avcodec/noise_bsf: Check for wrapped frames

Wrapped frames contain pointers so they need specific code to noise them, the generic code would lead to segfaults Signed-off-by: Michael Niedermayer <michael@niedermayer.cc> (cherry picked from commit 0889ebc577) Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Changelog: update
2025-12-06 14:59:59 +01:00 · 2023-06-07 01:33:16 +02:00 · 2023-06-04 18:35:46 +02:00 · 2023-05-27 22:38:20 -03:00 · 2023-05-26 18:18:40 +02:00 · 2023-05-26 14:45:28 +02:00
2479 changed files with 45838 additions and 148702 deletions
--- a/.gitignore
+++ b/.gitignore
@@ -36,4 +36,3 @@
 /lcov/
 /src
 /mapfile
-/tools/python/__pycache__/
--- a/.mailmap
+++ b/.mailmap
@@ -1,21 +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.fu@intel.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>
-<zhong.li@intel.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>
--- a/.travis.yml
+++ b/.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:
--- a/4
+++ b/4
@@ -1,6 +1,6 @@
-See the Git history of the project (git://source.ffmpeg.org/ffmpeg) to
+See the Git history of the project (https://git.ffmpeg.org/ffmpeg) to
 get the names of people who have contributed to FFmpeg.

 To check the log, you can type the command "git log" in the FFmpeg
 source directory, or browse the online repository at
-http://source.ffmpeg.org.
+https://git.ffmpeg.org/ffmpeg
--- a/1736
+++ b/1736
--- a/INSTALL.md
+++ b/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`.
--- a/LICENSE.md
+++ b/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.
--- a/35
+++ b/35
@@ -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
@@ -52,9 +52,9 @@ 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
+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, Rodger Combs, wm4
@@ -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
@@ -145,7 +143,6 @@ Codecs:
  asv*                                  Michael Niedermayer
  atrac3plus*                           Maxim Poliakovski
  audiotoolbox*                         Rodger Combs
-  avs2*                                 Huiwen Ren
  bgmc.c, bgmc.h                        Thilo Borgmann
  binkaudio.c                           Peter Ross
  cavs*                                 Stefan Gehrer
@@ -170,6 +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
@@ -191,17 +189,14 @@ Codecs:
  libcelt_dec.c                         Nicolas George
  libcodec2.c                           Tomas Härdin
  libdirac*                             David Conrad
-  libdavs2.c                            Huiwen Ren
  libgsm.c                              Michel Bardiaux
  libkvazaar.c                          Arttu Ylä-Outinen
-  libopenh264enc.c                      Martin Storsjo, Linjie Fu
  libopenjpeg.c                         Jaikrishnan Menon
  libopenjpegenc.c                      Michael Bradshaw
  libtheoraenc.c                        David Conrad
  libvorbis.c                           David Conrad
  libvpx*                               James Zern
  libxavs.c                             Stefan Gehrer
-  libxavs2.c                            Huiwen Ren
  libzvbi-teletextdec.c                 Marton Balint
  lzo.h, lzo.c                          Reimar Doeffinger
  mdec.c                                Michael Niedermayer
@@ -217,7 +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
@@ -366,15 +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
 ===========

@@ -436,9 +427,9 @@ Muxers/Demuxers:
  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
@@ -453,7 +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
@@ -461,6 +452,7 @@ 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
  psxstr.c                              Mike Melanson
@@ -508,7 +500,6 @@ Protocols:
  ftp.c                                 Lukasz Marek
  http.c                                Ronald S. Bultje
  libssh.c                              Lukasz Marek
-  libzmq.c                              Andriy Gelman
  mms*.c                                Ronald S. Bultje
  udp.c                                 Luca Abeni
  icecast.c                             Marvin Scholz
@@ -535,7 +526,6 @@ Alpha                                   Falk Hueffner
 MIPS                                    Manojkumar Bhosale, Shiyou Yin
 Mac OS X / PowerPC                      Romain Dolbeau, Guillaume Poirier
 Amiga / PowerPC                         Colin Ward
-Linux / PowerPC                         Lauri Kasanen
 Windows MinGW                           Alex Beregszaszi, Ramiro Polla
 Windows Cygwin                          Victor Paesa
 Windows MSVC                            Matthew Oliver, Hendrik Leppkes
@@ -564,7 +554,6 @@ Joakim Plate
 Jun Zhao
 Kieran Kunhya
 Kirill Gavrilov
-Limin Wang
 Martin Storsjö
 Panagiotis Issaris
 Pedro Arthur
@@ -608,14 +597,13 @@ James Almer                   7751 2E8C FD94 A169 57E6 9A7A 1463 01AD 7376 59E0
 Jean Delvare                  7CA6 9F44 60F1 BDC4 1FD2 C858 A552 6B9B B3CD 4E6A
 Loren Merritt                 ABD9 08F4 C920 3F65 D8BE 35D7 1540 DAA7 060F 56DE
 Lou Logan (llogan)            7D68 DC73 CBEF EABB 671A B6CF 621C 2E28 82F8 DC3A
-Lynne                         FE50 139C 6805 72CA FD52 1F8D A2FE A5F0 3F03 4464
 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
 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
@@ -624,7 +612,6 @@ 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
--- a/11
+++ b/11
@@ -50,12 +50,6 @@ $(TOOLS): %$(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) $(LIBFUZZER_PATH)
-
-tools/target_dem_fuzzer$(EXESUF): tools/target_dem_fuzzer.o $(FF_DEP_LIBS)
-	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
-
 tools/sofa2wavs$(EXESUF): ELIBS = $(FF_EXTRALIBS)
 tools/uncoded_frame$(EXESUF): $(FF_DEP_LIBS)
 tools/uncoded_frame$(EXESUF): ELIBS = $(FF_EXTRALIBS)
@@ -141,7 +135,7 @@ 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

@@ -151,7 +145,6 @@ distclean:: clean
 		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
@@ -166,7 +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
--- a/2
+++ b/2
@@ -1 +1 @@
-4.3.2
+4.1.11
--- a/12
+++ b/12
@@ -1,15 +1,15 @@

-              ┌────────────────────────────────────┐
-              │ RELEASE NOTES for FFmpeg 4.3 "4:3" │
-              └────────────────────────────────────┘
+              ┌─────────────────────────────────────────────┐
+              │ RELEASE NOTES for FFmpeg 4.1 "al-Khwarizmi" │
+              └─────────────────────────────────────────────┘

-   The FFmpeg Project proudly presents FFmpeg 4.3 "4:3", about 10
-   months after the release of FFmpeg 4.2.
+   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.freenode.net) or ask
+   feel free to join us on the #ffmpeg IRC channel (on irc.libera.chat) or ask
   on the mailing-lists.
--- a/compat/avisynth/avisynth_c.h
+++ b/compat/avisynth/avisynth_c.h
--- a/compat/avisynth/avs/capi.h
+++ b/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
--- a/compat/avisynth/avs/config.h
+++ b/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
--- a/compat/avisynth/avs/types.h
+++ b/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
--- a/compat/avisynth/avxsynth_c.h
+++ b/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__
--- a/compat/avisynth/windowsPorts/basicDataTypeConversions.h
+++ b/compat/avisynth/windowsPorts/basicDataTypeConversions.h
@@ -0,0 +1,85 @@
+#ifndef __DATA_TYPE_CONVERSIONS_H__
+#define __DATA_TYPE_CONVERSIONS_H__
+
+#include <stdint.h>
+#include <wchar.h>
+
+#ifdef __cplusplus
+namespace avxsynth {
+#endif // __cplusplus
+
+typedef int64_t __int64;
+typedef int32_t __int32;
+#ifdef __cplusplus
+typedef bool BOOL;
+#else
+typedef uint32_t BOOL;
+#endif // __cplusplus
+typedef void* HMODULE;
+typedef void* LPVOID;
+typedef void* PVOID;
+typedef PVOID HANDLE;
+typedef HANDLE HWND;
+typedef HANDLE HINSTANCE;
+typedef void* HDC;
+typedef void* HBITMAP;
+typedef void* HICON;
+typedef void* HFONT;
+typedef void* HGDIOBJ;
+typedef void* HBRUSH;
+typedef void* HMMIO;
+typedef void* HACMSTREAM;
+typedef void* HACMDRIVER;
+typedef void* HIC;
+typedef void* HACMOBJ;
+typedef HACMSTREAM* LPHACMSTREAM;
+typedef void* HACMDRIVERID;
+typedef void* LPHACMDRIVER;
+typedef unsigned char BYTE;
+typedef BYTE* LPBYTE;
+typedef char TCHAR;
+typedef TCHAR* LPTSTR;
+typedef const TCHAR* LPCTSTR;
+typedef char* LPSTR;
+typedef LPSTR LPOLESTR;
+typedef const char* LPCSTR;
+typedef LPCSTR LPCOLESTR;
+typedef wchar_t WCHAR;
+typedef unsigned short WORD;
+typedef unsigned int UINT;
+typedef UINT MMRESULT;
+typedef uint32_t DWORD;
+typedef DWORD COLORREF;
+typedef DWORD FOURCC;
+typedef DWORD HRESULT;
+typedef DWORD* LPDWORD;
+typedef DWORD* DWORD_PTR;
+typedef int32_t LONG;
+typedef int32_t* LONG_PTR;
+typedef LONG_PTR LRESULT;
+typedef uint32_t ULONG;
+typedef uint32_t* ULONG_PTR;
+//typedef __int64_t intptr_t;
+typedef uint64_t _fsize_t;
+
+
+//
+// Structures
+//
+
+typedef struct _GUID {
+  DWORD Data1;
+  WORD  Data2;
+  WORD  Data3;
+  BYTE  Data4[8];
+} GUID;
+
+typedef GUID REFIID;
+typedef GUID CLSID;
+typedef CLSID* LPCLSID;
+typedef GUID IID;
+
+#ifdef __cplusplus
+}; // namespace avxsynth
+#endif // __cplusplus
+#endif //  __DATA_TYPE_CONVERSIONS_H__
--- a/compat/avisynth/windowsPorts/windows2linux.h
+++ b/compat/avisynth/windowsPorts/windows2linux.h
@@ -0,0 +1,77 @@
+#ifndef __WINDOWS2LINUX_H__
+#define __WINDOWS2LINUX_H__
+
+/*
+ * LINUX SPECIFIC DEFINITIONS
+*/
+//
+// Data types conversions
+//
+#include <stdlib.h>
+#include <string.h>
+#include "basicDataTypeConversions.h"
+
+#ifdef __cplusplus
+namespace avxsynth {
+#endif // __cplusplus
+//
+// purposefully define the following MSFT definitions
+// to mean nothing (as they do not mean anything on Linux)
+//
+#define __stdcall
+#define __cdecl
+#define noreturn
+#define __declspec(x)
+#define STDAPI       extern "C" HRESULT
+#define STDMETHODIMP HRESULT __stdcall
+#define STDMETHODIMP_(x) x __stdcall
+
+#define STDMETHOD(x)    virtual HRESULT x
+#define STDMETHOD_(a, x) virtual a x
+
+#ifndef TRUE
+#define TRUE  true
+#endif
+
+#ifndef FALSE
+#define FALSE false
+#endif
+
+#define S_OK                (0x00000000)
+#define S_FALSE             (0x00000001)
+#define E_NOINTERFACE       (0X80004002)
+#define E_POINTER           (0x80004003)
+#define E_FAIL              (0x80004005)
+#define E_OUTOFMEMORY       (0x8007000E)
+
+#define INVALID_HANDLE_VALUE    ((HANDLE)((LONG_PTR)-1))
+#define FAILED(hr)              ((hr) & 0x80000000)
+#define SUCCEEDED(hr)           (!FAILED(hr))
+
+
+//
+// Functions
+//
+#define MAKEDWORD(a,b,c,d) (((a) << 24) | ((b) << 16) | ((c) << 8) | (d))
+#define MAKEWORD(a,b) (((a) << 8) | (b))
+
+#define lstrlen                             strlen
+#define lstrcpy                             strcpy
+#define lstrcmpi                            strcasecmp
+#define _stricmp                            strcasecmp
+#define InterlockedIncrement(x)             __sync_fetch_and_add((x), 1)
+#define InterlockedDecrement(x)             __sync_fetch_and_sub((x), 1)
+// Windows uses (new, old) ordering but GCC has (old, new)
+#define InterlockedCompareExchange(x,y,z)   __sync_val_compare_and_swap(x,z,y)
+
+#define UInt32x32To64(a, b)                 ( (uint64_t) ( ((uint64_t)((uint32_t)(a))) * ((uint32_t)(b))  ) )
+#define Int64ShrlMod32(a, b)                ( (uint64_t) ( (uint64_t)(a) >> (b) ) )
+#define Int32x32To64(a, b)                  ((__int64)(((__int64)((long)(a))) * ((long)(b))))
+
+#define MulDiv(nNumber, nNumerator, nDenominator)   (int32_t) (((int64_t) (nNumber) * (int64_t) (nNumerator) + (int64_t) ((nDenominator)/2)) / (int64_t) (nDenominator))
+
+#ifdef __cplusplus
+}; // namespace avxsynth
+#endif // __cplusplus
+
+#endif //  __WINDOWS2LINUX_H__
--- a/compat/cuda/cuda_runtime.h
+++ b/compat/cuda/cuda_runtime.h
@@ -1,131 +0,0 @@
-/*
- * Minimum CUDA compatibility definitions header
- *
- * Copyright (c) 2019 Rodger Combs
- *
- * 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__ uint3
-{
-    unsigned int x, y, z;
-} uint3;
-
-typedef struct uint3 dim3;
-
-typedef struct __device_builtin__ __align__(8) int2
-{
-    int x, y;
-} int2;
-
-typedef struct __device_builtin__ __align__(4) uchar4
-{
-    unsigned char x, y, z, w;
-} uchar4;
-
-typedef struct __device_builtin__ __align__(8) ushort4
-{
-    unsigned char x, y, z, w;
-} ushort4;
-
-typedef struct __device_builtin__ __align__(16) int4
-{
-    int x, y, z, w;
-} int4;
-
-// 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_uchar2(a, b) ((uchar2){.x = a, .y = b})
-#define make_ushort2(a, b) ((ushort2){.x = a, .y = b})
-#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})
-
-// 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(uchar2, make_uchar2(a & 0xFF, b & 0xFF))
-TEX2D(ushort2, make_ushort2(a & 0xFFFF, b & 0xFFFF))
-TEX2D(uchar4, make_uchar4(a & 0xFF, b & 0xFF, c & 0xFF, d & 0xFF))
-TEX2D(ushort4, make_ushort4(a & 0xFFFF, b & 0xFFFF, c & 0xFFFF, d & 0xFFFF))
-
-// Template calling tex instruction and converting the output to the selected type
-template <class T>
-static 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;
-}
-
-#endif /* COMPAT_CUDA_CUDA_RUNTIME_H */
--- a/compat/cuda/dynlink_loader.h
+++ b/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
--- a/compat/cuda/ptx2c.sh
+++ b/compat/cuda/ptx2c.sh
@@ -27,8 +27,10 @@ IN="$2"
 NAME="$(basename "$IN" | sed 's/\..*//')"

 printf "const char %s_ptx[] = \\" "$NAME" > "$OUT"
-echo >> "$OUT"
-sed -e "$(printf 's/\r//g')" -e 's/["\\]/\\&/g' -e "$(printf 's/^/\t"/')" -e 's/$/\\n"/' < "$IN" >> "$OUT"
-echo ";" >> "$OUT"
+while read LINE
+do
+    printf "\n\t\"%s\\\n\"" "$(printf "%s" "$LINE" | sed -e 's/\r//g' -e 's/["\\]/\\&/g')" >> "$OUT"
+done < "$IN"
+printf ";\n" >> "$OUT"

 exit 0
--- a/compat/djgpp/math.c
+++ b/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);
-}
--- a/compat/djgpp/math.h
+++ b/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);
--- a/compat/os2threads.h
+++ b/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)
 {
--- a/compat/w32pthreads.h
+++ b/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 */
--- a/compat/windows/makedef
+++ b/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)
--- a/compat/windows/mslink
+++ b/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 $?
--- a/544
+++ b/544
--- a/doc/APIchanges
+++ b/doc/APIchanges
@@ -15,151 +15,6 @@ libavutil:     2017-10-21

 API changes, most recent first:

-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
--- a/doc/Doxyfile
+++ b/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         = 4.3.2
+PROJECT_NUMBER         = 4.1.11

 # 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
--- a/doc/authors.texi
+++ b/doc/authors.texi
@@ -3,9 +3,9 @@
 The FFmpeg developers.

 For details about the authorship, see the Git history of the project
-(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
+(https://git.ffmpeg.org/ffmpeg), e.g. by typing the command
@command{git log} in the FFmpeg source directory, or browsing the
-online repository at @url{http://source.ffmpeg.org}.
+online repository at @url{https://git.ffmpeg.org/ffmpeg}.

 Maintainers for the specific components are listed in the file
@file{MAINTAINERS} in the source code tree.
--- a/doc/bitstream_filters.texi
+++ b/doc/bitstream_filters.texi
@@ -87,9 +87,6 @@ the timing info in the sequence header.
 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
@@ -224,10 +219,6 @@ Insert or remove AUD NAL units in all access units of the stream.
@item sample_aspect_ratio
 Set the sample aspect ratio of the stream in the VUI parameters.

-@item overscan_appropriate_flag
-Set whether the stream is suitable for display using overscan
-or not (see H.264 section E.2.1).
-
@item video_format
@item video_full_range_flag
 Set the video format in the stream (see H.264 section E.2.1 and
@@ -367,15 +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
@@ -548,111 +530,6 @@ ffmpeg -i INPUT -c copy -bsf noise[=1] output.mkv
@section null
 This bitstream filter passes the packets through unchanged.

-@section pcm_rechunk
-
-Repacketize PCM audio to a fixed number of samples per packet or a fixed packet
-rate per second. This is similar to the @ref{asetnsamples,,asetnsamples audio
-filter,ffmpeg-filters} but works on audio packets instead of audio frames.
-
-@table @option
-@item nb_out_samples, n
-Set the number of samples per each output audio packet. The number is intended
-as the number of samples @emph{per each channel}. Default value is 1024.
-
-@item pad, p
-If set to 1, the filter will pad the last audio packet with silence, so that it
-will contain the same number of samples (or roughly the same number of samples,
-see @option{frame_rate}) as the previous ones. Default value is 1.
-
-@item frame_rate, r
-This option makes the filter output a fixed number of packets per second instead
-of a fixed number of samples per packet. If the audio sample rate is not
-divisible by the frame rate then the number of samples will not be constant but
-will vary slightly so that each packet will start as close to the frame
-boundary as possible. Using this option has precedence over @option{nb_out_samples}.
-@end table
-
-You can generate the well known 1602-1601-1602-1601-1602 pattern of 48kHz audio
-for NTSC frame rate using the @option{frame_rate} option.
-@example
-ffmpeg -f lavfi -i sine=r=48000:d=1 -c pcm_s16le -bsf pcm_rechunk=r=30000/1001 -f framecrc -
-@end example
-
-@section prores_metadata
-
-Modify color property metadata embedded in prores stream.
-
-@table @option
-@item color_primaries
-Set the color primaries.
-Available values are:
-
-@table @samp
-@item auto
-Keep the same color primaries property (default).
-
-@item unknown
-@item bt709
-@item bt470bg
-BT601 625
-
-@item smpte170m
-BT601 525
-
-@item bt2020
-@item smpte431
-DCI P3
-
-@item smpte432
-P3 D65
-
-@end table
-
-@item transfer_characteristics
-Set the color transfer.
-Available values are:
-
-@table @samp
-@item auto
-Keep the same transfer characteristics property (default).
-
-@item unknown
-@item bt709
-BT 601, BT 709, BT 2020
-@item smpte2084
-SMPTE ST 2084
-@item arib-std-b67
-ARIB STD-B67
-@end table
-
-
-@item matrix_coefficients
-Set the matrix coefficient.
-Available values are:
-
-@table @samp
-@item auto
-Keep the same colorspace property (default).
-
-@item unknown
-@item bt709
-@item smpte170m
-BT 601
-
-@item bt2020nc
-@end table
-@end table
-
-Set Rec709 colorspace for each frame of the file
-@example
-ffmpeg -i INPUT -c copy -bsf:v prores_metadata=color_primaries=bt709:color_trc=bt709:colorspace=bt709 output.mov
-@end example
-
-Set Hybrid Log-Gamma parameters for each frame of the file
-@example
-ffmpeg -i INPUT -c copy -bsf:v prores_metadata=color_primaries=bt2020:color_trc=arib-std-b67:colorspace=bt2020nc output.mov
-@end example
-
@section remove_extra

 Remove extradata from packets.
@@ -689,12 +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

@@ -702,9 +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
@@ -716,8 +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
--- a/doc/build_system.txt
+++ b/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>
--- a/doc/codecs.texi
+++ b/doc/codecs.texi
@@ -55,10 +55,6 @@ 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.
@@ -80,8 +76,6 @@ Deprecated, use mpegvideo private options instead.
 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})
@@ -646,24 +640,6 @@ 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.
@@ -781,12 +757,14 @@ Set noise reduction.
 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
@@ -797,25 +775,11 @@ 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})
-
-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.
@end table

@item error @var{integer} (@emph{encoding,video})
@@ -855,8 +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})

@@ -957,9 +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
@@ -1230,7 +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
--- a/doc/decoders.texi
+++ b/doc/decoders.texi
@@ -47,39 +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).
-
-@item tilethreads
-Set amount of tile threads to use during decoding. The default value is 0 (autodetect).
-
-@item filmgrain
-Apply film grain to the decoded video if present in the bitstream. Defaults to the
-internal default of the library.
-
-@item oppoint
-Select an operating point of a scalable AV1 bitstream (0 - 31). Defaults to the
-internal default of the library.
-
-@item alllayers
-Output all spatial layers of a scalable AV1 bitstream. The default value is false.
-
-@end table
-
@section libdavs2

 AVS2-P2/IEEE1857.4 video decoder wrapper.
@@ -227,31 +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
@@ -287,7 +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}.

@@ -316,11 +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
--- a/doc/demuxers.texi
+++ b/doc/demuxers.texi
@@ -25,6 +25,17 @@ Audible Format 2, 3, and 4 demuxer.

 This demuxer is used to demux Audible Format 2, 3, and 4 (.aa) files.

+@section applehttp
+
+Apple HTTP Live Streaming demuxer.
+
+This demuxer presents all AVStreams from all variant streams.
+The id field is set to the bitrate variant index number. By setting
+the discard flags on AVStreams (by pressing 'a' or 'v' in ffplay),
+the caller can decide which variant streams to actually receive.
+The total bitrate of the variant that the stream belongs to is
+available in a metadata key named "variant_bitrate".
+
@section apng

 Animated Portable Network Graphics demuxer.
@@ -309,15 +320,6 @@ 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
@@ -331,10 +333,6 @@ segment index to start live streams at (negative values are from the end).
 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.
@@ -342,10 +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.
@end table

@section image2
@@ -456,17 +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
@@ -498,84 +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

@@ -604,13 +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
@@ -621,73 +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 export_all
-Export unrecognized boxes within the @var{udta} box as metadata entries. The first four
-characters of the box type are set as the key. Default is false.
-
-@item export_xmp
-Export entire contents of @var{XMP_} box and @var{uuid} box as a string with key @code{xmp}. Note that
-if @code{export_all} is set and this option isn't, the contents of @var{XMP_} box are still exported
-but with key @code{XMP_}. Default is false.
-
-@item activation_bytes
-4-byte key required to decrypt Audible AAX and AAX+ files. See Audible AAX subsection below.
-
-@item audible_fixed_key
-Fixed key used for handling Audible AAX/AAX+ files. It has been pre-set so should not be necessary to
-specify.
-
-@item decryption_key
-16-byte key, in hex, to decrypt files encrypted using ISO Common Encryption (CENC/AES-128 CTR; ISO/IEC 23001-7).
-@end table
-
-@subsection Audible AAX
-
-Audible AAX files are encrypted M4B files, and they can be decrypted by specifying a 4 byte activation secret.
-@example
-ffmpeg -activation_bytes 1CEB00DA -i test.aax -vn -c:a copy output.mp4
-@end example
-
@section mpegts

 MPEG-2 transport stream demuxer.
@@ -816,20 +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
--- a/doc/developer.texi
+++ b/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
@@ -625,7 +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?
--- a/doc/encoders.texi
+++ b/doc/encoders.texi
@@ -30,7 +30,11 @@ follows.

 Advanced Audio Coding (AAC) encoder.

-This encoder is the default AAC encoder, natively implemented into FFmpeg.
+This encoder is the default AAC encoder, natively implemented into FFmpeg. Its
+quality is on par or better than libfdk_aac at the default bitrate of 128kbps.
+This encoder also implements more options, profiles and samplerates than
+other encoders (with only the AAC-HE profile pending to be implemented) so this
+encoder has become the default and is the recommended choice.

@subsection Options

@@ -647,7 +651,10 @@ configuration. You need to explicitly configure the build with
 so if you allow the use of GPL, you should configure with
@code{--enable-gpl --enable-nonfree --enable-libfdk-aac}.

-This encoder has support for the AAC-HE profiles.
+This encoder is considered to produce output on par or worse at 128kbps to the
+@ref{aacenc,,the native FFmpeg AAC encoder} but can often produce better
+sounding audio at identical or lower bitrates and has support for the
+AAC-HE profiles.

 VBR encoding, enabled through the @option{vbr} or @option{flags
 +qscale} options, is experimental and only works with some
@@ -726,14 +733,6 @@ if set to 0.

 Default value is 0.

-@item eld_v2
-Enable ELDv2 (LD-MPS extension for ELD stereo signals) for ELDv2 if set to 1,
-disabled if set to 0.
-
-Note that option is available when fdk-aac version (AACENCODER_LIB_VL0.AACENCODER_LIB_VL1.AACENCODER_LIB_VL2) > (4.0.0).
-
-Default value is 0.
-
@item signaling
 Set SBR/PS signaling style.

@@ -1371,236 +1370,6 @@ makes it possible to store non-rgb pix_fmts.

@end table

-@section librav1e
-
-rav1e AV1 encoder wrapper.
-
-Requires the presence of the rav1e headers and library during configuration.
-You need to explicitly configure the build with @code{--enable-librav1e}.
-
-@subsection Options
-
-@table @option
-@item qmax
-Sets the maximum quantizer to use when using bitrate mode.
-
-@item qmin
-Sets the minimum quantizer to use when using bitrate mode.
-
-@item qp
-Uses quantizer mode to encode at the given quantizer (0-255).
-
-@item speed
-Selects the speed preset (0-10) to encode with.
-
-@item tiles
-Selects how many tiles to encode with.
-
-@item tile-rows
-Selects how many rows of tiles to encode with.
-
-@item tile-columns
-Selects how many columns of tiles to encode with.
-
-@item rav1e-params
-Set rav1e options using a list of @var{key}=@var{value} pairs separated
-by ":". See @command{rav1e --help} for a list of options.
-
-For example to specify librav1e encoding options with @option{-rav1e-params}:
-
-@example
-ffmpeg -i input -c:v librav1e -b:v 500K -rav1e-params speed=5:low_latency=true output.mp4
-@end example
-
-@end table
-
-@section libaom-av1
-
-libaom AV1 encoder wrapper.
-
-Requires the presence of the libaom headers and library during
-configuration.  You need to explicitly configure the build with
-@code{--enable-libaom}.
-
-@subsection Options
-
-The wrapper supports the following standard libavcodec options:
-
-@table @option
-
-@item b
-Set bitrate target in bits/second.  By default this will use
-variable-bitrate mode.  If @option{maxrate} and @option{minrate} are
-also set to the same value then it will use constant-bitrate mode,
-otherwise if @option{crf} is set as well then it will use
-constrained-quality mode.
-
-@item g keyint_min
-Set key frame placement.  The GOP size sets the maximum distance between
-key frames; if zero the output stream will be intra-only.  The minimum
-distance is ignored unless it is the same as the GOP size, in which case
-key frames will always appear at a fixed interval.  Not set by default,
-so without this option the library has completely free choice about
-where to place key frames.
-
-@item qmin qmax
-Set minimum/maximum quantisation values.  Valid range is from 0 to 63
-(warning: this does not match the quantiser values actually used by AV1
- divide by four to map real quantiser values to this range).  Defaults
-to min/max (no constraint).
-
-@item minrate maxrate bufsize rc_init_occupancy
-Set rate control buffering parameters.  Not used if not set - defaults
-to unconstrained variable bitrate.
-
-@item threads
-Set the number of threads to use while encoding.  This may require the
-@option{tiles} or @option{row-mt} options to also be set to actually
-use the specified number of threads fully. Defaults to the number of
-hardware threads supported by the host machine.
-
-@item profile
-Set the encoding profile.  Defaults to using the profile which matches
-the bit depth and chroma subsampling of the input.
-
-@end table
-
-The wrapper also has some specific options:
-
-@table @option
-
-@item cpu-used
-Set the quality/encoding speed tradeoff.  Valid range is from 0 to 8,
-higher numbers indicating greater speed and lower quality.  The default
-value is 1, which will be slow and high quality.
-
-@item auto-alt-ref
-Enable use of alternate reference frames.  Defaults to the internal
-default of the library.
-
-@item arnr-max-frames (@emph{frames})
-Set altref noise reduction max frame count. Default is -1.
-
-@item arnr-strength (@emph{strength})
-Set altref noise reduction filter strength. Range is -1 to 6. Default is -1.
-
-@item aq-mode (@emph{aq-mode})
-Set adaptive quantization mode. Possible values:
-
-@table @samp
-@item none (@emph{0})
-Disabled.
-
-@item variance (@emph{1})
-Variance-based.
-
-@item complexity (@emph{2})
-Complexity-based.
-
-@item cyclic (@emph{3})
-Cyclic refresh.
-@end table
-
-@item tune (@emph{tune})
-Set the distortion metric the encoder is tuned with. Default is @code{psnr}.
-
-@table @samp
-@item psnr (@emph{0})
-
-@item ssim (@emph{1})
-@end table
-
-@item lag-in-frames
-Set the maximum number of frames which the encoder may keep in flight
-at any one time for lookahead purposes.  Defaults to the internal
-default of the library.
-
-@item error-resilience
-Enable error resilience features:
-@table @option
-@item default
-Improve resilience against losses of whole frames.
-@end table
-Not enabled by default.
-
-@item crf
-Set the quality/size tradeoff for constant-quality (no bitrate target)
-and constrained-quality (with maximum bitrate target) modes. Valid
-range is 0 to 63, higher numbers indicating lower quality and smaller
-output size.  Only used if set; by default only the bitrate target is
-used.
-
-@item static-thresh
-Set a change threshold on blocks below which they will be skipped by
-the encoder.  Defined in arbitrary units as a nonnegative integer,
-defaulting to zero (no blocks are skipped).
-
-@item drop-threshold
-Set a threshold for dropping frames when close to rate control bounds.
-Defined as a percentage of the target buffer - when the rate control
-buffer falls below this percentage, frames will be dropped until it
-has refilled above the threshold.  Defaults to zero (no frames are
-dropped).
-
-@item denoise-noise-level (@emph{level})
-Amount of noise to be removed for grain synthesis. Grain synthesis is disabled if
-this option is not set or set to 0.
-
-@item denoise-block-size (@emph{pixels})
-Block size used for denoising for grain synthesis. If not set, AV1 codec
-uses the default value of 32.
-
-@item undershoot-pct (@emph{pct})
-Set datarate undershoot (min) percentage of the target bitrate. Range is -1 to 100.
-Default is -1.
-
-@item overshoot-pct (@emph{pct})
-Set datarate overshoot (max) percentage of the target bitrate. Range is -1 to 1000.
-Default is -1.
-
-@item minsection-pct (@emph{pct})
-Minimum percentage variation of the GOP bitrate from the target bitrate. If minsection-pct
-is not set, the libaomenc wrapper computes it as follows: @code{(minrate * 100 / bitrate)}.
-Range is -1 to 100. Default is -1 (unset).
-
-@item maxsection-pct (@emph{pct})
-Maximum percentage variation of the GOP bitrate from the target bitrate. If maxsection-pct
-is not set, the libaomenc wrapper computes it as follows: @code{(maxrate * 100 / bitrate)}.
-Range is -1 to 5000. Default is -1 (unset).
-
-@item frame-parallel (@emph{boolean})
-Enable frame parallel decodability features. Default is true.
-
-@item tiles
-Set the number of tiles to encode the input video with, as columns x
-rows.  Larger numbers allow greater parallelism in both encoding and
-decoding, but may decrease coding efficiency.  Defaults to the minimum
-number of tiles required by the size of the input video (this is 1x1
-(that is, a single tile) for sizes up to and including 4K).
-
-@item tile-columns tile-rows
-Set the number of tiles as log2 of the number of tile rows and columns.
-Provided for compatibility with libvpx/VP9.
-
-@item row-mt (Requires libaom >= 1.0.0-759-g90a15f4f2)
-Enable row based multi-threading. Disabled by default.
-
-@item enable-cdef (@emph{boolean})
-Enable Constrained Directional Enhancement Filter. The libaom-av1
-encoder enables CDEF by default.
-
-@item enable-restoration (@emph{boolean})
-Enable Loop Restoration Filter. Default is true for libaom-av1.
-
-@item enable-global-motion (@emph{boolean})
-Enable the use of global motion for block prediction. Default is true.
-
-@item enable-intrabc (@emph{boolean})
-Enable block copy mode for intra block prediction. This mode is
-useful for screen content. Default is true.
-
-@end table
-
@section libkvazaar

 Kvazaar H.265/HEVC encoder.
@@ -1872,8 +1641,7 @@ means unlimited.
@table @option
@item auto-alt-ref
 Enable use of alternate reference frames (2-pass only).
-Values greater than 1 enable multi-layer alternate reference frames (VP9 only).
-@item arnr-maxframes
+@item arnr-max-frames
 Set altref noise reduction max frame count.
@item arnr-type
 Set altref noise reduction filter type: backward, forward, centered.
@@ -1886,61 +1654,6 @@ Set number of frames to look ahead for frametype and ratecontrol.
@item error-resilient
 Enable error resiliency features.

-@item sharpness @var{integer}
-Increase sharpness at the expense of lower PSNR.
-The valid range is [0, 7].
-
-@item ts-parameters
-Sets the temporal scalability configuration using a :-separated list of
-key=value pairs. For example, to specify temporal scalability parameters
-with @code{ffmpeg}:
-@example
-ffmpeg -i INPUT -c:v libvpx -ts-parameters ts_number_layers=3:\
-ts_target_bitrate=250,500,1000:ts_rate_decimator=4,2,1:\
-ts_periodicity=4:ts_layer_id=0,2,1,2:ts_layering_mode=3 OUTPUT
-@end example
-Below is a brief explanation of each of the parameters, please
-refer to @code{struct vpx_codec_enc_cfg} in @code{vpx/vpx_encoder.h} for more
-details.
-@table @option
-@item ts_number_layers
-Number of temporal coding layers.
-@item ts_target_bitrate
-Target bitrate for each temporal layer (in kbps).
-(bitrate should be inclusive of the lower temporal layer).
-@item ts_rate_decimator
-Frame rate decimation factor for each temporal layer.
-@item ts_periodicity
-Length of the sequence defining frame temporal layer membership.
-@item ts_layer_id
-Template defining the membership of frames to temporal layers.
-@item ts_layering_mode
-(optional) Selecting the temporal structure from a set of pre-defined temporal layering modes.
-Currently supports the following options.
-@table @option
-@item 0
-No temporal layering flags are provided internally,
-relies on flags being passed in using @code{metadata} field in @code{AVFrame}
-with following keys.
-@table @option
-@item vp8-flags
-Sets the flags passed into the encoder to indicate the referencing scheme for
-the current frame.
-Refer to function @code{vpx_codec_encode} in @code{vpx/vpx_encoder.h} for more
-details.
-@item temporal_id
-Explicitly sets the temporal id of the current frame to encode.
-@end table
-@item 2
-Two temporal layers. 0-1...
-@item 3
-Three temporal layers. 0-2-1-2...; with single reference frame.
-@item 4
-Same as option "3", except there is a dependency between
-the two temporal layer 2 frames within the temporal period.
-@end table
-@end table
-
@item VP9-specific options
@table @option
@item lossless
@@ -1979,8 +1692,6 @@ Corpus VBR mode is a variant of standard VBR where the complexity distribution
 midpoint is passed in rather than calculated for a specific clip or chunk.

 The valid range is [0, 10000]. 0 (default) uses standard VBR.
-@item enable-tpl @var{boolean}
-Enable temporal dependency model.
@end table

@end table
@@ -2442,20 +2153,6 @@ during configuration. You need to explicitly configure the build with
@subsection Options

@table @option
-@item b
-Sets target video bitrate.
-
-@item bf
-
-@item g
-Set the GOP size.
-
-@item keyint_min
-Minimum GOP size.
-
-@item refs
-Number of reference frames each P-frame can use. The range is from @var{1-16}.
-
@item preset
 Set the x265 preset.

@@ -2468,28 +2165,6 @@ Set profile restrictions.
@item crf
 Set the quality for constant quality mode.

-@item qp
-Set constant quantization rate control method parameter.
-
-@item qmin
-Minimum quantizer scale.
-
-@item qmax
-Maximum quantizer scale.
-
-@item qdiff
-Maximum difference between quantizer scales.
-
-@item qblur
-Quantizer curve blur
-
-@item qcomp
-Quantizer curve compression factor
-
-@item i_qfactor
-
-@item b_qfactor
-
@item forced-idr
 Normally, when forcing a I-frame type, the encoder can select any type
 of I-frame. This option forces it to choose an IDR-frame.
@@ -2505,63 +2180,6 @@ ffmpeg -i input -c:v libx265 -x265-params crf=26:psy-rd=1 output.mp4
@end example
@end table

-@section libxavs2
-
-xavs2 AVS2-P2/IEEE1857.4 encoder wrapper.
-
-This encoder requires the presence of the libxavs2 headers and library
-during configuration. You need to explicitly configure the build with
-@option{--enable-libxavs2}.
-
-The following standard libavcodec options are used:
-@itemize
-@item
-@option{b} / @option{bit_rate}
-@item
-@option{g} / @option{gop_size}
-@item
-@option{bf} / @option{max_b_frames}
-@end itemize
-
-The encoder also has its own specific options:
-@subsection Options
-
-@table @option
-@item lcu_row_threads
-Set the number of parallel threads for rows from 1 to 8 (default 5).
-
-@item initial_qp
-Set the xavs2 quantization parameter from 1 to 63 (default 34). This is
-used to set the initial qp for the first frame.
-
-@item qp
-Set the xavs2 quantization parameter from 1 to 63 (default 34). This is
-used to set the qp value under constant-QP mode.
-
-@item max_qp
-Set the max qp for rate control from 1 to 63 (default 55).
-
-@item min_qp
-Set the min qp for rate control from 1 to 63 (default 20).
-
-@item speed_level
-Set the Speed level from 0 to 9 (default 0). Higher is better but slower.
-
-@item log_level
-Set the log level from -1 to 3 (default 0). -1: none, 0: error,
-1: warning, 2: info, 3: debug.
-
-@item xavs2-params
-Set xavs2 options using a list of @var{key}=@var{value} couples separated
-by ":".
-
-For example to specify libxavs2 encoding options with @option{-xavs2-params}:
-
-@example
-ffmpeg -i input -c:v libxavs2 -xavs2-params RdoqLevel=0 output.avs2
-@end example
-@end table
-
@section libxvid

 Xvid MPEG-4 Part 2 encoder wrapper.
@@ -2725,14 +2343,6 @@ fastest.

@end table

-@section MediaFoundation
-
-This provides wrappers to encoders (both audio and video) in the
-MediaFoundation framework. It can access both SW and HW encoders.
-Video encoders can take input in either of nv12 or yuv420p form
-(some encoders support both, some support only either - in practice,
-nv12 is the safer choice, especially among HW encoders).
-
@section mpeg2

 MPEG-2 video encoder.
@@ -2740,20 +2350,6 @@ MPEG-2 video encoder.
@subsection Options

@table @option
-@item profile @var{integer}
-Select the mpeg2 profile to encode:
-
-@table @samp
-@item 422
-@item main
-@item ss
-Spatially Scalable
-@item snr
-SNR Scalable
-@item high
-@item simple
-@end table
-
@item seq_disp_ext @var{integer}
 Specifies if the encoder should write a sequence_display_extension to the
 output.
@@ -2774,9 +2370,6 @@ Specifies the video_format written into the sequence display extension
 indicating the source of the video pictures. The default is @samp{unspecified},
 can be @samp{component}, @samp{pal}, @samp{ntsc}, @samp{secam} or @samp{mac}.
 For maximum compatibility, use @samp{component}.
-@item a53cc @var{boolean}
-Import closed captions (which must be ATSC compatible format) into output.
-Default is 1 (on).
@end table

@section png
@@ -2862,7 +2455,7 @@ recommended value) and do not set a size constraint.

@section QSV encoders

-The family of Intel QuickSync Video encoders (MPEG-2, H.264, HEVC, JPEG/MJPEG and VP9)
+The family of Intel QuickSync Video encoders (MPEG-2, H.264 and HEVC)

 The ratecontrol method is selected as follows:

@@ -3010,48 +2603,15 @@ Size / quality tradeoff: higher values are smaller / worse quality.
@end itemize

 All encoders support the following options:
-@table @option
-@item low_power
+@itemize
+@item
+@option{low_power}
+
 Some drivers/platforms offer a second encoder for some codecs intended to use
 less power than the default encoder; setting this option will attempt to use
 that encoder.  Note that it may support a reduced feature set, so some other
 options may not be available in this mode.
-
-@item idr_interval
-Set the number of normal intra frames between full-refresh (IDR) frames in
-open-GOP mode.  The intra frames are still IRAPs, but will not include global
-headers and may have non-decodable leading pictures.
-
-@item b_depth
-Set the B-frame reference depth.  When set to one (the default), all B-frames
-will refer only to P- or I-frames.  When set to greater values multiple layers
-of B-frames will be present, frames in each layer only referring to frames in
-higher layers.
-
-@item rc_mode
-Set the rate control mode to use.  A given driver may only support a subset of
-modes.
-
-Possible modes:
-@table @option
-@item auto
-Choose the mode automatically based on driver support and the other options.
-This is the default.
-@item CQP
-Constant-quality.
-@item CBR
-Constant-bitrate.
-@item VBR
-Variable-bitrate.
-@item ICQ
-Intelligent constant-quality.
-@item QVBR
-Quality-defined variable-bitrate.
-@item AVBR
-Average variable bitrate.
-@end table
-
-@end table
+@end itemize

 Each encoder also has its own specific options:
@table @option
@@ -3231,6 +2791,52 @@ Reduces detail but attempts to preserve color at extremely low bitrates.

@end table

+@section libxavs2
+
+xavs2 AVS2-P2/IEEE1857.4 encoder wrapper.
+
+This encoder requires the presence of the libxavs2 headers and library
+during configuration. You need to explicitly configure the build with
+@option{--enable-libxavs2}.
+
+@subsection Options
+
+@table @option
+@item lcu_row_threads
+Set the number of parallel threads for rows from 1 to 8 (default 5).
+
+@item initial_qp
+Set the xavs2 quantization parameter from 1 to 63 (default 34). This is
+used to set the initial qp for the first frame.
+
+@item qp
+Set the xavs2 quantization parameter from 1 to 63 (default 34). This is
+used to set the qp value under constant-QP mode.
+
+@item max_qp
+Set the max qp for rate control from 1 to 63 (default 55).
+
+@item min_qp
+Set the min qp for rate control from 1 to 63 (default 20).
+
+@item speed_level
+Set the Speed level from 0 to 9 (default 0). Higher is better but slower.
+
+@item log_level
+Set the log level from -1 to 3 (default 0). -1: none, 0: error,
+1: warning, 2: info, 3: debug.
+
+@item xavs2-params
+Set xavs2 options using a list of @var{key}=@var{value} couples separated
+by ":".
+
+For example to specify libxavs2 encoding options with @option{-xavs2-params}:
+
+@example
+ffmpeg -i input -c:v libxavs2 -xavs2-params preset_level=5 output.avs2
+@end example
+@end table
+
@c man end VIDEO ENCODERS

@chapter Subtitles Encoders
@@ -3245,14 +2851,6 @@ and they can also be used in Matroska files.
@subsection Options

@table @option
-@item palette
-Specify the global palette used by the bitmaps.
-
-The format for this option is a string containing 16 24-bits hexadecimal
-numbers (without 0x prefix) separated by commas, for example @code{0d00ee,
-ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1,
-7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b}.
-
@item even_rows_fix
 When set to 1, enable a work-around that makes the number of pixel rows
 even in all subtitles.  This fixes a problem with some players that
--- a/doc/examples/.gitignore
+++ b/doc/examples/.gitignore
@@ -1,4 +1,4 @@
-/avio_list_dir
+/avio_dir_cmd
 /avio_reading
 /decode_audio
 /decode_video
--- a/doc/examples/Makefile
+++ b/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)

--- a/doc/examples/Makefile.example
+++ b/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                       \
--- a/doc/examples/avio_list_dir.c
+++ b/doc/examples/avio_list_dir.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();

--- a/doc/examples/avio_reading.c
+++ b/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) {
--- a/doc/examples/decode_audio.c
+++ b/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)
 {
@@ -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->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);

--- a/doc/examples/decode_video.c
+++ b/doc/examples/decode_video.c
@@ -95,8 +95,7 @@ int main(int argc, char **argv)
    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];
--- a/doc/examples/demuxing_decoding.c
+++ b/doc/examples/demuxing_decoding.c
@@ -55,93 +55,95 @@ 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,
@@ -184,7 +186,8 @@ static int open_codec_context(int *stream_idx,
            return ret;
        }

-        /* Init the decoders */
+        /* 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));
@@ -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];
@@ -315,22 +325,23 @@ int main (int argc, char **argv)

    /* 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;
+        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");

--- a/doc/examples/encode_video.c
+++ b/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);
@@ -186,8 +186,7 @@ int main(int argc, char **argv)
    encode(c, NULL, pkt, f);

    /* add sequence end code to have a real MPEG file */
-    if (codec->id == AV_CODEC_ID_MPEG1VIDEO || codec->id == AV_CODEC_ID_MPEG2VIDEO)
-        fwrite(endcode, 1, sizeof(endcode), f);
+    fwrite(endcode, 1, sizeof(endcode), f);
    fclose(f);

    avcodec_free_context(&c);
--- a/doc/examples/metadata.c
+++ b/doc/examples/metadata.c
@@ -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);

--- a/doc/examples/muxing.c
+++ b/doc/examples/muxing.c
@@ -78,45 +78,15 @@ 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)
+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) {
-        AVPacket pkt = { 0 };
-
-        ret = avcodec_receive_packet(c, &pkt);
-        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
-            break;
-        else if (ret < 0) {
-            fprintf(stderr, "Error encoding a frame: %s\n", av_err2str(ret));
-            exit(1);
-        }
-
-        /* rescale output packet timestamp values from codec to stream timebase */
-        av_packet_rescale_ts(&pkt, c->time_base, st->time_base);
-        pkt.stream_index = st->index;
-
-        /* Write the compressed frame to the media file. */
-        log_packet(fmt_ctx, &pkt);
-        ret = av_interleaved_write_frame(fmt_ctx, &pkt);
-        av_packet_unref(&pkt);
-        if (ret < 0) {
-            fprintf(stderr, "Error while writing output packet: %s\n", av_err2str(ret));
-            exit(1);
-        }
-    }
-
-    return ret == AVERROR_EOF ? 1 : 0;
+    /* 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. */
@@ -315,7 +285,7 @@ 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++) {
@@ -339,10 +309,13 @@ 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);
@@ -376,7 +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);
+    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;
 }

 /**************************************************************/
@@ -396,7 +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);
@@ -476,7 +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
@@ -518,8 +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));
+    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)
--- a/doc/examples/vaapi_encode.c
+++ b/doc/examples/vaapi_encode.c
@@ -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;
--- a/doc/faq.texi
+++ b/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?

--- a/doc/fate.texi
+++ b/doc/fate.texi
@@ -149,8 +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.
@@ -159,8 +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:
--- a/doc/ffmpeg.texi
+++ b/doc/ffmpeg.texi
@@ -523,9 +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.

@@ -617,13 +614,8 @@ they do not conflict with the standard, as in:
 ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg
@end example

-@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})
@@ -783,13 +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})
@@ -879,19 +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})
+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
@@ -905,11 +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
@@ -937,12 +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.
-
-@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.
@@ -1029,35 +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}
@@ -1149,13 +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}.
@@ -1190,13 +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
@@ -1430,7 +1360,7 @@ 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})
@@ -1443,6 +1373,10 @@ 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.
@@ -1562,13 +1496,9 @@ Enable bitexact mode for (de)muxer and (de/en)coder
 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
@@ -1690,10 +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
@@ -1721,8 +1649,6 @@ 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 -xerror (@emph{global})
--- a/doc/ffplay.texi
+++ b/doc/ffplay.texi
@@ -66,8 +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
@@ -133,9 +131,8 @@ 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.
@@ -198,12 +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
--- a/doc/ffprobe.texi
+++ b/doc/ffprobe.texi
@@ -425,7 +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
@@ -591,7 +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
--- a/doc/ffprobe.xsd
+++ b/doc/ffprobe.xsd
@@ -147,25 +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" />
@@ -226,7 +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="has_b_frames"         type="xsd:int"/>
      <xsd:attribute name="sample_aspect_ratio"  type="xsd:string"/>
      <xsd:attribute name="display_aspect_ratio" type="xsd:string"/>
--- a/doc/fftools-common-opts.texi
+++ b/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).
@@ -109,10 +112,6 @@ Print detailed information about the muxer named @var{muxer_name}. Use the
@item filter=@var{filter_name}
 Print detailed information about the filter name @var{filter_name}. Use the
@option{-filters} option to get a list of all filters.
-
-@item bsf=@var{bitstream_filter_name}
-Print detailed information about the bitstream filter name @var{bitstream_filter_name}.
-Use the @option{-bsfs} option to get a list of all bitstream filters.
@end table

@item -version
@@ -236,15 +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
@@ -370,15 +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}.
--- a/doc/filters.texi
+++ b/doc/filters.texi
--- a/doc/formats.texi
+++ b/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.

@@ -143,7 +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
@@ -215,7 +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
@@ -228,28 +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
@@ -260,10 +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
--- a/doc/general.texi
+++ b/doc/general.texi
@@ -17,107 +17,21 @@ for more formats. None of them are used by default, their use has to be
 explicitly requested by passing the appropriate flags to
@command{./configure}.

-@section Alliance for Open Media (AOM)
+@section libxavs2

-FFmpeg can make use of the AOM library for AV1 decoding and encoding.
+FFmpeg can make use of the xavs2 library for AVS2-P2/IEEE1857.4 video encoding.

-Go to @url{http://aomedia.org/} and follow the instructions for
-installing the library. Then pass @code{--enable-libaom} to configure to
+Go to @url{https://github.com/pkuvcl/xavs2} and follow the instructions for
+installing the library. Then pass @code{--enable-libxavs2} to configure to
 enable it.

-@section AMD AMF/VCE
-
-FFmpeg can use the AMD Advanced Media Framework library
-for accelerated H.264 and HEVC(only windows) encoding on hardware with Video Coding Engine (VCE).
-
-To enable support you must obtain the AMF framework header files(version 1.4.9+) from
-@url{https://github.com/GPUOpen-LibrariesAndSDKs/AMF.git}.
-
-Create an @code{AMF/} directory in the system include path.
-Copy the contents of @code{AMF/amf/public/include/} into that directory.
-Then configure FFmpeg with @code{--enable-amf}.
-
-Initialization of amf encoder occurs in this order:
-1) trying to initialize through dx11(only windows)
-2) trying to initialize through dx9(only windows)
-3) trying to initialize through vulkan
-
-To use h.264(AMD VCE) encoder on linux amdgru-pro version 19.20+ and amf-amdgpu-pro
-package(amdgru-pro contains, but does not install automatically) are required.
-
-This driver can be installed using amdgpu-pro-install script in official amd driver archive.
-
-@section AviSynth
-
-FFmpeg can read AviSynth scripts as input. To enable support, pass
-@code{--enable-avisynth} to configure after installing the headers
-provided by @url{https://github.com/AviSynth/AviSynthPlus, AviSynth+}.
-AviSynth+ can be configured to install only the headers by either
-passing @code{-DHEADERS_ONLY:bool=on} to the normal CMake-based build
-system, or by using the supplied @code{GNUmakefile}.
-
-For Windows, supported AviSynth variants are
-@url{http://avisynth.nl, AviSynth 2.6 RC1 or higher} for 32-bit builds and
-@url{http://avisynth.nl/index.php/AviSynth+, AviSynth+ r1718 or higher} for 32-bit and 64-bit builds.
-
-For Linux, macOS, and BSD, the only supported AviSynth variant is
-@url{https://github.com/AviSynth/AviSynthPlus, AviSynth+}, starting with version 3.5.
-
@float NOTE
-In 2016, AviSynth+ added support for building with GCC. However, due to
-the eccentricities of Windows' calling conventions, 32-bit GCC builds
-of AviSynth+ are not compatible with typical 32-bit builds of FFmpeg.
-
-By default, FFmpeg assumes compatibility with 32-bit MSVC builds of
-AviSynth+ since that is the most widely-used and entrenched build
-configuration.  Users can override this and enable support for 32-bit
-GCC builds of AviSynth+ by passing @code{-DAVSC_WIN32_GCC32} to
-@code{--extra-cflags} when configuring FFmpeg.
-
-64-bit builds of FFmpeg are not affected, and can use either MSVC or
-GCC builds of AviSynth+ without any special flags.
+libxavs2 is under the GNU Public License Version 2 or later
+(see @url{http://www.gnu.org/licenses/old-licenses/gpl-2.0.html} for
+details), you must upgrade FFmpeg's license to GPL in order to use it.
@end float

-@float NOTE
-AviSynth(+) is loaded dynamically.  Distributors can build FFmpeg
-with @code{--enable-avisynth}, and the binaries will work regardless
-of the end user having AviSynth installed.  If/when an end user
-would like to use AviSynth scripts, then they can install AviSynth(+)
-and FFmpeg will be able to find and use it to open scripts.
-@end float
-
-@section Chromaprint
-
-FFmpeg can make use of the Chromaprint library for generating audio fingerprints.
-Pass @code{--enable-chromaprint} to configure to
-enable it. See @url{https://acoustid.org/chromaprint}.
-
-@section codec2
-
-FFmpeg can make use of the codec2 library for codec2 decoding and encoding.
-There is currently no native decoder, so libcodec2 must be used for decoding.
-
-Go to @url{http://freedv.org/}, download "Codec 2 source archive".
-Build and install using CMake. Debian users can install the libcodec2-dev package instead.
-Once libcodec2 is installed you can pass @code{--enable-libcodec2} to configure to enable it.
-
-The easiest way to use codec2 is with .c2 files, since they contain the mode information required for decoding.
-To encode such a file, use a .c2 file extension and give the libcodec2 encoder the -mode option:
-@code{ffmpeg -i input.wav -mode 700C output.c2}.
-Playback is as simple as @code{ffplay output.c2}.
-For a list of supported modes, run @code{ffmpeg -h encoder=libcodec2}.
-Raw codec2 files are also supported.
-To make sense of them the mode in use needs to be specified as a format option:
-@code{ffmpeg -f codec2raw -mode 1300 -i input.raw output.wav}.
-
-@section dav1d
-
-FFmpeg can make use of the dav1d library for AV1 video decoding.
-
-Go to @url{https://code.videolan.org/videolan/dav1d} and follow the instructions for
-installing the library. Then pass @code{--enable-libdav1d} to configure to enable it.
-
-@section davs2
+@section libdavs2

 FFmpeg can make use of the davs2 library for AVS2-P2/IEEE1857.4 video decoding.

@@ -131,63 +45,21 @@ libdavs2 is under the GNU Public License Version 2 or later
 details), you must upgrade FFmpeg's license to GPL in order to use it.
@end float

-@section Game Music Emu
+@section Alliance for Open Media libaom

-FFmpeg can make use of the Game Music Emu library to read audio from supported video game
-music file formats. Pass @code{--enable-libgme} to configure to
-enable it. See @url{https://bitbucket.org/mpyne/game-music-emu/overview}.
+FFmpeg can make use of the libaom library for AV1 decoding.

-@section Intel QuickSync Video
-
-FFmpeg can use Intel QuickSync Video (QSV) for accelerated decoding and encoding
-of multiple codecs. To use QSV, FFmpeg must be linked against the @code{libmfx}
-dispatcher, which loads the actual decoding libraries.
-
-The dispatcher is open source and can be downloaded from
-@url{https://github.com/lu-zero/mfx_dispatch.git}. FFmpeg needs to be configured
-with the @code{--enable-libmfx} option and @code{pkg-config} needs to be able to
-locate the dispatcher's @code{.pc} files.
-
-@section Kvazaar
-
-FFmpeg can make use of the Kvazaar library for HEVC encoding.
-
-Go to @url{https://github.com/ultravideo/kvazaar} and follow the
-instructions for installing the library. Then pass
-@code{--enable-libkvazaar} to configure to enable it.
-
-@section LAME
-
-FFmpeg can make use of the LAME library for MP3 encoding.
-
-Go to @url{http://lame.sourceforge.net/} and follow the
-instructions for installing the library.
-Then pass @code{--enable-libmp3lame} to configure to enable it.
-
-@section libilbc
-
-iLBC is a narrowband speech codec that has been made freely available
-by Google as part of the WebRTC project. libilbc is a packaging friendly
-copy of the iLBC codec. FFmpeg can make use of the libilbc library for
-iLBC decoding and encoding.
-
-Go to @url{https://github.com/TimothyGu/libilbc} and follow the instructions for
-installing the library. Then pass @code{--enable-libilbc} to configure to
+Go to @url{http://aomedia.org/} and follow the instructions for
+installing the library. Then pass @code{--enable-libaom} to configure to
 enable it.

-@section libvpx
+@section OpenJPEG

-FFmpeg can make use of the libvpx library for VP8/VP9 decoding and encoding.
+FFmpeg can use the OpenJPEG libraries for encoding/decoding J2K videos.  Go to
+@url{http://www.openjpeg.org/} to get the libraries and follow the installation
+instructions.  To enable using OpenJPEG in FFmpeg, pass @code{--enable-libopenjpeg} to
+@file{./configure}.

-Go to @url{http://www.webmproject.org/} and follow the instructions for
-installing the library. Then pass @code{--enable-libvpx} to configure to
-enable it.
-
-@section ModPlug
-
-FFmpeg can make use of this library, originating in Modplug-XMMS, to read from MOD-like music files.
-See @url{https://github.com/Konstanty/libmodplug}. Pass @code{--enable-libmodplug} to configure to
-enable it.

@section OpenCORE, VisualOn, and Fraunhofer libraries

@@ -234,9 +106,67 @@ Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the
 instructions for installing the library.
 Then pass @code{--enable-libfdk-aac} to configure to enable it.

+@section LAME
+
+FFmpeg can make use of the LAME library for MP3 encoding.
+
+Go to @url{http://lame.sourceforge.net/} and follow the
+instructions for installing the library.
+Then pass @code{--enable-libmp3lame} to configure to enable it.
+
+@section TwoLAME
+
+FFmpeg can make use of the TwoLAME library for MP2 encoding.
+
+Go to @url{http://www.twolame.org/} and follow the
+instructions for installing the library.
+Then pass @code{--enable-libtwolame} to configure to enable it.
+
+@section libcodec2 / codec2 general
+
+FFmpeg can make use of libcodec2 for codec2 encoding and decoding.
+There is currently no native decoder, so libcodec2 must be used for decoding.
+
+Go to @url{http://freedv.org/}, download "Codec 2 source archive".
+Build and install using CMake. Debian users can install the libcodec2-dev package instead.
+Once libcodec2 is installed you can pass @code{--enable-libcodec2} to configure to enable it.
+
+The easiest way to use codec2 is with .c2 files, since they contain the mode information required for decoding.
+To encode such a file, use a .c2 file extension and give the libcodec2 encoder the -mode option:
+@code{ffmpeg -i input.wav -mode 700C output.c2}.
+Playback is as simple as @code{ffplay output.c2}.
+For a list of supported modes, run @code{ffmpeg -h encoder=libcodec2}.
+Raw codec2 files are also supported.
+To make sense of them the mode in use needs to be specified as a format option:
+@code{ffmpeg -f codec2raw -mode 1300 -i input.raw output.wav}.
+
+@section libvpx
+
+FFmpeg can make use of the libvpx library for VP8/VP9 encoding.
+
+Go to @url{http://www.webmproject.org/} and follow the instructions for
+installing the library. Then pass @code{--enable-libvpx} to configure to
+enable it.
+
+@section libwavpack
+
+FFmpeg can make use of the libwavpack library for WavPack encoding.
+
+Go to @url{http://www.wavpack.com/} and follow the instructions for
+installing the library. Then pass @code{--enable-libwavpack} to configure to
+enable it.
+
+@section libxavs
+
+FFmpeg can make use of the libxavs library for Xavs encoding.
+
+Go to @url{http://xavs.sf.net/} and follow the instructions for
+installing the library. Then pass @code{--enable-libxavs} to configure to
+enable it.
+
@section OpenH264

-FFmpeg can make use of the OpenH264 library for H.264 decoding and encoding.
+FFmpeg can make use of the OpenH264 library for H.264 encoding and decoding.

 Go to @url{http://www.openh264.org/} and follow the instructions for
 installing the library. Then pass @code{--enable-libopenh264} to configure to
@@ -249,47 +179,6 @@ constrained baseline profile and CABAC.) Using it is mostly useful for
 testing and for taking advantage of Cisco's patent portfolio license
 (@url{http://www.openh264.org/BINARY_LICENSE.txt}).

-@section OpenJPEG
-
-FFmpeg can use the OpenJPEG libraries for decoding/encoding J2K videos.  Go to
-@url{http://www.openjpeg.org/} to get the libraries and follow the installation
-instructions.  To enable using OpenJPEG in FFmpeg, pass @code{--enable-libopenjpeg} to
-@file{./configure}.
-
-@section rav1e
-
-FFmpeg can make use of rav1e (Rust AV1 Encoder) via its C bindings to encode videos.
-Go to @url{https://github.com/xiph/rav1e/} and follow the instructions to build
-the C library. To enable using rav1e in FFmpeg, pass @code{--enable-librav1e}
-to @file{./configure}.
-
-@section TwoLAME
-
-FFmpeg can make use of the TwoLAME library for MP2 encoding.
-
-Go to @url{http://www.twolame.org/} and follow the
-instructions for installing the library.
-Then pass @code{--enable-libtwolame} to configure to enable it.
-
-@section VapourSynth
-
-FFmpeg can read VapourSynth scripts as input. To enable support, pass
-@code{--enable-vapoursynth} to configure. Vapoursynth is detected via
-@code{pkg-config}. Versions 42 or greater supported.
-See @url{http://www.vapoursynth.com/}.
-
-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}.
-
-@section WavPack
-
-FFmpeg can make use of the libwavpack library for WavPack encoding.
-
-Go to @url{http://www.wavpack.com/} and follow the instructions for
-installing the library. Then pass @code{--enable-libwavpack} to configure to
-enable it.
-
@section x264

 FFmpeg can make use of the x264 library for H.264 encoding.
@@ -318,37 +207,92 @@ x265 is under the GNU Public License Version 2 or later
 details), you must upgrade FFmpeg's license to GPL in order to use it.
@end float

-@section xavs
+@section kvazaar

-FFmpeg can make use of the xavs library for AVS encoding.
+FFmpeg can make use of the kvazaar library for HEVC encoding.

-Go to @url{http://xavs.sf.net/} and follow the instructions for
-installing the library. Then pass @code{--enable-libxavs} to configure to
+Go to @url{https://github.com/ultravideo/kvazaar} and follow the
+instructions for installing the library. Then pass
+@code{--enable-libkvazaar} to configure to enable it.
+
+@section libilbc
+
+iLBC is a narrowband speech codec that has been made freely available
+by Google as part of the WebRTC project. libilbc is a packaging friendly
+copy of the iLBC codec. FFmpeg can make use of the libilbc library for
+iLBC encoding and decoding.
+
+Go to @url{https://github.com/TimothyGu/libilbc} and follow the instructions for
+installing the library. Then pass @code{--enable-libilbc} to configure to
 enable it.

-@section xavs2
+@section libzvbi

-FFmpeg can make use of the xavs2 library for AVS2-P2/IEEE1857.4 video encoding.
-
-Go to @url{https://github.com/pkuvcl/xavs2} and follow the instructions for
-installing the library. Then pass @code{--enable-libxavs2} to configure to
-enable it.
-
-@float NOTE
-libxavs2 is under the GNU Public License Version 2 or later
-(see @url{http://www.gnu.org/licenses/old-licenses/gpl-2.0.html} for
-details), you must upgrade FFmpeg's license to GPL in order to use it.
-@end float
-
-@section ZVBI
-
-ZVBI is a VBI decoding library which can be used by FFmpeg to decode DVB
+libzvbi is a VBI decoding library which can be used by FFmpeg to decode DVB
 teletext pages and DVB teletext subtitles.

 Go to @url{http://sourceforge.net/projects/zapping/} and follow the instructions for
 installing the library. Then pass @code{--enable-libzvbi} to configure to
 enable it.

+@section AviSynth
+
+FFmpeg can read AviSynth scripts as input. To enable support, pass
+@code{--enable-avisynth} to configure.  The correct headers are
+included in compat/avisynth/, which allows the user to enable support
+without needing to search for these headers themselves.
+
+For Windows, supported AviSynth variants are
+@url{http://avisynth.nl, AviSynth 2.6 RC1 or higher} for 32-bit builds and
+@url{http://avs-plus.net, AviSynth+ r1718 or higher} for 32-bit and 64-bit builds.
+
+For Linux and OS X, the supported AviSynth variant is
+@url{https://github.com/avxsynth/avxsynth, AvxSynth}.
+
+@float NOTE
+There is currently a regression in AviSynth+'s @code{capi.h} header as of
+October 2016, which interferes with the ability for builds of FFmpeg to use
+MSVC-built binaries of AviSynth. Until this is resolved, you can make sure
+a known good version is installed by checking out a version from before
+the regression occurred:
+
+@code{git clone -b MT git://github.com/AviSynth/AviSynthPlus.git @*
+cd AviSynthPlus @*
+git checkout -b oldheader b4f292b4dbfad149697fb65c6a037bb3810813f9 @*
+make install PREFIX=/install/prefix}
+@end float
+
+@float NOTE
+AviSynth and AvxSynth are loaded dynamically.  Distributors can build FFmpeg
+with @code{--enable-avisynth}, and the binaries will work regardless of the
+end user having AviSynth or AvxSynth installed - they'll only need to be
+installed to use AviSynth scripts (obviously).
+@end float
+
+@section Intel QuickSync Video
+
+FFmpeg can use Intel QuickSync Video (QSV) for accelerated encoding and decoding
+of multiple codecs. To use QSV, FFmpeg must be linked against the @code{libmfx}
+dispatcher, which loads the actual decoding libraries.
+
+The dispatcher is open source and can be downloaded from
+@url{https://github.com/lu-zero/mfx_dispatch.git}. FFmpeg needs to be configured
+with the @code{--enable-libmfx} option and @code{pkg-config} needs to be able to
+locate the dispatcher's @code{.pc} files.
+
+@section AMD VCE
+
+FFmpeg can use the AMD Advanced Media Framework library for accelerated H.264
+and HEVC encoding on VCE enabled hardware under Windows.
+
+To enable support you must obtain the AMF framework header files from
+@url{https://github.com/GPUOpen-LibrariesAndSDKs/AMF.git}.
+
+Create an @code{AMF/} directory in the system include path.
+Copy the contents of @code{AMF/amf/public/include/} into that directory.
+Then configure FFmpeg with @code{--enable-amf}.
+
+
@chapter Supported File Formats, Codecs or Features

 You can use the @code{-formats} and @code{-codecs} options to have an exhaustive list.
@@ -419,9 +363,6 @@ library:
    @tab Contains header with version and mode info, simplifying playback.
@item CRI ADX                   @tab X @tab X
    @tab Audio-only format used in console video games.
-@item CRI AIX                   @tab   @tab X
-@item CRI HCA                   @tab   @tab X
-    @tab Audio-only format used in console video games.
@item Discworld II BMV          @tab   @tab X
@item Interplay C93             @tab   @tab X
    @tab Used in the game Cyberia from Interplay.
@@ -491,8 +432,6 @@ library:
@item IEC61937 encapsulation @tab X @tab X
@item IFF                       @tab   @tab X
    @tab Interchange File Format
-@item IFV                       @tab   @tab X
-    @tab A format used by some old CCTV DVRs.
@item iLBC                      @tab X @tab X
@item Interplay MVE             @tab   @tab X
    @tab Format used in various Interplay computer games.
@@ -577,6 +516,7 @@ library:
@item raw aptX                  @tab X @tab X
@item raw aptX HD               @tab X @tab X
@item raw Chinese AVS video     @tab X @tab X
+@item raw CRI ADX               @tab X @tab X
@item raw Dirac                 @tab X @tab X
@item raw DNxHD                 @tab X @tab X
@item raw DTS                   @tab X @tab X
@@ -710,7 +650,7 @@ library:
@item Psygnosis YOP             @tab   @tab X
@end multitable

-@code{X} means that the feature in that column (encoding / decoding) is supported.
+@code{X} means that encoding (resp. decoding) is supported.

@section Image Formats

@@ -780,7 +720,7 @@ following image formats are supported:
    @tab X Window Dump image format
@end multitable

-@code{X} means that the feature in that column (encoding / decoding) is supported.
+@code{X} means that encoding (resp. decoding) is supported.

@code{E} means that support is provided through an external library.

@@ -818,14 +758,12 @@ following image formats are supported:
@item Autodesk Animator Flic video  @tab     @tab  X
@item Autodesk RLE           @tab     @tab  X
    @tab fourcc: AASC
-@item AV1                    @tab  E  @tab  E
-    @tab Supported through external libraries libaom, libdav1d and librav1e
+@item AV1                    @tab     @tab  E
+    @tab Supported through external library libaom
@item Avid 1:1 10-bit RGB Packer  @tab  X  @tab  X
    @tab fourcc: AVrp
@item AVS (Audio Video Standard) video  @tab     @tab  X
    @tab Video encoding used by the Creature Shock game.
-@item AVS2-P2/IEEE1857.4     @tab  E  @tab  E
-    @tab Supported through external libraries libxavs2 and libdavs2
@item AYUV                   @tab  X  @tab  X
    @tab Microsoft uncompressed packed 4:4:4:4
@item Beam Software VB       @tab     @tab  X
@@ -851,8 +789,6 @@ following image formats are supported:
    @tab Codec used in Delphine Software International games.
@item Discworld II BMV Video @tab     @tab  X
@item Canopus Lossless Codec @tab     @tab  X
-@item CDToons                @tab     @tab  X
-    @tab Codec used in various Broderbund games.
@item Cinepak                @tab     @tab  X
@item Cirrus Logic AccuPak   @tab  X  @tab  X
    @tab fourcc: CLJR
@@ -972,8 +908,6 @@ following image formats are supported:
    @tab Video encoding used in NuppelVideo files.
@item On2 VP3                @tab     @tab  X
    @tab still experimental
-@item On2 VP4                @tab     @tab  X
-    @tab fourcc: VP40
@item On2 VP5                @tab     @tab  X
    @tab fourcc: VP50
@item On2 VP6                @tab     @tab  X
@@ -1068,7 +1002,7 @@ following image formats are supported:
    @tab Encoder works only in PAL8.
@end multitable

-@code{X} means that the feature in that column (encoding / decoding) is supported.
+@code{X} means that encoding (resp. decoding) is supported.

@code{E} means that support is provided through an external library.

@@ -1083,10 +1017,8 @@ following image formats are supported:
@item AAC+                   @tab  E  @tab  IX
    @tab encoding supported through external library libfdk-aac
@item AC-3                   @tab IX  @tab  IX
-@item ACELP.KELVIN           @tab     @tab  X
@item ADPCM 4X Movie         @tab     @tab  X
@item APDCM Yamaha AICA      @tab     @tab  X
-@item ADPCM Argonaut Games   @tab     @tab  X
@item ADPCM CDROM XA         @tab     @tab  X
@item ADPCM Creative Technology @tab     @tab  X
    @tab 16 -> 4, 8 -> 4, 8 -> 3, 8 -> 2
@@ -1102,14 +1034,10 @@ following image formats are supported:
@item ADPCM G.726            @tab  X  @tab  X
@item ADPCM IMA AMV          @tab     @tab  X
    @tab Used in AMV files
-@item ADPCM IMA Cunning Developments  @tab     @tab  X
@item ADPCM IMA Electronic Arts EACS  @tab     @tab  X
@item ADPCM IMA Electronic Arts SEAD  @tab     @tab  X
@item ADPCM IMA Funcom       @tab     @tab  X
-@item ADPCM IMA High Voltage Software ALP   @tab     @tab  X
@item ADPCM IMA QuickTime    @tab  X  @tab  X
-@item ADPCM IMA Simon & Schuster Interactive   @tab  X  @tab  X
-@item ADPCM IMA Ubisoft APM  @tab     @tab X
@item ADPCM IMA Loki SDL MJPEG  @tab     @tab  X
@item ADPCM IMA WAV          @tab  X  @tab  X
@item ADPCM IMA Westwood     @tab     @tab  X
@@ -1139,7 +1067,6 @@ following image formats are supported:
@item ADPCM Westwood Studios IMA @tab     @tab  X
    @tab Used in Westwood Studios games like Command and Conquer.
@item ADPCM Yamaha           @tab  X  @tab  X
-@item ADPCM Zork             @tab     @tab  X
@item AMR-NB                 @tab  E  @tab  X
    @tab encoding supported through external library libopencore-amrnb
@item AMR-WB                 @tab  E  @tab  X
@@ -1161,7 +1088,6 @@ following image formats are supported:
    @tab decoding supported through external library libcelt
@item codec2                 @tab  E  @tab  E
    @tab en/decoding supported through external library libcodec2
-@item CRI HCA                @tab     @tab X
@item Delphine Software International CIN audio  @tab     @tab  X
    @tab Codec used in Delphine Software International games.
@item Digital Speech Standard - Standard Play mode (DSS SP) @tab     @tab  X
@@ -1171,7 +1097,6 @@ following image formats are supported:
@item DCA (DTS Coherent Acoustics)  @tab  X  @tab  X
    @tab supported extensions: XCh, XXCH, X96, XBR, XLL, LBR (partially)
@item Dolby E  @tab     @tab  X
-@item DPCM Gremlin           @tab     @tab  X
@item DPCM id RoQ            @tab  X  @tab  X
    @tab Used in Quake III, Jedi Knight 2 and other computer games.
@item DPCM Interplay         @tab     @tab  X
@@ -1183,11 +1108,10 @@ following image formats are supported:
@item DPCM Sol               @tab     @tab  X
@item DPCM Xan               @tab     @tab  X
    @tab Used in Origin's Wing Commander IV AVI files.
-@item DPCM Xilam DERF        @tab     @tab  X
-@item DSD (Direct Stream Digital), least significant bit first  @tab  @tab  X
-@item DSD (Direct Stream Digital), most significant bit first   @tab  @tab  X
-@item DSD (Direct Stream Digital), least significant bit first, planar  @tab  @tab  X
-@item DSD (Direct Stream Digital), most significant bit first, planar   @tab  @tab  X
+@item DSD (Direct Stream Digitial), least significant bit first  @tab  @tab  X
+@item DSD (Direct Stream Digitial), most significant bit first   @tab  @tab  X
+@item DSD (Direct Stream Digitial), least significant bit first, planar  @tab  @tab  X
+@item DSD (Direct Stream Digitial), most significant bit first, planar   @tab  @tab  X
@item DSP Group TrueSpeech   @tab     @tab  X
@item DST (Direct Stream Transfer) @tab  @tab  X
@item DV audio               @tab     @tab  X
@@ -1250,6 +1174,7 @@ following image formats are supported:
@item PCM unsigned 24-bit little-endian  @tab  X  @tab  X
@item PCM unsigned 32-bit big-endian  @tab  X  @tab  X
@item PCM unsigned 32-bit little-endian  @tab  X  @tab  X
+@item PCM Zork               @tab     @tab  X
@item QCELP / PureVoice      @tab     @tab  X
@item QDesign Music Codec 1  @tab     @tab  X
@item QDesign Music Codec 2  @tab     @tab  X
@@ -1296,7 +1221,7 @@ following image formats are supported:
@item Xbox Media Audio 2     @tab     @tab  X
@end multitable

-@code{X} means that the feature in that column (encoding / decoding) is supported.
+@code{X} means that encoding (resp. decoding) is supported.

@code{E} means that support is provided through an external library.

@@ -1340,7 +1265,6 @@ performance on systems without hardware floating point support).

@multitable @columnfractions .4 .1
@item Name         @tab Support
-@item AMQP         @tab E
@item file         @tab X
@item FTP          @tab X
@item Gopher       @tab X
@@ -1365,7 +1289,6 @@ performance on systems without hardware floating point support).
@item TCP          @tab X
@item TLS          @tab X
@item UDP          @tab X
-@item ZMQ          @tab E
@end multitable

@code{X} means that the protocol is supported.
--- a/doc/git-howto.texi
+++ b/doc/git-howto.texi
@@ -53,7 +53,7 @@ Most distribution and operating system provide a package for it.
@section Cloning the source tree

@example
-git clone git://source.ffmpeg.org/ffmpeg <target>
+git clone https://git.ffmpeg.org/ffmpeg.git <target>
@end example

 This will put the FFmpeg sources into the directory @var{<target>}.
@@ -187,11 +187,18 @@ 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 and email address
+Make sure you have told Git your name, email address and GPG key

@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.
@@ -393,6 +400,19 @@ 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,
--- a/doc/indevs.texi
+++ b/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.
@@ -292,6 +282,11 @@ 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:
@@ -390,14 +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}.
-
@end table

@subsection Examples
@@ -407,7 +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
@@ -425,7 +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
@@ -800,7 +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
@@ -923,14 +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
@@ -1071,21 +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
@@ -1527,7 +1556,7 @@ ffmpeg -f x11grab -follow_mouse centered -show_region 1 -framerate 25 -video_siz
@end example

@item video_size
-Set the video frame size. Default is the full desktop.
+Set the video frame size. Default value is @code{vga}.

@item grab_x
@item grab_y
--- a/doc/libav-merge.txt
+++ b/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:
 ------------------------------------------
--- a/doc/mailing-list-faq.texi
+++ b/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 Freenode 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?
@@ -157,12 +157,11 @@ Perform a site search using your favorite search engine. Example:

 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?

--- a/doc/metadata.texi
+++ b/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
--- a/doc/multithreading.txt
+++ b/doc/multithreading.txt
@@ -51,14 +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
--- a/doc/muxers.texi
+++ b/doc/muxers.texi
@@ -94,25 +94,21 @@ compatibility with software that only supports a single audio stream in AVI
@anchor{chromaprint}
@section chromaprint

-Chromaprint fingerprinter.
+Chromaprint fingerprinter

-This muxer feeds audio data to the Chromaprint library,
-which generates a fingerprint for the provided audio data. See @url{https://acoustid.org/chromaprint}
-
-It takes a single signed native-endian 16-bit raw audio stream of at most 2 channels.
+This muxer feeds audio data to the Chromaprint library, which generates
+a fingerprint for the provided audio data. It takes a single signed
+native-endian 16-bit raw audio stream.

@subsection Options

@table @option
@item silence_threshold
-Threshold for detecting silence. Range is from -1 to 32767, where -1 disables
-silence detection. Silence detection can only be used with version 3 of the
-algorithm.
-Silence detection must be disabled for use with the AcoustID service. Default is -1.
+Threshold for detecting silence, ranges from 0 to 32767. -1 for default
+(required for use with the AcoustID service).

@item algorithm
-Version of algorithm to fingerprint with. Range is 0 to 4.
-Version 3 enables silence detection. Default is 1.
+Algorithm index to fingerprint with.

@item fp_format
 Format to output the fingerprint as. Accepts the following options:
@@ -124,7 +120,7 @@ Binary raw fingerprint
 Binary compressed fingerprint

@item base64
-Base64 compressed fingerprint @emph{(default)}
+Base64 compressed fingerprint

@end table

@@ -218,79 +214,66 @@ It creates a MPD manifest file and segment files for each stream.
 The segment filename might contain pre-defined identifiers used with SegmentTemplate
 as defined in section 5.3.9.4.4 of the standard. Available identifiers are "$RepresentationID$",
 "$Number$", "$Bandwidth$" and "$Time$".
-In addition to the standard identifiers, an ffmpeg-specific "$ext$" identifier is also supported.
-When specified ffmpeg will replace $ext$ in the file name with muxing format's extensions such as mp4, webm etc.,

@example
-ffmpeg -re -i <input> -map 0 -map 0 -c:a libfdk_aac -c:v libx264 \
-b:v:0 800k -b:v:1 300k -s:v:1 320x170 -profile:v:1 baseline \
-profile:v:0 main -bf 1 -keyint_min 120 -g 120 -sc_threshold 0 \
-b_strategy 0 -ar:a:1 22050 -use_timeline 1 -use_template 1 \
-window_size 5 -adaptation_sets "id=0,streams=v id=1,streams=a" \
+ffmpeg -re -i <input> -map 0 -map 0 -c:a libfdk_aac -c:v libx264
+-b:v:0 800k -b:v:1 300k -s:v:1 320x170 -profile:v:1 baseline
+-profile:v:0 main -bf 1 -keyint_min 120 -g 120 -sc_threshold 0
+-b_strategy 0 -ar:a:1 22050 -use_timeline 1 -use_template 1
+-window_size 5 -adaptation_sets "id=0,streams=v id=1,streams=a"
 -f dash /path/to/out.mpd
@end example

@table @option
-@item min_seg_duration @var{microseconds}
+@item -min_seg_duration @var{microseconds}
 This is a deprecated option to set the segment length in microseconds, use @var{seg_duration} instead.
-@item seg_duration @var{duration}
+@item -seg_duration @var{duration}
 Set the segment length in seconds (fractional value can be set). The value is
 treated as average segment duration when @var{use_template} is enabled and
-@item frag_duration @var{duration}
-Set the length in seconds of fragments within segments (fractional value can be set).
-@item frag_type @var{type}
-Set the type of interval for fragmentation.
-@item window_size @var{size}
+@var{use_timeline} is disabled and as minimum segment duration for all the other
+use cases.
+@item -window_size @var{size}
 Set the maximum number of segments kept in the manifest.
-@item extra_window_size @var{size}
+@item -extra_window_size @var{size}
 Set the maximum number of segments kept outside of the manifest before removing from disk.
-@item remove_at_exit @var{remove}
+@item -remove_at_exit @var{remove}
 Enable (1) or disable (0) removal of all segments when finished.
-@item use_template @var{template}
+@item -use_template @var{template}
 Enable (1) or disable (0) use of SegmentTemplate instead of SegmentList.
-@item use_timeline @var{timeline}
+@item -use_timeline @var{timeline}
 Enable (1) or disable (0) use of SegmentTimeline in SegmentTemplate.
-@item single_file @var{single_file}
+@item -single_file @var{single_file}
 Enable (1) or disable (0) storing all segments in one file, accessed using byte ranges.
-@item single_file_name @var{file_name}
-DASH-templated name to be used for baseURL. Implies @var{single_file} set to "1". In the template, "$ext$" is replaced with the file name extension specific for the segment format.
-@item init_seg_name @var{init_name}
-DASH-templated name to used for the initialization segment. Default is "init-stream$RepresentationID$.$ext$". "$ext$" is replaced with the file name extension specific for the segment format.
-@item media_seg_name @var{segment_name}
-DASH-templated name to used for the media segments. Default is "chunk-stream$RepresentationID$-$Number%05d$.$ext$". "$ext$" is replaced with the file name extension specific for the segment format.
-@item utc_timing_url @var{utc_url}
+@item -single_file_name @var{file_name}
+DASH-templated name to be used for baseURL. Implies @var{single_file} set to "1".
+@item -init_seg_name @var{init_name}
+DASH-templated name to used for the initialization segment. Default is "init-stream$RepresentationID$.m4s"
+@item -media_seg_name @var{segment_name}
+DASH-templated name to used for the media segments. Default is "chunk-stream$RepresentationID$-$Number%05d$.m4s"
+@item -utc_timing_url @var{utc_url}
 URL of the page that will return the UTC timestamp in ISO format. Example: "https://time.akamai.com/?iso"
@item method @var{method}
 Use the given HTTP method to create output files. Generally set to PUT or POST.
-@item http_user_agent @var{user_agent}
+@item -http_user_agent @var{user_agent}
 Override User-Agent field in HTTP header. Applicable only for HTTP output.
-@item http_persistent @var{http_persistent}
+@item -http_persistent @var{http_persistent}
 Use persistent HTTP connections. Applicable only for HTTP output.
-@item hls_playlist @var{hls_playlist}
+@item -hls_playlist @var{hls_playlist}
 Generate HLS playlist files as well. The master playlist is generated with the filename master.m3u8.
 One media playlist file is generated for each stream with filenames media_0.m3u8, media_1.m3u8, etc.
-@item streaming @var{streaming}
+@item -streaming @var{streaming}
 Enable (1) or disable (0) chunk streaming mode of output. In chunk streaming
 mode, each frame will be a moof fragment which forms a chunk.
-@item adaptation_sets @var{adaptation_sets}
+@item -adaptation_sets @var{adaptation_sets}
 Assign streams to AdaptationSets. Syntax is "id=x,streams=a,b,c id=y,streams=d,e" with x and y being the IDs
 of the adaptation sets and a,b,c,d and e are the indices of the mapped streams.

 To map all video (or audio) streams to an AdaptationSet, "v" (or "a") can be used as stream identifier instead of IDs.

 When no assignment is defined, this defaults to an AdaptationSet for each stream.
-
-Optional syntax is "id=x,seg_duration=x,frag_duration=x,frag_type=type,descriptor=descriptor_string,streams=a,b,c id=y,seg_duration=y,frag_type=type,streams=d,e" and so on,
-descriptor is useful to the scheme defined by ISO/IEC 23009-1:2014/Amd.2:2015.
-For example, -adaptation_sets "id=0,descriptor=<SupplementalProperty schemeIdUri=\"urn:mpeg:dash:srd:2014\" value=\"0,0,0,1,1,2,2\"/>,streams=v".
-Please note that descriptor string should be a self-closing xml tag.
-seg_duration, frag_duration and frag_type override the global option values for each adaptation set.
-For example, -adaptation_sets "id=0,seg_duration=2,frag_duration=1,frag_type=duration,streams=v id=1,seg_duration=2,frag_type=none,streams=a"
-type_id marks an adaptation set as containing streams meant to be used for Trick Mode for the referenced adaptation set.
-For example, -adaptation_sets "id=0,seg_duration=2,frag_type=none,streams=0 id=1,seg_duration=10,frag_type=none,trick_id=0,streams=1"
-@item timeout @var{timeout}
+@item -timeout @var{timeout}
 Set timeout for socket I/O operations. Applicable only for HTTP output.
-@item index_correction @var{index_correction}
+@item -index_correction @var{index_correction}
 Enable (1) or Disable (0) segment index correction logic. Applicable only when
@var{use_template} is enabled and @var{use_timeline} is disabled.

@@ -301,68 +284,18 @@ corrects that index value.
 Typically this logic is needed in live streaming use cases. The network bandwidth
 fluctuations are common during long run streaming. Each fluctuation can cause
 the segment indexes fall behind the expected real time position.
-@item format_options @var{options_list}
+@item -format_options @var{options_list}
 Set container format (mp4/webm) options using a @code{:} separated list of
 key=value parameters. Values containing @code{:} special characters must be
 escaped.

-@item global_sidx @var{global_sidx}
-Write global SIDX atom. Applicable only for single file, mp4 output, non-streaming mode.
-
@item dash_segment_type @var{dash_segment_type}
 Possible values:
-@table @option
-@item auto
-If this flag is set, the dash segment files format will be selected based on the stream codec. This is the default mode.
-
@item mp4
-If this flag is set, the dash segment files will be in in ISOBMFF format.
+If this flag is set, the dash segment files will be in in ISOBMFF format. This is the default format.

@item webm
 If this flag is set, the dash segment files will be in in WebM format.
-@end table
-
-@item ignore_io_errors @var{ignore_io_errors}
-Ignore IO errors during open and write. Useful for long-duration runs with network output.
-
-@item lhls @var{lhls}
-Enable Low-latency HLS(LHLS). Adds #EXT-X-PREFETCH tag with current segment's URI.
-Apple doesn't have an official spec for LHLS. Meanwhile hls.js player folks are
-trying to standardize a open LHLS spec. The draft spec is available in https://github.com/video-dev/hlsjs-rfcs/blob/lhls-spec/proposals/0001-lhls.md
-This option will also try to comply with the above open spec, till Apple's spec officially supports it.
-Applicable only when @var{streaming} and @var{hls_playlist} options are enabled.
-This is an experimental feature.
-
-@item ldash @var{ldash}
-Enable Low-latency Dash by constraining the presence and values of some elements.
-
-@item master_m3u8_publish_rate @var{master_m3u8_publish_rate}
-Publish master playlist repeatedly every after specified number of segment intervals.
-
-@item write_prft @var{write_prft}
-Write Producer Reference Time elements on supported streams. This also enables writing
-prft boxes in the underlying muxer. Applicable only when the @var{utc_url} option is enabled.
-It's set to auto by default, in which case the muxer will attempt to enable it only in modes
-that require it.
-
-@item mpd_profile @var{mpd_profile}
-Set one or more manifest profiles.
-
-@item http_opts @var{http_opts}
-A :-separated list of key=value options to pass to the underlying HTTP
-protocol. Applicable only for HTTP output.
-
-@item target_latency @var{target_latency}
-Set an intended target latency in seconds (fractional value can be set) for serving. Applicable only when @var{streaming} and @var{write_prft} options are enabled.
-This is an informative fields clients can use to measure the latency of the service.
-
-@item min_playback_rate @var{min_playback_rate}
-Set the minimum playback rate indicated as appropriate for the purposes of automatically
-adjusting playback latency and buffer occupancy during normal playback by clients.
-
-@item max_playback_rate @var{max_playback_rate}
-Set the maximum playback rate indicated as appropriate for the purposes of automatically
-adjusting playback latency and buffer occupancy during normal playback by clients.

@end table

@@ -648,9 +581,6 @@ Set the starting sequence numbers according to @var{start_number} option value.
@item epoch
 The start number will be the seconds since epoch (1970-01-01 00:00:00)

-@item epoch_us
-The start number will be the microseconds since epoch (1970-01-01 00:00:00)
-
@item datetime
 The start number will be based on the current date/time as YYYYmmddHHMMSS. e.g. 20161231235759.

@@ -702,8 +632,7 @@ This example will produce the playlists segment file sets:
@file{file_1_000.ts}, @file{file_1_001.ts}, @file{file_1_002.ts}, etc.

 The string "%v" may be present in the filename or in the last directory name
-containing the file, but only in one of them. (Additionally, %v may appear multiple times in the last
-sub-directory or filename.) If the string %v is present in the directory name, then
+containing the file. If the string is present in the directory name, then
 sub-directories are created after expanding the directory name pattern. This
 enables creation of segments corresponding to different variant streams in
 subdirectories.
@@ -847,9 +776,6 @@ fmp4 files may be used in HLS version 7 and above.
@item hls_fmp4_init_filename @var{filename}
 Set filename to the fragment files header file, default filename is @file{init.mp4}.

-@item hls_fmp4_init_resend
-Resend init file after m3u8 file refresh every time, default is @var{0}.
-
 When @code{var_stream_map} is set with two or more variant streams, the
@var{filename} pattern must contain the string "%v", this string specifies
 the position of variant stream index in the generated init file names.
@@ -902,10 +828,6 @@ including the file containing the AES encryption key.
 Add the @code{#EXT-X-INDEPENDENT-SEGMENTS} to playlists that has video segments
 and when all the segments of that playlist are guaranteed to start with a Key frame.

-@item iframes_only
-Add the @code{#EXT-X-I-FRAMES-ONLY} to playlists that has video segments
-and can play only I-frames in the @code{#EXT-X-BYTERANGE} mode.
-
@item split_by_time
 Allow segments to start on frames other than keyframes. This improves
 behavior on some players when the time between keyframes is inconsistent,
@@ -942,11 +864,7 @@ This will produce segments like this:
@item temp_file
 Write segment data to filename.tmp and rename to filename only once the segment is complete. A webserver
 serving up segments can be configured to reject requests to *.tmp to prevent access to in-progress segments
-before they have been added to the m3u8 playlist. This flag also affects how m3u8 playlist files are created.
-If this flag is set, all playlist files will written into temporary file and renamed after they are complete, similarly as segments are handled.
-But playlists with @code{file} protocol and with type (@code{hls_playlist_type}) other than @code{vod}
-are always written into temporary file regardless of this flag. Master playlist files (@code{master_pl_name}), if any, with @code{file} protocol,
-are always written into temporary file regardless of this flag if @code{master_pl_publish_rate} value is other than zero.
+before they have been added to the m3u8 playlist.

@end table

@@ -997,21 +915,7 @@ This example creates two hls variant streams. The first variant stream will
 contain video stream of bitrate 1000k and audio stream of bitrate 64k and the
 second variant stream will contain video stream of bitrate 256k and audio
 stream of bitrate 32k. Here, two media playlist with file names out_0.m3u8 and
-out_1.m3u8 will be created. If you want something meaningful text instead of indexes
-in result names, you may specify names for each or some of the variants
-as in the following example.
-
-
-@example
-ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \
-  -map 0:v -map 0:a -map 0:v -map 0:a -f hls -var_stream_map "v:0,a:0,name:my_hd v:1,a:1,name:my_sd" \
-  http://example.com/live/out_%v.m3u8
-@end example
-
-This example creates two hls variant streams as in the previous one.
-But here, the two media playlist with file names out_my_hd.m3u8 and
-out_my_sd.m3u8 will be created.
-
+out_1.m3u8 will be created.
@example
 ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k \
  -map 0:v -map 0:a -map 0:v -f hls -var_stream_map "v:0 a:0 v:1" \
@@ -1045,52 +949,6 @@ and they are mapped to the two video only variant streams with audio group names

 By default, a single hls variant containing all the encoded streams is created.

-@example
-ffmpeg -re -i in.ts -b:a:0 32k -b:a:1 64k -b:v:0 1000k \
-  -map 0:a -map 0:a -map 0:v -f hls \
-  -var_stream_map "a:0,agroup:aud_low,default:yes a:1,agroup:aud_low v:0,agroup:aud_low" \
-  -master_pl_name master.m3u8 \
-  http://example.com/live/out_%v.m3u8
-@end example
-This example creates two audio only and one video only variant streams. In
-addition to the #EXT-X-STREAM-INF tag for each variant stream in the master
-playlist, #EXT-X-MEDIA tag is also added for the two audio only variant streams
-and they are mapped to the one video only variant streams with audio group name
-'aud_low', and the audio group have default stat is NO or YES.
-
-By default, a single hls variant containing all the encoded streams is created.
-
-@example
-ffmpeg -re -i in.ts -b:a:0 32k -b:a:1 64k -b:v:0 1000k \
-  -map 0:a -map 0:a -map 0:v -f hls \
-  -var_stream_map "a:0,agroup:aud_low,default:yes,language:ENG a:1,agroup:aud_low,language:CHN v:0,agroup:aud_low" \
-  -master_pl_name master.m3u8 \
-  http://example.com/live/out_%v.m3u8
-@end example
-This example creates two audio only and one video only variant streams. In
-addition to the #EXT-X-STREAM-INF tag for each variant stream in the master
-playlist, #EXT-X-MEDIA tag is also added for the two audio only variant streams
-and they are mapped to the one video only variant streams with audio group name
-'aud_low', and the audio group have default stat is NO or YES, and one audio
-have and language is named ENG, the other audio language is named CHN.
-
-By default, a single hls variant containing all the encoded streams is created.
-
-@example
-ffmpeg -y -i input_with_subtitle.mkv \
- -b:v:0 5250k -c:v h264 -pix_fmt yuv420p -profile:v main -level 4.1 \
- -b:a:0 256k \
- -c:s webvtt -c:a mp2 -ar 48000 -ac 2 -map 0:v -map 0:a:0 -map 0:s:0 \
- -f hls -var_stream_map "v:0,a:0,s:0,sgroup:subtitle" \
- -master_pl_name master.m3u8 -t 300 -hls_time 10 -hls_init_time 4 -hls_list_size \
- 10 -master_pl_publish_rate 10  -hls_flags \
- delete_segments+discont_start+split_by_time ./tmp/video.m3u8
-@end example
-
-This example adds @code{#EXT-X-MEDIA} tag with @code{TYPE=SUBTITLES} in
-the master playlist with webvtt subtitle group name 'subtitle'. Please make sure
-the input file has one text subtitle stream at least.
-
@item cc_stream_map
 Map string which specifies different closed captions groups and their
 attributes. The closed captions stream groups are separated by space.
@@ -1111,7 +969,7 @@ ffmpeg -re -i in.ts -b:v 1000k -b:a 64k -a53cc 1 -f hls \
  http://example.com/live/out.m3u8
@end example
 This example adds @code{#EXT-X-MEDIA} tag with @code{TYPE=CLOSED-CAPTIONS} in
-the master playlist with group name 'cc', language 'en' (english) and
+the master playlist with group name 'cc', langauge 'en' (english) and
 INSTREAM-ID 'CC1'. Also, it adds @code{CLOSED-CAPTIONS} attribute with group
 name 'cc' for the output variant stream.
@example
@@ -1154,12 +1012,6 @@ Use persistent HTTP connections. Applicable only for HTTP output.
@item timeout
 Set timeout for socket I/O operations. Applicable only for HTTP output.

-@item -ignore_io_errors
-Ignore IO errors during open, write and delete. Useful for long-duration runs with network output.
-
-@item headers
-Set custom HTTP headers, can override built in default headers. Applicable only for HTTP output.
-
@end table

@anchor{ico}
@@ -1225,37 +1077,6 @@ The pattern "img%%-%d.jpg" will specify a sequence of filenames of the
 form @file{img%-1.jpg}, @file{img%-2.jpg}, ..., @file{img%-10.jpg},
 etc.

-The image muxer supports the .Y.U.V image file format. This format is
-special in that that each image frame consists of three files, for
-each of the YUV420P components. To read or write this image file format,
-specify the name of the '.Y' file. The muxer will automatically open the
-'.U' and '.V' files as required.
-
-@subsection Options
-
-@table @option
-@item frame_pts
-If set to 1, expand the filename with pts from pkt->pts.
-Default value is 0.
-
-@item start_number
-Start the sequence from the specified number. Default value is 1.
-
-@item update
-If set to 1, the filename will always be interpreted as just a
-filename, not a pattern, and the corresponding file will be continuously
-overwritten with new images. Default value is 0.
-
-@item strftime
-If set to 1, expand the filename with date and time information from
-@code{strftime()}. Default value is 0.
-
-@item protocol_opts @var{options_list}
-Set protocol options as a :-separated list of key=value parameters. Values
-containing the @code{:} special character must be escaped.
-
-@end table
-
@subsection Examples

 The following example shows how to use @command{ffmpeg} for creating a
@@ -1296,11 +1117,31 @@ You can set the file name with current frame's PTS:
 ffmpeg -f v4l2 -r 1 -i /dev/video0 -copyts -f image2 -frame_pts true %d.jpg"
@end example

-A more complex example is to publish contents of your desktop directly to a
-WebDAV server every second:
-@example
-ffmpeg -f x11grab -framerate 1 -i :0.0 -q:v 6 -update 1 -protocol_opts method=PUT http://example.com/desktop.jpg
-@end example
+@subsection Options
+
+@table @option
+@item frame_pts
+If set to 1, expand the filename with pts from pkt->pts.
+Default value is 0.
+
+@item start_number
+Start the sequence from the specified number. Default value is 1.
+
+@item update
+If set to 1, the filename will always be interpreted as just a
+filename, not a pattern, and the corresponding file will be continuously
+overwritten with new images. Default value is 0.
+
+@item strftime
+If set to 1, expand the filename with date and time information from
+@code{strftime()}. Default value is 0.
+@end table
+
+The image muxer supports the .Y.U.V image file format. This format is
+special in that that each image frame consists of three files, for
+each of the YUV420P components. To read or write this image file format,
+specify the name of the '.Y' file. The muxer will automatically open the
+'.U' and '.V' files as required.

@section matroska

@@ -1314,8 +1155,7 @@ The recognized metadata settings in this muxer are:

@table @option
@item title
-Set title name provided to a single track. This gets mapped to
-the FileDescription element for a stream written as attachment.
+Set title name provided to a single track.

@item language
 Specify the language of the track in the Matroska languages form.
@@ -1382,31 +1222,11 @@ index at the beginning of the file.

 If this option is set to a non-zero value, the muxer will reserve a given amount
 of space in the file header and then try to write the cues there when the muxing
-finishes. If the reserved space does not suffice, no Cues will be written, the
-file will be finalized and writing the trailer will return an error.
-A safe size for most use cases should be about 50kB per hour of video.
+finishes. If the available space does not suffice, muxing will fail. A safe size
+for most use cases should be about 50kB per hour of video.

 Note that cues are only written if the output is seekable and this option will
 have no effect if it is not.
-@item default_mode
-This option controls how the FlagDefault of the output tracks will be set.
-It influences which tracks players should play by default. The default mode
-is @samp{infer}.
-@table @samp
-@item infer
-In this mode, for each type of track (audio, video or subtitle), if there is
-a track with disposition default of this type, then the first such track
-(i.e. the one with the lowest index) will be marked as default; if no such
-track exists, the first track of this type will be marked as default instead
-(if existing). This ensures that the default flag is set in a sensible way even
-if the input originated from containers that lack the concept of default tracks.
-@item infer_no_subs
-This mode is the same as infer except that if no subtitle track with
-disposition default exists, no subtitle track will be marked as default.
-@item passthrough
-In this mode the FlagDefault is set if and only if the AV_DISPOSITION_DEFAULT
-flag is set in the disposition of the corresponding stream.
-@end table
@end table

@anchor{md5}
@@ -1499,10 +1319,6 @@ more efficient), but with this option set, the muxer writes one moof/mdat
 pair for each track, making it easier to separate tracks.

 This option is implicitly set when writing ismv (Smooth Streaming) files.
-@item -movflags skip_sidx
-Skip writing of sidx atom. When bitrate overhead due to sidx atom is high,
-this option could be used for cases where sidx atom is not mandatory.
-When global_sidx flag is enabled, this option will be ignored.
@item -movflags faststart
 Run a second pass moving the index (moov atom) to the beginning of the file.
 This operation can take a while, and will not work in various situations such
@@ -1556,6 +1372,13 @@ point on IIS with this muxer. Example:
 ffmpeg -re @var{<normal input/transcoding options>} -movflags isml+frag_keyframe -f ismv http://server/publishingpoint.isml/Streams(Encoder1)
@end example

+@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 mp3

 The MP3 muxer writes a raw MP3 stream with the following optional features:
@@ -1644,7 +1467,7 @@ Set the program @samp{service_type}. Default is @code{digital_tv}.
 Accepts the following options:
@table @samp
@item hex_value
-Any hexadecimal value between @code{0x01} and @code{0xff} as defined in
+Any hexdecimal value between @code{0x01} to @code{0xff} as defined in
 ETSI 300 468.
@item digital_tv
 Digital TV service.
@@ -1663,14 +1486,11 @@ Advanced Codec Digital HDTV service.
@end table

@item mpegts_pmt_start_pid @var{integer}
-Set the first PID for PMTs. Default is @code{0x1000}, minimum is @code{0x0020},
-maximum is @code{0x1ffa}. This option has no effect in m2ts mode where the PMT
-PID is fixed @code{0x0100}.
+Set the first PID for PMT. Default is @code{0x1000}. Max is @code{0x1f00}.

@item mpegts_start_pid @var{integer}
-Set the first PID for elementary streams. Default is @code{0x0100}, minimum is
-@code{0x0020}, maximum is @code{0x1ffa}. This option has no effect in m2ts mode
-where the elementary stream PIDs are fixed.
+Set the first PID for data packets. Default is @code{0x0100}. Max is
+@code{0x0f00}.

@item mpegts_m2ts_mode @var{boolean}
 Enable m2ts mode if set to @code{1}. Default value is @code{-1} which
@@ -1697,6 +1517,10 @@ Conform to System B (DVB) instead of System A (ATSC).
 Mark the initial packet of each stream as discontinuity.
@end table

+@item resend_headers @var{integer}
+Reemit PAT/PMT before writing the next packet. This option is deprecated:
+use @option{mpegts_flags} instead.
+
@item mpegts_copyts @var{boolean}
 Preserve original timestamps, if value is set to @code{1}. Default value
 is @code{-1}, which results in shifting timestamps so that they start from 0.
@@ -1705,16 +1529,14 @@ is @code{-1}, which results in shifting timestamps so that they start from 0.
 Omit the PES packet length for video packets. Default is @code{1} (true).

@item pcr_period @var{integer}
-Override the default PCR retransmission time in milliseconds. Default is
-@code{-1} which means that the PCR interval will be determined automatically:
-20 ms is used for CBR streams, the highest multiple of the frame duration which
-is less than 100 ms is used for VBR streams.
+Override the default PCR retransmission time in milliseconds. Ignored if
+variable muxrate is selected. Default is @code{20}.

-@item pat_period @var{duration}
-Maximum time in seconds between PAT/PMT tables. Default is @code{0.1}.
+@item pat_period @var{double}
+Maximum time in seconds between PAT/PMT tables.

-@item sdt_period @var{duration}
-Maximum time in seconds between SDT tables. Default is @code{0.5}.
+@item sdt_period @var{double}
+Maximum time in seconds between SDT tables.

@item tables_version @var{integer}
 Set PAT, PMT and SDT version (default @code{0}, valid values are from 0 to 31, inclusively).
@@ -1748,7 +1570,7 @@ ffmpeg -i file.mpg -c copy \
     out.ts
@end example

-@section mxf, mxf_d10, mxf_opatom
+@section mxf, mxf_d10

 MXF muxer.

@@ -1760,7 +1582,7 @@ The muxer options are:
@item store_user_comments @var{bool}
 Set if user comments should be stored if available or never.
 IRT D-10 does not allow user comments. The default is thus to write them for
-mxf and mxf_opatom but not for mxf_d10
+mxf but not for mxf_d10
@end table

@section null
@@ -2150,53 +1972,6 @@ Specify whether to remove all fragments when finished. Default 0 (do not remove)

@end table

-@anchor{streamhash}
-@section streamhash
-
-Per stream hash testing format.
-
-This muxer computes and prints a cryptographic hash of all the input frames,
-on a per-stream basis. This can be used for equality checks without having
-to do a complete binary comparison.
-
-By default audio frames are converted to signed 16-bit raw audio and
-video frames to raw video before computing the hash, but the output
-of explicit conversions to other codecs can also be used. Timestamps
-are ignored. It uses the SHA-256 cryptographic hash function by default,
-but supports several other algorithms.
-
-The output of the muxer consists of one line per stream of the form:
-@var{streamindex},@var{streamtype},@var{algo}=@var{hash}, where
-@var{streamindex} is the index of the mapped stream, @var{streamtype} is a
-single character indicating the type of stream, @var{algo} is a short string
-representing the hash function used, and @var{hash} is a hexadecimal number
-representing the computed hash.
-
-@table @option
-@item hash @var{algorithm}
-Use the cryptographic hash function specified by the string @var{algorithm}.
-Supported values include @code{MD5}, @code{murmur3}, @code{RIPEMD128},
-@code{RIPEMD160}, @code{RIPEMD256}, @code{RIPEMD320}, @code{SHA160},
-@code{SHA224}, @code{SHA256} (default), @code{SHA512/224}, @code{SHA512/256},
-@code{SHA384}, @code{SHA512}, @code{CRC32} and @code{adler32}.
-
-@end table
-
-@subsection Examples
-
-To compute the SHA-256 hash of the input converted to raw audio and
-video, and store it in the file @file{out.sha256}:
-@example
-ffmpeg -i INPUT -f streamhash out.sha256
-@end example
-
-To print an MD5 hash to stdout use the command:
-@example
-ffmpeg -i INPUT -f streamhash -hash md5 -
-@end example
-
-See also the @ref{hash} and @ref{framehash} muxers.
-
@anchor{fifo}
@section fifo

--- a/doc/outdevs.texi
+++ b/doc/outdevs.texi
@@ -140,8 +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.
@@ -155,10 +155,6 @@ Defaults to @option{0.5}.
 Sets the decklink device duplex mode. Must be @samp{unset}, @samp{half} or @samp{full}.
 Defaults to @samp{unset}.

-@item timing_offset
-Sets the genlock timing pixel offset on the used output.
-Defaults to @samp{unset}.
-
@end table

@subsection Examples
@@ -168,7 +164,7 @@ Defaults to @samp{unset}.
@item
 List output devices:
@example
-ffmpeg -sinks decklink
+ffmpeg -i test.avi -f decklink -list_devices 1 dummy
@end example

@item
@@ -220,6 +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.

@@ -329,8 +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.
--- a/doc/protocols.texi
+++ b/doc/protocols.texi
@@ -51,66 +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]
-@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".
-
-Muliple subscribers may stream from the broker using the command:
-@example
-ffplay amqp://[[user]:[password]@@]hostname[:port]
-@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.
-
-@end table
-
@section async

 Asynchronous data filling wrapper for input stream.
@@ -253,20 +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
@@ -288,14 +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.
@@ -311,6 +229,17 @@ 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.
@@ -461,11 +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.
-
@end table

@subsection HTTP Cookies
@@ -1255,7 +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
@@ -1286,7 +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}
@@ -1297,7 +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}
@@ -1342,26 +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 payload_size=@var{bytes}
 Sets the maximum declared size of a packet transferred
 during the single call to the sending function in Live
@@ -1377,7 +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.

@@ -1389,7 +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.
@@ -1407,10 +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
@@ -1504,12 +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}.
-
@end table

 For more information see: @url{https://github.com/Haivision/srt}.
@@ -1711,7 +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
@@ -1820,51 +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
--- a/doc/scaler.texi
+++ b/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
--- a/doc/snow.txt
+++ b/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:
--- a/doc/utils.texi
+++ b/doc/utils.texi
@@ -126,15 +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

@@ -713,8 +704,6 @@ 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
@end table
@@ -931,9 +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}.

--- a/doc/writing_filters.txt
+++ b/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.freenode.net.
+channel, #ffmpeg-devel on irc.libera.chat.
--- a/ffbuild/common.mak
+++ b/ffbuild/common.mak
@@ -38,6 +38,7 @@ OBJCCFLAGS  = $(CPPFLAGS) $(CFLAGS) $(OBJCFLAGS)
 ASFLAGS    := $(CPPFLAGS) $(ASFLAGS)
 CXXFLAGS   := $(CPPFLAGS) $(CFLAGS) $(CXXFLAGS)
 X86ASMFLAGS += $(IFLAGS:%=%/) -I$(<D)/ -Pconfig.asm
+NVCCFLAGS  += -ptx

 HOSTCCFLAGS = $(IFLAGS) $(HOSTCPPFLAGS) $(HOSTCFLAGS)
 LDFLAGS    := $(ALLFFLIBS:%=$(LD_PATH)lib%) $(LDFLAGS)
@@ -90,7 +91,7 @@ COMPILE_NVCC = $(call COMPILE,NVCC)
 %.h.c:
 	$(Q)echo '#include "$*.h"' >$@

-%.ptx: %.cu $(SRC_PATH)/compat/cuda/cuda_runtime.h
+%.ptx: %.cu
 	$(COMPILE_NVCC)

 %.ptx.c: %.ptx
@@ -160,9 +161,9 @@ $(SLIBOBJS): | $(sort $(dir $(SLIBOBJS)))
 $(TESTOBJS): | $(sort $(dir $(TESTOBJS)))
 $(TOOLOBJS): | tools

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

-CLEANSUFFIXES     = *.d *.gcda *.gcno *.h.c *.ho *.map *.o *.pc *.ptx *.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
--- a/fftools/Makefile
+++ b/fftools/Makefile
@@ -10,6 +10,7 @@ ALLAVPROGS   = $(AVBASENAMES:%=%$(PROGSSUF)$(EXESUF))
 ALLAVPROGS_G = $(AVBASENAMES:%=%$(PROGSSUF)_g$(EXESUF))

 OBJS-ffmpeg                        += fftools/ffmpeg_opt.o fftools/ffmpeg_filter.o fftools/ffmpeg_hw.o
+OBJS-ffmpeg-$(CONFIG_CUVID)        += fftools/ffmpeg_cuvid.o
 OBJS-ffmpeg-$(CONFIG_LIBMFX)       += fftools/ffmpeg_qsv.o
 ifndef CONFIG_VIDEOTOOLBOX
 OBJS-ffmpeg-$(CONFIG_VDA)          += fftools/ffmpeg_videotoolbox.o
@@ -31,7 +32,7 @@ $(foreach P,$(AVPROGS-yes),$(eval $(call DOFFTOOL,$(P))))
 all: $(AVPROGS)

 fftools/ffprobe.o fftools/cmdutils.o: libavutil/ffversion.h | fftools
-OUTDIRS += fftools
+OBJDIRS += fftools

 ifdef AVPROGS
 install: install-progs install-data
--- a/fftools/cmdutils.c
+++ b/fftools/cmdutils.c
@@ -55,6 +55,9 @@
 #include "libavutil/ffversion.h"
 #include "libavutil/version.h"
 #include "cmdutils.h"
+#if CONFIG_NETWORK
+#include "libavformat/network.h"
+#endif
 #if HAVE_SYS_RESOURCE_H
 #include <sys/time.h>
 #include <sys/resource.h>
@@ -116,7 +119,7 @@ static void log_callback_report(void *ptr, int level, const char *fmt, va_list v

 void init_dynload(void)
 {
-#if HAVE_SETDLLDIRECTORY && defined(_WIN32)
+#ifdef _WIN32
    /* Calling SetDllDirectory with the empty string (but not NULL) removes the
     * current working directory from the DLL search path as a security pre-caution. */
    SetDllDirectory("");
@@ -179,7 +182,7 @@ void show_help_options(const OptionDef *options, const char *msg, int req_flags,

    first = 1;
    for (po = options; po->name; po++) {
-        char buf[128];
+        char buf[64];

        if (((po->flags & req_flags) != req_flags) ||
            (alt_flags && !(po->flags & alt_flags)) ||
@@ -845,8 +848,8 @@ do {                                                                           \
    }

    if (octx->cur_group.nb_opts || codec_opts || format_opts || resample_opts)
-        av_log(NULL, AV_LOG_WARNING, "Trailing option(s) found in the "
-               "command: may be ignored.\n");
+        av_log(NULL, AV_LOG_WARNING, "Trailing options were found on the "
+               "commandline.\n");

    av_log(NULL, AV_LOG_DEBUG, "Finished splitting the commandline.\n");

@@ -977,7 +980,6 @@ static int init_report(const char *env)
    char *filename_template = NULL;
    char *key, *val;
    int ret, count = 0;
-    int prog_loglevel, envlevel = 0;
    time_t now;
    struct tm *tm;
    AVBPrint filename;
@@ -1009,7 +1011,6 @@ static int init_report(const char *env)
                av_log(NULL, AV_LOG_FATAL, "Invalid report file level\n");
                exit_program(1);
            }
-            envlevel = 1;
        } else {
            av_log(NULL, AV_LOG_ERROR, "Unknown key '%s' in FFREPORT\n", key);
        }
@@ -1026,10 +1027,6 @@ static int init_report(const char *env)
        return AVERROR(ENOMEM);
    }

-    prog_loglevel = av_log_get_level();
-    if (!envlevel)
-        report_file_level = FFMAX(report_file_level, prog_loglevel);
-
    report_file = fopen(filename.str, "w");
    if (!report_file) {
        int ret = AVERROR(errno);
@@ -1040,17 +1037,16 @@ static int init_report(const char *env)
    av_log_set_callback(log_callback_report);
    av_log(NULL, AV_LOG_INFO,
           "%s started on %04d-%02d-%02d at %02d:%02d:%02d\n"
-           "Report written to \"%s\"\n"
-           "Log level: %d\n",
+           "Report written to \"%s\"\n",
           program_name,
           tm->tm_year + 1900, tm->tm_mon + 1, tm->tm_mday,
           tm->tm_hour, tm->tm_min, tm->tm_sec,
-           filename.str, report_file_level);
+           filename.str);
    av_bprint_finalize(&filename, NULL);
    return 0;
 }

-int opt_report(void *optctx, const char *opt, const char *arg)
+int opt_report(const char *opt)
 {
    return init_report(NULL);
 }
@@ -1420,6 +1416,10 @@ static void print_codec(const AVCodec *c)
        printf("threads ");
    if (c->capabilities & AV_CODEC_CAP_AVOID_PROBING)
        printf("avoidprobe ");
+    if (c->capabilities & AV_CODEC_CAP_INTRA_ONLY)
+        printf("intraonly ");
+    if (c->capabilities & AV_CODEC_CAP_LOSSLESS)
+        printf("lossless ");
    if (c->capabilities & AV_CODEC_CAP_HARDWARE)
        printf("hardware ");
    if (c->capabilities & AV_CODEC_CAP_HYBRID)
@@ -1493,14 +1493,13 @@ static char get_media_type_char(enum AVMediaType type)
    }
 }

-static const AVCodec *next_codec_for_id(enum AVCodecID id, void **iter,
+static const AVCodec *next_codec_for_id(enum AVCodecID id, const AVCodec *prev,
                                        int encoder)
 {
-    const AVCodec *c;
-    while ((c = av_codec_iterate(iter))) {
-        if (c->id == id &&
-            (encoder ? av_codec_is_encoder(c) : av_codec_is_decoder(c)))
-            return c;
+    while ((prev = av_codec_next(prev))) {
+        if (prev->id == id &&
+            (encoder ? av_codec_is_encoder(prev) : av_codec_is_decoder(prev)))
+            return prev;
    }
    return NULL;
 }
@@ -1537,12 +1536,11 @@ static unsigned get_codecs_sorted(const AVCodecDescriptor ***rcodecs)

 static void print_codecs_for_id(enum AVCodecID id, int encoder)
 {
-    void *iter = NULL;
-    const AVCodec *codec;
+    const AVCodec *codec = NULL;

    printf(" (%s: ", encoder ? "encoders" : "decoders");

-    while ((codec = next_codec_for_id(id, &iter, encoder)))
+    while ((codec = next_codec_for_id(id, codec, encoder)))
        printf("%s ", codec->name);

    printf(")");
@@ -1565,8 +1563,7 @@ int show_codecs(void *optctx, const char *opt, const char *arg)
           " -------\n");
    for (i = 0; i < nb_codecs; i++) {
        const AVCodecDescriptor *desc = codecs[i];
-        const AVCodec *codec;
-        void *iter = NULL;
+        const AVCodec *codec = NULL;

        if (strstr(desc->name, "_deprecated"))
            continue;
@@ -1584,14 +1581,14 @@ int show_codecs(void *optctx, const char *opt, const char *arg)

        /* print decoders/encoders when there's more than one or their
         * names are different from codec name */
-        while ((codec = next_codec_for_id(desc->id, &iter, 0))) {
+        while ((codec = next_codec_for_id(desc->id, codec, 0))) {
            if (strcmp(codec->name, desc->name)) {
                print_codecs_for_id(desc->id, 0);
                break;
            }
        }
-        iter = NULL;
-        while ((codec = next_codec_for_id(desc->id, &iter, 1))) {
+        codec = NULL;
+        while ((codec = next_codec_for_id(desc->id, codec, 1))) {
            if (strcmp(codec->name, desc->name)) {
                print_codecs_for_id(desc->id, 1);
                break;
@@ -1622,10 +1619,9 @@ static void print_codecs(int encoder)
           encoder ? "Encoders" : "Decoders");
    for (i = 0; i < nb_codecs; i++) {
        const AVCodecDescriptor *desc = codecs[i];
-        const AVCodec *codec;
-        void *iter = NULL;
+        const AVCodec *codec = NULL;

-        while ((codec = next_codec_for_id(desc->id, &iter, encoder))) {
+        while ((codec = next_codec_for_id(desc->id, codec, encoder))) {
            printf(" %c", get_media_type_char(desc->type));
            printf((codec->capabilities & AV_CODEC_CAP_FRAME_THREADS) ? "F" : ".");
            printf((codec->capabilities & AV_CODEC_CAP_SLICE_THREADS) ? "S" : ".");
@@ -1830,10 +1826,9 @@ static void show_help_codec(const char *name, int encoder)
    if (codec)
        print_codec(codec);
    else if ((desc = avcodec_descriptor_get_by_name(name))) {
-        void *iter = NULL;
        int printed = 0;

-        while ((codec = next_codec_for_id(desc->id, &iter, encoder))) {
+        while ((codec = next_codec_for_id(desc->id, codec, encoder))) {
            printed = 1;
            print_codec(codec);
        }
@@ -1868,24 +1863,6 @@ static void show_help_demuxer(const char *name)
        show_help_children(fmt->priv_class, AV_OPT_FLAG_DECODING_PARAM);
 }

-static void show_help_protocol(const char *name)
-{
-    const AVClass *proto_class;
-
-    if (!name) {
-        av_log(NULL, AV_LOG_ERROR, "No protocol name specified.\n");
-        return;
-    }
-
-    proto_class = avio_protocol_get_class(name);
-    if (!proto_class) {
-        av_log(NULL, AV_LOG_ERROR, "Unknown protocol '%s'.\n", name);
-        return;
-    }
-
-    show_help_children(proto_class, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_ENCODING_PARAM);
-}
-
 static void show_help_muxer(const char *name)
 {
    const AVCodecDescriptor *desc;
@@ -2016,8 +1993,6 @@ int show_help(void *optctx, const char *opt, const char *arg)
        show_help_demuxer(par);
    } else if (!strcmp(topic, "muxer")) {
        show_help_muxer(par);
-    } else if (!strcmp(topic, "protocol")) {
-        show_help_protocol(par);
 #if CONFIG_AVFILTER
    } else if (!strcmp(topic, "filter")) {
        show_help_filter(par);
@@ -2057,7 +2032,7 @@ FILE *get_preset_file(char *filename, size_t filename_size,
        av_strlcpy(filename, preset_name, filename_size);
        f = fopen(filename, "r");
    } else {
-#if HAVE_GETMODULEHANDLE && defined(_WIN32)
+#ifdef _WIN32
        char datadir[MAX_PATH], *ls;
        base[2] = NULL;

@@ -2210,7 +2185,7 @@ double get_rotation(AVStream *st)
    if (fabs(theta - 90*round(theta/90)) > 2)
        av_log(NULL, AV_LOG_WARNING, "Odd rotation angle.\n"
               "If you want to help, upload a sample "
-               "of this file to https://streams.videolan.org/upload/ "
+               "of this file to ftp://upload.ffmpeg.org/incoming/ "
               "and contact the ffmpeg-devel mailing list. (ffmpeg-devel@ffmpeg.org)");

    return theta;
--- a/fftools/cmdutils.h
+++ b/fftools/cmdutils.h
@@ -99,7 +99,7 @@ int opt_default(void *optctx, const char *opt, const char *arg);
 */
 int opt_loglevel(void *optctx, const char *opt, const char *arg);

-int opt_report(void *optctx, const char *opt, const char *arg);
+int opt_report(const char *opt);

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

@@ -236,7 +236,7 @@ void show_help_options(const OptionDef *options, const char *msg, int req_flags,
    { "colors",      OPT_EXIT,             { .func_arg = show_colors },      "show available color names" },            \
    { "loglevel",    HAS_ARG,              { .func_arg = opt_loglevel },     "set logging level", "loglevel" },         \
    { "v",           HAS_ARG,              { .func_arg = opt_loglevel },     "set logging level", "loglevel" },         \
-    { "report",      0,                    { .func_arg = opt_report },       "generate a report" },                     \
+    { "report",      0,                    { (void*)opt_report },            "generate a report" },                     \
    { "max_alloc",   HAS_ARG,              { .func_arg = opt_max_alloc },    "set maximum size of a single allocated block", "bytes" }, \
    { "cpuflags",    HAS_ARG | OPT_EXPERT, { .func_arg = opt_cpuflags },     "force specific cpu flags", "flags" },     \
    { "hide_banner", OPT_BOOL | OPT_EXPERT, {&hide_banner},     "do not show program banner", "hide_banner" },          \
--- a/fftools/ffmpeg.c
+++ b/fftools/ffmpeg.c
@@ -182,7 +182,7 @@ static int sub2video_get_blank_frame(InputStream *ist)
    ist->sub2video.frame->width  = ist->dec_ctx->width  ? ist->dec_ctx->width  : ist->sub2video.w;
    ist->sub2video.frame->height = ist->dec_ctx->height ? ist->dec_ctx->height : ist->sub2video.h;
    ist->sub2video.frame->format = AV_PIX_FMT_RGB32;
-    if ((ret = av_frame_get_buffer(frame, 0)) < 0)
+    if ((ret = av_frame_get_buffer(frame, 32)) < 0)
        return ret;
    memset(frame->data[0], 0, frame->height * frame->linesize[0]);
    return 0;
@@ -237,7 +237,7 @@ static void sub2video_push_ref(InputStream *ist, int64_t pts)
    }
 }

-void sub2video_update(InputStream *ist, int64_t heartbeat_pts, AVSubtitle *sub)
+void sub2video_update(InputStream *ist, AVSubtitle *sub)
 {
    AVFrame *frame = ist->sub2video.frame;
    int8_t *dst;
@@ -254,12 +254,7 @@ void sub2video_update(InputStream *ist, int64_t heartbeat_pts, AVSubtitle *sub)
                                 AV_TIME_BASE_Q, ist->st->time_base);
        num_rects = sub->num_rects;
    } else {
-        /* If we are initializing the system, utilize current heartbeat
-           PTS as the start time, and show until the following subpicture
-           is received. Otherwise, utilize the previous subpicture's end time
-           as the fall-back value. */
-        pts       = ist->sub2video.initialize ?
-                    heartbeat_pts : ist->sub2video.end_pts;
+        pts       = ist->sub2video.end_pts;
        end_pts   = INT64_MAX;
        num_rects = 0;
    }
@@ -274,7 +269,6 @@ void sub2video_update(InputStream *ist, int64_t heartbeat_pts, AVSubtitle *sub)
        sub2video_copy_rect(dst, dst_linesize, frame->width, frame->height, sub->rects[i]);
    sub2video_push_ref(ist, pts);
    ist->sub2video.end_pts = end_pts;
-    ist->sub2video.initialize = 0;
 }

 static void sub2video_heartbeat(InputStream *ist, int64_t pts)
@@ -297,11 +291,9 @@ static void sub2video_heartbeat(InputStream *ist, int64_t pts)
        /* do not send the heartbeat frame if the subtitle is already ahead */
        if (pts2 <= ist2->sub2video.last_pts)
            continue;
-        if (pts2 >= ist2->sub2video.end_pts || ist2->sub2video.initialize)
-            /* if we have hit the end of the current displayed subpicture,
-               or if we need to initialize the system, update the
-               overlayed subpicture and its start/end times */
-            sub2video_update(ist2, pts2 + 1, NULL);
+        if (pts2 >= ist2->sub2video.end_pts ||
+            (!ist2->sub2video.frame->data[0] && ist2->sub2video.end_pts < INT64_MAX))
+            sub2video_update(ist2, NULL);
        for (j = 0, nb_reqs = 0; j < ist2->nb_filters; j++)
            nb_reqs += av_buffersrc_get_nb_failed_requests(ist2->filters[j]->filter);
        if (nb_reqs)
@@ -315,7 +307,7 @@ static void sub2video_flush(InputStream *ist)
    int ret;

    if (ist->sub2video.end_pts < INT64_MAX)
-        sub2video_update(ist, INT64_MAX, NULL);
+        sub2video_update(ist, NULL);
    for (i = 0; i < ist->nb_filters; i++) {
        ret = av_buffersrc_add_frame(ist->filters[i]->filter, NULL);
        if (ret != AVERROR_EOF && ret < 0)
@@ -501,37 +493,32 @@ static void ffmpeg_cleanup(int ret)
        FilterGraph *fg = filtergraphs[i];
        avfilter_graph_free(&fg->graph);
        for (j = 0; j < fg->nb_inputs; j++) {
-            InputFilter *ifilter = fg->inputs[j];
-            struct InputStream *ist = ifilter->ist;
-
-            while (av_fifo_size(ifilter->frame_queue)) {
+            while (av_fifo_size(fg->inputs[j]->frame_queue)) {
                AVFrame *frame;
-                av_fifo_generic_read(ifilter->frame_queue, &frame,
+                av_fifo_generic_read(fg->inputs[j]->frame_queue, &frame,
                                     sizeof(frame), NULL);
                av_frame_free(&frame);
            }
-            av_fifo_freep(&ifilter->frame_queue);
-            if (ist->sub2video.sub_queue) {
-                while (av_fifo_size(ist->sub2video.sub_queue)) {
+            av_fifo_freep(&fg->inputs[j]->frame_queue);
+            if (fg->inputs[j]->ist->sub2video.sub_queue) {
+                while (av_fifo_size(fg->inputs[j]->ist->sub2video.sub_queue)) {
                    AVSubtitle sub;
-                    av_fifo_generic_read(ist->sub2video.sub_queue,
+                    av_fifo_generic_read(fg->inputs[j]->ist->sub2video.sub_queue,
                                         &sub, sizeof(sub), NULL);
                    avsubtitle_free(&sub);
                }
-                av_fifo_freep(&ist->sub2video.sub_queue);
+                av_fifo_freep(&fg->inputs[j]->ist->sub2video.sub_queue);
            }
-            av_buffer_unref(&ifilter->hw_frames_ctx);
-            av_freep(&ifilter->name);
+            av_buffer_unref(&fg->inputs[j]->hw_frames_ctx);
+            av_freep(&fg->inputs[j]->name);
            av_freep(&fg->inputs[j]);
        }
        av_freep(&fg->inputs);
        for (j = 0; j < fg->nb_outputs; j++) {
-            OutputFilter *ofilter = fg->outputs[j];
-
-            av_freep(&ofilter->name);
-            av_freep(&ofilter->formats);
-            av_freep(&ofilter->channel_layouts);
-            av_freep(&ofilter->sample_rates);
+            av_freep(&fg->outputs[j]->name);
+            av_freep(&fg->outputs[j]->formats);
+            av_freep(&fg->outputs[j]->channel_layouts);
+            av_freep(&fg->outputs[j]->sample_rates);
            av_freep(&fg->outputs[j]);
        }
        av_freep(&fg->outputs);
@@ -563,7 +550,9 @@ static void ffmpeg_cleanup(int ret)
        if (!ost)
            continue;

-        av_bsf_free(&ost->bsf_ctx);
+        for (j = 0; j < ost->nb_bitstream_filters; j++)
+            av_bsf_free(&ost->bsf_ctx[j]);
+        av_freep(&ost->bsf_ctx);

        av_frame_free(&ost->filtered_frame);
        av_frame_free(&ost->last_frame);
@@ -791,8 +780,6 @@ static void write_packet(OutputFile *of, AVPacket *pkt, OutputStream *ost, int u
            int64_t max = ost->last_mux_dts + !(s->oformat->flags & AVFMT_TS_NONSTRICT);
            if (pkt->dts < max) {
                int loglevel = max - pkt->dts > 2 || st->codecpar->codec_type == AVMEDIA_TYPE_VIDEO ? AV_LOG_WARNING : AV_LOG_DEBUG;
-                if (exit_on_error)
-                    loglevel = AV_LOG_ERROR;
                av_log(s, loglevel, "Non-monotonous DTS in output stream "
                       "%d:%d; previous: %"PRId64", current: %"PRId64"; ",
                       ost->file_index, ost->st->index, ost->last_mux_dts, pkt->dts);
@@ -862,15 +849,40 @@ static void output_packet(OutputFile *of, AVPacket *pkt,
 {
    int ret = 0;

-    /* apply the output bitstream filters */
-    if (ost->bsf_ctx) {
-        ret = av_bsf_send_packet(ost->bsf_ctx, eof ? NULL : pkt);
+    /* apply the output bitstream filters, if any */
+    if (ost->nb_bitstream_filters) {
+        int idx;
+
+        ret = av_bsf_send_packet(ost->bsf_ctx[0], eof ? NULL : pkt);
        if (ret < 0)
            goto finish;
-        while ((ret = av_bsf_receive_packet(ost->bsf_ctx, pkt)) >= 0)
-            write_packet(of, pkt, ost, 0);
-        if (ret == AVERROR(EAGAIN))
-            ret = 0;
+
+        eof = 0;
+        idx = 1;
+        while (idx) {
+            /* get a packet from the previous filter up the chain */
+            ret = av_bsf_receive_packet(ost->bsf_ctx[idx - 1], pkt);
+            if (ret == AVERROR(EAGAIN)) {
+                ret = 0;
+                idx--;
+                continue;
+            } else if (ret == AVERROR_EOF) {
+                eof = 1;
+            } else if (ret < 0)
+                goto finish;
+
+            /* send it to the next filter down the chain or to the muxer */
+            if (idx < ost->nb_bitstream_filters) {
+                ret = av_bsf_send_packet(ost->bsf_ctx[idx], eof ? NULL : pkt);
+                if (ret < 0)
+                    goto finish;
+                idx++;
+                eof = 0;
+            } else if (eof)
+                goto finish;
+            else
+                write_packet(of, pkt, ost, 0);
+        }
    } else if (!eof)
        write_packet(of, pkt, ost, 0);

@@ -1068,7 +1080,6 @@ static void do_video_out(OutputFile *of,

    if (!ost->filters_script &&
        !ost->filters &&
-        (nb_filtergraphs == 0 || !filtergraphs[0]->graph_desc) &&
        next_picture &&
        ist &&
        lrintf(next_picture->pkt_duration * av_q2d(ist->st->time_base) / av_q2d(enc->time_base)) > 0) {
@@ -1125,7 +1136,7 @@ static void do_video_out(OutputFile *of,
                av_log(NULL, AV_LOG_DEBUG, "Not duplicating %d initial frames\n", (int)lrintf(delta0));
                delta = duration;
                delta0 = 0;
-                ost->sync_opts = llrint(sync_ipts);
+                ost->sync_opts = lrint(sync_ipts);
            }
        case VSYNC_CFR:
            // FIXME set to 0.5 after we fix some dts/pts bugs like in avidec.c
@@ -1136,18 +1147,18 @@ static void do_video_out(OutputFile *of,
            else if (delta > 1.1) {
                nb_frames = lrintf(delta);
                if (delta0 > 1.1)
-                    nb0_frames = llrintf(delta0 - 0.6);
+                    nb0_frames = lrintf(delta0 - 0.6);
            }
            break;
        case VSYNC_VFR:
            if (delta <= -0.6)
                nb_frames = 0;
            else if (delta > 0.6)
-                ost->sync_opts = llrint(sync_ipts);
+                ost->sync_opts = lrint(sync_ipts);
            break;
        case VSYNC_DROP:
        case VSYNC_PASSTHROUGH:
-            ost->sync_opts = llrint(sync_ipts);
+            ost->sync_opts = lrint(sync_ipts);
            break;
        default:
            av_assert0(0);
@@ -1183,27 +1194,33 @@ static void do_video_out(OutputFile *of,
    }
    ost->last_dropped = nb_frames == nb0_frames && next_picture;

-    /* duplicates frame if needed */
-    for (i = 0; i < nb_frames; i++) {
-        AVFrame *in_picture;
+  /* duplicates frame if needed */
+  for (i = 0; i < nb_frames; i++) {
+    AVFrame *in_picture;
+    av_init_packet(&pkt);
+    pkt.data = NULL;
+    pkt.size = 0;
+
+    if (i < nb0_frames && ost->last_frame) {
+        in_picture = ost->last_frame;
+    } else
+        in_picture = next_picture;
+
+    if (!in_picture)
+        return;
+
+    in_picture->pts = ost->sync_opts;
+
+#if 1
+    if (!check_recording_time(ost))
+#else
+    if (ost->frame_number >= ost->max_frames)
+#endif
+        return;
+
+    {
        int forced_keyframe = 0;
        double pts_time;
-        av_init_packet(&pkt);
-        pkt.data = NULL;
-        pkt.size = 0;
-
-        if (i < nb0_frames && ost->last_frame) {
-            in_picture = ost->last_frame;
-        } else
-            in_picture = next_picture;
-
-        if (!in_picture)
-            return;
-
-        in_picture->pts = ost->sync_opts;
-
-        if (!check_recording_time(ost))
-            return;

        if (enc->flags & (AV_CODEC_FLAG_INTERLACED_DCT | AV_CODEC_FLAG_INTERLACED_ME) &&
            ost->top_field_first >= 0)
@@ -1254,8 +1271,7 @@ static void do_video_out(OutputFile *of,
            ost->forced_keyframes_expr_const_values[FKF_N] += 1;
        } else if (   ost->forced_keyframes
                   && !strncmp(ost->forced_keyframes, "source", 6)
-                   && in_picture->key_frame==1
-                   && !i) {
+                   && in_picture->key_frame==1) {
            forced_keyframe = 1;
        }

@@ -1277,8 +1293,6 @@ static void do_video_out(OutputFile *of,
        ret = avcodec_send_frame(enc, in_picture);
        if (ret < 0)
            goto error;
-        // Make sure Closed Captions will not be duplicated
-        av_frame_remove_side_data(in_picture, AV_FRAME_DATA_A53_CC);

        while (1) {
            ret = avcodec_receive_packet(enc, &pkt);
@@ -1315,17 +1329,18 @@ static void do_video_out(OutputFile *of,
                fprintf(ost->logfile, "%s", enc->stats_out);
            }
        }
-        ost->sync_opts++;
-        /*
-         * For video, number of frames in == number of packets out.
-         * But there may be reordering, so we can't throw away frames on encoder
-         * flush, we need to limit them here, before they go into encoder.
-         */
-        ost->frame_number++;
-
-        if (vstats_filename && frame_size)
-            do_video_stats(ost, frame_size);
    }
+    ost->sync_opts++;
+    /*
+     * For video, number of frames in == number of packets out.
+     * But there may be reordering, so we can't throw away frames on encoder
+     * flush, we need to limit them here, before they go into encoder.
+     */
+    ost->frame_number++;
+
+    if (vstats_filename && frame_size)
+        do_video_stats(ost, frame_size);
+  }

    if (!ost->last_frame)
        ost->last_frame = av_frame_alloc();
@@ -1478,6 +1493,8 @@ static int reap_filters(int flush)
                    av_rescale_q(filtered_frame->pts, filter_tb, enc->time_base) -
                    av_rescale_q(start_time, AV_TIME_BASE_Q, enc->time_base);
            }
+            //if (ost->source_index >= 0)
+            //    *filtered_frame= *input_streams[ost->source_index]->decoded_frame; //for me_threshold

            switch (av_buffersink_get_type(filter)) {
            case AVMEDIA_TYPE_VIDEO:
@@ -1808,7 +1825,7 @@ static void print_report(int is_last_report, int64_t timer_start, int64_t cur_ti
        } else
            av_log(NULL, AV_LOG_INFO, "%s    %c", buf.str, end);

-        fflush(stderr);
+    fflush(stderr);
    }
    av_bprint_finalize(&buf, NULL);

@@ -1893,6 +1910,9 @@ static void flush_encoders(void)
            }
        }

+        if (enc->codec_type == AVMEDIA_TYPE_AUDIO && enc->frame_size <= 1)
+            continue;
+
        if (enc->codec_type != AVMEDIA_TYPE_VIDEO && enc->codec_type != AVMEDIA_TYPE_AUDIO)
            continue;

@@ -1912,46 +1932,46 @@ static void flush_encoders(void)
                av_assert0(0);
            }

-            av_init_packet(&pkt);
-            pkt.data = NULL;
-            pkt.size = 0;
+                av_init_packet(&pkt);
+                pkt.data = NULL;
+                pkt.size = 0;

-            update_benchmark(NULL);
+                update_benchmark(NULL);

-            while ((ret = avcodec_receive_packet(enc, &pkt)) == AVERROR(EAGAIN)) {
-                ret = avcodec_send_frame(enc, NULL);
-                if (ret < 0) {
+                while ((ret = avcodec_receive_packet(enc, &pkt)) == AVERROR(EAGAIN)) {
+                    ret = avcodec_send_frame(enc, NULL);
+                    if (ret < 0) {
+                        av_log(NULL, AV_LOG_FATAL, "%s encoding failed: %s\n",
+                               desc,
+                               av_err2str(ret));
+                        exit_program(1);
+                    }
+                }
+
+                update_benchmark("flush_%s %d.%d", desc, ost->file_index, ost->index);
+                if (ret < 0 && ret != AVERROR_EOF) {
                    av_log(NULL, AV_LOG_FATAL, "%s encoding failed: %s\n",
                           desc,
                           av_err2str(ret));
                    exit_program(1);
                }
-            }
-
-            update_benchmark("flush_%s %d.%d", desc, ost->file_index, ost->index);
-            if (ret < 0 && ret != AVERROR_EOF) {
-                av_log(NULL, AV_LOG_FATAL, "%s encoding failed: %s\n",
-                       desc,
-                       av_err2str(ret));
-                exit_program(1);
-            }
-            if (ost->logfile && enc->stats_out) {
-                fprintf(ost->logfile, "%s", enc->stats_out);
-            }
-            if (ret == AVERROR_EOF) {
-                output_packet(of, &pkt, ost, 1);
-                break;
-            }
-            if (ost->finished & MUXER_FINISHED) {
-                av_packet_unref(&pkt);
-                continue;
-            }
-            av_packet_rescale_ts(&pkt, enc->time_base, ost->mux_timebase);
-            pkt_size = pkt.size;
-            output_packet(of, &pkt, ost, 0);
-            if (ost->enc_ctx->codec_type == AVMEDIA_TYPE_VIDEO && vstats_filename) {
-                do_video_stats(ost, pkt_size);
-            }
+                if (ost->logfile && enc->stats_out) {
+                    fprintf(ost->logfile, "%s", enc->stats_out);
+                }
+                if (ret == AVERROR_EOF) {
+                    output_packet(of, &pkt, ost, 1);
+                    break;
+                }
+                if (ost->finished & MUXER_FINISHED) {
+                    av_packet_unref(&pkt);
+                    continue;
+                }
+                av_packet_rescale_ts(&pkt, enc->time_base, ost->mux_timebase);
+                pkt_size = pkt.size;
+                output_packet(of, &pkt, ost, 0);
+                if (ost->enc_ctx->codec_type == AVMEDIA_TYPE_VIDEO && vstats_filename) {
+                    do_video_stats(ost, pkt_size);
+                }
        }
    }
 }
@@ -1982,13 +2002,12 @@ static void do_streamcopy(InputStream *ist, OutputStream *ost, const AVPacket *p
    InputFile   *f = input_files [ist->file_index];
    int64_t start_time = (of->start_time == AV_NOPTS_VALUE) ? 0 : of->start_time;
    int64_t ost_tb_start_time = av_rescale_q(start_time, AV_TIME_BASE_Q, ost->mux_timebase);
-    AVPacket opkt;
+    AVPacket opkt = { 0 };
+
+    av_init_packet(&opkt);

    // EOF: flush output bitstream filters.
    if (!pkt) {
-        av_init_packet(&opkt);
-        opkt.data = NULL;
-        opkt.size = 0;
        output_packet(of, &opkt, ost, 1);
        return;
    }
@@ -2027,29 +2046,40 @@ static void do_streamcopy(InputStream *ist, OutputStream *ost, const AVPacket *p
    if (ost->enc_ctx->codec_type == AVMEDIA_TYPE_VIDEO)
        ost->sync_opts++;

-    if (av_packet_ref(&opkt, pkt) < 0)
-        exit_program(1);
-
    if (pkt->pts != AV_NOPTS_VALUE)
        opkt.pts = av_rescale_q(pkt->pts, ist->st->time_base, ost->mux_timebase) - ost_tb_start_time;
+    else
+        opkt.pts = AV_NOPTS_VALUE;

-    if (pkt->dts == AV_NOPTS_VALUE) {
+    if (pkt->dts == AV_NOPTS_VALUE)
        opkt.dts = av_rescale_q(ist->dts, AV_TIME_BASE_Q, ost->mux_timebase);
-    } else if (ost->st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
-        int duration = av_get_audio_frame_duration(ist->dec_ctx, pkt->size);
-        if(!duration)
-            duration = ist->dec_ctx->frame_size;
-        opkt.dts = av_rescale_delta(ist->st->time_base, pkt->dts,
-                                    (AVRational){1, ist->dec_ctx->sample_rate}, duration,
-                                    &ist->filter_in_rescale_delta_last, ost->mux_timebase);
-        /* dts will be set immediately afterwards to what pts is now */
-        opkt.pts = opkt.dts - ost_tb_start_time;
-    } else
+    else
        opkt.dts = av_rescale_q(pkt->dts, ist->st->time_base, ost->mux_timebase);
    opkt.dts -= ost_tb_start_time;

+    if (ost->st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO && pkt->dts != AV_NOPTS_VALUE) {
+        int duration = av_get_audio_frame_duration(ist->dec_ctx, pkt->size);
+        if(!duration)
+            duration = ist->dec_ctx->frame_size;
+        opkt.dts = opkt.pts = av_rescale_delta(ist->st->time_base, pkt->dts,
+                                               (AVRational){1, ist->dec_ctx->sample_rate}, duration, &ist->filter_in_rescale_delta_last,
+                                               ost->mux_timebase) - ost_tb_start_time;
+    }
+
    opkt.duration = av_rescale_q(pkt->duration, ist->st->time_base, ost->mux_timebase);

+    opkt.flags    = pkt->flags;
+
+    if (pkt->buf) {
+        opkt.buf = av_buffer_ref(pkt->buf);
+        if (!opkt.buf)
+            exit_program(1);
+    }
+    opkt.data = pkt->data;
+    opkt.size = pkt->size;
+
+    av_copy_packet_side_data(&opkt, pkt);
+
    output_packet(of, &opkt, ost, 0);
 }

@@ -2290,12 +2320,14 @@ static int decode_audio(InputStream *ist, AVPacket *pkt, int *got_output,
    ist->samples_decoded += decoded_frame->nb_samples;
    ist->frames_decoded++;

+#if 1
    /* increment next_dts to use for the case where the input stream does not
       have timestamps or there are multiple frames in the packet */
    ist->next_pts += ((int64_t)AV_TIME_BASE * decoded_frame->nb_samples) /
                     avctx->sample_rate;
    ist->next_dts += ((int64_t)AV_TIME_BASE * decoded_frame->nb_samples) /
                     avctx->sample_rate;
+#endif

    if (decoded_frame->pts != AV_NOPTS_VALUE) {
        decoded_frame_tb   = ist->st->time_base;
@@ -2370,7 +2402,7 @@ static int decode_video(InputStream *ist, AVPacket *pkt, int *got_output, int64_
            av_log(ist->dec_ctx, AV_LOG_WARNING,
                   "video_delay is larger in decoder than demuxer %d > %d.\n"
                   "If you want to help, upload a sample "
-                   "of this file to https://streams.videolan.org/upload/ "
+                   "of this file to ftp://upload.ffmpeg.org/incoming/ "
                   "and contact the ffmpeg-devel mailing list. (ffmpeg-devel@ffmpeg.org)\n",
                   ist->dec_ctx->has_b_frames,
                   ist->st->codecpar->video_delay);
@@ -2492,7 +2524,7 @@ static int transcode_subtitles(InputStream *ist, AVPacket *pkt, int *got_output,
        return ret;

    if (ist->sub2video.frame) {
-        sub2video_update(ist, INT64_MIN, &subtitle);
+        sub2video_update(ist, &subtitle);
    } else if (ist->nb_filters) {
        if (!ist->sub2video.sub_queue)
            ist->sub2video.sub_queue = av_fifo_alloc(8 * sizeof(AVSubtitle));
@@ -2761,7 +2793,7 @@ static void print_sdp(void)
        if (avio_open2(&sdp_pb, sdp_filename, AVIO_FLAG_WRITE, &int_cb, NULL) < 0) {
            av_log(NULL, AV_LOG_ERROR, "Failed to open sdp file '%s'\n", sdp_filename);
        } else {
-            avio_print(sdp_pb, sdp);
+            avio_printf(sdp_pb, "SDP:\n%s", sdp);
            avio_closep(&sdp_pb);
            av_freep(&sdp_filename);
        }
@@ -2993,28 +3025,35 @@ static int check_init_output_file(OutputFile *of, int file_index)

 static int init_output_bsfs(OutputStream *ost)
 {
-    AVBSFContext *ctx = ost->bsf_ctx;
-    int ret;
+    AVBSFContext *ctx;
+    int i, ret;

-    if (!ctx)
+    if (!ost->nb_bitstream_filters)
        return 0;

-    ret = avcodec_parameters_copy(ctx->par_in, ost->st->codecpar);
-    if (ret < 0)
-        return ret;
+    for (i = 0; i < ost->nb_bitstream_filters; i++) {
+        ctx = ost->bsf_ctx[i];

-    ctx->time_base_in = ost->st->time_base;
+        ret = avcodec_parameters_copy(ctx->par_in,
+                                      i ? ost->bsf_ctx[i - 1]->par_out : ost->st->codecpar);
+        if (ret < 0)
+            return ret;

-    ret = av_bsf_init(ctx);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Error initializing bitstream filter: %s\n",
-               ctx->filter->name);
-        return ret;
+        ctx->time_base_in = i ? ost->bsf_ctx[i - 1]->time_base_out : ost->st->time_base;
+
+        ret = av_bsf_init(ctx);
+        if (ret < 0) {
+            av_log(NULL, AV_LOG_ERROR, "Error initializing bitstream filter: %s\n",
+                   ost->bsf_ctx[i]->filter->name);
+            return ret;
+        }
    }

+    ctx = ost->bsf_ctx[ost->nb_bitstream_filters - 1];
    ret = avcodec_parameters_copy(ost->st->codecpar, ctx->par_out);
    if (ret < 0)
        return ret;
+
    ost->st->time_base = ctx->time_base_out;

    return 0;
@@ -3311,7 +3350,7 @@ static int init_output_stream_encode(OutputStream *ost)
                   "if you want a different framerate.\n",
                   ost->file_index, ost->index);
        }
-
+//      ost->frame_rate = ist->st->avg_frame_rate.num ? ist->st->avg_frame_rate : (AVRational){25, 1};
        if (ost->enc->supported_framerates && !ost->force_fps) {
            int idx = av_find_nearest_q_idx(ost->frame_rate, ost->enc->supported_framerates);
            ost->frame_rate = ost->enc->supported_framerates[idx];
@@ -3346,6 +3385,10 @@ static int init_output_stream_encode(OutputStream *ost)
            av_log(oc, AV_LOG_WARNING, "Frame rate very high for a muxer not efficiently supporting it.\n"
                                       "Please consider specifying a lower framerate, a different muxer or -vsync 2\n");
        }
+        for (j = 0; j < ost->forced_kf_count; j++)
+            ost->forced_kf_pts[j] = av_rescale_q(ost->forced_kf_pts[j],
+                                                 AV_TIME_BASE_Q,
+                                                 enc_ctx->time_base);

        enc_ctx->width  = av_buffersink_get_w(ost->filter->filter);
        enc_ctx->height = av_buffersink_get_h(ost->filter->filter);
@@ -3390,8 +3433,8 @@ static int init_output_stream_encode(OutputStream *ost)
                ost->forced_keyframes_expr_const_values[FKF_PREV_FORCED_N] = NAN;
                ost->forced_keyframes_expr_const_values[FKF_PREV_FORCED_T] = NAN;

-                // Don't parse the 'forced_keyframes' in case of 'keep-source-keyframes',
-                // parse it only for static kf timings
+            // Don't parse the 'forced_keyframes' in case of 'keep-source-keyframes',
+            // parse it only for static kf timings
            } else if(strncmp(ost->forced_keyframes, "source", 6)) {
                parse_forced_key_frames(ost->forced_keyframes, ost, ost->enc_ctx);
            }
@@ -3447,14 +3490,21 @@ static int init_output_stream(OutputStream *ost, char *error, int error_len)
            !av_dict_get(ost->encoder_opts, "ab", NULL, 0))
            av_dict_set(&ost->encoder_opts, "b", "128000", 0);

-        ret = hw_device_setup_for_encode(ost);
-        if (ret < 0) {
-            snprintf(error, error_len, "Device setup failed for "
-                     "encoder on output stream #%d:%d : %s",
+        if (ost->filter && av_buffersink_get_hw_frames_ctx(ost->filter->filter) &&
+            ((AVHWFramesContext*)av_buffersink_get_hw_frames_ctx(ost->filter->filter)->data)->format ==
+            av_buffersink_get_format(ost->filter->filter)) {
+            ost->enc_ctx->hw_frames_ctx = av_buffer_ref(av_buffersink_get_hw_frames_ctx(ost->filter->filter));
+            if (!ost->enc_ctx->hw_frames_ctx)
+                return AVERROR(ENOMEM);
+        } else {
+            ret = hw_device_setup_for_encode(ost);
+            if (ret < 0) {
+                snprintf(error, error_len, "Device setup failed for "
+                         "encoder on output stream #%d:%d : %s",
                     ost->file_index, ost->index, av_err2str(ret));
-            return ret;
+                return ret;
+            }
        }
-
        if (ist && ist->dec->type == AVMEDIA_TYPE_SUBTITLE && ost->enc->type == AVMEDIA_TYPE_SUBTITLE) {
            int input_props = 0, output_props = 0;
            AVCodecDescriptor const *input_descriptor =
@@ -3530,14 +3580,12 @@ static int init_output_stream(OutputStream *ost, char *error, int error_len)
            int i;
            for (i = 0; i < ist->st->nb_side_data; i++) {
                AVPacketSideData *sd = &ist->st->side_data[i];
-                if (sd->type != AV_PKT_DATA_CPB_PROPERTIES) {
-                    uint8_t *dst = av_stream_new_side_data(ost->st, sd->type, sd->size);
-                    if (!dst)
-                        return AVERROR(ENOMEM);
-                    memcpy(dst, sd->data, sd->size);
-                    if (ist->autorotate && sd->type == AV_PKT_DATA_DISPLAYMATRIX)
-                        av_display_rotation_set((uint32_t *)dst, 0);
-                }
+                uint8_t *dst = av_stream_new_side_data(ost->st, sd->type, sd->size);
+                if (!dst)
+                    return AVERROR(ENOMEM);
+                memcpy(dst, sd->data, sd->size);
+                if (ist->autorotate && sd->type == AV_PKT_DATA_DISPLAYMATRIX)
+                    av_display_rotation_set((uint32_t *)dst, 0);
            }
        }

@@ -3836,9 +3884,7 @@ static OutputStream *choose_output(void)
                       av_rescale_q(ost->st->cur_dts, ost->st->time_base,
                                    AV_TIME_BASE_Q);
        if (ost->st->cur_dts == AV_NOPTS_VALUE)
-            av_log(NULL, AV_LOG_DEBUG,
-                "cur_dts is invalid st:%d (%d) [init:%d i_done:%d finish:%d] (this is harmless if it occurs once at the start per stream)\n",
-                ost->st->index, ost->st->id, ost->initialized, ost->inputs_done, ost->finished);
+            av_log(NULL, AV_LOG_DEBUG, "cur_dts is invalid (this is harmless if it occurs once at the start per stream)\n");

        if (!ost->initialized && !ost->inputs_done)
            return ost;
@@ -4131,7 +4177,7 @@ static void reset_eagain(void)

 // set duration to max(tmp, duration) in a proper time base and return duration's time_base
 static AVRational duration_max(int64_t tmp, int64_t *duration, AVRational tmp_time_base,
-                               AVRational time_base)
+                                AVRational time_base)
 {
    int ret;

@@ -4156,7 +4202,7 @@ static int seek_to_start(InputFile *ifile, AVFormatContext *is)
    int i, ret, has_audio = 0;
    int64_t duration = 0;

-    ret = avformat_seek_file(is, -1, INT64_MIN, is->start_time, is->start_time, 0);
+    ret = av_seek_frame(is, -1, is->start_time, 0);
    if (ret < 0)
        return ret;

@@ -4224,7 +4270,6 @@ static int process_input(int file_index)
    int ret, thread_ret, i, j;
    int64_t duration;
    int64_t pkt_dts;
-    int disable_discontinuity_correction = copy_ts;

    is  = ifile->ctx;
    ret = get_input_packet(ifile, &pkt);
@@ -4426,20 +4471,10 @@ static int process_input(int file_index)
        pkt.dts += duration;

    pkt_dts = av_rescale_q_rnd(pkt.dts, ist->st->time_base, AV_TIME_BASE_Q, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
-
-    if (copy_ts && pkt_dts != AV_NOPTS_VALUE && ist->next_dts != AV_NOPTS_VALUE &&
-        (is->iformat->flags & AVFMT_TS_DISCONT) && ist->st->pts_wrap_bits < 60) {
-        int64_t wrap_dts = av_rescale_q_rnd(pkt.dts + (1LL<<ist->st->pts_wrap_bits),
-                                            ist->st->time_base, AV_TIME_BASE_Q,
-                                            AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
-        if (FFABS(wrap_dts - ist->next_dts) < FFABS(pkt_dts - ist->next_dts)/10)
-            disable_discontinuity_correction = 0;
-    }
-
    if ((ist->dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO ||
         ist->dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) &&
         pkt_dts != AV_NOPTS_VALUE && ist->next_dts != AV_NOPTS_VALUE &&
-        !disable_discontinuity_correction) {
+        !copy_ts) {
        int64_t delta   = pkt_dts - ist->next_dts;
        if (is->iformat->flags & AVFMT_TS_DISCONT) {
            if (delta < -1LL*dts_delta_threshold*AV_TIME_BASE ||
@@ -4447,10 +4482,7 @@ static int process_input(int file_index)
                pkt_dts + AV_TIME_BASE/10 < FFMAX(ist->pts, ist->dts)) {
                ifile->ts_offset -= delta;
                av_log(NULL, AV_LOG_DEBUG,
-                       "timestamp discontinuity for stream #%d:%d "
-                       "(id=%d, type=%s): %"PRId64", new offset= %"PRId64"\n",
-                       ist->file_index, ist->st->index, ist->st->id,
-                       av_get_media_type_string(ist->dec_ctx->codec_type),
+                       "timestamp discontinuity %"PRId64", new offset= %"PRId64"\n",
                       delta, ifile->ts_offset);
                pkt.dts -= av_rescale_q(delta, AV_TIME_BASE_Q, ist->st->time_base);
                if (pkt.pts != AV_NOPTS_VALUE)
@@ -4713,10 +4745,6 @@ static int transcode(void)
            av_freep(&ost->enc_ctx->stats_in);
        }
        total_packets_written += ost->packets_written;
-        if (!ost->packets_written && (abort_on_flags & ABORT_ON_FLAG_EMPTY_OUTPUT_STREAM)) {
-            av_log(NULL, AV_LOG_FATAL, "Empty output on stream %d.\n", i);
-            exit_program(1);
-        }
    }

    if (!total_packets_written && (abort_on_flags & ABORT_ON_FLAG_EMPTY_OUTPUT)) {
@@ -4734,6 +4762,7 @@ static int transcode(void)
        }
    }

+    av_buffer_unref(&hw_device_ctx);
    hw_device_free_all();

    /* finished ! */
@@ -4861,6 +4890,11 @@ int main(int argc, char **argv)
        exit_program(1);
    }

+//     if (nb_input_files == 0) {
+//         av_log(NULL, AV_LOG_FATAL, "At least one input file must be specified\n");
+//         exit_program(1);
+//     }
+
    for (i = 0; i < nb_output_files; i++) {
        if (strcmp(output_files[i]->ctx->oformat->name, "rtp"))
            want_sdp = 0;
--- a/fftools/ffmpeg.h
+++ b/fftools/ffmpeg.h
@@ -61,6 +61,7 @@ enum HWAccelID {
    HWACCEL_GENERIC,
    HWACCEL_VIDEOTOOLBOX,
    HWACCEL_QSV,
+    HWACCEL_CUVID,
 };

 typedef struct HWAccel {
@@ -71,7 +72,7 @@ typedef struct HWAccel {
 } HWAccel;

 typedef struct HWDevice {
-    const char *name;
+    char *name;
    enum AVHWDeviceType type;
    AVBufferRef *device_ref;
 } HWDevice;
@@ -348,7 +349,6 @@ typedef struct InputStream {
        AVFifoBuffer *sub_queue;    ///< queue of AVSubtitle* before filter init
        AVFrame *frame;
        int w, h;
-        unsigned int initialize; ///< marks if sub2video_update should force an initialization
    } sub2video;

    int dr1;
@@ -430,8 +430,7 @@ enum forced_keyframes_const {
    FKF_NB
 };

-#define ABORT_ON_FLAG_EMPTY_OUTPUT        (1 <<  0)
-#define ABORT_ON_FLAG_EMPTY_OUTPUT_STREAM (1 <<  1)
+#define ABORT_ON_FLAG_EMPTY_OUTPUT (1 <<  0)

 extern const char *const forced_keyframes_const_names[];

@@ -460,7 +459,8 @@ typedef struct OutputStream {
    AVRational mux_timebase;
    AVRational enc_timebase;

-    AVBSFContext            *bsf_ctx;
+    int                    nb_bitstream_filters;
+    AVBSFContext            **bsf_ctx;

    AVCodecContext *enc_ctx;
    AVCodecParameters *ref_par; /* associated input codec parameters with encoders options applied */
@@ -615,6 +615,7 @@ extern const AVIOInterruptCB int_cb;

 extern const OptionDef options[];
 extern const HWAccel hwaccels[];
+extern AVBufferRef *hw_device_ctx;
 #if CONFIG_QSV
 extern char *qsv_device;
 #endif
@@ -645,7 +646,7 @@ int filtergraph_is_simple(FilterGraph *fg);
 int init_simple_filtergraph(InputStream *ist, OutputStream *ost);
 int init_complex_filtergraph(FilterGraph *fg);

-void sub2video_update(InputStream *ist, int64_t heartbeat_pts, AVSubtitle *sub);
+void sub2video_update(InputStream *ist, AVSubtitle *sub);

 int ifilter_parameters_from_frame(InputFilter *ifilter, const AVFrame *frame);

@@ -653,6 +654,7 @@ int ffmpeg_parse_options(int argc, char **argv);

 int videotoolbox_init(AVCodecContext *s);
 int qsv_init(AVCodecContext *s);
+int cuvid_init(AVCodecContext *s);

 HWDevice *hw_device_get_by_name(const char *name);
 int hw_device_init_from_string(const char *arg, HWDevice **dev);
@@ -660,7 +662,6 @@ void hw_device_free_all(void);

 int hw_device_setup_for_decode(InputStream *ist);
 int hw_device_setup_for_encode(OutputStream *ost);
-int hw_device_setup_for_filter(FilterGraph *fg);

 int hwaccel_decode_init(AVCodecContext *avctx);

--- a/fftools/ffmpeg_cuvid.c
+++ b/fftools/ffmpeg_cuvid.c
@@ -0,0 +1,73 @@
+/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "libavutil/hwcontext.h"
+#include "libavutil/pixdesc.h"
+
+#include "ffmpeg.h"
+
+static void cuvid_uninit(AVCodecContext *avctx)
+{
+    InputStream *ist = avctx->opaque;
+    av_buffer_unref(&ist->hw_frames_ctx);
+}
+
+int cuvid_init(AVCodecContext *avctx)
+{
+    InputStream *ist = avctx->opaque;
+    AVHWFramesContext *frames_ctx;
+    int ret;
+
+    av_log(avctx, AV_LOG_VERBOSE, "Initializing cuvid hwaccel\n");
+
+    if (!hw_device_ctx) {
+        ret = av_hwdevice_ctx_create(&hw_device_ctx, AV_HWDEVICE_TYPE_CUDA,
+                                     ist->hwaccel_device, NULL, 0);
+        if (ret < 0) {
+            av_log(avctx, AV_LOG_ERROR, "Error creating a CUDA device\n");
+            return ret;
+        }
+    }
+
+    av_buffer_unref(&ist->hw_frames_ctx);
+    ist->hw_frames_ctx = av_hwframe_ctx_alloc(hw_device_ctx);
+    if (!ist->hw_frames_ctx) {
+        av_log(avctx, AV_LOG_ERROR, "Error creating a CUDA frames context\n");
+        return AVERROR(ENOMEM);
+    }
+
+    frames_ctx = (AVHWFramesContext*)ist->hw_frames_ctx->data;
+
+    frames_ctx->format = AV_PIX_FMT_CUDA;
+    frames_ctx->sw_format = avctx->sw_pix_fmt;
+    frames_ctx->width = avctx->width;
+    frames_ctx->height = avctx->height;
+
+    av_log(avctx, AV_LOG_DEBUG, "Initializing CUDA frames context: sw_format = %s, width = %d, height = %d\n",
+           av_get_pix_fmt_name(frames_ctx->sw_format), frames_ctx->width, frames_ctx->height);
+
+    ret = av_hwframe_ctx_init(ist->hw_frames_ctx);
+    if (ret < 0) {
+        av_log(avctx, AV_LOG_ERROR, "Error initializing a CUDA frame pool\n");
+        return ret;
+    }
+
+    ist->hwaccel_uninit = cuvid_uninit;
+
+    return 0;
+}
--- a/fftools/ffmpeg_filter.c
+++ b/fftools/ffmpeg_filter.c
@@ -99,8 +99,7 @@ void choose_sample_fmt(AVStream *st, AVCodec *codec)
                break;
        }
        if (*p == -1) {
-            const AVCodecDescriptor *desc = avcodec_descriptor_get(codec->id);
-            if(desc && (desc->props & AV_CODEC_PROP_LOSSLESS) && av_get_sample_fmt_name(st->codecpar->format) > av_get_sample_fmt_name(codec->sample_fmts[0]))
+            if((codec->capabilities & AV_CODEC_CAP_LOSSLESS) && av_get_sample_fmt_name(st->codecpar->format) > av_get_sample_fmt_name(codec->sample_fmts[0]))
                av_log(NULL, AV_LOG_ERROR, "Conversion will not be lossless.\n");
            if(av_get_sample_fmt_name(st->codecpar->format))
            av_log(NULL, AV_LOG_WARNING,
@@ -294,17 +293,10 @@ static void init_input_filter(FilterGraph *fg, AVFilterInOut *in)
            exit_program(1);
        }
        ist = input_streams[input_files[file_idx]->ist_index + st->index];
-        if (ist->user_set_discard == AVDISCARD_ALL) {
-            av_log(NULL, AV_LOG_FATAL, "Stream specifier '%s' in filtergraph description %s "
-                   "matches a disabled input stream.\n", p, fg->graph_desc);
-            exit_program(1);
-        }
    } else {
        /* find the first unused stream of corresponding type */
        for (i = 0; i < nb_input_streams; i++) {
            ist = input_streams[i];
-            if (ist->user_set_discard == AVDISCARD_ALL)
-                continue;
            if (ist->dec_ctx->codec_type == type && ist->discard)
                break;
        }
@@ -740,13 +732,6 @@ static int sub2video_prepare(InputStream *ist, InputFilter *ifilter)
    if (!ist->sub2video.frame)
        return AVERROR(ENOMEM);
    ist->sub2video.last_pts = INT64_MIN;
-    ist->sub2video.end_pts  = INT64_MIN;
-
-    /* sub2video structure has been (re-)initialized.
-       Mark it as such so that the system will be
-       initialized with the first received heartbeat. */
-    ist->sub2video.initialize = 1;
-
    return 0;
 }

@@ -793,9 +778,10 @@ static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter,
    av_bprint_init(&args, 0, AV_BPRINT_SIZE_AUTOMATIC);
    av_bprintf(&args,
             "video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:"
-             "pixel_aspect=%d/%d",
+             "pixel_aspect=%d/%d:sws_param=flags=%d",
             ifilter->width, ifilter->height, ifilter->format,
-             tb.num, tb.den, sar.num, sar.den);
+             tb.num, tb.den, sar.num, sar.den,
+             SWS_BILINEAR + ((ist->dec_ctx->flags&AV_CODEC_FLAG_BITEXACT) ? SWS_BITEXACT:0));
    if (fr.num && fr.den)
        av_bprintf(&args, ":frame_rate=%d/%d", fr.num, fr.den);
    snprintf(name, sizeof(name), "graph %d input from stream %d:%d", fg->index,
@@ -1062,9 +1048,17 @@ int configure_filtergraph(FilterGraph *fg)
    if ((ret = avfilter_graph_parse2(fg->graph, graph_desc, &inputs, &outputs)) < 0)
        goto fail;

-    ret = hw_device_setup_for_filter(fg);
-    if (ret < 0)
-        goto fail;
+    if (filter_hw_device || hw_device_ctx) {
+        AVBufferRef *device = filter_hw_device ? filter_hw_device->device_ref
+                                               : hw_device_ctx;
+        for (i = 0; i < fg->graph->nb_filters; i++) {
+            fg->graph->filters[i]->hw_device_ctx = av_buffer_ref(device);
+            if (!fg->graph->filters[i]->hw_device_ctx) {
+                ret = AVERROR(ENOMEM);
+                goto fail;
+            }
+        }
+    }

    if (simple && (!inputs || inputs->next || !outputs || outputs->next)) {
        const char *num_inputs;
@@ -1167,7 +1161,7 @@ int configure_filtergraph(FilterGraph *fg)
            while (av_fifo_size(ist->sub2video.sub_queue)) {
                AVSubtitle tmp;
                av_fifo_generic_read(ist->sub2video.sub_queue, &tmp, sizeof(tmp), NULL);
-                sub2video_update(ist, INT64_MIN, &tmp);
+                sub2video_update(ist, &tmp);
                avsubtitle_free(&tmp);
            }
        }
--- a/fftools/ffmpeg_hw.c
+++ b/fftools/ffmpeg_hw.c
@@ -19,8 +19,6 @@
 #include <string.h>

 #include "libavutil/avstring.h"
-#include "libavutil/pixdesc.h"
-#include "libavfilter/buffersink.h"

 #include "ffmpeg.h"

@@ -101,7 +99,7 @@ int hw_device_init_from_string(const char *arg, HWDevice **dev_out)
    // -> av_hwdevice_ctx_create_derived()

    AVDictionary *options = NULL;
-    const char *type_name = NULL, *name = NULL, *device = NULL;
+    char *type_name = NULL, *name = NULL, *device = NULL;
    enum AVHWDeviceType type;
    HWDevice *dev, *src;
    AVBufferRef *device_ref = NULL;
@@ -157,12 +155,10 @@ int hw_device_init_from_string(const char *arg, HWDevice **dev_out)
        ++p;
        q = strchr(p, ',');
        if (q) {
-            if (q - p > 0) {
-                device = av_strndup(p, q - p);
-                if (!device) {
-                    err = AVERROR(ENOMEM);
-                    goto fail;
-                }
+            device = av_strndup(p, q - p);
+            if (!device) {
+                err = AVERROR(ENOMEM);
+                goto fail;
            }
            err = av_dict_parse_string(&options, q + 1, "=", ",", 0);
            if (err < 0) {
@@ -172,8 +168,7 @@ int hw_device_init_from_string(const char *arg, HWDevice **dev_out)
        }

        err = av_hwdevice_ctx_create(&device_ref, type,
-                                     q ? device : p[0] ? p : NULL,
-                                     options, 0);
+                                     device ? device : p, options, 0);
        if (err < 0)
            goto fail;

@@ -418,57 +413,18 @@ int hw_device_setup_for_decode(InputStream *ist)

 int hw_device_setup_for_encode(OutputStream *ost)
 {
-    const AVCodecHWConfig *config;
-    HWDevice *dev = NULL;
-    AVBufferRef *frames_ref = NULL;
-    int i;
-
-    if (ost->filter) {
-        frames_ref = av_buffersink_get_hw_frames_ctx(ost->filter->filter);
-        if (frames_ref &&
-            ((AVHWFramesContext*)frames_ref->data)->format ==
-            ost->enc_ctx->pix_fmt) {
-            // Matching format, will try to use hw_frames_ctx.
-        } else {
-            frames_ref = NULL;
-        }
-    }
-
-    for (i = 0;; i++) {
-        config = avcodec_get_hw_config(ost->enc, i);
-        if (!config)
-            break;
-
-        if (frames_ref &&
-            config->methods & AV_CODEC_HW_CONFIG_METHOD_HW_FRAMES_CTX &&
-            (config->pix_fmt == AV_PIX_FMT_NONE ||
-             config->pix_fmt == ost->enc_ctx->pix_fmt)) {
-            av_log(ost->enc_ctx, AV_LOG_VERBOSE, "Using input "
-                   "frames context (format %s) with %s encoder.\n",
-                   av_get_pix_fmt_name(ost->enc_ctx->pix_fmt),
-                   ost->enc->name);
-            ost->enc_ctx->hw_frames_ctx = av_buffer_ref(frames_ref);
-            if (!ost->enc_ctx->hw_frames_ctx)
-                return AVERROR(ENOMEM);
-            return 0;
-        }
-
-        if (!dev &&
-            config->methods & AV_CODEC_HW_CONFIG_METHOD_HW_DEVICE_CTX)
-            dev = hw_device_get_by_type(config->device_type);
-    }
+    HWDevice *dev;

+    dev = hw_device_match_by_codec(ost->enc);
    if (dev) {
-        av_log(ost->enc_ctx, AV_LOG_VERBOSE, "Using device %s "
-               "(type %s) with %s encoder.\n", dev->name,
-               av_hwdevice_get_type_name(dev->type), ost->enc->name);
        ost->enc_ctx->hw_device_ctx = av_buffer_ref(dev->device_ref);
        if (!ost->enc_ctx->hw_device_ctx)
            return AVERROR(ENOMEM);
+        return 0;
    } else {
        // No device required, or no device available.
+        return 0;
    }
-    return 0;
 }

 static int hwaccel_retrieve_data(AVCodecContext *avctx, AVFrame *input)
@@ -521,31 +477,3 @@ int hwaccel_decode_init(AVCodecContext *avctx)

    return 0;
 }
-
-int hw_device_setup_for_filter(FilterGraph *fg)
-{
-    HWDevice *dev;
-    int i;
-
-    // If the user has supplied exactly one hardware device then just
-    // give it straight to every filter for convenience.  If more than
-    // one device is available then the user needs to pick one explcitly
-    // with the filter_hw_device option.
-    if (filter_hw_device)
-        dev = filter_hw_device;
-    else if (nb_hw_devices == 1)
-        dev = hw_devices[0];
-    else
-        dev = NULL;
-
-    if (dev) {
-        for (i = 0; i < fg->graph->nb_filters; i++) {
-            fg->graph->filters[i]->hw_device_ctx =
-                av_buffer_ref(dev->device_ref);
-            if (!fg->graph->filters[i]->hw_device_ctx)
-                return AVERROR(ENOMEM);
-        }
-    }
-
-    return 0;
-}
--- a/fftools/ffmpeg_opt.c
+++ b/fftools/ffmpeg_opt.c
@@ -44,80 +44,16 @@

 #define DEFAULT_PASS_LOGFILENAME_PREFIX "ffmpeg2pass"

-#define SPECIFIER_OPT_FMT_str  "%s"
-#define SPECIFIER_OPT_FMT_i    "%i"
-#define SPECIFIER_OPT_FMT_i64  "%"PRId64
-#define SPECIFIER_OPT_FMT_ui64 "%"PRIu64
-#define SPECIFIER_OPT_FMT_f    "%f"
-#define SPECIFIER_OPT_FMT_dbl  "%lf"
-
-static const char *opt_name_codec_names[]               = {"c", "codec", "acodec", "vcodec", "scodec", "dcodec", NULL};
-static const char *opt_name_audio_channels[]            = {"ac", NULL};
-static const char *opt_name_audio_sample_rate[]         = {"ar", NULL};
-static const char *opt_name_frame_rates[]               = {"r", NULL};
-static const char *opt_name_frame_sizes[]               = {"s", NULL};
-static const char *opt_name_frame_pix_fmts[]            = {"pix_fmt", NULL};
-static const char *opt_name_ts_scale[]                  = {"itsscale", NULL};
-static const char *opt_name_hwaccels[]                  = {"hwaccel", NULL};
-static const char *opt_name_hwaccel_devices[]           = {"hwaccel_device", NULL};
-static const char *opt_name_hwaccel_output_formats[]    = {"hwaccel_output_format", NULL};
-static const char *opt_name_autorotate[]                = {"autorotate", NULL};
-static const char *opt_name_max_frames[]                = {"frames", "aframes", "vframes", "dframes", NULL};
-static const char *opt_name_bitstream_filters[]         = {"bsf", "absf", "vbsf", NULL};
-static const char *opt_name_codec_tags[]                = {"tag", "atag", "vtag", "stag", NULL};
-static const char *opt_name_sample_fmts[]               = {"sample_fmt", NULL};
-static const char *opt_name_qscale[]                    = {"q", "qscale", NULL};
-static const char *opt_name_forced_key_frames[]         = {"forced_key_frames", NULL};
-static const char *opt_name_force_fps[]                 = {"force_fps", NULL};
-static const char *opt_name_frame_aspect_ratios[]       = {"aspect", NULL};
-static const char *opt_name_rc_overrides[]              = {"rc_override", NULL};
-static const char *opt_name_intra_matrices[]            = {"intra_matrix", NULL};
-static const char *opt_name_inter_matrices[]            = {"inter_matrix", NULL};
-static const char *opt_name_chroma_intra_matrices[]     = {"chroma_intra_matrix", NULL};
-static const char *opt_name_top_field_first[]           = {"top", NULL};
-static const char *opt_name_presets[]                   = {"pre", "apre", "vpre", "spre", NULL};
-static const char *opt_name_copy_initial_nonkeyframes[] = {"copyinkfr", NULL};
-static const char *opt_name_copy_prior_start[]          = {"copypriorss", NULL};
-static const char *opt_name_filters[]                   = {"filter", "af", "vf", NULL};
-static const char *opt_name_filter_scripts[]            = {"filter_script", NULL};
-static const char *opt_name_reinit_filters[]            = {"reinit_filter", NULL};
-static const char *opt_name_fix_sub_duration[]          = {"fix_sub_duration", NULL};
-static const char *opt_name_canvas_sizes[]              = {"canvas_size", NULL};
-static const char *opt_name_pass[]                      = {"pass", NULL};
-static const char *opt_name_passlogfiles[]              = {"passlogfile", NULL};
-static const char *opt_name_max_muxing_queue_size[]     = {"max_muxing_queue_size", NULL};
-static const char *opt_name_guess_layout_max[]          = {"guess_layout_max", NULL};
-static const char *opt_name_apad[]                      = {"apad", NULL};
-static const char *opt_name_discard[]                   = {"discard", NULL};
-static const char *opt_name_disposition[]               = {"disposition", NULL};
-static const char *opt_name_time_bases[]                = {"time_base", NULL};
-static const char *opt_name_enc_time_bases[]            = {"enc_time_base", NULL};
-
-#define WARN_MULTIPLE_OPT_USAGE(name, type, so, st)\
-{\
-    char namestr[128] = "";\
-    const char *spec = so->specifier && so->specifier[0] ? so->specifier : "";\
-    for (i = 0; opt_name_##name[i]; i++)\
-        av_strlcatf(namestr, sizeof(namestr), "-%s%s", opt_name_##name[i], opt_name_##name[i+1] ? (opt_name_##name[i+2] ? ", " : " or ") : "");\
-    av_log(NULL, AV_LOG_WARNING, "Multiple %s options specified for stream %d, only the last option '-%s%s%s "SPECIFIER_OPT_FMT_##type"' will be used.\n",\
-           namestr, st->index, opt_name_##name[0], spec[0] ? ":" : "", spec, so->u.type);\
-}
-
 #define MATCH_PER_STREAM_OPT(name, type, outvar, fmtctx, st)\
 {\
-    int i, ret, matches = 0;\
-    SpecifierOpt *so;\
+    int i, ret;\
    for (i = 0; i < o->nb_ ## name; i++) {\
        char *spec = o->name[i].specifier;\
-        if ((ret = check_stream_specifier(fmtctx, st, spec)) > 0) {\
+        if ((ret = check_stream_specifier(fmtctx, st, spec)) > 0)\
            outvar = o->name[i].u.type;\
-            so = &o->name[i];\
-            matches++;\
-        } else if (ret < 0)\
+        else if (ret < 0)\
            exit_program(1);\
    }\
-    if (matches > 1)\
-       WARN_MULTIPLE_OPT_USAGE(name, type, so, st);\
 }

 #define MATCH_PER_TYPE_OPT(name, type, outvar, fmtctx, mediatype)\
@@ -136,9 +72,13 @@ const HWAccel hwaccels[] = {
 #endif
 #if CONFIG_LIBMFX
    { "qsv",   qsv_init,   HWACCEL_QSV,   AV_PIX_FMT_QSV },
+#endif
+#if CONFIG_CUVID
+    { "cuvid", cuvid_init, HWACCEL_CUVID, AV_PIX_FMT_CUDA },
 #endif
    { 0 },
 };
+AVBufferRef *hw_device_ctx;
 HWDevice *filter_hw_device;

 char *vstats_filename;
@@ -232,11 +172,14 @@ static void init_options(OptionsContext *o)
 static int show_hwaccels(void *optctx, const char *opt, const char *arg)
 {
    enum AVHWDeviceType type = AV_HWDEVICE_TYPE_NONE;
+    int i;

    printf("Hardware acceleration methods:\n");
    while ((type = av_hwdevice_iterate_types(type)) !=
           AV_HWDEVICE_TYPE_NONE)
        printf("%s\n", av_hwdevice_get_type_name(type));
+    for (i = 0; hwaccels[i].name; i++)
+        printf("%s\n", hwaccels[i].name);
    printf("\n");
    return 0;
 }
@@ -262,9 +205,8 @@ static AVDictionary *strip_specifiers(AVDictionary *dict)
 static int opt_abort_on(void *optctx, const char *opt, const char *arg)
 {
    static const AVOption opts[] = {
-        { "abort_on"           , NULL, 0, AV_OPT_TYPE_FLAGS, { .i64 = 0 }, INT64_MIN, INT64_MAX,           .unit = "flags" },
-        { "empty_output"       , NULL, 0, AV_OPT_TYPE_CONST, { .i64 = ABORT_ON_FLAG_EMPTY_OUTPUT        }, .unit = "flags" },
-        { "empty_output_stream", NULL, 0, AV_OPT_TYPE_CONST, { .i64 = ABORT_ON_FLAG_EMPTY_OUTPUT_STREAM }, .unit = "flags" },
+        { "abort_on"        , NULL, 0, AV_OPT_TYPE_FLAGS, { .i64 = 0 }, INT64_MIN, INT64_MAX, .unit = "flags" },
+        { "empty_output"    , NULL, 0, AV_OPT_TYPE_CONST, { .i64 = ABORT_ON_FLAG_EMPTY_OUTPUT     },    .unit = "flags" },
        { NULL },
    };
    static const AVClass class = {
@@ -327,7 +269,7 @@ static int opt_map(void *optctx, const char *opt, const char *arg)
 {
    OptionsContext *o = optctx;
    StreamMap *m = NULL;
-    int i, negative = 0, file_idx, disabled = 0;
+    int i, negative = 0, file_idx;
    int sync_file_idx = -1, sync_stream_idx = 0;
    char *p, *sync;
    char *map;
@@ -362,11 +304,6 @@ static int opt_map(void *optctx, const char *opt, const char *arg)
                                       "match any streams.\n", arg);
            exit_program(1);
        }
-        if (input_streams[input_files[sync_file_idx]->ist_index + sync_stream_idx]->user_set_discard == AVDISCARD_ALL) {
-            av_log(NULL, AV_LOG_FATAL, "Sync stream specification in map %s matches a disabled input "
-                                       "stream.\n", arg);
-            exit_program(1);
-        }
    }


@@ -403,10 +340,6 @@ static int opt_map(void *optctx, const char *opt, const char *arg)
                if (check_stream_specifier(input_files[file_idx]->ctx, input_files[file_idx]->ctx->streams[i],
                            *p == ':' ? p + 1 : p) <= 0)
                    continue;
-                if (input_streams[input_files[file_idx]->ist_index + i]->user_set_discard == AVDISCARD_ALL) {
-                    disabled = 1;
-                    continue;
-                }
                GROW_ARRAY(o->stream_maps, o->nb_stream_maps);
                m = &o->stream_maps[o->nb_stream_maps - 1];

@@ -426,10 +359,6 @@ static int opt_map(void *optctx, const char *opt, const char *arg)
    if (!m) {
        if (allow_unused) {
            av_log(NULL, AV_LOG_VERBOSE, "Stream map '%s' matches no streams; ignoring.\n", arg);
-        } else if (disabled) {
-            av_log(NULL, AV_LOG_FATAL, "Stream map '%s' matches disabled streams.\n"
-                                       "To ignore this, add a trailing '?' to the map.\n", arg);
-            exit_program(1);
        } else {
            av_log(NULL, AV_LOG_FATAL, "Stream map '%s' matches no streams.\n"
                                       "To ignore this, add a trailing '?' to the map.\n", arg);
@@ -509,8 +438,7 @@ static int opt_map_channel(void *optctx, const char *opt, const char *arg)
    /* allow trailing ? to map_channel */
    if (allow_unused = strchr(mapchan, '?'))
        *allow_unused = 0;
-    if (m->channel_idx < 0 || m->channel_idx >= st->codecpar->channels ||
-        input_streams[input_files[m->file_idx]->ist_index + m->stream_idx]->user_set_discard == AVDISCARD_ALL) {
+    if (m->channel_idx < 0 || m->channel_idx >= st->codecpar->channels) {
        if (allow_unused) {
            av_log(NULL, AV_LOG_VERBOSE, "mapchan: invalid audio channel #%d.%d.%d\n",
                    m->file_idx, m->stream_idx, m->channel_idx);
@@ -536,15 +464,21 @@ static int opt_sdp_file(void *optctx, const char *opt, const char *arg)
 #if CONFIG_VAAPI
 static int opt_vaapi_device(void *optctx, const char *opt, const char *arg)
 {
+    HWDevice *dev;
    const char *prefix = "vaapi:";
    char *tmp;
    int err;
    tmp = av_asprintf("%s%s", prefix, arg);
    if (!tmp)
        return AVERROR(ENOMEM);
-    err = hw_device_init_from_string(tmp, NULL);
+    err = hw_device_init_from_string(tmp, &dev);
    av_free(tmp);
-    return err;
+    if (err < 0)
+        return err;
+    hw_device_ctx = av_buffer_ref(dev->device_ref);
+    if (!hw_device_ctx)
+        return AVERROR(ENOMEM);
+    return 0;
 }
 #endif

@@ -813,13 +747,6 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic)

        MATCH_PER_STREAM_OPT(discard, str, discard_str, ic, st);
        ist->user_set_discard = AVDISCARD_NONE;
-
-        if ((o->video_disable && ist->st->codecpar->codec_type == AVMEDIA_TYPE_VIDEO) ||
-            (o->audio_disable && ist->st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) ||
-            (o->subtitle_disable && ist->st->codecpar->codec_type == AVMEDIA_TYPE_SUBTITLE) ||
-            (o->data_disable && ist->st->codecpar->codec_type == AVMEDIA_TYPE_DATA))
-                ist->user_set_discard = AVDISCARD_ALL;
-
        if (discard_str && av_opt_eval_int(&cc, discard_opt, discard_str, &ist->user_set_discard) < 0) {
            av_log(NULL, AV_LOG_ERROR, "Error parsing discard %s.\n",
                    discard_str);
@@ -872,28 +799,9 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic)
            MATCH_PER_STREAM_OPT(top_field_first, i, ist->top_field_first, ic, st);

            MATCH_PER_STREAM_OPT(hwaccels, str, hwaccel, ic, st);
-            MATCH_PER_STREAM_OPT(hwaccel_output_formats, str,
-                                 hwaccel_output_format, ic, st);
-
-            if (!hwaccel_output_format && hwaccel && !strcmp(hwaccel, "cuvid")) {
-                av_log(NULL, AV_LOG_WARNING,
-                    "WARNING: defaulting hwaccel_output_format to cuda for compatibility "
-                    "with old commandlines. This behaviour is DEPRECATED and will be removed "
-                    "in the future. Please explicitly set \"-hwaccel_output_format cuda\".\n");
-                ist->hwaccel_output_format = AV_PIX_FMT_CUDA;
-            } else if (hwaccel_output_format) {
-                ist->hwaccel_output_format = av_get_pix_fmt(hwaccel_output_format);
-                if (ist->hwaccel_output_format == AV_PIX_FMT_NONE) {
-                    av_log(NULL, AV_LOG_FATAL, "Unrecognised hwaccel output "
-                           "format: %s", hwaccel_output_format);
-                }
-            } else {
-                ist->hwaccel_output_format = AV_PIX_FMT_NONE;
-            }
-
            if (hwaccel) {
                // The NVDEC hwaccels use a CUDA device, so remap the name here.
-                if (!strcmp(hwaccel, "nvdec") || !strcmp(hwaccel, "cuvid"))
+                if (!strcmp(hwaccel, "nvdec"))
                    hwaccel = "cuda";

                if (!strcmp(hwaccel, "none"))
@@ -927,6 +835,8 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic)
                               AV_HWDEVICE_TYPE_NONE)
                            av_log(NULL, AV_LOG_FATAL, "%s ",
                                   av_hwdevice_get_type_name(type));
+                        for (i = 0; hwaccels[i].name; i++)
+                            av_log(NULL, AV_LOG_FATAL, "%s ", hwaccels[i].name);
                        av_log(NULL, AV_LOG_FATAL, "\n");
                        exit_program(1);
                    }
@@ -940,6 +850,18 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic)
                    exit_program(1);
            }

+            MATCH_PER_STREAM_OPT(hwaccel_output_formats, str,
+                                 hwaccel_output_format, ic, st);
+            if (hwaccel_output_format) {
+                ist->hwaccel_output_format = av_get_pix_fmt(hwaccel_output_format);
+                if (ist->hwaccel_output_format == AV_PIX_FMT_NONE) {
+                    av_log(NULL, AV_LOG_FATAL, "Unrecognised hwaccel output "
+                           "format: %s", hwaccel_output_format);
+                }
+            } else {
+                ist->hwaccel_output_format = AV_PIX_FMT_NONE;
+            }
+
            ist->hwaccel_pix_fmt = AV_PIX_FMT_NONE;

            break;
@@ -989,7 +911,7 @@ static void assert_file_overwrite(const char *filename)
    if (!file_overwrite) {
        if (proto_name && !strcmp(proto_name, "file") && avio_check(filename, 0) == 0) {
            if (stdin_interaction && !no_file_overwrite) {
-                fprintf(stderr,"File '%s' already exists. Overwrite? [y/N] ", filename);
+                fprintf(stderr,"File '%s' already exists. Overwrite ? [y/N] ", filename);
                fflush(stderr);
                term_exit();
                signal(SIGINT, SIG_DFL);
@@ -1529,12 +1451,54 @@ static OutputStream *new_output_stream(OptionsContext *o, AVFormatContext *oc, e
    MATCH_PER_STREAM_OPT(copy_prior_start, i, ost->copy_prior_start, oc ,st);

    MATCH_PER_STREAM_OPT(bitstream_filters, str, bsfs, oc, st);
-    if (bsfs && *bsfs) {
-        ret = av_bsf_list_parse_str(bsfs, &ost->bsf_ctx);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Error parsing bitstream filter sequence '%s': %s\n", bsfs, av_err2str(ret));
+    while (bsfs && *bsfs) {
+        const AVBitStreamFilter *filter;
+        char *bsf, *bsf_options_str, *bsf_name;
+
+        bsf = av_get_token(&bsfs, ",");
+        if (!bsf)
+            exit_program(1);
+        bsf_name = av_strtok(bsf, "=", &bsf_options_str);
+        if (!bsf_name)
+            exit_program(1);
+
+        filter = av_bsf_get_by_name(bsf_name);
+        if (!filter) {
+            av_log(NULL, AV_LOG_FATAL, "Unknown bitstream filter %s\n", bsf_name);
            exit_program(1);
        }
+
+        ost->bsf_ctx = av_realloc_array(ost->bsf_ctx,
+                                        ost->nb_bitstream_filters + 1,
+                                        sizeof(*ost->bsf_ctx));
+        if (!ost->bsf_ctx)
+            exit_program(1);
+
+        ret = av_bsf_alloc(filter, &ost->bsf_ctx[ost->nb_bitstream_filters]);
+        if (ret < 0) {
+            av_log(NULL, AV_LOG_ERROR, "Error allocating a bitstream filter context\n");
+            exit_program(1);
+        }
+
+        ost->nb_bitstream_filters++;
+
+        if (bsf_options_str && filter->priv_class) {
+            const AVOption *opt = av_opt_next(ost->bsf_ctx[ost->nb_bitstream_filters-1]->priv_data, NULL);
+            const char * shorthand[2] = {NULL};
+
+            if (opt)
+                shorthand[0] = opt->name;
+
+            ret = av_opt_set_from_string(ost->bsf_ctx[ost->nb_bitstream_filters-1]->priv_data, bsf_options_str, shorthand, "=", ":");
+            if (ret < 0) {
+                av_log(NULL, AV_LOG_ERROR, "Error parsing options for bitstream filter %s\n", bsf_name);
+                exit_program(1);
+            }
+        }
+        av_freep(&bsf);
+
+        if (*bsfs)
+            bsfs++;
    }

    MATCH_PER_STREAM_OPT(codec_tags, str, codec_tag, oc, st);
@@ -2210,10 +2174,7 @@ static int open_output_file(OptionsContext *o, const char *filename)
            for (i = 0; i < nb_input_streams; i++) {
                int new_area;
                ist = input_streams[i];
-                new_area = ist->st->codecpar->width * ist->st->codecpar->height + 100000000*!!ist->st->codec_info_nb_frames
-                           + 5000000*!!(ist->st->disposition & AV_DISPOSITION_DEFAULT);
-                if (ist->user_set_discard == AVDISCARD_ALL)
-                    continue;
+                new_area = ist->st->codecpar->width * ist->st->codecpar->height + 100000000*!!ist->st->codec_info_nb_frames;
                if((qcr!=MKTAG('A', 'P', 'I', 'C')) && (ist->st->disposition & AV_DISPOSITION_ATTACHED_PIC))
                    new_area = 1;
                if (ist->st->codecpar->codec_type == AVMEDIA_TYPE_VIDEO &&
@@ -2234,10 +2195,7 @@ static int open_output_file(OptionsContext *o, const char *filename)
            for (i = 0; i < nb_input_streams; i++) {
                int score;
                ist = input_streams[i];
-                score = ist->st->codecpar->channels + 100000000*!!ist->st->codec_info_nb_frames
-                        + 5000000*!!(ist->st->disposition & AV_DISPOSITION_DEFAULT);
-                if (ist->user_set_discard == AVDISCARD_ALL)
-                    continue;
+                score = ist->st->codecpar->channels + 100000000*!!ist->st->codec_info_nb_frames;
                if (ist->st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO &&
                    score > best_score) {
                    best_score = score;
@@ -2259,8 +2217,6 @@ static int open_output_file(OptionsContext *o, const char *filename)
                    AVCodec const *output_codec =
                        avcodec_find_encoder(oc->oformat->subtitle_codec);
                    int input_props = 0, output_props = 0;
-                    if (input_streams[i]->user_set_discard == AVDISCARD_ALL)
-                        continue;
                    if (output_codec)
                        output_descriptor = avcodec_descriptor_get(output_codec->id);
                    if (input_descriptor)
@@ -2282,8 +2238,6 @@ static int open_output_file(OptionsContext *o, const char *filename)
        if (!o->data_disable ) {
            enum AVCodecID codec_id = av_guess_codec(oc->oformat, NULL, filename, NULL, AVMEDIA_TYPE_DATA);
            for (i = 0; codec_id != AV_CODEC_ID_NONE && i < nb_input_streams; i++) {
-                if (input_streams[i]->user_set_discard == AVDISCARD_ALL)
-                    continue;
                if (input_streams[i]->st->codecpar->codec_type == AVMEDIA_TYPE_DATA
                    && input_streams[i]->st->codecpar->codec_id == codec_id )
                    new_data_stream(o, oc, i);
@@ -2322,11 +2276,6 @@ loop_end:
                int src_idx = input_files[map->file_index]->ist_index + map->stream_index;

                ist = input_streams[input_files[map->file_index]->ist_index + map->stream_index];
-                if (ist->user_set_discard == AVDISCARD_ALL) {
-                    av_log(NULL, AV_LOG_FATAL, "Stream #%d:%d is disabled and cannot be mapped.\n",
-                           map->file_index, map->stream_index);
-                    exit_program(1);
-                }
                if(o->subtitle_disable && ist->st->codecpar->codec_type == AVMEDIA_TYPE_SUBTITLE)
                    continue;
                if(o->   audio_disable && ist->st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO)
@@ -3020,11 +2969,8 @@ static int opt_preset(void *optctx, const char *opt, const char *arg)
 static int opt_old2new(void *optctx, const char *opt, const char *arg)
 {
    OptionsContext *o = optctx;
-    int ret;
    char *s = av_asprintf("%s:%c", opt + 1, *opt);
-    if (!s)
-        return AVERROR(ENOMEM);
-    ret = parse_option(o, s, arg, options);
+    int ret = parse_option(o, s, arg, options);
    av_free(s);
    return ret;
 }
@@ -3055,8 +3001,6 @@ static int opt_qscale(void *optctx, const char *opt, const char *arg)
        return parse_option(o, "q:v", arg, options);
    }
    s = av_asprintf("q%s", opt + 6);
-    if (!s)
-        return AVERROR(ENOMEM);
    ret = parse_option(o, s, arg, options);
    av_free(s);
    return ret;
@@ -3101,11 +3045,8 @@ static int opt_vsync(void *optctx, const char *opt, const char *arg)
 static int opt_timecode(void *optctx, const char *opt, const char *arg)
 {
    OptionsContext *o = optctx;
-    int ret;
    char *tcr = av_asprintf("timecode=%s", arg);
-    if (!tcr)
-        return AVERROR(ENOMEM);
-    ret = parse_option(o, "metadata:g", tcr, options);
+    int ret = parse_option(o, "metadata:g", tcr, options);
    if (ret >= 0)
        ret = av_dict_set(&o->g->codec_opts, "gop_timecode", arg, 0);
    av_free(tcr);
@@ -3207,7 +3148,7 @@ void show_help_default(const char *opt, const char *arg)
           "    -h      -- print basic options\n"
           "    -h long -- print more options\n"
           "    -h full -- print all options (including all format and codec specific options, very long)\n"
-           "    -h type=name -- print all options for the named decoder/encoder/demuxer/muxer/filter/bsf/protocol\n"
+           "    -h type=name -- print all options for the named decoder/encoder/demuxer/muxer/filter/bsf\n"
           "    See man %s for detailed description of the options.\n"
           "\n", program_name);

@@ -3215,7 +3156,7 @@ void show_help_default(const char *opt, const char *arg)
                      OPT_EXIT, 0, 0);

    show_help_options(options, "Global options (affect whole program "
-                      "instead of just one file):",
+                      "instead of just one file:",
                      0, per_file | OPT_EXIT | OPT_EXPERT, 0);
    if (show_advanced)
        show_help_options(options, "Advanced global options:", OPT_EXPERT,
@@ -3465,7 +3406,7 @@ const OptionDef options[] = {
    { "stdin",          OPT_BOOL | OPT_EXPERT,                       { &stdin_interaction },
      "enable or disable interaction on standard input" },
    { "timelimit",      HAS_ARG | OPT_EXPERT,                        { .func_arg = opt_timelimit },
-        "set max runtime in seconds in CPU user time", "limit" },
+        "set max runtime in seconds", "limit" },
    { "dump",           OPT_BOOL | OPT_EXPERT,                       { &do_pkt_dump },
        "dump each input packet" },
    { "hex",            OPT_BOOL | OPT_EXPERT,                       { &do_hex_dump },
--- a/fftools/ffmpeg_qsv.c
+++ b/fftools/ffmpeg_qsv.c
@@ -28,7 +28,6 @@

 #include "ffmpeg.h"

-static AVBufferRef *hw_device_ctx;
 char *qsv_device = NULL;

 static int qsv_get_buffer(AVCodecContext *s, AVFrame *frame, int flags)
--- a/fftools/ffmpeg_videotoolbox.c
+++ b/fftools/ffmpeg_videotoolbox.c
@@ -51,12 +51,7 @@ static int videotoolbox_retrieve_data(AVCodecContext *s, AVFrame *frame)
    case kCVPixelFormatType_422YpCbCr8:       vt->tmp_frame->format = AV_PIX_FMT_UYVY422; break;
    case kCVPixelFormatType_32BGRA:           vt->tmp_frame->format = AV_PIX_FMT_BGRA; break;
 #ifdef kCFCoreFoundationVersionNumber10_7
-    case kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange:
-    case kCVPixelFormatType_420YpCbCr8BiPlanarFullRange: vt->tmp_frame->format = AV_PIX_FMT_NV12; break;
-#endif
-#if HAVE_KCVPIXELFORMATTYPE_420YPCBCR10BIPLANARVIDEORANGE
-    case kCVPixelFormatType_420YpCbCr10BiPlanarVideoRange:
-    case kCVPixelFormatType_420YpCbCr10BiPlanarFullRange: vt->tmp_frame->format = AV_PIX_FMT_P010; break;
+    case kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange: vt->tmp_frame->format = AV_PIX_FMT_NV12; break;
 #endif
    default:
        av_log(NULL, AV_LOG_ERROR,
@@ -67,7 +62,7 @@ static int videotoolbox_retrieve_data(AVCodecContext *s, AVFrame *frame)

    vt->tmp_frame->width  = frame->width;
    vt->tmp_frame->height = frame->height;
-    ret = av_frame_get_buffer(vt->tmp_frame, 0);
+    ret = av_frame_get_buffer(vt->tmp_frame, 32);
    if (ret < 0)
        return ret;

--- a/fftools/ffplay.c
+++ b/fftools/ffplay.c
@@ -40,7 +40,6 @@
 #include "libavutil/samplefmt.h"
 #include "libavutil/avassert.h"
 #include "libavutil/time.h"
-#include "libavutil/bprint.h"
 #include "libavformat/avformat.h"
 #include "libavdevice/avdevice.h"
 #include "libswscale/swscale.h"
@@ -325,9 +324,8 @@ static int seek_by_bytes = -1;
 static float seek_interval = 10;
 static int display_disable;
 static int borderless;
-static int alwaysontop;
 static int startup_volume = 100;
-static int show_status = -1;
+static int show_status = 1;
 static int av_sync_type = AV_SYNC_AUDIO_MASTER;
 static int64_t start_time = AV_NOPTS_VALUE;
 static int64_t duration = AV_NOPTS_VALUE;
@@ -355,7 +353,6 @@ static char *afilters = NULL;
 #endif
 static int autorotate = 1;
 static int find_stream_info = 1;
-static int filter_nbthreads = 0;

 /* current context */
 static int is_full_screen;
@@ -645,10 +642,7 @@ static int decoder_decode_frame(Decoder *d, AVFrame *frame, AVSubtitle *sub) {
                if (packet_queue_get(d->queue, &pkt, 1, &d->pkt_serial) < 0)
                    return -1;
            }
-            if (d->queue->serial == d->pkt_serial)
-                break;
-            av_packet_unref(&pkt);
-        } while (1);
+        } while (d->queue->serial != d->pkt_serial);

        if (pkt.data == flush_pkt.data) {
            avcodec_flush_buffers(d->avctx);
@@ -867,27 +861,31 @@ static void calculate_display_rect(SDL_Rect *rect,
                                   int scr_xleft, int scr_ytop, int scr_width, int scr_height,
                                   int pic_width, int pic_height, AVRational pic_sar)
 {
-    AVRational aspect_ratio = pic_sar;
-    int64_t width, height, x, y;
+    float aspect_ratio;
+    int width, height, x, y;

-    if (av_cmp_q(aspect_ratio, av_make_q(0, 1)) <= 0)
-        aspect_ratio = av_make_q(1, 1);
+    if (pic_sar.num == 0)
+        aspect_ratio = 0;
+    else
+        aspect_ratio = av_q2d(pic_sar);

-    aspect_ratio = av_mul_q(aspect_ratio, av_make_q(pic_width, pic_height));
+    if (aspect_ratio <= 0.0)
+        aspect_ratio = 1.0;
+    aspect_ratio *= (float)pic_width / (float)pic_height;

    /* XXX: we suppose the screen has a 1.0 pixel ratio */
    height = scr_height;
-    width = av_rescale(height, aspect_ratio.num, aspect_ratio.den) & ~1;
+    width = lrint(height * aspect_ratio) & ~1;
    if (width > scr_width) {
        width = scr_width;
-        height = av_rescale(width, aspect_ratio.den, aspect_ratio.num) & ~1;
+        height = lrint(width / aspect_ratio) & ~1;
    }
    x = (scr_width - width) / 2;
    y = (scr_height - height) / 2;
    rect->x = scr_xleft + x;
    rect->y = scr_ytop  + y;
-    rect->w = FFMAX((int)width,  1);
-    rect->h = FFMAX((int)height, 1);
+    rect->w = FFMAX(width,  1);
+    rect->h = FFMAX(height, 1);
 }

 static void get_sdl_pix_fmt_and_blendmode(int format, Uint32 *sdl_pix_fmt, SDL_BlendMode *sdl_blendmode)
@@ -1328,11 +1326,7 @@ static void sigterm_handler(int sig)
 static void set_default_window_size(int width, int height, AVRational sar)
 {
    SDL_Rect rect;
-    int max_width  = screen_width  ? screen_width  : INT_MAX;
-    int max_height = screen_height ? screen_height : INT_MAX;
-    if (max_width == INT_MAX && max_height == INT_MAX)
-        max_height = height;
-    calculate_display_rect(&rect, 0, 0, max_width, max_height, width, height, sar);
+    calculate_display_rect(&rect, 0, 0, INT_MAX, height, width, height, sar);
    default_width  = rect.w;
    default_height = rect.h;
 }
@@ -1341,8 +1335,13 @@ static int video_open(VideoState *is)
 {
    int w,h;

-    w = screen_width ? screen_width : default_width;
-    h = screen_height ? screen_height : default_height;
+    if (screen_width) {
+        w = screen_width;
+        h = screen_height;
+    } else {
+        w = default_width;
+        h = default_height;
+    }

    if (!window_title)
        window_title = input_filename;
@@ -1693,7 +1692,6 @@ display:
    }
    is->force_refresh = 0;
    if (show_status) {
-        AVBPrint buf;
        static int64_t last_time;
        int64_t cur_time;
        int aqsize, vqsize, sqsize;
@@ -1717,28 +1715,18 @@ display:
                av_diff = get_master_clock(is) - get_clock(&is->vidclk);
            else if (is->audio_st)
                av_diff = get_master_clock(is) - get_clock(&is->audclk);
-
-            av_bprint_init(&buf, 0, AV_BPRINT_SIZE_AUTOMATIC);
-            av_bprintf(&buf,
-                      "%7.2f %s:%7.3f fd=%4d aq=%5dKB vq=%5dKB sq=%5dB f=%"PRId64"/%"PRId64"   \r",
-                      get_master_clock(is),
-                      (is->audio_st && is->video_st) ? "A-V" : (is->video_st ? "M-V" : (is->audio_st ? "M-A" : "   ")),
-                      av_diff,
-                      is->frame_drops_early + is->frame_drops_late,
-                      aqsize / 1024,
-                      vqsize / 1024,
-                      sqsize,
-                      is->video_st ? is->viddec.avctx->pts_correction_num_faulty_dts : 0,
-                      is->video_st ? is->viddec.avctx->pts_correction_num_faulty_pts : 0);
-
-            if (show_status == 1 && AV_LOG_INFO > av_log_get_level())
-                fprintf(stderr, "%s", buf.str);
-            else
-                av_log(NULL, AV_LOG_INFO, "%s", buf.str);
-
-            fflush(stderr);
-            av_bprint_finalize(&buf, NULL);
-
+            av_log(NULL, AV_LOG_INFO,
+                   "%7.2f %s:%7.3f fd=%4d aq=%5dKB vq=%5dKB sq=%5dB f=%"PRId64"/%"PRId64"   \r",
+                   get_master_clock(is),
+                   (is->audio_st && is->video_st) ? "A-V" : (is->video_st ? "M-V" : (is->audio_st ? "M-A" : "   ")),
+                   av_diff,
+                   is->frame_drops_early + is->frame_drops_late,
+                   aqsize / 1024,
+                   vqsize / 1024,
+                   sqsize,
+                   is->video_st ? is->viddec.avctx->pts_correction_num_faulty_dts : 0,
+                   is->video_st ? is->viddec.avctx->pts_correction_num_faulty_pts : 0);
+            fflush(stdout);
            last_time = cur_time;
        }
    }
@@ -1971,7 +1959,6 @@ static int configure_audio_filters(VideoState *is, const char *afilters, int for
    avfilter_graph_free(&is->agraph);
    if (!(is->agraph = avfilter_graph_alloc()))
        return AVERROR(ENOMEM);
-    is->agraph->nb_threads = filter_nbthreads;

    while ((e = av_dict_get(swr_opts, "", e, AV_DICT_IGNORE_SUFFIX)))
        av_strlcatf(aresample_swr_opts, sizeof(aresample_swr_opts), "%s=%s:", e->key, e->value);
@@ -2121,10 +2108,10 @@ static int audio_thread(void *arg)
    return ret;
 }

-static int decoder_start(Decoder *d, int (*fn)(void *), const char *thread_name, void* arg)
+static int decoder_start(Decoder *d, int (*fn)(void *), void *arg)
 {
    packet_queue_start(d->queue);
-    d->decoder_tid = SDL_CreateThread(fn, thread_name, arg);
+    d->decoder_tid = SDL_CreateThread(fn, "decoder", arg);
    if (!d->decoder_tid) {
        av_log(NULL, AV_LOG_ERROR, "SDL_CreateThread(): %s\n", SDL_GetError());
        return AVERROR(ENOMEM);
@@ -2143,17 +2130,26 @@ static int video_thread(void *arg)
    AVRational frame_rate = av_guess_frame_rate(is->ic, is->video_st, NULL);

 #if CONFIG_AVFILTER
-    AVFilterGraph *graph = NULL;
+    AVFilterGraph *graph = avfilter_graph_alloc();
    AVFilterContext *filt_out = NULL, *filt_in = NULL;
    int last_w = 0;
    int last_h = 0;
    enum AVPixelFormat last_format = -2;
    int last_serial = -1;
    int last_vfilter_idx = 0;
+    if (!graph) {
+        av_frame_free(&frame);
+        return AVERROR(ENOMEM);
+    }
+
 #endif

-    if (!frame)
+    if (!frame) {
+#if CONFIG_AVFILTER
+        avfilter_graph_free(&graph);
+#endif
        return AVERROR(ENOMEM);
+    }

    for (;;) {
        ret = get_video_frame(is, frame);
@@ -2176,11 +2172,6 @@ static int video_thread(void *arg)
                   (const char *)av_x_if_null(av_get_pix_fmt_name(frame->format), "none"), is->viddec.pkt_serial);
            avfilter_graph_free(&graph);
            graph = avfilter_graph_alloc();
-            if (!graph) {
-                ret = AVERROR(ENOMEM);
-                goto the_end;
-            }
-            graph->nb_threads = filter_nbthreads;
            if ((ret = configure_video_filters(graph, is, vfilters_list ? vfilters_list[is->vfilter_idx] : NULL, frame)) < 0) {
                SDL_Event event;
                event.type = FF_QUIT_EVENT;
@@ -2690,7 +2681,7 @@ static int stream_component_open(VideoState *is, int stream_index)
            is->auddec.start_pts = is->audio_st->start_time;
            is->auddec.start_pts_tb = is->audio_st->time_base;
        }
-        if ((ret = decoder_start(&is->auddec, audio_thread, "audio_decoder", is)) < 0)
+        if ((ret = decoder_start(&is->auddec, audio_thread, is)) < 0)
            goto out;
        SDL_PauseAudioDevice(audio_dev, 0);
        break;
@@ -2699,7 +2690,7 @@ static int stream_component_open(VideoState *is, int stream_index)
        is->video_st = ic->streams[stream_index];

        decoder_init(&is->viddec, avctx, &is->videoq, is->continue_read_thread);
-        if ((ret = decoder_start(&is->viddec, video_thread, "video_decoder", is)) < 0)
+        if ((ret = decoder_start(&is->viddec, video_thread, is)) < 0)
            goto out;
        is->queue_attachments_req = 1;
        break;
@@ -2708,7 +2699,7 @@ static int stream_component_open(VideoState *is, int stream_index)
        is->subtitle_st = ic->streams[stream_index];

        decoder_init(&is->subdec, avctx, &is->subtitleq, is->continue_read_thread);
-        if ((ret = decoder_start(&is->subdec, subtitle_thread, "subtitle_decoder", is)) < 0)
+        if ((ret = decoder_start(&is->subdec, subtitle_thread, is)) < 0)
            goto out;
        break;
    default:
@@ -2775,6 +2766,9 @@ static int read_thread(void *arg)
    }

    memset(st_index, -1, sizeof(st_index));
+    is->last_video_stream = is->video_stream = -1;
+    is->last_audio_stream = is->audio_stream = -1;
+    is->last_subtitle_stream = is->subtitle_stream = -1;
    is->eof = 0;

    ic = avformat_alloc_context();
@@ -2986,7 +2980,7 @@ static int read_thread(void *arg)
        }
        if (is->queue_attachments_req) {
            if (is->video_st && is->video_st->disposition & AV_DISPOSITION_ATTACHED_PIC) {
-                AVPacket copy;
+                AVPacket copy = { 0 };
                if ((ret = av_packet_ref(&copy, &is->video_st->attached_pic)) < 0)
                    goto fail;
                packet_queue_put(&is->videoq, &copy);
@@ -3080,9 +3074,6 @@ static VideoState *stream_open(const char *filename, AVInputFormat *iformat)
    is = av_mallocz(sizeof(VideoState));
    if (!is)
        return NULL;
-    is->last_video_stream = is->video_stream = -1;
-    is->last_audio_stream = is->audio_stream = -1;
-    is->last_subtitle_stream = is->subtitle_stream = -1;
    is->filename = av_strdup(filename);
    if (!is->filename)
        goto fail;
@@ -3451,7 +3442,7 @@ static void event_loop(VideoState *cur_stream)
            break;
        case SDL_WINDOWEVENT:
            switch (event.window.event) {
-                case SDL_WINDOWEVENT_SIZE_CHANGED:
+                case SDL_WINDOWEVENT_RESIZED:
                    screen_width  = cur_stream->width  = event.window.data1;
                    screen_height = cur_stream->height = event.window.data2;
                    if (cur_stream->vis_texture) {
@@ -3597,7 +3588,6 @@ static const OptionDef options[] = {
    { "seek_interval", OPT_FLOAT | HAS_ARG, { &seek_interval }, "set seek interval for left/right keys, in seconds", "seconds" },
    { "nodisp", OPT_BOOL, { &display_disable }, "disable graphical display" },
    { "noborder", OPT_BOOL, { &borderless }, "borderless window" },
-    { "alwaysontop", OPT_BOOL, { &alwaysontop }, "window always on top" },
    { "volume", OPT_INT | HAS_ARG, { &startup_volume}, "set startup volume 0=min 100=max", "volume" },
    { "f", HAS_ARG, { .func_arg = opt_format }, "force format", "fmt" },
    { "pix_fmt", HAS_ARG | OPT_EXPERT | OPT_VIDEO, { .func_arg = opt_frame_pix_fmt }, "set pixel format", "format" },
@@ -3631,7 +3621,6 @@ static const OptionDef options[] = {
    { "autorotate", OPT_BOOL, { &autorotate }, "automatically rotate video", "" },
    { "find_stream_info", OPT_BOOL | OPT_INPUT | OPT_EXPERT, { &find_stream_info },
        "read and decode the streams to fill missing information with heuristics" },
-    { "filter_threads", HAS_ARG | OPT_INT | OPT_EXPERT, { &filter_nbthreads }, "number of filter threads per graph" },
    { NULL, },
 };

@@ -3739,12 +3728,6 @@ int main(int argc, char **argv)

    if (!display_disable) {
        int flags = SDL_WINDOW_HIDDEN;
-        if (alwaysontop)
-#if SDL_VERSION_ATLEAST(2,0,5)
-            flags |= SDL_WINDOW_ALWAYS_ON_TOP;
-#else
-            av_log(NULL, AV_LOG_WARNING, "Your SDL version doesn't support SDL_WINDOW_ALWAYS_ON_TOP. Feature will be inactive.\n");
-#endif
        if (borderless)
            flags |= SDL_WINDOW_BORDERLESS;
        else
--- a/fftools/ffprobe.c
+++ b/fftools/ffprobe.c
@@ -36,7 +36,6 @@
 #include "libavutil/display.h"
 #include "libavutil/hash.h"
 #include "libavutil/mastering_display_metadata.h"
-#include "libavutil/dovi_meta.h"
 #include "libavutil/opt.h"
 #include "libavutil/pixdesc.h"
 #include "libavutil/spherical.h"
@@ -166,8 +165,6 @@ typedef enum {
    SECTION_ID_FRAME_TAGS,
    SECTION_ID_FRAME_SIDE_DATA_LIST,
    SECTION_ID_FRAME_SIDE_DATA,
-    SECTION_ID_FRAME_SIDE_DATA_TIMECODE_LIST,
-    SECTION_ID_FRAME_SIDE_DATA_TIMECODE,
    SECTION_ID_FRAME_LOG,
    SECTION_ID_FRAME_LOGS,
    SECTION_ID_LIBRARY_VERSION,
@@ -212,9 +209,7 @@ static struct section sections[] = {
    [SECTION_ID_FRAME] =              { SECTION_ID_FRAME, "frame", 0, { SECTION_ID_FRAME_TAGS, SECTION_ID_FRAME_SIDE_DATA_LIST, SECTION_ID_FRAME_LOGS, -1 } },
    [SECTION_ID_FRAME_TAGS] =         { SECTION_ID_FRAME_TAGS, "tags", SECTION_FLAG_HAS_VARIABLE_FIELDS, { -1 }, .element_name = "tag", .unique_name = "frame_tags" },
    [SECTION_ID_FRAME_SIDE_DATA_LIST] ={ SECTION_ID_FRAME_SIDE_DATA_LIST, "side_data_list", SECTION_FLAG_IS_ARRAY, { SECTION_ID_FRAME_SIDE_DATA, -1 }, .element_name = "side_data", .unique_name = "frame_side_data_list" },
-    [SECTION_ID_FRAME_SIDE_DATA] =     { SECTION_ID_FRAME_SIDE_DATA, "side_data", 0, { SECTION_ID_FRAME_SIDE_DATA_TIMECODE_LIST, -1 } },
-    [SECTION_ID_FRAME_SIDE_DATA_TIMECODE_LIST] =     { SECTION_ID_FRAME_SIDE_DATA_TIMECODE_LIST, "timecodes", SECTION_FLAG_IS_ARRAY, { SECTION_ID_FRAME_SIDE_DATA_TIMECODE, -1 } },
-    [SECTION_ID_FRAME_SIDE_DATA_TIMECODE] =     { SECTION_ID_FRAME_SIDE_DATA_TIMECODE, "timecode", 0, { -1 } },
+    [SECTION_ID_FRAME_SIDE_DATA] =     { SECTION_ID_FRAME_SIDE_DATA, "side_data", 0, { -1 } },
    [SECTION_ID_FRAME_LOGS] =         { SECTION_ID_FRAME_LOGS, "logs", SECTION_FLAG_IS_ARRAY, { SECTION_ID_FRAME_LOG, -1 } },
    [SECTION_ID_FRAME_LOG] =          { SECTION_ID_FRAME_LOG, "log", 0, { -1 },  },
    [SECTION_ID_LIBRARY_VERSIONS] =   { SECTION_ID_LIBRARY_VERSIONS, "library_versions", SECTION_FLAG_IS_ARRAY, { SECTION_ID_LIBRARY_VERSION, -1 } },
@@ -255,7 +250,6 @@ static const OptionDef *options;

 /* FFprobe context */
 static const char *input_filename;
-static const char *print_input_filename;
 static AVInputFormat *iformat = NULL;

 static struct AVHashContext *hash;
@@ -1085,12 +1079,12 @@ typedef struct CompactContext {
 #define OFFSET(x) offsetof(CompactContext, x)

 static const AVOption compact_options[]= {
-    {"item_sep", "set item separator",    OFFSET(item_sep_str),    AV_OPT_TYPE_STRING, {.str="|"},  0, 0 },
-    {"s",        "set item separator",    OFFSET(item_sep_str),    AV_OPT_TYPE_STRING, {.str="|"},  0, 0 },
+    {"item_sep", "set item separator",    OFFSET(item_sep_str),    AV_OPT_TYPE_STRING, {.str="|"},  CHAR_MIN, CHAR_MAX },
+    {"s",        "set item separator",    OFFSET(item_sep_str),    AV_OPT_TYPE_STRING, {.str="|"},  CHAR_MIN, CHAR_MAX },
    {"nokey",    "force no key printing", OFFSET(nokey),           AV_OPT_TYPE_BOOL,   {.i64=0},    0,        1        },
    {"nk",       "force no key printing", OFFSET(nokey),           AV_OPT_TYPE_BOOL,   {.i64=0},    0,        1        },
-    {"escape",   "set escape mode",       OFFSET(escape_mode_str), AV_OPT_TYPE_STRING, {.str="c"},  0, 0 },
-    {"e",        "set escape mode",       OFFSET(escape_mode_str), AV_OPT_TYPE_STRING, {.str="c"},  0, 0 },
+    {"escape",   "set escape mode",       OFFSET(escape_mode_str), AV_OPT_TYPE_STRING, {.str="c"},  CHAR_MIN, CHAR_MAX },
+    {"e",        "set escape mode",       OFFSET(escape_mode_str), AV_OPT_TYPE_STRING, {.str="c"},  CHAR_MIN, CHAR_MAX },
    {"print_section", "print section name", OFFSET(print_section), AV_OPT_TYPE_BOOL,   {.i64=1},    0,        1        },
    {"p",             "print section name", OFFSET(print_section), AV_OPT_TYPE_BOOL,   {.i64=1},    0,        1        },
    {NULL},
@@ -1201,12 +1195,12 @@ static const Writer compact_writer = {
 #define OFFSET(x) offsetof(CompactContext, x)

 static const AVOption csv_options[] = {
-    {"item_sep", "set item separator",    OFFSET(item_sep_str),    AV_OPT_TYPE_STRING, {.str=","},  0, 0 },
-    {"s",        "set item separator",    OFFSET(item_sep_str),    AV_OPT_TYPE_STRING, {.str=","},  0, 0 },
+    {"item_sep", "set item separator",    OFFSET(item_sep_str),    AV_OPT_TYPE_STRING, {.str=","},  CHAR_MIN, CHAR_MAX },
+    {"s",        "set item separator",    OFFSET(item_sep_str),    AV_OPT_TYPE_STRING, {.str=","},  CHAR_MIN, CHAR_MAX },
    {"nokey",    "force no key printing", OFFSET(nokey),           AV_OPT_TYPE_BOOL,   {.i64=1},    0,        1        },
    {"nk",       "force no key printing", OFFSET(nokey),           AV_OPT_TYPE_BOOL,   {.i64=1},    0,        1        },
-    {"escape",   "set escape mode",       OFFSET(escape_mode_str), AV_OPT_TYPE_STRING, {.str="csv"}, 0, 0 },
-    {"e",        "set escape mode",       OFFSET(escape_mode_str), AV_OPT_TYPE_STRING, {.str="csv"}, 0, 0 },
+    {"escape",   "set escape mode",       OFFSET(escape_mode_str), AV_OPT_TYPE_STRING, {.str="csv"}, CHAR_MIN, CHAR_MAX },
+    {"e",        "set escape mode",       OFFSET(escape_mode_str), AV_OPT_TYPE_STRING, {.str="csv"}, CHAR_MIN, CHAR_MAX },
    {"print_section", "print section name", OFFSET(print_section), AV_OPT_TYPE_BOOL,   {.i64=1},    0,        1        },
    {"p",             "print section name", OFFSET(print_section), AV_OPT_TYPE_BOOL,   {.i64=1},    0,        1        },
    {NULL},
@@ -1239,8 +1233,8 @@ typedef struct FlatContext {
 #define OFFSET(x) offsetof(FlatContext, x)

 static const AVOption flat_options[]= {
-    {"sep_char", "set separator",    OFFSET(sep_str),    AV_OPT_TYPE_STRING, {.str="."},  0, 0 },
-    {"s",        "set separator",    OFFSET(sep_str),    AV_OPT_TYPE_STRING, {.str="."},  0, 0 },
+    {"sep_char", "set separator",    OFFSET(sep_str),    AV_OPT_TYPE_STRING, {.str="."},  CHAR_MIN, CHAR_MAX },
+    {"s",        "set separator",    OFFSET(sep_str),    AV_OPT_TYPE_STRING, {.str="."},  CHAR_MIN, CHAR_MAX },
    {"hierarchical", "specify if the section specification should be hierarchical", OFFSET(hierarchical), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1 },
    {"h",            "specify if the section specification should be hierarchical", OFFSET(hierarchical), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1 },
    {NULL},
@@ -1537,7 +1531,7 @@ static void json_print_section_header(WriterContext *wctx)
            if (parent_section && parent_section->id == SECTION_ID_PACKETS_AND_FRAMES) {
                if (!json->compact)
                    JSON_INDENT();
-                printf("\"type\": \"%s\"", section->name);
+                printf("\"type\": \"%s\"%s", section->name, json->item_sep);
            }
        }
        av_bprint_finalize(&buf, NULL);
@@ -1581,10 +1575,8 @@ static inline void json_print_item_str(WriterContext *wctx,
 static void json_print_str(WriterContext *wctx, const char *key, const char *value)
 {
    JSONContext *json = wctx->priv;
-    const struct section *parent_section = wctx->level ?
-        wctx->section[wctx->level-1] : NULL;

-    if (wctx->nb_item[wctx->level] || (parent_section && parent_section->id == SECTION_ID_PACKETS_AND_FRAMES))
+    if (wctx->nb_item[wctx->level])
        printf("%s", json->item_sep);
    if (!json->compact)
        JSON_INDENT();
@@ -1594,11 +1586,9 @@ static void json_print_str(WriterContext *wctx, const char *key, const char *val
 static void json_print_int(WriterContext *wctx, const char *key, long long int value)
 {
    JSONContext *json = wctx->priv;
-    const struct section *parent_section = wctx->level ?
-        wctx->section[wctx->level-1] : NULL;
    AVBPrint buf;

-    if (wctx->nb_item[wctx->level] || (parent_section && parent_section->id == SECTION_ID_PACKETS_AND_FRAMES))
+    if (wctx->nb_item[wctx->level])
        printf("%s", json->item_sep);
    if (!json->compact)
        JSON_INDENT();
@@ -1929,16 +1919,6 @@ static void print_pkt_side_data(WriterContext *w,
            AVContentLightMetadata *metadata = (AVContentLightMetadata *)sd->data;
            print_int("max_content", metadata->MaxCLL);
            print_int("max_average", metadata->MaxFALL);
-        } else if (sd->type == AV_PKT_DATA_DOVI_CONF) {
-            AVDOVIDecoderConfigurationRecord *dovi = (AVDOVIDecoderConfigurationRecord *)sd->data;
-            print_int("dv_version_major", dovi->dv_version_major);
-            print_int("dv_version_minor", dovi->dv_version_minor);
-            print_int("dv_profile", dovi->dv_profile);
-            print_int("dv_level", dovi->dv_level);
-            print_int("rpu_present_flag", dovi->rpu_present_flag);
-            print_int("el_present_flag", dovi->el_present_flag);
-            print_int("bl_present_flag", dovi->bl_present_flag);
-            print_int("dv_bl_signal_compatibility_id", dovi->dv_bl_signal_compatibility_id);
        }
        writer_print_section_footer(w);
    }
@@ -2219,18 +2199,6 @@ static void show_frame(WriterContext *w, AVFrame *frame, AVStream *stream,
                char tcbuf[AV_TIMECODE_STR_SIZE];
                av_timecode_make_mpeg_tc_string(tcbuf, *(int64_t *)(sd->data));
                print_str("timecode", tcbuf);
-            } else if (sd->type == AV_FRAME_DATA_S12M_TIMECODE && sd->size == 16) {
-                uint32_t *tc = (uint32_t*)sd->data;
-                int m = FFMIN(tc[0],3);
-                writer_print_section_header(w, SECTION_ID_FRAME_SIDE_DATA_TIMECODE_LIST);
-                for (int j = 1; j <= m ; j++) {
-                    char tcbuf[AV_TIMECODE_STR_SIZE];
-                    av_timecode_make_smpte_tc_string(tcbuf, tc[j], 0);
-                    writer_print_section_header(w, SECTION_ID_FRAME_SIDE_DATA_TIMECODE);
-                    print_str("value", tcbuf);
-                    writer_print_section_footer(w);
-                }
-                writer_print_section_footer(w);
            } else if (sd->type == AV_FRAME_DATA_MASTERING_DISPLAY_METADATA) {
                AVMasteringDisplayMetadata *metadata = (AVMasteringDisplayMetadata *)sd->data;

@@ -2445,7 +2413,9 @@ static int read_interval_packets(WriterContext *w, InputFile *ifile,
        }
        av_packet_unref(&pkt);
    }
-    av_packet_unref(&pkt);
+    av_init_packet(&pkt);
+    pkt.data = NULL;
+    pkt.size = 0;
    //Flush remaining frames that are cached in the decoder
    for (i = 0; i < fmt_ctx->nb_streams; i++) {
        pkt.stream_index = i;
@@ -2547,7 +2517,6 @@ static int show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_id
        if (dec_ctx) {
            print_int("coded_width",  dec_ctx->coded_width);
            print_int("coded_height", dec_ctx->coded_height);
-            print_int("closed_captions", !!(dec_ctx->properties & FF_CODEC_PROPERTY_CLOSED_CAPTIONS));
        }
 #endif
        print_int("has_b_frames", par->video_delay);
@@ -2677,20 +2646,20 @@ static int show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_id
    } while (0)

    if (do_show_stream_disposition) {
-        writer_print_section_header(w, in_program ? SECTION_ID_PROGRAM_STREAM_DISPOSITION : SECTION_ID_STREAM_DISPOSITION);
-        PRINT_DISPOSITION(DEFAULT,          "default");
-        PRINT_DISPOSITION(DUB,              "dub");
-        PRINT_DISPOSITION(ORIGINAL,         "original");
-        PRINT_DISPOSITION(COMMENT,          "comment");
-        PRINT_DISPOSITION(LYRICS,           "lyrics");
-        PRINT_DISPOSITION(KARAOKE,          "karaoke");
-        PRINT_DISPOSITION(FORCED,           "forced");
-        PRINT_DISPOSITION(HEARING_IMPAIRED, "hearing_impaired");
-        PRINT_DISPOSITION(VISUAL_IMPAIRED,  "visual_impaired");
-        PRINT_DISPOSITION(CLEAN_EFFECTS,    "clean_effects");
-        PRINT_DISPOSITION(ATTACHED_PIC,     "attached_pic");
-        PRINT_DISPOSITION(TIMED_THUMBNAILS, "timed_thumbnails");
-        writer_print_section_footer(w);
+    writer_print_section_header(w, in_program ? SECTION_ID_PROGRAM_STREAM_DISPOSITION : SECTION_ID_STREAM_DISPOSITION);
+    PRINT_DISPOSITION(DEFAULT,          "default");
+    PRINT_DISPOSITION(DUB,              "dub");
+    PRINT_DISPOSITION(ORIGINAL,         "original");
+    PRINT_DISPOSITION(COMMENT,          "comment");
+    PRINT_DISPOSITION(LYRICS,           "lyrics");
+    PRINT_DISPOSITION(KARAOKE,          "karaoke");
+    PRINT_DISPOSITION(FORCED,           "forced");
+    PRINT_DISPOSITION(HEARING_IMPAIRED, "hearing_impaired");
+    PRINT_DISPOSITION(VISUAL_IMPAIRED,  "visual_impaired");
+    PRINT_DISPOSITION(CLEAN_EFFECTS,    "clean_effects");
+    PRINT_DISPOSITION(ATTACHED_PIC,     "attached_pic");
+    PRINT_DISPOSITION(TIMED_THUMBNAILS, "timed_thumbnails");
+    writer_print_section_footer(w);
    }

    if (do_show_stream_tags)
@@ -2849,8 +2818,7 @@ static void show_error(WriterContext *w, int err)
    writer_print_section_footer(w);
 }

-static int open_input_file(InputFile *ifile, const char *filename,
-                           const char *print_filename)
+static int open_input_file(InputFile *ifile, const char *filename)
 {
    int err, i;
    AVFormatContext *fmt_ctx = NULL;
@@ -2872,10 +2840,6 @@ static int open_input_file(InputFile *ifile, const char *filename,
        print_error(filename, err);
        return err;
    }
-    if (print_filename) {
-        av_freep(&fmt_ctx->url);
-        fmt_ctx->url = av_strdup(print_filename);
-    }
    ifile->fmt_ctx = fmt_ctx;
    if (scan_all_pmts_set)
        av_dict_set(&format_opts, "scan_all_pmts", NULL, AV_DICT_MATCH_CASE);
@@ -2952,7 +2916,6 @@ static int open_input_file(InputFile *ifile, const char *filename,
            ist->dec_ctx->pkt_timebase = stream->time_base;
            ist->dec_ctx->framerate = stream->avg_frame_rate;
 #if FF_API_LAVF_AVCTX
-            ist->dec_ctx->properties = stream->codec->properties;
            ist->dec_ctx->coded_width = stream->codec->coded_width;
            ist->dec_ctx->coded_height = stream->codec->coded_height;
 #endif
@@ -2990,8 +2953,7 @@ static void close_input_file(InputFile *ifile)
    avformat_close_input(&ifile->fmt_ctx);
 }

-static int probe_file(WriterContext *wctx, const char *filename,
-                      const char *print_filename)
+static int probe_file(WriterContext *wctx, const char *filename)
 {
    InputFile ifile = { 0 };
    int ret, i;
@@ -3000,7 +2962,7 @@ static int probe_file(WriterContext *wctx, const char *filename,
    do_read_frames = do_show_frames || do_count_frames;
    do_read_packets = do_show_packets || do_count_packets;

-    ret = open_input_file(&ifile, filename, print_filename);
+    ret = open_input_file(&ifile, filename);
    if (ret < 0)
        goto end;

@@ -3306,12 +3268,6 @@ static int opt_input_file_i(void *optctx, const char *opt, const char *arg)
    return 0;
 }

-static int opt_print_filename(void *optctx, const char *opt, const char *arg)
-{
-    print_input_filename = arg;
-    return 0;
-}
-
 void show_help_default(const char *opt, const char *arg)
 {
    av_log_set_callback(log_callback_help);
@@ -3501,7 +3457,7 @@ static int opt_sections(void *optctx, const char *opt, const char *arg)
    return 0;
 }

-static int opt_show_versions(void *optctx, const char *opt, const char *arg)
+static int opt_show_versions(const char *opt, const char *arg)
 {
    mark_section_show_entries(SECTION_ID_PROGRAM_VERSION, 1, NULL);
    mark_section_show_entries(SECTION_ID_LIBRARY_VERSION, 1, NULL);
@@ -3509,7 +3465,7 @@ static int opt_show_versions(void *optctx, const char *opt, const char *arg)
 }

 #define DEFINE_OPT_SHOW_SECTION(section, target_section_id)             \
-    static int opt_show_##section(void *optctx, const char *opt, const char *arg) \
+    static int opt_show_##section(const char *opt, const char *arg)     \
    {                                                                   \
        mark_section_show_entries(SECTION_ID_##target_section_id, 1, NULL); \
        return 0;                                                       \
@@ -3537,40 +3493,39 @@ static const OptionDef real_options[] = {
      "use sexagesimal format HOURS:MM:SS.MICROSECONDS for time units" },
    { "pretty", 0, {.func_arg = opt_pretty},
      "prettify the format of displayed values, make it more human readable" },
-    { "print_format", OPT_STRING | HAS_ARG, { &print_format },
+    { "print_format", OPT_STRING | HAS_ARG, {(void*)&print_format},
      "set the output printing format (available formats are: default, compact, csv, flat, ini, json, xml)", "format" },
-    { "of", OPT_STRING | HAS_ARG, { &print_format }, "alias for -print_format", "format" },
-    { "select_streams", OPT_STRING | HAS_ARG, { &stream_specifier }, "select the specified streams", "stream_specifier" },
+    { "of", OPT_STRING | HAS_ARG, {(void*)&print_format}, "alias for -print_format", "format" },
+    { "select_streams", OPT_STRING | HAS_ARG, {(void*)&stream_specifier}, "select the specified streams", "stream_specifier" },
    { "sections", OPT_EXIT, {.func_arg = opt_sections}, "print sections structure and section information, and exit" },
-    { "show_data",    OPT_BOOL, { &do_show_data }, "show packets data" },
-    { "show_data_hash", OPT_STRING | HAS_ARG, { &show_data_hash }, "show packets data hash" },
-    { "show_error",   0, { .func_arg = &opt_show_error },  "show probing error" },
-    { "show_format",  0, { .func_arg = &opt_show_format }, "show format/container info" },
-    { "show_frames",  0, { .func_arg = &opt_show_frames }, "show frames info" },
+    { "show_data",    OPT_BOOL, {(void*)&do_show_data}, "show packets data" },
+    { "show_data_hash", OPT_STRING | HAS_ARG, {(void*)&show_data_hash}, "show packets data hash" },
+    { "show_error",   0, {(void*)&opt_show_error},  "show probing error" },
+    { "show_format",  0, {(void*)&opt_show_format}, "show format/container info" },
+    { "show_frames",  0, {(void*)&opt_show_frames}, "show frames info" },
    { "show_format_entry", HAS_ARG, {.func_arg = opt_show_format_entry},
      "show a particular entry from the format/container info", "entry" },
    { "show_entries", HAS_ARG, {.func_arg = opt_show_entries},
      "show a set of specified entries", "entry_list" },
 #if HAVE_THREADS
-    { "show_log", OPT_INT|HAS_ARG, { &do_show_log }, "show log" },
+    { "show_log", OPT_INT|HAS_ARG, {(void*)&do_show_log}, "show log" },
 #endif
-    { "show_packets", 0, { .func_arg = &opt_show_packets }, "show packets info" },
-    { "show_programs", 0, { .func_arg = &opt_show_programs }, "show programs info" },
-    { "show_streams", 0, { .func_arg = &opt_show_streams }, "show streams info" },
-    { "show_chapters", 0, { .func_arg = &opt_show_chapters }, "show chapters info" },
-    { "count_frames", OPT_BOOL, { &do_count_frames }, "count the number of frames per stream" },
-    { "count_packets", OPT_BOOL, { &do_count_packets }, "count the number of packets per stream" },
-    { "show_program_version",  0, { .func_arg = &opt_show_program_version },  "show ffprobe version" },
-    { "show_library_versions", 0, { .func_arg = &opt_show_library_versions }, "show library versions" },
-    { "show_versions",         0, { .func_arg = &opt_show_versions }, "show program and library versions" },
-    { "show_pixel_formats", 0, { .func_arg = &opt_show_pixel_formats }, "show pixel format descriptions" },
-    { "show_private_data", OPT_BOOL, { &show_private_data }, "show private data" },
-    { "private",           OPT_BOOL, { &show_private_data }, "same as show_private_data" },
+    { "show_packets", 0, {(void*)&opt_show_packets}, "show packets info" },
+    { "show_programs", 0, {(void*)&opt_show_programs}, "show programs info" },
+    { "show_streams", 0, {(void*)&opt_show_streams}, "show streams info" },
+    { "show_chapters", 0, {(void*)&opt_show_chapters}, "show chapters info" },
+    { "count_frames", OPT_BOOL, {(void*)&do_count_frames}, "count the number of frames per stream" },
+    { "count_packets", OPT_BOOL, {(void*)&do_count_packets}, "count the number of packets per stream" },
+    { "show_program_version",  0, {(void*)&opt_show_program_version},  "show ffprobe version" },
+    { "show_library_versions", 0, {(void*)&opt_show_library_versions}, "show library versions" },
+    { "show_versions",         0, {(void*)&opt_show_versions}, "show program and library versions" },
+    { "show_pixel_formats", 0, {(void*)&opt_show_pixel_formats}, "show pixel format descriptions" },
+    { "show_private_data", OPT_BOOL, {(void*)&show_private_data}, "show private data" },
+    { "private",           OPT_BOOL, {(void*)&show_private_data}, "same as show_private_data" },
    { "bitexact", OPT_BOOL, {&do_bitexact}, "force bitexact output" },
    { "read_intervals", HAS_ARG, {.func_arg = opt_read_intervals}, "set read intervals", "read_intervals" },
    { "default", HAS_ARG | OPT_AUDIO | OPT_VIDEO | OPT_EXPERT, {.func_arg = opt_default}, "generic catch all option", "" },
    { "i", HAS_ARG, {.func_arg = opt_input_file_i}, "read specified file", "input_file"},
-    { "print_filename", HAS_ARG, {.func_arg = opt_print_filename}, "override the printed input filename", "print_file"},
    { "find_stream_info", OPT_BOOL | OPT_INPUT | OPT_EXPERT, { &find_stream_info },
        "read and decode the streams to fill missing information with heuristics" },
    { NULL, },
@@ -3719,7 +3674,7 @@ int main(int argc, char **argv)
            av_log(NULL, AV_LOG_ERROR, "Use -h to get full help or, even better, run 'man %s'.\n", program_name);
            ret = AVERROR(EINVAL);
        } else if (input_filename) {
-            ret = probe_file(wctx, input_filename, print_input_filename);
+            ret = probe_file(wctx, input_filename);
            if (ret < 0 && do_show_error)
                show_error(wctx, ret);
        }
--- a/libavcodec/012v.c
+++ b/libavcodec/012v.c
@@ -131,8 +131,8 @@ static int zero12v_decode_frame(AVCodecContext *avctx, void *data,
            u = x/2 + (uint16_t *)(pic->data[1] + line * pic->linesize[1]);
            v = x/2 + (uint16_t *)(pic->data[2] + line * pic->linesize[2]);
            memcpy(y, y_temp, sizeof(*y) * (width - x));
-            memcpy(u, u_temp, sizeof(*u) * (width - x + 1) / 2);
-            memcpy(v, v_temp, sizeof(*v) * (width - x + 1) / 2);
+            memcpy(u, u_temp, sizeof(*u) * ((width - x + 1) / 2));
+            memcpy(v, v_temp, sizeof(*v) * ((width - x + 1) / 2));
        }

        line_end += stride;
--- a/libavcodec/4xm.c
+++ b/libavcodec/4xm.c
@@ -145,7 +145,7 @@ typedef struct FourXContext {
    int mv[256];
    VLC pre_vlc;
    int last_dc;
-    DECLARE_ALIGNED(32, int16_t, block)[6][64];
+    DECLARE_ALIGNED(16, int16_t, block)[6][64];
    void *bitstream_buffer;
    unsigned int bitstream_buffer_size;
    int version;
@@ -498,8 +498,8 @@ static int decode_i_block(FourXContext *f, int16_t *block)
 {
    int code, i, j, level, val;

-    if (get_bits_left(&f->gb) < 2){
-        av_log(f->avctx, AV_LOG_ERROR, "%d bits left before decode_i_block()\n", get_bits_left(&f->gb));
+    if (get_bits_left(&f->pre_gb) < 2) {
+        av_log(f->avctx, AV_LOG_ERROR, "%d bits left before decode_i_block()\n", get_bits_left(&f->pre_gb));
        return AVERROR_INVALIDDATA;
    }

@@ -703,7 +703,6 @@ static const uint8_t *read_huffman_tables(FourXContext *f,
        len_tab[j]  = len;
    }

-    ff_free_vlc(&f->pre_vlc);
    if (init_vlc(&f->pre_vlc, ACDC_VLC_BITS, 257, len_tab, 1, 1,
                 bits_tab, 4, 4, 0))
        return NULL;
--- a/libavcodec/8bps.c
+++ b/libavcodec/8bps.c
@@ -70,6 +70,9 @@ static int decode_frame(AVCodecContext *avctx, void *data,
    unsigned char *planemap = c->planemap;
    int ret;

+    if (buf_size < planes * height *2)
+        return AVERROR_INVALIDDATA;
+
    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
        return ret;

--- a/libavcodec/8svx.c
+++ b/libavcodec/8svx.c
@@ -164,7 +164,8 @@ static av_cold int eightsvx_decode_init(AVCodecContext *avctx)
    case AV_CODEC_ID_8SVX_FIB: esc->table = fibonacci;    break;
    case AV_CODEC_ID_8SVX_EXP: esc->table = exponential;  break;
    default:
-        av_assert1(0);
+        av_log(avctx, AV_LOG_ERROR, "Invalid codec id %d.\n", avctx->codec->id);
+        return AVERROR_INVALIDDATA;
    }
    avctx->sample_fmt = AV_SAMPLE_FMT_U8P;

--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -6,18 +6,12 @@ HEADERS = ac3_parser.h                                                  \
          avcodec.h                                                     \
          avdct.h                                                       \
          avfft.h                                                       \
-          bsf.h                                                         \
-          codec.h                                                       \
-          codec_desc.h                                                  \
-          codec_id.h                                                    \
-          codec_par.h                                                   \
          d3d11va.h                                                     \
          dirac.h                                                       \
          dv_profile.h                                                  \
          dxva2.h                                                       \
          jni.h                                                         \
          mediacodec.h                                                  \
-          packet.h                                                      \
          qsv.h                                                         \
          vaapi.h                                                       \
          vdpau.h                                                       \
@@ -173,16 +167,12 @@ OBJS-$(CONFIG_AAC_ENCODER)             += aacenc.o aaccoder.o aacenctab.o    \
                                          aacenc_ltp.o \
                                          aacenc_pred.o \
                                          psymodel.o mpeg4audio.o kbdwin.o cbrt_data.o
-OBJS-$(CONFIG_AAC_MF_ENCODER)          += mfenc.o mf_utils.o
 OBJS-$(CONFIG_AASC_DECODER)            += aasc.o msrledec.o
 OBJS-$(CONFIG_AC3_DECODER)             += ac3dec_float.o ac3dec_data.o ac3.o kbdwin.o ac3tab.o
 OBJS-$(CONFIG_AC3_FIXED_DECODER)       += ac3dec_fixed.o ac3dec_data.o ac3.o kbdwin.o ac3tab.o
 OBJS-$(CONFIG_AC3_ENCODER)             += ac3enc_float.o ac3enc.o ac3tab.o \
                                          ac3.o kbdwin.o
 OBJS-$(CONFIG_AC3_FIXED_ENCODER)       += ac3enc_fixed.o ac3enc.o ac3tab.o ac3.o
-OBJS-$(CONFIG_AC3_MF_ENCODER)          += mfenc.o mf_utils.o
-OBJS-$(CONFIG_ACELP_KELVIN_DECODER)    += g729dec.o lsp.o celp_math.o celp_filters.o acelp_filters.o acelp_pitch_delay.o acelp_vectors.o g729postfilter.o
-OBJS-$(CONFIG_AGM_DECODER)             += agm.o
 OBJS-$(CONFIG_AIC_DECODER)             += aic.o
 OBJS-$(CONFIG_ALAC_DECODER)            += alac.o alac_data.o alacdsp.o
 OBJS-$(CONFIG_ALAC_ENCODER)            += alacenc.o alac_data.o
@@ -202,13 +192,12 @@ OBJS-$(CONFIG_AMV_ENCODER)             += mjpegenc.o mjpegenc_common.o \
 OBJS-$(CONFIG_ANM_DECODER)             += anm.o
 OBJS-$(CONFIG_ANSI_DECODER)            += ansi.o cga_data.o
 OBJS-$(CONFIG_APE_DECODER)             += apedec.o
-OBJS-$(CONFIG_APTX_DECODER)            += aptxdec.o aptx.o
-OBJS-$(CONFIG_APTX_ENCODER)            += aptxenc.o aptx.o
-OBJS-$(CONFIG_APTX_HD_DECODER)         += aptxdec.o aptx.o
-OBJS-$(CONFIG_APTX_HD_ENCODER)         += aptxenc.o aptx.o
+OBJS-$(CONFIG_APTX_DECODER)            += aptx.o
+OBJS-$(CONFIG_APTX_ENCODER)            += aptx.o
+OBJS-$(CONFIG_APTX_HD_DECODER)         += aptx.o
+OBJS-$(CONFIG_APTX_HD_ENCODER)         += aptx.o
 OBJS-$(CONFIG_APNG_DECODER)            += png.o pngdec.o pngdsp.o
 OBJS-$(CONFIG_APNG_ENCODER)            += png.o pngenc.o
-OBJS-$(CONFIG_ARBC_DECODER)            += arbc.o
 OBJS-$(CONFIG_SSA_DECODER)             += assdec.o ass.o
 OBJS-$(CONFIG_SSA_ENCODER)             += assenc.o ass.o
 OBJS-$(CONFIG_ASS_DECODER)             += assdec.o ass.o
@@ -250,9 +239,8 @@ OBJS-$(CONFIG_BRENDER_PIX_DECODER)     += brenderpix.o
 OBJS-$(CONFIG_C93_DECODER)             += c93.o
 OBJS-$(CONFIG_CAVS_DECODER)            += cavs.o cavsdec.o cavsdsp.o \
                                          cavsdata.o
-OBJS-$(CONFIG_CCAPTION_DECODER)        += ccaption_dec.o ass.o
+OBJS-$(CONFIG_CCAPTION_DECODER)        += ccaption_dec.o
 OBJS-$(CONFIG_CDGRAPHICS_DECODER)      += cdgraphics.o
-OBJS-$(CONFIG_CDTOONS_DECODER)         += cdtoons.o
 OBJS-$(CONFIG_CDXL_DECODER)            += cdxl.o
 OBJS-$(CONFIG_CFHD_DECODER)            += cfhd.o cfhddata.o
 OBJS-$(CONFIG_CINEPAK_DECODER)         += cinepak.o
@@ -273,7 +261,6 @@ OBJS-$(CONFIG_DCA_DECODER)             += dcadec.o dca.o dcadata.o dcahuff.o \
 OBJS-$(CONFIG_DCA_ENCODER)             += dcaenc.o dca.o dcadata.o dcahuff.o \
                                          dcaadpcm.o
 OBJS-$(CONFIG_DDS_DECODER)             += dds.o
-OBJS-$(CONFIG_DERF_DPCM_DECODER)       += dpcm.o
 OBJS-$(CONFIG_DIRAC_DECODER)           += diracdec.o dirac.o diracdsp.o diractab.o \
                                          dirac_arith.o dirac_dwt.o dirac_vlc.o
 OBJS-$(CONFIG_DFA_DECODER)             += dfa.o
@@ -292,8 +279,8 @@ OBJS-$(CONFIG_DSS_SP_DECODER)          += dss_sp.o
 OBJS-$(CONFIG_DST_DECODER)             += dstdec.o dsd.o
 OBJS-$(CONFIG_DVBSUB_DECODER)          += dvbsubdec.o
 OBJS-$(CONFIG_DVBSUB_ENCODER)          += dvbsub.o
-OBJS-$(CONFIG_DVDSUB_DECODER)          += dvdsubdec.o dvdsub.o
-OBJS-$(CONFIG_DVDSUB_ENCODER)          += dvdsubenc.o dvdsub.o
+OBJS-$(CONFIG_DVDSUB_DECODER)          += dvdsubdec.o
+OBJS-$(CONFIG_DVDSUB_ENCODER)          += dvdsubenc.o
 OBJS-$(CONFIG_DVAUDIO_DECODER)         += dvaudiodec.o
 OBJS-$(CONFIG_DVVIDEO_DECODER)         += dvdec.o dv.o dvdata.o
 OBJS-$(CONFIG_DVVIDEO_ENCODER)         += dvenc.o dv.o dvdata.o
@@ -361,7 +348,6 @@ OBJS-$(CONFIG_H264_DECODER)            += h264dec.o h264_cabac.o h264_cavlc.o \
 OBJS-$(CONFIG_H264_AMF_ENCODER)        += amfenc_h264.o
 OBJS-$(CONFIG_H264_CUVID_DECODER)      += cuviddec.o
 OBJS-$(CONFIG_H264_MEDIACODEC_DECODER) += mediacodecdec.o
-OBJS-$(CONFIG_H264_MF_ENCODER)         += mfenc.o mf_utils.o
 OBJS-$(CONFIG_H264_MMAL_DECODER)       += mmaldec.o
 OBJS-$(CONFIG_H264_NVENC_ENCODER)      += nvenc_h264.o
 OBJS-$(CONFIG_NVENC_ENCODER)           += nvenc_h264.o
@@ -376,15 +362,12 @@ OBJS-$(CONFIG_H264_V4L2M2M_DECODER)    += v4l2_m2m_dec.o
 OBJS-$(CONFIG_H264_V4L2M2M_ENCODER)    += v4l2_m2m_enc.o
 OBJS-$(CONFIG_HAP_DECODER)             += hapdec.o hap.o
 OBJS-$(CONFIG_HAP_ENCODER)             += hapenc.o hap.o
-OBJS-$(CONFIG_HCA_DECODER)             += hcadec.o
-OBJS-$(CONFIG_HCOM_DECODER)            += hcom.o
 OBJS-$(CONFIG_HEVC_DECODER)            += hevcdec.o hevc_mvs.o \
                                          hevc_cabac.o hevc_refs.o hevcpred.o    \
                                          hevcdsp.o hevc_filter.o hevc_data.o
 OBJS-$(CONFIG_HEVC_AMF_ENCODER)        += amfenc_hevc.o
 OBJS-$(CONFIG_HEVC_CUVID_DECODER)      += cuviddec.o
 OBJS-$(CONFIG_HEVC_MEDIACODEC_DECODER) += mediacodecdec.o
-OBJS-$(CONFIG_HEVC_MF_ENCODER)         += mfenc.o mf_utils.o
 OBJS-$(CONFIG_HEVC_NVENC_ENCODER)      += nvenc_hevc.o
 OBJS-$(CONFIG_NVENC_HEVC_ENCODER)      += nvenc_hevc.o
 OBJS-$(CONFIG_HEVC_QSV_DECODER)        += qsvdec_h2645.o
@@ -400,14 +383,12 @@ OBJS-$(CONFIG_HQ_HQA_DECODER)          += hq_hqa.o hq_hqadata.o hq_hqadsp.o \
 OBJS-$(CONFIG_HQX_DECODER)             += hqx.o hqxvlc.o hqxdsp.o canopus.o
 OBJS-$(CONFIG_HUFFYUV_DECODER)         += huffyuv.o huffyuvdec.o
 OBJS-$(CONFIG_HUFFYUV_ENCODER)         += huffyuv.o huffyuvenc.o
-OBJS-$(CONFIG_HYMT_DECODER)            += huffyuv.o huffyuvdec.o
 OBJS-$(CONFIG_IDCIN_DECODER)           += idcinvideo.o
 OBJS-$(CONFIG_IDF_DECODER)             += bintext.o cga_data.o
 OBJS-$(CONFIG_IFF_ILBM_DECODER)        += iff.o
 OBJS-$(CONFIG_ILBC_DECODER)            += ilbcdec.o
 OBJS-$(CONFIG_IMC_DECODER)             += imc.o
 OBJS-$(CONFIG_IMM4_DECODER)            += imm4.o
-OBJS-$(CONFIG_IMM5_DECODER)            += imm5.o
 OBJS-$(CONFIG_INDEO2_DECODER)          += indeo2.o
 OBJS-$(CONFIG_INDEO3_DECODER)          += indeo3.o
 OBJS-$(CONFIG_INDEO4_DECODER)          += indeo4.o ivi.o
@@ -428,7 +409,6 @@ OBJS-$(CONFIG_KMVC_DECODER)            += kmvc.o
 OBJS-$(CONFIG_LAGARITH_DECODER)        += lagarith.o lagarithrac.o
 OBJS-$(CONFIG_LJPEG_ENCODER)           += ljpegenc.o mjpegenc_common.o
 OBJS-$(CONFIG_LOCO_DECODER)            += loco.o
-OBJS-$(CONFIG_LSCR_DECODER)            += png.o pngdec.o pngdsp.o
 OBJS-$(CONFIG_M101_DECODER)            += m101.o
 OBJS-$(CONFIG_MACE3_DECODER)           += mace.o
 OBJS-$(CONFIG_MACE6_DECODER)           += mace.o
@@ -440,7 +420,6 @@ OBJS-$(CONFIG_METASOUND_DECODER)       += metasound.o metasound_data.o \
 OBJS-$(CONFIG_MICRODVD_DECODER)        += microdvddec.o ass.o
 OBJS-$(CONFIG_MIMIC_DECODER)           += mimic.o
 OBJS-$(CONFIG_MJPEG_DECODER)           += mjpegdec.o
-OBJS-$(CONFIG_MJPEG_QSV_DECODER)       += qsvdec_other.o
 OBJS-$(CONFIG_MJPEG_ENCODER)           += mjpegenc.o mjpegenc_common.o \
                                          mjpegenc_huffman.o
 OBJS-$(CONFIG_MJPEGB_DECODER)          += mjpegbdec.o
@@ -462,7 +441,6 @@ OBJS-$(CONFIG_MP2FIXED_ENCODER)        += mpegaudioenc_fixed.o mpegaudio.o \
                                          mpegaudiodata.o mpegaudiodsp_data.o
 OBJS-$(CONFIG_MP2FLOAT_DECODER)        += mpegaudiodec_float.o
 OBJS-$(CONFIG_MP3_DECODER)             += mpegaudiodec_fixed.o
-OBJS-$(CONFIG_MP3_MF_ENCODER)          += mfenc.o mf_utils.o
 OBJS-$(CONFIG_MP3ADU_DECODER)          += mpegaudiodec_fixed.o
 OBJS-$(CONFIG_MP3ADUFLOAT_DECODER)     += mpegaudiodec_float.o
 OBJS-$(CONFIG_MP3FLOAT_DECODER)        += mpegaudiodec_float.o
@@ -505,23 +483,18 @@ OBJS-$(CONFIG_MSVIDEO1_DECODER)        += msvideo1.o
 OBJS-$(CONFIG_MSVIDEO1_ENCODER)        += msvideo1enc.o elbg.o
 OBJS-$(CONFIG_MSZH_DECODER)            += lcldec.o
 OBJS-$(CONFIG_MTS2_DECODER)            += mss4.o
-OBJS-$(CONFIG_MV30_DECODER)            += mv30.o
 OBJS-$(CONFIG_MVC1_DECODER)            += mvcdec.o
 OBJS-$(CONFIG_MVC2_DECODER)            += mvcdec.o
-OBJS-$(CONFIG_MVDV_DECODER)            += midivid.o
-OBJS-$(CONFIG_MVHA_DECODER)            += mvha.o
 OBJS-$(CONFIG_MWSC_DECODER)            += mwsc.o
 OBJS-$(CONFIG_MXPEG_DECODER)           += mxpegdec.o
 OBJS-$(CONFIG_NELLYMOSER_DECODER)      += nellymoserdec.o nellymoser.o
 OBJS-$(CONFIG_NELLYMOSER_ENCODER)      += nellymoserenc.o nellymoser.o
-OBJS-$(CONFIG_NOTCHLC_DECODER)         += notchlc.o
 OBJS-$(CONFIG_NUV_DECODER)             += nuv.o rtjpeg.o
 OBJS-$(CONFIG_ON2AVC_DECODER)          += on2avc.o on2avcdata.o
 OBJS-$(CONFIG_OPUS_DECODER)            += opusdec.o opus.o opus_celt.o opus_rc.o \
-                                          opus_pvq.o opus_silk.o opustab.o vorbis_data.o \
-                                          opusdsp.o
+                                          opus_pvq.o opus_silk.o opustab.o vorbis_data.o
 OBJS-$(CONFIG_OPUS_ENCODER)            += opusenc.o opus.o opus_rc.o opustab.o opus_pvq.o \
-                                          opusenc_psy.o vorbis_data.o
+                                          opusenc_psy.o
 OBJS-$(CONFIG_PAF_AUDIO_DECODER)       += pafaudio.o
 OBJS-$(CONFIG_PAF_VIDEO_DECODER)       += pafvideo.o
 OBJS-$(CONFIG_PAM_DECODER)             += pnmdec.o pnm.o
@@ -530,7 +503,6 @@ OBJS-$(CONFIG_PBM_DECODER)             += pnmdec.o pnm.o
 OBJS-$(CONFIG_PBM_ENCODER)             += pnmenc.o
 OBJS-$(CONFIG_PCX_DECODER)             += pcx.o
 OBJS-$(CONFIG_PCX_ENCODER)             += pcxenc.o
-OBJS-$(CONFIG_PFM_DECODER)             += pnmdec.o pnm.o
 OBJS-$(CONFIG_PGM_DECODER)             += pnmdec.o pnm.o
 OBJS-$(CONFIG_PGM_ENCODER)             += pnmenc.o
 OBJS-$(CONFIG_PGMYUV_DECODER)          += pnmdec.o pnm.o
@@ -600,14 +572,13 @@ OBJS-$(CONFIG_SIPR_DECODER)            += sipr.o acelp_pitch_delay.o \
                                          celp_math.o acelp_vectors.o \
                                          acelp_filters.o celp_filters.o \
                                          sipr16k.o
-OBJS-$(CONFIG_SIREN_DECODER)           += siren.o
 OBJS-$(CONFIG_SMACKAUD_DECODER)        += smacker.o
 OBJS-$(CONFIG_SMACKER_DECODER)         += smacker.o
 OBJS-$(CONFIG_SMC_DECODER)             += smc.o
 OBJS-$(CONFIG_SMVJPEG_DECODER)         += smvjpegdec.o
 OBJS-$(CONFIG_SNOW_DECODER)            += snowdec.o snow.o snow_dwt.o
 OBJS-$(CONFIG_SNOW_ENCODER)            += snowenc.o snow.o snow_dwt.o             \
-                                          h263.o h263data.o ituh263enc.o
+                                          h263.o ituh263enc.o
 OBJS-$(CONFIG_SOL_DPCM_DECODER)        += dpcm.o
 OBJS-$(CONFIG_SONIC_DECODER)           += sonic.o
 OBJS-$(CONFIG_SONIC_ENCODER)           += sonic.o
@@ -627,10 +598,10 @@ OBJS-$(CONFIG_SUNRAST_ENCODER)         += sunrastenc.o
 OBJS-$(CONFIG_LIBRSVG_DECODER)         += librsvgdec.o
 OBJS-$(CONFIG_SBC_DECODER)             += sbcdec.o sbcdec_data.o sbc.o
 OBJS-$(CONFIG_SBC_ENCODER)             += sbcenc.o sbc.o sbcdsp.o sbcdsp_data.o
-OBJS-$(CONFIG_SVQ1_DECODER)            += svq1dec.o svq1.o h263data.o
+OBJS-$(CONFIG_SVQ1_DECODER)            += svq1dec.o svq1.o svq13.o h263data.o
 OBJS-$(CONFIG_SVQ1_ENCODER)            += svq1enc.o svq1.o  h263data.o  \
                                          h263.o ituh263enc.o
-OBJS-$(CONFIG_SVQ3_DECODER)            += svq3.o mpegutils.o h264data.o
+OBJS-$(CONFIG_SVQ3_DECODER)            += svq3.o svq13.o mpegutils.o h264data.o
 OBJS-$(CONFIG_TEXT_DECODER)            += textdec.o ass.o
 OBJS-$(CONFIG_TEXT_ENCODER)            += srtenc.o ass_split.o
 OBJS-$(CONFIG_TAK_DECODER)             += takdec.o tak.o takdsp.o
@@ -639,7 +610,7 @@ OBJS-$(CONFIG_TARGA_ENCODER)           += targaenc.o rle.o
 OBJS-$(CONFIG_TARGA_Y216_DECODER)      += targa_y216dec.o
 OBJS-$(CONFIG_TDSC_DECODER)            += tdsc.o
 OBJS-$(CONFIG_TIERTEXSEQVIDEO_DECODER) += tiertexseqv.o
-OBJS-$(CONFIG_TIFF_DECODER)            += tiff.o lzw.o faxcompr.o tiff_data.o tiff_common.o mjpegdec.o
+OBJS-$(CONFIG_TIFF_DECODER)            += tiff.o lzw.o faxcompr.o tiff_data.o tiff_common.o
 OBJS-$(CONFIG_TIFF_ENCODER)            += tiffenc.o rle.o lzwenc.o tiff_data.o
 OBJS-$(CONFIG_TMV_DECODER)             += tmv.o cga_data.o
 OBJS-$(CONFIG_TRUEHD_DECODER)          += mlpdec.o mlpdsp.o
@@ -705,11 +676,10 @@ OBJS-$(CONFIG_VP9_CUVID_DECODER)       += cuviddec.o
 OBJS-$(CONFIG_VP9_MEDIACODEC_DECODER)  += mediacodecdec.o
 OBJS-$(CONFIG_VP9_RKMPP_DECODER)       += rkmppdec.o
 OBJS-$(CONFIG_VP9_VAAPI_ENCODER)       += vaapi_encode_vp9.o
-OBJS-$(CONFIG_VP9_QSV_ENCODER)         += qsvenc_vp9.o
 OBJS-$(CONFIG_VPLAYER_DECODER)         += textdec.o ass.o
 OBJS-$(CONFIG_VP9_V4L2M2M_DECODER)     += v4l2_m2m_dec.o
 OBJS-$(CONFIG_VQA_DECODER)             += vqavideo.o
-OBJS-$(CONFIG_WAVPACK_DECODER)         += wavpack.o dsd.o
+OBJS-$(CONFIG_WAVPACK_DECODER)         += wavpack.o
 OBJS-$(CONFIG_WAVPACK_ENCODER)         += wavpackenc.o
 OBJS-$(CONFIG_WCMV_DECODER)            += wcmv.o
 OBJS-$(CONFIG_WEBP_DECODER)            += webp.o
@@ -725,7 +695,7 @@ OBJS-$(CONFIG_WMAVOICE_DECODER)        += wmavoice.o \
                                          celp_filters.o \
                                          acelp_vectors.o acelp_filters.o
 OBJS-$(CONFIG_WMV1_DECODER)            += msmpeg4dec.o msmpeg4.o msmpeg4data.o
-OBJS-$(CONFIG_WMV1_ENCODER)            += msmpeg4enc.o msmpeg4.o msmpeg4data.o
+OBJS-$(CONFIG_WMV1_ENCODER)            += msmpeg4enc.o
 OBJS-$(CONFIG_WMV2_DECODER)            += wmv2dec.o wmv2.o wmv2data.o \
                                          msmpeg4dec.o msmpeg4.o msmpeg4data.o
 OBJS-$(CONFIG_WMV2_ENCODER)            += wmv2enc.o wmv2.o wmv2data.o \
@@ -767,7 +737,6 @@ OBJS-$(CONFIG_PCM_ALAW_DECODER)           += pcm.o
 OBJS-$(CONFIG_PCM_ALAW_ENCODER)           += pcm.o
 OBJS-$(CONFIG_PCM_BLURAY_DECODER)         += pcm-bluray.o
 OBJS-$(CONFIG_PCM_DVD_DECODER)            += pcm-dvd.o
-OBJS-$(CONFIG_PCM_DVD_ENCODER)            += pcm-dvdenc.o
 OBJS-$(CONFIG_PCM_F16LE_DECODER)          += pcm.o
 OBJS-$(CONFIG_PCM_F24LE_DECODER)          += pcm.o
 OBJS-$(CONFIG_PCM_F32BE_DECODER)          += pcm.o
@@ -827,14 +796,13 @@ OBJS-$(CONFIG_PCM_U32LE_DECODER)          += pcm.o
 OBJS-$(CONFIG_PCM_U32LE_ENCODER)          += pcm.o
 OBJS-$(CONFIG_PCM_VIDC_DECODER)           += pcm.o
 OBJS-$(CONFIG_PCM_VIDC_ENCODER)           += pcm.o
+OBJS-$(CONFIG_PCM_ZORK_DECODER)           += pcm.o

 OBJS-$(CONFIG_ADPCM_4XM_DECODER)          += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_ADX_DECODER)          += adxdec.o adx.o
 OBJS-$(CONFIG_ADPCM_ADX_ENCODER)          += adxenc.o adx.o
 OBJS-$(CONFIG_ADPCM_AFC_DECODER)          += adpcm.o adpcm_data.o
-OBJS-$(CONFIG_ADPCM_AGM_DECODER)          += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_AICA_DECODER)         += adpcm.o adpcm_data.o
-OBJS-$(CONFIG_ADPCM_ARGO_DECODER)         += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_CT_DECODER)           += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_DTK_DECODER)          += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_EA_DECODER)           += adpcm.o adpcm_data.o
@@ -850,23 +818,17 @@ OBJS-$(CONFIG_ADPCM_G726_ENCODER)         += g726.o
 OBJS-$(CONFIG_ADPCM_G726LE_DECODER)       += g726.o
 OBJS-$(CONFIG_ADPCM_G726LE_ENCODER)       += g726.o
 OBJS-$(CONFIG_ADPCM_IMA_AMV_DECODER)      += adpcm.o adpcm_data.o
-OBJS-$(CONFIG_ADPCM_IMA_ALP_DECODER)      += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_APC_DECODER)      += adpcm.o adpcm_data.o
-OBJS-$(CONFIG_ADPCM_IMA_APM_DECODER)      += adpcm.o adpcm_data.o
-OBJS-$(CONFIG_ADPCM_IMA_CUNNING_DECODER)  += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_DAT4_DECODER)     += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_DK3_DECODER)      += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_DK4_DECODER)      += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_EA_EACS_DECODER)  += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_EA_SEAD_DECODER)  += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_ISS_DECODER)      += adpcm.o adpcm_data.o
-OBJS-$(CONFIG_ADPCM_IMA_MTF_DECODER)      += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_OKI_DECODER)      += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_QT_DECODER)       += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_QT_ENCODER)       += adpcmenc.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_RAD_DECODER)      += adpcm.o adpcm_data.o
-OBJS-$(CONFIG_ADPCM_IMA_SSI_DECODER)      += adpcm.o adpcm_data.o
-OBJS-$(CONFIG_ADPCM_IMA_SSI_ENCODER)      += adpcmenc.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_SMJPEG_DECODER)   += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_WAV_DECODER)      += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_WAV_ENCODER)      += adpcmenc.o adpcm_data.o
@@ -886,7 +848,6 @@ OBJS-$(CONFIG_ADPCM_VIMA_DECODER)         += vima.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_XA_DECODER)           += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_YAMAHA_DECODER)       += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_YAMAHA_ENCODER)       += adpcmenc.o adpcm_data.o
-OBJS-$(CONFIG_ADPCM_ZORK_DECODER)         += adpcm.o adpcm_data.o

 # hardware accelerators
 OBJS-$(CONFIG_D3D11VA)                    += dxva2.o
@@ -909,7 +870,7 @@ OBJS-$(CONFIG_HEVC_D3D11VA_HWACCEL)       += dxva2_hevc.o
 OBJS-$(CONFIG_HEVC_DXVA2_HWACCEL)         += dxva2_hevc.o
 OBJS-$(CONFIG_HEVC_NVDEC_HWACCEL)         += nvdec_hevc.o
 OBJS-$(CONFIG_HEVC_QSV_HWACCEL)           += qsvdec_h2645.o
-OBJS-$(CONFIG_HEVC_VAAPI_HWACCEL)         += vaapi_hevc.o h265_profile_level.o
+OBJS-$(CONFIG_HEVC_VAAPI_HWACCEL)         += vaapi_hevc.o
 OBJS-$(CONFIG_HEVC_VDPAU_HWACCEL)         += vdpau_hevc.o
 OBJS-$(CONFIG_MJPEG_NVDEC_HWACCEL)        += nvdec_mjpeg.o
 OBJS-$(CONFIG_MJPEG_VAAPI_HWACCEL)        += vaapi_mjpeg.o
@@ -941,7 +902,6 @@ OBJS-$(CONFIG_VP9_D3D11VA_HWACCEL)        += dxva2_vp9.o
 OBJS-$(CONFIG_VP9_DXVA2_HWACCEL)          += dxva2_vp9.o
 OBJS-$(CONFIG_VP9_NVDEC_HWACCEL)          += nvdec_vp9.o
 OBJS-$(CONFIG_VP9_VAAPI_HWACCEL)          += vaapi_vp9.o
-OBJS-$(CONFIG_VP9_VDPAU_HWACCEL)          += vdpau_vp9.o
 OBJS-$(CONFIG_VP8_QSV_HWACCEL)            += qsvdec_other.o

 # libavformat dependencies
@@ -993,11 +953,9 @@ OBJS-$(CONFIG_PCM_ALAW_AT_ENCODER)        += audiotoolboxenc.o
 OBJS-$(CONFIG_PCM_MULAW_AT_ENCODER)       += audiotoolboxenc.o
 OBJS-$(CONFIG_LIBAOM_AV1_DECODER)         += libaomdec.o
 OBJS-$(CONFIG_LIBAOM_AV1_ENCODER)         += libaomenc.o
-OBJS-$(CONFIG_LIBARIBB24_DECODER)         += libaribb24.o ass.o
 OBJS-$(CONFIG_LIBCELT_DECODER)            += libcelt_dec.o
 OBJS-$(CONFIG_LIBCODEC2_DECODER)          += libcodec2.o codec2utils.o
 OBJS-$(CONFIG_LIBCODEC2_ENCODER)          += libcodec2.o codec2utils.o
-OBJS-$(CONFIG_LIBDAV1D_DECODER)           += libdav1d.o
 OBJS-$(CONFIG_LIBDAVS2_DECODER)           += libdavs2.o
 OBJS-$(CONFIG_LIBFDK_AAC_DECODER)         += libfdk-aacdec.o
 OBJS-$(CONFIG_LIBFDK_AAC_ENCODER)         += libfdk-aacenc.o
@@ -1020,7 +978,6 @@ OBJS-$(CONFIG_LIBOPUS_DECODER)            += libopusdec.o libopus.o     \
                                             vorbis_data.o
 OBJS-$(CONFIG_LIBOPUS_ENCODER)            += libopusenc.o libopus.o     \
                                             vorbis_data.o
-OBJS-$(CONFIG_LIBRAV1E_ENCODER)           += librav1e.o
 OBJS-$(CONFIG_LIBSHINE_ENCODER)           += libshine.o
 OBJS-$(CONFIG_LIBSPEEX_DECODER)           += libspeexdec.o
 OBJS-$(CONFIG_LIBSPEEX_ENCODER)           += libspeexenc.o
@@ -1066,20 +1023,18 @@ OBJS-$(CONFIG_DVD_NAV_PARSER)          += dvd_nav_parser.o
 OBJS-$(CONFIG_DVDSUB_PARSER)           += dvdsub_parser.o
 OBJS-$(CONFIG_FLAC_PARSER)             += flac_parser.o flacdata.o flac.o \
                                          vorbis_data.o
-OBJS-$(CONFIG_G723_1_PARSER)           += g723_1_parser.o
 OBJS-$(CONFIG_G729_PARSER)             += g729_parser.o
-OBJS-$(CONFIG_GIF_PARSER)              += gif_parser.o
 OBJS-$(CONFIG_GSM_PARSER)              += gsm_parser.o
 OBJS-$(CONFIG_H261_PARSER)             += h261_parser.o
 OBJS-$(CONFIG_H263_PARSER)             += h263_parser.o
 OBJS-$(CONFIG_H264_PARSER)             += h264_parser.o h264_sei.o h264data.o
 OBJS-$(CONFIG_HEVC_PARSER)             += hevc_parser.o hevc_data.o
-OBJS-$(CONFIG_JPEG2000_PARSER)         += jpeg2000_parser.o
 OBJS-$(CONFIG_MJPEG_PARSER)            += mjpeg_parser.o
-OBJS-$(CONFIG_MLP_PARSER)              += mlp_parse.o mlp_parser.o mlp.o
+OBJS-$(CONFIG_MLP_PARSER)              += mlp_parser.o mlp.o
 OBJS-$(CONFIG_MPEG4VIDEO_PARSER)       += mpeg4video_parser.o h263.o \
                                          mpeg4videodec.o mpeg4video.o \
                                          ituh263dec.o h263dec.o h263data.o
+OBJS-$(CONFIG_PNG_PARSER)              += png_parser.o
 OBJS-$(CONFIG_MPEGAUDIO_PARSER)        += mpegaudio_parser.o
 OBJS-$(CONFIG_MPEGVIDEO_PARSER)        += mpegvideo_parser.o    \
                                          mpeg12.o mpeg12data.o
@@ -1097,14 +1052,11 @@ OBJS-$(CONFIG_VC1_PARSER)              += vc1_parser.o vc1.o vc1data.o  \
 OBJS-$(CONFIG_VP3_PARSER)              += vp3_parser.o
 OBJS-$(CONFIG_VP8_PARSER)              += vp8_parser.o
 OBJS-$(CONFIG_VP9_PARSER)              += vp9_parser.o
-OBJS-$(CONFIG_WEBP_PARSER)             += webp_parser.o
 OBJS-$(CONFIG_XMA_PARSER)              += xma_parser.o

 # bitstream filters
 OBJS-$(CONFIG_AAC_ADTSTOASC_BSF)          += aac_adtstoasc_bsf.o mpeg4audio.o
 OBJS-$(CONFIG_AV1_METADATA_BSF)           += av1_metadata_bsf.o
-OBJS-$(CONFIG_AV1_FRAME_MERGE_BSF)        += av1_frame_merge_bsf.o
-OBJS-$(CONFIG_AV1_FRAME_SPLIT_BSF)        += av1_frame_split_bsf.o
 OBJS-$(CONFIG_CHOMP_BSF)                  += chomp_bsf.o
 OBJS-$(CONFIG_DUMP_EXTRADATA_BSF)         += dump_extradata_bsf.o
 OBJS-$(CONFIG_DCA_CORE_BSF)               += dca_core_bsf.o
@@ -1116,7 +1068,7 @@ OBJS-$(CONFIG_H264_METADATA_BSF)          += h264_metadata_bsf.o h264_levels.o
 OBJS-$(CONFIG_H264_MP4TOANNEXB_BSF)       += h264_mp4toannexb_bsf.o
 OBJS-$(CONFIG_H264_REDUNDANT_PPS_BSF)     += h264_redundant_pps_bsf.o
 OBJS-$(CONFIG_HAPQA_EXTRACT_BSF)          += hapqa_extract_bsf.o hap.o
-OBJS-$(CONFIG_HEVC_METADATA_BSF)          += h265_metadata_bsf.o h265_profile_level.o
+OBJS-$(CONFIG_HEVC_METADATA_BSF)          += h265_metadata_bsf.o
 OBJS-$(CONFIG_HEVC_MP4TOANNEXB_BSF)       += hevc_mp4toannexb_bsf.o
 OBJS-$(CONFIG_IMX_DUMP_HEADER_BSF)        += imx_dump_header_bsf.o
 OBJS-$(CONFIG_MJPEG2JPEG_BSF)             += mjpeg2jpeg_bsf.o
@@ -1128,13 +1080,9 @@ OBJS-$(CONFIG_MP3_HEADER_DECOMPRESS_BSF)  += mp3_header_decompress_bsf.o \
 OBJS-$(CONFIG_MPEG2_METADATA_BSF)         += mpeg2_metadata_bsf.o
 OBJS-$(CONFIG_NOISE_BSF)                  += noise_bsf.o
 OBJS-$(CONFIG_NULL_BSF)                   += null_bsf.o
-OBJS-$(CONFIG_OPUS_METADATA_BSF)          += opus_metadata_bsf.o
-OBJS-$(CONFIG_PCM_RECHUNK_BSF)            += pcm_rechunk_bsf.o
-OBJS-$(CONFIG_PRORES_METADATA_BSF)        += prores_metadata_bsf.o
 OBJS-$(CONFIG_REMOVE_EXTRADATA_BSF)       += remove_extradata_bsf.o
 OBJS-$(CONFIG_TEXT2MOVSUB_BSF)            += movsub_bsf.o
 OBJS-$(CONFIG_TRACE_HEADERS_BSF)          += trace_headers_bsf.o
-OBJS-$(CONFIG_TRUEHD_CORE_BSF)            += truehd_core_bsf.o mlp_parse.o mlp.o
 OBJS-$(CONFIG_VP9_METADATA_BSF)           += vp9_metadata_bsf.o
 OBJS-$(CONFIG_VP9_RAW_REORDER_BSF)        += vp9_raw_reorder_bsf.o
 OBJS-$(CONFIG_VP9_SUPERFRAME_BSF)         += vp9_superframe_bsf.o
@@ -1167,7 +1115,6 @@ SKIPHEADERS-$(CONFIG_JNI)              += ffjni.h
 SKIPHEADERS-$(CONFIG_LIBVPX)           += libvpx.h
 SKIPHEADERS-$(CONFIG_LIBWEBP_ENCODER)  += libwebpenc_common.h
 SKIPHEADERS-$(CONFIG_MEDIACODEC)       += mediacodecdec_common.h mediacodec_surface.h mediacodec_wrapper.h mediacodec_sw_buffer.h
-SKIPHEADERS-$(CONFIG_MEDIAFOUNDATION)  += mf_utils.h
 SKIPHEADERS-$(CONFIG_NVDEC)            += nvdec.h
 SKIPHEADERS-$(CONFIG_NVENC)            += nvenc.h
 SKIPHEADERS-$(CONFIG_QSV)              += qsv.h qsv_internal.h
@@ -1199,7 +1146,6 @@ TESTPROGS-$(CONFIG_IIRFILTER)             += iirfilter
 TESTPROGS-$(HAVE_MMX)                     += motion
 TESTPROGS-$(CONFIG_MPEGVIDEO)             += mpeg12framerate
 TESTPROGS-$(CONFIG_H264_METADATA_BSF)     += h264_levels
-TESTPROGS-$(CONFIG_HEVC_METADATA_BSF)     += h265_levels
 TESTPROGS-$(CONFIG_RANGECODER)            += rangecoder
 TESTPROGS-$(CONFIG_SNOW_ENCODER)          += snowenc

--- a/libavcodec/a64multienc.c
+++ b/libavcodec/a64multienc.c
@@ -60,11 +60,11 @@ typedef struct A64Context {
 } A64Context;

 /* gray gradient */
-static const uint8_t mc_colors[5]={0x0,0xb,0xc,0xf,0x1};
+static const int mc_colors[5]={0x0,0xb,0xc,0xf,0x1};

 /* other possible gradients - to be tested */
-//static const uint8_t mc_colors[5]={0x0,0x8,0xa,0xf,0x7};
-//static const uint8_t mc_colors[5]={0x0,0x9,0x8,0xa,0x3};
+//static const int mc_colors[5]={0x0,0x8,0xa,0xf,0x7};
+//static const int mc_colors[5]={0x0,0x9,0x8,0xa,0x3};

 static void to_meta_with_crop(AVCodecContext *avctx,
                              const AVFrame *p, int *dest)
--- a/libavcodec/aac.h
+++ b/libavcodec/aac.h
@@ -356,7 +356,7 @@ struct AACContext {
    OutputConfiguration oc[2];
    int warned_num_aac_frames;
    int warned_960_sbr;
-    unsigned warned_71_wide;
+
    int warned_gain_control;

    /* aacdec functions pointers */
@@ -368,7 +368,7 @@ struct AACContext {
                                   INTFLOAT *in, IndividualChannelStream *ics);
    void (*update_ltp)(AACContext *ac, SingleChannelElement *sce);
    void (*vector_pow43)(int *coefs, int len);
-    void (*subband_scale)(int *dst, int *src, int scale, int offset, int len, void *log_context);
+    void (*subband_scale)(int *dst, int *src, int scale, int offset, int len);

 };

--- a/libavcodec/aac_adtstoasc_bsf.c
+++ b/libavcodec/aac_adtstoasc_bsf.c
@@ -21,8 +21,8 @@

 #include "adts_header.h"
 #include "adts_parser.h"
+#include "avcodec.h"
 #include "bsf.h"
-#include "bsf_internal.h"
 #include "put_bits.h"
 #include "get_bits.h"
 #include "mpeg4audio.h"
@@ -134,8 +134,8 @@ static int aac_adtstoasc_init(AVBSFContext *ctx)
    /* Validate the extradata if the stream is already MPEG-4 AudioSpecificConfig */
    if (ctx->par_in->extradata) {
        MPEG4AudioConfig mp4ac;
-        int ret = avpriv_mpeg4audio_get_config2(&mp4ac, ctx->par_in->extradata,
-                                                ctx->par_in->extradata_size, 1, ctx);
+        int ret = avpriv_mpeg4audio_get_config(&mp4ac, ctx->par_in->extradata,
+                                               ctx->par_in->extradata_size * 8, 1);
        if (ret < 0) {
            av_log(ctx, AV_LOG_ERROR, "Error parsing AudioSpecificConfig extradata!\n");
            return ret;
--- a/libavcodec/aaccoder.c
+++ b/libavcodec/aaccoder.c
@@ -843,25 +843,25 @@ static void search_for_ms(AACEncContext *s, ChannelElement *cpe)
                                                    sce0->ics.swb_sizes[g],
                                                    sce0->sf_idx[w*16+g],
                                                    sce0->band_type[w*16+g],
-                                                    lambda / band0->threshold, INFINITY, &b1, NULL, 0);
+                                                    lambda / (band0->threshold + FLT_MIN), INFINITY, &b1, NULL, 0);
                        dist1 += quantize_band_cost(s, &sce1->coeffs[start + (w+w2)*128],
                                                    R34,
                                                    sce1->ics.swb_sizes[g],
                                                    sce1->sf_idx[w*16+g],
                                                    sce1->band_type[w*16+g],
-                                                    lambda / band1->threshold, INFINITY, &b2, NULL, 0);
+                                                    lambda / (band1->threshold + FLT_MIN), INFINITY, &b2, NULL, 0);
                        dist2 += quantize_band_cost(s, M,
                                                    M34,
                                                    sce0->ics.swb_sizes[g],
                                                    mididx,
                                                    midcb,
-                                                    lambda / minthr, INFINITY, &b3, NULL, 0);
+                                                    lambda / (minthr + FLT_MIN), INFINITY, &b3, NULL, 0);
                        dist2 += quantize_band_cost(s, S,
                                                    S34,
                                                    sce1->ics.swb_sizes[g],
                                                    sididx,
                                                    sidcb,
-                                                    mslambda / (minthr * bmax), INFINITY, &b4, NULL, 0);
+                                                    mslambda / (minthr * bmax + FLT_MIN), INFINITY, &b4, NULL, 0);
                        B0 += b1+b2;
                        B1 += b3+b4;
                        dist1 -= b1+b2;
--- a/libavcodec/aacdec.c
+++ b/libavcodec/aacdec.c
@@ -247,12 +247,14 @@ static void apply_independent_coupling(AACContext *ac,
                                       SingleChannelElement *target,
                                       ChannelElement *cce, int index)
 {
+    int i;
    const float gain = cce->coup.gain[index][0];
    const float *src = cce->ch[0].ret;
    float *dest = target->ret;
    const int len = 1024 << (ac->oc[1].m4ac.sbr == 1);

-    ac->fdsp->vector_fmac_scalar(dest, src, gain, len);
+    for (i = 0; i < len; i++)
+        dest[i] += gain * src[i];
 }

 #include "aacdec_template.c"
--- a/libavcodec/aacdec_fixed.c
+++ b/libavcodec/aacdec_fixed.c
@@ -162,7 +162,7 @@ static void vector_pow43(int *coefs, int len)
    }
 }

-static void subband_scale(int *dst, int *src, int scale, int offset, int len, void *log_context)
+static void subband_scale(int *dst, int *src, int scale, int offset, int len)
 {
    int ssign = scale < 0 ? -1 : 1;
    int s = FFABS(scale);
@@ -189,7 +189,7 @@ static void subband_scale(int *dst, int *src, int scale, int offset, int len, vo
            dst[i] = out * (unsigned)ssign;
        }
    } else {
-        av_log(log_context, AV_LOG_ERROR, "Overflow in subband_scale()\n");
+        av_log(NULL, AV_LOG_ERROR, "Overflow in subband_scale()\n");
    }
 }

@@ -461,7 +461,7 @@ AVCodec ff_aac_fixed_decoder = {
        AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_NONE
    },
    .capabilities    = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
-    .caps_internal   = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
+    .caps_internal   = FF_CODEC_CAP_INIT_THREADSAFE,
    .channel_layouts = aac_channel_layout,
    .profiles        = NULL_IF_CONFIG_SMALL(ff_aac_profiles),
    .flush = flush,
--- a/libavcodec/aacdec_template.c
+++ b/libavcodec/aacdec_template.c
@@ -520,7 +520,7 @@ static void flush(AVCodecContext *avctx)
 *
 * @return  Returns error status. 0 - OK, !0 - error
 */
-static int set_default_channel_config(AACContext *ac, AVCodecContext *avctx,
+static int set_default_channel_config(AVCodecContext *avctx,
                                      uint8_t (*layout_map)[3],
                                      int *tags,
                                      int channel_config)
@@ -547,7 +547,7 @@ static int set_default_channel_config(AACContext *ac, AVCodecContext *avctx,
     * As actual intended 7.1(wide) streams are very rare, default to assuming a
     * 7.1 layout was intended.
     */
-    if (channel_config == 7 && avctx->strict_std_compliance < FF_COMPLIANCE_STRICT && (!ac || !ac->warned_71_wide++)) {
+    if (channel_config == 7 && avctx->strict_std_compliance < FF_COMPLIANCE_STRICT) {
        av_log(avctx, AV_LOG_INFO, "Assuming an incorrectly encoded 7.1 channel layout"
               " instead of a spec-compliant 7.1(wide) layout, use -strict %d to decode"
               " according to the specification instead.\n", FF_COMPLIANCE_STRICT);
@@ -573,7 +573,7 @@ static ChannelElement *get_che(AACContext *ac, int type, int elem_id)

        av_log(ac->avctx, AV_LOG_DEBUG, "mono with CPE\n");

-        if (set_default_channel_config(ac, ac->avctx, layout_map,
+        if (set_default_channel_config(ac->avctx, layout_map,
                                       &layout_map_tags, 2) < 0)
            return NULL;
        if (output_configure(ac, layout_map, layout_map_tags,
@@ -592,7 +592,7 @@ static ChannelElement *get_che(AACContext *ac, int type, int elem_id)

        av_log(ac->avctx, AV_LOG_DEBUG, "stereo with SCE\n");

-        if (set_default_channel_config(ac, ac->avctx, layout_map,
+        if (set_default_channel_config(ac->avctx, layout_map,
                                       &layout_map_tags, 1) < 0)
            return NULL;
        if (output_configure(ac, layout_map, layout_map_tags,
@@ -841,7 +841,7 @@ static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
        if (tags < 0)
            return tags;
    } else {
-        if ((ret = set_default_channel_config(ac, avctx, layout_map,
+        if ((ret = set_default_channel_config(avctx, layout_map,
                                              &tags, channel_config)))
            return ret;
    }
@@ -937,7 +937,7 @@ static int decode_eld_specific_config(AACContext *ac, AVCodecContext *avctx,
        skip_bits_long(gb, 8 * len);
    }

-    if ((ret = set_default_channel_config(ac, avctx, layout_map,
+    if ((ret = set_default_channel_config(avctx, layout_map,
                                          &tags, channel_config)))
        return ret;

@@ -975,7 +975,7 @@ static int decode_audio_specific_config_gb(AACContext *ac,
    int i, ret;
    GetBitContext gbc = *gb;

-    if ((i = ff_mpeg4audio_get_config_gb(m4ac, &gbc, sync_extension, avctx)) < 0)
+    if ((i = ff_mpeg4audio_get_config_gb(m4ac, &gbc, sync_extension)) < 0)
        return AVERROR_INVALIDDATA;

    if (m4ac->sampling_index > 12) {
@@ -1200,7 +1200,7 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
        ac->oc[1].m4ac.chan_config = i;

        if (ac->oc[1].m4ac.chan_config) {
-            int ret = set_default_channel_config(ac, avctx, layout_map,
+            int ret = set_default_channel_config(avctx, layout_map,
                &layout_map_tags, ac->oc[1].m4ac.chan_config);
            if (!ret)
                output_configure(ac, layout_map, layout_map_tags,
@@ -1929,7 +1929,7 @@ static int decode_spectrum_and_dequant(AACContext *ac, INTFLOAT coef[1024],
            if (cbt_m1 < NOISE_BT - 1) {
                for (group = 0; group < (int)g_len; group++, cfo+=128) {
                    ac->vector_pow43(cfo, off_len);
-                    ac->subband_scale(cfo, cfo, sf[idx], 34, off_len, ac->avctx);
+                    ac->subband_scale(cfo, cfo, sf[idx], 34, off_len);
                }
            }
        }
@@ -2160,7 +2160,7 @@ static void apply_intensity_stereo(AACContext *ac,
                                      coef0 + group * 128 + offsets[i],
                                      scale,
                                      23,
-                                      offsets[i + 1] - offsets[i] ,ac->avctx);
+                                      offsets[i + 1] - offsets[i]);
 #else
                        ac->fdsp->vector_fmul_scalar(coef1 + group * 128 + offsets[i],
                                                    coef0 + group * 128 + offsets[i],
@@ -2807,7 +2807,7 @@ static void imdct_and_windowing_ld(AACContext *ac, SingleChannelElement *sce)

 static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce)
 {
-    INTFLOAT *in    = sce->coeffs;
+    UINTFLOAT *in   = sce->coeffs;
    INTFLOAT *out   = sce->ret;
    INTFLOAT *saved = sce->saved;
    INTFLOAT *buf  = ac->buf_mdct;
@@ -3002,7 +3002,7 @@ static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
        push_output_configuration(ac);
        if (hdr_info.chan_config) {
            ac->oc[1].m4ac.chan_config = hdr_info.chan_config;
-            if ((ret = set_default_channel_config(ac, ac->avctx,
+            if ((ret = set_default_channel_config(ac->avctx,
                                                  layout_map,
                                                  &layout_map_tags,
                                                  hdr_info.chan_config)) < 0)
@@ -3249,15 +3249,9 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
                    err = AVERROR_INVALIDDATA;
                    goto fail;
            }
-            err = 0;
-            while (elem_id > 0) {
-                int ret = decode_extension_payload(ac, gb, elem_id, che_prev, che_prev_type);
-                if (ret < 0) {
-                    err = ret;
-                    break;
-                }
-                elem_id -= ret;
-            }
+            while (elem_id > 0)
+                elem_id -= decode_extension_payload(ac, gb, elem_id, che_prev, che_prev_type);
+            err = 0; /* FIXME */
            break;

        default:
--- a/libavcodec/aacenc.c
+++ b/libavcodec/aacenc.c
@@ -28,6 +28,7 @@
 *              TODOs:
 * add sane pulse detection
 ***********************************/
+#include <float.h>

 #include "libavutil/libm.h"
 #include "libavutil/thread.h"
@@ -39,7 +40,6 @@
 #include "mpeg4audio.h"
 #include "kbdwin.h"
 #include "sinewin.h"
-#include "profiles.h"

 #include "aac.h"
 #include "aactab.h"
@@ -856,7 +856,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
                /* Not so fast though */
                ratio = sqrtf(ratio);
            }
-            s->lambda = FFMIN(s->lambda * ratio, 65536.f);
+            s->lambda = av_clipf(s->lambda * ratio, FLT_EPSILON, 65536.f);

            /* Keep iterating if we must reduce and lambda is in the sky */
            if (ratio > 0.9f && ratio < 1.1f) {
@@ -901,7 +901,7 @@ static av_cold int aac_encode_end(AVCodecContext *avctx)
 {
    AACEncContext *s = avctx->priv_data;

-    av_log(avctx, AV_LOG_INFO, "Qavg: %.3f\n", s->lambda_sum / s->lambda_count);
+    av_log(avctx, AV_LOG_INFO, "Qavg: %.3f\n", s->lambda_count ? s->lambda_sum / s->lambda_count : NAN);

    ff_mdct_end(&s->mdct1024);
    ff_mdct_end(&s->mdct128);
@@ -1132,7 +1132,6 @@ static const AVOption aacenc_options[] = {
    {"aac_ltp", "Long term prediction", offsetof(AACEncContext, options.ltp), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS},
    {"aac_pred", "AAC-Main prediction", offsetof(AACEncContext, options.pred), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS},
    {"aac_pce", "Forces the use of PCEs", offsetof(AACEncContext, options.pce), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS},
-    FF_AAC_PROFILE_OPTS
    {NULL}
 };

--- a/Show More
+++ b/Show More
@@ -1 +1 @@
 .3.2
 .1.11