Changelog: Update for the last 3 commits

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

.gitignore

@@ -29,6 +29,7 @@
 /ffmpeg
 /ffplay
 /ffprobe
+/ffserver
 /config.asm
 /config.h
 /coverage.info
@@ -36,4 +37,3 @@
 /lcov/
 /src
 /mapfile
-/tools/python/__pycache__/

.travis.yml

@@ -11,15 +11,11 @@ addons:
 compiler:
   - clang
   - gcc
-matrix:
-    exclude:
-        - os: osx
-          compiler: gcc
 cache:
   directories:
     - ffmpeg-samples
 before_install:
-  - if [ "$TRAVIS_OS_NAME" == "osx" ]; then brew update; fi
+  - if [ "$TRAVIS_OS_NAME" == "osx" ]; then brew update --all; fi
 install:
   - if [ "$TRAVIS_OS_NAME" == "osx" ]; then brew install nasm; fi
 script:

Changelog

@@ -1,143 +1,82 @@
 Entries are sorted chronologically from oldest to youngest within each release,
 releases are sorted from youngest to oldest.
 
-version 4.2:
-- tpad filter
-- AV1 decoding support through libdav1d
-- dedot filter
-- chromashift and rgbashift filters
-- freezedetect filter
-- truehd_core bitstream filter
-- dhav demuxer
-- PCM-DVD encoder
-- GIF parser
-- vividas demuxer
-- hymt decoder
-- anlmdn filter
-- maskfun filter
-- hcom demuxer and decoder
-- ARBC decoder
-- libaribb24 based ARIB STD-B24 caption support (profiles A and C)
-- Support decoding of HEVC 4:4:4 content in nvdec and cuviddec
-- removed libndi-newtek
-- agm decoder
-- KUX demuxer
-- AV1 frame split bitstream filter
-- lscr decoder
-- lagfun filter
-- asoftclip filter
-- Support decoding of HEVC 4:4:4 content in vdpau
-- colorhold filter
-- xmedian filter
-- asr filter
-- showspatial multimedia filter
-- VP4 video decoder
-- IFV demuxer
-- derain filter
-- deesser filter
-- mov muxer writes tracks with unspecified language instead of English by default
-- add support for using clang to compile CUDA kernels
-
-
-version 4.1:
-- deblock filter
-- tmix filter
-- amplify filter
-- fftdnoiz filter
-- aderivative and aintegral audio filters
-- pal75bars and pal100bars video filter sources
-- support mbedTLS based TLS
-- adeclick filter
-- adeclip filter
-- libtensorflow backend for DNN based filters like srcnn
-- vc1 decoder is now bit-exact
-- ATRAC9 decoder
-- lensfun wrapper filter
-- colorconstancy filter
-- AVS2 video decoder via libdavs2
-- IMM4 video decoder
-- Brooktree ProSumer video decoder
-- MatchWare Screen Capture Codec decoder
-- WinCam Motion Video decoder
-- 1D LUT filter (lut1d)
-- RemotelyAnywhere Screen Capture decoder
-- cue and acue filters
-- support for AV1 in MP4
-- transpose_npp filter
-- AVS2 video encoder via libxavs2
-- amultiply filter
-- Block-Matching 3d (bm3d) denoising filter
-- acrossover filter
-- ilbc decoder
-- audio denoiser as afftdn filter
-- AV1 parser
-- SER demuxer
-- sinc audio filter source
-- chromahold filter
-- setparams filter
-- vibrance filter
-- decoding S12M timecode in h264
-- xstack filter
-- pcm vidc decoder and encoder
-- (a)graphmonitor filter
-- yadif_cuda filter
-
-
-version 4.0:
-- Bitstream filters for editing metadata in H.264, HEVC and MPEG-2 streams
-- Dropped support for OpenJPEG versions 2.0 and below. Using OpenJPEG now
-  requires 2.1 (or later) and pkg-config.
-- VDA dropped (use VideoToolbox instead)
-- MagicYUV encoder
-- Raw AMR-NB and AMR-WB demuxers
-- TiVo ty/ty+ demuxer
-- Intel QSV-accelerated MJPEG encoding
-- PCE support for extended channel layouts in the AAC encoder
-- native aptX and aptX HD encoder and decoder
-- Raw aptX and aptX HD muxer and demuxer
-- NVIDIA NVDEC-accelerated H.264, HEVC, MJPEG, MPEG-1/2/4, VC1, VP8/9 hwaccel decoding
-- Intel QSV-accelerated overlay filter
-- mcompand audio filter
-- acontrast audio filter
-- OpenCL overlay filter
-- video mix filter
-- video normalize filter
-- audio lv2 wrapper filter
-- VAAPI MJPEG and VP8 decoding
-- AMD AMF H.264 and HEVC encoders
-- video fillborders filter
-- video setrange filter
-- nsp demuxer
-- support LibreSSL (via libtls)
-- AVX-512/ZMM support added
-- Dropped support for building for Windows XP. The minimum supported Windows
-  version is Windows Vista.
-- deconvolve video filter
-- entropy video filter
-- hilbert audio filter source
-- aiir audio filter
-- aiff: add support for CD-ROM XA ADPCM
-- Removed the ffserver program
-- Removed the ffmenc and ffmdec muxer and demuxer
-- VideoToolbox HEVC encoder and hwaccel
-- VAAPI-accelerated ProcAmp (color balance), denoise and sharpness filters
-- Add android_camera indev
-- codec2 en/decoding via libcodec2
-- muxer/demuxer for raw codec2 files and .c2 files
-- Moved nvidia codec headers into an external repository.
-  They can be found at http://git.videolan.org/?p=ffmpeg/nv-codec-headers.git
-- native SBC encoder and decoder
-- drmeter audio filter
-- hapqa_extract bitstream filter
-- filter_units bitstream filter
-- AV1 Support through libaom
-- E-AC-3 dependent frames support
-- bitstream filter for extracting E-AC-3 core
-- Haivision SRT protocol via libsrt
-- segafilm muxer
-- vfrdet filter
-- SRCNN filter
-
+version 3.4.1:
+- avcodec/vp9_superframe_split_bsf: Fix integer overflow in frame_size/total_size checks
+- avcodec/amrwbdec: Fix division by 0 in voice_factor()
+- avformat/utils: Fix warning: ISO C90 forbids mixed declarations and code
+- avcodec/decode: reset codec on receiving packet after EOF in compat_decode
+- avcodec/diracdsp: Fix integer overflow in PUT_SIGNED_RECT_CLAMPED()
+- avcodec/dirac_dwt: Fix integer overflows in COMPOSE_DAUB97*
+- avcodec/extract_extradata_bsf: Fix leak discovered via fuzzing
+- avcodec/vorbis: Fix another 1 << 31 > int32_t::max() with 1u.
+- avcodec/vorbis: 1 << 31 > int32_t::max(), so use 1u << 31 instead.
+- avformat/utils: Prevent undefined shift with wrap_bits > 64.
+- avcodec/j2kenc: Fix out of array access in encode_cblk()
+- avcodec/hevcdsp_template: Fix undefined shift in put_hevc_epel_bi_w_h()
+- lavf/mov: fix huge alloc in mov_read_ctts
+- avcodec/mlpdsp: Fix signed integer overflow, 2nd try
+- avcodec/h264idct_template: Fix integer overflow in ff_h264_idct8_add
+- avcodec/kgv1dec: Check that there is enough input for maximum RLE compression
+- avformat/aacdec: Fix leak in adts_aac_read_packet()
+- avcodec/dirac_dwt: Fix integer overflow in COMPOSE_FIDELITYi*
+- avcodec/sbrdsp_fixed: Fix integer overflow
+- avcodec/mpeg4videodec: Check also for negative versions in the validity check
+- Close ogg stream upon error when using AV_EF_EXPLODE.
+- Fix undefined shift on assumed 8-bit input.
+- Use ff_thread_once for fixed, float table init.
+- Fix leak of frame_duration_buffer in mov_fix_index().
+- avformat/mov: Propagate errors in mov_switch_root.
+- avcodec/hevcdsp_template: Fix invalid shift in put_hevc_epel_bi_w_v()
+- avcodec/mlpdsp: Fix undefined shift ff_mlp_pack_output()
+- avcodec/zmbv: Check that the buffer is large enough for mvec
+- avcodec/dirac_dwt: Fix integer overflow in COMPOSE_DD137iL0()
+- avcodec/wmv2dec: Check end of bitstream in parse_mb_skip() and ff_wmv2_decode_mb()
+- avcodec/snowdec: Check for remaining bitstream in decode_blocks()
+- avcodec/snowdec: Check intra block dc differences.
+- avformat/mov: Check size of STSC allocation
+- avcodec/vc2enc: Clear coef_buf on allocation
+- avcodec/h264dec: Fix potential array overread
+- avcodec/x86/mpegvideodsp: Fix signedness bug in need_emu
+- avcodec/aacpsdsp_template: Fix integer overflows in ps_decorrelate_c()
+- avcodec/aacdec_fixed: Fix undefined shift
+- avcodec/mdct_*: Fix integer overflow in addition in RESCALE()
+- avcodec/snowdec: Fix integer overflow in header parsing
+- avcodec/cngdec: Fix integer clipping
+- avcodec/sbrdsp_fixed: Fix integer overflow in shift in sbr_hf_g_filt_c()
+- avcodec/aacsbr_fixed: Fix division by zero in sbr_gain_calc()
+- avutil/softfloat: Add FLOAT_MIN
+- avcodec/h264idct_template: Fix integer overflows in ff_h264_idct8_add()
+- avcodec/xan: Check for bitstream end in xan_huffman_decode()
+- avcodec/exr: fix undefined shift in pxr24_uncompress()
+- avformat: Free the internal codec context at the end
+- avcodec/h264idct_template: Fix integer overflows in ff_h264_idct8_add()
+- avcodec/xan: Improve overlapping check
+- avcodec/aacdec_fixed: Fix integer overflow in apply_dependent_coupling_fixed()
+- avcodec/aacdec_fixed: Fix integer overflow in predict()
+- avcodec/jpeglsdec: Check for end of bitstream in ls_decode_line()
+- avcodec/jpeglsdec: Check ilv for being a supported value
+- tests/ffserver.regression.ref: update checksums to what ffserver currently produces
+- ffserver: Fix off by 1 error in path
+- avcodec/proresdec: align dequantization matrix buffers
+- avformat/matroskaenc: add missing allocation failure checks for stream durations
+- avformat/matroskaenc: actually enforce the stream limit
+- configure: Fix dependencies of aac_at decoder.
+- Don't manipulate duration when it's AV_NOPTS_VALUE.
+- lavfi/af_pan: fix sign handling in channel coefficient parser
+- avformat/hlsenc: write fmp4 init header after first AV frame
+- avformat/hlsenc: allocate space for terminating null
+- avformat/hlsenc: reindent hlsenc code
+- avformat/hlsenc: check hls segment mode for ignore the init filename
+- avformat/hlsenc: reindent hlsenc code
+- avformat/hlsenc: fix missing first segment bug in fmp4 mode
+- avformat/hlsenc: fix base_output_dirname is null when basename_size is 0 bug
+- ffplay: use SDL2 audio API
+- ffplay: only use hardware accelerated SDL texture formats
+- ffplay: create the window and the renderer before starting playback
+- ffmpeg: always init output stream before reaping filters
+- vc2enc_dwt: pad the temporary buffer by the slice size
+- lavu/arm: Check for have_vfp_vm instead of !have_vfpv3 for float_dsp_vfp
 
 version 3.4:
 - deflicker video filter

INSTALL.md

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

LICENSE.md

@@ -103,9 +103,18 @@ license version needs to be upgraded by passing `--enable-version3` to configure
 There are certain libraries you can combine with FFmpeg whose licenses are not
 compatible with the GPL and/or the LGPL. If you wish to enable these
 libraries, even in circumstances that their license may be incompatible, pass
-`--enable-nonfree` to configure. This will cause the resulting binary to be
+`--enable-nonfree` to configure. But note that if you enable any of these
+libraries the resulting binary will be under a complex license mix that is
+more restrictive than the LGPL and that may result in additional obligations.
+It is possible that these restrictions cause the resulting binary to be
 unredistributable.
 
 The Fraunhofer FDK AAC and OpenSSL libraries are under licenses which are
 incompatible with the GPLv2 and v3. To the best of our knowledge, they are
 compatible with the LGPL.
+
+The NVENC library, while its header file is licensed under the compatible MIT
+license, requires a proprietary binary blob at run time, and is deemed to be
+incompatible with the GPL. We are not certain if it is compatible with the
+LGPL, but we require `--enable-nonfree` even with LGPL configurations in case
+it is not.

MAINTAINERS

@@ -29,6 +29,9 @@ ffplay:
 ffprobe:
   ffprobe.c                             Stefano Sabatini
 
+ffserver:
+  ffserver.c                            Reynaldo H. Verdejo Pinochet
+
 Commandline utility code:
   cmdutils.c, cmdutils.h                Michael Niedermayer
 
@@ -39,7 +42,7 @@ QuickTime faststart:
 Miscellaneous Areas
 ===================
 
-documentation                           Stefano Sabatini, Mike Melanson, Timothy Gu, Gyan Doshi
+documentation                           Stefano Sabatini, Mike Melanson, Timothy Gu, Lou Logan
 project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser, Nikolay Aleksandrov
 presets                                 Robert Swain
 metadata subsystem                      Aurelien Jacobs
@@ -52,8 +55,8 @@ Communication
 
 website                                 Deby Barbara Lepage
 fate.ffmpeg.org                         Timothy Gu
-Trac bug tracker                        Alexander Strasser, Michael Niedermayer, Carl Eugen Hoyos
-mailing lists                           Baptiste Coudurier
+Trac bug tracker                        Alexander Strasser, Michael Niedermayer, Carl Eugen Hoyos, Lou Logan
+mailing lists                           Baptiste Coudurier, Lou Logan
 Google+                                 Paul B Mahol, Michael Niedermayer, Alexander Strasser
 Twitter                                 Lou Logan, Reynaldo H. Verdejo Pinochet
 Launchpad                               Timothy Gu
@@ -121,6 +124,7 @@ Generic Parts:
     motion*                             Michael Niedermayer
   rate control:
     ratecontrol.c                       Michael Niedermayer
+    libxvid_rc.c                        Michael Niedermayer
   simple IDCT:
     simple_idct.c, simple_idct.h        Michael Niedermayer
   postprocessing:
@@ -138,12 +142,10 @@ Codecs:
   aacenc*, aaccoder.c                   Rostislav Pehlivanov
   alacenc.c                             Jaikrishnan Menon
   alsdec.c                              Thilo Borgmann, Umair Khan
-  aptx.c                                Aurelien Jacobs
   ass*                                  Aurelien Jacobs
   asv*                                  Michael Niedermayer
   atrac3plus*                           Maxim Poliakovski
   audiotoolbox*                         Rodger Combs
-  avs2*                                 Huiwen Ren
   bgmc.c, bgmc.h                        Thilo Borgmann
   binkaudio.c                           Peter Ross
   cavs*                                 Stefan Gehrer
@@ -156,7 +158,7 @@ Codecs:
   cpia.c                                Stephan Hilb
   crystalhd.c                           Philip Langdale
   cscd.c                                Reimar Doeffinger
-  cuviddec.c                            Timo Rothenpieler
+  cuvid.c                               Timo Rothenpieler
   dca*                                  foo86
   dirac*                                Rostislav Pehlivanov
   dnxhd*                                Baptiste Coudurier
@@ -187,9 +189,7 @@ Codecs:
   jvdec.c                               Peter Ross
   lcl*.c                                Roberto Togni, Reimar Doeffinger
   libcelt_dec.c                         Nicolas George
-  libcodec2.c                           Tomas Härdin
   libdirac*                             David Conrad
-  libdavs2.c                            Huiwen Ren
   libgsm.c                              Michel Bardiaux
   libkvazaar.c                          Arttu Ylä-Outinen
   libopenjpeg.c                         Jaikrishnan Menon
@@ -212,7 +212,7 @@ Codecs:
   msrle.c                               Mike Melanson
   msvideo1.c                            Mike Melanson
   nuv.c                                 Reimar Doeffinger
-  nvdec*, nvenc*                        Timo Rothenpieler
+  nvenc*                                Timo Rothenpieler
   opus*                                 Rostislav Pehlivanov
   paf.*                                 Paul B Mahol
   pcx.c                                 Ivo van Poorten
@@ -220,7 +220,7 @@ Codecs:
   ptx.c                                 Ivo van Poorten
   qcelp*                                Reynaldo H. Verdejo Pinochet
   qdm2.c, qdm2data.h                    Roberto Togni
-  qsv*                                  Mark Thompson, Zhong Li
+  qsv*                                  Mark Thompson
   qtrle.c                               Mike Melanson
   ra144.c, ra144.h, ra288.c, ra288.h    Roberto Togni
   resample2.c                           Michael Niedermayer
@@ -242,10 +242,10 @@ Codecs:
   tta.c                                 Alex Beregszaszi, Jaikrishnan Menon
   ttaenc.c                              Paul B Mahol
   txd.c                                 Ivo van Poorten
-  v4l2_*                                Jorge Ramirez-Ortiz
   vc2*                                  Rostislav Pehlivanov
   vcr1.c                                Michael Niedermayer
-  videotoolboxenc.c                     Rick Kern, Aman Gupta
+  vda_h264_dec.c                        Xidorn Quan
+  videotoolboxenc.c                     Rick Kern
   vima.c                                Paul B Mahol
   vorbisdec.c                           Denes Balatoni, David Conrad
   vorbisenc.c                           Oded Shimon
@@ -268,11 +268,11 @@ Hardware acceleration:
   crystalhd.c                           Philip Langdale
   dxva2*                                Hendrik Leppkes, Laurent Aimar, Steve Lhomme
   d3d11va*                              Steve Lhomme
-  mediacodec*                           Matthieu Bouron, Aman Gupta
+  mediacodec*                           Matthieu Bouron
   vaapi*                                Gwenole Beauchesne
   vaapi_encode*                         Mark Thompson
   vdpau*                                Philip Langdale, Carl Eugen Hoyos
-  videotoolbox*                         Rick Kern, Aman Gupta
+  videotoolbox*                         Rick Kern
 
 
 libavdevice
@@ -282,7 +282,6 @@ libavdevice
 
 
   avfoundation.m                        Thilo Borgmann
-  android_camera.c                      Felix Matouschek
   decklink*                             Marton Balint
   dshow.c                               Roger Pack (CC rogerdpack@gmail.com)
   fbdev_enc.c                           Lukasz Marek
@@ -333,7 +332,6 @@ Filters:
   vf_bwdif                              Thomas Mundt (CC <thomas.mundt@hr.de>)
   vf_chromakey.c                        Timo Rothenpieler
   vf_colorchannelmixer.c                Paul B Mahol
-  vf_colorconstancy.c                   Mina Sami    (CC <minas.gorgy@gmail.com>)
   vf_colorbalance.c                     Paul B Mahol
   vf_colorkey.c                         Timo Rothenpieler
   vf_colorlevels.c                      Paul B Mahol
@@ -361,7 +359,6 @@ Filters:
   vf_ssim.c                             Paul B Mahol
   vf_stereo3d.c                         Paul B Mahol
   vf_telecine.c                         Paul B Mahol
-  vf_tonemap_opencl.c                   Ruiling Song
   vf_yadif.c                            Michael Niedermayer
   vf_zoompan.c                          Paul B Mahol
 
@@ -398,10 +395,8 @@ Muxers/Demuxers:
   brstm.c                               Paul B Mahol
   caf*                                  Peter Ross
   cdxl.c                                Paul B Mahol
-  codec2.c                              Tomas Härdin
   crc.c                                 Michael Niedermayer
   dashdec.c                             Steven Liu
-  dashenc.c                             Karthick Jeyapal
   daud.c                                Reimar Doeffinger
   dss.c                                 Oleksij Rempel
   dtsdec.c                              foo86
@@ -415,6 +410,7 @@ Muxers/Demuxers:
   flvenc.c                              Michael Niedermayer, Steven Liu
   gxf.c                                 Reimar Doeffinger
   gxfenc.c                              Baptiste Coudurier
+  hls.c                                 Anssi Hannula
   hlsenc.c                              Christian Suloway, Steven Liu
   idcin.c                               Mike Melanson
   idroqdec.c                            Mike Melanson
@@ -525,10 +521,9 @@ Operating systems / CPU architectures
 =====================================
 
 Alpha                                   Falk Hueffner
-MIPS                                    Manojkumar Bhosale, Shiyou Yin
+MIPS                                    Manojkumar Bhosale
 Mac OS X / PowerPC                      Romain Dolbeau, Guillaume Poirier
 Amiga / PowerPC                         Colin Ward
-Linux / PowerPC                         Lauri Kasanen
 Windows MinGW                           Alex Beregszaszi, Ramiro Polla
 Windows Cygwin                          Victor Paesa
 Windows MSVC                            Matthew Oliver, Hendrik Leppkes
@@ -554,7 +549,6 @@ Ivan Uskov
 James Darnley
 Jan Ekström
 Joakim Plate
-Jun Zhao
 Kieran Kunhya
 Kirill Gavrilov
 Martin Storsjö
@@ -577,11 +571,8 @@ Releases
 If you want to maintain an older release, please contact us
 
 
-GnuPG Fingerprints and IRC nicknames of maintainers and contributors
-====================================================================
-
-IRC nicknames are in parentheses. These apply
-to the IRC channels listed on the website.
+GnuPG Fingerprints of maintainers and contributors
+==================================================
 
 Alexander Strasser            1C96 78B7 83CB 8AA7 9AF5 D1EB A7D8 A57B A876 E58F
 Anssi Hannula                 1A92 FF42 2DD9 8D2E 8AF7 65A9 4278 C520 513D F3CB
@@ -596,10 +587,9 @@ FFmpeg release signing key    FCF9 86EA 15E6 E293 A564 4F10 B432 2F04 D676 58D8
 Ganesh Ajjanagadde            C96A 848E 97C3 CEA2 AB72 5CE4 45F9 6A2D 3C36 FB1B
 Gwenole Beauchesne            2E63 B3A6 3E44 37E2 017D 2704 53C7 6266 B153 99C4
 Jaikrishnan Menon             61A1 F09F 01C9 2D45 78E1 C862 25DC 8831 AF70 D368
-James Almer                   7751 2E8C FD94 A169 57E6 9A7A 1463 01AD 7376 59E0
 Jean Delvare                  7CA6 9F44 60F1 BDC4 1FD2 C858 A552 6B9B B3CD 4E6A
 Loren Merritt                 ABD9 08F4 C920 3F65 D8BE 35D7 1540 DAA7 060F 56DE
-Lou Logan (llogan)            7D68 DC73 CBEF EABB 671A B6CF 621C 2E28 82F8 DC3A
+Lou Logan                     7D68 DC73 CBEF EABB 671A B6CF 621C 2E28 82F8 DC3A
 Michael Niedermayer           9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
 Nicolas George                24CE 01CE 9ACC 5CEB 74D8 8D9D B063 D997 36E5 4C93
 Nikolay Aleksandrov           8978 1D8C FB71 588E 4B27 EAA8 C4F0 B5FC E011 13B1
@@ -616,5 +606,5 @@ Steinar H. Gunderson          C2E9 004F F028 C18E 4EAD DB83 7F61 7561 7797 8F76
 Stephan Hilb                  4F38 0B3A 5F39 B99B F505 E562 8D5C 5554 4E17 8863
 Tiancheng "Timothy" Gu        9456 AFC0 814A 8139 E994 8351 7FE6 B095 B582 B0D4
 Tim Nicholson                 38CF DB09 3ED0 F607 8B67 6CED 0C0B FC44 8B0B FC83
-Tomas Härdin (thardin)        A79D 4E3D F38F 763F 91F5 8B33 A01E 8AE0 41BB 2551
+Tomas Härdin                  A79D 4E3D F38F 763F 91F5 8B33 A01E 8AE0 41BB 2551
 Wei Gao                       4269 7741 857A 0E60 9EC5 08D2 4744 4EFA 62C1 87B9

Makefile

@@ -45,14 +45,12 @@ FF_DEP_LIBS  := $(DEP_LIBS)
 FF_STATIC_DEP_LIBS := $(STATIC_DEP_LIBS)
 
 $(TOOLS): %$(EXESUF): %.o
-	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(EXTRALIBS-$(*F)) $(EXTRALIBS) $(ELIBS)
+	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS)
 
 target_dec_%_fuzzer$(EXESUF): target_dec_%_fuzzer.o $(FF_DEP_LIBS)
 	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
 
-tools/target_dem_fuzzer$(EXESUF): tools/target_dem_fuzzer.o $(FF_DEP_LIBS)
-	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
-
+tools/cws2fws$(EXESUF): ELIBS = $(ZLIB)
 tools/sofa2wavs$(EXESUF): ELIBS = $(FF_EXTRALIBS)
 tools/uncoded_frame$(EXESUF): $(FF_DEP_LIBS)
 tools/uncoded_frame$(EXESUF): ELIBS = $(FF_EXTRALIBS)
@@ -61,7 +59,6 @@ tools/target_dec_%_fuzzer$(EXESUF): $(FF_DEP_LIBS)
 CONFIGURABLE_COMPONENTS =                                           \
     $(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c))                 \
     $(SRC_PATH)/libavcodec/bitstream_filters.c                      \
-    $(SRC_PATH)/libavcodec/parsers.c                                \
     $(SRC_PATH)/libavformat/protocols.c                             \
 
 config.h: ffbuild/.config
@@ -131,24 +128,25 @@ install-data: $(DATA_FILES)
 	$(Q)mkdir -p "$(DATADIR)"
 	$(INSTALL) -m 644 $(DATA_FILES) "$(DATADIR)"
 
-uninstall: uninstall-data uninstall-headers uninstall-libs uninstall-pkgconfig
+uninstall: uninstall-libs uninstall-headers uninstall-data
 
 uninstall-data:
 	$(RM) -r "$(DATADIR)"
 
 clean::
 	$(RM) $(CLEANSUFFIXES)
-	$(RM) $(addprefix compat/,$(CLEANSUFFIXES)) $(addprefix compat/*/,$(CLEANSUFFIXES)) $(addprefix compat/*/*/,$(CLEANSUFFIXES))
+	$(RM) $(CLEANSUFFIXES:%=compat/msvcrt/%)
+	$(RM) $(CLEANSUFFIXES:%=compat/atomics/pthread/%)
+	$(RM) $(CLEANSUFFIXES:%=compat/%)
 	$(RM) -r coverage-html
 	$(RM) -rf coverage.info coverage.info.in lcov
 
-distclean:: clean
+distclean::
+	$(RM) $(DISTCLEANSUFFIXES)
 	$(RM) .version avversion.h config.asm config.h mapfile  \
 		ffbuild/.config ffbuild/config.* libavutil/avconfig.h \
 		version.h libavutil/ffversion.h libavcodec/codec_names.h \
-		libavcodec/bsf_list.c libavformat/protocol_list.c \
-		libavcodec/codec_list.c libavcodec/parser_list.c \
-		libavformat/muxer_list.c libavformat/demuxer_list.c
+		libavcodec/bsf_list.c libavformat/protocol_list.c
 ifeq ($(SRC_LINK),src)
 	$(RM) src
 endif
@@ -157,12 +155,11 @@ endif
 config:
 	$(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION)
 
-build: all alltools examples testprogs
 check: all alltools examples testprogs fate
 
 include $(SRC_PATH)/tests/Makefile
 
-$(sort $(OUTDIRS)):
+$(sort $(OBJDIRS)):
 	$(Q)mkdir -p $@
 
 # Dummy rule to stop make trying to rebuild removed or renamed headers
@@ -173,5 +170,4 @@ $(sort $(OUTDIRS)):
 # so this saves some time on slow systems.
 .SUFFIXES:
 
-.PHONY: all all-yes alltools build check config testprogs
-.PHONY: *clean install* uninstall*
+.PHONY: all all-yes alltools check *clean config install* testprogs uninstall*

README.md

@@ -21,6 +21,8 @@ such as audio, video, subtitles and related metadata.
 * [ffplay](https://ffmpeg.org/ffplay.html) is a minimalistic multimedia player.
 * [ffprobe](https://ffmpeg.org/ffprobe.html) is a simple analysis tool to inspect
   multimedia content.
+* [ffserver](https://ffmpeg.org/ffserver.html) is a multimedia streaming server
+  for live broadcasts.
 * Additional small tools such as `aviocat`, `ismindex` and `qt-faststart`.
 
 ## Documentation

RELEASE

@@ -1 +1 @@
-4.2
+3.4.1

RELEASE_NOTES

@@ -1,13 +1,13 @@
 
-              ┌────────────────────────────────────┐
-              │ RELEASE NOTES for FFmpeg 4.2 "Ada" │
-              └────────────────────────────────────┘
+              ┌───────────────────────────────────────┐
+              │ RELEASE NOTES for FFmpeg 3.4 "Cantor" │
+              └───────────────────────────────────────┘
 
-   The FFmpeg Project proudly presents FFmpeg 4.2 "Ada", about 8
-   months after the release of FFmpeg 4.1.
+   The FFmpeg Project proudly presents FFmpeg 3.4 "Cantor", about 6
+   months after the release of FFmpeg 3.3.
 
    A complete Changelog is available at the root of the project, and the
-   complete Git history on https://git.ffmpeg.org/gitweb/ffmpeg.git
+   complete Git history on http://source.ffmpeg.org.
 
    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,

compat/avisynth/avisynth_c.h

@@ -34,22 +34,6 @@
 // NOTE: this is a partial update of the Avisynth C interface to recognize
 // new color spaces added in Avisynth 2.60. By no means is this document
 // completely Avisynth 2.60 compliant.
-// 170103: added new CPU constants (FMA4, AVX512xx)
-// 171102: define SIZETMOD. do not use yet, experimental. Offsets are size_t instead of int. Affects x64.
-// 171106: avs_get_row_size calls into avs_get_row_size_p, instead of direct field access
-// 171106: avs_get_height calls into avs_get_row_size_p, instead of direct field access
-// 180524: AVSC_EXPORT to dllexport in capi.h for avisynth_c_plugin_init
-// 180524: avs_is_same_colorspace VideoInfo parameters to const
-// 181230: Readability: functions regrouped to mix less AVSC_API and AVSC_INLINE, put together Avisynth+ specific stuff
-// 181230: use #ifndef AVSC_NO_DECLSPEC for AVSC_INLINE functions which are calling API functions
-// 181230: comments on avs_load_library (helper for loading API entries dynamically into a struct using AVSC_NO_DECLSPEC define)
-// 181230: define alias AVS_FRAME_ALIGN as FRAME_ALIGN
-// 181230: remove unused form of avs_get_rowsize and avs_get_height (kept earlier for reference)
-// 190104: avs_load_library: smart fallback mechanism for Avisynth+ specific functions:
-//         if they are not loadable, they will work in a classic Avisynth compatible mode
-//         Example#1: e.g. avs_is_444 will call the existing avs_is_yv24 instead
-//         Example#2: avs_bits_per_component will return 8 for all colorspaces (Classic Avisynth supports only 8 bits/pixel)
-//         Thus the Avisynth+ specific API functions are safely callable even when connected to classic Avisynth DLL
 
 #ifndef __AVISYNTH_C__
 #define __AVISYNTH_C__
@@ -58,7 +42,7 @@
 #include "avs/capi.h"
 #include "avs/types.h"
 
-#define AVS_FRAME_ALIGN FRAME_ALIGN
+
 /////////////////////////////////////////////////////////////////////
 //
 // Constants
@@ -140,7 +124,7 @@ enum {
     AVS_CS_GENERIC_YUVA444 = AVS_CS_PLANAR | AVS_CS_YUVA | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_1 }; // 4:4:4:A planar
 
 
-  // Specific color formats
+  // Specific colorformats
 enum {
   AVS_CS_UNKNOWN = 0,
   AVS_CS_BGR24 = AVS_CS_RGB_TYPE  | AVS_CS_BGR | AVS_CS_INTERLEAVED,
@@ -150,18 +134,18 @@ enum {
   //  AVS_CS_I420  = 1<<4  Reserved
   AVS_CS_RAW32 = 1<<5 | AVS_CS_INTERLEAVED,
 
-  AVS_CS_YV24  = AVS_CS_GENERIC_YUV444 | AVS_CS_SAMPLE_BITS_8,  // YUV 4:4:4 planar
-  AVS_CS_YV16  = AVS_CS_GENERIC_YUV422 | AVS_CS_SAMPLE_BITS_8,  // YUV 4:2:2 planar
-  AVS_CS_YV12  = AVS_CS_GENERIC_YUV420 | AVS_CS_SAMPLE_BITS_8,  // YUV 4:2:0 planar
+  AVS_CS_YV24  = AVS_CS_GENERIC_YUV444 | AVS_CS_SAMPLE_BITS_8,  // YVU 4:4:4 planar
+  AVS_CS_YV16  = AVS_CS_GENERIC_YUV422 | AVS_CS_SAMPLE_BITS_8,  // YVU 4:2:2 planar
+  AVS_CS_YV12  = AVS_CS_GENERIC_YUV420 | AVS_CS_SAMPLE_BITS_8,  // YVU 4:2:0 planar
   AVS_CS_I420  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_UPLANEFIRST | AVS_CS_SUB_HEIGHT_2 | AVS_CS_SUB_WIDTH_2,  // YUV 4:2:0 planar
   AVS_CS_IYUV  = AVS_CS_I420,
-  AVS_CS_YV411 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_4,  // YUV 4:1:1 planar
-  AVS_CS_YUV9  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_4 | AVS_CS_SUB_WIDTH_4,  // YUV 4:1:0 planar
+  AVS_CS_YV411 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_4,  // YVU 4:1:1 planar
+  AVS_CS_YUV9  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_4 | AVS_CS_SUB_WIDTH_4,  // YVU 4:1:0 planar
   AVS_CS_Y8    = AVS_CS_GENERIC_Y | AVS_CS_SAMPLE_BITS_8,       // Y   4:0:0 planar
 
   //-------------------------
   // AVS16: new planar constants go live! Experimental PF 160613
-  // 10-12-14-16 bit + planar RGB + BGR48/64 160725
+  // 10-12-14 bit + planar RGB + BRG48/64 160725
   AVS_CS_YUV444P10 = AVS_CS_GENERIC_YUV444 | AVS_CS_SAMPLE_BITS_10, // YUV 4:4:4 10bit samples
   AVS_CS_YUV422P10 = AVS_CS_GENERIC_YUV422 | AVS_CS_SAMPLE_BITS_10, // YUV 4:2:2 10bit samples
   AVS_CS_YUV420P10 = AVS_CS_GENERIC_YUV420 | AVS_CS_SAMPLE_BITS_10, // YUV 4:2:0 10bit samples
@@ -262,9 +246,9 @@ enum {  //SUBTYPES
 enum {
   // New 2.6 explicitly defined cache hints.
   AVS_CACHE_NOTHING=10, // Do not cache video.
-  AVS_CACHE_WINDOW=11, // Hard protect up to X frames within a range of X from the current frame N.
-  AVS_CACHE_GENERIC=12, // LRU cache up to X frames.
-  AVS_CACHE_FORCE_GENERIC=13, // LRU cache up to X frames, override any previous CACHE_WINDOW.
+  AVS_CACHE_WINDOW=11, // Hard protect upto X frames within a range of X from the current frame N.
+  AVS_CACHE_GENERIC=12, // LRU cache upto X frames.
+  AVS_CACHE_FORCE_GENERIC=13, // LRU cache upto X frames, override any previous CACHE_WINDOW.
 
   AVS_CACHE_GET_POLICY=30, // Get the current policy.
   AVS_CACHE_GET_WINDOW=31, // Get the current window h_span.
@@ -272,8 +256,8 @@ enum {
 
   AVS_CACHE_AUDIO=50, // Explicitly do cache audio, X byte cache.
   AVS_CACHE_AUDIO_NOTHING=51, // Explicitly do not cache audio.
-  AVS_CACHE_AUDIO_NONE=52, // Audio cache off (auto mode), X byte initial cache.
-  AVS_CACHE_AUDIO_AUTO=53, // Audio cache on (auto mode), X byte initial cache.
+  AVS_CACHE_AUDIO_NONE=52, // Audio cache off (auto mode), X byte intial cache.
+  AVS_CACHE_AUDIO_AUTO=53, // Audio cache on (auto mode), X byte intial cache.
 
   AVS_CACHE_GET_AUDIO_POLICY=70, // Get the current audio policy.
   AVS_CACHE_GET_AUDIO_SIZE=71, // Get the current audio cache size.
@@ -300,7 +284,7 @@ enum {
     AVS_CACHE_COST_MED=224, // Child response of medium cost. (Real time)
     AVS_CACHE_COST_HI=225, // Child response of heavy cost. (Slow)
 
-  AVS_CACHE_GETCHILD_THREAD_MODE=240, // Cache ask Child for thread safety.
+  AVS_CACHE_GETCHILD_THREAD_MODE=240, // Cache ask Child for thread safetyness.
     AVS_CACHE_THREAD_UNSAFE=241, // Only 1 thread allowed for all instances. 2.5 filters default!
     AVS_CACHE_THREAD_CLASS=242, // Only 1 thread allowed for each instance. 2.6 filters default!
     AVS_CACHE_THREAD_SAFE=243, //  Allow all threads in any instance.
@@ -313,8 +297,6 @@ enum {
   };
 
 #ifdef BUILDING_AVSCORE
-AVSValue create_c_video_filter(AVSValue args, void * user_data, IScriptEnvironment * e0);
-
 struct AVS_ScriptEnvironment {
         IScriptEnvironment * env;
         const char * error;
@@ -331,7 +313,7 @@ typedef struct AVS_ScriptEnvironment AVS_ScriptEnvironment;
 // AVS_VideoInfo
 //
 
-// AVS_VideoInfo is laid out identically to VideoInfo
+// AVS_VideoInfo is layed out identicly to VideoInfo
 typedef struct AVS_VideoInfo {
   int width, height;    // width=0 means no video
   unsigned fps_numerator, fps_denominator;
@@ -344,7 +326,7 @@ typedef struct AVS_VideoInfo {
   INT64 num_audio_samples;
   int nchannels;
 
-  // Image type properties
+  // Imagetype properties
 
   int image_type;
 } AVS_VideoInfo;
@@ -371,20 +353,77 @@ AVSC_INLINE int avs_is_yuv(const AVS_VideoInfo * p)
 AVSC_INLINE int avs_is_yuy2(const AVS_VideoInfo * p)
         { return (p->pixel_type & AVS_CS_YUY2) == AVS_CS_YUY2; }
 
-AVSC_API(int, avs_is_yv24)(const AVS_VideoInfo * p); // avs+: for generic 444 check, use avs_is_yuv444
+AVSC_API(int, avs_is_rgb48)(const AVS_VideoInfo * p);
 
-AVSC_API(int, avs_is_yv16)(const AVS_VideoInfo * p); // avs+: for generic 422 check, use avs_is_yuv422
+AVSC_API(int, avs_is_rgb64)(const AVS_VideoInfo * p);
 
-AVSC_API(int, avs_is_yv12)(const AVS_VideoInfo * p) ; // avs+: for generic 420 check, use avs_is_yuv420
+AVSC_API(int, avs_is_yv24)(const AVS_VideoInfo * p);
+
+AVSC_API(int, avs_is_yv16)(const AVS_VideoInfo * p);
+
+AVSC_API(int, avs_is_yv12)(const AVS_VideoInfo * p) ;
 
 AVSC_API(int, avs_is_yv411)(const AVS_VideoInfo * p);
 
-AVSC_API(int, avs_is_y8)(const AVS_VideoInfo * p); // avs+: for generic grayscale, use avs_is_y
+AVSC_API(int, avs_is_y8)(const AVS_VideoInfo * p);
+
+AVSC_API(int, avs_is_yuv444p16)(const AVS_VideoInfo * p);
+
+AVSC_API(int, avs_is_yuv422p16)(const AVS_VideoInfo * p);
+
+AVSC_API(int, avs_is_yuv420p16)(const AVS_VideoInfo * p);
+
+AVSC_API(int, avs_is_y16)(const AVS_VideoInfo * p);
+
+AVSC_API(int, avs_is_yuv444ps)(const AVS_VideoInfo * p);
+
+AVSC_API(int, avs_is_yuv422ps)(const AVS_VideoInfo * p);
+
+AVSC_API(int, avs_is_yuv420ps)(const AVS_VideoInfo * p);
+
+AVSC_API(int, avs_is_y32)(const AVS_VideoInfo * p);
+
+AVSC_API(int, avs_is_444)(const AVS_VideoInfo * p);
+
+AVSC_API(int, avs_is_422)(const AVS_VideoInfo * p);
+
+AVSC_API(int, avs_is_420)(const AVS_VideoInfo * p);
+
+AVSC_API(int, avs_is_y)(const AVS_VideoInfo * p);
+
+AVSC_API(int, avs_is_yuva)(const AVS_VideoInfo * p);
+
+AVSC_API(int, avs_is_planar_rgb)(const AVS_VideoInfo * p);
+
+AVSC_API(int, avs_is_planar_rgba)(const AVS_VideoInfo * p);
+
+
+
+AVSC_INLINE int avs_is_property(const AVS_VideoInfo * p, int property)
+        { return ((p->image_type & property)==property ); }
+
+AVSC_INLINE int avs_is_planar(const AVS_VideoInfo * p)
+        { return !!(p->pixel_type & AVS_CS_PLANAR); }
+
+AVSC_API(int, avs_is_color_space)(const AVS_VideoInfo * p, int c_space);
+
+AVSC_INLINE int avs_is_field_based(const AVS_VideoInfo * p)
+        { return !!(p->image_type & AVS_IT_FIELDBASED); }
+
+AVSC_INLINE int avs_is_parity_known(const AVS_VideoInfo * p)
+        { return ((p->image_type & AVS_IT_FIELDBASED)&&(p->image_type & (AVS_IT_BFF | AVS_IT_TFF))); }
+
+AVSC_INLINE int avs_is_bff(const AVS_VideoInfo * p)
+        { return !!(p->image_type & AVS_IT_BFF); }
+
+AVSC_INLINE int avs_is_tff(const AVS_VideoInfo * p)
+        { return !!(p->image_type & AVS_IT_TFF); }
 
 AVSC_API(int, avs_get_plane_width_subsampling)(const AVS_VideoInfo * p, int plane);
 
 AVSC_API(int, avs_get_plane_height_subsampling)(const AVS_VideoInfo * p, int plane);
 
+
 AVSC_API(int, avs_bits_per_pixel)(const AVS_VideoInfo * p);
 
 AVSC_API(int, avs_bytes_from_pixels)(const AVS_VideoInfo * p, int pixels);
@@ -393,42 +432,10 @@ AVSC_API(int, avs_row_size)(const AVS_VideoInfo * p, int plane);
 
 AVSC_API(int, avs_bmp_size)(const AVS_VideoInfo * vi);
 
-AVSC_API(int, avs_is_color_space)(const AVS_VideoInfo * p, int c_space);
-
-// no API for these, inline helper functions
-AVSC_INLINE int avs_is_property(const AVS_VideoInfo * p, int property)
-{
-  return ((p->image_type & property) == property);
-}
-
-AVSC_INLINE int avs_is_planar(const AVS_VideoInfo * p)
-{
-  return !!(p->pixel_type & AVS_CS_PLANAR);
-}
-
-AVSC_INLINE int avs_is_field_based(const AVS_VideoInfo * p)
-{
-  return !!(p->image_type & AVS_IT_FIELDBASED);
-}
-
-AVSC_INLINE int avs_is_parity_known(const AVS_VideoInfo * p)
-{
-  return ((p->image_type & AVS_IT_FIELDBASED) && (p->image_type & (AVS_IT_BFF | AVS_IT_TFF)));
-}
-
-AVSC_INLINE int avs_is_bff(const AVS_VideoInfo * p)
-{
-  return !!(p->image_type & AVS_IT_BFF);
-}
-
-AVSC_INLINE int avs_is_tff(const AVS_VideoInfo * p)
-{
-  return !!(p->image_type & AVS_IT_TFF);
-}
-
 AVSC_INLINE int avs_samples_per_second(const AVS_VideoInfo * p)
         { return p->audio_samples_per_second; }
 
+
 AVSC_INLINE int avs_bytes_per_channel_sample(const AVS_VideoInfo * p)
 {
     switch (p->sample_type) {
@@ -440,7 +447,6 @@ AVSC_INLINE int avs_bytes_per_channel_sample(const AVS_VideoInfo * p)
       default: return 0;
     }
 }
-
 AVSC_INLINE int avs_bytes_per_audio_sample(const AVS_VideoInfo * p)
         { return p->nchannels*avs_bytes_per_channel_sample(p);}
 
@@ -482,56 +488,19 @@ AVSC_INLINE void avs_set_fps(AVS_VideoInfo * p, unsigned numerator, unsigned den
     p->fps_denominator = denominator/x;
 }
 
-#ifndef AVSC_NO_DECLSPEC
-// this inline function is calling an API function
-AVSC_INLINE int avs_is_same_colorspace(const AVS_VideoInfo * x, const AVS_VideoInfo * y)
+#ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
+AVSC_INLINE int avs_is_same_colorspace(AVS_VideoInfo * x, AVS_VideoInfo * y)
 {
         return (x->pixel_type == y->pixel_type)
                 || (avs_is_yv12(x) && avs_is_yv12(y));
 }
 #endif
 
-// Avisynth+ extensions
-AVSC_API(int, avs_is_rgb48)(const AVS_VideoInfo * p);
-
-AVSC_API(int, avs_is_rgb64)(const AVS_VideoInfo * p);
-
-AVSC_API(int, avs_is_yuv444p16)(const AVS_VideoInfo * p); // obsolete, use avs_is_yuv444
-
-AVSC_API(int, avs_is_yuv422p16)(const AVS_VideoInfo * p); // obsolete, use avs_is_yuv422
-
-AVSC_API(int, avs_is_yuv420p16)(const AVS_VideoInfo * p); // obsolete, use avs_is_yuv420
-
-AVSC_API(int, avs_is_y16)(const AVS_VideoInfo * p); // obsolete, use avs_is_y
-
-AVSC_API(int, avs_is_yuv444ps)(const AVS_VideoInfo * p); // obsolete, use avs_is_yuv444
-
-AVSC_API(int, avs_is_yuv422ps)(const AVS_VideoInfo * p); // obsolete, use avs_is_yuv422
-
-AVSC_API(int, avs_is_yuv420ps)(const AVS_VideoInfo * p); // obsolete, use avs_is_yuv420
-
-AVSC_API(int, avs_is_y32)(const AVS_VideoInfo * p); // obsolete, use avs_is_y
-
-AVSC_API(int, avs_is_444)(const AVS_VideoInfo * p);
-
-AVSC_API(int, avs_is_422)(const AVS_VideoInfo * p);
-
-AVSC_API(int, avs_is_420)(const AVS_VideoInfo * p);
-
-AVSC_API(int, avs_is_y)(const AVS_VideoInfo * p);
-
-AVSC_API(int, avs_is_yuva)(const AVS_VideoInfo * p);
-
-AVSC_API(int, avs_is_planar_rgb)(const AVS_VideoInfo * p);
-
-AVSC_API(int, avs_is_planar_rgba)(const AVS_VideoInfo * p);
-
 AVSC_API(int, avs_num_components)(const AVS_VideoInfo * p);
 
 AVSC_API(int, avs_component_size)(const AVS_VideoInfo * p);
 
 AVSC_API(int, avs_bits_per_component)(const AVS_VideoInfo * p);
-// end of Avisynth+ specific
 
 /////////////////////////////////////////////////////////////////////
 //
@@ -544,15 +513,11 @@ AVSC_API(int, avs_bits_per_component)(const AVS_VideoInfo * p);
 // to be reused.  The instances are deleted when the corresponding AVS
 // file is closed.
 
-// AVS_VideoFrameBuffer is laid out identically to VideoFrameBuffer
+// AVS_VideoFrameBuffer is layed out identicly to VideoFrameBuffer
 // DO NOT USE THIS STRUCTURE DIRECTLY
 typedef struct AVS_VideoFrameBuffer {
   BYTE * data;
-#ifdef SIZETMOD
-  size_t data_size;
-#else
   int data_size;
-#endif
   // sequence_number is incremented every time the buffer is changed, so
   // that stale views can tell they're no longer valid.
   volatile long sequence_number;
@@ -562,94 +527,56 @@ typedef struct AVS_VideoFrameBuffer {
 
 // VideoFrame holds a "window" into a VideoFrameBuffer.
 
-// AVS_VideoFrame is laid out identically to IVideoFrame
+// AVS_VideoFrame is layed out identicly to IVideoFrame
 // DO NOT USE THIS STRUCTURE DIRECTLY
 typedef struct AVS_VideoFrame {
   volatile long refcount;
   AVS_VideoFrameBuffer * vfb;
-#ifdef SIZETMOD
-  size_t offset;
-#else
-  int offset;
-#endif
-  int pitch, row_size, height;
-#ifdef SIZETMOD
-  size_t offsetU, offsetV;
-#else
-  int offsetU, offsetV;
-#endif
-  int pitchUV;  // U&V offsets are from top of picture.
-  int row_sizeUV, heightUV; // for Planar RGB offsetU, offsetV is for the 2nd and 3rd Plane.
-                            // for Planar RGB pitchUV and row_sizeUV = 0, because when no VideoInfo (MakeWriteable)
-                            // the decision on existence of UV is checked by zero pitch
-  // AVS+ extension, avisynth.h: class does not break plugins if appended here
-#ifdef SIZETMOD
-  size_t offsetA;
-#else
-  int offsetA;
-#endif
-  int pitchA, row_sizeA; // 4th alpha plane support, pitch and row_size is 0 is none
+  int offset, pitch, row_size, height, offsetU, offsetV, pitchUV;  // U&V offsets are from top of picture.
+  int row_sizeUV, heightUV;
 } AVS_VideoFrame;
 
 // Access functions for AVS_VideoFrame
 AVSC_API(int, avs_get_pitch_p)(const AVS_VideoFrame * p, int plane);
 
+#ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
+AVSC_INLINE int avs_get_pitch(const AVS_VideoFrame * p) {
+        return avs_get_pitch_p(p, 0);}
+#endif
+
 AVSC_API(int, avs_get_row_size_p)(const AVS_VideoFrame * p, int plane);
 
+AVSC_INLINE int avs_get_row_size(const AVS_VideoFrame * p) {
+        return p->row_size; }
+
 AVSC_API(int, avs_get_height_p)(const AVS_VideoFrame * p, int plane);
 
+AVSC_INLINE int avs_get_height(const AVS_VideoFrame * p) {
+        return p->height;}
+
 AVSC_API(const BYTE *, avs_get_read_ptr_p)(const AVS_VideoFrame * p, int plane);
 
+#ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
+AVSC_INLINE const BYTE* avs_get_read_ptr(const AVS_VideoFrame * p) {
+        return avs_get_read_ptr_p(p, 0);}
+#endif
+
 AVSC_API(int, avs_is_writable)(const AVS_VideoFrame * p);
 
 AVSC_API(BYTE *, avs_get_write_ptr_p)(const AVS_VideoFrame * p, int plane);
 
-AVSC_API(void, avs_release_video_frame)(AVS_VideoFrame *);
-// makes a shallow copy of a video frame
-AVSC_API(AVS_VideoFrame *, avs_copy_video_frame)(AVS_VideoFrame *);
-
-// no API for these, inline helper functions
-#ifndef AVSC_NO_DECLSPEC
-// this inline function is calling an API function
-AVSC_INLINE int avs_get_pitch(const AVS_VideoFrame * p) {
-  return avs_get_pitch_p(p, 0);
-}
-#endif
-
-#ifndef AVSC_NO_DECLSPEC
-// this inline function is calling an API function
-AVSC_INLINE int avs_get_row_size(const AVS_VideoFrame * p) {
-        return avs_get_row_size_p(p, 0); }
-#endif
-
-
-#ifndef AVSC_NO_DECLSPEC
-// this inline function is calling an API function
-AVSC_INLINE int avs_get_height(const AVS_VideoFrame * p) {
-  return avs_get_height_p(p, 0);
-}
-#endif
-
-#ifndef AVSC_NO_DECLSPEC
-// this inline function is calling an API function
-AVSC_INLINE const BYTE* avs_get_read_ptr(const AVS_VideoFrame * p) {
-        return avs_get_read_ptr_p(p, 0);}
-#endif
-
-#ifndef AVSC_NO_DECLSPEC
-// this inline function is calling an API function
+#ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
 AVSC_INLINE BYTE* avs_get_write_ptr(const AVS_VideoFrame * p) {
         return avs_get_write_ptr_p(p, 0);}
 #endif
 
-#ifndef AVSC_NO_DECLSPEC
-// this inline function is calling an API function
-AVSC_INLINE void avs_release_frame(AVS_VideoFrame * f)
-  {avs_release_video_frame(f);}
-#endif
+AVSC_API(void, avs_release_video_frame)(AVS_VideoFrame *);
+// makes a shallow copy of a video frame
+AVSC_API(AVS_VideoFrame *, avs_copy_video_frame)(AVS_VideoFrame *);
 
 #ifndef AVSC_NO_DECLSPEC
-// this inline function is calling an API function
+AVSC_INLINE void avs_release_frame(AVS_VideoFrame * f)
+  {avs_release_video_frame(f);}
 AVSC_INLINE AVS_VideoFrame * avs_copy_frame(AVS_VideoFrame * f)
   {return avs_copy_video_frame(f);}
 #endif
@@ -660,14 +587,14 @@ AVSC_INLINE AVS_VideoFrame * avs_copy_frame(AVS_VideoFrame * f)
 //
 
 // Treat AVS_Value as a fat pointer.  That is use avs_copy_value
-// and avs_release_value appropriately as you would if AVS_Value was
+// and avs_release_value appropiaty as you would if AVS_Value was
 // a pointer.
 
 // To maintain source code compatibility with future versions of the
 // avisynth_c API don't use the AVS_Value directly.  Use the helper
 // functions below.
 
-// AVS_Value is laid out identically to AVSValue
+// AVS_Value is layed out identicly to AVSValue
 typedef struct AVS_Value AVS_Value;
 struct AVS_Value {
   short type;  // 'a'rray, 'c'lip, 'b'ool, 'i'nt, 'f'loat, 's'tring, 'v'oid, or 'l'ong
@@ -683,19 +610,15 @@ struct AVS_Value {
   } d;
 };
 
-// AVS_Value should be initialized with avs_void.
+// AVS_Value should be initilized with avs_void.
 // Should also set to avs_void after the value is released
-// with avs_copy_value.  Consider it the equivalent of setting
+// with avs_copy_value.  Consider it the equalvent of setting
 // a pointer to NULL
 static const AVS_Value avs_void = {'v'};
 
 AVSC_API(void, avs_copy_value)(AVS_Value * dest, AVS_Value src);
 AVSC_API(void, avs_release_value)(AVS_Value);
-AVSC_API(AVS_Clip *, avs_take_clip)(AVS_Value, AVS_ScriptEnvironment *);
-AVSC_API(void, avs_set_to_clip)(AVS_Value *, AVS_Clip *);
 
-
-// no API for these, inline helper functions
 AVSC_INLINE int avs_defined(AVS_Value v) { return v.type != 'v'; }
 AVSC_INLINE int avs_is_clip(AVS_Value v) { return v.type == 'c'; }
 AVSC_INLINE int avs_is_bool(AVS_Value v) { return v.type == 'b'; }
@@ -705,6 +628,9 @@ AVSC_INLINE int avs_is_string(AVS_Value v) { return v.type == 's'; }
 AVSC_INLINE int avs_is_array(AVS_Value v) { return v.type == 'a'; }
 AVSC_INLINE int avs_is_error(AVS_Value v) { return v.type == 'e'; }
 
+AVSC_API(AVS_Clip *, avs_take_clip)(AVS_Value, AVS_ScriptEnvironment *);
+AVSC_API(void, avs_set_to_clip)(AVS_Value *, AVS_Clip *);
+
 AVSC_INLINE int avs_as_bool(AVS_Value v)
         { return v.d.boolean; }
 AVSC_INLINE int avs_as_int(AVS_Value v)
@@ -735,13 +661,11 @@ AVSC_INLINE AVS_Value avs_new_value_float(float v0)
 AVSC_INLINE AVS_Value avs_new_value_error(const char * v0)
         { AVS_Value v; v.type = 'e'; v.d.string = v0; return v; }
 #ifndef AVSC_NO_DECLSPEC
-// this inline function is calling an API function
 AVSC_INLINE AVS_Value avs_new_value_clip(AVS_Clip * v0)
         { AVS_Value v; avs_set_to_clip(&v, v0); return v; }
 #endif
 AVSC_INLINE AVS_Value avs_new_value_array(AVS_Value * v0, int size)
         { AVS_Value v; v.type = 'a'; v.d.array = v0; v.array_size = (short)size; return v; }
-// end of inline helper functions
 
 /////////////////////////////////////////////////////////////////////
 //
@@ -798,7 +722,7 @@ struct AVS_FilterInfo
 
 // Create a new filter
 // fi is set to point to the AVS_FilterInfo so that you can
-//   modify it once it is initialized.
+//   modify it once it is initilized.
 // store_child should generally be set to true.  If it is not
 //    set than ALL methods (the function pointers) must be defined
 // If it is set than you do not need to worry about freeing the child
@@ -829,26 +753,10 @@ enum {
   AVS_CPUF_SSSE3      = 0x200,   //  Core 2
   AVS_CPUF_SSE4       = 0x400,   //  Penryn, Wolfdale, Yorkfield
   AVS_CPUF_SSE4_1     = 0x400,
-  AVS_CPUF_AVX        = 0x800,   //  Sandy Bridge, Bulldozer
+//AVS_CPUF_AVX        = 0x800,   //  Sandy Bridge, Bulldozer
   AVS_CPUF_SSE4_2    = 0x1000,   //  Nehalem
-  // AVS+
-  AVS_CPUF_AVX2      = 0x2000,   //  Haswell
-  AVS_CPUF_FMA3      = 0x4000,
-  AVS_CPUF_F16C      = 0x8000,
-  AVS_CPUF_MOVBE     = 0x10000,   // Big Endian Move
-  AVS_CPUF_POPCNT    = 0x20000,
-  AVS_CPUF_AES       = 0x40000,
-  AVS_CPUF_FMA4      = 0x80000,
-
-  AVS_CPUF_AVX512F    = 0x100000,  // AVX-512 Foundation.
-  AVS_CPUF_AVX512DQ   = 0x200000,  // AVX-512 DQ (Double/Quad granular) Instructions
-  AVS_CPUF_AVX512PF   = 0x400000,  // AVX-512 Prefetch
-  AVS_CPUF_AVX512ER   = 0x800000,  // AVX-512 Exponential and Reciprocal
-  AVS_CPUF_AVX512CD   = 0x1000000, // AVX-512 Conflict Detection
-  AVS_CPUF_AVX512BW   = 0x2000000, // AVX-512 BW (Byte/Word granular) Instructions
-  AVS_CPUF_AVX512VL   = 0x4000000, // AVX-512 VL (128/256 Vector Length) Extensions
-  AVS_CPUF_AVX512IFMA = 0x8000000, // AVX-512 IFMA integer 52 bit
-  AVS_CPUF_AVX512VBMI = 0x10000000 // AVX-512 VBMI
+//AVS_CPUF_AVX2      = 0x2000,   //  Haswell
+//AVS_CPUF_AVX512    = 0x4000,   //  Knights Landing
 };
 
 
@@ -885,23 +793,20 @@ AVSC_API(int, avs_set_global_var)(AVS_ScriptEnvironment *, const char* name, con
 
 AVSC_API(AVS_VideoFrame *, avs_new_video_frame_a)(AVS_ScriptEnvironment *,
                                           const AVS_VideoInfo * vi, int align);
-// align should be at least 16 for classic Avisynth
-// Avisynth+: any value, Avs+ ensures a minimum alignment if too small align is provided
+// align should be at least 16
 
-// no API for these, inline helper functions
 #ifndef AVSC_NO_DECLSPEC
-// this inline function is calling an API function
-AVSC_INLINE AVS_VideoFrame * avs_new_video_frame(AVS_ScriptEnvironment * env,
+AVSC_INLINE
+AVS_VideoFrame * avs_new_video_frame(AVS_ScriptEnvironment * env,
                                      const AVS_VideoInfo * vi)
-  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
+  {return avs_new_video_frame_a(env,vi,FRAME_ALIGN);}
 
-// an older compatibility alias
-// this inline function is calling an API function
-AVSC_INLINE AVS_VideoFrame * avs_new_frame(AVS_ScriptEnvironment * env,
+AVSC_INLINE
+AVS_VideoFrame * avs_new_frame(AVS_ScriptEnvironment * env,
                                const AVS_VideoInfo * vi)
-  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
+  {return avs_new_video_frame_a(env,vi,FRAME_ALIGN);}
 #endif
-// end of inline helper functions
+
 
 AVSC_API(int, avs_make_writable)(AVS_ScriptEnvironment *, AVS_VideoFrame * * pvf);
 
@@ -934,10 +839,7 @@ AVSC_API(AVS_VideoFrame *, avs_subframe_planar)(AVS_ScriptEnvironment *, AVS_Vid
 // The returned video frame must be be released
 
 #ifdef AVSC_NO_DECLSPEC
-// This part uses LoadLibrary and related functions to dynamically load Avisynth instead of declspec(dllimport)
-// When AVSC_NO_DECLSPEC is defined, you can use avs_load_library to populate API functions into a struct
-// AVSC_INLINE functions which call onto an API functions should be treated specially (todo)
-
+// use LoadLibrary and related functions to dynamically load Avisynth instead of declspec(dllimport)
 /*
   The following functions needs to have been declared, probably from windows.h
 
@@ -954,14 +856,6 @@ typedef struct AVS_Library AVS_Library;
 
 #define AVSC_DECLARE_FUNC(name) name##_func name
 
-// AVSC_DECLARE_FUNC helps keeping naming convention: type is xxxxx_func, function name is xxxxx
-// e.g. "AVSC_DECLARE_FUNC(avs_add_function);"
-// is a shortcut for "avs_add_function_func avs_add_function;"
-
-// Note: AVSC_INLINE functions which call into API,
-// are guarded by #ifndef AVSC_NO_DECLSPEC
-// They should call the appropriate library-> API entry
-
 struct AVS_Library {
   HMODULE handle;
 
@@ -1004,30 +898,13 @@ struct AVS_Library {
   AVSC_DECLARE_FUNC(avs_vsprintf);
 
   AVSC_DECLARE_FUNC(avs_get_error);
+  AVSC_DECLARE_FUNC(avs_is_rgb48);
+  AVSC_DECLARE_FUNC(avs_is_rgb64);
   AVSC_DECLARE_FUNC(avs_is_yv24);
   AVSC_DECLARE_FUNC(avs_is_yv16);
   AVSC_DECLARE_FUNC(avs_is_yv12);
   AVSC_DECLARE_FUNC(avs_is_yv411);
   AVSC_DECLARE_FUNC(avs_is_y8);
-  AVSC_DECLARE_FUNC(avs_is_color_space);
-
-  AVSC_DECLARE_FUNC(avs_get_plane_width_subsampling);
-  AVSC_DECLARE_FUNC(avs_get_plane_height_subsampling);
-  AVSC_DECLARE_FUNC(avs_bits_per_pixel);
-  AVSC_DECLARE_FUNC(avs_bytes_from_pixels);
-  AVSC_DECLARE_FUNC(avs_row_size);
-  AVSC_DECLARE_FUNC(avs_bmp_size);
-  AVSC_DECLARE_FUNC(avs_get_pitch_p);
-  AVSC_DECLARE_FUNC(avs_get_row_size_p);
-  AVSC_DECLARE_FUNC(avs_get_height_p);
-  AVSC_DECLARE_FUNC(avs_get_read_ptr_p);
-  AVSC_DECLARE_FUNC(avs_is_writable);
-  AVSC_DECLARE_FUNC(avs_get_write_ptr_p);
-
-  // Avisynth+ specific
-  // Note: these functions are simulated/use fallback to existing functions
-  AVSC_DECLARE_FUNC(avs_is_rgb48);
-  AVSC_DECLARE_FUNC(avs_is_rgb64);
   AVSC_DECLARE_FUNC(avs_is_yuv444p16);
   AVSC_DECLARE_FUNC(avs_is_yuv422p16);
   AVSC_DECLARE_FUNC(avs_is_yuv420p16);
@@ -1043,54 +920,29 @@ struct AVS_Library {
   AVSC_DECLARE_FUNC(avs_is_yuva);
   AVSC_DECLARE_FUNC(avs_is_planar_rgb);
   AVSC_DECLARE_FUNC(avs_is_planar_rgba);
+  AVSC_DECLARE_FUNC(avs_is_color_space);
+
+  AVSC_DECLARE_FUNC(avs_get_plane_width_subsampling);
+  AVSC_DECLARE_FUNC(avs_get_plane_height_subsampling);
+  AVSC_DECLARE_FUNC(avs_bits_per_pixel);
+  AVSC_DECLARE_FUNC(avs_bytes_from_pixels);
+  AVSC_DECLARE_FUNC(avs_row_size);
+  AVSC_DECLARE_FUNC(avs_bmp_size);
+  AVSC_DECLARE_FUNC(avs_get_pitch_p);
+  AVSC_DECLARE_FUNC(avs_get_row_size_p);
+  AVSC_DECLARE_FUNC(avs_get_height_p);
+  AVSC_DECLARE_FUNC(avs_get_read_ptr_p);
+  AVSC_DECLARE_FUNC(avs_is_writable);
+  AVSC_DECLARE_FUNC(avs_get_write_ptr_p);
+
   AVSC_DECLARE_FUNC(avs_num_components);
   AVSC_DECLARE_FUNC(avs_component_size);
   AVSC_DECLARE_FUNC(avs_bits_per_component);
-  // end of Avisynth+ specific
 
 };
 
 #undef AVSC_DECLARE_FUNC
 
-// Helper functions for fallback simulation
-// Avisynth+ extensions do not exist in classic Avisynth so they are simulated
-AVSC_INLINE int avs_is_xx_fallback_return_false(const AVS_VideoInfo * p)
-{
-  return 0;
-}
-
-// Avisynth+ extensions do not exist in classic Avisynth so they are simulated
-AVSC_INLINE int avs_num_components_fallback(const AVS_VideoInfo * p)
-{
-  switch (p->pixel_type) {
-  case AVS_CS_UNKNOWN:
-    return 0;
-  case AVS_CS_RAW32:
-  case AVS_CS_Y8:
-    return 1;
-  case AVS_CS_BGR32:
-    return 4; // not planar but return the count
-  default:
-    return 3;
-  }
-}
-
-// Avisynth+ extensions do not exist in classic Avisynth so they are simulated
-AVSC_INLINE int avs_component_size_fallback(const AVS_VideoInfo * p)
-{
-  return 1;
-}
-
-// Avisynth+ extensions do not exist in classic Avisynth so they are simulated
-AVSC_INLINE int avs_bits_per_component_fallback(const AVS_VideoInfo * p)
-{
-  return 8;
-}
-// End of helper functions for fallback simulation
-
-// avs_load_library() allocates an array for API procedure entries
-// reads and fills the entries with live procedure addresses.
-// AVSC_INLINE helpers which are calling into API procedures are not treated here (todo)
 
 AVSC_INLINE AVS_Library * avs_load_library() {
   AVS_Library *library = (AVS_Library *)malloc(sizeof(AVS_Library));
@@ -1108,55 +960,6 @@ AVSC_INLINE AVS_Library * avs_load_library() {
     goto fail;\
 }
 
-#if 0
-// FFmpeg-specific: we don't use the FALLBACK stuff, and it causes build errors,
-// so ifdef it out on our side.
-
-// When an API function is not loadable, let's try a replacement
-// Missing Avisynth+ functions will be substituted with classic Avisynth compatible methods
-/*
-Avisynth+                 When method is missing (classic Avisynth)
-avs_is_rgb48              constant false
-avs_is_rgb64              constant false
-avs_is_yuv444p16          constant false
-avs_is_yuv422p16          constant false
-avs_is_yuv420p16          constant false
-avs_is_y16                constant false
-avs_is_yuv444ps           constant false
-avs_is_yuv422ps           constant false
-avs_is_yuv420ps           constant false
-avs_is_y32                constant false
-avs_is_444                avs_is_yv24
-avs_is_422                avs_is_yv16
-avs_is_420                avs_is_yv12
-avs_is_y                  avs_is_y8
-avs_is_yuva               constant false
-avs_is_planar_rgb         constant false
-avs_is_planar_rgba        constant false
-avs_num_components        special: avs_num_components_fake Y8:1 RGB32:4 else 3
-avs_component_size        constant 1 (1 bytes/component)
-avs_bits_per_component    constant 8 (8 bits/component)
-*/
-
-  // try to load an alternative function
-#define AVSC_LOAD_FUNC_FALLBACK(name,name2) {\
-  library->name = (name##_func) GetProcAddress(library->handle, AVSC_STRINGIFY(name));\
-  if (library->name == NULL)\
-    library->name = (name##_func) GetProcAddress(library->handle, AVSC_STRINGIFY(name2));\
-  if (library->name == NULL)\
-    goto fail;\
-}
-
-  // try to assign a replacement function
-#define AVSC_LOAD_FUNC_FALLBACK_SIMULATED(name,name2) {\
-  library->name = (name##_func) GetProcAddress(library->handle, AVSC_STRINGIFY(name));\
-  if (library->name == NULL)\
-    library->name = name2;\
-  if (library->name == NULL)\
-    goto fail;\
-}
-#endif
-
   AVSC_LOAD_FUNC(avs_add_function);
   AVSC_LOAD_FUNC(avs_at_exit);
   AVSC_LOAD_FUNC(avs_bit_blt);
@@ -1196,11 +999,28 @@ avs_bits_per_component    constant 8 (8 bits/component)
   AVSC_LOAD_FUNC(avs_vsprintf);
 
   AVSC_LOAD_FUNC(avs_get_error);
+  AVSC_LOAD_FUNC(avs_is_rgb48);
+  AVSC_LOAD_FUNC(avs_is_rgb64);
   AVSC_LOAD_FUNC(avs_is_yv24);
   AVSC_LOAD_FUNC(avs_is_yv16);
   AVSC_LOAD_FUNC(avs_is_yv12);
   AVSC_LOAD_FUNC(avs_is_yv411);
   AVSC_LOAD_FUNC(avs_is_y8);
+  AVSC_LOAD_FUNC(avs_is_yuv444p16);
+  AVSC_LOAD_FUNC(avs_is_yuv422p16);
+  AVSC_LOAD_FUNC(avs_is_yuv420p16);
+  AVSC_LOAD_FUNC(avs_is_y16);
+  AVSC_LOAD_FUNC(avs_is_yuv444ps);
+  AVSC_LOAD_FUNC(avs_is_yuv422ps);
+  AVSC_LOAD_FUNC(avs_is_yuv420ps);
+  AVSC_LOAD_FUNC(avs_is_y32);
+  AVSC_LOAD_FUNC(avs_is_444);
+  AVSC_LOAD_FUNC(avs_is_422);
+  AVSC_LOAD_FUNC(avs_is_420);
+  AVSC_LOAD_FUNC(avs_is_y);
+  AVSC_LOAD_FUNC(avs_is_yuva);
+  AVSC_LOAD_FUNC(avs_is_planar_rgb);
+  AVSC_LOAD_FUNC(avs_is_planar_rgba);
   AVSC_LOAD_FUNC(avs_is_color_space);
 
   AVSC_LOAD_FUNC(avs_get_plane_width_subsampling);
@@ -1216,35 +1036,15 @@ avs_bits_per_component    constant 8 (8 bits/component)
   AVSC_LOAD_FUNC(avs_is_writable);
   AVSC_LOAD_FUNC(avs_get_write_ptr_p);
 
-#if 0
-  // Avisynth+ specific but made them callable for classic Avisynth hosts
-  AVSC_LOAD_FUNC_FALLBACK_SIMULATED(avs_is_rgb48, avs_is_xx_fallback_return_false);
-  AVSC_LOAD_FUNC_FALLBACK_SIMULATED(avs_is_rgb64, avs_is_xx_fallback_return_false);
-  AVSC_LOAD_FUNC_FALLBACK_SIMULATED(avs_is_yuv444p16, avs_is_xx_fallback_return_false);
-  AVSC_LOAD_FUNC_FALLBACK_SIMULATED(avs_is_yuv422p16, avs_is_xx_fallback_return_false);
-  AVSC_LOAD_FUNC_FALLBACK_SIMULATED(avs_is_yuv420p16, avs_is_xx_fallback_return_false);
-  AVSC_LOAD_FUNC_FALLBACK_SIMULATED(avs_is_y16, avs_is_xx_fallback_return_false);
-  AVSC_LOAD_FUNC_FALLBACK_SIMULATED(avs_is_yuv444ps, avs_is_xx_fallback_return_false);
-  AVSC_LOAD_FUNC_FALLBACK_SIMULATED(avs_is_yuv422ps, avs_is_xx_fallback_return_false);
-  AVSC_LOAD_FUNC_FALLBACK_SIMULATED(avs_is_yuv420ps, avs_is_xx_fallback_return_false);
-  AVSC_LOAD_FUNC_FALLBACK_SIMULATED(avs_is_y32, avs_is_xx_fallback_return_false);
-  AVSC_LOAD_FUNC_FALLBACK(avs_is_444, avs_is_yv24);
-  AVSC_LOAD_FUNC_FALLBACK(avs_is_422, avs_is_yv16);
-  AVSC_LOAD_FUNC_FALLBACK(avs_is_420, avs_is_yv12);
-  AVSC_LOAD_FUNC_FALLBACK(avs_is_y, avs_is_y8);
-  AVSC_LOAD_FUNC_FALLBACK_SIMULATED(avs_is_yuva, avs_is_xx_fallback_return_false);
-  AVSC_LOAD_FUNC_FALLBACK_SIMULATED(avs_is_planar_rgb, avs_is_xx_fallback_return_false);
-  AVSC_LOAD_FUNC_FALLBACK_SIMULATED(avs_is_planar_rgba, avs_is_xx_fallback_return_false);
-  AVSC_LOAD_FUNC_FALLBACK_SIMULATED(avs_num_components, avs_num_components_fallback);
-  AVSC_LOAD_FUNC_FALLBACK_SIMULATED(avs_component_size, avs_component_size_fallback);
-  AVSC_LOAD_FUNC_FALLBACK_SIMULATED(avs_bits_per_component, avs_bits_per_component_fallback);
-#endif
+  AVSC_LOAD_FUNC(avs_num_components);
+  AVSC_LOAD_FUNC(avs_component_size);
+  AVSC_LOAD_FUNC(avs_bits_per_component);
+
+
 
 #undef __AVSC_STRINGIFY
 #undef AVSC_STRINGIFY
 #undef AVSC_LOAD_FUNC
-#undef AVSC_LOAD_FUNC_FALLBACK
-#undef AVSC_LOAD_FUNC_FALLBACK_SIMULATED
 
   return library;
 

compat/avisynth/avs/capi.h

@@ -39,49 +39,17 @@
 #  define EXTERN_C
 #endif
 
-#ifdef BUILDING_AVSCORE
-#  if defined(GCC) && defined(X86_32)
-#    define AVSC_CC
-#  else // MSVC builds and 64-bit GCC
-#    ifndef AVSC_USE_STDCALL
-#      define AVSC_CC __cdecl
-#    else
-#      define AVSC_CC __stdcall
-#    endif
-#  endif
-#else // needed for programs that talk to AviSynth+
-#  ifndef AVSC_WIN32_GCC32 // see comment below
-#    ifndef AVSC_USE_STDCALL
-#      define AVSC_CC __cdecl
-#    else
-#      define AVSC_CC __stdcall
-#    endif
-#  else
-#    define AVSC_CC
-#  endif
+#ifndef AVSC_USE_STDCALL
+#  define AVSC_CC __cdecl
+#else
+#  define AVSC_CC __stdcall
 #endif
 
-// On 64-bit Windows, there's only one calling convention,
-// so there is no difference between MSVC and GCC. On 32-bit,
-// this isn't true. The convention that GCC needs to use to
-// even build AviSynth+ as 32-bit makes anything that uses
-// it incompatible with 32-bit MSVC builds of AviSynth+.
-// The AVSC_WIN32_GCC32 define is meant to provide a user
-// switchable way to make builds of FFmpeg to test 32-bit
-// GCC builds of AviSynth+ without having to screw around
-// with alternate headers, while still default to the usual
-// situation of using 32-bit MSVC builds of AviSynth+.
-
-// Hopefully, this situation will eventually be resolved
-// and a broadly compatible solution will arise so the
-// same 32-bit FFmpeg build can handle either MSVC or GCC
-// builds of AviSynth+.
-
 #define AVSC_INLINE static __inline
 
 #ifdef BUILDING_AVSCORE
-#  define AVSC_EXPORT __declspec(dllexport)
-#  define AVSC_API(ret, name) EXTERN_C AVSC_EXPORT ret AVSC_CC name
+#  define AVSC_EXPORT EXTERN_C
+#  define AVSC_API(ret, name) EXTERN_C __declspec(dllexport) ret AVSC_CC name
 #else
 #  define AVSC_EXPORT EXTERN_C __declspec(dllexport)
 #  ifndef AVSC_NO_DECLSPEC

compat/avisynth/avs/config.h

@@ -42,7 +42,7 @@
 // alignment. They should always request the exact alignment value they need.
 // This is to make sure that plugins work over the widest range of AviSynth
 // builds possible.
-#define FRAME_ALIGN 64
+#define FRAME_ALIGN 32
 
 #if   defined(_M_AMD64) || defined(__x86_64)
 #   define X86_64
@@ -52,19 +52,4 @@
 #   error Unsupported CPU architecture.
 #endif
 
-#if   defined(_MSC_VER)
-#   define MSVC
-#elif defined(__GNUC__)
-#   define GCC
-#elif defined(__clang__)
-#   define CLANG
-#else
-#   error Unsupported compiler.
-#endif
-
-#if   defined(GCC)
-#   undef __forceinline
-#   define __forceinline inline
-#endif
-
 #endif //AVS_CONFIG_H

compat/avisynth/avs/types.h

@@ -35,12 +35,6 @@
 
 // Define all types necessary for interfacing with avisynth.dll
 
-#ifdef __cplusplus
-  #include <cstddef>
-#else
-  #include <stddef.h>
-#endif
-
 // Raster types used by VirtualDub & Avisynth
 typedef unsigned int    Pixel32;
 typedef unsigned char   BYTE;

compat/cuda/cuda_runtime.h

@@ -1,131 +0,0 @@
-/*
- * Minimum CUDA compatibility definitions header
- *
- * Copyright (c) 2019 Rodger Combs
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#ifndef COMPAT_CUDA_CUDA_RUNTIME_H
-#define COMPAT_CUDA_CUDA_RUNTIME_H
-
-// Common macros
-#define __global__ __attribute__((global))
-#define __device__ __attribute__((device))
-#define __device_builtin__ __attribute__((device_builtin))
-#define __align__(N) __attribute__((aligned(N)))
-#define __inline__ __inline__ __attribute__((always_inline))
-
-#define max(a, b) ((a) > (b) ? (a) : (b))
-#define min(a, b) ((a) < (b) ? (a) : (b))
-#define abs(x) ((x) < 0 ? -(x) : (x))
-
-#define atomicAdd(a, b) (__atomic_fetch_add(a, b, __ATOMIC_SEQ_CST))
-
-// Basic typedefs
-typedef __device_builtin__ unsigned long long cudaTextureObject_t;
-
-typedef struct __device_builtin__ __align__(2) uchar2
-{
-    unsigned char x, y;
-} uchar2;
-
-typedef struct __device_builtin__ __align__(4) ushort2
-{
-    unsigned short x, y;
-} ushort2;
-
-typedef struct __device_builtin__ uint3
-{
-    unsigned int x, y, z;
-} uint3;
-
-typedef struct uint3 dim3;
-
-typedef struct __device_builtin__ __align__(8) int2
-{
-    int x, y;
-} int2;
-
-typedef struct __device_builtin__ __align__(4) uchar4
-{
-    unsigned char x, y, z, w;
-} uchar4;
-
-typedef struct __device_builtin__ __align__(8) ushort4
-{
-    unsigned char x, y, z, w;
-} ushort4;
-
-typedef struct __device_builtin__ __align__(16) int4
-{
-    int x, y, z, w;
-} int4;
-
-// Accessors for special registers
-#define GETCOMP(reg, comp) \
-    asm("mov.u32 %0, %%" #reg "." #comp ";" : "=r"(tmp)); \
-    ret.comp = tmp;
-
-#define GET(name, reg) static inline __device__ uint3 name() {\
-    uint3 ret; \
-    unsigned tmp; \
-    GETCOMP(reg, x) \
-    GETCOMP(reg, y) \
-    GETCOMP(reg, z) \
-    return ret; \
-}
-
-GET(getBlockIdx, ctaid)
-GET(getBlockDim, ntid)
-GET(getThreadIdx, tid)
-
-// Instead of externs for these registers, we turn access to them into calls into trivial ASM
-#define blockIdx (getBlockIdx())
-#define blockDim (getBlockDim())
-#define threadIdx (getThreadIdx())
-
-// Basic initializers (simple macros rather than inline functions)
-#define make_uchar2(a, b) ((uchar2){.x = a, .y = b})
-#define make_ushort2(a, b) ((ushort2){.x = a, .y = b})
-#define make_uchar4(a, b, c, d) ((uchar4){.x = a, .y = b, .z = c, .w = d})
-#define make_ushort4(a, b, c, d) ((ushort4){.x = a, .y = b, .z = c, .w = d})
-
-// Conversions from the tex instruction's 4-register output to various types
-#define TEX2D(type, ret) static inline __device__ void conv(type* out, unsigned a, unsigned b, unsigned c, unsigned d) {*out = (ret);}
-
-TEX2D(unsigned char, a & 0xFF)
-TEX2D(unsigned short, a & 0xFFFF)
-TEX2D(uchar2, make_uchar2(a & 0xFF, b & 0xFF))
-TEX2D(ushort2, make_ushort2(a & 0xFFFF, b & 0xFFFF))
-TEX2D(uchar4, make_uchar4(a & 0xFF, b & 0xFF, c & 0xFF, d & 0xFF))
-TEX2D(ushort4, make_ushort4(a & 0xFFFF, b & 0xFFFF, c & 0xFFFF, d & 0xFFFF))
-
-// Template calling tex instruction and converting the output to the selected type
-template <class T>
-static inline __device__ T tex2D(cudaTextureObject_t texObject, float x, float y)
-{
-  T ret;
-  unsigned ret1, ret2, ret3, ret4;
-  asm("tex.2d.v4.u32.f32 {%0, %1, %2, %3}, [%4, {%5, %6}];" :
-      "=r"(ret1), "=r"(ret2), "=r"(ret3), "=r"(ret4) :
-      "l"(texObject), "f"(x), "f"(y));
-  conv(&ret, ret1, ret2, ret3, ret4);
-  return ret;
-}
-
-#endif /* COMPAT_CUDA_CUDA_RUNTIME_H */

compat/cuda/dynlink_cuda.h

@@ -0,0 +1,98 @@
+/*
+ * This copyright notice applies to this header file only:
+ *
+ * Copyright (c) 2016
+ *
+ * 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.
+ */
+
+#if !defined(AV_COMPAT_DYNLINK_CUDA_H) && !defined(CUDA_VERSION)
+#define AV_COMPAT_DYNLINK_CUDA_H
+
+#include <stddef.h>
+
+#define CUDA_VERSION 7050
+
+#if defined(_WIN32) || defined(__CYGWIN__)
+#define CUDAAPI __stdcall
+#else
+#define CUDAAPI
+#endif
+
+#define CU_CTX_SCHED_BLOCKING_SYNC 4
+
+typedef int CUdevice;
+typedef void* CUarray;
+typedef void* CUcontext;
+typedef void* CUstream;
+#if defined(__x86_64) || defined(AMD64) || defined(_M_AMD64)
+typedef unsigned long long CUdeviceptr;
+#else
+typedef unsigned int CUdeviceptr;
+#endif
+
+typedef enum cudaError_enum {
+    CUDA_SUCCESS = 0
+} CUresult;
+
+typedef enum CUmemorytype_enum {
+    CU_MEMORYTYPE_HOST = 1,
+    CU_MEMORYTYPE_DEVICE = 2
+} CUmemorytype;
+
+typedef struct CUDA_MEMCPY2D_st {
+    size_t srcXInBytes;
+    size_t srcY;
+    CUmemorytype srcMemoryType;
+    const void *srcHost;
+    CUdeviceptr srcDevice;
+    CUarray srcArray;
+    size_t srcPitch;
+
+    size_t dstXInBytes;
+    size_t dstY;
+    CUmemorytype dstMemoryType;
+    void *dstHost;
+    CUdeviceptr dstDevice;
+    CUarray dstArray;
+    size_t dstPitch;
+
+    size_t WidthInBytes;
+    size_t Height;
+} CUDA_MEMCPY2D;
+
+typedef CUresult CUDAAPI tcuInit(unsigned int Flags);
+typedef CUresult CUDAAPI tcuDeviceGetCount(int *count);
+typedef CUresult CUDAAPI tcuDeviceGet(CUdevice *device, int ordinal);
+typedef CUresult CUDAAPI tcuDeviceGetName(char *name, int len, CUdevice dev);
+typedef CUresult CUDAAPI tcuDeviceComputeCapability(int *major, int *minor, CUdevice dev);
+typedef CUresult CUDAAPI tcuCtxCreate_v2(CUcontext *pctx, unsigned int flags, CUdevice dev);
+typedef CUresult CUDAAPI tcuCtxPushCurrent_v2(CUcontext *pctx);
+typedef CUresult CUDAAPI tcuCtxPopCurrent_v2(CUcontext *pctx);
+typedef CUresult CUDAAPI tcuCtxDestroy_v2(CUcontext ctx);
+typedef CUresult CUDAAPI tcuMemAlloc_v2(CUdeviceptr *dptr, size_t bytesize);
+typedef CUresult CUDAAPI tcuMemFree_v2(CUdeviceptr dptr);
+typedef CUresult CUDAAPI tcuMemcpy2D_v2(const CUDA_MEMCPY2D *pcopy);
+typedef CUresult CUDAAPI tcuGetErrorName(CUresult error, const char** pstr);
+typedef CUresult CUDAAPI tcuGetErrorString(CUresult error, const char** pstr);
+
+#endif

compat/cuda/dynlink_cuviddec.h

@@ -0,0 +1,886 @@
+/*
+ * This copyright notice applies to this header file only:
+ *
+ * Copyright (c) 2010-2017 NVIDIA Corporation
+ *
+ * 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 cuviddec.h
+//! NVDECODE API provides video decoding interface to NVIDIA GPU devices.
+//! \date 2015-2017
+//! This file contains constants, structure definitions and function prototypes used for decoding.
+/*****************************************************************************************************/
+
+#if !defined(__CUDA_VIDEO_H__)
+#define __CUDA_VIDEO_H__
+
+#if defined(_WIN64) || defined(__LP64__) || defined(__x86_64) || defined(AMD64) || defined(_M_AMD64)
+#if (CUDA_VERSION >= 3020) && (!defined(CUDA_FORCE_API_VERSION) || (CUDA_FORCE_API_VERSION >= 3020))
+#define __CUVID_DEVPTR64
+#endif
+#endif
+
+#if defined(__cplusplus)
+extern "C" {
+#endif /* __cplusplus */
+
+#if defined(__CYGWIN__)
+typedef unsigned int tcu_ulong;
+#else
+typedef unsigned long tcu_ulong;
+#endif
+
+typedef void *CUvideodecoder;
+typedef struct _CUcontextlock_st *CUvideoctxlock;
+
+/*********************************************************************************/
+//! \enum cudaVideoCodec
+//! Video codec enums
+//! These enums are used in CUVIDDECODECREATEINFO and CUVIDDECODECAPS structures
+/*********************************************************************************/
+typedef enum cudaVideoCodec_enum {
+    cudaVideoCodec_MPEG1=0,                                         /**<  MPEG1             */
+    cudaVideoCodec_MPEG2,                                           /**<  MPEG2             */
+    cudaVideoCodec_MPEG4,                                           /**<  MPEG4             */
+    cudaVideoCodec_VC1,                                             /**<  VC1               */
+    cudaVideoCodec_H264,                                            /**<  H264              */
+    cudaVideoCodec_JPEG,                                            /**<  JPEG              */
+    cudaVideoCodec_H264_SVC,                                        /**<  H264-SVC          */
+    cudaVideoCodec_H264_MVC,                                        /**<  H264-MVC          */
+    cudaVideoCodec_HEVC,                                            /**<  HEVC              */
+    cudaVideoCodec_VP8,                                             /**<  VP8               */
+    cudaVideoCodec_VP9,                                             /**<  VP9               */
+    cudaVideoCodec_NumCodecs,                                       /**<  Max codecs        */
+    // Uncompressed YUV
+    cudaVideoCodec_YUV420 = (('I'<<24)|('Y'<<16)|('U'<<8)|('V')),   /**< Y,U,V (4:2:0)      */
+    cudaVideoCodec_YV12   = (('Y'<<24)|('V'<<16)|('1'<<8)|('2')),   /**< Y,V,U (4:2:0)      */
+    cudaVideoCodec_NV12   = (('N'<<24)|('V'<<16)|('1'<<8)|('2')),   /**< Y,UV  (4:2:0)      */
+    cudaVideoCodec_YUYV   = (('Y'<<24)|('U'<<16)|('Y'<<8)|('V')),   /**< YUYV/YUY2 (4:2:2)  */
+    cudaVideoCodec_UYVY   = (('U'<<24)|('Y'<<16)|('V'<<8)|('Y'))    /**< UYVY (4:2:2)       */
+} cudaVideoCodec;
+
+/*********************************************************************************/
+//! \enum cudaVideoSurfaceFormat
+//! Video surface format enums used for output format of decoded output
+//! These enums are used in CUVIDDECODECREATEINFO structure
+/*********************************************************************************/
+typedef enum cudaVideoSurfaceFormat_enum {
+    cudaVideoSurfaceFormat_NV12=0,       /**< NV12 format          */
+    cudaVideoSurfaceFormat_P016=1        /**< 16 bit semiplaner format. Can be used for 10 bit(6LSB bits 0),
+                                              12 bit (4LSB bits 0) */
+} cudaVideoSurfaceFormat;
+
+/******************************************************************************************************************/
+//! \enum cudaVideoDeinterlaceMode
+//! Deinterlacing mode enums
+//! These enums are used in CUVIDDECODECREATEINFO structure
+//! Use cudaVideoDeinterlaceMode_Weave for progressive content and for content that doesn't need deinterlacing
+//! cudaVideoDeinterlaceMode_Adaptive needs more video memory than other DImodes
+/******************************************************************************************************************/
+typedef enum cudaVideoDeinterlaceMode_enum {
+    cudaVideoDeinterlaceMode_Weave=0,   /**< Weave both fields (no deinterlacing) */
+    cudaVideoDeinterlaceMode_Bob,       /**< Drop one field                       */
+    cudaVideoDeinterlaceMode_Adaptive   /**< Adaptive deinterlacing               */
+} cudaVideoDeinterlaceMode;
+
+/**************************************************************************************************************/
+//! \enum cudaVideoChromaFormat
+//! Chroma format enums
+//! These enums are used in CUVIDDECODECREATEINFO and CUVIDDECODECAPS structures
+//! JPEG supports Monochrome, YUV 4:2:0, YUV 4:2:2 and YUV 4:4:4 chroma formats.
+//! H264, HEVC, VP9, VP8, VC1, MPEG1, MPEG2 and MPEG4 support YUV 4:2:0 chroma format only.
+/**************************************************************************************************************/
+typedef enum cudaVideoChromaFormat_enum {
+    cudaVideoChromaFormat_Monochrome=0,  /**< MonoChrome */
+    cudaVideoChromaFormat_420,           /**< YUV 4:2:0  */
+    cudaVideoChromaFormat_422,           /**< YUV 4:2:2  */
+    cudaVideoChromaFormat_444            /**< YUV 4:4:4  */
+} cudaVideoChromaFormat;
+
+/*************************************************************************************************************/
+//! \enum cudaVideoCreateFlags
+//! Decoder flag enums to select preferred decode path
+//! cudaVideoCreate_Default and cudaVideoCreate_PreferCUVID are most optimized, use these whenever possible
+/*************************************************************************************************************/
+typedef enum cudaVideoCreateFlags_enum {
+    cudaVideoCreate_Default     = 0x00,     /**< Default operation mode: use dedicated video engines                        */
+    cudaVideoCreate_PreferCUDA  = 0x01,     /**< Use CUDA-based decoder (requires valid vidLock object for multi-threading) */
+    cudaVideoCreate_PreferDXVA  = 0x02,     /**< Go through DXVA internally if possible (requires D3D9 interop)             */
+    cudaVideoCreate_PreferCUVID = 0x04      /**< Use dedicated video engines directly                                       */
+} cudaVideoCreateFlags;
+
+
+/**************************************************************************************************************/
+//! \struct CUVIDDECODECAPS;
+//! This structure is used in cuvidGetDecoderCaps API
+/**************************************************************************************************************/
+typedef struct _CUVIDDECODECAPS
+{
+    cudaVideoCodec          eCodecType;                 /**< IN: cudaVideoCodec_XXX                                 */
+    cudaVideoChromaFormat   eChromaFormat;              /**< IN: cudaVideoChromaFormat_XXX                          */
+    unsigned int            nBitDepthMinus8;            /**< IN: The Value "BitDepth minus 8"                       */
+    unsigned int            reserved1[3];               /**< Reserved for future use - set to zero                  */
+
+    unsigned char           bIsSupported;               /**< OUT: 1 if codec supported, 0 if not supported          */
+    unsigned char           reserved2[3];               /**< Reserved for future use - set to zero                  */
+    unsigned int            nMaxWidth;                  /**< OUT: Max supported coded width in pixels               */
+    unsigned int            nMaxHeight;                 /**< OUT: Max supported coded height in pixels              */
+    unsigned int            nMaxMBCount;                /**< OUT: Max supported macroblock count
+                                                                  CodedWidth*CodedHeight/256 must be <= nMaxMBCount */
+    unsigned short          nMinWidth;                  /**< OUT: Min supported coded width in pixels               */
+    unsigned short          nMinHeight;                 /**< OUT: Min supported coded height in pixels              */
+    unsigned int            reserved3[11];              /**< Reserved for future use - set to zero                  */
+} CUVIDDECODECAPS;
+
+/**************************************************************************************************************/
+//! \struct CUVIDDECODECREATEINFO
+//! This structure is used in cuvidCreateDecoder API
+/**************************************************************************************************************/
+typedef struct _CUVIDDECODECREATEINFO
+{
+    tcu_ulong ulWidth;                  /**< IN: Coded sequence width in pixels                                             */
+    tcu_ulong ulHeight;                 /**< IN: Coded sequence height in pixels                                            */
+    tcu_ulong ulNumDecodeSurfaces;      /**< IN: Maximum number of internal decode surfaces                                 */
+    cudaVideoCodec CodecType;           /**< IN: cudaVideoCodec_XXX                                                         */
+    cudaVideoChromaFormat ChromaFormat; /**< IN: cudaVideoChromaFormat_XXX                                                  */
+    tcu_ulong ulCreationFlags;          /**< IN: Decoder creation flags (cudaVideoCreateFlags_XXX)                          */
+    tcu_ulong bitDepthMinus8;           /**< IN: The value "BitDepth minus 8"                                               */
+    tcu_ulong ulIntraDecodeOnly;        /**< IN: Set 1 only if video has all intra frames (default value is 0). This will
+                                             optimize video memory for Intra frames only decoding. The support is limited
+                                             to specific codecs(H264 rightnow), the flag will be ignored for codecs which
+                                             are not supported. However decoding might fail if the flag is enabled in case
+                                             of supported codecs for regular bit streams having P and/or B frames.          */
+    tcu_ulong Reserved1[3];             /**< Reserved for future use - set to zero                                          */
+    /**
+    * IN: area of the frame that should be displayed
+    */
+    struct {
+        short left;
+        short top;
+        short right;
+        short bottom;
+    } display_area;
+
+    cudaVideoSurfaceFormat OutputFormat;       /**< IN: cudaVideoSurfaceFormat_XXX                                     */
+    cudaVideoDeinterlaceMode DeinterlaceMode;  /**< IN: cudaVideoDeinterlaceMode_XXX                                   */
+    tcu_ulong ulTargetWidth;                   /**< IN: Post-processed output width (Should be aligned to 2)           */
+    tcu_ulong ulTargetHeight;                  /**< IN: Post-processed output height (Should be aligbed to 2)          */
+    tcu_ulong ulNumOutputSurfaces;             /**< IN: Maximum number of output surfaces simultaneously mapped        */
+    CUvideoctxlock vidLock;                    /**< IN: If non-NULL, context lock used for synchronizing ownership of
+                                                    the cuda context. Needed for cudaVideoCreate_PreferCUDA decode     */
+    /**
+    * IN: target rectangle in the output frame (for aspect ratio conversion)
+    * if a null rectangle is specified, {0,0,ulTargetWidth,ulTargetHeight} will be used
+    */
+    struct {
+        short left;
+        short top;
+        short right;
+        short bottom;
+    } target_rect;
+    tcu_ulong Reserved2[5];                    /**< Reserved for future use - set to zero */
+} CUVIDDECODECREATEINFO;
+
+/*********************************************************/
+//! \struct CUVIDH264DPBENTRY
+//! H.264 DPB entry
+//! This structure is used in CUVIDH264PICPARAMS structure
+/*********************************************************/
+typedef struct _CUVIDH264DPBENTRY
+{
+    int PicIdx;                 /**< picture index of reference frame                                        */
+    int FrameIdx;               /**< frame_num(short-term) or LongTermFrameIdx(long-term)                    */
+    int is_long_term;           /**< 0=short term reference, 1=long term reference                           */
+    int not_existing;           /**< non-existing reference frame (corresponding PicIdx should be set to -1) */
+    int used_for_reference;     /**< 0=unused, 1=top_field, 2=bottom_field, 3=both_fields                    */
+    int FieldOrderCnt[2];       /**< field order count of top and bottom fields                              */
+} CUVIDH264DPBENTRY;
+
+/************************************************************/
+//! \struct CUVIDH264MVCEXT
+//! H.264 MVC picture parameters ext
+//! This structure is used in CUVIDH264PICPARAMS structure
+/************************************************************/
+typedef struct _CUVIDH264MVCEXT
+{
+    int num_views_minus1;                  /**< Max number of coded views minus 1 in video : Range - 0 to 1023              */
+    int view_id;                           /**< view identifier                                                             */
+    unsigned char inter_view_flag;         /**< 1 if used for inter-view prediction, 0 if not                               */
+    unsigned char num_inter_view_refs_l0;  /**< number of inter-view ref pics in RefPicList0                                */
+    unsigned char num_inter_view_refs_l1;  /**< number of inter-view ref pics in RefPicList1                                */
+    unsigned char MVCReserved8Bits;        /**< Reserved bits                                                               */
+    int InterViewRefsL0[16];               /**< view id of the i-th view component for inter-view prediction in RefPicList0 */
+    int InterViewRefsL1[16];               /**< view id of the i-th view component for inter-view prediction in RefPicList1 */
+} CUVIDH264MVCEXT;
+
+/*********************************************************/
+//! \struct CUVIDH264SVCEXT
+//! H.264 SVC picture parameters ext
+//! This structure is used in CUVIDH264PICPARAMS structure
+/*********************************************************/
+typedef struct _CUVIDH264SVCEXT
+{
+    unsigned char profile_idc;
+    unsigned char level_idc;
+    unsigned char DQId;
+    unsigned char DQIdMax;
+    unsigned char disable_inter_layer_deblocking_filter_idc;
+    unsigned char ref_layer_chroma_phase_y_plus1;
+    signed char   inter_layer_slice_alpha_c0_offset_div2;
+    signed char   inter_layer_slice_beta_offset_div2;
+
+    unsigned short DPBEntryValidFlag;
+    unsigned char inter_layer_deblocking_filter_control_present_flag;
+    unsigned char extended_spatial_scalability_idc;
+    unsigned char adaptive_tcoeff_level_prediction_flag;
+    unsigned char slice_header_restriction_flag;
+    unsigned char chroma_phase_x_plus1_flag;
+    unsigned char chroma_phase_y_plus1;
+
+    unsigned char tcoeff_level_prediction_flag;
+    unsigned char constrained_intra_resampling_flag;
+    unsigned char ref_layer_chroma_phase_x_plus1_flag;
+    unsigned char store_ref_base_pic_flag;
+    unsigned char Reserved8BitsA;
+    unsigned char Reserved8BitsB;
+
+    short scaled_ref_layer_left_offset;
+    short scaled_ref_layer_top_offset;
+    short scaled_ref_layer_right_offset;
+    short scaled_ref_layer_bottom_offset;
+    unsigned short Reserved16Bits;
+    struct _CUVIDPICPARAMS *pNextLayer; /**< Points to the picparams for the next layer to be decoded.
+                                             Linked list ends at the target layer. */
+    int bRefBaseLayer;                  /**< whether to store ref base pic */
+} CUVIDH264SVCEXT;
+
+/******************************************************/
+//! \struct CUVIDH264PICPARAMS
+//! H.264 picture parameters
+//! This structure is used in CUVIDPICPARAMS structure
+/******************************************************/
+typedef struct _CUVIDH264PICPARAMS
+{
+    // SPS
+    int log2_max_frame_num_minus4;
+    int pic_order_cnt_type;
+    int log2_max_pic_order_cnt_lsb_minus4;
+    int delta_pic_order_always_zero_flag;
+    int frame_mbs_only_flag;
+    int direct_8x8_inference_flag;
+    int num_ref_frames;             // NOTE: shall meet level 4.1 restrictions
+    unsigned char residual_colour_transform_flag;
+    unsigned char bit_depth_luma_minus8;    // Must be 0 (only 8-bit supported)
+    unsigned char bit_depth_chroma_minus8;  // Must be 0 (only 8-bit supported)
+    unsigned char qpprime_y_zero_transform_bypass_flag;
+    // PPS
+    int entropy_coding_mode_flag;
+    int pic_order_present_flag;
+    int num_ref_idx_l0_active_minus1;
+    int num_ref_idx_l1_active_minus1;
+    int weighted_pred_flag;
+    int weighted_bipred_idc;
+    int pic_init_qp_minus26;
+    int deblocking_filter_control_present_flag;
+    int redundant_pic_cnt_present_flag;
+    int transform_8x8_mode_flag;
+    int MbaffFrameFlag;
+    int constrained_intra_pred_flag;
+    int chroma_qp_index_offset;
+    int second_chroma_qp_index_offset;
+    int ref_pic_flag;
+    int frame_num;
+    int CurrFieldOrderCnt[2];
+    // DPB
+    CUVIDH264DPBENTRY dpb[16];          // List of reference frames within the DPB
+    // Quantization Matrices (raster-order)
+    unsigned char WeightScale4x4[6][16];
+    unsigned char WeightScale8x8[2][64];
+    // FMO/ASO
+    unsigned char fmo_aso_enable;
+    unsigned char num_slice_groups_minus1;
+    unsigned char slice_group_map_type;
+    signed char pic_init_qs_minus26;
+    unsigned int slice_group_change_rate_minus1;
+    union
+    {
+        unsigned long long slice_group_map_addr;
+        const unsigned char *pMb2SliceGroupMap;
+    } fmo;
+    unsigned int Reserved[12];
+    // SVC/MVC
+    union
+    {
+        CUVIDH264MVCEXT mvcext;
+        CUVIDH264SVCEXT svcext;
+    };
+} CUVIDH264PICPARAMS;
+
+
+/********************************************************/
+//! \struct CUVIDMPEG2PICPARAMS
+//! MPEG-2 picture parameters
+//! This structure is used in CUVIDPICPARAMS structure
+/********************************************************/
+typedef struct _CUVIDMPEG2PICPARAMS
+{
+    int ForwardRefIdx;          // Picture index of forward reference (P/B-frames)
+    int BackwardRefIdx;         // Picture index of backward reference (B-frames)
+    int picture_coding_type;
+    int full_pel_forward_vector;
+    int full_pel_backward_vector;
+    int f_code[2][2];
+    int intra_dc_precision;
+    int frame_pred_frame_dct;
+    int concealment_motion_vectors;
+    int q_scale_type;
+    int intra_vlc_format;
+    int alternate_scan;
+    int top_field_first;
+    // Quantization matrices (raster order)
+    unsigned char QuantMatrixIntra[64];
+    unsigned char QuantMatrixInter[64];
+} CUVIDMPEG2PICPARAMS;
+
+// MPEG-4 has VOP types instead of Picture types
+#define I_VOP 0
+#define P_VOP 1
+#define B_VOP 2
+#define S_VOP 3
+
+/*******************************************************/
+//! \struct CUVIDMPEG4PICPARAMS
+//! MPEG-4 picture parameters
+//! This structure is used in CUVIDPICPARAMS structure
+/*******************************************************/
+typedef struct _CUVIDMPEG4PICPARAMS
+{
+    int ForwardRefIdx;          // Picture index of forward reference (P/B-frames)
+    int BackwardRefIdx;         // Picture index of backward reference (B-frames)
+    // VOL
+    int video_object_layer_width;
+    int video_object_layer_height;
+    int vop_time_increment_bitcount;
+    int top_field_first;
+    int resync_marker_disable;
+    int quant_type;
+    int quarter_sample;
+    int short_video_header;
+    int divx_flags;
+    // VOP
+    int vop_coding_type;
+    int vop_coded;
+    int vop_rounding_type;
+    int alternate_vertical_scan_flag;
+    int interlaced;
+    int vop_fcode_forward;
+    int vop_fcode_backward;
+    int trd[2];
+    int trb[2];
+    // Quantization matrices (raster order)
+    unsigned char QuantMatrixIntra[64];
+    unsigned char QuantMatrixInter[64];
+    int gmc_enabled;
+} CUVIDMPEG4PICPARAMS;
+
+/********************************************************/
+//! \struct CUVIDVC1PICPARAMS
+//! VC1 picture parameters
+//! This structure is used in CUVIDPICPARAMS structure
+/********************************************************/
+typedef struct _CUVIDVC1PICPARAMS
+{
+    int ForwardRefIdx;      /**< Picture index of forward reference (P/B-frames) */
+    int BackwardRefIdx;     /**< Picture index of backward reference (B-frames)  */
+    int FrameWidth;         /**< Actual frame width                              */
+    int FrameHeight;        /**< Actual frame height                             */
+    // PICTURE
+    int intra_pic_flag;     /**< Set to 1 for I,BI frames */
+    int ref_pic_flag;       /**< Set to 1 for I,P frames  */
+    int progressive_fcm;    /**< Progressive frame        */
+    // SEQUENCE
+    int profile;
+    int postprocflag;
+    int pulldown;
+    int interlace;
+    int tfcntrflag;
+    int finterpflag;
+    int psf;
+    int multires;
+    int syncmarker;
+    int rangered;
+    int maxbframes;
+    // ENTRYPOINT
+    int panscan_flag;
+    int refdist_flag;
+    int extended_mv;
+    int dquant;
+    int vstransform;
+    int loopfilter;
+    int fastuvmc;
+    int overlap;
+    int quantizer;
+    int extended_dmv;
+    int range_mapy_flag;
+    int range_mapy;
+    int range_mapuv_flag;
+    int range_mapuv;
+    int rangeredfrm;    // range reduction state
+} CUVIDVC1PICPARAMS;
+
+/***********************************************************/
+//! \struct CUVIDJPEGPICPARAMS
+//! JPEG picture parameters
+//! This structure is used in CUVIDPICPARAMS structure
+/***********************************************************/
+typedef struct _CUVIDJPEGPICPARAMS
+{
+    int Reserved;
+} CUVIDJPEGPICPARAMS;
+
+
+/*******************************************************/
+//! \struct CUVIDHEVCPICPARAMS
+//! HEVC picture parameters
+//! This structure is used in CUVIDPICPARAMS structure
+/*******************************************************/
+typedef struct _CUVIDHEVCPICPARAMS
+{
+    // sps
+    int pic_width_in_luma_samples;
+    int pic_height_in_luma_samples;
+    unsigned char log2_min_luma_coding_block_size_minus3;
+    unsigned char log2_diff_max_min_luma_coding_block_size;
+    unsigned char log2_min_transform_block_size_minus2;
+    unsigned char log2_diff_max_min_transform_block_size;
+    unsigned char pcm_enabled_flag;
+    unsigned char log2_min_pcm_luma_coding_block_size_minus3;
+    unsigned char log2_diff_max_min_pcm_luma_coding_block_size;
+    unsigned char pcm_sample_bit_depth_luma_minus1;
+
+    unsigned char pcm_sample_bit_depth_chroma_minus1;
+    unsigned char pcm_loop_filter_disabled_flag;
+    unsigned char strong_intra_smoothing_enabled_flag;
+    unsigned char max_transform_hierarchy_depth_intra;
+    unsigned char max_transform_hierarchy_depth_inter;
+    unsigned char amp_enabled_flag;
+    unsigned char separate_colour_plane_flag;
+    unsigned char log2_max_pic_order_cnt_lsb_minus4;
+
+    unsigned char num_short_term_ref_pic_sets;
+    unsigned char long_term_ref_pics_present_flag;
+    unsigned char num_long_term_ref_pics_sps;
+    unsigned char sps_temporal_mvp_enabled_flag;
+    unsigned char sample_adaptive_offset_enabled_flag;
+    unsigned char scaling_list_enable_flag;
+    unsigned char IrapPicFlag;
+    unsigned char IdrPicFlag;
+
+    unsigned char bit_depth_luma_minus8;
+    unsigned char bit_depth_chroma_minus8;
+    unsigned char reserved1[14];
+
+    // pps
+    unsigned char dependent_slice_segments_enabled_flag;
+    unsigned char slice_segment_header_extension_present_flag;
+    unsigned char sign_data_hiding_enabled_flag;
+    unsigned char cu_qp_delta_enabled_flag;
+    unsigned char diff_cu_qp_delta_depth;
+    signed char init_qp_minus26;
+    signed char pps_cb_qp_offset;
+    signed char pps_cr_qp_offset;
+
+    unsigned char constrained_intra_pred_flag;
+    unsigned char weighted_pred_flag;
+    unsigned char weighted_bipred_flag;
+    unsigned char transform_skip_enabled_flag;
+    unsigned char transquant_bypass_enabled_flag;
+    unsigned char entropy_coding_sync_enabled_flag;
+    unsigned char log2_parallel_merge_level_minus2;
+    unsigned char num_extra_slice_header_bits;
+
+    unsigned char loop_filter_across_tiles_enabled_flag;
+    unsigned char loop_filter_across_slices_enabled_flag;
+    unsigned char output_flag_present_flag;
+    unsigned char num_ref_idx_l0_default_active_minus1;
+    unsigned char num_ref_idx_l1_default_active_minus1;
+    unsigned char lists_modification_present_flag;
+    unsigned char cabac_init_present_flag;
+    unsigned char pps_slice_chroma_qp_offsets_present_flag;
+
+    unsigned char deblocking_filter_override_enabled_flag;
+    unsigned char pps_deblocking_filter_disabled_flag;
+    signed char   pps_beta_offset_div2;
+    signed char   pps_tc_offset_div2;
+    unsigned char tiles_enabled_flag;
+    unsigned char uniform_spacing_flag;
+    unsigned char num_tile_columns_minus1;
+    unsigned char num_tile_rows_minus1;
+
+    unsigned short column_width_minus1[21];
+    unsigned short row_height_minus1[21];
+    unsigned int   reserved3[15];
+
+    // RefPicSets
+    int NumBitsForShortTermRPSInSlice;
+    int NumDeltaPocsOfRefRpsIdx;
+    int NumPocTotalCurr;
+    int NumPocStCurrBefore;
+    int NumPocStCurrAfter;
+    int NumPocLtCurr;
+    int CurrPicOrderCntVal;
+    int RefPicIdx[16];                      // [refpic] Indices of valid reference pictures (-1 if unused for reference)
+    int PicOrderCntVal[16];                 // [refpic]
+    unsigned char IsLongTerm[16];           // [refpic] 0=not a long-term reference, 1=long-term reference
+    unsigned char RefPicSetStCurrBefore[8]; // [0..NumPocStCurrBefore-1] -> refpic (0..15)
+    unsigned char RefPicSetStCurrAfter[8];  // [0..NumPocStCurrAfter-1] -> refpic (0..15)
+    unsigned char RefPicSetLtCurr[8];       // [0..NumPocLtCurr-1] -> refpic (0..15)
+    unsigned char RefPicSetInterLayer0[8];
+    unsigned char RefPicSetInterLayer1[8];
+    unsigned int  reserved4[12];
+
+    // scaling lists (diag order)
+    unsigned char ScalingList4x4[6][16];       // [matrixId][i]
+    unsigned char ScalingList8x8[6][64];       // [matrixId][i]
+    unsigned char ScalingList16x16[6][64];     // [matrixId][i]
+    unsigned char ScalingList32x32[2][64];     // [matrixId][i]
+    unsigned char ScalingListDCCoeff16x16[6];  // [matrixId]
+    unsigned char ScalingListDCCoeff32x32[2];  // [matrixId]
+} CUVIDHEVCPICPARAMS;
+
+
+/***********************************************************/
+//! \struct CUVIDVP8PICPARAMS
+//! VP8 picture parameters
+//! This structure is used in CUVIDPICPARAMS structure
+/***********************************************************/
+typedef struct _CUVIDVP8PICPARAMS
+{
+    int width;
+    int height;
+    unsigned int first_partition_size;
+    //Frame Indexes
+    unsigned char LastRefIdx;
+    unsigned char GoldenRefIdx;
+    unsigned char AltRefIdx;
+    union {
+        struct {
+            unsigned char frame_type : 1;    /**< 0 = KEYFRAME, 1 = INTERFRAME  */
+            unsigned char version : 3;
+            unsigned char show_frame : 1;
+            unsigned char update_mb_segmentation_data : 1;    /**< Must be 0 if segmentation is not enabled */
+            unsigned char Reserved2Bits : 2;
+        };
+        unsigned char wFrameTagFlags;
+    };
+    unsigned char Reserved1[4];
+    unsigned int  Reserved2[3];
+} CUVIDVP8PICPARAMS;
+
+/***********************************************************/
+//! \struct CUVIDVP9PICPARAMS
+//! VP9 picture parameters
+//! This structure is used in CUVIDPICPARAMS structure
+/***********************************************************/
+typedef struct _CUVIDVP9PICPARAMS
+{
+    unsigned int width;
+    unsigned int height;
+
+    //Frame Indices
+    unsigned char LastRefIdx;
+    unsigned char GoldenRefIdx;
+    unsigned char AltRefIdx;
+    unsigned char colorSpace;
+
+    unsigned short profile : 3;
+    unsigned short frameContextIdx : 2;
+    unsigned short frameType : 1;
+    unsigned short showFrame : 1;
+    unsigned short errorResilient : 1;
+    unsigned short frameParallelDecoding : 1;
+    unsigned short subSamplingX : 1;
+    unsigned short subSamplingY : 1;
+    unsigned short intraOnly : 1;
+    unsigned short allow_high_precision_mv : 1;
+    unsigned short refreshEntropyProbs : 1;
+    unsigned short reserved2Bits : 2;
+
+    unsigned short reserved16Bits;
+
+    unsigned char  refFrameSignBias[4];
+
+    unsigned char bitDepthMinus8Luma;
+    unsigned char bitDepthMinus8Chroma;
+    unsigned char loopFilterLevel;
+    unsigned char loopFilterSharpness;
+
+    unsigned char modeRefLfEnabled;
+    unsigned char log2_tile_columns;
+    unsigned char log2_tile_rows;
+
+    unsigned char segmentEnabled : 1;
+    unsigned char segmentMapUpdate : 1;
+    unsigned char segmentMapTemporalUpdate : 1;
+    unsigned char segmentFeatureMode : 1;
+    unsigned char reserved4Bits : 4;
+
+
+    unsigned char segmentFeatureEnable[8][4];
+    short         segmentFeatureData[8][4];
+    unsigned char mb_segment_tree_probs[7];
+    unsigned char segment_pred_probs[3];
+    unsigned char reservedSegment16Bits[2];
+
+    int qpYAc;
+    int qpYDc;
+    int qpChDc;
+    int qpChAc;
+
+    unsigned int activeRefIdx[3];
+    unsigned int resetFrameContext;
+    unsigned int mcomp_filter_type;
+    unsigned int mbRefLfDelta[4];
+    unsigned int mbModeLfDelta[2];
+    unsigned int frameTagSize;
+    unsigned int offsetToDctParts;
+    unsigned int reserved128Bits[4];
+
+} CUVIDVP9PICPARAMS;
+
+
+/******************************************************************************************/
+//! \struct CUVIDPICPARAMS
+//! Picture parameters for decoding
+//! This structure is used in cuvidDecodePicture API
+//! IN  for cuvidDecodePicture
+/******************************************************************************************/
+typedef struct _CUVIDPICPARAMS
+{
+    int PicWidthInMbs;                     /**< IN: Coded frame size in macroblocks                           */
+    int FrameHeightInMbs;                  /**< IN: Coded frame height in macroblocks                         */
+    int CurrPicIdx;                        /**< IN: Output index of the current picture                       */
+    int field_pic_flag;                    /**< IN: 0=frame picture, 1=field picture                          */
+    int bottom_field_flag;                 /**< IN: 0=top field, 1=bottom field (ignored if field_pic_flag=0) */
+    int second_field;                      /**< IN: Second field of a complementary field pair                */
+    // Bitstream data
+    unsigned int nBitstreamDataLen;        /**< IN: Number of bytes in bitstream data buffer                  */
+    const unsigned char *pBitstreamData;   /**< IN: Ptr to bitstream data for this picture (slice-layer)      */
+    unsigned int nNumSlices;               /**< IN: Number of slices in this picture                          */
+    const unsigned int *pSliceDataOffsets; /**< IN: nNumSlices entries, contains offset of each slice within
+                                                        the bitstream data buffer                             */
+    int ref_pic_flag;                      /**< IN: This picture is a reference picture                       */
+    int intra_pic_flag;                    /**< IN: This picture is entirely intra coded                      */
+    unsigned int Reserved[30];             /**< Reserved for future use                                       */
+    // IN: Codec-specific data
+    union {
+        CUVIDMPEG2PICPARAMS mpeg2;         /**< Also used for MPEG-1 */
+        CUVIDH264PICPARAMS  h264;
+        CUVIDVC1PICPARAMS   vc1;
+        CUVIDMPEG4PICPARAMS mpeg4;
+        CUVIDJPEGPICPARAMS  jpeg;
+        CUVIDHEVCPICPARAMS  hevc;
+        CUVIDVP8PICPARAMS   vp8;
+        CUVIDVP9PICPARAMS   vp9;
+        unsigned int CodecReserved[1024];
+    } CodecSpecific;
+} CUVIDPICPARAMS;
+
+
+/******************************************************/
+//! \struct CUVIDPROCPARAMS
+//! Picture parameters for postprocessing
+//! This structure is used in cuvidMapVideoFrame API
+/******************************************************/
+typedef struct _CUVIDPROCPARAMS
+{
+    int progressive_frame;              /**< IN: Input is progressive (deinterlace_mode will be ignored)                */
+    int second_field;                   /**< IN: Output the second field (ignored if deinterlace mode is Weave)         */
+    int top_field_first;                /**< IN: Input frame is top field first (1st field is top, 2nd field is bottom) */
+    int unpaired_field;                 /**< IN: Input only contains one field (2nd field is invalid)                   */
+    // The fields below are used for raw YUV input
+    unsigned int reserved_flags;        /**< Reserved for future use (set to zero)                                      */
+    unsigned int reserved_zero;         /**< Reserved (set to zero)                                                     */
+    unsigned long long raw_input_dptr;  /**< IN: Input CUdeviceptr for raw YUV extensions                               */
+    unsigned int raw_input_pitch;       /**< IN: pitch in bytes of raw YUV input (should be aligned appropriately)      */
+    unsigned int raw_input_format;      /**< IN: Input YUV format (cudaVideoCodec_enum)                                 */
+    unsigned long long raw_output_dptr; /**< IN: Output CUdeviceptr for raw YUV extensions                              */
+    unsigned int raw_output_pitch;      /**< IN: pitch in bytes of raw YUV output (should be aligned appropriately)     */
+    unsigned int Reserved1;             /**< Reserved for future use (set to zero)                                      */
+    CUstream output_stream;             /**< IN: stream object used by cuvidMapVideoFrame                               */
+    unsigned int Reserved[46];          /**< Reserved for future use (set to zero)                                      */
+    void *Reserved2[2];                 /**< Reserved for future use (set to zero)                                      */
+} CUVIDPROCPARAMS;
+
+
+/***********************************************************************************************************/
+//! VIDEO_DECODER
+//!
+//! In order to minimize decode latencies, there should be always at least 2 pictures in the decode
+//! queue at any time, in order to make sure that all decode engines are always busy.
+//!
+//! Overall data flow:
+//!  - cuvidGetDecoderCaps(...)
+//!  - cuvidCreateDecoder(...)
+//!  - For each picture:
+//!    + cuvidDecodePicture(N)
+//!    + cuvidMapVideoFrame(N-4)
+//!    + do some processing in cuda
+//!    + cuvidUnmapVideoFrame(N-4)
+//!    + cuvidDecodePicture(N+1)
+//!    + cuvidMapVideoFrame(N-3)
+//!    + ...
+//!  - cuvidDestroyDecoder(...)
+//!
+//! NOTE:
+//! - When the cuda context is created from a D3D device, the D3D device must also be created
+//!   with the D3DCREATE_MULTITHREADED flag.
+//! - There is a limit to how many pictures can be mapped simultaneously (ulNumOutputSurfaces)
+//! - cuvidDecodePicture may block the calling thread if there are too many pictures pending
+//!   in the decode queue
+/***********************************************************************************************************/
+
+
+/**********************************************************************************************************************/
+//! \fn CUresult CUDAAPI cuvidGetDecoderCaps(CUVIDDECODECAPS *pdc)
+//! Queries decode capabilities of NVDEC-HW based on CodecType, ChromaFormat and BitDepthMinus8 parameters.
+//! 1. Application fills IN parameters CodecType, ChromaFormat and BitDepthMinus8 of CUVIDDECODECAPS structure
+//! 2. On calling cuvidGetDecoderCaps, driver fills OUT parameters if the IN parameters are supported
+//!    If IN parameters passed to the driver are not supported by NVDEC-HW, then all OUT params are set to 0.
+//! E.g. on Geforce GTX 960:
+//!   App fills - eCodecType = cudaVideoCodec_H264; eChromaFormat = cudaVideoChromaFormat_420; nBitDepthMinus8 = 0;
+//!   Given IN parameters are supported, hence driver fills: bIsSupported = 1; nMinWidth   = 48; nMinHeight  = 16;
+//!   nMaxWidth = 4096; nMaxHeight = 4096; nMaxMBCount = 65536;
+//! CodedWidth*CodedHeight/256 must be less than or equal to nMaxMBCount
+/**********************************************************************************************************************/
+typedef CUresult CUDAAPI tcuvidGetDecoderCaps(CUVIDDECODECAPS *pdc);
+
+/********************************************************************************************************************/
+//! \fn CUresult CUDAAPI cuvidCreateDecoder(CUvideodecoder *phDecoder, CUVIDDECODECREATEINFO *pdci)
+//! Create the decoder object based on pdci. A handle to the created decoder is returned
+/********************************************************************************************************************/
+typedef CUresult CUDAAPI tcuvidCreateDecoder(CUvideodecoder *phDecoder, CUVIDDECODECREATEINFO *pdci);
+/********************************************************************************************************************/
+//! \fn CUresult CUDAAPI cuvidDestroyDecoder(CUvideodecoder hDecoder)
+//! Destroy the decoder object.
+/********************************************************************************************************************/
+typedef CUresult CUDAAPI tcuvidDestroyDecoder(CUvideodecoder hDecoder);
+
+/********************************************************************************************************************/
+//! \fn CUresult CUDAAPI cuvidDecodePicture(CUvideodecoder hDecoder, CUVIDPICPARAMS *pPicParams)
+//! Decode a single picture (field or frame)
+//! Kicks off HW decoding
+/********************************************************************************************************************/
+typedef CUresult CUDAAPI tcuvidDecodePicture(CUvideodecoder hDecoder, CUVIDPICPARAMS *pPicParams);
+
+
+#if !defined(__CUVID_DEVPTR64) || defined(__CUVID_INTERNAL)
+/************************************************************************************************************************/
+//! \fn CUresult CUDAAPI cuvidMapVideoFrame(CUvideodecoder hDecoder, int nPicIdx, unsigned int *pDevPtr,
+//!                                         unsigned int  *pPitch, CUVIDPROCPARAMS *pVPP);
+//! Post-process and map video frame corresponding to nPicIdx for use in cuda. Returns cuda device pointer and associated
+//! pitch of the video frame
+/************************************************************************************************************************/
+typedef CUresult CUDAAPI tcuvidMapVideoFrame(CUvideodecoder hDecoder, int nPicIdx,
+                                           unsigned int *pDevPtr, unsigned int *pPitch,
+                                           CUVIDPROCPARAMS *pVPP);
+
+/********************************************************************************************************************/
+//! \fn CUresult CUDAAPI cuvidUnmapVideoFrame(CUvideodecoder hDecoder, unsigned int DevPtr)
+//! Unmap a previously mapped video frame
+/********************************************************************************************************************/
+typedef CUresult CUDAAPI tcuvidUnmapVideoFrame(CUvideodecoder hDecoder, unsigned int DevPtr);
+#endif
+
+#if defined(_WIN64) || defined(__LP64__) || defined(__x86_64) || defined(AMD64) || defined(_M_AMD64)
+/************************************************************************************************************************/
+//! \fn CUresult CUDAAPI cuvidMapVideoFrame64(CUvideodecoder hDecoder, int nPicIdx, unsigned long long *pDevPtr,
+//!                                           unsigned int *pPitch, CUVIDPROCPARAMS *pVPP);
+//! Post-process and map video frame corresponding to nPicIdx for use in cuda. Returns cuda device pointer and associated
+//! pitch of the video frame
+/************************************************************************************************************************/
+typedef CUresult CUDAAPI tcuvidMapVideoFrame64(CUvideodecoder hDecoder, int nPicIdx, unsigned long long *pDevPtr,
+                                             unsigned int *pPitch, CUVIDPROCPARAMS *pVPP);
+
+/********************************************************************************************************************/
+//! \fn CUresult CUDAAPI cuvidUnmapVideoFrame64(CUvideodecoder hDecoder, unsigned long long DevPtr);
+//! Unmap a previously mapped video frame
+/********************************************************************************************************************/
+typedef CUresult CUDAAPI tcuvidUnmapVideoFrame64(CUvideodecoder hDecoder, unsigned long long DevPtr);
+
+#if defined(__CUVID_DEVPTR64) && !defined(__CUVID_INTERNAL)
+#define tcuvidMapVideoFrame      tcuvidMapVideoFrame64
+#define tcuvidUnmapVideoFrame    tcuvidUnmapVideoFrame64
+#endif
+#endif
+
+
+
+/********************************************************************************************************************/
+//!
+//! Context-locking: to facilitate multi-threaded implementations, the following 4 functions
+//! provide a simple mutex-style host synchronization. If a non-NULL context is specified
+//! in CUVIDDECODECREATEINFO, the codec library will acquire the mutex associated with the given
+//! context before making any cuda calls.
+//! A multi-threaded application could create a lock associated with a context handle so that
+//! multiple threads can safely share the same cuda context:
+//!  - use cuCtxPopCurrent immediately after context creation in order to create a 'floating' context
+//!    that can be passed to cuvidCtxLockCreate.
+//!  - When using a floating context, all cuda calls should only be made within a cuvidCtxLock/cuvidCtxUnlock section.
+//!
+//! NOTE: This is a safer alternative to cuCtxPushCurrent and cuCtxPopCurrent, and is not related to video
+//! decoder in any way (implemented as a critical section associated with cuCtx{Push|Pop}Current calls).
+/********************************************************************************************************************/
+
+/********************************************************************************************************************/
+//! \fn CUresult CUDAAPI cuvidCtxLockCreate(CUvideoctxlock *pLock, CUcontext ctx)
+//! This API is used to create CtxLock object
+/********************************************************************************************************************/
+typedef CUresult CUDAAPI tcuvidCtxLockCreate(CUvideoctxlock *pLock, CUcontext ctx);
+
+/********************************************************************************************************************/
+//! \fn CUresult CUDAAPI cuvidCtxLockDestroy(CUvideoctxlock lck)
+//! This API is used to free CtxLock object
+/********************************************************************************************************************/
+typedef CUresult CUDAAPI tcuvidCtxLockDestroy(CUvideoctxlock lck);
+
+/********************************************************************************************************************/
+//! \fn CUresult CUDAAPI cuvidCtxLock(CUvideoctxlock lck, unsigned int reserved_flags)
+//! This API is used to acquire ctxlock
+/********************************************************************************************************************/
+typedef CUresult CUDAAPI tcuvidCtxLock(CUvideoctxlock lck, unsigned int reserved_flags);
+
+/********************************************************************************************************************/
+//! \fn CUresult CUDAAPI cuvidCtxUnlock(CUvideoctxlock lck, unsigned int reserved_flags)
+//! This API is used to release ctxlock
+/********************************************************************************************************************/
+typedef CUresult CUDAAPI tcuvidCtxUnlock(CUvideoctxlock lck, unsigned int reserved_flags);
+
+/**********************************************************************************************/
+
+#if defined(__cplusplus)
+}
+#endif /* __cplusplus */
+
+#endif // __CUDA_VIDEO_H__

compat/cuda/dynlink_loader.h

@@ -1,33 +1,268 @@
 /*
- * This file is part of FFmpeg.
+ * This copyright notice applies to this header file only:
  *
- * 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.
+ * Copyright (c) 2016
  *
- * 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.
+ * 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:
  *
- * 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
+ * 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.
  */
 
-#ifndef COMPAT_CUDA_DYNLINK_LOADER_H
-#define COMPAT_CUDA_DYNLINK_LOADER_H
+#ifndef AV_COMPAT_CUDA_DYNLINK_LOADER_H
+#define AV_COMPAT_CUDA_DYNLINK_LOADER_H
 
-#include "libavutil/log.h"
+#include "compat/cuda/dynlink_cuda.h"
+#include "compat/cuda/dynlink_nvcuvid.h"
+#include "compat/nvenc/nvEncodeAPI.h"
 #include "compat/w32dlfcn.h"
 
-#define FFNV_LOAD_FUNC(path) dlopen((path), RTLD_LAZY)
-#define FFNV_SYM_FUNC(lib, sym) dlsym((lib), (sym))
-#define FFNV_FREE_FUNC(lib) dlclose(lib)
-#define FFNV_LOG_FUNC(logctx, msg, ...) av_log(logctx, AV_LOG_ERROR, msg,  __VA_ARGS__)
-#define FFNV_DEBUG_LOG_FUNC(logctx, msg, ...) av_log(logctx, AV_LOG_DEBUG, msg,  __VA_ARGS__)
+#include "libavutil/log.h"
+#include "libavutil/error.h"
 
-#include <ffnvcodec/dynlink_loader.h>
+#if defined(_WIN32)
+# define LIB_HANDLE HMODULE
+#else
+# define LIB_HANDLE void*
+#endif
+
+#if defined(_WIN32) || defined(__CYGWIN__)
+# define CUDA_LIBNAME "nvcuda.dll"
+# define NVCUVID_LIBNAME "nvcuvid.dll"
+# if ARCH_X86_64
+#  define NVENC_LIBNAME "nvEncodeAPI64.dll"
+# else
+#  define NVENC_LIBNAME "nvEncodeAPI.dll"
+# endif
+#else
+# define CUDA_LIBNAME "libcuda.so.1"
+# define NVCUVID_LIBNAME "libnvcuvid.so.1"
+# define NVENC_LIBNAME "libnvidia-encode.so.1"
+#endif
+
+#define LOAD_LIBRARY(l, path)                                     \
+    do {                                                          \
+        if (!((l) = dlopen(path, RTLD_LAZY))) {                   \
+            av_log(NULL, AV_LOG_ERROR, "Cannot load %s\n", path); \
+            ret = AVERROR_UNKNOWN;                                \
+            goto error;                                           \
+        }                                                         \
+        av_log(NULL, AV_LOG_TRACE, "Loaded lib: %s\n", path);     \
+    } while (0)
+
+#define LOAD_SYMBOL(fun, tp, symbol)                                \
+    do {                                                            \
+        if (!((f->fun) = (tp*)dlsym(f->lib, symbol))) {             \
+            av_log(NULL, AV_LOG_ERROR, "Cannot load %s\n", symbol); \
+            ret = AVERROR_UNKNOWN;                                  \
+            goto error;                                             \
+        }                                                           \
+        av_log(NULL, AV_LOG_TRACE, "Loaded sym: %s\n", symbol);     \
+    } while (0)
+
+#define LOAD_SYMBOL_OPT(fun, tp, symbol)                                     \
+    do {                                                                     \
+        if (!((f->fun) = (tp*)dlsym(f->lib, symbol))) {                      \
+            av_log(NULL, AV_LOG_DEBUG, "Cannot load optional %s\n", symbol); \
+        } else {                                                             \
+            av_log(NULL, AV_LOG_TRACE, "Loaded sym: %s\n", symbol);          \
+        }                                                                    \
+    } while (0)
+
+#define GENERIC_LOAD_FUNC_PREAMBLE(T, n, N)  \
+    T *f;                                    \
+    int ret;                                 \
+                                             \
+    n##_free_functions(functions);           \
+                                             \
+    f = *functions = av_mallocz(sizeof(*f)); \
+    if (!f)                                  \
+        return AVERROR(ENOMEM);              \
+                                             \
+    LOAD_LIBRARY(f->lib, N);
+
+#define GENERIC_LOAD_FUNC_FINALE(n) \
+    return 0;                       \
+error:                              \
+    n##_free_functions(functions);  \
+    return ret;
+
+#define GENERIC_FREE_FUNC()              \
+    if (!functions)                      \
+        return;                          \
+    if (*functions && (*functions)->lib) \
+        dlclose((*functions)->lib);      \
+    av_freep(functions);
+
+#ifdef AV_COMPAT_DYNLINK_CUDA_H
+typedef struct CudaFunctions {
+    tcuInit *cuInit;
+    tcuDeviceGetCount *cuDeviceGetCount;
+    tcuDeviceGet *cuDeviceGet;
+    tcuDeviceGetName *cuDeviceGetName;
+    tcuDeviceComputeCapability *cuDeviceComputeCapability;
+    tcuCtxCreate_v2 *cuCtxCreate;
+    tcuCtxPushCurrent_v2 *cuCtxPushCurrent;
+    tcuCtxPopCurrent_v2 *cuCtxPopCurrent;
+    tcuCtxDestroy_v2 *cuCtxDestroy;
+    tcuMemAlloc_v2 *cuMemAlloc;
+    tcuMemFree_v2 *cuMemFree;
+    tcuMemcpy2D_v2 *cuMemcpy2D;
+    tcuGetErrorName *cuGetErrorName;
+    tcuGetErrorString *cuGetErrorString;
+
+    LIB_HANDLE lib;
+} CudaFunctions;
+#else
+typedef struct CudaFunctions CudaFunctions;
+#endif
+
+typedef struct CuvidFunctions {
+    tcuvidGetDecoderCaps *cuvidGetDecoderCaps;
+    tcuvidCreateDecoder *cuvidCreateDecoder;
+    tcuvidDestroyDecoder *cuvidDestroyDecoder;
+    tcuvidDecodePicture *cuvidDecodePicture;
+    tcuvidMapVideoFrame *cuvidMapVideoFrame;
+    tcuvidUnmapVideoFrame *cuvidUnmapVideoFrame;
+    tcuvidCtxLockCreate *cuvidCtxLockCreate;
+    tcuvidCtxLockDestroy *cuvidCtxLockDestroy;
+    tcuvidCtxLock *cuvidCtxLock;
+    tcuvidCtxUnlock *cuvidCtxUnlock;
+
+    tcuvidCreateVideoSource *cuvidCreateVideoSource;
+    tcuvidCreateVideoSourceW *cuvidCreateVideoSourceW;
+    tcuvidDestroyVideoSource *cuvidDestroyVideoSource;
+    tcuvidSetVideoSourceState *cuvidSetVideoSourceState;
+    tcuvidGetVideoSourceState *cuvidGetVideoSourceState;
+    tcuvidGetSourceVideoFormat *cuvidGetSourceVideoFormat;
+    tcuvidGetSourceAudioFormat *cuvidGetSourceAudioFormat;
+    tcuvidCreateVideoParser *cuvidCreateVideoParser;
+    tcuvidParseVideoData *cuvidParseVideoData;
+    tcuvidDestroyVideoParser *cuvidDestroyVideoParser;
+
+    LIB_HANDLE lib;
+} CuvidFunctions;
+
+typedef NVENCSTATUS NVENCAPI tNvEncodeAPICreateInstance(NV_ENCODE_API_FUNCTION_LIST *functionList);
+typedef NVENCSTATUS NVENCAPI tNvEncodeAPIGetMaxSupportedVersion(uint32_t* version);
+
+typedef struct NvencFunctions {
+    tNvEncodeAPICreateInstance *NvEncodeAPICreateInstance;
+    tNvEncodeAPIGetMaxSupportedVersion *NvEncodeAPIGetMaxSupportedVersion;
+
+    LIB_HANDLE lib;
+} NvencFunctions;
+
+#ifdef AV_COMPAT_DYNLINK_CUDA_H
+static inline void cuda_free_functions(CudaFunctions **functions)
+{
+    GENERIC_FREE_FUNC();
+}
+#endif
+
+static inline void cuvid_free_functions(CuvidFunctions **functions)
+{
+    GENERIC_FREE_FUNC();
+}
+
+static inline void nvenc_free_functions(NvencFunctions **functions)
+{
+    GENERIC_FREE_FUNC();
+}
+
+#ifdef AV_COMPAT_DYNLINK_CUDA_H
+static inline int cuda_load_functions(CudaFunctions **functions)
+{
+    GENERIC_LOAD_FUNC_PREAMBLE(CudaFunctions, cuda, CUDA_LIBNAME);
+
+    LOAD_SYMBOL(cuInit, tcuInit, "cuInit");
+    LOAD_SYMBOL(cuDeviceGetCount, tcuDeviceGetCount, "cuDeviceGetCount");
+    LOAD_SYMBOL(cuDeviceGet, tcuDeviceGet, "cuDeviceGet");
+    LOAD_SYMBOL(cuDeviceGetName, tcuDeviceGetName, "cuDeviceGetName");
+    LOAD_SYMBOL(cuDeviceComputeCapability, tcuDeviceComputeCapability, "cuDeviceComputeCapability");
+    LOAD_SYMBOL(cuCtxCreate, tcuCtxCreate_v2, "cuCtxCreate_v2");
+    LOAD_SYMBOL(cuCtxPushCurrent, tcuCtxPushCurrent_v2, "cuCtxPushCurrent_v2");
+    LOAD_SYMBOL(cuCtxPopCurrent, tcuCtxPopCurrent_v2, "cuCtxPopCurrent_v2");
+    LOAD_SYMBOL(cuCtxDestroy, tcuCtxDestroy_v2, "cuCtxDestroy_v2");
+    LOAD_SYMBOL(cuMemAlloc, tcuMemAlloc_v2, "cuMemAlloc_v2");
+    LOAD_SYMBOL(cuMemFree, tcuMemFree_v2, "cuMemFree_v2");
+    LOAD_SYMBOL(cuMemcpy2D, tcuMemcpy2D_v2, "cuMemcpy2D_v2");
+    LOAD_SYMBOL(cuGetErrorName, tcuGetErrorName, "cuGetErrorName");
+    LOAD_SYMBOL(cuGetErrorString, tcuGetErrorString, "cuGetErrorString");
+
+    GENERIC_LOAD_FUNC_FINALE(cuda);
+}
+#endif
+
+static inline int cuvid_load_functions(CuvidFunctions **functions)
+{
+    GENERIC_LOAD_FUNC_PREAMBLE(CuvidFunctions, cuvid, NVCUVID_LIBNAME);
+
+    LOAD_SYMBOL_OPT(cuvidGetDecoderCaps, tcuvidGetDecoderCaps, "cuvidGetDecoderCaps");
+    LOAD_SYMBOL(cuvidCreateDecoder, tcuvidCreateDecoder, "cuvidCreateDecoder");
+    LOAD_SYMBOL(cuvidDestroyDecoder, tcuvidDestroyDecoder, "cuvidDestroyDecoder");
+    LOAD_SYMBOL(cuvidDecodePicture, tcuvidDecodePicture, "cuvidDecodePicture");
+#ifdef __CUVID_DEVPTR64
+    LOAD_SYMBOL(cuvidMapVideoFrame, tcuvidMapVideoFrame, "cuvidMapVideoFrame64");
+    LOAD_SYMBOL(cuvidUnmapVideoFrame, tcuvidUnmapVideoFrame, "cuvidUnmapVideoFrame64");
+#else
+    LOAD_SYMBOL(cuvidMapVideoFrame, tcuvidMapVideoFrame, "cuvidMapVideoFrame");
+    LOAD_SYMBOL(cuvidUnmapVideoFrame, tcuvidUnmapVideoFrame, "cuvidUnmapVideoFrame");
+#endif
+    LOAD_SYMBOL(cuvidCtxLockCreate, tcuvidCtxLockCreate, "cuvidCtxLockCreate");
+    LOAD_SYMBOL(cuvidCtxLockDestroy, tcuvidCtxLockDestroy, "cuvidCtxLockDestroy");
+    LOAD_SYMBOL(cuvidCtxLock, tcuvidCtxLock, "cuvidCtxLock");
+    LOAD_SYMBOL(cuvidCtxUnlock, tcuvidCtxUnlock, "cuvidCtxUnlock");
+
+    LOAD_SYMBOL(cuvidCreateVideoSource, tcuvidCreateVideoSource, "cuvidCreateVideoSource");
+    LOAD_SYMBOL(cuvidCreateVideoSourceW, tcuvidCreateVideoSourceW, "cuvidCreateVideoSourceW");
+    LOAD_SYMBOL(cuvidDestroyVideoSource, tcuvidDestroyVideoSource, "cuvidDestroyVideoSource");
+    LOAD_SYMBOL(cuvidSetVideoSourceState, tcuvidSetVideoSourceState, "cuvidSetVideoSourceState");
+    LOAD_SYMBOL(cuvidGetVideoSourceState, tcuvidGetVideoSourceState, "cuvidGetVideoSourceState");
+    LOAD_SYMBOL(cuvidGetSourceVideoFormat, tcuvidGetSourceVideoFormat, "cuvidGetSourceVideoFormat");
+    LOAD_SYMBOL(cuvidGetSourceAudioFormat, tcuvidGetSourceAudioFormat, "cuvidGetSourceAudioFormat");
+    LOAD_SYMBOL(cuvidCreateVideoParser, tcuvidCreateVideoParser, "cuvidCreateVideoParser");
+    LOAD_SYMBOL(cuvidParseVideoData, tcuvidParseVideoData, "cuvidParseVideoData");
+    LOAD_SYMBOL(cuvidDestroyVideoParser, tcuvidDestroyVideoParser, "cuvidDestroyVideoParser");
+
+    GENERIC_LOAD_FUNC_FINALE(cuvid);
+}
+
+static inline int nvenc_load_functions(NvencFunctions **functions)
+{
+    GENERIC_LOAD_FUNC_PREAMBLE(NvencFunctions, nvenc, NVENC_LIBNAME);
+
+    LOAD_SYMBOL(NvEncodeAPICreateInstance, tNvEncodeAPICreateInstance, "NvEncodeAPICreateInstance");
+    LOAD_SYMBOL(NvEncodeAPIGetMaxSupportedVersion, tNvEncodeAPIGetMaxSupportedVersion, "NvEncodeAPIGetMaxSupportedVersion");
+
+    GENERIC_LOAD_FUNC_FINALE(nvenc);
+}
+
+#undef GENERIC_LOAD_FUNC_PREAMBLE
+#undef LOAD_LIBRARY
+#undef LOAD_SYMBOL
+#undef GENERIC_LOAD_FUNC_FINALE
+#undef GENERIC_FREE_FUNC
+#undef CUDA_LIBNAME
+#undef NVCUVID_LIBNAME
+#undef NVENC_LIBNAME
+#undef LIB_HANDLE
+
+#endif
 
-#endif /* COMPAT_CUDA_DYNLINK_LOADER_H */

compat/cuda/dynlink_nvcuvid.h

@@ -0,0 +1,356 @@
+/*
+ * This copyright notice applies to this header file only:
+ *
+ * Copyright (c) 2010-2017 NVIDIA Corporation
+ *
+ * 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 nvcuvid.h
+//!   NVDECODE API provides video decoding interface to NVIDIA GPU devices.
+//! \date 2015-2017
+//!  This file contains the interface constants, structure definitions and function prototypes.
+/********************************************************************************************************************/
+
+#if !defined(__NVCUVID_H__)
+#define __NVCUVID_H__
+
+#include "compat/cuda/dynlink_cuviddec.h"
+
+#if defined(__cplusplus)
+extern "C" {
+#endif /* __cplusplus */
+
+/*********************************
+** Initialization
+*********************************/
+CUresult  CUDAAPI cuvidInit(unsigned int Flags);
+
+/***********************************************/
+//!
+//! High-level helper APIs for video sources
+//!
+/***********************************************/
+
+typedef void *CUvideosource;
+typedef void *CUvideoparser;
+typedef long long CUvideotimestamp;
+
+
+/************************************************************************/
+//! \enum cudaVideoState
+//! Video source state enums
+//! Used in cuvidSetVideoSourceState and cuvidGetVideoSourceState APIs
+/************************************************************************/
+typedef enum {
+    cudaVideoState_Error   = -1,    /**< Error state (invalid source)                  */
+    cudaVideoState_Stopped = 0,     /**< Source is stopped (or reached end-of-stream)  */
+    cudaVideoState_Started = 1      /**< Source is running and delivering data         */
+} cudaVideoState;
+
+/************************************************************************/
+//! \enum cudaAudioCodec
+//! Audio compression enums
+//! Used in CUAUDIOFORMAT structure
+/************************************************************************/
+typedef enum {
+    cudaAudioCodec_MPEG1=0,         /**< MPEG-1 Audio               */
+    cudaAudioCodec_MPEG2,           /**< MPEG-2 Audio               */
+    cudaAudioCodec_MP3,             /**< MPEG-1 Layer III Audio     */
+    cudaAudioCodec_AC3,             /**< Dolby Digital (AC3) Audio  */
+    cudaAudioCodec_LPCM,            /**< PCM Audio                  */
+    cudaAudioCodec_AAC,             /**< AAC Audio                  */
+} cudaAudioCodec;
+
+/************************************************************************************************/
+//! \ingroup STRUCTS
+//! \struct CUVIDEOFORMAT
+//! Video format
+//! Used in cuvidGetSourceVideoFormat API
+/************************************************************************************************/
+typedef struct
+{
+    cudaVideoCodec codec;                   /**< OUT: Compression format          */
+   /**
+    * OUT: frame rate = numerator / denominator (for example: 30000/1001)
+    */
+    struct {
+        /**< OUT: frame rate numerator   (0 = unspecified or variable frame rate) */
+        unsigned int numerator;
+        /**< OUT: frame rate denominator (0 = unspecified or variable frame rate) */
+        unsigned int denominator;
+    } frame_rate;
+    unsigned char progressive_sequence;     /**< OUT: 0=interlaced, 1=progressive                                      */
+    unsigned char bit_depth_luma_minus8;    /**< OUT: high bit depth luma. E.g, 2 for 10-bitdepth, 4 for 12-bitdepth   */
+    unsigned char bit_depth_chroma_minus8;  /**< OUT: high bit depth chroma. E.g, 2 for 10-bitdepth, 4 for 12-bitdepth */
+    unsigned char reserved1;                /**< Reserved for future use                                               */
+    unsigned int coded_width;               /**< OUT: coded frame width in pixels                                      */
+    unsigned int coded_height;              /**< OUT: coded frame height in pixels                                     */
+   /**
+    * area of the frame that should be displayed
+    * typical example:
+    * coded_width = 1920, coded_height = 1088
+    * display_area = { 0,0,1920,1080 }
+    */
+    struct {
+        int left;                           /**< OUT: left position of display rect    */
+        int top;                            /**< OUT: top position of display rect     */
+        int right;                          /**< OUT: right position of display rect   */
+        int bottom;                         /**< OUT: bottom position of display rect  */
+    } display_area;
+    cudaVideoChromaFormat chroma_format;    /**< OUT:  Chroma format                   */
+    unsigned int bitrate;                   /**< OUT: video bitrate (bps, 0=unknown)   */
+   /**
+    * OUT: Display Aspect Ratio = x:y (4:3, 16:9, etc)
+    */
+    struct {
+        int x;
+        int y;
+    } display_aspect_ratio;
+    /**
+    * Video Signal Description
+    * Refer section E.2.1 (VUI parameters semantics) of H264 spec file
+    */
+    struct {
+        unsigned char video_format          : 3; /**< OUT: 0-Component, 1-PAL, 2-NTSC, 3-SECAM, 4-MAC, 5-Unspecified     */
+        unsigned char video_full_range_flag : 1; /**< OUT: indicates the black level and luma and chroma range           */
+        unsigned char reserved_zero_bits    : 4; /**< Reserved bits                                                      */
+        unsigned char color_primaries;           /**< OUT: chromaticity coordinates of source primaries                  */
+        unsigned char transfer_characteristics;  /**< OUT: opto-electronic transfer characteristic of the source picture */
+        unsigned char matrix_coefficients;       /**< OUT: used in deriving luma and chroma signals from RGB primaries   */
+    } video_signal_description;
+    unsigned int seqhdr_data_length;             /**< OUT: Additional bytes following (CUVIDEOFORMATEX)                  */
+} CUVIDEOFORMAT;
+
+/****************************************************************/
+//! \ingroup STRUCTS
+//! \struct CUVIDEOFORMATEX
+//! Video format including raw sequence header information
+//! Used in cuvidGetSourceVideoFormat API
+/****************************************************************/
+typedef struct
+{
+    CUVIDEOFORMAT format;                 /**< OUT: CUVIDEOFORMAT structure */
+    unsigned char raw_seqhdr_data[1024];  /**< OUT: Sequence header data    */
+} CUVIDEOFORMATEX;
+
+/****************************************************************/
+//! \ingroup STRUCTS
+//! \struct CUAUDIOFORMAT
+//! Audio formats
+//! Used in cuvidGetSourceAudioFormat API
+/****************************************************************/
+typedef struct
+{
+    cudaAudioCodec codec;       /**< OUT: Compression format                                              */
+    unsigned int channels;      /**< OUT: number of audio channels                                        */
+    unsigned int samplespersec; /**< OUT: sampling frequency                                              */
+    unsigned int bitrate;       /**< OUT: For uncompressed, can also be used to determine bits per sample */
+    unsigned int reserved1;     /**< Reserved for future use                                              */
+    unsigned int reserved2;     /**< Reserved for future use                                              */
+} CUAUDIOFORMAT;
+
+
+/***************************************************************/
+//! \enum CUvideopacketflags
+//! Data packet flags
+//! Used in CUVIDSOURCEDATAPACKET structure
+/***************************************************************/
+typedef enum {
+    CUVID_PKT_ENDOFSTREAM   = 0x01,   /**< Set when this is the last packet for this stream  */
+    CUVID_PKT_TIMESTAMP     = 0x02,   /**< Timestamp is valid                                */
+    CUVID_PKT_DISCONTINUITY = 0x04,   /**< Set when a discontinuity has to be signalled      */
+    CUVID_PKT_ENDOFPICTURE  = 0x08,   /**< Set when the packet contains exactly one frame    */
+} CUvideopacketflags;
+
+/*****************************************************************************/
+//! \ingroup STRUCTS
+//! \struct CUVIDSOURCEDATAPACKET
+//! Data Packet
+//! Used in cuvidParseVideoData API
+//! IN for cuvidParseVideoData
+/*****************************************************************************/
+typedef struct _CUVIDSOURCEDATAPACKET
+{
+    tcu_ulong flags;                /**< IN: Combination of CUVID_PKT_XXX flags                              */
+    tcu_ulong payload_size;         /**< IN: number of bytes in the payload (may be zero if EOS flag is set) */
+    const unsigned char *payload;   /**< IN: Pointer to packet payload data (may be NULL if EOS flag is set) */
+    CUvideotimestamp timestamp;     /**< IN: Presentation time stamp (10MHz clock), only valid if
+                                             CUVID_PKT_TIMESTAMP flag is set                                 */
+} CUVIDSOURCEDATAPACKET;
+
+// Callback for packet delivery
+typedef int (CUDAAPI *PFNVIDSOURCECALLBACK)(void *, CUVIDSOURCEDATAPACKET *);
+
+/**************************************************************************************************************************/
+//! \ingroup STRUCTS
+//! \struct CUVIDSOURCEPARAMS
+//! Describes parameters needed in cuvidCreateVideoSource API
+//! NVDECODE API is intended for HW accelerated video decoding so CUvideosource doesn't have audio demuxer for all supported
+//! containers. It's recommended to clients to use their own or third party demuxer if audio support is needed.
+/**************************************************************************************************************************/
+typedef struct _CUVIDSOURCEPARAMS
+{
+    unsigned int ulClockRate;                   /**< IN: Time stamp units in Hz (0=default=10000000Hz)      */
+    unsigned int uReserved1[7];                 /**< Reserved for future use - set to zero                  */
+    void *pUserData;                            /**< IN: User private data passed in to the data handlers   */
+    PFNVIDSOURCECALLBACK pfnVideoDataHandler;   /**< IN: Called to deliver video packets                    */
+    PFNVIDSOURCECALLBACK pfnAudioDataHandler;   /**< IN: Called to deliver audio packets.                   */
+    void *pvReserved2[8];                       /**< Reserved for future use - set to NULL                  */
+} CUVIDSOURCEPARAMS;
+
+
+/**********************************************/
+//! \ingroup ENUMS
+//! \enum CUvideosourceformat_flags
+//! CUvideosourceformat_flags
+//! Used in cuvidGetSourceVideoFormat API
+/**********************************************/
+typedef enum {
+    CUVID_FMT_EXTFORMATINFO = 0x100             /**< Return extended format structure (CUVIDEOFORMATEX) */
+} CUvideosourceformat_flags;
+
+#if !defined(__APPLE__)
+/**************************************************************************************************************************/
+//! \fn CUresult CUDAAPI cuvidCreateVideoSource(CUvideosource *pObj, const char *pszFileName, CUVIDSOURCEPARAMS *pParams)
+//! Create CUvideosource object. CUvideosource spawns demultiplexer thread that provides two callbacks:
+//! pfnVideoDataHandler() and pfnAudioDataHandler()
+//! NVDECODE API is intended for HW accelerated video decoding so CUvideosource doesn't have audio demuxer for all supported
+//! containers. It's recommended to clients to use their own or third party demuxer if audio support is needed.
+/**************************************************************************************************************************/
+typedef CUresult CUDAAPI tcuvidCreateVideoSource(CUvideosource *pObj, const char *pszFileName, CUVIDSOURCEPARAMS *pParams);
+
+/****************************************************************************************************************************/
+//! \fn CUresult CUDAAPI cuvidCreateVideoSourceW(CUvideosource *pObj, const wchar_t *pwszFileName, CUVIDSOURCEPARAMS *pParams)
+//! Create video source object and initialize
+/****************************************************************************************************************************/
+typedef CUresult CUDAAPI tcuvidCreateVideoSourceW(CUvideosource *pObj, const wchar_t *pwszFileName, CUVIDSOURCEPARAMS *pParams);
+
+/*********************************************************************/
+//! \fn CUresult CUDAAPI cuvidDestroyVideoSource(CUvideosource obj)
+//! Destroy video source
+/*********************************************************************/
+typedef CUresult CUDAAPI tcuvidDestroyVideoSource(CUvideosource obj);
+
+/******************************************************************************************/
+//! \fn CUresult CUDAAPI cuvidSetVideoSourceState(CUvideosource obj, cudaVideoState state)
+//! Set video source state
+/******************************************************************************************/
+typedef CUresult CUDAAPI tcuvidSetVideoSourceState(CUvideosource obj, cudaVideoState state);
+
+/******************************************************************************************/
+//! \fn cudaVideoState CUDAAPI cuvidGetVideoSourceState(CUvideosource obj)
+//! Get video source state
+/******************************************************************************************/
+typedef cudaVideoState CUDAAPI tcuvidGetVideoSourceState(CUvideosource obj);
+
+/****************************************************************************************************************/
+//! \fn CUresult CUDAAPI cuvidGetSourceVideoFormat(CUvideosource obj, CUVIDEOFORMAT *pvidfmt, unsigned int flags)
+//! Gets details of video stream in pvidfmt
+/****************************************************************************************************************/
+typedef CUresult CUDAAPI tcuvidGetSourceVideoFormat(CUvideosource obj, CUVIDEOFORMAT *pvidfmt, unsigned int flags);
+
+/****************************************************************************************************************/
+//! \fn CUresult CUDAAPI cuvidGetSourceAudioFormat(CUvideosource obj, CUAUDIOFORMAT *paudfmt, unsigned int flags)
+//! Get audio source format
+//! NVDECODE API is intended for HW accelarated video decoding so CUvideosource doesn't have audio demuxer for all suppported
+//! containers. It's recommended to clients to use their own or third party demuxer if audio support is needed.
+/****************************************************************************************************************/
+typedef CUresult CUDAAPI tcuvidGetSourceAudioFormat(CUvideosource obj, CUAUDIOFORMAT *paudfmt, unsigned int flags);
+
+#endif
+/**********************************************************************************/
+//! \ingroup STRUCTS
+//! \struct CUVIDPARSERDISPINFO
+//! Used in cuvidParseVideoData API with PFNVIDDISPLAYCALLBACK pfnDisplayPicture
+/**********************************************************************************/
+typedef struct _CUVIDPARSERDISPINFO
+{
+    int picture_index;          /**< OUT: Index of the current picture                                                         */
+    int progressive_frame;      /**< OUT: 1 if progressive frame; 0 otherwise                                                  */
+    int top_field_first;        /**< OUT: 1 if top field is displayed first; 0 otherwise                                       */
+    int repeat_first_field;     /**< OUT: Number of additional fields (1=ivtc, 2=frame doubling, 4=frame tripling,
+                                     -1=unpaired field)                                                                        */
+    CUvideotimestamp timestamp; /**< OUT: Presentation time stamp                                                              */
+} CUVIDPARSERDISPINFO;
+
+/***********************************************************************************************************************/
+//! Parser callbacks
+//! The parser will call these synchronously from within cuvidParseVideoData(), whenever a picture is ready to
+//! be decoded and/or displayed. First argument in functions is "void *pUserData" member of structure CUVIDSOURCEPARAMS
+/***********************************************************************************************************************/
+typedef int (CUDAAPI *PFNVIDSEQUENCECALLBACK)(void *, CUVIDEOFORMAT *);
+typedef int (CUDAAPI *PFNVIDDECODECALLBACK)(void *, CUVIDPICPARAMS *);
+typedef int (CUDAAPI *PFNVIDDISPLAYCALLBACK)(void *, CUVIDPARSERDISPINFO *);
+
+/**************************************/
+//! \ingroup STRUCTS
+//! \struct CUVIDPARSERPARAMS
+//! Used in cuvidCreateVideoParser API
+/**************************************/
+typedef struct _CUVIDPARSERPARAMS
+{
+    cudaVideoCodec CodecType;                   /**< IN: cudaVideoCodec_XXX                                                  */
+    unsigned int ulMaxNumDecodeSurfaces;        /**< IN: Max # of decode surfaces (parser will cycle through these)          */
+    unsigned int ulClockRate;                   /**< IN: Timestamp units in Hz (0=default=10000000Hz)                        */
+    unsigned int ulErrorThreshold;              /**< IN: % Error threshold (0-100) for calling pfnDecodePicture (100=always
+                                                     IN: call pfnDecodePicture even if picture bitstream is fully corrupted) */
+    unsigned int ulMaxDisplayDelay;             /**< IN: Max display queue delay (improves pipelining of decode with display)
+                                                         0=no delay (recommended values: 2..4)                               */
+    unsigned int uReserved1[5];                 /**< IN: Reserved for future use - set to 0                                  */
+    void *pUserData;                            /**< IN: User data for callbacks                                             */
+    PFNVIDSEQUENCECALLBACK pfnSequenceCallback; /**< IN: Called before decoding frames and/or whenever there is a fmt change */
+    PFNVIDDECODECALLBACK pfnDecodePicture;      /**< IN: Called when a picture is ready to be decoded (decode order)         */
+    PFNVIDDISPLAYCALLBACK pfnDisplayPicture;    /**< IN: Called whenever a picture is ready to be displayed (display order)  */
+    void *pvReserved2[7];                       /**< Reserved for future use - set to NULL                                   */
+    CUVIDEOFORMATEX *pExtVideoInfo;             /**< IN: [Optional] sequence header data from system layer                   */
+} CUVIDPARSERPARAMS;
+
+/************************************************************************************************/
+//! \fn CUresult CUDAAPI cuvidCreateVideoParser(CUvideoparser *pObj, CUVIDPARSERPARAMS *pParams)
+//! Create video parser object and initialize
+/************************************************************************************************/
+typedef CUresult CUDAAPI tcuvidCreateVideoParser(CUvideoparser *pObj, CUVIDPARSERPARAMS *pParams);
+
+/************************************************************************************************/
+//! \fn CUresult CUDAAPI cuvidParseVideoData(CUvideoparser obj, CUVIDSOURCEDATAPACKET *pPacket)
+//! Parse the video data from source data packet in pPacket
+//! Extracts parameter sets like SPS, PPS, bitstream etc. from pPacket and
+//! calls back pfnDecodePicture with CUVIDPICPARAMS data for kicking of HW decoding
+/************************************************************************************************/
+typedef CUresult CUDAAPI tcuvidParseVideoData(CUvideoparser obj, CUVIDSOURCEDATAPACKET *pPacket);
+
+/*******************************************************************/
+//! \fn CUresult CUDAAPI cuvidDestroyVideoParser(CUvideoparser obj)
+/*******************************************************************/
+typedef CUresult CUDAAPI tcuvidDestroyVideoParser(CUvideoparser obj);
+
+/**********************************************************************************************/
+
+#if defined(__cplusplus)
+}
+#endif /* __cplusplus */
+
+#endif // __NVCUVID_H__
+
+

compat/cuda/ptx2c.sh

@@ -27,7 +27,7 @@ IN="$2"
 NAME="$(basename "$IN" | sed 's/\..*//')"
 
 printf "const char %s_ptx[] = \\" "$NAME" > "$OUT"
-while IFS= read -r LINE
+while read LINE
 do
     printf "\n\t\"%s\\\n\"" "$(printf "%s" "$LINE" | sed -e 's/\r//g' -e 's/["\\]/\\&/g')" >> "$OUT"
 done < "$IN"

compat/os2threads.h

@@ -1,5 +1,5 @@
 /*
- * Copyright (c) 2011-2017 KO Myung-Hun <komh@chollian.net>
+ * Copyright (c) 2011 KO Myung-Hun <komh@chollian.net>
  *
  * This file is part of FFmpeg.
  *
@@ -46,11 +46,9 @@ typedef struct {
 
 typedef void pthread_attr_t;
 
-typedef _fmutex pthread_mutex_t;
+typedef HMTX pthread_mutex_t;
 typedef void pthread_mutexattr_t;
 
-#define PTHREAD_MUTEX_INITIALIZER _FMUTEX_INITIALIZER
-
 typedef struct {
     HEV event_sem;
     HEV ack_sem;
@@ -100,28 +98,28 @@ static av_always_inline int pthread_join(pthread_t thread, void **value_ptr)
 static av_always_inline int pthread_mutex_init(pthread_mutex_t *mutex,
                                                const pthread_mutexattr_t *attr)
 {
-    _fmutex_create(mutex, 0);
+    DosCreateMutexSem(NULL, (PHMTX)mutex, 0, FALSE);
 
     return 0;
 }
 
 static av_always_inline int pthread_mutex_destroy(pthread_mutex_t *mutex)
 {
-    _fmutex_close(mutex);
+    DosCloseMutexSem(*(PHMTX)mutex);
 
     return 0;
 }
 
 static av_always_inline int pthread_mutex_lock(pthread_mutex_t *mutex)
 {
-    _fmutex_request(mutex, 0);
+    DosRequestMutexSem(*(PHMTX)mutex, SEM_INDEFINITE_WAIT);
 
     return 0;
 }
 
 static av_always_inline int pthread_mutex_unlock(pthread_mutex_t *mutex)
 {
-    _fmutex_release(mutex);
+    DosReleaseMutexSem(*(PHMTX)mutex);
 
     return 0;
 }

compat/tms470/math.h

@@ -16,21 +16,15 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#ifndef AVUTIL_HWCONTEXT_MEDIACODEC_H
-#define AVUTIL_HWCONTEXT_MEDIACODEC_H
+#ifndef COMPAT_TMS470_MATH_H
+#define COMPAT_TMS470_MATH_H
 
-/**
- * MediaCodec details.
- *
- * Allocated as AVHWDeviceContext.hwctx
- */
-typedef struct AVMediaCodecDeviceContext {
-    /**
-     * android/view/Surface handle, to be filled by the user.
-     *
-     * This is the default surface used by decoders on this device.
-     */
-    void *surface;
-} AVMediaCodecDeviceContext;
+#include_next <math.h>
 
-#endif /* AVUTIL_HWCONTEXT_MEDIACODEC_H */
+#undef INFINITY
+#undef NAN
+
+#define INFINITY (*(const float*)((const unsigned []){ 0x7f800000 }))
+#define NAN      (*(const float*)((const unsigned []){ 0x7fc00000 }))
+
+#endif /* COMPAT_TMS470_MATH_H */

compat/w32dlfcn.h

@@ -21,7 +21,6 @@
 
 #ifdef _WIN32
 #include <windows.h>
-#include "config.h"
 #if (_WIN32_WINNT < 0x0602) || HAVE_WINRT
 #include "libavutil/wchar_filename.h"
 #endif

compat/w32pthreads.h

@@ -39,6 +39,11 @@
 #include <windows.h>
 #include <process.h>
 
+#if _WIN32_WINNT < 0x0600 && defined(__MINGW32__)
+#undef MemoryBarrier
+#define MemoryBarrier __sync_synchronize
+#endif
+
 #include "libavutil/attributes.h"
 #include "libavutil/common.h"
 #include "libavutil/internal.h"
@@ -51,15 +56,24 @@ typedef struct pthread_t {
     void *ret;
 } pthread_t;
 
-/* use light weight mutex/condition variable API for Windows Vista and later */
-typedef SRWLOCK pthread_mutex_t;
+/* the conditional variable api for windows 6.0+ uses critical sections and
+ * not mutexes */
+typedef CRITICAL_SECTION pthread_mutex_t;
+
+/* This is the CONDITION_VARIABLE typedef for using Windows' native
+ * conditional variables on kernels 6.0+. */
+#if HAVE_CONDITION_VARIABLE_PTR
 typedef CONDITION_VARIABLE pthread_cond_t;
+#else
+typedef struct pthread_cond_t {
+    void *Ptr;
+} pthread_cond_t;
+#endif
 
-#define PTHREAD_MUTEX_INITIALIZER SRWLOCK_INIT
-#define PTHREAD_COND_INITIALIZER CONDITION_VARIABLE_INIT
-
+#if _WIN32_WINNT >= 0x0600
 #define InitializeCriticalSection(x) InitializeCriticalSectionEx(x, 0, 0)
 #define WaitForSingleObject(a, b) WaitForSingleObjectEx(a, b, FALSE)
+#endif
 
 static av_unused unsigned __stdcall attribute_align_arg win32thread_worker(void *arg)
 {
@@ -100,25 +114,26 @@ static av_unused int pthread_join(pthread_t thread, void **value_ptr)
 
 static inline int pthread_mutex_init(pthread_mutex_t *m, void* attr)
 {
-    InitializeSRWLock(m);
+    InitializeCriticalSection(m);
     return 0;
 }
 static inline int pthread_mutex_destroy(pthread_mutex_t *m)
 {
-    /* Unlocked SWR locks use no resources */
+    DeleteCriticalSection(m);
     return 0;
 }
 static inline int pthread_mutex_lock(pthread_mutex_t *m)
 {
-    AcquireSRWLockExclusive(m);
+    EnterCriticalSection(m);
     return 0;
 }
 static inline int pthread_mutex_unlock(pthread_mutex_t *m)
 {
-    ReleaseSRWLockExclusive(m);
+    LeaveCriticalSection(m);
     return 0;
 }
 
+#if _WIN32_WINNT >= 0x0600
 typedef INIT_ONCE pthread_once_t;
 #define PTHREAD_ONCE_INIT INIT_ONCE_STATIC_INIT
 
@@ -152,7 +167,7 @@ static inline int pthread_cond_broadcast(pthread_cond_t *cond)
 
 static inline int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
 {
-    SleepConditionVariableSRW(cond, mutex, INFINITE, 0);
+    SleepConditionVariableCS(cond, mutex, INFINITE);
     return 0;
 }
 
@@ -162,4 +177,242 @@ static inline int pthread_cond_signal(pthread_cond_t *cond)
     return 0;
 }
 
+#else // _WIN32_WINNT < 0x0600
+
+/* atomic init state of dynamically loaded functions */
+static LONG w32thread_init_state = 0;
+static av_unused void w32thread_init(void);
+
+/* for pre-Windows 6.0 platforms, define INIT_ONCE struct,
+ * compatible to the one used in the native API */
+
+typedef union pthread_once_t  {
+    void * Ptr;    ///< For the Windows 6.0+ native functions
+    LONG state;    ///< For the pre-Windows 6.0 compat code
+} pthread_once_t;
+
+#define PTHREAD_ONCE_INIT {0}
+
+/* function pointers to init once API on windows 6.0+ kernels */
+static BOOL (WINAPI *initonce_begin)(pthread_once_t *lpInitOnce, DWORD dwFlags, BOOL *fPending, void **lpContext);
+static BOOL (WINAPI *initonce_complete)(pthread_once_t *lpInitOnce, DWORD dwFlags, void *lpContext);
+
+/* pre-Windows 6.0 compat using a spin-lock */
+static inline void w32thread_once_fallback(LONG volatile *state, void (*init_routine)(void))
+{
+    switch (InterlockedCompareExchange(state, 1, 0)) {
+    /* Initial run */
+    case 0:
+        init_routine();
+        InterlockedExchange(state, 2);
+        break;
+    /* Another thread is running init */
+    case 1:
+        while (1) {
+            MemoryBarrier();
+            if (*state == 2)
+                break;
+            Sleep(0);
+        }
+        break;
+    /* Initialization complete */
+    case 2:
+        break;
+    }
+}
+
+static av_unused int pthread_once(pthread_once_t *once_control, void (*init_routine)(void))
+{
+    w32thread_once_fallback(&w32thread_init_state, w32thread_init);
+
+    /* Use native functions on Windows 6.0+ */
+    if (initonce_begin && initonce_complete) {
+        BOOL pending = FALSE;
+        initonce_begin(once_control, 0, &pending, NULL);
+        if (pending)
+            init_routine();
+        initonce_complete(once_control, 0, NULL);
+        return 0;
+    }
+
+    w32thread_once_fallback(&once_control->state, init_routine);
+    return 0;
+}
+
+/* for pre-Windows 6.0 platforms we need to define and use our own condition
+ * variable and api */
+
+typedef struct  win32_cond_t {
+    pthread_mutex_t mtx_broadcast;
+    pthread_mutex_t mtx_waiter_count;
+    volatile int waiter_count;
+    HANDLE semaphore;
+    HANDLE waiters_done;
+    volatile int is_broadcast;
+} win32_cond_t;
+
+/* function pointers to conditional variable API on windows 6.0+ kernels */
+static void (WINAPI *cond_broadcast)(pthread_cond_t *cond);
+static void (WINAPI *cond_init)(pthread_cond_t *cond);
+static void (WINAPI *cond_signal)(pthread_cond_t *cond);
+static BOOL (WINAPI *cond_wait)(pthread_cond_t *cond, pthread_mutex_t *mutex,
+                                DWORD milliseconds);
+
+static av_unused int pthread_cond_init(pthread_cond_t *cond, const void *unused_attr)
+{
+    win32_cond_t *win32_cond = NULL;
+
+    w32thread_once_fallback(&w32thread_init_state, w32thread_init);
+
+    if (cond_init) {
+        cond_init(cond);
+        return 0;
+    }
+
+    /* non native condition variables */
+    win32_cond = (win32_cond_t*)av_mallocz(sizeof(win32_cond_t));
+    if (!win32_cond)
+        return ENOMEM;
+    cond->Ptr = win32_cond;
+    win32_cond->semaphore = CreateSemaphore(NULL, 0, 0x7fffffff, NULL);
+    if (!win32_cond->semaphore)
+        return ENOMEM;
+    win32_cond->waiters_done = CreateEvent(NULL, TRUE, FALSE, NULL);
+    if (!win32_cond->waiters_done)
+        return ENOMEM;
+
+    pthread_mutex_init(&win32_cond->mtx_waiter_count, NULL);
+    pthread_mutex_init(&win32_cond->mtx_broadcast, NULL);
+    return 0;
+}
+
+static av_unused int pthread_cond_destroy(pthread_cond_t *cond)
+{
+    win32_cond_t *win32_cond = (win32_cond_t*)cond->Ptr;
+    /* native condition variables do not destroy */
+    if (cond_init)
+        return 0;
+
+    /* non native condition variables */
+    CloseHandle(win32_cond->semaphore);
+    CloseHandle(win32_cond->waiters_done);
+    pthread_mutex_destroy(&win32_cond->mtx_waiter_count);
+    pthread_mutex_destroy(&win32_cond->mtx_broadcast);
+    av_freep(&win32_cond);
+    cond->Ptr = NULL;
+    return 0;
+}
+
+static av_unused int pthread_cond_broadcast(pthread_cond_t *cond)
+{
+    win32_cond_t *win32_cond = (win32_cond_t*)cond->Ptr;
+    int have_waiter;
+
+    if (cond_broadcast) {
+        cond_broadcast(cond);
+        return 0;
+    }
+
+    /* non native condition variables */
+    pthread_mutex_lock(&win32_cond->mtx_broadcast);
+    pthread_mutex_lock(&win32_cond->mtx_waiter_count);
+    have_waiter = 0;
+
+    if (win32_cond->waiter_count) {
+        win32_cond->is_broadcast = 1;
+        have_waiter = 1;
+    }
+
+    if (have_waiter) {
+        ReleaseSemaphore(win32_cond->semaphore, win32_cond->waiter_count, NULL);
+        pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
+        WaitForSingleObject(win32_cond->waiters_done, INFINITE);
+        ResetEvent(win32_cond->waiters_done);
+        win32_cond->is_broadcast = 0;
+    } else
+        pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
+    pthread_mutex_unlock(&win32_cond->mtx_broadcast);
+    return 0;
+}
+
+static av_unused int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
+{
+    win32_cond_t *win32_cond = (win32_cond_t*)cond->Ptr;
+    int last_waiter;
+    if (cond_wait) {
+        cond_wait(cond, mutex, INFINITE);
+        return 0;
+    }
+
+    /* non native condition variables */
+    pthread_mutex_lock(&win32_cond->mtx_broadcast);
+    pthread_mutex_lock(&win32_cond->mtx_waiter_count);
+    win32_cond->waiter_count++;
+    pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
+    pthread_mutex_unlock(&win32_cond->mtx_broadcast);
+
+    // unlock the external mutex
+    pthread_mutex_unlock(mutex);
+    WaitForSingleObject(win32_cond->semaphore, INFINITE);
+
+    pthread_mutex_lock(&win32_cond->mtx_waiter_count);
+    win32_cond->waiter_count--;
+    last_waiter = !win32_cond->waiter_count || !win32_cond->is_broadcast;
+    pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
+
+    if (last_waiter)
+        SetEvent(win32_cond->waiters_done);
+
+    // lock the external mutex
+    return pthread_mutex_lock(mutex);
+}
+
+static av_unused int pthread_cond_signal(pthread_cond_t *cond)
+{
+    win32_cond_t *win32_cond = (win32_cond_t*)cond->Ptr;
+    int have_waiter;
+    if (cond_signal) {
+        cond_signal(cond);
+        return 0;
+    }
+
+    pthread_mutex_lock(&win32_cond->mtx_broadcast);
+
+    /* non-native condition variables */
+    pthread_mutex_lock(&win32_cond->mtx_waiter_count);
+    have_waiter = win32_cond->waiter_count;
+    pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
+
+    if (have_waiter) {
+        ReleaseSemaphore(win32_cond->semaphore, 1, NULL);
+        WaitForSingleObject(win32_cond->waiters_done, INFINITE);
+        ResetEvent(win32_cond->waiters_done);
+    }
+
+    pthread_mutex_unlock(&win32_cond->mtx_broadcast);
+    return 0;
+}
+#endif
+
+static av_unused void w32thread_init(void)
+{
+#if _WIN32_WINNT < 0x0600
+    HMODULE kernel_dll = GetModuleHandle(TEXT("kernel32.dll"));
+    /* if one is available, then they should all be available */
+    cond_init      = (void (WINAPI*)(pthread_cond_t *))
+        GetProcAddress(kernel_dll, "InitializeConditionVariable");
+    cond_broadcast = (void (WINAPI*)(pthread_cond_t *))
+        GetProcAddress(kernel_dll, "WakeAllConditionVariable");
+    cond_signal    = (void (WINAPI*)(pthread_cond_t *))
+        GetProcAddress(kernel_dll, "WakeConditionVariable");
+    cond_wait      = (BOOL (WINAPI*)(pthread_cond_t *, pthread_mutex_t *, DWORD))
+        GetProcAddress(kernel_dll, "SleepConditionVariableCS");
+    initonce_begin = (BOOL (WINAPI*)(pthread_once_t *, DWORD, BOOL *, void **))
+        GetProcAddress(kernel_dll, "InitOnceBeginInitialize");
+    initonce_complete = (BOOL (WINAPI*)(pthread_once_t *, DWORD, void *))
+        GetProcAddress(kernel_dll, "InitOnceComplete");
+#endif
+
+}
+
 #endif /* COMPAT_W32PTHREADS_H */

compat/windows/makedef

@@ -45,11 +45,7 @@ libname=$(mktemp -u "library").lib
 
 trap 'rm -f -- $libname' EXIT
 
-if [ -n "$AR" ]; then
-    $AR rcs ${libname} $@ >/dev/null
-else
-    lib.exe -out:${libname} $@ >/dev/null
-fi
+lib -out:${libname} $@ >/dev/null
 if [ $? != 0 ]; then
     echo "Could not create temporary library." >&2
     exit 1
@@ -58,7 +54,23 @@ fi
 IFS='
 '
 
-prefix="$EXTERN_PREFIX"
+# Determine if we're building for x86 or x86_64 and
+# set the symbol prefix accordingly.
+prefix=""
+arch=$(dumpbin -headers ${libname} |
+       tr '\t' ' ' |
+       grep '^ \+.\+machine \+(.\+)' |
+       head -1 |
+       sed -e 's/^ \{1,\}.\{1,\} \{1,\}machine \{1,\}(\(...\)).*/\1/')
+
+if [ "${arch}" = "x86" ]; then
+    prefix="_"
+else
+    if [ "${arch}" != "ARM" ] && [ "${arch}" != "x64" ]; then
+        echo "Unknown machine type." >&2
+        exit 1
+    fi
+fi
 
 started=0
 regex="none"
@@ -100,19 +112,7 @@ for line in $(cat ${vscript} | tr '\t' ' '); do
 '
 done
 
-if [ -n "$NM" ]; then
-    # Use eval, since NM="nm -g"
-    dump=$(eval "$NM --defined-only -g ${libname}" |
-              grep -v : |
-              grep -v ^$ |
-              cut -d' ' -f3 |
-              sed -e "s/^${prefix}//")
-else
-    dump=$(dumpbin.exe -linkermember:1 ${libname} |
-              sed -e '/public symbols/,$!d' -e '/^ \{1,\}Summary/,$d' -e "s/ \{1,\}${prefix}/ /" -e 's/ \{1,\}/ /g' |
-              tail -n +2 |
-              cut -d' ' -f3)
-fi
+dump=$(dumpbin -linkermember:1 ${libname})
 
 rm ${libname}
 
@@ -121,6 +121,9 @@ list=""
 for exp in ${regex}; do
     list="${list}"'
 '$(echo "${dump}" |
+          sed -e '/public symbols/,$!d' -e '/^ \{1,\}Summary/,$d' -e "s/ \{1,\}${prefix}/ /" -e 's/ \{1,\}/ /g' |
+          tail -n +2 |
+          cut -d' ' -f3 |
           grep "^${exp}" |
           sed -e 's/^/    /')
 done

compat/windows/mslink

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

doc/APIchanges

@@ -2,229 +2,19 @@ Never assume the API of libav* to be stable unless at least 1 month has passed
 since the last major version increase or the API was added.
 
 The last version increases were:
-libavcodec:    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
+libavcodec:    2015-08-28
+libavdevice:   2015-08-28
+libavfilter:   2015-08-28
+libavformat:   2015-08-28
+libavresample: 2015-08-28
+libpostproc:   2015-08-28
+libswresample: 2015-08-28
+libswscale:    2015-08-28
+libavutil:     2015-08-28
 
 
 API changes, most recent first:
 
--------- 8< --------- FFmpeg 4.2 was cut here -------- 8< ---------
-
-2019-06-21 - a30e44098a - lavu 56.30.100 - frame.h
-  Add FF_DECODE_ERROR_DECODE_SLICES
-
-2019-06-14 - edfced8c04 - lavu 56.29.100 - frame.h
-  Add FF_DECODE_ERROR_CONCEALMENT_ACTIVE
-
-2019-05-15 - b79b29ddb1 - lavu 56.28.100 - tx.h
-  Add av_tx_init(), av_tx_uninit() and related definitions.
-
-2019-04-20 - 3153a6502a - lavc 58.52.100 - avcodec.h
-  Add AV_CODEC_FLAG_DROPCHANGED to allow avcodec_receive_frame to drop
-  frames whose parameters differ from first decoded frame in stream.
-
-2019-04-12 - abfeba9724 - lavf 58.27.102
-  Rename hls,applehttp demuxer to hls
-
-2019-01-27 - 5bcefceec8 - lavc 58.46.100 - avcodec.h
-  Add discard_damaged_percentage
-
-2019-01-08 - 1ef4828276 - lavu 56.26.100 - frame.h
-  Add AV_FRAME_DATA_REGIONS_OF_INTEREST
-
-2018-12-21 - 2744d6b364 - lavu 56.25.100 - hdr_dynamic_metadata.h
-  Add AV_FRAME_DATA_DYNAMIC_HDR_PLUS enum value, av_dynamic_hdr_plus_alloc(),
-  av_dynamic_hdr_plus_create_side_data() functions, and related structs.
-
--------- 8< --------- FFmpeg 4.1 was cut here -------- 8< ---------
-
-2018-10-27 - 718044dc19 - lavu 56.21.100 - pixdesc.h
-  Add av_read_image_line2(), av_write_image_line2()
-
-2018-10-24 - f9d4126f28 - lavu 56.20.100 - frame.h
-  Add AV_FRAME_DATA_S12M_TIMECODE
-
-2018-10-11 - f6d48b618a - lavc 58.33.100 - mediacodec.h
-  Add av_mediacodec_render_buffer_at_time().
-
-2018-09-09 - 35498c124a - lavc 58.29.100 - avcodec.h
-  Add AV_PKT_DATA_AFD
-
-2018-08-16 - b33f5299a5 - lavc 58.23.100 - avcodec.h
-  Add av_bsf_flush().
-
-2018-05-18 - 2b2f2f65f3 - lavf 58.15.100 - avformat.h
-  Add pmt_version field to AVProgram
-
-2018-05-17 - 5dfeb7f081 - lavf 58.14.100 - avformat.h
-  Add AV_DISPOSITION_STILL_IMAGE
-
-2018-05-10 - c855683427 - lavu 56.18.101 - hwcontext_cuda.h
-  Add AVCUDADeviceContext.stream.
-
-2018-04-30 - 56b081da57 - lavu 56.18.100 - pixdesc.h
-  Add AV_PIX_FMT_FLAG_ALPHA to AV_PIX_FMT_PAL8.
-
-2018-04-26 - 5be0410cb3 - lavu 56.17.100 - opt.h
-  Add AV_OPT_FLAG_DEPRECATED.
-
-2018-04-26 - 71fa82bed6 - lavu 56.16.100 - threadmessage.h
-  Add av_thread_message_queue_nb_elems().
-
--------- 8< --------- FFmpeg 4.0 was cut here -------- 8< ---------
-
-2018-04-03 - d6fc031caf - lavu 56.13.100 - pixdesc.h
-  Deprecate AV_PIX_FMT_FLAG_PSEUDOPAL and make allocating a pseudo palette
-  optional for API users (see AV_PIX_FMT_FLAG_PSEUDOPAL doxygen for details).
-
-2018-04-01 - 860086ee16 - lavc 58.17.100 - avcodec.h
-  Add av_packet_make_refcounted().
-
-2018-04-01 - f1805d160d - lavfi 7.14.100 - avfilter.h
-  Deprecate use of avfilter_register(), avfilter_register_all(),
-  avfilter_next(). Add av_filter_iterate().
-
-2018-03-25 - b7d0d912ef - lavc 58.16.100 - avcodec.h
-  Add FF_SUB_CHARENC_MODE_IGNORE.
-
-2018-03-23 - db2a7c947e - lavu 56.12.100 - encryption_info.h
-  Add AVEncryptionInitInfo and AVEncryptionInfo structures to hold new side-data
-  for encryption info.
-
-2018-03-21 - f14ca60001 - lavc 58.15.100 - avcodec.h
-  Add av_packet_make_writable().
-
-2018-03-18 - 4b86ac27a0 - lavu 56.11.100 - frame.h
-  Add AV_FRAME_DATA_QP_TABLE_PROPERTIES and AV_FRAME_DATA_QP_TABLE_DATA.
-
-2018-03-15 - e0e72539cf - lavu 56.10.100 - opt.h
-  Add AV_OPT_FLAG_BSF_PARAM
-
-2018-03-07 - 950170bd3b - lavu 56.9.100 - crc.h
-  Add AV_CRC_8_EBU crc variant.
-
-2018-03-07 - 2a0eb86857 - lavc 58.14.100 - mediacodec.h
-  Change the default behavior of avcodec_flush() on mediacodec
-  video decoders. To restore the previous behavior, use the new
-  delay_flush=1 option.
-
-2018-03-01 - 6731f60598 - lavu 56.8.100 - frame.h
-  Add av_frame_new_side_data_from_buf().
-
-2018-02-15 - 8a8d0b319a
-  Change av_ripemd_update(), av_murmur3_update() and av_hash_update() length
-  parameter type to size_t at next major bump.
-
-2018-02-12 - bcab11a1a2 - lavfi 7.12.100 - avfilter.h
-  Add AVFilterContext.extra_hw_frames.
-
-2018-02-12 - d23fff0d8a - lavc 58.11.100 - avcodec.h
-  Add AVCodecContext.extra_hw_frames.
-
-2018-02-06 - 0694d87024 - lavf 58.9.100 - avformat.h
-  Deprecate use of av_register_input_format(), av_register_output_format(),
-  av_register_all(), av_iformat_next(), av_oformat_next().
-  Add av_demuxer_iterate(), and av_muxer_iterate().
-
-2018-02-06 - 36c85d6e77 - lavc 58.10.100 - avcodec.h
-  Deprecate use of avcodec_register(), avcodec_register_all(),
-  av_codec_next(), av_register_codec_parser(), and av_parser_next().
-  Add av_codec_iterate() and av_parser_iterate().
-
-2018-02-04 - ff46124b0d - lavf 58.8.100 - avformat.h
-  Deprecate the current names of the RTSP "timeout", "stimeout", "user-agent"
-  options. Introduce "listen_timeout" as replacement for the current "timeout"
-  option, and "user_agent" as replacement for "user-agent". Once the deprecation
-  is over, the old "timeout" option will be removed, and "stimeout" will be
-  renamed to "stimeout" (the "timeout" option will essentially change semantics).
-
-2018-01-28 - ea3672b7d6 - lavf 58.7.100 - avformat.h
-  Deprecate AVFormatContext filename field which had limited length, use the
-  new dynamically allocated url field instead.
-
-2018-01-28 - ea3672b7d6 - lavf 58.7.100 - avformat.h
-  Add url field to AVFormatContext and add ff_format_set_url helper function.
-
-2018-01-27 - 6194d7e564 - lavf 58.6.100 - avformat.h
-  Add AVFMTCTX_UNSEEKABLE (for HLS demuxer).
-
-2018-01-23 - 9f07cf7c00 - lavu 56.9.100 - aes_ctr.h
-  Add method to set the 16-byte IV.
-
-2018-01-16 - 631c56a8e4 - lavf 58.5.100 - avformat.h
-  Explicitly make avformat_network_init() and avformat_network_deinit() optional.
-  If these are not called, network initialization and deinitialization is
-  automatic, and unlike in older versions, fully supported, unless libavformat
-  is linked to ancient GnuTLS and OpenSSL.
-
-2018-01-16 - 6512ff72f9 - lavf 58.4.100 - avformat.h
-  Deprecate AVStream.recommended_encoder_configuration. It was useful only for
-  FFserver, which has been removed.
-
-2018-01-05 - 798dcf2432 - lavfi 7.11.101 - avfilter.h
-  Deprecate avfilter_link_get_channels(). Use av_buffersink_get_channels().
-
-2017-01-04 - c29038f304 - lavr 4.0.0 - avresample.h
-  Deprecate the entire library. Merged years ago to provide compatibility
-  with Libav, it remained unmaintained by the FFmpeg project and duplicated
-  functionality provided by libswresample.
-
-  In order to improve consistency and reduce attack surface, it has been deprecated.
-  Users of this library are asked to migrate to libswresample, which, as well as
-  providing more functionality, is faster and has higher accuracy.
-
-2017-12-26 - a04c2c707d - lavc 58.9.100 - avcodec.h
-  Deprecate av_lockmgr_register(). You need to build FFmpeg with threading
-  support enabled to get basic thread-safety (which is the default build
-  configuration).
-
-2017-12-24 - 8b81eabe57 - lavu 56.7.100 - cpu.h
-  AVX-512 flags added.
-
-2017-12-16 - 8bf4e6d3ce - lavc 58.8.100 - avcodec.h
-  The MediaCodec decoders now support AVCodecContext.hw_device_ctx.
-
-2017-12-16 - e4d9f05ca7 - lavu 56.6.100 - hwcontext.h hwcontext_mediacodec.h
-  Add AV_HWDEVICE_TYPE_MEDIACODEC and a new installed header with
-  MediaCodec-specific hwcontext definitions.
-
-2017-12-14 - b945fed629 - lavc 58.7.100 - avcodec.h
-  Add AV_CODEC_CAP_HARDWARE, AV_CODEC_CAP_HYBRID, and AVCodec.wrapper_name,
-  and mark all AVCodecs accordingly.
-
-2017-11-29 - d268094f88 - lavu 56.4.100 / 56.7.0 - stereo3d.h
-  Add view field to AVStereo3D structure and AVStereo3DView enum.
-
-2017-11-26 - 3a71bcc213 - lavc 58.6.100 - avcodec.h
-  Add const to AVCodecContext.hwaccel.
-
-2017-11-26 - 3536a3efb9 - lavc 58.5.100 - avcodec.h
-  Deprecate user visibility of the AVHWAccel structure and the functions
-  av_register_hwaccel() and av_hwaccel_next().
-
-2017-11-26 - 24cc0a53e9 - lavc 58.4.100 - avcodec.h
-  Add AVCodecHWConfig and avcodec_get_hw_config().
-
-2017-11-22 - 3650cb2dfa - lavu 56.3.100 - opencl.h
-  Remove experimental OpenCL API (av_opencl_*).
-
-2017-11-22 - b25d8ef0a7 - lavu 56.2.100 - hwcontext.h hwcontext_opencl.h
-  Add AV_HWDEVICE_TYPE_OPENCL and a new installed header with
-  OpenCL-specific hwcontext definitions.
-
-2017-11-22 - a050f56c09 - lavu 56.1.100 - pixfmt.h
-  Add AV_PIX_FMT_OPENCL.
-
-2017-11-11 - 48e4eda11d - lavc 58.3.100 - avcodec.h
-  Add avcodec_get_hw_frames_parameters().
-
 -------- 8< --------- FFmpeg 3.4 was cut here -------- 8< ---------
 
 2017-09-28 - b6cf66ae1c - lavc 57.106.104 - avcodec.h
@@ -952,7 +742,7 @@ API changes, most recent first:
   Add av_opt_get_dict_val/set_dict_val with AV_OPT_TYPE_DICT to support
   dictionary types being set as options.
 
-2014-08-13 - afbd4b7e09 - lavf 56.01.0 - avformat.h
+2014-08-13 - afbd4b8 - lavf 56.01.0 - avformat.h
   Add AVFormatContext.event_flags and AVStream.event_flags for signaling to
   the user when events happen in the file/stream.
 
@@ -969,7 +759,7 @@ API changes, most recent first:
 2014-08-08 - 5c3c671 - lavf 55.53.100 - avio.h
   Add avio_feof() and deprecate url_feof().
 
-2014-08-07 - bb789016d4 - lsws 2.1.3 - swscale.h
+2014-08-07 - bb78903 - lsws 2.1.3 - swscale.h
   sws_getContext is not going to be removed in the future.
 
 2014-08-07 - a561662 / ad1ee5f - lavc 55.73.101 / 55.57.3 - avcodec.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         = 4.2
+PROJECT_NUMBER         = 3.4.1
 
 # Using the PROJECT_BRIEF tag one can provide an optional one line description
 # for a project that appears at the top of each page and should give viewer a

doc/bitstream_filters.texi

@@ -37,61 +37,6 @@ raw ADTS AAC or an MPEG-TS container to MP4A-LATM, to an FLV file, or
 to MOV/MP4 files and related formats such as 3GP or M4A. Please note
 that it is auto-inserted for MP4A-LATM and MOV/MP4 and related formats.
 
-@section av1_metadata
-
-Modify metadata embedded in an AV1 stream.
-
-@table @option
-@item td
-Insert or remove temporal delimiter OBUs in all temporal units of the
-stream.
-
-@table @samp
-@item insert
-Insert a TD at the beginning of every TU which does not already have one.
-@item remove
-Remove the TD from the beginning of every TU which has one.
-@end table
-
-@item color_primaries
-@item transfer_characteristics
-@item matrix_coefficients
-Set the color description fields in the stream (see AV1 section 6.4.2).
-
-@item color_range
-Set the color range in the stream (see AV1 section 6.4.2; note that
-this cannot be set for streams using BT.709 primaries, sRGB transfer
-characteristic and identity (RGB) matrix coefficients).
-@table @samp
-@item tv
-Limited range.
-@item pc
-Full range.
-@end table
-
-@item chroma_sample_position
-Set the chroma sample location in the stream (see AV1 section 6.4.2).
-This can only be set for 4:2:0 streams.
-
-@table @samp
-@item vertical
-Left position (matching the default in MPEG-2 and H.264).
-@item colocated
-Top-left position.
-@end table
-
-@item tick_rate
-Set the tick rate (@emph{num_units_in_display_tick / time_scale}) in
-the timing info in the sequence header.
-@item num_ticks_per_picture
-Set the number of ticks in each picture, to indicate that the stream
-has a fixed framerate.  Ignored if @option{tick_rate} is not also set.
-
-@item delete_padding
-Deletes Padding OBUs.
-
-@end table
-
 @section chomp
 
 Remove zero padding at the end of a packet.
@@ -103,24 +48,21 @@ DTS-HD.
 
 @section dump_extra
 
-Add extradata to the beginning of the filtered packets except when
-said packets already exactly begin with the extradata that is intended
-to be added.
+Add extradata to the beginning of the filtered packets.
 
-@table @option
-@item freq
 The additional argument specifies which packets should be filtered.
 It accepts the values:
 @table @samp
+@item a
+add extradata to all key packets, but only if @var{local_header} is
+set in the @option{flags2} codec context field
+
 @item k
-@item keyframe
 add extradata to all key packets
 
 @item e
-@item all
 add extradata to all packets
 @end table
-@end table
 
 If not specified it is assumed @samp{k}.
 
@@ -132,10 +74,6 @@ 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 eac3_core
-
-Extract the core from a E-AC-3 stream, dropping extra channels.
-
 @section extract_extradata
 
 Extract the in-band extradata.
@@ -154,135 +92,6 @@ When this option is enabled, the long-term headers are removed from the
 bitstream after extraction.
 @end table
 
-@section filter_units
-
-Remove units with types in or not in a given set from the stream.
-
-@table @option
-@item pass_types
-List of unit types or ranges of unit types to pass through while removing
-all others.  This is specified as a '|'-separated list of unit type values
-or ranges of values with '-'.
-
-@item remove_types
-Identical to @option{pass_types}, except the units in the given set
-removed and all others passed through.
-@end table
-
-Extradata is unchanged by this transformation, but note that if the stream
-contains inline parameter sets then the output may be unusable if they are
-removed.
-
-For example, to remove all non-VCL NAL units from an H.264 stream:
-@example
-ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=pass_types=1-5' OUTPUT
-@end example
-
-To remove all AUDs, SEI and filler from an H.265 stream:
-@example
-ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=remove_types=35|38-40' OUTPUT
-@end example
-
-@section hapqa_extract
-
-Extract Rgb or Alpha part of an HAPQA file, without recompression, in order to create an HAPQ or an HAPAlphaOnly file.
-
-@table @option
-@item texture
-Specifies the texture to keep.
-
-@table @option
-@item color
-@item alpha
-@end table
-
-@end table
-
-Convert HAPQA to HAPQ
-@example
-ffmpeg -i hapqa_inputfile.mov -c copy -bsf:v hapqa_extract=texture=color -tag:v HapY -metadata:s:v:0 encoder="HAPQ" hapq_file.mov
-@end example
-
-Convert HAPQA to HAPAlphaOnly
-@example
-ffmpeg -i hapqa_inputfile.mov -c copy -bsf:v hapqa_extract=texture=alpha -tag:v HapA -metadata:s:v:0 encoder="HAPAlpha Only" hapalphaonly_file.mov
-@end example
-
-@section h264_metadata
-
-Modify metadata embedded in an H.264 stream.
-
-@table @option
-@item aud
-Insert or remove AUD NAL units in all access units of the stream.
-
-@table @samp
-@item insert
-@item remove
-@end table
-
-@item sample_aspect_ratio
-Set the sample aspect ratio of the stream in the VUI parameters.
-
-@item video_format
-@item video_full_range_flag
-Set the video format in the stream (see H.264 section E.2.1 and
-table E-2).
-
-@item colour_primaries
-@item transfer_characteristics
-@item matrix_coefficients
-Set the colour description in the stream (see H.264 section E.2.1
-and tables E-3, E-4 and E-5).
-
-@item chroma_sample_loc_type
-Set the chroma sample location in the stream (see H.264 section
-E.2.1 and figure E-1).
-
-@item tick_rate
-Set the tick rate (num_units_in_tick / time_scale) in the VUI
-parameters.  This is the smallest time unit representable in the
-stream, and in many cases represents the field rate of the stream
-(double the frame rate).
-@item fixed_frame_rate_flag
-Set whether the stream has fixed framerate - typically this indicates
-that the framerate is exactly half the tick rate, but the exact
-meaning is dependent on interlacing and the picture structure (see
-H.264 section E.2.1 and table E-6).
-
-@item crop_left
-@item crop_right
-@item crop_top
-@item crop_bottom
-Set the frame cropping offsets in the SPS.  These values will replace
-the current ones if the stream is already cropped.
-
-These fields are set in pixels.  Note that some sizes may not be
-representable if the chroma is subsampled or the stream is interlaced
-(see H.264 section 7.4.2.1.1).
-
-@item sei_user_data
-Insert a string as SEI unregistered user data.  The argument must
-be of the form @emph{UUID+string}, where the UUID is as hex digits
-possibly separated by hyphens, and the string can be anything.
-
-For example, @samp{086f3693-b7b3-4f2c-9653-21492feee5b8+hello} will
-insert the string ``hello'' associated with the given UUID.
-
-@item delete_filler
-Deletes both filler NAL units and filler SEI messages.
-
-@item level
-Set the level in the SPS.  Refer to H.264 section A.3 and tables A-1
-to A-5.
-
-The argument must be the name of a level (for example, @samp{4.2}), a
-level_idc value (for example, @samp{42}), or the special name @samp{auto}
-indicating that the filter should attempt to guess the level from the
-input stream properties.
-
-@end table
-
 @section h264_mp4toannexb
 
 Convert an H.264 bitstream from length prefixed mode to start code
@@ -302,78 +111,6 @@ ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts
 Please note that this filter is auto-inserted for MPEG-TS (muxer
 @code{mpegts}) and raw H.264 (muxer @code{h264}) output formats.
 
-@section h264_redundant_pps
-
-This applies a specific fixup to some Blu-ray streams which contain
-redundant PPSs modifying irrelevant parameters of the stream which
-confuse other transformations which require correct extradata.
-
-A new single global PPS is created, and all of the redundant PPSs
-within the stream are removed.
-
-@section hevc_metadata
-
-Modify metadata embedded in an HEVC stream.
-
-@table @option
-@item aud
-Insert or remove AUD NAL units in all access units of the stream.
-
-@table @samp
-@item insert
-@item remove
-@end table
-
-@item sample_aspect_ratio
-Set the sample aspect ratio in the stream in the VUI parameters.
-
-@item video_format
-@item video_full_range_flag
-Set the video format in the stream (see H.265 section E.3.1 and
-table E.2).
-
-@item colour_primaries
-@item transfer_characteristics
-@item matrix_coefficients
-Set the colour description in the stream (see H.265 section E.3.1
-and tables E.3, E.4 and E.5).
-
-@item chroma_sample_loc_type
-Set the chroma sample location in the stream (see H.265 section
-E.3.1 and figure E.1).
-
-@item tick_rate
-Set the tick rate in the VPS and VUI parameters (num_units_in_tick /
-time_scale).  Combined with @option{num_ticks_poc_diff_one}, this can
-set a constant framerate in the stream.  Note that it is likely to be
-overridden by container parameters when the stream is in a container.
-
-@item num_ticks_poc_diff_one
-Set poc_proportional_to_timing_flag in VPS and VUI and use this value
-to set num_ticks_poc_diff_one_minus1 (see H.265 sections 7.4.3.1 and
-E.3.1).  Ignored if @option{tick_rate} is not also set.
-
-@item crop_left
-@item crop_right
-@item crop_top
-@item crop_bottom
-Set the conformance window cropping offsets in the SPS.  These values
-will replace the current ones if the stream is already cropped.
-
-These fields are set in pixels.  Note that some sizes may not be
-representable if the chroma is subsampled (H.265 section 7.4.3.2.1).
-
-@item level
-Set the level in the VPS and SPS.  See H.265 section A.4 and tables
-A.6 and A.7.
-
-The argument must be the name of a level (for example, @samp{5.1}), a
-@emph{general_level_idc} value (for example, @samp{153} for level 5.1),
-or the special name @samp{auto} indicating that the filter should
-attempt to guess the level from the input stream properties.
-
-@end table
-
 @section hevc_mp4toannexb
 
 Convert an HEVC/H.265 bitstream from length prefixed mode to start code
@@ -461,42 +198,6 @@ See also the @ref{text2movsub} filter.
 
 Decompress non-standard compressed MP3 audio headers.
 
-@section mpeg2_metadata
-
-Modify metadata embedded in an MPEG-2 stream.
-
-@table @option
-@item display_aspect_ratio
-Set the display aspect ratio in the stream.
-
-The following fixed values are supported:
-@table @option
-@item 4/3
-@item 16/9
-@item 221/100
-@end table
-Any other value will result in square pixels being signalled instead
-(see H.262 section 6.3.3 and table 6-3).
-
-@item frame_rate
-Set the frame rate in the stream.  This is constructed from a table
-of known values combined with a small multiplier and divisor - if
-the supplied value is not exactly representable, the nearest
-representable value will be used instead (see H.262 section 6.3.3
-and table 6-4).
-
-@item video_format
-Set the video format in the stream (see H.262 section 6.3.6 and
-table 6-6).
-
-@item colour_primaries
-@item transfer_characteristics
-@item matrix_coefficients
-Set the colour description in the stream (see H.262 section 6.3.6
-and tables 6-7, 6-8 and 6-9).
-
-@end table
-
 @section mpeg4_unpack_bframes
 
 Unpack DivX-style packed B-frames.
@@ -544,72 +245,6 @@ ffmpeg -i INPUT -c copy -bsf noise[=1] output.mkv
 @section null
 This bitstream filter passes the packets through unchanged.
 
-@section prores_metadata
-
-Modify color property metadata embedded in prores stream.
-
-@table @option
-@item color_primaries
-Set the color primaries.
-Available values are:
-
-@table @samp
-@item auto
-Keep the same color primaries property (default).
-
-@item unknown
-@item bt709
-@item bt470bg
-BT601 625
-
-@item smpte170m
-BT601 525
-
-@item bt2020
-@item smpte431
-DCI P3
-
-@item smpte432
-P3 D65
-
-@end table
-
-@item transfer_characteristics
-Set the color transfer.
-Available values are:
-
-@table @samp
-@item auto
-Keep the same transfer characteristics property (default).
-
-@item unknown
-@item bt709
-BT 601, BT 709, BT 2020
-@end table
-
-
-@item matrix_coefficients
-Set the matrix coefficient.
-Available values are:
-
-@table @samp
-@item auto
-Keep the same transfer characteristics property (default).
-
-@item unknown
-@item bt709
-@item smpte170m
-BT 601
-
-@item bt2020nc
-@end table
-@end table
-
-Set Rec709 colorspace for each frame of the file
-@example
-ffmpeg -i INPUT -c copy -bsf:v prores_metadata=color_primaries=bt709:color_trc=bt709:colorspace=bt709 output.mov
-@end example
-
 @section remove_extra
 
 Remove extradata from packets.
@@ -640,45 +275,6 @@ codec) with metadata headers.
 
 See also the @ref{mov2textsub} filter.
 
-@section trace_headers
-
-Log trace output containing all syntax elements in the coded stream
-headers (everything above the level of individual coded blocks).
-This can be useful for debugging low-level stream issues.
-
-Supports AV1, H.264, H.265, (M)JPEG, MPEG-2 and VP9, but depending
-on the build only a subset of these may be available.
-
-@section truehd_core
-
-Extract the core from a TrueHD stream, dropping ATMOS data.
-
-@section vp9_metadata
-
-Modify metadata embedded in a VP9 stream.
-
-@table @option
-@item color_space
-Set the color space value in the frame header.
-@table @samp
-@item unknown
-@item bt601
-@item bt709
-@item smpte170
-@item smpte240
-@item bt2020
-@item rgb
-@end table
-
-@item color_range
-Set the color range value in the frame header.  Note that this cannot
-be set in RGB streams.
-@table @samp
-@item tv
-@item pc
-@end table
-@end table
-
 @section vp9_superframe
 
 Merge VP9 invisible (alt-ref) frames back into VP9 superframes. This

doc/build_system.txt

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

doc/codecs.texi

@@ -44,6 +44,12 @@ Use 1/4 pel motion compensation.
 Use loop filter.
 @item qscale
 Use fixed qscale.
+@item gmc
+Use gmc.
+@item mv0
+Always try a mb with mv=<0,0>.
+@item input_preserved
+
 @item pass1
 Use internal 2pass ratecontrol in first pass mode.
 @item pass2
@@ -55,10 +61,9 @@ Do not draw edges.
 @item psnr
 Set error[?] variables during encoding.
 @item truncated
-@item drop_changed
-Don't output frames whose parameters differ from first decoded frame in stream.
-Error AVERROR_INPUT_CHANGED is returned when a frame is dropped.
 
+@item naq
+Normalize adaptive quantization.
 @item ildct
 Use interlaced DCT.
 @item low_delay
@@ -470,6 +475,8 @@ rate control
 macroblock (MB) type
 @item qp
 per-block quantization parameter (QP)
+@item mv
+motion vector
 @item dct_coeff
 
 @item green_metadata
@@ -479,12 +486,18 @@ display complexity metadata for the upcoming frame, GoP or for a given duration.
 
 @item startcode
 
+@item pts
+
 @item er
 error recognition
 @item mmco
 memory management control operations (H.264)
 @item bugs
 
+@item vis_qp
+visualize quantization parameter (QP), lower QP are tinted greener
+@item vis_mb_type
+visualize block types
 @item buffers
 picture buffer allocations
 @item thread_ops
@@ -493,6 +506,21 @@ threading operations
 skip motion compensation
 @end table
 
+@item vismv @var{integer} (@emph{decoding,video})
+Visualize motion vectors (MVs).
+
+This option is deprecated, see the codecview filter instead.
+
+Possible values:
+@table @samp
+@item pf
+forward predicted MVs of P-frames
+@item bf
+forward predicted MVs of B-frames
+@item bb
+backward predicted MVs of B-frames
+@end table
+
 @item cmp @var{integer} (@emph{encoding,video})
 Set full pel me compare function.
 
@@ -729,6 +757,8 @@ Set context model.
 
 @item slice_flags @var{integer}
 
+@item xvmc_acceleration @var{integer}
+
 @item mbd @var{integer} (@emph{encoding,video})
 Set macroblock decision algorithm (high quality mode).
 
@@ -778,6 +808,8 @@ Place global headers at every keyframe instead of in extradata.
 Frame data might be split into multiple chunks.
 @item showall
 Show all frames before the first keyframe.
+@item skiprd
+Deprecated, use mpegvideo private options instead.
 @item export_mvs
 Export motion vectors into frame side-data (see @code{AV_FRAME_DATA_MOTION_VECTORS})
 for codecs that support it. See also @file{doc/examples/export_mvs.c}.
@@ -963,9 +995,6 @@ Discard all bidirectional frames.
 @item nokey
 Discard all frames excepts keyframes.
 
-@item nointra
-Discard all frames except I frames.
-
 @item all
 Discard all frames.
 @end table
@@ -990,6 +1019,10 @@ Set chroma qp offset from luma.
 @item trellis @var{integer} (@emph{encoding,audio,video})
 Set rate-distortion optimal quantization.
 
+@item sc_factor @var{integer} (@emph{encoding,video})
+Set value multiplied by qscale for each frame and added to
+scene_change_score.
+
 @item mv0_threshold @var{integer} (@emph{encoding,video})
 @item b_sensitivity @var{integer} (@emph{encoding,video})
 Adjust sensitivity of b_frame_strategy 1.
@@ -1236,7 +1269,7 @@ instead of alpha. Default is 0.
 @item dump_separator @var{string} (@emph{input})
 Separator used to separate the fields printed on the command line about the
 Stream parameters.
-For example, to separate the fields with newlines and indentation:
+For example to separate the fields with newlines and indention:
 @example
 ffprobe -dump_separator "
                           "  -i ~/videos/matrixbench_mpeg2.mpg

doc/decoders.texi

@@ -25,6 +25,13 @@ enabled decoders.
 A description of some of the currently available video decoders
 follows.
 
+@section hevc
+
+HEVC / H.265 decoder.
+
+Note: the @option{skip_loop_filter} option has effect only at level
+@code{all}.
+
 @section rawvideo
 
 Raw video decoder.
@@ -47,38 +54,6 @@ top-field-first is assumed
 
 @end table
 
-@section libdav1d
-
-dav1d AV1 decoder.
-
-libdav1d allows libavcodec to decode the AOMedia Video 1 (AV1) codec.
-Requires the presence of the libdav1d headers and library during configuration.
-You need to explicitly configure the build with @code{--enable-libdav1d}.
-
-@subsection Options
-
-The following option is supported by the libdav1d wrapper.
-
-@table @option
-
-@item framethreads
-Set amount of frame threads to use during decoding. The default value is 0 (autodetect).
-
-@item tilethreads
-Set amount of tile threads to use during decoding. The default value is 0 (autodetect).
-
-@item filmgrain
-Apply film grain to the decoded video if present in the bitstream. The default value
-is true.
-
-@end table
-
-@section libdavs2
-
-AVS2-P2/IEEE1857.4 video decoder wrapper.
-
-This decoder allows libavcodec to decode AVS2 streams with davs2 library.
-
 @c man end VIDEO DECODERS
 
 @chapter Audio Decoders
@@ -220,31 +195,6 @@ without this library.
 @chapter Subtitles Decoders
 @c man begin SUBTILES DECODERS
 
-@section libaribb24
-
-ARIB STD-B24 caption decoder.
-
-Implements profiles A and C of the ARIB STD-B24 standard.
-
-@subsection libaribb24 Decoder Options
-
-@table @option
-
-@item -aribb24-base-path @var{path}
-Sets the base path for the libaribb24 library. This is utilized for reading of
-configuration files (for custom unicode conversions), and for dumping of
-non-text symbols as images under that location.
-
-Unset by default.
-
-@item -aribb24-skip-ruby-text @var{boolean}
-Tells the decoder wrapper to skip text blocks that contain half-height ruby
-text.
-
-Enabled by default.
-
-@end table
-
 @section dvbsub
 
 @subsection Options
@@ -305,25 +255,18 @@ configuration. You need to explicitly configure the build with
 
 @table @option
 @item txt_page
-List of teletext page numbers to decode. Pages that do not match the specified
-list are dropped. You may use the special @code{*} string to match all pages,
-or @code{subtitle} to match all subtitle pages.
+List of teletext page numbers to decode. You may use the special * string to
+match all pages. Pages that do not match the specified list are dropped.
 Default value is *.
 @item txt_chop_top
 Discards the top teletext line. Default value is 1.
 @item txt_format
-Specifies the format of the decoded subtitles.
-@table @option
-@item bitmap
-The default format, you should use this for teletext pages, because certain
-graphics and colors cannot be expressed in simple text or even ASS.
-@item text
-Simple text based output without formatting.
-@item ass
-Formatted ASS output, subtitle pages and teletext pages are returned in
-different styles, subtitle pages are stripped down to text, but an effort is
-made to keep the text alignment and the formatting.
-@end table
+Specifies the format of the decoded subtitles. The teletext decoder is capable
+of decoding the teletext pages to bitmaps or to simple text, you should use
+"bitmap" for teletext pages, because certain graphics and colors cannot be
+expressed in simple text. You might use "text" for teletext based subtitles if
+your application can handle simple text based subtitles. Default value is
+bitmap.
 @item txt_left
 X offset of generated bitmaps, default is 0.
 @item txt_top
@@ -336,8 +279,7 @@ present between the subtitle lines because of double-sized teletext characters.
 Default value is 1.
 @item txt_duration
 Sets the display duration of the decoded teletext pages or subtitles in
-milliseconds. Default value is -1 which means infinity or until the next
-subtitle event comes.
+milliseconds. Default value is 30000 which is 30 seconds.
 @item txt_transparent
 Force transparent background of the generated teletext bitmaps. Default value
 is 0 which means an opaque background.

doc/demuxers.texi

@@ -25,6 +25,17 @@ Audible Format 2, 3, and 4 demuxer.
 
 This demuxer is used to demux Audible Format 2, 3, and 4 (.aa) files.
 
+@section applehttp
+
+Apple HTTP Live Streaming demuxer.
+
+This demuxer presents all AVStreams from all variant streams.
+The id field is set to the bitrate variant index number. By setting
+the discard flags on AVStreams (by pressing 'a' or 'v' in ffplay),
+the caller can decide which variant streams to actually receive.
+The total bitrate of the variant that the stream belongs to is
+available in a metadata key named "variant_bitrate".
+
 @section apng
 
 Animated Portable Network Graphics demuxer.
@@ -233,16 +244,6 @@ file subdir/file-2.wav
 @end example
 @end itemize
 
-@section dash
-
-Dynamic Adaptive Streaming over HTTP demuxer.
-
-This demuxer presents all AVStreams found in the manifest.
-By setting the discard flags on AVStreams the caller can decide
-which streams to actually receive.
-Each stream mirrors the @code{id} and @code{bandwidth} properties from the
-@code{<Representation>} as metadata keys named "id" and "variant_bitrate" respectively.
-
 @section flv, live_flv
 
 Adobe Flash Video Format demuxer.
@@ -258,12 +259,6 @@ ffmpeg -f live_flv -i rtmp://<any.server>/anything/key ....
 @table @option
 @item -flv_metadata @var{bool}
 Allocate the streams according to the onMetaData array content.
-
-@item -flv_ignore_prevtag @var{bool}
-Ignore the size of previous tag value.
-
-@item -flv_full_metadata @var{bool}
-Output all context of the onMetadata.
 @end table
 
 @section gif
@@ -309,15 +304,6 @@ infinitely.
 
 HLS demuxer
 
-Apple HTTP Live Streaming demuxer.
-
-This demuxer presents all AVStreams from all variant streams.
-The id field is set to the bitrate variant index number. By setting
-the discard flags on AVStreams (by pressing 'a' or 'v' in ffplay),
-the caller can decide which variant streams to actually receive.
-The total bitrate of the variant that the stream belongs to is
-available in a metadata key named "variant_bitrate".
-
 It accepts the following options:
 
 @table @option
@@ -330,14 +316,6 @@ segment index to start live streams at (negative values are from the end).
 @item max_reload
 Maximum number of times a insufficient list is attempted to be reloaded.
 Default value is 1000.
-
-@item http_persistent
-Use persistent HTTP connections. Applicable only for HTTP streams.
-Enabled by default.
-
-@item http_multiple
-Use multiple HTTP connections for downloading HTTP segments.
-Enabled by default for HTTP/1.1 servers.
 @end table
 
 @section image2
@@ -479,84 +457,14 @@ ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv
 
 The Game Music Emu library is a collection of video game music file emulators.
 
-See @url{https://bitbucket.org/mpyne/game-music-emu/overview} for more information.
+See @url{http://code.google.com/p/game-music-emu/} for more information.
 
-It accepts the following options:
+Some files have multiple tracks. The demuxer will pick the first track by
+default. The @option{track_index} option can be used to select a different
+track. Track indexes start at 0. The demuxer exports the number of tracks as
+@var{tracks} meta data entry.
 
-@table @option
-
-@item track_index
-Set the index of which track to demux. The demuxer can only export one track.
-Track indexes start at 0. Default is to pick the first track. Number of tracks
-is exported as @var{tracks} metadata entry.
-
-@item sample_rate
-Set the sampling rate of the exported track. Range is 1000 to 999999. Default is 44100.
-
-@item max_size @emph{(bytes)}
-The demuxer buffers the entire file into memory. Adjust this value to set the maximum buffer size,
-which in turn, acts as a ceiling for the size of files that can be read.
-Default is 50 MiB.
-
-@end table
-
-@section libmodplug
-
-ModPlug based module demuxer
-
-See @url{https://github.com/Konstanty/libmodplug}
-
-It will export one 2-channel 16-bit 44.1 kHz audio stream.
-Optionally, a @code{pal8} 16-color video stream can be exported with or without printed metadata.
-
-It accepts the following options:
-
-@table @option
-@item noise_reduction
-Apply a simple low-pass filter. Can be 1 (on) or 0 (off). Default is 0.
-
-@item reverb_depth
-Set amount of reverb. Range 0-100. Default is 0.
-
-@item reverb_delay
-Set delay in ms, clamped to 40-250 ms. Default is 0.
-
-@item bass_amount
-Apply bass expansion a.k.a. XBass or megabass. Range is 0 (quiet) to 100 (loud). Default is 0.
-
-@item bass_range
-Set cutoff i.e. upper-bound for bass frequencies. Range is 10-100 Hz. Default is 0.
-
-@item surround_depth
-Apply a Dolby Pro-Logic surround effect. Range is 0 (quiet) to 100 (heavy). Default is 0.
-
-@item surround_delay
-Set surround delay in ms, clamped to 5-40 ms. Default is 0.
-
-@item max_size
-The demuxer buffers the entire file into memory. Adjust this value to set the maximum buffer size,
-which in turn, acts as a ceiling for the size of files that can be read. Range is 0 to 100 MiB.
-0 removes buffer size limit (not recommended). Default is 5 MiB.
-
-@item video_stream_expr
-String which is evaluated using the eval API to assign colors to the generated video stream.
-Variables which can be used are @code{x}, @code{y}, @code{w}, @code{h}, @code{t}, @code{speed},
-@code{tempo}, @code{order}, @code{pattern} and @code{row}.
-
-@item video_stream
-Generate video stream. Can be 1 (on) or 0 (off). Default is 0.
-
-@item video_stream_w
-Set video frame width in 'chars' where one char indicates 8 pixels. Range is 20-512. Default is 30.
-
-@item video_stream_h
-Set video frame height in 'chars' where one char indicates 8 pixels. Range is 20-512. Default is 30.
-
-@item video_stream_ptxt
-Print metadata on video stream. Includes @code{speed}, @code{tempo}, @code{order}, @code{pattern},
-@code{row} and @code{ts} (time in ms). Can be 1 (on) or 0 (off). Default is 1.
-
-@end table
+For very large files, the @option{max_size} option may have to be adjusted.
 
 @section libopenmpt
 
@@ -612,9 +520,6 @@ This demuxer accepts the following options:
 Set size limit for looking up a new synchronization. Default value is
 65536.
 
-@item skip_unknown_pmt
-Skip PMTs for programs not defined in the PAT. Default value is 0.
-
 @item fix_teletext_pts
 Override teletext packet PTS and DTS values with the timestamps calculated
 from the PCR of the first program which the teletext stream is part of and is
@@ -629,10 +534,6 @@ Show the detected raw packet size, cannot be set by the user.
 Scan and combine all PMTs. The value is an integer with value from -1
 to 1 (-1 means automatic setting, 1 means enabled, 0 means
 disabled). Default value is -1.
-
-@item merge_pmt_versions
-Re-use existing streams when a PMT's version is updated and elementary
-streams move to different PIDs. Default value is 0.
 @end table
 
 @section mpjpeg
@@ -730,20 +631,4 @@ Example: convert the captions to a format most players understand:
 ffmpeg -i http://www.ted.com/talks/subtitles/id/1/lang/en talk1-en.srt
 @end example
 
-@section vapoursynth
-
-Vapoursynth wrapper.
-
-Due to security concerns, Vapoursynth scripts will not
-be autodetected so the input format has to be forced. For ff* CLI tools,
-add @code{-f vapoursynth} before the input @code{-i yourscript.vpy}.
-
-This demuxer accepts the following option:
-@table @option
-@item max_script_size
-The demuxer buffers the entire script into memory. Adjust this value to set the maximum buffer size,
-which in turn, acts as a ceiling for the size of scripts that can be read.
-Default is 1 MiB.
-@end table
-
 @c man end DEMUXERS

doc/developer.texi

@@ -10,7 +10,9 @@
 
 @contents
 
-@chapter Notes for external developers
+@chapter Developers Guide
+
+@section Notes for external developers
 
 This document is mostly useful for internal FFmpeg developers.
 External developers who need to use the API in their application should
@@ -28,13 +30,15 @@ For more detailed legal information about the use of FFmpeg in
 external programs read the @file{LICENSE} file in the source tree and
 consult @url{https://ffmpeg.org/legal.html}.
 
-@chapter Contributing
+@section Contributing
 
-There are 2 ways by which code gets into FFmpeg:
+There are 3 ways by which code gets into FFmpeg.
 @itemize @bullet
-@item Submitting patches to the ffmpeg-devel mailing list.
+@item Submitting patches to the main developer mailing list.
       See @ref{Submitting patches} for details.
 @item Directly committing changes to the main tree.
+@item Committing changes to a git clone, for example on github.com or
+      gitorious.org. And asking us to merge these changes.
 @end itemize
 
 Whichever way, changes should be reviewed by the maintainer of the code
@@ -43,9 +47,9 @@ 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 Coding Rules
 
-@section Code formatting conventions
+@subsection Code formatting conventions
 
 There are the following guidelines regarding the indentation in files:
 
@@ -70,7 +74,7 @@ The presentation is one inspired by 'indent -i4 -kr -nut'.
 The main priority in FFmpeg is simplicity and small code size in order to
 minimize the bug count.
 
-@section Comments
+@subsection Comments
 Use the JavaDoc/Doxygen  format (see examples below) so that code documentation
 can be generated automatically. All nontrivial functions should have a comment
 above them explaining what the function does, even if it is just one sentence.
@@ -110,7 +114,7 @@ int myfunc(int my_parameter)
 ...
 @end example
 
-@section C language features
+@subsection C language features
 
 FFmpeg is programmed in the ISO C90 language with a few additional
 features from ISO C99, namely:
@@ -128,9 +132,6 @@ 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
 Implementation defined behavior for signed integers is assumed to match the
 expected behavior for two's complement. Non representable values in integer
@@ -159,7 +160,7 @@ mixing statements and declarations;
 GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}).
 @end itemize
 
-@section Naming conventions
+@subsection 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
@@ -183,7 +184,7 @@ 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}.
+@samp{avpriv_aac_parse_header}.
 
 @item
 Each library has its own prefix for public symbols, in addition to the
@@ -203,7 +204,7 @@ letter as they are reserved by the C standard. Names starting with @code{_}
 are reserved at the file level and may not be used for externally visible
 symbols. If in doubt, just avoid names starting with @code{_} altogether.
 
-@section Miscellaneous conventions
+@subsection Miscellaneous conventions
 
 @itemize @bullet
 @item
@@ -215,7 +216,7 @@ Casts should be used only when necessary. Unneeded parentheses
 should also be avoided if they don't make the code easier to understand.
 @end itemize
 
-@section Editor configuration
+@subsection Editor configuration
 In order to configure Vim to follow FFmpeg formatting conventions, paste
 the following snippet into your @file{.vimrc}:
 @example
@@ -248,9 +249,9 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
 (setq c-default-style "ffmpeg")
 @end lisp
 
-@chapter Development Policy
+@section Development Policy
 
-@section Patches/Committing
+@subsection Patches/Committing
 @subheading Licenses for patches must be compatible with FFmpeg.
 Contributions should be licensed under the
 @uref{http://www.gnu.org/licenses/lgpl-2.1.html, LGPL 2.1},
@@ -349,7 +350,7 @@ time-frame (12h for build failures and security fixes, 3 days small changes,
 1 week for big patches) then commit your patch if you think it is OK.
 Also note, the maintainer can simply ask for more time to review!
 
-@section Code
+@subsection Code
 @subheading API/ABI changes should be discussed before they are made.
 Do not change behavior of the programs (renaming options etc) or public
 API or ABI without first discussing it on the ffmpeg-devel mailing list.
@@ -380,29 +381,12 @@ Never write to unallocated memory, never write over the end of arrays,
 always check values read from some untrusted source before using them
 as array index or other risky things.
 
-@section Documentation/Other
-@subheading Subscribe to the ffmpeg-devel mailing list.
-It is important to be subscribed to the
-@uref{https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel, ffmpeg-devel}
-mailing list. Almost any non-trivial patch is to be sent there for review.
-Other developers may have comments about your contribution. We expect you see
-those comments, and to improve it if requested. (N.B. Experienced committers
-have other channels, and may sometimes skip review for trivial fixes.) Also,
-discussion here about bug fixes and FFmpeg improvements by other developers may
-be helpful information for you. Finally, by being a list subscriber, your
-contribution will be posted immediately to the list, without the moderation
-hold which messages from non-subscribers experience.
-
-However, it is more important to the project that we receive your patch than
-that you be subscribed to the ffmpeg-devel list. If you have a patch, and don't
-want to subscribe and discuss the patch, then please do send it to the list
-anyway.
-
+@subsection Documentation/Other
 @subheading Subscribe to the ffmpeg-cvslog mailing list.
-Diffs of all commits are sent to the
-@uref{https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-cvslog, ffmpeg-cvslog}
-mailing list. Some developers read this list to review all code base changes
-from all sources. Subscribing to this list is not mandatory.
+It is important to do this as the diffs of all commits are sent there and
+reviewed by all the other developers. Bugs and possible improvements or
+general questions regarding commits are discussed there. We expect you to
+react if problems with your code are uncovered.
 
 @subheading Keep the documentation up to date.
 Update the documentation if you change behavior or add features. If you are
@@ -422,7 +406,7 @@ finding a new maintainer and also don't forget to update the @file{MAINTAINERS}
 
 We think our rules are not too hard. If you have comments, contact us.
 
-@chapter Code of conduct
+@section Code of conduct
 
 Be friendly and respectful towards others and third parties.
 Treat others the way you yourself want to be treated.
@@ -452,7 +436,7 @@ Finally, keep in mind the immortal words of Bill and Ted,
 "Be excellent to each other."
 
 @anchor{Submitting patches}
-@chapter Submitting patches
+@section Submitting patches
 
 First, read the @ref{Coding Rules} above if you did not yet, in particular
 the rules regarding patch submission.
@@ -501,7 +485,7 @@ Give us a few days to react. But if some time passes without reaction,
 send a reminder by email. Your patch should eventually be dealt with.
 
 
-@chapter New codecs or formats checklist
+@section New codecs or formats checklist
 
 @enumerate
 @item
@@ -553,7 +537,7 @@ Did you make sure it compiles standalone, i.e. with
 @end enumerate
 
 
-@chapter Patch submission checklist
+@section patch submission checklist
 
 @enumerate
 @item
@@ -563,9 +547,9 @@ Does @code{make fate} pass with the patch applied?
 Was the patch generated with git format-patch or send-email?
 
 @item
-Did you sign-off your patch? (@code{git commit -s})
-See @uref{https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/plain/Documentation/process/submitting-patches.rst, Sign your work} for the meaning
-of @dfn{sign-off}.
+Did you sign off your patch? (git commit -s)
+See @url{http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob_plain;f=Documentation/SubmittingPatches} for the meaning
+of sign off.
 
 @item
 Did you provide a clear git commit log message?
@@ -666,7 +650,7 @@ Test your code with valgrind and or Address Sanitizer to ensure it's free
 of leaks, out of array accesses, etc.
 @end enumerate
 
-@chapter Patch review process
+@section Patch review process
 
 All patches posted to ffmpeg-devel will be reviewed, unless they contain a
 clear note that the patch is not for the git master branch.
@@ -697,7 +681,7 @@ to be reviewed, please consider helping to review other patches, that is a great
 way to get everyone's patches reviewed sooner.
 
 @anchor{Regression tests}
-@chapter Regression tests
+@section Regression tests
 
 Before submitting a patch (or committing to the repository), you should at least
 test that you did not break anything.
@@ -708,7 +692,7 @@ Running 'make fate' accomplishes this, please see @url{fate.html} for details.
 this case, the reference results of the regression tests shall be modified
 accordingly].
 
-@section Adding files to the fate-suite dataset
+@subsection Adding files to the fate-suite dataset
 
 When there is no muxer or encoder available to generate test media for a
 specific test then the media has to be included in the fate-suite.
@@ -719,7 +703,7 @@ Once you have a working fate test and fate sample, provide in the commit
 message or introductory message for the patch series that you post to
 the ffmpeg-devel mailing list, a direct link to download the sample media.
 
-@section Visualizing Test Coverage
+@subsection Visualizing Test Coverage
 
 The FFmpeg build system allows visualizing the test coverage in an easy
 manner with the coverage tools @code{gcov}/@code{lcov}.  This involves
@@ -746,7 +730,7 @@ You can use the command @code{make lcov-reset} to reset the coverage
 measurements. You will need to rerun @code{make lcov} after running a
 new test.
 
-@section Using Valgrind
+@subsection Using Valgrind
 
 The configure script provides a shortcut for using valgrind to spot bugs
 related to memory handling. Just add the option
@@ -760,7 +744,7 @@ In case you need finer control over how valgrind is invoked, use the
 your configure line instead.
 
 @anchor{Release process}
-@chapter Release process
+@section Release process
 
 FFmpeg maintains a set of @strong{release branches}, which are the
 recommended deliverable for system integrators and distributors (such as
@@ -792,7 +776,7 @@ adjustments to the symbol versioning file. Please discuss such changes
 on the @strong{ffmpeg-devel} mailing list in time to allow forward planning.
 
 @anchor{Criteria for Point Releases}
-@section Criteria for Point Releases
+@subsection Criteria for Point Releases
 
 Changes that match the following criteria are valid candidates for
 inclusion into a point release:
@@ -816,7 +800,7 @@ point releases of the same release branch.
 The order for checking the rules is (1 OR 2 OR 3) AND 4.
 
 
-@section Release Checklist
+@subsection Release Checklist
 
 The release process involves the following steps:
 

doc/encoders.texi

@@ -64,6 +64,7 @@ to find an optimal combination by adding or subtracting a specific value from
 all quantizers and adjusting some individual quantizer a little.  Will tune
 itself based on whether @option{aac_is}, @option{aac_ms} and @option{aac_pns}
 are enabled.
+This is the default choice for a coder.
 
 @item anmr
 Average noise to mask ratio (ANMR) trellis-based solution.
@@ -76,10 +77,10 @@ Not currently recommended.
 @item fast
 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
+This method sets a constant quantizer for all bands. This is the fastest of all
+the methods and has no rate control or support for @option{aac_is} or
+@option{aac_pns}.
+Not recommended.
 
 @end table
 
@@ -733,14 +734,6 @@ if set to 0.
 
 Default value is 0.
 
-@item eld_v2
-Enable ELDv2 (LD-MPS extension for ELD stereo signals) for ELDv2 if set to 1,
-disabled if set to 0.
-
-Note that option is available when fdk-aac version (AACENCODER_LIB_VL0.AACENCODER_LIB_VL1.AACENCODER_LIB_VL2) > (4.0.0).
-
-Default value is 0.
-
 @item signaling
 Set SBR/PS signaling style.
 
@@ -989,11 +982,6 @@ Other values include 0 for mono and stereo, 1 for surround sound with masking
 and LFE bandwidth optimizations, and 255 for independent streams with an
 unspecified channel layout.
 
-@item apply_phase_inv (N.A.) (requires libopus >= 1.2)
-If set to 0, disables the use of phase inversion for intensity stereo,
-improving the quality of mono downmixes, but slightly reducing normal stereo
-quality. The default is 1 (phase inversion enabled).
-
 @end table
 
 @anchor{libshine}
@@ -1378,181 +1366,6 @@ makes it possible to store non-rgb pix_fmts.
 
 @end table
 
-@section libaom-av1
-
-libaom AV1 encoder wrapper.
-
-Requires the presence of the libaom headers and library during
-configuration.  You need to explicitly configure the build with
-@code{--enable-libaom}.
-
-@subsection Options
-
-The wrapper supports the following standard libavcodec options:
-
-@table @option
-
-@item b
-Set bitrate target in bits/second.  By default this will use
-variable-bitrate mode.  If @option{maxrate} and @option{minrate} are
-also set to the same value then it will use constant-bitrate mode,
-otherwise if @option{crf} is set as well then it will use
-constrained-quality mode.
-
-@item g keyint_min
-Set key frame placement.  The GOP size sets the maximum distance between
-key frames; if zero the output stream will be intra-only.  The minimum
-distance is ignored unless it is the same as the GOP size, in which case
-key frames will always appear at a fixed interval.  Not set by default,
-so without this option the library has completely free choice about
-where to place key frames.
-
-@item qmin qmax
-Set minimum/maximum quantisation values.  Valid range is from 0 to 63
-(warning: this does not match the quantiser values actually used by AV1
-- divide by four to map real quantiser values to this range).  Defaults
-to min/max (no constraint).
-
-@item minrate maxrate bufsize rc_init_occupancy
-Set rate control buffering parameters.  Not used if not set - defaults
-to unconstrained variable bitrate.
-
-@item threads
-Set the number of threads to use while encoding.  This may require the
-@option{tiles} or @option{row-mt} options to also be set to actually
-use the specified number of threads fully. Defaults to the number of
-hardware threads supported by the host machine.
-
-@item profile
-Set the encoding profile.  Defaults to using the profile which matches
-the bit depth and chroma subsampling of the input.
-
-@end table
-
-The wrapper also has some specific options:
-
-@table @option
-
-@item cpu-used
-Set the quality/encoding speed tradeoff.  Valid range is from 0 to 8,
-higher numbers indicating greater speed and lower quality.  The default
-value is 1, which will be slow and high quality.
-
-@item auto-alt-ref
-Enable use of alternate reference frames.  Defaults to the internal
-default of the library.
-
-@item arnr-max-frames (@emph{frames})
-Set altref noise reduction max frame count. Default is -1.
-
-@item arnr-strength (@emph{strength})
-Set altref noise reduction filter strength. Range is -1 to 6. Default is -1.
-
-@item aq-mode (@emph{aq-mode})
-Set adaptive quantization mode. Possible values:
-
-@table @samp
-@item none (@emph{0})
-Disabled.
-
-@item variance (@emph{1})
-Variance-based.
-
-@item complexity (@emph{2})
-Complexity-based.
-
-@item cyclic (@emph{3})
-Cyclic refresh.
-@end table
-
-@item lag-in-frames
-Set the maximum number of frames which the encoder may keep in flight
-at any one time for lookahead purposes.  Defaults to the internal
-default of the library.
-
-@item error-resilience
-Enable error resilience features:
-@table @option
-@item default
-Improve resilience against losses of whole frames.
-@end table
-Not enabled by default.
-
-@item crf
-Set the quality/size tradeoff for constant-quality (no bitrate target)
-and constrained-quality (with maximum bitrate target) modes. Valid
-range is 0 to 63, higher numbers indicating lower quality and smaller
-output size.  Only used if set; by default only the bitrate target is
-used.
-
-@item static-thresh
-Set a change threshold on blocks below which they will be skipped by
-the encoder.  Defined in arbitrary units as a nonnegative integer,
-defaulting to zero (no blocks are skipped).
-
-@item drop-threshold
-Set a threshold for dropping frames when close to rate control bounds.
-Defined as a percentage of the target buffer - when the rate control
-buffer falls below this percentage, frames will be dropped until it
-has refilled above the threshold.  Defaults to zero (no frames are
-dropped).
-
-@item denoise-noise-level (@emph{level})
-Amount of noise to be removed for grain synthesis. Grain synthesis is disabled if
-this option is not set or set to 0.
-
-@item denoise-block-size (@emph{pixels})
-Block size used for denoising for grain synthesis. If not set, AV1 codec
-uses the default value of 32.
-
-@item undershoot-pct (@emph{pct})
-Set datarate undershoot (min) percentage of the target bitrate. Range is -1 to 100.
-Default is -1.
-
-@item overshoot-pct (@emph{pct})
-Set datarate overshoot (max) percentage of the target bitrate. Range is -1 to 1000.
-Default is -1.
-
-@item minsection-pct (@emph{pct})
-Minimum percentage variation of the GOP bitrate from the target bitrate. If minsection-pct
-is not set, the libaomenc wrapper computes it as follows: @code{(minrate * 100 / bitrate)}.
-Range is -1 to 100. Default is -1 (unset).
-
-@item maxsection-pct (@emph{pct})
-Maximum percentage variation of the GOP bitrate from the target bitrate. If maxsection-pct
-is not set, the libaomenc wrapper computes it as follows: @code{(maxrate * 100 / bitrate)}.
-Range is -1 to 5000. Default is -1 (unset).
-
-@item frame-parallel (@emph{boolean})
-Enable frame parallel decodability features. Default is true.
-
-@item tiles
-Set the number of tiles to encode the input video with, as columns x
-rows.  Larger numbers allow greater parallelism in both encoding and
-decoding, but may decrease coding efficiency.  Defaults to the minimum
-number of tiles required by the size of the input video (this is 1x1
-(that is, a single tile) for sizes up to and including 4K).
-
-@item tile-columns tile-rows
-Set the number of tiles as log2 of the number of tile rows and columns.
-Provided for compatibility with libvpx/VP9.
-
-@item row-mt (Requires libaom >= 1.0.0-759-g90a15f4f2)
-Enable row based multi-threading. Disabled by default.
-
-@item enable-cdef (@emph{boolean})
-Enable Constrained Directional Enhancement Filter. The libaom-av1
-encoder enables CDEF by default.
-
-@item enable-global-motion (@emph{boolean})
-Enable the use of global motion for block prediction. Default is true.
-
-@item enable-intrabc (@emph{boolean})
-Enable block copy mode for intra block prediction. This mode is
-useful for screen content. Default is true.
-
-@end table
-
 @section libkvazaar
 
 Kvazaar H.265/HEVC encoder.
@@ -1824,8 +1637,7 @@ means unlimited.
 @table @option
 @item auto-alt-ref
 Enable use of alternate reference frames (2-pass only).
-Values greater than 1 enable multi-layer alternate reference frames (VP9 only).
-@item arnr-maxframes
+@item arnr-max-frames
 Set altref noise reduction max frame count.
 @item arnr-type
 Set altref noise reduction filter type: backward, forward, centered.
@@ -1838,38 +1650,6 @@ Set number of frames to look ahead for frametype and ratecontrol.
 @item error-resilient
 Enable error resiliency features.
 
-@item sharpness @var{integer}
-Increase sharpness at the expense of lower PSNR.
-The valid range is [0, 7].
-
-@item VP8-specific options
-@table @option
-@item ts-parameters
-Sets the temporal scalability configuration using a :-separated list of
-key=value pairs. For example, to specify temporal scalability parameters
-with @code{ffmpeg}:
-@example
-ffmpeg -i INPUT -c:v libvpx -ts-parameters ts_number_layers=3:\
-ts_target_bitrate=250000,500000,1000000:ts_rate_decimator=4,2,1:\
-ts_periodicity=4:ts_layer_id=0,2,1,2 OUTPUT
-@end example
-Below is a brief explanation of each of the parameters, please
-refer to @code{struct vpx_codec_enc_cfg} in @code{vpx/vpx_encoder.h} for more
-details.
-@table @option
-@item ts_number_layers
-Number of temporal coding layers.
-@item ts_target_bitrate
-Target bitrate for each temporal layer.
-@item ts_rate_decimator
-Frame rate decimation factor for each temporal layer.
-@item ts_periodicity
-Length of the sequence defining frame temporal layer membership.
-@item ts_layer_id
-Template defining the membership of frames to temporal layers.
-@end table
-@end table
-
 @item VP9-specific options
 @table @option
 @item lossless
@@ -1901,15 +1681,6 @@ colorspaces:
 @end table
 @item row-mt @var{boolean}
 Enable row based multi-threading.
-@item tune-content
-Set content type: default (0), screen (1), film (2).
-@item corpus-complexity
-Corpus VBR mode is a variant of standard VBR where the complexity distribution
-midpoint is passed in rather than calculated for a specific clip or chunk.
-
-The valid range is [0, 10000]. 0 (default) uses standard VBR.
-@item enable-tpl @var{boolean}
-Enable temporal dependency model.
 @end table
 
 @end table
@@ -2377,12 +2148,6 @@ Set the x265 preset.
 @item tune
 Set the x265 tune parameter.
 
-@item profile
-Set profile restrictions.
-
-@item crf
-Set the quality for constant quality mode.
-
 @item forced-idr
 Normally, when forcing a I-frame type, the encoder can select any type
 of I-frame. This option forces it to choose an IDR-frame.
@@ -2398,63 +2163,6 @@ ffmpeg -i input -c:v libx265 -x265-params crf=26:psy-rd=1 output.mp4
 @end example
 @end table
 
-@section libxavs2
-
-xavs2 AVS2-P2/IEEE1857.4 encoder wrapper.
-
-This encoder requires the presence of the libxavs2 headers and library
-during configuration. You need to explicitly configure the build with
-@option{--enable-libxavs2}.
-
-The following standard libavcodec options are used:
-@itemize
-@item
-@option{b} / @option{bit_rate}
-@item
-@option{g} / @option{gop_size}
-@item
-@option{bf} / @option{max_b_frames}
-@end itemize
-
-The encoder also has its own specific options:
-@subsection Options
-
-@table @option
-@item lcu_row_threads
-Set the number of parallel threads for rows from 1 to 8 (default 5).
-
-@item initial_qp
-Set the xavs2 quantization parameter from 1 to 63 (default 34). This is
-used to set the initial qp for the first frame.
-
-@item qp
-Set the xavs2 quantization parameter from 1 to 63 (default 34). This is
-used to set the qp value under constant-QP mode.
-
-@item max_qp
-Set the max qp for rate control from 1 to 63 (default 55).
-
-@item min_qp
-Set the min qp for rate control from 1 to 63 (default 20).
-
-@item speed_level
-Set the Speed level from 0 to 9 (default 0). Higher is better but slower.
-
-@item log_level
-Set the log level from -1 to 3 (default 0). -1: none, 0: error,
-1: warning, 2: info, 3: debug.
-
-@item xavs2-params
-Set xavs2 options using a list of @var{key}=@var{value} couples separated
-by ":".
-
-For example to specify libxavs2 encoding options with @option{-xavs2-params}:
-
-@example
-ffmpeg -i input -c:v libxavs2 -xavs2-params RdoqLevel=0 output.avs2
-@end example
-@end table
-
 @section libxvid
 
 Xvid MPEG-4 Part 2 encoder wrapper.
@@ -2640,14 +2348,6 @@ Never write it.
 @itemx always
 Always write it.
 @end table
-@item video_format @var{integer}
-Specifies the video_format written into the sequence display extension
-indicating the source of the video pictures. The default is @samp{unspecified},
-can be @samp{component}, @samp{pal}, @samp{ntsc}, @samp{secam} or @samp{mac}.
-For maximum compatibility, use @samp{component}.
-@item a53cc @var{boolean}
-Import closed captions (which must be ATSC compatible format) into output.
-Default is 1 (on).
 @end table
 
 @section png
@@ -2843,9 +2543,6 @@ The following standard libavcodec options are used:
 @option{bf} / @option{max_b_frames}
 @item
 @option{profile}
-
-If not set, this will be determined automatically from the format of the input
-frames and the profiles supported by the driver.
 @item
 @option{level}
 @item
@@ -2866,8 +2563,7 @@ Speed / quality tradeoff: higher values are faster / worse quality.
 Size / quality tradeoff: higher values are smaller / worse quality.
 @item
 @option{qmin}
-@item
-@option{qmax}
+(only: @option{qmax} is not supported)
 @item
 @option{i_qfactor} / @option{i_quant_factor}
 @item
@@ -2876,55 +2572,8 @@ Size / quality tradeoff: higher values are smaller / worse quality.
 @option{b_qfactor} / @option{b_quant_factor}
 @item
 @option{b_qoffset} / @option{b_quant_offset}
-@item
-@option{slices}
 @end itemize
 
-All encoders support the following options:
-@table @option
-@item low_power
-Some drivers/platforms offer a second encoder for some codecs intended to use
-less power than the default encoder; setting this option will attempt to use
-that encoder.  Note that it may support a reduced feature set, so some other
-options may not be available in this mode.
-
-@item idr_interval
-Set the number of normal intra frames between full-refresh (IDR) frames in
-open-GOP mode.  The intra frames are still IRAPs, but will not include global
-headers and may have non-decodable leading pictures.
-
-@item b_depth
-Set the B-frame reference depth.  When set to one (the default), all B-frames
-will refer only to P- or I-frames.  When set to greater values multiple layers
-of B-frames will be present, frames in each layer only referring to frames in
-higher layers.
-
-@item rc_mode
-Set the rate control mode to use.  A given driver may only support a subset of
-modes.
-
-Possible modes:
-@table @option
-@item auto
-Choose the mode automatically based on driver support and the other options.
-This is the default.
-@item CQP
-Constant-quality.
-@item CBR
-Constant-bitrate.
-@item VBR
-Variable-bitrate.
-@item ICQ
-Intelligent constant-quality.
-@item QVBR
-Quality-defined variable-bitrate.
-@item AVBR
-Average variable bitrate.
-@end table
-
-@end table
-
-Each encoder also has its own specific options:
 @table @option
 
 @item h264_vaapi
@@ -2932,6 +2581,8 @@ Each encoder also has its own specific options:
 @option{level} sets the value of @emph{level_idc}.
 
 @table @option
+@item low_power
+Use low-power encoding mode.
 @item coder
 Set entropy encoder (default is @emph{cabac}).  Possible values:
 
@@ -2944,70 +2595,21 @@ Use CABAC.
 @item cavlc
 Use CAVLC.
 @end table
-
-@item aud
-Include access unit delimiters in the stream (not included by default).
-
-@item sei
-Set SEI message types to include.
-Some combination of the following values:
-@table @samp
-@item identifier
-Include a @emph{user_data_unregistered} message containing information about
-the encoder.
-@item timing
-Include picture timing parameters (@emph{buffering_period} and
-@emph{pic_timing} messages).
-@item recovery_point
-Include recovery points where appropriate (@emph{recovery_point} messages).
-@end table
-
 @end table
 
 @item hevc_vaapi
 @option{profile} and @option{level} set the values of
 @emph{general_profile_idc} and @emph{general_level_idc} respectively.
 
-@table @option
-@item aud
-Include access unit delimiters in the stream (not included by default).
-
-@item tier
-Set @emph{general_tier_flag}.  This may affect the level chosen for the stream
-if it is not explicitly specified.
-
-@item sei
-Set SEI message types to include.
-Some combination of the following values:
-@table @samp
-@item hdr
-Include HDR metadata if the input frames have it
-(@emph{mastering_display_colour_volume} and @emph{content_light_level}
-messages).
-@end table
-
-@end table
-
 @item mjpeg_vaapi
-Only baseline DCT encoding is supported.  The encoder always uses the standard
-quantisation and huffman tables - @option{global_quality} scales the standard
-quantisation table (range 1-100).
-
-For YUV, 4:2:0, 4:2:2 and 4:4:4 subsampling modes are supported.  RGB is also
-supported, and will create an RGB JPEG.
-
-@table @option
-@item jfif
-Include JFIF header in each frame (not included by default).
-@item huffman
-Include standard huffman tables (on by default).  Turning this off will save
-a few hundred bytes in each output frame, but may lose compatibility with some
-JPEG decoders which don't fully handle MJPEG.
-@end table
+Always encodes using the standard quantisation and huffman tables -
+@option{global_quality} scales the standard quantisation table (range 1-100).
 
 @item mpeg2_vaapi
 @option{profile} and @option{level} set the value of @emph{profile_and_level_indication}.
 
+No rate control is supported.
+
 @item vp8_vaapi
 B-frames are not supported.
 

doc/examples/.gitignore

@@ -20,5 +20,3 @@
 /scaling_video
 /transcode_aac
 /transcoding
-/vaapi_encode
-/vaapi_transcode

doc/examples/Makefile

@@ -19,8 +19,6 @@ EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE)  += resampling_audio
 EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE)     += scaling_video
 EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE)     += transcode_aac
 EXAMPLES-$(CONFIG_TRANSCODING_EXAMPLE)       += transcoding
-EXAMPLES-$(CONFIG_VAAPI_ENCODE_EXAMPLE)      += vaapi_encode
-EXAMPLES-$(CONFIG_VAAPI_TRANSCODE_EXAMPLE)   += vaapi_transcode
 
 EXAMPLES       := $(EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)$(EXESUF))
 EXAMPLES_G     := $(EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)_g$(EXESUF))
@@ -37,7 +35,7 @@ $(EXAMPLES_G): %$(PROGSSUF)_g$(EXESUF): %.o
 examples: $(EXAMPLES)
 
 $(EXAMPLES:%$(PROGSSUF)$(EXESUF)=%.o): | doc/examples
-OUTDIRS += doc/examples
+OBJDIRS += doc/examples
 
 DOXY_INPUT += $(EXAMPLES:%$(PROGSSUF)$(EXESUF)=%.c)
 

doc/examples/avio_dir_cmd.c

@@ -143,6 +143,8 @@ int main(int argc, char *argv[])
         return 1;
     }
 
+    /* register codecs and formats and other lavf/lavc components*/
+    av_register_all();
     avformat_network_init();
 
     op = argv[1];

doc/examples/avio_reading.c

@@ -44,8 +44,6 @@ static int read_packet(void *opaque, uint8_t *buf, int buf_size)
     struct buffer_data *bd = (struct buffer_data *)opaque;
     buf_size = FFMIN(buf_size, bd->size);
 
-    if (!buf_size)
-        return AVERROR_EOF;
     printf("ptr:%p size:%zu\n", bd->ptr, bd->size);
 
     /* copy internal buffer data to buf */
@@ -74,6 +72,9 @@ int main(int argc, char *argv[])
     }
     input_filename = argv[1];
 
+    /* register codecs and formats and other lavf/lavc components*/
+    av_register_all();
+
     /* slurp file content into buffer */
     ret = av_file_map(input_filename, &buffer, &buffer_size, 0, NULL);
     if (ret < 0)
@@ -117,12 +118,11 @@ int main(int argc, char *argv[])
 
 end:
     avformat_close_input(&fmt_ctx);
-
     /* note: the internal buffer could have changed, and be != avio_ctx_buffer */
-    if (avio_ctx)
+    if (avio_ctx) {
         av_freep(&avio_ctx->buffer);
-    avio_context_free(&avio_ctx);
-
+        av_freep(&avio_ctx);
+    }
     av_file_unmap(buffer, buffer_size);
 
     if (ret < 0) {

doc/examples/decode_audio.c

@@ -94,6 +94,9 @@ int main(int argc, char **argv)
     filename    = argv[1];
     outfilename = argv[2];
 
+    /* register all the codecs */
+    avcodec_register_all();
+
     pkt = av_packet_alloc();
 
     /* find the MPEG audio decoder */

doc/examples/decode_video.c

@@ -101,6 +101,8 @@ int main(int argc, char **argv)
     filename    = argv[1];
     outfilename = argv[2];
 
+    avcodec_register_all();
+
     pkt = av_packet_alloc();
     if (!pkt)
         exit(1);

doc/examples/demuxing_decoding.c

@@ -252,6 +252,9 @@ int main (int argc, char **argv)
     video_dst_filename = argv[2];
     audio_dst_filename = argv[3];
 
+    /* register all formats and codecs */
+    av_register_all();
+
     /* open input file, and allocate format context */
     if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
         fprintf(stderr, "Could not open source file %s\n", src_filename);

doc/examples/encode_audio.c

@@ -138,6 +138,9 @@ int main(int argc, char **argv)
     }
     filename = argv[1];
 
+    /* register all the codecs */
+    avcodec_register_all();
+
     /* find the MP2 encoder */
     codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
     if (!codec) {

doc/examples/encode_video.c

@@ -84,6 +84,8 @@ int main(int argc, char **argv)
     filename = argv[1];
     codec_name = argv[2];
 
+    avcodec_register_all();
+
     /* find the mpeg1video encoder */
     codec = avcodec_find_encoder_by_name(codec_name);
     if (!codec) {

doc/examples/extract_mvs.c

@@ -129,6 +129,8 @@ int main(int argc, char **argv)
     }
     src_filename = argv[1];
 
+    av_register_all();
+
     if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
         fprintf(stderr, "Could not open source file %s\n", src_filename);
         exit(1);

doc/examples/filter_audio.c

@@ -64,13 +64,13 @@ static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
 {
     AVFilterGraph *filter_graph;
     AVFilterContext *abuffer_ctx;
-    const AVFilter  *abuffer;
+    AVFilter        *abuffer;
     AVFilterContext *volume_ctx;
-    const AVFilter  *volume;
+    AVFilter        *volume;
     AVFilterContext *aformat_ctx;
-    const AVFilter  *aformat;
+    AVFilter        *aformat;
     AVFilterContext *abuffersink_ctx;
-    const AVFilter  *abuffersink;
+    AVFilter        *abuffersink;
 
     AVDictionary *options_dict = NULL;
     uint8_t options_str[1024];
@@ -289,6 +289,8 @@ int main(int argc, char *argv[])
         return 1;
     }
 
+    avfilter_register_all();
+
     /* Allocate the frame we will be using to store the data. */
     frame  = av_frame_alloc();
     if (!frame) {

doc/examples/filtering_audio.c

@@ -32,6 +32,7 @@
 
 #include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
+#include <libavfilter/avfiltergraph.h>
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
 #include <libavutil/opt.h>
@@ -74,6 +75,7 @@ static int open_input_file(const char *filename)
     if (!dec_ctx)
         return AVERROR(ENOMEM);
     avcodec_parameters_to_context(dec_ctx, fmt_ctx->streams[audio_stream_index]->codecpar);
+    av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
 
     /* init the audio decoder */
     if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
@@ -88,8 +90,8 @@ static int init_filters(const char *filters_descr)
 {
     char args[512];
     int ret = 0;
-    const AVFilter *abuffersrc  = avfilter_get_by_name("abuffer");
-    const AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
+    AVFilter *abuffersrc  = avfilter_get_by_name("abuffer");
+    AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
     AVFilterInOut *outputs = avfilter_inout_alloc();
     AVFilterInOut *inputs  = avfilter_inout_alloc();
     static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
@@ -227,6 +229,9 @@ int main(int argc, char **argv)
         exit(1);
     }
 
+    av_register_all();
+    avfilter_register_all();
+
     if ((ret = open_input_file(argv[1])) < 0)
         goto end;
     if ((ret = init_filters(filter_descr)) < 0)

doc/examples/filtering_video.c

@@ -29,11 +29,10 @@
 
 #define _XOPEN_SOURCE 600 /* for usleep */
 #include <unistd.h>
-#include <stdio.h>
-#include <stdlib.h>
 
 #include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
+#include <libavfilter/avfiltergraph.h>
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
 #include <libavutil/opt.h>
@@ -79,6 +78,7 @@ static int open_input_file(const char *filename)
     if (!dec_ctx)
         return AVERROR(ENOMEM);
     avcodec_parameters_to_context(dec_ctx, fmt_ctx->streams[video_stream_index]->codecpar);
+    av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
 
     /* init the video decoder */
     if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
@@ -93,8 +93,8 @@ static int init_filters(const char *filters_descr)
 {
     char args[512];
     int ret = 0;
-    const AVFilter *buffersrc  = avfilter_get_by_name("buffer");
-    const AVFilter *buffersink = avfilter_get_by_name("buffersink");
+    AVFilter *buffersrc  = avfilter_get_by_name("buffer");
+    AVFilter *buffersink = avfilter_get_by_name("buffersink");
     AVFilterInOut *outputs = avfilter_inout_alloc();
     AVFilterInOut *inputs  = avfilter_inout_alloc();
     AVRational time_base = fmt_ctx->streams[video_stream_index]->time_base;
@@ -211,20 +211,20 @@ int main(int argc, char **argv)
 {
     int ret;
     AVPacket packet;
-    AVFrame *frame;
-    AVFrame *filt_frame;
+    AVFrame *frame = av_frame_alloc();
+    AVFrame *filt_frame = av_frame_alloc();
 
+    if (!frame || !filt_frame) {
+        perror("Could not allocate frame");
+        exit(1);
+    }
     if (argc != 2) {
         fprintf(stderr, "Usage: %s file\n", argv[0]);
         exit(1);
     }
 
-    frame = av_frame_alloc();
-    filt_frame = av_frame_alloc();
-    if (!frame || !filt_frame) {
-        perror("Could not allocate frame");
-        exit(1);
-    }
+    av_register_all();
+    avfilter_register_all();
 
     if ((ret = open_input_file(argv[1])) < 0)
         goto end;
@@ -252,25 +252,27 @@ int main(int argc, char **argv)
                     goto end;
                 }
 
-                frame->pts = frame->best_effort_timestamp;
+                if (ret >= 0) {
+                    frame->pts = frame->best_effort_timestamp;
 
-                /* push the decoded frame into the filtergraph */
-                if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
-                    av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
-                    break;
-                }
-
-                /* pull filtered frames from the filtergraph */
-                while (1) {
-                    ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
-                    if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
+                    /* push the decoded frame into the filtergraph */
+                    if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
+                        av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
                         break;
-                    if (ret < 0)
-                        goto end;
-                    display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base);
-                    av_frame_unref(filt_frame);
+                    }
+
+                    /* pull filtered frames from the filtergraph */
+                    while (1) {
+                        ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
+                        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
+                            break;
+                        if (ret < 0)
+                            goto end;
+                        display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base);
+                        av_frame_unref(filt_frame);
+                    }
+                    av_frame_unref(frame);
                 }
-                av_frame_unref(frame);
             }
         }
         av_packet_unref(&packet);

doc/examples/http_multiclient.c

@@ -114,6 +114,7 @@ int main(int argc, char **argv)
     in_uri = argv[1];
     out_uri = argv[2];
 
+    av_register_all();
     avformat_network_init();
 
     if ((ret = av_dict_set(&options, "listen", "2", 0)) < 0) {

doc/examples/hw_decode.c

@@ -4,23 +4,21 @@
  *
  * HW Acceleration API (video decoding) decode sample
  *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to deal
- * in the Software without restriction, including without limitation the rights
- * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- * copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
+ * This file is part of FFmpeg.
  *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
+ * 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.
  *
- * 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.
+ * 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
  */
 
 /**
@@ -46,6 +44,34 @@ static AVBufferRef *hw_device_ctx = NULL;
 static enum AVPixelFormat hw_pix_fmt;
 static FILE *output_file = NULL;
 
+static enum AVPixelFormat find_fmt_by_hw_type(const enum AVHWDeviceType type)
+{
+    enum AVPixelFormat fmt;
+
+    switch (type) {
+    case AV_HWDEVICE_TYPE_VAAPI:
+        fmt = AV_PIX_FMT_VAAPI;
+        break;
+    case AV_HWDEVICE_TYPE_DXVA2:
+        fmt = AV_PIX_FMT_DXVA2_VLD;
+        break;
+    case AV_HWDEVICE_TYPE_D3D11VA:
+        fmt = AV_PIX_FMT_D3D11;
+        break;
+    case AV_HWDEVICE_TYPE_VDPAU:
+        fmt = AV_PIX_FMT_VDPAU;
+        break;
+    case AV_HWDEVICE_TYPE_VIDEOTOOLBOX:
+        fmt = AV_PIX_FMT_VIDEOTOOLBOX;
+        break;
+    default:
+        fmt = AV_PIX_FMT_NONE;
+        break;
+    }
+
+    return fmt;
+}
+
 static int hw_decoder_init(AVCodecContext *ctx, const enum AVHWDeviceType type)
 {
     int err = 0;
@@ -88,7 +114,7 @@ static int decode_write(AVCodecContext *avctx, AVPacket *packet)
         return ret;
     }
 
-    while (1) {
+    while (ret >= 0) {
         if (!(frame = av_frame_alloc()) || !(sw_frame = av_frame_alloc())) {
             fprintf(stderr, "Can not alloc frame\n");
             ret = AVERROR(ENOMEM);
@@ -140,10 +166,13 @@ static int decode_write(AVCodecContext *avctx, AVPacket *packet)
     fail:
         av_frame_free(&frame);
         av_frame_free(&sw_frame);
-        av_freep(&buffer);
+        if (buffer)
+            av_freep(&buffer);
         if (ret < 0)
             return ret;
     }
+
+    return 0;
 }
 
 int main(int argc, char *argv[])
@@ -155,20 +184,18 @@ int main(int argc, char *argv[])
     AVCodec *decoder = NULL;
     AVPacket packet;
     enum AVHWDeviceType type;
-    int i;
 
     if (argc < 4) {
-        fprintf(stderr, "Usage: %s <device type> <input file> <output file>\n", argv[0]);
+        fprintf(stderr, "Usage: %s <vaapi|vdpau|dxva2|d3d11va> <input file> <output file>\n", argv[0]);
         return -1;
     }
 
+    av_register_all();
+
     type = av_hwdevice_find_type_by_name(argv[1]);
-    if (type == AV_HWDEVICE_TYPE_NONE) {
-        fprintf(stderr, "Device type %s is not supported.\n", argv[1]);
-        fprintf(stderr, "Available device types:");
-        while((type = av_hwdevice_iterate_types(type)) != AV_HWDEVICE_TYPE_NONE)
-            fprintf(stderr, " %s", av_hwdevice_get_type_name(type));
-        fprintf(stderr, "\n");
+    hw_pix_fmt = find_fmt_by_hw_type(type);
+    if (hw_pix_fmt == -1) {
+        fprintf(stderr, "Cannot support '%s' in this example.\n", argv[1]);
         return -1;
     }
 
@@ -191,20 +218,6 @@ int main(int argc, char *argv[])
     }
     video_stream = ret;
 
-    for (i = 0;; i++) {
-        const AVCodecHWConfig *config = avcodec_get_hw_config(decoder, i);
-        if (!config) {
-            fprintf(stderr, "Decoder %s does not support device type %s.\n",
-                    decoder->name, av_hwdevice_get_type_name(type));
-            return -1;
-        }
-        if (config->methods & AV_CODEC_HW_CONFIG_METHOD_HW_DEVICE_CTX &&
-            config->device_type == type) {
-            hw_pix_fmt = config->pix_fmt;
-            break;
-        }
-    }
-
     if (!(decoder_ctx = avcodec_alloc_context3(decoder)))
         return AVERROR(ENOMEM);
 
@@ -213,6 +226,7 @@ int main(int argc, char *argv[])
         return -1;
 
     decoder_ctx->get_format  = get_hw_format;
+    av_opt_set_int(decoder_ctx, "refcounted_frames", 1, 0);
 
     if (hw_decoder_init(decoder_ctx, type) < 0)
         return -1;

doc/examples/metadata.c

@@ -44,14 +44,10 @@ int main (int argc, char **argv)
         return 1;
     }
 
+    av_register_all();
     if ((ret = avformat_open_input(&fmt_ctx, argv[1], NULL, NULL)))
         return ret;
 
-    if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
-        return ret;
-    }
-
     while ((tag = av_dict_get(fmt_ctx->metadata, "", tag, AV_DICT_IGNORE_SUFFIX)))
         printf("%s=%s\n", tag->key, tag->value);
 

doc/examples/muxing.c

@@ -488,9 +488,9 @@ static AVFrame *get_video_frame(OutputStream *ost)
             }
         }
         fill_yuv_image(ost->tmp_frame, ost->next_pts, c->width, c->height);
-        sws_scale(ost->sws_ctx, (const uint8_t * const *) ost->tmp_frame->data,
-                  ost->tmp_frame->linesize, 0, c->height, ost->frame->data,
-                  ost->frame->linesize);
+        sws_scale(ost->sws_ctx,
+                  (const uint8_t * const *)ost->tmp_frame->data, ost->tmp_frame->linesize,
+                  0, c->height, ost->frame->data, ost->frame->linesize);
     } else {
         fill_yuv_image(ost->frame, ost->next_pts, c->width, c->height);
     }
@@ -564,6 +564,9 @@ int main(int argc, char **argv)
     AVDictionary *opt = NULL;
     int i;
 
+    /* Initialize libavcodec, and register all codecs and formats. */
+    av_register_all();
+
     if (argc < 2) {
         printf("usage: %s output_file\n"
                "API example program to output a media file with libavformat.\n"

doc/examples/qsvdec.c

@@ -150,6 +150,8 @@ int main(int argc, char **argv)
 
     int ret, i;
 
+    av_register_all();
+
     if (argc < 3) {
         fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
         return 1;
@@ -208,6 +210,7 @@ int main(int argc, char **argv)
                video_st->codecpar->extradata_size);
         decoder_ctx->extradata_size = video_st->codecpar->extradata_size;
     }
+    decoder_ctx->refcounted_frames = 1;
 
     decoder_ctx->opaque      = &decode;
     decoder_ctx->get_format  = get_format;

doc/examples/remuxing.c

@@ -65,6 +65,8 @@ int main(int argc, char **argv)
     in_filename  = argv[1];
     out_filename = argv[2];
 
+    av_register_all();
+
     if ((ret = avformat_open_input(&ifmt_ctx, in_filename, 0, 0)) < 0) {
         fprintf(stderr, "Could not open input file '%s'", in_filename);
         goto end;

doc/examples/transcode_aac.c

@@ -1,6 +1,4 @@
 /*
- * Copyright (c) 2013-2018 Andreas Unterweger
- *
  * This file is part of FFmpeg.
  *
  * FFmpeg is free software; you can redistribute it and/or
@@ -10,7 +8,7 @@
  *
  * 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
+ * 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
@@ -20,11 +18,10 @@
 
 /**
  * @file
- * Simple audio converter
+ * simple audio converter
  *
  * @example transcode_aac.c
  * Convert an input audio file to AAC in an MP4 container using FFmpeg.
- * Formats other than MP4 are supported based on the output file extension.
  * @author Andreas Unterweger (dustsigns@gmail.com)
  */
 
@@ -43,18 +40,12 @@
 
 #include "libswresample/swresample.h"
 
-/* The output bit rate in bit/s */
+/** The output bit rate in kbit/s */
 #define OUTPUT_BIT_RATE 96000
-/* The number of output channels */
+/** The number of output channels */
 #define OUTPUT_CHANNELS 2
 
-/**
- * Open an input file and the required decoder.
- * @param      filename             File to be opened
- * @param[out] input_format_context Format context of opened file
- * @param[out] input_codec_context  Codec context of opened file
- * @return Error code (0 if successful)
- */
+/** Open an input file and the required decoder. */
 static int open_input_file(const char *filename,
                            AVFormatContext **input_format_context,
                            AVCodecContext **input_codec_context)
@@ -63,7 +54,7 @@ static int open_input_file(const char *filename,
     AVCodec *input_codec;
     int error;
 
-    /* Open the input file to read from it. */
+    /** Open the input file to read from it. */
     if ((error = avformat_open_input(input_format_context, filename, NULL,
                                      NULL)) < 0) {
         fprintf(stderr, "Could not open input file '%s' (error '%s')\n",
@@ -72,7 +63,7 @@ static int open_input_file(const char *filename,
         return error;
     }
 
-    /* Get information on the input file (number of streams etc.). */
+    /** Get information on the input file (number of streams etc.). */
     if ((error = avformat_find_stream_info(*input_format_context, NULL)) < 0) {
         fprintf(stderr, "Could not open find stream info (error '%s')\n",
                 av_err2str(error));
@@ -80,7 +71,7 @@ static int open_input_file(const char *filename,
         return error;
     }
 
-    /* Make sure that there is only one stream in the input file. */
+    /** Make sure that there is only one stream in the input file. */
     if ((*input_format_context)->nb_streams != 1) {
         fprintf(stderr, "Expected one audio input stream, but found %d\n",
                 (*input_format_context)->nb_streams);
@@ -88,14 +79,14 @@ static int open_input_file(const char *filename,
         return AVERROR_EXIT;
     }
 
-    /* Find a decoder for the audio stream. */
+    /** Find a decoder for the audio stream. */
     if (!(input_codec = avcodec_find_decoder((*input_format_context)->streams[0]->codecpar->codec_id))) {
         fprintf(stderr, "Could not find input codec\n");
         avformat_close_input(input_format_context);
         return AVERROR_EXIT;
     }
 
-    /* Allocate a new decoding context. */
+    /** allocate a new decoding context */
     avctx = avcodec_alloc_context3(input_codec);
     if (!avctx) {
         fprintf(stderr, "Could not allocate a decoding context\n");
@@ -103,7 +94,7 @@ static int open_input_file(const char *filename,
         return AVERROR(ENOMEM);
     }
 
-    /* Initialize the stream parameters with demuxer information. */
+    /** initialize the stream parameters with demuxer information */
     error = avcodec_parameters_to_context(avctx, (*input_format_context)->streams[0]->codecpar);
     if (error < 0) {
         avformat_close_input(input_format_context);
@@ -111,7 +102,7 @@ static int open_input_file(const char *filename,
         return error;
     }
 
-    /* Open the decoder for the audio stream to use it later. */
+    /** Open the decoder for the audio stream to use it later. */
     if ((error = avcodec_open2(avctx, input_codec, NULL)) < 0) {
         fprintf(stderr, "Could not open input codec (error '%s')\n",
                 av_err2str(error));
@@ -120,7 +111,7 @@ static int open_input_file(const char *filename,
         return error;
     }
 
-    /* Save the decoder context for easier access later. */
+    /** Save the decoder context for easier access later. */
     *input_codec_context = avctx;
 
     return 0;
@@ -130,11 +121,6 @@ static int open_input_file(const char *filename,
  * Open an output file and the required encoder.
  * Also set some basic encoder parameters.
  * Some of these parameters are based on the input file's parameters.
- * @param      filename              File to be opened
- * @param      input_codec_context   Codec context of input file
- * @param[out] output_format_context Format context of output file
- * @param[out] output_codec_context  Codec context of output file
- * @return Error code (0 if successful)
  */
 static int open_output_file(const char *filename,
                             AVCodecContext *input_codec_context,
@@ -147,7 +133,7 @@ static int open_output_file(const char *filename,
     AVCodec *output_codec          = NULL;
     int error;
 
-    /* Open the output file to write to it. */
+    /** Open the output file to write to it. */
     if ((error = avio_open(&output_io_context, filename,
                            AVIO_FLAG_WRITE)) < 0) {
         fprintf(stderr, "Could not open output file '%s' (error '%s')\n",
@@ -155,35 +141,32 @@ static int open_output_file(const char *filename,
         return error;
     }
 
-    /* Create a new format context for the output container format. */
+    /** Create a new format context for the output container format. */
     if (!(*output_format_context = avformat_alloc_context())) {
         fprintf(stderr, "Could not allocate output format context\n");
         return AVERROR(ENOMEM);
     }
 
-    /* Associate the output file (pointer) with the container format context. */
+    /** Associate the output file (pointer) with the container format context. */
     (*output_format_context)->pb = output_io_context;
 
-    /* Guess the desired container format based on the file extension. */
+    /** Guess the desired container format based on the file extension. */
     if (!((*output_format_context)->oformat = av_guess_format(NULL, filename,
                                                               NULL))) {
         fprintf(stderr, "Could not find output file format\n");
         goto cleanup;
     }
 
-    if (!((*output_format_context)->url = av_strdup(filename))) {
-        fprintf(stderr, "Could not allocate url.\n");
-        error = AVERROR(ENOMEM);
-        goto cleanup;
-    }
+    av_strlcpy((*output_format_context)->filename, filename,
+               sizeof((*output_format_context)->filename));
 
-    /* Find the encoder to be used by its name. */
+    /** Find the encoder to be used by its name. */
     if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC))) {
         fprintf(stderr, "Could not find an AAC encoder.\n");
         goto cleanup;
     }
 
-    /* Create a new audio stream in the output file container. */
+    /** Create a new audio stream in the output file container. */
     if (!(stream = avformat_new_stream(*output_format_context, NULL))) {
         fprintf(stderr, "Could not create new stream\n");
         error = AVERROR(ENOMEM);
@@ -197,27 +180,31 @@ static int open_output_file(const char *filename,
         goto cleanup;
     }
 
-    /* Set the basic encoder parameters.
-     * The input file's sample rate is used to avoid a sample rate conversion. */
+    /**
+     * Set the basic encoder parameters.
+     * The input file's sample rate is used to avoid a sample rate conversion.
+     */
     avctx->channels       = OUTPUT_CHANNELS;
     avctx->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
     avctx->sample_rate    = input_codec_context->sample_rate;
     avctx->sample_fmt     = output_codec->sample_fmts[0];
     avctx->bit_rate       = OUTPUT_BIT_RATE;
 
-    /* Allow the use of the experimental AAC encoder. */
+    /** Allow the use of the experimental AAC encoder */
     avctx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
 
-    /* Set the sample rate for the container. */
+    /** Set the sample rate for the container. */
     stream->time_base.den = input_codec_context->sample_rate;
     stream->time_base.num = 1;
 
-    /* Some container formats (like MP4) require global headers to be present.
-     * Mark the encoder so that it behaves accordingly. */
+    /**
+     * Some container formats (like MP4) require global headers to be present
+     * Mark the encoder so that it behaves accordingly.
+     */
     if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
         avctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
 
-    /* Open the encoder for the audio stream to use it later. */
+    /** Open the encoder for the audio stream to use it later. */
     if ((error = avcodec_open2(avctx, output_codec, NULL)) < 0) {
         fprintf(stderr, "Could not open output codec (error '%s')\n",
                 av_err2str(error));
@@ -230,7 +217,7 @@ static int open_output_file(const char *filename,
         goto cleanup;
     }
 
-    /* Save the encoder context for easier access later. */
+    /** Save the encoder context for easier access later. */
     *output_codec_context = avctx;
 
     return 0;
@@ -243,23 +230,16 @@ cleanup:
     return error < 0 ? error : AVERROR_EXIT;
 }
 
-/**
- * Initialize one data packet for reading or writing.
- * @param packet Packet to be initialized
- */
+/** Initialize one data packet for reading or writing. */
 static void init_packet(AVPacket *packet)
 {
     av_init_packet(packet);
-    /* Set the packet data and size so that it is recognized as being empty. */
+    /** Set the packet data and size so that it is recognized as being empty. */
     packet->data = NULL;
     packet->size = 0;
 }
 
-/**
- * Initialize one audio frame for reading from the input file.
- * @param[out] frame Frame to be initialized
- * @return Error code (0 if successful)
- */
+/** Initialize one audio frame for reading from the input file */
 static int init_input_frame(AVFrame **frame)
 {
     if (!(*frame = av_frame_alloc())) {
@@ -273,10 +253,6 @@ static int init_input_frame(AVFrame **frame)
  * Initialize the audio resampler based on the input and output codec settings.
  * If the input and output sample formats differ, a conversion is required
  * libswresample takes care of this, but requires initialization.
- * @param      input_codec_context  Codec context of the input file
- * @param      output_codec_context Codec context of the output file
- * @param[out] resample_context     Resample context for the required conversion
- * @return Error code (0 if successful)
  */
 static int init_resampler(AVCodecContext *input_codec_context,
                           AVCodecContext *output_codec_context,
@@ -284,7 +260,7 @@ static int init_resampler(AVCodecContext *input_codec_context,
 {
         int error;
 
-        /*
+        /**
          * Create a resampler context for the conversion.
          * Set the conversion parameters.
          * Default channel layouts based on the number of channels
@@ -303,14 +279,14 @@ static int init_resampler(AVCodecContext *input_codec_context,
             fprintf(stderr, "Could not allocate resample context\n");
             return AVERROR(ENOMEM);
         }
-        /*
+        /**
         * Perform a sanity check so that the number of converted samples is
         * not greater than the number of samples to be converted.
         * If the sample rates differ, this case has to be handled differently
         */
         av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);
 
-        /* Open the resampler with the specified parameters. */
+        /** Open the resampler with the specified parameters. */
         if ((error = swr_init(*resample_context)) < 0) {
             fprintf(stderr, "Could not open resample context\n");
             swr_free(resample_context);
@@ -319,15 +295,10 @@ static int init_resampler(AVCodecContext *input_codec_context,
     return 0;
 }
 
-/**
- * Initialize a FIFO buffer for the audio samples to be encoded.
- * @param[out] fifo                 Sample buffer
- * @param      output_codec_context Codec context of the output file
- * @return Error code (0 if successful)
- */
+/** Initialize a FIFO buffer for the audio samples to be encoded. */
 static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
 {
-    /* Create the FIFO buffer based on the specified output sample format. */
+    /** Create the FIFO buffer based on the specified output sample format. */
     if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
                                       output_codec_context->channels, 1))) {
         fprintf(stderr, "Could not allocate FIFO\n");
@@ -336,11 +307,7 @@ static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
     return 0;
 }
 
-/**
- * Write the header of the output file container.
- * @param output_format_context Format context of the output file
- * @return Error code (0 if successful)
- */
+/** Write the header of the output file container. */
 static int write_output_file_header(AVFormatContext *output_format_context)
 {
     int error;
@@ -352,32 +319,20 @@ static int write_output_file_header(AVFormatContext *output_format_context)
     return 0;
 }
 
-/**
- * Decode one audio frame from the input file.
- * @param      frame                Audio frame to be decoded
- * @param      input_format_context Format context of the input file
- * @param      input_codec_context  Codec context of the input file
- * @param[out] data_present         Indicates whether data has been decoded
- * @param[out] finished             Indicates whether the end of file has
- *                                  been reached and all data has been
- *                                  decoded. If this flag is false, there
- *                                  is more data to be decoded, i.e., this
- *                                  function has to be called again.
- * @return Error code (0 if successful)
- */
+/** Decode one audio frame from the input file. */
 static int decode_audio_frame(AVFrame *frame,
                               AVFormatContext *input_format_context,
                               AVCodecContext *input_codec_context,
                               int *data_present, int *finished)
 {
-    /* Packet used for temporary storage. */
+    /** Packet used for temporary storage. */
     AVPacket input_packet;
     int error;
     init_packet(&input_packet);
 
-    /* Read one audio frame from the input file into a temporary packet. */
+    /** Read one audio frame from the input file into a temporary packet. */
     if ((error = av_read_frame(input_format_context, &input_packet)) < 0) {
-        /* If we are at the end of the file, flush the decoder below. */
+        /** If we are at the end of the file, flush the decoder below. */
         if (error == AVERROR_EOF)
             *finished = 1;
         else {
@@ -387,52 +342,34 @@ static int decode_audio_frame(AVFrame *frame,
         }
     }
 
-    /* Send the audio frame stored in the temporary packet to the decoder.
-     * The input audio stream decoder is used to do this. */
-    if ((error = avcodec_send_packet(input_codec_context, &input_packet)) < 0) {
-        fprintf(stderr, "Could not send packet for decoding (error '%s')\n",
+    /**
+     * Decode the audio frame stored in the temporary packet.
+     * The input audio stream decoder is used to do this.
+     * If we are at the end of the file, pass an empty packet to the decoder
+     * to flush it.
+     */
+    if ((error = avcodec_decode_audio4(input_codec_context, frame,
+                                       data_present, &input_packet)) < 0) {
+        fprintf(stderr, "Could not decode frame (error '%s')\n",
                 av_err2str(error));
+        av_packet_unref(&input_packet);
         return error;
     }
 
-    /* Receive one frame from the decoder. */
-    error = avcodec_receive_frame(input_codec_context, frame);
-    /* If the decoder asks for more data to be able to decode a frame,
-     * return indicating that no data is present. */
-    if (error == AVERROR(EAGAIN)) {
-        error = 0;
-        goto cleanup;
-    /* If the end of the input file is reached, stop decoding. */
-    } else if (error == AVERROR_EOF) {
-        *finished = 1;
-        error = 0;
-        goto cleanup;
-    } else if (error < 0) {
-        fprintf(stderr, "Could not decode frame (error '%s')\n",
-                av_err2str(error));
-        goto cleanup;
-    /* Default case: Return decoded data. */
-    } else {
-        *data_present = 1;
-        goto cleanup;
-    }
-
-cleanup:
+    /**
+     * If the decoder has not been flushed completely, we are not finished,
+     * so that this function has to be called again.
+     */
+    if (*finished && *data_present)
+        *finished = 0;
     av_packet_unref(&input_packet);
-    return error;
+    return 0;
 }
 
 /**
  * Initialize a temporary storage for the specified number of audio samples.
  * The conversion requires temporary storage due to the different format.
  * The number of audio samples to be allocated is specified in frame_size.
- * @param[out] converted_input_samples Array of converted samples. The
- *                                     dimensions are reference, channel
- *                                     (for multi-channel audio), sample.
- * @param      output_codec_context    Codec context of the output file
- * @param      frame_size              Number of samples to be converted in
- *                                     each round
- * @return Error code (0 if successful)
  */
 static int init_converted_samples(uint8_t ***converted_input_samples,
                                   AVCodecContext *output_codec_context,
@@ -440,7 +377,8 @@ static int init_converted_samples(uint8_t ***converted_input_samples,
 {
     int error;
 
-    /* Allocate as many pointers as there are audio channels.
+    /**
+     * Allocate as many pointers as there are audio channels.
      * Each pointer will later point to the audio samples of the corresponding
      * channels (although it may be NULL for interleaved formats).
      */
@@ -450,8 +388,10 @@ static int init_converted_samples(uint8_t ***converted_input_samples,
         return AVERROR(ENOMEM);
     }
 
-    /* Allocate memory for the samples of all channels in one consecutive
-     * block for convenience. */
+    /**
+     * Allocate memory for the samples of all channels in one consecutive
+     * block for convenience.
+     */
     if ((error = av_samples_alloc(*converted_input_samples, NULL,
                                   output_codec_context->channels,
                                   frame_size,
@@ -468,15 +408,8 @@ static int init_converted_samples(uint8_t ***converted_input_samples,
 
 /**
  * Convert the input audio samples into the output sample format.
- * The conversion happens on a per-frame basis, the size of which is
- * specified by frame_size.
- * @param      input_data       Samples to be decoded. The dimensions are
- *                              channel (for multi-channel audio), sample.
- * @param[out] converted_data   Converted samples. The dimensions are channel
- *                              (for multi-channel audio), sample.
- * @param      frame_size       Number of samples to be converted
- * @param      resample_context Resample context for the conversion
- * @return Error code (0 if successful)
+ * The conversion happens on a per-frame basis, the size of which is specified
+ * by frame_size.
  */
 static int convert_samples(const uint8_t **input_data,
                            uint8_t **converted_data, const int frame_size,
@@ -484,7 +417,7 @@ static int convert_samples(const uint8_t **input_data,
 {
     int error;
 
-    /* Convert the samples using the resampler. */
+    /** Convert the samples using the resampler. */
     if ((error = swr_convert(resample_context,
                              converted_data, frame_size,
                              input_data    , frame_size)) < 0) {
@@ -496,28 +429,23 @@ static int convert_samples(const uint8_t **input_data,
     return 0;
 }
 
-/**
- * Add converted input audio samples to the FIFO buffer for later processing.
- * @param fifo                    Buffer to add the samples to
- * @param converted_input_samples Samples to be added. The dimensions are channel
- *                                (for multi-channel audio), sample.
- * @param frame_size              Number of samples to be converted
- * @return Error code (0 if successful)
- */
+/** Add converted input audio samples to the FIFO buffer for later processing. */
 static int add_samples_to_fifo(AVAudioFifo *fifo,
                                uint8_t **converted_input_samples,
                                const int frame_size)
 {
     int error;
 
-    /* Make the FIFO as large as it needs to be to hold both,
-     * the old and the new samples. */
+    /**
+     * Make the FIFO as large as it needs to be to hold both,
+     * the old and the new samples.
+     */
     if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
         fprintf(stderr, "Could not reallocate FIFO\n");
         return error;
     }
 
-    /* Store the new samples in the FIFO buffer. */
+    /** Store the new samples in the FIFO buffer. */
     if (av_audio_fifo_write(fifo, (void **)converted_input_samples,
                             frame_size) < frame_size) {
         fprintf(stderr, "Could not write data to FIFO\n");
@@ -527,20 +455,8 @@ static int add_samples_to_fifo(AVAudioFifo *fifo,
 }
 
 /**
- * Read one audio frame from the input file, decode, convert and store
+ * Read one audio frame from the input file, decodes, converts and stores
  * it in the FIFO buffer.
- * @param      fifo                 Buffer used for temporary storage
- * @param      input_format_context Format context of the input file
- * @param      input_codec_context  Codec context of the input file
- * @param      output_codec_context Codec context of the output file
- * @param      resampler_context    Resample context for the conversion
- * @param[out] finished             Indicates whether the end of file has
- *                                  been reached and all data has been
- *                                  decoded. If this flag is false,
- *                                  there is more data to be decoded,
- *                                  i.e., this function has to be called
- *                                  again.
- * @return Error code (0 if successful)
  */
 static int read_decode_convert_and_store(AVAudioFifo *fifo,
                                          AVFormatContext *input_format_context,
@@ -549,41 +465,45 @@ static int read_decode_convert_and_store(AVAudioFifo *fifo,
                                          SwrContext *resampler_context,
                                          int *finished)
 {
-    /* Temporary storage of the input samples of the frame read from the file. */
+    /** Temporary storage of the input samples of the frame read from the file. */
     AVFrame *input_frame = NULL;
-    /* Temporary storage for the converted input samples. */
+    /** Temporary storage for the converted input samples. */
     uint8_t **converted_input_samples = NULL;
-    int data_present = 0;
+    int data_present;
     int ret = AVERROR_EXIT;
 
-    /* Initialize temporary storage for one input frame. */
+    /** Initialize temporary storage for one input frame. */
     if (init_input_frame(&input_frame))
         goto cleanup;
-    /* Decode one frame worth of audio samples. */
+    /** Decode one frame worth of audio samples. */
     if (decode_audio_frame(input_frame, input_format_context,
                            input_codec_context, &data_present, finished))
         goto cleanup;
-    /* If we are at the end of the file and there are no more samples
+    /**
+     * If we are at the end of the file and there are no more samples
      * in the decoder which are delayed, we are actually finished.
-     * This must not be treated as an error. */
-    if (*finished) {
+     * This must not be treated as an error.
+     */
+    if (*finished && !data_present) {
         ret = 0;
         goto cleanup;
     }
-    /* If there is decoded data, convert and store it. */
+    /** If there is decoded data, convert and store it */
     if (data_present) {
-        /* Initialize the temporary storage for the converted input samples. */
+        /** Initialize the temporary storage for the converted input samples. */
         if (init_converted_samples(&converted_input_samples, output_codec_context,
                                    input_frame->nb_samples))
             goto cleanup;
 
-        /* Convert the input samples to the desired output sample format.
-         * This requires a temporary storage provided by converted_input_samples. */
+        /**
+         * Convert the input samples to the desired output sample format.
+         * This requires a temporary storage provided by converted_input_samples.
+         */
         if (convert_samples((const uint8_t**)input_frame->extended_data, converted_input_samples,
                             input_frame->nb_samples, resampler_context))
             goto cleanup;
 
-        /* Add the converted input samples to the FIFO buffer for later processing. */
+        /** Add the converted input samples to the FIFO buffer for later processing. */
         if (add_samples_to_fifo(fifo, converted_input_samples,
                                 input_frame->nb_samples))
             goto cleanup;
@@ -604,10 +524,6 @@ cleanup:
 /**
  * Initialize one input frame for writing to the output file.
  * The frame will be exactly frame_size samples large.
- * @param[out] frame                Frame to be initialized
- * @param      output_codec_context Codec context of the output file
- * @param      frame_size           Size of the frame
- * @return Error code (0 if successful)
  */
 static int init_output_frame(AVFrame **frame,
                              AVCodecContext *output_codec_context,
@@ -615,24 +531,28 @@ static int init_output_frame(AVFrame **frame,
 {
     int error;
 
-    /* Create a new frame to store the audio samples. */
+    /** Create a new frame to store the audio samples. */
     if (!(*frame = av_frame_alloc())) {
         fprintf(stderr, "Could not allocate output frame\n");
         return AVERROR_EXIT;
     }
 
-    /* Set the frame's parameters, especially its size and format.
+    /**
+     * Set the frame's parameters, especially its size and format.
      * av_frame_get_buffer needs this to allocate memory for the
      * audio samples of the frame.
      * Default channel layouts based on the number of channels
-     * are assumed for simplicity. */
+     * are assumed for simplicity.
+     */
     (*frame)->nb_samples     = frame_size;
     (*frame)->channel_layout = output_codec_context->channel_layout;
     (*frame)->format         = output_codec_context->sample_fmt;
     (*frame)->sample_rate    = output_codec_context->sample_rate;
 
-    /* Allocate the samples of the created frame. This call will make
-     * sure that the audio frame can hold as many samples as specified. */
+    /**
+     * Allocate the samples of the created frame. This call will make
+     * sure that the audio frame can hold as many samples as specified.
+     */
     if ((error = av_frame_get_buffer(*frame, 0)) < 0) {
         fprintf(stderr, "Could not allocate output frame samples (error '%s')\n",
                 av_err2str(error));
@@ -643,114 +563,87 @@ static int init_output_frame(AVFrame **frame,
     return 0;
 }
 
-/* Global timestamp for the audio frames. */
+/** Global timestamp for the audio frames */
 static int64_t pts = 0;
 
-/**
- * Encode one frame worth of audio to the output file.
- * @param      frame                 Samples to be encoded
- * @param      output_format_context Format context of the output file
- * @param      output_codec_context  Codec context of the output file
- * @param[out] data_present          Indicates whether data has been
- *                                   encoded
- * @return Error code (0 if successful)
- */
+/** Encode one frame worth of audio to the output file. */
 static int encode_audio_frame(AVFrame *frame,
                               AVFormatContext *output_format_context,
                               AVCodecContext *output_codec_context,
                               int *data_present)
 {
-    /* Packet used for temporary storage. */
+    /** Packet used for temporary storage. */
     AVPacket output_packet;
     int error;
     init_packet(&output_packet);
 
-    /* Set a timestamp based on the sample rate for the container. */
+    /** Set a timestamp based on the sample rate for the container. */
     if (frame) {
         frame->pts = pts;
         pts += frame->nb_samples;
     }
 
-    /* Send the audio frame stored in the temporary packet to the encoder.
-     * The output audio stream encoder is used to do this. */
-    error = avcodec_send_frame(output_codec_context, frame);
-    /* The encoder signals that it has nothing more to encode. */
-    if (error == AVERROR_EOF) {
-        error = 0;
-        goto cleanup;
-    } else if (error < 0) {
-        fprintf(stderr, "Could not send packet for encoding (error '%s')\n",
+    /**
+     * Encode the audio frame and store it in the temporary packet.
+     * The output audio stream encoder is used to do this.
+     */
+    if ((error = avcodec_encode_audio2(output_codec_context, &output_packet,
+                                       frame, data_present)) < 0) {
+        fprintf(stderr, "Could not encode frame (error '%s')\n",
                 av_err2str(error));
+        av_packet_unref(&output_packet);
         return error;
     }
 
-    /* Receive one encoded frame from the encoder. */
-    error = avcodec_receive_packet(output_codec_context, &output_packet);
-    /* If the encoder asks for more data to be able to provide an
-     * encoded frame, return indicating that no data is present. */
-    if (error == AVERROR(EAGAIN)) {
-        error = 0;
-        goto cleanup;
-    /* If the last frame has been encoded, stop encoding. */
-    } else if (error == AVERROR_EOF) {
-        error = 0;
-        goto cleanup;
-    } else if (error < 0) {
-        fprintf(stderr, "Could not encode frame (error '%s')\n",
-                av_err2str(error));
-        goto cleanup;
-    /* Default case: Return encoded data. */
-    } else {
-        *data_present = 1;
+    /** Write one audio frame from the temporary packet to the output file. */
+    if (*data_present) {
+        if ((error = av_write_frame(output_format_context, &output_packet)) < 0) {
+            fprintf(stderr, "Could not write frame (error '%s')\n",
+                    av_err2str(error));
+            av_packet_unref(&output_packet);
+            return error;
+        }
+
+        av_packet_unref(&output_packet);
     }
 
-    /* Write one audio frame from the temporary packet to the output file. */
-    if (*data_present &&
-        (error = av_write_frame(output_format_context, &output_packet)) < 0) {
-        fprintf(stderr, "Could not write frame (error '%s')\n",
-                av_err2str(error));
-        goto cleanup;
-    }
-
-cleanup:
-    av_packet_unref(&output_packet);
-    return error;
+    return 0;
 }
 
 /**
  * Load one audio frame from the FIFO buffer, encode and write it to the
  * output file.
- * @param fifo                  Buffer used for temporary storage
- * @param output_format_context Format context of the output file
- * @param output_codec_context  Codec context of the output file
- * @return Error code (0 if successful)
  */
 static int load_encode_and_write(AVAudioFifo *fifo,
                                  AVFormatContext *output_format_context,
                                  AVCodecContext *output_codec_context)
 {
-    /* Temporary storage of the output samples of the frame written to the file. */
+    /** Temporary storage of the output samples of the frame written to the file. */
     AVFrame *output_frame;
-    /* Use the maximum number of possible samples per frame.
+    /**
+     * Use the maximum number of possible samples per frame.
      * If there is less than the maximum possible frame size in the FIFO
-     * buffer use this number. Otherwise, use the maximum possible frame size. */
+     * buffer use this number. Otherwise, use the maximum possible frame size
+     */
     const int frame_size = FFMIN(av_audio_fifo_size(fifo),
                                  output_codec_context->frame_size);
     int data_written;
 
-    /* Initialize temporary storage for one output frame. */
+    /** Initialize temporary storage for one output frame. */
     if (init_output_frame(&output_frame, output_codec_context, frame_size))
         return AVERROR_EXIT;
 
-    /* Read as many samples from the FIFO buffer as required to fill the frame.
-     * The samples are stored in the frame temporarily. */
+    /**
+     * Read as many samples from the FIFO buffer as required to fill the frame.
+     * The samples are stored in the frame temporarily.
+     */
     if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) {
         fprintf(stderr, "Could not read data from FIFO\n");
         av_frame_free(&output_frame);
         return AVERROR_EXIT;
     }
 
-    /* Encode one frame worth of audio samples. */
+    /** Encode one frame worth of audio samples. */
     if (encode_audio_frame(output_frame, output_format_context,
                            output_codec_context, &data_written)) {
         av_frame_free(&output_frame);
@@ -760,11 +653,7 @@ static int load_encode_and_write(AVAudioFifo *fifo,
     return 0;
 }
 
-/**
- * Write the trailer of the output file container.
- * @param output_format_context Format context of the output file
- * @return Error code (0 if successful)
- */
+/** Write the trailer of the output file container. */
 static int write_output_file_trailer(AVFormatContext *output_format_context)
 {
     int error;
@@ -776,6 +665,7 @@ static int write_output_file_trailer(AVFormatContext *output_format_context)
     return 0;
 }
 
+/** Convert an audio file to an AAC file in an MP4 container. */
 int main(int argc, char **argv)
 {
     AVFormatContext *input_format_context = NULL, *output_format_context = NULL;
@@ -784,75 +674,90 @@ int main(int argc, char **argv)
     AVAudioFifo *fifo = NULL;
     int ret = AVERROR_EXIT;
 
-    if (argc != 3) {
+    if (argc < 3) {
         fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
         exit(1);
     }
 
-    /* Open the input file for reading. */
+    /** Register all codecs and formats so that they can be used. */
+    av_register_all();
+    /** Open the input file for reading. */
     if (open_input_file(argv[1], &input_format_context,
                         &input_codec_context))
         goto cleanup;
-    /* Open the output file for writing. */
+    /** Open the output file for writing. */
     if (open_output_file(argv[2], input_codec_context,
                          &output_format_context, &output_codec_context))
         goto cleanup;
-    /* Initialize the resampler to be able to convert audio sample formats. */
+    /** Initialize the resampler to be able to convert audio sample formats. */
     if (init_resampler(input_codec_context, output_codec_context,
                        &resample_context))
         goto cleanup;
-    /* Initialize the FIFO buffer to store audio samples to be encoded. */
+    /** Initialize the FIFO buffer to store audio samples to be encoded. */
     if (init_fifo(&fifo, output_codec_context))
         goto cleanup;
-    /* Write the header of the output file container. */
+    /** Write the header of the output file container. */
     if (write_output_file_header(output_format_context))
         goto cleanup;
 
-    /* Loop as long as we have input samples to read or output samples
-     * to write; abort as soon as we have neither. */
+    /**
+     * Loop as long as we have input samples to read or output samples
+     * to write; abort as soon as we have neither.
+     */
     while (1) {
-        /* Use the encoder's desired frame size for processing. */
+        /** Use the encoder's desired frame size for processing. */
         const int output_frame_size = output_codec_context->frame_size;
         int finished                = 0;
 
-        /* Make sure that there is one frame worth of samples in the FIFO
+        /**
+         * Make sure that there is one frame worth of samples in the FIFO
          * buffer so that the encoder can do its work.
          * Since the decoder's and the encoder's frame size may differ, we
          * need to FIFO buffer to store as many frames worth of input samples
-         * that they make up at least one frame worth of output samples. */
+         * that they make up at least one frame worth of output samples.
+         */
         while (av_audio_fifo_size(fifo) < output_frame_size) {
-            /* Decode one frame worth of audio samples, convert it to the
-             * output sample format and put it into the FIFO buffer. */
+            /**
+             * Decode one frame worth of audio samples, convert it to the
+             * output sample format and put it into the FIFO buffer.
+             */
             if (read_decode_convert_and_store(fifo, input_format_context,
                                               input_codec_context,
                                               output_codec_context,
                                               resample_context, &finished))
                 goto cleanup;
 
-            /* If we are at the end of the input file, we continue
-             * encoding the remaining audio samples to the output file. */
+            /**
+             * If we are at the end of the input file, we continue
+             * encoding the remaining audio samples to the output file.
+             */
             if (finished)
                 break;
         }
 
-        /* If we have enough samples for the encoder, we encode them.
+        /**
+         * If we have enough samples for the encoder, we encode them.
          * At the end of the file, we pass the remaining samples to
-         * the encoder. */
+         * the encoder.
+         */
         while (av_audio_fifo_size(fifo) >= output_frame_size ||
                (finished && av_audio_fifo_size(fifo) > 0))
-            /* Take one frame worth of audio samples from the FIFO buffer,
-             * encode it and write it to the output file. */
+            /**
+             * Take one frame worth of audio samples from the FIFO buffer,
+             * encode it and write it to the output file.
+             */
             if (load_encode_and_write(fifo, output_format_context,
                                       output_codec_context))
                 goto cleanup;
 
-        /* If we are at the end of the input file and have encoded
-         * all remaining samples, we can exit this loop and finish. */
+        /**
+         * If we are at the end of the input file and have encoded
+         * all remaining samples, we can exit this loop and finish.
+         */
         if (finished) {
             int data_written;
-            /* Flush the encoder as it may have delayed frames. */
+            /** 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;
@@ -861,7 +766,7 @@ int main(int argc, char **argv)
         }
     }
 
-    /* Write the trailer of the output file container. */
+    /** Write the trailer of the output file container. */
     if (write_output_file_trailer(output_format_context))
         goto cleanup;
     ret = 0;

doc/examples/transcoding.c

@@ -30,6 +30,7 @@
 
 #include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
+#include <libavfilter/avfiltergraph.h>
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
 #include <libavutil/opt.h>
@@ -172,9 +173,6 @@ static int open_output_file(const char *filename)
                 enc_ctx->time_base = (AVRational){1, enc_ctx->sample_rate};
             }
 
-            if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
-                enc_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
-
             /* Third parameter can be used to pass settings to encoder */
             ret = avcodec_open2(enc_ctx, encoder, NULL);
             if (ret < 0) {
@@ -186,6 +184,8 @@ static int open_output_file(const char *filename)
                 av_log(NULL, AV_LOG_ERROR, "Failed to copy encoder parameters to output stream #%u\n", i);
                 return ret;
             }
+            if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
+                enc_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
 
             out_stream->time_base = enc_ctx->time_base;
             stream_ctx[i].enc_ctx = enc_ctx;
@@ -228,8 +228,8 @@ static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
 {
     char args[512];
     int ret = 0;
-    const AVFilter *buffersrc = NULL;
-    const AVFilter *buffersink = NULL;
+    AVFilter *buffersrc = NULL;
+    AVFilter *buffersink = NULL;
     AVFilterContext *buffersrc_ctx = NULL;
     AVFilterContext *buffersink_ctx = NULL;
     AVFilterInOut *outputs = avfilter_inout_alloc();
@@ -518,6 +518,9 @@ int main(int argc, char **argv)
         return 1;
     }
 
+    av_register_all();
+    avfilter_register_all();
+
     if ((ret = open_input_file(argv[1])) < 0)
         goto end;
     if ((ret = open_output_file(argv[2])) < 0)

doc/examples/vaapi_encode.c

@@ -1,224 +0,0 @@
-/*
- * Video Acceleration API (video encoding) encode sample
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to deal
- * in the Software without restriction, including without limitation the rights
- * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- * copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
- * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
- * THE SOFTWARE.
- */
-
-/**
- * @file
- * Intel VAAPI-accelerated encoding example.
- *
- * @example vaapi_encode.c
- * This example shows how to do VAAPI-accelerated encoding. now only support NV12
- * raw file, usage like: vaapi_encode 1920 1080 input.yuv output.h264
- *
- */
-
-#include <stdio.h>
-#include <string.h>
-#include <errno.h>
-
-#include <libavcodec/avcodec.h>
-#include <libavutil/pixdesc.h>
-#include <libavutil/hwcontext.h>
-
-static int width, height;
-static AVBufferRef *hw_device_ctx = NULL;
-
-static int set_hwframe_ctx(AVCodecContext *ctx, AVBufferRef *hw_device_ctx)
-{
-    AVBufferRef *hw_frames_ref;
-    AVHWFramesContext *frames_ctx = NULL;
-    int err = 0;
-
-    if (!(hw_frames_ref = av_hwframe_ctx_alloc(hw_device_ctx))) {
-        fprintf(stderr, "Failed to create VAAPI frame context.\n");
-        return -1;
-    }
-    frames_ctx = (AVHWFramesContext *)(hw_frames_ref->data);
-    frames_ctx->format    = AV_PIX_FMT_VAAPI;
-    frames_ctx->sw_format = AV_PIX_FMT_NV12;
-    frames_ctx->width     = width;
-    frames_ctx->height    = height;
-    frames_ctx->initial_pool_size = 20;
-    if ((err = av_hwframe_ctx_init(hw_frames_ref)) < 0) {
-        fprintf(stderr, "Failed to initialize VAAPI frame context."
-                "Error code: %s\n",av_err2str(err));
-        av_buffer_unref(&hw_frames_ref);
-        return err;
-    }
-    ctx->hw_frames_ctx = av_buffer_ref(hw_frames_ref);
-    if (!ctx->hw_frames_ctx)
-        err = AVERROR(ENOMEM);
-
-    av_buffer_unref(&hw_frames_ref);
-    return err;
-}
-
-static int encode_write(AVCodecContext *avctx, AVFrame *frame, FILE *fout)
-{
-    int ret = 0;
-    AVPacket enc_pkt;
-
-    av_init_packet(&enc_pkt);
-    enc_pkt.data = NULL;
-    enc_pkt.size = 0;
-
-    if ((ret = avcodec_send_frame(avctx, frame)) < 0) {
-        fprintf(stderr, "Error code: %s\n", av_err2str(ret));
-        goto end;
-    }
-    while (1) {
-        ret = avcodec_receive_packet(avctx, &enc_pkt);
-        if (ret)
-            break;
-
-        enc_pkt.stream_index = 0;
-        ret = fwrite(enc_pkt.data, enc_pkt.size, 1, fout);
-        av_packet_unref(&enc_pkt);
-    }
-
-end:
-    ret = ((ret == AVERROR(EAGAIN)) ? 0 : -1);
-    return ret;
-}
-
-int main(int argc, char *argv[])
-{
-    int size, err;
-    FILE *fin = NULL, *fout = NULL;
-    AVFrame *sw_frame = NULL, *hw_frame = NULL;
-    AVCodecContext *avctx = NULL;
-    AVCodec *codec = NULL;
-    const char *enc_name = "h264_vaapi";
-
-    if (argc < 5) {
-        fprintf(stderr, "Usage: %s <width> <height> <input file> <output file>\n", argv[0]);
-        return -1;
-    }
-
-    width  = atoi(argv[1]);
-    height = atoi(argv[2]);
-    size   = width * height;
-
-    if (!(fin = fopen(argv[3], "r"))) {
-        fprintf(stderr, "Fail to open input file : %s\n", strerror(errno));
-        return -1;
-    }
-    if (!(fout = fopen(argv[4], "w+b"))) {
-        fprintf(stderr, "Fail to open output file : %s\n", strerror(errno));
-        err = -1;
-        goto close;
-    }
-
-    err = av_hwdevice_ctx_create(&hw_device_ctx, AV_HWDEVICE_TYPE_VAAPI,
-                                 NULL, NULL, 0);
-    if (err < 0) {
-        fprintf(stderr, "Failed to create a VAAPI device. Error code: %s\n", av_err2str(err));
-        goto close;
-    }
-
-    if (!(codec = avcodec_find_encoder_by_name(enc_name))) {
-        fprintf(stderr, "Could not find encoder.\n");
-        err = -1;
-        goto close;
-    }
-
-    if (!(avctx = avcodec_alloc_context3(codec))) {
-        err = AVERROR(ENOMEM);
-        goto close;
-    }
-
-    avctx->width     = width;
-    avctx->height    = height;
-    avctx->time_base = (AVRational){1, 25};
-    avctx->framerate = (AVRational){25, 1};
-    avctx->sample_aspect_ratio = (AVRational){1, 1};
-    avctx->pix_fmt   = AV_PIX_FMT_VAAPI;
-
-    /* set hw_frames_ctx for encoder's AVCodecContext */
-    if ((err = set_hwframe_ctx(avctx, hw_device_ctx)) < 0) {
-        fprintf(stderr, "Failed to set hwframe context.\n");
-        goto close;
-    }
-
-    if ((err = avcodec_open2(avctx, codec, NULL)) < 0) {
-        fprintf(stderr, "Cannot open video encoder codec. Error code: %s\n", av_err2str(err));
-        goto close;
-    }
-
-    while (1) {
-        if (!(sw_frame = av_frame_alloc())) {
-            err = AVERROR(ENOMEM);
-            goto close;
-        }
-        /* read data into software frame, and transfer them into hw frame */
-        sw_frame->width  = width;
-        sw_frame->height = height;
-        sw_frame->format = AV_PIX_FMT_NV12;
-        if ((err = av_frame_get_buffer(sw_frame, 32)) < 0)
-            goto close;
-        if ((err = fread((uint8_t*)(sw_frame->data[0]), size, 1, fin)) <= 0)
-            break;
-        if ((err = fread((uint8_t*)(sw_frame->data[1]), size/2, 1, fin)) <= 0)
-            break;
-
-        if (!(hw_frame = av_frame_alloc())) {
-            err = AVERROR(ENOMEM);
-            goto close;
-        }
-        if ((err = av_hwframe_get_buffer(avctx->hw_frames_ctx, hw_frame, 0)) < 0) {
-            fprintf(stderr, "Error code: %s.\n", av_err2str(err));
-            goto close;
-        }
-        if (!hw_frame->hw_frames_ctx) {
-            err = AVERROR(ENOMEM);
-            goto close;
-        }
-        if ((err = av_hwframe_transfer_data(hw_frame, sw_frame, 0)) < 0) {
-            fprintf(stderr, "Error while transferring frame data to surface."
-                    "Error code: %s.\n", av_err2str(err));
-            goto close;
-        }
-
-        if ((err = (encode_write(avctx, hw_frame, fout))) < 0) {
-            fprintf(stderr, "Failed to encode.\n");
-            goto close;
-        }
-        av_frame_free(&hw_frame);
-        av_frame_free(&sw_frame);
-    }
-
-    /* flush encoder */
-    err = encode_write(avctx, NULL, fout);
-    if (err == AVERROR_EOF)
-        err = 0;
-
-close:
-    if (fin)
-        fclose(fin);
-    if (fout)
-        fclose(fout);
-    av_frame_free(&sw_frame);
-    av_frame_free(&hw_frame);
-    avcodec_free_context(&avctx);
-    av_buffer_unref(&hw_device_ctx);
-
-    return err;
-}

doc/examples/vaapi_transcode.c

@@ -1,306 +0,0 @@
-/*
- * Video Acceleration API (video transcoding) transcode sample
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to deal
- * in the Software without restriction, including without limitation the rights
- * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- * copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
- * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
- * THE SOFTWARE.
- */
-
-/**
- * @file
- * Intel VAAPI-accelerated transcoding example.
- *
- * @example vaapi_transcode.c
- * This example shows how to do VAAPI-accelerated transcoding.
- * Usage: vaapi_transcode input_stream codec output_stream
- * e.g: - vaapi_transcode input.mp4 h264_vaapi output_h264.mp4
- *      - vaapi_transcode input.mp4 vp9_vaapi output_vp9.ivf
- */
-
-#include <stdio.h>
-#include <errno.h>
-
-#include <libavutil/hwcontext.h>
-#include <libavcodec/avcodec.h>
-#include <libavformat/avformat.h>
-
-static AVFormatContext *ifmt_ctx = NULL, *ofmt_ctx = NULL;
-static AVBufferRef *hw_device_ctx = NULL;
-static AVCodecContext *decoder_ctx = NULL, *encoder_ctx = NULL;
-static int video_stream = -1;
-static AVStream *ost;
-static int initialized = 0;
-
-static enum AVPixelFormat get_vaapi_format(AVCodecContext *ctx,
-                                           const enum AVPixelFormat *pix_fmts)
-{
-    const enum AVPixelFormat *p;
-
-    for (p = pix_fmts; *p != AV_PIX_FMT_NONE; p++) {
-        if (*p == AV_PIX_FMT_VAAPI)
-            return *p;
-    }
-
-    fprintf(stderr, "Unable to decode this file using VA-API.\n");
-    return AV_PIX_FMT_NONE;
-}
-
-static int open_input_file(const char *filename)
-{
-    int ret;
-    AVCodec *decoder = NULL;
-    AVStream *video = NULL;
-
-    if ((ret = avformat_open_input(&ifmt_ctx, filename, NULL, NULL)) < 0) {
-        fprintf(stderr, "Cannot open input file '%s', Error code: %s\n",
-                filename, av_err2str(ret));
-        return ret;
-    }
-
-    if ((ret = avformat_find_stream_info(ifmt_ctx, NULL)) < 0) {
-        fprintf(stderr, "Cannot find input stream information. Error code: %s\n",
-                av_err2str(ret));
-        return ret;
-    }
-
-    ret = av_find_best_stream(ifmt_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, &decoder, 0);
-    if (ret < 0) {
-        fprintf(stderr, "Cannot find a video stream in the input file. "
-                "Error code: %s\n", av_err2str(ret));
-        return ret;
-    }
-    video_stream = ret;
-
-    if (!(decoder_ctx = avcodec_alloc_context3(decoder)))
-        return AVERROR(ENOMEM);
-
-    video = ifmt_ctx->streams[video_stream];
-    if ((ret = avcodec_parameters_to_context(decoder_ctx, video->codecpar)) < 0) {
-        fprintf(stderr, "avcodec_parameters_to_context error. Error code: %s\n",
-                av_err2str(ret));
-        return ret;
-    }
-
-    decoder_ctx->hw_device_ctx = av_buffer_ref(hw_device_ctx);
-    if (!decoder_ctx->hw_device_ctx) {
-        fprintf(stderr, "A hardware device reference create failed.\n");
-        return AVERROR(ENOMEM);
-    }
-    decoder_ctx->get_format    = get_vaapi_format;
-
-    if ((ret = avcodec_open2(decoder_ctx, decoder, NULL)) < 0)
-        fprintf(stderr, "Failed to open codec for decoding. Error code: %s\n",
-                av_err2str(ret));
-
-    return ret;
-}
-
-static int encode_write(AVFrame *frame)
-{
-    int ret = 0;
-    AVPacket enc_pkt;
-
-    av_init_packet(&enc_pkt);
-    enc_pkt.data = NULL;
-    enc_pkt.size = 0;
-
-    if ((ret = avcodec_send_frame(encoder_ctx, frame)) < 0) {
-        fprintf(stderr, "Error during encoding. Error code: %s\n", av_err2str(ret));
-        goto end;
-    }
-    while (1) {
-        ret = avcodec_receive_packet(encoder_ctx, &enc_pkt);
-        if (ret)
-            break;
-
-        enc_pkt.stream_index = 0;
-        av_packet_rescale_ts(&enc_pkt, ifmt_ctx->streams[video_stream]->time_base,
-                             ofmt_ctx->streams[0]->time_base);
-        ret = av_interleaved_write_frame(ofmt_ctx, &enc_pkt);
-        if (ret < 0) {
-            fprintf(stderr, "Error during writing data to output file. "
-                    "Error code: %s\n", av_err2str(ret));
-            return -1;
-        }
-    }
-
-end:
-    if (ret == AVERROR_EOF)
-        return 0;
-    ret = ((ret == AVERROR(EAGAIN)) ? 0:-1);
-    return ret;
-}
-
-static int dec_enc(AVPacket *pkt, AVCodec *enc_codec)
-{
-    AVFrame *frame;
-    int ret = 0;
-
-    ret = avcodec_send_packet(decoder_ctx, pkt);
-    if (ret < 0) {
-        fprintf(stderr, "Error during decoding. Error code: %s\n", av_err2str(ret));
-        return ret;
-    }
-
-    while (ret >= 0) {
-        if (!(frame = av_frame_alloc()))
-            return AVERROR(ENOMEM);
-
-        ret = avcodec_receive_frame(decoder_ctx, frame);
-        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
-            av_frame_free(&frame);
-            return 0;
-        } else if (ret < 0) {
-            fprintf(stderr, "Error while decoding. Error code: %s\n", av_err2str(ret));
-            goto fail;
-        }
-
-        if (!initialized) {
-            /* we need to ref hw_frames_ctx of decoder to initialize encoder's codec.
-               Only after we get a decoded frame, can we obtain its hw_frames_ctx */
-            encoder_ctx->hw_frames_ctx = av_buffer_ref(decoder_ctx->hw_frames_ctx);
-            if (!encoder_ctx->hw_frames_ctx) {
-                ret = AVERROR(ENOMEM);
-                goto fail;
-            }
-            /* set AVCodecContext Parameters for encoder, here we keep them stay
-             * the same as decoder.
-             * xxx: now the sample can't handle resolution change case.
-             */
-            encoder_ctx->time_base = av_inv_q(decoder_ctx->framerate);
-            encoder_ctx->pix_fmt   = AV_PIX_FMT_VAAPI;
-            encoder_ctx->width     = decoder_ctx->width;
-            encoder_ctx->height    = decoder_ctx->height;
-
-            if ((ret = avcodec_open2(encoder_ctx, enc_codec, NULL)) < 0) {
-                fprintf(stderr, "Failed to open encode codec. Error code: %s\n",
-                        av_err2str(ret));
-                goto fail;
-            }
-
-            if (!(ost = avformat_new_stream(ofmt_ctx, enc_codec))) {
-                fprintf(stderr, "Failed to allocate stream for output format.\n");
-                ret = AVERROR(ENOMEM);
-                goto fail;
-            }
-
-            ost->time_base = encoder_ctx->time_base;
-            ret = avcodec_parameters_from_context(ost->codecpar, encoder_ctx);
-            if (ret < 0) {
-                fprintf(stderr, "Failed to copy the stream parameters. "
-                        "Error code: %s\n", av_err2str(ret));
-                goto fail;
-            }
-
-            /* write the stream header */
-            if ((ret = avformat_write_header(ofmt_ctx, NULL)) < 0) {
-                fprintf(stderr, "Error while writing stream header. "
-                        "Error code: %s\n", av_err2str(ret));
-                goto fail;
-            }
-
-            initialized = 1;
-        }
-
-        if ((ret = encode_write(frame)) < 0)
-            fprintf(stderr, "Error during encoding and writing.\n");
-
-fail:
-        av_frame_free(&frame);
-        if (ret < 0)
-            return ret;
-    }
-    return 0;
-}
-
-int main(int argc, char **argv)
-{
-    int ret = 0;
-    AVPacket dec_pkt;
-    AVCodec *enc_codec;
-
-    if (argc != 4) {
-        fprintf(stderr, "Usage: %s <input file> <encode codec> <output file>\n"
-                "The output format is guessed according to the file extension.\n"
-                "\n", argv[0]);
-        return -1;
-    }
-
-    ret = av_hwdevice_ctx_create(&hw_device_ctx, AV_HWDEVICE_TYPE_VAAPI, NULL, NULL, 0);
-    if (ret < 0) {
-        fprintf(stderr, "Failed to create a VAAPI device. Error code: %s\n", av_err2str(ret));
-        return -1;
-    }
-
-    if ((ret = open_input_file(argv[1])) < 0)
-        goto end;
-
-    if (!(enc_codec = avcodec_find_encoder_by_name(argv[2]))) {
-        fprintf(stderr, "Could not find encoder '%s'\n", argv[2]);
-        ret = -1;
-        goto end;
-    }
-
-    if ((ret = (avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, argv[3]))) < 0) {
-        fprintf(stderr, "Failed to deduce output format from file extension. Error code: "
-                "%s\n", av_err2str(ret));
-        goto end;
-    }
-
-    if (!(encoder_ctx = avcodec_alloc_context3(enc_codec))) {
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-
-    ret = avio_open(&ofmt_ctx->pb, argv[3], AVIO_FLAG_WRITE);
-    if (ret < 0) {
-        fprintf(stderr, "Cannot open output file. "
-                "Error code: %s\n", av_err2str(ret));
-        goto end;
-    }
-
-    /* read all packets and only transcoding video */
-    while (ret >= 0) {
-        if ((ret = av_read_frame(ifmt_ctx, &dec_pkt)) < 0)
-            break;
-
-        if (video_stream == dec_pkt.stream_index)
-            ret = dec_enc(&dec_pkt, enc_codec);
-
-        av_packet_unref(&dec_pkt);
-    }
-
-    /* flush decoder */
-    dec_pkt.data = NULL;
-    dec_pkt.size = 0;
-    ret = dec_enc(&dec_pkt, enc_codec);
-    av_packet_unref(&dec_pkt);
-
-    /* flush encoder */
-    ret = encode_write(NULL);
-
-    /* write the trailer for output stream */
-    av_write_trailer(ofmt_ctx);
-
-end:
-    avformat_close_input(&ifmt_ctx);
-    avformat_close_input(&ofmt_ctx);
-    avcodec_free_context(&decoder_ctx);
-    avcodec_free_context(&encoder_ctx);
-    av_buffer_unref(&hw_device_ctx);
-    return ret;
-}

doc/faq.texi

@@ -76,7 +76,7 @@ the gcc developers. Note that we will not add workarounds for gcc bugs.
 
 Also note that (some of) the gcc developers believe this is not a bug or
 not a bug they should fix:
-@url{https://gcc.gnu.org/bugzilla/show_bug.cgi?id=11203}.
+@url{http://gcc.gnu.org/bugzilla/show_bug.cgi?id=11203}.
 Then again, some of them do not know the difference between an undecidable
 problem and an NP-hard problem...
 
@@ -257,13 +257,13 @@ default.
 @section Which are good parameters for encoding high quality MPEG-4?
 
 '-mbd rd -flags +mv4+aic -trellis 2 -cmp 2 -subcmp 2 -g 300 -pass 1/2',
-things to try: '-bf 2', '-mpv_flags qp_rd', '-mpv_flags mv0', '-mpv_flags skip_rd'.
+things to try: '-bf 2', '-flags qprd', '-flags mv0', '-flags skiprd'.
 
 @section Which are good parameters for encoding high quality MPEG-1/MPEG-2?
 
 '-mbd rd -trellis 2 -cmp 2 -subcmp 2 -g 100 -pass 1/2'
 but beware the '-g 100' might cause problems with some decoders.
-Things to try: '-bf 2', '-mpv_flags qp_rd', '-mpv_flags mv0', '-mpv_flags skip_rd'.
+Things to try: '-bf 2', '-flags qprd', '-flags mv0', '-flags skiprd.
 
 @section Interlaced video looks very bad when encoded with ffmpeg, what is wrong?
 
@@ -501,71 +501,6 @@ ffmpeg -i ega_screen.nut -vf setdar=4/3 ega_screen_anamorphic.nut
 ffmpeg -i ega_screen.nut -aspect 4/3 -c copy ega_screen_overridden.nut
 @end example
 
-@anchor{background task}
-@section How do I run ffmpeg as a background task?
-
-ffmpeg normally checks the console input, for entries like "q" to stop
-and "?" to give help, while performing operations. ffmpeg does not have a way of
-detecting when it is running as a background task.
-When it checks the console input, that can cause the process running ffmpeg
-in the background to suspend.
-
-To prevent those input checks, allowing ffmpeg to run as a background task,
-use the @url{ffmpeg.html#stdin-option, @code{-nostdin} option}
-in the ffmpeg invocation. This is effective whether you run ffmpeg in a shell
-or invoke ffmpeg in its own process via an operating system API.
-
-As an alternative, when you are running ffmpeg in a shell, you can redirect
-standard input to @code{/dev/null} (on Linux and macOS)
-or @code{NUL} (on Windows). You can do this redirect either
-on the ffmpeg invocation, or from a shell script which calls ffmpeg.
-
-For example:
-
-@example
-ffmpeg -nostdin -i INPUT OUTPUT
-@end example
-
-or (on Linux, macOS, and other UNIX-like shells):
-
-@example
-ffmpeg -i INPUT OUTPUT </dev/null
-@end example
-
-or (on Windows):
-
-@example
-ffmpeg -i INPUT OUTPUT <NUL
-@end example
-
-@section How do I prevent ffmpeg from suspending with a message like @emph{suspended (tty output)}?
-
-If you run ffmpeg in the background, you may find that its process suspends.
-There may be a message like @emph{suspended (tty output)}. The question is how
-to prevent the process from being suspended.
-
-For example:
-
-@example
-% ffmpeg -i INPUT OUTPUT &> ~/tmp/log.txt &
-[1] 93352
-%
-[1]  + suspended (tty output)  ffmpeg -i INPUT OUTPUT &>
-@end example
-
-The message "tty output" notwithstanding, the problem here is that
-ffmpeg normally checks the console input when it runs. The operating system
-detects this, and suspends the process until you can bring it to the
-foreground and attend to it.
-
-The solution is to use the right techniques to tell ffmpeg not to consult
-console input. You can use the
-@url{ffmpeg.html#stdin-option, @code{-nostdin} option},
-or redirect standard input with @code{< /dev/null}.
-See FAQ
-@ref{background task, @emph{How do I run ffmpeg as a background task?}}
-for details.
-
 @chapter Development
 
 @section Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat?
@@ -601,7 +536,7 @@ No. These tools are too bloated and they complicate the build.
 FFmpeg is already organized in a highly modular manner and does not need to
 be rewritten in a formal object language. Further, many of the developers
 favor straight C; it works for them. For more arguments on this matter,
-read @uref{https://web.archive.org/web/20111004021423/http://kernel.org/pub/linux/docs/lkml/#s15, "Programming Religion"}.
+read @uref{http://www.tux.org/lkml/#s15, "Programming Religion"}.
 
 @section Why are the ffmpeg programs devoid of debugging symbols?
 

doc/fate.texi

@@ -147,28 +147,6 @@ process.
 The only thing left is to automate the execution of the fate.sh script and
 the synchronisation of the samples directory.
 
-@chapter Uploading new samples to the fate suite
-
-This is for developers who have an account on the fate suite server.
-If you upload new samples, please make sure they are as small as possible,
-space on each client, network bandwidth and so on benefit from smaller test cases.
-Also keep in mind older checkouts use existing sample files, that means in
-practice generally do not replace, remove or overwrite files as it likely would
-break older checkouts or releases.
-Also all needed samples for a commit should be uploaded, ideally 24
-hours, before the push.
-
-@example
-#First update your local samples copy:
-rsync -vauL --chmod=Dg+s,Duo+x,ug+rw,o+r,o-w,+X fate-suite.ffmpeg.org:/home/samples/fate-suite/ ~/fate-suite
-
-#Then do a dry run checking what would be uploaded:
-rsync -vanL --no-g --chmod=Dg+s,Duo+x,ug+rw,o+r,o-w,+X ~/fate-suite/ fate-suite.ffmpeg.org:/home/samples/fate-suite
-
-#Upload the files:
-rsync -vaL  --no-g --chmod=Dg+s,Duo+x,ug+rw,o+r,o-w,+X ~/fate-suite/ fate-suite.ffmpeg.org:/home/samples/fate-suite
-@end example
-
 
 @chapter FATE makefile targets and variables
 
@@ -224,11 +202,6 @@ Set to @samp{1} to generate the missing or mismatched references.
 Specify which hardware acceleration to use while running regression tests,
 by default @samp{none} is used.
 
-@item KEEP
-Set to @samp{1} to keep temp files generated by fate test(s) when test is successful.
-Default is @samp{0}, which removes these files. Files are always kept when a test
-fails.
-
 @end table
 
 @section Examples

doc/ffmpeg-bitstream-filters.texi

@@ -26,12 +26,12 @@ bitstream level modifications without performing decoding.
 @chapter See Also
 
 @ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
 @url{libavcodec.html,libavcodec}
 @end ifhtml
 
 @ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), libavcodec(3)
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavcodec(3)
 @end ifnothtml
 
 @include authors.texi

doc/ffmpeg-codecs.texi

@@ -23,12 +23,12 @@ the libavcodec library.
 @chapter See Also
 
 @ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
 @url{libavcodec.html,libavcodec}
 @end ifhtml
 
 @ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), libavcodec(3)
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavcodec(3)
 @end ifnothtml
 
 @include authors.texi

doc/ffmpeg-devices.texi

@@ -23,12 +23,12 @@ libavdevice library.
 @chapter See Also
 
 @ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
 @url{libavdevice.html,libavdevice}
 @end ifhtml
 
 @ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), libavdevice(3)
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavdevice(3)
 @end ifnothtml
 
 @include authors.texi

doc/ffmpeg-filters.texi

@@ -23,12 +23,12 @@ libavfilter library.
 @chapter See Also
 
 @ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
 @url{libavfilter.html,libavfilter}
 @end ifhtml
 
 @ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), libavfilter(3)
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavfilter(3)
 @end ifnothtml
 
 @include authors.texi

doc/ffmpeg-formats.texi

@@ -23,12 +23,12 @@ provided by the libavformat library.
 @chapter See Also
 
 @ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
 @url{libavformat.html,libavformat}
 @end ifhtml
 
 @ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), libavformat(3)
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavformat(3)
 @end ifnothtml
 
 @include authors.texi

doc/ffmpeg-protocols.texi

@@ -23,12 +23,12 @@ libavformat library.
 @chapter See Also
 
 @ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
 @url{libavformat.html,libavformat}
 @end ifhtml
 
 @ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), libavformat(3)
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavformat(3)
 @end ifnothtml
 
 @include authors.texi

doc/ffmpeg-resampler.texi

@@ -25,12 +25,12 @@ and convert audio format and packing layout.
 @chapter See Also
 
 @ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
 @url{libswresample.html,libswresample}
 @end ifhtml
 
 @ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), libswresample(3)
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libswresample(3)
 @end ifnothtml
 
 @include authors.texi

doc/ffmpeg-scaler.texi

@@ -24,12 +24,12 @@ image rescaling and pixel format conversion.
 @chapter See Also
 
 @ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
 @url{libswscale.html,libswscale}
 @end ifhtml
 
 @ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), libswscale(3)
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libswscale(3)
 @end ifnothtml
 
 @include authors.texi

doc/ffmpeg-utils.texi

@@ -23,12 +23,12 @@ by the libavutil library.
 @chapter See Also
 
 @ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
 @url{libavutil.html,libavutil}
 @end ifhtml
 
 @ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), libavutil(3)
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavutil(3)
 @end ifnothtml
 
 @include authors.texi

doc/ffmpeg.texi

@@ -216,208 +216,16 @@ filters is obviously also impossible, since filters work on uncompressed data.
 @chapter Stream selection
 @c man begin STREAM SELECTION
 
-@command{ffmpeg} provides the @code{-map} option for manual control of stream selection in each
-output file. Users can skip @code{-map} and let ffmpeg perform automatic stream selection as
-described below. The @code{-vn / -an / -sn / -dn} options can be used to skip inclusion of
-video, audio, subtitle and data streams respectively, whether manually mapped or automatically
-selected, except for those streams which are outputs of complex filtergraphs.
+By default, @command{ffmpeg} includes only one stream of each type (video, audio, subtitle)
+present in the input files and adds them to each output file.  It picks the
+"best" of each based upon the following criteria: for video, it is the stream
+with the highest resolution, for audio, it is the stream with the most channels, for
+subtitles, it is the first subtitle stream. In the case where several streams of
+the same type rate equally, the stream with the lowest index is chosen.
 
-@section Description
-The sub-sections that follow describe the various rules that are involved in stream selection.
-The examples that follow next show how these rules are applied in practice.
-
-While every effort is made to accurately reflect the behavior of the program, FFmpeg is under
-continuous development and the code may have changed since the time of this writing.
-
-@subsection Automatic stream selection
-
-In the absence of any map options for a particular output file, ffmpeg inspects the output
-format to check which type of streams can be included in it, viz. video, audio and/or
-subtitles. For each acceptable stream type, ffmpeg will pick one stream, when available,
-from among all the inputs.
-
-It will select that stream based upon the following criteria:
-@itemize
-@item
-for video, it is the stream with the highest resolution,
-@item
-for audio, it is the stream with the most channels,
-@item
-for subtitles, it is the first subtitle stream found but there's a caveat.
-The output format's default subtitle encoder can be either text-based or image-based,
-and only a subtitle stream of the same type will be chosen.
-@end itemize
-
-In the case where several streams of the same type rate equally, the stream with the lowest
-index is chosen.
-
-Data or attachment streams are not automatically selected and can only be included
-using @code{-map}.
-@subsection Manual stream selection
-
-When @code{-map} is used, only user-mapped streams are included in that output file,
-with one possible exception for filtergraph outputs described below.
-
-@subsection Complex filtergraphs
-
-If there are any complex filtergraph output streams with unlabeled pads, they will be added
-to the first output file. This will lead to a fatal error if the stream type is not supported
-by the output format. In the absence of the map option, the inclusion of these streams leads
-to the automatic stream selection of their types being skipped. If map options are present,
-these filtergraph streams are included in addition to the mapped streams.
-
-Complex filtergraph output streams with labeled pads must be mapped once and exactly once.
-
-@subsection Stream handling
-
-Stream handling is independent of stream selection, with an exception for subtitles described
-below. Stream handling is set via the @code{-codec} option addressed to streams within a
-specific @emph{output} file. In particular, codec options are applied by ffmpeg after the
-stream selection process and thus do not influence the latter. If no @code{-codec} option is
-specified for a stream type, ffmpeg will select the default encoder registered by the output
-file muxer.
-
-An exception exists for subtitles. If a subtitle encoder is specified for an output file, the
-first subtitle stream found of any type, text or image, will be included. ffmpeg does not validate
-if the specified encoder can convert the selected stream or if the converted stream is acceptable
-within the output format. This applies generally as well: when the user sets an encoder manually,
-the stream selection process cannot check if the encoded stream can be muxed into the output file.
-If it cannot, ffmpeg will abort and @emph{all} output files will fail to be processed.
-
-@section Examples
-
-The following examples illustrate the behavior, quirks and limitations of ffmpeg's stream
-selection methods.
-
-They assume the following three input files.
-
-@verbatim
-
-input file 'A.avi'
-      stream 0: video 640x360
-      stream 1: audio 2 channels
-
-input file 'B.mp4'
-      stream 0: video 1920x1080
-      stream 1: audio 2 channels
-      stream 2: subtitles (text)
-      stream 3: audio 5.1 channels
-      stream 4: subtitles (text)
-
-input file 'C.mkv'
-      stream 0: video 1280x720
-      stream 1: audio 2 channels
-      stream 2: subtitles (image)
-@end verbatim
-
-@subsubheading Example: automatic stream selection
-@example
-ffmpeg -i A.avi -i B.mp4 out1.mkv out2.wav -map 1:a -c:a copy out3.mov
-@end example
-There are three output files specified, and for the first two, no @code{-map} options
-are set, so ffmpeg will select streams for these two files automatically.
-
-@file{out1.mkv} is a Matroska container file and accepts video, audio and subtitle streams,
-so ffmpeg will try to select one of each type.@*
-For video, it will select @code{stream 0} from @file{B.mp4}, which has the highest
-resolution among all the input video streams.@*
-For audio, it will select @code{stream 3} from @file{B.mp4}, since it has the greatest
-number of channels.@*
-For subtitles, it will select @code{stream 2} from @file{B.mp4}, which is the first subtitle
-stream from among @file{A.avi} and @file{B.mp4}.
-
-@file{out2.wav} accepts only audio streams, so only @code{stream 3} from @file{B.mp4} is
-selected.
-
-For @file{out3.mov}, since a @code{-map} option is set, no automatic stream selection will
-occur. The @code{-map 1:a} option will select all audio streams from the second input
-@file{B.mp4}. No other streams will be included in this output file.
-
-For the first two outputs, all included streams will be transcoded. The encoders chosen will
-be the default ones registered by each output format, which may not match the codec of the
-selected input streams.
-
-For the third output, codec option for audio streams has been set
-to @code{copy}, so no decoding-filtering-encoding operations will occur, or @emph{can} occur.
-Packets of selected streams shall be conveyed from the input file and muxed within the output
-file.
-
-@subsubheading Example: automatic subtitles selection
-@example
-ffmpeg -i C.mkv out1.mkv -c:s dvdsub -an out2.mkv
-@end example
-Although @file{out1.mkv} is a Matroska container file which accepts subtitle streams, only a
-video and audio stream shall be selected. The subtitle stream of @file{C.mkv} is image-based
-and the default subtitle encoder of the Matroska muxer is text-based, so a transcode operation
-for the subtitles is expected to fail and hence the stream isn't selected. However, in
-@file{out2.mkv}, a subtitle encoder is specified in the command and so, the subtitle stream is
-selected, in addition to the video stream. The presence of @code{-an} disables audio stream
-selection for @file{out2.mkv}.
-
-@subsubheading Example: unlabeled filtergraph outputs
-@example
-ffmpeg -i A.avi -i C.mkv -i B.mp4 -filter_complex "overlay" out1.mp4 out2.srt
-@end example
-A filtergraph is setup here using the @code{-filter_complex} option and consists of a single
-video filter. The @code{overlay} filter requires exactly two video inputs, but none are
-specified, so the first two available video streams are used, those of @file{A.avi} and
-@file{C.mkv}. The output pad of the filter has no label and so is sent to the first output file
-@file{out1.mp4}. Due to this, automatic selection of the video stream is skipped, which would
-have selected the stream in @file{B.mp4}. The audio stream with most channels viz. @code{stream 3}
-in @file{B.mp4}, is chosen automatically. No subtitle stream is chosen however, since the MP4
-format has no default subtitle encoder registered, and the user hasn't specified a subtitle encoder.
-
-The 2nd output file, @file{out2.srt}, only accepts text-based subtitle streams. So, even though
-the first subtitle stream available belongs to @file{C.mkv}, it is image-based and hence skipped.
-The selected stream, @code{stream 2} in @file{B.mp4}, is the first text-based subtitle stream.
-
-@subsubheading Example: labeled filtergraph outputs
-@example
-ffmpeg -i A.avi -i B.mp4 -i C.mkv -filter_complex "[1:v]hue=s=0[outv];overlay;aresample" \
-       -map '[outv]' -an        out1.mp4 \
-                                out2.mkv \
-       -map '[outv]' -map 1:a:0 out3.mkv
-@end example
-
-The above command will fail, as the output pad labelled @code{[outv]} has been mapped twice.
-None of the output files shall be processed.
-
-@example
-ffmpeg -i A.avi -i B.mp4 -i C.mkv -filter_complex "[1:v]hue=s=0[outv];overlay;aresample" \
-       -an        out1.mp4 \
-                  out2.mkv \
-       -map 1:a:0 out3.mkv
-@end example
-
-This command above will also fail as the hue filter output has a label, @code{[outv]},
-and hasn't been mapped anywhere.
-
-The command should be modified as follows,
-@example
-ffmpeg -i A.avi -i B.mp4 -i C.mkv -filter_complex "[1:v]hue=s=0,split=2[outv1][outv2];overlay;aresample" \
-        -map '[outv1]' -an        out1.mp4 \
-                                  out2.mkv \
-        -map '[outv2]' -map 1:a:0 out3.mkv
-@end example
-The video stream from @file{B.mp4} is sent to the hue filter, whose output is cloned once using
-the split filter, and both outputs labelled. Then a copy each is mapped to the first and third
-output files.
-
-The overlay filter, requiring two video inputs, uses the first two unused video streams. Those
-are the streams from @file{A.avi} and @file{C.mkv}. The overlay output isn't labelled, so it is
-sent to the first output file @file{out1.mp4}, regardless of the presence of the @code{-map} option.
-
-The aresample filter is sent the first unused audio stream, that of @file{A.avi}. Since this filter
-output is also unlabelled, it too is mapped to the first output file. The presence of @code{-an}
-only suppresses automatic or manual stream selection of audio streams, not outputs sent from
-filtergraphs. Both these mapped streams shall be ordered before the mapped stream in @file{out1.mp4}.
-
-The video, audio and subtitle streams mapped to @code{out2.mkv} are entirely determined by
-automatic stream selection.
-
-@file{out3.mkv} consists of the cloned video output from the hue filter and the first audio
-stream from @file{B.mp4}.
-@*
+You can disable some of those defaults by using the @code{-vn/-an/-sn/-dn} options. For
+full manual control, use the @code{-map} option, which disables the defaults just
+described.
 
 @c man end STREAM SELECTION
 
@@ -481,8 +289,8 @@ see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1)
 
 -to and -t are mutually exclusive and -t has priority.
 
-@item -to @var{position} (@emph{input/output})
-Stop writing the output or reading the input at @var{position}.
+@item -to @var{position} (@emph{output})
+Stop writing the output at @var{position}.
 @var{position} must be a time duration specification,
 see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
 
@@ -508,7 +316,7 @@ input until the timestamps reach @var{position}.
 @var{position} must be a time duration specification,
 see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
 
-@item -sseof @var{position} (@emph{input})
+@item -sseof @var{position} (@emph{input/output})
 
 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.
@@ -523,9 +331,6 @@ The offset is added to the timestamps of the input files. Specifying
 a positive offset means that the corresponding streams are delayed by
 the time duration specified in @var{offset}.
 
-@item -itsscale @var{scale} (@emph{input,per-stream})
-Rescale input timestamps. @var{scale} should be a floating point number.
-
 @item -timestamp @var{date} (@emph{output})
 Set the recording timestamp in the container.
 
@@ -570,31 +375,22 @@ The following dispositions are recognized:
 @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
-ffmpeg -i in.mkv -c copy -disposition:a:1 default out.mkv
+ffmpeg -i in.mkv -disposition:a:1 default out.mkv
 @end example
 
 To make the second subtitle stream the default stream and remove the default
 disposition from the first subtitle stream:
 @example
-ffmpeg -i in.mkv -c copy -disposition:s:0 0 -disposition:s:1 default out.mkv
+ffmpeg -i INPUT -disposition:s:0 0 -disposition:s:1 default OUTPUT
 @end example
 
-To add an embedded cover/thumbnail:
-@example
-ffmpeg -i in.mp4 -i IMAGE -map 0 -map 1 -c copy -c:v:1 png -disposition:v:1 attached_pic out.mp4
-@end example
-
-Not all muxers support embedded thumbnails, and those who do, only support a few formats, like JPEG or PNG.
-
 @item -program [title=@var{title}:][program_num=@var{program_num}:]st=@var{stream}[:st=@var{stream}...] (@emph{output})
 
 Creates a program with the specified @var{title}, @var{program_num} and adds the specified
@@ -617,15 +413,6 @@ they do not conflict with the standard, as in:
 ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg
 @end example
 
-@item -dn (@emph{input/output})
-As an input option, blocks all data streams of a file from being filtered or
-being automatically selected or mapped for any output. See @code{-discard}
-option to disable streams individually.
-
-As an output option, disables data recording i.e. automatic selection or
-mapping of any data stream. For full manual control see the @code{-map}
-option.
-
 @item -dframes @var{number} (@emph{output})
 Set the number of data frames to output. This is an obsolete alias for
 @code{-frames:d}, which you should use instead.
@@ -683,7 +470,6 @@ the encoding process. It is made of "@var{key}=@var{value}" lines. @var{key}
 consists of only alphanumeric characters. The last key of a sequence of
 progress information is always "progress".
 
-@anchor{stdin option}
 @item -stdin
 Enable interaction on standard input. On by default unless standard input is
 used as an input. To explicitly disable interaction you need to specify
@@ -783,14 +569,8 @@ If used together with @option{-vcodec copy}, it will affect the aspect ratio
 stored at container level, but not the aspect ratio stored in encoded
 frames, if it exists.
 
-@item -vn (@emph{input/output})
-As an input option, blocks all video streams of a file from being filtered or
-being automatically selected or mapped for any output. See @code{-discard}
-option to disable streams individually.
-
-As an output option, disables video recording i.e. automatic selection or
-mapping of any video stream. For full manual control see the @code{-map}
-option.
+@item -vn (@emph{output})
+Disable video recording.
 
 @item -vcodec @var{codec} (@emph{output})
 Set the video codec. This is an alias for @code{-codec:v}.
@@ -837,6 +617,8 @@ as the input (or graph output) and automatic conversions are disabled.
 
 @item -sws_flags @var{flags} (@emph{input/output})
 Set SwScaler flags.
+@item -vdt @var{n}
+Discard threshold.
 
 @item -rc_override[:@var{stream_specifier}] @var{override} (@emph{output,per-stream})
 Rate control override for specific intervals, formatted as "int,int,int"
@@ -974,43 +756,6 @@ If not specified, @samp{auto_any} is used.
 platform-appropriate subdevice (@samp{dxva2} or @samp{vaapi}) and then deriving a
 QSV device from that.)
 
-@item opencl
-@var{device} selects the platform and device as @emph{platform_index.device_index}.
-
-The set of devices can also be filtered using the key-value pairs to find only
-devices matching particular platform or device strings.
-
-The strings usable as filters are:
-@table @option
-@item platform_profile
-@item platform_version
-@item platform_name
-@item platform_vendor
-@item platform_extensions
-@item device_name
-@item device_vendor
-@item driver_version
-@item device_version
-@item device_profile
-@item device_extensions
-@item device_type
-@end table
-
-The indices and filters must together uniquely select a device.
-
-Examples:
-@table @emph
-@item -init_hw_device opencl:0.1
-Choose the second device on the first platform.
-
-@item -init_hw_device opencl:,device_name=Foo9000
-Choose the device with a name containing the string @emph{Foo9000}.
-
-@item -init_hw_device opencl:1,device_type=gpu,device_extensions=cl_khr_fp16
-Choose the GPU device on the second platform supporting the @emph{cl_khr_fp16}
-extension.
-@end table
-
 @end table
 
 @item -init_hw_device @var{type}[=@var{name}]@@@var{source}
@@ -1041,6 +786,9 @@ Do not use any hardware acceleration (the default).
 @item auto
 Automatically select the hardware acceleration method.
 
+@item vda
+Use Apple VDA hardware acceleration.
+
 @item vdpau
 Use VDPAU (Video Decode and Presentation API for Unix) hardware acceleration.
 
@@ -1102,14 +850,8 @@ Set the number of audio channels. For output streams it is set by
 default to the number of input audio channels. For input streams
 this option only makes sense for audio grabbing devices and raw demuxers
 and is mapped to the corresponding demuxer options.
-@item -an (@emph{input/output})
-As an input option, blocks all audio streams of a file from being filtered or
-being automatically selected or mapped for any output. See @code{-discard}
-option to disable streams individually.
-
-As an output option, disables audio recording i.e. automatic selection or
-mapping of any audio stream. For full manual control see the @code{-map}
-option.
+@item -an (@emph{output})
+Disable audio recording.
 @item -acodec @var{codec} (@emph{input/output})
 Set the audio codec. This is an alias for @code{-codec:a}.
 @item -sample_fmt[:@var{stream_specifier}] @var{sample_fmt} (@emph{output,per-stream})
@@ -1143,14 +885,8 @@ stereo but not 6 channels as 5.1. The default is to always try to guess. Use
 @table @option
 @item -scodec @var{codec} (@emph{input/output})
 Set the subtitle codec. This is an alias for @code{-codec:s}.
-@item -sn (@emph{input/output})
-As an input option, blocks all subtitle streams of a file from being filtered or
-being automatically selected or mapped for any output. See @code{-discard}
-option to disable streams individually.
-
-As an output option, disables subtitle recording i.e. automatic selection or
-mapping of any subtitle stream. For full manual control see the @code{-map}
-option.
+@item -sn (@emph{output})
+Disable subtitle recording.
 @item -sbsf @var{bitstream_filter}
 Deprecated, see -bsf
 @end table
@@ -1376,12 +1112,12 @@ disable any chapter copying.
 
 @item -benchmark (@emph{global})
 Show benchmarking information at the end of an encode.
-Shows real, system and user time used and maximum memory consumption.
+Shows CPU time used and maximum memory consumption.
 Maximum memory consumption is not supported on all systems,
 it will usually display as 0 if not supported.
 @item -benchmark_all (@emph{global})
 Show benchmarking information during the encode.
-Shows real, system and user time used in various steps (audio/video encode/decode).
+Shows CPU time used in various steps (audio/video encode/decode).
 @item -timelimit @var{duration} (@emph{global})
 Exit after ffmpeg has been running for @var{duration} seconds.
 @item -dump (@emph{global})
@@ -1396,6 +1132,14 @@ loss).
 By default @command{ffmpeg} attempts to read the input(s) as fast as possible.
 This option will slow down the reading of the input(s) to the native frame rate
 of the input(s). It is useful for real-time output (e.g. live streaming).
+@item -loop_input
+Loop over the input stream. Currently it works only for image
+streams. This option is used for automatic FFserver testing.
+This option is deprecated, use -loop 1.
+@item -loop_output @var{number_of_times}
+Repeatedly loop output for formats that support looping such as animated GIF
+(0 will loop the output infinitely).
+This option is deprecated, use -loop.
 @item -vsync @var{parameter}
 Video sync method.
 For compatibility reasons old values can be specified as numbers.
@@ -1509,15 +1253,13 @@ or as a floating point number (e.g. 0.04166, 2.0833e-5)
 
 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 input stream ends.
 @item -dts_delta_threshold
 Timestamp discontinuity delta threshold.
-@item -muxdelay @var{seconds} (@emph{output})
+@item -muxdelay @var{seconds} (@emph{input})
 Set the maximum demux-decode delay.
-@item -muxpreload @var{seconds} (@emph{output})
+@item -muxpreload @var{seconds} (@emph{input})
 Set the initial demux-decode delay.
 @item -streamid @var{output-stream-index}:@var{new-value} (@emph{output})
 Assign a new stream-id value to an output stream. This option should be
@@ -1633,16 +1375,24 @@ file or device. With low latency / high rate live streams, packets may be
 discarded if they are not read in a timely manner; raising this value can
 avoid it.
 
+@item -override_ffserver (@emph{global})
+Overrides the input specifications from @command{ffserver}. Using this
+option you can map any input stream to @command{ffserver} and control
+many aspects of the encoding from @command{ffmpeg}. Without this
+option @command{ffmpeg} will transmit to @command{ffserver} what is
+requested by @command{ffserver}.
+
+The option is intended for cases where features are needed that cannot be
+specified to @command{ffserver} but can be to @command{ffmpeg}.
+
 @item -sdp_file @var{file} (@emph{global})
 Print sdp information for an output stream to @var{file}.
 This allows dumping sdp information when at least one output isn't an
 rtp stream. (Requires at least one of the output formats to be rtp).
 
 @item -discard (@emph{input})
-Allows discarding specific streams or frames from streams.
-Any input stream can be fully discarded, using value @code{all} whereas
-selective discarding of frames from a stream occurs at the demuxer
-and is not supported by all demuxers.
+Allows discarding specific streams or frames of streams at the demuxer.
+Not all demuxers support this.
 
 @table @option
 @item none
@@ -1989,7 +1739,7 @@ ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
 @ifset config-not-all
 @url{ffmpeg-all.html,ffmpeg-all},
 @end ifset
-@url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
+@url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
 @url{ffmpeg-utils.html,ffmpeg-utils},
 @url{ffmpeg-scaler.html,ffmpeg-scaler},
 @url{ffmpeg-resampler.html,ffmpeg-resampler},
@@ -2008,7 +1758,7 @@ ffmpeg(1),
 @ifset config-not-all
 ffmpeg-all(1),
 @end ifset
-ffplay(1), ffprobe(1),
+ffplay(1), ffprobe(1), ffserver(1),
 ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
 ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
 ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)

doc/ffplay.texi

@@ -60,14 +60,10 @@ Play @var{duration} seconds of audio/video.
 see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
 @item -bytes
 Seek by bytes.
-@item -seek_interval
-Set custom interval, in seconds, for seeking using left/right keys. Default is 10 seconds.
 @item -nodisp
 Disable graphical display.
 @item -noborder
 Borderless window.
-@item -alwaysontop
-Window always on top. Available on: X11 with SDL >= 2.0.5, Windows SDL >= 2.0.6.
 @item -volume
 Set the startup volume. 0 means silence, 100 means no volume reduction or
 amplification. Negative values are treated as 0, values above 100 are treated
@@ -76,10 +72,6 @@ as 100.
 Force format.
 @item -window_title @var{title}
 Set window title (default is the input filename).
-@item -left @var{title}
-Set the x position for the left of the window (default is a centered window).
-@item -top @var{title}
-Set the y position for the top of the window (default is a centered window).
 @item -loop @var{number}
 Loops movie playback <number> times. 0 means forever.
 @item -showmode @var{mode}
@@ -197,12 +189,6 @@ input as soon as possible. Enabled by default for realtime streams, where data
 may be dropped if not read in time. Use this option to enable infinite buffers
 for all inputs, use @option{-noinfbuf} to disable it.
 
-@item -filter_threads @var{nb_threads}
-Defines how many threads are used to process a filter pipeline. Each pipeline
-will produce a thread pool with this many threads available for parallel
-processing. The default is 0 which means that the thread count will be
-determined by the number of available CPUs.
-
 @end table
 
 @section While playing
@@ -305,7 +291,7 @@ Toggle full screen.
 @ifset config-not-all
 @url{ffplay-all.html,ffmpeg-all},
 @end ifset
-@url{ffmpeg.html,ffmpeg}, @url{ffprobe.html,ffprobe},
+@url{ffmpeg.html,ffmpeg}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
 @url{ffmpeg-utils.html,ffmpeg-utils},
 @url{ffmpeg-scaler.html,ffmpeg-scaler},
 @url{ffmpeg-resampler.html,ffmpeg-resampler},
@@ -324,7 +310,7 @@ ffplay(1),
 @ifset config-not-all
 ffplay-all(1),
 @end ifset
-ffmpeg(1), ffprobe(1),
+ffmpeg(1), ffprobe(1), ffserver(1),
 ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
 ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
 ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)

doc/ffprobe.texi

@@ -425,7 +425,7 @@ The @code{csv} writer is equivalent to @code{compact}, but supports
 different defaults.
 
 Each section is printed on a single line.
-If no option is specified, the output has the form:
+If no option is specifid, the output has the form:
 @example
 section|key1=val1| ... |keyN=valN
 @end example
@@ -584,14 +584,14 @@ value is 0.
 This is required for generating an XML file which can be validated
 through an XSD file.
 
-@item xsd_strict, x
+@item xsd_compliant, x
 If set to 1 perform more checks for ensuring that the output is XSD
 compliant. Default value is 0.
 This option automatically sets @option{fully_qualified} to 1.
 @end table
 
 For more information about the XML format, see
-@url{https://www.w3.org/XML/}.
+@url{http://www.w3.org/XML/}.
 @c man end WRITERS
 
 @chapter Timecode
@@ -653,7 +653,7 @@ DV, GXF and AVI timecodes are available in format metadata
 @ifset config-not-all
 @url{ffprobe-all.html,ffprobe-all},
 @end ifset
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay},
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffserver.html,ffserver},
 @url{ffmpeg-utils.html,ffmpeg-utils},
 @url{ffmpeg-scaler.html,ffmpeg-scaler},
 @url{ffmpeg-resampler.html,ffmpeg-resampler},
@@ -672,7 +672,7 @@ ffprobe(1),
 @ifset config-not-all
 ffprobe-all(1),
 @end ifset
-ffmpeg(1), ffplay(1),
+ffmpeg(1), ffplay(1), ffserver(1),
 ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
 ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
 ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)

doc/ffprobe.xsd

@@ -147,25 +147,11 @@
         </xsd:sequence>
     </xsd:complexType>
     <xsd:complexType name="frameSideDataType">
-        <xsd:sequence>
-            <xsd:element name="timecodes" type="ffprobe:frameSideDataTimecodeList" minOccurs="0" maxOccurs="1"/>
-        </xsd:sequence>
-
         <xsd:attribute name="side_data_type"              type="xsd:string"/>
         <xsd:attribute name="side_data_size"              type="xsd:int"   />
         <xsd:attribute name="timecode"                    type="xsd:string"/>
     </xsd:complexType>
 
-    <xsd:complexType name="frameSideDataTimecodeList">
-        <xsd:sequence>
-            <xsd:element name="timecode" type="ffprobe:frameSideDataTimecodeType" minOccurs="0" maxOccurs="unbounded"/>
-        </xsd:sequence>
-    </xsd:complexType>
-
-    <xsd:complexType name="frameSideDataTimecodeType">
-        <xsd:attribute name="value"              type="xsd:string"/>
-    </xsd:complexType>
-
     <xsd:complexType name="subtitleType">
       <xsd:attribute name="media_type"         type="xsd:string" fixed="subtitle" use="required"/>
       <xsd:attribute name="pts"                type="xsd:long" />

doc/ffserver.conf

@@ -0,0 +1,372 @@
+# Port on which the server is listening. You must select a different
+# port from your standard HTTP web server if it is running on the same
+# computer.
+HTTPPort 8090
+
+# Address on which the server is bound. Only useful if you have
+# several network interfaces.
+HTTPBindAddress 0.0.0.0
+
+# Number of simultaneous HTTP connections that can be handled. It has
+# to be defined *before* the MaxClients parameter, since it defines the
+# MaxClients maximum limit.
+MaxHTTPConnections 2000
+
+# Number of simultaneous requests that can be handled. Since FFServer
+# is very fast, it is more likely that you will want to leave this high
+# and use MaxBandwidth, below.
+MaxClients 1000
+
+# This the maximum amount of kbit/sec that you are prepared to
+# consume when streaming to clients.
+MaxBandwidth 1000
+
+# Access log file (uses standard Apache log file format)
+# '-' is the standard output.
+CustomLog -
+
+##################################################################
+# Definition of the live feeds. Each live feed contains one video
+# and/or audio sequence coming from an ffmpeg encoder or another
+# ffserver. This sequence may be encoded simultaneously with several
+# codecs at several resolutions.
+
+<Feed feed1.ffm>
+
+# You must use 'ffmpeg' to send a live feed to ffserver. In this
+# example, you can type:
+#
+# ffmpeg http://localhost:8090/feed1.ffm
+
+# ffserver can also do time shifting. It means that it can stream any
+# previously recorded live stream. The request should contain:
+# "http://xxxx?date=[YYYY-MM-DDT][[HH:]MM:]SS[.m...]".You must specify
+# a path where the feed is stored on disk. You also specify the
+# maximum size of the feed, where zero means unlimited. Default:
+# File=/tmp/feed_name.ffm FileMaxSize=5M
+File /tmp/feed1.ffm
+FileMaxSize 200K
+
+# You could specify
+# ReadOnlyFile /saved/specialvideo.ffm
+# This marks the file as readonly and it will not be deleted or updated.
+
+# Specify launch in order to start ffmpeg automatically.
+# First ffmpeg must be defined with an appropriate path if needed,
+# after that options can follow, but avoid adding the http:// field
+#Launch ffmpeg
+
+# Only allow connections from localhost to the feed.
+ACL allow 127.0.0.1
+
+</Feed>
+
+
+##################################################################
+# Now you can define each stream which will be generated from the
+# original audio and video stream. Each format has a filename (here
+# 'test1.mpg'). FFServer will send this stream when answering a
+# request containing this filename.
+
+<Stream test1.mpg>
+
+# coming from live feed 'feed1'
+Feed feed1.ffm
+
+# Format of the stream : you can choose among:
+# mpeg       : MPEG-1 multiplexed video and audio
+# mpegvideo  : only MPEG-1 video
+# mp2        : MPEG-2 audio (use AudioCodec to select layer 2 and 3 codec)
+# ogg        : Ogg format (Vorbis audio codec)
+# rm         : RealNetworks-compatible stream. Multiplexed audio and video.
+# ra         : RealNetworks-compatible stream. Audio only.
+# mpjpeg     : Multipart JPEG (works with Netscape without any plugin)
+# jpeg       : Generate a single JPEG image.
+# mjpeg      : Generate a M-JPEG stream.
+# asf        : ASF compatible streaming (Windows Media Player format).
+# swf        : Macromedia Flash compatible stream
+# avi        : AVI format (MPEG-4 video, MPEG audio sound)
+Format mpeg
+
+# Bitrate for the audio stream. Codecs usually support only a few
+# different bitrates.
+AudioBitRate 32
+
+# Number of audio channels: 1 = mono, 2 = stereo
+AudioChannels 1
+
+# Sampling frequency for audio. When using low bitrates, you should
+# lower this frequency to 22050 or 11025. The supported frequencies
+# depend on the selected audio codec.
+AudioSampleRate 44100
+
+# Bitrate for the video stream
+VideoBitRate 64
+
+# Ratecontrol buffer size
+VideoBufferSize 40
+
+# Number of frames per second
+VideoFrameRate 3
+
+# Size of the video frame: WxH (default: 160x128)
+# The following abbreviations are defined: sqcif, qcif, cif, 4cif, qqvga,
+# qvga, vga, svga, xga, uxga, qxga, sxga, qsxga, hsxga, wvga, wxga, wsxga,
+# wuxga, woxga, wqsxga, wquxga, whsxga, whuxga, cga, ega, hd480, hd720,
+# hd1080
+VideoSize 160x128
+
+# Transmit only intra frames (useful for low bitrates, but kills frame rate).
+#VideoIntraOnly
+
+# If non-intra only, an intra frame is transmitted every VideoGopSize
+# frames. Video synchronization can only begin at an intra frame.
+VideoGopSize 12
+
+# More MPEG-4 parameters
+# VideoHighQuality
+# Video4MotionVector
+
+# Choose your codecs:
+#AudioCodec mp2
+#VideoCodec mpeg1video
+
+# Suppress audio
+#NoAudio
+
+# Suppress video
+#NoVideo
+
+#VideoQMin 3
+#VideoQMax 31
+
+# Set this to the number of seconds backwards in time to start. Note that
+# most players will buffer 5-10 seconds of video, and also you need to allow
+# for a keyframe to appear in the data stream.
+#Preroll 15
+
+# ACL:
+
+# You can allow ranges of addresses (or single addresses)
+#ACL ALLOW <first address> <last address>
+
+# You can deny ranges of addresses (or single addresses)
+#ACL DENY <first address> <last address>
+
+# You can repeat the ACL allow/deny as often as you like. It is on a per
+# stream basis. The first match defines the action. If there are no matches,
+# then the default is the inverse of the last ACL statement.
+#
+# Thus 'ACL allow localhost' only allows access from localhost.
+# 'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and
+# allow everybody else.
+
+</Stream>
+
+
+##################################################################
+# Example streams
+
+
+# Multipart JPEG
+
+#<Stream test.mjpg>
+#Feed feed1.ffm
+#Format mpjpeg
+#VideoFrameRate 2
+#VideoIntraOnly
+#NoAudio
+#Strict -1
+#</Stream>
+
+
+# Single JPEG
+
+#<Stream test.jpg>
+#Feed feed1.ffm
+#Format jpeg
+#VideoFrameRate 2
+#VideoIntraOnly
+##VideoSize 352x240
+#NoAudio
+#Strict -1
+#</Stream>
+
+
+# Flash
+
+#<Stream test.swf>
+#Feed feed1.ffm
+#Format swf
+#VideoFrameRate 2
+#VideoIntraOnly
+#NoAudio
+#</Stream>
+
+
+# ASF compatible
+
+<Stream test.asf>
+Feed feed1.ffm
+Format asf
+VideoFrameRate 15
+VideoSize 352x240
+VideoBitRate 256
+VideoBufferSize 40
+VideoGopSize 30
+AudioBitRate 64
+StartSendOnKey
+</Stream>
+
+
+# MP3 audio
+
+#<Stream test.mp3>
+#Feed feed1.ffm
+#Format mp2
+#AudioCodec mp3
+#AudioBitRate 64
+#AudioChannels 1
+#AudioSampleRate 44100
+#NoVideo
+#</Stream>
+
+
+# Ogg Vorbis audio
+
+#<Stream test.ogg>
+#Feed feed1.ffm
+#Metadata title "Stream title"
+#AudioBitRate 64
+#AudioChannels 2
+#AudioSampleRate 44100
+#NoVideo
+#</Stream>
+
+
+# Real with audio only at 32 kbits
+
+#<Stream test.ra>
+#Feed feed1.ffm
+#Format rm
+#AudioBitRate 32
+#NoVideo
+#NoAudio
+#</Stream>
+
+
+# Real with audio and video at 64 kbits
+
+#<Stream test.rm>
+#Feed feed1.ffm
+#Format rm
+#AudioBitRate 32
+#VideoBitRate 128
+#VideoFrameRate 25
+#VideoGopSize 25
+#NoAudio
+#</Stream>
+
+
+##################################################################
+# A stream coming from a file: you only need to set the input
+# filename and optionally a new format. Supported conversions:
+#    AVI -> ASF
+
+#<Stream file.rm>
+#File "/usr/local/httpd/htdocs/tlive.rm"
+#NoAudio
+#</Stream>
+
+#<Stream file.asf>
+#File "/usr/local/httpd/htdocs/test.asf"
+#NoAudio
+#Metadata author "Me"
+#Metadata copyright "Super MegaCorp"
+#Metadata title "Test stream from disk"
+#Metadata comment "Test comment"
+#</Stream>
+
+
+##################################################################
+# RTSP examples
+#
+# You can access this stream with the RTSP URL:
+#   rtsp://localhost:5454/test1-rtsp.mpg
+#
+# A non-standard RTSP redirector is also created. Its URL is:
+#   http://localhost:8090/test1-rtsp.rtsp
+
+#<Stream test1-rtsp.mpg>
+#Format rtp
+#File "/usr/local/httpd/htdocs/test1.mpg"
+#</Stream>
+
+
+# Transcode an incoming live feed to another live feed,
+# using libx264 and video presets
+
+#<Stream live.h264>
+#Format rtp
+#Feed feed1.ffm
+#VideoCodec libx264
+#VideoFrameRate 24
+#VideoBitRate 100
+#VideoSize 480x272
+#AVPresetVideo default
+#AVPresetVideo baseline
+#AVOptionVideo flags +global_header
+#
+#AudioCodec aac
+#AudioBitRate 32
+#AudioChannels 2
+#AudioSampleRate 22050
+#AVOptionAudio flags +global_header
+#</Stream>
+
+##################################################################
+# SDP/multicast examples
+#
+# If you want to send your stream in multicast, you must set the
+# multicast address with MulticastAddress. The port and the TTL can
+# also be set.
+#
+# An SDP file is automatically generated by ffserver by adding the
+# 'sdp' extension to the stream name (here
+# http://localhost:8090/test1-sdp.sdp). You should usually give this
+# file to your player to play the stream.
+#
+# The 'NoLoop' option can be used to avoid looping when the stream is
+# terminated.
+
+#<Stream test1-sdp.mpg>
+#Format rtp
+#File "/usr/local/httpd/htdocs/test1.mpg"
+#MulticastAddress 224.124.0.1
+#MulticastPort 5000
+#MulticastTTL 16
+#NoLoop
+#</Stream>
+
+
+##################################################################
+# Special streams
+
+# Server status
+
+<Stream stat.html>
+Format status
+
+# Only allow local people to get the status
+ACL allow localhost
+ACL allow 192.168.0.0 192.168.255.255
+
+#FaviconURL http://pond1.gladstonefamily.net:8080/favicon.ico
+</Stream>
+
+
+# Redirect index.html to the appropriate site
+
+<Redirect index.html>
+URL http://www.ffmpeg.org/
+</Redirect>

doc/ffserver.texi

@@ -0,0 +1,923 @@
+\input texinfo @c -*- texinfo -*-
+@documentencoding UTF-8
+
+@settitle ffserver Documentation
+@titlepage
+@center @titlefont{ffserver Documentation}
+@end titlepage
+
+@top
+
+@contents
+
+@chapter Synopsis
+
+ffserver [@var{options}]
+
+@chapter Description
+@c man begin DESCRIPTION
+
+@command{ffserver} is a streaming server for both audio and video.
+It supports several live feeds, streaming from files and time shifting
+on live feeds. You can seek to positions in the past on each live
+feed, provided you specify a big enough feed storage.
+
+@command{ffserver} is configured through a configuration file, which
+is read at startup. If not explicitly specified, it will read from
+@file{/etc/ffserver.conf}.
+
+@command{ffserver} receives prerecorded files or FFM streams from some
+@command{ffmpeg} instance as input, then streams them over
+RTP/RTSP/HTTP.
+
+An @command{ffserver} instance will listen on some port as specified
+in the configuration file. You can launch one or more instances of
+@command{ffmpeg} and send one or more FFM streams to the port where
+ffserver is expecting to receive them. Alternately, you can make
+@command{ffserver} launch such @command{ffmpeg} instances at startup.
+
+Input streams are called feeds, and each one is specified by a
+@code{<Feed>} section in the configuration file.
+
+For each feed you can have different output streams in various
+formats, each one specified by a @code{<Stream>} section in the
+configuration file.
+
+@chapter Detailed description
+
+@command{ffserver} works by forwarding streams encoded by
+@command{ffmpeg}, or pre-recorded streams which are read from disk.
+
+Precisely, @command{ffserver} acts as an HTTP server, accepting POST
+requests from @command{ffmpeg} to acquire the stream to publish, and
+serving RTSP clients or HTTP clients GET requests with the stream
+media content.
+
+A feed is an @ref{FFM} stream created by @command{ffmpeg}, and sent to
+a port where @command{ffserver} is listening.
+
+Each feed is identified by a unique name, corresponding to the name
+of the resource published on @command{ffserver}, and is configured by
+a dedicated @code{Feed} section in the configuration file.
+
+The feed publish URL is given by:
+@example
+http://@var{ffserver_ip_address}:@var{http_port}/@var{feed_name}
+@end example
+
+where @var{ffserver_ip_address} is the IP address of the machine where
+@command{ffserver} is installed, @var{http_port} is the port number of
+the HTTP server (configured through the @option{HTTPPort} option), and
+@var{feed_name} is the name of the corresponding feed defined in the
+configuration file.
+
+Each feed is associated to a file which is stored on disk. This stored
+file is used to send pre-recorded data to a player as fast as
+possible when new content is added in real-time to the stream.
+
+A "live-stream" or "stream" is a resource published by
+@command{ffserver}, and made accessible through the HTTP protocol to
+clients.
+
+A stream can be connected to a feed, or to a file. In the first case,
+the published stream is forwarded from the corresponding feed
+generated by a running instance of @command{ffmpeg}, in the second
+case the stream is read from a pre-recorded file.
+
+Each stream is identified by a unique name, corresponding to the name
+of the resource served by @command{ffserver}, and is configured by
+a dedicated @code{Stream} section in the configuration file.
+
+The stream access HTTP URL is given by:
+@example
+http://@var{ffserver_ip_address}:@var{http_port}/@var{stream_name}[@var{options}]
+@end example
+
+The stream access RTSP URL is given by:
+@example
+http://@var{ffserver_ip_address}:@var{rtsp_port}/@var{stream_name}[@var{options}]
+@end example
+
+@var{stream_name} is the name of the corresponding stream defined in
+the configuration file. @var{options} is a list of options specified
+after the URL which affects how the stream is served by
+@command{ffserver}. @var{http_port} and @var{rtsp_port} are the HTTP
+and RTSP ports configured with the options @var{HTTPPort} and
+@var{RTSPPort} respectively.
+
+In case the stream is associated to a feed, the encoding parameters
+must be configured in the stream configuration. They are sent to
+@command{ffmpeg} when setting up the encoding. This allows
+@command{ffserver} to define the encoding parameters used by
+the @command{ffmpeg} encoders.
+
+The @command{ffmpeg} @option{override_ffserver} commandline option
+allows one to override the encoding parameters set by the server.
+
+Multiple streams can be connected to the same feed.
+
+For example, you can have a situation described by the following
+graph:
+
+@verbatim
+               _________       __________
+              |         |     |          |
+ffmpeg 1 -----| feed 1  |-----| stream 1 |
+    \         |_________|\    |__________|
+     \                    \
+      \                    \   __________
+       \                    \ |          |
+        \                    \| stream 2 |
+         \                    |__________|
+          \
+           \   _________       __________
+            \ |         |     |          |
+             \| feed 2  |-----| stream 3 |
+              |_________|     |__________|
+
+               _________       __________
+              |         |     |          |
+ffmpeg 2 -----| feed 3  |-----| stream 4 |
+              |_________|     |__________|
+
+               _________       __________
+              |         |     |          |
+              | file 1  |-----| stream 5 |
+              |_________|     |__________|
+
+@end verbatim
+
+@anchor{FFM}
+@section FFM, FFM2 formats
+
+FFM and FFM2 are formats used by ffserver. They allow storing a wide variety of
+video and audio streams and encoding options, and can store a moving time segment
+of an infinite movie or a whole movie.
+
+FFM is version specific, and there is limited compatibility of FFM files
+generated by one version of ffmpeg/ffserver and another version of
+ffmpeg/ffserver. It may work but it is not guaranteed to work.
+
+FFM2 is extensible while maintaining compatibility and should work between
+differing versions of tools. FFM2 is the default.
+
+@section Status stream
+
+@command{ffserver} supports an HTTP interface which exposes the
+current status of the server.
+
+Simply point your browser to the address of the special status stream
+specified in the configuration file.
+
+For example if you have:
+@example
+<Stream status.html>
+Format status
+
+# Only allow local people to get the status
+ACL allow localhost
+ACL allow 192.168.0.0 192.168.255.255
+</Stream>
+@end example
+
+then the server will post a page with the status information when
+the special stream @file{status.html} is requested.
+
+@section How do I make it work?
+
+As a simple test, just run the following two command lines where INPUTFILE
+is some file which you can decode with ffmpeg:
+
+@example
+ffserver -f doc/ffserver.conf &
+ffmpeg -i INPUTFILE http://localhost:8090/feed1.ffm
+@end example
+
+At this point you should be able to go to your Windows machine and fire up
+Windows Media Player (WMP). Go to Open URL and enter
+
+@example
+    http://<linuxbox>:8090/test.asf
+@end example
+
+You should (after a short delay) see video and hear audio.
+
+WARNING: trying to stream test1.mpg doesn't work with WMP as it tries to
+transfer the entire file before starting to play.
+The same is true of AVI files.
+
+You should edit the @file{ffserver.conf} file to suit your needs (in
+terms of frame rates etc). Then install @command{ffserver} and
+@command{ffmpeg}, write a script to start them up, and off you go.
+
+@section What else can it do?
+
+You can replay video from .ffm files that was recorded earlier.
+However, there are a number of caveats, including the fact that the
+ffserver parameters must match the original parameters used to record the
+file. If they do not, then ffserver deletes the file before recording into it.
+(Now that I write this, it seems broken).
+
+You can fiddle with many of the codec choices and encoding parameters, and
+there are a bunch more parameters that you cannot control. Post a message
+to the mailing list if there are some 'must have' parameters. Look in
+ffserver.conf for a list of the currently available controls.
+
+It will automatically generate the ASX or RAM files that are often used
+in browsers. These files are actually redirections to the underlying ASF
+or RM file. The reason for this is that the browser often fetches the
+entire file before starting up the external viewer. The redirection files
+are very small and can be transferred quickly. [The stream itself is
+often 'infinite' and thus the browser tries to download it and never
+finishes.]
+
+@section Tips
+
+* When you connect to a live stream, most players (WMP, RA, etc) want to
+buffer a certain number of seconds of material so that they can display the
+signal continuously. However, ffserver (by default) starts sending data
+in realtime. This means that there is a pause of a few seconds while the
+buffering is being done by the player. The good news is that this can be
+cured by adding a '?buffer=5' to the end of the URL. This means that the
+stream should start 5 seconds in the past -- and so the first 5 seconds
+of the stream are sent as fast as the network will allow. It will then
+slow down to real time. This noticeably improves the startup experience.
+
+You can also add a 'Preroll 15' statement into the ffserver.conf that will
+add the 15 second prebuffering on all requests that do not otherwise
+specify a time. In addition, ffserver will skip frames until a key_frame
+is found. This further reduces the startup delay by not transferring data
+that will be discarded.
+
+@section Why does the ?buffer / Preroll stop working after a time?
+
+It turns out that (on my machine at least) the number of frames successfully
+grabbed is marginally less than the number that ought to be grabbed. This
+means that the timestamp in the encoded data stream gets behind realtime.
+This means that if you say 'Preroll 10', then when the stream gets 10
+or more seconds behind, there is no Preroll left.
+
+Fixing this requires a change in the internals of how timestamps are
+handled.
+
+@section Does the @code{?date=} stuff work.
+
+Yes (subject to the limitation outlined above). Also note that whenever you
+start ffserver, it deletes the ffm file (if any parameters have changed),
+thus wiping out what you had recorded before.
+
+The format of the @code{?date=xxxxxx} is fairly flexible. You should use one
+of the following formats (the 'T' is literal):
+
+@example
+* YYYY-MM-DDTHH:MM:SS     (localtime)
+* YYYY-MM-DDTHH:MM:SSZ    (UTC)
+@end example
+
+You can omit the YYYY-MM-DD, and then it refers to the current day. However
+note that @samp{?date=16:00:00} refers to 16:00 on the current day -- this
+may be in the future and so is unlikely to be useful.
+
+You use this by adding the ?date= to the end of the URL for the stream.
+For example:   @samp{http://localhost:8080/test.asf?date=2002-07-26T23:05:00}.
+@c man end
+
+@chapter Options
+@c man begin OPTIONS
+
+@include fftools-common-opts.texi
+
+@section Main options
+
+@table @option
+@item -f @var{configfile}
+Read configuration file @file{configfile}. If not specified it will
+read by default from @file{/etc/ffserver.conf}.
+
+@item -n
+Enable no-launch mode. This option disables all the @code{Launch}
+directives within the various @code{<Feed>} sections. Since
+@command{ffserver} will not launch any @command{ffmpeg} instances, you
+will have to launch them manually.
+
+@item -d
+Enable debug mode. This option increases log verbosity, and directs
+log messages to stdout. When specified, the @option{CustomLog} option
+is ignored.
+@end table
+
+@chapter Configuration file syntax
+
+@command{ffserver} reads a configuration file containing global
+options and settings for each stream and feed.
+
+The configuration file consists of global options and dedicated
+sections, which must be introduced by "<@var{SECTION_NAME}
+@var{ARGS}>" on a separate line and must be terminated by a line in
+the form "</@var{SECTION_NAME}>". @var{ARGS} is optional.
+
+Currently the following sections are recognized: @samp{Feed},
+@samp{Stream}, @samp{Redirect}.
+
+A line starting with @code{#} is ignored and treated as a comment.
+
+Name of options and sections are case-insensitive.
+
+@section ACL syntax
+An ACL (Access Control List) specifies the address which are allowed
+to access a given stream, or to write a given feed.
+
+It accepts the following forms
+@itemize
+@item
+Allow/deny access to @var{address}.
+@example
+ACL ALLOW <address>
+ACL DENY <address>
+@end example
+
+@item
+Allow/deny access to ranges of addresses from @var{first_address} to
+@var{last_address}.
+@example
+ACL ALLOW <first_address> <last_address>
+ACL DENY <first_address> <last_address>
+@end example
+@end itemize
+
+You can repeat the ACL allow/deny as often as you like. It is on a per
+stream basis. The first match defines the action. If there are no matches,
+then the default is the inverse of the last ACL statement.
+
+Thus 'ACL allow localhost' only allows access from localhost.
+'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and
+allow everybody else.
+
+@section Global options
+@table @option
+@item HTTPPort @var{port_number}
+@item Port @var{port_number}
+@item RTSPPort @var{port_number}
+
+@var{HTTPPort} sets the HTTP server listening TCP port number,
+@var{RTSPPort} sets the RTSP server listening TCP port number.
+
+@var{Port} is the equivalent of @var{HTTPPort} and is deprecated.
+
+You must select a different port from your standard HTTP web server if
+it is running on the same computer.
+
+If not specified, no corresponding server will be created.
+
+@item HTTPBindAddress @var{ip_address}
+@item BindAddress @var{ip_address}
+@item RTSPBindAddress @var{ip_address}
+Set address on which the HTTP/RTSP server is bound. Only useful if you
+have several network interfaces.
+
+@var{BindAddress} is the equivalent of @var{HTTPBindAddress} and is
+deprecated.
+
+@item MaxHTTPConnections @var{n}
+Set number of simultaneous HTTP connections that can be handled. It
+has to be defined @emph{before} the @option{MaxClients} parameter,
+since it defines the @option{MaxClients} maximum limit.
+
+Default value is 2000.
+
+@item MaxClients @var{n}
+Set number of simultaneous requests that can be handled. Since
+@command{ffserver} is very fast, it is more likely that you will want
+to leave this high and use @option{MaxBandwidth}.
+
+Default value is 5.
+
+@item MaxBandwidth @var{kbps}
+Set the maximum amount of kbit/sec that you are prepared to consume
+when streaming to clients.
+
+Default value is 1000.
+
+@item CustomLog @var{filename}
+Set access log file (uses standard Apache log file format). '-' is the
+standard output.
+
+If not specified @command{ffserver} will produce no log.
+
+In case the commandline option @option{-d} is specified this option is
+ignored, and the log is written to standard output.
+
+@item NoDaemon
+Set no-daemon mode. This option is currently ignored since now
+@command{ffserver} will always work in no-daemon mode, and is
+deprecated.
+
+@item UseDefaults
+@item NoDefaults
+Control whether default codec options are used for the all streams or not.
+Each stream may overwrite this setting for its own. Default is @var{UseDefaults}.
+The last occurrence overrides the previous if multiple definitions exist.
+@end table
+
+@section Feed section
+
+A Feed section defines a feed provided to @command{ffserver}.
+
+Each live feed contains one video and/or audio sequence coming from an
+@command{ffmpeg} encoder or another @command{ffserver}. This sequence
+may be encoded simultaneously with several codecs at several
+resolutions.
+
+A feed instance specification is introduced by a line in the form:
+@example
+<Feed FEED_FILENAME>
+@end example
+
+where @var{FEED_FILENAME} specifies the unique name of the FFM stream.
+
+The following options are recognized within a Feed section.
+
+@table @option
+@item File @var{filename}
+@item ReadOnlyFile @var{filename}
+Set the path where the feed file is stored on disk.
+
+If not specified, the @file{/tmp/FEED.ffm} is assumed, where
+@var{FEED} is the feed name.
+
+If @option{ReadOnlyFile} is used the file is marked as read-only and
+it will not be deleted or updated.
+
+@item Truncate
+Truncate the feed file, rather than appending to it. By default
+@command{ffserver} will append data to the file, until the maximum
+file size value is reached (see @option{FileMaxSize} option).
+
+@item FileMaxSize @var{size}
+Set maximum size of the feed file in bytes. 0 means unlimited. The
+postfixes @code{K} (2^10), @code{M} (2^20), and @code{G} (2^30) are
+recognized.
+
+Default value is 5M.
+
+@item Launch @var{args}
+Launch an @command{ffmpeg} command when creating @command{ffserver}.
+
+@var{args} must be a sequence of arguments to be provided to an
+@command{ffmpeg} instance. The first provided argument is ignored, and
+it is replaced by a path with the same dirname of the @command{ffserver}
+instance, followed by the remaining argument and terminated with a
+path corresponding to the feed.
+
+When the launched process exits, @command{ffserver} will launch
+another program instance.
+
+In case you need a more complex @command{ffmpeg} configuration,
+e.g. if you need to generate multiple FFM feeds with a single
+@command{ffmpeg} instance, you should launch @command{ffmpeg} by hand.
+
+This option is ignored in case the commandline option @option{-n} is
+specified.
+
+@item ACL @var{spec}
+Specify the list of IP address which are allowed or denied to write
+the feed. Multiple ACL options can be specified.
+@end table
+
+@section Stream section
+
+A Stream section defines a stream provided by @command{ffserver}, and
+identified by a single name.
+
+The stream is sent when answering a request containing the stream
+name.
+
+A stream section must be introduced by the line:
+@example
+<Stream STREAM_NAME>
+@end example
+
+where @var{STREAM_NAME} specifies the unique name of the stream.
+
+The following options are recognized within a Stream section.
+
+Encoding options are marked with the @emph{encoding} tag, and they are
+used to set the encoding parameters, and are mapped to libavcodec
+encoding options. Not all encoding options are supported, in
+particular it is not possible to set encoder private options. In order
+to override the encoding options specified by @command{ffserver}, you
+can use the @command{ffmpeg} @option{override_ffserver} commandline
+option.
+
+Only one of the @option{Feed} and @option{File} options should be set.
+
+@table @option
+@item Feed @var{feed_name}
+Set the input feed. @var{feed_name} must correspond to an existing
+feed defined in a @code{Feed} section.
+
+When this option is set, encoding options are used to setup the
+encoding operated by the remote @command{ffmpeg} process.
+
+@item File @var{filename}
+Set the filename of the pre-recorded input file to stream.
+
+When this option is set, encoding options are ignored and the input
+file content is re-streamed as is.
+
+@item Format @var{format_name}
+Set the format of the output stream.
+
+Must be the name of a format recognized by FFmpeg. If set to
+@samp{status}, it is treated as a status stream.
+
+@item InputFormat @var{format_name}
+Set input format. If not specified, it is automatically guessed.
+
+@item Preroll @var{n}
+Set this to the number of seconds backwards in time to start. Note that
+most players will buffer 5-10 seconds of video, and also you need to allow
+for a keyframe to appear in the data stream.
+
+Default value is 0.
+
+@item StartSendOnKey
+Do not send stream until it gets the first key frame. By default
+@command{ffserver} will send data immediately.
+
+@item MaxTime @var{n}
+Set the number of seconds to run. This value set the maximum duration
+of the stream a client will be able to receive.
+
+A value of 0 means that no limit is set on the stream duration.
+
+@item ACL @var{spec}
+Set ACL for the stream.
+
+@item DynamicACL @var{spec}
+
+@item RTSPOption @var{option}
+
+@item MulticastAddress @var{address}
+
+@item MulticastPort @var{port}
+
+@item MulticastTTL @var{integer}
+
+@item NoLoop
+
+@item FaviconURL @var{url}
+Set favicon (favourite icon) for the server status page. It is ignored
+for regular streams.
+
+@item Author @var{value}
+@item Comment @var{value}
+@item Copyright @var{value}
+@item Title @var{value}
+Set metadata corresponding to the option. All these options are
+deprecated in favor of @option{Metadata}.
+
+@item Metadata @var{key} @var{value}
+Set metadata value on the output stream.
+
+@item UseDefaults
+@item NoDefaults
+Control whether default codec options are used for the stream or not.
+Default is @var{UseDefaults} unless disabled globally.
+
+@item NoAudio
+@item NoVideo
+Suppress audio/video.
+
+@item AudioCodec @var{codec_name} (@emph{encoding,audio})
+Set audio codec.
+
+@item AudioBitRate @var{rate} (@emph{encoding,audio})
+Set bitrate for the audio stream in kbits per second.
+
+@item AudioChannels @var{n} (@emph{encoding,audio})
+Set number of audio channels.
+
+@item AudioSampleRate @var{n} (@emph{encoding,audio})
+Set sampling frequency for audio. When using low bitrates, you should
+lower this frequency to 22050 or 11025. The supported frequencies
+depend on the selected audio codec.
+
+@item AVOptionAudio [@var{codec}:]@var{option} @var{value} (@emph{encoding,audio})
+Set generic or private option for audio stream.
+Private option must be prefixed with codec name or codec must be defined before.
+
+@item AVPresetAudio @var{preset} (@emph{encoding,audio})
+Set preset for audio stream.
+
+@item VideoCodec @var{codec_name} (@emph{encoding,video})
+Set video codec.
+
+@item VideoBitRate @var{n} (@emph{encoding,video})
+Set bitrate for the video stream in kbits per second.
+
+@item VideoBitRateRange @var{range} (@emph{encoding,video})
+Set video bitrate range.
+
+A range must be specified in the form @var{minrate}-@var{maxrate}, and
+specifies the @option{minrate} and @option{maxrate} encoding options
+expressed in kbits per second.
+
+@item VideoBitRateRangeTolerance @var{n} (@emph{encoding,video})
+Set video bitrate tolerance in kbits per second.
+
+@item PixelFormat @var{pixel_format} (@emph{encoding,video})
+Set video pixel format.
+
+@item Debug @var{integer} (@emph{encoding,video})
+Set video @option{debug} encoding option.
+
+@item Strict @var{integer} (@emph{encoding,video})
+Set video @option{strict} encoding option.
+
+@item VideoBufferSize @var{n} (@emph{encoding,video})
+Set ratecontrol buffer size, expressed in KB.
+
+@item VideoFrameRate @var{n} (@emph{encoding,video})
+Set number of video frames per second.
+
+@item VideoSize (@emph{encoding,video})
+Set size of the video frame, must be an abbreviation or in the form
+@var{W}x@var{H}.  See @ref{video size syntax,,the Video size section
+in the ffmpeg-utils(1) manual,ffmpeg-utils}.
+
+Default value is @code{160x128}.
+
+@item VideoIntraOnly (@emph{encoding,video})
+Transmit only intra frames (useful for low bitrates, but kills frame rate).
+
+@item VideoGopSize @var{n} (@emph{encoding,video})
+If non-intra only, an intra frame is transmitted every VideoGopSize
+frames. Video synchronization can only begin at an intra frame.
+
+@item VideoTag @var{tag} (@emph{encoding,video})
+Set video tag.
+
+@item VideoHighQuality (@emph{encoding,video})
+@item Video4MotionVector (@emph{encoding,video})
+
+@item BitExact (@emph{encoding,video})
+Set bitexact encoding flag.
+
+@item IdctSimple (@emph{encoding,video})
+Set simple IDCT algorithm.
+
+@item Qscale @var{n} (@emph{encoding,video})
+Enable constant quality encoding, and set video qscale (quantization
+scale) value, expressed in @var{n} QP units.
+
+@item VideoQMin @var{n} (@emph{encoding,video})
+@item VideoQMax @var{n} (@emph{encoding,video})
+Set video qmin/qmax.
+
+@item VideoQDiff @var{integer} (@emph{encoding,video})
+Set video @option{qdiff} encoding option.
+
+@item LumiMask @var{float} (@emph{encoding,video})
+@item DarkMask @var{float} (@emph{encoding,video})
+Set @option{lumi_mask}/@option{dark_mask} encoding options.
+
+@item AVOptionVideo [@var{codec}:]@var{option} @var{value} (@emph{encoding,video})
+Set generic or private option for video stream.
+Private option must be prefixed with codec name or codec must be defined before.
+
+@item AVPresetVideo @var{preset} (@emph{encoding,video})
+Set preset for video stream.
+
+@var{preset} must be the path of a preset file.
+@end table
+
+@subsection Server status stream
+
+A server status stream is a special stream which is used to show
+statistics about the @command{ffserver} operations.
+
+It must be specified setting the option @option{Format} to
+@samp{status}.
+
+@section Redirect section
+
+A redirect section specifies where to redirect the requested URL to
+another page.
+
+A redirect section must be introduced by the line:
+@example
+<Redirect NAME>
+@end example
+
+where @var{NAME} is the name of the page which should be redirected.
+
+It only accepts the option @option{URL}, which specify the redirection
+URL.
+
+@chapter Stream examples
+
+@itemize
+@item
+Multipart JPEG
+@example
+<Stream test.mjpg>
+Feed feed1.ffm
+Format mpjpeg
+VideoFrameRate 2
+VideoIntraOnly
+NoAudio
+Strict -1
+</Stream>
+@end example
+
+@item
+Single JPEG
+@example
+<Stream test.jpg>
+Feed feed1.ffm
+Format jpeg
+VideoFrameRate 2
+VideoIntraOnly
+VideoSize 352x240
+NoAudio
+Strict -1
+</Stream>
+@end example
+
+@item
+Flash
+@example
+<Stream test.swf>
+Feed feed1.ffm
+Format swf
+VideoFrameRate 2
+VideoIntraOnly
+NoAudio
+</Stream>
+@end example
+
+@item
+ASF compatible
+@example
+<Stream test.asf>
+Feed feed1.ffm
+Format asf
+VideoFrameRate 15
+VideoSize 352x240
+VideoBitRate 256
+VideoBufferSize 40
+VideoGopSize 30
+AudioBitRate 64
+StartSendOnKey
+</Stream>
+@end example
+
+@item
+MP3 audio
+@example
+<Stream test.mp3>
+Feed feed1.ffm
+Format mp2
+AudioCodec mp3
+AudioBitRate 64
+AudioChannels 1
+AudioSampleRate 44100
+NoVideo
+</Stream>
+@end example
+
+@item
+Ogg Vorbis audio
+@example
+<Stream test.ogg>
+Feed feed1.ffm
+Metadata title "Stream title"
+AudioBitRate 64
+AudioChannels 2
+AudioSampleRate 44100
+NoVideo
+</Stream>
+@end example
+
+@item
+Real with audio only at 32 kbits
+@example
+<Stream test.ra>
+Feed feed1.ffm
+Format rm
+AudioBitRate 32
+NoVideo
+</Stream>
+@end example
+
+@item
+Real with audio and video at 64 kbits
+@example
+<Stream test.rm>
+Feed feed1.ffm
+Format rm
+AudioBitRate 32
+VideoBitRate 128
+VideoFrameRate 25
+VideoGopSize 25
+</Stream>
+@end example
+
+@item
+For stream coming from a file: you only need to set the input filename
+and optionally a new format.
+
+@example
+<Stream file.rm>
+File "/usr/local/httpd/htdocs/tlive.rm"
+NoAudio
+</Stream>
+@end example
+
+@example
+<Stream file.asf>
+File "/usr/local/httpd/htdocs/test.asf"
+NoAudio
+Metadata author "Me"
+Metadata copyright "Super MegaCorp"
+Metadata title "Test stream from disk"
+Metadata comment "Test comment"
+</Stream>
+@end example
+@end itemize
+
+@c man end
+
+@include config.texi
+@ifset config-all
+@ifset config-avutil
+@include utils.texi
+@end ifset
+@ifset config-avcodec
+@include codecs.texi
+@include bitstream_filters.texi
+@end ifset
+@ifset config-avformat
+@include formats.texi
+@include protocols.texi
+@end ifset
+@ifset config-avdevice
+@include devices.texi
+@end ifset
+@ifset config-swresample
+@include resampler.texi
+@end ifset
+@ifset config-swscale
+@include scaler.texi
+@end ifset
+@ifset config-avfilter
+@include filters.texi
+@end ifset
+@end ifset
+
+@chapter See Also
+
+@ifhtml
+@ifset config-all
+@url{ffserver.html,ffserver},
+@end ifset
+@ifset config-not-all
+@url{ffserver-all.html,ffserver-all},
+@end ifset
+the @file{doc/ffserver.conf} example,
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
+@url{ffmpeg-utils.html,ffmpeg-utils},
+@url{ffmpeg-scaler.html,ffmpeg-scaler},
+@url{ffmpeg-resampler.html,ffmpeg-resampler},
+@url{ffmpeg-codecs.html,ffmpeg-codecs},
+@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters},
+@url{ffmpeg-formats.html,ffmpeg-formats},
+@url{ffmpeg-devices.html,ffmpeg-devices},
+@url{ffmpeg-protocols.html,ffmpeg-protocols},
+@url{ffmpeg-filters.html,ffmpeg-filters}
+@end ifhtml
+
+@ifnothtml
+@ifset config-all
+ffserver(1),
+@end ifset
+@ifset config-not-all
+ffserver-all(1),
+@end ifset
+the @file{doc/ffserver.conf} example, ffmpeg(1), ffplay(1), ffprobe(1),
+ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
+ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
+ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)
+@end ifnothtml
+
+@include authors.texi
+
+@ignore
+
+@setfilename ffserver
+@settitle ffserver video server
+
+@end ignore
+
+@bye

doc/fftools-common-opts.texi

@@ -34,25 +34,18 @@ Possible forms of stream specifiers are:
 @table @option
 @item @var{stream_index}
 Matches the stream with this index. E.g. @code{-threads:1 4} would set the
-thread count for the second stream to 4. If @var{stream_index} is used as an
-additional stream specifier (see below), then it selects stream number
-@var{stream_index} from the matching streams. Stream numbering is based on the
-order of the streams as detected by libavformat except when a program ID is
-also specified. In this case it is based on the ordering of the streams in the
-program.
-@item @var{stream_type}[:@var{additional_stream_specifier}]
+thread count for the second stream to 4.
+@item @var{stream_type}[:@var{stream_index}]
 @var{stream_type} is one of following: 'v' or 'V' for video, 'a' for audio, 's'
 for subtitle, 'd' for data, and 't' for attachments. 'v' matches all video
 streams, 'V' only matches video streams which are not attached pictures, video
-thumbnails or cover arts. If @var{additional_stream_specifier} is used, then
-it matches streams which both have this type and match the
-@var{additional_stream_specifier}. Otherwise, it matches all streams of the
-specified type.
-@item p:@var{program_id}[:@var{additional_stream_specifier}]
-Matches streams which are in the program with the id @var{program_id}. If
-@var{additional_stream_specifier} is used, then it matches streams which both
-are part of the program and match the @var{additional_stream_specifier}.
-
+thumbnails or cover arts.  If @var{stream_index} is given, then it matches
+stream number @var{stream_index} of this type. Otherwise, it matches all
+streams of this type.
+@item p:@var{program_id}[:@var{stream_index}]
+If @var{stream_index} is given, then it matches the stream with number @var{stream_index}
+in the program with the id @var{program_id}. Otherwise, it matches all streams in the
+program.
 @item #@var{stream_id} or i:@var{stream_id}
 Match the stream by stream id (e.g. PID in MPEG-TS container).
 @item m:@var{key}[:@var{value}]
@@ -109,10 +102,6 @@ Print detailed information about the muxer named @var{muxer_name}. Use the
 @item filter=@var{filter_name}
 Print detailed information about the filter name @var{filter_name}. Use the
 @option{-filters} option to get a list of all filters.
-
-@item bsf=@var{bitstream_filter_name}
-Print detailed information about the bitstream filter name @var{bitstream_filter_name}.
-Use the @option{-bsfs} option to get a list of all bitstream filters.
 @end table
 
 @item -version
@@ -179,24 +168,14 @@ The returned list cannot be assumed to be always complete.
 ffmpeg -sinks pulse,server=192.168.0.4
 @end example
 
-@item -loglevel [@var{flags}+]@var{loglevel} | -v [@var{flags}+]@var{loglevel}
-Set logging level and flags used by the library.
-
-The optional @var{flags} prefix can consist of the following values:
-@table @samp
-@item repeat
-Indicates that repeated log output should not be compressed to the first line
-and the "Last message repeated n times" line will be omitted.
-@item level
-Indicates that log output should add a @code{[level]} prefix to each message
-line. This can be used as an alternative to log coloring, e.g. when dumping the
-log to file.
-@end table
-Flags can also be used alone by adding a '+'/'-' prefix to set/reset a single
-flag without affecting other @var{flags} or changing @var{loglevel}. When
-setting both @var{flags} and @var{loglevel}, a '+' separator is expected
-between the last @var{flags} value and before @var{loglevel}.
-
+@item -loglevel [repeat+]@var{loglevel} | -v [repeat+]@var{loglevel}
+Set the logging level used by the library.
+Adding "repeat+" indicates that repeated log output should not be compressed
+to the first line and the "Last message repeated n times" line will be
+omitted. "repeat" can also be used alone.
+If "repeat" is used alone, and with no prior loglevel set, the default
+loglevel will be used. If multiple loglevel parameters are given, using
+'repeat' will not change the loglevel.
 @var{loglevel} is a string or a number containing one of the following values:
 @table @samp
 @item quiet, -8
@@ -222,17 +201,6 @@ Show everything, including debugging information.
 @item trace, 56
 @end table
 
-For example to enable repeated log output, add the @code{level} prefix, and set
-@var{loglevel} to @code{verbose}:
-@example
-ffmpeg -loglevel repeat+level+verbose -i input output
-@end example
-Another example that enables repeated log output without affecting current
-state of @code{level} prefix flag or @var{loglevel}:
-@example
-ffmpeg [...] -loglevel +repeat
-@end example
-
 By default the program logs to stderr. If coloring is supported by the
 terminal, colors are used to mark errors and warnings. Log coloring
 can be disabled setting the environment variable
@@ -246,7 +214,7 @@ Dump full command line and console output to a file named
 @code{@var{program}-@var{YYYYMMDD}-@var{HHMMSS}.log} in the current
 directory.
 This file can be useful for bug reports.
-It also implies @code{-loglevel debug}.
+It also implies @code{-loglevel verbose}.
 
 Setting the environment variable @env{FFREPORT} to any value has the
 same effect. If the value is a ':'-separated key=value sequence, these
@@ -347,6 +315,51 @@ Possible flags for this option are:
 @item k8
 @end table
 @end table
+
+@item -opencl_bench
+This option is used to benchmark all available OpenCL devices and print the
+results. This option is only available when FFmpeg has been compiled with
+@code{--enable-opencl}.
+
+When FFmpeg is configured with @code{--enable-opencl}, the options for the
+global OpenCL context are set via @option{-opencl_options}. See the
+"OpenCL Options" section in the ffmpeg-utils manual for the complete list of
+supported options. Amongst others, these options include the ability to select
+a specific platform and device to run the OpenCL code on. By default, FFmpeg
+will run on the first device of the first platform. While the options for the
+global OpenCL context provide flexibility to the user in selecting the OpenCL
+device of their choice, most users would probably want to select the fastest
+OpenCL device for their system.
+
+This option assists the selection of the most efficient configuration by
+identifying the appropriate device for the user's system. The built-in
+benchmark is run on all the OpenCL devices and the performance is measured for
+each device. The devices in the results list are sorted based on their
+performance with the fastest device listed first. The user can subsequently
+invoke @command{ffmpeg} using the device deemed most appropriate via
+@option{-opencl_options} to obtain the best performance for the OpenCL
+accelerated code.
+
+Typical usage to use the fastest OpenCL device involve the following steps.
+
+Run the command:
+@example
+ffmpeg -opencl_bench
+@end example
+Note down the platform ID (@var{pidx}) and device ID (@var{didx}) of the first
+i.e. fastest device in the list.
+Select the platform and device using the command:
+@example
+ffmpeg -opencl_options platform_idx=@var{pidx}:device_idx=@var{didx} ...
+@end example
+
+@item -opencl_options options (@emph{global})
+Set OpenCL environment options. This option is only available when
+FFmpeg has been compiled with @code{--enable-opencl}.
+
+@var{options} must be a list of @var{key}=@var{value} option pairs
+separated by ':'. See the ``OpenCL Options'' section in the
+ffmpeg-utils manual for the list of supported options.
 @end table
 
 @section AVOptions
@@ -372,15 +385,7 @@ ffmpeg -i input.flac -id3v2_version 3 out.mp3
 @end example
 
 All codec AVOptions are per-stream, and thus a stream specifier
-should be attached to them:
-@example
-ffmpeg -i multichannel.mxf -map 0:v:0 -map 0:a:0 -map 0:a:0 -c:a:0 ac3 -b:a:0 640k -ac:a:1 2 -c:a:1 aac -b:2 128k out.mp4
-@end example
-
-In the above example, a multichannel audio stream is mapped twice for output.
-The first instance is encoded with codec ac3 and bitrate 640k.
-The second instance is downmixed to 2 channels and encoded with codec aac. A bitrate of 128k is specified for it using
-absolute index of the output stream.
+should be attached to them.
 
 Note: the @option{-nooption} syntax cannot be used for boolean
 AVOptions, use @option{-option 0}/@option{-option 1}.

doc/formats.texi

@@ -30,43 +30,37 @@ latency. Must be an integer not lesser than 32. It is 5000000 by default.
 @item packetsize @var{integer} (@emph{output})
 Set packet size.
 
-@item fflags @var{flags}
-Set format flags. Some are implemented for a limited number of formats.
+@item fflags @var{flags} (@emph{input/output})
+Set format flags.
 
-Possible values for input files:
+Possible values:
 @table @samp
-@item discardcorrupt
-Discard corrupted packets.
+@item ignidx
+Ignore index.
 @item fastseek
 Enable fast, but inaccurate seeks for some formats.
 @item genpts
-Generate missing PTS if DTS is present.
-@item igndts
-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.
+Generate PTS.
 @item nofillin
-Do not fill in missing values in packet fields that can be exactly calculated.
+Do not fill in missing values that can be exactly calculated.
 @item noparse
 Disable AVParsers, this needs @code{+nofillin} too.
+@item igndts
+Ignore DTS.
+@item discardcorrupt
+Discard corrupted frames.
 @item sortdts
-Try to interleave output packets by DTS. At present, available only for AVIs with an index.
-@end table
-
-Possible values for output files:
-@table @samp
-@item autobsf
-Automatically apply bitstream filters as required by the output format. Enabled by default.
+Try to interleave output packets by DTS.
+@item keepside
+Do not merge side data.
+@item latm
+Enable RTP MP4A-LATM payload.
+@item nobuffer
+Reduce the latency introduced by optional buffering
 @item bitexact
 Only write platform-, build- and time-independent data.
 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
@@ -211,7 +205,7 @@ is @code{0} (meaning that no offset is applied).
 @item dump_separator @var{string} (@emph{input})
 Separator used to separate the fields printed on the command line about the
 Stream parameters.
-For example, to separate the fields with newlines and indentation:
+For example to separate the fields with newlines and indention:
 @example
 ffprobe -dump_separator "
                           "  -i ~/videos/matrixbench_mpeg2.mpg
@@ -220,32 +214,6 @@ ffprobe -dump_separator "
 @item max_streams @var{integer} (@emph{input})
 Specifies the maximum number of streams. This can be used to reject files that
 would require too many resources due to a large number of streams.
-
-@item skip_estimate_duration_from_pts @var{bool} (@emph{input})
-Skip estimation of input duration when calculated using PTS.
-At present, applicable for MPEG-PS and MPEG-TS.
-
-@item strict, f_strict @var{integer} (@emph{input/output})
-Specify how strictly to follow the standards. @code{f_strict} is deprecated and
-should be used only via the @command{ffmpeg} tool.
-
-Possible values:
-@table @samp
-@item very
-strictly conform to an older more strict version of the spec or reference software
-@item strict
-strictly conform to all the things in the spec no matter what consequences
-@item normal
-
-@item unofficial
-allow unofficial extensions
-@item experimental
-allow non standardized experimental things, experimental
-(unfinished/work in progress/not well tested) decoders and encoders.
-Note: experimental decoders can pose a security risk, do not use this for
-decoding untrusted input.
-@end table
-
 @end table
 
 @c man end FORMAT OPTIONS
@@ -256,10 +224,30 @@ decoding untrusted input.
 Format stream specifiers allow selection of one or more streams that
 match specific properties.
 
+Possible forms of stream specifiers are:
+@table @option
+@item @var{stream_index}
+Matches the stream with this index.
+
+@item @var{stream_type}[:@var{stream_index}]
+@var{stream_type} is one of following: 'v' for video, 'a' for audio,
+'s' for subtitle, 'd' for data, and 't' for attachments. If
+@var{stream_index} is given, then it matches the stream number
+@var{stream_index} of this type. Otherwise, it matches all streams of
+this type.
+
+@item p:@var{program_id}[:@var{stream_index}]
+If @var{stream_index} is given, then it matches the stream with number
+@var{stream_index} in the program with the id
+@var{program_id}. Otherwise, it matches all streams in the program.
+
+@item #@var{stream_id}
+Matches the stream by a format-specific ID.
+@end table
+
 The exact semantics of stream specifiers is defined by the
 @code{avformat_match_stream_specifier()} function declared in the
-@file{libavformat/avformat.h} header and documented in the
-@ref{Stream specifiers,,Stream specifiers section in the ffmpeg(1) manual,ffmpeg}.
+@file{libavformat/avformat.h} header.
 
 @ifclear config-writeonly
 @include demuxers.texi

doc/general.texi

@@ -17,164 +17,13 @@ for more formats. None of them are used by default, their use has to be
 explicitly requested by passing the appropriate flags to
 @command{./configure}.
 
-@section Alliance for Open Media (AOM)
+@section OpenJPEG
 
-FFmpeg can make use of the AOM library for AV1 decoding and encoding.
+FFmpeg can use the OpenJPEG libraries for encoding/decoding J2K videos.  Go to
+@url{http://www.openjpeg.org/} to get the libraries and follow the installation
+instructions.  To enable using OpenJPEG in FFmpeg, pass @code{--enable-libopenjpeg} to
+@file{./configure}.
 
-Go to @url{http://aomedia.org/} and follow the instructions for
-installing the library. Then pass @code{--enable-libaom} to configure to
-enable it.
-
-@section AMD AMF/VCE
-
-FFmpeg can use the AMD Advanced Media Framework library under Windows
-for accelerated H.264 and HEVC encoding on hardware with Video Coding Engine (VCE).
-
-To enable support you must obtain the AMF framework header files from
-@url{https://github.com/GPUOpen-LibrariesAndSDKs/AMF.git}.
-
-Create an @code{AMF/} directory in the system include path.
-Copy the contents of @code{AMF/amf/public/include/} into that directory.
-Then configure FFmpeg with @code{--enable-amf}.
-
-@section AviSynth
-
-FFmpeg can read AviSynth scripts as input. To enable support, pass
-@code{--enable-avisynth} to configure.  The correct headers are
-included in compat/avisynth/, which allows the user to enable support
-without needing to search for these headers themselves.
-
-For Windows, supported AviSynth variants are
-@url{http://avisynth.nl, AviSynth 2.6 RC1 or higher} for 32-bit builds and
-@url{http://avisynth.nl/index.php/AviSynth+, AviSynth+ r1718 or higher} for 32-bit and 64-bit builds.
-
-For Linux and OS X, the supported AviSynth variant is
-@url{https://github.com/avxsynth/avxsynth, AvxSynth}.
-
-@float NOTE
-In 2016, AviSynth+ added support for building with GCC. However, due to
-the eccentricities of Windows' calling conventions, 32-bit GCC builds
-of AviSynth+ are not compatible with typical 32-bit builds of FFmpeg.
-
-By default, FFmpeg assumes compatibility with 32-bit MSVC builds of
-AviSynth+ since that is the most widely-used and entrenched build
-configuration.  Users can override this and enable support for 32-bit
-GCC builds of AviSynth+ by passing @code{-DAVSC_WIN32_GCC32} to
-@code{--extra-cflags} when configuring FFmpeg.
-
-64-bit builds of FFmpeg are not affected, and can use either MSVC or
-GCC builds of AviSynth+ without any special flags.
-@end float
-
-@float NOTE
-AviSynth and AvxSynth are loaded dynamically.  Distributors can build FFmpeg
-with @code{--enable-avisynth}, and the binaries will work regardless of the
-end user having AviSynth or AvxSynth installed - they'll only need to be
-installed to use AviSynth scripts (obviously).
-@end float
-
-@section Chromaprint
-
-FFmpeg can make use of the Chromaprint library for generating audio fingerprints.
-Pass @code{--enable-chromaprint} to configure to
-enable it. See @url{https://acoustid.org/chromaprint}.
-
-@section codec2
-
-FFmpeg can make use of the codec2 library for codec2 decoding and encoding.
-There is currently no native decoder, so libcodec2 must be used for decoding.
-
-Go to @url{http://freedv.org/}, download "Codec 2 source archive".
-Build and install using CMake. Debian users can install the libcodec2-dev package instead.
-Once libcodec2 is installed you can pass @code{--enable-libcodec2} to configure to enable it.
-
-The easiest way to use codec2 is with .c2 files, since they contain the mode information required for decoding.
-To encode such a file, use a .c2 file extension and give the libcodec2 encoder the -mode option:
-@code{ffmpeg -i input.wav -mode 700C output.c2}.
-Playback is as simple as @code{ffplay output.c2}.
-For a list of supported modes, run @code{ffmpeg -h encoder=libcodec2}.
-Raw codec2 files are also supported.
-To make sense of them the mode in use needs to be specified as a format option:
-@code{ffmpeg -f codec2raw -mode 1300 -i input.raw output.wav}.
-
-@section dav1d
-
-FFmpeg can make use of the dav1d library for AV1 video decoding.
-
-Go to @url{https://code.videolan.org/videolan/dav1d} and follow the instructions for
-installing the library. Then pass @code{--enable-libdav1d} to configure to enable it.
-
-@section davs2
-
-FFmpeg can make use of the davs2 library for AVS2-P2/IEEE1857.4 video decoding.
-
-Go to @url{https://github.com/pkuvcl/davs2} and follow the instructions for
-installing the library. Then pass @code{--enable-libdavs2} to configure to
-enable it.
-
-@float NOTE
-libdavs2 is under the GNU Public License Version 2 or later
-(see @url{http://www.gnu.org/licenses/old-licenses/gpl-2.0.html} for
-details), you must upgrade FFmpeg's license to GPL in order to use it.
-@end float
-
-@section Game Music Emu
-
-FFmpeg can make use of the Game Music Emu library to read audio from supported video game
-music file formats. Pass @code{--enable-libgme} to configure to
-enable it. See @url{https://bitbucket.org/mpyne/game-music-emu/overview}.
-
-@section Intel QuickSync Video
-
-FFmpeg can use Intel QuickSync Video (QSV) for accelerated decoding and encoding
-of multiple codecs. To use QSV, FFmpeg must be linked against the @code{libmfx}
-dispatcher, which loads the actual decoding libraries.
-
-The dispatcher is open source and can be downloaded from
-@url{https://github.com/lu-zero/mfx_dispatch.git}. FFmpeg needs to be configured
-with the @code{--enable-libmfx} option and @code{pkg-config} needs to be able to
-locate the dispatcher's @code{.pc} files.
-
-@section Kvazaar
-
-FFmpeg can make use of the Kvazaar library for HEVC encoding.
-
-Go to @url{https://github.com/ultravideo/kvazaar} and follow the
-instructions for installing the library. Then pass
-@code{--enable-libkvazaar} to configure to enable it.
-
-@section LAME
-
-FFmpeg can make use of the LAME library for MP3 encoding.
-
-Go to @url{http://lame.sourceforge.net/} and follow the
-instructions for installing the library.
-Then pass @code{--enable-libmp3lame} to configure to enable it.
-
-@section libilbc
-
-iLBC is a narrowband speech codec that has been made freely available
-by Google as part of the WebRTC project. libilbc is a packaging friendly
-copy of the iLBC codec. FFmpeg can make use of the libilbc library for
-iLBC decoding and encoding.
-
-Go to @url{https://github.com/TimothyGu/libilbc} and follow the instructions for
-installing the library. Then pass @code{--enable-libilbc} to configure to
-enable it.
-
-@section libvpx
-
-FFmpeg can make use of the libvpx library for VP8/VP9 decoding and encoding.
-
-Go to @url{http://www.webmproject.org/} and follow the instructions for
-installing the library. Then pass @code{--enable-libvpx} to configure to
-enable it.
-
-@section ModPlug
-
-FFmpeg can make use of this library, originating in Modplug-XMMS, to read from MOD-like music files.
-See @url{https://github.com/Konstanty/libmodplug}. Pass @code{--enable-libmodplug} to configure to
-enable it.
 
 @section OpenCORE, VisualOn, and Fraunhofer libraries
 
@@ -189,10 +38,9 @@ upgrade FFmpeg's license to LGPL version 3 (or if you have enabled
 GPL components, GPL version 3) by passing @code{--enable-version3} to configure in
 order to use it.
 
-The license of the Fraunhofer AAC library is incompatible with the GPL.
-Therefore, for GPL builds, you have to pass @code{--enable-nonfree} to
-configure in order to use it. To the best of our knowledge, it is
-compatible with the LGPL.
+The Fraunhofer AAC library is licensed under a license incompatible to the GPL
+and is not known to be compatible to the LGPL. Therefore, you have to pass
+@code{--enable-nonfree} to configure to use it.
 @end float
 
 @subsection OpenCORE AMR
@@ -215,15 +63,55 @@ Then pass @code{--enable-libvo-amrwbenc} to configure to enable it.
 
 @subsection Fraunhofer AAC library
 
-FFmpeg can make use of the Fraunhofer AAC library for AAC decoding & encoding.
+FFmpeg can make use of the Fraunhofer AAC library for AAC encoding.
 
 Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the
 instructions for installing the library.
 Then pass @code{--enable-libfdk-aac} to configure to enable it.
 
+@section LAME
+
+FFmpeg can make use of the LAME library for MP3 encoding.
+
+Go to @url{http://lame.sourceforge.net/} and follow the
+instructions for installing the library.
+Then pass @code{--enable-libmp3lame} to configure to enable it.
+
+@section TwoLAME
+
+FFmpeg can make use of the TwoLAME library for MP2 encoding.
+
+Go to @url{http://www.twolame.org/} and follow the
+instructions for installing the library.
+Then pass @code{--enable-libtwolame} to configure to enable it.
+
+@section libvpx
+
+FFmpeg can make use of the libvpx library for VP8/VP9 encoding.
+
+Go to @url{http://www.webmproject.org/} and follow the instructions for
+installing the library. Then pass @code{--enable-libvpx} to configure to
+enable it.
+
+@section libwavpack
+
+FFmpeg can make use of the libwavpack library for WavPack encoding.
+
+Go to @url{http://www.wavpack.com/} and follow the instructions for
+installing the library. Then pass @code{--enable-libwavpack} to configure to
+enable it.
+
+@section libxavs
+
+FFmpeg can make use of the libxavs library for Xavs encoding.
+
+Go to @url{http://xavs.sf.net/} and follow the instructions for
+installing the library. Then pass @code{--enable-libxavs} to configure to
+enable it.
+
 @section OpenH264
 
-FFmpeg can make use of the OpenH264 library for H.264 decoding and encoding.
+FFmpeg can make use of the OpenH264 library for H.264 encoding and decoding.
 
 Go to @url{http://www.openh264.org/} and follow the instructions for
 installing the library. Then pass @code{--enable-libopenh264} to configure to
@@ -236,40 +124,6 @@ constrained baseline profile and CABAC.) Using it is mostly useful for
 testing and for taking advantage of Cisco's patent portfolio license
 (@url{http://www.openh264.org/BINARY_LICENSE.txt}).
 
-@section OpenJPEG
-
-FFmpeg can use the OpenJPEG libraries for decoding/encoding J2K videos.  Go to
-@url{http://www.openjpeg.org/} to get the libraries and follow the installation
-instructions.  To enable using OpenJPEG in FFmpeg, pass @code{--enable-libopenjpeg} to
-@file{./configure}.
-
-@section TwoLAME
-
-FFmpeg can make use of the TwoLAME library for MP2 encoding.
-
-Go to @url{http://www.twolame.org/} and follow the
-instructions for installing the library.
-Then pass @code{--enable-libtwolame} to configure to enable it.
-
-@section VapourSynth
-
-FFmpeg can read VapourSynth scripts as input. To enable support, pass
-@code{--enable-vapoursynth} to configure. Vapoursynth is detected via
-@code{pkg-config}. Versions 42 or greater supported.
-See @url{http://www.vapoursynth.com/}.
-
-Due to security concerns, Vapoursynth scripts will not
-be autodetected so the input format has to be forced. For ff* CLI tools,
-add @code{-f vapoursynth} before the input @code{-i yourscript.vpy}.
-
-@section WavPack
-
-FFmpeg can make use of the libwavpack library for WavPack encoding.
-
-Go to @url{http://www.wavpack.com/} and follow the instructions for
-installing the library. Then pass @code{--enable-libwavpack} to configure to
-enable it.
-
 @section x264
 
 FFmpeg can make use of the x264 library for H.264 encoding.
@@ -298,37 +152,80 @@ x265 is under the GNU Public License Version 2 or later
 details), you must upgrade FFmpeg's license to GPL in order to use it.
 @end float
 
-@section xavs
+@section kvazaar
 
-FFmpeg can make use of the xavs library for AVS encoding.
+FFmpeg can make use of the kvazaar library for HEVC encoding.
 
-Go to @url{http://xavs.sf.net/} and follow the instructions for
-installing the library. Then pass @code{--enable-libxavs} to configure to
+Go to @url{https://github.com/ultravideo/kvazaar} and follow the
+instructions for installing the library. Then pass
+@code{--enable-libkvazaar} to configure to enable it.
+
+@section libilbc
+
+iLBC is a narrowband speech codec that has been made freely available
+by Google as part of the WebRTC project. libilbc is a packaging friendly
+copy of the iLBC codec. FFmpeg can make use of the libilbc library for
+iLBC encoding and decoding.
+
+Go to @url{https://github.com/TimothyGu/libilbc} and follow the instructions for
+installing the library. Then pass @code{--enable-libilbc} to configure to
 enable it.
 
-@section xavs2
+@section libzvbi
 
-FFmpeg can make use of the xavs2 library for AVS2-P2/IEEE1857.4 video encoding.
-
-Go to @url{https://github.com/pkuvcl/xavs2} and follow the instructions for
-installing the library. Then pass @code{--enable-libxavs2} to configure to
-enable it.
-
-@float NOTE
-libxavs2 is under the GNU Public License Version 2 or later
-(see @url{http://www.gnu.org/licenses/old-licenses/gpl-2.0.html} for
-details), you must upgrade FFmpeg's license to GPL in order to use it.
-@end float
-
-@section ZVBI
-
-ZVBI is a VBI decoding library which can be used by FFmpeg to decode DVB
+libzvbi is a VBI decoding library which can be used by FFmpeg to decode DVB
 teletext pages and DVB teletext subtitles.
 
 Go to @url{http://sourceforge.net/projects/zapping/} and follow the instructions for
 installing the library. Then pass @code{--enable-libzvbi} to configure to
 enable it.
 
+@section AviSynth
+
+FFmpeg can read AviSynth scripts as input. To enable support, pass
+@code{--enable-avisynth} to configure.  The correct headers are
+included in compat/avisynth/, which allows the user to enable support
+without needing to search for these headers themselves.
+
+For Windows, supported AviSynth variants are
+@url{http://avisynth.nl, AviSynth 2.6 RC1 or higher} for 32-bit builds and
+@url{http://avs-plus.net, AviSynth+ r1718 or higher} for 32-bit and 64-bit builds.
+
+For Linux and OS X, the supported AviSynth variant is
+@url{https://github.com/avxsynth/avxsynth, AvxSynth}.
+
+@float NOTE
+There is currently a regression in AviSynth+'s @code{capi.h} header as of
+October 2016, which interferes with the ability for builds of FFmpeg to use
+MSVC-built binaries of AviSynth. Until this is resolved, you can make sure
+a known good version is installed by checking out a version from before
+the regression occurred:
+
+@code{git clone -b MT git://github.com/AviSynth/AviSynthPlus.git @*
+cd AviSynthPlus @*
+git checkout -b oldheader b4f292b4dbfad149697fb65c6a037bb3810813f9 @*
+make install PREFIX=/install/prefix}
+@end float
+
+@float NOTE
+AviSynth and AvxSynth are loaded dynamically.  Distributors can build FFmpeg
+with @code{--enable-avisynth}, and the binaries will work regardless of the
+end user having AviSynth or AvxSynth installed - they'll only need to be
+installed to use AviSynth scripts (obviously).
+@end float
+
+@section Intel QuickSync Video
+
+FFmpeg can use Intel QuickSync Video (QSV) for accelerated encoding and decoding
+of multiple codecs. To use QSV, FFmpeg must be linked against the @code{libmfx}
+dispatcher, which loads the actual decoding libraries.
+
+The dispatcher is open source and can be downloaded from
+@url{https://github.com/lu-zero/mfx_dispatch.git}. FFmpeg needs to be configured
+with the @code{--enable-libmfx} option and @code{pkg-config} needs to be able to
+locate the dispatcher's @code{.pc} files.
+
+
 @chapter Supported File Formats, Codecs or Features
 
 You can use the @code{-formats} and @code{-codecs} options to have an exhaustive list.
@@ -393,10 +290,6 @@ library:
 @item BRSTM                     @tab   @tab X
     @tab Audio format used on the Nintendo Wii.
 @item BWF                       @tab X @tab X
-@item codec2 (raw)              @tab X @tab X
-    @tab Must be given -mode format option to decode correctly.
-@item codec2 (.c2 files)        @tab X @tab X
-    @tab Contains header with version and mode info, simplifying playback.
 @item CRI ADX                   @tab X @tab X
     @tab Audio-only format used in console video games.
 @item Discworld II BMV          @tab   @tab X
@@ -468,13 +361,10 @@ library:
 @item IEC61937 encapsulation @tab X @tab X
 @item IFF                       @tab   @tab X
     @tab Interchange File Format
-@item IFV                       @tab   @tab X
-    @tab A format used by some old CCTV DVRs.
 @item iLBC                      @tab X @tab X
 @item Interplay MVE             @tab   @tab X
     @tab Format used in various Interplay computer games.
 @item Iterated Systems ClearVideo @tab     @tab  X
-    @tab I-frames only
 @item IV8                       @tab   @tab X
     @tab A format generated by IndigoVision 8000 video server.
 @item IVF (On2)                 @tab X @tab X
@@ -534,7 +424,6 @@ library:
 @item NC camera feed            @tab   @tab X
     @tab NC (AVIP NC4600) camera streams
 @item NIST SPeech HEader REsources @tab   @tab X
-@item Computerized Speech Lab NSP @tab   @tab X
 @item NTT TwinVQ (VQF)          @tab   @tab X
     @tab Nippon Telegraph and Telephone Corporation TwinVQ.
 @item Nullsoft Streaming Video  @tab   @tab X
@@ -549,10 +438,6 @@ library:
 @item QCP                       @tab   @tab X
 @item raw ADTS (AAC)            @tab X @tab X
 @item raw AC-3                  @tab X @tab X
-@item raw AMR-NB                @tab   @tab X
-@item raw AMR-WB                @tab   @tab X
-@item raw aptX                  @tab X @tab X
-@item raw aptX HD               @tab X @tab X
 @item raw Chinese AVS video     @tab X @tab X
 @item raw CRI ADX               @tab X @tab X
 @item raw Dirac                 @tab X @tab X
@@ -576,14 +461,12 @@ library:
 @item raw NULL                  @tab X @tab
 @item raw video                 @tab X @tab X
 @item raw id RoQ                @tab X @tab
-@item raw SBC                   @tab X @tab X
 @item raw Shorten               @tab   @tab X
 @item raw TAK                   @tab   @tab X
 @item raw TrueHD                @tab X @tab X
 @item raw VC-1                  @tab X @tab X
 @item raw PCM A-law             @tab X @tab X
 @item raw PCM mu-law            @tab X @tab X
-@item raw PCM Archimedes VIDC   @tab X @tab X
 @item raw PCM signed 8 bit      @tab X @tab X
 @item raw PCM signed 16 bit big-endian  @tab X @tab X
 @item raw PCM signed 16 bit little-endian  @tab X @tab X
@@ -627,8 +510,7 @@ library:
 @item SAP                       @tab X @tab X
 @item SBG                       @tab   @tab X
 @item SDP                       @tab   @tab X
-@item SER                       @tab   @tab X
-@item Sega FILM/CPK             @tab X @tab X
+@item Sega FILM/CPK             @tab   @tab X
     @tab Used in many Sega Saturn console games.
 @item Silicon Graphics Movie    @tab   @tab X
 @item Sierra SOL                @tab   @tab X
@@ -688,7 +570,7 @@ library:
 @item Psygnosis YOP             @tab   @tab X
 @end multitable
 
-@code{X} means that the feature in that column (encoding / decoding) is supported.
+@code{X} means that encoding (resp. decoding) is supported.
 
 @section Image Formats
 
@@ -758,7 +640,7 @@ following image formats are supported:
     @tab X Window Dump image format
 @end multitable
 
-@code{X} means that the feature in that column (encoding / decoding) is supported.
+@code{X} means that encoding (resp. decoding) is supported.
 
 @code{E} means that support is provided through an external library.
 
@@ -796,8 +678,6 @@ following image formats are supported:
 @item Autodesk Animator Flic video  @tab     @tab  X
 @item Autodesk RLE           @tab     @tab  X
     @tab fourcc: AASC
-@item AV1                    @tab  E  @tab  E
-    @tab Supported through external libraries libaom and libdav1d
 @item Avid 1:1 10-bit RGB Packer  @tab  X  @tab  X
     @tab fourcc: AVrp
 @item AVS (Audio Video Standard) video  @tab     @tab  X
@@ -871,7 +751,6 @@ following image formats are supported:
     @tab fourcc: G2M2, G2M3
 @item Go2Webinar             @tab     @tab  X
     @tab fourcc: G2M4
-@item Gremlin Digital Video  @tab     @tab  X
 @item H.261                  @tab  X  @tab  X
 @item H.263 / H.263-1996     @tab  X  @tab  X
 @item H.263+ / H.263-1998 / H.263 version 2  @tab  X  @tab  X
@@ -892,7 +771,6 @@ following image formats are supported:
     @tab IFF interleaved bitmap
 @item IFF ByteRun1           @tab     @tab  X
     @tab IFF run length encoded bitmap
-@item Infinity IMM4          @tab     @tab  X
 @item Intel H.263            @tab     @tab  X
 @item Intel Indeo 2          @tab     @tab  X
 @item Intel Indeo 3          @tab     @tab  X
@@ -914,7 +792,7 @@ following image formats are supported:
 @item LucasArts SANM/Smush   @tab     @tab  X
     @tab Used in LucasArts games / SMUSH animations.
 @item lossless MJPEG         @tab  X  @tab  X
-@item MagicYUV Video         @tab  X  @tab  X
+@item MagicYUV Video         @tab     @tab  X
 @item Mandsoft Screen Capture Codec  @tab     @tab  X
 @item Microsoft ATC Screen   @tab     @tab  X
     @tab Also known as Microsoft Screen 3.
@@ -946,8 +824,6 @@ following image formats are supported:
     @tab Video encoding used in NuppelVideo files.
 @item On2 VP3                @tab     @tab  X
     @tab still experimental
-@item On2 VP4                @tab     @tab  X
-    @tab fourcc: VP40
 @item On2 VP5                @tab     @tab  X
     @tab fourcc: VP50
 @item On2 VP6                @tab     @tab  X
@@ -1042,7 +918,7 @@ following image formats are supported:
     @tab Encoder works only in PAL8.
 @end multitable
 
-@code{X} means that the feature in that column (encoding / decoding) is supported.
+@code{X} means that encoding (resp. decoding) is supported.
 
 @code{E} means that support is provided through an external library.
 
@@ -1114,20 +990,13 @@ following image formats are supported:
 @item Amazing Studio PAF Audio @tab     @tab  X
 @item Apple lossless audio   @tab  X  @tab  X
     @tab QuickTime fourcc 'alac'
-@item aptX                   @tab  X  @tab  X
-    @tab Used in Bluetooth A2DP
-@item aptX HD                @tab  X  @tab  X
-    @tab Used in Bluetooth A2DP
 @item ATRAC1                 @tab     @tab  X
 @item ATRAC3                 @tab     @tab  X
 @item ATRAC3+                @tab     @tab  X
-@item ATRAC9                 @tab     @tab  X
 @item Bink Audio             @tab     @tab  X
     @tab Used in Bink and Smacker files in many games.
 @item CELT                   @tab     @tab  E
     @tab decoding supported through external library libcelt
-@item codec2                 @tab  E  @tab  E
-    @tab en/decoding supported through external library libcodec2
 @item Delphine Software International CIN audio  @tab     @tab  X
     @tab Codec used in Delphine Software International games.
 @item Digital Speech Standard - Standard Play mode (DSS SP) @tab     @tab  X
@@ -1148,10 +1017,10 @@ following image formats are supported:
 @item DPCM Sol               @tab     @tab  X
 @item DPCM Xan               @tab     @tab  X
     @tab Used in Origin's Wing Commander IV AVI files.
-@item DSD (Direct Stream Digital), least significant bit first  @tab  @tab  X
-@item DSD (Direct Stream Digital), most significant bit first   @tab  @tab  X
-@item DSD (Direct Stream Digital), least significant bit first, planar  @tab  @tab  X
-@item DSD (Direct Stream Digital), most significant bit first, planar   @tab  @tab  X
+@item DSD (Direct Stream Digitial), least significant bit first  @tab  @tab  X
+@item DSD (Direct Stream Digitial), most significant bit first   @tab  @tab  X
+@item DSD (Direct Stream Digitial), least significant bit first, planar  @tab  @tab  X
+@item DSD (Direct Stream Digitial), most significant bit first, planar   @tab  @tab  X
 @item DSP Group TrueSpeech   @tab     @tab  X
 @item DST (Direct Stream Transfer) @tab  @tab  X
 @item DV audio               @tab     @tab  X
@@ -1188,7 +1057,6 @@ following image formats are supported:
     @tab encoding supported through external library libopus
 @item PCM A-law              @tab  X  @tab  X
 @item PCM mu-law             @tab  X  @tab  X
-@item PCM Archimedes VIDC    @tab  X  @tab  X
 @item PCM signed 8-bit planar  @tab  X  @tab  X
 @item PCM signed 16-bit big-endian planar  @tab  X  @tab  X
 @item PCM signed 16-bit little-endian planar  @tab  X  @tab  X
@@ -1227,8 +1095,6 @@ 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 SBC (low-complexity subband codec) @tab  X  @tab  X
-    @tab Used in Bluetooth A2DP
 @item Shorten                @tab     @tab  X
 @item Sierra VMD audio       @tab     @tab  X
     @tab Used in Sierra VMD files.
@@ -1261,7 +1127,7 @@ following image formats are supported:
 @item Xbox Media Audio 2     @tab     @tab  X
 @end multitable
 
-@code{X} means that the feature in that column (encoding / decoding) is supported.
+@code{X} means that encoding (resp. decoding) is supported.
 
 @code{E} means that support is provided through an external library.
 

doc/indevs.texi

@@ -63,46 +63,6 @@ Set the number of channels. Default is 2.
 
 @end table
 
-@section android_camera
-
-Android camera input device.
-
-This input devices uses the Android Camera2 NDK API which is
-available on devices with API level 24+. The availability of
-android_camera is autodetected during configuration.
-
-This device allows capturing from all cameras on an Android device,
-which are integrated into the Camera2 NDK API.
-
-The available cameras are enumerated internally and can be selected
-with the @var{camera_index} parameter. The input file string is
-discarded.
-
-Generally the back facing camera has index 0 while the front facing
-camera has index 1.
-
-@subsection Options
-
-@table @option
-
-@item video_size
-Set the video size given as a string such as 640x480 or hd720.
-Falls back to the first available configuration reported by
-Android if requested video size is not available or by default.
-
-@item framerate
-Set the video framerate.
-Falls back to the first available configuration reported by
-Android if requested framerate is not available or by default (-1).
-
-@item camera_index
-Set the index of the camera to use. Default is 0.
-
-@item input_queue_size
-Set the maximum number of frames to buffer. Default is 5.
-
-@end table
-
 @section avfoundation
 
 AVFoundation input device.
@@ -178,9 +138,6 @@ Capture the mouse pointer. Default is 0.
 @item -capture_mouse_clicks
 Capture the screen mouse clicks. Default is 0.
 
-@item -capture_raw_data
-Capture the raw device data. Default is 0.
-Using this option may result in receiving the underlying data delivered to the AVFoundation framework. E.g. for muxed devices that sends raw DV data to the framework (like tape-based camcorders), setting this option to false results in extracted video frames captured in the designated pixel format only. Setting this option to true results in receiving the raw DV stream untouched.
 @end table
 
 @subsection Examples
@@ -211,13 +168,6 @@ Record video from the system default video device using the pixel format bgr0 an
 $ ffmpeg -f avfoundation -pixel_format bgr0 -i "default:none" out.avi
 @end example
 
-@item
-Record raw DV data from a suitable input device and write the output into out.dv:
-@example
-$ ffmpeg -f avfoundation -capture_raw_data true -i "zr100:none" out.dv
-@end example
-
-
 @end itemize
 
 @section bktr
@@ -277,8 +227,7 @@ audio track.
 
 @item list_devices
 If set to @option{true}, print a list of devices and exit.
-Defaults to @option{false}. Alternatively you can use the @code{-sources}
-option of ffmpeg to list the available input devices.
+Defaults to @option{false}.
 
 @item list_formats
 If set to @option{true}, print a list of supported formats and exit.
@@ -289,8 +238,6 @@ This sets the input video format to the format given by the FourCC. To see
 the supported values of your device(s) use @option{list_formats}.
 Note that there is a FourCC @option{'pal '} that can also be used
 as @option{pal} (3 letters).
-Default behavior is autodetection of the input video format, if the hardware
-supports it.
 
 @item bm_v210
 This is a deprecated option, you can use @option{raw_format} instead.
@@ -337,12 +284,6 @@ Defaults to @samp{2}.
 Sets the decklink device duplex mode. Must be @samp{unset}, @samp{half} or @samp{full}.
 Defaults to @samp{unset}.
 
-@item timecode_format
-Timecode type to include in the frame and video stream metadata. Must be
-@samp{none}, @samp{rp188vitc}, @samp{rp188vitc2}, @samp{rp188ltc},
-@samp{rp188any}, @samp{vitc}, @samp{vitc2}, or @samp{serial}. Defaults to
-@samp{none} (not included).
-
 @item video_input
 Sets the video input source. Must be @samp{unset}, @samp{sdi}, @samp{hdmi},
 @samp{optical_sdi}, @samp{component}, @samp{composite} or @samp{s_video}.
@@ -355,13 +296,11 @@ Sets the audio input source. Must be @samp{unset}, @samp{embedded},
 
 @item video_pts
 Sets the video packet timestamp source. Must be @samp{video}, @samp{audio},
-@samp{reference}, @samp{wallclock} or @samp{abs_wallclock}.
-Defaults to @samp{video}.
+@samp{reference} or @samp{wallclock}. Defaults to @samp{video}.
 
 @item audio_pts
 Sets the audio packet timestamp source. Must be @samp{video}, @samp{audio},
-@samp{reference}, @samp{wallclock} or @samp{abs_wallclock}.
-Defaults to @samp{audio}.
+@samp{reference} or @samp{wallclock}. Defaults to @samp{audio}.
 
 @item draw_bars
 If set to @samp{true}, color bars are drawn in the event of a signal loss.
@@ -372,29 +311,6 @@ Sets maximum input buffer size in bytes. If the buffering reaches this value,
 incoming frames will be dropped.
 Defaults to @samp{1073741824}.
 
-@item audio_depth
-Sets the audio sample bit depth. Must be @samp{16} or @samp{32}.
-Defaults to @samp{16}.
-
-@item decklink_copyts
-If set to @option{true}, timestamps are forwarded as they are without removing
-the initial offset.
-Defaults to @option{false}.
-
-@item timestamp_align
-Capture start time alignment in seconds. If set to nonzero, input frames are
-dropped till the system timestamp aligns with configured value.
-Alignment difference of up to one frame duration is tolerated.
-This is useful for maintaining input synchronization across N different
-hardware devices deployed for 'N-way' redundancy. The system time of different
-hardware devices should be synchronized with protocols such as NTP or PTP,
-before using this option.
-Note that this method is not foolproof. In some border cases input
-synchronization may not happen due to thread scheduling jitters in the OS.
-Either sync could go wrong by 1 frame or in a rarer case
-@option{timestamp_align} seconds.
-Defaults to @samp{0}.
-
 @end table
 
 @subsection Examples
@@ -433,6 +349,116 @@ ffmpeg -channels 16 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder'
 
 @end itemize
 
+@section kmsgrab
+
+KMS video input device.
+
+Captures the KMS scanout framebuffer associated with a specified CRTC or plane as a
+DRM object that can be passed to other hardware functions.
+
+Requires either DRM master or CAP_SYS_ADMIN to run.
+
+If you don't understand what all of that means, you probably don't want this.  Look at
+@option{x11grab} instead.
+
+@subsection Options
+
+@table @option
+
+@item device
+DRM device to capture on.  Defaults to @option{/dev/dri/card0}.
+
+@item format
+Pixel format of the framebuffer.  Defaults to @option{bgr0}.
+
+@item format_modifier
+Format modifier to signal on output frames.  This is necessary to import correctly into
+some APIs, but can't be autodetected.  See the libdrm documentation for possible values.
+
+@item crtc_id
+KMS CRTC ID to define the capture source.  The first active plane on the given CRTC
+will be used.
+
+@item plane_id
+KMS plane ID to define the capture source.  Defaults to the first active plane found if
+neither @option{crtc_id} nor @option{plane_id} are specified.
+
+@item framerate
+Framerate to capture at.  This is not synchronised to any page flipping or framebuffer
+changes - it just defines the interval at which the framebuffer is sampled.  Sampling
+faster than the framebuffer update rate will generate independent frames with the same
+content.  Defaults to @code{30}.
+
+@end table
+
+@subsection Examples
+
+@itemize
+
+@item
+Capture from the first active plane, download the result to normal frames and encode.
+This will only work if the framebuffer is both linear and mappable - if not, the result
+may be scrambled or fail to download.
+@example
+ffmpeg -f kmsgrab -i - -vf 'hwdownload,format=bgr0' output.mp4
+@end example
+
+@item
+Capture from CRTC ID 42 at 60fps, map the result to VAAPI, convert to NV12 and encode as H.264.
+@example
+ffmpeg -crtc_id 42 -framerate 60 -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,scale_vaapi=w=1920:h=1080:format=nv12' -c:v h264_vaapi output.mp4
+@end example
+
+@end itemize
+
+@section libndi_newtek
+
+The libndi_newtek input device provides capture capabilities for using NDI (Network
+Device Interface, standard created by NewTek).
+
+Input filename is a NDI source name that could be found by sending -find_sources 1
+to command line - it has no specific syntax but human-readable formatted.
+
+To enable this input device, you need the NDI SDK and you
+need to configure with the appropriate @code{--extra-cflags}
+and @code{--extra-ldflags}.
+
+@subsection Options
+
+@table @option
+
+@item find_sources
+If set to @option{true}, print a list of found/available NDI sources and exit.
+Defaults to @option{false}.
+
+@item wait_sources
+Override time to wait until the number of online sources have changed.
+Defaults to @option{0.5}.
+
+@item allow_video_fields
+When this flag is @option{false}, all video that you receive will be progressive.
+Defaults to @option{true}.
+
+@end table
+
+@subsection Examples
+
+@itemize
+
+@item
+List input devices:
+@example
+ffmpeg -f libndi_newtek -find_sources 1 -i dummy
+@end example
+
+@item
+Restream to NDI:
+@example
+ffmpeg -f libndi_newtek -i "DEV-5.INTERNAL.M1STEREO.TV (NDI_SOURCE_NAME_1)" -f libndi_newtek -y NDI_SOURCE_NAME_2
+@end example
+
+@end itemize
+
 @section dshow
 
 Windows DirectShow input device.
@@ -797,7 +823,7 @@ ffplay -f iec61883 -i auto
 Grab and record the input of a FireWire DV/HDV device,
 using a packet buffer of 100000 packets if the source is HDV.
 @example
-ffmpeg -f iec61883 -i auto -dvbuffer 100000 out.mpg
+ffmpeg -f iec61883 -i auto -hdvbuffer 100000 out.mpg
 @end example
 
 @end itemize
@@ -860,76 +886,6 @@ Set the number of channels. Default is 2.
 
 @end table
 
-@section kmsgrab
-
-KMS video input device.
-
-Captures the KMS scanout framebuffer associated with a specified CRTC or plane as a
-DRM object that can be passed to other hardware functions.
-
-Requires either DRM master or CAP_SYS_ADMIN to run.
-
-If you don't understand what all of that means, you probably don't want this.  Look at
-@option{x11grab} instead.
-
-@subsection Options
-
-@table @option
-
-@item device
-DRM device to capture on.  Defaults to @option{/dev/dri/card0}.
-
-@item format
-Pixel format of the framebuffer.  Defaults to @option{bgr0}.
-
-@item format_modifier
-Format modifier to signal on output frames.  This is necessary to import correctly into
-some APIs, but can't be autodetected.  See the libdrm documentation for possible values.
-
-@item crtc_id
-KMS CRTC ID to define the capture source.  The first active plane on the given CRTC
-will be used.
-
-@item plane_id
-KMS plane ID to define the capture source.  Defaults to the first active plane found if
-neither @option{crtc_id} nor @option{plane_id} are specified.
-
-@item framerate
-Framerate to capture at.  This is not synchronised to any page flipping or framebuffer
-changes - it just defines the interval at which the framebuffer is sampled.  Sampling
-faster than the framebuffer update rate will generate independent frames with the same
-content.  Defaults to @code{30}.
-
-@end table
-
-@subsection Examples
-
-@itemize
-
-@item
-Capture from the first active plane, download the result to normal frames and encode.
-This will only work if the framebuffer is both linear and mappable - if not, the result
-may be scrambled or fail to download.
-@example
-ffmpeg -f kmsgrab -i - -vf 'hwdownload,format=bgr0' output.mp4
-@end example
-
-@item
-Capture from CRTC ID 42 at 60fps, map the result to VAAPI, convert to NV12 and encode as H.264.
-@example
-ffmpeg -crtc_id 42 -framerate 60 -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,scale_vaapi=w=1920:h=1080:format=nv12' -c:v h264_vaapi output.mp4
-@end example
-
-@item
-To capture only part of a plane the output can be cropped - this can be used to capture
-a single window, as long as it has a known absolute position and size.  For example, to
-capture and encode the middle quarter of a 1920x1080 plane:
-@example
-ffmpeg -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,crop=960:540:480:270,scale_vaapi=960:540:nv12' -c:v h264_vaapi output.mp4
-@end example
-
-@end itemize
-
 @section lavfi
 
 Libavfilter input virtual device.
@@ -1068,21 +1024,6 @@ IIDC1394 input device, based on libdc1394 and libraw1394.
 
 Requires the configure option @code{--enable-libdc1394}.
 
-@subsection Options
-@table @option
-
-@item framerate
-Set the frame rate. Default is @code{ntsc}, corresponding to a frame
-rate of @code{30000/1001}.
-
-@item pixel_format
-Select the pixel format. Default is @code{uyvy422}.
-
-@item video_size
-Set the video size given as a string such as @code{640x480} or @code{hd720}.
-Default is @code{qvga}.
-@end table
-
 @section openal
 
 The OpenAL input device provides audio capture on all systems with a
@@ -1201,6 +1142,7 @@ Set the number of channels. Default is 2.
 
 @end table
 
+
 @section pulse
 
 PulseAudio input device.

doc/issue_tracker.txt

@@ -193,6 +193,9 @@ ffplay
 ffprobe
     issues in or related to ffprobe.c
 
+ffserver
+    issues in or related to ffserver.c
+
 postproc
     issues in libpostproc/*
 

doc/libav-merge.txt

@@ -94,17 +94,18 @@ Stuff that didn't reach the codebase:
   - a853388d2 hevc: change the stride of the MC buffer to be in bytes instead of elements
   - 0cef06df0 checkasm: add HEVC MC tests
   - e7078e842 hevcdsp: add x86 SIMD for MC
-  - 7993ec19a hevc: Add hevc_get_pixel_4/8/12/16/24/32/48/64
+- VAAPI VP8 decode hwaccel (currently under review: http://ffmpeg.org/pipermail/ffmpeg-devel/2017-February/thread.html#207348)
+- Removal of the custom atomic API (5cc0057f49, see http://ffmpeg.org/pipermail/ffmpeg-devel/2017-March/209003.html)
+- new bitstream reader (see http://ffmpeg.org/pipermail/ffmpeg-devel/2017-April/209609.html)
+- use of the bsf instead of our parser for vp9 superframes (see fa1749dd34)
 - use av_cpu_max_align() instead of hardcoding alignment requirements (see https://ffmpeg.org/pipermail/ffmpeg-devel/2017-September/215834.html)
   - f44ec22e0 lavc: use av_cpu_max_align() instead of hardcoding alignment requirements
   - 4de220d2e frame: allow align=0 (meaning automatic) for av_frame_get_buffer()
-- Support recovery from an already present HLS playlist (see 16cb06bb30)
-- Remove all output devices (see 8e7e042d41, 8d3db95f20, 6ce13070bd, d46cd24986 and https://ffmpeg.org/pipermail/ffmpeg-devel/2017-September/216904.html)
-- avcodec/libaomenc: export the Sequence Header OBU as extradata (See a024c3ce9a)
 
 Collateral damage that needs work locally:
 ------------------------------------------
 
+- Merge proresdec2.c and proresdec_lgpl.c
 - Merge proresenc_anatoliy.c and proresenc_kostya.c
 - Fix MIPS AC3 downmix
 

doc/libavcodec.texi

@@ -26,13 +26,13 @@ implementing robust and fast codecs as well as for experimentation.
 @chapter See Also
 
 @ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
 @url{ffmpeg-codecs.html,ffmpeg-codecs}, @url{ffmpeg-bitstream-filters.html,bitstream-filters},
 @url{libavutil.html,libavutil}
 @end ifhtml
 
 @ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1),
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
 ffmpeg-codecs(1), ffmpeg-bitstream-filters(1),
 libavutil(3)
 @end ifnothtml

doc/libavdevice.texi

@@ -23,13 +23,13 @@ VfW, DShow, and ALSA.
 @chapter See Also
 
 @ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
 @url{ffmpeg-devices.html,ffmpeg-devices},
 @url{libavutil.html,libavutil}, @url{libavcodec.html,libavcodec}, @url{libavformat.html,libavformat}
 @end ifhtml
 
 @ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1),
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
 ffmpeg-devices(1),
 libavutil(3), libavcodec(3), libavformat(3)
 @end ifnothtml

doc/libavfilter.texi

@@ -21,14 +21,14 @@ framework containing several filters, sources and sinks.
 @chapter See Also
 
 @ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
 @url{ffmpeg-filters.html,ffmpeg-filters},
 @url{libavutil.html,libavutil}, @url{libswscale.html,libswscale}, @url{libswresample.html,libswresample},
 @url{libavcodec.html,libavcodec}, @url{libavformat.html,libavformat}, @url{libavdevice.html,libavdevice}
 @end ifhtml
 
 @ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1),
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
 ffmpeg-filters(1),
 libavutil(3), libswscale(3), libswresample(3), libavcodec(3), libavformat(3), libavdevice(3)
 @end ifnothtml

doc/libavformat.texi

@@ -26,13 +26,13 @@ resource.
 @chapter See Also
 
 @ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
 @url{ffmpeg-formats.html,ffmpeg-formats}, @url{ffmpeg-protocols.html,ffmpeg-protocols},
 @url{libavutil.html,libavutil}, @url{libavcodec.html,libavcodec}
 @end ifhtml
 
 @ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1),
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
 ffmpeg-formats(1), ffmpeg-protocols(1),
 libavutil(3), libavcodec(3)
 @end ifnothtml

doc/libavutil.texi

@@ -42,12 +42,12 @@ It should avoid useless features that almost no one needs.
 @chapter See Also
 
 @ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
 @url{ffmpeg-utils.html,ffmpeg-utils}
 @end ifhtml
 
 @ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1),
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
 ffmpeg-utils(1)
 @end ifnothtml
 

doc/libswresample.texi

@@ -48,13 +48,13 @@ enabled through dedicated options.
 @chapter See Also
 
 @ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
 @url{ffmpeg-resampler.html,ffmpeg-resampler},
 @url{libavutil.html,libavutil}
 @end ifhtml
 
 @ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1),
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
 ffmpeg-resampler(1),
 libavutil(3)
 @end ifnothtml

doc/libswscale.texi

@@ -41,13 +41,13 @@ colorspaces differ.
 @chapter See Also
 
 @ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
 @url{ffmpeg-scaler.html,ffmpeg-scaler},
 @url{libavutil.html,libavutil}
 @end ifhtml
 
 @ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1),
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
 ffmpeg-scaler(1),
 libavutil(3)
 @end ifnothtml

doc/mailing-list-faq.texi

@@ -27,7 +27,8 @@ for examples.
 @item
 @url{https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-user/, ffmpeg-user}:
 For questions involving unscripted usage or compilation of the FFmpeg
-command-line tools (@command{ffmpeg}, @command{ffprobe}, @command{ffplay}).
+command-line tools (@command{ffmpeg}, @command{ffprobe}, @command{ffplay},
+@command{ffserver}).
 
 @item
 @url{https://lists.ffmpeg.org/mailman/listinfo/libav-user/, libav-user}:
@@ -47,8 +48,7 @@ We cannot provide help for scripts and/or third-party tools.
 @anchor{How do I ask a question or send a message to a mailing list?}
 @section How do I ask a question or send a message to a mailing list?
 
-First you must @ref{How do I subscribe?, subscribe}. Then all you have to do is
-send an email:
+All you have to do is send an email:
 
 @itemize
 @item
@@ -58,14 +58,49 @@ ffmpeg-user mailing list.
 @item
 Email @email{libav-user@@ffmpeg.org} to send a message to the
 libav-user mailing list.
-
-@item
-Email @email{ffmpeg-devel@@ffmpeg.org} to send a message to the
-ffmpeg-devel mailing list.
 @end itemize
 
+If you are not subscribed to the mailing list then your question must be
+manually approved. Approval may take several days, but the wait is
+usually less. If you want the message to be sent with no delay then you
+must subscribe first. See @ref{How do I subscribe?}
+
+Please do not send a message, subscribe, and re-send the message: this
+results in duplicates, causes more work for the admins, and may lower
+your chance at getting an answer. However, you may do so if you first
+@ref{How do I delete my message in the moderation queue?, delete your original message from the moderation queue}.
+
 @chapter Subscribing / Unsubscribing
 
+@section What does subscribing do?
+
+Subscribing allows two things:
+
+@itemize
+@item
+Your messages will show up in the mailing list without waiting in the
+moderation queue and needing to be manually approved by a mailing list
+admin.
+
+@item
+You will receive all messages to the mailing list including replies to
+your messages. Non-subscribed users do not receive any messages.
+@end itemize
+
+@section Do I need to subscribe?
+
+No. You can still send a message to the mailing list without
+subscribing. See @ref{How do I ask a question or send a message to a mailing list?}
+
+However, your message will need to be manually approved by a mailing
+list admin, and you will not receive any mailing list messages or
+replies.
+
+You can ask to be CCd in your message, but replying users will
+sometimes forget to do so.
+
+You may also view and reply to messages via the @ref{Where are the archives?, archives}.
+
 @anchor{How do I subscribe?}
 @section How do I subscribe?
 
@@ -90,9 +125,6 @@ The process is the same for the other mailing lists.
 Please avoid asking a mailing list admin to unsubscribe you unless you
 are absolutely unable to do so by yourself. See @ref{Who do I contact if I have a problem with the mailing list?}
 
-Note that it is possible to temporarily halt message delivery (vacation mode).
-See @ref{How do I disable mail delivery without unsubscribing?}
-
 @chapter Moderation Queue
 @anchor{Why is my message awaiting moderator approval?}
 @section Why is my message awaiting moderator approval?
@@ -103,6 +135,8 @@ must be manually approved by a mailing list admin:
 These are:
 
 @itemize
+@item
+Messages from users who are @strong{not} subscribed.
 
 @item
 Messages that exceed the @ref{What is the message size limit?, message size limit}.
@@ -115,13 +149,13 @@ or is abusive towards others).
 
 @section How long does it take for my message in the moderation queue to be approved?
 
-The queue is not checked on a regular basis. You can ask on the
-@t{#ffmpeg-devel} IRC channel on Freenode for someone to approve your message.
+The queue is usually checked once or twice a day, but on occasion
+several days may pass before someone checks the queue.
 
 @anchor{How do I delete my message in the moderation queue?}
 @section How do I delete my message in the moderation queue?
 
-You should have received an email with the subject @emph{Your message to <mailing list name> awaits moderator approval}.
+You should have received an email with the subject @emph{Your message to ffmpeg-user awaits moderator approval}.
 A link is in the message that will allow you to delete your message
 unless a mailing list admin already approved or rejected it.
 
@@ -142,9 +176,6 @@ Click the email link at the top of the message just under the subject
 title. The link will provide the proper headers to keep the message
 within the thread.
 
-Note that you must be subscribed to send a message to the ffmpeg-user or
-libav-user mailing lists.
-
 @section How do I search the archives?
 
 Perform a site search using your favorite search engine. Example:
@@ -157,12 +188,11 @@ Perform a site search using your favorite search engine. Example:
 
 You can ask for help in the official @t{#ffmpeg} IRC channel on Freenode.
 
-Some users prefer the third-party @url{http://www.ffmpeg-archive.org/, Nabble}
-interface which presents the mailing lists in a typical forum layout.
+Some users prefer the third-party Nabble interface which presents the
+mailing lists in a typical forum layout.
 
-There are also numerous third-party help sites such as
-@url{https://superuser.com/tags/ffmpeg, Super User} and
-@url{https://www.reddit.com/r/ffmpeg/, r/ffmpeg on reddit}.
+There are also numerous third-party help sites such as Super User and
+r/ffmpeg on reddit.
 
 @anchor{What is top-posting?}
 @section What is top-posting?
@@ -174,15 +204,16 @@ Instead, use trimmed interleaved/inline replies (@url{https://lists.ffmpeg.org/p
 @anchor{What is the message size limit?}
 @section What is the message size limit?
 
-The message size limit is 1000 kilobytes. Please provide links to larger files
-instead of attaching them.
+The message size limit is 500 kilobytes for the user lists and 1000
+kilobytes for ffmpeg-devel. Please provide links to larger files instead
+of attaching them.
 
 @section Where can I upload sample files?
 
 Anywhere that is not too annoying for us to use.
 
 Google Drive and Dropbox are acceptable if you need a file host, and
-@url{https://0x0.st/, 0x0.st} is good for files under 256 MiB.
+0x0.st is good for files under 256 MiB.
 
 Small, short samples are preferred if possible.
 
@@ -229,54 +260,6 @@ or headers.
 
 You can then filter the mailing list messages to their own folder.
 
-@anchor{How do I disable mail delivery without unsubscribing?}
-@section How do I disable mail delivery without unsubscribing?
-
-Sometimes you may want to temporarily stop receiving all mailing list
-messages. This "vacation mode" is simple to do:
-
-@enumerate
-@item
-Go to the @url{https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-user/, ffmpeg-user mailing list info page}
-
-@item
-Enter your email address in the box at very bottom of the page and click the
-@emph{Unsubscribe or edit options} box.
-
-@item
-Enter your password and click the @emph{Log in} button.
-
-@item
-Look for the @emph{Mail delivery} option. Here you can disable/enable mail
-delivery. If you check @emph{Set globally} it will apply your choice to all
-other FFmpeg mailing lists you are subscribed to.
-@end enumerate
-
-Alternatively, from your subscribed address, send a message to @email{ffmpeg-user-request@@ffmpeg.org}
-with the subject @emph{set delivery off}. To re-enable mail delivery send a
-message to @email{ffmpeg-user-request@@ffmpeg.org} with the subject
-@emph{set delivery on}.
-
-@anchor{Why is the mailing list munging my address?}
-@section Why is the mailing list munging my address?
-
-This is due to subscribers that use an email service with a DMARC reject policy
-which adds difficulties to mailing list operators.
-
-The mailing list must re-write (munge) the @emph{From:} header for such users;
-otherwise their email service will reject and bounce the message resulting in
-automatic unsubscribing from the mailing list.
-
-When sending a message these users will see @emph{via <mailing list name>}
-added to their name and the @emph{From:} address munged to the address of
-the particular mailing list.
-
-If you want to avoid this then please use a different email service.
-
-Note that ffmpeg-devel does not apply any munging as it causes issues with
-patch authorship. As a result users with an email service with a DMARC reject
-policy may be automatically unsubscribed due to rejected and bounced messages.
-
 @chapter Rules and Etiquette
 
 @section What are the rules and the proper etiquette?
@@ -375,15 +358,6 @@ form a multi-part message is recommended by email standards.
 Check your spam folder.
 @end itemize
 
-@anchor{Why do I keep getting unsubscribed from ffmpeg-devel?}
-@section Why do I keep getting unsubscribed from ffmpeg-devel?
-
-Users with an email service that has a DMARC reject or quarantine policy may be
-automatically unsubscribed from the ffmpeg-devel mailing list due to the mailing
-list messages being continuously rejected and bounced back.
-
-Consider using a different email service.
-
 @anchor{Who do I contact if I have a problem with the mailing list?}
 @section Who do I contact if I have a problem with the mailing list?
 

doc/metadata.texi

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

doc/muxers.texi

@@ -94,23 +94,21 @@ compatibility with software that only supports a single audio stream in AVI
 @anchor{chromaprint}
 @section chromaprint
 
-Chromaprint fingerprinter.
+Chromaprint fingerprinter
 
-This muxer feeds audio data to the Chromaprint library,
-which generates a fingerprint for the provided audio data. See @url{https://acoustid.org/chromaprint}
-
-It takes a single signed native-endian 16-bit raw audio stream of at most 2 channels.
+This muxer feeds audio data to the Chromaprint library, which generates
+a fingerprint for the provided audio data. It takes a single signed
+native-endian 16-bit raw audio stream.
 
 @subsection Options
 
 @table @option
 @item silence_threshold
-Threshold for detecting silence, ranges from -1 to 32767. -1 disables silence detection and
-is required for use with the AcoustID service. Default is -1.
+Threshold for detecting silence, ranges from 0 to 32767. -1 for default
+(required for use with the AcoustID service).
 
 @item algorithm
-Version of algorithm to fingerprint with. Range is 0 to 4. Version 2 requires that silence
-detection be enabled. Default is 1.
+Algorithm index to fingerprint with.
 
 @item fp_format
 Format to output the fingerprint as. Accepts the following options:
@@ -122,7 +120,7 @@ Binary raw fingerprint
 Binary compressed fingerprint
 
 @item base64
-Base64 compressed fingerprint @emph{(default)}
+Base64 compressed fingerprint
 
 @end table
 
@@ -216,113 +214,46 @@ It creates a MPD manifest file and segment files for each stream.
 The segment filename might contain pre-defined identifiers used with SegmentTemplate
 as defined in section 5.3.9.4.4 of the standard. Available identifiers are "$RepresentationID$",
 "$Number$", "$Bandwidth$" and "$Time$".
-In addition to the standard identifiers, an ffmpeg-specific "$ext$" identifier is also supported.
-When specified ffmpeg will replace $ext$ in the file name with muxing format's extensions such as mp4, webm etc.,
 
 @example
-ffmpeg -re -i <input> -map 0 -map 0 -c:a libfdk_aac -c:v libx264 \
--b:v:0 800k -b:v:1 300k -s:v:1 320x170 -profile:v:1 baseline \
--profile:v:0 main -bf 1 -keyint_min 120 -g 120 -sc_threshold 0 \
--b_strategy 0 -ar:a:1 22050 -use_timeline 1 -use_template 1 \
--window_size 5 -adaptation_sets "id=0,streams=v id=1,streams=a" \
+ffmpeg -re -i <input> -map 0 -map 0 -c:a libfdk_aac -c:v libx264
+-b:v:0 800k -b:v:1 300k -s:v:1 320x170 -profile:v:1 baseline
+-profile:v:0 main -bf 1 -keyint_min 120 -g 120 -sc_threshold 0
+-b_strategy 0 -ar:a:1 22050 -use_timeline 1 -use_template 1
+-window_size 5 -adaptation_sets "id=0,streams=v id=1,streams=a"
 -f dash /path/to/out.mpd
 @end example
 
 @table @option
-@item min_seg_duration @var{microseconds}
-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
-@var{use_timeline} is disabled and as minimum segment duration for all the other
-use cases.
-@item window_size @var{size}
+@item -min_seg_duration @var{microseconds}
+Set the segment length in microseconds.
+@item -window_size @var{size}
 Set the maximum number of segments kept in the manifest.
-@item extra_window_size @var{size}
+@item -extra_window_size @var{size}
 Set the maximum number of segments kept outside of the manifest before removing from disk.
-@item remove_at_exit @var{remove}
+@item -remove_at_exit @var{remove}
 Enable (1) or disable (0) removal of all segments when finished.
-@item use_template @var{template}
+@item -use_template @var{template}
 Enable (1) or disable (0) use of SegmentTemplate instead of SegmentList.
-@item use_timeline @var{timeline}
+@item -use_timeline @var{timeline}
 Enable (1) or disable (0) use of SegmentTimeline in SegmentTemplate.
-@item single_file @var{single_file}
+@item -single_file @var{single_file}
 Enable (1) or disable (0) storing all segments in one file, accessed using byte ranges.
-@item single_file_name @var{file_name}
-DASH-templated name to be used for baseURL. Implies @var{single_file} set to "1". In the template, "$ext$" is replaced with the file name extension specific for the segment format.
-@item init_seg_name @var{init_name}
-DASH-templated name to used for the initialization segment. Default is "init-stream$RepresentationID$.$ext$". "$ext$" is replaced with the file name extension specific for the segment format.
-@item media_seg_name @var{segment_name}
-DASH-templated name to used for the media segments. Default is "chunk-stream$RepresentationID$-$Number%05d$.$ext$". "$ext$" is replaced with the file name extension specific for the segment format.
-@item utc_timing_url @var{utc_url}
+@item -single_file_name @var{file_name}
+DASH-templated name to be used for baseURL. Implies @var{single_file} set to "1".
+@item -init_seg_name @var{init_name}
+DASH-templated name to used for the initialization segment. Default is "init-stream$RepresentationID$.m4s"
+@item -media_seg_name @var{segment_name}
+DASH-templated name to used for the media segments. Default is "chunk-stream$RepresentationID$-$Number%05d$.m4s"
+@item -utc_timing_url @var{utc_url}
 URL of the page that will return the UTC timestamp in ISO format. Example: "https://time.akamai.com/?iso"
-@item method @var{method}
-Use the given HTTP method to create output files. Generally set to PUT or POST.
-@item http_user_agent @var{user_agent}
-Override User-Agent field in HTTP header. Applicable only for HTTP output.
-@item http_persistent @var{http_persistent}
-Use persistent HTTP connections. Applicable only for HTTP output.
-@item hls_playlist @var{hls_playlist}
-Generate HLS playlist files as well. The master playlist is generated with the filename master.m3u8.
-One media playlist file is generated for each stream with filenames media_0.m3u8, media_1.m3u8, etc.
-@item streaming @var{streaming}
-Enable (1) or disable (0) chunk streaming mode of output. In chunk streaming
-mode, each frame will be a moof fragment which forms a chunk.
-@item adaptation_sets @var{adaptation_sets}
+@item -adaptation_sets @var{adaptation_sets}
 Assign streams to AdaptationSets. Syntax is "id=x,streams=a,b,c id=y,streams=d,e" with x and y being the IDs
 of the adaptation sets and a,b,c,d and e are the indices of the mapped streams.
 
 To map all video (or audio) streams to an AdaptationSet, "v" (or "a") can be used as stream identifier instead of IDs.
 
 When no assignment is defined, this defaults to an AdaptationSet for each stream.
-@item timeout @var{timeout}
-Set timeout for socket I/O operations. Applicable only for HTTP output.
-@item index_correction @var{index_correction}
-Enable (1) or Disable (0) segment index correction logic. Applicable only when
-@var{use_template} is enabled and @var{use_timeline} is disabled.
-
-When enabled, the logic monitors the flow of segment indexes. If a streams's
-segment index value is not at the expected real time position, then the logic
-corrects that index value.
-
-Typically this logic is needed in live streaming use cases. The network bandwidth
-fluctuations are common during long run streaming. Each fluctuation can cause
-the segment indexes fall behind the expected real time position.
-@item format_options @var{options_list}
-Set container format (mp4/webm) options using a @code{:} separated list of
-key=value parameters. Values containing @code{:} special characters must be
-escaped.
-
-@item global_sidx @var{global_sidx}
-Write global SIDX atom. Applicable only for single file, mp4 output, non-streaming mode.
-
-@item dash_segment_type @var{dash_segment_type}
-Possible values:
-@table @option
-@item auto
-If this flag is set, the dash segment files format will be selected based on the stream codec. This is the default mode.
-
-@item mp4
-If this flag is set, the dash segment files will be in in ISOBMFF format.
-
-@item webm
-If this flag is set, the dash segment files will be in in WebM format.
-@end table
-
-@item ignore_io_errors @var{ignore_io_errors}
-Ignore IO errors during open and write. Useful for long-duration runs with network output.
-
-@item lhls @var{lhls}
-Enable Low-latency HLS(LHLS). Adds #EXT-X-PREFETCH tag with current segment's URI.
-Apple doesn't have an official spec for LHLS. Meanwhile hls.js player folks are
-trying to standardize a open LHLS spec. The draft spec is available in https://github.com/video-dev/hlsjs-rfcs/blob/lhls-spec/proposals/0001-lhls.md
-This option will also try to comply with the above open spec, till Apple's spec officially supports it.
-Applicable only when @var{streaming} and @var{hls_playlist} options are enabled.
-This is an experimental feature.
-
-@item master_m3u8_publish_rate @var{master_m3u8_publish_rate}
-Publish master playlist repeatedly every after specified number of segment intervals.
-
 @end table
 
 @anchor{framecrc}
@@ -537,12 +468,9 @@ By default, the muxer creates a file for each segment produced. These files
 have the same name as the playlist, followed by a sequential number and a
 .ts extension.
 
-Make sure to require a closed GOP when encoding and to set the GOP
-size to fit your segment time constraint.
-
 For example, to convert an input file with @command{ffmpeg}:
 @example
-ffmpeg -i in.mkv -c:v h264 -flags +cgop -g 30 -hls_time 1 out.m3u8
+ffmpeg -i in.nut out.m3u8
 @end example
 This example will produce the playlist, @file{out.m3u8}, and segment files:
 @file{out0.ts}, @file{out1.ts}, @file{out2.ts}, etc.
@@ -570,12 +498,6 @@ Segment will be cut on the next key frame after this time has passed.
 Set the maximum number of playlist entries. If set to 0 the list file
 will contain all the segments. Default value is 5.
 
-@item hls_delete_threshold @var{size}
-Set the number of unreferenced segments to keep on disk before @code{hls_flags delete_segments}
-deletes them. Increase this to allow continue clients to download segments which
-were recently referenced in the playlist. Default value is 1, meaning segments older than
-@code{hls_list_size+1} will be deleted.
-
 @item hls_ts_options @var{options_list}
 Set output format options using a :-separated list of key=value
 parameters. Values containing @code{:} special characters must be
@@ -643,70 +565,38 @@ This example will produce the playlist, @file{out.m3u8}, and segment files:
 but only the file name part without any path info will be contained in the m3u8 segment list.
 Should a relative path be specified, the path of the created segment
 files will be relative to the current working directory.
-When strftime_mkdir is set, the whole expanded value of @var{filename} will be written into the m3u8 segment list.
+When use_localtime_mkdir is set, the whole expanded value of @var{filename} will be written into the m3u8 segment list.
 
-When @code{var_stream_map} is set with two or more variant streams, the
-@var{filename} pattern must contain the string "%v", this string specifies
-the position of variant stream index in the generated segment file names.
-@example
-ffmpeg -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \
-  -map 0:v -map 0:a -map 0:v -map 0:a -f hls -var_stream_map "v:0,a:0 v:1,a:1" \
-  -hls_segment_filename 'file_%v_%03d.ts' out_%v.m3u8
-@end example
-This example will produce the playlists segment file sets:
-@file{file_0_000.ts}, @file{file_0_001.ts}, @file{file_0_002.ts}, etc. and
-@file{file_1_000.ts}, @file{file_1_001.ts}, @file{file_1_002.ts}, etc.
-
-The string "%v" may be present in the filename or in the last directory name
-containing the file, but only in one of them. (Additionally, %v may appear multiple times in the last
-sub-directory or filename.) If the string %v is present in the directory name, then
-sub-directories are created after expanding the directory name pattern. This
-enables creation of segments corresponding to different variant streams in
-subdirectories.
-@example
-ffmpeg -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \
-  -map 0:v -map 0:a -map 0:v -map 0:a -f hls -var_stream_map "v:0,a:0 v:1,a:1" \
-  -hls_segment_filename 'vs%v/file_%03d.ts' vs%v/out.m3u8
-@end example
-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
 hls_flag and %%d will be the specifier.
 @example
-ffmpeg -i in.nut -strftime 1 -hls_segment_filename 'file-%Y%m%d-%s.ts' out.m3u8
+ffmpeg -i in.nut -use_localtime 1 -hls_segment_filename 'file-%Y%m%d-%s.ts' out.m3u8
 @end example
 This example will produce the playlist, @file{out.m3u8}, and segment files:
 @file{file-20160215-1455569023.ts}, @file{file-20160215-1455569024.ts}, etc.
 Note: On some systems/environments, the @code{%s} specifier is not available. See
   @code{strftime()} documentation.
 @example
-ffmpeg -i in.nut -strftime 1 -hls_flags second_level_segment_index -hls_segment_filename 'file-%Y%m%d-%%04d.ts' out.m3u8
+ffmpeg -i in.nut -use_localtime 1 -hls_flags second_level_segment_index -hls_segment_filename 'file-%Y%m%d-%%04d.ts' out.m3u8
 @end example
 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
+Used together with -use_localtime, it will create all subdirectories which
 is expanded in @var{filename}.
 @example
-ffmpeg -i in.nut -strftime 1 -strftime_mkdir 1 -hls_segment_filename '%Y%m%d/file-%Y%m%d-%s.ts' out.m3u8
+ffmpeg -i in.nut -use_localtime 1 -use_localtime_mkdir 1 -hls_segment_filename '%Y%m%d/file-%Y%m%d-%s.ts' out.m3u8
 @end example
 This example will create a directory 201560215 (if it does not exist), and then
 produce the playlist, @file{out.m3u8}, and segment files:
 @file{20160215/file-20160215-1455569023.ts}, @file{20160215/file-20160215-1455569024.ts}, etc.
 
 @example
-ffmpeg -i in.nut -strftime 1 -strftime_mkdir 1 -hls_segment_filename '%Y/%m/%d/file-%Y%m%d-%s.ts' out.m3u8
+ffmpeg -i in.nut -use_localtime 1 -use_localtime_mkdir 1 -hls_segment_filename '%Y/%m/%d/file-%Y%m%d-%s.ts' out.m3u8
 @end example
 This example will create a directory hierarchy 2016/02/15 (if any of them do not exist), and then
 produce the playlist, @file{out.m3u8}, and segment files:
@@ -791,26 +681,17 @@ Possible values:
 
 @table @samp
 @item mpegts
-Output segment files in MPEG-2 Transport Stream format. This is
-compatible with all HLS versions.
+If this flag is set, the hls segment files will format to mpegts.
+the mpegts files is used in all hls versions.
 
 @item fmp4
-Output segment files in fragmented MP4 format, similar to MPEG-DASH.
-fmp4 files may be used in HLS version 7 and above.
+If this flag is set, the hls segment files will format to fragment mp4 looks like dash.
+the fmp4 files is used in hls after version 7.
 
 @end table
 
 @item hls_fmp4_init_filename @var{filename}
-Set filename to the fragment files header file, default filename is @file{init.mp4}.
-
-When @code{var_stream_map} is set with two or more variant streams, the
-@var{filename} pattern must contain the string "%v", this string specifies
-the position of variant stream index in the generated init file names.
-The string "%v" may be present in the filename or in the last directory name
-containing the file. If the string is present in the directory name, then
-sub-directories are created after expanding the directory name pattern. This
-enables creation of init files corresponding to different variant streams in
-subdirectories.
+set filename to the fragment files header file, default filename is @file{init.mp4}.
 
 @item hls_flags @var{flags}
 Possible values:
@@ -851,14 +732,6 @@ The file specified by @code{hls_key_info_file} will be checked periodically and
 detect updates to the encryption info. Be sure to replace this file atomically,
 including the file containing the AES encryption key.
 
-@item independent_segments
-Add the @code{#EXT-X-INDEPENDENT-SEGMENTS} to playlists that has video segments
-and when all the segments of that playlist are guaranteed to start with a Key frame.
-
-@item iframes_only
-Add the @code{#EXT-X-I-FRAMES-ONLY} to playlists that has video segments
-and can play only I-frames in the @code{#EXT-X-BYTERANGE} mode.
-
 @item split_by_time
 Allow segments to start on frames other than keyframes. This improves
 behavior on some players when the time between keyframes is inconsistent,
@@ -870,24 +743,24 @@ Generate @code{EXT-X-PROGRAM-DATE-TIME} tags.
 
 @item second_level_segment_index
 Makes it possible to use segment indexes as %%d in hls_segment_filename expression
-besides date/time values when strftime is on.
+besides date/time values when use_localtime is on.
 To get fixed width numbers with trailing zeroes, %%0xd format is available where x is the required width.
 
 @item second_level_segment_size
 Makes it possible to use segment sizes (counted in bytes) as %%s in hls_segment_filename
-expression besides date/time values when strftime is on.
+expression besides date/time values when use_localtime is on.
 To get fixed width numbers with trailing zeroes, %%0xs format is available where x is the required width.
 
 @item second_level_segment_duration
 Makes it possible to use segment duration (calculated  in microseconds) as %%t in hls_segment_filename
-expression besides date/time values when strftime is on.
+expression besides date/time values when use_localtime is on.
 To get fixed width numbers with trailing zeroes, %%0xt format is available where x is the required width.
 
 @example
 ffmpeg -i sample.mpeg \
    -f hls -hls_time 3 -hls_list_size 5 \
    -hls_flags second_level_segment_index+second_level_segment_size+second_level_segment_duration \
-   -strftime 1 -strftime_mkdir 1 -hls_segment_filename "segment_%Y%m%d%H%M%S_%%04d_%%08s_%%013t.ts" stream.m3u8
+   -use_localtime 1 -use_localtime_mkdir 1 -hls_segment_filename "segment_%Y%m%d%H%M%S_%%04d_%%08s_%%013t.ts" stream.m3u8
 @end example
 This will produce segments like this:
 @file{segment_20170102194334_0003_00122200_0000003000000.ts}, @file{segment_20170102194334_0004_00120072_0000003000000.ts} etc.
@@ -895,11 +768,7 @@ This will produce segments like this:
 @item temp_file
 Write segment data to filename.tmp and rename to filename only once the segment is complete. A webserver
 serving up segments can be configured to reject requests to *.tmp to prevent access to in-progress segments
-before they have been added to the m3u8 playlist. This flag also affects how m3u8 playlist files are created.
-If this flag is set, all playlist files will written into temporary file and renamed after they are complete, similarly as segments are handled.
-But playlists with @code{file} protocol and with type (@code{hls_playlist_type}) other than @code{vod}
-are always written into temporary file regardles of this flag. Master playlist files (@code{master_pl_name}), if any, with @code{file} protocol,
-are always written into temporary file regardles of this flag if @code{master_pl_publish_rate} value is other than zero.
+before they have been added to the m3u8 playlist.
 
 @end table
 
@@ -925,179 +794,6 @@ files.
 @item http_user_agent
 Override User-Agent field in HTTP header. Applicable only for HTTP output.
 
-@item var_stream_map
-Map string which specifies how to group the audio, video and subtitle streams
-into different variant streams. The variant stream groups are separated
-by space.
-Expected string format is like this "a:0,v:0 a:1,v:1 ....". Here a:, v:, s: are
-the keys to specify audio, video and subtitle streams respectively.
-Allowed values are 0 to 9 (limited just based on practical usage).
-
-When there are two or more variant streams, the output filename pattern must
-contain the string "%v", this string specifies the position of variant stream
-index in the output media playlist filenames. The string "%v" may be present in
-the filename or in the last directory name containing the file. If the string is
-present in the directory name, then sub-directories are created after expanding
-the directory name pattern. This enables creation of variant streams in
-subdirectories.
-
-@example
-ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \
-  -map 0:v -map 0:a -map 0:v -map 0:a -f hls -var_stream_map "v:0,a:0 v:1,a:1" \
-  http://example.com/live/out_%v.m3u8
-@end example
-This example creates two hls variant streams. The first variant stream will
-contain video stream of bitrate 1000k and audio stream of bitrate 64k and the
-second variant stream will contain video stream of bitrate 256k and audio
-stream of bitrate 32k. Here, two media playlist with file names out_0.m3u8 and
-out_1.m3u8 will be created. If you want something meaningful text instead of indexes
-in result names, you may specify names for each or some of the variants
-as in the following example.
-
-
-@example
-ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \
-  -map 0:v -map 0:a -map 0:v -map 0:a -f hls -var_stream_map "v:0,a:0,name:my_hd v:1,a:1,name:my_sd" \
-  http://example.com/live/out_%v.m3u8
-@end example
-
-This example creates two hls variant streams as in the previous one.
-But here, the two media playlist with file names out_my_hd.m3u8 and
-out_my_sd.m3u8 will be created.
-
-@example
-ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k \
-  -map 0:v -map 0:a -map 0:v -f hls -var_stream_map "v:0 a:0 v:1" \
-  http://example.com/live/out_%v.m3u8
-@end example
-This example creates three hls variant streams. The first variant stream will
-be a video only stream with video bitrate 1000k, the second variant stream will
-be an audio only stream with bitrate 64k and the third variant stream will be a
-video only stream with bitrate 256k. Here, three media playlist with file names
-out_0.m3u8, out_1.m3u8 and out_2.m3u8 will be created.
-@example
-ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \
-  -map 0:v -map 0:a -map 0:v -map 0:a -f hls -var_stream_map "v:0,a:0 v:1,a:1" \
-  http://example.com/live/vs_%v/out.m3u8
-@end example
-This example creates the variant streams in subdirectories. Here, the first
-media playlist is created at @file{http://example.com/live/vs_0/out.m3u8} and
-the second one at @file{http://example.com/live/vs_1/out.m3u8}.
-@example
-ffmpeg -re -i in.ts -b:a:0 32k -b:a:1 64k -b:v:0 1000k -b:v:1 3000k  \
-  -map 0:a -map 0:a -map 0:v -map 0:v -f hls \
-  -var_stream_map "a:0,agroup:aud_low a:1,agroup:aud_high v:0,agroup:aud_low v:1,agroup:aud_high" \
-  -master_pl_name master.m3u8 \
-  http://example.com/live/out_%v.m3u8
-@end example
-This example creates two audio only and two video only variant streams. In
-addition to the #EXT-X-STREAM-INF tag for each variant stream in the master
-playlist, #EXT-X-MEDIA tag is also added for the two audio only variant streams
-and they are mapped to the two video only variant streams with audio group names
-'aud_low' and 'aud_high'.
-
-By default, a single hls variant containing all the encoded streams is created.
-
-@example
-ffmpeg -re -i in.ts -b:a:0 32k -b:a:1 64k -b:v:0 1000k \
-  -map 0:a -map 0:a -map 0:v -f hls \
-  -var_stream_map "a:0,agroup:aud_low,default:yes a:1,agroup:aud_low v:0,agroup:aud_low" \
-  -master_pl_name master.m3u8 \
-  http://example.com/live/out_%v.m3u8
-@end example
-This example creates two audio only and one video only variant streams. In
-addition to the #EXT-X-STREAM-INF tag for each variant stream in the master
-playlist, #EXT-X-MEDIA tag is also added for the two audio only variant streams
-and they are mapped to the one video only variant streams with audio group name
-'aud_low', and the audio group have default stat is NO or YES.
-
-By default, a single hls variant containing all the encoded streams is created.
-
-@example
-ffmpeg -re -i in.ts -b:a:0 32k -b:a:1 64k -b:v:0 1000k \
-  -map 0:a -map 0:a -map 0:v -f hls \
-  -var_stream_map "a:0,agroup:aud_low,default:yes,language:ENG a:1,agroup:aud_low,language:CHN v:0,agroup:aud_low" \
-  -master_pl_name master.m3u8 \
-  http://example.com/live/out_%v.m3u8
-@end example
-This example creates two audio only and one video only variant streams. In
-addition to the #EXT-X-STREAM-INF tag for each variant stream in the master
-playlist, #EXT-X-MEDIA tag is also added for the two audio only variant streams
-and they are mapped to the one video only variant streams with audio group name
-'aud_low', and the audio group have default stat is NO or YES, and one audio
-have and language is named ENG, the other audio language is named CHN.
-
-By default, a single hls variant containing all the encoded streams is created.
-
-@item cc_stream_map
-Map string which specifies different closed captions groups and their
-attributes. The closed captions stream groups are separated by space.
-Expected string format is like this
-"ccgroup:<group name>,instreamid:<INSTREAM-ID>,language:<language code> ....".
-'ccgroup' and 'instreamid' are mandatory attributes. 'language' is an optional
-attribute.
-The closed captions groups configured using this option are mapped to different
-variant streams by providing the same 'ccgroup' name in the
-@code{var_stream_map} string. If @code{var_stream_map} is not set, then the
-first available ccgroup in @code{cc_stream_map} is mapped to the output variant
-stream. The examples for these two use cases are given below.
-
-@example
-ffmpeg -re -i in.ts -b:v 1000k -b:a 64k -a53cc 1 -f hls \
-  -cc_stream_map "ccgroup:cc,instreamid:CC1,language:en" \
-  -master_pl_name master.m3u8 \
-  http://example.com/live/out.m3u8
-@end example
-This example adds @code{#EXT-X-MEDIA} tag with @code{TYPE=CLOSED-CAPTIONS} in
-the master playlist with group name 'cc', language 'en' (english) and
-INSTREAM-ID 'CC1'. Also, it adds @code{CLOSED-CAPTIONS} attribute with group
-name 'cc' for the output variant stream.
-@example
-ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \
-  -a53cc:0 1 -a53cc:1 1\
-  -map 0:v -map 0:a -map 0:v -map 0:a -f hls \
-  -cc_stream_map "ccgroup:cc,instreamid:CC1,language:en ccgroup:cc,instreamid:CC2,language:sp" \
-  -var_stream_map "v:0,a:0,ccgroup:cc v:1,a:1,ccgroup:cc" \
-  -master_pl_name master.m3u8 \
-  http://example.com/live/out_%v.m3u8
-@end example
-This example adds two @code{#EXT-X-MEDIA} tags with @code{TYPE=CLOSED-CAPTIONS} in
-the master playlist for the INSTREAM-IDs 'CC1' and 'CC2'. Also, it adds
-@code{CLOSED-CAPTIONS} attribute with group name 'cc' for the two output variant
-streams.
-
-@item master_pl_name
-Create HLS master playlist with the given name.
-
-@example
-ffmpeg -re -i in.ts -f hls -master_pl_name master.m3u8 http://example.com/live/out.m3u8
-@end example
-This example creates HLS master playlist with name master.m3u8 and it is
-published at http://example.com/live/
-
-@item master_pl_publish_rate
-Publish master play list repeatedly every after specified number of segment intervals.
-
-@example
-ffmpeg -re -i in.ts -f hls -master_pl_name master.m3u8 \
--hls_time 2 -master_pl_publish_rate 30 http://example.com/live/out.m3u8
-@end example
-
-This example creates HLS master playlist with name master.m3u8 and keep
-publishing it repeatedly every after 30 segments i.e. every after 60s.
-
-@item http_persistent
-Use persistent HTTP connections. Applicable only for HTTP output.
-
-@item timeout
-Set timeout for socket I/O operations. Applicable only for HTTP output.
-
-@item -ignore_io_errors
-Ignore IO errors during open, write and delete. Useful for long-duration runs with network output.
-
-@item headers
-Set custom HTTP headers, can override built in default headers. Applicable only for HTTP output.
-
 @end table
 
 @anchor{ico}
@@ -1198,18 +894,9 @@ can be used:
 ffmpeg -f v4l2 -r 1 -i /dev/video0 -f image2 -strftime 1 "%Y-%m-%d_%H-%M-%S.jpg"
 @end example
 
-You can set the file name with current frame's PTS:
-@example
-ffmpeg -f v4l2 -r 1 -i /dev/video0 -copyts -f image2 -frame_pts true %d.jpg"
-@end example
-
 @subsection Options
 
 @table @option
-@item frame_pts
-If set to 1, expand the filename with pts from pkt->pts.
-Default value is 0.
-
 @item start_number
 Start the sequence from the specified number. Default value is 1.
 
@@ -1405,10 +1092,6 @@ more efficient), but with this option set, the muxer writes one moof/mdat
 pair for each track, making it easier to separate tracks.
 
 This option is implicitly set when writing ismv (Smooth Streaming) files.
-@item -movflags skip_sidx
-Skip writing of sidx atom. When bitrate overhead due to sidx atom is high,
-this option could be used for cases where sidx atom is not mandatory.
-When global_sidx flag is enabled, this option will be ignored.
 @item -movflags faststart
 Run a second pass moving the index (moov atom) to the beginning of the file.
 This operation can take a while, and will not work in various situations such
@@ -1440,18 +1123,6 @@ be negative. This enables the initial sample to have DTS/CTS of zero, and
 reduces the need for edit lists for some cases such as video tracks with
 B-frames. Additionally, eases conformance with the DASH-IF interoperability
 guidelines.
-
-This option is implicitly set when writing ismv (Smooth Streaming) files.
-@item -write_prft
-Write producer time reference box (PRFT) with a specified time source for the
-NTP field in the PRFT box. Set value as @samp{wallclock} to specify timesource
-as wallclock time and @samp{pts} to specify timesource as input packets' PTS
-values.
-
-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}.
 @end table
 
 @subsection Example
@@ -1557,7 +1228,7 @@ Set the program @samp{service_type}. Default is @code{digital_tv}.
 Accepts the following options:
 @table @samp
 @item hex_value
-Any hexadecimal value between @code{0x01} and @code{0xff} as defined in
+Any hexdecimal value between @code{0x01} to @code{0xff} as defined in
 ETSI 300 468.
 @item digital_tv
 Digital TV service.
@@ -1660,7 +1331,7 @@ ffmpeg -i file.mpg -c copy \
      out.ts
 @end example
 
-@section mxf, mxf_d10, mxf_opatom
+@section mxf, mxf_d10
 
 MXF muxer.
 
@@ -1672,7 +1343,7 @@ The muxer options are:
 @item store_user_comments @var{bool}
 Set if user comments should be stored if available or never.
 IRT D-10 does not allow user comments. The default is thus to write them for
-mxf and mxf_opatom but not for mxf_d10
+mxf but not for mxf_d10
 @end table
 
 @section null
@@ -1978,9 +1649,6 @@ segment would usually span. Otherwise, the segment will be filled with the next
 packet written. Defaults to @code{0}.
 @end table
 
-Make sure to require a closed GOP when encoding and to set the GOP
-size to fit your segment time constraint.
-
 @subsection Examples
 
 @itemize
@@ -1989,7 +1657,7 @@ Remux the content of file @file{in.mkv} to a list of segments
 @file{out-000.nut}, @file{out-001.nut}, etc., and write the list of
 generated segments to @file{out.list}:
 @example
-ffmpeg -i in.mkv -codec hevc -flags +cgop -g 60 -map 0 -f segment -segment_list out.list out%03d.nut
+ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.list out%03d.nut
 @end example
 
 @item
@@ -2160,35 +1828,20 @@ ffmpeg -re -i ... -c:v libx264 -c:a aac -f fifo -fifo_format flv -map 0:v -map 0
 @anchor{tee}
 @section tee
 
-The tee muxer can be used to write the same data to several outputs, such as files or streams.
-It can be used, for example, to stream a video over a network and save it to disk at the same time.
+The tee muxer can be used to write the same data to several files or any
+other kind of muxer. It can be used, for example, to both stream a video to
+the network and save it to disk at the same time.
 
 It is different from specifying several outputs to the @command{ffmpeg}
-command-line tool. With the tee muxer, the audio and video data will be encoded only once.
-With conventional multiple outputs, multiple encoding operations in parallel are initiated,
-which can be a very expensive process. The tee muxer is not useful when using the libavformat API
-directly because it is then possible to feed the same packets to several muxers directly.
-
-Since the tee muxer does not represent any particular output format, ffmpeg cannot auto-select
-output streams. So all streams intended for output must be specified using @code{-map}. See
-the examples below.
-
-Some encoders may need different options depending on the output format;
-the auto-detection of this can not work with the tee muxer, so they need to be explicitly specified.
-The main example is the @option{global_header} flag.
-
-The slave outputs are specified in the file name given to the muxer,
-separated by '|'. If any of the slave name contains the '|' separator,
-leading or trailing spaces or any special character, those must be
-escaped (see @ref{quoting_and_escaping,,the "Quoting and escaping"
-section in the ffmpeg-utils(1) manual,ffmpeg-utils}).
-
-@subsection Options
+command-line tool because the audio and video data will be encoded only once
+with the tee muxer; encoding can be a very expensive process. It is not
+useful when using the libavformat API directly because it is then possible
+to feed the same packets to several muxers directly.
 
 @table @option
 
 @item use_fifo @var{bool}
-If set to 1, slave outputs will be processed in separate threads using the @ref{fifo}
+If set to 1, slave outputs will be processed in separate thread using @ref{fifo}
 muxer. This allows to compensate for different speed/latency/reliability of
 outputs and setup transparent recovery. By default this feature is turned off.
 
@@ -2197,6 +1850,12 @@ Options to pass to fifo pseudo-muxer instances. See @ref{fifo}.
 
 @end table
 
+The slave outputs are specified in the file name given to the muxer,
+separated by '|'. If any of the slave name contains the '|' separator,
+leading or trailing spaces or any special character, it must be
+escaped (see @ref{quoting_and_escaping,,the "Quoting and escaping"
+section in the ffmpeg-utils(1) manual,ffmpeg-utils}).
+
 Muxer options can be specified for each slave by prepending them as a list of
 @var{key}=@var{value} pairs separated by ':', between square brackets. If
 the options values contain a special character or the ':' separator, they
@@ -2205,27 +1864,13 @@ must be escaped; note that this is a second level escaping.
 The following special options are also recognized:
 @table @option
 @item f
-Specify the format name. Required if it cannot be guessed from the
-output URL.
+Specify the format name. Useful if it cannot be guessed from the
+output name suffix.
 
 @item bsfs[/@var{spec}]
 Specify a list of bitstream filters to apply to the specified
 output.
 
-It is possible to specify to which streams a given bitstream filter
-applies, by appending a stream specifier to the option separated by
-@code{/}. @var{spec} must be a stream specifier (see @ref{Format
-stream specifiers}).
-
-If the stream specifier is not specified, the bitstream filters will be
-applied to all streams in the output. This will cause that output operation
-to fail if the output contains streams to which the bitstream filter cannot
-be applied e.g. @code{h264_mp4toannexb} being applied to an output containing an audio stream.
-
-Options for a bitstream filter must be specified in the form of @code{opt=value}.
-
-Several bitstream filters can be specified, separated by ",".
-
 @item use_fifo @var{bool}
 This allows to override tee muxer use_fifo option for individual slave muxer.
 
@@ -2233,13 +1878,19 @@ This allows to override tee muxer use_fifo option for individual slave muxer.
 This allows to override tee muxer fifo_options for individual slave muxer.
 See @ref{fifo}.
 
+It is possible to specify to which streams a given bitstream filter
+applies, by appending a stream specifier to the option separated by
+@code{/}. @var{spec} must be a stream specifier (see @ref{Format
+stream specifiers}).  If the stream specifier is not specified, the
+bitstream filters will be applied to all streams in the output.
+
+Several bitstream filters can be specified, separated by ",".
+
 @item select
 Select the streams that should be mapped to the slave output,
 specified by a stream specifier. If not specified, this defaults to
-all the mapped streams. This will cause that output operation to fail
-if the output format does not accept all mapped streams.
-
-You may use multiple stream specifiers separated by commas (@code{,}) e.g.: @code{a:0,v}
+all the input streams. You may use multiple stream specifiers
+separated by commas (@code{,}) e.g.: @code{a:0,v}
 
 @item onfail
 Specify behaviour on output failure. This can be set to either @code{abort} (which is
@@ -2253,7 +1904,7 @@ will continue without being affected.
 @itemize
 @item
 Encode something and both archive it in a WebM file and stream it
-as MPEG-TS over UDP:
+as MPEG-TS over UDP (the streams need to be explicitly mapped):
 @example
 ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a
   "archive-20121107.mkv|[f=mpegts]udp://10.0.1.255:1234/"
@@ -2276,19 +1927,23 @@ option is applied to @file{out.aac} in order to make it contain only
 audio packets.
 @example
 ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac
-       -f tee "[bsfs/v=dump_extra=freq=keyframe]out.ts|[movflags=+faststart]out.mp4|[select=a]out.aac"
+       -f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=a]out.aac"
 @end example
 
 @item
-As above, but select only stream @code{a:1} for the audio output. Note
+As below, but select only stream @code{a:1} for the audio output. Note
 that a second level escaping must be performed, as ":" is a special
 character used to separate options.
 @example
 ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac
-       -f tee "[bsfs/v=dump_extra=freq=keyframe]out.ts|[movflags=+faststart]out.mp4|[select=\'a:1\']out.aac"
+       -f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=\'a:1\']out.aac"
 @end example
 @end itemize
 
+Note: some codecs may need different options depending on the output format;
+the auto-detection of this can not work with the tee muxer. The main example
+is the @option{global_header} flag.
+
 @section webm_dash_manifest
 
 WebM DASH Manifest muxer.

doc/outdevs.texi

@@ -140,8 +140,7 @@ device with @command{-list_formats 1}. Audio sample rate is always 48 kHz.
 
 @item list_devices
 If set to @option{true}, print a list of devices and exit.
-Defaults to @option{false}. Alternatively you can use the @code{-sinks}
-option of ffmpeg to list the available output devices.
+Defaults to @option{false}.
 
 @item list_formats
 If set to @option{true}, print a list of supported formats and exit.
@@ -151,14 +150,6 @@ Defaults to @option{false}.
 Amount of time to preroll video in seconds.
 Defaults to @option{0.5}.
 
-@item duplex_mode
-Sets the decklink device duplex mode. Must be @samp{unset}, @samp{half} or @samp{full}.
-Defaults to @samp{unset}.
-
-@item timing_offset
-Sets the genlock timing pixel offset on the used output.
-Defaults to @samp{unset}.
-
 @end table
 
 @subsection Examples
@@ -191,6 +182,51 @@ ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 -s 720x486 -r 24000/1001 'DeckLi
 
 @end itemize
 
+@section libndi_newtek
+
+The libndi_newtek output device provides playback capabilities for using NDI (Network
+Device Interface, standard created by NewTek).
+
+Output filename is a NDI name.
+
+To enable this output device, you need the NDI SDK and you
+need to configure with the appropriate @code{--extra-cflags}
+and @code{--extra-ldflags}.
+
+NDI uses uyvy422 pixel format natively, but also supports bgra, bgr0, rgba and
+rgb0.
+
+@subsection Options
+
+@table @option
+
+@item reference_level
+The audio reference level in dB. This specifies how many dB above the
+reference level (+4dBU) is the full range of 16 bit audio.
+Defaults to @option{0}.
+
+@item clock_video
+These specify whether video "clock" themselves.
+Defaults to @option{false}.
+
+@item clock_audio
+These specify whether audio "clock" themselves.
+Defaults to @option{false}.
+
+@end table
+
+@subsection Examples
+
+@itemize
+
+@item
+Play video clip:
+@example
+ffmpeg -i "udp://@@239.1.1.1:10480?fifo_size=1000000&overrun_nonfatal=1" -vf "scale=720:576,fps=fps=25,setdar=dar=16/9,format=pix_fmts=uyvy422" -f libndi_newtek NEW_NDI1
+@end example
+
+@end itemize
+
 @section fbdev
 
 Linux framebuffer output device.
@@ -357,18 +393,9 @@ Set the SDL window size, can be a string of the form
 If not specified it defaults to the size of the input video,
 downscaled according to the aspect ratio.
 
-@item window_x
-@item window_y
-Set the position of the window on the screen.
-
 @item window_fullscreen
 Set fullscreen mode when non-zero value is provided.
 Default value is zero.
-
-@item window_enable_quit
-Enable quit action (using window button or keyboard key)
-when non-zero value is provided.
-Default value is 1 (enable quit action)
 @end table
 
 @subsection Interactive commands
@@ -393,10 +420,6 @@ ffmpeg -i INPUT -c:v rawvideo -pix_fmt yuv420p -window_size qcif -f sdl "SDL out
 
 sndio audio output device.
 
-@section v4l2
-
-Video4Linux2 output device.
-
 @section xv
 
 XV (XVideo) output device.

doc/platform.texi

@@ -148,11 +148,16 @@ To target 32 bits replace @code{x86_64} with @code{i686} in the command above.
 
 @section Microsoft Visual C++ or Intel C++ Compiler for Windows
 
-FFmpeg can be built with MSVC 2013 or later.
+FFmpeg can be built with MSVC 2012 or earlier using a C99-to-C89 conversion utility
+and wrapper, or with MSVC 2013 and ICL natively.
 
 You will need the following prerequisites:
 
 @itemize
+@item @uref{https://github.com/libav/c99-to-c89/, C99-to-C89 Converter & Wrapper}
+(if using MSVC 2012 or earlier)
+@item @uref{http://code.google.com/p/msinttypes/, msinttypes}
+(if using MSVC 2012 or earlier)
 @item @uref{http://msys2.github.io/, MSYS2}
 @item @uref{http://www.nasm.us/, NASM}
 (Also available via MSYS2's package manager.)
@@ -161,13 +166,16 @@ You will need the following prerequisites:
 To set up a proper environment in MSYS2, you need to run @code{msys_shell.bat} from
 the Visual Studio or Intel Compiler command prompt.
 
-Place @code{yasm.exe} somewhere in your @code{PATH}.
+Place @code{yasm.exe} somewhere in your @code{PATH}. If using MSVC 2012 or
+earlier, place @code{c99wrap.exe} and @code{c99conv.exe} somewhere in your
+@code{PATH} as well.
 
 Next, make sure any other headers and libs you want to use, such as zlib, are
 located in a spot that the compiler can see. Do so by modifying the @code{LIB}
 and @code{INCLUDE} environment variables to include the @strong{Windows-style}
 paths to these directories. Alternatively, you can try to use the
-@code{--extra-cflags}/@code{--extra-ldflags} configure options.
+@code{--extra-cflags}/@code{--extra-ldflags} configure options. If using MSVC
+2012 or earlier, place @code{inttypes.h} somewhere the compiler can see too.
 
 Finally, run:
 
@@ -209,6 +217,8 @@ can see.
 
 @item FFmpeg has been tested with the following on i686 and x86_64:
 @itemize
+@item Visual Studio 2010 Pro and Express
+@item Visual Studio 2012 Pro and Express
 @item Visual Studio 2013 Pro and Express
 @item Intel Composer XE 2013
 @item Intel Composer XE 2013 SP1

doc/protocols.texi

@@ -193,20 +193,6 @@ Set I/O operation maximum block size, in bytes. Default value is
 @code{INT_MAX}, which results in not limiting the requested block size.
 Setting this value reasonably low improves user termination request reaction
 time, which is valuable for files on slow medium.
-
-@item follow
-If set to 1, the protocol will retry reading at the end of the file, allowing
-reading files that still are being written. In order for this to terminate,
-you either need to use the rw_timeout option, or use the interrupt callback
-(for API users).
-
-@item seekable
-Controls if seekability is advertised on the file. 0 means non-seekable, -1
-means auto (seekable for normal files, non-seekable for named pipes).
-
-Many demuxers handle seekable and non-seekable resources differently,
-overriding this might speed up opening certain files at the cost of losing some
-features (e.g. accurate seeking).
 @end table
 
 @section ftp
@@ -243,6 +229,17 @@ it, unless special care is taken (tests, customized server configuration
 etc.). Different FTP servers behave in different way during seek
 operation. ff* tools may produce incomplete content due to server limitations.
 
+This protocol accepts the following options:
+
+@table @option
+@item follow
+If set to 1, the protocol will retry reading at the end of the file, allowing
+reading files that still are being written. In order for this to terminate,
+you either need to use the rw_timeout option, or use the interrupt callback
+(for API users).
+
+@end table
+
 @section gopher
 
 Gopher protocol.
@@ -299,9 +296,6 @@ Use persistent connections if set to 1, default is 0.
 @item post_data
 Set custom HTTP post data.
 
-@item referer
-Set the Referer header. Include 'Referer: URL' header in HTTP request.
-
 @item user_agent
 Override the User-Agent header. If not specified the protocol will use a
 string describing the libavformat build. ("Lavf/<version>")
@@ -327,9 +321,6 @@ Sets the maximum delay in seconds after which to give up reconnecting
 @item mime_type
 Export the MIME type.
 
-@item http_version
-Exports the HTTP response version number. Usually "1.0" or "1.1".
-
 @item icy
 If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
 supports this, the metadata has to be retrieved by the application by reading
@@ -372,7 +363,7 @@ If set to 1 enables experimental HTTP server. This can be used to send data when
 used as an output option, or read data from a client with HTTP POST when used as
 an input option.
 If set to 2 enables experimental multi-client HTTP server. This is not yet implemented
-in ffmpeg.c and thus must not be used as a command line option.
+in ffmpeg.c or ffserver.c and thus must not be used as a command line option.
 @example
 # Server side (sending):
 ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://@var{server}:@var{port}
@@ -393,11 +384,6 @@ ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://@var{server}:@var{p
 wget --post-file=somefile.ogg http://@var{server}:@var{port}
 @end example
 
-@item send_expect_100
-Send an Expect: 100-continue header for POST. If set to 1 it will send, if set
-to 0 it won't, if set to -1 it will try to send if it is applicable. Default
-value is -1.
-
 @end table
 
 @subsection HTTP Cookies
@@ -1163,265 +1149,6 @@ If set to any value, listen for an incoming connection. Outgoing connection is d
 Set the maximum number of streams. By default no limit is set.
 @end table
 
-@section srt
-
-Haivision Secure Reliable Transport Protocol via libsrt.
-
-The supported syntax for a SRT URL is:
-@example
-srt://@var{hostname}:@var{port}[?@var{options}]
-@end example
-
-@var{options} contains a list of &-separated options of the form
-@var{key}=@var{val}.
-
-or
-
-@example
-@var{options} srt://@var{hostname}:@var{port}
-@end example
-
-@var{options} contains a list of '-@var{key} @var{val}'
-options.
-
-This protocol accepts the following options.
-
-@table @option
-@item connect_timeout
-Connection timeout; SRT cannot connect for RTT > 1500 msec
-(2 handshake exchanges) with the default connect timeout of
-3 seconds. This option applies to the caller and rendezvous
-connection modes. The connect timeout is 10 times the value
-set for the rendezvous mode (which can be used as a
-workaround for this connection problem with earlier versions).
-
-@item ffs=@var{bytes}
-Flight Flag Size (Window Size), in bytes. FFS is actually an
-internal parameter and you should set it to not less than
-@option{recv_buffer_size} and @option{mss}. The default value
-is relatively large, therefore unless you set a very large receiver buffer,
-you do not need to change this option. Default value is 25600.
-
-@item inputbw=@var{bytes/seconds}
-Sender nominal input rate, in bytes per seconds. Used along with
-@option{oheadbw}, when @option{maxbw} is set to relative (0), to
-calculate maximum sending rate when recovery packets are sent
-along with the main media stream:
-@option{inputbw} * (100 + @option{oheadbw}) / 100
-if @option{inputbw} is not set while @option{maxbw} is set to
-relative (0), the actual input rate is evaluated inside
-the library. Default value is 0.
-
-@item iptos=@var{tos}
-IP Type of Service. Applies to sender only. Default value is 0xB8.
-
-@item ipttl=@var{ttl}
-IP Time To Live. Applies to sender only. Default value is 64.
-
-@item latency
-Timestamp-based Packet Delivery Delay.
-Used to absorb bursts of missed packet retransmissions.
-This flag sets both @option{rcvlatency} and @option{peerlatency}
-to the same value. Note that prior to version 1.3.0
-this is the only flag to set the latency, however
-this is effectively equivalent to setting @option{peerlatency},
-when side is sender and @option{rcvlatency}
-when side is receiver, and the bidirectional stream
-sending is not supported.
-
-@item listen_timeout
-Set socket listen timeout.
-
-@item maxbw=@var{bytes/seconds}
-Maximum sending bandwidth, in bytes per seconds.
--1 infinite (CSRTCC limit is 30mbps)
-0 relative to input rate (see @option{inputbw})
->0 absolute limit value
-Default value is 0 (relative)
-
-@item mode=@var{caller|listener|rendezvous}
-Connection mode.
-@option{caller} opens client connection.
-@option{listener} starts server to listen for incoming connections.
-@option{rendezvous} use Rendez-Vous connection mode.
-Default value is caller.
-
-@item mss=@var{bytes}
-Maximum Segment Size, in bytes. Used for buffer allocation
-and rate calculation using a packet counter assuming fully
-filled packets. The smallest MSS between the peers is
-used. This is 1500 by default in the overall internet.
-This is the maximum size of the UDP packet and can be
-only decreased, unless you have some unusual dedicated
-network settings. Default value is 1500.
-
-@item nakreport=@var{1|0}
-If set to 1, Receiver will send `UMSG_LOSSREPORT` messages
-periodically until a lost packet is retransmitted or
-intentionally dropped. Default value is 1.
-
-@item oheadbw=@var{percents}
-Recovery bandwidth overhead above input rate, in percents.
-See @option{inputbw}. Default value is 25%.
-
-@item passphrase=@var{string}
-HaiCrypt Encryption/Decryption Passphrase string, length
-from 10 to 79 characters. The passphrase is the shared
-secret between the sender and the receiver. It is used
-to generate the Key Encrypting Key using PBKDF2
-(Password-Based Key Derivation Function). It is used
-only if @option{pbkeylen} is non-zero. It is used on
-the receiver only if the received data is encrypted.
-The configured passphrase cannot be recovered (write-only).
-
-@item payload_size=@var{bytes}
-Sets the maximum declared size of a packet transferred
-during the single call to the sending function in Live
-mode. Use 0 if this value isn't used (which is default in
-file mode).
-Default is -1 (automatic), which typically means MPEG-TS;
-if you are going to use SRT
-to send any different kind of payload, such as, for example,
-wrapping a live stream in very small frames, then you can
-use a bigger maximum frame size, though not greater than
-1456 bytes.
-
-@item pkt_size=@var{bytes}
-Alias for @samp{payload_size}.
-
-@item peerlatency
-The latency value (as described in @option{rcvlatency}) that is
-set by the sender side as a minimum value for the receiver.
-
-@item pbkeylen=@var{bytes}
-Sender encryption key length, in bytes.
-Only can be set to 0, 16, 24 and 32.
-Enable sender encryption if not 0.
-Not required on receiver (set to 0),
-key size obtained from sender in HaiCrypt handshake.
-Default value is 0.
-
-@item rcvlatency
-The time that should elapse since the moment when the
-packet was sent and the moment when it's delivered to
-the receiver application in the receiving function.
-This time should be a buffer time large enough to cover
-the time spent for sending, unexpectedly extended RTT
-time, and the time needed to retransmit the lost UDP
-packet. The effective latency value will be the maximum
-of this options' value and the value of @option{peerlatency}
-set by the peer side. Before version 1.3.0 this option
-is only available as @option{latency}.
-
-@item recv_buffer_size=@var{bytes}
-Set UDP receive buffer size, expressed in bytes.
-
-@item send_buffer_size=@var{bytes}
-Set UDP send buffer size, expressed in bytes.
-
-@item rw_timeout
-Set raise error timeout for read/write optations.
-
-This option is only relevant in read mode:
-if no data arrived in more than this time
-interval, raise error.
-
-@item tlpktdrop=@var{1|0}
-Too-late Packet Drop. When enabled on receiver, it skips
-missing packets that have not been delivered in time and
-delivers the following packets to the application when
-their time-to-play has come. It also sends a fake ACK to
-the sender. When enabled on sender and enabled on the
-receiving peer, the sender drops the older packets that
-have no chance of being delivered in time. It was
-automatically enabled in the sender if the receiver
-supports it.
-
-@item sndbuf=@var{bytes}
-Set send buffer size, expressed in bytes.
-
-@item rcvbuf=@var{bytes}
-Set receive buffer size, expressed in bytes.
-
-Receive buffer must not be greater than @option{ffs}.
-
-@item lossmaxttl=@var{packets}
-The value up to which the Reorder Tolerance may grow. When
-Reorder Tolerance is > 0, then packet loss report is delayed
-until that number of packets come in. Reorder Tolerance
-increases every time a "belated" packet has come, but it
-wasn't due to retransmission (that is, when UDP packets tend
-to come out of order), with the difference between the latest
-sequence and this packet's sequence, and not more than the
-value of this option. By default it's 0, which means that this
-mechanism is turned off, and the loss report is always sent
-immediately upon experiencing a "gap" in sequences.
-
-@item minversion
-The minimum SRT version that is required from the peer. A connection
-to a peer that does not satisfy the minimum version requirement
-will be rejected.
-
-The version format in hex is 0xXXYYZZ for x.y.z in human readable
-form.
-
-@item streamid=@var{string}
-A string limited to 512 characters that can be set on the socket prior
-to connecting. This stream ID will be able to be retrieved by the
-listener side from the socket that is returned from srt_accept and
-was connected by a socket with that set stream ID. SRT does not enforce
-any special interpretation of the contents of this string.
-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 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
-type must be exactly the same on both connecting parties, otherwise
-the connection is rejected.
-
-@item messageapi=@var{1|0}
-When set, this socket uses the Message API, otherwise it uses Buffer
-API. Note that in live mode (see @option{transtype}) there’s only
-message API available. In File mode you can chose to use one of two modes:
-
-Stream API (default, when this option is false). In this mode you may
-send as many data as you wish with one sending instruction, or even use
-dedicated functions that read directly from a file. The internal facility
-will take care of any speed and congestion control. When receiving, you
-can also receive as many data as desired, the data not extracted will be
-waiting for the next call. There is no boundary between data portions in
-the Stream mode.
-
-Message API. In this mode your single sending instruction passes exactly
-one piece of data that has boundaries (a message). Contrary to Live mode,
-this message may span across multiple UDP packets and the only size
-limitation is that it shall fit as a whole in the sending buffer. The
-receiver shall use as large buffer as necessary to receive the message,
-otherwise the message will not be given up. When the message is not
-complete (not all packets received or there was a packet loss) it will
-not be given up.
-
-@item transtype=@var{live|file}
-Sets the transmission type for the socket, in particular, setting this
-option sets multiple other parameters to their default values as required
-for a particular transmission type.
-
-live: Set options as for live transmission. In this mode, you should
-send by one sending instruction only so many data that fit in one UDP packet,
-and limited to the value defined first in @option{payload_size} (1316 is
-default in this mode). There is no speed control in this mode, only the
-bandwidth control, if configured, in order to not exceed the bandwidth with
-the overhead transmission (retransmitted and control packets).
-
-file: Set options as for non-live transmission. See @option{messageapi}
-for further explanations
-
-@end table
-
-For more information see: @url{https://github.com/Haivision/srt}.
-
 @section srtp
 
 Secure Real-time Transport Protocol.
@@ -1459,7 +1186,6 @@ Accepted options:
 Start offset of the extracted segment, in bytes.
 @item end
 End offset of the extracted segment, in bytes.
-If set to 0, extract till end of file.
 @end table
 
 Examples:
@@ -1475,11 +1201,6 @@ Play an AVI file directly from a TAR archive:
 subfile,,start,183241728,end,366490624,,:archive.tar
 @end example
 
-Play a MPEG-TS file from start offset till end:
-@example
-subfile,,start,32815239,end,0,,:video.ts
-@end example
-
 @section tee
 
 Writes the output to multiple protocols. The individual outputs are separated
@@ -1521,12 +1242,6 @@ Set receive buffer size, expressed bytes.
 
 @item send_buffer_size=@var{bytes}
 Set send buffer size, expressed bytes.
-
-@item tcp_nodelay=@var{1|0}
-Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
-
-@item tcp_mss=@var{bytes}
-Set maximum segment size for outgoing TCP packets, expressed in bytes.
 @end table
 
 The following example shows how to setup a listening TCP connection
@@ -1562,7 +1277,7 @@ If enabled, try to verify the peer that we are communicating with.
 Note, if using OpenSSL, this currently only makes sure that the
 peer certificate is signed by one of the root certificates in the CA
 database, but it does not validate that the certificate actually
-matches the host name we are trying to connect to. (With other backends,
+matches the host name we are trying to connect to. (With GnuTLS,
 the host name is validated as well.)
 
 This is disabled by default since it requires a CA database to be
@@ -1633,8 +1348,9 @@ packet bursts.
 Override the local UDP port to bind with.
 
 @item localaddr=@var{addr}
-Local IP address of a network interface used for sending packets or joining
-multicast groups.
+Choose the local IP address. This is useful e.g. if sending multicast
+and the host has multiple interfaces, where the user can choose
+which interface to send on by specifying the IP address of that interface.
 
 @item pkt_size=@var{size}
 Set the size in bytes of UDP packets.
@@ -1657,12 +1373,12 @@ For receiving, this gives the benefit of only receiving packets from
 the specified peer address/port.
 
 @item sources=@var{address}[,@var{address}]
-Only receive packets sent from the specified addresses. In case of multicast,
-also subscribe to multicast traffic coming from these addresses only.
+Only receive packets sent to the multicast group from one of the
+specified sender IP addresses.
 
 @item block=@var{address}[,@var{address}]
-Ignore packets sent from the specified addresses. In case of multicast, also
-exclude the source addresses in the multicast subscription.
+Ignore packets sent to the multicast group from the specified
+sender IP addresses.
 
 @item fifo_size=@var{units}
 Set the UDP receiving circular buffer size, expressed as a number of

doc/scaler.texi

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

doc/snow.txt

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

doc/utils.texi

@@ -1057,3 +1057,33 @@ indication of the corresponding powers of 10 and of 2.
 @end table
 
 @c man end EXPRESSION EVALUATION
+
+@chapter OpenCL Options
+@c man begin OPENCL OPTIONS
+
+When FFmpeg is configured with @code{--enable-opencl}, it is possible
+to set the options for the global OpenCL context.
+
+The list of supported options follows:
+
+@table @option
+@item build_options
+Set build options used to compile the registered kernels.
+
+See reference "OpenCL Specification Version: 1.2 chapter 5.6.4".
+
+@item platform_idx
+Select the index of the platform to run OpenCL code.
+
+The specified index must be one of the indexes in the device list
+which can be obtained with @code{ffmpeg -opencl_bench} or @code{av_opencl_get_device_list()}.
+
+@item device_idx
+Select the index of the device used to run OpenCL code.
+
+The specified index must be one of the indexes in the device list which
+can be obtained with @code{ffmpeg -opencl_bench} or @code{av_opencl_get_device_list()}.
+
+@end table
+
+@c man end OPENCL OPTIONS