RELEASE_NOTES: Based on the version from 4.3

Name suggested by Lynne, Gyan, Reto, Zane, Jan, Derek

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

.gitignore

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

.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>

Changelog

@@ -1,126 +1,7 @@
 Entries are sorted chronologically from oldest to youngest within each release,
 releases are sorted from youngest to oldest.
 
-version 6.0:
-- Radiance HDR image support
-- ddagrab (Desktop Duplication) video capture filter
-- ffmpeg -shortest_buf_duration option
-- ffmpeg now requires threading to be built
-- ffmpeg now runs every muxer in a separate thread
-- Add new mode to cropdetect filter to detect crop-area based on motion vectors and edges
-- VAAPI decoding and encoding for 10/12bit 422, 10/12bit 444 HEVC and VP9
-- WBMP (Wireless Application Protocol Bitmap) image format
-- a3dscope filter
-- bonk decoder and demuxer
-- Micronas SC-4 audio decoder
-- LAF demuxer
-- APAC decoder and demuxer
-- Media 100i decoders
-- DTS to PTS reorder bsf
-- ViewQuest VQC decoder
-- backgroundkey filter
-- nvenc AV1 encoding support
-- MediaCodec decoder via NDKMediaCodec
-- MediaCodec encoder
-- oneVPL support for QSV
-- QSV AV1 encoder
-- QSV decoding and encoding for 10/12bit 422, 10/12bit 444 HEVC and VP9
-- showcwt multimedia filter
-- corr video filter
-- adrc audio filter
-- afdelaysrc audio filter
-- WADY DPCM decoder and demuxer
-- CBD2 DPCM decoder
-- ssim360 video filter
-- ffmpeg CLI new options: -stats_enc_pre[_fmt], -stats_enc_post[_fmt],
-  -stats_mux_pre[_fmt]
-- hstack_vaapi, vstack_vaapi and xstack_vaapi filters
-- XMD ADPCM decoder and demuxer
-- media100 to mjpegb bsf
-- ffmpeg CLI new option: -fix_sub_duration_heartbeat
-- WavArc decoder and demuxer
-- CrystalHD decoders deprecated
-- SDNS demuxer
-- RKA decoder and demuxer
-- filtergraph syntax in ffmpeg CLI now supports passing file contents
-  as option values, by prefixing option name with '/'
-- hstack_qsv, vstack_qsv and xstack_qsv filters
-
-
-version 5.1:
-- add ipfs/ipns gateway support
-- dialogue enhance audio filter
-- dropped obsolete XvMC hwaccel
-- pcm-bluray encoder
-- DFPWM audio encoder/decoder and raw muxer/demuxer
-- SITI filter
-- Vizrt Binary Image encoder/decoder
-- avsynctest source filter
-- feedback video filter
-- pixelize video filter
-- colormap video filter
-- colorchart video source filter
-- multiply video filter
-- PGS subtitle frame merge bitstream filter
-- blurdetect filter
-- tiltshelf audio filter
-- QOI image format support
-- ffprobe -o option
-- virtualbass audio filter
-- VDPAU AV1 hwaccel
-- PHM image format support
-- remap_opencl filter
-- added chromakey_cuda filter
-- added bilateral_cuda filter
-
-
-version 5.0:
-- ADPCM IMA Westwood encoder
-- Westwood AUD muxer
-- ADPCM IMA Acorn Replay decoder
-- Argonaut Games CVG demuxer
-- Argonaut Games CVG muxer
-- Concatf protocol
-- afwtdn audio filter
-- audio and video segment filters
-- Apple Graphics (SMC) encoder
-- hsvkey and hsvhold video filters
-- adecorrelate audio filter
-- atilt audio filter
-- grayworld video filter
-- AV1 Low overhead bitstream format muxer
-- swscale slice threading
-- MSN Siren decoder
-- scharr video filter
-- apsyclip audio filter
-- morpho video filter
-- amr parser
-- (a)latency filters
-- GEM Raster image decoder
-- asdr audio filter
-- speex decoder
-- limitdiff video filter
-- xcorrelate video filter
-- varblur video filter
-- huesaturation video filter
-- colorspectrum source video filter
-- RTP packetizer for uncompressed video (RFC 4175)
-- bitpacked encoder
-- VideoToolbox VP9 hwaccel
-- VideoToolbox ProRes hwaccel
-- support loongarch.
-- aspectralstats audio filter
-- adynamicsmooth audio filter
-- libplacebo filter
-- vflip_vulkan, hflip_vulkan and flip_vulkan filters
-- adynamicequalizer audio filter
-- yadif_videotoolbox filter
-- VideoToolbox ProRes encoder
-- anlmf audio filter
-- IMF demuxer (experimental)
-
-
-version 4.4:
+version <next>:
 - AudioToolbox output device
 - MacCaption demuxer
 - PGX decoder

MAINTAINERS

@@ -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
@@ -187,7 +192,6 @@ Codecs:
   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
@@ -221,7 +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
@@ -261,14 +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
 
@@ -347,7 +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
@@ -393,7 +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
@@ -410,7 +414,6 @@ Muxers/Demuxers:
   dashdec.c                             Steven Liu
   dashenc.c                             Karthick Jeyapal
   daud.c                                Reimar Doeffinger
-  dfpwmdec.c                            Jack Bruienne
   dss.c                                 Oleksij Rempel
   dtsdec.c                              foo86
   dtshddec.c                            Paul B Mahol
@@ -427,12 +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_probe.*                        Leo Izen
   jvdec.c                               Peter Ross
   kvag.c                                Zane van Iperen
   libmodplug.c                          Clément Bœsch
@@ -512,7 +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
@@ -539,7 +539,6 @@ 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
@@ -610,22 +609,17 @@ Daniel Verkamp                78A6 07ED 782C 653E C628 B8B9 F0EB 8DD8 2F0E 21C7
 FFmpeg release signing key    FCF9 86EA 15E6 E293 A564 4F10 B432 2F04 D676 58D8
 Ganesh Ajjanagadde            C96A 848E 97C3 CEA2 AB72 5CE4 45F9 6A2D 3C36 FB1B
 Gwenole Beauchesne            2E63 B3A6 3E44 37E2 017D 2704 53C7 6266 B153 99C4
-Haihao Xiang (haihao)         1F0C 31E8 B4FE F7A4 4DC1 DC99 E0F5 76D4 76FC 437F
 Jaikrishnan Menon             61A1 F09F 01C9 2D45 78E1 C862 25DC 8831 AF70 D368
 James Almer                   7751 2E8C FD94 A169 57E6 9A7A 1463 01AD 7376 59E0
 Jean Delvare                  7CA6 9F44 60F1 BDC4 1FD2 C858 A552 6B9B B3CD 4E6A
-Leo Izen (thebombzen)         B6FD 3CFC 7ACF 83FC 9137 6945 5A71 C331 FD2F A19A
 Loren Merritt                 ABD9 08F4 C920 3F65 D8BE 35D7 1540 DAA7 060F 56DE
 Lynne                         FE50 139C 6805 72CA FD52 1F8D A2FE A5F0 3F03 4464
 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

Makefile

@@ -13,19 +13,17 @@ vpath %.v    $(SRC_PATH)
 vpath %.texi $(SRC_PATH)
 vpath %.cu   $(SRC_PATH)
 vpath %.ptx  $(SRC_PATH)
-vpath %.metal $(SRC_PATH)
 vpath %/fate_config.sh.template $(SRC_PATH)
 
 TESTTOOLS   = audiogen videogen rotozoom tiny_psnr tiny_ssim base64 audiomatch
 HOSTPROGS  := $(TESTTOOLS:%=tests/%) doc/print_options
 
-ALLFFLIBS = avcodec avdevice avfilter avformat avutil postproc swscale swresample
-
 # $(FFLIBS-yes) needs to be in linking order
 FFLIBS-$(CONFIG_AVDEVICE)   += avdevice
 FFLIBS-$(CONFIG_AVFILTER)   += avfilter
 FFLIBS-$(CONFIG_AVFORMAT)   += avformat
 FFLIBS-$(CONFIG_AVCODEC)    += avcodec
+FFLIBS-$(CONFIG_AVRESAMPLE) += avresample
 FFLIBS-$(CONFIG_POSTPROC)   += postproc
 FFLIBS-$(CONFIG_SWRESAMPLE) += swresample
 FFLIBS-$(CONFIG_SWSCALE)    += swscale
@@ -67,8 +65,6 @@ tools/target_io_dem_fuzzer$(EXESUF): tools/target_io_dem_fuzzer.o $(FF_DEP_LIBS)
 
 tools/enum_options$(EXESUF): ELIBS = $(FF_EXTRALIBS)
 tools/enum_options$(EXESUF): $(FF_DEP_LIBS)
-tools/scale_slice_test$(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 = $(FF_EXTRALIBS)
@@ -78,23 +74,21 @@ tools/target_dem_%_fuzzer$(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) :=
@@ -116,13 +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
@@ -163,7 +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 \

README.md

@@ -9,7 +9,7 @@ such as audio, video, subtitles and related metadata.
 * `libavcodec` provides implementation of a wider range of codecs.
 * `libavformat` implements streaming protocols, container formats and basic I/O access.
 * `libavutil` includes hashers, decompressors and miscellaneous utility functions.
-* `libavfilter` provides means to alter decoded audio and video through a directed graph of connected filters.
+* `libavfilter` provides a mean to alter decoded Audio and Video through chain of filters.
 * `libavdevice` provides an abstraction to access capture and playback devices.
 * `libswresample` implements audio mixing and resampling routines.
 * `libswscale` implements color conversion and scaling routines.

RELEASE

@@ -1 +1 @@
-5.1.git
+4.4

RELEASE_NOTES

@@ -0,0 +1,15 @@
+
+              ┌────────────────────────────────────┐
+              │ RELEASE NOTES for FFmpeg 4.4 "Rao" │
+              └────────────────────────────────────┘
+
+   The FFmpeg Project proudly presents FFmpeg 4.4 "Rao", about 10
+   months after the release of FFmpeg 4.3.
+
+   A complete Changelog is available at the root of the project, and the
+   complete Git history on https://git.ffmpeg.org/gitweb/ffmpeg.git
+
+   We hope you will like this release as much as we enjoyed working on it, and
+   as usual, if you have any questions about it, or any FFmpeg related topic,
+   feel free to join us on the #ffmpeg IRC channel (on irc.freenode.net) or ask
+   on the mailing-lists.

compat/atomics/win32/stdatomic.h

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

compat/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 */

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

compat/msvcrt/snprintf.c

@@ -59,7 +59,7 @@ int avpriv_vsnprintf(char *s, size_t n, const char *fmt,
      * recommends to provide _snprintf/_vsnprintf() a buffer size that
      * is one less than the actual buffer, and zero it before calling
      * _snprintf/_vsnprintf() to workaround this problem.
-     * See https://web.archive.org/web/20151214111935/http://msdn.microsoft.com/en-us/library/1kt27hek(v=vs.80).aspx */
+     * See http://msdn.microsoft.com/en-us/library/1kt27hek(v=vs.80).aspx */
     memset(s, 0, n);
     va_copy(ap_copy, ap);
     ret = _vsnprintf(s, n - 1, fmt, ap_copy);

compat/w32dlfcn.h

@@ -20,40 +20,11 @@
 #define COMPAT_W32DLFCN_H
 
 #ifdef _WIN32
-#include <stdint.h>
-
 #include <windows.h>
-
 #include "config.h"
-#include "libavutil/macros.h"
+#if (_WIN32_WINNT < 0x0602) || HAVE_WINRT
 #include "libavutil/wchar_filename.h"
-
-static inline wchar_t *get_module_filename(HMODULE module)
-{
-    wchar_t *path = NULL, *new_path;
-    DWORD path_size = 0, path_len;
-
-    do {
-        path_size = path_size ? FFMIN(2 * path_size, INT16_MAX + 1) : MAX_PATH;
-        new_path = av_realloc_array(path, path_size, sizeof *path);
-        if (!new_path) {
-            av_free(path);
-            return NULL;
-        }
-        path = new_path;
-        // Returns path_size in case of insufficient buffer.
-        // Whether the error is set or not and whether the output
-        // is null-terminated or not depends on the version of Windows.
-        path_len = GetModuleFileNameW(module, path, path_size);
-    } while (path_len && path_size <= INT16_MAX && path_size <= path_len);
-
-    if (!path_len) {
-        av_free(path);
-        return NULL;
-    }
-    return path;
-}
-
+#endif
 /**
  * Safe function used to open dynamic libs. This attempts to improve program security
  * by removing the current directory from the dll search path. Only dll's found in the
@@ -63,53 +34,29 @@ static inline wchar_t *get_module_filename(HMODULE module)
  */
 static inline HMODULE win32_dlopen(const char *name)
 {
-    wchar_t *name_w;
-    HMODULE module = NULL;
-    if (utf8towchar(name, &name_w))
-        name_w = NULL;
 #if _WIN32_WINNT < 0x0602
-    // On Win7 and earlier we check if KB2533623 is available
+    // Need to check if KB2533623 is available
     if (!GetProcAddress(GetModuleHandleW(L"kernel32.dll"), "SetDefaultDllDirectories")) {
-        wchar_t *path = NULL, *new_path;
-        DWORD pathlen, pathsize, namelen;
-        if (!name_w)
+        HMODULE module = NULL;
+        wchar_t *path = NULL, *name_w = NULL;
+        DWORD pathlen;
+        if (utf8towchar(name, &name_w))
             goto exit;
-        namelen = wcslen(name_w);
+        path = (wchar_t *)av_mallocz_array(MAX_PATH, sizeof(wchar_t));
         // Try local directory first
-        path = get_module_filename(NULL);
-        if (!path)
+        pathlen = GetModuleFileNameW(NULL, path, MAX_PATH);
+        pathlen = wcsrchr(path, '\\') - path;
+        if (pathlen == 0 || pathlen + wcslen(name_w) + 2 > MAX_PATH)
             goto exit;
-        new_path = wcsrchr(path, '\\');
-        if (!new_path)
-            goto exit;
-        pathlen = new_path - path;
-        pathsize = pathlen + namelen + 2;
-        new_path = av_realloc_array(path, pathsize, sizeof *path);
-        if (!new_path)
-            goto exit;
-        path = new_path;
+        path[pathlen] = '\\';
         wcscpy(path + pathlen + 1, name_w);
         module = LoadLibraryExW(path, NULL, LOAD_WITH_ALTERED_SEARCH_PATH);
         if (module == NULL) {
             // Next try System32 directory
-            pathlen = GetSystemDirectoryW(path, pathsize);
-            if (!pathlen)
+            pathlen = GetSystemDirectoryW(path, MAX_PATH);
+            if (pathlen == 0 || pathlen + wcslen(name_w) + 2 > MAX_PATH)
                 goto exit;
-            // Buffer is not enough in two cases:
-            // 1. system directory + \ + module name
-            // 2. system directory even without the module name.
-            if (pathlen + namelen + 2 > pathsize) {
-                pathsize = pathlen + namelen + 2;
-                new_path = av_realloc_array(path, pathsize, sizeof *path);
-                if (!new_path)
-                    goto exit;
-                path = new_path;
-                // Query again to handle the case #2.
-                pathlen = GetSystemDirectoryW(path, pathsize);
-                if (!pathlen)
-                    goto exit;
-            }
-            path[pathlen] = L'\\';
+            path[pathlen] = '\\';
             wcscpy(path + pathlen + 1, name_w);
             module = LoadLibraryExW(path, NULL, LOAD_WITH_ALTERED_SEARCH_PATH);
         }
@@ -126,19 +73,16 @@ exit:
 #   define LOAD_LIBRARY_SEARCH_SYSTEM32        0x00000800
 #endif
 #if HAVE_WINRT
-    if (!name_w)
+    wchar_t *name_w = NULL;
+    int ret;
+    if (utf8towchar(name, &name_w))
         return NULL;
-    module = LoadPackagedLibrary(name_w, 0);
-#else
-#define LOAD_FLAGS (LOAD_LIBRARY_SEARCH_APPLICATION_DIR | LOAD_LIBRARY_SEARCH_SYSTEM32)
-    /* filename may be be in CP_ACP */
-    if (!name_w)
-        return LoadLibraryExA(name, NULL, LOAD_FLAGS);
-    module = LoadLibraryExW(name_w, NULL, LOAD_FLAGS);
-#undef LOAD_FLAGS
-#endif
+    ret = LoadPackagedLibrary(name_w, 0);
     av_free(name_w);
-    return module;
+    return ret;
+#else
+    return LoadLibraryExA(name, NULL, LOAD_LIBRARY_SEARCH_APPLICATION_DIR | LOAD_LIBRARY_SEARCH_SYSTEM32);
+#endif
 }
 #define dlopen(name, flags) win32_dlopen(name)
 #define dlclose FreeLibrary

compat/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"

doc/APIchanges

@@ -1,420 +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 6.0 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
@@ -1810,7 +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

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
 
 # 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

doc/Makefile

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

doc/bitstream_filters.texi

@@ -81,7 +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}

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})
@@ -706,12 +761,64 @@ Possible values:
 @item lowres @var{integer} (@emph{decoding,audio,video})
 Decode at 1= 1/2, 2=1/4, 3=1/8 resolutions.
 
+@item skip_threshold @var{integer} (@emph{encoding,video})
+Set frame skip threshold.
+
+@item skip_factor @var{integer} (@emph{encoding,video})
+Set frame skip factor.
+
+@item skip_exp @var{integer} (@emph{encoding,video})
+Set frame skip exponent.
+Negative values behave identical to the corresponding positive ones, except
+that the score is normalized.
+Positive values exist primarily for compatibility reasons and are not so useful.
+
+@item skipcmp @var{integer} (@emph{encoding,video})
+Set frame skip compare function.
+
+Possible values:
+@table @samp
+@item sad
+sum of absolute differences, fast (default)
+@item sse
+sum of squared errors
+@item satd
+sum of absolute Hadamard transformed differences
+@item dct
+sum of absolute DCT transformed differences
+@item psnr
+sum of squared quantization errors (avoid, low quality)
+@item bit
+number of bits needed for the block
+@item rd
+rate distortion optimal, slow
+@item zero
+0
+@item vsad
+sum of absolute vertical differences
+@item vsse
+sum of squared vertical differences
+@item nsse
+noise preserving sum of squared differences
+@item w53
+5/3 wavelet, only used in snow
+@item w97
+9/7 wavelet, only used in snow
+@item dctmax
+
+@item chroma
+
+@end table
+
 @item 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})
@@ -751,17 +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})
 

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
@@ -360,8 +293,6 @@ Enabled by default.
 @table @option
 @item compute_clut
 @table @option
-@item -2
-Compute clut once if no matching CLUT is in the stream.
 @item -1
 Compute clut if no matching CLUT is in the stream.
 @item 0

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.
@@ -274,47 +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.
-
-@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 ...
@@ -391,9 +324,6 @@ 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.
 
@@ -416,13 +346,6 @@ 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
@@ -738,12 +661,6 @@ 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.
@@ -762,15 +679,6 @@ 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.
 @end table
 
 @subsection Audible AAX
@@ -811,10 +719,6 @@ 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

doc/dev_community/community.md

@@ -25,9 +25,6 @@ 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`.
-
 ## Voting
 
 Voting is done using a ranked voting system, currently running on https://vote.ffmpeg.org/ .

doc/developer.texi

@@ -10,79 +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.
 
-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:
-@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.
-@end itemize
+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}.
 
-@section Contributing code
+@chapter Contributing
 
-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}.
+There are 2 ways by which code gets into FFmpeg:
+@itemize @bullet
+@item Submitting patches to the ffmpeg-devel mailing list.
+      See @ref{Submitting patches} for details.
+@item Directly committing changes to the main tree.
+@end itemize
+
+Whichever way, changes should be reviewed by the maintainer of the code
+before they are committed. And they should follow the @ref{Coding Rules}.
 The developer making the commit and the author are responsible for their changes
 and should try to fix issues their commit causes.
 
 @anchor{Coding Rules}
 @chapter Coding Rules
 
-@section C language features
-
-FFmpeg is 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
-
 @section Code formatting conventions
 
 There are the following guidelines regarding the indentation in files:
@@ -105,39 +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
@@ -179,51 +110,92 @@ int myfunc(int my_parameter)
 ...
 @end example
 
-@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.
@@ -246,7 +218,39 @@ 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 Patches/Committing
@@ -490,22 +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

doc/doxy-wrapper.sh

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

doc/encoders.texi

@@ -53,7 +53,7 @@ Set AAC encoder coding method. Possible values:
 
 @table @samp
 @item twoloop
-Two loop searching (TLS) method. This is the default method.
+Two loop searching (TLS) method.
 
 This method first sets quantizers depending on band thresholds and then tries
 to find an optimal combination by adding or subtracting a specific value from
@@ -75,6 +75,7 @@ Constant quantizer method.
 Uses a cheaper version of twoloop algorithm that doesn't try to do as many
 clever adjustments. Worse with low bitrates (less than 64kbps), but is better
 and much faster at higher bitrates.
+This is the default choice for a coder
 
 @end table
 
@@ -1267,59 +1268,6 @@ disabled
 A description of some of the currently available video encoders
 follows.
 
-@section a64_multi, a64_multi5
-
-A64 / Commodore 64 multicolor charset encoder. @code{a64_multi5} is extended with 5th color (colram).
-
-@section Cinepak
-
-Cinepak aka CVID encoder.
-Compatible with Windows 3.1 and vintage MacOS.
-
-@subsection Options
-
-@table @option
-@item g @var{integer}
-Keyframe interval.
-A keyframe is inserted at least every @code{-g} frames, sometimes sooner.
-
-@item q:v @var{integer}
-Quality factor. Lower is better. Higher gives lower bitrate.
-The following table lists bitrates when encoding akiyo_cif.y4m for various values of @code{-q:v} with @code{-g 100}:
-
-@table @option
-@item @code{-q:v 1} 1918 kb/s
-@item @code{-q:v 2} 1735 kb/s
-@item @code{-q:v 4} 1500 kb/s
-@item @code{-q:v 10} 1041 kb/s
-@item @code{-q:v 20} 826 kb/s
-@item @code{-q:v 40} 553 kb/s
-@item @code{-q:v 100} 394 kb/s
-@item @code{-q:v 200} 312 kb/s
-@item @code{-q:v 400} 266 kb/s
-@item @code{-q:v 1000} 237 kb/s
-@end table
-
-@item max_extra_cb_iterations @var{integer}
-Max extra codebook recalculation passes, more is better and slower.
-
-@item skip_empty_cb @var{boolean}
-Avoid wasting bytes, ignore vintage MacOS decoder.
-
-@item max_strips @var{integer}
-@itemx min_strips @var{integer}
-The minimum and maximum number of strips to use.
-Wider range sometimes improves quality.
-More strips is generally better quality but costs more bits.
-Fewer strips tend to yield more keyframes.
-Vintage compatible is 1..3.
-
-@item strip_number_adaptivity @var{integer}
-How much number of strips is allowed to change between frames.
-Higher is better but slower.
-
-@end table
-
 @section GIF
 
 GIF image/animation encoder.
@@ -1799,29 +1747,27 @@ You need to explicitly configure the build with @code{--enable-libsvtav1}.
 @table @option
 @item profile
 Set the encoding profile.
-@table @samp
-@item main
-@item high
-@item professional
-@end table
 
 @item level
-Set the operating point level. For example: '4.0'
-
-@item hielevel
-Set the Hierarchical prediction levels.
-@table @samp
-@item 3level
-@item 4level
-This is the default.
-@end table
+Set the operating point level.
 
 @item tier
 Set the operating point tier.
-@table @samp
-@item main
-This is the default.
-@item high
+
+@item rc
+Set the rate control mode to use.
+
+Possible modes:
+@table @option
+@item cqp
+Constant quantizer: use fixed values of qindex (dependent on the frame type)
+throughout the stream.  This mode is the default.
+
+@item vbr
+Variable bitrate: use a target bitrate for the whole stream.
+
+@item cvbr
+Constrained variable bitrate: use a target bitrate for each GOP.
 @end table
 
 @item qmax
@@ -1830,9 +1776,6 @@ Set the maximum quantizer to use when using a bitrate mode.
 @item qmin
 Set the minimum quantizer to use when using a bitrate mode.
 
-@item crf
-Constant rate factor value used in crf rate control mode (0-63).
-
 @item qp
 Set the quantizer used in cqp rate control mode (0-63).
 
@@ -1843,8 +1786,8 @@ Enable scene change detection.
 Set number of frames to look ahead (0-120).
 
 @item preset
-Set the quality-speed tradeoff, in the range 0 to 13.  Higher values are
-faster but lower quality.
+Set the quality-speed tradeoff, in the range 0 to 8.  Higher values are
+faster but lower quality.  Defaults to 8 (highest speed).
 
 @item tile_rows
 Set log2 of the number of rows of tiles to use (0-6).
@@ -1852,45 +1795,6 @@ Set log2 of the number of rows of tiles to use (0-6).
 @item tile_columns
 Set log2 of the number of columns of tiles to use (0-4).
 
-@item svtav1-params
-Set SVT-AV1 options using a list of @var{key}=@var{value} pairs separated
-by ":". See the SVT-AV1 encoder user guide for a list of accepted parameters.
-
-@end table
-
-@section libjxl
-
-libjxl JPEG XL encoder wrapper.
-
-Requires the presence of the libjxl headers and library during
-configuration. You need to explicitly configure the build with
-@code{--enable-libjxl}.
-
-@subsection Options
-
-The libjxl wrapper supports the following options:
-
-@table @option
-
-@item distance
-Set the target Butteraugli distance. This is a quality setting: lower
-distance yields higher quality, with distance=1.0 roughly comparable to
-libjpeg Quality 90 for photographic content. Setting distance=0.0 yields
-true lossless encoding. Valid values range between 0.0 and 15.0, and sane
-values rarely exceed 5.0. Setting distance=0.1 usually attains
-transparency for most input. The default is 1.0.
-
-@item effort
-Set the encoding effort used. Higher effort values produce more consistent
-quality and usually produces a better quality/bpp curve, at the cost of
-more CPU time required. Valid values range from 1 to 9, and the default is 7.
-
-@item modular
-Force the encoder to use Modular mode instead of choosing automatically. The
-default is to use VarDCT for lossy encoding and Modular for lossless. VarDCT
-is generally superior to Modular for lossy encoding but does not support
-lossless encoding.
-
 @end table
 
 @section libkvazaar
@@ -2080,11 +1984,8 @@ kilobits/s.
 @item keyint_min (@emph{kf-min-dist})
 
 @item qmin (@emph{min-q})
-Minimum (Best Quality) Quantizer.
 
 @item qmax (@emph{max-q})
-Maximum (Worst Quality) Quantizer.
-Can be changed per-frame.
 
 @item bufsize (@emph{buf-sz}, @emph{buf-optimal-sz})
 Set ratecontrol buffer size (in bits). Note @command{vpxenc}'s options are
@@ -2176,8 +2077,6 @@ Set altref noise reduction filter type: backward, forward, centered.
 Set altref noise reduction filter strength.
 @item rc-lookahead, lag-in-frames (@emph{lag-in-frames})
 Set number of frames to look ahead for frametype and ratecontrol.
-@item min-gf-interval
-Set minimum golden/alternate reference frame interval (VP9 only).
 @end table
 
 @item error-resilient
@@ -2350,10 +2249,11 @@ and compression tools used, and varies the combination of these tools. This
 maps to the @var{method} option in libwebp. The valid range is 0 to 6.
 Default is 4.
 
-@item -quality @var{float}
-For lossy encoding, this controls image quality. For lossless encoding, this
-controls the effort and time spent in compression.
-Range is 0 to 100. Default is 75.
+@item -qscale @var{float}
+For lossy encoding, this controls image quality, 0 to 100. For lossless
+encoding, this controls the effort and time spent at compressing more. The
+default value is 75. Note that for usage via libavcodec, this option is called
+@var{global_quality} and must be multiplied by @var{FF_QP2LAMBDA}.
 
 @item -preset @var{type}
 Configuration preset. This does some automatic settings based on the general
@@ -2739,9 +2639,6 @@ ffmpeg -i foo.mpg -c:v libx264 -x264opts keyint=123:min-keyint=20 -an out.mkv
 Import closed captions (which must be ATSC compatible format) into output.
 Only the mpeg2 and h264 decoders provide these. Default is 1 (on).
 
-@item udu_sei @var{boolean}
-Import user data unregistered SEI if available into output. Default is 0 (off).
-
 @item x264-params (N.A.)
 Override the x264 configuration using a :-separated list of key=value
 parameters.
@@ -2823,9 +2720,6 @@ Quantizer curve compression factor
 Normally, when forcing a I-frame type, the encoder can select any type
 of I-frame. This option forces it to choose an IDR-frame.
 
-@item udu_sei @var{boolean}
-Import user data unregistered SEI if available into output. Default is 0 (off).
-
 @item x265-params
 Set x265 options using a list of @var{key}=@var{value} couples separated
 by ":". See @command{x265 --help} for a list of options.
@@ -3202,13 +3096,12 @@ Setting a higher @option{bits_per_mb} limit will improve the speed.
 For the fastest encoding speed set the @option{qscale} parameter (4 is the
 recommended value) and do not set a size constraint.
 
-@section QSV Encoders
+@section QSV encoders
 
-The family of Intel QuickSync Video encoders (MPEG-2, H.264, HEVC, JPEG/MJPEG
-and VP9)
+The family of Intel QuickSync Video encoders (MPEG-2, H.264, HEVC, JPEG/MJPEG and VP9)
 
-@subsection Ratecontrol Method
 The ratecontrol method is selected as follows:
+
 @itemize @bullet
 @item
 When @option{global_quality} is specified, a quality-based mode is used.
@@ -3223,8 +3116,7 @@ also set (the @option{-qscale} ffmpeg option).
 @option{look_ahead} option is also set.
 
 @item
-@var{ICQ} -- intelligent constant quality otherwise. For the ICQ modes, global
-quality range is 1 to 51, with 1 being the best quality.
+@var{ICQ} -- intelligent constant quality otherwise.
 @end itemize
 
 @item
@@ -3246,9 +3138,9 @@ the average bitrate.
 than the average bitrate.
 
 @item
-@var{AVBR} - average VBR mode, when @option{maxrate} is not specified, both
-@option{avbr_accuracy} and @option{avbr_convergence} are set to non-zero. This
-mode is available for H264 and HEVC on Windows.
+@var{AVBR} - average VBR mode, when @option{maxrate} is not specified. This mode
+is further configured by the @option{avbr_accuracy} and
+@option{avbr_convergence} options.
 @end itemize
 @end itemize
 
@@ -3256,7 +3148,6 @@ Note that depending on your system, a different mode than the one you specified
 may be selected by the encoder. Set the verbosity level to @var{verbose} or
 higher to see the actual settings used by the QSV runtime.
 
-@subsection Global Options -> MSDK Options
 Additional libavcodec global options are mapped to MSDK options as follows:
 
 @itemize
@@ -3293,575 +3184,6 @@ encoder use CAVLC instead of CABAC.
 
 @end itemize
 
-@subsection Common Options
-Following options are used by all qsv encoders.
-
-@table @option
-@item @var{async_depth}
-Specifies how many asynchronous operations an application performs
-before the application explicitly synchronizes the result. If zero,
-the value is not specified.
-
-@item @var{preset}
-This option itemizes a range of choices from veryfast (best speed) to veryslow
-(best quality).
-@table @samp
-@item veryfast
-@item faster
-@item fast
-@item medium
-@item slow
-@item slower
-@item veryslow
-@end table
-
-@item @var{forced_idr}
-Forcing I frames as IDR frames.
-
-@item @var{low_power}
-For encoders set this flag to ON to reduce power consumption and GPU usage.
-@end table
-
-@subsection Runtime Options
-Following options can be used durning qsv encoding.
-
-@table @option
-@item @var{global_quality}
-@item @var{i_quant_factor}
-@item @var{i_quant_offset}
-@item @var{b_quant_factor}
-@item @var{b_quant_offset}
-Supported in h264_qsv and hevc_qsv.
-Change these value to reset qsv codec's qp configuration.
-
-@item @var{max_frame_size}
-Supported in h264_qsv and hevc_qsv.
-Change this value to reset qsv codec's MaxFrameSize configuration.
-
-@item @var{gop_size}
-Change this value to reset qsv codec's gop configuration.
-
-@item @var{int_ref_type}
-@item @var{int_ref_cycle_size}
-@item @var{int_ref_qp_delta}
-@item @var{int_ref_cycle_dist}
-Supported in h264_qsv and hevc_qsv.
-Change these value to reset qsv codec's Intra Refresh configuration.
-
-@item @var{qmax}
-@item @var{qmin}
-@item @var{max_qp_i}
-@item @var{min_qp_i}
-@item @var{max_qp_p}
-@item @var{min_qp_p}
-@item @var{max_qp_b}
-@item @var{min_qp_b}
-Supported in h264_qsv.
-Change these value to reset qsv codec's max/min qp configuration.
-
-@item @var{low_delay_brc}
-Supported in h264_qsv and hevc_qsv.
-Change this value to reset qsv codec's low_delay_brc configuration.
-
-@item @var{framerate}
-Change this value to reset qsv codec's framerate configuration.
-
-@item @var{bit_rate}
-@item @var{rc_buffer_size}
-@item @var{rc_initial_buffer_occupancy}
-@item @var{rc_max_rate}
-Change these value to reset qsv codec's bitrate control configuration.
-
-@item @var{pic_timing_sei}
-Supported in h264_qsv and hevc_qsv.
-Change this value to reset qsv codec's pic_timing_sei configuration.
-@end table
-
-@subsection H264 options
-These options are used by h264_qsv
-
-@table @option
-@item @var{extbrc}
-Extended bitrate control.
-
-@item @var{recovery_point_sei}
-Set this flag to insert the recovery point SEI message at the beginning of every
-intra refresh cycle.
-
-@item @var{rdo}
-Enable rate distortion optimization.
-
-@item @var{max_frame_size}
-Maximum encoded frame size in bytes.
-
-@item @var{max_frame_size_i}
-Maximum encoded frame size for I frames in bytes. If this value is set as larger
-than zero, then for I frames the value set by max_frame_size is ignored.
-
-@item @var{max_frame_size_p}
-Maximum encoded frame size for P frames in bytes. If this value is set as larger
-than zero, then for P frames the value set by max_frame_size is ignored.
-
-@item @var{max_slice_size}
-Maximum encoded slice size in bytes.
-
-@item @var{bitrate_limit}
-Toggle bitrate limitations.
-Modifies bitrate to be in the range imposed by the QSV encoder. Setting this
-flag off may lead to violation of HRD conformance. Mind that specifying bitrate
-below the QSV encoder range might significantly affect quality. If on this
-option takes effect in non CQP modes: if bitrate is not in the range imposed
-by the QSV encoder, it will be changed to be in the range.
-
-@item @var{mbbrc}
-Setting this flag enables macroblock level bitrate control that generally
-improves subjective visual quality. Enabling this flag may have negative impact
-on performance and objective visual quality metric.
-
-@item @var{low_delay_brc}
-Setting this flag turns on or off LowDelayBRC feautre in qsv plugin, which provides
-more accurate bitrate control to minimize the variance of bitstream size frame
-by frame. Value: -1-default 0-off 1-on
-
-@item @var{adaptive_i}
-This flag controls insertion of I frames by the QSV encoder. Turn ON this flag
-to allow changing of frame type from P and B to I.
-
-@item @var{adaptive_b}
-This flag controls changing of frame type from B to P.
-
-@item @var{p_strategy}
-Enable P-pyramid: 0-default 1-simple 2-pyramid(bf need to be set to 0).
-
-@item @var{b_strategy}
-This option controls usage of B frames as reference.
-
-@item @var{dblk_idc}
-This option disable deblocking. It has value in range 0~2.
-
-@item @var{cavlc}
-If set, CAVLC is used; if unset, CABAC is used for encoding.
-
-@item @var{vcm}
-Video conferencing mode, please see ratecontrol method.
-
-@item @var{idr_interval}
-Distance (in I-frames) between IDR frames.
-
-@item @var{pic_timing_sei}
-Insert picture timing SEI with pic_struct_syntax element.
-
-@item @var{single_sei_nal_unit}
-Put all the SEI messages into one NALU.
-
-@item @var{max_dec_frame_buffering}
-Maximum number of frames buffered in the DPB.
-
-@item @var{look_ahead}
-Use VBR algorithm with look ahead.
-
-@item @var{look_ahead_depth}
-Depth of look ahead in number frames.
-
-@item @var{look_ahead_downsampling}
-Downscaling factor for the frames saved for the lookahead analysis.
-@table @samp
-@item unknown
-@item auto
-@item off
-@item 2x
-@item 4x
-@end table
-
-@item @var{int_ref_type}
-Specifies intra refresh type. The major goal of intra refresh is improvement of
-error resilience without significant impact on encoded bitstream size caused by
-I frames. The SDK encoder achieves this by encoding part of each frame in
-refresh cycle using intra MBs. @var{none} means no refresh. @var{vertical} means
-vertical refresh, by column of MBs. @var{horizontal} means horizontal refresh,
-by rows of MBs. @var{slice} means horizontal refresh by slices without
-overlapping. In case of @var{slice}, in_ref_cycle_size is ignored. To enable
-intra refresh, B frame should be set to 0.
-
-@item @var{int_ref_cycle_size}
-Specifies number of pictures within refresh cycle starting from 2. 0 and 1 are
-invalid values.
-
-@item @var{int_ref_qp_delta}
-Specifies QP difference for inserted intra MBs. This is signed value in
-[-51, 51] range if target encoding bit-depth for luma samples is 8 and this
-range is [-63, 63] for 10 bit-depth or [-75, 75] for 12 bit-depth respectively.
-
-@item @var{int_ref_cycle_dist}
-Distance between the beginnings of the intra-refresh cycles in frames.
-
-@item @var{profile}
-@table @samp
-@item unknown
-@item baseline
-@item main
-@item high
-@end table
-
-@item @var{a53cc}
-Use A53 Closed Captions (if available).
-
-@item @var{aud}
-Insert the Access Unit Delimiter NAL.
-
-@item @var{mfmode}
-Multi-Frame Mode.
-@table @samp
-@item off
-@item auto
-@end table
-
-@item @var{repeat_pps}
-Repeat pps for every frame.
-
-@item @var{max_qp_i}
-Maximum video quantizer scale for I frame.
-
-@item @var{min_qp_i}
-Minimum video quantizer scale for I frame.
-
-@item @var{max_qp_p}
-Maximum video quantizer scale for P frame.
-
-@item @var{min_qp_p}
-Minimum video quantizer scale for P frame.
-
-@item @var{max_qp_b}
-Maximum video quantizer scale for B frame.
-
-@item @var{min_qp_b}
-Minimum video quantizer scale for B frame.
-
-@item @var{scenario}
-Provides a hint to encoder about the scenario for the encoding session.
-@table @samp
-@item unknown
-@item displayremoting
-@item videoconference
-@item archive
-@item livestreaming
-@item cameracapture
-@item videosurveillance
-@item gamestreaming
-@item remotegaming
-@end table
-
-@item @var{avbr_accuracy}
-Accuracy of the AVBR ratecontrol (unit of tenth of percent).
-
-@item @var{avbr_convergence}
-Convergence of the AVBR ratecontrol (unit of 100 frames)
-
-The parameters @var{avbr_accuracy} and @var{avbr_convergence} are for the
-average variable bitrate control (AVBR) algorithm.
-The algorithm focuses on overall encoding quality while meeting the specified
-bitrate, @var{target_bitrate}, within the accuracy range @var{avbr_accuracy},
-after a @var{avbr_Convergence} period. This method does not follow HRD and the
-instant bitrate is not capped or padded.
-
-@item @var{skip_frame}
-Use per-frame metadata "qsv_skip_frame" to skip frame when encoding. This option
-defines the usage of this metadata.
-@table @samp
-@item no_skip
-Frame skipping is disabled.
-@item insert_dummy
-Encoder inserts into bitstream frame where all macroblocks are encoded as
-skipped.
-@item insert_nothing
-Similar to insert_dummy, but encoder inserts nothing into bitstream. The skipped
-frames are still used in brc. For example, gop still include skipped frames, and
-the frames after skipped frames will be larger in size.
-@item brc_only
-skip_frame metadata indicates the number of missed frames before the current
-frame.
-@end table
-
-@end table
-
-@subsection HEVC Options
-These options are used by hevc_qsv
-
-@table @option
-@item @var{extbrc}
-Extended bitrate control.
-
-@item @var{recovery_point_sei}
-Set this flag to insert the recovery point SEI message at the beginning of every
-intra refresh cycle.
-
-@item @var{rdo}
-Enable rate distortion optimization.
-
-@item @var{max_frame_size}
-Maximum encoded frame size in bytes.
-
-@item @var{max_frame_size_i}
-Maximum encoded frame size for I frames in bytes. If this value is set as larger
-than zero, then for I frames the value set by max_frame_size is ignored.
-
-@item @var{max_frame_size_p}
-Maximum encoded frame size for P frames in bytes. If this value is set as larger
-than zero, then for P frames the value set by max_frame_size is ignored.
-
-@item @var{max_slice_size}
-Maximum encoded slice size in bytes.
-
-@item @var{mbbrc}
-Setting this flag enables macroblock level bitrate control that generally
-improves subjective visual quality. Enabling this flag may have negative impact
-on performance and objective visual quality metric.
-
-@item @var{low_delay_brc}
-Setting this flag turns on or off LowDelayBRC feautre in qsv plugin, which provides
-more accurate bitrate control to minimize the variance of bitstream size frame
-by frame. Value: -1-default 0-off 1-on
-
-@item @var{adaptive_i}
-This flag controls insertion of I frames by the QSV encoder. Turn ON this flag
-to allow changing of frame type from P and B to I.
-
-@item @var{adaptive_b}
-This flag controls changing of frame type from B to P.
-
-@item @var{p_strategy}
-Enable P-pyramid: 0-default 1-simple 2-pyramid(bf need to be set to 0).
-
-@item @var{b_strategy}
-This option controls usage of B frames as reference.
-
-@item @var{dblk_idc}
-This option disable deblocking. It has value in range 0~2.
-
-@item @var{idr_interval}
-Distance (in I-frames) between IDR frames.
-@table @samp
-@item begin_only
-Output an IDR-frame only at the beginning of the stream.
-@end table
-
-@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.
-
-@item @var{look_ahead_depth}
-Depth of look ahead in number frames, available when extbrc option is enabled.
-
-@item @var{profile}
-Set the encoding profile (scc requires libmfx >= 1.32).
-
-@table @samp
-@item unknown
-@item main
-@item main10
-@item mainsp
-@item rext
-@item scc
-@end table
-
-@item @var{tier}
-Set the encoding tier (only level >= 4 can support high tier).
-This option only takes effect when the level option is specified.
-
-@table @samp
-@item main
-@item high
-@end table
-
-@item @var{gpb}
-1: GPB (generalized P/B frame)
-
-0: regular P frame.
-
-@item @var{tile_cols}
-Number of columns for tiled encoding.
-
-@item @var{tile_rows}
-Number of rows for tiled encoding.
-
-@item @var{aud}
-Insert the Access Unit Delimiter NAL.
-
-@item @var{pic_timing_sei}
-Insert picture timing SEI with pic_struct_syntax element.
-
-@item @var{transform_skip}
-Turn this option ON to enable transformskip. It is supported on platform equal
-or newer than ICL.
-
-@item @var{int_ref_type}
-Specifies intra refresh type. The major goal of intra refresh is improvement of
-error resilience without significant impact on encoded bitstream size caused by
-I frames. The SDK encoder achieves this by encoding part of each frame in
-refresh cycle using intra MBs. @var{none} means no refresh. @var{vertical} means
-vertical refresh, by column of MBs. @var{horizontal} means horizontal refresh,
-by rows of MBs. @var{slice} means horizontal refresh by slices without
-overlapping. In case of @var{slice}, in_ref_cycle_size is ignored. To enable
-intra refresh, B frame should be set to 0.
-
-@item @var{int_ref_cycle_size}
-Specifies number of pictures within refresh cycle starting from 2. 0 and 1 are
-invalid values.
-
-@item @var{int_ref_qp_delta}
-Specifies QP difference for inserted intra MBs. This is signed value in
-[-51, 51] range if target encoding bit-depth for luma samples is 8 and this
-range is [-63, 63] for 10 bit-depth or [-75, 75] for 12 bit-depth respectively.
-
-@item @var{int_ref_cycle_dist}
-Distance between the beginnings of the intra-refresh cycles in frames.
-
-@item @var{max_qp_i}
-Maximum video quantizer scale for I frame.
-
-@item @var{min_qp_i}
-Minimum video quantizer scale for I frame.
-
-@item @var{max_qp_p}
-Maximum video quantizer scale for P frame.
-
-@item @var{min_qp_p}
-Minimum video quantizer scale for P frame.
-
-@item @var{max_qp_b}
-Maximum video quantizer scale for B frame.
-
-@item @var{min_qp_b}
-Minimum video quantizer scale for B frame.
-
-@item @var{scenario}
-Provides a hint to encoder about the scenario for the encoding session.
-@table @samp
-@item unknown
-@item displayremoting
-@item videoconference
-@item archive
-@item livestreaming
-@item cameracapture
-@item videosurveillance
-@item gamestreaming
-@item remotegaming
-@end table
-
-@item @var{avbr_accuracy}
-Accuracy of the AVBR ratecontrol (unit of tenth of percent).
-
-@item @var{avbr_convergence}
-Convergence of the AVBR ratecontrol (unit of 100 frames)
-
-The parameters @var{avbr_accuracy} and @var{avbr_convergence} are for the
-average variable bitrate control (AVBR) algorithm.
-The algorithm focuses on overall encoding quality while meeting the specified
-bitrate, @var{target_bitrate}, within the accuracy range @var{avbr_accuracy},
-after a @var{avbr_Convergence} period. This method does not follow HRD and the
-instant bitrate is not capped or padded.
-
-@item @var{skip_frame}
-Use per-frame metadata "qsv_skip_frame" to skip frame when encoding. This option
-defines the usage of this metadata.
-@table @samp
-@item no_skip
-Frame skipping is disabled.
-@item insert_dummy
-Encoder inserts into bitstream frame where all macroblocks are encoded as
-skipped.
-@item insert_nothing
-Similar to insert_dummy, but encoder inserts nothing into bitstream. The skipped
-frames are still used in brc. For example, gop still include skipped frames, and
-the frames after skipped frames will be larger in size.
-@item brc_only
-skip_frame metadata indicates the number of missed frames before the current
-frame.
-@end table
-
-@end table
-
-@subsection MPEG2 Options
-These options are used by mpeg2_qsv
-@table @option
-@item @var{profile}
-@table @samp
-@item unknown
-@item simple
-@item main
-@item high
-@end table
-@end table
-
-@subsection VP9 Options
-These options are used by vp9_qsv
-@table @option
-@item @var{profile}
-@table @samp
-@item unknown
-@item profile0
-@item profile1
-@item profile2
-@item profile3
-@end table
-
-@item @var{tile_cols}
-Number of columns for tiled encoding (requires libmfx >= 1.29).
-
-@item @var{tile_rows}
-Number of rows for tiled encoding (requires libmfx  >= 1.29).
-@end table
-
-@subsection AV1 Options
-These options are used by av1_qsv (requires libvpl).
-@table @option
-@item @var{profile}
-@table @samp
-@item unknown
-@item main
-@end table
-
-@item @var{tile_cols}
-Number of columns for tiled encoding.
-
-@item @var{tile_rows}
-Number of rows for tiled encoding.
-
-@item @var{adaptive_i}
-This flag controls insertion of I frames by the QSV encoder. Turn ON this flag
-to allow changing of frame type from P and B to I.
-
-@item @var{adaptive_b}
-This flag controls changing of frame type from B to P.
-
-@item @var{b_strategy}
-This option controls usage of B frames as reference.
-
-@item @var{extbrc}
-Extended bitrate control.
-
-@item @var{look_ahead_depth}
-Depth of look ahead in number frames, available when extbrc option is enabled.
-
-@item @var{low_delay_brc}
-Setting this flag turns on or off LowDelayBRC feautre in qsv plugin, which provides
-more accurate bitrate control to minimize the variance of bitstream size frame
-by frame. Value: -1-default 0-off 1-on
-
-@item max_frame_size
-Set the allowed max size in bytes for each frame. If the frame size exceeds
-the limitation, encoder will adjust the QP value to control the frame size.
-Invalid in CQP rate control mode.
-@end table
-
 @section snow
 
 @subsection Options
@@ -3942,17 +3264,6 @@ will refer only to P- or I-frames.  When set to greater values multiple layers
 of B-frames will be present, frames in each layer only referring to frames in
 higher layers.
 
-@item async_depth
-Maximum processing parallelism. Increase this to improve single channel
-performance. This option doesn't work if driver doesn't implement vaSyncBuffer
-function. Please make sure there are enough hw_frames allocated if a large
-number of async_depth is used.
-
-@item max_frame_size
-Set the allowed max size in bytes for each frame. If the frame size exceeds
-the limitation, encoder will adjust the QP value to control the frame size.
-Invalid in CQP rate control mode.
-
 @item rc_mode
 Set the rate control mode to use.  A given driver may only support a subset of
 modes.
@@ -4096,22 +3407,6 @@ required to produce a stream usable with all decoders.
 
 @end table
 
-@section vbn
-
-Vizrt Binary Image encoder.
-
-This format is used by the broadcast vendor Vizrt for quick texture streaming.
-Advanced features of the format such as LZW compression of texture data or
-generation of mipmaps are not supported.
-
-@subsection Options
-
-@table @option
-@item format @var{string}
-Sets the texture compression used by the VBN file. Can be @var{dxt1},
-@var{dxt5} or @var{raw}. Default is @var{dxt5}.
-@end table
-
 @section vc2
 
 SMPTE VC-2 (previously BBC Dirac Pro). This codec was primarily aimed at

doc/examples/.gitignore

@@ -22,4 +22,3 @@
 /transcoding
 /vaapi_encode
 /vaapi_transcode
-/qsv_transcode

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))

doc/examples/Makefile.example

@@ -11,32 +11,25 @@ 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))
 

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.

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>

doc/examples/avio_reading.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>

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;
 

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);

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;
@@ -73,8 +72,8 @@ 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 */
@@ -150,7 +149,8 @@ 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 +185,7 @@ 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 +345,7 @@ 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)) {

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;
         }

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);

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;
 }

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;
 

doc/examples/filtering_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);
 

doc/examples/filtering_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));

doc/examples/http_multiclient.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>

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);

doc/examples/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);

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;
@@ -373,7 +376,7 @@ 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);
 }
 
 /**************************************************************/
@@ -402,8 +405,7 @@ static AVFrame *alloc_picture(enum AVPixelFormat pix_fmt, int width, int height)
     return picture;
 }
 
-static void open_video(AVFormatContext *oc, const AVCodec *codec,
-                       OutputStream *ost, AVDictionary *opt_arg)
+static void open_video(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
 {
     int ret;
     AVCodecContext *c = ost->enc;
@@ -516,7 +518,7 @@ 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 +526,6 @@ 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 +536,10 @@ 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 +626,10 @@ int main(int argc, char **argv)
         }
     }
 
+    /* Write the trailer, if any. The trailer must be written before you
+     * close the CodecContexts open when you wrote the header; otherwise
+     * av_write_trailer() may try to use memory that was freed on
+     * av_codec_close(). */
     av_write_trailer(oc);
 
     /* Close each codec. */

doc/examples/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;
-}

doc/examples/qsvdec.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);
 

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);
 

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);

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>

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. */
@@ -450,7 +449,7 @@ static int init_converted_samples(uint8_t ***converted_input_samples,
      * Each pointer will later point to the audio samples of the corresponding
      * channels (although it may be NULL for interleaved formats).
      */
-    if (!(*converted_input_samples = calloc(output_codec_context->ch_layout.nb_channels,
+    if (!(*converted_input_samples = calloc(output_codec_context->channels,
                                             sizeof(**converted_input_samples)))) {
         fprintf(stderr, "Could not allocate converted input sample pointers\n");
         return AVERROR(ENOMEM);
@@ -459,7 +458,7 @@ static int init_converted_samples(uint8_t ***converted_input_samples,
     /* Allocate memory for the samples of all channels in one consecutive
      * block for convenience. */
     if ((error = av_samples_alloc(*converted_input_samples, NULL,
-                                  output_codec_context->ch_layout.nb_channels,
+                                  output_codec_context->channels,
                                   frame_size,
                                   output_codec_context->sample_fmt, 0)) < 0) {
         fprintf(stderr,
@@ -559,7 +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. */
@@ -633,7 +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;
 
@@ -680,16 +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. */
@@ -860,6 +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;

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);
@@ -125,7 +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;
 
@@ -177,9 +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};
@@ -292,7 +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) {
@@ -301,14 +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",
+                "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) {
@@ -331,9 +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;

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>
@@ -102,7 +105,7 @@ int main(int argc, char *argv[])
     FILE *fin = NULL, *fout = NULL;
     AVFrame *sw_frame = NULL, *hw_frame = NULL;
     AVCodecContext *avctx = NULL;
-    const AVCodec *codec = NULL;
+    AVCodec *codec = NULL;
     const char *enc_name = "h264_vaapi";
 
     if (argc < 5) {

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;
@@ -223,9 +226,9 @@ fail:
 
 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"

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.
 

doc/ffmpeg.texi

@@ -449,11 +449,6 @@ output file already exists.
 Set number of times input stream shall be looped. Loop 0 means no loop,
 loop -1 means infinite loop.
 
-@item -recast_media (@emph{global})
-Allow forcing a decoder of a different media type than the one
-detected or designated by the demuxer. Useful for decoding media
-data muxed as data streams.
-
 @item -c[:@var{stream_specifier}] @var{codec} (@emph{input/output,per-stream})
 @itemx -codec[:@var{stream_specifier}] @var{codec} (@emph{input/output,per-stream})
 Select an encoder (when used before an output file) or a decoder (when used
@@ -518,21 +513,6 @@ see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1)
 Like the @code{-ss} option but relative to the "end of file". That is negative
 values are earlier in the file, 0 is at EOF.
 
-@item -isync @var{input_index} (@emph{input})
-Assign an input as a sync source.
-
-This will take the difference between the start times of the target and reference inputs and
-offset the timestamps of the target file by that difference. The source timestamps of the two
-inputs should derive from the same clock source for expected results. If @code{copyts} is set
-then @code{start_at_zero} must also be set. If either of the inputs has no starting timestamp
-then no sync adjustment is made.
-
-Acceptable values are those that refer to a valid ffmpeg input index. If the sync reference is
-the target index itself or @var{-1}, then no adjustment is made to target timestamps. A sync
-reference may not itself be synced to any other input.
-
-Default value is @var{-1}.
-
 @item -itsoffset @var{offset} (@emph{input})
 Set the input time offset.
 
@@ -575,22 +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
@@ -639,21 +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
@@ -774,16 +759,6 @@ This option is similar to @option{-filter}, the only difference is that its
 argument is the name of the file from which a filtergraph description is to be
 read.
 
-@item -reinit_filter[:@var{stream_specifier}] @var{integer} (@emph{input,per-stream})
-This boolean option determines if the filtergraph(s) to which this stream is fed gets
-reinitialized when input frame parameters change mid-stream. This option is enabled by
-default as most video and all audio filters cannot handle deviation in input frame properties.
-Upon reinitialization, existing filter state is lost, like e.g. the frame count @code{n}
-reference available in some filters. Any frames buffered at time of reinitialization are lost.
-The properties where a change triggers reinitialization are,
-for video, frame resolution or pixel format;
-for audio, sample format, sample rate, channel count or channel layout.
-
 @item -filter_threads @var{nb_threads} (@emph{global})
 Defines how many threads are used to process a filter pipeline. Each pipeline
 will produce a thread pool with this many threads available for parallel processing.
@@ -877,20 +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).
 
@@ -923,32 +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}
@@ -1022,9 +960,14 @@ 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}.
 @item -vstats_file @var{file}
@@ -1041,6 +984,8 @@ 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})
@@ -1051,7 +996,6 @@ Deprecated see -bsf
 @item -force_key_frames[:@var{stream_specifier}] @var{time}[,@var{time}...] (@emph{output,per-stream})
 @item -force_key_frames[:@var{stream_specifier}] expr:@var{expr} (@emph{output,per-stream})
 @item -force_key_frames[:@var{stream_specifier}] source (@emph{output,per-stream})
-@item -force_key_frames[:@var{stream_specifier}] source_no_drop (@emph{output,per-stream})
 
 @var{force_key_frames} can take arguments of the following form:
 
@@ -1113,12 +1057,6 @@ starting from second 13:
 If the argument is @code{source}, ffmpeg will force a key frame if
 the current frame being encoded is marked as a key frame in its source.
 
-@item source_no_drop
-If the argument is @code{source_no_drop}, ffmpeg will force a key frame if
-the current frame being encoded is marked as a key frame in its source.
-In cases where this particular source frame has to be dropped,
-enforce the next available frame to become a key frame instead.
-
 @end table
 
 Note that forcing too many keyframes is very harmful for the lookahead
@@ -1141,27 +1079,9 @@ device type:
 @item cuda
 @var{device} is the number of the CUDA device.
 
-The following options are recognized:
-@table @option
-@item primary_ctx
-If set to 1, uses the primary device context instead of creating a new one.
-@end table
-
-Examples:
-@table @emph
-@item -init_hw_device cuda:1
-Choose the second device on the system.
-
-@item -init_hw_device cuda:0,primary_ctx=1
-Choose the first device and use the primary device context.
-@end table
-
 @item dxva2
 @var{device} is the number of the Direct3D 9 display adapter.
 
-@item d3d11va
-@var{device} is the number of the Direct3D 11 display adapter.
-
 @item vaapi
 @var{device} is either an X11 display name or a DRM render node.
 If not specified, it will attempt to open the default X11 display (@emph{$DISPLAY})
@@ -1185,21 +1105,9 @@ If not specified, it will attempt to open the default X11 display (@emph{$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}.
 
@@ -1302,9 +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.
 
@@ -1338,25 +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
 
@@ -1456,18 +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.
@@ -1481,56 +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:
@@ -1538,13 +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
@@ -1555,10 +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.
@@ -1686,44 +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 -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
 
@@ -1742,6 +1596,18 @@ 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
@@ -1820,23 +1686,7 @@ 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.
-
+Finish encoding when the shortest input stream ends.
 @item -dts_delta_threshold
 Timestamp discontinuity delta threshold.
 @item -dts_error_threshold @var{seconds}
@@ -1971,15 +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}.
@@ -2054,123 +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.
-
-@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
-Encoder timebase, as a rational number @var{num/den}. Note that this may be
-different from the timebase used by the muxer.
-
-@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
-Decoding timestamp of the packet, as an integer. Should be multiplied by the
-timebase to compute presentation time. Post-encoding only.
-
-@item dt
-Decoding time of the frame or packet, as a decimal number. Equal to
-@var{dts} multiplied by @var{tb}.
-
-@item sn
-Number of audio samples sent to the encoder so far. Audio and pre-encoding only.
-
-@item samp
-Number of audio samples in the frame. Audio and pre-encoding only.
-
-@item size
-Size of the encoded packet in bytes. Post-encoding only.
-
-@item br
-Current bitrate in bits per second. Post-encoding only.
-
-@item abr
-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
-
-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

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.
 

doc/ffprobe.texi

@@ -12,7 +12,7 @@
 
 @chapter Synopsis
 
-ffprobe [@var{options}] @file{input_url}
+ffprobe [@var{options}] [@file{input_url}]
 
 @chapter Description
 @c man begin DESCRIPTION
@@ -28,9 +28,6 @@ If a url is specified in input, ffprobe will try to open and
 probe the url content. If the url cannot be opened or recognized as
 a multimedia file, a positive exit code is returned.
 
-If no output is specified as output with @option{o} ffprobe will write
-to stdout.
-
 ffprobe may be employed both as a standalone application or in
 combination with a textual filter, which may perform more
 sophisticated processing, e.g. statistical processing or plotting.
@@ -338,12 +335,6 @@ Show information about all pixel formats supported by FFmpeg.
 Pixel format information for each format is printed within a section
 with name "PIXEL_FORMAT".
 
-@item -show_optional_fields @var{value}
-Some writers viz. JSON and XML, omit the printing of fields with invalid or non-applicable values,
-while other writers always print them. This option enables one to control this behaviour.
-Valid values are @code{always}/@code{1}, @code{never}/@code{0} and @code{auto}/@code{-1}.
-Default is @var{auto}.
-
 @item -bitexact
 Force bitexact output, useful to produce output which is not dependent
 on the specific build.
@@ -351,10 +342,6 @@ on the specific build.
 @item -i @var{input_url}
 Read @var{input_url}.
 
-@item -o @var{output_url}
-Write output to @var{output_url}. If not specified, the output is sent
-to stdout.
-
 @end table
 @c man end
 

doc/ffprobe.xsd

@@ -29,18 +29,22 @@
     </xsd:complexType>
 
     <xsd:complexType name="framesType">
-        <xsd:choice minOccurs="0" maxOccurs="unbounded">
-            <xsd:element name="frame"    type="ffprobe:frameType"/>
-            <xsd:element name="subtitle" type="ffprobe:subtitleType"/>
-        </xsd:choice>
+        <xsd:sequence>
+            <xsd:choice minOccurs="0" maxOccurs="unbounded">
+                <xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
+                <xsd:element name="subtitle" type="ffprobe:subtitleType" minOccurs="0" maxOccurs="unbounded"/>
+            </xsd:choice>
+        </xsd:sequence>
     </xsd:complexType>
 
     <xsd:complexType name="packetsAndFramesType">
-        <xsd:choice minOccurs="0" maxOccurs="unbounded">
-            <xsd:element name="packet"   type="ffprobe:packetType"/>
-            <xsd:element name="frame"    type="ffprobe:frameType"/>
-            <xsd:element name="subtitle" type="ffprobe:subtitleType"/>
-        </xsd:choice>
+        <xsd:sequence>
+            <xsd:choice minOccurs="0" maxOccurs="unbounded">
+                <xsd:element name="packet" type="ffprobe:packetType" minOccurs="0" maxOccurs="unbounded"/>
+                <xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
+                <xsd:element name="subtitle" type="ffprobe:subtitleType" minOccurs="0" maxOccurs="unbounded"/>
+            </xsd:choice>
+        </xsd:sequence>
     </xsd:complexType>
 
     <xsd:complexType name="packetType">
@@ -86,14 +90,14 @@
       <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="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" />
 
@@ -195,11 +199,6 @@
       <xsd:attribute name="clean_effects"    type="xsd:int" use="required" />
       <xsd:attribute name="attached_pic"     type="xsd:int" use="required" />
       <xsd:attribute name="timed_thumbnails" type="xsd:int" use="required" />
-      <xsd:attribute name="captions"         type="xsd:int" use="required" />
-      <xsd:attribute name="descriptions"     type="xsd:int" use="required" />
-      <xsd:attribute name="metadata"         type="xsd:int" use="required" />
-      <xsd:attribute name="dependent"        type="xsd:int" use="required" />
-      <xsd:attribute name="still_image"      type="xsd:int" use="required" />
     </xsd:complexType>
 
     <xsd:complexType name="streamType">
@@ -217,7 +216,6 @@
       <xsd:attribute name="codec_tag"        type="xsd:string" use="required"/>
       <xsd:attribute name="codec_tag_string" type="xsd:string" use="required"/>
       <xsd:attribute name="extradata"        type="xsd:string" />
-      <xsd:attribute name="extradata_size"   type="xsd:int"    />
       <xsd:attribute name="extradata_hash"   type="xsd:string" />
 
       <!-- video attributes -->
@@ -226,7 +224,6 @@
       <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"/>
@@ -246,7 +243,6 @@
       <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:attribute name="id"               type="xsd:string"/>
       <xsd:attribute name="r_frame_rate"     type="xsd:string" use="required"/>
@@ -273,6 +269,10 @@
       <xsd:attribute name="program_id"           type="xsd:int"    use="required"/>
       <xsd:attribute name="program_num"          type="xsd:int"    use="required"/>
       <xsd:attribute name="nb_streams"           type="xsd:int"    use="required"/>
+      <xsd:attribute name="start_time"           type="xsd:float"/>
+      <xsd:attribute name="start_pts"            type="xsd:long"/>
+      <xsd:attribute name="end_time"             type="xsd:float"/>
+      <xsd:attribute name="end_pts"              type="xsd:long"/>
       <xsd:attribute name="pmt_pid"              type="xsd:int"    use="required"/>
       <xsd:attribute name="pcr_pid"              type="xsd:int"    use="required"/>
     </xsd:complexType>

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

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

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
@@ -614,7 +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 SBC                   @tab X @tab X
 @item raw Shorten               @tab   @tab X
 @item raw TAK                   @tab   @tab X
@@ -666,10 +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
@@ -707,21 +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
@@ -761,17 +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
@@ -790,8 +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
@@ -802,8 +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
@@ -812,10 +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
@@ -1052,7 +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
@@ -1160,7 +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
@@ -1168,7 +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
@@ -1198,8 +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
@@ -1220,7 +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
@@ -1228,7 +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
@@ -1236,12 +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
@@ -1262,7 +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
@@ -1270,14 +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
@@ -1287,7 +1242,6 @@ 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
@@ -1322,7 +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
@@ -1335,7 +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
@@ -1347,7 +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
@@ -1356,11 +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

doc/git-howto.texi

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

doc/indevs.texi

@@ -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
@@ -1289,11 +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.

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.

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
 
@@ -799,7 +642,15 @@ were recently referenced in the playlist. Default value is 1, meaning segments o
 Set output format options using a :-separated list of key=value
 parameters. Values containing @code{:} special characters must be
 escaped.
-@code{hls_ts_options} is deprecated, use hls_segment_options instead of it..
+
+@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.
@@ -886,6 +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
@@ -903,6 +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}.
@@ -920,10 +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
@@ -1060,8 +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
@@ -1427,10 +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.
@@ -1569,27 +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.
@@ -1733,14 +1571,6 @@ B-frames. Additionally, eases conformance with the DASH-IF interoperability
 guidelines.
 
 This option is implicitly set when writing ismv (Smooth Streaming) files.
-
-@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
 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
@@ -1751,19 +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
@@ -1913,10 +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}
@@ -1938,11 +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
@@ -2054,182 +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 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
 
@@ -2369,11 +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
@@ -2603,6 +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
 
@@ -2735,49 +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.
@@ -2844,4 +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

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

doc/outdevs.texi

@@ -198,43 +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}.
-
 @end table
 
 @subsection Examples

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.
 
@@ -941,11 +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
@@ -1153,10 +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}.
 
@@ -1204,59 +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
@@ -1282,10 +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
@@ -1303,9 +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}.
@@ -1318,7 +1154,6 @@ The following flags are accepted:
 @item video
 @item audio
 @item data
-@item subtitle
 @end table
 
 By default it accepts all media types.
@@ -1329,23 +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
@@ -1630,12 +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
@@ -1735,9 +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
@@ -1787,11 +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}.
@@ -1900,8 +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
@@ -2155,4 +1972,5 @@ decoding errors.
 
 @end table
 
+
 @c man end PROTOCOLS

doc/t2h.pm

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

doc/utils.texi

@@ -713,42 +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 7.1(top)
-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 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
@@ -765,7 +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
 
@@ -1077,13 +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

doc/writing_filters.txt

@@ -418,4 +418,4 @@ done:
 
 When all of this is done, you can submit your patch to the ffmpeg-devel
 mailing-list for review.  If you need any help, feel free to come on our IRC
-channel, #ffmpeg-devel on irc.libera.chat.
+channel, #ffmpeg-devel on irc.freenode.net.

ffbuild/.gitignore

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

ffbuild/arch.mak

@@ -8,14 +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)

ffbuild/bin2c.c

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

ffbuild/common.mak

@@ -12,13 +12,10 @@ endif
 
 ifndef SUBDIR
 
-BIN2CEXE = ffbuild/bin2c$(HOSTEXESUF)
-BIN2C = $(BIN2CEXE)
-
 ifndef V
 Q      = @
 ECHO   = printf "$(1)\t%s\n" $(2)
-BRIEF  = CC CXX OBJCC HOSTCC HOSTLD AS X86ASM AR LD STRIP CP WINDRES NVCC BIN2C
+BRIEF  = CC CXX OBJCC HOSTCC HOSTLD AS X86ASM AR LD STRIP CP WINDRES NVCC
 SILENT = DEPCC DEPHOSTCC DEPAS DEPX86ASM RANLIB RM
 
 MSG    = $@
@@ -29,8 +26,7 @@ $(foreach VAR,$(SILENT),$(eval override $(VAR) = @$($(VAR))))
 $(eval INSTALL = @$(call ECHO,INSTALL,$$(^:$(SRC_DIR)/%=%)); $(INSTALL))
 endif
 
-# Prepend to a recursively expanded variable without making it simply expanded.
-PREPEND = $(eval $(1) = $(patsubst %,$$(%), $(2)) $(value $(1)))
+ALLFFLIBS = avcodec avdevice avfilter avformat avresample avutil postproc swscale swresample
 
 # NASM requires -I path terminated with /
 IFLAGS     := -I. -I$(SRC_LINK)/
@@ -40,9 +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)
 
@@ -104,7 +90,7 @@ COMPILE_LASX = $(call COMPILE,CC,LASXFLAGS)
 	-$(if $(ASMSTRIPFLAGS), $(STRIP) $(ASMSTRIPFLAGS) $@)
 
 %.o: %.rc
-	$(WINDRES) $(IFLAGS) $(foreach ARG,$(CC_DEPFLAGS),--preprocessor-arg "$(ARG)") -o $@ $<
+	$(WINDRES) $(IFLAGS) --preprocessor "$(DEPWINDRES) -E -xc-header -DRC_INVOKED $(CC_DEPFLAGS)" -o $@ $<
 
 %.i: %.c
 	$(CC) $(CCFLAGS) $(CC_E) $<
@@ -112,35 +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)

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)

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"

fftools/Makefile

@@ -9,22 +9,15 @@ AVBASENAMES  = ffmpeg ffplay ffprobe
 ALLAVPROGS   = $(AVBASENAMES:%=%$(PROGSSUF)$(EXESUF))
 ALLAVPROGS_G = $(AVBASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
 
-OBJS-ffmpeg +=                  \
-    fftools/ffmpeg_demux.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))

fftools/cmdutils.h

@@ -44,9 +44,11 @@ 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;
 
 /**
@@ -54,17 +56,6 @@ extern int hide_banner;
  */
 void register_exit(void (*cb)(int ret));
 
-/**
- * Reports an error corresponding to the provided
- * AVERROR code and calls exit_program() with the
- * corresponding POSIX error code.
- * @note ret must be an AVERROR-value of a POSIX error code
- *       (i.e. AVERROR(EFOO) and not AVERROR_FOO).
- *       library functions can return both, so call this only
- *       with AVERROR(EFOO) of your own.
- */
-void report_and_exit(int ret) av_noreturn;
-
 /**
  * Wraps exit with a program-specific cleanup routine.
  */
@@ -75,6 +66,11 @@ void exit_program(int ret) av_noreturn;
  */
 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.
@@ -87,12 +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.
  */
@@ -189,6 +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.
@@ -201,6 +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.
  *
@@ -259,6 +317,7 @@ typedef struct OptionGroup {
 
     AVDictionary *codec_opts;
     AVDictionary *format_opts;
+    AVDictionary *resample_opts;
     AVDictionary *sws_dict;
     AVDictionary *swr_opts;
 } OptionGroup;
@@ -365,8 +424,8 @@ AVDictionary *filter_codec_opts(AVDictionary *opts, enum AVCodecID codec_id,
  * 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.
- * Calls exit() on failure.
+ * @return pointer to the created array of dictionaries, NULL if it
+ * cannot be created
  */
 AVDictionary **setup_find_stream_info_opts(AVFormatContext *s,
                                            AVDictionary *codec_opts);
@@ -389,6 +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.
@@ -428,26 +617,11 @@ FILE *get_preset_file(char *filename, size_t filename_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.
- * Calls exit() on failure.
- *
- * @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
- */
-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)\
     array = grow_array(array, sizeof(*array), &nb_elems, nb_elems + 1)
 
-#define ALLOC_ARRAY_ELEM(array, nb_elems)\
-    allocate_array_elem(&array, sizeof(*array[0]), &nb_elems)
-
 #define GET_PIX_FMT_NAME(pix_fmt)\
     const char *name = av_get_pix_fmt_name(pix_fmt);
 
@@ -461,6 +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(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 */

fftools/ffmpeg.h

@@ -21,19 +21,16 @@
 
 #include "config.h"
 
-#include <stdatomic.h>
 #include <stdint.h>
 #include <stdio.h>
 #include <signal.h>
 
 #include "cmdutils.h"
-#include "sync_queue.h"
 
 #include "libavformat/avformat.h"
 #include "libavformat/avio.h"
 
 #include "libavcodec/avcodec.h"
-#include "libavcodec/bsf.h"
 
 #include "libavfilter/avfilter.h"
 
@@ -49,20 +46,12 @@
 
 #include "libswresample/swresample.h"
 
-// deprecated features
-#define FFMPEG_OPT_PSNR 1
-#define FFMPEG_OPT_MAP_CHANNEL 1
-#define FFMPEG_OPT_MAP_SYNC 1
-#define FFMPEG_ROTATION_METADATA 1
-
-enum VideoSyncMethod {
-    VSYNC_AUTO = -1,
-    VSYNC_PASSTHROUGH,
-    VSYNC_CFR,
-    VSYNC_VFR,
-    VSYNC_VSCFR,
-    VSYNC_DROP,
-};
+#define VSYNC_AUTO       -1
+#define VSYNC_PASSTHROUGH 0
+#define VSYNC_CFR         1
+#define VSYNC_VFR         2
+#define VSYNC_VSCFR       0xfe
+#define VSYNC_DROP        0xff
 
 #define MAX_STREAMS 1024    /* arbitrary sanity check value */
 
@@ -70,8 +59,17 @@ enum HWAccelID {
     HWACCEL_NONE = 0,
     HWACCEL_AUTO,
     HWACCEL_GENERIC,
+    HWACCEL_VIDEOTOOLBOX,
+    HWACCEL_QSV,
 };
 
+typedef struct HWAccel {
+    const char *name;
+    int (*init)(AVCodecContext *s);
+    enum HWAccelID id;
+    enum AVPixelFormat pix_fmt;
+} HWAccel;
+
 typedef struct HWDevice {
     const char *name;
     enum AVHWDeviceType type;
@@ -83,15 +81,15 @@ typedef struct StreamMap {
     int disabled;           /* 1 is this mapping is disabled by a negative map */
     int file_index;
     int stream_index;
+    int sync_file_index;
+    int sync_stream_index;
     char *linklabel;       /* name of an output link, for mapping lavfi outputs */
 } StreamMap;
 
-#if FFMPEG_OPT_MAP_CHANNEL
 typedef struct {
     int  file_idx,  stream_idx,  channel_idx; // input
     int ofile_idx, ostream_idx;               // output
 } AudioChannelMap;
-#endif
 
 typedef struct OptionsContext {
     OptionGroup *g;
@@ -104,8 +102,6 @@ typedef struct OptionsContext {
 
     SpecifierOpt *codec_names;
     int        nb_codec_names;
-    SpecifierOpt *audio_ch_layouts;
-    int        nb_audio_ch_layouts;
     SpecifierOpt *audio_channels;
     int        nb_audio_channels;
     SpecifierOpt *audio_sample_rate;
@@ -123,11 +119,8 @@ typedef struct OptionsContext {
     int64_t input_ts_offset;
     int loop;
     int rate_emu;
-    float readrate;
     int accurate_seek;
     int thread_queue_size;
-    int input_sync_ref;
-    int find_stream_info;
 
     SpecifierOpt *ts_scale;
     int        nb_ts_scale;
@@ -145,10 +138,11 @@ typedef struct OptionsContext {
     /* output options */
     StreamMap *stream_maps;
     int     nb_stream_maps;
-#if FFMPEG_OPT_MAP_CHANNEL
     AudioChannelMap *audio_channel_maps; /* one info entry per -map_channel */
     int           nb_audio_channel_maps; /* number of (valid) -map_channel settings */
-#endif
+    int metadata_global_manual;
+    int metadata_streams_manual;
+    int metadata_chapters_manual;
     const char **attachments;
     int       nb_attachments;
 
@@ -156,10 +150,9 @@ typedef struct OptionsContext {
 
     int64_t recording_time;
     int64_t stop_time;
-    int64_t limit_filesize;
+    uint64_t limit_filesize;
     float mux_preload;
     float mux_max_delay;
-    float shortest_buf_duration;
     int shortest;
     int bitexact;
 
@@ -186,18 +179,10 @@ typedef struct OptionsContext {
     int        nb_qscale;
     SpecifierOpt *forced_key_frames;
     int        nb_forced_key_frames;
-    SpecifierOpt *fps_mode;
-    int        nb_fps_mode;
     SpecifierOpt *force_fps;
     int        nb_force_fps;
     SpecifierOpt *frame_aspect_ratios;
     int        nb_frame_aspect_ratios;
-    SpecifierOpt *display_rotations;
-    int        nb_display_rotations;
-    SpecifierOpt *display_hflips;
-    int        nb_display_hflips;
-    SpecifierOpt *display_vflips;
-    int        nb_display_vflips;
     SpecifierOpt *rc_overrides;
     int        nb_rc_overrides;
     SpecifierOpt *intra_matrices;
@@ -224,8 +209,6 @@ typedef struct OptionsContext {
     int        nb_reinit_filters;
     SpecifierOpt *fix_sub_duration;
     int        nb_fix_sub_duration;
-    SpecifierOpt *fix_sub_duration_heartbeat;
-    int        nb_fix_sub_duration_heartbeat;
     SpecifierOpt *canvas_sizes;
     int        nb_canvas_sizes;
     SpecifierOpt *pass;
@@ -252,20 +235,6 @@ typedef struct OptionsContext {
     int        nb_enc_time_bases;
     SpecifierOpt *autoscale;
     int        nb_autoscale;
-    SpecifierOpt *bits_per_raw_sample;
-    int        nb_bits_per_raw_sample;
-    SpecifierOpt *enc_stats_pre;
-    int        nb_enc_stats_pre;
-    SpecifierOpt *enc_stats_post;
-    int        nb_enc_stats_post;
-    SpecifierOpt *mux_stats;
-    int        nb_mux_stats;
-    SpecifierOpt *enc_stats_pre_fmt;
-    int        nb_enc_stats_pre_fmt;
-    SpecifierOpt *enc_stats_post_fmt;
-    int        nb_enc_stats_post_fmt;
-    SpecifierOpt *mux_stats_fmt;
-    int        nb_mux_stats_fmt;
 } OptionsContext;
 
 typedef struct InputFilter {
@@ -275,7 +244,7 @@ typedef struct InputFilter {
     uint8_t            *name;
     enum AVMediaType    type;   // AVMEDIA_TYPE_SUBTITLE for sub2video
 
-    AVFifo *frame_queue;
+    AVFifoBuffer *frame_queue;
 
     // parameters configured for this input
     int format;
@@ -284,10 +253,10 @@ typedef struct InputFilter {
     AVRational sample_aspect_ratio;
 
     int sample_rate;
-    AVChannelLayout ch_layout;
+    int channels;
+    uint64_t channel_layout;
 
     AVBufferRef *hw_frames_ctx;
-    int32_t *displaymatrix;
 
     int eof;
 } InputFilter;
@@ -307,13 +276,12 @@ typedef struct OutputFilter {
     AVRational frame_rate;
     int format;
     int sample_rate;
-    AVChannelLayout ch_layout;
+    uint64_t channel_layout;
 
     // those are only set if no format is specified and the encoder gives us multiple options
-    // They point directly to the relevant lists of the encoder.
-    const int *formats;
-    const AVChannelLayout *ch_layouts;
-    const int *sample_rates;
+    int *formats;
+    uint64_t *channel_layouts;
+    int *sample_rates;
 } OutputFilter;
 
 typedef struct FilterGraph {
@@ -322,9 +290,6 @@ typedef struct FilterGraph {
 
     AVFilterGraph *graph;
     int reconfiguration;
-    // true when the filtergraph contains only meta filters
-    // that do not modify the frame data
-    int is_meta;
 
     InputFilter   **inputs;
     int          nb_inputs;
@@ -340,41 +305,23 @@ typedef struct InputStream {
     int decoding_needed;     /* non zero if the packets must be decoded in 'raw_fifo', see DECODING_FOR_* */
 #define DECODING_FOR_OST    1
 #define DECODING_FOR_FILTER 2
-    int processing_needed;   /* non zero if the packets must be processed */
-    // should attach FrameData as opaque_ref after decoding
-    int want_frame_data;
 
-    /**
-     * Codec parameters - to be used by the decoding/streamcopy code.
-     * st->codecpar should not be accessed, because it may be modified
-     * concurrently by the demuxing thread.
-     */
-    AVCodecParameters *par;
     AVCodecContext *dec_ctx;
     const AVCodec *dec;
     AVFrame *decoded_frame;
+    AVFrame *filter_frame; /* a ref of decoded_frame, to be sent to filters */
     AVPacket *pkt;
 
-    AVRational framerate_guessed;
-
-    int64_t       prev_pkt_pts;
     int64_t       start;     /* time when read started */
     /* predicted dts of the next packet read for this stream or (when there are
      * several frames in a packet) of the next frame in current packet (in AV_TIME_BASE units) */
     int64_t       next_dts;
-    int64_t first_dts;       ///< dts of the first packet read for this stream (in AV_TIME_BASE units)
     int64_t       dts;       ///< dts of the last packet read for this stream (in AV_TIME_BASE units)
 
     int64_t       next_pts;  ///< synthetic pts for the next decode frame (in AV_TIME_BASE units)
     int64_t       pts;       ///< current pts of the decoded frame  (in AV_TIME_BASE units)
     int           wrap_correction_done;
 
-    // the value of AVCodecParserContext.repeat_pict from the AVStream parser
-    // for the last packet returned from ifile_get_packet()
-    // -1 if unknown
-    // FIXME: this is a hack, the avstream parser should not be used
-    int last_pkt_repeat_pict;
-
     int64_t filter_in_rescale_delta_last;
 
     int64_t min_pts; /* pts with the smallest value in a current stream */
@@ -405,12 +352,14 @@ typedef struct InputStream {
     struct sub2video {
         int64_t last_pts;
         int64_t end_pts;
-        AVFifo *sub_queue;    ///< queue of AVSubtitle* before filter init
+        AVFifoBuffer *sub_queue;    ///< queue of AVSubtitle* before filter init
         AVFrame *frame;
         int w, h;
         unsigned int initialize; ///< marks if sub2video_update should force an initialization
     } sub2video;
 
+    int dr1;
+
     /* decoded data from this stream goes into all those filters
      * currently video and audio only */
     InputFilter **filters;
@@ -424,8 +373,14 @@ typedef struct InputStream {
     char  *hwaccel_device;
     enum AVPixelFormat hwaccel_output_format;
 
+    /* hwaccel context */
+    void  *hwaccel_ctx;
+    void (*hwaccel_uninit)(AVCodecContext *s);
+    int  (*hwaccel_get_buffer)(AVCodecContext *s, AVFrame *frame, int flags);
     int  (*hwaccel_retrieve_data)(AVCodecContext *s, AVFrame *frame);
     enum AVPixelFormat hwaccel_pix_fmt;
+    enum AVPixelFormat hwaccel_retrieved_pix_fmt;
+    AVBufferRef *hw_frames_ctx;
 
     /* stats */
     // combined size of all the packets read
@@ -442,46 +397,37 @@ typedef struct InputStream {
     int got_output;
 } InputStream;
 
-typedef struct LastFrameDuration {
-    int     stream_idx;
-    int64_t duration;
-} LastFrameDuration;
-
 typedef struct InputFile {
-    int index;
-
     AVFormatContext *ctx;
     int eof_reached;      /* true if eof reached */
     int eagain;           /* true if last read attempt returned EAGAIN */
+    int ist_index;        /* index of first stream in input_streams */
+    int loop;             /* set number of times input stream should be looped */
+    int64_t duration;     /* actual duration of the longest stream in a file
+                             at the moment when looping happens */
+    AVRational time_base; /* time base of the duration */
     int64_t input_ts_offset;
-    int input_sync_ref;
-    /**
-     * Effective format start time based on enabled streams.
-     */
-    int64_t start_time_effective;
+
     int64_t ts_offset;
-    /**
-     * Extra timestamp offset added by discontinuity handling.
-     */
-    int64_t ts_offset_discont;
     int64_t last_ts;
     int64_t start_time;   /* user-specified start time in AV_TIME_BASE or AV_NOPTS_VALUE */
+    int seek_timestamp;
     int64_t recording_time;
-
-    /* streams that ffmpeg is aware of;
-     * there may be extra streams in ctx that are not mapped to an InputStream
-     * if new streams appear dynamically during demuxing */
-    InputStream **streams;
-    int        nb_streams;
-
+    int nb_streams;       /* number of stream that ffmpeg is aware of; may be different
+                             from ctx.nb_streams if new streams appear during av_read_frame() */
+    int nb_streams_warn;  /* number of streams that the user was warned of */
     int rate_emu;
-    float readrate;
     int accurate_seek;
 
-    /* when looping the input file, this queue is used by decoders to report
-     * the last frame duration back to the demuxer thread */
-    AVThreadMessageQueue *audio_duration_queue;
-    int                   audio_duration_queue_size;
+    AVPacket *pkt;
+
+#if HAVE_THREADS
+    AVThreadMessageQueue *in_thread_queue;
+    pthread_t thread;           /* thread reading from this file */
+    int non_blocking;           /* reading packets from the thread should not block */
+    int joined;                 /* the thread has been joined */
+    int thread_queue_size;      /* maximum number of queued packets */
+#endif
 } InputFile;
 
 enum forced_keyframes_const {
@@ -496,41 +442,6 @@ enum forced_keyframes_const {
 #define ABORT_ON_FLAG_EMPTY_OUTPUT        (1 <<  0)
 #define ABORT_ON_FLAG_EMPTY_OUTPUT_STREAM (1 <<  1)
 
-enum EncStatsType {
-    ENC_STATS_LITERAL = 0,
-    ENC_STATS_FILE_IDX,
-    ENC_STATS_STREAM_IDX,
-    ENC_STATS_FRAME_NUM,
-    ENC_STATS_FRAME_NUM_IN,
-    ENC_STATS_TIMEBASE,
-    ENC_STATS_TIMEBASE_IN,
-    ENC_STATS_PTS,
-    ENC_STATS_PTS_TIME,
-    ENC_STATS_PTS_IN,
-    ENC_STATS_PTS_TIME_IN,
-    ENC_STATS_DTS,
-    ENC_STATS_DTS_TIME,
-    ENC_STATS_SAMPLE_NUM,
-    ENC_STATS_NB_SAMPLES,
-    ENC_STATS_PKT_SIZE,
-    ENC_STATS_BITRATE,
-    ENC_STATS_AVG_BITRATE,
-};
-
-typedef struct EncStatsComponent {
-    enum EncStatsType type;
-
-    uint8_t *str;
-    size_t   str_len;
-} EncStatsComponent;
-
-typedef struct EncStats {
-    EncStatsComponent  *components;
-    int              nb_components;
-
-    AVIOContext        *io;
-} EncStats;
-
 extern const char *const forced_keyframes_const_names[];
 
 typedef enum {
@@ -538,92 +449,64 @@ typedef enum {
     MUXER_FINISHED = 2,
 } OSTFinished ;
 
-enum {
-    KF_FORCE_SOURCE         = 1,
-    KF_FORCE_SOURCE_NO_DROP = 2,
-};
-
-typedef struct KeyframeForceCtx {
-    int          type;
-
-    int64_t      ref_pts;
-
-    // timestamps of the forced keyframes, in AV_TIME_BASE_Q
-    int64_t     *pts;
-    int       nb_pts;
-    int          index;
-
-    AVExpr      *pexpr;
-    double       expr_const_values[FKF_NB];
-
-    int          dropped_keyframe;
-} KeyframeForceCtx;
-
 typedef struct OutputStream {
-    const AVClass *class;
-
     int file_index;          /* file index */
     int index;               /* stream index in the output file */
-
-    /* input stream that is the source for this output stream;
-     * may be NULL for streams with no well-defined source, e.g.
-     * attachments or outputs from complex filtergraphs */
-    InputStream *ist;
-
+    int source_index;        /* InputStream index */
     AVStream *st;            /* stream in the output file */
-    /* number of frames emitted by the video-encoding sync code */
-    int64_t vsync_frame_number;
-    /* predicted pts of the next frame to be encoded
-     * audio/video encoding only */
-    int64_t next_pts;
-    /* dts of the last packet sent to the muxing queue, in AV_TIME_BASE_Q */
+    int encoding_needed;     /* true if encoding needed for this stream */
+    int frame_number;
+    /* input pts and corresponding output pts
+       for A/V sync */
+    struct InputStream *sync_ist; /* input stream to sync against */
+    int64_t sync_opts;       /* output frame counter, could be changed to some true timestamp */ // FIXME look at frame_number
+    /* pts of the first frame encoded for this stream, used for limiting
+     * recording time */
+    int64_t first_pts;
+    /* dts of the last packet sent to the muxer */
     int64_t last_mux_dts;
-    /* pts of the last frame received from the filters, in AV_TIME_BASE_Q */
-    int64_t last_filter_pts;
-
-    // timestamp from which the streamcopied streams should start,
-    // in AV_TIME_BASE_Q;
-    // everything before it should be discarded
-    int64_t ts_copy_start;
-
     // the timebase of the packets sent to the muxer
     AVRational mux_timebase;
     AVRational enc_timebase;
 
+    AVBSFContext            *bsf_ctx;
+
     AVCodecContext *enc_ctx;
+    AVCodecParameters *ref_par; /* associated input codec parameters with encoders options applied */
+    const AVCodec *enc;
+    int64_t max_frames;
     AVFrame *filtered_frame;
     AVFrame *last_frame;
-    AVFrame *sq_frame;
     AVPacket *pkt;
-    int64_t last_dropped;
-    int64_t last_nb0_frames[3];
+    int last_dropped;
+    int last_nb0_frames[3];
+
+    void  *hwaccel_ctx;
 
     /* video only */
     AVRational frame_rate;
     AVRational max_frame_rate;
-    enum VideoSyncMethod vsync_method;
     int is_cfr;
     int force_fps;
     int top_field_first;
-#if FFMPEG_ROTATION_METADATA
     int rotate_overridden;
-#endif
     int autoscale;
-    int bitexact;
-    int bits_per_raw_sample;
-#if FFMPEG_ROTATION_METADATA
     double rotate_override_value;
-#endif
 
     AVRational frame_aspect_ratio;
 
-    KeyframeForceCtx kf;
+    /* forced key frames */
+    int64_t forced_kf_ref_pts;
+    int64_t *forced_kf_pts;
+    int forced_kf_count;
+    int forced_kf_index;
+    char *forced_keyframes;
+    AVExpr *forced_keyframes_pexpr;
+    double forced_keyframes_expr_const_values[FKF_NB];
 
     /* audio only */
-#if FFMPEG_OPT_MAP_CHANNEL
     int *audio_channels_map;             /* list of the channels id to pick from the source stream */
     int audio_channels_mapped;           /* number of channels in audio_channels_map */
-#endif
 
     char *logfile_prefix;
     FILE *logfile;
@@ -636,9 +519,11 @@ typedef struct OutputStream {
     AVDictionary *encoder_opts;
     AVDictionary *sws_dict;
     AVDictionary *swr_opts;
+    AVDictionary *resample_opts;
     char *apad;
     OSTFinished finished;        /* no more packets should be written for this stream */
     int unavailable;                     /* true if the steram is unavailable (possibly temporarily) */
+    int stream_copy;
 
     // init_output_stream() has been called for this stream
     // The encoder and the bitstream filters have been initialized and the stream
@@ -648,70 +533,65 @@ typedef struct OutputStream {
     int inputs_done;
 
     const char *attachment_filename;
-    int streamcopy_started;
     int copy_initial_nonkeyframes;
     int copy_prior_start;
+    char *disposition;
 
     int keep_pix_fmt;
 
     /* stats */
-    // combined size of all the packets sent to the muxer
-    uint64_t data_size_mux;
-    // combined size of all the packets received from the encoder
-    uint64_t data_size_enc;
+    // combined size of all the packets written
+    uint64_t data_size;
     // number of packets send to the muxer
-    atomic_uint_least64_t packets_written;
+    uint64_t packets_written;
     // number of frames/samples sent to the encoder
     uint64_t frames_encoded;
     uint64_t samples_encoded;
-    // number of packets received from the encoder
-    uint64_t packets_encoded;
 
     /* packet quality factor */
     int quality;
 
+    int max_muxing_queue_size;
+
+    /* the packets are buffered here until the muxer is ready to be initialized */
+    AVFifoBuffer *muxing_queue;
+
+    /*
+     * The size of the AVPackets' buffers in queue.
+     * Updated when a packet is either pushed or pulled from the queue.
+     */
+    size_t muxing_queue_data_size;
+
+    /* Threshold after which max_muxing_queue_size will be in effect */
+    size_t muxing_queue_data_threshold;
+
     /* packet picture type */
     int pict_type;
 
     /* frame encode sum of squared error values */
     int64_t error[4];
-
-    int sq_idx_encode;
-    int sq_idx_mux;
-
-    EncStats enc_stats_pre;
-    EncStats enc_stats_post;
-
-    /*
-     * bool on whether this stream should be utilized for splitting
-     * subtitles utilizing fix_sub_duration at random access points.
-     */
-    unsigned int fix_sub_duration_heartbeat;
 } OutputStream;
 
 typedef struct OutputFile {
-    const AVClass *class;
-
-    int index;
-
-    const AVOutputFormat *format;
-    const char           *url;
-
-    OutputStream **streams;
-    int         nb_streams;
-
-    SyncQueue *sq_encode;
-
+    AVFormatContext *ctx;
+    AVDictionary *opts;
+    int ost_index;       /* index of the first stream in output_streams */
     int64_t recording_time;  ///< desired length of the resulting file in microseconds == AV_TIME_BASE units
     int64_t start_time;      ///< start time in microseconds == AV_TIME_BASE units
+    uint64_t limit_filesize; /* filesize limit expressed in bytes */
 
     int shortest;
-    int bitexact;
+
+    int header_written;
 } OutputFile;
 
+extern InputStream **input_streams;
+extern int        nb_input_streams;
 extern InputFile   **input_files;
 extern int        nb_input_files;
 
+extern OutputStream **output_streams;
+extern int         nb_output_streams;
 extern OutputFile   **output_files;
 extern int         nb_output_files;
 
@@ -725,10 +605,13 @@ extern float audio_drift_threshold;
 extern float dts_delta_threshold;
 extern float dts_error_threshold;
 
-extern enum VideoSyncMethod video_sync_method;
+extern int audio_volume;
+extern int audio_sync_method;
+extern int video_sync_method;
 extern float frame_drop_threshold;
 extern int do_benchmark;
 extern int do_benchmark_all;
+extern int do_deinterlace;
 extern int do_hex_dump;
 extern int do_pkt_dump;
 extern int copy_ts;
@@ -741,10 +624,12 @@ extern int print_stats;
 extern int64_t stats_period;
 extern int qp_hist;
 extern int stdin_interaction;
+extern int frame_bits_per_raw_sample;
 extern AVIOContext *progress_avio;
 extern float max_error_rate;
+extern char *videotoolbox_pixfmt;
 
-extern char *filter_nbthreads;
+extern int filter_nbthreads;
 extern int filter_complex_nbthreads;
 extern int vstats_version;
 extern int auto_conversion_filters;
@@ -752,19 +637,12 @@ extern int auto_conversion_filters;
 extern const AVIOInterruptCB int_cb;
 
 extern const OptionDef options[];
+extern const HWAccel hwaccels[];
+#if CONFIG_QSV
+extern char *qsv_device;
+#endif
 extern HWDevice *filter_hw_device;
 
-extern unsigned nb_output_dumped;
-extern int main_return_code;
-
-extern int ignore_unknown_streams;
-extern int copy_unknown_streams;
-
-extern int recast_media;
-
-#if FFMPEG_OPT_PSNR
-extern int do_psnr;
-#endif
 
 void term_init(void);
 void term_exit(void);
@@ -774,12 +652,7 @@ void show_usage(void);
 void remove_avoptions(AVDictionary **a, AVDictionary *b);
 void assert_avoptions(AVDictionary *m);
 
-void assert_file_overwrite(const char *filename);
-char *file_read(const char *filename);
-AVDictionary *strip_specifiers(const AVDictionary *dict);
-const AVCodec *find_codec_or_die(void *logctx, const char *name,
-                                 enum AVMediaType type, int encoder);
-int parse_and_set_vsync(const char *arg, int *vsync_var, int file_idx, int st_idx, int is_global);
+int guess_input_channel_layout(InputStream *ist);
 
 int configure_filtergraph(FilterGraph *fg);
 void check_filter_outputs(void);
@@ -793,9 +666,8 @@ int ifilter_parameters_from_frame(InputFilter *ifilter, const AVFrame *frame);
 
 int ffmpeg_parse_options(int argc, char **argv);
 
-void enc_stats_write(OutputStream *ost, EncStats *es,
-                     const AVFrame *frame, const AVPacket *pkt,
-                     uint64_t frame_num);
+int videotoolbox_init(AVCodecContext *s);
+int qsv_init(AVCodecContext *s);
 
 HWDevice *hw_device_get_by_name(const char *name);
 int hw_device_init_from_string(const char *arg, HWDevice **dev);
@@ -807,99 +679,4 @@ int hw_device_setup_for_filter(FilterGraph *fg);
 
 int hwaccel_decode_init(AVCodecContext *avctx);
 
-/*
- * Initialize muxing state for the given stream, should be called
- * after the codec/streamcopy setup has been done.
- *
- * Open the muxer once all the streams have been initialized.
- */
-int of_stream_init(OutputFile *of, OutputStream *ost);
-int of_write_trailer(OutputFile *of);
-int of_open(const OptionsContext *o, const char *filename);
-void of_close(OutputFile **pof);
-
-void of_enc_stats_close(void);
-
-/*
- * Send a single packet to the output, applying any bitstream filters
- * associated with the output stream.  This may result in any number
- * of packets actually being written, depending on what bitstream
- * filters are applied.  The supplied packet is consumed and will be
- * blank (as if newly-allocated) when this function returns.
- *
- * If eof is set, instead indicate EOF to all bitstream filters and
- * therefore flush any delayed packets to the output.  A blank packet
- * must be supplied in this case.
- */
-void of_output_packet(OutputFile *of, AVPacket *pkt, OutputStream *ost, int eof);
-int64_t of_filesize(OutputFile *of);
-
-int ifile_open(const OptionsContext *o, const char *filename);
-void ifile_close(InputFile **f);
-
-/**
- * Get next input packet from the demuxer.
- *
- * @param pkt the packet is written here when this function returns 0
- * @return
- * - 0 when a packet has been read successfully
- * - 1 when stream end was reached, but the stream is looped;
- *     caller should flush decoders and read from this demuxer again
- * - a negative error code on failure
- */
-int ifile_get_packet(InputFile *f, AVPacket **pkt);
-
-/* iterate over all input streams in all input files;
- * pass NULL to start iteration */
-InputStream *ist_iter(InputStream *prev);
-
-#define SPECIFIER_OPT_FMT_str  "%s"
-#define SPECIFIER_OPT_FMT_i    "%i"
-#define SPECIFIER_OPT_FMT_i64  "%"PRId64
-#define SPECIFIER_OPT_FMT_ui64 "%"PRIu64
-#define SPECIFIER_OPT_FMT_f    "%f"
-#define SPECIFIER_OPT_FMT_dbl  "%lf"
-
-#define WARN_MULTIPLE_OPT_USAGE(name, type, so, st)\
-{\
-    char namestr[128] = "";\
-    const char *spec = so->specifier && so->specifier[0] ? so->specifier : "";\
-    for (int _i = 0; opt_name_##name[_i]; _i++)\
-        av_strlcatf(namestr, sizeof(namestr), "-%s%s", opt_name_##name[_i], opt_name_##name[_i+1] ? (opt_name_##name[_i+2] ? ", " : " or ") : "");\
-    av_log(NULL, AV_LOG_WARNING, "Multiple %s options specified for stream %d, only the last option '-%s%s%s "SPECIFIER_OPT_FMT_##type"' will be used.\n",\
-           namestr, st->index, opt_name_##name[0], spec[0] ? ":" : "", spec, so->u.type);\
-}
-
-#define MATCH_PER_STREAM_OPT(name, type, outvar, fmtctx, st)\
-{\
-    int _ret, _matches = 0;\
-    SpecifierOpt *so;\
-    for (int _i = 0; _i < o->nb_ ## name; _i++) {\
-        char *spec = o->name[_i].specifier;\
-        if ((_ret = check_stream_specifier(fmtctx, st, spec)) > 0) {\
-            outvar = o->name[_i].u.type;\
-            so = &o->name[_i];\
-            _matches++;\
-        } else if (_ret < 0)\
-            exit_program(1);\
-    }\
-    if (_matches > 1)\
-       WARN_MULTIPLE_OPT_USAGE(name, type, so, st);\
-}
-
-#define MATCH_PER_TYPE_OPT(name, type, outvar, fmtctx, mediatype)\
-{\
-    int i;\
-    for (i = 0; i < o->nb_ ## name; i++) {\
-        char *spec = o->name[i].specifier;\
-        if (!strcmp(spec, mediatype))\
-            outvar = o->name[i].u.type;\
-    }\
-}
-
-extern const char * const opt_name_codec_names[];
-extern const char * const opt_name_codec_tags[];
-extern const char * const opt_name_frame_rates[];
-extern const char * const opt_name_top_field_first[];
-
 #endif /* FFTOOLS_FFMPEG_H */

fftools/ffmpeg_hw.c

@@ -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;
@@ -339,7 +325,7 @@ int hw_device_setup_for_decode(InputStream *ist)
             if (ist->hwaccel_id == HWACCEL_AUTO) {
                 ist->hwaccel_device_type = dev->type;
             } else if (ist->hwaccel_device_type != dev->type) {
-                av_log(NULL, AV_LOG_ERROR, "Invalid hwaccel device "
+                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),
@@ -353,18 +339,6 @@ int hw_device_setup_for_decode(InputStream *ist)
         } else if (ist->hwaccel_id == HWACCEL_GENERIC) {
             type = ist->hwaccel_device_type;
             dev = hw_device_get_by_type(type);
-
-            // When "-qsv_device device" is used, an internal QSV device named
-            // as "__qsv_device" is created. Another QSV device is created too
-            // if "-init_hw_device qsv=name:device" is used. There are 2 QSV devices
-            // if both "-qsv_device device" and "-init_hw_device qsv=name:device"
-            // are used, hw_device_get_by_type(AV_HWDEVICE_TYPE_QSV) returns NULL.
-            // To keep back-compatibility with the removed ad-hoc libmfx setup code,
-            // call hw_device_get_by_name("__qsv_device") to select the internal QSV
-            // device.
-            if (!dev && type == AV_HWDEVICE_TYPE_QSV)
-                dev = hw_device_get_by_name("__qsv_device");
-
             if (!dev)
                 err = hw_device_init_from_type(type, NULL, &dev);
         } else {
@@ -390,7 +364,7 @@ int hw_device_setup_for_decode(InputStream *ist)
             type = config->device_type;
             dev = hw_device_get_by_type(type);
             if (dev) {
-                av_log(NULL, AV_LOG_INFO, "Using auto "
+                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);
             }
@@ -408,12 +382,12 @@ int hw_device_setup_for_decode(InputStream *ist)
                 continue;
             }
             if (ist->hwaccel_device) {
-                av_log(NULL, AV_LOG_INFO, "Using auto "
+                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(NULL, AV_LOG_INFO, "Using auto "
+                av_log(ist->dec_ctx, AV_LOG_INFO, "Using auto "
                        "hwaccel type %s with new default device.\n",
                        av_hwdevice_get_type_name(type));
             }
@@ -421,7 +395,7 @@ int hw_device_setup_for_decode(InputStream *ist)
         if (dev) {
             ist->hwaccel_device_type = type;
         } else {
-            av_log(NULL, AV_LOG_INFO, "Auto hwaccel "
+            av_log(ist->dec_ctx, AV_LOG_INFO, "Auto hwaccel "
                    "disabled: no device found.\n");
             ist->hwaccel_id = HWACCEL_NONE;
             return 0;
@@ -429,7 +403,7 @@ int hw_device_setup_for_decode(InputStream *ist)
     }
 
     if (!dev) {
-        av_log(NULL, AV_LOG_ERROR, "No device available "
+        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;
@@ -461,7 +435,7 @@ int hw_device_setup_for_encode(OutputStream *ost)
     }
 
     for (i = 0;; i++) {
-        config = avcodec_get_hw_config(ost->enc_ctx->codec, i);
+        config = avcodec_get_hw_config(ost->enc, i);
         if (!config)
             break;
 
@@ -472,7 +446,7 @@ int hw_device_setup_for_encode(OutputStream *ost)
             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->name);
             ost->enc_ctx->hw_frames_ctx = av_buffer_ref(frames_ref);
             if (!ost->enc_ctx->hw_frames_ctx)
                 return AVERROR(ENOMEM);
@@ -487,7 +461,7 @@ int hw_device_setup_for_encode(OutputStream *ost)
     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);
+               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);
@@ -553,21 +527,15 @@ int hw_device_setup_for_filter(FilterGraph *fg)
     HWDevice *dev;
     int i;
 
-    // Pick the last hardware device if the user doesn't pick the device for
-    // filters explicitly with the filter_hw_device option.
+    // If the user has supplied exactly one hardware device then just
+    // give it straight to every filter for convenience.  If more than
+    // one device is available then the user needs to pick one explcitly
+    // with the filter_hw_device option.
     if (filter_hw_device)
         dev = filter_hw_device;
-    else if (nb_hw_devices > 0) {
-        dev = hw_devices[nb_hw_devices - 1];
-
-        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);
-    } else
+    else if (nb_hw_devices == 1)
+        dev = hw_devices[0];
+    else
         dev = NULL;
 
     if (dev) {

fftools/ffmpeg_mux.c

@@ -1,751 +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;
-    AVStream *st = ost->st;
-    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 (st->codecpar->codec_type == AVMEDIA_TYPE_VIDEO && ost->vsync_method == VSYNC_DROP)
-        pkt->pts = pkt->dts = AV_NOPTS_VALUE;
-
-    if (st->codecpar->codec_type == AVMEDIA_TYPE_VIDEO) {
-        if (ost->frame_rate.num && ost->is_cfr) {
-            if (pkt->duration > 0)
-                av_log(ost, AV_LOG_WARNING, "Overriding packet duration by frame rate, this should not happen\n");
-            pkt->duration = av_rescale_q(1, av_inv_q(ost->frame_rate),
-                                         pkt->time_base);
-        }
-    }
-
-    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 ((st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO || st->codecpar->codec_type == AVMEDIA_TYPE_VIDEO || st->codecpar->codec_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 || st->codecpar->codec_type == AVMEDIA_TYPE_VIDEO ? AV_LOG_WARNING : AV_LOG_DEBUG;
-                if (exit_on_error)
-                    loglevel = AV_LOG_ERROR;
-                av_log(s, loglevel, "Non-monotonous DTS in output stream "
-                       "%d:%d; previous: %"PRId64", current: %"PRId64"; ",
-                       ost->file_index, ost->st->index, 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;
-
-    ost->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(st->codecpar->codec_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) {
-        print_error("av_interleaved_write_frame()", 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)
-                return (ret == AVERROR_EOF || 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 && stream_eof)
-            tq_receive_finish(mux->tq, stream_idx);
-        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(Muxer *mux, 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(mux, ost, pkt);
-        if (ret < 0) {
-            if (pkt)
-                av_packet_unref(pkt);
-            return ret;
-        }
-    }
-
-    return 0;
-}
-
-void of_output_packet(OutputFile *of, AVPacket *pkt, OutputStream *ost, int eof)
-{
-    Muxer *mux = mux_from_of(of);
-    MuxStream *ms = ms_from_ost(ost);
-    const char *err_msg;
-    int ret = 0;
-
-    if (!eof && 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;
-
-        ret = av_bsf_send_packet(ms->bsf_ctx, eof ? NULL : 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, pkt);
-            if (ret == AVERROR(EAGAIN))
-                return;
-            else if (ret == AVERROR_EOF)
-                bsf_eof = 1;
-            else if (ret < 0) {
-                err_msg = "applying bitstream filters to a packet";
-                goto fail;
-            }
-
-            ret = submit_packet(mux, bsf_eof ? NULL : pkt, ost);
-            if (ret < 0)
-                goto mux_fail;
-        }
-    } else {
-        ret = submit_packet(mux, eof ? NULL : pkt, ost);
-        if (ret < 0)
-            goto mux_fail;
-    }
-
-    return;
-
-mux_fail:
-    err_msg = "submitting a packet to the muxer";
-
-fail:
-    av_log(ost, AV_LOG_ERROR, "Error %s\n", err_msg);
-    if (exit_on_error)
-        exit_program(1);
-
-}
-
-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 void pkt_move(void *dst, void *src)
-{
-    av_packet_move_ref(dst, src);
-}
-
-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;
-
-        /* try to improve muxing time_base (only possible if nothing has been written yet) */
-        if (!av_fifo_can_read(ms->muxing_queue))
-            ost->mux_timebase = ost->st->time_base;
-
-        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 0;
-
-    ret = avcodec_parameters_copy(ctx->par_in, ost->st->codecpar);
-    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;
-
-    return 0;
-}
-
-int of_stream_init(OutputFile *of, OutputStream *ost)
-{
-    Muxer *mux = mux_from_of(of);
-    MuxStream *ms = ms_from_ost(ost);
-    int ret;
-
-    if (ost->sq_idx_mux >= 0)
-        sq_set_tb(mux->sq_mux, ost->sq_idx_mux, ost->mux_timebase);
-
-    /* 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;
-
-    ost->initialized = 1;
-
-    return mux_check_init(mux);
-}
-
-int of_write_trailer(OutputFile *of)
-{
-    Muxer *mux = mux_from_of(of);
-    AVFormatContext *fc = mux->fc;
-    int ret;
-
-    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);
-    }
-
-    ret = thread_stop(mux);
-    if (ret < 0)
-        main_return_code = ret;
-
-    ret = av_write_trailer(fc);
-    if (ret < 0) {
-        av_log(mux, AV_LOG_ERROR, "Error writing trailer: %s\n", av_err2str(ret));
-        return 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));
-            return ret;
-        }
-    }
-
-    return 0;
-}
-
-static void ost_free(OutputStream **post)
-{
-    OutputStream *ost = *post;
-    MuxStream *ms;
-
-    if (!ost)
-        return;
-    ms = ms_from_ost(ost);
-
-    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);
-    }
-
-    av_bsf_free(&ms->bsf_ctx);
-
-    av_frame_free(&ost->filtered_frame);
-    av_frame_free(&ost->sq_frame);
-    av_frame_free(&ost->last_frame);
-    av_packet_free(&ost->pkt);
-    av_dict_free(&ost->encoder_opts);
-
-    av_freep(&ost->kf.pts);
-    av_expr_free(ost->kf.pexpr);
-
-    av_freep(&ost->avfilter);
-    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_close(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);
-}

fftools/ffmpeg_mux.h

@@ -1,102 +0,0 @@
-/*
- * Muxer internal APIs - should not be included outside of ffmpeg_mux*
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#ifndef FFTOOLS_FFMPEG_MUX_H
-#define FFTOOLS_FFMPEG_MUX_H
-
-#include <stdatomic.h>
-#include <stdint.h>
-
-#include "thread_queue.h"
-
-#include "libavformat/avformat.h"
-
-#include "libavcodec/packet.h"
-
-#include "libavutil/dict.h"
-#include "libavutil/fifo.h"
-#include "libavutil/thread.h"
-
-typedef struct MuxStream {
-    OutputStream ost;
-
-    // name used for logging
-    char log_name[32];
-
-    /* the packets are buffered here until the muxer is ready to be initialized */
-    AVFifo *muxing_queue;
-
-    AVBSFContext *bsf_ctx;
-
-    EncStats stats;
-
-    int64_t max_frames;
-
-    /*
-     * The size of the AVPackets' buffers in queue.
-     * Updated when a packet is either pushed or pulled from the queue.
-     */
-    size_t muxing_queue_data_size;
-
-    int max_muxing_queue_size;
-
-    /* Threshold after which max_muxing_queue_size will be in effect */
-    size_t muxing_queue_data_threshold;
-
-    /* dts of the last packet sent to the muxer, in the stream timebase
-     * used for making up missing dts values */
-    int64_t last_mux_dts;
-} MuxStream;
-
-typedef struct Muxer {
-    OutputFile of;
-
-    // name used for logging
-    char log_name[32];
-
-    AVFormatContext *fc;
-
-    pthread_t    thread;
-    ThreadQueue *tq;
-
-    AVDictionary *opts;
-
-    int thread_queue_size;
-
-    /* filesize limit expressed in bytes */
-    int64_t limit_filesize;
-    atomic_int_least64_t last_filesize;
-    int header_written;
-
-    SyncQueue *sq_mux;
-    AVPacket *sq_pkt;
-} Muxer;
-
-/* whether we want to print an SDP, set in of_open() */
-extern int want_sdp;
-
-int mux_check_init(Muxer *mux);
-
-static MuxStream *ms_from_ost(OutputStream *ost)
-{
-    return (MuxStream*)ost;
-}
-
-#endif /* FFTOOLS_FFMPEG_MUX_H */

fftools/ffmpeg_qsv.c

@@ -0,0 +1,110 @@
+/*
+ * 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 <mfx/mfxvideo.h>
+#include <stdlib.h>
+
+#include "libavutil/dict.h"
+#include "libavutil/hwcontext.h"
+#include "libavutil/hwcontext_qsv.h"
+#include "libavutil/mem.h"
+#include "libavutil/opt.h"
+#include "libavcodec/qsv.h"
+
+#include "ffmpeg.h"
+
+static AVBufferRef *hw_device_ctx;
+char *qsv_device = NULL;
+
+static int qsv_get_buffer(AVCodecContext *s, AVFrame *frame, int flags)
+{
+    InputStream *ist = s->opaque;
+
+    return av_hwframe_get_buffer(ist->hw_frames_ctx, frame, 0);
+}
+
+static void qsv_uninit(AVCodecContext *s)
+{
+    InputStream *ist = s->opaque;
+    av_buffer_unref(&ist->hw_frames_ctx);
+}
+
+static int qsv_device_init(InputStream *ist)
+{
+    int err;
+    AVDictionary *dict = NULL;
+
+    if (qsv_device) {
+        err = av_dict_set(&dict, "child_device", qsv_device, 0);
+        if (err < 0)
+            return err;
+    }
+
+    err = av_hwdevice_ctx_create(&hw_device_ctx, AV_HWDEVICE_TYPE_QSV,
+                                 ist->hwaccel_device, dict, 0);
+    if (err < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Error creating a QSV device\n");
+        goto err_out;
+    }
+
+err_out:
+    if (dict)
+        av_dict_free(&dict);
+
+    return err;
+}
+
+int qsv_init(AVCodecContext *s)
+{
+    InputStream *ist = s->opaque;
+    AVHWFramesContext *frames_ctx;
+    AVQSVFramesContext *frames_hwctx;
+    int ret;
+
+    if (!hw_device_ctx) {
+        ret = qsv_device_init(ist);
+        if (ret < 0)
+            return ret;
+    }
+
+    av_buffer_unref(&ist->hw_frames_ctx);
+    ist->hw_frames_ctx = av_hwframe_ctx_alloc(hw_device_ctx);
+    if (!ist->hw_frames_ctx)
+        return AVERROR(ENOMEM);
+
+    frames_ctx   = (AVHWFramesContext*)ist->hw_frames_ctx->data;
+    frames_hwctx = frames_ctx->hwctx;
+
+    frames_ctx->width             = FFALIGN(s->coded_width,  32);
+    frames_ctx->height            = FFALIGN(s->coded_height, 32);
+    frames_ctx->format            = AV_PIX_FMT_QSV;
+    frames_ctx->sw_format         = s->sw_pix_fmt;
+    frames_ctx->initial_pool_size = 64 + s->extra_hw_frames;
+    frames_hwctx->frame_type      = MFX_MEMTYPE_VIDEO_MEMORY_DECODER_TARGET;
+
+    ret = av_hwframe_ctx_init(ist->hw_frames_ctx);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Error initializing a QSV frame pool\n");
+        return ret;
+    }
+
+    ist->hwaccel_get_buffer = qsv_get_buffer;
+    ist->hwaccel_uninit     = qsv_uninit;
+
+    return 0;
+}

fftools/ffmpeg_videotoolbox.c

@@ -0,0 +1,169 @@
+/*
+ * 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 "config.h"
+
+#if HAVE_UTGETOSTYPEFROMSTRING
+#include <CoreServices/CoreServices.h>
+#endif
+
+#include "libavcodec/avcodec.h"
+#include "libavcodec/videotoolbox.h"
+#include "libavutil/imgutils.h"
+#include "ffmpeg.h"
+
+typedef struct VTContext {
+    AVFrame *tmp_frame;
+} VTContext;
+
+char *videotoolbox_pixfmt;
+
+static int videotoolbox_retrieve_data(AVCodecContext *s, AVFrame *frame)
+{
+    InputStream *ist = s->opaque;
+    VTContext  *vt = ist->hwaccel_ctx;
+    CVPixelBufferRef pixbuf = (CVPixelBufferRef)frame->data[3];
+    OSType pixel_format = CVPixelBufferGetPixelFormatType(pixbuf);
+    CVReturn err;
+    uint8_t *data[4] = { 0 };
+    int linesize[4] = { 0 };
+    int planes, ret, i;
+
+    av_frame_unref(vt->tmp_frame);
+
+    switch (pixel_format) {
+    case kCVPixelFormatType_420YpCbCr8Planar: vt->tmp_frame->format = AV_PIX_FMT_YUV420P; break;
+    case kCVPixelFormatType_422YpCbCr8:       vt->tmp_frame->format = AV_PIX_FMT_UYVY422; break;
+    case kCVPixelFormatType_32BGRA:           vt->tmp_frame->format = AV_PIX_FMT_BGRA; break;
+#ifdef kCFCoreFoundationVersionNumber10_7
+    case kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange:
+    case kCVPixelFormatType_420YpCbCr8BiPlanarFullRange: vt->tmp_frame->format = AV_PIX_FMT_NV12; break;
+#endif
+#if HAVE_KCVPIXELFORMATTYPE_420YPCBCR10BIPLANARVIDEORANGE
+    case kCVPixelFormatType_420YpCbCr10BiPlanarVideoRange:
+    case kCVPixelFormatType_420YpCbCr10BiPlanarFullRange: vt->tmp_frame->format = AV_PIX_FMT_P010; break;
+#endif
+    default:
+        av_log(NULL, AV_LOG_ERROR,
+               "%s: Unsupported pixel format: %s\n",
+               av_fourcc2str(s->codec_tag), videotoolbox_pixfmt);
+        return AVERROR(ENOSYS);
+    }
+
+    vt->tmp_frame->width  = frame->width;
+    vt->tmp_frame->height = frame->height;
+    ret = av_frame_get_buffer(vt->tmp_frame, 0);
+    if (ret < 0)
+        return ret;
+
+    err = CVPixelBufferLockBaseAddress(pixbuf, kCVPixelBufferLock_ReadOnly);
+    if (err != kCVReturnSuccess) {
+        av_log(NULL, AV_LOG_ERROR, "Error locking the pixel buffer.\n");
+        return AVERROR_UNKNOWN;
+    }
+
+    if (CVPixelBufferIsPlanar(pixbuf)) {
+
+        planes = CVPixelBufferGetPlaneCount(pixbuf);
+        for (i = 0; i < planes; i++) {
+            data[i]     = CVPixelBufferGetBaseAddressOfPlane(pixbuf, i);
+            linesize[i] = CVPixelBufferGetBytesPerRowOfPlane(pixbuf, i);
+        }
+    } else {
+        data[0] = CVPixelBufferGetBaseAddress(pixbuf);
+        linesize[0] = CVPixelBufferGetBytesPerRow(pixbuf);
+    }
+
+    av_image_copy(vt->tmp_frame->data, vt->tmp_frame->linesize,
+                  (const uint8_t **)data, linesize, vt->tmp_frame->format,
+                  frame->width, frame->height);
+
+    ret = av_frame_copy_props(vt->tmp_frame, frame);
+    CVPixelBufferUnlockBaseAddress(pixbuf, kCVPixelBufferLock_ReadOnly);
+    if (ret < 0)
+        return ret;
+
+    av_frame_unref(frame);
+    av_frame_move_ref(frame, vt->tmp_frame);
+
+    return 0;
+}
+
+static void videotoolbox_uninit(AVCodecContext *s)
+{
+    InputStream *ist = s->opaque;
+    VTContext  *vt = ist->hwaccel_ctx;
+
+    ist->hwaccel_uninit        = NULL;
+    ist->hwaccel_retrieve_data = NULL;
+
+    av_frame_free(&vt->tmp_frame);
+
+    av_videotoolbox_default_free(s);
+    av_freep(&ist->hwaccel_ctx);
+}
+
+int videotoolbox_init(AVCodecContext *s)
+{
+    InputStream *ist = s->opaque;
+    int loglevel = (ist->hwaccel_id == HWACCEL_AUTO) ? AV_LOG_VERBOSE : AV_LOG_ERROR;
+    int ret = 0;
+    VTContext *vt;
+
+    vt = av_mallocz(sizeof(*vt));
+    if (!vt)
+        return AVERROR(ENOMEM);
+
+    ist->hwaccel_ctx           = vt;
+    ist->hwaccel_uninit        = videotoolbox_uninit;
+    ist->hwaccel_retrieve_data = videotoolbox_retrieve_data;
+
+    vt->tmp_frame = av_frame_alloc();
+    if (!vt->tmp_frame) {
+        ret = AVERROR(ENOMEM);
+        goto fail;
+    }
+
+    // TODO: reindent
+        if (!videotoolbox_pixfmt) {
+            ret = av_videotoolbox_default_init(s);
+        } else {
+            AVVideotoolboxContext *vtctx = av_videotoolbox_alloc_context();
+            CFStringRef pixfmt_str = CFStringCreateWithCString(kCFAllocatorDefault,
+                                                               videotoolbox_pixfmt,
+                                                               kCFStringEncodingUTF8);
+#if HAVE_UTGETOSTYPEFROMSTRING
+            vtctx->cv_pix_fmt_type = UTGetOSTypeFromString(pixfmt_str);
+#else
+            av_log(s, loglevel, "UTGetOSTypeFromString() is not available "
+                   "on this platform, %s pixel format can not be honored from "
+                   "the command line\n", videotoolbox_pixfmt);
+#endif
+            ret = av_videotoolbox_default_init2(s, vtctx);
+            CFRelease(pixfmt_str);
+        }
+    if (ret < 0) {
+        av_log(NULL, loglevel, "Error creating Videotoolbox decoder.\n");
+        goto fail;
+    }
+
+    return 0;
+fail:
+    videotoolbox_uninit(s);
+    return ret;
+}

fftools/ffplay.c

@@ -24,7 +24,6 @@
  */
 
 #include "config.h"
-#include "config_components.h"
 #include <inttypes.h>
 #include <math.h>
 #include <limits.h>
@@ -32,7 +31,6 @@
 #include <stdint.h>
 
 #include "libavutil/avstring.h"
-#include "libavutil/channel_layout.h"
 #include "libavutil/eval.h"
 #include "libavutil/mathematics.h"
 #include "libavutil/pixdesc.h"
@@ -41,6 +39,7 @@
 #include "libavutil/fifo.h"
 #include "libavutil/parseutils.h"
 #include "libavutil/samplefmt.h"
+#include "libavutil/avassert.h"
 #include "libavutil/time.h"
 #include "libavutil/bprint.h"
 #include "libavformat/avformat.h"
@@ -60,7 +59,8 @@
 #include <SDL_thread.h>
 
 #include "cmdutils.h"
-#include "opt_common.h"
+
+#include <assert.h>
 
 const char program_name[] = "ffplay";
 const int program_birth_year = 2003;
@@ -117,7 +117,7 @@ typedef struct MyAVPacketList {
 } MyAVPacketList;
 
 typedef struct PacketQueue {
-    AVFifo *pkt_list;
+    AVFifoBuffer *pkt_list;
     int nb_packets;
     int size;
     int64_t duration;
@@ -134,7 +134,8 @@ typedef struct PacketQueue {
 
 typedef struct AudioParams {
     int freq;
-    AVChannelLayout ch_layout;
+    int channels;
+    int64_t channel_layout;
     enum AVSampleFormat fmt;
     int frame_size;
     int bytes_per_sec;
@@ -202,7 +203,7 @@ typedef struct Decoder {
 
 typedef struct VideoState {
     SDL_Thread *read_tid;
-    const AVInputFormat *iformat;
+    AVInputFormat *iformat;
     int abort_request;
     int force_refresh;
     int paused;
@@ -307,7 +308,7 @@ typedef struct VideoState {
 } VideoState;
 
 /* options specified by the user */
-static const AVInputFormat *file_iformat;
+static AVInputFormat *file_iformat;
 static const char *input_filename;
 static const char *window_title;
 static int default_width  = 640;
@@ -413,21 +414,31 @@ int cmp_audio_fmts(enum AVSampleFormat fmt1, int64_t channel_count1,
         return channel_count1 != channel_count2 || fmt1 != fmt2;
 }
 
+static inline
+int64_t get_valid_channel_layout(int64_t channel_layout, int channels)
+{
+    if (channel_layout && av_get_channel_layout_nb_channels(channel_layout) == channels)
+        return channel_layout;
+    else
+        return 0;
+}
+
 static int packet_queue_put_private(PacketQueue *q, AVPacket *pkt)
 {
     MyAVPacketList pkt1;
-    int ret;
 
     if (q->abort_request)
        return -1;
 
+    if (av_fifo_space(q->pkt_list) < sizeof(pkt1)) {
+        if (av_fifo_grow(q->pkt_list, sizeof(pkt1)) < 0)
+            return -1;
+    }
 
     pkt1.pkt = pkt;
     pkt1.serial = q->serial;
 
-    ret = av_fifo_write(q->pkt_list, &pkt1, 1);
-    if (ret < 0)
-        return ret;
+    av_fifo_generic_write(q->pkt_list, &pkt1, sizeof(pkt1), NULL);
     q->nb_packets++;
     q->size += pkt1.pkt->size + sizeof(pkt1);
     q->duration += pkt1.pkt->duration;
@@ -468,7 +479,7 @@ static int packet_queue_put_nullpacket(PacketQueue *q, AVPacket *pkt, int stream
 static int packet_queue_init(PacketQueue *q)
 {
     memset(q, 0, sizeof(PacketQueue));
-    q->pkt_list = av_fifo_alloc2(1, sizeof(MyAVPacketList), AV_FIFO_FLAG_AUTO_GROW);
+    q->pkt_list = av_fifo_alloc(sizeof(MyAVPacketList));
     if (!q->pkt_list)
         return AVERROR(ENOMEM);
     q->mutex = SDL_CreateMutex();
@@ -490,8 +501,10 @@ static void packet_queue_flush(PacketQueue *q)
     MyAVPacketList pkt1;
 
     SDL_LockMutex(q->mutex);
-    while (av_fifo_read(q->pkt_list, &pkt1, 1) >= 0)
+    while (av_fifo_size(q->pkt_list) >= sizeof(pkt1)) {
+        av_fifo_generic_read(q->pkt_list, &pkt1, sizeof(pkt1), NULL);
         av_packet_free(&pkt1.pkt);
+    }
     q->nb_packets = 0;
     q->size = 0;
     q->duration = 0;
@@ -502,7 +515,7 @@ static void packet_queue_flush(PacketQueue *q)
 static void packet_queue_destroy(PacketQueue *q)
 {
     packet_queue_flush(q);
-    av_fifo_freep2(&q->pkt_list);
+    av_fifo_freep(&q->pkt_list);
     SDL_DestroyMutex(q->mutex);
     SDL_DestroyCond(q->cond);
 }
@@ -540,7 +553,8 @@ static int packet_queue_get(PacketQueue *q, AVPacket *pkt, int block, int *seria
             break;
         }
 
-        if (av_fifo_read(q->pkt_list, &pkt1, 1) >= 0) {
+        if (av_fifo_size(q->pkt_list) >= sizeof(pkt1)) {
+            av_fifo_generic_read(q->pkt_list, &pkt1, sizeof(pkt1), NULL);
             q->nb_packets--;
             q->size -= pkt1.pkt->size + sizeof(pkt1);
             q->duration -= pkt1.pkt->duration;
@@ -951,10 +965,10 @@ static void set_sdl_yuv_conversion_mode(AVFrame *frame)
             mode = SDL_YUV_CONVERSION_JPEG;
         else if (frame->colorspace == AVCOL_SPC_BT709)
             mode = SDL_YUV_CONVERSION_BT709;
-        else if (frame->colorspace == AVCOL_SPC_BT470BG || frame->colorspace == AVCOL_SPC_SMPTE170M)
+        else if (frame->colorspace == AVCOL_SPC_BT470BG || frame->colorspace == AVCOL_SPC_SMPTE170M || frame->colorspace == AVCOL_SPC_SMPTE240M)
             mode = SDL_YUV_CONVERSION_BT601;
     }
-    SDL_SetYUVConversionMode(mode); /* FIXME: no support for linear transfer */
+    SDL_SetYUVConversionMode(mode);
 #endif
 }
 
@@ -1011,17 +1025,15 @@ static void video_image_display(VideoState *is)
     }
 
     calculate_display_rect(&rect, is->xleft, is->ytop, is->width, is->height, vp->width, vp->height, vp->sar);
-    set_sdl_yuv_conversion_mode(vp->frame);
 
     if (!vp->uploaded) {
-        if (upload_texture(&is->vid_texture, vp->frame, &is->img_convert_ctx) < 0) {
-            set_sdl_yuv_conversion_mode(NULL);
+        if (upload_texture(&is->vid_texture, vp->frame, &is->img_convert_ctx) < 0)
             return;
-        }
         vp->uploaded = 1;
         vp->flip_v = vp->frame->linesize[0] < 0;
     }
 
+    set_sdl_yuv_conversion_mode(vp->frame);
     SDL_RenderCopyEx(renderer, is->vid_texture, NULL, &rect, 0, NULL, vp->flip_v ? SDL_FLIP_VERTICAL : 0);
     set_sdl_yuv_conversion_mode(NULL);
     if (sp) {
@@ -1060,7 +1072,7 @@ static void video_audio_display(VideoState *s)
     nb_freq = 1 << (rdft_bits - 1);
 
     /* compute display index : center on currently output samples */
-    channels = s->audio_tgt.ch_layout.nb_channels;
+    channels = s->audio_tgt.channels;
     nb_display_channels = channels;
     if (!s->paused) {
         int data_used= s->show_mode == SHOW_MODE_WAVES ? s->width : (2*nb_freq);
@@ -1458,13 +1470,13 @@ static void check_external_clock_speed(VideoState *is) {
 }
 
 /* seek in the stream */
-static void stream_seek(VideoState *is, int64_t pos, int64_t rel, int by_bytes)
+static void stream_seek(VideoState *is, int64_t pos, int64_t rel, int seek_by_bytes)
 {
     if (!is->seek_req) {
         is->seek_pos = pos;
         is->seek_rel = rel;
         is->seek_flags &= ~AVSEEK_FLAG_BYTE;
-        if (by_bytes)
+        if (seek_by_bytes)
             is->seek_flags |= AVSEEK_FLAG_BYTE;
         is->seek_req = 1;
         SDL_CondSignal(is->continue_read_thread);
@@ -1846,7 +1858,7 @@ static int configure_video_filters(AVFilterGraph *graph, VideoState *is, const c
     AVFilterContext *filt_src = NULL, *filt_out = NULL, *last_filter = NULL;
     AVCodecParameters *codecpar = is->video_st->codecpar;
     AVRational fr = av_guess_frame_rate(is->ic, is->video_st, NULL);
-    const AVDictionaryEntry *e = NULL;
+    AVDictionaryEntry *e = NULL;
     int nb_pix_fmts = 0;
     int i, j;
 
@@ -1860,7 +1872,7 @@ static int configure_video_filters(AVFilterGraph *graph, VideoState *is, const c
     }
     pix_fmts[nb_pix_fmts] = AV_PIX_FMT_NONE;
 
-    while ((e = av_dict_iterate(sws_dict, e))) {
+    while ((e = av_dict_get(sws_dict, "", e, AV_DICT_IGNORE_SUFFIX))) {
         if (!strcmp(e->key, "sws_flags")) {
             av_strlcatf(sws_flags_str, sizeof(sws_flags_str), "%s=%s:", "flags", e->value);
         } else
@@ -1915,14 +1927,7 @@ static int configure_video_filters(AVFilterGraph *graph, VideoState *is, const c
 } while (0)
 
     if (autorotate) {
-        double theta = 0.0;
-        int32_t *displaymatrix = NULL;
-        AVFrameSideData *sd = av_frame_get_side_data(frame, AV_FRAME_DATA_DISPLAYMATRIX);
-        if (sd)
-            displaymatrix = (int32_t *)sd->data;
-        if (!displaymatrix)
-            displaymatrix = (int32_t *)av_stream_get_side_data(is->video_st, AV_PKT_DATA_DISPLAYMATRIX, NULL);
-        theta = get_rotation(displaymatrix);
+        double theta  = get_rotation(is->video_st);
 
         if (fabs(theta - 90) < 1.0) {
             INSERT_FILT("transpose", "clock");
@@ -1952,10 +1957,11 @@ static int configure_audio_filters(VideoState *is, const char *afilters, int for
 {
     static const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE };
     int sample_rates[2] = { 0, -1 };
+    int64_t channel_layouts[2] = { 0, -1 };
+    int channels[2] = { 0, -1 };
     AVFilterContext *filt_asrc = NULL, *filt_asink = NULL;
     char aresample_swr_opts[512] = "";
-    const AVDictionaryEntry *e = NULL;
-    AVBPrint bp;
+    AVDictionaryEntry *e = NULL;
     char asrc_args[256];
     int ret;
 
@@ -1964,20 +1970,20 @@ static int configure_audio_filters(VideoState *is, const char *afilters, int for
         return AVERROR(ENOMEM);
     is->agraph->nb_threads = filter_nbthreads;
 
-    av_bprint_init(&bp, 0, AV_BPRINT_SIZE_AUTOMATIC);
-
-    while ((e = av_dict_iterate(swr_opts, e)))
+    while ((e = av_dict_get(swr_opts, "", e, AV_DICT_IGNORE_SUFFIX)))
         av_strlcatf(aresample_swr_opts, sizeof(aresample_swr_opts), "%s=%s:", e->key, e->value);
     if (strlen(aresample_swr_opts))
         aresample_swr_opts[strlen(aresample_swr_opts)-1] = '\0';
     av_opt_set(is->agraph, "aresample_swr_opts", aresample_swr_opts, 0);
 
-    av_channel_layout_describe_bprint(&is->audio_filter_src.ch_layout, &bp);
-
     ret = snprintf(asrc_args, sizeof(asrc_args),
-                   "sample_rate=%d:sample_fmt=%s:time_base=%d/%d:channel_layout=%s",
+                   "sample_rate=%d:sample_fmt=%s:channels=%d:time_base=%d/%d",
                    is->audio_filter_src.freq, av_get_sample_fmt_name(is->audio_filter_src.fmt),
-                   1, is->audio_filter_src.freq, bp.str);
+                   is->audio_filter_src.channels,
+                   1, is->audio_filter_src.freq);
+    if (is->audio_filter_src.channel_layout)
+        snprintf(asrc_args + ret, sizeof(asrc_args) - ret,
+                 ":channel_layout=0x%"PRIx64,  is->audio_filter_src.channel_layout);
 
     ret = avfilter_graph_create_filter(&filt_asrc,
                                        avfilter_get_by_name("abuffer"), "ffplay_abuffer",
@@ -1998,10 +2004,14 @@ static int configure_audio_filters(VideoState *is, const char *afilters, int for
         goto end;
 
     if (force_output_format) {
+        channel_layouts[0] = is->audio_tgt.channel_layout;
+        channels       [0] = is->audio_tgt.channel_layout ? -1 : is->audio_tgt.channels;
         sample_rates   [0] = is->audio_tgt.freq;
         if ((ret = av_opt_set_int(filt_asink, "all_channel_counts", 0, AV_OPT_SEARCH_CHILDREN)) < 0)
             goto end;
-        if ((ret = av_opt_set(filt_asink, "ch_layouts", bp.str, AV_OPT_SEARCH_CHILDREN)) < 0)
+        if ((ret = av_opt_set_int_list(filt_asink, "channel_layouts", channel_layouts,  -1, AV_OPT_SEARCH_CHILDREN)) < 0)
+            goto end;
+        if ((ret = av_opt_set_int_list(filt_asink, "channel_counts" , channels       ,  -1, AV_OPT_SEARCH_CHILDREN)) < 0)
             goto end;
         if ((ret = av_opt_set_int_list(filt_asink, "sample_rates"   , sample_rates   ,  -1, AV_OPT_SEARCH_CHILDREN)) < 0)
             goto end;
@@ -2017,8 +2027,6 @@ static int configure_audio_filters(VideoState *is, const char *afilters, int for
 end:
     if (ret < 0)
         avfilter_graph_free(&is->agraph);
-    av_bprint_finalize(&bp, NULL);
-
     return ret;
 }
 #endif  /* CONFIG_AVFILTER */
@@ -2030,6 +2038,7 @@ static int audio_thread(void *arg)
     Frame *af;
 #if CONFIG_AVFILTER
     int last_serial = -1;
+    int64_t dec_channel_layout;
     int reconfigure;
 #endif
     int got_frame = 0;
@@ -2047,26 +2056,27 @@ static int audio_thread(void *arg)
                 tb = (AVRational){1, frame->sample_rate};
 
 #if CONFIG_AVFILTER
+                dec_channel_layout = get_valid_channel_layout(frame->channel_layout, frame->channels);
+
                 reconfigure =
-                    cmp_audio_fmts(is->audio_filter_src.fmt, is->audio_filter_src.ch_layout.nb_channels,
-                                   frame->format, frame->ch_layout.nb_channels)    ||
-                    av_channel_layout_compare(&is->audio_filter_src.ch_layout, &frame->ch_layout) ||
+                    cmp_audio_fmts(is->audio_filter_src.fmt, is->audio_filter_src.channels,
+                                   frame->format, frame->channels)    ||
+                    is->audio_filter_src.channel_layout != dec_channel_layout ||
                     is->audio_filter_src.freq           != frame->sample_rate ||
                     is->auddec.pkt_serial               != last_serial;
 
                 if (reconfigure) {
                     char buf1[1024], buf2[1024];
-                    av_channel_layout_describe(&is->audio_filter_src.ch_layout, buf1, sizeof(buf1));
-                    av_channel_layout_describe(&frame->ch_layout, buf2, sizeof(buf2));
+                    av_get_channel_layout_string(buf1, sizeof(buf1), -1, is->audio_filter_src.channel_layout);
+                    av_get_channel_layout_string(buf2, sizeof(buf2), -1, dec_channel_layout);
                     av_log(NULL, AV_LOG_DEBUG,
                            "Audio frame changed from rate:%d ch:%d fmt:%s layout:%s serial:%d to rate:%d ch:%d fmt:%s layout:%s serial:%d\n",
-                           is->audio_filter_src.freq, is->audio_filter_src.ch_layout.nb_channels, av_get_sample_fmt_name(is->audio_filter_src.fmt), buf1, last_serial,
-                           frame->sample_rate, frame->ch_layout.nb_channels, av_get_sample_fmt_name(frame->format), buf2, is->auddec.pkt_serial);
+                           is->audio_filter_src.freq, is->audio_filter_src.channels, av_get_sample_fmt_name(is->audio_filter_src.fmt), buf1, last_serial,
+                           frame->sample_rate, frame->channels, av_get_sample_fmt_name(frame->format), buf2, is->auddec.pkt_serial);
 
                     is->audio_filter_src.fmt            = frame->format;
-                    ret = av_channel_layout_copy(&is->audio_filter_src.ch_layout, &frame->ch_layout);
-                    if (ret < 0)
-                        goto the_end;
+                    is->audio_filter_src.channels       = frame->channels;
+                    is->audio_filter_src.channel_layout = dec_channel_layout;
                     is->audio_filter_src.freq           = frame->sample_rate;
                     last_serial                         = is->auddec.pkt_serial;
 
@@ -2332,6 +2342,7 @@ static int synchronize_audio(VideoState *is, int nb_samples)
 static int audio_decode_frame(VideoState *is)
 {
     int data_size, resampled_data_size;
+    int64_t dec_channel_layout;
     av_unused double audio_clock0;
     int wanted_nb_samples;
     Frame *af;
@@ -2352,31 +2363,34 @@ static int audio_decode_frame(VideoState *is)
         frame_queue_next(&is->sampq);
     } while (af->serial != is->audioq.serial);
 
-    data_size = av_samples_get_buffer_size(NULL, af->frame->ch_layout.nb_channels,
+    data_size = av_samples_get_buffer_size(NULL, af->frame->channels,
                                            af->frame->nb_samples,
                                            af->frame->format, 1);
 
+    dec_channel_layout =
+        (af->frame->channel_layout && af->frame->channels == av_get_channel_layout_nb_channels(af->frame->channel_layout)) ?
+        af->frame->channel_layout : av_get_default_channel_layout(af->frame->channels);
     wanted_nb_samples = synchronize_audio(is, af->frame->nb_samples);
 
     if (af->frame->format        != is->audio_src.fmt            ||
-        av_channel_layout_compare(&af->frame->ch_layout, &is->audio_src.ch_layout) ||
+        dec_channel_layout       != is->audio_src.channel_layout ||
         af->frame->sample_rate   != is->audio_src.freq           ||
         (wanted_nb_samples       != af->frame->nb_samples && !is->swr_ctx)) {
         swr_free(&is->swr_ctx);
-        swr_alloc_set_opts2(&is->swr_ctx,
-                            &is->audio_tgt.ch_layout, is->audio_tgt.fmt, is->audio_tgt.freq,
-                            &af->frame->ch_layout, af->frame->format, af->frame->sample_rate,
-                            0, NULL);
+        is->swr_ctx = swr_alloc_set_opts(NULL,
+                                         is->audio_tgt.channel_layout, is->audio_tgt.fmt, is->audio_tgt.freq,
+                                         dec_channel_layout,           af->frame->format, af->frame->sample_rate,
+                                         0, NULL);
         if (!is->swr_ctx || swr_init(is->swr_ctx) < 0) {
             av_log(NULL, AV_LOG_ERROR,
                    "Cannot create sample rate converter for conversion of %d Hz %s %d channels to %d Hz %s %d channels!\n",
-                    af->frame->sample_rate, av_get_sample_fmt_name(af->frame->format), af->frame->ch_layout.nb_channels,
-                    is->audio_tgt.freq, av_get_sample_fmt_name(is->audio_tgt.fmt), is->audio_tgt.ch_layout.nb_channels);
+                    af->frame->sample_rate, av_get_sample_fmt_name(af->frame->format), af->frame->channels,
+                    is->audio_tgt.freq, av_get_sample_fmt_name(is->audio_tgt.fmt), is->audio_tgt.channels);
             swr_free(&is->swr_ctx);
             return -1;
         }
-        if (av_channel_layout_copy(&is->audio_src.ch_layout, &af->frame->ch_layout) < 0)
-            return -1;
+        is->audio_src.channel_layout = dec_channel_layout;
+        is->audio_src.channels       = af->frame->channels;
         is->audio_src.freq = af->frame->sample_rate;
         is->audio_src.fmt = af->frame->format;
     }
@@ -2385,7 +2399,7 @@ static int audio_decode_frame(VideoState *is)
         const uint8_t **in = (const uint8_t **)af->frame->extended_data;
         uint8_t **out = &is->audio_buf1;
         int out_count = (int64_t)wanted_nb_samples * is->audio_tgt.freq / af->frame->sample_rate + 256;
-        int out_size  = av_samples_get_buffer_size(NULL, is->audio_tgt.ch_layout.nb_channels, out_count, is->audio_tgt.fmt, 0);
+        int out_size  = av_samples_get_buffer_size(NULL, is->audio_tgt.channels, out_count, is->audio_tgt.fmt, 0);
         int len2;
         if (out_size < 0) {
             av_log(NULL, AV_LOG_ERROR, "av_samples_get_buffer_size() failed\n");
@@ -2412,7 +2426,7 @@ static int audio_decode_frame(VideoState *is)
                 swr_free(&is->swr_ctx);
         }
         is->audio_buf = is->audio_buf1;
-        resampled_data_size = len2 * is->audio_tgt.ch_layout.nb_channels * av_get_bytes_per_sample(is->audio_tgt.fmt);
+        resampled_data_size = len2 * is->audio_tgt.channels * av_get_bytes_per_sample(is->audio_tgt.fmt);
     } else {
         is->audio_buf = af->frame->data[0];
         resampled_data_size = data_size;
@@ -2481,26 +2495,24 @@ static void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
     }
 }
 
-static int audio_open(void *opaque, AVChannelLayout *wanted_channel_layout, int wanted_sample_rate, struct AudioParams *audio_hw_params)
+static int audio_open(void *opaque, int64_t wanted_channel_layout, int wanted_nb_channels, int wanted_sample_rate, struct AudioParams *audio_hw_params)
 {
     SDL_AudioSpec wanted_spec, spec;
     const char *env;
     static const int next_nb_channels[] = {0, 0, 1, 6, 2, 6, 4, 6};
     static const int next_sample_rates[] = {0, 44100, 48000, 96000, 192000};
     int next_sample_rate_idx = FF_ARRAY_ELEMS(next_sample_rates) - 1;
-    int wanted_nb_channels = wanted_channel_layout->nb_channels;
 
     env = SDL_getenv("SDL_AUDIO_CHANNELS");
     if (env) {
         wanted_nb_channels = atoi(env);
-        av_channel_layout_uninit(wanted_channel_layout);
-        av_channel_layout_default(wanted_channel_layout, wanted_nb_channels);
+        wanted_channel_layout = av_get_default_channel_layout(wanted_nb_channels);
     }
-    if (wanted_channel_layout->order != AV_CHANNEL_ORDER_NATIVE) {
-        av_channel_layout_uninit(wanted_channel_layout);
-        av_channel_layout_default(wanted_channel_layout, wanted_nb_channels);
+    if (!wanted_channel_layout || wanted_nb_channels != av_get_channel_layout_nb_channels(wanted_channel_layout)) {
+        wanted_channel_layout = av_get_default_channel_layout(wanted_nb_channels);
+        wanted_channel_layout &= ~AV_CH_LAYOUT_STEREO_DOWNMIX;
     }
-    wanted_nb_channels = wanted_channel_layout->nb_channels;
+    wanted_nb_channels = av_get_channel_layout_nb_channels(wanted_channel_layout);
     wanted_spec.channels = wanted_nb_channels;
     wanted_spec.freq = wanted_sample_rate;
     if (wanted_spec.freq <= 0 || wanted_spec.channels <= 0) {
@@ -2527,7 +2539,7 @@ static int audio_open(void *opaque, AVChannelLayout *wanted_channel_layout, int
                 return -1;
             }
         }
-        av_channel_layout_default(wanted_channel_layout, wanted_spec.channels);
+        wanted_channel_layout = av_get_default_channel_layout(wanted_spec.channels);
     }
     if (spec.format != AUDIO_S16SYS) {
         av_log(NULL, AV_LOG_ERROR,
@@ -2535,9 +2547,8 @@ static int audio_open(void *opaque, AVChannelLayout *wanted_channel_layout, int
         return -1;
     }
     if (spec.channels != wanted_spec.channels) {
-        av_channel_layout_uninit(wanted_channel_layout);
-        av_channel_layout_default(wanted_channel_layout, spec.channels);
-        if (wanted_channel_layout->order != AV_CHANNEL_ORDER_NATIVE) {
+        wanted_channel_layout = av_get_default_channel_layout(spec.channels);
+        if (!wanted_channel_layout) {
             av_log(NULL, AV_LOG_ERROR,
                    "SDL advised channel count %d is not supported!\n", spec.channels);
             return -1;
@@ -2546,10 +2557,10 @@ static int audio_open(void *opaque, AVChannelLayout *wanted_channel_layout, int
 
     audio_hw_params->fmt = AV_SAMPLE_FMT_S16;
     audio_hw_params->freq = spec.freq;
-    if (av_channel_layout_copy(&audio_hw_params->ch_layout, wanted_channel_layout) < 0)
-        return -1;
-    audio_hw_params->frame_size = av_samples_get_buffer_size(NULL, audio_hw_params->ch_layout.nb_channels, 1, audio_hw_params->fmt, 1);
-    audio_hw_params->bytes_per_sec = av_samples_get_buffer_size(NULL, audio_hw_params->ch_layout.nb_channels, audio_hw_params->freq, audio_hw_params->fmt, 1);
+    audio_hw_params->channel_layout = wanted_channel_layout;
+    audio_hw_params->channels =  spec.channels;
+    audio_hw_params->frame_size = av_samples_get_buffer_size(NULL, audio_hw_params->channels, 1, audio_hw_params->fmt, 1);
+    audio_hw_params->bytes_per_sec = av_samples_get_buffer_size(NULL, audio_hw_params->channels, audio_hw_params->freq, audio_hw_params->fmt, 1);
     if (audio_hw_params->bytes_per_sec <= 0 || audio_hw_params->frame_size <= 0) {
         av_log(NULL, AV_LOG_ERROR, "av_samples_get_buffer_size failed\n");
         return -1;
@@ -2565,9 +2576,9 @@ static int stream_component_open(VideoState *is, int stream_index)
     const AVCodec *codec;
     const char *forced_codec_name = NULL;
     AVDictionary *opts = NULL;
-    const AVDictionaryEntry *t = NULL;
-    int sample_rate;
-    AVChannelLayout ch_layout = { 0 };
+    AVDictionaryEntry *t = NULL;
+    int sample_rate, nb_channels;
+    int64_t channel_layout;
     int ret = 0;
     int stream_lowres = lowres;
 
@@ -2635,27 +2646,24 @@ static int stream_component_open(VideoState *is, int stream_index)
             AVFilterContext *sink;
 
             is->audio_filter_src.freq           = avctx->sample_rate;
-            ret = av_channel_layout_copy(&is->audio_filter_src.ch_layout, &avctx->ch_layout);
-            if (ret < 0)
-                goto fail;
+            is->audio_filter_src.channels       = avctx->channels;
+            is->audio_filter_src.channel_layout = get_valid_channel_layout(avctx->channel_layout, avctx->channels);
             is->audio_filter_src.fmt            = avctx->sample_fmt;
             if ((ret = configure_audio_filters(is, afilters, 0)) < 0)
                 goto fail;
             sink = is->out_audio_filter;
             sample_rate    = av_buffersink_get_sample_rate(sink);
-            ret = av_buffersink_get_ch_layout(sink, &ch_layout);
-            if (ret < 0)
-                goto fail;
+            nb_channels    = av_buffersink_get_channels(sink);
+            channel_layout = av_buffersink_get_channel_layout(sink);
         }
 #else
         sample_rate    = avctx->sample_rate;
-        ret = av_channel_layout_copy(&ch_layout, &avctx->ch_layout);
-        if (ret < 0)
-            goto fail;
+        nb_channels    = avctx->channels;
+        channel_layout = avctx->channel_layout;
 #endif
 
         /* prepare audio output */
-        if ((ret = audio_open(is, &ch_layout, sample_rate, &is->audio_tgt)) < 0)
+        if ((ret = audio_open(is, channel_layout, nb_channels, sample_rate, &is->audio_tgt)) < 0)
             goto fail;
         is->audio_hw_buf_size = ret;
         is->audio_src = is->audio_tgt;
@@ -2709,7 +2717,6 @@ static int stream_component_open(VideoState *is, int stream_index)
 fail:
     avcodec_free_context(&avctx);
 out:
-    av_channel_layout_uninit(&ch_layout);
     av_dict_free(&opts);
 
     return ret;
@@ -2754,7 +2761,7 @@ static int read_thread(void *arg)
     AVPacket *pkt = NULL;
     int64_t stream_start_time;
     int pkt_in_play_range = 0;
-    const AVDictionaryEntry *t;
+    AVDictionaryEntry *t;
     SDL_mutex *wait_mutex = SDL_CreateMutex();
     int scan_all_pmts_set = 0;
     int64_t pkt_ts;
@@ -2829,9 +2836,7 @@ static int read_thread(void *arg)
         ic->pb->eof_reached = 0; // FIXME hack, ffplay maybe should not use avio_feof() to test for the end
 
     if (seek_by_bytes < 0)
-        seek_by_bytes = !(ic->iformat->flags & AVFMT_NO_BYTE_SEEK) &&
-                        !!(ic->iformat->flags & AVFMT_TS_DISCONT) &&
-                        strcmp("ogg", ic->iformat->name);
+        seek_by_bytes = !!(ic->iformat->flags & AVFMT_TS_DISCONT) && strcmp("ogg", ic->iformat->name);
 
     is->max_frame_duration = (ic->iformat->flags & AVFMT_TS_DISCONT) ? 10.0 : 3600.0;
 
@@ -3070,8 +3075,7 @@ static int read_thread(void *arg)
     return 0;
 }
 
-static VideoState *stream_open(const char *filename,
-                               const AVInputFormat *iformat)
+static VideoState *stream_open(const char *filename, AVInputFormat *iformat)
 {
     VideoState *is;
 
@@ -3184,7 +3188,7 @@ static void stream_cycle_channel(VideoState *is, int codec_type)
             switch (codec_type) {
             case AVMEDIA_TYPE_AUDIO:
                 if (st->codecpar->sample_rate != 0 &&
-                    st->codecpar->ch_layout.nb_channels != 0)
+                    st->codecpar->channels != 0)
                     goto the_end;
                 break;
             case AVMEDIA_TYPE_VIDEO:
@@ -3470,6 +3474,12 @@ static void event_loop(VideoState *cur_stream)
     }
 }
 
+static int opt_frame_size(void *optctx, const char *opt, const char *arg)
+{
+    av_log(NULL, AV_LOG_WARNING, "Option -s is deprecated, use -video_size.\n");
+    return opt_default(NULL, "video_size", arg);
+}
+
 static int opt_width(void *optctx, const char *opt, const char *arg)
 {
     screen_width = parse_number_or_die(opt, arg, OPT_INT64, 1, INT_MAX);
@@ -3492,6 +3502,12 @@ static int opt_format(void *optctx, const char *opt, const char *arg)
     return 0;
 }
 
+static int opt_frame_pix_fmt(void *optctx, const char *opt, const char *arg)
+{
+    av_log(NULL, AV_LOG_WARNING, "Option -pix_fmt is deprecated, use -pixel_format.\n");
+    return opt_default(NULL, "pixel_format", arg);
+}
+
 static int opt_sync(void *optctx, const char *opt, const char *arg)
 {
     if (!strcmp(arg, "audio"))
@@ -3537,7 +3553,7 @@ static void opt_input_file(void *optctx, const char *filename)
         exit(1);
     }
     if (!strcmp(filename, "-"))
-        filename = "fd:";
+        filename = "pipe:";
     input_filename = filename;
 }
 
@@ -3569,6 +3585,7 @@ static const OptionDef options[] = {
     CMDUTILS_COMMON_OPTIONS
     { "x", HAS_ARG, { .func_arg = opt_width }, "force displayed width", "width" },
     { "y", HAS_ARG, { .func_arg = opt_height }, "force displayed height", "height" },
+    { "s", HAS_ARG | OPT_VIDEO, { .func_arg = opt_frame_size }, "set frame size (WxH or abbreviation)", "size" },
     { "fs", OPT_BOOL, { &is_full_screen }, "force full screen" },
     { "an", OPT_BOOL, { &audio_disable }, "disable audio" },
     { "vn", OPT_BOOL, { &video_disable }, "disable video" },
@@ -3585,6 +3602,7 @@ static const OptionDef options[] = {
     { "alwaysontop", OPT_BOOL, { &alwaysontop }, "window always on top" },
     { "volume", OPT_INT | HAS_ARG, { &startup_volume}, "set startup volume 0=min 100=max", "volume" },
     { "f", HAS_ARG, { .func_arg = opt_format }, "force format", "fmt" },
+    { "pix_fmt", HAS_ARG | OPT_EXPERT | OPT_VIDEO, { .func_arg = opt_frame_pix_fmt }, "set pixel format", "format" },
     { "stats", OPT_BOOL | OPT_EXPERT, { &show_status }, "show status", "" },
     { "fast", OPT_BOOL | OPT_EXPERT, { &fast }, "non spec compliant optimizations", "" },
     { "genpts", OPT_BOOL | OPT_EXPERT, { &genpts }, "generate pts", "" },
@@ -3606,6 +3624,7 @@ static const OptionDef options[] = {
 #endif
     { "rdftspeed", OPT_INT | HAS_ARG| OPT_AUDIO | OPT_EXPERT, { &rdftspeed }, "rdft speed", "msecs" },
     { "showmode", HAS_ARG, { .func_arg = opt_show_mode}, "select show mode (0 = video, 1 = waves, 2 = RDFT)", "mode" },
+    { "default", HAS_ARG | OPT_AUDIO | OPT_VIDEO | OPT_EXPERT, { .func_arg = opt_default }, "generic catch all option", "" },
     { "i", OPT_BOOL, { &dummy}, "read specified file", "input_file"},
     { "codec", HAS_ARG, { .func_arg = opt_codec}, "force decoder", "decoder_name" },
     { "acodec", HAS_ARG | OPT_STRING | OPT_EXPERT, {    &audio_codec_name }, "force audio decoder",    "decoder_name" },
@@ -3677,6 +3696,8 @@ int main(int argc, char **argv)
 #endif
     avformat_network_init();
 
+    init_opts();
+
     signal(SIGINT , sigterm_handler); /* Interrupt (ANSI).    */
     signal(SIGTERM, sigterm_handler); /* Termination (ANSI).  */
 
@@ -3727,10 +3748,6 @@ int main(int argc, char **argv)
             flags |= SDL_WINDOW_BORDERLESS;
         else
             flags |= SDL_WINDOW_RESIZABLE;
-
-#ifdef SDL_HINT_VIDEO_X11_NET_WM_BYPASS_COMPOSITOR
-        SDL_SetHint(SDL_HINT_VIDEO_X11_NET_WM_BYPASS_COMPOSITOR, "0");
-#endif
         window = SDL_CreateWindow(program_name, SDL_WINDOWPOS_UNDEFINED, SDL_WINDOWPOS_UNDEFINED, default_width, default_height, flags);
         SDL_SetHint(SDL_HINT_RENDER_SCALE_QUALITY, "linear");
         if (window) {

fftools/fftools.manifest

@@ -1,9 +0,0 @@
-<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
-<assembly xmlns="urn:schemas-microsoft-com:asm.v1" manifestVersion="1.0" xmlns:asmv3="urn:schemas-microsoft-com:asm.v3">
-  <asmv3:application>
-    <asmv3:windowsSettings>
-      <dpiAware xmlns="http://schemas.microsoft.com/SMI/2005/WindowsSettings">true</dpiAware>
-      <dpiAwareness xmlns="http://schemas.microsoft.com/SMI/2016/WindowsSettings">PerMonitorV2</dpiAwareness>
-    </asmv3:windowsSettings>
-  </asmv3:application>
-</assembly>

fftools/fftoolsres.rc

@@ -1,2 +0,0 @@
-#include <windows.h>
-1 RT_MANIFEST fftools.manifest

fftools/fopen_utf8.h

@@ -1,71 +0,0 @@
-/*
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#ifndef FFTOOLS_FOPEN_UTF8_H
-#define FFTOOLS_FOPEN_UTF8_H
-
-#include <stdio.h>
-
-/* The fopen_utf8 function here is essentially equivalent to avpriv_fopen_utf8,
- * except that it doesn't set O_CLOEXEC, and that it isn't exported
- * from a different library. (On Windows, each DLL might use a different
- * CRT, and FILE* handles can't be shared across them.) */
-
-#ifdef _WIN32
-#include "libavutil/wchar_filename.h"
-
-static inline FILE *fopen_utf8(const char *path_utf8, const char *mode)
-{
-    wchar_t *path_w, *mode_w;
-    FILE *f;
-
-    /* convert UTF-8 to wide chars */
-    if (get_extended_win32_path(path_utf8, &path_w)) /* This sets errno on error. */
-        return NULL;
-    if (!path_w)
-        goto fallback;
-
-    if (utf8towchar(mode, &mode_w))
-        return NULL;
-    if (!mode_w) {
-        /* If failing to interpret the mode string as utf8, it is an invalid
-         * parameter. */
-        av_freep(&path_w);
-        errno = EINVAL;
-        return NULL;
-    }
-
-    f = _wfopen(path_w, mode_w);
-    av_freep(&path_w);
-    av_freep(&mode_w);
-
-    return f;
-fallback:
-    /* path may be in CP_ACP */
-    return fopen(path_utf8, mode);
-}
-
-#else
-
-static inline FILE *fopen_utf8(const char *path, const char *mode)
-{
-    return fopen(path, mode);
-}
-#endif
-
-#endif /* FFTOOLS_FOPEN_UTF8_H */

fftools/objpool.c

@@ -1,131 +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 <stdint.h>
-
-#include "libavcodec/packet.h"
-
-#include "libavutil/common.h"
-#include "libavutil/error.h"
-#include "libavutil/frame.h"
-#include "libavutil/mem.h"
-
-#include "objpool.h"
-
-struct ObjPool {
-    void        *pool[32];
-    unsigned int pool_count;
-
-    ObjPoolCBAlloc alloc;
-    ObjPoolCBReset reset;
-    ObjPoolCBFree  free;
-};
-
-ObjPool *objpool_alloc(ObjPoolCBAlloc cb_alloc, ObjPoolCBReset cb_reset,
-                       ObjPoolCBFree cb_free)
-{
-    ObjPool *op = av_mallocz(sizeof(*op));
-
-    if (!op)
-        return NULL;
-
-    op->alloc = cb_alloc;
-    op->reset = cb_reset;
-    op->free  = cb_free;
-
-    return op;
-}
-
-void objpool_free(ObjPool **pop)
-{
-    ObjPool *op = *pop;
-
-    if (!op)
-        return;
-
-    for (unsigned int i = 0; i < op->pool_count; i++)
-        op->free(&op->pool[i]);
-
-    av_freep(pop);
-}
-
-int  objpool_get(ObjPool *op, void **obj)
-{
-    if (op->pool_count) {
-        *obj = op->pool[--op->pool_count];
-        op->pool[op->pool_count] = NULL;
-    } else
-        *obj = op->alloc();
-
-    return *obj ? 0 : AVERROR(ENOMEM);
-}
-
-void objpool_release(ObjPool *op, void **obj)
-{
-    if (!*obj)
-        return;
-
-    op->reset(*obj);
-
-    if (op->pool_count < FF_ARRAY_ELEMS(op->pool))
-        op->pool[op->pool_count++] = *obj;
-    else
-        op->free(obj);
-
-    *obj = NULL;
-}
-
-static void *alloc_packet(void)
-{
-    return av_packet_alloc();
-}
-static void *alloc_frame(void)
-{
-    return av_frame_alloc();
-}
-
-static void reset_packet(void *obj)
-{
-    av_packet_unref(obj);
-}
-static void reset_frame(void *obj)
-{
-    av_frame_unref(obj);
-}
-
-static void free_packet(void **obj)
-{
-    AVPacket *pkt = *obj;
-    av_packet_free(&pkt);
-    *obj = NULL;
-}
-static void free_frame(void **obj)
-{
-    AVFrame *frame = *obj;
-    av_frame_free(&frame);
-    *obj = NULL;
-}
-
-ObjPool *objpool_alloc_packets(void)
-{
-    return objpool_alloc(alloc_packet, reset_packet, free_packet);
-}
-ObjPool *objpool_alloc_frames(void)
-{
-    return objpool_alloc(alloc_frame, reset_frame, free_frame);
-}