Changelog: Update

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
avcodec/takdec: Check remaining space for first predictors
2025-12-10 00:40:01 +01:00 · 2025-05-16 01:38:40 +02:00 · 2025-05-16 01:35:55 +02:00 · 2025-05-16 01:34:47 +02:00 · 2025-05-16 01:33:17 +02:00 · 2025-05-16 01:18:31 +02:00
4748 changed files with 202189 additions and 394858 deletions
--- a/.gitignore
+++ b/.gitignore
@@ -19,12 +19,8 @@
 *.swp
 *.ver
 *.version
-*.metal.air
-*.metallib
-*.metallib.c
 *.ptx
 *.ptx.c
-*.ptx.gz
 *_g
 \#*
 .\#*
@@ -35,8 +31,8 @@
 /ffprobe
 /config.asm
 /config.h
-/config_components.h
 /coverage.info
+/avversion.h
 /lcov/
 /src
 /mapfile
--- a/.mailmap
+++ b/.mailmap
@@ -1,3 +1,4 @@
+<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>
--- a/1035
+++ b/1035
--- a/INSTALL.md
+++ b/INSTALL.md
@@ -15,3 +15,11 @@ NOTICE
 ------

 - Non system dependencies (e.g. libx264, libvpx) are disabled by default.
+
+NOTICE for Package Maintainers
+------------------------------
+
+ - It is recommended to build FFmpeg twice, first with minimal external dependencies so
+   that 3rd party packages, which depend on FFmpegs libavutil/libavfilter/libavcodec/libavformat
+   can then be built. And last build FFmpeg with full dependancies (which may in turn depend on
+   some of these 3rd party packages). This avoids circular dependencies during build.
--- a/49
+++ b/49
@@ -11,11 +11,17 @@ A (CC <address>) after the name means that the maintainer prefers to be CC-ed on
 patches and related discussions.


+Project Leader
+==============
+
+  final design decisions
+
+
 Applications
 ============

 ffmpeg:
-  ffmpeg.c                              Michael Niedermayer, Anton Khirnov
+  ffmpeg.c                              Michael Niedermayer

 ffplay:
  ffplay.c                              Marton Balint
@@ -34,8 +40,7 @@ Miscellaneous Areas
 ===================

 documentation                           Stefano Sabatini, Mike Melanson, Timothy Gu, Gyan Doshi
-project server day to day operations    Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser, Nikolay Aleksandrov
-project server emergencies              Árpád Gereöffy, Reimar Doeffinger, Alexander Strasser, Nikolay Aleksandrov
+project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser, Nikolay Aleksandrov
 presets                                 Robert Swain
 metadata subsystem                      Aurelien Jacobs
 release management                      Michael Niedermayer
@@ -110,6 +115,8 @@ Generic Parts:
    lzw.*                               Michael Niedermayer
  floating point AAN DCT:
    faandct.c, faandct.h                Michael Niedermayer
+  Non-power-of-two MDCT:
+    mdct15.c, mdct15.h                  Rostislav Pehlivanov
  Golomb coding:
    golomb.c, golomb.h                  Michael Niedermayer
  motion estimation:
@@ -131,10 +138,8 @@ Codecs:
  8bps.c                                Roberto Togni
  8svx.c                                Jaikrishnan Menon
  aacenc*, aaccoder.c                   Rostislav Pehlivanov
-  adpcm.c                               Zane van Iperen
  alacenc.c                             Jaikrishnan Menon
  alsdec.c                              Thilo Borgmann, Umair Khan
-  amfenc*                               Dmitrii Ovchinnikov
  aptx.c                                Aurelien Jacobs
  ass*                                  Aurelien Jacobs
  asv*                                  Michael Niedermayer
@@ -151,10 +156,10 @@ Codecs:
  ccaption_dec.c                        Anshul Maheshwari, Aman Gupta
  cljr                                  Alex Beregszaszi
  cpia.c                                Stephan Hilb
+  crystalhd.c                           Philip Langdale
  cscd.c                                Reimar Doeffinger
  cuviddec.c                            Timo Rothenpieler
  dca*                                  foo86
-  dfpwm*                                Jack Bruienne
  dirac*                                Rostislav Pehlivanov
  dnxhd*                                Baptiste Coudurier
  dolby_e*                              foo86
@@ -181,14 +186,12 @@ Codecs:
  interplayvideo.c                      Mike Melanson
  jni*, ffjni*                          Matthieu Bouron
  jpeg2000*                             Nicolas Bertrand
-  jpegxl*                               Leo Izen
  jvdec.c                               Peter Ross
  lcl*.c                                Roberto Togni, Reimar Doeffinger
  libcelt_dec.c                         Nicolas George
  libcodec2.c                           Tomas Härdin
  libdirac*                             David Conrad
  libdavs2.c                            Huiwen Ren
-  libjxl*.c, libjxl.h                   Leo Izen
  libgsm.c                              Michel Bardiaux
  libkvazaar.c                          Arttu Ylä-Outinen
  libopenh264enc.c                      Martin Storsjo, Linjie Fu
@@ -211,7 +214,6 @@ Codecs:
  mqc*                                  Nicolas Bertrand
  msmpeg4.c, msmpeg4data.h              Michael Niedermayer
  msrle.c                               Mike Melanson
-  msrleenc.c                            Tomas Härdin
  msvideo1.c                            Mike Melanson
  nuv.c                                 Reimar Doeffinger
  nvdec*, nvenc*                        Timo Rothenpieler
@@ -223,7 +225,7 @@ Codecs:
  ptx.c                                 Ivo van Poorten
  qcelp*                                Reynaldo H. Verdejo Pinochet
  qdm2.c, qdm2data.h                    Roberto Togni
-  qsv*                                  Mark Thompson, Zhong Li, Haihao Xiang
+  qsv*                                  Mark Thompson, Zhong Li
  qtrle.c                               Mike Melanson
  ra144.c, ra144.h, ra288.c, ra288.h    Roberto Togni
  resample2.c                           Michael Niedermayer
@@ -263,14 +265,16 @@ Codecs:
  xan.c                                 Mike Melanson
  xbm*                                  Paul B Mahol
  xface                                 Stefano Sabatini
+  xvmc.c                                Ivan Kalvachev
  xwd*                                  Paul B Mahol

 Hardware acceleration:
+  crystalhd.c                           Philip Langdale
  dxva2*                                Hendrik Leppkes, Laurent Aimar, Steve Lhomme
  d3d11va*                              Steve Lhomme
  mediacodec*                           Matthieu Bouron, Aman Gupta
-  vaapi*                                Haihao Xiang
-  vaapi_encode*                         Mark Thompson, Haihao Xiang
+  vaapi*                                Gwenole Beauchesne
+  vaapi_encode*                         Mark Thompson
  vdpau*                                Philip Langdale, Carl Eugen Hoyos
  videotoolbox*                         Rick Kern, Aman Gupta

@@ -349,7 +353,6 @@ Filters:
  vf_il.c                               Paul B Mahol
  vf_(t)interlace                       Thomas Mundt (CC <thomas.mundt@hr.de>)
  vf_lenscorrection.c                   Daniel Oberhoff
-  vf_libplacebo.c                       Niklas Haas
  vf_mergeplanes.c                      Paul B Mahol
  vf_mestimate.c                        Davinder Singh
  vf_minterpolate.c                     Davinder Singh
@@ -395,7 +398,6 @@ Muxers/Demuxers:
  apngdec.c                             Benoit Fouet
  argo_asf.c                            Zane van Iperen
  argo_brp.c                            Zane van Iperen
-  argo_cvg.c                            Zane van Iperen
  ass*                                  Aurelien Jacobs
  astdec.c                              Paul B Mahol
  astenc.c                              James Almer
@@ -412,14 +414,12 @@ Muxers/Demuxers:
  dashdec.c                             Steven Liu
  dashenc.c                             Karthick Jeyapal
  daud.c                                Reimar Doeffinger
-  dfpwmdec.c                            Jack Bruienne
  dss.c                                 Oleksij Rempel
  dtsdec.c                              foo86
  dtshddec.c                            Paul B Mahol
  dv.c                                  Roman Shaposhnik
  electronicarts.c                      Peter Ross
  epafdec.c                             Paul B Mahol
-  evc*                                  Samsung (Dawid Kozinski)
  ffm*                                  Baptiste Coudurier
  flic.c                                Mike Melanson
  flvdec.c                              Michael Niedermayer
@@ -430,12 +430,10 @@ Muxers/Demuxers:
  idcin.c                               Mike Melanson
  idroqdec.c                            Mike Melanson
  iff.c                                 Jaikrishnan Menon
-  imf*                                  Pierre-Anthony Lemieux
  img2*.c                               Michael Niedermayer
  ipmovie.c                             Mike Melanson
  ircam*                                Paul B Mahol
  iss.c                                 Stefan Gehrer
-  jpegxl*                               Leo Izen
  jvdec.c                               Peter Ross
  kvag.c                                Zane van Iperen
  libmodplug.c                          Clément Bœsch
@@ -515,7 +513,6 @@ Protocols:
  bluray.c                              Petri Hintukainen
  ftp.c                                 Lukasz Marek
  http.c                                Ronald S. Bultje
-  libsrt.c                              Zhao Zhili
  libssh.c                              Lukasz Marek
  libzmq.c                              Andriy Gelman
  mms*.c                                Ronald S. Bultje
@@ -542,11 +539,9 @@ Operating systems / CPU architectures

 Alpha                                   Falk Hueffner
 MIPS                                    Manojkumar Bhosale, Shiyou Yin
-LoongArch                               Shiyou Yin
 Mac OS X / PowerPC                      Romain Dolbeau, Guillaume Poirier
 Amiga / PowerPC                         Colin Ward
 Linux / PowerPC                         Lauri Kasanen
-RISC-V                                  Rémi Denis-Courmont
 Windows MinGW                           Alex Beregszaszi, Ramiro Polla
 Windows Cygwin                          Victor Paesa
 Windows MSVC                            Matthew Oliver, Hendrik Leppkes
@@ -588,10 +583,12 @@ wm4
 Releases
 ========

+7.0                                     Michael Niedermayer
+6.1                                     Michael Niedermayer
+5.1                                     Michael Niedermayer
+4.4                                     Michael Niedermayer
+3.4                                     Michael Niedermayer
 2.8                                     Michael Niedermayer
-2.7                                     Michael Niedermayer
-2.6                                     Michael Niedermayer
-2.5                                     Michael Niedermayer

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

@@ -614,22 +611,18 @@ Daniel Verkamp                78A6 07ED 782C 653E C628 B8B9 F0EB 8DD8 2F0E 21C7
 FFmpeg release signing key    FCF9 86EA 15E6 E293 A564 4F10 B432 2F04 D676 58D8
 Ganesh Ajjanagadde            C96A 848E 97C3 CEA2 AB72 5CE4 45F9 6A2D 3C36 FB1B
 Gwenole Beauchesne            2E63 B3A6 3E44 37E2 017D 2704 53C7 6266 B153 99C4
-Haihao Xiang (haihao)         1F0C 31E8 B4FE F7A4 4DC1 DC99 E0F5 76D4 76FC 437F
 Jaikrishnan Menon             61A1 F09F 01C9 2D45 78E1 C862 25DC 8831 AF70 D368
 James Almer                   7751 2E8C FD94 A169 57E6 9A7A 1463 01AD 7376 59E0
 Jean Delvare                  7CA6 9F44 60F1 BDC4 1FD2 C858 A552 6B9B B3CD 4E6A
-Leo Izen (Traneptora)         B6FD 3CFC 7ACF 83FC 9137 6945 5A71 C331 FD2F A19A
 Loren Merritt                 ABD9 08F4 C920 3F65 D8BE 35D7 1540 DAA7 060F 56DE
 Lynne                         FE50 139C 6805 72CA FD52 1F8D A2FE A5F0 3F03 4464
 Michael Niedermayer           9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
                              DD1E C9E8 DE08 5C62 9B3E 1846 B18E 8928 B394 8D64
 Nicolas George                24CE 01CE 9ACC 5CEB 74D8 8D9D B063 D997 36E5 4C93
-Niklas Haas (haasn)           1DDB 8076 B14D 5B48 32FC 99D9 EB52 DA9C 02BA 6FB4
 Nikolay Aleksandrov           8978 1D8C FB71 588E 4B27 EAA8 C4F0 B5FC E011 13B1
 Panagiotis Issaris            6571 13A3 33D9 3726 F728 AA98 F643 B12E ECF3 E029
 Peter Ross                    A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B
 Philip Langdale               5DC5 8D66 5FBA 3A43 18EC 045E F8D6 B194 6A75 682E
-Pierre-Anthony Lemieux (pal)  F4B3 9492 E6F2 E4AF AEC8 46CB 698F A1F0 F8D4 EED4
 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
--- a/25
+++ b/25
@@ -13,19 +13,17 @@ vpath %.v    $(SRC_PATH)
 vpath %.texi $(SRC_PATH)
 vpath %.cu   $(SRC_PATH)
 vpath %.ptx  $(SRC_PATH)
-vpath %.metal $(SRC_PATH)
 vpath %/fate_config.sh.template $(SRC_PATH)

 TESTTOOLS   = audiogen videogen rotozoom tiny_psnr tiny_ssim base64 audiomatch
 HOSTPROGS  := $(TESTTOOLS:%=tests/%) doc/print_options

-ALLFFLIBS = avcodec avdevice avfilter avformat avutil postproc swscale swresample
-
 # $(FFLIBS-yes) needs to be in linking order
 FFLIBS-$(CONFIG_AVDEVICE)   += avdevice
 FFLIBS-$(CONFIG_AVFILTER)   += avfilter
 FFLIBS-$(CONFIG_AVFORMAT)   += avformat
 FFLIBS-$(CONFIG_AVCODEC)    += avcodec
+FFLIBS-$(CONFIG_AVRESAMPLE) += avresample
 FFLIBS-$(CONFIG_POSTPROC)   += postproc
 FFLIBS-$(CONFIG_SWRESAMPLE) += swresample
 FFLIBS-$(CONFIG_SWSCALE)    += swscale
@@ -47,7 +45,7 @@ FF_DEP_LIBS  := $(DEP_LIBS)
 FF_STATIC_DEP_LIBS := $(STATIC_DEP_LIBS)

 $(TOOLS): %$(EXESUF): %.o
-	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $(filter-out $(FF_DEP_LIBS), $^) $(EXTRALIBS-$(*F)) $(EXTRALIBS) $(ELIBS)
+	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(EXTRALIBS-$(*F)) $(EXTRALIBS) $(ELIBS)

 target_dec_%_fuzzer$(EXESUF): target_dec_%_fuzzer.o $(FF_DEP_LIBS)
 	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
@@ -67,10 +65,6 @@ tools/target_io_dem_fuzzer$(EXESUF): tools/target_io_dem_fuzzer.o $(FF_DEP_LIBS)

 tools/enum_options$(EXESUF): ELIBS = $(FF_EXTRALIBS)
 tools/enum_options$(EXESUF): $(FF_DEP_LIBS)
-tools/enc_recon_frame_test$(EXESUF): $(FF_DEP_LIBS)
-tools/enc_recon_frame_test$(EXESUF): ELIBS = $(FF_EXTRALIBS)
-tools/scale_slice_test$(EXESUF): $(FF_DEP_LIBS)
-tools/scale_slice_test$(EXESUF): ELIBS = $(FF_EXTRALIBS)
 tools/sofa2wavs$(EXESUF): ELIBS = $(FF_EXTRALIBS)
 tools/uncoded_frame$(EXESUF): $(FF_DEP_LIBS)
 tools/uncoded_frame$(EXESUF): ELIBS = $(FF_EXTRALIBS)
@@ -80,23 +74,21 @@ tools/target_dem_%_fuzzer$(EXESUF): $(FF_DEP_LIBS)
 CONFIGURABLE_COMPONENTS =                                           \
    $(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c))                 \
    $(SRC_PATH)/libavcodec/bitstream_filters.c                      \
-    $(SRC_PATH)/libavcodec/hwaccels.h                               \
    $(SRC_PATH)/libavcodec/parsers.c                                \
    $(SRC_PATH)/libavformat/protocols.c                             \

-config_components.h: ffbuild/.config
+config.h: ffbuild/.config
 ffbuild/.config: $(CONFIGURABLE_COMPONENTS)
 	@-tput bold 2>/dev/null
-	@-printf '\nWARNING: $(?) newer than config_components.h, rerun configure\n\n'
+	@-printf '\nWARNING: $(?) newer than config.h, rerun configure\n\n'
 	@-tput sgr0 2>/dev/null

 SUBDIR_VARS := CLEANFILES FFLIBS HOSTPROGS TESTPROGS TOOLS               \
               HEADERS ARCH_HEADERS BUILT_HEADERS SKIPHEADERS            \
               ARMV5TE-OBJS ARMV6-OBJS ARMV8-OBJS VFP-OBJS NEON-OBJS     \
-               ALTIVEC-OBJS VSX-OBJS RVV-OBJS MMX-OBJS X86ASM-OBJS       \
+               ALTIVEC-OBJS VSX-OBJS MMX-OBJS X86ASM-OBJS                \
               MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSP-OBJS MSA-OBJS         \
-               MMI-OBJS LSX-OBJS LASX-OBJS OBJS SLIBOBJS SHLIBOBJS       \
-               STLIBOBJS HOSTOBJS TESTOBJS
+               MMI-OBJS OBJS SLIBOBJS HOSTOBJS TESTOBJS

 define RESET
 $(1) :=
@@ -118,13 +110,12 @@ include $(SRC_PATH)/fftools/Makefile
 include $(SRC_PATH)/doc/Makefile
 include $(SRC_PATH)/doc/examples/Makefile

-$(ALLFFLIBS:%=lib%/version.o): libavutil/ffversion.h
+libavcodec/avcodec.o libavformat/utils.o libavdevice/avdevice.o libavfilter/avfilter.o libavutil/utils.o libpostproc/postprocess.o libswresample/swresample.o libswscale/utils.o : libavutil/ffversion.h

 $(PROGS): %$(PROGSSUF)$(EXESUF): %$(PROGSSUF)_g$(EXESUF)
 ifeq ($(STRIPTYPE),direct)
 	$(STRIP) -o $@ $<
 else
-	$(RM) $@
 	$(CP) $< $@
 	$(STRIP) $@
 endif
@@ -165,7 +156,7 @@ clean::
 	$(RM) -rf coverage.info coverage.info.in lcov

 distclean:: clean
-	$(RM) .version config.asm config.h config_components.h mapfile  \
+	$(RM) .version avversion.h config.asm config.h mapfile  \
 		ffbuild/.config ffbuild/config.* libavutil/avconfig.h \
 		version.h libavutil/ffversion.h libavcodec/codec_names.h \
 		libavcodec/bsf_list.c libavformat/protocol_list.c \
--- a/README.md
+++ b/README.md
@@ -9,7 +9,7 @@ such as audio, video, subtitles and related metadata.
 * `libavcodec` provides implementation of a wider range of codecs.
 * `libavformat` implements streaming protocols, container formats and basic I/O access.
 * `libavutil` includes hashers, decompressors and miscellaneous utility functions.
-* `libavfilter` provides means to alter decoded audio and video through a directed graph of connected filters.
+* `libavfilter` provides a mean to alter decoded Audio and Video through chain of filters.
 * `libavdevice` provides an abstraction to access capture and playback devices.
 * `libswresample` implements audio mixing and resampling routines.
 * `libswscale` implements color conversion and scaling routines.
--- a/2
+++ b/2
@@ -1 +1 @@
-5.1.git
+4.4.6
--- a/15
+++ b/15
@@ -0,0 +1,15 @@
+
+              ┌────────────────────────────────────┐
+              │ RELEASE NOTES for FFmpeg 4.4 "Rao" │
+              └────────────────────────────────────┘
+
+   The FFmpeg Project proudly presents FFmpeg 4.4 "Rao", about 10
+   months after the release of FFmpeg 4.3.
+
+   A complete Changelog is available at the root of the project, and the
+   complete Git history on https://git.ffmpeg.org/gitweb/ffmpeg.git
+
+   We hope you will like this release as much as we enjoyed working on it, and
+   as usual, if you have any questions about it, or any FFmpeg related topic,
+   feel free to join us on the #ffmpeg IRC channel (on irc.libera.chat) or ask
+   on the mailing-lists.
--- a/compat/atomics/win32/stdatomic.h
+++ b/compat/atomics/win32/stdatomic.h
@@ -19,6 +19,7 @@
 #ifndef COMPAT_ATOMICS_WIN32_STDATOMIC_H
 #define COMPAT_ATOMICS_WIN32_STDATOMIC_H

+#define WIN32_LEAN_AND_MEAN
 #include <stddef.h>
 #include <stdint.h>
 #include <windows.h>
@@ -95,7 +96,7 @@ do {                                    \
    atomic_load(object)

 #define atomic_exchange(object, desired) \
-    InterlockedExchangePointer((PVOID volatile *)object, (PVOID)desired)
+    InterlockedExchangePointer(object, desired);

 #define atomic_exchange_explicit(object, desired, order) \
    atomic_exchange(object, desired)
--- a/compat/cuda/cuda_runtime.h
+++ b/compat/cuda/cuda_runtime.h
@@ -181,12 +181,8 @@ static inline __device__ double trunc(double a) { return __builtin_trunc(a); }
 static inline __device__ float fabsf(float a) { return __builtin_fabsf(a); }
 static inline __device__ float fabs(float a) { return __builtin_fabsf(a); }
 static inline __device__ double fabs(double a) { return __builtin_fabs(a); }
-static inline __device__ float sqrtf(float a) { return __builtin_sqrtf(a); }

-static inline __device__ float __saturatef(float a) { return __nvvm_saturate_f(a); }
 static inline __device__ float __sinf(float a) { return __nvvm_sin_approx_f(a); }
 static inline __device__ float __cosf(float a) { return __nvvm_cos_approx_f(a); }
-static inline __device__ float __expf(float a) { return __nvvm_ex2_approx_f(a * (float)__builtin_log2(__builtin_exp(1))); }
-static inline __device__ float __powf(float a, float b) { return __nvvm_ex2_approx_f(__nvvm_lg2_approx_f(a) * b); }

 #endif /* COMPAT_CUDA_CUDA_RUNTIME_H */
--- a/compat/cuda/ptx2c.sh
+++ b/compat/cuda/ptx2c.sh
@@ -0,0 +1,34 @@
+#!/bin/sh
+
+# Copyright (c) 2017, NVIDIA CORPORATION. All rights reserved.
+#
+# Permission is hereby granted, free of charge, to any person obtaining a
+# copy of this software and associated documentation files (the "Software"),
+# to deal in the Software without restriction, including without limitation
+# the rights to use, copy, modify, merge, publish, distribute, sublicense,
+# and/or sell copies of the Software, and to permit persons to whom the
+# Software is furnished to do so, subject to the following conditions:
+#
+# The above copyright notice and this permission notice shall be included in
+# all copies or substantial portions of the Software.
+#
+# THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+# IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+# FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.  IN NO EVENT SHALL
+# THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+# LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
+# FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
+# DEALINGS IN THE SOFTWARE.
+
+set -e
+
+OUT="$1"
+IN="$2"
+NAME="$(basename "$IN" | sed 's/\..*//')"
+
+printf "const char %s_ptx[] = \\" "$NAME" > "$OUT"
+echo >> "$OUT"
+sed -e "$(printf 's/\r//g')" -e 's/["\\]/\\&/g' -e "$(printf 's/^/\t"/')" -e 's/$/\\n"/' < "$IN" >> "$OUT"
+echo ";" >> "$OUT"
+
+exit 0
--- a/compat/msvcrt/snprintf.c
+++ b/compat/msvcrt/snprintf.c
@@ -59,7 +59,7 @@ int avpriv_vsnprintf(char *s, size_t n, const char *fmt,
     * recommends to provide _snprintf/_vsnprintf() a buffer size that
     * is one less than the actual buffer, and zero it before calling
     * _snprintf/_vsnprintf() to workaround this problem.
-     * See https://web.archive.org/web/20151214111935/http://msdn.microsoft.com/en-us/library/1kt27hek(v=vs.80).aspx */
+     * See http://msdn.microsoft.com/en-us/library/1kt27hek(v=vs.80).aspx */
    memset(s, 0, n);
    va_copy(ap_copy, ap);
    ret = _vsnprintf(s, n - 1, fmt, ap_copy);
--- a/compat/w32dlfcn.h
+++ b/compat/w32dlfcn.h
@@ -20,40 +20,11 @@
 #define COMPAT_W32DLFCN_H

 #ifdef _WIN32
-#include <stdint.h>
-
 #include <windows.h>
-
 #include "config.h"
-#include "libavutil/macros.h"
+#if (_WIN32_WINNT < 0x0602) || HAVE_WINRT
 #include "libavutil/wchar_filename.h"
-
-static inline wchar_t *get_module_filename(HMODULE module)
-{
-    wchar_t *path = NULL, *new_path;
-    DWORD path_size = 0, path_len;
-
-    do {
-        path_size = path_size ? FFMIN(2 * path_size, INT16_MAX + 1) : MAX_PATH;
-        new_path = av_realloc_array(path, path_size, sizeof *path);
-        if (!new_path) {
-            av_free(path);
-            return NULL;
-        }
-        path = new_path;
-        // Returns path_size in case of insufficient buffer.
-        // Whether the error is set or not and whether the output
-        // is null-terminated or not depends on the version of Windows.
-        path_len = GetModuleFileNameW(module, path, path_size);
-    } while (path_len && path_size <= INT16_MAX && path_size <= path_len);
-
-    if (!path_len) {
-        av_free(path);
-        return NULL;
-    }
-    return path;
-}
-
+#endif
 /**
 * Safe function used to open dynamic libs. This attempts to improve program security
 * by removing the current directory from the dll search path. Only dll's found in the
@@ -63,53 +34,29 @@ static inline wchar_t *get_module_filename(HMODULE module)
 */
 static inline HMODULE win32_dlopen(const char *name)
 {
-    wchar_t *name_w;
-    HMODULE module = NULL;
-    if (utf8towchar(name, &name_w))
-        name_w = NULL;
 #if _WIN32_WINNT < 0x0602
-    // On Win7 and earlier we check if KB2533623 is available
+    // Need to check if KB2533623 is available
    if (!GetProcAddress(GetModuleHandleW(L"kernel32.dll"), "SetDefaultDllDirectories")) {
-        wchar_t *path = NULL, *new_path;
-        DWORD pathlen, pathsize, namelen;
-        if (!name_w)
+        HMODULE module = NULL;
+        wchar_t *path = NULL, *name_w = NULL;
+        DWORD pathlen;
+        if (utf8towchar(name, &name_w))
            goto exit;
-        namelen = wcslen(name_w);
+        path = (wchar_t *)av_mallocz_array(MAX_PATH, sizeof(wchar_t));
        // Try local directory first
-        path = get_module_filename(NULL);
-        if (!path)
+        pathlen = GetModuleFileNameW(NULL, path, MAX_PATH);
+        pathlen = wcsrchr(path, '\\') - path;
+        if (pathlen == 0 || pathlen + wcslen(name_w) + 2 > MAX_PATH)
            goto exit;
-        new_path = wcsrchr(path, '\\');
-        if (!new_path)
-            goto exit;
-        pathlen = new_path - path;
-        pathsize = pathlen + namelen + 2;
-        new_path = av_realloc_array(path, pathsize, sizeof *path);
-        if (!new_path)
-            goto exit;
-        path = new_path;
+        path[pathlen] = '\\';
        wcscpy(path + pathlen + 1, name_w);
        module = LoadLibraryExW(path, NULL, LOAD_WITH_ALTERED_SEARCH_PATH);
        if (module == NULL) {
            // Next try System32 directory
-            pathlen = GetSystemDirectoryW(path, pathsize);
-            if (!pathlen)
+            pathlen = GetSystemDirectoryW(path, MAX_PATH);
+            if (pathlen == 0 || pathlen + wcslen(name_w) + 2 > MAX_PATH)
                goto exit;
-            // Buffer is not enough in two cases:
-            // 1. system directory + \ + module name
-            // 2. system directory even without the module name.
-            if (pathlen + namelen + 2 > pathsize) {
-                pathsize = pathlen + namelen + 2;
-                new_path = av_realloc_array(path, pathsize, sizeof *path);
-                if (!new_path)
-                    goto exit;
-                path = new_path;
-                // Query again to handle the case #2.
-                pathlen = GetSystemDirectoryW(path, pathsize);
-                if (!pathlen)
-                    goto exit;
-            }
-            path[pathlen] = L'\\';
+            path[pathlen] = '\\';
            wcscpy(path + pathlen + 1, name_w);
            module = LoadLibraryExW(path, NULL, LOAD_WITH_ALTERED_SEARCH_PATH);
        }
@@ -126,19 +73,16 @@ exit:
 #   define LOAD_LIBRARY_SEARCH_SYSTEM32        0x00000800
 #endif
 #if HAVE_WINRT
-    if (!name_w)
+    wchar_t *name_w = NULL;
+    int ret;
+    if (utf8towchar(name, &name_w))
        return NULL;
-    module = LoadPackagedLibrary(name_w, 0);
-#else
-#define LOAD_FLAGS (LOAD_LIBRARY_SEARCH_APPLICATION_DIR | LOAD_LIBRARY_SEARCH_SYSTEM32)
-    /* filename may be be in CP_ACP */
-    if (!name_w)
-        return LoadLibraryExA(name, NULL, LOAD_FLAGS);
-    module = LoadLibraryExW(name_w, NULL, LOAD_FLAGS);
-#undef LOAD_FLAGS
-#endif
+    ret = LoadPackagedLibrary(name_w, 0);
    av_free(name_w);
-    return module;
+    return ret;
+#else
+    return LoadLibraryExA(name, NULL, LOAD_LIBRARY_SEARCH_APPLICATION_DIR | LOAD_LIBRARY_SEARCH_SYSTEM32);
+#endif
 }
 #define dlopen(name, flags) win32_dlopen(name)
 #define dlclose FreeLibrary
--- a/compat/w32pthreads.h
+++ b/compat/w32pthreads.h
@@ -35,6 +35,7 @@
 * As most functions here are used without checking return values,
 * only implement return values as necessary. */

+#define WIN32_LEAN_AND_MEAN
 #include <windows.h>
 #include <process.h>
 #include <time.h>
@@ -65,14 +66,7 @@ typedef CONDITION_VARIABLE pthread_cond_t;
 #define PTHREAD_CANCEL_ENABLE 1
 #define PTHREAD_CANCEL_DISABLE 0

-#if HAVE_WINRT
-#define THREADFUNC_RETTYPE DWORD
-#else
-#define THREADFUNC_RETTYPE unsigned
-#endif
-
-static av_unused THREADFUNC_RETTYPE
-__stdcall attribute_align_arg win32thread_worker(void *arg)
+static av_unused unsigned __stdcall attribute_align_arg win32thread_worker(void *arg)
 {
    pthread_t *h = (pthread_t*)arg;
    h->ret = h->func(h->arg);
--- a/compat/windows/mswindres
+++ b/compat/windows/mswindres
@@ -1,32 +0,0 @@
-#!/bin/sh
-
-if [ "$1" = "--version" ]; then
-    rc.exe -?
-    exit $?
-fi
-
-if [ $# -lt 2 ]; then
-    echo "Usage: mswindres [-I/include/path ...] [-DSOME_DEFINE ...] [-o output.o] input.rc [output.o]" >&2
-    exit 0
-fi
-
-EXTRA_OPTS="-nologo"
-
-while [ $# -gt 2 ]; do
-    case $1 in
-    -D*) EXTRA_OPTS="$EXTRA_OPTS -d$(echo $1 | sed -e "s/^..//" -e "s/ /\\\\ /g")" ;;
-    -I*) EXTRA_OPTS="$EXTRA_OPTS -i$(echo $1 | sed -e "s/^..//" -e "s/ /\\\\ /g")" ;;
-    -o)  OPT_OUT="$2"; shift ;;
-    esac
-    shift
-done
-
-IN="$1"
-if [ -z "$OPT_OUT" ]; then
-    OUT="$2"
-else
-    OUT="$OPT_OUT"
-fi
-
-eval set -- $EXTRA_OPTS
-rc.exe "$@" -fo "$OUT" "$IN"
--- a/1292
+++ b/1292
--- a/doc/APIchanges
+++ b/doc/APIchanges
@@ -1,592 +1,20 @@
-The last version increases of all libraries were on 2023-02-09
+Never assume the API of libav* to be stable unless at least 1 month has passed
+since the last major version increase or the API was added.
+
+The last version increases were:
+libavcodec:    2017-10-21
+libavdevice:   2017-10-21
+libavfilter:   2017-10-21
+libavformat:   2017-10-21
+libavresample: 2017-10-21
+libpostproc:   2017-10-21
+libswresample: 2017-10-21
+libswscale:    2017-10-21
+libavutil:     2017-10-21
+

 API changes, most recent first:

-------- 8< --------- FFmpeg 6.1 was cut here -------- 8< ---------
-
-2023-10-27 - xxxxxxxxxx - lavu 58.28.100 - channel_layout.h
-  Add AV_CH_LAYOUT_3POINT1POINT2 and AV_CHANNEL_LAYOUT_3POINT1POINT2.
-  Add AV_CH_LAYOUT_5POINT1POINT2_BACK and AV_CHANNEL_LAYOUT_5POINT1POINT2_BACK.
-  Add AV_CH_LAYOUT_5POINT1POINT4_BACK and AV_CHANNEL_LAYOUT_5POINT1POINT4_BACK.
-  Add AV_CH_LAYOUT_7POINT1POINT2 and AV_CHANNEL_LAYOUT_7POINT1POINT2.
-  Add AV_CH_LAYOUT_7POINT1POINT4_BACK and AV_CHANNEL_LAYOUT_7POINT1POINT4_BACK.
-
-2023-10-06 - xxxxxxxxxx - lavc 60.30.101 - avcodec.h
-  AVCodecContext.coded_side_data may now be used during decoding, to be set
-  by user before calling avcodec_open2() for initialization.
-
-2023-10-06 - xxxxxxxxxx - lavc 60.15.100 - avformat.h
-  Deprecate AVFormatContext.{nb_,}side_data, av_stream_add_side_data(),
-  av_stream_new_side_data(), and av_stream_get_side_data(). Side data fields
-  from AVFormatContext.codecpar should be used from now on.
-
-2023-10-06 - xxxxxxxxxx - lavc 60.30.100 - codec_par.h
-  Added {nb_,}coded_side_data to AVCodecParameters.
-  The AVCodecParameters helpers will copy it to and from its AVCodecContext
-  namesake.
-
-2023-10-06 - xxxxxxxxxx - lavc 60.29.100 - packet.h
-  Added av_packet_side_data_new(), av_packet_side_data_add(),
-  av_packet_side_data_get(), av_packet_side_data_remove, and
-  av_packet_side_data_free().
-
-2023-10-03 - xxxxxxxxxx - lavc 60.28.100 - codec_par.h defs.h
-  Move the definition of enum AVFieldOrder from codec_par.h to defs.h.
-
-2023-10-03 - xxxxxxxxxx - lavf 60.14.100 - avformat.h
-  Deprecate AVFMT_ALLOW_FLUSH without replacement. Users can always
-  flush any muxer by sending a NULL packet.
-
-2023-09-28 - xxxxxxxxxx - lavu 58.27.100 - pixfmt.h
-  Add AV_PIX_FMT_GBRAP14BE, AV_PIX_FMT_GBRAP14LE pixel formats.
-
-2023-09-28 - xxxxxxxxxx - lavu 58.26.100 - hwcontext_cuda.h
-  Add AV_CUDA_USE_CURRENT_CONTEXT.
-
-2023-09-19 - xxxxxxxxxx - lavu 58.25.100 - avutil.h
-  Make AV_TIME_BASE_Q compatible with C++.
-
-2023-09-xx - xxxxxxxxxx - lavf 60 - avformat.h
-  Deprecate AVFMT_FLAG_SHORTEST without replacement.
-
-2023-09-07 - xxxxxxxxxx - lavu 58.24.100 - imgutils.h
-  Add av_image_copy2(), a wrapper around the av_image_copy()
-  to overcome limitations of automatic conversions.
-
-2023-09-07 - xxxxxxxxxx - lavu 58.23.100 - fifo.h
-  Constify the AVFifo pointees in av_fifo_peek() and av_fifo_peek_to_cb().
-
-2023-09-07 - xxxxxxxxxx - lavu 58.22.100 - audio_fifo.h
-  Constify some pointees in av_audio_fifo_write(), av_audio_fifo_read(),
-  av_audio_fifo_peek() and av_audio_fifo_peek_at().
-
-2023-09-07 - xxxxxxxxxx - lavu 58.21.100 - samplefmt.h
-  Constify some pointees in av_samples_copy() and av_samples_set_silence().
-
-2023-09-07 - xxxxxxxxxx - lavu 58.20.100 - imgutils.h
-  Constify some pointees in av_image_copy(), av_image_copy_uc_from() and
-  av_image_fill_black().
-
-2023-09-07 - xxxxxxxxxx - lavf 60.12.100 - avio.h
-  Constify the buffer pointees in the write_packet and write_data_type
-  callbacks of AVIOContext on the next major bump.
-
-2023-09-07 - xxxxxxxxxx - lavc 60.26.100 - defs.h
-  Add AV_PROFILE_* and AV_LEVEL_* replacements in defs.h for the
-  defines from avcodec.h. The latter are deprecated.
-
-2023-09-06 - xxxxxxxxxx - lavc 60.25.101 - avcodec.h
-  AVCodecContext.rc_buffer_size may now be set by decoders.
-
-2023-09-02 - xxxxxxxxxx - lavu 58.19.100 - executor.h
-  Add AVExecutor API
-
-2023-09-xx - xxxxxxxxxx - lavc 60.25.100 - avfft.h
-  The entire header will be deprecated and removed in two major bumps.
-  For a replacement to av_dct, av_rdft, av_fft and av_mdct, use
-  the new API from libavutil/tx.h.
-
-2023-07-xx - xxxxxxxxxx - lavu 58.18.100 - tx.h
-  Add AV_TX_REAL_TO_REAL and AV_TX_REAL_TO_IMAGINARY
-
-2023-08-18 - xxxxxxxxxx - lavu 58.17.100 - channel_layout.h
-  All AV_CHANNEL_LAYOUT_* macros are now compatible with C++ 17 and older.
-
-2023-08-08 - xxxxxxxxxx - lavu 58.15.100 - video_hint.h
-  Add AVVideoHint API.
-
-2023-07-xx - xxxxxxxxxx - lavc 60 - avcodec.h
-  Deprecate AV_CODEC_FLAG_DROPCHANGED without replacement.
-
-2023-07-05 - xxxxxxxxxx - lavu 58.14.100 - random_seed.h
-  Add av_random_bytes()
-
-2023-05-29 - xxxxxxxxxx - lavc 60.16.100 - avcodec.h codec_id.h
-  Add AV_CODEC_ID_EVC, FF_PROFILE_EVC_BASELINE, and FF_PROFILE_EVC_MAIN.
-
-2023-05-29 - xxxxxxxxxx - lavu 58.12.100 - mathematics.h
-  Add av_bessel_i0()
-
-2023-05-xx - xxxxxxxxxx - lavc 60.15.100 - avcodec.h
-  Add AVHWAccel.update_thread_context, AVHWAccel.free_frame_priv,
-  AVHWAccel.flush.
-
-2023-05-xx - xxxxxxxxxx - lavu 58.11.100 - hwcontext_vulkan.h
-  Add AVVulkanDeviceContext.lock_queue, AVVulkanDeviceContext.unlock_queue,
-  AVVulkanFramesContext.format, AVVulkanFramesContext.lock_frame,
-  AVVulkanFramesContext.unlock_frame, AVVkFrame.queue_family.
-  Deprecate AV_VK_FRAME_FLAG_CONTIGUOUS_MEMORY (use multiplane images instead).
-
-2023-05-xx - xxxxxxxxxx - lavu 58.10.100 - pixfmt.h
-  Add AV_PIX_FMT_P212BE, AV_PIX_FMT_P212LE, AV_PIX_FMT_P412BE,
-  AV_PIX_FMT_P412LE.
-
-2023-05-xx - xxxxxxxxxx - lavu 58.8.100 - frame.h
-  Add av_frame_replace().
-
-2023-05-xx - xxxxxxxxxx - lavu 58 - frame.h
-  Deprecate AVFrame.palette_has_changed without replacement.
-
-2023-05-xx - xxxxxxxxxx - lavc 60 - avcodec.h
-  Depreate AVCodecContext.ticks_per_frame in favor of
-  AVCodecContext.framerate (encoding) and
-  AV_CODEC_PROP_FIELDS (decoding).
-
-2023-05-xx - xxxxxxxxxx - lavc 60.12.100 - codec_desc.h
-  Add AV_CODEC_PROP_FIELDS.
-
-2023-05-xx - xxxxxxxxxx - lavc 60 - codec.h
-  Depreate AV_CODEC_CAP_SUBFRAMES without replacement.
-
-2023-05-xx - xxxxxxxxxx - lavc 60.11.100 - codec_par.h
-  Add AVCodecParameters.framerate.
-
-2023-05-04 - xxxxxxxxxx - lavu 58.7.100 - frame.h
-  Deprecate AVFrame.interlaced_frame, AVFrame.top_field_first, and
-  AVFrame.key_frame.
-  Add AV_FRAME_FLAG_INTERLACED, AV_FRAME_FLAG_TOP_FIELD_FIRST, and
-  AV_FRAME_FLAG_KEY flags as replacement.
-
-2023-04-10 - xxxxxxxxxx - lavu 58.6.100 - frame.h
-  av_frame_get_plane_buffer() now accepts const AVFrame*.
-
-2023-04-04 - xxxxxxxxxx - lavu 58.6.100 - hdr_dynamic_metadata.h
-  Add AV_HDR_PLUS_MAX_PAYLOAD_SIZE.
-  av_dynamic_hdr_plus_create_side_data() now accepts a user provided
-  buffer.
-
-2023-03-xx - xxxxxxxxxx - lavfi 9.5.100 - avfilter.h
-  Add AVFILTER_FLAG_HWDEVICE.
-
-2023-03-21 - xxxxxxxxxx - lavu 58.5.100 - hdr_dynamic_metadata.h
-  Add av_dynamic_hdr_plus_from_t35() and av_dynamic_hdr_plus_to_t35()
-  functions to convert between raw T.35 payloads containing dynamic
-  HDR10+ metadata and their parsed representations as AVDynamicHDRPlus.
-
-2023-03-17 - xxxxxxxxxx - lavu 58.4.100 - hdr_dynamic_vivid_metadata.h
-  Add two group of three spline params.
-  Deprecate previous define which only supports one group of params.
-
-2023-03-02 - xxxxxxxxxx - lavc 60.6.100 - avcodec.h
-  Add FF_PROFILE_EAC3_DDP_ATMOS, FF_PROFILE_TRUEHD_ATMOS,
-  FF_PROFILE_DTS_HD_MA_X and FF_PROFILE_DTS_HD_MA_X_IMAX.
-
-2023-02-25 - xxxxxxxxxx - lavc 60.5.100 - avcodec.h
-  Add FF_PROFILE_HEVC_SCC.
-
-------- 8< --------- FFmpeg 6.0 was cut here -------- 8< ---------
-
-2023-02-16 - 927042b409 - lavf 60.2.100 - avformat.h
-  Deprecate AVFormatContext io_close callback.
-  The superior io_close2 callback should be used instead.
-
-2023-02-13 - 2296078397 - lavu 58.1.100 - frame.h
-  Deprecate AVFrame.coded_picture_number and display_picture_number.
-  Their usefulness is questionable and very few decoders set them.
-
-2023-02-13 - 6b6f7db819 - lavc 60.2.100 - avcodec.h
-  Add AVCodecContext.frame_num as a 64bit version of frame_number.
-  Deprecate AVCodecContext.frame_number.
-
-2023-02-12 - d1b9a3ddb4 - lavfi 9.1.100 - avfilter.h
-  Add filtergraph segment parsing API.
-  New structs:
-    - AVFilterGraphSegment
-    - AVFilterChain
-    - AVFilterParams
-    - AVFilterPadParams
-  New functions:
-    - avfilter_graph_segment_parse()
-    - avfilter_graph_segment_create_filters()
-    - avfilter_graph_segment_apply_opts()
-    - avfilter_graph_segment_init()
-    - avfilter_graph_segment_link()
-    - avfilter_graph_segment_apply()
-
-2023-02-09 - 719a93f4e4 - lavu 58.0.100 - csp.h
-  Add av_csp_approximate_trc_gamma() and av_csp_trc_func_from_id().
-  Add av_csp_trc_function.
-
-2023-02-09 - 868a31b42d - lavc 60.0.100 - avcodec.h
-  avcodec_decode_subtitle2() now accepts const AVPacket*.
-
-2023-02-04 - d02340b9e3 - lavc 59.63.100
-  Allow AV_CODEC_FLAG_COPY_OPAQUE to be used with decoders.
-
-2023-01-29 - a1a80f2e64 - lavc 59.59.100 - avcodec.h
-  Add AV_CODEC_FLAG_COPY_OPAQUE and AV_CODEC_FLAG_FRAME_DURATION.
-
-2023-01-13 - 002d0ec740 - lavu 57.44.100 - ambient_viewing_environment.h frame.h
-  Adds a new structure for holding H.274 Ambient Viewing Environment metadata,
-  AVAmbientViewingEnvironment.
-  Adds a new AVFrameSideDataType entry AV_FRAME_DATA_AMBIENT_VIEWING_ENVIRONMENT
-  for it.
-
-2022-12-10 - 7a8d78f7e3 - lavc 59.55.100 - avcodec.h
-  Add AV_HWACCEL_FLAG_UNSAFE_OUTPUT.
-
-2022-11-24 - e97368eba5 - lavu 57.43.100 - tx.h
-  Add AV_TX_FLOAT_DCT, AV_TX_DOUBLE_DCT and AV_TX_INT32_DCT.
-
-2022-11-06 - 9dad237928 - lavu 57.42.100 - dict.h
-  Add av_dict_iterate().
-
-2022-11-03 - 6228ba141d - lavu 57.41.100 - channel_layout.h
-  Add AV_CH_LAYOUT_7POINT1_TOP_BACK and AV_CHANNEL_LAYOUT_7POINT1_TOP_BACK.
-
-2022-10-30 - 83e918de71 - lavu 57.40.100 - channel_layout.h
-  Add AV_CH_LAYOUT_CUBE and AV_CHANNEL_LAYOUT_CUBE.
-
-2022-10-11 - 479747645f - lavu 57.39.101 - pixfmt.h
-  Add AV_PIX_FMT_RGBF32 and AV_PIX_FMT_RGBAF32.
-
-2022-10-05 - 37d5ddc317 - lavu 57.39.100 - cpu.h
-  Add AV_CPU_FLAG_RVB_BASIC.
-
-2022-10-03 - d09776d486 - lavf 59.34.100 - avio.h
-  Make AVIODirContext an opaque type in a future major version bump.
-
-2022-09-27 - 0c0a3deb18 - lavu 57.38.100 - cpu.h
-  Add CPU flags for RISC-V vector extensions:
-  AV_CPU_FLAG_RVV_I32, AV_CPU_FLAG_RVV_F32, AV_CPU_FLAG_RVV_I64,
-  AV_CPU_FLAG_RVV_F64
-
-2022-09-26 - a02a0e8db4 - lavc 59.48.100 - avcodec.h
-  Deprecate avcodec_enum_to_chroma_pos() and avcodec_chroma_pos_to_enum().
-  Use av_chroma_location_enum_to_pos() or av_chroma_location_pos_to_enum()
-  instead.
-
-2022-09-26 - xxxxxxxxxx - lavu 57.37.100 - pixdesc.h pixfmt.h
-  Add av_chroma_location_enum_to_pos() and av_chroma_location_pos_to_enum().
-  Add AV_PIX_FMT_RGBF32BE, AV_PIX_FMT_RGBF32LE, AV_PIX_FMT_RGBAF32BE,
-  AV_PIX_FMT_RGBAF32LE.
-
-2022-09-26 - cf856d8957 - lavc 59.47.100 - avcodec.h defs.h
-  Move the AV_EF_* and FF_COMPLIANCE_* defines from avcodec.h to defs.h.
-
-2022-09-03 - d75c4693fe - lavu 57.36.100 - pixfmt.h
-  Add AV_PIX_FMT_P012, AV_PIX_FMT_Y212, AV_PIX_FMT_XV30, AV_PIX_FMT_XV36
-
-2022-09-03 - dea9744560 - lavu 57.35.100 - file.h
-  Deprecate av_tempfile() without replacement.
-
-2022-08-03 - cc5a5c9860 - lavu 57.34.100 - pixfmt.h
-  Add AV_PIX_FMT_VUYX.
-
-2022-08-22 - 14726571dd - lavf 59 - avformat.h
-  Deprecate av_stream_get_end_pts() without replacement.
-
-2022-08-19 - 352799dca8 - lavc 59.42.102 - codec_id.h
-  Deprecate AV_CODEC_ID_AYUV and ayuv decoder/encoder. The rawvideo codec
-  and vuya pixel format combination will be used instead from now on.
-
-2022-08-07 - e95b08a7dd - lavu 57.33.101 - pixfmt.h
-  Add AV_PIX_FMT_RGBAF16{BE,LE} pixel formats.
-
-2022-08-12 - e0bbdbe0a6 - lavu 57.33.100 - hwcontext_qsv.h
-  Add loader field to AVQSVDeviceContext
-
-2022-08-03 - 6ab8a9d375 - lavu 57.32.100 - pixfmt.h
-  Add AV_PIX_FMT_VUYA.
-
-2022-08-02 - e3838b856f - lavc 59.41.100 - avcodec.h codec.h
-  Add AV_CODEC_FLAG_RECON_FRAME and AV_CODEC_CAP_ENCODER_RECON_FRAME.
-  avcodec_receive_frame() may now be used on encoders when
-  AV_CODEC_FLAG_RECON_FRAME is active.
-
-2022-08-02 - eede1d2927 - lavu 57.31.100 - frame.h
-  av_frame_make_writable() may now be called on non-refcounted
-  frames and will make a refcounted copy out of them.
-  Previously an error was returned in such cases.
-
-2022-07-30 - e1a0f2df3d - lavc 59.40.100 - avcodec.h
-  Add the AV_CODEC_FLAG2_ICC_PROFILES flag to AVCodecContext, to enable
-  automatic reading and writing of embedded ICC profiles in image files.
-  The "flags2" option now supports the corresponding flag "icc_profiles".
-
-2022-07-19 - 4397f9a5a0 - lavu 57.30.100 - frame.h
-  Add AVFrame.duration, deprecate AVFrame.pkt_duration.
-
-------- 8< --------- FFmpeg 5.1 was cut here -------- 8< ---------
-
-2022-06-12 - 7cae3d8b76 - lavf 59.25.100 - avio.h
-  Add avio_vprintf(), similar to avio_printf() but allow to use it
-  from within a function taking a variable argument list as input.
-
-2022-06-12 - ff59ecc4de - lavu 57.27.100 - uuid.h
-  Add UUID handling functions.
-  Add av_uuid_parse(), av_uuid_urn_parse(), av_uuid_parse_range(),
-  av_uuid_parse_range(), av_uuid_equal(), av_uuid_copy(), and av_uuid_nil().
-
-2022-06-01 - d42b410e05 - lavu 57.26.100 - csp.h
-  Add public API for colorspace structs.
-  Add av_csp_luma_coeffs_from_avcsp(), av_csp_primaries_desc_from_id(),
-  and av_csp_primaries_id_from_desc().
-
-2022-05-23 - 4cdc14aa95 - lavu 57.25.100 - avutil.h
-  Deprecate av_fopen_utf8() without replacement.
-
-2022-03-16 - f3a0e2ee2b - all libraries - version_major.h
-  Add lib<name>/version_major.h as new installed headers, which only
-  contain the major version number (and corresponding API deprecation
-  defines).
-
-2022-03-15 - cdba98bb80 - swr 4.5.100 - swresample.h
-  Add swr_alloc_set_opts2() and swr_build_matrix2().
-  Deprecate swr_alloc_set_opts() and swr_build_matrix().
-
-2022-03-15 - cdba98bb80 - lavfi 8.28.100 - avfilter.h buffersink.h buffersrc.h
-  Update AVFilterLink for the new channel layout API: add ch_layout,
-  deprecate channel_layout.
-
-  Update the buffersink filter sink for the new channel layout API:
-  add av_buffersink_get_ch_layout() and the ch_layouts option,
-  deprecate av_buffersink_get_channel_layout() and the channel_layouts option.
-
-  Update AVBufferSrcParameters for the new channel layout API:
-  add ch_layout, deprecate channel_layout.
-
-2022-03-15 - cdba98bb80 - lavf 59.19.100 - avformat.h
-  Add AV_DISPOSITION_NON_DIEGETIC.
-
-2022-03-15 - cdba98bb80 - lavc 59.24.100 - avcodec.h codec_par.h
-  Update AVCodecParameters for the new channel layout API: add ch_layout,
-  deprecate channels/channel_layout.
-
-  Update AVCodecContext for the new channel layout API: add ch_layout,
-  deprecate channels/channel_layout.
-
-  Update AVCodec for the new channel layout API: add ch_layouts,
-  deprecate channel_layouts.
-
-2022-03-15 - cdba98bb80 - lavu 57.24.100 - channel_layout.h frame.h opt.h
-  Add new channel layout API based on the AVChannelLayout struct.
-  Add support for Ambisonic audio.
-  Deprecate previous channel layout API based on uint64 bitmasks.
-
-  Add AV_OPT_TYPE_CHLAYOUT option type, deprecate AV_OPT_TYPE_CHANNEL_LAYOUT.
-  Update AVFrame for the new channel layout API: add ch_layout, deprecate
-  channels/channel_layout.
-
-2022-03-10 - f629ea2e18 - lavu 57.23.100 - cpu.h
-  Add AV_CPU_FLAG_AVX512ICL.
-
-2022-02-07 - a10f1aec1f - lavu 57.21.100 - fifo.h
-  Deprecate AVFifoBuffer and the API around it, namely av_fifo_alloc(),
-  av_fifo_alloc_array(), av_fifo_free(), av_fifo_freep(), av_fifo_reset(),
-  av_fifo_size(), av_fifo_space(), av_fifo_generic_peek_at(),
-  av_fifo_generic_peek(), av_fifo_generic_read(), av_fifo_generic_write(),
-  av_fifo_realloc2(), av_fifo_grow(), av_fifo_drain() and av_fifo_peek2().
-  Users should switch to the AVFifo-API.
-
-2022-02-07 - 7329b22c05 - lavu 57.20.100 - fifo.h
-  Add a new FIFO API, which allows setting a FIFO element size.
-  This API operates on these elements rather than on bytes.
-  Add av_fifo_alloc2(), av_fifo_elem_size(), av_fifo_can_read(),
-  av_fifo_can_write(), av_fifo_grow2(), av_fifo_drain2(), av_fifo_write(),
-  av_fifo_write_from_cb(), av_fifo_read(), av_fifo_read_to_cb(),
-  av_fifo_peek(), av_fifo_peek_to_cb(), av_fifo_drain2(), av_fifo_reset2(),
-  av_fifo_freep2(), av_fifo_auto_grow_limit().
-
-2022-01-26 - af94ab7c7c0 - lavu 57.19.100 - tx.h
-  Add AV_TX_FLOAT_RDFT, AV_TX_DOUBLE_RDFT and AV_TX_INT32_RDFT.
-
-------- 8< --------- FFmpeg 5.0 was cut here -------- 8< ---------
-
-2022-01-04 - 78dc21b123e - lavu 57.16.100 - frame.h
-  Add AV_FRAME_DATA_DOVI_METADATA.
-
-2022-01-03 - 70f318e6b6c - lavf 59.13.100 - avformat.h
-  Add AVFMT_EXPERIMENTAL flag.
-
-2021-12-22 - b7e1ec7bda9 - lavu 57.13.100 - hwcontext_videotoolbox.h
-  Add av_vt_pixbuf_set_attachments
-
-2021-12-22 - 69bd95dcd8d - lavu 57.13.100 - hwcontext_videotoolbox.h
-  Add av_map_videotoolbox_chroma_loc_from_av
-  Add av_map_videotoolbox_color_matrix_from_av
-  Add av_map_videotoolbox_color_primaries_from_av
-  Add av_map_videotoolbox_color_trc_from_av
-
-2021-12-21 - ffbab99f2c2 - lavu 57.12.100 - cpu.h
-  Add AV_CPU_FLAG_SLOW_GATHER.
-
-2021-12-20 - 278068dc60d - lavu 57.11.101 - display.h
-  Modified the documentation of av_display_rotation_set()
-  to match its longstanding actual behaviour of treating
-  the angle as directed clockwise.
-
-2021-12-12 - 64834bb86a1 - lavf 59.10.100 - avformat.h
-  Add AVFormatContext io_close2 which returns an int
-
-2021-12-10 - f45cbb775e4 - lavu 57.11.100 - hwcontext_vulkan.h
-  Add AVVkFrame.offset and AVVulkanFramesContext.flags.
-
-2021-12-04 - b9c928a486f - lavfi 8.19.100 - avfilter.h
-  Add AVFILTER_FLAG_METADATA_ONLY.
-
-2021-12-03 - b236ef0a594 - lavu 57.10.100 - frame.h
-  Add AVFrame.time_base
-
-2021-11-22 - b2cd1fb2ec6 - lavu 57.9.100 - pixfmt.h
-  Add AV_PIX_FMT_P210, AV_PIX_FMT_P410, AV_PIX_FMT_P216, and AV_PIX_FMT_P416.
-
-2021-11-17 - 54e65aa38ab - lavf 57.9.100 - frame.h
-  Add AV_FRAME_DATA_DOVI_RPU_BUFFER.
-
-2021-11-16 - ed75a08d36c - lavf 59.9.100 - avformat.h
-  Add av_stream_get_class(). Schedule adding AVStream.av_class at libavformat
-  major version 60.
-  Add av_disposition_to_string() and av_disposition_from_string().
-  Add "disposition" AVOption to AVStream's class.
-
-2021-11-12 - 8478d60d5b5 - lavu 57.8.100 - hwcontext_vulkan.h
-  Added AVVkFrame.sem_value, AVVulkanDeviceContext.queue_family_encode_index,
-  nb_encode_queues, queue_family_decode_index, and nb_decode_queues.
-
-2021-10-18 - 682bafdb125 - lavf 59.8.100 - avio.h
-  Introduce public bytes_{read,written} statistic fields to AVIOContext.
-
-2021-10-13 - a5622ed16f8 - lavf 59.7.100 - avio.h
-  Deprecate AVIOContext.written. Originally added as a private entry in
-  commit 3f75e5116b900f1428aa13041fc7d6301bf1988a, its grouping with
-  the comment noting its private state was missed during merging of the field
-  from Libav (most likely due to an already existing field in between).
-
-2021-09-21 - 0760d9153c3 - lavu 57.7.100 - pixfmt.h
-  Add AV_PIX_FMT_X2BGR10.
-
-2021-09-20 - 8d5de914d31 - lavu 57.6.100 - mem.h
-  Deprecate av_mallocz_array() as it is identical to av_calloc().
-
-2021-09-20 - 176b8d785bf - lavc 59.9.100 - avcodec.h
-  Deprecate AVCodecContext.sub_text_format and the corresponding
-  AVOptions. It is unused since the last major bump.
-
-2021-09-20 - dd846bc4a91 - lavc 59.8.100 - avcodec.h codec.h
-  Deprecate AV_CODEC_FLAG_TRUNCATED and AV_CODEC_CAP_TRUNCATED,
-  as they are redundant with parsers.
-
-2021-09-17 - ccfdef79b13 - lavu 57.5.101 - buffer.h
-  Constified the input parameters in av_buffer_replace(), av_buffer_ref(),
-  and av_buffer_pool_buffer_get_opaque().
-
-2021-09-08 - 4f78711f9c2 - lavu 57.5.100 - hwcontext_d3d11va.h
-  Add AVD3D11VAFramesContext.texture_infos
-
-2021-09-06 - 42cd64c1826 - lsws 6.1.100 - swscale.h
-  Add AVFrame-based scaling API:
-    - sws_scale_frame()
-    - sws_frame_start()
-    - sws_frame_end()
-    - sws_send_slice()
-    - sws_receive_slice()
-    - sws_receive_slice_alignment()
-
-2021-09-02 - cbf111059d2 - lavc 59.7.100 - avcodec.h
-  Incremented the number of elements of AVCodecParser.codec_ids to seven.
-
-2021-08-24 - 590a7e02f04 - lavc 59.6.100 - avcodec.h
-  Add FF_CODEC_PROPERTY_FILM_GRAIN
-
-2021-08-20 - 7c5f998196d - lavfi 8.3.100 - avfilter.H
-  Add avfilter_filter_pad_count() as a replacement for avfilter_pad_count().
-  Deprecate avfilter_pad_count().
-
-2021-08-17 - 8c53b145993 - lavu 57.4.101 - opt.h
-  av_opt_copy() now guarantees that allocated src and dst options
-  don't alias each other even on error.
-
-2021-08-14 - d5de9965ef6 - lavu 57.4.100 - imgutils.h
-  Add av_image_copy_plane_uc_from()
-
-2021-08-02 - a1a0fddfd05 - lavc 59.4.100 - packet.h
-  Add AVPacket.opaque, AVPacket.opaque_ref, AVPacket.time_base.
-
-2021-07-23 - 2dd8acbe800 - lavu 57.3.100 - common.h macros.h
-  Move several macros (AV_NE, FFDIFFSIGN, FFMAX, FFMAX3, FFMIN, FFMIN3,
-  FFSWAP, FF_ARRAY_ELEMS, MKTAG, MKBETAG) from common.h to macros.h.
-
-2021-07-22 - e3b5ff17c2e - lavu 57.2.100 - film_grain_params.h
-  Add AV_FILM_GRAIN_PARAMS_H274, AVFilmGrainH274Params
-
-2021-07-19 - c1bf56a526f - lavu 57.1.100 - cpu.h
-  Add av_cpu_force_count()
-
-2021-06-17 - aca923b3653 - lavc 59.2.100 - packet.h
-  Add AV_PKT_DATA_DYNAMIC_HDR10_PLUS
-
-2021-06-09 - 2cccab96f6f - lavf 59.3.100 - avformat.h
-  Add pts_wrap_bits to AVStream
-
-2021-06-10 - 7c9763070d9 - lavc 59.1.100 - avcodec.h codec.h
-  Move av_get_profile_name() from avcodec.h to codec.h.
-
-2021-06-10 - bb3648e6766 - lavc 59.1.100 - avcodec.h codec_par.h
-  Move av_get_audio_frame_duration2() from avcodec.h to codec_par.h.
-
-2021-06-10 - 881db34f6a0 - lavc 59.1.100 - avcodec.h codec_id.h
-  Move av_get_bits_per_sample(), av_get_exact_bits_per_sample(),
-  avcodec_profile_name(), and av_get_pcm_codec() from avcodec.h
-  to codec_id.h.
-
-2021-06-10 - ff0a96046d8 - lavc 59.1.100 - avcodec.h defs.h
-  Add new installed header defs.h. The following definitions are moved
-  into it from avcodec.h:
-    - AVDiscard
-    - AVAudioServiceType
-    - AVPanScan
-    - AVCPBProperties and av_cpb_properties_alloc()
-    - AVProducerReferenceTime
-    - av_xiphlacing()
-
-2021-04-27 - cb3ac722f4 - lavc 59.0.100 - avcodec.h
-  Constified AVCodecParserContext.parser.
-
-2021-04-27 - 8b3e6ce5f4 - lavd 59.0.100 - avdevice.h
-  The av_*_device_next API functions now accept and return
-  pointers to const AVInputFormat resp. AVOutputFormat.
-
-2021-04-27 - d7e0d428fa - lavd 59.0.100 - avdevice.h
-  avdevice_list_input_sources and avdevice_list_output_sinks now accept
-  pointers to const AVInputFormat resp. const AVOutputFormat.
-
-2021-04-27 - 46dac8cf3d - lavf 59.0.100 - avformat.h
-  av_find_best_stream now uses a const AVCodec ** parameter
-  for the returned decoder.
-
-2021-04-27 - 626535f6a1 - lavc 59.0.100 - codec.h
-  avcodec_find_encoder_by_name(), avcodec_find_encoder(),
-  avcodec_find_decoder_by_name() and avcodec_find_decoder()
-  now return a pointer to const AVCodec.
-
-2021-04-27 - 14fa0a4efb - lavf 59.0.100 - avformat.h
-  Constified AVFormatContext.*_codec.
-
-2021-04-27 - 56450a0ee4 - lavf 59.0.100 - avformat.h
-  Constified the pointers to AVInputFormats and AVOutputFormats
-  in AVFormatContext, avformat_alloc_output_context2(),
-  av_find_input_format(), av_probe_input_format(),
-  av_probe_input_format2(), av_probe_input_format3(),
-  av_probe_input_buffer2(), av_probe_input_buffer(),
-  avformat_open_input(), av_guess_format() and av_guess_codec().
-  Furthermore, constified the AVProbeData in av_probe_input_format(),
-  av_probe_input_format2() and av_probe_input_format3().
-
-2021-04-19 - 18af1ea8d1 - lavu 56.74.100 - tx.h
-  Add AV_TX_FULL_IMDCT and AV_TX_UNALIGNED.
-
-2021-04-17 - f1bf465aa0 - lavu 56.73.100 - frame.h detection_bbox.h
-  Add AV_FRAME_DATA_DETECTION_BBOXES
-
-2021-04-06 - 557953a397 - lavf 58.78.100 - avformat.h
-  Add avformat_index_get_entries_count(), avformat_index_get_entry(),
-  and avformat_index_get_entry_from_timestamp().
-
-2021-03-21 - a77beea6c8 - lavu 56.72.100 - frame.h
-  Deprecated av_get_colorspace_name().
-  Use av_color_space_name() instead.
-
 -------- 8< --------- FFmpeg 4.4 was cut here -------- 8< ---------

 2021-03-19 - e8c0bca6bd - lavu 56.69.100 - adler32.h
@@ -1982,7 +1410,7 @@ API changes, most recent first:
 2014-04-15 - ef818d8 - lavf 55.37.101 - avformat.h
  Add av_format_inject_global_side_data()

-2014-04-12 - 4f698be8f - lavu 52.76.100 - log.h
+2014-04-12 - 4f698be - lavu 52.76.100 - log.h
  Add av_log_get_flags()

 2014-04-11 - 6db42a2b - lavd 55.12.100 - avdevice.h
--- 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         =
+PROJECT_NUMBER         = 4.4.6

 # Using the PROJECT_BRIEF tag one can provide an optional one line description
 # for a project that appears at the top of each page and should give viewer a
@@ -1980,7 +1980,6 @@ PREDEFINED             = __attribute__(x)= \
                         av_alloc_size(...)= \
                         AV_GCC_VERSION_AT_LEAST(x,y)=1 \
                         AV_GCC_VERSION_AT_MOST(x,y)=0 \
-                         "FF_PAD_STRUCTURE(name,size,...)=typedef struct name { __VA_ARGS__ } name;" \
                         __GNUC__

 # If the MACRO_EXPANSION and EXPAND_ONLY_PREDEF tags are set to YES then this
--- a/doc/Makefile
+++ b/doc/Makefile
@@ -19,7 +19,6 @@ MANPAGES3   = $(LIBRARIES-yes:%=doc/%.3)
 MANPAGES    = $(MANPAGES1) $(MANPAGES3)
 PODPAGES    = $(AVPROGS-yes:%=doc/%.pod)  $(AVPROGS-yes:%=doc/%-all.pod)  $(COMPONENTS-yes:%=doc/%.pod)  $(LIBRARIES-yes:%=doc/%.pod)
 HTMLPAGES   = $(AVPROGS-yes:%=doc/%.html) $(AVPROGS-yes:%=doc/%-all.html) $(COMPONENTS-yes:%=doc/%.html) $(LIBRARIES-yes:%=doc/%.html) \
-              doc/community.html                                        \
              doc/developer.html                                        \
              doc/faq.html                                              \
              doc/fate.html                                             \
@@ -28,9 +27,6 @@ HTMLPAGES   = $(AVPROGS-yes:%=doc/%.html) $(AVPROGS-yes:%=doc/%-all.html) $(COMP
              doc/mailing-list-faq.html                                 \
              doc/nut.html                                              \
              doc/platform.html                                         \
-              $(SRC_PATH)/doc/bootstrap.min.css                         \
-              $(SRC_PATH)/doc/style.min.css                             \
-              $(SRC_PATH)/doc/default.css                               \

 TXTPAGES    = doc/fate.txt                                              \

@@ -106,7 +102,7 @@ DOXY_INPUT_DEPS = $(addprefix $(SRC_PATH)/, $(DOXY_INPUT)) ffbuild/config.mak

 doc/doxy/html: TAG = DOXY
 doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(SRC_PATH)/doc/doxy-wrapper.sh $(DOXY_INPUT_DEPS)
-	$(M)$(SRC_PATH)/doc/doxy-wrapper.sh $$PWD/doc/doxy $(SRC_PATH) doc/Doxyfile $(DOXYGEN) $(DOXY_INPUT);
+	$(M)OUT_DIR=$$PWD/doc/doxy; cd $(SRC_PATH); ./doc/doxy-wrapper.sh $$OUT_DIR $< $(DOXYGEN) $(DOXY_INPUT);

 install-doc: install-html install-man

--- a/doc/bitstream_filters.texi
+++ b/doc/bitstream_filters.texi
@@ -81,7 +81,7 @@ Top-left position.
@end table

@item tick_rate
-Set the tick rate (@emph{time_scale / num_units_in_display_tick}) in
+Set the tick rate (@emph{num_units_in_display_tick / time_scale}) in
 the timing info in the sequence header.
@item num_ticks_per_picture
 Set the number of ticks in each picture, to indicate that the stream
@@ -132,36 +132,6 @@ the header stored in extradata to the key packets:
 ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts
@end example

-@section dv_error_marker
-
-Blocks in DV which are marked as damaged are replaced by blocks of the specified color.
-
-@table @option
-@item color
-The color to replace damaged blocks by
-@item sta
-A 16 bit mask which specifies which of the 16 possible error status values are
-to be replaced by colored blocks. 0xFFFE is the default which replaces all non 0
-error status values.
-@table @samp
-@item ok
-No error, no concealment
-@item err
-Error, No concealment
-@item res
-Reserved
-@item notok
-Error or concealment
-@item notres
-Not reserved
-@item Aa, Ba, Ca, Ab, Bb, Cb, A, B, C, a, b, erri, erru
-The specific error status code
-@end table
-see page 44-46 or section 5.5 of
-@url{http://web.archive.org/web/20060927044735/http://www.smpte.org/smpte_store/standards/pdf/s314m.pdf}
-
-@end table
-
@section eac3_core

 Extract the core from a E-AC-3 stream, dropping extra channels.
@@ -247,16 +217,12 @@ Modify metadata embedded in an H.264 stream.
 Insert or remove AUD NAL units in all access units of the stream.

@table @samp
-@item pass
@item insert
@item remove
@end table

-Default is pass.
-
@item sample_aspect_ratio
 Set the sample aspect ratio of the stream in the VUI parameters.
-See H.264 table E-1.

@item overscan_appropriate_flag
 Set whether the stream is suitable for display using overscan
@@ -278,7 +244,7 @@ Set the chroma sample location in the stream (see H.264 section
 E.2.1 and figure E-1).

@item tick_rate
-Set the tick rate (time_scale / num_units_in_tick) in the VUI
+Set the tick rate (num_units_in_tick / time_scale) in the VUI
 parameters.  This is the smallest time unit representable in the
 stream, and in many cases represents the field rate of the stream
 (double the frame rate).
@@ -287,11 +253,6 @@ Set whether the stream has fixed framerate - typically this indicates
 that the framerate is exactly half the tick rate, but the exact
 meaning is dependent on interlacing and the picture structure (see
 H.264 section E.2.1 and table E-6).
-@item zero_new_constraint_set_flags
-Zero constraint_set4_flag and constraint_set5_flag in the SPS. These
-bits were reserved in a previous version of the H.264 spec, and thus
-some hardware decoders require these to be zero. The result of zeroing
-this is still a valid bitstream.

@item crop_left
@item crop_right
@@ -315,37 +276,6 @@ insert the string ``hello'' associated with the given UUID.
@item delete_filler
 Deletes both filler NAL units and filler SEI messages.

-@item display_orientation
-Insert, extract or remove Display orientation SEI messages.
-See H.264 section D.1.27 and D.2.27 for syntax and semantics.
-
-@table @samp
-@item pass
-@item insert
-@item remove
-@item extract
-@end table
-
-Default is pass.
-
-Insert mode works in conjunction with @code{rotate} and @code{flip} options.
-Any pre-existing Display orientation messages will be removed in insert or remove mode.
-Extract mode attaches the display matrix to the packet as side data.
-
-@item rotate
-Set rotation in display orientation SEI (anticlockwise angle in degrees).
-Range is -360 to +360. Default is NaN.
-
-@item flip
-Set flip in display orientation SEI.
-
-@table @samp
-@item horizontal
-@item vertical
-@end table
-
-Default is unset.
-
@item level
 Set the level in the SPS.  Refer to H.264 section A.3 and tables A-1
 to A-5.
@@ -382,6 +312,9 @@ This applies a specific fixup to some Blu-ray streams which contain
 redundant PPSs modifying irrelevant parameters of the stream which
 confuse other transformations which require correct extradata.

+A new single global PPS is created, and all of the redundant PPSs
+within the stream are removed.
+
@section hevc_metadata

 Modify metadata embedded in an HEVC stream.
@@ -414,8 +347,8 @@ Set the chroma sample location in the stream (see H.265 section
 E.3.1 and figure E.1).

@item tick_rate
-Set the tick rate in the VPS and VUI parameters (time_scale /
-num_units_in_tick). Combined with @option{num_ticks_poc_diff_one}, this can
+Set the tick rate in the VPS and VUI parameters (num_units_in_tick /
+time_scale).  Combined with @option{num_ticks_poc_diff_one}, this can
 set a constant framerate in the stream.  Note that it is likely to be
 overridden by container parameters when the stream is in a container.

@@ -596,67 +529,20 @@ container. Can be used for fuzzing or testing error resilience/concealment.
 Parameters:
@table @option
@item amount
-Accepts an expression whose evaluation per-packet determines how often bytes in that
-packet will be modified. A value below 0 will result in a variable frequency.
-Default is 0 which results in no modification. However, if neither amount nor drop is specified,
-amount will be set to @var{-1}. See below for accepted variables.
-@item drop
-Accepts an expression evaluated per-packet whose value determines whether that packet is dropped.
-Evaluation to a positive value results in the packet being dropped. Evaluation to a negative
-value results in a variable chance of it being dropped, roughly inverse in proportion to the magnitude
-of the value. Default is 0 which results in no drops. See below for accepted variables.
+A numeral string, whose value is related to how often output bytes will
+be modified. Therefore, values below or equal to 0 are forbidden, and
+the lower the more frequent bytes will be modified, with 1 meaning
+every byte is modified.
@item dropamount
-Accepts a non-negative integer, which assigns a variable chance of it being dropped, roughly inverse
-in proportion to the value. Default is 0 which results in no drops. This option is kept for backwards
-compatibility and is equivalent to setting drop to a negative value with the same magnitude
-i.e. @code{dropamount=4} is the same as @code{drop=-4}. Ignored if drop is also specified.
+A numeral string, whose value is related to how often packets will be dropped.
+Therefore, values below or equal to 0 are forbidden, and the lower the more
+frequent packets will be dropped, with 1 meaning every packet is dropped.
@end table

-Both @code{amount} and @code{drop} accept expressions containing the following variables:
-
-@table @samp
-@item n
-The index of the packet, starting from zero.
-@item tb
-The timebase for packet timestamps.
-@item pts
-Packet presentation timestamp.
-@item dts
-Packet decoding timestamp.
-@item nopts
-Constant representing AV_NOPTS_VALUE.
-@item startpts
-First non-AV_NOPTS_VALUE PTS seen in the stream.
-@item startdts
-First non-AV_NOPTS_VALUE DTS seen in the stream.
-@item duration
-@itemx d
-Packet duration, in timebase units.
-@item pos
-Packet position in input; may be -1 when unknown or not set.
-@item size
-Packet size, in bytes.
-@item key
-Whether packet is marked as a keyframe.
-@item state
-A pseudo random integer, primarily derived from the content of packet payload.
-@end table
-
-@subsection Examples
-Apply modification to every byte but don't drop any packets.
+The following example applies the modification to every byte but does not drop
+any packets.
@example
-ffmpeg -i INPUT -c copy -bsf noise=1 output.mkv
-@end example
-
-Drop every video packet not marked as a keyframe after timestamp 30s but do not
-modify any of the remaining packets.
-@example
-ffmpeg -i INPUT -c copy -bsf:v noise=drop='gt(t\,30)*not(key)' output.mkv
-@end example
-
-Drop one second of audio every 10 seconds and add some random noise to the rest.
-@example
-ffmpeg -i INPUT -c copy -bsf:a noise=amount=-1:drop='between(mod(t\,10)\,9\,10)' output.mkv
+ffmpeg -i INPUT -c copy -bsf noise[=1] output.mkv
@end example

@section null
@@ -692,14 +578,6 @@ for NTSC frame rate using the @option{frame_rate} option.
 ffmpeg -f lavfi -i sine=r=48000:d=1 -c pcm_s16le -bsf pcm_rechunk=r=30000/1001 -f framecrc -
@end example

-@section pgs_frame_merge
-
-Merge a sequence of PGS Subtitle segments ending with an "end of display set"
-segment into a single packet.
-
-This is required by some containers that support PGS subtitles
-(muxer @code{matroska}).
-
@section prores_metadata

 Modify color property metadata embedded in prores stream.
@@ -806,10 +684,6 @@ It accepts the following parameters:
@item pts
@item dts
 Set expressions for PTS, DTS or both.
-@item duration
-Set expression for duration.
-@item time_base
-Set output time base.
@end table

 The expressions are evaluated through the eval API and can contain the following
@@ -833,9 +707,6 @@ The demux timestamp in input.
@item PTS
 The presentation timestamp in input.

-@item DURATION
-The duration in input.
-
@item STARTDTS
 The DTS of the first packet.

@@ -848,38 +719,17 @@ The previous input DTS.
@item PREV_INPTS
 The previous input PTS.

-@item PREV_INDURATION
-The previous input duration.
-
@item PREV_OUTDTS
 The previous output DTS.

@item PREV_OUTPTS
 The previous output PTS.

-@item PREV_OUTDURATION
-The previous output duration.
-
-@item NEXT_DTS
-The next input DTS.
-
-@item NEXT_PTS
-The next input PTS.
-
-@item NEXT_DURATION
-The next input duration.
-
@item TB
 The timebase of stream packet belongs.

-@item TB_OUT
-The output timebase.
-
@item SR
 The sample rate of stream packet belongs.
-
-@item NOPTS
-The AV_NOPTS_VALUE constant.
@end table

@anchor{text2movsub}
--- a/doc/bootstrap.min.css
+++ b/doc/bootstrap.min.css
--- a/doc/codecs.texi
+++ b/doc/codecs.texi
@@ -144,6 +144,21 @@ Default value is 0.
@item b_qfactor @var{float} (@emph{encoding,video})
 Set qp factor between P and B frames.

+@item b_strategy @var{integer} (@emph{encoding,video})
+Set strategy to choose between I/P/B-frames.
+
+@item ps @var{integer} (@emph{encoding,video})
+Set RTP payload size in bytes.
+
+@item mv_bits @var{integer}
+@item header_bits @var{integer}
+@item i_tex_bits @var{integer}
+@item p_tex_bits @var{integer}
+@item i_count @var{integer}
+@item p_count @var{integer}
+@item skip_count @var{integer}
+@item misc_bits @var{integer}
+@item frame_bits @var{integer}
@item codec_tag @var{integer}
@item bug @var{flags} (@emph{decoding,video})
 Workaround not auto detected encoder bugs.
@@ -233,6 +248,9 @@ consider things that a sane encoder should not do as an error

@item block_align @var{integer}

+@item mpeg_quant @var{integer} (@emph{encoding,video})
+Use MPEG quantizers instead of H.263.
+
@item rc_override_count @var{integer}

@item maxrate @var{integer} (@emph{encoding,audio,video})
@@ -338,6 +356,19 @@ favor predicting from the previous frame instead of the current

@item bits_per_coded_sample @var{integer}

+@item pred @var{integer} (@emph{encoding,video})
+Set prediction method.
+
+Possible values:
+@table @samp
+@item left
+
+@item plane
+
+@item median
+
+@end table
+
@item aspect @var{rational number} (@emph{encoding,video})
 Set sample aspect ratio.

@@ -554,6 +585,9 @@ sab diamond motion estimation
@item last_pred @var{integer} (@emph{encoding,video})
 Set amount of motion predictors from the previous frame.

+@item preme @var{integer} (@emph{encoding,video})
+Set pre motion estimation.
+
@item precmp @var{integer} (@emph{encoding,video})
 Set pre motion estimation compare function.

@@ -602,6 +636,23 @@ Set limit motion vectors range (1023 for DivX player).

@item global_quality @var{integer} (@emph{encoding,audio,video})

+@item coder @var{integer} (@emph{encoding,video})
+
+Possible values:
+@table @samp
+@item vlc
+variable length coder / huffman coder
+@item ac
+arithmetic coder
+@item raw
+raw (no encoding)
+@item rle
+run-length coder
+@end table
+
+@item context @var{integer} (@emph{encoding,video})
+Set context model.
+
@item slice_flags @var{integer}

@item mbd @var{integer} (@emph{encoding,video})
@@ -617,6 +668,12 @@ use fewest bits
 use best rate distortion
@end table

+@item sc_threshold @var{integer} (@emph{encoding,video})
+Set scene change threshold.
+
+@item nr @var{integer} (@emph{encoding,video})
+Set noise reduction.
+
@item rc_init_occupancy @var{integer} (@emph{encoding,video})
 Set number of bits which should be loaded into the rc buffer before
 decoding starts.
@@ -644,8 +701,6 @@ for codecs that support it. See also @file{doc/examples/export_mvs.c}.
 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.
-@item icc_profiles
-Generate/parse embedded ICC profiles from/to colorimetry tags.
@end table

@item export_side_data @var{flags} (@emph{decoding/encoding,audio,video,subtitles})
@@ -697,24 +752,73 @@ profiles are documented in the relevant encoder documentation.

@item level @var{integer} (@emph{encoding,audio,video})

-Set the encoder level. This level depends on the specific codec, and
-might correspond to the profile level. It is set by default to
-@samp{unknown}.
-
 Possible values:
@table @samp
@item unknown
+
@end table

@item lowres @var{integer} (@emph{decoding,audio,video})
 Decode at 1= 1/2, 2=1/4, 3=1/8 resolutions.

+@item skip_threshold @var{integer} (@emph{encoding,video})
+Set frame skip threshold.
+
+@item skip_factor @var{integer} (@emph{encoding,video})
+Set frame skip factor.
+
+@item skip_exp @var{integer} (@emph{encoding,video})
+Set frame skip exponent.
+Negative values behave identical to the corresponding positive ones, except
+that the score is normalized.
+Positive values exist primarily for compatibility reasons and are not so useful.
+
+@item skipcmp @var{integer} (@emph{encoding,video})
+Set frame skip compare function.
+
+Possible values:
+@table @samp
+@item sad
+sum of absolute differences, fast (default)
+@item sse
+sum of squared errors
+@item satd
+sum of absolute Hadamard transformed differences
+@item dct
+sum of absolute DCT transformed differences
+@item psnr
+sum of squared quantization errors (avoid, low quality)
+@item bit
+number of bits needed for the block
+@item rd
+rate distortion optimal, slow
+@item zero
+0
+@item vsad
+sum of absolute vertical differences
+@item vsse
+sum of squared vertical differences
+@item nsse
+noise preserving sum of squared differences
+@item w53
+5/3 wavelet, only used in snow
+@item w97
+9/7 wavelet, only used in snow
+@item dctmax
+
+@item chroma
+
+@end table
+
@item mblmin @var{integer} (@emph{encoding,video})
 Set min macroblock lagrange factor (VBR).

@item mblmax @var{integer} (@emph{encoding,video})
 Set max macroblock lagrange factor (VBR).

+@item mepc @var{integer} (@emph{encoding,video})
+Set motion estimation bitrate penalty compensation (1.0 = 256).
+
@item skip_loop_filter @var{integer} (@emph{decoding,video})
@item skip_idct        @var{integer} (@emph{decoding,video})
@item skip_frame       @var{integer} (@emph{decoding,video})
@@ -754,17 +858,31 @@ Default value is @samp{default}.
@item bidir_refine @var{integer} (@emph{encoding,video})
 Refine the two motion vectors used in bidirectional macroblocks.

+@item brd_scale @var{integer} (@emph{encoding,video})
+Downscale frames for dynamic B-frame decision.
+
@item keyint_min @var{integer} (@emph{encoding,video})
 Set minimum interval between IDR-frames.

@item refs @var{integer} (@emph{encoding,video})
 Set reference frames to consider for motion compensation.

+@item chromaoffset @var{integer} (@emph{encoding,video})
+Set chroma qp offset from luma.
+
@item trellis @var{integer} (@emph{encoding,audio,video})
 Set rate-distortion optimal quantization.

@item mv0_threshold @var{integer} (@emph{encoding,video})
+@item b_sensitivity @var{integer} (@emph{encoding,video})
+Adjust sensitivity of b_frame_strategy 1.
+
@item compression_level @var{integer} (@emph{encoding,audio,video})
+@item min_prediction_order @var{integer} (@emph{encoding,audio})
+@item max_prediction_order @var{integer} (@emph{encoding,audio})
+@item timecode_frame_start @var{integer} (@emph{encoding,video})
+Set GOP timecode frame start number, in non drop frame format.
+
@item bits_per_raw_sample @var{integer}
@item channel_layout @var{integer} (@emph{decoding/encoding,audio})

@@ -778,6 +896,7 @@ Possible values:
@end table
@item rc_max_vbv_use @var{float} (@emph{encoding,video})
@item rc_min_vbv_use @var{float} (@emph{encoding,video})
+@item ticks_per_frame @var{integer} (@emph{decoding/encoding,audio,video})

@item color_primaries @var{integer} (@emph{decoding/encoding,video})
 Possible values:
@@ -892,11 +1011,9 @@ Possible values:
@table @samp
@item tv
@item mpeg
-@item limited
 MPEG (219*2^(n-8))
@item pc
@item jpeg
-@item full
 JPEG (2^n-1)
@end table

--- a/doc/community.texi
+++ b/doc/community.texi
@@ -1,175 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8
-
-@settitle Community
-@titlepage
-@center @titlefont{Community}
-@end titlepage
-
-@top
-
-@contents
-
-@anchor{Organisation}
-@chapter Organisation
-
-The FFmpeg project is organized through a community working on global consensus.
-
-Decisions are taken by the ensemble of active members, through voting and are aided by two committees.
-
-@anchor{General Assembly}
-@chapter General Assembly
-
-The ensemble of active members is called the General Assembly (GA).
-
-The General Assembly is sovereign and legitimate for all its decisions regarding the FFmpeg project.
-
-The General Assembly is made up of active contributors.
-
-Contributors are considered "active contributors" if they have pushed more than 20 patches in the last 36 months in the main FFmpeg repository, or if they have been voted in by the GA.
-
-Additional members are added to the General Assembly through a vote after proposal by a member of the General Assembly. They are part of the GA for two years, after which they need a confirmation by the GA.
-
-A script to generate the current members of the general assembly (minus members voted in) can be found in `tools/general_assembly.pl`.
-
-@anchor{Voting}
-@chapter Voting
-
-Voting is done using a ranked voting system, currently running on https://vote.ffmpeg.org/ .
-
-Majority vote means more than 50% of the expressed ballots.
-
-@anchor{Technical Committee}
-@chapter Technical Committee
-
-The Technical Committee (TC) is here to arbitrate and make decisions when technical conflicts occur in the project. They will consider the merits of all the positions, judge them and make a decision.
-
-The TC resolves technical conflicts but is not a technical steering committee.
-
-Decisions by the TC are binding for all the contributors.
-
-Decisions made by the TC can be re-opened after 1 year or by a majority vote of the General Assembly, requested by one of the member of the GA.
-
-The TC is elected by the General Assembly for a duration of 1 year, and is composed of 5 members. Members can be re-elected if they wish. A majority vote in the General Assembly can trigger a new election of the TC.
-
-The members of the TC can be elected from outside of the GA. Candidates for election can either be suggested or self-nominated.
-
-The conflict resolution process is detailed in the resolution process document.
-
-The TC can be contacted at <tc@@ffmpeg>.
-
-@anchor{Resolution Process}
-@section Resolution Process
-
-The Technical Committee (TC) is here to arbitrate and make decisions when technical conflicts occur in the project.
-
-The TC main role is to resolve technical conflicts. It is therefore not a technical steering committee, but it is understood that some decisions might impact the future of the project.
-
-@subsection Seizing
-
-The TC can take possession of any technical matter that it sees fit.
-
-To involve the TC in a matter, email tc@ or CC them on an ongoing discussion.
-
-As members of TC are developers, they also can email tc@ to raise an issue.
-@subsection Announcement
-
-The TC, once seized, must announce itself on the main mailing list, with a [TC] tag.
-
-The TC has 2 modes of operation: a RFC one and an internal one.
-
-If the TC thinks it needs the input from the larger community, the TC can call for a RFC. Else, it can decide by itself.
-
-If the disagreement involves a member of the TC, that member should recuse themselves from the decision.
-
-The decision to use a RFC process or an internal discussion is a discretionary decision of the TC.
-
-The TC can also reject a seizure for a few reasons such as: the matter was not discussed enough previously; it lacks expertise to reach a beneficial decision on the matter; or the matter is too trivial.
-@subsection RFC call
-
-In the RFC mode, one person from the TC posts on the mailing list the technical question and will request input from the community.
-
-The mail will have the following specification:
-
-    a precise title
-    a specific tag [TC RFC]
-    a top-level email
-    contain a precise question that does not exceed 100 words and that is answerable by developers
-    may have an extra description, or a link to a previous discussion, if deemed necessary,
-    contain a precise end date for the answers.
-
-The answers from the community must be on the main mailing list and must have the following specification:
-
-    keep the tag and the title unchanged
-    limited to 400 words
-    a first-level, answering directly to the main email
-    answering to the question.
-
-Further replies to answers are permitted, as long as they conform to the community standards of politeness, they are limited to 100 words, and are not nested more than once. (max-depth=2)
-
-After the end-date, mails on the thread will be ignored.
-
-Violations of those rules will be escalated through the Community Committee.
-
-After all the emails are in, the TC has 96 hours to give its final decision. Exceptionally, the TC can request an extra delay, that will be notified on the mailing list.
-@subsection Within TC
-
-In the internal case, the TC has 96 hours to give its final decision. Exceptionally, the TC can request an extra delay.
-@subsection Decisions
-
-The decisions from the TC will be sent on the mailing list, with the [TC] tag.
-
-Internally, the TC should take decisions with a majority, or using ranked-choice voting.
-
-The decision from the TC should be published with a summary of the reasons that lead to this decision.
-
-The decisions from the TC are final, until the matters are reopened after no less than one year.
-
-@anchor{Community Committee}
-@chapter Community Committee
-
-The Community Committee (CC) is here to arbitrage and make decisions when inter-personal conflicts occur in the project. It will decide quickly and take actions, for the sake of the project.
-
-The CC can remove privileges of offending members, including removal of commit access and temporary ban from the community.
-
-Decisions made by the CC can be re-opened after 1 year or by a majority vote of the General Assembly. Indefinite bans from the community must be confirmed by the General Assembly, in a majority vote.
-
-The CC is elected by the General Assembly for a duration of 1 year, and is composed of 5 members. Members can be re-elected if they wish. A majority vote in the General Assembly can trigger a new election of the CC.
-
-The members of the CC can be elected from outside of the GA. Candidates for election can either be suggested or self-nominated.
-
-The CC is governed by and responsible for enforcing the Code of Conduct.
-
-The CC can be contacted at <cc@@ffmpeg>.
-
-@anchor{Code of Conduct}
-@chapter Code of Conduct
-
-Be friendly and respectful towards others and third parties.
-Treat others the way you yourself want to be treated.
-
-Be considerate. Not everyone shares the same viewpoint and priorities as you do.
-Different opinions and interpretations help the project.
-Looking at issues from a different perspective assists development.
-
-Do not assume malice for things that can be attributed to incompetence. Even if
-it is malice, it's rarely good to start with that as initial assumption.
-
-Stay friendly even if someone acts contrarily. Everyone has a bad day
-once in a while.
-If you yourself have a bad day or are angry then try to take a break and reply
-once you are calm and without anger if you have to.
-
-Try to help other team members and cooperate if you can.
-
-The goal of software development is to create technical excellence, not for any
-individual to be better and "win" against the others. Large software projects
-are only possible and successful through teamwork.
-
-If someone struggles do not put them down. Give them a helping hand
-instead and point them in the right direction.
-
-Finally, keep in mind the immortal words of Bill and Ted,
-"Be excellent to each other."
-
-@bye
--- a/doc/decoders.texi
+++ b/doc/decoders.texi
@@ -76,23 +76,13 @@ The following options are supported by the libdav1d wrapper.

@item framethreads
 Set amount of frame threads to use during decoding. The default value is 0 (autodetect).
-This option is deprecated for libdav1d >= 1.0 and will be removed in the future. Use the
-option @code{max_frame_delay} and the global option @code{threads} instead.

@item tilethreads
 Set amount of tile threads to use during decoding. The default value is 0 (autodetect).
-This option is deprecated for libdav1d >= 1.0 and will be removed in the future. Use the
-global option @code{threads} instead.
-
-@item max_frame_delay
-Set max amount of frames the decoder may buffer internally. 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.
-This option is deprecated and will be removed in the future. See the global option
-@code{export_side_data} to export Film Grain parameters instead of applying it.

@item oppoint
 Select an operating point of a scalable AV1 bitstream (0 - 31). Defaults to the
@@ -130,63 +120,6 @@ Set amount of frame threads to use during decoding. The default value is 0 (auto

@end table

-@section QSV Decoders
-
-The family of Intel QuickSync Video decoders (VC1, MPEG-2, H.264, HEVC,
-JPEG/MJPEG, VP8, VP9, AV1).
-
-@subsection Common Options
-
-The following options are supported by all qsv decoders.
-
-@table @option
-
-@item @var{async_depth}
-Internal parallelization depth, the higher the value the higher the latency.
-
-@item @var{gpu_copy}
-A GPU-accelerated copy between video and system memory
-@table @samp
-@item default
-@item on
-@item off
-@end table
-
-@end table
-
-@subsection HEVC Options
-Extra options for hevc_qsv.
-
-@table @option
-
-@item @var{load_plugin}
-A user plugin to load in an internal session
-@table @samp
-@item none
-@item hevc_sw
-@item hevc_hw
-@end table
-
-@item @var{load_plugins}
-A :-separate list of hexadecimal plugin UIDs to load in an internal session
-
-@end table
-
-@section v210
-
-Uncompressed 4:2:2 10-bit decoder.
-
-@subsection Options
-
-@table @option
-
-@item custom_stride
-Set the line size of the v210 data in bytes. The default value is 0
-(autodetect). You can use the special -1 value for a strideless v210 as seen in
-BOXX files.
-
-@end table
-
@c man end VIDEO DECODERS

@chapter Audio Decoders
@@ -353,156 +286,6 @@ Enabled by default.

@end table

-@section libaribcaption
-
-Yet another ARIB STD-B24 caption decoder using external @dfn{libaribcaption}
-library.
-
-Implements profiles A and C of the Japanse ARIB STD-B24 standard,
-Brazilian ABNT NBR 15606-1, and Philippines version of ISDB-T.
-
-Requires the presence of the libaribcaption headers and library
-(@url{https://github.com/xqq/libaribcaption}) during configuration.
-You need to explicitly configure the build with @code{--enable-libaribcaption}.
-If both @dfn{libaribb24} and @dfn{libaribcaption} are enabled, @dfn{libaribcaption}
-decoder precedes.
-
-@subsection libaribcaption Decoder Options
-
-@table @option
-
-@item -sub_type @var{subtitle_type}
-Specifies the format of the decoded subtitles.
-
-@table @samp
-@item bitmap
-Graphical image.
-@item ass
-ASS formatted text.
-@item text
-Simple text based output without formatting.
-@end table
-
-The default is @dfn{ass} as same as @dfn{libaribb24} decoder.
-Some present players (e.g., @dfn{mpv}) expect ASS format for ARIB caption.
-
-@item -caption_encoding @var{encoding_scheme}
-Specifies the encoding scheme of input subtitle text.
-
-@table @samp
-@item auto
-Automatically detect text encoding.
-@item jis
-8bit-char JIS encoding defined in ARIB STD B24.
-This encoding used in Japan for ISDB captions.
-@item utf8
-UTF-8 encoding defined in ARIB STD B24.
-This encoding is used in Philippines for ISDB-T captions.
-@item latin
-Latin character encoding defined in ABNT NBR 15606-1.
-This encoding is used in South America for SBTVD / ISDB-Tb captions.
-@end table
-
-The default is @dfn{ass} as same as @dfn{libaribb24} decoder.
-Some present players (e.g., @dfn{mpv}) expect ASS format for ARIB caption.
-
-@item -font @var{font_name[,font_name2,...]}
-Specify comma-separated list of font family names to be used for @dfn{bitmap}
-or @dfn{ass} type subtitle rendering.
-Only first font name is used for @dfn{ass} type subtitle.
-
-If not specified, use internaly defined default font family.
-
-@item -ass_single_rect @var{boolean}
-ARIB STD-B24 specifies that some captions may be displayed at different
-positions at a time (multi-rectangle subtitle).
-Since some players (e.g., old @dfn{mpv}) can't handle multiple ASS rectangles
-in a single AVSubtitle, or multiple ASS rectangles of indeterminate duration
-with the same start timestamp, this option can change the behavior so that
-all the texts are displayed in a single ASS rectangle.
-
-The default is @var{false}.
-
-If your player cannot handle AVSubtitles with multiple ASS rectangles properly,
-set this option to @var{true} or define @env{ASS_SINGLE_RECT=1} to change
-default behavior at compilation.
-
-@item -replace_fullwidth_ascii @var{boolean}
-Specify whether to replace MSZ (Middle Size, half width) fullwidth
-alphanumerics with halfwidth alphanumerics.
-
-The default is @var{true}.
-
-@item -force_outline_text @var{boolean}
-Specify whether always render outline text for all characters regardless of
-the indication by charactor style.
-
-The default is @var{false}.
-
-@item -outline_width @var{number} (0.0 - 3.0)
-Specify width for outline text, in dots (relative).
-
-The default is @var{1.5}.
-
-@item -ignore_background @var{boolean}
-Specify whether to ignore background color rendering.
-
-The default is @var{false}.
-
-@item -ignore_ruby @var{boolean}
-Specify whether to ignore rendering for ruby-like (furigana) characters.
-
-The default is @var{false}.
-
-@item -replace_drcs @var{boolean}
-Specify whether to render replaced DRCS characters as Unicode characters.
-
-The default is @var{true}.
-
-@item -canvas_size @var{image_size}
-Specify the resolution of the canvas to render subtitles to; usually, this
-should be frame size of input video.
-This only applies when @code{-subtitle_type} is set to @var{bitmap}.
-
-The libaribcaption decoder assumes input frame size for bitmap rendering as below:
-@enumerate
-@item
-PROFILE_A : 1440 x 1080 with SAR (PAR) 4:3
-@item
-PROFILE_C : 320 x 180 with SAR (PAR) 1:1
-@end enumerate
-
-If actual frame size of input video does not match above assumption,
-the rendered captions may be distorted.
-To make the captions undistorted, add @code{-canvas_size} option to specify
-actual input video size.
-
-Note that the @code{-canvas_size} option is not required for video with
-different size but same aspect ratio.
-In such cases, the caption will be stretched or shrunk to actual video size
-if @code{-canvas_size} option is not specified.
-If @code{-canvas_size} option is specified with different size,
-the caption will be stretched or shrunk as specified size with calculated SAR.
-
-@end table
-
-@subsection libaribcaption decoder usage examples
-
-Display MPEG-TS file with ARIB subtitle by @code{ffplay} tool:
-@example
-ffplay -sub_type bitmap MPEG.TS
-@end example
-
-Display MPEG-TS file with input frame size 1920x1080 by @code{ffplay} tool:
-@example
-ffplay -sub_type bitmap -canvas_size 1920x1080 MPEG.TS
-@end example
-
-Embed ARIB subtitle in transcoded video:
-@example
-ffmpeg -sub_type bitmap -i src.m2t -filter_complex "[0:v][0:s]overlay" -vcodec h264 dest.mp4
-@end example
-
@section dvbsub

@subsection Options
@@ -510,8 +293,6 @@ ffmpeg -sub_type bitmap -i src.m2t -filter_complex "[0:v][0:s]overlay" -vcodec h
@table @option
@item compute_clut
@table @option
-@item -2
-Compute clut once if no matching CLUT is in the stream.
@item -1
 Compute clut if no matching CLUT is in the stream.
@item 0
--- a/doc/demuxers.texi
+++ b/doc/demuxers.texi
@@ -25,13 +25,6 @@ Audible Format 2, 3, and 4 demuxer.

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

-@section aac
-
-Raw Audio Data Transport Stream AAC demuxer.
-
-This demuxer is used to demux an ADTS input containing a single AAC stream
-alongwith any ID3v1/2 or APE tags in it.
-
@section apng

 Animated Portable Network Graphics demuxer.
@@ -44,15 +37,12 @@ between the last fcTL and IEND chunks.

@table @option
@item -ignore_loop @var{bool}
-Ignore the loop variable in the file if set. Default is enabled.
-
+Ignore the loop variable in the file if set.
@item -max_fps @var{int}
-Maximum framerate in frames per second. Default of 0 imposes no limit.
-
+Maximum framerate in frames per second (0 for no limit).
@item -default_fps @var{int}
 Default framerate in frames per second when none is specified in the file
-(0 meaning as fast as possible). Default is 15.
-
+(0 meaning as fast as possible).
@end table

@section asf
@@ -103,7 +93,8 @@ backslash or single quotes.
 All subsequent file-related directives apply to that file.

@item @code{ffconcat version 1.0}
-Identify the script type and version.
+Identify the script type and version. It also sets the @option{safe} option
+to 1 if it was -1.

 To make FFmpeg recognize the format automatically, this directive must
 appear exactly as is (no extra space or byte-order-mark) on the very first
@@ -157,16 +148,6 @@ directive) will be reduced based on their specified Out point.
 Metadata of the packets of the file. The specified metadata will be set for
 each file packet. You can specify this directive multiple times to add multiple
 metadata entries.
-This directive is deprecated, use @code{file_packet_meta} instead.
-
-@item @code{file_packet_meta @var{key} @var{value}}
-Metadata of the packets of the file. The specified metadata will be set for
-each file packet. You can specify this directive multiple times to add multiple
-metadata entries.
-
-@item @code{option @var{key} @var{value}}
-Option to access, open and probe the file.
-Can be present multiple times.

@item @code{stream}
 Introduce a stream in the virtual file.
@@ -184,20 +165,6 @@ subfiles will be used.
 This is especially useful for MPEG-PS (VOB) files, where the order of the
 streams is not reliable.

-@item @code{stream_meta @var{key} @var{value}}
-Metadata for the stream.
-Can be present multiple times.
-
-@item @code{stream_codec @var{value}}
-Codec for the stream.
-
-@item @code{stream_extradata @var{hex_string}}
-Extradata for the string, encoded in hexadecimal.
-
-@item @code{chapter @var{id} @var{start} @var{end}}
-Add a chapter. @var{id} is an unique identifier, possibly small and
-consecutive.
-
@end table

@subsection Options
@@ -207,8 +174,7 @@ This demuxer accepts the following option:
@table @option

@item safe
-If set to 1, reject unsafe file paths and directives.
-A file path is considered safe if it
+If set to 1, reject unsafe file paths. A file path is considered safe if it
 does not contain a protocol specification and is relative and all components
 only contain characters from the portable character set (letters, digits,
 period, underscore and hyphen) and have no period at the beginning of a
@@ -218,6 +184,9 @@ If set to 0, any file name is accepted.

 The default is 1.

+-1 is equivalent to 1 if the format was automatically
+probed and 0 otherwise.
+
@item auto_convert
 If set to 1, try to perform automatic conversions on packet data to make the
 streams concatenable.
@@ -274,55 +243,11 @@ which streams to actually receive.
 Each stream mirrors the @code{id} and @code{bandwidth} properties from the
@code{<Representation>} as metadata keys named "id" and "variant_bitrate" respectively.

-@subsection Options
-
-This demuxer accepts the following option:
-
-@table @option
-
-@item cenc_decryption_key
-16-byte key, in hex, to decrypt files encrypted using ISO Common Encryption (CENC/AES-128 CTR; ISO/IEC 23001-7).
-
-@end table
-
-@section ea
-
-Electronic Arts Multimedia format demuxer.
-
-This format is used by various Electronic Arts games.
-
-@subsection Options
-
-@table @option
-
-@item merge_alpha @var{bool}
-
-Normally the VP6 alpha channel (if exists) is returned as a secondary video
-stream, by setting this option you can make the demuxer return a single video
-stream which contains the alpha channel in addition to the ordinary video.
-
-@end table
-
-@section imf
-
-Interoperable Master Format demuxer.
-
-This demuxer presents audio and video streams found in an IMF Composition, as
-specified in @url{https://doi.org/10.5594/SMPTE.ST2067-2.2020, SMPTE ST 2067-2}.
-
-@example
-ffmpeg [-assetmaps <path of ASSETMAP1>,<path of ASSETMAP2>,...] -i <path of CPL> ...
-@end example
-
-If @code{-assetmaps} is not specified, the demuxer looks for a file called
-@file{ASSETMAP.xml} in the same directory as the CPL.
-
-@section flv, live_flv, kux
+@section flv, live_flv

 Adobe Flash Video Format demuxer.

 This demuxer is used to demux FLV files and RTMP network streams. In case of live network streams, if you force format, you may use live_flv option instead of flv to survive timestamp discontinuities.
-KUX is a flv variant used on the Youku platform.

@example
 ffmpeg -f flv -i myfile.flv ...
@@ -399,12 +324,16 @@ It accepts the following options:
@item live_start_index
 segment index to start live streams at (negative values are from the end).

-@item prefer_x_start
-prefer to use #EXT-X-START if it's in playlist instead of live_start_index.
-
@item allowed_extensions
 ',' separated list of file extensions that hls is allowed to access.

+@item extension_picky
+This blocks disallowed extensions from probing
+It also requires all available segments to have matching extensions to the format
+except mpegts, which is always allowed.
+It is recommended to set the whitelists correctly instead of depending on extensions
+Enabled by default.
+
@item max_reload
 Maximum number of times a insufficient list is attempted to be reloaded.
 Default value is 1000.
@@ -424,13 +353,6 @@ Enabled by default for HTTP/1.1 servers.
@item http_seekable
 Use HTTP partial requests for downloading HTTP segments.
 0 = disable, 1 = enable, -1 = auto, Default is auto.
-
-@item seg_format_options
-Set options for the demuxer of media segments using a list of key=value pairs separated by @code{:}.
-
-@item seg_max_retry
-Maximum number of times to reload a segment on error, useful when segment skip on network error is not desired.
-Default value is 0.
@end table

@section image2
@@ -746,12 +668,6 @@ Set mfra timestamps as PTS
 Don't use mfra box to set timestamps
@end table

-@item use_tfdt
-For fragmented input, set fragment's starting timestamp to @code{baseMediaDecodeTime} from the @code{tfdt} box.
-Default is enabled, which will prefer to use the @code{tfdt} box to set DTS. Disable to use the @code{earliest_presentation_time} from the @code{sidx} box.
-In either case, the timestamp from the @code{mfra} box will be used if it's available and @code{use_mfra_for} is
-set to pts or dts.
-
@item export_all
 Export unrecognized boxes within the @var{udta} box as metadata entries. The first four
 characters of the box type are set as the key. Default is false.
@@ -770,22 +686,6 @@ specify.

@item decryption_key
 16-byte key, in hex, to decrypt files encrypted using ISO Common Encryption (CENC/AES-128 CTR; ISO/IEC 23001-7).
-
-@item max_stts_delta
-Very high sample deltas written in a trak's stts box may occasionally be intended but usually they are written in
-error or used to store a negative value for dts correction when treated as signed 32-bit integers. This option lets
-the user set an upper limit, beyond which the delta is clamped to 1. Values greater than the limit if negative when
-cast to int32 are used to adjust onward dts.
-
-Unit is the track time scale. Range is 0 to UINT_MAX. Default is @code{UINT_MAX - 48000*10} which allows upto
-a 10 second dts correction for 48 kHz audio streams while accommodating 99.9% of @code{uint32} range.
-
-@item interleaved_read
-Interleave packets from multiple tracks at demuxer level. For badly interleaved files, this prevents playback issues
-caused by large gaps between packets in different tracks, as MOV/MP4 do not have packet placement requirements.
-However, this can cause excessive seeking on very badly interleaved files, due to seeking between tracks, so disabling
-it may prevent I/O issues, at the expense of playback.
-
@end table

@subsection Audible AAX
@@ -826,10 +726,6 @@ disabled). Default value is -1.
@item merge_pmt_versions
 Re-use existing streams when a PMT's version is updated and elementary
 streams move to different PIDs. Default value is 0.
-
-@item max_packet_size
-Set maximum size, in bytes, of packet emitted by the demuxer. Payloads above this size
-are split across multiple packets. Range is 1 to INT_MAX/2. Default is 204800 bytes.
@end table

@section mpjpeg
--- a/doc/dev_community/community.md
+++ b/doc/dev_community/community.md
@@ -0,0 +1,79 @@
+# FFmpeg project
+
+## Organisation
+
+The FFmpeg project is organized through a community working on global consensus.
+
+Decisions are taken by the ensemble of active members, through voting and
+are aided by two committees.
+
+## General Assembly
+
+The ensemble of active members is called the General Assembly (GA).
+
+The General Assembly is sovereign and legitimate for all its decisions
+regarding the FFmpeg project.
+
+The General Assembly is made up of active contributors.
+
+Contributors are considered "active contributors" if they have pushed more
+than 20 patches in the last 36 months in the main FFmpeg repository, or
+if they have been voted in by the GA.
+
+Additional members are added to the General Assembly through a vote after
+proposal by a member of the General Assembly.
+They are part of the GA for two years, after which they need a confirmation by
+the GA.
+
+## Voting
+
+Voting is done using a ranked voting system, currently running on https://vote.ffmpeg.org/ .
+
+Majority vote means more than 50% of the expressed ballots.
+
+## Technical Committee
+
+The Technical Committee (TC) is here to arbitrate and make decisions when
+technical conflicts occur in the project.
+They will consider the merits of all the positions, judge them and make a
+decision.
+
+The TC resolves technical conflicts but is not a technical steering committee.
+
+Decisions by the TC are binding for all the contributors.
+
+Decisions made by the TC can be re-opened after 1 year or by a majority vote
+of the General Assembly, requested by one of the member of the GA.
+
+The TC is elected by the General Assembly for a duration of 1 year, and
+is composed of 5 members.
+Members can be re-elected if they wish. A majority vote in the General Assembly
+can trigger a new election of the TC.
+
+The members of the TC can be elected from outside of the GA.
+Candidates for election can either be suggested or self-nominated.
+
+The conflict resolution process is detailed in the [resolution process](resolution_process.md) document.
+
+## Community committee
+
+The Community Committee (CC) is here to arbitrage and make decisions when
+inter-personal conflicts occur in the project. It will decide quickly and
+take actions, for the sake of the project.
+
+The CC can remove privileges of offending members, including removal of
+commit access and temporary ban from the community.
+
+Decisions made by the CC can be re-opened after 1 year or by a majority vote
+of the General Assembly. Indefinite bans from the community must be confirmed
+by the General Assembly, in a majority vote.
+
+The CC is elected by the General Assembly for a duration of 1 year, and is
+composed of 5 members.
+Members can be re-elected if they wish. A majority vote in the General Assembly
+can trigger a new election of the CC.
+
+The members of the CC can be elected from outside of the GA.
+Candidates for election can either be suggested or self-nominated.
+
+The CC is governed by and responsible for enforcing the Code of Conduct.
--- a/doc/dev_community/resolution_process.md
+++ b/doc/dev_community/resolution_process.md
@@ -0,0 +1,91 @@
+# Technical Committee
+
+_This document only makes sense with the rules from [the community document](community)_.
+
+The Technical Committee (**TC**) is here to arbitrate and make decisions when
+technical conflicts occur in the project.
+
+The TC main role is to resolve technical conflicts.
+It is therefore not a technical steering committee, but it is understood that
+some decisions might impact the future of the project.
+
+# Process
+
+## Seizing
+
+The TC can take possession of any technical matter that it sees fit.
+
+To involve the TC in a matter, email tc@ or CC them on an ongoing discussion.
+
+As members of TC are developers, they also can email tc@ to raise an issue.
+
+## Announcement
+
+The TC, once seized, must announce itself on the main mailing list, with a _[TC]_ tag.
+
+The TC has 2 modes of operation: a RFC one and an internal one.
+
+If the TC thinks it needs the input from the larger community, the TC can call
+for a RFC. Else, it can decide by itself.
+
+If the disagreement involves a member of the TC, that member should recuse
+themselves from the decision.
+
+The decision to use a RFC process or an internal discussion is a discretionary
+decision of the TC.
+
+The TC can also reject a seizure for a few reasons such as:
+the matter was not discussed enough previously; it lacks expertise to reach a
+beneficial decision on the matter; or the matter is too trivial.
+
+### RFC call
+
+In the RFC mode, one person from the TC posts on the mailing list the
+technical question and will request input from the community.
+
+The mail will have the following specification:
+* a precise title
+* a specific tag [TC RFC]
+* a top-level email
+* contain a precise question that does not exceed 100 words and that is answerable by developers
+* may have an extra description, or a link to a previous discussion, if deemed necessary,
+* contain a precise end date for the answers.
+
+The answers from the community must be on the main mailing list and must have
+the following specification:
+* keep the tag and the title unchanged
+* limited to 400 words
+* a first-level, answering directly to the main email
+* answering to the question.
+
+Further replies to answers are permitted, as long as they conform to the
+community standards of politeness, they are limited to 100 words, and are not
+nested more than once. (max-depth=2)
+
+After the end-date, mails on the thread will be ignored.
+
+Violations of those rules will be escalated through the Community Committee.
+
+After all the emails are in, the TC has 96 hours to give its final decision.
+Exceptionally, the TC can request an extra delay, that will be notified on the
+mailing list.
+
+### Within TC
+
+In the internal case, the TC has 96 hours to give its final decision.
+Exceptionally, the TC can request an extra delay.
+
+
+## Decisions
+
+The decisions from the TC will be sent on the mailing list, with the _[TC]_ tag.
+
+Internally, the TC should take decisions with a majority, or using
+ranked-choice voting.
+
+The decision from the TC should be published with a summary of the reasons that
+lead to this decision.
+
+The decisions from the TC are final, until the matters are reopened after
+no less than one year.
+
--- a/doc/developer.texi
+++ b/doc/developer.texi
@@ -10,115 +10,41 @@

@contents

-@chapter Introduction
+@chapter Notes for external developers

-This text is concerned with the development @emph{of} FFmpeg itself. Information
-on using the FFmpeg libraries in other programs can be found elsewhere, e.g. in:
-@itemize @bullet
-@item
-the installed header files
-@item
-@url{http://ffmpeg.org/doxygen/trunk/index.html, the Doxygen documentation}
-generated from the headers
-@item
-the examples under @file{doc/examples}
-@end itemize
+This document is mostly useful for internal FFmpeg developers.
+External developers who need to use the API in their application should
+refer to the API doxygen documentation in the public headers, and
+check the examples in @file{doc/examples} and in the source code to
+see how the public API is employed.
+
+You can use the FFmpeg libraries in your commercial program, but you
+are encouraged to @emph{publish any patch you make}. In this case the
+best way to proceed is to send your patches to the ffmpeg-devel
+mailing list following the guidelines illustrated in the remainder of
+this document.

 For more detailed legal information about the use of FFmpeg in
 external programs read the @file{LICENSE} file in the source tree and
 consult @url{https://ffmpeg.org/legal.html}.

-If you modify FFmpeg code for your own use case, you are highly encouraged to
-@emph{submit your changes back to us}, using this document as a guide. There are
-both pragmatic and ideological reasons to do so:
+@chapter Contributing
+
+There are 2 ways by which code gets into FFmpeg:
@itemize @bullet
-@item
-Maintaining external changes to keep up with upstream development is
-time-consuming and error-prone. With your code in the main tree, it will be
-maintained by FFmpeg developers.
-@item
-FFmpeg developers include leading experts in the field who can find bugs or
-design flaws in your code.
-@item
-By supporting the project you find useful you ensure it continues to be
-maintained and developed.
+@item Submitting patches to the ffmpeg-devel mailing list.
+      See @ref{Submitting patches} for details.
+@item Directly committing changes to the main tree.
@end itemize

-All proposed code changes should be submitted for review to
-@url{mailto:ffmpeg-devel@@ffmpeg.org, the development mailing list}, as
-described in more detail in the @ref{Submitting patches} chapter. The code
-should comply with the @ref{Development Policy} and follow the @ref{Coding Rules}.
+Whichever way, changes should be reviewed by the maintainer of the code
+before they are committed. And they should follow the @ref{Coding Rules}.
 The developer making the commit and the author are responsible for their changes
 and should try to fix issues their commit causes.

@anchor{Coding Rules}
@chapter Coding Rules

-@section Language
-
-FFmpeg is mainly programmed in the ISO C99 language, extended with:
-@itemize @bullet
-@item
-Atomic operations from C11 @file{stdatomic.h}. They are emulated on
-architectures/compilers that do not support them, so all FFmpeg-internal code
-may use atomics without any extra checks. However, @file{stdatomic.h} must not
-be included in public headers, so they stay C99-compatible.
-@end itemize
-
-Compiler-specific extensions may be used with good reason, but must not be
-depended on, i.e. the code must still compile and work with compilers lacking
-the extension.
-
-The following C99 features must not be used anywhere in the codebase:
-@itemize @bullet
-@item
-variable-length arrays;
-
-@item
-complex numbers;
-
-@item
-mixed statements and declarations.
-@end itemize
-
-@subsection SIMD/DSP
-@anchor{SIMD/DSP}
-
-As modern compilers are unable to generate efficient SIMD or other
-performance-critical DSP code from plain C, handwritten assembly is used.
-Usually such code is isolated in a separate function. Then the standard approach
-is writing multiple versions of this function – a plain C one that works
-everywhere and may also be useful for debugging, and potentially multiple
-architecture-specific optimized implementations. Initialization code then
-chooses the best available version at runtime and loads it into a function
-pointer; the function in question is then always called through this pointer.
-
-The specific syntax used for writing assembly is:
-@itemize @bullet
-@item
-NASM on x86;
-
-@item
-GAS on ARM.
-@end itemize
-
-A unit testing framework for assembly called @code{checkasm} lives under
-@file{tests/checkasm}. All new assembly should come with @code{checkasm} tests;
-adding tests for existing assembly that lacks them is also strongly encouraged.
-
-@subsection Other languages
-
-Other languages than C may be used in special cases:
-@itemize @bullet
-@item
-Compiler intrinsics or inline assembly when the code in question cannot be
-written in the standard way described in the @ref{SIMD/DSP} section. This
-typically applies to code that needs to be inlined.
-
-@item
-Objective-C where required for interacting with macOS-specific interfaces.
-@end itemize
-
@section Code formatting conventions

 There are the following guidelines regarding the indentation in files:
@@ -141,39 +67,8 @@ K&R coding style is used.
@end itemize
 The presentation is one inspired by 'indent -i4 -kr -nut'.

-@subsection Vim configuration
-In order to configure Vim to follow FFmpeg formatting conventions, paste
-the following snippet into your @file{.vimrc}:
-@example
-" indentation rules for FFmpeg: 4 spaces, no tabs
-set expandtab
-set shiftwidth=4
-set softtabstop=4
-set cindent
-set cinoptions=(0
-" Allow tabs in Makefiles.
-autocmd FileType make,automake set noexpandtab shiftwidth=8 softtabstop=8
-" Trailing whitespace and tabs are forbidden, so highlight them.
-highlight ForbiddenWhitespace ctermbg=red guibg=red
-match ForbiddenWhitespace /\s\+$\|\t/
-" Do not highlight spaces at the end of line while typing on that line.
-autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@@<!$/
-@end example
-
-@subsection Emacs configuration
-For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
-@lisp
-(c-add-style "ffmpeg"
-             '("k&r"
-               (c-basic-offset . 4)
-               (indent-tabs-mode . nil)
-               (show-trailing-whitespace . t)
-               (c-offsets-alist
-                (statement-cont . (c-lineup-assignments +)))
-               )
-             )
-(setq c-default-style "ffmpeg")
-@end lisp
+The main priority in FFmpeg is simplicity and small code size in order to
+minimize the bug count.

@section Comments
 Use the JavaDoc/Doxygen  format (see examples below) so that code documentation
@@ -215,52 +110,92 @@ int myfunc(int my_parameter)
 ...
@end example

-@anchor{Naming conventions}
-@section Naming conventions
+@section C language features

-Names of functions, variables, and struct members must be lowercase, using
-underscores (_) to separate words. For example, @samp{avfilter_get_video_buffer}
-is an acceptable function name and @samp{AVFilterGetVideo} is not.
+FFmpeg is programmed in the ISO C90 language with a few additional
+features from ISO C99, namely:

-Struct, union, enum, and typedeffed type names must use CamelCase. All structs
-and unions should be typedeffed to the same name as the struct/union tag, e.g.
-@code{typedef struct AVFoo @{ ... @} AVFoo;}. Enums are typically not
-typedeffed.
-
-Enumeration constants and macros must be UPPERCASE, except for macros
-masquerading as functions, which should use the function naming convention.
-
-All identifiers in the libraries should be namespaced as follows:
@itemize @bullet
@item
-No namespacing for identifiers with file and lower scope (e.g. local variables,
-static functions), and struct and union members,
+the @samp{inline} keyword;

@item
-The @code{ff_} prefix must be used for variables and functions visible outside
-of file scope, but only used internally within a single library, e.g.
-@samp{ff_w64_demuxer}. This prevents name collisions when FFmpeg is statically
-linked.
+@samp{//} comments;
+
+@item
+designated struct initializers (@samp{struct s x = @{ .i = 17 @};});
+
+@item
+compound literals (@samp{x = (struct s) @{ 17, 23 @};}).
+
+@item
+for loops with variable definition (@samp{for (int i = 0; i < 8; i++)});
+
+@item
+Variadic macros (@samp{#define ARRAY(nb, ...) (int[nb + 1])@{ nb, __VA_ARGS__ @}});
+
+@item
+Implementation defined behavior for signed integers is assumed to match the
+expected behavior for two's complement. Non representable values in integer
+casts are binary truncated. Shift right of signed values uses sign extension.
+@end itemize
+
+These features are supported by all compilers we care about, so we will not
+accept patches to remove their use unless they absolutely do not impair
+clarity and performance.
+
+All code must compile with recent versions of GCC and a number of other
+currently supported compilers. To ensure compatibility, please do not use
+additional C99 features or GCC extensions. Especially watch out for:
+
+@itemize @bullet
+@item
+mixing statements and declarations;
+
+@item
+@samp{long long} (use @samp{int64_t} instead);
+
+@item
+@samp{__attribute__} not protected by @samp{#ifdef __GNUC__} or similar;
+
+@item
+GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}).
+@end itemize
+
+@section Naming conventions
+All names should be composed with underscores (_), not CamelCase. For example,
+@samp{avfilter_get_video_buffer} is an acceptable function name and
+@samp{AVFilterGetVideo} is not. The exception from this are type names, like
+for example structs and enums; they should always be in CamelCase.
+
+There are the following conventions for naming variables and functions:
+
+@itemize @bullet
+@item
+For local variables no prefix is required.
+
+@item
+For file-scope variables and functions declared as @code{static}, no prefix
+is required.
+
+@item
+For variables and functions visible outside of file scope, but only used
+internally by a library, an @code{ff_} prefix should be used,
+e.g. @samp{ff_w64_demuxer}.

@item
 For variables and functions visible outside of file scope, used internally
 across multiple libraries, use @code{avpriv_} as prefix, for example,
@samp{avpriv_report_missing_feature}.

-@item
-All other internal identifiers, like private type or macro names, should be
-namespaced only to avoid possible internal conflicts. E.g. @code{H264_NAL_SPS}
-vs. @code{HEVC_NAL_SPS}.
-
@item
 Each library has its own prefix for public symbols, in addition to the
 commonly used @code{av_} (@code{avformat_} for libavformat,
@code{avcodec_} for libavcodec, @code{swr_} for libswresample, etc).
 Check the existing code and choose names accordingly.
-
-@item
-Other public identifiers (struct, union, enum, macro, type names) must use their
-library's public prefix (@code{AV}, @code{Sws}, or @code{Swr}).
+Note that some symbols without these prefixes are also exported for
+retro-compatibility reasons. These exceptions are declared in the
+@code{lib<name>/lib<name>.v} files.
@end itemize

 Furthermore, name space reserved for the system should not be invaded.
@@ -274,50 +209,50 @@ symbols. If in doubt, just avoid names starting with @code{_} altogether.
@section Miscellaneous conventions

@itemize @bullet
+@item
+fprintf and printf are forbidden in libavformat and libavcodec,
+please use av_log() instead.
+
@item
 Casts should be used only when necessary. Unneeded parentheses
 should also be avoided if they don't make the code easier to understand.
@end itemize

-@anchor{Development Policy}
+@section Editor configuration
+In order to configure Vim to follow FFmpeg formatting conventions, paste
+the following snippet into your @file{.vimrc}:
+@example
+" indentation rules for FFmpeg: 4 spaces, no tabs
+set expandtab
+set shiftwidth=4
+set softtabstop=4
+set cindent
+set cinoptions=(0
+" Allow tabs in Makefiles.
+autocmd FileType make,automake set noexpandtab shiftwidth=8 softtabstop=8
+" Trailing whitespace and tabs are forbidden, so highlight them.
+highlight ForbiddenWhitespace ctermbg=red guibg=red
+match ForbiddenWhitespace /\s\+$\|\t/
+" Do not highlight spaces at the end of line while typing on that line.
+autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@@<!$/
+@end example
+
+For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
+@lisp
+(c-add-style "ffmpeg"
+             '("k&r"
+               (c-basic-offset . 4)
+               (indent-tabs-mode . nil)
+               (show-trailing-whitespace . t)
+               (c-offsets-alist
+                (statement-cont . (c-lineup-assignments +)))
+               )
+             )
+(setq c-default-style "ffmpeg")
+@end lisp
+
@chapter Development Policy

-@section Code behaviour
-
-@subheading Correctness
-The code must be valid. It must not crash, abort, access invalid pointers, leak
-memory, cause data races or signed integer overflow, or otherwise cause
-undefined behaviour. Error codes should be checked and, when applicable,
-forwarded to the caller.
-
-@subheading Thread- and library-safety
-Our libraries may be called by multiple independent callers in the same process.
-These calls may happen from any number of threads and the different call sites
-may not be aware of each other - e.g. a user program may be calling our
-libraries directly, and use one or more libraries that also call our libraries.
-The code must behave correctly under such conditions.
-
-@subheading Robustness
-The code must treat as untrusted any bytestream received from a caller or read
-from a file, network, etc. It must not misbehave when arbitrary data is sent to
-it - typically it should print an error message and return
-@code{AVERROR_INVALIDDATA} on encountering invalid input data.
-
-@subheading Memory allocation
-The code must use the @code{av_malloc()} family of functions from
-@file{libavutil/mem.h} to perform all memory allocation, except in special cases
-(e.g. when interacting with an external library that requires a specific
-allocator to be used).
-
-All allocations should be checked and @code{AVERROR(ENOMEM)} returned on
-failure. A common mistake is that error paths leak memory - make sure that does
-not happen.
-
-@subheading stdio
-Our libraries must not access the stdio streams stdin/stdout/stderr directly
-(e.g. via @code{printf()} family of functions), as that is not library-safe. For
-logging, use @code{av_log()}.
-
@section Patches/Committing
@subheading Licenses for patches must be compatible with FFmpeg.
 Contributions should be licensed under the
@@ -340,24 +275,13 @@ missing samples or an implementation with a small subset of features.
 Always check the mailing list for any reviewers with issues and test
 FATE before you push.

-@subheading Commit messages
-Commit messages are highly important tools for informing other developers on
-what a given change does and why. Every commit must always have a properly
-filled out commit message with the following format:
-@example
-area changed: short 1 line description
-
-details describing what and why and giving references.
-@end example
-
-If the commit addresses a known bug on our bug tracker or other external issue
-(e.g. CVE), the commit message should include the relevant bug ID(s) or other
-external identifiers. Note that this should be done in addition to a proper
-explanation and not instead of it. Comments such as "fixed!" or "Changed it."
-are not acceptable.
-
-When applying patches that have been discussed at length on the mailing list,
-reference the thread in the commit message.
+@subheading Keep the main commit message short with an extended description below.
+The commit message should have a short first line in the form of
+a @samp{topic: short description} as a header, separated by a newline
+from the body consisting of an explanation of why the change is necessary.
+If the commit fixes a known bug on the bug tracker, the commit message
+should include its bug ID. Referring to the issue on the bug tracker does
+not exempt you from writing an excerpt of the bug in the commit message.

@subheading Testing must be adequate but not excessive.
 If it works for you, others, and passes FATE then it should be OK to commit
@@ -376,6 +300,15 @@ later on.
 Also if you have doubts about splitting or not splitting, do not hesitate to
 ask/discuss it on the developer mailing list.

+@subheading Ask before you change the build system (configure, etc).
+Do not commit changes to the build system (Makefiles, configure script)
+which change behavior, defaults etc, without asking first. The same
+applies to compiler warning fixes, trivial looking fixes and to code
+maintained by other developers. We usually have a reason for doing things
+the way we do. Send your changes as patches to the ffmpeg-devel mailing
+list, and if the code maintainers say OK, you may commit. This does not
+apply to files you wrote and/or maintain.
+
@subheading Cosmetic changes should be kept in separate patches.
 We refuse source indentation and other cosmetic changes if they are mixed
 with functional changes, such commits will be rejected and removed. Every
@@ -390,12 +323,28 @@ NOTE: If you had to put if()@{ .. @} over a large (> 5 lines) chunk of code,
 then either do NOT change the indentation of the inner part within (do not
 move it to the right)! or do so in a separate commit

+@subheading Commit messages should always be filled out properly.
+Always fill out the commit log message. Describe in a few lines what you
+changed and why. You can refer to mailing list postings if you fix a
+particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.
+Recommended format:
+
+@example
+area changed: Short 1 line description
+
+details describing what and why and giving references.
+@end example
+
@subheading Credit the author of the patch.
 Make sure the author of the commit is set correctly. (see git commit --author)
 If you apply a patch, send an
 answer to ffmpeg-devel (or wherever you got the patch from) saying that
 you applied the patch.

+@subheading Complex patches should refer to discussion surrounding them.
+When applying patches that have been discussed (at length) on the mailing
+list, reference the thread in the log message.
+
@subheading Always wait long enough before pushing changes
 Do NOT commit to code actively maintained by others without permission.
 Send a patch to ffmpeg-devel. If no one answers within a reasonable
@@ -404,6 +353,22 @@ time-frame (12h for build failures and security fixes, 3 days small changes,
 Also note, the maintainer can simply ask for more time to review!

@section Code
+@subheading API/ABI changes should be discussed before they are made.
+Do not change behavior of the programs (renaming options etc) or public
+API or ABI without first discussing it on the ffmpeg-devel mailing list.
+Do not remove widely used functionality or features (redundant code can be removed).
+
+@subheading Remember to check if you need to bump versions for libav*.
+Depending on the change, you may need to change the version integer.
+Incrementing the first component means no backward compatibility to
+previous versions (e.g. removal of a function from the public API).
+Incrementing the second component means backward compatible change
+(e.g. addition of a function to the public API or extension of an
+existing data structure).
+Incrementing the third component means a noteworthy binary compatible
+change (e.g. encoder bug fix that matters for the decoder). The third
+component always starts at 100 to distinguish FFmpeg from Libav.
+
@subheading Warnings for correct code may be disabled if there is no other option.
 Compiler warnings indicate potential bugs or code with bad style. If a type of
 warning always points to correct and clean code, that warning should
@@ -413,150 +378,10 @@ If it is a bug, the bug has to be fixed. If it is not, the code should
 be changed to not generate a warning unless that causes a slowdown
 or obfuscates the code.

-@section Library public interfaces
-Every library in FFmpeg provides a set of public APIs in its installed headers,
-which are those listed in the variable @code{HEADERS} in that library's
-@file{Makefile}. All identifiers defined in those headers (except for those
-explicitly documented otherwise), and corresponding symbols exported from
-compiled shared or static libraries are considered public interfaces and must
-comply with the API and ABI compatibility rules described in this section.
-
-Public APIs must be backward compatible within a given major version. I.e. any
-valid user code that compiles and works with a given library version must still
-compile and work with any later version, as long as the major version number is
-unchanged. "Valid user code" here means code that is calling our APIs in a
-documented and/or intended manner and is not relying on any undefined behavior.
-Incrementing the major version may break backward compatibility, but only to the
-extent described in @ref{Major version bumps}.
-
-We also guarantee backward ABI compatibility for shared and static libraries.
-I.e. it should be possible to replace a shared or static build of our library
-with a build of any later version (re-linking the user binary in the static
-case) without breaking any valid user binaries, as long as the major version
-number remains unchanged.
-
-@subsection Adding new interfaces
-Any new public identifiers in installed headers are considered new API - this
-includes new functions, structs, macros, enum values, typedefs, new fields in
-existing structs, new installed headers, etc. Consider the following
-guidelines when adding new APIs.
-
-@subsubheading Motivation
-While new APIs can be added relatively easily, changing or removing them is much
-harder due to abovementioned compatibility requirements. You should then
-consider carefully whether the functionality you are adding really needs to be
-exposed to our callers as new public API.
-
-Your new API should have at least one well-established use case outside of the
-library that cannot be easily achieved with existing APIs. Every library in
-FFmpeg also has a defined scope - your new API must fit within it.
-
-@subsubheading Replacing existing APIs
-If your new API is replacing an existing one, it should be strictly superior to
-it, so that the advantages of using the new API outweight the cost to the
-callers of changing their code. After adding the new API you should then
-deprecate the old one and schedule it for removal, as described in
-@ref{Removing interfaces}.
-
-If you deem an existing API deficient and want to fix it, the preferred approach
-in most cases is to add a differently-named replacement and deprecate the
-existing API rather than modify it. It is important to make the changes visible
-to our callers (e.g. through compile- or run-time deprecation warnings) and make
-it clear how to transition to the new API (e.g. in the Doxygen documentation or
-on the wiki).
-
-@subsubheading API design
-The FFmpeg libraries are used by a variety of callers to perform a wide range of
-multimedia-related processing tasks. You should therefore - within reason - try
-to design your new API for the broadest feasible set of use cases and avoid
-unnecessarily limiting it to a specific type of callers (e.g. just media
-playback or just transcoding).
-
-@subsubheading Consistency
-Check whether similar APIs already exist in FFmpeg. If they do, try to model
-your new addition on them to achieve better overall consistency.
-
-The naming of your new identifiers should follow the @ref{Naming conventions}
-and be aligned with other similar APIs, if applicable.
-
-@subsubheading Extensibility
-You should also consider how your API might be extended in the future in a
-backward-compatible way. If you are adding a new struct @code{AVFoo}, the
-standard approach is requiring the caller to always allocate it through a
-constructor function, typically named @code{av_foo_alloc()}. This way new fields
-may be added to the end of the struct without breaking ABI compatibility.
-Typically you will also want a destructor - @code{av_foo_free(AVFoo**)} that
-frees the indirectly supplied object (and its contents, if applicable) and
-writes @code{NULL} to the supplied pointer, thus eliminating the potential
-dangling pointer in the caller's memory.
-
-If you are adding new functions, consider whether it might be desirable to tweak
-their behavior in the future - you may want to add a flags argument, even though
-it would be unused initially.
-
-@subsubheading Documentation
-All new APIs must be documented as Doxygen-formatted comments above the
-identifiers you add to the public headers. You should also briefly mention the
-change in @file{doc/APIchanges}.
-
-@subsubheading Bump the version
-Backward-incompatible API or ABI changes require incrementing (bumping) the
-major version number, as described in @ref{Major version bumps}. Major
-bumps are significant events that happen on a schedule - so if your change
-strictly requires one you should add it under @code{#if} preprocesor guards that
-disable it until the next major bump happens.
-
-New APIs that can be added without breaking API or ABI compatibility require
-bumping the minor version number.
-
-Incrementing the third (micro) version component means a noteworthy binary
-compatible change (e.g. encoder bug fix that matters for the decoder). The third
-component always starts at 100 to distinguish FFmpeg from Libav.
-
-@anchor{Removing interfaces}
-@subsection Removing interfaces
-Due to abovementioned compatibility guarantees, removing APIs is an involved
-process that should only be undertaken with good reason. Typically a deficient,
-restrictive, or otherwise inadequate API is replaced by a superior one, though
-it does at times happen that we remove an API without any replacement (e.g. when
-the feature it provides is deemed not worth the maintenance effort, out of scope
-of the project, fundamentally flawed, etc.).
-
-The removal has two steps - first the API is deprecated and scheduled for
-removal, but remains present and functional. The second step is actually
-removing the API - this is described in @ref{Major version bumps}.
-
-To deprecate an API you should signal to our users that they should stop using
-it. E.g. if you intend to remove struct members or functions, you should mark
-them with @code{attribute_deprecated}. When this cannot be done, it may be
-possible to detect the use of the deprecated API at runtime and print a warning
-(though take care not to print it too often). You should also document the
-deprecation (and the replacement, if applicable) in the relevant Doxygen
-documentation block.
-
-Finally, you should define a deprecation guard along the lines of
-@code{#define FF_API_<FOO> (LIBAVBAR_VERSION_MAJOR < XX)} (where XX is the major
-version in which the API will be removed) in @file{libavbar/version_major.h}
-(@file{version.h} in case of @code{libavutil}). Then wrap all uses of the
-deprecated API in @code{#if FF_API_<FOO> .... #endif}, so that the code will
-automatically get disabled once the major version reaches XX. You can also use
-@code{FF_DISABLE_DEPRECATION_WARNINGS} and @code{FF_ENABLE_DEPRECATION_WARNINGS}
-to suppress compiler deprecation warnings inside these guards. You should test
-that the code compiles and works with the guard macro evaluating to both true
-and false.
-
-@anchor{Major version bumps}
-@subsection Major version bumps
-A major version bump signifies an API and/or ABI compatibility break. To reduce
-the negative effects on our callers, who are required to adapt their code,
-backward-incompatible changes during a major bump should be limited to:
-@itemize @bullet
-@item
-Removing previously deprecated APIs.
-
-@item
-Performing ABI- but not API-breaking changes, like reordering struct contents.
-@end itemize
+@subheading Check untrusted input properly.
+Never write to unallocated memory, never write over the end of arrays,
+always check values read from some untrusted source before using them
+as array index or other risky things.

@section Documentation/Other
@subheading Subscribe to the ffmpeg-devel mailing list.
@@ -600,6 +425,35 @@ finding a new maintainer and also don't forget to update the @file{MAINTAINERS}

 We think our rules are not too hard. If you have comments, contact us.

+@chapter Code of conduct
+
+Be friendly and respectful towards others and third parties.
+Treat others the way you yourself want to be treated.
+
+Be considerate. Not everyone shares the same viewpoint and priorities as you do.
+Different opinions and interpretations help the project.
+Looking at issues from a different perspective assists development.
+
+Do not assume malice for things that can be attributed to incompetence. Even if
+it is malice, it's rarely good to start with that as initial assumption.
+
+Stay friendly even if someone acts contrarily. Everyone has a bad day
+once in a while.
+If you yourself have a bad day or are angry then try to take a break and reply
+once you are calm and without anger if you have to.
+
+Try to help other team members and cooperate if you can.
+
+The goal of software development is to create technical excellence, not for any
+individual to be better and "win" against the others. Large software projects
+are only possible and successful through teamwork.
+
+If someone struggles do not put them down. Give them a helping hand
+instead and point them in the right direction.
+
+Finally, keep in mind the immortal words of Bill and Ted,
+"Be excellent to each other."
+
@anchor{Submitting patches}
@chapter Submitting patches

@@ -640,22 +494,6 @@ patch is inline or attached per mail.
 You can check @url{https://patchwork.ffmpeg.org}, if your patch does not show up, its mime type
 likely was wrong.

-@subheading Sending patches from email clients
-Using @code{git send-email} might not be desirable for everyone. The
-following trick allows to send patches via email clients in a safe
-way. It has been tested with Outlook and Thunderbird (with X-Unsent
-extension) and might work with other applications.
-
-Create your patch like this:
-
-@verbatim
-git format-patch -s -o "outputfolder" --add-header "X-Unsent: 1" --suffix .eml --to ffmpeg-devel@ffmpeg.org -1 1a2b3c4d
-@end verbatim
-
-Now you'll just need to open the eml file with the email application
-and execute 'Send'.
-
-@subheading Reviews
 Your patch will be reviewed on the mailing list. You will likely be asked
 to make some changes and are expected to send in an improved version that
 incorporates the requests from the review. This process may go through
@@ -816,14 +654,16 @@ Lines with similar content should be aligned vertically when doing so
 improves readability.

@item
-Consider adding a regression test for your code. All new modules
-should be covered by tests. That includes demuxers, muxers, decoders, encoders
-filters, bitstream filters, parsers. If its not possible to do that, add
-an explanation why to your patchset, its ok to not test if theres a reason.
+Consider adding a regression test for your code.

@item
 If you added YASM code please check that things still work with --disable-yasm.

+@item
+Make sure you check the return values of function and return appropriate
+error codes. Especially memory allocation functions like @code{av_malloc()}
+are notoriously left unchecked, which is a serious problem.
+
@item
 Test your code with valgrind and or Address Sanitizer to ensure it's free
 of leaks, out of array accesses, etc.
@@ -873,8 +713,6 @@ accordingly].

@section Adding files to the fate-suite dataset

-If you need a sample uploaded send a mail to samples-request.
-
 When there is no muxer or encoder available to generate test media for a
 specific test then the media has to be included in the fate-suite.
 First please make sure that the sample file is as small as possible to test the
@@ -924,6 +762,25 @@ In case you need finer control over how valgrind is invoked, use the
@code{--target-exec='valgrind <your_custom_valgrind_options>} option in
 your configure line instead.

+@anchor{Maintenance}
+@chapter Maintenance process
+
+@anchor{MAINTAINERS}
+@section MAINTAINERS
+
+The developers maintaining each part of the codebase are listed in @file{MAINTAINERS}.
+Being listed in @file{MAINTAINERS}, gives one the right to have git write access to
+the specific repository.
+
+@anchor{Becoming a maintainer}
+@section Becoming a maintainer
+
+People add themselves to @file{MAINTAINERS} by sending a patch like any other code
+change. These get reviewed by the community like any other patch. It is expected
+that, if someone has an objection to a new maintainer, she is willing to object
+in public with her full name and is willing to take over maintainership for the area.
+
+
@anchor{Release process}
@chapter Release process

--- a/doc/doxy-wrapper.sh
+++ b/doc/doxy-wrapper.sh
@@ -1,13 +1,10 @@
 #!/bin/sh

 OUT_DIR="${1}"
-SRC_DIR="${2}"
-DOXYFILE="${3}"
-DOXYGEN="${4}"
+DOXYFILE="${2}"
+DOXYGEN="${3}"

-shift 4
-
-cd ${SRC_DIR}
+shift 3

 if [ -e "VERSION" ]; then
    VERSION=`cat "VERSION"`
--- a/doc/encoders.texi
+++ b/doc/encoders.texi
--- a/doc/examples/.gitignore
+++ b/doc/examples/.gitignore
@@ -22,4 +22,3 @@
 /transcoding
 /vaapi_encode
 /vaapi_transcode
-/qsv_transcode
--- a/doc/examples/Makefile
+++ b/doc/examples/Makefile
@@ -1,27 +1,26 @@
-EXAMPLES-$(CONFIG_AVIO_HTTP_SERVE_FILES)     += avio_http_serve_files
 EXAMPLES-$(CONFIG_AVIO_LIST_DIR_EXAMPLE)     += avio_list_dir
-EXAMPLES-$(CONFIG_AVIO_READ_CALLBACK_EXAMPLE) += avio_read_callback
+EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE)      += avio_reading
 EXAMPLES-$(CONFIG_DECODE_AUDIO_EXAMPLE)      += decode_audio
-EXAMPLES-$(CONFIG_DECODE_FILTER_AUDIO_EXAMPLE) += decode_filter_audio
-EXAMPLES-$(CONFIG_DECODE_FILTER_VIDEO_EXAMPLE) += decode_filter_video
 EXAMPLES-$(CONFIG_DECODE_VIDEO_EXAMPLE)      += decode_video
-EXAMPLES-$(CONFIG_DEMUX_DECODE_EXAMPLE)      += demux_decode
+EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding
 EXAMPLES-$(CONFIG_ENCODE_AUDIO_EXAMPLE)      += encode_audio
 EXAMPLES-$(CONFIG_ENCODE_VIDEO_EXAMPLE)      += encode_video
 EXAMPLES-$(CONFIG_EXTRACT_MVS_EXAMPLE)       += extract_mvs
 EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE)      += filter_audio
+EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE)   += filtering_audio
+EXAMPLES-$(CONFIG_FILTERING_VIDEO_EXAMPLE)   += filtering_video
+EXAMPLES-$(CONFIG_HTTP_MULTICLIENT_EXAMPLE)  += http_multiclient
 EXAMPLES-$(CONFIG_HW_DECODE_EXAMPLE)         += hw_decode
-EXAMPLES-$(CONFIG_MUX_EXAMPLE)               += mux
-EXAMPLES-$(CONFIG_QSV_DECODE_EXAMPLE)        += qsv_decode
-EXAMPLES-$(CONFIG_REMUX_EXAMPLE)             += remux
-EXAMPLES-$(CONFIG_RESAMPLE_AUDIO_EXAMPLE)    += resample_audio
-EXAMPLES-$(CONFIG_SCALE_VIDEO_EXAMPLE)       += scale_video
-EXAMPLES-$(CONFIG_SHOW_METADATA_EXAMPLE)     += show_metadata
+EXAMPLES-$(CONFIG_METADATA_EXAMPLE)          += metadata
+EXAMPLES-$(CONFIG_MUXING_EXAMPLE)            += muxing
+EXAMPLES-$(CONFIG_QSVDEC_EXAMPLE)            += qsvdec
+EXAMPLES-$(CONFIG_REMUXING_EXAMPLE)          += remuxing
+EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE)  += resampling_audio
+EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE)     += scaling_video
 EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE)     += transcode_aac
-EXAMPLES-$(CONFIG_TRANSCODE_EXAMPLE)         += transcode
+EXAMPLES-$(CONFIG_TRANSCODING_EXAMPLE)       += transcoding
 EXAMPLES-$(CONFIG_VAAPI_ENCODE_EXAMPLE)      += vaapi_encode
 EXAMPLES-$(CONFIG_VAAPI_TRANSCODE_EXAMPLE)   += vaapi_transcode
-EXAMPLES-$(CONFIG_QSV_TRANSCODE_EXAMPLE)     += qsv_transcode

 EXAMPLES       := $(EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)$(EXESUF))
 EXAMPLES_G     := $(EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)_g$(EXESUF))
--- a/doc/examples/Makefile.example
+++ b/doc/examples/Makefile.example
@@ -11,40 +11,33 @@ CFLAGS += -Wall -g
 CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
 LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)

-# missing the following targets, since they need special options in the FFmpeg build:
-# qsv_decode
-# qsv_transcode
-# vaapi_encode
-# vaapi_transcode
-
-EXAMPLES=\
-                avio_http_serve_files              \
-                avio_list_dir                      \
-                avio_read_callback                 \
+EXAMPLES=       avio_list_dir                      \
+                avio_reading                       \
                decode_audio                       \
-                decode_filter_audio                \
-                decode_filter_video                \
                decode_video                       \
-                demux_decode                       \
+                demuxing_decoding                  \
                encode_audio                       \
                encode_video                       \
                extract_mvs                        \
+                filtering_video                    \
+                filtering_audio                    \
+                http_multiclient                   \
                hw_decode                          \
-                mux                                \
-                remux                              \
-                resample_audio                     \
-                scale_video                        \
-                show_metadata                      \
+                metadata                           \
+                muxing                             \
+                remuxing                           \
+                resampling_audio                   \
+                scaling_video                      \
                transcode_aac                      \
-                transcode
+                transcoding                        \

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

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

 .phony: all clean-test clean

--- a/doc/examples/README
+++ b/doc/examples/README
@@ -7,10 +7,8 @@ that you have them installed and working on your system.

 Method 1: build the installed examples in a generic read/write user directory

-Copy to a read/write user directory and run:
-make -f Makefile.example
-
-It will link to the libraries on your system, assuming the PKG_CONFIG_PATH is
+Copy to a read/write user directory and just use "make", it will link
+to the libraries on your system, assuming the PKG_CONFIG_PATH is
 correctly configured.

 Method 2: build the examples in-tree
@@ -22,4 +20,4 @@ examples using "make examplesclean"

 If you want to try the dedicated Makefile examples (to emulate the first
 method), go into doc/examples and run a command such as
-PKG_CONFIG_PATH=pc-uninstalled make -f Makefile.example
+PKG_CONFIG_PATH=pc-uninstalled make.
--- a/doc/examples/avio_list_dir.c
+++ b/doc/examples/avio_list_dir.c
@@ -20,13 +20,6 @@
 * THE SOFTWARE.
 */

-/**
- * @file libavformat AVIOContext list directory API usage example
- * @example avio_list_dir.c
- *
- * Show how to list directories through the libavformat AVIOContext API.
- */
-
 #include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
 #include <libavformat/avio.h>
--- a/doc/examples/avio_read_callback.c
+++ b/doc/examples/avio_read_callback.c
@@ -21,11 +21,12 @@
 */

 /**
- * @file libavformat AVIOContext read callback API usage example
- * @example avio_read_callback.c
+ * @file
+ * libavformat AVIOContext API example.
 *
 * Make libavformat demuxer access media content through a custom
 * AVIOContext read callback.
+ * @example avio_reading.c
 */

 #include <libavcodec/avcodec.h>
--- a/doc/examples/decode_audio.c
+++ b/doc/examples/decode_audio.c
@@ -21,11 +21,10 @@
 */

 /**
- * @file libavcodec audio decoding API usage example
- * @example decode_audio.c
+ * @file
+ * audio decoding with libavcodec API example
 *
- * Decode data from an MP2 input file and generate a raw audio file to
- * be played with ffplay.
+ * @example decode_audio.c
 */

 #include <stdio.h>
@@ -98,7 +97,7 @@ static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame,
            exit(1);
        }
        for (i = 0; i < frame->nb_samples; i++)
-            for (ch = 0; ch < dec_ctx->ch_layout.nb_channels; ch++)
+            for (ch = 0; ch < dec_ctx->channels; ch++)
                fwrite(frame->data[ch] + data_size*i, 1, data_size, outfile);
    }
 }
@@ -216,7 +215,7 @@ int main(int argc, char **argv)
        sfmt = av_get_packed_sample_fmt(sfmt);
    }

-    n_channels = c->ch_layout.nb_channels;
+    n_channels = c->channels;
    if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
        goto end;

--- a/doc/examples/decode_video.c
+++ b/doc/examples/decode_video.c
@@ -21,11 +21,10 @@
 */

 /**
- * @file libavcodec video decoding API usage example
- * @example decode_video.c *
+ * @file
+ * video decoding with libavcodec API example
 *
- * Read from an MPEG1 video file, decode frames, and generate PGM images as
- * output.
+ * @example decode_video.c
 */

 #include <stdio.h>
@@ -70,12 +69,12 @@ static void decode(AVCodecContext *dec_ctx, AVFrame *frame, AVPacket *pkt,
            exit(1);
        }

-        printf("saving frame %3"PRId64"\n", dec_ctx->frame_num);
+        printf("saving frame %3d\n", dec_ctx->frame_number);
        fflush(stdout);

        /* the picture is allocated by the decoder. no need to
           free it */
-        snprintf(buf, sizeof(buf), "%s-%"PRId64, filename, dec_ctx->frame_num);
+        snprintf(buf, sizeof(buf), "%s-%d", filename, dec_ctx->frame_number);
        pgm_save(frame->data[0], frame->linesize[0],
                 frame->width, frame->height, buf);
    }
@@ -93,7 +92,6 @@ int main(int argc, char **argv)
    uint8_t *data;
    size_t   data_size;
    int ret;
-    int eof;
    AVPacket *pkt;

    if (argc <= 2) {
@@ -152,16 +150,15 @@ int main(int argc, char **argv)
        exit(1);
    }

-    do {
+    while (!feof(f)) {
        /* read raw data from the input file */
        data_size = fread(inbuf, 1, INBUF_SIZE, f);
-        if (ferror(f))
+        if (!data_size)
            break;
-        eof = !data_size;

        /* use the parser to split the data into frames */
        data = inbuf;
-        while (data_size > 0 || eof) {
+        while (data_size > 0) {
            ret = av_parser_parse2(parser, c, &pkt->data, &pkt->size,
                                   data, data_size, AV_NOPTS_VALUE, AV_NOPTS_VALUE, 0);
            if (ret < 0) {
@@ -173,10 +170,8 @@ int main(int argc, char **argv)

            if (pkt->size)
                decode(c, frame, pkt, outfilename);
-            else if (eof)
-                break;
        }
-    } while (!eof);
+    }

    /* flush the decoder */
    decode(c, frame, NULL, outfilename);
--- a/doc/examples/demuxing_decoding.c
+++ b/doc/examples/demuxing_decoding.c
@@ -21,18 +21,17 @@
 */

 /**
- * @file libavformat and libavcodec demuxing and decoding API usage example
- * @example demux_decode.c
+ * @file
+ * Demuxing and decoding example.
 *
- * Show how to use the libavformat and libavcodec API to demux and decode audio
- * and video data. Write the output as raw audio and input files to be played by
- * ffplay.
+ * Show how to use the libavformat and libavcodec API to demux and
+ * decode audio and video data.
+ * @example demuxing_decoding.c
 */

 #include <libavutil/imgutils.h>
 #include <libavutil/samplefmt.h>
 #include <libavutil/timestamp.h>
-#include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>

 static AVFormatContext *fmt_ctx = NULL;
@@ -73,14 +72,14 @@ static int output_video_frame(AVFrame *frame)
        return -1;
    }

-    printf("video_frame n:%d\n",
-           video_frame_count++);
+    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_copy2(video_dst_data, video_dst_linesize,
-                   frame->data, frame->linesize,
-                   pix_fmt, width, height);
+    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);
@@ -138,11 +137,9 @@ static int decode_packet(AVCodecContext *dec, const AVPacket *pkt)
            ret = output_audio_frame(frame);

        av_frame_unref(frame);
-        if (ret < 0)
-            return ret;
    }

-    return 0;
+    return ret;
 }

 static int open_codec_context(int *stream_idx,
@@ -150,7 +147,8 @@ static int open_codec_context(int *stream_idx,
 {
    int ret, stream_index;
    AVStream *st;
-    const AVCodec *dec = NULL;
+    AVCodec *dec = NULL;
+    AVDictionary *opts = NULL;

    ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
    if (ret < 0) {
@@ -185,7 +183,7 @@ static int open_codec_context(int *stream_idx,
        }

        /* Init the decoders */
-        if ((ret = avcodec_open2(*dec_ctx, dec, NULL)) < 0) {
+        if ((ret = avcodec_open2(*dec_ctx, dec, &opts)) < 0) {
            fprintf(stderr, "Failed to open %s codec\n",
                    av_get_media_type_string(type));
            return ret;
@@ -345,7 +343,7 @@ int main (int argc, char **argv)

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

        if (av_sample_fmt_is_planar(sfmt)) {
--- a/doc/examples/encode_audio.c
+++ b/doc/examples/encode_audio.c
@@ -21,10 +21,10 @@
 */

 /**
- * @file libavcodec encoding audio API usage examples
- * @example encode_audio.c
+ * @file
+ * audio encoding with libavcodec API example.
 *
- * Generate a synthetic audio signal and encode it to an output MP2 file.
+ * @example encode_audio.c
 */

 #include <stdint.h>
@@ -70,25 +70,26 @@ static int select_sample_rate(const AVCodec *codec)
 }

 /* select layout with the highest channel count */
-static int select_channel_layout(const AVCodec *codec, AVChannelLayout *dst)
+static int select_channel_layout(const AVCodec *codec)
 {
-    const AVChannelLayout *p, *best_ch_layout;
+    const uint64_t *p;
+    uint64_t best_ch_layout = 0;
    int best_nb_channels   = 0;

-    if (!codec->ch_layouts)
-        return av_channel_layout_copy(dst, &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO);
+    if (!codec->channel_layouts)
+        return AV_CH_LAYOUT_STEREO;

-    p = codec->ch_layouts;
-    while (p->nb_channels) {
-        int nb_channels = p->nb_channels;
+    p = codec->channel_layouts;
+    while (*p) {
+        int nb_channels = av_get_channel_layout_nb_channels(*p);

        if (nb_channels > best_nb_channels) {
-            best_ch_layout   = p;
+            best_ch_layout    = *p;
            best_nb_channels = nb_channels;
        }
        p++;
    }
-    return av_channel_layout_copy(dst, best_ch_layout);
+    return best_ch_layout;
 }

 static void encode(AVCodecContext *ctx, AVFrame *frame, AVPacket *pkt,
@@ -163,9 +164,8 @@ int main(int argc, char **argv)

    /* select other audio parameters supported by the encoder */
    c->sample_rate    = select_sample_rate(codec);
-    ret = select_channel_layout(codec, &c->ch_layout);
-    if (ret < 0)
-        exit(1);
+    c->channel_layout = select_channel_layout(codec);
+    c->channels       = av_get_channel_layout_nb_channels(c->channel_layout);

    /* open it */
    if (avcodec_open2(c, codec, NULL) < 0) {
@@ -195,9 +195,7 @@ int main(int argc, char **argv)

    frame->nb_samples     = c->frame_size;
    frame->format         = c->sample_fmt;
-    ret = av_channel_layout_copy(&frame->ch_layout, &c->ch_layout);
-    if (ret < 0)
-        exit(1);
+    frame->channel_layout = c->channel_layout;

    /* allocate the data buffers */
    ret = av_frame_get_buffer(frame, 0);
@@ -220,7 +218,7 @@ int main(int argc, char **argv)
        for (j = 0; j < c->frame_size; j++) {
            samples[2*j] = (int)(sin(t) * 10000);

-            for (k = 1; k < c->ch_layout.nb_channels; k++)
+            for (k = 1; k < c->channels; k++)
                samples[2*j + k] = samples[2*j];
            t += tincr;
        }
--- a/doc/examples/encode_video.c
+++ b/doc/examples/encode_video.c
@@ -21,10 +21,10 @@
 */

 /**
- * @file libavcodec encoding video API usage example
- * @example encode_video.c
+ * @file
+ * video encoding with libavcodec API example
 *
- * Generate synthetic video data and encode it to an output file.
+ * @example encode_video.c
 */

 #include <stdio.h>
@@ -155,25 +155,12 @@ int main(int argc, char **argv)
    for (i = 0; i < 25; i++) {
        fflush(stdout);

-        /* Make sure the frame data is writable.
-           On the first round, the frame is fresh from av_frame_get_buffer()
-           and therefore we know it is writable.
-           But on the next rounds, encode() will have called
-           avcodec_send_frame(), and the codec may have kept a reference to
-           the frame in its internal structures, that makes the frame
-           unwritable.
-           av_frame_make_writable() checks that and allocates a new buffer
-           for the frame only if necessary.
-         */
+        /* make sure the frame data is writable */
        ret = av_frame_make_writable(frame);
        if (ret < 0)
            exit(1);

-        /* Prepare a dummy image.
-           In real code, this is where you would have your own logic for
-           filling the frame. FFmpeg does not care what you put in the
-           frame.
-         */
+        /* prepare a dummy image */
        /* Y */
        for (y = 0; y < c->height; y++) {
            for (x = 0; x < c->width; x++) {
@@ -198,12 +185,7 @@ int main(int argc, char **argv)
    /* flush the encoder */
    encode(c, NULL, pkt, f);

-    /* Add sequence end code to have a real MPEG file.
-       It makes only sense because this tiny examples writes packets
-       directly. This is called "elementary stream" and only works for some
-       codecs. To create a valid file, you usually need to write packets
-       into a proper file format or protocol; see mux.c.
-     */
+    /* add sequence end code to have a real MPEG file */
    if (codec->id == AV_CODEC_ID_MPEG1VIDEO || codec->id == AV_CODEC_ID_MPEG2VIDEO)
        fwrite(endcode, 1, sizeof(endcode), f);
    fclose(f);
--- a/doc/examples/extract_mvs.c
+++ b/doc/examples/extract_mvs.c
@@ -21,16 +21,7 @@
 * THE SOFTWARE.
 */

-/**
- * @file libavcodec motion vectors extraction API usage example
- * @example extract_mvs.c
- *
- * Read from input file, decode video stream and print a motion vectors
- * representation to stdout.
- */
-
 #include <libavutil/motion_vector.h>
-#include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>

 static AVFormatContext *fmt_ctx = NULL;
@@ -69,11 +60,10 @@ static int decode_packet(const AVPacket *pkt)
                const AVMotionVector *mvs = (const AVMotionVector *)sd->data;
                for (i = 0; i < sd->size / sizeof(*mvs); i++) {
                    const AVMotionVector *mv = &mvs[i];
-                    printf("%d,%2d,%2d,%2d,%4d,%4d,%4d,%4d,0x%"PRIx64",%4d,%4d,%4d\n",
+                    printf("%d,%2d,%2d,%2d,%4d,%4d,%4d,%4d,0x%"PRIx64"\n",
                        video_frame_count, mv->source,
                        mv->w, mv->h, mv->src_x, mv->src_y,
-                        mv->dst_x, mv->dst_y, mv->flags,
-                        mv->motion_x, mv->motion_y, mv->motion_scale);
+                        mv->dst_x, mv->dst_y, mv->flags);
                }
            }
            av_frame_unref(frame);
@@ -88,7 +78,7 @@ static int open_codec_context(AVFormatContext *fmt_ctx, enum AVMediaType type)
    int ret;
    AVStream *st;
    AVCodecContext *dec_ctx = NULL;
-    const AVCodec *dec = NULL;
+    AVCodec *dec = NULL;
    AVDictionary *opts = NULL;

    ret = av_find_best_stream(fmt_ctx, type, -1, -1, &dec, 0);
@@ -114,9 +104,7 @@ static int open_codec_context(AVFormatContext *fmt_ctx, enum AVMediaType type)

        /* Init the video decoder */
        av_dict_set(&opts, "flags2", "+export_mvs", 0);
-        ret = avcodec_open2(dec_ctx, dec, &opts);
-        av_dict_free(&opts);
-        if (ret < 0) {
+        if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) {
            fprintf(stderr, "Failed to open %s codec\n",
                    av_get_media_type_string(type));
            return ret;
@@ -133,7 +121,7 @@ static int open_codec_context(AVFormatContext *fmt_ctx, enum AVMediaType type)
 int main(int argc, char **argv)
 {
    int ret = 0;
-    AVPacket *pkt = NULL;
+    AVPacket pkt = { 0 };

    if (argc != 2) {
        fprintf(stderr, "Usage: %s <video>\n", argv[0]);
@@ -168,20 +156,13 @@ int main(int argc, char **argv)
        goto end;
    }

-    pkt = av_packet_alloc();
-    if (!pkt) {
-        fprintf(stderr, "Could not allocate AVPacket\n");
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-
-    printf("framenum,source,blockw,blockh,srcx,srcy,dstx,dsty,flags,motion_x,motion_y,motion_scale\n");
+    printf("framenum,source,blockw,blockh,srcx,srcy,dstx,dsty,flags\n");

    /* read frames from the file */
-    while (av_read_frame(fmt_ctx, pkt) >= 0) {
-        if (pkt->stream_index == video_stream_idx)
-            ret = decode_packet(pkt);
-        av_packet_unref(pkt);
+    while (av_read_frame(fmt_ctx, &pkt) >= 0) {
+        if (pkt.stream_index == video_stream_idx)
+            ret = decode_packet(&pkt);
+        av_packet_unref(&pkt);
        if (ret < 0)
            break;
    }
@@ -193,6 +174,5 @@ end:
    avcodec_free_context(&video_dec_ctx);
    avformat_close_input(&fmt_ctx);
    av_frame_free(&frame);
-    av_packet_free(&pkt);
    return ret < 0;
 }
--- a/doc/examples/filter_audio.c
+++ b/doc/examples/filter_audio.c
@@ -19,11 +19,13 @@
 */

 /**
- * @file libavfilter audio filtering API usage example
- * @example filter_audio.c
+ * @file
+ * libavfilter API usage example.
 *
- * This example will generate a sine wave audio, pass it through a simple filter
- * chain, and then compute the MD5 checksum of the output data.
+ * @example filter_audio.c
+ * This example will generate a sine wave audio,
+ * pass it through a simple filter chain, and then compute the MD5 checksum of
+ * the output data.
 *
 * The filter chain it uses is:
 * (input) -> abuffer -> volume -> aformat -> abuffersink -> (output)
@@ -53,7 +55,7 @@

 #define INPUT_SAMPLERATE     48000
 #define INPUT_FORMAT         AV_SAMPLE_FMT_FLTP
-#define INPUT_CHANNEL_LAYOUT (AVChannelLayout)AV_CHANNEL_LAYOUT_5POINT0
+#define INPUT_CHANNEL_LAYOUT AV_CH_LAYOUT_5POINT0

 #define VOLUME_VAL 0.90

@@ -98,7 +100,7 @@ static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
    }

    /* Set the filter options through the AVOptions API. */
-    av_channel_layout_describe(&INPUT_CHANNEL_LAYOUT, ch_layout, sizeof(ch_layout));
+    av_get_channel_layout_string(ch_layout, sizeof(ch_layout), 0, INPUT_CHANNEL_LAYOUT);
    av_opt_set    (abuffer_ctx, "channel_layout", ch_layout,                            AV_OPT_SEARCH_CHILDREN);
    av_opt_set    (abuffer_ctx, "sample_fmt",     av_get_sample_fmt_name(INPUT_FORMAT), AV_OPT_SEARCH_CHILDREN);
    av_opt_set_q  (abuffer_ctx, "time_base",      (AVRational){ 1, INPUT_SAMPLERATE },  AV_OPT_SEARCH_CHILDREN);
@@ -152,8 +154,9 @@ static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
    /* A third way of passing the options is in a string of the form
     * key1=value1:key2=value2.... */
    snprintf(options_str, sizeof(options_str),
-             "sample_fmts=%s:sample_rates=%d:channel_layouts=stereo",
-             av_get_sample_fmt_name(AV_SAMPLE_FMT_S16), 44100);
+             "sample_fmts=%s:sample_rates=%d:channel_layouts=0x%"PRIx64,
+             av_get_sample_fmt_name(AV_SAMPLE_FMT_S16), 44100,
+             (uint64_t)AV_CH_LAYOUT_STEREO);
    err = avfilter_init_str(aformat_ctx, options_str);
    if (err < 0) {
        av_log(NULL, AV_LOG_ERROR, "Could not initialize the aformat filter.\n");
@@ -212,7 +215,7 @@ static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
 static int process_output(struct AVMD5 *md5, AVFrame *frame)
 {
    int planar     = av_sample_fmt_is_planar(frame->format);
-    int channels   = frame->ch_layout.nb_channels;
+    int channels   = av_get_channel_layout_nb_channels(frame->channel_layout);
    int planes     = planar ? channels : 1;
    int bps        = av_get_bytes_per_sample(frame->format);
    int plane_size = bps * frame->nb_samples * (planar ? 1 : channels);
@@ -245,7 +248,7 @@ static int get_input(AVFrame *frame, int frame_num)
    /* Set up the frame properties and allocate the buffer for the data. */
    frame->sample_rate    = INPUT_SAMPLERATE;
    frame->format         = INPUT_FORMAT;
-    av_channel_layout_copy(&frame->ch_layout, &INPUT_CHANNEL_LAYOUT);
+    frame->channel_layout = INPUT_CHANNEL_LAYOUT;
    frame->nb_samples     = FRAME_SIZE;
    frame->pts            = frame_num * FRAME_SIZE;

--- a/doc/examples/decode_filter_audio.c
+++ b/doc/examples/decode_filter_audio.c
@@ -23,11 +23,9 @@
 */

 /**
- * @file audio decoding and filtering usage example
- * @example decode_filter_audio.c
- *
- * Demux, decode and filter audio input file, generate a raw audio
- * file to be played with ffplay.
+ * @file
+ * API example for audio decoding and filtering
+ * @example filtering_audio.c
 */

 #include <unistd.h>
@@ -36,7 +34,6 @@
 #include <libavformat/avformat.h>
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
-#include <libavutil/channel_layout.h>
 #include <libavutil/opt.h>

 static const char *filter_descr = "aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono";
@@ -51,8 +48,8 @@ static int audio_stream_index = -1;

 static int open_input_file(const char *filename)
 {
-    const AVCodec *dec;
    int ret;
+    AVCodec *dec;

    if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
@@ -96,6 +93,7 @@ static int init_filters(const char *filters_descr)
    AVFilterInOut *outputs = avfilter_inout_alloc();
    AVFilterInOut *inputs  = avfilter_inout_alloc();
    static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
+    static const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 };
    static const int out_sample_rates[] = { 8000, -1 };
    const AVFilterLink *outlink;
    AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;
@@ -107,13 +105,12 @@ static int init_filters(const char *filters_descr)
    }

    /* buffer audio source: the decoded frames from the decoder will be inserted here. */
-    if (dec_ctx->ch_layout.order == AV_CHANNEL_ORDER_UNSPEC)
-        av_channel_layout_default(&dec_ctx->ch_layout, dec_ctx->ch_layout.nb_channels);
-    ret = snprintf(args, sizeof(args),
-            "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=",
+    if (!dec_ctx->channel_layout)
+        dec_ctx->channel_layout = av_get_default_channel_layout(dec_ctx->channels);
+    snprintf(args, sizeof(args),
+            "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
             time_base.num, time_base.den, dec_ctx->sample_rate,
-             av_get_sample_fmt_name(dec_ctx->sample_fmt));
-    av_channel_layout_describe(&dec_ctx->ch_layout, args + ret, sizeof(args) - ret);
+             av_get_sample_fmt_name(dec_ctx->sample_fmt), dec_ctx->channel_layout);
    ret = avfilter_graph_create_filter(&buffersrc_ctx, abuffersrc, "in",
                                       args, NULL, filter_graph);
    if (ret < 0) {
@@ -136,7 +133,7 @@ static int init_filters(const char *filters_descr)
        goto end;
    }

-    ret = av_opt_set(buffersink_ctx, "ch_layouts", "mono",
+    ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts", out_channel_layouts, -1,
                              AV_OPT_SEARCH_CHILDREN);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
@@ -187,7 +184,7 @@ static int init_filters(const char *filters_descr)
    /* Print summary of the sink buffer
     * Note: args buffer is reused to store channel layout string */
    outlink = buffersink_ctx->inputs[0];
-    av_channel_layout_describe(&outlink->ch_layout, args, sizeof(args));
+    av_get_channel_layout_string(args, sizeof(args), -1, outlink->channel_layout);
    av_log(NULL, AV_LOG_INFO, "Output: srate:%dHz fmt:%s chlayout:%s\n",
           (int)outlink->sample_rate,
           (char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
@@ -202,7 +199,7 @@ end:

 static void print_frame(const AVFrame *frame)
 {
-    const int n = frame->nb_samples * frame->ch_layout.nb_channels;
+    const int n = frame->nb_samples * av_get_channel_layout_nb_channels(frame->channel_layout);
    const uint16_t *p     = (uint16_t*)frame->data[0];
    const uint16_t *p_end = p + n;

@@ -217,12 +214,12 @@ static void print_frame(const AVFrame *frame)
 int main(int argc, char **argv)
 {
    int ret;
-    AVPacket *packet = av_packet_alloc();
+    AVPacket packet;
    AVFrame *frame = av_frame_alloc();
    AVFrame *filt_frame = av_frame_alloc();

-    if (!packet || !frame || !filt_frame) {
-        fprintf(stderr, "Could not allocate frame or packet\n");
+    if (!frame || !filt_frame) {
+        perror("Could not allocate frame");
        exit(1);
    }
    if (argc != 2) {
@@ -237,11 +234,11 @@ int main(int argc, char **argv)

    /* read all packets */
    while (1) {
-        if ((ret = av_read_frame(fmt_ctx, packet)) < 0)
+        if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
            break;

-        if (packet->stream_index == audio_stream_index) {
-            ret = avcodec_send_packet(dec_ctx, packet);
+        if (packet.stream_index == audio_stream_index) {
+            ret = avcodec_send_packet(dec_ctx, &packet);
            if (ret < 0) {
                av_log(NULL, AV_LOG_ERROR, "Error while sending a packet to the decoder\n");
                break;
@@ -277,13 +274,12 @@ int main(int argc, char **argv)
                }
            }
        }
-        av_packet_unref(packet);
+        av_packet_unref(&packet);
    }
 end:
    avfilter_graph_free(&filter_graph);
    avcodec_free_context(&dec_ctx);
    avformat_close_input(&fmt_ctx);
-    av_packet_free(&packet);
    av_frame_free(&frame);
    av_frame_free(&filt_frame);

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

 #define _XOPEN_SOURCE 600 /* for usleep */
@@ -53,8 +53,8 @@ static int64_t last_pts = AV_NOPTS_VALUE;

 static int open_input_file(const char *filename)
 {
-    const AVCodec *dec;
    int ret;
+    AVCodec *dec;

    if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
@@ -210,7 +210,7 @@ static void display_frame(const AVFrame *frame, AVRational time_base)
 int main(int argc, char **argv)
 {
    int ret;
-    AVPacket *packet;
+    AVPacket packet;
    AVFrame *frame;
    AVFrame *filt_frame;

@@ -221,9 +221,8 @@ int main(int argc, char **argv)

    frame = av_frame_alloc();
    filt_frame = av_frame_alloc();
-    packet = av_packet_alloc();
-    if (!frame || !filt_frame || !packet) {
-        fprintf(stderr, "Could not allocate frame or packet\n");
+    if (!frame || !filt_frame) {
+        perror("Could not allocate frame");
        exit(1);
    }

@@ -234,11 +233,11 @@ int main(int argc, char **argv)

    /* read all packets */
    while (1) {
-        if ((ret = av_read_frame(fmt_ctx, packet)) < 0)
+        if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
            break;

-        if (packet->stream_index == video_stream_index) {
-            ret = avcodec_send_packet(dec_ctx, packet);
+        if (packet.stream_index == video_stream_index) {
+            ret = avcodec_send_packet(dec_ctx, &packet);
            if (ret < 0) {
                av_log(NULL, AV_LOG_ERROR, "Error while sending a packet to the decoder\n");
                break;
@@ -274,7 +273,7 @@ int main(int argc, char **argv)
                av_frame_unref(frame);
            }
        }
-        av_packet_unref(packet);
+        av_packet_unref(&packet);
    }
 end:
    avfilter_graph_free(&filter_graph);
@@ -282,7 +281,6 @@ end:
    avformat_close_input(&fmt_ctx);
    av_frame_free(&frame);
    av_frame_free(&filt_frame);
-    av_packet_free(&packet);

    if (ret < 0 && ret != AVERROR_EOF) {
        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
--- a/doc/examples/avio_http_serve_files.c
+++ b/doc/examples/avio_http_serve_files.c
@@ -21,11 +21,12 @@
 */

 /**
- * @file libavformat multi-client network API usage example
- * @example avio_http_serve_files.c
+ * @file
+ * libavformat multi-client network API usage example.
 *
- * Serve a file without decoding or demuxing it over the HTTP protocol. Multiple
- * clients can connect and will receive the same file.
+ * @example http_multiclient.c
+ * This example will serve a file without decoding or demuxing it over http.
+ * Multiple clients can connect and will receive the same file.
 */

 #include <libavformat/avformat.h>
--- a/doc/examples/hw_decode.c
+++ b/doc/examples/hw_decode.c
@@ -24,11 +24,12 @@
 */

 /**
- * @file HW-accelerated decoding API usage.example
- * @example hw_decode.c
+ * @file
+ * HW-Accelerated decoding example.
 *
- * Perform HW-accelerated decoding with output frames from HW video
- * surfaces.
+ * @example hw_decode.c
+ * This example shows how to do HW-accelerated decoding with output
+ * frames from the HW video surfaces.
 */

 #include <stdio.h>
@@ -151,8 +152,8 @@ int main(int argc, char *argv[])
    int video_stream, ret;
    AVStream *video = NULL;
    AVCodecContext *decoder_ctx = NULL;
-    const AVCodec *decoder = NULL;
-    AVPacket *packet = NULL;
+    AVCodec *decoder = NULL;
+    AVPacket packet;
    enum AVHWDeviceType type;
    int i;

@@ -171,12 +172,6 @@ int main(int argc, char *argv[])
        return -1;
    }

-    packet = av_packet_alloc();
-    if (!packet) {
-        fprintf(stderr, "Failed to allocate AVPacket\n");
-        return -1;
-    }
-
    /* open the input file */
    if (avformat_open_input(&input_ctx, argv[2], NULL, NULL) != 0) {
        fprintf(stderr, "Cannot open input file '%s'\n", argv[2]);
@@ -232,21 +227,23 @@ int main(int argc, char *argv[])

    /* actual decoding and dump the raw data */
    while (ret >= 0) {
-        if ((ret = av_read_frame(input_ctx, packet)) < 0)
+        if ((ret = av_read_frame(input_ctx, &packet)) < 0)
            break;

-        if (video_stream == packet->stream_index)
-            ret = decode_write(decoder_ctx, packet);
+        if (video_stream == packet.stream_index)
+            ret = decode_write(decoder_ctx, &packet);

-        av_packet_unref(packet);
+        av_packet_unref(&packet);
    }

    /* flush the decoder */
-    ret = decode_write(decoder_ctx, NULL);
+    packet.data = NULL;
+    packet.size = 0;
+    ret = decode_write(decoder_ctx, &packet);
+    av_packet_unref(&packet);

    if (output_file)
        fclose(output_file);
-    av_packet_free(&packet);
    avcodec_free_context(&decoder_ctx);
    avformat_close_input(&input_ctx);
    av_buffer_unref(&hw_device_ctx);
--- a/doc/examples/show_metadata.c
+++ b/doc/examples/show_metadata.c
@@ -21,10 +21,9 @@
 */

 /**
- * @file libavformat metadata extraction API usage example
- * @example show_metadata.c
- *
- * Show metadata from an input file.
+ * @file
+ * Shows how the metadata API can be used in application programs.
+ * @example metadata.c
 */

 #include <stdio.h>
@@ -35,7 +34,7 @@
 int main (int argc, char **argv)
 {
    AVFormatContext *fmt_ctx = NULL;
-    const AVDictionaryEntry *tag = NULL;
+    AVDictionaryEntry *tag = NULL;
    int ret;

    if (argc != 2) {
@@ -53,7 +52,7 @@ int main (int argc, char **argv)
        return ret;
    }

-    while ((tag = av_dict_iterate(fmt_ctx->metadata, tag)))
+    while ((tag = av_dict_get(fmt_ctx->metadata, "", tag, AV_DICT_IGNORE_SUFFIX)))
        printf("%s=%s\n", tag->key, tag->value);

    avformat_close_input(&fmt_ctx);
--- a/doc/examples/muxing.c
+++ b/doc/examples/muxing.c
@@ -21,11 +21,12 @@
 */

 /**
- * @file libavformat muxing API usage example
- * @example mux.c
+ * @file
+ * libavformat API example.
 *
- * Generate a synthetic audio and video signal and mux them to a media file in
- * any supported libavformat format. The default codecs are used.
+ * Output a media file in any supported libavformat format. The default
+ * codecs are used.
+ * @example muxing.c
 */

 #include <stdlib.h>
@@ -38,7 +39,6 @@
 #include <libavutil/opt.h>
 #include <libavutil/mathematics.h>
 #include <libavutil/timestamp.h>
-#include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
 #include <libswscale/swscale.h>
 #include <libswresample/swresample.h>
@@ -61,8 +61,6 @@ typedef struct OutputStream {
    AVFrame *frame;
    AVFrame *tmp_frame;

-    AVPacket *tmp_pkt;
-
    float t, tincr, tincr2;

    struct SwsContext *sws_ctx;
@@ -81,7 +79,7 @@ static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt)
 }

 static int write_frame(AVFormatContext *fmt_ctx, AVCodecContext *c,
-                       AVStream *st, AVFrame *frame, AVPacket *pkt)
+                       AVStream *st, AVFrame *frame)
 {
    int ret;

@@ -94,7 +92,9 @@ static int write_frame(AVFormatContext *fmt_ctx, AVCodecContext *c,
    }

    while (ret >= 0) {
-        ret = avcodec_receive_packet(c, pkt);
+        AVPacket pkt = { 0 };
+
+        ret = avcodec_receive_packet(c, &pkt);
        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
            break;
        else if (ret < 0) {
@@ -103,15 +103,13 @@ static int write_frame(AVFormatContext *fmt_ctx, AVCodecContext *c,
        }

        /* 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;
+        av_packet_rescale_ts(&pkt, c->time_base, st->time_base);
+        pkt.stream_index = st->index;

        /* Write the compressed frame to the media file. */
-        log_packet(fmt_ctx, pkt);
-        ret = av_interleaved_write_frame(fmt_ctx, pkt);
-        /* pkt is now blank (av_interleaved_write_frame() takes ownership of
-         * its contents and resets pkt), so that no unreferencing is necessary.
-         * This would be different if one used av_write_frame(). */
+        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);
@@ -123,7 +121,7 @@ static int write_frame(AVFormatContext *fmt_ctx, AVCodecContext *c,

 /* Add an output stream. */
 static void add_stream(OutputStream *ost, AVFormatContext *oc,
-                       const AVCodec **codec,
+                       AVCodec **codec,
                       enum AVCodecID codec_id)
 {
    AVCodecContext *c;
@@ -137,12 +135,6 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
        exit(1);
    }

-    ost->tmp_pkt = av_packet_alloc();
-    if (!ost->tmp_pkt) {
-        fprintf(stderr, "Could not allocate AVPacket\n");
-        exit(1);
-    }
-
    ost->st = avformat_new_stream(oc, NULL);
    if (!ost->st) {
        fprintf(stderr, "Could not allocate stream\n");
@@ -169,7 +161,16 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
                    c->sample_rate = 44100;
            }
        }
-        av_channel_layout_copy(&c->ch_layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO);
+        c->channels        = av_get_channel_layout_nb_channels(c->channel_layout);
+        c->channel_layout = AV_CH_LAYOUT_STEREO;
+        if ((*codec)->channel_layouts) {
+            c->channel_layout = (*codec)->channel_layouts[0];
+            for (i = 0; (*codec)->channel_layouts[i]; i++) {
+                if ((*codec)->channel_layouts[i] == AV_CH_LAYOUT_STEREO)
+                    c->channel_layout = AV_CH_LAYOUT_STEREO;
+            }
+        }
+        c->channels        = av_get_channel_layout_nb_channels(c->channel_layout);
        ost->st->time_base = (AVRational){ 1, c->sample_rate };
        break;

@@ -214,22 +215,25 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
 /* audio output */

 static AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt,
-                                  const AVChannelLayout *channel_layout,
+                                  uint64_t channel_layout,
                                  int sample_rate, int nb_samples)
 {
    AVFrame *frame = av_frame_alloc();
+    int ret;
+
    if (!frame) {
        fprintf(stderr, "Error allocating an audio frame\n");
        exit(1);
    }

    frame->format = sample_fmt;
-    av_channel_layout_copy(&frame->ch_layout, channel_layout);
+    frame->channel_layout = channel_layout;
    frame->sample_rate = sample_rate;
    frame->nb_samples = nb_samples;

    if (nb_samples) {
-        if (av_frame_get_buffer(frame, 0) < 0) {
+        ret = av_frame_get_buffer(frame, 0);
+        if (ret < 0) {
            fprintf(stderr, "Error allocating an audio buffer\n");
            exit(1);
        }
@@ -238,8 +242,7 @@ static AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt,
    return frame;
 }

-static void open_audio(AVFormatContext *oc, const AVCodec *codec,
-                       OutputStream *ost, AVDictionary *opt_arg)
+static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
 {
    AVCodecContext *c;
    int nb_samples;
@@ -268,9 +271,9 @@ static void open_audio(AVFormatContext *oc, const AVCodec *codec,
    else
        nb_samples = c->frame_size;

-    ost->frame     = alloc_audio_frame(c->sample_fmt, &c->ch_layout,
+    ost->frame     = alloc_audio_frame(c->sample_fmt, c->channel_layout,
                                       c->sample_rate, nb_samples);
-    ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, &c->ch_layout,
+    ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, c->channel_layout,
                                       c->sample_rate, nb_samples);

    /* copy the stream parameters to the muxer */
@@ -288,10 +291,10 @@ static void open_audio(AVFormatContext *oc, const AVCodec *codec,
    }

    /* set options */
-    av_opt_set_chlayout  (ost->swr_ctx, "in_chlayout",       &c->ch_layout,      0);
+    av_opt_set_int       (ost->swr_ctx, "in_channel_count",   c->channels,       0);
    av_opt_set_int       (ost->swr_ctx, "in_sample_rate",     c->sample_rate,    0);
    av_opt_set_sample_fmt(ost->swr_ctx, "in_sample_fmt",      AV_SAMPLE_FMT_S16, 0);
-    av_opt_set_chlayout  (ost->swr_ctx, "out_chlayout",      &c->ch_layout,      0);
+    av_opt_set_int       (ost->swr_ctx, "out_channel_count",  c->channels,       0);
    av_opt_set_int       (ost->swr_ctx, "out_sample_rate",    c->sample_rate,    0);
    av_opt_set_sample_fmt(ost->swr_ctx, "out_sample_fmt",     c->sample_fmt,     0);

@@ -317,7 +320,7 @@ static AVFrame *get_audio_frame(OutputStream *ost)

    for (j = 0; j <frame->nb_samples; j++) {
        v = (int)(sin(ost->t) * 10000);
-        for (i = 0; i < ost->enc->ch_layout.nb_channels; i++)
+        for (i = 0; i < ost->enc->channels; i++)
            *q++ = v;
        ost->t     += ost->tincr;
        ost->tincr += ost->tincr2;
@@ -347,8 +350,7 @@ static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
    if (frame) {
        /* convert samples from native format to destination codec format, using the resampler */
        /* compute destination number of samples */
-        dst_nb_samples = av_rescale_rnd(swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples,
-                                        c->sample_rate, c->sample_rate, AV_ROUND_UP);
+        dst_nb_samples = swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples;
        av_assert0(dst_nb_samples == frame->nb_samples);

        /* when we pass a frame to the encoder, it may keep a reference to it
@@ -373,37 +375,36 @@ static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
        ost->samples_count += dst_nb_samples;
    }

-    return write_frame(oc, c, ost->st, frame, ost->tmp_pkt);
+    return write_frame(oc, c, ost->st, frame);
 }

 /**************************************************************/
 /* video output */

-static AVFrame *alloc_frame(enum AVPixelFormat pix_fmt, int width, int height)
+static AVFrame *alloc_picture(enum AVPixelFormat pix_fmt, int width, int height)
 {
-    AVFrame *frame;
+    AVFrame *picture;
    int ret;

-    frame = av_frame_alloc();
-    if (!frame)
+    picture = av_frame_alloc();
+    if (!picture)
        return NULL;

-    frame->format = pix_fmt;
-    frame->width  = width;
-    frame->height = height;
+    picture->format = pix_fmt;
+    picture->width  = width;
+    picture->height = height;

    /* allocate the buffers for the frame data */
-    ret = av_frame_get_buffer(frame, 0);
+    ret = av_frame_get_buffer(picture, 0);
    if (ret < 0) {
        fprintf(stderr, "Could not allocate frame data.\n");
        exit(1);
    }

-    return frame;
+    return picture;
 }

-static void open_video(AVFormatContext *oc, const AVCodec *codec,
-                       OutputStream *ost, AVDictionary *opt_arg)
+static void open_video(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
 {
    int ret;
    AVCodecContext *c = ost->enc;
@@ -420,7 +421,7 @@ static void open_video(AVFormatContext *oc, const AVCodec *codec,
    }

    /* allocate and init a re-usable frame */
-    ost->frame = alloc_frame(c->pix_fmt, c->width, c->height);
+    ost->frame = alloc_picture(c->pix_fmt, c->width, c->height);
    if (!ost->frame) {
        fprintf(stderr, "Could not allocate video frame\n");
        exit(1);
@@ -431,9 +432,9 @@ static void open_video(AVFormatContext *oc, const AVCodec *codec,
     * output format. */
    ost->tmp_frame = NULL;
    if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
-        ost->tmp_frame = alloc_frame(AV_PIX_FMT_YUV420P, c->width, c->height);
+        ost->tmp_frame = alloc_picture(AV_PIX_FMT_YUV420P, c->width, c->height);
        if (!ost->tmp_frame) {
-            fprintf(stderr, "Could not allocate temporary video frame\n");
+            fprintf(stderr, "Could not allocate temporary picture\n");
            exit(1);
        }
    }
@@ -516,7 +517,7 @@ static AVFrame *get_video_frame(OutputStream *ost)
 */
 static int write_video_frame(AVFormatContext *oc, OutputStream *ost)
 {
-    return write_frame(oc, ost->enc, ost->st, get_video_frame(ost), ost->tmp_pkt);
+    return write_frame(oc, ost->enc, ost->st, get_video_frame(ost));
 }

 static void close_stream(AVFormatContext *oc, OutputStream *ost)
@@ -524,7 +525,6 @@ static void close_stream(AVFormatContext *oc, OutputStream *ost)
    avcodec_free_context(&ost->enc);
    av_frame_free(&ost->frame);
    av_frame_free(&ost->tmp_frame);
-    av_packet_free(&ost->tmp_pkt);
    sws_freeContext(ost->sws_ctx);
    swr_free(&ost->swr_ctx);
 }
@@ -535,10 +535,10 @@ static void close_stream(AVFormatContext *oc, OutputStream *ost)
 int main(int argc, char **argv)
 {
    OutputStream video_st = { 0 }, audio_st = { 0 };
-    const AVOutputFormat *fmt;
    const char *filename;
+    AVOutputFormat *fmt;
    AVFormatContext *oc;
-    const AVCodec *audio_codec, *video_codec;
+    AVCodec *audio_codec, *video_codec;
    int ret;
    int have_video = 0, have_audio = 0;
    int encode_video = 0, encode_audio = 0;
@@ -625,6 +625,10 @@ int main(int argc, char **argv)
        }
    }

+    /* Write the trailer, if any. The trailer must be written before you
+     * close the CodecContexts open when you wrote the header; otherwise
+     * av_write_trailer() may try to use memory that was freed on
+     * av_codec_close(). */
    av_write_trailer(oc);

    /* Close each codec. */
--- a/doc/examples/qsv_transcode.c
+++ b/doc/examples/qsv_transcode.c
@@ -1,438 +0,0 @@
-/*
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to deal
- * in the Software without restriction, including without limitation the rights
- * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- * copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
- * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
- * THE SOFTWARE.
- */
-
-/**
- * @file Intel QSV-accelerated video transcoding API usage example
- * @example qsv_transcode.c
- *
- * Perform QSV-accelerated transcoding and show to dynamically change
- * encoder's options.
- *
- * Usage: qsv_transcode input_stream codec output_stream initial option
- *                      { frame_number new_option }
- * e.g: - qsv_transcode input.mp4 h264_qsv output_h264.mp4 "g 60"
- *      - qsv_transcode input.mp4 hevc_qsv output_hevc.mp4 "g 60 async_depth 1"
- *                      100 "g 120"
- *         (initialize codec with gop_size 60 and change it to 120 after 100
- *          frames)
- */
-
-#include <stdio.h>
-#include <errno.h>
-
-#include <libavutil/hwcontext.h>
-#include <libavcodec/avcodec.h>
-#include <libavformat/avformat.h>
-#include <libavutil/opt.h>
-
-static AVFormatContext *ifmt_ctx = NULL, *ofmt_ctx = NULL;
-static AVBufferRef *hw_device_ctx = NULL;
-static AVCodecContext *decoder_ctx = NULL, *encoder_ctx = NULL;
-static int video_stream = -1;
-
-typedef struct DynamicSetting {
-    int frame_number;
-    char* optstr;
-} DynamicSetting;
-static DynamicSetting *dynamic_setting;
-static int setting_number;
-static int current_setting_number;
-
-static int str_to_dict(char* optstr, AVDictionary **opt)
-{
-    char *key, *value;
-    if (strlen(optstr) == 0)
-        return 0;
-    key = strtok(optstr, " ");
-    if (key == NULL)
-        return AVERROR(ENAVAIL);
-    value = strtok(NULL, " ");
-    if (value == NULL)
-        return AVERROR(ENAVAIL);
-    av_dict_set(opt, key, value, 0);
-    do {
-        key = strtok(NULL, " ");
-        if (key == NULL)
-            return 0;
-        value = strtok(NULL, " ");
-        if (value == NULL)
-            return AVERROR(ENAVAIL);
-        av_dict_set(opt, key, value, 0);
-    } while(key != NULL);
-    return 0;
-}
-
-static int dynamic_set_parameter(AVCodecContext *avctx)
-{
-    AVDictionary *opts = NULL;
-    int ret = 0;
-    static int frame_number = 0;
-    frame_number++;
-    if (current_setting_number < setting_number &&
-        frame_number == dynamic_setting[current_setting_number].frame_number) {
-        AVDictionaryEntry *e = NULL;
-        ret = str_to_dict(dynamic_setting[current_setting_number++].optstr, &opts);
-        if (ret < 0) {
-            fprintf(stderr, "The dynamic parameter is wrong\n");
-            goto fail;
-        }
-        /* Set common option. The dictionary will be freed and replaced
-         * by a new one containing all options not found in common option list.
-         * Then this new dictionary is used to set private option. */
-        if ((ret = av_opt_set_dict(avctx, &opts)) < 0)
-            goto fail;
-        /* Set codec specific option */
-        if ((ret = av_opt_set_dict(avctx->priv_data, &opts)) < 0)
-            goto fail;
-        /* There is no "framerate" option in commom option list. Use "-r" to set
-         * framerate, which is compatible with ffmpeg commandline. The video is
-         * assumed to be average frame rate, so set time_base to 1/framerate. */
-        e = av_dict_get(opts, "r", NULL, 0);
-        if (e) {
-            avctx->framerate = av_d2q(atof(e->value), INT_MAX);
-            encoder_ctx->time_base = av_inv_q(encoder_ctx->framerate);
-        }
-    }
-fail:
-    av_dict_free(&opts);
-    return ret;
-}
-
-static int get_format(AVCodecContext *avctx, const enum AVPixelFormat *pix_fmts)
-{
-    while (*pix_fmts != AV_PIX_FMT_NONE) {
-        if (*pix_fmts == AV_PIX_FMT_QSV) {
-            return AV_PIX_FMT_QSV;
-        }
-
-        pix_fmts++;
-    }
-
-    fprintf(stderr, "The QSV pixel format not offered in get_format()\n");
-
-    return AV_PIX_FMT_NONE;
-}
-
-static int open_input_file(char *filename)
-{
-    int ret;
-    const AVCodec *decoder = NULL;
-    AVStream *video = NULL;
-
-    if ((ret = avformat_open_input(&ifmt_ctx, filename, NULL, NULL)) < 0) {
-        fprintf(stderr, "Cannot open input file '%s', Error code: %s\n",
-                filename, av_err2str(ret));
-        return ret;
-    }
-
-    if ((ret = avformat_find_stream_info(ifmt_ctx, NULL)) < 0) {
-        fprintf(stderr, "Cannot find input stream information. Error code: %s\n",
-                av_err2str(ret));
-        return ret;
-    }
-
-    ret = av_find_best_stream(ifmt_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, NULL, 0);
-    if (ret < 0) {
-        fprintf(stderr, "Cannot find a video stream in the input file. "
-                "Error code: %s\n", av_err2str(ret));
-        return ret;
-    }
-    video_stream = ret;
-    video = ifmt_ctx->streams[video_stream];
-
-    switch(video->codecpar->codec_id) {
-    case AV_CODEC_ID_H264:
-        decoder = avcodec_find_decoder_by_name("h264_qsv");
-        break;
-    case AV_CODEC_ID_HEVC:
-        decoder = avcodec_find_decoder_by_name("hevc_qsv");
-        break;
-    case AV_CODEC_ID_VP9:
-        decoder = avcodec_find_decoder_by_name("vp9_qsv");
-        break;
-    case AV_CODEC_ID_VP8:
-        decoder = avcodec_find_decoder_by_name("vp8_qsv");
-        break;
-    case AV_CODEC_ID_AV1:
-        decoder = avcodec_find_decoder_by_name("av1_qsv");
-        break;
-    case AV_CODEC_ID_MPEG2VIDEO:
-        decoder = avcodec_find_decoder_by_name("mpeg2_qsv");
-        break;
-    case AV_CODEC_ID_MJPEG:
-        decoder = avcodec_find_decoder_by_name("mjpeg_qsv");
-        break;
-    default:
-        fprintf(stderr, "Codec is not supportted by qsv\n");
-        return AVERROR(ENAVAIL);
-    }
-
-    if (!(decoder_ctx = avcodec_alloc_context3(decoder)))
-        return AVERROR(ENOMEM);
-
-    if ((ret = avcodec_parameters_to_context(decoder_ctx, video->codecpar)) < 0) {
-        fprintf(stderr, "avcodec_parameters_to_context error. Error code: %s\n",
-                av_err2str(ret));
-        return ret;
-    }
-    decoder_ctx->framerate = av_guess_frame_rate(ifmt_ctx, video, NULL);
-
-    decoder_ctx->hw_device_ctx = av_buffer_ref(hw_device_ctx);
-    if (!decoder_ctx->hw_device_ctx) {
-        fprintf(stderr, "A hardware device reference create failed.\n");
-        return AVERROR(ENOMEM);
-    }
-    decoder_ctx->get_format    = get_format;
-    decoder_ctx->pkt_timebase = video->time_base;
-    if ((ret = avcodec_open2(decoder_ctx, decoder, NULL)) < 0)
-        fprintf(stderr, "Failed to open codec for decoding. Error code: %s\n",
-                av_err2str(ret));
-
-    return ret;
-}
-
-static int encode_write(AVPacket *enc_pkt, AVFrame *frame)
-{
-    int ret = 0;
-
-    av_packet_unref(enc_pkt);
-
-    if((ret = dynamic_set_parameter(encoder_ctx)) < 0) {
-        fprintf(stderr, "Failed to set dynamic parameter. Error code: %s\n",
-                av_err2str(ret));
-        goto end;
-    }
-
-    if ((ret = avcodec_send_frame(encoder_ctx, frame)) < 0) {
-        fprintf(stderr, "Error during encoding. Error code: %s\n", av_err2str(ret));
-        goto end;
-    }
-    while (1) {
-        if (ret = avcodec_receive_packet(encoder_ctx, enc_pkt))
-            break;
-        enc_pkt->stream_index = 0;
-        av_packet_rescale_ts(enc_pkt, encoder_ctx->time_base,
-                             ofmt_ctx->streams[0]->time_base);
-        if ((ret = av_interleaved_write_frame(ofmt_ctx, enc_pkt)) < 0) {
-            fprintf(stderr, "Error during writing data to output file. "
-                    "Error code: %s\n", av_err2str(ret));
-            return ret;
-        }
-    }
-
-end:
-    if (ret == AVERROR_EOF)
-        return 0;
-    ret = ((ret == AVERROR(EAGAIN)) ? 0:-1);
-    return ret;
-}
-
-static int dec_enc(AVPacket *pkt, const AVCodec *enc_codec, char *optstr)
-{
-    AVFrame *frame;
-    int ret = 0;
-
-    ret = avcodec_send_packet(decoder_ctx, pkt);
-    if (ret < 0) {
-        fprintf(stderr, "Error during decoding. Error code: %s\n", av_err2str(ret));
-        return ret;
-    }
-
-    while (ret >= 0) {
-        if (!(frame = av_frame_alloc()))
-            return AVERROR(ENOMEM);
-
-        ret = avcodec_receive_frame(decoder_ctx, frame);
-        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
-            av_frame_free(&frame);
-            return 0;
-        } else if (ret < 0) {
-            fprintf(stderr, "Error while decoding. Error code: %s\n", av_err2str(ret));
-            goto fail;
-        }
-        if (!encoder_ctx->hw_frames_ctx) {
-            AVDictionaryEntry *e = NULL;
-            AVDictionary *opts = NULL;
-            AVStream *ost;
-            /* we need to ref hw_frames_ctx of decoder to initialize encoder's codec.
-               Only after we get a decoded frame, can we obtain its hw_frames_ctx */
-            encoder_ctx->hw_frames_ctx = av_buffer_ref(decoder_ctx->hw_frames_ctx);
-            if (!encoder_ctx->hw_frames_ctx) {
-                ret = AVERROR(ENOMEM);
-                goto fail;
-            }
-            /* set AVCodecContext Parameters for encoder, here we keep them stay
-             * the same as decoder.
-             */
-            encoder_ctx->time_base = av_inv_q(decoder_ctx->framerate);
-            encoder_ctx->pix_fmt   = AV_PIX_FMT_QSV;
-            encoder_ctx->width     = decoder_ctx->width;
-            encoder_ctx->height    = decoder_ctx->height;
-            if ((ret = str_to_dict(optstr, &opts)) < 0) {
-                fprintf(stderr, "Failed to set encoding parameter.\n");
-                goto fail;
-            }
-            /* There is no "framerate" option in commom option list. Use "-r" to
-            * set framerate, which is compatible with ffmpeg commandline. The
-            * video is assumed to be average frame rate, so set time_base to
-            * 1/framerate. */
-            e = av_dict_get(opts, "r", NULL, 0);
-            if (e) {
-                encoder_ctx->framerate = av_d2q(atof(e->value), INT_MAX);
-                encoder_ctx->time_base = av_inv_q(encoder_ctx->framerate);
-            }
-            if ((ret = avcodec_open2(encoder_ctx, enc_codec, &opts)) < 0) {
-                fprintf(stderr, "Failed to open encode codec. Error code: %s\n",
-                        av_err2str(ret));
-                av_dict_free(&opts);
-                goto fail;
-            }
-            av_dict_free(&opts);
-
-            if (!(ost = avformat_new_stream(ofmt_ctx, enc_codec))) {
-                fprintf(stderr, "Failed to allocate stream for output format.\n");
-                ret = AVERROR(ENOMEM);
-                goto fail;
-            }
-
-            ost->time_base = encoder_ctx->time_base;
-            ret = avcodec_parameters_from_context(ost->codecpar, encoder_ctx);
-            if (ret < 0) {
-                fprintf(stderr, "Failed to copy the stream parameters. "
-                        "Error code: %s\n", av_err2str(ret));
-                goto fail;
-            }
-
-            /* write the stream header */
-            if ((ret = avformat_write_header(ofmt_ctx, NULL)) < 0) {
-                fprintf(stderr, "Error while writing stream header. "
-                        "Error code: %s\n", av_err2str(ret));
-                goto fail;
-            }
-        }
-        frame->pts = av_rescale_q(frame->pts, decoder_ctx->pkt_timebase,
-                                  encoder_ctx->time_base);
-        if ((ret = encode_write(pkt, frame)) < 0)
-            fprintf(stderr, "Error during encoding and writing.\n");
-
-fail:
-        av_frame_free(&frame);
-        if (ret < 0)
-            return ret;
-    }
-    return 0;
-}
-
-int main(int argc, char **argv)
-{
-    const AVCodec *enc_codec;
-    int ret = 0;
-    AVPacket *dec_pkt;
-
-    if (argc < 5 || (argc - 5) % 2) {
-        av_log(NULL, AV_LOG_ERROR, "Usage: %s <input file> <encoder> <output file>"
-               " <\"encoding option set 0\"> [<frame_number> <\"encoding options set 1\">]...\n", argv[0]);
-        return 1;
-    }
-    setting_number = (argc - 5) / 2;
-    dynamic_setting = av_malloc(setting_number * sizeof(*dynamic_setting));
-    current_setting_number = 0;
-    for (int i = 0; i < setting_number; i++) {
-        dynamic_setting[i].frame_number = atoi(argv[i*2 + 5]);
-        dynamic_setting[i].optstr = argv[i*2 + 6];
-    }
-
-    ret = av_hwdevice_ctx_create(&hw_device_ctx, AV_HWDEVICE_TYPE_QSV, NULL, NULL, 0);
-    if (ret < 0) {
-        fprintf(stderr, "Failed to create a QSV device. Error code: %s\n", av_err2str(ret));
-        goto end;
-    }
-
-    dec_pkt = av_packet_alloc();
-    if (!dec_pkt) {
-        fprintf(stderr, "Failed to allocate decode packet\n");
-        goto end;
-    }
-
-    if ((ret = open_input_file(argv[1])) < 0)
-        goto end;
-
-    if (!(enc_codec = avcodec_find_encoder_by_name(argv[2]))) {
-        fprintf(stderr, "Could not find encoder '%s'\n", argv[2]);
-        ret = -1;
-        goto end;
-    }
-
-    if ((ret = (avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, argv[3]))) < 0) {
-        fprintf(stderr, "Failed to deduce output format from file extension. Error code: "
-                "%s\n", av_err2str(ret));
-        goto end;
-    }
-
-    if (!(encoder_ctx = avcodec_alloc_context3(enc_codec))) {
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-
-    ret = avio_open(&ofmt_ctx->pb, argv[3], AVIO_FLAG_WRITE);
-    if (ret < 0) {
-        fprintf(stderr, "Cannot open output file. "
-                "Error code: %s\n", av_err2str(ret));
-        goto end;
-    }
-
-    /* read all packets and only transcoding video */
-    while (ret >= 0) {
-        if ((ret = av_read_frame(ifmt_ctx, dec_pkt)) < 0)
-            break;
-
-        if (video_stream == dec_pkt->stream_index)
-            ret = dec_enc(dec_pkt, enc_codec, argv[4]);
-
-        av_packet_unref(dec_pkt);
-    }
-
-    /* flush decoder */
-    av_packet_unref(dec_pkt);
-    if ((ret = dec_enc(dec_pkt, enc_codec, argv[4])) < 0) {
-        fprintf(stderr, "Failed to flush decoder %s\n", av_err2str(ret));
-        goto end;
-    }
-
-    /* flush encoder */
-    if ((ret = encode_write(dec_pkt, NULL)) < 0) {
-        fprintf(stderr, "Failed to flush encoder %s\n", av_err2str(ret));
-        goto end;
-    }
-
-    /* write the trailer for output stream */
-    if ((ret = av_write_trailer(ofmt_ctx)) < 0)
-        fprintf(stderr, "Failed to write trailer %s\n", av_err2str(ret));
-
-end:
-    avformat_close_input(&ifmt_ctx);
-    avformat_close_input(&ofmt_ctx);
-    avcodec_free_context(&decoder_ctx);
-    avcodec_free_context(&encoder_ctx);
-    av_buffer_unref(&hw_device_ctx);
-    av_packet_free(&dec_pkt);
-    av_freep(&dynamic_setting);
-    return ret;
-}
--- a/doc/examples/qsv_decode.c
+++ b/doc/examples/qsv_decode.c
@@ -21,11 +21,12 @@
 */

 /**
- * @file Intel QSV-accelerated H.264 decoding API usage example
- * @example qsv_decode.c
+ * @file
+ * Intel QSV-accelerated H.264 decoding example.
 *
- * Perform QSV-accelerated H.264 decoding with output frames in the
- * GPU video surfaces, write the decoded frames to an output file.
+ * @example qsvdec.c
+ * This example shows how to do QSV-accelerated H.264 decoding with output
+ * frames in the GPU video surfaces.
 */

 #include "config.h"
@@ -43,10 +44,38 @@
 #include "libavutil/hwcontext_qsv.h"
 #include "libavutil/mem.h"

+typedef struct DecodeContext {
+    AVBufferRef *hw_device_ref;
+} DecodeContext;
+
 static int get_format(AVCodecContext *avctx, const enum AVPixelFormat *pix_fmts)
 {
    while (*pix_fmts != AV_PIX_FMT_NONE) {
        if (*pix_fmts == AV_PIX_FMT_QSV) {
+            DecodeContext *decode = avctx->opaque;
+            AVHWFramesContext  *frames_ctx;
+            AVQSVFramesContext *frames_hwctx;
+            int ret;
+
+            /* create a pool of surfaces to be used by the decoder */
+            avctx->hw_frames_ctx = av_hwframe_ctx_alloc(decode->hw_device_ref);
+            if (!avctx->hw_frames_ctx)
+                return AV_PIX_FMT_NONE;
+            frames_ctx   = (AVHWFramesContext*)avctx->hw_frames_ctx->data;
+            frames_hwctx = frames_ctx->hwctx;
+
+            frames_ctx->format            = AV_PIX_FMT_QSV;
+            frames_ctx->sw_format         = avctx->sw_pix_fmt;
+            frames_ctx->width             = FFALIGN(avctx->coded_width,  32);
+            frames_ctx->height            = FFALIGN(avctx->coded_height, 32);
+            frames_ctx->initial_pool_size = 32;
+
+            frames_hwctx->frame_type = MFX_MEMTYPE_VIDEO_MEMORY_DECODER_TARGET;
+
+            ret = av_hwframe_ctx_init(avctx->hw_frames_ctx);
+            if (ret < 0)
+                return AV_PIX_FMT_NONE;
+
            return AV_PIX_FMT_QSV;
        }

@@ -58,7 +87,7 @@ static int get_format(AVCodecContext *avctx, const enum AVPixelFormat *pix_fmts)
    return AV_PIX_FMT_NONE;
 }

-static int decode_packet(AVCodecContext *decoder_ctx,
+static int decode_packet(DecodeContext *decode, AVCodecContext *decoder_ctx,
                         AVFrame *frame, AVFrame *sw_frame,
                         AVPacket *pkt, AVIOContext *output_ctx)
 {
@@ -112,15 +141,15 @@ int main(int argc, char **argv)
    AVCodecContext *decoder_ctx = NULL;
    const AVCodec *decoder;

-    AVPacket *pkt = NULL;
+    AVPacket pkt = { 0 };
    AVFrame *frame = NULL, *sw_frame = NULL;

+    DecodeContext decode = { NULL };
+
    AVIOContext *output_ctx = NULL;

    int ret, i;

-    AVBufferRef *device_ref = NULL;
-
    if (argc < 3) {
        fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
        return 1;
@@ -148,7 +177,7 @@ int main(int argc, char **argv)
    }

    /* open the hardware device */
-    ret = av_hwdevice_ctx_create(&device_ref, AV_HWDEVICE_TYPE_QSV,
+    ret = av_hwdevice_ctx_create(&decode.hw_device_ref, AV_HWDEVICE_TYPE_QSV,
                                 "auto", NULL, 0);
    if (ret < 0) {
        fprintf(stderr, "Cannot open the hardware device\n");
@@ -180,8 +209,7 @@ int main(int argc, char **argv)
        decoder_ctx->extradata_size = video_st->codecpar->extradata_size;
    }

-
-    decoder_ctx->hw_device_ctx = av_buffer_ref(device_ref);
+    decoder_ctx->opaque      = &decode;
    decoder_ctx->get_format  = get_format;

    ret = avcodec_open2(decoder_ctx, NULL, NULL);
@@ -199,26 +227,27 @@ int main(int argc, char **argv)

    frame    = av_frame_alloc();
    sw_frame = av_frame_alloc();
-    pkt      = av_packet_alloc();
-    if (!frame || !sw_frame || !pkt) {
+    if (!frame || !sw_frame) {
        ret = AVERROR(ENOMEM);
        goto finish;
    }

    /* actual decoding */
    while (ret >= 0) {
-        ret = av_read_frame(input_ctx, pkt);
+        ret = av_read_frame(input_ctx, &pkt);
        if (ret < 0)
            break;

-        if (pkt->stream_index == video_st->index)
-            ret = decode_packet(decoder_ctx, frame, sw_frame, pkt, output_ctx);
+        if (pkt.stream_index == video_st->index)
+            ret = decode_packet(&decode, decoder_ctx, frame, sw_frame, &pkt, output_ctx);

-        av_packet_unref(pkt);
+        av_packet_unref(&pkt);
    }

    /* flush the decoder */
-    ret = decode_packet(decoder_ctx, frame, sw_frame, NULL, output_ctx);
+    pkt.data = NULL;
+    pkt.size = 0;
+    ret = decode_packet(&decode, decoder_ctx, frame, sw_frame, &pkt, output_ctx);

 finish:
    if (ret < 0) {
@@ -231,11 +260,10 @@ finish:

    av_frame_free(&frame);
    av_frame_free(&sw_frame);
-    av_packet_free(&pkt);

    avcodec_free_context(&decoder_ctx);

-    av_buffer_unref(&device_ref);
+    av_buffer_unref(&decode.hw_device_ref);

    avio_close(output_ctx);

--- a/doc/examples/remuxing.c
+++ b/doc/examples/remuxing.c
@@ -21,11 +21,11 @@
 */

 /**
- * @file libavformat/libavcodec demuxing and muxing API usage example
- * @example remux.c
+ * @file
+ * libavformat/libavcodec demuxing and muxing API example.
 *
- * Remux streams from one container format to another. Data is copied from the
- * input to the output without transcoding.
+ * Remux streams from one container format to another.
+ * @example remuxing.c
 */

 #include <libavutil/timestamp.h>
@@ -45,9 +45,9 @@ static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt, cons

 int main(int argc, char **argv)
 {
-    const AVOutputFormat *ofmt = NULL;
+    AVOutputFormat *ofmt = NULL;
    AVFormatContext *ifmt_ctx = NULL, *ofmt_ctx = NULL;
-    AVPacket *pkt = NULL;
+    AVPacket pkt;
    const char *in_filename, *out_filename;
    int ret, i;
    int stream_index = 0;
@@ -65,12 +65,6 @@ int main(int argc, char **argv)
    in_filename  = argv[1];
    out_filename = argv[2];

-    pkt = av_packet_alloc();
-    if (!pkt) {
-        fprintf(stderr, "Could not allocate AVPacket\n");
-        return 1;
-    }
-
    if ((ret = avformat_open_input(&ifmt_ctx, in_filename, 0, 0)) < 0) {
        fprintf(stderr, "Could not open input file '%s'", in_filename);
        goto end;
@@ -91,7 +85,7 @@ int main(int argc, char **argv)
    }

    stream_mapping_size = ifmt_ctx->nb_streams;
-    stream_mapping = av_calloc(stream_mapping_size, sizeof(*stream_mapping));
+    stream_mapping = av_mallocz_array(stream_mapping_size, sizeof(*stream_mapping));
    if (!stream_mapping) {
        ret = AVERROR(ENOMEM);
        goto end;
@@ -146,39 +140,38 @@ int main(int argc, char **argv)
    while (1) {
        AVStream *in_stream, *out_stream;

-        ret = av_read_frame(ifmt_ctx, pkt);
+        ret = av_read_frame(ifmt_ctx, &pkt);
        if (ret < 0)
            break;

-        in_stream  = ifmt_ctx->streams[pkt->stream_index];
-        if (pkt->stream_index >= stream_mapping_size ||
-            stream_mapping[pkt->stream_index] < 0) {
-            av_packet_unref(pkt);
+        in_stream  = ifmt_ctx->streams[pkt.stream_index];
+        if (pkt.stream_index >= stream_mapping_size ||
+            stream_mapping[pkt.stream_index] < 0) {
+            av_packet_unref(&pkt);
            continue;
        }

-        pkt->stream_index = stream_mapping[pkt->stream_index];
-        out_stream = ofmt_ctx->streams[pkt->stream_index];
-        log_packet(ifmt_ctx, pkt, "in");
+        pkt.stream_index = stream_mapping[pkt.stream_index];
+        out_stream = ofmt_ctx->streams[pkt.stream_index];
+        log_packet(ifmt_ctx, &pkt, "in");

        /* copy packet */
-        av_packet_rescale_ts(pkt, in_stream->time_base, out_stream->time_base);
-        pkt->pos = -1;
-        log_packet(ofmt_ctx, pkt, "out");
+        pkt.pts = av_rescale_q_rnd(pkt.pts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
+        pkt.dts = av_rescale_q_rnd(pkt.dts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
+        pkt.duration = av_rescale_q(pkt.duration, in_stream->time_base, out_stream->time_base);
+        pkt.pos = -1;
+        log_packet(ofmt_ctx, &pkt, "out");

-        ret = av_interleaved_write_frame(ofmt_ctx, pkt);
-        /* pkt is now blank (av_interleaved_write_frame() takes ownership of
-         * its contents and resets pkt), so that no unreferencing is necessary.
-         * This would be different if one used av_write_frame(). */
+        ret = av_interleaved_write_frame(ofmt_ctx, &pkt);
        if (ret < 0) {
            fprintf(stderr, "Error muxing packet\n");
            break;
        }
+        av_packet_unref(&pkt);
    }

    av_write_trailer(ofmt_ctx);
 end:
-    av_packet_free(&pkt);

    avformat_close_input(&ifmt_ctx);

--- a/doc/examples/resampling_audio.c
+++ b/doc/examples/resampling_audio.c
@@ -21,12 +21,8 @@
 */

 /**
- * @file audio resampling API usage example
- * @example resample_audio.c
- *
- * Generate a synthetic audio signal, and Use libswresample API to perform audio
- * resampling. The output is written to a raw audio file to be played with
- * ffplay.
+ * @example resampling_audio.c
+ * libswresample API use example.
 */

 #include <libavutil/opt.h>
@@ -84,7 +80,7 @@ static void fill_samples(double *dst, int nb_samples, int nb_channels, int sampl

 int main(int argc, char **argv)
 {
-    AVChannelLayout src_ch_layout = AV_CHANNEL_LAYOUT_STEREO, dst_ch_layout = AV_CHANNEL_LAYOUT_SURROUND;
+    int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND;
    int src_rate = 48000, dst_rate = 44100;
    uint8_t **src_data = NULL, **dst_data = NULL;
    int src_nb_channels = 0, dst_nb_channels = 0;
@@ -96,7 +92,6 @@ int main(int argc, char **argv)
    int dst_bufsize;
    const char *fmt;
    struct SwrContext *swr_ctx;
-    char buf[64];
    double t;
    int ret;

@@ -125,11 +120,11 @@ int main(int argc, char **argv)
    }

    /* set options */
-    av_opt_set_chlayout(swr_ctx, "in_chlayout",    &src_ch_layout, 0);
+    av_opt_set_int(swr_ctx, "in_channel_layout",    src_ch_layout, 0);
    av_opt_set_int(swr_ctx, "in_sample_rate",       src_rate, 0);
    av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);

-    av_opt_set_chlayout(swr_ctx, "out_chlayout",    &dst_ch_layout, 0);
+    av_opt_set_int(swr_ctx, "out_channel_layout",    dst_ch_layout, 0);
    av_opt_set_int(swr_ctx, "out_sample_rate",       dst_rate, 0);
    av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);

@@ -141,7 +136,7 @@ int main(int argc, char **argv)

    /* allocate source and destination samples buffers */

-    src_nb_channels = src_ch_layout.nb_channels;
+    src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
    ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_nb_channels,
                                             src_nb_samples, src_sample_fmt, 0);
    if (ret < 0) {
@@ -156,7 +151,7 @@ int main(int argc, char **argv)
        av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);

    /* buffer is going to be directly written to a rawaudio file, no alignment */
-    dst_nb_channels = dst_ch_layout.nb_channels;
+    dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
    ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels,
                                             dst_nb_samples, dst_sample_fmt, 0);
    if (ret < 0) {
@@ -199,10 +194,9 @@ int main(int argc, char **argv)

    if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0)
        goto end;
-    av_channel_layout_describe(&dst_ch_layout, buf, sizeof(buf));
    fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"
-            "ffplay -f %s -channel_layout %s -channels %d -ar %d %s\n",
-            fmt, buf, dst_nb_channels, dst_rate, dst_filename);
+            "ffplay -f %s -channel_layout %"PRId64" -channels %d -ar %d %s\n",
+            fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);

 end:
    fclose(dst_file);
--- a/doc/examples/scaling_video.c
+++ b/doc/examples/scaling_video.c
@@ -21,10 +21,9 @@
 */

 /**
- * @file libswscale API usage example
- * @example scale_video.c
- *
- * Generate a synthetic video signal and use libswscale to perform rescaling.
+ * @file
+ * libswscale API use example.
+ * @example scaling_video.c
 */

 #include <libavutil/imgutils.h>
--- a/doc/examples/transcode_aac.c
+++ b/doc/examples/transcode_aac.c
@@ -1,5 +1,5 @@
 /*
- * Copyright (c) 2013-2022 Andreas Unterweger
+ * Copyright (c) 2013-2018 Andreas Unterweger
 *
 * This file is part of FFmpeg.
 *
@@ -19,11 +19,12 @@
 */

 /**
- * @file audio transcoding to MPEG/AAC API usage example
- * @example transcode_aac.c
+ * @file
+ * Simple audio converter
 *
- * Convert an input audio file to AAC in an MP4 container. Formats other than
- * MP4 are supported based on the output file extension.
+ * @example transcode_aac.c
+ * Convert an input audio file to AAC in an MP4 container using FFmpeg.
+ * Formats other than MP4 are supported based on the output file extension.
 * @author Andreas Unterweger (dustsigns@gmail.com)
 */

@@ -37,7 +38,6 @@
 #include "libavutil/audio_fifo.h"
 #include "libavutil/avassert.h"
 #include "libavutil/avstring.h"
-#include "libavutil/channel_layout.h"
 #include "libavutil/frame.h"
 #include "libavutil/opt.h"

@@ -60,8 +60,7 @@ static int open_input_file(const char *filename,
                           AVCodecContext **input_codec_context)
 {
    AVCodecContext *avctx;
-    const AVCodec *input_codec;
-    const AVStream *stream;
+    AVCodec *input_codec;
    int error;

    /* Open the input file to read from it. */
@@ -89,10 +88,8 @@ static int open_input_file(const char *filename,
        return AVERROR_EXIT;
    }

-    stream = (*input_format_context)->streams[0];
-
    /* Find a decoder for the audio stream. */
-    if (!(input_codec = avcodec_find_decoder(stream->codecpar->codec_id))) {
+    if (!(input_codec = avcodec_find_decoder((*input_format_context)->streams[0]->codecpar->codec_id))) {
        fprintf(stderr, "Could not find input codec\n");
        avformat_close_input(input_format_context);
        return AVERROR_EXIT;
@@ -107,7 +104,7 @@ static int open_input_file(const char *filename,
    }

    /* Initialize the stream parameters with demuxer information. */
-    error = avcodec_parameters_to_context(avctx, stream->codecpar);
+    error = avcodec_parameters_to_context(avctx, (*input_format_context)->streams[0]->codecpar);
    if (error < 0) {
        avformat_close_input(input_format_context);
        avcodec_free_context(&avctx);
@@ -123,9 +120,6 @@ static int open_input_file(const char *filename,
        return error;
    }

-    /* Set the packet timebase for the decoder. */
-    avctx->pkt_timebase = stream->time_base;
-
    /* Save the decoder context for easier access later. */
    *input_codec_context = avctx;

@@ -150,7 +144,7 @@ static int open_output_file(const char *filename,
    AVCodecContext *avctx          = NULL;
    AVIOContext *output_io_context = NULL;
    AVStream *stream               = NULL;
-    const AVCodec *output_codec    = NULL;
+    AVCodec *output_codec          = NULL;
    int error;

    /* Open the output file to write to it. */
@@ -205,11 +199,15 @@ static int open_output_file(const char *filename,

    /* Set the basic encoder parameters.
     * The input file's sample rate is used to avoid a sample rate conversion. */
-    av_channel_layout_default(&avctx->ch_layout, OUTPUT_CHANNELS);
+    avctx->channels       = OUTPUT_CHANNELS;
+    avctx->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
    avctx->sample_rate    = input_codec_context->sample_rate;
    avctx->sample_fmt     = output_codec->sample_fmts[0];
    avctx->bit_rate       = OUTPUT_BIT_RATE;

+    /* Allow the use of the experimental AAC encoder. */
+    avctx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
+
    /* Set the sample rate for the container. */
    stream->time_base.den = input_codec_context->sample_rate;
    stream->time_base.num = 1;
@@ -291,18 +289,21 @@ static int init_resampler(AVCodecContext *input_codec_context,
        /*
         * Create a resampler context for the conversion.
         * Set the conversion parameters.
+         * Default channel layouts based on the number of channels
+         * are assumed for simplicity (they are sometimes not detected
+         * properly by the demuxer and/or decoder).
         */
-        error = swr_alloc_set_opts2(resample_context,
-                                             &output_codec_context->ch_layout,
+        *resample_context = swr_alloc_set_opts(NULL,
+                                              av_get_default_channel_layout(output_codec_context->channels),
                                              output_codec_context->sample_fmt,
                                              output_codec_context->sample_rate,
-                                             &input_codec_context->ch_layout,
+                                              av_get_default_channel_layout(input_codec_context->channels),
                                              input_codec_context->sample_fmt,
                                              input_codec_context->sample_rate,
                                              0, NULL);
-        if (error < 0) {
+        if (!*resample_context) {
            fprintf(stderr, "Could not allocate resample context\n");
-            return error;
+            return AVERROR(ENOMEM);
        }
        /*
        * Perform a sanity check so that the number of converted samples is
@@ -330,7 +331,7 @@ static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
 {
    /* Create the FIFO buffer based on the specified output sample format. */
    if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
-                                      output_codec_context->ch_layout.nb_channels, 1))) {
+                                      output_codec_context->channels, 1))) {
        fprintf(stderr, "Could not allocate FIFO\n");
        return AVERROR(ENOMEM);
    }
@@ -379,8 +380,6 @@ static int decode_audio_frame(AVFrame *frame,
    if (error < 0)
        return error;

-    *data_present = 0;
-    *finished = 0;
    /* Read one audio frame from the input file into a temporary packet. */
    if ((error = av_read_frame(input_format_context, input_packet)) < 0) {
        /* If we are at the end of the file, flush the decoder below. */
@@ -447,17 +446,26 @@ static int init_converted_samples(uint8_t ***converted_input_samples,
    int error;

    /* Allocate as many pointers as there are audio channels.
-     * Each pointer will point to the audio samples of the corresponding
+     * Each pointer will later point to the audio samples of the corresponding
     * channels (although it may be NULL for interleaved formats).
-     * Allocate memory for the samples of all channels in one consecutive
+     */
+    if (!(*converted_input_samples = calloc(output_codec_context->channels,
+                                            sizeof(**converted_input_samples)))) {
+        fprintf(stderr, "Could not allocate converted input sample pointers\n");
+        return AVERROR(ENOMEM);
+    }
+
+    /* Allocate memory for the samples of all channels in one consecutive
     * block for convenience. */
-    if ((error = av_samples_alloc_array_and_samples(converted_input_samples, NULL,
-                                  output_codec_context->ch_layout.nb_channels,
+    if ((error = av_samples_alloc(*converted_input_samples, NULL,
+                                  output_codec_context->channels,
                                  frame_size,
                                  output_codec_context->sample_fmt, 0)) < 0) {
        fprintf(stderr,
                "Could not allocate converted input samples (error '%s')\n",
                av_err2str(error));
+        av_freep(&(*converted_input_samples)[0]);
+        free(*converted_input_samples);
        return error;
    }
    return 0;
@@ -550,7 +558,7 @@ static int read_decode_convert_and_store(AVAudioFifo *fifo,
    AVFrame *input_frame = NULL;
    /* Temporary storage for the converted input samples. */
    uint8_t **converted_input_samples = NULL;
-    int data_present;
+    int data_present = 0;
    int ret = AVERROR_EXIT;

    /* Initialize temporary storage for one input frame. */
@@ -589,9 +597,10 @@ static int read_decode_convert_and_store(AVAudioFifo *fifo,
    ret = 0;

 cleanup:
-    if (converted_input_samples)
+    if (converted_input_samples) {
        av_freep(&converted_input_samples[0]);
-    av_freep(&converted_input_samples);
+        free(converted_input_samples);
+    }
    av_frame_free(&input_frame);

    return ret;
@@ -623,7 +632,7 @@ static int init_output_frame(AVFrame **frame,
     * Default channel layouts based on the number of channels
     * are assumed for simplicity. */
    (*frame)->nb_samples     = frame_size;
-    av_channel_layout_copy(&(*frame)->ch_layout, &output_codec_context->ch_layout);
+    (*frame)->channel_layout = output_codec_context->channel_layout;
    (*frame)->format         = output_codec_context->sample_fmt;
    (*frame)->sample_rate    = output_codec_context->sample_rate;

@@ -670,16 +679,17 @@ static int encode_audio_frame(AVFrame *frame,
        pts += frame->nb_samples;
    }

-    *data_present = 0;
    /* Send the audio frame stored in the temporary packet to the encoder.
     * The output audio stream encoder is used to do this. */
    error = avcodec_send_frame(output_codec_context, frame);
-    /* Check for errors, but proceed with fetching encoded samples if the
-     *  encoder signals that it has nothing more to encode. */
-    if (error < 0 && error != AVERROR_EOF) {
-      fprintf(stderr, "Could not send packet for encoding (error '%s')\n",
-              av_err2str(error));
-      goto cleanup;
+    /* The encoder signals that it has nothing more to encode. */
+    if (error == AVERROR_EOF) {
+        error = 0;
+        goto cleanup;
+    } else if (error < 0) {
+        fprintf(stderr, "Could not send packet for encoding (error '%s')\n",
+                av_err2str(error));
+        goto cleanup;
    }

    /* Receive one encoded frame from the encoder. */
@@ -850,6 +860,7 @@ int main(int argc, char **argv)
            int data_written;
            /* Flush the encoder as it may have delayed frames. */
            do {
+                data_written = 0;
                if (encode_audio_frame(NULL, output_format_context,
                                       output_codec_context, &data_written))
                    goto cleanup;
--- a/doc/examples/transcoding.c
+++ b/doc/examples/transcoding.c
@@ -23,18 +23,15 @@
 */

 /**
- * @file demuxing, decoding, filtering, encoding and muxing API usage example
- * @example transcode.c
- *
- * Convert input to output file, applying some hard-coded filter-graph on both
- * audio and video streams.
+ * @file
+ * API example for demuxing, decoding, filtering, encoding and muxing
+ * @example transcoding.c
 */

 #include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
-#include <libavutil/channel_layout.h>
 #include <libavutil/opt.h>
 #include <libavutil/pixdesc.h>

@@ -74,13 +71,13 @@ static int open_input_file(const char *filename)
        return ret;
    }

-    stream_ctx = av_calloc(ifmt_ctx->nb_streams, sizeof(*stream_ctx));
+    stream_ctx = av_mallocz_array(ifmt_ctx->nb_streams, sizeof(*stream_ctx));
    if (!stream_ctx)
        return AVERROR(ENOMEM);

    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
        AVStream *stream = ifmt_ctx->streams[i];
-        const AVCodec *dec = avcodec_find_decoder(stream->codecpar->codec_id);
+        AVCodec *dec = avcodec_find_decoder(stream->codecpar->codec_id);
        AVCodecContext *codec_ctx;
        if (!dec) {
            av_log(NULL, AV_LOG_ERROR, "Failed to find decoder for stream #%u\n", i);
@@ -97,11 +94,6 @@ static int open_input_file(const char *filename)
                   "for stream #%u\n", i);
            return ret;
        }
-
-        /* Inform the decoder about the timebase for the packet timestamps.
-         * This is highly recommended, but not mandatory. */
-        codec_ctx->pkt_timebase = stream->time_base;
-
        /* Reencode video & audio and remux subtitles etc. */
        if (codec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
                || codec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
@@ -130,7 +122,7 @@ static int open_output_file(const char *filename)
    AVStream *out_stream;
    AVStream *in_stream;
    AVCodecContext *dec_ctx, *enc_ctx;
-    const AVCodec *encoder;
+    AVCodec *encoder;
    int ret;
    unsigned int i;

@@ -182,9 +174,8 @@ static int open_output_file(const char *filename)
                enc_ctx->time_base = av_inv_q(dec_ctx->framerate);
            } else {
                enc_ctx->sample_rate = dec_ctx->sample_rate;
-                ret = av_channel_layout_copy(&enc_ctx->ch_layout, &dec_ctx->ch_layout);
-                if (ret < 0)
-                    return ret;
+                enc_ctx->channel_layout = dec_ctx->channel_layout;
+                enc_ctx->channels = av_get_channel_layout_nb_channels(enc_ctx->channel_layout);
                /* take first format from list of supported formats */
                enc_ctx->sample_fmt = encoder->sample_fmts[0];
                enc_ctx->time_base = (AVRational){1, enc_ctx->sample_rate};
@@ -271,7 +262,7 @@ static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
        snprintf(args, sizeof(args),
                "video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
                dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
-                dec_ctx->pkt_timebase.num, dec_ctx->pkt_timebase.den,
+                dec_ctx->time_base.num, dec_ctx->time_base.den,
                dec_ctx->sample_aspect_ratio.num,
                dec_ctx->sample_aspect_ratio.den);

@@ -297,7 +288,6 @@ static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
            goto end;
        }
    } else if (dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
-        char buf[64];
        buffersrc = avfilter_get_by_name("abuffer");
        buffersink = avfilter_get_by_name("abuffersink");
        if (!buffersrc || !buffersink) {
@@ -306,14 +296,14 @@ static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
            goto end;
        }

-        if (dec_ctx->ch_layout.order == AV_CHANNEL_ORDER_UNSPEC)
-            av_channel_layout_default(&dec_ctx->ch_layout, dec_ctx->ch_layout.nb_channels);
-        av_channel_layout_describe(&dec_ctx->ch_layout, buf, sizeof(buf));
+        if (!dec_ctx->channel_layout)
+            dec_ctx->channel_layout =
+                av_get_default_channel_layout(dec_ctx->channels);
        snprintf(args, sizeof(args),
-                "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=%s",
-                dec_ctx->pkt_timebase.num, dec_ctx->pkt_timebase.den, dec_ctx->sample_rate,
+                "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
+                dec_ctx->time_base.num, dec_ctx->time_base.den, dec_ctx->sample_rate,
                av_get_sample_fmt_name(dec_ctx->sample_fmt),
-                buf);
+                dec_ctx->channel_layout);
        ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
                args, NULL, filter_graph);
        if (ret < 0) {
@@ -336,9 +326,9 @@ static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
            goto end;
        }

-        av_channel_layout_describe(&enc_ctx->ch_layout, buf, sizeof(buf));
-        ret = av_opt_set(buffersink_ctx, "ch_layouts",
-                         buf, AV_OPT_SEARCH_CHILDREN);
+        ret = av_opt_set_bin(buffersink_ctx, "channel_layouts",
+                (uint8_t*)&enc_ctx->channel_layout,
+                sizeof(enc_ctx->channel_layout), AV_OPT_SEARCH_CHILDREN);
        if (ret < 0) {
            av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
            goto end;
@@ -441,10 +431,6 @@ static int encode_write_frame(unsigned int stream_index, int flush)
    /* encode filtered frame */
    av_packet_unref(enc_pkt);

-    if (filt_frame && filt_frame->pts != AV_NOPTS_VALUE)
-        filt_frame->pts = av_rescale_q(filt_frame->pts, filt_frame->time_base,
-                                       stream->enc_ctx->time_base);
-
    ret = avcodec_send_frame(stream->enc_ctx, filt_frame);

    if (ret < 0)
@@ -499,7 +485,6 @@ static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index)
            break;
        }

-        filter->filtered_frame->time_base = av_buffersink_get_time_base(filter->buffersink_ctx);;
        filter->filtered_frame->pict_type = AV_PICTURE_TYPE_NONE;
        ret = encode_write_frame(stream_index, 0);
        av_frame_unref(filter->filtered_frame);
@@ -554,6 +539,9 @@ int main(int argc, char **argv)

            av_log(NULL, AV_LOG_DEBUG, "Going to reencode&filter the frame\n");

+            av_packet_rescale_ts(packet,
+                                 ifmt_ctx->streams[stream_index]->time_base,
+                                 stream->dec_ctx->time_base);
            ret = avcodec_send_packet(stream->dec_ctx, packet);
            if (ret < 0) {
                av_log(NULL, AV_LOG_ERROR, "Decoding failed\n");
@@ -585,38 +573,11 @@ int main(int argc, char **argv)
        av_packet_unref(packet);
    }

-    /* flush decoders, filters and encoders */
+    /* flush filters and encoders */
    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
-        StreamContext *stream;
-
+        /* flush filter */
        if (!filter_ctx[i].filter_graph)
            continue;
-
-        stream = &stream_ctx[i];
-
-        av_log(NULL, AV_LOG_INFO, "Flushing stream %u decoder\n", i);
-
-        /* flush decoder */
-        ret = avcodec_send_packet(stream->dec_ctx, NULL);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Flushing decoding failed\n");
-            goto end;
-        }
-
-        while (ret >= 0) {
-            ret = avcodec_receive_frame(stream->dec_ctx, stream->dec_frame);
-            if (ret == AVERROR_EOF)
-                break;
-            else if (ret < 0)
-                goto end;
-
-            stream->dec_frame->pts = stream->dec_frame->best_effort_timestamp;
-            ret = filter_encode_write_frame(stream->dec_frame, i);
-            if (ret < 0)
-                goto end;
-        }
-
-        /* flush filter */
        ret = filter_encode_write_frame(NULL, i);
        if (ret < 0) {
            av_log(NULL, AV_LOG_ERROR, "Flushing filter failed\n");
--- a/doc/examples/vaapi_encode.c
+++ b/doc/examples/vaapi_encode.c
@@ -1,4 +1,6 @@
 /*
+ * Video Acceleration API (video encoding) encode sample
+ *
 * Permission is hereby granted, free of charge, to any person obtaining a copy
 * of this software and associated documentation files (the "Software"), to deal
 * in the Software without restriction, including without limitation the rights
@@ -19,12 +21,13 @@
 */

 /**
- * @file Intel VAAPI-accelerated encoding API usage example
- * @example vaapi_encode.c
+ * @file
+ * Intel VAAPI-accelerated encoding example.
+ *
+ * @example vaapi_encode.c
+ * This example shows how to do VAAPI-accelerated encoding. now only support NV12
+ * raw file, usage like: vaapi_encode 1920 1080 input.yuv output.h264
 *
- * Perform VAAPI-accelerated encoding. Read input from an NV12 raw
- * file, and write the H.264 encoded data to an output raw file.
- * Usage: vaapi_encode 1920 1080 input.yuv output.h264
 */

 #include <stdio.h>
@@ -88,6 +91,10 @@ static int encode_write(AVCodecContext *avctx, AVFrame *frame, FILE *fout)
        enc_pkt->stream_index = 0;
        ret = fwrite(enc_pkt->data, enc_pkt->size, 1, fout);
        av_packet_unref(enc_pkt);
+        if (ret != enc_pkt->size) {
+            ret = AVERROR(errno);
+            break;
+        }
    }

 end:
@@ -102,7 +109,7 @@ int main(int argc, char *argv[])
    FILE *fin = NULL, *fout = NULL;
    AVFrame *sw_frame = NULL, *hw_frame = NULL;
    AVCodecContext *avctx = NULL;
-    const AVCodec *codec = NULL;
+    AVCodec *codec = NULL;
    const char *enc_name = "h264_vaapi";

    if (argc < 5) {
--- a/doc/examples/vaapi_transcode.c
+++ b/doc/examples/vaapi_transcode.c
@@ -1,4 +1,6 @@
 /*
+ * Video Acceleration API (video transcoding) transcode sample
+ *
 * Permission is hereby granted, free of charge, to any person obtaining a copy
 * of this software and associated documentation files (the "Software"), to deal
 * in the Software without restriction, including without limitation the rights
@@ -19,10 +21,11 @@
 */

 /**
- * @file Intel VAAPI-accelerated transcoding API usage example
- * @example vaapi_transcode.c
+ * @file
+ * Intel VAAPI-accelerated transcoding example.
 *
- * Perform VAAPI-accelerated transcoding.
+ * @example vaapi_transcode.c
+ * This example shows how to do VAAPI-accelerated transcoding.
 * Usage: vaapi_transcode input_stream codec output_stream
 * e.g: - vaapi_transcode input.mp4 h264_vaapi output_h264.mp4
 *      - vaapi_transcode input.mp4 vp9_vaapi output_vp9.ivf
@@ -59,7 +62,7 @@ static enum AVPixelFormat get_vaapi_format(AVCodecContext *ctx,
 static int open_input_file(const char *filename)
 {
    int ret;
-    const AVCodec *decoder = NULL;
+    AVCodec *decoder = NULL;
    AVStream *video = NULL;

    if ((ret = avformat_open_input(&ifmt_ctx, filename, NULL, NULL)) < 0) {
@@ -139,7 +142,7 @@ end:
    return ret;
 }

-static int dec_enc(AVPacket *pkt, const AVCodec *enc_codec)
+static int dec_enc(AVPacket *pkt, AVCodec *enc_codec)
 {
    AVFrame *frame;
    int ret = 0;
@@ -215,17 +218,15 @@ static int dec_enc(AVPacket *pkt, const AVCodec *enc_codec)

 fail:
        av_frame_free(&frame);
-        if (ret < 0)
-            return ret;
    }
-    return 0;
+    return ret;
 }

 int main(int argc, char **argv)
 {
-    const AVCodec *enc_codec;
    int ret = 0;
    AVPacket *dec_pkt;
+    AVCodec *enc_codec;

    if (argc != 4) {
        fprintf(stderr, "Usage: %s <input file> <encode codec> <output file>\n"
--- a/doc/fate.texi
+++ b/doc/fate.texi
@@ -79,21 +79,6 @@ Do not put a '~' character in the samples path to indicate a home
 directory. Because of shell nuances, this will cause FATE to fail.
@end float

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

@@ -223,14 +208,6 @@ meaning only while running the regression tests.
 Specify how many threads to use while running regression tests, it is
 quite useful to detect thread-related regressions.

-This variable may be set to the string "random", optionally followed by a
-number, like "random99", This will cause each test to use a random number of
-threads. If a number is specified, it is used as a maximum number of threads,
-otherwise 16 is the maximum.
-
-In case a test fails, the thread count used for it will be written into the
-errfile.
-
@item THREAD_TYPE
 Specify which threading strategy test, either @samp{slice} or @samp{frame},
 by default @samp{slice+frame}
--- a/doc/fate_config.sh.template
+++ b/doc/fate_config.sh.template
@@ -1,5 +1,5 @@
 slot=                                    # some unique identifier
-repo=git://source.ffmpeg.org/ffmpeg.git  # the source repository
+repo=https://git.ffmpeg.org/ffmpeg.git   # the source repository
 #branch=release/2.6                       # the branch to test
 samples=                                 # path to samples directory
 workdir=                                 # directory in which to do all the work
--- a/doc/ffmpeg.texi
+++ b/doc/ffmpeg.texi
@@ -17,9 +17,9 @@ ffmpeg [@var{global_options}] @{[@var{input_file_options}] -i @file{input_url}@}
@chapter Description
@c man begin DESCRIPTION

-@command{ffmpeg} is a universal media converter. It can read a wide variety of
-inputs - including live grabbing/recording devices - filter, and transcode them
-into a plethora of output formats.
+@command{ffmpeg} is a very fast video and audio converter that can also grab from
+a live audio/video source. It can also convert between arbitrary sample
+rates and resize video on the fly with a high quality polyphase filter.

@command{ffmpeg} reads from an arbitrary number of input "files" (which can be regular
 files, pipes, network streams, grabbing devices, etc.), specified by the
@@ -49,32 +49,24 @@ Do not mix input and output files -- first specify all input files, then all
 output files. Also do not mix options which belong to different files. All
 options apply ONLY to the next input or output file and are reset between files.

-Some simple examples follow.
-
@itemize
@item
-Convert an input media file to a different format, by re-encoding media streams:
+To set the video bitrate of the output file to 64 kbit/s:
@example
-ffmpeg -i input.avi output.mp4
+ffmpeg -i input.avi -b:v 64k -bufsize 64k output.avi
@end example

@item
-Set the video bitrate of the output file to 64 kbit/s:
+To force the frame rate of the output file to 24 fps:
@example
-ffmpeg -i input.avi -b:v 64k -bufsize 64k output.mp4
+ffmpeg -i input.avi -r 24 output.avi
@end example

@item
-Force the frame rate of the output file to 24 fps:
+To force the frame rate of the input file (valid for raw formats only)
+to 1 fps and the frame rate of the output file to 24 fps:
@example
-ffmpeg -i input.avi -r 24 output.mp4
-@end example
-
-@item
-Force the frame rate of the input file (valid for raw formats only) to 1 fps and
-the frame rate of the output file to 24 fps:
-@example
-ffmpeg -r 1 -i input.m2v -r 24 output.mp4
+ffmpeg -r 1 -i input.m2v -r 24 output.avi
@end example
@end itemize

@@ -457,11 +449,6 @@ output file already exists.
 Set number of times input stream shall be looped. Loop 0 means no loop,
 loop -1 means infinite loop.

-@item -recast_media (@emph{global})
-Allow forcing a decoder of a different media type than the one
-detected or designated by the demuxer. Useful for decoding media
-data muxed as data streams.
-
@item -c[:@var{stream_specifier}] @var{codec} (@emph{input/output,per-stream})
@itemx -codec[:@var{stream_specifier}] @var{codec} (@emph{input/output,per-stream})
 Select an encoder (when used before an output file) or a decoder (when used
@@ -526,21 +513,6 @@ see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1)
 Like the @code{-ss} option but relative to the "end of file". That is negative
 values are earlier in the file, 0 is at EOF.

-@item -isync @var{input_index} (@emph{input})
-Assign an input as a sync source.
-
-This will take the difference between the start times of the target and reference inputs and
-offset the timestamps of the target file by that difference. The source timestamps of the two
-inputs should derive from the same clock source for expected results. If @code{copyts} is set
-then @code{start_at_zero} must also be set. If either of the inputs has no starting timestamp
-then no sync adjustment is made.
-
-Acceptable values are those that refer to a valid ffmpeg input index. If the sync reference is
-the target index itself or @var{-1}, then no adjustment is made to target timestamps. A sync
-reference may not itself be synced to any other input.
-
-Default value is @var{-1}.
-
@item -itsoffset @var{offset} (@emph{input})
 Set the input time offset.

@@ -583,22 +555,27 @@ ffmpeg -i INPUT -metadata:s:a:0 language=eng OUTPUT
@item -disposition[:stream_specifier] @var{value} (@emph{output,per-stream})
 Sets the disposition for a stream.

-By default, the disposition is copied from the input stream, unless the output
-stream this option applies to is fed by a complex filtergraph - in that case the
-disposition is unset by default.
+This option overrides the disposition copied from the input stream. It is also
+possible to delete the disposition by setting it to 0.

-@var{value} is a sequence of items separated by '+' or '-'. The first item may
-also be prefixed with '+' or '-', in which case this option modifies the default
-value. Otherwise (the first item is not prefixed) this options overrides the
-default value. A '+' prefix adds the given disposition, '-' removes it. It is
-also possible to clear the disposition by setting it to 0.
-
-If no @code{-disposition} options were specified for an output file, ffmpeg will
-automatically set the 'default' disposition on the first stream of each type,
-when there are multiple streams of this type in the output file and no stream of
-that type is already marked as default.
-
-The @code{-dispositions} option lists the known dispositions.
+The following dispositions are recognized:
+@table @option
+@item default
+@item dub
+@item original
+@item comment
+@item lyrics
+@item karaoke
+@item forced
+@item hearing_impaired
+@item visual_impaired
+@item clean_effects
+@item attached_pic
+@item captions
+@item descriptions
+@item dependent
+@item metadata
+@end table

 For example, to make the second audio stream the default stream:
@example
@@ -647,21 +624,21 @@ The parameters set for each target are as follows.
@var{pal}:
 -f vcd -muxrate 1411200 -muxpreload 0.44 -packetsize 2324
 -s 352x288 -r 25
-codec:v mpeg1video -g 15 -b:v 1150k -maxrate:v 1150k -minrate:v 1150k -bufsize:v 327680
+-codec:v mpeg1video -g 15 -b:v 1150k -maxrate:v 1150v -minrate:v 1150k -bufsize:v 327680
 -ar 44100 -ac 2
 -codec:a mp2 -b:a 224k

@var{ntsc}:
 -f vcd -muxrate 1411200 -muxpreload 0.44 -packetsize 2324
 -s 352x240 -r 30000/1001
-codec:v mpeg1video -g 18 -b:v 1150k -maxrate:v 1150k -minrate:v 1150k -bufsize:v 327680
+-codec:v mpeg1video -g 18 -b:v 1150k -maxrate:v 1150v -minrate:v 1150k -bufsize:v 327680
 -ar 44100 -ac 2
 -codec:a mp2 -b:a 224k

@var{film}:
 -f vcd -muxrate 1411200 -muxpreload 0.44 -packetsize 2324
 -s 352x240 -r 24000/1001
-codec:v mpeg1video -g 18 -b:v 1150k -maxrate:v 1150k -minrate:v 1150k -bufsize:v 327680
+-codec:v mpeg1video -g 18 -b:v 1150k -maxrate:v 1150v -minrate:v 1150k -bufsize:v 327680
 -ar 44100 -ac 2
 -codec:a mp2 -b:a 224k
@end example
@@ -777,22 +754,11 @@ syntax.
 See the @ref{filter_complex_option,,-filter_complex option} if you
 want to create filtergraphs with multiple inputs and/or outputs.

-@anchor{filter_script option}
@item -filter_script[:@var{stream_specifier}] @var{filename} (@emph{output,per-stream})
 This option is similar to @option{-filter}, the only difference is that its
 argument is the name of the file from which a filtergraph description is to be
 read.

-@item -reinit_filter[:@var{stream_specifier}] @var{integer} (@emph{input,per-stream})
-This boolean option determines if the filtergraph(s) to which this stream is fed gets
-reinitialized when input frame parameters change mid-stream. This option is enabled by
-default as most video and all audio filters cannot handle deviation in input frame properties.
-Upon reinitialization, existing filter state is lost, like e.g. the frame count @code{n}
-reference available in some filters. Any frames buffered at time of reinitialization are lost.
-The properties where a change triggers reinitialization are,
-for video, frame resolution or pixel format;
-for audio, sample format, sample rate, channel count or channel layout.
-
@item -filter_threads @var{nb_threads} (@emph{global})
 Defines how many threads are used to process a filter pipeline. Each pipeline
 will produce a thread pool with this many threads available for parallel processing.
@@ -886,20 +852,9 @@ This is not the same as the @option{-framerate} option used for some input forma
 like image2 or v4l2 (it used to be the same in older versions of FFmpeg).
 If in doubt use @option{-framerate} instead of the input option @option{-r}.

-As an output option:
-@table @option
-@item video encoding
-Duplicate or drop frames right before encoding them to achieve constant output
+As an output option, duplicate or drop input frames to achieve constant output
 frame rate @var{fps}.

-@item video streamcopy
-Indicate to the muxer that @var{fps} is the stream frame rate. No data is
-dropped or duplicated in this case. This may produce invalid files if @var{fps}
-does not match the actual stream frame rate as determined by packet timestamps.
-See also the @code{setts} bitstream filter.
-
-@end table
-
@item -fpsmax[:@var{stream_specifier}] @var{fps} (@emph{output,per-stream})
 Set maximum frame rate (Hz value, fraction or abbreviation).

@@ -932,32 +887,6 @@ 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 -display_rotation[:@var{stream_specifier}] @var{rotation} (@emph{input,per-stream})
-Set video rotation metadata.
-
-@var{rotation} is a decimal number specifying the amount in degree by
-which the video should be rotated counter-clockwise before being
-displayed.
-
-This option overrides the rotation/display transform metadata stored in
-the file, if any. When the video is being transcoded (rather than
-copied) and @code{-autorotate} is enabled, the video will be rotated at
-the filtering stage. Otherwise, the metadata will be written into the
-output file if the muxer supports it.
-
-If the @code{-display_hflip} and/or @code{-display_vflip} options are
-given, they are applied after the rotation specified by this option.
-
-@item -display_hflip[:@var{stream_specifier}] (@emph{input,per-stream})
-Set whether on display the image should be horizontally flipped.
-
-See the @code{-display_rotation} option for more details.
-
-@item -display_vflip[:@var{stream_specifier}] (@emph{input,per-stream})
-Set whether on display the image should be vertically flipped.
-
-See the @code{-display_rotation} option for more details.
-
@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}
@@ -1023,12 +952,7 @@ If @var{pix_fmt} is a single @code{+}, ffmpeg selects the same pixel format
 as the input (or graph output) and automatic conversions are disabled.

@item -sws_flags @var{flags} (@emph{input/output})
-Set default flags for the libswscale library. These flags are used by
-automatically inserted @code{scale} filters and those within simple
-filtergraphs, if not overridden within the filtergraph definition.
-
-See the @ref{scaler_options,,ffmpeg-scaler manual,ffmpeg-scaler} for a list
-of scaler options.
+Set SwScaler flags.

@item -rc_override[:@var{stream_specifier}] @var{override} (@emph{output,per-stream})
 Rate control override for specific intervals, formatted as "int,int,int"
@@ -1036,24 +960,36 @@ list separated with slashes. Two first values are the beginning and
 end frame numbers, last one is quantizer to use if positive, or quality
 factor if negative.

+@item -ilme
+Force interlacing support in encoder (MPEG-2 and MPEG-4 only).
+Use this option if your input file is interlaced and you want
+to keep the interlaced format for minimum losses.
+The alternative is to deinterlace the input stream by use of a filter
+such as @code{yadif} or @code{bwdif}, but deinterlacing introduces losses.
@item -psnr
-Calculate PSNR of compressed frames. This option is deprecated, pass the
-PSNR flag to the encoder instead, using @code{-flags +psnr}.
-
+Calculate PSNR of compressed frames.
@item -vstats
-Dump video coding statistics to @file{vstats_HHMMSS.log}. See the
-@ref{vstats_file_format,,vstats file format} section for the format description.
-
+Dump video coding statistics to @file{vstats_HHMMSS.log}.
@item -vstats_file @var{file}
-Dump video coding statistics to @var{file}. See the
-@ref{vstats_file_format,,vstats file format} section for the format description.
-
+Dump video coding statistics to @var{file}.
@item -vstats_version @var{file}
-Specify which version of the vstats format to use. Default is @code{2}. See the
-@ref{vstats_file_format,,vstats file format} section for the format description.
+Specifies which version of the vstats format to use. Default is 2.

+version = 1 :
+
+@code{frame= %5d q= %2.1f PSNR= %6.2f f_size= %6d s_size= %8.0fkB time= %0.3f br= %7.1fkbits/s avg_br= %7.1fkbits/s}
+
+version > 1:
+
+@code{out= %2d st= %2d frame= %5d q= %2.1f PSNR= %6.2f f_size= %6d s_size= %8.0fkB time= %0.3f br= %7.1fkbits/s avg_br= %7.1fkbits/s}
+@item -top[:@var{stream_specifier}] @var{n} (@emph{output,per-stream})
+top=1/bottom=0/auto=-1 field first
+@item -dc @var{precision}
+Intra_dc_precision.
@item -vtag @var{fourcc/tag} (@emph{output})
 Force video tag/fourcc. This is an alias for @code{-tag:v}.
+@item -qphist (@emph{global})
+Show QP histogram
@item -vbsf @var{bitstream_filter}
 Deprecated see -bsf

@@ -1120,8 +1056,6 @@ starting from second 13:
@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.
-In cases where this particular source frame has to be dropped,
-enforce the next available frame to become a key frame instead.

@end table

@@ -1145,32 +1079,13 @@ device type:
@item cuda
@var{device} is the number of the CUDA device.

-The following options are recognized:
-@table @option
-@item primary_ctx
-If set to 1, uses the primary device context instead of creating a new one.
-@end table
-
-Examples:
-@table @emph
-@item -init_hw_device cuda:1
-Choose the second device on the system.
-
-@item -init_hw_device cuda:0,primary_ctx=1
-Choose the first device and use the primary device context.
-@end table
-
@item dxva2
@var{device} is the number of the Direct3D 9 display adapter.

-@item d3d11va
-@var{device} is the number of the Direct3D 11 display adapter.
-
@item vaapi
-@var{device} is either an X11 display name, a DRM render node or a DirectX adapter index.
+@var{device} is either an X11 display name or a DRM render node.
 If not specified, it will attempt to open the default X11 display (@emph{$DISPLAY})
-and then the first DRM render node (@emph{/dev/dri/renderD128}), or the default
-DirectX adapter on Windows.
+and then the first DRM render node (@emph{/dev/dri/renderD128}).

@item vdpau
@var{device} is an X11 display name.
@@ -1190,21 +1105,9 @@ If not specified, it will attempt to open the default X11 display (@emph{$DISPLA
@end table
 If not specified, @samp{auto_any} is used.
 (Note that it may be easier to achieve the desired result for QSV by creating the
-platform-appropriate subdevice (@samp{dxva2} or @samp{d3d11va} or @samp{vaapi}) and then deriving a
+platform-appropriate subdevice (@samp{dxva2} or @samp{vaapi}) and then deriving a
 QSV device from that.)

-Alternatively, @samp{child_device_type} helps to choose platform-appropriate subdevice type.
-On Windows @samp{d3d11va} is used as default subdevice type.
-
-Examples:
-@table @emph
-@item -init_hw_device qsv:hw,child_device_type=d3d11va
-Choose the GPU subdevice with type @samp{d3d11va} and create QSV device with @samp{MFX_IMPL_HARDWARE}.
-
-@item -init_hw_device qsv:hw,child_device_type=dxva2
-Choose the GPU subdevice with type @samp{dxva2} and create QSV device with @samp{MFX_IMPL_HARDWARE}.
-@end table
-
@item opencl
@var{device} selects the platform and device as @emph{platform_index.device_index}.

@@ -1307,9 +1210,6 @@ Use VDPAU (Video Decode and Presentation API for Unix) hardware acceleration.
@item dxva2
 Use DXVA2 (DirectX Video Acceleration) hardware acceleration.

-@item d3d11va
-Use D3D11VA (DirectX Video Acceleration) hardware acceleration.
-
@item vaapi
 Use VAAPI (Video Acceleration API) hardware acceleration.

@@ -1343,25 +1243,7 @@ by name, or it can create a new device as if
 were called immediately before.

@item -hwaccels
-List all hardware acceleration components enabled in this build of ffmpeg.
-Actual runtime availability depends on the hardware and its suitable driver
-being installed.
-
-@item -fix_sub_duration_heartbeat[:@var{stream_specifier}]
-Set a specific output video stream as the heartbeat stream according to which
-to split and push through currently in-progress subtitle upon receipt of a
-random access packet.
-
-This lowers the latency of subtitles for which the end packet or the following
-subtitle has not yet been received. As a drawback, this will most likely lead
-to duplication of subtitle events in order to cover the full duration, so
-when dealing with use cases where latency of when the subtitle event is passed
-on to output is not relevant this option should not be utilized.
-
-Requires @option{-fix_sub_duration} to be set for the relevant input subtitle
-stream for this to have any effect, as well as for the input subtitle stream
-having to be directly mapped to the same output in which the heartbeat stream
-resides.
+List all hardware acceleration methods supported in this build of ffmpeg.

@end table

@@ -1461,18 +1343,18 @@ Set the size of the canvas used to render subtitles.
@section Advanced options

@table @option
-@item -map [-]@var{input_file_id}[:@var{stream_specifier}][?] | @var{[linklabel]} (@emph{output})
+@item -map [-]@var{input_file_id}[:@var{stream_specifier}][?][,@var{sync_file_id}[:@var{stream_specifier}]] | @var{[linklabel]} (@emph{output})

-Create one or more streams in the output file. This option has two forms for
-specifying the data source(s): the first selects one or more streams from some
-input file (specified with @code{-i}), the second takes an output from some
-complex filtergraph (specified with @code{-filter_complex} or
-@code{-filter_complex_script}).
+Designate one or more input streams as a source for the output file. Each input
+stream is identified by the input file index @var{input_file_id} and
+the input stream index @var{input_stream_id} within the input
+file. Both indices start at 0. If specified,
+@var{sync_file_id}:@var{stream_specifier} sets which input stream
+is used as a presentation sync reference.

-In the first form, an output stream is created for every stream from the input
-file with the index @var{input_file_id}. If @var{stream_specifier} is given,
-only those streams that match the specifier are used (see the
-@ref{Stream specifiers} section for the @var{stream_specifier} syntax).
+The first @code{-map} option on the command line specifies the
+source for output stream 0, the second @code{-map} option specifies
+the source for output stream 1, etc.

 A @code{-} character before the stream identifier creates a "negative" mapping.
 It disables matching streams from already created mappings.
@@ -1486,56 +1368,39 @@ An alternative @var{[linklabel]} form will map outputs from complex filter
 graphs (see the @option{-filter_complex} option) to the output file.
@var{linklabel} must correspond to a defined output link label in the graph.

-This option may be specified multiple times, each adding more streams to the
-output file. Any given input stream may also be mapped any number of times as a
-source for different output streams, e.g. in order to use different encoding
-options and/or filters. The streams are created in the output in the same order
-in which the @code{-map} options are given on the commandline.
-
-Using this option disables the default mappings for this output file.
-
-Examples:
-
-@table @emph
-
-@item map everything
-To map ALL streams from the first input file to output
+For example, to map ALL streams from the first input file to output
@example
 ffmpeg -i INPUT -map 0 output
@end example

-@item select specific stream
-If you have two audio streams in the first input file, these streams are
-identified by @var{0:0} and @var{0:1}. You can use @code{-map} to select which
-streams to place in an output file. For example:
+For example, if you have two audio streams in the first input file,
+these streams are identified by "0:0" and "0:1". You can use
+@code{-map} to select which streams to place in an output file. For
+example:
@example
 ffmpeg -i INPUT -map 0:1 out.wav
@end example
-will map the second input stream in @file{INPUT} to the (single) output stream
-in @file{out.wav}.
+will map the input stream in @file{INPUT} identified by "0:1" to
+the (single) output stream in @file{out.wav}.

-@item create multiple streams
-To select the stream with index 2 from input file @file{a.mov} (specified by the
-identifier @var{0:2}), and stream with index 6 from input @file{b.mov}
-(specified by the identifier @var{1:6}), and copy them to the output file
-@file{out.mov}:
+For example, to select the stream with index 2 from input file
+@file{a.mov} (specified by the identifier "0:2"), and stream with
+index 6 from input @file{b.mov} (specified by the identifier "1:6"),
+and copy them to the output file @file{out.mov}:
@example
 ffmpeg -i a.mov -i b.mov -c copy -map 0:2 -map 1:6 out.mov
@end example

-@item create multiple streams 2
 To select all video and the third audio stream from an input file:
@example
 ffmpeg -i INPUT -map 0:v -map 0:a:2 OUTPUT
@end example

-@item negative map
 To map all the streams except the second audio, use negative mappings
@example
 ffmpeg -i INPUT -map 0 -map -0:a:1 OUTPUT
@end example

-@item optional map
 To map the video and audio streams from the first input, and using the
 trailing @code{?}, ignore the audio mapping if no audio streams exist in
 the first input:
@@ -1543,13 +1408,12 @@ the first input:
 ffmpeg -i INPUT -map 0:v -map 0:a? OUTPUT
@end example

-@item map by language
 To pick the English audio stream:
@example
 ffmpeg -i INPUT -map 0:m:language:eng OUTPUT
@end example

-@end table
+Note that using this option disables the default mappings for this output file.

@item -ignore_unknown
 Ignore input streams with unknown type instead of failing if copying
@@ -1560,10 +1424,6 @@ Allow input streams with unknown type to be copied instead of failing if copying
 such streams is attempted.

@item -map_channel [@var{input_file_id}.@var{stream_specifier}.@var{channel_id}|-1][?][:@var{output_file_id}.@var{stream_specifier}]
-This option is deprecated and will be removed. It can be replaced by the
-@var{pan} filter. In some cases it may be easier to use some combination of the
-@var{channelsplit}, @var{channelmap}, or @var{amerge} filters.
-
 Map an audio channel from a given input to an output. If
@var{output_file_id}.@var{stream_specifier} is not set, the audio channel will
 be mapped on all the audio streams.
@@ -1691,47 +1551,33 @@ Exit after ffmpeg has been running for @var{duration} seconds in CPU user time.
 Dump each input packet to stderr.
@item -hex (@emph{global})
 When dumping packets, also dump the payload.
-@item -readrate @var{speed} (@emph{input})
-Limit input read speed.
-
-Its value is a floating-point positive number which represents the maximum duration of
-media, in seconds, that should be ingested in one second of wallclock time.
-Default value is zero and represents no imposed limitation on speed of ingestion.
-Value @code{1} represents real-time speed and is equivalent to @code{-re}.
-
-Mainly used to simulate a capture device or live input stream (e.g. when reading from a file).
-Should not be used with a low value when input is an actual capture device or live stream as
-it may cause packet loss.
-
-It is useful for when flow speed of output packets is important, such as live streaming.
@item -re (@emph{input})
-Read input at native frame rate. This is equivalent to setting @code{-readrate 1}.
-@item -readrate_initial_burst @var{seconds}
-Set an initial read burst time, in seconds, after which @option{-re/-readrate}
-will be enforced.
-@item -vsync @var{parameter} (@emph{global})
-@itemx -fps_mode[:@var{stream_specifier}] @var{parameter} (@emph{output,per-stream})
-Set video sync method / framerate mode. vsync is applied to all output video streams
-but can be overridden for a stream by setting fps_mode. vsync is deprecated and will be
-removed in the future.
-
-For compatibility reasons some of the values for vsync can be specified as numbers (shown
-in parentheses in the following table).
+Read input at native frame rate. Mainly used to simulate a grab device,
+or live input stream (e.g. when reading from a file). Should not be used
+with actual grab devices or live input streams (where it can cause packet
+loss).
+By default @command{ffmpeg} attempts to read the input(s) as fast as possible.
+This option will slow down the reading of the input(s) to the native frame rate
+of the input(s). It is useful for real-time output (e.g. live streaming).
+@item -vsync @var{parameter}
+Video sync method.
+For compatibility reasons old values can be specified as numbers.
+Newly added values will have to be specified as strings always.

@table @option
-@item passthrough (0)
+@item 0, passthrough
 Each frame is passed with its timestamp from the demuxer to the muxer.
-@item cfr (1)
+@item 1, cfr
 Frames will be duplicated and dropped to achieve exactly the requested
 constant frame rate.
-@item vfr (2)
+@item 2, vfr
 Frames are passed through with their timestamp or dropped so as to
 prevent 2 frames from having the same timestamp.
@item drop
 As passthrough but destroys all timestamps, making the muxer generate
 fresh timestamps based on frame-rate.
-@item auto (-1)
-Chooses between cfr and vfr depending on muxer capabilities. This is the
+@item -1, auto
+Chooses between 1 and 2 depending on muxer capabilities. This is the
 default method.
@end table

@@ -1750,6 +1596,24 @@ The default is -1.1. One possible usecase is to avoid framedrops in case
 of noisy timestamps or to increase frame drop precision in case of exact
 timestamps.

+@item -async @var{samples_per_second}
+Audio sync method. "Stretches/squeezes" the audio stream to match the timestamps,
+the parameter is the maximum samples per second by which the audio is changed.
+-async 1 is a special case where only the start of the audio stream is corrected
+without any later correction.
+
+Note that the timestamps may be further modified by the muxer, after this.
+For example, in the case that the format option @option{avoid_negative_ts}
+is enabled.
+
+This option has been deprecated. Use the @code{aresample} audio filter instead.
+
+@item -adrift_threshold @var{time}
+Set the minimum difference between timestamps and audio data (in seconds) to trigger
+adding/dropping samples to make it match the timestamps. This option effectively is
+a threshold to select between hard (add/drop) and soft (squeeze/stretch) compensation.
+@code{-async} must be set to a positive value.
+
@item -apad @var{parameters} (@emph{output,per-stream})
 Pad the output audio stream(s). This is the same as applying @code{-af apad}.
 Argument is a string of filter parameters composed the same as with the @code{apad} filter.
@@ -1796,7 +1660,8 @@ Try to make the choice automatically, in order to generate a sane output.
 Default value is -1.

@item -enc_time_base[:@var{stream_specifier}] @var{timebase} (@emph{output,per-stream})
-Set the encoder timebase. @var{timebase} can assume one of the following values:
+Set the encoder timebase. @var{timebase} is a floating point number,
+and can assume one of the following values:

@table @option
@item 0
@@ -1804,17 +1669,16 @@ Assign a default value according to the media type.

 For video - use 1/framerate, for audio - use 1/samplerate.

-@item demux
-Use the timebase from the demuxer.
+@item -1
+Use the input stream timebase when possible.

-@item filter
-Use the timebase from the filtergraph.
+If an input stream is not available, the default timebase will be used.

-@item a positive number
+@item >0
 Use the provided number as the timebase.

 This field can be provided as a ratio of two integers (e.g. 1:24, 1:48000)
-or as a decimal number (e.g. 0.04166, 2.0833e-5)
+or as a floating point number (e.g. 0.04166, 2.0833e-5)
@end table

 Default value is 0.
@@ -1822,55 +1686,13 @@ Default value is 0.
@item -bitexact (@emph{input/output})
 Enable bitexact mode for (de)muxer and (de/en)coder
@item -shortest (@emph{output})
-Finish encoding when the shortest output stream ends.
-
-Note that this option may require buffering frames, which introduces extra
-latency. The maximum amount of this latency may be controlled with the
-@code{-shortest_buf_duration} option.
-
-@item -shortest_buf_duration @var{duration} (@emph{output})
-The @code{-shortest} option may require buffering potentially large amounts
-of data when at least one of the streams is "sparse" (i.e. has large gaps
-between frames – this is typically the case for subtitles).
-
-This option controls the maximum duration of buffered frames in seconds.
-Larger values may allow the @code{-shortest} option to produce more accurate
-results, but increase memory use and latency.
-
-The default value is 10 seconds.
-
-@item -dts_delta_threshold @var{threshold}
-Timestamp discontinuity delta threshold, expressed as a decimal number
-of seconds.
-
-The timestamp discontinuity correction enabled by this option is only
-applied to input formats accepting timestamp discontinuity (for which
-the @code{AV_FMT_DISCONT} flag is enabled), e.g. MPEG-TS and HLS, and
-is automatically disabled when employing the @code{-copy_ts} option
-(unless wrapping is detected).
-
-If a timestamp discontinuity is detected whose absolute value is
-greater than @var{threshold}, ffmpeg will remove the discontinuity by
-decreasing/increasing the current DTS and PTS by the corresponding
-delta value.
-
-The default value is 10.
-
-@item -dts_error_threshold @var{threshold}
-Timestamp error delta threshold, expressed as a decimal number of
-seconds.
-
-The timestamp correction enabled by this option is only applied to
-input formats not accepting timestamp discontinuity (for which the
-@code{AV_FMT_DISCONT} flag is not enabled).
-
-If a timestamp discontinuity is detected whose absolute value is
-greater than @var{threshold}, ffmpeg will drop the PTS/DTS timestamp
-value.
-
-The default value is @code{3600*30} (30 hours), which is arbitrarily
-picked and quite conservative.
-
+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})
 Set the maximum demux-decode delay.
@item -muxpreload @var{seconds} (@emph{output})
@@ -1981,7 +1803,6 @@ The default is the number of available CPUs.
 Define a complex filtergraph, i.e. one with arbitrary number of inputs and/or
 outputs. Equivalent to @option{-filter_complex}.

-@anchor{filter_complex_script option}
@item -filter_complex_script @var{filename} (@emph{global})
 This option is similar to @option{-filter_complex}, the only difference is that
 its argument is the name of the file from which a complex filtergraph
@@ -2000,15 +1821,12 @@ to the @option{-ss} option is considered an actual timestamp, and is not
 offset by the start time of the file. This matters only for files which do
 not start from timestamp 0, such as transport streams.

-@item -thread_queue_size @var{size} (@emph{input/output})
-For input, this option sets the maximum number of queued packets when reading
-from the file or device. With low latency / high rate live streams, packets may
-be discarded if they are not read in a timely manner; setting this value can
+@item -thread_queue_size @var{size} (@emph{input})
+This option sets the maximum number of queued packets when reading from the
+file or device. With low latency / high rate live streams, packets may be
+discarded if they are not read in a timely manner; setting this value can
 force ffmpeg to use a separate input thread and read packets as soon as they
-arrive. By default ffmpeg only does this if multiple inputs are specified.
-
-For output, this option specified the maximum number of packets that may be
-queued to each muxing thread.
+arrive. By default ffmpeg only do this if multiple inputs are specified.

@item -sdp_file @var{file} (@emph{global})
 Print sdp information for an output stream to @var{file}.
@@ -2083,132 +1901,6 @@ filter (scale, aresample) in the graph.
 On by default, to explicitly disable it you need to specify
@code{-noauto_conversion_filters}.

-@item -bits_per_raw_sample[:@var{stream_specifier}] @var{value} (@emph{output,per-stream})
-Declare the number of bits per raw sample in the given output stream to be
-@var{value}. Note that this option sets the information provided to the
-encoder/muxer, it does not change the stream to conform to this value. Setting
-values that do not match the stream properties may result in encoding failures
-or invalid output files.
-
-@anchor{stats_enc_options}
-@item -stats_enc_pre[:@var{stream_specifier}] @var{path} (@emph{output,per-stream})
-@item -stats_enc_post[:@var{stream_specifier}] @var{path} (@emph{output,per-stream})
-@item -stats_mux_pre[:@var{stream_specifier}] @var{path} (@emph{output,per-stream})
-Write per-frame encoding information about the matching streams into the file
-given by @var{path}.
-
-@option{-stats_enc_pre} writes information about raw video or audio frames right
-before they are sent for encoding, while @option{-stats_enc_post} writes
-information about encoded packets as they are received from the encoder.
-@option{-stats_mux_pre} writes information about packets just as they are about to
-be sent to the muxer. Every frame or packet produces one line in the specified
-file. The format of this line is controlled by @option{-stats_enc_pre_fmt} /
-@option{-stats_enc_post_fmt} / @option{-stats_mux_pre_fmt}.
-
-When stats for multiple streams are written into a single file, the lines
-corresponding to different streams will be interleaved. The precise order of
-this interleaving is not specified and not guaranteed to remain stable between
-different invocations of the program, even with the same options.
-
-@item -stats_enc_pre_fmt[:@var{stream_specifier}] @var{format_spec} (@emph{output,per-stream})
-@item -stats_enc_post_fmt[:@var{stream_specifier}] @var{format_spec} (@emph{output,per-stream})
-@item -stats_mux_pre_fmt[:@var{stream_specifier}] @var{format_spec} (@emph{output,per-stream})
-Specify the format for the lines written with @option{-stats_enc_pre} /
-@option{-stats_enc_post} / @option{-stats_mux_pre}.
-
-@var{format_spec} is a string that may contain directives of the form
-@var{@{fmt@}}. @var{format_spec} is backslash-escaped --- use \@{, \@}, and \\
-to write a literal @{, @}, or \, respectively, into the output.
-
-The directives given with @var{fmt} may be one of the following:
-@table @option
-@item fidx
-Index of the output file.
-
-@item sidx
-Index of the output stream in the file.
-
-@item n
-Frame number. Pre-encoding: number of frames sent to the encoder so far.
-Post-encoding: number of packets received from the encoder so far.
-Muxing: number of packets submitted to the muxer for this stream so far.
-
-@item ni
-Input frame number. Index of the input frame (i.e. output by a decoder) that
-corresponds to this output frame or packet. -1 if unavailable.
-
-@item tb
-Timebase in which this frame/packet's timestamps are expressed, as a rational
-number @var{num/den}. Note that encoder and muxer may use different timebases.
-
-@item tbi
-Timebase for @var{ptsi}, as a rational number @var{num/den}. Available when
-@var{ptsi} is available, @var{0/1} otherwise.
-
-@item pts
-Presentation timestamp of the frame or packet, as an integer. Should be
-multiplied by the timebase to compute presentation time.
-
-@item ptsi
-Presentation timestamp of the input frame (see @var{ni}), as an integer. Should
-be multiplied by @var{tbi} to compute presentation time. Printed as
-(2^63 - 1 = 9223372036854775807) when not available.
-
-@item t
-Presentation time of the frame or packet, as a decimal number. Equal to
-@var{pts} multiplied by @var{tb}.
-
-@item ti
-Presentation time of the input frame (see @var{ni}), as a decimal number. Equal
-to @var{ptsi} multiplied by @var{tbi}. Printed as inf when not available.
-
-@item dts (@emph{packet})
-Decoding timestamp of the packet, as an integer. Should be multiplied by the
-timebase to compute presentation time.
-
-@item dt (@emph{packet})
-Decoding time of the frame or packet, as a decimal number. Equal to
-@var{dts} multiplied by @var{tb}.
-
-@item sn (@emph{frame,audio})
-Number of audio samples sent to the encoder so far.
-
-@item samp (@emph{frame,audio})
-Number of audio samples in the frame.
-
-@item size (@emph{packet})
-Size of the encoded packet in bytes.
-
-@item br (@emph{packet})
-Current bitrate in bits per second. Post-encoding only.
-
-@item abr (@emph{packet})
-Average bitrate for the whole stream so far, in bits per second, -1 if it cannot
-be determined at this point. Post-encoding only.
-@end table
-
-Directives tagged with @emph{packet} may only be used with
-@option{-stats_enc_post_fmt} and @option{-stats_mux_pre_fmt}.
-
-Directives tagged with @emph{frame} may only be used with
-@option{-stats_enc_pre_fmt}.
-
-Directives tagged with @emph{audio} may only be used with audio streams.
-
-The default format strings are:
-@table @option
-@item pre-encoding
-@{fidx@} @{sidx@} @{n@} @{t@}
-@item post-encoding
-@{fidx@} @{sidx@} @{n@} @{t@}
-@end table
-In the future, new items may be added to the end of the default formatting
-strings. Users who depend on the format staying exactly the same, should
-prescribe it manually.
-
-Note that stats for different streams written into the same file may have
-different formats.
-
@end table

@section Preset files
@@ -2266,63 +1958,6 @@ search for the file @file{libvpx-1080p.avpreset}.
 If no such file is found, then ffmpeg will search for a file named
@var{arg}.avpreset in the same directories.

-@anchor{vstats_file_format}
-@section vstats file format
-The @code{-vstats} and @code{-vstats_file} options enable generation of a file
-containing statistics about the generated video outputs.
-
-The @code{-vstats_version} option controls the format version of the generated
-file.
-
-With version @code{1} the format is:
-@example
-frame= @var{FRAME} q= @var{FRAME_QUALITY} PSNR= @var{PSNR} f_size= @var{FRAME_SIZE} s_size= @var{STREAM_SIZE}kB time= @var{TIMESTAMP} br= @var{BITRATE}kbits/s avg_br= @var{AVERAGE_BITRATE}kbits/s
-@end example
-
-With version @code{2} the format is:
-@example
-out= @var{OUT_FILE_INDEX} st= @var{OUT_FILE_STREAM_INDEX} frame= @var{FRAME_NUMBER} q= @var{FRAME_QUALITY}f PSNR= @var{PSNR} f_size= @var{FRAME_SIZE} s_size= @var{STREAM_SIZE}kB time= @var{TIMESTAMP} br= @var{BITRATE}kbits/s avg_br= @var{AVERAGE_BITRATE}kbits/s
-@end example
-
-The value corresponding to each key is described below:
-@table @option
-@item avg_br
-average bitrate expressed in Kbits/s
-
-@item br
-bitrate expressed in Kbits/s
-
-@item frame
-number of encoded frame
-
-@item out
-out file index
-
-@item PSNR
-Peak Signal to Noise Ratio
-
-@item q
-quality of the frame
-
-@item f_size
-encoded packet size expressed as number of bytes
-
-@item s_size
-stream size expressed in KiB
-
-@item st
-out file stream index
-
-@item time
-time of the packet
-
-@item type
-picture type
-@end table
-
-See also the @ref{stats_enc_options,,-stats_enc options} for an alternative way
-to show encoding statistics.
-
@c man end OPTIONS

@chapter Examples
@@ -2586,7 +2221,7 @@ ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)
@ignore

@setfilename ffmpeg
-@settitle ffmpeg media converter
+@settitle ffmpeg video converter

@end ignore

--- a/doc/ffplay.texi
+++ b/doc/ffplay.texi
@@ -34,6 +34,10 @@ various FFmpeg APIs.
 Force displayed width.
@item -y @var{height}
 Force displayed height.
+@item -s @var{size}
+Set frame size (WxH or abbreviation), needed for videos which do
+not contain a header with the frame size like raw YUV.  This option
+has been deprecated in favor of private options, try -video_size.
@item -fs
 Start in fullscreen mode.
@item -an
@@ -122,6 +126,10 @@ Read @var{input_url}.

@section Advanced options
@table @option
+@item -pix_fmt @var{format}
+Set pixel format.
+This option has been deprecated in favor of private options, try -pixel_format.
+
@item -stats
 Print several playback statistics, in particular show the stream
 duration, the codec parameters, the current position in the stream and
@@ -214,6 +222,8 @@ Pause.
 Toggle mute.

@item 9, 0
+Decrease and increase volume respectively.
+
@item /, *
 Decrease and increase volume respectively.

--- a/doc/ffprobe.texi
+++ b/doc/ffprobe.texi
@@ -12,7 +12,7 @@

@chapter Synopsis

-ffprobe [@var{options}] @file{input_url}
+ffprobe [@var{options}] [@file{input_url}]

@chapter Description
@c man begin DESCRIPTION
@@ -28,9 +28,6 @@ If a url is specified in input, ffprobe will try to open and
 probe the url content. If the url cannot be opened or recognized as
 a multimedia file, a positive exit code is returned.

-If no output is specified as output with @option{o} ffprobe will write
-to stdout.
-
 ffprobe may be employed both as a standalone application or in
 combination with a textual filter, which may perform more
 sophisticated processing, e.g. statistical processing or plotting.
@@ -41,7 +38,7 @@ ffprobe will show it.

 ffprobe output is designed to be easily parsable by a textual filter,
 and consists of one or more sections of a form defined by the selected
-writer, which is specified by the @option{output_format} option.
+writer, which is specified by the @option{print_format} option.

 Sections may contain other nested sections, and are identified by a
 name (which may be shared by other sections), and an unique
@@ -83,7 +80,7 @@ Use sexagesimal format HH:MM:SS.MICROSECONDS for time values.
 Prettify the format of the displayed values, it corresponds to the
 options "-unit -prefix -byte_binary_prefix -sexagesimal".

-@item -output_format, -of, -print_format @var{writer_name}[=@var{writer_options}]
+@item -of, -print_format @var{writer_name}[=@var{writer_options}]
 Set the output printing format.

@var{writer_name} specifies the name of the writer, and
@@ -91,7 +88,7 @@ Set the output printing format.

 For example for printing the output in JSON format, specify:
@example
-output_format json
+-print_format json
@end example

 For more details on the available output printing formats, see the
@@ -338,12 +335,6 @@ Show information about all pixel formats supported by FFmpeg.
 Pixel format information for each format is printed within a section
 with name "PIXEL_FORMAT".

-@item -show_optional_fields @var{value}
-Some writers viz. JSON and XML, omit the printing of fields with invalid or non-applicable values,
-while other writers always print them. This option enables one to control this behaviour.
-Valid values are @code{always}/@code{1}, @code{never}/@code{0} and @code{auto}/@code{-1}.
-Default is @var{auto}.
-
@item -bitexact
 Force bitexact output, useful to produce output which is not dependent
 on the specific build.
@@ -351,10 +342,6 @@ on the specific build.
@item -i @var{input_url}
 Read @var{input_url}.

-@item -o @var{output_url}
-Write output to @var{output_url}. If not specified, the output is sent
-to stdout.
-
@end table
@c man end

--- a/doc/ffprobe.xsd
+++ b/doc/ffprobe.xsd
@@ -1,409 +1,389 @@
 <?xml version="1.0" encoding="UTF-8"?>

 <xsd:schema xmlns:xsd="http://www.w3.org/2001/XMLSchema"
-            targetNamespace="http://www.ffmpeg.org/schema/ffprobe"
-            xmlns:ffprobe="http://www.ffmpeg.org/schema/ffprobe">
+    targetNamespace="http://www.ffmpeg.org/schema/ffprobe"
+    xmlns:ffprobe="http://www.ffmpeg.org/schema/ffprobe">

-  <xsd:element name="ffprobe" type="ffprobe:ffprobeType"/>
+    <xsd:element name="ffprobe" type="ffprobe:ffprobeType"/>

-  <xsd:complexType name="ffprobeType">
-    <xsd:sequence>
-      <xsd:element name="program_version"  type="ffprobe:programVersionType"  minOccurs="0" maxOccurs="1" />
-      <xsd:element name="library_versions" type="ffprobe:libraryVersionsType" minOccurs="0" maxOccurs="1" />
-      <xsd:element name="pixel_formats"    type="ffprobe:pixelFormatsType"    minOccurs="0" maxOccurs="1" />
-      <xsd:element name="packets"  type="ffprobe:packetsType" minOccurs="0" maxOccurs="1" />
-      <xsd:element name="frames"   type="ffprobe:framesType"  minOccurs="0" maxOccurs="1" />
-      <xsd:element name="packets_and_frames" type="ffprobe:packetsAndFramesType" minOccurs="0" maxOccurs="1" />
-      <xsd:element name="programs" type="ffprobe:programsType" minOccurs="0" maxOccurs="1" />
-      <xsd:element name="streams"  type="ffprobe:streamsType" minOccurs="0" maxOccurs="1" />
-      <xsd:element name="chapters" type="ffprobe:chaptersType" minOccurs="0" maxOccurs="1" />
-      <xsd:element name="format"   type="ffprobe:formatType"  minOccurs="0" maxOccurs="1" />
-      <xsd:element name="error"    type="ffprobe:errorType"   minOccurs="0" maxOccurs="1" />
-    </xsd:sequence>
-  </xsd:complexType>
+    <xsd:complexType name="ffprobeType">
+        <xsd:sequence>
+            <xsd:element name="program_version"  type="ffprobe:programVersionType"  minOccurs="0" maxOccurs="1" />
+            <xsd:element name="library_versions" type="ffprobe:libraryVersionsType" minOccurs="0" maxOccurs="1" />
+            <xsd:element name="pixel_formats"    type="ffprobe:pixelFormatsType"    minOccurs="0" maxOccurs="1" />
+            <xsd:element name="packets"  type="ffprobe:packetsType" minOccurs="0" maxOccurs="1" />
+            <xsd:element name="frames"   type="ffprobe:framesType"  minOccurs="0" maxOccurs="1" />
+            <xsd:element name="packets_and_frames" type="ffprobe:packetsAndFramesType" minOccurs="0" maxOccurs="1" />
+            <xsd:element name="programs" type="ffprobe:programsType" minOccurs="0" maxOccurs="1" />
+            <xsd:element name="streams"  type="ffprobe:streamsType" minOccurs="0" maxOccurs="1" />
+            <xsd:element name="chapters" type="ffprobe:chaptersType" minOccurs="0" maxOccurs="1" />
+            <xsd:element name="format"   type="ffprobe:formatType"  minOccurs="0" maxOccurs="1" />
+            <xsd:element name="error"    type="ffprobe:errorType"   minOccurs="0" maxOccurs="1" />
+        </xsd:sequence>
+    </xsd:complexType>

-  <xsd:complexType name="packetsType">
-    <xsd:sequence>
-      <xsd:element name="packet" type="ffprobe:packetType" minOccurs="0" maxOccurs="unbounded"/>
-    </xsd:sequence>
-  </xsd:complexType>
+    <xsd:complexType name="packetsType">
+        <xsd:sequence>
+            <xsd:element name="packet" type="ffprobe:packetType" minOccurs="0" maxOccurs="unbounded"/>
+        </xsd:sequence>
+    </xsd:complexType>

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

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

-  <xsd:complexType name="tagsType">
-    <xsd:sequence>
-      <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
-    </xsd:sequence>
-  </xsd:complexType>
+    <xsd:complexType name="packetType">
+      <xsd:sequence>
+        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
+        <xsd:element name="side_data_list" type="ffprobe:packetSideDataListType"   minOccurs="0" maxOccurs="1" />
+      </xsd:sequence>

-  <xsd:complexType name="packetType">
-    <xsd:sequence>
-      <xsd:element name="tags" type="ffprobe:tagsType" minOccurs="0" maxOccurs="1"/>
-      <xsd:element name="side_data_list" type="ffprobe:packetSideDataListType"   minOccurs="0" maxOccurs="1" />
-    </xsd:sequence>
+      <xsd:attribute name="codec_type"    type="xsd:string" use="required" />
+      <xsd:attribute name="stream_index"  type="xsd:int" use="required" />
+      <xsd:attribute name="pts"           type="xsd:long"  />
+      <xsd:attribute name="pts_time"      type="xsd:float" />
+      <xsd:attribute name="dts"           type="xsd:long"  />
+      <xsd:attribute name="dts_time"      type="xsd:float" />
+      <xsd:attribute name="duration"      type="xsd:long"  />
+      <xsd:attribute name="duration_time" type="xsd:float" />
+      <xsd:attribute name="size"          type="xsd:long" use="required" />
+      <xsd:attribute name="pos"           type="xsd:long"  />
+      <xsd:attribute name="flags"         type="xsd:string" use="required" />
+      <xsd:attribute name="data"          type="xsd:string" />
+      <xsd:attribute name="data_hash"     type="xsd:string" />
+    </xsd:complexType>

-    <xsd:attribute name="codec_type"    type="xsd:string" use="required" />
-    <xsd:attribute name="stream_index"  type="xsd:int" use="required" />
-    <xsd:attribute name="pts"           type="xsd:long"  />
-    <xsd:attribute name="pts_time"      type="xsd:float" />
-    <xsd:attribute name="dts"           type="xsd:long"  />
-    <xsd:attribute name="dts_time"      type="xsd:float" />
-    <xsd:attribute name="duration"      type="xsd:long"  />
-    <xsd:attribute name="duration_time" type="xsd:float" />
-    <xsd:attribute name="size"          type="xsd:long" use="required" />
-    <xsd:attribute name="pos"           type="xsd:long"  />
-    <xsd:attribute name="flags"         type="xsd:string" use="required" />
-    <xsd:attribute name="data"          type="xsd:string" />
-    <xsd:attribute name="data_hash"     type="xsd:string" />
-  </xsd:complexType>
+    <xsd:complexType name="packetSideDataListType">
+        <xsd:sequence>
+            <xsd:element name="side_data" type="ffprobe:packetSideDataType" minOccurs="1" maxOccurs="unbounded"/>
+        </xsd:sequence>
+    </xsd:complexType>
+    <xsd:complexType name="packetSideDataType">
+        <xsd:attribute name="side_data_type"              type="xsd:string"/>
+        <xsd:attribute name="side_data_size"              type="xsd:int"   />
+    </xsd:complexType>

-  <xsd:complexType name="packetSideDataListType">
-    <xsd:sequence>
-      <xsd:element name="side_data" type="ffprobe:packetSideDataType" minOccurs="1" maxOccurs="unbounded"/>
-    </xsd:sequence>
-  </xsd:complexType>
+    <xsd:complexType name="frameType">
+      <xsd:sequence>
+            <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
+            <xsd:element name="logs" type="ffprobe:logsType" minOccurs="0" maxOccurs="1"/>
+            <xsd:element name="side_data_list" type="ffprobe:frameSideDataListType"   minOccurs="0" maxOccurs="1" />
+      </xsd:sequence>

-  <xsd:complexType name="packetSideDataType">
-    <xsd:sequence>
-      <xsd:element name="side_datum" type="ffprobe:packetSideDatumType" minOccurs="1" maxOccurs="unbounded"/>
-    </xsd:sequence>
+      <xsd:attribute name="media_type"    type="xsd:string" use="required"/>
+      <xsd:attribute name="stream_index"  type="xsd:int"  />
+      <xsd:attribute name="key_frame"     type="xsd:int"    use="required"/>
+      <xsd:attribute name="pts"           type="xsd:long" />
+      <xsd:attribute name="pts_time"      type="xsd:float"/>
+      <xsd:attribute name="pkt_pts"       type="xsd:long" />
+      <xsd:attribute name="pkt_pts_time"  type="xsd:float"/>
+      <xsd:attribute name="pkt_dts"       type="xsd:long" />
+      <xsd:attribute name="pkt_dts_time"  type="xsd:float"/>
+      <xsd:attribute name="best_effort_timestamp"      type="xsd:long" />
+      <xsd:attribute name="best_effort_timestamp_time" type="xsd:float" />
+      <xsd:attribute name="pkt_duration"  type="xsd:long" />
+      <xsd:attribute name="pkt_duration_time" type="xsd:float"/>
+      <xsd:attribute name="pkt_pos"       type="xsd:long" />
+      <xsd:attribute name="pkt_size"      type="xsd:int" />

-    <xsd:attribute name="type" type="xsd:string"/>
-  </xsd:complexType>
+      <!-- audio attributes -->
+      <xsd:attribute name="sample_fmt"             type="xsd:string"/>
+      <xsd:attribute name="nb_samples"             type="xsd:long"  />
+      <xsd:attribute name="channels"               type="xsd:int"   />
+      <xsd:attribute name="channel_layout"         type="xsd:string"/>

-  <xsd:complexType name="packetSideDatumType">
-    <xsd:attribute name="key"   type="xsd:string"/>
-    <xsd:attribute name="value" type="xsd:string"/>
-  </xsd:complexType>
+      <!-- video attributes -->
+      <xsd:attribute name="width"                  type="xsd:long"  />
+      <xsd:attribute name="height"                 type="xsd:long"  />
+      <xsd:attribute name="pix_fmt"                type="xsd:string"/>
+      <xsd:attribute name="sample_aspect_ratio"    type="xsd:string"/>
+      <xsd:attribute name="pict_type"              type="xsd:string"/>
+      <xsd:attribute name="coded_picture_number"   type="xsd:long"  />
+      <xsd:attribute name="display_picture_number" type="xsd:long"  />
+      <xsd:attribute name="interlaced_frame"       type="xsd:int"   />
+      <xsd:attribute name="top_field_first"        type="xsd:int"   />
+      <xsd:attribute name="repeat_pict"            type="xsd:int"   />
+      <xsd:attribute name="color_range"            type="xsd:string"/>
+      <xsd:attribute name="color_space"            type="xsd:string"/>
+      <xsd:attribute name="color_primaries"        type="xsd:string"/>
+      <xsd:attribute name="color_transfer"         type="xsd:string"/>
+      <xsd:attribute name="chroma_location"        type="xsd:string"/>
+    </xsd:complexType>

-  <xsd:complexType name="frameType">
-    <xsd:sequence>
-      <xsd:element name="tags" type="ffprobe:tagsType" minOccurs="0" maxOccurs="1"/>
-      <xsd:element name="logs" type="ffprobe:logsType" minOccurs="0" maxOccurs="1"/>
-      <xsd:element name="side_data_list" type="ffprobe:frameSideDataListType"   minOccurs="0" maxOccurs="1" />
-    </xsd:sequence>
+    <xsd:complexType name="logsType">
+        <xsd:sequence>
+            <xsd:element name="log" type="ffprobe:logType" minOccurs="1" maxOccurs="unbounded"/>
+        </xsd:sequence>
+    </xsd:complexType>
+    <xsd:complexType name="logType">
+        <xsd:attribute name="context"                     type="xsd:string"/>
+        <xsd:attribute name="level"                       type="xsd:int"   />
+        <xsd:attribute name="category"                    type="xsd:int"   />
+        <xsd:attribute name="parent_context"              type="xsd:string"/>
+        <xsd:attribute name="parent_category"             type="xsd:int"   />
+        <xsd:attribute name="message"                     type="xsd:string"/>
+    </xsd:complexType>

-    <xsd:attribute name="media_type"    type="xsd:string" use="required"/>
-    <xsd:attribute name="stream_index"  type="xsd:int"  />
-    <xsd:attribute name="key_frame"     type="xsd:int"    use="required"/>
-    <xsd:attribute name="pts"           type="xsd:long" />
-    <xsd:attribute name="pts_time"      type="xsd:float"/>
-    <xsd:attribute name="pkt_dts"       type="xsd:long" />
-    <xsd:attribute name="pkt_dts_time"  type="xsd:float"/>
-    <xsd:attribute name="best_effort_timestamp"      type="xsd:long" />
-    <xsd:attribute name="best_effort_timestamp_time" type="xsd:float" />
-    <xsd:attribute name="pkt_duration"  type="xsd:long" />
-    <xsd:attribute name="pkt_duration_time" type="xsd:float"/>
-    <xsd:attribute name="duration"      type="xsd:long" />
-    <xsd:attribute name="duration_time" type="xsd:float"/>
-    <xsd:attribute name="pkt_pos"       type="xsd:long" />
-    <xsd:attribute name="pkt_size"      type="xsd:int" />
+    <xsd:complexType name="frameSideDataListType">
+        <xsd:sequence>
+            <xsd:element name="side_data" type="ffprobe:frameSideDataType" minOccurs="1" maxOccurs="unbounded"/>
+        </xsd:sequence>
+    </xsd:complexType>
+    <xsd:complexType name="frameSideDataType">
+        <xsd:sequence>
+            <xsd:element name="timecodes" type="ffprobe:frameSideDataTimecodeList" minOccurs="0" maxOccurs="1"/>
+        </xsd:sequence>

-    <!-- audio attributes -->
-    <xsd:attribute name="sample_fmt"             type="xsd:string"/>
-    <xsd:attribute name="nb_samples"             type="xsd:long"  />
-    <xsd:attribute name="channels"               type="xsd:int"   />
-    <xsd:attribute name="channel_layout"         type="xsd:string"/>
+        <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>

-    <!-- video attributes -->
-    <xsd:attribute name="width"                  type="xsd:long"  />
-    <xsd:attribute name="height"                 type="xsd:long"  />
-    <xsd:attribute name="crop_top"               type="xsd:long"  />
-    <xsd:attribute name="crop_bottom"            type="xsd:long"  />
-    <xsd:attribute name="crop_left"              type="xsd:long"  />
-    <xsd:attribute name="crop_right"             type="xsd:long"  />
-    <xsd:attribute name="pix_fmt"                type="xsd:string"/>
-    <xsd:attribute name="sample_aspect_ratio"    type="xsd:string"/>
-    <xsd:attribute name="pict_type"              type="xsd:string"/>
-    <xsd:attribute name="coded_picture_number"   type="xsd:long"  />
-    <xsd:attribute name="display_picture_number" type="xsd:long"  />
-    <xsd:attribute name="interlaced_frame"       type="xsd:int"   />
-    <xsd:attribute name="top_field_first"        type="xsd:int"   />
-    <xsd:attribute name="repeat_pict"            type="xsd:int"   />
-    <xsd:attribute name="color_range"            type="xsd:string"/>
-    <xsd:attribute name="color_space"            type="xsd:string"/>
-    <xsd:attribute name="color_primaries"        type="xsd:string"/>
-    <xsd:attribute name="color_transfer"         type="xsd:string"/>
-    <xsd:attribute name="chroma_location"        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="logsType">
-    <xsd:sequence>
-      <xsd:element name="log" type="ffprobe:logType" minOccurs="1" maxOccurs="unbounded"/>
-    </xsd:sequence>
-  </xsd:complexType>
-  <xsd:complexType name="logType">
-    <xsd:attribute name="context"                     type="xsd:string"/>
-    <xsd:attribute name="level"                       type="xsd:int"   />
-    <xsd:attribute name="category"                    type="xsd:int"   />
-    <xsd:attribute name="parent_context"              type="xsd:string"/>
-    <xsd:attribute name="parent_category"             type="xsd:int"   />
-    <xsd:attribute name="message"                     type="xsd:string"/>
-  </xsd:complexType>
+    <xsd:complexType name="frameSideDataTimecodeType">
+        <xsd:attribute name="value"              type="xsd:string"/>
+    </xsd:complexType>

-  <xsd:complexType name="frameSideDataListType">
-    <xsd:sequence>
-      <xsd:element name="side_data" type="ffprobe:frameSideDataType" minOccurs="1" maxOccurs="unbounded"/>
-    </xsd:sequence>
-  </xsd:complexType>
-  <xsd:complexType name="frameSideDataType">
-    <xsd:sequence>
-      <xsd:element name="timecodes" type="ffprobe:frameSideDataTimecodeList" minOccurs="0" maxOccurs="1"/>
-    </xsd:sequence>
+    <xsd:complexType name="subtitleType">
+      <xsd:attribute name="media_type"         type="xsd:string" fixed="subtitle" use="required"/>
+      <xsd:attribute name="pts"                type="xsd:long" />
+      <xsd:attribute name="pts_time"           type="xsd:float"/>
+      <xsd:attribute name="format"             type="xsd:int"  />
+      <xsd:attribute name="start_display_time" type="xsd:int"  />
+      <xsd:attribute name="end_display_time"   type="xsd:int"  />
+      <xsd:attribute name="num_rects"          type="xsd:int"  />
+    </xsd:complexType>

-    <xsd: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="streamsType">
+        <xsd:sequence>
+            <xsd:element name="stream" type="ffprobe:streamType" minOccurs="0" maxOccurs="unbounded"/>
+        </xsd:sequence>
+    </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="programsType">
+        <xsd:sequence>
+            <xsd:element name="program" type="ffprobe:programType" minOccurs="0" maxOccurs="unbounded"/>
+        </xsd:sequence>
+    </xsd:complexType>

-  <xsd:complexType name="frameSideDataTimecodeType">
-    <xsd:attribute name="value"              type="xsd:string"/>
-  </xsd:complexType>
+    <xsd:complexType name="streamDispositionType">
+      <xsd:attribute name="default"          type="xsd:int" use="required" />
+      <xsd:attribute name="dub"              type="xsd:int" use="required" />
+      <xsd:attribute name="original"         type="xsd:int" use="required" />
+      <xsd:attribute name="comment"          type="xsd:int" use="required" />
+      <xsd:attribute name="lyrics"           type="xsd:int" use="required" />
+      <xsd:attribute name="karaoke"          type="xsd:int" use="required" />
+      <xsd:attribute name="forced"           type="xsd:int" use="required" />
+      <xsd:attribute name="hearing_impaired" type="xsd:int" use="required" />
+      <xsd:attribute name="visual_impaired"  type="xsd:int" use="required" />
+      <xsd:attribute name="clean_effects"    type="xsd:int" use="required" />
+      <xsd:attribute name="attached_pic"     type="xsd:int" use="required" />
+      <xsd:attribute name="timed_thumbnails" type="xsd:int" use="required" />
+    </xsd:complexType>

-  <xsd:complexType name="subtitleType">
-    <xsd:attribute name="media_type"         type="xsd:string" fixed="subtitle" use="required"/>
-    <xsd:attribute name="pts"                type="xsd:long" />
-    <xsd:attribute name="pts_time"           type="xsd:float"/>
-    <xsd:attribute name="format"             type="xsd:int"  />
-    <xsd:attribute name="start_display_time" type="xsd:int"  />
-    <xsd:attribute name="end_display_time"   type="xsd:int"  />
-    <xsd:attribute name="num_rects"          type="xsd:int"  />
-  </xsd:complexType>
+    <xsd:complexType name="streamType">
+      <xsd:sequence>
+        <xsd:element name="disposition" type="ffprobe:streamDispositionType" minOccurs="0" maxOccurs="1"/>
+        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
+        <xsd:element name="side_data_list" type="ffprobe:packetSideDataListType"   minOccurs="0" maxOccurs="1" />
+      </xsd:sequence>

-  <xsd:complexType name="streamsType">
-    <xsd:sequence>
-      <xsd:element name="stream" type="ffprobe:streamType" minOccurs="0" maxOccurs="unbounded"/>
-    </xsd:sequence>
-  </xsd:complexType>
+      <xsd:attribute name="index"            type="xsd:int" use="required"/>
+      <xsd:attribute name="codec_name"       type="xsd:string" />
+      <xsd:attribute name="codec_long_name"  type="xsd:string" />
+      <xsd:attribute name="profile"          type="xsd:string" />
+      <xsd:attribute name="codec_type"       type="xsd:string" />
+      <xsd:attribute name="codec_tag"        type="xsd:string" use="required"/>
+      <xsd:attribute name="codec_tag_string" type="xsd:string" use="required"/>
+      <xsd:attribute name="extradata"        type="xsd:string" />
+      <xsd:attribute name="extradata_hash"   type="xsd:string" />

-  <xsd:complexType name="programsType">
-    <xsd:sequence>
-      <xsd:element name="program" type="ffprobe:programType" minOccurs="0" maxOccurs="unbounded"/>
-    </xsd:sequence>
-  </xsd:complexType>
+      <!-- video attributes -->
+      <xsd:attribute name="width"                type="xsd:int"/>
+      <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"/>
+      <xsd:attribute name="pix_fmt"              type="xsd:string"/>
+      <xsd:attribute name="level"                type="xsd:int"/>
+      <xsd:attribute name="color_range"          type="xsd:string"/>
+      <xsd:attribute name="color_space"          type="xsd:string"/>
+      <xsd:attribute name="color_transfer"       type="xsd:string"/>
+      <xsd:attribute name="color_primaries"      type="xsd:string"/>
+      <xsd:attribute name="chroma_location"      type="xsd:string"/>
+      <xsd:attribute name="field_order"          type="xsd:string"/>
+      <xsd:attribute name="refs"                 type="xsd:int"/>

-  <xsd:complexType name="streamDispositionType">
-    <xsd:attribute name="default"          type="xsd:int" use="required" />
-    <xsd:attribute name="dub"              type="xsd:int" use="required" />
-    <xsd:attribute name="original"         type="xsd:int" use="required" />
-    <xsd:attribute name="comment"          type="xsd:int" use="required" />
-    <xsd:attribute name="lyrics"           type="xsd:int" use="required" />
-    <xsd:attribute name="karaoke"          type="xsd:int" use="required" />
-    <xsd:attribute name="forced"           type="xsd:int" use="required" />
-    <xsd:attribute name="hearing_impaired" type="xsd:int" use="required" />
-    <xsd:attribute name="visual_impaired"  type="xsd:int" use="required" />
-    <xsd:attribute name="clean_effects"    type="xsd:int" use="required" />
-    <xsd:attribute name="attached_pic"     type="xsd:int" use="required" />
-    <xsd:attribute name="timed_thumbnails" type="xsd:int" use="required" />
-    <xsd:attribute name="non_diegetic"     type="xsd:int" use="required" />
-    <xsd:attribute name="captions"         type="xsd:int" use="required" />
-    <xsd:attribute name="descriptions"     type="xsd:int" use="required" />
-    <xsd:attribute name="metadata"         type="xsd:int" use="required" />
-    <xsd:attribute name="dependent"        type="xsd:int" use="required" />
-    <xsd:attribute name="still_image"      type="xsd:int" use="required" />
-  </xsd:complexType>
+      <!-- audio attributes -->
+      <xsd:attribute name="sample_fmt"       type="xsd:string"/>
+      <xsd:attribute name="sample_rate"      type="xsd:int"/>
+      <xsd:attribute name="channels"         type="xsd:int"/>
+      <xsd:attribute name="channel_layout"   type="xsd:string"/>
+      <xsd:attribute name="bits_per_sample"  type="xsd:int"/>

-  <xsd:complexType name="streamType">
-    <xsd:sequence>
-      <xsd:element name="disposition" type="ffprobe:streamDispositionType" minOccurs="0" maxOccurs="1"/>
-      <xsd:element name="tags" type="ffprobe:tagsType" minOccurs="0" maxOccurs="1"/>
-      <xsd:element name="side_data_list" type="ffprobe:packetSideDataListType"   minOccurs="0" maxOccurs="1" />
-    </xsd:sequence>
+      <xsd:attribute name="id"               type="xsd:string"/>
+      <xsd:attribute name="r_frame_rate"     type="xsd:string" use="required"/>
+      <xsd:attribute name="avg_frame_rate"   type="xsd:string" use="required"/>
+      <xsd:attribute name="time_base"        type="xsd:string" use="required"/>
+      <xsd:attribute name="start_pts"        type="xsd:long"/>
+      <xsd:attribute name="start_time"       type="xsd:float"/>
+      <xsd:attribute name="duration_ts"      type="xsd:long"/>
+      <xsd:attribute name="duration"         type="xsd:float"/>
+      <xsd:attribute name="bit_rate"         type="xsd:int"/>
+      <xsd:attribute name="max_bit_rate"     type="xsd:int"/>
+      <xsd:attribute name="bits_per_raw_sample" type="xsd:int"/>
+      <xsd:attribute name="nb_frames"        type="xsd:int"/>
+      <xsd:attribute name="nb_read_frames"   type="xsd:int"/>
+      <xsd:attribute name="nb_read_packets"  type="xsd:int"/>
+    </xsd:complexType>

-    <xsd:attribute name="index"            type="xsd:int" use="required"/>
-    <xsd:attribute name="codec_name"       type="xsd:string" />
-    <xsd:attribute name="codec_long_name"  type="xsd:string" />
-    <xsd:attribute name="profile"          type="xsd:string" />
-    <xsd:attribute name="codec_type"       type="xsd:string" />
-    <xsd:attribute name="codec_tag"        type="xsd:string" use="required"/>
-    <xsd:attribute name="codec_tag_string" type="xsd:string" use="required"/>
-    <xsd:attribute name="extradata"        type="xsd:string" />
-    <xsd:attribute name="extradata_size"   type="xsd:int"    />
-    <xsd:attribute name="extradata_hash"   type="xsd:string" />
+    <xsd:complexType name="programType">
+      <xsd:sequence>
+        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
+        <xsd:element name="streams" type="ffprobe:streamsType" minOccurs="0" maxOccurs="1"/>
+      </xsd:sequence>

-    <!-- video attributes -->
-    <xsd:attribute name="width"                type="xsd:int"/>
-    <xsd:attribute name="height"               type="xsd:int"/>
-    <xsd:attribute name="coded_width"          type="xsd:int"/>
-    <xsd:attribute name="coded_height"         type="xsd:int"/>
-    <xsd:attribute name="closed_captions"      type="xsd:boolean"/>
-    <xsd:attribute name="film_grain"           type="xsd:boolean"/>
-    <xsd:attribute name="has_b_frames"         type="xsd:int"/>
-    <xsd:attribute name="sample_aspect_ratio"  type="xsd:string"/>
-    <xsd:attribute name="display_aspect_ratio" type="xsd:string"/>
-    <xsd:attribute name="pix_fmt"              type="xsd:string"/>
-    <xsd:attribute name="level"                type="xsd:int"/>
-    <xsd:attribute name="color_range"          type="xsd:string"/>
-    <xsd:attribute name="color_space"          type="xsd:string"/>
-    <xsd:attribute name="color_transfer"       type="xsd:string"/>
-    <xsd:attribute name="color_primaries"      type="xsd:string"/>
-    <xsd:attribute name="chroma_location"      type="xsd:string"/>
-    <xsd:attribute name="field_order"          type="xsd:string"/>
-    <xsd:attribute name="refs"                 type="xsd:int"/>
+      <xsd:attribute name="program_id"           type="xsd:int"    use="required"/>
+      <xsd:attribute name="program_num"          type="xsd:int"    use="required"/>
+      <xsd:attribute name="nb_streams"           type="xsd:int"    use="required"/>
+      <xsd:attribute name="start_time"           type="xsd:float"/>
+      <xsd:attribute name="start_pts"            type="xsd:long"/>
+      <xsd:attribute name="end_time"             type="xsd:float"/>
+      <xsd:attribute name="end_pts"              type="xsd:long"/>
+      <xsd:attribute name="pmt_pid"              type="xsd:int"    use="required"/>
+      <xsd:attribute name="pcr_pid"              type="xsd:int"    use="required"/>
+    </xsd:complexType>

-    <!-- audio attributes -->
-    <xsd:attribute name="sample_fmt"       type="xsd:string"/>
-    <xsd:attribute name="sample_rate"      type="xsd:int"/>
-    <xsd:attribute name="channels"         type="xsd:int"/>
-    <xsd:attribute name="channel_layout"   type="xsd:string"/>
-    <xsd:attribute name="bits_per_sample"  type="xsd:int"/>
-    <xsd:attribute name="initial_padding"  type="xsd:int"/>
+    <xsd:complexType name="formatType">
+      <xsd:sequence>
+        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
+      </xsd:sequence>

-    <xsd:attribute name="id"               type="xsd:string"/>
-    <xsd:attribute name="r_frame_rate"     type="xsd:string" use="required"/>
-    <xsd:attribute name="avg_frame_rate"   type="xsd:string" use="required"/>
-    <xsd:attribute name="time_base"        type="xsd:string" use="required"/>
-    <xsd:attribute name="start_pts"        type="xsd:long"/>
-    <xsd:attribute name="start_time"       type="xsd:float"/>
-    <xsd:attribute name="duration_ts"      type="xsd:long"/>
-    <xsd:attribute name="duration"         type="xsd:float"/>
-    <xsd:attribute name="bit_rate"         type="xsd:int"/>
-    <xsd:attribute name="max_bit_rate"     type="xsd:int"/>
-    <xsd:attribute name="bits_per_raw_sample" type="xsd:int"/>
-    <xsd:attribute name="nb_frames"        type="xsd:int"/>
-    <xsd:attribute name="nb_read_frames"   type="xsd:int"/>
-    <xsd:attribute name="nb_read_packets"  type="xsd:int"/>
-  </xsd:complexType>
+      <xsd:attribute name="filename"         type="xsd:string" use="required"/>
+      <xsd:attribute name="nb_streams"       type="xsd:int"    use="required"/>
+      <xsd:attribute name="nb_programs"      type="xsd:int"    use="required"/>
+      <xsd:attribute name="format_name"      type="xsd:string" use="required"/>
+      <xsd:attribute name="format_long_name" type="xsd:string"/>
+      <xsd:attribute name="start_time"       type="xsd:float"/>
+      <xsd:attribute name="duration"         type="xsd:float"/>
+      <xsd:attribute name="size"             type="xsd:long"/>
+      <xsd:attribute name="bit_rate"         type="xsd:long"/>
+      <xsd:attribute name="probe_score"      type="xsd:int"/>
+    </xsd:complexType>

-  <xsd:complexType name="programType">
-    <xsd:sequence>
-      <xsd:element name="tags" type="ffprobe:tagsType" minOccurs="0" maxOccurs="1"/>
-      <xsd:element name="streams" type="ffprobe:streamsType" minOccurs="0" maxOccurs="1"/>
-    </xsd:sequence>
+    <xsd:complexType name="tagType">
+      <xsd:attribute name="key"   type="xsd:string" use="required"/>
+      <xsd:attribute name="value" type="xsd:string" use="required"/>
+    </xsd:complexType>

-    <xsd:attribute name="program_id"           type="xsd:int"    use="required"/>
-    <xsd:attribute name="program_num"          type="xsd:int"    use="required"/>
-    <xsd:attribute name="nb_streams"           type="xsd:int"    use="required"/>
-    <xsd:attribute name="pmt_pid"              type="xsd:int"    use="required"/>
-    <xsd:attribute name="pcr_pid"              type="xsd:int"    use="required"/>
-  </xsd:complexType>
+    <xsd:complexType name="errorType">
+      <xsd:attribute name="code"   type="xsd:int"    use="required"/>
+      <xsd:attribute name="string" type="xsd:string" use="required"/>
+    </xsd:complexType>

-  <xsd:complexType name="formatType">
-    <xsd:sequence>
-      <xsd:element name="tags" type="ffprobe:tagsType" minOccurs="0" maxOccurs="1"/>
-    </xsd:sequence>
+    <xsd:complexType name="programVersionType">
+      <xsd:attribute name="version"          type="xsd:string" use="required"/>
+      <xsd:attribute name="copyright"        type="xsd:string" use="required"/>
+      <xsd:attribute name="build_date"       type="xsd:string"/>
+      <xsd:attribute name="build_time"       type="xsd:string"/>
+      <xsd:attribute name="compiler_ident"   type="xsd:string" use="required"/>
+      <xsd:attribute name="configuration"    type="xsd:string" use="required"/>
+    </xsd:complexType>

-    <xsd:attribute name="filename"         type="xsd:string" use="required"/>
-    <xsd:attribute name="nb_streams"       type="xsd:int"    use="required"/>
-    <xsd:attribute name="nb_programs"      type="xsd:int"    use="required"/>
-    <xsd:attribute name="format_name"      type="xsd:string" use="required"/>
-    <xsd:attribute name="format_long_name" type="xsd:string"/>
-    <xsd:attribute name="start_time"       type="xsd:float"/>
-    <xsd:attribute name="duration"         type="xsd:float"/>
-    <xsd:attribute name="size"             type="xsd:long"/>
-    <xsd:attribute name="bit_rate"         type="xsd:long"/>
-    <xsd:attribute name="probe_score"      type="xsd:int"/>
-  </xsd:complexType>
+    <xsd:complexType name="chaptersType">
+      <xsd:sequence>
+        <xsd:element name="chapter" type="ffprobe:chapterType" minOccurs="0" maxOccurs="unbounded"/>
+      </xsd:sequence>
+    </xsd:complexType>

-  <xsd:complexType name="tagType">
-    <xsd:attribute name="key"   type="xsd:string" use="required"/>
-    <xsd:attribute name="value" type="xsd:string" use="required"/>
-  </xsd:complexType>
+    <xsd:complexType name="chapterType">
+      <xsd:sequence>
+        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
+      </xsd:sequence>

-  <xsd:complexType name="errorType">
-    <xsd:attribute name="code"   type="xsd:int"    use="required"/>
-    <xsd:attribute name="string" type="xsd:string" use="required"/>
-  </xsd:complexType>
+      <xsd:attribute name="id"         type="xsd:int" use="required"/>
+      <xsd:attribute name="time_base"  type="xsd:string" use="required"/>
+      <xsd:attribute name="start"      type="xsd:int" use="required"/>
+      <xsd:attribute name="start_time" type="xsd:float"/>
+      <xsd:attribute name="end"        type="xsd:int" use="required"/>
+      <xsd:attribute name="end_time"   type="xsd:float" use="required"/>
+    </xsd:complexType>

-  <xsd:complexType name="programVersionType">
-    <xsd:attribute name="version"          type="xsd:string" use="required"/>
-    <xsd:attribute name="copyright"        type="xsd:string" use="required"/>
-    <xsd:attribute name="build_date"       type="xsd:string"/>
-    <xsd:attribute name="build_time"       type="xsd:string"/>
-    <xsd:attribute name="compiler_ident"   type="xsd:string" use="required"/>
-    <xsd:attribute name="configuration"    type="xsd:string" use="required"/>
-  </xsd:complexType>
+    <xsd:complexType name="libraryVersionType">
+      <xsd:attribute name="name"        type="xsd:string" use="required"/>
+      <xsd:attribute name="major"       type="xsd:int"    use="required"/>
+      <xsd:attribute name="minor"       type="xsd:int"    use="required"/>
+      <xsd:attribute name="micro"       type="xsd:int"    use="required"/>
+      <xsd:attribute name="version"     type="xsd:int"    use="required"/>
+      <xsd:attribute name="ident"       type="xsd:string" use="required"/>
+    </xsd:complexType>

-  <xsd:complexType name="chaptersType">
-    <xsd:sequence>
-      <xsd:element name="chapter" type="ffprobe:chapterType" minOccurs="0" maxOccurs="unbounded"/>
-    </xsd:sequence>
-  </xsd:complexType>
+    <xsd:complexType name="libraryVersionsType">
+        <xsd:sequence>
+          <xsd:element name="library_version" type="ffprobe:libraryVersionType" minOccurs="0" maxOccurs="unbounded"/>
+        </xsd:sequence>
+    </xsd:complexType>

-  <xsd:complexType name="chapterType">
-    <xsd:sequence>
-      <xsd:element name="tags" type="ffprobe:tagsType" minOccurs="0" maxOccurs="unbounded"/>
-    </xsd:sequence>
+    <xsd:complexType name="pixelFormatFlagsType">
+      <xsd:attribute name="big_endian" type="xsd:int" use="required"/>
+      <xsd:attribute name="palette"    type="xsd:int" use="required"/>
+      <xsd:attribute name="bitstream"  type="xsd:int" use="required"/>
+      <xsd:attribute name="hwaccel"    type="xsd:int" use="required"/>
+      <xsd:attribute name="planar"     type="xsd:int" use="required"/>
+      <xsd:attribute name="rgb"        type="xsd:int" use="required"/>
+      <xsd:attribute name="alpha"      type="xsd:int" use="required"/>
+    </xsd:complexType>

-    <xsd:attribute name="id"         type="xsd:int" use="required"/>
-    <xsd:attribute name="time_base"  type="xsd:string" use="required"/>
-    <xsd:attribute name="start"      type="xsd:int" use="required"/>
-    <xsd:attribute name="start_time" type="xsd:float"/>
-    <xsd:attribute name="end"        type="xsd:int" use="required"/>
-    <xsd:attribute name="end_time"   type="xsd:float" use="required"/>
-  </xsd:complexType>
+    <xsd:complexType name="pixelFormatComponentType">
+      <xsd:attribute name="index"      type="xsd:int" use="required"/>
+      <xsd:attribute name="bit_depth"  type="xsd:int" use="required"/>
+    </xsd:complexType>

-  <xsd:complexType name="libraryVersionType">
-    <xsd:attribute name="name"        type="xsd:string" use="required"/>
-    <xsd:attribute name="major"       type="xsd:int"    use="required"/>
-    <xsd:attribute name="minor"       type="xsd:int"    use="required"/>
-    <xsd:attribute name="micro"       type="xsd:int"    use="required"/>
-    <xsd:attribute name="version"     type="xsd:int"    use="required"/>
-    <xsd:attribute name="ident"       type="xsd:string" use="required"/>
-  </xsd:complexType>
+    <xsd:complexType name="pixelFormatComponentsType">
+      <xsd:sequence>
+        <xsd:element name="component" type="ffprobe:pixelFormatComponentType" minOccurs="0" maxOccurs="unbounded"/>
+      </xsd:sequence>
+    </xsd:complexType>

-  <xsd:complexType name="libraryVersionsType">
-    <xsd:sequence>
-      <xsd:element name="library_version" type="ffprobe:libraryVersionType" minOccurs="0" maxOccurs="unbounded"/>
-    </xsd:sequence>
-  </xsd:complexType>
+    <xsd:complexType name="pixelFormatType">
+      <xsd:sequence>
+        <xsd:element name="flags" type="ffprobe:pixelFormatFlagsType" minOccurs="0" maxOccurs="1"/>
+        <xsd:element name="components" type="ffprobe:pixelFormatComponentsType" minOccurs="0" maxOccurs="1"/>
+      </xsd:sequence>

-  <xsd:complexType name="pixelFormatFlagsType">
-    <xsd:attribute name="big_endian" type="xsd:int" use="required"/>
-    <xsd:attribute name="palette"    type="xsd:int" use="required"/>
-    <xsd:attribute name="bitstream"  type="xsd:int" use="required"/>
-    <xsd:attribute name="hwaccel"    type="xsd:int" use="required"/>
-    <xsd:attribute name="planar"     type="xsd:int" use="required"/>
-    <xsd:attribute name="rgb"        type="xsd:int" use="required"/>
-    <xsd:attribute name="alpha"      type="xsd:int" use="required"/>
-  </xsd:complexType>
+      <xsd:attribute name="name"                type="xsd:string" use="required"/>
+      <xsd:attribute name="nb_components"       type="xsd:int"    use="required"/>
+      <xsd:attribute name="log2_chroma_w"       type="xsd:int"/>
+      <xsd:attribute name="log2_chroma_h"       type="xsd:int"/>
+      <xsd:attribute name="bits_per_pixel"      type="xsd:int"/>
+    </xsd:complexType>

-  <xsd:complexType name="pixelFormatComponentType">
-    <xsd:attribute name="index"      type="xsd:int" use="required"/>
-    <xsd:attribute name="bit_depth"  type="xsd:int" use="required"/>
-  </xsd:complexType>
-
-  <xsd:complexType name="pixelFormatComponentsType">
-    <xsd:sequence>
-      <xsd:element name="component" type="ffprobe:pixelFormatComponentType" minOccurs="0" maxOccurs="unbounded"/>
-    </xsd:sequence>
-  </xsd:complexType>
-
-  <xsd:complexType name="pixelFormatType">
-    <xsd:sequence>
-      <xsd:element name="flags" type="ffprobe:pixelFormatFlagsType" minOccurs="0" maxOccurs="1"/>
-      <xsd:element name="components" type="ffprobe:pixelFormatComponentsType" minOccurs="0" maxOccurs="1"/>
-    </xsd:sequence>
-
-    <xsd:attribute name="name"                type="xsd:string" use="required"/>
-    <xsd:attribute name="nb_components"       type="xsd:int"    use="required"/>
-    <xsd:attribute name="log2_chroma_w"       type="xsd:int"/>
-    <xsd:attribute name="log2_chroma_h"       type="xsd:int"/>
-    <xsd:attribute name="bits_per_pixel"      type="xsd:int"/>
-  </xsd:complexType>
-
-  <xsd:complexType name="pixelFormatsType">
-    <xsd:sequence>
-      <xsd:element name="pixel_format" type="ffprobe:pixelFormatType" minOccurs="0" maxOccurs="unbounded"/>
-    </xsd:sequence>
-  </xsd:complexType>
+    <xsd:complexType name="pixelFormatsType">
+      <xsd:sequence>
+        <xsd:element name="pixel_format" type="ffprobe:pixelFormatType" minOccurs="0" maxOccurs="unbounded"/>
+      </xsd:sequence>
+    </xsd:complexType>
 </xsd:schema>
--- a/doc/fftools-common-opts.texi
+++ b/doc/fftools-common-opts.texi
@@ -167,9 +167,6 @@ Show available sample formats.
@item -layouts
 Show channel names and standard channel layouts.

-@item -dispositions
-Show stream dispositions.
-
@item -colors
 Show recognized color names.

@@ -356,13 +353,6 @@ Possible flags for this option are:
@end table
@end table

-@item -cpucount @var{count} (@emph{global})
-Override detection of CPU count. This option is intended
-for testing. Do not use it unless you know what you're doing.
-@example
-ffmpeg -cpucount 2
-@end example
-
@item -max_alloc @var{bytes}
 Set the maximum size limit for allocating a block on the heap by ffmpeg's
 family of malloc functions. Exercise @strong{extreme caution} when using
--- a/doc/filters.texi
+++ b/doc/filters.texi
--- a/doc/formats.texi
+++ b/doc/formats.texi
@@ -49,6 +49,7 @@ Generate missing PTS if DTS is present.
 Ignore DTS if PTS is set. Inert when nofillin is set.
@item ignidx
 Ignore index.
+@item keepside (@emph{deprecated},@emph{inert})
@item nobuffer
 Reduce the latency introduced by buffering during initial input streams analysis.
@item nofillin
@@ -69,6 +70,7 @@ This ensures that file and data checksums are reproducible and match between
 platforms. Its primary use is for regression testing.
@item flush_packets
 Write out packets immediately.
+@item latm (@emph{deprecated},@emph{inert})
@item shortest
 Stop muxing at the end of the shortest stream.
 It may be needed to increase max_interleave_delta to avoid flushing the longer
--- a/doc/general_contents.texi
+++ b/doc/general_contents.texi
@@ -171,13 +171,6 @@ 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 libjxl
-
-JPEG XL is an image format intended to fully replace legacy JPEG for an extended
-period of life. See @url{https://jpegxl.info/} for more information, and see
-@url{https://github.com/libjxl/libjxl} for the library source. You can pass
-@code{--enable-libjxl} to configure in order enable the libjxl wrapper.
-
@section libvpx

 FFmpeg can make use of the libvpx library for VP8/VP9 decoding and encoding.
@@ -270,7 +263,7 @@ to @file{./configure}.

 FFmpeg can make use of the Scalable Video Technology for AV1 library for AV1 encoding.

-Go to @url{https://gitlab.com/AOMediaCodec/SVT-AV1/} and follow the instructions
+Go to @url{https://github.com/OpenVisualCloud/SVT-AV1/} and follow the instructions
 for installing the library. Then pass @code{--enable-libsvtav1} to configure to
 enable it.

@@ -510,8 +503,6 @@ library:
    @tab A format used by libvpx
@item Internet Video Recording  @tab   @tab X
@item IRCAM                     @tab X @tab X
-@item LAF                       @tab   @tab X
-    @tab Limitless Audio Format
@item LATM                      @tab X @tab X
@item LMLM4                     @tab   @tab X
    @tab Used by Linux Media Labs MPEG-4 PCI boards
@@ -534,8 +525,6 @@ library:
@item Metal Gear Solid: The Twin Snakes @tab @tab X
@item Megalux Frame             @tab   @tab X
    @tab Used by Megalux Ultimate Paint
-@item MobiClip MODS             @tab   @tab X
-@item MobiClip MOFLEX           @tab   @tab X
@item Mobotix .mxg              @tab   @tab X
@item Monkey's Audio            @tab   @tab X
@item Motion Pixels MVI         @tab   @tab X
@@ -579,7 +568,6 @@ library:
@item Ogg                       @tab X @tab X
@item Playstation Portable PMP  @tab   @tab X
@item Portable Voice Format     @tab   @tab X
-@item RK Audio (RKA)            @tab   @tab X
@item TechnoTrend PVA           @tab   @tab X
    @tab Used by TechnoTrend DVB PCI boards.
@item QCP                       @tab   @tab X
@@ -587,12 +575,9 @@ library:
@item raw AC-3                  @tab X @tab X
@item raw AMR-NB                @tab   @tab X
@item raw AMR-WB                @tab   @tab X
-@item raw APAC                  @tab   @tab X
@item raw aptX                  @tab X @tab X
@item raw aptX HD               @tab X @tab X
-@item raw Bonk                  @tab   @tab X
@item raw Chinese AVS video     @tab X @tab X
-@item raw DFPWM                 @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
@@ -614,8 +599,6 @@ library:
@item raw NULL                  @tab X @tab
@item raw video                 @tab X @tab X
@item raw id RoQ                @tab X @tab
-@item raw OBU                   @tab X @tab X
-@item raw OSQ                   @tab   @tab X
@item raw SBC                   @tab X @tab X
@item raw Shorten               @tab   @tab X
@item raw TAK                   @tab   @tab X
@@ -667,10 +650,8 @@ library:
@item Sample Dump eXchange      @tab   @tab X
@item SAP                       @tab X @tab X
@item SBG                       @tab   @tab X
-@item SDNS                      @tab   @tab X
@item SDP                       @tab   @tab X
@item SER                       @tab   @tab X
-@item Digital Pictures SGA      @tab   @tab X
@item Sega FILM/CPK             @tab X @tab X
    @tab Used in many Sega Saturn console games.
@item Silicon Graphics Movie    @tab   @tab X
@@ -708,21 +689,18 @@ library:
@item Vivo                      @tab   @tab X
@item VPK                       @tab   @tab X
    @tab Audio format used in Sony PS games.
-@item Marble WADY               @tab   @tab X
@item WAV                       @tab X @tab X
-@item Waveform Archiver         @tab   @tab X
@item WavPack                   @tab X @tab X
@item WebM                      @tab X @tab X
@item Windows Televison (WTV)   @tab X @tab X
@item Wing Commander III movie  @tab   @tab X
    @tab Multimedia format used in Origin's Wing Commander III computer game.
-@item Westwood Studios audio    @tab X @tab X
+@item Westwood Studios audio    @tab   @tab X
    @tab Multimedia format used in Westwood Studios games.
@item Westwood Studios VQA      @tab   @tab X
    @tab Multimedia format used in Westwood Studios games.
@item Wideband Single-bit Data (WSD) @tab   @tab X
@item WVE                       @tab   @tab X
-@item Konami XMD                @tab   @tab X
@item XMV                       @tab   @tab X
    @tab Microsoft video container used in Xbox games.
@item XVAG                      @tab   @tab X
@@ -762,17 +740,12 @@ following image formats are supported:
    @tab OpenEXR
@item FITS         @tab X @tab X
    @tab Flexible Image Transport System
-@item HDR          @tab X @tab X
-    @tab Radiance HDR RGBE Image format
-@item IMG          @tab   @tab X
-    @tab GEM Raster image
@item JPEG         @tab X @tab X
    @tab Progressive JPEG is not supported.
@item JPEG 2000    @tab X @tab X
@item JPEG-LS      @tab X @tab X
@item LJPEG        @tab X @tab
    @tab Lossless JPEG
-@item Media 100    @tab   @tab X
@item MSP          @tab   @tab X
    @tab Microsoft Paint image
@item PAM          @tab X @tab X
@@ -791,8 +764,6 @@ following image formats are supported:
    @tab PGM with U and V components in YUV 4:2:0
@item PGX          @tab   @tab X
    @tab PGX file decoder
-@item PHM          @tab X @tab X
-    @tab Portable HalfFloatMap image
@item PIC          @tab @tab X
    @tab Pictor/PC Paint
@item PNG          @tab X @tab X
@@ -803,8 +774,6 @@ following image formats are supported:
    @tab Photoshop
@item PTX          @tab   @tab X
    @tab V.Flash PTX format
-@item QOI          @tab X @tab X
-    @tab Quite OK Image format
@item SGI          @tab X @tab X
    @tab SGI RGB image format
@item Sun Rasterfile  @tab X @tab X
@@ -813,10 +782,6 @@ following image formats are supported:
    @tab YUV, JPEG and some extension is not supported yet.
@item Truevision Targa  @tab X @tab X
    @tab Targa (.TGA) image format
-@item VBN  @tab X @tab X
-    @tab Vizrt Binary Image format
-@item WBMP         @tab X @tab X
-    @tab Wireless Application Protocol Bitmap image format
@item WebP         @tab E @tab X
    @tab WebP image format, encoding supported through external library libwebp
@item XBM  @tab X @tab X
@@ -1007,7 +972,7 @@ following image formats are supported:
    @tab Also known as Microsoft Screen 3.
@item Microsoft Expression Encoder Screen  @tab     @tab  X
    @tab Also known as Microsoft Titanium Screen 2.
-@item Microsoft RLE          @tab  X  @tab  X
+@item Microsoft RLE          @tab     @tab  X
@item Microsoft Screen 1     @tab     @tab  X
    @tab Also known as Windows Media Video V7 Screen.
@item Microsoft Screen 2     @tab     @tab  X
@@ -1053,7 +1018,7 @@ following image formats are supported:
@item QuickTime 8BPS video   @tab     @tab  X
@item QuickTime Animation (RLE) video  @tab  X  @tab  X
    @tab fourcc: 'rle '
-@item QuickTime Graphics (SMC)  @tab  X  @tab  X
+@item QuickTime Graphics (SMC)  @tab     @tab  X
    @tab fourcc: 'smc '
@item QuickTime video (RPZA) @tab  X  @tab  X
    @tab fourcc: rpza
@@ -1067,8 +1032,6 @@ following image formats are supported:
@item RealVideo 4.0          @tab     @tab  X
@item Renderware TXD (TeXture Dictionary)  @tab     @tab  X
    @tab Texture dictionaries used by the Renderware Engine.
-@item RivaTuner Video        @tab     @tab  X
-    @tab fourcc: 'RTV1'
@item RL2 video              @tab     @tab  X
    @tab used in some games by Entertainment Software Partners
@item ScreenPressor          @tab     @tab  X
@@ -1105,8 +1068,6 @@ following image formats are supported:
@item v408 QuickTime uncompressed 4:4:4:4          @tab  X  @tab  X
@item v410 QuickTime uncompressed 4:4:4 10-bit     @tab  X  @tab  X
@item VBLE Lossless Codec    @tab     @tab  X
-@item vMix Video             @tab     @tab  X
-    @tab fourcc: 'VMX1'
@item VMware Screen Codec / VMware Video  @tab     @tab  X
    @tab Codec used in videos captured by VMware.
@item Westwood Studios VQA (Vector Quantized Animation) video  @tab     @tab  X
@@ -1165,7 +1126,6 @@ following image formats are supported:
@item ADPCM Electronic Arts XAS @tab     @tab  X
@item ADPCM G.722            @tab  X  @tab  X
@item ADPCM G.726            @tab  X  @tab  X
-@item ADPCM IMA Acorn Replay @tab     @tab  X
@item ADPCM IMA AMV          @tab  X  @tab  X
    @tab Used in AMV files
@item ADPCM IMA Cunning Developments  @tab     @tab  X
@@ -1173,7 +1133,6 @@ following image formats are supported:
@item ADPCM IMA Electronic Arts SEAD  @tab     @tab  X
@item ADPCM IMA Funcom       @tab     @tab  X
@item ADPCM IMA High Voltage Software ALP      @tab  X  @tab  X
-@item ADPCM IMA Mobiclip MOFLEX  @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  X  @tab  X
@@ -1203,8 +1162,7 @@ following image formats are supported:
@item ADPCM Sound Blaster Pro 4-bit  @tab     @tab  X
@item ADPCM VIMA             @tab     @tab  X
    @tab Used in LucasArts SMUSH animations.
-@item ADPCM Konami XMD       @tab     @tab  X
-@item ADPCM Westwood Studios IMA      @tab  X @tab  X
+@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
@@ -1225,7 +1183,6 @@ following image formats are supported:
@item ATRAC9                 @tab     @tab  X
@item Bink Audio             @tab     @tab  X
    @tab Used in Bink and Smacker files in many games.
-@item Bonk audio             @tab     @tab  X
@item CELT                   @tab     @tab  E
    @tab decoding supported through external library libcelt
@item codec2                 @tab  E  @tab  E
@@ -1233,7 +1190,6 @@ following image formats are supported:
@item CRI HCA                @tab     @tab X
@item Delphine Software International CIN audio  @tab     @tab  X
    @tab Codec used in Delphine Software International games.
-@item DFPWM                  @tab  X  @tab  X
@item Digital Speech Standard - Standard Play mode (DSS SP) @tab     @tab  X
@item Discworld II BMV Audio @tab     @tab  X
@item COOK                   @tab     @tab  X
@@ -1241,12 +1197,9 @@ 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 Cuberoot-Delta-Exact @tab  @tab  X
-    @tab Used in few games.
@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 Marble WADY       @tab     @tab  X
@item DPCM Interplay         @tab     @tab  X
    @tab Used in various Interplay computer games.
@item DPCM Squareroot-Delta-Exact  @tab  @tab  X
@@ -1267,7 +1220,6 @@ following image formats are supported:
@item Enhanced AC-3          @tab  X  @tab  X
@item EVRC (Enhanced Variable Rate Codec) @tab     @tab  X
@item FLAC (Free Lossless Audio Codec)  @tab  X  @tab  IX
-@item FTR Voice              @tab     @tab  X
@item G.723.1                @tab X   @tab  X
@item G.729                  @tab     @tab  X
@item GSM                    @tab  E  @tab  X
@@ -1275,14 +1227,12 @@ following image formats are supported:
@item GSM Microsoft variant  @tab  E  @tab  X
    @tab encoding supported through external library libgsm
@item IAC (Indeo Audio Coder)  @tab     @tab  X
-@item iLBC (Internet Low Bitrate Codec) @tab  E  @tab  EX
+@item iLBC (Internet Low Bitrate Codec) @tab  E  @tab  E
    @tab encoding and decoding supported through external library libilbc
@item IMC (Intel Music Coder)  @tab     @tab  X
@item Interplay ACM            @tab     @tab  X
@item MACE (Macintosh Audio Compression/Expansion) 3:1  @tab     @tab  X
@item MACE (Macintosh Audio Compression/Expansion) 6:1  @tab     @tab  X
-@item Marian's A-pac audio     @tab     @tab  X
-@item MI-SC4 (Micronas SC-4 Audio)  @tab     @tab  X
@item MLP (Meridian Lossless Packing)  @tab  X  @tab  X
    @tab Used in DVD-Audio discs.
@item Monkey's Audio         @tab     @tab  X
@@ -1292,14 +1242,12 @@ following image formats are supported:
@item MP3 (MPEG audio layer 3)  @tab  E  @tab IX
    @tab encoding supported through external library LAME, ADU MP3 and MP3onMP4 also supported
@item MPEG-4 Audio Lossless Coding (ALS)  @tab     @tab  X
-@item MobiClip FastAudio     @tab     @tab  X
@item Musepack SV7           @tab     @tab  X
@item Musepack SV8           @tab     @tab  X
@item Nellymoser Asao        @tab  X  @tab  X
@item On2 AVC (Audio for Video Codec) @tab     @tab  X
@item Opus                   @tab  E  @tab  X
    @tab encoding supported through external library libopus
-@item OSQ (Original Sound Quality)  @tab     @tab  X
@item PCM A-law              @tab  X  @tab  X
@item PCM mu-law             @tab  X  @tab  X
@item PCM Archimedes VIDC    @tab  X  @tab  X
@@ -1328,7 +1276,6 @@ 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 SGA                @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
@@ -1341,7 +1288,6 @@ following image formats are supported:
    @tab Real low bitrate AC-3 codec
@item RealAudio Lossless     @tab     @tab  X
@item RealAudio SIPR / ACELP.NET @tab     @tab  X
-@item RK Audio (RKA)         @tab     @tab  X
@item SBC (low-complexity subband codec) @tab  X  @tab  X
    @tab Used in Bluetooth A2DP
@item Shorten                @tab     @tab  X
@@ -1353,7 +1299,7 @@ following image formats are supported:
    @tab experimental codec
@item Sonic lossless         @tab  X  @tab  X
    @tab experimental codec
-@item Speex                  @tab  E  @tab  EX
+@item Speex                  @tab  E  @tab  E
    @tab supported through external library libspeex
@item TAK (Tom's lossless Audio Kompressor)  @tab     @tab  X
@item True Audio (TTA)       @tab  X  @tab  X
@@ -1362,11 +1308,9 @@ following image formats are supported:
@item TwinVQ (VQF flavor)    @tab     @tab  X
@item VIMA                   @tab     @tab  X
    @tab Used in LucasArts SMUSH animations.
-@item ViewQuest VQC          @tab     @tab  X
@item Vorbis                 @tab  E  @tab  X
    @tab A native but very primitive encoder exists.
@item Voxware MetaSound      @tab     @tab  X
-@item Waveform Archiver      @tab     @tab  X
@item WavPack                @tab  X  @tab  X
@item Westwood Audio (SND1)  @tab     @tab  X
@item Windows Media Audio 1  @tab  X  @tab  X
--- a/doc/git-howto.texi
+++ b/doc/git-howto.texi
@@ -143,7 +143,7 @@ git log <filename(s)>
@end example

 You may also use the graphical tools like @command{gitview} or @command{gitk}
-or the web interface available at @url{http://source.ffmpeg.org/}.
+or the web interface available at @url{https://git.ffmpeg.org/ffmpeg.git}.

@section Checking source tree status

@@ -224,46 +224,16 @@ git config --global core.editor
 or set by one of the following environment variables:
@var{GIT_EDITOR}, @var{VISUAL} or @var{EDITOR}.

-@section Writing a commit message
+Log messages should be concise but descriptive. Explain why you made a change,
+what you did will be obvious from the changes themselves most of the time.
+Saying just "bug fix" or "10l" is bad. Remember that people of varying skill
+levels look at and educate themselves while reading through your code. Don't
+include filenames in log messages, Git provides that information.

-Log messages should be concise but descriptive.
-
-The first line must contain the context, a colon and a very short
-summary of what the commit does. Details can be added, if necessary,
-separated by an empty line. These details should not exceed 60-72 characters
-per line, except when containing code.
-
-Example of a good commit message:
-
-@example
-avcodec/cbs: add a helper to read extradata within packet side data
-
-Using ff_cbs_read() on the raw buffer will not parse it as extradata,
-resulting in parsing errors for example when handling ISOBMFF avcC.
-This helper works around that.
-@end example
-
-@example
-ptr might be NULL
-@end example
-
-If the summary on the first line is not enough, in the body of the message,
-explain why you made a change, what you did will be obvious from the changes
-themselves most of the time. Saying just "bug fix" or "10l" is bad. Remember
-that people of varying skill levels look at and educate themselves while
-reading through your code. Don't include filenames in log messages except in
-the context, Git provides that information.
-
-If the commit fixes a registered issue, state it in a separate line of the
-body: @code{Fix Trac ticket #42.}
-
-The first line will be used to name
+Possibly make the commit message have a terse, descriptive first line, an
+empty line and then a full description. The first line will be used to name
 the patch by @command{git format-patch}.

-Common mistakes for the first line, as seen in @command{git log --oneline}
-include: missing context at the beginning; description of what the code did
-before the patch; line too long or wrapped to the second line.
-
@section Preparing a patchset

@example
--- a/doc/indevs.texi
+++ b/doc/indevs.texi
@@ -344,23 +344,9 @@ Defines number of audio channels to capture. Must be @samp{2}, @samp{8} or @samp
 Defaults to @samp{2}.

@item duplex_mode
-Sets the decklink device duplex/profile mode. Must be @samp{unset}, @samp{half}, @samp{full},
-@samp{one_sub_device_full}, @samp{one_sub_device_half}, @samp{two_sub_device_full},
-@samp{four_sub_device_half}
+Sets the decklink device duplex mode. Must be @samp{unset}, @samp{half} or @samp{full}.
 Defaults to @samp{unset}.

-Note: DeckLink SDK 11.0 have replaced the duplex property by a profile property.
-For the DeckLink Duo 2 and DeckLink Quad 2, a profile is shared between any 2
-sub-devices that utilize the same connectors. For the DeckLink 8K Pro, a profile
-is shared between all 4 sub-devices. So DeckLink 8K Pro support four profiles.
-
-Valid profile modes for DeckLink 8K Pro(with DeckLink SDK >= 11.0):
-@samp{one_sub_device_full}, @samp{one_sub_device_half}, @samp{two_sub_device_full},
-@samp{four_sub_device_half}
-
-Valid profile modes for DeckLink Quad 2 and DeckLink Duo 2:
-@samp{half}, @samp{full}
-
@item timecode_format
 Timecode type to include in the frame and video stream metadata. Must be
@samp{none}, @samp{rp188vitc}, @samp{rp188vitc2}, @samp{rp188ltc},
@@ -625,12 +611,6 @@ Save the currently used video capture filter device and its
 parameters (if the filter supports it) to a file.
 If a file with the same name exists it will be overwritten.

-@item use_video_device_timestamps
-If set to @option{false}, the timestamp for video frames will be
-derived from the wallclock instead of the timestamp provided by
-the capture device. This allows working around devices that
-provide unreliable timestamps.
-
@end table

@subsection Examples
@@ -991,8 +971,9 @@ This input device reads data from the open output pads of a libavfilter
 filtergraph.

 For each filtergraph open output, the input device will create a
-corresponding stream which is mapped to the generated output.
-The filtergraph is specified through the option @option{graph}.
+corresponding stream which is mapped to the generated output. Currently
+only video data is supported. The filtergraph is specified through the
+option @option{graph}.

@subsection Options

@@ -1288,11 +1269,11 @@ Specify the samplerate in Hz, by default 48kHz is used.
 Specify the channels in use, by default 2 (stereo) is set.

@item frame_size
-This option does nothing and is deprecated.
+Specify the number of bytes per frame, by default it is set to 1024.

@item fragment_size
-Specify the size in bytes of the minimal buffering fragment in PulseAudio, it
-will affect the audio latency. By default it is set to 50 ms amount of data.
+Specify the minimal buffering fragment in PulseAudio, it will affect the
+audio latency. By default it is unset.

@item wallclock
 Set the initial PTS using the current time. Default is 1.
--- a/doc/mailing-list-faq.texi
+++ b/doc/mailing-list-faq.texi
@@ -116,7 +116,7 @@ or is abusive towards others).
@section How long does it take for my message in the moderation queue to be approved?

 The queue is not checked on a regular basis. You can ask on the
-@t{#ffmpeg-devel} IRC channel on Libera Chat for someone to approve your message.
+@t{#ffmpeg-devel} IRC channel on Freenode for someone to approve your message.

@anchor{How do I delete my message in the moderation queue?}
@section How do I delete my message in the moderation queue?
@@ -155,7 +155,7 @@ Perform a site search using your favorite search engine. Example:

@section Is there an alternative to the mailing list?

-You can ask for help in the official @t{#ffmpeg} IRC channel on Libera Chat.
+You can ask for help in the official @t{#ffmpeg} IRC channel on Freenode.

 Some users prefer the third-party @url{http://www.ffmpeg-archive.org/, Nabble}
 interface which presents the mailing lists in a typical forum layout.
@@ -344,7 +344,7 @@ recommended.
 Avoid sending the same message to multiple mailing lists.

@item
-Please follow our @url{https://ffmpeg.org/community.html#Code-of-conduct, Code of Conduct}.
+Please follow our @url{https://ffmpeg.org/developer.html#Code-of-conduct, Code of Conduct}.
@end itemize

@chapter Help
--- a/doc/mips.txt
+++ b/doc/mips.txt
@@ -48,6 +48,11 @@ Files that have MIPS copyright notice in them:
      float_dsp_mips.c
      libm_mips.h
      softfloat_tables.h
+* libavcodec/
+      fft_fixed_32.c
+      fft_init_table.c
+      fft_table.h
+      mdct_fixed_32.c
 * libavcodec/mips/
      aacdec_fixed.c
      aacsbr_fixed.c
@@ -65,6 +70,9 @@ Files that have MIPS copyright notice in them:
      compute_antialias_float.h
      lsp_mips.h
      dsputil_mips.c
+      fft_mips.c
+      fft_table.h
+      fft_init_table.c
      fmtconvert_mips.c
      iirfilter_mips.c
      mpegaudiodsp_mips_fixed.c
--- a/doc/multithreading.txt
+++ b/doc/multithreading.txt
@@ -55,7 +55,8 @@ 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
-FFCodec.caps_internal and use ff_thread_get_buffer() to allocate frames.
+AVCodec.caps_internal and use ff_thread_get_buffer() to allocate frames. The
+frames must then be freed with ff_thread_release_buffer().
 Otherwise decode directly into the user-supplied frames.

 Call ff_thread_report_progress() after some part of the current picture has decoded.
--- a/doc/muxers.texi
+++ b/doc/muxers.texi
@@ -19,33 +19,6 @@ enabled demuxers and muxers.

 A description of some of the currently available muxers follows.

-@anchor{a64}
-@section a64
-
-A64 muxer for Commodore 64 video. Accepts a single @code{a64_multi} or @code{a64_multi5} codec video stream.
-
-@anchor{adts}
-@section adts
-
-Audio Data Transport Stream muxer. It accepts a single AAC stream.
-
-@subsection Options
-
-It accepts the following options:
-
-@table @option
-
-@item write_id3v2 @var{bool}
-Enable to write ID3v2.4 tags at the start of the stream. Default is disabled.
-
-@item write_apetag @var{bool}
-Enable to write APE tags at the end of the stream. Default is disabled.
-
-@item write_mpeg2 @var{bool}
-Enable to set MPEG version bit in the ADTS frame header to 1 which indicates MPEG-2. Default is 0, which indicates MPEG-4.
-
-@end table
-
@anchor{aiff}
@section aiff

@@ -65,37 +38,6 @@ ID3v2.3 and ID3v2.4) are supported. The default is version 4.

@end table

-@anchor{alp}
-@section alp
-
-Muxer for audio of High Voltage Software's Lego Racers game. It accepts a single ADPCM_IMA_ALP stream
-with no more than 2 channels nor a sample rate greater than 44100 Hz.
-
-Extensions: tun, pcm
-
-@subsection Options
-
-It accepts the following options:
-
-@table @option
-
-@item type @var{type}
-Set file type.
-
-@table @samp
-@item tun
-Set file type as music. Must have a sample rate of 22050 Hz.
-
-@item pcm
-Set file type as sfx.
-
-@item auto
-Set file type as per output file extension. @code{.pcm} results in type @code{pcm} else type @code{tun} is set. @var{(default)}
-
-@end table
-
-@end table
-
@anchor{asf}
@section asf

@@ -231,6 +173,37 @@ and the input video converted to MPEG-2 video, use the command:
 ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f crc -
@end example

+@section flv
+
+Adobe Flash Video Format muxer.
+
+This muxer accepts the following options:
+
+@table @option
+
+@item flvflags @var{flags}
+Possible values:
+
+@table @samp
+
+@item aac_seq_header_detect
+Place AAC sequence header based on audio stream data.
+
+@item no_sequence_end
+Disable sequence end tag.
+
+@item no_metadata
+Disable metadata tag.
+
+@item no_duration_filesize
+Disable duration and filesize in metadata when they are equal to zero
+at the end of stream. (Be used to non-seekable living stream).
+
+@item add_keyframe_index
+Used to facilitate seeking; particularly for HTTP pseudo streaming.
+@end table
+@end table
+
@anchor{dash}
@section dash

@@ -264,6 +237,8 @@ ffmpeg -re -i <input> -map 0 -map 0 -c:a libfdk_aac -c:v libx264 \
@end example

@table @option
+@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}
 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
@@ -362,13 +337,12 @@ Ignore IO errors during open and write. Useful for long-duration runs with netwo

@item lhls @var{lhls}
 Enable Low-latency HLS(LHLS). Adds #EXT-X-PREFETCH tag with current segment's URI.
-hls.js player folks are trying to standardize an 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 tries to comply with the above open spec.
-It enables @var{streaming} and @var{hls_playlist} options automatically.
+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.

-Note: This is not Apple's version LHLS. See @url{https://datatracker.ietf.org/doc/html/draft-pantos-hls-rfc8216bis}
-
@item ldash @var{ldash}
 Enable Low-latency Dash by constraining the presence and values of some elements.

@@ -406,137 +380,6 @@ adjusting playback latency and buffer occupancy during normal playback by client

@end table

-@anchor{fifo}
-@section fifo
-
-The fifo pseudo-muxer allows the separation of encoding and muxing by using
-first-in-first-out queue and running the actual muxer in a separate thread. This
-is especially useful in combination with the @ref{tee} muxer and can be used to
-send data to several destinations with different reliability/writing speed/latency.
-
-API users should be aware that callback functions (interrupt_callback,
-io_open and io_close) used within its AVFormatContext must be thread-safe.
-
-The behavior of the fifo muxer if the queue fills up or if the output fails is
-selectable,
-
-@itemize @bullet
-
-@item
-output can be transparently restarted with configurable delay between retries
-based on real time or time of the processed stream.
-
-@item
-encoding can be blocked during temporary failure, or continue transparently
-dropping packets in case fifo queue fills up.
-
-@end itemize
-
-@table @option
-
-@item fifo_format
-Specify the format name. Useful if it cannot be guessed from the
-output name suffix.
-
-@item queue_size
-Specify size of the queue (number of packets). Default value is 60.
-
-@item format_opts
-Specify format options for the underlying muxer. Muxer options can be specified
-as a list of @var{key}=@var{value} pairs separated by ':'.
-
-@item drop_pkts_on_overflow @var{bool}
-If set to 1 (true), in case the fifo queue fills up, packets will be dropped
-rather than blocking the encoder. This makes it possible to continue streaming without
-delaying the input, at the cost of omitting part of the stream. By default
-this option is set to 0 (false), so in such cases the encoder will be blocked
-until the muxer processes some of the packets and none of them is lost.
-
-@item attempt_recovery @var{bool}
-If failure occurs, attempt to recover the output. This is especially useful
-when used with network output, since it makes it possible to restart streaming transparently.
-By default this option is set to 0 (false).
-
-@item max_recovery_attempts
-Sets maximum number of successive unsuccessful recovery attempts after which
-the output fails permanently. By default this option is set to 0 (unlimited).
-
-@item recovery_wait_time @var{duration}
-Waiting time before the next recovery attempt after previous unsuccessful
-recovery attempt. Default value is 5 seconds.
-
-@item recovery_wait_streamtime @var{bool}
-If set to 0 (false), the real time is used when waiting for the recovery
-attempt (i.e. the recovery will be attempted after at least
-recovery_wait_time seconds).
-If set to 1 (true), the time of the processed stream is taken into account
-instead (i.e. the recovery will be attempted after at least @var{recovery_wait_time}
-seconds of the stream is omitted).
-By default, this option is set to 0 (false).
-
-@item recover_any_error @var{bool}
-If set to 1 (true), recovery will be attempted regardless of type of the error
-causing the failure. By default this option is set to 0 (false) and in case of
-certain (usually permanent) errors the recovery is not attempted even when
-@var{attempt_recovery} is set to 1.
-
-@item restart_with_keyframe @var{bool}
-Specify whether to wait for the keyframe after recovering from
-queue overflow or failure. This option is set to 0 (false) by default.
-
-@item timeshift @var{duration}
-Buffer the specified amount of packets and delay writing the output. Note that
-@var{queue_size} must be big enough to store the packets for timeshift. At the
-end of the input the fifo buffer is flushed at realtime speed.
-
-@end table
-
-@subsection Examples
-
-@itemize
-
-@item
-Stream something to rtmp server, continue processing the stream at real-time
-rate even in case of temporary failure (network outage) and attempt to recover
-streaming every second indefinitely.
-@example
-ffmpeg -re -i ... -c:v libx264 -c:a aac -f fifo -fifo_format flv -map 0:v -map 0:a
-  -drop_pkts_on_overflow 1 -attempt_recovery 1 -recovery_wait_time 1 rtmp://example.com/live/stream_name
-@end example
-
-@end itemize
-
-@section flv
-
-Adobe Flash Video Format muxer.
-
-This muxer accepts the following options:
-
-@table @option
-
-@item flvflags @var{flags}
-Possible values:
-
-@table @samp
-
-@item aac_seq_header_detect
-Place AAC sequence header based on audio stream data.
-
-@item no_sequence_end
-Disable sequence end tag.
-
-@item no_metadata
-Disable metadata tag.
-
-@item no_duration_filesize
-Disable duration and filesize in metadata when they are equal to zero
-at the end of stream. (Be used to non-seekable living stream).
-
-@item add_keyframe_index
-Used to facilitate seeking; particularly for HTTP pseudo streaming.
-@end table
-@end table
-
@anchor{framecrc}
@section framecrc

@@ -795,6 +638,20 @@ deletes them. Increase this to allow continue clients to download segments which
 were recently referenced in the playlist. Default value is 1, meaning segments older than
@code{hls_list_size+1} will be deleted.

+@item hls_ts_options @var{options_list}
+Set output format options using a :-separated list of key=value
+parameters. Values containing @code{:} special characters must be
+escaped.
+
+@item hls_wrap @var{wrap}
+This is a deprecated option, you can use @code{hls_list_size}
+and @code{hls_flags delete_segments} instead it
+
+This option is useful to avoid to fill the disk with many segment
+files, and limits the maximum number of segment files written to disk
+to @var{wrap}.
+
+
@item hls_start_number_source
 Start the playlist sequence number (@code{#EXT-X-MEDIA-SEQUENCE}) according to the specified source.
 Unless @code{hls_flags single_file} is set, it also specifies source of starting sequence numbers of
@@ -880,6 +737,9 @@ This example will produce the playlists segment file sets:
@file{vs0/file_000.ts}, @file{vs0/file_001.ts}, @file{vs0/file_002.ts}, etc. and
@file{vs1/file_000.ts}, @file{vs1/file_001.ts}, @file{vs1/file_002.ts}, etc.

+@item use_localtime
+Same as strftime option, will be deprecated.
+
@item strftime
 Use strftime() on @var{filename} to expand the segment filename with localtime.
 The segment number is also available in this mode, but to use it, you need to specify second_level_segment_index
@@ -897,6 +757,9 @@ ffmpeg -i in.nut -strftime 1 -hls_flags second_level_segment_index -hls_segment_
 This example will produce the playlist, @file{out.m3u8}, and segment files:
@file{file-20160215-0001.ts}, @file{file-20160215-0002.ts}, etc.

+@item use_localtime_mkdir
+Same as strftime_mkdir option, will be deprecated .
+
@item strftime_mkdir
 Used together with -strftime_mkdir, it will create all subdirectories which
 is expanded in @var{filename}.
@@ -914,10 +777,6 @@ This example will create a directory hierarchy 2016/02/15 (if any of them do not
 produce the playlist, @file{out.m3u8}, and segment files:
@file{2016/02/15/file-20160215-1455569023.ts}, @file{2016/02/15/file-20160215-1455569024.ts}, etc.

-@item hls_segment_options @var{options_list}
-Set output format options using a :-separated list of key=value
-parameters. Values containing @code{:} special characters must be
-escaped.

@item hls_key_info_file @var{key_info_file}
 Use the information in @var{key_info_file} for segment encryption. The first
@@ -1054,8 +913,6 @@ and remove the @code{#EXT-X-ENDLIST} from the old segment list.
@item round_durations
 Round the duration info in the playlist file segment info to integer
 values, instead of using floating point.
-If there are no other features requiring higher HLS versions be used,
-then this will allow ffmpeg to output a HLS version 2 m3u8.

@item discont_start
 Add the @code{#EXT-X-DISCONTINUITY} tag to the playlist, before the
@@ -1325,7 +1182,7 @@ 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
+@item -ignore_io_errors
 Ignore IO errors during open, write and delete. Useful for long-duration runs with network output.

@item headers
@@ -1421,10 +1278,6 @@ overwritten with new images. Default value is 0.
 If set to 1, expand the filename with date and time information from
@code{strftime()}. Default value is 0.

-@item atomic_writing
-Write output to a temporary file, which is renamed to target filename once
-writing is completed. Default is disabled.
-
@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.
@@ -1468,7 +1321,7 @@ ffmpeg -f v4l2 -r 1 -i /dev/video0 -f image2 -strftime 1 "%Y-%m-%d_%H-%M-%S.jpg"

 You can set the file name with current frame's PTS:
@example
-ffmpeg -f v4l2 -r 1 -i /dev/video0 -copyts -f image2 -frame_pts true %d.jpg
+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
@@ -1563,27 +1416,18 @@ 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 cues_to_front
-If set, the muxer will write the index at the beginning of the file
-by shifting the main data if necessary. This can be combined with
-reserve_index_space in which case the data is only shifted if
-the initially reserved space turns out to be insufficient.
-
-This option is ignored if the output is unseekable.
-
@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{passthrough}.
+is @samp{infer}.
@table @samp
@item infer
-Every track with disposition default will have the FlagDefault set.
-Additionally, for each type of track (audio, video or subtitle), if no track
-with disposition default of this type exists, then the first track of this type
-will be marked as default (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.
+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.
@@ -1630,10 +1474,8 @@ MOV/MP4/ISMV (Smooth Streaming) muxer.
 The mov/mp4/ismv muxer supports fragmentation. Normally, a MOV/MP4
 file has all the metadata about all packets stored in one location
 (written at the end of the file, it can be moved to the start for
-better playback by adding @code{+faststart} to the @code{-movflags}, or
-using the @command{qt-faststart} tool).
-
-A fragmented
+better playback by adding @var{faststart} to the @var{movflags}, or
+using the @command{qt-faststart} tool). A fragmented
 file consists of a number of fragments, where packets and metadata
 about these packets are stored together. Writing a fragmented
 file has the advantage that the file is decodable even if the
@@ -1643,34 +1485,40 @@ very long files (since writing normal MOV/MP4 files stores info about
 every single packet in memory until the file is closed). The downside
 is that it is less compatible with other applications.

-Fragmentation is enabled by setting one of the options that define
-how to cut the file into fragments: @code{-frag_duration}, @code{-frag_size},
-@code{-min_frag_duration}, @code{-movflags +frag_keyframe} and
-@code{-movflags +frag_custom}. If more than one condition is specified,
-fragments are cut when one of the specified conditions is fulfilled. The
-exception to this is @code{-min_frag_duration}, which has to be fulfilled for
-any of the other conditions to apply.
-
@subsection Options

+Fragmentation is enabled by setting one of the AVOptions that define
+how to cut the file into fragments:
+
@table @option
-@item frag_duration @var{duration}
-Create fragments that are @var{duration} microseconds long.
-@item frag_size @var{size}
-Create fragments that contain up to @var{size} bytes of payload data.
-@item min_frag_duration @var{duration}
-Don't create fragments that are shorter than @var{duration} microseconds long.
-@item movflags @var{flags}
-Set various muxing switches. The following flags can be used:
-@table @samp
-@item frag_keyframe
+@item -moov_size @var{bytes}
+Reserves space for the moov atom at the beginning of the file instead of placing the
+moov atom at the end. If the space reserved is insufficient, muxing will fail.
+@item -movflags frag_keyframe
 Start a new fragment at each video keyframe.
-@item frag_custom
+@item -frag_duration @var{duration}
+Create fragments that are @var{duration} microseconds long.
+@item -frag_size @var{size}
+Create fragments that contain up to @var{size} bytes of payload data.
+@item -movflags frag_custom
 Allow the caller to manually choose when to cut fragments, by
 calling @code{av_write_frame(ctx, NULL)} to write a fragment with
 the packets written so far. (This is only useful with other
 applications integrating libavformat, not from @command{ffmpeg}.)
-@item empty_moov
+@item -min_frag_duration @var{duration}
+Don't create fragments that are shorter than @var{duration} microseconds long.
+@end table
+
+If more than one condition is specified, fragments are cut when
+one of the specified conditions is fulfilled. The exception to this is
+@code{-min_frag_duration}, which has to be fulfilled for any of the other
+conditions to apply.
+
+Additionally, the way the output file is written can be adjusted
+through a few other options:
+
+@table @option
+@item -movflags empty_moov
 Write an initial moov atom directly at the start of the file, without
 describing any samples in it. Generally, an mdat/moov pair is written
 at the start of the file, as a normal MOV/MP4 file, containing only
@@ -1679,40 +1527,43 @@ mdat atom, and the moov atom only describes the tracks but has
 a zero duration.

 This option is implicitly set when writing ismv (Smooth Streaming) files.
-@item separate_moof
+@item -movflags separate_moof
 Write a separate moof (movie fragment) atom for each track. Normally,
 packets for all tracks are written in a moof atom (which is slightly
 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 skip_sidx
+@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 faststart
+@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
 as fragmented output, thus it is not enabled by default.
-@item rtphint
+@item -movflags rtphint
 Add RTP hinting tracks to the output file.
-@item disable_chpl
+@item -movflags disable_chpl
 Disable Nero chapter markers (chpl atom).  Normally, both Nero chapters
 and a QuickTime chapter track are written to the file. With this option
 set, only the QuickTime chapter track will be written. Nero chapters can
 cause failures when the file is reprocessed with certain tagging programs, like
 mp3Tag 2.61a and iTunes 11.3, most likely other versions are affected as well.
-@item omit_tfhd_offset
+@item -movflags omit_tfhd_offset
 Do not write any absolute base_data_offset in tfhd atoms. This avoids
 tying fragments to absolute byte positions in the file/streams.
-@item default_base_moof
+@item -movflags default_base_moof
 Similarly to the omit_tfhd_offset, this flag avoids writing the
 absolute base_data_offset field in tfhd atoms, but does so by using
 the new default-base-is-moof flag instead. This flag is new from
 14496-12:2012. This may make the fragments easier to parse in certain
 circumstances (avoiding basing track fragment location calculations
 on the implicit end of the previous track fragment).
-@item negative_cts_offsets
+@item -write_tmcd
+Specify @code{on} to force writing a timecode track, @code{off} to disable it
+and @code{auto} to write a timecode track only for mov and mp4 output (default).
+@item -movflags negative_cts_offsets
 Enables utilization of version 1 of the CTTS box, in which the CTS offsets can
 be negative. This enables the initial sample to have DTS/CTS of zero, and
 reduces the need for edit lists for some cases such as video tracks with
@@ -1720,24 +1571,7 @@ B-frames. Additionally, eases conformance with the DASH-IF interoperability
 guidelines.

 This option is implicitly set when writing ismv (Smooth Streaming) files.
-@end table
-
-@item moov_size @var{bytes}
-Reserves space for the moov atom at the beginning of the file instead of placing the
-moov atom at the end. If the space reserved is insufficient, muxing will fail.
-
-@item write_tmcd
-Specify @code{on} to force writing a timecode track, @code{off} to disable it
-and @code{auto} to write a timecode track only for mov and mp4 output (default).
-
-@item write_btrt @var{bool}
-Force or disable writing bitrate box inside stsd box of a track.
-The box contains decoding buffer size (in bytes), maximum bitrate and
-average bitrate for the track. The box will be skipped if none of these values
-can be computed.
-Default is @code{-1} or @code{auto}, which will write the box only in MP4 mode.
-
-@item write_prft
+@item -write_prft
 Write producer time reference box (PRFT) with a specified time source for the
 NTP field in the PRFT box. Set value as @samp{wallclock} to specify timesource
 as wallclock time and @samp{pts} to specify timesource as input packets' PTS
@@ -1747,19 +1581,6 @@ Setting value to @samp{pts} is applicable only for a live encoding use case,
 where PTS values are set as as wallclock time at the source. For example, an
 encoding use case with decklink capture source where @option{video_pts} and
@option{audio_pts} are set to @samp{abs_wallclock}.
-
-@item empty_hdlr_name @var{bool}
-Enable to skip writing the name inside a @code{hdlr} box.
-Default is @code{false}.
-
-@item movie_timescale @var{scale}
-Set the timescale written in the movie header box (@code{mvhd}).
-Range is 1 to INT_MAX. Default is 1000.
-
-@item video_track_timescale @var{scale}
-Set the timescale used for video tracks. Range is 0 to INT_MAX.
-If set to @code{0}, the timescale is automatically set based on
-the native stream time base. Default is 0.
@end table

@subsection Example
@@ -1909,10 +1730,6 @@ Reemit PAT and PMT at each video frame.
 Conform to System B (DVB) instead of System A (ATSC).
@item initial_discontinuity
 Mark the initial packet of each stream as discontinuity.
-@item nit
-Emit NIT table.
-@item omit_rai
-Disable writing of random access indicator.
@end table

@item mpegts_copyts @var{boolean}
@@ -1934,11 +1751,8 @@ Maximum time in seconds between PAT/PMT tables. Default is @code{0.1}.
@item sdt_period @var{duration}
 Maximum time in seconds between SDT tables. Default is @code{0.5}.

-@item nit_period @var{duration}
-Maximum time in seconds between NIT tables. Default is @code{0.5}.
-
@item tables_version @var{integer}
-Set PAT, PMT, SDT and NIT version (default @code{0}, valid values are from 0 to 31, inclusively).
+Set PAT, PMT and SDT version (default @code{0}, valid values are from 0 to 31, inclusively).
 This option allows updating stream structure so that standard consumer may
 detect the change. To do so, reopen output @code{AVFormatContext} (in case of API
 usage) or restart @command{ffmpeg} instance, cyclically changing
@@ -2050,188 +1864,6 @@ ogg files can be safely chained.

@end table

-@anchor{raw muxers}
-@section raw muxers
-
-Raw muxers accept a single stream matching the designated codec. They do not store timestamps or metadata.
-The recognized extension is the same as the muxer name unless indicated otherwise.
-
-@subsection ac3
-
-Dolby Digital, also known as AC-3, audio.
-
-@subsection adx
-
-CRI Middleware ADX audio.
-
-This muxer will write out the total sample count near the start of the first packet
-when the output is seekable and the count can be stored in 32 bits.
-
-@subsection aptx
-
-aptX (Audio Processing Technology for Bluetooth) audio.
-
-@subsection aptx_hd
-
-aptX HD (Audio Processing Technology for Bluetooth) audio.
-
-Extensions: aptxhd
-
-@subsection avs2
-
-AVS2-P2/IEEE1857.4 video.
-
-Extensions: avs, avs2
-
-@subsection cavsvideo
-
-Chinese AVS (Audio Video Standard) video.
-
-Extensions: cavs
-
-@subsection codec2raw
-
-Codec 2 audio.
-
-No extension is registered so format name has to be supplied e.g. with the ffmpeg CLI tool @code{-f codec2raw}.
-
-@subsection data
-
-Data muxer accepts a single stream with any codec of any type.
-The input stream has to be selected using the @code{-map} option with the ffmpeg CLI tool.
-
-No extension is registered so format name has to be supplied e.g. with the ffmpeg CLI tool @code{-f data}.
-
-@subsection dirac
-
-BBC Dirac video. The Dirac Pro codec is a subset and is standardized as SMPTE VC-2.
-
-Extensions: drc, vc2
-
-@subsection dnxhd
-
-Avid DNxHD video. It is standardized as SMPTE VC-3. Accepts DNxHR streams.
-
-Extensions: dnxhd, dnxhr
-
-@subsection dts
-
-DTS Coherent Acoustics (DCA) audio.
-
-@subsection eac3
-
-Dolby Digital Plus, also known as Enhanced AC-3, audio.
-
-@subsection evc
-
-MPEG-5 Essential Video Coding (EVC) / EVC / MPEG-5 Part 1 EVC video.
-
-Extensions: evc
-
-@subsection g722
-
-ITU-T G.722 audio.
-
-@subsection g723_1
-
-ITU-T G.723.1 audio.
-
-Extensions: tco, rco
-
-@subsection g726
-
-ITU-T G.726 big-endian ("left-justified") audio.
-
-No extension is registered so format name has to be supplied e.g. with the ffmpeg CLI tool @code{-f g726}.
-
-@subsection g726le
-
-ITU-T G.726 little-endian ("right-justified") audio.
-
-No extension is registered so format name has to be supplied e.g. with the ffmpeg CLI tool @code{-f g726le}.
-
-@subsection gsm
-
-Global System for Mobile Communications audio.
-
-@subsection h261
-
-ITU-T H.261 video.
-
-@subsection h263
-
-ITU-T H.263 / H.263-1996, H.263+ / H.263-1998 / H.263 version 2 video.
-
-@subsection h264
-
-ITU-T H.264 / MPEG-4 Part 10 AVC video. Bitstream shall be converted to Annex B syntax if it's in length-prefixed mode.
-
-Extensions: h264, 264
-
-@subsection hevc
-
-ITU-T H.265 / MPEG-H Part 2 HEVC video. Bitstream shall be converted to Annex B syntax if it's in length-prefixed mode.
-
-Extensions: hevc, h265, 265
-
-@subsection m4v
-
-MPEG-4 Part 2 video.
-
-@subsection mjpeg
-
-Motion JPEG video.
-
-Extensions: mjpg, mjpeg
-
-@subsection mlp
-
-Meridian Lossless Packing, also known as Packed PCM, audio.
-
-@subsection mp2
-
-MPEG-1 Audio Layer II audio.
-
-Extensions: mp2, m2a, mpa
-
-@subsection mpeg1video
-
-MPEG-1 Part 2 video.
-
-Extensions: mpg, mpeg, m1v
-
-@subsection mpeg2video
-
-ITU-T H.262 / MPEG-2 Part 2 video.
-
-Extensions: m2v
-
-@subsection obu
-
-AV1 low overhead Open Bitstream Units muxer. Temporal delimiter OBUs will be inserted in all temporal units of the stream.
-
-@subsection rawvideo
-
-Raw uncompressed video.
-
-Extensions: yuv, rgb
-
-@subsection sbc
-
-Bluetooth SIG low-complexity subband codec audio.
-
-Extensions: sbc, msbc
-
-@subsection truehd
-
-Dolby TrueHD audio.
-
-Extensions: thd
-
-@subsection vc1
-
-SMPTE 421M / VC-1 video.
-
@anchor{segment}
@section segment, stream_segment, ssegment

@@ -2371,11 +2003,6 @@ Note that splitting may not be accurate, unless you force the
 reference stream key-frames at the given time. See the introductory
 notice and the examples below.

-@item min_seg_duration @var{time}
-Set minimum segment duration to @var{time}, the value must be a duration
-specification. This prevents the muxer ending segments at a duration below
-this value. Only effective with @code{segment_time}. Default value is "0".
-
@item segment_atclocktime @var{1|0}
 If set to "1" split at regular clock time intervals starting from 00:00
 o'clock. The @var{time} value specified in @option{segment_time} is
@@ -2605,6 +2232,106 @@ ffmpeg -i INPUT -f streamhash -hash md5 -

 See also the @ref{hash} and @ref{framehash} muxers.

+@anchor{fifo}
+@section fifo
+
+The fifo pseudo-muxer allows the separation of encoding and muxing by using
+first-in-first-out queue and running the actual muxer in a separate thread. This
+is especially useful in combination with the @ref{tee} muxer and can be used to
+send data to several destinations with different reliability/writing speed/latency.
+
+API users should be aware that callback functions (interrupt_callback,
+io_open and io_close) used within its AVFormatContext must be thread-safe.
+
+The behavior of the fifo muxer if the queue fills up or if the output fails is
+selectable,
+
+@itemize @bullet
+
+@item
+output can be transparently restarted with configurable delay between retries
+based on real time or time of the processed stream.
+
+@item
+encoding can be blocked during temporary failure, or continue transparently
+dropping packets in case fifo queue fills up.
+
+@end itemize
+
+@table @option
+
+@item fifo_format
+Specify the format name. Useful if it cannot be guessed from the
+output name suffix.
+
+@item queue_size
+Specify size of the queue (number of packets). Default value is 60.
+
+@item format_opts
+Specify format options for the underlying muxer. Muxer options can be specified
+as a list of @var{key}=@var{value} pairs separated by ':'.
+
+@item drop_pkts_on_overflow @var{bool}
+If set to 1 (true), in case the fifo queue fills up, packets will be dropped
+rather than blocking the encoder. This makes it possible to continue streaming without
+delaying the input, at the cost of omitting part of the stream. By default
+this option is set to 0 (false), so in such cases the encoder will be blocked
+until the muxer processes some of the packets and none of them is lost.
+
+@item attempt_recovery @var{bool}
+If failure occurs, attempt to recover the output. This is especially useful
+when used with network output, since it makes it possible to restart streaming transparently.
+By default this option is set to 0 (false).
+
+@item max_recovery_attempts
+Sets maximum number of successive unsuccessful recovery attempts after which
+the output fails permanently. By default this option is set to 0 (unlimited).
+
+@item recovery_wait_time @var{duration}
+Waiting time before the next recovery attempt after previous unsuccessful
+recovery attempt. Default value is 5 seconds.
+
+@item recovery_wait_streamtime @var{bool}
+If set to 0 (false), the real time is used when waiting for the recovery
+attempt (i.e. the recovery will be attempted after at least
+recovery_wait_time seconds).
+If set to 1 (true), the time of the processed stream is taken into account
+instead (i.e. the recovery will be attempted after at least @var{recovery_wait_time}
+seconds of the stream is omitted).
+By default, this option is set to 0 (false).
+
+@item recover_any_error @var{bool}
+If set to 1 (true), recovery will be attempted regardless of type of the error
+causing the failure. By default this option is set to 0 (false) and in case of
+certain (usually permanent) errors the recovery is not attempted even when
+@var{attempt_recovery} is set to 1.
+
+@item restart_with_keyframe @var{bool}
+Specify whether to wait for the keyframe after recovering from
+queue overflow or failure. This option is set to 0 (false) by default.
+
+@item timeshift @var{duration}
+Buffer the specified amount of packets and delay writing the output. Note that
+@var{queue_size} must be big enough to store the packets for timeshift. At the
+end of the input the fifo buffer is flushed at realtime speed.
+
+@end table
+
+@subsection Examples
+
+@itemize
+
+@item
+Stream something to rtmp server, continue processing the stream at real-time
+rate even in case of temporary failure (network outage) and attempt to recover
+streaming every second indefinitely.
+@example
+ffmpeg -re -i ... -c:v libx264 -c:a aac -f fifo -fifo_format flv -map 0:v -map 0:a
+  -drop_pkts_on_overflow 1 -attempt_recovery 1 -recovery_wait_time 1 rtmp://example.com/live/stream_name
+@end example
+
+@end itemize
+
@anchor{tee}
@section tee

@@ -2737,49 +2464,6 @@ ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac
@end example
@end itemize

-@section webm_chunk
-
-WebM Live Chunk Muxer.
-
-This muxer writes out WebM headers and chunks as separate files which can be
-consumed by clients that support WebM Live streams via DASH.
-
-@subsection Options
-
-This muxer supports the following options:
-
-@table @option
-@item chunk_start_index
-Index of the first chunk (defaults to 0).
-
-@item header
-Filename of the header where the initialization data will be written.
-
-@item audio_chunk_duration
-Duration of each audio chunk in milliseconds (defaults to 5000).
-@end table
-
-@subsection Example
-@example
-ffmpeg -f v4l2 -i /dev/video0 \
-       -f alsa -i hw:0 \
-       -map 0:0 \
-       -c:v libvpx-vp9 \
-       -s 640x360 -keyint_min 30 -g 30 \
-       -f webm_chunk \
-       -header webm_live_video_360.hdr \
-       -chunk_start_index 1 \
-       webm_live_video_360_%d.chk \
-       -map 1:0 \
-       -c:a libvorbis \
-       -b:a 128k \
-       -f webm_chunk \
-       -header webm_live_audio_128.hdr \
-       -chunk_start_index 1 \
-       -audio_chunk_duration 1000 \
-       webm_live_audio_128_%d.chk
-@end example
-
@section webm_dash_manifest

 WebM DASH Manifest muxer.
@@ -2846,4 +2530,47 @@ ffmpeg -f webm_dash_manifest -i video1.webm \
       manifest.xml
@end example

+@section webm_chunk
+
+WebM Live Chunk Muxer.
+
+This muxer writes out WebM headers and chunks as separate files which can be
+consumed by clients that support WebM Live streams via DASH.
+
+@subsection Options
+
+This muxer supports the following options:
+
+@table @option
+@item chunk_start_index
+Index of the first chunk (defaults to 0).
+
+@item header
+Filename of the header where the initialization data will be written.
+
+@item audio_chunk_duration
+Duration of each audio chunk in milliseconds (defaults to 5000).
+@end table
+
+@subsection Example
+@example
+ffmpeg -f v4l2 -i /dev/video0 \
+       -f alsa -i hw:0 \
+       -map 0:0 \
+       -c:v libvpx-vp9 \
+       -s 640x360 -keyint_min 30 -g 30 \
+       -f webm_chunk \
+       -header webm_live_video_360.hdr \
+       -chunk_start_index 1 \
+       webm_live_video_360_%d.chk \
+       -map 1:0 \
+       -c:a libvorbis \
+       -b:a 128k \
+       -f webm_chunk \
+       -header webm_live_audio_128.hdr \
+       -chunk_start_index 1 \
+       -audio_chunk_duration 1000 \
+       webm_live_audio_128_%d.chk
+@end example
+
@c man end MUXERS
--- a/doc/optimization.txt
+++ b/doc/optimization.txt
@@ -267,11 +267,6 @@ CELL/SPU:
 http://www-01.ibm.com/chips/techlib/techlib.nsf/techdocs/30B3520C93F437AB87257060006FFE5E/$file/Language_Extensions_for_CBEA_2.4.pdf
 http://www-01.ibm.com/chips/techlib/techlib.nsf/techdocs/9F820A5FFA3ECE8C8725716A0062585F/$file/CBE_Handbook_v1.1_24APR2007_pub.pdf

-RISC-V-specific:
----------------
-The RISC-V Instruction Set Manual, Volume 1, Unprivileged ISA:
-https://riscv.org/technical/specifications/
-
 GCC asm links:
 --------------
 official doc but quite ugly
--- a/doc/outdevs.texi
+++ b/doc/outdevs.texi
@@ -198,48 +198,13 @@ Amount of time to preroll video in seconds.
 Defaults to @option{0.5}.

@item duplex_mode
-Sets the decklink device duplex/profile mode. Must be @samp{unset}, @samp{half}, @samp{full},
-@samp{one_sub_device_full}, @samp{one_sub_device_half}, @samp{two_sub_device_full},
-@samp{four_sub_device_half}
+Sets the decklink device duplex mode. Must be @samp{unset}, @samp{half} or @samp{full}.
 Defaults to @samp{unset}.

-Note: DeckLink SDK 11.0 have replaced the duplex property by a profile property.
-For the DeckLink Duo 2 and DeckLink Quad 2, a profile is shared between any 2
-sub-devices that utilize the same connectors. For the DeckLink 8K Pro, a profile
-is shared between all 4 sub-devices. So DeckLink 8K Pro support four profiles.
-
-Valid profile modes for DeckLink 8K Pro(with DeckLink SDK >= 11.0):
-@samp{one_sub_device_full}, @samp{one_sub_device_half}, @samp{two_sub_device_full},
-@samp{four_sub_device_half}
-
-Valid profile modes for DeckLink Quad 2 and DeckLink Duo 2:
-@samp{half}, @samp{full}
-
@item timing_offset
 Sets the genlock timing pixel offset on the used output.
 Defaults to @samp{unset}.

-@item link
-Sets the SDI video link configuration on the used output. Must be
-@samp{unset}, @samp{single} link SDI, @samp{dual} link SDI or @samp{quad} link
-SDI.
-Defaults to @samp{unset}.
-
-@item sqd
-Enable Square Division Quad Split mode for Quad-link SDI output.
-Must be @samp{unset}, @samp{true} or @samp{false}.
-Defaults to @option{unset}.
-
-@item level_a
-Enable SMPTE Level A mode on the used output.
-Must be @samp{unset}, @samp{true} or @samp{false}.
-Defaults to @option{unset}.
-
-@item vanc_queue_size
-Sets maximum output buffer size in bytes for VANC data. If the buffering reaches this value,
-outgoing VANC data will be dropped.
-Defaults to @samp{1048576}.
-
@end table

@subsection Examples
@@ -426,18 +391,13 @@ For more information about SDL, check:

@table @option

-@item window_borderless
-Set SDL window border off.
-Default value is 0 (enable window border).
+@item window_title
+Set the SDL window title, if not specified default to the filename
+specified for the output device.

-@item window_enable_quit
-Enable quit action (using window button or keyboard key)
-when non-zero value is provided.
-Default value is 1 (enable quit action).
-
-@item window_fullscreen
-Set fullscreen mode when non-zero value is provided.
-Default value is zero.
+@item icon_title
+Set the name of the iconified SDL window, if not specified it is set
+to the same value of @var{window_title}.

@item window_size
 Set the SDL window size, can be a string of the form
@@ -445,13 +405,18 @@ Set the SDL window size, can be a string of the form
 If not specified it defaults to the size of the input video,
 downscaled according to the aspect ratio.

-@item window_title
-Set the SDL window title, if not specified default to the filename
-specified for the output device.
-
@item window_x
@item window_y
 Set the position of the window on the screen.
+
+@item window_fullscreen
+Set fullscreen mode when non-zero value is provided.
+Default value is zero.
+
+@item window_enable_quit
+Enable quit action (using window button or keyboard key)
+when non-zero value is provided.
+Default value is 1 (enable quit action)
@end table

@subsection Interactive commands
--- a/doc/platform.texi
+++ b/doc/platform.texi
@@ -92,6 +92,9 @@ For information about compiling FFmpeg on OS/2 see

@chapter Windows

+To get help and instructions for building FFmpeg under Windows, check out
+the FFmpeg Windows Help Forum at @url{http://ffmpeg.zeranoe.com/forum/}.
+
@section Native Windows compilation using MinGW or MinGW-w64

 FFmpeg can be built to run natively on Windows using the MinGW-w64
--- a/doc/protocols.texi
+++ b/doc/protocols.texi
@@ -215,38 +215,6 @@ ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
 Note that you may need to escape the character "|" which is special for
 many shells.

-@section concatf
-
-Physical concatenation protocol using a line break delimited list of
-resources.
-
-Read and seek from many resources in sequence as if they were
-a unique resource.
-
-A URL accepted by this protocol has the syntax:
-@example
-concatf:@var{URL}
-@end example
-
-where @var{URL} is the url containing a line break delimited list of
-resources to be concatenated, each one possibly specifying a distinct
-protocol. Special characters must be escaped with backslash or single
-quotes. See @ref{quoting_and_escaping,,the "Quoting and escaping"
-section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
-
-For example to read a sequence of files @file{split1.mpeg},
-@file{split2.mpeg}, @file{split3.mpeg} listed in separate lines within
-a file @file{split.txt} with @command{ffplay} use the command:
-@example
-ffplay concatf:split.txt
-@end example
-Where @file{split.txt} contains the lines:
-@example
-split1.mpeg
-split2.mpeg
-split3.mpeg
-@end example
-
@section crypto

 AES-encrypted stream reading protocol.
@@ -275,33 +243,6 @@ For example, to convert a GIF file given inline with @command{ffmpeg}:
 ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png
@end example

-@section fd
-
-File descriptor access protocol.
-
-The accepted syntax is:
-@example
-fd: -fd @var{file_descriptor}
-@end example
-
-If @option{fd} is not specified, by default the stdout file descriptor will be
-used for writing, stdin for reading. Unlike the pipe protocol, fd protocol has
-seek support if it corresponding to a regular file. fd protocol doesn't support
-pass file descriptor via URL for security.
-
-This protocol accepts the following options:
-
-@table @option
-@item blocksize
-Set I/O operation maximum block size, in bytes. Default value is
-@code{INT_MAX}, which results in not limiting the requested block size.
-Setting this value reasonably low improves user termination request reaction
-time, which is valuable if data transmission is slow.
-
-@item fd
-Set file descriptor.
-@end table
-
@section file

 File access protocol.
@@ -465,6 +406,9 @@ Set the Referer header. Include 'Referer: URL' header in HTTP request.
 Override the User-Agent header. If not specified the protocol will use a
 string describing the libavformat build. ("Lavf/<version>")

+@item user-agent
+This is a deprecated option, you can use user_agent instead it.
+
@item reconnect_at_eof
 If set then eof is treated like an error and causes reconnection, this is useful
 for live / endless streams.
@@ -641,35 +585,6 @@ Establish a TLS (HTTPS) connection to Icecast.
 icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint}
@end example

-@section ipfs
-
-InterPlanetary File System (IPFS) protocol support. One can access files stored
-on the IPFS network through so-called gateways. These are http(s) endpoints.
-This protocol wraps the IPFS native protocols (ipfs:// and ipns://) to be sent
-to such a gateway. Users can (and should) host their own node which means this
-protocol will use one's local gateway to access files on the IPFS network.
-
-This protocol accepts the following options:
-
-@table @option
-
-@item gateway
-Defines the gateway to use. When not set, the protocol will first try
-locating the local gateway by looking at @code{$IPFS_GATEWAY}, @code{$IPFS_PATH}
-and @code{$HOME/.ipfs/}, in that order.
-
-@end table
-
-One can use this protocol in 2 ways. Using IPFS:
-@example
-ffplay ipfs://<hash>
-@end example
-
-Or the IPNS protocol (IPNS is mutable IPFS):
-@example
-ffplay ipns://<hash>
-@end example
-
@section mmst

 MMS (Microsoft Media Server) protocol over TCP.
@@ -714,7 +629,7 @@ The accepted syntax is:
 pipe:[@var{number}]
@end example

-If @option{fd} isn't specified, @var{number} is the number corresponding to the file descriptor of the
+@var{number} is the number corresponding to the file descriptor of the
 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr).  If @var{number}
 is not specified, by default the stdout file descriptor will be used
 for writing, stdin for reading.
@@ -741,8 +656,6 @@ 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 if data transmission is slow.
-@item fd
-Set file descriptor.
@end table

 Note that some formats (typically MOV), require the output protocol to
@@ -803,14 +716,6 @@ Set internal RIST buffer size in milliseconds for retransmission of data.
 Default value is 0 which means the librist default (1 sec). Maximum value is 30
 seconds.

-@item fifo_size
-Size of the librist receiver output fifo in number of packets. This must be a
-power of 2.
-Defaults to 8192 (vs the librist default of 1024).
-
-@item overrun_nonfatal=@var{1|0}
-Survive in case of librist fifo buffer overrun. Default value is 0.
-
@item pkt_size
 Set maximum packet size for sending data. 1316 by default.

@@ -896,13 +801,6 @@ be named, by prefixing the type with 'N' and specifying the name before
 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
 times to construct arbitrary AMF sequences.

-@item rtmp_enhanced_codecs
-Specify the list of codecs the client advertises to support in an
-enhanced RTMP stream. This option should be set to a comma separated
-list of fourcc values, like @code{hvc1,av01,vp09} for multiple codecs
-or @code{hvc1} for only one codec. The specified list will be presented
-in the "fourCcLive" property of the Connect Command Message.
-
@item rtmp_flashver
 Version of the Flash plugin used to run the SWF player. The default
 is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
@@ -948,11 +846,6 @@ URL to player swf file, compute hash/size automatically.
@item rtmp_tcurl
 URL of the target stream. Defaults to proto://host[:port]/app.

-@item tcp_nodelay=@var{1|0}
-Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
-
-@emph{Remark: Writing to the socket is currently not optimized to minimize system calls and reduces the efficiency / effect of TCP_NODELAY.}
-
@end table

 For example to read with @command{ffplay} a multimedia resource named
@@ -1160,10 +1053,6 @@ set to 1) or to a default remote address (if set to 0).
@item localport=@var{n}
 Set the local RTP port to @var{n}.

-@item localaddr=@var{addr}
-Local IP address of a network interface used for sending packets or joining
-multicast groups.
-
@item timeout=@var{n}
 Set timeout (in microseconds) of socket I/O operations to @var{n}.

@@ -1211,59 +1100,6 @@ Options can be set on the @command{ffmpeg}/@command{ffplay} command
 line, or set in code via @code{AVOption}s or in
@code{avformat_open_input}.

-@subsection Muxer
-The following options are supported.
-
-@table @option
-@item rtsp_transport
-Set RTSP transport protocols.
-
-It accepts the following values:
-@table @samp
-@item udp
-Use UDP as lower transport protocol.
-
-@item tcp
-Use TCP (interleaving within the RTSP control channel) as lower
-transport protocol.
-@end table
-
-Default value is @samp{0}.
-
-@item rtsp_flags
-Set RTSP flags.
-
-The following values are accepted:
-@table @samp
-@item latm
-Use MP4A-LATM packetization instead of MPEG4-GENERIC for AAC.
-@item rfc2190
-Use RFC 2190 packetization instead of RFC 4629 for H.263.
-@item skip_rtcp
-Don't send RTCP sender reports.
-@item h264_mode0
-Use mode 0 for H.264 in RTP.
-@item send_bye
-Send RTCP BYE packets when finishing.
-@end table
-
-Default value is @samp{0}.
-
-
-@item min_port
-Set minimum local UDP port. Default value is 5000.
-
-@item max_port
-Set maximum local UDP port. Default value is 65000.
-
-@item buffer_size
-Set the maximum socket buffer size in bytes.
-
-@item pkt_size
-Set max send packet size (in bytes). Default value is 1472.
-@end table
-
-@subsection Demuxer
 The following options are supported.

@table @option
@@ -1289,10 +1125,6 @@ Use UDP multicast as lower transport protocol.
@item http
 Use HTTP tunneling as lower transport protocol, which is useful for
 passing proxies.
-
-@item https
-Use HTTPs tunneling as lower transport protocol, which is useful for
-passing proxies and widely used for security consideration.
@end table

 Multiple lower transport protocols may be specified, in that case they are
@@ -1310,9 +1142,6 @@ Accept packets only from negotiated peer address and port.
 Act as a server, listening for an incoming connection.
@item prefer_tcp
 Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
-@item satip_raw
-Export raw MPEG-TS stream instead of demuxing. The flag will simply write out
-the raw stream, with the original PAT/PMT/PIDs intact.
@end table

 Default value is @samp{none}.
@@ -1325,7 +1154,6 @@ The following flags are accepted:
@item video
@item audio
@item data
-@item subtitle
@end table

 By default it accepts all media types.
@@ -1336,23 +1164,21 @@ Set minimum local UDP port. Default value is 5000.
@item max_port
 Set maximum local UDP port. Default value is 65000.

-@item listen_timeout
-Set maximum timeout (in seconds) to establish an initial connection. Setting
-@option{listen_timeout} > 0 sets @option{rtsp_flags} to @samp{listen}. Default is -1
-which means an infinite timeout when @samp{listen} mode is set.
+@item timeout
+Set maximum timeout (in seconds) to wait for incoming connections.
+
+A value of -1 means infinite (default). This option implies the
+@option{rtsp_flags} set to @samp{listen}.

@item reorder_queue_size
 Set number of packets to buffer for handling of reordered packets.

-@item timeout
+@item stimeout
 Set socket TCP I/O timeout in microseconds.

-@item user_agent
+@item user-agent
 Override User-Agent header. If not specified, it defaults to the
 libavformat identifier string.
-
-@item buffer_size
-Set the maximum socket buffer size in bytes.
@end table

 When receiving data over UDP, the demuxer tries to reorder received packets
@@ -1637,12 +1463,6 @@ when the old encryption key is decommissioned. Default is -1.
 -1 means auto (0x1000 in srt library). The range for
 this option is integers in the 0 - @code{INT_MAX}.

-@item snddropdelay=@var{microseconds}
-The sender's extra delay before dropping packets. This delay is
-added to the default drop delay time interval value.
-
-Special value -1: Do not drop packets on the sender at all.
-
@item payload_size=@var{bytes}
 Sets the maximum declared size of a packet transferred
 during the single call to the sending function in Live
@@ -1742,9 +1562,6 @@ This option doesn’t make sense in Rendezvous connection; the result
 might be that simply one side will override the value from the other
 side and it’s the matter of luck which one would win

-@item srt_streamid=@var{string}
-Alias for @samp{streamid} to avoid conflict with ffmpeg command line option.
-
@item smoother=@var{live|file}
 The type of Smoother used for the transmission for that socket, which
 is responsible for the transmission and congestion control. The Smoother
@@ -1794,11 +1611,6 @@ Default is -1. -1 means auto (off with 0 seconds in live mode, on with 180
 seconds in file mode). The range for this option is integers in the
 0 - @code{INT_MAX}.

-@item tsbpd=@var{1|0}
-When true, use Timestamp-based Packet Delivery mode. The default behavior
-depends on the transmission type: enabled in live mode, disabled in file
-mode.
-
@end table

 For more information see: @url{https://github.com/Haivision/srt}.
@@ -1889,12 +1701,6 @@ The list of supported options follows.
 Listen for an incoming connection. 0 disables listen, 1 enables listen in
 single client mode, 2 enables listen in multi-client mode. Default value is 0.

-@item local_addr=@var{addr}
-Local IP address of a network interface used for tcp socket connect.
-
-@item local_port=@var{port}
-Local port used for tcp socket connect.
-
@item timeout=@var{microseconds}
 Set raise error timeout, expressed in microseconds.

@@ -1913,8 +1719,6 @@ Set send buffer size, expressed bytes.
@item tcp_nodelay=@var{1|0}
 Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.

-@emph{Remark: Writing to the socket is currently not optimized to minimize system calls and reduces the efficiency / effect of TCP_NODELAY.}
-
@item tcp_mss=@var{bytes}
 Set maximum segment size for outgoing TCP packets, expressed in bytes.
@end table
@@ -2168,4 +1972,5 @@ decoding errors.

@end table

+
@c man end PROTOCOLS
--- a/doc/resampler.texi
+++ b/doc/resampler.texi
@@ -11,8 +11,18 @@ programmatic use.

@table @option

-@item uchl, used_chlayout
-Set used input channel layout. Default is unset. This option is
+@item ich, in_channel_count
+Set the number of input channels. Default value is 0. Setting this
+value is not mandatory if the corresponding channel layout
+@option{in_channel_layout} is set.
+
+@item och, out_channel_count
+Set the number of output channels. Default value is 0. Setting this
+value is not mandatory if the corresponding channel layout
+@option{out_channel_layout} is set.
+
+@item uch, used_channel_count
+Set the number of used input channels. Default value is 0. This option is
 only used for special remapping.

@item isr, in_sample_rate
@@ -31,8 +41,8 @@ Specify the output sample format. It is set by default to @code{none}.
 Set the internal sample format. Default value is @code{none}.
 This will automatically be chosen when it is not explicitly set.

-@item ichl, in_chlayout
-@item ochl, out_chlayout
+@item icl, in_channel_layout
+@item ocl, out_channel_layout
 Set the input/output channel layout.

 See @ref{channel layout syntax,,the Channel Layout section in the ffmpeg-utils(1) manual,ffmpeg-utils}
--- a/doc/t2h.pm
+++ b/doc/t2h.pm
@@ -20,8 +20,45 @@
 # License along with FFmpeg; if not, write to the Free Software
 # Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA

+# Texinfo 7.0 changed the syntax of various functions.
+# Provide a shim for older versions.
+sub ff_set_from_init_file($$) {
+    my $key = shift;
+    my $value = shift;
+    if (exists &{'texinfo_set_from_init_file'}) {
+        texinfo_set_from_init_file($key, $value);
+    } else {
+        set_from_init_file($key, $value);
+    }
+}
+
+sub ff_get_conf($) {
+    my $key = shift;
+    if (exists &{'texinfo_get_conf'}) {
+        texinfo_get_conf($key);
+    } else {
+        get_conf($key);
+    }
+}
+
+sub get_formatting_function($$) {
+    my $obj = shift;
+    my $func = shift;
+
+    my $sub = $obj->can('formatting_function');
+    if ($sub) {
+        return $obj->formatting_function($func);
+    } else {
+        return $obj->{$func};
+    }
+}
+
+# determine texinfo version
+my $program_version_num = version->declare(ff_get_conf('PACKAGE_VERSION'))->numify;
+my $program_version_6_8 = $program_version_num >= 6.008000;
+
 # no navigation elements
-set_from_init_file('HEADERS', 0);
+ff_set_from_init_file('HEADERS', 0);

 sub ffmpeg_heading_command($$$$$)
 {
@@ -55,7 +92,7 @@ sub ffmpeg_heading_command($$$$$)
        $element = $command->{'parent'};
    }
    if ($element) {
-        $result .= &{$self->{'format_element_header'}}($self, $cmdname,
+        $result .= &{get_formatting_function($self, 'format_element_header')}($self, $cmdname,
                                                       $command, $element);
    }

@@ -112,7 +149,11 @@ sub ffmpeg_heading_command($$$$$)
                $cmdname
                    = $Texinfo::Common::level_to_structuring_command{$cmdname}->[$heading_level];
            }
-            $result .= &{$self->{'format_heading_text'}}(
+            # format_heading_text expects an array of headings for texinfo >= 7.0
+            if ($program_version_num >= 7.000000) {
+                $heading = [$heading];
+            }
+            $result .= &{get_formatting_function($self,'format_heading_text')}(
                        $self, $cmdname, $heading,
                        $heading_level +
                        $self->get_conf('CHAPTER_HEADER_LEVEL') - 1, $command);
@@ -126,23 +167,19 @@ foreach my $command (keys(%Texinfo::Common::sectioning_commands), 'node') {
    texinfo_register_command_formatting($command, \&ffmpeg_heading_command);
 }

-# determine if texinfo is at least version 6.8
-my $program_version_num = version->declare(get_conf('PACKAGE_VERSION'))->numify;
-my $program_version_6_8 = $program_version_num >= 6.008000;
-
 # print the TOC where @contents is used
 if ($program_version_6_8) {
-    set_from_init_file('CONTENTS_OUTPUT_LOCATION', 'inline');
+    ff_set_from_init_file('CONTENTS_OUTPUT_LOCATION', 'inline');
 } else {
-    set_from_init_file('INLINE_CONTENTS', 1);
+    ff_set_from_init_file('INLINE_CONTENTS', 1);
 }

 # make chapters <h2>
-set_from_init_file('CHAPTER_HEADER_LEVEL', 2);
+ff_set_from_init_file('CHAPTER_HEADER_LEVEL', 2);

 # Do not add <hr>
-set_from_init_file('DEFAULT_RULE', '');
-set_from_init_file('BIG_RULE', '');
+ff_set_from_init_file('DEFAULT_RULE', '');
+ff_set_from_init_file('BIG_RULE', '');

 # Customized file beginning
 sub ffmpeg_begin_file($$$)
@@ -159,7 +196,18 @@ sub ffmpeg_begin_file($$$)
    my ($title, $description, $encoding, $date, $css_lines,
        $doctype, $bodytext, $copying_comment, $after_body_open,
        $extra_head, $program_and_version, $program_homepage,
-        $program, $generator) = $self->_file_header_informations($command);
+        $program, $generator);
+    if ($program_version_num >= 7.000000) {
+        ($title, $description, $encoding, $date, $css_lines,
+         $doctype, $bodytext, $copying_comment, $after_body_open,
+         $extra_head, $program_and_version, $program_homepage,
+         $program, $generator) = $self->_file_header_information($command);
+    } else {
+        ($title, $description, $encoding, $date, $css_lines,
+         $doctype, $bodytext, $copying_comment, $after_body_open,
+         $extra_head, $program_and_version, $program_homepage,
+         $program, $generator) = $self->_file_header_informations($command);
+    }

    my $links = $self->_get_links ($filename, $element);

@@ -223,7 +271,7 @@ if ($program_version_6_8) {
 sub ffmpeg_end_file($)
 {
    my $self = shift;
-    my $program_string = &{$self->{'format_program_string'}}($self);
+    my $program_string = &{get_formatting_function($self,'format_program_string')}($self);
    my $program_text = <<EOT;
      <p style="font-size: small;">
        $program_string
@@ -244,7 +292,7 @@ if ($program_version_6_8) {

 # Dummy title command
 # Ignore title. Title is handled through ffmpeg_begin_file().
-set_from_init_file('USE_TITLEPAGE_FOR_TITLE', 1);
+ff_set_from_init_file('USE_TITLEPAGE_FOR_TITLE', 1);
 sub ffmpeg_title($$$$)
 {
    return '';
@@ -262,8 +310,14 @@ sub ffmpeg_float($$$$$)
    my $args = shift;
    my $content = shift;

-    my ($caption, $prepended) = Texinfo::Common::float_name_caption($self,
-                                                                $command);
+    my ($caption, $prepended);
+    if ($program_version_num >= 7.000000) {
+        ($caption, $prepended) = Texinfo::Convert::Converter::float_name_caption($self,
+                                                                                 $command);
+    } else {
+        ($caption, $prepended) = Texinfo::Common::float_name_caption($self,
+                                                                     $command);
+    }
    my $caption_text = '';
    my $prepended_text;
    my $prepended_save = '';
@@ -335,8 +389,13 @@ sub ffmpeg_float($$$$$)
            $caption->{'args'}->[0], 'float caption');
    }
    if ($prepended_text.$caption_text ne '') {
-        $prepended_text = $self->_attribute_class('div','float-caption'). '>'
-                . $prepended_text;
+        if ($program_version_num >= 7.000000) {
+            $prepended_text = $self->html_attribute_class('div',['float-caption']). '>'
+                    . $prepended_text;
+        } else {
+            $prepended_text = $self->_attribute_class('div','float-caption'). '>'
+                    . $prepended_text;
+        }
        $caption_text .= '</div>';
    }
    my $html_class = '';
@@ -349,8 +408,13 @@ sub ffmpeg_float($$$$$)
        $prepended_text = '';
        $caption_text   = '';
    }
-    return $self->_attribute_class('div', $html_class). '>' . "\n" .
-        $prepended_text . $caption_text . $content . '</div>';
+    if ($program_version_num >= 7.000000) {
+        return $self->html_attribute_class('div', [$html_class]). '>' . "\n" .
+            $prepended_text . $caption_text . $content . '</div>';
+    } else {
+        return $self->_attribute_class('div', $html_class). '>' . "\n" .
+            $prepended_text . $caption_text . $content . '</div>';
+    }
 }

 texinfo_register_command_formatting('float',
--- a/doc/transforms.md
+++ b/doc/transforms.md
--- a/doc/utils.texi
+++ b/doc/utils.texi
@@ -695,8 +695,6 @@ FL+FR+FC+LFE+SL+SR
 FL+FR+FC+BC+SL+SR
@item 6.0(front)
 FL+FR+FLC+FRC+SL+SR
-@item 3.1.2
-FL+FR+FC+LFE+TFL+TFR
@item hexagonal
 FL+FR+FC+BL+BR+BC
@item 6.1
@@ -715,48 +713,28 @@ FL+FR+FC+LFE+BL+BR+SL+SR
 FL+FR+FC+LFE+BL+BR+FLC+FRC
@item 7.1(wide-side)
 FL+FR+FC+LFE+FLC+FRC+SL+SR
-@item 5.1.2
-FL+FR+FC+LFE+BL+BR+TFL+TFR
@item octagonal
 FL+FR+FC+BL+BR+BC+SL+SR
-@item cube
-FL+FR+BL+BR+TFL+TFR+TBL+TBR
-@item 5.1.4
-FL+FR+FC+LFE+BL+BR+TFL+TFR+TBL+TBR
-@item 7.1.2
-FL+FR+FC+LFE+BL+BR+SL+SR+TFL+TFR
-@item 7.1.4
-FL+FR+FC+LFE+BL+BR+SL+SR+TFL+TFR+TBL+TBR
@item hexadecagonal
 FL+FR+FC+BL+BR+BC+SL+SR+WL+WR+TBL+TBR+TBC+TFC+TFL+TFR
@item downmix
 DL+DR
-@item 22.2
-FL+FR+FC+LFE+BL+BR+FLC+FRC+BC+SL+SR+TC+TFL+TFC+TFR+TBL+TBC+TBR+LFE2+TSL+TSR+BFC+BFL+BFR
@end table

-A custom channel layout can be specified as a sequence of terms, separated by '+'.
-Each term can be:
+A custom channel layout can be specified as a sequence of terms, separated by
+'+' or '|'. Each term can be:
@itemize
-@item
-the name of a single channel (e.g. @samp{FL}, @samp{FR}, @samp{FC}, @samp{LFE}, etc.),
-each optionally containing a custom name after a '@@', (e.g. @samp{FL@@Left},
-@samp{FR@@Right}, @samp{FC@@Center}, @samp{LFE@@Low_Frequency}, etc.)
-@end itemize
-
-A standard channel layout can be specified by the following:
-@itemize
-@item
-the name of a single channel (e.g. @samp{FL}, @samp{FR}, @samp{FC}, @samp{LFE}, etc.)
-
@item
 the name of a standard channel layout (e.g. @samp{mono},
@samp{stereo}, @samp{4.0}, @samp{quad}, @samp{5.0}, etc.)

+@item
+the name of a single channel (e.g. @samp{FL}, @samp{FR}, @samp{FC}, @samp{LFE}, etc.)
+
@item
 a number of channels, in decimal, followed by 'c', yielding the default channel
 layout for that number of channels (see the function
-@code{av_channel_layout_default}). Note that not all channel counts have a
+@code{av_get_default_channel_layout}). Note that not all channel counts have a
 default layout.

@item
@@ -773,7 +751,7 @@ Before libavutil version 53 the trailing character "c" to specify a number of
 channels was optional, but now it is required, while a channel layout mask can
 also be specified as a decimal number (if and only if not followed by "c" or "C").

-See also the function @code{av_channel_layout_from_string} defined in
+See also the function @code{av_get_channel_layout} defined in
@file{libavutil/channel_layout.h}.
@c man end SYNTAX

@@ -1085,13 +1063,13 @@ indication of the corresponding powers of 10 and of 2.
@item T
 10^12 / 2^40
@item P
-10^15 / 2^50
+10^15 / 2^40
@item E
-10^18 / 2^60
+10^18 / 2^50
@item Z
-10^21 / 2^70
+10^21 / 2^60
@item Y
-10^24 / 2^80
+10^24 / 2^70
@end table

@c man end EXPRESSION EVALUATION
--- a/ffbuild/.gitignore
+++ b/ffbuild/.gitignore
@@ -1,6 +1,4 @@
 /.config
-/bin2c
-/bin2c.exe
 /config.fate
 /config.log
 /config.mak
--- a/ffbuild/arch.mak
+++ b/ffbuild/arch.mak
@@ -8,14 +8,10 @@ OBJS-$(HAVE_MIPSFPU)   += $(MIPSFPU-OBJS)    $(MIPSFPU-OBJS-yes)
 OBJS-$(HAVE_MIPSDSP)   += $(MIPSDSP-OBJS)    $(MIPSDSP-OBJS-yes)
 OBJS-$(HAVE_MIPSDSPR2) += $(MIPSDSPR2-OBJS)  $(MIPSDSPR2-OBJS-yes)
 OBJS-$(HAVE_MSA)       += $(MSA-OBJS)        $(MSA-OBJS-yes)
-OBJS-$(HAVE_MMI)       += $(MMI-OBJS)        $(MMI-OBJS-yes)
-OBJS-$(HAVE_LSX)       += $(LSX-OBJS)        $(LSX-OBJS-yes)
-OBJS-$(HAVE_LASX)      += $(LASX-OBJS)       $(LASX-OBJS-yes)
+OBJS-$(HAVE_MMI)   += $(MMI-OBJS)   $(MMI-OBJS-yes)

 OBJS-$(HAVE_ALTIVEC) += $(ALTIVEC-OBJS) $(ALTIVEC-OBJS-yes)
 OBJS-$(HAVE_VSX)     += $(VSX-OBJS) $(VSX-OBJS-yes)

-OBJS-$(HAVE_RVV)     += $(RVV-OBJS)     $(RVV-OBJS-yes)
-
 OBJS-$(HAVE_MMX)     += $(MMX-OBJS)     $(MMX-OBJS-yes)
 OBJS-$(HAVE_X86ASM)  += $(X86ASM-OBJS)  $(X86ASM-OBJS-yes)
--- a/ffbuild/bin2c.c
+++ b/ffbuild/bin2c.c
@@ -1,76 +0,0 @@
-/*
- * This file is part of FFmpeg.
- *
- * Permission is hereby granted, free of charge, to any person obtaining a
- * copy of this software and associated documentation files (the "Software"),
- * to deal in the Software without restriction, including without limitation
- * the rights to use, copy, modify, merge, publish, distribute, sublicense,
- * and/or sell copies of the Software, and to permit persons to whom the
- * Software is furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.  IN NO EVENT SHALL
- * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
- * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
- * DEALINGS IN THE SOFTWARE.
- */
-
-#include <string.h>
-#include <stdio.h>
-
-int main(int argc, char **argv)
-{
-    const char *name;
-    FILE *input, *output;
-    unsigned int length = 0;
-    unsigned char data;
-
-    if (argc < 3 || argc > 4)
-        return 1;
-
-    input = fopen(argv[1], "rb");
-    if (!input)
-        return -1;
-
-    output = fopen(argv[2], "wb");
-    if (!output)
-        return -1;
-
-    if (argc == 4) {
-        name = argv[3];
-    } else {
-        size_t arglen = strlen(argv[1]);
-        name = argv[1];
-
-        for (int i = 0; i < arglen; i++) {
-            if (argv[1][i] == '.')
-                argv[1][i] = '_';
-            else if (argv[1][i] == '/')
-                name = &argv[1][i+1];
-        }
-    }
-
-    fprintf(output, "const unsigned char ff_%s_data[] = { ", name);
-
-    while (fread(&data, 1, 1, input) > 0) {
-        fprintf(output, "0x%02x, ", data);
-        length++;
-    }
-
-    fprintf(output, "0x00 };\n");
-    fprintf(output, "const unsigned int ff_%s_len = %u;\n", name, length);
-
-    fclose(output);
-
-    if (ferror(input) || !feof(input))
-        return -1;
-
-    fclose(input);
-
-    return 0;
-}
--- a/ffbuild/common.mak
+++ b/ffbuild/common.mak
@@ -12,13 +12,10 @@ endif

 ifndef SUBDIR

-BIN2CEXE = ffbuild/bin2c$(HOSTEXESUF)
-BIN2C = $(BIN2CEXE)
-
 ifndef V
 Q      = @
 ECHO   = printf "$(1)\t%s\n" $(2)
-BRIEF  = CC CXX OBJCC HOSTCC HOSTLD AS X86ASM AR LD STRIP CP WINDRES NVCC BIN2C
+BRIEF  = CC CXX OBJCC HOSTCC HOSTLD AS X86ASM AR LD STRIP CP WINDRES NVCC
 SILENT = DEPCC DEPHOSTCC DEPAS DEPX86ASM RANLIB RM

 MSG    = $@
@@ -29,8 +26,7 @@ $(foreach VAR,$(SILENT),$(eval override $(VAR) = @$($(VAR))))
 $(eval INSTALL = @$(call ECHO,INSTALL,$$(^:$(SRC_DIR)/%=%)); $(INSTALL))
 endif

-# Prepend to a recursively expanded variable without making it simply expanded.
-PREPEND = $(eval $(1) = $(patsubst %,$$(%), $(2)) $(value $(1)))
+ALLFFLIBS = avcodec avdevice avfilter avformat avresample avutil postproc swscale swresample

 # NASM requires -I path terminated with /
 IFLAGS     := -I. -I$(SRC_LINK)/
@@ -40,9 +36,7 @@ CCFLAGS     = $(CPPFLAGS) $(CFLAGS)
 OBJCFLAGS  += $(EOBJCFLAGS)
 OBJCCFLAGS  = $(CPPFLAGS) $(CFLAGS) $(OBJCFLAGS)
 ASFLAGS    := $(CPPFLAGS) $(ASFLAGS)
-# Use PREPEND here so that later (target-dependent) additions to CPPFLAGS
-# end up in CXXFLAGS.
-$(call PREPEND,CXXFLAGS, CPPFLAGS CFLAGS)
+CXXFLAGS   := $(CPPFLAGS) $(CFLAGS) $(CXXFLAGS)
 X86ASMFLAGS += $(IFLAGS:%=%/) -I$(<D)/ -Pconfig.asm

 HOSTCCFLAGS = $(IFLAGS) $(HOSTCPPFLAGS) $(HOSTCFLAGS)
@@ -62,8 +56,6 @@ COMPILE_HOSTC = $(call COMPILE,HOSTCC)
 COMPILE_NVCC = $(call COMPILE,NVCC)
 COMPILE_MMI = $(call COMPILE,CC,MMIFLAGS)
 COMPILE_MSA = $(call COMPILE,CC,MSAFLAGS)
-COMPILE_LSX = $(call COMPILE,CC,LSXFLAGS)
-COMPILE_LASX = $(call COMPILE,CC,LASXFLAGS)

 %_mmi.o: %_mmi.c
 	$(COMPILE_MMI)
@@ -71,12 +63,6 @@ COMPILE_LASX = $(call COMPILE,CC,LASXFLAGS)
 %_msa.o: %_msa.c
 	$(COMPILE_MSA)

-%_lsx.o: %_lsx.c
-	$(COMPILE_LSX)
-
-%_lasx.o: %_lasx.c
-	$(COMPILE_LASX)
-
 %.o: %.c
 	$(COMPILE_C)

@@ -112,35 +98,11 @@ COMPILE_LASX = $(call COMPILE,CC,LASXFLAGS)
 %.h.c:
 	$(Q)echo '#include "$*.h"' >$@

-$(BIN2CEXE): ffbuild/bin2c_host.o
-	$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $^ $(HOSTEXTRALIBS)
-
-%.metal.air: %.metal
-	$(METALCC) $< -o $@
-
-%.metallib: %.metal.air
-	$(METALLIB) --split-module-without-linking $< -o $@
-
-%.metallib.c: %.metallib $(BIN2CEXE)
-	$(BIN2C) $< $@ $(subst .,_,$(basename $(notdir $@)))
-
 %.ptx: %.cu $(SRC_PATH)/compat/cuda/cuda_runtime.h
 	$(COMPILE_NVCC)

-ifdef CONFIG_PTX_COMPRESSION
-%.ptx.gz: TAG = GZIP
-%.ptx.gz: %.ptx
-	$(M)gzip -c9 $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<) >$@
-
-%.ptx.c: %.ptx.gz $(BIN2CEXE)
-	$(BIN2C) $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<) $@ $(subst .,_,$(basename $(notdir $@)))
-else
-%.ptx.c: %.ptx $(BIN2CEXE)
-	$(BIN2C) $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<) $@ $(subst .,_,$(basename $(notdir $@)))
-endif
-
-clean::
-	$(RM) $(BIN2CEXE)
+%.ptx.c: %.ptx
+	$(Q)sh $(SRC_PATH)/compat/cuda/ptx2c.sh $@ $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<)

 %.c %.h %.pc %.ver %.version: TAG = GEN

@@ -160,8 +122,6 @@ include $(SRC_PATH)/ffbuild/arch.mak

 OBJS      += $(OBJS-yes)
 SLIBOBJS  += $(SLIBOBJS-yes)
-SHLIBOBJS += $(SHLIBOBJS-yes)
-STLIBOBJS += $(STLIBOBJS-yes)
 FFLIBS    := $($(NAME)_FFLIBS) $(FFLIBS-yes) $(FFLIBS)
 TESTPROGS += $(TESTPROGS-yes)

@@ -170,8 +130,6 @@ FFEXTRALIBS := $(LDLIBS:%=$(LD_LIB)) $(foreach lib,EXTRALIBS-$(NAME) $(FFLIBS:%=

 OBJS      := $(sort $(OBJS:%=$(SUBDIR)%))
 SLIBOBJS  := $(sort $(SLIBOBJS:%=$(SUBDIR)%))
-SHLIBOBJS := $(sort $(SHLIBOBJS:%=$(SUBDIR)%))
-STLIBOBJS := $(sort $(STLIBOBJS:%=$(SUBDIR)%))
 TESTOBJS  := $(TESTOBJS:%=$(SUBDIR)tests/%) $(TESTPROGS:%=$(SUBDIR)tests/%.o)
 TESTPROGS := $(TESTPROGS:%=$(SUBDIR)tests/%$(EXESUF))
 HOSTOBJS  := $(HOSTPROGS:%=$(SUBDIR)%.o)
@@ -193,7 +151,7 @@ HOBJS        = $(filter-out $(SKIPHEADERS:.h=.h.o),$(ALLHEADERS:.h=.h.o))
 PTXOBJS      = $(filter %.ptx.o,$(OBJS))
 $(HOBJS):     CCFLAGS += $(CFLAGS_HEADERS)
 checkheaders: $(HOBJS)
-.SECONDARY:   $(HOBJS:.o=.c) $(PTXOBJS:.o=.c) $(PTXOBJS:.o=.gz) $(PTXOBJS:.o=)
+.SECONDARY:   $(HOBJS:.o=.c) $(PTXOBJS:.o=.c) $(PTXOBJS:.o=)

 alltools: $(TOOLS)

@@ -207,14 +165,12 @@ $(OBJS):     | $(sort $(dir $(OBJS)))
 $(HOBJS):    | $(sort $(dir $(HOBJS)))
 $(HOSTOBJS): | $(sort $(dir $(HOSTOBJS)))
 $(SLIBOBJS): | $(sort $(dir $(SLIBOBJS)))
-$(SHLIBOBJS): | $(sort $(dir $(SHLIBOBJS)))
-$(STLIBOBJS): | $(sort $(dir $(STLIBOBJS)))
 $(TESTOBJS): | $(sort $(dir $(TESTOBJS)))
 $(TOOLOBJS): | tools

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

-CLEANSUFFIXES     = *.d *.gcda *.gcno *.h.c *.ho *.map *.o *.pc *.ptx *.ptx.gz *.ptx.c *.ver *.version *$(DEFAULT_X86ASMD).asm *~ *.ilk *.pdb
+CLEANSUFFIXES     = *.d *.gcda *.gcno *.h.c *.ho *.map *.o *.pc *.ptx *.ptx.c *.ver *.version *$(DEFAULT_X86ASMD).asm *~ *.ilk *.pdb
 LIBSUFFIXES       = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a

 define RULES
@@ -224,4 +180,4 @@ endef

 $(eval $(RULES))

-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d) $(SHLIBOBJS:.o=.d) $(STLIBOBJS:.o=.d) $(SLIBOBJS:.o=.d)) $(OBJS:.o=$(DEFAULT_X86ASMD).d)
+-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d) $(SLIBOBJS:.o=.d)) $(OBJS:.o=$(DEFAULT_X86ASMD).d)
--- a/ffbuild/library.mak
+++ b/ffbuild/library.mak
@@ -14,26 +14,10 @@ INSTHEADERS := $(INSTHEADERS) $(HEADERS:%=$(SUBDIR)%)
 all-$(CONFIG_STATIC): $(SUBDIR)$(LIBNAME)  $(SUBDIR)lib$(FULLNAME).pc
 all-$(CONFIG_SHARED): $(SUBDIR)$(SLIBNAME) $(SUBDIR)lib$(FULLNAME).pc

-LIBOBJS := $(OBJS) $(SHLIBOBJS) $(STLIBOBJS) $(SUBDIR)%.h.o $(TESTOBJS)
+LIBOBJS := $(OBJS) $(SUBDIR)%.h.o $(TESTOBJS)
 $(LIBOBJS) $(LIBOBJS:.o=.s) $(LIBOBJS:.o=.i):   CPPFLAGS += -DHAVE_AV_CONFIG_H

-ifdef CONFIG_SHARED
-# In case both shared libs and static libs are enabled, it can happen
-# that a user might want to link e.g. libavformat statically, but
-# libavcodec and the other libs dynamically. In this case
-# libavformat won't be able to access libavcodec's internal symbols,
-# so that they have to be duplicated into the archive just like
-# for purely shared builds.
-# Test programs are always statically linked against their library
-# to be able to access their library's internals, even with shared builds.
-# Yet linking against dependend libraries still uses dynamic linking.
-# This means that we are in the scenario described above.
-# In case only static libs are used, the linker will only use
-# one of these copies; this depends on the duplicated object files
-# containing exactly the same symbols.
-OBJS += $(SHLIBOBJS)
-endif
-$(SUBDIR)$(LIBNAME): $(OBJS) $(STLIBOBJS)
+$(SUBDIR)$(LIBNAME): $(OBJS)
 	$(RM) $@
 	$(AR) $(ARFLAGS) $(AR_O) $^
 	$(RANLIB) $@
@@ -52,8 +36,8 @@ $(LIBOBJS): CPPFLAGS += -DBUILDING_$(NAME)
 $(TESTPROGS) $(TOOLS): %$(EXESUF): %.o
 	$$(LD) $(LDFLAGS) $(LDEXEFLAGS) $$(LD_O) $$(filter %.o,$$^) $$(THISLIB) $(FFEXTRALIBS) $$(EXTRALIBS-$$(*F)) $$(ELIBS)

-$(SUBDIR)lib$(NAME).version: $(SUBDIR)version.h $(SUBDIR)version_major.h | $(SUBDIR)
-	$$(M) $$(SRC_PATH)/ffbuild/libversion.sh $(NAME) $$^ > $$@
+$(SUBDIR)lib$(NAME).version: $(SUBDIR)version.h | $(SUBDIR)
+	$$(M) $$(SRC_PATH)/ffbuild/libversion.sh $(NAME) $$< > $$@

 $(SUBDIR)lib$(FULLNAME).pc: $(SUBDIR)version.h ffbuild/config.sh | $(SUBDIR)
 	$$(M) $$(SRC_PATH)/ffbuild/pkgconfig_generate.sh $(NAME) "$(DESC)"
@@ -64,7 +48,7 @@ $(SUBDIR)lib$(NAME).ver: $(SUBDIR)lib$(NAME).v $(OBJS)
 $(SUBDIR)$(SLIBNAME): $(SUBDIR)$(SLIBNAME_WITH_MAJOR)
 	$(Q)cd ./$(SUBDIR) && $(LN_S) $(SLIBNAME_WITH_MAJOR) $(SLIBNAME)

-$(SUBDIR)$(SLIBNAME_WITH_MAJOR): $(OBJS) $(SHLIBOBJS) $(SLIBOBJS) $(SUBDIR)lib$(NAME).ver
+$(SUBDIR)$(SLIBNAME_WITH_MAJOR): $(OBJS) $(SLIBOBJS) $(SUBDIR)lib$(NAME).ver
 	$(SLIB_CREATE_DEF_CMD)
 	$$(LD) $(SHFLAGS) $(LDFLAGS) $(LDSOFLAGS) $$(LD_O) $$(filter %.o,$$^) $(FFEXTRALIBS)
 	$(SLIB_EXTRA_CMD)
--- a/ffbuild/libversion.sh
+++ b/ffbuild/libversion.sh
@@ -5,12 +5,8 @@ toupper(){
 name=lib$1
 ucname=$(toupper ${name})
 file=$2
-file2=$3

 eval $(awk "/#define ${ucname}_VERSION_M/ { print \$2 \"=\" \$3 }" "$file")
-if [ -f "$file2" ]; then
-  eval $(awk "/#define ${ucname}_VERSION_M/ { print \$2 \"=\" \$3 }" "$file2")
-fi
 eval ${ucname}_VERSION=\$${ucname}_VERSION_MAJOR.\$${ucname}_VERSION_MINOR.\$${ucname}_VERSION_MICRO
 eval echo "${name}_VERSION=\$${ucname}_VERSION"
 eval echo "${name}_VERSION_MAJOR=\$${ucname}_VERSION_MAJOR"
--- a/fftools/Makefile
+++ b/fftools/Makefile
@@ -9,24 +9,15 @@ AVBASENAMES  = ffmpeg ffplay ffprobe
 ALLAVPROGS   = $(AVBASENAMES:%=%$(PROGSSUF)$(EXESUF))
 ALLAVPROGS_G = $(AVBASENAMES:%=%$(PROGSSUF)_g$(EXESUF))

-OBJS-ffmpeg +=                  \
-    fftools/ffmpeg_dec.o        \
-    fftools/ffmpeg_demux.o      \
-    fftools/ffmpeg_enc.o        \
-    fftools/ffmpeg_filter.o     \
-    fftools/ffmpeg_hw.o         \
-    fftools/ffmpeg_mux.o        \
-    fftools/ffmpeg_mux_init.o   \
-    fftools/ffmpeg_opt.o        \
-    fftools/objpool.o           \
-    fftools/sync_queue.o        \
-    fftools/thread_queue.o      \
+OBJS-ffmpeg                        += fftools/ffmpeg_opt.o fftools/ffmpeg_filter.o fftools/ffmpeg_hw.o
+OBJS-ffmpeg-$(CONFIG_LIBMFX)       += fftools/ffmpeg_qsv.o
+ifndef CONFIG_VIDEOTOOLBOX
+OBJS-ffmpeg-$(CONFIG_VDA)          += fftools/ffmpeg_videotoolbox.o
+endif
+OBJS-ffmpeg-$(CONFIG_VIDEOTOOLBOX) += fftools/ffmpeg_videotoolbox.o

 define DOFFTOOL
-OBJS-$(1) += fftools/cmdutils.o fftools/opt_common.o fftools/$(1).o $(OBJS-$(1)-yes)
-ifdef HAVE_GNU_WINDRES
-OBJS-$(1) += fftools/fftoolsres.o
-endif
+OBJS-$(1) += fftools/cmdutils.o fftools/$(1).o $(OBJS-$(1)-yes)
 $(1)$(PROGSSUF)_g$(EXESUF): $$(OBJS-$(1))
 $$(OBJS-$(1)): | fftools
 $$(OBJS-$(1)): CFLAGS  += $(CFLAGS-$(1))
--- a/fftools/cmdutils.c
+++ b/fftools/cmdutils.c
--- a/fftools/cmdutils.h
+++ b/fftools/cmdutils.h
@@ -44,16 +44,33 @@ extern const char program_name[];
 */
 extern const int program_birth_year;

+extern AVCodecContext *avcodec_opts[AVMEDIA_TYPE_NB];
+extern AVFormatContext *avformat_opts;
 extern AVDictionary *sws_dict;
 extern AVDictionary *swr_opts;
-extern AVDictionary *format_opts, *codec_opts;
+extern AVDictionary *format_opts, *codec_opts, *resample_opts;
 extern int hide_banner;

+/**
+ * Register a program-specific cleanup routine.
+ */
+void register_exit(void (*cb)(int ret));
+
+/**
+ * Wraps exit with a program-specific cleanup routine.
+ */
+void exit_program(int ret) av_noreturn;
+
 /**
 * Initialize dynamic library loading
 */
 void init_dynload(void);

+/**
+ * Initialize the cmdutils option system, in particular
+ * allocate the *_opts contexts.
+ */
+void init_opts(void);
 /**
 * Uninitialize the cmdutils option system, in particular
 * free the *_opts contexts and their contents.
@@ -66,12 +83,28 @@ void uninit_opts(void);
 */
 void log_callback_help(void* ptr, int level, const char* fmt, va_list vl);

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

+/**
+ * Set the libav* libraries log level.
+ */
+int opt_loglevel(void *optctx, const char *opt, const char *arg);
+
+int opt_report(void *optctx, const char *opt, const char *arg);
+
+int opt_max_alloc(void *optctx, const char *opt, const char *arg);
+
+int opt_codec_debug(void *optctx, const char *opt, const char *arg);
+
 /**
 * Limit the execution time.
 */
@@ -79,6 +112,8 @@ int opt_timelimit(void *optctx, const char *opt, const char *arg);

 /**
 * Parse a string and return its corresponding value as a double.
+ * Exit from the application if the string cannot be correctly
+ * parsed or the corresponding value is invalid.
 *
 * @param context the context of the value to be set (e.g. the
 * corresponding command line option name)
@@ -88,8 +123,25 @@ int opt_timelimit(void *optctx, const char *opt, const char *arg);
 * @param min the minimum valid accepted value
 * @param max the maximum valid accepted value
 */
-int parse_number(const char *context, const char *numstr, int type,
-                 double min, double max, double *dst);
+double parse_number_or_die(const char *context, const char *numstr, int type,
+                           double min, double max);
+
+/**
+ * Parse a string specifying a time and return its corresponding
+ * value as a number of microseconds. Exit from the application if
+ * the string cannot be correctly parsed.
+ *
+ * @param context the context of the value to be set (e.g. the
+ * corresponding command line option name)
+ * @param timestr the string to be parsed
+ * @param is_duration a flag which tells how to interpret timestr, if
+ * not zero timestr is interpreted as a duration, otherwise as a
+ * date
+ *
+ * @see av_parse_time()
+ */
+int64_t parse_time_or_die(const char *context, const char *timestr,
+                          int is_duration);

 typedef struct SpecifierOpt {
    char *specifier;    /**< stream/chapter/program/... specifier */
@@ -149,6 +201,47 @@ typedef struct OptionDef {
 void show_help_options(const OptionDef *options, const char *msg, int req_flags,
                       int rej_flags, int alt_flags);

+#if CONFIG_AVDEVICE
+#define CMDUTILS_COMMON_OPTIONS_AVDEVICE                                                                                \
+    { "sources"    , OPT_EXIT | HAS_ARG, { .func_arg = show_sources },                                                  \
+      "list sources of the input device", "device" },                                                                   \
+    { "sinks"      , OPT_EXIT | HAS_ARG, { .func_arg = show_sinks },                                                    \
+      "list sinks of the output device", "device" },                                                                    \
+
+#else
+#define CMDUTILS_COMMON_OPTIONS_AVDEVICE
+#endif
+
+#define CMDUTILS_COMMON_OPTIONS                                                                                         \
+    { "L",           OPT_EXIT,             { .func_arg = show_license },     "show license" },                          \
+    { "h",           OPT_EXIT,             { .func_arg = show_help },        "show help", "topic" },                    \
+    { "?",           OPT_EXIT,             { .func_arg = show_help },        "show help", "topic" },                    \
+    { "help",        OPT_EXIT,             { .func_arg = show_help },        "show help", "topic" },                    \
+    { "-help",       OPT_EXIT,             { .func_arg = show_help },        "show help", "topic" },                    \
+    { "version",     OPT_EXIT,             { .func_arg = show_version },     "show version" },                          \
+    { "buildconf",   OPT_EXIT,             { .func_arg = show_buildconf },   "show build configuration" },              \
+    { "formats",     OPT_EXIT,             { .func_arg = show_formats },     "show available formats" },                \
+    { "muxers",      OPT_EXIT,             { .func_arg = show_muxers },      "show available muxers" },                 \
+    { "demuxers",    OPT_EXIT,             { .func_arg = show_demuxers },    "show available demuxers" },               \
+    { "devices",     OPT_EXIT,             { .func_arg = show_devices },     "show available devices" },                \
+    { "codecs",      OPT_EXIT,             { .func_arg = show_codecs },      "show available codecs" },                 \
+    { "decoders",    OPT_EXIT,             { .func_arg = show_decoders },    "show available decoders" },               \
+    { "encoders",    OPT_EXIT,             { .func_arg = show_encoders },    "show available encoders" },               \
+    { "bsfs",        OPT_EXIT,             { .func_arg = show_bsfs },        "show available bit stream filters" },     \
+    { "protocols",   OPT_EXIT,             { .func_arg = show_protocols },   "show available protocols" },              \
+    { "filters",     OPT_EXIT,             { .func_arg = show_filters },     "show available filters" },                \
+    { "pix_fmts",    OPT_EXIT,             { .func_arg = show_pix_fmts },    "show available pixel formats" },          \
+    { "layouts",     OPT_EXIT,             { .func_arg = show_layouts },     "show standard channel layouts" },         \
+    { "sample_fmts", OPT_EXIT,             { .func_arg = show_sample_fmts }, "show available audio sample formats" },   \
+    { "colors",      OPT_EXIT,             { .func_arg = show_colors },      "show available color names" },            \
+    { "loglevel",    HAS_ARG,              { .func_arg = opt_loglevel },     "set logging level", "loglevel" },         \
+    { "v",           HAS_ARG,              { .func_arg = opt_loglevel },     "set logging level", "loglevel" },         \
+    { "report",      0,                    { .func_arg = 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" },          \
+    CMDUTILS_COMMON_OPTIONS_AVDEVICE                                                                                    \
+
 /**
 * Show help for all options with given flags in class and all its
 * children.
@@ -161,6 +254,11 @@ void show_help_children(const AVClass *class, int flags);
 */
 void show_help_default(const char *opt, const char *arg);

+/**
+ * Generic -h handler common to all fftools.
+ */
+int show_help(void *optctx, const char *opt, const char *arg);
+
 /**
 * Parse the command line arguments.
 *
@@ -173,8 +271,8 @@ void show_help_default(const char *opt, const char *arg);
 * argument without a leading option name flag. NULL if such arguments do
 * not have to be processed.
 */
-int parse_options(void *optctx, int argc, char **argv, const OptionDef *options,
-                  int (* parse_arg_function)(void *optctx, const char*));
+void parse_options(void *optctx, int argc, char **argv, const OptionDef *options,
+                   void (* parse_arg_function)(void *optctx, const char*));

 /**
 * Parse one given option.
@@ -219,6 +317,7 @@ typedef struct OptionGroup {

    AVDictionary *codec_opts;
    AVDictionary *format_opts;
+    AVDictionary *resample_opts;
    AVDictionary *sws_dict;
    AVDictionary *swr_opts;
 } OptionGroup;
@@ -312,12 +411,10 @@ int check_stream_specifier(AVFormatContext *s, AVStream *st, const char *spec);
 * @param st A stream from s for which the options should be filtered.
 * @param codec The particular codec for which the options should be filtered.
 *              If null, the default one is looked up according to the codec id.
- * @param dst a pointer to the created dictionary
- * @return a non-negative number on success, a negative error code on failure
+ * @return a pointer to the created dictionary
 */
-int filter_codec_opts(const AVDictionary *opts, enum AVCodecID codec_id,
-                      AVFormatContext *s, AVStream *st, const AVCodec *codec,
-                      AVDictionary **dst);
+AVDictionary *filter_codec_opts(AVDictionary *opts, enum AVCodecID codec_id,
+                                AVFormatContext *s, AVStream *st, const AVCodec *codec);

 /**
 * Setup AVCodecContext options for avformat_find_stream_info().
@@ -326,10 +423,12 @@ int filter_codec_opts(const AVDictionary *opts, enum AVCodecID codec_id,
 * contained in s.
 * Each dictionary will contain the options from codec_opts which can
 * be applied to the corresponding stream codec context.
+ *
+ * @return pointer to the created array of dictionaries, NULL if it
+ * cannot be created
 */
-int setup_find_stream_info_opts(AVFormatContext *s,
-                                AVDictionary *codec_opts,
-                                AVDictionary ***dst);
+AVDictionary **setup_find_stream_info_opts(AVFormatContext *s,
+                                           AVDictionary *codec_opts);

 /**
 * Print an error message to stderr, indicating filename and a human
@@ -349,6 +448,136 @@ void print_error(const char *filename, int err);
 */
 void show_banner(int argc, char **argv, const OptionDef *options);

+/**
+ * Print the version of the program to stdout. The version message
+ * depends on the current versions of the repository and of the libav*
+ * libraries.
+ * This option processing function does not utilize the arguments.
+ */
+int show_version(void *optctx, const char *opt, const char *arg);
+
+/**
+ * Print the build configuration of the program to stdout. The contents
+ * depend on the definition of FFMPEG_CONFIGURATION.
+ * This option processing function does not utilize the arguments.
+ */
+int show_buildconf(void *optctx, const char *opt, const char *arg);
+
+/**
+ * Print the license of the program to stdout. The license depends on
+ * the license of the libraries compiled into the program.
+ * This option processing function does not utilize the arguments.
+ */
+int show_license(void *optctx, const char *opt, const char *arg);
+
+/**
+ * Print a listing containing all the formats supported by the
+ * program (including devices).
+ * This option processing function does not utilize the arguments.
+ */
+int show_formats(void *optctx, const char *opt, const char *arg);
+
+/**
+ * Print a listing containing all the muxers supported by the
+ * program (including devices).
+ * This option processing function does not utilize the arguments.
+ */
+int show_muxers(void *optctx, const char *opt, const char *arg);
+
+/**
+ * Print a listing containing all the demuxer supported by the
+ * program (including devices).
+ * This option processing function does not utilize the arguments.
+ */
+int show_demuxers(void *optctx, const char *opt, const char *arg);
+
+/**
+ * Print a listing containing all the devices supported by the
+ * program.
+ * This option processing function does not utilize the arguments.
+ */
+int show_devices(void *optctx, const char *opt, const char *arg);
+
+#if CONFIG_AVDEVICE
+/**
+ * Print a listing containing autodetected sinks of the output device.
+ * Device name with options may be passed as an argument to limit results.
+ */
+int show_sinks(void *optctx, const char *opt, const char *arg);
+
+/**
+ * Print a listing containing autodetected sources of the input device.
+ * Device name with options may be passed as an argument to limit results.
+ */
+int show_sources(void *optctx, const char *opt, const char *arg);
+#endif
+
+/**
+ * Print a listing containing all the codecs supported by the
+ * program.
+ * This option processing function does not utilize the arguments.
+ */
+int show_codecs(void *optctx, const char *opt, const char *arg);
+
+/**
+ * Print a listing containing all the decoders supported by the
+ * program.
+ */
+int show_decoders(void *optctx, const char *opt, const char *arg);
+
+/**
+ * Print a listing containing all the encoders supported by the
+ * program.
+ */
+int show_encoders(void *optctx, const char *opt, const char *arg);
+
+/**
+ * Print a listing containing all the filters supported by the
+ * program.
+ * This option processing function does not utilize the arguments.
+ */
+int show_filters(void *optctx, const char *opt, const char *arg);
+
+/**
+ * Print a listing containing all the bit stream filters supported by the
+ * program.
+ * This option processing function does not utilize the arguments.
+ */
+int show_bsfs(void *optctx, const char *opt, const char *arg);
+
+/**
+ * Print a listing containing all the protocols supported by the
+ * program.
+ * This option processing function does not utilize the arguments.
+ */
+int show_protocols(void *optctx, const char *opt, const char *arg);
+
+/**
+ * Print a listing containing all the pixel formats supported by the
+ * program.
+ * This option processing function does not utilize the arguments.
+ */
+int show_pix_fmts(void *optctx, const char *opt, const char *arg);
+
+/**
+ * Print a listing containing all the standard channel layouts supported by
+ * the program.
+ * This option processing function does not utilize the arguments.
+ */
+int show_layouts(void *optctx, const char *opt, const char *arg);
+
+/**
+ * Print a listing containing all the sample formats supported by the
+ * program.
+ */
+int show_sample_fmts(void *optctx, const char *opt, const char *arg);
+
+/**
+ * Print a listing containing all the color names and values recognized
+ * by the program.
+ */
+int show_colors(void *optctx, const char *opt, const char *arg);
+
 /**
 * Return a positive value if a line read from standard input
 * starts with [yY], otherwise return 0.
@@ -378,31 +607,20 @@ FILE *get_preset_file(char *filename, size_t filename_size,

 /**
 * Realloc array to hold new_size elements of elem_size.
+ * Calls exit() on failure.
 *
- * @param array pointer to the array to reallocate, will be updated
- *              with a new pointer on success
+ * @param array array to reallocate
 * @param elem_size size in bytes of each element
 * @param size new element count will be written here
 * @param new_size number of elements to place in reallocated array
- * @return a non-negative number on success, a negative error code on failure
+ * @return reallocated array
 */
-int grow_array(void **array, int elem_size, int *size, int new_size);
+void *grow_array(void *array, int elem_size, int *size, int new_size);

-/**
- * Atomically add a new element to an array of pointers, i.e. allocate
- * a new entry, reallocate the array of pointers and make the new last
- * member of this array point to the newly allocated buffer.
- *
- * @param array     array of pointers to reallocate
- * @param elem_size size of the new element to allocate
- * @param nb_elems  pointer to the number of elements of the array array;
- *                  *nb_elems will be incremented by one by this function.
- * @return pointer to the newly allocated entry or NULL on failure
- */
-void *allocate_array_elem(void *array, size_t elem_size, int *nb_elems);
+#define media_type_string av_get_media_type_string

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

 #define GET_PIX_FMT_NAME(pix_fmt)\
    const char *name = av_get_pix_fmt_name(pix_fmt);
@@ -417,6 +635,14 @@ void *allocate_array_elem(void *array, size_t elem_size, int *nb_elems);
    char name[16];\
    snprintf(name, sizeof(name), "%d", rate);

-double get_rotation(const int32_t *displaymatrix);
+#define GET_CH_LAYOUT_NAME(ch_layout)\
+    char name[16];\
+    snprintf(name, sizeof(name), "0x%"PRIx64, ch_layout);
+
+#define GET_CH_LAYOUT_DESC(ch_layout)\
+    char name[128];\
+    av_get_channel_layout_string(name, sizeof(name), 0, ch_layout);
+
+double get_rotation(AVStream *st);

 #endif /* FFTOOLS_CMDUTILS_H */
--- a/fftools/ffmpeg.c
+++ b/fftools/ffmpeg.c
--- a/fftools/ffmpeg.h
+++ b/fftools/ffmpeg.h
--- a/fftools/ffmpeg_dec.c
+++ b/fftools/ffmpeg_dec.c
--- a/fftools/ffmpeg_demux.c
+++ b/fftools/ffmpeg_demux.c
--- a/fftools/ffmpeg_enc.c
+++ b/fftools/ffmpeg_enc.c
@@ -1,888 +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>
-#include <stdint.h>
-
-#include "ffmpeg.h"
-
-#include "libavutil/avassert.h"
-#include "libavutil/avstring.h"
-#include "libavutil/avutil.h"
-#include "libavutil/dict.h"
-#include "libavutil/display.h"
-#include "libavutil/eval.h"
-#include "libavutil/frame.h"
-#include "libavutil/intreadwrite.h"
-#include "libavutil/log.h"
-#include "libavutil/pixdesc.h"
-#include "libavutil/rational.h"
-#include "libavutil/timestamp.h"
-
-#include "libavcodec/avcodec.h"
-
-#include "libavformat/avformat.h"
-
-struct Encoder {
-    AVFrame *sq_frame;
-
-    // packet for receiving encoded output
-    AVPacket *pkt;
-
-    // combined size of all the packets received from the encoder
-    uint64_t data_size;
-
-    // number of packets received from the encoder
-    uint64_t packets_encoded;
-
-    int opened;
-};
-
-void enc_free(Encoder **penc)
-{
-    Encoder *enc = *penc;
-
-    if (!enc)
-        return;
-
-    av_frame_free(&enc->sq_frame);
-
-    av_packet_free(&enc->pkt);
-
-    av_freep(penc);
-}
-
-int enc_alloc(Encoder **penc, const AVCodec *codec)
-{
-    Encoder *enc;
-
-    *penc = NULL;
-
-    enc = av_mallocz(sizeof(*enc));
-    if (!enc)
-        return AVERROR(ENOMEM);
-
-    enc->pkt = av_packet_alloc();
-    if (!enc->pkt)
-        goto fail;
-
-    *penc = enc;
-
-    return 0;
-fail:
-    enc_free(&enc);
-    return AVERROR(ENOMEM);
-}
-
-static int hw_device_setup_for_encode(OutputStream *ost, AVBufferRef *frames_ref)
-{
-    const AVCodecHWConfig *config;
-    HWDevice *dev = NULL;
-    int i;
-
-    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_ctx->codec, 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_ctx->codec->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);
-    }
-
-    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_ctx->codec->name);
-        ost->enc_ctx->hw_device_ctx = av_buffer_ref(dev->device_ref);
-        if (!ost->enc_ctx->hw_device_ctx)
-            return AVERROR(ENOMEM);
-    } else {
-        // No device required, or no device available.
-    }
-    return 0;
-}
-
-static int set_encoder_id(OutputFile *of, OutputStream *ost)
-{
-    const char *cname = ost->enc_ctx->codec->name;
-    uint8_t *encoder_string;
-    int encoder_string_len;
-
-    if (av_dict_get(ost->st->metadata, "encoder",  NULL, 0))
-        return 0;
-
-    encoder_string_len = sizeof(LIBAVCODEC_IDENT) + strlen(cname) + 2;
-    encoder_string     = av_mallocz(encoder_string_len);
-    if (!encoder_string)
-        return AVERROR(ENOMEM);
-
-    if (!of->bitexact && !ost->bitexact)
-        av_strlcpy(encoder_string, LIBAVCODEC_IDENT " ", encoder_string_len);
-    else
-        av_strlcpy(encoder_string, "Lavc ", encoder_string_len);
-    av_strlcat(encoder_string, cname, encoder_string_len);
-    av_dict_set(&ost->st->metadata, "encoder",  encoder_string,
-                AV_DICT_DONT_STRDUP_VAL | AV_DICT_DONT_OVERWRITE);
-
-    return 0;
-}
-
-int enc_open(OutputStream *ost, const AVFrame *frame)
-{
-    InputStream *ist = ost->ist;
-    Encoder              *e = ost->enc;
-    AVCodecContext *enc_ctx = ost->enc_ctx;
-    AVCodecContext *dec_ctx = NULL;
-    const AVCodec      *enc = enc_ctx->codec;
-    OutputFile      *of = output_files[ost->file_index];
-    FrameData *fd;
-    int ret;
-
-    if (e->opened)
-        return 0;
-
-    // frame is always non-NULL for audio and video
-    av_assert0(frame || (enc->type != AVMEDIA_TYPE_VIDEO && enc->type != AVMEDIA_TYPE_AUDIO));
-
-    if (frame) {
-        av_assert0(frame->opaque_ref);
-        fd = (FrameData*)frame->opaque_ref->data;
-    }
-
-    ret = set_encoder_id(output_files[ost->file_index], ost);
-    if (ret < 0)
-        return ret;
-
-    if (ist) {
-        dec_ctx = ist->dec_ctx;
-    }
-
-    // the timebase is chosen by filtering code
-    if (ost->type == AVMEDIA_TYPE_AUDIO || ost->type == AVMEDIA_TYPE_VIDEO) {
-        enc_ctx->time_base      = frame->time_base;
-        enc_ctx->framerate      = fd->frame_rate_filter;
-        ost->st->avg_frame_rate = fd->frame_rate_filter;
-    }
-
-    switch (enc_ctx->codec_type) {
-    case AVMEDIA_TYPE_AUDIO:
-        enc_ctx->sample_fmt     = frame->format;
-        enc_ctx->sample_rate    = frame->sample_rate;
-        ret = av_channel_layout_copy(&enc_ctx->ch_layout, &frame->ch_layout);
-        if (ret < 0)
-            return ret;
-
-        if (ost->bits_per_raw_sample)
-            enc_ctx->bits_per_raw_sample = ost->bits_per_raw_sample;
-        else
-            enc_ctx->bits_per_raw_sample = FFMIN(fd->bits_per_raw_sample,
-                                                 av_get_bytes_per_sample(enc_ctx->sample_fmt) << 3);
-        break;
-
-    case AVMEDIA_TYPE_VIDEO: {
-        enc_ctx->width  = frame->width;
-        enc_ctx->height = frame->height;
-        enc_ctx->sample_aspect_ratio = ost->st->sample_aspect_ratio =
-            ost->frame_aspect_ratio.num ? // overridden by the -aspect cli option
-            av_mul_q(ost->frame_aspect_ratio, (AVRational){ enc_ctx->height, enc_ctx->width }) :
-            frame->sample_aspect_ratio;
-
-        enc_ctx->pix_fmt = frame->format;
-
-        if (ost->bits_per_raw_sample)
-            enc_ctx->bits_per_raw_sample = ost->bits_per_raw_sample;
-        else
-            enc_ctx->bits_per_raw_sample = FFMIN(fd->bits_per_raw_sample,
-                                                 av_pix_fmt_desc_get(enc_ctx->pix_fmt)->comp[0].depth);
-
-        enc_ctx->color_range            = frame->color_range;
-        enc_ctx->color_primaries        = frame->color_primaries;
-        enc_ctx->color_trc              = frame->color_trc;
-        enc_ctx->colorspace             = frame->colorspace;
-        enc_ctx->chroma_sample_location = frame->chroma_location;
-
-        if (enc_ctx->flags & (AV_CODEC_FLAG_INTERLACED_DCT | AV_CODEC_FLAG_INTERLACED_ME) ||
-            (frame->flags & AV_FRAME_FLAG_INTERLACED)
-#if FFMPEG_OPT_TOP
-            || ost->top_field_first >= 0
-#endif
-            ) {
-            int top_field_first =
-#if FFMPEG_OPT_TOP
-                ost->top_field_first >= 0 ?
-                ost->top_field_first :
-#endif
-                !!(frame->flags & AV_FRAME_FLAG_TOP_FIELD_FIRST);
-
-            if (enc->id == AV_CODEC_ID_MJPEG)
-                enc_ctx->field_order = top_field_first ? AV_FIELD_TT : AV_FIELD_BB;
-            else
-                enc_ctx->field_order = top_field_first ? AV_FIELD_TB : AV_FIELD_BT;
-        } else
-            enc_ctx->field_order = AV_FIELD_PROGRESSIVE;
-
-        break;
-        }
-    case AVMEDIA_TYPE_SUBTITLE:
-        if (ost->enc_timebase.num)
-            av_log(ost, AV_LOG_WARNING,
-                   "-enc_time_base not supported for subtitles, ignoring\n");
-        enc_ctx->time_base = AV_TIME_BASE_Q;
-
-        if (!enc_ctx->width) {
-            enc_ctx->width     = ost->ist->par->width;
-            enc_ctx->height    = ost->ist->par->height;
-        }
-        if (dec_ctx && dec_ctx->subtitle_header) {
-            /* ASS code assumes this buffer is null terminated so add extra byte. */
-            enc_ctx->subtitle_header = av_mallocz(dec_ctx->subtitle_header_size + 1);
-            if (!enc_ctx->subtitle_header)
-                return AVERROR(ENOMEM);
-            memcpy(enc_ctx->subtitle_header, dec_ctx->subtitle_header,
-                   dec_ctx->subtitle_header_size);
-            enc_ctx->subtitle_header_size = dec_ctx->subtitle_header_size;
-        }
-
-        break;
-    default:
-        av_assert0(0);
-        break;
-    }
-
-    if (ost->bitexact)
-        enc_ctx->flags |= AV_CODEC_FLAG_BITEXACT;
-
-    if (!av_dict_get(ost->encoder_opts, "threads", NULL, 0))
-        av_dict_set(&ost->encoder_opts, "threads", "auto", 0);
-
-    if (enc->capabilities & AV_CODEC_CAP_ENCODER_REORDERED_OPAQUE) {
-        ret = av_dict_set(&ost->encoder_opts, "flags", "+copy_opaque", AV_DICT_MULTIKEY);
-        if (ret < 0)
-            return ret;
-    }
-
-    av_dict_set(&ost->encoder_opts, "flags", "+frame_duration", AV_DICT_MULTIKEY);
-
-    ret = hw_device_setup_for_encode(ost, frame ? frame->hw_frames_ctx : NULL);
-    if (ret < 0) {
-        av_log(ost, AV_LOG_ERROR,
-               "Encoding hardware device setup failed: %s\n", av_err2str(ret));
-        return ret;
-    }
-
-    if ((ret = avcodec_open2(ost->enc_ctx, enc, &ost->encoder_opts)) < 0) {
-        if (ret != AVERROR_EXPERIMENTAL)
-            av_log(ost, AV_LOG_ERROR, "Error while opening encoder - maybe "
-                   "incorrect parameters such as bit_rate, rate, width or height.\n");
-        return ret;
-    }
-
-    e->opened = 1;
-
-    if (ost->sq_idx_encode >= 0) {
-        e->sq_frame = av_frame_alloc();
-        if (!e->sq_frame)
-            return AVERROR(ENOMEM);
-    }
-
-    if (ost->enc_ctx->frame_size) {
-        av_assert0(ost->sq_idx_encode >= 0);
-        sq_frame_samples(output_files[ost->file_index]->sq_encode,
-                         ost->sq_idx_encode, ost->enc_ctx->frame_size);
-    }
-
-    ret = check_avoptions(ost->encoder_opts);
-    if (ret < 0)
-        return ret;
-
-    if (ost->enc_ctx->bit_rate && ost->enc_ctx->bit_rate < 1000 &&
-        ost->enc_ctx->codec_id != AV_CODEC_ID_CODEC2 /* don't complain about 700 bit/s modes */)
-        av_log(ost, AV_LOG_WARNING, "The bitrate parameter is set too low."
-                                    " It takes bits/s as argument, not kbits/s\n");
-
-    ret = avcodec_parameters_from_context(ost->par_in, ost->enc_ctx);
-    if (ret < 0) {
-        av_log(ost, AV_LOG_FATAL,
-               "Error initializing the output stream codec context.\n");
-        return ret;
-    }
-
-    /*
-     * Add global input side data. For now this is naive, and copies it
-     * from the input stream's global side data. All side data should
-     * really be funneled over AVFrame and libavfilter, then added back to
-     * packet side data, and then potentially using the first packet for
-     * global side data.
-     */
-    if (ist) {
-        int i;
-        for (i = 0; i < ist->st->codecpar->nb_coded_side_data; i++) {
-            AVPacketSideData *sd_src = &ist->st->codecpar->coded_side_data[i];
-            if (sd_src->type != AV_PKT_DATA_CPB_PROPERTIES) {
-                AVPacketSideData *sd_dst = av_packet_side_data_new(&ost->par_in->coded_side_data,
-                                                                   &ost->par_in->nb_coded_side_data,
-                                                                   sd_src->type, sd_src->size, 0);
-                if (!sd_dst)
-                    return AVERROR(ENOMEM);
-                memcpy(sd_dst->data, sd_src->data, sd_src->size);
-                if (ist->autorotate && sd_src->type == AV_PKT_DATA_DISPLAYMATRIX)
-                    av_display_rotation_set((int32_t *)sd_dst->data, 0);
-            }
-        }
-    }
-
-    // copy timebase while removing common factors
-    if (ost->st->time_base.num <= 0 || ost->st->time_base.den <= 0)
-        ost->st->time_base = av_add_q(ost->enc_ctx->time_base, (AVRational){0, 1});
-
-    ret = of_stream_init(of, ost);
-    if (ret < 0)
-        return ret;
-
-    return 0;
-}
-
-static int check_recording_time(OutputStream *ost, int64_t ts, AVRational tb)
-{
-    OutputFile *of = output_files[ost->file_index];
-
-    if (of->recording_time != INT64_MAX &&
-        av_compare_ts(ts, tb, of->recording_time, AV_TIME_BASE_Q) >= 0) {
-        close_output_stream(ost);
-        return 0;
-    }
-    return 1;
-}
-
-int enc_subtitle(OutputFile *of, OutputStream *ost, const AVSubtitle *sub)
-{
-    Encoder *e = ost->enc;
-    int subtitle_out_max_size = 1024 * 1024;
-    int subtitle_out_size, nb, i, ret;
-    AVCodecContext *enc;
-    AVPacket *pkt = e->pkt;
-    int64_t pts;
-
-    if (sub->pts == AV_NOPTS_VALUE) {
-        av_log(ost, AV_LOG_ERROR, "Subtitle packets must have a pts\n");
-        return exit_on_error ? AVERROR(EINVAL) : 0;
-    }
-    if (ost->finished ||
-        (of->start_time != AV_NOPTS_VALUE && sub->pts < of->start_time))
-        return 0;
-
-    enc = ost->enc_ctx;
-
-    /* Note: DVB subtitle need one packet to draw them and one other
-       packet to clear them */
-    /* XXX: signal it in the codec context ? */
-    if (enc->codec_id == AV_CODEC_ID_DVB_SUBTITLE)
-        nb = 2;
-    else if (enc->codec_id == AV_CODEC_ID_ASS)
-        nb = FFMAX(sub->num_rects, 1);
-    else
-        nb = 1;
-
-    /* shift timestamp to honor -ss and make check_recording_time() work with -t */
-    pts = sub->pts;
-    if (output_files[ost->file_index]->start_time != AV_NOPTS_VALUE)
-        pts -= output_files[ost->file_index]->start_time;
-    for (i = 0; i < nb; i++) {
-        AVSubtitle local_sub = *sub;
-
-        if (!check_recording_time(ost, pts, AV_TIME_BASE_Q))
-            return 0;
-
-        ret = av_new_packet(pkt, subtitle_out_max_size);
-        if (ret < 0)
-            return AVERROR(ENOMEM);
-
-        local_sub.pts = pts;
-        // start_display_time is required to be 0
-        local_sub.pts               += av_rescale_q(sub->start_display_time, (AVRational){ 1, 1000 }, AV_TIME_BASE_Q);
-        local_sub.end_display_time  -= sub->start_display_time;
-        local_sub.start_display_time = 0;
-
-        if (enc->codec_id == AV_CODEC_ID_DVB_SUBTITLE && i == 1)
-            local_sub.num_rects = 0;
-        else if (enc->codec_id == AV_CODEC_ID_ASS && sub->num_rects > 0) {
-            local_sub.num_rects = 1;
-            local_sub.rects += i;
-        }
-
-        ost->frames_encoded++;
-
-        subtitle_out_size = avcodec_encode_subtitle(enc, pkt->data, pkt->size, &local_sub);
-        if (subtitle_out_size < 0) {
-            av_log(ost, AV_LOG_FATAL, "Subtitle encoding failed\n");
-            return subtitle_out_size;
-        }
-
-        av_shrink_packet(pkt, subtitle_out_size);
-        pkt->time_base = AV_TIME_BASE_Q;
-        pkt->pts       = sub->pts;
-        pkt->duration = av_rescale_q(sub->end_display_time, (AVRational){ 1, 1000 }, pkt->time_base);
-        if (enc->codec_id == AV_CODEC_ID_DVB_SUBTITLE) {
-            /* XXX: the pts correction is handled here. Maybe handling
-               it in the codec would be better */
-            if (i == 0)
-                pkt->pts += av_rescale_q(sub->start_display_time, (AVRational){ 1, 1000 }, pkt->time_base);
-            else
-                pkt->pts += av_rescale_q(sub->end_display_time, (AVRational){ 1, 1000 }, pkt->time_base);
-        }
-        pkt->dts = pkt->pts;
-
-        ret = of_output_packet(of, ost, pkt);
-        if (ret < 0)
-            return ret;
-    }
-
-    return 0;
-}
-
-void enc_stats_write(OutputStream *ost, EncStats *es,
-                     const AVFrame *frame, const AVPacket *pkt,
-                     uint64_t frame_num)
-{
-    Encoder      *e = ost->enc;
-    AVIOContext *io = es->io;
-    AVRational   tb = frame ? frame->time_base : pkt->time_base;
-    int64_t     pts = frame ? frame->pts : pkt->pts;
-
-    AVRational  tbi = (AVRational){ 0, 1};
-    int64_t    ptsi = INT64_MAX;
-
-    const FrameData *fd;
-
-    if ((frame && frame->opaque_ref) || (pkt && pkt->opaque_ref)) {
-        fd   = (const FrameData*)(frame ? frame->opaque_ref->data : pkt->opaque_ref->data);
-        tbi  = fd->dec.tb;
-        ptsi = fd->dec.pts;
-    }
-
-    for (size_t i = 0; i < es->nb_components; i++) {
-        const EncStatsComponent *c = &es->components[i];
-
-        switch (c->type) {
-        case ENC_STATS_LITERAL:         avio_write (io, c->str,     c->str_len);                    continue;
-        case ENC_STATS_FILE_IDX:        avio_printf(io, "%d",       ost->file_index);               continue;
-        case ENC_STATS_STREAM_IDX:      avio_printf(io, "%d",       ost->index);                    continue;
-        case ENC_STATS_TIMEBASE:        avio_printf(io, "%d/%d",    tb.num, tb.den);                continue;
-        case ENC_STATS_TIMEBASE_IN:     avio_printf(io, "%d/%d",    tbi.num, tbi.den);              continue;
-        case ENC_STATS_PTS:             avio_printf(io, "%"PRId64,  pts);                           continue;
-        case ENC_STATS_PTS_IN:          avio_printf(io, "%"PRId64,  ptsi);                          continue;
-        case ENC_STATS_PTS_TIME:        avio_printf(io, "%g",       pts * av_q2d(tb));              continue;
-        case ENC_STATS_PTS_TIME_IN:     avio_printf(io, "%g",       ptsi == INT64_MAX ?
-                                                                    INFINITY : ptsi * av_q2d(tbi)); continue;
-        case ENC_STATS_FRAME_NUM:       avio_printf(io, "%"PRIu64,  frame_num);                     continue;
-        case ENC_STATS_FRAME_NUM_IN:    avio_printf(io, "%"PRIu64,  fd ? fd->dec.frame_num : -1);   continue;
-        }
-
-        if (frame) {
-            switch (c->type) {
-            case ENC_STATS_SAMPLE_NUM:  avio_printf(io, "%"PRIu64,  ost->samples_encoded);          continue;
-            case ENC_STATS_NB_SAMPLES:  avio_printf(io, "%d",       frame->nb_samples);             continue;
-            default: av_assert0(0);
-            }
-        } else {
-            switch (c->type) {
-            case ENC_STATS_DTS:         avio_printf(io, "%"PRId64,  pkt->dts);                      continue;
-            case ENC_STATS_DTS_TIME:    avio_printf(io, "%g",       pkt->dts * av_q2d(tb));         continue;
-            case ENC_STATS_PKT_SIZE:    avio_printf(io, "%d",       pkt->size);                     continue;
-            case ENC_STATS_BITRATE: {
-                double duration = FFMAX(pkt->duration, 1) * av_q2d(tb);
-                avio_printf(io, "%g",  8.0 * pkt->size / duration);
-                continue;
-            }
-            case ENC_STATS_AVG_BITRATE: {
-                double duration = pkt->dts * av_q2d(tb);
-                avio_printf(io, "%g",  duration > 0 ? 8.0 * e->data_size / duration : -1.);
-                continue;
-            }
-            default: av_assert0(0);
-            }
-        }
-    }
-    avio_w8(io, '\n');
-    avio_flush(io);
-}
-
-static inline double psnr(double d)
-{
-    return -10.0 * log10(d);
-}
-
-static int update_video_stats(OutputStream *ost, const AVPacket *pkt, int write_vstats)
-{
-    Encoder        *e = ost->enc;
-    const uint8_t *sd = av_packet_get_side_data(pkt, AV_PKT_DATA_QUALITY_STATS,
-                                                NULL);
-    AVCodecContext *enc = ost->enc_ctx;
-    enum AVPictureType pict_type;
-    int64_t frame_number;
-    double ti1, bitrate, avg_bitrate;
-    double psnr_val = -1;
-
-    ost->quality   = sd ? AV_RL32(sd) : -1;
-    pict_type      = sd ? sd[4] : AV_PICTURE_TYPE_NONE;
-
-    if ((enc->flags & AV_CODEC_FLAG_PSNR) && sd && sd[5]) {
-        // FIXME the scaling assumes 8bit
-        double error = AV_RL64(sd + 8) / (enc->width * enc->height * 255.0 * 255.0);
-        if (error >= 0 && error <= 1)
-            psnr_val = psnr(error);
-    }
-
-    if (!write_vstats)
-        return 0;
-
-    /* this is executed just the first time update_video_stats is called */
-    if (!vstats_file) {
-        vstats_file = fopen(vstats_filename, "w");
-        if (!vstats_file) {
-            perror("fopen");
-            return AVERROR(errno);
-        }
-    }
-
-    frame_number = e->packets_encoded;
-    if (vstats_version <= 1) {
-        fprintf(vstats_file, "frame= %5"PRId64" q= %2.1f ", frame_number,
-                ost->quality / (float)FF_QP2LAMBDA);
-    } else  {
-        fprintf(vstats_file, "out= %2d st= %2d frame= %5"PRId64" q= %2.1f ", ost->file_index, ost->index, frame_number,
-                ost->quality / (float)FF_QP2LAMBDA);
-    }
-
-    if (psnr_val >= 0)
-        fprintf(vstats_file, "PSNR= %6.2f ", psnr_val);
-
-    fprintf(vstats_file,"f_size= %6d ", pkt->size);
-    /* compute pts value */
-    ti1 = pkt->dts * av_q2d(pkt->time_base);
-    if (ti1 < 0.01)
-        ti1 = 0.01;
-
-    bitrate     = (pkt->size * 8) / av_q2d(enc->time_base) / 1000.0;
-    avg_bitrate = (double)(e->data_size * 8) / ti1 / 1000.0;
-    fprintf(vstats_file, "s_size= %8.0fkB time= %0.3f br= %7.1fkbits/s avg_br= %7.1fkbits/s ",
-           (double)e->data_size / 1024, ti1, bitrate, avg_bitrate);
-    fprintf(vstats_file, "type= %c\n", av_get_picture_type_char(pict_type));
-
-    return 0;
-}
-
-static int encode_frame(OutputFile *of, OutputStream *ost, AVFrame *frame)
-{
-    Encoder            *e = ost->enc;
-    AVCodecContext   *enc = ost->enc_ctx;
-    AVPacket         *pkt = e->pkt;
-    const char *type_desc = av_get_media_type_string(enc->codec_type);
-    const char    *action = frame ? "encode" : "flush";
-    int ret;
-
-    if (frame) {
-        if (ost->enc_stats_pre.io)
-            enc_stats_write(ost, &ost->enc_stats_pre, frame, NULL,
-                            ost->frames_encoded);
-
-        ost->frames_encoded++;
-        ost->samples_encoded += frame->nb_samples;
-
-        if (debug_ts) {
-            av_log(ost, AV_LOG_INFO, "encoder <- type:%s "
-                   "frame_pts:%s frame_pts_time:%s time_base:%d/%d\n",
-                   type_desc,
-                   av_ts2str(frame->pts), av_ts2timestr(frame->pts, &enc->time_base),
-                   enc->time_base.num, enc->time_base.den);
-        }
-
-        if (frame->sample_aspect_ratio.num && !ost->frame_aspect_ratio.num)
-            enc->sample_aspect_ratio = frame->sample_aspect_ratio;
-    }
-
-    update_benchmark(NULL);
-
-    ret = avcodec_send_frame(enc, frame);
-    if (ret < 0 && !(ret == AVERROR_EOF && !frame)) {
-        av_log(ost, AV_LOG_ERROR, "Error submitting %s frame to the encoder\n",
-               type_desc);
-        return ret;
-    }
-
-    while (1) {
-        av_packet_unref(pkt);
-
-        ret = avcodec_receive_packet(enc, pkt);
-        update_benchmark("%s_%s %d.%d", action, type_desc,
-                         ost->file_index, ost->index);
-
-        pkt->time_base = enc->time_base;
-
-        /* if two pass, output log on success and EOF */
-        if ((ret >= 0 || ret == AVERROR_EOF) && ost->logfile && enc->stats_out)
-            fprintf(ost->logfile, "%s", enc->stats_out);
-
-        if (ret == AVERROR(EAGAIN)) {
-            av_assert0(frame); // should never happen during flushing
-            return 0;
-        } else if (ret == AVERROR_EOF) {
-            ret = of_output_packet(of, ost, NULL);
-            return ret < 0 ? ret : AVERROR_EOF;
-        } else if (ret < 0) {
-            av_log(ost, AV_LOG_ERROR, "%s encoding failed\n", type_desc);
-            return ret;
-        }
-
-        if (enc->codec_type == AVMEDIA_TYPE_VIDEO) {
-            ret = update_video_stats(ost, pkt, !!vstats_filename);
-            if (ret < 0)
-                return ret;
-        }
-
-        if (ost->enc_stats_post.io)
-            enc_stats_write(ost, &ost->enc_stats_post, NULL, pkt,
-                            e->packets_encoded);
-
-        if (debug_ts) {
-            av_log(ost, AV_LOG_INFO, "encoder -> type:%s "
-                   "pkt_pts:%s pkt_pts_time:%s pkt_dts:%s pkt_dts_time:%s "
-                   "duration:%s duration_time:%s\n",
-                   type_desc,
-                   av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, &enc->time_base),
-                   av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, &enc->time_base),
-                   av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, &enc->time_base));
-        }
-
-        if ((ret = trigger_fix_sub_duration_heartbeat(ost, pkt)) < 0) {
-            av_log(NULL, AV_LOG_ERROR,
-                   "Subtitle heartbeat logic failed in %s! (%s)\n",
-                   __func__, av_err2str(ret));
-            return ret;
-        }
-
-        e->data_size += pkt->size;
-
-        e->packets_encoded++;
-
-        ret = of_output_packet(of, ost, pkt);
-        if (ret < 0)
-            return ret;
-    }
-
-    av_assert0(0);
-}
-
-static int submit_encode_frame(OutputFile *of, OutputStream *ost,
-                               AVFrame *frame)
-{
-    Encoder *e = ost->enc;
-    int ret;
-
-    if (ost->sq_idx_encode < 0)
-        return encode_frame(of, ost, frame);
-
-    if (frame) {
-        ret = av_frame_ref(e->sq_frame, frame);
-        if (ret < 0)
-            return ret;
-        frame = e->sq_frame;
-    }
-
-    ret = sq_send(of->sq_encode, ost->sq_idx_encode,
-                  SQFRAME(frame));
-    if (ret < 0) {
-        if (frame)
-            av_frame_unref(frame);
-        if (ret != AVERROR_EOF)
-            return ret;
-    }
-
-    while (1) {
-        AVFrame *enc_frame = e->sq_frame;
-
-        ret = sq_receive(of->sq_encode, ost->sq_idx_encode,
-                               SQFRAME(enc_frame));
-        if (ret == AVERROR_EOF) {
-            enc_frame = NULL;
-        } else if (ret < 0) {
-            return (ret == AVERROR(EAGAIN)) ? 0 : ret;
-        }
-
-        ret = encode_frame(of, ost, enc_frame);
-        if (enc_frame)
-            av_frame_unref(enc_frame);
-        if (ret < 0) {
-            if (ret == AVERROR_EOF)
-                close_output_stream(ost);
-            return ret;
-        }
-    }
-}
-
-static int do_audio_out(OutputFile *of, OutputStream *ost,
-                        AVFrame *frame)
-{
-    AVCodecContext *enc = ost->enc_ctx;
-    int ret;
-
-    if (!(enc->codec->capabilities & AV_CODEC_CAP_PARAM_CHANGE) &&
-        enc->ch_layout.nb_channels != frame->ch_layout.nb_channels) {
-        av_log(ost, AV_LOG_ERROR,
-               "Audio channel count changed and encoder does not support parameter changes\n");
-        return 0;
-    }
-
-    if (!check_recording_time(ost, frame->pts, frame->time_base))
-        return 0;
-
-    ret = submit_encode_frame(of, ost, frame);
-    return (ret < 0 && ret != AVERROR_EOF) ? ret : 0;
-}
-
-static enum AVPictureType forced_kf_apply(void *logctx, KeyframeForceCtx *kf,
-                                          AVRational tb, const AVFrame *in_picture)
-{
-    double pts_time;
-
-    if (kf->ref_pts == AV_NOPTS_VALUE)
-        kf->ref_pts = in_picture->pts;
-
-    pts_time = (in_picture->pts - kf->ref_pts) * av_q2d(tb);
-    if (kf->index < kf->nb_pts &&
-        av_compare_ts(in_picture->pts, tb, kf->pts[kf->index], AV_TIME_BASE_Q) >= 0) {
-        kf->index++;
-        goto force_keyframe;
-    } else if (kf->pexpr) {
-        double res;
-        kf->expr_const_values[FKF_T] = pts_time;
-        res = av_expr_eval(kf->pexpr,
-                           kf->expr_const_values, NULL);
-        av_log(logctx, AV_LOG_TRACE,
-               "force_key_frame: n:%f n_forced:%f prev_forced_n:%f t:%f prev_forced_t:%f -> res:%f\n",
-               kf->expr_const_values[FKF_N],
-               kf->expr_const_values[FKF_N_FORCED],
-               kf->expr_const_values[FKF_PREV_FORCED_N],
-               kf->expr_const_values[FKF_T],
-               kf->expr_const_values[FKF_PREV_FORCED_T],
-               res);
-
-        kf->expr_const_values[FKF_N] += 1;
-
-        if (res) {
-            kf->expr_const_values[FKF_PREV_FORCED_N] = kf->expr_const_values[FKF_N] - 1;
-            kf->expr_const_values[FKF_PREV_FORCED_T] = kf->expr_const_values[FKF_T];
-            kf->expr_const_values[FKF_N_FORCED]     += 1;
-            goto force_keyframe;
-        }
-    } else if (kf->type == KF_FORCE_SOURCE && (in_picture->flags & AV_FRAME_FLAG_KEY)) {
-        goto force_keyframe;
-    }
-
-    return AV_PICTURE_TYPE_NONE;
-
-force_keyframe:
-    av_log(logctx, AV_LOG_DEBUG, "Forced keyframe at time %f\n", pts_time);
-    return AV_PICTURE_TYPE_I;
-}
-
-/* May modify/reset frame */
-static int do_video_out(OutputFile *of, OutputStream *ost, AVFrame *in_picture)
-{
-    int ret;
-    AVCodecContext *enc = ost->enc_ctx;
-
-    if (!check_recording_time(ost, in_picture->pts, ost->enc_ctx->time_base))
-        return 0;
-
-    in_picture->quality = enc->global_quality;
-    in_picture->pict_type = forced_kf_apply(ost, &ost->kf, enc->time_base, in_picture);
-
-#if FFMPEG_OPT_TOP
-    if (ost->top_field_first >= 0) {
-        in_picture->flags &= ~AV_FRAME_FLAG_TOP_FIELD_FIRST;
-        in_picture->flags |= AV_FRAME_FLAG_TOP_FIELD_FIRST * (!!ost->top_field_first);
-    }
-#endif
-
-    ret = submit_encode_frame(of, ost, in_picture);
-    return (ret == AVERROR_EOF) ? 0 : ret;
-}
-
-int enc_frame(OutputStream *ost, AVFrame *frame)
-{
-    OutputFile *of = output_files[ost->file_index];
-    int ret;
-
-    ret = enc_open(ost, frame);
-    if (ret < 0)
-        return ret;
-
-    return ost->enc_ctx->codec_type == AVMEDIA_TYPE_VIDEO ?
-           do_video_out(of, ost, frame) : do_audio_out(of, ost, frame);
-}
-
-int enc_flush(void)
-{
-    int ret;
-
-    for (OutputStream *ost = ost_iter(NULL); ost; ost = ost_iter(ost)) {
-        OutputFile      *of = output_files[ost->file_index];
-        if (ost->sq_idx_encode >= 0)
-            sq_send(of->sq_encode, ost->sq_idx_encode, SQFRAME(NULL));
-    }
-
-    for (OutputStream *ost = ost_iter(NULL); ost; ost = ost_iter(ost)) {
-        Encoder          *e = ost->enc;
-        AVCodecContext *enc = ost->enc_ctx;
-        OutputFile      *of = output_files[ost->file_index];
-
-        if (!enc || !e->opened ||
-            (enc->codec_type != AVMEDIA_TYPE_VIDEO && enc->codec_type != AVMEDIA_TYPE_AUDIO))
-            continue;
-
-        ret = submit_encode_frame(of, ost, NULL);
-        if (ret != AVERROR_EOF)
-            return ret;
-    }
-
-    return 0;
-}
--- a/fftools/ffmpeg_filter.c
+++ b/fftools/ffmpeg_filter.c
--- a/fftools/ffmpeg_hw.c
+++ b/fftools/ffmpeg_hw.c
@@ -27,7 +27,7 @@
 static int nb_hw_devices;
 static HWDevice **hw_devices;

-HWDevice *hw_device_get_by_type(enum AVHWDeviceType type)
+static HWDevice *hw_device_get_by_type(enum AVHWDeviceType type)
 {
    HWDevice *found = NULL;
    int i;
@@ -93,8 +93,6 @@ static char *hw_device_default_name(enum AVHWDeviceType type)

 int hw_device_init_from_string(const char *arg, HWDevice **dev_out)
 {
-    // "type=name"
-    // "type=name,key=value,key2=value2"
    // "type=name:device,key=value,key2=value2"
    // "type:device,key=value,key2=value2"
    // -> av_hwdevice_ctx_create()
@@ -126,7 +124,7 @@ int hw_device_init_from_string(const char *arg, HWDevice **dev_out)
    }

    if (*p == '=') {
-        k = strcspn(p + 1, ":@,");
+        k = strcspn(p + 1, ":@");

        name = av_strndup(p + 1, k);
        if (!name) {
@@ -192,18 +190,6 @@ int hw_device_init_from_string(const char *arg, HWDevice **dev_out)
                                             src->device_ref, 0);
        if (err < 0)
            goto fail;
-    } else if (*p == ',') {
-        err = av_dict_parse_string(&options, p + 1, "=", ",", 0);
-
-        if (err < 0) {
-            errmsg = "failed to parse options";
-            goto invalid;
-        }
-
-        err = av_hwdevice_ctx_create(&device_ref, type,
-                                     NULL, options, 0);
-        if (err < 0)
-            goto fail;
    } else {
        errmsg = "parse error";
        goto invalid;
@@ -242,9 +228,9 @@ fail:
    goto done;
 }

-int hw_device_init_from_type(enum AVHWDeviceType type,
-                             const char *device,
-                             HWDevice **dev_out)
+static int hw_device_init_from_type(enum AVHWDeviceType type,
+                                    const char *device,
+                                    HWDevice **dev_out)
 {
    AVBufferRef *device_ref = NULL;
    HWDevice *dev;
@@ -297,7 +283,195 @@ void hw_device_free_all(void)
    nb_hw_devices = 0;
 }

-int hwaccel_retrieve_data(AVCodecContext *avctx, AVFrame *input)
+static HWDevice *hw_device_match_by_codec(const AVCodec *codec)
+{
+    const AVCodecHWConfig *config;
+    HWDevice *dev;
+    int i;
+    for (i = 0;; i++) {
+        config = avcodec_get_hw_config(codec, i);
+        if (!config)
+            return NULL;
+        if (!(config->methods & AV_CODEC_HW_CONFIG_METHOD_HW_DEVICE_CTX))
+            continue;
+        dev = hw_device_get_by_type(config->device_type);
+        if (dev)
+            return dev;
+    }
+}
+
+int hw_device_setup_for_decode(InputStream *ist)
+{
+    const AVCodecHWConfig *config;
+    enum AVHWDeviceType type;
+    HWDevice *dev = NULL;
+    int err, auto_device = 0;
+
+    if (ist->hwaccel_device) {
+        dev = hw_device_get_by_name(ist->hwaccel_device);
+        if (!dev) {
+            if (ist->hwaccel_id == HWACCEL_AUTO) {
+                auto_device = 1;
+            } else if (ist->hwaccel_id == HWACCEL_GENERIC) {
+                type = ist->hwaccel_device_type;
+                err = hw_device_init_from_type(type, ist->hwaccel_device,
+                                               &dev);
+            } else {
+                // This will be dealt with by API-specific initialisation
+                // (using hwaccel_device), so nothing further needed here.
+                return 0;
+            }
+        } else {
+            if (ist->hwaccel_id == HWACCEL_AUTO) {
+                ist->hwaccel_device_type = dev->type;
+            } else if (ist->hwaccel_device_type != dev->type) {
+                av_log(ist->dec_ctx, AV_LOG_ERROR, "Invalid hwaccel device "
+                       "specified for decoder: device %s of type %s is not "
+                       "usable with hwaccel %s.\n", dev->name,
+                       av_hwdevice_get_type_name(dev->type),
+                       av_hwdevice_get_type_name(ist->hwaccel_device_type));
+                return AVERROR(EINVAL);
+            }
+        }
+    } else {
+        if (ist->hwaccel_id == HWACCEL_AUTO) {
+            auto_device = 1;
+        } else if (ist->hwaccel_id == HWACCEL_GENERIC) {
+            type = ist->hwaccel_device_type;
+            dev = hw_device_get_by_type(type);
+            if (!dev)
+                err = hw_device_init_from_type(type, NULL, &dev);
+        } else {
+            dev = hw_device_match_by_codec(ist->dec);
+            if (!dev) {
+                // No device for this codec, but not using generic hwaccel
+                // and therefore may well not need one - ignore.
+                return 0;
+            }
+        }
+    }
+
+    if (auto_device) {
+        int i;
+        if (!avcodec_get_hw_config(ist->dec, 0)) {
+            // Decoder does not support any hardware devices.
+            return 0;
+        }
+        for (i = 0; !dev; i++) {
+            config = avcodec_get_hw_config(ist->dec, i);
+            if (!config)
+                break;
+            type = config->device_type;
+            dev = hw_device_get_by_type(type);
+            if (dev) {
+                av_log(ist->dec_ctx, AV_LOG_INFO, "Using auto "
+                       "hwaccel type %s with existing device %s.\n",
+                       av_hwdevice_get_type_name(type), dev->name);
+            }
+        }
+        for (i = 0; !dev; i++) {
+            config = avcodec_get_hw_config(ist->dec, i);
+            if (!config)
+                break;
+            type = config->device_type;
+            // Try to make a new device of this type.
+            err = hw_device_init_from_type(type, ist->hwaccel_device,
+                                           &dev);
+            if (err < 0) {
+                // Can't make a device of this type.
+                continue;
+            }
+            if (ist->hwaccel_device) {
+                av_log(ist->dec_ctx, AV_LOG_INFO, "Using auto "
+                       "hwaccel type %s with new device created "
+                       "from %s.\n", av_hwdevice_get_type_name(type),
+                       ist->hwaccel_device);
+            } else {
+                av_log(ist->dec_ctx, AV_LOG_INFO, "Using auto "
+                       "hwaccel type %s with new default device.\n",
+                       av_hwdevice_get_type_name(type));
+            }
+        }
+        if (dev) {
+            ist->hwaccel_device_type = type;
+        } else {
+            av_log(ist->dec_ctx, AV_LOG_INFO, "Auto hwaccel "
+                   "disabled: no device found.\n");
+            ist->hwaccel_id = HWACCEL_NONE;
+            return 0;
+        }
+    }
+
+    if (!dev) {
+        av_log(ist->dec_ctx, AV_LOG_ERROR, "No device available "
+               "for decoder: device type %s needed for codec %s.\n",
+               av_hwdevice_get_type_name(type), ist->dec->name);
+        return err;
+    }
+
+    ist->dec_ctx->hw_device_ctx = av_buffer_ref(dev->device_ref);
+    if (!ist->dec_ctx->hw_device_ctx)
+        return AVERROR(ENOMEM);
+
+    return 0;
+}
+
+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);
+    }
+
+    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);
+    } else {
+        // No device required, or no device available.
+    }
+    return 0;
+}
+
+static int hwaccel_retrieve_data(AVCodecContext *avctx, AVFrame *input)
 {
    InputStream *ist = avctx->opaque;
    AVFrame *output = NULL;
@@ -339,25 +513,39 @@ fail:
    return err;
 }

-AVBufferRef *hw_device_for_filter(void)
+int hwaccel_decode_init(AVCodecContext *avctx)
 {
-    // Pick the last hardware device if the user doesn't pick the device for
-    // filters explicitly with the filter_hw_device option.
+    InputStream *ist = avctx->opaque;
+
+    ist->hwaccel_retrieve_data = &hwaccel_retrieve_data;
+
+    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)
-        return filter_hw_device->device_ref;
-    else if (nb_hw_devices > 0) {
-        HWDevice *dev = hw_devices[nb_hw_devices - 1];
+        dev = filter_hw_device;
+    else if (nb_hw_devices == 1)
+        dev = hw_devices[0];
+    else
+        dev = NULL;

-        if (nb_hw_devices > 1)
-            av_log(NULL, AV_LOG_WARNING, "There are %d hardware devices. device "
-                   "%s of type %s is picked for filters by default. Set hardware "
-                   "device explicitly with the filter_hw_device option if device "
-                   "%s is not usable for filters.\n",
-                   nb_hw_devices, dev->name,
-                   av_hwdevice_get_type_name(dev->type), dev->name);
-
-        return dev->device_ref;
+    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 NULL;
+    return 0;
 }
--- a/fftools/ffmpeg_mux.c
+++ b/fftools/ffmpeg_mux.c
@@ -1,951 +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 <stdatomic.h>
-#include <stdio.h>
-#include <string.h>
-
-#include "ffmpeg.h"
-#include "ffmpeg_mux.h"
-#include "objpool.h"
-#include "sync_queue.h"
-#include "thread_queue.h"
-
-#include "libavutil/fifo.h"
-#include "libavutil/intreadwrite.h"
-#include "libavutil/log.h"
-#include "libavutil/mem.h"
-#include "libavutil/timestamp.h"
-#include "libavutil/thread.h"
-
-#include "libavcodec/packet.h"
-
-#include "libavformat/avformat.h"
-#include "libavformat/avio.h"
-
-int want_sdp = 1;
-
-static Muxer *mux_from_of(OutputFile *of)
-{
-    return (Muxer*)of;
-}
-
-static int64_t filesize(AVIOContext *pb)
-{
-    int64_t ret = -1;
-
-    if (pb) {
-        ret = avio_size(pb);
-        if (ret <= 0) // FIXME improve avio_size() so it works with non seekable output too
-            ret = avio_tell(pb);
-    }
-
-    return ret;
-}
-
-static int write_packet(Muxer *mux, OutputStream *ost, AVPacket *pkt)
-{
-    MuxStream *ms = ms_from_ost(ost);
-    AVFormatContext *s = mux->fc;
-    int64_t fs;
-    uint64_t frame_num;
-    int ret;
-
-    fs = filesize(s->pb);
-    atomic_store(&mux->last_filesize, fs);
-    if (fs >= mux->limit_filesize) {
-        ret = AVERROR_EOF;
-        goto fail;
-    }
-
-    if (ost->type == AVMEDIA_TYPE_VIDEO && ost->vsync_method == VSYNC_DROP)
-        pkt->pts = pkt->dts = AV_NOPTS_VALUE;
-
-    // rescale timestamps to the stream timebase
-    if (ost->type == AVMEDIA_TYPE_AUDIO && !ost->enc) {
-        // use av_rescale_delta() for streamcopying audio, to preserve
-        // accuracy with coarse input timebases
-        int duration = av_get_audio_frame_duration2(ost->st->codecpar, pkt->size);
-
-        if (!duration)
-            duration = ost->st->codecpar->frame_size;
-
-        pkt->dts = av_rescale_delta(pkt->time_base, pkt->dts,
-                                    (AVRational){1, ost->st->codecpar->sample_rate}, duration,
-                                    &ms->ts_rescale_delta_last, ost->st->time_base);
-        pkt->pts = pkt->dts;
-
-        pkt->duration = av_rescale_q(pkt->duration, pkt->time_base, ost->st->time_base);
-    } else
-        av_packet_rescale_ts(pkt, pkt->time_base, ost->st->time_base);
-    pkt->time_base = ost->st->time_base;
-
-    if (!(s->oformat->flags & AVFMT_NOTIMESTAMPS)) {
-        if (pkt->dts != AV_NOPTS_VALUE &&
-            pkt->pts != AV_NOPTS_VALUE &&
-            pkt->dts > pkt->pts) {
-            av_log(s, AV_LOG_WARNING, "Invalid DTS: %"PRId64" PTS: %"PRId64" in output stream %d:%d, replacing by guess\n",
-                   pkt->dts, pkt->pts,
-                   ost->file_index, ost->st->index);
-            pkt->pts =
-            pkt->dts = pkt->pts + pkt->dts + ms->last_mux_dts + 1
-                     - FFMIN3(pkt->pts, pkt->dts, ms->last_mux_dts + 1)
-                     - FFMAX3(pkt->pts, pkt->dts, ms->last_mux_dts + 1);
-        }
-        if ((ost->type == AVMEDIA_TYPE_AUDIO || ost->type == AVMEDIA_TYPE_VIDEO || ost->type == AVMEDIA_TYPE_SUBTITLE) &&
-            pkt->dts != AV_NOPTS_VALUE &&
-            ms->last_mux_dts != AV_NOPTS_VALUE) {
-            int64_t max = ms->last_mux_dts + !(s->oformat->flags & AVFMT_TS_NONSTRICT);
-            if (pkt->dts < max) {
-                int loglevel = max - pkt->dts > 2 || ost->type == AVMEDIA_TYPE_VIDEO ? AV_LOG_WARNING : AV_LOG_DEBUG;
-                if (exit_on_error)
-                    loglevel = AV_LOG_ERROR;
-                av_log(s, loglevel, "Non-monotonic DTS in output stream "
-                       "%d:%d; previous: %"PRId64", current: %"PRId64"; ",
-                       ost->file_index, ost->st->index, ms->last_mux_dts, pkt->dts);
-                if (exit_on_error) {
-                    ret = AVERROR(EINVAL);
-                    goto fail;
-                }
-
-                av_log(s, loglevel, "changing to %"PRId64". This may result "
-                       "in incorrect timestamps in the output file.\n",
-                       max);
-                if (pkt->pts >= pkt->dts)
-                    pkt->pts = FFMAX(pkt->pts, max);
-                pkt->dts = max;
-            }
-        }
-    }
-    ms->last_mux_dts = pkt->dts;
-
-    ms->data_size_mux += pkt->size;
-    frame_num = atomic_fetch_add(&ost->packets_written, 1);
-
-    pkt->stream_index = ost->index;
-
-    if (debug_ts) {
-        av_log(ost, AV_LOG_INFO, "muxer <- type:%s "
-                "pkt_pts:%s pkt_pts_time:%s pkt_dts:%s pkt_dts_time:%s duration:%s duration_time:%s size:%d\n",
-                av_get_media_type_string(ost->type),
-                av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, &ost->st->time_base),
-                av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, &ost->st->time_base),
-                av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, &ost->st->time_base),
-                pkt->size
-              );
-    }
-
-    if (ms->stats.io)
-        enc_stats_write(ost, &ms->stats, NULL, pkt, frame_num);
-
-    ret = av_interleaved_write_frame(s, pkt);
-    if (ret < 0) {
-        av_log(ost, AV_LOG_ERROR,
-               "Error submitting a packet to the muxer: %s\n",
-               av_err2str(ret));
-        goto fail;
-    }
-
-    return 0;
-fail:
-    av_packet_unref(pkt);
-    return ret;
-}
-
-static int sync_queue_process(Muxer *mux, OutputStream *ost, AVPacket *pkt, int *stream_eof)
-{
-    OutputFile *of = &mux->of;
-
-    if (ost->sq_idx_mux >= 0) {
-        int ret = sq_send(mux->sq_mux, ost->sq_idx_mux, SQPKT(pkt));
-        if (ret < 0) {
-            if (ret == AVERROR_EOF)
-                *stream_eof = 1;
-
-            return ret;
-        }
-
-        while (1) {
-            ret = sq_receive(mux->sq_mux, -1, SQPKT(mux->sq_pkt));
-            if (ret < 0) {
-                /* n.b.: We forward EOF from the sync queue, terminating muxing.
-                 * This assumes that if a muxing sync queue is present, then all
-                 * the streams use it. That is true currently, but may change in
-                 * the future, then this code needs to be revisited.
-                 */
-                return ret == AVERROR(EAGAIN) ? 0 : ret;
-            }
-
-            ret = write_packet(mux, of->streams[ret],
-                               mux->sq_pkt);
-            if (ret < 0)
-                return ret;
-        }
-    } else if (pkt)
-        return write_packet(mux, ost, pkt);
-
-    return 0;
-}
-
-static void thread_set_name(OutputFile *of)
-{
-    char name[16];
-    snprintf(name, sizeof(name), "mux%d:%s", of->index, of->format->name);
-    ff_thread_setname(name);
-}
-
-static void *muxer_thread(void *arg)
-{
-    Muxer     *mux = arg;
-    OutputFile *of = &mux->of;
-    AVPacket  *pkt = NULL;
-    int        ret = 0;
-
-    pkt = av_packet_alloc();
-    if (!pkt) {
-        ret = AVERROR(ENOMEM);
-        goto finish;
-    }
-
-    thread_set_name(of);
-
-    while (1) {
-        OutputStream *ost;
-        int stream_idx, stream_eof = 0;
-
-        ret = tq_receive(mux->tq, &stream_idx, pkt);
-        if (stream_idx < 0) {
-            av_log(mux, AV_LOG_VERBOSE, "All streams finished\n");
-            ret = 0;
-            break;
-        }
-
-        ost = of->streams[stream_idx];
-        ret = sync_queue_process(mux, ost, ret < 0 ? NULL : pkt, &stream_eof);
-        av_packet_unref(pkt);
-        if (ret == AVERROR_EOF) {
-            if (stream_eof) {
-                tq_receive_finish(mux->tq, stream_idx);
-            } else {
-                av_log(mux, AV_LOG_VERBOSE, "Muxer returned EOF\n");
-                ret = 0;
-                break;
-            }
-        } else if (ret < 0) {
-            av_log(mux, AV_LOG_ERROR, "Error muxing a packet\n");
-            break;
-        }
-    }
-
-finish:
-    av_packet_free(&pkt);
-
-    for (unsigned int i = 0; i < mux->fc->nb_streams; i++)
-        tq_receive_finish(mux->tq, i);
-
-    av_log(mux, AV_LOG_VERBOSE, "Terminating muxer thread\n");
-
-    return (void*)(intptr_t)ret;
-}
-
-static int thread_submit_packet(Muxer *mux, OutputStream *ost, AVPacket *pkt)
-{
-    int ret = 0;
-
-    if (!pkt || ost->finished & MUXER_FINISHED)
-        goto finish;
-
-    ret = tq_send(mux->tq, ost->index, pkt);
-    if (ret < 0)
-        goto finish;
-
-    return 0;
-
-finish:
-    if (pkt)
-        av_packet_unref(pkt);
-
-    ost->finished |= MUXER_FINISHED;
-    tq_send_finish(mux->tq, ost->index);
-    return ret == AVERROR_EOF ? 0 : ret;
-}
-
-static int queue_packet(OutputStream *ost, AVPacket *pkt)
-{
-    MuxStream *ms = ms_from_ost(ost);
-    AVPacket *tmp_pkt = NULL;
-    int ret;
-
-    if (!av_fifo_can_write(ms->muxing_queue)) {
-        size_t cur_size = av_fifo_can_read(ms->muxing_queue);
-        size_t pkt_size = pkt ? pkt->size : 0;
-        unsigned int are_we_over_size =
-            (ms->muxing_queue_data_size + pkt_size) > ms->muxing_queue_data_threshold;
-        size_t limit    = are_we_over_size ? ms->max_muxing_queue_size : SIZE_MAX;
-        size_t new_size = FFMIN(2 * cur_size, limit);
-
-        if (new_size <= cur_size) {
-            av_log(ost, AV_LOG_ERROR,
-                   "Too many packets buffered for output stream %d:%d.\n",
-                   ost->file_index, ost->st->index);
-            return AVERROR(ENOSPC);
-        }
-        ret = av_fifo_grow2(ms->muxing_queue, new_size - cur_size);
-        if (ret < 0)
-            return ret;
-    }
-
-    if (pkt) {
-        ret = av_packet_make_refcounted(pkt);
-        if (ret < 0)
-            return ret;
-
-        tmp_pkt = av_packet_alloc();
-        if (!tmp_pkt)
-            return AVERROR(ENOMEM);
-
-        av_packet_move_ref(tmp_pkt, pkt);
-        ms->muxing_queue_data_size += tmp_pkt->size;
-    }
-    av_fifo_write(ms->muxing_queue, &tmp_pkt, 1);
-
-    return 0;
-}
-
-static int submit_packet(Muxer *mux, AVPacket *pkt, OutputStream *ost)
-{
-    int ret;
-
-    if (mux->tq) {
-        return thread_submit_packet(mux, ost, pkt);
-    } else {
-        /* the muxer is not initialized yet, buffer the packet */
-        ret = queue_packet(ost, pkt);
-        if (ret < 0) {
-            if (pkt)
-                av_packet_unref(pkt);
-            return ret;
-        }
-    }
-
-    return 0;
-}
-
-int of_output_packet(OutputFile *of, OutputStream *ost, AVPacket *pkt)
-{
-    Muxer *mux = mux_from_of(of);
-    MuxStream *ms = ms_from_ost(ost);
-    const char *err_msg;
-    int ret = 0;
-
-    if (pkt && pkt->dts != AV_NOPTS_VALUE)
-        ost->last_mux_dts = av_rescale_q(pkt->dts, pkt->time_base, AV_TIME_BASE_Q);
-
-    /* apply the output bitstream filters */
-    if (ms->bsf_ctx) {
-        int bsf_eof = 0;
-
-        if (pkt)
-            av_packet_rescale_ts(pkt, pkt->time_base, ms->bsf_ctx->time_base_in);
-
-        ret = av_bsf_send_packet(ms->bsf_ctx, pkt);
-        if (ret < 0) {
-            err_msg = "submitting a packet for bitstream filtering";
-            goto fail;
-        }
-
-        while (!bsf_eof) {
-            ret = av_bsf_receive_packet(ms->bsf_ctx, ms->bsf_pkt);
-            if (ret == AVERROR(EAGAIN))
-                return 0;
-            else if (ret == AVERROR_EOF)
-                bsf_eof = 1;
-            else if (ret < 0) {
-                err_msg = "applying bitstream filters to a packet";
-                goto fail;
-            }
-
-            if (!bsf_eof)
-                ms->bsf_pkt->time_base = ms->bsf_ctx->time_base_out;
-
-            ret = submit_packet(mux, bsf_eof ? NULL : ms->bsf_pkt, ost);
-            if (ret < 0)
-                goto mux_fail;
-        }
-    } else {
-        ret = submit_packet(mux, pkt, ost);
-        if (ret < 0)
-            goto mux_fail;
-    }
-
-    return 0;
-
-mux_fail:
-    err_msg = "submitting a packet to the muxer";
-
-fail:
-    av_log(ost, AV_LOG_ERROR, "Error %s\n", err_msg);
-    return exit_on_error ? ret : 0;
-}
-
-int of_streamcopy(OutputStream *ost, const AVPacket *pkt, int64_t dts)
-{
-    OutputFile *of = output_files[ost->file_index];
-    MuxStream  *ms = ms_from_ost(ost);
-    int64_t start_time = (of->start_time == AV_NOPTS_VALUE) ? 0 : of->start_time;
-    int64_t ts_offset;
-    AVPacket *opkt = ms->pkt;
-    int ret;
-
-    av_packet_unref(opkt);
-
-    if (of->recording_time != INT64_MAX &&
-        dts >= of->recording_time + start_time)
-        pkt = NULL;
-
-    // EOF: flush output bitstream filters.
-    if (!pkt)
-        return of_output_packet(of, ost, NULL);
-
-    if (!ms->streamcopy_started && !(pkt->flags & AV_PKT_FLAG_KEY) &&
-        !ms->copy_initial_nonkeyframes)
-        return 0;
-
-    if (!ms->streamcopy_started) {
-        if (!ms->copy_prior_start &&
-            (pkt->pts == AV_NOPTS_VALUE ?
-             dts < ms->ts_copy_start :
-             pkt->pts < av_rescale_q(ms->ts_copy_start, AV_TIME_BASE_Q, pkt->time_base)))
-            return 0;
-
-        if (of->start_time != AV_NOPTS_VALUE && dts < of->start_time)
-            return 0;
-    }
-
-    ret = av_packet_ref(opkt, pkt);
-    if (ret < 0)
-        return ret;
-
-    ts_offset = av_rescale_q(start_time, AV_TIME_BASE_Q, opkt->time_base);
-
-    if (pkt->pts != AV_NOPTS_VALUE)
-        opkt->pts -= ts_offset;
-
-    if (pkt->dts == AV_NOPTS_VALUE) {
-        opkt->dts = av_rescale_q(dts, AV_TIME_BASE_Q, opkt->time_base);
-    } else if (ost->st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
-        opkt->pts = opkt->dts - ts_offset;
-    }
-    opkt->dts -= ts_offset;
-
-    {
-        int ret = trigger_fix_sub_duration_heartbeat(ost, pkt);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR,
-                   "Subtitle heartbeat logic failed in %s! (%s)\n",
-                   __func__, av_err2str(ret));
-            return ret;
-        }
-    }
-
-    ret = of_output_packet(of, ost, opkt);
-    if (ret < 0)
-        return ret;
-
-    ms->streamcopy_started = 1;
-
-    return 0;
-}
-
-static int thread_stop(Muxer *mux)
-{
-    void *ret;
-
-    if (!mux || !mux->tq)
-        return 0;
-
-    for (unsigned int i = 0; i < mux->fc->nb_streams; i++)
-        tq_send_finish(mux->tq, i);
-
-    pthread_join(mux->thread, &ret);
-
-    tq_free(&mux->tq);
-
-    return (int)(intptr_t)ret;
-}
-
-static int thread_start(Muxer *mux)
-{
-    AVFormatContext *fc = mux->fc;
-    ObjPool *op;
-    int ret;
-
-    op = objpool_alloc_packets();
-    if (!op)
-        return AVERROR(ENOMEM);
-
-    mux->tq = tq_alloc(fc->nb_streams, mux->thread_queue_size, op, pkt_move);
-    if (!mux->tq) {
-        objpool_free(&op);
-        return AVERROR(ENOMEM);
-    }
-
-    ret = pthread_create(&mux->thread, NULL, muxer_thread, (void*)mux);
-    if (ret) {
-        tq_free(&mux->tq);
-        return AVERROR(ret);
-    }
-
-    /* flush the muxing queues */
-    for (int i = 0; i < fc->nb_streams; i++) {
-        OutputStream *ost = mux->of.streams[i];
-        MuxStream     *ms = ms_from_ost(ost);
-        AVPacket *pkt;
-
-        while (av_fifo_read(ms->muxing_queue, &pkt, 1) >= 0) {
-            ret = thread_submit_packet(mux, ost, pkt);
-            if (pkt) {
-                ms->muxing_queue_data_size -= pkt->size;
-                av_packet_free(&pkt);
-            }
-            if (ret < 0)
-                return ret;
-        }
-    }
-
-    return 0;
-}
-
-static int print_sdp(void)
-{
-    char sdp[16384];
-    int i;
-    int j, ret;
-    AVIOContext *sdp_pb;
-    AVFormatContext **avc;
-
-    for (i = 0; i < nb_output_files; i++) {
-        if (!mux_from_of(output_files[i])->header_written)
-            return 0;
-    }
-
-    avc = av_malloc_array(nb_output_files, sizeof(*avc));
-    if (!avc)
-        return AVERROR(ENOMEM);
-    for (i = 0, j = 0; i < nb_output_files; i++) {
-        if (!strcmp(output_files[i]->format->name, "rtp")) {
-            avc[j] = mux_from_of(output_files[i])->fc;
-            j++;
-        }
-    }
-
-    if (!j) {
-        av_log(NULL, AV_LOG_ERROR, "No output streams in the SDP.\n");
-        ret = AVERROR(EINVAL);
-        goto fail;
-    }
-
-    ret = av_sdp_create(avc, j, sdp, sizeof(sdp));
-    if (ret < 0)
-        goto fail;
-
-    if (!sdp_filename) {
-        printf("SDP:\n%s\n", sdp);
-        fflush(stdout);
-    } else {
-        ret = avio_open2(&sdp_pb, sdp_filename, AVIO_FLAG_WRITE, &int_cb, NULL);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Failed to open sdp file '%s'\n", sdp_filename);
-            goto fail;
-        }
-
-        avio_print(sdp_pb, sdp);
-        avio_closep(&sdp_pb);
-        av_freep(&sdp_filename);
-    }
-
-    // SDP successfully written, allow muxer threads to start
-    ret = 1;
-
-fail:
-    av_freep(&avc);
-    return ret;
-}
-
-int mux_check_init(Muxer *mux)
-{
-    OutputFile *of = &mux->of;
-    AVFormatContext *fc = mux->fc;
-    int ret, i;
-
-    for (i = 0; i < fc->nb_streams; i++) {
-        OutputStream *ost = of->streams[i];
-        if (!ost->initialized)
-            return 0;
-    }
-
-    ret = avformat_write_header(fc, &mux->opts);
-    if (ret < 0) {
-        av_log(mux, AV_LOG_ERROR, "Could not write header (incorrect codec "
-               "parameters ?): %s\n", av_err2str(ret));
-        return ret;
-    }
-    //assert_avoptions(of->opts);
-    mux->header_written = 1;
-
-    av_dump_format(fc, of->index, fc->url, 1);
-    nb_output_dumped++;
-
-    if (sdp_filename || want_sdp) {
-        ret = print_sdp();
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Error writing the SDP.\n");
-            return ret;
-        } else if (ret == 1) {
-            /* SDP is written only after all the muxers are ready, so now we
-             * start ALL the threads */
-            for (i = 0; i < nb_output_files; i++) {
-                ret = thread_start(mux_from_of(output_files[i]));
-                if (ret < 0)
-                    return ret;
-            }
-        }
-    } else {
-        ret = thread_start(mux_from_of(of));
-        if (ret < 0)
-            return ret;
-    }
-
-    return 0;
-}
-
-static int bsf_init(MuxStream *ms)
-{
-    OutputStream *ost = &ms->ost;
-    AVBSFContext *ctx = ms->bsf_ctx;
-    int ret;
-
-    if (!ctx)
-        return avcodec_parameters_copy(ost->st->codecpar, ost->par_in);
-
-    ret = avcodec_parameters_copy(ctx->par_in, ost->par_in);
-    if (ret < 0)
-        return ret;
-
-    ctx->time_base_in = ost->st->time_base;
-
-    ret = av_bsf_init(ctx);
-    if (ret < 0) {
-        av_log(ms, AV_LOG_ERROR, "Error initializing bitstream filter: %s\n",
-               ctx->filter->name);
-        return ret;
-    }
-
-    ret = avcodec_parameters_copy(ost->st->codecpar, ctx->par_out);
-    if (ret < 0)
-        return ret;
-    ost->st->time_base = ctx->time_base_out;
-
-    ms->bsf_pkt = av_packet_alloc();
-    if (!ms->bsf_pkt)
-        return AVERROR(ENOMEM);
-
-    return 0;
-}
-
-int of_stream_init(OutputFile *of, OutputStream *ost)
-{
-    Muxer *mux = mux_from_of(of);
-    MuxStream *ms = ms_from_ost(ost);
-    int ret;
-
-    /* initialize bitstream filters for the output stream
-     * needs to be done here, because the codec id for streamcopy is not
-     * known until now */
-    ret = bsf_init(ms);
-    if (ret < 0)
-        return ret;
-
-    if (ms->stream_duration) {
-        ost->st->duration = av_rescale_q(ms->stream_duration, ms->stream_duration_tb,
-                                         ost->st->time_base);
-    }
-
-    ost->initialized = 1;
-
-    return mux_check_init(mux);
-}
-
-static int check_written(OutputFile *of)
-{
-    int64_t total_packets_written = 0;
-    int pass1_used = 1;
-    int ret = 0;
-
-    for (int i = 0; i < of->nb_streams; i++) {
-        OutputStream *ost = of->streams[i];
-        uint64_t packets_written = atomic_load(&ost->packets_written);
-
-        total_packets_written += packets_written;
-
-        if (ost->enc_ctx &&
-            (ost->enc_ctx->flags & (AV_CODEC_FLAG_PASS1 | AV_CODEC_FLAG_PASS2))
-             != AV_CODEC_FLAG_PASS1)
-            pass1_used = 0;
-
-        if (!packets_written &&
-            (abort_on_flags & ABORT_ON_FLAG_EMPTY_OUTPUT_STREAM)) {
-            av_log(ost, AV_LOG_FATAL, "Empty output stream\n");
-            ret = err_merge(ret, AVERROR(EINVAL));
-        }
-    }
-
-    if (!total_packets_written) {
-        int level = AV_LOG_WARNING;
-
-        if (abort_on_flags & ABORT_ON_FLAG_EMPTY_OUTPUT) {
-            ret = err_merge(ret, AVERROR(EINVAL));
-            level = AV_LOG_FATAL;
-        }
-
-        av_log(of, level, "Output file is empty, nothing was encoded%s\n",
-               pass1_used ? "" : "(check -ss / -t / -frames parameters if used)");
-    }
-
-    return ret;
-}
-
-static void mux_final_stats(Muxer *mux)
-{
-    OutputFile *of = &mux->of;
-    uint64_t total_packets = 0, total_size = 0;
-    uint64_t video_size = 0, audio_size = 0, subtitle_size = 0,
-             extra_size = 0, other_size = 0;
-
-    uint8_t overhead[16] = "unknown";
-    int64_t file_size = of_filesize(of);
-
-    av_log(of, AV_LOG_VERBOSE, "Output file #%d (%s):\n",
-           of->index, of->url);
-
-    for (int j = 0; j < of->nb_streams; j++) {
-        OutputStream *ost = of->streams[j];
-        MuxStream     *ms = ms_from_ost(ost);
-        const AVCodecParameters *par = ost->st->codecpar;
-        const  enum AVMediaType type = par->codec_type;
-        const uint64_t s = ms->data_size_mux;
-
-        switch (type) {
-        case AVMEDIA_TYPE_VIDEO:    video_size    += s; break;
-        case AVMEDIA_TYPE_AUDIO:    audio_size    += s; break;
-        case AVMEDIA_TYPE_SUBTITLE: subtitle_size += s; break;
-        default:                    other_size    += s; break;
-        }
-
-        extra_size    += par->extradata_size;
-        total_size    += s;
-        total_packets += atomic_load(&ost->packets_written);
-
-        av_log(of, AV_LOG_VERBOSE, "  Output stream #%d:%d (%s): ",
-               of->index, j, av_get_media_type_string(type));
-        if (ost->enc) {
-            av_log(of, AV_LOG_VERBOSE, "%"PRIu64" frames encoded",
-                   ost->frames_encoded);
-            if (type == AVMEDIA_TYPE_AUDIO)
-                av_log(of, AV_LOG_VERBOSE, " (%"PRIu64" samples)", ost->samples_encoded);
-            av_log(of, AV_LOG_VERBOSE, "; ");
-        }
-
-        av_log(of, AV_LOG_VERBOSE, "%"PRIu64" packets muxed (%"PRIu64" bytes); ",
-               atomic_load(&ost->packets_written), s);
-
-        av_log(of, AV_LOG_VERBOSE, "\n");
-    }
-
-    av_log(of, AV_LOG_VERBOSE, "  Total: %"PRIu64" packets (%"PRIu64" bytes) muxed\n",
-           total_packets, total_size);
-
-    if (total_size && file_size > 0 && file_size >= total_size) {
-        snprintf(overhead, sizeof(overhead), "%f%%",
-                 100.0 * (file_size - total_size) / total_size);
-    }
-
-    av_log(of, AV_LOG_INFO,
-           "video:%1.0fkB audio:%1.0fkB subtitle:%1.0fkB other streams:%1.0fkB "
-           "global headers:%1.0fkB muxing overhead: %s\n",
-           video_size    / 1024.0,
-           audio_size    / 1024.0,
-           subtitle_size / 1024.0,
-           other_size    / 1024.0,
-           extra_size    / 1024.0,
-           overhead);
-}
-
-int of_write_trailer(OutputFile *of)
-{
-    Muxer *mux = mux_from_of(of);
-    AVFormatContext *fc = mux->fc;
-    int ret, mux_result = 0;
-
-    if (!mux->tq) {
-        av_log(mux, AV_LOG_ERROR,
-               "Nothing was written into output file, because "
-               "at least one of its streams received no packets.\n");
-        return AVERROR(EINVAL);
-    }
-
-    mux_result = thread_stop(mux);
-
-    ret = av_write_trailer(fc);
-    if (ret < 0) {
-        av_log(mux, AV_LOG_ERROR, "Error writing trailer: %s\n", av_err2str(ret));
-        mux_result = err_merge(mux_result, ret);
-    }
-
-    mux->last_filesize = filesize(fc->pb);
-
-    if (!(of->format->flags & AVFMT_NOFILE)) {
-        ret = avio_closep(&fc->pb);
-        if (ret < 0) {
-            av_log(mux, AV_LOG_ERROR, "Error closing file: %s\n", av_err2str(ret));
-            mux_result = err_merge(mux_result, ret);
-        }
-    }
-
-    mux_final_stats(mux);
-
-    // check whether anything was actually written
-    ret = check_written(of);
-    mux_result = err_merge(mux_result, ret);
-
-    return mux_result;
-}
-
-static void ost_free(OutputStream **post)
-{
-    OutputStream *ost = *post;
-    MuxStream *ms;
-
-    if (!ost)
-        return;
-    ms = ms_from_ost(ost);
-
-    enc_free(&ost->enc);
-
-    if (ost->logfile) {
-        if (fclose(ost->logfile))
-            av_log(ms, AV_LOG_ERROR,
-                   "Error closing logfile, loss of information possible: %s\n",
-                   av_err2str(AVERROR(errno)));
-        ost->logfile = NULL;
-    }
-
-    if (ms->muxing_queue) {
-        AVPacket *pkt;
-        while (av_fifo_read(ms->muxing_queue, &pkt, 1) >= 0)
-            av_packet_free(&pkt);
-        av_fifo_freep2(&ms->muxing_queue);
-    }
-
-    avcodec_parameters_free(&ost->par_in);
-
-    av_bsf_free(&ms->bsf_ctx);
-    av_packet_free(&ms->bsf_pkt);
-
-    av_packet_free(&ms->pkt);
-    av_dict_free(&ost->encoder_opts);
-
-    av_freep(&ost->kf.pts);
-    av_expr_free(ost->kf.pexpr);
-
-    av_freep(&ost->logfile_prefix);
-    av_freep(&ost->apad);
-
-#if FFMPEG_OPT_MAP_CHANNEL
-    av_freep(&ost->audio_channels_map);
-    ost->audio_channels_mapped = 0;
-#endif
-
-    av_dict_free(&ost->sws_dict);
-    av_dict_free(&ost->swr_opts);
-
-    if (ost->enc_ctx)
-        av_freep(&ost->enc_ctx->stats_in);
-    avcodec_free_context(&ost->enc_ctx);
-
-    for (int i = 0; i < ost->enc_stats_pre.nb_components; i++)
-        av_freep(&ost->enc_stats_pre.components[i].str);
-    av_freep(&ost->enc_stats_pre.components);
-
-    for (int i = 0; i < ost->enc_stats_post.nb_components; i++)
-        av_freep(&ost->enc_stats_post.components[i].str);
-    av_freep(&ost->enc_stats_post.components);
-
-    for (int i = 0; i < ms->stats.nb_components; i++)
-        av_freep(&ms->stats.components[i].str);
-    av_freep(&ms->stats.components);
-
-    av_freep(post);
-}
-
-static void fc_close(AVFormatContext **pfc)
-{
-    AVFormatContext *fc = *pfc;
-
-    if (!fc)
-        return;
-
-    if (!(fc->oformat->flags & AVFMT_NOFILE))
-        avio_closep(&fc->pb);
-    avformat_free_context(fc);
-
-    *pfc = NULL;
-}
-
-void of_free(OutputFile **pof)
-{
-    OutputFile *of = *pof;
-    Muxer *mux;
-
-    if (!of)
-        return;
-    mux = mux_from_of(of);
-
-    thread_stop(mux);
-
-    sq_free(&of->sq_encode);
-    sq_free(&mux->sq_mux);
-
-    for (int i = 0; i < of->nb_streams; i++)
-        ost_free(&of->streams[i]);
-    av_freep(&of->streams);
-
-    av_dict_free(&mux->opts);
-
-    av_packet_free(&mux->sq_pkt);
-
-    fc_close(&mux->fc);
-
-    av_freep(pof);
-}
-
-int64_t of_filesize(OutputFile *of)
-{
-    Muxer *mux = mux_from_of(of);
-    return atomic_load(&mux->last_filesize);
-}
--- a/Show More
+++ b/Show More
@@ -1 +1 @@
 .1.git
 .4.6