Changelog: update

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>

.gitignore

@@ -19,12 +19,8 @@
 *.swp
 *.ver
 *.version
-*.metal.air
-*.metallib
-*.metallib.c
 *.ptx
 *.ptx.c
-*.ptx.gz
 *_g
 \#*
 .\#*
@@ -35,9 +31,8 @@
 /ffprobe
 /config.asm
 /config.h
-/config_components.h
 /coverage.info
+/avversion.h
 /lcov/
 /src
 /mapfile
-/tools/python/__pycache__/

.mailmap

@@ -1,25 +0,0 @@
-<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>

.travis.yml

@@ -19,7 +19,7 @@ cache:
   directories:
     - ffmpeg-samples
 before_install:
-  - if [ "$TRAVIS_OS_NAME" == "osx" ]; then brew update; fi
+  - if [ "$TRAVIS_OS_NAME" == "osx" ]; then brew update --all; fi
 install:
   - if [ "$TRAVIS_OS_NAME" == "osx" ]; then brew install nasm; fi
 script:

INSTALL.md

@@ -1,4 +1,4 @@
-## Installing FFmpeg
+#Installing FFmpeg:
 
 1. Type `./configure` to create the configuration. A list of configure
 options is printed by running `configure --help`.

LICENSE.md

@@ -21,11 +21,10 @@ Specifically, the GPL parts of FFmpeg are:
     - `compat/solaris/make_sunver.pl`
     - `doc/t2h.pm`
     - `doc/texi2pod.pl`
-    - `libswresample/tests/swresample.c`
+    - `libswresample/swresample-test.c`
     - `tests/checkasm/*`
     - `tests/tiny_ssim.c`
 - the following filters in libavfilter:
-    - `signature_lookup.c`
     - `vf_blackframe.c`
     - `vf_boxblur.c`
     - `vf_colormatrix.c`
@@ -35,13 +34,13 @@ Specifically, the GPL parts of FFmpeg are:
     - `vf_eq.c`
     - `vf_find_rect.c`
     - `vf_fspp.c`
+    - `vf_geq.c`
     - `vf_histeq.c`
     - `vf_hqdn3d.c`
+    - `vf_interlace.c`
     - `vf_kerndeint.c`
-    - `vf_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`
@@ -50,14 +49,12 @@ Specifically, the GPL parts of FFmpeg are:
     - `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
@@ -83,47 +80,41 @@ affect the licensing of binaries resulting from the combination.
 
 ### Compatible libraries
 
-The following libraries are under GPL version 2:
-- avisynth
+The following libraries are under GPL:
 - 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
+The OpenCORE and VisualOn libraries are under the Apache License 2.0. That
+license is incompatible with the LGPL v2.1 and the GPL v2, but not with
 version 3 of those licenses. So to combine these libraries with FFmpeg, the
 license version needs to be upgraded by passing `--enable-version3` to configure.
 
-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
+`--enable-nonfree` to configure. But note that if you enable any of these
+libraries the resulting binary will be under a complex license mix that is
+more restrictive than the LGPL and that may result in additional obligations.
+It is possible that these restrictions cause the resulting binary to be
 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.
+
+The NVENC library, while its header file is licensed under the compatible MIT
+license, requires a proprietary binary blob at run time, and is deemed to be
+incompatible with the GPL. We are not certain if it is compatible with the
+LGPL, but we require `--enable-nonfree` even with LGPL configurations in case
+it is not.

MAINTAINERS

@@ -39,7 +39,7 @@ QuickTime faststart:
 Miscellaneous Areas
 ===================
 
-documentation                           Stefano Sabatini, Mike Melanson, Timothy Gu, Gyan Doshi
+documentation                           Stefano Sabatini, Mike Melanson, Timothy Gu, Lou Logan, Gyan Doshi
 project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser, Nikolay Aleksandrov
 presets                                 Robert Swain
 metadata subsystem                      Aurelien Jacobs
@@ -52,12 +52,12 @@ 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
+Trac bug tracker                        Alexander Strasser, Michael Niedermayer, Carl Eugen Hoyos, Lou Logan
+mailing lists                           Baptiste Coudurier, Lou Logan
+Google+                                 Paul B Mahol, Michael Niedermayer, Alexander Strasser
+Twitter                                 Lou Logan, Reynaldo H. Verdejo Pinochet
 Launchpad                               Timothy Gu
-ffmpeg-security                         Andreas Cadhalpun, Carl Eugen Hoyos, Clément Bœsch, Michael Niedermayer, Reimar Doeffinger, rcombs, wm4
+ffmpeg-security                         Andreas Cadhalpun, Carl Eugen Hoyos, Clément Bœsch, Michael Niedermayer, Reimar Doeffinger, Rodger Combs, wm4
 
 
 libavutil
@@ -78,7 +78,6 @@ Other:
   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
@@ -89,7 +88,6 @@ Other:
   rational.c, rational.h                Michael Niedermayer
   rc4                                   Reimar Doeffinger
   ripemd.c, ripemd.h                    James Almer
-  tx*                                   Lynne
 
 
 libavcodec
@@ -138,15 +136,13 @@ Codecs:
   8bps.c                                Roberto Togni
   8svx.c                                Jaikrishnan Menon
   aacenc*, aaccoder.c                   Rostislav Pehlivanov
-  adpcm.c                               Zane van Iperen
   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
+  audiotoolbox*                         Rodger Combs
   bgmc.c, bgmc.h                        Thilo Borgmann
   binkaudio.c                           Peter Ross
   cavs*                                 Stefan Gehrer
@@ -161,7 +157,6 @@ Codecs:
   cscd.c                                Reimar Doeffinger
   cuviddec.c                            Timo Rothenpieler
   dca*                                  foo86
-  dfpwm*                                Jack Bruienne
   dirac*                                Rostislav Pehlivanov
   dnxhd*                                Baptiste Coudurier
   dolby_e*                              foo86
@@ -172,6 +167,7 @@ Codecs:
   eacmv*, eaidct*, eat*                 Peter Ross
   evrc*                                 Paul B Mahol
   exif.c, exif.h                        Thilo Borgmann
+  exr.c                                 Martin Vignali
   ffv1*                                 Michael Niedermayer
   ffwavesynth.c                         Nicolas George
   fifo.c                                Jan Sebechlebsky
@@ -193,18 +189,14 @@ Codecs:
   libcelt_dec.c                         Nicolas George
   libcodec2.c                           Tomas Härdin
   libdirac*                             David Conrad
-  libdavs2.c                            Huiwen Ren
-  libjxl*.c, libjxl.h                   Leo Izen
   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
@@ -220,7 +212,6 @@ Codecs:
   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
@@ -228,7 +219,7 @@ Codecs:
   ptx.c                                 Ivo van Poorten
   qcelp*                                Reynaldo H. Verdejo Pinochet
   qdm2.c, qdm2data.h                    Roberto Togni
-  qsv*                                  Mark Thompson, Zhong Li, Haihao Xiang
+  qsv*                                  Mark Thompson, Zhong Li
   qtrle.c                               Mike Melanson
   ra144.c, ra144.h, ra288.c, ra288.h    Roberto Togni
   resample2.c                           Michael Niedermayer
@@ -238,6 +229,7 @@ Codecs:
   rv10.c                                Michael Niedermayer
   s3tc*                                 Ivo van Poorten
   smc.c                                 Mike Melanson
+  smvjpegdec.c                          Ash Hughes
   snow*                                 Michael Niedermayer, Loren Merritt
   sonic.c                               Alex Beregszaszi
   speedhq.c                             Steinar H. Gunderson
@@ -268,6 +260,7 @@ Codecs:
   xan.c                                 Mike Melanson
   xbm*                                  Paul B Mahol
   xface                                 Stefano Sabatini
+  xvmc.c                                Ivan Kalvachev
   xwd*                                  Paul B Mahol
 
 Hardware acceleration:
@@ -275,8 +268,8 @@ Hardware acceleration:
   dxva2*                                Hendrik Leppkes, Laurent Aimar, Steve Lhomme
   d3d11va*                              Steve Lhomme
   mediacodec*                           Matthieu Bouron, Aman Gupta
-  vaapi*                                Haihao Xiang
-  vaapi_encode*                         Mark Thompson, Haihao Xiang
+  vaapi*                                Gwenole Beauchesne
+  vaapi_encode*                         Mark Thompson
   vdpau*                                Philip Langdale, Carl Eugen Hoyos
   videotoolbox*                         Rick Kern, Aman Gupta
 
@@ -355,7 +348,6 @@ Filters:
   vf_il.c                               Paul B Mahol
   vf_(t)interlace                       Thomas Mundt (CC <thomas.mundt@hr.de>)
   vf_lenscorrection.c                   Daniel Oberhoff
-  vf_libplacebo.c                       Niklas Haas
   vf_mergeplanes.c                      Paul B Mahol
   vf_mestimate.c                        Davinder Singh
   vf_minterpolate.c                     Davinder Singh
@@ -368,15 +360,12 @@ Filters:
   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
 ===========
 
@@ -395,13 +384,7 @@ Muxers/Demuxers:
   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
-  argo_cvg.c                            Zane van Iperen
   ass*                                  Aurelien Jacobs
   astdec.c                              Paul B Mahol
   astenc.c                              James Almer
@@ -418,7 +401,6 @@ Muxers/Demuxers:
   dashdec.c                             Steven Liu
   dashenc.c                             Karthick Jeyapal
   daud.c                                Reimar Doeffinger
-  dfpwmdec.c                            Jack Bruienne
   dss.c                                 Oleksij Rempel
   dtsdec.c                              foo86
   dtshddec.c                            Paul B Mahol
@@ -439,17 +421,15 @@ Muxers/Demuxers:
   ipmovie.c                             Mike Melanson
   ircam*                                Paul B Mahol
   iss.c                                 Stefan Gehrer
-  jpegxl_probe.*                        Leo Izen
   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.c                            Aurelien Jacobs
+  matroskadec.c                         Aurelien Jacobs
+  matroskaenc.c                         David Conrad
   matroska subtitles (matroskaenc.c)    John Peebles
   metadata*                             Aurelien Jacobs
   mgsts.c                               Paul B Mahol
@@ -464,7 +444,7 @@ Muxers/Demuxers:
   mpegtsenc.c                           Baptiste Coudurier
   msnwc_tcp.c                           Ramiro Polla
   mtv.c                                 Reynaldo H. Verdejo Pinochet
-  mxf*                                  Baptiste Coudurier, Tomas Härdin
+  mxf*                                  Baptiste Coudurier
   nistspheredec.c                       Paul B Mahol
   nsvdec.c                              Francois Revol
   nut*                                  Michael Niedermayer
@@ -472,9 +452,9 @@ Muxers/Demuxers:
   oggdec.c, oggdec.h                    David Conrad
   oggenc.c                              Baptiste Coudurier
   oggparse*.c                           David Conrad
+  oggparsedaala*                        Rostislav Pehlivanov
   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
@@ -519,9 +499,7 @@ Protocols:
   bluray.c                              Petri Hintukainen
   ftp.c                                 Lukasz Marek
   http.c                                Ronald S. Bultje
-  libsrt.c                              Zhao Zhili
   libssh.c                              Lukasz Marek
-  libzmq.c                              Andriy Gelman
   mms*.c                                Ronald S. Bultje
   udp.c                                 Luca Abeni
   icecast.c                             Marvin Scholz
@@ -546,10 +524,8 @@ Operating systems / CPU architectures
 
 Alpha                                   Falk Hueffner
 MIPS                                    Manojkumar Bhosale, Shiyou Yin
-LoongArch                               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
@@ -578,7 +554,6 @@ Joakim Plate
 Jun Zhao
 Kieran Kunhya
 Kirill Gavrilov
-Limin Wang
 Martin Storsjö
 Panagiotis Issaris
 Pedro Arthur
@@ -617,22 +592,17 @@ 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
-Haihao Xiang (haihao)         1F0C 31E8 B4FE F7A4 4DC1 DC99 E0F5 76D4 76FC 437F
 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
-Leo Izen (thebombzen)         B6FD 3CFC 7ACF 83FC 9137 6945 5A71 C331 FD2F A19A
 Loren Merritt                 ABD9 08F4 C920 3F65 D8BE 35D7 1540 DAA7 060F 56DE
-Lynne                         FE50 139C 6805 72CA FD52 1F8D A2FE A5F0 3F03 4464
+Lou Logan (llogan)            7D68 DC73 CBEF EABB 671A B6CF 621C 2E28 82F8 DC3A
 Michael Niedermayer           9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
-                              DD1E C9E8 DE08 5C62 9B3E 1846 B18E 8928 B394 8D64
 Nicolas George                24CE 01CE 9ACC 5CEB 74D8 8D9D B063 D997 36E5 4C93
-Niklas Haas (haasn)           1DDB 8076 B14D 5B48 32FC 99D9 EB52 DA9C 02BA 6FB4
 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
@@ -641,9 +611,7 @@ 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

Makefile

@@ -13,19 +13,17 @@ vpath %.v    $(SRC_PATH)
 vpath %.texi $(SRC_PATH)
 vpath %.cu   $(SRC_PATH)
 vpath %.ptx  $(SRC_PATH)
-vpath %.metal $(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
 
-ALLFFLIBS = avcodec avdevice avfilter avformat avutil postproc swscale swresample
-
 # $(FFLIBS-yes) needs to be in linking order
 FFLIBS-$(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
@@ -52,40 +50,21 @@ $(TOOLS): %$(EXESUF): %.o
 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): $(FF_DEP_LIBS)
-tools/scale_slice_test$(EXESUF): $(FF_DEP_LIBS)
-tools/scale_slice_test$(EXESUF): ELIBS = $(FF_EXTRALIBS)
 tools/sofa2wavs$(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/hwaccels.h                               \
     $(SRC_PATH)/libavcodec/parsers.c                                \
     $(SRC_PATH)/libavformat/protocols.c                             \
 
-config_components.h: ffbuild/.config
+config.h: ffbuild/.config
 ffbuild/.config: $(CONFIGURABLE_COMPONENTS)
 	@-tput bold 2>/dev/null
-	@-printf '\nWARNING: $(?) newer than config_components.h, rerun configure\n\n'
+	@-printf '\nWARNING: $(?) newer than config.h, rerun configure\n\n'
 	@-tput sgr0 2>/dev/null
 
 SUBDIR_VARS := CLEANFILES FFLIBS HOSTPROGS TESTPROGS TOOLS               \
@@ -93,8 +72,7 @@ SUBDIR_VARS := CLEANFILES FFLIBS HOSTPROGS TESTPROGS TOOLS               \
                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 LSX-OBJS LASX-OBJS OBJS SLIBOBJS SHLIBOBJS       \
-               STLIBOBJS HOSTOBJS TESTOBJS
+               MMI-OBJS OBJS SLIBOBJS HOSTOBJS TESTOBJS
 
 define RESET
 $(1) :=
@@ -116,13 +94,12 @@ include $(SRC_PATH)/fftools/Makefile
 include $(SRC_PATH)/doc/Makefile
 include $(SRC_PATH)/doc/examples/Makefile
 
-$(ALLFFLIBS:%=lib%/version.o): libavutil/ffversion.h
+libavcodec/utils.o libavformat/utils.o libavdevice/avdevice.o libavfilter/avfilter.o libavutil/utils.o libpostproc/postprocess.o libswresample/swresample.o libswscale/utils.o : libavutil/ffversion.h
 
 $(PROGS): %$(PROGSSUF)$(EXESUF): %$(PROGSSUF)_g$(EXESUF)
 ifeq ($(STRIPTYPE),direct)
 	$(STRIP) -o $@ $<
 else
-	$(RM) $@
 	$(CP) $< $@
 	$(STRIP) $@
 endif
@@ -158,17 +135,16 @@ uninstall-data:
 
 clean::
 	$(RM) $(CLEANSUFFIXES)
-	$(RM) $(addprefix compat/,$(CLEANSUFFIXES)) $(addprefix compat/*/,$(CLEANSUFFIXES)) $(addprefix compat/*/*/,$(CLEANSUFFIXES))
+	$(RM) $(addprefix compat/,$(CLEANSUFFIXES)) $(addprefix compat/*/,$(CLEANSUFFIXES))
 	$(RM) -r coverage-html
 	$(RM) -rf coverage.info coverage.info.in lcov
 
 distclean:: clean
-	$(RM) .version config.asm config.h config_components.h mapfile  \
+	$(RM) .version avversion.h config.asm config.h mapfile  \
 		ffbuild/.config ffbuild/config.* libavutil/avconfig.h \
 		version.h libavutil/ffversion.h libavcodec/codec_names.h \
 		libavcodec/bsf_list.c libavformat/protocol_list.c \
 		libavcodec/codec_list.c libavcodec/parser_list.c \
-		libavfilter/filter_list.c libavdevice/indev_list.c libavdevice/outdev_list.c \
 		libavformat/muxer_list.c libavformat/demuxer_list.c
 ifeq ($(SRC_LINK),src)
 	$(RM) src
@@ -183,7 +159,7 @@ check: all alltools examples testprogs fate
 
 include $(SRC_PATH)/tests/Makefile
 
-$(sort $(OUTDIRS)):
+$(sort $(OBJDIRS)):
 	$(Q)mkdir -p $@
 
 # Dummy rule to stop make trying to rebuild removed or renamed headers

README.md

@@ -9,7 +9,7 @@ such as audio, video, subtitles and related metadata.
 * `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 means to alter decoded audio and video through a directed graph of connected filters.
+* `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.

RELEASE

@@ -1 +1 @@
-5.1.2
+4.1.6

RELEASE_NOTES

@@ -1,18 +1,15 @@
 
-            ┌────────────────────────────────────────────┐
-            │ RELEASE NOTES for FFmpeg 5.1 "Riemann" LTS │
-            └────────────────────────────────────────────┘
+              ┌─────────────────────────────────────────────┐
+              │ RELEASE NOTES for FFmpeg 4.1 "al-Khwarizmi" │
+              └─────────────────────────────────────────────┘
 
-   The FFmpeg Project proudly presents FFmpeg 5.1 "Riemann" LTS, about 6
-   months after the release of FFmpeg 5.0, our first Long Term Support
-   release. While several past FFmpeg releases have enjoyed long term
-   support, this is the first release where such an intention is made
-   clear at release.
+   The FFmpeg Project proudly presents FFmpeg 4.1 "al-Khwarizmi", about 6
+   months after the release of FFmpeg 4.0.
 
    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.libera.chat) or ask
+   feel free to join us on the #ffmpeg IRC channel (on irc.freenode.net) or ask
    on the mailing-lists.

compat/atomics/win32/stdatomic.h

@@ -96,7 +96,7 @@ do {                                    \
     atomic_load(object)
 
 #define atomic_exchange(object, desired) \
-    InterlockedExchangePointer((PVOID volatile *)object, (PVOID)desired)
+    InterlockedExchangePointer(object, desired);
 
 #define atomic_exchange_explicit(object, desired, order) \
     atomic_exchange(object, desired)

compat/avisynth/avs/capi.h

@@ -0,0 +1,62 @@
+// Avisynth C Interface Version 0.20
+// Copyright 2003 Kevin Atkinson
+
+// This program is free software; you can redistribute it and/or modify
+// it under the terms of the GNU General Public License as published by
+// the Free Software Foundation; either version 2 of the License, or
+// (at your option) any later version.
+//
+// This program is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+// GNU General Public License for more details.
+//
+// You should have received a copy of the GNU General Public License
+// along with this program; if not, write to the Free Software
+// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
+// http://www.gnu.org/copyleft/gpl.html .
+//
+// As a special exception, I give you permission to link to the
+// Avisynth C interface with independent modules that communicate with
+// the Avisynth C interface solely through the interfaces defined in
+// avisynth_c.h, regardless of the license terms of these independent
+// modules, and to copy and distribute the resulting combined work
+// under terms of your choice, provided that every copy of the
+// combined work is accompanied by a complete copy of the source code
+// of the Avisynth C interface and Avisynth itself (with the version
+// used to produce the combined work), being distributed under the
+// terms of the GNU General Public License plus this exception.  An
+// independent module is a module which is not derived from or based
+// on Avisynth C Interface, such as 3rd-party filters, import and
+// export plugins, or graphical user interfaces.
+
+#ifndef AVS_CAPI_H
+#define AVS_CAPI_H
+
+#ifdef __cplusplus
+#  define EXTERN_C extern "C"
+#else
+#  define EXTERN_C
+#endif
+
+#ifndef AVSC_USE_STDCALL
+#  define AVSC_CC __cdecl
+#else
+#  define AVSC_CC __stdcall
+#endif
+
+#define AVSC_INLINE static __inline
+
+#ifdef BUILDING_AVSCORE
+#  define AVSC_EXPORT EXTERN_C
+#  define AVSC_API(ret, name) EXTERN_C __declspec(dllexport) ret AVSC_CC name
+#else
+#  define AVSC_EXPORT EXTERN_C __declspec(dllexport)
+#  ifndef AVSC_NO_DECLSPEC
+#    define AVSC_API(ret, name) EXTERN_C __declspec(dllimport) ret AVSC_CC name
+#  else
+#    define AVSC_API(ret, name) typedef ret (AVSC_CC *name##_func)
+#  endif
+#endif
+
+#endif //AVS_CAPI_H

compat/avisynth/avs/config.h

@@ -0,0 +1,55 @@
+// Avisynth C Interface Version 0.20
+// Copyright 2003 Kevin Atkinson
+
+// This program is free software; you can redistribute it and/or modify
+// it under the terms of the GNU General Public License as published by
+// the Free Software Foundation; either version 2 of the License, or
+// (at your option) any later version.
+//
+// This program is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+// GNU General Public License for more details.
+//
+// You should have received a copy of the GNU General Public License
+// along with this program; if not, write to the Free Software
+// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
+// http://www.gnu.org/copyleft/gpl.html .
+//
+// As a special exception, I give you permission to link to the
+// Avisynth C interface with independent modules that communicate with
+// the Avisynth C interface solely through the interfaces defined in
+// avisynth_c.h, regardless of the license terms of these independent
+// modules, and to copy and distribute the resulting combined work
+// under terms of your choice, provided that every copy of the
+// combined work is accompanied by a complete copy of the source code
+// of the Avisynth C interface and Avisynth itself (with the version
+// used to produce the combined work), being distributed under the
+// terms of the GNU General Public License plus this exception.  An
+// independent module is a module which is not derived from or based
+// on Avisynth C Interface, such as 3rd-party filters, import and
+// export plugins, or graphical user interfaces.
+
+#ifndef AVS_CONFIG_H
+#define AVS_CONFIG_H
+
+// Undefine this to get cdecl calling convention
+#define AVSC_USE_STDCALL 1
+
+// NOTE TO PLUGIN AUTHORS:
+// Because FRAME_ALIGN can be substantially higher than the alignment
+// a plugin actually needs, plugins should not use FRAME_ALIGN to check for
+// alignment. They should always request the exact alignment value they need.
+// This is to make sure that plugins work over the widest range of AviSynth
+// builds possible.
+#define FRAME_ALIGN 32
+
+#if   defined(_M_AMD64) || defined(__x86_64)
+#   define X86_64
+#elif defined(_M_IX86) || defined(__i386__)
+#   define X86_32
+#else
+#   error Unsupported CPU architecture.
+#endif
+
+#endif //AVS_CONFIG_H

compat/avisynth/avs/types.h

@@ -0,0 +1,51 @@
+// Avisynth C Interface Version 0.20
+// Copyright 2003 Kevin Atkinson
+
+// This program is free software; you can redistribute it and/or modify
+// it under the terms of the GNU General Public License as published by
+// the Free Software Foundation; either version 2 of the License, or
+// (at your option) any later version.
+//
+// This program is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+// GNU General Public License for more details.
+//
+// You should have received a copy of the GNU General Public License
+// along with this program; if not, write to the Free Software
+// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
+// http://www.gnu.org/copyleft/gpl.html .
+//
+// As a special exception, I give you permission to link to the
+// Avisynth C interface with independent modules that communicate with
+// the Avisynth C interface solely through the interfaces defined in
+// avisynth_c.h, regardless of the license terms of these independent
+// modules, and to copy and distribute the resulting combined work
+// under terms of your choice, provided that every copy of the
+// combined work is accompanied by a complete copy of the source code
+// of the Avisynth C interface and Avisynth itself (with the version
+// used to produce the combined work), being distributed under the
+// terms of the GNU General Public License plus this exception.  An
+// independent module is a module which is not derived from or based
+// on Avisynth C Interface, such as 3rd-party filters, import and
+// export plugins, or graphical user interfaces.
+
+#ifndef AVS_TYPES_H
+#define AVS_TYPES_H
+
+// Define all types necessary for interfacing with avisynth.dll
+
+// Raster types used by VirtualDub & Avisynth
+typedef unsigned int    Pixel32;
+typedef unsigned char   BYTE;
+
+// Audio Sample information
+typedef float SFLOAT;
+
+#ifdef __GNUC__
+typedef long long int INT64;
+#else
+typedef __int64 INT64;
+#endif
+
+#endif //AVS_TYPES_H

compat/avisynth/avxsynth_c.h

@@ -0,0 +1,728 @@
+// Avisynth C Interface Version 0.20
+// Copyright 2003 Kevin Atkinson
+
+// This program is free software; you can redistribute it and/or modify
+// it under the terms of the GNU General Public License as published by
+// the Free Software Foundation; either version 2 of the License, or
+// (at your option) any later version.
+//
+// This program is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+// GNU General Public License for more details.
+//
+// You should have received a copy of the GNU General Public License
+// along with this program; if not, write to the Free Software
+// Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston,
+// MA 02110-1301 USA, or visit
+// http://www.gnu.org/copyleft/gpl.html .
+//
+// As a special exception, I give you permission to link to the
+// Avisynth C interface with independent modules that communicate with
+// the Avisynth C interface solely through the interfaces defined in
+// avisynth_c.h, regardless of the license terms of these independent
+// modules, and to copy and distribute the resulting combined work
+// under terms of your choice, provided that every copy of the
+// combined work is accompanied by a complete copy of the source code
+// of the Avisynth C interface and Avisynth itself (with the version
+// used to produce the combined work), being distributed under the
+// terms of the GNU General Public License plus this exception.  An
+// independent module is a module which is not derived from or based
+// on Avisynth C Interface, such as 3rd-party filters, import and
+// export plugins, or graphical user interfaces.
+
+#ifndef __AVXSYNTH_C__
+#define __AVXSYNTH_C__
+
+#include "windowsPorts/windows2linux.h"
+#include <stdarg.h>
+
+#ifdef __cplusplus
+#  define EXTERN_C extern "C"
+#else
+#  define EXTERN_C
+#endif
+
+#define AVSC_USE_STDCALL 1
+
+#ifndef AVSC_USE_STDCALL
+#  define AVSC_CC __cdecl
+#else
+#  define AVSC_CC __stdcall
+#endif
+
+#define AVSC_INLINE static __inline
+
+#ifdef AVISYNTH_C_EXPORTS
+#  define AVSC_EXPORT EXTERN_C
+#  define AVSC_API(ret, name) EXTERN_C __declspec(dllexport) ret AVSC_CC name
+#else
+#  define AVSC_EXPORT EXTERN_C __declspec(dllexport)
+#  ifndef AVSC_NO_DECLSPEC
+#    define AVSC_API(ret, name) EXTERN_C __declspec(dllimport) ret AVSC_CC name
+#  else
+#    define AVSC_API(ret, name) typedef ret (AVSC_CC *name##_func)
+#  endif
+#endif
+
+#ifdef __GNUC__
+typedef long long int INT64;
+#else
+typedef __int64 INT64;
+#endif
+
+
+/////////////////////////////////////////////////////////////////////
+//
+// Constants
+//
+
+#ifndef __AVXSYNTH_H__
+enum { AVISYNTH_INTERFACE_VERSION = 3 };
+#endif
+
+enum {AVS_SAMPLE_INT8  = 1<<0,
+      AVS_SAMPLE_INT16 = 1<<1,
+      AVS_SAMPLE_INT24 = 1<<2,
+      AVS_SAMPLE_INT32 = 1<<3,
+      AVS_SAMPLE_FLOAT = 1<<4};
+
+enum {AVS_PLANAR_Y=1<<0,
+      AVS_PLANAR_U=1<<1,
+      AVS_PLANAR_V=1<<2,
+      AVS_PLANAR_ALIGNED=1<<3,
+      AVS_PLANAR_Y_ALIGNED=AVS_PLANAR_Y|AVS_PLANAR_ALIGNED,
+      AVS_PLANAR_U_ALIGNED=AVS_PLANAR_U|AVS_PLANAR_ALIGNED,
+      AVS_PLANAR_V_ALIGNED=AVS_PLANAR_V|AVS_PLANAR_ALIGNED};
+
+  // Colorspace properties.
+enum {AVS_CS_BGR = 1<<28,
+      AVS_CS_YUV = 1<<29,
+      AVS_CS_INTERLEAVED = 1<<30,
+      AVS_CS_PLANAR = 1<<31};
+
+  // Specific colorformats
+enum {
+  AVS_CS_UNKNOWN = 0,
+  AVS_CS_BGR24 = 1<<0 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
+  AVS_CS_BGR32 = 1<<1 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
+  AVS_CS_YUY2 = 1<<2 | AVS_CS_YUV | AVS_CS_INTERLEAVED,
+  AVS_CS_YV12 = 1<<3 | AVS_CS_YUV | AVS_CS_PLANAR,  // y-v-u, planar
+  AVS_CS_I420 = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR,  // y-u-v, planar
+  AVS_CS_IYUV = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR  // same as above
+};
+
+enum {
+  AVS_IT_BFF = 1<<0,
+  AVS_IT_TFF = 1<<1,
+  AVS_IT_FIELDBASED = 1<<2};
+
+enum {
+  AVS_FILTER_TYPE=1,
+  AVS_FILTER_INPUT_COLORSPACE=2,
+  AVS_FILTER_OUTPUT_TYPE=9,
+  AVS_FILTER_NAME=4,
+  AVS_FILTER_AUTHOR=5,
+  AVS_FILTER_VERSION=6,
+  AVS_FILTER_ARGS=7,
+  AVS_FILTER_ARGS_INFO=8,
+  AVS_FILTER_ARGS_DESCRIPTION=10,
+  AVS_FILTER_DESCRIPTION=11};
+
+enum {  //SUBTYPES
+  AVS_FILTER_TYPE_AUDIO=1,
+  AVS_FILTER_TYPE_VIDEO=2,
+  AVS_FILTER_OUTPUT_TYPE_SAME=3,
+  AVS_FILTER_OUTPUT_TYPE_DIFFERENT=4};
+
+enum {
+  AVS_CACHE_NOTHING=0,
+  AVS_CACHE_RANGE=1,
+  AVS_CACHE_ALL=2,
+  AVS_CACHE_AUDIO=3,
+  AVS_CACHE_AUDIO_NONE=4,
+  AVS_CACHE_AUDIO_AUTO=5
+};
+
+#define AVS_FRAME_ALIGN 16
+
+typedef struct AVS_Clip AVS_Clip;
+typedef struct AVS_ScriptEnvironment AVS_ScriptEnvironment;
+
+/////////////////////////////////////////////////////////////////////
+//
+// AVS_VideoInfo
+//
+
+// AVS_VideoInfo is layed out identicly to VideoInfo
+typedef struct AVS_VideoInfo {
+  int width, height;    // width=0 means no video
+  unsigned fps_numerator, fps_denominator;
+  int num_frames;
+
+  int pixel_type;
+
+  int audio_samples_per_second;   // 0 means no audio
+  int sample_type;
+  INT64 num_audio_samples;
+  int nchannels;
+
+  // Imagetype properties
+
+  int image_type;
+} AVS_VideoInfo;
+
+// useful functions of the above
+AVSC_INLINE int avs_has_video(const AVS_VideoInfo * p)
+        { return (p->width!=0); }
+
+AVSC_INLINE int avs_has_audio(const AVS_VideoInfo * p)
+        { return (p->audio_samples_per_second!=0); }
+
+AVSC_INLINE int avs_is_rgb(const AVS_VideoInfo * p)
+        { return !!(p->pixel_type&AVS_CS_BGR); }
+
+AVSC_INLINE int avs_is_rgb24(const AVS_VideoInfo * p)
+        { return (p->pixel_type&AVS_CS_BGR24)==AVS_CS_BGR24; } // Clear out additional properties
+
+AVSC_INLINE int avs_is_rgb32(const AVS_VideoInfo * p)
+        { return (p->pixel_type & AVS_CS_BGR32) == AVS_CS_BGR32 ; }
+
+AVSC_INLINE int avs_is_yuv(const AVS_VideoInfo * p)
+        { return !!(p->pixel_type&AVS_CS_YUV ); }
+
+AVSC_INLINE int avs_is_yuy2(const AVS_VideoInfo * p)
+        { return (p->pixel_type & AVS_CS_YUY2) == AVS_CS_YUY2; }
+
+AVSC_INLINE int avs_is_yv12(const AVS_VideoInfo * p)
+        { return ((p->pixel_type & AVS_CS_YV12) == AVS_CS_YV12)||((p->pixel_type & AVS_CS_I420) == AVS_CS_I420); }
+
+AVSC_INLINE int avs_is_color_space(const AVS_VideoInfo * p, int c_space)
+        { return ((p->pixel_type & c_space) == c_space); }
+
+AVSC_INLINE int avs_is_property(const AVS_VideoInfo * p, int property)
+        { return ((p->pixel_type & property)==property ); }
+
+AVSC_INLINE int avs_is_planar(const AVS_VideoInfo * p)
+        { return !!(p->pixel_type & AVS_CS_PLANAR); }
+
+AVSC_INLINE int avs_is_field_based(const AVS_VideoInfo * p)
+        { return !!(p->image_type & AVS_IT_FIELDBASED); }
+
+AVSC_INLINE int avs_is_parity_known(const AVS_VideoInfo * p)
+        { return ((p->image_type & AVS_IT_FIELDBASED)&&(p->image_type & (AVS_IT_BFF | AVS_IT_TFF))); }
+
+AVSC_INLINE int avs_is_bff(const AVS_VideoInfo * p)
+        { return !!(p->image_type & AVS_IT_BFF); }
+
+AVSC_INLINE int avs_is_tff(const AVS_VideoInfo * p)
+        { return !!(p->image_type & AVS_IT_TFF); }
+
+AVSC_INLINE int avs_bits_per_pixel(const AVS_VideoInfo * p)
+{
+  switch (p->pixel_type) {
+      case AVS_CS_BGR24: return 24;
+      case AVS_CS_BGR32: return 32;
+      case AVS_CS_YUY2:  return 16;
+      case AVS_CS_YV12:
+      case AVS_CS_I420:  return 12;
+      default:           return 0;
+    }
+}
+AVSC_INLINE int avs_bytes_from_pixels(const AVS_VideoInfo * p, int pixels)
+        { return pixels * (avs_bits_per_pixel(p)>>3); }   // Will work on planar images, but will return only luma planes
+
+AVSC_INLINE int avs_row_size(const AVS_VideoInfo * p)
+        { return avs_bytes_from_pixels(p,p->width); }  // Also only returns first plane on planar images
+
+AVSC_INLINE int avs_bmp_size(const AVS_VideoInfo * vi)
+        { if (avs_is_planar(vi)) {int p = vi->height * ((avs_row_size(vi)+3) & ~3); p+=p>>1; return p;  } return vi->height * ((avs_row_size(vi)+3) & ~3); }
+
+AVSC_INLINE int avs_samples_per_second(const AVS_VideoInfo * p)
+        { return p->audio_samples_per_second; }
+
+
+AVSC_INLINE int avs_bytes_per_channel_sample(const AVS_VideoInfo * p)
+{
+    switch (p->sample_type) {
+      case AVS_SAMPLE_INT8:  return sizeof(signed char);
+      case AVS_SAMPLE_INT16: return sizeof(signed short);
+      case AVS_SAMPLE_INT24: return 3;
+      case AVS_SAMPLE_INT32: return sizeof(signed int);
+      case AVS_SAMPLE_FLOAT: return sizeof(float);
+      default: return 0;
+    }
+}
+AVSC_INLINE int avs_bytes_per_audio_sample(const AVS_VideoInfo * p)
+        { return p->nchannels*avs_bytes_per_channel_sample(p);}
+
+AVSC_INLINE INT64 avs_audio_samples_from_frames(const AVS_VideoInfo * p, INT64 frames)
+        { return ((INT64)(frames) * p->audio_samples_per_second * p->fps_denominator / p->fps_numerator); }
+
+AVSC_INLINE int avs_frames_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
+        { return (int)(samples * (INT64)p->fps_numerator / (INT64)p->fps_denominator / (INT64)p->audio_samples_per_second); }
+
+AVSC_INLINE INT64 avs_audio_samples_from_bytes(const AVS_VideoInfo * p, INT64 bytes)
+        { return bytes / avs_bytes_per_audio_sample(p); }
+
+AVSC_INLINE INT64 avs_bytes_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
+        { return samples * avs_bytes_per_audio_sample(p); }
+
+AVSC_INLINE int avs_audio_channels(const AVS_VideoInfo * p)
+        { return p->nchannels; }
+
+AVSC_INLINE int avs_sample_type(const AVS_VideoInfo * p)
+        { return p->sample_type;}
+
+// useful mutator
+AVSC_INLINE void avs_set_property(AVS_VideoInfo * p, int property)
+        { p->image_type|=property; }
+
+AVSC_INLINE void avs_clear_property(AVS_VideoInfo * p, int property)
+        { p->image_type&=~property; }
+
+AVSC_INLINE void avs_set_field_based(AVS_VideoInfo * p, int isfieldbased)
+        { if (isfieldbased) p->image_type|=AVS_IT_FIELDBASED; else p->image_type&=~AVS_IT_FIELDBASED; }
+
+AVSC_INLINE void avs_set_fps(AVS_VideoInfo * p, unsigned numerator, unsigned denominator)
+{
+    unsigned x=numerator, y=denominator;
+    while (y) {   // find gcd
+      unsigned t = x%y; x = y; y = t;
+    }
+    p->fps_numerator = numerator/x;
+    p->fps_denominator = denominator/x;
+}
+
+AVSC_INLINE int avs_is_same_colorspace(AVS_VideoInfo * x, AVS_VideoInfo * y)
+{
+        return (x->pixel_type == y->pixel_type)
+                || (avs_is_yv12(x) && avs_is_yv12(y));
+}
+
+/////////////////////////////////////////////////////////////////////
+//
+// AVS_VideoFrame
+//
+
+// VideoFrameBuffer holds information about a memory block which is used
+// for video data.  For efficiency, instances of this class are not deleted
+// when the refcount reaches zero; instead they're stored in a linked list
+// to be reused.  The instances are deleted when the corresponding AVS
+// file is closed.
+
+// AVS_VideoFrameBuffer is layed out identicly to VideoFrameBuffer
+// DO NOT USE THIS STRUCTURE DIRECTLY
+typedef struct AVS_VideoFrameBuffer {
+  unsigned char * data;
+  int data_size;
+  // sequence_number is incremented every time the buffer is changed, so
+  // that stale views can tell they're no longer valid.
+  long sequence_number;
+
+  long refcount;
+} AVS_VideoFrameBuffer;
+
+// VideoFrame holds a "window" into a VideoFrameBuffer.
+
+// AVS_VideoFrame is layed out identicly to IVideoFrame
+// DO NOT USE THIS STRUCTURE DIRECTLY
+typedef struct AVS_VideoFrame {
+  int refcount;
+  AVS_VideoFrameBuffer * vfb;
+  int offset, pitch, row_size, height, offsetU, offsetV, pitchUV;  // U&V offsets are from top of picture.
+} AVS_VideoFrame;
+
+// Access functions for AVS_VideoFrame
+AVSC_INLINE int avs_get_pitch(const AVS_VideoFrame * p) {
+        return p->pitch;}
+
+AVSC_INLINE int avs_get_pitch_p(const AVS_VideoFrame * p, int plane) {
+  switch (plane) {
+  case AVS_PLANAR_U: case AVS_PLANAR_V: return p->pitchUV;}
+  return p->pitch;}
+
+AVSC_INLINE int avs_get_row_size(const AVS_VideoFrame * p) {
+        return p->row_size; }
+
+AVSC_INLINE int avs_get_row_size_p(const AVS_VideoFrame * p, int plane) {
+        int r;
+    switch (plane) {
+    case AVS_PLANAR_U: case AVS_PLANAR_V:
+                if (p->pitchUV) return p->row_size>>1;
+                else            return 0;
+    case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED:
+                if (p->pitchUV) {
+                        r = ((p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)) )>>1; // Aligned rowsize
+                        if (r < p->pitchUV)
+                                return r;
+                        return p->row_size>>1;
+                } else return 0;
+    case AVS_PLANAR_Y_ALIGNED:
+                r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
+                if (r <= p->pitch)
+                        return r;
+                return p->row_size;
+    }
+    return p->row_size;
+}
+
+AVSC_INLINE int avs_get_height(const AVS_VideoFrame * p) {
+        return p->height;}
+
+AVSC_INLINE int avs_get_height_p(const AVS_VideoFrame * p, int plane) {
+        switch (plane) {
+                case AVS_PLANAR_U: case AVS_PLANAR_V:
+                        if (p->pitchUV) return p->height>>1;
+                        return 0;
+        }
+        return p->height;}
+
+AVSC_INLINE const unsigned char* avs_get_read_ptr(const AVS_VideoFrame * p) {
+        return p->vfb->data + p->offset;}
+
+AVSC_INLINE const unsigned char* avs_get_read_ptr_p(const AVS_VideoFrame * p, int plane)
+{
+        switch (plane) {
+                case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
+                case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
+                default:           return p->vfb->data + p->offset;}
+}
+
+AVSC_INLINE int avs_is_writable(const AVS_VideoFrame * p) {
+        return (p->refcount == 1 && p->vfb->refcount == 1);}
+
+AVSC_INLINE unsigned char* avs_get_write_ptr(const AVS_VideoFrame * p)
+{
+        if (avs_is_writable(p)) {
+                ++p->vfb->sequence_number;
+                return p->vfb->data + p->offset;
+        } else
+                return 0;
+}
+
+AVSC_INLINE unsigned char* avs_get_write_ptr_p(const AVS_VideoFrame * p, int plane)
+{
+        if (plane==AVS_PLANAR_Y && avs_is_writable(p)) {
+                ++p->vfb->sequence_number;
+                return p->vfb->data + p->offset;
+        } else if (plane==AVS_PLANAR_Y) {
+                return 0;
+        } else {
+                switch (plane) {
+                        case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
+                        case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
+                        default:       return p->vfb->data + p->offset;
+                }
+        }
+}
+
+#if defined __cplusplus
+extern "C"
+{
+#endif // __cplusplus
+AVSC_API(void, avs_release_video_frame)(AVS_VideoFrame *);
+// makes a shallow copy of a video frame
+AVSC_API(AVS_VideoFrame *, avs_copy_video_frame)(AVS_VideoFrame *);
+#if defined __cplusplus
+}
+#endif // __cplusplus
+
+#ifndef AVSC_NO_DECLSPEC
+AVSC_INLINE void avs_release_frame(AVS_VideoFrame * f)
+  {avs_release_video_frame(f);}
+AVSC_INLINE AVS_VideoFrame * avs_copy_frame(AVS_VideoFrame * f)
+  {return avs_copy_video_frame(f);}
+#endif
+
+/////////////////////////////////////////////////////////////////////
+//
+// AVS_Value
+//
+
+// Treat AVS_Value as a fat pointer.  That is use avs_copy_value
+// and avs_release_value appropiaty as you would if AVS_Value was
+// a pointer.
+
+// To maintain source code compatibility with future versions of the
+// avisynth_c API don't use the AVS_Value directly.  Use the helper
+// functions below.
+
+// AVS_Value is layed out identicly to AVSValue
+typedef struct AVS_Value AVS_Value;
+struct AVS_Value {
+  short type;  // 'a'rray, 'c'lip, 'b'ool, 'i'nt, 'f'loat, 's'tring, 'v'oid, or 'l'ong
+               // for some function e'rror
+  short array_size;
+  union {
+    void * clip; // do not use directly, use avs_take_clip
+    char boolean;
+    int integer;
+    INT64 integer64; // match addition of __int64 to avxplugin.h
+    float floating_pt;
+    const char * string;
+    const AVS_Value * array;
+  } d;
+};
+
+// AVS_Value should be initilized with avs_void.
+// Should also set to avs_void after the value is released
+// with avs_copy_value.  Consider it the equalvent of setting
+// a pointer to NULL
+static const AVS_Value avs_void = {'v'};
+
+AVSC_API(void, avs_copy_value)(AVS_Value * dest, AVS_Value src);
+AVSC_API(void, avs_release_value)(AVS_Value);
+
+AVSC_INLINE int avs_defined(AVS_Value v) { return v.type != 'v'; }
+AVSC_INLINE int avs_is_clip(AVS_Value v) { return v.type == 'c'; }
+AVSC_INLINE int avs_is_bool(AVS_Value v) { return v.type == 'b'; }
+AVSC_INLINE int avs_is_int(AVS_Value v) { return v.type == 'i'; }
+AVSC_INLINE int avs_is_float(AVS_Value v) { return v.type == 'f' || v.type == 'i'; }
+AVSC_INLINE int avs_is_string(AVS_Value v) { return v.type == 's'; }
+AVSC_INLINE int avs_is_array(AVS_Value v) { return v.type == 'a'; }
+AVSC_INLINE int avs_is_error(AVS_Value v) { return v.type == 'e'; }
+
+#if defined __cplusplus
+extern "C"
+{
+#endif // __cplusplus
+AVSC_API(AVS_Clip *, avs_take_clip)(AVS_Value, AVS_ScriptEnvironment *);
+AVSC_API(void, avs_set_to_clip)(AVS_Value *, AVS_Clip *);
+#if defined __cplusplus
+}
+#endif // __cplusplus
+
+AVSC_INLINE int avs_as_bool(AVS_Value v)
+        { return v.d.boolean; }
+AVSC_INLINE int avs_as_int(AVS_Value v)
+        { return v.d.integer; }
+AVSC_INLINE const char * avs_as_string(AVS_Value v)
+        { return avs_is_error(v) || avs_is_string(v) ? v.d.string : 0; }
+AVSC_INLINE double avs_as_float(AVS_Value v)
+        { return avs_is_int(v) ? v.d.integer : v.d.floating_pt; }
+AVSC_INLINE const char * avs_as_error(AVS_Value v)
+        { return avs_is_error(v) ? v.d.string : 0; }
+AVSC_INLINE const AVS_Value * avs_as_array(AVS_Value v)
+        { return v.d.array; }
+AVSC_INLINE int avs_array_size(AVS_Value v)
+        { return avs_is_array(v) ? v.array_size : 1; }
+AVSC_INLINE AVS_Value avs_array_elt(AVS_Value v, int index)
+        { return avs_is_array(v) ? v.d.array[index] : v; }
+
+// only use these functions on am AVS_Value that does not already have
+// an active value.  Remember, treat AVS_Value as a fat pointer.
+AVSC_INLINE AVS_Value avs_new_value_bool(int v0)
+        { AVS_Value v = {0}; v.type = 'b'; v.d.boolean = v0 == 0 ? 0 : 1; return v; }
+AVSC_INLINE AVS_Value avs_new_value_int(int v0)
+        { AVS_Value v = {0}; v.type = 'i'; v.d.integer = v0; return v; }
+AVSC_INLINE AVS_Value avs_new_value_string(const char * v0)
+        { AVS_Value v = {0}; v.type = 's'; v.d.string = v0; return v; }
+AVSC_INLINE AVS_Value avs_new_value_float(float v0)
+        { AVS_Value v = {0}; v.type = 'f'; v.d.floating_pt = v0; return v;}
+AVSC_INLINE AVS_Value avs_new_value_error(const char * v0)
+        { AVS_Value v = {0}; v.type = 'e'; v.d.string = v0; return v; }
+#ifndef AVSC_NO_DECLSPEC
+AVSC_INLINE AVS_Value avs_new_value_clip(AVS_Clip * v0)
+        { AVS_Value v = {0}; avs_set_to_clip(&v, v0); return v; }
+#endif
+AVSC_INLINE AVS_Value avs_new_value_array(AVS_Value * v0, int size)
+        { AVS_Value v = {0}; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; }
+
+/////////////////////////////////////////////////////////////////////
+//
+// AVS_Clip
+//
+#if defined __cplusplus
+extern "C"
+{
+#endif // __cplusplus
+AVSC_API(void, avs_release_clip)(AVS_Clip *);
+AVSC_API(AVS_Clip *, avs_copy_clip)(AVS_Clip *);
+
+AVSC_API(const char *, avs_clip_get_error)(AVS_Clip *); // return 0 if no error
+
+AVSC_API(const AVS_VideoInfo *, avs_get_video_info)(AVS_Clip *);
+
+AVSC_API(int, avs_get_version)(AVS_Clip *);
+
+AVSC_API(AVS_VideoFrame *, avs_get_frame)(AVS_Clip *, int n);
+// The returned video frame must be released with avs_release_video_frame
+
+AVSC_API(int, avs_get_parity)(AVS_Clip *, int n);
+// return field parity if field_based, else parity of first field in frame
+
+AVSC_API(int, avs_get_audio)(AVS_Clip *, void * buf,
+                                  INT64 start, INT64 count);
+// start and count are in samples
+
+AVSC_API(int, avs_set_cache_hints)(AVS_Clip *,
+                                        int cachehints, size_t frame_range);
+#if defined __cplusplus
+}
+#endif // __cplusplus
+
+// This is the callback type used by avs_add_function
+typedef AVS_Value (AVSC_CC * AVS_ApplyFunc)
+                        (AVS_ScriptEnvironment *, AVS_Value args, void * user_data);
+
+typedef struct AVS_FilterInfo AVS_FilterInfo;
+struct AVS_FilterInfo
+{
+  // these members should not be modified outside of the AVS_ApplyFunc callback
+  AVS_Clip * child;
+  AVS_VideoInfo vi;
+  AVS_ScriptEnvironment * env;
+  AVS_VideoFrame * (AVSC_CC * get_frame)(AVS_FilterInfo *, int n);
+  int (AVSC_CC * get_parity)(AVS_FilterInfo *, int n);
+  int (AVSC_CC * get_audio)(AVS_FilterInfo *, void * buf,
+                                  INT64 start, INT64 count);
+  int (AVSC_CC * set_cache_hints)(AVS_FilterInfo *, int cachehints,
+                                        int frame_range);
+  void (AVSC_CC * free_filter)(AVS_FilterInfo *);
+
+  // Should be set when ever there is an error to report.
+  // It is cleared before any of the above methods are called
+  const char * error;
+  // this is to store whatever and may be modified at will
+  void * user_data;
+};
+
+// Create a new filter
+// fi is set to point to the AVS_FilterInfo so that you can
+//   modify it once it is initilized.
+// store_child should generally be set to true.  If it is not
+//    set than ALL methods (the function pointers) must be defined
+// If it is set than you do not need to worry about freeing the child
+//    clip.
+#if defined __cplusplus
+extern "C"
+{
+#endif // __cplusplus
+AVSC_API(AVS_Clip *, avs_new_c_filter)(AVS_ScriptEnvironment * e,
+                                      AVS_FilterInfo * * fi,
+                                      AVS_Value child, int store_child);
+#if defined __cplusplus
+}
+#endif // __cplusplus
+
+
+/////////////////////////////////////////////////////////////////////
+//
+// AVS_ScriptEnvironment
+//
+
+// For GetCPUFlags.  These are backwards-compatible with those in VirtualDub.
+enum {
+                                /* slowest CPU to support extension */
+  AVS_CPU_FORCE        = 0x01,   // N/A
+  AVS_CPU_FPU          = 0x02,   // 386/486DX
+  AVS_CPU_MMX          = 0x04,   // P55C, K6, PII
+  AVS_CPU_INTEGER_SSE  = 0x08,   // PIII, Athlon
+  AVS_CPU_SSE          = 0x10,   // PIII, Athlon XP/MP
+  AVS_CPU_SSE2         = 0x20,   // PIV, Hammer
+  AVS_CPU_3DNOW        = 0x40,   // K6-2
+  AVS_CPU_3DNOW_EXT    = 0x80,   // Athlon
+  AVS_CPU_X86_64       = 0xA0,   // Hammer (note: equiv. to 3DNow + SSE2,
+                                 // which only Hammer will have anyway)
+};
+
+#if defined __cplusplus
+extern "C"
+{
+#endif // __cplusplus
+AVSC_API(const char *, avs_get_error)(AVS_ScriptEnvironment *); // return 0 if no error
+
+AVSC_API(long, avs_get_cpu_flags)(AVS_ScriptEnvironment *);
+AVSC_API(int, avs_check_version)(AVS_ScriptEnvironment *, int version);
+
+AVSC_API(char *, avs_save_string)(AVS_ScriptEnvironment *, const char* s, int length);
+AVSC_API(char *, avs_sprintf)(AVS_ScriptEnvironment *, const char * fmt, ...);
+
+AVSC_API(char *, avs_vsprintf)(AVS_ScriptEnvironment *, const char * fmt, va_list val);
+ // note: val is really a va_list; I hope everyone typedefs va_list to a pointer
+
+AVSC_API(int, avs_add_function)(AVS_ScriptEnvironment *,
+                                     const char * name, const char * params,
+                                     AVS_ApplyFunc apply, void * user_data);
+
+AVSC_API(int, avs_function_exists)(AVS_ScriptEnvironment *, const char * name);
+
+AVSC_API(AVS_Value, avs_invoke)(AVS_ScriptEnvironment *, const char * name,
+                               AVS_Value args, const char** arg_names);
+// The returned value must be be released with avs_release_value
+
+AVSC_API(AVS_Value, avs_get_var)(AVS_ScriptEnvironment *, const char* name);
+// The returned value must be be released with avs_release_value
+
+AVSC_API(int, avs_set_var)(AVS_ScriptEnvironment *, const char* name, AVS_Value val);
+
+AVSC_API(int, avs_set_global_var)(AVS_ScriptEnvironment *, const char* name, const AVS_Value val);
+
+//void avs_push_context(AVS_ScriptEnvironment *, int level=0);
+//void avs_pop_context(AVS_ScriptEnvironment *);
+
+AVSC_API(AVS_VideoFrame *, avs_new_video_frame_a)(AVS_ScriptEnvironment *,
+                                          const AVS_VideoInfo * vi, int align);
+// align should be at least 16
+#if defined __cplusplus
+}
+#endif // __cplusplus
+
+#ifndef AVSC_NO_DECLSPEC
+AVSC_INLINE
+AVS_VideoFrame * avs_new_video_frame(AVS_ScriptEnvironment * env,
+                                     const AVS_VideoInfo * vi)
+  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
+
+AVSC_INLINE
+AVS_VideoFrame * avs_new_frame(AVS_ScriptEnvironment * env,
+                               const AVS_VideoInfo * vi)
+  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
+#endif
+
+#if defined __cplusplus
+extern "C"
+{
+#endif // __cplusplus
+AVSC_API(int, avs_make_writable)(AVS_ScriptEnvironment *, AVS_VideoFrame * * pvf);
+
+AVSC_API(void, avs_bit_blt)(AVS_ScriptEnvironment *, unsigned char* dstp, int dst_pitch, const unsigned char* srcp, int src_pitch, int row_size, int height);
+
+typedef void (AVSC_CC *AVS_ShutdownFunc)(void* user_data, AVS_ScriptEnvironment * env);
+AVSC_API(void, avs_at_exit)(AVS_ScriptEnvironment *, AVS_ShutdownFunc function, void * user_data);
+
+AVSC_API(AVS_VideoFrame *, avs_subframe)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height);
+// The returned video frame must be be released
+
+AVSC_API(int, avs_set_memory_max)(AVS_ScriptEnvironment *, int mem);
+
+AVSC_API(int, avs_set_working_dir)(AVS_ScriptEnvironment *, const char * newdir);
+
+// avisynth.dll exports this; it's a way to use it as a library, without
+// writing an AVS script or without going through AVIFile.
+AVSC_API(AVS_ScriptEnvironment *, avs_create_script_environment)(int version);
+#if defined __cplusplus
+}
+#endif // __cplusplus
+
+// this symbol is the entry point for the plugin and must
+// be defined
+AVSC_EXPORT
+const char * AVSC_CC avisynth_c_plugin_init(AVS_ScriptEnvironment* env);
+
+
+#if defined __cplusplus
+extern "C"
+{
+#endif // __cplusplus
+AVSC_API(void, avs_delete_script_environment)(AVS_ScriptEnvironment *);
+
+
+AVSC_API(AVS_VideoFrame *, avs_subframe_planar)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height, int rel_offsetU, int rel_offsetV, int new_pitchUV);
+// The returned video frame must be be released
+#if defined __cplusplus
+}
+#endif // __cplusplus
+
+#endif //__AVXSYNTH_C__

compat/avisynth/windowsPorts/basicDataTypeConversions.h

@@ -0,0 +1,85 @@
+#ifndef __DATA_TYPE_CONVERSIONS_H__
+#define __DATA_TYPE_CONVERSIONS_H__
+
+#include <stdint.h>
+#include <wchar.h>
+
+#ifdef __cplusplus
+namespace avxsynth {
+#endif // __cplusplus
+
+typedef int64_t __int64;
+typedef int32_t __int32;
+#ifdef __cplusplus
+typedef bool BOOL;
+#else
+typedef uint32_t BOOL;
+#endif // __cplusplus
+typedef void* HMODULE;
+typedef void* LPVOID;
+typedef void* PVOID;
+typedef PVOID HANDLE;
+typedef HANDLE HWND;
+typedef HANDLE HINSTANCE;
+typedef void* HDC;
+typedef void* HBITMAP;
+typedef void* HICON;
+typedef void* HFONT;
+typedef void* HGDIOBJ;
+typedef void* HBRUSH;
+typedef void* HMMIO;
+typedef void* HACMSTREAM;
+typedef void* HACMDRIVER;
+typedef void* HIC;
+typedef void* HACMOBJ;
+typedef HACMSTREAM* LPHACMSTREAM;
+typedef void* HACMDRIVERID;
+typedef void* LPHACMDRIVER;
+typedef unsigned char BYTE;
+typedef BYTE* LPBYTE;
+typedef char TCHAR;
+typedef TCHAR* LPTSTR;
+typedef const TCHAR* LPCTSTR;
+typedef char* LPSTR;
+typedef LPSTR LPOLESTR;
+typedef const char* LPCSTR;
+typedef LPCSTR LPCOLESTR;
+typedef wchar_t WCHAR;
+typedef unsigned short WORD;
+typedef unsigned int UINT;
+typedef UINT MMRESULT;
+typedef uint32_t DWORD;
+typedef DWORD COLORREF;
+typedef DWORD FOURCC;
+typedef DWORD HRESULT;
+typedef DWORD* LPDWORD;
+typedef DWORD* DWORD_PTR;
+typedef int32_t LONG;
+typedef int32_t* LONG_PTR;
+typedef LONG_PTR LRESULT;
+typedef uint32_t ULONG;
+typedef uint32_t* ULONG_PTR;
+//typedef __int64_t intptr_t;
+typedef uint64_t _fsize_t;
+
+
+//
+// Structures
+//
+
+typedef struct _GUID {
+  DWORD Data1;
+  WORD  Data2;
+  WORD  Data3;
+  BYTE  Data4[8];
+} GUID;
+
+typedef GUID REFIID;
+typedef GUID CLSID;
+typedef CLSID* LPCLSID;
+typedef GUID IID;
+
+#ifdef __cplusplus
+}; // namespace avxsynth
+#endif // __cplusplus
+#endif //  __DATA_TYPE_CONVERSIONS_H__

compat/avisynth/windowsPorts/windows2linux.h

@@ -0,0 +1,77 @@
+#ifndef __WINDOWS2LINUX_H__
+#define __WINDOWS2LINUX_H__
+
+/*
+ * LINUX SPECIFIC DEFINITIONS
+*/
+//
+// Data types conversions
+//
+#include <stdlib.h>
+#include <string.h>
+#include "basicDataTypeConversions.h"
+
+#ifdef __cplusplus
+namespace avxsynth {
+#endif // __cplusplus
+//
+// purposefully define the following MSFT definitions
+// to mean nothing (as they do not mean anything on Linux)
+//
+#define __stdcall
+#define __cdecl
+#define noreturn
+#define __declspec(x)
+#define STDAPI       extern "C" HRESULT
+#define STDMETHODIMP HRESULT __stdcall
+#define STDMETHODIMP_(x) x __stdcall
+
+#define STDMETHOD(x)    virtual HRESULT x
+#define STDMETHOD_(a, x) virtual a x
+
+#ifndef TRUE
+#define TRUE  true
+#endif
+
+#ifndef FALSE
+#define FALSE false
+#endif
+
+#define S_OK                (0x00000000)
+#define S_FALSE             (0x00000001)
+#define E_NOINTERFACE       (0X80004002)
+#define E_POINTER           (0x80004003)
+#define E_FAIL              (0x80004005)
+#define E_OUTOFMEMORY       (0x8007000E)
+
+#define INVALID_HANDLE_VALUE    ((HANDLE)((LONG_PTR)-1))
+#define FAILED(hr)              ((hr) & 0x80000000)
+#define SUCCEEDED(hr)           (!FAILED(hr))
+
+
+//
+// Functions
+//
+#define MAKEDWORD(a,b,c,d) (((a) << 24) | ((b) << 16) | ((c) << 8) | (d))
+#define MAKEWORD(a,b) (((a) << 8) | (b))
+
+#define lstrlen                             strlen
+#define lstrcpy                             strcpy
+#define lstrcmpi                            strcasecmp
+#define _stricmp                            strcasecmp
+#define InterlockedIncrement(x)             __sync_fetch_and_add((x), 1)
+#define InterlockedDecrement(x)             __sync_fetch_and_sub((x), 1)
+// Windows uses (new, old) ordering but GCC has (old, new)
+#define InterlockedCompareExchange(x,y,z)   __sync_val_compare_and_swap(x,z,y)
+
+#define UInt32x32To64(a, b)                 ( (uint64_t) ( ((uint64_t)((uint32_t)(a))) * ((uint32_t)(b))  ) )
+#define Int64ShrlMod32(a, b)                ( (uint64_t) ( (uint64_t)(a) >> (b) ) )
+#define Int32x32To64(a, b)                  ((__int64)(((__int64)((long)(a))) * ((long)(b))))
+
+#define MulDiv(nNumber, nNumerator, nDenominator)   (int32_t) (((int64_t) (nNumber) * (int64_t) (nNumerator) + (int64_t) ((nDenominator)/2)) / (int64_t) (nDenominator))
+
+#ifdef __cplusplus
+}; // namespace avxsynth
+#endif // __cplusplus
+
+#endif //  __WINDOWS2LINUX_H__

compat/cuda/cuda_runtime.h

@@ -1,191 +0,0 @@
-/*
- * 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 sqrtf(float a) { return __builtin_sqrtf(a); }
-
-static inline __device__ float __saturatef(float a) { return __nvvm_saturate_f(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); }
-static inline __device__ float __expf(float a) { return __nvvm_ex2_approx_f(a * (float)__builtin_log2(__builtin_exp(1))); }
-
-#endif /* COMPAT_CUDA_CUDA_RUNTIME_H */

compat/cuda/dynlink_loader.h

@@ -16,8 +16,8 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#ifndef COMPAT_CUDA_DYNLINK_LOADER_H
-#define COMPAT_CUDA_DYNLINK_LOADER_H
+#ifndef AV_COMPAT_CUDA_DYNLINK_LOADER_H
+#define AV_COMPAT_CUDA_DYNLINK_LOADER_H
 
 #include "libavutil/log.h"
 #include "compat/w32dlfcn.h"
@@ -30,4 +30,4 @@
 
 #include <ffnvcodec/dynlink_loader.h>
 
-#endif /* COMPAT_CUDA_DYNLINK_LOADER_H */
+#endif

compat/cuda/ptx2c.sh

@@ -0,0 +1,36 @@
+#!/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" > "$OUT"
+while read LINE
+do
+    printf "\n\t\"%s\\\n\"" "$(printf "%s" "$LINE" | sed -e 's/\r//g' -e 's/["\\]/\\&/g')" >> "$OUT"
+done < "$IN"
+printf ";\n" >> "$OUT"
+
+exit 0

compat/djgpp/math.c

@@ -1,47 +0,0 @@
-/*
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#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);
-}

compat/djgpp/math.h

@@ -1,25 +0,0 @@
-/*
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-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);

compat/msvcrt/snprintf.c

@@ -59,7 +59,7 @@ int avpriv_vsnprintf(char *s, size_t n, const char *fmt,
      * 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 https://web.archive.org/web/20151214111935/http://msdn.microsoft.com/en-us/library/1kt27hek(v=vs.80).aspx */
+     * 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);

compat/os2threads.h

@@ -27,19 +27,15 @@
 #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;
@@ -167,28 +163,6 @@ static av_always_inline int pthread_cond_broadcast(pthread_cond_t *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)
 {

compat/w32dlfcn.h

@@ -20,40 +20,11 @@
 #define COMPAT_W32DLFCN_H
 
 #ifdef _WIN32
-#include <stdint.h>
-
 #include <windows.h>
-
 #include "config.h"
-#include "libavutil/macros.h"
+#if (_WIN32_WINNT < 0x0602) || HAVE_WINRT
 #include "libavutil/wchar_filename.h"
-
-static inline wchar_t *get_module_filename(HMODULE module)
-{
-    wchar_t *path = NULL, *new_path;
-    DWORD path_size = 0, path_len;
-
-    do {
-        path_size = path_size ? FFMIN(2 * path_size, INT16_MAX + 1) : MAX_PATH;
-        new_path = av_realloc_array(path, path_size, sizeof *path);
-        if (!new_path) {
-            av_free(path);
-            return NULL;
-        }
-        path = new_path;
-        // Returns path_size in case of insufficient buffer.
-        // Whether the error is set or not and whether the output
-        // is null-terminated or not depends on the version of Windows.
-        path_len = GetModuleFileNameW(module, path, path_size);
-    } while (path_len && path_size <= INT16_MAX && path_size <= path_len);
-
-    if (!path_len) {
-        av_free(path);
-        return NULL;
-    }
-    return path;
-}
-
+#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
@@ -63,53 +34,29 @@ static inline wchar_t *get_module_filename(HMODULE module)
  */
 static inline HMODULE win32_dlopen(const char *name)
 {
-    wchar_t *name_w;
-    HMODULE module = NULL;
-    if (utf8towchar(name, &name_w))
-        name_w = NULL;
 #if _WIN32_WINNT < 0x0602
-    // On Win7 and earlier we check if KB2533623 is available
+    // Need to check if KB2533623 is available
     if (!GetProcAddress(GetModuleHandleW(L"kernel32.dll"), "SetDefaultDllDirectories")) {
-        wchar_t *path = NULL, *new_path;
-        DWORD pathlen, pathsize, namelen;
-        if (!name_w)
+        HMODULE module = NULL;
+        wchar_t *path = NULL, *name_w = NULL;
+        DWORD pathlen;
+        if (utf8towchar(name, &name_w))
             goto exit;
-        namelen = wcslen(name_w);
+        path = (wchar_t *)av_mallocz_array(MAX_PATH, sizeof(wchar_t));
         // Try local directory first
-        path = get_module_filename(NULL);
-        if (!path)
+        pathlen = GetModuleFileNameW(NULL, path, MAX_PATH);
+        pathlen = wcsrchr(path, '\\') - path;
+        if (pathlen == 0 || pathlen + wcslen(name_w) + 2 > MAX_PATH)
             goto exit;
-        new_path = wcsrchr(path, '\\');
-        if (!new_path)
-            goto exit;
-        pathlen = new_path - path;
-        pathsize = pathlen + namelen + 2;
-        new_path = av_realloc_array(path, pathsize, sizeof *path);
-        if (!new_path)
-            goto exit;
-        path = new_path;
+        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, pathsize);
-            if (!pathlen)
+            pathlen = GetSystemDirectoryW(path, MAX_PATH);
+            if (pathlen == 0 || pathlen + wcslen(name_w) + 2 > MAX_PATH)
                 goto exit;
-            // Buffer is not enough in two cases:
-            // 1. system directory + \ + module name
-            // 2. system directory even without the module name.
-            if (pathlen + namelen + 2 > pathsize) {
-                pathsize = pathlen + namelen + 2;
-                new_path = av_realloc_array(path, pathsize, sizeof *path);
-                if (!new_path)
-                    goto exit;
-                path = new_path;
-                // Query again to handle the case #2.
-                pathlen = GetSystemDirectoryW(path, pathsize);
-                if (!pathlen)
-                    goto exit;
-            }
-            path[pathlen] = L'\\';
+            path[pathlen] = '\\';
             wcscpy(path + pathlen + 1, name_w);
             module = LoadLibraryExW(path, NULL, LOAD_WITH_ALTERED_SEARCH_PATH);
         }
@@ -126,19 +73,16 @@ exit:
 #   define LOAD_LIBRARY_SEARCH_SYSTEM32        0x00000800
 #endif
 #if HAVE_WINRT
-    if (!name_w)
+    wchar_t *name_w = NULL;
+    int ret;
+    if (utf8towchar(name, &name_w))
         return NULL;
-    module = LoadPackagedLibrary(name_w, 0);
-#else
-#define LOAD_FLAGS (LOAD_LIBRARY_SEARCH_APPLICATION_DIR | LOAD_LIBRARY_SEARCH_SYSTEM32)
-    /* filename may be be in CP_ACP */
-    if (!name_w)
-        return LoadLibraryExA(name, NULL, LOAD_FLAGS);
-    module = LoadLibraryExW(name_w, NULL, LOAD_FLAGS);
-#undef LOAD_FLAGS
-#endif
+    ret = LoadPackagedLibrary(name_w, 0);
     av_free(name_w);
-    return module;
+    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

compat/w32pthreads.h

@@ -38,13 +38,11 @@
 #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;
@@ -63,9 +61,6 @@ typedef CONDITION_VARIABLE pthread_cond_t;
 #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;
@@ -161,31 +156,10 @@ static inline int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex
     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 */

compat/windows/makedef

@@ -48,7 +48,7 @@ trap 'rm -f -- $libname' EXIT
 if [ -n "$AR" ]; then
     $AR rcs ${libname} $@ >/dev/null
 else
-    lib.exe -out:${libname} $@ >/dev/null
+    lib -out:${libname} $@ >/dev/null
 fi
 if [ $? != 0 ]; then
     echo "Could not create temporary library." >&2
@@ -108,7 +108,7 @@ if [ -n "$NM" ]; then
               cut -d' ' -f3 |
               sed -e "s/^${prefix}//")
 else
-    dump=$(dumpbin.exe -linkermember:1 ${libname} |
+    dump=$(dumpbin -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)

compat/windows/mslink

@@ -4,6 +4,6 @@ LINK_EXE_PATH=$(dirname "$(command -v cl)")/link
 if [ -x "$LINK_EXE_PATH" ]; then
     "$LINK_EXE_PATH" $@
 else
-    link.exe $@
+    link $@
 fi
 exit $?

doc/APIchanges

@@ -2,571 +2,19 @@ Never assume the API of libav* to be stable unless at least 1 month has passed
 since the last major version increase or the API was added.
 
 The last version increases were:
-libavcodec:    2021-04-27
-libavdevice:   2021-04-27
-libavfilter:   2021-04-27
-libavformat:   2021-04-27
-libpostproc:   2021-04-27
-libswresample: 2021-04-27
-libswscale:    2021-04-27
-libavutil:     2021-04-27
+libavcodec:    2017-10-21
+libavdevice:   2017-10-21
+libavfilter:   2017-10-21
+libavformat:   2017-10-21
+libavresample: 2017-10-21
+libpostproc:   2017-10-21
+libswresample: 2017-10-21
+libswscale:    2017-10-21
+libavutil:     2017-10-21
 
 
 API changes, most recent first:
 
--------- 8< --------- FFmpeg 5.1 was cut here -------- 8< ---------
-
-2022-06-12 - 7cae3d8b76 - lavf 59.25.100 - avio.h
-  Add avio_vprintf(), similar to avio_printf() but allow to use it
-  from within a function taking a variable argument list as input.
-
-2022-06-12 - ff59ecc4de - lavu 57.27.100 - uuid.h
-  Add UUID handling functions.
-  Add av_uuid_parse(), av_uuid_urn_parse(), av_uuid_parse_range(),
-  av_uuid_parse_range(), av_uuid_equal(), av_uuid_copy(), and av_uuid_nil().
-
-2022-06-01 - d42b410e05 - lavu 57.26.100 - csp.h
-  Add public API for colorspace structs.
-  Add av_csp_luma_coeffs_from_avcsp(), av_csp_primaries_desc_from_id(),
-  and av_csp_primaries_id_from_desc().
-
-2022-05-23 - 4cdc14aa95 - lavu 57.25.100 - avutil.h
-  Deprecate av_fopen_utf8() without replacement.
-
-2022-03-16 - f3a0e2ee2b - all libraries - version_major.h
-  Add lib<name>/version_major.h as new installed headers, which only
-  contain the major version number (and corresponding API deprecation
-  defines).
-
-2022-03-15 - cdba98bb80 - swr 4.5.100 - swresample.h
-  Add swr_alloc_set_opts2() and swr_build_matrix2().
-  Deprecate swr_alloc_set_opts() and swr_build_matrix().
-
-2022-03-15 - cdba98bb80 - lavfi 8.28.100 - avfilter.h buffersink.h buffersrc.h
-  Update AVFilterLink for the new channel layout API: add ch_layout,
-  deprecate channel_layout.
-
-  Update the buffersink filter sink for the new channel layout API:
-  add av_buffersink_get_ch_layout() and the ch_layouts option,
-  deprecate av_buffersink_get_channel_layout() and the channel_layouts option.
-
-  Update AVBufferSrcParameters for the new channel layout API:
-  add ch_layout, deprecate channel_layout.
-
-2022-03-15 - cdba98bb80 - lavf 59.19.100 - avformat.h
-  Add AV_DISPOSITION_NON_DIEGETIC.
-
-2022-03-15 - cdba98bb80 - lavc 59.24.100 - avcodec.h codec_par.h
-  Update AVCodecParameters for the new channel layout API: add ch_layout,
-  deprecate channels/channel_layout.
-
-  Update AVCodecContext for the new channel layout API: add ch_layout,
-  deprecate channels/channel_layout.
-
-  Update AVCodec for the new channel layout API: add ch_layouts,
-  deprecate channel_layouts.
-
-2022-03-15 - cdba98bb80 - lavu 57.24.100 - channel_layout.h frame.h opt.h
-  Add new channel layout API based on the AVChannelLayout struct.
-  Add support for Ambisonic audio.
-  Deprecate previous channel layout API based on uint64 bitmasks.
-
-  Add AV_OPT_TYPE_CHLAYOUT option type, deprecate AV_OPT_TYPE_CHANNEL_LAYOUT.
-  Update AVFrame for the new channel layout API: add ch_layout, deprecate
-  channels/channel_layout.
-
-2022-03-10 - f629ea2e18 - lavu 57.23.100 - cpu.h
-  Add AV_CPU_FLAG_AVX512ICL.
-
-2022-02-07 - a10f1aec1f - lavu 57.21.100 - fifo.h
-  Deprecate AVFifoBuffer and the API around it, namely av_fifo_alloc(),
-  av_fifo_alloc_array(), av_fifo_free(), av_fifo_freep(), av_fifo_reset(),
-  av_fifo_size(), av_fifo_space(), av_fifo_generic_peek_at(),
-  av_fifo_generic_peek(), av_fifo_generic_read(), av_fifo_generic_write(),
-  av_fifo_realloc2(), av_fifo_grow(), av_fifo_drain() and av_fifo_peek2().
-  Users should switch to the AVFifo-API.
-
-2022-02-07 - 7329b22c05 - lavu 57.20.100 - fifo.h
-  Add a new FIFO API, which allows setting a FIFO element size.
-  This API operates on these elements rather than on bytes.
-  Add av_fifo_alloc2(), av_fifo_elem_size(), av_fifo_can_read(),
-  av_fifo_can_write(), av_fifo_grow2(), av_fifo_drain2(), av_fifo_write(),
-  av_fifo_write_from_cb(), av_fifo_read(), av_fifo_read_to_cb(),
-  av_fifo_peek(), av_fifo_peek_to_cb(), av_fifo_drain2(), av_fifo_reset2(),
-  av_fifo_freep2(), av_fifo_auto_grow_limit().
-
-2022-01-26 - af94ab7c7c0 - lavu 57.19.100 - tx.h
-  Add AV_TX_FLOAT_RDFT, AV_TX_DOUBLE_RDFT and AV_TX_INT32_RDFT.
-
--------- 8< --------- FFmpeg 5.0 was cut here -------- 8< ---------
-
-2022-01-04 - 78dc21b123e - lavu 57.16.100 - frame.h
-  Add AV_FRAME_DATA_DOVI_METADATA.
-
-2022-01-03 - 70f318e6b6c - lavf 59.13.100 - avformat.h
-  Add AVFMT_EXPERIMENTAL flag.
-
-2021-12-22 - b7e1ec7bda9 - lavu 57.13.100 - hwcontext_videotoolbox.h
-  Add av_vt_pixbuf_set_attachments
-
-2021-12-22 - 69bd95dcd8d - lavu 57.13.100 - hwcontext_videotoolbox.h
-  Add av_map_videotoolbox_chroma_loc_from_av
-  Add av_map_videotoolbox_color_matrix_from_av
-  Add av_map_videotoolbox_color_primaries_from_av
-  Add av_map_videotoolbox_color_trc_from_av
-
-2021-12-21 - ffbab99f2c2 - lavu 57.12.100 - cpu.h
-  Add AV_CPU_FLAG_SLOW_GATHER.
-
-2021-12-20 - 278068dc60d - lavu 57.11.101 - display.h
-  Modified the documentation of av_display_rotation_set()
-  to match its longstanding actual behaviour of treating
-  the angle as directed clockwise.
-
-2021-12-12 - 64834bb86a1 - lavf 59.10.100 - avformat.h
-  Add AVFormatContext io_close2 which returns an int
-
-2021-12-10 - f45cbb775e4 - lavu 57.11.100 - hwcontext_vulkan.h
-  Add AVVkFrame.offset and AVVulkanFramesContext.flags.
-
-2021-12-04 - b9c928a486f - lavfi 8.19.100 - avfilter.h
-  Add AVFILTER_FLAG_METADATA_ONLY.
-
-2021-12-03 - b236ef0a594 - lavu 57.10.100 - frame.h
-  Add AVFrame.time_base
-
-2021-11-22 - b2cd1fb2ec6 - lavu 57.9.100 - pixfmt.h
-  Add AV_PIX_FMT_P210, AV_PIX_FMT_P410, AV_PIX_FMT_P216, and AV_PIX_FMT_P416.
-
-2021-11-17 - 54e65aa38ab - lavf 57.9.100 - frame.h
-  Add AV_FRAME_DATA_DOVI_RPU_BUFFER.
-
-2021-11-16 - ed75a08d36c - lavf 59.9.100 - avformat.h
-  Add av_stream_get_class(). Schedule adding AVStream.av_class at libavformat
-  major version 60.
-  Add av_disposition_to_string() and av_disposition_from_string().
-  Add "disposition" AVOption to AVStream's class.
-
-2021-11-12 - 8478d60d5b5 - lavu 57.8.100 - hwcontext_vulkan.h
-  Added AVVkFrame.sem_value, AVVulkanDeviceContext.queue_family_encode_index,
-  nb_encode_queues, queue_family_decode_index, and nb_decode_queues.
-
-2021-10-18 - 682bafdb125 - lavf 59.8.100 - avio.h
-  Introduce public bytes_{read,written} statistic fields to AVIOContext.
-
-2021-10-13 - a5622ed16f8 - lavf 59.7.100 - avio.h
-  Deprecate AVIOContext.written. Originally added as a private entry in
-  commit 3f75e5116b900f1428aa13041fc7d6301bf1988a, its grouping with
-  the comment noting its private state was missed during merging of the field
-  from Libav (most likely due to an already existing field in between).
-
-2021-09-21 - 0760d9153c3 - lavu 57.7.100 - pixfmt.h
-  Add AV_PIX_FMT_X2BGR10.
-
-2021-09-20 - 8d5de914d31 - lavu 57.6.100 - mem.h
-  Deprecate av_mallocz_array() as it is identical to av_calloc().
-
-2021-09-20 - 176b8d785bf - lavc 59.9.100 - avcodec.h
-  Deprecate AVCodecContext.sub_text_format and the corresponding
-  AVOptions. It is unused since the last major bump.
-
-2021-09-20 - dd846bc4a91 - lavc 59.8.100 - avcodec.h codec.h
-  Deprecate AV_CODEC_FLAG_TRUNCATED and AV_CODEC_CAP_TRUNCATED,
-  as they are redundant with parsers.
-
-2021-09-17 - ccfdef79b13 - lavu 57.5.101 - buffer.h
-  Constified the input parameters in av_buffer_replace(), av_buffer_ref(),
-  and av_buffer_pool_buffer_get_opaque().
-
-2021-09-08 - 4f78711f9c2 - lavu 57.5.100 - hwcontext_d3d11va.h
-  Add AVD3D11VAFramesContext.texture_infos
-
-2021-09-06 - 42cd64c1826 - lsws 6.1.100 - swscale.h
-  Add AVFrame-based scaling API:
-    - sws_scale_frame()
-    - sws_frame_start()
-    - sws_frame_end()
-    - sws_send_slice()
-    - sws_receive_slice()
-    - sws_receive_slice_alignment()
-
-2021-09-02 - cbf111059d2 - lavc 59.7.100 - avcodec.h
-  Incremented the number of elements of AVCodecParser.codec_ids to seven.
-
-2021-08-24 - 590a7e02f04 - lavc 59.6.100 - avcodec.h
-  Add FF_CODEC_PROPERTY_FILM_GRAIN
-
-2021-08-20 - 7c5f998196d - lavfi 8.3.100 - avfilter.H
-  Add avfilter_filter_pad_count() as a replacement for avfilter_pad_count().
-  Deprecate avfilter_pad_count().
-
-2021-08-17 - 8c53b145993 - lavu 57.4.101 - opt.h
-  av_opt_copy() now guarantees that allocated src and dst options
-  don't alias each other even on error.
-
-2021-08-14 - d5de9965ef6 - lavu 57.4.100 - imgutils.h
-  Add av_image_copy_plane_uc_from()
-
-2021-08-02 - a1a0fddfd05 - lavc 59.4.100 - packet.h
-  Add AVPacket.opaque, AVPacket.opaque_ref, AVPacket.time_base.
-
-2021-07-23 - 2dd8acbe800 - lavu 57.3.100 - common.h macros.h
-  Move several macros (AV_NE, FFDIFFSIGN, FFMAX, FFMAX3, FFMIN, FFMIN3,
-  FFSWAP, FF_ARRAY_ELEMS, MKTAG, MKBETAG) from common.h to macros.h.
-
-2021-07-22 - e3b5ff17c2e - lavu 57.2.100 - film_grain_params.h
-  Add AV_FILM_GRAIN_PARAMS_H274, AVFilmGrainH274Params
-
-2021-07-19 - c1bf56a526f - lavu 57.1.100 - cpu.h
-  Add av_cpu_force_count()
-
-2021-06-17 - aca923b3653 - lavc 59.2.100 - packet.h
-  Add AV_PKT_DATA_DYNAMIC_HDR10_PLUS
-
-2021-06-09 - 2cccab96f6f - lavf 59.3.100 - avformat.h
-  Add pts_wrap_bits to AVStream
-
-2021-06-10 - 7c9763070d9 - lavc 59.1.100 - avcodec.h codec.h
-  Move av_get_profile_name() from avcodec.h to codec.h.
-
-2021-06-10 - bb3648e6766 - lavc 59.1.100 - avcodec.h codec_par.h
-  Move av_get_audio_frame_duration2() from avcodec.h to codec_par.h.
-
-2021-06-10 - 881db34f6a0 - lavc 59.1.100 - avcodec.h codec_id.h
-  Move av_get_bits_per_sample(), av_get_exact_bits_per_sample(),
-  avcodec_profile_name(), and av_get_pcm_codec() from avcodec.h
-  to codec_id.h.
-
-2021-06-10 - ff0a96046d8 - lavc 59.1.100 - avcodec.h defs.h
-  Add new installed header defs.h. The following definitions are moved
-  into it from avcodec.h:
-    - AVDiscard
-    - AVAudioServiceType
-    - AVPanScan
-    - AVCPBProperties and av_cpb_properties_alloc()
-    - AVProducerReferenceTime
-    - av_xiphlacing()
-
-2021-04-27 - cb3ac722f4 - lavc 59.0.100 - avcodec.h
-  Constified AVCodecParserContext.parser.
-
-2021-04-27 - 8b3e6ce5f4 - lavd 59.0.100 - avdevice.h
-  The av_*_device_next API functions now accept and return
-  pointers to const AVInputFormat resp. AVOutputFormat.
-
-2021-04-27 - d7e0d428fa - lavd 59.0.100 - avdevice.h
-  avdevice_list_input_sources and avdevice_list_output_sinks now accept
-  pointers to const AVInputFormat resp. const AVOutputFormat.
-
-2021-04-27 - 46dac8cf3d - lavf 59.0.100 - avformat.h
-  av_find_best_stream now uses a const AVCodec ** parameter
-  for the returned decoder.
-
-2021-04-27 - 626535f6a1 - lavc 59.0.100 - codec.h
-  avcodec_find_encoder_by_name(), avcodec_find_encoder(),
-  avcodec_find_decoder_by_name() and avcodec_find_decoder()
-  now return a pointer to const AVCodec.
-
-2021-04-27 - 14fa0a4efb - lavf 59.0.100 - avformat.h
-  Constified AVFormatContext.*_codec.
-
-2021-04-27 - 56450a0ee4 - lavf 59.0.100 - avformat.h
-  Constified the pointers to AVInputFormats and AVOutputFormats
-  in AVFormatContext, avformat_alloc_output_context2(),
-  av_find_input_format(), av_probe_input_format(),
-  av_probe_input_format2(), av_probe_input_format3(),
-  av_probe_input_buffer2(), av_probe_input_buffer(),
-  avformat_open_input(), av_guess_format() and av_guess_codec().
-  Furthermore, constified the AVProbeData in av_probe_input_format(),
-  av_probe_input_format2() and av_probe_input_format3().
-
-2021-04-19 - 18af1ea8d1 - lavu 56.74.100 - tx.h
-  Add AV_TX_FULL_IMDCT and AV_TX_UNALIGNED.
-
-2021-04-17 - f1bf465aa0 - lavu 56.73.100 - frame.h detection_bbox.h
-  Add AV_FRAME_DATA_DETECTION_BBOXES
-
-2021-04-06 - 557953a397 - lavf 58.78.100 - avformat.h
-  Add avformat_index_get_entries_count(), avformat_index_get_entry(),
-  and avformat_index_get_entry_from_timestamp().
-
-2021-03-21 - a77beea6c8 - lavu 56.72.100 - frame.h
-  Deprecated av_get_colorspace_name().
-  Use av_color_space_name() instead.
-
--------- 8< --------- FFmpeg 4.4 was cut here -------- 8< ---------
-
-2021-03-19 - e8c0bca6bd - lavu 56.69.100 - adler32.h
-  Added a typedef for the type of the Adler-32 checksums
-  used by av_adler32_update(). It will be changed to uint32_t
-  at the next major bump.
-  The type of the parameter for the length of the input buffer
-  will also be changed to size_t at the next major bump.
-
-2021-03-19 - e318438f2f - lavf 58.75.100  - avformat.h
-  AVChapter.id will be changed from int to int64_t
-  on the next major version bump.
-
-2021-03-17 - f7db77bd87 - lavc 58.133.100 - codec.h
-  Deprecated av_init_packet(). Once removed, sizeof(AVPacket) will
-  no longer be a part of the public ABI.
-  Deprecated AVPacketList.
-
-2021-03-16 - 7d09579190 - lavc 58.132.100 - codec.h
-  Add AV_CODEC_CAP_OTHER_THREADS as a new name for
-  AV_CODEC_CAP_AUTO_THREADS. AV_CODEC_CAP_AUTO_THREADS
-  is now deprecated.
-
-2021-03-12 - 6e7e3a3820 - lavc 58.131.100 - avcodec.h codec.h
-  Add a get_encode_buffer callback to AVCodecContext, similar to
-  get_buffer2 but for encoders.
-  Add avcodec_default_get_encode_buffer().
-  Add AV_GET_ENCODE_BUFFER_FLAG_REF.
-  Encoders may now be flagged as AV_CODEC_CAP_DR1 capable.
-
-2021-03-10 - 42e68fe015 - lavf 58.72.100 - avformat.h
-  Change AVBufferRef related AVStream function and struct size
-  parameter and fields type to size_t at next major bump.
-
-2021-03-10 - d79e0fe65c - lavc 58.130.100 - packet.h
-  Change AVBufferRef related AVPacket function and struct size
-  parameter and fields type to size_t at next major bump.
-
-2021-03-10 - 14040a1d91 - lavu 56.68.100 - buffer.h frame.h
-  Change AVBufferRef and relevant AVFrame function and struct size
-  parameter and fields type to size_t at next major bump.
-
-2021-03-04 - a0eec776b6 - lavc 58.128.101 - avcodec.h
-  Enable err_recognition to be set for encoders.
-
-2021-03-03 - 2ff40b98ec - lavf 58.70.100 - avformat.h
-  Deprecate AVFMT_FLAG_PRIV_OPT. It will do nothing
-  as soon as av_demuxer_open() is removed.
-
-2021-02-27 - dd9227e48f - lavc 58.126.100 - avcodec.h
-  Deprecated avcodec_get_frame_class().
-
-2021-02-21 - 5ca40d6d94 - lavu 56.66.100 - tx.h
-  Add enum AVTXFlags and AVTXFlags.AV_TX_INPLACE
-
-2021-02-14 - 4f49ca7bbc - lavd 58.12.100 - avdevice.h
-  Deprecated avdevice_capabilities_create() and
-  avdevice_capabilities_free().
-
-2021-02-10 - 1bda9bb68a - lavu 56.65.100 - common.h
-  Add FFABS64U()
-
-2021-01-26 - 5dd9567080 - lavu 56.64.100 - common.h
-  Add FFABSU()
-
-2021-01-25 - 56709ca8aa - lavc 58.119.100 - avcodec.h
-  Deprecate AVCodecContext.debug_mv, FF_DEBUG_VIS_MV_P_FOR, FF_DEBUG_VIS_MV_B_FOR,
-  FF_DEBUG_VIS_MV_B_BACK
-
-2021-01-11 - ebdd33086a - lavc 58.116.100 - avcodec.h
-  Add FF_PROFILE_VVC_MAIN_10 and FF_PROFILE_VVC_MAIN_10_444.
-
-2020-01-01 - baecaa16c1 - lavu 56.63.100 - video_enc_params.h
-  Add AV_VIDEO_ENC_PARAMS_MPEG2
-
-2020-12-03 - eca12f4d5a - lavu 56.62.100 - timecode.h
-  Add av_timecode_init_from_components.
-
-2020-11-27 - a83098ab03 - lavc 58.114.100 - avcodec.h
-  Deprecate AVCodecContext.thread_safe_callbacks. Starting with
-  LIBAVCODEC_VERSION_MAJOR=60, user callbacks must always be
-  thread-safe when frame threading is used.
-
-2020-11-25 - d243dd540a - lavc 58.113.100 - avcodec.h
-  Adds a new flag AV_CODEC_EXPORT_DATA_FILM_GRAIN for export_side_data.
-
-2020-11-25 - 4f9ee87253 - lavu 56.61.100 - film_grain_params.h
-  Adds a new API for extracting codec film grain parameters as side data.
-  Adds a new AVFrameSideDataType entry AV_FRAME_DATA_FILM_GRAIN_PARAMS for it.
-
-2020-10-28 - f95d9510ff - lavf 58.64.100 - avformat.h
-  Add AVSTREAM_EVENT_FLAG_NEW_PACKETS.
-
-2020-09-28 - 68918d3b7f - lavu 56.60.100 - buffer.h
-  Add a av_buffer_replace() convenience function.
-
-2020-09-13 - 837b6eb90e - lavu 56.59.100 - timecode.h
-  Add av_timecode_make_smpte_tc_string2.
-
-2020-08-21 - 06f2651204 - lavu 56.58.100 - avstring.h
-  Deprecate av_d2str(). Use av_asprintf() instead.
-
-2020-08-04 - 34de0abbe7 - lavu 56.58.100 - channel_layout.h
-  Add AV_CH_LAYOUT_22POINT2 together with its newly required pieces:
-  AV_CH_TOP_SIDE_LEFT, AV_CH_TOP_SIDE_RIGHT, AV_CH_BOTTOM_FRONT_CENTER,
-  AV_CH_BOTTOM_FRONT_LEFT, AV_CH_BOTTOM_FRONT_RIGHT.
-
-2020-07-23 - 84655b7101 - lavu 56.57.100 - cpu.h
-  Add AV_CPU_FLAG_MMI and AV_CPU_FLAG_MSA.
-
-2020-07-22 - 3a8e927176 - lavu 56.56.100 - imgutils.h
-  Add av_image_fill_plane_sizes().
-
-2020-07-15 - 448a9aaa78 - lavc 58.96.100 - packet.h
-  Add AV_PKT_DATA_S12M_TIMECODE.
-
-2020-06-12 - b09fb030c1 - lavu 56.55.100 - pixdesc.h
-  Add AV_PIX_FMT_X2RGB10.
-
-2020-06-11 - bc8ab084fb - lavu 56.54.100 - frame.h
-  Add AV_FRAME_DATA_SEI_UNREGISTERED.
-
-2020-06-10 - 1b4a98b029 - lavu 56.53.100 - log.h opt.h
-  Add av_opt_child_class_iterate() and AVClass.child_class_iterate().
-  Deprecate av_opt_child_class_next() and AVClass.child_class_next().
-
--------- 8< --------- FFmpeg 4.3 was cut here -------- 8< ---------
-
-2020-06-05 - ec39c2276a - lavu 56.50.100 - buffer.h
-  Passing NULL as alloc argument to av_buffer_pool_init2() is now allowed.
-
-2020-05-27 - ba6cada92e - lavc 58.88.100 - avcodec.h codec.h
-  Move AVCodec-related public API to new header codec.h.
-
-2020-05-23 - 064b875e89 - lavu 56.49.100 - video_enc_params.h
-  Add AV_VIDEO_ENC_PARAMS_H264.
-
-2020-05-23 - 2e08b39444 - lavu 56.48.100 - hwcontext.h
-  Add av_hwdevice_ctx_create_derived_opts.
-
-2020-05-23 - 6b65c4ec54 - lavu 56.47.100 - rational.h
-  Add av_gcd_q().
-
-2020-05-22 - af9e622776 - lavu 56.46.101 - opt.h
-  Add AV_OPT_FLAG_CHILD_CONSTS.
-
-2020-05-22 - 9d443c3e68 - lavc 58.87.100 - avcodec.h codec_par.h
-  Move AVBitstreamFilter-related public API to new header bsf.h.
-  Move AVCodecParameters-related public API to new header codec_par.h.
-
-2020-05-21 - 13b1bbff0b - lavc 58.86.101 - avcodec.h
-  Deprecated AV_CODEC_CAP_INTRA_ONLY and AV_CODEC_CAP_LOSSLESS.
-
-2020-05-17 - 84af196c65 - lavu 56.46.100 - common.h
-  Add av_sat_add64() and av_sat_sub64()
-
-2020-05-12 - 991d417692 - lavu 56.45.100 - video_enc_params.h
-                          lavc 58.84.100 - avcodec.h
-  Add a new API for exporting video encoding information.
-  Replaces the deprecated API for exporting QP tables from decoders.
-  Add AV_CODEC_EXPORT_DATA_VIDEO_ENC_PARAMS to request this information from
-  decoders.
-
-2020-05-10 - dccd07f66d - lavu 56.44.100 - hwcontext_vulkan.h
-  Add enabled_inst_extensions, num_enabled_inst_extensions, enabled_dev_extensions
-  and num_enabled_dev_extensions fields to AVVulkanDeviceContext
-
-2020-04-22 - 0e1db79e37 - lavc 58.81.100 - packet.h
-                        - lavu 56.43.100 - dovi_meta.h
-  Add AV_PKT_DATA_DOVI_CONF and AVDOVIDecoderConfigurationRecord.
-
-2020-04-15 - 22b25b3ea5 - lavc 58.79.100 - avcodec.h
-  Add formal support for calling avcodec_flush_buffers() on encoders.
-  Encoders that set the cap AV_CODEC_CAP_ENCODER_FLUSH will be flushed.
-  For all other encoders, the call is now a no-op rather than undefined
-  behaviour.
-
-2020-04-10 - 672946c7fe - lavc 58.78.100 - avcodec.h codec_desc.h codec_id.h packet.h
-  Move AVCodecDesc-related public API to new header codec_desc.h.
-  Move AVCodecID enum to new header codec_id.h.
-  Move AVPacket-related public API to new header packet.h.
-
-2020-03-29 - 4cb0dda555 - lavf 58.42.100 - avformat.h
-  av_read_frame() now guarantees to handle uninitialized input packets
-  and to return refcounted packets on success.
-
-2020-03-27 - c52ec0367d - lavc 58.77.100 - avcodec.h
-  av_packet_ref() now guarantees to return the destination packet
-  in a blank state on error.
-
-2020-03-10 - 05d27f342b - lavc 58.75.100 - avcodec.h
-  Add AV_PKT_DATA_ICC_PROFILE.
-
-2020-02-21 - d005a7cdfd - lavc 58.73.101 - avcodec.h
-  Add AV_CODEC_EXPORT_DATA_PRFT.
-
-2020-02-21 - c666689491 - lavc 58.73.100 - avcodec.h
-  Add AVCodecContext.export_side_data and AV_CODEC_EXPORT_DATA_MVS.
-
-2020-02-13 - e8f054b095 - lavu 56.41.100 - tx.h
-  Add AV_TX_INT32_FFT and AV_TX_INT32_MDCT
-
-2020-02-12 - 3182114f88 - lavu 56.40.100 - log.h
-  Add av_log_once().
-
-2020-02-04 - a88449ffb2 - lavu 56.39.100 - hwcontext.h
-  Add AV_PIX_FMT_VULKAN
-  Add AV_HWDEVICE_TYPE_VULKAN and implementation.
-
-2020-01-30 - 27529eeb27 - lavf 58.37.100 - avio.h
-  Add avio_protocol_get_class().
-
-2020-01-15 - 717b2074ec - lavc 58.66.100 - avcodec.h
-  Add AV_PKT_DATA_PRFT and AVProducerReferenceTime.
-
-2019-12-27 - 45259a0ee4 - lavu 56.38.100 - eval.h
-  Add av_expr_count_func().
-
-2019-12-26 - 16685114d5 - lavu 56.37.100 - buffer.h
-  Add av_buffer_pool_buffer_get_opaque().
-
-2019-11-17 - 1c23abc88f - lavu 56.36.100 - eval API
-  Add av_expr_count_vars().
-
-2019-10-14 - f3746d31f9 - lavu 56.35.101 - opt.h
-  Add AV_OPT_FLAG_RUNTIME_PARAM.
-
-2019-09-25 - f8406ab4b9 - lavc 58.59.100 - avcodec.h
-  Add max_samples
-
-2019-09-04 - 2a9d461abc - lavu 56.35.100 - hwcontext_videotoolbox.h
-  Add av_map_videotoolbox_format_from_pixfmt2() for full range pixfmt
-
-2019-09-01 - 8821d1f56e - lavu 56.34.100 - pixfmt.h
-  Add EBU Tech. 3213-E AVColorPrimaries value
-
-2019-08-17 - 95fa73a2b4 - lavf 58.31.101 - avio.h
-  4K limit removed from avio_printf.
-
-2019-08-17 - a82f8f2f10 - lavf 58.31.100 - avio.h
-  Add avio_print_string_array and avio_print.
-
-2019-07-27 - 42e2319ba9 - lavu 56.33.100 - tx.h
-  Add AV_TX_DOUBLE_FFT and AV_TX_DOUBLE_MDCT
-
--------- 8< --------- FFmpeg 4.2 was cut here -------- 8< ---------
-
-2019-06-21 - a30e44098a - lavu 56.30.100 - frame.h
-  Add FF_DECODE_ERROR_DECODE_SLICES
-
-2019-06-14 - edfced8c04 - lavu 56.29.100 - frame.h
-  Add FF_DECODE_ERROR_CONCEALMENT_ACTIVE
-
-2019-05-15 - b79b29ddb1 - lavu 56.28.100 - tx.h
-  Add av_tx_init(), av_tx_uninit() and related definitions.
-
-2019-04-20 - 3153a6502a - lavc 58.52.100 - avcodec.h
-  Add AV_CODEC_FLAG_DROPCHANGED to allow avcodec_receive_frame to drop
-  frames whose parameters differ from first decoded frame in stream.
-
-2019-04-12 - abfeba9724 - lavf 58.27.102
-  Rename hls,applehttp demuxer to hls
-
-2019-01-27 - 5bcefceec8 - lavc 58.46.100 - avcodec.h
-  Add discard_damaged_percentage
-
-2019-01-08 - 1ef4828276 - lavu 56.26.100 - frame.h
-  Add AV_FRAME_DATA_REGIONS_OF_INTEREST
-
-2018-12-21 - 2744d6b364 - lavu 56.25.100 - hdr_dynamic_metadata.h
-  Add AV_FRAME_DATA_DYNAMIC_HDR_PLUS enum value, av_dynamic_hdr_plus_alloc(),
-  av_dynamic_hdr_plus_create_side_data() functions, and related structs.
-
 -------- 8< --------- FFmpeg 4.1 was cut here -------- 8< ---------
 
 2018-10-27 - 718044dc19 - lavu 56.21.100 - pixdesc.h
@@ -1689,7 +1137,7 @@ API changes, most recent first:
 2014-04-15 - ef818d8 - lavf 55.37.101 - avformat.h
   Add av_format_inject_global_side_data()
 
-2014-04-12 - 4f698be8f - lavu 52.76.100 - log.h
+2014-04-12 - 4f698be - lavu 52.76.100 - log.h
   Add av_log_get_flags()
 
 2014-04-11 - 6db42a2b - lavd 55.12.100 - avdevice.h

doc/Doxyfile

@@ -38,7 +38,7 @@ PROJECT_NAME           = FFmpeg
 # could be handy for archiving the generated documentation or if some version
 # control system is used.
 
-PROJECT_NUMBER         = 5.1.2
+PROJECT_NUMBER         = 4.1.6
 
 # Using the PROJECT_BRIEF tag one can provide an optional one line description
 # for a project that appears at the top of each page and should give viewer a

doc/Makefile

@@ -27,9 +27,6 @@ HTMLPAGES   = $(AVPROGS-yes:%=doc/%.html) $(AVPROGS-yes:%=doc/%-all.html) $(COMP
               doc/mailing-list-faq.html                                 \
               doc/nut.html                                              \
               doc/platform.html                                         \
-              $(SRC_PATH)/doc/bootstrap.min.css                         \
-              $(SRC_PATH)/doc/style.min.css                             \
-              $(SRC_PATH)/doc/default.css                               \
 
 TXTPAGES    = doc/fate.txt                                              \
 
@@ -105,7 +102,7 @@ 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)
-	$(M)$(SRC_PATH)/doc/doxy-wrapper.sh $$PWD/doc/doxy $(SRC_PATH) doc/Doxyfile $(DOXYGEN) $(DOXY_INPUT);
+	$(M)OUT_DIR=$$PWD/doc/doxy; cd $(SRC_PATH); ./doc/doxy-wrapper.sh $$OUT_DIR $< $(DOXYGEN) $(DOXY_INPUT);
 
 install-doc: install-html install-man
 

doc/bitstream_filters.texi

@@ -81,15 +81,12 @@ Top-left position.
 @end table
 
 @item tick_rate
-Set the tick rate (@emph{time_scale / num_units_in_display_tick}) in
+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
@@ -103,9 +100,7 @@ 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.
+Add extradata to the beginning of the filtered packets.
 
 @table @option
 @item freq
@@ -122,7 +117,7 @@ add extradata to all packets
 @end table
 @end table
 
-If not specified it is assumed @samp{k}.
+If not specified it is assumed @samp{e}.
 
 For example the following @command{ffmpeg} command forces a global
 header (thus disabling individual packet headers) in the H.264 packets
@@ -132,36 +127,6 @@ the header stored in extradata to the key packets:
 ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts
 @end example
 
-@section dv_error_marker
-
-Blocks in DV which are marked as damaged are replaced by blocks of the specified color.
-
-@table @option
-@item color
-The color to replace damaged blocks by
-@item sta
-A 16 bit mask which specifies which of the 16 possible error status values are
-to be replaced by colored blocks. 0xFFFE is the default which replaces all non 0
-error status values.
-@table @samp
-@item ok
-No error, no concealment
-@item err
-Error, No concealment
-@item res
-Reserved
-@item notok
-Error or concealment
-@item notres
-Not reserved
-@item Aa, Ba, Ca, Ab, Bb, Cb, A, B, C, a, b, erri, erru
-The specific error status code
-@end table
-see page 44-46 or section 5.5 of
-@url{http://web.archive.org/web/20060927044735/http://www.smpte.org/smpte_store/standards/pdf/s314m.pdf}
-
-@end table
-
 @section eac3_core
 
 Extract the core from a E-AC-3 stream, dropping extra channels.
@@ -247,20 +212,12 @@ Modify metadata embedded in an H.264 stream.
 Insert or remove AUD NAL units in all access units of the stream.
 
 @table @samp
-@item pass
 @item insert
 @item remove
 @end table
 
-Default is pass.
-
 @item sample_aspect_ratio
 Set the sample aspect ratio of the stream in the VUI parameters.
-See H.264 table E-1.
-
-@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
@@ -278,7 +235,7 @@ 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 (time_scale / num_units_in_tick) in the VUI
+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).
@@ -287,11 +244,6 @@ 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 zero_new_constraint_set_flags
-Zero constraint_set4_flag and constraint_set5_flag in the SPS. These
-bits were reserved in a previous version of the H.264 spec, and thus
-some hardware decoders require these to be zero. The result of zeroing
-this is still a valid bitstream.
 
 @item crop_left
 @item crop_right
@@ -315,37 +267,6 @@ insert the string ``hello'' associated with the given UUID.
 @item delete_filler
 Deletes both filler NAL units and filler SEI messages.
 
-@item display_orientation
-Insert, extract or remove Display orientation SEI messages.
-See H.264 section D.1.27 and D.2.27 for syntax and semantics.
-
-@table @samp
-@item pass
-@item insert
-@item remove
-@item extract
-@end table
-
-Default is pass.
-
-Insert mode works in conjunction with @code{rotate} and @code{flip} options.
-Any pre-existing Display orientation messages will be removed in insert or remove mode.
-Extract mode attaches the display matrix to the packet as side data.
-
-@item rotate
-Set rotation in display orientation SEI (anticlockwise angle in degrees).
-Range is -360 to +360. Default is NaN.
-
-@item flip
-Set flip in display orientation SEI.
-
-@table @samp
-@item horizontal
-@item vertical
-@end table
-
-Default is unset.
-
 @item level
 Set the level in the SPS.  Refer to H.264 section A.3 and tables A-1
 to A-5.
@@ -417,8 +338,8 @@ 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 (time_scale /
-num_units_in_tick). Combined with @option{num_ticks_poc_diff_one}, this can
+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.
 
@@ -437,15 +358,6 @@ 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
@@ -599,185 +511,25 @@ container. Can be used for fuzzing or testing error resilience/concealment.
 Parameters:
 @table @option
 @item amount
-Accepts an expression whose evaluation per-packet determines how often bytes in that
-packet will be modified. A value below 0 will result in a variable frequency.
-Default is 0 which results in no modification. However, if neither amount nor drop is specified,
-amount will be set to @var{-1}. See below for accepted variables.
-@item drop
-Accepts an expression evaluated per-packet whose value determines whether that packet is dropped.
-Evaluation to a positive value results in the packet being dropped. Evaluation to a negative
-value results in a variable chance of it being dropped, roughly inverse in proportion to the magnitude
-of the value. Default is 0 which results in no drops. See below for accepted variables.
+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
-Accepts a non-negative integer, which assigns a variable chance of it being dropped, roughly inverse
-in proportion to the value. Default is 0 which results in no drops. This option is kept for backwards
-compatibility and is equivalent to setting drop to a negative value with the same magnitude
-i.e. @code{dropamount=4} is the same as @code{drop=-4}. Ignored if drop is also specified.
+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
 
-Both @code{amount} and @code{drop} accept expressions containing the following variables:
-
-@table @samp
-@item n
-The index of the packet, starting from zero.
-@item tb
-The timebase for packet timestamps.
-@item pts
-Packet presentation timestamp.
-@item dts
-Packet decoding timestamp.
-@item nopts
-Constant representing AV_NOPTS_VALUE.
-@item startpts
-First non-AV_NOPTS_VALUE PTS seen in the stream.
-@item startdts
-First non-AV_NOPTS_VALUE DTS seen in the stream.
-@item duration
-@itemx d
-Packet duration, in timebase units.
-@item pos
-Packet position in input; may be -1 when unknown or not set.
-@item size
-Packet size, in bytes.
-@item key
-Whether packet is marked as a keyframe.
-@item state
-A pseudo random integer, primarily derived from the content of packet payload.
-@end table
-
-@subsection Examples
-Apply modification to every byte but don't drop any packets.
+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
-
-Drop every video packet not marked as a keyframe after timestamp 30s but do not
-modify any of the remaining packets.
-@example
-ffmpeg -i INPUT -c copy -bsf:v noise=drop='gt(t\,30)*not(key)' output.mkv
-@end example
-
-Drop one second of audio every 10 seconds and add some random noise to the rest.
-@example
-ffmpeg -i INPUT -c copy -bsf:a noise=amount=-1:drop='between(mod(t\,10)\,9\,10)' output.mkv
+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 pgs_frame_merge
-
-Merge a sequence of PGS Subtitle segments ending with an "end of display set"
-segment into a single packet.
-
-This is required by some containers that support PGS subtitles
-(muxer @code{matroska}).
-
-@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.
@@ -800,91 +552,6 @@ 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.
-@item duration
-Set expression for duration.
-@item time_base
-Set output time base.
-@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 DURATION
-The duration 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_INDURATION
-The previous input duration.
-
-@item PREV_OUTDTS
-The previous output DTS.
-
-@item PREV_OUTPTS
-The previous output PTS.
-
-@item PREV_OUTDURATION
-The previous output duration.
-
-@item NEXT_DTS
-The next input DTS.
-
-@item NEXT_PTS
-The next input PTS.
-
-@item NEXT_DURATION
-The next input duration.
-
-@item TB
-The timebase of stream packet belongs.
-
-@item TB_OUT
-The output timebase.
-
-@item SR
-The sample rate of stream packet belongs.
-
-@item NOPTS
-The AV_NOPTS_VALUE constant.
-@end table
-
 @anchor{text2movsub}
 @section text2movsub
 
@@ -899,12 +566,7 @@ 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.
+Supports H.264, H.265, MPEG-2 and VP9.
 
 @section vp9_metadata
 
@@ -912,9 +574,7 @@ 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.
+Set the color space value in the frame header.
 @table @samp
 @item unknown
 @item bt601
@@ -926,8 +586,8 @@ incompatible with profiles 0 and 2.
 @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.
+Set the color range value in the frame header.  Note that this cannot
+be set in RGB streams.
 @table @samp
 @item tv
 @item pc

doc/build_system.txt

@@ -36,11 +36,11 @@ install
 examples
     Build all examples located in doc/examples.
 
-checkheaders
-    Check headers dependencies.
+libavformat/output-example
+    Build the libavformat basic example.
 
-alltools
-    Build all tools in tools directory.
+libswscale/swscale-test
+    Build the swscale self-test (useful also as an example).
 
 config
     Reconfigure the project with the current configuration.
@@ -48,8 +48,6 @@ config
 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>

doc/codecs.texi

@@ -50,13 +50,11 @@ Use internal 2pass ratecontrol in first pass mode.
 Use internal 2pass ratecontrol in second pass mode.
 @item gray
 Only decode/encode grayscale.
+@item emu_edge
+Do not draw edges.
 @item psnr
 Set error[?] variables during encoding.
 @item truncated
-Input bitstream might be randomly truncated.
-@item drop_changed
-Don't output frames whose parameters differ from first decoded frame in stream.
-Error AVERROR_INPUT_CHANGED is returned when a frame is dropped.
 
 @item ildct
 Use interlaced DCT.
@@ -70,14 +68,50 @@ This ensures that file and data checksums are reproducible and match between
 platforms. Its primary use is for regression testing.
 @item aic
 Apply H263 advanced intra coding / mpeg4 ac prediction.
+@item cbp
+Deprecated, use mpegvideo private options instead.
+@item qprd
+Deprecated, use mpegvideo private options instead.
 @item ilme
 Apply interlaced motion estimation.
 @item cgop
 Use closed gop.
-@item output_corrupt
-Output even potentially corrupted frames.
 @end table
 
+@item me_method @var{integer} (@emph{encoding,video})
+Set motion estimation method.
+
+Possible values:
+@table @samp
+@item zero
+zero motion estimation (fastest)
+@item full
+full motion estimation (slowest)
+@item epzs
+EPZS motion estimation (default)
+@item esa
+esa motion estimation (alias for full)
+@item tesa
+tesa motion estimation
+@item dia
+dia motion estimation (alias for epzs)
+@item log
+log motion estimation
+@item phods
+phods motion estimation
+@item x1
+X1 motion estimation
+@item hex
+hex motion estimation
+@item umh
+umh motion estimation
+@item iter
+iter motion estimation
+@end table
+
+@item extradata_size @var{integer}
+Set extradata size.
+
 @item time_base @var{rational number}
 Set codec time base.
 
@@ -144,6 +178,24 @@ Default value is 0.
 @item b_qfactor @var{float} (@emph{encoding,video})
 Set qp factor between P and B frames.
 
+@item rc_strategy @var{integer} (@emph{encoding,video})
+Set ratecontrol method.
+
+@item b_strategy @var{integer} (@emph{encoding,video})
+Set strategy to choose between I/P/B-frames.
+
+@item ps @var{integer} (@emph{encoding,video})
+Set RTP payload size in bytes.
+
+@item mv_bits @var{integer}
+@item header_bits @var{integer}
+@item i_tex_bits @var{integer}
+@item p_tex_bits @var{integer}
+@item i_count @var{integer}
+@item p_count @var{integer}
+@item skip_count @var{integer}
+@item misc_bits @var{integer}
+@item frame_bits @var{integer}
 @item codec_tag @var{integer}
 @item bug @var{flags} (@emph{decoding,video})
 Workaround not auto detected encoder bugs.
@@ -152,6 +204,8 @@ Possible values:
 @table @samp
 @item autodetect
 
+@item old_msmpeg4
+some old lavc generated msmpeg4v3 files (no autodetection)
 @item xvid_ilace
 Xvid interlacing bug (autodetected if fourcc==XVIX)
 @item ump4
@@ -160,6 +214,8 @@ Xvid interlacing bug (autodetected if fourcc==XVIX)
 padding bug (autodetected)
 @item amv
 
+@item ac_vlc
+illegal vlc bug (autodetected per fourcc)
 @item qpel_chroma
 
 @item std_qpel
@@ -180,6 +236,14 @@ Workaround various bugs in microsoft broken decoders.
 trancated frames
 @end table
 
+@item lelim @var{integer} (@emph{encoding,video})
+Set single coefficient elimination threshold for luminance (negative
+values also consider DC coefficient).
+
+@item celim @var{integer} (@emph{encoding,video})
+Set single coefficient elimination threshold for chrominance (negative
+values also consider dc coefficient)
+
 @item strict @var{integer} (@emph{decoding/encoding,audio,video})
 Specify how strictly to follow the standards.
 
@@ -233,8 +297,29 @@ consider things that a sane encoder should not do as an error
 
 @item block_align @var{integer}
 
+@item mpeg_quant @var{integer} (@emph{encoding,video})
+Use MPEG quantizers instead of H.263.
+
+@item qsquish @var{float} (@emph{encoding,video})
+How to keep quantizer between qmin and qmax (0 = clip, 1 = use
+differentiable function).
+
+@item rc_qmod_amp @var{float} (@emph{encoding,video})
+Set experimental quantizer modulation.
+
+@item rc_qmod_freq @var{integer} (@emph{encoding,video})
+Set experimental quantizer modulation.
+
 @item rc_override_count @var{integer}
 
+@item rc_eq @var{string} (@emph{encoding,video})
+Set rate control equation. When computing the expression, besides the
+standard functions defined in the section 'Expression Evaluation', the
+following functions are available: bits2qp(bits), qp2bits(qp). Also
+the following constants are available: iTex pTex tex mv fCode iCount
+mcVar var isI isP isB avgQP qComp avgIITex avgPITex avgPPTex avgBPTex
+avgTex.
+
 @item maxrate @var{integer} (@emph{encoding,audio,video})
 Set max bitrate tolerance (in bits/s). Requires bufsize to be set.
 
@@ -245,12 +330,18 @@ encode. It is of little use elsewise.
 @item bufsize @var{integer} (@emph{encoding,audio,video})
 Set ratecontrol buffer size (in bits).
 
+@item rc_buf_aggressivity @var{float} (@emph{encoding,video})
+Currently useless.
+
 @item i_qfactor @var{float} (@emph{encoding,video})
 Set QP factor between P and I frames.
 
 @item i_qoffset @var{float} (@emph{encoding,video})
 Set QP offset between P and I frames.
 
+@item rc_init_cplx @var{float} (@emph{encoding,video})
+Set initial complexity for 1-pass encoding.
+
 @item dct @var{integer} (@emph{encoding,video})
 Set DCT algorithm.
 
@@ -315,7 +406,11 @@ Automatically pick a IDCT compatible with the simple one
 
 @item simpleneon
 
-@item xvid
+@item simplealpha
+
+@item ipp
+
+@item xvidmmx
 
 @item faani
 floating point AAN IDCT
@@ -338,6 +433,19 @@ favor predicting from the previous frame instead of the current
 
 @item bits_per_coded_sample @var{integer}
 
+@item pred @var{integer} (@emph{encoding,video})
+Set prediction method.
+
+Possible values:
+@table @samp
+@item left
+
+@item plane
+
+@item median
+
+@end table
+
 @item aspect @var{rational number} (@emph{encoding,video})
 Set sample aspect ratio.
 
@@ -532,28 +640,13 @@ noise preserving sum of squared differences
 
 @item dia_size @var{integer} (@emph{encoding,video})
 Set diamond type & size for motion estimation.
-@table @samp
-@item (1024, INT_MAX)
-full motion estimation(slowest)
-@item (768, 1024]
-umh motion estimation
-@item (512, 768]
-hex motion estimation
-@item (256, 512]
-l2s diamond motion estimation
-@item [2,256]
-var diamond motion estimation
-@item (-1,  2)
-small diamond motion estimation
-@item -1
-funny diamond motion estimation
-@item (INT_MIN, -1)
-sab diamond motion estimation
-@end table
 
 @item last_pred @var{integer} (@emph{encoding,video})
 Set amount of motion predictors from the previous frame.
 
+@item preme @var{integer} (@emph{encoding,video})
+Set pre motion estimation.
+
 @item precmp @var{integer} (@emph{encoding,video})
 Set pre motion estimation compare function.
 
@@ -597,11 +690,40 @@ Set diamond type & size for motion estimation pre-pass.
 @item subq @var{integer} (@emph{encoding,video})
 Set sub pel motion estimation quality.
 
+@item dtg_active_format @var{integer}
+
 @item me_range @var{integer} (@emph{encoding,video})
 Set limit motion vectors range (1023 for DivX player).
 
+@item ibias @var{integer} (@emph{encoding,video})
+Set intra quant bias.
+
+@item pbias @var{integer} (@emph{encoding,video})
+Set inter quant bias.
+
+@item color_table_id @var{integer}
+
 @item global_quality @var{integer} (@emph{encoding,audio,video})
 
+@item coder @var{integer} (@emph{encoding,video})
+
+Possible values:
+@table @samp
+@item vlc
+variable length coder / huffman coder
+@item ac
+arithmetic coder
+@item raw
+raw (no encoding)
+@item rle
+run-length coder
+@item deflate
+deflate-based coder
+@end table
+
+@item context @var{integer} (@emph{encoding,video})
+Set context model.
+
 @item slice_flags @var{integer}
 
 @item mbd @var{integer} (@emph{encoding,video})
@@ -617,16 +739,32 @@ use fewest bits
 use best rate distortion
 @end table
 
+@item stream_codec_tag @var{integer}
+
+@item sc_threshold @var{integer} (@emph{encoding,video})
+Set scene change threshold.
+
+@item lmin @var{integer} (@emph{encoding,video})
+Set min lagrange factor (VBR).
+
+@item lmax @var{integer} (@emph{encoding,video})
+Set max lagrange factor (VBR).
+
+@item nr @var{integer} (@emph{encoding,video})
+Set noise reduction.
+
 @item rc_init_occupancy @var{integer} (@emph{encoding,video})
 Set number of bits which should be loaded into the rc buffer before
 decoding starts.
 
-@item flags2 @var{flags} (@emph{decoding/encoding,audio,video,subtitles})
+@item flags2 @var{flags} (@emph{decoding/encoding,audio,video})
 
 Possible values:
 @table @samp
 @item fast
 Allow non spec compliant speedup tricks.
+@item sgop
+Deprecated, use mpegvideo private options instead.
 @item noout
 Skip bitstream encoding.
 @item ignorecrop
@@ -637,32 +775,17 @@ Place global headers at every keyframe instead of in extradata.
 Frame data might be split into multiple chunks.
 @item showall
 Show all frames before the first keyframe.
+@item skiprd
+Deprecated, use mpegvideo private options instead.
 @item export_mvs
 Export motion vectors into frame side-data (see @code{AV_FRAME_DATA_MOTION_VECTORS})
 for codecs that support it. See also @file{doc/examples/export_mvs.c}.
-@item skip_manual
-Do not skip samples and export skip information as frame side data.
-@item ass_ro_flush_noop
-Do not reset ASS ReadOrder field on flush.
 @end table
 
-@item export_side_data @var{flags} (@emph{decoding/encoding,audio,video,subtitles})
+@item error @var{integer} (@emph{encoding,video})
 
-Possible values:
-@table @samp
-@item mvs
-Export motion vectors into frame side-data (see @code{AV_FRAME_DATA_MOTION_VECTORS})
-for codecs that support it. See also @file{doc/examples/export_mvs.c}.
-@item prft
-Export encoder Producer Reference Time into packet side-data (see @code{AV_PKT_DATA_PRFT})
-for codecs that support it.
-@item venc_params
-Export video encoding parameters through frame side data (see @code{AV_FRAME_DATA_VIDEO_ENC_PARAMS})
-for codecs that support it. At present, those are H.264 and VP9.
-@item film_grain
-Export film grain parameters through frame side data (see @code{AV_FRAME_DATA_FILM_GRAIN_PARAMS}).
-Supported at present by AV1 decoders.
-@end table
+@item qns @var{integer} (@emph{encoding,video})
+Deprecated, use mpegvideo private options instead.
 
 @item threads @var{integer} (@emph{decoding/encoding,video})
 Set the number of threads to be used, in case the selected codec
@@ -676,6 +799,12 @@ automatically select the number of threads to set
 
 Default value is @samp{auto}.
 
+@item me_threshold @var{integer} (@emph{encoding,video})
+Set motion estimation threshold.
+
+@item mb_threshold @var{integer} (@emph{encoding,video})
+Set macroblock threshold.
+
 @item dc @var{integer} (@emph{encoding,video})
 Set intra_dc_precision.
 
@@ -690,8 +819,49 @@ Set number of macroblock rows at the bottom which are skipped.
 
 @item profile @var{integer} (@emph{encoding,audio,video})
 
-Set encoder codec profile. Default value is @samp{unknown}. Encoder specific
-profiles are documented in the relevant encoder documentation.
+Possible values:
+@table @samp
+@item unknown
+
+@item aac_main
+
+@item aac_low
+
+@item aac_ssr
+
+@item aac_ltp
+
+@item aac_he
+
+@item aac_he_v2
+
+@item aac_ld
+
+@item aac_eld
+
+@item mpeg2_aac_low
+
+@item mpeg2_aac_he
+
+@item mpeg4_sp
+
+@item mpeg4_core
+
+@item mpeg4_main
+
+@item mpeg4_asp
+
+@item dts
+
+@item dts_es
+
+@item dts_96_24
+
+@item dts_hd_hra
+
+@item dts_hd_ma
+
+@end table
 
 @item level @var{integer} (@emph{encoding,audio,video})
 
@@ -704,12 +874,67 @@ Possible values:
 @item lowres @var{integer} (@emph{decoding,audio,video})
 Decode at 1= 1/2, 2=1/4, 3=1/8 resolutions.
 
+@item skip_threshold @var{integer} (@emph{encoding,video})
+Set frame skip threshold.
+
+@item skip_factor @var{integer} (@emph{encoding,video})
+Set frame skip factor.
+
+@item skip_exp @var{integer} (@emph{encoding,video})
+Set frame skip exponent.
+Negative values behave identical to the corresponding positive ones, except
+that the score is normalized.
+Positive values exist primarily for compatibility reasons and are not so useful.
+
+@item skipcmp @var{integer} (@emph{encoding,video})
+Set frame skip compare function.
+
+Possible values:
+@table @samp
+@item sad
+sum of absolute differences, fast (default)
+@item sse
+sum of squared errors
+@item satd
+sum of absolute Hadamard transformed differences
+@item dct
+sum of absolute DCT transformed differences
+@item psnr
+sum of squared quantization errors (avoid, low quality)
+@item bit
+number of bits needed for the block
+@item rd
+rate distortion optimal, slow
+@item zero
+0
+@item vsad
+sum of absolute vertical differences
+@item vsse
+sum of squared vertical differences
+@item nsse
+noise preserving sum of squared differences
+@item w53
+5/3 wavelet, only used in snow
+@item w97
+9/7 wavelet, only used in snow
+@item dctmax
+
+@item chroma
+
+@end table
+
+@item border_mask @var{float} (@emph{encoding,video})
+Increase the quantizer for macroblocks close to borders.
+
 @item mblmin @var{integer} (@emph{encoding,video})
 Set min macroblock lagrange factor (VBR).
 
 @item mblmax @var{integer} (@emph{encoding,video})
 Set max macroblock lagrange factor (VBR).
 
+@item mepc @var{integer} (@emph{encoding,video})
+Set motion estimation bitrate penalty compensation (1.0 = 256).
+
 @item skip_loop_filter @var{integer} (@emph{decoding,video})
 @item skip_idct        @var{integer} (@emph{decoding,video})
 @item skip_frame       @var{integer} (@emph{decoding,video})
@@ -737,9 +962,6 @@ Discard all bidirectional frames.
 @item nokey
 Discard all frames excepts keyframes.
 
-@item nointra
-Discard all frames except I frames.
-
 @item all
 Discard all frames.
 @end table
@@ -749,17 +971,34 @@ Default value is @samp{default}.
 @item bidir_refine @var{integer} (@emph{encoding,video})
 Refine the two motion vectors used in bidirectional macroblocks.
 
+@item brd_scale @var{integer} (@emph{encoding,video})
+Downscale frames for dynamic B-frame decision.
+
 @item keyint_min @var{integer} (@emph{encoding,video})
 Set minimum interval between IDR-frames.
 
 @item refs @var{integer} (@emph{encoding,video})
 Set reference frames to consider for motion compensation.
 
+@item chromaoffset @var{integer} (@emph{encoding,video})
+Set chroma qp offset from luma.
+
 @item trellis @var{integer} (@emph{encoding,audio,video})
 Set rate-distortion optimal quantization.
 
 @item mv0_threshold @var{integer} (@emph{encoding,video})
+@item b_sensitivity @var{integer} (@emph{encoding,video})
+Adjust sensitivity of b_frame_strategy 1.
+
 @item compression_level @var{integer} (@emph{encoding,audio,video})
+@item min_prediction_order @var{integer} (@emph{encoding,audio})
+@item max_prediction_order @var{integer} (@emph{encoding,audio})
+@item timecode_frame_start @var{integer} (@emph{encoding,video})
+Set GOP timecode frame start number, in non drop frame format.
+
+@item request_channels @var{integer} (@emph{decoding,audio})
+Set desired number of audio channels.
+
 @item bits_per_raw_sample @var{integer}
 @item channel_layout @var{integer} (@emph{decoding/encoding,audio})
 
@@ -873,12 +1112,6 @@ BT.2020 NCL
 BT.2020 CL
 @item smpte2085
 SMPTE 2085
-@item chroma-derived-nc
-Chroma-derived NCL
-@item chroma-derived-c
-Chroma-derived CL
-@item ictcp
-ICtCp
 @end table
 
 @item color_range @var{integer} (@emph{decoding/encoding,video})
@@ -999,7 +1232,7 @@ instead of alpha. Default is 0.
 @item dump_separator @var{string} (@emph{input})
 Separator used to separate the fields printed on the command line about the
 Stream parameters.
-For example, to separate the fields with newlines and indentation:
+For example to separate the fields with newlines and indention:
 @example
 ffprobe -dump_separator "
                           "  -i ~/videos/matrixbench_mpeg2.mpg

doc/decoders.texi

@@ -25,19 +25,6 @@ enabled 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.
@@ -60,45 +47,6 @@ top-field-first is assumed
 
 @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).
-This option is deprecated for libdav1d >= 1.0 and will be removed in the future. Use the
-global option @code{threads} instead.
-
-@item tilethreads
-Set amount of tile threads to use during decoding. The default value is 0 (autodetect).
-This option is deprecated for libdav1d >= 1.0 and will be removed in the future. Use the
-global option @code{threads} instead.
-
-@item filmgrain
-Apply film grain to the decoded video if present in the bitstream. Defaults to the
-internal default of the library.
-This option is deprecated and will be removed in the future. See the global option
-@code{export_side_data} to export Film Grain parameters instead of applying it.
-
-@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.
@@ -107,84 +55,6 @@ 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
-
-@section QSV Decoders
-
-The family of Intel QuickSync Video decoders (VC1, MPEG-2, H.264, HEVC,
-JPEG/MJPEG, VP8, VP9, AV1).
-
-@subsection Common Options
-
-The following options are supported by all qsv decoders.
-
-@table @option
-
-@item @var{async_depth}
-Internal parallelization depth, the higher the value the higher the latency.
-
-@item @var{gpu_copy}
-A GPU-accelerated copy between video and system memory
-@table @samp
-@item default
-@item on
-@item off
-@end table
-
-@end table
-
-@subsection HEVC Options
-Extra options for hevc_qsv.
-
-@table @option
-
-@item @var{load_plugin}
-A user plugin to load in an internal session
-@table @samp
-@item none
-@item hevc_sw
-@item hevc_hw
-@end table
-
-@item @var{load_plugins}
-A :-separate list of hexadecimal plugin UIDs to load in an internal session
-
-@end table
-
-@section v210
-
-Uncompressed 4:2:2 10-bit decoder.
-
-@subsection Options
-
-@table @option
-
-@item custom_stride
-Set the line size of the v210 data in bytes. The default value is 0
-(autodetect). You can use the special -1 value for a strideless v210 as seen in
-BOXX files.
-
-@end table
-
-@c man end VIDEO DECODERS
-
 @chapter Audio Decoders
 @c man begin AUDIO DECODERS
 
@@ -204,7 +74,7 @@ the undocumented RealAudio 3 (a.k.a. dnet).
 
 @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.
+from the AC-3 stream. This factor is applied exponentially.
 There are 3 notable scale factor ranges:
 @table @option
 @item drc_scale == 0
@@ -324,31 +194,6 @@ without this library.
 @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
@@ -356,8 +201,6 @@ Enabled by default.
 @table @option
 @item compute_clut
 @table @option
-@item -2
-Compute clut once if no matching CLUT is in the stream.
 @item -1
 Compute clut if no matching CLUT is in the stream.
 @item 0
@@ -386,7 +229,7 @@ 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,
+numbers (without 0x prefix) separated by comas, for example @code{0d00ee,
 ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1,
 7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b}.
 
@@ -415,11 +258,6 @@ 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

doc/demuxers.texi

@@ -25,12 +25,16 @@ Audible Format 2, 3, and 4 demuxer.
 
 This demuxer is used to demux Audible Format 2, 3, and 4 (.aa) files.
 
-@section aac
+@section applehttp
 
-Raw Audio Data Transport Stream AAC demuxer.
+Apple HTTP Live Streaming demuxer.
 
-This demuxer is used to demux an ADTS input containing a single AAC stream
-alongwith any ID3v1/2 or APE tags in it.
+This demuxer presents all AVStreams from all variant streams.
+The id field is set to the bitrate variant index number. By setting
+the discard flags on AVStreams (by pressing 'a' or 'v' in ffplay),
+the caller can decide which variant streams to actually receive.
+The total bitrate of the variant that the stream belongs to is
+available in a metadata key named "variant_bitrate".
 
 @section apng
 
@@ -44,15 +48,12 @@ between the last fcTL and IEND chunks.
 
 @table @option
 @item -ignore_loop @var{bool}
-Ignore the loop variable in the file if set. Default is enabled.
-
+Ignore the loop variable in the file if set.
 @item -max_fps @var{int}
-Maximum framerate in frames per second. Default of 0 imposes no limit.
-
+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). Default is 15.
-
+(0 meaning as fast as possible).
 @end table
 
 @section asf
@@ -103,7 +104,8 @@ 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.
+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
@@ -157,16 +159,6 @@ directive) will be reduced based on their specified Out point.
 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.
-This directive is deprecated, use @code{file_packet_meta} instead.
-
-@item @code{file_packet_meta @var{key} @var{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{option @var{key} @var{value}}
-Option to access, open and probe the file.
-Can be present multiple times.
 
 @item @code{stream}
 Introduce a stream in the virtual file.
@@ -184,20 +176,6 @@ subfiles will be used.
 This is especially useful for MPEG-PS (VOB) files, where the order of the
 streams is not reliable.
 
-@item @code{stream_meta @var{key} @var{value}}
-Metadata for the stream.
-Can be present multiple times.
-
-@item @code{stream_codec @var{value}}
-Codec for the stream.
-
-@item @code{stream_extradata @var{hex_string}}
-Extradata for the string, encoded in hexadecimal.
-
-@item @code{chapter @var{id} @var{start} @var{end}}
-Add a chapter. @var{id} is an unique identifier, possibly small and
-consecutive.
-
 @end table
 
 @subsection Options
@@ -207,8 +185,7 @@ This demuxer accepts the following option:
 @table @option
 
 @item safe
-If set to 1, reject unsafe file paths and directives.
-A file path is considered safe if it
+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
@@ -218,6 +195,9 @@ 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.
@@ -274,29 +254,11 @@ 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.
 
-@subsection Options
-
-This demuxer accepts the following option:
-
-@table @option
-
-@item cenc_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
-
-@section imf
-
-Interoperable Master Format demuxer.
-
-This demuxer presents audio and video streams found in an IMF Composition.
-
-@section flv, live_flv, kux
+@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.
-KUX is a flv variant used on the Youku platform.
 
 @example
 ffmpeg -f flv -i myfile.flv ...
@@ -358,24 +320,12 @@ infinitely.
 
 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 prefer_x_start
-prefer to use #EXT-X-START if it's in playlist instead of live_start_index.
-
 @item allowed_extensions
 ',' separated list of file extensions that hls is allowed to access.
 
@@ -383,10 +333,6 @@ prefer to use #EXT-X-START if it's in playlist instead of live_start_index.
 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.
@@ -394,13 +340,6 @@ 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.
-
-@item seg_format_options
-Set options for the demuxer of media segments using a list of key=value pairs separated by @code{:}.
 @end table
 
 @section image2
@@ -511,17 +450,6 @@ 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
@@ -553,84 +481,14 @@ ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv
 
 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.
+See @url{http://code.google.com/p/game-music-emu/} for more information.
 
-It accepts the following options:
+Some files have multiple tracks. The demuxer will pick the first track by
+default. The @option{track_index} option can be used to select a different
+track. Track indexes start at 0. The demuxer exports the number of tracks as
+@var{tracks} meta data entry.
 
-@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
+For very large files, the @option{max_size} option may have to be adjusted.
 
 @section libopenmpt
 
@@ -659,13 +517,9 @@ 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
+@section mov/mp4/3gp/QuickTime
 
-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
+QuickTime / MP4 demuxer.
 
 This demuxer accepts the following options:
 @table @option
@@ -676,88 +530,10 @@ 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.
+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 use_tfdt
-For fragmented input, set fragment's starting timestamp to @code{baseMediaDecodeTime} from the @code{tfdt} box.
-Default is enabled, which will prefer to use the @code{tfdt} box to set DTS. Disable to use the @code{earliest_presentation_time} from the @code{sidx} box.
-In either case, the timestamp from the @code{mfra} box will be used if it's available and @code{use_mfra_for} is
-set to pts or dts.
-
-@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).
-
-@item max_stts_delta
-Very high sample deltas written in a trak's stts box may occasionally be intended but usually they are written in
-error or used to store a negative value for dts correction when treated as signed 32-bit integers. This option lets
-the user set an upper limit, beyond which the delta is clamped to 1. Values greater than the limit if negative when
-cast to int32 are used to adjust onward dts.
-
-Unit is the track time scale. Range is 0 to UINT_MAX. Default is @code{UINT_MAX - 48000*10} which allows upto
-a 10 second dts correction for 48 kHz audio streams while accommodating 99.9% of @code{uint32} range.
-@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.
@@ -789,10 +565,6 @@ 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.
-
-@item max_packet_size
-Set maximum size, in bytes, of packet emitted by the demuxer. Payloads above this size
-are split across multiple packets. Range is 1 to INT_MAX/2. Default is 204800 bytes.
 @end table
 
 @section mpjpeg
@@ -890,20 +662,4 @@ Example: convert the captions to a format most players understand:
 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

doc/dev_community/community.md

@@ -1,79 +0,0 @@
-# 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.

doc/dev_community/resolution_process.md

@@ -1,91 +0,0 @@
-# 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.
-

doc/developer.texi

@@ -131,9 +131,6 @@ 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
@@ -494,22 +491,6 @@ 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.
 
-@subheading Sending patches from email clients
-Using @code{git send-email} might not be desirable for everyone. The
-following trick allows to send patches via email clients in a safe
-way. It has been tested with Outlook and Thunderbird (with X-Unsent
-extension) and might work with other applications.
-
-Create your patch like this:
-
-@verbatim
-git format-patch -s -o "outputfolder" --add-header "X-Unsent: 1" --suffix .eml --to ffmpeg-devel@ffmpeg.org -1 1a2b3c4d
-@end verbatim
-
-Now you'll just need to open the eml file with the email application
-and execute 'Send'.
-
-@subheading Reviews
 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
@@ -641,7 +622,7 @@ If the patch fixes a bug, did you provide a verbose analysis of the bug?
 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/}.
+URL, you can upload to ftp://upload.ffmpeg.org.
 
 @item
 Did you provide a verbose summary about what the patch does change?

doc/doxy-wrapper.sh

@@ -1,13 +1,10 @@
 #!/bin/sh
 
 OUT_DIR="${1}"
-SRC_DIR="${2}"
-DOXYFILE="${3}"
-DOXYGEN="${4}"
+DOXYFILE="${2}"
+DOXYGEN="${3}"
 
-shift 4
-
-cd ${SRC_DIR}
+shift 3
 
 if [ -e "VERSION" ]; then
     VERSION=`cat "VERSION"`

doc/examples/.gitignore

@@ -1,4 +1,4 @@
-/avio_list_dir
+/avio_dir_cmd
 /avio_reading
 /decode_audio
 /decode_video

doc/examples/Makefile

@@ -1,4 +1,4 @@
-EXAMPLES-$(CONFIG_AVIO_LIST_DIR_EXAMPLE)     += avio_list_dir
+EXAMPLES-$(CONFIG_AVIO_DIR_CMD_EXAMPLE)      += avio_dir_cmd
 EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE)      += avio_reading
 EXAMPLES-$(CONFIG_DECODE_AUDIO_EXAMPLE)      += decode_audio
 EXAMPLES-$(CONFIG_DECODE_VIDEO_EXAMPLE)      += decode_video
@@ -37,7 +37,7 @@ $(EXAMPLES_G): %$(PROGSSUF)_g$(EXESUF): %.o
 examples: $(EXAMPLES)
 
 $(EXAMPLES:%$(PROGSSUF)$(EXESUF)=%.o): | doc/examples
-OUTDIRS += doc/examples
+OBJDIRS += doc/examples
 
 DOXY_INPUT += $(EXAMPLES:%$(PROGSSUF)$(EXESUF)=%.c)
 

doc/examples/Makefile.example

@@ -11,7 +11,7 @@ CFLAGS += -Wall -g
 CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
 LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
 
-EXAMPLES=       avio_list_dir                      \
+EXAMPLES=       avio_dir_cmd                       \
                 avio_reading                       \
                 decode_audio                       \
                 decode_video                       \

doc/examples/avio_dir_cmd.c

@@ -102,15 +102,38 @@ static int list_op(const char *input_dir)
     return ret;
 }
 
+static int del_op(const char *url)
+{
+    int ret = avpriv_io_delete(url);
+    if (ret < 0)
+        av_log(NULL, AV_LOG_ERROR, "Cannot delete '%s': %s.\n", url, av_err2str(ret));
+    return ret;
+}
+
+static int move_op(const char *src, const char *dst)
+{
+    int ret = avpriv_io_move(src, dst);
+    if (ret < 0)
+        av_log(NULL, AV_LOG_ERROR, "Cannot move '%s' into '%s': %s.\n", src, dst, av_err2str(ret));
+    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);
+    fprintf(stderr, "usage: %s OPERATION entry1 [entry2]\n"
+            "API example program to show how to manipulate resources "
+            "accessed through AVIOContext.\n"
+            "OPERATIONS:\n"
+            "list      list content of the directory\n"
+            "move      rename content in directory\n"
+            "del       delete content in directory\n",
+            program_name);
 }
 
 int main(int argc, char *argv[])
 {
+    const char *op = NULL;
     int ret;
 
     av_log_set_level(AV_LOG_DEBUG);
@@ -122,7 +145,32 @@ int main(int argc, char *argv[])
 
     avformat_network_init();
 
-    ret = list_op(argv[1]);
+    op = argv[1];
+    if (strcmp(op, "list") == 0) {
+        if (argc < 3) {
+            av_log(NULL, AV_LOG_INFO, "Missing argument for list operation.\n");
+            ret = AVERROR(EINVAL);
+        } else {
+            ret = list_op(argv[2]);
+        }
+    } else if (strcmp(op, "del") == 0) {
+        if (argc < 3) {
+            av_log(NULL, AV_LOG_INFO, "Missing argument for del operation.\n");
+            ret = AVERROR(EINVAL);
+        } else {
+            ret = del_op(argv[2]);
+        }
+    } else if (strcmp(op, "move") == 0) {
+        if (argc < 4) {
+            av_log(NULL, AV_LOG_INFO, "Missing argument for move operation.\n");
+            ret = AVERROR(EINVAL);
+        } else {
+            ret = move_op(argv[2], argv[3]);
+        }
+    } else {
+        av_log(NULL, AV_LOG_INFO, "Invalid operation %s\n", op);
+        ret = AVERROR(EINVAL);
+    }
 
     avformat_network_deinit();
 

doc/examples/avio_reading.c

@@ -117,12 +117,11 @@ int main(int argc, char *argv[])
 
 end:
     avformat_close_input(&fmt_ctx);
-
     /* note: the internal buffer could have changed, and be != avio_ctx_buffer */
-    if (avio_ctx)
+    if (avio_ctx) {
         av_freep(&avio_ctx->buffer);
-    avio_context_free(&avio_ctx);
-
+        av_freep(&avio_ctx);
+    }
     av_file_unmap(buffer, buffer_size);
 
     if (ret < 0) {

doc/examples/decode_audio.c

@@ -39,35 +39,6 @@
 #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)
 {
@@ -97,7 +68,7 @@ static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame,
             exit(1);
         }
         for (i = 0; i < frame->nb_samples; i++)
-            for (ch = 0; ch < dec_ctx->ch_layout.nb_channels; ch++)
+            for (ch = 0; ch < dec_ctx->channels; ch++)
                 fwrite(frame->data[ch] + data_size*i, 1, data_size, outfile);
     }
 }
@@ -115,9 +86,6 @@ int main(int argc, char **argv)
     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]);
@@ -204,26 +172,6 @@ int main(int argc, char **argv)
     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->ch_layout.nb_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);
 

doc/examples/decode_video.c

@@ -41,7 +41,7 @@ static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
     FILE *f;
     int i;
 
-    f = fopen(filename,"wb");
+    f = fopen(filename,"w");
     fprintf(f, "P5\n%d %d\n%d\n", xsize, ysize, 255);
     for (i = 0; i < ysize; i++)
         fwrite(buf + i * wrap, 1, xsize, f);
@@ -92,12 +92,10 @@ int main(int argc, char **argv)
     uint8_t *data;
     size_t   data_size;
     int ret;
-    int eof;
     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]);
+        fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
         exit(0);
     }
     filename    = argv[1];
@@ -151,16 +149,15 @@ int main(int argc, char **argv)
         exit(1);
     }
 
-    do {
+    while (!feof(f)) {
         /* read raw data from the input file */
         data_size = fread(inbuf, 1, INBUF_SIZE, f);
-        if (ferror(f))
+        if (!data_size)
             break;
-        eof = !data_size;
 
         /* use the parser to split the data into frames */
         data = inbuf;
-        while (data_size > 0 || eof) {
+        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) {
@@ -172,10 +169,8 @@ int main(int argc, char **argv)
 
             if (pkt->size)
                 decode(c, frame, pkt, outfilename);
-            else if (eof)
-                break;
         }
-    } while (!eof);
+    }
 
     /* flush the decoder */
     decode(c, frame, NULL, outfilename);

doc/examples/demuxing_decoding.c

@@ -32,7 +32,6 @@
 #include <libavutil/imgutils.h>
 #include <libavutil/samplefmt.h>
 #include <libavutil/timestamp.h>
-#include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
 
 static AVFormatContext *fmt_ctx = NULL;
@@ -52,97 +51,99 @@ static int video_dst_bufsize;
 
 static int video_stream_idx = -1, audio_stream_idx = -1;
 static AVFrame *frame = NULL;
-static AVPacket *pkt = NULL;
+static AVPacket pkt;
 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;
-    }
+/* Enable or disable frame reference counting. You are not supposed to support
+ * both paths in your application but pick the one most appropriate to your
+ * needs. Look for the use of refcount in this example to see what are the
+ * differences of API usage between them. */
+static int refcount = 0;
 
-    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)
+static int decode_packet(int *got_frame, int cached)
 {
     int ret = 0;
+    int decoded = pkt.size;
 
-    // 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;
-    }
+    *got_frame = 0;
 
-    // get all the available frames from the decoder
-    while (ret >= 0) {
-        ret = avcodec_receive_frame(dec, frame);
+    if (pkt.stream_index == video_stream_idx) {
+        /* decode video frame */
+        ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
         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));
+            fprintf(stderr, "Error decoding video frame (%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);
+        if (*got_frame) {
 
-        av_frame_unref(frame);
-        if (ret < 0)
+            if (frame->width != width || frame->height != height ||
+                frame->format != pix_fmt) {
+                /* To handle this change, one could call av_image_alloc again and
+                 * decode the following frames into another rawvideo file. */
+                fprintf(stderr, "Error: Width, height and pixel format have to be "
+                        "constant in a rawvideo file, but the width, height or "
+                        "pixel format of the input video changed:\n"
+                        "old: width = %d, height = %d, format = %s\n"
+                        "new: width = %d, height = %d, format = %s\n",
+                        width, height, av_get_pix_fmt_name(pix_fmt),
+                        frame->width, frame->height,
+                        av_get_pix_fmt_name(frame->format));
+                return -1;
+            }
+
+            printf("video_frame%s n:%d coded_n:%d\n",
+                   cached ? "(cached)" : "",
+                   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);
+        }
+    } else if (pkt.stream_index == audio_stream_idx) {
+        /* decode audio frame */
+        ret = avcodec_decode_audio4(audio_dec_ctx, frame, got_frame, &pkt);
+        if (ret < 0) {
+            fprintf(stderr, "Error decoding audio frame (%s)\n", av_err2str(ret));
             return ret;
+        }
+        /* Some audio decoders decode only part of the packet, and have to be
+         * called again with the remainder of the packet data.
+         * Sample: fate-suite/lossless-audio/luckynight-partial.shn
+         * Also, some decoders might over-read the packet. */
+        decoded = FFMIN(ret, pkt.size);
+
+        if (*got_frame) {
+            size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format);
+            printf("audio_frame%s n:%d nb_samples:%d pts:%s\n",
+                   cached ? "(cached)" : "",
+                   audio_frame_count++, frame->nb_samples,
+                   av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));
+
+            /* Write the raw audio data samples of the first plane. This works
+             * fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However,
+             * most audio decoders output planar audio, which uses a separate
+             * plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P).
+             * In other words, this code will write only the first audio channel
+             * in these cases.
+             * You should use libswresample or libavfilter to convert the frame
+             * to packed data. */
+            fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file);
+        }
     }
 
-    return 0;
+    /* If we use frame reference counting, we own the data and need
+     * to de-reference it when we don't use it anymore */
+    if (*got_frame && refcount)
+        av_frame_unref(frame);
+
+    return decoded;
 }
 
 static int open_codec_context(int *stream_idx,
@@ -150,7 +151,8 @@ static int open_codec_context(int *stream_idx,
 {
     int ret, stream_index;
     AVStream *st;
-    const AVCodec *dec = NULL;
+    AVCodec *dec = NULL;
+    AVDictionary *opts = NULL;
 
     ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
     if (ret < 0) {
@@ -184,8 +186,9 @@ static int open_codec_context(int *stream_idx,
             return ret;
         }
 
-        /* Init the decoders */
-        if ((ret = avcodec_open2(*dec_ctx, dec, NULL)) < 0) {
+        /* Init the decoders, with or without reference counting */
+        av_dict_set(&opts, "refcounted_frames", refcount ? "1" : "0", 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;
@@ -227,17 +230,24 @@ static int get_format_from_sample_fmt(const char **fmt,
 
 int main (int argc, char **argv)
 {
-    int ret = 0;
+    int ret = 0, got_frame;
 
-    if (argc != 4) {
-        fprintf(stderr, "usage: %s  input_file video_output_file audio_output_file\n"
+    if (argc != 4 && argc != 5) {
+        fprintf(stderr, "usage: %s [-refcount] 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]);
+                "audio frames to a rawaudio file named audio_output_file.\n\n"
+                "If the -refcount option is specified, the program use the\n"
+                "reference counting frame system which allows keeping a copy of\n"
+                "the data for longer than one decode call.\n"
+                "\n", argv[0]);
         exit(1);
     }
+    if (argc == 5 && !strcmp(argv[1], "-refcount")) {
+        refcount = 1;
+        argv++;
+    }
     src_filename = argv[1];
     video_dst_filename = argv[2];
     audio_dst_filename = argv[3];
@@ -303,12 +313,10 @@ int main (int argc, char **argv)
         goto end;
     }
 
-    pkt = av_packet_alloc();
-    if (!pkt) {
-        fprintf(stderr, "Could not allocate packet\n");
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
+    /* initialize packet, set data to NULL, let the demuxer fill it */
+    av_init_packet(&pkt);
+    pkt.data = NULL;
+    pkt.size = 0;
 
     if (video_stream)
         printf("Demuxing video from file '%s' into '%s'\n", src_filename, video_dst_filename);
@@ -316,23 +324,24 @@ int main (int argc, char **argv)
         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;
+    while (av_read_frame(fmt_ctx, &pkt) >= 0) {
+        AVPacket orig_pkt = pkt;
+        do {
+            ret = decode_packet(&got_frame, 0);
+            if (ret < 0)
+                break;
+            pkt.data += ret;
+            pkt.size -= ret;
+        } while (pkt.size > 0);
+        av_packet_unref(&orig_pkt);
     }
 
-    /* flush the decoders */
-    if (video_dec_ctx)
-        decode_packet(video_dec_ctx, NULL);
-    if (audio_dec_ctx)
-        decode_packet(audio_dec_ctx, NULL);
+    /* flush cached frames */
+    pkt.data = NULL;
+    pkt.size = 0;
+    do {
+        decode_packet(&got_frame, 1);
+    } while (got_frame);
 
     printf("Demuxing succeeded.\n");
 
@@ -345,7 +354,7 @@ int main (int argc, char **argv)
 
     if (audio_stream) {
         enum AVSampleFormat sfmt = audio_dec_ctx->sample_fmt;
-        int n_channels = audio_dec_ctx->ch_layout.nb_channels;
+        int n_channels = audio_dec_ctx->channels;
         const char *fmt;
 
         if (av_sample_fmt_is_planar(sfmt)) {
@@ -374,7 +383,6 @@ end:
         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]);
 

doc/examples/encode_audio.c

@@ -70,25 +70,26 @@ static int select_sample_rate(const AVCodec *codec)
 }
 
 /* select layout with the highest channel count */
-static int select_channel_layout(const AVCodec *codec, AVChannelLayout *dst)
+static int select_channel_layout(const AVCodec *codec)
 {
-    const AVChannelLayout *p, *best_ch_layout;
+    const uint64_t *p;
+    uint64_t best_ch_layout = 0;
     int best_nb_channels   = 0;
 
-    if (!codec->ch_layouts)
-        return av_channel_layout_copy(dst, &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO);
+    if (!codec->channel_layouts)
+        return AV_CH_LAYOUT_STEREO;
 
-    p = codec->ch_layouts;
-    while (p->nb_channels) {
-        int nb_channels = p->nb_channels;
+    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_ch_layout    = *p;
             best_nb_channels = nb_channels;
         }
         p++;
     }
-    return av_channel_layout_copy(dst, best_ch_layout);
+    return best_ch_layout;
 }
 
 static void encode(AVCodecContext *ctx, AVFrame *frame, AVPacket *pkt,
@@ -163,9 +164,8 @@ int main(int argc, char **argv)
 
     /* select other audio parameters supported by the encoder */
     c->sample_rate    = select_sample_rate(codec);
-    ret = select_channel_layout(codec, &c->ch_layout);
-    if (ret < 0)
-        exit(1);
+    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) {
@@ -195,9 +195,7 @@ int main(int argc, char **argv)
 
     frame->nb_samples     = c->frame_size;
     frame->format         = c->sample_fmt;
-    ret = av_channel_layout_copy(&frame->ch_layout, &c->ch_layout);
-    if (ret < 0)
-        exit(1);
+    frame->channel_layout = c->channel_layout;
 
     /* allocate the data buffers */
     ret = av_frame_get_buffer(frame, 0);
@@ -220,7 +218,7 @@ int main(int argc, char **argv)
         for (j = 0; j < c->frame_size; j++) {
             samples[2*j] = (int)(sin(t) * 10000);
 
-            for (k = 1; k < c->ch_layout.nb_channels; k++)
+            for (k = 1; k < c->channels; k++)
                 samples[2*j + k] = samples[2*j];
             t += tincr;
         }

doc/examples/encode_video.c

@@ -145,7 +145,7 @@ int main(int argc, char **argv)
     frame->width  = c->width;
     frame->height = c->height;
 
-    ret = av_frame_get_buffer(frame, 0);
+    ret = av_frame_get_buffer(frame, 32);
     if (ret < 0) {
         fprintf(stderr, "Could not allocate the video frame data\n");
         exit(1);
@@ -155,25 +155,12 @@ int main(int argc, char **argv)
     for (i = 0; i < 25; i++) {
         fflush(stdout);
 
-        /* Make sure the frame data is writable.
-           On the first round, the frame is fresh from av_frame_get_buffer()
-           and therefore we know it is writable.
-           But on the next rounds, encode() will have called
-           avcodec_send_frame(), and the codec may have kept a reference to
-           the frame in its internal structures, that makes the frame
-           unwritable.
-           av_frame_make_writable() checks that and allocates a new buffer
-           for the frame only if necessary.
-         */
+        /* make sure the frame data is writable */
         ret = av_frame_make_writable(frame);
         if (ret < 0)
             exit(1);
 
-        /* Prepare a dummy image.
-           In real code, this is where you would have your own logic for
-           filling the frame. FFmpeg does not care what you put in the
-           frame.
-         */
+        /* prepare a dummy image */
         /* Y */
         for (y = 0; y < c->height; y++) {
             for (x = 0; x < c->width; x++) {
@@ -198,14 +185,8 @@ int main(int argc, char **argv)
     /* flush the encoder */
     encode(c, NULL, pkt, f);
 
-    /* Add sequence end code to have a real MPEG file.
-       It makes only sense because this tiny examples writes packets
-       directly. This is called "elementary stream" and only works for some
-       codecs. To create a valid file, you usually need to write packets
-       into a proper file format or protocol; see muxing.c.
-     */
-    if (codec->id == AV_CODEC_ID_MPEG1VIDEO || codec->id == AV_CODEC_ID_MPEG2VIDEO)
-        fwrite(endcode, 1, sizeof(endcode), f);
+    /* add sequence end code to have a real MPEG file */
+    fwrite(endcode, 1, sizeof(endcode), f);
     fclose(f);
 
     avcodec_free_context(&c);

doc/examples/extract_mvs.c

@@ -22,7 +22,6 @@
  */
 
 #include <libavutil/motion_vector.h>
-#include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
 
 static AVFormatContext *fmt_ctx = NULL;
@@ -79,7 +78,7 @@ static int open_codec_context(AVFormatContext *fmt_ctx, enum AVMediaType type)
     int ret;
     AVStream *st;
     AVCodecContext *dec_ctx = NULL;
-    const AVCodec *dec = NULL;
+    AVCodec *dec = NULL;
     AVDictionary *opts = NULL;
 
     ret = av_find_best_stream(fmt_ctx, type, -1, -1, &dec, 0);
@@ -105,9 +104,7 @@ static int open_codec_context(AVFormatContext *fmt_ctx, enum AVMediaType type)
 
         /* Init the video decoder */
         av_dict_set(&opts, "flags2", "+export_mvs", 0);
-        ret = avcodec_open2(dec_ctx, dec, &opts);
-        av_dict_free(&opts);
-        if (ret < 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;
@@ -124,7 +121,7 @@ static int open_codec_context(AVFormatContext *fmt_ctx, enum AVMediaType type)
 int main(int argc, char **argv)
 {
     int ret = 0;
-    AVPacket *pkt = NULL;
+    AVPacket pkt = { 0 };
 
     if (argc != 2) {
         fprintf(stderr, "Usage: %s <video>\n", argv[0]);
@@ -159,20 +156,13 @@ int main(int argc, char **argv)
         goto end;
     }
 
-    pkt = av_packet_alloc();
-    if (!pkt) {
-        fprintf(stderr, "Could not allocate AVPacket\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);
+    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;
     }
@@ -184,6 +174,5 @@ end:
     avcodec_free_context(&video_dec_ctx);
     avformat_close_input(&fmt_ctx);
     av_frame_free(&frame);
-    av_packet_free(&pkt);
     return ret < 0;
 }

doc/examples/filter_audio.c

@@ -55,7 +55,7 @@
 
 #define INPUT_SAMPLERATE     48000
 #define INPUT_FORMAT         AV_SAMPLE_FMT_FLTP
-#define INPUT_CHANNEL_LAYOUT (AVChannelLayout)AV_CHANNEL_LAYOUT_5POINT0
+#define INPUT_CHANNEL_LAYOUT AV_CH_LAYOUT_5POINT0
 
 #define VOLUME_VAL 0.90
 
@@ -100,7 +100,7 @@ static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
     }
 
     /* Set the filter options through the AVOptions API. */
-    av_channel_layout_describe(&INPUT_CHANNEL_LAYOUT, ch_layout, sizeof(ch_layout));
+    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);
@@ -154,8 +154,9 @@ static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
     /* 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=stereo",
-             av_get_sample_fmt_name(AV_SAMPLE_FMT_S16), 44100);
+             "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");
@@ -214,7 +215,7 @@ static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
 static int process_output(struct AVMD5 *md5, AVFrame *frame)
 {
     int planar     = av_sample_fmt_is_planar(frame->format);
-    int channels   = frame->ch_layout.nb_channels;
+    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);
@@ -247,7 +248,7 @@ static int get_input(AVFrame *frame, int frame_num)
     /* Set up the frame properties and allocate the buffer for the data. */
     frame->sample_rate    = INPUT_SAMPLERATE;
     frame->format         = INPUT_FORMAT;
-    av_channel_layout_copy(&frame->ch_layout, &INPUT_CHANNEL_LAYOUT);
+    frame->channel_layout = INPUT_CHANNEL_LAYOUT;
     frame->nb_samples     = FRAME_SIZE;
     frame->pts            = frame_num * FRAME_SIZE;
 

doc/examples/filtering_audio.c

@@ -34,7 +34,6 @@
 #include <libavformat/avformat.h>
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
-#include <libavutil/channel_layout.h>
 #include <libavutil/opt.h>
 
 static const char *filter_descr = "aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono";
@@ -49,8 +48,8 @@ static int audio_stream_index = -1;
 
 static int open_input_file(const char *filename)
 {
-    const AVCodec *dec;
     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");
@@ -94,6 +93,7 @@ static int init_filters(const char *filters_descr)
     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;
@@ -105,13 +105,12 @@ static int init_filters(const char *filters_descr)
     }
 
     /* buffer audio source: the decoded frames from the decoder will be inserted here. */
-    if (dec_ctx->ch_layout.order == AV_CHANNEL_ORDER_UNSPEC)
-        av_channel_layout_default(&dec_ctx->ch_layout, dec_ctx->ch_layout.nb_channels);
-    ret = snprintf(args, sizeof(args),
-            "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=",
+    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));
-    av_channel_layout_describe(&dec_ctx->ch_layout, args + ret, sizeof(args) - ret);
+             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) {
@@ -134,7 +133,7 @@ static int init_filters(const char *filters_descr)
         goto end;
     }
 
-    ret = av_opt_set(buffersink_ctx, "ch_layouts", "mono",
+    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");
@@ -185,7 +184,7 @@ static int init_filters(const char *filters_descr)
     /* Print summary of the sink buffer
      * Note: args buffer is reused to store channel layout string */
     outlink = buffersink_ctx->inputs[0];
-    av_channel_layout_describe(&outlink->ch_layout, args, sizeof(args));
+    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), "?"),
@@ -200,7 +199,7 @@ end:
 
 static void print_frame(const AVFrame *frame)
 {
-    const int n = frame->nb_samples * frame->ch_layout.nb_channels;
+    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;
 
@@ -215,12 +214,12 @@ static void print_frame(const AVFrame *frame)
 int main(int argc, char **argv)
 {
     int ret;
-    AVPacket *packet = av_packet_alloc();
+    AVPacket packet;
     AVFrame *frame = av_frame_alloc();
     AVFrame *filt_frame = av_frame_alloc();
 
-    if (!packet || !frame || !filt_frame) {
-        fprintf(stderr, "Could not allocate frame or packet\n");
+    if (!frame || !filt_frame) {
+        perror("Could not allocate frame");
         exit(1);
     }
     if (argc != 2) {
@@ -235,11 +234,11 @@ int main(int argc, char **argv)
 
     /* read all packets */
     while (1) {
-        if ((ret = av_read_frame(fmt_ctx, packet)) < 0)
+        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 (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;
@@ -275,13 +274,12 @@ int main(int argc, char **argv)
                 }
             }
         }
-        av_packet_unref(packet);
+        av_packet_unref(&packet);
     }
 end:
     avfilter_graph_free(&filter_graph);
     avcodec_free_context(&dec_ctx);
     avformat_close_input(&fmt_ctx);
-    av_packet_free(&packet);
     av_frame_free(&frame);
     av_frame_free(&filt_frame);
 

doc/examples/filtering_video.c

@@ -53,8 +53,8 @@ static int64_t last_pts = AV_NOPTS_VALUE;
 
 static int open_input_file(const char *filename)
 {
-    const AVCodec *dec;
     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");
@@ -210,7 +210,7 @@ static void display_frame(const AVFrame *frame, AVRational time_base)
 int main(int argc, char **argv)
 {
     int ret;
-    AVPacket *packet;
+    AVPacket packet;
     AVFrame *frame;
     AVFrame *filt_frame;
 
@@ -221,9 +221,8 @@ int main(int argc, char **argv)
 
     frame = av_frame_alloc();
     filt_frame = av_frame_alloc();
-    packet = av_packet_alloc();
-    if (!frame || !filt_frame || !packet) {
-        fprintf(stderr, "Could not allocate frame or packet\n");
+    if (!frame || !filt_frame) {
+        perror("Could not allocate frame");
         exit(1);
     }
 
@@ -234,11 +233,11 @@ int main(int argc, char **argv)
 
     /* read all packets */
     while (1) {
-        if ((ret = av_read_frame(fmt_ctx, packet)) < 0)
+        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 (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;
@@ -274,7 +273,7 @@ int main(int argc, char **argv)
                 av_frame_unref(frame);
             }
         }
-        av_packet_unref(packet);
+        av_packet_unref(&packet);
     }
 end:
     avfilter_graph_free(&filter_graph);
@@ -282,7 +281,6 @@ end:
     avformat_close_input(&fmt_ctx);
     av_frame_free(&frame);
     av_frame_free(&filt_frame);
-    av_packet_free(&packet);
 
     if (ret < 0 && ret != AVERROR_EOF) {
         fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));

doc/examples/hw_decode.c

@@ -152,8 +152,8 @@ int main(int argc, char *argv[])
     int video_stream, ret;
     AVStream *video = NULL;
     AVCodecContext *decoder_ctx = NULL;
-    const AVCodec *decoder = NULL;
-    AVPacket *packet = NULL;
+    AVCodec *decoder = NULL;
+    AVPacket packet;
     enum AVHWDeviceType type;
     int i;
 
@@ -172,12 +172,6 @@ int main(int argc, char *argv[])
         return -1;
     }
 
-    packet = av_packet_alloc();
-    if (!packet) {
-        fprintf(stderr, "Failed to allocate AVPacket\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]);
@@ -229,25 +223,27 @@ int main(int argc, char *argv[])
     }
 
     /* open the file to dump raw data */
-    output_file = fopen(argv[3], "w+b");
+    output_file = fopen(argv[3], "w+");
 
     /* actual decoding and dump the raw data */
     while (ret >= 0) {
-        if ((ret = av_read_frame(input_ctx, packet)) < 0)
+        if ((ret = av_read_frame(input_ctx, &packet)) < 0)
             break;
 
-        if (video_stream == packet->stream_index)
-            ret = decode_write(decoder_ctx, packet);
+        if (video_stream == packet.stream_index)
+            ret = decode_write(decoder_ctx, &packet);
 
-        av_packet_unref(packet);
+        av_packet_unref(&packet);
     }
 
     /* flush the decoder */
-    ret = decode_write(decoder_ctx, NULL);
+    packet.data = NULL;
+    packet.size = 0;
+    ret = decode_write(decoder_ctx, &packet);
+    av_packet_unref(&packet);
 
     if (output_file)
         fclose(output_file);
-    av_packet_free(&packet);
     avcodec_free_context(&decoder_ctx);
     avformat_close_input(&input_ctx);
     av_buffer_unref(&hw_device_ctx);

doc/examples/metadata.c

@@ -34,7 +34,7 @@
 int main (int argc, char **argv)
 {
     AVFormatContext *fmt_ctx = NULL;
-    const AVDictionaryEntry *tag = NULL;
+    AVDictionaryEntry *tag = NULL;
     int ret;
 
     if (argc != 2) {
@@ -47,11 +47,6 @@ int main (int argc, char **argv)
     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);
 

doc/examples/muxing.c

@@ -39,7 +39,6 @@
 #include <libavutil/opt.h>
 #include <libavutil/mathematics.h>
 #include <libavutil/timestamp.h>
-#include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
 #include <libswscale/swscale.h>
 #include <libswresample/swresample.h>
@@ -62,8 +61,6 @@ typedef struct OutputStream {
     AVFrame *frame;
     AVFrame *tmp_frame;
 
-    AVPacket *tmp_pkt;
-
     float t, tincr, tincr2;
 
     struct SwsContext *sws_ctx;
@@ -81,50 +78,20 @@ static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt)
            pkt->stream_index);
 }
 
-static int write_frame(AVFormatContext *fmt_ctx, AVCodecContext *c,
-                       AVStream *st, AVFrame *frame, AVPacket *pkt)
+static int write_frame(AVFormatContext *fmt_ctx, const AVRational *time_base, AVStream *st, AVPacket *pkt)
 {
-    int ret;
+    /* rescale output packet timestamp values from codec to stream timebase */
+    av_packet_rescale_ts(pkt, *time_base, st->time_base);
+    pkt->stream_index = st->index;
 
-    // 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) {
-        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);
-        /* pkt is now blank (av_interleaved_write_frame() takes ownership of
-         * its contents and resets pkt), so that no unreferencing is necessary.
-         * This would be different if one used av_write_frame(). */
-        if (ret < 0) {
-            fprintf(stderr, "Error while writing output packet: %s\n", av_err2str(ret));
-            exit(1);
-        }
-    }
-
-    return ret == AVERROR_EOF ? 1 : 0;
+    /* Write the compressed frame to the media file. */
+    log_packet(fmt_ctx, pkt);
+    return av_interleaved_write_frame(fmt_ctx, pkt);
 }
 
 /* Add an output stream. */
 static void add_stream(OutputStream *ost, AVFormatContext *oc,
-                       const AVCodec **codec,
+                       AVCodec **codec,
                        enum AVCodecID codec_id)
 {
     AVCodecContext *c;
@@ -138,12 +105,6 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
         exit(1);
     }
 
-    ost->tmp_pkt = av_packet_alloc();
-    if (!ost->tmp_pkt) {
-        fprintf(stderr, "Could not allocate AVPacket\n");
-        exit(1);
-    }
-
     ost->st = avformat_new_stream(oc, NULL);
     if (!ost->st) {
         fprintf(stderr, "Could not allocate stream\n");
@@ -170,7 +131,16 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
                     c->sample_rate = 44100;
             }
         }
-        av_channel_layout_copy(&c->ch_layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO);
+        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;
 
@@ -200,7 +170,7 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
              * the motion of the chroma plane does not match the luma plane. */
             c->mb_decision = 2;
         }
-        break;
+    break;
 
     default:
         break;
@@ -215,7 +185,7 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
 /* audio output */
 
 static AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt,
-                                  const AVChannelLayout *channel_layout,
+                                  uint64_t channel_layout,
                                   int sample_rate, int nb_samples)
 {
     AVFrame *frame = av_frame_alloc();
@@ -227,7 +197,7 @@ static AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt,
     }
 
     frame->format = sample_fmt;
-    av_channel_layout_copy(&frame->ch_layout, channel_layout);
+    frame->channel_layout = channel_layout;
     frame->sample_rate = sample_rate;
     frame->nb_samples = nb_samples;
 
@@ -242,8 +212,7 @@ static AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt,
     return frame;
 }
 
-static void open_audio(AVFormatContext *oc, const AVCodec *codec,
-                       OutputStream *ost, AVDictionary *opt_arg)
+static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
 {
     AVCodecContext *c;
     int nb_samples;
@@ -272,9 +241,9 @@ static void open_audio(AVFormatContext *oc, const AVCodec *codec,
     else
         nb_samples = c->frame_size;
 
-    ost->frame     = alloc_audio_frame(c->sample_fmt, &c->ch_layout,
+    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->ch_layout,
+    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 */
@@ -285,25 +254,25 @@ static void open_audio(AVFormatContext *oc, const AVCodec *codec,
     }
 
     /* create resampler context */
-    ost->swr_ctx = swr_alloc();
-    if (!ost->swr_ctx) {
-        fprintf(stderr, "Could not allocate resampler context\n");
-        exit(1);
-    }
+        ost->swr_ctx = swr_alloc();
+        if (!ost->swr_ctx) {
+            fprintf(stderr, "Could not allocate resampler context\n");
+            exit(1);
+        }
 
-    /* set options */
-    av_opt_set_chlayout  (ost->swr_ctx, "in_chlayout",       &c->ch_layout,      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_chlayout  (ost->swr_ctx, "out_chlayout",      &c->ch_layout,      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);
+        /* 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);
-    }
+        /* 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
@@ -316,12 +285,12 @@ static AVFrame *get_audio_frame(OutputStream *ost)
 
     /* 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)
+                      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->ch_layout.nb_channels; i++)
+        for (i = 0; i < ost->enc->channels; i++)
             *q++ = v;
         ost->t     += ost->tincr;
         ost->tincr += ost->tincr2;
@@ -340,20 +309,23 @@ static AVFrame *get_audio_frame(OutputStream *ost)
 static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
 {
     AVCodecContext *c;
+    AVPacket pkt = { 0 }; // data and size must be 0;
     AVFrame *frame;
     int ret;
+    int got_packet;
     int dst_nb_samples;
 
+    av_init_packet(&pkt);
     c = ost->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);
+            /* 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;
@@ -377,7 +349,22 @@ static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
         ost->samples_count += dst_nb_samples;
     }
 
-    return write_frame(oc, c, ost->st, frame, ost->tmp_pkt);
+    ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
+    if (ret < 0) {
+        fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
+        exit(1);
+    }
+
+    if (got_packet) {
+        ret = write_frame(oc, &c->time_base, ost->st, &pkt);
+        if (ret < 0) {
+            fprintf(stderr, "Error while writing audio frame: %s\n",
+                    av_err2str(ret));
+            exit(1);
+        }
+    }
+
+    return (frame || got_packet) ? 0 : 1;
 }
 
 /**************************************************************/
@@ -397,7 +384,7 @@ static AVFrame *alloc_picture(enum AVPixelFormat pix_fmt, int width, int height)
     picture->height = height;
 
     /* allocate the buffers for the frame data */
-    ret = av_frame_get_buffer(picture, 0);
+    ret = av_frame_get_buffer(picture, 32);
     if (ret < 0) {
         fprintf(stderr, "Could not allocate frame data.\n");
         exit(1);
@@ -406,8 +393,7 @@ static AVFrame *alloc_picture(enum AVPixelFormat pix_fmt, int width, int height)
     return picture;
 }
 
-static void open_video(AVFormatContext *oc, const AVCodec *codec,
-                       OutputStream *ost, AVDictionary *opt_arg)
+static void open_video(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
 {
     int ret;
     AVCodecContext *c = ost->enc;
@@ -478,7 +464,7 @@ static AVFrame *get_video_frame(OutputStream *ost)
 
     /* check if we want to generate more frames */
     if (av_compare_ts(ost->next_pts, c->time_base,
-                      STREAM_DURATION, (AVRational){ 1, 1 }) > 0)
+                      STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
         return NULL;
 
     /* when we pass a frame to the encoder, it may keep a reference to it
@@ -520,7 +506,37 @@ static AVFrame *get_video_frame(OutputStream *ost)
  */
 static int write_video_frame(AVFormatContext *oc, OutputStream *ost)
 {
-    return write_frame(oc, ost->enc, ost->st, get_video_frame(ost), ost->tmp_pkt);
+    int ret;
+    AVCodecContext *c;
+    AVFrame *frame;
+    int got_packet = 0;
+    AVPacket pkt = { 0 };
+
+    c = ost->enc;
+
+    frame = get_video_frame(ost);
+
+    av_init_packet(&pkt);
+
+    /* encode the image */
+    ret = avcodec_encode_video2(c, &pkt, frame, &got_packet);
+    if (ret < 0) {
+        fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret));
+        exit(1);
+    }
+
+    if (got_packet) {
+        ret = write_frame(oc, &c->time_base, ost->st, &pkt);
+    } else {
+        ret = 0;
+    }
+
+    if (ret < 0) {
+        fprintf(stderr, "Error while writing video frame: %s\n", av_err2str(ret));
+        exit(1);
+    }
+
+    return (frame || got_packet) ? 0 : 1;
 }
 
 static void close_stream(AVFormatContext *oc, OutputStream *ost)
@@ -528,7 +544,6 @@ static void close_stream(AVFormatContext *oc, OutputStream *ost)
     avcodec_free_context(&ost->enc);
     av_frame_free(&ost->frame);
     av_frame_free(&ost->tmp_frame);
-    av_packet_free(&ost->tmp_pkt);
     sws_freeContext(ost->sws_ctx);
     swr_free(&ost->swr_ctx);
 }
@@ -539,10 +554,10 @@ static void close_stream(AVFormatContext *oc, OutputStream *ost)
 int main(int argc, char **argv)
 {
     OutputStream video_st = { 0 }, audio_st = { 0 };
-    const AVOutputFormat *fmt;
     const char *filename;
+    AVOutputFormat *fmt;
     AVFormatContext *oc;
-    const AVCodec *audio_codec, *video_codec;
+    AVCodec *audio_codec, *video_codec;
     int ret;
     int have_video = 0, have_audio = 0;
     int encode_video = 0, encode_audio = 0;
@@ -629,6 +644,10 @@ int main(int argc, char **argv)
         }
     }
 
+    /* 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. */

doc/examples/qsvdec.c

@@ -44,10 +44,38 @@
 #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;
         }
 
@@ -59,7 +87,7 @@ static int get_format(AVCodecContext *avctx, const enum AVPixelFormat *pix_fmts)
     return AV_PIX_FMT_NONE;
 }
 
-static int decode_packet(AVCodecContext *decoder_ctx,
+static int decode_packet(DecodeContext *decode, AVCodecContext *decoder_ctx,
                          AVFrame *frame, AVFrame *sw_frame,
                          AVPacket *pkt, AVIOContext *output_ctx)
 {
@@ -113,15 +141,15 @@ int main(int argc, char **argv)
     AVCodecContext *decoder_ctx = NULL;
     const AVCodec *decoder;
 
-    AVPacket *pkt = NULL;
+    AVPacket pkt = { 0 };
     AVFrame *frame = NULL, *sw_frame = NULL;
 
+    DecodeContext decode = { NULL };
+
     AVIOContext *output_ctx = NULL;
 
     int ret, i;
 
-    AVBufferRef *device_ref = NULL;
-
     if (argc < 3) {
         fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
         return 1;
@@ -149,7 +177,7 @@ int main(int argc, char **argv)
     }
 
     /* open the hardware device */
-    ret = av_hwdevice_ctx_create(&device_ref, AV_HWDEVICE_TYPE_QSV,
+    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");
@@ -181,8 +209,7 @@ int main(int argc, char **argv)
         decoder_ctx->extradata_size = video_st->codecpar->extradata_size;
     }
 
-
-    decoder_ctx->hw_device_ctx = av_buffer_ref(device_ref);
+    decoder_ctx->opaque      = &decode;
     decoder_ctx->get_format  = get_format;
 
     ret = avcodec_open2(decoder_ctx, NULL, NULL);
@@ -200,26 +227,27 @@ int main(int argc, char **argv)
 
     frame    = av_frame_alloc();
     sw_frame = av_frame_alloc();
-    pkt      = av_packet_alloc();
-    if (!frame || !sw_frame || !pkt) {
+    if (!frame || !sw_frame) {
         ret = AVERROR(ENOMEM);
         goto finish;
     }
 
     /* actual decoding */
     while (ret >= 0) {
-        ret = av_read_frame(input_ctx, pkt);
+        ret = av_read_frame(input_ctx, &pkt);
         if (ret < 0)
             break;
 
-        if (pkt->stream_index == video_st->index)
-            ret = decode_packet(decoder_ctx, frame, sw_frame, pkt, output_ctx);
+        if (pkt.stream_index == video_st->index)
+            ret = decode_packet(&decode, decoder_ctx, frame, sw_frame, &pkt, output_ctx);
 
-        av_packet_unref(pkt);
+        av_packet_unref(&pkt);
     }
 
     /* flush the decoder */
-    ret = decode_packet(decoder_ctx, frame, sw_frame, NULL, output_ctx);
+    pkt.data = NULL;
+    pkt.size = 0;
+    ret = decode_packet(&decode, decoder_ctx, frame, sw_frame, &pkt, output_ctx);
 
 finish:
     if (ret < 0) {
@@ -232,11 +260,10 @@ finish:
 
     av_frame_free(&frame);
     av_frame_free(&sw_frame);
-    av_packet_free(&pkt);
 
     avcodec_free_context(&decoder_ctx);
 
-    av_buffer_unref(&device_ref);
+    av_buffer_unref(&decode.hw_device_ref);
 
     avio_close(output_ctx);
 

doc/examples/remuxing.c

@@ -45,9 +45,9 @@ static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt, cons
 
 int main(int argc, char **argv)
 {
-    const AVOutputFormat *ofmt = NULL;
+    AVOutputFormat *ofmt = NULL;
     AVFormatContext *ifmt_ctx = NULL, *ofmt_ctx = NULL;
-    AVPacket *pkt = NULL;
+    AVPacket pkt;
     const char *in_filename, *out_filename;
     int ret, i;
     int stream_index = 0;
@@ -65,12 +65,6 @@ int main(int argc, char **argv)
     in_filename  = argv[1];
     out_filename = argv[2];
 
-    pkt = av_packet_alloc();
-    if (!pkt) {
-        fprintf(stderr, "Could not allocate AVPacket\n");
-        return 1;
-    }
-
     if ((ret = avformat_open_input(&ifmt_ctx, in_filename, 0, 0)) < 0) {
         fprintf(stderr, "Could not open input file '%s'", in_filename);
         goto end;
@@ -91,7 +85,7 @@ int main(int argc, char **argv)
     }
 
     stream_mapping_size = ifmt_ctx->nb_streams;
-    stream_mapping = av_calloc(stream_mapping_size, sizeof(*stream_mapping));
+    stream_mapping = av_mallocz_array(stream_mapping_size, sizeof(*stream_mapping));
     if (!stream_mapping) {
         ret = AVERROR(ENOMEM);
         goto end;
@@ -146,39 +140,38 @@ int main(int argc, char **argv)
     while (1) {
         AVStream *in_stream, *out_stream;
 
-        ret = av_read_frame(ifmt_ctx, pkt);
+        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);
+        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");
+        pkt.stream_index = stream_mapping[pkt.stream_index];
+        out_stream = ofmt_ctx->streams[pkt.stream_index];
+        log_packet(ifmt_ctx, &pkt, "in");
 
         /* copy packet */
-        av_packet_rescale_ts(pkt, in_stream->time_base, out_stream->time_base);
-        pkt->pos = -1;
-        log_packet(ofmt_ctx, pkt, "out");
+        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);
-        /* pkt is now blank (av_interleaved_write_frame() takes ownership of
-         * its contents and resets pkt), so that no unreferencing is necessary.
-         * This would be different if one used av_write_frame(). */
+        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:
-    av_packet_free(&pkt);
 
     avformat_close_input(&ifmt_ctx);
 

doc/examples/resampling_audio.c

@@ -80,7 +80,7 @@ static void fill_samples(double *dst, int nb_samples, int nb_channels, int sampl
 
 int main(int argc, char **argv)
 {
-    AVChannelLayout src_ch_layout = AV_CHANNEL_LAYOUT_STEREO, dst_ch_layout = AV_CHANNEL_LAYOUT_SURROUND;
+    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;
@@ -92,7 +92,6 @@ int main(int argc, char **argv)
     int dst_bufsize;
     const char *fmt;
     struct SwrContext *swr_ctx;
-    char buf[64];
     double t;
     int ret;
 
@@ -121,11 +120,11 @@ int main(int argc, char **argv)
     }
 
     /* set options */
-    av_opt_set_chlayout(swr_ctx, "in_chlayout",    &src_ch_layout, 0);
+    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_chlayout(swr_ctx, "out_chlayout",    &dst_ch_layout, 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);
 
@@ -137,7 +136,7 @@ int main(int argc, char **argv)
 
     /* allocate source and destination samples buffers */
 
-    src_nb_channels = src_ch_layout.nb_channels;
+    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) {
@@ -152,7 +151,7 @@ int main(int argc, char **argv)
         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 = dst_ch_layout.nb_channels;
+    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) {
@@ -195,10 +194,9 @@ int main(int argc, char **argv)
 
     if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0)
         goto end;
-    av_channel_layout_describe(&dst_ch_layout, buf, sizeof(buf));
     fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"
-            "ffplay -f %s -channel_layout %s -channels %d -ar %d %s\n",
-            fmt, buf, dst_nb_channels, dst_rate, dst_filename);
+            "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);

doc/examples/transcode_aac.c

@@ -1,5 +1,5 @@
 /*
- * Copyright (c) 2013-2022 Andreas Unterweger
+ * Copyright (c) 2013-2018 Andreas Unterweger
  *
  * This file is part of FFmpeg.
  *
@@ -38,7 +38,6 @@
 #include "libavutil/audio_fifo.h"
 #include "libavutil/avassert.h"
 #include "libavutil/avstring.h"
-#include "libavutil/channel_layout.h"
 #include "libavutil/frame.h"
 #include "libavutil/opt.h"
 
@@ -61,8 +60,7 @@ static int open_input_file(const char *filename,
                            AVCodecContext **input_codec_context)
 {
     AVCodecContext *avctx;
-    const AVCodec *input_codec;
-    const AVStream *stream;
+    AVCodec *input_codec;
     int error;
 
     /* Open the input file to read from it. */
@@ -90,10 +88,8 @@ static int open_input_file(const char *filename,
         return AVERROR_EXIT;
     }
 
-    stream = (*input_format_context)->streams[0];
-
     /* Find a decoder for the audio stream. */
-    if (!(input_codec = avcodec_find_decoder(stream->codecpar->codec_id))) {
+    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;
@@ -108,7 +104,7 @@ static int open_input_file(const char *filename,
     }
 
     /* Initialize the stream parameters with demuxer information. */
-    error = avcodec_parameters_to_context(avctx, stream->codecpar);
+    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);
@@ -124,9 +120,6 @@ static int open_input_file(const char *filename,
         return error;
     }
 
-    /* Set the packet timebase for the decoder. */
-    avctx->pkt_timebase = stream->time_base;
-
     /* Save the decoder context for easier access later. */
     *input_codec_context = avctx;
 
@@ -151,7 +144,7 @@ static int open_output_file(const char *filename,
     AVCodecContext *avctx          = NULL;
     AVIOContext *output_io_context = NULL;
     AVStream *stream               = NULL;
-    const AVCodec *output_codec    = NULL;
+    AVCodec *output_codec          = NULL;
     int error;
 
     /* Open the output file to write to it. */
@@ -206,11 +199,15 @@ static int open_output_file(const char *filename,
 
     /* Set the basic encoder parameters.
      * The input file's sample rate is used to avoid a sample rate conversion. */
-    av_channel_layout_default(&avctx->ch_layout, OUTPUT_CHANNELS);
+    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;
@@ -248,16 +245,14 @@ cleanup:
 
 /**
  * Initialize one data packet for reading or writing.
- * @param[out] packet Packet to be initialized
- * @return Error code (0 if successful)
+ * @param packet Packet to be initialized
  */
-static int init_packet(AVPacket **packet)
+static void init_packet(AVPacket *packet)
 {
-    if (!(*packet = av_packet_alloc())) {
-        fprintf(stderr, "Could not allocate packet\n");
-        return AVERROR(ENOMEM);
-    }
-    return 0;
+    av_init_packet(packet);
+    /* Set the packet data and size so that it is recognized as being empty. */
+    packet->data = NULL;
+    packet->size = 0;
 }
 
 /**
@@ -292,18 +287,21 @@ static int init_resampler(AVCodecContext *input_codec_context,
         /*
          * 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).
          */
-        error = swr_alloc_set_opts2(resample_context,
-                                             &output_codec_context->ch_layout,
+        *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,
-                                             &input_codec_context->ch_layout,
+                                              av_get_default_channel_layout(input_codec_context->channels),
                                               input_codec_context->sample_fmt,
                                               input_codec_context->sample_rate,
                                               0, NULL);
-        if (error < 0) {
+        if (!*resample_context) {
             fprintf(stderr, "Could not allocate resample context\n");
-            return error;
+            return AVERROR(ENOMEM);
         }
         /*
         * Perform a sanity check so that the number of converted samples is
@@ -331,7 +329,7 @@ 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->ch_layout.nb_channels, 1))) {
+                                      output_codec_context->channels, 1))) {
         fprintf(stderr, "Could not allocate FIFO\n");
         return AVERROR(ENOMEM);
     }
@@ -373,33 +371,28 @@ static int decode_audio_frame(AVFrame *frame,
                               int *data_present, int *finished)
 {
     /* Packet used for temporary storage. */
-    AVPacket *input_packet;
+    AVPacket input_packet;
     int error;
+    init_packet(&input_packet);
 
-    error = init_packet(&input_packet);
-    if (error < 0)
-        return error;
-
-    *data_present = 0;
-    *finished = 0;
     /* Read one audio frame from the input file into a temporary packet. */
-    if ((error = av_read_frame(input_format_context, input_packet)) < 0) {
+    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;
+            return error;
         }
     }
 
     /* 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) {
+    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;
+        return error;
     }
 
     /* Receive one frame from the decoder. */
@@ -425,7 +418,7 @@ static int decode_audio_frame(AVFrame *frame,
     }
 
 cleanup:
-    av_packet_free(&input_packet);
+    av_packet_unref(&input_packet);
     return error;
 }
 
@@ -451,7 +444,7 @@ static int init_converted_samples(uint8_t ***converted_input_samples,
      * 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->ch_layout.nb_channels,
+    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);
@@ -460,7 +453,7 @@ static int init_converted_samples(uint8_t ***converted_input_samples,
     /* 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->ch_layout.nb_channels,
+                                  output_codec_context->channels,
                                   frame_size,
                                   output_codec_context->sample_fmt, 0)) < 0) {
         fprintf(stderr,
@@ -560,7 +553,7 @@ static int read_decode_convert_and_store(AVAudioFifo *fifo,
     AVFrame *input_frame = NULL;
     /* Temporary storage for the converted input samples. */
     uint8_t **converted_input_samples = NULL;
-    int data_present;
+    int data_present = 0;
     int ret = AVERROR_EXIT;
 
     /* Initialize temporary storage for one input frame. */
@@ -634,7 +627,7 @@ static int init_output_frame(AVFrame **frame,
      * Default channel layouts based on the number of channels
      * are assumed for simplicity. */
     (*frame)->nb_samples     = frame_size;
-    av_channel_layout_copy(&(*frame)->ch_layout, &output_codec_context->ch_layout);
+    (*frame)->channel_layout = output_codec_context->channel_layout;
     (*frame)->format         = output_codec_context->sample_fmt;
     (*frame)->sample_rate    = output_codec_context->sample_rate;
 
@@ -668,12 +661,9 @@ static int encode_audio_frame(AVFrame *frame,
                               int *data_present)
 {
     /* Packet used for temporary storage. */
-    AVPacket *output_packet;
+    AVPacket output_packet;
     int error;
-
-    error = init_packet(&output_packet);
-    if (error < 0)
-        return error;
+    init_packet(&output_packet);
 
     /* Set a timestamp based on the sample rate for the container. */
     if (frame) {
@@ -681,20 +671,21 @@ static int encode_audio_frame(AVFrame *frame,
         pts += frame->nb_samples;
     }
 
-    *data_present = 0;
     /* 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);
-    /* Check for errors, but proceed with fetching encoded samples if the
-     *  encoder signals that it has nothing more to encode. */
-    if (error < 0 && error != AVERROR_EOF) {
-      fprintf(stderr, "Could not send packet for encoding (error '%s')\n",
-              av_err2str(error));
-      goto cleanup;
+    /* 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));
+        return error;
     }
 
     /* Receive one encoded frame from the encoder. */
-    error = avcodec_receive_packet(output_codec_context, output_packet);
+    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)) {
@@ -715,14 +706,14 @@ static int encode_audio_frame(AVFrame *frame,
 
     /* 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) {
+        (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);
+    av_packet_unref(&output_packet);
     return error;
 }
 
@@ -861,6 +852,7 @@ int main(int argc, char **argv)
             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;

doc/examples/transcoding.c

@@ -32,7 +32,6 @@
 #include <libavformat/avformat.h>
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
-#include <libavutil/channel_layout.h>
 #include <libavutil/opt.h>
 #include <libavutil/pixdesc.h>
 
@@ -42,17 +41,12 @@ 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;
 
@@ -72,13 +66,13 @@ static int open_input_file(const char *filename)
         return ret;
     }
 
-    stream_ctx = av_calloc(ifmt_ctx->nb_streams, sizeof(*stream_ctx));
+    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];
-        const AVCodec *dec = avcodec_find_decoder(stream->codecpar->codec_id);
+        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);
@@ -108,10 +102,6 @@ static int open_input_file(const char *filename)
             }
         }
         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);
@@ -123,7 +113,7 @@ static int open_output_file(const char *filename)
     AVStream *out_stream;
     AVStream *in_stream;
     AVCodecContext *dec_ctx, *enc_ctx;
-    const AVCodec *encoder;
+    AVCodec *encoder;
     int ret;
     unsigned int i;
 
@@ -175,9 +165,8 @@ static int open_output_file(const char *filename)
                 enc_ctx->time_base = av_inv_q(dec_ctx->framerate);
             } else {
                 enc_ctx->sample_rate = dec_ctx->sample_rate;
-                ret = av_channel_layout_copy(&enc_ctx->ch_layout, &dec_ctx->ch_layout);
-                if (ret < 0)
-                    return ret;
+                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};
@@ -290,7 +279,6 @@ static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
             goto end;
         }
     } else if (dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
-        char buf[64];
         buffersrc = avfilter_get_by_name("abuffer");
         buffersink = avfilter_get_by_name("abuffersink");
         if (!buffersrc || !buffersink) {
@@ -299,14 +287,14 @@ static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
             goto end;
         }
 
-        if (dec_ctx->ch_layout.order == AV_CHANNEL_ORDER_UNSPEC)
-            av_channel_layout_default(&dec_ctx->ch_layout, dec_ctx->ch_layout.nb_channels);
-        av_channel_layout_describe(&dec_ctx->ch_layout, buf, sizeof(buf));
+        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=%s",
+                "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),
-                buf);
+                dec_ctx->channel_layout);
         ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
                 args, NULL, filter_graph);
         if (ret < 0) {
@@ -329,9 +317,9 @@ static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
             goto end;
         }
 
-        av_channel_layout_describe(&enc_ctx->ch_layout, buf, sizeof(buf));
-        ret = av_opt_set(buffersink_ctx, "ch_layouts",
-                         buf, AV_OPT_SEARCH_CHILDREN);
+        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;
@@ -410,63 +398,54 @@ static int init_filters(void)
                 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;
+static int encode_write_frame(AVFrame *filt_frame, unsigned int stream_index, int *got_frame) {
     int ret;
+    int got_frame_local;
+    AVPacket enc_pkt;
+    int (*enc_func)(AVCodecContext *, AVPacket *, const AVFrame *, int *) =
+        (ifmt_ctx->streams[stream_index]->codecpar->codec_type ==
+         AVMEDIA_TYPE_VIDEO) ? avcodec_encode_video2 : avcodec_encode_audio2;
+
+    if (!got_frame)
+        got_frame = &got_frame_local;
 
     av_log(NULL, AV_LOG_INFO, "Encoding frame\n");
     /* encode filtered frame */
-    av_packet_unref(enc_pkt);
-
-    ret = avcodec_send_frame(stream->enc_ctx, filt_frame);
-
+    enc_pkt.data = NULL;
+    enc_pkt.size = 0;
+    av_init_packet(&enc_pkt);
+    ret = enc_func(stream_ctx[stream_index].enc_ctx, &enc_pkt,
+            filt_frame, got_frame);
+    av_frame_free(&filt_frame);
     if (ret < 0)
         return ret;
+    if (!(*got_frame))
+        return 0;
 
-    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);
-    }
+    /* prepare packet for muxing */
+    enc_pkt.stream_index = stream_index;
+    av_packet_rescale_ts(&enc_pkt,
+                         stream_ctx[stream_index].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;
+    AVFrame *filt_frame;
 
     av_log(NULL, AV_LOG_INFO, "Pushing decoded frame to filters\n");
     /* push the decoded frame into the filtergraph */
-    ret = av_buffersrc_add_frame_flags(filter->buffersrc_ctx,
+    ret = av_buffersrc_add_frame_flags(filter_ctx[stream_index].buffersrc_ctx,
             frame, 0);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
@@ -475,9 +454,14 @@ static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index)
 
     /* pull filtered frames from the filtergraph */
     while (1) {
+        filt_frame = av_frame_alloc();
+        if (!filt_frame) {
+            ret = AVERROR(ENOMEM);
+            break;
+        }
         av_log(NULL, AV_LOG_INFO, "Pulling filtered frame from filters\n");
-        ret = av_buffersink_get_frame(filter->buffersink_ctx,
-                                      filter->filtered_frame);
+        ret = av_buffersink_get_frame(filter_ctx[stream_index].buffersink_ctx,
+                filt_frame);
         if (ret < 0) {
             /* if no more frames for output - returns AVERROR(EAGAIN)
              * if flushed and no more frames for output - returns AVERROR_EOF
@@ -485,12 +469,12 @@ static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index)
              */
             if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
                 ret = 0;
+            av_frame_free(&filt_frame);
             break;
         }
 
-        filter->filtered_frame->pict_type = AV_PICTURE_TYPE_NONE;
-        ret = encode_write_frame(stream_index, 0);
-        av_frame_unref(filter->filtered_frame);
+        filt_frame->pict_type = AV_PICTURE_TYPE_NONE;
+        ret = encode_write_frame(filt_frame, stream_index, NULL);
         if (ret < 0)
             break;
     }
@@ -500,20 +484,34 @@ static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index)
 
 static int flush_encoder(unsigned int stream_index)
 {
+    int ret;
+    int got_frame;
+
     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);
+    while (1) {
+        av_log(NULL, AV_LOG_INFO, "Flushing stream #%u encoder\n", stream_index);
+        ret = encode_write_frame(NULL, stream_index, &got_frame);
+        if (ret < 0)
+            break;
+        if (!got_frame)
+            return 0;
+    }
+    return ret;
 }
 
 int main(int argc, char **argv)
 {
     int ret;
-    AVPacket *packet = NULL;
+    AVPacket packet = { .data = NULL, .size = 0 };
+    AVFrame *frame = NULL;
+    enum AVMediaType type;
     unsigned int stream_index;
     unsigned int i;
+    int got_frame;
+    int (*dec_func)(AVCodecContext *, AVFrame *, int *, const AVPacket *);
 
     if (argc != 3) {
         av_log(NULL, AV_LOG_ERROR, "Usage: %s <input file> <output file>\n", argv[0]);
@@ -526,54 +524,56 @@ int main(int argc, char **argv)
         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)
+        if ((ret = av_read_frame(ifmt_ctx, &packet)) < 0)
             break;
-        stream_index = packet->stream_index;
+        stream_index = packet.stream_index;
+        type = ifmt_ctx->streams[packet.stream_index]->codecpar->codec_type;
         av_log(NULL, AV_LOG_DEBUG, "Demuxer gave frame of stream_index %u\n",
                 stream_index);
 
         if (filter_ctx[stream_index].filter_graph) {
-            StreamContext *stream = &stream_ctx[stream_index];
-
             av_log(NULL, AV_LOG_DEBUG, "Going to reencode&filter the frame\n");
-
-            av_packet_rescale_ts(packet,
+            frame = av_frame_alloc();
+            if (!frame) {
+                ret = AVERROR(ENOMEM);
+                break;
+            }
+            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);
+                                 stream_ctx[stream_index].dec_ctx->time_base);
+            dec_func = (type == AVMEDIA_TYPE_VIDEO) ? avcodec_decode_video2 :
+                avcodec_decode_audio4;
+            ret = dec_func(stream_ctx[stream_index].dec_ctx, frame,
+                    &got_frame, &packet);
             if (ret < 0) {
+                av_frame_free(&frame);
                 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 (got_frame) {
+                frame->pts = frame->best_effort_timestamp;
+                ret = filter_encode_write_frame(frame, stream_index);
+                av_frame_free(&frame);
                 if (ret < 0)
                     goto end;
+            } else {
+                av_frame_free(&frame);
             }
         } else {
             /* remux this frame without reencoding */
-            av_packet_rescale_ts(packet,
+            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);
+            ret = av_interleaved_write_frame(ofmt_ctx, &packet);
             if (ret < 0)
                 goto end;
         }
-        av_packet_unref(packet);
+        av_packet_unref(&packet);
     }
 
     /* flush filters and encoders */
@@ -597,18 +597,14 @@ int main(int argc, char **argv)
 
     av_write_trailer(ofmt_ctx);
 end:
-    av_packet_free(&packet);
+    av_packet_unref(&packet);
+    av_frame_free(&frame);
     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) {
+        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);

doc/examples/vaapi_encode.c

@@ -74,27 +74,27 @@ static int set_hwframe_ctx(AVCodecContext *ctx, AVBufferRef *hw_device_ctx)
 static int encode_write(AVCodecContext *avctx, AVFrame *frame, FILE *fout)
 {
     int ret = 0;
-    AVPacket *enc_pkt;
+    AVPacket enc_pkt;
 
-    if (!(enc_pkt = av_packet_alloc()))
-        return AVERROR(ENOMEM);
+    av_init_packet(&enc_pkt);
+    enc_pkt.data = NULL;
+    enc_pkt.size = 0;
 
     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);
+        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);
+        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;
 }
@@ -105,7 +105,7 @@ int main(int argc, char *argv[])
     FILE *fin = NULL, *fout = NULL;
     AVFrame *sw_frame = NULL, *hw_frame = NULL;
     AVCodecContext *avctx = NULL;
-    const AVCodec *codec = NULL;
+    AVCodec *codec = NULL;
     const char *enc_name = "h264_vaapi";
 
     if (argc < 5) {
@@ -172,7 +172,7 @@ int main(int argc, char *argv[])
         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)
+        if ((err = av_frame_get_buffer(sw_frame, 32)) < 0)
             goto close;
         if ((err = fread((uint8_t*)(sw_frame->data[0]), size, 1, fin)) <= 0)
             break;

doc/examples/vaapi_transcode.c

@@ -62,7 +62,7 @@ static enum AVPixelFormat get_vaapi_format(AVCodecContext *ctx,
 static int open_input_file(const char *filename)
 {
     int ret;
-    const AVCodec *decoder = NULL;
+    AVCodec *decoder = NULL;
     AVStream *video = NULL;
 
     if ((ret = avformat_open_input(&ifmt_ctx, filename, NULL, NULL)) < 0) {
@@ -109,25 +109,28 @@ static int open_input_file(const char *filename)
     return ret;
 }
 
-static int encode_write(AVPacket *enc_pkt, AVFrame *frame)
+static int encode_write(AVFrame *frame)
 {
     int ret = 0;
+    AVPacket enc_pkt;
 
-    av_packet_unref(enc_pkt);
+    av_init_packet(&enc_pkt);
+    enc_pkt.data = NULL;
+    enc_pkt.size = 0;
 
     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);
+        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,
+        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);
+        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));
@@ -142,7 +145,7 @@ end:
     return ret;
 }
 
-static int dec_enc(AVPacket *pkt, const AVCodec *enc_codec)
+static int dec_enc(AVPacket *pkt, AVCodec *enc_codec)
 {
     AVFrame *frame;
     int ret = 0;
@@ -213,7 +216,7 @@ static int dec_enc(AVPacket *pkt, const AVCodec *enc_codec)
             initialized = 1;
         }
 
-        if ((ret = encode_write(pkt, frame)) < 0)
+        if ((ret = encode_write(frame)) < 0)
             fprintf(stderr, "Error during encoding and writing.\n");
 
 fail:
@@ -226,9 +229,9 @@ fail:
 
 int main(int argc, char **argv)
 {
-    const AVCodec *enc_codec;
     int ret = 0;
-    AVPacket *dec_pkt;
+    AVPacket dec_pkt;
+    AVCodec *enc_codec;
 
     if (argc != 4) {
         fprintf(stderr, "Usage: %s <input file> <encode codec> <output file>\n"
@@ -243,12 +246,6 @@ int main(int argc, char **argv)
         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;
 
@@ -278,21 +275,23 @@ int main(int argc, char **argv)
 
     /* read all packets and only transcoding video */
     while (ret >= 0) {
-        if ((ret = av_read_frame(ifmt_ctx, dec_pkt)) < 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);
+        if (video_stream == dec_pkt.stream_index)
+            ret = dec_enc(&dec_pkt, enc_codec);
 
-        av_packet_unref(dec_pkt);
+        av_packet_unref(&dec_pkt);
     }
 
     /* flush decoder */
-    av_packet_unref(dec_pkt);
-    ret = dec_enc(dec_pkt, enc_codec);
+    dec_pkt.data = NULL;
+    dec_pkt.size = 0;
+    ret = dec_enc(&dec_pkt, enc_codec);
+    av_packet_unref(&dec_pkt);
 
     /* flush encoder */
-    ret = encode_write(dec_pkt, NULL);
+    ret = encode_write(NULL);
 
     /* write the trailer for output stream */
     av_write_trailer(ofmt_ctx);
@@ -303,6 +302,5 @@ end:
     avcodec_free_context(&decoder_ctx);
     avcodec_free_context(&encoder_ctx);
     av_buffer_unref(&hw_device_ctx);
-    av_packet_free(&dec_pkt);
     return ret;
 }

doc/faq.texi

@@ -76,7 +76,7 @@ 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}.
+@url{http://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...
 
@@ -257,13 +257,13 @@ 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'.
+things to try: '-bf 2', '-flags qprd', '-flags mv0', '-flags skiprd'.
 
 @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'.
+Things to try: '-bf 2', '-flags qprd', '-flags mv0', '-flags skiprd.
 
 @section Interlaced video looks very bad when encoded with ffmpeg, what is wrong?
 
@@ -516,7 +516,7 @@ 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)
+standard input to @code{/dev/null} (on Linux and Mac OS)
 or @code{NUL} (on Windows). You can do this redirect either
 on the ffmpeg invocation, or from a shell script which calls ffmpeg.
 
@@ -526,7 +526,7 @@ For example:
 ffmpeg -nostdin -i INPUT OUTPUT
 @end example
 
-or (on Linux, macOS, and other UNIX-like shells):
+or (on Linux, Mac OS, and other UNIX-like shells):
 
 @example
 ffmpeg -i INPUT OUTPUT </dev/null
@@ -601,7 +601,7 @@ No. These tools are too bloated and they complicate the build.
 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"}.
+read @uref{http://www.tux.org/lkml/#s15, "Programming Religion"}.
 
 @section Why are the ffmpeg programs devoid of debugging symbols?
 

doc/fate.texi

@@ -79,21 +79,6 @@ 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 get the complete list of tests, run the command:
-@example
-make fate-list
-@end example
-
-You can specify a subset of tests to run by specifying the
-corresponding elements from the list with the @code{fate-} prefix,
-e.g. as in:
-@example
-make fate-ffprobe_compact fate-ffprobe_xml
-@end example
-
-This makes it easier to run a few tests in case of failure without
-running the complete test suite.
-
 To use a custom wrapper to run the test, pass @option{--target-exec} to
 @command{configure} or set the @var{TARGET_EXEC} Make variable.
 
@@ -164,8 +149,6 @@ 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.
@@ -174,8 +157,6 @@ 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:

doc/ffmpeg.texi

@@ -449,11 +449,6 @@ output file already exists.
 Set number of times input stream shall be looped. Loop 0 means no loop,
 loop -1 means infinite loop.
 
-@item -recast_media (@emph{global})
-Allow forcing a decoder of a different media type than the one
-detected or designated by the demuxer. Useful for decoding media
-data muxed as data streams.
-
 @item -c[:@var{stream_specifier}] @var{codec} (@emph{input/output,per-stream})
 @itemx -codec[:@var{stream_specifier}] @var{codec} (@emph{input/output,per-stream})
 Select an encoder (when used before an output file) or a decoder (when used
@@ -518,21 +513,6 @@ see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1)
 Like the @code{-ss} option but relative to the "end of file". That is negative
 values are earlier in the file, 0 is at EOF.
 
-@item -isync @var{input_index} (@emph{input})
-Assign an input as a sync source.
-
-This will take the difference between the start times of the target and reference inputs and
-offset the timestamps of the target file by that difference. The source timestamps of the two
-inputs should derive from the same clock source for expected results. If @code{copyts} is set
-then @code{start_at_zero} must also be set. If either of the inputs has no starting timestamp
-then no sync adjustment is made.
-
-Acceptable values are those that refer to a valid ffmpeg input index. If the sync reference is
-the target index itself or @var{-1}, then no adjustment is made to target timestamps. A sync
-reference may not itself be synced to any other input.
-
-Default value is @var{-1}.
-
 @item -itsoffset @var{offset} (@emph{input})
 Set the input time offset.
 
@@ -543,9 +523,6 @@ The offset is added to the timestamps of the input files. Specifying
 a positive offset means that the corresponding streams are delayed by
 the time duration specified in @var{offset}.
 
-@item -itsscale @var{scale} (@emph{input,per-stream})
-Rescale input timestamps. @var{scale} should be a floating point number.
-
 @item -timestamp @var{date} (@emph{output})
 Set the recording timestamp in the container.
 
@@ -575,22 +552,27 @@ ffmpeg -i INPUT -metadata:s:a:0 language=eng OUTPUT
 @item -disposition[:stream_specifier] @var{value} (@emph{output,per-stream})
 Sets the disposition for a stream.
 
-By default, the disposition is copied from the input stream, unless the output
-stream this option applies to is fed by a complex filtergraph - in that case the
-disposition is unset by default.
+This option overrides the disposition copied from the input stream. It is also
+possible to delete the disposition by setting it to 0.
 
-@var{value} is a sequence of items separated by '+' or '-'. The first item may
-also be prefixed with '+' or '-', in which case this option modifies the default
-value. Otherwise (the first item is not prefixed) this options overrides the
-default value. A '+' prefix adds the given disposition, '-' removes it. It is
-also possible to clear the disposition by setting it to 0.
-
-If no @code{-disposition} options were specified for an output file, ffmpeg will
-automatically set the 'default' disposition on the first stream of each type,
-when there are multiple streams of this type in the output file and no stream of
-that type is already marked as default.
-
-The @code{-dispositions} option lists the known dispositions.
+The following dispositions are recognized:
+@table @option
+@item default
+@item dub
+@item original
+@item comment
+@item lyrics
+@item karaoke
+@item forced
+@item hearing_impaired
+@item visual_impaired
+@item clean_effects
+@item attached_pic
+@item captions
+@item descriptions
+@item dependent
+@item metadata
+@end table
 
 For example, to make the second audio stream the default stream:
 @example
@@ -632,109 +614,8 @@ they do not conflict with the standard, as in:
 ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg
 @end example
 
-The parameters set for each target are as follows.
-
-@strong{VCD}
-@example
-@var{pal}:
--f vcd -muxrate 1411200 -muxpreload 0.44 -packetsize 2324
--s 352x288 -r 25
--codec:v mpeg1video -g 15 -b:v 1150k -maxrate:v 1150k -minrate:v 1150k -bufsize:v 327680
--ar 44100 -ac 2
--codec:a mp2 -b:a 224k
-
-@var{ntsc}:
--f vcd -muxrate 1411200 -muxpreload 0.44 -packetsize 2324
--s 352x240 -r 30000/1001
--codec:v mpeg1video -g 18 -b:v 1150k -maxrate:v 1150k -minrate:v 1150k -bufsize:v 327680
--ar 44100 -ac 2
--codec:a mp2 -b:a 224k
-
-@var{film}:
--f vcd -muxrate 1411200 -muxpreload 0.44 -packetsize 2324
--s 352x240 -r 24000/1001
--codec:v mpeg1video -g 18 -b:v 1150k -maxrate:v 1150k -minrate:v 1150k -bufsize:v 327680
--ar 44100 -ac 2
--codec:a mp2 -b:a 224k
-@end example
-
-@strong{SVCD}
-@example
-@var{pal}:
--f svcd -packetsize 2324
--s 480x576 -pix_fmt yuv420p -r 25
--codec:v mpeg2video -g 15 -b:v 2040k -maxrate:v 2516k -minrate:v 0 -bufsize:v 1835008 -scan_offset 1
--ar 44100
--codec:a mp2 -b:a 224k
-
-@var{ntsc}:
--f svcd -packetsize 2324
--s 480x480 -pix_fmt yuv420p -r 30000/1001
--codec:v mpeg2video -g 18 -b:v 2040k -maxrate:v 2516k -minrate:v 0 -bufsize:v 1835008 -scan_offset 1
--ar 44100
--codec:a mp2 -b:a 224k
-
-@var{film}:
--f svcd -packetsize 2324
--s 480x480 -pix_fmt yuv420p -r 24000/1001
--codec:v mpeg2video -g 18 -b:v 2040k -maxrate:v 2516k -minrate:v 0 -bufsize:v 1835008 -scan_offset 1
--ar 44100
--codec:a mp2 -b:a 224k
-@end example
-
-@strong{DVD}
-@example
-@var{pal}:
--f dvd -muxrate 10080k -packetsize 2048
--s 720x576 -pix_fmt yuv420p -r 25
--codec:v mpeg2video -g 15 -b:v 6000k -maxrate:v 9000k -minrate:v 0 -bufsize:v 1835008
--ar 48000
--codec:a ac3 -b:a 448k
-
-@var{ntsc}:
--f dvd -muxrate 10080k -packetsize 2048
--s 720x480 -pix_fmt yuv420p -r 30000/1001
--codec:v mpeg2video -g 18 -b:v 6000k -maxrate:v 9000k -minrate:v 0 -bufsize:v 1835008
--ar 48000
--codec:a ac3 -b:a 448k
-
-@var{film}:
--f dvd -muxrate 10080k -packetsize 2048
--s 720x480 -pix_fmt yuv420p -r 24000/1001
--codec:v mpeg2video -g 18 -b:v 6000k -maxrate:v 9000k -minrate:v 0 -bufsize:v 1835008
--ar 48000
--codec:a ac3 -b:a 448k
-@end example
-
-@strong{DV}
-@example
-@var{pal}:
--f dv
--s 720x576 -pix_fmt yuv420p -r 25
--ar 48000 -ac 2
-
-@var{ntsc}:
--f dv
--s 720x480 -pix_fmt yuv411p -r 30000/1001
--ar 48000 -ac 2
-
-@var{film}:
--f dv
--s 720x480 -pix_fmt yuv411p -r 24000/1001
--ar 48000 -ac 2
-@end example
-The @code{dv50} target is identical to the @code{dv} target except that the pixel format set is @code{yuv422p} for all three standards.
-
-Any user-set value for a parameter above will override the target preset value. In that case, the output may
-not comply with the target standard.
-
-@item -dn (@emph{input/output})
-As an input option, blocks all data streams of a file from being filtered or
-being automatically selected or mapped for any output. See @code{-discard}
-option to disable streams individually.
-
-As an output option, disables data recording i.e. automatic selection or
-mapping of any data stream. For full manual control see the @code{-map}
+@item -dn (@emph{output})
+Disable data recording. For full manual control see the @code{-map}
 option.
 
 @item -dframes @var{number} (@emph{output})
@@ -774,16 +655,6 @@ This option is similar to @option{-filter}, the only difference is that its
 argument is the name of the file from which a filtergraph description is to be
 read.
 
-@item -reinit_filter[:@var{stream_specifier}] @var{integer} (@emph{input,per-stream})
-This boolean option determines if the filtergraph(s) to which this stream is fed gets
-reinitialized when input frame parameters change mid-stream. This option is enabled by
-default as most video and all audio filters cannot handle deviation in input frame properties.
-Upon reinitialization, existing filter state is lost, like e.g. the frame count @code{n}
-reference available in some filters. Any frames buffered at time of reinitialization are lost.
-The properties where a change triggers reinitialization are,
-for video, frame resolution or pixel format;
-for audio, sample format, sample rate, channel count or channel layout.
-
 @item -filter_threads @var{nb_threads} (@emph{global})
 Defines how many threads are used to process a filter pipeline. Each pipeline
 will produce a thread pool with this many threads available for parallel processing.
@@ -796,19 +667,14 @@ Specify the preset for matching stream(s).
 Print encoding progress/statistics. It is on by default, to explicitly
 disable it you need to specify @code{-nostats}.
 
-@item -stats_period @var{time} (@emph{global})
-Set period at which encoding progress/statistics are updated. Default is 0.5 seconds.
-
 @item -progress @var{url} (@emph{global})
 Send program-friendly progress information to @var{url}.
 
-Progress information is written periodically and at the end of
+Progress information is written approximately every second and at the end of
 the encoding process. It is made of "@var{key}=@var{value}" lines. @var{key}
 consists of only alphanumeric characters. The last key of a sequence of
 progress information is always "progress".
 
-The update period is set using @code{-stats_period}.
-
 @anchor{stdin option}
 @item -stdin
 Enable interaction on standard input. On by default unless standard input is
@@ -860,6 +726,10 @@ ffmpeg -dump_attachment:t "" -i INPUT
 Technical note -- attachments are implemented as codec extradata, so this
 option can actually be used to extract extradata from any stream, not just
 attachments.
+
+@item -noautorotate
+Disable automatically rotating video based on file metadata.
+
 @end table
 
 @section Video Options
@@ -880,13 +750,6 @@ If in doubt use @option{-framerate} instead of the input option @option{-r}.
 As an output option, duplicate or drop input frames to achieve constant output
 frame rate @var{fps}.
 
-@item -fpsmax[:@var{stream_specifier}] @var{fps} (@emph{output,per-stream})
-Set maximum frame rate (Hz value, fraction or abbreviation).
-
-Clamps output frame rate when output framerate is auto-set and is higher than this value.
-Useful in batch processing or when input framerate is wrongly detected as very high.
-It cannot be set together with @code{-r}. It is ignored during streamcopy.
-
 @item -s[:@var{stream_specifier}] @var{size} (@emph{input/output,per-stream})
 Set frame size.
 
@@ -912,13 +775,8 @@ If used together with @option{-vcodec copy}, it will affect the aspect ratio
 stored at container level, but not the aspect ratio stored in encoded
 frames, if it exists.
 
-@item -vn (@emph{input/output})
-As an input option, blocks all video streams of a file from being filtered or
-being automatically selected or mapped for any output. See @code{-discard}
-option to disable streams individually.
-
-As an output option, disables video recording i.e. automatic selection or
-mapping of any video stream. For full manual control see the @code{-map}
+@item -vn (@emph{output})
+Disable video recording. For full manual control see the @code{-map}
 option.
 
 @item -vcodec @var{codec} (@emph{output})
@@ -948,18 +806,6 @@ Create the filtergraph specified by @var{filtergraph} and use it to
 filter the stream.
 
 This is an alias for @code{-filter:v}, see the @ref{filter_option,,-filter option}.
-
-@item -autorotate
-Automatically rotate the video according to file metadata. Enabled by
-default, use @option{-noautorotate} to disable it.
-
-@item -autoscale
-Automatically scale the video according to the resolution of first frame.
-Enabled by default, use @option{-noautoscale} to disable it. When autoscale is
-disabled, all output frames of filter graph might not be in the same resolution
-and may be inadequate for some encoder/muxer. Therefore, it is not recommended
-to disable it unless you really know what you are doing.
-Disable autoscale at your own risk.
 @end table
 
 @section Advanced Video options
@@ -989,8 +835,8 @@ factor if negative.
 Force interlacing support in encoder (MPEG-2 and MPEG-4 only).
 Use this option if your input file is interlaced and you want
 to keep the interlaced format for minimum losses.
-The alternative is to deinterlace the input stream by use of a filter
-such as @code{yadif} or @code{bwdif}, but deinterlacing introduces losses.
+The alternative is to deinterlace the input stream with
+@option{-deinterlace}, but deinterlacing introduces losses.
 @item -psnr
 Calculate PSNR of compressed frames.
 @item -vstats
@@ -1020,20 +866,12 @@ Deprecated see -bsf
 
 @item -force_key_frames[:@var{stream_specifier}] @var{time}[,@var{time}...] (@emph{output,per-stream})
 @item -force_key_frames[:@var{stream_specifier}] expr:@var{expr} (@emph{output,per-stream})
-@item -force_key_frames[:@var{stream_specifier}] source (@emph{output,per-stream})
-@item -force_key_frames[:@var{stream_specifier}] source_no_drop (@emph{output,per-stream})
+Force key frames at the specified timestamps, more precisely at the first
+frames after each specified time.
 
-@var{force_key_frames} can take arguments of the following form:
-
-@table @option
-
-@item @var{time}[,@var{time}...]
-If the argument consists of timestamps, ffmpeg will round the specified times to the nearest
-output timestamp as per the encoder time base and force a keyframe at the first frame having
-timestamp equal or greater than the computed timestamp. Note that if the encoder time base is too
-coarse, then the keyframes may be forced on frames with timestamps lower than the specified time.
-The default encoder time base is the inverse of the output framerate but may be set otherwise
-via @code{-enc_time_base}.
+If the argument is prefixed with @code{expr:}, the string @var{expr}
+is interpreted like an expression and is evaluated for each frame. A
+key frame is forced in case the evaluation is non-zero.
 
 If one of the times is "@code{chapters}[@var{delta}]", it is expanded into
 the time of the beginning of all chapters in the file, shifted by
@@ -1047,11 +885,6 @@ before the beginning of every chapter:
 -force_key_frames 0:05:00,chapters-0.1
 @end example
 
-@item expr:@var{expr}
-If the argument is prefixed with @code{expr:}, the string @var{expr}
-is interpreted like an expression and is evaluated for each frame. A
-key frame is forced in case the evaluation is non-zero.
-
 The expression in @var{expr} can contain the following constants:
 @table @option
 @item n
@@ -1079,18 +912,6 @@ starting from second 13:
 -force_key_frames expr:if(isnan(prev_forced_t),gte(t,13),gte(t,prev_forced_t+5))
 @end example
 
-@item source
-If the argument is @code{source}, ffmpeg will force a key frame if
-the current frame being encoded is marked as a key frame in its source.
-
-@item source_no_drop
-If the argument is @code{source_no_drop}, ffmpeg will force a key frame if
-the current frame being encoded is marked as a key frame in its source.
-In cases where this particular source frame has to be dropped,
-enforce the next available frame to become a key frame instead.
-
-@end table
-
 Note that forcing too many keyframes is very harmful for the lookahead
 algorithms of certain encoders: using fixed-GOP options or similar
 would be more efficient.
@@ -1111,27 +932,9 @@ device type:
 @item cuda
 @var{device} is the number of the CUDA device.
 
-The following options are recognized:
-@table @option
-@item primary_ctx
-If set to 1, uses the primary device context instead of creating a new one.
-@end table
-
-Examples:
-@table @emph
-@item -init_hw_device cuda:1
-Choose the second device on the system.
-
-@item -init_hw_device cuda:0,primary_ctx=1
-Choose the first device and use the primary device context.
-@end table
-
 @item dxva2
 @var{device} is the number of the Direct3D 9 display adapter.
 
-@item d3d11va
-@var{device} is the number of the Direct3D 11 display adapter.
-
 @item vaapi
 @var{device} is either an X11 display name or a DRM render node.
 If not specified, it will attempt to open the default X11 display (@emph{$DISPLAY})
@@ -1155,21 +958,9 @@ If not specified, it will attempt to open the default X11 display (@emph{$DISPLA
 @end table
 If not specified, @samp{auto_any} is used.
 (Note that it may be easier to achieve the desired result for QSV by creating the
-platform-appropriate subdevice (@samp{dxva2} or @samp{d3d11va} or @samp{vaapi}) and then deriving a
+platform-appropriate subdevice (@samp{dxva2} or @samp{vaapi}) and then deriving a
 QSV device from that.)
 
-Alternatively, @samp{child_device_type} helps to choose platform-appropriate subdevice type.
-On Windows @samp{d3d11va} is used as default subdevice type.
-
-Examples:
-@table @emph
-@item -init_hw_device qsv:hw,child_device_type=d3d11va
-Choose the GPU subdevice with type @samp{d3d11va} and create QSV device with @samp{MFX_IMPL_HARDWARE}.
-
-@item -init_hw_device qsv:hw,child_device_type=dxva2
-Choose the GPU subdevice with type @samp{dxva2} and create QSV device with @samp{MFX_IMPL_HARDWARE}.
-@end table
-
 @item opencl
 @var{device} selects the platform and device as @emph{platform_index.device_index}.
 
@@ -1207,35 +998,6 @@ Choose the GPU device on the second platform supporting the @emph{cl_khr_fp16}
 extension.
 @end table
 
-@item vulkan
-If @var{device} is an integer, it selects the device by its index in a
-system-dependent list of devices.  If @var{device} is any other string, it
-selects the first device with a name containing that string as a substring.
-
-The following options are recognized:
-@table @option
-@item debug
-If set to 1, enables the validation layer, if installed.
-@item linear_images
-If set to 1, images allocated by the hwcontext will be linear and locally mappable.
-@item instance_extensions
-A plus separated list of additional instance extensions to enable.
-@item device_extensions
-A plus separated list of additional device extensions to enable.
-@end table
-
-Examples:
-@table @emph
-@item -init_hw_device vulkan:1
-Choose the second device on the system.
-
-@item -init_hw_device vulkan:RADV
-Choose the first device with a name containing the string @emph{RADV}.
-
-@item -init_hw_device vulkan:0,instance_extensions=VK_KHR_wayland_surface+VK_KHR_xcb_surface
-Choose the first device and enable the Wayland and XCB instance extensions.
-@end table
-
 @end table
 
 @item -init_hw_device @var{type}[=@var{name}]@@@var{source}
@@ -1272,9 +1034,6 @@ Use VDPAU (Video Decode and Presentation API for Unix) hardware acceleration.
 @item dxva2
 Use DXVA2 (DirectX Video Acceleration) hardware acceleration.
 
-@item d3d11va
-Use D3D11VA (DirectX Video Acceleration) hardware acceleration.
-
 @item vaapi
 Use VAAPI (Video Acceleration API) hardware acceleration.
 
@@ -1308,9 +1067,7 @@ by name, or it can create a new device as if
 were called immediately before.
 
 @item -hwaccels
-List all hardware acceleration components enabled in this build of ffmpeg.
-Actual runtime availability depends on the hardware and its suitable driver
-being installed.
+List all hardware acceleration methods supported in this build of ffmpeg.
 
 @end table
 
@@ -1332,13 +1089,8 @@ Set the number of audio channels. For output streams it is set by
 default to the number of input audio channels. For input streams
 this option only makes sense for audio grabbing devices and raw demuxers
 and is mapped to the corresponding demuxer options.
-@item -an (@emph{input/output})
-As an input option, blocks all audio streams of a file from being filtered or
-being automatically selected or mapped for any output. See @code{-discard}
-option to disable streams individually.
-
-As an output option, disables audio recording i.e. automatic selection or
-mapping of any audio stream. For full manual control see the @code{-map}
+@item -an (@emph{output})
+Disable audio recording. For full manual control see the @code{-map}
 option.
 @item -acodec @var{codec} (@emph{input/output})
 Set the audio codec. This is an alias for @code{-codec:a}.
@@ -1373,13 +1125,8 @@ stereo but not 6 channels as 5.1. The default is to always try to guess. Use
 @table @option
 @item -scodec @var{codec} (@emph{input/output})
 Set the subtitle codec. This is an alias for @code{-codec:s}.
-@item -sn (@emph{input/output})
-As an input option, blocks all subtitle streams of a file from being filtered or
-being automatically selected or mapped for any output. See @code{-discard}
-option to disable streams individually.
-
-As an output option, disables subtitle recording i.e. automatic selection or
-mapping of any subtitle stream. For full manual control see the @code{-map}
+@item -sn (@emph{output})
+Disable subtitle recording. For full manual control see the @code{-map}
 option.
 @item -sbsf @var{bitstream_filter}
 Deprecated, see -bsf
@@ -1613,49 +1360,42 @@ it will usually display as 0 if not supported.
 Show benchmarking information during the encode.
 Shows real, system and user time used in various steps (audio/video encode/decode).
 @item -timelimit @var{duration} (@emph{global})
-Exit after ffmpeg has been running for @var{duration} seconds in CPU user time.
+Exit after ffmpeg has been running for @var{duration} seconds.
 @item -dump (@emph{global})
 Dump each input packet to stderr.
 @item -hex (@emph{global})
 When dumping packets, also dump the payload.
-@item -readrate @var{speed} (@emph{input})
-Limit input read speed.
-
-Its value is a floating-point positive number which represents the maximum duration of
-media, in seconds, that should be ingested in one second of wallclock time.
-Default value is zero and represents no imposed limitation on speed of ingestion.
-Value @code{1} represents real-time speed and is equivalent to @code{-re}.
-
-Mainly used to simulate a capture device or live input stream (e.g. when reading from a file).
-Should not be used with a low value when input is an actual capture device or live stream as
-it may cause packet loss.
-
-It is useful for when flow speed of output packets is important, such as live streaming.
 @item -re (@emph{input})
-Read input at native frame rate. This is equivalent to setting @code{-readrate 1}.
-@item -vsync @var{parameter} (@emph{global})
-@itemx -fps_mode[:@var{stream_specifier}] @var{parameter} (@emph{output,per-stream})
-Set video sync method / framerate mode. vsync is applied to all output video streams
-but can be overridden for a stream by setting fps_mode. vsync is deprecated and will be
-removed in the future.
-
-For compatibility reasons some of the values for vsync can be specified as numbers (shown
-in parentheses in the following table).
+Read input at native frame rate. Mainly used to simulate a grab device,
+or live input stream (e.g. when reading from a file). Should not be used
+with actual grab devices or live input streams (where it can cause packet
+loss).
+By default @command{ffmpeg} attempts to read the input(s) as fast as possible.
+This option will slow down the reading of the input(s) to the native frame rate
+of the input(s). It is useful for real-time output (e.g. live streaming).
+@item -loop_output @var{number_of_times}
+Repeatedly loop output for formats that support looping such as animated GIF
+(0 will loop the output infinitely).
+This option is deprecated, use -loop.
+@item -vsync @var{parameter}
+Video sync method.
+For compatibility reasons old values can be specified as numbers.
+Newly added values will have to be specified as strings always.
 
 @table @option
-@item passthrough (0)
+@item 0, passthrough
 Each frame is passed with its timestamp from the demuxer to the muxer.
-@item cfr (1)
+@item 1, cfr
 Frames will be duplicated and dropped to achieve exactly the requested
 constant frame rate.
-@item vfr (2)
+@item 2, vfr
 Frames are passed through with their timestamp or dropped so as to
 prevent 2 frames from having the same timestamp.
 @item drop
 As passthrough but destroys all timestamps, making the muxer generate
 fresh timestamps based on frame-rate.
-@item auto (-1)
-Chooses between cfr and vfr depending on muxer capabilities. This is the
+@item -1, auto
+Chooses between 1 and 2 depending on muxer capabilities. This is the
 default method.
 @end table
 
@@ -1686,17 +1426,6 @@ is enabled.
 
 This option has been deprecated. Use the @code{aresample} audio filter instead.
 
-@item -adrift_threshold @var{time}
-Set the minimum difference between timestamps and audio data (in seconds) to trigger
-adding/dropping samples to make it match the timestamps. This option effectively is
-a threshold to select between hard (add/drop) and soft (squeeze/stretch) compensation.
-@code{-async} must be set to a positive value.
-
-@item -apad @var{parameters} (@emph{output,per-stream})
-Pad the output audio stream(s). This is the same as applying @code{-af apad}.
-Argument is a string of filter parameters composed the same as with the @code{apad} filter.
-@code{-shortest} must be set for this output for the option to take effect.
-
 @item -copyts
 Do not process input timestamps, but keep their values without trying
 to sanitize them. In particular, do not remove the initial start time
@@ -1764,16 +1493,12 @@ Default value is 0.
 @item -bitexact (@emph{input/output})
 Enable bitexact mode for (de)muxer and (de/en)coder
 @item -shortest (@emph{output})
-Finish encoding when the shortest output stream ends.
+Finish encoding when the shortest input stream ends.
 @item -dts_delta_threshold
 Timestamp discontinuity delta threshold.
-@item -dts_error_threshold @var{seconds}
-Timestamp error delta threshold. This threshold use to discard crazy/damaged
-timestamps and the default is 30 hours which is arbitrarily picked and quite
-conservative.
-@item -muxdelay @var{seconds} (@emph{output})
+@item -muxdelay @var{seconds} (@emph{input})
 Set the maximum demux-decode delay.
-@item -muxpreload @var{seconds} (@emph{output})
+@item -muxpreload @var{seconds} (@emph{input})
 Set the initial demux-decode delay.
 @item -streamid @var{output-stream-index}:@var{new-value} (@emph{output})
 Assign a new stream-id value to an output stream. This option should be
@@ -1851,22 +1576,6 @@ graph will be added to the output file automatically, so we can simply write
 ffmpeg -i video.mkv -i image.png -filter_complex 'overlay' out.mkv
 @end example
 
-As a special exception, you can use a bitmap subtitle stream as input: it
-will be converted into a video with the same size as the largest video in
-the file, or 720x576 if no video is present. Note that this is an
-experimental and temporary solution. It will be removed once libavfilter has
-proper support for subtitles.
-
-For example, to hardcode subtitles on top of a DVB-T recording stored in
-MPEG-TS format, delaying the subtitles by 1 second:
-@example
-ffmpeg -i input.ts -filter_complex \
-  '[#0x2ef] setpts=PTS+1/TB [sub] ; [#0x2d0] [sub] overlay' \
-  -sn -map '#0x2dc' output.mkv
-@end example
-(0x2d0, 0x2dc and 0x2ef are the MPEG-TS PIDs of respectively the video,
-audio and subtitles streams; 0:0, 0:3 and 0:7 would have worked too)
-
 To generate 5 seconds of pure red video using lavfi @code{color} source:
 @example
 ffmpeg -filter_complex 'color=c=red' -t 5 out.mkv
@@ -1902,9 +1611,8 @@ not start from timestamp 0, such as transport streams.
 @item -thread_queue_size @var{size} (@emph{input})
 This option sets the maximum number of queued packets when reading from the
 file or device. With low latency / high rate live streams, packets may be
-discarded if they are not read in a timely manner; setting this value can
-force ffmpeg to use a separate input thread and read packets as soon as they
-arrive. By default ffmpeg only does this if multiple inputs are specified.
+discarded if they are not read in a timely manner; raising this value can
+avoid it.
 
 @item -sdp_file @var{file} (@emph{global})
 Print sdp information for an output stream to @var{file}.
@@ -1912,10 +1620,8 @@ This allows dumping sdp information when at least one output isn't an
 rtp stream. (Requires at least one of the output formats to be rtp).
 
 @item -discard (@emph{input})
-Allows discarding specific streams or frames from streams.
-Any input stream can be fully discarded, using value @code{all} whereas
-selective discarding of frames from a stream occurs at the demuxer
-and is not supported by all demuxers.
+Allows discarding specific streams or frames of streams at the demuxer.
+Not all demuxers support this.
 
 @table @option
 @item none
@@ -1943,15 +1649,8 @@ Stop and abort on various conditions. The following flags are available:
 @table @option
 @item empty_output
 No packets were passed to the muxer, the output is empty.
-@item empty_output_stream
-No packets were passed to the muxer in some of the output streams.
 @end table
 
-@item -max_error_rate (@emph{global})
-Set fraction of decoding frame failures across all inputs which when crossed
-ffmpeg will return exit code 69. Crossing this threshold does not terminate
-processing. Range is a floating-point number between 0 to 1. Default is 2/3.
-
 @item -xerror (@emph{global})
 Stop and exit on error
 
@@ -1964,30 +1663,24 @@ this buffer, in packets, for the matching output stream.
 The default value of this option should be high enough for most uses, so only
 touch this option if you are sure that you need it.
 
-@item -muxing_queue_data_threshold @var{bytes} (@emph{output,per-stream})
-This is a minimum threshold until which the muxing queue size is not taken into
-account. Defaults to 50 megabytes per stream, and is based on the overall size
-of packets passed to the muxer.
-
-@item -auto_conversion_filters (@emph{global})
-Enable automatically inserting format conversion filters in all filter
-graphs, including those defined by @option{-vf}, @option{-af},
-@option{-filter_complex} and @option{-lavfi}. If filter format negotiation
-requires a conversion, the initialization of the filters will fail.
-Conversions can still be performed by inserting the relevant conversion
-filter (scale, aresample) in the graph.
-On by default, to explicitly disable it you need to specify
-@code{-noauto_conversion_filters}.
-
-@item -bits_per_raw_sample[:@var{stream_specifier}] @var{value} (@emph{output,per-stream})
-Declare the number of bits per raw sample in the given output stream to be
-@var{value}. Note that this option sets the information provided to the
-encoder/muxer, it does not change the stream to conform to this value. Setting
-values that do not match the stream properties may result in encoding failures
-or invalid output files.
-
 @end table
 
+As a special exception, you can use a bitmap subtitle stream as input: it
+will be converted into a video with the same size as the largest video in
+the file, or 720x576 if no video is present. Note that this is an
+experimental and temporary solution. It will be removed once libavfilter has
+proper support for subtitles.
+
+For example, to hardcode subtitles on top of a DVB-T recording stored in
+MPEG-TS format, delaying the subtitles by 1 second:
+@example
+ffmpeg -i input.ts -filter_complex \
+  '[#0x2ef] setpts=PTS+1/TB [sub] ; [#0x2d0] [sub] overlay' \
+  -sn -map '#0x2dc' output.mkv
+@end example
+(0x2d0, 0x2dc and 0x2ef are the MPEG-TS PIDs of respectively the video,
+audio and subtitles streams; 0:0, 0:3 and 0:7 would have worked too)
+
 @section Preset files
 A preset file contains a sequence of @var{option}=@var{value} pairs,
 one for each line, specifying a sequence of options which would be
@@ -2264,7 +1957,6 @@ ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
 @ifset config-avfilter
 @include filters.texi
 @end ifset
-@include general_contents.texi
 @end ifset
 
 @chapter See Also

doc/ffplay.texi

@@ -34,6 +34,10 @@ various FFmpeg APIs.
 Force displayed width.
 @item -y @var{height}
 Force displayed height.
+@item -s @var{size}
+Set frame size (WxH or abbreviation), needed for videos which do
+not contain a header with the frame size like raw YUV.  This option
+has been deprecated in favor of private options, try -video_size.
 @item -fs
 Start in fullscreen mode.
 @item -an
@@ -62,8 +66,6 @@ Set custom interval, in seconds, for seeking using left/right keys. Default is 1
 Disable graphical display.
 @item -noborder
 Borderless window.
-@item -alwaysontop
-Window always on top. Available on: X11 with SDL >= 2.0.5, Windows SDL >= 2.0.6.
 @item -volume
 Set the startup volume. 0 means silence, 100 means no volume reduction or
 amplification. Negative values are treated as 0, values above 100 are treated
@@ -122,12 +124,15 @@ Read @var{input_url}.
 
 @section Advanced options
 @table @option
+@item -pix_fmt @var{format}
+Set pixel format.
+This option has been deprecated in favor of private options, try -pixel_format.
+
 @item -stats
 Print several playback statistics, in particular show the stream
 duration, the codec parameters, the current position in the stream and
-the audio/video synchronisation drift. It is shown by default, unless the
-log level is lower than @code{info}. Its display can be forced by manually
-specifying this option. To disable it, you need to specify @code{-nostats}.
+the audio/video synchronisation drift. It is on by default, to
+explicitly disable it you need to specify @code{-nostats}.
 
 @item -fast
 Non-spec-compliant optimizations.
@@ -190,12 +195,6 @@ input as soon as possible. Enabled by default for realtime streams, where data
 may be dropped if not read in time. Use this option to enable infinite buffers
 for all inputs, use @option{-noinfbuf} to disable it.
 
-@item -filter_threads @var{nb_threads}
-Defines how many threads are used to process a filter pipeline. Each pipeline
-will produce a thread pool with this many threads available for parallel
-processing. The default is 0 which means that the thread count will be
-determined by the number of available CPUs.
-
 @end table
 
 @section While playing
@@ -214,6 +213,8 @@ Pause.
 Toggle mute.
 
 @item 9, 0
+Decrease and increase volume respectively.
+
 @item /, *
 Decrease and increase volume respectively.
 
@@ -285,7 +286,6 @@ Toggle full screen.
 @ifset config-avfilter
 @include filters.texi
 @end ifset
-@include general_contents.texi
 @end ifset
 
 @chapter See Also

doc/ffprobe.texi

@@ -12,7 +12,7 @@
 
 @chapter Synopsis
 
-ffprobe [@var{options}] @file{input_url}
+ffprobe [@var{options}] [@file{input_url}]
 
 @chapter Description
 @c man begin DESCRIPTION
@@ -28,9 +28,6 @@ If a url is specified in input, ffprobe will try to open and
 probe the url content. If the url cannot be opened or recognized as
 a multimedia file, a positive exit code is returned.
 
-If no output is specified as output with @option{o} ffprobe will write
-to stdout.
-
 ffprobe may be employed both as a standalone application or in
 combination with a textual filter, which may perform more
 sophisticated processing, e.g. statistical processing or plotting.
@@ -338,12 +335,6 @@ Show information about all pixel formats supported by FFmpeg.
 Pixel format information for each format is printed within a section
 with name "PIXEL_FORMAT".
 
-@item -show_optional_fields @var{value}
-Some writers viz. JSON and XML, omit the printing of fields with invalid or non-applicable values,
-while other writers always print them. This option enables one to control this behaviour.
-Valid values are @code{always}/@code{1}, @code{never}/@code{0} and @code{auto}/@code{-1}.
-Default is @var{auto}.
-
 @item -bitexact
 Force bitexact output, useful to produce output which is not dependent
 on the specific build.
@@ -351,10 +342,6 @@ on the specific build.
 @item -i @var{input_url}
 Read @var{input_url}.
 
-@item -o @var{output_url}
-Write output to @var{output_url}. If not specified, the output is sent
-to stdout.
-
 @end table
 @c man end
 
@@ -438,7 +425,7 @@ The @code{csv} writer is equivalent to @code{compact}, but supports
 different defaults.
 
 Each section is printed on a single line.
-If no option is specified, the output has the form:
+If no option is specifid, the output has the form:
 @example
 section|key1=val1| ... |keyN=valN
 @end example
@@ -604,7 +591,7 @@ This option automatically sets @option{fully_qualified} to 1.
 @end table
 
 For more information about the XML format, see
-@url{https://www.w3.org/XML/}.
+@url{http://www.w3.org/XML/}.
 @c man end WRITERS
 
 @chapter Timecode
@@ -655,7 +642,6 @@ DV, GXF and AVI timecodes are available in format metadata
 @ifset config-avfilter
 @include filters.texi
 @end ifset
-@include general_contents.texi
 @end ifset
 
 @chapter See Also

doc/ffprobe.xsd

@@ -29,18 +29,22 @@
     </xsd:complexType>
 
     <xsd:complexType name="framesType">
-        <xsd:choice minOccurs="0" maxOccurs="unbounded">
-            <xsd:element name="frame"    type="ffprobe:frameType"/>
-            <xsd:element name="subtitle" type="ffprobe:subtitleType"/>
-        </xsd:choice>
+        <xsd:sequence>
+            <xsd:choice minOccurs="0" maxOccurs="unbounded">
+                <xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
+                <xsd:element name="subtitle" type="ffprobe:subtitleType" minOccurs="0" maxOccurs="unbounded"/>
+            </xsd:choice>
+        </xsd:sequence>
     </xsd:complexType>
 
     <xsd:complexType name="packetsAndFramesType">
-        <xsd:choice minOccurs="0" maxOccurs="unbounded">
-            <xsd:element name="packet"   type="ffprobe:packetType"/>
-            <xsd:element name="frame"    type="ffprobe:frameType"/>
-            <xsd:element name="subtitle" type="ffprobe:subtitleType"/>
-        </xsd:choice>
+        <xsd:sequence>
+            <xsd:choice minOccurs="0" maxOccurs="unbounded">
+                <xsd:element name="packet" type="ffprobe:packetType" minOccurs="0" maxOccurs="unbounded"/>
+                <xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
+                <xsd:element name="subtitle" type="ffprobe:subtitleType" minOccurs="0" maxOccurs="unbounded"/>
+            </xsd:choice>
+        </xsd:sequence>
     </xsd:complexType>
 
     <xsd:complexType name="packetType">
@@ -57,6 +61,8 @@
       <xsd:attribute name="dts_time"      type="xsd:float" />
       <xsd:attribute name="duration"      type="xsd:long"  />
       <xsd:attribute name="duration_time" type="xsd:float" />
+      <xsd:attribute name="convergence_duration"      type="xsd:long"  />
+      <xsd:attribute name="convergence_duration_time" type="xsd:float" />
       <xsd:attribute name="size"          type="xsd:long" use="required" />
       <xsd:attribute name="pos"           type="xsd:long"  />
       <xsd:attribute name="flags"         type="xsd:string" use="required" />
@@ -86,6 +92,8 @@
       <xsd:attribute name="key_frame"     type="xsd:int"    use="required"/>
       <xsd:attribute name="pts"           type="xsd:long" />
       <xsd:attribute name="pts_time"      type="xsd:float"/>
+      <xsd:attribute name="pkt_pts"       type="xsd:long" />
+      <xsd:attribute name="pkt_pts_time"  type="xsd:float"/>
       <xsd:attribute name="pkt_dts"       type="xsd:long" />
       <xsd:attribute name="pkt_dts_time"  type="xsd:float"/>
       <xsd:attribute name="best_effort_timestamp"      type="xsd:long" />
@@ -139,25 +147,11 @@
         </xsd:sequence>
     </xsd:complexType>
     <xsd:complexType name="frameSideDataType">
-        <xsd:sequence>
-            <xsd:element name="timecodes" type="ffprobe:frameSideDataTimecodeList" minOccurs="0" maxOccurs="1"/>
-        </xsd:sequence>
-
         <xsd:attribute name="side_data_type"              type="xsd:string"/>
         <xsd:attribute name="side_data_size"              type="xsd:int"   />
         <xsd:attribute name="timecode"                    type="xsd:string"/>
     </xsd:complexType>
 
-    <xsd:complexType name="frameSideDataTimecodeList">
-        <xsd:sequence>
-            <xsd:element name="timecode" type="ffprobe:frameSideDataTimecodeType" minOccurs="0" maxOccurs="unbounded"/>
-        </xsd:sequence>
-    </xsd:complexType>
-
-    <xsd:complexType name="frameSideDataTimecodeType">
-        <xsd:attribute name="value"              type="xsd:string"/>
-    </xsd:complexType>
-
     <xsd:complexType name="subtitleType">
       <xsd:attribute name="media_type"         type="xsd:string" fixed="subtitle" use="required"/>
       <xsd:attribute name="pts"                type="xsd:long" />
@@ -193,11 +187,6 @@
       <xsd:attribute name="clean_effects"    type="xsd:int" use="required" />
       <xsd:attribute name="attached_pic"     type="xsd:int" use="required" />
       <xsd:attribute name="timed_thumbnails" type="xsd:int" use="required" />
-      <xsd:attribute name="captions"         type="xsd:int" use="required" />
-      <xsd:attribute name="descriptions"     type="xsd:int" use="required" />
-      <xsd:attribute name="metadata"         type="xsd:int" use="required" />
-      <xsd:attribute name="dependent"        type="xsd:int" use="required" />
-      <xsd:attribute name="still_image"      type="xsd:int" use="required" />
     </xsd:complexType>
 
     <xsd:complexType name="streamType">
@@ -212,10 +201,10 @@
       <xsd:attribute name="codec_long_name"  type="xsd:string" />
       <xsd:attribute name="profile"          type="xsd:string" />
       <xsd:attribute name="codec_type"       type="xsd:string" />
+      <xsd:attribute name="codec_time_base"  type="xsd:string" use="required"/>
       <xsd:attribute name="codec_tag"        type="xsd:string" use="required"/>
       <xsd:attribute name="codec_tag_string" type="xsd:string" use="required"/>
       <xsd:attribute name="extradata"        type="xsd:string" />
-      <xsd:attribute name="extradata_size"   type="xsd:int"    />
       <xsd:attribute name="extradata_hash"   type="xsd:string" />
 
       <!-- video attributes -->
@@ -223,8 +212,6 @@
       <xsd:attribute name="height"               type="xsd:int"/>
       <xsd:attribute name="coded_width"          type="xsd:int"/>
       <xsd:attribute name="coded_height"         type="xsd:int"/>
-      <xsd:attribute name="closed_captions"      type="xsd:boolean"/>
-      <xsd:attribute name="film_grain"           type="xsd:boolean"/>
       <xsd:attribute name="has_b_frames"         type="xsd:int"/>
       <xsd:attribute name="sample_aspect_ratio"  type="xsd:string"/>
       <xsd:attribute name="display_aspect_ratio" type="xsd:string"/>
@@ -236,6 +223,7 @@
       <xsd:attribute name="color_primaries"      type="xsd:string"/>
       <xsd:attribute name="chroma_location"      type="xsd:string"/>
       <xsd:attribute name="field_order"          type="xsd:string"/>
+      <xsd:attribute name="timecode"             type="xsd:string"/>
       <xsd:attribute name="refs"                 type="xsd:int"/>
 
       <!-- audio attributes -->
@@ -270,6 +258,10 @@
       <xsd:attribute name="program_id"           type="xsd:int"    use="required"/>
       <xsd:attribute name="program_num"          type="xsd:int"    use="required"/>
       <xsd:attribute name="nb_streams"           type="xsd:int"    use="required"/>
+      <xsd:attribute name="start_time"           type="xsd:float"/>
+      <xsd:attribute name="start_pts"            type="xsd:long"/>
+      <xsd:attribute name="end_time"             type="xsd:float"/>
+      <xsd:attribute name="end_pts"              type="xsd:long"/>
       <xsd:attribute name="pmt_pid"              type="xsd:int"    use="required"/>
       <xsd:attribute name="pcr_pid"              type="xsd:int"    use="required"/>
     </xsd:complexType>
@@ -351,6 +343,7 @@
       <xsd:attribute name="hwaccel"    type="xsd:int" use="required"/>
       <xsd:attribute name="planar"     type="xsd:int" use="required"/>
       <xsd:attribute name="rgb"        type="xsd:int" use="required"/>
+      <xsd:attribute name="pseudopal"  type="xsd:int" use="required"/>
       <xsd:attribute name="alpha"      type="xsd:int" use="required"/>
     </xsd:complexType>
 

doc/fftools-common-opts.texi

@@ -34,24 +34,27 @@ Possible forms of stream specifiers are:
 @table @option
 @item @var{stream_index}
 Matches the stream with this index. E.g. @code{-threads:1 4} would set the
-thread count for the second stream to 4. If @var{stream_index} is used as an
-additional stream specifier (see below), then it selects stream number
-@var{stream_index} from the matching streams. Stream numbering is based on the
-order of the streams as detected by libavformat except when a program ID is
-also specified. In this case it is based on the ordering of the streams in the
-program.
-@item @var{stream_type}[:@var{additional_stream_specifier}]
+thread count for the second stream to 4.
+@item @var{stream_type}[:@var{stream_index}]
 @var{stream_type} is one of following: 'v' or 'V' for video, 'a' for audio, 's'
 for subtitle, 'd' for data, and 't' for attachments. 'v' matches all video
 streams, 'V' only matches video streams which are not attached pictures, video
-thumbnails or cover arts. If @var{additional_stream_specifier} is used, then
-it matches streams which both have this type and match the
-@var{additional_stream_specifier}. Otherwise, it matches all streams of the
-specified type.
-@item p:@var{program_id}[:@var{additional_stream_specifier}]
-Matches streams which are in the program with the id @var{program_id}. If
-@var{additional_stream_specifier} is used, then it matches streams which both
-are part of the program and match the @var{additional_stream_specifier}.
+thumbnails or cover arts.  If @var{stream_index} is given, then it matches
+stream number @var{stream_index} of this type. Otherwise, it matches all
+streams of this type.
+@item p:@var{program_id}[:@var{stream_index}] or p:@var{program_id}[:@var{stream_type}[:@var{stream_index}]] or
+p:@var{program_id}:m:@var{key}[:@var{value}]
+In first version, if @var{stream_index} is given, then it matches the stream with number @var{stream_index}
+in the program with the id @var{program_id}. Otherwise, it matches all streams in the
+program. In the second version, @var{stream_type} is one of following: 'v' for video, 'a' for audio, 's'
+for subtitle, 'd' for data. If @var{stream_index} is also given, then it matches
+stream number @var{stream_index} of this type in the program with the id @var{program_id}.
+Otherwise, if only @var{stream_type} is given, it matches all
+streams of this type in the program with the id @var{program_id}.
+In the third version matches streams in the program with the id @var{program_id} with the metadata
+tag @var{key} having the specified value. If
+@var{value} is not given, matches streams that contain the given tag with any
+value.
 
 @item #@var{stream_id} or i:@var{stream_id}
 Match the stream by stream id (e.g. PID in MPEG-TS container).
@@ -107,24 +110,13 @@ Print detailed information about the muxer named @var{muxer_name}. Use the
 @option{-formats} option to get a list of all muxers and demuxers.
 
 @item filter=@var{filter_name}
-Print detailed information about the filter named @var{filter_name}. Use the
+Print detailed information about the filter name @var{filter_name}. Use the
 @option{-filters} option to get a list of all filters.
-
-@item bsf=@var{bitstream_filter_name}
-Print detailed information about the bitstream filter named @var{bitstream_filter_name}.
-Use the @option{-bsfs} option to get a list of all bitstream filters.
-
-@item protocol=@var{protocol_name}
-Print detailed information about the protocol named @var{protocol_name}.
-Use the @option{-protocols} option to get a list of all protocols.
 @end table
 
 @item -version
 Show version.
 
-@item -buildconf
-Show the build configuration, one option per line.
-
 @item -formats
 Show available formats (including devices).
 
@@ -167,9 +159,6 @@ Show available sample formats.
 @item -layouts
 Show channel names and standard channel layouts.
 
-@item -dispositions
-Show stream dispositions.
-
 @item -colors
 Show recognized color names.
 
@@ -246,15 +235,17 @@ ffmpeg [...] -loglevel +repeat
 By default the program logs to stderr. If coloring is supported by the
 terminal, colors are used to mark errors and warnings. Log coloring
 can be disabled setting the environment variable
-@env{AV_LOG_FORCE_NOCOLOR}, or can be forced setting
+@env{AV_LOG_FORCE_NOCOLOR} or @env{NO_COLOR}, or can be forced setting
 the environment variable @env{AV_LOG_FORCE_COLOR}.
+The use of the environment variable @env{NO_COLOR} is deprecated and
+will be dropped in a future FFmpeg version.
 
 @item -report
-Dump full command line and log output to a file named
+Dump full command line and console output to a file named
 @code{@var{program}-@var{YYYYMMDD}-@var{HHMMSS}.log} in the current
 directory.
 This file can be useful for bug reports.
-It also implies @code{-loglevel debug}.
+It also implies @code{-loglevel verbose}.
 
 Setting the environment variable @env{FFREPORT} to any value has the
 same effect. If the value is a ':'-separated key=value sequence, these
@@ -355,19 +346,6 @@ Possible flags for this option are:
 @item k8
 @end table
 @end table
-
-@item -cpucount @var{count} (@emph{global})
-Override detection of CPU count. This option is intended
-for testing. Do not use it unless you know what you're doing.
-@example
-ffmpeg -cpucount 2
-@end example
-
-@item -max_alloc @var{bytes}
-Set the maximum size limit for allocating a block on the heap by ffmpeg's
-family of malloc functions. Exercise @strong{extreme caution} when using
-this option. Don't use if you do not understand the full consequence of doing so.
-Default is INT_MAX.
 @end table
 
 @section AVOptions
@@ -393,15 +371,7 @@ ffmpeg -i input.flac -id3v2_version 3 out.mp3
 @end example
 
 All codec AVOptions are per-stream, and thus a stream specifier
-should be attached to them:
-@example
-ffmpeg -i multichannel.mxf -map 0:v:0 -map 0:a:0 -map 0:a:0 -c:a:0 ac3 -b:a:0 640k -ac:a:1 2 -c:a:1 aac -b:2 128k out.mp4
-@end example
-
-In the above example, a multichannel audio stream is mapped twice for output.
-The first instance is encoded with codec ac3 and bitrate 640k.
-The second instance is downmixed to 2 channels and encoded with codec aac. A bitrate of 128k is specified for it using
-absolute index of the output stream.
+should be attached to them.
 
 Note: the @option{-nooption} syntax cannot be used for boolean
 AVOptions, use @option{-option 0}/@option{-option 1}.

doc/formats.texi

@@ -27,10 +27,6 @@ stream information. A higher value will enable detecting more
 information in case it is dispersed into the stream, but will increase
 latency. Must be an integer not lesser than 32. It is 5000000 by default.
 
-@item max_probe_packets @var{integer} (@emph{input})
-Set the maximum number of buffered packets when probing a codec.
-Default is 2500 packets.
-
 @item packetsize @var{integer} (@emph{output})
 Set packet size.
 
@@ -49,6 +45,7 @@ Generate missing PTS if DTS is present.
 Ignore DTS if PTS is set. Inert when nofillin is set.
 @item ignidx
 Ignore index.
+@item keepside (@emph{deprecated},@emph{inert})
 @item nobuffer
 Reduce the latency introduced by buffering during initial input streams analysis.
 @item nofillin
@@ -69,6 +66,7 @@ This ensures that file and data checksums are reproducible and match between
 platforms. Its primary use is for regression testing.
 @item flush_packets
 Write out packets immediately.
+@item latm (@emph{deprecated},@emph{inert})
 @item shortest
 Stop muxing at the end of the shortest stream.
 It may be needed to increase max_interleave_delta to avoid flushing the longer
@@ -141,7 +139,7 @@ Consider things that a sane encoder should not do as an error.
 
 @item max_interleave_delta @var{integer} (@emph{output})
 Set maximum buffering duration for interleaving. The duration is
-expressed in microseconds, and defaults to 10000000 (10 seconds).
+expressed in microseconds, and defaults to 1000000 (1 second).
 
 To ensure all the streams are interleaved correctly, libavformat will
 wait until it has at least one packet for each stream before actually
@@ -213,7 +211,7 @@ is @code{0} (meaning that no offset is applied).
 @item dump_separator @var{string} (@emph{input})
 Separator used to separate the fields printed on the command line about the
 Stream parameters.
-For example, to separate the fields with newlines and indentation:
+For example to separate the fields with newlines and indention:
 @example
 ffprobe -dump_separator "
                           "  -i ~/videos/matrixbench_mpeg2.mpg
@@ -226,28 +224,6 @@ would require too many resources due to a large number of streams.
 @item skip_estimate_duration_from_pts @var{bool} (@emph{input})
 Skip estimation of input duration when calculated using PTS.
 At present, applicable for MPEG-PS and MPEG-TS.
-
-@item strict, f_strict @var{integer} (@emph{input/output})
-Specify how strictly to follow the standards. @code{f_strict} is deprecated and
-should be used only via the @command{ffmpeg} tool.
-
-Possible values:
-@table @samp
-@item very
-strictly conform to an older more strict version of the spec or reference software
-@item strict
-strictly conform to all the things in the spec no matter what consequences
-@item normal
-
-@item unofficial
-allow unofficial extensions
-@item experimental
-allow non standardized experimental things, experimental
-(unfinished/work in progress/not well tested) decoders and encoders.
-Note: experimental decoders can pose a security risk, do not use this for
-decoding untrusted input.
-@end table
-
 @end table
 
 @c man end FORMAT OPTIONS
@@ -258,10 +234,30 @@ decoding untrusted input.
 Format stream specifiers allow selection of one or more streams that
 match specific properties.
 
+Possible forms of stream specifiers are:
+@table @option
+@item @var{stream_index}
+Matches the stream with this index.
+
+@item @var{stream_type}[:@var{stream_index}]
+@var{stream_type} is one of following: 'v' for video, 'a' for audio,
+'s' for subtitle, 'd' for data, and 't' for attachments. If
+@var{stream_index} is given, then it matches the stream number
+@var{stream_index} of this type. Otherwise, it matches all streams of
+this type.
+
+@item p:@var{program_id}[:@var{stream_index}]
+If @var{stream_index} is given, then it matches the stream with number
+@var{stream_index} in the program with the id
+@var{program_id}. Otherwise, it matches all streams in the program.
+
+@item #@var{stream_id}
+Matches the stream by a format-specific ID.
+@end table
+
 The exact semantics of stream specifiers is defined by the
 @code{avformat_match_stream_specifier()} function declared in the
-@file{libavformat/avformat.h} header and documented in the
-@ref{Stream specifiers,,Stream specifiers section in the ffmpeg(1) manual,ffmpeg}.
+@file{libavformat/avformat.h} header.
 
 @ifclear config-writeonly
 @include demuxers.texi

doc/git-howto.texi

@@ -187,18 +187,11 @@ to make sure you don't have untracked files or deletions.
 git add [-i|-p|-A] <filenames/dirnames>
 @end example
 
-Make sure you have told Git your name, email address and GPG key
+Make sure you have told Git your name and email address
 
 @example
 git config --global user.name "My Name"
 git config --global user.email my@@email.invalid
-git config --global user.signingkey ABCDEF0123245
-@end example
-
-Enable signing all commits or use -S
-
-@example
-git config --global commit.gpgsign true
 @end example
 
 Use @option{--global} to set the global configuration for all your Git checkouts.
@@ -224,46 +217,16 @@ git config --global core.editor
 or set by one of the following environment variables:
 @var{GIT_EDITOR}, @var{VISUAL} or @var{EDITOR}.
 
-@section Writing a commit message
+Log messages should be concise but descriptive. Explain why you made a change,
+what you did will be obvious from the changes themselves most of the time.
+Saying just "bug fix" or "10l" is bad. Remember that people of varying skill
+levels look at and educate themselves while reading through your code. Don't
+include filenames in log messages, Git provides that information.
 
-Log messages should be concise but descriptive.
-
-The first line must contain the context, a colon and a very short
-summary of what the commit does. Details can be added, if necessary,
-separated by an empty line. These details should not exceed 60-72 characters
-per line, except when containing code.
-
-Example of a good commit message:
-
-@example
-avcodec/cbs: add a helper to read extradata within packet side data
-
-Using ff_cbs_read() on the raw buffer will not parse it as extradata,
-resulting in parsing errors for example when handling ISOBMFF avcC.
-This helper works around that.
-@end example
-
-@example
-ptr might be NULL
-@end example
-
-If the summary on the first line is not enough, in the body of the message,
-explain why you made a change, what you did will be obvious from the changes
-themselves most of the time. Saying just "bug fix" or "10l" is bad. Remember
-that people of varying skill levels look at and educate themselves while
-reading through your code. Don't include filenames in log messages except in
-the context, Git provides that information.
-
-If the commit fixes a registered issue, state it in a separate line of the
-body: @code{Fix Trac ticket #42.}
-
-The first line will be used to name
+Possibly make the commit message have a terse, descriptive first line, an
+empty line and then a full description. The first line will be used to name
 the patch by @command{git format-patch}.
 
-Common mistakes for the first line, as seen in @command{git log --oneline}
-include: missing context at the beginning; description of what the code did
-before the patch; line too long or wrapped to the second line.
-
 @section Preparing a patchset
 
 @example
@@ -430,19 +393,6 @@ git checkout -b svn_23456 $SHA1
 where @var{$SHA1} is the commit hash from the @command{git log} output.
 
 
-@chapter gpg key generation
-
-If you have no gpg key yet, we recommend that you create a ed25519 based key as it
-is small, fast and secure. Especially it results in small signatures in git.
-
-@example
-gpg --default-new-key-algo "ed25519/cert,sign+cv25519/encr" --quick-generate-key "human@@server.com"
-@end example
-
-When generating a key, make sure the email specified matches the email used in git as some sites like
-github consider mismatches a reason to declare such commits unverified. After generating a key you
-can add it to the MAINTAINER file and upload it to a keyserver.
-
 @chapter Pre-push checklist
 
 Once you have a set of commits that you feel are ready for pushing,

doc/indevs.texi

@@ -178,9 +178,6 @@ Capture the mouse pointer. Default is 0.
 @item -capture_mouse_clicks
 Capture the screen mouse clicks. Default is 0.
 
-@item -capture_raw_data
-Capture the raw device data. Default is 0.
-Using this option may result in receiving the underlying data delivered to the AVFoundation framework. E.g. for muxed devices that sends raw DV data to the framework (like tape-based camcorders), setting this option to false results in extracted video frames captured in the designated pixel format only. Setting this option to true results in receiving the raw DV stream untouched.
 @end table
 
 @subsection Examples
@@ -211,13 +208,6 @@ Record video from the system default video device using the pixel format bgr0 an
 $ ffmpeg -f avfoundation -pixel_format bgr0 -i "default:none" out.avi
 @end example
 
-@item
-Record raw DV data from a suitable input device and write the output into out.dv:
-@example
-$ ffmpeg -f avfoundation -capture_raw_data true -i "zr100:none" out.dv
-@end example
-
-
 @end itemize
 
 @section bktr
@@ -277,8 +267,8 @@ audio track.
 
 @item list_devices
 If set to @option{true}, print a list of devices and exit.
-Defaults to @option{false}. This option is deprecated, please use the
-@code{-sources} option of ffmpeg to list the available input devices.
+Defaults to @option{false}. Alternatively you can use the @code{-sources}
+option of ffmpeg to list the available input devices.
 
 @item list_formats
 If set to @option{true}, print a list of supported formats and exit.
@@ -292,35 +282,25 @@ as @option{pal} (3 letters).
 Default behavior is autodetection of the input video format, if the hardware
 supports it.
 
+@item bm_v210
+This is a deprecated option, you can use @option{raw_format} instead.
+If set to @samp{1}, video is captured in 10 bit v210 instead
+of uyvy422. Not all Blackmagic devices support this option.
+
 @item raw_format
 Set the pixel format of the captured video.
 Available values are:
 @table @samp
-@item auto
-
-This is the default which means 8-bit YUV 422 or 8-bit ARGB if format
-autodetection is used, 8-bit YUV 422 otherwise.
-
 @item uyvy422
 
-8-bit YUV 422.
-
 @item yuv422p10
 
-10-bit YUV 422.
-
 @item argb
 
-8-bit RGB.
-
 @item bgra
 
-8-bit RGB.
-
 @item rgb10
 
-10-bit RGB.
-
 @end table
 
 @item teletext_lines
@@ -344,33 +324,14 @@ Defines number of audio channels to capture. Must be @samp{2}, @samp{8} or @samp
 Defaults to @samp{2}.
 
 @item duplex_mode
-Sets the decklink device duplex/profile mode. Must be @samp{unset}, @samp{half}, @samp{full},
-@samp{one_sub_device_full}, @samp{one_sub_device_half}, @samp{two_sub_device_full},
-@samp{four_sub_device_half}
+Sets the decklink device duplex mode. Must be @samp{unset}, @samp{half} or @samp{full}.
 Defaults to @samp{unset}.
 
-Note: DeckLink SDK 11.0 have replaced the duplex property by a profile property.
-For the DeckLink Duo 2 and DeckLink Quad 2, a profile is shared between any 2
-sub-devices that utilize the same connectors. For the DeckLink 8K Pro, a profile
-is shared between all 4 sub-devices. So DeckLink 8K Pro support four profiles.
-
-Valid profile modes for DeckLink 8K Pro(with DeckLink SDK >= 11.0):
-@samp{one_sub_device_full}, @samp{one_sub_device_half}, @samp{two_sub_device_full},
-@samp{four_sub_device_half}
-
-Valid profile modes for DeckLink Quad 2 and DeckLink Duo 2:
-@samp{half}, @samp{full}
-
 @item timecode_format
 Timecode type to include in the frame and video stream metadata. Must be
 @samp{none}, @samp{rp188vitc}, @samp{rp188vitc2}, @samp{rp188ltc},
-@samp{rp188hfr}, @samp{rp188any}, @samp{vitc}, @samp{vitc2}, or @samp{serial}.
-Defaults to @samp{none} (not included).
-
-In order to properly support 50/60 fps timecodes, the ordering of the queried
-timecode types for @samp{rp188any} is HFR, VITC1, VITC2 and LTC for >30 fps
-content. Note that this is slightly different to the ordering used by the
-DeckLink API, which is HFR, VITC1, LTC, VITC2.
+@samp{rp188any}, @samp{vitc}, @samp{vitc2}, or @samp{serial}. Defaults to
+@samp{none} (not included).
 
 @item video_input
 Sets the video input source. Must be @samp{unset}, @samp{sdi}, @samp{hdmi},
@@ -424,20 +385,6 @@ Either sync could go wrong by 1 frame or in a rarer case
 @option{timestamp_align} seconds.
 Defaults to @samp{0}.
 
-@item wait_for_tc (@emph{bool})
-Drop frames till a frame with timecode is received. Sometimes serial timecode
-isn't received with the first input frame. If that happens, the stored stream
-timecode will be inaccurate. If this option is set to @option{true}, input frames
-are dropped till a frame with timecode is received.
-Option @var{timecode_format} must be specified.
-Defaults to @option{false}.
-
-@item enable_klv(@emph{bool})
-If set to @option{true}, extracts KLV data from VANC and outputs KLV packets.
-KLV VANC packets are joined based on MID and PSC fields and aggregated into
-one KLV packet.
-Defaults to @option{false}.
-
 @end table
 
 @subsection Examples
@@ -447,7 +394,7 @@ Defaults to @option{false}.
 @item
 List input devices:
 @example
-ffmpeg -sources decklink
+ffmpeg -f decklink -list_devices 1 -i dummy
 @end example
 
 @item
@@ -465,7 +412,7 @@ ffmpeg -format_code Hi50 -f decklink -i 'Intensity Pro' -c:a copy -c:v copy outp
 @item
 Capture video clip at 1080i50 10 bit:
 @example
-ffmpeg -raw_format yuv422p10 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder' -c:a copy -c:v copy output.avi
+ffmpeg -bm_v210 1 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder' -c:a copy -c:v copy output.avi
 @end example
 
 @item
@@ -625,12 +572,6 @@ Save the currently used video capture filter device and its
 parameters (if the filter supports it) to a file.
 If a file with the same name exists it will be overwritten.
 
-@item use_video_device_timestamps
-If set to @option{false}, the timestamp for video frames will be
-derived from the wallclock instead of the timestamp provided by
-the capture device. This allows working around devices that
-provide unreliable timestamps.
-
 @end table
 
 @subsection Examples
@@ -846,7 +787,7 @@ ffplay -f iec61883 -i auto
 Grab and record the input of a FireWire DV/HDV device,
 using a packet buffer of 100000 packets if the source is HDV.
 @example
-ffmpeg -f iec61883 -i auto -dvbuffer 100000 out.mpg
+ffmpeg -f iec61883 -i auto -hdvbuffer 100000 out.mpg
 @end example
 
 @end itemize
@@ -929,15 +870,11 @@ If you don't understand what all of that means, you probably don't want this.  L
 DRM device to capture on.  Defaults to @option{/dev/dri/card0}.
 
 @item format
-Pixel format of the framebuffer.  This can be autodetected if you are running Linux 5.7
-or later, but needs to be provided for earlier versions.  Defaults to @option{bgr0},
-which is the most common format used by the Linux console and Xorg X server.
+Pixel format of the framebuffer.  Defaults to @option{bgr0}.
 
 @item format_modifier
 Format modifier to signal on output frames.  This is necessary to import correctly into
-some APIs.  It can be autodetected if you are running Linux 5.7 or later, but will need
-to be provided explicitly when needed in earlier versions.  See the libdrm documentation
-for possible values.
+some APIs, but can't be autodetected.  See the libdrm documentation for possible values.
 
 @item crtc_id
 KMS CRTC ID to define the capture source.  The first active plane on the given CRTC
@@ -973,14 +910,6 @@ Capture from CRTC ID 42 at 60fps, map the result to VAAPI, convert to NV12 and e
 ffmpeg -crtc_id 42 -framerate 60 -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,scale_vaapi=w=1920:h=1080:format=nv12' -c:v h264_vaapi output.mp4
 @end example
 
-@item
-To capture only part of a plane the output can be cropped - this can be used to capture
-a single window, as long as it has a known absolute position and size.  For example, to
-capture and encode the middle quarter of a 1920x1080 plane:
-@example
-ffmpeg -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,crop=960:540:480:270,scale_vaapi=960:540:nv12' -c:v h264_vaapi output.mp4
-@end example
-
 @end itemize
 
 @section lavfi
@@ -1121,21 +1050,71 @@ IIDC1394 input device, based on libdc1394 and libraw1394.
 
 Requires the configure option @code{--enable-libdc1394}.
 
+@section libndi_newtek
+
+The libndi_newtek input device provides capture capabilities for using NDI (Network
+Device Interface, standard created by NewTek).
+
+Input filename is a NDI source name that could be found by sending -find_sources 1
+to command line - it has no specific syntax but human-readable formatted.
+
+To enable this input device, you need the NDI SDK and you
+need to configure with the appropriate @code{--extra-cflags}
+and @code{--extra-ldflags}.
+
 @subsection Options
+
 @table @option
 
-@item framerate
-Set the frame rate. Default is @code{ntsc}, corresponding to a frame
-rate of @code{30000/1001}.
+@item find_sources
+If set to @option{true}, print a list of found/available NDI sources and exit.
+Defaults to @option{false}.
 
-@item pixel_format
-Select the pixel format. Default is @code{uyvy422}.
+@item wait_sources
+Override time to wait until the number of online sources have changed.
+Defaults to @option{0.5}.
+
+@item allow_video_fields
+When this flag is @option{false}, all video that you receive will be progressive.
+Defaults to @option{true}.
+
+@item extra_ips
+If is set to list of comma separated ip addresses, scan for sources not only
+using mDNS but also use unicast ip addresses specified by this list.
 
-@item video_size
-Set the video size given as a string such as @code{640x480} or @code{hd720}.
-Default is @code{qvga}.
 @end table
 
+@subsection Examples
+
+@itemize
+
+@item
+List input devices:
+@example
+ffmpeg -f libndi_newtek -find_sources 1 -i dummy
+@end example
+
+@item
+List local and remote input devices:
+@example
+ffmpeg -f libndi_newtek -extra_ips "192.168.10.10" -find_sources 1 -i dummy
+@end example
+
+@item
+Restream to NDI:
+@example
+ffmpeg -f libndi_newtek -i "DEV-5.INTERNAL.M1STEREO.TV (NDI_SOURCE_NAME_1)" -f libndi_newtek -y NDI_SOURCE_NAME_2
+@end example
+
+@item
+Restream remote NDI to local NDI:
+@example
+ffmpeg -f libndi_newtek -extra_ips "192.168.10.10" -i "DEV-5.REMOTE.M1STEREO.TV (NDI_SOURCE_NAME_1)" -f libndi_newtek -y NDI_SOURCE_NAME_2
+@end example
+
+
+@end itemize
+
 @section openal
 
 The OpenAL input device provides audio capture on all systems with a
@@ -1289,11 +1268,11 @@ Specify the samplerate in Hz, by default 48kHz is used.
 Specify the channels in use, by default 2 (stereo) is set.
 
 @item frame_size
-This option does nothing and is deprecated.
+Specify the number of bytes per frame, by default it is set to 1024.
 
 @item fragment_size
-Specify the size in bytes of the minimal buffering fragment in PulseAudio, it
-will affect the audio latency. By default it is set to 50 ms amount of data.
+Specify the minimal buffering fragment in PulseAudio, it will affect the
+audio latency. By default it is unset.
 
 @item wallclock
 Set the initial PTS using the current time. Default is 1.
@@ -1528,14 +1507,6 @@ ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0+10,20 out.mpg
 @subsection Options
 
 @table @option
-@item select_region
-Specify whether to select the grabbing area graphically using the pointer.
-A value of @code{1} prompts the user to select the grabbing area graphically
-by clicking and dragging. A single click with no dragging will select the
-whole screen. A region with zero width or height will also select the whole
-screen. This option overwrites the @var{video_size}, @var{grab_x}, and
-@var{grab_y} options. Default value is @code{0}.
-
 @item draw_mouse
 Specify whether to draw the mouse pointer. A value of @code{0} specifies
 not to draw the pointer. Default value is @code{1}.
@@ -1584,21 +1555,8 @@ With @var{follow_mouse}:
 ffmpeg -f x11grab -follow_mouse centered -show_region 1 -framerate 25 -video_size cif -i :0.0 out.mpg
 @end example
 
-@item window_id
-Grab this window, instead of the whole screen. Default value is 0, which maps to
-the whole screen (root window).
-
-The id of a window can be found using the @command{xwininfo} program, possibly with options -tree and
--root.
-
-If the window is later enlarged, the new area is not recorded. Video ends when
-the window is closed, unmapped (i.e., iconified) or shrunk beyond the video
-size (which defaults to the initial window size).
-
-This option disables options @option{follow_mouse} and @option{select_region}.
-
 @item video_size
-Set the video frame size. Default is the full desktop or window.
+Set the video frame size. Default value is @code{vga}.
 
 @item grab_x
 @item grab_y

doc/libav-merge.txt

@@ -100,7 +100,6 @@ Stuff that didn't reach the codebase:
   - 4de220d2e frame: allow align=0 (meaning automatic) for av_frame_get_buffer()
 - Support recovery from an already present HLS playlist (see 16cb06bb30)
 - Remove all output devices (see 8e7e042d41, 8d3db95f20, 6ce13070bd, d46cd24986 and https://ffmpeg.org/pipermail/ffmpeg-devel/2017-September/216904.html)
-- avcodec/libaomenc: export the Sequence Header OBU as extradata (See a024c3ce9a)
 
 Collateral damage that needs work locally:
 ------------------------------------------

doc/mailing-list-faq.texi

@@ -64,6 +64,10 @@ Email @email{ffmpeg-devel@@ffmpeg.org} to send a message to the
 ffmpeg-devel mailing list.
 @end itemize
 
+Note that the ffmpeg-devel mailing list does not require you to subscribe
+to send a message or patch, but ffmpeg-user and libav-user do require
+subscription.
+
 @chapter Subscribing / Unsubscribing
 
 @anchor{How do I subscribe?}
@@ -90,9 +94,6 @@ The process is the same for the other mailing lists.
 Please avoid asking a mailing list admin to unsubscribe you unless you
 are absolutely unable to do so by yourself. See @ref{Who do I contact if I have a problem with the mailing list?}
 
-Note that it is possible to temporarily halt message delivery (vacation mode).
-See @ref{How do I disable mail delivery without unsubscribing?}
-
 @chapter Moderation Queue
 @anchor{Why is my message awaiting moderator approval?}
 @section Why is my message awaiting moderator approval?
@@ -115,8 +116,7 @@ or is abusive towards others).
 
 @section How long does it take for my message in the moderation queue to be approved?
 
-The queue is not checked on a regular basis. You can ask on the
-@t{#ffmpeg-devel} IRC channel on Libera Chat for someone to approve your message.
+The queue is usually checked daily to several times a week.
 
 @anchor{How do I delete my message in the moderation queue?}
 @section How do I delete my message in the moderation queue?
@@ -155,14 +155,13 @@ Perform a site search using your favorite search engine. Example:
 
 @section Is there an alternative to the mailing list?
 
-You can ask for help in the official @t{#ffmpeg} IRC channel on Libera Chat.
+You can ask for help in the official @t{#ffmpeg} IRC channel on Freenode.
 
-Some users prefer the third-party @url{http://www.ffmpeg-archive.org/, Nabble}
-interface which presents the mailing lists in a typical forum layout.
+Some users prefer the third-party Nabble interface which presents the
+mailing lists in a typical forum layout.
 
-There are also numerous third-party help sites such as
-@url{https://superuser.com/tags/ffmpeg, Super User} and
-@url{https://www.reddit.com/r/ffmpeg/, r/ffmpeg on reddit}.
+There are also numerous third-party help sites such as Super User and
+r/ffmpeg on reddit.
 
 @anchor{What is top-posting?}
 @section What is top-posting?
@@ -182,7 +181,7 @@ instead of attaching them.
 Anywhere that is not too annoying for us to use.
 
 Google Drive and Dropbox are acceptable if you need a file host, and
-@url{https://0x0.st/, 0x0.st} is good for files under 256 MiB.
+0x0.st is good for files under 256 MiB.
 
 Small, short samples are preferred if possible.
 
@@ -229,54 +228,6 @@ or headers.
 
 You can then filter the mailing list messages to their own folder.
 
-@anchor{How do I disable mail delivery without unsubscribing?}
-@section How do I disable mail delivery without unsubscribing?
-
-Sometimes you may want to temporarily stop receiving all mailing list
-messages. This "vacation mode" is simple to do:
-
-@enumerate
-@item
-Go to the @url{https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-user/, ffmpeg-user mailing list info page}
-
-@item
-Enter your email address in the box at very bottom of the page and click the
-@emph{Unsubscribe or edit options} box.
-
-@item
-Enter your password and click the @emph{Log in} button.
-
-@item
-Look for the @emph{Mail delivery} option. Here you can disable/enable mail
-delivery. If you check @emph{Set globally} it will apply your choice to all
-other FFmpeg mailing lists you are subscribed to.
-@end enumerate
-
-Alternatively, from your subscribed address, send a message to @email{ffmpeg-user-request@@ffmpeg.org}
-with the subject @emph{set delivery off}. To re-enable mail delivery send a
-message to @email{ffmpeg-user-request@@ffmpeg.org} with the subject
-@emph{set delivery on}.
-
-@anchor{Why is the mailing list munging my address?}
-@section Why is the mailing list munging my address?
-
-This is due to subscribers that use an email service with a DMARC reject policy
-which adds difficulties to mailing list operators.
-
-The mailing list must re-write (munge) the @emph{From:} header for such users;
-otherwise their email service will reject and bounce the message resulting in
-automatic unsubscribing from the mailing list.
-
-When sending a message these users will see @emph{via <mailing list name>}
-added to their name and the @emph{From:} address munged to the address of
-the particular mailing list.
-
-If you want to avoid this then please use a different email service.
-
-Note that ffmpeg-devel does not apply any munging as it causes issues with
-patch authorship. As a result users with an email service with a DMARC reject
-policy may be automatically unsubscribed due to rejected and bounced messages.
-
 @chapter Rules and Etiquette
 
 @section What are the rules and the proper etiquette?
@@ -375,15 +326,6 @@ form a multi-part message is recommended by email standards.
 Check your spam folder.
 @end itemize
 
-@anchor{Why do I keep getting unsubscribed from ffmpeg-devel?}
-@section Why do I keep getting unsubscribed from ffmpeg-devel?
-
-Users with an email service that has a DMARC reject or quarantine policy may be
-automatically unsubscribed from the ffmpeg-devel mailing list due to the mailing
-list messages being continuously rejected and bounced back.
-
-Consider using a different email service.
-
 @anchor{Who do I contact if I have a problem with the mailing list?}
 @section Who do I contact if I have a problem with the mailing list?
 

doc/metadata.texi

@@ -33,7 +33,7 @@ At the beginning of a chapter section there may be an optional timebase to be
 used for start/end values. It must be in form
 @samp{TIMEBASE=@var{num}/@var{den}}, where @var{num} and @var{den} are
 integers. If the timebase is missing then start/end times are assumed to
-be in nanoseconds.
+be in milliseconds.
 
 Next a chapter section must contain chapter start and end times in form
 @samp{START=@var{num}}, @samp{END=@var{num}}, where @var{num} is a positive

doc/multithreading.txt

@@ -20,7 +20,8 @@ Slice threading -
 
 Frame threading -
 * Restrictions with slice threading also apply.
-* Custom get_buffer2() and get_format() callbacks must be thread-safe.
+* For best performance, the client should set thread_safe_callbacks if it
+  provides a thread-safe get_buffer() callback.
 * There is one frame of delay added for every thread beyond the first one.
   Clients must be able to handle this; the pkt_dts and pkt_pts fields in
   AVFrame will work as usual.
@@ -50,14 +51,16 @@ the decode process starts. Call ff_thread_finish_setup() afterwards. If
 some code can't be moved, have update_thread_context() run it in the next
 thread.
 
+If the codec allocates writable tables in its init(), add an init_thread_copy()
+which re-allocates them for other threads.
+
 Add AV_CODEC_CAP_FRAME_THREADS to the codec capabilities. There will be very little
 speed gain at this point but it should work.
 
 If there are inter-frame dependencies, so the codec calls
-ff_thread_report/await_progress(), set FF_CODEC_CAP_ALLOCATE_PROGRESS in
-AVCodec.caps_internal and use ff_thread_get_buffer() to allocate frames. The
+ff_thread_report/await_progress(), set AVCodecInternal.allocate_progress. The
 frames must then be freed with ff_thread_release_buffer().
-Otherwise decode directly into the user-supplied frames.
+Otherwise leave it at zero and decode directly into the user-supplied frames.
 
 Call ff_thread_report_progress() after some part of the current picture has decoded.
 A good place to put this is where draw_horiz_band() is called - add this if it isn't

doc/outdevs.texi

@@ -38,52 +38,6 @@ ffmpeg -i INPUT -f alsa hw:1,7
 @end example
 @end itemize
 
-@section AudioToolbox
-
-AudioToolbox output device.
-
-Allows native output to CoreAudio devices on OSX.
-
-The output filename can be empty (or @code{-}) to refer to the default system output device or a number that refers to the device index as shown using: @code{-list_devices true}.
-
-Alternatively, the audio input device can be chosen by index using the
-@option{
-    -audio_device_index <INDEX>
-}
-, overriding any device name or index given in the input filename.
-
-All available devices can be enumerated by using @option{-list_devices true}, listing
-all device names, UIDs and corresponding indices.
-
-@subsection Options
-
-AudioToolbox supports the following options:
-
-@table @option
-
-@item -audio_device_index <INDEX>
-Specify the audio device by its index. Overrides anything given in the output filename.
-
-@end table
-
-@subsection Examples
-
-@itemize
-
-@item
-Print the list of supported devices and output a sine wave to the default device:
-@example
-$ ffmpeg -f lavfi -i sine=r=44100 -f audiotoolbox -list_devices true -
-@end example
-
-@item
-Output a sine wave to the device with the index 2, overriding any output filename:
-@example
-$ ffmpeg -f lavfi -i sine=r=44100 -f audiotoolbox -audio_device_index 2 -
-@end example
-
-@end itemize
-
 @section caca
 
 CACA output device.
@@ -186,8 +140,8 @@ device with @command{-list_formats 1}. Audio sample rate is always 48 kHz.
 
 @item list_devices
 If set to @option{true}, print a list of devices and exit.
-Defaults to @option{false}. This option is deprecated, please use the
-@code{-sinks} option of ffmpeg to list the available output devices.
+Defaults to @option{false}. Alternatively you can use the @code{-sinks}
+option of ffmpeg to list the available output devices.
 
 @item list_formats
 If set to @option{true}, print a list of supported formats and exit.
@@ -198,43 +152,9 @@ Amount of time to preroll video in seconds.
 Defaults to @option{0.5}.
 
 @item duplex_mode
-Sets the decklink device duplex/profile mode. Must be @samp{unset}, @samp{half}, @samp{full},
-@samp{one_sub_device_full}, @samp{one_sub_device_half}, @samp{two_sub_device_full},
-@samp{four_sub_device_half}
+Sets the decklink device duplex mode. Must be @samp{unset}, @samp{half} or @samp{full}.
 Defaults to @samp{unset}.
 
-Note: DeckLink SDK 11.0 have replaced the duplex property by a profile property.
-For the DeckLink Duo 2 and DeckLink Quad 2, a profile is shared between any 2
-sub-devices that utilize the same connectors. For the DeckLink 8K Pro, a profile
-is shared between all 4 sub-devices. So DeckLink 8K Pro support four profiles.
-
-Valid profile modes for DeckLink 8K Pro(with DeckLink SDK >= 11.0):
-@samp{one_sub_device_full}, @samp{one_sub_device_half}, @samp{two_sub_device_full},
-@samp{four_sub_device_half}
-
-Valid profile modes for DeckLink Quad 2 and DeckLink Duo 2:
-@samp{half}, @samp{full}
-
-@item timing_offset
-Sets the genlock timing pixel offset on the used output.
-Defaults to @samp{unset}.
-
-@item link
-Sets the SDI video link configuration on the used output. Must be
-@samp{unset}, @samp{single} link SDI, @samp{dual} link SDI or @samp{quad} link
-SDI.
-Defaults to @samp{unset}.
-
-@item sqd
-Enable Square Division Quad Split mode for Quad-link SDI output.
-Must be @samp{unset}, @samp{true} or @samp{false}.
-Defaults to @option{unset}.
-
-@item level_a
-Enable SMPTE Level A mode on the used output.
-Must be @samp{unset}, @samp{true} or @samp{false}.
-Defaults to @option{unset}.
-
 @end table
 
 @subsection Examples
@@ -244,7 +164,7 @@ Defaults to @option{unset}.
 @item
 List output devices:
 @example
-ffmpeg -sinks decklink
+ffmpeg -i test.avi -f decklink -list_devices 1 dummy
 @end example
 
 @item
@@ -296,6 +216,51 @@ ffmpeg -re -i INPUT -c:v rawvideo -pix_fmt bgra -f fbdev /dev/fb0
 
 See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1).
 
+@section libndi_newtek
+
+The libndi_newtek output device provides playback capabilities for using NDI (Network
+Device Interface, standard created by NewTek).
+
+Output filename is a NDI name.
+
+To enable this output device, you need the NDI SDK and you
+need to configure with the appropriate @code{--extra-cflags}
+and @code{--extra-ldflags}.
+
+NDI uses uyvy422 pixel format natively, but also supports bgra, bgr0, rgba and
+rgb0.
+
+@subsection Options
+
+@table @option
+
+@item reference_level
+The audio reference level in dB. This specifies how many dB above the
+reference level (+4dBU) is the full range of 16 bit audio.
+Defaults to @option{0}.
+
+@item clock_video
+These specify whether video "clock" themselves.
+Defaults to @option{false}.
+
+@item clock_audio
+These specify whether audio "clock" themselves.
+Defaults to @option{false}.
+
+@end table
+
+@subsection Examples
+
+@itemize
+
+@item
+Play video clip:
+@example
+ffmpeg -i "udp://@@239.1.1.1:10480?fifo_size=1000000&overrun_nonfatal=1" -vf "scale=720:576,fps=fps=25,setdar=dar=16/9,format=pix_fmts=uyvy422" -f libndi_newtek NEW_NDI1
+@end example
+
+@end itemize
+
 @section opengl
 OpenGL output device.
 
@@ -405,8 +370,6 @@ ffmpeg  -i INPUT -f pulse "stream name"
 
 SDL (Simple DirectMedia Layer) output device.
 
-"sdl2" can be used as alias for "sdl".
-
 This output device allows one to show a video stream in an SDL
 window. Only one SDL window is allowed per application, so you can
 have only one instance of this output device in an application.

doc/protocols.texi

@@ -51,82 +51,6 @@ in microseconds.
 
 A description of the currently available protocols follows.
 
-@section amqp
-
-Advanced Message Queueing Protocol (AMQP) version 0-9-1 is a broker based
-publish-subscribe communication protocol.
-
-FFmpeg must be compiled with --enable-librabbitmq to support AMQP. A separate
-AMQP broker must also be run. An example open-source AMQP broker is RabbitMQ.
-
-After starting the broker, an FFmpeg client may stream data to the broker using
-the command:
-
-@example
-ffmpeg -re -i input -f mpegts amqp://[[user]:[password]@@]hostname[:port][/vhost]
-@end example
-
-Where hostname and port (default is 5672) is the address of the broker. The
-client may also set a user/password for authentication. The default for both
-fields is "guest". Name of virtual host on broker can be set with vhost. The
-default value is "/".
-
-Muliple subscribers may stream from the broker using the command:
-@example
-ffplay amqp://[[user]:[password]@@]hostname[:port][/vhost]
-@end example
-
-In RabbitMQ all data published to the broker flows through a specific exchange,
-and each subscribing client has an assigned queue/buffer. When a packet arrives
-at an exchange, it may be copied to a client's queue depending on the exchange
-and routing_key fields.
-
-The following options are supported:
-
-@table @option
-
-@item exchange
-Sets the exchange to use on the broker. RabbitMQ has several predefined
-exchanges: "amq.direct" is the default exchange, where the publisher and
-subscriber must have a matching routing_key; "amq.fanout" is the same as a
-broadcast operation (i.e. the data is forwarded to all queues on the fanout
-exchange independent of the routing_key); and "amq.topic" is similar to
-"amq.direct", but allows for more complex pattern matching (refer to the RabbitMQ
-documentation).
-
-@item routing_key
-Sets the routing key. The default value is "amqp". The routing key is used on
-the "amq.direct" and "amq.topic" exchanges to decide whether packets are written
-to the queue of a subscriber.
-
-@item pkt_size
-Maximum size of each packet sent/received to the broker. Default is 131072.
-Minimum is 4096 and max is any large value (representable by an int). When
-receiving packets, this sets an internal buffer size in FFmpeg. It should be
-equal to or greater than the size of the published packets to the broker. Otherwise
-the received message may be truncated causing decoding errors.
-
-@item connection_timeout
-The timeout in seconds during the initial connection to the broker. The
-default value is rw_timeout, or 5 seconds if rw_timeout is not set.
-
-@item delivery_mode @var{mode}
-Sets the delivery mode of each message sent to broker.
-The following values are accepted:
-@table @samp
-@item persistent
-Delivery mode set to "persistent" (2). This is the default value.
-Messages may be written to the broker's disk depending on its setup.
-
-@item non-persistent
-Delivery mode set to "non-persistent" (1).
-Messages will stay in broker's memory unless the broker is under memory
-pressure.
-
-@end table
-
-@end table
-
 @section async
 
 Asynchronous data filling wrapper for input stream.
@@ -175,16 +99,6 @@ Caching wrapper for input stream.
 
 Cache the input stream to temporary file. It brings seeking capability to live streams.
 
-The accepted options are:
-@table @option
-
-@item read_ahead_limit
-Amount in bytes that may be read ahead when seeking isn't supported. Range is -1 to INT_MAX.
--1 for unlimited. Default is 65536.
-
-@end table
-
-URL Syntax is
 @example
 cache:@var{URL}
 @end example
@@ -215,38 +129,6 @@ ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
 Note that you may need to escape the character "|" which is special for
 many shells.
 
-@section concatf
-
-Physical concatenation protocol using a line break delimited list of
-resources.
-
-Read and seek from many resources in sequence as if they were
-a unique resource.
-
-A URL accepted by this protocol has the syntax:
-@example
-concatf:@var{URL}
-@end example
-
-where @var{URL} is the url containing a line break delimited list of
-resources to be concatenated, each one possibly specifying a distinct
-protocol. Special characters must be escaped with backslash or single
-quotes. See @ref{quoting_and_escaping,,the "Quoting and escaping"
-section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
-
-For example to read a sequence of files @file{split1.mpeg},
-@file{split2.mpeg}, @file{split3.mpeg} listed in separate lines within
-a file @file{split.txt} with @command{ffplay} use the command:
-@example
-ffplay concatf:split.txt
-@end example
-Where @file{split.txt} contains the lines:
-@example
-split1.mpeg
-split2.mpeg
-split3.mpeg
-@end example
-
 @section crypto
 
 AES-encrypted stream reading protocol.
@@ -311,20 +193,6 @@ Set I/O operation maximum block size, in bytes. Default value is
 @code{INT_MAX}, which results in not limiting the requested block size.
 Setting this value reasonably low improves user termination request reaction
 time, which is valuable for files on slow medium.
-
-@item follow
-If set to 1, the protocol will retry reading at the end of the file, allowing
-reading files that still are being written. In order for this to terminate,
-you either need to use the rw_timeout option, or use the interrupt callback
-(for API users).
-
-@item seekable
-Controls if seekability is advertised on the file. 0 means non-seekable, -1
-means auto (seekable for normal files, non-seekable for named pipes).
-
-Many demuxers handle seekable and non-seekable resources differently,
-overriding this might speed up opening certain files at the cost of losing some
-features (e.g. accurate seeking).
 @end table
 
 @section ftp
@@ -346,14 +214,6 @@ Set timeout in microseconds of socket I/O operations used by the underlying low
 operation. By default it is set to -1, which means that the timeout is
 not specified.
 
-@item ftp-user
-Set a user to be used for authenticating to the FTP server. This is overridden by the
-user in the FTP URL.
-
-@item ftp-password
-Set a password to be used for authenticating to the FTP server. This is overridden by
-the password in the FTP URL, or by @option{ftp-anonymous-password} if no user is set.
-
 @item ftp-anonymous-password
 Password used when login as anonymous user. Typically an e-mail address
 should be used.
@@ -369,16 +229,21 @@ it, unless special care is taken (tests, customized server configuration
 etc.). Different FTP servers behave in different way during seek
 operation. ff* tools may produce incomplete content due to server limitations.
 
+This protocol accepts the following options:
+
+@table @option
+@item follow
+If set to 1, the protocol will retry reading at the end of the file, allowing
+reading files that still are being written. In order for this to terminate,
+you either need to use the rw_timeout option, or use the interrupt callback
+(for API users).
+
+@end table
+
 @section gopher
 
 Gopher protocol.
 
-@section gophers
-
-Gophers protocol.
-
-The Gopher protocol with TLS encapsulation.
-
 @section hls
 
 Read Apple HTTP Live Streaming compliant segmented stream as
@@ -438,6 +303,14 @@ Set the Referer header. Include 'Referer: URL' header in HTTP request.
 Override the User-Agent header. If not specified the protocol will use a
 string describing the libavformat build. ("Lavf/<version>")
 
+@item user-agent
+This is a deprecated option, you can use user_agent instead it.
+
+@item timeout
+Set timeout in microseconds of socket I/O operations used by the underlying low level
+operation. By default it is set to -1, which means that the timeout is
+not specified.
+
 @item reconnect_at_eof
 If set then eof is treated like an error and causes reconnection, this is useful
 for live / endless streams.
@@ -445,13 +318,6 @@ for live / endless streams.
 @item reconnect_streamed
 If set then even streamed/non seekable streams will be reconnected on errors.
 
-@item reconnect_on_network_error
-Reconnect automatically in case of TCP/TLS errors during connect.
-
-@item reconnect_on_http_error
-A comma separated list of HTTP status codes to reconnect on. The list can
-include specific status codes (e.g. '503') or the strings '4xx' / '5xx'.
-
 @item reconnect_delay_max
 Sets the maximum delay in seconds after which to give up reconnecting
 
@@ -524,33 +390,6 @@ ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://@var{server}:@var{p
 wget --post-file=somefile.ogg http://@var{server}:@var{port}
 @end example
 
-@item send_expect_100
-Send an Expect: 100-continue header for POST. If set to 1 it will send, if set
-to 0 it won't, if set to -1 it will try to send if it is applicable. Default
-value is -1.
-
-@item auth_type
-
-Set HTTP authentication type. No option for Digest, since this method requires
-getting nonce parameters from the server first and can't be used straight away like
-Basic.
-
-@table @option
-@item none
-Choose the HTTP authentication type automatically. This is the default.
-@item basic
-
-Choose the HTTP basic authentication.
-
-Basic authentication sends a Base64-encoded string that contains a user name and password
-for the client. Base64 is not a form of encryption and should be considered the same as
-sending the user name and password in clear text (Base64 is a reversible encoding).
-If a resource needs to be protected, strongly consider using an authentication scheme
-other than basic authentication. HTTPS/TLS should be used with basic authentication.
-Without these additional security enhancements, basic authentication should not be used
-to protect sensitive or valuable information.
-@end table
-
 @end table
 
 @subsection HTTP Cookies
@@ -605,47 +444,12 @@ audio/mpeg.
 This enables support for Icecast versions < 2.4.0, that do not support the
 HTTP PUT method but the SOURCE method.
 
-@item tls
-Establish a TLS (HTTPS) connection to Icecast.
-
 @end table
 
 @example
 icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint}
 @end example
 
-@section ipfs
-
-InterPlanetary File System (IPFS) protocol support. One can access files stored
-on the IPFS network through so-called gateways. These are http(s) endpoints.
-This protocol wraps the IPFS native protocols (ipfs:// and ipns://) to be sent
-to such a gateway. Users can (and should) host their own node which means this
-protocol will use one's local gateway to access files on the IPFS network.
-
-If a user doesn't have a node of their own then the public gateway @code{https://dweb.link}
-is used by default.
-
-This protocol accepts the following options:
-
-@table @option
-
-@item gateway
-Defines the gateway to use. When not set, the protocol will first try
-locating the local gateway by looking at @code{$IPFS_GATEWAY}, @code{$IPFS_PATH}
-and @code{$HOME/.ipfs/}, in that order. If that fails @code{https://dweb.link} will be used.
-
-@end table
-
-One can use this protocol in 2 ways. Using IPFS:
-@example
-ffplay ipfs://QmbGtJg23skhvFmu9mJiePVByhfzu5rwo74MEkVDYAmF5T
-@end example
-
-Or the IPNS protocol (IPNS is mutable IPFS):
-@example
-ffplay ipns://QmbGtJg23skhvFmu9mJiePVByhfzu5rwo74MEkVDYAmF5T
-@end example
-
 @section mmst
 
 MMS (Microsoft Media Server) protocol over TCP.
@@ -757,50 +561,6 @@ Example usage:
 -f rtp_mpegts -fec prompeg=l=8:d=4 rtp://@var{hostname}:@var{port}
 @end example
 
-@section rist
-
-Reliable Internet Streaming Transport protocol
-
-The accepted options are:
-@table @option
-@item rist_profile
-Supported values:
-@table @samp
-@item simple
-@item main
-This one is default.
-@item advanced
-@end table
-
-@item buffer_size
-Set internal RIST buffer size in milliseconds for retransmission of data.
-Default value is 0 which means the librist default (1 sec). Maximum value is 30
-seconds.
-
-@item fifo_size
-Size of the librist receiver output fifo in number of packets. This must be a
-power of 2.
-Defaults to 8192 (vs the librist default of 1024).
-
-@item overrun_nonfatal=@var{1|0}
-Survive in case of librist fifo buffer overrun. Default value is 0.
-
-@item pkt_size
-Set maximum packet size for sending data. 1316 by default.
-
-@item log_level
-Set loglevel for RIST logging messages. You only need to set this if you
-explicitly want to enable debug level messages or packet loss simulation,
-otherwise the regular loglevel is respected.
-
-@item secret
-Set override of encryption secret, by default is unset.
-
-@item encryption
-Set encryption type, by default is disabled.
-Acceptable values are 128 and 256.
-@end table
-
 @section rtmp
 
 Real-Time Messaging Protocol.
@@ -915,11 +675,6 @@ URL to player swf file, compute hash/size automatically.
 @item rtmp_tcurl
 URL of the target stream. Defaults to proto://host[:port]/app.
 
-@item tcp_nodelay=@var{1|0}
-Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
-
-@emph{Remark: Writing to the socket is currently not optimized to minimize system calls and reduces the efficiency / effect of TCP_NODELAY.}
-
 @end table
 
 For example to read with @command{ffplay} a multimedia resource named
@@ -1107,9 +862,6 @@ Set the local RTCP port to @var{n}.
 @item pkt_size=@var{n}
 Set max packet size (in bytes) to @var{n}.
 
-@item buffer_size=@var{size}
-Set the maximum UDP socket buffer size in bytes.
-
 @item connect=0|1
 Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
 to 0).
@@ -1127,13 +879,6 @@ set to 1) or to a default remote address (if set to 0).
 @item localport=@var{n}
 Set the local RTP port to @var{n}.
 
-@item localaddr=@var{addr}
-Local IP address of a network interface used for sending packets or joining
-multicast groups.
-
-@item timeout=@var{n}
-Set timeout (in microseconds) of socket I/O operations to @var{n}.
-
 This is a deprecated option. Instead, @option{localrtpport} should be
 used.
 
@@ -1242,18 +987,19 @@ Set minimum local UDP port. Default value is 5000.
 @item max_port
 Set maximum local UDP port. Default value is 65000.
 
-@item listen_timeout
-Set maximum timeout (in seconds) to establish an initial connection. Setting
-@option{listen_timeout} > 0 sets @option{rtsp_flags} to @samp{listen}. Default is -1
-which means an infinite timeout when @samp{listen} mode is set.
+@item timeout
+Set maximum timeout (in seconds) to wait for incoming connections.
+
+A value of -1 means infinite (default). This option implies the
+@option{rtsp_flags} set to @samp{listen}.
 
 @item reorder_queue_size
 Set number of packets to buffer for handling of reordered packets.
 
-@item timeout
+@item stimeout
 Set socket TCP I/O timeout in microseconds.
 
-@item user_agent
+@item user-agent
 Override User-Agent header. If not specified, it defaults to the
 libavformat identifier string.
 @end table
@@ -1433,7 +1179,7 @@ options.
 This protocol accepts the following options.
 
 @table @option
-@item connect_timeout=@var{milliseconds}
+@item connect_timeout
 Connection timeout; SRT cannot connect for RTT > 1500 msec
 (2 handshake exchanges) with the default connect timeout of
 3 seconds. This option applies to the caller and rendezvous
@@ -1464,7 +1210,7 @@ IP Type of Service. Applies to sender only. Default value is 0xB8.
 @item ipttl=@var{ttl}
 IP Time To Live. Applies to sender only. Default value is 64.
 
-@item latency=@var{microseconds}
+@item latency
 Timestamp-based Packet Delivery Delay.
 Used to absorb bursts of missed packet retransmissions.
 This flag sets both @option{rcvlatency} and @option{peerlatency}
@@ -1475,7 +1221,7 @@ when side is sender and @option{rcvlatency}
 when side is receiver, and the bidirectional stream
 sending is not supported.
 
-@item listen_timeout=@var{microseconds}
+@item listen_timeout
 Set socket listen timeout.
 
 @item maxbw=@var{bytes/seconds}
@@ -1520,32 +1266,6 @@ only if @option{pbkeylen} is non-zero. It is used on
 the receiver only if the received data is encrypted.
 The configured passphrase cannot be recovered (write-only).
 
-@item enforced_encryption=@var{1|0}
-If true, both connection parties must have the same password
-set (including empty, that is, with no encryption). If the
-password doesn't match or only one side is unencrypted,
-the connection is rejected. Default is true.
-
-@item kmrefreshrate=@var{packets}
-The number of packets to be transmitted after which the
-encryption key is switched to a new key. Default is -1.
--1 means auto (0x1000000 in srt library). The range for
-this option is integers in the 0 - @code{INT_MAX}.
-
-@item kmpreannounce=@var{packets}
-The interval between when a new encryption key is sent and
-when switchover occurs. This value also applies to the
-subsequent interval between when switchover occurs and
-when the old encryption key is decommissioned. Default is -1.
--1 means auto (0x1000 in srt library). The range for
-this option is integers in the 0 - @code{INT_MAX}.
-
-@item snddropdelay=@var{microseconds}
-The sender's extra delay before dropping packets. This delay is
-added to the default drop delay time interval value.
-
-Special value -1: Do not drop packets on the sender at all.
-
 @item payload_size=@var{bytes}
 Sets the maximum declared size of a packet transferred
 during the single call to the sending function in Live
@@ -1561,7 +1281,7 @@ use a bigger maximum frame size, though not greater than
 @item pkt_size=@var{bytes}
 Alias for @samp{payload_size}.
 
-@item peerlatency=@var{microseconds}
+@item peerlatency
 The latency value (as described in @option{rcvlatency}) that is
 set by the sender side as a minimum value for the receiver.
 
@@ -1573,7 +1293,7 @@ Not required on receiver (set to 0),
 key size obtained from sender in HaiCrypt handshake.
 Default value is 0.
 
-@item rcvlatency=@var{microseconds}
+@item rcvlatency
 The time that should elapse since the moment when the
 packet was sent and the moment when it's delivered to
 the receiver application in the receiving function.
@@ -1591,10 +1311,12 @@ Set UDP receive buffer size, expressed in bytes.
 @item send_buffer_size=@var{bytes}
 Set UDP send buffer size, expressed in bytes.
 
-@item timeout=@var{microseconds}
-Set raise error timeouts for read, write and connect operations. Note that the
-SRT library has internal timeouts which can be controlled separately, the
-value set here is only a cap on those.
+@item rw_timeout
+Set raise error timeout for read/write optations.
+
+This option is only relevant in read mode:
+if no data arrived in more than this time
+interval, raise error.
 
 @item tlpktdrop=@var{1|0}
 Too-late Packet Drop. When enabled on receiver, it skips
@@ -1645,9 +1367,6 @@ This option doesn’t make sense in Rendezvous connection; the result
 might be that simply one side will override the value from the other
 side and it’s the matter of luck which one would win
 
-@item srt_streamid=@var{string}
-Alias for @samp{streamid} to avoid conflict with ffmpeg command line option.
-
 @item smoother=@var{live|file}
 The type of Smoother used for the transmission for that socket, which
 is responsible for the transmission and congestion control. The Smoother
@@ -1691,17 +1410,6 @@ the overhead transmission (retransmitted and control packets).
 file: Set options as for non-live transmission. See @option{messageapi}
 for further explanations
 
-@item linger=@var{seconds}
-The number of seconds that the socket waits for unsent data when closing.
-Default is -1. -1 means auto (off with 0 seconds in live mode, on with 180
-seconds in file mode). The range for this option is integers in the
-0 - @code{INT_MAX}.
-
-@item tsbpd=@var{1|0}
-When true, use Timestamp-based Packet Delivery mode. The default behavior
-depends on the transmission type: enabled in live mode, disabled in file
-mode.
-
 @end table
 
 For more information see: @url{https://github.com/Haivision/srt}.
@@ -1788,9 +1496,8 @@ tcp://@var{hostname}:@var{port}[?@var{options}]
 The list of supported options follows.
 
 @table @option
-@item listen=@var{2|1|0}
-Listen for an incoming connection. 0 disables listen, 1 enables listen in
-single client mode, 2 enables listen in multi-client mode. Default value is 0.
+@item listen=@var{1|0}
+Listen for an incoming connection. Default value is 0.
 
 @item timeout=@var{microseconds}
 Set raise error timeout, expressed in microseconds.
@@ -1810,8 +1517,6 @@ Set send buffer size, expressed bytes.
 @item tcp_nodelay=@var{1|0}
 Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
 
-@emph{Remark: Writing to the socket is currently not optimized to minimize system calls and reduces the efficiency / effect of TCP_NODELAY.}
-
 @item tcp_mss=@var{bytes}
 Set maximum segment size for outgoing TCP packets, expressed in bytes.
 @end table
@@ -1868,10 +1573,6 @@ A file containing the private key for the certificate.
 If enabled, listen for connections on the provided port, and assume
 the server role in the handshake instead of the client role.
 
-@item http_proxy
-The HTTP proxy to tunnel through, e.g. @code{http://example.com:1234}.
-The proxy must support the CONNECT method.
-
 @end table
 
 Example command lines:
@@ -1910,7 +1611,7 @@ The list of supported options follows.
 @item buffer_size=@var{size}
 Set the UDP maximum socket buffer size in bytes. This is used to set either
 the receive or send buffer size, depending on what the socket is used for.
-Default is 32 KB for output, 384 KB for input.  See also @var{fifo_size}.
+Default is 64KB.  See also @var{fifo_size}.
 
 @item bitrate=@var{bitrate}
 If set to nonzero, the output will have the specified constant bitrate if the
@@ -2019,50 +1720,4 @@ Timeout in ms.
 Create the Unix socket in listening mode.
 @end table
 
-@section zmq
-
-ZeroMQ asynchronous messaging using the libzmq library.
-
-This library supports unicast streaming to multiple clients without relying on
-an external server.
-
-The required syntax for streaming or connecting to a stream is:
-@example
-zmq:tcp://ip-address:port
-@end example
-
-Example:
-Create a localhost stream on port 5555:
-@example
-ffmpeg -re -i input -f mpegts zmq:tcp://127.0.0.1:5555
-@end example
-
-Multiple clients may connect to the stream using:
-@example
-ffplay zmq:tcp://127.0.0.1:5555
-@end example
-
-Streaming to multiple clients is implemented using a ZeroMQ Pub-Sub pattern.
-The server side binds to a port and publishes data. Clients connect to the
-server (via IP address/port) and subscribe to the stream. The order in which
-the server and client start generally does not matter.
-
-ffmpeg must be compiled with the --enable-libzmq option to support
-this protocol.
-
-Options can be set on the @command{ffmpeg}/@command{ffplay} command
-line. The following options are supported:
-
-@table @option
-
-@item pkt_size
-Forces the maximum packet size for sending/receiving data. The default value is
-131,072 bytes. On the server side, this sets the maximum size of sent packets
-via ZeroMQ. On the clients, it sets an internal buffer size for receiving
-packets. Note that pkt_size on the clients should be equal to or greater than
-pkt_size on the server. Otherwise the received message may be truncated causing
-decoding errors.
-
-@end table
-
 @c man end PROTOCOLS

doc/scaler.texi

@@ -5,8 +5,7 @@
 The video scaler supports the following named options.
 
 Options may be set by specifying -@var{option} @var{value} in the
-FFmpeg tools, with a few API-only exceptions noted below.
-For programmatic use, they can be set explicitly in the
+FFmpeg tools. For programmatic use, they can be set explicitly in the
 @code{SwsContext} options or through the @file{libavutil/opt.h} API.
 
 @table @option
@@ -48,8 +47,7 @@ Select Gaussian rescaling algorithm.
 Select sinc rescaling algorithm.
 
 @item lanczos
-Select Lanczos rescaling algorithm. The default width (alpha) is 3 and can be
-changed by setting @code{param0}.
+Select Lanczos rescaling algorithm.
 
 @item spline
 Select natural bicubic spline rescaling algorithm.
@@ -70,31 +68,29 @@ Select full chroma input.
 Enable bitexact output.
 @end table
 
-@item srcw @var{(API only)}
+@item srcw
 Set source width.
 
-@item srch @var{(API only)}
+@item srch
 Set source height.
 
-@item dstw @var{(API only)}
+@item dstw
 Set destination width.
 
-@item dsth @var{(API only)}
+@item dsth
 Set destination height.
 
-@item src_format @var{(API only)}
+@item src_format
 Set source pixel format (must be expressed as an integer).
 
-@item dst_format @var{(API only)}
+@item dst_format
 Set destination pixel format (must be expressed as an integer).
 
-@item src_range @var{(boolean)}
-If value is set to @code{1}, indicates source is full range. Default value is
-@code{0}, which indicates source is limited range.
+@item src_range
+Select source range.
 
-@item dst_range @var{(boolean)}
-If value is set to @code{1}, enable full range for destination. Default value
-is @code{0}, which enables limited range.
+@item dst_range
+Select destination range.
 
 @anchor{sws_params}
 @item param0, param1

doc/snow.txt

@@ -172,7 +172,7 @@ spatial_decomposition_count
     FIXME
 
 colorspace_type
-    0   unspecified YCbCr
+    0   unspecified YcbCr
     1   Gray
     2   Gray + Alpha
     3   GBR
@@ -235,7 +235,7 @@ spatial_decomposition_type
     stored as delta from last, last is reset to 0 if always_reset || keyframe
 
 qlog
-    quality (logarithmic quantizer scale)
+    quality (logarthmic quantizer scale)
     stored as delta from last, last is reset to 0 if always_reset || keyframe
 
 mv_scale
@@ -251,11 +251,11 @@ block_max_depth
     stored as delta from last, last is reset to 0 if always_reset || keyframe
 
 quant_table
-    quantization table
+    quantiztation table
 
 
 Highlevel bitstream structure:
-==============================
+=============================
  --------------------------------------------
 |                   Header                   |
  --------------------------------------------
@@ -303,7 +303,7 @@ Decoding process:
                   |  Intra DC  |               |
                   |            |    LL0 subband prediction
                    ------------                |
-                                \        Dequantization
+                                \        Dequantizaton
  -------------------             \             |
 |  Reference frames |             \           IDWT
 | -------   ------- |    Motion    \           |
@@ -390,8 +390,8 @@ motion vector prediction
    (mvx_diff, mvy_diff)*mv_scale
 
 
-Intra DC Prediction:
-====================
+Intra DC Predicton:
+======================
 the luma and chroma values of the left block are used as predictors
 
 the used luma and chroma is the sum of the predictor and y_diff, cb_diff, cr_diff
@@ -407,7 +407,7 @@ Motion Compensation:
 
 Halfpel interpolation:
 ----------------------
-Halfpel interpolation is done by convolution with the halfpel filter stored
+halfpel interpolation is done by convolution with the halfpel filter stored
 in the header:
 
 horizontal halfpel samples are found by
@@ -463,8 +463,8 @@ to the closest available fullpel sample
 Smaller pel interpolation:
 --------------------------
 if diag_mc is set then points which lie on a line between 2 vertically,
-horizontally or diagonally adjacent halfpel points shall be interpolated
-linearly with rounding to nearest and halfway values rounded up.
+horiziontally or diagonally adjacent halfpel points shall be interpolated
+linearls with rounding to nearest and halfway values rounded up.
 points which lie on 2 diagonals at the same time should only use the one
 diagonal not containing the fullpel point
 
@@ -519,8 +519,8 @@ width,height here are the width and height of the LL0 subband not of the final
 video
 
 
-Dequantization:
-===============
+Dequantizaton:
+==============
 FIXME
 
 Wavelet Transform:

doc/t2h.pm

@@ -126,16 +126,8 @@ foreach my $command (keys(%Texinfo::Common::sectioning_commands), 'node') {
     texinfo_register_command_formatting($command, \&ffmpeg_heading_command);
 }
 
-# determine if texinfo is at least version 6.8
-my $program_version_num = version->declare(get_conf('PACKAGE_VERSION'))->numify;
-my $program_version_6_8 = $program_version_num >= 6.008000;
-
 # print the TOC where @contents is used
-if ($program_version_6_8) {
-    set_from_init_file('CONTENTS_OUTPUT_LOCATION', 'inline');
-} else {
-    set_from_init_file('INLINE_CONTENTS', 1);
-}
+set_from_init_file('INLINE_CONTENTS', 1);
 
 # make chapters <h2>
 set_from_init_file('CHAPTER_HEADER_LEVEL', 2);
@@ -192,11 +184,7 @@ EOT
 
     return $head1 . $head_title . $head2 . $head_title . $head3;
 }
-if ($program_version_6_8) {
-    texinfo_register_formatting_function('format_begin_file', \&ffmpeg_begin_file);
-} else {
-    texinfo_register_formatting_function('begin_file', \&ffmpeg_begin_file);
-}
+texinfo_register_formatting_function('begin_file', \&ffmpeg_begin_file);
 
 sub ffmpeg_program_string($)
 {
@@ -213,11 +201,7 @@ sub ffmpeg_program_string($)
       $self->gdt('This document was generated automatically.'));
   }
 }
-if ($program_version_6_8) {
-    texinfo_register_formatting_function('format_program_string', \&ffmpeg_program_string);
-} else {
-    texinfo_register_formatting_function('program_string', \&ffmpeg_program_string);
-}
+texinfo_register_formatting_function('program_string', \&ffmpeg_program_string);
 
 # Customized file ending
 sub ffmpeg_end_file($)
@@ -236,11 +220,7 @@ EOT
 EOT
     return $program_text . $footer;
 }
-if ($program_version_6_8) {
-    texinfo_register_formatting_function('format_end_file', \&ffmpeg_end_file);
-} else {
-    texinfo_register_formatting_function('end_file', \&ffmpeg_end_file);
-}
+texinfo_register_formatting_function('end_file', \&ffmpeg_end_file);
 
 # Dummy title command
 # Ignore title. Title is handled through ffmpeg_begin_file().

doc/texi2pod.pl

@@ -172,9 +172,6 @@ INF: while(<$inf>) {
         } elsif ($ended =~ /^(?:itemize|enumerate|(?:multi|[fv])?table)$/) {
             $_ = "\n=back\n";
             $ic = pop @icstack;
-        } elsif ($ended =~ /^float$/) {
-            $_ = "\n=back\n";
-            $ic = pop @icstack;
         } else {
             die "unknown command \@end $ended at line $.\n";
         }
@@ -300,12 +297,6 @@ INF: while(<$inf>) {
         $_ = "";        # need a paragraph break
     };
 
-    /^\@(float)\s+\w+/ and do {
-        push @endwstack, $endw;
-        $endw = $1;
-        $_ = "\n=over 4\n";
-    };
-
     /^\@item\s+(.*\S)\s*$/ and $endw eq "multitable" and do {
         my $columns = $1;
         $columns =~ s/\@tab/ : /;

doc/transforms.md

@@ -1,706 +0,0 @@
-The basis transforms used for FFT and various other derived functions are based
-on the following unrollings.
-The functions can be easily adapted to double precision floats as well.
-
-# Parity permutation
-The basis transforms described here all use the following permutation:
-
-``` C
-void ff_tx_gen_split_radix_parity_revtab(int *revtab, int len, int inv,
-                                         int basis, int dual_stride);
-```
-Parity means even and odd complex numbers will be split, e.g. the even
-coefficients will come first, after which the odd coefficients will be
-placed. For example, a 4-point transform's coefficients after reordering:
-`z[0].re, z[0].im, z[2].re, z[2].im, z[1].re, z[1].im, z[3].re, z[3].im`
-
-The basis argument is the length of the largest non-composite transform
-supported, and also implies that the basis/2 transform is supported as well,
-as the split-radix algorithm requires it to be.
-
-The dual_stride argument indicates that both the basis, as well as the
-basis/2 transforms support doing two transforms at once, and the coefficients
-will be interleaved between each pair in a split-radix like so (stride == 2):
-`tx1[0], tx1[2], tx2[0], tx2[2], tx1[1], tx1[3], tx2[1], tx2[3]`
-A non-zero number switches this on, with the value indicating the stride
-(how many values of 1 transform to put first before switching to the other).
-Must be a power of two or 0. Must be less than the basis.
-Value will be clipped to the transform size, so for a basis of 16 and a
-dual_stride of 8, dual 8-point transforms will be laid out as if dual_stride
-was set to 4.
-Usually you'll set this to half the complex numbers that fit in a single
-register or 0. This allows to reuse SSE functions as dual-transform
-functions in AVX mode.
-If length is smaller than basis/2 this function will not do anything.
-
-# 4-point FFT transform
-The only permutation this transform needs is to swap the `z[1]` and `z[2]`
-elements when performing an inverse transform, which in the assembly code is
-hardcoded with the function itself being templated and duplicated for each
-direction.
-
-``` C
-static void fft4(FFTComplex *z)
-{
-    FFTSample r1 = z[0].re - z[2].re;
-    FFTSample r2 = z[0].im - z[2].im;
-    FFTSample r3 = z[1].re - z[3].re;
-    FFTSample r4 = z[1].im - z[3].im;
-    /* r5-r8 second transform */
-
-    FFTSample t1 = z[0].re + z[2].re;
-    FFTSample t2 = z[0].im + z[2].im;
-    FFTSample t3 = z[1].re + z[3].re;
-    FFTSample t4 = z[1].im + z[3].im;
-    /* t5-t8 second transform */
-
-    /* 1sub + 1add = 2 instructions */
-
-    /* 2 shufs */
-    FFTSample a3 = t1 - t3;
-    FFTSample a4 = t2 - t4;
-    FFTSample b3 = r1 - r4;
-    FFTSample b2 = r2 - r3;
-
-    FFTSample a1 = t1 + t3;
-    FFTSample a2 = t2 + t4;
-    FFTSample b1 = r1 + r4;
-    FFTSample b4 = r2 + r3;
-    /* 1 add 1 sub 3 shufs */
-
-    z[0].re = a1;
-    z[0].im = a2;
-    z[2].re = a3;
-    z[2].im = a4;
-
-    z[1].re = b1;
-    z[1].im = b2;
-    z[3].re = b3;
-    z[3].im = b4;
-}
-```
-
-# 8-point AVX FFT transform
-Input must be pre-permuted using the parity lookup table, generated via
-`ff_tx_gen_split_radix_parity_revtab`.
-
-``` C
-static void fft8(FFTComplex *z)
-{
-    FFTSample r1 = z[0].re - z[4].re;
-    FFTSample r2 = z[0].im - z[4].im;
-    FFTSample r3 = z[1].re - z[5].re;
-    FFTSample r4 = z[1].im - z[5].im;
-
-    FFTSample r5 = z[2].re - z[6].re;
-    FFTSample r6 = z[2].im - z[6].im;
-    FFTSample r7 = z[3].re - z[7].re;
-    FFTSample r8 = z[3].im - z[7].im;
-
-    FFTSample q1 = z[0].re + z[4].re;
-    FFTSample q2 = z[0].im + z[4].im;
-    FFTSample q3 = z[1].re + z[5].re;
-    FFTSample q4 = z[1].im + z[5].im;
-
-    FFTSample q5 = z[2].re + z[6].re;
-    FFTSample q6 = z[2].im + z[6].im;
-    FFTSample q7 = z[3].re + z[7].re;
-    FFTSample q8 = z[3].im + z[7].im;
-
-    FFTSample s3 = q1 - q3;
-    FFTSample s1 = q1 + q3;
-    FFTSample s4 = q2 - q4;
-    FFTSample s2 = q2 + q4;
-
-    FFTSample s7 = q5 - q7;
-    FFTSample s5 = q5 + q7;
-    FFTSample s8 = q6 - q8;
-    FFTSample s6 = q6 + q8;
-
-    FFTSample e1 = s1 * -1;
-    FFTSample e2 = s2 * -1;
-    FFTSample e3 = s3 * -1;
-    FFTSample e4 = s4 * -1;
-
-    FFTSample e5 = s5 *  1;
-    FFTSample e6 = s6 *  1;
-    FFTSample e7 = s7 * -1;
-    FFTSample e8 = s8 *  1;
-
-    FFTSample w1 =  e5 - e1;
-    FFTSample w2 =  e6 - e2;
-    FFTSample w3 =  e8 - e3;
-    FFTSample w4 =  e7 - e4;
-
-    FFTSample w5 =  s1 - e5;
-    FFTSample w6 =  s2 - e6;
-    FFTSample w7 =  s3 - e8;
-    FFTSample w8 =  s4 - e7;
-
-    z[0].re = w1;
-    z[0].im = w2;
-    z[2].re = w3;
-    z[2].im = w4;
-    z[4].re = w5;
-    z[4].im = w6;
-    z[6].re = w7;
-    z[6].im = w8;
-
-    FFTSample z1 = r1 - r4;
-    FFTSample z2 = r1 + r4;
-    FFTSample z3 = r3 - r2;
-    FFTSample z4 = r3 + r2;
-
-    FFTSample z5 = r5 - r6;
-    FFTSample z6 = r5 + r6;
-    FFTSample z7 = r7 - r8;
-    FFTSample z8 = r7 + r8;
-
-    z3 *= -1;
-    z5 *= -M_SQRT1_2;
-    z6 *= -M_SQRT1_2;
-    z7 *=  M_SQRT1_2;
-    z8 *=  M_SQRT1_2;
-
-    FFTSample t5 = z7 - z6;
-    FFTSample t6 = z8 + z5;
-    FFTSample t7 = z8 - z5;
-    FFTSample t8 = z7 + z6;
-
-    FFTSample u1 =  z2 + t5;
-    FFTSample u2 =  z3 + t6;
-    FFTSample u3 =  z1 - t7;
-    FFTSample u4 =  z4 + t8;
-
-    FFTSample u5 =  z2 - t5;
-    FFTSample u6 =  z3 - t6;
-    FFTSample u7 =  z1 + t7;
-    FFTSample u8 =  z4 - t8;
-
-    z[1].re = u1;
-    z[1].im = u2;
-    z[3].re = u3;
-    z[3].im = u4;
-    z[5].re = u5;
-    z[5].im = u6;
-    z[7].re = u7;
-    z[7].im = u8;
-}
-```
-
-As you can see, there are 2 independent paths, one for even and one for odd coefficients.
-This theme continues throughout the document. Note that in the actual assembly code,
-the paths are interleaved to improve unit saturation and CPU dependency tracking, so
-to more clearly see them, you'll need to deinterleave the instructions.
-
-# 8-point SSE/ARM64 FFT transform
-Input must be pre-permuted using the parity lookup table, generated via
-`ff_tx_gen_split_radix_parity_revtab`.
-
-``` C
-static void fft8(FFTComplex *z)
-{
-    FFTSample r1 = z[0].re - z[4].re;
-    FFTSample r2 = z[0].im - z[4].im;
-    FFTSample r3 = z[1].re - z[5].re;
-    FFTSample r4 = z[1].im - z[5].im;
-
-    FFTSample j1 = z[2].re - z[6].re;
-    FFTSample j2 = z[2].im - z[6].im;
-    FFTSample j3 = z[3].re - z[7].re;
-    FFTSample j4 = z[3].im - z[7].im;
-
-    FFTSample q1 = z[0].re + z[4].re;
-    FFTSample q2 = z[0].im + z[4].im;
-    FFTSample q3 = z[1].re + z[5].re;
-    FFTSample q4 = z[1].im + z[5].im;
-
-    FFTSample k1 = z[2].re + z[6].re;
-    FFTSample k2 = z[2].im + z[6].im;
-    FFTSample k3 = z[3].re + z[7].re;
-    FFTSample k4 = z[3].im + z[7].im;
-    /* 2 add 2 sub = 4 */
-
-    /* 2 shufs, 1 add 1 sub = 4 */
-    FFTSample s1 = q1 + q3;
-    FFTSample s2 = q2 + q4;
-    FFTSample g1 = k3 + k1;
-    FFTSample g2 = k2 + k4;
-
-    FFTSample s3 = q1 - q3;
-    FFTSample s4 = q2 - q4;
-    FFTSample g4 = k3 - k1;
-    FFTSample g3 = k2 - k4;
-
-    /* 1 unpack + 1 shuffle = 2 */
-
-    /* 1 add */
-    FFTSample w1 =  s1 + g1;
-    FFTSample w2 =  s2 + g2;
-    FFTSample w3 =  s3 + g3;
-    FFTSample w4 =  s4 + g4;
-
-    /* 1 sub */
-    FFTSample h1 =  s1 - g1;
-    FFTSample h2 =  s2 - g2;
-    FFTSample h3 =  s3 - g3;
-    FFTSample h4 =  s4 - g4;
-
-    z[0].re = w1;
-    z[0].im = w2;
-    z[2].re = w3;
-    z[2].im = w4;
-    z[4].re = h1;
-    z[4].im = h2;
-    z[6].re = h3;
-    z[6].im = h4;
-
-    /* 1 shuf + 1 shuf + 1 xor + 1 addsub */
-    FFTSample z1 = r1 + r4;
-    FFTSample z2 = r2 - r3;
-    FFTSample z3 = r1 - r4;
-    FFTSample z4 = r2 + r3;
-
-    /* 1 mult */
-    j1 *=  M_SQRT1_2;
-    j2 *= -M_SQRT1_2;
-    j3 *= -M_SQRT1_2;
-    j4 *=  M_SQRT1_2;
-
-    /* 1 shuf + 1 addsub */
-    FFTSample l2 = j1 - j2;
-    FFTSample l1 = j2 + j1;
-    FFTSample l4 = j3 - j4;
-    FFTSample l3 = j4 + j3;
-
-    /* 1 shuf + 1 addsub */
-    FFTSample t1 = l3 - l2;
-    FFTSample t2 = l4 + l1;
-    FFTSample t3 = l1 - l4;
-    FFTSample t4 = l2 + l3;
-
-    /* 1 add */
-    FFTSample u1 =  z1 - t1;
-    FFTSample u2 =  z2 - t2;
-    FFTSample u3 =  z3 - t3;
-    FFTSample u4 =  z4 - t4;
-
-    /* 1 sub */
-    FFTSample o1 =  z1 + t1;
-    FFTSample o2 =  z2 + t2;
-    FFTSample o3 =  z3 + t3;
-    FFTSample o4 =  z4 + t4;
-
-    z[1].re = u1;
-    z[1].im = u2;
-    z[3].re = u3;
-    z[3].im = u4;
-    z[5].re = o1;
-    z[5].im = o2;
-    z[7].re = o3;
-    z[7].im = o4;
-}
-```
-
-Most functions here are highly tuned to use x86's addsub instruction to save on
-external sign mask loading.
-
-# 16-point AVX FFT transform
-This version expects the output of the 8 and 4-point transforms to follow the
-even/odd convention established above.
-
-``` C
-static void fft16(FFTComplex *z)
-{
-    FFTSample cos_16_1 = 0.92387950420379638671875f;
-    FFTSample cos_16_3 = 0.3826834261417388916015625f;
-
-    fft8(z);
-    fft4(z+8);
-    fft4(z+10);
-
-    FFTSample s[32];
-
-    /*
-        xorps m1, m1 - free
-        mulps m0
-        shufps m1, m1, m0
-        xorps
-        addsub
-        shufps
-        mulps
-        mulps
-        addps
-        or (fma3)
-        shufps
-        shufps
-        mulps
-        mulps
-        fma
-        fma
-     */
-
-    s[0]  =  z[8].re*( 1) - z[8].im*( 0);
-    s[1]  =  z[8].im*( 1) + z[8].re*( 0);
-    s[2]  =  z[9].re*( 1) - z[9].im*(-1);
-    s[3]  =  z[9].im*( 1) + z[9].re*(-1);
-
-    s[4]  = z[10].re*( 1) - z[10].im*( 0);
-    s[5]  = z[10].im*( 1) + z[10].re*( 0);
-    s[6]  = z[11].re*( 1) - z[11].im*( 1);
-    s[7]  = z[11].im*( 1) + z[11].re*( 1);
-
-    s[8]  = z[12].re*(  cos_16_1) - z[12].im*( -cos_16_3);
-    s[9]  = z[12].im*(  cos_16_1) + z[12].re*( -cos_16_3);
-    s[10] = z[13].re*(  cos_16_3) - z[13].im*( -cos_16_1);
-    s[11] = z[13].im*(  cos_16_3) + z[13].re*( -cos_16_1);
-
-    s[12] = z[14].re*(  cos_16_1) - z[14].im*(  cos_16_3);
-    s[13] = z[14].im*( -cos_16_1) + z[14].re*( -cos_16_3);
-    s[14] = z[15].re*(  cos_16_3) - z[15].im*(  cos_16_1);
-    s[15] = z[15].im*( -cos_16_3) + z[15].re*( -cos_16_1);
-
-    s[2] *=  M_SQRT1_2;
-    s[3] *=  M_SQRT1_2;
-    s[5] *= -1;
-    s[6] *=  M_SQRT1_2;
-    s[7] *= -M_SQRT1_2;
-
-    FFTSample w5 =  s[0] + s[4];
-    FFTSample w6 =  s[1] - s[5];
-    FFTSample x5 =  s[2] + s[6];
-    FFTSample x6 =  s[3] - s[7];
-
-    FFTSample w3 =  s[4] - s[0];
-    FFTSample w4 =  s[5] + s[1];
-    FFTSample x3 =  s[6] - s[2];
-    FFTSample x4 =  s[7] + s[3];
-
-    FFTSample y5 =  s[8] + s[12];
-    FFTSample y6 =  s[9] - s[13];
-    FFTSample u5 = s[10] + s[14];
-    FFTSample u6 = s[11] - s[15];
-
-    FFTSample y3 = s[12] - s[8];
-    FFTSample y4 = s[13] + s[9];
-    FFTSample u3 = s[14] - s[10];
-    FFTSample u4 = s[15] + s[11];
-
-    /* 2xorps, 2vperm2fs, 2 adds, 2 vpermilps = 8 */
-
-    FFTSample o1  = z[0].re + w5;
-    FFTSample o2  = z[0].im + w6;
-    FFTSample o5  = z[1].re + x5;
-    FFTSample o6  = z[1].im + x6;
-    FFTSample o9  = z[2].re + w4; //h
-    FFTSample o10 = z[2].im + w3;
-    FFTSample o13 = z[3].re + x4;
-    FFTSample o14 = z[3].im + x3;
-
-    FFTSample o17 = z[0].re - w5;
-    FFTSample o18 = z[0].im - w6;
-    FFTSample o21 = z[1].re - x5;
-    FFTSample o22 = z[1].im - x6;
-    FFTSample o25 = z[2].re - w4; //h
-    FFTSample o26 = z[2].im - w3;
-    FFTSample o29 = z[3].re - x4;
-    FFTSample o30 = z[3].im - x3;
-
-    FFTSample o3  = z[4].re + y5;
-    FFTSample o4  = z[4].im + y6;
-    FFTSample o7  = z[5].re + u5;
-    FFTSample o8  = z[5].im + u6;
-    FFTSample o11 = z[6].re + y4; //h
-    FFTSample o12 = z[6].im + y3;
-    FFTSample o15 = z[7].re + u4;
-    FFTSample o16 = z[7].im + u3;
-
-    FFTSample o19 = z[4].re - y5;
-    FFTSample o20 = z[4].im - y6;
-    FFTSample o23 = z[5].re - u5;
-    FFTSample o24 = z[5].im - u6;
-    FFTSample o27 = z[6].re - y4; //h
-    FFTSample o28 = z[6].im - y3;
-    FFTSample o31 = z[7].re - u4;
-    FFTSample o32 = z[7].im - u3;
-
-    /* This is just deinterleaving, happens separately */
-    z[0]  = (FFTComplex){  o1,  o2 };
-    z[1]  = (FFTComplex){  o3,  o4 };
-    z[2]  = (FFTComplex){  o5,  o6 };
-    z[3]  = (FFTComplex){  o7,  o8 };
-    z[4]  = (FFTComplex){  o9, o10 };
-    z[5]  = (FFTComplex){ o11, o12 };
-    z[6]  = (FFTComplex){ o13, o14 };
-    z[7]  = (FFTComplex){ o15, o16 };
-
-    z[8]  = (FFTComplex){ o17, o18 };
-    z[9]  = (FFTComplex){ o19, o20 };
-    z[10] = (FFTComplex){ o21, o22 };
-    z[11] = (FFTComplex){ o23, o24 };
-    z[12] = (FFTComplex){ o25, o26 };
-    z[13] = (FFTComplex){ o27, o28 };
-    z[14] = (FFTComplex){ o29, o30 };
-    z[15] = (FFTComplex){ o31, o32 };
-}
-```
-
-# AVX split-radix synthesis
-To create larger transforms, the following unrolling of the C split-radix
-function is used.
-
-``` C
-#define BF(x, y, a, b)                           \
-    do {                                         \
-        x = (a) - (b);                           \
-        y = (a) + (b);                           \
-    } while (0)
-
-#define BUTTERFLIES(a0,a1,a2,a3)               \
-    do {                                       \
-        r0=a0.re;                              \
-        i0=a0.im;                              \
-        r1=a1.re;                              \
-        i1=a1.im;                              \
-        BF(q3, q5, q5, q1);                    \
-        BF(a2.re, a0.re, r0, q5);              \
-        BF(a3.im, a1.im, i1, q3);              \
-        BF(q4, q6, q2, q6);                    \
-        BF(a3.re, a1.re, r1, q4);              \
-        BF(a2.im, a0.im, i0, q6);              \
-    } while (0)
-
-#undef TRANSFORM
-#define TRANSFORM(a0,a1,a2,a3,wre,wim)         \
-    do {                                       \
-        CMUL(q1, q2, a2.re, a2.im, wre, -wim); \
-        CMUL(q5, q6, a3.re, a3.im, wre,  wim); \
-        BUTTERFLIES(a0, a1, a2, a3);           \
-    } while (0)
-
-#define CMUL(dre, dim, are, aim, bre, bim)       \
-    do {                                         \
-        (dre) = (are) * (bre) - (aim) * (bim);   \
-        (dim) = (are) * (bim) + (aim) * (bre);   \
-    } while (0)
-
-static void recombine(FFTComplex *z, const FFTSample *cos,
-                      unsigned int n)
-{
-    const int o1 = 2*n;
-    const int o2 = 4*n;
-    const int o3 = 6*n;
-    const FFTSample *wim = cos + o1 - 7;
-    FFTSample q1, q2, q3, q4, q5, q6, r0, i0, r1, i1;
-
-#if 0
-    for (int i = 0; i < n; i += 4) {
-#endif
-
-#if 0
-        TRANSFORM(z[ 0 + 0], z[ 0 + 4], z[o2 + 0], z[o2 + 2], cos[0], wim[7]);
-        TRANSFORM(z[ 0 + 1], z[ 0 + 5], z[o2 + 1], z[o2 + 3], cos[2], wim[5]);
-        TRANSFORM(z[ 0 + 2], z[ 0 + 6], z[o2 + 4], z[o2 + 6], cos[4], wim[3]);
-        TRANSFORM(z[ 0 + 3], z[ 0 + 7], z[o2 + 5], z[o2 + 7], cos[6], wim[1]);
-
-        TRANSFORM(z[o1 + 0], z[o1 + 4], z[o3 + 0], z[o3 + 2], cos[1], wim[6]);
-        TRANSFORM(z[o1 + 1], z[o1 + 5], z[o3 + 1], z[o3 + 3], cos[3], wim[4]);
-        TRANSFORM(z[o1 + 2], z[o1 + 6], z[o3 + 4], z[o3 + 6], cos[5], wim[2]);
-        TRANSFORM(z[o1 + 3], z[o1 + 7], z[o3 + 5], z[o3 + 7], cos[7], wim[0]);
-#else
-        FFTSample h[8], j[8], r[8], w[8];
-        FFTSample t[8];
-        FFTComplex *m0 = &z[0];
-        FFTComplex *m1 = &z[4];
-        FFTComplex *m2 = &z[o2 + 0];
-        FFTComplex *m3 = &z[o2 + 4];
-
-        const FFTSample *t1  = &cos[0];
-        const FFTSample *t2  = &wim[0];
-
-        /* 2 loads (tabs) */
-
-        /* 2 vperm2fs, 2 shufs (im), 2 shufs (tabs) */
-        /* 1 xor, 1 add, 1 sub, 4 mults OR 2 mults, 2 fmas */
-        /* 13 OR 10ish (-2 each for second passovers!) */
-
-        w[0] = m2[0].im*t1[0] - m2[0].re*t2[7];
-        w[1] = m2[0].re*t1[0] + m2[0].im*t2[7];
-        w[2] = m2[1].im*t1[2] - m2[1].re*t2[5];
-        w[3] = m2[1].re*t1[2] + m2[1].im*t2[5];
-        w[4] = m3[0].im*t1[4] - m3[0].re*t2[3];
-        w[5] = m3[0].re*t1[4] + m3[0].im*t2[3];
-        w[6] = m3[1].im*t1[6] - m3[1].re*t2[1];
-        w[7] = m3[1].re*t1[6] + m3[1].im*t2[1];
-
-        j[0] = m2[2].im*t1[0] + m2[2].re*t2[7];
-        j[1] = m2[2].re*t1[0] - m2[2].im*t2[7];
-        j[2] = m2[3].im*t1[2] + m2[3].re*t2[5];
-        j[3] = m2[3].re*t1[2] - m2[3].im*t2[5];
-        j[4] = m3[2].im*t1[4] + m3[2].re*t2[3];
-        j[5] = m3[2].re*t1[4] - m3[2].im*t2[3];
-        j[6] = m3[3].im*t1[6] + m3[3].re*t2[1];
-        j[7] = m3[3].re*t1[6] - m3[3].im*t2[1];
-
-        /* 1 add + 1 shuf */
-        t[1] = j[0] + w[0];
-        t[0] = j[1] + w[1];
-        t[3] = j[2] + w[2];
-        t[2] = j[3] + w[3];
-        t[5] = j[4] + w[4];
-        t[4] = j[5] + w[5];
-        t[7] = j[6] + w[6];
-        t[6] = j[7] + w[7];
-
-        /* 1 sub + 1 xor */
-        r[0] =  (w[0] - j[0]);
-        r[1] = -(w[1] - j[1]);
-        r[2] =  (w[2] - j[2]);
-        r[3] = -(w[3] - j[3]);
-        r[4] =  (w[4] - j[4]);
-        r[5] = -(w[5] - j[5]);
-        r[6] =  (w[6] - j[6]);
-        r[7] = -(w[7] - j[7]);
-
-        /* Min: 2 subs, 2 adds, 2 vperm2fs (OPTIONAL) */
-        m2[0].re = m0[0].re - t[0];
-        m2[0].im = m0[0].im - t[1];
-        m2[1].re = m0[1].re - t[2];
-        m2[1].im = m0[1].im - t[3];
-        m3[0].re = m0[2].re - t[4];
-        m3[0].im = m0[2].im - t[5];
-        m3[1].re = m0[3].re - t[6];
-        m3[1].im = m0[3].im - t[7];
-
-        m2[2].re = m1[0].re - r[0];
-        m2[2].im = m1[0].im - r[1];
-        m2[3].re = m1[1].re - r[2];
-        m2[3].im = m1[1].im - r[3];
-        m3[2].re = m1[2].re - r[4];
-        m3[2].im = m1[2].im - r[5];
-        m3[3].re = m1[3].re - r[6];
-        m3[3].im = m1[3].im - r[7];
-
-        m0[0].re = m0[0].re + t[0];
-        m0[0].im = m0[0].im + t[1];
-        m0[1].re = m0[1].re + t[2];
-        m0[1].im = m0[1].im + t[3];
-        m0[2].re = m0[2].re + t[4];
-        m0[2].im = m0[2].im + t[5];
-        m0[3].re = m0[3].re + t[6];
-        m0[3].im = m0[3].im + t[7];
-
-        m1[0].re = m1[0].re + r[0];
-        m1[0].im = m1[0].im + r[1];
-        m1[1].re = m1[1].re + r[2];
-        m1[1].im = m1[1].im + r[3];
-        m1[2].re = m1[2].re + r[4];
-        m1[2].im = m1[2].im + r[5];
-        m1[3].re = m1[3].re + r[6];
-        m1[3].im = m1[3].im + r[7];
-
-        /* Identical for below, but with the following parameters */
-        m0 = &z[o1];
-        m1 = &z[o1 + 4];
-        m2 = &z[o3 + 0];
-        m3 = &z[o3 + 4];
-        t1  = &cos[1];
-        t2  = &wim[-1];
-
-        w[0] = m2[0].im*t1[0] - m2[0].re*t2[7];
-        w[1] = m2[0].re*t1[0] + m2[0].im*t2[7];
-        w[2] = m2[1].im*t1[2] - m2[1].re*t2[5];
-        w[3] = m2[1].re*t1[2] + m2[1].im*t2[5];
-        w[4] = m3[0].im*t1[4] - m3[0].re*t2[3];
-        w[5] = m3[0].re*t1[4] + m3[0].im*t2[3];
-        w[6] = m3[1].im*t1[6] - m3[1].re*t2[1];
-        w[7] = m3[1].re*t1[6] + m3[1].im*t2[1];
-
-        j[0] = m2[2].im*t1[0] + m2[2].re*t2[7];
-        j[1] = m2[2].re*t1[0] - m2[2].im*t2[7];
-        j[2] = m2[3].im*t1[2] + m2[3].re*t2[5];
-        j[3] = m2[3].re*t1[2] - m2[3].im*t2[5];
-        j[4] = m3[2].im*t1[4] + m3[2].re*t2[3];
-        j[5] = m3[2].re*t1[4] - m3[2].im*t2[3];
-        j[6] = m3[3].im*t1[6] + m3[3].re*t2[1];
-        j[7] = m3[3].re*t1[6] - m3[3].im*t2[1];
-
-        /* 1 add + 1 shuf */
-        t[1] = j[0] + w[0];
-        t[0] = j[1] + w[1];
-        t[3] = j[2] + w[2];
-        t[2] = j[3] + w[3];
-        t[5] = j[4] + w[4];
-        t[4] = j[5] + w[5];
-        t[7] = j[6] + w[6];
-        t[6] = j[7] + w[7];
-
-        /* 1 sub + 1 xor */
-        r[0] =  (w[0] - j[0]);
-        r[1] = -(w[1] - j[1]);
-        r[2] =  (w[2] - j[2]);
-        r[3] = -(w[3] - j[3]);
-        r[4] =  (w[4] - j[4]);
-        r[5] = -(w[5] - j[5]);
-        r[6] =  (w[6] - j[6]);
-        r[7] = -(w[7] - j[7]);
-
-        /* Min: 2 subs, 2 adds, 2 vperm2fs (OPTIONAL) */
-        m2[0].re = m0[0].re - t[0];
-        m2[0].im = m0[0].im - t[1];
-        m2[1].re = m0[1].re - t[2];
-        m2[1].im = m0[1].im - t[3];
-        m3[0].re = m0[2].re - t[4];
-        m3[0].im = m0[2].im - t[5];
-        m3[1].re = m0[3].re - t[6];
-        m3[1].im = m0[3].im - t[7];
-
-        m2[2].re = m1[0].re - r[0];
-        m2[2].im = m1[0].im - r[1];
-        m2[3].re = m1[1].re - r[2];
-        m2[3].im = m1[1].im - r[3];
-        m3[2].re = m1[2].re - r[4];
-        m3[2].im = m1[2].im - r[5];
-        m3[3].re = m1[3].re - r[6];
-        m3[3].im = m1[3].im - r[7];
-
-        m0[0].re = m0[0].re + t[0];
-        m0[0].im = m0[0].im + t[1];
-        m0[1].re = m0[1].re + t[2];
-        m0[1].im = m0[1].im + t[3];
-        m0[2].re = m0[2].re + t[4];
-        m0[2].im = m0[2].im + t[5];
-        m0[3].re = m0[3].re + t[6];
-        m0[3].im = m0[3].im + t[7];
-
-        m1[0].re = m1[0].re + r[0];
-        m1[0].im = m1[0].im + r[1];
-        m1[1].re = m1[1].re + r[2];
-        m1[1].im = m1[1].im + r[3];
-        m1[2].re = m1[2].re + r[4];
-        m1[2].im = m1[2].im + r[5];
-        m1[3].re = m1[3].re + r[6];
-        m1[3].im = m1[3].im + r[7];
-#endif
-
-#if 0
-        z   +=   4; // !!!
-        cos += 2*4;
-        wim -= 2*4;
-    }
-#endif
-}
-```
-
-The macros used are identical to those in the generic C version, only with all
-variable declarations exported to the function body.
-An important point here is that the high frequency registers (m2 and m3) have
-their high and low halves swapped in the output. This is intentional, as the
-inputs must also have the same layout, and therefore, the input swapping is only
-performed once for the bottom-most basis transform, with all subsequent combinations
-using the already swapped halves.
-
-Also note that this function requires a special iteration way, due to coefficients
-beginning to overlap, particularly `[o1]` with `[0]` after the second iteration.
-To iterate further, set `z = &z[16]` via `z += 8` for the second iteration. After
-the 4th iteration, the layout resets, so repeat the same.

doc/utils.texi

@@ -110,13 +110,11 @@ maximum of 2 digits. The @var{m} at the end expresses decimal value for
 @emph{or}
 
 @example
-[-]@var{S}+[.@var{m}...][s|ms|us]
+[-]@var{S}+[.@var{m}...]
 @end example
 
 @var{S} expresses the number of seconds, with the optional decimal part
-@var{m}.  The optional literal suffixes @samp{s}, @samp{ms} or @samp{us}
-indicate to interpret the value as seconds, milliseconds or microseconds,
-respectively.
+@var{m}.
 
 In both expressions, the optional @samp{-} indicates negative duration.
 
@@ -128,15 +126,6 @@ The following examples are all valid time duration:
 @item 55
 55 seconds
 
-@item 0.2
-0.2 seconds
-
-@item 200ms
-200 milliseconds, that's 0.2s
-
-@item 200000us
-200000 microseconds, that's 0.2s
-
 @item 12:03:45
 12 hours, 03 minutes and 45 seconds
 
@@ -715,36 +704,24 @@ FL+FR+FC+LFE+BL+BR+FLC+FRC
 FL+FR+FC+LFE+FLC+FRC+SL+SR
 @item octagonal
 FL+FR+FC+BL+BR+BC+SL+SR
-@item hexadecagonal
-FL+FR+FC+BL+BR+BC+SL+SR+WL+WR+TBL+TBR+TBC+TFC+TFL+TFR
 @item downmix
 DL+DR
-@item 22.2
-FL+FR+FC+LFE+BL+BR+FLC+FRC+BC+SL+SR+TC+TFL+TFC+TFR+TBL+TBC+TBR+LFE2+TSL+TSR+BFC+BFL+BFR
 @end table
 
-A custom channel layout can be specified as a sequence of terms, separated by '+'.
-Each term can be:
+A custom channel layout can be specified as a sequence of terms, separated by
+'+' or '|'. Each term can be:
 @itemize
-@item
-the name of a single channel (e.g. @samp{FL}, @samp{FR}, @samp{FC}, @samp{LFE}, etc.),
-each optionally containing a custom name after a '@@', (e.g. @samp{FL@@Left},
-@samp{FR@@Right}, @samp{FC@@Center}, @samp{LFE@@Low_Frequency}, etc.)
-@end itemize
-
-A standard channel layout can be specified by the following:
-@itemize
-@item
-the name of a single channel (e.g. @samp{FL}, @samp{FR}, @samp{FC}, @samp{LFE}, etc.)
-
 @item
 the name of a standard channel layout (e.g. @samp{mono},
 @samp{stereo}, @samp{4.0}, @samp{quad}, @samp{5.0}, etc.)
 
+@item
+the name of a single channel (e.g. @samp{FL}, @samp{FR}, @samp{FC}, @samp{LFE}, etc.)
+
 @item
 a number of channels, in decimal, followed by 'c', yielding the default channel
 layout for that number of channels (see the function
-@code{av_channel_layout_default}). Note that not all channel counts have a
+@code{av_get_default_channel_layout}). Note that not all channel counts have a
 default layout.
 
 @item
@@ -761,7 +738,7 @@ Before libavutil version 53 the trailing character "c" to specify a number of
 channels was optional, but now it is required, while a channel layout mask can
 also be specified as a decimal number (if and only if not followed by "c" or "C").
 
-See also the function @code{av_channel_layout_from_string} defined in
+See also the function @code{av_get_channel_layout} defined in
 @file{libavutil/channel_layout.h}.
 @c man end SYNTAX
 
@@ -943,9 +920,6 @@ corresponding input value will be returned.
 @item round(expr)
 Round the value of expression @var{expr} to the nearest integer. For example, "round(1.5)" is "2.0".
 
-@item sgn(x)
-Compute sign of @var{x}.
-
 @item sin(x)
 Compute sine of @var{x}.
 

doc/writing_filters.txt

@@ -389,7 +389,7 @@ distributor with something like this:
 
     td.in  = in;
     td.out = out;
-    ctx->internal->execute(ctx, filter_slice, &td, NULL, FFMIN(outlink->h, ff_filter_get_nb_threads(ctx)));
+    ctx->internal->execute(ctx, filter_slice, &td, NULL, FFMIN(outlink->h, ctx->graph->nb_threads));
 
     // ...
 
@@ -418,4 +418,4 @@ done:
 
 When all of this is done, you can submit your patch to the ffmpeg-devel
 mailing-list for review.  If you need any help, feel free to come on our IRC
-channel, #ffmpeg-devel on irc.libera.chat.
+channel, #ffmpeg-devel on irc.freenode.net.

ffbuild/.gitignore

@@ -1,6 +1,4 @@
 /.config
-/bin2c
-/bin2c.exe
 /config.fate
 /config.log
 /config.mak

ffbuild/arch.mak

@@ -8,9 +8,7 @@ OBJS-$(HAVE_MIPSFPU)   += $(MIPSFPU-OBJS)    $(MIPSFPU-OBJS-yes)
 OBJS-$(HAVE_MIPSDSP)   += $(MIPSDSP-OBJS)    $(MIPSDSP-OBJS-yes)
 OBJS-$(HAVE_MIPSDSPR2) += $(MIPSDSPR2-OBJS)  $(MIPSDSPR2-OBJS-yes)
 OBJS-$(HAVE_MSA)       += $(MSA-OBJS)        $(MSA-OBJS-yes)
-OBJS-$(HAVE_MMI)       += $(MMI-OBJS)        $(MMI-OBJS-yes)
-OBJS-$(HAVE_LSX)       += $(LSX-OBJS)        $(LSX-OBJS-yes)
-OBJS-$(HAVE_LASX)      += $(LASX-OBJS)       $(LASX-OBJS-yes)
+OBJS-$(HAVE_MMI)   += $(MMI-OBJS)   $(MMI-OBJS-yes)
 
 OBJS-$(HAVE_ALTIVEC) += $(ALTIVEC-OBJS) $(ALTIVEC-OBJS-yes)
 OBJS-$(HAVE_VSX)     += $(VSX-OBJS) $(VSX-OBJS-yes)

ffbuild/bin2c.c

@@ -1,76 +0,0 @@
-/*
- * This file is part of FFmpeg.
- *
- * 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 <string.h>
-#include <stdio.h>
-
-int main(int argc, char **argv)
-{
-    const char *name;
-    FILE *input, *output;
-    unsigned int length = 0;
-    unsigned char data;
-
-    if (argc < 3 || argc > 4)
-        return 1;
-
-    input = fopen(argv[1], "rb");
-    if (!input)
-        return -1;
-
-    output = fopen(argv[2], "wb");
-    if (!output)
-        return -1;
-
-    if (argc == 4) {
-        name = argv[3];
-    } else {
-        size_t arglen = strlen(argv[1]);
-        name = argv[1];
-
-        for (int i = 0; i < arglen; i++) {
-            if (argv[1][i] == '.')
-                argv[1][i] = '_';
-            else if (argv[1][i] == '/')
-                name = &argv[1][i+1];
-        }
-    }
-
-    fprintf(output, "const unsigned char ff_%s_data[] = { ", name);
-
-    while (fread(&data, 1, 1, input) > 0) {
-        fprintf(output, "0x%02x, ", data);
-        length++;
-    }
-
-    fprintf(output, "0x00 };\n");
-    fprintf(output, "const unsigned int ff_%s_len = %u;\n", name, length);
-
-    fclose(output);
-
-    if (ferror(input) || !feof(input))
-        return -1;
-
-    fclose(input);
-
-    return 0;
-}

ffbuild/common.mak

@@ -12,13 +12,10 @@ endif
 
 ifndef SUBDIR
 
-BIN2CEXE = ffbuild/bin2c$(HOSTEXESUF)
-BIN2C = $(BIN2CEXE)
-
 ifndef V
 Q      = @
 ECHO   = printf "$(1)\t%s\n" $(2)
-BRIEF  = CC CXX OBJCC HOSTCC HOSTLD AS X86ASM AR LD STRIP CP WINDRES NVCC BIN2C
+BRIEF  = CC CXX OBJCC HOSTCC HOSTLD AS X86ASM AR LD STRIP CP WINDRES NVCC
 SILENT = DEPCC DEPHOSTCC DEPAS DEPX86ASM RANLIB RM
 
 MSG    = $@
@@ -29,8 +26,7 @@ $(foreach VAR,$(SILENT),$(eval override $(VAR) = @$($(VAR))))
 $(eval INSTALL = @$(call ECHO,INSTALL,$$(^:$(SRC_DIR)/%=%)); $(INSTALL))
 endif
 
-# Prepend to a recursively expanded variable without making it simply expanded.
-PREPEND = $(eval $(1) = $(patsubst %,$$(%), $(2)) $(value $(1)))
+ALLFFLIBS = avcodec avdevice avfilter avformat avresample avutil postproc swscale swresample
 
 # NASM requires -I path terminated with /
 IFLAGS     := -I. -I$(SRC_LINK)/
@@ -40,17 +36,16 @@ CCFLAGS     = $(CPPFLAGS) $(CFLAGS)
 OBJCFLAGS  += $(EOBJCFLAGS)
 OBJCCFLAGS  = $(CPPFLAGS) $(CFLAGS) $(OBJCFLAGS)
 ASFLAGS    := $(CPPFLAGS) $(ASFLAGS)
-# Use PREPEND here so that later (target-dependent) additions to CPPFLAGS
-# end up in CXXFLAGS.
-$(call PREPEND,CXXFLAGS, CPPFLAGS CFLAGS)
+CXXFLAGS   := $(CPPFLAGS) $(CFLAGS) $(CXXFLAGS)
 X86ASMFLAGS += $(IFLAGS:%=%/) -I$(<D)/ -Pconfig.asm
+NVCCFLAGS  += -ptx
 
 HOSTCCFLAGS = $(IFLAGS) $(HOSTCPPFLAGS) $(HOSTCFLAGS)
 LDFLAGS    := $(ALLFFLIBS:%=$(LD_PATH)lib%) $(LDFLAGS)
 
 define COMPILE
        $(call $(1)DEP,$(1))
-       $($(1)) $($(1)FLAGS) $($(2)) $($(1)_DEPFLAGS) $($(1)_C) $($(1)_O) $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<)
+       $($(1)) $($(1)FLAGS) $($(1)_DEPFLAGS) $($(1)_C) $($(1)_O) $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<)
 endef
 
 COMPILE_C = $(call COMPILE,CC)
@@ -60,22 +55,6 @@ COMPILE_M = $(call COMPILE,OBJCC)
 COMPILE_X86ASM = $(call COMPILE,X86ASM)
 COMPILE_HOSTC = $(call COMPILE,HOSTCC)
 COMPILE_NVCC = $(call COMPILE,NVCC)
-COMPILE_MMI = $(call COMPILE,CC,MMIFLAGS)
-COMPILE_MSA = $(call COMPILE,CC,MSAFLAGS)
-COMPILE_LSX = $(call COMPILE,CC,LSXFLAGS)
-COMPILE_LASX = $(call COMPILE,CC,LASXFLAGS)
-
-%_mmi.o: %_mmi.c
-	$(COMPILE_MMI)
-
-%_msa.o: %_msa.c
-	$(COMPILE_MSA)
-
-%_lsx.o: %_lsx.c
-	$(COMPILE_LSX)
-
-%_lasx.o: %_lasx.c
-	$(COMPILE_LASX)
 
 %.o: %.c
 	$(COMPILE_C)
@@ -104,7 +83,7 @@ COMPILE_LASX = $(call COMPILE,CC,LASXFLAGS)
 	-$(if $(ASMSTRIPFLAGS), $(STRIP) $(ASMSTRIPFLAGS) $@)
 
 %.o: %.rc
-	$(WINDRES) $(IFLAGS) $(foreach ARG,$(CC_DEPFLAGS),--preprocessor-arg "$(ARG)") -o $@ $<
+	$(WINDRES) $(IFLAGS) --preprocessor "$(DEPWINDRES) -E -xc-header -DRC_INVOKED $(CC_DEPFLAGS)" -o $@ $<
 
 %.i: %.c
 	$(CC) $(CCFLAGS) $(CC_E) $<
@@ -112,40 +91,16 @@ COMPILE_LASX = $(call COMPILE,CC,LASXFLAGS)
 %.h.c:
 	$(Q)echo '#include "$*.h"' >$@
 
-$(BIN2CEXE): ffbuild/bin2c_host.o
-	$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $^ $(HOSTEXTRALIBS)
-
-%.metal.air: %.metal
-	$(METALCC) $< -o $@
-
-%.metallib: %.metal.air
-	$(METALLIB) --split-module-without-linking $< -o $@
-
-%.metallib.c: %.metallib $(BIN2CEXE)
-	$(BIN2C) $< $@ $(subst .,_,$(basename $(notdir $@)))
-
-%.ptx: %.cu $(SRC_PATH)/compat/cuda/cuda_runtime.h
+%.ptx: %.cu
 	$(COMPILE_NVCC)
 
-ifdef CONFIG_PTX_COMPRESSION
-%.ptx.gz: TAG = GZIP
-%.ptx.gz: %.ptx
-	$(M)gzip -c9 $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<) >$@
-
-%.ptx.c: %.ptx.gz $(BIN2CEXE)
-	$(BIN2C) $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<) $@ $(subst .,_,$(basename $(notdir $@)))
-else
-%.ptx.c: %.ptx $(BIN2CEXE)
-	$(BIN2C) $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<) $@ $(subst .,_,$(basename $(notdir $@)))
-endif
-
-clean::
-	$(RM) $(BIN2CEXE)
+%.ptx.c: %.ptx
+	$(Q)sh $(SRC_PATH)/compat/cuda/ptx2c.sh $@ $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<)
 
 %.c %.h %.pc %.ver %.version: TAG = GEN
 
 # Dummy rule to stop make trying to rebuild removed or renamed headers
-%.h %_template.c:
+%.h:
 	@:
 
 # Disable suffix rules.  Most of the builtin rules are suffix rules,
@@ -160,8 +115,6 @@ include $(SRC_PATH)/ffbuild/arch.mak
 
 OBJS      += $(OBJS-yes)
 SLIBOBJS  += $(SLIBOBJS-yes)
-SHLIBOBJS += $(SHLIBOBJS-yes)
-STLIBOBJS += $(STLIBOBJS-yes)
 FFLIBS    := $($(NAME)_FFLIBS) $(FFLIBS-yes) $(FFLIBS)
 TESTPROGS += $(TESTPROGS-yes)
 
@@ -170,8 +123,6 @@ FFEXTRALIBS := $(LDLIBS:%=$(LD_LIB)) $(foreach lib,EXTRALIBS-$(NAME) $(FFLIBS:%=
 
 OBJS      := $(sort $(OBJS:%=$(SUBDIR)%))
 SLIBOBJS  := $(sort $(SLIBOBJS:%=$(SUBDIR)%))
-SHLIBOBJS := $(sort $(SHLIBOBJS:%=$(SUBDIR)%))
-STLIBOBJS := $(sort $(STLIBOBJS:%=$(SUBDIR)%))
 TESTOBJS  := $(TESTOBJS:%=$(SUBDIR)tests/%) $(TESTPROGS:%=$(SUBDIR)tests/%.o)
 TESTPROGS := $(TESTPROGS:%=$(SUBDIR)tests/%$(EXESUF))
 HOSTOBJS  := $(HOSTPROGS:%=$(SUBDIR)%.o)
@@ -193,7 +144,7 @@ HOBJS        = $(filter-out $(SKIPHEADERS:.h=.h.o),$(ALLHEADERS:.h=.h.o))
 PTXOBJS      = $(filter %.ptx.o,$(OBJS))
 $(HOBJS):     CCFLAGS += $(CFLAGS_HEADERS)
 checkheaders: $(HOBJS)
-.SECONDARY:   $(HOBJS:.o=.c) $(PTXOBJS:.o=.c) $(PTXOBJS:.o=.gz) $(PTXOBJS:.o=)
+.SECONDARY:   $(HOBJS:.o=.c) $(PTXOBJS:.o=.c) $(PTXOBJS:.o=)
 
 alltools: $(TOOLS)
 
@@ -207,14 +158,12 @@ $(OBJS):     | $(sort $(dir $(OBJS)))
 $(HOBJS):    | $(sort $(dir $(HOBJS)))
 $(HOSTOBJS): | $(sort $(dir $(HOSTOBJS)))
 $(SLIBOBJS): | $(sort $(dir $(SLIBOBJS)))
-$(SHLIBOBJS): | $(sort $(dir $(SHLIBOBJS)))
-$(STLIBOBJS): | $(sort $(dir $(STLIBOBJS)))
 $(TESTOBJS): | $(sort $(dir $(TESTOBJS)))
 $(TOOLOBJS): | tools
 
-OUTDIRS := $(OUTDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(SLIBOBJS) $(SHLIBOBJS) $(STLIBOBJS) $(TESTOBJS))
+OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(SLIBOBJS) $(TESTOBJS))
 
-CLEANSUFFIXES     = *.d *.gcda *.gcno *.h.c *.ho *.map *.o *.pc *.ptx *.ptx.gz *.ptx.c *.ver *.version *$(DEFAULT_X86ASMD).asm *~ *.ilk *.pdb
+CLEANSUFFIXES     = *.d *.gcda *.gcno *.h.c *.ho *.map *.o *.pc *.ptx *.ptx.c *.ver *.version *$(DEFAULT_X86ASMD).asm *~
 LIBSUFFIXES       = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a
 
 define RULES
@@ -224,4 +173,4 @@ endef
 
 $(eval $(RULES))
 
--include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d) $(SHLIBOBJS:.o=.d) $(STLIBOBJS:.o=.d) $(SLIBOBJS:.o=.d)) $(OBJS:.o=$(DEFAULT_X86ASMD).d)
+-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d) $(SLIBOBJS:.o=.d)) $(OBJS:.o=$(DEFAULT_X86ASMD).d)

ffbuild/library.mak

@@ -14,26 +14,10 @@ INSTHEADERS := $(INSTHEADERS) $(HEADERS:%=$(SUBDIR)%)
 all-$(CONFIG_STATIC): $(SUBDIR)$(LIBNAME)  $(SUBDIR)lib$(FULLNAME).pc
 all-$(CONFIG_SHARED): $(SUBDIR)$(SLIBNAME) $(SUBDIR)lib$(FULLNAME).pc
 
-LIBOBJS := $(OBJS) $(SHLIBOBJS) $(STLIBOBJS) $(SUBDIR)%.h.o $(TESTOBJS)
+LIBOBJS := $(OBJS) $(SUBDIR)%.h.o $(TESTOBJS)
 $(LIBOBJS) $(LIBOBJS:.o=.s) $(LIBOBJS:.o=.i):   CPPFLAGS += -DHAVE_AV_CONFIG_H
 
-ifdef CONFIG_SHARED
-# In case both shared libs and static libs are enabled, it can happen
-# that a user might want to link e.g. libavformat statically, but
-# libavcodec and the other libs dynamically. In this case
-# libavformat won't be able to access libavcodec's internal symbols,
-# so that they have to be duplicated into the archive just like
-# for purely shared builds.
-# Test programs are always statically linked against their library
-# to be able to access their library's internals, even with shared builds.
-# Yet linking against dependend libraries still uses dynamic linking.
-# This means that we are in the scenario described above.
-# In case only static libs are used, the linker will only use
-# one of these copies; this depends on the duplicated object files
-# containing exactly the same symbols.
-OBJS += $(SHLIBOBJS)
-endif
-$(SUBDIR)$(LIBNAME): $(OBJS) $(STLIBOBJS)
+$(SUBDIR)$(LIBNAME): $(OBJS)
 	$(RM) $@
 	$(AR) $(ARFLAGS) $(AR_O) $^
 	$(RANLIB) $@
@@ -52,8 +36,8 @@ $(LIBOBJS): CPPFLAGS += -DBUILDING_$(NAME)
 $(TESTPROGS) $(TOOLS): %$(EXESUF): %.o
 	$$(LD) $(LDFLAGS) $(LDEXEFLAGS) $$(LD_O) $$(filter %.o,$$^) $$(THISLIB) $(FFEXTRALIBS) $$(EXTRALIBS-$$(*F)) $$(ELIBS)
 
-$(SUBDIR)lib$(NAME).version: $(SUBDIR)version.h $(SUBDIR)version_major.h | $(SUBDIR)
-	$$(M) $$(SRC_PATH)/ffbuild/libversion.sh $(NAME) $$^ > $$@
+$(SUBDIR)lib$(NAME).version: $(SUBDIR)version.h | $(SUBDIR)
+	$$(M) $$(SRC_PATH)/ffbuild/libversion.sh $(NAME) $$< > $$@
 
 $(SUBDIR)lib$(FULLNAME).pc: $(SUBDIR)version.h ffbuild/config.sh | $(SUBDIR)
 	$$(M) $$(SRC_PATH)/ffbuild/pkgconfig_generate.sh $(NAME) "$(DESC)"
@@ -64,7 +48,7 @@ $(SUBDIR)lib$(NAME).ver: $(SUBDIR)lib$(NAME).v $(OBJS)
 $(SUBDIR)$(SLIBNAME): $(SUBDIR)$(SLIBNAME_WITH_MAJOR)
 	$(Q)cd ./$(SUBDIR) && $(LN_S) $(SLIBNAME_WITH_MAJOR) $(SLIBNAME)
 
-$(SUBDIR)$(SLIBNAME_WITH_MAJOR): $(OBJS) $(SHLIBOBJS) $(SLIBOBJS) $(SUBDIR)lib$(NAME).ver
+$(SUBDIR)$(SLIBNAME_WITH_MAJOR): $(OBJS) $(SLIBOBJS) $(SUBDIR)lib$(NAME).ver
 	$(SLIB_CREATE_DEF_CMD)
 	$$(LD) $(SHFLAGS) $(LDFLAGS) $(LDSOFLAGS) $$(LD_O) $$(filter %.o,$$^) $(FFEXTRALIBS)
 	$(SLIB_EXTRA_CMD)