early-access version 2433

README.md

@@ -1,7 +1,7 @@
 yuzu emulator early access
 =============
 
-This is the source code for early-access 2432.
+This is the source code for early-access 2433.
 
 ## Legal Notice
 

externals/ffmpeg/ffmpeg/.gitattributes

@@ -0,0 +1,2 @@
+*.pnm -diff -text
+tests/ref/fate/sub-scc eol=crlf

externals/ffmpeg/ffmpeg/.gitignore

@@ -0,0 +1,39 @@
+*.a
+*.o
+*.o.*
+*.d
+*.def
+*.dll
+*.dylib
+*.exe
+*.exp
+*.gcda
+*.gcno
+*.h.c
+*.ilk
+*.lib
+*.pc
+*.pdb
+*.so
+*.so.*
+*.swp
+*.ver
+*.version
+*.ptx
+*.ptx.c
+*_g
+\#*
+.\#*
+/.config
+/.version
+/ffmpeg
+/ffplay
+/ffprobe
+/config.asm
+/config.h
+/coverage.info
+/avversion.h
+/lcov/
+/src
+/mapfile
+/tools/python/__pycache__/

externals/ffmpeg/ffmpeg/.mailmap

@@ -0,0 +1,25 @@
+<james.darnley@gmail.com> <jdarnley@obe.tv>
+<jeebjp@gmail.com> <jan.ekstrom@aminocom.com>
+<sw@jkqxz.net> <mrt@jkqxz.net>
+<u@pkh.me> <cboesch@gopro.com>
+<zhilizhao@tencent.com> <quinkblack@foxmail.com>
+<zhilizhao@tencent.com> <wantlamy@gmail.com>
+<modmaker@google.com> <modmaker-at-google.com@ffmpeg.org>
+<stebbins@jetheaddev.com> <jstebbins@jetheaddev.com>
+<barryjzhao@tencent.com> <mypopydev@gmail.com>
+<barryjzhao@tencent.com> <jun.zhao@intel.com>
+<josh@itanimul.li> <joshdk@obe.tv>
+<michael@niedermayer.cc> <michaelni@gmx.at>
+<linjie.justin.fu@gmail.com> <linjie.fu@intel.com>
+<linjie.justin.fu@gmail.com> <fulinjie@zju.edu.cn>
+<ceffmpeg@gmail.com> <cehoyos@ag.or.at>
+<ceffmpeg@gmail.com> <cehoyos@rainbow.studorg.tuwien.ac.at>
+<ffmpeg@gyani.pro> <gyandoshi@gmail.com>
+<atomnuker@gmail.com> <rpehlivanov@obe.tv>
+<lizhong1008@gmail.com> <zhong.li@intel.com>
+<lizhong1008@gmail.com> <zhongli_dev@126.com>
+<andreas.rheinhardt@gmail.com> <andreas.rheinhardt@googlemail.com>
+rcombs <rcombs@rcombs.me> <rodger.combs@gmail.com>
+<thilo.borgmann@mail.de> <thilo.borgmann@googlemail.com>
+<liuqi05@kuaishou.com> <lq@chinaffmpeg.org>
+<ruiling.song83@gmail.com> <ruiling.song@intel.com>

externals/ffmpeg/ffmpeg/.travis.yml

@@ -0,0 +1,30 @@
+language: c
+sudo: false
+os:
+  - linux
+  - osx
+addons:
+  apt:
+    packages:
+      - nasm
+      - diffutils
+compiler:
+  - clang
+  - gcc
+matrix:
+    exclude:
+        - os: osx
+          compiler: gcc
+cache:
+  directories:
+    - ffmpeg-samples
+before_install:
+  - if [ "$TRAVIS_OS_NAME" == "osx" ]; then brew update; fi
+install:
+  - if [ "$TRAVIS_OS_NAME" == "osx" ]; then brew install nasm; fi
+script:
+  - mkdir -p ffmpeg-samples
+  - ./configure --samples=ffmpeg-samples --cc=$CC
+  - make -j 8
+  - make fate-rsync
+  - make check -j 8

externals/ffmpeg/ffmpeg/CONTRIBUTING.md

@@ -0,0 +1,4 @@
+# Note to Github users
+Patches should be submitted to the [ffmpeg-devel mailing list](https://ffmpeg.org/mailman/listinfo/ffmpeg-devel) using `git format-patch` or `git send-email`. Github pull requests should be avoided because they are not part of our review process and **will be ignored**.
+
+See [https://ffmpeg.org/developer.html#Contributing](https://ffmpeg.org/developer.html#Contributing) for more information.

externals/ffmpeg/ffmpeg/COPYING.GPLv2

@@ -0,0 +1,339 @@
+                    GNU GENERAL PUBLIC LICENSE
+                       Version 2, June 1991
+
+ Copyright (C) 1989, 1991 Free Software Foundation, Inc.,
+ 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ Everyone is permitted to copy and distribute verbatim copies
+ of this license document, but changing it is not allowed.
+
+                            Preamble
+
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+   TERMS AND CONDITIONS FOR COPYING, DISTRIBUTION AND MODIFICATION
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+Public License instead of this License.

externals/ffmpeg/ffmpeg/COPYING.GPLv3

@@ -0,0 +1,674 @@
+                    GNU GENERAL PUBLIC LICENSE
+                       Version 3, 29 June 2007
+
+ Copyright (C) 2007 Free Software Foundation, Inc. <http://fsf.org/>
+ Everyone is permitted to copy and distribute verbatim copies
+ of this license document, but changing it is not allowed.
+
+                            Preamble
+
+  The GNU General Public License is a free, copyleft license for
+software and other kinds of works.
+
+  The licenses for most software and other practical works are designed
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+have the freedom to distribute copies of free software (and charge for
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+free programs, and that you know you can do these things.
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externals/ffmpeg/ffmpeg/COPYING.LGPLv2.1

@@ -0,0 +1,502 @@
+                  GNU LESSER GENERAL PUBLIC LICENSE
+                       Version 2.1, February 1999
+
+ Copyright (C) 1991, 1999 Free Software Foundation, Inc.
+ 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ Everyone is permitted to copy and distribute verbatim copies
+ of this license document, but changing it is not allowed.
+
+[This is the first released version of the Lesser GPL.  It also counts
+ as the successor of the GNU Library Public License, version 2, hence
+ the version number 2.1.]
+
+                            Preamble
+
+  The licenses for most software are designed to take away your
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+
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+
+  When we speak of free software, we are referring to freedom of use,
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+
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externals/ffmpeg/ffmpeg/COPYING.LGPLv3

@@ -0,0 +1,165 @@
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+
+  If the Library as you received it specifies that a proxy can decide
+whether future versions of the GNU Lesser General Public License shall
+apply, that proxy's public statement of acceptance of any version is
+permanent authorization for you to choose that version for the
+Library.

externals/ffmpeg/ffmpeg/CREDITS

@@ -0,0 +1,6 @@
+See the Git history of the project (git://source.ffmpeg.org/ffmpeg) to
+get the names of people who have contributed to FFmpeg.
+
+To check the log, you can type the command "git log" in the FFmpeg
+source directory, or browse the online repository at
+http://source.ffmpeg.org.

externals/ffmpeg/ffmpeg/INSTALL.md

@@ -0,0 +1,17 @@
+## 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.

externals/ffmpeg/ffmpeg/LICENSE.md

@@ -0,0 +1,129 @@
+# License
+
+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 optimization in the files
+    - `libavcodec/x86/flac_dsp_gpl.asm`
+    - `libavcodec/x86/idct_mmx.c`
+    - `libavfilter/x86/vf_removegrain.asm`
+- the following building and testing tools
+    - `compat/solaris/make_sunver.pl`
+    - `doc/t2h.pm`
+    - `doc/texi2pod.pl`
+    - `libswresample/tests/swresample.c`
+    - `tests/checkasm/*`
+    - `tests/tiny_ssim.c`
+- the following filters in libavfilter:
+    - `signature_lookup.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_histeq.c`
+    - `vf_hqdn3d.c`
+    - `vf_kerndeint.c`
+    - `vf_lensfun.c` (GPL version 3 or later)
+    - `vf_mcdeint.c`
+    - `vf_mpdecimate.c`
+    - `vf_nnedi.c`
+    - `vf_owdenoise.c`
+    - `vf_perspective.c`
+    - `vf_phase.c`
+    - `vf_pp.c`
+    - `vf_pp7.c`
+    - `vf_pullup.c`
+    - `vf_repeatfields.c`
+    - `vf_sab.c`
+    - `vf_signature.c`
+    - `vf_smartblur.c`
+    - `vf_spp.c`
+    - `vf_stereo3d.c`
+    - `vf_super2xsai.c`
+    - `vf_tinterlace.c`
+    - `vf_uspp.c`
+    - `vf_vaguedenoiser.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 version 2:
+- avisynth
+- frei0r
+- libcdio
+- libdavs2
+- librubberband
+- libvidstab
+- libx264
+- libx265
+- libxavs
+- libxavs2
+- libxvid
+
+When combining them with FFmpeg, FFmpeg needs to be licensed as GPL as well by
+passing `--enable-gpl` to configure.
+
+The following libraries are under LGPL version 3:
+- gmp
+- libaribb24
+- liblensfun
+
+When combining them with FFmpeg, use the configure option `--enable-version3` to
+upgrade FFmpeg to the LGPL v3.
+
+The VMAF, mbedTLS, RK MPI, 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.
+
+The smbclient library is under the GPL v3, to combine it with FFmpeg,
+the options `--enable-gpl` and `--enable-version3` have to be passed to
+configure to upgrade FFmpeg to the GPL v3.
+
+### Incompatible libraries
+
+There are certain libraries you can combine with FFmpeg whose licenses are not
+compatible with the GPL and/or the LGPL. If you wish to enable these
+libraries, even in circumstances that their license may be incompatible, pass
+`--enable-nonfree` to configure. This will cause the resulting binary to be
+unredistributable.
+
+The Fraunhofer FDK AAC and OpenSSL libraries are under licenses which are
+incompatible with the GPLv2 and v3. To the best of our knowledge, they are
+compatible with the LGPL.

externals/ffmpeg/ffmpeg/MAINTAINERS

@@ -0,0 +1,637 @@
+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
+==============
+
+  final design decisions
+
+
+Applications
+============
+
+ffmpeg:
+  ffmpeg.c                              Michael Niedermayer
+
+ffplay:
+  ffplay.c                              Marton Balint
+
+ffprobe:
+  ffprobe.c                             Stefano Sabatini
+
+Commandline utility code:
+  cmdutils.c, cmdutils.h                Michael Niedermayer
+
+QuickTime faststart:
+  tools/qt-faststart.c                  Baptiste Coudurier
+
+
+Miscellaneous Areas
+===================
+
+documentation                           Stefano Sabatini, Mike Melanson, Timothy Gu, Gyan Doshi
+project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser, Nikolay Aleksandrov
+presets                                 Robert Swain
+metadata subsystem                      Aurelien Jacobs
+release management                      Michael Niedermayer
+API tests                               Ludmila Glinskih
+
+
+Communication
+=============
+
+website                                 Deby Barbara Lepage
+fate.ffmpeg.org                         Timothy Gu
+Trac bug tracker                        Alexander Strasser, Michael Niedermayer, Carl Eugen Hoyos
+Patchwork                               Andriy Gelman
+mailing lists                           Baptiste Coudurier
+Twitter                                 Reynaldo H. Verdejo Pinochet
+Launchpad                               Timothy Gu
+ffmpeg-security                         Andreas Cadhalpun, Carl Eugen Hoyos, Clément Bœsch, Michael Niedermayer, Reimar Doeffinger, rcombs, wm4
+
+
+libavutil
+=========
+
+External Interfaces:
+  libavutil/avutil.h                    Michael Niedermayer
+Internal Interfaces:
+  libavutil/common.h                    Michael Niedermayer
+
+Other:
+  aes_ctr.c, aes_ctr.h                  Eran Kornblau
+  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
+  hwcontext_cuda*                       Timo Rothenpieler
+  hwcontext_vulkan*                     Lynne
+  intfloat*                             Michael Niedermayer
+  integer.c, integer.h                  Michael Niedermayer
+  lzo                                   Reimar Doeffinger
+  mathematics.c, mathematics.h          Michael Niedermayer
+  mem.c, mem.h                          Michael Niedermayer
+  opencl.c, opencl.h                    Wei Gao
+  opt.c, opt.h                          Michael Niedermayer
+  rational.c, rational.h                Michael Niedermayer
+  rc4                                   Reimar Doeffinger
+  ripemd.c, ripemd.h                    James Almer
+  tx*                                   Lynne
+
+
+libavcodec
+==========
+
+Generic Parts:
+  External Interfaces:
+    avcodec.h                           Michael Niedermayer
+  utility code:
+    utils.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:
+    rangecoder.c, rangecoder.h          Michael Niedermayer
+    lzw.*                               Michael Niedermayer
+  floating point AAN DCT:
+    faandct.c, faandct.h                Michael Niedermayer
+  Non-power-of-two MDCT:
+    mdct15.c, mdct15.h                  Rostislav Pehlivanov
+  Golomb coding:
+    golomb.c, golomb.h                  Michael Niedermayer
+  motion estimation:
+    motion*                             Michael Niedermayer
+  rate control:
+    ratecontrol.c                       Michael Niedermayer
+  simple IDCT:
+    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
+  aacenc*, aaccoder.c                   Rostislav Pehlivanov
+  alacenc.c                             Jaikrishnan Menon
+  alsdec.c                              Thilo Borgmann, Umair Khan
+  aptx.c                                Aurelien Jacobs
+  ass*                                  Aurelien Jacobs
+  asv*                                  Michael Niedermayer
+  atrac3plus*                           Maxim Poliakovski
+  audiotoolbox*                         rcombs
+  avs2*                                 Huiwen Ren
+  bgmc.c, bgmc.h                        Thilo Borgmann
+  binkaudio.c                           Peter Ross
+  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, Aman Gupta
+  cljr                                  Alex Beregszaszi
+  cpia.c                                Stephan Hilb
+  crystalhd.c                           Philip Langdale
+  cscd.c                                Reimar Doeffinger
+  cuviddec.c                            Timo Rothenpieler
+  dca*                                  foo86
+  dirac*                                Rostislav Pehlivanov
+  dnxhd*                                Baptiste Coudurier
+  dolby_e*                              foo86
+  dpcm.c                                Mike Melanson
+  dss_sp.c                              Oleksij Rempel
+  dv.c                                  Roman Shaposhnik
+  dvbsubdec.c                           Anshul Maheshwari
+  eacmv*, eaidct*, eat*                 Peter Ross
+  evrc*                                 Paul B Mahol
+  exif.c, exif.h                        Thilo Borgmann
+  ffv1*                                 Michael Niedermayer
+  ffwavesynth.c                         Nicolas George
+  fifo.c                                Jan Sebechlebsky
+  flicvideo.c                           Mike Melanson
+  g722.c                                Martin Storsjo
+  g726.c                                Roman Shaposhnik
+  gifdec.c                              Baptiste Coudurier
+  h261*                                 Michael Niedermayer
+  h263*                                 Michael Niedermayer
+  h264*                                 Loren Merritt, Michael Niedermayer
+  hap*                                  Tom Butterworth
+  huffyuv*                              Michael Niedermayer
+  idcinvideo.c                          Mike Melanson
+  interplayvideo.c                      Mike Melanson
+  jni*, ffjni*                          Matthieu Bouron
+  jpeg2000*                             Nicolas Bertrand
+  jvdec.c                               Peter Ross
+  lcl*.c                                Roberto Togni, Reimar Doeffinger
+  libcelt_dec.c                         Nicolas George
+  libcodec2.c                           Tomas Härdin
+  libdirac*                             David Conrad
+  libdavs2.c                            Huiwen Ren
+  libgsm.c                              Michel Bardiaux
+  libkvazaar.c                          Arttu Ylä-Outinen
+  libopenh264enc.c                      Martin Storsjo, Linjie Fu
+  libopenjpeg.c                         Jaikrishnan Menon
+  libopenjpegenc.c                      Michael Bradshaw
+  libtheoraenc.c                        David Conrad
+  libvorbis.c                           David Conrad
+  libvpx*                               James Zern
+  libxavs.c                             Stefan Gehrer
+  libxavs2.c                            Huiwen Ren
+  libzvbi-teletextdec.c                 Marton Balint
+  lzo.h, lzo.c                          Reimar Doeffinger
+  mdec.c                                Michael Niedermayer
+  mimic.c                               Ramiro Polla
+  mjpeg*.c                              Michael Niedermayer
+  mlp*                                  Ramiro Polla, Jai Luthra
+  mmvideo.c                             Peter Ross
+  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
+  nuv.c                                 Reimar Doeffinger
+  nvdec*, nvenc*                        Timo Rothenpieler
+  omx.c                                 Martin Storsjo, Aman Gupta
+  opus*                                 Rostislav Pehlivanov
+  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
+  qsv*                                  Mark Thompson, Zhong Li
+  qtrle.c                               Mike Melanson
+  ra144.c, ra144.h, ra288.c, ra288.h    Roberto Togni
+  resample2.c                           Michael Niedermayer
+  rl2.c                                 Sascha Sommer
+  rpza.c                                Roberto Togni
+  rtjpeg.c, rtjpeg.h                    Reimar Doeffinger
+  rv10.c                                Michael Niedermayer
+  s3tc*                                 Ivo van Poorten
+  smc.c                                 Mike Melanson
+  snow*                                 Michael Niedermayer, Loren Merritt
+  sonic.c                               Alex Beregszaszi
+  speedhq.c                             Steinar H. Gunderson
+  srt*                                  Aurelien Jacobs
+  sunrast.c                             Ivo van Poorten
+  svq3.c                                Michael Niedermayer
+  tak*                                  Paul B Mahol
+  truemotion1*                          Mike Melanson
+  tta.c                                 Alex Beregszaszi, Jaikrishnan Menon
+  ttaenc.c                              Paul B Mahol
+  txd.c                                 Ivo van Poorten
+  v4l2_*                                Jorge Ramirez-Ortiz
+  vc2*                                  Rostislav Pehlivanov
+  vcr1.c                                Michael Niedermayer
+  videotoolboxenc.c                     Rick Kern, Aman Gupta
+  vima.c                                Paul B Mahol
+  vorbisdec.c                           Denes Balatoni, David Conrad
+  vorbisenc.c                           Oded Shimon
+  vp3*                                  Mike Melanson
+  vp5                                   Aurelien Jacobs
+  vp6                                   Aurelien Jacobs
+  vp8                                   David Conrad, Ronald Bultje
+  vp9                                   Ronald Bultje
+  vqavideo.c                            Mike Melanson
+  wmaprodec.c                           Sascha Sommer
+  wmavoice.c                            Ronald S. Bultje
+  wmv2.c                                Michael Niedermayer
+  xan.c                                 Mike Melanson
+  xbm*                                  Paul B Mahol
+  xface                                 Stefano Sabatini
+  xvmc.c                                Ivan Kalvachev
+  xwd*                                  Paul B Mahol
+
+Hardware acceleration:
+  crystalhd.c                           Philip Langdale
+  dxva2*                                Hendrik Leppkes, Laurent Aimar, Steve Lhomme
+  d3d11va*                              Steve Lhomme
+  mediacodec*                           Matthieu Bouron, Aman Gupta
+  vaapi*                                Gwenole Beauchesne
+  vaapi_encode*                         Mark Thompson
+  vdpau*                                Philip Langdale, Carl Eugen Hoyos
+  videotoolbox*                         Rick Kern, Aman Gupta
+
+
+libavdevice
+===========
+  External Interface:
+    libavdevice/avdevice.h
+
+
+  avfoundation.m                        Thilo Borgmann
+  android_camera.c                      Felix Matouschek
+  decklink*                             Marton Balint
+  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
+  pulse_audio_enc.c                     Lukasz Marek
+  sdl                                   Stefano Sabatini
+  sdl2.c                                Josh de Kock
+  v4l2.c                                Giorgio Vazzana
+  vfwcap.c                              Ramiro Polla
+  xv.c                                  Lukasz Marek
+
+libavfilter
+===========
+
+Generic parts:
+  graphdump.c                           Nicolas George
+
+  motion_estimation.c                   Davinder Singh
+
+Filters:
+  f_drawgraph.c                         Paul B Mahol
+  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_atempo.c                           Pavel Koshevoy
+  af_biquads.c                          Paul B Mahol
+  af_chorus.c                           Paul B Mahol
+  af_compand.c                          Paul B Mahol
+  af_firequalizer.c                     Muhammad Faiz
+  af_hdcd.c                             Burt P.
+  af_ladspa.c                           Paul B Mahol
+  af_loudnorm.c                         Kyle Swanson
+  af_pan.c                              Nicolas George
+  af_sidechaincompress.c                Paul B Mahol
+  af_silenceremove.c                    Paul B Mahol
+  avf_aphasemeter.c                     Paul B Mahol
+  avf_avectorscope.c                    Paul B Mahol
+  avf_showcqt.c                         Muhammad Faiz
+  vf_blend.c                            Paul B Mahol
+  vf_bwdif                              Thomas Mundt (CC <thomas.mundt@hr.de>)
+  vf_chromakey.c                        Timo Rothenpieler
+  vf_colorchannelmixer.c                Paul B Mahol
+  vf_colorconstancy.c                   Mina Sami    (CC <minas.gorgy@gmail.com>)
+  vf_colorbalance.c                     Paul B Mahol
+  vf_colorkey.c                         Timo Rothenpieler
+  vf_colorlevels.c                      Paul B Mahol
+  vf_coreimage.m                        Thilo Borgmann
+  vf_deband.c                           Paul B Mahol
+  vf_dejudder.c                         Nicholas Robbins
+  vf_delogo.c                           Jean Delvare (CC <jdelvare@suse.com>)
+  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_(t)interlace                       Thomas Mundt (CC <thomas.mundt@hr.de>)
+  vf_lenscorrection.c                   Daniel Oberhoff
+  vf_mergeplanes.c                      Paul B Mahol
+  vf_mestimate.c                        Davinder Singh
+  vf_minterpolate.c                     Davinder Singh
+  vf_neighbor.c                         Paul B Mahol
+  vf_psnr.c                             Paul B Mahol
+  vf_random.c                           Paul B Mahol
+  vf_readvitc.c                         Tobias Rapp (CC t.rapp at noa-archive dot com)
+  vf_scale.c                            Michael Niedermayer
+  vf_separatefields.c                   Paul B Mahol
+  vf_ssim.c                             Paul B Mahol
+  vf_stereo3d.c                         Paul B Mahol
+  vf_telecine.c                         Paul B Mahol
+  vf_tonemap_opencl.c                   Ruiling Song
+  vf_yadif.c                            Michael Niedermayer
+  vf_zoompan.c                          Paul B Mahol
+
+Sources:
+  vsrc_mandelbrot.c                     Michael Niedermayer
+
+dnn                                     Yejun Guo
+
+libavformat
+===========
+
+Generic parts:
+  External Interface:
+    libavformat/avformat.h              Michael Niedermayer
+  Utility Code:
+    libavformat/utils.c                 Michael Niedermayer
+  Text Subtitles                        Clément Bœsch
+
+
+Muxers/Demuxers:
+  4xm.c                                 Mike Melanson
+  aadec.c                               Vesselin Bontchev (vesselin.bontchev at yandex dot com)
+  adtsenc.c                             Robert Swain
+  afc.c                                 Paul B Mahol
+  aiffdec.c                             Baptiste Coudurier, Matthieu Bouron
+  aiffenc.c                             Baptiste Coudurier, Matthieu Bouron
+  alp.c                                 Zane van Iperen
+  amvenc.c                              Zane van Iperen
+  apm.c                                 Zane van Iperen
+  apngdec.c                             Benoit Fouet
+  argo_asf.c                            Zane van Iperen
+  argo_brp.c                            Zane van Iperen
+  ass*                                  Aurelien Jacobs
+  astdec.c                              Paul B Mahol
+  astenc.c                              James Almer
+  avi*                                  Michael Niedermayer
+  avisynth.c                            Stephen Hutchinson
+  avr.c                                 Paul B Mahol
+  bink.c                                Peter Ross
+  boadec.c                              Michael Niedermayer
+  brstm.c                               Paul B Mahol
+  caf*                                  Peter Ross
+  cdxl.c                                Paul B Mahol
+  codec2.c                              Tomas Härdin
+  crc.c                                 Michael Niedermayer
+  dashdec.c                             Steven Liu
+  dashenc.c                             Karthick Jeyapal
+  daud.c                                Reimar Doeffinger
+  dss.c                                 Oleksij Rempel
+  dtsdec.c                              foo86
+  dtshddec.c                            Paul B Mahol
+  dv.c                                  Roman Shaposhnik
+  electronicarts.c                      Peter Ross
+  epafdec.c                             Paul B Mahol
+  ffm*                                  Baptiste Coudurier
+  flic.c                                Mike Melanson
+  flvdec.c                              Michael Niedermayer
+  flvenc.c                              Michael Niedermayer, Steven Liu
+  gxf.c                                 Reimar Doeffinger
+  gxfenc.c                              Baptiste Coudurier
+  hlsenc.c                              Christian Suloway, Steven Liu
+  idcin.c                               Mike Melanson
+  idroqdec.c                            Mike Melanson
+  iff.c                                 Jaikrishnan Menon
+  img2*.c                               Michael Niedermayer
+  ipmovie.c                             Mike Melanson
+  ircam*                                Paul B Mahol
+  iss.c                                 Stefan Gehrer
+  jvdec.c                               Peter Ross
+  kvag.c                                Zane van Iperen
+  libmodplug.c                          Clément Bœsch
+  libopenmpt.c                          Josh de Kock
+  lmlm4.c                               Ivo van Poorten
+  lvfdec.c                              Paul B Mahol
+  lxfdec.c                              Tomas Härdin
+  matroska.c                            Aurelien Jacobs, Andreas Rheinhardt
+  matroskadec.c                         Aurelien Jacobs, Andreas Rheinhardt
+  matroskaenc.c                         David Conrad, Andreas Rheinhardt
+  matroska subtitles (matroskaenc.c)    John Peebles
+  metadata*                             Aurelien Jacobs
+  mgsts.c                               Paul B Mahol
+  microdvd*                             Aurelien Jacobs
+  mm.c                                  Peter Ross
+  mov.c                                 Baptiste Coudurier
+  movenc.c                              Baptiste Coudurier, Matthieu Bouron
+  movenccenc.c                          Eran Kornblau
+  mpeg.c                                Michael Niedermayer
+  mpegenc.c                             Michael Niedermayer
+  mpegts.c                              Marton Balint
+  mpegtsenc.c                           Baptiste Coudurier
+  msnwc_tcp.c                           Ramiro Polla
+  mtv.c                                 Reynaldo H. Verdejo Pinochet
+  mxf*                                  Baptiste Coudurier, Tomas Härdin
+  nistspheredec.c                       Paul B Mahol
+  nsvdec.c                              Francois Revol
+  nut*                                  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
+  pp_bnk.c                              Zane van Iperen
+  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
+  rtp.c, rtpenc.c                       Martin Storsjo
+  rtpdec_ac3.*                          Gilles Chanteperdrix
+  rtpdec_dv.*                           Thomas Volkert
+  rtpdec_h261.*, rtpenc_h261.*          Thomas Volkert
+  rtpdec_hevc.*, rtpenc_hevc.*          Thomas Volkert
+  rtpdec_mpa_robust.*                   Gilles Chanteperdrix
+  rtpdec_asf.*                          Ronald S. Bultje
+  rtpdec_vc2hq.*, rtpenc_vc2hq.*        Thomas Volkert
+  rtpdec_vp9.c                          Thomas Volkert
+  rtpenc_mpv.*, rtpenc_aac.*            Martin Storsjo
+  s337m.c                               foo86
+  sbgdec.c                              Nicolas George
+  sdp.c                                 Martin Storsjo
+  segafilm.c                            Mike Melanson
+  segment.c                             Stefano Sabatini
+  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
+  wvenc.c                               Paul B Mahol
+
+Protocols:
+  async.c                               Zhang Rui
+  bluray.c                              Petri Hintukainen
+  ftp.c                                 Lukasz Marek
+  http.c                                Ronald S. Bultje
+  libssh.c                              Lukasz Marek
+  libzmq.c                              Andriy Gelman
+  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
+=====================================
+
+Alpha                                   Falk Hueffner
+MIPS                                    Manojkumar Bhosale, Shiyou Yin
+Mac OS X / PowerPC                      Romain Dolbeau, Guillaume Poirier
+Amiga / PowerPC                         Colin Ward
+Linux / PowerPC                         Lauri Kasanen
+Windows MinGW                           Alex Beregszaszi, Ramiro Polla
+Windows Cygwin                          Victor Paesa
+Windows MSVC                            Matthew Oliver, Hendrik Leppkes
+Windows ICL                             Matthew Oliver
+ADI/Blackfin DSP                        Marc Hoffman
+Sparc                                   Roman Shaposhnik
+OS/2                                    KO Myung-Hun
+
+
+Developers with git write access who are currently not maintaining any specific part
+====================================================================================
+Alex Converse
+Andreas Cadhalpun
+Anuradha Suraparaju
+Ben Littler
+Benjamin Larsson
+Bobby Bingham
+Daniel Verkamp
+Derek Buitenhuis
+Ganesh Ajjanagadde
+Henrik Gramner
+Ivan Uskov
+James Darnley
+Jan Ekström
+Joakim Plate
+Jun Zhao
+Kieran Kunhya
+Kirill Gavrilov
+Limin Wang
+Martin Storsjö
+Panagiotis Issaris
+Pedro Arthur
+Sebastien Zwickert
+Vittorio Giovara
+wm4
+(this list is incomplete)
+
+
+Releases
+========
+
+2.8                                     Michael Niedermayer
+2.7                                     Michael Niedermayer
+2.6                                     Michael Niedermayer
+2.5                                     Michael Niedermayer
+
+If you want to maintain an older release, please contact us
+
+
+GnuPG Fingerprints and IRC nicknames of maintainers and contributors
+====================================================================
+
+IRC nicknames are in parentheses. These apply
+to the IRC channels listed on the website.
+
+Alexander Strasser            1C96 78B7 83CB 8AA7 9AF5 D1EB A7D8 A57B A876 E58F
+Anssi Hannula                 1A92 FF42 2DD9 8D2E 8AF7 65A9 4278 C520 513D F3CB
+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
+FFmpeg release signing key    FCF9 86EA 15E6 E293 A564 4F10 B432 2F04 D676 58D8
+Ganesh Ajjanagadde            C96A 848E 97C3 CEA2 AB72 5CE4 45F9 6A2D 3C36 FB1B
+Gwenole Beauchesne            2E63 B3A6 3E44 37E2 017D 2704 53C7 6266 B153 99C4
+Jaikrishnan Menon             61A1 F09F 01C9 2D45 78E1 C862 25DC 8831 AF70 D368
+James Almer                   7751 2E8C FD94 A169 57E6 9A7A 1463 01AD 7376 59E0
+Jean Delvare                  7CA6 9F44 60F1 BDC4 1FD2 C858 A552 6B9B B3CD 4E6A
+Loren Merritt                 ABD9 08F4 C920 3F65 D8BE 35D7 1540 DAA7 060F 56DE
+Lynne                         FE50 139C 6805 72CA FD52 1F8D A2FE A5F0 3F03 4464
+Michael Niedermayer           9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
+Nicolas George                24CE 01CE 9ACC 5CEB 74D8 8D9D B063 D997 36E5 4C93
+Nikolay Aleksandrov           8978 1D8C FB71 588E 4B27 EAA8 C4F0 B5FC E011 13B1
+Panagiotis Issaris            6571 13A3 33D9 3726 F728 AA98 F643 B12E ECF3 E029
+Peter Ross                    A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B
+Philip Langdale               5DC5 8D66 5FBA 3A43 18EC 045E F8D6 B194 6A75 682E
+Ramiro Polla                  7859 C65B 751B 1179 792E DAE8 8E95 8B2F 9B6C 5700
+Reimar Doeffinger             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
+Steinar H. Gunderson          C2E9 004F F028 C18E 4EAD DB83 7F61 7561 7797 8F76
+Stephan Hilb                  4F38 0B3A 5F39 B99B F505 E562 8D5C 5554 4E17 8863
+Thilo Borgmann (thilo)        CE1D B7F4 4D20 FC3A DD9F FE5A 257C 5B8F 1D20 B92F
+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 (thardin)        A79D 4E3D F38F 763F 91F5 8B33 A01E 8AE0 41BB 2551
+Wei Gao                       4269 7741 857A 0E60 9EC5 08D2 4744 4EFA 62C1 87B9
+Zane van Iperen (zane)        61AE D40F 368B 6F26 9DAE 3892 6861 6B2D 8AC4 DCC5

externals/ffmpeg/ffmpeg/Makefile

@@ -0,0 +1,191 @@
+MAIN_MAKEFILE=1
+include ffbuild/config.mak
+
+vpath %.c    $(SRC_PATH)
+vpath %.cpp  $(SRC_PATH)
+vpath %.h    $(SRC_PATH)
+vpath %.inc  $(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 %.cu   $(SRC_PATH)
+vpath %.ptx  $(SRC_PATH)
+vpath %/fate_config.sh.template $(SRC_PATH)
+
+TESTTOOLS   = audiogen videogen rotozoom tiny_psnr tiny_ssim base64 audiomatch
+HOSTPROGS  := $(TESTTOOLS:%=tests/%) doc/print_options
+
+# $(FFLIBS-yes) needs to be in linking order
+FFLIBS-$(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
+
+SKIPHEADERS = compat/w32pthreads.h
+
+# first so "all" becomes default target
+all: all-yes
+
+include $(SRC_PATH)/tools/Makefile
+include $(SRC_PATH)/ffbuild/common.mak
+
+FF_EXTRALIBS := $(FFEXTRALIBS)
+FF_DEP_LIBS  := $(DEP_LIBS)
+FF_STATIC_DEP_LIBS := $(STATIC_DEP_LIBS)
+
+$(TOOLS): %$(EXESUF): %.o
+	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(EXTRALIBS-$(*F)) $(EXTRALIBS) $(ELIBS)
+
+target_dec_%_fuzzer$(EXESUF): target_dec_%_fuzzer.o $(FF_DEP_LIBS)
+	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
+
+tools/target_bsf_%_fuzzer$(EXESUF): tools/target_bsf_%_fuzzer.o $(FF_DEP_LIBS)
+	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
+
+target_dem_%_fuzzer$(EXESUF): target_dem_%_fuzzer.o $(FF_DEP_LIBS)
+	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
+
+tools/target_dem_fuzzer$(EXESUF): tools/target_dem_fuzzer.o $(FF_DEP_LIBS)
+	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
+
+tools/target_io_dem_fuzzer$(EXESUF): tools/target_io_dem_fuzzer.o $(FF_DEP_LIBS)
+	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
+
+
+tools/enum_options$(EXESUF): ELIBS = $(FF_EXTRALIBS)
+tools/enum_options$(EXESUF): $(FF_DEP_LIBS)
+tools/sofa2wavs$(EXESUF): ELIBS = $(FF_EXTRALIBS)
+tools/uncoded_frame$(EXESUF): $(FF_DEP_LIBS)
+tools/uncoded_frame$(EXESUF): ELIBS = $(FF_EXTRALIBS)
+tools/target_dec_%_fuzzer$(EXESUF): $(FF_DEP_LIBS)
+tools/target_dem_%_fuzzer$(EXESUF): $(FF_DEP_LIBS)
+
+CONFIGURABLE_COMPONENTS =                                           \
+    $(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c))                 \
+    $(SRC_PATH)/libavcodec/bitstream_filters.c                      \
+    $(SRC_PATH)/libavcodec/parsers.c                                \
+    $(SRC_PATH)/libavformat/protocols.c                             \
+
+config.h: ffbuild/.config
+ffbuild/.config: $(CONFIGURABLE_COMPONENTS)
+	@-tput bold 2>/dev/null
+	@-printf '\nWARNING: $(?) newer than config.h, rerun configure\n\n'
+	@-tput sgr0 2>/dev/null
+
+SUBDIR_VARS := CLEANFILES FFLIBS HOSTPROGS TESTPROGS TOOLS               \
+               HEADERS ARCH_HEADERS BUILT_HEADERS SKIPHEADERS            \
+               ARMV5TE-OBJS ARMV6-OBJS ARMV8-OBJS VFP-OBJS NEON-OBJS     \
+               ALTIVEC-OBJS VSX-OBJS MMX-OBJS X86ASM-OBJS                \
+               MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSP-OBJS MSA-OBJS         \
+               MMI-OBJS OBJS SLIBOBJS HOSTOBJS TESTOBJS
+
+define RESET
+$(1) :=
+$(1)-yes :=
+endef
+
+define DOSUBDIR
+$(foreach V,$(SUBDIR_VARS),$(eval $(call RESET,$(V))))
+SUBDIR := $(1)/
+include $(SRC_PATH)/$(1)/Makefile
+-include $(SRC_PATH)/$(1)/$(ARCH)/Makefile
+-include $(SRC_PATH)/$(1)/$(INTRINSICS)/Makefile
+include $(SRC_PATH)/ffbuild/library.mak
+endef
+
+$(foreach D,$(FFLIBS),$(eval $(call DOSUBDIR,lib$(D))))
+
+include $(SRC_PATH)/fftools/Makefile
+include $(SRC_PATH)/doc/Makefile
+include $(SRC_PATH)/doc/examples/Makefile
+
+libavcodec/avcodec.o libavformat/utils.o libavdevice/avdevice.o libavfilter/avfilter.o libavutil/utils.o libpostproc/postprocess.o libswresample/swresample.o libswscale/utils.o : libavutil/ffversion.h
+
+$(PROGS): %$(PROGSSUF)$(EXESUF): %$(PROGSSUF)_g$(EXESUF)
+ifeq ($(STRIPTYPE),direct)
+	$(STRIP) -o $@ $<
+else
+	$(CP) $< $@
+	$(STRIP) $@
+endif
+
+%$(PROGSSUF)_g$(EXESUF): $(FF_DEP_LIBS)
+	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
+
+VERSION_SH  = $(SRC_PATH)/ffbuild/version.sh
+GIT_LOG     = $(SRC_PATH)/.git/logs/HEAD
+
+.version: $(wildcard $(GIT_LOG)) $(VERSION_SH) ffbuild/config.mak
+.version: M=@
+
+libavutil/ffversion.h .version:
+	$(M)$(VERSION_SH) $(SRC_PATH) libavutil/ffversion.h $(EXTRA_VERSION)
+	$(Q)touch .version
+
+# force version.sh to run whenever version might have changed
+-include .version
+
+install: install-libs install-headers
+
+install-libs: install-libs-yes
+
+install-data: $(DATA_FILES)
+	$(Q)mkdir -p "$(DATADIR)"
+	$(INSTALL) -m 644 $(DATA_FILES) "$(DATADIR)"
+
+uninstall: uninstall-data uninstall-headers uninstall-libs uninstall-pkgconfig
+
+uninstall-data:
+	$(RM) -r "$(DATADIR)"
+
+clean::
+	$(RM) $(CLEANSUFFIXES)
+	$(RM) $(addprefix compat/,$(CLEANSUFFIXES)) $(addprefix compat/*/,$(CLEANSUFFIXES)) $(addprefix compat/*/*/,$(CLEANSUFFIXES))
+	$(RM) -r coverage-html
+	$(RM) -rf coverage.info coverage.info.in lcov
+
+distclean:: clean
+	$(RM) .version avversion.h config.asm config.h mapfile  \
+		ffbuild/.config ffbuild/config.* libavutil/avconfig.h \
+		version.h libavutil/ffversion.h libavcodec/codec_names.h \
+		libavcodec/bsf_list.c libavformat/protocol_list.c \
+		libavcodec/codec_list.c libavcodec/parser_list.c \
+		libavfilter/filter_list.c libavdevice/indev_list.c libavdevice/outdev_list.c \
+		libavformat/muxer_list.c libavformat/demuxer_list.c
+ifeq ($(SRC_LINK),src)
+	$(RM) src
+endif
+	$(RM) -rf doc/examples/pc-uninstalled
+
+config:
+	$(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION)
+
+build: all alltools examples testprogs
+check: all alltools examples testprogs fate
+
+include $(SRC_PATH)/tests/Makefile
+
+$(sort $(OUTDIRS)):
+	$(Q)mkdir -p $@
+
+# Dummy rule to stop make trying to rebuild removed or renamed headers
+%.h:
+	@:
+
+# Disable suffix rules.  Most of the builtin rules are suffix rules,
+# so this saves some time on slow systems.
+.SUFFIXES:
+
+.PHONY: all all-yes alltools build check config testprogs
+.PHONY: *clean install* uninstall*

externals/ffmpeg/ffmpeg/README.md

@@ -0,0 +1,46 @@
+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](https://ffmpeg.org/ffmpeg.html) is a command line toolbox to
+  manipulate, convert and stream multimedia content.
+* [ffplay](https://ffmpeg.org/ffplay.html) is a minimalistic multimedia player.
+* [ffprobe](https://ffmpeg.org/ffprobe.html) is a simple analysis tool to inspect
+  multimedia content.
+* 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](https://ffmpeg.org)
+and in the [wiki](https://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.
+
+## Contributing
+
+Patches should be submitted to the ffmpeg-devel mailing list using
+`git format-patch` or `git send-email`. Github pull requests should be
+avoided because they are not part of our review process and will be ignored.

externals/ffmpeg/ffmpeg/RELEASE

@@ -0,0 +1 @@
+4.4

externals/ffmpeg/ffmpeg/RELEASE_NOTES

@@ -0,0 +1,15 @@
+
+              ┌────────────────────────────────────┐
+              │ RELEASE NOTES for FFmpeg 4.4 "Rao" │
+              └────────────────────────────────────┘
+
+   The FFmpeg Project proudly presents FFmpeg 4.4 "Rao", about 10
+   months after the release of FFmpeg 4.3.
+
+   A complete Changelog is available at the root of the project, and the
+   complete Git history on https://git.ffmpeg.org/gitweb/ffmpeg.git
+
+   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.

externals/ffmpeg/ffmpeg/compat/aix/math.h

@@ -0,0 +1,31 @@
+/*
+ * Work around the class() function in AIX math.h clashing with
+ * identifiers named "class".
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef COMPAT_AIX_MATH_H
+#define COMPAT_AIX_MATH_H
+
+#define class class_in_math_h_causes_problems
+
+#include_next <math.h>
+
+#undef class
+
+#endif /* COMPAT_AIX_MATH_H */

externals/ffmpeg/ffmpeg/compat/atomics/dummy/stdatomic.h

@@ -0,0 +1,176 @@
+/*
+ * 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
+ */
+
+/*
+ * based on vlc_atomic.h from VLC
+ * Copyright (C) 2010 Rémi Denis-Courmont
+ */
+
+#ifndef COMPAT_ATOMICS_DUMMY_STDATOMIC_H
+#define COMPAT_ATOMICS_DUMMY_STDATOMIC_H
+
+#include <stdint.h>
+
+#define ATOMIC_FLAG_INIT 0
+
+#define ATOMIC_VAR_INIT(value) (value)
+
+#define atomic_init(obj, value) \
+do {                            \
+    *(obj) = (value);           \
+} while(0)
+
+#define kill_dependency(y) ((void)0)
+
+#define atomic_thread_fence(order) \
+    ((void)0)
+
+#define atomic_signal_fence(order) \
+    ((void)0)
+
+#define atomic_is_lock_free(obj) 0
+
+typedef intptr_t atomic_flag;
+typedef intptr_t atomic_bool;
+typedef intptr_t atomic_char;
+typedef intptr_t atomic_schar;
+typedef intptr_t atomic_uchar;
+typedef intptr_t atomic_short;
+typedef intptr_t atomic_ushort;
+typedef intptr_t atomic_int;
+typedef intptr_t atomic_uint;
+typedef intptr_t atomic_long;
+typedef intptr_t atomic_ulong;
+typedef intptr_t atomic_llong;
+typedef intptr_t atomic_ullong;
+typedef intptr_t atomic_wchar_t;
+typedef intptr_t atomic_int_least8_t;
+typedef intptr_t atomic_uint_least8_t;
+typedef intptr_t atomic_int_least16_t;
+typedef intptr_t atomic_uint_least16_t;
+typedef intptr_t atomic_int_least32_t;
+typedef intptr_t atomic_uint_least32_t;
+typedef intptr_t atomic_int_least64_t;
+typedef intptr_t atomic_uint_least64_t;
+typedef intptr_t atomic_int_fast8_t;
+typedef intptr_t atomic_uint_fast8_t;
+typedef intptr_t atomic_int_fast16_t;
+typedef intptr_t atomic_uint_fast16_t;
+typedef intptr_t atomic_int_fast32_t;
+typedef intptr_t atomic_uint_fast32_t;
+typedef intptr_t atomic_int_fast64_t;
+typedef intptr_t atomic_uint_fast64_t;
+typedef intptr_t atomic_intptr_t;
+typedef intptr_t atomic_uintptr_t;
+typedef intptr_t atomic_size_t;
+typedef intptr_t atomic_ptrdiff_t;
+typedef intptr_t atomic_intmax_t;
+typedef intptr_t atomic_uintmax_t;
+
+#define atomic_store(object, desired)   \
+do {                                    \
+    *(object) = (desired);              \
+} while (0)
+
+#define atomic_store_explicit(object, desired, order) \
+    atomic_store(object, desired)
+
+#define atomic_load(object) \
+    (*(object))
+
+#define atomic_load_explicit(object, order) \
+    atomic_load(object)
+
+static inline intptr_t atomic_exchange(intptr_t *object, intptr_t desired)
+{
+    intptr_t ret = *object;
+    *object = desired;
+    return ret;
+}
+
+#define atomic_exchange_explicit(object, desired, order) \
+    atomic_exchange(object, desired)
+
+static inline int atomic_compare_exchange_strong(intptr_t *object, intptr_t *expected,
+                                                 intptr_t desired)
+{
+    int ret;
+    if (*object == *expected) {
+        *object = desired;
+        ret = 1;
+    } else {
+        *expected = *object;
+        ret = 0;
+    }
+    return ret;
+}
+
+#define atomic_compare_exchange_strong_explicit(object, expected, desired, success, failure) \
+    atomic_compare_exchange_strong(object, expected, desired)
+
+#define atomic_compare_exchange_weak(object, expected, desired) \
+    atomic_compare_exchange_strong(object, expected, desired)
+
+#define atomic_compare_exchange_weak_explicit(object, expected, desired, success, failure) \
+    atomic_compare_exchange_weak(object, expected, desired)
+
+#define FETCH_MODIFY(opname, op)                                            \
+static inline intptr_t atomic_fetch_ ## opname(intptr_t *object, intptr_t operand) \
+{                                                                    \
+    intptr_t ret;                                                    \
+    ret = *object;                                                   \
+    *object = *object op operand;                                    \
+    return ret;                                                      \
+}
+
+FETCH_MODIFY(add, +)
+FETCH_MODIFY(sub, -)
+FETCH_MODIFY(or,  |)
+FETCH_MODIFY(xor, ^)
+FETCH_MODIFY(and, &)
+
+#undef FETCH_MODIFY
+
+#define atomic_fetch_add_explicit(object, operand, order) \
+    atomic_fetch_add(object, operand)
+
+#define atomic_fetch_sub_explicit(object, operand, order) \
+    atomic_fetch_sub(object, operand)
+
+#define atomic_fetch_or_explicit(object, operand, order) \
+    atomic_fetch_or(object, operand)
+
+#define atomic_fetch_xor_explicit(object, operand, order) \
+    atomic_fetch_xor(object, operand)
+
+#define atomic_fetch_and_explicit(object, operand, order) \
+    atomic_fetch_and(object, operand)
+
+#define atomic_flag_test_and_set(object) \
+    atomic_exchange(object, 1)
+
+#define atomic_flag_test_and_set_explicit(object, order) \
+    atomic_flag_test_and_set(object)
+
+#define atomic_flag_clear(object) \
+    atomic_store(object, 0)
+
+#define atomic_flag_clear_explicit(object, order) \
+    atomic_flag_clear(object)
+
+#endif /* COMPAT_ATOMICS_DUMMY_STDATOMIC_H */

externals/ffmpeg/ffmpeg/compat/atomics/gcc/stdatomic.h

@@ -0,0 +1,173 @@
+/*
+ * 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
+ */
+
+/*
+ * based on vlc_atomic.h from VLC
+ * Copyright (C) 2010 Rémi Denis-Courmont
+ */
+
+#ifndef COMPAT_ATOMICS_GCC_STDATOMIC_H
+#define COMPAT_ATOMICS_GCC_STDATOMIC_H
+
+#include <stddef.h>
+#include <stdint.h>
+
+#define ATOMIC_FLAG_INIT 0
+
+#define ATOMIC_VAR_INIT(value) (value)
+
+#define atomic_init(obj, value) \
+do {                            \
+    *(obj) = (value);           \
+} while(0)
+
+#define kill_dependency(y) ((void)0)
+
+#define atomic_thread_fence(order) \
+    __sync_synchronize()
+
+#define atomic_signal_fence(order) \
+    ((void)0)
+
+#define atomic_is_lock_free(obj) 0
+
+typedef         _Bool      atomic_flag;
+typedef         _Bool      atomic_bool;
+typedef          char      atomic_char;
+typedef   signed char      atomic_schar;
+typedef unsigned char      atomic_uchar;
+typedef          short     atomic_short;
+typedef unsigned short     atomic_ushort;
+typedef          int       atomic_int;
+typedef unsigned int       atomic_uint;
+typedef          long      atomic_long;
+typedef unsigned long      atomic_ulong;
+typedef          long long atomic_llong;
+typedef unsigned long long atomic_ullong;
+typedef          wchar_t   atomic_wchar_t;
+typedef       int_least8_t atomic_int_least8_t;
+typedef      uint_least8_t atomic_uint_least8_t;
+typedef      int_least16_t atomic_int_least16_t;
+typedef     uint_least16_t atomic_uint_least16_t;
+typedef      int_least32_t atomic_int_least32_t;
+typedef     uint_least32_t atomic_uint_least32_t;
+typedef      int_least64_t atomic_int_least64_t;
+typedef     uint_least64_t atomic_uint_least64_t;
+typedef       int_fast8_t atomic_int_fast8_t;
+typedef      uint_fast8_t atomic_uint_fast8_t;
+typedef      int_fast16_t atomic_int_fast16_t;
+typedef     uint_fast16_t atomic_uint_fast16_t;
+typedef      int_fast32_t atomic_int_fast32_t;
+typedef     uint_fast32_t atomic_uint_fast32_t;
+typedef      int_fast64_t atomic_int_fast64_t;
+typedef     uint_fast64_t atomic_uint_fast64_t;
+typedef          intptr_t atomic_intptr_t;
+typedef         uintptr_t atomic_uintptr_t;
+typedef            size_t atomic_size_t;
+typedef         ptrdiff_t atomic_ptrdiff_t;
+typedef          intmax_t atomic_intmax_t;
+typedef         uintmax_t atomic_uintmax_t;
+
+#define atomic_store(object, desired)   \
+do {                                    \
+    *(object) = (desired);              \
+    __sync_synchronize();               \
+} while (0)
+
+#define atomic_store_explicit(object, desired, order) \
+    atomic_store(object, desired)
+
+#define atomic_load(object) \
+    (__sync_synchronize(), *(object))
+
+#define atomic_load_explicit(object, order) \
+    atomic_load(object)
+
+#define atomic_exchange(object, desired)                            \
+({                                                                  \
+    __typeof__(object) _obj = (object);                             \
+    __typeof__(*object) _old;                                       \
+    do                                                              \
+        _old = atomic_load(_obj);                                   \
+    while (!__sync_bool_compare_and_swap(_obj, _old, (desired)));   \
+    _old;                                                           \
+})
+
+#define atomic_exchange_explicit(object, desired, order) \
+    atomic_exchange(object, desired)
+
+#define atomic_compare_exchange_strong(object, expected, desired)   \
+({                                                                  \
+    __typeof__(object) _exp = (expected);                           \
+    __typeof__(*object) _old = *_exp;                               \
+    *_exp = __sync_val_compare_and_swap((object), _old, (desired)); \
+    *_exp == _old;                                                  \
+})
+
+#define atomic_compare_exchange_strong_explicit(object, expected, desired, success, failure) \
+    atomic_compare_exchange_strong(object, expected, desired)
+
+#define atomic_compare_exchange_weak(object, expected, desired) \
+    atomic_compare_exchange_strong(object, expected, desired)
+
+#define atomic_compare_exchange_weak_explicit(object, expected, desired, success, failure) \
+    atomic_compare_exchange_weak(object, expected, desired)
+
+#define atomic_fetch_add(object, operand) \
+    __sync_fetch_and_add(object, operand)
+
+#define atomic_fetch_add_explicit(object, operand, order) \
+    atomic_fetch_add(object, operand)
+
+#define atomic_fetch_sub(object, operand) \
+    __sync_fetch_and_sub(object, operand)
+
+#define atomic_fetch_sub_explicit(object, operand, order) \
+    atomic_fetch_sub(object, operand)
+
+#define atomic_fetch_or(object, operand) \
+    __sync_fetch_and_or(object, operand)
+
+#define atomic_fetch_or_explicit(object, operand, order) \
+    atomic_fetch_or(object, operand)
+
+#define atomic_fetch_xor(object, operand) \
+    __sync_fetch_and_xor(object, operand)
+
+#define atomic_fetch_xor_explicit(object, operand, order) \
+    atomic_fetch_xor(object, operand)
+
+#define atomic_fetch_and(object, operand) \
+    __sync_fetch_and_and(object, operand)
+
+#define atomic_fetch_and_explicit(object, operand, order) \
+    atomic_fetch_and(object, operand)
+
+#define atomic_flag_test_and_set(object) \
+    atomic_exchange(object, 1)
+
+#define atomic_flag_test_and_set_explicit(object, order) \
+    atomic_flag_test_and_set(object)
+
+#define atomic_flag_clear(object) \
+    atomic_store(object, 0)
+
+#define atomic_flag_clear_explicit(object, order) \
+    atomic_flag_clear(object)
+
+#endif /* COMPAT_ATOMICS_GCC_STDATOMIC_H */

externals/ffmpeg/ffmpeg/compat/atomics/pthread/stdatomic.c

@@ -0,0 +1,39 @@
+/*
+ * 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
+ */
+
+/*
+ * based on vlc_atomic.h from VLC
+ * Copyright (C) 2010 Rémi Denis-Courmont
+ */
+
+#include <pthread.h>
+#include <stdint.h>
+
+#include "stdatomic.h"
+
+static pthread_mutex_t atomic_lock = PTHREAD_MUTEX_INITIALIZER;
+
+void avpriv_atomic_lock(void)
+{
+    pthread_mutex_lock(&atomic_lock);
+}
+
+void avpriv_atomic_unlock(void)
+{
+    pthread_mutex_unlock(&atomic_lock);
+}

externals/ffmpeg/ffmpeg/compat/atomics/pthread/stdatomic.h

@@ -0,0 +1,197 @@
+/*
+ * 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
+ */
+
+/*
+ * based on vlc_atomic.h from VLC
+ * Copyright (C) 2010 Rémi Denis-Courmont
+ */
+
+#ifndef COMPAT_ATOMICS_PTHREAD_STDATOMIC_H
+#define COMPAT_ATOMICS_PTHREAD_STDATOMIC_H
+
+#include <stdint.h>
+
+#define ATOMIC_FLAG_INIT 0
+
+#define ATOMIC_VAR_INIT(value) (value)
+
+#define atomic_init(obj, value) \
+do {                            \
+    *(obj) = (value);           \
+} while(0)
+
+#define kill_dependency(y) ((void)0)
+
+#define atomic_signal_fence(order) \
+    ((void)0)
+
+#define atomic_is_lock_free(obj) 0
+
+typedef intptr_t atomic_flag;
+typedef intptr_t atomic_bool;
+typedef intptr_t atomic_char;
+typedef intptr_t atomic_schar;
+typedef intptr_t atomic_uchar;
+typedef intptr_t atomic_short;
+typedef intptr_t atomic_ushort;
+typedef intptr_t atomic_int;
+typedef intptr_t atomic_uint;
+typedef intptr_t atomic_long;
+typedef intptr_t atomic_ulong;
+typedef intptr_t atomic_llong;
+typedef intptr_t atomic_ullong;
+typedef intptr_t atomic_wchar_t;
+typedef intptr_t atomic_int_least8_t;
+typedef intptr_t atomic_uint_least8_t;
+typedef intptr_t atomic_int_least16_t;
+typedef intptr_t atomic_uint_least16_t;
+typedef intptr_t atomic_int_least32_t;
+typedef intptr_t atomic_uint_least32_t;
+typedef intptr_t atomic_int_least64_t;
+typedef intptr_t atomic_uint_least64_t;
+typedef intptr_t atomic_int_fast8_t;
+typedef intptr_t atomic_uint_fast8_t;
+typedef intptr_t atomic_int_fast16_t;
+typedef intptr_t atomic_uint_fast16_t;
+typedef intptr_t atomic_int_fast32_t;
+typedef intptr_t atomic_uint_fast32_t;
+typedef intptr_t atomic_int_fast64_t;
+typedef intptr_t atomic_uint_fast64_t;
+typedef intptr_t atomic_intptr_t;
+typedef intptr_t atomic_uintptr_t;
+typedef intptr_t atomic_size_t;
+typedef intptr_t atomic_ptrdiff_t;
+typedef intptr_t atomic_intmax_t;
+typedef intptr_t atomic_uintmax_t;
+
+void avpriv_atomic_lock(void);
+void avpriv_atomic_unlock(void);
+
+static inline void atomic_thread_fence(int order)
+{
+    avpriv_atomic_lock();
+    avpriv_atomic_unlock();
+}
+
+static inline void atomic_store(intptr_t *object, intptr_t desired)
+{
+    avpriv_atomic_lock();
+    *object = desired;
+    avpriv_atomic_unlock();
+}
+
+#define atomic_store_explicit(object, desired, order) \
+    atomic_store(object, desired)
+
+static inline intptr_t atomic_load(intptr_t *object)
+{
+    intptr_t ret;
+    avpriv_atomic_lock();
+    ret = *object;
+    avpriv_atomic_unlock();
+    return ret;
+}
+
+#define atomic_load_explicit(object, order) \
+    atomic_load(object)
+
+static inline intptr_t atomic_exchange(intptr_t *object, intptr_t desired)
+{
+    intptr_t ret;
+    avpriv_atomic_lock();
+    ret     = *object;
+    *object = desired;
+    avpriv_atomic_unlock();
+    return ret;
+}
+
+#define atomic_exchange_explicit(object, desired, order) \
+    atomic_exchange(object, desired)
+
+static inline int atomic_compare_exchange_strong(intptr_t *object, intptr_t *expected,
+                                                 intptr_t desired)
+{
+    int ret;
+    avpriv_atomic_lock();
+    if (*object == *expected) {
+        ret     = 1;
+        *object = desired;
+    } else {
+        ret = 0;
+        *expected = *object;
+    }
+    avpriv_atomic_unlock();
+    return ret;
+}
+
+#define atomic_compare_exchange_strong_explicit(object, expected, desired, success, failure) \
+    atomic_compare_exchange_strong(object, expected, desired)
+
+#define atomic_compare_exchange_weak(object, expected, desired) \
+    atomic_compare_exchange_strong(object, expected, desired)
+
+#define atomic_compare_exchange_weak_explicit(object, expected, desired, success, failure) \
+    atomic_compare_exchange_weak(object, expected, desired)
+
+#define FETCH_MODIFY(opname, op)                                                   \
+static inline intptr_t atomic_fetch_ ## opname(intptr_t *object, intptr_t operand) \
+{                                                                                  \
+    intptr_t ret;                                                                  \
+    avpriv_atomic_lock();                                                          \
+    ret = *object;                                                                 \
+    *object = *object op operand;                                                  \
+    avpriv_atomic_unlock();                                                        \
+    return ret;                                                                    \
+}
+
+FETCH_MODIFY(add, +)
+FETCH_MODIFY(sub, -)
+FETCH_MODIFY(or,  |)
+FETCH_MODIFY(xor, ^)
+FETCH_MODIFY(and, &)
+
+#undef FETCH_MODIFY
+
+#define atomic_fetch_add_explicit(object, operand, order) \
+    atomic_fetch_add(object, operand)
+
+#define atomic_fetch_sub_explicit(object, operand, order) \
+    atomic_fetch_sub(object, operand)
+
+#define atomic_fetch_or_explicit(object, operand, order) \
+    atomic_fetch_or(object, operand)
+
+#define atomic_fetch_xor_explicit(object, operand, order) \
+    atomic_fetch_xor(object, operand)
+
+#define atomic_fetch_and_explicit(object, operand, order) \
+    atomic_fetch_and(object, operand)
+
+#define atomic_flag_test_and_set(object) \
+    atomic_exchange(object, 1)
+
+#define atomic_flag_test_and_set_explicit(object, order) \
+    atomic_flag_test_and_set(object)
+
+#define atomic_flag_clear(object) \
+    atomic_store(object, 0)
+
+#define atomic_flag_clear_explicit(object, order) \
+    atomic_flag_clear(object)
+
+#endif /* COMPAT_ATOMICS_PTHREAD_STDATOMIC_H */

externals/ffmpeg/ffmpeg/compat/atomics/suncc/stdatomic.h

@@ -0,0 +1,186 @@
+/*
+ * 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_ATOMICS_SUNCC_STDATOMIC_H
+#define COMPAT_ATOMICS_SUNCC_STDATOMIC_H
+
+#include <atomic.h>
+#include <mbarrier.h>
+#include <stddef.h>
+#include <stdint.h>
+
+#define ATOMIC_FLAG_INIT 0
+
+#define ATOMIC_VAR_INIT(value) (value)
+
+#define atomic_init(obj, value) \
+do {                            \
+    *(obj) = (value);           \
+} while(0)
+
+#define kill_dependency(y) ((void)0)
+
+#define atomic_thread_fence(order) \
+    __machine_rw_barrier();
+
+#define atomic_signal_fence(order) \
+    ((void)0)
+
+#define atomic_is_lock_free(obj) 0
+
+typedef intptr_t atomic_flag;
+typedef intptr_t atomic_bool;
+typedef intptr_t atomic_char;
+typedef intptr_t atomic_schar;
+typedef intptr_t atomic_uchar;
+typedef intptr_t atomic_short;
+typedef intptr_t atomic_ushort;
+typedef intptr_t atomic_int;
+typedef intptr_t atomic_uint;
+typedef intptr_t atomic_long;
+typedef intptr_t atomic_ulong;
+typedef intptr_t atomic_llong;
+typedef intptr_t atomic_ullong;
+typedef intptr_t atomic_wchar_t;
+typedef intptr_t atomic_int_least8_t;
+typedef intptr_t atomic_uint_least8_t;
+typedef intptr_t atomic_int_least16_t;
+typedef intptr_t atomic_uint_least16_t;
+typedef intptr_t atomic_int_least32_t;
+typedef intptr_t atomic_uint_least32_t;
+typedef intptr_t atomic_int_least64_t;
+typedef intptr_t atomic_uint_least64_t;
+typedef intptr_t atomic_int_fast8_t;
+typedef intptr_t atomic_uint_fast8_t;
+typedef intptr_t atomic_int_fast16_t;
+typedef intptr_t atomic_uint_fast16_t;
+typedef intptr_t atomic_int_fast32_t;
+typedef intptr_t atomic_uint_fast32_t;
+typedef intptr_t atomic_int_fast64_t;
+typedef intptr_t atomic_uint_fast64_t;
+typedef intptr_t atomic_intptr_t;
+typedef intptr_t atomic_uintptr_t;
+typedef intptr_t atomic_size_t;
+typedef intptr_t atomic_ptrdiff_t;
+typedef intptr_t atomic_intmax_t;
+typedef intptr_t atomic_uintmax_t;
+
+static inline void atomic_store(intptr_t *object, intptr_t desired)
+{
+    *object = desired;
+    __machine_rw_barrier();
+}
+
+#define atomic_store_explicit(object, desired, order) \
+    atomic_store(object, desired)
+
+static inline intptr_t atomic_load(intptr_t *object)
+{
+    __machine_rw_barrier();
+    return *object;
+}
+
+#define atomic_load_explicit(object, order) \
+    atomic_load(object)
+
+#define atomic_exchange(object, desired) \
+    atomic_swap_ptr(object, desired)
+
+#define atomic_exchange_explicit(object, desired, order) \
+    atomic_exchange(object, desired)
+
+static inline int atomic_compare_exchange_strong(intptr_t *object, intptr_t *expected,
+                                                 intptr_t desired)
+{
+    intptr_t  old = *expected;
+    *expected = (intptr_t)atomic_cas_ptr(object, (void *)old, (void *)desired);
+    return *expected == old;
+}
+
+#define atomic_compare_exchange_strong_explicit(object, expected, desired, success, failure) \
+    atomic_compare_exchange_strong(object, expected, desired)
+
+#define atomic_compare_exchange_weak(object, expected, desired) \
+    atomic_compare_exchange_strong(object, expected, desired)
+
+#define atomic_compare_exchange_weak_explicit(object, expected, desired, success, failure) \
+    atomic_compare_exchange_weak(object, expected, desired)
+
+static inline intptr_t atomic_fetch_add(intptr_t *object, intptr_t operand)
+{
+    return atomic_add_ptr_nv(object, operand) - operand;
+}
+
+#define atomic_fetch_sub(object, operand) \
+    atomic_fetch_add(object, -(operand))
+
+static inline intptr_t atomic_fetch_or(intptr_t *object, intptr_t operand)
+{
+    intptr_t old;
+    do {
+        old = atomic_load(object);
+    } while (!atomic_compare_exchange_strong(object, old, old | operand));
+    return old;
+}
+
+static inline intptr_t atomic_fetch_xor(intptr_t *object, intptr_t operand)
+{
+    intptr_t old;
+    do {
+        old = atomic_load(object);
+    } while (!atomic_compare_exchange_strong(object, old, old ^ operand));
+    return old;
+}
+
+static inline intptr_t atomic_fetch_and(intptr_t *object, intptr_t operand)
+{
+    intptr_t old;
+    do {
+        old = atomic_load(object);
+    } while (!atomic_compare_exchange_strong(object, old, old & operand));
+    return old;
+}
+
+#define atomic_fetch_add_explicit(object, operand, order) \
+    atomic_fetch_add(object, operand)
+
+#define atomic_fetch_sub_explicit(object, operand, order) \
+    atomic_fetch_sub(object, operand)
+
+#define atomic_fetch_or_explicit(object, operand, order) \
+    atomic_fetch_or(object, operand)
+
+#define atomic_fetch_xor_explicit(object, operand, order) \
+    atomic_fetch_xor(object, operand)
+
+#define atomic_fetch_and_explicit(object, operand, order) \
+    atomic_fetch_and(object, operand)
+
+#define atomic_flag_test_and_set(object) \
+    atomic_exchange(object, 1)
+
+#define atomic_flag_test_and_set_explicit(object, order) \
+    atomic_flag_test_and_set(object)
+
+#define atomic_flag_clear(object) \
+    atomic_store(object, 0)
+
+#define atomic_flag_clear_explicit(object, order) \
+    atomic_flag_clear(object)
+
+#endif /* COMPAT_ATOMICS_SUNCC_STDATOMIC_H */

externals/ffmpeg/ffmpeg/compat/atomics/win32/stdatomic.h

@@ -0,0 +1,181 @@
+/*
+ * 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_ATOMICS_WIN32_STDATOMIC_H
+#define COMPAT_ATOMICS_WIN32_STDATOMIC_H
+
+#define WIN32_LEAN_AND_MEAN
+#include <stddef.h>
+#include <stdint.h>
+#include <windows.h>
+
+#define ATOMIC_FLAG_INIT 0
+
+#define ATOMIC_VAR_INIT(value) (value)
+
+#define atomic_init(obj, value) \
+do {                            \
+    *(obj) = (value);           \
+} while(0)
+
+#define kill_dependency(y) ((void)0)
+
+#define atomic_thread_fence(order) \
+    MemoryBarrier();
+
+#define atomic_signal_fence(order) \
+    ((void)0)
+
+#define atomic_is_lock_free(obj) 0
+
+typedef intptr_t atomic_flag;
+typedef intptr_t atomic_bool;
+typedef intptr_t atomic_char;
+typedef intptr_t atomic_schar;
+typedef intptr_t atomic_uchar;
+typedef intptr_t atomic_short;
+typedef intptr_t atomic_ushort;
+typedef intptr_t atomic_int;
+typedef intptr_t atomic_uint;
+typedef intptr_t atomic_long;
+typedef intptr_t atomic_ulong;
+typedef intptr_t atomic_llong;
+typedef intptr_t atomic_ullong;
+typedef intptr_t atomic_wchar_t;
+typedef intptr_t atomic_int_least8_t;
+typedef intptr_t atomic_uint_least8_t;
+typedef intptr_t atomic_int_least16_t;
+typedef intptr_t atomic_uint_least16_t;
+typedef intptr_t atomic_int_least32_t;
+typedef intptr_t atomic_uint_least32_t;
+typedef intptr_t atomic_int_least64_t;
+typedef intptr_t atomic_uint_least64_t;
+typedef intptr_t atomic_int_fast8_t;
+typedef intptr_t atomic_uint_fast8_t;
+typedef intptr_t atomic_int_fast16_t;
+typedef intptr_t atomic_uint_fast16_t;
+typedef intptr_t atomic_int_fast32_t;
+typedef intptr_t atomic_uint_fast32_t;
+typedef intptr_t atomic_int_fast64_t;
+typedef intptr_t atomic_uint_fast64_t;
+typedef intptr_t atomic_intptr_t;
+typedef intptr_t atomic_uintptr_t;
+typedef intptr_t atomic_size_t;
+typedef intptr_t atomic_ptrdiff_t;
+typedef intptr_t atomic_intmax_t;
+typedef intptr_t atomic_uintmax_t;
+
+#define atomic_store(object, desired)   \
+do {                                    \
+    *(object) = (desired);              \
+    MemoryBarrier();                    \
+} while (0)
+
+#define atomic_store_explicit(object, desired, order) \
+    atomic_store(object, desired)
+
+#define atomic_load(object) \
+    (MemoryBarrier(), *(object))
+
+#define atomic_load_explicit(object, order) \
+    atomic_load(object)
+
+#define atomic_exchange(object, desired) \
+    InterlockedExchangePointer(object, desired);
+
+#define atomic_exchange_explicit(object, desired, order) \
+    atomic_exchange(object, desired)
+
+static inline int atomic_compare_exchange_strong(intptr_t *object, intptr_t *expected,
+                                                 intptr_t desired)
+{
+    intptr_t old = *expected;
+    *expected = (intptr_t)InterlockedCompareExchangePointer(
+        (PVOID *)object, (PVOID)desired, (PVOID)old);
+    return *expected == old;
+}
+
+#define atomic_compare_exchange_strong_explicit(object, expected, desired, success, failure) \
+    atomic_compare_exchange_strong(object, expected, desired)
+
+#define atomic_compare_exchange_weak(object, expected, desired) \
+    atomic_compare_exchange_strong(object, expected, desired)
+
+#define atomic_compare_exchange_weak_explicit(object, expected, desired, success, failure) \
+    atomic_compare_exchange_weak(object, expected, desired)
+
+#ifdef _WIN64
+#define atomic_fetch_add(object, operand) \
+    InterlockedExchangeAdd64(object, operand)
+
+#define atomic_fetch_sub(object, operand) \
+    InterlockedExchangeAdd64(object, -(operand))
+
+#define atomic_fetch_or(object, operand) \
+    InterlockedOr64(object, operand)
+
+#define atomic_fetch_xor(object, operand) \
+    InterlockedXor64(object, operand)
+
+#define atomic_fetch_and(object, operand) \
+    InterlockedAnd64(object, operand)
+#else
+#define atomic_fetch_add(object, operand) \
+    InterlockedExchangeAdd(object, operand)
+
+#define atomic_fetch_sub(object, operand) \
+    InterlockedExchangeAdd(object, -(operand))
+
+#define atomic_fetch_or(object, operand) \
+    InterlockedOr(object, operand)
+
+#define atomic_fetch_xor(object, operand) \
+    InterlockedXor(object, operand)
+
+#define atomic_fetch_and(object, operand) \
+    InterlockedAnd(object, operand)
+#endif /* _WIN64 */
+
+#define atomic_fetch_add_explicit(object, operand, order) \
+    atomic_fetch_add(object, operand)
+
+#define atomic_fetch_sub_explicit(object, operand, order) \
+    atomic_fetch_sub(object, operand)
+
+#define atomic_fetch_or_explicit(object, operand, order) \
+    atomic_fetch_or(object, operand)
+
+#define atomic_fetch_xor_explicit(object, operand, order) \
+    atomic_fetch_xor(object, operand)
+
+#define atomic_fetch_and_explicit(object, operand, order) \
+    atomic_fetch_and(object, operand)
+
+#define atomic_flag_test_and_set(object) \
+    atomic_exchange(object, 1)
+
+#define atomic_flag_test_and_set_explicit(object, order) \
+    atomic_flag_test_and_set(object)
+
+#define atomic_flag_clear(object) \
+    atomic_store(object, 0)
+
+#define atomic_flag_clear_explicit(object, order) \
+    atomic_flag_clear(object)
+
+#endif /* COMPAT_ATOMICS_WIN32_STDATOMIC_H */

externals/ffmpeg/ffmpeg/compat/cuda/cuda_runtime.h

@@ -0,0 +1,188 @@
+/*
+ * Minimum CUDA compatibility definitions header
+ *
+ * Copyright (c) 2019 rcombs
+ *
+ * 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_CUDA_CUDA_RUNTIME_H
+#define COMPAT_CUDA_CUDA_RUNTIME_H
+
+// Common macros
+#define __global__ __attribute__((global))
+#define __device__ __attribute__((device))
+#define __device_builtin__ __attribute__((device_builtin))
+#define __align__(N) __attribute__((aligned(N)))
+#define __inline__ __inline__ __attribute__((always_inline))
+
+#define max(a, b) ((a) > (b) ? (a) : (b))
+#define min(a, b) ((a) < (b) ? (a) : (b))
+#define abs(x) ((x) < 0 ? -(x) : (x))
+
+#define atomicAdd(a, b) (__atomic_fetch_add(a, b, __ATOMIC_SEQ_CST))
+
+// Basic typedefs
+typedef __device_builtin__ unsigned long long cudaTextureObject_t;
+
+typedef struct __device_builtin__ __align__(2) uchar2
+{
+    unsigned char x, y;
+} uchar2;
+
+typedef struct __device_builtin__ __align__(4) ushort2
+{
+    unsigned short x, y;
+} ushort2;
+
+typedef struct __device_builtin__ __align__(8) float2
+{
+    float x, y;
+} float2;
+
+typedef struct __device_builtin__ __align__(8) int2
+{
+    int x, y;
+} int2;
+
+typedef struct __device_builtin__ uint3
+{
+    unsigned int x, y, z;
+} uint3;
+
+typedef struct uint3 dim3;
+
+typedef struct __device_builtin__ __align__(4) uchar4
+{
+    unsigned char x, y, z, w;
+} uchar4;
+
+typedef struct __device_builtin__ __align__(8) ushort4
+{
+    unsigned short x, y, z, w;
+} ushort4;
+
+typedef struct __device_builtin__ __align__(16) int4
+{
+    int x, y, z, w;
+} int4;
+
+typedef struct __device_builtin__ __align__(16) float4
+{
+    float x, y, z, w;
+} float4;
+
+// Accessors for special registers
+#define GETCOMP(reg, comp) \
+    asm("mov.u32 %0, %%" #reg "." #comp ";" : "=r"(tmp)); \
+    ret.comp = tmp;
+
+#define GET(name, reg) static inline __device__ uint3 name() {\
+    uint3 ret; \
+    unsigned tmp; \
+    GETCOMP(reg, x) \
+    GETCOMP(reg, y) \
+    GETCOMP(reg, z) \
+    return ret; \
+}
+
+GET(getBlockIdx, ctaid)
+GET(getBlockDim, ntid)
+GET(getThreadIdx, tid)
+
+// Instead of externs for these registers, we turn access to them into calls into trivial ASM
+#define blockIdx (getBlockIdx())
+#define blockDim (getBlockDim())
+#define threadIdx (getThreadIdx())
+
+// Basic initializers (simple macros rather than inline functions)
+#define make_int2(a, b) ((int2){.x = a, .y = b})
+#define make_uchar2(a, b) ((uchar2){.x = a, .y = b})
+#define make_ushort2(a, b) ((ushort2){.x = a, .y = b})
+#define make_float2(a, b) ((float2){.x = a, .y = b})
+#define make_int4(a, b, c, d) ((int4){.x = a, .y = b, .z = c, .w = d})
+#define make_uchar4(a, b, c, d) ((uchar4){.x = a, .y = b, .z = c, .w = d})
+#define make_ushort4(a, b, c, d) ((ushort4){.x = a, .y = b, .z = c, .w = d})
+#define make_float4(a, b, c, d) ((float4){.x = a, .y = b, .z = c, .w = d})
+
+// Conversions from the tex instruction's 4-register output to various types
+#define TEX2D(type, ret) static inline __device__ void conv(type* out, unsigned a, unsigned b, unsigned c, unsigned d) {*out = (ret);}
+
+TEX2D(unsigned char, a & 0xFF)
+TEX2D(unsigned short, a & 0xFFFF)
+TEX2D(float, a)
+TEX2D(uchar2, make_uchar2(a & 0xFF, b & 0xFF))
+TEX2D(ushort2, make_ushort2(a & 0xFFFF, b & 0xFFFF))
+TEX2D(float2, make_float2(a, b))
+TEX2D(uchar4, make_uchar4(a & 0xFF, b & 0xFF, c & 0xFF, d & 0xFF))
+TEX2D(ushort4, make_ushort4(a & 0xFFFF, b & 0xFFFF, c & 0xFFFF, d & 0xFFFF))
+TEX2D(float4, make_float4(a, b, c, d))
+
+// Template calling tex instruction and converting the output to the selected type
+template<typename T>
+inline __device__ T tex2D(cudaTextureObject_t texObject, float x, float y)
+{
+  T ret;
+  unsigned ret1, ret2, ret3, ret4;
+  asm("tex.2d.v4.u32.f32 {%0, %1, %2, %3}, [%4, {%5, %6}];" :
+      "=r"(ret1), "=r"(ret2), "=r"(ret3), "=r"(ret4) :
+      "l"(texObject), "f"(x), "f"(y));
+  conv(&ret, ret1, ret2, ret3, ret4);
+  return ret;
+}
+
+template<>
+inline __device__ float4 tex2D<float4>(cudaTextureObject_t texObject, float x, float y)
+{
+    float4 ret;
+    asm("tex.2d.v4.f32.f32 {%0, %1, %2, %3}, [%4, {%5, %6}];" :
+        "=r"(ret.x), "=r"(ret.y), "=r"(ret.z), "=r"(ret.w) :
+        "l"(texObject), "f"(x), "f"(y));
+    return ret;
+}
+
+template<>
+inline __device__ float tex2D<float>(cudaTextureObject_t texObject, float x, float y)
+{
+    return tex2D<float4>(texObject, x, y).x;
+}
+
+template<>
+inline __device__ float2 tex2D<float2>(cudaTextureObject_t texObject, float x, float y)
+{
+    float4 ret = tex2D<float4>(texObject, x, y);
+    return make_float2(ret.x, ret.y);
+}
+
+// Math helper functions
+static inline __device__ float floorf(float a) { return __builtin_floorf(a); }
+static inline __device__ float floor(float a) { return __builtin_floorf(a); }
+static inline __device__ double floor(double a) { return __builtin_floor(a); }
+static inline __device__ float ceilf(float a) { return __builtin_ceilf(a); }
+static inline __device__ float ceil(float a) { return __builtin_ceilf(a); }
+static inline __device__ double ceil(double a) { return __builtin_ceil(a); }
+static inline __device__ float truncf(float a) { return __builtin_truncf(a); }
+static inline __device__ float trunc(float a) { return __builtin_truncf(a); }
+static inline __device__ double trunc(double a) { return __builtin_trunc(a); }
+static inline __device__ float fabsf(float a) { return __builtin_fabsf(a); }
+static inline __device__ float fabs(float a) { return __builtin_fabsf(a); }
+static inline __device__ double fabs(double a) { return __builtin_fabs(a); }
+
+static inline __device__ float __sinf(float a) { return __nvvm_sin_approx_f(a); }
+static inline __device__ float __cosf(float a) { return __nvvm_cos_approx_f(a); }
+
+#endif /* COMPAT_CUDA_CUDA_RUNTIME_H */

externals/ffmpeg/ffmpeg/compat/cuda/dynlink_loader.h

@@ -0,0 +1,33 @@
+/*
+ * 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_CUDA_DYNLINK_LOADER_H
+#define COMPAT_CUDA_DYNLINK_LOADER_H
+
+#include "libavutil/log.h"
+#include "compat/w32dlfcn.h"
+
+#define FFNV_LOAD_FUNC(path) dlopen((path), RTLD_LAZY)
+#define FFNV_SYM_FUNC(lib, sym) dlsym((lib), (sym))
+#define FFNV_FREE_FUNC(lib) dlclose(lib)
+#define FFNV_LOG_FUNC(logctx, msg, ...) av_log(logctx, AV_LOG_ERROR, msg,  __VA_ARGS__)
+#define FFNV_DEBUG_LOG_FUNC(logctx, msg, ...) av_log(logctx, AV_LOG_DEBUG, msg,  __VA_ARGS__)
+
+#include <ffnvcodec/dynlink_loader.h>
+
+#endif /* COMPAT_CUDA_DYNLINK_LOADER_H */

externals/ffmpeg/ffmpeg/compat/cuda/ptx2c.sh

@@ -0,0 +1,34 @@
+#!/bin/sh
+
+# Copyright (c) 2017, NVIDIA CORPORATION. All rights reserved.
+#
+# 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.
+
+set -e
+
+OUT="$1"
+IN="$2"
+NAME="$(basename "$IN" | sed 's/\..*//')"
+
+printf "const char %s_ptx[] = \\" "$NAME" > "$OUT"
+echo >> "$OUT"
+sed -e "$(printf 's/\r//g')" -e 's/["\\]/\\&/g' -e "$(printf 's/^/\t"/')" -e 's/$/\\n"/' < "$IN" >> "$OUT"
+echo ";" >> "$OUT"
+
+exit 0

externals/ffmpeg/ffmpeg/compat/dispatch_semaphore/semaphore.h

@@ -0,0 +1,42 @@
+/*
+ * 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_DISPATCH_SEMAPHORE_SEMAPHORE_H
+#define COMPAT_DISPATCH_SEMAPHORE_SEMAPHORE_H
+
+#include <dispatch/dispatch.h>
+#include <errno.h>
+
+#define sem_t dispatch_semaphore_t
+#define sem_post(psem)              dispatch_semaphore_signal(*psem)
+#define sem_wait(psem)              dispatch_semaphore_wait(*psem, DISPATCH_TIME_FOREVER)
+#define sem_timedwait(psem, val)    dispatch_semaphore_wait(*psem, dispatch_walltime(val, 0))
+#define sem_destroy(psem)           dispatch_release(*psem)
+
+static inline int compat_sem_init(dispatch_semaphore_t *psem,
+                                  int unused, int val)
+{
+    int ret = !!(*psem = dispatch_semaphore_create(val)) - 1;
+    if (ret < 0)
+        errno = ENOMEM;
+    return ret;
+}
+
+#define sem_init compat_sem_init
+
+#endif /* COMPAT_DISPATCH_SEMAPHORE_SEMAPHORE_H */

externals/ffmpeg/ffmpeg/compat/djgpp/math.c

@@ -0,0 +1,47 @@
+/*
+ * 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 <math.h>
+
+#define FUN(name, type, op) \
+type name(type x, type y) \
+{ \
+    if (fpclassify(x) == FP_NAN) return y; \
+    if (fpclassify(y) == FP_NAN) return x; \
+    return x op y ? x : y; \
+}
+
+FUN(fmin, double, <)
+FUN(fmax, double, >)
+FUN(fminf, float, <)
+FUN(fmaxf, float, >)
+
+long double fmodl(long double x, long double y)
+{
+    return fmod(x, y);
+}
+
+long double scalbnl(long double x, int exp)
+{
+    return scalbn(x, exp);
+}
+
+long double copysignl(long double x, long double y)
+{
+    return copysign(x, y);
+}

externals/ffmpeg/ffmpeg/compat/djgpp/math.h

@@ -0,0 +1,25 @@
+/*
+ * 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
+ */
+
+double fmin(double, double);
+double fmax(double, double);
+float fminf(float, float);
+float fmaxf(float, float);
+long double fmodl(long double, long double);
+long double scalbnl(long double, int);
+long double copysignl(long double, long double);

externals/ffmpeg/ffmpeg/compat/float/float.h

@@ -0,0 +1,35 @@
+/*
+ * 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

externals/ffmpeg/ffmpeg/compat/float/limits.h

@@ -0,0 +1,22 @@
+/*
+ * 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>

externals/ffmpeg/ffmpeg/compat/getopt.c

@@ -0,0 +1,84 @@
+/*
+ * 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;
+}

externals/ffmpeg/ffmpeg/compat/msvcrt/snprintf.c

@@ -0,0 +1,71 @@
+/*
+ * 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;
+}

externals/ffmpeg/ffmpeg/compat/msvcrt/snprintf.h

@@ -0,0 +1,38 @@
+/*
+ * 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_MSVCRT_SNPRINTF_H
+#define COMPAT_MSVCRT_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_MSVCRT_SNPRINTF_H */

externals/ffmpeg/ffmpeg/compat/os2threads.h

@@ -0,0 +1,229 @@
+/*
+ * Copyright (c) 2011-2017 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 COMPAT_OS2THREADS_H
+#define COMPAT_OS2THREADS_H
+
+#define INCL_DOS
+#define INCL_DOSERRORS
+#include <os2.h>
+
+#undef __STRICT_ANSI__          /* for _beginthread() */
+#include <stdlib.h>
+#include <time.h>
+
+#include <sys/builtin.h>
+#include <sys/fmutex.h>
+
+#include "libavutil/attributes.h"
+#include "libavutil/common.h"
+#include "libavutil/time.h"
+
+typedef struct {
+    TID tid;
+    void *(*start_routine)(void *);
+    void *arg;
+    void *result;
+} pthread_t;
+
+typedef void pthread_attr_t;
+
+typedef _fmutex pthread_mutex_t;
+typedef void pthread_mutexattr_t;
+
+#define PTHREAD_MUTEX_INITIALIZER _FMUTEX_INITIALIZER
+
+typedef struct {
+    HEV event_sem;
+    HEV ack_sem;
+    volatile unsigned  wait_count;
+} pthread_cond_t;
+
+typedef void pthread_condattr_t;
+
+typedef struct {
+    volatile int done;
+    _fmutex mtx;
+} pthread_once_t;
+
+#define PTHREAD_ONCE_INIT {0, _FMUTEX_INITIALIZER}
+
+static void thread_entry(void *arg)
+{
+    pthread_t *thread = arg;
+
+    thread->result = thread->start_routine(thread->arg);
+}
+
+static av_always_inline int pthread_create(pthread_t *thread,
+                                           const pthread_attr_t *attr,
+                                           void *(*start_routine)(void*),
+                                           void *arg)
+{
+    thread->start_routine = start_routine;
+    thread->arg = arg;
+    thread->result = NULL;
+
+    thread->tid = _beginthread(thread_entry, NULL, 1024 * 1024, thread);
+
+    return 0;
+}
+
+static av_always_inline int pthread_join(pthread_t thread, void **value_ptr)
+{
+    DosWaitThread(&thread.tid, DCWW_WAIT);
+
+    if (value_ptr)
+        *value_ptr = thread.result;
+
+    return 0;
+}
+
+static av_always_inline int pthread_mutex_init(pthread_mutex_t *mutex,
+                                               const pthread_mutexattr_t *attr)
+{
+    _fmutex_create(mutex, 0);
+
+    return 0;
+}
+
+static av_always_inline int pthread_mutex_destroy(pthread_mutex_t *mutex)
+{
+    _fmutex_close(mutex);
+
+    return 0;
+}
+
+static av_always_inline int pthread_mutex_lock(pthread_mutex_t *mutex)
+{
+    _fmutex_request(mutex, 0);
+
+    return 0;
+}
+
+static av_always_inline int pthread_mutex_unlock(pthread_mutex_t *mutex)
+{
+    _fmutex_release(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);
+    DosCreateEventSem(NULL, &cond->ack_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);
+    DosCloseEventSem(cond->ack_sem);
+
+    return 0;
+}
+
+static av_always_inline int pthread_cond_signal(pthread_cond_t *cond)
+{
+    if (!__atomic_cmpxchg32(&cond->wait_count, 0, 0)) {
+        DosPostEventSem(cond->event_sem);
+        DosWaitEventSem(cond->ack_sem, SEM_INDEFINITE_WAIT);
+    }
+
+    return 0;
+}
+
+static av_always_inline int pthread_cond_broadcast(pthread_cond_t *cond)
+{
+    while (!__atomic_cmpxchg32(&cond->wait_count, 0, 0))
+        pthread_cond_signal(cond);
+
+    return 0;
+}
+
+static av_always_inline int pthread_cond_timedwait(pthread_cond_t *cond,
+                                                   pthread_mutex_t *mutex,
+                                                   const struct timespec *abstime)
+{
+    int64_t abs_milli = abstime->tv_sec * 1000LL + abstime->tv_nsec / 1000000;
+    ULONG t = av_clip64(abs_milli - av_gettime() / 1000, 0, ULONG_MAX);
+
+    __atomic_increment(&cond->wait_count);
+
+    pthread_mutex_unlock(mutex);
+
+    APIRET ret = DosWaitEventSem(cond->event_sem, t);
+
+    __atomic_decrement(&cond->wait_count);
+
+    DosPostEventSem(cond->ack_sem);
+
+    pthread_mutex_lock(mutex);
+
+    return (ret == ERROR_TIMEOUT) ? ETIMEDOUT : 0;
+}
+
+static av_always_inline int pthread_cond_wait(pthread_cond_t *cond,
+                                              pthread_mutex_t *mutex)
+{
+    __atomic_increment(&cond->wait_count);
+
+    pthread_mutex_unlock(mutex);
+
+    DosWaitEventSem(cond->event_sem, SEM_INDEFINITE_WAIT);
+
+    __atomic_decrement(&cond->wait_count);
+
+    DosPostEventSem(cond->ack_sem);
+
+    pthread_mutex_lock(mutex);
+
+    return 0;
+}
+
+static av_always_inline int pthread_once(pthread_once_t *once_control,
+                                         void (*init_routine)(void))
+{
+    if (!once_control->done)
+    {
+        _fmutex_request(&once_control->mtx, 0);
+
+        if (!once_control->done)
+        {
+            init_routine();
+
+            once_control->done = 1;
+        }
+
+        _fmutex_release(&once_control->mtx);
+    }
+
+    return 0;
+}
+#endif /* COMPAT_OS2THREADS_H */

externals/ffmpeg/ffmpeg/compat/solaris/make_sunver.pl

@@ -0,0 +1,352 @@
+#!/usr/bin/env perl
+
+# make_sunver.pl
+#
+#   Copyright (C) 2010, 2011, 2012, 2013
+#   Free Software Foundation, Inc.
+#
+# This file 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 3 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; see the file COPYING.GPLv3.  If not see
+# <http://www.gnu.org/licenses/>.
+
+# This script takes at least two arguments, a GNU style version script and
+# a list of object and archive files, and generates a corresponding Sun
+# style version script as follows:
+#
+# Each glob pattern, C++ mangled pattern or literal in the input script is
+# matched against all global symbols in the input objects, emitting those
+# that matched (or nothing if no match was found).
+# A comment with the original pattern and its type is left in the output
+# file to make it easy to understand the matches.
+#
+# It uses elfdump when present (native), GNU readelf otherwise.
+# It depends on the GNU version of c++filt, since it must understand the
+# GNU mangling style.
+
+use FileHandle;
+use IPC::Open2;
+
+# Enforce C locale.
+$ENV{'LC_ALL'} = "C";
+$ENV{'LANG'} = "C";
+
+# Input version script, GNU style.
+my $symvers = shift;
+
+##########
+# Get all the symbols from the library, match them, and add them to a hash.
+
+my %sym_hash = ();
+
+# List of objects and archives to process.
+my @OBJECTS = ();
+
+# List of shared objects to omit from processing.
+my @SHAREDOBJS = ();
+
+# Filter out those input archives that have corresponding shared objects to
+# avoid adding all symbols matched in the archive to the output map.
+foreach $file (@ARGV) {
+    if (($so = $file) =~ s/\.a$/.so/ && -e $so) {
+    printf STDERR "omitted $file -> $so\n";
+    push (@SHAREDOBJS, $so);
+    } else {
+    push (@OBJECTS, $file);
+    }
+}
+
+# We need to detect and ignore hidden symbols.  Solaris nm can only detect
+# this in the harder to parse default output format, and GNU nm not at all,
+# so use elfdump -s in the native case and GNU readelf -s otherwise.
+# GNU objdump -t cannot be used since it produces a variable number of
+# columns.
+
+# The path to elfdump.
+my $elfdump = "/usr/ccs/bin/elfdump";
+
+if (-f $elfdump) {
+    open ELFDUMP,$elfdump.' -s '.(join ' ',@OBJECTS).'|' or die $!;
+    my $skip_arsym = 0;
+
+    while (<ELFDUMP>) {
+    chomp;
+
+    # Ignore empty lines.
+    if (/^$/) {
+        # End of archive symbol table, stop skipping.
+        $skip_arsym = 0 if $skip_arsym;
+        next;
+    }
+
+    # Keep skipping until end of archive symbol table.
+    next if ($skip_arsym);
+
+    # Ignore object name header for individual objects and archives.
+    next if (/:$/);
+
+    # Ignore table header lines.
+    next if (/^Symbol Table Section:/);
+    next if (/index.*value.*size/);
+
+    # Start of archive symbol table: start skipping.
+    if (/^Symbol Table: \(archive/) {
+        $skip_arsym = 1;
+        next;
+    }
+
+    # Split table.
+    (undef, undef, undef, undef, $bind, $oth, undef, $shndx, $name) = split;
+
+    # Error out for unknown input.
+    die "unknown input line:\n$_" unless defined($bind);
+
+    # Ignore local symbols.
+    next if ($bind eq "LOCL");
+    # Ignore hidden symbols.
+    next if ($oth eq "H");
+    # Ignore undefined symbols.
+    next if ($shndx eq "UNDEF");
+    # Error out for unhandled cases.
+    if ($bind !~ /^(GLOB|WEAK)/ or $oth ne "D") {
+        die "unhandled symbol:\n$_";
+    }
+
+    # Remember symbol.
+    $sym_hash{$name}++;
+    }
+    close ELFDUMP or die "$elfdump error";
+} else {
+    open READELF, 'readelf -s -W '.(join ' ',@OBJECTS).'|' or die $!;
+    # Process each symbol.
+    while (<READELF>) {
+    chomp;
+
+    # Ignore empty lines.
+    next if (/^$/);
+
+    # Ignore object name header.
+    next if (/^File: .*$/);
+
+    # Ignore table header lines.
+    next if (/^Symbol table.*contains.*:/);
+    next if (/Num:.*Value.*Size/);
+
+    # Split table.
+    (undef, undef, undef, undef, $bind, $vis, $ndx, $name) = split;
+
+    # Error out for unknown input.
+    die "unknown input line:\n$_" unless defined($bind);
+
+    # Ignore local symbols.
+    next if ($bind eq "LOCAL");
+    # Ignore hidden symbols.
+    next if ($vis eq "HIDDEN");
+    # Ignore undefined symbols.
+    next if ($ndx eq "UND");
+    # Error out for unhandled cases.
+    if ($bind !~ /^(GLOBAL|WEAK)/ or $vis ne "DEFAULT") {
+        die "unhandled symbol:\n$_";
+    }
+
+    # Remember symbol.
+    $sym_hash{$name}++;
+    }
+    close READELF or die "readelf error";
+}
+
+##########
+# The various types of glob patterns.
+#
+# A glob pattern that is to be applied to the demangled name: 'cxx'.
+# A glob patterns that applies directly to the name in the .o files: 'glob'.
+# This pattern is ignored; used for local variables (usually just '*'): 'ign'.
+
+# The type of the current pattern.
+my $glob = 'glob';
+
+# We're currently inside `extern "C++"', which Sun ld doesn't understand.
+my $in_extern = 0;
+
+# The c++filt command to use.  This *must* be GNU c++filt; the Sun Studio
+# c++filt doesn't handle the GNU mangling style.
+my $cxxfilt = $ENV{'CXXFILT'} || "c++filt";
+
+# The current version name.
+my $current_version = "";
+
+# Was there any attempt to match a symbol to this version?
+my $matches_attempted;
+
+# The number of versions which matched this symbol.
+my $matched_symbols;
+
+open F,$symvers or die $!;
+
+# Print information about generating this file
+print "# This file was generated by make_sunver.pl.  DO NOT EDIT!\n";
+print "# It was generated by:\n";
+printf "# %s %s %s\n", $0, $symvers, (join ' ',@ARGV);
+printf "# Omitted archives with corresponding shared libraries: %s\n",
+    (join ' ', @SHAREDOBJS) if $#SHAREDOBJS >= 0;
+print "#\n\n";
+
+print "\$mapfile_version 2\n";
+
+while (<F>) {
+    # Lines of the form '};'
+    if (/^([ \t]*)(\}[ \t]*;[ \t]*)$/) {
+    $glob = 'glob';
+    if ($in_extern) {
+        $in_extern--;
+        print "$1##$2\n";
+    } else {
+        print;
+    }
+    next;
+    }
+
+    # Lines of the form '} SOME_VERSION_NAME_1.0;'
+    if (/^[ \t]*\}[ \tA-Z0-9_.a-z]+;[ \t]*$/) {
+    $glob = 'glob';
+    # We tried to match symbols agains this version, but none matched.
+    # Emit dummy hidden symbol to avoid marking this version WEAK.
+    if ($matches_attempted && $matched_symbols == 0) {
+        print "  hidden:\n";
+        print "    .force_WEAK_off_$current_version = DATA S0x0 V0x0;\n";
+    }
+    print; next;
+    }
+
+    # Comment and blank lines
+    if (/^[ \t]*\#/) { print; next; }
+    if (/^[ \t]*$/) { print; next; }
+
+    # Lines of the form '{'
+    if (/^([ \t]*){$/) {
+    if ($in_extern) {
+        print "$1##{\n";
+    } else {
+        print;
+    }
+    next;
+    }
+
+    # Lines of the form 'SOME_VERSION_NAME_1.1 {'
+    if (/^([A-Z0-9_.]+)[ \t]+{$/) {
+    # Record version name.
+    $current_version = $1;
+    # Reset match attempts, #matched symbols for this version.
+    $matches_attempted = 0;
+    $matched_symbols = 0;
+    print "SYMBOL_VERSION $1 {\n";
+    next;
+    }
+
+    # Ignore 'global:'
+    if (/^[ \t]*global:$/) { print; next; }
+
+    # After 'local:', globs should be ignored, they won't be exported.
+    if (/^[ \t]*local:$/) {
+    $glob = 'ign';
+    print;
+    next;
+    }
+
+    # After 'extern "C++"', globs are C++ patterns
+    if (/^([ \t]*)(extern \"C\+\+\"[ \t]*)$/) {
+    $in_extern++;
+    $glob = 'cxx';
+    # Need to comment, Sun ld cannot handle this.
+    print "$1##$2\n"; next;
+    }
+
+    # Chomp newline now we're done with passing through the input file.
+    chomp;
+
+    # Catch globs.  Note that '{}' is not allowed in globs by this script,
+    # so only '*' and '[]' are available.
+    if (/^([ \t]*)([^ \t;{}#]+);?[ \t]*$/) {
+    my $ws = $1;
+    my $ptn = $2;
+    # Turn the glob into a regex by replacing '*' with '.*', '?' with '.'.
+    # Keep $ptn so we can still print the original form.
+    ($pattern = $ptn) =~ s/\*/\.\*/g;
+    $pattern =~ s/\?/\./g;
+
+    if ($glob eq 'ign') {
+        # We're in a local: * section; just continue.
+        print "$_\n";
+        next;
+    }
+
+    # Print the glob commented for human readers.
+    print "$ws##$ptn ($glob)\n";
+    # We tried to match a symbol to this version.
+    $matches_attempted++;
+
+    if ($glob eq 'glob') {
+        my %ptn_syms = ();
+
+        # Match ptn against symbols in %sym_hash.
+        foreach my $sym (keys %sym_hash) {
+        # Maybe it matches one of the patterns based on the symbol in
+        # the .o file.
+        $ptn_syms{$sym}++ if ($sym =~ /^$pattern$/);
+        }
+
+        foreach my $sym (sort keys(%ptn_syms)) {
+        $matched_symbols++;
+        print "$ws$sym;\n";
+        }
+    } elsif ($glob eq 'cxx') {
+        my %dem_syms = ();
+
+        # Verify that we're actually using GNU c++filt.  Other versions
+        # most likely cannot handle GNU style symbol mangling.
+        my $cxxout = `$cxxfilt --version 2>&1`;
+        $cxxout =~ m/GNU/ or die "$0 requires GNU c++filt to function";
+
+        # Talk to c++filt through a pair of file descriptors.
+        # Need to start a fresh instance per pattern, otherwise the
+        # process grows to 500+ MB.
+        my $pid = open2(*FILTIN, *FILTOUT, $cxxfilt) or die $!;
+
+        # Match ptn against symbols in %sym_hash.
+        foreach my $sym (keys %sym_hash) {
+        # No?  Well, maybe its demangled form matches one of those
+        # patterns.
+        printf FILTOUT "%s\n",$sym;
+        my $dem = <FILTIN>;
+        chomp $dem;
+        $dem_syms{$sym}++ if ($dem =~ /^$pattern$/);
+        }
+
+        close FILTOUT or die "c++filt error";
+        close FILTIN or die "c++filt error";
+        # Need to wait for the c++filt process to avoid lots of zombies.
+        waitpid $pid, 0;
+
+        foreach my $sym (sort keys(%dem_syms)) {
+        $matched_symbols++;
+        print "$ws$sym;\n";
+        }
+    } else {
+        # No?  Well, then ignore it.
+    }
+    next;
+    }
+    # Important sanity check.  This script can't handle lots of formats
+    # that GNU ld can, so be sure to error out if one is seen!
+    die "strange line `$_'";
+}
+close F;

externals/ffmpeg/ffmpeg/compat/strtod.c

@@ -0,0 +1,93 @@
+/*
+ * 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 const char *check_nan_suffix(const char *s)
+{
+    const 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)
+{
+    const 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, (char **)&end, 16);
+    } else {
+        res = strtod(nptr, (char **)&end);
+    }
+
+    if (endptr)
+        *endptr = (char *)end;
+
+    return res;
+}

externals/ffmpeg/ffmpeg/compat/va_copy.h

@@ -0,0 +1,34 @@
+/*
+ * 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
+ */
+
+#ifndef COMPAT_VA_COPY_H
+#define COMPAT_VA_COPY_H
+
+#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
+
+#endif /* COMPAT_VA_COPY_H */

externals/ffmpeg/ffmpeg/compat/w32dlfcn.h

@@ -0,0 +1,94 @@
+/*
+ * 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_W32DLFCN_H
+#define COMPAT_W32DLFCN_H
+
+#ifdef _WIN32
+#include <windows.h>
+#include "config.h"
+#if (_WIN32_WINNT < 0x0602) || HAVE_WINRT
+#include "libavutil/wchar_filename.h"
+#endif
+/**
+ * Safe function used to open dynamic libs. This attempts to improve program security
+ * by removing the current directory from the dll search path. Only dll's found in the
+ * executable or system directory are allowed to be loaded.
+ * @param name  The dynamic lib name.
+ * @return A handle to the opened lib.
+ */
+static inline HMODULE win32_dlopen(const char *name)
+{
+#if _WIN32_WINNT < 0x0602
+    // Need to check if KB2533623 is available
+    if (!GetProcAddress(GetModuleHandleW(L"kernel32.dll"), "SetDefaultDllDirectories")) {
+        HMODULE module = NULL;
+        wchar_t *path = NULL, *name_w = NULL;
+        DWORD pathlen;
+        if (utf8towchar(name, &name_w))
+            goto exit;
+        path = (wchar_t *)av_mallocz_array(MAX_PATH, sizeof(wchar_t));
+        // Try local directory first
+        pathlen = GetModuleFileNameW(NULL, path, MAX_PATH);
+        pathlen = wcsrchr(path, '\\') - path;
+        if (pathlen == 0 || pathlen + wcslen(name_w) + 2 > MAX_PATH)
+            goto exit;
+        path[pathlen] = '\\';
+        wcscpy(path + pathlen + 1, name_w);
+        module = LoadLibraryExW(path, NULL, LOAD_WITH_ALTERED_SEARCH_PATH);
+        if (module == NULL) {
+            // Next try System32 directory
+            pathlen = GetSystemDirectoryW(path, MAX_PATH);
+            if (pathlen == 0 || pathlen + wcslen(name_w) + 2 > MAX_PATH)
+                goto exit;
+            path[pathlen] = '\\';
+            wcscpy(path + pathlen + 1, name_w);
+            module = LoadLibraryExW(path, NULL, LOAD_WITH_ALTERED_SEARCH_PATH);
+        }
+exit:
+        av_free(path);
+        av_free(name_w);
+        return module;
+    }
+#endif
+#ifndef LOAD_LIBRARY_SEARCH_APPLICATION_DIR
+#   define LOAD_LIBRARY_SEARCH_APPLICATION_DIR 0x00000200
+#endif
+#ifndef LOAD_LIBRARY_SEARCH_SYSTEM32
+#   define LOAD_LIBRARY_SEARCH_SYSTEM32        0x00000800
+#endif
+#if HAVE_WINRT
+    wchar_t *name_w = NULL;
+    int ret;
+    if (utf8towchar(name, &name_w))
+        return NULL;
+    ret = LoadPackagedLibrary(name_w, 0);
+    av_free(name_w);
+    return ret;
+#else
+    return LoadLibraryExA(name, NULL, LOAD_LIBRARY_SEARCH_APPLICATION_DIR | LOAD_LIBRARY_SEARCH_SYSTEM32);
+#endif
+}
+#define dlopen(name, flags) win32_dlopen(name)
+#define dlclose FreeLibrary
+#define dlsym GetProcAddress
+#else
+#include <dlfcn.h>
+#endif
+
+#endif /* COMPAT_W32DLFCN_H */

externals/ffmpeg/ffmpeg/compat/w32pthreads.h

@@ -0,0 +1,191 @@
+/*
+ * 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 COMPAT_W32PTHREADS_H
+#define 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 <time.h>
+
+#include "libavutil/attributes.h"
+#include "libavutil/common.h"
+#include "libavutil/internal.h"
+#include "libavutil/mem.h"
+#include "libavutil/time.h"
+
+typedef struct pthread_t {
+    void *handle;
+    void *(*func)(void* arg);
+    void *arg;
+    void *ret;
+} pthread_t;
+
+/* use light weight mutex/condition variable API for Windows Vista and later */
+typedef SRWLOCK pthread_mutex_t;
+typedef CONDITION_VARIABLE pthread_cond_t;
+
+#define PTHREAD_MUTEX_INITIALIZER SRWLOCK_INIT
+#define PTHREAD_COND_INITIALIZER CONDITION_VARIABLE_INIT
+
+#define InitializeCriticalSection(x) InitializeCriticalSectionEx(x, 0, 0)
+#define WaitForSingleObject(a, b) WaitForSingleObjectEx(a, b, FALSE)
+
+#define PTHREAD_CANCEL_ENABLE 1
+#define PTHREAD_CANCEL_DISABLE 0
+
+static av_unused unsigned __stdcall attribute_align_arg win32thread_worker(void *arg)
+{
+    pthread_t *h = (pthread_t*)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;
+#if HAVE_WINRT
+    thread->handle = (void*)CreateThread(NULL, 0, win32thread_worker, thread,
+                                           0, NULL);
+#else
+    thread->handle = (void*)_beginthreadex(NULL, 0, win32thread_worker, thread,
+                                           0, NULL);
+#endif
+    return !thread->handle;
+}
+
+static av_unused int pthread_join(pthread_t thread, void **value_ptr)
+{
+    DWORD ret = WaitForSingleObject(thread.handle, INFINITE);
+    if (ret != WAIT_OBJECT_0) {
+        if (ret == WAIT_ABANDONED)
+            return EINVAL;
+        else
+            return EDEADLK;
+    }
+    if (value_ptr)
+        *value_ptr = thread.ret;
+    CloseHandle(thread.handle);
+    return 0;
+}
+
+static inline int pthread_mutex_init(pthread_mutex_t *m, void* attr)
+{
+    InitializeSRWLock(m);
+    return 0;
+}
+static inline int pthread_mutex_destroy(pthread_mutex_t *m)
+{
+    /* Unlocked SWR locks use no resources */
+    return 0;
+}
+static inline int pthread_mutex_lock(pthread_mutex_t *m)
+{
+    AcquireSRWLockExclusive(m);
+    return 0;
+}
+static inline int pthread_mutex_unlock(pthread_mutex_t *m)
+{
+    ReleaseSRWLockExclusive(m);
+    return 0;
+}
+
+typedef INIT_ONCE pthread_once_t;
+#define PTHREAD_ONCE_INIT INIT_ONCE_STATIC_INIT
+
+static av_unused int pthread_once(pthread_once_t *once_control, void (*init_routine)(void))
+{
+    BOOL pending = FALSE;
+    InitOnceBeginInitialize(once_control, 0, &pending, NULL);
+    if (pending)
+        init_routine();
+    InitOnceComplete(once_control, 0, NULL);
+    return 0;
+}
+
+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 int pthread_cond_destroy(pthread_cond_t *cond)
+{
+    return 0;
+}
+
+static inline int pthread_cond_broadcast(pthread_cond_t *cond)
+{
+    WakeAllConditionVariable(cond);
+    return 0;
+}
+
+static inline int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
+{
+    SleepConditionVariableSRW(cond, mutex, INFINITE, 0);
+    return 0;
+}
+
+static inline int pthread_cond_timedwait(pthread_cond_t *cond, pthread_mutex_t *mutex,
+                                         const struct timespec *abstime)
+{
+    int64_t abs_milli = abstime->tv_sec * 1000LL + abstime->tv_nsec / 1000000;
+    DWORD t = av_clip64(abs_milli - av_gettime() / 1000, 0, UINT32_MAX);
+
+    if (!SleepConditionVariableSRW(cond, mutex, t, 0)) {
+        DWORD err = GetLastError();
+        if (err == ERROR_TIMEOUT)
+            return ETIMEDOUT;
+        else
+            return EINVAL;
+    }
+    return 0;
+}
+
+static inline int pthread_cond_signal(pthread_cond_t *cond)
+{
+    WakeConditionVariable(cond);
+    return 0;
+}
+
+static inline int pthread_setcancelstate(int state, int *oldstate)
+{
+    return 0;
+}
+
+#endif /* COMPAT_W32PTHREADS_H */

externals/ffmpeg/ffmpeg/compat/windows/makedef

@@ -0,0 +1,129 @@
+#!/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 -- $libname' EXIT
+
+if [ -n "$AR" ]; then
+    $AR rcs ${libname} $@ >/dev/null
+else
+    lib.exe -out:${libname} $@ >/dev/null
+fi
+if [ $? != 0 ]; then
+    echo "Could not create temporary library." >&2
+    exit 1
+fi
+
+IFS='
+'
+
+prefix="$EXTERN_PREFIX"
+
+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
+
+if [ -n "$NM" ]; then
+    # Use eval, since NM="nm -g"
+    dump=$(eval "$NM --defined-only -g ${libname}" |
+              grep -v : |
+              grep -v ^$ |
+              cut -d' ' -f3 |
+              sed -e "s/^${prefix}//")
+else
+    dump=$(dumpbin.exe -linkermember:1 ${libname} |
+              sed -e '/public symbols/,$!d' -e '/^ \{1,\}Summary/,$d' -e "s/ \{1,\}${prefix}/ /" -e 's/ \{1,\}/ /g' |
+              tail -n +2 |
+              cut -d' ' -f3)
+fi
+
+rm ${libname}
+
+IFS=';'
+list=""
+for exp in ${regex}; do
+    list="${list}"'
+'$(echo "${dump}" |
+          grep "^${exp}" |
+          sed -e 's/^/    /')
+done
+
+echo "EXPORTS"
+echo "${list}" | sort | uniq | tail -n +2

externals/ffmpeg/ffmpeg/compat/windows/mslink

@@ -0,0 +1,9 @@
+#!/bin/sh
+
+LINK_EXE_PATH=$(dirname "$(command -v cl)")/link
+if [ -x "$LINK_EXE_PATH" ]; then
+    "$LINK_EXE_PATH" $@
+else
+    link.exe $@
+fi
+exit $?

externals/ffmpeg/ffmpeg/doc/.gitignore

@@ -0,0 +1,9 @@
+/*.1
+/*.3
+/*.html
+/*.pod
+/config.texi
+/avoptions_codec.texi
+/avoptions_format.texi
+/fate.txt
+/print_options

externals/ffmpeg/ffmpeg/doc/Makefile

@@ -0,0 +1,154 @@
+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) $(LIBRARIES-yes:%=doc/%.html) \
+              doc/developer.html                                        \
+              doc/faq.html                                              \
+              doc/fate.html                                             \
+              doc/general.html                                          \
+              doc/git-howto.html                                        \
+              doc/mailing-list-faq.html                                 \
+              doc/nut.html                                              \
+              doc/platform.html                                         \
+
+TXTPAGES    = doc/fate.txt                                              \
+
+
+DOCS-$(CONFIG_HTMLPAGES) += $(HTMLPAGES)
+DOCS-$(CONFIG_PODPAGES)  += $(PODPAGES)
+DOCS-$(CONFIG_MANPAGES)  += $(MANPAGES)
+DOCS-$(CONFIG_TXTPAGES)  += $(TXTPAGES)
+DOCS = $(DOCS-yes)
+
+all-$(CONFIG_DOC): doc
+
+doc: documentation
+
+apidoc: doc/doxy/html
+documentation: $(DOCS)
+
+TEXIDEP = 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/
+
+DOXY_INPUT      = $(INSTHEADERS)
+DOXY_INPUT_DEPS = $(addprefix $(SRC_PATH)/, $(DOXY_INPUT)) ffbuild/config.mak
+
+doc/doxy/html: TAG = DOXY
+doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(SRC_PATH)/doc/doxy-wrapper.sh $(DOXY_INPUT_DEPS)
+	$(M)OUT_DIR=$$PWD/doc/doxy; cd $(SRC_PATH); ./doc/doxy-wrapper.sh $$OUT_DIR $< $(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
+
+docclean::
+	$(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

externals/ffmpeg/ffmpeg/doc/authors.texi

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

externals/ffmpeg/ffmpeg/doc/bitstream_filters.texi

@@ -0,0 +1,799 @@
+@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 an MPEG-4 Audio Specific Configuration
+bitstream.
+
+This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4
+ADTS header and removes the ADTS header.
+
+This filter is required for example when copying an AAC stream from a
+raw ADTS AAC or an MPEG-TS container to MP4A-LATM, to an FLV file, or
+to MOV/MP4 files and related formats such as 3GP or M4A. Please note
+that it is auto-inserted for MP4A-LATM and MOV/MP4 and related formats.
+
+@section av1_metadata
+
+Modify metadata embedded in an AV1 stream.
+
+@table @option
+@item td
+Insert or remove temporal delimiter OBUs in all temporal units of the
+stream.
+
+@table @samp
+@item insert
+Insert a TD at the beginning of every TU which does not already have one.
+@item remove
+Remove the TD from the beginning of every TU which has one.
+@end table
+
+@item color_primaries
+@item transfer_characteristics
+@item matrix_coefficients
+Set the color description fields in the stream (see AV1 section 6.4.2).
+
+@item color_range
+Set the color range in the stream (see AV1 section 6.4.2; note that
+this cannot be set for streams using BT.709 primaries, sRGB transfer
+characteristic and identity (RGB) matrix coefficients).
+@table @samp
+@item tv
+Limited range.
+@item pc
+Full range.
+@end table
+
+@item chroma_sample_position
+Set the chroma sample location in the stream (see AV1 section 6.4.2).
+This can only be set for 4:2:0 streams.
+
+@table @samp
+@item vertical
+Left position (matching the default in MPEG-2 and H.264).
+@item colocated
+Top-left position.
+@end table
+
+@item tick_rate
+Set the tick rate (@emph{num_units_in_display_tick / time_scale}) in
+the timing info in the sequence header.
+@item num_ticks_per_picture
+Set the number of ticks in each picture, to indicate that the stream
+has a fixed framerate.  Ignored if @option{tick_rate} is not also set.
+
+@item delete_padding
+Deletes Padding OBUs.
+
+@end table
+
+@section chomp
+
+Remove zero padding at the end of a packet.
+
+@section dca_core
+
+Extract the core from a DCA/DTS stream, dropping extensions such as
+DTS-HD.
+
+@section dump_extra
+
+Add extradata to the beginning of the filtered packets except when
+said packets already exactly begin with the extradata that is intended
+to be added.
+
+@table @option
+@item freq
+The additional argument specifies which packets should be filtered.
+It accepts the values:
+@table @samp
+@item k
+@item keyframe
+add extradata to all key packets
+
+@item e
+@item all
+add extradata to all packets
+@end table
+@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 eac3_core
+
+Extract the core from a E-AC-3 stream, dropping extra channels.
+
+@section extract_extradata
+
+Extract the in-band extradata.
+
+Certain codecs allow the long-term headers (e.g. MPEG-2 sequence headers,
+or H.264/HEVC (VPS/)SPS/PPS) to be transmitted either "in-band" (i.e. as a part
+of the bitstream containing the coded frames) or "out of band" (e.g. on the
+container level). This latter form is called "extradata" in FFmpeg terminology.
+
+This bitstream filter detects the in-band headers and makes them available as
+extradata.
+
+@table @option
+@item remove
+When this option is enabled, the long-term headers are removed from the
+bitstream after extraction.
+@end table
+
+@section filter_units
+
+Remove units with types in or not in a given set from the stream.
+
+@table @option
+@item pass_types
+List of unit types or ranges of unit types to pass through while removing
+all others.  This is specified as a '|'-separated list of unit type values
+or ranges of values with '-'.
+
+@item remove_types
+Identical to @option{pass_types}, except the units in the given set
+removed and all others passed through.
+@end table
+
+Extradata is unchanged by this transformation, but note that if the stream
+contains inline parameter sets then the output may be unusable if they are
+removed.
+
+For example, to remove all non-VCL NAL units from an H.264 stream:
+@example
+ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=pass_types=1-5' OUTPUT
+@end example
+
+To remove all AUDs, SEI and filler from an H.265 stream:
+@example
+ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=remove_types=35|38-40' OUTPUT
+@end example
+
+@section hapqa_extract
+
+Extract Rgb or Alpha part of an HAPQA file, without recompression, in order to create an HAPQ or an HAPAlphaOnly file.
+
+@table @option
+@item texture
+Specifies the texture to keep.
+
+@table @option
+@item color
+@item alpha
+@end table
+
+@end table
+
+Convert HAPQA to HAPQ
+@example
+ffmpeg -i hapqa_inputfile.mov -c copy -bsf:v hapqa_extract=texture=color -tag:v HapY -metadata:s:v:0 encoder="HAPQ" hapq_file.mov
+@end example
+
+Convert HAPQA to HAPAlphaOnly
+@example
+ffmpeg -i hapqa_inputfile.mov -c copy -bsf:v hapqa_extract=texture=alpha -tag:v HapA -metadata:s:v:0 encoder="HAPAlpha Only" hapalphaonly_file.mov
+@end example
+
+@section h264_metadata
+
+Modify metadata embedded in an H.264 stream.
+
+@table @option
+@item aud
+Insert or remove AUD NAL units in all access units of the stream.
+
+@table @samp
+@item insert
+@item remove
+@end table
+
+@item sample_aspect_ratio
+Set the sample aspect ratio of the stream in the VUI parameters.
+
+@item overscan_appropriate_flag
+Set whether the stream is suitable for display using overscan
+or not (see H.264 section E.2.1).
+
+@item video_format
+@item video_full_range_flag
+Set the video format in the stream (see H.264 section E.2.1 and
+table E-2).
+
+@item colour_primaries
+@item transfer_characteristics
+@item matrix_coefficients
+Set the colour description in the stream (see H.264 section E.2.1
+and tables E-3, E-4 and E-5).
+
+@item chroma_sample_loc_type
+Set the chroma sample location in the stream (see H.264 section
+E.2.1 and figure E-1).
+
+@item tick_rate
+Set the tick rate (num_units_in_tick / time_scale) in the VUI
+parameters.  This is the smallest time unit representable in the
+stream, and in many cases represents the field rate of the stream
+(double the frame rate).
+@item fixed_frame_rate_flag
+Set whether the stream has fixed framerate - typically this indicates
+that the framerate is exactly half the tick rate, but the exact
+meaning is dependent on interlacing and the picture structure (see
+H.264 section E.2.1 and table E-6).
+
+@item crop_left
+@item crop_right
+@item crop_top
+@item crop_bottom
+Set the frame cropping offsets in the SPS.  These values will replace
+the current ones if the stream is already cropped.
+
+These fields are set in pixels.  Note that some sizes may not be
+representable if the chroma is subsampled or the stream is interlaced
+(see H.264 section 7.4.2.1.1).
+
+@item sei_user_data
+Insert a string as SEI unregistered user data.  The argument must
+be of the form @emph{UUID+string}, where the UUID is as hex digits
+possibly separated by hyphens, and the string can be anything.
+
+For example, @samp{086f3693-b7b3-4f2c-9653-21492feee5b8+hello} will
+insert the string ``hello'' associated with the given UUID.
+
+@item delete_filler
+Deletes both filler NAL units and filler SEI messages.
+
+@item level
+Set the level in the SPS.  Refer to H.264 section A.3 and tables A-1
+to A-5.
+
+The argument must be the name of a level (for example, @samp{4.2}), a
+level_idc value (for example, @samp{42}), or the special name @samp{auto}
+indicating that the filter should attempt to guess the level from the
+input stream properties.
+
+@end table
+
+@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 (muxer @code{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
+
+Please note that this filter is auto-inserted for MPEG-TS (muxer
+@code{mpegts}) and raw H.264 (muxer @code{h264}) output formats.
+
+@section h264_redundant_pps
+
+This applies a specific fixup to some Blu-ray streams which contain
+redundant PPSs modifying irrelevant parameters of the stream which
+confuse other transformations which require correct extradata.
+
+A new single global PPS is created, and all of the redundant PPSs
+within the stream are removed.
+
+@section hevc_metadata
+
+Modify metadata embedded in an HEVC stream.
+
+@table @option
+@item aud
+Insert or remove AUD NAL units in all access units of the stream.
+
+@table @samp
+@item insert
+@item remove
+@end table
+
+@item sample_aspect_ratio
+Set the sample aspect ratio in the stream in the VUI parameters.
+
+@item video_format
+@item video_full_range_flag
+Set the video format in the stream (see H.265 section E.3.1 and
+table E.2).
+
+@item colour_primaries
+@item transfer_characteristics
+@item matrix_coefficients
+Set the colour description in the stream (see H.265 section E.3.1
+and tables E.3, E.4 and E.5).
+
+@item chroma_sample_loc_type
+Set the chroma sample location in the stream (see H.265 section
+E.3.1 and figure E.1).
+
+@item tick_rate
+Set the tick rate in the VPS and VUI parameters (num_units_in_tick /
+time_scale).  Combined with @option{num_ticks_poc_diff_one}, this can
+set a constant framerate in the stream.  Note that it is likely to be
+overridden by container parameters when the stream is in a container.
+
+@item num_ticks_poc_diff_one
+Set poc_proportional_to_timing_flag in VPS and VUI and use this value
+to set num_ticks_poc_diff_one_minus1 (see H.265 sections 7.4.3.1 and
+E.3.1).  Ignored if @option{tick_rate} is not also set.
+
+@item crop_left
+@item crop_right
+@item crop_top
+@item crop_bottom
+Set the conformance window cropping offsets in the SPS.  These values
+will replace the current ones if the stream is already cropped.
+
+These fields are set in pixels.  Note that some sizes may not be
+representable if the chroma is subsampled (H.265 section 7.4.3.2.1).
+
+@item level
+Set the level in the VPS and SPS.  See H.265 section A.4 and tables
+A.6 and A.7.
+
+The argument must be the name of a level (for example, @samp{5.1}), a
+@emph{general_level_idc} value (for example, @samp{153} for level 5.1),
+or the special name @samp{auto} indicating that the filter should
+attempt to guess the level from the input stream properties.
+
+@end table
+
+@section hevc_mp4toannexb
+
+Convert an HEVC/H.265 bitstream from length prefixed mode to start code
+prefixed mode (as defined in the Annex B of the ITU-T H.265
+specification).
+
+This is required by some streaming formats, typically the MPEG-2
+transport stream format (muxer @code{mpegts}).
+
+For example to remux an MP4 file containing an HEVC stream to mpegts
+format with @command{ffmpeg}, you can use the command:
+
+@example
+ffmpeg -i INPUT.mp4 -codec copy -bsf:v hevc_mp4toannexb OUTPUT.ts
+@end example
+
+Please note that this filter is auto-inserted for MPEG-TS (muxer
+@code{mpegts}) and raw HEVC/H.265 (muxer @code{h265} or
+@code{hevc}) output formats.
+
+@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 mjpegadump
+
+Add an MJPEG A header to the bitstream, to enable decoding by
+Quicktime.
+
+@anchor{mov2textsub}
+@section mov2textsub
+
+Extract a representable text file from MOV subtitles, stripping the
+metadata header from each subtitle packet.
+
+See also the @ref{text2movsub} filter.
+
+@section mp3decomp
+
+Decompress non-standard compressed MP3 audio headers.
+
+@section mpeg2_metadata
+
+Modify metadata embedded in an MPEG-2 stream.
+
+@table @option
+@item display_aspect_ratio
+Set the display aspect ratio in the stream.
+
+The following fixed values are supported:
+@table @option
+@item 4/3
+@item 16/9
+@item 221/100
+@end table
+Any other value will result in square pixels being signalled instead
+(see H.262 section 6.3.3 and table 6-3).
+
+@item frame_rate
+Set the frame rate in the stream.  This is constructed from a table
+of known values combined with a small multiplier and divisor - if
+the supplied value is not exactly representable, the nearest
+representable value will be used instead (see H.262 section 6.3.3
+and table 6-4).
+
+@item video_format
+Set the video format in the stream (see H.262 section 6.3.6 and
+table 6-6).
+
+@item colour_primaries
+@item transfer_characteristics
+@item matrix_coefficients
+Set the colour description in the stream (see H.262 section 6.3.6
+and tables 6-7, 6-8 and 6-9).
+
+@end table
+
+@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 or simply drops them without damaging the
+container. Can be used for fuzzing or testing error resilience/concealment.
+
+Parameters:
+@table @option
+@item amount
+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.
+@item dropamount
+A numeral string, whose value is related to how often packets will be dropped.
+Therefore, values below or equal to 0 are forbidden, and the lower the more
+frequent packets will be dropped, with 1 meaning every packet is dropped.
+@end table
+
+The following example applies the modification to every byte but does not drop
+any packets.
+@example
+ffmpeg -i INPUT -c copy -bsf noise[=1] output.mkv
+@end example
+
+@section null
+This bitstream filter passes the packets through unchanged.
+
+@section pcm_rechunk
+
+Repacketize PCM audio to a fixed number of samples per packet or a fixed packet
+rate per second. This is similar to the @ref{asetnsamples,,asetnsamples audio
+filter,ffmpeg-filters} but works on audio packets instead of audio frames.
+
+@table @option
+@item nb_out_samples, n
+Set the number of samples per each output audio packet. The number is intended
+as the number of samples @emph{per each channel}. Default value is 1024.
+
+@item pad, p
+If set to 1, the filter will pad the last audio packet with silence, so that it
+will contain the same number of samples (or roughly the same number of samples,
+see @option{frame_rate}) as the previous ones. Default value is 1.
+
+@item frame_rate, r
+This option makes the filter output a fixed number of packets per second instead
+of a fixed number of samples per packet. If the audio sample rate is not
+divisible by the frame rate then the number of samples will not be constant but
+will vary slightly so that each packet will start as close to the frame
+boundary as possible. Using this option has precedence over @option{nb_out_samples}.
+@end table
+
+You can generate the well known 1602-1601-1602-1601-1602 pattern of 48kHz audio
+for NTSC frame rate using the @option{frame_rate} option.
+@example
+ffmpeg -f lavfi -i sine=r=48000:d=1 -c pcm_s16le -bsf pcm_rechunk=r=30000/1001 -f framecrc -
+@end example
+
+@section prores_metadata
+
+Modify color property metadata embedded in prores stream.
+
+@table @option
+@item color_primaries
+Set the color primaries.
+Available values are:
+
+@table @samp
+@item auto
+Keep the same color primaries property (default).
+
+@item unknown
+@item bt709
+@item bt470bg
+BT601 625
+
+@item smpte170m
+BT601 525
+
+@item bt2020
+@item smpte431
+DCI P3
+
+@item smpte432
+P3 D65
+
+@end table
+
+@item transfer_characteristics
+Set the color transfer.
+Available values are:
+
+@table @samp
+@item auto
+Keep the same transfer characteristics property (default).
+
+@item unknown
+@item bt709
+BT 601, BT 709, BT 2020
+@item smpte2084
+SMPTE ST 2084
+@item arib-std-b67
+ARIB STD-B67
+@end table
+
+
+@item matrix_coefficients
+Set the matrix coefficient.
+Available values are:
+
+@table @samp
+@item auto
+Keep the same colorspace property (default).
+
+@item unknown
+@item bt709
+@item smpte170m
+BT 601
+
+@item bt2020nc
+@end table
+@end table
+
+Set Rec709 colorspace for each frame of the file
+@example
+ffmpeg -i INPUT -c copy -bsf:v prores_metadata=color_primaries=bt709:color_trc=bt709:colorspace=bt709 output.mov
+@end example
+
+Set Hybrid Log-Gamma parameters for each frame of the file
+@example
+ffmpeg -i INPUT -c copy -bsf:v prores_metadata=color_primaries=bt2020:color_trc=arib-std-b67:colorspace=bt2020nc output.mov
+@end example
+
+@section remove_extra
+
+Remove extradata from packets.
+
+It accepts the following parameter:
+@table @option
+@item freq
+Set which frame types to remove extradata from.
+
+@table @samp
+@item k
+Remove extradata from non-keyframes only.
+
+@item keyframe
+Remove extradata from keyframes only.
+
+@item e, all
+Remove extradata from all frames.
+
+@end table
+@end table
+
+@section setts
+Set PTS and DTS in packets.
+
+It accepts the following parameters:
+@table @option
+@item ts
+@item pts
+@item dts
+Set expressions for PTS, DTS or both.
+@end table
+
+The expressions are evaluated through the eval API and can contain the following
+constants:
+
+@table @option
+@item N
+The count of the input packet. Starting from 0.
+
+@item TS
+The demux timestamp in input in case of @code{ts} or @code{dts} option or presentation
+timestamp in case of @code{pts} option.
+
+@item POS
+The original position in the file of the packet, or undefined if undefined
+for the current packet
+
+@item DTS
+The demux timestamp in input.
+
+@item PTS
+The presentation timestamp in input.
+
+@item STARTDTS
+The DTS of the first packet.
+
+@item STARTPTS
+The PTS of the first packet.
+
+@item PREV_INDTS
+The previous input DTS.
+
+@item PREV_INPTS
+The previous input PTS.
+
+@item PREV_OUTDTS
+The previous output DTS.
+
+@item PREV_OUTPTS
+The previous output PTS.
+
+@item TB
+The timebase of stream packet belongs.
+
+@item SR
+The sample rate of stream packet belongs.
+@end table
+
+@anchor{text2movsub}
+@section text2movsub
+
+Convert text subtitles to MOV subtitles (as used by the @code{mov_text}
+codec) with metadata headers.
+
+See also the @ref{mov2textsub} filter.
+
+@section trace_headers
+
+Log trace output containing all syntax elements in the coded stream
+headers (everything above the level of individual coded blocks).
+This can be useful for debugging low-level stream issues.
+
+Supports AV1, H.264, H.265, (M)JPEG, MPEG-2 and VP9, but depending
+on the build only a subset of these may be available.
+
+@section truehd_core
+
+Extract the core from a TrueHD stream, dropping ATMOS data.
+
+@section vp9_metadata
+
+Modify metadata embedded in a VP9 stream.
+
+@table @option
+@item color_space
+Set the color space value in the frame header.  Note that any frame
+set to RGB will be implicitly set to PC range and that RGB is
+incompatible with profiles 0 and 2.
+@table @samp
+@item unknown
+@item bt601
+@item bt709
+@item smpte170
+@item smpte240
+@item bt2020
+@item rgb
+@end table
+
+@item color_range
+Set the color range value in the frame header.  Note that any value
+imposed by the color space will take precedence over this value.
+@table @samp
+@item tv
+@item pc
+@end table
+@end table
+
+@section vp9_superframe
+
+Merge VP9 invisible (alt-ref) frames back into VP9 superframes. This
+fixes merging of split/segmented VP9 streams where the alt-ref frame
+was split from its visible counterpart.
+
+@section vp9_superframe_split
+
+Split VP9 superframes into single frames.
+
+@section vp9_raw_reorder
+
+Given a VP9 stream with correct timestamps but possibly out of order,
+insert additional show-existing-frame packets to correct the ordering.
+
+@c man end BITSTREAM FILTERS

externals/ffmpeg/ffmpeg/doc/build_system.txt

@@ -0,0 +1,66 @@
+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 in developing 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 that you must have installed it.
+
+fate-list
+    List all fate/regression test targets.
+
+install
+    Install headers, libraries and programs.
+
+examples
+    Build all examples located in doc/examples.
+
+checkheaders
+    Check headers dependencies.
+
+alltools
+    Build all tools in tools directory.
+
+config
+    Reconfigure the project with the current configuration.
+
+tools/target_dec_<decoder>_fuzzer
+    Build fuzzer to fuzz the specified decoder.
+
+tools/target_bsf_<filter>_fuzzer
+    Build fuzzer to fuzz the specified bitstream filter.
+
+Useful standard make commands:
+make -t <target>
+    Touch all files that otherwise would be built, this is useful to reduce
+    unneeded rebuilding when changing headers, but note that 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 that it will still not run all reg tests.
+

externals/ffmpeg/ffmpeg/doc/decoders.texi

@@ -0,0 +1,398 @@
+@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 av1
+
+AOMedia Video 1 (AV1) decoder.
+
+@subsection Options
+
+@table @option
+
+@item operating_point
+Select an operating point of a scalable AV1 bitstream (0 - 31). Default is 0.
+
+@end table
+
+@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
+
+@section libdav1d
+
+dav1d AV1 decoder.
+
+libdav1d allows libavcodec to decode the AOMedia Video 1 (AV1) codec.
+Requires the presence of the libdav1d headers and library during configuration.
+You need to explicitly configure the build with @code{--enable-libdav1d}.
+
+@subsection Options
+
+The following options are supported by the libdav1d wrapper.
+
+@table @option
+
+@item framethreads
+Set amount of frame threads to use during decoding. The default value is 0 (autodetect).
+
+@item tilethreads
+Set amount of tile threads to use during decoding. The default value is 0 (autodetect).
+
+@item filmgrain
+Apply film grain to the decoded video if present in the bitstream. Defaults to the
+internal default of the library.
+
+@item oppoint
+Select an operating point of a scalable AV1 bitstream (0 - 31). Defaults to the
+internal default of the library.
+
+@item alllayers
+Output all spatial layers of a scalable AV1 bitstream. The default value is false.
+
+@end table
+
+@section libdavs2
+
+AVS2-P2/IEEE1857.4 video decoder wrapper.
+
+This decoder allows libavcodec to decode AVS2 streams with davs2 library.
+
+@c man end VIDEO DECODERS
+
+@section libuavs3d
+
+AVS3-P2/IEEE1857.10 video decoder.
+
+libuavs3d allows libavcodec to decode AVS3 streams.
+Requires the presence of the libuavs3d headers and library during configuration.
+You need to explicitly configure the build with @code{--enable-libuavs3d}.
+
+@subsection Options
+
+The following option is supported by the libuavs3d wrapper.
+
+@table @option
+
+@item frame_threads
+Set amount of frame threads to use during decoding. The default value is 0 (autodetect).
+
+@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. The default value is 1.
+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 synthesizer.
+
+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 libaribb24
+
+ARIB STD-B24 caption decoder.
+
+Implements profiles A and C of the ARIB STD-B24 standard.
+
+@subsection libaribb24 Decoder Options
+
+@table @option
+
+@item -aribb24-base-path @var{path}
+Sets the base path for the libaribb24 library. This is utilized for reading of
+configuration files (for custom unicode conversions), and for dumping of
+non-text symbols as images under that location.
+
+Unset by default.
+
+@item -aribb24-skip-ruby-text @var{boolean}
+Tells the decoder wrapper to skip text blocks that contain half-height ruby
+text.
+
+Enabled by default.
+
+@end table
+
+@section dvbsub
+
+@subsection Options
+
+@table @option
+@item compute_clut
+@table @option
+@item -1
+Compute clut if no matching CLUT is in the stream.
+@item 0
+Never compute CLUT
+@item 1
+Always compute CLUT and override the one provided in the stream.
+@end table
+@item dvb_substream
+Selects the dvb substream, or all substreams if -1 which is default.
+
+@end table
+
+@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 commas, 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. Pages that do not match the specified
+list are dropped. You may use the special @code{*} string to match all pages,
+or @code{subtitle} to match all subtitle pages.
+Default value is *.
+@item txt_default_region
+Set default character set used for decoding, a value between 0 and 87 (see
+ETS 300 706, Section 15, Table 32). Default value is -1, which does not
+override the libzvbi default. This option is needed for some legacy level 1.0
+transmissions which cannot signal the proper charset.
+@item txt_chop_top
+Discards the top teletext line. Default value is 1.
+@item txt_format
+Specifies the format of the decoded subtitles.
+@table @option
+@item bitmap
+The default format, you should use this for teletext pages, because certain
+graphics and colors cannot be expressed in simple text or even ASS.
+@item text
+Simple text based output without formatting.
+@item ass
+Formatted ASS output, subtitle pages and teletext pages are returned in
+different styles, subtitle pages are stripped down to text, but an effort is
+made to keep the text alignment and the formatting.
+@end table
+@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 characters.
+Default value is 1.
+@item txt_duration
+Sets the display duration of the decoded teletext pages or subtitles in
+milliseconds. Default value is -1 which means infinity or until the next
+subtitle event comes.
+@item txt_transparent
+Force transparent background of the generated teletext bitmaps. Default value
+is 0 which means an opaque background.
+@item txt_opacity
+Sets the opacity (0-255) of the teletext background. If
+@option{txt_transparent} is not set, it only affects characters between a start
+box and an end box, typically subtitles. Default value is 0 if
+@option{txt_transparent} is set, 255 otherwise.
+
+@end table
+
+@c man end SUBTILES DECODERS

externals/ffmpeg/ffmpeg/doc/default.css

@@ -0,0 +1,165 @@
+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;
+}
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+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;
+}
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+h3 {
+    color: #313131;
+    font-size: 0.9em;
+    margin-bottom: -6px;
+    background-color: #BBF3BB;
+}
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+    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;
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+#repos th {
+    background-color: #7BB37B;
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+    border: solid 1px #6A996A;
+}

externals/ffmpeg/ffmpeg/doc/demuxers.texi

@@ -0,0 +1,835 @@
+@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{-demuxers} of the ff* tools will display the list of
+enabled demuxers. Use @code{-formats} to view a combined list of
+enabled demuxers and muxers.
+
+The description of some of the currently available demuxers follows.
+
+@section aa
+
+Audible Format 2, 3, and 4 demuxer.
+
+This demuxer is used to demux Audible Format 2, 3, and 4 (.aa) files.
+
+@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 packets 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 -1.
+
+To make FFmpeg recognize the format automatically, this directive must
+appear 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{inpoint @var{timestamp}}
+In point of the file. When the demuxer opens the file it instantly seeks to the
+specified timestamp. Seeking is done so that all streams can be presented
+successfully at In point.
+
+This directive works best with intra frame codecs, because for non-intra frame
+ones you will usually get extra packets before the actual In point and the
+decoded content will most likely contain frames before In point too.
+
+For each file, packets before the file In point will have timestamps less than
+the calculated start timestamp of the file (negative in case of the first
+file), and the duration of the files (if not specified by the @code{duration}
+directive) will be reduced based on their specified In point.
+
+Because of potential packets before the specified In point, packet timestamps
+may overlap between two concatenated files.
+
+@item @code{outpoint @var{timestamp}}
+Out point of the file. When the demuxer reaches the specified decoding
+timestamp in any of the streams, it handles it as an end of file condition and
+skips the current and all the remaining packets from all streams.
+
+Out point is exclusive, which means that the demuxer will not output packets
+with a decoding timestamp greater or equal to Out point.
+
+This directive works best with intra frame codecs and formats where all streams
+are tightly interleaved. For non-intra frame codecs you will usually get
+additional packets with presentation timestamp after Out point therefore the
+decoded content will most likely contain frames after Out point too. If your
+streams are not tightly interleaved you may not get all the packets from all
+streams before Out point and you may only will be able to decode the earliest
+stream until Out point.
+
+The duration of the files (if not specified by the @code{duration}
+directive) will be reduced based on their specified Out point.
+
+@item @code{file_packet_metadata @var{key=value}}
+Metadata of the packets of the file. The specified metadata will be set for
+each file packet. You can specify this directive multiple times to add multiple
+metadata entries.
+
+@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.
+
+-1 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.
+
+@item segment_time_metadata
+If set to 1, every packet will contain the @var{lavf.concat.start_time} and the
+@var{lavf.concat.duration} packet metadata values which are the start_time and
+the duration of the respective file segments in the concatenated output
+expressed in microseconds. The duration metadata is only set if it is known
+based on the concat file.
+The default is 0.
+
+@end table
+
+@subsection Examples
+
+@itemize
+@item
+Use absolute filenames and include some comments:
+@example
+# my first filename
+file /mnt/share/file-1.wav
+# my second filename including whitespace
+file '/mnt/share/file 2.wav'
+# my third filename including whitespace plus single quote
+file '/mnt/share/file 3'\''.wav'
+@end example
+
+@item
+Allow for input format auto-probing, use safe filenames and set the duration of
+the first file:
+@example
+ffconcat version 1.0
+
+file file-1.wav
+duration 20.0
+
+file subdir/file-2.wav
+@end example
+@end itemize
+
+@section dash
+
+Dynamic Adaptive Streaming over HTTP demuxer.
+
+This demuxer presents all AVStreams found in the manifest.
+By setting the discard flags on AVStreams the caller can decide
+which streams to actually receive.
+Each stream mirrors the @code{id} and @code{bandwidth} properties from the
+@code{<Representation>} as metadata keys named "id" and "variant_bitrate" respectively.
+
+@section flv, live_flv
+
+Adobe Flash Video Format demuxer.
+
+This demuxer is used to demux FLV files and RTMP network streams. In case of live network streams, if you force format, you may use live_flv option instead of flv to survive timestamp discontinuities.
+
+@example
+ffmpeg -f flv -i myfile.flv ...
+ffmpeg -f live_flv -i rtmp://<any.server>/anything/key ....
+@end example
+
+
+@table @option
+@item -flv_metadata @var{bool}
+Allocate the streams according to the onMetaData array content.
+
+@item -flv_ignore_prevtag @var{bool}
+Ignore the size of previous tag value.
+
+@item -flv_full_metadata @var{bool}
+Output all context of the onMetadata.
+@end table
+
+@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 hls
+
+HLS demuxer
+
+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".
+
+It accepts the following options:
+
+@table @option
+@item live_start_index
+segment index to start live streams at (negative values are from the end).
+
+@item allowed_extensions
+',' separated list of file extensions that hls is allowed to access.
+
+@item max_reload
+Maximum number of times a insufficient list is attempted to be reloaded.
+Default value is 1000.
+
+@item m3u8_hold_counters
+The maximum number of times to load m3u8 when it refreshes without new segments.
+Default value is 1000.
+
+@item http_persistent
+Use persistent HTTP connections. Applicable only for HTTP streams.
+Enabled by default.
+
+@item http_multiple
+Use multiple HTTP connections for downloading HTTP segments.
+Enabled by default for HTTP/1.1 servers.
+
+@item http_seekable
+Use HTTP partial requests for downloading HTTP segments.
+0 = disable, 1 = enable, -1 = auto, Default is auto.
+@end table
+
+@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.
+@item export_path_metadata
+If set to 1, will add two extra fields to the metadata found in input, making them
+also available for other filters (see @var{drawtext} filter for examples). Default
+value is 0. The extra fields are described below:
+@table @option
+@item lavf.image2dec.source_path
+Corresponds to the full path to the input file being read.
+@item lavf.image2dec.source_basename
+Corresponds to the name of the file being read.
+@end table
+
+@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 libgme
+
+The Game Music Emu library is a collection of video game music file emulators.
+
+See @url{https://bitbucket.org/mpyne/game-music-emu/overview} for more information.
+
+It accepts the following options:
+
+@table @option
+
+@item track_index
+Set the index of which track to demux. The demuxer can only export one track.
+Track indexes start at 0. Default is to pick the first track. Number of tracks
+is exported as @var{tracks} metadata entry.
+
+@item sample_rate
+Set the sampling rate of the exported track. Range is 1000 to 999999. Default is 44100.
+
+@item max_size @emph{(bytes)}
+The demuxer buffers the entire file into memory. Adjust this value to set the maximum buffer size,
+which in turn, acts as a ceiling for the size of files that can be read.
+Default is 50 MiB.
+
+@end table
+
+@section libmodplug
+
+ModPlug based module demuxer
+
+See @url{https://github.com/Konstanty/libmodplug}
+
+It will export one 2-channel 16-bit 44.1 kHz audio stream.
+Optionally, a @code{pal8} 16-color video stream can be exported with or without printed metadata.
+
+It accepts the following options:
+
+@table @option
+@item noise_reduction
+Apply a simple low-pass filter. Can be 1 (on) or 0 (off). Default is 0.
+
+@item reverb_depth
+Set amount of reverb. Range 0-100. Default is 0.
+
+@item reverb_delay
+Set delay in ms, clamped to 40-250 ms. Default is 0.
+
+@item bass_amount
+Apply bass expansion a.k.a. XBass or megabass. Range is 0 (quiet) to 100 (loud). Default is 0.
+
+@item bass_range
+Set cutoff i.e. upper-bound for bass frequencies. Range is 10-100 Hz. Default is 0.
+
+@item surround_depth
+Apply a Dolby Pro-Logic surround effect. Range is 0 (quiet) to 100 (heavy). Default is 0.
+
+@item surround_delay
+Set surround delay in ms, clamped to 5-40 ms. Default is 0.
+
+@item max_size
+The demuxer buffers the entire file into memory. Adjust this value to set the maximum buffer size,
+which in turn, acts as a ceiling for the size of files that can be read. Range is 0 to 100 MiB.
+0 removes buffer size limit (not recommended). Default is 5 MiB.
+
+@item video_stream_expr
+String which is evaluated using the eval API to assign colors to the generated video stream.
+Variables which can be used are @code{x}, @code{y}, @code{w}, @code{h}, @code{t}, @code{speed},
+@code{tempo}, @code{order}, @code{pattern} and @code{row}.
+
+@item video_stream
+Generate video stream. Can be 1 (on) or 0 (off). Default is 0.
+
+@item video_stream_w
+Set video frame width in 'chars' where one char indicates 8 pixels. Range is 20-512. Default is 30.
+
+@item video_stream_h
+Set video frame height in 'chars' where one char indicates 8 pixels. Range is 20-512. Default is 30.
+
+@item video_stream_ptxt
+Print metadata on video stream. Includes @code{speed}, @code{tempo}, @code{order}, @code{pattern},
+@code{row} and @code{ts} (time in ms). Can be 1 (on) or 0 (off). Default is 1.
+
+@end table
+
+@section libopenmpt
+
+libopenmpt based module demuxer
+
+See @url{https://lib.openmpt.org/libopenmpt/} for more information.
+
+Some files have multiple subsongs (tracks) this can be set with the @option{subsong}
+option.
+
+It accepts the following options:
+
+@table @option
+@item subsong
+Set the subsong index. This can be either  'all', 'auto', or the index of the
+subsong. Subsong indexes start at 0. The default is 'auto'.
+
+The default value is to let libopenmpt choose.
+
+@item layout
+Set the channel layout. Valid values are 1, 2, and 4 channel layouts.
+The default value is STEREO.
+
+@item sample_rate
+Set the sample rate for libopenmpt to output.
+Range is from 1000 to INT_MAX. The value default is 48000.
+@end table
+
+@section mov/mp4/3gp
+
+Demuxer for Quicktime File Format & ISO/IEC Base Media File Format (ISO/IEC 14496-12 or MPEG-4 Part 12, ISO/IEC 15444-12 or JPEG 2000 Part 12).
+
+Registered extensions: mov, mp4, m4a, 3gp, 3g2, mj2, psp, m4b, ism, ismv, isma, f4v
+
+@subsection Options
+
+This demuxer accepts the following options:
+@table @option
+@item enable_drefs
+Enable loading of external tracks, disabled by default.
+Enabling this can theoretically leak information in some use cases.
+
+@item use_absolute_path
+Allows loading of external tracks via absolute paths, disabled by default.
+Enabling this poses a security risk. It should only be enabled if the source
+is known to be non-malicious.
+
+@item seek_streams_individually
+When seeking, identify the closest point in each stream individually and demux packets in
+that stream from identified point. This can lead to a different sequence of packets compared
+to demuxing linearly from the beginning. Default is true.
+
+@item ignore_editlist
+Ignore any edit list atoms. The demuxer, by default, modifies the stream index to reflect the
+timeline described by the edit list. Default is false.
+
+@item advanced_editlist
+Modify the stream index to reflect the timeline described by the edit list. @code{ignore_editlist}
+must be set to false for this option to be effective.
+If both @code{ignore_editlist} and this option are set to false, then only the
+start of the stream index is modified to reflect initial dwell time or starting timestamp
+described by the edit list. Default is true.
+
+@item ignore_chapters
+Don't parse chapters. This includes GoPro 'HiLight' tags/moments. Note that chapters are
+only parsed when input is seekable. Default is false.
+
+@item use_mfra_for
+For seekable fragmented input, set fragment's starting timestamp from media fragment random access box, if present.
+
+Following options are available:
+@table @samp
+@item auto
+Auto-detect whether to set mfra timestamps as PTS or DTS @emph{(default)}
+
+@item dts
+Set mfra timestamps as DTS
+
+@item pts
+Set mfra timestamps as PTS
+
+@item 0
+Don't use mfra box to set timestamps
+@end table
+
+@item export_all
+Export unrecognized boxes within the @var{udta} box as metadata entries. The first four
+characters of the box type are set as the key. Default is false.
+
+@item export_xmp
+Export entire contents of @var{XMP_} box and @var{uuid} box as a string with key @code{xmp}. Note that
+if @code{export_all} is set and this option isn't, the contents of @var{XMP_} box are still exported
+but with key @code{XMP_}. Default is false.
+
+@item activation_bytes
+4-byte key required to decrypt Audible AAX and AAX+ files. See Audible AAX subsection below.
+
+@item audible_fixed_key
+Fixed key used for handling Audible AAX/AAX+ files. It has been pre-set so should not be necessary to
+specify.
+
+@item decryption_key
+16-byte key, in hex, to decrypt files encrypted using ISO Common Encryption (CENC/AES-128 CTR; ISO/IEC 23001-7).
+@end table
+
+@subsection Audible AAX
+
+Audible AAX files are encrypted M4B files, and they can be decrypted by specifying a 4 byte activation secret.
+@example
+ffmpeg -activation_bytes 1CEB00DA -i test.aax -vn -c:a copy output.mp4
+@end example
+
+@section mpegts
+
+MPEG-2 transport stream demuxer.
+
+This demuxer accepts the following options:
+@table @option
+@item resync_size
+Set size limit for looking up a new synchronization. Default value is
+65536.
+
+@item skip_unknown_pmt
+Skip PMTs for programs not defined in the PAT. Default value is 0.
+
+@item fix_teletext_pts
+Override 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.
+
+@item ts_packetsize
+Output option carrying the raw packet size in bytes.
+Show the detected raw packet size, cannot be set by the user.
+
+@item scan_all_pmts
+Scan and combine all PMTs. The value is an integer with value from -1
+to 1 (-1 means automatic setting, 1 means enabled, 0 means
+disabled). Default value is -1.
+
+@item merge_pmt_versions
+Re-use existing streams when a PMT's version is updated and elementary
+streams move to different PIDs. Default value is 0.
+@end table
+
+@section mpjpeg
+
+MJPEG encapsulated in multi-part MIME demuxer.
+
+This demuxer allows reading of MJPEG, where each frame is represented as a part of
+multipart/x-mixed-replace stream.
+@table @option
+
+@item strict_mime_boundary
+Default implementation applies a relaxed standard to multi-part MIME boundary detection,
+to prevent regression with numerous existing endpoints not generating a proper MIME
+MJPEG stream. Turning this option on by setting it to 1 will result in a stricter check
+of the boundary value.
+@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
+
+@section vapoursynth
+
+Vapoursynth wrapper.
+
+Due to security concerns, Vapoursynth scripts will not
+be autodetected so the input format has to be forced. For ff* CLI tools,
+add @code{-f vapoursynth} before the input @code{-i yourscript.vpy}.
+
+This demuxer accepts the following option:
+@table @option
+@item max_script_size
+The demuxer buffers the entire script into memory. Adjust this value to set the maximum buffer size,
+which in turn, acts as a ceiling for the size of scripts that can be read.
+Default is 1 MiB.
+@end table
+
+@c man end DEMUXERS

externals/ffmpeg/ffmpeg/doc/dev_community/community.md

@@ -0,0 +1,79 @@
+# FFmpeg project
+
+## Organisation
+
+The FFmpeg project is organized through a community working on global consensus.
+
+Decisions are taken by the ensemble of active members, through voting and
+are aided by two committees.
+
+## General Assembly
+
+The ensemble of active members is called the General Assembly (GA).
+
+The General Assembly is sovereign and legitimate for all its decisions
+regarding the FFmpeg project.
+
+The General Assembly is made up of active contributors.
+
+Contributors are considered "active contributors" if they have pushed more
+than 20 patches in the last 36 months in the main FFmpeg repository, or
+if they have been voted in by the GA.
+
+Additional members are added to the General Assembly through a vote after
+proposal by a member of the General Assembly.
+They are part of the GA for two years, after which they need a confirmation by
+the GA.
+
+## Voting
+
+Voting is done using a ranked voting system, currently running on https://vote.ffmpeg.org/ .
+
+Majority vote means more than 50% of the expressed ballots.
+
+## Technical Committee
+
+The Technical Committee (TC) is here to arbitrate and make decisions when
+technical conflicts occur in the project.
+They will consider the merits of all the positions, judge them and make a
+decision.
+
+The TC resolves technical conflicts but is not a technical steering committee.
+
+Decisions by the TC are binding for all the contributors.
+
+Decisions made by the TC can be re-opened after 1 year or by a majority vote
+of the General Assembly, requested by one of the member of the GA.
+
+The TC is elected by the General Assembly for a duration of 1 year, and
+is composed of 5 members.
+Members can be re-elected if they wish. A majority vote in the General Assembly
+can trigger a new election of the TC.
+
+The members of the TC can be elected from outside of the GA.
+Candidates for election can either be suggested or self-nominated.
+
+The conflict resolution process is detailed in the [resolution process](resolution_process.md) document.
+
+## Community committee
+
+The Community Committee (CC) is here to arbitrage and make decisions when
+inter-personal conflicts occur in the project. It will decide quickly and
+take actions, for the sake of the project.
+
+The CC can remove privileges of offending members, including removal of
+commit access and temporary ban from the community.
+
+Decisions made by the CC can be re-opened after 1 year or by a majority vote
+of the General Assembly. Indefinite bans from the community must be confirmed
+by the General Assembly, in a majority vote.
+
+The CC is elected by the General Assembly for a duration of 1 year, and is
+composed of 5 members.
+Members can be re-elected if they wish. A majority vote in the General Assembly
+can trigger a new election of the CC.
+
+The members of the CC can be elected from outside of the GA.
+Candidates for election can either be suggested or self-nominated.
+
+The CC is governed by and responsible for enforcing the Code of Conduct.

externals/ffmpeg/ffmpeg/doc/dev_community/resolution_process.md

@@ -0,0 +1,91 @@
+# Technical Committee
+
+_This document only makes sense with the rules from [the community document](community)_.
+
+The Technical Committee (**TC**) is here to arbitrate and make decisions when
+technical conflicts occur in the project.
+
+The TC main role is to resolve technical conflicts.
+It is therefore not a technical steering committee, but it is understood that
+some decisions might impact the future of the project.
+
+# Process
+
+## Seizing
+
+The TC can take possession of any technical matter that it sees fit.
+
+To involve the TC in a matter, email tc@ or CC them on an ongoing discussion.
+
+As members of TC are developers, they also can email tc@ to raise an issue.
+
+## Announcement
+
+The TC, once seized, must announce itself on the main mailing list, with a _[TC]_ tag.
+
+The TC has 2 modes of operation: a RFC one and an internal one.
+
+If the TC thinks it needs the input from the larger community, the TC can call
+for a RFC. Else, it can decide by itself.
+
+If the disagreement involves a member of the TC, that member should recuse
+themselves from the decision.
+
+The decision to use a RFC process or an internal discussion is a discretionary
+decision of the TC.
+
+The TC can also reject a seizure for a few reasons such as:
+the matter was not discussed enough previously; it lacks expertise to reach a
+beneficial decision on the matter; or the matter is too trivial.
+
+### RFC call
+
+In the RFC mode, one person from the TC posts on the mailing list the
+technical question and will request input from the community.
+
+The mail will have the following specification:
+* a precise title
+* a specific tag [TC RFC]
+* a top-level email
+* contain a precise question that does not exceed 100 words and that is answerable by developers
+* may have an extra description, or a link to a previous discussion, if deemed necessary,
+* contain a precise end date for the answers.
+
+The answers from the community must be on the main mailing list and must have
+the following specification:
+* keep the tag and the title unchanged
+* limited to 400 words
+* a first-level, answering directly to the main email
+* answering to the question.
+
+Further replies to answers are permitted, as long as they conform to the
+community standards of politeness, they are limited to 100 words, and are not
+nested more than once. (max-depth=2)
+
+After the end-date, mails on the thread will be ignored.
+
+Violations of those rules will be escalated through the Community Committee.
+
+After all the emails are in, the TC has 96 hours to give its final decision.
+Exceptionally, the TC can request an extra delay, that will be notified on the
+mailing list.
+
+### Within TC
+
+In the internal case, the TC has 96 hours to give its final decision.
+Exceptionally, the TC can request an extra delay.
+
+
+## Decisions
+
+The decisions from the TC will be sent on the mailing list, with the _[TC]_ tag.
+
+Internally, the TC should take decisions with a majority, or using
+ranked-choice voting.
+
+The decision from the TC should be published with a summary of the reasons that
+lead to this decision.
+
+The decisions from the TC are final, until the matters are reopened after
+no less than one year.
+

externals/ffmpeg/ffmpeg/doc/developer.texi

@@ -0,0 +1,866 @@
+\input texinfo @c -*- texinfo -*-
+@documentencoding UTF-8
+
+@settitle Developer Documentation
+@titlepage
+@center @titlefont{Developer Documentation}
+@end titlepage
+
+@top
+
+@contents
+
+@chapter Notes for external developers
+
+This document is mostly useful for internal FFmpeg developers.
+External developers who need to use the API in their application should
+refer to the API doxygen documentation in the public headers, and
+check the examples in @file{doc/examples} and in the source code to
+see how the public API is employed.
+
+You can use the FFmpeg libraries in your commercial program, but you
+are encouraged to @emph{publish any patch you make}. In this case the
+best way to proceed is to send your patches to the ffmpeg-devel
+mailing list following the guidelines illustrated in the remainder of
+this document.
+
+For more detailed legal information about the use of FFmpeg in
+external programs read the @file{LICENSE} file in the source tree and
+consult @url{https://ffmpeg.org/legal.html}.
+
+@chapter Contributing
+
+There are 2 ways by which code gets into FFmpeg:
+@itemize @bullet
+@item Submitting patches to the ffmpeg-devel mailing list.
+      See @ref{Submitting patches} for details.
+@item Directly committing changes to the main tree.
+@end itemize
+
+Whichever way, changes should be reviewed by the maintainer of the code
+before they are committed. And they should follow the @ref{Coding Rules}.
+The developer making the commit and the author are responsible for their changes
+and should try to fix issues their commit causes.
+
+@anchor{Coding Rules}
+@chapter Coding Rules
+
+@section Code formatting conventions
+
+There are the following guidelines regarding the indentation in files:
+
+@itemize @bullet
+@item
+Indent size is 4.
+
+@item
+The TAB character is forbidden outside of Makefiles as is any
+form of trailing whitespace. Commits containing either will be
+rejected by the git repository.
+
+@item
+You should try to limit your code lines to 80 characters; however, do so if
+and only if this improves readability.
+
+@item
+K&R coding style is used.
+@end itemize
+The presentation is one inspired by 'indent -i4 -kr -nut'.
+
+The main priority in FFmpeg is simplicity and small code size in order to
+minimize the bug count.
+
+@section Comments
+Use the JavaDoc/Doxygen  format (see examples below) so that code documentation
+can be generated automatically. All nontrivial functions should have a comment
+above them explaining what the function does, even if it is just one sentence.
+All structures and their member variables should be documented, too.
+
+Avoid Qt-style and similar Doxygen syntax with @code{!} in it, i.e. replace
+@code{//!} with @code{///} and similar.  Also @@ syntax should be employed
+for markup commands, i.e. use @code{@@param} and not @code{\param}.
+
+@example
+/**
+ * @@file
+ * MPEG codec.
+ * @@author ...
+ */
+
+/**
+ * Summary sentence.
+ * more text ...
+ * ...
+ */
+typedef struct Foobar @{
+    int var1; /**< var1 description */
+    int var2; ///< var2 description
+    /** var3 description */
+    int var3;
+@} Foobar;
+
+/**
+ * Summary sentence.
+ * more text ...
+ * ...
+ * @@param my_parameter description of my_parameter
+ * @@return return value description
+ */
+int myfunc(int my_parameter)
+...
+@end example
+
+@section C language features
+
+FFmpeg is programmed in the ISO C90 language with a few additional
+features from ISO C99, namely:
+
+@itemize @bullet
+@item
+the @samp{inline} keyword;
+
+@item
+@samp{//} comments;
+
+@item
+designated struct initializers (@samp{struct s x = @{ .i = 17 @};});
+
+@item
+compound literals (@samp{x = (struct s) @{ 17, 23 @};}).
+
+@item
+for loops with variable definition (@samp{for (int i = 0; i < 8; i++)});
+
+@item
+Variadic macros (@samp{#define ARRAY(nb, ...) (int[nb + 1])@{ nb, __VA_ARGS__ @}});
+
+@item
+Implementation defined behavior for signed integers is assumed to match the
+expected behavior for two's complement. Non representable values in integer
+casts are binary truncated. Shift right of signed values uses sign extension.
+@end itemize
+
+These features are supported by all compilers we care about, so we will not
+accept patches to remove their use unless they absolutely do not impair
+clarity and performance.
+
+All code must compile with recent versions of GCC and a number of other
+currently supported compilers. To ensure compatibility, please do not use
+additional C99 features or GCC extensions. Especially watch out for:
+
+@itemize @bullet
+@item
+mixing statements and declarations;
+
+@item
+@samp{long long} (use @samp{int64_t} instead);
+
+@item
+@samp{__attribute__} not protected by @samp{#ifdef __GNUC__} or similar;
+
+@item
+GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}).
+@end itemize
+
+@section Naming conventions
+All names should be composed with underscores (_), not CamelCase. For example,
+@samp{avfilter_get_video_buffer} is an acceptable function name and
+@samp{AVFilterGetVideo} is not. The exception from this are type names, like
+for example structs and enums; they should always be in CamelCase.
+
+There are the following conventions for naming variables and functions:
+
+@itemize @bullet
+@item
+For local variables no prefix is required.
+
+@item
+For file-scope variables and functions declared as @code{static}, no prefix
+is required.
+
+@item
+For variables and functions visible outside of file scope, but only used
+internally by a library, an @code{ff_} prefix should be used,
+e.g. @samp{ff_w64_demuxer}.
+
+@item
+For variables and functions visible outside of file scope, used internally
+across multiple libraries, use @code{avpriv_} as prefix, for example,
+@samp{avpriv_report_missing_feature}.
+
+@item
+Each library has its own prefix for public symbols, in addition to the
+commonly used @code{av_} (@code{avformat_} for libavformat,
+@code{avcodec_} for libavcodec, @code{swr_} for libswresample, etc).
+Check the existing code and choose names accordingly.
+Note that some symbols without these prefixes are also exported for
+retro-compatibility reasons. These exceptions are declared in the
+@code{lib<name>/lib<name>.v} files.
+@end itemize
+
+Furthermore, name space reserved for the system should not be invaded.
+Identifiers ending in @code{_t} are reserved by
+@url{http://pubs.opengroup.org/onlinepubs/007904975/functions/xsh_chap02_02.html#tag_02_02_02, POSIX}.
+Also avoid names starting with @code{__} or @code{_} followed by an uppercase
+letter as they are reserved by the C standard. Names starting with @code{_}
+are reserved at the file level and may not be used for externally visible
+symbols. If in doubt, just avoid names starting with @code{_} altogether.
+
+@section Miscellaneous conventions
+
+@itemize @bullet
+@item
+fprintf and printf are forbidden in libavformat and libavcodec,
+please use av_log() instead.
+
+@item
+Casts should be used only when necessary. Unneeded parentheses
+should also be avoided if they don't make the code easier to understand.
+@end itemize
+
+@section Editor configuration
+In order to configure Vim to follow FFmpeg formatting conventions, paste
+the following snippet into your @file{.vimrc}:
+@example
+" indentation rules for FFmpeg: 4 spaces, no tabs
+set expandtab
+set shiftwidth=4
+set softtabstop=4
+set cindent
+set cinoptions=(0
+" Allow tabs in Makefiles.
+autocmd FileType make,automake set noexpandtab shiftwidth=8 softtabstop=8
+" Trailing whitespace and tabs are forbidden, so highlight them.
+highlight ForbiddenWhitespace ctermbg=red guibg=red
+match ForbiddenWhitespace /\s\+$\|\t/
+" Do not highlight spaces at the end of line while typing on that line.
+autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@@<!$/
+@end example
+
+For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
+@lisp
+(c-add-style "ffmpeg"
+             '("k&r"
+               (c-basic-offset . 4)
+               (indent-tabs-mode . nil)
+               (show-trailing-whitespace . t)
+               (c-offsets-alist
+                (statement-cont . (c-lineup-assignments +)))
+               )
+             )
+(setq c-default-style "ffmpeg")
+@end lisp
+
+@chapter Development Policy
+
+@section Patches/Committing
+@subheading Licenses for patches must be compatible with FFmpeg.
+Contributions should be licensed under the
+@uref{http://www.gnu.org/licenses/lgpl-2.1.html, LGPL 2.1},
+including an "or any later version" clause, or, if you prefer
+a gift-style license, the
+@uref{http://opensource.org/licenses/isc-license.txt, ISC} or
+@uref{http://mit-license.org/, MIT} license.
+@uref{http://www.gnu.org/licenses/gpl-2.0.html, GPL 2} including
+an "or any later version" clause is also acceptable, but LGPL is
+preferred.
+If you add a new file, give it a proper license header. Do not copy and
+paste it from a random place, use an existing file as template.
+
+@subheading You must not commit code which breaks FFmpeg!
+This means unfinished code which is enabled and breaks compilation,
+or compiles but does not work/breaks the regression tests. Code which
+is unfinished but disabled may be permitted under-circumstances, like
+missing samples or an implementation with a small subset of features.
+Always check the mailing list for any reviewers with issues and test
+FATE before you push.
+
+@subheading Keep the main commit message short with an extended description below.
+The commit message should have a short first line in the form of
+a @samp{topic: short description} as a header, separated by a newline
+from the body consisting of an explanation of why the change is necessary.
+If the commit fixes a known bug on the bug tracker, the commit message
+should include its bug ID. Referring to the issue on the bug tracker does
+not exempt you from writing an excerpt of the bug in the commit message.
+
+@subheading Testing must be adequate but not excessive.
+If it works for you, others, and passes FATE then it should be OK to commit
+it, provided it fits the other committing criteria. You should not worry about
+over-testing things. If your code has problems (portability, triggers
+compiler bugs, unusual environment etc) they will be reported and eventually
+fixed.
+
+@subheading Do not commit unrelated changes together.
+They should be split them into self-contained pieces. Also do not forget
+that if part B depends on part A, but A does not depend on B, then A can
+and should be committed first and separate from B. Keeping changes well
+split into self-contained parts makes reviewing and understanding them on
+the commit log mailing list easier. This also helps in case of debugging
+later on.
+Also if you have doubts about splitting or not splitting, do not hesitate to
+ask/discuss it on the developer mailing list.
+
+@subheading Ask before you change the build system (configure, etc).
+Do not commit changes to the build system (Makefiles, configure script)
+which change behavior, defaults etc, without asking first. The same
+applies to compiler warning fixes, trivial looking fixes and to code
+maintained by other developers. We usually have a reason for doing things
+the way we do. Send your changes as patches to the ffmpeg-devel mailing
+list, and if the code maintainers say OK, you may commit. This does not
+apply to files you wrote and/or maintain.
+
+@subheading Cosmetic changes should be kept in separate patches.
+We refuse source indentation and other cosmetic changes if they are mixed
+with functional changes, such commits will be rejected and removed. Every
+developer has his own indentation style, you should not change it. Of course
+if you (re)write something, you can use your own style, even though we would
+prefer if the indentation throughout FFmpeg was consistent (Many projects
+force a given indentation style - we do not.). If you really need to make
+indentation changes (try to avoid this), separate them strictly from real
+changes.
+
+NOTE: If you had to put if()@{ .. @} over a large (> 5 lines) chunk of code,
+then either do NOT change the indentation of the inner part within (do not
+move it to the right)! or do so in a separate commit
+
+@subheading Commit messages should always be filled out properly.
+Always fill out the commit log message. Describe in a few lines what you
+changed and why. You can refer to mailing list postings if you fix a
+particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.
+Recommended format:
+
+@example
+area changed: Short 1 line description
+
+details describing what and why and giving references.
+@end example
+
+@subheading Credit the author of the patch.
+Make sure the author of the commit is set correctly. (see git commit --author)
+If you apply a patch, send an
+answer to ffmpeg-devel (or wherever you got the patch from) saying that
+you applied the patch.
+
+@subheading Complex patches should refer to discussion surrounding them.
+When applying patches that have been discussed (at length) on the mailing
+list, reference the thread in the log message.
+
+@subheading Always wait long enough before pushing changes
+Do NOT commit to code actively maintained by others without permission.
+Send a patch to ffmpeg-devel. If no one answers within a reasonable
+time-frame (12h for build failures and security fixes, 3 days small changes,
+1 week for big patches) then commit your patch if you think it is OK.
+Also note, the maintainer can simply ask for more time to review!
+
+@section Code
+@subheading API/ABI changes should be discussed before they are made.
+Do not change behavior of the programs (renaming options etc) or public
+API or ABI without first discussing it on the ffmpeg-devel mailing list.
+Do not remove widely used functionality or features (redundant code can be removed).
+
+@subheading Remember to check if you need to bump versions for libav*.
+Depending on the change, you may need to change the version integer.
+Incrementing the first component means no backward compatibility to
+previous versions (e.g. removal of a function from the public API).
+Incrementing the second component means backward compatible change
+(e.g. addition of a function to the public API or extension of an
+existing data structure).
+Incrementing the third component means a noteworthy binary compatible
+change (e.g. encoder bug fix that matters for the decoder). The third
+component always starts at 100 to distinguish FFmpeg from Libav.
+
+@subheading Warnings for correct code may be disabled if there is no other option.
+Compiler warnings indicate potential bugs or code with bad style. If a type of
+warning always points to correct and clean code, that warning should
+be disabled, not the code changed.
+Thus the remaining warnings can either be bugs or correct code.
+If it is a bug, the bug has to be fixed. If it is not, the code should
+be changed to not generate a warning unless that causes a slowdown
+or obfuscates the code.
+
+@subheading Check untrusted input properly.
+Never write to unallocated memory, never write over the end of arrays,
+always check values read from some untrusted source before using them
+as array index or other risky things.
+
+@section Documentation/Other
+@subheading Subscribe to the ffmpeg-devel mailing list.
+It is important to be subscribed to the
+@uref{https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel, ffmpeg-devel}
+mailing list. Almost any non-trivial patch is to be sent there for review.
+Other developers may have comments about your contribution. We expect you see
+those comments, and to improve it if requested. (N.B. Experienced committers
+have other channels, and may sometimes skip review for trivial fixes.) Also,
+discussion here about bug fixes and FFmpeg improvements by other developers may
+be helpful information for you. Finally, by being a list subscriber, your
+contribution will be posted immediately to the list, without the moderation
+hold which messages from non-subscribers experience.
+
+However, it is more important to the project that we receive your patch than
+that you be subscribed to the ffmpeg-devel list. If you have a patch, and don't
+want to subscribe and discuss the patch, then please do send it to the list
+anyway.
+
+@subheading Subscribe to the ffmpeg-cvslog mailing list.
+Diffs of all commits are sent to the
+@uref{https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-cvslog, ffmpeg-cvslog}
+mailing list. Some developers read this list to review all code base changes
+from all sources. Subscribing to this list is not mandatory.
+
+@subheading Keep the documentation up to date.
+Update the documentation if you change behavior or add features. If you are
+unsure how best to do this, send a patch to ffmpeg-devel, the documentation
+maintainer(s) will review and commit your stuff.
+
+@subheading Important discussions should be accessible to all.
+Try to keep important discussions and requests (also) on the public
+developer mailing list, so that all developers can benefit from them.
+
+@subheading Check your entries in MAINTAINERS.
+Make sure that no parts of the codebase that you maintain are missing from the
+@file{MAINTAINERS} file. If something that you want to maintain is missing add it with
+your name after it.
+If at some point you no longer want to maintain some code, then please help in
+finding a new maintainer and also don't forget to update the @file{MAINTAINERS} file.
+
+We think our rules are not too hard. If you have comments, contact us.
+
+@chapter Code of conduct
+
+Be friendly and respectful towards others and third parties.
+Treat others the way you yourself want to be treated.
+
+Be considerate. Not everyone shares the same viewpoint and priorities as you do.
+Different opinions and interpretations help the project.
+Looking at issues from a different perspective assists development.
+
+Do not assume malice for things that can be attributed to incompetence. Even if
+it is malice, it's rarely good to start with that as initial assumption.
+
+Stay friendly even if someone acts contrarily. Everyone has a bad day
+once in a while.
+If you yourself have a bad day or are angry then try to take a break and reply
+once you are calm and without anger if you have to.
+
+Try to help other team members and cooperate if you can.
+
+The goal of software development is to create technical excellence, not for any
+individual to be better and "win" against the others. Large software projects
+are only possible and successful through teamwork.
+
+If someone struggles do not put them down. Give them a helping hand
+instead and point them in the right direction.
+
+Finally, keep in mind the immortal words of Bill and Ted,
+"Be excellent to each other."
+
+@anchor{Submitting patches}
+@chapter Submitting patches
+
+First, read the @ref{Coding Rules} above if you did not yet, in particular
+the rules regarding patch submission.
+
+When you submit your patch, please use @code{git format-patch} or
+@code{git send-email}. We cannot read other diffs :-).
+
+Also please do not submit a patch which contains several unrelated changes.
+Split it into separate, self-contained pieces. This does not mean splitting
+file by file. Instead, make the patch as small as possible while still
+keeping it as a logical unit that contains an individual change, even
+if it spans multiple files. This makes reviewing your patches much easier
+for us and greatly increases your chances of getting your patch applied.
+
+Use the patcheck tool of FFmpeg to check your patch.
+The tool is located in the tools directory.
+
+Run the @ref{Regression tests} before submitting a patch in order to verify
+it does not cause unexpected problems.
+
+It also helps quite a bit if you tell us what the patch does (for example
+'replaces lrint by lrintf'), and why (for example '*BSD isn't C99 compliant
+and has no lrint()')
+
+Also please if you send several patches, send each patch as a separate mail,
+do not attach several unrelated patches to the same mail.
+
+Patches should be posted to the
+@uref{https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel, ffmpeg-devel}
+mailing list. Use @code{git send-email} when possible since it will properly
+send patches without requiring extra care. If you cannot, then send patches
+as base64-encoded attachments, so your patch is not trashed during
+transmission. Also ensure the correct mime type is used
+(text/x-diff or text/x-patch or at least text/plain) and that only one
+patch is inline or attached per mail.
+You can check @url{https://patchwork.ffmpeg.org}, if your patch does not show up, its mime type
+likely was wrong.
+
+Your patch will be reviewed on the mailing list. You will likely be asked
+to make some changes and are expected to send in an improved version that
+incorporates the requests from the review. This process may go through
+several iterations. Once your patch is deemed good enough, some developer
+will pick it up and commit it to the official FFmpeg tree.
+
+Give us a few days to react. But if some time passes without reaction,
+send a reminder by email. Your patch should eventually be dealt with.
+
+
+@chapter New codecs or formats checklist
+
+@enumerate
+@item
+Did you use av_cold for codec initialization and close functions?
+
+@item
+Did you add a long_name under NULL_IF_CONFIG_SMALL to the AVCodec or
+AVInputFormat/AVOutputFormat struct?
+
+@item
+Did you bump the minor version number (and reset the micro version
+number) in @file{libavcodec/version.h} or @file{libavformat/version.h}?
+
+@item
+Did you register it in @file{allcodecs.c} or @file{allformats.c}?
+
+@item
+Did you add the AVCodecID to @file{avcodec.h}?
+When adding new codec IDs, also add an entry to the codec descriptor
+list in @file{libavcodec/codec_desc.c}.
+
+@item
+If it has a FourCC, did you add it to @file{libavformat/riff.c},
+even if it is only a decoder?
+
+@item
+Did you add a rule to compile the appropriate files in the Makefile?
+Remember to do this even if you're just adding a format to a file that is
+already being compiled by some other rule, like a raw demuxer.
+
+@item
+Did you add an entry to the table of supported formats or codecs in
+@file{doc/general.texi}?
+
+@item
+Did you add an entry in the Changelog?
+
+@item
+If it depends on a parser or a library, did you add that dependency in
+configure?
+
+@item
+Did you @code{git add} the appropriate files before committing?
+
+@item
+Did you make sure it compiles standalone, i.e. with
+@code{configure --disable-everything --enable-decoder=foo}
+(or @code{--enable-demuxer} or whatever your component is)?
+@end enumerate
+
+
+@chapter Patch submission checklist
+
+@enumerate
+@item
+Does @code{make fate} pass with the patch applied?
+
+@item
+Was the patch generated with git format-patch or send-email?
+
+@item
+Did you sign-off your patch? (@code{git commit -s})
+See @uref{https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/plain/Documentation/process/submitting-patches.rst, Sign your work} for the meaning
+of @dfn{sign-off}.
+
+@item
+Did you provide a clear git commit log message?
+
+@item
+Is the patch against latest FFmpeg git master branch?
+
+@item
+Are you subscribed to ffmpeg-devel?
+(the list is subscribers only due to spam)
+
+@item
+Have you checked that the changes are minimal, so that the same cannot be
+achieved with a smaller patch and/or simpler final code?
+
+@item
+If the change is to speed critical code, did you benchmark it?
+
+@item
+If you did any benchmarks, did you provide them in the mail?
+
+@item
+Have you checked that the patch does not introduce buffer overflows or
+other security issues?
+
+@item
+Did you test your decoder or demuxer against damaged data? If no, see
+tools/trasher, the noise bitstream filter, and
+@uref{http://caca.zoy.org/wiki/zzuf, zzuf}. Your decoder or demuxer
+should not crash, end in a (near) infinite loop, or allocate ridiculous
+amounts of memory when fed damaged data.
+
+@item
+Did you test your decoder or demuxer against sample files?
+Samples may be obtained at @url{https://samples.ffmpeg.org}.
+
+@item
+Does the patch not mix functional and cosmetic changes?
+
+@item
+Did you add tabs or trailing whitespace to the code? Both are forbidden.
+
+@item
+Is the patch attached to the email you send?
+
+@item
+Is the mime type of the patch correct? It should be text/x-diff or
+text/x-patch or at least text/plain and not application/octet-stream.
+
+@item
+If the patch fixes a bug, did you provide a verbose analysis of the bug?
+
+@item
+If the patch fixes a bug, did you provide enough information, including
+a sample, so the bug can be reproduced and the fix can be verified?
+Note please do not attach samples >100k to mails but rather provide a
+URL, you can upload to @url{https://streams.videolan.org/upload/}.
+
+@item
+Did you provide a verbose summary about what the patch does change?
+
+@item
+Did you provide a verbose explanation why it changes things like it does?
+
+@item
+Did you provide a verbose summary of the user visible advantages and
+disadvantages if the patch is applied?
+
+@item
+Did you provide an example so we can verify the new feature added by the
+patch easily?
+
+@item
+If you added a new file, did you insert a license header? It should be
+taken from FFmpeg, not randomly copied and pasted from somewhere else.
+
+@item
+You should maintain alphabetical order in alphabetically ordered lists as
+long as doing so does not break API/ABI compatibility.
+
+@item
+Lines with similar content should be aligned vertically when doing so
+improves readability.
+
+@item
+Consider adding a regression test for your code.
+
+@item
+If you added YASM code please check that things still work with --disable-yasm.
+
+@item
+Make sure you check the return values of function and return appropriate
+error codes. Especially memory allocation functions like @code{av_malloc()}
+are notoriously left unchecked, which is a serious problem.
+
+@item
+Test your code with valgrind and or Address Sanitizer to ensure it's free
+of leaks, out of array accesses, etc.
+@end enumerate
+
+@chapter Patch review process
+
+All patches posted to ffmpeg-devel will be reviewed, unless they contain a
+clear note that the patch is not for the git master branch.
+Reviews and comments will be posted as replies to the patch on the
+mailing list. The patch submitter then has to take care of every comment,
+that can be by resubmitting a changed patch or by discussion. Resubmitted
+patches will themselves be reviewed like any other patch. If at some point
+a patch passes review with no comments then it is approved, that can for
+simple and small patches happen immediately while large patches will generally
+have to be changed and reviewed many times before they are approved.
+After a patch is approved it will be committed to the repository.
+
+We will review all submitted patches, but sometimes we are quite busy so
+especially for large patches this can take several weeks.
+
+If you feel that the review process is too slow and you are willing to try to
+take over maintainership of the area of code you change then just clone
+git master and maintain the area of code there. We will merge each area from
+where its best maintained.
+
+When resubmitting patches, please do not make any significant changes
+not related to the comments received during review. Such patches will
+be rejected. Instead, submit significant changes or new features as
+separate patches.
+
+Everyone is welcome to review patches. Also if you are waiting for your patch
+to be reviewed, please consider helping to review other patches, that is a great
+way to get everyone's patches reviewed sooner.
+
+@anchor{Regression tests}
+@chapter Regression tests
+
+Before submitting a patch (or committing to the repository), you should at least
+test that you did not break anything.
+
+Running 'make fate' accomplishes this, please see @url{fate.html} for details.
+
+[Of course, some patches may change the results of the regression tests. In
+this case, the reference results of the regression tests shall be modified
+accordingly].
+
+@section Adding files to the fate-suite dataset
+
+When there is no muxer or encoder available to generate test media for a
+specific test then the media has to be included in the fate-suite.
+First please make sure that the sample file is as small as possible to test the
+respective decoder or demuxer sufficiently. Large files increase network
+bandwidth and disk space requirements.
+Once you have a working fate test and fate sample, provide in the commit
+message or introductory message for the patch series that you post to
+the ffmpeg-devel mailing list, a direct link to download the sample media.
+
+@section Visualizing Test Coverage
+
+The FFmpeg build system allows visualizing the test coverage in an easy
+manner with the coverage tools @code{gcov}/@code{lcov}.  This involves
+the following steps:
+
+@enumerate
+@item
+    Configure to compile with instrumentation enabled:
+    @code{configure --toolchain=gcov}.
+
+@item
+    Run your test case, either manually or via FATE. This can be either
+    the full FATE regression suite, or any arbitrary invocation of any
+    front-end tool provided by FFmpeg, in any combination.
+
+@item
+    Run @code{make lcov} to generate coverage data in HTML format.
+
+@item
+    View @code{lcov/index.html} in your preferred HTML viewer.
+@end enumerate
+
+You can use the command @code{make lcov-reset} to reset the coverage
+measurements. You will need to rerun @code{make lcov} after running a
+new test.
+
+@section Using Valgrind
+
+The configure script provides a shortcut for using valgrind to spot bugs
+related to memory handling. Just add the option
+@code{--toolchain=valgrind-memcheck} or @code{--toolchain=valgrind-massif}
+to your configure line, and reasonable defaults will be set for running
+FATE under the supervision of either the @strong{memcheck} or the
+@strong{massif} tool of the valgrind suite.
+
+In case you need finer control over how valgrind is invoked, use the
+@code{--target-exec='valgrind <your_custom_valgrind_options>} option in
+your configure line instead.
+
+@anchor{Release process}
+@chapter Release process
+
+FFmpeg maintains a set of @strong{release branches}, which are the
+recommended deliverable for system integrators and distributors (such as
+Linux distributions, etc.). At regular times, a @strong{release
+manager} prepares, tests and publishes tarballs on the
+@url{https://ffmpeg.org} website.
+
+There are two kinds of releases:
+
+@enumerate
+@item
+@strong{Major releases} always include the latest and greatest
+features and functionality.
+
+@item
+@strong{Point releases} are cut from @strong{release} branches,
+which are named @code{release/X}, with @code{X} being the release
+version number.
+@end enumerate
+
+Note that we promise to our users that shared libraries from any FFmpeg
+release never break programs that have been @strong{compiled} against
+previous versions of @strong{the same release series} in any case!
+
+However, from time to time, we do make API changes that require adaptations
+in applications. Such changes are only allowed in (new) major releases and
+require further steps such as bumping library version numbers and/or
+adjustments to the symbol versioning file. Please discuss such changes
+on the @strong{ffmpeg-devel} mailing list in time to allow forward planning.
+
+@anchor{Criteria for Point Releases}
+@section Criteria for Point Releases
+
+Changes that match the following criteria are valid candidates for
+inclusion into a point release:
+
+@enumerate
+@item
+Fixes a security issue, preferably identified by a @strong{CVE
+number} issued by @url{http://cve.mitre.org/}.
+
+@item
+Fixes a documented bug in @url{https://trac.ffmpeg.org}.
+
+@item
+Improves the included documentation.
+
+@item
+Retains both source code and binary compatibility with previous
+point releases of the same release branch.
+@end enumerate
+
+The order for checking the rules is (1 OR 2 OR 3) AND 4.
+
+
+@section Release Checklist
+
+The release process involves the following steps:
+
+@enumerate
+@item
+Ensure that the @file{RELEASE} file contains the version number for
+the upcoming release.
+
+@item
+Add the release at @url{https://trac.ffmpeg.org/admin/ticket/versions}.
+
+@item
+Announce the intent to do a release to the mailing list.
+
+@item
+Make sure all relevant security fixes have been backported. See
+@url{https://ffmpeg.org/security.html}.
+
+@item
+Ensure that the FATE regression suite still passes in the release
+branch on at least @strong{i386} and @strong{amd64}
+(cf. @ref{Regression tests}).
+
+@item
+Prepare the release tarballs in @code{bz2} and @code{gz} formats, and
+supplementing files that contain @code{gpg} signatures
+
+@item
+Publish the tarballs at @url{https://ffmpeg.org/releases}. Create and
+push an annotated tag in the form @code{nX}, with @code{X}
+containing the version number.
+
+@item
+Propose and send a patch to the @strong{ffmpeg-devel} mailing list
+with a news entry for the website.
+
+@item
+Publish the news entry.
+
+@item
+Send an announcement to the mailing list.
+@end enumerate
+
+@bye

externals/ffmpeg/ffmpeg/doc/devices.texi

@@ -0,0 +1,25 @@
+@chapter Device Options
+@c man begin DEVICE OPTIONS
+
+The libavdevice library provides the same interface as
+libavformat. Namely, an input device is considered like a demuxer, and
+an output device like a muxer, and the interface and generic device
+options are the same provided by libavformat (see the ffmpeg-formats
+manual).
+
+In addition each input or output device may support so-called private
+options, which are specific for that component.
+
+Options may be set by specifying -@var{option} @var{value} in the
+FFmpeg tools, or by setting the value explicitly in the device
+@code{AVFormatContext} options or using the @file{libavutil/opt.h} API
+for programmatic use.
+
+@c man end DEVICE OPTIONS
+
+@ifclear config-writeonly
+@include indevs.texi
+@end ifclear
+@ifclear config-readonly
+@include outdevs.texi
+@end ifclear

externals/ffmpeg/ffmpeg/doc/doxy-wrapper.sh

@@ -0,0 +1,21 @@
+#!/bin/sh
+
+OUT_DIR="${1}"
+DOXYFILE="${2}"
+DOXYGEN="${3}"
+
+shift 3
+
+if [ -e "VERSION" ]; then
+    VERSION=`cat "VERSION"`
+else
+    VERSION=`git describe`
+fi
+
+$DOXYGEN - <<EOF
+@INCLUDE        = ${DOXYFILE}
+INPUT           = $@
+HTML_TIMESTAMP  = NO
+PROJECT_NUMBER  = $VERSION
+OUTPUT_DIRECTORY = $OUT_DIR
+EOF

externals/ffmpeg/ffmpeg/doc/doxy/.gitignore

@@ -0,0 +1 @@
+/html/

externals/ffmpeg/ffmpeg/doc/errno.txt

@@ -0,0 +1,174 @@
+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  -  ++++++  File name 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

externals/ffmpeg/ffmpeg/doc/examples/.gitignore

@@ -0,0 +1,24 @@
+/avio_list_dir
+/avio_reading
+/decode_audio
+/decode_video
+/demuxing_decoding
+/encode_audio
+/encode_video
+/extract_mvs
+/filter_audio
+/filtering_audio
+/filtering_video
+/http_multiclient
+/hw_decode
+/metadata
+/muxing
+/pc-uninstalled
+/qsvdec
+/remuxing
+/resampling_audio
+/scaling_video
+/transcode_aac
+/transcoding
+/vaapi_encode
+/vaapi_transcode

externals/ffmpeg/ffmpeg/doc/examples/Makefile

@@ -0,0 +1,64 @@
+EXAMPLES-$(CONFIG_AVIO_LIST_DIR_EXAMPLE)     += avio_list_dir
+EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE)      += avio_reading
+EXAMPLES-$(CONFIG_DECODE_AUDIO_EXAMPLE)      += decode_audio
+EXAMPLES-$(CONFIG_DECODE_VIDEO_EXAMPLE)      += decode_video
+EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding
+EXAMPLES-$(CONFIG_ENCODE_AUDIO_EXAMPLE)      += encode_audio
+EXAMPLES-$(CONFIG_ENCODE_VIDEO_EXAMPLE)      += encode_video
+EXAMPLES-$(CONFIG_EXTRACT_MVS_EXAMPLE)       += extract_mvs
+EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE)      += filter_audio
+EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE)   += filtering_audio
+EXAMPLES-$(CONFIG_FILTERING_VIDEO_EXAMPLE)   += filtering_video
+EXAMPLES-$(CONFIG_HTTP_MULTICLIENT_EXAMPLE)  += http_multiclient
+EXAMPLES-$(CONFIG_HW_DECODE_EXAMPLE)         += hw_decode
+EXAMPLES-$(CONFIG_METADATA_EXAMPLE)          += metadata
+EXAMPLES-$(CONFIG_MUXING_EXAMPLE)            += muxing
+EXAMPLES-$(CONFIG_QSVDEC_EXAMPLE)            += qsvdec
+EXAMPLES-$(CONFIG_REMUXING_EXAMPLE)          += remuxing
+EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE)  += resampling_audio
+EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE)     += scaling_video
+EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE)     += transcode_aac
+EXAMPLES-$(CONFIG_TRANSCODING_EXAMPLE)       += transcoding
+EXAMPLES-$(CONFIG_VAAPI_ENCODE_EXAMPLE)      += vaapi_encode
+EXAMPLES-$(CONFIG_VAAPI_TRANSCODE_EXAMPLE)   += vaapi_transcode
+
+EXAMPLES       := $(EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)$(EXESUF))
+EXAMPLES_G     := $(EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)_g$(EXESUF))
+ALL_EXAMPLES   := $(EXAMPLES) $(EXAMPLES-:%=doc/examples/%$(PROGSSUF)$(EXESUF))
+ALL_EXAMPLES_G := $(EXAMPLES_G) $(EXAMPLES-:%=doc/examples/%$(PROGSSUF)_g$(EXESUF))
+PROGS          += $(EXAMPLES)
+
+EXAMPLE_MAKEFILE := $(SRC_PATH)/doc/examples/Makefile
+EXAMPLES_FILES := $(wildcard $(SRC_PATH)/doc/examples/*.c) $(SRC_PATH)/doc/examples/README $(EXAMPLE_MAKEFILE)
+
+$(foreach P,$(EXAMPLES),$(eval OBJS-$(P:%$(PROGSSUF)$(EXESUF)=%) = $(P:%$(PROGSSUF)$(EXESUF)=%).o))
+$(EXAMPLES_G): %$(PROGSSUF)_g$(EXESUF): %.o
+
+examples: $(EXAMPLES)
+
+$(EXAMPLES:%$(PROGSSUF)$(EXESUF)=%.o): | doc/examples
+OUTDIRS += doc/examples
+
+DOXY_INPUT += $(EXAMPLES:%$(PROGSSUF)$(EXESUF)=%.c)
+
+install: install-examples
+
+install-examples: $(EXAMPLES_FILES)
+	$(Q)mkdir -p "$(DATADIR)/examples"
+	$(INSTALL) -m 644 $(EXAMPLES_FILES) "$(DATADIR)/examples"
+	$(INSTALL) -m 644 $(EXAMPLE_MAKEFILE:%=%.example) "$(DATADIR)/examples/Makefile"
+
+uninstall: uninstall-examples
+
+uninstall-examples:
+	$(RM) -r "$(DATADIR)/examples"
+
+examplesclean:
+	$(RM) $(ALL_EXAMPLES) $(ALL_EXAMPLES_G)
+	$(RM) $(CLEANSUFFIXES:%=doc/examples/%)
+
+docclean:: examplesclean
+
+-include $(wildcard $(EXAMPLES:%$(PROGSSUF)$(EXESUF)=%.d))
+
+.PHONY: examples

externals/ffmpeg/ffmpeg/doc/examples/Makefile.example

@@ -0,0 +1,50 @@
+# 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)) $(LDLIBS)
+
+EXAMPLES=       avio_list_dir                      \
+                avio_reading                       \
+                decode_audio                       \
+                decode_video                       \
+                demuxing_decoding                  \
+                encode_audio                       \
+                encode_video                       \
+                extract_mvs                        \
+                filtering_video                    \
+                filtering_audio                    \
+                http_multiclient                   \
+                hw_decode                          \
+                metadata                           \
+                muxing                             \
+                remuxing                           \
+                resampling_audio                   \
+                scaling_video                      \
+                transcode_aac                      \
+                transcoding                        \
+
+OBJS=$(addsuffix .o,$(EXAMPLES))
+
+# the following examples make explicit use of the math library
+avcodec:           LDLIBS += -lm
+encode_audio:      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)

externals/ffmpeg/ffmpeg/doc/examples/README

@@ -0,0 +1,23 @@
+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.

externals/ffmpeg/ffmpeg/doc/examples/avio_list_dir.c

@@ -0,0 +1,130 @@
+/*
+ * 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>";
+}
+
+static int list_op(const char *input_dir)
+{
+    AVIODirEntry *entry = NULL;
+    AVIODirContext *ctx = NULL;
+    int cnt, ret;
+    char filemode[4], uid_and_gid[20];
+
+    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);
+    return ret;
+}
+
+static void usage(const char *program_name)
+{
+    fprintf(stderr, "usage: %s input_dir\n"
+            "API example program to show how to list files in directory "
+            "accessed through AVIOContext.\n", program_name);
+}
+
+int main(int argc, char *argv[])
+{
+    int ret;
+
+    av_log_set_level(AV_LOG_DEBUG);
+
+    if (argc < 2) {
+        usage(argv[0]);
+        return 1;
+    }
+
+    avformat_network_init();
+
+    ret = list_op(argv[1]);
+
+    avformat_network_deinit();
+
+    return ret < 0 ? 1 : 0;
+}

externals/ffmpeg/ffmpeg/doc/examples/avio_reading.c

@@ -0,0 +1,134 @@
+/*
+ * Copyright (c) 2014 Stefano Sabatini
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * libavformat AVIOContext API example.
+ *
+ * Make libavformat demuxer access media content through a custom
+ * AVIOContext read callback.
+ * @example avio_reading.c
+ */
+
+#include <libavcodec/avcodec.h>
+#include <libavformat/avformat.h>
+#include <libavformat/avio.h>
+#include <libavutil/file.h>
+
+struct buffer_data {
+    uint8_t *ptr;
+    size_t size; ///< size left in the buffer
+};
+
+static int read_packet(void *opaque, uint8_t *buf, int buf_size)
+{
+    struct buffer_data *bd = (struct buffer_data *)opaque;
+    buf_size = FFMIN(buf_size, bd->size);
+
+    if (!buf_size)
+        return AVERROR_EOF;
+    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];
+
+    /* 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);
+    avio_context_free(&avio_ctx);
+
+    av_file_unmap(buffer, buffer_size);
+
+    if (ret < 0) {
+        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
+        return 1;
+    }
+
+    return 0;
+}

externals/ffmpeg/ffmpeg/doc/examples/decode_audio.c

@@ -0,0 +1,236 @@
+/*
+ * 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
+ * audio decoding with libavcodec API example
+ *
+ * @example decode_audio.c
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include <libavutil/frame.h>
+#include <libavutil/mem.h>
+
+#include <libavcodec/avcodec.h>
+
+#define AUDIO_INBUF_SIZE 20480
+#define AUDIO_REFILL_THRESH 4096
+
+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;
+}
+
+static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame,
+                   FILE *outfile)
+{
+    int i, ch;
+    int ret, data_size;
+
+    /* send the packet with the compressed data to the decoder */
+    ret = avcodec_send_packet(dec_ctx, pkt);
+    if (ret < 0) {
+        fprintf(stderr, "Error submitting the packet to the decoder\n");
+        exit(1);
+    }
+
+    /* read all the output frames (in general there may be any number of them */
+    while (ret >= 0) {
+        ret = avcodec_receive_frame(dec_ctx, frame);
+        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
+            return;
+        else if (ret < 0) {
+            fprintf(stderr, "Error during decoding\n");
+            exit(1);
+        }
+        data_size = av_get_bytes_per_sample(dec_ctx->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 < frame->nb_samples; i++)
+            for (ch = 0; ch < dec_ctx->channels; ch++)
+                fwrite(frame->data[ch] + data_size*i, 1, data_size, outfile);
+    }
+}
+
+int main(int argc, char **argv)
+{
+    const char *outfilename, *filename;
+    const AVCodec *codec;
+    AVCodecContext *c= NULL;
+    AVCodecParserContext *parser = NULL;
+    int len, ret;
+    FILE *f, *outfile;
+    uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
+    uint8_t *data;
+    size_t   data_size;
+    AVPacket *pkt;
+    AVFrame *decoded_frame = NULL;
+    enum AVSampleFormat sfmt;
+    int n_channels = 0;
+    const char *fmt;
+
+    if (argc <= 2) {
+        fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
+        exit(0);
+    }
+    filename    = argv[1];
+    outfilename = argv[2];
+
+    pkt = av_packet_alloc();
+
+    /* find the MPEG audio decoder */
+    codec = avcodec_find_decoder(AV_CODEC_ID_MP2);
+    if (!codec) {
+        fprintf(stderr, "Codec not found\n");
+        exit(1);
+    }
+
+    parser = av_parser_init(codec->id);
+    if (!parser) {
+        fprintf(stderr, "Parser 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 */
+    data      = inbuf;
+    data_size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
+
+    while (data_size > 0) {
+        if (!decoded_frame) {
+            if (!(decoded_frame = av_frame_alloc())) {
+                fprintf(stderr, "Could not allocate audio frame\n");
+                exit(1);
+            }
+        }
+
+        ret = av_parser_parse2(parser, c, &pkt->data, &pkt->size,
+                               data, data_size,
+                               AV_NOPTS_VALUE, AV_NOPTS_VALUE, 0);
+        if (ret < 0) {
+            fprintf(stderr, "Error while parsing\n");
+            exit(1);
+        }
+        data      += ret;
+        data_size -= ret;
+
+        if (pkt->size)
+            decode(c, pkt, decoded_frame, outfile);
+
+        if (data_size < AUDIO_REFILL_THRESH) {
+            memmove(inbuf, data, data_size);
+            data = inbuf;
+            len = fread(data + data_size, 1,
+                        AUDIO_INBUF_SIZE - data_size, f);
+            if (len > 0)
+                data_size += len;
+        }
+    }
+
+    /* flush the decoder */
+    pkt->data = NULL;
+    pkt->size = 0;
+    decode(c, pkt, decoded_frame, outfile);
+
+    /* print output pcm infomations, because there have no metadata of pcm */
+    sfmt = c->sample_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 = c->channels;
+    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, c->sample_rate,
+           outfilename);
+end:
+    fclose(outfile);
+    fclose(f);
+
+    avcodec_free_context(&c);
+    av_parser_close(parser);
+    av_frame_free(&decoded_frame);
+    av_packet_free(&pkt);
+
+    return 0;
+}

externals/ffmpeg/ffmpeg/doc/examples/decode_video.c

@@ -0,0 +1,187 @@
+/*
+ * 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
+ * video decoding with libavcodec API example
+ *
+ * @example decode_video.c
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include <libavcodec/avcodec.h>
+
+#define INBUF_SIZE 4096
+
+static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
+                     char *filename)
+{
+    FILE *f;
+    int i;
+
+    f = fopen(filename,"wb");
+    fprintf(f, "P5\n%d %d\n%d\n", xsize, ysize, 255);
+    for (i = 0; i < ysize; i++)
+        fwrite(buf + i * wrap, 1, xsize, f);
+    fclose(f);
+}
+
+static void decode(AVCodecContext *dec_ctx, AVFrame *frame, AVPacket *pkt,
+                   const char *filename)
+{
+    char buf[1024];
+    int ret;
+
+    ret = avcodec_send_packet(dec_ctx, pkt);
+    if (ret < 0) {
+        fprintf(stderr, "Error sending a packet for decoding\n");
+        exit(1);
+    }
+
+    while (ret >= 0) {
+        ret = avcodec_receive_frame(dec_ctx, frame);
+        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
+            return;
+        else if (ret < 0) {
+            fprintf(stderr, "Error during decoding\n");
+            exit(1);
+        }
+
+        printf("saving frame %3d\n", dec_ctx->frame_number);
+        fflush(stdout);
+
+        /* the picture is allocated by the decoder. no need to
+           free it */
+        snprintf(buf, sizeof(buf), "%s-%d", filename, dec_ctx->frame_number);
+        pgm_save(frame->data[0], frame->linesize[0],
+                 frame->width, frame->height, buf);
+    }
+}
+
+int main(int argc, char **argv)
+{
+    const char *filename, *outfilename;
+    const AVCodec *codec;
+    AVCodecParserContext *parser;
+    AVCodecContext *c= NULL;
+    FILE *f;
+    AVFrame *frame;
+    uint8_t inbuf[INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
+    uint8_t *data;
+    size_t   data_size;
+    int ret;
+    AVPacket *pkt;
+
+    if (argc <= 2) {
+        fprintf(stderr, "Usage: %s <input file> <output file>\n"
+                "And check your input file is encoded by mpeg1video please.\n", argv[0]);
+        exit(0);
+    }
+    filename    = argv[1];
+    outfilename = argv[2];
+
+    pkt = av_packet_alloc();
+    if (!pkt)
+        exit(1);
+
+    /* set end of buffer to 0 (this ensures that no overreading happens for damaged MPEG streams) */
+    memset(inbuf + INBUF_SIZE, 0, AV_INPUT_BUFFER_PADDING_SIZE);
+
+    /* find the MPEG-1 video decoder */
+    codec = avcodec_find_decoder(AV_CODEC_ID_MPEG1VIDEO);
+    if (!codec) {
+        fprintf(stderr, "Codec not found\n");
+        exit(1);
+    }
+
+    parser = av_parser_init(codec->id);
+    if (!parser) {
+        fprintf(stderr, "parser not found\n");
+        exit(1);
+    }
+
+    c = avcodec_alloc_context3(codec);
+    if (!c) {
+        fprintf(stderr, "Could not allocate video codec context\n");
+        exit(1);
+    }
+
+    /* 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);
+    }
+
+    while (!feof(f)) {
+        /* read raw data from the input file */
+        data_size = fread(inbuf, 1, INBUF_SIZE, f);
+        if (!data_size)
+            break;
+
+        /* use the parser to split the data into frames */
+        data = inbuf;
+        while (data_size > 0) {
+            ret = av_parser_parse2(parser, c, &pkt->data, &pkt->size,
+                                   data, data_size, AV_NOPTS_VALUE, AV_NOPTS_VALUE, 0);
+            if (ret < 0) {
+                fprintf(stderr, "Error while parsing\n");
+                exit(1);
+            }
+            data      += ret;
+            data_size -= ret;
+
+            if (pkt->size)
+                decode(c, frame, pkt, outfilename);
+        }
+    }
+
+    /* flush the decoder */
+    decode(c, frame, NULL, outfilename);
+
+    fclose(f);
+
+    av_parser_close(parser);
+    avcodec_free_context(&c);
+    av_frame_free(&frame);
+    av_packet_free(&pkt);
+
+    return 0;
+}

externals/ffmpeg/ffmpeg/doc/examples/demuxing_decoding.c

@@ -0,0 +1,382 @@
+/*
+ * 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 = NULL;
+static int video_frame_count = 0;
+static int audio_frame_count = 0;
+
+static int output_video_frame(AVFrame *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 n:%d coded_n:%d\n",
+           video_frame_count++, frame->coded_picture_number);
+
+    /* 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);
+    return 0;
+}
+
+static int output_audio_frame(AVFrame *frame)
+{
+    size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format);
+    printf("audio_frame n:%d nb_samples:%d pts:%s\n",
+           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);
+
+    return 0;
+}
+
+static int decode_packet(AVCodecContext *dec, const AVPacket *pkt)
+{
+    int ret = 0;
+
+    // submit the packet to the decoder
+    ret = avcodec_send_packet(dec, pkt);
+    if (ret < 0) {
+        fprintf(stderr, "Error submitting a packet for decoding (%s)\n", av_err2str(ret));
+        return ret;
+    }
+
+    // get all the available frames from the decoder
+    while (ret >= 0) {
+        ret = avcodec_receive_frame(dec, frame);
+        if (ret < 0) {
+            // those two return values are special and mean there is no output
+            // frame available, but there were no errors during decoding
+            if (ret == AVERROR_EOF || ret == AVERROR(EAGAIN))
+                return 0;
+
+            fprintf(stderr, "Error during decoding (%s)\n", av_err2str(ret));
+            return ret;
+        }
+
+        // write the frame data to output file
+        if (dec->codec->type == AVMEDIA_TYPE_VIDEO)
+            ret = output_video_frame(frame);
+        else
+            ret = output_audio_frame(frame);
+
+        av_frame_unref(frame);
+        if (ret < 0)
+            return ret;
+    }
+
+    return 0;
+}
+
+static int open_codec_context(int *stream_idx,
+                              AVCodecContext **dec_ctx, AVFormatContext *fmt_ctx, enum AVMediaType type)
+{
+    int ret, stream_index;
+    AVStream *st;
+    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 = avcodec_find_decoder(st->codecpar->codec_id);
+        if (!dec) {
+            fprintf(stderr, "Failed to find %s codec\n",
+                    av_get_media_type_string(type));
+            return AVERROR(EINVAL);
+        }
+
+        /* Allocate a codec context for the decoder */
+        *dec_ctx = avcodec_alloc_context3(dec);
+        if (!*dec_ctx) {
+            fprintf(stderr, "Failed to allocate the %s codec context\n",
+                    av_get_media_type_string(type));
+            return AVERROR(ENOMEM);
+        }
+
+        /* Copy codec parameters from input stream to output codec context */
+        if ((ret = avcodec_parameters_to_context(*dec_ctx, st->codecpar)) < 0) {
+            fprintf(stderr, "Failed to copy %s codec parameters to decoder context\n",
+                    av_get_media_type_string(type));
+            return ret;
+        }
+
+        /* Init the decoders */
+        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;
+
+    if (argc != 4) {
+        fprintf(stderr, "usage: %s  input_file video_output_file audio_output_file\n"
+                "API example program to show how to read frames from an input file.\n"
+                "This program reads frames from a file, decodes them, and writes decoded\n"
+                "video frames to a rawvideo file named video_output_file, and decoded\n"
+                "audio frames to a rawaudio file named audio_output_file.\n",
+                argv[0]);
+        exit(1);
+    }
+    src_filename = argv[1];
+    video_dst_filename = argv[2];
+    audio_dst_filename = argv[3];
+
+    /* 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, &video_dec_ctx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) {
+        video_stream = fmt_ctx->streams[video_stream_idx];
+
+        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, &audio_dec_ctx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) {
+        audio_stream = fmt_ctx->streams[audio_stream_idx];
+        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;
+    }
+
+    frame = av_frame_alloc();
+    if (!frame) {
+        fprintf(stderr, "Could not allocate frame\n");
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+
+    pkt = av_packet_alloc();
+    if (!pkt) {
+        fprintf(stderr, "Could not allocate packet\n");
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+
+    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) {
+        // check if the packet belongs to a stream we are interested in, otherwise
+        // skip it
+        if (pkt->stream_index == video_stream_idx)
+            ret = decode_packet(video_dec_ctx, pkt);
+        else if (pkt->stream_index == audio_stream_idx)
+            ret = decode_packet(audio_dec_ctx, pkt);
+        av_packet_unref(pkt);
+        if (ret < 0)
+            break;
+    }
+
+    /* flush the decoders */
+    if (video_dec_ctx)
+        decode_packet(video_dec_ctx, NULL);
+    if (audio_dec_ctx)
+        decode_packet(audio_dec_ctx, NULL);
+
+    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_free_context(&video_dec_ctx);
+    avcodec_free_context(&audio_dec_ctx);
+    avformat_close_input(&fmt_ctx);
+    if (video_dst_file)
+        fclose(video_dst_file);
+    if (audio_dst_file)
+        fclose(audio_dst_file);
+    av_packet_free(&pkt);
+    av_frame_free(&frame);
+    av_free(video_dst_data[0]);
+
+    return ret < 0;
+}

externals/ffmpeg/ffmpeg/doc/examples/encode_audio.c

@@ -0,0 +1,238 @@
+/*
+ * 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
+ * audio encoding with libavcodec API example.
+ *
+ * @example encode_audio.c
+ */
+
+#include <stdint.h>
+#include <stdio.h>
+#include <stdlib.h>
+
+#include <libavcodec/avcodec.h>
+
+#include <libavutil/channel_layout.h>
+#include <libavutil/common.h>
+#include <libavutil/frame.h>
+#include <libavutil/samplefmt.h>
+
+/* check that a given sample format is supported by the encoder */
+static int check_sample_fmt(const 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(const AVCodec *codec)
+{
+    const int *p;
+    int best_samplerate = 0;
+
+    if (!codec->supported_samplerates)
+        return 44100;
+
+    p = codec->supported_samplerates;
+    while (*p) {
+        if (!best_samplerate || abs(44100 - *p) < abs(44100 - best_samplerate))
+            best_samplerate = *p;
+        p++;
+    }
+    return best_samplerate;
+}
+
+/* select layout with the highest channel count */
+static int select_channel_layout(const 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;
+}
+
+static void encode(AVCodecContext *ctx, AVFrame *frame, AVPacket *pkt,
+                   FILE *output)
+{
+    int ret;
+
+    /* send the frame for encoding */
+    ret = avcodec_send_frame(ctx, frame);
+    if (ret < 0) {
+        fprintf(stderr, "Error sending the frame to the encoder\n");
+        exit(1);
+    }
+
+    /* read all the available output packets (in general there may be any
+     * number of them */
+    while (ret >= 0) {
+        ret = avcodec_receive_packet(ctx, pkt);
+        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
+            return;
+        else if (ret < 0) {
+            fprintf(stderr, "Error encoding audio frame\n");
+            exit(1);
+        }
+
+        fwrite(pkt->data, 1, pkt->size, output);
+        av_packet_unref(pkt);
+    }
+}
+
+int main(int argc, char **argv)
+{
+    const char *filename;
+    const AVCodec *codec;
+    AVCodecContext *c= NULL;
+    AVFrame *frame;
+    AVPacket *pkt;
+    int i, j, k, ret;
+    FILE *f;
+    uint16_t *samples;
+    float t, tincr;
+
+    if (argc <= 1) {
+        fprintf(stderr, "Usage: %s <output file>\n", argv[0]);
+        return 0;
+    }
+    filename = argv[1];
+
+    /* 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);
+    }
+
+    /* packet for holding encoded output */
+    pkt = av_packet_alloc();
+    if (!pkt) {
+        fprintf(stderr, "could not allocate the packet\n");
+        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;
+
+    /* allocate the data buffers */
+    ret = av_frame_get_buffer(frame, 0);
+    if (ret < 0) {
+        fprintf(stderr, "Could not allocate audio data buffers\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++) {
+        /* make sure the frame is writable -- makes a copy if the encoder
+         * kept a reference internally */
+        ret = av_frame_make_writable(frame);
+        if (ret < 0)
+            exit(1);
+        samples = (uint16_t*)frame->data[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(c, frame, pkt, f);
+    }
+
+    /* flush the encoder */
+    encode(c, NULL, pkt, f);
+
+    fclose(f);
+
+    av_frame_free(&frame);
+    av_packet_free(&pkt);
+    avcodec_free_context(&c);
+
+    return 0;
+}

externals/ffmpeg/ffmpeg/doc/examples/encode_video.c

@@ -0,0 +1,198 @@
+/*
+ * 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
+ * video encoding with libavcodec API example
+ *
+ * @example encode_video.c
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include <libavcodec/avcodec.h>
+
+#include <libavutil/opt.h>
+#include <libavutil/imgutils.h>
+
+static void encode(AVCodecContext *enc_ctx, AVFrame *frame, AVPacket *pkt,
+                   FILE *outfile)
+{
+    int ret;
+
+    /* send the frame to the encoder */
+    if (frame)
+        printf("Send frame %3"PRId64"\n", frame->pts);
+
+    ret = avcodec_send_frame(enc_ctx, frame);
+    if (ret < 0) {
+        fprintf(stderr, "Error sending a frame for encoding\n");
+        exit(1);
+    }
+
+    while (ret >= 0) {
+        ret = avcodec_receive_packet(enc_ctx, pkt);
+        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
+            return;
+        else if (ret < 0) {
+            fprintf(stderr, "Error during encoding\n");
+            exit(1);
+        }
+
+        printf("Write packet %3"PRId64" (size=%5d)\n", pkt->pts, pkt->size);
+        fwrite(pkt->data, 1, pkt->size, outfile);
+        av_packet_unref(pkt);
+    }
+}
+
+int main(int argc, char **argv)
+{
+    const char *filename, *codec_name;
+    const AVCodec *codec;
+    AVCodecContext *c= NULL;
+    int i, ret, x, y;
+    FILE *f;
+    AVFrame *frame;
+    AVPacket *pkt;
+    uint8_t endcode[] = { 0, 0, 1, 0xb7 };
+
+    if (argc <= 2) {
+        fprintf(stderr, "Usage: %s <output file> <codec name>\n", argv[0]);
+        exit(0);
+    }
+    filename = argv[1];
+    codec_name = argv[2];
+
+    /* find the mpeg1video encoder */
+    codec = avcodec_find_encoder_by_name(codec_name);
+    if (!codec) {
+        fprintf(stderr, "Codec '%s' not found\n", codec_name);
+        exit(1);
+    }
+
+    c = avcodec_alloc_context3(codec);
+    if (!c) {
+        fprintf(stderr, "Could not allocate video codec context\n");
+        exit(1);
+    }
+
+    pkt = av_packet_alloc();
+    if (!pkt)
+        exit(1);
+
+    /* put sample parameters */
+    c->bit_rate = 400000;
+    /* resolution must be a multiple of two */
+    c->width = 352;
+    c->height = 288;
+    /* frames per second */
+    c->time_base = (AVRational){1, 25};
+    c->framerate = (AVRational){25, 1};
+
+    /* 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 */
+    ret = avcodec_open2(c, codec, NULL);
+    if (ret < 0) {
+        fprintf(stderr, "Could not open codec: %s\n", av_err2str(ret));
+        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;
+
+    ret = av_frame_get_buffer(frame, 0);
+    if (ret < 0) {
+        fprintf(stderr, "Could not allocate the video frame data\n");
+        exit(1);
+    }
+
+    /* encode 1 second of video */
+    for (i = 0; i < 25; i++) {
+        fflush(stdout);
+
+        /* make sure the frame data is writable */
+        ret = av_frame_make_writable(frame);
+        if (ret < 0)
+            exit(1);
+
+        /* 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 */
+        encode(c, frame, pkt, f);
+    }
+
+    /* flush the encoder */
+    encode(c, NULL, pkt, f);
+
+    /* add sequence end code to have a real MPEG file */
+    if (codec->id == AV_CODEC_ID_MPEG1VIDEO || codec->id == AV_CODEC_ID_MPEG2VIDEO)
+        fwrite(endcode, 1, sizeof(endcode), f);
+    fclose(f);
+
+    avcodec_free_context(&c);
+    av_frame_free(&frame);
+    av_packet_free(&pkt);
+
+    return 0;
+}

externals/ffmpeg/ffmpeg/doc/examples/extract_mvs.c

@@ -0,0 +1,178 @@
+/*
+ * 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 int video_frame_count = 0;
+
+static int decode_packet(const AVPacket *pkt)
+{
+    int ret = avcodec_send_packet(video_dec_ctx, pkt);
+    if (ret < 0) {
+        fprintf(stderr, "Error while sending a packet to the decoder: %s\n", av_err2str(ret));
+        return ret;
+    }
+
+    while (ret >= 0)  {
+        ret = avcodec_receive_frame(video_dec_ctx, frame);
+        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
+            break;
+        } else if (ret < 0) {
+            fprintf(stderr, "Error while receiving a frame from the decoder: %s\n", av_err2str(ret));
+            return ret;
+        }
+
+        if (ret >= 0) {
+            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);
+                }
+            }
+            av_frame_unref(frame);
+        }
+    }
+
+    return 0;
+}
+
+static int open_codec_context(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, &dec, 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 {
+        int stream_idx = ret;
+        st = fmt_ctx->streams[stream_idx];
+
+        dec_ctx = avcodec_alloc_context3(dec);
+        if (!dec_ctx) {
+            fprintf(stderr, "Failed to allocate codec\n");
+            return AVERROR(EINVAL);
+        }
+
+        ret = avcodec_parameters_to_context(dec_ctx, st->codecpar);
+        if (ret < 0) {
+            fprintf(stderr, "Failed to copy codec parameters to codec context\n");
+            return ret;
+        }
+
+        /* 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;
+        }
+
+        video_stream_idx = stream_idx;
+        video_stream = fmt_ctx->streams[video_stream_idx];
+        video_dec_ctx = dec_ctx;
+    }
+
+    return 0;
+}
+
+int main(int argc, char **argv)
+{
+    int ret = 0;
+    AVPacket pkt = { 0 };
+
+    if (argc != 2) {
+        fprintf(stderr, "Usage: %s <video>\n", argv[0]);
+        exit(1);
+    }
+    src_filename = argv[1];
+
+    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);
+    }
+
+    open_codec_context(fmt_ctx, AVMEDIA_TYPE_VIDEO);
+
+    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");
+
+    /* read frames from the file */
+    while (av_read_frame(fmt_ctx, &pkt) >= 0) {
+        if (pkt.stream_index == video_stream_idx)
+            ret = decode_packet(&pkt);
+        av_packet_unref(&pkt);
+        if (ret < 0)
+            break;
+    }
+
+    /* flush cached frames */
+    decode_packet(NULL);
+
+end:
+    avcodec_free_context(&video_dec_ctx);
+    avformat_close_input(&fmt_ctx);
+    av_frame_free(&frame);
+    return ret < 0;
+}

externals/ffmpeg/ffmpeg/doc/examples/filter_audio.c

@@ -0,0 +1,363 @@
+/*
+ * 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;
+    const AVFilter  *abuffer;
+    AVFilterContext *volume_ctx;
+    const AVFilter  *volume;
+    AVFilterContext *aformat_ctx;
+    const AVFilter  *aformat;
+    AVFilterContext *abuffersink_ctx;
+    const 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;
+    }
+
+    /* 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;
+}

externals/ffmpeg/ffmpeg/doc/examples/filtering_audio.c

@@ -0,0 +1,292 @@
+/*
+ * 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/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 an audio stream in the input file\n");
+        return ret;
+    }
+    audio_stream_index = ret;
+
+    /* create decoding context */
+    dec_ctx = avcodec_alloc_context3(dec);
+    if (!dec_ctx)
+        return AVERROR(ENOMEM);
+    avcodec_parameters_to_context(dec_ctx, fmt_ctx->streams[audio_stream_index]->codecpar);
+
+    /* 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;
+    const AVFilter *abuffersrc  = avfilter_get_by_name("abuffer");
+    const 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(frame->channel_layout);
+    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 packet;
+    AVFrame *frame = av_frame_alloc();
+    AVFrame *filt_frame = av_frame_alloc();
+
+    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);
+    }
+
+    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 == audio_stream_index) {
+            ret = avcodec_send_packet(dec_ctx, &packet);
+            if (ret < 0) {
+                av_log(NULL, AV_LOG_ERROR, "Error while sending a packet to the decoder\n");
+                break;
+            }
+
+            while (ret >= 0) {
+                ret = avcodec_receive_frame(dec_ctx, frame);
+                if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
+                    break;
+                } else if (ret < 0) {
+                    av_log(NULL, AV_LOG_ERROR, "Error while receiving a frame from the decoder\n");
+                    goto end;
+                }
+
+                if (ret >= 0) {
+                    /* push the audio data from 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 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);
+                    }
+                    av_frame_unref(frame);
+                }
+            }
+        }
+        av_packet_unref(&packet);
+    }
+end:
+    avfilter_graph_free(&filter_graph);
+    avcodec_free_context(&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);
+}

externals/ffmpeg/ffmpeg/doc/examples/filtering_video.c

@@ -0,0 +1,291 @@
+/*
+ * 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 <stdio.h>
+#include <stdlib.h>
+
+#include <libavcodec/avcodec.h>
+#include <libavformat/avformat.h>
+#include <libavfilter/buffersink.h>
+#include <libavfilter/buffersrc.h>
+#include <libavutil/opt.h>
+
+const char *filter_descr = "scale=78:24,transpose=cclock";
+/* other way:
+   scale=78:24 [scl]; [scl] transpose=cclock // assumes "[in]" and "[out]" to be input output pads respectively
+ */
+
+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;
+
+    /* create decoding context */
+    dec_ctx = avcodec_alloc_context3(dec);
+    if (!dec_ctx)
+        return AVERROR(ENOMEM);
+    avcodec_parameters_to_context(dec_ctx, fmt_ctx->streams[video_stream_index]->codecpar);
+
+    /* 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;
+    const AVFilter *buffersrc  = avfilter_get_by_name("buffer");
+    const 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;
+    AVFrame *filt_frame;
+
+    if (argc != 2) {
+        fprintf(stderr, "Usage: %s file\n", argv[0]);
+        exit(1);
+    }
+
+    frame = av_frame_alloc();
+    filt_frame = av_frame_alloc();
+    if (!frame || !filt_frame) {
+        perror("Could not allocate frame");
+        exit(1);
+    }
+
+    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) {
+            ret = avcodec_send_packet(dec_ctx, &packet);
+            if (ret < 0) {
+                av_log(NULL, AV_LOG_ERROR, "Error while sending a packet to the decoder\n");
+                break;
+            }
+
+            while (ret >= 0) {
+                ret = avcodec_receive_frame(dec_ctx, frame);
+                if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
+                    break;
+                } else if (ret < 0) {
+                    av_log(NULL, AV_LOG_ERROR, "Error while receiving a frame from the decoder\n");
+                    goto end;
+                }
+
+                frame->pts = frame->best_effort_timestamp;
+
+                /* 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_packet_unref(&packet);
+    }
+end:
+    avfilter_graph_free(&filter_graph);
+    avcodec_free_context(&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);
+}

externals/ffmpeg/ffmpeg/doc/examples/http_multiclient.c

@@ -0,0 +1,156 @@
+/*
+ * Copyright (c) 2015 Stephan Holljes
+ *
+ * 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 multi-client network API usage example.
+ *
+ * @example http_multiclient.c
+ * This example will serve a file without decoding or demuxing it over http.
+ * Multiple clients can connect and will receive the same file.
+ */
+
+#include <libavformat/avformat.h>
+#include <libavutil/opt.h>
+#include <unistd.h>
+
+static void process_client(AVIOContext *client, const char *in_uri)
+{
+    AVIOContext *input = NULL;
+    uint8_t buf[1024];
+    int ret, n, reply_code;
+    uint8_t *resource = NULL;
+    while ((ret = avio_handshake(client)) > 0) {
+        av_opt_get(client, "resource", AV_OPT_SEARCH_CHILDREN, &resource);
+        // check for strlen(resource) is necessary, because av_opt_get()
+        // may return empty string.
+        if (resource && strlen(resource))
+            break;
+        av_freep(&resource);
+    }
+    if (ret < 0)
+        goto end;
+    av_log(client, AV_LOG_TRACE, "resource=%p\n", resource);
+    if (resource && resource[0] == '/' && !strcmp((resource + 1), in_uri)) {
+        reply_code = 200;
+    } else {
+        reply_code = AVERROR_HTTP_NOT_FOUND;
+    }
+    if ((ret = av_opt_set_int(client, "reply_code", reply_code, AV_OPT_SEARCH_CHILDREN)) < 0) {
+        av_log(client, AV_LOG_ERROR, "Failed to set reply_code: %s.\n", av_err2str(ret));
+        goto end;
+    }
+    av_log(client, AV_LOG_TRACE, "Set reply code to %d\n", reply_code);
+
+    while ((ret = avio_handshake(client)) > 0);
+
+    if (ret < 0)
+        goto end;
+
+    fprintf(stderr, "Handshake performed.\n");
+    if (reply_code != 200)
+        goto end;
+    fprintf(stderr, "Opening input file.\n");
+    if ((ret = avio_open2(&input, in_uri, AVIO_FLAG_READ, NULL, NULL)) < 0) {
+        av_log(input, AV_LOG_ERROR, "Failed to open input: %s: %s.\n", in_uri,
+               av_err2str(ret));
+        goto end;
+    }
+    for(;;) {
+        n = avio_read(input, buf, sizeof(buf));
+        if (n < 0) {
+            if (n == AVERROR_EOF)
+                break;
+            av_log(input, AV_LOG_ERROR, "Error reading from input: %s.\n",
+                   av_err2str(n));
+            break;
+        }
+        avio_write(client, buf, n);
+        avio_flush(client);
+    }
+end:
+    fprintf(stderr, "Flushing client\n");
+    avio_flush(client);
+    fprintf(stderr, "Closing client\n");
+    avio_close(client);
+    fprintf(stderr, "Closing input\n");
+    avio_close(input);
+    av_freep(&resource);
+}
+
+int main(int argc, char **argv)
+{
+    AVDictionary *options = NULL;
+    AVIOContext *client = NULL, *server = NULL;
+    const char *in_uri, *out_uri;
+    int ret, pid;
+    av_log_set_level(AV_LOG_TRACE);
+    if (argc < 3) {
+        printf("usage: %s input http://hostname[:port]\n"
+               "API example program to serve http to multiple clients.\n"
+               "\n", argv[0]);
+        return 1;
+    }
+
+    in_uri = argv[1];
+    out_uri = argv[2];
+
+    avformat_network_init();
+
+    if ((ret = av_dict_set(&options, "listen", "2", 0)) < 0) {
+        fprintf(stderr, "Failed to set listen mode for server: %s\n", av_err2str(ret));
+        return ret;
+    }
+    if ((ret = avio_open2(&server, out_uri, AVIO_FLAG_WRITE, NULL, &options)) < 0) {
+        fprintf(stderr, "Failed to open server: %s\n", av_err2str(ret));
+        return ret;
+    }
+    fprintf(stderr, "Entering main loop.\n");
+    for(;;) {
+        if ((ret = avio_accept(server, &client)) < 0)
+            goto end;
+        fprintf(stderr, "Accepted client, forking process.\n");
+        // XXX: Since we don't reap our children and don't ignore signals
+        //      this produces zombie processes.
+        pid = fork();
+        if (pid < 0) {
+            perror("Fork failed");
+            ret = AVERROR(errno);
+            goto end;
+        }
+        if (pid == 0) {
+            fprintf(stderr, "In child.\n");
+            process_client(client, in_uri);
+            avio_close(server);
+            exit(0);
+        }
+        if (pid > 0)
+            avio_close(client);
+    }
+end:
+    avio_close(server);
+    if (ret < 0 && ret != AVERROR_EOF) {
+        fprintf(stderr, "Some errors occurred: %s\n", av_err2str(ret));
+        return 1;
+    }
+    return 0;
+}

externals/ffmpeg/ffmpeg/doc/examples/hw_decode.c

@@ -0,0 +1,252 @@
+/*
+ * Copyright (c) 2017 Jun Zhao
+ * Copyright (c) 2017 Kaixuan Liu
+ *
+ * HW Acceleration API (video decoding) decode sample
+ *
+ * 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
+ * HW-Accelerated decoding example.
+ *
+ * @example hw_decode.c
+ * This example shows how to do HW-accelerated decoding with output
+ * frames from the HW video surfaces.
+ */
+
+#include <stdio.h>
+
+#include <libavcodec/avcodec.h>
+#include <libavformat/avformat.h>
+#include <libavutil/pixdesc.h>
+#include <libavutil/hwcontext.h>
+#include <libavutil/opt.h>
+#include <libavutil/avassert.h>
+#include <libavutil/imgutils.h>
+
+static AVBufferRef *hw_device_ctx = NULL;
+static enum AVPixelFormat hw_pix_fmt;
+static FILE *output_file = NULL;
+
+static int hw_decoder_init(AVCodecContext *ctx, const enum AVHWDeviceType type)
+{
+    int err = 0;
+
+    if ((err = av_hwdevice_ctx_create(&hw_device_ctx, type,
+                                      NULL, NULL, 0)) < 0) {
+        fprintf(stderr, "Failed to create specified HW device.\n");
+        return err;
+    }
+    ctx->hw_device_ctx = av_buffer_ref(hw_device_ctx);
+
+    return err;
+}
+
+static enum AVPixelFormat get_hw_format(AVCodecContext *ctx,
+                                        const enum AVPixelFormat *pix_fmts)
+{
+    const enum AVPixelFormat *p;
+
+    for (p = pix_fmts; *p != -1; p++) {
+        if (*p == hw_pix_fmt)
+            return *p;
+    }
+
+    fprintf(stderr, "Failed to get HW surface format.\n");
+    return AV_PIX_FMT_NONE;
+}
+
+static int decode_write(AVCodecContext *avctx, AVPacket *packet)
+{
+    AVFrame *frame = NULL, *sw_frame = NULL;
+    AVFrame *tmp_frame = NULL;
+    uint8_t *buffer = NULL;
+    int size;
+    int ret = 0;
+
+    ret = avcodec_send_packet(avctx, packet);
+    if (ret < 0) {
+        fprintf(stderr, "Error during decoding\n");
+        return ret;
+    }
+
+    while (1) {
+        if (!(frame = av_frame_alloc()) || !(sw_frame = av_frame_alloc())) {
+            fprintf(stderr, "Can not alloc frame\n");
+            ret = AVERROR(ENOMEM);
+            goto fail;
+        }
+
+        ret = avcodec_receive_frame(avctx, frame);
+        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
+            av_frame_free(&frame);
+            av_frame_free(&sw_frame);
+            return 0;
+        } else if (ret < 0) {
+            fprintf(stderr, "Error while decoding\n");
+            goto fail;
+        }
+
+        if (frame->format == hw_pix_fmt) {
+            /* retrieve data from GPU to CPU */
+            if ((ret = av_hwframe_transfer_data(sw_frame, frame, 0)) < 0) {
+                fprintf(stderr, "Error transferring the data to system memory\n");
+                goto fail;
+            }
+            tmp_frame = sw_frame;
+        } else
+            tmp_frame = frame;
+
+        size = av_image_get_buffer_size(tmp_frame->format, tmp_frame->width,
+                                        tmp_frame->height, 1);
+        buffer = av_malloc(size);
+        if (!buffer) {
+            fprintf(stderr, "Can not alloc buffer\n");
+            ret = AVERROR(ENOMEM);
+            goto fail;
+        }
+        ret = av_image_copy_to_buffer(buffer, size,
+                                      (const uint8_t * const *)tmp_frame->data,
+                                      (const int *)tmp_frame->linesize, tmp_frame->format,
+                                      tmp_frame->width, tmp_frame->height, 1);
+        if (ret < 0) {
+            fprintf(stderr, "Can not copy image to buffer\n");
+            goto fail;
+        }
+
+        if ((ret = fwrite(buffer, 1, size, output_file)) < 0) {
+            fprintf(stderr, "Failed to dump raw data.\n");
+            goto fail;
+        }
+
+    fail:
+        av_frame_free(&frame);
+        av_frame_free(&sw_frame);
+        av_freep(&buffer);
+        if (ret < 0)
+            return ret;
+    }
+}
+
+int main(int argc, char *argv[])
+{
+    AVFormatContext *input_ctx = NULL;
+    int video_stream, ret;
+    AVStream *video = NULL;
+    AVCodecContext *decoder_ctx = NULL;
+    AVCodec *decoder = NULL;
+    AVPacket packet;
+    enum AVHWDeviceType type;
+    int i;
+
+    if (argc < 4) {
+        fprintf(stderr, "Usage: %s <device type> <input file> <output file>\n", argv[0]);
+        return -1;
+    }
+
+    type = av_hwdevice_find_type_by_name(argv[1]);
+    if (type == AV_HWDEVICE_TYPE_NONE) {
+        fprintf(stderr, "Device type %s is not supported.\n", argv[1]);
+        fprintf(stderr, "Available device types:");
+        while((type = av_hwdevice_iterate_types(type)) != AV_HWDEVICE_TYPE_NONE)
+            fprintf(stderr, " %s", av_hwdevice_get_type_name(type));
+        fprintf(stderr, "\n");
+        return -1;
+    }
+
+    /* open the input file */
+    if (avformat_open_input(&input_ctx, argv[2], NULL, NULL) != 0) {
+        fprintf(stderr, "Cannot open input file '%s'\n", argv[2]);
+        return -1;
+    }
+
+    if (avformat_find_stream_info(input_ctx, NULL) < 0) {
+        fprintf(stderr, "Cannot find input stream information.\n");
+        return -1;
+    }
+
+    /* find the video stream information */
+    ret = av_find_best_stream(input_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, &decoder, 0);
+    if (ret < 0) {
+        fprintf(stderr, "Cannot find a video stream in the input file\n");
+        return -1;
+    }
+    video_stream = ret;
+
+    for (i = 0;; i++) {
+        const AVCodecHWConfig *config = avcodec_get_hw_config(decoder, i);
+        if (!config) {
+            fprintf(stderr, "Decoder %s does not support device type %s.\n",
+                    decoder->name, av_hwdevice_get_type_name(type));
+            return -1;
+        }
+        if (config->methods & AV_CODEC_HW_CONFIG_METHOD_HW_DEVICE_CTX &&
+            config->device_type == type) {
+            hw_pix_fmt = config->pix_fmt;
+            break;
+        }
+    }
+
+    if (!(decoder_ctx = avcodec_alloc_context3(decoder)))
+        return AVERROR(ENOMEM);
+
+    video = input_ctx->streams[video_stream];
+    if (avcodec_parameters_to_context(decoder_ctx, video->codecpar) < 0)
+        return -1;
+
+    decoder_ctx->get_format  = get_hw_format;
+
+    if (hw_decoder_init(decoder_ctx, type) < 0)
+        return -1;
+
+    if ((ret = avcodec_open2(decoder_ctx, decoder, NULL)) < 0) {
+        fprintf(stderr, "Failed to open codec for stream #%u\n", video_stream);
+        return -1;
+    }
+
+    /* open the file to dump raw data */
+    output_file = fopen(argv[3], "w+b");
+
+    /* actual decoding and dump the raw data */
+    while (ret >= 0) {
+        if ((ret = av_read_frame(input_ctx, &packet)) < 0)
+            break;
+
+        if (video_stream == packet.stream_index)
+            ret = decode_write(decoder_ctx, &packet);
+
+        av_packet_unref(&packet);
+    }
+
+    /* flush the decoder */
+    packet.data = NULL;
+    packet.size = 0;
+    ret = decode_write(decoder_ctx, &packet);
+    av_packet_unref(&packet);
+
+    if (output_file)
+        fclose(output_file);
+    avcodec_free_context(&decoder_ctx);
+    avformat_close_input(&input_ctx);
+    av_buffer_unref(&hw_device_ctx);
+
+    return 0;
+}

externals/ffmpeg/ffmpeg/doc/examples/metadata.c

@@ -0,0 +1,60 @@
+/*
+ * 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;
+    }
+
+    if ((ret = avformat_open_input(&fmt_ctx, argv[1], NULL, NULL)))
+        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;
+    }
+
+    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;
+}

externals/ffmpeg/ffmpeg/doc/examples/muxing.c

@@ -0,0 +1,649 @@
+/*
+ * 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;
+    AVCodecContext *enc;
+
+    /* 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, AVCodecContext *c,
+                       AVStream *st, AVFrame *frame)
+{
+    int ret;
+
+    // send the frame to the encoder
+    ret = avcodec_send_frame(c, frame);
+    if (ret < 0) {
+        fprintf(stderr, "Error sending a frame to the encoder: %s\n",
+                av_err2str(ret));
+        exit(1);
+    }
+
+    while (ret >= 0) {
+        AVPacket pkt = { 0 };
+
+        ret = avcodec_receive_packet(c, &pkt);
+        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
+            break;
+        else if (ret < 0) {
+            fprintf(stderr, "Error encoding a frame: %s\n", av_err2str(ret));
+            exit(1);
+        }
+
+        /* rescale output packet timestamp values from codec to stream timebase */
+        av_packet_rescale_ts(&pkt, c->time_base, st->time_base);
+        pkt.stream_index = st->index;
+
+        /* Write the compressed frame to the media file. */
+        log_packet(fmt_ctx, &pkt);
+        ret = av_interleaved_write_frame(fmt_ctx, &pkt);
+        av_packet_unref(&pkt);
+        if (ret < 0) {
+            fprintf(stderr, "Error while writing output packet: %s\n", av_err2str(ret));
+            exit(1);
+        }
+    }
+
+    return ret == AVERROR_EOF ? 1 : 0;
+}
+
+/* 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, NULL);
+    if (!ost->st) {
+        fprintf(stderr, "Could not allocate stream\n");
+        exit(1);
+    }
+    ost->st->id = oc->nb_streams-1;
+    c = avcodec_alloc_context3(*codec);
+    if (!c) {
+        fprintf(stderr, "Could not alloc an encoding context\n");
+        exit(1);
+    }
+    ost->enc = c;
+
+    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 |= AV_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->enc;
+
+    /* 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 & AV_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);
+
+    /* copy the stream parameters to the muxer */
+    ret = avcodec_parameters_from_context(ost->st->codecpar, c);
+    if (ret < 0) {
+        fprintf(stderr, "Could not copy the stream parameters\n");
+        exit(1);
+    }
+
+    /* 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->enc->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->enc->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;
+    AVFrame *frame;
+    int ret;
+    int dst_nb_samples;
+
+    c = ost->enc;
+
+    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;
+    }
+
+    return write_frame(oc, c, ost->st, frame);
+}
+
+/**************************************************************/
+/* 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, 0);
+    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->enc;
+    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);
+        }
+    }
+
+    /* copy the stream parameters to the muxer */
+    ret = avcodec_parameters_from_context(ost->st->codecpar, c);
+    if (ret < 0) {
+        fprintf(stderr, "Could not copy the stream parameters\n");
+        exit(1);
+    }
+}
+
+/* Prepare a dummy image. */
+static void fill_yuv_image(AVFrame *pict, int frame_index,
+                           int width, int height)
+{
+    int x, y, i;
+
+    i = frame_index;
+
+    /* Y */
+    for (y = 0; y < height; y++)
+        for (x = 0; x < width; x++)
+            pict->data[0][y * pict->linesize[0] + x] = x + y + i * 3;
+
+    /* Cb and Cr */
+    for (y = 0; y < height / 2; y++) {
+        for (x = 0; x < width / 2; x++) {
+            pict->data[1][y * pict->linesize[1] + x] = 128 + y + i * 2;
+            pict->data[2][y * pict->linesize[2] + x] = 64 + x + i * 5;
+        }
+    }
+}
+
+static AVFrame *get_video_frame(OutputStream *ost)
+{
+    AVCodecContext *c = ost->enc;
+
+    /* check if we want to generate more frames */
+    if (av_compare_ts(ost->next_pts, c->time_base,
+                      STREAM_DURATION, (AVRational){ 1, 1 }) > 0)
+        return NULL;
+
+    /* when we pass a frame to the encoder, it may keep a reference to it
+     * internally; make sure we do not overwrite it here */
+    if (av_frame_make_writable(ost->frame) < 0)
+        exit(1);
+
+    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)
+{
+    return write_frame(oc, ost->enc, ost->st, get_video_frame(ost));
+}
+
+static void close_stream(AVFormatContext *oc, OutputStream *ost)
+{
+    avcodec_free_context(&ost->enc);
+    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;
+    int i;
+
+    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];
+    for (i = 2; i+1 < argc; i+=2) {
+        if (!strcmp(argv[i], "-flags") || !strcmp(argv[i], "-fflags"))
+            av_dict_set(&opt, argv[i]+1, argv[i+1], 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.enc->time_base,
+                                            audio_st.next_pts, audio_st.enc->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;
+}

externals/ffmpeg/ffmpeg/doc/examples/qsvdec.c

@@ -0,0 +1,271 @@
+/*
+ * 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 GPU video surfaces.
+ */
+
+#include "config.h"
+
+#include <stdio.h>
+
+#include "libavformat/avformat.h"
+#include "libavformat/avio.h"
+
+#include "libavcodec/avcodec.h"
+
+#include "libavutil/buffer.h"
+#include "libavutil/error.h"
+#include "libavutil/hwcontext.h"
+#include "libavutil/hwcontext_qsv.h"
+#include "libavutil/mem.h"
+
+typedef struct DecodeContext {
+    AVBufferRef *hw_device_ref;
+} DecodeContext;
+
+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) {
+            DecodeContext *decode = avctx->opaque;
+            AVHWFramesContext  *frames_ctx;
+            AVQSVFramesContext *frames_hwctx;
+            int ret;
+
+            /* create a pool of surfaces to be used by the decoder */
+            avctx->hw_frames_ctx = av_hwframe_ctx_alloc(decode->hw_device_ref);
+            if (!avctx->hw_frames_ctx)
+                return AV_PIX_FMT_NONE;
+            frames_ctx   = (AVHWFramesContext*)avctx->hw_frames_ctx->data;
+            frames_hwctx = frames_ctx->hwctx;
+
+            frames_ctx->format            = AV_PIX_FMT_QSV;
+            frames_ctx->sw_format         = avctx->sw_pix_fmt;
+            frames_ctx->width             = FFALIGN(avctx->coded_width,  32);
+            frames_ctx->height            = FFALIGN(avctx->coded_height, 32);
+            frames_ctx->initial_pool_size = 32;
+
+            frames_hwctx->frame_type = MFX_MEMTYPE_VIDEO_MEMORY_DECODER_TARGET;
+
+            ret = av_hwframe_ctx_init(avctx->hw_frames_ctx);
+            if (ret < 0)
+                return AV_PIX_FMT_NONE;
+
+            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, AVFrame *sw_frame,
+                         AVPacket *pkt, AVIOContext *output_ctx)
+{
+    int ret = 0;
+
+    ret = avcodec_send_packet(decoder_ctx, pkt);
+    if (ret < 0) {
+        fprintf(stderr, "Error during decoding\n");
+        return ret;
+    }
+
+    while (ret >= 0) {
+        int i, j;
+
+        ret = avcodec_receive_frame(decoder_ctx, frame);
+        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
+            break;
+        else if (ret < 0) {
+            fprintf(stderr, "Error during decoding\n");
+            return 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. */
+        ret = av_hwframe_transfer_data(sw_frame, frame, 0);
+        if (ret < 0) {
+            fprintf(stderr, "Error transferring the data to system memory\n");
+            goto fail;
+        }
+
+        for (i = 0; i < FF_ARRAY_ELEMS(sw_frame->data) && sw_frame->data[i]; i++)
+            for (j = 0; j < (sw_frame->height >> (i > 0)); j++)
+                avio_write(output_ctx, sw_frame->data[i] + j * sw_frame->linesize[i], sw_frame->width);
+
+fail:
+        av_frame_unref(sw_frame);
+        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, *sw_frame = NULL;
+
+    DecodeContext decode = { NULL };
+
+    AVIOContext *output_ctx = NULL;
+
+    int ret, i;
+
+    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->codecpar->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;
+    }
+
+    /* open the hardware device */
+    ret = av_hwdevice_ctx_create(&decode.hw_device_ref, AV_HWDEVICE_TYPE_QSV,
+                                 "auto", NULL, 0);
+    if (ret < 0) {
+        fprintf(stderr, "Cannot open the hardware device\n");
+        goto finish;
+    }
+
+    /* 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->codecpar->extradata_size) {
+        decoder_ctx->extradata = av_mallocz(video_st->codecpar->extradata_size +
+                                            AV_INPUT_BUFFER_PADDING_SIZE);
+        if (!decoder_ctx->extradata) {
+            ret = AVERROR(ENOMEM);
+            goto finish;
+        }
+        memcpy(decoder_ctx->extradata, video_st->codecpar->extradata,
+               video_st->codecpar->extradata_size);
+        decoder_ctx->extradata_size = video_st->codecpar->extradata_size;
+    }
+
+    decoder_ctx->opaque      = &decode;
+    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();
+    sw_frame = av_frame_alloc();
+    if (!frame || !sw_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, sw_frame, &pkt, output_ctx);
+
+        av_packet_unref(&pkt);
+    }
+
+    /* flush the decoder */
+    pkt.data = NULL;
+    pkt.size = 0;
+    ret = decode_packet(&decode, decoder_ctx, frame, sw_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);
+    av_frame_free(&sw_frame);
+
+    avcodec_free_context(&decoder_ctx);
+
+    av_buffer_unref(&decode.hw_device_ref);
+
+    avio_close(output_ctx);
+
+    return ret;
+}

externals/ffmpeg/ffmpeg/doc/examples/remuxing.c

@@ -0,0 +1,191 @@
+/*
+ * 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;
+    int stream_index = 0;
+    int *stream_mapping = NULL;
+    int stream_mapping_size = 0;
+
+    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];
+
+    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;
+    }
+
+    stream_mapping_size = ifmt_ctx->nb_streams;
+    stream_mapping = av_mallocz_array(stream_mapping_size, sizeof(*stream_mapping));
+    if (!stream_mapping) {
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+
+    ofmt = ofmt_ctx->oformat;
+
+    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
+        AVStream *out_stream;
+        AVStream *in_stream = ifmt_ctx->streams[i];
+        AVCodecParameters *in_codecpar = in_stream->codecpar;
+
+        if (in_codecpar->codec_type != AVMEDIA_TYPE_AUDIO &&
+            in_codecpar->codec_type != AVMEDIA_TYPE_VIDEO &&
+            in_codecpar->codec_type != AVMEDIA_TYPE_SUBTITLE) {
+            stream_mapping[i] = -1;
+            continue;
+        }
+
+        stream_mapping[i] = stream_index++;
+
+        out_stream = avformat_new_stream(ofmt_ctx, NULL);
+        if (!out_stream) {
+            fprintf(stderr, "Failed allocating output stream\n");
+            ret = AVERROR_UNKNOWN;
+            goto end;
+        }
+
+        ret = avcodec_parameters_copy(out_stream->codecpar, in_codecpar);
+        if (ret < 0) {
+            fprintf(stderr, "Failed to copy codec parameters\n");
+            goto end;
+        }
+        out_stream->codecpar->codec_tag = 0;
+    }
+    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];
+        if (pkt.stream_index >= stream_mapping_size ||
+            stream_mapping[pkt.stream_index] < 0) {
+            av_packet_unref(&pkt);
+            continue;
+        }
+
+        pkt.stream_index = stream_mapping[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_packet_unref(&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);
+
+    av_freep(&stream_mapping);
+
+    if (ret < 0 && ret != AVERROR_EOF) {
+        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
+        return 1;
+    }
+
+    return 0;
+}

externals/ffmpeg/ffmpeg/doc/examples/resampling_audio.c

@@ -0,0 +1,214 @@
+/*
+ * 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;
+}

externals/ffmpeg/ffmpeg/doc/examples/scaling_video.c

@@ -0,0 +1,140 @@
+/*
+ * 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;
+}

externals/ffmpeg/ffmpeg/doc/examples/transcode_aac.c

@@ -0,0 +1,893 @@
+/*
+ * Copyright (c) 2013-2018 Andreas Unterweger
+ *
+ * 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.
+ * Formats other than MP4 are supported based on the output file extension.
+ * @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 bit/s */
+#define OUTPUT_BIT_RATE 96000
+/* The number of output channels */
+#define OUTPUT_CHANNELS 2
+
+/**
+ * Open an input file and the required decoder.
+ * @param      filename             File to be opened
+ * @param[out] input_format_context Format context of opened file
+ * @param[out] input_codec_context  Codec context of opened file
+ * @return Error code (0 if successful)
+ */
+static int open_input_file(const char *filename,
+                           AVFormatContext **input_format_context,
+                           AVCodecContext **input_codec_context)
+{
+    AVCodecContext *avctx;
+    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, av_err2str(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",
+                av_err2str(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]->codecpar->codec_id))) {
+        fprintf(stderr, "Could not find input codec\n");
+        avformat_close_input(input_format_context);
+        return AVERROR_EXIT;
+    }
+
+    /* Allocate a new decoding context. */
+    avctx = avcodec_alloc_context3(input_codec);
+    if (!avctx) {
+        fprintf(stderr, "Could not allocate a decoding context\n");
+        avformat_close_input(input_format_context);
+        return AVERROR(ENOMEM);
+    }
+
+    /* Initialize the stream parameters with demuxer information. */
+    error = avcodec_parameters_to_context(avctx, (*input_format_context)->streams[0]->codecpar);
+    if (error < 0) {
+        avformat_close_input(input_format_context);
+        avcodec_free_context(&avctx);
+        return error;
+    }
+
+    /* Open the decoder for the audio stream to use it later. */
+    if ((error = avcodec_open2(avctx, input_codec, NULL)) < 0) {
+        fprintf(stderr, "Could not open input codec (error '%s')\n",
+                av_err2str(error));
+        avcodec_free_context(&avctx);
+        avformat_close_input(input_format_context);
+        return error;
+    }
+
+    /* Save the decoder context for easier access later. */
+    *input_codec_context = avctx;
+
+    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.
+ * @param      filename              File to be opened
+ * @param      input_codec_context   Codec context of input file
+ * @param[out] output_format_context Format context of output file
+ * @param[out] output_codec_context  Codec context of output file
+ * @return Error code (0 if successful)
+ */
+static int open_output_file(const char *filename,
+                            AVCodecContext *input_codec_context,
+                            AVFormatContext **output_format_context,
+                            AVCodecContext **output_codec_context)
+{
+    AVCodecContext *avctx          = NULL;
+    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, av_err2str(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;
+    }
+
+    if (!((*output_format_context)->url = av_strdup(filename))) {
+        fprintf(stderr, "Could not allocate url.\n");
+        error = AVERROR(ENOMEM);
+        goto cleanup;
+    }
+
+    /* 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, NULL))) {
+        fprintf(stderr, "Could not create new stream\n");
+        error = AVERROR(ENOMEM);
+        goto cleanup;
+    }
+
+    avctx = avcodec_alloc_context3(output_codec);
+    if (!avctx) {
+        fprintf(stderr, "Could not allocate an encoding context\n");
+        error = AVERROR(ENOMEM);
+        goto cleanup;
+    }
+
+    /* Set the basic encoder parameters.
+     * The input file's sample rate is used to avoid a sample rate conversion. */
+    avctx->channels       = OUTPUT_CHANNELS;
+    avctx->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
+    avctx->sample_rate    = input_codec_context->sample_rate;
+    avctx->sample_fmt     = output_codec->sample_fmts[0];
+    avctx->bit_rate       = OUTPUT_BIT_RATE;
+
+    /* Allow the use of the experimental AAC encoder. */
+    avctx->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)
+        avctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
+
+    /* Open the encoder for the audio stream to use it later. */
+    if ((error = avcodec_open2(avctx, output_codec, NULL)) < 0) {
+        fprintf(stderr, "Could not open output codec (error '%s')\n",
+                av_err2str(error));
+        goto cleanup;
+    }
+
+    error = avcodec_parameters_from_context(stream->codecpar, avctx);
+    if (error < 0) {
+        fprintf(stderr, "Could not initialize stream parameters\n");
+        goto cleanup;
+    }
+
+    /* Save the encoder context for easier access later. */
+    *output_codec_context = avctx;
+
+    return 0;
+
+cleanup:
+    avcodec_free_context(&avctx);
+    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.
+ * @param[out] packet Packet to be initialized
+ * @return Error code (0 if successful)
+ */
+static int init_packet(AVPacket **packet)
+{
+    if (!(*packet = av_packet_alloc())) {
+        fprintf(stderr, "Could not allocate packet\n");
+        return AVERROR(ENOMEM);
+    }
+    return 0;
+}
+
+/**
+ * Initialize one audio frame for reading from the input file.
+ * @param[out] frame Frame to be initialized
+ * @return Error code (0 if successful)
+ */
+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.
+ * @param      input_codec_context  Codec context of the input file
+ * @param      output_codec_context Codec context of the output file
+ * @param[out] resample_context     Resample context for the required conversion
+ * @return Error code (0 if successful)
+ */
+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.
+ * @param[out] fifo                 Sample buffer
+ * @param      output_codec_context Codec context of the output file
+ * @return Error code (0 if successful)
+ */
+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.
+ * @param output_format_context Format context of the output file
+ * @return Error code (0 if successful)
+ */
+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",
+                av_err2str(error));
+        return error;
+    }
+    return 0;
+}
+
+/**
+ * Decode one audio frame from the input file.
+ * @param      frame                Audio frame to be decoded
+ * @param      input_format_context Format context of the input file
+ * @param      input_codec_context  Codec context of the input file
+ * @param[out] data_present         Indicates whether data has been decoded
+ * @param[out] finished             Indicates whether the end of file has
+ *                                  been reached and all data has been
+ *                                  decoded. If this flag is false, there
+ *                                  is more data to be decoded, i.e., this
+ *                                  function has to be called again.
+ * @return Error code (0 if successful)
+ */
+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;
+
+    error = init_packet(&input_packet);
+    if (error < 0)
+        return error;
+
+    /* 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",
+                    av_err2str(error));
+            goto cleanup;
+        }
+    }
+
+    /* Send the audio frame stored in the temporary packet to the decoder.
+     * The input audio stream decoder is used to do this. */
+    if ((error = avcodec_send_packet(input_codec_context, input_packet)) < 0) {
+        fprintf(stderr, "Could not send packet for decoding (error '%s')\n",
+                av_err2str(error));
+        goto cleanup;
+    }
+
+    /* Receive one frame from the decoder. */
+    error = avcodec_receive_frame(input_codec_context, frame);
+    /* If the decoder asks for more data to be able to decode a frame,
+     * return indicating that no data is present. */
+    if (error == AVERROR(EAGAIN)) {
+        error = 0;
+        goto cleanup;
+    /* If the end of the input file is reached, stop decoding. */
+    } else if (error == AVERROR_EOF) {
+        *finished = 1;
+        error = 0;
+        goto cleanup;
+    } else if (error < 0) {
+        fprintf(stderr, "Could not decode frame (error '%s')\n",
+                av_err2str(error));
+        goto cleanup;
+    /* Default case: Return decoded data. */
+    } else {
+        *data_present = 1;
+        goto cleanup;
+    }
+
+cleanup:
+    av_packet_free(&input_packet);
+    return error;
+}
+
+/**
+ * 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.
+ * @param[out] converted_input_samples Array of converted samples. The
+ *                                     dimensions are reference, channel
+ *                                     (for multi-channel audio), sample.
+ * @param      output_codec_context    Codec context of the output file
+ * @param      frame_size              Number of samples to be converted in
+ *                                     each round
+ * @return Error code (0 if successful)
+ */
+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",
+                av_err2str(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.
+ * @param      input_data       Samples to be decoded. The dimensions are
+ *                              channel (for multi-channel audio), sample.
+ * @param[out] converted_data   Converted samples. The dimensions are channel
+ *                              (for multi-channel audio), sample.
+ * @param      frame_size       Number of samples to be converted
+ * @param      resample_context Resample context for the conversion
+ * @return Error code (0 if successful)
+ */
+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",
+                av_err2str(error));
+        return error;
+    }
+
+    return 0;
+}
+
+/**
+ * Add converted input audio samples to the FIFO buffer for later processing.
+ * @param fifo                    Buffer to add the samples to
+ * @param converted_input_samples Samples to be added. The dimensions are channel
+ *                                (for multi-channel audio), sample.
+ * @param frame_size              Number of samples to be converted
+ * @return Error code (0 if successful)
+ */
+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, decode, convert and store
+ * it in the FIFO buffer.
+ * @param      fifo                 Buffer used for temporary storage
+ * @param      input_format_context Format context of the input file
+ * @param      input_codec_context  Codec context of the input file
+ * @param      output_codec_context Codec context of the output file
+ * @param      resampler_context    Resample context for the conversion
+ * @param[out] finished             Indicates whether the end of file has
+ *                                  been reached and all data has been
+ *                                  decoded. If this flag is false,
+ *                                  there is more data to be decoded,
+ *                                  i.e., this function has to be called
+ *                                  again.
+ * @return Error code (0 if successful)
+ */
+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 = 0;
+    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) {
+        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.
+ * @param[out] frame                Frame to be initialized
+ * @param      output_codec_context Codec context of the output file
+ * @param      frame_size           Size of the frame
+ * @return Error code (0 if successful)
+ */
+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 not allocate output frame samples (error '%s')\n",
+                av_err2str(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.
+ * @param      frame                 Samples to be encoded
+ * @param      output_format_context Format context of the output file
+ * @param      output_codec_context  Codec context of the output file
+ * @param[out] data_present          Indicates whether data has been
+ *                                   encoded
+ * @return Error code (0 if successful)
+ */
+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;
+
+    error = init_packet(&output_packet);
+    if (error < 0)
+        return error;
+
+    /* Set a timestamp based on the sample rate for the container. */
+    if (frame) {
+        frame->pts = pts;
+        pts += frame->nb_samples;
+    }
+
+    /* Send the audio frame stored in the temporary packet to the encoder.
+     * The output audio stream encoder is used to do this. */
+    error = avcodec_send_frame(output_codec_context, frame);
+    /* The encoder signals that it has nothing more to encode. */
+    if (error == AVERROR_EOF) {
+        error = 0;
+        goto cleanup;
+    } else if (error < 0) {
+        fprintf(stderr, "Could not send packet for encoding (error '%s')\n",
+                av_err2str(error));
+        goto cleanup;
+    }
+
+    /* Receive one encoded frame from the encoder. */
+    error = avcodec_receive_packet(output_codec_context, output_packet);
+    /* If the encoder asks for more data to be able to provide an
+     * encoded frame, return indicating that no data is present. */
+    if (error == AVERROR(EAGAIN)) {
+        error = 0;
+        goto cleanup;
+    /* If the last frame has been encoded, stop encoding. */
+    } else if (error == AVERROR_EOF) {
+        error = 0;
+        goto cleanup;
+    } else if (error < 0) {
+        fprintf(stderr, "Could not encode frame (error '%s')\n",
+                av_err2str(error));
+        goto cleanup;
+    /* Default case: Return encoded data. */
+    } else {
+        *data_present = 1;
+    }
+
+    /* Write one audio frame from the temporary packet to the output file. */
+    if (*data_present &&
+        (error = av_write_frame(output_format_context, output_packet)) < 0) {
+        fprintf(stderr, "Could not write frame (error '%s')\n",
+                av_err2str(error));
+        goto cleanup;
+    }
+
+cleanup:
+    av_packet_free(&output_packet);
+    return error;
+}
+
+/**
+ * Load one audio frame from the FIFO buffer, encode and write it to the
+ * output file.
+ * @param fifo                  Buffer used for temporary storage
+ * @param output_format_context Format context of the output file
+ * @param output_codec_context  Codec context of the output file
+ * @return Error code (0 if successful)
+ */
+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.
+ * @param output_format_context Format context of the output file
+ * @return Error code (0 if successful)
+ */
+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",
+                av_err2str(error));
+        return error;
+    }
+    return 0;
+}
+
+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);
+    }
+
+    /* 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 {
+                data_written = 0;
+                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_free_context(&output_codec_context);
+    if (output_format_context) {
+        avio_closep(&output_format_context->pb);
+        avformat_free_context(output_format_context);
+    }
+    if (input_codec_context)
+        avcodec_free_context(&input_codec_context);
+    if (input_format_context)
+        avformat_close_input(&input_format_context);
+
+    return ret;
+}

externals/ffmpeg/ffmpeg/doc/examples/transcoding.c

@@ -0,0 +1,621 @@
+/*
+ * 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/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;
+
+    AVPacket *enc_pkt;
+    AVFrame *filtered_frame;
+} FilteringContext;
+static FilteringContext *filter_ctx;
+
+typedef struct StreamContext {
+    AVCodecContext *dec_ctx;
+    AVCodecContext *enc_ctx;
+
+    AVFrame *dec_frame;
+} StreamContext;
+static StreamContext *stream_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;
+    }
+
+    stream_ctx = av_mallocz_array(ifmt_ctx->nb_streams, sizeof(*stream_ctx));
+    if (!stream_ctx)
+        return AVERROR(ENOMEM);
+
+    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
+        AVStream *stream = ifmt_ctx->streams[i];
+        AVCodec *dec = avcodec_find_decoder(stream->codecpar->codec_id);
+        AVCodecContext *codec_ctx;
+        if (!dec) {
+            av_log(NULL, AV_LOG_ERROR, "Failed to find decoder for stream #%u\n", i);
+            return AVERROR_DECODER_NOT_FOUND;
+        }
+        codec_ctx = avcodec_alloc_context3(dec);
+        if (!codec_ctx) {
+            av_log(NULL, AV_LOG_ERROR, "Failed to allocate the decoder context for stream #%u\n", i);
+            return AVERROR(ENOMEM);
+        }
+        ret = avcodec_parameters_to_context(codec_ctx, stream->codecpar);
+        if (ret < 0) {
+            av_log(NULL, AV_LOG_ERROR, "Failed to copy decoder parameters to input decoder context "
+                   "for stream #%u\n", i);
+            return ret;
+        }
+        /* Reencode video & audio and remux subtitles etc. */
+        if (codec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
+                || codec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
+            if (codec_ctx->codec_type == AVMEDIA_TYPE_VIDEO)
+                codec_ctx->framerate = av_guess_frame_rate(ifmt_ctx, stream, NULL);
+            /* Open decoder */
+            ret = avcodec_open2(codec_ctx, dec, NULL);
+            if (ret < 0) {
+                av_log(NULL, AV_LOG_ERROR, "Failed to open decoder for stream #%u\n", i);
+                return ret;
+            }
+        }
+        stream_ctx[i].dec_ctx = codec_ctx;
+
+        stream_ctx[i].dec_frame = av_frame_alloc();
+        if (!stream_ctx[i].dec_frame)
+            return AVERROR(ENOMEM);
+    }
+
+    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 = stream_ctx[i].dec_ctx;
+
+        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;
+            }
+            enc_ctx = avcodec_alloc_context3(encoder);
+            if (!enc_ctx) {
+                av_log(NULL, AV_LOG_FATAL, "Failed to allocate the encoder context\n");
+                return AVERROR(ENOMEM);
+            }
+
+            /* 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 */
+                if (encoder->pix_fmts)
+                    enc_ctx->pix_fmt = encoder->pix_fmts[0];
+                else
+                    enc_ctx->pix_fmt = dec_ctx->pix_fmt;
+                /* video time_base can be set to whatever is handy and supported by encoder */
+                enc_ctx->time_base = av_inv_q(dec_ctx->framerate);
+            } 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};
+            }
+
+            if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
+                enc_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
+
+            /* 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;
+            }
+            ret = avcodec_parameters_from_context(out_stream->codecpar, enc_ctx);
+            if (ret < 0) {
+                av_log(NULL, AV_LOG_ERROR, "Failed to copy encoder parameters to output stream #%u\n", i);
+                return ret;
+            }
+
+            out_stream->time_base = enc_ctx->time_base;
+            stream_ctx[i].enc_ctx = enc_ctx;
+        } 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_parameters_copy(out_stream->codecpar, in_stream->codecpar);
+            if (ret < 0) {
+                av_log(NULL, AV_LOG_ERROR, "Copying parameters for stream #%u failed\n", i);
+                return ret;
+            }
+            out_stream->time_base = in_stream->time_base;
+        }
+
+    }
+    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;
+    const AVFilter *buffersrc = NULL;
+    const 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]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO
+                || ifmt_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_VIDEO))
+            continue;
+
+
+        if (ifmt_ctx->streams[i]->codecpar->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], stream_ctx[i].dec_ctx,
+                stream_ctx[i].enc_ctx, filter_spec);
+        if (ret)
+            return ret;
+
+        filter_ctx[i].enc_pkt = av_packet_alloc();
+        if (!filter_ctx[i].enc_pkt)
+            return AVERROR(ENOMEM);
+
+        filter_ctx[i].filtered_frame = av_frame_alloc();
+        if (!filter_ctx[i].filtered_frame)
+            return AVERROR(ENOMEM);
+    }
+    return 0;
+}
+
+static int encode_write_frame(unsigned int stream_index, int flush)
+{
+    StreamContext *stream = &stream_ctx[stream_index];
+    FilteringContext *filter = &filter_ctx[stream_index];
+    AVFrame *filt_frame = flush ? NULL : filter->filtered_frame;
+    AVPacket *enc_pkt = filter->enc_pkt;
+    int ret;
+
+    av_log(NULL, AV_LOG_INFO, "Encoding frame\n");
+    /* encode filtered frame */
+    av_packet_unref(enc_pkt);
+
+    ret = avcodec_send_frame(stream->enc_ctx, filt_frame);
+
+    if (ret < 0)
+        return ret;
+
+    while (ret >= 0) {
+        ret = avcodec_receive_packet(stream->enc_ctx, enc_pkt);
+
+        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
+            return 0;
+
+        /* prepare packet for muxing */
+        enc_pkt->stream_index = stream_index;
+        av_packet_rescale_ts(enc_pkt,
+                             stream->enc_ctx->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)
+{
+    FilteringContext *filter = &filter_ctx[stream_index];
+    int ret;
+
+    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->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) {
+        av_log(NULL, AV_LOG_INFO, "Pulling filtered frame from filters\n");
+        ret = av_buffersink_get_frame(filter->buffersink_ctx,
+                                      filter->filtered_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;
+            break;
+        }
+
+        filter->filtered_frame->pict_type = AV_PICTURE_TYPE_NONE;
+        ret = encode_write_frame(stream_index, 0);
+        av_frame_unref(filter->filtered_frame);
+        if (ret < 0)
+            break;
+    }
+
+    return ret;
+}
+
+static int flush_encoder(unsigned int stream_index)
+{
+    if (!(stream_ctx[stream_index].enc_ctx->codec->capabilities &
+                AV_CODEC_CAP_DELAY))
+        return 0;
+
+    av_log(NULL, AV_LOG_INFO, "Flushing stream #%u encoder\n", stream_index);
+    return encode_write_frame(stream_index, 1);
+}
+
+int main(int argc, char **argv)
+{
+    int ret;
+    AVPacket *packet = NULL;
+    unsigned int stream_index;
+    unsigned int i;
+
+    if (argc != 3) {
+        av_log(NULL, AV_LOG_ERROR, "Usage: %s <input file> <output file>\n", argv[0]);
+        return 1;
+    }
+
+    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;
+    if (!(packet = av_packet_alloc()))
+        goto end;
+
+    /* read all packets */
+    while (1) {
+        if ((ret = av_read_frame(ifmt_ctx, packet)) < 0)
+            break;
+        stream_index = packet->stream_index;
+        av_log(NULL, AV_LOG_DEBUG, "Demuxer gave frame of stream_index %u\n",
+                stream_index);
+
+        if (filter_ctx[stream_index].filter_graph) {
+            StreamContext *stream = &stream_ctx[stream_index];
+
+            av_log(NULL, AV_LOG_DEBUG, "Going to reencode&filter the frame\n");
+
+            av_packet_rescale_ts(packet,
+                                 ifmt_ctx->streams[stream_index]->time_base,
+                                 stream->dec_ctx->time_base);
+            ret = avcodec_send_packet(stream->dec_ctx, packet);
+            if (ret < 0) {
+                av_log(NULL, AV_LOG_ERROR, "Decoding failed\n");
+                break;
+            }
+
+            while (ret >= 0) {
+                ret = avcodec_receive_frame(stream->dec_ctx, stream->dec_frame);
+                if (ret == AVERROR_EOF || ret == AVERROR(EAGAIN))
+                    break;
+                else if (ret < 0)
+                    goto end;
+
+                stream->dec_frame->pts = stream->dec_frame->best_effort_timestamp;
+                ret = filter_encode_write_frame(stream->dec_frame, stream_index);
+                if (ret < 0)
+                    goto end;
+            }
+        } 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_packet_unref(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_packet_free(&packet);
+    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
+        avcodec_free_context(&stream_ctx[i].dec_ctx);
+        if (ofmt_ctx && ofmt_ctx->nb_streams > i && ofmt_ctx->streams[i] && stream_ctx[i].enc_ctx)
+            avcodec_free_context(&stream_ctx[i].enc_ctx);
+        if (filter_ctx && filter_ctx[i].filter_graph) {
+            avfilter_graph_free(&filter_ctx[i].filter_graph);
+            av_packet_free(&filter_ctx[i].enc_pkt);
+            av_frame_free(&filter_ctx[i].filtered_frame);
+        }
+
+        av_frame_free(&stream_ctx[i].dec_frame);
+    }
+    av_free(filter_ctx);
+    av_free(stream_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;
+}

externals/ffmpeg/ffmpeg/doc/examples/vaapi_encode.c

@@ -0,0 +1,224 @@
+/*
+ * Video Acceleration API (video encoding) encode sample
+ *
+ * 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 VAAPI-accelerated encoding example.
+ *
+ * @example vaapi_encode.c
+ * This example shows how to do VAAPI-accelerated encoding. now only support NV12
+ * raw file, usage like: vaapi_encode 1920 1080 input.yuv output.h264
+ *
+ */
+
+#include <stdio.h>
+#include <string.h>
+#include <errno.h>
+
+#include <libavcodec/avcodec.h>
+#include <libavutil/pixdesc.h>
+#include <libavutil/hwcontext.h>
+
+static int width, height;
+static AVBufferRef *hw_device_ctx = NULL;
+
+static int set_hwframe_ctx(AVCodecContext *ctx, AVBufferRef *hw_device_ctx)
+{
+    AVBufferRef *hw_frames_ref;
+    AVHWFramesContext *frames_ctx = NULL;
+    int err = 0;
+
+    if (!(hw_frames_ref = av_hwframe_ctx_alloc(hw_device_ctx))) {
+        fprintf(stderr, "Failed to create VAAPI frame context.\n");
+        return -1;
+    }
+    frames_ctx = (AVHWFramesContext *)(hw_frames_ref->data);
+    frames_ctx->format    = AV_PIX_FMT_VAAPI;
+    frames_ctx->sw_format = AV_PIX_FMT_NV12;
+    frames_ctx->width     = width;
+    frames_ctx->height    = height;
+    frames_ctx->initial_pool_size = 20;
+    if ((err = av_hwframe_ctx_init(hw_frames_ref)) < 0) {
+        fprintf(stderr, "Failed to initialize VAAPI frame context."
+                "Error code: %s\n",av_err2str(err));
+        av_buffer_unref(&hw_frames_ref);
+        return err;
+    }
+    ctx->hw_frames_ctx = av_buffer_ref(hw_frames_ref);
+    if (!ctx->hw_frames_ctx)
+        err = AVERROR(ENOMEM);
+
+    av_buffer_unref(&hw_frames_ref);
+    return err;
+}
+
+static int encode_write(AVCodecContext *avctx, AVFrame *frame, FILE *fout)
+{
+    int ret = 0;
+    AVPacket *enc_pkt;
+
+    if (!(enc_pkt = av_packet_alloc()))
+        return AVERROR(ENOMEM);
+
+    if ((ret = avcodec_send_frame(avctx, frame)) < 0) {
+        fprintf(stderr, "Error code: %s\n", av_err2str(ret));
+        goto end;
+    }
+    while (1) {
+        ret = avcodec_receive_packet(avctx, enc_pkt);
+        if (ret)
+            break;
+
+        enc_pkt->stream_index = 0;
+        ret = fwrite(enc_pkt->data, enc_pkt->size, 1, fout);
+        av_packet_unref(enc_pkt);
+    }
+
+end:
+    av_packet_free(&enc_pkt);
+    ret = ((ret == AVERROR(EAGAIN)) ? 0 : -1);
+    return ret;
+}
+
+int main(int argc, char *argv[])
+{
+    int size, err;
+    FILE *fin = NULL, *fout = NULL;
+    AVFrame *sw_frame = NULL, *hw_frame = NULL;
+    AVCodecContext *avctx = NULL;
+    AVCodec *codec = NULL;
+    const char *enc_name = "h264_vaapi";
+
+    if (argc < 5) {
+        fprintf(stderr, "Usage: %s <width> <height> <input file> <output file>\n", argv[0]);
+        return -1;
+    }
+
+    width  = atoi(argv[1]);
+    height = atoi(argv[2]);
+    size   = width * height;
+
+    if (!(fin = fopen(argv[3], "r"))) {
+        fprintf(stderr, "Fail to open input file : %s\n", strerror(errno));
+        return -1;
+    }
+    if (!(fout = fopen(argv[4], "w+b"))) {
+        fprintf(stderr, "Fail to open output file : %s\n", strerror(errno));
+        err = -1;
+        goto close;
+    }
+
+    err = av_hwdevice_ctx_create(&hw_device_ctx, AV_HWDEVICE_TYPE_VAAPI,
+                                 NULL, NULL, 0);
+    if (err < 0) {
+        fprintf(stderr, "Failed to create a VAAPI device. Error code: %s\n", av_err2str(err));
+        goto close;
+    }
+
+    if (!(codec = avcodec_find_encoder_by_name(enc_name))) {
+        fprintf(stderr, "Could not find encoder.\n");
+        err = -1;
+        goto close;
+    }
+
+    if (!(avctx = avcodec_alloc_context3(codec))) {
+        err = AVERROR(ENOMEM);
+        goto close;
+    }
+
+    avctx->width     = width;
+    avctx->height    = height;
+    avctx->time_base = (AVRational){1, 25};
+    avctx->framerate = (AVRational){25, 1};
+    avctx->sample_aspect_ratio = (AVRational){1, 1};
+    avctx->pix_fmt   = AV_PIX_FMT_VAAPI;
+
+    /* set hw_frames_ctx for encoder's AVCodecContext */
+    if ((err = set_hwframe_ctx(avctx, hw_device_ctx)) < 0) {
+        fprintf(stderr, "Failed to set hwframe context.\n");
+        goto close;
+    }
+
+    if ((err = avcodec_open2(avctx, codec, NULL)) < 0) {
+        fprintf(stderr, "Cannot open video encoder codec. Error code: %s\n", av_err2str(err));
+        goto close;
+    }
+
+    while (1) {
+        if (!(sw_frame = av_frame_alloc())) {
+            err = AVERROR(ENOMEM);
+            goto close;
+        }
+        /* read data into software frame, and transfer them into hw frame */
+        sw_frame->width  = width;
+        sw_frame->height = height;
+        sw_frame->format = AV_PIX_FMT_NV12;
+        if ((err = av_frame_get_buffer(sw_frame, 0)) < 0)
+            goto close;
+        if ((err = fread((uint8_t*)(sw_frame->data[0]), size, 1, fin)) <= 0)
+            break;
+        if ((err = fread((uint8_t*)(sw_frame->data[1]), size/2, 1, fin)) <= 0)
+            break;
+
+        if (!(hw_frame = av_frame_alloc())) {
+            err = AVERROR(ENOMEM);
+            goto close;
+        }
+        if ((err = av_hwframe_get_buffer(avctx->hw_frames_ctx, hw_frame, 0)) < 0) {
+            fprintf(stderr, "Error code: %s.\n", av_err2str(err));
+            goto close;
+        }
+        if (!hw_frame->hw_frames_ctx) {
+            err = AVERROR(ENOMEM);
+            goto close;
+        }
+        if ((err = av_hwframe_transfer_data(hw_frame, sw_frame, 0)) < 0) {
+            fprintf(stderr, "Error while transferring frame data to surface."
+                    "Error code: %s.\n", av_err2str(err));
+            goto close;
+        }
+
+        if ((err = (encode_write(avctx, hw_frame, fout))) < 0) {
+            fprintf(stderr, "Failed to encode.\n");
+            goto close;
+        }
+        av_frame_free(&hw_frame);
+        av_frame_free(&sw_frame);
+    }
+
+    /* flush encoder */
+    err = encode_write(avctx, NULL, fout);
+    if (err == AVERROR_EOF)
+        err = 0;
+
+close:
+    if (fin)
+        fclose(fin);
+    if (fout)
+        fclose(fout);
+    av_frame_free(&sw_frame);
+    av_frame_free(&hw_frame);
+    avcodec_free_context(&avctx);
+    av_buffer_unref(&hw_device_ctx);
+
+    return err;
+}

externals/ffmpeg/ffmpeg/doc/examples/vaapi_transcode.c

@@ -0,0 +1,308 @@
+/*
+ * Video Acceleration API (video transcoding) transcode sample
+ *
+ * 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 VAAPI-accelerated transcoding example.
+ *
+ * @example vaapi_transcode.c
+ * This example shows how to do VAAPI-accelerated transcoding.
+ * Usage: vaapi_transcode input_stream codec output_stream
+ * e.g: - vaapi_transcode input.mp4 h264_vaapi output_h264.mp4
+ *      - vaapi_transcode input.mp4 vp9_vaapi output_vp9.ivf
+ */
+
+#include <stdio.h>
+#include <errno.h>
+
+#include <libavutil/hwcontext.h>
+#include <libavcodec/avcodec.h>
+#include <libavformat/avformat.h>
+
+static AVFormatContext *ifmt_ctx = NULL, *ofmt_ctx = NULL;
+static AVBufferRef *hw_device_ctx = NULL;
+static AVCodecContext *decoder_ctx = NULL, *encoder_ctx = NULL;
+static int video_stream = -1;
+static AVStream *ost;
+static int initialized = 0;
+
+static enum AVPixelFormat get_vaapi_format(AVCodecContext *ctx,
+                                           const enum AVPixelFormat *pix_fmts)
+{
+    const enum AVPixelFormat *p;
+
+    for (p = pix_fmts; *p != AV_PIX_FMT_NONE; p++) {
+        if (*p == AV_PIX_FMT_VAAPI)
+            return *p;
+    }
+
+    fprintf(stderr, "Unable to decode this file using VA-API.\n");
+    return AV_PIX_FMT_NONE;
+}
+
+static int open_input_file(const char *filename)
+{
+    int ret;
+    AVCodec *decoder = NULL;
+    AVStream *video = NULL;
+
+    if ((ret = avformat_open_input(&ifmt_ctx, filename, NULL, NULL)) < 0) {
+        fprintf(stderr, "Cannot open input file '%s', Error code: %s\n",
+                filename, av_err2str(ret));
+        return ret;
+    }
+
+    if ((ret = avformat_find_stream_info(ifmt_ctx, NULL)) < 0) {
+        fprintf(stderr, "Cannot find input stream information. Error code: %s\n",
+                av_err2str(ret));
+        return ret;
+    }
+
+    ret = av_find_best_stream(ifmt_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, &decoder, 0);
+    if (ret < 0) {
+        fprintf(stderr, "Cannot find a video stream in the input file. "
+                "Error code: %s\n", av_err2str(ret));
+        return ret;
+    }
+    video_stream = ret;
+
+    if (!(decoder_ctx = avcodec_alloc_context3(decoder)))
+        return AVERROR(ENOMEM);
+
+    video = ifmt_ctx->streams[video_stream];
+    if ((ret = avcodec_parameters_to_context(decoder_ctx, video->codecpar)) < 0) {
+        fprintf(stderr, "avcodec_parameters_to_context error. Error code: %s\n",
+                av_err2str(ret));
+        return ret;
+    }
+
+    decoder_ctx->hw_device_ctx = av_buffer_ref(hw_device_ctx);
+    if (!decoder_ctx->hw_device_ctx) {
+        fprintf(stderr, "A hardware device reference create failed.\n");
+        return AVERROR(ENOMEM);
+    }
+    decoder_ctx->get_format    = get_vaapi_format;
+
+    if ((ret = avcodec_open2(decoder_ctx, decoder, NULL)) < 0)
+        fprintf(stderr, "Failed to open codec for decoding. Error code: %s\n",
+                av_err2str(ret));
+
+    return ret;
+}
+
+static int encode_write(AVPacket *enc_pkt, AVFrame *frame)
+{
+    int ret = 0;
+
+    av_packet_unref(enc_pkt);
+
+    if ((ret = avcodec_send_frame(encoder_ctx, frame)) < 0) {
+        fprintf(stderr, "Error during encoding. Error code: %s\n", av_err2str(ret));
+        goto end;
+    }
+    while (1) {
+        ret = avcodec_receive_packet(encoder_ctx, enc_pkt);
+        if (ret)
+            break;
+
+        enc_pkt->stream_index = 0;
+        av_packet_rescale_ts(enc_pkt, ifmt_ctx->streams[video_stream]->time_base,
+                             ofmt_ctx->streams[0]->time_base);
+        ret = av_interleaved_write_frame(ofmt_ctx, enc_pkt);
+        if (ret < 0) {
+            fprintf(stderr, "Error during writing data to output file. "
+                    "Error code: %s\n", av_err2str(ret));
+            return -1;
+        }
+    }
+
+end:
+    if (ret == AVERROR_EOF)
+        return 0;
+    ret = ((ret == AVERROR(EAGAIN)) ? 0:-1);
+    return ret;
+}
+
+static int dec_enc(AVPacket *pkt, AVCodec *enc_codec)
+{
+    AVFrame *frame;
+    int ret = 0;
+
+    ret = avcodec_send_packet(decoder_ctx, pkt);
+    if (ret < 0) {
+        fprintf(stderr, "Error during decoding. Error code: %s\n", av_err2str(ret));
+        return ret;
+    }
+
+    while (ret >= 0) {
+        if (!(frame = av_frame_alloc()))
+            return AVERROR(ENOMEM);
+
+        ret = avcodec_receive_frame(decoder_ctx, frame);
+        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
+            av_frame_free(&frame);
+            return 0;
+        } else if (ret < 0) {
+            fprintf(stderr, "Error while decoding. Error code: %s\n", av_err2str(ret));
+            goto fail;
+        }
+
+        if (!initialized) {
+            /* we need to ref hw_frames_ctx of decoder to initialize encoder's codec.
+               Only after we get a decoded frame, can we obtain its hw_frames_ctx */
+            encoder_ctx->hw_frames_ctx = av_buffer_ref(decoder_ctx->hw_frames_ctx);
+            if (!encoder_ctx->hw_frames_ctx) {
+                ret = AVERROR(ENOMEM);
+                goto fail;
+            }
+            /* set AVCodecContext Parameters for encoder, here we keep them stay
+             * the same as decoder.
+             * xxx: now the sample can't handle resolution change case.
+             */
+            encoder_ctx->time_base = av_inv_q(decoder_ctx->framerate);
+            encoder_ctx->pix_fmt   = AV_PIX_FMT_VAAPI;
+            encoder_ctx->width     = decoder_ctx->width;
+            encoder_ctx->height    = decoder_ctx->height;
+
+            if ((ret = avcodec_open2(encoder_ctx, enc_codec, NULL)) < 0) {
+                fprintf(stderr, "Failed to open encode codec. Error code: %s\n",
+                        av_err2str(ret));
+                goto fail;
+            }
+
+            if (!(ost = avformat_new_stream(ofmt_ctx, enc_codec))) {
+                fprintf(stderr, "Failed to allocate stream for output format.\n");
+                ret = AVERROR(ENOMEM);
+                goto fail;
+            }
+
+            ost->time_base = encoder_ctx->time_base;
+            ret = avcodec_parameters_from_context(ost->codecpar, encoder_ctx);
+            if (ret < 0) {
+                fprintf(stderr, "Failed to copy the stream parameters. "
+                        "Error code: %s\n", av_err2str(ret));
+                goto fail;
+            }
+
+            /* write the stream header */
+            if ((ret = avformat_write_header(ofmt_ctx, NULL)) < 0) {
+                fprintf(stderr, "Error while writing stream header. "
+                        "Error code: %s\n", av_err2str(ret));
+                goto fail;
+            }
+
+            initialized = 1;
+        }
+
+        if ((ret = encode_write(pkt, frame)) < 0)
+            fprintf(stderr, "Error during encoding and writing.\n");
+
+fail:
+        av_frame_free(&frame);
+        if (ret < 0)
+            return ret;
+    }
+    return 0;
+}
+
+int main(int argc, char **argv)
+{
+    int ret = 0;
+    AVPacket *dec_pkt;
+    AVCodec *enc_codec;
+
+    if (argc != 4) {
+        fprintf(stderr, "Usage: %s <input file> <encode codec> <output file>\n"
+                "The output format is guessed according to the file extension.\n"
+                "\n", argv[0]);
+        return -1;
+    }
+
+    ret = av_hwdevice_ctx_create(&hw_device_ctx, AV_HWDEVICE_TYPE_VAAPI, NULL, NULL, 0);
+    if (ret < 0) {
+        fprintf(stderr, "Failed to create a VAAPI device. Error code: %s\n", av_err2str(ret));
+        return -1;
+    }
+
+    dec_pkt = av_packet_alloc();
+    if (!dec_pkt) {
+        fprintf(stderr, "Failed to allocate decode packet\n");
+        goto end;
+    }
+
+    if ((ret = open_input_file(argv[1])) < 0)
+        goto end;
+
+    if (!(enc_codec = avcodec_find_encoder_by_name(argv[2]))) {
+        fprintf(stderr, "Could not find encoder '%s'\n", argv[2]);
+        ret = -1;
+        goto end;
+    }
+
+    if ((ret = (avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, argv[3]))) < 0) {
+        fprintf(stderr, "Failed to deduce output format from file extension. Error code: "
+                "%s\n", av_err2str(ret));
+        goto end;
+    }
+
+    if (!(encoder_ctx = avcodec_alloc_context3(enc_codec))) {
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+
+    ret = avio_open(&ofmt_ctx->pb, argv[3], AVIO_FLAG_WRITE);
+    if (ret < 0) {
+        fprintf(stderr, "Cannot open output file. "
+                "Error code: %s\n", av_err2str(ret));
+        goto end;
+    }
+
+    /* read all packets and only transcoding video */
+    while (ret >= 0) {
+        if ((ret = av_read_frame(ifmt_ctx, dec_pkt)) < 0)
+            break;
+
+        if (video_stream == dec_pkt->stream_index)
+            ret = dec_enc(dec_pkt, enc_codec);
+
+        av_packet_unref(dec_pkt);
+    }
+
+    /* flush decoder */
+    av_packet_unref(dec_pkt);
+    ret = dec_enc(dec_pkt, enc_codec);
+
+    /* flush encoder */
+    ret = encode_write(dec_pkt, NULL);
+
+    /* write the trailer for output stream */
+    av_write_trailer(ofmt_ctx);
+
+end:
+    avformat_close_input(&ifmt_ctx);
+    avformat_close_input(&ofmt_ctx);
+    avcodec_free_context(&decoder_ctx);
+    avcodec_free_context(&encoder_ctx);
+    av_buffer_unref(&hw_device_ctx);
+    av_packet_free(&dec_pkt);
+    return ret;
+}

externals/ffmpeg/ffmpeg/doc/faq.texi

@@ -0,0 +1,686 @@
+\input texinfo @c -*- texinfo -*-
+@documentencoding UTF-8
+
+@settitle FFmpeg FAQ
+@titlepage
+@center @titlefont{FFmpeg FAQ}
+@end titlepage
+
+@top
+
+@contents
+
+@chapter General Questions
+
+@section Why doesn't FFmpeg support feature [xyz]?
+
+Because no one has taken on that task yet. FFmpeg development is
+driven by the tasks that are important to the individual developers.
+If there is a feature that is important to you, the best way to get
+it implemented is to undertake the task yourself or sponsor a developer.
+
+@section FFmpeg does not support codec XXX. Can you include a Windows DLL loader to support it?
+
+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 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
+codecs. Please consult the supported codec list in the ffmpeg
+documentation.
+
+@section Which codecs are supported by Windows?
+
+Windows does not support standard formats like MPEG very well, unless you
+install some additional codecs.
+
+The following list of video codecs should work on most Windows systems:
+@table @option
+@item msmpeg4v2
+.avi/.asf
+@item msmpeg4
+.asf only
+@item wmv1
+.asf only
+@item wmv2
+.asf only
+@item mpeg4
+Only if you have some MPEG-4 codec like ffdshow or Xvid installed.
+@item mpeg1video
+.mpg only
+@end table
+Note, ASF files often have .wmv or .wma extensions in Windows. It should also
+be mentioned that Microsoft claims a patent on the ASF format, and may sue
+or threaten users who create ASF files with non-Microsoft software. It is
+strongly advised to avoid ASF where possible.
+
+The following list of audio codecs should work on most Windows systems:
+@table @option
+@item adpcm_ima_wav
+@item adpcm_ms
+@item pcm_s16le
+always
+@item libmp3lame
+If some MP3 codec like LAME is installed.
+@end table
+
+
+@chapter Compilation
+
+@section @code{error: can't find a register in class 'GENERAL_REGS' while reloading 'asm'}
+
+This is a bug in gcc. Do not report it to us. Instead, please report it to
+the gcc developers. Note that we will not add workarounds for gcc bugs.
+
+Also note that (some of) the gcc developers believe this is not a bug or
+not a bug they should fix:
+@url{https://gcc.gnu.org/bugzilla/show_bug.cgi?id=11203}.
+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 "$@@"
+@end example
+
+@chapter Usage
+
+@section ffmpeg does not work; what is wrong?
+
+Try a @code{make distclean} in the ffmpeg source directory before the build.
+If this does not help see
+(@url{https://ffmpeg.org/bugreports.html}).
+
+@section How do I encode single pictures into movies?
+
+First, rename your pictures to follow a numerical sequence.
+For example, img1.jpg, img2.jpg, img3.jpg,...
+Then you may run:
+
+@example
+ffmpeg -f image2 -i img%d.jpg /tmp/a.mpg
+@end example
+
+Notice that @samp{%d} is replaced by the image number.
+
+@file{img%03d.jpg} means the sequence @file{img001.jpg}, @file{img002.jpg}, etc.
+
+Use the @option{-start_number} option to declare a starting number for
+the sequence. This is useful if your sequence does not start with
+@file{img001.jpg} but is still in a numerical order. The following
+example will start with @file{img100.jpg}:
+
+@example
+ffmpeg -f image2 -start_number 100 -i img%d.jpg /tmp/a.mpg
+@end example
+
+If you have large number of pictures to rename, you can use the
+following command to ease the burden. The command, using the bourne
+shell syntax, symbolically links all files in the current directory
+that match @code{*jpg} to the @file{/tmp} directory in the sequence of
+@file{img001.jpg}, @file{img002.jpg} and so on.
+
+@example
+x=1; for i in *jpg; do counter=$(printf %03d $x); ln -s "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done
+@end example
+
+If you want to sequence them by oldest modified first, substitute
+@code{$(ls -r -t *jpg)} in place of @code{*jpg}.
+
+Then run:
+
+@example
+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
+@end example
+
+The @file{movie.mpg} used as input will be converted to
+@file{movie1.jpg}, @file{movie2.jpg}, etc...
+
+Instead of relying on file format self-recognition, you may also use
+@table @option
+@item -c:v ppm
+@item -c:v png
+@item -c:v mjpeg
+@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
+@end example
+
+Beware that there is no "jpeg" codec. Use "mjpeg" instead.
+
+@section Why do I see a slight quality degradation with multithreaded MPEG* encoding?
+
+For multithreaded MPEG* encoding, the encoded slices must be independent,
+otherwise thread n would practically have to wait for n-1 to finish, so it's
+quite logical that there is a small reduction of quality. This is not a bug.
+
+@section How can I read from the standard input or write to the standard output?
+
+Use @file{-} as file name.
+
+@section -f jpeg doesn't work.
+
+Try '-f image2 test%d.jpg'.
+
+@section Why can I not change the frame rate?
+
+Some codecs, like MPEG-1/2, only allow a small number of fixed frame rates.
+Choose a different codec with the -c:v command line option.
+
+@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
+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 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',
+things to try: '-bf 2', '-mpv_flags qp_rd', '-mpv_flags mv0', '-mpv_flags skip_rd'.
+
+@section Which are good parameters for encoding high quality MPEG-1/MPEG-2?
+
+'-mbd rd -trellis 2 -cmp 2 -subcmp 2 -g 100 -pass 1/2'
+but beware the '-g 100' might cause problems with some decoders.
+Things to try: '-bf 2', '-mpv_flags qp_rd', '-mpv_flags mv0', '-mpv_flags skip_rd'.
+
+@section Interlaced video looks very bad when encoded with ffmpeg, what is wrong?
+
+You should use '-flags +ilme+ildct' and maybe '-flags +alt' for interlaced
+material, and try '-top 0/1' if the result looks really messed-up.
+
+@section How can I read DirectShow files?
+
+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.
+
+Just create an "input.avs" text file with this single line ...
+@example
+DirectShowSource("C:\path to your file\yourfile.asf")
+@end example
+... and then feed that text file to ffmpeg:
+@example
+ffmpeg -i input.avs
+@end example
+
+For ANY other help on AviSynth, please visit the
+@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
+different kinds of "joining" and points out how those are addressed in
+FFmpeg. To join video files may mean:
+
+@itemize
+
+@item
+To put them one after the other: this is called to @emph{concatenate} them
+(in short: concat) and is addressed
+@ref{How can I concatenate video files, in this very faq}.
+
+@item
+To put them together in the same file, to let the user choose between the
+different versions (example: different audio languages): this is called to
+@emph{multiplex} them together (in short: mux), and is done by simply
+invoking ffmpeg with several @option{-i} options.
+
+@item
+For audio, to put all channels together in a single stream (example: two
+mono streams into one stereo stream): this is sometimes called to
+@emph{merge} them, and can be done using the
+@url{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{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{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{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{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{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
+equally humble @code{copy} under Windows), and finally transcoding back to your
+format of choice.
+
+@example
+ffmpeg -i input1.avi -qscale:v 1 intermediate1.mpg
+ffmpeg -i input2.avi -qscale:v 1 intermediate2.mpg
+cat intermediate1.mpg intermediate2.mpg > intermediate_all.mpg
+ffmpeg -i intermediate_all.mpg -qscale:v 2 output.avi
+@end example
+
+Additionally, you can use the @code{concat} protocol instead of @code{cat} or
+@code{copy} which will avoid creation of a potentially huge intermediate file.
+
+@example
+ffmpeg -i input1.avi -qscale:v 1 intermediate1.mpg
+ffmpeg -i input2.avi -qscale:v 1 intermediate2.mpg
+ffmpeg -i concat:"intermediate1.mpg|intermediate2.mpg" -c copy intermediate_all.mpg
+ffmpeg -i intermediate_all.mpg -qscale:v 2 output.avi
+@end example
+
+Note that you may need to escape the character "|" which is special for many
+shells.
+
+Another option is usage of named pipes, should your platform support it:
+
+@example
+mkfifo intermediate1.mpg
+mkfifo intermediate2.mpg
+ffmpeg -i input1.avi -qscale:v 1 -y intermediate1.mpg < /dev/null &
+ffmpeg -i input2.avi -qscale:v 1 -y intermediate2.mpg < /dev/null &
+cat intermediate1.mpg intermediate2.mpg |\
+ffmpeg -f mpeg -i - -c:v mpeg4 -c:a 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
+from all but the first stream. This can be accomplished by piping through
+@code{tail} as seen below. Note that when piping through @code{tail} you
+must use command grouping, @code{@{  ;@}}, to background properly.
+
+For example, let's say we want to concatenate two FLV files into an
+output.flv file:
+
+@example
+mkfifo temp1.a
+mkfifo temp1.v
+mkfifo temp2.a
+mkfifo temp2.v
+mkfifo all.a
+mkfifo all.v
+ffmpeg -i input1.flv -vn -f u16le -c:a pcm_s16le -ac 2 -ar 44100 - > temp1.a < /dev/null &
+ffmpeg -i input2.flv -vn -f u16le -c:a pcm_s16le -ac 2 -ar 44100 - > temp2.a < /dev/null &
+ffmpeg -i input1.flv -an -f yuv4mpegpipe - > temp1.v < /dev/null &
+@{ ffmpeg -i input2.flv -an -f yuv4mpegpipe - < /dev/null | tail -n +2 > temp2.v ; @} &
+cat temp1.a temp2.a > all.a &
+cat temp1.v temp2.v > all.v &
+ffmpeg -f u16le -c:a pcm_s16le -ac 2 -ar 44100 -i all.a \
+       -f yuv4mpegpipe -i all.v \
+       -y output.flv
+rm temp[12].[av] all.[av]
+@end example
+
+@section Using @option{-f lavfi}, audio becomes mono for no apparent reason.
+
+Use @option{-dumpgraph -} to find out exactly where the channel layout is
+lost.
+
+Most likely, it is through @code{auto-inserted aresample}. Try to understand
+why the converting filter was needed at that place.
+
+Just before the output is a likely place, as @option{-f lavfi} currently
+only support packed S16.
+
+Then insert the correct @code{aformat} explicitly in the filtergraph,
+specifying the exact format.
+
+@example
+aformat=sample_fmts=s16:channel_layouts=stereo
+@end example
+
+@section Why does FFmpeg not see the subtitles in my VOB file?
+
+VOB and a few other formats do not have a global header that describes
+everything present in the file. Instead, applications are supposed to scan
+the file to see what it contains. Since VOB files are frequently large, only
+the beginning is scanned. If the subtitles happen only later in the file,
+they will not be 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{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
+
+@anchor{background task}
+@section How do I run ffmpeg as a background task?
+
+ffmpeg normally checks the console input, for entries like "q" to stop
+and "?" to give help, while performing operations. ffmpeg does not have a way of
+detecting when it is running as a background task.
+When it checks the console input, that can cause the process running ffmpeg
+in the background to suspend.
+
+To prevent those input checks, allowing ffmpeg to run as a background task,
+use the @url{ffmpeg.html#stdin-option, @code{-nostdin} option}
+in the ffmpeg invocation. This is effective whether you run ffmpeg in a shell
+or invoke ffmpeg in its own process via an operating system API.
+
+As an alternative, when you are running ffmpeg in a shell, you can redirect
+standard input to @code{/dev/null} (on Linux and macOS)
+or @code{NUL} (on Windows). You can do this redirect either
+on the ffmpeg invocation, or from a shell script which calls ffmpeg.
+
+For example:
+
+@example
+ffmpeg -nostdin -i INPUT OUTPUT
+@end example
+
+or (on Linux, macOS, and other UNIX-like shells):
+
+@example
+ffmpeg -i INPUT OUTPUT </dev/null
+@end example
+
+or (on Windows):
+
+@example
+ffmpeg -i INPUT OUTPUT <NUL
+@end example
+
+@section How do I prevent ffmpeg from suspending with a message like @emph{suspended (tty output)}?
+
+If you run ffmpeg in the background, you may find that its process suspends.
+There may be a message like @emph{suspended (tty output)}. The question is how
+to prevent the process from being suspended.
+
+For example:
+
+@example
+% ffmpeg -i INPUT OUTPUT &> ~/tmp/log.txt &
+[1] 93352
+%
+[1]  + suspended (tty output)  ffmpeg -i INPUT OUTPUT &>
+@end example
+
+The message "tty output" notwithstanding, the problem here is that
+ffmpeg normally checks the console input when it runs. The operating system
+detects this, and suspends the process until you can bring it to the
+foreground and attend to it.
+
+The solution is to use the right techniques to tell ffmpeg not to consult
+console input. You can use the
+@url{ffmpeg.html#stdin-option, @code{-nostdin} option},
+or redirect standard input with @code{< /dev/null}.
+See FAQ
+@ref{background task, @emph{How do I run ffmpeg as a background task?}}
+for details.
+
+@chapter Development
+
+@section Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat?
+
+Yes. Check the @file{doc/examples} directory in the source
+repository, also available online at:
+@url{https://github.com/FFmpeg/FFmpeg/tree/master/doc/examples}.
+
+Examples are also installed by default, usually in
+@code{$PREFIX/share/ffmpeg/examples}.
+
+Also you may read the Developers Guide of the FFmpeg documentation. Alternatively,
+examine the source code for one of the many open source projects that
+already incorporate FFmpeg at (@url{projects.html}).
+
+@section Can you support my C compiler XXX?
+
+It depends. If your compiler is C99-compliant, then patches to support
+it are likely to be welcome if they do not pollute the source code
+with @code{#ifdef}s related to the compiler.
+
+@section Is Microsoft Visual C++ supported?
+
+Yes. Please see the @uref{platform.html, Microsoft Visual C++}
+section in the FFmpeg documentation.
+
+@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++?
+
+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{https://web.archive.org/web/20111004021423/http://kernel.org/pub/linux/docs/lkml/#s15, "Programming Religion"}.
+
+@section Why are the ffmpeg programs devoid of debugging symbols?
+
+The build process creates @command{ffmpeg_g}, @command{ffplay_g}, etc. which
+contain full debug information. Those binaries are stripped to create
+@command{ffmpeg}, @command{ffplay}, etc. If you need the debug information, use
+the *_g versions.
+
+@section I do not like the LGPL, can I contribute code under the GPL instead?
+
+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
+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.
+
+FFmpeg builds static libraries by default. In static libraries, dependencies
+are not handled. That has two consequences. First, you must specify the
+libraries in dependency order: @code{-lavdevice} must come before
+@code{-lavformat}, @code{-lavutil} must come after everything else, etc.
+Second, external libraries that are used in FFmpeg have to be specified too.
+
+An easy way to get the full list of required libraries in dependency order
+is to use @code{pkg-config}.
+
+@example
+c99 -o program program.c $(pkg-config --cflags --libs libavformat libavcodec)
+@end example
+
+See @file{doc/example/Makefile} and @file{doc/example/pc-uninstalled} for
+more details.
+
+@section I'm using FFmpeg from within my C++ application but the linker complains about missing symbols which seem to be available.
+
+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
+encompassing your FFmpeg includes using @code{extern "C"}.
+
+See @url{http://www.parashift.com/c++-faq-lite/mixing-c-and-cpp.html#faq-32.3}
+
+@section I'm using libavutil from within my C++ application but the compiler complains about 'UINT64_C' was not declared in this scope
+
+FFmpeg is a pure C project using C99 math features, in order to enable C++
+to use them you have to append -D__STDC_CONSTANT_MACROS to your CXXFLAGS
+
+@section I have a file in memory / a API different from *open/*read/ libc how do I use it with libavformat?
+
+You have to create a custom AVIOContext using @code{avio_alloc_context},
+see @file{libavformat/aviobuf.c} in FFmpeg and @file{libmpdemux/demux_lavf.c} in MPlayer or MPlayer2 sources.
+
+@section Where is the documentation about ffv1, msmpeg4, asv1, 4xm?
+
+see @url{https://www.ffmpeg.org/~michael/}
+
+@section How do I feed H.263-RTP (and other codecs in RTP) to libavcodec?
+
+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.
+
+@code{r_frame_rate} is NOT the average frame rate, it is the smallest frame rate
+that can accurately represent all timestamps. So no, it is not
+wrong if it is larger than the average!
+For example, if you have mixed 25 and 30 fps content, then @code{r_frame_rate}
+will be 150 (it is the least common multiple).
+If you are looking for the average frame rate, see @code{AVStream.avg_frame_rate}.
+
+@section Why is @code{make fate} not running all tests?
+
+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

externals/ffmpeg/ffmpeg/doc/fate.texi

@@ -0,0 +1,242 @@
+\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 Uploading new samples to the fate suite
+
+If you need a sample uploaded send a mail to samples-request.
+
+This is for developers who have an account on the fate suite server.
+If you upload new samples, please make sure they are as small as possible,
+space on each client, network bandwidth and so on benefit from smaller test cases.
+Also keep in mind older checkouts use existing sample files, that means in
+practice generally do not replace, remove or overwrite files as it likely would
+break older checkouts or releases.
+Also all needed samples for a commit should be uploaded, ideally 24
+hours, before the push.
+If you need an account for frequently uploading samples or you wish to help
+others by doing that send a mail to ffmpeg-devel.
+
+@example
+#First update your local samples copy:
+rsync -vauL --chmod=Dg+s,Duo+x,ug+rw,o+r,o-w,+X fate-suite.ffmpeg.org:/home/samples/fate-suite/ ~/fate-suite
+
+#Then do a dry run checking what would be uploaded:
+rsync -vanL --no-g --chmod=Dg+s,Duo+x,ug+rw,o+r,o-w,+X ~/fate-suite/ fate-suite.ffmpeg.org:/home/samples/fate-suite
+
+#Upload the files:
+rsync -vaL  --no-g --chmod=Dg+s,Duo+x,ug+rw,o+r,o-w,+X ~/fate-suite/ fate-suite.ffmpeg.org:/home/samples/fate-suite
+@end example
+
+
+@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.
+
+@item HWACCEL
+Specify which hardware acceleration to use while running regression tests,
+by default @samp{none} is used.
+
+@item KEEP
+Set to @samp{1} to keep temp files generated by fate test(s) when test is successful.
+Default is @samp{0}, which removes these files. Files are always kept when a test
+fails.
+
+@end table
+
+@section Examples
+
+@example
+make V=1 SAMPLES=/var/fate/samples THREADS=2 CPUFLAGS=mmx fate
+@end example

externals/ffmpeg/ffmpeg/doc/fate_config.sh.template

@@ -0,0 +1,33 @@
+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
+ignore_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'
+#makeopts_fate= # extra options passed to 'make' when running tests,
+                # defaulting to makeopts above if this is not set
+#tar=           # command to create a tar archive from its arguments on stdout,
+                # defaults to 'tar c'

externals/ffmpeg/ffmpeg/doc/ffmpeg-bitstream-filters.texi

@@ -0,0 +1,46 @@
+\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{libavcodec.html,libavcodec}
+@end ifhtml
+
+@ifnothtml
+ffmpeg(1), ffplay(1), ffprobe(1), libavcodec(3)
+@end ifnothtml
+
+@include authors.texi
+
+@ignore
+
+@setfilename ffmpeg-bitstream-filters
+@settitle FFmpeg bitstream filters
+
+@end ignore
+
+@bye

externals/ffmpeg/ffmpeg/doc/ffmpeg-codecs.texi

@@ -0,0 +1,43 @@
+\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{libavcodec.html,libavcodec}
+@end ifhtml
+
+@ifnothtml
+ffmpeg(1), ffplay(1), ffprobe(1), libavcodec(3)
+@end ifnothtml
+
+@include authors.texi
+
+@ignore
+
+@setfilename ffmpeg-codecs
+@settitle FFmpeg codecs
+
+@end ignore
+
+@bye

externals/ffmpeg/ffmpeg/doc/ffmpeg-devices.texi

@@ -0,0 +1,43 @@
+\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{libavdevice.html,libavdevice}
+@end ifhtml
+
+@ifnothtml
+ffmpeg(1), ffplay(1), ffprobe(1), libavdevice(3)
+@end ifnothtml
+
+@include authors.texi
+
+@ignore
+
+@setfilename ffmpeg-devices
+@settitle FFmpeg devices
+
+@end ignore
+
+@bye

externals/ffmpeg/ffmpeg/doc/ffmpeg-filters.texi

@@ -0,0 +1,43 @@
+\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{libavfilter.html,libavfilter}
+@end ifhtml
+
+@ifnothtml
+ffmpeg(1), ffplay(1), ffprobe(1), libavfilter(3)
+@end ifnothtml
+
+@include authors.texi
+
+@ignore
+
+@setfilename ffmpeg-filters
+@settitle FFmpeg filters
+
+@end ignore
+
+@bye