update for FFmpeg 2.3.4

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
avcodec/mjpegdec: check bits per pixel for changes similar to dimensions
2025-12-07 15:30:00 +01:00 · 2014-10-03 03:41:00 +02:00 · 2014-10-03 03:40:59 +02:00 · 2014-10-03 03:40:59 +02:00 · 2014-10-03 03:40:59 +02:00 · 2014-10-03 03:40:59 +02:00
7695 changed files with 326027 additions and 1129337 deletions
--- a/.gitattributes
+++ b/.gitattributes
@@ -1,2 +0,0 @@
 *.pnm -diff -text
 tests/ref/fate/sub-scc eol=crlf
--- a/.gitignore
+++ b/.gitignore
@@ -1,6 +1,5 @@
 *.a
 *.o
 *.o.*
 *.d
 *.def
 *.dll
@@ -18,26 +17,70 @@
 *.so.*
 *.swp
 *.ver
-*.version
+*-example
-*.metal.air
+*-test
 *.metallib
 *.metallib.c
 *.ptx
 *.ptx.c
 *.ptx.gz
 *_g
 \#*
 .\#*
 /.config
 /.version
 /ffmpeg
 /ffplay
 /ffprobe
-/config.asm
+/ffserver
-/config.h
+/config.*
 /config_components.h
 /coverage.info
 /doc/*.1
 /doc/*.3
 /doc/*.html
 /doc/*.pod
 /doc/config.texi
 /doc/avoptions_codec.texi
 /doc/avoptions_format.texi
 /doc/doxy/html/
 /doc/examples/avio_reading
 /doc/examples/decoding_encoding
 /doc/examples/demuxing_decoding
 /doc/examples/filter_audio
 /doc/examples/filtering_audio
 /doc/examples/filtering_video
 /doc/examples/metadata
 /doc/examples/muxing
 /doc/examples/pc-uninstalled
 /doc/examples/remuxing
 /doc/examples/resampling_audio
 /doc/examples/scaling_video
 /doc/examples/transcode_aac
 /doc/examples/transcoding
 /doc/fate.txt
 /doc/print_options
 /lcov/
-/src
+/libavcodec/*_tablegen
-/mapfile
+/libavcodec/*_tables.c
-/tools/python/__pycache__/
+/libavcodec/*_tables.h
 /libavutil/avconfig.h
 /libavutil/ffversion.h
 /tests/audiogen
 /tests/base64
 /tests/data/
 /tests/rotozoom
 /tests/tiny_psnr
 /tests/tiny_ssim
 /tests/videogen
 /tests/vsynth1/
 /tools/aviocat
 /tools/ffbisect
 /tools/bisect.need
 /tools/crypto_bench
 /tools/cws2fws
 /tools/fourcc2pixfmt
 /tools/ffescape
 /tools/ffeval
 /tools/ffhash
 /tools/graph2dot
 /tools/ismindex
 /tools/pktdumper
 /tools/probetest
 /tools/qt-faststart
 /tools/trasher
 /tools/seek_print
 /tools/uncoded_frame
 /tools/zmqsend
--- a/.mailmap
+++ b/.mailmap
@@ -1,25 +0,0 @@
 <james.darnley@gmail.com> <jdarnley@obe.tv>
 <jeebjp@gmail.com> <jan.ekstrom@aminocom.com>
 <sw@jkqxz.net> <mrt@jkqxz.net>
 <u@pkh.me> <cboesch@gopro.com>
 <zhilizhao@tencent.com> <quinkblack@foxmail.com>
 <zhilizhao@tencent.com> <wantlamy@gmail.com>
 <modmaker@google.com> <modmaker-at-google.com@ffmpeg.org>
 <stebbins@jetheaddev.com> <jstebbins@jetheaddev.com>
 <barryjzhao@tencent.com> <mypopydev@gmail.com>
 <barryjzhao@tencent.com> <jun.zhao@intel.com>
 <josh@itanimul.li> <joshdk@obe.tv>
 <michael@niedermayer.cc> <michaelni@gmx.at>
 <linjie.justin.fu@gmail.com> <linjie.fu@intel.com>
 <linjie.justin.fu@gmail.com> <fulinjie@zju.edu.cn>
 <ceffmpeg@gmail.com> <cehoyos@ag.or.at>
 <ceffmpeg@gmail.com> <cehoyos@rainbow.studorg.tuwien.ac.at>
 <ffmpeg@gyani.pro> <gyandoshi@gmail.com>
 <atomnuker@gmail.com> <rpehlivanov@obe.tv>
 <lizhong1008@gmail.com> <zhong.li@intel.com>
 <lizhong1008@gmail.com> <zhongli_dev@126.com>
 <andreas.rheinhardt@gmail.com> <andreas.rheinhardt@googlemail.com>
 rcombs <rcombs@rcombs.me> <rodger.combs@gmail.com>
 <thilo.borgmann@mail.de> <thilo.borgmann@googlemail.com>
 <liuqi05@kuaishou.com> <lq@chinaffmpeg.org>
 <ruiling.song83@gmail.com> <ruiling.song@intel.com>
--- a/.travis.yml
+++ b/.travis.yml
@@ -1,30 +0,0 @@
 language: c
 sudo: false
 os:
  - linux
  - osx
 addons:
  apt:
    packages:
      - nasm
      - diffutils
 compiler:
  - clang
  - gcc
 matrix:
    exclude:
        - os: osx
          compiler: gcc
 cache:
  directories:
    - ffmpeg-samples
 before_install:
  - if [ "$TRAVIS_OS_NAME" == "osx" ]; then brew update; fi
 install:
  - if [ "$TRAVIS_OS_NAME" == "osx" ]; then brew install nasm; fi
 script:
  - mkdir -p ffmpeg-samples
  - ./configure --samples=ffmpeg-samples --cc=$CC
  - make -j 8
  - make fate-rsync
  - make check -j 8
--- a/CONTRIBUTING.md
+++ b/CONTRIBUTING.md
@@ -1,4 +0,0 @@
 # Note to Github users
 Patches should be submitted to the [ffmpeg-devel mailing list](https://ffmpeg.org/mailman/listinfo/ffmpeg-devel) using `git format-patch` or `git send-email`. Github pull requests should be avoided because they are not part of our review process and **will be ignored**.
 See [https://ffmpeg.org/developer.html#Contributing](https://ffmpeg.org/developer.html#Contributing) for more information.
--- a/785
+++ b/785
@@ -1,761 +1,29 @@
 Entries are sorted chronologically from oldest to youngest within each release,
 releases are sorted from youngest to oldest.
-version 5.1:
+version 2.3.3:
- add ipfs/ipns protocol support
+- h264: fix grayscale only decoding with weighted prediction
- dialogue enhance audio filter
+- mjpegdec: support AV_PIX_FMT_YUV420P16 with upscale_h
- dropped obsolete XvMC hwaccel
+- proresenc_ks: fix buffer overflow
- pcm-bluray encoder
+- matroskadec: fix crash
 - DFPWM audio encoder/decoder and raw muxer/demuxer
 - SITI filter
 - Vizrt Binary Image encoder/decoder
 - avsynctest source filter
 - feedback video filter
 - pixelize video filter
 - colormap video filter
 - colorchart video source filter
 - multiply video filter
 - PGS subtitle frame merge bitstream filter
 - blurdetect filter
 - tiltshelf audio filter
 - QOI image format support
 - ffprobe -o option
 - virtualbass audio filter
 - VDPAU AV1 hwaccel
 - PHM image format support
 - remap_opencl filter
 - added chromakey_cuda filter
 version 2.3.2:
 - snow: fix null pointer dereference
 - huffyucdec: fix overread
 - vc1dec: fix crash
 - iff: fix out of array access
 - matroskaenc: fix assertion failure
 - cdgraphics: fix infinite loop
 - dvdsub_parser: fix infinite loop
 - mpeg12dec: support decoding some broken files
 - v4l2enc: fix crash
 - h264_parser: fix handling huge resolutions
 - h264_mp4toannexb_bsf: multiple bugfixes
-version 5.0:
+version 2.3.1:
- ADPCM IMA Westwood encoder
+- public AVDCT API/ABI for DCT functions
- Westwood AUD muxer
+- g2meet: allow size changes within original sizes
- ADPCM IMA Acorn Replay decoder
+- dv: improved error resilience, fixing Ticket2340 and Ticket2341
 - Argonaut Games CVG demuxer
 - Argonaut Games CVG muxer
 - Concatf protocol
 - afwtdn audio filter
 - audio and video segment filters
 - Apple Graphics (SMC) encoder
 - hsvkey and hsvhold video filters
 - adecorrelate audio filter
 - atilt audio filter
 - grayworld video filter
 - AV1 Low overhead bitstream format muxer
 - swscale slice threading
 - MSN Siren decoder
 - scharr video filter
 - apsyclip audio filter
 - morpho video filter
 - amr parser
 - (a)latency filters
 - GEM Raster image decoder
 - asdr audio filter
 - speex decoder
 - limitdiff video filter
 - xcorrelate video filter
 - varblur video filter
 - huesaturation video filter
 - colorspectrum source video filter
 - RTP packetizer for uncompressed video (RFC 4175)
 - bitpacked encoder
 - VideoToolbox VP9 hwaccel
 - VideoToolbox ProRes hwaccel
 - support loongarch.
 - aspectralstats audio filter
 - adynamicsmooth audio filter
 - libplacebo filter
 - vflip_vulkan, hflip_vulkan and flip_vulkan filters
 - adynamicequalizer audio filter
 - yadif_videotoolbox filter
 - VideoToolbox ProRes encoder
 - anlmf audio filter
 - IMF demuxer (experimental)
 version 4.4:
 - AudioToolbox output device
 - MacCaption demuxer
 - PGX decoder
 - chromanr video filter
 - VDPAU accelerated HEVC 10/12bit decoding
 - ADPCM IMA Ubisoft APM encoder
 - Rayman 2 APM muxer
 - AV1 encoding support SVT-AV1
 - Cineform HD encoder
 - ADPCM Argonaut Games encoder
 - Argonaut Games ASF muxer
 - AV1 Low overhead bitstream format demuxer
 - RPZA video encoder
 - ADPCM IMA MOFLEX decoder
 - MobiClip FastAudio decoder
 - MobiClip video decoder
 - MOFLEX demuxer
 - MODS demuxer
 - PhotoCD decoder
 - MCA demuxer
 - AV1 decoder (Hardware acceleration used only)
 - SVS demuxer
 - Argonaut Games BRP demuxer
 - DAT demuxer
 - aax demuxer
 - IPU decoder, parser and demuxer
 - Intel QSV-accelerated AV1 decoding
 - Argonaut Games Video decoder
 - libwavpack encoder removed
 - ACE demuxer
 - AVS3 demuxer
 - AVS3 video decoder via libuavs3d
 - Cintel RAW decoder
 - VDPAU accelerated VP9 10/12bit decoding
 - afreqshift and aphaseshift filters
 - High Voltage Software ADPCM encoder
 - LEGO Racers ALP (.tun & .pcm) muxer
 - AV1 VAAPI decoder
 - adenorm filter
 - ADPCM IMA AMV encoder
 - AMV muxer
 - NVDEC AV1 hwaccel
 - DXVA2/D3D11VA hardware accelerated AV1 decoding
 - speechnorm filter
 - SpeedHQ encoder
 - asupercut filter
 - asubcut filter
 - Microsoft Paint (MSP) version 2 decoder
 - Microsoft Paint (MSP) demuxer
 - AV1 monochrome encoding support via libaom >= 2.0.1
 - asuperpass and asuperstop filter
 - shufflepixels filter
 - tmidequalizer filter
 - estdif filter
 - epx filter
 - Dolby E parser
 - shear filter
 - kirsch filter
 - colortemperature filter
 - colorcontrast filter
 - PFM encoder
 - colorcorrect filter
 - binka demuxer
 - XBM parser
 - xbm_pipe demuxer
 - colorize filter
 - CRI parser
 - aexciter audio filter
 - exposure video filter
 - monochrome video filter
 - setts bitstream filter
 - vif video filter
 - OpenEXR image encoder
 - Simbiosis IMX decoder
 - Simbiosis IMX demuxer
 - Digital Pictures SGA demuxer and decoders
 - TTML subtitle encoder and muxer
 - identity video filter
 - msad video filter
 - gophers protocol
 - RIST protocol via librist
 version 4.3:
 - v360 filter
 - Intel QSV-accelerated MJPEG decoding
 - Intel QSV-accelerated VP9 decoding
 - Support for TrueHD in mp4
 - Support AMD AMF encoder on Linux (via Vulkan)
 - IMM5 video decoder
 - ZeroMQ protocol
 - support Sipro ACELP.KELVIN decoding
 - streamhash muxer
 - sierpinski video source
 - scroll video filter
 - photosensitivity filter
 - anlms filter
 - arnndn filter
 - bilateral filter
 - maskedmin and maskedmax filters
 - VDPAU VP9 hwaccel
 - median filter
 - QSV-accelerated VP9 encoding
 - AV1 encoding support via librav1e
 - AV1 frame merge bitstream filter
 - AV1 Annex B demuxer
 - axcorrelate filter
 - mvdv decoder
 - mvha decoder
 - MPEG-H 3D Audio support in mp4
 - thistogram filter
 - freezeframes filter
 - Argonaut Games ADPCM decoder
 - Argonaut Games ASF demuxer
 - xfade video filter
 - xfade_opencl filter
 - afirsrc audio filter source
 - pad_opencl filter
 - Simon & Schuster Interactive ADPCM decoder
 - Real War KVAG demuxer
 - CDToons video decoder
 - siren audio decoder
 - Rayman 2 ADPCM decoder
 - Rayman 2 APM demuxer
 - cas video filter
 - High Voltage Software ADPCM decoder
 - LEGO Racers ALP (.tun & .pcm) demuxer
 - AMQP 0-9-1 protocol (RabbitMQ)
 - Vulkan support
 - avgblur_vulkan, overlay_vulkan, scale_vulkan and chromaber_vulkan filters
 - ADPCM IMA MTF decoder
 - FWSE demuxer
 - DERF DPCM decoder
 - DERF demuxer
 - CRI HCA decoder
 - CRI HCA demuxer
 - overlay_cuda filter
 - switch from AvxSynth to AviSynth+ on Linux
 - mv30 decoder
 - Expanded styling support for 3GPP Timed Text Subtitles (movtext)
 - WebP parser
 - tmedian filter
 - maskedthreshold filter
 - Support for muxing pcm and pgs in m2ts
 - Cunning Developments ADPCM decoder
 - asubboost filter
 - Pro Pinball Series Soundbank demuxer
 - pcm_rechunk bitstream filter
 - scdet filter
 - NotchLC decoder
 - gradients source video filter
 - MediaFoundation encoder wrapper
 - untile filter
 - Simon & Schuster Interactive ADPCM encoder
 - PFM decoder
 - dblur video filter
 - Real War KVAG muxer
 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:
 - deflicker video filter
 - doubleweave video filter
 - lumakey video filter
 - pixscope video filter
 - oscilloscope video filter
 - config.log and other configuration files moved into ffbuild/ directory
 - update cuvid/nvenc headers to Video Codec SDK 8.0.14
 - afir audio filter
 - scale_cuda CUDA based video scale filter
 - librsvg support for svg rasterization
 - crossfeed audio filter
 - spec compliant VP9 muxing support in MP4
 - remove the libnut muxer/demuxer wrappers
 - remove the libschroedinger encoder/decoder wrappers
 - surround audio filter
 - sofalizer filter switched to libmysofa
 - Gremlin Digital Video demuxer and decoder
 - headphone audio filter
 - superequalizer audio filter
 - roberts video filter
 - The x86 assembler default switched from yasm to nasm, pass
  --x86asmexe=yasm to configure to restore the old behavior.
 - additional frame format support for Interplay MVE movies
 - support for decoding through D3D11VA in ffmpeg
 - limiter video filter
 - libvmaf video filter
 - Dolby E decoder and SMPTE 337M demuxer
 - unpremultiply video filter
 - tlut2 video filter
 - floodfill video filter
 - pseudocolor video filter
 - raw G.726 muxer and demuxer, left- and right-justified
 - NewTek NDI input/output device
 - Some video filters with several inputs now use a common set of options:
  blend, libvmaf, lut3d, overlay, psnr, ssim.
  They must always be used by name.
 - FITS demuxer and decoder
 - FITS muxer and encoder
 - add --disable-autodetect build switch
 - drop deprecated qtkit input device (use avfoundation instead)
 - despill video filter
 - haas audio filter
 - SUP/PGS subtitle muxer
 - convolve video filter
 - VP9 tile threading support
 - KMS screen grabber
 - CUDA thumbnail filter
 - V4L2 mem2mem HW assisted codecs
 - Rockchip MPP hardware decoding
 - vmafmotion video filter
 - use MIME type "G726" for little-endian G.726, "AAL2-G726" for big-endian G.726
 version 3.3:
 - CrystalHD decoder moved to new decode API
 - add internal ebur128 library, remove external libebur128 dependency
 - Pro-MPEG CoP #3-R2 FEC protocol
 - premultiply video filter
 - Support for spherical videos
 - configure now fails if autodetect-libraries are requested but not found
 - PSD Decoder
 - 16.8 floating point pcm decoder
 - 24.0 floating point pcm decoder
 - Apple Pixlet decoder
 - QDMC audio decoder
 - NewTek SpeedHQ decoder
 - MIDI Sample Dump Standard demuxer
 - readeia608 filter
 - Sample Dump eXchange demuxer
 - abitscope multimedia filter
 - Scenarist Closed Captions demuxer and muxer
 - threshold filter
 - midequalizer filter
 - Optimal Huffman tables for (M)JPEG encoding
 - VAAPI-accelerated MPEG-2 and VP8 encoding
 - FM Screen Capture Codec decoder
 - native Opus encoder
 - ScreenPressor decoder
 - incomplete ClearVideo decoder
 - Intel QSV video scaling and deinterlacing filters
 - Support MOV with multiple sample description tables
 - XPM decoder
 - Removed the legacy X11 screen grabber, use XCB instead
 - MPEG-7 Video Signature filter
 - Removed asyncts filter (use af_aresample instead)
 - Intel QSV-accelerated VP8 video decoding
 - VAAPI-accelerated deinterlacing
 version 3.2:
 - libopenmpt demuxer
 - tee protocol
 - Changed metadata print option to accept general urls
 - Alias muxer for Ogg Video (.ogv)
 - VP8 in Ogg muxing
 - curves filter doesn't automatically insert points at x=0 and x=1 anymore
 - 16-bit support in curves filter and selectivecolor filter
 - OpenH264 decoder wrapper
 - MediaCodec H.264/HEVC/MPEG-4/VP8/VP9 hwaccel
 - True Audio (TTA) muxer
 - crystalizer audio filter
 - acrusher audio filter
 - bitplanenoise video filter
 - floating point support in als decoder
 - fifo muxer
 - maskedclamp filter
 - hysteresis filter
 - lut2 filter
 - yuvtestsrc filter
 - CUDA CUVID H.263/VP8/VP9/10 bit HEVC (Dithered) Decoding
 - vaguedenoiser filter
 - added threads option per filter instance
 - weave filter
 - gblur filter
 - avgblur filter
 - sobel and prewitt filter
 - MediaCodec HEVC/MPEG-4/VP8/VP9 decoding
 - Meridian Lossless Packing (MLP) / TrueHD encoder
 - Non-Local Means (nlmeans) denoising filter
 - sdl2 output device and ffplay support
 - sdl1 output device and sdl1 support removed
 - extended mov edit list support
 - libfaac encoder removed
 - Matroska muxer now writes CRC32 elements by default in all Level 1 elements
 - sidedata video and asidedata audio filter
 - Changed mapping of rtp MIME type G726 to codec g726le.
 - spec compliant VAAPI/DXVA2 VC-1 decoding of slices in frame-coded images
 version 3.1:
 - DXVA2-accelerated HEVC Main10 decoding
 - fieldhint filter
 - loop video filter and aloop audio filter
 - Bob Weaver deinterlacing filter
 - firequalizer filter
 - datascope filter
 - bench and abench filters
 - ciescope filter
 - protocol blacklisting API
 - MediaCodec H264 decoding
 - VC-2 HQ RTP payload format (draft v1) depacketizer and packetizer
 - VP9 RTP payload format (draft v2) packetizer
 - AudioToolbox audio decoders
 - AudioToolbox audio encoders
 - coreimage filter (GPU based image filtering on OSX)
 - libdcadec removed
 - bitstream filter for extracting DTS core
 - ADPCM IMA DAT4 decoder
 - musx demuxer
 - aix demuxer
 - remap filter
 - hash and framehash muxers
 - colorspace filter
 - hdcd filter
 - readvitc filter
 - VAAPI-accelerated format conversion and scaling
 - libnpp/CUDA-accelerated format conversion and scaling
 - Duck TrueMotion 2.0 Real Time decoder
 - Wideband Single-bit Data (WSD) demuxer
 - VAAPI-accelerated H.264/HEVC/MJPEG encoding
 - DTS Express (LBR) decoder
 - Generic OpenMAX IL encoder with support for Raspberry Pi
 - IFF ANIM demuxer & decoder
 - Direct Stream Transfer (DST) decoder
 - loudnorm filter
 - MTAF demuxer and decoder
 - MagicYUV decoder
 - OpenExr improvements (tile data and B44/B44A support)
 - BitJazz SheerVideo decoder
 - CUDA CUVID H264/HEVC decoder
 - 10-bit depth support in native utvideo decoder
 - libutvideo wrapper removed
 - YUY2 Lossless Codec decoder
 - VideoToolbox H.264 encoder
 version 3.0:
 - Common Encryption (CENC) MP4 encoding and decoding support
 - DXV decoding
 - extrastereo filter
 - ocr filter
 - alimiter filter
 - stereowiden filter
 - stereotools filter
 - rubberband filter
 - tremolo filter
 - agate filter
 - chromakey filter
 - maskedmerge filter
 - Screenpresso SPV1 decoding
 - chromaprint fingerprinting muxer
 - ffplay dynamic volume control
 - displace filter
 - selectivecolor filter
 - extensive native AAC encoder improvements and removal of experimental flag
 - ADPCM PSX decoder
 - 3dostr, dcstr, fsb, genh, vag, xvag, ads, msf, svag & vpk demuxer
 - zscale filter
 - wve demuxer
 - zero-copy Intel QSV transcoding in ffmpeg
 - shuffleframes filter
 - SDX2 DPCM decoder
 - vibrato filter
 - innoHeim/Rsupport Screen Capture Codec decoder
 - ADPCM AICA decoder
 - Interplay ACM demuxer and audio decoder
 - XMA1 & XMA2 decoder
 - realtime filter
 - anoisesrc audio filter source
 - IVR demuxer
 - compensationdelay filter
 - acompressor filter
 - support encoding 16-bit RLE SGI images
 - apulsator filter
 - sidechaingate audio filter
 - mipsdspr1 option has been renamed to mipsdsp
 - aemphasis filter
 - mips32r5 option has been removed
 - mips64r6 option has been removed
 - DXVA2-accelerated VP9 decoding
 - SOFAlizer: virtual binaural acoustics filter
 - VAAPI VP9 hwaccel
 - audio high-order multiband parametric equalizer
 - automatic bitstream filtering
 - showspectrumpic filter
 - libstagefright support removed
 - spectrumsynth filter
 - ahistogram filter
 - only seek with the right mouse button in ffplay
 - toggle full screen when double-clicking with the left mouse button in ffplay
 - afftfilt filter
 - convolution filter
 - libquvi support removed
 - support for dvaudio in wav and avi
 - libaacplus and libvo-aacenc support removed
 - Cineform HD decoder
 - new DCA decoder with full support for DTS-HD extensions
 - significant performance improvements in Windows Television (WTV) demuxer
 - nnedi deinterlacer
 - streamselect video and astreamselect audio filter
 - swaprect filter
 - metadata video and ametadata audio filter
 - SMPTE VC-2 HQ profile support for the Dirac decoder
 - SMPTE VC-2 native encoder supporting the HQ profile
 version 2.8:
 - colorkey video filter
 - BFSTM/BCSTM demuxer
 - little-endian ADPCM_THP decoder
 - Hap decoder and encoder
 - DirectDraw Surface image/texture decoder
 - ssim filter
 - optional new ASF demuxer
 - showvolume filter
 - Many improvements to the JPEG 2000 decoder
 - Go2Meeting decoding support
 - adrawgraph audio and drawgraph video filter
 - removegrain video filter
 - Intel QSV-accelerated MPEG-2 video and HEVC encoding
 - Intel QSV-accelerated MPEG-2 video and HEVC decoding
 - Intel QSV-accelerated VC-1 video decoding
 - libkvazaar HEVC encoder
 - erosion, dilation, deflate and inflate video filters
 - Dynamic Audio Normalizer as dynaudnorm filter
 - Reverse video and areverse audio filter
 - Random filter
 - deband filter
 - AAC fixed-point decoding
 - sidechaincompress audio filter
 - bitstream filter for converting HEVC from MP4 to Annex B
 - acrossfade audio filter
 - allyuv and allrgb video sources
 - atadenoise video filter
 - OS X VideoToolbox support
 - aphasemeter filter
 - showfreqs filter
 - vectorscope filter
 - waveform filter
 - hstack and vstack filter
 - Support DNx100 (1440x1080@8)
 - VAAPI hevc hwaccel
 - VDPAU hevc hwaccel
 - framerate filter
 - Switched default encoders for webm to VP9 and Opus
 - Removed experimental flag from the JPEG 2000 encoder
 version 2.7:
 - FFT video filter
 - TDSC decoder
 - DTS lossless extension (XLL) decoding (not lossless, disabled by default)
 - showwavespic filter
 - DTS decoding through libdcadec
 - Drop support for nvenc API before 5.0
 - nvenc HEVC encoder
 - Detelecine filter
 - Intel QSV-accelerated H.264 encoding
 - MMAL-accelerated H.264 decoding
 - basic APNG encoder and muxer with default extension "apng"
 - unpack DivX-style packed B-frames in MPEG-4 bitstream filter
 - WebM Live Chunk Muxer
 - nvenc level and tier options
 - chorus filter
 - Canopus HQ/HQA decoder
 - Automatically rotate videos based on metadata in ffmpeg
 - improved Quickdraw compatibility
 - VP9 high bit-depth and extended colorspaces decoding support
 - WebPAnimEncoder API when available for encoding and muxing WebP
 - Direct3D11-accelerated decoding
 - Support Secure Transport
 - Multipart JPEG demuxer
 version 2.6:
 - nvenc encoder
 - 10bit spp filter
 - colorlevels filter
 - RIFX format for *.wav files
 - RTP/mpegts muxer
 - non continuous cache protocol support
 - tblend filter
 - cropdetect support for non 8bpp, absolute (if limit >= 1) and relative (if limit < 1.0) threshold
 - Camellia symmetric block cipher
 - OpenH264 encoder wrapper
 - VOC seeking support
 - Closed caption Decoder
 - fspp, uspp, pp7 MPlayer postprocessing filters ported to native filters
 - showpalette filter
 - Twofish symmetric block cipher
 - Support DNx100 (960x720@8)
 - eq2 filter ported from libmpcodecs as eq filter
 - removed libmpcodecs
 - Changed default DNxHD colour range in QuickTime .mov derivatives to mpeg range
 - ported softpulldown filter from libmpcodecs as repeatfields filter
 - dcshift filter
 - RTP depacketizer for loss tolerant payload format for MP3 audio (RFC 5219)
 - RTP depacketizer for AC3 payload format (RFC 4184)
 - palettegen and paletteuse filters
 - VP9 RTP payload format (draft 0) experimental depacketizer
 - RTP depacketizer for DV (RFC 6469)
 - DXVA2-accelerated HEVC decoding
 - AAC ELD 480 decoding
 - Intel QSV-accelerated H.264 decoding
 - DSS SP decoder and DSS demuxer
 - Fix stsd atom corruption in DNxHD QuickTimes
 - Canopus HQX decoder
 - RTP depacketization of T.140 text (RFC 4103)
 - Port MIPS optimizations to 64-bit
 version 2.5:
 - HEVC/H.265 RTP payload format (draft v6) packetizer
 - SUP/PGS subtitle demuxer
 - ffprobe -show_pixel_formats option
 - CAST128 symmetric block cipher, ECB mode
 - STL subtitle demuxer and decoder
 - libutvideo YUV 4:2:2 10bit support
 - XCB-based screen-grabber
 - UDP-Lite support (RFC 3828)
 - xBR scaling filter
 - AVFoundation screen capturing support
 - ffserver supports codec private options
 - creating DASH compatible fragmented MP4, MPEG-DASH segmenting muxer
 - WebP muxer with animated WebP support
 - zygoaudio decoding support
 - APNG demuxer
 - postproc visualization support
 version 2.4:
 - Icecast protocol
 - ported lenscorrection filter from frei0r filter
 - large optimizations in dctdnoiz to make it usable
 - ICY metadata are now requested by default with the HTTP protocol
 - support for using metadata in stream specifiers in fftools
 - LZMA compression support in TIFF decoder
 - H.261 RTP payload format (RFC 4587) depacketizer and experimental packetizer
 - HEVC/H.265 RTP payload format (draft v6) depacketizer
 - added codecview filter to visualize information exported by some codecs
 - Matroska 3D support thorugh side data
 - HTML generation using texi2html is deprecated in favor of makeinfo/texi2any
 - silenceremove filter
 version 2.3:
 - AC3 fixed-point decoding
@@ -780,7 +48,7 @@ version 2.3:
 - libbs2b-based stereo-to-binaural audio filter
 - libx264 reference frames count limiting depending on level
 - native Opus decoder
- display matrix export and rotation API
+- display matrix export and rotation api
 - WebVTT encoder
 - showcqt multimedia filter
 - zoompan filter
@@ -822,7 +90,6 @@ version 2.2:
 - libx265 encoder
 - dejudder filter
 - Autodetect VDA like all other hardware accelerations
 - aliases and defaults for Ogg subtypes (opus, spx)
 version 2.1:
@@ -1229,7 +496,7 @@ easier to use. The changes are:
          all the stream in the first input file, except for the second audio
          stream'.
    * There is a new option -c (or -codec) for choosing the decoder/encoder to
-      use, which makes it possible to precisely specify target stream(s) consistently with
+      use, which allows to precisely specify target stream(s) consistently with
      other options. E.g. -c:v lib264 sets the codec for all video streams, -c:a:0
      libvorbis sets the codec for the first audio stream and -c copy copies all
      the streams without reencoding. Old -vcodec/-acodec/-scodec options are now
@@ -1418,8 +685,8 @@ version 0.8:
 - showinfo filter added
 - SMPTE 302M AES3 audio decoder
 - Apple Core Audio Format muxer
- 9 bits and 10 bits per sample support in the H.264 decoder
+- 9bit and 10bit per sample support in the H.264 decoder
- 9 bits and 10 bits FFV1 encoding / decoding
+- 9bit and 10bit FFV1 encoding / decoding
 - split filter added
 - select filter added
 - sdl output device added
@@ -1712,7 +979,7 @@ version 0.4.9-pre1:
 - rate distorted optimal lambda->qp support
 - AAC encoding with libfaac
 - Sunplus JPEG codec (SP5X) support
- use Lagrange multiplier instead of QP for ratecontrol
+- use Lagrange multipler instead of QP for ratecontrol
 - Theora/VP3 decoding support
 - XA and ADX ADPCM codecs
 - export MPEG-2 active display area / pan scan
--- a/INSTALL.md
+++ b/INSTALL.md
@@ -1,4 +1,4 @@
-## Installing FFmpeg
+#Installing FFmpeg:
 1. Type `./configure` to create the configuration. A list of configure
 options is printed by running `configure --help`.
--- a/LICENSE.md
+++ b/LICENSE.md
@@ -1,129 +1,103 @@
-# License
+#FFmpeg:
 Most files in FFmpeg are under the GNU Lesser General Public License version 2.1
-or later (LGPL v2.1+). Read the file `COPYING.LGPLv2.1` for details. Some other
+or later (LGPL v2.1+). Read the file COPYING.LGPLv2.1 for details. Some other
 files have MIT/X11/BSD-style licenses. In combination the LGPL v2.1+ applies to
 FFmpeg.
 Some optional parts of FFmpeg are licensed under the GNU General Public License
-version 2 or later (GPL v2+). See the file `COPYING.GPLv2` for details. None of
+version 2 or later (GPL v2+). See the file COPYING.GPLv2 for details. None of
-these parts are used by default, you have to explicitly pass `--enable-gpl` to
+these parts are used by default, you have to explicitly pass --enable-gpl to
 configure to activate them. In this case, FFmpeg's license changes to GPL v2+.
 Specifically, the GPL parts of FFmpeg are:
 - libpostproc
- optional x86 optimization in the files
+- libmpcodecs
-    - `libavcodec/x86/flac_dsp_gpl.asm`
+- optional x86 optimizations in the files
-    - `libavcodec/x86/idct_mmx.c`
+  libavcodec/x86/idct_mmx.c
-    - `libavfilter/x86/vf_removegrain.asm`
+- libutvideo encoding/decoding wrappers in
- the following building and testing tools
+  libavcodec/libutvideo*.cpp
-    - `compat/solaris/make_sunver.pl`
+- the X11 grabber in libavdevice/x11grab.c
-    - `doc/t2h.pm`
+- the swresample test app in
-    - `doc/texi2pod.pl`
+  libswresample/swresample-test.c
-    - `libswresample/tests/swresample.c`
+- the texi2pod.pl tool
    - `tests/checkasm/*`
    - `tests/tiny_ssim.c`
 - the following filters in libavfilter:
-    - `signature_lookup.c`
+    - f_ebur128.c
-    - `vf_blackframe.c`
+    - vf_blackframe.c
-    - `vf_boxblur.c`
+    - vf_boxblur.c
-    - `vf_colormatrix.c`
+    - vf_colormatrix.c
-    - `vf_cover_rect.c`
+    - vf_cropdetect.c
-    - `vf_cropdetect.c`
+    - vf_decimate.c
-    - `vf_delogo.c`
+    - vf_delogo.c
-    - `vf_eq.c`
+    - vf_geq.c
-    - `vf_find_rect.c`
+    - vf_histeq.c
-    - `vf_fspp.c`
+    - vf_hqdn3d.c
-    - `vf_histeq.c`
+    - vf_kerndeint.c
-    - `vf_hqdn3d.c`
+    - vf_mcdeint.c
-    - `vf_kerndeint.c`
+    - vf_mp.c
-    - `vf_lensfun.c` (GPL version 3 or later)
+    - vf_owdenoise.c
-    - `vf_mcdeint.c`
+    - vf_perspective.c
-    - `vf_mpdecimate.c`
+    - vf_phase.c
-    - `vf_nnedi.c`
+    - vf_pp.c
-    - `vf_owdenoise.c`
+    - vf_pullup.c
-    - `vf_perspective.c`
+    - vf_sab.c
-    - `vf_phase.c`
+    - vf_smartblur.c
-    - `vf_pp.c`
+    - vf_spp.c
-    - `vf_pp7.c`
+    - vf_stereo3d.c
-    - `vf_pullup.c`
+    - vf_super2xsai.c
-    - `vf_repeatfields.c`
+    - vf_tinterlace.c
-    - `vf_sab.c`
+    - vsrc_mptestsrc.c
    - `vf_signature.c`
    - `vf_smartblur.c`
    - `vf_spp.c`
    - `vf_stereo3d.c`
    - `vf_super2xsai.c`
    - `vf_tinterlace.c`
    - `vf_uspp.c`
    - `vf_vaguedenoiser.c`
    - `vsrc_mptestsrc.c`
 Should you, for whatever reason, prefer to use version 3 of the (L)GPL, then
-the configure parameter `--enable-version3` will activate this licensing option
+the configure parameter --enable-version3 will activate this licensing option
-for you. Read the file `COPYING.LGPLv3` or, if you have enabled GPL parts,
+for you. Read the file COPYING.LGPLv3 or, if you have enabled GPL parts,
-`COPYING.GPLv3` to learn the exact legal terms that apply in this case.
+COPYING.GPLv3 to learn the exact legal terms that apply in this case.
 There are a handful of files under other licensing terms, namely:
-* The files `libavcodec/jfdctfst.c`, `libavcodec/jfdctint_template.c` and
+* The files libavcodec/jfdctfst.c, libavcodec/jfdctint_template.c and
-  `libavcodec/jrevdct.c` are taken from libjpeg, see the top of the files for
+  libavcodec/jrevdct.c are taken from libjpeg, see the top of the files for
  licensing details. Specifically note that you must credit the IJG in the
  documentation accompanying your program if you only distribute executables.
  You must also indicate any changes including additions and deletions to
  those three files in the documentation.
 * `tests/reference.pnm` is under the expat license.
-## External libraries
+external libraries
 ==================
 FFmpeg can be combined with a number of external libraries, which sometimes
 affect the licensing of binaries resulting from the combination.
-### Compatible libraries
+compatible libraries
-
+--------------------
 The following libraries are under GPL version 2:
 - avisynth
 - frei0r
 - libcdio
 - libdavs2
 - librubberband
 - libvidstab
 - libx264
 - libx265
 - libxavs
 - libxavs2
 - libxvid
 The following libraries are under GPL:
    - frei0r
    - libcdio
    - libutvideo
    - libvidstab
    - libx264
    - libx265
    - libxavs
    - libxvid
 When combining them with FFmpeg, FFmpeg needs to be licensed as GPL as well by
-passing `--enable-gpl` to configure.
+passing --enable-gpl to configure.
-The following libraries are under LGPL version 3:
+The OpenCORE and VisualOn libraries are under the Apache License 2.0. That
- gmp
+license is incompatible with the LGPL v2.1 and the GPL v2, but not with
 - libaribb24
 - liblensfun
 When combining them with FFmpeg, use the configure option `--enable-version3` to
 upgrade FFmpeg to the LGPL v3.
 The VMAF, mbedTLS, RK MPI, OpenCORE and VisualOn libraries are under the Apache License
 2.0. That license is incompatible with the LGPL v2.1 and the GPL v2, but not with
 version 3 of those licenses. So to combine these libraries with FFmpeg, the
-license version needs to be upgraded by passing `--enable-version3` to configure.
+license version needs to be upgraded by passing --enable-version3 to configure.
-The smbclient library is under the GPL v3, to combine it with FFmpeg,
+incompatible libraries
-the options `--enable-gpl` and `--enable-version3` have to be passed to
+----------------------
 configure to upgrade FFmpeg to the GPL v3.
-### Incompatible libraries
+The Fraunhofer AAC library, FAAC and aacplus are under licenses which
-
+are incompatible with the GPLv2 and v3. We do not know for certain if their
-There are certain libraries you can combine with FFmpeg whose licenses are not
+licenses are compatible with the LGPL.
-compatible with the GPL and/or the LGPL. If you wish to enable these
+If you wish to enable these libraries, pass --enable-nonfree to configure.
-libraries, even in circumstances that their license may be incompatible, pass
+But note that if you enable any of these libraries the resulting binary will
-`--enable-nonfree` to configure. This will cause the resulting binary to be
+be under a complex license mix that is more restrictive than the LGPL and that
-unredistributable.
+may result in additional obligations. It is possible that these
-
+restrictions cause the resulting binary to be unredistributeable.
 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.
--- a/318
+++ b/318
@@ -14,6 +14,7 @@ patches and related discussions.
 Project Leader
 ==============
 Michael Niedermayer
  final design decisions
@@ -29,6 +30,9 @@ ffplay:
 ffprobe:
  ffprobe.c                             Stefano Sabatini
 ffserver:
  ffserver.c                            Reynaldo H. Verdejo Pinochet
 Commandline utility code:
  cmdutils.c, cmdutils.h                Michael Niedermayer
@@ -39,25 +43,24 @@ QuickTime faststart:
 Miscellaneous Areas
 ===================
-documentation                           Stefano Sabatini, Mike Melanson, Timothy Gu, Gyan Doshi
+documentation                           Stefano Sabatini, Mike Melanson, Timothy Gu
-project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser, Nikolay Aleksandrov
+build system (configure, makefiles)     Diego Biurrun, Mans Rullgard
 project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser
 presets                                 Robert Swain
 metadata subsystem                      Aurelien Jacobs
 release management                      Michael Niedermayer
 API tests                               Ludmila Glinskih
 Communication
 =============
-website                                 Deby Barbara Lepage
+website                                 Robert Swain
 fate.ffmpeg.org                         Timothy Gu
-Trac bug tracker                        Alexander Strasser, Michael Niedermayer, Carl Eugen Hoyos
+Trac bug tracker                        Alexander Strasser, Michael Niedermayer, Carl Eugen Hoyos, Lou Logan
-Patchwork                               Andriy Gelman
+mailing lists                           Michael Niedermayer, Baptiste Coudurier, Lou Logan
-mailing lists                           Baptiste Coudurier
+Google+                                 Paul B Mahol, Michael Niedermayer, Alexander Strasser
-Twitter                                 Reynaldo H. Verdejo Pinochet
+Twitter                                 Lou Logan
 Launchpad                               Timothy Gu
 ffmpeg-security                         Andreas Cadhalpun, Carl Eugen Hoyos, Clément Bœsch, Michael Niedermayer, Reimar Doeffinger, rcombs, wm4
 libavutil
@@ -69,7 +72,6 @@ Internal Interfaces:
  libavutil/common.h                    Michael Niedermayer
 Other:
  aes_ctr.c, aes_ctr.h                  Eran Kornblau
  bprint                                Nicolas George
  bswap.h
  des                                   Reimar Doeffinger
@@ -77,8 +79,6 @@ Other:
  eval.c, eval.h                        Michael Niedermayer
  float_dsp                             Loren Merritt
  hash                                  Reimar Doeffinger
  hwcontext_cuda*                       Timo Rothenpieler
  hwcontext_vulkan*                     Lynne
  intfloat*                             Michael Niedermayer
  integer.c, integer.h                  Michael Niedermayer
  lzo                                   Reimar Doeffinger
@@ -89,7 +89,7 @@ Other:
  rational.c, rational.h                Michael Niedermayer
  rc4                                   Reimar Doeffinger
  ripemd.c, ripemd.h                    James Almer
-  tx*                                   Lynne
+  timecode                              Clément Bœsch
 libavcodec
@@ -115,14 +115,15 @@ Generic Parts:
    lzw.*                               Michael Niedermayer
  floating point AAN DCT:
    faandct.c, faandct.h                Michael Niedermayer
  Non-power-of-two MDCT:
    mdct15.c, mdct15.h                  Rostislav Pehlivanov
  Golomb coding:
    golomb.c, golomb.h                  Michael Niedermayer
  LPC:
    lpc.c, lpc.h                        Justin Ruggles
  motion estimation:
    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:
@@ -137,44 +138,43 @@ Codecs:
  4xm.c                                 Michael Niedermayer
  8bps.c                                Roberto Togni
  8svx.c                                Jaikrishnan Menon
-  aacenc*, aaccoder.c                   Rostislav Pehlivanov
+  aasc.c                                Kostya Shishkov
-  adpcm.c                               Zane van Iperen
+  ac3*                                  Justin Ruggles
  alacenc.c                             Jaikrishnan Menon
-  alsdec.c                              Thilo Borgmann, Umair Khan
+  alsdec.c                              Thilo Borgmann
-  aptx.c                                Aurelien Jacobs
+  apedec.c                              Kostya Shishkov
  ass*                                  Aurelien Jacobs
  asv*                                  Michael Niedermayer
  atrac3*                               Benjamin Larsson
  atrac3plus*                           Maxim Poliakovski
  audiotoolbox*                         rcombs
  avs2*                                 Huiwen Ren
  bgmc.c, bgmc.h                        Thilo Borgmann
  bink.c                                Kostya Shishkov
  binkaudio.c                           Peter Ross
  bmp.c                                 Mans Rullgard, Kostya Shishkov
  cavs*                                 Stefan Gehrer
  cdxl.c                                Paul B Mahol
  celp_filters.*                        Vitor Sessak
  cinepak.c                             Roberto Togni
  cinepakenc.c                          Rl / Aetey G.T. AB
  ccaption_dec.c                        Anshul Maheshwari, Aman Gupta
  cljr                                  Alex Beregszaszi
  cllc.c                                Derek Buitenhuis
  cook.c, cookdata.h                    Benjamin Larsson
  cpia.c                                Stephan Hilb
  crystalhd.c                           Philip Langdale
  cscd.c                                Reimar Doeffinger
-  cuviddec.c                            Timo Rothenpieler
+  dca.c                                 Kostya Shishkov, Benjamin Larsson
  dca*                                  foo86
  dfpwm*                                Jack Bruienne
  dirac*                                Rostislav Pehlivanov
  dnxhd*                                Baptiste Coudurier
  dolby_e*                              foo86
  dpcm.c                                Mike Melanson
  dss_sp.c                              Oleksij Rempel
  dv.c                                  Roman Shaposhnik
  dvbsubdec.c                           Anshul Maheshwari
  dxa.c                                 Kostya Shishkov
  eacmv*, eaidct*, eat*                 Peter Ross
  evrc*                                 Paul B Mahol
  exif.c, exif.h                        Thilo Borgmann
  ffv1*                                 Michael Niedermayer
  ffwavesynth.c                         Nicolas George
-  fifo.c                                Jan Sebechlebsky
+  fic.c                                 Derek Buitenhuis
  flac*                                 Justin Ruggles
  flashsv*                              Benjamin Larsson
  flicvideo.c                           Mike Melanson
  g722.c                                Martin Storsjo
  g726.c                                Roman Shaposhnik
@@ -182,53 +182,58 @@ Codecs:
  h261*                                 Michael Niedermayer
  h263*                                 Michael Niedermayer
  h264*                                 Loren Merritt, Michael Niedermayer
-  hap*                                  Tom Butterworth
+  huffyuv*                              Michael Niedermayer, Christophe Gisquet
  huffyuv*                              Michael Niedermayer
  idcinvideo.c                          Mike Melanson
  imc*                                  Benjamin Larsson
  indeo2*                               Kostya Shishkov
  indeo5*                               Kostya Shishkov
  interplayvideo.c                      Mike Melanson
-  jni*, ffjni*                          Matthieu Bouron
+  ivi*                                  Kostya Shishkov
  jacosub*                              Clément Bœsch
  jpeg2000*                             Nicolas Bertrand
  jpeg_ls.c                             Kostya Shishkov
  jvdec.c                               Peter Ross
  kmvc.c                                Kostya Shishkov
  lcl*.c                                Roberto Togni, Reimar Doeffinger
  libcelt_dec.c                         Nicolas George
  libcodec2.c                           Tomas Härdin
  libdirac*                             David Conrad
  libdavs2.c                            Huiwen Ren
  libjxl*.c, libjxl.h                   Leo Izen
  libgsm.c                              Michel Bardiaux
  libkvazaar.c                          Arttu Ylä-Outinen
  libopenh264enc.c                      Martin Storsjo, Linjie Fu
  libopenjpeg.c                         Jaikrishnan Menon
  libopenjpegenc.c                      Michael Bradshaw
  libschroedinger*                      David Conrad
  libspeexdec.c                         Justin Ruggles
  libtheoraenc.c                        David Conrad
  libutvideo*                           Derek Buitenhuis
  libvorbis.c                           David Conrad
  libvpx*                               James Zern
  libx264.c                             Mans Rullgard, Jason Garrett-Glaser
  libx265.c                             Derek Buitenhuis
  libxavs.c                             Stefan Gehrer
  libxavs2.c                            Huiwen Ren
  libzvbi-teletextdec.c                 Marton Balint
  loco.c                                Kostya Shishkov
  lzo.h, lzo.c                          Reimar Doeffinger
  mdec.c                                Michael Niedermayer
  mimic.c                               Ramiro Polla
  mjpeg*.c                              Michael Niedermayer
-  mlp*                                  Ramiro Polla, Jai Luthra
+  mlp*                                  Ramiro Polla
  mmvideo.c                             Peter Ross
  mpc*                                  Kostya Shishkov
  mpeg12.c, mpeg12data.h                Michael Niedermayer
  mpegvideo.c, mpegvideo.h              Michael Niedermayer
  mqc*                                  Nicolas Bertrand
  msmpeg4.c, msmpeg4data.h              Michael Niedermayer
  msrle.c                               Mike Melanson
  msvideo1.c                            Mike Melanson
  nellymoserdec.c                       Benjamin Larsson
  nuv.c                                 Reimar Doeffinger
  nvdec*, nvenc*                        Timo Rothenpieler
  omx.c                                 Martin Storsjo, Aman Gupta
  opus*                                 Rostislav Pehlivanov
  paf.*                                 Paul B Mahol
  pcx.c                                 Ivo van Poorten
  pgssubdec.c                           Reimar Doeffinger
  ptx.c                                 Ivo van Poorten
  qcelp*                                Reynaldo H. Verdejo Pinochet
-  qdm2.c, qdm2data.h                    Roberto Togni
+  qdm2.c, qdm2data.h                    Roberto Togni, Benjamin Larsson
-  qsv*                                  Mark Thompson, Zhong Li, Haihao Xiang
+  qdrw.c                                Kostya Shishkov
  qpeg.c                                Kostya Shishkov
  qtrle.c                               Mike Melanson
  ra144.c, ra144.h, ra288.c, ra288.h    Roberto Togni
  resample2.c                           Michael Niedermayer
@@ -236,49 +241,65 @@ Codecs:
  rpza.c                                Roberto Togni
  rtjpeg.c, rtjpeg.h                    Reimar Doeffinger
  rv10.c                                Michael Niedermayer
  rv3*                                  Kostya Shishkov
  rv4*                                  Kostya Shishkov, Christophe Gisquet
  s3tc*                                 Ivo van Poorten
  smacker.c                             Kostya Shishkov
  smc.c                                 Mike Melanson
  smvjpegdec.c                          Ash Hughes
  snow*                                 Michael Niedermayer, Loren Merritt
  sonic.c                               Alex Beregszaszi
  speedhq.c                             Steinar H. Gunderson
  srt*                                  Aurelien Jacobs
  sunrast.c                             Ivo van Poorten
  svq3.c                                Michael Niedermayer
  tak*                                  Paul B Mahol
  targa.c                               Kostya Shishkov
  tiff.c                                Kostya Shishkov
  truemotion1*                          Mike Melanson
  truemotion2*                          Kostya Shishkov
  truespeech.c                          Kostya Shishkov
  tscc.c                                Kostya Shishkov
  tta.c                                 Alex Beregszaszi, Jaikrishnan Menon
  ttaenc.c                              Paul B Mahol
  txd.c                                 Ivo van Poorten
-  v4l2_*                                Jorge Ramirez-Ortiz
+  ulti*                                 Kostya Shishkov
-  vc2*                                  Rostislav Pehlivanov
+  v410*.c                               Derek Buitenhuis
  vb.c                                  Kostya Shishkov
  vble.c                                Derek Buitenhuis
  vc1*                                  Kostya Shishkov, Christophe Gisquet
  vcr1.c                                Michael Niedermayer
-  videotoolboxenc.c                     Rick Kern, Aman Gupta
+  vda_h264_dec.c                        Xidorn Quan
  vima.c                                Paul B Mahol
  vmnc.c                                Kostya Shishkov
  vorbisdec.c                           Denes Balatoni, David Conrad
  vorbisenc.c                           Oded Shimon
  vp3*                                  Mike Melanson
  vp5                                   Aurelien Jacobs
  vp6                                   Aurelien Jacobs
-  vp8                                   David Conrad, Ronald Bultje
+  vp8                                   David Conrad, Jason Garrett-Glaser, Ronald Bultje
-  vp9                                   Ronald Bultje
+  vp9                                   Ronald Bultje, Clément Bœsch
  vqavideo.c                            Mike Melanson
  wavpack.c                             Kostya Shishkov
  wmaprodec.c                           Sascha Sommer
  wmavoice.c                            Ronald S. Bultje
  wmv2.c                                Michael Niedermayer
  wnv1.c                                Kostya Shishkov
  xan.c                                 Mike Melanson
  xbm*                                  Paul B Mahol
  xface                                 Stefano Sabatini
  xl.c                                  Kostya Shishkov
  xvmc.c                                Ivan Kalvachev
  xwd*                                  Paul B Mahol
  zerocodec.c                           Derek Buitenhuis
  zmbv*                                 Kostya Shishkov
 Hardware acceleration:
  crystalhd.c                           Philip Langdale
-  dxva2*                                Hendrik Leppkes, Laurent Aimar, Steve Lhomme
+  dxva2*                                Laurent Aimar
-  d3d11va*                              Steve Lhomme
+  libstagefright.cpp                    Mohamed Naufal
-  mediacodec*                           Matthieu Bouron, Aman Gupta
+  vaapi*                                Gwenole Beauchesne
-  vaapi*                                Haihao Xiang
+  vda*                                  Sebastien Zwickert
-  vaapi_encode*                         Mark Thompson, Haihao Xiang
+  vdpau*                                Carl Eugen Hoyos
  vdpau*                                Philip Langdale, Carl Eugen Hoyos
  videotoolbox*                         Rick Kern, Aman Gupta
 libavdevice
@@ -288,8 +309,6 @@ 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
  gdigrab.c                             Roger Pack (CC rogerdpack@gmail.com)
@@ -298,9 +317,9 @@ libavdevice
  libdc1394.c                           Roman Shaposhnik
  opengl_enc.c                          Lukasz Marek
  pulse_audio_enc.c                     Lukasz Marek
  qtkit.m                               Thilo Borgmann
  sdl                                   Stefano Sabatini
-  sdl2.c                                Josh de Kock
+  v4l2.c                                Luca Abeni
  v4l2.c                                Giorgio Vazzana
  vfwcap.c                              Ramiro Polla
  xv.c                                  Lukasz Marek
@@ -310,10 +329,7 @@ libavfilter
 Generic parts:
  graphdump.c                           Nicolas George
  motion_estimation.c                   Davinder Singh
 Filters:
  f_drawgraph.c                         Paul B Mahol
  af_adelay.c                           Paul B Mahol
  af_aecho.c                            Paul B Mahol
  af_afade.c                            Paul B Mahol
@@ -321,62 +337,34 @@ Filters:
  af_aphaser.c                          Paul B Mahol
  af_aresample.c                        Michael Niedermayer
  af_astats.c                           Paul B Mahol
  af_astreamsync.c                      Nicolas George
  af_atempo.c                           Pavel Koshevoy
  af_biquads.c                          Paul B Mahol
  af_chorus.c                           Paul B Mahol
  af_compand.c                          Paul B Mahol
  af_firequalizer.c                     Muhammad Faiz
  af_hdcd.c                             Burt P.
  af_ladspa.c                           Paul B Mahol
  af_loudnorm.c                         Kyle Swanson
  af_pan.c                              Nicolas George
  af_sidechaincompress.c                Paul B Mahol
  af_silenceremove.c                    Paul B Mahol
  avf_aphasemeter.c                     Paul B Mahol
  avf_avectorscope.c                    Paul B Mahol
  avf_showcqt.c                         Muhammad Faiz
  vf_blend.c                            Paul B Mahol
  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
  vf_coreimage.m                        Thilo Borgmann
  vf_deband.c                           Paul B Mahol
  vf_dejudder.c                         Nicholas Robbins
-  vf_delogo.c                           Jean Delvare (CC <jdelvare@suse.com>)
+  vf_delogo.c                           Jean Delvare (CC <khali@linux-fr.org>)
  vf_drawbox.c/drawgrid                 Andrey Utkin
  vf_extractplanes.c                    Paul B Mahol
  vf_histogram.c                        Paul B Mahol
  vf_hqx.c                              Clément Bœsch
  vf_idet.c                             Pascal Massimino
  vf_il.c                               Paul B Mahol
  vf_(t)interlace                       Thomas Mundt (CC <thomas.mundt@hr.de>)
  vf_lenscorrection.c                   Daniel Oberhoff
  vf_libplacebo.c                       Niklas Haas
  vf_mergeplanes.c                      Paul B Mahol
  vf_mestimate.c                        Davinder Singh
  vf_minterpolate.c                     Davinder Singh
  vf_neighbor.c                         Paul B Mahol
  vf_psnr.c                             Paul B Mahol
  vf_random.c                           Paul B Mahol
  vf_readvitc.c                         Tobias Rapp (CC t.rapp at noa-archive dot com)
  vf_scale.c                            Michael Niedermayer
  vf_separatefields.c                   Paul B Mahol
  vf_ssim.c                             Paul B Mahol
  vf_stereo3d.c                         Paul B Mahol
  vf_telecine.c                         Paul B Mahol
  vf_tonemap_opencl.c                   Ruiling Song
  vf_yadif.c                            Michael Niedermayer
  vf_zoompan.c                          Paul B Mahol
 Sources:
  vsrc_mandelbrot.c                     Michael Niedermayer
 dnn                                     Yejun Guo
 libavformat
 ===========
@@ -385,53 +373,39 @@ Generic parts:
    libavformat/avformat.h              Michael Niedermayer
  Utility Code:
    libavformat/utils.c                 Michael Niedermayer
  Text Subtitles                        Clément Bœsch
 Muxers/Demuxers:
  4xm.c                                 Mike Melanson
  aadec.c                               Vesselin Bontchev (vesselin.bontchev at yandex dot com)
  adtsenc.c                             Robert Swain
  afc.c                                 Paul B Mahol
  aiffdec.c                             Baptiste Coudurier, Matthieu Bouron
  aiffenc.c                             Baptiste Coudurier, Matthieu Bouron
-  alp.c                                 Zane van Iperen
+  ape.c                                 Kostya Shishkov
  amvenc.c                              Zane van Iperen
  apm.c                                 Zane van Iperen
  apngdec.c                             Benoit Fouet
  argo_asf.c                            Zane van Iperen
  argo_brp.c                            Zane van Iperen
  argo_cvg.c                            Zane van Iperen
  ass*                                  Aurelien Jacobs
  astdec.c                              Paul B Mahol
  astenc.c                              James Almer
  avi*                                  Michael Niedermayer
-  avisynth.c                            Stephen Hutchinson
+  avisynth.c                            AvxSynth Team (avxsynth.testing at gmail dot com)
  avr.c                                 Paul B Mahol
  bink.c                                Peter Ross
  boadec.c                              Michael Niedermayer
  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
  dfpwmdec.c                            Jack Bruienne
  dss.c                                 Oleksij Rempel
  dtsdec.c                              foo86
  dtshddec.c                            Paul B Mahol
  dv.c                                  Roman Shaposhnik
  dxa.c                                 Kostya Shishkov
  electronicarts.c                      Peter Ross
  epafdec.c                             Paul B Mahol
  ffm*                                  Baptiste Coudurier
  flac*                                 Justin Ruggles
  flic.c                                Mike Melanson
-  flvdec.c                              Michael Niedermayer
+  flvdec.c, flvenc.c                    Michael Niedermayer
  flvenc.c                              Michael Niedermayer, Steven Liu
  gxf.c                                 Reimar Doeffinger
  gxfenc.c                              Baptiste Coudurier
-  hlsenc.c                              Christian Suloway, Steven Liu
+  hls.c                                 Anssi Hannula
  idcin.c                               Mike Melanson
  idroqdec.c                            Mike Melanson
  iff.c                                 Jaikrishnan Menon
@@ -439,32 +413,32 @@ Muxers/Demuxers:
  ipmovie.c                             Mike Melanson
  ircam*                                Paul B Mahol
  iss.c                                 Stefan Gehrer
-  jpegxl_probe.*                        Leo Izen
+  jacosub*                              Clément Bœsch
  jvdec.c                               Peter Ross
  kvag.c                                Zane van Iperen
  libmodplug.c                          Clément Bœsch
-  libopenmpt.c                          Josh de Kock
+  libnut.c                              Oded Shimon
  lmlm4.c                               Ivo van Poorten
  lvfdec.c                              Paul B Mahol
  lxfdec.c                              Tomas Härdin
-  matroska.c                            Aurelien Jacobs, Andreas Rheinhardt
+  matroska.c                            Aurelien Jacobs
-  matroskadec.c                         Aurelien Jacobs, Andreas Rheinhardt
+  matroskadec.c                         Aurelien Jacobs
-  matroskaenc.c                         David Conrad, Andreas Rheinhardt
+  matroskaenc.c                         David Conrad
  matroska subtitles (matroskaenc.c)    John Peebles
  metadata*                             Aurelien Jacobs
  mgsts.c                               Paul B Mahol
  microdvd*                             Aurelien Jacobs
  mm.c                                  Peter Ross
-  mov.c                                 Baptiste Coudurier
+  mov.c                                 Michael Niedermayer, Baptiste Coudurier
  movenc.c                              Baptiste Coudurier, Matthieu Bouron
-  movenccenc.c                          Eran Kornblau
+  mpc.c                                 Kostya Shishkov
  mpeg.c                                Michael Niedermayer
  mpegenc.c                             Michael Niedermayer
  mpegts.c                              Marton Balint
  mpegtsenc.c                           Baptiste Coudurier
  msnwc_tcp.c                           Ramiro Polla
  mtv.c                                 Reynaldo H. Verdejo Pinochet
-  mxf*                                  Baptiste Coudurier, Tomas Härdin
+  mxf*                                  Baptiste Coudurier
  mxfdec.c                              Tomas Härdin
  nistspheredec.c                       Paul B Mahol
  nsvdec.c                              Francois Revol
  nut*                                  Michael Niedermayer
@@ -474,7 +448,6 @@ Muxers/Demuxers:
  oggparse*.c                           David Conrad
  oma.c                                 Maxim Poliakovski
  paf.c                                 Paul B Mahol
  pp_bnk.c                              Zane van Iperen
  psxstr.c                              Mike Melanson
  pva.c                                 Ivo van Poorten
  pvfdec.c                              Paul B Mahol
@@ -482,22 +455,17 @@ Muxers/Demuxers:
  raw.c                                 Michael Niedermayer
  rdt.c                                 Ronald S. Bultje
  rl2.c                                 Sascha Sommer
-  rmdec.c, rmenc.c                      Ronald S. Bultje
+  rmdec.c, rmenc.c                      Ronald S. Bultje, Kostya Shishkov
  rtmp*                                 Kostya Shishkov
  rtp.c, rtpenc.c                       Martin Storsjo
  rtpdec_ac3.*                          Gilles Chanteperdrix
  rtpdec_dv.*                           Thomas Volkert
  rtpdec_h261.*, rtpenc_h261.*          Thomas Volkert
  rtpdec_hevc.*, rtpenc_hevc.*          Thomas Volkert
  rtpdec_mpa_robust.*                   Gilles Chanteperdrix
  rtpdec_asf.*                          Ronald S. Bultje
  rtpdec_vc2hq.*, rtpenc_vc2hq.*        Thomas Volkert
  rtpdec_vp9.c                          Thomas Volkert
  rtpenc_mpv.*, rtpenc_aac.*            Martin Storsjo
-  s337m.c                               foo86
+  rtsp.c                                Luca Barbato
  sbgdec.c                              Nicolas George
  sdp.c                                 Martin Storsjo
  segafilm.c                            Mike Melanson
-  segment.c                             Stefano Sabatini
+  siff.c                                Kostya Shishkov
  smacker.c                             Kostya Shishkov
  smjpeg*                               Paul B Mahol
  spdif*                                Anssi Hannula
  srtdec.c                              Aurelien Jacobs
@@ -512,19 +480,16 @@ Muxers/Demuxers:
  webvtt*                               Matthew J Heaney
  westwood.c                            Mike Melanson
  wtv.c                                 Peter Ross
  wv.c                                  Kostya Shishkov
  wvenc.c                               Paul B Mahol
 Protocols:
  async.c                               Zhang Rui
  bluray.c                              Petri Hintukainen
  ftp.c                                 Lukasz Marek
  http.c                                Ronald S. Bultje
  libsrt.c                              Zhao Zhili
  libssh.c                              Lukasz Marek
  libzmq.c                              Andriy Gelman
  mms*.c                                Ronald S. Bultje
  udp.c                                 Luca Abeni
  icecast.c                             Marvin Scholz
 libswresample
@@ -544,69 +509,38 @@ Resamplers:
 Operating systems / CPU architectures
 =====================================
-Alpha                                   Falk Hueffner
+Alpha                                   Mans Rullgard, Falk Hueffner
-MIPS                                    Manojkumar Bhosale, Shiyou Yin
+ARM                                     Mans Rullgard
-LoongArch                               Shiyou Yin
+AVR32                                   Mans Rullgard
 MIPS                                    Mans Rullgard, Nedeljko Babic
 Mac OS X / PowerPC                      Romain Dolbeau, Guillaume Poirier
 Amiga / PowerPC                         Colin Ward
-Linux / PowerPC                         Lauri Kasanen
+Linux / PowerPC                         Luca Barbato
 Windows MinGW                           Alex Beregszaszi, Ramiro Polla
 Windows Cygwin                          Victor Paesa
-Windows MSVC                            Matthew Oliver, Hendrik Leppkes
+Windows MSVC                            Matthew Oliver
 Windows ICL                             Matthew Oliver
 ADI/Blackfin DSP                        Marc Hoffman
 Sparc                                   Roman Shaposhnik
-OS/2                                    KO Myung-Hun
+x86                                     Michael Niedermayer
 Developers with git write access who are currently not maintaining any specific part
 ====================================================================================
 Alex Converse
 Andreas Cadhalpun
 Anuradha Suraparaju
 Ben Littler
 Benjamin Larsson
 Bobby Bingham
 Daniel Verkamp
 Derek Buitenhuis
 Ganesh Ajjanagadde
 Henrik Gramner
 Ivan Uskov
 James Darnley
 Jan Ekström
 Joakim Plate
 Jun Zhao
 Kieran Kunhya
 Kirill Gavrilov
 Limin Wang
 Martin Storsjö
 Panagiotis Issaris
 Pedro Arthur
 Sebastien Zwickert
 Vittorio Giovara
 wm4
 (this list is incomplete)
 Releases
 ========
-2.8                                     Michael Niedermayer
+2.3                                     Michael Niedermayer
-2.7                                     Michael Niedermayer
+2.2                                     Michael Niedermayer
-2.6                                     Michael Niedermayer
+1.2                                     Michael Niedermayer
 2.5                                     Michael Niedermayer
 If you want to maintain an older release, please contact us
-GnuPG Fingerprints and IRC nicknames of maintainers and contributors
+GnuPG Fingerprints of maintainers and contributors
-====================================================================
+==================================================
 IRC nicknames are in parentheses. These apply
 to the IRC channels listed on the website.
 Alexander Strasser            1C96 78B7 83CB 8AA7 9AF5 D1EB A7D8 A57B A876 E58F
 Anssi Hannula                 1A92 FF42 2DD9 8D2E 8AF7 65A9 4278 C520 513D F3CB
 Anton Khirnov                 6D0C 6625 56F8 65D1 E5F5 814B B50A 1241 C067 07AB
 Ash Hughes                    694D 43D2 D180 C7C7 6421 ABD3 A641 D0B7 623D 6029
 Attila Kinali                 11F0 F9A6 A1D2 11F6 C745 D10C 6520 BCDD F2DF E765
 Baptiste Coudurier            8D77 134D 20CC 9220 201F C5DB 0AC9 325C 5C1A BAAA
@@ -614,35 +548,27 @@ Ben Littler                   3EE3 3723 E560 3214 A8CD 4DEB 2CDB FCE7 768C 8D2C
 Benoit Fouet                  B22A 4F4F 43EF 636B BB66 FCDC 0023 AE1E 2985 49C8
 Clément Bœsch                 52D0 3A82 D445 F194 DB8B 2B16 87EE 2CB8 F4B8 FCF9
 Daniel Verkamp                78A6 07ED 782C 653E C628 B8B9 F0EB 8DD8 2F0E 21C7
 Diego Biurrun                 8227 1E31 B6D9 4994 7427 E220 9CAE D6CC 4757 FCC5
 FFmpeg release signing key    FCF9 86EA 15E6 E293 A564 4F10 B432 2F04 D676 58D8
 Ganesh Ajjanagadde            C96A 848E 97C3 CEA2 AB72 5CE4 45F9 6A2D 3C36 FB1B
 Gwenole Beauchesne            2E63 B3A6 3E44 37E2 017D 2704 53C7 6266 B153 99C4
 Haihao Xiang (haihao)         1F0C 31E8 B4FE F7A4 4DC1 DC99 E0F5 76D4 76FC 437F
 Jaikrishnan Menon             61A1 F09F 01C9 2D45 78E1 C862 25DC 8831 AF70 D368
 James Almer                   7751 2E8C FD94 A169 57E6 9A7A 1463 01AD 7376 59E0
 Jean Delvare                  7CA6 9F44 60F1 BDC4 1FD2 C858 A552 6B9B B3CD 4E6A
-Leo Izen (thebombzen)         B6FD 3CFC 7ACF 83FC 9137 6945 5A71 C331 FD2F A19A
+Justin Ruggles                3136 ECC0 C10D 6C04 5F43 CA29 FCBE CD2A 3787 1EBF
 Loren Merritt                 ABD9 08F4 C920 3F65 D8BE 35D7 1540 DAA7 060F 56DE
-Lynne                         FE50 139C 6805 72CA FD52 1F8D A2FE A5F0 3F03 4464
+Lou Logan                     7D68 DC73 CBEF EABB 671A B6CF 621C 2E28 82F8 DC3A
 Luca Barbato                  6677 4209 213C 8843 5B67 29E7 E84C 78C2 84E9 0E34
 Michael Niedermayer           9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
 Nicolas George                24CE 01CE 9ACC 5CEB 74D8 8D9D B063 D997 36E5 4C93
 Niklas Haas (haasn)           1DDB 8076 B14D 5B48 32FC 99D9 EB52 DA9C 02BA 6FB4
 Nikolay Aleksandrov           8978 1D8C FB71 588E 4B27 EAA8 C4F0 B5FC E011 13B1
 Panagiotis Issaris            6571 13A3 33D9 3726 F728 AA98 F643 B12E ECF3 E029
 Peter Ross                    A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B
 Philip Langdale               5DC5 8D66 5FBA 3A43 18EC 045E F8D6 B194 6A75 682E
 Ramiro Polla                  7859 C65B 751B 1179 792E DAE8 8E95 8B2F 9B6C 5700
 Reimar Doeffinger             C61D 16E5 9E2C D10C 8958 38A4 0899 A2B9 06D4 D9C7
 Reinhard Tartler              9300 5DC2 7E87 6C37 ED7B CA9A 9808 3544 9453 48A4
 Reynaldo H. Verdejo Pinochet  6E27 CD34 170C C78E 4D4F 5F40 C18E 077F 3114 452A
 Robert Swain                  EE7A 56EA 4A81 A7B5 2001 A521 67FA 362D A2FC 3E71
 Sascha Sommer                 38A0 F88B 868E 9D3A 97D4 D6A0 E823 706F 1E07 0D3C
 Stefano Sabatini              0D0B AD6B 5330 BBAD D3D6 6A0C 719C 2839 FC43 2D5F
 Steinar H. Gunderson          C2E9 004F F028 C18E 4EAD DB83 7F61 7561 7797 8F76
 Stephan Hilb                  4F38 0B3A 5F39 B99B F505 E562 8D5C 5554 4E17 8863
 Thilo Borgmann (thilo)        CE1D B7F4 4D20 FC3A DD9F FE5A 257C 5B8F 1D20 B92F
 Tiancheng "Timothy" Gu        9456 AFC0 814A 8139 E994 8351 7FE6 B095 B582 B0D4
 Tim Nicholson                 38CF DB09 3ED0 F607 8B67 6CED 0C0B FC44 8B0B FC83
-Tomas Härdin (thardin)        A79D 4E3D F38F 763F 91F5 8B33 A01E 8AE0 41BB 2551
+Tomas Härdin                  A79D 4E3D F38F 763F 91F5 8B33 A01E 8AE0 41BB 2551
 Wei Gao                       4269 7741 857A 0E60 9EC5 08D2 4744 4EFA 62C1 87B9
 Zane van Iperen (zane)        61AE D40F 368B 6F26 9DAE 3892 6861 6B2D 8AC4 DCC5
--- a/179
+++ b/179
@@ -1,31 +1,49 @@
 MAIN_MAKEFILE=1
-include ffbuild/config.mak
+include config.mak
 vpath %.c    $(SRC_PATH)
 vpath %.cpp  $(SRC_PATH)
 vpath %.h    $(SRC_PATH)
 vpath %.inc  $(SRC_PATH)
 vpath %.m    $(SRC_PATH)
 vpath %.S    $(SRC_PATH)
 vpath %.asm  $(SRC_PATH)
 vpath %.rc   $(SRC_PATH)
 vpath %.v    $(SRC_PATH)
 vpath %.texi $(SRC_PATH)
 vpath %.cu   $(SRC_PATH)
 vpath %.ptx  $(SRC_PATH)
 vpath %.metal $(SRC_PATH)
 vpath %/fate_config.sh.template $(SRC_PATH)
-TESTTOOLS   = audiogen videogen rotozoom tiny_psnr tiny_ssim base64 audiomatch
+AVPROGS-$(CONFIG_FFMPEG)   += ffmpeg
-HOSTPROGS  := $(TESTTOOLS:%=tests/%) doc/print_options
+AVPROGS-$(CONFIG_FFPLAY)   += ffplay
 AVPROGS-$(CONFIG_FFPROBE)  += ffprobe
 AVPROGS-$(CONFIG_FFSERVER) += ffserver
-ALLFFLIBS = avcodec avdevice avfilter avformat avutil postproc swscale swresample
+AVPROGS    := $(AVPROGS-yes:%=%$(PROGSSUF)$(EXESUF))
 INSTPROGS   = $(AVPROGS-yes:%=%$(PROGSSUF)$(EXESUF))
 PROGS      += $(AVPROGS)
 AVBASENAMES  = ffmpeg ffplay ffprobe ffserver
 ALLAVPROGS   = $(AVBASENAMES:%=%$(PROGSSUF)$(EXESUF))
 ALLAVPROGS_G = $(AVBASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
 $(foreach prog,$(AVBASENAMES),$(eval OBJS-$(prog) += cmdutils.o))
 $(foreach prog,$(AVBASENAMES),$(eval OBJS-$(prog)-$(CONFIG_OPENCL) += cmdutils_opencl.o))
 OBJS-ffmpeg                   += ffmpeg_opt.o ffmpeg_filter.o
 OBJS-ffmpeg-$(HAVE_VDPAU_X11) += ffmpeg_vdpau.o
 OBJS-ffmpeg-$(HAVE_DXVA2_LIB) += ffmpeg_dxva2.o
 OBJS-ffmpeg-$(CONFIG_VDA)     += ffmpeg_vda.o
 TESTTOOLS   = audiogen videogen rotozoom tiny_psnr tiny_ssim base64
 HOSTPROGS  := $(TESTTOOLS:%=tests/%) doc/print_options
 TOOLS       = qt-faststart trasher uncoded_frame
 TOOLS-$(CONFIG_ZLIB) += cws2fws
 # $(FFLIBS-yes) needs to be in linking order
 FFLIBS-$(CONFIG_AVDEVICE)   += avdevice
 FFLIBS-$(CONFIG_AVFILTER)   += avfilter
 FFLIBS-$(CONFIG_AVFORMAT)   += avformat
 FFLIBS-$(CONFIG_AVCODEC)    += avcodec
 FFLIBS-$(CONFIG_AVRESAMPLE) += avresample
 FFLIBS-$(CONFIG_POSTPROC)   += postproc
 FFLIBS-$(CONFIG_SWRESAMPLE) += swresample
 FFLIBS-$(CONFIG_SWSCALE)    += swscale
@@ -33,68 +51,36 @@ FFLIBS-$(CONFIG_SWSCALE)    += swscale
 FFLIBS := avutil
 DATA_FILES := $(wildcard $(SRC_PATH)/presets/*.ffpreset) $(SRC_PATH)/doc/ffprobe.xsd
 EXAMPLES_FILES := $(wildcard $(SRC_PATH)/doc/examples/*.c) $(SRC_PATH)/doc/examples/Makefile $(SRC_PATH)/doc/examples/README
-SKIPHEADERS = compat/w32pthreads.h
+SKIPHEADERS = cmdutils_common_opts.h compat/w32pthreads.h
-# first so "all" becomes default target
+include $(SRC_PATH)/common.mak
 all: all-yes
 include $(SRC_PATH)/tools/Makefile
 include $(SRC_PATH)/ffbuild/common.mak
 FF_EXTRALIBS := $(FFEXTRALIBS)
 FF_DEP_LIBS  := $(DEP_LIBS)
 FF_STATIC_DEP_LIBS := $(STATIC_DEP_LIBS)
-$(TOOLS): %$(EXESUF): %.o
+all: $(AVPROGS)
 	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(EXTRALIBS-$(*F)) $(EXTRALIBS) $(ELIBS)
-target_dec_%_fuzzer$(EXESUF): target_dec_%_fuzzer.o $(FF_DEP_LIBS)
+$(TOOLS): %$(EXESUF): %.o $(EXEOBJS)
-	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
+	$(LD) $(LDFLAGS) $(LD_O) $^ $(ELIBS)
-tools/target_bsf_%_fuzzer$(EXESUF): tools/target_bsf_%_fuzzer.o $(FF_DEP_LIBS)
+tools/cws2fws$(EXESUF): ELIBS = $(ZLIB)
 	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
 target_dem_%_fuzzer$(EXESUF): target_dem_%_fuzzer.o $(FF_DEP_LIBS)
 	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
 tools/target_dem_fuzzer$(EXESUF): tools/target_dem_fuzzer.o $(FF_DEP_LIBS)
 	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
 tools/target_io_dem_fuzzer$(EXESUF): tools/target_io_dem_fuzzer.o $(FF_DEP_LIBS)
 	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
 tools/enum_options$(EXESUF): ELIBS = $(FF_EXTRALIBS)
 tools/enum_options$(EXESUF): $(FF_DEP_LIBS)
 tools/scale_slice_test$(EXESUF): $(FF_DEP_LIBS)
 tools/scale_slice_test$(EXESUF): ELIBS = $(FF_EXTRALIBS)
 tools/sofa2wavs$(EXESUF): ELIBS = $(FF_EXTRALIBS)
 tools/uncoded_frame$(EXESUF): $(FF_DEP_LIBS)
 tools/uncoded_frame$(EXESUF): ELIBS = $(FF_EXTRALIBS)
 tools/target_dec_%_fuzzer$(EXESUF): $(FF_DEP_LIBS)
 tools/target_dem_%_fuzzer$(EXESUF): $(FF_DEP_LIBS)
-CONFIGURABLE_COMPONENTS =                                           \
+config.h: .config
-    $(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c))                 \
+.config: $(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c))
    $(SRC_PATH)/libavcodec/bitstream_filters.c                      \
    $(SRC_PATH)/libavcodec/hwaccels.h                               \
    $(SRC_PATH)/libavcodec/parsers.c                                \
    $(SRC_PATH)/libavformat/protocols.c                             \
 config_components.h: ffbuild/.config
 ffbuild/.config: $(CONFIGURABLE_COMPONENTS)
 	@-tput bold 2>/dev/null
-	@-printf '\nWARNING: $(?) newer than config_components.h, rerun configure\n\n'
+	@-printf '\nWARNING: $(?F) newer than config.h, rerun configure\n\n'
 	@-tput sgr0 2>/dev/null
-SUBDIR_VARS := CLEANFILES FFLIBS HOSTPROGS TESTPROGS TOOLS               \
+SUBDIR_VARS := CLEANFILES EXAMPLES FFLIBS HOSTPROGS TESTPROGS TOOLS      \
               HEADERS ARCH_HEADERS BUILT_HEADERS SKIPHEADERS            \
               ARMV5TE-OBJS ARMV6-OBJS ARMV8-OBJS VFP-OBJS NEON-OBJS     \
-               ALTIVEC-OBJS VSX-OBJS MMX-OBJS X86ASM-OBJS                \
+               ALTIVEC-OBJS MMX-OBJS YASM-OBJS                           \
-               MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSP-OBJS MSA-OBJS         \
+               MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSPR1-OBJS MIPS32R2-OBJS  \
-               MMI-OBJS LSX-OBJS LASX-OBJS OBJS SLIBOBJS SHLIBOBJS       \
+               OBJS SLIBOBJS HOSTOBJS TESTOBJS
               STLIBOBJS HOSTOBJS TESTOBJS
 define RESET
 $(1) :=
@@ -106,34 +92,41 @@ $(foreach V,$(SUBDIR_VARS),$(eval $(call RESET,$(V))))
 SUBDIR := $(1)/
 include $(SRC_PATH)/$(1)/Makefile
 -include $(SRC_PATH)/$(1)/$(ARCH)/Makefile
-include $(SRC_PATH)/$(1)/$(INTRINSICS)/Makefile
+include $(SRC_PATH)/library.mak
 include $(SRC_PATH)/ffbuild/library.mak
 endef
 $(foreach D,$(FFLIBS),$(eval $(call DOSUBDIR,lib$(D))))
 include $(SRC_PATH)/fftools/Makefile
 include $(SRC_PATH)/doc/Makefile
 include $(SRC_PATH)/doc/examples/Makefile
-$(ALLFFLIBS:%=lib%/version.o): libavutil/ffversion.h
+define DOPROG
 OBJS-$(1) += $(1).o $(EXEOBJS) $(OBJS-$(1)-yes)
 $(1)$(PROGSSUF)_g$(EXESUF): $$(OBJS-$(1))
 $$(OBJS-$(1)): CFLAGS  += $(CFLAGS-$(1))
 $(1)$(PROGSSUF)_g$(EXESUF): LDFLAGS += $(LDFLAGS-$(1))
 $(1)$(PROGSSUF)_g$(EXESUF): FF_EXTRALIBS += $(LIBS-$(1))
 -include $$(OBJS-$(1):.o=.d)
 endef
 $(foreach P,$(PROGS),$(eval $(call DOPROG,$(P:$(PROGSSUF)$(EXESUF)=))))
 ffprobe.o cmdutils.o : libavutil/ffversion.h
 $(PROGS): %$(PROGSSUF)$(EXESUF): %$(PROGSSUF)_g$(EXESUF)
 ifeq ($(STRIPTYPE),direct)
 	$(STRIP) -o $@ $<
 else
 	$(RM) $@
 	$(CP) $< $@
 	$(STRIP) $@
 endif
-%$(PROGSSUF)_g$(EXESUF): $(FF_DEP_LIBS)
+%$(PROGSSUF)_g$(EXESUF): %.o $(FF_DEP_LIBS)
-	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
+	$(LD) $(LDFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
-VERSION_SH  = $(SRC_PATH)/ffbuild/version.sh
+OBJDIRS += tools
 -include $(wildcard tools/*.d)
 VERSION_SH  = $(SRC_PATH)/version.sh
 GIT_LOG     = $(SRC_PATH)/.git/logs/HEAD
-.version: $(wildcard $(GIT_LOG)) $(VERSION_SH) ffbuild/config.mak
+.version: $(wildcard $(GIT_LOG)) $(VERSION_SH) config.mak
 .version: M=@
 libavutil/ffversion.h .version:
@@ -143,47 +136,53 @@ libavutil/ffversion.h .version:
 # force version.sh to run whenever version might have changed
 -include .version
 ifdef AVPROGS
 install: install-progs install-data
 endif
 install: install-libs install-headers
 install-libs: install-libs-yes
-install-data: $(DATA_FILES)
+install-progs-yes:
-	$(Q)mkdir -p "$(DATADIR)"
+install-progs-$(CONFIG_SHARED): install-libs
 	$(INSTALL) -m 644 $(DATA_FILES) "$(DATADIR)"
-uninstall: uninstall-data uninstall-headers uninstall-libs uninstall-pkgconfig
+install-progs: install-progs-yes $(AVPROGS)
 	$(Q)mkdir -p "$(BINDIR)"
 	$(INSTALL) -c -m 755 $(INSTPROGS) "$(BINDIR)"
 install-data: $(DATA_FILES) $(EXAMPLES_FILES)
 	$(Q)mkdir -p "$(DATADIR)/examples"
 	$(INSTALL) -m 644 $(DATA_FILES) "$(DATADIR)"
 	$(INSTALL) -m 644 $(EXAMPLES_FILES) "$(DATADIR)/examples"
 uninstall: uninstall-libs uninstall-headers uninstall-progs uninstall-data
 uninstall-progs:
 	$(RM) $(addprefix "$(BINDIR)/", $(ALLAVPROGS))
 uninstall-data:
 	$(RM) -r "$(DATADIR)"
 clean::
 	$(RM) $(ALLAVPROGS) $(ALLAVPROGS_G)
 	$(RM) $(CLEANSUFFIXES)
-	$(RM) $(addprefix compat/,$(CLEANSUFFIXES)) $(addprefix compat/*/,$(CLEANSUFFIXES)) $(addprefix compat/*/*/,$(CLEANSUFFIXES))
+	$(RM) $(CLEANSUFFIXES:%=tools/%)
 	$(RM) -r coverage-html
-	$(RM) -rf coverage.info coverage.info.in lcov
+	$(RM) -rf coverage.info lcov
-distclean:: clean
+distclean::
-	$(RM) .version config.asm config.h config_components.h mapfile  \
+	$(RM) $(DISTCLEANSUFFIXES)
-		ffbuild/.config ffbuild/config.* libavutil/avconfig.h \
+	$(RM) config.* .config libavutil/avconfig.h .version version.h libavutil/ffversion.h libavcodec/codec_names.h
 		version.h libavutil/ffversion.h libavcodec/codec_names.h \
 		libavcodec/bsf_list.c libavformat/protocol_list.c \
 		libavcodec/codec_list.c libavcodec/parser_list.c \
 		libavfilter/filter_list.c libavdevice/indev_list.c libavdevice/outdev_list.c \
 		libavformat/muxer_list.c libavformat/demuxer_list.c
 ifeq ($(SRC_LINK),src)
 	$(RM) src
 endif
 	$(RM) -rf doc/examples/pc-uninstalled
 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
@@ -194,5 +193,5 @@ $(sort $(OUTDIRS)):
 # so this saves some time on slow systems.
 .SUFFIXES:
-.PHONY: all all-yes alltools build check config testprogs
+.PHONY: all all-yes alltools check *clean config install*
-.PHONY: *clean install* uninstall*
+.PHONY: testprogs uninstall*
--- a/README.md
+++ b/README.md
@@ -1,46 +1,19 @@
 FFmpeg README
 =============
-FFmpeg is a collection of libraries and tools to process multimedia content
+1) Documentation
-such as audio, video, subtitles and related metadata.
+----------------
-## Libraries
+* Read the documentation in the doc/ directory in git.
-* `libavcodec` provides implementation of a wider range of codecs.
+  You can also view it online at http://ffmpeg.org/documentation.html
 * `libavformat` implements streaming protocols, container formats and basic I/O access.
 * `libavutil` includes hashers, decompressors and miscellaneous utility functions.
 * `libavfilter` provides means to alter decoded audio and video through a directed graph of connected filters.
 * `libavdevice` provides an abstraction to access capture and playback devices.
 * `libswresample` implements audio mixing and resampling routines.
 * `libswscale` implements color conversion and scaling routines.
-## Tools
+2) Licensing
 ------------
-* [ffmpeg](https://ffmpeg.org/ffmpeg.html) is a command line toolbox to
+* See the LICENSE file.
  manipulate, convert and stream multimedia content.
 * [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.
 * Additional small tools such as `aviocat`, `ismindex` and `qt-faststart`.
-## Documentation
+3) Build and Install
 --------------------
-The offline documentation is available in the **doc/** directory.
+* See the INSTALL file.
 The online documentation is available in the main [website](https://ffmpeg.org)
 and in the [wiki](https://trac.ffmpeg.org).
 ### Examples
 Coding examples are available in the **doc/examples** directory.
 ## License
 FFmpeg codebase is mainly LGPL-licensed with optional components licensed under
 GPL. Please refer to the LICENSE file for detailed information.
 ## Contributing
 Patches should be submitted to the ffmpeg-devel mailing list using
 `git format-patch` or `git send-email`. Github pull requests should be
 avoided because they are not part of our review process and will be ignored.
--- a/2
+++ b/2
@@ -1 +1 @@
-4.4.git
+2.3.4
--- a/177
+++ b/177
@@ -0,0 +1,177 @@
 ┌───────────────────────────────────────────┐
 │ RELEASE NOTES for FFmpeg 2.3 "Mandelbrot" │
 └───────────────────────────────────────────┘
   The FFmpeg Project proudly presents FFmpeg 2.3 "Mandelbrot", a major
   release with all the great features committed during the three-month period
   since the release of FFmpeg 2.2.
   In this release, there are lots of internal overhauls that make FFmpeg a
   more accessible project for new developers. Many important new
   optimizations and features like QTKit and AVFoundation input devices are
   committed. Contributions done by Libav such as a new native Opus decoder
   are also merged.
   Because of the increasing difficulty to maintain and lack of maintainers,
   we are very sorry to say that we have removed all Blackfin and SPARC
   architecture assembly optimizations with the cleanups done. If you are
   interested in maintaining optimization for these two architecture, feel
   free to contact us and we will restore the code!
   Oh, and since this release, this modern-looking release note is provided in
   addition to the old-style Changelog file, to make it easier for you to
   focus on the most important features in this release.
   Enjoy!
   ┌────────────────────────────┐
   │ *  API Information         │
   └────────────────────────────┘
     FFmpeg 2.3 is completely source-compatible to the FFmpeg 2.2 series. There
     are however some API deprecations that you need to take care of. Use `git
     diff n2.2 n2.3 doc/APIchanges` to show the list of added and deprecated
     APIs. FFmpeg 2.3 includes the following library versions:
       • libavutil      52.92.100
       • libavcodec     55.69.100
       • libavformat    55.48.100
       • libavdevice    55.13.102
       • libavfilter     4.11.100
       • libswscale      2. 6.100
       • libswresample   0.19.100
       • libpostproc    52. 3.100
     Please refer to the doc/APIChanges file for more information.
   ┌────────────────────────────┐
   │ New Optimization           │
   └────────────────────────────┘
     We are excited to announce that we have committed new x86 assembly
     optimization for HEVC, and FFmpeg's audio resampler libswresample. ARM
     users will get a boost in MLP/TrueHD decoding thanks to new optimization.
     Decoding Huffyuv also got a major boost from optimization on the C code.
     Of special interest for Microsoft Visual Studio users, we have also
     converted some preexisting x86 assembly to NASM/Yasm format compatible
     with MSVC setup, especially in the area of audio resampling.
     Another major feature in this release is the introduction of AArch64
     (ARMv8) assembly optimization. AArch64 is another name for the first
     64-bit ARM architecture, used by Apple A7 SoC inside iPhone 5S. Some
     32-bit ARM assembly has already been ported to AArch64, but more work is
     underway.
   ┌────────────────────────────┐
   │ Native Opus decoder        │
   └────────────────────────────┘
     Opus is an open audio format jointly developed by Xiph.Org, Mozilla,
     Skype/Microsoft, and Broadcom. It combines the features of the Skype Cilk
     speech codec and the Xiph.Org CELT music codec into one low-latency
     codec. Decoding Opus is already possible since FFmpeg 1.0 using the
     libopus library, but the new Opus native decoder brings a higher level of
     stability and speed.
   ┌────────────────────────────┐
   │ QTKit and AVFoundation     │
   └────────────────────────────┘
     For OS X users, the new QTKit and AVFoundation devices allow you to use
     the integrated camera on Macs. AVFoundation is a newer API only available
     on OS X 10.7 "Lion" or newer. For users with older OS X systems, the
     QTKit device using the older OS X API is for you.
   ┌────────────────────────────┐
   │ API Additions              │
   └────────────────────────────┘
     In this release, stream side data are introduced as AVStream.side_data as
     a way to store miscellaneous stream-wide information. The format is
     similar to the previously anonymous structure AVPacket.side_data (now
     named as AVPacketSideData). With this change, audio ReplayGain
     information and video rotation matrix are now exported through this API,
     if available in the demuxer.
     We also have improved libswresample's Doxygen API documentation, so new
     developers wishing to use FFmpeg's excellent libraries can get started
     more easily and faster.
   ┌────────────────────────────┐
   │ Last But Not Least         │
   └────────────────────────────┘
     Other interesting new features including hqx video filter, a pixel art
     scaling filter; a fixed-point AC-3 decoder contributed by Imagination
     Technologies; an On2 TrueMotion VP7 video decoder; an HTML5 WebVTT
     subtitle encoder that allows creation of WebVTT from any text-based
     subtitles; and an 1-bit Direct Stream Digital audio decoder.
 ┌────────────────────────────┐
 │ ★  List of New Features    │
 └────────────────────────────┘
   ┌────────────────────────────┐
   │ Command line tools         │
   └────────────────────────────┘
    • Support for decoding through DXVA2 in ffmpeg
   ┌────────────────────────────┐
   │ libavcodec                 │
   └────────────────────────────┘
    • AC3 fixed-point decoding
    • VP7 video decoder
    • Alias PIX image encoder and decoder
    • Improvements to the BRender PIX image decoder
    • Improvements to the XBM decoder
    • Improvements to OpenEXR image decoder
    • Support decoding 16-bit RLE SGI images
    • Direct Stream Digital (DSD) decoder
    • On2 AVC (Audio for Video) decoder
    • Native Opus decoder
    • WebVTT encoder
   ┌────────────────────────────┐
   │ libavdevice                │
   └────────────────────────────┘
    • QTKit input device
    • GDI screen grabbing for Windows
    • AVFoundation input device
   ┌────────────────────────────┐
   │ libavformat                │
   └────────────────────────────┘
    • subfile protocol
    • Phantom Cine demuxer
    • Alternative rendition support for HTTP Live Streaming
    • Magic Lantern Video (MLV) demuxer
    • Image format auto-detection
    • LRC lyric file demuxer and muxer
    • Samba protocol (via libsmbclient)
    • WebM DASH Manifest muxer
   ┌────────────────────────────┐
   │ libavfilter                │
   └────────────────────────────┘
    • shuffleplanes filter
    • libbs2b-based stereo-to-binaural audio filter
    • showcqt multimedia filter
    • zoompan filter
    • signalstats filter
    • hqx filter (hq2x, hq3x, hq4x)
    • flanger filter
    • libfribidi support in drawtext
 ┌────────────────────────────┐
 │ ⚠  Behaviour changes       │
 └────────────────────────────┘
  • libx264 reference frames count is now limited depending on level chosen
  • Because of the new image format auto-detection feature, you don't need to
    specify image format when decoding an image with no extension.
--- a/arch.mak
+++ b/arch.mak
@@ -0,0 +1,15 @@
 OBJS-$(HAVE_ARMV5TE) += $(ARMV5TE-OBJS) $(ARMV5TE-OBJS-yes)
 OBJS-$(HAVE_ARMV6)   += $(ARMV6-OBJS)   $(ARMV6-OBJS-yes)
 OBJS-$(HAVE_ARMV8)   += $(ARMV8-OBJS)   $(ARMV8-OBJS-yes)
 OBJS-$(HAVE_VFP)     += $(VFP-OBJS)     $(VFP-OBJS-yes)
 OBJS-$(HAVE_NEON)    += $(NEON-OBJS)    $(NEON-OBJS-yes)
 OBJS-$(HAVE_MIPSFPU)   += $(MIPSFPU-OBJS)    $(MIPSFPU-OBJS-yes)
 OBJS-$(HAVE_MIPS32R2)  += $(MIPS32R2-OBJS)   $(MIPS32R2-OBJS-yes)
 OBJS-$(HAVE_MIPSDSPR1) += $(MIPSDSPR1-OBJS)  $(MIPSDSPR1-OBJS-yes)
 OBJS-$(HAVE_MIPSDSPR2) += $(MIPSDSPR2-OBJS)  $(MIPSDSPR2-OBJS-yes)
 OBJS-$(HAVE_ALTIVEC) += $(ALTIVEC-OBJS) $(ALTIVEC-OBJS-yes)
 OBJS-$(HAVE_MMX)     += $(MMX-OBJS)     $(MMX-OBJS-yes)
 OBJS-$(HAVE_YASM)    += $(YASM-OBJS)    $(YASM-OBJS-yes)
--- a/cmdutils.c
+++ b/cmdutils.c
--- a/cmdutils.h
+++ b/cmdutils.h
@@ -0,0 +1,586 @@
 /*
 * Various utilities for command line tools
 * copyright (c) 2003 Fabrice Bellard
 *
 * 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 FFMPEG_CMDUTILS_H
 #define FFMPEG_CMDUTILS_H
 #include <stdint.h>
 #include "config.h"
 #include "libavcodec/avcodec.h"
 #include "libavfilter/avfilter.h"
 #include "libavformat/avformat.h"
 #include "libswscale/swscale.h"
 #ifdef _WIN32
 #undef main /* We don't want SDL to override our main() */
 #endif
 /**
 * program name, defined by the program for show_version().
 */
 extern const char program_name[];
 /**
 * program birth year, defined by the program for show_banner()
 */
 extern const int program_birth_year;
 extern AVCodecContext *avcodec_opts[AVMEDIA_TYPE_NB];
 extern AVFormatContext *avformat_opts;
 extern struct SwsContext *sws_opts;
 extern AVDictionary *swr_opts;
 extern AVDictionary *format_opts, *codec_opts, *resample_opts;
 extern int hide_banner;
 /**
 * Register a program-specific cleanup routine.
 */
 void register_exit(void (*cb)(int ret));
 /**
 * Wraps exit with a program-specific cleanup routine.
 */
 void exit_program(int ret) av_noreturn;
 /**
 * Initialize the cmdutils option system, in particular
 * allocate the *_opts contexts.
 */
 void init_opts(void);
 /**
 * Uninitialize the cmdutils option system, in particular
 * free the *_opts contexts and their contents.
 */
 void uninit_opts(void);
 /**
 * Trivial log callback.
 * Only suitable for opt_help and similar since it lacks prefix handling.
 */
 void log_callback_help(void* ptr, int level, const char* fmt, va_list vl);
 /**
 * Override the cpuflags.
 */
 int opt_cpuflags(void *optctx, const char *opt, const char *arg);
 /**
 * Fallback for options that are not explicitly handled, these will be
 * parsed through AVOptions.
 */
 int opt_default(void *optctx, const char *opt, const char *arg);
 /**
 * Set the libav* libraries log level.
 */
 int opt_loglevel(void *optctx, const char *opt, const char *arg);
 int opt_report(const char *opt);
 int opt_max_alloc(void *optctx, const char *opt, const char *arg);
 int opt_codec_debug(void *optctx, const char *opt, const char *arg);
 #if CONFIG_OPENCL
 int opt_opencl(void *optctx, const char *opt, const char *arg);
 int opt_opencl_bench(void *optctx, const char *opt, const char *arg);
 #endif
 /**
 * Limit the execution time.
 */
 int opt_timelimit(void *optctx, const char *opt, const char *arg);
 /**
 * Parse a string and return its corresponding value as a double.
 * Exit from the application if the string cannot be correctly
 * parsed or the corresponding value is invalid.
 *
 * @param context the context of the value to be set (e.g. the
 * corresponding command line option name)
 * @param numstr the string to be parsed
 * @param type the type (OPT_INT64 or OPT_FLOAT) as which the
 * string should be parsed
 * @param min the minimum valid accepted value
 * @param max the maximum valid accepted value
 */
 double parse_number_or_die(const char *context, const char *numstr, int type,
                           double min, double max);
 /**
 * Parse a string specifying a time and return its corresponding
 * value as a number of microseconds. Exit from the application if
 * the string cannot be correctly parsed.
 *
 * @param context the context of the value to be set (e.g. the
 * corresponding command line option name)
 * @param timestr the string to be parsed
 * @param is_duration a flag which tells how to interpret timestr, if
 * not zero timestr is interpreted as a duration, otherwise as a
 * date
 *
 * @see av_parse_time()
 */
 int64_t parse_time_or_die(const char *context, const char *timestr,
                          int is_duration);
 typedef struct SpecifierOpt {
    char *specifier;    /**< stream/chapter/program/... specifier */
    union {
        uint8_t *str;
        int        i;
        int64_t  i64;
        float      f;
        double   dbl;
    } u;
 } SpecifierOpt;
 typedef struct OptionDef {
    const char *name;
    int flags;
 #define HAS_ARG    0x0001
 #define OPT_BOOL   0x0002
 #define OPT_EXPERT 0x0004
 #define OPT_STRING 0x0008
 #define OPT_VIDEO  0x0010
 #define OPT_AUDIO  0x0020
 #define OPT_INT    0x0080
 #define OPT_FLOAT  0x0100
 #define OPT_SUBTITLE 0x0200
 #define OPT_INT64  0x0400
 #define OPT_EXIT   0x0800
 #define OPT_DATA   0x1000
 #define OPT_PERFILE  0x2000     /* the option is per-file (currently ffmpeg-only).
                                   implied by OPT_OFFSET or OPT_SPEC */
 #define OPT_OFFSET 0x4000       /* option is specified as an offset in a passed optctx */
 #define OPT_SPEC   0x8000       /* option is to be stored in an array of SpecifierOpt.
                                   Implies OPT_OFFSET. Next element after the offset is
                                   an int containing element count in the array. */
 #define OPT_TIME  0x10000
 #define OPT_DOUBLE 0x20000
 #define OPT_INPUT  0x40000
 #define OPT_OUTPUT 0x80000
     union {
        void *dst_ptr;
        int (*func_arg)(void *, const char *, const char *);
        size_t off;
    } u;
    const char *help;
    const char *argname;
 } OptionDef;
 /**
 * Print help for all options matching specified flags.
 *
 * @param options a list of options
 * @param msg title of this group. Only printed if at least one option matches.
 * @param req_flags print only options which have all those flags set.
 * @param rej_flags don't print options which have any of those flags set.
 * @param alt_flags print only options that have at least one of those flags set
 */
 void show_help_options(const OptionDef *options, const char *msg, int req_flags,
                       int rej_flags, int alt_flags);
 /**
 * Show help for all options with given flags in class and all its
 * children.
 */
 void show_help_children(const AVClass *class, int flags);
 /**
 * Per-fftool specific help handler. Implemented in each
 * fftool, called by show_help().
 */
 void show_help_default(const char *opt, const char *arg);
 /**
 * Generic -h handler common to all fftools.
 */
 int show_help(void *optctx, const char *opt, const char *arg);
 /**
 * Parse the command line arguments.
 *
 * @param optctx an opaque options context
 * @param argc   number of command line arguments
 * @param argv   values of command line arguments
 * @param options Array with the definitions required to interpret every
 * option of the form: -option_name [argument]
 * @param parse_arg_function Name of the function called to process every
 * argument without a leading option name flag. NULL if such arguments do
 * not have to be processed.
 */
 void parse_options(void *optctx, int argc, char **argv, const OptionDef *options,
                   void (* parse_arg_function)(void *optctx, const char*));
 /**
 * Parse one given option.
 *
 * @return on success 1 if arg was consumed, 0 otherwise; negative number on error
 */
 int parse_option(void *optctx, const char *opt, const char *arg,
                 const OptionDef *options);
 /**
 * An option extracted from the commandline.
 * Cannot use AVDictionary because of options like -map which can be
 * used multiple times.
 */
 typedef struct Option {
    const OptionDef  *opt;
    const char       *key;
    const char       *val;
 } Option;
 typedef struct OptionGroupDef {
    /**< group name */
    const char *name;
    /**
     * Option to be used as group separator. Can be NULL for groups which
     * are terminated by a non-option argument (e.g. ffmpeg output files)
     */
    const char *sep;
    /**
     * Option flags that must be set on each option that is
     * applied to this group
     */
    int flags;
 } OptionGroupDef;
 typedef struct OptionGroup {
    const OptionGroupDef *group_def;
    const char *arg;
    Option *opts;
    int  nb_opts;
    AVDictionary *codec_opts;
    AVDictionary *format_opts;
    AVDictionary *resample_opts;
    struct SwsContext *sws_opts;
    AVDictionary *swr_opts;
 } OptionGroup;
 /**
 * A list of option groups that all have the same group type
 * (e.g. input files or output files)
 */
 typedef struct OptionGroupList {
    const OptionGroupDef *group_def;
    OptionGroup *groups;
    int       nb_groups;
 } OptionGroupList;
 typedef struct OptionParseContext {
    OptionGroup global_opts;
    OptionGroupList *groups;
    int           nb_groups;
    /* parsing state */
    OptionGroup cur_group;
 } OptionParseContext;
 /**
 * Parse an options group and write results into optctx.
 *
 * @param optctx an app-specific options context. NULL for global options group
 */
 int parse_optgroup(void *optctx, OptionGroup *g);
 /**
 * Split the commandline into an intermediate form convenient for further
 * processing.
 *
 * The commandline is assumed to be composed of options which either belong to a
 * group (those with OPT_SPEC, OPT_OFFSET or OPT_PERFILE) or are global
 * (everything else).
 *
 * A group (defined by an OptionGroupDef struct) is a sequence of options
 * terminated by either a group separator option (e.g. -i) or a parameter that
 * is not an option (doesn't start with -). A group without a separator option
 * must always be first in the supplied groups list.
 *
 * All options within the same group are stored in one OptionGroup struct in an
 * OptionGroupList, all groups with the same group definition are stored in one
 * OptionGroupList in OptionParseContext.groups. The order of group lists is the
 * same as the order of group definitions.
 */
 int split_commandline(OptionParseContext *octx, int argc, char *argv[],
                      const OptionDef *options,
                      const OptionGroupDef *groups, int nb_groups);
 /**
 * Free all allocated memory in an OptionParseContext.
 */
 void uninit_parse_context(OptionParseContext *octx);
 /**
 * Find the '-loglevel' option in the command line args and apply it.
 */
 void parse_loglevel(int argc, char **argv, const OptionDef *options);
 /**
 * Return index of option opt in argv or 0 if not found.
 */
 int locate_option(int argc, char **argv, const OptionDef *options,
                  const char *optname);
 /**
 * Check if the given stream matches a stream specifier.
 *
 * @param s  Corresponding format context.
 * @param st Stream from s to be checked.
 * @param spec A stream specifier of the [v|a|s|d]:[\<stream index\>] form.
 *
 * @return 1 if the stream matches, 0 if it doesn't, <0 on error
 */
 int check_stream_specifier(AVFormatContext *s, AVStream *st, const char *spec);
 /**
 * Filter out options for given codec.
 *
 * Create a new options dictionary containing only the options from
 * opts which apply to the codec with ID codec_id.
 *
 * @param opts     dictionary to place options in
 * @param codec_id ID of the codec that should be filtered for
 * @param s Corresponding format context.
 * @param st A stream from s for which the options should be filtered.
 * @param codec The particular codec for which the options should be filtered.
 *              If null, the default one is looked up according to the codec id.
 * @return a pointer to the created dictionary
 */
 AVDictionary *filter_codec_opts(AVDictionary *opts, enum AVCodecID codec_id,
                                AVFormatContext *s, AVStream *st, AVCodec *codec);
 /**
 * Setup AVCodecContext options for avformat_find_stream_info().
 *
 * Create an array of dictionaries, one dictionary for each stream
 * contained in s.
 * Each dictionary will contain the options from codec_opts which can
 * be applied to the corresponding stream codec context.
 *
 * @return pointer to the created array of dictionaries, NULL if it
 * cannot be created
 */
 AVDictionary **setup_find_stream_info_opts(AVFormatContext *s,
                                           AVDictionary *codec_opts);
 /**
 * Print an error message to stderr, indicating filename and a human
 * readable description of the error code err.
 *
 * If strerror_r() is not available the use of this function in a
 * multithreaded application may be unsafe.
 *
 * @see av_strerror()
 */
 void print_error(const char *filename, int err);
 /**
 * Print the program banner to stderr. The banner contents depend on the
 * current version of the repository and of the libav* libraries used by
 * the program.
 */
 void show_banner(int argc, char **argv, const OptionDef *options);
 /**
 * Print the version of the program to stdout. The version message
 * depends on the current versions of the repository and of the libav*
 * libraries.
 * This option processing function does not utilize the arguments.
 */
 int show_version(void *optctx, const char *opt, const char *arg);
 /**
 * Print the build configuration of the program to stdout. The contents
 * depend on the definition of FFMPEG_CONFIGURATION.
 * This option processing function does not utilize the arguments.
 */
 int show_buildconf(void *optctx, const char *opt, const char *arg);
 /**
 * Print the license of the program to stdout. The license depends on
 * the license of the libraries compiled into the program.
 * This option processing function does not utilize the arguments.
 */
 int show_license(void *optctx, const char *opt, const char *arg);
 /**
 * Print a listing containing all the formats supported by the
 * program (including devices).
 * This option processing function does not utilize the arguments.
 */
 int show_formats(void *optctx, const char *opt, const char *arg);
 /**
 * Print a listing containing all the devices supported by the
 * program.
 * This option processing function does not utilize the arguments.
 */
 int show_devices(void *optctx, const char *opt, const char *arg);
 /**
 * Print a listing containing all the codecs supported by the
 * program.
 * This option processing function does not utilize the arguments.
 */
 int show_codecs(void *optctx, const char *opt, const char *arg);
 /**
 * Print a listing containing all the decoders supported by the
 * program.
 */
 int show_decoders(void *optctx, const char *opt, const char *arg);
 /**
 * Print a listing containing all the encoders supported by the
 * program.
 */
 int show_encoders(void *optctx, const char *opt, const char *arg);
 /**
 * Print a listing containing all the filters supported by the
 * program.
 * This option processing function does not utilize the arguments.
 */
 int show_filters(void *optctx, const char *opt, const char *arg);
 /**
 * Print a listing containing all the bit stream filters supported by the
 * program.
 * This option processing function does not utilize the arguments.
 */
 int show_bsfs(void *optctx, const char *opt, const char *arg);
 /**
 * Print a listing containing all the protocols supported by the
 * program.
 * This option processing function does not utilize the arguments.
 */
 int show_protocols(void *optctx, const char *opt, const char *arg);
 /**
 * Print a listing containing all the pixel formats supported by the
 * program.
 * This option processing function does not utilize the arguments.
 */
 int show_pix_fmts(void *optctx, const char *opt, const char *arg);
 /**
 * Print a listing containing all the standard channel layouts supported by
 * the program.
 * This option processing function does not utilize the arguments.
 */
 int show_layouts(void *optctx, const char *opt, const char *arg);
 /**
 * Print a listing containing all the sample formats supported by the
 * program.
 */
 int show_sample_fmts(void *optctx, const char *opt, const char *arg);
 /**
 * Print a listing containing all the color names and values recognized
 * by the program.
 */
 int show_colors(void *optctx, const char *opt, const char *arg);
 /**
 * Return a positive value if a line read from standard input
 * starts with [yY], otherwise return 0.
 */
 int read_yesno(void);
 /**
 * Read the file with name filename, and put its content in a newly
 * allocated 0-terminated buffer.
 *
 * @param filename file to read from
 * @param bufptr location where pointer to buffer is returned
 * @param size   location where size of buffer is returned
 * @return >= 0 in case of success, a negative value corresponding to an
 * AVERROR error code in case of failure.
 */
 int cmdutils_read_file(const char *filename, char **bufptr, size_t *size);
 /**
 * Get a file corresponding to a preset file.
 *
 * If is_path is non-zero, look for the file in the path preset_name.
 * Otherwise search for a file named arg.ffpreset in the directories
 * $FFMPEG_DATADIR (if set), $HOME/.ffmpeg, and in the datadir defined
 * at configuration time or in a "ffpresets" folder along the executable
 * on win32, in that order. If no such file is found and
 * codec_name is defined, then search for a file named
 * codec_name-preset_name.avpreset in the above-mentioned directories.
 *
 * @param filename buffer where the name of the found filename is written
 * @param filename_size size in bytes of the filename buffer
 * @param preset_name name of the preset to search
 * @param is_path tell if preset_name is a filename path
 * @param codec_name name of the codec for which to look for the
 * preset, may be NULL
 */
 FILE *get_preset_file(char *filename, size_t filename_size,
                      const char *preset_name, int is_path, const char *codec_name);
 /**
 * Realloc array to hold new_size elements of elem_size.
 * Calls exit() on failure.
 *
 * @param array array to reallocate
 * @param elem_size size in bytes of each element
 * @param size new element count will be written here
 * @param new_size number of elements to place in reallocated array
 * @return reallocated array
 */
 void *grow_array(void *array, int elem_size, int *size, int new_size);
 #define media_type_string av_get_media_type_string
 #define GROW_ARRAY(array, nb_elems)\
    array = grow_array(array, sizeof(*array), &nb_elems, nb_elems + 1)
 #define GET_PIX_FMT_NAME(pix_fmt)\
    const char *name = av_get_pix_fmt_name(pix_fmt);
 #define GET_SAMPLE_FMT_NAME(sample_fmt)\
    const char *name = av_get_sample_fmt_name(sample_fmt)
 #define GET_SAMPLE_RATE_NAME(rate)\
    char name[16];\
    snprintf(name, sizeof(name), "%d", rate);
 #define GET_CH_LAYOUT_NAME(ch_layout)\
    char name[16];\
    snprintf(name, sizeof(name), "0x%"PRIx64, ch_layout);
 #define GET_CH_LAYOUT_DESC(ch_layout)\
    char name[128];\
    av_get_channel_layout_string(name, sizeof(name), 0, ch_layout);
 #endif /* CMDUTILS_H */
--- a/cmdutils_common_opts.h
+++ b/cmdutils_common_opts.h
@@ -0,0 +1,29 @@
    { "L"          , OPT_EXIT, {.func_arg = show_license},      "show license" },
    { "h"          , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
    { "?"          , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
    { "help"       , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
    { "-help"      , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
    { "version"    , OPT_EXIT, {.func_arg = show_version},      "show version" },
    { "buildconf"  , OPT_EXIT, {.func_arg = show_buildconf},    "show build configuration" },
    { "formats"    , OPT_EXIT, {.func_arg = show_formats  },    "show available formats" },
    { "devices"    , OPT_EXIT, {.func_arg = show_devices  },    "show available devices" },
    { "codecs"     , OPT_EXIT, {.func_arg = show_codecs   },    "show available codecs" },
    { "decoders"   , OPT_EXIT, {.func_arg = show_decoders },    "show available decoders" },
    { "encoders"   , OPT_EXIT, {.func_arg = show_encoders },    "show available encoders" },
    { "bsfs"       , OPT_EXIT, {.func_arg = show_bsfs     },    "show available bit stream filters" },
    { "protocols"  , OPT_EXIT, {.func_arg = show_protocols},    "show available protocols" },
    { "filters"    , OPT_EXIT, {.func_arg = show_filters  },    "show available filters" },
    { "pix_fmts"   , OPT_EXIT, {.func_arg = show_pix_fmts },    "show available pixel formats" },
    { "layouts"    , OPT_EXIT, {.func_arg = show_layouts  },    "show standard channel layouts" },
    { "sample_fmts", OPT_EXIT, {.func_arg = show_sample_fmts }, "show available audio sample formats" },
    { "colors"     , OPT_EXIT, {.func_arg = show_colors },      "show available color names" },
    { "loglevel"   , HAS_ARG,  {.func_arg = opt_loglevel},      "set logging level", "loglevel" },
    { "v",           HAS_ARG,  {.func_arg = opt_loglevel},      "set logging level", "loglevel" },
    { "report"     , 0,        {(void*)opt_report}, "generate a report" },
    { "max_alloc"  , HAS_ARG,  {.func_arg = opt_max_alloc},     "set maximum size of a single allocated block", "bytes" },
    { "cpuflags"   , HAS_ARG | OPT_EXPERT, { .func_arg = opt_cpuflags }, "force specific cpu flags", "flags" },
    { "hide_banner", OPT_BOOL | OPT_EXPERT, {&hide_banner},     "do not show program banner", "hide_banner" },
 #if CONFIG_OPENCL
    { "opencl_bench", OPT_EXIT, {.func_arg = opt_opencl_bench}, "run benchmark on all OpenCL devices and show results" },
    { "opencl_options", HAS_ARG, {.func_arg = opt_opencl},      "set OpenCL environment options" },
 #endif
--- a/cmdutils_opencl.c
+++ b/cmdutils_opencl.c
@@ -0,0 +1,274 @@
 /*
 * Copyright (C) 2013 Lenny Wang
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
 #include "libavutil/opt.h"
 #include "libavutil/time.h"
 #include "libavutil/log.h"
 #include "libavutil/opencl.h"
 #include "cmdutils.h"
 typedef struct {
    int platform_idx;
    int device_idx;
    char device_name[64];
    int64_t runtime;
 } OpenCLDeviceBenchmark;
 const char *ocl_bench_source = AV_OPENCL_KERNEL(
 inline unsigned char clip_uint8(int a)
 {
    if (a & (~0xFF))
        return (-a)>>31;
    else
        return a;
 }
 kernel void unsharp_bench(
                    global unsigned char *src,
                    global unsigned char *dst,
                    global int *mask,
                    int width,
                    int height)
 {
    int i, j, local_idx, lc_idx, sum = 0;
    int2 thread_idx, block_idx, global_idx, lm_idx;
    thread_idx.x = get_local_id(0);
    thread_idx.y = get_local_id(1);
    block_idx.x = get_group_id(0);
    block_idx.y = get_group_id(1);
    global_idx.x = get_global_id(0);
    global_idx.y = get_global_id(1);
    local uchar data[32][32];
    local int lc[128];
    for (i = 0; i <= 1; i++) {
        lm_idx.y = -8 + (block_idx.y + i) * 16 + thread_idx.y;
        lm_idx.y = lm_idx.y < 0 ? 0 : lm_idx.y;
        lm_idx.y = lm_idx.y >= height ? height - 1: lm_idx.y;
        for (j = 0; j <= 1; j++) {
            lm_idx.x = -8 + (block_idx.x + j) * 16 + thread_idx.x;
            lm_idx.x = lm_idx.x < 0 ? 0 : lm_idx.x;
            lm_idx.x = lm_idx.x >= width ? width - 1: lm_idx.x;
            data[i*16 + thread_idx.y][j*16 + thread_idx.x] = src[lm_idx.y*width + lm_idx.x];
        }
    }
    local_idx = thread_idx.y*16 + thread_idx.x;
    if (local_idx < 128)
        lc[local_idx] = mask[local_idx];
    barrier(CLK_LOCAL_MEM_FENCE);
    \n#pragma unroll\n
    for (i = -4; i <= 4; i++) {
        lm_idx.y = 8 + i + thread_idx.y;
        \n#pragma unroll\n
        for (j = -4; j <= 4; j++) {
            lm_idx.x = 8 + j + thread_idx.x;
            lc_idx = (i + 4)*8 + j + 4;
            sum += (int)data[lm_idx.y][lm_idx.x] * lc[lc_idx];
        }
    }
    int temp = (int)data[thread_idx.y + 8][thread_idx.x + 8];
    int res = temp + (((temp - (int)((sum + 1<<15) >> 16))) >> 16);
    if (global_idx.x < width && global_idx.y < height)
        dst[global_idx.x + global_idx.y*width] = clip_uint8(res);
 }
 );
 #define OCLCHECK(method, ... )                                                 \
 do {                                                                           \
    status = method(__VA_ARGS__);                                              \
    if (status != CL_SUCCESS) {                                                \
        av_log(NULL, AV_LOG_ERROR, # method " error '%s'\n",                   \
               av_opencl_errstr(status));                                      \
        ret = AVERROR_EXTERNAL;                                                \
        goto end;                                                              \
    }                                                                          \
 } while (0)
 #define CREATEBUF(out, flags, size)                                            \
 do {                                                                           \
    out = clCreateBuffer(ext_opencl_env->context, flags, size, NULL, &status); \
    if (status != CL_SUCCESS) {                                                \
        av_log(NULL, AV_LOG_ERROR, "Could not create OpenCL buffer\n");        \
        ret = AVERROR_EXTERNAL;                                                \
        goto end;                                                              \
    }                                                                          \
 } while (0)
 static void fill_rand_int(int *data, int n)
 {
    int i;
    srand(av_gettime());
    for (i = 0; i < n; i++)
        data[i] = rand();
 }
 #define OPENCL_NB_ITER 5
 static int64_t run_opencl_bench(AVOpenCLExternalEnv *ext_opencl_env)
 {
    int i, arg = 0, width = 1920, height = 1088;
    int64_t start, ret = 0;
    cl_int status;
    size_t kernel_len;
    char *inbuf;
    int *mask;
    int buf_size = width * height * sizeof(char);
    int mask_size = sizeof(uint32_t) * 128;
    cl_mem cl_mask, cl_inbuf, cl_outbuf;
    cl_kernel kernel = NULL;
    cl_program program = NULL;
    size_t local_work_size_2d[2] = {16, 16};
    size_t global_work_size_2d[2] = {(size_t)width, (size_t)height};
    if (!(inbuf = av_malloc(buf_size)) || !(mask = av_malloc(mask_size))) {
        av_log(NULL, AV_LOG_ERROR, "Out of memory\n");
        ret = AVERROR(ENOMEM);
        goto end;
    }
    fill_rand_int((int*)inbuf, buf_size/4);
    fill_rand_int(mask, mask_size/4);
    CREATEBUF(cl_mask, CL_MEM_READ_ONLY, mask_size);
    CREATEBUF(cl_inbuf, CL_MEM_READ_ONLY, buf_size);
    CREATEBUF(cl_outbuf, CL_MEM_READ_WRITE, buf_size);
    kernel_len = strlen(ocl_bench_source);
    program = clCreateProgramWithSource(ext_opencl_env->context, 1, &ocl_bench_source,
                                        &kernel_len, &status);
    if (status != CL_SUCCESS || !program) {
        av_log(NULL, AV_LOG_ERROR, "OpenCL unable to create benchmark program\n");
        ret = AVERROR_EXTERNAL;
        goto end;
    }
    status = clBuildProgram(program, 1, &(ext_opencl_env->device_id), NULL, NULL, NULL);
    if (status != CL_SUCCESS) {
        av_log(NULL, AV_LOG_ERROR, "OpenCL unable to build benchmark program\n");
        ret = AVERROR_EXTERNAL;
        goto end;
    }
    kernel = clCreateKernel(program, "unsharp_bench", &status);
    if (status != CL_SUCCESS) {
        av_log(NULL, AV_LOG_ERROR, "OpenCL unable to create benchmark kernel\n");
        ret = AVERROR_EXTERNAL;
        goto end;
    }
    OCLCHECK(clEnqueueWriteBuffer, ext_opencl_env->command_queue, cl_inbuf, CL_TRUE, 0,
             buf_size, inbuf, 0, NULL, NULL);
    OCLCHECK(clEnqueueWriteBuffer, ext_opencl_env->command_queue, cl_mask, CL_TRUE, 0,
             mask_size, mask, 0, NULL, NULL);
    OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_mem), &cl_inbuf);
    OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_mem), &cl_outbuf);
    OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_mem), &cl_mask);
    OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_int), &width);
    OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_int), &height);
    start = av_gettime_relative();
    for (i = 0; i < OPENCL_NB_ITER; i++)
        OCLCHECK(clEnqueueNDRangeKernel, ext_opencl_env->command_queue, kernel, 2, NULL,
                 global_work_size_2d, local_work_size_2d, 0, NULL, NULL);
    clFinish(ext_opencl_env->command_queue);
    ret = (av_gettime_relative() - start)/OPENCL_NB_ITER;
 end:
    if (kernel)
        clReleaseKernel(kernel);
    if (program)
        clReleaseProgram(program);
    if (cl_inbuf)
        clReleaseMemObject(cl_inbuf);
    if (cl_outbuf)
        clReleaseMemObject(cl_outbuf);
    if (cl_mask)
        clReleaseMemObject(cl_mask);
    av_free(inbuf);
    av_free(mask);
    return ret;
 }
 static int compare_ocl_device_desc(const void *a, const void *b)
 {
    return ((OpenCLDeviceBenchmark*)a)->runtime - ((OpenCLDeviceBenchmark*)b)->runtime;
 }
 int opt_opencl_bench(void *optctx, const char *opt, const char *arg)
 {
    int i, j, nb_devices = 0, count = 0;
    int64_t score = 0;
    AVOpenCLDeviceList *device_list;
    AVOpenCLDeviceNode *device_node = NULL;
    OpenCLDeviceBenchmark *devices = NULL;
    cl_platform_id platform;
    av_opencl_get_device_list(&device_list);
    for (i = 0; i < device_list->platform_num; i++)
        nb_devices += device_list->platform_node[i]->device_num;
    if (!nb_devices) {
        av_log(NULL, AV_LOG_ERROR, "No OpenCL device detected!\n");
        return AVERROR(EINVAL);
    }
    if (!(devices = av_malloc_array(nb_devices, sizeof(OpenCLDeviceBenchmark)))) {
        av_log(NULL, AV_LOG_ERROR, "Could not allocate buffer\n");
        return AVERROR(ENOMEM);
    }
    for (i = 0; i < device_list->platform_num; i++) {
        for (j = 0; j < device_list->platform_node[i]->device_num; j++) {
            device_node = device_list->platform_node[i]->device_node[j];
            platform = device_list->platform_node[i]->platform_id;
            score = av_opencl_benchmark(device_node, platform, run_opencl_bench);
            if (score > 0) {
                devices[count].platform_idx = i;
                devices[count].device_idx = j;
                devices[count].runtime = score;
                strcpy(devices[count].device_name, device_node->device_name);
                count++;
            }
        }
    }
    qsort(devices, count, sizeof(OpenCLDeviceBenchmark), compare_ocl_device_desc);
    fprintf(stderr, "platform_idx\tdevice_idx\tdevice_name\truntime\n");
    for (i = 0; i < count; i++)
        fprintf(stdout, "%d\t%d\t%s\t%"PRId64"\n",
                devices[i].platform_idx, devices[i].device_idx,
                devices[i].device_name, devices[i].runtime);
    av_opencl_free_device_list(&device_list);
    av_free(devices);
    return 0;
 }
 int opt_opencl(void *optctx, const char *opt, const char *arg)
 {
    char *key, *value;
    const char *opts = arg;
    int ret = 0;
    while (*opts) {
        ret = av_opt_get_key_value(&opts, "=", ":", 0, &key, &value);
        if (ret < 0)
            return ret;
        ret = av_opencl_set_option(key, value);
        if (ret < 0)
            return ret;
        if (*opts)
            opts++;
    }
    return ret;
 }
--- a/common.mak
+++ b/common.mak
@@ -0,0 +1,154 @@
 #
 # common bits used by all libraries
 #
 # first so "all" becomes default target
 all: all-yes
 ifndef SUBDIR
 ifndef V
 Q      = @
 ECHO   = printf "$(1)\t%s\n" $(2)
 BRIEF  = CC CXX HOSTCC HOSTLD AS YASM AR LD STRIP CP WINDRES
 SILENT = DEPCC DEPHOSTCC DEPAS DEPYASM RANLIB RM
 MSG    = $@
 M      = @$(call ECHO,$(TAG),$@);
 $(foreach VAR,$(BRIEF), \
    $(eval override $(VAR) = @$$(call ECHO,$(VAR),$$(MSG)); $($(VAR))))
 $(foreach VAR,$(SILENT),$(eval override $(VAR) = @$($(VAR))))
 $(eval INSTALL = @$(call ECHO,INSTALL,$$(^:$(SRC_DIR)/%=%)); $(INSTALL))
 endif
 ALLFFLIBS = avcodec avdevice avfilter avformat avresample avutil postproc swscale swresample
 # NASM requires -I path terminated with /
 IFLAGS     := -I. -I$(SRC_PATH)/
 CPPFLAGS   := $(IFLAGS) $(CPPFLAGS)
 CFLAGS     += $(ECFLAGS)
 CCFLAGS     = $(CPPFLAGS) $(CFLAGS)
 ASFLAGS    := $(CPPFLAGS) $(ASFLAGS)
 CXXFLAGS   += $(CPPFLAGS) $(CFLAGS)
 YASMFLAGS  += $(IFLAGS:%=%/) -Pconfig.asm
 HOSTCCFLAGS = $(IFLAGS) $(HOSTCPPFLAGS) $(HOSTCFLAGS)
 LDFLAGS    := $(ALLFFLIBS:%=$(LD_PATH)lib%) $(LDFLAGS)
 define COMPILE
       $(call $(1)DEP,$(1))
       $($(1)) $($(1)FLAGS) $($(1)_DEPFLAGS) $($(1)_C) $($(1)_O) $<
 endef
 COMPILE_C = $(call COMPILE,CC)
 COMPILE_CXX = $(call COMPILE,CXX)
 COMPILE_S = $(call COMPILE,AS)
 COMPILE_HOSTC = $(call COMPILE,HOSTCC)
 %.o: %.c
 	$(COMPILE_C)
 %.o: %.cpp
 	$(COMPILE_CXX)
 %.o: %.m
 	$(COMPILE_C)
 %.s: %.c
 	$(CC) $(CPPFLAGS) $(CFLAGS) -S -o $@ $<
 %.o: %.S
 	$(COMPILE_S)
 %_host.o: %.c
 	$(COMPILE_HOSTC)
 %.o: %.rc
 	$(WINDRES) $(IFLAGS) --preprocessor "$(DEPWINDRES) -E -xc-header -DRC_INVOKED $(CC_DEPFLAGS)" -o $@ $<
 %.i: %.c
 	$(CC) $(CCFLAGS) $(CC_E) $<
 %.h.c:
 	$(Q)echo '#include "$*.h"' >$@
 %.ver: %.v
 	$(Q)sed 's/$$MAJOR/$($(basename $(@F))_VERSION_MAJOR)/' $^ > $@
 %.c %.h: TAG = GEN
 # Dummy rule to stop make trying to rebuild removed or renamed headers
 %.h:
 	@:
 # Disable suffix rules.  Most of the builtin rules are suffix rules,
 # so this saves some time on slow systems.
 .SUFFIXES:
 # Do not delete intermediate files from chains of implicit rules
 $(OBJS):
 endif
 include $(SRC_PATH)/arch.mak
 OBJS      += $(OBJS-yes)
 SLIBOBJS  += $(SLIBOBJS-yes)
 FFLIBS    := $($(NAME)_FFLIBS) $(FFLIBS-yes) $(FFLIBS)
 TESTPROGS += $(TESTPROGS-yes)
 LDLIBS       = $(FFLIBS:%=%$(BUILDSUF))
 FFEXTRALIBS := $(LDLIBS:%=$(LD_LIB)) $(EXTRALIBS)
 OBJS      := $(sort $(OBJS:%=$(SUBDIR)%))
 SLIBOBJS  := $(sort $(SLIBOBJS:%=$(SUBDIR)%))
 TESTOBJS  := $(TESTOBJS:%=$(SUBDIR)%) $(TESTPROGS:%=$(SUBDIR)%-test.o)
 TESTPROGS := $(TESTPROGS:%=$(SUBDIR)%-test$(EXESUF))
 HOSTOBJS  := $(HOSTPROGS:%=$(SUBDIR)%.o)
 HOSTPROGS := $(HOSTPROGS:%=$(SUBDIR)%$(HOSTEXESUF))
 TOOLS     += $(TOOLS-yes)
 TOOLOBJS  := $(TOOLS:%=tools/%.o)
 TOOLS     := $(TOOLS:%=tools/%$(EXESUF))
 HEADERS   += $(HEADERS-yes)
 PATH_LIBNAME = $(foreach NAME,$(1),lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME))
 DEP_LIBS := $(foreach lib,$(FFLIBS),$(call PATH_LIBNAME,$(lib)))
 SRC_DIR    := $(SRC_PATH)/lib$(NAME)
 ALLHEADERS := $(subst $(SRC_DIR)/,$(SUBDIR),$(wildcard $(SRC_DIR)/*.h $(SRC_DIR)/$(ARCH)/*.h))
 SKIPHEADERS += $(ARCH_HEADERS:%=$(ARCH)/%) $(SKIPHEADERS-)
 SKIPHEADERS := $(SKIPHEADERS:%=$(SUBDIR)%)
 HOBJS        = $(filter-out $(SKIPHEADERS:.h=.h.o),$(ALLHEADERS:.h=.h.o))
 checkheaders: $(HOBJS)
 .SECONDARY:   $(HOBJS:.o=.c)
 alltools: $(TOOLS)
 $(HOSTOBJS): %.o: %.c
 	$(COMPILE_HOSTC)
 $(HOSTPROGS): %$(HOSTEXESUF): %.o
 	$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $^ $(HOSTLIBS)
 $(OBJS):     | $(sort $(dir $(OBJS)))
 $(HOBJS):    | $(sort $(dir $(HOBJS)))
 $(HOSTOBJS): | $(sort $(dir $(HOSTOBJS)))
 $(SLIBOBJS): | $(sort $(dir $(SLIBOBJS)))
 $(TESTOBJS): | $(sort $(dir $(TESTOBJS)))
 $(TOOLOBJS): | tools
 OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(SLIBOBJS) $(TESTOBJS))
 CLEANSUFFIXES     = *.d *.o *~ *.h.c *.map *.ver *.ho *.gcno *.gcda
 DISTCLEANSUFFIXES = *.pc
 LIBSUFFIXES       = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a
 define RULES
 clean::
 	$(RM) $(OBJS) $(OBJS:.o=.d)
 	$(RM) $(HOSTPROGS)
 	$(RM) $(TOOLS)
 endef
 $(eval $(RULES))
 -include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d) $(SLIBOBJS:.o=.d))
--- a/compat/aix/math.h
+++ b/compat/aix/math.h
@@ -19,8 +19,8 @@
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
-#ifndef COMPAT_AIX_MATH_H
+#ifndef FFMPEG_COMPAT_AIX_MATH_H
-#define COMPAT_AIX_MATH_H
+#define FFMPEG_COMPAT_AIX_MATH_H
 #define class class_in_math_h_causes_problems
@@ -28,4 +28,4 @@
 #undef class
-#endif /* COMPAT_AIX_MATH_H */
+#endif /* FFMPEG_COMPAT_AIX_MATH_H */
--- a/compat/atomics/dummy/stdatomic.h
+++ b/compat/atomics/dummy/stdatomic.h
@@ -1,176 +0,0 @@
 /*
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
 /*
 * based on vlc_atomic.h from VLC
 * Copyright (C) 2010 Rémi Denis-Courmont
 */
 #ifndef COMPAT_ATOMICS_DUMMY_STDATOMIC_H
 #define COMPAT_ATOMICS_DUMMY_STDATOMIC_H
 #include <stdint.h>
 #define ATOMIC_FLAG_INIT 0
 #define ATOMIC_VAR_INIT(value) (value)
 #define atomic_init(obj, value) \
 do {                            \
    *(obj) = (value);           \
 } while(0)
 #define kill_dependency(y) ((void)0)
 #define atomic_thread_fence(order) \
    ((void)0)
 #define atomic_signal_fence(order) \
    ((void)0)
 #define atomic_is_lock_free(obj) 0
 typedef intptr_t atomic_flag;
 typedef intptr_t atomic_bool;
 typedef intptr_t atomic_char;
 typedef intptr_t atomic_schar;
 typedef intptr_t atomic_uchar;
 typedef intptr_t atomic_short;
 typedef intptr_t atomic_ushort;
 typedef intptr_t atomic_int;
 typedef intptr_t atomic_uint;
 typedef intptr_t atomic_long;
 typedef intptr_t atomic_ulong;
 typedef intptr_t atomic_llong;
 typedef intptr_t atomic_ullong;
 typedef intptr_t atomic_wchar_t;
 typedef intptr_t atomic_int_least8_t;
 typedef intptr_t atomic_uint_least8_t;
 typedef intptr_t atomic_int_least16_t;
 typedef intptr_t atomic_uint_least16_t;
 typedef intptr_t atomic_int_least32_t;
 typedef intptr_t atomic_uint_least32_t;
 typedef intptr_t atomic_int_least64_t;
 typedef intptr_t atomic_uint_least64_t;
 typedef intptr_t atomic_int_fast8_t;
 typedef intptr_t atomic_uint_fast8_t;
 typedef intptr_t atomic_int_fast16_t;
 typedef intptr_t atomic_uint_fast16_t;
 typedef intptr_t atomic_int_fast32_t;
 typedef intptr_t atomic_uint_fast32_t;
 typedef intptr_t atomic_int_fast64_t;
 typedef intptr_t atomic_uint_fast64_t;
 typedef intptr_t atomic_intptr_t;
 typedef intptr_t atomic_uintptr_t;
 typedef intptr_t atomic_size_t;
 typedef intptr_t atomic_ptrdiff_t;
 typedef intptr_t atomic_intmax_t;
 typedef intptr_t atomic_uintmax_t;
 #define atomic_store(object, desired)   \
 do {                                    \
    *(object) = (desired);              \
 } while (0)
 #define atomic_store_explicit(object, desired, order) \
    atomic_store(object, desired)
 #define atomic_load(object) \
    (*(object))
 #define atomic_load_explicit(object, order) \
    atomic_load(object)
 static inline intptr_t atomic_exchange(intptr_t *object, intptr_t desired)
 {
    intptr_t ret = *object;
    *object = desired;
    return ret;
 }
 #define atomic_exchange_explicit(object, desired, order) \
    atomic_exchange(object, desired)
 static inline int atomic_compare_exchange_strong(intptr_t *object, intptr_t *expected,
                                                 intptr_t desired)
 {
    int ret;
    if (*object == *expected) {
        *object = desired;
        ret = 1;
    } else {
        *expected = *object;
        ret = 0;
    }
    return ret;
 }
 #define atomic_compare_exchange_strong_explicit(object, expected, desired, success, failure) \
    atomic_compare_exchange_strong(object, expected, desired)
 #define atomic_compare_exchange_weak(object, expected, desired) \
    atomic_compare_exchange_strong(object, expected, desired)
 #define atomic_compare_exchange_weak_explicit(object, expected, desired, success, failure) \
    atomic_compare_exchange_weak(object, expected, desired)
 #define FETCH_MODIFY(opname, op)                                            \
 static inline intptr_t atomic_fetch_ ## opname(intptr_t *object, intptr_t operand) \
 {                                                                    \
    intptr_t ret;                                                    \
    ret = *object;                                                   \
    *object = *object op operand;                                    \
    return ret;                                                      \
 }
 FETCH_MODIFY(add, +)
 FETCH_MODIFY(sub, -)
 FETCH_MODIFY(or,  |)
 FETCH_MODIFY(xor, ^)
 FETCH_MODIFY(and, &)
 #undef FETCH_MODIFY
 #define atomic_fetch_add_explicit(object, operand, order) \
    atomic_fetch_add(object, operand)
 #define atomic_fetch_sub_explicit(object, operand, order) \
    atomic_fetch_sub(object, operand)
 #define atomic_fetch_or_explicit(object, operand, order) \
    atomic_fetch_or(object, operand)
 #define atomic_fetch_xor_explicit(object, operand, order) \
    atomic_fetch_xor(object, operand)
 #define atomic_fetch_and_explicit(object, operand, order) \
    atomic_fetch_and(object, operand)
 #define atomic_flag_test_and_set(object) \
    atomic_exchange(object, 1)
 #define atomic_flag_test_and_set_explicit(object, order) \
    atomic_flag_test_and_set(object)
 #define atomic_flag_clear(object) \
    atomic_store(object, 0)
 #define atomic_flag_clear_explicit(object, order) \
    atomic_flag_clear(object)
 #endif /* COMPAT_ATOMICS_DUMMY_STDATOMIC_H */
--- a/compat/atomics/gcc/stdatomic.h
+++ b/compat/atomics/gcc/stdatomic.h
@@ -1,173 +0,0 @@
 /*
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
 /*
 * based on vlc_atomic.h from VLC
 * Copyright (C) 2010 Rémi Denis-Courmont
 */
 #ifndef COMPAT_ATOMICS_GCC_STDATOMIC_H
 #define COMPAT_ATOMICS_GCC_STDATOMIC_H
 #include <stddef.h>
 #include <stdint.h>
 #define ATOMIC_FLAG_INIT 0
 #define ATOMIC_VAR_INIT(value) (value)
 #define atomic_init(obj, value) \
 do {                            \
    *(obj) = (value);           \
 } while(0)
 #define kill_dependency(y) ((void)0)
 #define atomic_thread_fence(order) \
    __sync_synchronize()
 #define atomic_signal_fence(order) \
    ((void)0)
 #define atomic_is_lock_free(obj) 0
 typedef         _Bool      atomic_flag;
 typedef         _Bool      atomic_bool;
 typedef          char      atomic_char;
 typedef   signed char      atomic_schar;
 typedef unsigned char      atomic_uchar;
 typedef          short     atomic_short;
 typedef unsigned short     atomic_ushort;
 typedef          int       atomic_int;
 typedef unsigned int       atomic_uint;
 typedef          long      atomic_long;
 typedef unsigned long      atomic_ulong;
 typedef          long long atomic_llong;
 typedef unsigned long long atomic_ullong;
 typedef          wchar_t   atomic_wchar_t;
 typedef       int_least8_t atomic_int_least8_t;
 typedef      uint_least8_t atomic_uint_least8_t;
 typedef      int_least16_t atomic_int_least16_t;
 typedef     uint_least16_t atomic_uint_least16_t;
 typedef      int_least32_t atomic_int_least32_t;
 typedef     uint_least32_t atomic_uint_least32_t;
 typedef      int_least64_t atomic_int_least64_t;
 typedef     uint_least64_t atomic_uint_least64_t;
 typedef       int_fast8_t atomic_int_fast8_t;
 typedef      uint_fast8_t atomic_uint_fast8_t;
 typedef      int_fast16_t atomic_int_fast16_t;
 typedef     uint_fast16_t atomic_uint_fast16_t;
 typedef      int_fast32_t atomic_int_fast32_t;
 typedef     uint_fast32_t atomic_uint_fast32_t;
 typedef      int_fast64_t atomic_int_fast64_t;
 typedef     uint_fast64_t atomic_uint_fast64_t;
 typedef          intptr_t atomic_intptr_t;
 typedef         uintptr_t atomic_uintptr_t;
 typedef            size_t atomic_size_t;
 typedef         ptrdiff_t atomic_ptrdiff_t;
 typedef          intmax_t atomic_intmax_t;
 typedef         uintmax_t atomic_uintmax_t;
 #define atomic_store(object, desired)   \
 do {                                    \
    *(object) = (desired);              \
    __sync_synchronize();               \
 } while (0)
 #define atomic_store_explicit(object, desired, order) \
    atomic_store(object, desired)
 #define atomic_load(object) \
    (__sync_synchronize(), *(object))
 #define atomic_load_explicit(object, order) \
    atomic_load(object)
 #define atomic_exchange(object, desired)                            \
 ({                                                                  \
    __typeof__(object) _obj = (object);                             \
    __typeof__(*object) _old;                                       \
    do                                                              \
        _old = atomic_load(_obj);                                   \
    while (!__sync_bool_compare_and_swap(_obj, _old, (desired)));   \
    _old;                                                           \
 })
 #define atomic_exchange_explicit(object, desired, order) \
    atomic_exchange(object, desired)
 #define atomic_compare_exchange_strong(object, expected, desired)   \
 ({                                                                  \
    __typeof__(object) _exp = (expected);                           \
    __typeof__(*object) _old = *_exp;                               \
    *_exp = __sync_val_compare_and_swap((object), _old, (desired)); \
    *_exp == _old;                                                  \
 })
 #define atomic_compare_exchange_strong_explicit(object, expected, desired, success, failure) \
    atomic_compare_exchange_strong(object, expected, desired)
 #define atomic_compare_exchange_weak(object, expected, desired) \
    atomic_compare_exchange_strong(object, expected, desired)
 #define atomic_compare_exchange_weak_explicit(object, expected, desired, success, failure) \
    atomic_compare_exchange_weak(object, expected, desired)
 #define atomic_fetch_add(object, operand) \
    __sync_fetch_and_add(object, operand)
 #define atomic_fetch_add_explicit(object, operand, order) \
    atomic_fetch_add(object, operand)
 #define atomic_fetch_sub(object, operand) \
    __sync_fetch_and_sub(object, operand)
 #define atomic_fetch_sub_explicit(object, operand, order) \
    atomic_fetch_sub(object, operand)
 #define atomic_fetch_or(object, operand) \
    __sync_fetch_and_or(object, operand)
 #define atomic_fetch_or_explicit(object, operand, order) \
    atomic_fetch_or(object, operand)
 #define atomic_fetch_xor(object, operand) \
    __sync_fetch_and_xor(object, operand)
 #define atomic_fetch_xor_explicit(object, operand, order) \
    atomic_fetch_xor(object, operand)
 #define atomic_fetch_and(object, operand) \
    __sync_fetch_and_and(object, operand)
 #define atomic_fetch_and_explicit(object, operand, order) \
    atomic_fetch_and(object, operand)
 #define atomic_flag_test_and_set(object) \
    atomic_exchange(object, 1)
 #define atomic_flag_test_and_set_explicit(object, order) \
    atomic_flag_test_and_set(object)
 #define atomic_flag_clear(object) \
    atomic_store(object, 0)
 #define atomic_flag_clear_explicit(object, order) \
    atomic_flag_clear(object)
 #endif /* COMPAT_ATOMICS_GCC_STDATOMIC_H */
--- a/compat/atomics/pthread/stdatomic.c
+++ b/compat/atomics/pthread/stdatomic.c
@@ -1,39 +0,0 @@
 /*
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
 /*
 * based on vlc_atomic.h from VLC
 * Copyright (C) 2010 Rémi Denis-Courmont
 */
 #include <pthread.h>
 #include <stdint.h>
 #include "stdatomic.h"
 static pthread_mutex_t atomic_lock = PTHREAD_MUTEX_INITIALIZER;
 void avpriv_atomic_lock(void)
 {
    pthread_mutex_lock(&atomic_lock);
 }
 void avpriv_atomic_unlock(void)
 {
    pthread_mutex_unlock(&atomic_lock);
 }
--- a/compat/atomics/pthread/stdatomic.h
+++ b/compat/atomics/pthread/stdatomic.h
@@ -1,197 +0,0 @@
 /*
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
 /*
 * based on vlc_atomic.h from VLC
 * Copyright (C) 2010 Rémi Denis-Courmont
 */
 #ifndef COMPAT_ATOMICS_PTHREAD_STDATOMIC_H
 #define COMPAT_ATOMICS_PTHREAD_STDATOMIC_H
 #include <stdint.h>
 #define ATOMIC_FLAG_INIT 0
 #define ATOMIC_VAR_INIT(value) (value)
 #define atomic_init(obj, value) \
 do {                            \
    *(obj) = (value);           \
 } while(0)
 #define kill_dependency(y) ((void)0)
 #define atomic_signal_fence(order) \
    ((void)0)
 #define atomic_is_lock_free(obj) 0
 typedef intptr_t atomic_flag;
 typedef intptr_t atomic_bool;
 typedef intptr_t atomic_char;
 typedef intptr_t atomic_schar;
 typedef intptr_t atomic_uchar;
 typedef intptr_t atomic_short;
 typedef intptr_t atomic_ushort;
 typedef intptr_t atomic_int;
 typedef intptr_t atomic_uint;
 typedef intptr_t atomic_long;
 typedef intptr_t atomic_ulong;
 typedef intptr_t atomic_llong;
 typedef intptr_t atomic_ullong;
 typedef intptr_t atomic_wchar_t;
 typedef intptr_t atomic_int_least8_t;
 typedef intptr_t atomic_uint_least8_t;
 typedef intptr_t atomic_int_least16_t;
 typedef intptr_t atomic_uint_least16_t;
 typedef intptr_t atomic_int_least32_t;
 typedef intptr_t atomic_uint_least32_t;
 typedef intptr_t atomic_int_least64_t;
 typedef intptr_t atomic_uint_least64_t;
 typedef intptr_t atomic_int_fast8_t;
 typedef intptr_t atomic_uint_fast8_t;
 typedef intptr_t atomic_int_fast16_t;
 typedef intptr_t atomic_uint_fast16_t;
 typedef intptr_t atomic_int_fast32_t;
 typedef intptr_t atomic_uint_fast32_t;
 typedef intptr_t atomic_int_fast64_t;
 typedef intptr_t atomic_uint_fast64_t;
 typedef intptr_t atomic_intptr_t;
 typedef intptr_t atomic_uintptr_t;
 typedef intptr_t atomic_size_t;
 typedef intptr_t atomic_ptrdiff_t;
 typedef intptr_t atomic_intmax_t;
 typedef intptr_t atomic_uintmax_t;
 void avpriv_atomic_lock(void);
 void avpriv_atomic_unlock(void);
 static inline void atomic_thread_fence(int order)
 {
    avpriv_atomic_lock();
    avpriv_atomic_unlock();
 }
 static inline void atomic_store(intptr_t *object, intptr_t desired)
 {
    avpriv_atomic_lock();
    *object = desired;
    avpriv_atomic_unlock();
 }
 #define atomic_store_explicit(object, desired, order) \
    atomic_store(object, desired)
 static inline intptr_t atomic_load(intptr_t *object)
 {
    intptr_t ret;
    avpriv_atomic_lock();
    ret = *object;
    avpriv_atomic_unlock();
    return ret;
 }
 #define atomic_load_explicit(object, order) \
    atomic_load(object)
 static inline intptr_t atomic_exchange(intptr_t *object, intptr_t desired)
 {
    intptr_t ret;
    avpriv_atomic_lock();
    ret     = *object;
    *object = desired;
    avpriv_atomic_unlock();
    return ret;
 }
 #define atomic_exchange_explicit(object, desired, order) \
    atomic_exchange(object, desired)
 static inline int atomic_compare_exchange_strong(intptr_t *object, intptr_t *expected,
                                                 intptr_t desired)
 {
    int ret;
    avpriv_atomic_lock();
    if (*object == *expected) {
        ret     = 1;
        *object = desired;
    } else {
        ret = 0;
        *expected = *object;
    }
    avpriv_atomic_unlock();
    return ret;
 }
 #define atomic_compare_exchange_strong_explicit(object, expected, desired, success, failure) \
    atomic_compare_exchange_strong(object, expected, desired)
 #define atomic_compare_exchange_weak(object, expected, desired) \
    atomic_compare_exchange_strong(object, expected, desired)
 #define atomic_compare_exchange_weak_explicit(object, expected, desired, success, failure) \
    atomic_compare_exchange_weak(object, expected, desired)
 #define FETCH_MODIFY(opname, op)                                                   \
 static inline intptr_t atomic_fetch_ ## opname(intptr_t *object, intptr_t operand) \
 {                                                                                  \
    intptr_t ret;                                                                  \
    avpriv_atomic_lock();                                                          \
    ret = *object;                                                                 \
    *object = *object op operand;                                                  \
    avpriv_atomic_unlock();                                                        \
    return ret;                                                                    \
 }
 FETCH_MODIFY(add, +)
 FETCH_MODIFY(sub, -)
 FETCH_MODIFY(or,  |)
 FETCH_MODIFY(xor, ^)
 FETCH_MODIFY(and, &)
 #undef FETCH_MODIFY
 #define atomic_fetch_add_explicit(object, operand, order) \
    atomic_fetch_add(object, operand)
 #define atomic_fetch_sub_explicit(object, operand, order) \
    atomic_fetch_sub(object, operand)
 #define atomic_fetch_or_explicit(object, operand, order) \
    atomic_fetch_or(object, operand)
 #define atomic_fetch_xor_explicit(object, operand, order) \
    atomic_fetch_xor(object, operand)
 #define atomic_fetch_and_explicit(object, operand, order) \
    atomic_fetch_and(object, operand)
 #define atomic_flag_test_and_set(object) \
    atomic_exchange(object, 1)
 #define atomic_flag_test_and_set_explicit(object, order) \
    atomic_flag_test_and_set(object)
 #define atomic_flag_clear(object) \
    atomic_store(object, 0)
 #define atomic_flag_clear_explicit(object, order) \
    atomic_flag_clear(object)
 #endif /* COMPAT_ATOMICS_PTHREAD_STDATOMIC_H */
--- a/compat/atomics/suncc/stdatomic.h
+++ b/compat/atomics/suncc/stdatomic.h
@@ -1,186 +0,0 @@
 /*
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
 #ifndef COMPAT_ATOMICS_SUNCC_STDATOMIC_H
 #define COMPAT_ATOMICS_SUNCC_STDATOMIC_H
 #include <atomic.h>
 #include <mbarrier.h>
 #include <stddef.h>
 #include <stdint.h>
 #define ATOMIC_FLAG_INIT 0
 #define ATOMIC_VAR_INIT(value) (value)
 #define atomic_init(obj, value) \
 do {                            \
    *(obj) = (value);           \
 } while(0)
 #define kill_dependency(y) ((void)0)
 #define atomic_thread_fence(order) \
    __machine_rw_barrier();
 #define atomic_signal_fence(order) \
    ((void)0)
 #define atomic_is_lock_free(obj) 0
 typedef intptr_t atomic_flag;
 typedef intptr_t atomic_bool;
 typedef intptr_t atomic_char;
 typedef intptr_t atomic_schar;
 typedef intptr_t atomic_uchar;
 typedef intptr_t atomic_short;
 typedef intptr_t atomic_ushort;
 typedef intptr_t atomic_int;
 typedef intptr_t atomic_uint;
 typedef intptr_t atomic_long;
 typedef intptr_t atomic_ulong;
 typedef intptr_t atomic_llong;
 typedef intptr_t atomic_ullong;
 typedef intptr_t atomic_wchar_t;
 typedef intptr_t atomic_int_least8_t;
 typedef intptr_t atomic_uint_least8_t;
 typedef intptr_t atomic_int_least16_t;
 typedef intptr_t atomic_uint_least16_t;
 typedef intptr_t atomic_int_least32_t;
 typedef intptr_t atomic_uint_least32_t;
 typedef intptr_t atomic_int_least64_t;
 typedef intptr_t atomic_uint_least64_t;
 typedef intptr_t atomic_int_fast8_t;
 typedef intptr_t atomic_uint_fast8_t;
 typedef intptr_t atomic_int_fast16_t;
 typedef intptr_t atomic_uint_fast16_t;
 typedef intptr_t atomic_int_fast32_t;
 typedef intptr_t atomic_uint_fast32_t;
 typedef intptr_t atomic_int_fast64_t;
 typedef intptr_t atomic_uint_fast64_t;
 typedef intptr_t atomic_intptr_t;
 typedef intptr_t atomic_uintptr_t;
 typedef intptr_t atomic_size_t;
 typedef intptr_t atomic_ptrdiff_t;
 typedef intptr_t atomic_intmax_t;
 typedef intptr_t atomic_uintmax_t;
 static inline void atomic_store(intptr_t *object, intptr_t desired)
 {
    *object = desired;
    __machine_rw_barrier();
 }
 #define atomic_store_explicit(object, desired, order) \
    atomic_store(object, desired)
 static inline intptr_t atomic_load(intptr_t *object)
 {
    __machine_rw_barrier();
    return *object;
 }
 #define atomic_load_explicit(object, order) \
    atomic_load(object)
 #define atomic_exchange(object, desired) \
    atomic_swap_ptr(object, desired)
 #define atomic_exchange_explicit(object, desired, order) \
    atomic_exchange(object, desired)
 static inline int atomic_compare_exchange_strong(intptr_t *object, intptr_t *expected,
                                                 intptr_t desired)
 {
    intptr_t  old = *expected;
    *expected = (intptr_t)atomic_cas_ptr(object, (void *)old, (void *)desired);
    return *expected == old;
 }
 #define atomic_compare_exchange_strong_explicit(object, expected, desired, success, failure) \
    atomic_compare_exchange_strong(object, expected, desired)
 #define atomic_compare_exchange_weak(object, expected, desired) \
    atomic_compare_exchange_strong(object, expected, desired)
 #define atomic_compare_exchange_weak_explicit(object, expected, desired, success, failure) \
    atomic_compare_exchange_weak(object, expected, desired)
 static inline intptr_t atomic_fetch_add(intptr_t *object, intptr_t operand)
 {
    return atomic_add_ptr_nv(object, operand) - operand;
 }
 #define atomic_fetch_sub(object, operand) \
    atomic_fetch_add(object, -(operand))
 static inline intptr_t atomic_fetch_or(intptr_t *object, intptr_t operand)
 {
    intptr_t old;
    do {
        old = atomic_load(object);
    } while (!atomic_compare_exchange_strong(object, old, old | operand));
    return old;
 }
 static inline intptr_t atomic_fetch_xor(intptr_t *object, intptr_t operand)
 {
    intptr_t old;
    do {
        old = atomic_load(object);
    } while (!atomic_compare_exchange_strong(object, old, old ^ operand));
    return old;
 }
 static inline intptr_t atomic_fetch_and(intptr_t *object, intptr_t operand)
 {
    intptr_t old;
    do {
        old = atomic_load(object);
    } while (!atomic_compare_exchange_strong(object, old, old & operand));
    return old;
 }
 #define atomic_fetch_add_explicit(object, operand, order) \
    atomic_fetch_add(object, operand)
 #define atomic_fetch_sub_explicit(object, operand, order) \
    atomic_fetch_sub(object, operand)
 #define atomic_fetch_or_explicit(object, operand, order) \
    atomic_fetch_or(object, operand)
 #define atomic_fetch_xor_explicit(object, operand, order) \
    atomic_fetch_xor(object, operand)
 #define atomic_fetch_and_explicit(object, operand, order) \
    atomic_fetch_and(object, operand)
 #define atomic_flag_test_and_set(object) \
    atomic_exchange(object, 1)
 #define atomic_flag_test_and_set_explicit(object, order) \
    atomic_flag_test_and_set(object)
 #define atomic_flag_clear(object) \
    atomic_store(object, 0)
 #define atomic_flag_clear_explicit(object, order) \
    atomic_flag_clear(object)
 #endif /* COMPAT_ATOMICS_SUNCC_STDATOMIC_H */
--- a/compat/atomics/win32/stdatomic.h
+++ b/compat/atomics/win32/stdatomic.h
@@ -1,181 +0,0 @@
 /*
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
 #ifndef COMPAT_ATOMICS_WIN32_STDATOMIC_H
 #define COMPAT_ATOMICS_WIN32_STDATOMIC_H
 #define WIN32_LEAN_AND_MEAN
 #include <stddef.h>
 #include <stdint.h>
 #include <windows.h>
 #define ATOMIC_FLAG_INIT 0
 #define ATOMIC_VAR_INIT(value) (value)
 #define atomic_init(obj, value) \
 do {                            \
    *(obj) = (value);           \
 } while(0)
 #define kill_dependency(y) ((void)0)
 #define atomic_thread_fence(order) \
    MemoryBarrier();
 #define atomic_signal_fence(order) \
    ((void)0)
 #define atomic_is_lock_free(obj) 0
 typedef intptr_t atomic_flag;
 typedef intptr_t atomic_bool;
 typedef intptr_t atomic_char;
 typedef intptr_t atomic_schar;
 typedef intptr_t atomic_uchar;
 typedef intptr_t atomic_short;
 typedef intptr_t atomic_ushort;
 typedef intptr_t atomic_int;
 typedef intptr_t atomic_uint;
 typedef intptr_t atomic_long;
 typedef intptr_t atomic_ulong;
 typedef intptr_t atomic_llong;
 typedef intptr_t atomic_ullong;
 typedef intptr_t atomic_wchar_t;
 typedef intptr_t atomic_int_least8_t;
 typedef intptr_t atomic_uint_least8_t;
 typedef intptr_t atomic_int_least16_t;
 typedef intptr_t atomic_uint_least16_t;
 typedef intptr_t atomic_int_least32_t;
 typedef intptr_t atomic_uint_least32_t;
 typedef intptr_t atomic_int_least64_t;
 typedef intptr_t atomic_uint_least64_t;
 typedef intptr_t atomic_int_fast8_t;
 typedef intptr_t atomic_uint_fast8_t;
 typedef intptr_t atomic_int_fast16_t;
 typedef intptr_t atomic_uint_fast16_t;
 typedef intptr_t atomic_int_fast32_t;
 typedef intptr_t atomic_uint_fast32_t;
 typedef intptr_t atomic_int_fast64_t;
 typedef intptr_t atomic_uint_fast64_t;
 typedef intptr_t atomic_intptr_t;
 typedef intptr_t atomic_uintptr_t;
 typedef intptr_t atomic_size_t;
 typedef intptr_t atomic_ptrdiff_t;
 typedef intptr_t atomic_intmax_t;
 typedef intptr_t atomic_uintmax_t;
 #define atomic_store(object, desired)   \
 do {                                    \
    *(object) = (desired);              \
    MemoryBarrier();                    \
 } while (0)
 #define atomic_store_explicit(object, desired, order) \
    atomic_store(object, desired)
 #define atomic_load(object) \
    (MemoryBarrier(), *(object))
 #define atomic_load_explicit(object, order) \
    atomic_load(object)
 #define atomic_exchange(object, desired) \
    InterlockedExchangePointer((PVOID volatile *)object, (PVOID)desired)
 #define atomic_exchange_explicit(object, desired, order) \
    atomic_exchange(object, desired)
 static inline int atomic_compare_exchange_strong(intptr_t *object, intptr_t *expected,
                                                 intptr_t desired)
 {
    intptr_t old = *expected;
    *expected = (intptr_t)InterlockedCompareExchangePointer(
        (PVOID *)object, (PVOID)desired, (PVOID)old);
    return *expected == old;
 }
 #define atomic_compare_exchange_strong_explicit(object, expected, desired, success, failure) \
    atomic_compare_exchange_strong(object, expected, desired)
 #define atomic_compare_exchange_weak(object, expected, desired) \
    atomic_compare_exchange_strong(object, expected, desired)
 #define atomic_compare_exchange_weak_explicit(object, expected, desired, success, failure) \
    atomic_compare_exchange_weak(object, expected, desired)
 #ifdef _WIN64
 #define atomic_fetch_add(object, operand) \
    InterlockedExchangeAdd64(object, operand)
 #define atomic_fetch_sub(object, operand) \
    InterlockedExchangeAdd64(object, -(operand))
 #define atomic_fetch_or(object, operand) \
    InterlockedOr64(object, operand)
 #define atomic_fetch_xor(object, operand) \
    InterlockedXor64(object, operand)
 #define atomic_fetch_and(object, operand) \
    InterlockedAnd64(object, operand)
 #else
 #define atomic_fetch_add(object, operand) \
    InterlockedExchangeAdd(object, operand)
 #define atomic_fetch_sub(object, operand) \
    InterlockedExchangeAdd(object, -(operand))
 #define atomic_fetch_or(object, operand) \
    InterlockedOr(object, operand)
 #define atomic_fetch_xor(object, operand) \
    InterlockedXor(object, operand)
 #define atomic_fetch_and(object, operand) \
    InterlockedAnd(object, operand)
 #endif /* _WIN64 */
 #define atomic_fetch_add_explicit(object, operand, order) \
    atomic_fetch_add(object, operand)
 #define atomic_fetch_sub_explicit(object, operand, order) \
    atomic_fetch_sub(object, operand)
 #define atomic_fetch_or_explicit(object, operand, order) \
    atomic_fetch_or(object, operand)
 #define atomic_fetch_xor_explicit(object, operand, order) \
    atomic_fetch_xor(object, operand)
 #define atomic_fetch_and_explicit(object, operand, order) \
    atomic_fetch_and(object, operand)
 #define atomic_flag_test_and_set(object) \
    atomic_exchange(object, 1)
 #define atomic_flag_test_and_set_explicit(object, order) \
    atomic_flag_test_and_set(object)
 #define atomic_flag_clear(object) \
    atomic_store(object, 0)
 #define atomic_flag_clear_explicit(object, order) \
    atomic_flag_clear(object)
 #endif /* COMPAT_ATOMICS_WIN32_STDATOMIC_H */
--- a/compat/avisynth/avisynth_c.h
+++ b/compat/avisynth/avisynth_c.h
@@ -0,0 +1,880 @@
 // Avisynth C Interface Version 0.20
 // Copyright 2003 Kevin Atkinson
 // This program is free software; you can redistribute it and/or modify
 // it under the terms of the GNU General Public License as published by
 // the Free Software Foundation; either version 2 of the License, or
 // (at your option) any later version.
 //
 // This program is distributed in the hope that it will be useful,
 // but WITHOUT ANY WARRANTY; without even the implied warranty of
 // MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 // GNU General Public License for more details.
 //
 // You should have received a copy of the GNU General Public License
 // along with this program; if not, write to the Free Software
 // Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston,
 // MA 02110-1301 USA, or visit
 // http://www.gnu.org/copyleft/gpl.html .
 //
 // As a special exception, I give you permission to link to the
 // Avisynth C interface with independent modules that communicate with
 // the Avisynth C interface solely through the interfaces defined in
 // avisynth_c.h, regardless of the license terms of these independent
 // modules, and to copy and distribute the resulting combined work
 // under terms of your choice, provided that every copy of the
 // combined work is accompanied by a complete copy of the source code
 // of the Avisynth C interface and Avisynth itself (with the version
 // used to produce the combined work), being distributed under the
 // terms of the GNU General Public License plus this exception.  An
 // independent module is a module which is not derived from or based
 // on Avisynth C Interface, such as 3rd-party filters, import and
 // export plugins, or graphical user interfaces.
 // 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.
 #ifndef __AVISYNTH_C__
 #define __AVISYNTH_C__
 #ifdef __cplusplus
 #  define EXTERN_C extern "C"
 #else
 #  define EXTERN_C
 #endif
 #define AVSC_USE_STDCALL 1
 #ifndef AVSC_USE_STDCALL
 #  define AVSC_CC __cdecl
 #else
 #  define AVSC_CC __stdcall
 #endif
 #define AVSC_INLINE static __inline
 #ifdef AVISYNTH_C_EXPORTS
 #  define AVSC_EXPORT EXTERN_C
 #  define AVSC_API(ret, name) EXTERN_C __declspec(dllexport) ret AVSC_CC name
 #else
 #  define AVSC_EXPORT EXTERN_C __declspec(dllexport)
 #  ifndef AVSC_NO_DECLSPEC
 #    define AVSC_API(ret, name) EXTERN_C __declspec(dllimport) ret AVSC_CC name
 #  else
 #    define AVSC_API(ret, name) typedef ret (AVSC_CC *name##_func)
 #  endif
 #endif
 typedef unsigned char BYTE;
 #ifdef __GNUC__
 typedef long long int INT64;
 #else
 typedef __int64 INT64;
 #endif
 /////////////////////////////////////////////////////////////////////
 //
 // Constants
 //
 #ifndef __AVISYNTH_H__
 enum { AVISYNTH_INTERFACE_VERSION = 4 };
 #endif
 enum {AVS_SAMPLE_INT8  = 1<<0,
      AVS_SAMPLE_INT16 = 1<<1,
      AVS_SAMPLE_INT24 = 1<<2,
      AVS_SAMPLE_INT32 = 1<<3,
      AVS_SAMPLE_FLOAT = 1<<4};
 enum {AVS_PLANAR_Y=1<<0,
      AVS_PLANAR_U=1<<1,
      AVS_PLANAR_V=1<<2,
      AVS_PLANAR_ALIGNED=1<<3,
      AVS_PLANAR_Y_ALIGNED=AVS_PLANAR_Y|AVS_PLANAR_ALIGNED,
      AVS_PLANAR_U_ALIGNED=AVS_PLANAR_U|AVS_PLANAR_ALIGNED,
      AVS_PLANAR_V_ALIGNED=AVS_PLANAR_V|AVS_PLANAR_ALIGNED,
      AVS_PLANAR_A=1<<4,
      AVS_PLANAR_R=1<<5,
      AVS_PLANAR_G=1<<6,
      AVS_PLANAR_B=1<<7,
      AVS_PLANAR_A_ALIGNED=AVS_PLANAR_A|AVS_PLANAR_ALIGNED,
      AVS_PLANAR_R_ALIGNED=AVS_PLANAR_R|AVS_PLANAR_ALIGNED,
      AVS_PLANAR_G_ALIGNED=AVS_PLANAR_G|AVS_PLANAR_ALIGNED,
      AVS_PLANAR_B_ALIGNED=AVS_PLANAR_B|AVS_PLANAR_ALIGNED};
  // Colorspace properties.
 enum {AVS_CS_BGR = 1<<28,
      AVS_CS_YUV = 1<<29,
      AVS_CS_INTERLEAVED = 1<<30,
      AVS_CS_PLANAR = 1<<31,
      AVS_CS_SHIFT_SUB_WIDTH   = 0,
      AVS_CS_SHIFT_SUB_HEIGHT  = 1 << 3,
      AVS_CS_SHIFT_SAMPLE_BITS = 1 << 4,
      AVS_CS_SUB_WIDTH_MASK    = 7 << AVS_CS_SHIFT_SUB_WIDTH,
      AVS_CS_SUB_WIDTH_1       = 3 << AVS_CS_SHIFT_SUB_WIDTH, // YV24
      AVS_CS_SUB_WIDTH_2       = 0 << AVS_CS_SHIFT_SUB_WIDTH, // YV12, I420, YV16
      AVS_CS_SUB_WIDTH_4       = 1 << AVS_CS_SHIFT_SUB_WIDTH, // YUV9, YV411
      AVS_CS_VPLANEFIRST       = 1 << 3, // YV12, YV16, YV24, YV411, YUV9
      AVS_CS_UPLANEFIRST       = 1 << 4, // I420
      AVS_CS_SUB_HEIGHT_MASK   = 7 << AVS_CS_SHIFT_SUB_HEIGHT,
      AVS_CS_SUB_HEIGHT_1      = 3 << AVS_CS_SHIFT_SUB_HEIGHT, // YV16, YV24, YV411
      AVS_CS_SUB_HEIGHT_2      = 0 << AVS_CS_SHIFT_SUB_HEIGHT, // YV12, I420
      AVS_CS_SUB_HEIGHT_4      = 1 << AVS_CS_SHIFT_SUB_HEIGHT, // YUV9
      AVS_CS_SAMPLE_BITS_MASK  = 7 << AVS_CS_SHIFT_SAMPLE_BITS,
      AVS_CS_SAMPLE_BITS_8     = 0 << AVS_CS_SHIFT_SAMPLE_BITS,
      AVS_CS_SAMPLE_BITS_16    = 1 << AVS_CS_SHIFT_SAMPLE_BITS,
      AVS_CS_SAMPLE_BITS_32    = 2 << AVS_CS_SHIFT_SAMPLE_BITS,
      AVS_CS_PLANAR_MASK       = AVS_CS_PLANAR | AVS_CS_INTERLEAVED | AVS_CS_YUV | AVS_CS_BGR | AVS_CS_SAMPLE_BITS_MASK | AVS_CS_SUB_HEIGHT_MASK | AVS_CS_SUB_WIDTH_MASK,
      AVS_CS_PLANAR_FILTER     = ~( AVS_CS_VPLANEFIRST | AVS_CS_UPLANEFIRST )};
  // Specific colorformats
 enum {
  AVS_CS_UNKNOWN = 0,
  AVS_CS_BGR24 = 1<<0 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
  AVS_CS_BGR32 = 1<<1 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
  AVS_CS_YUY2 = 1<<2 | AVS_CS_YUV | AVS_CS_INTERLEAVED,
  //  AVS_CS_YV12  = 1<<3  Reserved
  //  AVS_CS_I420  = 1<<4  Reserved
  AVS_CS_RAW32 = 1<<5 | AVS_CS_INTERLEAVED,
  AVS_CS_YV24  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_1,  // YVU 4:4:4 planar
  AVS_CS_YV16  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_2,  // YVU 4:2:2 planar
  AVS_CS_YV12  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_2 | AVS_CS_SUB_WIDTH_2,  // 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,  // 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_PLANAR | AVS_CS_INTERLEAVED | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8                                              // Y   4:0:0 planar
 };
 enum {
  AVS_IT_BFF = 1<<0,
  AVS_IT_TFF = 1<<1,
  AVS_IT_FIELDBASED = 1<<2};
 enum {
  AVS_FILTER_TYPE=1,
  AVS_FILTER_INPUT_COLORSPACE=2,
  AVS_FILTER_OUTPUT_TYPE=9,
  AVS_FILTER_NAME=4,
  AVS_FILTER_AUTHOR=5,
  AVS_FILTER_VERSION=6,
  AVS_FILTER_ARGS=7,
  AVS_FILTER_ARGS_INFO=8,
  AVS_FILTER_ARGS_DESCRIPTION=10,
  AVS_FILTER_DESCRIPTION=11};
 enum {  //SUBTYPES
  AVS_FILTER_TYPE_AUDIO=1,
  AVS_FILTER_TYPE_VIDEO=2,
  AVS_FILTER_OUTPUT_TYPE_SAME=3,
  AVS_FILTER_OUTPUT_TYPE_DIFFERENT=4};
 enum {
  AVS_CACHE_NOTHING=0,
  AVS_CACHE_RANGE=1,
  AVS_CACHE_ALL=2,
  AVS_CACHE_AUDIO=3,
  AVS_CACHE_AUDIO_NONE=4,
  AVS_CACHE_AUDIO_AUTO=5
  };
 #define AVS_FRAME_ALIGN 16
 typedef struct AVS_Clip AVS_Clip;
 typedef struct AVS_ScriptEnvironment AVS_ScriptEnvironment;
 /////////////////////////////////////////////////////////////////////
 //
 // AVS_VideoInfo
 //
 // AVS_VideoInfo is layed out identicly to VideoInfo
 typedef struct AVS_VideoInfo {
  int width, height;    // width=0 means no video
  unsigned fps_numerator, fps_denominator;
  int num_frames;
  int pixel_type;
  int audio_samples_per_second;   // 0 means no audio
  int sample_type;
  INT64 num_audio_samples;
  int nchannels;
  // Imagetype properties
  int image_type;
 } AVS_VideoInfo;
 // useful functions of the above
 AVSC_INLINE int avs_has_video(const AVS_VideoInfo * p)
        { return (p->width!=0); }
 AVSC_INLINE int avs_has_audio(const AVS_VideoInfo * p)
        { return (p->audio_samples_per_second!=0); }
 AVSC_INLINE int avs_is_rgb(const AVS_VideoInfo * p)
        { return !!(p->pixel_type&AVS_CS_BGR); }
 AVSC_INLINE int avs_is_rgb24(const AVS_VideoInfo * p)
        { return (p->pixel_type&AVS_CS_BGR24)==AVS_CS_BGR24; } // Clear out additional properties
 AVSC_INLINE int avs_is_rgb32(const AVS_VideoInfo * p)
        { return (p->pixel_type & AVS_CS_BGR32) == AVS_CS_BGR32 ; }
 AVSC_INLINE int avs_is_yuv(const AVS_VideoInfo * p)
        { return !!(p->pixel_type&AVS_CS_YUV ); }
 AVSC_INLINE int avs_is_yuy2(const AVS_VideoInfo * p)
        { return (p->pixel_type & AVS_CS_YUY2) == AVS_CS_YUY2; }
 AVSC_INLINE int avs_is_yv24(const AVS_VideoInfo * p)
        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV24  & AVS_CS_PLANAR_FILTER); }
 AVSC_INLINE int avs_is_yv16(const AVS_VideoInfo * p)
        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV16  & AVS_CS_PLANAR_FILTER); }
 AVSC_INLINE int avs_is_yv12(const AVS_VideoInfo * p)
        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV12  & AVS_CS_PLANAR_FILTER); }
 AVSC_INLINE int avs_is_yv411(const AVS_VideoInfo * p)
        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV411 & AVS_CS_PLANAR_FILTER); }
 AVSC_INLINE int avs_is_y8(const AVS_VideoInfo * p)
        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_Y8    & AVS_CS_PLANAR_FILTER); }
 AVSC_INLINE int avs_is_property(const AVS_VideoInfo * p, int property)
        { return ((p->pixel_type & property)==property ); }
 AVSC_INLINE int avs_is_planar(const AVS_VideoInfo * p)
        { return !!(p->pixel_type & AVS_CS_PLANAR); }
 AVSC_INLINE int avs_is_color_space(const AVS_VideoInfo * p, int c_space)
        { return avs_is_planar(p) ? ((p->pixel_type & AVS_CS_PLANAR_MASK) == (c_space & AVS_CS_PLANAR_FILTER)) : ((p->pixel_type & c_space) == 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_INLINE int avs_bits_per_pixel(const AVS_VideoInfo * p)
 {
  switch (p->pixel_type) {
      case AVS_CS_BGR24: return 24;
      case AVS_CS_BGR32: return 32;
      case AVS_CS_YUY2:  return 16;
      case AVS_CS_YV12:
      case AVS_CS_I420:  return 12;
      default:           return 0;
    }
 }
 AVSC_INLINE int avs_bytes_from_pixels(const AVS_VideoInfo * p, int pixels)
        { return pixels * (avs_bits_per_pixel(p)>>3); }   // Will work on planar images, but will return only luma planes
 AVSC_INLINE int avs_row_size(const AVS_VideoInfo * p)
        { return avs_bytes_from_pixels(p,p->width); }  // Also only returns first plane on planar images
 AVSC_INLINE int avs_bmp_size(const AVS_VideoInfo * vi)
        { if (avs_is_planar(vi)) {int p = vi->height * ((avs_row_size(vi)+3) & ~3); p+=p>>1; return p;  } return vi->height * ((avs_row_size(vi)+3) & ~3); }
 AVSC_INLINE int avs_samples_per_second(const AVS_VideoInfo * p)
        { return p->audio_samples_per_second; }
 AVSC_INLINE int avs_bytes_per_channel_sample(const AVS_VideoInfo * p)
 {
    switch (p->sample_type) {
      case AVS_SAMPLE_INT8:  return sizeof(signed char);
      case AVS_SAMPLE_INT16: return sizeof(signed short);
      case AVS_SAMPLE_INT24: return 3;
      case AVS_SAMPLE_INT32: return sizeof(signed int);
      case AVS_SAMPLE_FLOAT: return sizeof(float);
      default: return 0;
    }
 }
 AVSC_INLINE int avs_bytes_per_audio_sample(const AVS_VideoInfo * p)
        { return p->nchannels*avs_bytes_per_channel_sample(p);}
 AVSC_INLINE INT64 avs_audio_samples_from_frames(const AVS_VideoInfo * p, INT64 frames)
        { return ((INT64)(frames) * p->audio_samples_per_second * p->fps_denominator / p->fps_numerator); }
 AVSC_INLINE int avs_frames_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
        { return (int)(samples * (INT64)p->fps_numerator / (INT64)p->fps_denominator / (INT64)p->audio_samples_per_second); }
 AVSC_INLINE INT64 avs_audio_samples_from_bytes(const AVS_VideoInfo * p, INT64 bytes)
        { return bytes / avs_bytes_per_audio_sample(p); }
 AVSC_INLINE INT64 avs_bytes_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
        { return samples * avs_bytes_per_audio_sample(p); }
 AVSC_INLINE int avs_audio_channels(const AVS_VideoInfo * p)
        { return p->nchannels; }
 AVSC_INLINE int avs_sample_type(const AVS_VideoInfo * p)
        { return p->sample_type;}
 // useful mutator
 AVSC_INLINE void avs_set_property(AVS_VideoInfo * p, int property)
        { p->image_type|=property; }
 AVSC_INLINE void avs_clear_property(AVS_VideoInfo * p, int property)
        { p->image_type&=~property; }
 AVSC_INLINE void avs_set_field_based(AVS_VideoInfo * p, int isfieldbased)
        { if (isfieldbased) p->image_type|=AVS_IT_FIELDBASED; else p->image_type&=~AVS_IT_FIELDBASED; }
 AVSC_INLINE void avs_set_fps(AVS_VideoInfo * p, unsigned numerator, unsigned denominator)
 {
    unsigned x=numerator, y=denominator;
    while (y) {   // find gcd
      unsigned t = x%y; x = y; y = t;
    }
    p->fps_numerator = numerator/x;
    p->fps_denominator = denominator/x;
 }
 AVSC_INLINE int avs_is_same_colorspace(AVS_VideoInfo * x, AVS_VideoInfo * y)
 {
        return (x->pixel_type == y->pixel_type)
                || (avs_is_yv12(x) && avs_is_yv12(y));
 }
 /////////////////////////////////////////////////////////////////////
 //
 // AVS_VideoFrame
 //
 // VideoFrameBuffer holds information about a memory block which is used
 // for video data.  For efficiency, instances of this class are not deleted
 // when the refcount reaches zero; instead they're stored in a linked list
 // to be reused.  The instances are deleted when the corresponding AVS
 // file is closed.
 // AVS_VideoFrameBuffer is layed out identicly to VideoFrameBuffer
 // DO NOT USE THIS STRUCTURE DIRECTLY
 typedef struct AVS_VideoFrameBuffer {
  BYTE * data;
  int data_size;
  // sequence_number is incremented every time the buffer is changed, so
  // that stale views can tell they're no longer valid.
  volatile long sequence_number;
  volatile long refcount;
 } AVS_VideoFrameBuffer;
 // VideoFrame holds a "window" into a VideoFrameBuffer.
 // AVS_VideoFrame is layed out identicly to IVideoFrame
 // DO NOT USE THIS STRUCTURE DIRECTLY
 typedef struct AVS_VideoFrame {
  volatile long refcount;
  AVS_VideoFrameBuffer * vfb;
  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_INLINE int avs_get_pitch(const AVS_VideoFrame * p) {
        return p->pitch;}
 AVSC_INLINE int avs_get_pitch_p(const AVS_VideoFrame * p, int plane) {
  switch (plane) {
  case AVS_PLANAR_U: case AVS_PLANAR_V: return p->pitchUV;}
  return p->pitch;}
 AVSC_INLINE int avs_get_row_size(const AVS_VideoFrame * p) {
        return p->row_size; }
 AVSC_INLINE int avs_get_row_size_p(const AVS_VideoFrame * p, int plane) {
        int r;
    switch (plane) {
    case AVS_PLANAR_U: case AVS_PLANAR_V:
                if (p->pitchUV) return p->row_sizeUV;
                else            return 0;
    case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED:
                if (p->pitchUV) {
                        r = (p->row_sizeUV+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
                        if (r < p->pitchUV)
                                return r;
                        return p->row_sizeUV;
                } else return 0;
    case AVS_PLANAR_Y_ALIGNED:
                r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
                if (r <= p->pitch)
                        return r;
                return p->row_size;
    }
    return p->row_size;
 }
 AVSC_INLINE int avs_get_height(const AVS_VideoFrame * p) {
        return p->height;}
 AVSC_INLINE int avs_get_height_p(const AVS_VideoFrame * p, int plane) {
        switch (plane) {
                case AVS_PLANAR_U: case AVS_PLANAR_V:
                        if (p->pitchUV) return p->heightUV;
                        return 0;
        }
        return p->height;}
 AVSC_INLINE const BYTE* avs_get_read_ptr(const AVS_VideoFrame * p) {
        return p->vfb->data + p->offset;}
 AVSC_INLINE const BYTE* avs_get_read_ptr_p(const AVS_VideoFrame * p, int plane)
 {
        switch (plane) {
                case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
                case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
                default:           return p->vfb->data + p->offset;}
 }
 AVSC_INLINE int avs_is_writable(const AVS_VideoFrame * p) {
        return (p->refcount == 1 && p->vfb->refcount == 1);}
 AVSC_INLINE BYTE* avs_get_write_ptr(const AVS_VideoFrame * p)
 {
        if (avs_is_writable(p)) {
                ++p->vfb->sequence_number;
                return p->vfb->data + p->offset;
        } else
                return 0;
 }
 AVSC_INLINE BYTE* avs_get_write_ptr_p(const AVS_VideoFrame * p, int plane)
 {
        if (plane==AVS_PLANAR_Y && avs_is_writable(p)) {
                ++p->vfb->sequence_number;
                return p->vfb->data + p->offset;
        } else if (plane==AVS_PLANAR_Y) {
                return 0;
        } else {
                switch (plane) {
                        case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
                        case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
                        default:       return p->vfb->data + p->offset;
                }
        }
 }
 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
 AVSC_INLINE void avs_release_frame(AVS_VideoFrame * f)
  {avs_release_video_frame(f);}
 AVSC_INLINE AVS_VideoFrame * avs_copy_frame(AVS_VideoFrame * f)
  {return avs_copy_video_frame(f);}
 #endif
 /////////////////////////////////////////////////////////////////////
 //
 // AVS_Value
 //
 // Treat AVS_Value as a fat pointer.  That is use avs_copy_value
 // and avs_release_value appropiaty as you would if AVS_Value was
 // a pointer.
 // To maintain source code compatibility with future versions of the
 // avisynth_c API don't use the AVS_Value directly.  Use the helper
 // functions below.
 // AVS_Value is layed out identicly to AVSValue
 typedef struct AVS_Value AVS_Value;
 struct AVS_Value {
  short type;  // 'a'rray, 'c'lip, 'b'ool, 'i'nt, 'f'loat, 's'tring, 'v'oid, or 'l'ong
               // for some function e'rror
  short array_size;
  union {
    void * clip; // do not use directly, use avs_take_clip
    char boolean;
    int integer;
    float floating_pt;
    const char * string;
    const AVS_Value * array;
  } d;
 };
 // AVS_Value should be initilized with avs_void.
 // Should also set to avs_void after the value is released
 // with avs_copy_value.  Consider it the equalvent of setting
 // a pointer to NULL
 static const AVS_Value avs_void = {'v'};
 AVSC_API(void, avs_copy_value)(AVS_Value * dest, AVS_Value src);
 AVSC_API(void, avs_release_value)(AVS_Value);
 AVSC_INLINE int avs_defined(AVS_Value v) { return v.type != 'v'; }
 AVSC_INLINE int avs_is_clip(AVS_Value v) { return v.type == 'c'; }
 AVSC_INLINE int avs_is_bool(AVS_Value v) { return v.type == 'b'; }
 AVSC_INLINE int avs_is_int(AVS_Value v) { return v.type == 'i'; }
 AVSC_INLINE int avs_is_float(AVS_Value v) { return v.type == 'f' || v.type == 'i'; }
 AVSC_INLINE int avs_is_string(AVS_Value v) { return v.type == 's'; }
 AVSC_INLINE int avs_is_array(AVS_Value v) { return v.type == 'a'; }
 AVSC_INLINE int avs_is_error(AVS_Value v) { return v.type == 'e'; }
 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)
        { return v.d.integer; }
 AVSC_INLINE const char * avs_as_string(AVS_Value v)
        { return avs_is_error(v) || avs_is_string(v) ? v.d.string : 0; }
 AVSC_INLINE double avs_as_float(AVS_Value v)
        { return avs_is_int(v) ? v.d.integer : v.d.floating_pt; }
 AVSC_INLINE const char * avs_as_error(AVS_Value v)
        { return avs_is_error(v) ? v.d.string : 0; }
 AVSC_INLINE const AVS_Value * avs_as_array(AVS_Value v)
        { return v.d.array; }
 AVSC_INLINE int avs_array_size(AVS_Value v)
        { return avs_is_array(v) ? v.array_size : 1; }
 AVSC_INLINE AVS_Value avs_array_elt(AVS_Value v, int index)
        { return avs_is_array(v) ? v.d.array[index] : v; }
 // only use these functions on an AVS_Value that does not already have
 // an active value.  Remember, treat AVS_Value as a fat pointer.
 AVSC_INLINE AVS_Value avs_new_value_bool(int v0)
        { AVS_Value v; v.type = 'b'; v.d.boolean = v0 == 0 ? 0 : 1; return v; }
 AVSC_INLINE AVS_Value avs_new_value_int(int v0)
        { AVS_Value v; v.type = 'i'; v.d.integer = v0; return v; }
 AVSC_INLINE AVS_Value avs_new_value_string(const char * v0)
        { AVS_Value v; v.type = 's'; v.d.string = v0; return v; }
 AVSC_INLINE AVS_Value avs_new_value_float(float v0)
        { AVS_Value v; v.type = 'f'; v.d.floating_pt = v0; return v;}
 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
 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 = size; return v; }
 /////////////////////////////////////////////////////////////////////
 //
 // AVS_Clip
 //
 AVSC_API(void, avs_release_clip)(AVS_Clip *);
 AVSC_API(AVS_Clip *, avs_copy_clip)(AVS_Clip *);
 AVSC_API(const char *, avs_clip_get_error)(AVS_Clip *); // return 0 if no error
 AVSC_API(const AVS_VideoInfo *, avs_get_video_info)(AVS_Clip *);
 AVSC_API(int, avs_get_version)(AVS_Clip *);
 AVSC_API(AVS_VideoFrame *, avs_get_frame)(AVS_Clip *, int n);
 // The returned video frame must be released with avs_release_video_frame
 AVSC_API(int, avs_get_parity)(AVS_Clip *, int n);
 // return field parity if field_based, else parity of first field in frame
 AVSC_API(int, avs_get_audio)(AVS_Clip *, void * buf,
                             INT64 start, INT64 count);
 // start and count are in samples
 AVSC_API(int, avs_set_cache_hints)(AVS_Clip *,
                                   int cachehints, int frame_range);
 // This is the callback type used by avs_add_function
 typedef AVS_Value (AVSC_CC * AVS_ApplyFunc)
                        (AVS_ScriptEnvironment *, AVS_Value args, void * user_data);
 typedef struct AVS_FilterInfo AVS_FilterInfo;
 struct AVS_FilterInfo
 {
  // these members should not be modified outside of the AVS_ApplyFunc callback
  AVS_Clip * child;
  AVS_VideoInfo vi;
  AVS_ScriptEnvironment * env;
  AVS_VideoFrame * (AVSC_CC * get_frame)(AVS_FilterInfo *, int n);
  int (AVSC_CC * get_parity)(AVS_FilterInfo *, int n);
  int (AVSC_CC * get_audio)(AVS_FilterInfo *, void * buf,
                                  INT64 start, INT64 count);
  int (AVSC_CC * set_cache_hints)(AVS_FilterInfo *, int cachehints,
                                        int frame_range);
  void (AVSC_CC * free_filter)(AVS_FilterInfo *);
  // Should be set when ever there is an error to report.
  // It is cleared before any of the above methods are called
  const char * error;
  // this is to store whatever and may be modified at will
  void * user_data;
 };
 // Create a new filter
 // fi is set to point to the AVS_FilterInfo so that you can
 //   modify it once it is initilized.
 // store_child should generally be set to true.  If it is not
 //    set than ALL methods (the function pointers) must be defined
 // If it is set than you do not need to worry about freeing the child
 //    clip.
 AVSC_API(AVS_Clip *, avs_new_c_filter)(AVS_ScriptEnvironment * e,
                                       AVS_FilterInfo * * fi,
                                       AVS_Value child, int store_child);
 /////////////////////////////////////////////////////////////////////
 //
 // AVS_ScriptEnvironment
 //
 // For GetCPUFlags.  These are backwards-compatible with those in VirtualDub.
 enum {
                                /* slowest CPU to support extension */
  AVS_CPU_FORCE        = 0x01,   // N/A
  AVS_CPU_FPU          = 0x02,   // 386/486DX
  AVS_CPU_MMX          = 0x04,   // P55C, K6, PII
  AVS_CPU_INTEGER_SSE  = 0x08,   // PIII, Athlon
  AVS_CPU_SSE          = 0x10,   // PIII, Athlon XP/MP
  AVS_CPU_SSE2         = 0x20,   // PIV, Hammer
  AVS_CPU_3DNOW        = 0x40,   // K6-2
  AVS_CPU_3DNOW_EXT    = 0x80,   // Athlon
  AVS_CPU_X86_64       = 0xA0,   // Hammer (note: equiv. to 3DNow + SSE2,
                                 // which only Hammer will have anyway)
  AVS_CPUF_SSE3       = 0x100,   //  PIV+, K8 Venice
  AVS_CPUF_SSSE3      = 0x200,   //  Core 2
  AVS_CPUF_SSE4       = 0x400,   //  Penryn, Wolfdale, Yorkfield
  AVS_CPUF_SSE4_1     = 0x400,
  AVS_CPUF_SSE4_2     = 0x800,   //  Nehalem
 };
 AVSC_API(const char *, avs_get_error)(AVS_ScriptEnvironment *); // return 0 if no error
 AVSC_API(long, avs_get_cpu_flags)(AVS_ScriptEnvironment *);
 AVSC_API(int, avs_check_version)(AVS_ScriptEnvironment *, int version);
 AVSC_API(char *, avs_save_string)(AVS_ScriptEnvironment *, const char* s, int length);
 AVSC_API(char *, avs_sprintf)(AVS_ScriptEnvironment *, const char * fmt, ...);
 AVSC_API(char *, avs_vsprintf)(AVS_ScriptEnvironment *, const char * fmt, void* val);
 // note: val is really a va_list; I hope everyone typedefs va_list to a pointer
 AVSC_API(int, avs_add_function)(AVS_ScriptEnvironment *,
                                const char * name, const char * params,
                                AVS_ApplyFunc apply, void * user_data);
 AVSC_API(int, avs_function_exists)(AVS_ScriptEnvironment *, const char * name);
 AVSC_API(AVS_Value, avs_invoke)(AVS_ScriptEnvironment *, const char * name,
                               AVS_Value args, const char** arg_names);
 // The returned value must be be released with avs_release_value
 AVSC_API(AVS_Value, avs_get_var)(AVS_ScriptEnvironment *, const char* name);
 // The returned value must be be released with avs_release_value
 AVSC_API(int, avs_set_var)(AVS_ScriptEnvironment *, const char* name, AVS_Value val);
 AVSC_API(int, avs_set_global_var)(AVS_ScriptEnvironment *, const char* name, const AVS_Value val);
 //void avs_push_context(AVS_ScriptEnvironment *, int level=0);
 //void avs_pop_context(AVS_ScriptEnvironment *);
 AVSC_API(AVS_VideoFrame *, avs_new_video_frame_a)(AVS_ScriptEnvironment *,
                                          const AVS_VideoInfo * vi, int align);
 // align should be at least 16
 #ifndef AVSC_NO_DECLSPEC
 AVSC_INLINE
 AVS_VideoFrame * avs_new_video_frame(AVS_ScriptEnvironment * env,
                                     const AVS_VideoInfo * vi)
  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
 AVSC_INLINE
 AVS_VideoFrame * avs_new_frame(AVS_ScriptEnvironment * env,
                               const AVS_VideoInfo * vi)
  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
 #endif
 AVSC_API(int, avs_make_writable)(AVS_ScriptEnvironment *, AVS_VideoFrame * * pvf);
 AVSC_API(void, avs_bit_blt)(AVS_ScriptEnvironment *, BYTE* dstp, int dst_pitch, const BYTE* srcp, int src_pitch, int row_size, int height);
 typedef void (AVSC_CC *AVS_ShutdownFunc)(void* user_data, AVS_ScriptEnvironment * env);
 AVSC_API(void, avs_at_exit)(AVS_ScriptEnvironment *, AVS_ShutdownFunc function, void * user_data);
 AVSC_API(AVS_VideoFrame *, avs_subframe)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height);
 // The returned video frame must be be released
 AVSC_API(int, avs_set_memory_max)(AVS_ScriptEnvironment *, int mem);
 AVSC_API(int, avs_set_working_dir)(AVS_ScriptEnvironment *, const char * newdir);
 // avisynth.dll exports this; it's a way to use it as a library, without
 // writing an AVS script or without going through AVIFile.
 AVSC_API(AVS_ScriptEnvironment *, avs_create_script_environment)(int version);
 // this symbol is the entry point for the plugin and must
 // be defined
 AVSC_EXPORT
 const char * AVSC_CC avisynth_c_plugin_init(AVS_ScriptEnvironment* env);
 AVSC_API(void, avs_delete_script_environment)(AVS_ScriptEnvironment *);
 AVSC_API(AVS_VideoFrame *, avs_subframe_planar)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height, int rel_offsetU, int rel_offsetV, int new_pitchUV);
 // The returned video frame must be be released
 #ifdef AVSC_NO_DECLSPEC
 // 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
  void* malloc(size_t)
  void free(void*);
  HMODULE LoadLibrary(const char*);
  void* GetProcAddress(HMODULE, const char*);
  FreeLibrary(HMODULE);
 */
 typedef struct AVS_Library AVS_Library;
 #define AVSC_DECLARE_FUNC(name) name##_func name
 struct AVS_Library {
  HMODULE handle;
  AVSC_DECLARE_FUNC(avs_add_function);
  AVSC_DECLARE_FUNC(avs_at_exit);
  AVSC_DECLARE_FUNC(avs_bit_blt);
  AVSC_DECLARE_FUNC(avs_check_version);
  AVSC_DECLARE_FUNC(avs_clip_get_error);
  AVSC_DECLARE_FUNC(avs_copy_clip);
  AVSC_DECLARE_FUNC(avs_copy_value);
  AVSC_DECLARE_FUNC(avs_copy_video_frame);
  AVSC_DECLARE_FUNC(avs_create_script_environment);
  AVSC_DECLARE_FUNC(avs_delete_script_environment);
  AVSC_DECLARE_FUNC(avs_function_exists);
  AVSC_DECLARE_FUNC(avs_get_audio);
  AVSC_DECLARE_FUNC(avs_get_cpu_flags);
  AVSC_DECLARE_FUNC(avs_get_error);
  AVSC_DECLARE_FUNC(avs_get_frame);
  AVSC_DECLARE_FUNC(avs_get_parity);
  AVSC_DECLARE_FUNC(avs_get_var);
  AVSC_DECLARE_FUNC(avs_get_version);
  AVSC_DECLARE_FUNC(avs_get_video_info);
  AVSC_DECLARE_FUNC(avs_invoke);
  AVSC_DECLARE_FUNC(avs_make_writable);
  AVSC_DECLARE_FUNC(avs_new_c_filter);
  AVSC_DECLARE_FUNC(avs_new_video_frame_a);
  AVSC_DECLARE_FUNC(avs_release_clip);
  AVSC_DECLARE_FUNC(avs_release_value);
  AVSC_DECLARE_FUNC(avs_release_video_frame);
  AVSC_DECLARE_FUNC(avs_save_string);
  AVSC_DECLARE_FUNC(avs_set_cache_hints);
  AVSC_DECLARE_FUNC(avs_set_global_var);
  AVSC_DECLARE_FUNC(avs_set_memory_max);
  AVSC_DECLARE_FUNC(avs_set_to_clip);
  AVSC_DECLARE_FUNC(avs_set_var);
  AVSC_DECLARE_FUNC(avs_set_working_dir);
  AVSC_DECLARE_FUNC(avs_sprintf);
  AVSC_DECLARE_FUNC(avs_subframe);
  AVSC_DECLARE_FUNC(avs_subframe_planar);
  AVSC_DECLARE_FUNC(avs_take_clip);
  AVSC_DECLARE_FUNC(avs_vsprintf);
 };
 #undef AVSC_DECLARE_FUNC
 AVSC_INLINE AVS_Library * avs_load_library() {
  AVS_Library *library = (AVS_Library *)malloc(sizeof(AVS_Library));
  if (library == NULL)
    return NULL;
  library->handle = LoadLibrary("avisynth");
  if (library->handle == NULL)
    goto fail;
 #define __AVSC_STRINGIFY(x) #x
 #define AVSC_STRINGIFY(x) __AVSC_STRINGIFY(x)
 #define AVSC_LOAD_FUNC(name) {\
  library->name = (name##_func) GetProcAddress(library->handle, AVSC_STRINGIFY(name));\
  if (library->name == NULL)\
    goto fail;\
 }
  AVSC_LOAD_FUNC(avs_add_function);
  AVSC_LOAD_FUNC(avs_at_exit);
  AVSC_LOAD_FUNC(avs_bit_blt);
  AVSC_LOAD_FUNC(avs_check_version);
  AVSC_LOAD_FUNC(avs_clip_get_error);
  AVSC_LOAD_FUNC(avs_copy_clip);
  AVSC_LOAD_FUNC(avs_copy_value);
  AVSC_LOAD_FUNC(avs_copy_video_frame);
  AVSC_LOAD_FUNC(avs_create_script_environment);
  AVSC_LOAD_FUNC(avs_delete_script_environment);
  AVSC_LOAD_FUNC(avs_function_exists);
  AVSC_LOAD_FUNC(avs_get_audio);
  AVSC_LOAD_FUNC(avs_get_cpu_flags);
  AVSC_LOAD_FUNC(avs_get_error);
  AVSC_LOAD_FUNC(avs_get_frame);
  AVSC_LOAD_FUNC(avs_get_parity);
  AVSC_LOAD_FUNC(avs_get_var);
  AVSC_LOAD_FUNC(avs_get_version);
  AVSC_LOAD_FUNC(avs_get_video_info);
  AVSC_LOAD_FUNC(avs_invoke);
  AVSC_LOAD_FUNC(avs_make_writable);
  AVSC_LOAD_FUNC(avs_new_c_filter);
  AVSC_LOAD_FUNC(avs_new_video_frame_a);
  AVSC_LOAD_FUNC(avs_release_clip);
  AVSC_LOAD_FUNC(avs_release_value);
  AVSC_LOAD_FUNC(avs_release_video_frame);
  AVSC_LOAD_FUNC(avs_save_string);
  AVSC_LOAD_FUNC(avs_set_cache_hints);
  AVSC_LOAD_FUNC(avs_set_global_var);
  AVSC_LOAD_FUNC(avs_set_memory_max);
  AVSC_LOAD_FUNC(avs_set_to_clip);
  AVSC_LOAD_FUNC(avs_set_var);
  AVSC_LOAD_FUNC(avs_set_working_dir);
  AVSC_LOAD_FUNC(avs_sprintf);
  AVSC_LOAD_FUNC(avs_subframe);
  AVSC_LOAD_FUNC(avs_subframe_planar);
  AVSC_LOAD_FUNC(avs_take_clip);
  AVSC_LOAD_FUNC(avs_vsprintf);
 #undef __AVSC_STRINGIFY
 #undef AVSC_STRINGIFY
 #undef AVSC_LOAD_FUNC
  return library;
 fail:
  free(library);
  return NULL;
 }
 AVSC_INLINE void avs_free_library(AVS_Library *library) {
  if (library == NULL)
    return;
  FreeLibrary(library->handle);
  free(library);
 }
 #endif
 #endif
--- a/compat/avisynth/avisynth_c_25.h
+++ b/compat/avisynth/avisynth_c_25.h
@@ -0,0 +1,68 @@
 //  Copyright (c) 2011 FFmpegSource Project
 //
 //  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.
 /* these are defines/functions that are used and were changed in the switch to 2.6
 * and are needed to maintain full compatility with 2.5 */
 enum {
  AVS_CS_YV12_25 = 1<<3 | AVS_CS_YUV | AVS_CS_PLANAR,  // y-v-u, planar
  AVS_CS_I420_25 = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR,  // y-u-v, planar
 };
 AVSC_INLINE int avs_get_height_p_25(const AVS_VideoFrame * p, int plane) {
    switch (plane)
    {
        case AVS_PLANAR_U: case AVS_PLANAR_V:
            if (p->pitchUV)
                return p->height>>1;
            return 0;
    }
    return p->height;}
 AVSC_INLINE int avs_get_row_size_p_25(const AVS_VideoFrame * p, int plane) {
    int r;
    switch (plane)
    {
    case AVS_PLANAR_U: case AVS_PLANAR_V:
        if (p->pitchUV)
            return p->row_size>>1;
        else
            return 0;
    case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED:
        if (p->pitchUV)
        {
            r = ((p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)) )>>1; // Aligned rowsize
            if (r < p->pitchUV)
                return r;
            return p->row_size>>1;
        }
        else
            return 0;
    case AVS_PLANAR_Y_ALIGNED:
        r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
        if (r <= p->pitch)
            return r;
        return p->row_size;
    }
    return p->row_size;
 }
 AVSC_INLINE int avs_is_yv12_25(const AVS_VideoInfo * p)
    { return ((p->pixel_type & AVS_CS_YV12_25) == AVS_CS_YV12_25)||((p->pixel_type & AVS_CS_I420_25) == AVS_CS_I420_25); }
--- a/compat/avisynth/avxsynth_c.h
+++ b/compat/avisynth/avxsynth_c.h
@@ -0,0 +1,728 @@
 // Avisynth C Interface Version 0.20
 // Copyright 2003 Kevin Atkinson
 // This program is free software; you can redistribute it and/or modify
 // it under the terms of the GNU General Public License as published by
 // the Free Software Foundation; either version 2 of the License, or
 // (at your option) any later version.
 //
 // This program is distributed in the hope that it will be useful,
 // but WITHOUT ANY WARRANTY; without even the implied warranty of
 // MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 // GNU General Public License for more details.
 //
 // You should have received a copy of the GNU General Public License
 // along with this program; if not, write to the Free Software
 // Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston,
 // MA 02110-1301 USA, or visit
 // http://www.gnu.org/copyleft/gpl.html .
 //
 // As a special exception, I give you permission to link to the
 // Avisynth C interface with independent modules that communicate with
 // the Avisynth C interface solely through the interfaces defined in
 // avisynth_c.h, regardless of the license terms of these independent
 // modules, and to copy and distribute the resulting combined work
 // under terms of your choice, provided that every copy of the
 // combined work is accompanied by a complete copy of the source code
 // of the Avisynth C interface and Avisynth itself (with the version
 // used to produce the combined work), being distributed under the
 // terms of the GNU General Public License plus this exception.  An
 // independent module is a module which is not derived from or based
 // on Avisynth C Interface, such as 3rd-party filters, import and
 // export plugins, or graphical user interfaces.
 #ifndef __AVXSYNTH_C__
 #define __AVXSYNTH_C__
 #include "windowsPorts/windows2linux.h"
 #include <stdarg.h>
 #ifdef __cplusplus
 #  define EXTERN_C extern "C"
 #else
 #  define EXTERN_C
 #endif
 #define AVSC_USE_STDCALL 1
 #ifndef AVSC_USE_STDCALL
 #  define AVSC_CC __cdecl
 #else
 #  define AVSC_CC __stdcall
 #endif
 #define AVSC_INLINE static __inline
 #ifdef AVISYNTH_C_EXPORTS
 #  define AVSC_EXPORT EXTERN_C
 #  define AVSC_API(ret, name) EXTERN_C __declspec(dllexport) ret AVSC_CC name
 #else
 #  define AVSC_EXPORT EXTERN_C __declspec(dllexport)
 #  ifndef AVSC_NO_DECLSPEC
 #    define AVSC_API(ret, name) EXTERN_C __declspec(dllimport) ret AVSC_CC name
 #  else
 #    define AVSC_API(ret, name) typedef ret (AVSC_CC *name##_func)
 #  endif
 #endif
 #ifdef __GNUC__
 typedef long long int INT64;
 #else
 typedef __int64 INT64;
 #endif
 /////////////////////////////////////////////////////////////////////
 //
 // Constants
 //
 #ifndef __AVXSYNTH_H__
 enum { AVISYNTH_INTERFACE_VERSION = 3 };
 #endif
 enum {AVS_SAMPLE_INT8  = 1<<0,
      AVS_SAMPLE_INT16 = 1<<1,
      AVS_SAMPLE_INT24 = 1<<2,
      AVS_SAMPLE_INT32 = 1<<3,
      AVS_SAMPLE_FLOAT = 1<<4};
 enum {AVS_PLANAR_Y=1<<0,
      AVS_PLANAR_U=1<<1,
      AVS_PLANAR_V=1<<2,
      AVS_PLANAR_ALIGNED=1<<3,
      AVS_PLANAR_Y_ALIGNED=AVS_PLANAR_Y|AVS_PLANAR_ALIGNED,
      AVS_PLANAR_U_ALIGNED=AVS_PLANAR_U|AVS_PLANAR_ALIGNED,
      AVS_PLANAR_V_ALIGNED=AVS_PLANAR_V|AVS_PLANAR_ALIGNED};
  // Colorspace properties.
 enum {AVS_CS_BGR = 1<<28,
      AVS_CS_YUV = 1<<29,
      AVS_CS_INTERLEAVED = 1<<30,
      AVS_CS_PLANAR = 1<<31};
  // Specific colorformats
 enum {
  AVS_CS_UNKNOWN = 0,
  AVS_CS_BGR24 = 1<<0 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
  AVS_CS_BGR32 = 1<<1 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
  AVS_CS_YUY2 = 1<<2 | AVS_CS_YUV | AVS_CS_INTERLEAVED,
  AVS_CS_YV12 = 1<<3 | AVS_CS_YUV | AVS_CS_PLANAR,  // y-v-u, planar
  AVS_CS_I420 = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR,  // y-u-v, planar
  AVS_CS_IYUV = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR  // same as above
 };
 enum {
  AVS_IT_BFF = 1<<0,
  AVS_IT_TFF = 1<<1,
  AVS_IT_FIELDBASED = 1<<2};
 enum {
  AVS_FILTER_TYPE=1,
  AVS_FILTER_INPUT_COLORSPACE=2,
  AVS_FILTER_OUTPUT_TYPE=9,
  AVS_FILTER_NAME=4,
  AVS_FILTER_AUTHOR=5,
  AVS_FILTER_VERSION=6,
  AVS_FILTER_ARGS=7,
  AVS_FILTER_ARGS_INFO=8,
  AVS_FILTER_ARGS_DESCRIPTION=10,
  AVS_FILTER_DESCRIPTION=11};
 enum {  //SUBTYPES
  AVS_FILTER_TYPE_AUDIO=1,
  AVS_FILTER_TYPE_VIDEO=2,
  AVS_FILTER_OUTPUT_TYPE_SAME=3,
  AVS_FILTER_OUTPUT_TYPE_DIFFERENT=4};
 enum {
  AVS_CACHE_NOTHING=0,
  AVS_CACHE_RANGE=1,
  AVS_CACHE_ALL=2,
  AVS_CACHE_AUDIO=3,
  AVS_CACHE_AUDIO_NONE=4,
  AVS_CACHE_AUDIO_AUTO=5
 };
 #define AVS_FRAME_ALIGN 16
 typedef struct AVS_Clip AVS_Clip;
 typedef struct AVS_ScriptEnvironment AVS_ScriptEnvironment;
 /////////////////////////////////////////////////////////////////////
 //
 // AVS_VideoInfo
 //
 // AVS_VideoInfo is layed out identicly to VideoInfo
 typedef struct AVS_VideoInfo {
  int width, height;    // width=0 means no video
  unsigned fps_numerator, fps_denominator;
  int num_frames;
  int pixel_type;
  int audio_samples_per_second;   // 0 means no audio
  int sample_type;
  INT64 num_audio_samples;
  int nchannels;
  // Imagetype properties
  int image_type;
 } AVS_VideoInfo;
 // useful functions of the above
 AVSC_INLINE int avs_has_video(const AVS_VideoInfo * p)
        { return (p->width!=0); }
 AVSC_INLINE int avs_has_audio(const AVS_VideoInfo * p)
        { return (p->audio_samples_per_second!=0); }
 AVSC_INLINE int avs_is_rgb(const AVS_VideoInfo * p)
        { return !!(p->pixel_type&AVS_CS_BGR); }
 AVSC_INLINE int avs_is_rgb24(const AVS_VideoInfo * p)
        { return (p->pixel_type&AVS_CS_BGR24)==AVS_CS_BGR24; } // Clear out additional properties
 AVSC_INLINE int avs_is_rgb32(const AVS_VideoInfo * p)
        { return (p->pixel_type & AVS_CS_BGR32) == AVS_CS_BGR32 ; }
 AVSC_INLINE int avs_is_yuv(const AVS_VideoInfo * p)
        { return !!(p->pixel_type&AVS_CS_YUV ); }
 AVSC_INLINE int avs_is_yuy2(const AVS_VideoInfo * p)
        { return (p->pixel_type & AVS_CS_YUY2) == AVS_CS_YUY2; }
 AVSC_INLINE int avs_is_yv12(const AVS_VideoInfo * p)
        { return ((p->pixel_type & AVS_CS_YV12) == AVS_CS_YV12)||((p->pixel_type & AVS_CS_I420) == AVS_CS_I420); }
 AVSC_INLINE int avs_is_color_space(const AVS_VideoInfo * p, int c_space)
        { return ((p->pixel_type & c_space) == c_space); }
 AVSC_INLINE int avs_is_property(const AVS_VideoInfo * p, int property)
        { return ((p->pixel_type & property)==property ); }
 AVSC_INLINE int avs_is_planar(const AVS_VideoInfo * p)
        { return !!(p->pixel_type & AVS_CS_PLANAR); }
 AVSC_INLINE int avs_is_field_based(const AVS_VideoInfo * p)
        { return !!(p->image_type & AVS_IT_FIELDBASED); }
 AVSC_INLINE int avs_is_parity_known(const AVS_VideoInfo * p)
        { return ((p->image_type & AVS_IT_FIELDBASED)&&(p->image_type & (AVS_IT_BFF | AVS_IT_TFF))); }
 AVSC_INLINE int avs_is_bff(const AVS_VideoInfo * p)
        { return !!(p->image_type & AVS_IT_BFF); }
 AVSC_INLINE int avs_is_tff(const AVS_VideoInfo * p)
        { return !!(p->image_type & AVS_IT_TFF); }
 AVSC_INLINE int avs_bits_per_pixel(const AVS_VideoInfo * p)
 {
  switch (p->pixel_type) {
      case AVS_CS_BGR24: return 24;
      case AVS_CS_BGR32: return 32;
      case AVS_CS_YUY2:  return 16;
      case AVS_CS_YV12:
      case AVS_CS_I420:  return 12;
      default:           return 0;
    }
 }
 AVSC_INLINE int avs_bytes_from_pixels(const AVS_VideoInfo * p, int pixels)
        { return pixels * (avs_bits_per_pixel(p)>>3); }   // Will work on planar images, but will return only luma planes
 AVSC_INLINE int avs_row_size(const AVS_VideoInfo * p)
        { return avs_bytes_from_pixels(p,p->width); }  // Also only returns first plane on planar images
 AVSC_INLINE int avs_bmp_size(const AVS_VideoInfo * vi)
        { if (avs_is_planar(vi)) {int p = vi->height * ((avs_row_size(vi)+3) & ~3); p+=p>>1; return p;  } return vi->height * ((avs_row_size(vi)+3) & ~3); }
 AVSC_INLINE int avs_samples_per_second(const AVS_VideoInfo * p)
        { return p->audio_samples_per_second; }
 AVSC_INLINE int avs_bytes_per_channel_sample(const AVS_VideoInfo * p)
 {
    switch (p->sample_type) {
      case AVS_SAMPLE_INT8:  return sizeof(signed char);
      case AVS_SAMPLE_INT16: return sizeof(signed short);
      case AVS_SAMPLE_INT24: return 3;
      case AVS_SAMPLE_INT32: return sizeof(signed int);
      case AVS_SAMPLE_FLOAT: return sizeof(float);
      default: return 0;
    }
 }
 AVSC_INLINE int avs_bytes_per_audio_sample(const AVS_VideoInfo * p)
        { return p->nchannels*avs_bytes_per_channel_sample(p);}
 AVSC_INLINE INT64 avs_audio_samples_from_frames(const AVS_VideoInfo * p, INT64 frames)
        { return ((INT64)(frames) * p->audio_samples_per_second * p->fps_denominator / p->fps_numerator); }
 AVSC_INLINE int avs_frames_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
        { return (int)(samples * (INT64)p->fps_numerator / (INT64)p->fps_denominator / (INT64)p->audio_samples_per_second); }
 AVSC_INLINE INT64 avs_audio_samples_from_bytes(const AVS_VideoInfo * p, INT64 bytes)
        { return bytes / avs_bytes_per_audio_sample(p); }
 AVSC_INLINE INT64 avs_bytes_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
        { return samples * avs_bytes_per_audio_sample(p); }
 AVSC_INLINE int avs_audio_channels(const AVS_VideoInfo * p)
        { return p->nchannels; }
 AVSC_INLINE int avs_sample_type(const AVS_VideoInfo * p)
        { return p->sample_type;}
 // useful mutator
 AVSC_INLINE void avs_set_property(AVS_VideoInfo * p, int property)
        { p->image_type|=property; }
 AVSC_INLINE void avs_clear_property(AVS_VideoInfo * p, int property)
        { p->image_type&=~property; }
 AVSC_INLINE void avs_set_field_based(AVS_VideoInfo * p, int isfieldbased)
        { if (isfieldbased) p->image_type|=AVS_IT_FIELDBASED; else p->image_type&=~AVS_IT_FIELDBASED; }
 AVSC_INLINE void avs_set_fps(AVS_VideoInfo * p, unsigned numerator, unsigned denominator)
 {
    unsigned x=numerator, y=denominator;
    while (y) {   // find gcd
      unsigned t = x%y; x = y; y = t;
    }
    p->fps_numerator = numerator/x;
    p->fps_denominator = denominator/x;
 }
 AVSC_INLINE int avs_is_same_colorspace(AVS_VideoInfo * x, AVS_VideoInfo * y)
 {
        return (x->pixel_type == y->pixel_type)
                || (avs_is_yv12(x) && avs_is_yv12(y));
 }
 /////////////////////////////////////////////////////////////////////
 //
 // AVS_VideoFrame
 //
 // VideoFrameBuffer holds information about a memory block which is used
 // for video data.  For efficiency, instances of this class are not deleted
 // when the refcount reaches zero; instead they're stored in a linked list
 // to be reused.  The instances are deleted when the corresponding AVS
 // file is closed.
 // AVS_VideoFrameBuffer is layed out identicly to VideoFrameBuffer
 // DO NOT USE THIS STRUCTURE DIRECTLY
 typedef struct AVS_VideoFrameBuffer {
  unsigned char * data;
  int data_size;
  // sequence_number is incremented every time the buffer is changed, so
  // that stale views can tell they're no longer valid.
  long sequence_number;
  long refcount;
 } AVS_VideoFrameBuffer;
 // VideoFrame holds a "window" into a VideoFrameBuffer.
 // AVS_VideoFrame is layed out identicly to IVideoFrame
 // DO NOT USE THIS STRUCTURE DIRECTLY
 typedef struct AVS_VideoFrame {
  int refcount;
  AVS_VideoFrameBuffer * vfb;
  int offset, pitch, row_size, height, offsetU, offsetV, pitchUV;  // U&V offsets are from top of picture.
 } AVS_VideoFrame;
 // Access functions for AVS_VideoFrame
 AVSC_INLINE int avs_get_pitch(const AVS_VideoFrame * p) {
        return p->pitch;}
 AVSC_INLINE int avs_get_pitch_p(const AVS_VideoFrame * p, int plane) {
  switch (plane) {
  case AVS_PLANAR_U: case AVS_PLANAR_V: return p->pitchUV;}
  return p->pitch;}
 AVSC_INLINE int avs_get_row_size(const AVS_VideoFrame * p) {
        return p->row_size; }
 AVSC_INLINE int avs_get_row_size_p(const AVS_VideoFrame * p, int plane) {
        int r;
    switch (plane) {
    case AVS_PLANAR_U: case AVS_PLANAR_V:
                if (p->pitchUV) return p->row_size>>1;
                else            return 0;
    case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED:
                if (p->pitchUV) {
                        r = ((p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)) )>>1; // Aligned rowsize
                        if (r < p->pitchUV)
                                return r;
                        return p->row_size>>1;
                } else return 0;
    case AVS_PLANAR_Y_ALIGNED:
                r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
                if (r <= p->pitch)
                        return r;
                return p->row_size;
    }
    return p->row_size;
 }
 AVSC_INLINE int avs_get_height(const AVS_VideoFrame * p) {
        return p->height;}
 AVSC_INLINE int avs_get_height_p(const AVS_VideoFrame * p, int plane) {
        switch (plane) {
                case AVS_PLANAR_U: case AVS_PLANAR_V:
                        if (p->pitchUV) return p->height>>1;
                        return 0;
        }
        return p->height;}
 AVSC_INLINE const unsigned char* avs_get_read_ptr(const AVS_VideoFrame * p) {
        return p->vfb->data + p->offset;}
 AVSC_INLINE const unsigned char* avs_get_read_ptr_p(const AVS_VideoFrame * p, int plane)
 {
        switch (plane) {
                case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
                case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
                default:           return p->vfb->data + p->offset;}
 }
 AVSC_INLINE int avs_is_writable(const AVS_VideoFrame * p) {
        return (p->refcount == 1 && p->vfb->refcount == 1);}
 AVSC_INLINE unsigned char* avs_get_write_ptr(const AVS_VideoFrame * p)
 {
        if (avs_is_writable(p)) {
                ++p->vfb->sequence_number;
                return p->vfb->data + p->offset;
        } else
                return 0;
 }
 AVSC_INLINE unsigned char* avs_get_write_ptr_p(const AVS_VideoFrame * p, int plane)
 {
        if (plane==AVS_PLANAR_Y && avs_is_writable(p)) {
                ++p->vfb->sequence_number;
                return p->vfb->data + p->offset;
        } else if (plane==AVS_PLANAR_Y) {
                return 0;
        } else {
                switch (plane) {
                        case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
                        case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
                        default:       return p->vfb->data + p->offset;
                }
        }
 }
 #if defined __cplusplus
 extern "C"
 {
 #endif // __cplusplus
 AVSC_API(void, avs_release_video_frame)(AVS_VideoFrame *);
 // makes a shallow copy of a video frame
 AVSC_API(AVS_VideoFrame *, avs_copy_video_frame)(AVS_VideoFrame *);
 #if defined __cplusplus
 }
 #endif // __cplusplus
 #ifndef AVSC_NO_DECLSPEC
 AVSC_INLINE void avs_release_frame(AVS_VideoFrame * f)
  {avs_release_video_frame(f);}
 AVSC_INLINE AVS_VideoFrame * avs_copy_frame(AVS_VideoFrame * f)
  {return avs_copy_video_frame(f);}
 #endif
 /////////////////////////////////////////////////////////////////////
 //
 // AVS_Value
 //
 // Treat AVS_Value as a fat pointer.  That is use avs_copy_value
 // and avs_release_value appropiaty as you would if AVS_Value was
 // a pointer.
 // To maintain source code compatibility with future versions of the
 // avisynth_c API don't use the AVS_Value directly.  Use the helper
 // functions below.
 // AVS_Value is layed out identicly to AVSValue
 typedef struct AVS_Value AVS_Value;
 struct AVS_Value {
  short type;  // 'a'rray, 'c'lip, 'b'ool, 'i'nt, 'f'loat, 's'tring, 'v'oid, or 'l'ong
               // for some function e'rror
  short array_size;
  union {
    void * clip; // do not use directly, use avs_take_clip
    char boolean;
    int integer;
    INT64 integer64; // match addition of __int64 to avxplugin.h
    float floating_pt;
    const char * string;
    const AVS_Value * array;
  } d;
 };
 // AVS_Value should be initilized with avs_void.
 // Should also set to avs_void after the value is released
 // with avs_copy_value.  Consider it the equalvent of setting
 // a pointer to NULL
 static const AVS_Value avs_void = {'v'};
 AVSC_API(void, avs_copy_value)(AVS_Value * dest, AVS_Value src);
 AVSC_API(void, avs_release_value)(AVS_Value);
 AVSC_INLINE int avs_defined(AVS_Value v) { return v.type != 'v'; }
 AVSC_INLINE int avs_is_clip(AVS_Value v) { return v.type == 'c'; }
 AVSC_INLINE int avs_is_bool(AVS_Value v) { return v.type == 'b'; }
 AVSC_INLINE int avs_is_int(AVS_Value v) { return v.type == 'i'; }
 AVSC_INLINE int avs_is_float(AVS_Value v) { return v.type == 'f' || v.type == 'i'; }
 AVSC_INLINE int avs_is_string(AVS_Value v) { return v.type == 's'; }
 AVSC_INLINE int avs_is_array(AVS_Value v) { return v.type == 'a'; }
 AVSC_INLINE int avs_is_error(AVS_Value v) { return v.type == 'e'; }
 #if defined __cplusplus
 extern "C"
 {
 #endif // __cplusplus
 AVSC_API(AVS_Clip *, avs_take_clip)(AVS_Value, AVS_ScriptEnvironment *);
 AVSC_API(void, avs_set_to_clip)(AVS_Value *, AVS_Clip *);
 #if defined __cplusplus
 }
 #endif // __cplusplus
 AVSC_INLINE int avs_as_bool(AVS_Value v)
        { return v.d.boolean; }
 AVSC_INLINE int avs_as_int(AVS_Value v)
        { return v.d.integer; }
 AVSC_INLINE const char * avs_as_string(AVS_Value v)
        { return avs_is_error(v) || avs_is_string(v) ? v.d.string : 0; }
 AVSC_INLINE double avs_as_float(AVS_Value v)
        { return avs_is_int(v) ? v.d.integer : v.d.floating_pt; }
 AVSC_INLINE const char * avs_as_error(AVS_Value v)
        { return avs_is_error(v) ? v.d.string : 0; }
 AVSC_INLINE const AVS_Value * avs_as_array(AVS_Value v)
        { return v.d.array; }
 AVSC_INLINE int avs_array_size(AVS_Value v)
        { return avs_is_array(v) ? v.array_size : 1; }
 AVSC_INLINE AVS_Value avs_array_elt(AVS_Value v, int index)
        { return avs_is_array(v) ? v.d.array[index] : v; }
 // only use these functions on am AVS_Value that does not already have
 // an active value.  Remember, treat AVS_Value as a fat pointer.
 AVSC_INLINE AVS_Value avs_new_value_bool(int v0)
        { AVS_Value v; v.type = 'b'; v.d.boolean = v0 == 0 ? 0 : 1; return v; }
 AVSC_INLINE AVS_Value avs_new_value_int(int v0)
        { AVS_Value v; v.type = 'i'; v.d.integer = v0; return v; }
 AVSC_INLINE AVS_Value avs_new_value_string(const char * v0)
        { AVS_Value v; v.type = 's'; v.d.string = v0; return v; }
 AVSC_INLINE AVS_Value avs_new_value_float(float v0)
        { AVS_Value v; v.type = 'f'; v.d.floating_pt = v0; return v;}
 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
 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 = size; return v; }
 /////////////////////////////////////////////////////////////////////
 //
 // AVS_Clip
 //
 #if defined __cplusplus
 extern "C"
 {
 #endif // __cplusplus
 AVSC_API(void, avs_release_clip)(AVS_Clip *);
 AVSC_API(AVS_Clip *, avs_copy_clip)(AVS_Clip *);
 AVSC_API(const char *, avs_clip_get_error)(AVS_Clip *); // return 0 if no error
 AVSC_API(const AVS_VideoInfo *, avs_get_video_info)(AVS_Clip *);
 AVSC_API(int, avs_get_version)(AVS_Clip *);
 AVSC_API(AVS_VideoFrame *, avs_get_frame)(AVS_Clip *, int n);
 // The returned video frame must be released with avs_release_video_frame
 AVSC_API(int, avs_get_parity)(AVS_Clip *, int n);
 // return field parity if field_based, else parity of first field in frame
 AVSC_API(int, avs_get_audio)(AVS_Clip *, void * buf,
                                  INT64 start, INT64 count);
 // start and count are in samples
 AVSC_API(int, avs_set_cache_hints)(AVS_Clip *,
                                        int cachehints, size_t frame_range);
 #if defined __cplusplus
 }
 #endif // __cplusplus
 // This is the callback type used by avs_add_function
 typedef AVS_Value (AVSC_CC * AVS_ApplyFunc)
                        (AVS_ScriptEnvironment *, AVS_Value args, void * user_data);
 typedef struct AVS_FilterInfo AVS_FilterInfo;
 struct AVS_FilterInfo
 {
  // these members should not be modified outside of the AVS_ApplyFunc callback
  AVS_Clip * child;
  AVS_VideoInfo vi;
  AVS_ScriptEnvironment * env;
  AVS_VideoFrame * (AVSC_CC * get_frame)(AVS_FilterInfo *, int n);
  int (AVSC_CC * get_parity)(AVS_FilterInfo *, int n);
  int (AVSC_CC * get_audio)(AVS_FilterInfo *, void * buf,
                                  INT64 start, INT64 count);
  int (AVSC_CC * set_cache_hints)(AVS_FilterInfo *, int cachehints,
                                        int frame_range);
  void (AVSC_CC * free_filter)(AVS_FilterInfo *);
  // Should be set when ever there is an error to report.
  // It is cleared before any of the above methods are called
  const char * error;
  // this is to store whatever and may be modified at will
  void * user_data;
 };
 // Create a new filter
 // fi is set to point to the AVS_FilterInfo so that you can
 //   modify it once it is initilized.
 // store_child should generally be set to true.  If it is not
 //    set than ALL methods (the function pointers) must be defined
 // If it is set than you do not need to worry about freeing the child
 //    clip.
 #if defined __cplusplus
 extern "C"
 {
 #endif // __cplusplus
 AVSC_API(AVS_Clip *, avs_new_c_filter)(AVS_ScriptEnvironment * e,
                                      AVS_FilterInfo * * fi,
                                      AVS_Value child, int store_child);
 #if defined __cplusplus
 }
 #endif // __cplusplus
 /////////////////////////////////////////////////////////////////////
 //
 // AVS_ScriptEnvironment
 //
 // For GetCPUFlags.  These are backwards-compatible with those in VirtualDub.
 enum {
                                /* slowest CPU to support extension */
  AVS_CPU_FORCE        = 0x01,   // N/A
  AVS_CPU_FPU          = 0x02,   // 386/486DX
  AVS_CPU_MMX          = 0x04,   // P55C, K6, PII
  AVS_CPU_INTEGER_SSE  = 0x08,   // PIII, Athlon
  AVS_CPU_SSE          = 0x10,   // PIII, Athlon XP/MP
  AVS_CPU_SSE2         = 0x20,   // PIV, Hammer
  AVS_CPU_3DNOW        = 0x40,   // K6-2
  AVS_CPU_3DNOW_EXT    = 0x80,   // Athlon
  AVS_CPU_X86_64       = 0xA0,   // Hammer (note: equiv. to 3DNow + SSE2,
                                 // which only Hammer will have anyway)
 };
 #if defined __cplusplus
 extern "C"
 {
 #endif // __cplusplus
 AVSC_API(const char *, avs_get_error)(AVS_ScriptEnvironment *); // return 0 if no error
 AVSC_API(long, avs_get_cpu_flags)(AVS_ScriptEnvironment *);
 AVSC_API(int, avs_check_version)(AVS_ScriptEnvironment *, int version);
 AVSC_API(char *, avs_save_string)(AVS_ScriptEnvironment *, const char* s, int length);
 AVSC_API(char *, avs_sprintf)(AVS_ScriptEnvironment *, const char * fmt, ...);
 AVSC_API(char *, avs_vsprintf)(AVS_ScriptEnvironment *, const char * fmt, va_list val);
 // note: val is really a va_list; I hope everyone typedefs va_list to a pointer
 AVSC_API(int, avs_add_function)(AVS_ScriptEnvironment *,
                                     const char * name, const char * params,
                                     AVS_ApplyFunc apply, void * user_data);
 AVSC_API(int, avs_function_exists)(AVS_ScriptEnvironment *, const char * name);
 AVSC_API(AVS_Value, avs_invoke)(AVS_ScriptEnvironment *, const char * name,
                               AVS_Value args, const char** arg_names);
 // The returned value must be be released with avs_release_value
 AVSC_API(AVS_Value, avs_get_var)(AVS_ScriptEnvironment *, const char* name);
 // The returned value must be be released with avs_release_value
 AVSC_API(int, avs_set_var)(AVS_ScriptEnvironment *, const char* name, AVS_Value val);
 AVSC_API(int, avs_set_global_var)(AVS_ScriptEnvironment *, const char* name, const AVS_Value val);
 //void avs_push_context(AVS_ScriptEnvironment *, int level=0);
 //void avs_pop_context(AVS_ScriptEnvironment *);
 AVSC_API(AVS_VideoFrame *, avs_new_video_frame_a)(AVS_ScriptEnvironment *,
                                          const AVS_VideoInfo * vi, int align);
 // align should be at least 16
 #if defined __cplusplus
 }
 #endif // __cplusplus
 #ifndef AVSC_NO_DECLSPEC
 AVSC_INLINE
 AVS_VideoFrame * avs_new_video_frame(AVS_ScriptEnvironment * env,
                                     const AVS_VideoInfo * vi)
  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
 AVSC_INLINE
 AVS_VideoFrame * avs_new_frame(AVS_ScriptEnvironment * env,
                               const AVS_VideoInfo * vi)
  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
 #endif
 #if defined __cplusplus
 extern "C"
 {
 #endif // __cplusplus
 AVSC_API(int, avs_make_writable)(AVS_ScriptEnvironment *, AVS_VideoFrame * * pvf);
 AVSC_API(void, avs_bit_blt)(AVS_ScriptEnvironment *, unsigned char* dstp, int dst_pitch, const unsigned char* srcp, int src_pitch, int row_size, int height);
 typedef void (AVSC_CC *AVS_ShutdownFunc)(void* user_data, AVS_ScriptEnvironment * env);
 AVSC_API(void, avs_at_exit)(AVS_ScriptEnvironment *, AVS_ShutdownFunc function, void * user_data);
 AVSC_API(AVS_VideoFrame *, avs_subframe)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height);
 // The returned video frame must be be released
 AVSC_API(int, avs_set_memory_max)(AVS_ScriptEnvironment *, int mem);
 AVSC_API(int, avs_set_working_dir)(AVS_ScriptEnvironment *, const char * newdir);
 // avisynth.dll exports this; it's a way to use it as a library, without
 // writing an AVS script or without going through AVIFile.
 AVSC_API(AVS_ScriptEnvironment *, avs_create_script_environment)(int version);
 #if defined __cplusplus
 }
 #endif // __cplusplus
 // this symbol is the entry point for the plugin and must
 // be defined
 AVSC_EXPORT
 const char * AVSC_CC avisynth_c_plugin_init(AVS_ScriptEnvironment* env);
 #if defined __cplusplus
 extern "C"
 {
 #endif // __cplusplus
 AVSC_API(void, avs_delete_script_environment)(AVS_ScriptEnvironment *);
 AVSC_API(AVS_VideoFrame *, avs_subframe_planar)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height, int rel_offsetU, int rel_offsetV, int new_pitchUV);
 // The returned video frame must be be released
 #if defined __cplusplus
 }
 #endif // __cplusplus
 #endif //__AVXSYNTH_C__
--- a/compat/avisynth/windowsPorts/basicDataTypeConversions.h
+++ b/compat/avisynth/windowsPorts/basicDataTypeConversions.h
@@ -0,0 +1,85 @@
 #ifndef __DATA_TYPE_CONVERSIONS_H__
 #define __DATA_TYPE_CONVERSIONS_H__
 #include <stdint.h>
 #include <wchar.h>
 #ifdef __cplusplus
 namespace avxsynth {
 #endif // __cplusplus
 typedef int64_t __int64;
 typedef int32_t __int32;
 #ifdef __cplusplus
 typedef bool BOOL;
 #else
 typedef uint32_t BOOL;
 #endif // __cplusplus
 typedef void* HMODULE;
 typedef void* LPVOID;
 typedef void* PVOID;
 typedef PVOID HANDLE;
 typedef HANDLE HWND;
 typedef HANDLE HINSTANCE;
 typedef void* HDC;
 typedef void* HBITMAP;
 typedef void* HICON;
 typedef void* HFONT;
 typedef void* HGDIOBJ;
 typedef void* HBRUSH;
 typedef void* HMMIO;
 typedef void* HACMSTREAM;
 typedef void* HACMDRIVER;
 typedef void* HIC;
 typedef void* HACMOBJ;
 typedef HACMSTREAM* LPHACMSTREAM;
 typedef void* HACMDRIVERID;
 typedef void* LPHACMDRIVER;
 typedef unsigned char BYTE;
 typedef BYTE* LPBYTE;
 typedef char TCHAR;
 typedef TCHAR* LPTSTR;
 typedef const TCHAR* LPCTSTR;
 typedef char* LPSTR;
 typedef LPSTR LPOLESTR;
 typedef const char* LPCSTR;
 typedef LPCSTR LPCOLESTR;
 typedef wchar_t WCHAR;
 typedef unsigned short WORD;
 typedef unsigned int UINT;
 typedef UINT MMRESULT;
 typedef uint32_t DWORD;
 typedef DWORD COLORREF;
 typedef DWORD FOURCC;
 typedef DWORD HRESULT;
 typedef DWORD* LPDWORD;
 typedef DWORD* DWORD_PTR;
 typedef int32_t LONG;
 typedef int32_t* LONG_PTR;
 typedef LONG_PTR LRESULT;
 typedef uint32_t ULONG;
 typedef uint32_t* ULONG_PTR;
 //typedef __int64_t intptr_t;
 typedef uint64_t _fsize_t;
 //
 // Structures
 //
 typedef struct _GUID {
  DWORD Data1;
  WORD  Data2;
  WORD  Data3;
  BYTE  Data4[8];
 } GUID;
 typedef GUID REFIID;
 typedef GUID CLSID;
 typedef CLSID* LPCLSID;
 typedef GUID IID;
 #ifdef __cplusplus
 }; // namespace avxsynth
 #endif // __cplusplus
 #endif //  __DATA_TYPE_CONVERSIONS_H__
--- a/compat/avisynth/windowsPorts/windows2linux.h
+++ b/compat/avisynth/windowsPorts/windows2linux.h
@@ -0,0 +1,77 @@
 #ifndef __WINDOWS2LINUX_H__
 #define __WINDOWS2LINUX_H__
 /*
 * LINUX SPECIFIC DEFINITIONS
 */
 //
 // Data types conversions
 //
 #include <stdlib.h>
 #include <string.h>
 #include "basicDataTypeConversions.h"
 #ifdef __cplusplus
 namespace avxsynth {
 #endif // __cplusplus
 //
 // purposefully define the following MSFT definitions
 // to mean nothing (as they do not mean anything on Linux)
 //
 #define __stdcall
 #define __cdecl
 #define noreturn
 #define __declspec(x)
 #define STDAPI       extern "C" HRESULT
 #define STDMETHODIMP HRESULT __stdcall
 #define STDMETHODIMP_(x) x __stdcall
 #define STDMETHOD(x)    virtual HRESULT x
 #define STDMETHOD_(a, x) virtual a x
 #ifndef TRUE
 #define TRUE  true
 #endif
 #ifndef FALSE
 #define FALSE false
 #endif
 #define S_OK                (0x00000000)
 #define S_FALSE             (0x00000001)
 #define E_NOINTERFACE       (0X80004002)
 #define E_POINTER           (0x80004003)
 #define E_FAIL              (0x80004005)
 #define E_OUTOFMEMORY       (0x8007000E)
 #define INVALID_HANDLE_VALUE    ((HANDLE)((LONG_PTR)-1))
 #define FAILED(hr)              ((hr) & 0x80000000)
 #define SUCCEEDED(hr)           (!FAILED(hr))
 //
 // Functions
 //
 #define MAKEDWORD(a,b,c,d) ((a << 24) | (b << 16) | (c << 8) | (d))
 #define MAKEWORD(a,b) ((a << 8) | (b))
 #define lstrlen                             strlen
 #define lstrcpy                             strcpy
 #define lstrcmpi                            strcasecmp
 #define _stricmp                            strcasecmp
 #define InterlockedIncrement(x)             __sync_fetch_and_add((x), 1)
 #define InterlockedDecrement(x)             __sync_fetch_and_sub((x), 1)
 // Windows uses (new, old) ordering but GCC has (old, new)
 #define InterlockedCompareExchange(x,y,z)   __sync_val_compare_and_swap(x,z,y)
 #define UInt32x32To64(a, b)                 ( (uint64_t) ( ((uint64_t)((uint32_t)(a))) * ((uint32_t)(b))  ) )
 #define Int64ShrlMod32(a, b)                ( (uint64_t) ( (uint64_t)(a) >> (b) ) )
 #define Int32x32To64(a, b)                  ((__int64)(((__int64)((long)(a))) * ((long)(b))))
 #define MulDiv(nNumber, nNumerator, nDenominator)   (int32_t) (((int64_t) (nNumber) * (int64_t) (nNumerator) + (int64_t) ((nDenominator)/2)) / (int64_t) (nDenominator))
 #ifdef __cplusplus
 }; // namespace avxsynth
 #endif // __cplusplus
 #endif //  __WINDOWS2LINUX_H__
--- a/compat/cuda/cuda_runtime.h
+++ b/compat/cuda/cuda_runtime.h
@@ -1,191 +0,0 @@
 /*
 * Minimum CUDA compatibility definitions header
 *
 * Copyright (c) 2019 rcombs
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
 #ifndef COMPAT_CUDA_CUDA_RUNTIME_H
 #define COMPAT_CUDA_CUDA_RUNTIME_H
 // Common macros
 #define __global__ __attribute__((global))
 #define __device__ __attribute__((device))
 #define __device_builtin__ __attribute__((device_builtin))
 #define __align__(N) __attribute__((aligned(N)))
 #define __inline__ __inline__ __attribute__((always_inline))
 #define max(a, b) ((a) > (b) ? (a) : (b))
 #define min(a, b) ((a) < (b) ? (a) : (b))
 #define abs(x) ((x) < 0 ? -(x) : (x))
 #define atomicAdd(a, b) (__atomic_fetch_add(a, b, __ATOMIC_SEQ_CST))
 // Basic typedefs
 typedef __device_builtin__ unsigned long long cudaTextureObject_t;
 typedef struct __device_builtin__ __align__(2) uchar2
 {
    unsigned char x, y;
 } uchar2;
 typedef struct __device_builtin__ __align__(4) ushort2
 {
    unsigned short x, y;
 } ushort2;
 typedef struct __device_builtin__ __align__(8) float2
 {
    float x, y;
 } float2;
 typedef struct __device_builtin__ __align__(8) int2
 {
    int x, y;
 } int2;
 typedef struct __device_builtin__ uint3
 {
    unsigned int x, y, z;
 } uint3;
 typedef struct uint3 dim3;
 typedef struct __device_builtin__ __align__(4) uchar4
 {
    unsigned char x, y, z, w;
 } uchar4;
 typedef struct __device_builtin__ __align__(8) ushort4
 {
    unsigned short x, y, z, w;
 } ushort4;
 typedef struct __device_builtin__ __align__(16) int4
 {
    int x, y, z, w;
 } int4;
 typedef struct __device_builtin__ __align__(16) float4
 {
    float x, y, z, w;
 } float4;
 // Accessors for special registers
 #define GETCOMP(reg, comp) \
    asm("mov.u32 %0, %%" #reg "." #comp ";" : "=r"(tmp)); \
    ret.comp = tmp;
 #define GET(name, reg) static inline __device__ uint3 name() {\
    uint3 ret; \
    unsigned tmp; \
    GETCOMP(reg, x) \
    GETCOMP(reg, y) \
    GETCOMP(reg, z) \
    return ret; \
 }
 GET(getBlockIdx, ctaid)
 GET(getBlockDim, ntid)
 GET(getThreadIdx, tid)
 // Instead of externs for these registers, we turn access to them into calls into trivial ASM
 #define blockIdx (getBlockIdx())
 #define blockDim (getBlockDim())
 #define threadIdx (getThreadIdx())
 // Basic initializers (simple macros rather than inline functions)
 #define make_int2(a, b) ((int2){.x = a, .y = b})
 #define make_uchar2(a, b) ((uchar2){.x = a, .y = b})
 #define make_ushort2(a, b) ((ushort2){.x = a, .y = b})
 #define make_float2(a, b) ((float2){.x = a, .y = b})
 #define make_int4(a, b, c, d) ((int4){.x = a, .y = b, .z = c, .w = d})
 #define make_uchar4(a, b, c, d) ((uchar4){.x = a, .y = b, .z = c, .w = d})
 #define make_ushort4(a, b, c, d) ((ushort4){.x = a, .y = b, .z = c, .w = d})
 #define make_float4(a, b, c, d) ((float4){.x = a, .y = b, .z = c, .w = d})
 // Conversions from the tex instruction's 4-register output to various types
 #define TEX2D(type, ret) static inline __device__ void conv(type* out, unsigned a, unsigned b, unsigned c, unsigned d) {*out = (ret);}
 TEX2D(unsigned char, a & 0xFF)
 TEX2D(unsigned short, a & 0xFFFF)
 TEX2D(float, a)
 TEX2D(uchar2, make_uchar2(a & 0xFF, b & 0xFF))
 TEX2D(ushort2, make_ushort2(a & 0xFFFF, b & 0xFFFF))
 TEX2D(float2, make_float2(a, b))
 TEX2D(uchar4, make_uchar4(a & 0xFF, b & 0xFF, c & 0xFF, d & 0xFF))
 TEX2D(ushort4, make_ushort4(a & 0xFFFF, b & 0xFFFF, c & 0xFFFF, d & 0xFFFF))
 TEX2D(float4, make_float4(a, b, c, d))
 // Template calling tex instruction and converting the output to the selected type
 template<typename T>
 inline __device__ T tex2D(cudaTextureObject_t texObject, float x, float y)
 {
  T ret;
  unsigned ret1, ret2, ret3, ret4;
  asm("tex.2d.v4.u32.f32 {%0, %1, %2, %3}, [%4, {%5, %6}];" :
      "=r"(ret1), "=r"(ret2), "=r"(ret3), "=r"(ret4) :
      "l"(texObject), "f"(x), "f"(y));
  conv(&ret, ret1, ret2, ret3, ret4);
  return ret;
 }
 template<>
 inline __device__ float4 tex2D<float4>(cudaTextureObject_t texObject, float x, float y)
 {
    float4 ret;
    asm("tex.2d.v4.f32.f32 {%0, %1, %2, %3}, [%4, {%5, %6}];" :
        "=r"(ret.x), "=r"(ret.y), "=r"(ret.z), "=r"(ret.w) :
        "l"(texObject), "f"(x), "f"(y));
    return ret;
 }
 template<>
 inline __device__ float tex2D<float>(cudaTextureObject_t texObject, float x, float y)
 {
    return tex2D<float4>(texObject, x, y).x;
 }
 template<>
 inline __device__ float2 tex2D<float2>(cudaTextureObject_t texObject, float x, float y)
 {
    float4 ret = tex2D<float4>(texObject, x, y);
    return make_float2(ret.x, ret.y);
 }
 // Math helper functions
 static inline __device__ float floorf(float a) { return __builtin_floorf(a); }
 static inline __device__ float floor(float a) { return __builtin_floorf(a); }
 static inline __device__ double floor(double a) { return __builtin_floor(a); }
 static inline __device__ float ceilf(float a) { return __builtin_ceilf(a); }
 static inline __device__ float ceil(float a) { return __builtin_ceilf(a); }
 static inline __device__ double ceil(double a) { return __builtin_ceil(a); }
 static inline __device__ float truncf(float a) { return __builtin_truncf(a); }
 static inline __device__ float trunc(float a) { return __builtin_truncf(a); }
 static inline __device__ double trunc(double a) { return __builtin_trunc(a); }
 static inline __device__ float fabsf(float a) { return __builtin_fabsf(a); }
 static inline __device__ float fabs(float a) { return __builtin_fabsf(a); }
 static inline __device__ double fabs(double a) { return __builtin_fabs(a); }
 static inline __device__ float sqrtf(float a) { return __builtin_sqrtf(a); }
 static inline __device__ float __saturatef(float a) { return __nvvm_saturate_f(a); }
 static inline __device__ float __sinf(float a) { return __nvvm_sin_approx_f(a); }
 static inline __device__ float __cosf(float a) { return __nvvm_cos_approx_f(a); }
 static inline __device__ float __expf(float a) { return __nvvm_ex2_approx_f(a * (float)__builtin_log2(__builtin_exp(1))); }
 #endif /* COMPAT_CUDA_CUDA_RUNTIME_H */
--- a/compat/cuda/dynlink_loader.h
+++ b/compat/cuda/dynlink_loader.h
@@ -1,33 +0,0 @@
 /*
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
 #ifndef COMPAT_CUDA_DYNLINK_LOADER_H
 #define COMPAT_CUDA_DYNLINK_LOADER_H
 #include "libavutil/log.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 <ffnvcodec/dynlink_loader.h>
 #endif /* COMPAT_CUDA_DYNLINK_LOADER_H */
--- a/compat/dispatch_semaphore/semaphore.h
+++ b/compat/dispatch_semaphore/semaphore.h
@@ -1,42 +0,0 @@
 /*
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
 #ifndef COMPAT_DISPATCH_SEMAPHORE_SEMAPHORE_H
 #define COMPAT_DISPATCH_SEMAPHORE_SEMAPHORE_H
 #include <dispatch/dispatch.h>
 #include <errno.h>
 #define sem_t dispatch_semaphore_t
 #define sem_post(psem)              dispatch_semaphore_signal(*psem)
 #define sem_wait(psem)              dispatch_semaphore_wait(*psem, DISPATCH_TIME_FOREVER)
 #define sem_timedwait(psem, val)    dispatch_semaphore_wait(*psem, dispatch_walltime(val, 0))
 #define sem_destroy(psem)           dispatch_release(*psem)
 static inline int compat_sem_init(dispatch_semaphore_t *psem,
                                  int unused, int val)
 {
    int ret = !!(*psem = dispatch_semaphore_create(val)) - 1;
    if (ret < 0)
        errno = ENOMEM;
    return ret;
 }
 #define sem_init compat_sem_init
 #endif /* COMPAT_DISPATCH_SEMAPHORE_SEMAPHORE_H */
--- a/compat/djgpp/math.c
+++ b/compat/djgpp/math.c
@@ -1,47 +0,0 @@
 /*
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
 #include <math.h>
 #define FUN(name, type, op) \
 type name(type x, type y) \
 { \
    if (fpclassify(x) == FP_NAN) return y; \
    if (fpclassify(y) == FP_NAN) return x; \
    return x op y ? x : y; \
 }
 FUN(fmin, double, <)
 FUN(fmax, double, >)
 FUN(fminf, float, <)
 FUN(fmaxf, float, >)
 long double fmodl(long double x, long double y)
 {
    return fmod(x, y);
 }
 long double scalbnl(long double x, int exp)
 {
    return scalbn(x, exp);
 }
 long double copysignl(long double x, long double y)
 {
    return copysign(x, y);
 }
--- a/compat/djgpp/math.h
+++ b/compat/djgpp/math.h
@@ -1,25 +0,0 @@
 /*
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
 double fmin(double, double);
 double fmax(double, double);
 float fminf(float, float);
 float fmaxf(float, float);
 long double fmodl(long double, long double);
 long double scalbnl(long double, int);
 long double copysignl(long double, long double);
--- a/compat/getopt.c
+++ b/compat/getopt.c
@@ -54,7 +54,7 @@ static int getopt(int argc, char *argv[], char *opts)
        }
    }
    optopt = c = argv[optind][sp];
-    if (c == ':' || !(cp = strchr(opts, c))) {
+    if (c == ':' || (cp = strchr(opts, c)) == NULL) {
        fprintf(stderr, ": illegal option -- %c\n", c);
        if (argv[optind][++sp] == '\0') {
            optind++;
--- a/compat/msvcrt/snprintf.c
+++ b/compat/msvcrt/snprintf.c
@@ -59,7 +59,7 @@ int avpriv_vsnprintf(char *s, size_t n, const char *fmt,
     * recommends to provide _snprintf/_vsnprintf() a buffer size that
     * is one less than the actual buffer, and zero it before calling
     * _snprintf/_vsnprintf() to workaround this problem.
-     * See https://web.archive.org/web/20151214111935/http://msdn.microsoft.com/en-us/library/1kt27hek(v=vs.80).aspx */
+     * See http://msdn.microsoft.com/en-us/library/1kt27hek(v=vs.80).aspx */
    memset(s, 0, n);
    va_copy(ap_copy, ap);
    ret = _vsnprintf(s, n - 1, fmt, ap_copy);
--- a/compat/msvcrt/snprintf.h
+++ b/compat/msvcrt/snprintf.h
@@ -19,8 +19,8 @@
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
-#ifndef COMPAT_MSVCRT_SNPRINTF_H
+#ifndef COMPAT_SNPRINTF_H
-#define COMPAT_MSVCRT_SNPRINTF_H
+#define COMPAT_SNPRINTF_H
 #include <stdarg.h>
 #include <stdio.h>
@@ -35,4 +35,4 @@ int avpriv_vsnprintf(char *s, size_t n, const char *fmt, va_list ap);
 #define _snprintf avpriv_snprintf
 #define vsnprintf avpriv_vsnprintf
-#endif /* COMPAT_MSVCRT_SNPRINTF_H */
+#endif /* COMPAT_SNPRINTF_H */
--- a/compat/os2threads.h
+++ b/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.
 *
@@ -23,118 +23,96 @@
 * os2threads to pthreads wrapper
 */
-#ifndef COMPAT_OS2THREADS_H
+#ifndef AVCODEC_OS2PTHREADS_H
-#define COMPAT_OS2THREADS_H
+#define AVCODEC_OS2PTHREADS_H
 #define INCL_DOS
 #define INCL_DOSERRORS
 #include <os2.h>
 #undef __STRICT_ANSI__          /* for _beginthread() */
 #include <stdlib.h>
 #include <time.h>
-#include <sys/builtin.h>
+#include "libavutil/mem.h"
 #include <sys/fmutex.h>
 #include "libavutil/attributes.h"
 #include "libavutil/common.h"
 #include "libavutil/time.h"
 typedef struct {
    TID tid;
    void *(*start_routine)(void *);
    void *arg;
    void *result;
 } pthread_t;
 typedef TID  pthread_t;
 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  event_sem;
-    HEV ack_sem;
+    int  wait_count;
    volatile unsigned  wait_count;
 } pthread_cond_t;
 typedef void pthread_condattr_t;
-typedef struct {
+struct thread_arg {
-    volatile int done;
+    void *(*start_routine)(void *);
-    _fmutex mtx;
+    void *arg;
-} pthread_once_t;
+};
 #define PTHREAD_ONCE_INIT {0, _FMUTEX_INITIALIZER}
 static void thread_entry(void *arg)
 {
-    pthread_t *thread = arg;
+    struct thread_arg *thread_arg = arg;
-    thread->result = thread->start_routine(thread->arg);
+    thread_arg->start_routine(thread_arg->arg);
    av_free(thread_arg);
 }
-static av_always_inline int pthread_create(pthread_t *thread,
+static av_always_inline int pthread_create(pthread_t *thread, const pthread_attr_t *attr, void *(*start_routine)(void*), void *arg)
                                           const pthread_attr_t *attr,
                                           void *(*start_routine)(void*),
                                           void *arg)
 {
-    thread->start_routine = start_routine;
+    struct thread_arg *thread_arg;
    thread->arg = arg;
    thread->result = NULL;
-    thread->tid = _beginthread(thread_entry, NULL, 1024 * 1024, thread);
+    thread_arg = av_mallocz(sizeof(struct thread_arg));
    thread_arg->start_routine = start_routine;
    thread_arg->arg = arg;
    *thread = _beginthread(thread_entry, NULL, 256 * 1024, thread_arg);
    return 0;
 }
 static av_always_inline int pthread_join(pthread_t thread, void **value_ptr)
 {
-    DosWaitThread(&thread.tid, DCWW_WAIT);
+    DosWaitThread((PTID)&thread, DCWW_WAIT);
    if (value_ptr)
        *value_ptr = thread.result;
    return 0;
 }
-static av_always_inline int pthread_mutex_init(pthread_mutex_t *mutex,
+static av_always_inline int pthread_mutex_init(pthread_mutex_t *mutex, const pthread_mutexattr_t *attr)
                                               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;
 }
-static av_always_inline int pthread_cond_init(pthread_cond_t *cond,
+static av_always_inline int pthread_cond_init(pthread_cond_t *cond, const pthread_condattr_t *attr)
                                              const pthread_condattr_t *attr)
 {
    DosCreateEventSem(NULL, &cond->event_sem, DCE_POSTONE, FALSE);
    DosCreateEventSem(NULL, &cond->ack_sem, DCE_POSTONE, FALSE);
    cond->wait_count = 0;
@@ -144,16 +122,16 @@ static av_always_inline int pthread_cond_init(pthread_cond_t *cond,
 static av_always_inline int pthread_cond_destroy(pthread_cond_t *cond)
 {
    DosCloseEventSem(cond->event_sem);
    DosCloseEventSem(cond->ack_sem);
    return 0;
 }
 static av_always_inline int pthread_cond_signal(pthread_cond_t *cond)
 {
-    if (!__atomic_cmpxchg32(&cond->wait_count, 0, 0)) {
+    if (cond->wait_count > 0) {
        DosPostEventSem(cond->event_sem);
-        DosWaitEventSem(cond->ack_sem, SEM_INDEFINITE_WAIT);
+
        cond->wait_count--;
    }
    return 0;
@@ -161,69 +139,26 @@ static av_always_inline int pthread_cond_signal(pthread_cond_t *cond)
 static av_always_inline int pthread_cond_broadcast(pthread_cond_t *cond)
 {
-    while (!__atomic_cmpxchg32(&cond->wait_count, 0, 0))
+    while (cond->wait_count > 0) {
-        pthread_cond_signal(cond);
+        DosPostEventSem(cond->event_sem);
        cond->wait_count--;
    }
    return 0;
 }
-static av_always_inline int pthread_cond_timedwait(pthread_cond_t *cond,
+static av_always_inline int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
                                                   pthread_mutex_t *mutex,
                                                   const struct timespec *abstime)
 {
-    int64_t abs_milli = abstime->tv_sec * 1000LL + abstime->tv_nsec / 1000000;
+    cond->wait_count++;
    ULONG t = av_clip64(abs_milli - av_gettime() / 1000, 0, ULONG_MAX);
    __atomic_increment(&cond->wait_count);
    pthread_mutex_unlock(mutex);
    APIRET ret = DosWaitEventSem(cond->event_sem, t);
    __atomic_decrement(&cond->wait_count);
    DosPostEventSem(cond->ack_sem);
    pthread_mutex_lock(mutex);
    return (ret == ERROR_TIMEOUT) ? ETIMEDOUT : 0;
 }
 static av_always_inline int pthread_cond_wait(pthread_cond_t *cond,
                                              pthread_mutex_t *mutex)
 {
    __atomic_increment(&cond->wait_count);
    pthread_mutex_unlock(mutex);
    DosWaitEventSem(cond->event_sem, SEM_INDEFINITE_WAIT);
    __atomic_decrement(&cond->wait_count);
    DosPostEventSem(cond->ack_sem);
    pthread_mutex_lock(mutex);
    return 0;
 }
-static av_always_inline int pthread_once(pthread_once_t *once_control,
+#endif /* AVCODEC_OS2PTHREADS_H */
                                         void (*init_routine)(void))
 {
    if (!once_control->done)
    {
        _fmutex_request(&once_control->mtx, 0);
        if (!once_control->done)
        {
            init_routine();
            once_control->done = 1;
        }
        _fmutex_release(&once_control->mtx);
    }
    return 0;
 }
 #endif /* COMPAT_OS2THREADS_H */
--- a/compat/plan9/head
+++ b/compat/plan9/head
@@ -0,0 +1,10 @@
 #!/bin/sh
 n=10
 case "$1" in
    -n)  n=$2;      shift 2 ;;
    -n*) n=${1#-n}; shift   ;;
 esac
 exec sed ${n}q "$@"
--- a/compat/plan9/main.c
+++ b/compat/plan9/main.c
@@ -0,0 +1,34 @@
 /*
 * 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
 */
 int plan9_main(int argc, char **argv);
 #undef main
 int main(int argc, char **argv)
 {
    /* The setfcr() function in lib9 is broken, must use asm. */
 #ifdef __i386
    short fcr;
    __asm__ volatile ("fstcw        %0 \n"
                      "or      $63, %0 \n"
                      "fldcw        %0 \n"
                      : "=m"(fcr));
 #endif
    return plan9_main(argc, argv);
 }
--- a/compat/plan9/printf
+++ b/compat/plan9/printf
@@ -0,0 +1,2 @@
 #!/bin/sh
 exec awk "BEGIN { for (i = 2; i < ARGC; i++) printf \"$1\", ARGV[i] }" "$@"
--- a/compat/solaris/make_sunver.pl
+++ b/compat/solaris/make_sunver.pl
@@ -1,352 +0,0 @@
 #!/usr/bin/env perl
 # make_sunver.pl
 #
 #   Copyright (C) 2010, 2011, 2012, 2013
 #   Free Software Foundation, Inc.
 #
 # This file is free software; you can redistribute it and/or modify it
 # under the terms of the GNU General Public License as published by
 # the Free Software Foundation; either version 3 of the License, or
 # (at your option) any later version.
 #
 # This program is distributed in the hope that it will be useful, but
 # WITHOUT ANY WARRANTY; without even the implied warranty of
 # MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 # General Public License for more details.
 #
 # You should have received a copy of the GNU General Public License
 # along with this program; see the file COPYING.GPLv3.  If not see
 # <http://www.gnu.org/licenses/>.
 # This script takes at least two arguments, a GNU style version script and
 # a list of object and archive files, and generates a corresponding Sun
 # style version script as follows:
 #
 # Each glob pattern, C++ mangled pattern or literal in the input script is
 # matched against all global symbols in the input objects, emitting those
 # that matched (or nothing if no match was found).
 # A comment with the original pattern and its type is left in the output
 # file to make it easy to understand the matches.
 #
 # It uses elfdump when present (native), GNU readelf otherwise.
 # It depends on the GNU version of c++filt, since it must understand the
 # GNU mangling style.
 use FileHandle;
 use IPC::Open2;
 # Enforce C locale.
 $ENV{'LC_ALL'} = "C";
 $ENV{'LANG'} = "C";
 # Input version script, GNU style.
 my $symvers = shift;
 ##########
 # Get all the symbols from the library, match them, and add them to a hash.
 my %sym_hash = ();
 # List of objects and archives to process.
 my @OBJECTS = ();
 # List of shared objects to omit from processing.
 my @SHAREDOBJS = ();
 # Filter out those input archives that have corresponding shared objects to
 # avoid adding all symbols matched in the archive to the output map.
 foreach $file (@ARGV) {
    if (($so = $file) =~ s/\.a$/.so/ && -e $so) {
    printf STDERR "omitted $file -> $so\n";
    push (@SHAREDOBJS, $so);
    } else {
    push (@OBJECTS, $file);
    }
 }
 # We need to detect and ignore hidden symbols.  Solaris nm can only detect
 # this in the harder to parse default output format, and GNU nm not at all,
 # so use elfdump -s in the native case and GNU readelf -s otherwise.
 # GNU objdump -t cannot be used since it produces a variable number of
 # columns.
 # The path to elfdump.
 my $elfdump = "/usr/ccs/bin/elfdump";
 if (-f $elfdump) {
    open ELFDUMP,$elfdump.' -s '.(join ' ',@OBJECTS).'|' or die $!;
    my $skip_arsym = 0;
    while (<ELFDUMP>) {
    chomp;
    # Ignore empty lines.
    if (/^$/) {
        # End of archive symbol table, stop skipping.
        $skip_arsym = 0 if $skip_arsym;
        next;
    }
    # Keep skipping until end of archive symbol table.
    next if ($skip_arsym);
    # Ignore object name header for individual objects and archives.
    next if (/:$/);
    # Ignore table header lines.
    next if (/^Symbol Table Section:/);
    next if (/index.*value.*size/);
    # Start of archive symbol table: start skipping.
    if (/^Symbol Table: \(archive/) {
        $skip_arsym = 1;
        next;
    }
    # Split table.
    (undef, undef, undef, undef, $bind, $oth, undef, $shndx, $name) = split;
    # Error out for unknown input.
    die "unknown input line:\n$_" unless defined($bind);
    # Ignore local symbols.
    next if ($bind eq "LOCL");
    # Ignore hidden symbols.
    next if ($oth eq "H");
    # Ignore undefined symbols.
    next if ($shndx eq "UNDEF");
    # Error out for unhandled cases.
    if ($bind !~ /^(GLOB|WEAK)/ or $oth ne "D") {
        die "unhandled symbol:\n$_";
    }
    # Remember symbol.
    $sym_hash{$name}++;
    }
    close ELFDUMP or die "$elfdump error";
 } else {
    open READELF, 'readelf -s -W '.(join ' ',@OBJECTS).'|' or die $!;
    # Process each symbol.
    while (<READELF>) {
    chomp;
    # Ignore empty lines.
    next if (/^$/);
    # Ignore object name header.
    next if (/^File: .*$/);
    # Ignore table header lines.
    next if (/^Symbol table.*contains.*:/);
    next if (/Num:.*Value.*Size/);
    # Split table.
    (undef, undef, undef, undef, $bind, $vis, $ndx, $name) = split;
    # Error out for unknown input.
    die "unknown input line:\n$_" unless defined($bind);
    # Ignore local symbols.
    next if ($bind eq "LOCAL");
    # Ignore hidden symbols.
    next if ($vis eq "HIDDEN");
    # Ignore undefined symbols.
    next if ($ndx eq "UND");
    # Error out for unhandled cases.
    if ($bind !~ /^(GLOBAL|WEAK)/ or $vis ne "DEFAULT") {
        die "unhandled symbol:\n$_";
    }
    # Remember symbol.
    $sym_hash{$name}++;
    }
    close READELF or die "readelf error";
 }
 ##########
 # The various types of glob patterns.
 #
 # A glob pattern that is to be applied to the demangled name: 'cxx'.
 # A glob patterns that applies directly to the name in the .o files: 'glob'.
 # This pattern is ignored; used for local variables (usually just '*'): 'ign'.
 # The type of the current pattern.
 my $glob = 'glob';
 # We're currently inside `extern "C++"', which Sun ld doesn't understand.
 my $in_extern = 0;
 # The c++filt command to use.  This *must* be GNU c++filt; the Sun Studio
 # c++filt doesn't handle the GNU mangling style.
 my $cxxfilt = $ENV{'CXXFILT'} || "c++filt";
 # The current version name.
 my $current_version = "";
 # Was there any attempt to match a symbol to this version?
 my $matches_attempted;
 # The number of versions which matched this symbol.
 my $matched_symbols;
 open F,$symvers or die $!;
 # Print information about generating this file
 print "# This file was generated by make_sunver.pl.  DO NOT EDIT!\n";
 print "# It was generated by:\n";
 printf "# %s %s %s\n", $0, $symvers, (join ' ',@ARGV);
 printf "# Omitted archives with corresponding shared libraries: %s\n",
    (join ' ', @SHAREDOBJS) if $#SHAREDOBJS >= 0;
 print "#\n\n";
 print "\$mapfile_version 2\n";
 while (<F>) {
    # Lines of the form '};'
    if (/^([ \t]*)(\}[ \t]*;[ \t]*)$/) {
    $glob = 'glob';
    if ($in_extern) {
        $in_extern--;
        print "$1##$2\n";
    } else {
        print;
    }
    next;
    }
    # Lines of the form '} SOME_VERSION_NAME_1.0;'
    if (/^[ \t]*\}[ \tA-Z0-9_.a-z]+;[ \t]*$/) {
    $glob = 'glob';
    # We tried to match symbols agains this version, but none matched.
    # Emit dummy hidden symbol to avoid marking this version WEAK.
    if ($matches_attempted && $matched_symbols == 0) {
        print "  hidden:\n";
        print "    .force_WEAK_off_$current_version = DATA S0x0 V0x0;\n";
    }
    print; next;
    }
    # Comment and blank lines
    if (/^[ \t]*\#/) { print; next; }
    if (/^[ \t]*$/) { print; next; }
    # Lines of the form '{'
    if (/^([ \t]*){$/) {
    if ($in_extern) {
        print "$1##{\n";
    } else {
        print;
    }
    next;
    }
    # Lines of the form 'SOME_VERSION_NAME_1.1 {'
    if (/^([A-Z0-9_.]+)[ \t]+{$/) {
    # Record version name.
    $current_version = $1;
    # Reset match attempts, #matched symbols for this version.
    $matches_attempted = 0;
    $matched_symbols = 0;
    print "SYMBOL_VERSION $1 {\n";
    next;
    }
    # Ignore 'global:'
    if (/^[ \t]*global:$/) { print; next; }
    # After 'local:', globs should be ignored, they won't be exported.
    if (/^[ \t]*local:$/) {
    $glob = 'ign';
    print;
    next;
    }
    # After 'extern "C++"', globs are C++ patterns
    if (/^([ \t]*)(extern \"C\+\+\"[ \t]*)$/) {
    $in_extern++;
    $glob = 'cxx';
    # Need to comment, Sun ld cannot handle this.
    print "$1##$2\n"; next;
    }
    # Chomp newline now we're done with passing through the input file.
    chomp;
    # Catch globs.  Note that '{}' is not allowed in globs by this script,
    # so only '*' and '[]' are available.
    if (/^([ \t]*)([^ \t;{}#]+);?[ \t]*$/) {
    my $ws = $1;
    my $ptn = $2;
    # Turn the glob into a regex by replacing '*' with '.*', '?' with '.'.
    # Keep $ptn so we can still print the original form.
    ($pattern = $ptn) =~ s/\*/\.\*/g;
    $pattern =~ s/\?/\./g;
    if ($glob eq 'ign') {
        # We're in a local: * section; just continue.
        print "$_\n";
        next;
    }
    # Print the glob commented for human readers.
    print "$ws##$ptn ($glob)\n";
    # We tried to match a symbol to this version.
    $matches_attempted++;
    if ($glob eq 'glob') {
        my %ptn_syms = ();
        # Match ptn against symbols in %sym_hash.
        foreach my $sym (keys %sym_hash) {
        # Maybe it matches one of the patterns based on the symbol in
        # the .o file.
        $ptn_syms{$sym}++ if ($sym =~ /^$pattern$/);
        }
        foreach my $sym (sort keys(%ptn_syms)) {
        $matched_symbols++;
        print "$ws$sym;\n";
        }
    } elsif ($glob eq 'cxx') {
        my %dem_syms = ();
        # Verify that we're actually using GNU c++filt.  Other versions
        # most likely cannot handle GNU style symbol mangling.
        my $cxxout = `$cxxfilt --version 2>&1`;
        $cxxout =~ m/GNU/ or die "$0 requires GNU c++filt to function";
        # Talk to c++filt through a pair of file descriptors.
        # Need to start a fresh instance per pattern, otherwise the
        # process grows to 500+ MB.
        my $pid = open2(*FILTIN, *FILTOUT, $cxxfilt) or die $!;
        # Match ptn against symbols in %sym_hash.
        foreach my $sym (keys %sym_hash) {
        # No?  Well, maybe its demangled form matches one of those
        # patterns.
        printf FILTOUT "%s\n",$sym;
        my $dem = <FILTIN>;
        chomp $dem;
        $dem_syms{$sym}++ if ($dem =~ /^$pattern$/);
        }
        close FILTOUT or die "c++filt error";
        close FILTIN or die "c++filt error";
        # Need to wait for the c++filt process to avoid lots of zombies.
        waitpid $pid, 0;
        foreach my $sym (sort keys(%dem_syms)) {
        $matched_symbols++;
        print "$ws$sym;\n";
        }
    } else {
        # No?  Well, then ignore it.
    }
    next;
    }
    # Important sanity check.  This script can't handle lots of formats
    # that GNU ld can, so be sure to error out if one is seen!
    die "strange line `$_'";
 }
 close F;
--- a/compat/strtod.c
+++ b/compat/strtod.c
@@ -25,9 +25,9 @@
 #include "libavutil/avstring.h"
 #include "libavutil/mathematics.h"
-static const char *check_nan_suffix(const char *s)
+static char *check_nan_suffix(char *s)
 {
-    const char *start = s;
+    char *start = s;
    if (*s++ != '(')
        return start;
@@ -44,7 +44,7 @@ double strtod(const char *, char **);
 double avpriv_strtod(const char *nptr, char **endptr)
 {
-    const char *end;
+    char *end;
    double res;
    /* Skip leading spaces */
@@ -81,13 +81,13 @@ double avpriv_strtod(const char *nptr, char **endptr)
               !av_strncasecmp(nptr, "+0x", 3)) {
        /* FIXME this doesn't handle exponents, non-integers (float/double)
         * and numbers too large for long long */
-        res = strtoll(nptr, (char **)&end, 16);
+        res = strtoll(nptr, &end, 16);
    } else {
-        res = strtod(nptr, (char **)&end);
+        res = strtod(nptr, &end);
    }
    if (endptr)
-        *endptr = (char *)end;
+        *endptr = end;
    return res;
 }
--- a/compat/tms470/math.h
+++ b/compat/tms470/math.h
@@ -0,0 +1,30 @@
 /*
 * 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 FFMPEG_COMPAT_TMS470_MATH_H
 #define FFMPEG_COMPAT_TMS470_MATH_H
 #include_next <math.h>
 #undef INFINITY
 #undef NAN
 #define INFINITY (*(const float*)((const unsigned []){ 0x7f800000 }))
 #define NAN      (*(const float*)((const unsigned []){ 0x7fc00000 }))
 #endif /* FFMPEG_COMPAT_TMS470_MATH_H */
--- a/compat/va_copy.h
+++ b/compat/va_copy.h
@@ -19,9 +19,6 @@
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
 #ifndef COMPAT_VA_COPY_H
 #define COMPAT_VA_COPY_H
 #include <stdarg.h>
 #if !defined(va_copy) && defined(_MSC_VER)
@@ -30,5 +27,3 @@
 #if !defined(va_copy) && defined(__GNUC__) && __GNUC__ < 3
 #define va_copy(dst, src) __va_copy(dst, src)
 #endif
 #endif /* COMPAT_VA_COPY_H */
--- a/compat/w32dlfcn.h
+++ b/compat/w32dlfcn.h
@@ -1,150 +0,0 @@
 /*
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
 #ifndef COMPAT_W32DLFCN_H
 #define COMPAT_W32DLFCN_H
 #ifdef _WIN32
 #include <stdint.h>
 #include <windows.h>
 #include "config.h"
 #include "libavutil/macros.h"
 #include "libavutil/wchar_filename.h"
 static inline wchar_t *get_module_filename(HMODULE module)
 {
    wchar_t *path = NULL, *new_path;
    DWORD path_size = 0, path_len;
    do {
        path_size = path_size ? FFMIN(2 * path_size, INT16_MAX + 1) : MAX_PATH;
        new_path = av_realloc_array(path, path_size, sizeof *path);
        if (!new_path) {
            av_free(path);
            return NULL;
        }
        path = new_path;
        // Returns path_size in case of insufficient buffer.
        // Whether the error is set or not and whether the output
        // is null-terminated or not depends on the version of Windows.
        path_len = GetModuleFileNameW(module, path, path_size);
    } while (path_len && path_size <= INT16_MAX && path_size <= path_len);
    if (!path_len) {
        av_free(path);
        return NULL;
    }
    return path;
 }
 /**
 * Safe function used to open dynamic libs. This attempts to improve program security
 * by removing the current directory from the dll search path. Only dll's found in the
 * executable or system directory are allowed to be loaded.
 * @param name  The dynamic lib name.
 * @return A handle to the opened lib.
 */
 static inline HMODULE win32_dlopen(const char *name)
 {
    wchar_t *name_w;
    HMODULE module = NULL;
    if (utf8towchar(name, &name_w))
        name_w = NULL;
 #if _WIN32_WINNT < 0x0602
    // On Win7 and earlier we check if KB2533623 is available
    if (!GetProcAddress(GetModuleHandleW(L"kernel32.dll"), "SetDefaultDllDirectories")) {
        wchar_t *path = NULL, *new_path;
        DWORD pathlen, pathsize, namelen;
        if (!name_w)
            goto exit;
        namelen = wcslen(name_w);
        // Try local directory first
        path = get_module_filename(NULL);
        if (!path)
            goto exit;
        new_path = wcsrchr(path, '\\');
        if (!new_path)
            goto exit;
        pathlen = new_path - path;
        pathsize = pathlen + namelen + 2;
        new_path = av_realloc_array(path, pathsize, sizeof *path);
        if (!new_path)
            goto exit;
        path = new_path;
        wcscpy(path + pathlen + 1, name_w);
        module = LoadLibraryExW(path, NULL, LOAD_WITH_ALTERED_SEARCH_PATH);
        if (module == NULL) {
            // Next try System32 directory
            pathlen = GetSystemDirectoryW(path, pathsize);
            if (!pathlen)
                goto exit;
            // Buffer is not enough in two cases:
            // 1. system directory + \ + module name
            // 2. system directory even without the module name.
            if (pathlen + namelen + 2 > pathsize) {
                pathsize = pathlen + namelen + 2;
                new_path = av_realloc_array(path, pathsize, sizeof *path);
                if (!new_path)
                    goto exit;
                path = new_path;
                // Query again to handle the case #2.
                pathlen = GetSystemDirectoryW(path, pathsize);
                if (!pathlen)
                    goto exit;
            }
            path[pathlen] = L'\\';
            wcscpy(path + pathlen + 1, name_w);
            module = LoadLibraryExW(path, NULL, LOAD_WITH_ALTERED_SEARCH_PATH);
        }
 exit:
        av_free(path);
        av_free(name_w);
        return module;
    }
 #endif
 #ifndef LOAD_LIBRARY_SEARCH_APPLICATION_DIR
 #   define LOAD_LIBRARY_SEARCH_APPLICATION_DIR 0x00000200
 #endif
 #ifndef LOAD_LIBRARY_SEARCH_SYSTEM32
 #   define LOAD_LIBRARY_SEARCH_SYSTEM32        0x00000800
 #endif
 #if HAVE_WINRT
    if (!name_w)
        return NULL;
    module = LoadPackagedLibrary(name_w, 0);
 #else
 #define LOAD_FLAGS (LOAD_LIBRARY_SEARCH_APPLICATION_DIR | LOAD_LIBRARY_SEARCH_SYSTEM32)
    /* filename may be be in CP_ACP */
    if (!name_w)
        return LoadLibraryExA(name, NULL, LOAD_FLAGS);
    module = LoadLibraryExW(name_w, NULL, LOAD_FLAGS);
 #undef LOAD_FLAGS
 #endif
    av_free(name_w);
    return module;
 }
 #define dlopen(name, flags) win32_dlopen(name)
 #define dlclose FreeLibrary
 #define dlsym GetProcAddress
 #else
 #include <dlfcn.h>
 #endif
 #endif /* COMPAT_W32DLFCN_H */
--- a/compat/w32pthreads.h
+++ b/compat/w32pthreads.h
@@ -26,8 +26,8 @@
 * w32threads to pthreads wrapper
 */
-#ifndef COMPAT_W32PTHREADS_H
+#ifndef FFMPEG_COMPAT_W32PTHREADS_H
-#define COMPAT_W32PTHREADS_H
+#define FFMPEG_COMPAT_W32PTHREADS_H
 /* Build up a pthread-like API using underlying Windows API. Have only static
 * methods so as to not conflict with a potentially linked in pthread-win32
@@ -38,13 +38,10 @@
 #define WIN32_LEAN_AND_MEAN
 #include <windows.h>
 #include <process.h>
 #include <time.h>
 #include "libavutil/attributes.h"
 #include "libavutil/common.h"
 #include "libavutil/internal.h"
 #include "libavutil/mem.h"
 #include "libavutil/time.h"
 typedef struct pthread_t {
    void *handle;
@@ -53,139 +50,234 @@ typedef struct pthread_t {
    void *ret;
 } pthread_t;
-/* use light weight mutex/condition variable API for Windows Vista and later */
+/* the conditional variable api for windows 6.0+ uses critical sections and
-typedef SRWLOCK pthread_mutex_t;
+ * not mutexes */
-typedef CONDITION_VARIABLE pthread_cond_t;
+typedef CRITICAL_SECTION pthread_mutex_t;
-#define PTHREAD_MUTEX_INITIALIZER SRWLOCK_INIT
+/* This is the CONDITIONAL_VARIABLE typedef for using Window's native
-#define PTHREAD_COND_INITIALIZER CONDITION_VARIABLE_INIT
+ * conditional variables on kernels 6.0+.
 * MinGW does not currently have this typedef. */
 typedef struct pthread_cond_t {
    void *ptr;
 } pthread_cond_t;
-#define InitializeCriticalSection(x) InitializeCriticalSectionEx(x, 0, 0)
+/* function pointers to conditional variable API on windows 6.0+ kernels */
-#define WaitForSingleObject(a, b) WaitForSingleObjectEx(a, b, FALSE)
+#if _WIN32_WINNT < 0x0600
 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);
 #else
 #define cond_init      InitializeConditionVariable
 #define cond_broadcast WakeAllConditionVariable
 #define cond_signal    WakeConditionVariable
 #define cond_wait      SleepConditionVariableCS
 #endif
-#define PTHREAD_CANCEL_ENABLE 1
+static unsigned __stdcall attribute_align_arg win32thread_worker(void *arg)
 #define PTHREAD_CANCEL_DISABLE 0
 static av_unused unsigned __stdcall attribute_align_arg win32thread_worker(void *arg)
 {
-    pthread_t *h = (pthread_t*)arg;
+    pthread_t *h = arg;
    h->ret = h->func(h->arg);
    return 0;
 }
-static av_unused int pthread_create(pthread_t *thread, const void *unused_attr,
+static int pthread_create(pthread_t *thread, const void *unused_attr,
-                                    void *(*start_routine)(void*), void *arg)
+                          void *(*start_routine)(void*), void *arg)
 {
    thread->func   = start_routine;
    thread->arg    = arg;
 #if HAVE_WINRT
    thread->handle = (void*)CreateThread(NULL, 0, win32thread_worker, thread,
                                           0, NULL);
 #else
    thread->handle = (void*)_beginthreadex(NULL, 0, win32thread_worker, thread,
                                           0, NULL);
 #endif
    return !thread->handle;
 }
-static av_unused int pthread_join(pthread_t thread, void **value_ptr)
+static void pthread_join(pthread_t thread, void **value_ptr)
 {
    DWORD ret = WaitForSingleObject(thread.handle, INFINITE);
-    if (ret != WAIT_OBJECT_0) {
+    if (ret != WAIT_OBJECT_0)
-        if (ret == WAIT_ABANDONED)
+        return;
            return EINVAL;
        else
            return EDEADLK;
    }
    if (value_ptr)
        *value_ptr = thread.ret;
    CloseHandle(thread.handle);
    return 0;
 }
 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;
 }
-typedef INIT_ONCE pthread_once_t;
+/* for pre-Windows 6.0 platforms we need to define and use our own condition
-#define PTHREAD_ONCE_INIT INIT_ONCE_STATIC_INIT
+ * 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;
-static av_unused int pthread_once(pthread_once_t *once_control, void (*init_routine)(void))
+static int pthread_cond_init(pthread_cond_t *cond, const void *unused_attr)
 {
-    BOOL pending = FALSE;
+    win32_cond_t *win32_cond = NULL;
-    InitOnceBeginInitialize(once_control, 0, &pending, NULL);
+    if (cond_init) {
-    if (pending)
+        cond_init(cond);
-        init_routine();
+        return 0;
    InitOnceComplete(once_control, 0, NULL);
    return 0;
 }
 static inline int pthread_cond_init(pthread_cond_t *cond, const void *unused_attr)
 {
    InitializeConditionVariable(cond);
    return 0;
 }
 /* native condition variables do not destroy */
 static inline int pthread_cond_destroy(pthread_cond_t *cond)
 {
    return 0;
 }
 static inline int pthread_cond_broadcast(pthread_cond_t *cond)
 {
    WakeAllConditionVariable(cond);
    return 0;
 }
 static inline int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
 {
    SleepConditionVariableSRW(cond, mutex, INFINITE, 0);
    return 0;
 }
 static inline int pthread_cond_timedwait(pthread_cond_t *cond, pthread_mutex_t *mutex,
                                         const struct timespec *abstime)
 {
    int64_t abs_milli = abstime->tv_sec * 1000LL + abstime->tv_nsec / 1000000;
    DWORD t = av_clip64(abs_milli - av_gettime() / 1000, 0, UINT32_MAX);
    if (!SleepConditionVariableSRW(cond, mutex, t, 0)) {
        DWORD err = GetLastError();
        if (err == ERROR_TIMEOUT)
            return ETIMEDOUT;
        else
            return EINVAL;
    }
    /* non native condition variables */
    win32_cond = 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 inline int pthread_cond_signal(pthread_cond_t *cond)
+static void pthread_cond_destroy(pthread_cond_t *cond)
 {
-    WakeConditionVariable(cond);
+    win32_cond_t *win32_cond = cond->ptr;
-    return 0;
+    /* native condition variables do not destroy */
    if (cond_init)
        return;
    /* 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;
 }
-static inline int pthread_setcancelstate(int state, int *oldstate)
+static void pthread_cond_broadcast(pthread_cond_t *cond)
 {
-    return 0;
+    win32_cond_t *win32_cond = cond->ptr;
    int have_waiter;
    if (cond_broadcast) {
        cond_broadcast(cond);
        return;
    }
    /* 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);
 }
-#endif /* COMPAT_W32PTHREADS_H */
+static int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
 {
    win32_cond_t *win32_cond = 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 void pthread_cond_signal(pthread_cond_t *cond)
 {
    win32_cond_t *win32_cond = cond->ptr;
    int have_waiter;
    if (cond_signal) {
        cond_signal(cond);
        return;
    }
    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);
 }
 static void w32thread_init(void)
 {
 #if _WIN32_WINNT < 0x0600
    HANDLE kernel_dll = GetModuleHandle(TEXT("kernel32.dll"));
    /* if one is available, then they should all be available */
    cond_init      =
        (void*)GetProcAddress(kernel_dll, "InitializeConditionVariable");
    cond_broadcast =
        (void*)GetProcAddress(kernel_dll, "WakeAllConditionVariable");
    cond_signal    =
        (void*)GetProcAddress(kernel_dll, "WakeConditionVariable");
    cond_wait      =
        (void*)GetProcAddress(kernel_dll, "SleepConditionVariableCS");
 #endif
 }
 #endif /* FFMPEG_COMPAT_W32PTHREADS_H */
--- a/compat/windows/makedef
+++ b/compat/windows/makedef
@@ -45,11 +45,7 @@ libname=$(mktemp -u "library").lib
 trap 'rm -f -- $libname' EXIT
-if [ -n "$AR" ]; then
+lib -out:${libname} $@ >/dev/null
    $AR rcs ${libname} $@ >/dev/null
 else
    lib.exe -out:${libname} $@ >/dev/null
 fi
 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
+dump=$(dumpbin -linkermember:1 ${libname})
    # Use eval, since NM="nm -g"
    dump=$(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
 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
--- a/compat/windows/mslink
+++ b/compat/windows/mslink
@@ -1,9 +0,0 @@
 #!/bin/sh
 LINK_EXE_PATH=$(dirname "$(command -v cl)")/link
 if [ -x "$LINK_EXE_PATH" ]; then
    "$LINK_EXE_PATH" $@
 else
    link.exe $@
 fi
 exit $?
--- a/5716
+++ b/5716
--- a/doc/.gitignore
+++ b/doc/.gitignore
@@ -1,9 +0,0 @@
 /*.1
 /*.3
 /*.html
 /*.pod
 /config.texi
 /avoptions_codec.texi
 /avoptions_format.texi
 /fate.txt
 /print_options
--- a/doc/APIchanges
+++ b/doc/APIchanges
--- a/doc/Doxyfile
+++ b/doc/Doxyfile
--- a/doc/Makefile
+++ b/doc/Makefile
@@ -24,12 +24,8 @@ HTMLPAGES   = $(AVPROGS-yes:%=doc/%.html) $(AVPROGS-yes:%=doc/%-all.html) $(COMP
              doc/fate.html                                             \
              doc/general.html                                          \
              doc/git-howto.html                                        \
              doc/mailing-list-faq.html                                 \
              doc/nut.html                                              \
              doc/platform.html                                         \
              $(SRC_PATH)/doc/bootstrap.min.css                         \
              $(SRC_PATH)/doc/style.min.css                             \
              $(SRC_PATH)/doc/default.css                               \
 TXTPAGES    = doc/fate.txt                                              \
@@ -40,6 +36,26 @@ DOCS-$(CONFIG_MANPAGES)  += $(MANPAGES)
 DOCS-$(CONFIG_TXTPAGES)  += $(TXTPAGES)
 DOCS = $(DOCS-yes)
 DOC_EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE)      += avio_reading
 DOC_EXAMPLES-$(CONFIG_AVCODEC_EXAMPLE)           += avcodec
 DOC_EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding
 DOC_EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE)      += filter_audio
 DOC_EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE)   += filtering_audio
 DOC_EXAMPLES-$(CONFIG_FILTERING_VIDEO_EXAMPLE)   += filtering_video
 DOC_EXAMPLES-$(CONFIG_METADATA_EXAMPLE)          += metadata
 DOC_EXAMPLES-$(CONFIG_MUXING_EXAMPLE)            += muxing
 DOC_EXAMPLES-$(CONFIG_REMUXING_EXAMPLE)          += remuxing
 DOC_EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE)  += resampling_audio
 DOC_EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE)     += scaling_video
 DOC_EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE)     += transcode_aac
 DOC_EXAMPLES-$(CONFIG_TRANSCODING_EXAMPLE)       += transcoding
 ALL_DOC_EXAMPLES_LIST = $(DOC_EXAMPLES-) $(DOC_EXAMPLES-yes)
 DOC_EXAMPLES       := $(DOC_EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)$(EXESUF))
 ALL_DOC_EXAMPLES   := $(ALL_DOC_EXAMPLES_LIST:%=doc/examples/%$(PROGSSUF)$(EXESUF))
 ALL_DOC_EXAMPLES_G := $(ALL_DOC_EXAMPLES_LIST:%=doc/examples/%$(PROGSSUF)_g$(EXESUF))
 PROGS              += $(DOC_EXAMPLES)
 all-$(CONFIG_DOC): doc
 doc: documentation
@@ -47,6 +63,8 @@ doc: documentation
 apidoc: doc/doxy/html
 documentation: $(DOCS)
 examples: $(DOC_EXAMPLES)
 TEXIDEP = perl $(SRC_PATH)/doc/texidep.pl $(SRC_PATH) $< $@ >$(@:%=%.d)
 doc/%.txt: TAG = TXT
@@ -62,25 +80,14 @@ $(GENTEXI): doc/avoptions_%.texi: doc/print_options$(HOSTEXESUF)
 	$(M)doc/print_options $* > $@
 doc/%.html: TAG = HTML
 doc/%-all.html: TAG = HTML
 ifdef HAVE_MAKEINFO_HTML
 doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.pm $(GENTEXI)
 	$(Q)$(TEXIDEP)
 	$(M)makeinfo --html -I doc --no-split -D config-not-all --init-file=$(SRC_PATH)/doc/t2h.pm --output $@ $<
 doc/%-all.html: doc/%.texi $(SRC_PATH)/doc/t2h.pm $(GENTEXI)
 	$(Q)$(TEXIDEP)
 	$(M)makeinfo --html -I doc --no-split -D config-all --init-file=$(SRC_PATH)/doc/t2h.pm --output $@ $<
 else
 doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
 	$(Q)$(TEXIDEP)
 	$(M)texi2html -I doc -monolithic --D=config-not-all --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
 doc/%-all.html: TAG = HTML
 doc/%-all.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
 	$(Q)$(TEXIDEP)
 	$(M)texi2html -I doc -monolithic --D=config-all --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
 endif
 doc/%.pod: TAG = POD
 doc/%.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI)
@@ -94,18 +101,19 @@ doc/%-all.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI)
 doc/%.1 doc/%.3: TAG = MAN
 doc/%.1: doc/%.pod $(GENTEXI)
-	$(M)pod2man --section=1 --center=" " --release=" " --date=" " $< > $@
+	$(M)pod2man --section=1 --center=" " --release=" " $< > $@
 doc/%.3: doc/%.pod $(GENTEXI)
-	$(M)pod2man --section=3 --center=" " --release=" " --date=" " $< > $@
+	$(M)pod2man --section=3 --center=" " --release=" " $< > $@
 $(DOCS) doc/doxy/html: | doc/
 $(DOC_EXAMPLES:%$(EXESUF)=%.o): | doc/examples
 OBJDIRS += doc/examples
-DOXY_INPUT      = $(INSTHEADERS)
+DOXY_INPUT      = $(addprefix $(SRC_PATH)/, $(INSTHEADERS) $(DOC_EXAMPLES:%$(EXESUF)=%.c) $(LIB_EXAMPLES:%$(EXESUF)=%.c))
 DOXY_INPUT_DEPS = $(addprefix $(SRC_PATH)/, $(DOXY_INPUT)) ffbuild/config.mak
 doc/doxy/html: TAG = DOXY
-doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(SRC_PATH)/doc/doxy-wrapper.sh $(DOXY_INPUT_DEPS)
+doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(SRC_PATH)/doc/doxy-wrapper.sh $(DOXY_INPUT)
-	$(M)$(SRC_PATH)/doc/doxy-wrapper.sh $$PWD/doc/doxy $(SRC_PATH) doc/Doxyfile $(DOXYGEN) $(DOXY_INPUT);
+	$(M)$(SRC_PATH)/doc/doxy-wrapper.sh $(SRC_PATH) $< $(DOXYGEN) $(DOXY_INPUT)
 install-doc: install-html install-man
@@ -147,7 +155,11 @@ clean:: docclean
 distclean:: docclean
 	$(RM) doc/config.texi
-docclean::
+examplesclean:
 	$(RM) $(ALL_DOC_EXAMPLES) $(ALL_DOC_EXAMPLES_G)
 	$(RM) $(CLEANSUFFIXES:%=doc/examples/%)
 docclean: examplesclean
 	$(RM) $(CLEANSUFFIXES:%=doc/%)
 	$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 doc/*.3 doc/avoptions_*.texi
 	$(RM) -r doc/doxy/html
--- a/doc/bitstream_filters.texi
+++ b/doc/bitstream_filters.texi
@@ -13,114 +13,40 @@ bitstream filter using the option @code{--disable-bsf=BSF}.
 The option @code{-bsfs} of the ff* tools will display the list of
 all the supported bitstream filters included in your build.
-The ff* tools have a -bsf option applied per stream, taking a
+Below is a description of the currently available bitstream filters.
 comma-separated list of filters, whose parameters follow the filter
 name after a '='.
@example
 ffmpeg -i INPUT -c:v copy -bsf:v filter1[=opt1=str1:opt2=str2][,filter2] OUTPUT
@end example
 Below is a description of the currently available bitstream filters,
 with their parameters, if any.
@section aac_adtstoasc
-Convert MPEG-2/4 AAC ADTS to an MPEG-4 Audio Specific Configuration
+Convert MPEG-2/4 AAC ADTS to MPEG-4 Audio Specific Configuration
-bitstream.
+bitstream filter.
 This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4
 ADTS header and removes the ADTS header.
-This filter is required for example when copying an AAC stream from a
+This is required for example when copying an AAC stream from a raw
-raw ADTS AAC or an MPEG-TS container to MP4A-LATM, to an FLV file, or
+ADTS AAC container to a FLV or a MOV/MP4 file.
 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{time_scale / num_units_in_display_tick}) 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.
@section dca_core
 Extract the core from a DCA/DTS stream, dropping extensions such as
 DTS-HD.
@section dump_extra
-Add extradata to the beginning of the filtered packets except when
+Add extradata to the beginning of the filtered packets.
 said packets already exactly begin with the extradata that is intended
 to be added.
@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}.
@@ -132,231 +58,6 @@ the header stored in extradata to the key packets:
 ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts
@end example
@section dv_error_marker
 Blocks in DV which are marked as damaged are replaced by blocks of the specified color.
@table @option
@item color
 The color to replace damaged blocks by
@item sta
 A 16 bit mask which specifies which of the 16 possible error status values are
 to be replaced by colored blocks. 0xFFFE is the default which replaces all non 0
 error status values.
@table @samp
@item ok
 No error, no concealment
@item err
 Error, No concealment
@item res
 Reserved
@item notok
 Error or concealment
@item notres
 Not reserved
@item Aa, Ba, Ca, Ab, Bb, Cb, A, B, C, a, b, erri, erru
 The specific error status code
@end table
 see page 44-46 or section 5.5 of
@url{http://web.archive.org/web/20060927044735/http://www.smpte.org/smpte_store/standards/pdf/s314m.pdf}
@end table
@section eac3_core
 Extract the core from a E-AC-3 stream, dropping extra channels.
@section extract_extradata
 Extract the in-band extradata.
 Certain codecs allow the long-term headers (e.g. MPEG-2 sequence headers,
 or H.264/HEVC (VPS/)SPS/PPS) to be transmitted either "in-band" (i.e. as a part
 of the bitstream containing the coded frames) or "out of band" (e.g. on the
 container level). This latter form is called "extradata" in FFmpeg terminology.
 This bitstream filter detects the in-band headers and makes them available as
 extradata.
@table @option
@item remove
 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 pass
@item insert
@item remove
@end table
 Default is pass.
@item sample_aspect_ratio
 Set the sample aspect ratio of the stream in the VUI parameters.
 See H.264 table E-1.
@item overscan_appropriate_flag
 Set whether the stream is suitable for display using overscan
 or not (see H.264 section E.2.1).
@item video_format
@item video_full_range_flag
 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 (time_scale / num_units_in_tick) 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 zero_new_constraint_set_flags
 Zero constraint_set4_flag and constraint_set5_flag in the SPS. These
 bits were reserved in a previous version of the H.264 spec, and thus
 some hardware decoders require these to be zero. The result of zeroing
 this is still a valid bitstream.
@item crop_left
@item crop_right
@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 display_orientation
 Insert, extract or remove Display orientation SEI messages.
 See H.264 section D.1.27 and D.2.27 for syntax and semantics.
@table @samp
@item pass
@item insert
@item remove
@item extract
@end table
 Default is pass.
 Insert mode works in conjunction with @code{rotate} and @code{flip} options.
 Any pre-existing Display orientation messages will be removed in insert or remove mode.
 Extract mode attaches the display matrix to the packet as side data.
@item rotate
 Set rotation in display orientation SEI (anticlockwise angle in degrees).
 Range is -360 to +360. Default is NaN.
@item flip
 Set flip in display orientation SEI.
@table @samp
@item horizontal
@item vertical
@end table
 Default is unset.
@item level
 Set the level in the SPS.  Refer to H.264 section A.3 and tables A-1
 to A-5.
 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
@@ -364,7 +65,7 @@ prefixed mode (as defined in the Annex B of the ITU-T H.264
 specification).
 This is required by some streaming formats, typically the MPEG-2
-transport stream format (muxer @code{mpegts}).
+transport stream format ("mpegts").
 For example to remux an MP4 file containing an H.264 stream to mpegts
 format with @command{ffmpeg}, you can use the command:
@@ -373,101 +74,6 @@ format with @command{ffmpeg}, you can use the command:
 ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts
@end example
 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 (time_scale /
 num_units_in_tick). 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
 prefixed mode (as defined in the Annex B of the ITU-T H.265
 specification).
 This is required by some streaming formats, typically the MPEG-2
 transport stream format (muxer @code{mpegts}).
 For example to remux an MP4 file containing an HEVC stream to mpegts
 format with @command{ffmpeg}, you can use the command:
@example
 ffmpeg -i INPUT.mp4 -codec copy -bsf:v hevc_mp4toannexb OUTPUT.ts
@end example
 Please note that this filter is auto-inserted for MPEG-TS (muxer
@code{mpegts}) and raw HEVC/H.265 (muxer @code{h265} or
@code{hevc}) output formats.
@section imxdump
 Modifies the bitstream to fit in MOV and to be usable by the Final Cut
@@ -518,435 +124,21 @@ exiftran -i -9 frame*.jpg
 ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
@end example
-@section mjpegadump
+@section mjpega_dump_header
-Add an MJPEG A header to the bitstream, to enable decoding by
+@section movsub
 Quicktime.
-@anchor{mov2textsub}
+@section mp3_header_decompress
@section mov2textsub
 Extract a representable text file from MOV subtitles, stripping the
 metadata header from each subtitle packet.
 See also the @ref{text2movsub} filter.
@section mp3decomp
 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.
 DivX-style packed B-frames are not valid MPEG-4 and were only a
 workaround for the broken Video for Windows subsystem.
 They use more space, can cause minor AV sync issues, require more
 CPU power to decode (unless the player has some decoded picture queue
 to compensate the 2,0,2,0 frame per packet style) and cause
 trouble if copied into a standard container like mp4 or mpeg-ps/ts,
 because MPEG-4 decoders may not be able to decode them, since they are
 not valid MPEG-4.
 For example to fix an AVI file containing an MPEG-4 stream with
 DivX-style packed B-frames using @command{ffmpeg}, you can use the command:
@example
 ffmpeg -i INPUT.avi -codec copy -bsf:v mpeg4_unpack_bframes OUTPUT.avi
@end example
@section noise
-Damages the contents of packets or simply drops them without damaging the
+Damages the contents of packets without damaging the container. Can be
-container. Can be used for fuzzing or testing error resilience/concealment.
+used for fuzzing or testing error resilience/concealment.
 Parameters:
@table @option
@item amount
 Accepts an expression whose evaluation per-packet determines how often bytes in that
 packet will be modified. A value below 0 will result in a variable frequency.
 Default is 0 which results in no modification. However, if neither amount nor drop is specified,
 amount will be set to @var{-1}. See below for accepted variables.
@item drop
 Accepts an expression evaluated per-packet whose value determines whether that packet is dropped.
 Evaluation to a positive value results in the packet being dropped. Evaluation to a negative
 value results in a variable chance of it being dropped, roughly inverse in proportion to the magnitude
 of the value. Default is 0 which results in no drops. See below for accepted variables.
@item dropamount
 Accepts a non-negative integer, which assigns a variable chance of it being dropped, roughly inverse
 in proportion to the value. Default is 0 which results in no drops. This option is kept for backwards
 compatibility and is equivalent to setting drop to a negative value with the same magnitude
 i.e. @code{dropamount=4} is the same as @code{drop=-4}. Ignored if drop is also specified.
@end table
 Both @code{amount} and @code{drop} accept expressions containing the following variables:
@table @samp
@item n
 The index of the packet, starting from zero.
@item tb
 The timebase for packet timestamps.
@item pts
 Packet presentation timestamp.
@item dts
 Packet decoding timestamp.
@item nopts
 Constant representing AV_NOPTS_VALUE.
@item startpts
 First non-AV_NOPTS_VALUE PTS seen in the stream.
@item startdts
 First non-AV_NOPTS_VALUE DTS seen in the stream.
@item duration
@itemx d
 Packet duration, in timebase units.
@item pos
 Packet position in input; may be -1 when unknown or not set.
@item size
 Packet size, in bytes.
@item key
 Whether packet is marked as a keyframe.
@item state
 A pseudo random integer, primarily derived from the content of packet payload.
@end table
@subsection Examples
 Apply modification to every byte but don't drop any packets.
@example
-ffmpeg -i INPUT -c copy -bsf noise=1 output.mkv
+ffmpeg -i INPUT -c copy -bsf noise output.mkv
@end example
 Drop every video packet not marked as a keyframe after timestamp 30s but do not
 modify any of the remaining packets.
@example
 ffmpeg -i INPUT -c copy -bsf:v noise=drop='gt(t\,30)*not(key)' output.mkv
@end example
 Drop one second of audio every 10 seconds and add some random noise to the rest.
@example
 ffmpeg -i INPUT -c copy -bsf:a noise=amount=-1:drop='between(mod(t\,10)\,9\,10)' output.mkv
@end example
@section null
 This bitstream filter passes the packets through unchanged.
@section pcm_rechunk
 Repacketize PCM audio to a fixed number of samples per packet or a fixed packet
 rate per second. This is similar to the @ref{asetnsamples,,asetnsamples audio
 filter,ffmpeg-filters} but works on audio packets instead of audio frames.
@table @option
@item nb_out_samples, n
 Set the number of samples per each output audio packet. The number is intended
 as the number of samples @emph{per each channel}. Default value is 1024.
@item pad, p
 If set to 1, the filter will pad the last audio packet with silence, so that it
 will contain the same number of samples (or roughly the same number of samples,
 see @option{frame_rate}) as the previous ones. Default value is 1.
@item frame_rate, r
 This option makes the filter output a fixed number of packets per second instead
 of a fixed number of samples per packet. If the audio sample rate is not
 divisible by the frame rate then the number of samples will not be constant but
 will vary slightly so that each packet will start as close to the frame
 boundary as possible. Using this option has precedence over @option{nb_out_samples}.
@end table
 You can generate the well known 1602-1601-1602-1601-1602 pattern of 48kHz audio
 for NTSC frame rate using the @option{frame_rate} option.
@example
 ffmpeg -f lavfi -i sine=r=48000:d=1 -c pcm_s16le -bsf pcm_rechunk=r=30000/1001 -f framecrc -
@end example
@section pgs_frame_merge
 Merge a sequence of PGS Subtitle segments ending with an "end of display set"
 segment into a single packet.
 This is required by some containers that support PGS subtitles
 (muxer @code{matroska}).
@section prores_metadata
 Modify color property metadata embedded in prores stream.
@table @option
@item color_primaries
 Set the color primaries.
 Available values are:
@table @samp
@item auto
 Keep the same color primaries property (default).
@item unknown
@item bt709
@item bt470bg
 BT601 625
@item smpte170m
 BT601 525
@item bt2020
@item smpte431
 DCI P3
@item smpte432
 P3 D65
@end table
@item transfer_characteristics
 Set the color transfer.
 Available values are:
@table @samp
@item auto
 Keep the same transfer characteristics property (default).
@item unknown
@item bt709
 BT 601, BT 709, BT 2020
@item smpte2084
 SMPTE ST 2084
@item arib-std-b67
 ARIB STD-B67
@end table
@item matrix_coefficients
 Set the matrix coefficient.
 Available values are:
@table @samp
@item auto
 Keep the same colorspace property (default).
@item unknown
@item bt709
@item smpte170m
 BT 601
@item bt2020nc
@end table
@end table
 Set Rec709 colorspace for each frame of the file
@example
 ffmpeg -i INPUT -c copy -bsf:v prores_metadata=color_primaries=bt709:color_trc=bt709:colorspace=bt709 output.mov
@end example
 Set Hybrid Log-Gamma parameters for each frame of the file
@example
 ffmpeg -i INPUT -c copy -bsf:v prores_metadata=color_primaries=bt2020:color_trc=arib-std-b67:colorspace=bt2020nc output.mov
@end example
@section remove_extra
 Remove extradata from packets.
 It accepts the following parameter:
@table @option
@item freq
 Set which frame types to remove extradata from.
@table @samp
@item k
 Remove extradata from non-keyframes only.
@item keyframe
 Remove extradata from keyframes only.
@item e, all
 Remove extradata from all frames.
@end table
@end table
@section setts
 Set PTS and DTS in packets.
 It accepts the following parameters:
@table @option
@item ts
@item pts
@item dts
 Set expressions for PTS, DTS or both.
@item duration
 Set expression for duration.
@item time_base
 Set output time base.
@end table
 The expressions are evaluated through the eval API and can contain the following
 constants:
@table @option
@item N
 The count of the input packet. Starting from 0.
@item TS
 The demux timestamp in input in case of @code{ts} or @code{dts} option or presentation
 timestamp in case of @code{pts} option.
@item POS
 The original position in the file of the packet, or undefined if undefined
 for the current packet
@item DTS
 The demux timestamp in input.
@item PTS
 The presentation timestamp in input.
@item DURATION
 The duration in input.
@item STARTDTS
 The DTS of the first packet.
@item STARTPTS
 The PTS of the first packet.
@item PREV_INDTS
 The previous input DTS.
@item PREV_INPTS
 The previous input PTS.
@item PREV_INDURATION
 The previous input duration.
@item PREV_OUTDTS
 The previous output DTS.
@item PREV_OUTPTS
 The previous output PTS.
@item PREV_OUTDURATION
 The previous output duration.
@item NEXT_DTS
 The next input DTS.
@item NEXT_PTS
 The next input PTS.
@item NEXT_DURATION
 The next input duration.
@item TB
 The timebase of stream packet belongs.
@item TB_OUT
 The output timebase.
@item SR
 The sample rate of stream packet belongs.
@item NOPTS
 The AV_NOPTS_VALUE constant.
@end table
@anchor{text2movsub}
@section text2movsub
 Convert text subtitles to MOV subtitles (as used by the @code{mov_text}
 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.  Note that any frame
 set to RGB will be implicitly set to PC range and that RGB is
 incompatible with profiles 0 and 2.
@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 any value
 imposed by the color space will take precedence over this value.
@table @samp
@item tv
@item pc
@end table
@end table
@section vp9_superframe
 Merge VP9 invisible (alt-ref) frames back into VP9 superframes. This
 fixes merging of split/segmented VP9 streams where the alt-ref frame
 was split from its visible counterpart.
@section vp9_superframe_split
 Split VP9 superframes into single frames.
@section vp9_raw_reorder
 Given a VP9 stream with correct timestamps but possibly out of order,
 insert additional show-existing-frame packets to correct the ordering.
@c man end BITSTREAM FILTERS
--- a/doc/bootstrap.min.css
+++ b/doc/bootstrap.min.css
--- a/doc/build_system.txt
+++ b/doc/build_system.txt
@@ -7,28 +7,20 @@ V
    Disable the default terse mode, the full command issued by make and its
    output will be shown on the screen.
 DBG
    Preprocess x86 external assembler files to a .dbg.asm file in the object
    directory, which then gets compiled. Helps in developing those assembler
    files.
 DESTDIR
    Destination directory for the install targets, useful to prepare packages
    or install FFmpeg in cross-environments.
 GEN
    Set to ‘1’ to generate the missing or mismatched references.
 Makefile targets:
 all
    Default target, builds all the libraries and the executables.
 fate
-    Run the fate test suite, note that you must have installed it.
+    Run the fate test suite, note you must have installed it
 fate-list
-    List all fate/regression test targets.
+    Will list all fate/regression test targets
 install
    Install headers, libraries and programs.
@@ -36,31 +28,29 @@ install
 examples
    Build all examples located in doc/examples.
-checkheaders
+libavformat/output-example
-    Check headers dependencies.
+    Build the libavformat basic example.
-alltools
+libavcodec/api-example
-    Build all tools in tools directory.
+    Build the libavcodec basic example.
 libswscale/swscale-test
    Build the swscale self-test (useful also as example).
 config
-    Reconfigure the project with the current configuration.
+    Reconfigure the project with current configuration.
 tools/target_dec_<decoder>_fuzzer
    Build fuzzer to fuzz the specified decoder.
 tools/target_bsf_<filter>_fuzzer
    Build fuzzer to fuzz the specified bitstream filter.
 Useful standard make commands:
 make -t <target>
-    Touch all files that otherwise would be built, this is useful to reduce
+    Touch all files that otherwise would be build, this is useful to reduce
-    unneeded rebuilding when changing headers, but note that you must force rebuilds
+    unneeded rebuilding when changing headers, but note you must force rebuilds
    of files that actually need it by hand then.
 make -j<num>
-    Rebuild with multiple jobs at the same time. Faster on multi processor systems.
+    rebuild with multiple jobs at the same time. Faster on multi processor systems
 make -k
-    Continue build in case of errors, this is useful for the regression tests
+    continue build in case of errors, this is useful for the regression tests
-    sometimes but note that it will still not run all reg tests.
+    sometimes but note it will still not run all reg tests.
--- a/doc/codecs.texi
+++ b/doc/codecs.texi
@@ -7,7 +7,7 @@ all the encoders and decoders. In addition each codec may support
 so-called private options, which are specific for a given codec.
 Sometimes, a global option may only affect a specific kind of codec,
-and may be nonsensical or ignored by another, so you need to be aware
+and may be unsensical or ignored by another, so you need to be aware
 of the meaning of the specified options. Also some options are
 meant only for decoding or encoding.
@@ -44,20 +44,26 @@ 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
 Use internal 2pass ratecontrol in second pass mode.
@item gray
 Only decode/encode grayscale.
@item emu_edge
 Do not draw edges.
@item psnr
 Set error[?] variables during encoding.
@item truncated
 Input bitstream might be randomly truncated.
@item drop_changed
 Don't output frames whose parameters differ from first decoded frame in stream.
 Error AVERROR_INPUT_CHANGED is returned when a frame is dropped.
@item naq
 Normalize adaptive quantization.
@item ildct
 Use interlaced DCT.
@item low_delay
@@ -65,19 +71,53 @@ Force low delay.
@item global_header
 Place global headers in extradata instead of every keyframe.
@item bitexact
-Only write platform-, build- and time-independent data. (except (I)DCT).
+Use only bitexact stuff (except (I)DCT).
 This ensures that file and data checksums are reproducible and match between
 platforms. Its primary use is for regression testing.
@item aic
 Apply H263 advanced intra coding / mpeg4 ac prediction.
@item cbp
 Deprecated, use mpegvideo private options instead.
@item qprd
 Deprecated, use mpegvideo private options instead.
@item ilme
 Apply interlaced motion estimation.
@item cgop
 Use closed gop.
@item output_corrupt
 Output even potentially corrupted frames.
@end table
@item me_method @var{integer} (@emph{encoding,video})
 Set motion estimation method.
 Possible values:
@table @samp
@item zero
 zero motion estimation (fastest)
@item full
 full motion estimation (slowest)
@item epzs
 EPZS motion estimation (default)
@item esa
 esa motion estimation (alias for full)
@item tesa
 tesa motion estimation
@item dia
 dia motion estimation (alias for epzs)
@item log
 log motion estimation
@item phods
 phods motion estimation
@item x1
 X1 motion estimation
@item hex
 hex motion estimation
@item umh
 umh motion estimation
@item iter
 iter motion estimation
@end table
@item extradata_size @var{integer}
 Set extradata size.
@item time_base @var{rational number}
 Set codec time base.
@@ -87,7 +127,7 @@ should be @code{1 / frame_rate} and timestamp increments should be
 identically 1.
@item g @var{integer} (@emph{encoding,video})
-Set the group of picture (GOP) size. Default value is 12.
+Set the group of picture size. Default value is 12.
@item ar @var{integer} (@emph{decoding/encoding,audio})
 Set audio sampling rate (in Hz).
@@ -96,8 +136,7 @@ Set audio sampling rate (in Hz).
 Set number of audio channels.
@item cutoff @var{integer} (@emph{encoding,audio})
-Set cutoff bandwidth. (Supported only by selected encoders, see
+Set cutoff bandwidth.
 their respective documentation sections.)
@item frame_size @var{integer} (@emph{encoding,audio})
 Set audio frame size.
@@ -144,6 +183,24 @@ Default value is 0.
@item b_qfactor @var{float} (@emph{encoding,video})
 Set qp factor between P and B frames.
@item rc_strategy @var{integer} (@emph{encoding,video})
 Set ratecontrol method.
@item b_strategy @var{integer} (@emph{encoding,video})
 Set strategy to choose between I/P/B-frames.
@item ps @var{integer} (@emph{encoding,video})
 Set RTP payload size in bytes.
@item mv_bits @var{integer}
@item header_bits @var{integer}
@item i_tex_bits @var{integer}
@item p_tex_bits @var{integer}
@item i_count @var{integer}
@item p_count @var{integer}
@item skip_count @var{integer}
@item misc_bits @var{integer}
@item frame_bits @var{integer}
@item codec_tag @var{integer}
@item bug @var{flags} (@emph{decoding,video})
 Workaround not auto detected encoder bugs.
@@ -152,6 +209,8 @@ Possible values:
@table @samp
@item autodetect
@item old_msmpeg4
 some old lavc generated msmpeg4v3 files (no autodetection)
@item xvid_ilace
 Xvid interlacing bug (autodetected if fourcc==XVIX)
@item ump4
@@ -160,6 +219,8 @@ Xvid interlacing bug (autodetected if fourcc==XVIX)
 padding bug (autodetected)
@item amv
@item ac_vlc
 illegal vlc bug (autodetected per fourcc)
@item qpel_chroma
@item std_qpel
@@ -180,13 +241,21 @@ Workaround various bugs in microsoft broken decoders.
 trancated frames
@end table
@item lelim @var{integer} (@emph{encoding,video})
 Set single coefficient elimination threshold for luminance (negative
 values also consider DC coefficient).
@item celim @var{integer} (@emph{encoding,video})
 Set single coefficient elimination threshold for chrominance (negative
 values also consider dc coefficient)
@item strict @var{integer} (@emph{decoding/encoding,audio,video})
 Specify how strictly to follow the standards.
 Possible values:
@table @samp
@item very
-strictly conform to an older more strict version of the spec or reference software
+strictly conform to a 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
@@ -233,8 +302,29 @@ consider things that a sane encoder should not do as an error
@item block_align @var{integer}
@item mpeg_quant @var{integer} (@emph{encoding,video})
 Use MPEG quantizers instead of H.263.
@item qsquish @var{float} (@emph{encoding,video})
 How to keep quantizer between qmin and qmax (0 = clip, 1 = use
 differentiable function).
@item rc_qmod_amp @var{float} (@emph{encoding,video})
 Set experimental quantizer modulation.
@item rc_qmod_freq @var{integer} (@emph{encoding,video})
 Set experimental quantizer modulation.
@item rc_override_count @var{integer}
@item rc_eq @var{string} (@emph{encoding,video})
 Set rate control equation. When computing the expression, besides the
 standard functions defined in the section 'Expression Evaluation', the
 following functions are available: bits2qp(bits), qp2bits(qp). Also
 the following constants are available: iTex pTex tex mv fCode iCount
 mcVar var isI isP isB avgQP qComp avgIITex avgPITex avgPPTex avgBPTex
 avgTex.
@item maxrate @var{integer} (@emph{encoding,audio,video})
 Set max bitrate tolerance (in bits/s). Requires bufsize to be set.
@@ -245,12 +335,18 @@ encode. It is of little use elsewise.
@item bufsize @var{integer} (@emph{encoding,audio,video})
 Set ratecontrol buffer size (in bits).
@item rc_buf_aggressivity @var{float} (@emph{encoding,video})
 Currently useless.
@item i_qfactor @var{float} (@emph{encoding,video})
 Set QP factor between P and I frames.
@item i_qoffset @var{float} (@emph{encoding,video})
 Set QP offset between P and I frames.
@item rc_init_cplx @var{float} (@emph{encoding,video})
 Set initial complexity for 1-pass encoding.
@item dct @var{integer} (@emph{encoding,video})
 Set DCT algorithm.
@@ -315,7 +411,11 @@ Automatically pick a IDCT compatible with the simple one
@item simpleneon
-@item xvid
+@item simplealpha
@item ipp
@item xvidmmx
@item faani
 floating point AAN IDCT
@@ -338,12 +438,22 @@ favor predicting from the previous frame instead of the current
@item bits_per_coded_sample @var{integer}
@item pred @var{integer} (@emph{encoding,video})
 Set prediction method.
 Possible values:
@table @samp
@item left
@item plane
@item median
@end table
@item aspect @var{rational number} (@emph{encoding,video})
 Set sample aspect ratio.
@item sar @var{rational number} (@emph{encoding,video})
 Set sample aspect ratio. Alias to @var{aspect}.
@item debug @var{flags} (@emph{decoding/encoding,audio,video,subtitles})
 Print specific debug info.
@@ -359,27 +469,43 @@ rate control
 macroblock (MB) type
@item qp
 per-block quantization parameter (QP)
@item mv
 motion vector
@item dct_coeff
@item green_metadata
 display complexity metadata for the upcoming frame, GoP or for a given duration.
@item skip
@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
 threading operations
-@item nomc
+@end table
-skip motion compensation
+
@item vismv @var{integer} (@emph{decoding,video})
 Visualize motion vectors (MVs).
 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})
@@ -532,28 +658,13 @@ noise preserving sum of squared differences
@item dia_size @var{integer} (@emph{encoding,video})
 Set diamond type & size for motion estimation.
@table @samp
@item (1024, INT_MAX)
 full motion estimation(slowest)
@item (768, 1024]
 umh motion estimation
@item (512, 768]
 hex motion estimation
@item (256, 512]
 l2s diamond motion estimation
@item [2,256]
 var diamond motion estimation
@item (-1,  2)
 small diamond motion estimation
@item -1
 funny diamond motion estimation
@item (INT_MIN, -1)
 sab diamond motion estimation
@end table
@item last_pred @var{integer} (@emph{encoding,video})
 Set amount of motion predictors from the previous frame.
@item preme @var{integer} (@emph{encoding,video})
 Set pre motion estimation.
@item precmp @var{integer} (@emph{encoding,video})
 Set pre motion estimation compare function.
@@ -597,13 +708,44 @@ Set diamond type & size for motion estimation pre-pass.
@item subq @var{integer} (@emph{encoding,video})
 Set sub pel motion estimation quality.
@item dtg_active_format @var{integer}
@item me_range @var{integer} (@emph{encoding,video})
 Set limit motion vectors range (1023 for DivX player).
@item ibias @var{integer} (@emph{encoding,video})
 Set intra quant bias.
@item pbias @var{integer} (@emph{encoding,video})
 Set inter quant bias.
@item color_table_id @var{integer}
@item global_quality @var{integer} (@emph{encoding,audio,video})
@item coder @var{integer} (@emph{encoding,video})
 Possible values:
@table @samp
@item vlc
 variable length coder / huffman coder
@item ac
 arithmetic coder
@item raw
 raw (no encoding)
@item rle
 run-length coder
@item deflate
 deflate-based coder
@end table
@item context @var{integer} (@emph{encoding,video})
 Set context model.
@item slice_flags @var{integer}
@item xvmc_acceleration @var{integer}
@item mbd @var{integer} (@emph{encoding,video})
 Set macroblock decision algorithm (high quality mode).
@@ -617,16 +759,32 @@ use fewest bits
 use best rate distortion
@end table
@item stream_codec_tag @var{integer}
@item sc_threshold @var{integer} (@emph{encoding,video})
 Set scene change threshold.
@item lmin @var{integer} (@emph{encoding,video})
 Set min lagrange factor (VBR).
@item lmax @var{integer} (@emph{encoding,video})
 Set max lagrange factor (VBR).
@item nr @var{integer} (@emph{encoding,video})
 Set noise reduction.
@item rc_init_occupancy @var{integer} (@emph{encoding,video})
 Set number of bits which should be loaded into the rc buffer before
 decoding starts.
-@item flags2 @var{flags} (@emph{decoding/encoding,audio,video,subtitles})
+@item flags2 @var{flags} (@emph{decoding/encoding,audio,video})
 Possible values:
@table @samp
@item fast
 Allow non spec compliant speedup tricks.
@item sgop
 Deprecated, use mpegvideo private options instead.
@item noout
 Skip bitstream encoding.
@item ignorecrop
@@ -637,44 +795,28 @@ 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 export_mvs
+@item skiprd
-Export motion vectors into frame side-data (see @code{AV_FRAME_DATA_MOTION_VECTORS})
+Deprecated, use mpegvideo private options instead.
 for codecs that support it. See also @file{doc/examples/export_mvs.c}.
@item skip_manual
 Do not skip samples and export skip information as frame side data.
@item ass_ro_flush_noop
 Do not reset ASS ReadOrder field on flush.
@end table
-@item export_side_data @var{flags} (@emph{decoding/encoding,audio,video,subtitles})
+@item error @var{integer} (@emph{encoding,video})
-Possible values:
+@item qns @var{integer} (@emph{encoding,video})
-@table @samp
+Deprecated, use mpegvideo private options instead.
@item mvs
 Export motion vectors into frame side-data (see @code{AV_FRAME_DATA_MOTION_VECTORS})
 for codecs that support it. See also @file{doc/examples/export_mvs.c}.
@item prft
 Export encoder Producer Reference Time into packet side-data (see @code{AV_PKT_DATA_PRFT})
 for codecs that support it.
@item venc_params
 Export video encoding parameters through frame side data (see @code{AV_FRAME_DATA_VIDEO_ENC_PARAMS})
 for codecs that support it. At present, those are H.264 and VP9.
@item film_grain
 Export film grain parameters through frame side data (see @code{AV_FRAME_DATA_FILM_GRAIN_PARAMS}).
 Supported at present by AV1 decoders.
@end table
@item threads @var{integer} (@emph{decoding/encoding,video})
 Set the number of threads to be used, in case the selected codec
 implementation supports multi-threading.
 Possible values:
@table @samp
-@item auto, 0
+@item auto
-automatically select the number of threads to set
+detect a good number of threads
@end table
-Default value is @samp{auto}.
+@item me_threshold @var{integer} (@emph{encoding,video})
 Set motion estimation threshold.
@item mb_threshold @var{integer} (@emph{encoding,video})
 Set macroblock threshold.
@item dc @var{integer} (@emph{encoding,video})
 Set intra_dc_precision.
@@ -690,8 +832,41 @@ Set number of macroblock rows at the bottom which are skipped.
@item profile @var{integer} (@emph{encoding,audio,video})
-Set encoder codec profile. Default value is @samp{unknown}. Encoder specific
+Possible values:
-profiles are documented in the relevant encoder documentation.
+@table @samp
@item unknown
@item aac_main
@item aac_low
@item aac_ssr
@item aac_ltp
@item aac_he
@item aac_he_v2
@item aac_ld
@item aac_eld
@item mpeg2_aac_low
@item mpeg2_aac_he
@item dts
@item dts_es
@item dts_96_24
@item dts_hd_hra
@item dts_hd_ma
@end table
@item level @var{integer} (@emph{encoding,audio,video})
@@ -704,12 +879,67 @@ Possible values:
@item lowres @var{integer} (@emph{decoding,audio,video})
 Decode at 1= 1/2, 2=1/4, 3=1/8 resolutions.
@item skip_threshold @var{integer} (@emph{encoding,video})
 Set frame skip threshold.
@item skip_factor @var{integer} (@emph{encoding,video})
 Set frame skip factor.
@item skip_exp @var{integer} (@emph{encoding,video})
 Set frame skip exponent.
 Negative values behave identical to the corresponding positive ones, except
 that the score is normalized.
 Positive values exist primarily for compatibility reasons and are not so useful.
@item skipcmp @var{integer} (@emph{encoding,video})
 Set frame skip compare function.
 Possible values:
@table @samp
@item sad
 sum of absolute differences, fast (default)
@item sse
 sum of squared errors
@item satd
 sum of absolute Hadamard transformed differences
@item dct
 sum of absolute DCT transformed differences
@item psnr
 sum of squared quantization errors (avoid, low quality)
@item bit
 number of bits needed for the block
@item rd
 rate distortion optimal, slow
@item zero
 0
@item vsad
 sum of absolute vertical differences
@item vsse
 sum of squared vertical differences
@item nsse
 noise preserving sum of squared differences
@item w53
 5/3 wavelet, only used in snow
@item w97
 9/7 wavelet, only used in snow
@item dctmax
@item chroma
@end table
@item border_mask @var{float} (@emph{encoding,video})
 Increase the quantizer for macroblocks close to borders.
@item mblmin @var{integer} (@emph{encoding,video})
 Set min macroblock lagrange factor (VBR).
@item mblmax @var{integer} (@emph{encoding,video})
 Set max macroblock lagrange factor (VBR).
@item mepc @var{integer} (@emph{encoding,video})
 Set motion estimation bitrate penalty compensation (1.0 = 256).
@item skip_loop_filter @var{integer} (@emph{decoding,video})
@item skip_idct        @var{integer} (@emph{decoding,video})
@item skip_frame       @var{integer} (@emph{decoding,video})
@@ -737,9 +967,6 @@ Discard all bidirectional frames.
@item nokey
 Discard all frames excepts keyframes.
@item nointra
 Discard all frames except I frames.
@item all
 Discard all frames.
@end table
@@ -749,17 +976,38 @@ Default value is @samp{default}.
@item bidir_refine @var{integer} (@emph{encoding,video})
 Refine the two motion vectors used in bidirectional macroblocks.
@item brd_scale @var{integer} (@emph{encoding,video})
 Downscale frames for dynamic B-frame decision.
@item keyint_min @var{integer} (@emph{encoding,video})
 Set minimum interval between IDR-frames.
@item refs @var{integer} (@emph{encoding,video})
 Set reference frames to consider for motion compensation.
@item chromaoffset @var{integer} (@emph{encoding,video})
 Set chroma qp offset from luma.
@item trellis @var{integer} (@emph{encoding,audio,video})
 Set rate-distortion optimal quantization.
@item 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.
@item compression_level @var{integer} (@emph{encoding,audio,video})
@item min_prediction_order @var{integer} (@emph{encoding,audio})
@item max_prediction_order @var{integer} (@emph{encoding,audio})
@item timecode_frame_start @var{integer} (@emph{encoding,video})
 Set GOP timecode frame start number, in non drop frame format.
@item request_channels @var{integer} (@emph{decoding,audio})
 Set desired number of audio channels.
@item bits_per_raw_sample @var{integer}
@item channel_layout @var{integer} (@emph{decoding/encoding,audio})
@@ -774,142 +1022,11 @@ Possible values:
@item rc_max_vbv_use @var{float} (@emph{encoding,video})
@item rc_min_vbv_use @var{float} (@emph{encoding,video})
@item ticks_per_frame @var{integer} (@emph{decoding/encoding,audio,video})
@item color_primaries @var{integer} (@emph{decoding/encoding,video})
 Possible values:
@table @samp
@item bt709
 BT.709
@item bt470m
 BT.470 M
@item bt470bg
 BT.470 BG
@item smpte170m
 SMPTE 170 M
@item smpte240m
 SMPTE 240 M
@item film
 Film
@item bt2020
 BT.2020
@item smpte428
@item smpte428_1
 SMPTE ST 428-1
@item smpte431
 SMPTE 431-2
@item smpte432
 SMPTE 432-1
@item jedec-p22
 JEDEC P22
@end table
@item color_trc @var{integer} (@emph{decoding/encoding,video})
 Possible values:
@table @samp
@item bt709
 BT.709
@item gamma22
 BT.470 M
@item gamma28
 BT.470 BG
@item smpte170m
 SMPTE 170 M
@item smpte240m
 SMPTE 240 M
@item linear
 Linear
@item log
@item log100
 Log
@item log_sqrt
@item log316
 Log square root
@item iec61966_2_4
@item iec61966-2-4
 IEC 61966-2-4
@item bt1361
@item bt1361e
 BT.1361
@item iec61966_2_1
@item iec61966-2-1
 IEC 61966-2-1
@item bt2020_10
@item bt2020_10bit
 BT.2020 - 10 bit
@item bt2020_12
@item bt2020_12bit
 BT.2020 - 12 bit
@item smpte2084
 SMPTE ST 2084
@item smpte428
@item smpte428_1
 SMPTE ST 428-1
@item arib-std-b67
 ARIB STD-B67
@end table
@item colorspace @var{integer} (@emph{decoding/encoding,video})
 Possible values:
@table @samp
@item rgb
 RGB
@item bt709
 BT.709
@item fcc
 FCC
@item bt470bg
 BT.470 BG
@item smpte170m
 SMPTE 170 M
@item smpte240m
 SMPTE 240 M
@item ycocg
 YCOCG
@item bt2020nc
@item bt2020_ncl
 BT.2020 NCL
@item bt2020c
@item bt2020_cl
 BT.2020 CL
@item smpte2085
 SMPTE 2085
@item chroma-derived-nc
 Chroma-derived NCL
@item chroma-derived-c
 Chroma-derived CL
@item ictcp
 ICtCp
@end table
@item color_range @var{integer} (@emph{decoding/encoding,video})
 If used as input parameter, it serves as a hint to the decoder, which
 color_range the input has.
 Possible values:
@table @samp
@item tv
@item mpeg
 MPEG (219*2^(n-8))
@item pc
@item jpeg
 JPEG (2^n-1)
@end table
@item chroma_sample_location @var{integer} (@emph{decoding/encoding,video})
 Possible values:
@table @samp
@item left
@item center
@item topleft
@item top
@item bottomleft
@item bottom
@end table
@item log_level_offset @var{integer}
 Set the log level offset.
@@ -988,37 +1105,10 @@ Interlaced video, top coded first, bottom displayed first
 Interlaced video, bottom coded first, top displayed first
@end table
-@item skip_alpha @var{bool} (@emph{decoding,video})
+@item skip_alpha @var{integer} (@emph{decoding,video})
 Set to 1 to disable processing alpha (transparency). This works like the
@samp{gray} flag in the @option{flags} option which skips chroma information
 instead of alpha. Default is 0.
@item codec_whitelist @var{list} (@emph{input})
 "," separated list of allowed decoders. By default all are allowed.
@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:
@example
 ffprobe -dump_separator "
                          "  -i ~/videos/matrixbench_mpeg2.mpg
@end example
@item max_pixels @var{integer} (@emph{decoding/encoding,video})
 Maximum number of pixels per image. This value can be used to avoid out of
 memory failures due to large images.
@item apply_cropping @var{bool} (@emph{decoding,video})
 Enable cropping if cropping parameters are multiples of the required
 alignment for the left and top parameters. If the alignment is not met the
 cropping will be partially applied to maintain alignment.
 Default is 1 (enabled).
 Note: The required alignment depends on if @code{AV_CODEC_FLAG_UNALIGNED} is set and the
 CPU. @code{AV_CODEC_FLAG_UNALIGNED} cannot be changed from the command line. Also hardware
 decoders will not apply left/top Cropping.
@end table
@c man end CODEC OPTIONS
--- a/doc/decoders.texi
+++ b/doc/decoders.texi
@@ -25,19 +25,6 @@ enabled decoders.
 A description of some of the currently available video decoders
 follows.
@section av1
 AOMedia Video 1 (AV1) decoder.
@subsection Options
@table @option
@item operating_point
 Select an operating point of a scalable AV1 bitstream (0 - 31). Default is 0.
@end table
@section rawvideo
 Raw video decoder.
@@ -60,129 +47,6 @@ top-field-first is assumed
@end table
@section libdav1d
 dav1d AV1 decoder.
 libdav1d allows libavcodec to decode the AOMedia Video 1 (AV1) codec.
 Requires the presence of the libdav1d headers and library during configuration.
 You need to explicitly configure the build with @code{--enable-libdav1d}.
@subsection Options
 The following options are supported by the libdav1d wrapper.
@table @option
@item framethreads
 Set amount of frame threads to use during decoding. The default value is 0 (autodetect).
 This option is deprecated for libdav1d >= 1.0 and will be removed in the future. Use the
 global option @code{threads} instead.
@item tilethreads
 Set amount of tile threads to use during decoding. The default value is 0 (autodetect).
 This option is deprecated for libdav1d >= 1.0 and will be removed in the future. Use the
 global option @code{threads} instead.
@item filmgrain
 Apply film grain to the decoded video if present in the bitstream. Defaults to the
 internal default of the library.
 This option is deprecated and will be removed in the future. See the global option
@code{export_side_data} to export Film Grain parameters instead of applying it.
@item oppoint
 Select an operating point of a scalable AV1 bitstream (0 - 31). Defaults to the
 internal default of the library.
@item alllayers
 Output all spatial layers of a scalable AV1 bitstream. The default value is false.
@end table
@section libdavs2
 AVS2-P2/IEEE1857.4 video decoder wrapper.
 This decoder allows libavcodec to decode AVS2 streams with davs2 library.
@c man end VIDEO DECODERS
@section libuavs3d
 AVS3-P2/IEEE1857.10 video decoder.
 libuavs3d allows libavcodec to decode AVS3 streams.
 Requires the presence of the libuavs3d headers and library during configuration.
 You need to explicitly configure the build with @code{--enable-libuavs3d}.
@subsection Options
 The following option is supported by the libuavs3d wrapper.
@table @option
@item frame_threads
 Set amount of frame threads to use during decoding. The default value is 0 (autodetect).
@end table
@section QSV Decoders
 The family of Intel QuickSync Video decoders (VC1, MPEG-2, H.264, HEVC,
 JPEG/MJPEG, VP8, VP9, AV1).
@subsection Common Options
 The following options are supported by all qsv decoders.
@table @option
@item @var{async_depth}
 Internal parallelization depth, the higher the value the higher the latency.
@item @var{gpu_copy}
 A GPU-accelerated copy between video and system memory
@table @samp
@item default
@item on
@item off
@end table
@end table
@subsection HEVC Options
 Extra options for hevc_qsv.
@table @option
@item @var{load_plugin}
 A user plugin to load in an internal session
@table @samp
@item none
@item hevc_sw
@item hevc_hw
@end table
@item @var{load_plugins}
 A :-separate list of hexadecimal plugin UIDs to load in an internal session
@end table
@section v210
 Uncompressed 4:2:2 10-bit decoder.
@subsection Options
@table @option
@item custom_stride
 Set the line size of the v210 data in bytes. The default value is 0
 (autodetect). You can use the special -1 value for a strideless v210 as seen in
 BOXX files.
@end table
@c man end VIDEO DECODERS
@chapter Audio Decoders
@@ -204,7 +68,7 @@ the undocumented RealAudio 3 (a.k.a. dnet).
@item -drc_scale @var{value}
 Dynamic Range Scale Factor. The factor to apply to dynamic range values
-from the AC-3 stream. This factor is applied exponentially. The default value is 1.
+from the AC-3 stream. This factor is applied exponentially.
 There are 3 notable scale factor ranges:
@table @option
@item drc_scale == 0
@@ -219,26 +83,9 @@ Loud sounds are fully compressed.  Soft sounds are enhanced.
@end table
@section flac
 FLAC audio decoder.
 This decoder aims to implement the complete FLAC specification from Xiph.
@subsection FLAC Decoder options
@table @option
@item -use_buggy_lpc
 The lavc FLAC encoder used to produce buggy streams with high lpc values
 (like the default value). This option makes it possible to decode such streams
 correctly by using lavc's old buggy lpc logic for decoding.
@end table
@section ffwavesynth
-Internal wave synthesizer.
+Internal wave synthetizer.
 This decoder generates wave patterns according to predefined sequences. Its
 use is purely internal and the format of the data it accepts is not publicly
@@ -316,60 +163,11 @@ Requires the presence of the libopus headers and library during
 configuration. You need to explicitly configure the build with
@code{--enable-libopus}.
 An FFmpeg native decoder for Opus exists, so users can decode Opus
 without this library.
@c man end AUDIO DECODERS
@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
@table @option
@item compute_clut
@table @option
@item -2
 Compute clut once if no matching CLUT is in the stream.
@item -1
 Compute clut if no matching CLUT is in the stream.
@item 0
 Never compute CLUT
@item 1
 Always compute CLUT and override the one provided in the stream.
@end table
@item dvb_substream
 Selects the dvb substream, or all substreams if -1 which is default.
@end table
@section dvdsub
 This codec decodes the bitmap subtitles used in DVDs; the same subtitles can
@@ -386,18 +184,9 @@ palette is stored in the IFO file, and therefore not available when reading
 from dumped VOB files.
 The format for this option is a string containing 16 24-bits hexadecimal
-numbers (without 0x prefix) separated by commas, for example @code{0d00ee,
+numbers (without 0x prefix) separated by comas, for example @code{0d00ee,
 ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1,
 7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b}.
@item ifo_palette
 Specify the IFO file from which the global palette is obtained.
 (experimental)
@item forced_subs_only
 Only decode subtitle entries marked as forced. Some titles have forced
 and non-forced subtitles in the same track. Setting this flag to @code{1}
 will only keep the forced subtitles. Default value is @code{0}.
@end table
@section libzvbi-teletext
@@ -411,30 +200,18 @@ 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 of teletext page numbers to decode. You may use the special * string to
-list are dropped. You may use the special @code{*} string to match all pages,
+match all pages. Pages that do not match the specified list are dropped.
 or @code{subtitle} to match all subtitle pages.
 Default value is *.
@item txt_default_region
 Set default character set used for decoding, a value between 0 and 87 (see
 ETS 300 706, Section 15, Table 32). Default value is -1, which does not
 override the libzvbi default. This option is needed for some legacy level 1.0
 transmissions which cannot signal the proper charset.
@item txt_chop_top
 Discards the top teletext line. Default value is 1.
@item txt_format
-Specifies the format of the decoded subtitles.
+Specifies the format of the decoded subtitles. The teletext decoder is capable
-@table @option
+of decoding the teletext pages to bitmaps or to simple text, you should use
-@item bitmap
+"bitmap" for teletext pages, because certain graphics and colors cannot be
-The default format, you should use this for teletext pages, because certain
+expressed in simple text. You might use "text" for teletext based subtitles if
-graphics and colors cannot be expressed in simple text or even ASS.
+your application can handle simple text based subtitles. Default value is
-@item text
+bitmap.
 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
@item txt_left
 X offset of generated bitmaps, default is 0.
@item txt_top
@@ -443,21 +220,14 @@ Y offset of generated bitmaps, default is 0.
 Chops leading and trailing spaces and removes empty lines from the generated
 text. This option is useful for teletext based subtitles where empty spaces may
 be present at the start or at the end of the lines or empty lines may be
-present between the subtitle lines because of double-sized teletext characters.
+present between the subtitle lines because of double-sized teletext charactes.
 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
+miliseconds. Default value is 30000 which is 30 seconds.
 subtitle event comes.
@item txt_transparent
 Force transparent background of the generated teletext bitmaps. Default value
-is 0 which means an opaque background.
+is 0 which means an opaque (black) background.
@item txt_opacity
 Sets the opacity (0-255) of the teletext background. If
@option{txt_transparent} is not set, it only affects characters between a start
 box and an end box, typically subtitles. Default value is 0 if
@option{txt_transparent} is set, 255 otherwise.
@end table
@c man end SUBTILES DECODERS
--- a/doc/demuxers.texi
+++ b/doc/demuxers.texi
@@ -13,47 +13,21 @@ You can disable all the demuxers using the configure option
 the option @code{--enable-demuxer=@var{DEMUXER}}, or disable it
 with the option @code{--disable-demuxer=@var{DEMUXER}}.
-The option @code{-demuxers} of the ff* tools will display the list of
+The option @code{-formats} of the ff* tools will display the list of
-enabled demuxers. Use @code{-formats} to view a combined list of
+enabled demuxers.
 enabled demuxers and muxers.
 The description of some of the currently available demuxers follows.
-@section aa
+@section applehttp
-Audible Format 2, 3, and 4 demuxer.
+Apple HTTP Live Streaming demuxer.
-This demuxer is used to demux Audible Format 2, 3, and 4 (.aa) files.
+This demuxer presents all AVStreams from all variant streams.
-
+The id field is set to the bitrate variant index number. By setting
-@section aac
+the discard flags on AVStreams (by pressing 'a' or 'v' in ffplay),
-
+the caller can decide which variant streams to actually receive.
-Raw Audio Data Transport Stream AAC demuxer.
+The total bitrate of the variant that the stream belongs to is
-
+available in a metadata key named "variant_bitrate".
 This demuxer is used to demux an ADTS input containing a single AAC stream
 alongwith any ID3v1/2 or APE tags in it.
@section apng
 Animated Portable Network Graphics demuxer.
 This demuxer is used to demux APNG files.
 All headers, but the PNG signature, up to (but not including) the first
 fcTL chunk are transmitted as extradata.
 Frames are then split as being all the chunks between two fcTL ones, or
 between the last fcTL and IEND chunks.
@table @option
@item -ignore_loop @var{bool}
 Ignore the loop variable in the file if set. Default is enabled.
@item -max_fps @var{int}
 Maximum framerate in frames per second. Default of 0 imposes no limit.
@item -default_fps @var{int}
 Default framerate in frames per second when none is specified in the file
 (0 meaning as fast as possible). Default is 15.
@end table
@section asf
@@ -72,7 +46,7 @@ Do not try to resynchronize by looking for a certain optional start code.
 Virtual concatenation script demuxer.
 This demuxer reads a list of files and other directives from a text file and
-demuxes them one after the other, as if all their packets had been muxed
+demuxes them one after the other, as if all their packet had been muxed
 together.
 The timestamps in the files are adjusted so that the first file starts at 0
@@ -103,10 +77,11 @@ backslash or single quotes.
 All subsequent file-related directives apply to that file.
@item @code{ffconcat version 1.0}
-Identify the script type and version.
+Identify the script type and version. It also sets the @option{safe} option
 to 1 if it was to its default -1.
 To make FFmpeg recognize the format automatically, this directive must
-appear exactly as is (no extra space or byte-order-mark) on the very first
+appears exactly as is (no extra space or byte-order-mark) on the very first
 line of the script.
@item @code{duration @var{dur}}
@@ -117,57 +92,6 @@ file is not available or accurate.
 If the duration is set for all files, then it is possible to seek in the
 whole concatenated video.
@item @code{inpoint @var{timestamp}}
 In point of the file. When the demuxer opens the file it instantly seeks to the
 specified timestamp. Seeking is done so that all streams can be presented
 successfully at In point.
 This directive works best with intra frame codecs, because for non-intra frame
 ones you will usually get extra packets before the actual In point and the
 decoded content will most likely contain frames before In point too.
 For each file, packets before the file In point will have timestamps less than
 the calculated start timestamp of the file (negative in case of the first
 file), and the duration of the files (if not specified by the @code{duration}
 directive) will be reduced based on their specified In point.
 Because of potential packets before the specified In point, packet timestamps
 may overlap between two concatenated files.
@item @code{outpoint @var{timestamp}}
 Out point of the file. When the demuxer reaches the specified decoding
 timestamp in any of the streams, it handles it as an end of file condition and
 skips the current and all the remaining packets from all streams.
 Out point is exclusive, which means that the demuxer will not output packets
 with a decoding timestamp greater or equal to Out point.
 This directive works best with intra frame codecs and formats where all streams
 are tightly interleaved. For non-intra frame codecs you will usually get
 additional packets with presentation timestamp after Out point therefore the
 decoded content will most likely contain frames after Out point too. If your
 streams are not tightly interleaved you may not get all the packets from all
 streams before Out point and you may only will be able to decode the earliest
 stream until Out point.
 The duration of the files (if not specified by the @code{duration}
 directive) will be reduced based on their specified Out point.
@item @code{file_packet_metadata @var{key=value}}
 Metadata of the packets of the file. The specified metadata will be set for
 each file packet. You can specify this directive multiple times to add multiple
 metadata entries.
 This directive is deprecated, use @code{file_packet_meta} instead.
@item @code{file_packet_meta @var{key} @var{value}}
 Metadata of the packets of the file. The specified metadata will be set for
 each file packet. You can specify this directive multiple times to add multiple
 metadata entries.
@item @code{option @var{key} @var{value}}
 Option to access, open and probe the file.
 Can be present multiple times.
@item @code{stream}
 Introduce a stream in the virtual file.
 All subsequent stream-related directives apply to the last introduced
@@ -184,20 +108,6 @@ subfiles will be used.
 This is especially useful for MPEG-PS (VOB) files, where the order of the
 streams is not reliable.
@item @code{stream_meta @var{key} @var{value}}
 Metadata for the stream.
 Can be present multiple times.
@item @code{stream_codec @var{value}}
 Codec for the stream.
@item @code{stream_extradata @var{hex_string}}
 Extradata for the string, encoded in hexadecimal.
@item @code{chapter @var{id} @var{start} @var{end}}
 Add a chapter. @var{id} is an unique identifier, possibly small and
 consecutive.
@end table
@subsection Options
@@ -207,8 +117,7 @@ This demuxer accepts the following option:
@table @option
@item safe
-If set to 1, reject unsafe file paths and directives.
+If set to 1, reject unsafe file paths. A file path is considered safe if it
 A file path is considered safe if it
 does not contain a protocol specification and is relative and all components
 only contain characters from the portable character set (letters, digits,
 period, underscore and hyphen) and have no period at the beginning of a
@@ -216,192 +125,54 @@ component.
 If set to 0, any file name is accepted.
-The default is 1.
+The default is -1, it is equivalent to 1 if the format was automatically
 probed and 0 otherwise.
@item auto_convert
 If set to 1, try to perform automatic conversions on packet data to make the
 streams concatenable.
 The default is 1.
 Currently, the only conversion is adding the h264_mp4toannexb bitstream
 filter to H.264 streams in MP4 format. This is necessary in particular if
 there are resolution changes.
@item segment_time_metadata
 If set to 1, every packet will contain the @var{lavf.concat.start_time} and the
@var{lavf.concat.duration} packet metadata values which are the start_time and
 the duration of the respective file segments in the concatenated output
 expressed in microseconds. The duration metadata is only set if it is known
 based on the concat file.
 The default is 0.
@end table
-@subsection Examples
+@section flv
@itemize
@item
 Use absolute filenames and include some comments:
@example
 # my first filename
 file /mnt/share/file-1.wav
 # my second filename including whitespace
 file '/mnt/share/file 2.wav'
 # my third filename including whitespace plus single quote
 file '/mnt/share/file 3'\''.wav'
@end example
@item
 Allow for input format auto-probing, use safe filenames and set the duration of
 the first file:
@example
 ffconcat version 1.0
 file file-1.wav
 duration 20.0
 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.
@subsection Options
 This demuxer accepts the following option:
@table @option
@item cenc_decryption_key
 16-byte key, in hex, to decrypt files encrypted using ISO Common Encryption (CENC/AES-128 CTR; ISO/IEC 23001-7).
@end table
@section imf
 Interoperable Master Format demuxer.
 This demuxer presents audio and video streams found in an IMF Composition.
@section flv, live_flv, kux
 Adobe Flash Video Format demuxer.
-This demuxer is used to demux FLV files and RTMP network streams. In case of live network streams, if you force format, you may use live_flv option instead of flv to survive timestamp discontinuities.
+This demuxer is used to demux FLV files and RTMP network streams.
 KUX is a flv variant used on the Youku platform.
@example
 ffmpeg -f flv -i myfile.flv ...
 ffmpeg -f live_flv -i rtmp://<any.server>/anything/key ....
@end example
@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
+@section libgme
-Animated GIF demuxer.
+The Game Music Emu library is a collection of video game music file emulators.
-It accepts the following options:
+See @url{http://code.google.com/p/game-music-emu/} for more information.
-@table @option
+Some files have multiple tracks. The demuxer will pick the first track by
-@item min_delay
+default. The @option{track_index} option can be used to select a different
-Set the minimum valid delay between frames in hundredths of seconds.
+track. Track indexes start at 0. The demuxer exports the number of tracks as
-Range is 0 to 6000. Default value is 2.
+@var{tracks} meta data entry.
-@item max_gif_delay
+For very large files, the @option{max_size} option may have to be adjusted.
 Set the maximum valid delay between frames in hundredth of seconds.
 Range is 0 to 65535. Default value is 65535 (nearly eleven minutes),
 the maximum value allowed by the specification.
-@item default_delay
+@section libquvi
 Set the default delay between frames in hundredths of seconds.
 Range is 0 to 6000. Default value is 10.
-@item ignore_loop
+Play media from Internet services using the quvi project.
 GIF files can contain information to loop a certain number of times (or
 infinitely). If @option{ignore_loop} is set to 1, then the loop setting
 from the input will be ignored and looping will not occur. If set to 0,
 then looping will occur and will cycle the number of times according to
 the GIF. Default value is 1.
@end table
-For example, with the overlay filter, place an infinitely looping GIF
+The demuxer accepts a @option{format} option to request a specific quality. It
-over another video:
+is by default set to @var{best}.
@example
 ffmpeg -i input.mp4 -ignore_loop 0 -i input.gif -filter_complex overlay=shortest=1 out.mkv
@end example
-Note that in the above example the shortest option for overlay filter is
+See @url{http://quvi.sourceforge.net/} for more information.
 used to end the output video at the length of the shortest input file,
 which in this case is @file{input.mp4} as the GIF in this example loops
 infinitely.
-@section hls
+FFmpeg needs to be built with @code{--enable-libquvi} for this demuxer to be
-
+enabled.
 HLS demuxer
 Apple HTTP Live Streaming demuxer.
 This demuxer presents all AVStreams from all variant streams.
 The id field is set to the bitrate variant index number. By setting
 the discard flags on AVStreams (by pressing 'a' or 'v' in ffplay),
 the caller can decide which variant streams to actually receive.
 The total bitrate of the variant that the stream belongs to is
 available in a metadata key named "variant_bitrate".
 It accepts the following options:
@table @option
@item live_start_index
 segment index to start live streams at (negative values are from the end).
@item prefer_x_start
 prefer to use #EXT-X-START if it's in playlist instead of live_start_index.
@item allowed_extensions
 ',' separated list of file extensions that hls is allowed to access.
@item max_reload
 Maximum number of times a insufficient list is attempted to be reloaded.
 Default value is 1000.
@item m3u8_hold_counters
 The maximum number of times to load m3u8 when it refreshes without new segments.
 Default value is 1000.
@item http_persistent
 Use persistent HTTP connections. Applicable only for HTTP streams.
 Enabled by default.
@item http_multiple
 Use multiple HTTP connections for downloading HTTP segments.
 Enabled by default for HTTP/1.1 servers.
@item http_seekable
 Use HTTP partial requests for downloading HTTP segments.
 0 = disable, 1 = enable, -1 = auto, Default is auto.
@item seg_format_options
 Set options for the demuxer of media segments using a list of key=value pairs separated by @code{:}.
@end table
@section image2
@@ -428,10 +199,6 @@ Select the pattern type used to interpret the provided filename.
@var{pattern_type} accepts one of the following values.
@table @option
@item none
 Disable pattern matching, therefore the video will only contain the specified
 image. You should use this option if you do not want to create sequences from
 multiple images and your filenames may contain special pattern characters.
@item sequence
 Select a sequence pattern type, used to specify a sequence of files
 indexed by sequential numbers.
@@ -511,17 +278,6 @@ nanosecond precision.
@item video_size
 Set the video size of the images to read. If not specified the video
 size is guessed from the first image file in the sequence.
@item export_path_metadata
 If set to 1, will add two extra fields to the metadata found in input, making them
 also available for other filters (see @var{drawtext} filter for examples). Default
 value is 0. The extra fields are described below:
@table @option
@item lavf.image2dec.source_path
 Corresponds to the full path to the input file being read.
@item lavf.image2dec.source_basename
 Corresponds to the name of the file being read.
@end table
@end table
@subsection Examples
@@ -549,265 +305,17 @@ ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv
@end example
@end itemize
@section libgme
 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.
 It accepts the following options:
@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
@section libopenmpt
 libopenmpt based module demuxer
 See @url{https://lib.openmpt.org/libopenmpt/} for more information.
 Some files have multiple subsongs (tracks) this can be set with the @option{subsong}
 option.
 It accepts the following options:
@table @option
@item subsong
 Set the subsong index. This can be either  'all', 'auto', or the index of the
 subsong. Subsong indexes start at 0. The default is 'auto'.
 The default value is to let libopenmpt choose.
@item layout
 Set the channel layout. Valid values are 1, 2, and 4 channel layouts.
 The default value is STEREO.
@item sample_rate
 Set the sample rate for libopenmpt to output.
 Range is from 1000 to INT_MAX. The value default is 48000.
@end table
@section mov/mp4/3gp
 Demuxer for Quicktime File Format & ISO/IEC Base Media File Format (ISO/IEC 14496-12 or MPEG-4 Part 12, ISO/IEC 15444-12 or JPEG 2000 Part 12).
 Registered extensions: mov, mp4, m4a, 3gp, 3g2, mj2, psp, m4b, ism, ismv, isma, f4v
@subsection Options
 This demuxer accepts the following options:
@table @option
@item enable_drefs
 Enable loading of external tracks, disabled by default.
 Enabling this can theoretically leak information in some use cases.
@item use_absolute_path
 Allows loading of external tracks via absolute paths, disabled by default.
 Enabling this poses a security risk. It should only be enabled if the source
 is known to be non-malicious.
@item seek_streams_individually
 When seeking, identify the closest point in each stream individually and demux packets in
 that stream from identified point. This can lead to a different sequence of packets compared
 to demuxing linearly from the beginning. Default is true.
@item ignore_editlist
 Ignore any edit list atoms. The demuxer, by default, modifies the stream index to reflect the
 timeline described by the edit list. Default is false.
@item advanced_editlist
 Modify the stream index to reflect the timeline described by the edit list. @code{ignore_editlist}
 must be set to false for this option to be effective.
 If both @code{ignore_editlist} and this option are set to false, then only the
 start of the stream index is modified to reflect initial dwell time or starting timestamp
 described by the edit list. Default is true.
@item ignore_chapters
 Don't parse chapters. This includes GoPro 'HiLight' tags/moments. Note that chapters are
 only parsed when input is seekable. Default is false.
@item use_mfra_for
 For seekable fragmented input, set fragment's starting timestamp from media fragment random access box, if present.
 Following options are available:
@table @samp
@item auto
 Auto-detect whether to set mfra timestamps as PTS or DTS @emph{(default)}
@item dts
 Set mfra timestamps as DTS
@item pts
 Set mfra timestamps as PTS
@item 0
 Don't use mfra box to set timestamps
@end table
@item use_tfdt
 For fragmented input, set fragment's starting timestamp to @code{baseMediaDecodeTime} from the @code{tfdt} box.
 Default is enabled, which will prefer to use the @code{tfdt} box to set DTS. Disable to use the @code{earliest_presentation_time} from the @code{sidx} box.
 In either case, the timestamp from the @code{mfra} box will be used if it's available and @code{use_mfra_for} is
 set to pts or dts.
@item export_all
 Export unrecognized boxes within the @var{udta} box as metadata entries. The first four
 characters of the box type are set as the key. Default is false.
@item export_xmp
 Export entire contents of @var{XMP_} box and @var{uuid} box as a string with key @code{xmp}. Note that
 if @code{export_all} is set and this option isn't, the contents of @var{XMP_} box are still exported
 but with key @code{XMP_}. Default is false.
@item activation_bytes
 4-byte key required to decrypt Audible AAX and AAX+ files. See Audible AAX subsection below.
@item audible_fixed_key
 Fixed key used for handling Audible AAX/AAX+ files. It has been pre-set so should not be necessary to
 specify.
@item decryption_key
 16-byte key, in hex, to decrypt files encrypted using ISO Common Encryption (CENC/AES-128 CTR; ISO/IEC 23001-7).
@item max_stts_delta
 Very high sample deltas written in a trak's stts box may occasionally be intended but usually they are written in
 error or used to store a negative value for dts correction when treated as signed 32-bit integers. This option lets
 the user set an upper limit, beyond which the delta is clamped to 1. Values greater than the limit if negative when
 cast to int32 are used to adjust onward dts.
 Unit is the track time scale. Range is 0 to UINT_MAX. Default is @code{UINT_MAX - 48000*10} which allows upto
 a 10 second dts correction for 48 kHz audio streams while accommodating 99.9% of @code{uint32} range.
@end table
@subsection Audible AAX
 Audible AAX files are encrypted M4B files, and they can be decrypted by specifying a 4 byte activation secret.
@example
 ffmpeg -activation_bytes 1CEB00DA -i test.aax -vn -c:a copy output.mp4
@end example
@section mpegts
 MPEG-2 transport stream demuxer.
 This demuxer accepts the following options:
@table @option
@item resync_size
 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
+Overrides 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
 not discarded. Default value is 1, set this option to 0 if you want your
 teletext packet PTS and DTS values untouched.
@item ts_packetsize
 Output option carrying the raw packet size in bytes.
 Show the detected raw packet size, cannot be set by the user.
@item scan_all_pmts
 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.
@item max_packet_size
 Set maximum size, in bytes, of packet emitted by the demuxer. Payloads above this size
 are split across multiple packets. Range is 1 to INT_MAX/2. Default is 204800 bytes.
@end table
@section mpjpeg
 MJPEG encapsulated in multi-part MIME demuxer.
 This demuxer allows reading of MJPEG, where each frame is represented as a part of
 multipart/x-mixed-replace stream.
@table @option
@item strict_mime_boundary
 Default implementation applies a relaxed standard to multi-part MIME boundary detection,
 to prevent regression with numerous existing endpoints not generating a proper MIME
 MJPEG stream. Turning this option on by setting it to 1 will result in a stricter check
 of the boundary value.
@end table
@section rawvideo
@@ -890,20 +398,4 @@ Example: convert the captions to a format most players understand:
 ffmpeg -i http://www.ted.com/talks/subtitles/id/1/lang/en talk1-en.srt
@end example
@section vapoursynth
 Vapoursynth wrapper.
 Due to security concerns, Vapoursynth scripts will not
 be autodetected so the input format has to be forced. For ff* CLI tools,
 add @code{-f vapoursynth} before the input @code{-i yourscript.vpy}.
 This demuxer accepts the following option:
@table @option
@item max_script_size
 The demuxer buffers the entire script into memory. Adjust this value to set the maximum buffer size,
 which in turn, acts as a ceiling for the size of scripts that can be read.
 Default is 1 MiB.
@end table
@c man end DEMUXERS
--- a/doc/dev_community/community.md
+++ b/doc/dev_community/community.md
@@ -1,79 +0,0 @@
 # FFmpeg project
 ## Organisation
 The FFmpeg project is organized through a community working on global consensus.
 Decisions are taken by the ensemble of active members, through voting and
 are aided by two committees.
 ## General Assembly
 The ensemble of active members is called the General Assembly (GA).
 The General Assembly is sovereign and legitimate for all its decisions
 regarding the FFmpeg project.
 The General Assembly is made up of active contributors.
 Contributors are considered "active contributors" if they have pushed more
 than 20 patches in the last 36 months in the main FFmpeg repository, or
 if they have been voted in by the GA.
 Additional members are added to the General Assembly through a vote after
 proposal by a member of the General Assembly.
 They are part of the GA for two years, after which they need a confirmation by
 the GA.
 ## Voting
 Voting is done using a ranked voting system, currently running on https://vote.ffmpeg.org/ .
 Majority vote means more than 50% of the expressed ballots.
 ## Technical Committee
 The Technical Committee (TC) is here to arbitrate and make decisions when
 technical conflicts occur in the project.
 They will consider the merits of all the positions, judge them and make a
 decision.
 The TC resolves technical conflicts but is not a technical steering committee.
 Decisions by the TC are binding for all the contributors.
 Decisions made by the TC can be re-opened after 1 year or by a majority vote
 of the General Assembly, requested by one of the member of the GA.
 The TC is elected by the General Assembly for a duration of 1 year, and
 is composed of 5 members.
 Members can be re-elected if they wish. A majority vote in the General Assembly
 can trigger a new election of the TC.
 The members of the TC can be elected from outside of the GA.
 Candidates for election can either be suggested or self-nominated.
 The conflict resolution process is detailed in the [resolution process](resolution_process.md) document.
 ## Community committee
 The Community Committee (CC) is here to arbitrage and make decisions when
 inter-personal conflicts occur in the project. It will decide quickly and
 take actions, for the sake of the project.
 The CC can remove privileges of offending members, including removal of
 commit access and temporary ban from the community.
 Decisions made by the CC can be re-opened after 1 year or by a majority vote
 of the General Assembly. Indefinite bans from the community must be confirmed
 by the General Assembly, in a majority vote.
 The CC is elected by the General Assembly for a duration of 1 year, and is
 composed of 5 members.
 Members can be re-elected if they wish. A majority vote in the General Assembly
 can trigger a new election of the CC.
 The members of the CC can be elected from outside of the GA.
 Candidates for election can either be suggested or self-nominated.
 The CC is governed by and responsible for enforcing the Code of Conduct.
--- a/doc/dev_community/resolution_process.md
+++ b/doc/dev_community/resolution_process.md
@@ -1,91 +0,0 @@
 # Technical Committee
 _This document only makes sense with the rules from [the community document](community)_.
 The Technical Committee (**TC**) is here to arbitrate and make decisions when
 technical conflicts occur in the project.
 The TC main role is to resolve technical conflicts.
 It is therefore not a technical steering committee, but it is understood that
 some decisions might impact the future of the project.
 # Process
 ## Seizing
 The TC can take possession of any technical matter that it sees fit.
 To involve the TC in a matter, email tc@ or CC them on an ongoing discussion.
 As members of TC are developers, they also can email tc@ to raise an issue.
 ## Announcement
 The TC, once seized, must announce itself on the main mailing list, with a _[TC]_ tag.
 The TC has 2 modes of operation: a RFC one and an internal one.
 If the TC thinks it needs the input from the larger community, the TC can call
 for a RFC. Else, it can decide by itself.
 If the disagreement involves a member of the TC, that member should recuse
 themselves from the decision.
 The decision to use a RFC process or an internal discussion is a discretionary
 decision of the TC.
 The TC can also reject a seizure for a few reasons such as:
 the matter was not discussed enough previously; it lacks expertise to reach a
 beneficial decision on the matter; or the matter is too trivial.
 ### RFC call
 In the RFC mode, one person from the TC posts on the mailing list the
 technical question and will request input from the community.
 The mail will have the following specification:
 * a precise title
 * a specific tag [TC RFC]
 * a top-level email
 * contain a precise question that does not exceed 100 words and that is answerable by developers
 * may have an extra description, or a link to a previous discussion, if deemed necessary,
 * contain a precise end date for the answers.
 The answers from the community must be on the main mailing list and must have
 the following specification:
 * keep the tag and the title unchanged
 * limited to 400 words
 * a first-level, answering directly to the main email
 * answering to the question.
 Further replies to answers are permitted, as long as they conform to the
 community standards of politeness, they are limited to 100 words, and are not
 nested more than once. (max-depth=2)
 After the end-date, mails on the thread will be ignored.
 Violations of those rules will be escalated through the Community Committee.
 After all the emails are in, the TC has 96 hours to give its final decision.
 Exceptionally, the TC can request an extra delay, that will be notified on the
 mailing list.
 ### Within TC
 In the internal case, the TC has 96 hours to give its final decision.
 Exceptionally, the TC can request an extra delay.
 ## Decisions
 The decisions from the TC will be sent on the mailing list, with the _[TC]_ tag.
 Internally, the TC should take decisions with a majority, or using
 ranked-choice voting.
 The decision from the TC should be published with a summary of the reasons that
 lead to this decision.
 The decisions from the TC are final, until the matters are reopened after
 no less than one year.
--- a/doc/developer.texi
+++ b/doc/developer.texi
@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle Developer Documentation
@titlepage
@@ -10,7 +9,9 @@
@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
@@ -26,15 +27,17 @@ this document.
 For more detailed legal information about the use of FFmpeg in
 external programs read the @file{LICENSE} file in the source tree and
-consult @url{https://ffmpeg.org/legal.html}.
+consult @url{http://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.
+      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 +46,9 @@ 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:
@@ -61,16 +64,13 @@ rejected by the git repository.
@item
 You should try to limit your code lines to 80 characters; however, do so if
 and only if this improves readability.
@item
 K&R coding style is used.
@end itemize
 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 +110,7 @@ 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:
@@ -123,21 +123,10 @@ the @samp{inline} keyword;
@samp{//} comments;
@item
-designated struct initializers (@samp{struct s x = @{ .i = 17 @};});
+designated struct initializers (@samp{struct s x = @{ .i = 17 @};})
@item
-compound literals (@samp{x = (struct s) @{ 17, 23 @};}).
+compound literals (@samp{x = (struct s) @{ 17, 23 @};})
@item
 for loops with variable definition (@samp{for (int i = 0; i < 8; i++)});
@item
 Variadic macros (@samp{#define ARRAY(nb, ...) (int[nb + 1])@{ nb, __VA_ARGS__ @}});
@item
 Implementation defined behavior for signed integers is assumed to match the
 expected behavior for two's complement. Non representable values in integer
 casts are binary truncated. Shift right of signed values uses sign extension.
@end itemize
 These features are supported by all compilers we care about, so we will not
@@ -162,11 +151,11 @@ 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
-for example structs and enums; they should always be in CamelCase.
+for example structs and enums; they should always be in the CamelCase
 There are the following conventions for naming variables and functions:
@@ -186,7 +175,7 @@ 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
@@ -206,7 +195,7 @@ 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
@@ -218,7 +207,7 @@ 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
@@ -238,7 +227,7 @@ autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@@<!$/
@end example
 For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
-@lisp
+@example
 (c-add-style "ffmpeg"
             '("k&r"
               (c-basic-offset . 4)
@@ -249,12 +238,12 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
               )
             )
 (setq c-default-style "ffmpeg")
-@end lisp
+@end example
-@chapter Development Policy
+@section Development Policy
-@section Patches/Committing
+@enumerate
-@subheading Licenses for patches must be compatible with FFmpeg.
+@item
 Contributions should be licensed under the
@uref{http://www.gnu.org/licenses/lgpl-2.1.html, LGPL 2.1},
 including an "or any later version" clause, or, if you prefer
@@ -267,15 +256,15 @@ preferred.
 If you add a new file, give it a proper license header. Do not copy and
 paste it from a random place, use an existing file as template.
-@subheading You must not commit code which breaks FFmpeg!
+@item
-This means unfinished code which is enabled and breaks compilation,
+You must not commit code which breaks FFmpeg! (Meaning unfinished but
-or compiles but does not work/breaks the regression tests. Code which
+enabled code which breaks compilation or compiles but does not work or
-is unfinished but disabled may be permitted under-circumstances, like
+breaks the regression tests)
-missing samples or an implementation with a small subset of features.
+You can commit unfinished stuff (for testing etc), but it must be disabled
-Always check the mailing list for any reviewers with issues and test
+(#ifdef etc) by default so it does not interfere with other developers'
-FATE before you push.
+work.
-@subheading Keep the main commit message short with an extended description below.
+@item
 The commit message should have a short first line in the form of
 a @samp{topic: short description} as a header, separated by a newline
 from the body consisting of an explanation of why the change is necessary.
@@ -283,24 +272,30 @@ If the commit fixes a known bug on the bug tracker, the commit message
 should include its bug ID. Referring to the issue on the bug tracker does
 not exempt you from writing an excerpt of the bug in the commit message.
-@subheading Testing must be adequate but not excessive.
+@item
-If it works for you, others, and passes FATE then it should be OK to commit
+You do not have to over-test things. If it works for you, and you think it
-it, provided it fits the other committing criteria. You should not worry about
+should work for others, then commit. If your code has problems
-over-testing things. If your code has problems (portability, triggers
+(portability, triggers compiler bugs, unusual environment etc) they will be
-compiler bugs, unusual environment etc) they will be reported and eventually
+reported and eventually fixed.
 fixed.
-@subheading Do not commit unrelated changes together.
+@item
-They should be split them into self-contained pieces. Also do not forget
+Do not commit unrelated changes together, split them into self-contained
-that if part B depends on part A, but A does not depend on B, then A can
+pieces. Also do not forget that if part B depends on part A, but A does not
-and should be committed first and separate from B. Keeping changes well
+depend on B, then A can and should be committed first and separate from B.
-split into self-contained parts makes reviewing and understanding them on
+Keeping changes well split into self-contained parts makes reviewing and
-the commit log mailing list easier. This also helps in case of debugging
+understanding them on the commit log mailing list easier. This also helps
-later on.
+in case of debugging later on.
 Also if you have doubts about splitting or not splitting, do not hesitate to
 ask/discuss it on the developer mailing list.
-@subheading Ask before you change the build system (configure, etc).
+@item
 Do not change behavior of the programs (renaming options etc) or public
 API or ABI without first discussing it on the ffmpeg-devel mailing list.
 Do not remove functionality from the code. Just improve!
 Note: Redundant code can be removed.
@item
 Do not commit changes to the build system (Makefiles, configure script)
 which change behavior, defaults etc, without asking first. The same
 applies to compiler warning fixes, trivial looking fixes and to code
@@ -309,7 +304,7 @@ the way we do. Send your changes as patches to the ffmpeg-devel mailing
 list, and if the code maintainers say OK, you may commit. This does not
 apply to files you wrote and/or maintain.
-@subheading Cosmetic changes should be kept in separate patches.
+@item
 We refuse source indentation and other cosmetic changes if they are mixed
 with functional changes, such commits will be rejected and removed. Every
 developer has his own indentation style, you should not change it. Of course
@@ -323,43 +318,56 @@ NOTE: If you had to put if()@{ .. @} over a large (> 5 lines) chunk of code,
 then either do NOT change the indentation of the inner part within (do not
 move it to the right)! or do so in a separate commit
-@subheading Commit messages should always be filled out properly.
+@item
 Always fill out the commit log message. Describe in a few lines what you
 changed and why. You can refer to mailing list postings if you fix a
 particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.
 Recommended format:
@example
 area changed: Short 1 line description
 details describing what and why and giving references.
@end example
-@subheading Credit the author of the patch.
+@item
 Make sure the author of the commit is set correctly. (see git commit --author)
 If you apply a patch, send an
 answer to ffmpeg-devel (or wherever you got the patch from) saying that
 you applied the patch.
-@subheading Complex patches should refer to discussion surrounding them.
+@item
 When applying patches that have been discussed (at length) on the mailing
 list, reference the thread in the log message.
-@subheading Always wait long enough before pushing changes
+@item
 Do NOT commit to code actively maintained by others without permission.
-Send a patch to ffmpeg-devel. If no one answers within a reasonable
+Send a patch to ffmpeg-devel instead. If no one answers within a reasonable
-time-frame (12h for build failures and security fixes, 3 days small changes,
+timeframe (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
+@item
-@subheading API/ABI changes should be discussed before they are made.
+Subscribe to the ffmpeg-cvslog mailing list. The diffs of all commits
-Do not change behavior of the programs (renaming options etc) or public
+are sent there and reviewed by all the other developers. Bugs and possible
-API or ABI without first discussing it on the ffmpeg-devel mailing list.
+improvements or general questions regarding commits are discussed there. We
-Do not remove widely used functionality or features (redundant code can be removed).
+expect you to react if problems with your code are uncovered.
-@subheading Remember to check if you need to bump versions for libav*.
+@item
-Depending on the change, you may need to change the version integer.
+Update the documentation if you change behavior or add features. If you are
 unsure how best to do this, send a patch to ffmpeg-devel, the documentation
 maintainer(s) will review and commit your stuff.
@item
 Try to keep important discussions and requests (also) on the public
 developer mailing list, so that all developers can benefit from them.
@item
 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.
@item
 Remember to check if you need to bump versions for the specific libav*
 parts (libavutil, libavcodec, libavformat) you are changing. You need
 to change the version integer.
 Incrementing the first component means no backward compatibility to
 previous versions (e.g. removal of a function from the public API).
 Incrementing the second component means backward compatible change
@@ -369,7 +377,7 @@ Incrementing the third component means a noteworthy binary compatible
 change (e.g. encoder bug fix that matters for the decoder). The third
 component always starts at 100 to distinguish FFmpeg from Libav.
-@subheading Warnings for correct code may be disabled if there is no other option.
+@item
 Compiler warnings indicate potential bugs or code with bad style. If a type of
 warning always points to correct and clean code, that warning should
 be disabled, not the code changed.
@@ -378,90 +386,24 @@ If it is a bug, the bug has to be fixed. If it is not, the code should
 be changed to not generate a warning unless that causes a slowdown
 or obfuscates the code.
-@subheading Check untrusted input properly.
+@item
 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.
@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.
@subheading Keep the documentation up to date.
 Update the documentation if you change behavior or add features. If you are
 unsure how best to do this, send a patch to ffmpeg-devel, the documentation
 maintainer(s) will review and commit your stuff.
@subheading Important discussions should be accessible to all.
 Try to keep important discussions and requests (also) on the public
 developer mailing list, so that all developers can benefit from them.
@subheading Check your entries in MAINTAINERS.
 Make sure that no parts of the codebase that you maintain are missing from the
@file{MAINTAINERS} file. If something that you want to maintain is missing add it with
 your name after it.
-If at some point you no longer want to maintain some code, then please help in
+If at some point you no longer want to maintain some code, then please help
-finding a new maintainer and also don't forget to update the @file{MAINTAINERS} file.
+finding a new maintainer and also don't forget updating the @file{MAINTAINERS} file.
@end enumerate
 We think our rules are not too hard. If you have comments, contact us.
@chapter Code of conduct
 Be friendly and respectful towards others and third parties.
 Treat others the way you yourself want to be treated.
 Be considerate. Not everyone shares the same viewpoint and priorities as you do.
 Different opinions and interpretations help the project.
 Looking at issues from a different perspective assists development.
 Do not assume malice for things that can be attributed to incompetence. Even if
 it is malice, it's rarely good to start with that as initial assumption.
 Stay friendly even if someone acts contrarily. Everyone has a bad day
 once in a while.
 If you yourself have a bad day or are angry then try to take a break and reply
 once you are calm and without anger if you have to.
 Try to help other team members and cooperate if you can.
 The goal of software development is to create technical excellence, not for any
 individual to be better and "win" against the others. Large software projects
 are only possible and successful through teamwork.
 If someone struggles do not put them down. Give them a helping hand
 instead and point them in the right direction.
 Finally, keep in mind the immortal words of Bill and Ted,
 "Be excellent to each other."
@anchor{Submitting patches}
-@chapter Submitting patches
+@section Submitting patches
 First, read the @ref{Coding Rules} above if you did not yet, in particular
 the rules regarding patch submission.
 When you submit your patch, please use @code{git format-patch} or
-@code{git send-email}. We cannot read other diffs :-).
+@code{git send-email}. We cannot read other diffs :-)
 Also please do not submit a patch which contains several unrelated changes.
 Split it into separate, self-contained pieces. This does not mean splitting
@@ -484,32 +426,12 @@ Also please if you send several patches, send each patch as a separate mail,
 do not attach several unrelated patches to the same mail.
 Patches should be posted to the
-@uref{https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel, ffmpeg-devel}
+@uref{http://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel, ffmpeg-devel}
 mailing list. Use @code{git send-email} when possible since it will properly
 send patches without requiring extra care. If you cannot, then send patches
 as base64-encoded attachments, so your patch is not trashed during
-transmission. Also ensure the correct mime type is used
+transmission.
 (text/x-diff or text/x-patch or at least text/plain) and that only one
 patch is inline or attached per mail.
 You can check @url{https://patchwork.ffmpeg.org}, if your patch does not show up, its mime type
 likely was wrong.
@subheading Sending patches from email clients
 Using @code{git send-email} might not be desirable for everyone. The
 following trick allows to send patches via email clients in a safe
 way. It has been tested with Outlook and Thunderbird (with X-Unsent
 extension) and might work with other applications.
 Create your patch like this:
@verbatim
 git format-patch -s -o "outputfolder" --add-header "X-Unsent: 1" --suffix .eml --to ffmpeg-devel@ffmpeg.org -1 1a2b3c4d
@end verbatim
 Now you'll just need to open the eml file with the email application
 and execute 'Send'.
@subheading Reviews
 Your patch will be reviewed on the mailing list. You will likely be asked
 to make some changes and are expected to send in an improved version that
 incorporates the requests from the review. This process may go through
@@ -520,7 +442,7 @@ 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
@@ -572,7 +494,7 @@ Did you make sure it compiles standalone, i.e. with
@end enumerate
-@chapter Patch submission checklist
+@section patch submission checklist
@enumerate
@item
@@ -582,9 +504,9 @@ 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})
+Did you sign off your patch? (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
+See @url{http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob_plain;f=Documentation/SubmittingPatches} for the meaning
-of @dfn{sign-off}.
+of sign off.
@item
 Did you provide a clear git commit log message?
@@ -617,10 +539,6 @@ tools/trasher, the noise bitstream filter, and
 should not crash, end in a (near) infinite loop, or allocate ridiculous
 amounts of memory when fed damaged data.
@item
 Did you test your decoder or demuxer against sample files?
 Samples may be obtained at @url{https://samples.ffmpeg.org}.
@item
 Does the patch not mix functional and cosmetic changes?
@@ -641,7 +559,7 @@ If the patch fixes a bug, did you provide a verbose analysis of the bug?
 If the patch fixes a bug, did you provide enough information, including
 a sample, so the bug can be reproduced and the fix can be verified?
 Note please do not attach samples >100k to mails but rather provide a
-URL, you can upload to @url{https://streams.videolan.org/upload/}.
+URL, you can upload to ftp://upload.ffmpeg.org
@item
 Did you provide a verbose summary about what the patch does change?
@@ -670,10 +588,10 @@ Lines with similar content should be aligned vertically when doing so
 improves readability.
@item
-Consider adding a regression test for your code.
+Consider to add a regression test for your code.
@item
-If you added YASM code please check that things still work with --disable-yasm.
+If you added YASM code please check that things still work with --disable-yasm
@item
 Make sure you check the return values of function and return appropriate
@@ -685,7 +603,7 @@ 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.
@@ -711,12 +629,8 @@ not related to the comments received during review. Such patches will
 be rejected. Instead, submit significant changes or new features as
 separate patches.
 Everyone is welcome to review patches. Also if you are waiting for your patch
 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.
@@ -727,18 +641,19 @@ 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.
+specific test then the media has to be inlcuded in the fate-suite.
 First please make sure that the sample file is as small as possible to test the
 respective decoder or demuxer sufficiently. Large files increase network
 bandwidth and disk space requirements.
 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
+message or introductionary 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
@@ -765,7 +680,7 @@ 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
@@ -779,13 +694,13 @@ 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
 Linux distributions, etc.). At regular times, a @strong{release
 manager} prepares, tests and publishes tarballs on the
-@url{https://ffmpeg.org} website.
+@url{http://ffmpeg.org} website.
 There are two kinds of releases:
@@ -811,7 +726,7 @@ 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:
@@ -835,7 +750,7 @@ 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:
@@ -864,7 +779,7 @@ Prepare the release tarballs in @code{bz2} and @code{gz} formats, and
 supplementing files that contain @code{gpg} signatures
@item
-Publish the tarballs at @url{https://ffmpeg.org/releases}. Create and
+Publish the tarballs at @url{http://ffmpeg.org/releases}. Create and
 push an annotated tag in the form @code{nX}, with @code{X}
 containing the version number.
@@ -876,7 +791,7 @@ with a news entry for the website.
 Publish the news entry.
@item
-Send an announcement to the mailing list.
+Send announcement to the mailing list.
@end enumerate
@bye
--- a/doc/doxy-wrapper.sh
+++ b/doc/doxy-wrapper.sh
@@ -1,24 +1,13 @@
 #!/bin/sh
-OUT_DIR="${1}"
+SRC_PATH="${1}"
-SRC_DIR="${2}"
+DOXYFILE="${2}"
-DOXYFILE="${3}"
+DOXYGEN="${3}"
 DOXYGEN="${4}"
-shift 4
+shift 3
 cd ${SRC_DIR}
 if [ -e "VERSION" ]; then
    VERSION=`cat "VERSION"`
 else
    VERSION=`git describe`
 fi
 $DOXYGEN - <<EOF
@INCLUDE        = ${DOXYFILE}
 INPUT           = $@
-HTML_TIMESTAMP  = NO
+EXAMPLE_PATH    = ${SRC_PATH}/doc/examples
 PROJECT_NUMBER  = $VERSION
 OUTPUT_DIRECTORY = $OUT_DIR
 EOF
--- a/doc/doxy/.gitignore
+++ b/doc/doxy/.gitignore
@@ -1 +0,0 @@
 /html/
--- a/doc/encoders.texi
+++ b/doc/encoders.texi
--- a/doc/errno.txt
+++ b/doc/errno.txt
@@ -76,7 +76,7 @@ EMFILE           POSIX     ++++++  Too many open files
 EMLINK           POSIX     ++++++  Too many links
 EMSGSIZE         POSIX     +++..+  Message too long
 EMULTIHOP        POSIX     ++4...  Multihop attempted
-ENAMETOOLONG     POSIX  -  ++++++  File name too long
+ENAMETOOLONG     POSIX  -  ++++++  Filen ame too long
 ENAVAIL                    +.....  No XENIX semaphores available
 ENEEDAUTH                  .++...  Need authenticator
 ENETDOWN         POSIX     +++..+  Network is down
--- a/doc/examples/.gitignore
+++ b/doc/examples/.gitignore
@@ -1,24 +0,0 @@
 /avio_list_dir
 /avio_reading
 /decode_audio
 /decode_video
 /demuxing_decoding
 /encode_audio
 /encode_video
 /extract_mvs
 /filter_audio
 /filtering_audio
 /filtering_video
 /http_multiclient
 /hw_decode
 /metadata
 /muxing
 /pc-uninstalled
 /qsvdec
 /remuxing
 /resampling_audio
 /scaling_video
 /transcode_aac
 /transcoding
 /vaapi_encode
 /vaapi_transcode
--- a/doc/examples/Makefile
+++ b/doc/examples/Makefile
@@ -1,64 +1,42 @@
-EXAMPLES-$(CONFIG_AVIO_LIST_DIR_EXAMPLE)     += avio_list_dir
+# use pkg-config for getting CFLAGS and LDLIBS
-EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE)      += avio_reading
+FFMPEG_LIBS=    libavdevice                        \
-EXAMPLES-$(CONFIG_DECODE_AUDIO_EXAMPLE)      += decode_audio
+                libavformat                        \
-EXAMPLES-$(CONFIG_DECODE_VIDEO_EXAMPLE)      += decode_video
+                libavfilter                        \
-EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding
+                libavcodec                         \
-EXAMPLES-$(CONFIG_ENCODE_AUDIO_EXAMPLE)      += encode_audio
+                libswresample                      \
-EXAMPLES-$(CONFIG_ENCODE_VIDEO_EXAMPLE)      += encode_video
+                libswscale                         \
-EXAMPLES-$(CONFIG_EXTRACT_MVS_EXAMPLE)       += extract_mvs
+                libavutil                          \
 EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE)      += filter_audio
 EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE)   += filtering_audio
 EXAMPLES-$(CONFIG_FILTERING_VIDEO_EXAMPLE)   += filtering_video
 EXAMPLES-$(CONFIG_HTTP_MULTICLIENT_EXAMPLE)  += http_multiclient
 EXAMPLES-$(CONFIG_HW_DECODE_EXAMPLE)         += hw_decode
 EXAMPLES-$(CONFIG_METADATA_EXAMPLE)          += metadata
 EXAMPLES-$(CONFIG_MUXING_EXAMPLE)            += muxing
 EXAMPLES-$(CONFIG_QSVDEC_EXAMPLE)            += qsvdec
 EXAMPLES-$(CONFIG_REMUXING_EXAMPLE)          += remuxing
 EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE)  += resampling_audio
 EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE)     += scaling_video
 EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE)     += transcode_aac
 EXAMPLES-$(CONFIG_TRANSCODING_EXAMPLE)       += transcoding
 EXAMPLES-$(CONFIG_VAAPI_ENCODE_EXAMPLE)      += vaapi_encode
 EXAMPLES-$(CONFIG_VAAPI_TRANSCODE_EXAMPLE)   += vaapi_transcode
-EXAMPLES       := $(EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)$(EXESUF))
+CFLAGS += -Wall -g
-EXAMPLES_G     := $(EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)_g$(EXESUF))
+CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
-ALL_EXAMPLES   := $(EXAMPLES) $(EXAMPLES-:%=doc/examples/%$(PROGSSUF)$(EXESUF))
+LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
 ALL_EXAMPLES_G := $(EXAMPLES_G) $(EXAMPLES-:%=doc/examples/%$(PROGSSUF)_g$(EXESUF))
 PROGS          += $(EXAMPLES)
-EXAMPLE_MAKEFILE := $(SRC_PATH)/doc/examples/Makefile
+EXAMPLES=       avio_reading                       \
-EXAMPLES_FILES := $(wildcard $(SRC_PATH)/doc/examples/*.c) $(SRC_PATH)/doc/examples/README $(EXAMPLE_MAKEFILE)
+                decoding_encoding                  \
                demuxing_decoding                  \
                filtering_video                    \
                filtering_audio                    \
                metadata                           \
                muxing                             \
                remuxing                           \
                resampling_audio                   \
                scaling_video                      \
                transcode_aac                      \
                transcoding                        \
-$(foreach P,$(EXAMPLES),$(eval OBJS-$(P:%$(PROGSSUF)$(EXESUF)=%) = $(P:%$(PROGSSUF)$(EXESUF)=%).o))
+OBJS=$(addsuffix .o,$(EXAMPLES))
 $(EXAMPLES_G): %$(PROGSSUF)_g$(EXESUF): %.o
-examples: $(EXAMPLES)
+# the following examples make explicit use of the math library
 avcodec:           LDLIBS += -lm
 muxing:            LDLIBS += -lm
 resampling_audio:  LDLIBS += -lm
-$(EXAMPLES:%$(PROGSSUF)$(EXESUF)=%.o): | doc/examples
+.phony: all clean-test clean
 OUTDIRS += doc/examples
-DOXY_INPUT += $(EXAMPLES:%$(PROGSSUF)$(EXESUF)=%.c)
+all: $(OBJS) $(EXAMPLES)
-install: install-examples
+clean-test:
 	$(RM) test*.pgm test.h264 test.mp2 test.sw test.mpg
-install-examples: $(EXAMPLES_FILES)
+clean: clean-test
-	$(Q)mkdir -p "$(DATADIR)/examples"
+	$(RM) $(EXAMPLES) $(OBJS)
 	$(INSTALL) -m 644 $(EXAMPLES_FILES) "$(DATADIR)/examples"
 	$(INSTALL) -m 644 $(EXAMPLE_MAKEFILE:%=%.example) "$(DATADIR)/examples/Makefile"
 uninstall: uninstall-examples
 uninstall-examples:
 	$(RM) -r "$(DATADIR)/examples"
 examplesclean:
 	$(RM) $(ALL_EXAMPLES) $(ALL_EXAMPLES_G)
 	$(RM) $(CLEANSUFFIXES:%=doc/examples/%)
 docclean:: examplesclean
 -include $(wildcard $(EXAMPLES:%$(PROGSSUF)$(EXESUF)=%.d))
 .PHONY: examples
--- a/doc/examples/Makefile.example
+++ b/doc/examples/Makefile.example
@@ -1,50 +0,0 @@
 # use pkg-config for getting CFLAGS and LDLIBS
 FFMPEG_LIBS=    libavdevice                        \
                libavformat                        \
                libavfilter                        \
                libavcodec                         \
                libswresample                      \
                libswscale                         \
                libavutil                          \
 CFLAGS += -Wall -g
 CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
 LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
 EXAMPLES=       avio_list_dir                      \
                avio_reading                       \
                decode_audio                       \
                decode_video                       \
                demuxing_decoding                  \
                encode_audio                       \
                encode_video                       \
                extract_mvs                        \
                filtering_video                    \
                filtering_audio                    \
                http_multiclient                   \
                hw_decode                          \
                metadata                           \
                muxing                             \
                remuxing                           \
                resampling_audio                   \
                scaling_video                      \
                transcode_aac                      \
                transcoding                        \
 OBJS=$(addsuffix .o,$(EXAMPLES))
 # the following examples make explicit use of the math library
 avcodec:           LDLIBS += -lm
 encode_audio:      LDLIBS += -lm
 muxing:            LDLIBS += -lm
 resampling_audio:  LDLIBS += -lm
 .phony: all clean-test clean
 all: $(OBJS) $(EXAMPLES)
 clean-test:
 	$(RM) test*.pgm test.h264 test.mp2 test.sw test.mpg
 clean: clean-test
 	$(RM) $(EXAMPLES) $(OBJS)
--- a/doc/examples/avio_list_dir.c
+++ b/doc/examples/avio_list_dir.c
@@ -1,130 +0,0 @@
 /*
 * Copyright (c) 2014 Lukasz Marek
 *
 * Permission is hereby granted, free of charge, to any person obtaining a copy
 * of this software and associated documentation files (the "Software"), to deal
 * in the Software without restriction, including without limitation the rights
 * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
 * copies of the Software, and to permit persons to whom the Software is
 * furnished to do so, subject to the following conditions:
 *
 * The above copyright notice and this permission notice shall be included in
 * all copies or substantial portions of the Software.
 *
 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
 * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
 * THE SOFTWARE.
 */
 #include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
 #include <libavformat/avio.h>
 static const char *type_string(int type)
 {
    switch (type) {
    case AVIO_ENTRY_DIRECTORY:
        return "<DIR>";
    case AVIO_ENTRY_FILE:
        return "<FILE>";
    case AVIO_ENTRY_BLOCK_DEVICE:
        return "<BLOCK DEVICE>";
    case AVIO_ENTRY_CHARACTER_DEVICE:
        return "<CHARACTER DEVICE>";
    case AVIO_ENTRY_NAMED_PIPE:
        return "<PIPE>";
    case AVIO_ENTRY_SYMBOLIC_LINK:
        return "<LINK>";
    case AVIO_ENTRY_SOCKET:
        return "<SOCKET>";
    case AVIO_ENTRY_SERVER:
        return "<SERVER>";
    case AVIO_ENTRY_SHARE:
        return "<SHARE>";
    case AVIO_ENTRY_WORKGROUP:
        return "<WORKGROUP>";
    case AVIO_ENTRY_UNKNOWN:
    default:
        break;
    }
    return "<UNKNOWN>";
 }
 static int list_op(const char *input_dir)
 {
    AVIODirEntry *entry = NULL;
    AVIODirContext *ctx = NULL;
    int cnt, ret;
    char filemode[4], uid_and_gid[20];
    if ((ret = avio_open_dir(&ctx, input_dir, NULL)) < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot open directory: %s.\n", av_err2str(ret));
        goto fail;
    }
    cnt = 0;
    for (;;) {
        if ((ret = avio_read_dir(ctx, &entry)) < 0) {
            av_log(NULL, AV_LOG_ERROR, "Cannot list directory: %s.\n", av_err2str(ret));
            goto fail;
        }
        if (!entry)
            break;
        if (entry->filemode == -1) {
            snprintf(filemode, 4, "???");
        } else {
            snprintf(filemode, 4, "%3"PRIo64, entry->filemode);
        }
        snprintf(uid_and_gid, 20, "%"PRId64"(%"PRId64")", entry->user_id, entry->group_id);
        if (cnt == 0)
            av_log(NULL, AV_LOG_INFO, "%-9s %12s %30s %10s %s %16s %16s %16s\n",
                   "TYPE", "SIZE", "NAME", "UID(GID)", "UGO", "MODIFIED",
                   "ACCESSED", "STATUS_CHANGED");
        av_log(NULL, AV_LOG_INFO, "%-9s %12"PRId64" %30s %10s %s %16"PRId64" %16"PRId64" %16"PRId64"\n",
               type_string(entry->type),
               entry->size,
               entry->name,
               uid_and_gid,
               filemode,
               entry->modification_timestamp,
               entry->access_timestamp,
               entry->status_change_timestamp);
        avio_free_directory_entry(&entry);
        cnt++;
    };
  fail:
    avio_close_dir(&ctx);
    return ret;
 }
 static void usage(const char *program_name)
 {
    fprintf(stderr, "usage: %s input_dir\n"
            "API example program to show how to list files in directory "
            "accessed through AVIOContext.\n", program_name);
 }
 int main(int argc, char *argv[])
 {
    int ret;
    av_log_set_level(AV_LOG_DEBUG);
    if (argc < 2) {
        usage(argv[0]);
        return 1;
    }
    avformat_network_init();
    ret = list_op(argv[1]);
    avformat_network_deinit();
    return ret < 0 ? 1 : 0;
 }
--- a/doc/examples/avio_reading.c
+++ b/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) {
--- a/doc/examples/decode_audio.c
+++ b/doc/examples/decode_audio.c
@@ -1,236 +0,0 @@
 /*
 * Copyright (c) 2001 Fabrice Bellard
 *
 * 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
 * audio decoding with libavcodec API example
 *
 * @example decode_audio.c
 */
 #include <stdio.h>
 #include <stdlib.h>
 #include <string.h>
 #include <libavutil/frame.h>
 #include <libavutil/mem.h>
 #include <libavcodec/avcodec.h>
 #define AUDIO_INBUF_SIZE 20480
 #define AUDIO_REFILL_THRESH 4096
 static int get_format_from_sample_fmt(const char **fmt,
                                      enum AVSampleFormat sample_fmt)
 {
    int i;
    struct sample_fmt_entry {
        enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
    } sample_fmt_entries[] = {
        { AV_SAMPLE_FMT_U8,  "u8",    "u8"    },
        { AV_SAMPLE_FMT_S16, "s16be", "s16le" },
        { AV_SAMPLE_FMT_S32, "s32be", "s32le" },
        { AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
        { AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
    };
    *fmt = NULL;
    for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
        struct sample_fmt_entry *entry = &sample_fmt_entries[i];
        if (sample_fmt == entry->sample_fmt) {
            *fmt = AV_NE(entry->fmt_be, entry->fmt_le);
            return 0;
        }
    }
    fprintf(stderr,
            "sample format %s is not supported as output format\n",
            av_get_sample_fmt_name(sample_fmt));
    return -1;
 }
 static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame,
                   FILE *outfile)
 {
    int i, ch;
    int ret, data_size;
    /* send the packet with the compressed data to the decoder */
    ret = avcodec_send_packet(dec_ctx, pkt);
    if (ret < 0) {
        fprintf(stderr, "Error submitting the packet to the decoder\n");
        exit(1);
    }
    /* read all the output frames (in general there may be any number of them */
    while (ret >= 0) {
        ret = avcodec_receive_frame(dec_ctx, frame);
        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
            return;
        else if (ret < 0) {
            fprintf(stderr, "Error during decoding\n");
            exit(1);
        }
        data_size = av_get_bytes_per_sample(dec_ctx->sample_fmt);
        if (data_size < 0) {
            /* This should not occur, checking just for paranoia */
            fprintf(stderr, "Failed to calculate data size\n");
            exit(1);
        }
        for (i = 0; i < frame->nb_samples; i++)
            for (ch = 0; ch < dec_ctx->ch_layout.nb_channels; ch++)
                fwrite(frame->data[ch] + data_size*i, 1, data_size, outfile);
    }
 }
 int main(int argc, char **argv)
 {
    const char *outfilename, *filename;
    const AVCodec *codec;
    AVCodecContext *c= NULL;
    AVCodecParserContext *parser = NULL;
    int len, ret;
    FILE *f, *outfile;
    uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
    uint8_t *data;
    size_t   data_size;
    AVPacket *pkt;
    AVFrame *decoded_frame = NULL;
    enum AVSampleFormat sfmt;
    int n_channels = 0;
    const char *fmt;
    if (argc <= 2) {
        fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
        exit(0);
    }
    filename    = argv[1];
    outfilename = argv[2];
    pkt = av_packet_alloc();
    /* find the MPEG audio decoder */
    codec = avcodec_find_decoder(AV_CODEC_ID_MP2);
    if (!codec) {
        fprintf(stderr, "Codec not found\n");
        exit(1);
    }
    parser = av_parser_init(codec->id);
    if (!parser) {
        fprintf(stderr, "Parser not found\n");
        exit(1);
    }
    c = avcodec_alloc_context3(codec);
    if (!c) {
        fprintf(stderr, "Could not allocate audio codec context\n");
        exit(1);
    }
    /* open it */
    if (avcodec_open2(c, codec, NULL) < 0) {
        fprintf(stderr, "Could not open codec\n");
        exit(1);
    }
    f = fopen(filename, "rb");
    if (!f) {
        fprintf(stderr, "Could not open %s\n", filename);
        exit(1);
    }
    outfile = fopen(outfilename, "wb");
    if (!outfile) {
        av_free(c);
        exit(1);
    }
    /* decode until eof */
    data      = inbuf;
    data_size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
    while (data_size > 0) {
        if (!decoded_frame) {
            if (!(decoded_frame = av_frame_alloc())) {
                fprintf(stderr, "Could not allocate audio frame\n");
                exit(1);
            }
        }
        ret = av_parser_parse2(parser, c, &pkt->data, &pkt->size,
                               data, data_size,
                               AV_NOPTS_VALUE, AV_NOPTS_VALUE, 0);
        if (ret < 0) {
            fprintf(stderr, "Error while parsing\n");
            exit(1);
        }
        data      += ret;
        data_size -= ret;
        if (pkt->size)
            decode(c, pkt, decoded_frame, outfile);
        if (data_size < AUDIO_REFILL_THRESH) {
            memmove(inbuf, data, data_size);
            data = inbuf;
            len = fread(data + data_size, 1,
                        AUDIO_INBUF_SIZE - data_size, f);
            if (len > 0)
                data_size += len;
        }
    }
    /* flush the decoder */
    pkt->data = NULL;
    pkt->size = 0;
    decode(c, pkt, decoded_frame, outfile);
    /* print output pcm infomations, because there have no metadata of pcm */
    sfmt = c->sample_fmt;
    if (av_sample_fmt_is_planar(sfmt)) {
        const char *packed = av_get_sample_fmt_name(sfmt);
        printf("Warning: the sample format the decoder produced is planar "
               "(%s). This example will output the first channel only.\n",
               packed ? packed : "?");
        sfmt = av_get_packed_sample_fmt(sfmt);
    }
    n_channels = c->ch_layout.nb_channels;
    if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
        goto end;
    printf("Play the output audio file with the command:\n"
           "ffplay -f %s -ac %d -ar %d %s\n",
           fmt, n_channels, c->sample_rate,
           outfilename);
 end:
    fclose(outfile);
    fclose(f);
    avcodec_free_context(&c);
    av_parser_close(parser);
    av_frame_free(&decoded_frame);
    av_packet_free(&pkt);
    return 0;
 }
--- a/doc/examples/decode_video.c
+++ b/doc/examples/decode_video.c
@@ -1,191 +0,0 @@
 /*
 * Copyright (c) 2001 Fabrice Bellard
 *
 * 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
 * video decoding with libavcodec API example
 *
 * @example decode_video.c
 */
 #include <stdio.h>
 #include <stdlib.h>
 #include <string.h>
 #include <libavcodec/avcodec.h>
 #define INBUF_SIZE 4096
 static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
                     char *filename)
 {
    FILE *f;
    int i;
    f = fopen(filename,"wb");
    fprintf(f, "P5\n%d %d\n%d\n", xsize, ysize, 255);
    for (i = 0; i < ysize; i++)
        fwrite(buf + i * wrap, 1, xsize, f);
    fclose(f);
 }
 static void decode(AVCodecContext *dec_ctx, AVFrame *frame, AVPacket *pkt,
                   const char *filename)
 {
    char buf[1024];
    int ret;
    ret = avcodec_send_packet(dec_ctx, pkt);
    if (ret < 0) {
        fprintf(stderr, "Error sending a packet for decoding\n");
        exit(1);
    }
    while (ret >= 0) {
        ret = avcodec_receive_frame(dec_ctx, frame);
        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
            return;
        else if (ret < 0) {
            fprintf(stderr, "Error during decoding\n");
            exit(1);
        }
        printf("saving frame %3d\n", dec_ctx->frame_number);
        fflush(stdout);
        /* the picture is allocated by the decoder. no need to
           free it */
        snprintf(buf, sizeof(buf), "%s-%d", filename, dec_ctx->frame_number);
        pgm_save(frame->data[0], frame->linesize[0],
                 frame->width, frame->height, buf);
    }
 }
 int main(int argc, char **argv)
 {
    const char *filename, *outfilename;
    const AVCodec *codec;
    AVCodecParserContext *parser;
    AVCodecContext *c= NULL;
    FILE *f;
    AVFrame *frame;
    uint8_t inbuf[INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
    uint8_t *data;
    size_t   data_size;
    int ret;
    int eof;
    AVPacket *pkt;
    if (argc <= 2) {
        fprintf(stderr, "Usage: %s <input file> <output file>\n"
                "And check your input file is encoded by mpeg1video please.\n", argv[0]);
        exit(0);
    }
    filename    = argv[1];
    outfilename = argv[2];
    pkt = av_packet_alloc();
    if (!pkt)
        exit(1);
    /* set end of buffer to 0 (this ensures that no overreading happens for damaged MPEG streams) */
    memset(inbuf + INBUF_SIZE, 0, AV_INPUT_BUFFER_PADDING_SIZE);
    /* find the MPEG-1 video decoder */
    codec = avcodec_find_decoder(AV_CODEC_ID_MPEG1VIDEO);
    if (!codec) {
        fprintf(stderr, "Codec not found\n");
        exit(1);
    }
    parser = av_parser_init(codec->id);
    if (!parser) {
        fprintf(stderr, "parser not found\n");
        exit(1);
    }
    c = avcodec_alloc_context3(codec);
    if (!c) {
        fprintf(stderr, "Could not allocate video codec context\n");
        exit(1);
    }
    /* For some codecs, such as msmpeg4 and mpeg4, width and height
       MUST be initialized there because this information is not
       available in the bitstream. */
    /* open it */
    if (avcodec_open2(c, codec, NULL) < 0) {
        fprintf(stderr, "Could not open codec\n");
        exit(1);
    }
    f = fopen(filename, "rb");
    if (!f) {
        fprintf(stderr, "Could not open %s\n", filename);
        exit(1);
    }
    frame = av_frame_alloc();
    if (!frame) {
        fprintf(stderr, "Could not allocate video frame\n");
        exit(1);
    }
    do {
        /* read raw data from the input file */
        data_size = fread(inbuf, 1, INBUF_SIZE, f);
        if (ferror(f))
            break;
        eof = !data_size;
        /* use the parser to split the data into frames */
        data = inbuf;
        while (data_size > 0 || eof) {
            ret = av_parser_parse2(parser, c, &pkt->data, &pkt->size,
                                   data, data_size, AV_NOPTS_VALUE, AV_NOPTS_VALUE, 0);
            if (ret < 0) {
                fprintf(stderr, "Error while parsing\n");
                exit(1);
            }
            data      += ret;
            data_size -= ret;
            if (pkt->size)
                decode(c, frame, pkt, outfilename);
            else if (eof)
                break;
        }
    } while (!eof);
    /* flush the decoder */
    decode(c, frame, NULL, outfilename);
    fclose(f);
    av_parser_close(parser);
    avcodec_free_context(&c);
    av_frame_free(&frame);
    av_packet_free(&pkt);
    return 0;
 }
--- a/doc/examples/decoding_encoding.c
+++ b/doc/examples/decoding_encoding.c
@@ -0,0 +1,664 @@
 /*
 * Copyright (c) 2001 Fabrice Bellard
 *
 * 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
 * libavcodec API use example.
 *
 * @example decoding_encoding.c
 * Note that libavcodec only handles codecs (mpeg, mpeg4, etc...),
 * not file formats (avi, vob, mp4, mov, mkv, mxf, flv, mpegts, mpegps, etc...). See library 'libavformat' for the
 * format handling
 */
 #include <math.h>
 #include <libavutil/opt.h>
 #include <libavcodec/avcodec.h>
 #include <libavutil/channel_layout.h>
 #include <libavutil/common.h>
 #include <libavutil/imgutils.h>
 #include <libavutil/mathematics.h>
 #include <libavutil/samplefmt.h>
 #define INBUF_SIZE 4096
 #define AUDIO_INBUF_SIZE 20480
 #define AUDIO_REFILL_THRESH 4096
 /* check that a given sample format is supported by the encoder */
 static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt)
 {
    const enum AVSampleFormat *p = codec->sample_fmts;
    while (*p != AV_SAMPLE_FMT_NONE) {
        if (*p == sample_fmt)
            return 1;
        p++;
    }
    return 0;
 }
 /* just pick the highest supported samplerate */
 static int select_sample_rate(AVCodec *codec)
 {
    const int *p;
    int best_samplerate = 0;
    if (!codec->supported_samplerates)
        return 44100;
    p = codec->supported_samplerates;
    while (*p) {
        best_samplerate = FFMAX(*p, best_samplerate);
        p++;
    }
    return best_samplerate;
 }
 /* select layout with the highest channel count */
 static int select_channel_layout(AVCodec *codec)
 {
    const uint64_t *p;
    uint64_t best_ch_layout = 0;
    int best_nb_channels   = 0;
    if (!codec->channel_layouts)
        return AV_CH_LAYOUT_STEREO;
    p = codec->channel_layouts;
    while (*p) {
        int nb_channels = av_get_channel_layout_nb_channels(*p);
        if (nb_channels > best_nb_channels) {
            best_ch_layout    = *p;
            best_nb_channels = nb_channels;
        }
        p++;
    }
    return best_ch_layout;
 }
 /*
 * Audio encoding example
 */
 static void audio_encode_example(const char *filename)
 {
    AVCodec *codec;
    AVCodecContext *c= NULL;
    AVFrame *frame;
    AVPacket pkt;
    int i, j, k, ret, got_output;
    int buffer_size;
    FILE *f;
    uint16_t *samples;
    float t, tincr;
    printf("Encode audio file %s\n", filename);
    /* find the MP2 encoder */
    codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
    if (!codec) {
        fprintf(stderr, "Codec not found\n");
        exit(1);
    }
    c = avcodec_alloc_context3(codec);
    if (!c) {
        fprintf(stderr, "Could not allocate audio codec context\n");
        exit(1);
    }
    /* put sample parameters */
    c->bit_rate = 64000;
    /* check that the encoder supports s16 pcm input */
    c->sample_fmt = AV_SAMPLE_FMT_S16;
    if (!check_sample_fmt(codec, c->sample_fmt)) {
        fprintf(stderr, "Encoder does not support sample format %s",
                av_get_sample_fmt_name(c->sample_fmt));
        exit(1);
    }
    /* select other audio parameters supported by the encoder */
    c->sample_rate    = select_sample_rate(codec);
    c->channel_layout = select_channel_layout(codec);
    c->channels       = av_get_channel_layout_nb_channels(c->channel_layout);
    /* open it */
    if (avcodec_open2(c, codec, NULL) < 0) {
        fprintf(stderr, "Could not open codec\n");
        exit(1);
    }
    f = fopen(filename, "wb");
    if (!f) {
        fprintf(stderr, "Could not open %s\n", filename);
        exit(1);
    }
    /* frame containing input raw audio */
    frame = av_frame_alloc();
    if (!frame) {
        fprintf(stderr, "Could not allocate audio frame\n");
        exit(1);
    }
    frame->nb_samples     = c->frame_size;
    frame->format         = c->sample_fmt;
    frame->channel_layout = c->channel_layout;
    /* the codec gives us the frame size, in samples,
     * we calculate the size of the samples buffer in bytes */
    buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size,
                                             c->sample_fmt, 0);
    if (buffer_size < 0) {
        fprintf(stderr, "Could not get sample buffer size\n");
        exit(1);
    }
    samples = av_malloc(buffer_size);
    if (!samples) {
        fprintf(stderr, "Could not allocate %d bytes for samples buffer\n",
                buffer_size);
        exit(1);
    }
    /* setup the data pointers in the AVFrame */
    ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
                                   (const uint8_t*)samples, buffer_size, 0);
    if (ret < 0) {
        fprintf(stderr, "Could not setup audio frame\n");
        exit(1);
    }
    /* encode a single tone sound */
    t = 0;
    tincr = 2 * M_PI * 440.0 / c->sample_rate;
    for (i = 0; i < 200; i++) {
        av_init_packet(&pkt);
        pkt.data = NULL; // packet data will be allocated by the encoder
        pkt.size = 0;
        for (j = 0; j < c->frame_size; j++) {
            samples[2*j] = (int)(sin(t) * 10000);
            for (k = 1; k < c->channels; k++)
                samples[2*j + k] = samples[2*j];
            t += tincr;
        }
        /* encode the samples */
        ret = avcodec_encode_audio2(c, &pkt, frame, &got_output);
        if (ret < 0) {
            fprintf(stderr, "Error encoding audio frame\n");
            exit(1);
        }
        if (got_output) {
            fwrite(pkt.data, 1, pkt.size, f);
            av_free_packet(&pkt);
        }
    }
    /* get the delayed frames */
    for (got_output = 1; got_output; i++) {
        ret = avcodec_encode_audio2(c, &pkt, NULL, &got_output);
        if (ret < 0) {
            fprintf(stderr, "Error encoding frame\n");
            exit(1);
        }
        if (got_output) {
            fwrite(pkt.data, 1, pkt.size, f);
            av_free_packet(&pkt);
        }
    }
    fclose(f);
    av_freep(&samples);
    av_frame_free(&frame);
    avcodec_close(c);
    av_free(c);
 }
 /*
 * Audio decoding.
 */
 static void audio_decode_example(const char *outfilename, const char *filename)
 {
    AVCodec *codec;
    AVCodecContext *c= NULL;
    int len;
    FILE *f, *outfile;
    uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
    AVPacket avpkt;
    AVFrame *decoded_frame = NULL;
    av_init_packet(&avpkt);
    printf("Decode audio file %s to %s\n", filename, outfilename);
    /* find the mpeg audio decoder */
    codec = avcodec_find_decoder(AV_CODEC_ID_MP2);
    if (!codec) {
        fprintf(stderr, "Codec not found\n");
        exit(1);
    }
    c = avcodec_alloc_context3(codec);
    if (!c) {
        fprintf(stderr, "Could not allocate audio codec context\n");
        exit(1);
    }
    /* open it */
    if (avcodec_open2(c, codec, NULL) < 0) {
        fprintf(stderr, "Could not open codec\n");
        exit(1);
    }
    f = fopen(filename, "rb");
    if (!f) {
        fprintf(stderr, "Could not open %s\n", filename);
        exit(1);
    }
    outfile = fopen(outfilename, "wb");
    if (!outfile) {
        av_free(c);
        exit(1);
    }
    /* decode until eof */
    avpkt.data = inbuf;
    avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
    while (avpkt.size > 0) {
        int got_frame = 0;
        if (!decoded_frame) {
            if (!(decoded_frame = av_frame_alloc())) {
                fprintf(stderr, "Could not allocate audio frame\n");
                exit(1);
            }
        }
        len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
        if (len < 0) {
            fprintf(stderr, "Error while decoding\n");
            exit(1);
        }
        if (got_frame) {
            /* if a frame has been decoded, output it */
            int data_size = av_samples_get_buffer_size(NULL, c->channels,
                                                       decoded_frame->nb_samples,
                                                       c->sample_fmt, 1);
            if (data_size < 0) {
                /* This should not occur, checking just for paranoia */
                fprintf(stderr, "Failed to calculate data size\n");
                exit(1);
            }
            fwrite(decoded_frame->data[0], 1, data_size, outfile);
        }
        avpkt.size -= len;
        avpkt.data += len;
        avpkt.dts =
        avpkt.pts = AV_NOPTS_VALUE;
        if (avpkt.size < AUDIO_REFILL_THRESH) {
            /* Refill the input buffer, to avoid trying to decode
             * incomplete frames. Instead of this, one could also use
             * a parser, or use a proper container format through
             * libavformat. */
            memmove(inbuf, avpkt.data, avpkt.size);
            avpkt.data = inbuf;
            len = fread(avpkt.data + avpkt.size, 1,
                        AUDIO_INBUF_SIZE - avpkt.size, f);
            if (len > 0)
                avpkt.size += len;
        }
    }
    fclose(outfile);
    fclose(f);
    avcodec_close(c);
    av_free(c);
    av_frame_free(&decoded_frame);
 }
 /*
 * Video encoding example
 */
 static void video_encode_example(const char *filename, int codec_id)
 {
    AVCodec *codec;
    AVCodecContext *c= NULL;
    int i, ret, x, y, got_output;
    FILE *f;
    AVFrame *frame;
    AVPacket pkt;
    uint8_t endcode[] = { 0, 0, 1, 0xb7 };
    printf("Encode video file %s\n", filename);
    /* find the mpeg1 video encoder */
    codec = avcodec_find_encoder(codec_id);
    if (!codec) {
        fprintf(stderr, "Codec not found\n");
        exit(1);
    }
    c = avcodec_alloc_context3(codec);
    if (!c) {
        fprintf(stderr, "Could not allocate video codec context\n");
        exit(1);
    }
    /* put sample parameters */
    c->bit_rate = 400000;
    /* resolution must be a multiple of two */
    c->width = 352;
    c->height = 288;
    /* frames per second */
    c->time_base = (AVRational){1,25};
    /* emit one intra frame every ten frames
     * check frame pict_type before passing frame
     * to encoder, if frame->pict_type is AV_PICTURE_TYPE_I
     * then gop_size is ignored and the output of encoder
     * will always be I frame irrespective to gop_size
     */
    c->gop_size = 10;
    c->max_b_frames = 1;
    c->pix_fmt = AV_PIX_FMT_YUV420P;
    if (codec_id == AV_CODEC_ID_H264)
        av_opt_set(c->priv_data, "preset", "slow", 0);
    /* open it */
    if (avcodec_open2(c, codec, NULL) < 0) {
        fprintf(stderr, "Could not open codec\n");
        exit(1);
    }
    f = fopen(filename, "wb");
    if (!f) {
        fprintf(stderr, "Could not open %s\n", filename);
        exit(1);
    }
    frame = av_frame_alloc();
    if (!frame) {
        fprintf(stderr, "Could not allocate video frame\n");
        exit(1);
    }
    frame->format = c->pix_fmt;
    frame->width  = c->width;
    frame->height = c->height;
    /* the image can be allocated by any means and av_image_alloc() is
     * just the most convenient way if av_malloc() is to be used */
    ret = av_image_alloc(frame->data, frame->linesize, c->width, c->height,
                         c->pix_fmt, 32);
    if (ret < 0) {
        fprintf(stderr, "Could not allocate raw picture buffer\n");
        exit(1);
    }
    /* encode 1 second of video */
    for (i = 0; i < 25; i++) {
        av_init_packet(&pkt);
        pkt.data = NULL;    // packet data will be allocated by the encoder
        pkt.size = 0;
        fflush(stdout);
        /* prepare a dummy image */
        /* Y */
        for (y = 0; y < c->height; y++) {
            for (x = 0; x < c->width; x++) {
                frame->data[0][y * frame->linesize[0] + x] = x + y + i * 3;
            }
        }
        /* Cb and Cr */
        for (y = 0; y < c->height/2; y++) {
            for (x = 0; x < c->width/2; x++) {
                frame->data[1][y * frame->linesize[1] + x] = 128 + y + i * 2;
                frame->data[2][y * frame->linesize[2] + x] = 64 + x + i * 5;
            }
        }
        frame->pts = i;
        /* encode the image */
        ret = avcodec_encode_video2(c, &pkt, frame, &got_output);
        if (ret < 0) {
            fprintf(stderr, "Error encoding frame\n");
            exit(1);
        }
        if (got_output) {
            printf("Write frame %3d (size=%5d)\n", i, pkt.size);
            fwrite(pkt.data, 1, pkt.size, f);
            av_free_packet(&pkt);
        }
    }
    /* get the delayed frames */
    for (got_output = 1; got_output; i++) {
        fflush(stdout);
        ret = avcodec_encode_video2(c, &pkt, NULL, &got_output);
        if (ret < 0) {
            fprintf(stderr, "Error encoding frame\n");
            exit(1);
        }
        if (got_output) {
            printf("Write frame %3d (size=%5d)\n", i, pkt.size);
            fwrite(pkt.data, 1, pkt.size, f);
            av_free_packet(&pkt);
        }
    }
    /* add sequence end code to have a real mpeg file */
    fwrite(endcode, 1, sizeof(endcode), f);
    fclose(f);
    avcodec_close(c);
    av_free(c);
    av_freep(&frame->data[0]);
    av_frame_free(&frame);
    printf("\n");
 }
 /*
 * Video decoding example
 */
 static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
                     char *filename)
 {
    FILE *f;
    int i;
    f = fopen(filename,"w");
    fprintf(f, "P5\n%d %d\n%d\n", xsize, ysize, 255);
    for (i = 0; i < ysize; i++)
        fwrite(buf + i * wrap, 1, xsize, f);
    fclose(f);
 }
 static int decode_write_frame(const char *outfilename, AVCodecContext *avctx,
                              AVFrame *frame, int *frame_count, AVPacket *pkt, int last)
 {
    int len, got_frame;
    char buf[1024];
    len = avcodec_decode_video2(avctx, frame, &got_frame, pkt);
    if (len < 0) {
        fprintf(stderr, "Error while decoding frame %d\n", *frame_count);
        return len;
    }
    if (got_frame) {
        printf("Saving %sframe %3d\n", last ? "last " : "", *frame_count);
        fflush(stdout);
        /* the picture is allocated by the decoder, no need to free it */
        snprintf(buf, sizeof(buf), outfilename, *frame_count);
        pgm_save(frame->data[0], frame->linesize[0],
                 avctx->width, avctx->height, buf);
        (*frame_count)++;
    }
    if (pkt->data) {
        pkt->size -= len;
        pkt->data += len;
    }
    return 0;
 }
 static void video_decode_example(const char *outfilename, const char *filename)
 {
    AVCodec *codec;
    AVCodecContext *c= NULL;
    int frame_count;
    FILE *f;
    AVFrame *frame;
    uint8_t inbuf[INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
    AVPacket avpkt;
    av_init_packet(&avpkt);
    /* set end of buffer to 0 (this ensures that no overreading happens for damaged mpeg streams) */
    memset(inbuf + INBUF_SIZE, 0, FF_INPUT_BUFFER_PADDING_SIZE);
    printf("Decode video file %s to %s\n", filename, outfilename);
    /* find the mpeg1 video decoder */
    codec = avcodec_find_decoder(AV_CODEC_ID_MPEG1VIDEO);
    if (!codec) {
        fprintf(stderr, "Codec not found\n");
        exit(1);
    }
    c = avcodec_alloc_context3(codec);
    if (!c) {
        fprintf(stderr, "Could not allocate video codec context\n");
        exit(1);
    }
    if(codec->capabilities&CODEC_CAP_TRUNCATED)
        c->flags|= CODEC_FLAG_TRUNCATED; /* we do not send complete frames */
    /* For some codecs, such as msmpeg4 and mpeg4, width and height
       MUST be initialized there because this information is not
       available in the bitstream. */
    /* open it */
    if (avcodec_open2(c, codec, NULL) < 0) {
        fprintf(stderr, "Could not open codec\n");
        exit(1);
    }
    f = fopen(filename, "rb");
    if (!f) {
        fprintf(stderr, "Could not open %s\n", filename);
        exit(1);
    }
    frame = av_frame_alloc();
    if (!frame) {
        fprintf(stderr, "Could not allocate video frame\n");
        exit(1);
    }
    frame_count = 0;
    for (;;) {
        avpkt.size = fread(inbuf, 1, INBUF_SIZE, f);
        if (avpkt.size == 0)
            break;
        /* NOTE1: some codecs are stream based (mpegvideo, mpegaudio)
           and this is the only method to use them because you cannot
           know the compressed data size before analysing it.
           BUT some other codecs (msmpeg4, mpeg4) are inherently frame
           based, so you must call them with all the data for one
           frame exactly. You must also initialize 'width' and
           'height' before initializing them. */
        /* NOTE2: some codecs allow the raw parameters (frame size,
           sample rate) to be changed at any frame. We handle this, so
           you should also take care of it */
        /* here, we use a stream based decoder (mpeg1video), so we
           feed decoder and see if it could decode a frame */
        avpkt.data = inbuf;
        while (avpkt.size > 0)
            if (decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 0) < 0)
                exit(1);
    }
    /* some codecs, such as MPEG, transmit the I and P frame with a
       latency of one frame. You must do the following to have a
       chance to get the last frame of the video */
    avpkt.data = NULL;
    avpkt.size = 0;
    decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 1);
    fclose(f);
    avcodec_close(c);
    av_free(c);
    av_frame_free(&frame);
    printf("\n");
 }
 int main(int argc, char **argv)
 {
    const char *output_type;
    /* register all the codecs */
    avcodec_register_all();
    if (argc < 2) {
        printf("usage: %s output_type\n"
               "API example program to decode/encode a media stream with libavcodec.\n"
               "This program generates a synthetic stream and encodes it to a file\n"
               "named test.h264, test.mp2 or test.mpg depending on output_type.\n"
               "The encoded stream is then decoded and written to a raw data output.\n"
               "output_type must be chosen between 'h264', 'mp2', 'mpg'.\n",
               argv[0]);
        return 1;
    }
    output_type = argv[1];
    if (!strcmp(output_type, "h264")) {
        video_encode_example("test.h264", AV_CODEC_ID_H264);
    } else if (!strcmp(output_type, "mp2")) {
        audio_encode_example("test.mp2");
        audio_decode_example("test.sw", "test.mp2");
    } else if (!strcmp(output_type, "mpg")) {
        video_encode_example("test.mpg", AV_CODEC_ID_MPEG1VIDEO);
        video_decode_example("test%02d.pgm", "test.mpg");
    } else {
        fprintf(stderr, "Invalid output type '%s', choose between 'h264', 'mp2', or 'mpg'\n",
                output_type);
        return 1;
    }
    return 0;
 }
--- a/doc/examples/demuxing_decoding.c
+++ b/doc/examples/demuxing_decoding.c
@@ -32,13 +32,10 @@
 #include <libavutil/imgutils.h>
 #include <libavutil/samplefmt.h>
 #include <libavutil/timestamp.h>
 #include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
 static AVFormatContext *fmt_ctx = NULL;
 static AVCodecContext *video_dec_ctx = NULL, *audio_dec_ctx;
 static int width, height;
 static enum AVPixelFormat pix_fmt;
 static AVStream *video_stream = NULL, *audio_stream = NULL;
 static const char *src_filename = NULL;
 static const char *video_dst_filename = NULL;
@@ -52,105 +49,100 @@ static int video_dst_bufsize;
 static int video_stream_idx = -1, audio_stream_idx = -1;
 static AVFrame *frame = NULL;
-static AVPacket *pkt = NULL;
+static AVPacket pkt;
 static int video_frame_count = 0;
 static int audio_frame_count = 0;
-static int output_video_frame(AVFrame *frame)
+/* The different ways of decoding and managing data memory. You are not
-{
+ * supposed to support all the modes in your application but pick the one most
-    if (frame->width != width || frame->height != height ||
+ * appropriate to your needs. Look for the use of api_mode in this example to
-        frame->format != pix_fmt) {
+ * see what are the differences of API usage between them */
-        /* To handle this change, one could call av_image_alloc again and
+enum {
-         * decode the following frames into another rawvideo file. */
+    API_MODE_OLD                  = 0, /* old method, deprecated */
-        fprintf(stderr, "Error: Width, height and pixel format have to be "
+    API_MODE_NEW_API_REF_COUNT    = 1, /* new method, using the frame reference counting */
-                "constant in a rawvideo file, but the width, height or "
+    API_MODE_NEW_API_NO_REF_COUNT = 2, /* new method, without reference counting */
-                "pixel format of the input video changed:\n"
+};
                "old: width = %d, height = %d, format = %s\n"
                "new: width = %d, height = %d, format = %s\n",
                width, height, av_get_pix_fmt_name(pix_fmt),
                frame->width, frame->height,
                av_get_pix_fmt_name(frame->format));
        return -1;
    }
-    printf("video_frame n:%d coded_n:%d\n",
+static int api_mode = API_MODE_OLD;
           video_frame_count++, frame->coded_picture_number);
-    /* copy decoded frame to destination buffer:
+static int decode_packet(int *got_frame, int cached)
     * this is required since rawvideo expects non aligned data */
    av_image_copy(video_dst_data, video_dst_linesize,
                  (const uint8_t **)(frame->data), frame->linesize,
                  pix_fmt, width, height);
    /* write to rawvideo file */
    fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file);
    return 0;
 }
 static int output_audio_frame(AVFrame *frame)
 {
    size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format);
    printf("audio_frame n:%d nb_samples:%d pts:%s\n",
           audio_frame_count++, frame->nb_samples,
           av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));
    /* Write the raw audio data samples of the first plane. This works
     * fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However,
     * most audio decoders output planar audio, which uses a separate
     * plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P).
     * In other words, this code will write only the first audio channel
     * in these cases.
     * You should use libswresample or libavfilter to convert the frame
     * to packed data. */
    fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file);
    return 0;
 }
 static int decode_packet(AVCodecContext *dec, const AVPacket *pkt)
 {
    int ret = 0;
    int decoded = pkt.size;
-    // submit the packet to the decoder
+    *got_frame = 0;
    ret = avcodec_send_packet(dec, pkt);
    if (ret < 0) {
        fprintf(stderr, "Error submitting a packet for decoding (%s)\n", av_err2str(ret));
        return ret;
    }
-    // get all the available frames from the decoder
+    if (pkt.stream_index == video_stream_idx) {
-    while (ret >= 0) {
+        /* decode video frame */
-        ret = avcodec_receive_frame(dec, frame);
+        ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
        if (ret < 0) {
-            // those two return values are special and mean there is no output
+            fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret));
            // frame available, but there were no errors during decoding
            if (ret == AVERROR_EOF || ret == AVERROR(EAGAIN))
                return 0;
            fprintf(stderr, "Error during decoding (%s)\n", av_err2str(ret));
            return ret;
        }
-        // write the frame data to output file
+        if (*got_frame) {
-        if (dec->codec->type == AVMEDIA_TYPE_VIDEO)
+            printf("video_frame%s n:%d coded_n:%d pts:%s\n",
-            ret = output_video_frame(frame);
+                   cached ? "(cached)" : "",
-        else
+                   video_frame_count++, frame->coded_picture_number,
-            ret = output_audio_frame(frame);
+                   av_ts2timestr(frame->pts, &video_dec_ctx->time_base));
-        av_frame_unref(frame);
+            /* copy decoded frame to destination buffer:
-        if (ret < 0)
+             * this is required since rawvideo expects non aligned data */
            av_image_copy(video_dst_data, video_dst_linesize,
                          (const uint8_t **)(frame->data), frame->linesize,
                          video_dec_ctx->pix_fmt, video_dec_ctx->width, video_dec_ctx->height);
            /* write to rawvideo file */
            fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file);
        }
    } else if (pkt.stream_index == audio_stream_idx) {
        /* decode audio frame */
        ret = avcodec_decode_audio4(audio_dec_ctx, frame, got_frame, &pkt);
        if (ret < 0) {
            fprintf(stderr, "Error decoding audio frame (%s)\n", av_err2str(ret));
            return ret;
        }
        /* Some audio decoders decode only part of the packet, and have to be
         * called again with the remainder of the packet data.
         * Sample: fate-suite/lossless-audio/luckynight-partial.shn
         * Also, some decoders might over-read the packet. */
        decoded = FFMIN(ret, pkt.size);
        if (*got_frame) {
            size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format);
            printf("audio_frame%s n:%d nb_samples:%d pts:%s\n",
                   cached ? "(cached)" : "",
                   audio_frame_count++, frame->nb_samples,
                   av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));
            /* Write the raw audio data samples of the first plane. This works
             * fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However,
             * most audio decoders output planar audio, which uses a separate
             * plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P).
             * In other words, this code will write only the first audio channel
             * in these cases.
             * You should use libswresample or libavfilter to convert the frame
             * to packed data. */
            fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file);
        }
    }
-    return 0;
+    /* If we use the new API with reference counting, we own the data and need
     * to de-reference it when we don't use it anymore */
    if (*got_frame && api_mode == API_MODE_NEW_API_REF_COUNT)
        av_frame_unref(frame);
    return decoded;
 }
 static int open_codec_context(int *stream_idx,
-                              AVCodecContext **dec_ctx, AVFormatContext *fmt_ctx, enum AVMediaType type)
+                              AVFormatContext *fmt_ctx, enum AVMediaType type)
 {
-    int ret, stream_index;
+    int ret;
    AVStream *st;
-    const AVCodec *dec = NULL;
+    AVCodecContext *dec_ctx = NULL;
    AVCodec *dec = NULL;
    AVDictionary *opts = NULL;
    ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
    if (ret < 0) {
@@ -158,39 +150,26 @@ static int open_codec_context(int *stream_idx,
                av_get_media_type_string(type), src_filename);
        return ret;
    } else {
-        stream_index = ret;
+        *stream_idx = ret;
-        st = fmt_ctx->streams[stream_index];
+        st = fmt_ctx->streams[*stream_idx];
        /* find decoder for the stream */
-        dec = avcodec_find_decoder(st->codecpar->codec_id);
+        dec_ctx = st->codec;
        dec = avcodec_find_decoder(dec_ctx->codec_id);
        if (!dec) {
            fprintf(stderr, "Failed to find %s codec\n",
                    av_get_media_type_string(type));
            return AVERROR(EINVAL);
        }
-        /* Allocate a codec context for the decoder */
+        /* Init the decoders, with or without reference counting */
-        *dec_ctx = avcodec_alloc_context3(dec);
+        if (api_mode == API_MODE_NEW_API_REF_COUNT)
-        if (!*dec_ctx) {
+            av_dict_set(&opts, "refcounted_frames", "1", 0);
-            fprintf(stderr, "Failed to allocate the %s codec context\n",
+        if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) {
                    av_get_media_type_string(type));
            return AVERROR(ENOMEM);
        }
        /* Copy codec parameters from input stream to output codec context */
        if ((ret = avcodec_parameters_to_context(*dec_ctx, st->codecpar)) < 0) {
            fprintf(stderr, "Failed to copy %s codec parameters to decoder context\n",
                    av_get_media_type_string(type));
            return ret;
        }
        /* Init the decoders */
        if ((ret = avcodec_open2(*dec_ctx, dec, NULL)) < 0) {
            fprintf(stderr, "Failed to open %s codec\n",
                    av_get_media_type_string(type));
            return ret;
        }
        *stream_idx = stream_index;
    }
    return 0;
@@ -227,21 +206,40 @@ static int get_format_from_sample_fmt(const char **fmt,
 int main (int argc, char **argv)
 {
-    int ret = 0;
+    int ret = 0, got_frame;
-    if (argc != 4) {
+    if (argc != 4 && argc != 5) {
-        fprintf(stderr, "usage: %s  input_file video_output_file audio_output_file\n"
+        fprintf(stderr, "usage: %s [-refcount=<old|new_norefcount|new_refcount>] "
                "input_file video_output_file audio_output_file\n"
                "API example program to show how to read frames from an input file.\n"
                "This program reads frames from a file, decodes them, and writes decoded\n"
                "video frames to a rawvideo file named video_output_file, and decoded\n"
-                "audio frames to a rawaudio file named audio_output_file.\n",
+                "audio frames to a rawaudio file named audio_output_file.\n\n"
-                argv[0]);
+                "If the -refcount option is specified, the program use the\n"
                "reference counting frame system which allows keeping a copy of\n"
                "the data for longer than one decode call. If unset, it's using\n"
                "the classic old method.\n"
                "\n", argv[0]);
        exit(1);
    }
    if (argc == 5) {
        const char *mode = argv[1] + strlen("-refcount=");
        if      (!strcmp(mode, "old"))            api_mode = API_MODE_OLD;
        else if (!strcmp(mode, "new_norefcount")) api_mode = API_MODE_NEW_API_NO_REF_COUNT;
        else if (!strcmp(mode, "new_refcount"))   api_mode = API_MODE_NEW_API_REF_COUNT;
        else {
            fprintf(stderr, "unknow mode '%s'\n", mode);
            exit(1);
        }
        argv++;
    }
    src_filename = argv[1];
    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);
@@ -254,8 +252,9 @@ int main (int argc, char **argv)
        exit(1);
    }
-    if (open_codec_context(&video_stream_idx, &video_dec_ctx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) {
+    if (open_codec_context(&video_stream_idx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) {
        video_stream = fmt_ctx->streams[video_stream_idx];
        video_dec_ctx = video_stream->codec;
        video_dst_file = fopen(video_dst_filename, "wb");
        if (!video_dst_file) {
@@ -265,11 +264,9 @@ int main (int argc, char **argv)
        }
        /* allocate image where the decoded image will be put */
        width = video_dec_ctx->width;
        height = video_dec_ctx->height;
        pix_fmt = video_dec_ctx->pix_fmt;
        ret = av_image_alloc(video_dst_data, video_dst_linesize,
-                             width, height, pix_fmt, 1);
+                             video_dec_ctx->width, video_dec_ctx->height,
                             video_dec_ctx->pix_fmt, 1);
        if (ret < 0) {
            fprintf(stderr, "Could not allocate raw video buffer\n");
            goto end;
@@ -277,8 +274,9 @@ int main (int argc, char **argv)
        video_dst_bufsize = ret;
    }
-    if (open_codec_context(&audio_stream_idx, &audio_dec_ctx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) {
+    if (open_codec_context(&audio_stream_idx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) {
        audio_stream = fmt_ctx->streams[audio_stream_idx];
        audio_dec_ctx = audio_stream->codec;
        audio_dst_file = fopen(audio_dst_filename, "wb");
        if (!audio_dst_file) {
            fprintf(stderr, "Could not open destination file %s\n", audio_dst_filename);
@@ -296,19 +294,22 @@ int main (int argc, char **argv)
        goto end;
    }
-    frame = av_frame_alloc();
+    /* When using the new API, you need to use the libavutil/frame.h API, while
     * the classic frame management is available in libavcodec */
    if (api_mode == API_MODE_OLD)
        frame = avcodec_alloc_frame();
    else
        frame = av_frame_alloc();
    if (!frame) {
        fprintf(stderr, "Could not allocate frame\n");
        ret = AVERROR(ENOMEM);
        goto end;
    }
-    pkt = av_packet_alloc();
+    /* initialize packet, set data to NULL, let the demuxer fill it */
-    if (!pkt) {
+    av_init_packet(&pkt);
-        fprintf(stderr, "Could not allocate packet\n");
+    pkt.data = NULL;
-        ret = AVERROR(ENOMEM);
+    pkt.size = 0;
        goto end;
    }
    if (video_stream)
        printf("Demuxing video from file '%s' into '%s'\n", src_filename, video_dst_filename);
@@ -316,36 +317,37 @@ int main (int argc, char **argv)
        printf("Demuxing audio from file '%s' into '%s'\n", src_filename, audio_dst_filename);
    /* read frames from the file */
-    while (av_read_frame(fmt_ctx, pkt) >= 0) {
+    while (av_read_frame(fmt_ctx, &pkt) >= 0) {
-        // check if the packet belongs to a stream we are interested in, otherwise
+        AVPacket orig_pkt = pkt;
-        // skip it
+        do {
-        if (pkt->stream_index == video_stream_idx)
+            ret = decode_packet(&got_frame, 0);
-            ret = decode_packet(video_dec_ctx, pkt);
+            if (ret < 0)
-        else if (pkt->stream_index == audio_stream_idx)
+                break;
-            ret = decode_packet(audio_dec_ctx, pkt);
+            pkt.data += ret;
-        av_packet_unref(pkt);
+            pkt.size -= ret;
-        if (ret < 0)
+        } while (pkt.size > 0);
-            break;
+        av_free_packet(&orig_pkt);
    }
-    /* flush the decoders */
+    /* flush cached frames */
-    if (video_dec_ctx)
+    pkt.data = NULL;
-        decode_packet(video_dec_ctx, NULL);
+    pkt.size = 0;
-    if (audio_dec_ctx)
+    do {
-        decode_packet(audio_dec_ctx, NULL);
+        decode_packet(&got_frame, 1);
    } while (got_frame);
    printf("Demuxing succeeded.\n");
    if (video_stream) {
        printf("Play the output video file with the command:\n"
               "ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
-               av_get_pix_fmt_name(pix_fmt), width, height,
+               av_get_pix_fmt_name(video_dec_ctx->pix_fmt), video_dec_ctx->width, video_dec_ctx->height,
               video_dst_filename);
    }
    if (audio_stream) {
        enum AVSampleFormat sfmt = audio_dec_ctx->sample_fmt;
-        int n_channels = audio_dec_ctx->ch_layout.nb_channels;
+        int n_channels = audio_dec_ctx->channels;
        const char *fmt;
        if (av_sample_fmt_is_planar(sfmt)) {
@@ -367,15 +369,17 @@ int main (int argc, char **argv)
    }
 end:
-    avcodec_free_context(&video_dec_ctx);
+    avcodec_close(video_dec_ctx);
-    avcodec_free_context(&audio_dec_ctx);
+    avcodec_close(audio_dec_ctx);
    avformat_close_input(&fmt_ctx);
    if (video_dst_file)
        fclose(video_dst_file);
    if (audio_dst_file)
        fclose(audio_dst_file);
-    av_packet_free(&pkt);
+    if (api_mode == API_MODE_OLD)
-    av_frame_free(&frame);
+        avcodec_free_frame(&frame);
    else
        av_frame_free(&frame);
    av_free(video_dst_data[0]);
    return ret < 0;
--- a/doc/examples/encode_audio.c
+++ b/doc/examples/encode_audio.c
@@ -1,240 +0,0 @@
 /*
 * Copyright (c) 2001 Fabrice Bellard
 *
 * 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
 * audio encoding with libavcodec API example.
 *
 * @example encode_audio.c
 */
 #include <stdint.h>
 #include <stdio.h>
 #include <stdlib.h>
 #include <libavcodec/avcodec.h>
 #include <libavutil/channel_layout.h>
 #include <libavutil/common.h>
 #include <libavutil/frame.h>
 #include <libavutil/samplefmt.h>
 /* check that a given sample format is supported by the encoder */
 static int check_sample_fmt(const AVCodec *codec, enum AVSampleFormat sample_fmt)
 {
    const enum AVSampleFormat *p = codec->sample_fmts;
    while (*p != AV_SAMPLE_FMT_NONE) {
        if (*p == sample_fmt)
            return 1;
        p++;
    }
    return 0;
 }
 /* just pick the highest supported samplerate */
 static int select_sample_rate(const AVCodec *codec)
 {
    const int *p;
    int best_samplerate = 0;
    if (!codec->supported_samplerates)
        return 44100;
    p = codec->supported_samplerates;
    while (*p) {
        if (!best_samplerate || abs(44100 - *p) < abs(44100 - best_samplerate))
            best_samplerate = *p;
        p++;
    }
    return best_samplerate;
 }
 /* select layout with the highest channel count */
 static int select_channel_layout(const AVCodec *codec, AVChannelLayout *dst)
 {
    const AVChannelLayout *p, *best_ch_layout;
    int best_nb_channels   = 0;
    if (!codec->ch_layouts)
        return av_channel_layout_copy(dst, &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO);
    p = codec->ch_layouts;
    while (p->nb_channels) {
        int nb_channels = p->nb_channels;
        if (nb_channels > best_nb_channels) {
            best_ch_layout   = p;
            best_nb_channels = nb_channels;
        }
        p++;
    }
    return av_channel_layout_copy(dst, best_ch_layout);
 }
 static void encode(AVCodecContext *ctx, AVFrame *frame, AVPacket *pkt,
                   FILE *output)
 {
    int ret;
    /* send the frame for encoding */
    ret = avcodec_send_frame(ctx, frame);
    if (ret < 0) {
        fprintf(stderr, "Error sending the frame to the encoder\n");
        exit(1);
    }
    /* read all the available output packets (in general there may be any
     * number of them */
    while (ret >= 0) {
        ret = avcodec_receive_packet(ctx, pkt);
        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
            return;
        else if (ret < 0) {
            fprintf(stderr, "Error encoding audio frame\n");
            exit(1);
        }
        fwrite(pkt->data, 1, pkt->size, output);
        av_packet_unref(pkt);
    }
 }
 int main(int argc, char **argv)
 {
    const char *filename;
    const AVCodec *codec;
    AVCodecContext *c= NULL;
    AVFrame *frame;
    AVPacket *pkt;
    int i, j, k, ret;
    FILE *f;
    uint16_t *samples;
    float t, tincr;
    if (argc <= 1) {
        fprintf(stderr, "Usage: %s <output file>\n", argv[0]);
        return 0;
    }
    filename = argv[1];
    /* find the MP2 encoder */
    codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
    if (!codec) {
        fprintf(stderr, "Codec not found\n");
        exit(1);
    }
    c = avcodec_alloc_context3(codec);
    if (!c) {
        fprintf(stderr, "Could not allocate audio codec context\n");
        exit(1);
    }
    /* put sample parameters */
    c->bit_rate = 64000;
    /* check that the encoder supports s16 pcm input */
    c->sample_fmt = AV_SAMPLE_FMT_S16;
    if (!check_sample_fmt(codec, c->sample_fmt)) {
        fprintf(stderr, "Encoder does not support sample format %s",
                av_get_sample_fmt_name(c->sample_fmt));
        exit(1);
    }
    /* select other audio parameters supported by the encoder */
    c->sample_rate    = select_sample_rate(codec);
    ret = select_channel_layout(codec, &c->ch_layout);
    if (ret < 0)
        exit(1);
    /* open it */
    if (avcodec_open2(c, codec, NULL) < 0) {
        fprintf(stderr, "Could not open codec\n");
        exit(1);
    }
    f = fopen(filename, "wb");
    if (!f) {
        fprintf(stderr, "Could not open %s\n", filename);
        exit(1);
    }
    /* packet for holding encoded output */
    pkt = av_packet_alloc();
    if (!pkt) {
        fprintf(stderr, "could not allocate the packet\n");
        exit(1);
    }
    /* frame containing input raw audio */
    frame = av_frame_alloc();
    if (!frame) {
        fprintf(stderr, "Could not allocate audio frame\n");
        exit(1);
    }
    frame->nb_samples     = c->frame_size;
    frame->format         = c->sample_fmt;
    ret = av_channel_layout_copy(&frame->ch_layout, &c->ch_layout);
    if (ret < 0)
        exit(1);
    /* allocate the data buffers */
    ret = av_frame_get_buffer(frame, 0);
    if (ret < 0) {
        fprintf(stderr, "Could not allocate audio data buffers\n");
        exit(1);
    }
    /* encode a single tone sound */
    t = 0;
    tincr = 2 * M_PI * 440.0 / c->sample_rate;
    for (i = 0; i < 200; i++) {
        /* make sure the frame is writable -- makes a copy if the encoder
         * kept a reference internally */
        ret = av_frame_make_writable(frame);
        if (ret < 0)
            exit(1);
        samples = (uint16_t*)frame->data[0];
        for (j = 0; j < c->frame_size; j++) {
            samples[2*j] = (int)(sin(t) * 10000);
            for (k = 1; k < c->ch_layout.nb_channels; k++)
                samples[2*j + k] = samples[2*j];
            t += tincr;
        }
        encode(c, frame, pkt, f);
    }
    /* flush the encoder */
    encode(c, NULL, pkt, f);
    fclose(f);
    av_frame_free(&frame);
    av_packet_free(&pkt);
    avcodec_free_context(&c);
    return 0;
 }
--- a/doc/examples/encode_video.c
+++ b/doc/examples/encode_video.c
@@ -1,216 +0,0 @@
 /*
 * Copyright (c) 2001 Fabrice Bellard
 *
 * 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
 * video encoding with libavcodec API example
 *
 * @example encode_video.c
 */
 #include <stdio.h>
 #include <stdlib.h>
 #include <string.h>
 #include <libavcodec/avcodec.h>
 #include <libavutil/opt.h>
 #include <libavutil/imgutils.h>
 static void encode(AVCodecContext *enc_ctx, AVFrame *frame, AVPacket *pkt,
                   FILE *outfile)
 {
    int ret;
    /* send the frame to the encoder */
    if (frame)
        printf("Send frame %3"PRId64"\n", frame->pts);
    ret = avcodec_send_frame(enc_ctx, frame);
    if (ret < 0) {
        fprintf(stderr, "Error sending a frame for encoding\n");
        exit(1);
    }
    while (ret >= 0) {
        ret = avcodec_receive_packet(enc_ctx, pkt);
        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
            return;
        else if (ret < 0) {
            fprintf(stderr, "Error during encoding\n");
            exit(1);
        }
        printf("Write packet %3"PRId64" (size=%5d)\n", pkt->pts, pkt->size);
        fwrite(pkt->data, 1, pkt->size, outfile);
        av_packet_unref(pkt);
    }
 }
 int main(int argc, char **argv)
 {
    const char *filename, *codec_name;
    const AVCodec *codec;
    AVCodecContext *c= NULL;
    int i, ret, x, y;
    FILE *f;
    AVFrame *frame;
    AVPacket *pkt;
    uint8_t endcode[] = { 0, 0, 1, 0xb7 };
    if (argc <= 2) {
        fprintf(stderr, "Usage: %s <output file> <codec name>\n", argv[0]);
        exit(0);
    }
    filename = argv[1];
    codec_name = argv[2];
    /* find the mpeg1video encoder */
    codec = avcodec_find_encoder_by_name(codec_name);
    if (!codec) {
        fprintf(stderr, "Codec '%s' not found\n", codec_name);
        exit(1);
    }
    c = avcodec_alloc_context3(codec);
    if (!c) {
        fprintf(stderr, "Could not allocate video codec context\n");
        exit(1);
    }
    pkt = av_packet_alloc();
    if (!pkt)
        exit(1);
    /* put sample parameters */
    c->bit_rate = 400000;
    /* resolution must be a multiple of two */
    c->width = 352;
    c->height = 288;
    /* frames per second */
    c->time_base = (AVRational){1, 25};
    c->framerate = (AVRational){25, 1};
    /* emit one intra frame every ten frames
     * check frame pict_type before passing frame
     * to encoder, if frame->pict_type is AV_PICTURE_TYPE_I
     * then gop_size is ignored and the output of encoder
     * will always be I frame irrespective to gop_size
     */
    c->gop_size = 10;
    c->max_b_frames = 1;
    c->pix_fmt = AV_PIX_FMT_YUV420P;
    if (codec->id == AV_CODEC_ID_H264)
        av_opt_set(c->priv_data, "preset", "slow", 0);
    /* open it */
    ret = avcodec_open2(c, codec, NULL);
    if (ret < 0) {
        fprintf(stderr, "Could not open codec: %s\n", av_err2str(ret));
        exit(1);
    }
    f = fopen(filename, "wb");
    if (!f) {
        fprintf(stderr, "Could not open %s\n", filename);
        exit(1);
    }
    frame = av_frame_alloc();
    if (!frame) {
        fprintf(stderr, "Could not allocate video frame\n");
        exit(1);
    }
    frame->format = c->pix_fmt;
    frame->width  = c->width;
    frame->height = c->height;
    ret = av_frame_get_buffer(frame, 0);
    if (ret < 0) {
        fprintf(stderr, "Could not allocate the video frame data\n");
        exit(1);
    }
    /* encode 1 second of video */
    for (i = 0; i < 25; i++) {
        fflush(stdout);
        /* Make sure the frame data is writable.
           On the first round, the frame is fresh from av_frame_get_buffer()
           and therefore we know it is writable.
           But on the next rounds, encode() will have called
           avcodec_send_frame(), and the codec may have kept a reference to
           the frame in its internal structures, that makes the frame
           unwritable.
           av_frame_make_writable() checks that and allocates a new buffer
           for the frame only if necessary.
         */
        ret = av_frame_make_writable(frame);
        if (ret < 0)
            exit(1);
        /* Prepare a dummy image.
           In real code, this is where you would have your own logic for
           filling the frame. FFmpeg does not care what you put in the
           frame.
         */
        /* Y */
        for (y = 0; y < c->height; y++) {
            for (x = 0; x < c->width; x++) {
                frame->data[0][y * frame->linesize[0] + x] = x + y + i * 3;
            }
        }
        /* Cb and Cr */
        for (y = 0; y < c->height/2; y++) {
            for (x = 0; x < c->width/2; x++) {
                frame->data[1][y * frame->linesize[1] + x] = 128 + y + i * 2;
                frame->data[2][y * frame->linesize[2] + x] = 64 + x + i * 5;
            }
        }
        frame->pts = i;
        /* encode the image */
        encode(c, frame, pkt, f);
    }
    /* flush the encoder */
    encode(c, NULL, pkt, f);
    /* Add sequence end code to have a real MPEG file.
       It makes only sense because this tiny examples writes packets
       directly. This is called "elementary stream" and only works for some
       codecs. To create a valid file, you usually need to write packets
       into a proper file format or protocol; see muxing.c.
     */
    if (codec->id == AV_CODEC_ID_MPEG1VIDEO || codec->id == AV_CODEC_ID_MPEG2VIDEO)
        fwrite(endcode, 1, sizeof(endcode), f);
    fclose(f);
    avcodec_free_context(&c);
    av_frame_free(&frame);
    av_packet_free(&pkt);
    return 0;
 }
--- a/doc/examples/extract_mvs.c
+++ b/doc/examples/extract_mvs.c
@@ -1,189 +0,0 @@
 /*
 * Copyright (c) 2012 Stefano Sabatini
 * Copyright (c) 2014 Clément Bœsch
 *
 * Permission is hereby granted, free of charge, to any person obtaining a copy
 * of this software and associated documentation files (the "Software"), to deal
 * in the Software without restriction, including without limitation the rights
 * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
 * copies of the Software, and to permit persons to whom the Software is
 * furnished to do so, subject to the following conditions:
 *
 * The above copyright notice and this permission notice shall be included in
 * all copies or substantial portions of the Software.
 *
 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
 * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
 * THE SOFTWARE.
 */
 #include <libavutil/motion_vector.h>
 #include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
 static AVFormatContext *fmt_ctx = NULL;
 static AVCodecContext *video_dec_ctx = NULL;
 static AVStream *video_stream = NULL;
 static const char *src_filename = NULL;
 static int video_stream_idx = -1;
 static AVFrame *frame = NULL;
 static int video_frame_count = 0;
 static int decode_packet(const AVPacket *pkt)
 {
    int ret = avcodec_send_packet(video_dec_ctx, pkt);
    if (ret < 0) {
        fprintf(stderr, "Error while sending a packet to the decoder: %s\n", av_err2str(ret));
        return ret;
    }
    while (ret >= 0)  {
        ret = avcodec_receive_frame(video_dec_ctx, frame);
        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
            break;
        } else if (ret < 0) {
            fprintf(stderr, "Error while receiving a frame from the decoder: %s\n", av_err2str(ret));
            return ret;
        }
        if (ret >= 0) {
            int i;
            AVFrameSideData *sd;
            video_frame_count++;
            sd = av_frame_get_side_data(frame, AV_FRAME_DATA_MOTION_VECTORS);
            if (sd) {
                const AVMotionVector *mvs = (const AVMotionVector *)sd->data;
                for (i = 0; i < sd->size / sizeof(*mvs); i++) {
                    const AVMotionVector *mv = &mvs[i];
                    printf("%d,%2d,%2d,%2d,%4d,%4d,%4d,%4d,0x%"PRIx64"\n",
                        video_frame_count, mv->source,
                        mv->w, mv->h, mv->src_x, mv->src_y,
                        mv->dst_x, mv->dst_y, mv->flags);
                }
            }
            av_frame_unref(frame);
        }
    }
    return 0;
 }
 static int open_codec_context(AVFormatContext *fmt_ctx, enum AVMediaType type)
 {
    int ret;
    AVStream *st;
    AVCodecContext *dec_ctx = NULL;
    const AVCodec *dec = NULL;
    AVDictionary *opts = NULL;
    ret = av_find_best_stream(fmt_ctx, type, -1, -1, &dec, 0);
    if (ret < 0) {
        fprintf(stderr, "Could not find %s stream in input file '%s'\n",
                av_get_media_type_string(type), src_filename);
        return ret;
    } else {
        int stream_idx = ret;
        st = fmt_ctx->streams[stream_idx];
        dec_ctx = avcodec_alloc_context3(dec);
        if (!dec_ctx) {
            fprintf(stderr, "Failed to allocate codec\n");
            return AVERROR(EINVAL);
        }
        ret = avcodec_parameters_to_context(dec_ctx, st->codecpar);
        if (ret < 0) {
            fprintf(stderr, "Failed to copy codec parameters to codec context\n");
            return ret;
        }
        /* Init the video decoder */
        av_dict_set(&opts, "flags2", "+export_mvs", 0);
        ret = avcodec_open2(dec_ctx, dec, &opts);
        av_dict_free(&opts);
        if (ret < 0) {
            fprintf(stderr, "Failed to open %s codec\n",
                    av_get_media_type_string(type));
            return ret;
        }
        video_stream_idx = stream_idx;
        video_stream = fmt_ctx->streams[video_stream_idx];
        video_dec_ctx = dec_ctx;
    }
    return 0;
 }
 int main(int argc, char **argv)
 {
    int ret = 0;
    AVPacket *pkt = NULL;
    if (argc != 2) {
        fprintf(stderr, "Usage: %s <video>\n", argv[0]);
        exit(1);
    }
    src_filename = argv[1];
    if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
        fprintf(stderr, "Could not open source file %s\n", src_filename);
        exit(1);
    }
    if (avformat_find_stream_info(fmt_ctx, NULL) < 0) {
        fprintf(stderr, "Could not find stream information\n");
        exit(1);
    }
    open_codec_context(fmt_ctx, AVMEDIA_TYPE_VIDEO);
    av_dump_format(fmt_ctx, 0, src_filename, 0);
    if (!video_stream) {
        fprintf(stderr, "Could not find video stream in the input, aborting\n");
        ret = 1;
        goto end;
    }
    frame = av_frame_alloc();
    if (!frame) {
        fprintf(stderr, "Could not allocate frame\n");
        ret = AVERROR(ENOMEM);
        goto end;
    }
    pkt = av_packet_alloc();
    if (!pkt) {
        fprintf(stderr, "Could not allocate AVPacket\n");
        ret = AVERROR(ENOMEM);
        goto end;
    }
    printf("framenum,source,blockw,blockh,srcx,srcy,dstx,dsty,flags\n");
    /* read frames from the file */
    while (av_read_frame(fmt_ctx, pkt) >= 0) {
        if (pkt->stream_index == video_stream_idx)
            ret = decode_packet(pkt);
        av_packet_unref(pkt);
        if (ret < 0)
            break;
    }
    /* flush cached frames */
    decode_packet(NULL);
 end:
    avcodec_free_context(&video_dec_ctx);
    avformat_close_input(&fmt_ctx);
    av_frame_free(&frame);
    av_packet_free(&pkt);
    return ret < 0;
 }
--- a/doc/examples/filter_audio.c
+++ b/doc/examples/filter_audio.c
@@ -45,7 +45,6 @@
 #include "libavutil/channel_layout.h"
 #include "libavutil/md5.h"
 #include "libavutil/mem.h"
 #include "libavutil/opt.h"
 #include "libavutil/samplefmt.h"
@@ -55,7 +54,7 @@
 #define INPUT_SAMPLERATE     48000
 #define INPUT_FORMAT         AV_SAMPLE_FMT_FLTP
-#define INPUT_CHANNEL_LAYOUT (AVChannelLayout)AV_CHANNEL_LAYOUT_5POINT0
+#define INPUT_CHANNEL_LAYOUT AV_CH_LAYOUT_5POINT0
 #define VOLUME_VAL 0.90
@@ -64,13 +63,13 @@ 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];
@@ -100,7 +99,7 @@ static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
    }
    /* Set the filter options through the AVOptions API. */
-    av_channel_layout_describe(&INPUT_CHANNEL_LAYOUT, ch_layout, sizeof(ch_layout));
+    av_get_channel_layout_string(ch_layout, sizeof(ch_layout), 0, INPUT_CHANNEL_LAYOUT);
    av_opt_set    (abuffer_ctx, "channel_layout", ch_layout,                            AV_OPT_SEARCH_CHILDREN);
    av_opt_set    (abuffer_ctx, "sample_fmt",     av_get_sample_fmt_name(INPUT_FORMAT), AV_OPT_SEARCH_CHILDREN);
    av_opt_set_q  (abuffer_ctx, "time_base",      (AVRational){ 1, INPUT_SAMPLERATE },  AV_OPT_SEARCH_CHILDREN);
@@ -154,8 +153,9 @@ static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
    /* A third way of passing the options is in a string of the form
     * key1=value1:key2=value2.... */
    snprintf(options_str, sizeof(options_str),
-             "sample_fmts=%s:sample_rates=%d:channel_layouts=stereo",
+             "sample_fmts=%s:sample_rates=%d:channel_layouts=0x%"PRIx64,
-             av_get_sample_fmt_name(AV_SAMPLE_FMT_S16), 44100);
+             av_get_sample_fmt_name(AV_SAMPLE_FMT_S16), 44100,
             (uint64_t)AV_CH_LAYOUT_STEREO);
    err = avfilter_init_str(aformat_ctx, options_str);
    if (err < 0) {
        av_log(NULL, AV_LOG_ERROR, "Could not initialize the aformat filter.\n");
@@ -214,7 +214,7 @@ static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
 static int process_output(struct AVMD5 *md5, AVFrame *frame)
 {
    int planar     = av_sample_fmt_is_planar(frame->format);
-    int channels   = frame->ch_layout.nb_channels;
+    int channels   = av_get_channel_layout_nb_channels(frame->channel_layout);
    int planes     = planar ? channels : 1;
    int bps        = av_get_bytes_per_sample(frame->format);
    int plane_size = bps * frame->nb_samples * (planar ? 1 : channels);
@@ -247,7 +247,7 @@ static int get_input(AVFrame *frame, int frame_num)
    /* Set up the frame properties and allocate the buffer for the data. */
    frame->sample_rate    = INPUT_SAMPLERATE;
    frame->format         = INPUT_FORMAT;
-    av_channel_layout_copy(&frame->ch_layout, &INPUT_CHANNEL_LAYOUT);
+    frame->channel_layout = INPUT_CHANNEL_LAYOUT;
    frame->nb_samples     = FRAME_SIZE;
    frame->pts            = frame_num * FRAME_SIZE;
@@ -288,6 +288,8 @@ 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) {
--- a/doc/examples/filtering_audio.c
+++ b/doc/examples/filtering_audio.c
@@ -32,9 +32,10 @@
 #include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
 #include <libavfilter/avfiltergraph.h>
 #include <libavfilter/avcodec.h>
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
 #include <libavutil/channel_layout.h>
 #include <libavutil/opt.h>
 static const char *filter_descr = "aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono";
@@ -49,8 +50,8 @@ static int audio_stream_index = -1;
 static int open_input_file(const char *filename)
 {
    const AVCodec *dec;
    int ret;
    AVCodec *dec;
    if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
@@ -65,16 +66,12 @@ static int open_input_file(const char *filename)
    /* select the audio stream */
    ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);
    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot find an audio stream in the input file\n");
+        av_log(NULL, AV_LOG_ERROR, "Cannot find a audio stream in the input file\n");
        return ret;
    }
    audio_stream_index = ret;
-
+    dec_ctx = fmt_ctx->streams[audio_stream_index]->codec;
-    /* create decoding context */
+    av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
    dec_ctx = avcodec_alloc_context3(dec);
    if (!dec_ctx)
        return AVERROR(ENOMEM);
    avcodec_parameters_to_context(dec_ctx, fmt_ctx->streams[audio_stream_index]->codecpar);
    /* init the audio decoder */
    if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
@@ -89,11 +86,12 @@ static int init_filters(const char *filters_descr)
 {
    char args[512];
    int ret = 0;
-    const AVFilter *abuffersrc  = avfilter_get_by_name("abuffer");
+    AVFilter *abuffersrc  = avfilter_get_by_name("abuffer");
-    const AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
+    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 };
    static const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 };
    static const int out_sample_rates[] = { 8000, -1 };
    const AVFilterLink *outlink;
    AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;
@@ -105,13 +103,12 @@ static int init_filters(const char *filters_descr)
    }
    /* buffer audio source: the decoded frames from the decoder will be inserted here. */
-    if (dec_ctx->ch_layout.order == AV_CHANNEL_ORDER_UNSPEC)
+    if (!dec_ctx->channel_layout)
-        av_channel_layout_default(&dec_ctx->ch_layout, dec_ctx->ch_layout.nb_channels);
+        dec_ctx->channel_layout = av_get_default_channel_layout(dec_ctx->channels);
-    ret = snprintf(args, sizeof(args),
+    snprintf(args, sizeof(args),
-            "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=",
+            "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
             time_base.num, time_base.den, dec_ctx->sample_rate,
-             av_get_sample_fmt_name(dec_ctx->sample_fmt));
+             av_get_sample_fmt_name(dec_ctx->sample_fmt), dec_ctx->channel_layout);
    av_channel_layout_describe(&dec_ctx->ch_layout, args + ret, sizeof(args) - ret);
    ret = avfilter_graph_create_filter(&buffersrc_ctx, abuffersrc, "in",
                                       args, NULL, filter_graph);
    if (ret < 0) {
@@ -134,7 +131,7 @@ static int init_filters(const char *filters_descr)
        goto end;
    }
-    ret = av_opt_set(buffersink_ctx, "ch_layouts", "mono",
+    ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts", out_channel_layouts, -1,
                              AV_OPT_SEARCH_CHILDREN);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
@@ -148,28 +145,12 @@ static int init_filters(const char *filters_descr)
        goto end;
    }
-    /*
+    /* Endpoints for the filter graph. */
     * Set the endpoints for the filter graph. The filter_graph will
     * be linked to the graph described by filters_descr.
     */
    /*
     * The buffer source output must be connected to the input pad of
     * the first filter described by filters_descr; since the first
     * filter input label is not specified, it is set to "in" by
     * default.
     */
    outputs->name       = av_strdup("in");
    outputs->filter_ctx = buffersrc_ctx;
    outputs->pad_idx    = 0;
    outputs->next       = NULL;
    /*
     * The buffer sink input must be connected to the output pad of
     * the last filter described by filters_descr; since the last
     * filter output label is not specified, it is set to "out" by
     * default.
     */
    inputs->name       = av_strdup("out");
    inputs->filter_ctx = buffersink_ctx;
    inputs->pad_idx    = 0;
@@ -185,7 +166,7 @@ static int init_filters(const char *filters_descr)
    /* Print summary of the sink buffer
     * Note: args buffer is reused to store channel layout string */
    outlink = buffersink_ctx->inputs[0];
-    av_channel_layout_describe(&outlink->ch_layout, args, sizeof(args));
+    av_get_channel_layout_string(args, sizeof(args), -1, outlink->channel_layout);
    av_log(NULL, AV_LOG_INFO, "Output: srate:%dHz fmt:%s chlayout:%s\n",
           (int)outlink->sample_rate,
           (char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
@@ -200,7 +181,7 @@ end:
 static void print_frame(const AVFrame *frame)
 {
-    const int n = frame->nb_samples * frame->ch_layout.nb_channels;
+    const int n = frame->nb_samples * av_get_channel_layout_nb_channels(av_frame_get_channel_layout(frame));
    const uint16_t *p     = (uint16_t*)frame->data[0];
    const uint16_t *p_end = p + n;
@@ -215,12 +196,13 @@ static void print_frame(const AVFrame *frame)
 int main(int argc, char **argv)
 {
    int ret;
-    AVPacket *packet = av_packet_alloc();
+    AVPacket packet0, packet;
    AVFrame *frame = av_frame_alloc();
    AVFrame *filt_frame = av_frame_alloc();
    int got_frame;
-    if (!packet || !frame || !filt_frame) {
+    if (!frame || !filt_frame) {
-        fprintf(stderr, "Could not allocate frame or packet\n");
+        perror("Could not allocate frame");
        exit(1);
    }
    if (argc != 2) {
@@ -228,60 +210,64 @@ 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)
        goto end;
    /* read all packets */
    packet0.data = NULL;
    packet.data = NULL;
    while (1) {
-        if ((ret = av_read_frame(fmt_ctx, packet)) < 0)
+        if (!packet0.data) {
-            break;
+            if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
        if (packet->stream_index == audio_stream_index) {
            ret = avcodec_send_packet(dec_ctx, packet);
            if (ret < 0) {
                av_log(NULL, AV_LOG_ERROR, "Error while sending a packet to the decoder\n");
                break;
-            }
+            packet0 = packet;
-
+        }
-            while (ret >= 0) {
+
-                ret = avcodec_receive_frame(dec_ctx, frame);
+        if (packet.stream_index == audio_stream_index) {
-                if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
+            got_frame = 0;
-                    break;
+            ret = avcodec_decode_audio4(dec_ctx, frame, &got_frame, &packet);
-                } else if (ret < 0) {
+            if (ret < 0) {
-                    av_log(NULL, AV_LOG_ERROR, "Error while receiving a frame from the decoder\n");
+                av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n");
-                    goto end;
+                continue;
-                }
+            }
-
+            packet.size -= ret;
-                if (ret >= 0) {
+            packet.data += ret;
-                    /* push the audio data from decoded frame into the filtergraph */
+
-                    if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
+            if (got_frame) {
-                        av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
+                /* push the audio data from decoded frame into the filtergraph */
-                        break;
+                if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, 0) < 0) {
-                    }
+                    av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
-
+                    break;
-                    /* pull filtered audio from the filtergraph */
+                }
-                    while (1) {
+
-                        ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
+                /* pull filtered audio from the filtergraph */
-                        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
+                while (1) {
-                            break;
+                    ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
-                        if (ret < 0)
+                    if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
-                            goto end;
+                        break;
-                        print_frame(filt_frame);
+                    if (ret < 0)
-                        av_frame_unref(filt_frame);
+                        goto end;
-                    }
+                    print_frame(filt_frame);
-                    av_frame_unref(frame);
+                    av_frame_unref(filt_frame);
-                }
+                }
-            }
+            }
            if (packet.size <= 0)
                av_free_packet(&packet0);
        } else {
            /* discard non-wanted packets */
            av_free_packet(&packet0);
        }
        av_packet_unref(packet);
    }
 end:
    avfilter_graph_free(&filter_graph);
-    avcodec_free_context(&dec_ctx);
+    avcodec_close(dec_ctx);
    avformat_close_input(&fmt_ctx);
    av_packet_free(&packet);
    av_frame_free(&frame);
    av_frame_free(&filt_frame);
--- a/doc/examples/filtering_video.c
+++ b/doc/examples/filtering_video.c
@@ -29,19 +29,16 @@
 #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/avcodec.h>
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
 #include <libavutil/opt.h>
-const char *filter_descr = "scale=78:24,transpose=cclock";
+const char *filter_descr = "scale=78:24";
 /* other way:
   scale=78:24 [scl]; [scl] transpose=cclock // assumes "[in]" and "[out]" to be input output pads respectively
 */
 static AVFormatContext *fmt_ctx;
 static AVCodecContext *dec_ctx;
@@ -53,8 +50,8 @@ static int64_t last_pts = AV_NOPTS_VALUE;
 static int open_input_file(const char *filename)
 {
    const AVCodec *dec;
    int ret;
    AVCodec *dec;
    if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
@@ -73,12 +70,8 @@ static int open_input_file(const char *filename)
        return ret;
    }
    video_stream_index = ret;
-
+    dec_ctx = fmt_ctx->streams[video_stream_index]->codec;
-    /* create decoding context */
+    av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
    dec_ctx = avcodec_alloc_context3(dec);
    if (!dec_ctx)
        return AVERROR(ENOMEM);
    avcodec_parameters_to_context(dec_ctx, fmt_ctx->streams[video_stream_index]->codecpar);
    /* init the video decoder */
    if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
@@ -93,11 +86,10 @@ static int init_filters(const char *filters_descr)
 {
    char args[512];
    int ret = 0;
-    const AVFilter *buffersrc  = avfilter_get_by_name("buffer");
+    AVFilter *buffersrc  = avfilter_get_by_name("buffer");
-    const AVFilter *buffersink = avfilter_get_by_name("buffersink");
+    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;
    enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE };
    filter_graph = avfilter_graph_alloc();
@@ -110,7 +102,7 @@ static int init_filters(const char *filters_descr)
    snprintf(args, sizeof(args),
            "video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
            dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
-            time_base.num, time_base.den,
+            dec_ctx->time_base.num, dec_ctx->time_base.den,
            dec_ctx->sample_aspect_ratio.num, dec_ctx->sample_aspect_ratio.den);
    ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
@@ -135,28 +127,12 @@ static int init_filters(const char *filters_descr)
        goto end;
    }
-    /*
+    /* Endpoints for the filter graph. */
     * Set the endpoints for the filter graph. The filter_graph will
     * be linked to the graph described by filters_descr.
     */
    /*
     * The buffer source output must be connected to the input pad of
     * the first filter described by filters_descr; since the first
     * filter input label is not specified, it is set to "in" by
     * default.
     */
    outputs->name       = av_strdup("in");
    outputs->filter_ctx = buffersrc_ctx;
    outputs->pad_idx    = 0;
    outputs->next       = NULL;
    /*
     * The buffer sink input must be connected to the output pad of
     * the last filter described by filters_descr; since the last
     * filter output label is not specified, it is set to "out" by
     * default.
     */
    inputs->name       = av_strdup("out");
    inputs->filter_ctx = buffersink_ctx;
    inputs->pad_idx    = 0;
@@ -210,22 +186,22 @@ static void display_frame(const AVFrame *frame, AVRational time_base)
 int main(int argc, char **argv)
 {
    int ret;
-    AVPacket *packet;
+    AVPacket packet;
-    AVFrame *frame;
+    AVFrame *frame = av_frame_alloc();
-    AVFrame *filt_frame;
+    AVFrame *filt_frame = av_frame_alloc();
    int got_frame;
    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();
+    av_register_all();
-    filt_frame = av_frame_alloc();
+    avfilter_register_all();
    packet = av_packet_alloc();
    if (!frame || !filt_frame || !packet) {
        fprintf(stderr, "Could not allocate frame or packet\n");
        exit(1);
    }
    if ((ret = open_input_file(argv[1])) < 0)
        goto end;
@@ -234,26 +210,19 @@ int main(int argc, char **argv)
    /* read all packets */
    while (1) {
-        if ((ret = av_read_frame(fmt_ctx, packet)) < 0)
+        if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
            break;
-        if (packet->stream_index == video_stream_index) {
+        if (packet.stream_index == video_stream_index) {
-            ret = avcodec_send_packet(dec_ctx, packet);
+            got_frame = 0;
            ret = avcodec_decode_video2(dec_ctx, frame, &got_frame, &packet);
            if (ret < 0) {
-                av_log(NULL, AV_LOG_ERROR, "Error while sending a packet to the decoder\n");
+                av_log(NULL, AV_LOG_ERROR, "Error decoding video\n");
                break;
            }
-            while (ret >= 0) {
+            if (got_frame) {
-                ret = avcodec_receive_frame(dec_ctx, frame);
+                frame->pts = av_frame_get_best_effort_timestamp(frame);
                if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
                    break;
                } else if (ret < 0) {
                    av_log(NULL, AV_LOG_ERROR, "Error while receiving a frame from the decoder\n");
                    goto end;
                }
                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) {
@@ -274,15 +243,14 @@ int main(int argc, char **argv)
                av_frame_unref(frame);
            }
        }
-        av_packet_unref(packet);
+        av_free_packet(&packet);
    }
 end:
    avfilter_graph_free(&filter_graph);
-    avcodec_free_context(&dec_ctx);
+    avcodec_close(dec_ctx);
    avformat_close_input(&fmt_ctx);
    av_frame_free(&frame);
    av_frame_free(&filt_frame);
    av_packet_free(&packet);
    if (ret < 0 && ret != AVERROR_EOF) {
        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
--- a/doc/examples/http_multiclient.c
+++ b/doc/examples/http_multiclient.c
@@ -1,156 +0,0 @@
 /*
 * Copyright (c) 2015 Stephan Holljes
 *
 * 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
 * libavformat multi-client network API usage example.
 *
 * @example http_multiclient.c
 * This example will serve a file without decoding or demuxing it over http.
 * Multiple clients can connect and will receive the same file.
 */
 #include <libavformat/avformat.h>
 #include <libavutil/opt.h>
 #include <unistd.h>
 static void process_client(AVIOContext *client, const char *in_uri)
 {
    AVIOContext *input = NULL;
    uint8_t buf[1024];
    int ret, n, reply_code;
    uint8_t *resource = NULL;
    while ((ret = avio_handshake(client)) > 0) {
        av_opt_get(client, "resource", AV_OPT_SEARCH_CHILDREN, &resource);
        // check for strlen(resource) is necessary, because av_opt_get()
        // may return empty string.
        if (resource && strlen(resource))
            break;
        av_freep(&resource);
    }
    if (ret < 0)
        goto end;
    av_log(client, AV_LOG_TRACE, "resource=%p\n", resource);
    if (resource && resource[0] == '/' && !strcmp((resource + 1), in_uri)) {
        reply_code = 200;
    } else {
        reply_code = AVERROR_HTTP_NOT_FOUND;
    }
    if ((ret = av_opt_set_int(client, "reply_code", reply_code, AV_OPT_SEARCH_CHILDREN)) < 0) {
        av_log(client, AV_LOG_ERROR, "Failed to set reply_code: %s.\n", av_err2str(ret));
        goto end;
    }
    av_log(client, AV_LOG_TRACE, "Set reply code to %d\n", reply_code);
    while ((ret = avio_handshake(client)) > 0);
    if (ret < 0)
        goto end;
    fprintf(stderr, "Handshake performed.\n");
    if (reply_code != 200)
        goto end;
    fprintf(stderr, "Opening input file.\n");
    if ((ret = avio_open2(&input, in_uri, AVIO_FLAG_READ, NULL, NULL)) < 0) {
        av_log(input, AV_LOG_ERROR, "Failed to open input: %s: %s.\n", in_uri,
               av_err2str(ret));
        goto end;
    }
    for(;;) {
        n = avio_read(input, buf, sizeof(buf));
        if (n < 0) {
            if (n == AVERROR_EOF)
                break;
            av_log(input, AV_LOG_ERROR, "Error reading from input: %s.\n",
                   av_err2str(n));
            break;
        }
        avio_write(client, buf, n);
        avio_flush(client);
    }
 end:
    fprintf(stderr, "Flushing client\n");
    avio_flush(client);
    fprintf(stderr, "Closing client\n");
    avio_close(client);
    fprintf(stderr, "Closing input\n");
    avio_close(input);
    av_freep(&resource);
 }
 int main(int argc, char **argv)
 {
    AVDictionary *options = NULL;
    AVIOContext *client = NULL, *server = NULL;
    const char *in_uri, *out_uri;
    int ret, pid;
    av_log_set_level(AV_LOG_TRACE);
    if (argc < 3) {
        printf("usage: %s input http://hostname[:port]\n"
               "API example program to serve http to multiple clients.\n"
               "\n", argv[0]);
        return 1;
    }
    in_uri = argv[1];
    out_uri = argv[2];
    avformat_network_init();
    if ((ret = av_dict_set(&options, "listen", "2", 0)) < 0) {
        fprintf(stderr, "Failed to set listen mode for server: %s\n", av_err2str(ret));
        return ret;
    }
    if ((ret = avio_open2(&server, out_uri, AVIO_FLAG_WRITE, NULL, &options)) < 0) {
        fprintf(stderr, "Failed to open server: %s\n", av_err2str(ret));
        return ret;
    }
    fprintf(stderr, "Entering main loop.\n");
    for(;;) {
        if ((ret = avio_accept(server, &client)) < 0)
            goto end;
        fprintf(stderr, "Accepted client, forking process.\n");
        // XXX: Since we don't reap our children and don't ignore signals
        //      this produces zombie processes.
        pid = fork();
        if (pid < 0) {
            perror("Fork failed");
            ret = AVERROR(errno);
            goto end;
        }
        if (pid == 0) {
            fprintf(stderr, "In child.\n");
            process_client(client, in_uri);
            avio_close(server);
            exit(0);
        }
        if (pid > 0)
            avio_close(client);
    }
 end:
    avio_close(server);
    if (ret < 0 && ret != AVERROR_EOF) {
        fprintf(stderr, "Some errors occurred: %s\n", av_err2str(ret));
        return 1;
    }
    return 0;
 }
--- a/doc/examples/hw_decode.c
+++ b/doc/examples/hw_decode.c
@@ -1,256 +0,0 @@
 /*
 * Copyright (c) 2017 Jun Zhao
 * Copyright (c) 2017 Kaixuan Liu
 *
 * 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:
 *
 * 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
 * HW-Accelerated decoding example.
 *
 * @example hw_decode.c
 * This example shows how to do HW-accelerated decoding with output
 * frames from the HW video surfaces.
 */
 #include <stdio.h>
 #include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
 #include <libavutil/pixdesc.h>
 #include <libavutil/hwcontext.h>
 #include <libavutil/opt.h>
 #include <libavutil/avassert.h>
 #include <libavutil/imgutils.h>
 static AVBufferRef *hw_device_ctx = NULL;
 static enum AVPixelFormat hw_pix_fmt;
 static FILE *output_file = NULL;
 static int hw_decoder_init(AVCodecContext *ctx, const enum AVHWDeviceType type)
 {
    int err = 0;
    if ((err = av_hwdevice_ctx_create(&hw_device_ctx, type,
                                      NULL, NULL, 0)) < 0) {
        fprintf(stderr, "Failed to create specified HW device.\n");
        return err;
    }
    ctx->hw_device_ctx = av_buffer_ref(hw_device_ctx);
    return err;
 }
 static enum AVPixelFormat get_hw_format(AVCodecContext *ctx,
                                        const enum AVPixelFormat *pix_fmts)
 {
    const enum AVPixelFormat *p;
    for (p = pix_fmts; *p != -1; p++) {
        if (*p == hw_pix_fmt)
            return *p;
    }
    fprintf(stderr, "Failed to get HW surface format.\n");
    return AV_PIX_FMT_NONE;
 }
 static int decode_write(AVCodecContext *avctx, AVPacket *packet)
 {
    AVFrame *frame = NULL, *sw_frame = NULL;
    AVFrame *tmp_frame = NULL;
    uint8_t *buffer = NULL;
    int size;
    int ret = 0;
    ret = avcodec_send_packet(avctx, packet);
    if (ret < 0) {
        fprintf(stderr, "Error during decoding\n");
        return ret;
    }
    while (1) {
        if (!(frame = av_frame_alloc()) || !(sw_frame = av_frame_alloc())) {
            fprintf(stderr, "Can not alloc frame\n");
            ret = AVERROR(ENOMEM);
            goto fail;
        }
        ret = avcodec_receive_frame(avctx, frame);
        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
            av_frame_free(&frame);
            av_frame_free(&sw_frame);
            return 0;
        } else if (ret < 0) {
            fprintf(stderr, "Error while decoding\n");
            goto fail;
        }
        if (frame->format == hw_pix_fmt) {
            /* retrieve data from GPU to CPU */
            if ((ret = av_hwframe_transfer_data(sw_frame, frame, 0)) < 0) {
                fprintf(stderr, "Error transferring the data to system memory\n");
                goto fail;
            }
            tmp_frame = sw_frame;
        } else
            tmp_frame = frame;
        size = av_image_get_buffer_size(tmp_frame->format, tmp_frame->width,
                                        tmp_frame->height, 1);
        buffer = av_malloc(size);
        if (!buffer) {
            fprintf(stderr, "Can not alloc buffer\n");
            ret = AVERROR(ENOMEM);
            goto fail;
        }
        ret = av_image_copy_to_buffer(buffer, size,
                                      (const uint8_t * const *)tmp_frame->data,
                                      (const int *)tmp_frame->linesize, tmp_frame->format,
                                      tmp_frame->width, tmp_frame->height, 1);
        if (ret < 0) {
            fprintf(stderr, "Can not copy image to buffer\n");
            goto fail;
        }
        if ((ret = fwrite(buffer, 1, size, output_file)) < 0) {
            fprintf(stderr, "Failed to dump raw data.\n");
            goto fail;
        }
    fail:
        av_frame_free(&frame);
        av_frame_free(&sw_frame);
        av_freep(&buffer);
        if (ret < 0)
            return ret;
    }
 }
 int main(int argc, char *argv[])
 {
    AVFormatContext *input_ctx = NULL;
    int video_stream, ret;
    AVStream *video = NULL;
    AVCodecContext *decoder_ctx = NULL;
    const AVCodec *decoder = NULL;
    AVPacket *packet = NULL;
    enum AVHWDeviceType type;
    int i;
    if (argc < 4) {
        fprintf(stderr, "Usage: %s <device type> <input file> <output file>\n", argv[0]);
        return -1;
    }
    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");
        return -1;
    }
    packet = av_packet_alloc();
    if (!packet) {
        fprintf(stderr, "Failed to allocate AVPacket\n");
        return -1;
    }
    /* open the input file */
    if (avformat_open_input(&input_ctx, argv[2], NULL, NULL) != 0) {
        fprintf(stderr, "Cannot open input file '%s'\n", argv[2]);
        return -1;
    }
    if (avformat_find_stream_info(input_ctx, NULL) < 0) {
        fprintf(stderr, "Cannot find input stream information.\n");
        return -1;
    }
    /* find the video stream information */
    ret = av_find_best_stream(input_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, &decoder, 0);
    if (ret < 0) {
        fprintf(stderr, "Cannot find a video stream in the input file\n");
        return -1;
    }
    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);
    video = input_ctx->streams[video_stream];
    if (avcodec_parameters_to_context(decoder_ctx, video->codecpar) < 0)
        return -1;
    decoder_ctx->get_format  = get_hw_format;
    if (hw_decoder_init(decoder_ctx, type) < 0)
        return -1;
    if ((ret = avcodec_open2(decoder_ctx, decoder, NULL)) < 0) {
        fprintf(stderr, "Failed to open codec for stream #%u\n", video_stream);
        return -1;
    }
    /* open the file to dump raw data */
    output_file = fopen(argv[3], "w+b");
    /* actual decoding and dump the raw data */
    while (ret >= 0) {
        if ((ret = av_read_frame(input_ctx, packet)) < 0)
            break;
        if (video_stream == packet->stream_index)
            ret = decode_write(decoder_ctx, packet);
        av_packet_unref(packet);
    }
    /* flush the decoder */
    ret = decode_write(decoder_ctx, NULL);
    if (output_file)
        fclose(output_file);
    av_packet_free(&packet);
    avcodec_free_context(&decoder_ctx);
    avformat_close_input(&input_ctx);
    av_buffer_unref(&hw_device_ctx);
    return 0;
 }
--- a/doc/examples/metadata.c
+++ b/doc/examples/metadata.c
@@ -34,7 +34,7 @@
 int main (int argc, char **argv)
 {
    AVFormatContext *fmt_ctx = NULL;
-    const AVDictionaryEntry *tag = NULL;
+    AVDictionaryEntry *tag = NULL;
    int ret;
    if (argc != 2) {
@@ -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);
--- a/doc/examples/muxing.c
+++ b/doc/examples/muxing.c
@@ -39,7 +39,6 @@
 #include <libavutil/opt.h>
 #include <libavutil/mathematics.h>
 #include <libavutil/timestamp.h>
 #include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
 #include <libswscale/swscale.h>
 #include <libswresample/swresample.h>
@@ -53,21 +52,16 @@
 // a wrapper around a single output AVStream
 typedef struct OutputStream {
    AVStream *st;
    AVCodecContext *enc;
    /* pts of the next frame that will be generated */
    int64_t next_pts;
    int samples_count;
    AVFrame *frame;
    AVFrame *tmp_frame;
    AVPacket *tmp_pkt;
    float t, tincr, tincr2;
    struct SwsContext *sws_ctx;
    struct SwrContext *swr_ctx;
 } OutputStream;
 static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt)
@@ -81,54 +75,23 @@ static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt)
           pkt->stream_index);
 }
-static int write_frame(AVFormatContext *fmt_ctx, AVCodecContext *c,
+static int write_frame(AVFormatContext *fmt_ctx, const AVRational *time_base, AVStream *st, AVPacket *pkt)
                       AVStream *st, AVFrame *frame, AVPacket *pkt)
 {
-    int ret;
+    /* rescale output packet timestamp values from codec to stream timebase */
    av_packet_rescale_ts(pkt, *time_base, st->time_base);
    pkt->stream_index = st->index;
-    // send the frame to the encoder
+    /* Write the compressed frame to the media file. */
-    ret = avcodec_send_frame(c, frame);
+    log_packet(fmt_ctx, pkt);
-    if (ret < 0) {
+    return av_interleaved_write_frame(fmt_ctx, pkt);
        fprintf(stderr, "Error sending a frame to the encoder: %s\n",
                av_err2str(ret));
        exit(1);
    }
    while (ret >= 0) {
        ret = avcodec_receive_packet(c, pkt);
        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
            break;
        else if (ret < 0) {
            fprintf(stderr, "Error encoding a frame: %s\n", av_err2str(ret));
            exit(1);
        }
        /* rescale output packet timestamp values from codec to stream timebase */
        av_packet_rescale_ts(pkt, c->time_base, st->time_base);
        pkt->stream_index = st->index;
        /* Write the compressed frame to the media file. */
        log_packet(fmt_ctx, pkt);
        ret = av_interleaved_write_frame(fmt_ctx, pkt);
        /* pkt is now blank (av_interleaved_write_frame() takes ownership of
         * its contents and resets pkt), so that no unreferencing is necessary.
         * This would be different if one used av_write_frame(). */
        if (ret < 0) {
            fprintf(stderr, "Error while writing output packet: %s\n", av_err2str(ret));
            exit(1);
        }
    }
    return ret == AVERROR_EOF ? 1 : 0;
 }
 /* Add an output stream. */
 static void add_stream(OutputStream *ost, AVFormatContext *oc,
-                       const AVCodec **codec,
+                       AVCodec **codec,
                       enum AVCodecID codec_id)
 {
    AVCodecContext *c;
    int i;
    /* find the encoder */
    *codec = avcodec_find_encoder(codec_id);
@@ -138,24 +101,13 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
        exit(1);
    }
-    ost->tmp_pkt = av_packet_alloc();
+    ost->st = avformat_new_stream(oc, *codec);
    if (!ost->tmp_pkt) {
        fprintf(stderr, "Could not allocate AVPacket\n");
        exit(1);
    }
    ost->st = avformat_new_stream(oc, NULL);
    if (!ost->st) {
        fprintf(stderr, "Could not allocate stream\n");
        exit(1);
    }
    ost->st->id = oc->nb_streams-1;
-    c = avcodec_alloc_context3(*codec);
+    c = ost->st->codec;
    if (!c) {
        fprintf(stderr, "Could not alloc an encoding context\n");
        exit(1);
    }
    ost->enc = c;
    switch ((*codec)->type) {
    case AVMEDIA_TYPE_AUDIO:
@@ -163,15 +115,8 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
            (*codec)->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
        c->bit_rate    = 64000;
        c->sample_rate = 44100;
-        if ((*codec)->supported_samplerates) {
+        c->channels    = 2;
-            c->sample_rate = (*codec)->supported_samplerates[0];
+        c->channel_layout = AV_CH_LAYOUT_STEREO;
            for (i = 0; (*codec)->supported_samplerates[i]; i++) {
                if ((*codec)->supported_samplerates[i] == 44100)
                    c->sample_rate = 44100;
            }
        }
        av_channel_layout_copy(&c->ch_layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO);
        ost->st->time_base = (AVRational){ 1, c->sample_rate };
        break;
    case AVMEDIA_TYPE_VIDEO:
@@ -185,13 +130,12 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
         * of which frame timestamps are represented. For fixed-fps content,
         * timebase should be 1/framerate and timestamp increments should be
         * identical to 1. */
-        ost->st->time_base = (AVRational){ 1, STREAM_FRAME_RATE };
+        c->time_base.den = STREAM_FRAME_RATE;
-        c->time_base       = ost->st->time_base;
+        c->time_base.num = 1;
        c->gop_size      = 12; /* emit one intra frame every twelve frames at most */
        c->pix_fmt       = STREAM_PIX_FMT;
        if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) {
-            /* just for testing, we also add B-frames */
+            /* just for testing, we also add B frames */
            c->max_b_frames = 2;
        }
        if (c->codec_id == AV_CODEC_ID_MPEG1VIDEO) {
@@ -200,7 +144,7 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
             * the motion of the chroma plane does not match the luma plane. */
            c->mb_decision = 2;
        }
-        break;
+    break;
    default:
        break;
@@ -208,54 +152,25 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
    /* Some formats want stream headers to be separate. */
    if (oc->oformat->flags & AVFMT_GLOBALHEADER)
-        c->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
+        c->flags |= CODEC_FLAG_GLOBAL_HEADER;
 }
 /**************************************************************/
 /* audio output */
-static AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt,
+int samples_count;
                                  const AVChannelLayout *channel_layout,
                                  int sample_rate, int nb_samples)
 {
    AVFrame *frame = av_frame_alloc();
    int ret;
-    if (!frame) {
+struct SwrContext *swr_ctx = NULL;
        fprintf(stderr, "Error allocating an audio frame\n");
        exit(1);
    }
-    frame->format = sample_fmt;
+static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost)
    av_channel_layout_copy(&frame->ch_layout, channel_layout);
    frame->sample_rate = sample_rate;
    frame->nb_samples = nb_samples;
    if (nb_samples) {
        ret = av_frame_get_buffer(frame, 0);
        if (ret < 0) {
            fprintf(stderr, "Error allocating an audio buffer\n");
            exit(1);
        }
    }
    return frame;
 }
 static void open_audio(AVFormatContext *oc, const AVCodec *codec,
                       OutputStream *ost, AVDictionary *opt_arg)
 {
    AVCodecContext *c;
    int nb_samples;
    int ret;
    AVDictionary *opt = NULL;
-    c = ost->enc;
+    c = ost->st->codec;
    /* open it */
-    av_dict_copy(&opt, opt_arg, 0);
+    ret = avcodec_open2(c, codec, NULL);
    ret = avcodec_open2(c, codec, &opt);
    av_dict_free(&opt);
    if (ret < 0) {
        fprintf(stderr, "Could not open audio codec: %s\n", av_err2str(ret));
        exit(1);
@@ -267,41 +182,59 @@ static void open_audio(AVFormatContext *oc, const AVCodec *codec,
    /* increment frequency by 110 Hz per second */
    ost->tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
-    if (c->codec->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE)
+    ost->frame = av_frame_alloc();
-        nb_samples = 10000;
+    if (!ost->frame)
    else
        nb_samples = c->frame_size;
    ost->frame     = alloc_audio_frame(c->sample_fmt, &c->ch_layout,
                                       c->sample_rate, nb_samples);
    ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, &c->ch_layout,
                                       c->sample_rate, nb_samples);
    /* copy the stream parameters to the muxer */
    ret = avcodec_parameters_from_context(ost->st->codecpar, c);
    if (ret < 0) {
        fprintf(stderr, "Could not copy the stream parameters\n");
        exit(1);
-    }
+
    ost->frame->sample_rate    = c->sample_rate;
    ost->frame->format         = AV_SAMPLE_FMT_S16;
    ost->frame->channel_layout = c->channel_layout;
    if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
        ost->frame->nb_samples = 10000;
    else
        ost->frame->nb_samples = c->frame_size;
    ost->tmp_frame = av_frame_alloc();
    if (!ost->frame)
        exit(1);
    ost->tmp_frame->sample_rate    = c->sample_rate;
    ost->tmp_frame->format         = c->sample_fmt;
    ost->tmp_frame->channel_layout = c->channel_layout;
    ost->tmp_frame->nb_samples     = ost->frame->nb_samples;
    /* create resampler context */
-    ost->swr_ctx = swr_alloc();
+    if (c->sample_fmt != AV_SAMPLE_FMT_S16) {
-    if (!ost->swr_ctx) {
+        swr_ctx = swr_alloc();
-        fprintf(stderr, "Could not allocate resampler context\n");
+        if (!swr_ctx) {
-        exit(1);
+            fprintf(stderr, "Could not allocate resampler context\n");
            exit(1);
        }
        /* set options */
        av_opt_set_int       (swr_ctx, "in_channel_count",   c->channels,       0);
        av_opt_set_int       (swr_ctx, "in_sample_rate",     c->sample_rate,    0);
        av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt",      AV_SAMPLE_FMT_S16, 0);
        av_opt_set_int       (swr_ctx, "out_channel_count",  c->channels,       0);
        av_opt_set_int       (swr_ctx, "out_sample_rate",    c->sample_rate,    0);
        av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt",     c->sample_fmt,     0);
        /* initialize the resampling context */
        if ((ret = swr_init(swr_ctx)) < 0) {
            fprintf(stderr, "Failed to initialize the resampling context\n");
            exit(1);
        }
    }
-    /* set options */
+    ret = av_frame_get_buffer(ost->frame, 0);
-    av_opt_set_chlayout  (ost->swr_ctx, "in_chlayout",       &c->ch_layout,      0);
+    if (ret < 0) {
-    av_opt_set_int       (ost->swr_ctx, "in_sample_rate",     c->sample_rate,    0);
+        fprintf(stderr, "Could not allocate an audio frame.\n");
-    av_opt_set_sample_fmt(ost->swr_ctx, "in_sample_fmt",      AV_SAMPLE_FMT_S16, 0);
+        exit(1);
-    av_opt_set_chlayout  (ost->swr_ctx, "out_chlayout",      &c->ch_layout,      0);
+    }
-    av_opt_set_int       (ost->swr_ctx, "out_sample_rate",    c->sample_rate,    0);
+    ret = av_frame_get_buffer(ost->tmp_frame, 0);
-    av_opt_set_sample_fmt(ost->swr_ctx, "out_sample_fmt",     c->sample_fmt,     0);
+    if (ret < 0) {
-
+        fprintf(stderr, "Could not allocate an audio frame.\n");
    /* initialize the resampling context */
    if ((ret = swr_init(ost->swr_ctx)) < 0) {
        fprintf(stderr, "Failed to initialize the resampling context\n");
        exit(1);
    }
 }
@@ -310,27 +243,34 @@ static void open_audio(AVFormatContext *oc, const AVCodec *codec,
 * 'nb_channels' channels. */
 static AVFrame *get_audio_frame(OutputStream *ost)
 {
-    AVFrame *frame = ost->tmp_frame;
+    int j, i, v, ret;
-    int j, i, v;
+    int16_t *q = (int16_t*)ost->frame->data[0];
    int16_t *q = (int16_t*)frame->data[0];
    /* check if we want to generate more frames */
-    if (av_compare_ts(ost->next_pts, ost->enc->time_base,
+    if (av_compare_ts(ost->next_pts, ost->st->codec->time_base,
-                      STREAM_DURATION, (AVRational){ 1, 1 }) > 0)
+                      STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
        return NULL;
-    for (j = 0; j <frame->nb_samples; j++) {
+    /* when we pass a frame to the encoder, it may keep a reference to it
     * internally;
     * make sure we do not overwrite it here
     */
    ret = av_frame_make_writable(ost->frame);
    if (ret < 0)
        exit(1);
    for (j = 0; j < ost->frame->nb_samples; j++) {
        v = (int)(sin(ost->t) * 10000);
-        for (i = 0; i < ost->enc->ch_layout.nb_channels; i++)
+        for (i = 0; i < ost->st->codec->channels; i++)
            *q++ = v;
        ost->t     += ost->tincr;
        ost->tincr += ost->tincr2;
    }
-    frame->pts = ost->next_pts;
+    ost->frame->pts = ost->next_pts;
-    ost->next_pts  += frame->nb_samples;
+    ost->next_pts  += ost->frame->nb_samples;
-    return frame;
+    return ost->frame;
 }
 /*
@@ -340,44 +280,58 @@ static AVFrame *get_audio_frame(OutputStream *ost)
 static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
 {
    AVCodecContext *c;
    AVPacket pkt = { 0 }; // data and size must be 0;
    AVFrame *frame;
    int ret;
    int got_packet;
    int dst_nb_samples;
-    c = ost->enc;
+    av_init_packet(&pkt);
    c = ost->st->codec;
    frame = get_audio_frame(ost);
    if (frame) {
        /* convert samples from native format to destination codec format, using the resampler */
-        /* compute destination number of samples */
+        if (swr_ctx) {
-        dst_nb_samples = av_rescale_rnd(swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples,
+            /* compute destination number of samples */
-                                        c->sample_rate, c->sample_rate, AV_ROUND_UP);
+            dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + frame->nb_samples,
-        av_assert0(dst_nb_samples == frame->nb_samples);
+                                            c->sample_rate, c->sample_rate, AV_ROUND_UP);
            av_assert0(dst_nb_samples == frame->nb_samples);
-        /* when we pass a frame to the encoder, it may keep a reference to it
+            /* convert to destination format */
-         * internally;
+            ret = swr_convert(swr_ctx,
-         * make sure we do not overwrite it here
+                              ost->tmp_frame->data, dst_nb_samples,
-         */
+                              (const uint8_t **)frame->data, frame->nb_samples);
-        ret = av_frame_make_writable(ost->frame);
+            if (ret < 0) {
-        if (ret < 0)
+                fprintf(stderr, "Error while converting\n");
-            exit(1);
+                exit(1);
-
+            }
-        /* convert to destination format */
+            frame = ost->tmp_frame;
-        ret = swr_convert(ost->swr_ctx,
+        } else {
-                          ost->frame->data, dst_nb_samples,
+            dst_nb_samples = frame->nb_samples;
                          (const uint8_t **)frame->data, frame->nb_samples);
        if (ret < 0) {
            fprintf(stderr, "Error while converting\n");
            exit(1);
        }
        frame = ost->frame;
-        frame->pts = av_rescale_q(ost->samples_count, (AVRational){1, c->sample_rate}, c->time_base);
+        frame->pts = av_rescale_q(samples_count, (AVRational){1, c->sample_rate}, c->time_base);
-        ost->samples_count += dst_nb_samples;
+        samples_count += dst_nb_samples;
    }
-    return write_frame(oc, c, ost->st, frame, ost->tmp_pkt);
+    ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
    if (ret < 0) {
        fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
        exit(1);
    }
    if (got_packet) {
        ret = write_frame(oc, &c->time_base, ost->st, &pkt);
        if (ret < 0) {
            fprintf(stderr, "Error while writing audio frame: %s\n",
                    av_err2str(ret));
            exit(1);
        }
    }
    return (frame || got_packet) ? 0 : 1;
 }
 /**************************************************************/
@@ -397,7 +351,7 @@ static AVFrame *alloc_picture(enum AVPixelFormat pix_fmt, int width, int height)
    picture->height = height;
    /* allocate the buffers for the frame data */
-    ret = av_frame_get_buffer(picture, 0);
+    ret = av_frame_get_buffer(picture, 32);
    if (ret < 0) {
        fprintf(stderr, "Could not allocate frame data.\n");
        exit(1);
@@ -406,18 +360,13 @@ static AVFrame *alloc_picture(enum AVPixelFormat pix_fmt, int width, int height)
    return picture;
 }
-static void open_video(AVFormatContext *oc, const AVCodec *codec,
+static void open_video(AVFormatContext *oc, AVCodec *codec, OutputStream *ost)
                       OutputStream *ost, AVDictionary *opt_arg)
 {
    int ret;
-    AVCodecContext *c = ost->enc;
+    AVCodecContext *c = ost->st->codec;
    AVDictionary *opt = NULL;
    av_dict_copy(&opt, opt_arg, 0);
    /* open the codec */
-    ret = avcodec_open2(c, codec, &opt);
+    ret = avcodec_open2(c, codec, NULL);
    av_dict_free(&opt);
    if (ret < 0) {
        fprintf(stderr, "Could not open video codec: %s\n", av_err2str(ret));
        exit(1);
@@ -441,20 +390,21 @@ static void open_video(AVFormatContext *oc, const AVCodec *codec,
            exit(1);
        }
    }
    /* copy the stream parameters to the muxer */
    ret = avcodec_parameters_from_context(ost->st->codecpar, c);
    if (ret < 0) {
        fprintf(stderr, "Could not copy the stream parameters\n");
        exit(1);
    }
 }
 /* Prepare a dummy image. */
 static void fill_yuv_image(AVFrame *pict, int frame_index,
                           int width, int height)
 {
-    int x, y, i;
+    int x, y, i, ret;
    /* when we pass a frame to the encoder, it may keep a reference to it
     * internally;
     * make sure we do not overwrite it here
     */
    ret = av_frame_make_writable(pict);
    if (ret < 0)
        exit(1);
    i = frame_index;
@@ -474,18 +424,13 @@ static void fill_yuv_image(AVFrame *pict, int frame_index,
 static AVFrame *get_video_frame(OutputStream *ost)
 {
-    AVCodecContext *c = ost->enc;
+    AVCodecContext *c = ost->st->codec;
    /* check if we want to generate more frames */
-    if (av_compare_ts(ost->next_pts, c->time_base,
+    if (av_compare_ts(ost->next_pts, ost->st->codec->time_base,
-                      STREAM_DURATION, (AVRational){ 1, 1 }) > 0)
+                      STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
        return NULL;
    /* when we pass a frame to the encoder, it may keep a reference to it
     * internally; make sure we do not overwrite it here */
    if (av_frame_make_writable(ost->frame) < 0)
        exit(1);
    if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
        /* as we only generate a YUV420P picture, we must convert it
         * to the codec pixel format if needed */
@@ -502,9 +447,9 @@ 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,
+        sws_scale(ost->sws_ctx,
-                  ost->tmp_frame->linesize, 0, c->height, ost->frame->data,
+                  (const uint8_t * const *)ost->tmp_frame->data, ost->tmp_frame->linesize,
-                  ost->frame->linesize);
+                  0, c->height, ost->frame->data, ost->frame->linesize);
    } else {
        fill_yuv_image(ost->frame, ost->next_pts, c->width, c->height);
    }
@@ -520,17 +465,64 @@ static AVFrame *get_video_frame(OutputStream *ost)
 */
 static int write_video_frame(AVFormatContext *oc, OutputStream *ost)
 {
-    return write_frame(oc, ost->enc, ost->st, get_video_frame(ost), ost->tmp_pkt);
+    int ret;
    AVCodecContext *c;
    AVFrame *frame;
    int got_packet = 0;
    c = ost->st->codec;
    frame = get_video_frame(ost);
    if (oc->oformat->flags & AVFMT_RAWPICTURE) {
        /* a hack to avoid data copy with some raw video muxers */
        AVPacket pkt;
        av_init_packet(&pkt);
        if (!frame)
            return 1;
        pkt.flags        |= AV_PKT_FLAG_KEY;
        pkt.stream_index  = ost->st->index;
        pkt.data          = (uint8_t *)frame;
        pkt.size          = sizeof(AVPicture);
        pkt.pts = pkt.dts = frame->pts;
        av_packet_rescale_ts(&pkt, c->time_base, ost->st->time_base);
        ret = av_interleaved_write_frame(oc, &pkt);
    } else {
        AVPacket pkt = { 0 };
        av_init_packet(&pkt);
        /* encode the image */
        ret = avcodec_encode_video2(c, &pkt, frame, &got_packet);
        if (ret < 0) {
            fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret));
            exit(1);
        }
        if (got_packet) {
            ret = write_frame(oc, &c->time_base, ost->st, &pkt);
        } else {
            ret = 0;
        }
    }
    if (ret < 0) {
        fprintf(stderr, "Error while writing video frame: %s\n", av_err2str(ret));
        exit(1);
    }
    return (frame || got_packet) ? 0 : 1;
 }
 static void close_stream(AVFormatContext *oc, OutputStream *ost)
 {
-    avcodec_free_context(&ost->enc);
+    avcodec_close(ost->st->codec);
    av_frame_free(&ost->frame);
    av_frame_free(&ost->tmp_frame);
    av_packet_free(&ost->tmp_pkt);
    sws_freeContext(ost->sws_ctx);
    swr_free(&ost->swr_ctx);
 }
 /**************************************************************/
@@ -539,17 +531,18 @@ static void close_stream(AVFormatContext *oc, OutputStream *ost)
 int main(int argc, char **argv)
 {
    OutputStream video_st = { 0 }, audio_st = { 0 };
    const AVOutputFormat *fmt;
    const char *filename;
    AVOutputFormat *fmt;
    AVFormatContext *oc;
-    const AVCodec *audio_codec, *video_codec;
+    AVCodec *audio_codec, *video_codec;
    int ret;
    int have_video = 0, have_audio = 0;
    int encode_video = 0, encode_audio = 0;
    AVDictionary *opt = NULL;
    int i;
-    if (argc < 2) {
+    /* 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"
               "This program generates a synthetic audio and video stream, encodes and\n"
@@ -561,10 +554,6 @@ int main(int argc, char **argv)
    }
    filename = argv[1];
    for (i = 2; i+1 < argc; i+=2) {
        if (!strcmp(argv[i], "-flags") || !strcmp(argv[i], "-fflags"))
            av_dict_set(&opt, argv[i]+1, argv[i+1], 0);
    }
    /* allocate the output media context */
    avformat_alloc_output_context2(&oc, NULL, NULL, filename);
@@ -593,10 +582,10 @@ int main(int argc, char **argv)
    /* Now that all the parameters are set, we can open the audio and
     * video codecs and allocate the necessary encode buffers. */
    if (have_video)
-        open_video(oc, video_codec, &video_st, opt);
+        open_video(oc, video_codec, &video_st);
    if (have_audio)
-        open_audio(oc, audio_codec, &audio_st, opt);
+        open_audio(oc, audio_codec, &audio_st);
    av_dump_format(oc, 0, filename, 1);
@@ -611,7 +600,7 @@ int main(int argc, char **argv)
    }
    /* Write the stream header, if any. */
-    ret = avformat_write_header(oc, &opt);
+    ret = avformat_write_header(oc, NULL);
    if (ret < 0) {
        fprintf(stderr, "Error occurred when opening output file: %s\n",
                av_err2str(ret));
@@ -621,14 +610,18 @@ int main(int argc, char **argv)
    while (encode_video || encode_audio) {
        /* select the stream to encode */
        if (encode_video &&
-            (!encode_audio || av_compare_ts(video_st.next_pts, video_st.enc->time_base,
+            (!encode_audio || av_compare_ts(video_st.next_pts, video_st.st->codec->time_base,
-                                            audio_st.next_pts, audio_st.enc->time_base) <= 0)) {
+                                            audio_st.next_pts, audio_st.st->codec->time_base) <= 0)) {
            encode_video = !write_video_frame(oc, &video_st);
        } else {
            encode_audio = !write_audio_frame(oc, &audio_st);
        }
    }
    /* Write the trailer, if any. The trailer must be written before you
     * close the CodecContexts open when you wrote the header; otherwise
     * av_write_trailer() may try to use memory that was freed on
     * av_codec_close(). */
    av_write_trailer(oc);
    /* Close each codec. */
@@ -639,7 +632,7 @@ int main(int argc, char **argv)
    if (!(fmt->flags & AVFMT_NOFILE))
        /* Close the output file. */
-        avio_closep(&oc->pb);
+        avio_close(oc->pb);
    /* free the stream */
    avformat_free_context(oc);
--- a/doc/examples/qsvdec.c
+++ b/doc/examples/qsvdec.c
@@ -1,244 +0,0 @@
 /*
 * Copyright (c) 2015 Anton Khirnov
 *
 * Permission is hereby granted, free of charge, to any person obtaining a copy
 * of this software and associated documentation files (the "Software"), to deal
 * in the Software without restriction, including without limitation the rights
 * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
 * copies of the Software, and to permit persons to whom the Software is
 * furnished to do so, subject to the following conditions:
 *
 * The above copyright notice and this permission notice shall be included in
 * all copies or substantial portions of the Software.
 *
 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
 * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
 * THE SOFTWARE.
 */
 /**
 * @file
 * Intel QSV-accelerated H.264 decoding example.
 *
 * @example qsvdec.c
 * This example shows how to do QSV-accelerated H.264 decoding with output
 * frames in the GPU video surfaces.
 */
 #include "config.h"
 #include <stdio.h>
 #include "libavformat/avformat.h"
 #include "libavformat/avio.h"
 #include "libavcodec/avcodec.h"
 #include "libavutil/buffer.h"
 #include "libavutil/error.h"
 #include "libavutil/hwcontext.h"
 #include "libavutil/hwcontext_qsv.h"
 #include "libavutil/mem.h"
 static int get_format(AVCodecContext *avctx, const enum AVPixelFormat *pix_fmts)
 {
    while (*pix_fmts != AV_PIX_FMT_NONE) {
        if (*pix_fmts == AV_PIX_FMT_QSV) {
            return AV_PIX_FMT_QSV;
        }
        pix_fmts++;
    }
    fprintf(stderr, "The QSV pixel format not offered in get_format()\n");
    return AV_PIX_FMT_NONE;
 }
 static int decode_packet(AVCodecContext *decoder_ctx,
                         AVFrame *frame, AVFrame *sw_frame,
                         AVPacket *pkt, AVIOContext *output_ctx)
 {
    int ret = 0;
    ret = avcodec_send_packet(decoder_ctx, pkt);
    if (ret < 0) {
        fprintf(stderr, "Error during decoding\n");
        return ret;
    }
    while (ret >= 0) {
        int i, j;
        ret = avcodec_receive_frame(decoder_ctx, frame);
        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
            break;
        else if (ret < 0) {
            fprintf(stderr, "Error during decoding\n");
            return ret;
        }
        /* A real program would do something useful with the decoded frame here.
         * We just retrieve the raw data and write it to a file, which is rather
         * useless but pedagogic. */
        ret = av_hwframe_transfer_data(sw_frame, frame, 0);
        if (ret < 0) {
            fprintf(stderr, "Error transferring the data to system memory\n");
            goto fail;
        }
        for (i = 0; i < FF_ARRAY_ELEMS(sw_frame->data) && sw_frame->data[i]; i++)
            for (j = 0; j < (sw_frame->height >> (i > 0)); j++)
                avio_write(output_ctx, sw_frame->data[i] + j * sw_frame->linesize[i], sw_frame->width);
 fail:
        av_frame_unref(sw_frame);
        av_frame_unref(frame);
        if (ret < 0)
            return ret;
    }
    return 0;
 }
 int main(int argc, char **argv)
 {
    AVFormatContext *input_ctx = NULL;
    AVStream *video_st = NULL;
    AVCodecContext *decoder_ctx = NULL;
    const AVCodec *decoder;
    AVPacket *pkt = NULL;
    AVFrame *frame = NULL, *sw_frame = NULL;
    AVIOContext *output_ctx = NULL;
    int ret, i;
    AVBufferRef *device_ref = NULL;
    if (argc < 3) {
        fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
        return 1;
    }
    /* open the input file */
    ret = avformat_open_input(&input_ctx, argv[1], NULL, NULL);
    if (ret < 0) {
        fprintf(stderr, "Cannot open input file '%s': ", argv[1]);
        goto finish;
    }
    /* find the first H.264 video stream */
    for (i = 0; i < input_ctx->nb_streams; i++) {
        AVStream *st = input_ctx->streams[i];
        if (st->codecpar->codec_id == AV_CODEC_ID_H264 && !video_st)
            video_st = st;
        else
            st->discard = AVDISCARD_ALL;
    }
    if (!video_st) {
        fprintf(stderr, "No H.264 video stream in the input file\n");
        goto finish;
    }
    /* open the hardware device */
    ret = av_hwdevice_ctx_create(&device_ref, AV_HWDEVICE_TYPE_QSV,
                                 "auto", NULL, 0);
    if (ret < 0) {
        fprintf(stderr, "Cannot open the hardware device\n");
        goto finish;
    }
    /* initialize the decoder */
    decoder = avcodec_find_decoder_by_name("h264_qsv");
    if (!decoder) {
        fprintf(stderr, "The QSV decoder is not present in libavcodec\n");
        goto finish;
    }
    decoder_ctx = avcodec_alloc_context3(decoder);
    if (!decoder_ctx) {
        ret = AVERROR(ENOMEM);
        goto finish;
    }
    decoder_ctx->codec_id = AV_CODEC_ID_H264;
    if (video_st->codecpar->extradata_size) {
        decoder_ctx->extradata = av_mallocz(video_st->codecpar->extradata_size +
                                            AV_INPUT_BUFFER_PADDING_SIZE);
        if (!decoder_ctx->extradata) {
            ret = AVERROR(ENOMEM);
            goto finish;
        }
        memcpy(decoder_ctx->extradata, video_st->codecpar->extradata,
               video_st->codecpar->extradata_size);
        decoder_ctx->extradata_size = video_st->codecpar->extradata_size;
    }
    decoder_ctx->hw_device_ctx = av_buffer_ref(device_ref);
    decoder_ctx->get_format  = get_format;
    ret = avcodec_open2(decoder_ctx, NULL, NULL);
    if (ret < 0) {
        fprintf(stderr, "Error opening the decoder: ");
        goto finish;
    }
    /* open the output stream */
    ret = avio_open(&output_ctx, argv[2], AVIO_FLAG_WRITE);
    if (ret < 0) {
        fprintf(stderr, "Error opening the output context: ");
        goto finish;
    }
    frame    = av_frame_alloc();
    sw_frame = av_frame_alloc();
    pkt      = av_packet_alloc();
    if (!frame || !sw_frame || !pkt) {
        ret = AVERROR(ENOMEM);
        goto finish;
    }
    /* actual decoding */
    while (ret >= 0) {
        ret = av_read_frame(input_ctx, pkt);
        if (ret < 0)
            break;
        if (pkt->stream_index == video_st->index)
            ret = decode_packet(decoder_ctx, frame, sw_frame, pkt, output_ctx);
        av_packet_unref(pkt);
    }
    /* flush the decoder */
    ret = decode_packet(decoder_ctx, frame, sw_frame, NULL, output_ctx);
 finish:
    if (ret < 0) {
        char buf[1024];
        av_strerror(ret, buf, sizeof(buf));
        fprintf(stderr, "%s\n", buf);
    }
    avformat_close_input(&input_ctx);
    av_frame_free(&frame);
    av_frame_free(&sw_frame);
    av_packet_free(&pkt);
    avcodec_free_context(&decoder_ctx);
    av_buffer_unref(&device_ref);
    avio_close(output_ctx);
    return ret;
 }
--- a/doc/examples/remuxing.c
+++ b/doc/examples/remuxing.c
@@ -45,14 +45,11 @@ static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt, cons
 int main(int argc, char **argv)
 {
-    const AVOutputFormat *ofmt = NULL;
+    AVOutputFormat *ofmt = NULL;
    AVFormatContext *ifmt_ctx = NULL, *ofmt_ctx = NULL;
-    AVPacket *pkt = NULL;
+    AVPacket pkt;
    const char *in_filename, *out_filename;
    int ret, i;
    int stream_index = 0;
    int *stream_mapping = NULL;
    int stream_mapping_size = 0;
    if (argc < 3) {
        printf("usage: %s input output\n"
@@ -65,11 +62,7 @@ int main(int argc, char **argv)
    in_filename  = argv[1];
    out_filename = argv[2];
-    pkt = av_packet_alloc();
+    av_register_all();
    if (!pkt) {
        fprintf(stderr, "Could not allocate AVPacket\n");
        return 1;
    }
    if ((ret = avformat_open_input(&ifmt_ctx, in_filename, 0, 0)) < 0) {
        fprintf(stderr, "Could not open input file '%s'", in_filename);
@@ -90,42 +83,25 @@ int main(int argc, char **argv)
        goto end;
    }
    stream_mapping_size = ifmt_ctx->nb_streams;
    stream_mapping = av_calloc(stream_mapping_size, sizeof(*stream_mapping));
    if (!stream_mapping) {
        ret = AVERROR(ENOMEM);
        goto end;
    }
    ofmt = ofmt_ctx->oformat;
    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
        AVStream *out_stream;
        AVStream *in_stream = ifmt_ctx->streams[i];
-        AVCodecParameters *in_codecpar = in_stream->codecpar;
+        AVStream *out_stream = avformat_new_stream(ofmt_ctx, in_stream->codec->codec);
        if (in_codecpar->codec_type != AVMEDIA_TYPE_AUDIO &&
            in_codecpar->codec_type != AVMEDIA_TYPE_VIDEO &&
            in_codecpar->codec_type != AVMEDIA_TYPE_SUBTITLE) {
            stream_mapping[i] = -1;
            continue;
        }
        stream_mapping[i] = stream_index++;
        out_stream = avformat_new_stream(ofmt_ctx, NULL);
        if (!out_stream) {
            fprintf(stderr, "Failed allocating output stream\n");
            ret = AVERROR_UNKNOWN;
            goto end;
        }
-        ret = avcodec_parameters_copy(out_stream->codecpar, in_codecpar);
+        ret = avcodec_copy_context(out_stream->codec, in_stream->codec);
        if (ret < 0) {
-            fprintf(stderr, "Failed to copy codec parameters\n");
+            fprintf(stderr, "Failed to copy context from input to output stream codec context\n");
            goto end;
        }
-        out_stream->codecpar->codec_tag = 0;
+        out_stream->codec->codec_tag = 0;
        if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
            out_stream->codec->flags |= CODEC_FLAG_GLOBAL_HEADER;
    }
    av_dump_format(ofmt_ctx, 0, out_filename, 1);
@@ -146,49 +122,40 @@ int main(int argc, char **argv)
    while (1) {
        AVStream *in_stream, *out_stream;
-        ret = av_read_frame(ifmt_ctx, pkt);
+        ret = av_read_frame(ifmt_ctx, &pkt);
        if (ret < 0)
            break;
-        in_stream  = ifmt_ctx->streams[pkt->stream_index];
+        in_stream  = ifmt_ctx->streams[pkt.stream_index];
-        if (pkt->stream_index >= stream_mapping_size ||
+        out_stream = ofmt_ctx->streams[pkt.stream_index];
            stream_mapping[pkt->stream_index] < 0) {
            av_packet_unref(pkt);
            continue;
        }
-        pkt->stream_index = stream_mapping[pkt->stream_index];
+        log_packet(ifmt_ctx, &pkt, "in");
        out_stream = ofmt_ctx->streams[pkt->stream_index];
        log_packet(ifmt_ctx, pkt, "in");
        /* copy packet */
-        av_packet_rescale_ts(pkt, in_stream->time_base, out_stream->time_base);
+        pkt.pts = av_rescale_q_rnd(pkt.pts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
-        pkt->pos = -1;
+        pkt.dts = av_rescale_q_rnd(pkt.dts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
-        log_packet(ofmt_ctx, pkt, "out");
+        pkt.duration = av_rescale_q(pkt.duration, in_stream->time_base, out_stream->time_base);
        pkt.pos = -1;
        log_packet(ofmt_ctx, &pkt, "out");
-        ret = av_interleaved_write_frame(ofmt_ctx, pkt);
+        ret = av_interleaved_write_frame(ofmt_ctx, &pkt);
        /* pkt is now blank (av_interleaved_write_frame() takes ownership of
         * its contents and resets pkt), so that no unreferencing is necessary.
         * This would be different if one used av_write_frame(). */
        if (ret < 0) {
            fprintf(stderr, "Error muxing packet\n");
            break;
        }
        av_free_packet(&pkt);
    }
    av_write_trailer(ofmt_ctx);
 end:
    av_packet_free(&pkt);
    avformat_close_input(&ifmt_ctx);
    /* close output */
    if (ofmt_ctx && !(ofmt->flags & AVFMT_NOFILE))
-        avio_closep(&ofmt_ctx->pb);
+        avio_close(ofmt_ctx->pb);
    avformat_free_context(ofmt_ctx);
    av_freep(&stream_mapping);
    if (ret < 0 && ret != AVERROR_EOF) {
        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
        return 1;
--- a/doc/examples/resampling_audio.c
+++ b/doc/examples/resampling_audio.c
@@ -80,7 +80,7 @@ static void fill_samples(double *dst, int nb_samples, int nb_channels, int sampl
 int main(int argc, char **argv)
 {
-    AVChannelLayout src_ch_layout = AV_CHANNEL_LAYOUT_STEREO, dst_ch_layout = AV_CHANNEL_LAYOUT_SURROUND;
+    int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND;
    int src_rate = 48000, dst_rate = 44100;
    uint8_t **src_data = NULL, **dst_data = NULL;
    int src_nb_channels = 0, dst_nb_channels = 0;
@@ -92,7 +92,6 @@ int main(int argc, char **argv)
    int dst_bufsize;
    const char *fmt;
    struct SwrContext *swr_ctx;
    char buf[64];
    double t;
    int ret;
@@ -121,11 +120,11 @@ int main(int argc, char **argv)
    }
    /* set options */
-    av_opt_set_chlayout(swr_ctx, "in_chlayout",    &src_ch_layout, 0);
+    av_opt_set_int(swr_ctx, "in_channel_layout",    src_ch_layout, 0);
    av_opt_set_int(swr_ctx, "in_sample_rate",       src_rate, 0);
    av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);
-    av_opt_set_chlayout(swr_ctx, "out_chlayout",    &dst_ch_layout, 0);
+    av_opt_set_int(swr_ctx, "out_channel_layout",    dst_ch_layout, 0);
    av_opt_set_int(swr_ctx, "out_sample_rate",       dst_rate, 0);
    av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);
@@ -137,7 +136,7 @@ int main(int argc, char **argv)
    /* allocate source and destination samples buffers */
-    src_nb_channels = src_ch_layout.nb_channels;
+    src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
    ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_nb_channels,
                                             src_nb_samples, src_sample_fmt, 0);
    if (ret < 0) {
@@ -152,7 +151,7 @@ int main(int argc, char **argv)
        av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
    /* buffer is going to be directly written to a rawaudio file, no alignment */
-    dst_nb_channels = dst_ch_layout.nb_channels;
+    dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
    ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels,
                                             dst_nb_samples, dst_sample_fmt, 0);
    if (ret < 0) {
@@ -195,13 +194,13 @@ int main(int argc, char **argv)
    if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0)
        goto end;
    av_channel_layout_describe(&dst_ch_layout, buf, sizeof(buf));
    fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"
-            "ffplay -f %s -channel_layout %s -channels %d -ar %d %s\n",
+            "ffplay -f %s -channel_layout %"PRId64" -channels %d -ar %d %s\n",
-            fmt, buf, dst_nb_channels, dst_rate, dst_filename);
+            fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);
 end:
-    fclose(dst_file);
+    if (dst_file)
        fclose(dst_file);
    if (src_data)
        av_freep(&src_data[0]);
--- a/doc/examples/scaling_video.c
+++ b/doc/examples/scaling_video.c
@@ -132,7 +132,8 @@ int main(int argc, char **argv)
           av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h, dst_filename);
 end:
-    fclose(dst_file);
+    if (dst_file)
        fclose(dst_file);
    av_freep(&src_data[0]);
    av_freep(&dst_data[0]);
    sws_freeContext(sws_ctx);
--- a/doc/examples/transcode_aac.c
+++ b/doc/examples/transcode_aac.c
--- a/doc/examples/transcoding.c
+++ b/doc/examples/transcoding.c
@@ -30,9 +30,10 @@
 #include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
 #include <libavfilter/avfiltergraph.h>
 #include <libavfilter/avcodec.h>
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
 #include <libavutil/channel_layout.h>
 #include <libavutil/opt.h>
 #include <libavutil/pixdesc.h>
@@ -42,20 +43,9 @@ typedef struct FilteringContext {
    AVFilterContext *buffersink_ctx;
    AVFilterContext *buffersrc_ctx;
    AVFilterGraph *filter_graph;
    AVPacket *enc_pkt;
    AVFrame *filtered_frame;
 } FilteringContext;
 static FilteringContext *filter_ctx;
 typedef struct StreamContext {
    AVCodecContext *dec_ctx;
    AVCodecContext *enc_ctx;
    AVFrame *dec_frame;
 } StreamContext;
 static StreamContext *stream_ctx;
 static int open_input_file(const char *filename)
 {
    int ret;
@@ -72,46 +62,22 @@ static int open_input_file(const char *filename)
        return ret;
    }
    stream_ctx = av_calloc(ifmt_ctx->nb_streams, sizeof(*stream_ctx));
    if (!stream_ctx)
        return AVERROR(ENOMEM);
    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
-        AVStream *stream = ifmt_ctx->streams[i];
+        AVStream *stream;
        const AVCodec *dec = avcodec_find_decoder(stream->codecpar->codec_id);
        AVCodecContext *codec_ctx;
-        if (!dec) {
+        stream = ifmt_ctx->streams[i];
-            av_log(NULL, AV_LOG_ERROR, "Failed to find decoder for stream #%u\n", i);
+        codec_ctx = stream->codec;
            return AVERROR_DECODER_NOT_FOUND;
        }
        codec_ctx = avcodec_alloc_context3(dec);
        if (!codec_ctx) {
            av_log(NULL, AV_LOG_ERROR, "Failed to allocate the decoder context for stream #%u\n", i);
            return AVERROR(ENOMEM);
        }
        ret = avcodec_parameters_to_context(codec_ctx, stream->codecpar);
        if (ret < 0) {
            av_log(NULL, AV_LOG_ERROR, "Failed to copy decoder parameters to input decoder context "
                   "for stream #%u\n", i);
            return ret;
        }
        /* Reencode video & audio and remux subtitles etc. */
        if (codec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
                || codec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
            if (codec_ctx->codec_type == AVMEDIA_TYPE_VIDEO)
                codec_ctx->framerate = av_guess_frame_rate(ifmt_ctx, stream, NULL);
            /* Open decoder */
-            ret = avcodec_open2(codec_ctx, dec, NULL);
+            ret = avcodec_open2(codec_ctx,
                    avcodec_find_decoder(codec_ctx->codec_id), NULL);
            if (ret < 0) {
                av_log(NULL, AV_LOG_ERROR, "Failed to open decoder for stream #%u\n", i);
                return ret;
            }
        }
        stream_ctx[i].dec_ctx = codec_ctx;
        stream_ctx[i].dec_frame = av_frame_alloc();
        if (!stream_ctx[i].dec_frame)
            return AVERROR(ENOMEM);
    }
    av_dump_format(ifmt_ctx, 0, filename, 0);
@@ -123,7 +89,7 @@ static int open_output_file(const char *filename)
    AVStream *out_stream;
    AVStream *in_stream;
    AVCodecContext *dec_ctx, *enc_ctx;
-    const AVCodec *encoder;
+    AVCodec *encoder;
    int ret;
    unsigned int i;
@@ -143,21 +109,13 @@ static int open_output_file(const char *filename)
        }
        in_stream = ifmt_ctx->streams[i];
-        dec_ctx = stream_ctx[i].dec_ctx;
+        dec_ctx = in_stream->codec;
        enc_ctx = out_stream->codec;
        if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
                || dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
            /* in this example, we choose transcoding to same codec */
            encoder = avcodec_find_encoder(dec_ctx->codec_id);
            if (!encoder) {
                av_log(NULL, AV_LOG_FATAL, "Necessary encoder not found\n");
                return AVERROR_INVALIDDATA;
            }
            enc_ctx = avcodec_alloc_context3(encoder);
            if (!enc_ctx) {
                av_log(NULL, AV_LOG_FATAL, "Failed to allocate the encoder context\n");
                return AVERROR(ENOMEM);
            }
            /* In this example, we transcode to same properties (picture size,
             * sample rate etc.). These properties can be changed for output
@@ -167,52 +125,40 @@ static int open_output_file(const char *filename)
                enc_ctx->width = dec_ctx->width;
                enc_ctx->sample_aspect_ratio = dec_ctx->sample_aspect_ratio;
                /* take first format from list of supported formats */
-                if (encoder->pix_fmts)
+                enc_ctx->pix_fmt = encoder->pix_fmts[0];
                    enc_ctx->pix_fmt = encoder->pix_fmts[0];
                else
                    enc_ctx->pix_fmt = dec_ctx->pix_fmt;
                /* video time_base can be set to whatever is handy and supported by encoder */
-                enc_ctx->time_base = av_inv_q(dec_ctx->framerate);
+                enc_ctx->time_base = dec_ctx->time_base;
            } else {
                enc_ctx->sample_rate = dec_ctx->sample_rate;
-                ret = av_channel_layout_copy(&enc_ctx->ch_layout, &dec_ctx->ch_layout);
+                enc_ctx->channel_layout = dec_ctx->channel_layout;
-                if (ret < 0)
+                enc_ctx->channels = av_get_channel_layout_nb_channels(enc_ctx->channel_layout);
                    return ret;
                /* take first format from list of supported formats */
                enc_ctx->sample_fmt = encoder->sample_fmts[0];
                enc_ctx->time_base = (AVRational){1, enc_ctx->sample_rate};
            }
            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) {
                av_log(NULL, AV_LOG_ERROR, "Cannot open video encoder for stream #%u\n", i);
                return ret;
            }
            ret = avcodec_parameters_from_context(out_stream->codecpar, enc_ctx);
            if (ret < 0) {
                av_log(NULL, AV_LOG_ERROR, "Failed to copy encoder parameters to output stream #%u\n", i);
                return ret;
            }
            out_stream->time_base = enc_ctx->time_base;
            stream_ctx[i].enc_ctx = enc_ctx;
        } else if (dec_ctx->codec_type == AVMEDIA_TYPE_UNKNOWN) {
            av_log(NULL, AV_LOG_FATAL, "Elementary stream #%d is of unknown type, cannot proceed\n", i);
            return AVERROR_INVALIDDATA;
        } else {
            /* if this stream must be remuxed */
-            ret = avcodec_parameters_copy(out_stream->codecpar, in_stream->codecpar);
+            ret = avcodec_copy_context(ofmt_ctx->streams[i]->codec,
                    ifmt_ctx->streams[i]->codec);
            if (ret < 0) {
-                av_log(NULL, AV_LOG_ERROR, "Copying parameters for stream #%u failed\n", i);
+                av_log(NULL, AV_LOG_ERROR, "Copying stream context failed\n");
                return ret;
            }
            out_stream->time_base = in_stream->time_base;
        }
        if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
            enc_ctx->flags |= CODEC_FLAG_GLOBAL_HEADER;
    }
    av_dump_format(ofmt_ctx, 0, filename, 1);
@@ -239,8 +185,8 @@ static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
 {
    char args[512];
    int ret = 0;
-    const AVFilter *buffersrc = NULL;
+    AVFilter *buffersrc = NULL;
-    const AVFilter *buffersink = NULL;
+    AVFilter *buffersink = NULL;
    AVFilterContext *buffersrc_ctx = NULL;
    AVFilterContext *buffersink_ctx = NULL;
    AVFilterInOut *outputs = avfilter_inout_alloc();
@@ -290,7 +236,6 @@ static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
            goto end;
        }
    } else if (dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
        char buf[64];
        buffersrc = avfilter_get_by_name("abuffer");
        buffersink = avfilter_get_by_name("abuffersink");
        if (!buffersrc || !buffersink) {
@@ -299,14 +244,14 @@ static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
            goto end;
        }
-        if (dec_ctx->ch_layout.order == AV_CHANNEL_ORDER_UNSPEC)
+        if (!dec_ctx->channel_layout)
-            av_channel_layout_default(&dec_ctx->ch_layout, dec_ctx->ch_layout.nb_channels);
+            dec_ctx->channel_layout =
-        av_channel_layout_describe(&dec_ctx->ch_layout, buf, sizeof(buf));
+                av_get_default_channel_layout(dec_ctx->channels);
        snprintf(args, sizeof(args),
-                "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=%s",
+                "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
                dec_ctx->time_base.num, dec_ctx->time_base.den, dec_ctx->sample_rate,
                av_get_sample_fmt_name(dec_ctx->sample_fmt),
-                buf);
+                dec_ctx->channel_layout);
        ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
                args, NULL, filter_graph);
        if (ret < 0) {
@@ -329,9 +274,9 @@ static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
            goto end;
        }
-        av_channel_layout_describe(&enc_ctx->ch_layout, buf, sizeof(buf));
+        ret = av_opt_set_bin(buffersink_ctx, "channel_layouts",
-        ret = av_opt_set(buffersink_ctx, "ch_layouts",
+                (uint8_t*)&enc_ctx->channel_layout,
-                         buf, AV_OPT_SEARCH_CHILDREN);
+                sizeof(enc_ctx->channel_layout), AV_OPT_SEARCH_CHILDREN);
        if (ret < 0) {
            av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
            goto end;
@@ -397,76 +342,75 @@ static int init_filters(void)
        filter_ctx[i].buffersrc_ctx  = NULL;
        filter_ctx[i].buffersink_ctx = NULL;
        filter_ctx[i].filter_graph   = NULL;
-        if (!(ifmt_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO
+        if (!(ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO
-                || ifmt_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_VIDEO))
+                || ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO))
            continue;
-        if (ifmt_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_VIDEO)
+        if (ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO)
            filter_spec = "null"; /* passthrough (dummy) filter for video */
        else
            filter_spec = "anull"; /* passthrough (dummy) filter for audio */
-        ret = init_filter(&filter_ctx[i], stream_ctx[i].dec_ctx,
+        ret = init_filter(&filter_ctx[i], ifmt_ctx->streams[i]->codec,
-                stream_ctx[i].enc_ctx, filter_spec);
+                ofmt_ctx->streams[i]->codec, filter_spec);
        if (ret)
            return ret;
        filter_ctx[i].enc_pkt = av_packet_alloc();
        if (!filter_ctx[i].enc_pkt)
            return AVERROR(ENOMEM);
        filter_ctx[i].filtered_frame = av_frame_alloc();
        if (!filter_ctx[i].filtered_frame)
            return AVERROR(ENOMEM);
    }
    return 0;
 }
-static int encode_write_frame(unsigned int stream_index, int flush)
+static int encode_write_frame(AVFrame *filt_frame, unsigned int stream_index, int *got_frame) {
 {
    StreamContext *stream = &stream_ctx[stream_index];
    FilteringContext *filter = &filter_ctx[stream_index];
    AVFrame *filt_frame = flush ? NULL : filter->filtered_frame;
    AVPacket *enc_pkt = filter->enc_pkt;
    int ret;
    int got_frame_local;
    AVPacket enc_pkt;
    int (*enc_func)(AVCodecContext *, AVPacket *, const AVFrame *, int *) =
        (ifmt_ctx->streams[stream_index]->codec->codec_type ==
         AVMEDIA_TYPE_VIDEO) ? avcodec_encode_video2 : avcodec_encode_audio2;
    if (!got_frame)
        got_frame = &got_frame_local;
    av_log(NULL, AV_LOG_INFO, "Encoding frame\n");
    /* encode filtered frame */
-    av_packet_unref(enc_pkt);
+    enc_pkt.data = NULL;
-
+    enc_pkt.size = 0;
-    ret = avcodec_send_frame(stream->enc_ctx, filt_frame);
+    av_init_packet(&enc_pkt);
-
+    ret = enc_func(ofmt_ctx->streams[stream_index]->codec, &enc_pkt,
            filt_frame, got_frame);
    av_frame_free(&filt_frame);
    if (ret < 0)
        return ret;
    if (!(*got_frame))
        return 0;
-    while (ret >= 0) {
+    /* prepare packet for muxing */
-        ret = avcodec_receive_packet(stream->enc_ctx, enc_pkt);
+    enc_pkt.stream_index = stream_index;
-
+    enc_pkt.dts = av_rescale_q_rnd(enc_pkt.dts,
-        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
+            ofmt_ctx->streams[stream_index]->codec->time_base,
-            return 0;
+            ofmt_ctx->streams[stream_index]->time_base,
-
+            AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
-        /* prepare packet for muxing */
+    enc_pkt.pts = av_rescale_q_rnd(enc_pkt.pts,
-        enc_pkt->stream_index = stream_index;
+            ofmt_ctx->streams[stream_index]->codec->time_base,
-        av_packet_rescale_ts(enc_pkt,
+            ofmt_ctx->streams[stream_index]->time_base,
-                             stream->enc_ctx->time_base,
+            AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
-                             ofmt_ctx->streams[stream_index]->time_base);
+    enc_pkt.duration = av_rescale_q(enc_pkt.duration,
-
+            ofmt_ctx->streams[stream_index]->codec->time_base,
-        av_log(NULL, AV_LOG_DEBUG, "Muxing frame\n");
+            ofmt_ctx->streams[stream_index]->time_base);
        /* mux encoded frame */
        ret = av_interleaved_write_frame(ofmt_ctx, enc_pkt);
    }
    av_log(NULL, AV_LOG_DEBUG, "Muxing frame\n");
    /* mux encoded frame */
    ret = av_interleaved_write_frame(ofmt_ctx, &enc_pkt);
    return ret;
 }
 static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index)
 {
    FilteringContext *filter = &filter_ctx[stream_index];
    int ret;
    AVFrame *filt_frame;
    av_log(NULL, AV_LOG_INFO, "Pushing decoded frame to filters\n");
    /* push the decoded frame into the filtergraph */
-    ret = av_buffersrc_add_frame_flags(filter->buffersrc_ctx,
+    ret = av_buffersrc_add_frame_flags(filter_ctx[stream_index].buffersrc_ctx,
            frame, 0);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
@@ -475,9 +419,14 @@ static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index)
    /* pull filtered frames from the filtergraph */
    while (1) {
        filt_frame = av_frame_alloc();
        if (!filt_frame) {
            ret = AVERROR(ENOMEM);
            break;
        }
        av_log(NULL, AV_LOG_INFO, "Pulling filtered frame from filters\n");
-        ret = av_buffersink_get_frame(filter->buffersink_ctx,
+        ret = av_buffersink_get_frame(filter_ctx[stream_index].buffersink_ctx,
-                                      filter->filtered_frame);
+                filt_frame);
        if (ret < 0) {
            /* if no more frames for output - returns AVERROR(EAGAIN)
             * if flushed and no more frames for output - returns AVERROR_EOF
@@ -485,12 +434,12 @@ static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index)
             */
            if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
                ret = 0;
            av_frame_free(&filt_frame);
            break;
        }
-        filter->filtered_frame->pict_type = AV_PICTURE_TYPE_NONE;
+        filt_frame->pict_type = AV_PICTURE_TYPE_NONE;
-        ret = encode_write_frame(stream_index, 0);
+        ret = encode_write_frame(filt_frame, stream_index, NULL);
        av_frame_unref(filter->filtered_frame);
        if (ret < 0)
            break;
    }
@@ -500,80 +449,109 @@ static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index)
 static int flush_encoder(unsigned int stream_index)
 {
-    if (!(stream_ctx[stream_index].enc_ctx->codec->capabilities &
+    int ret;
-                AV_CODEC_CAP_DELAY))
+    int got_frame;
    if (!(ofmt_ctx->streams[stream_index]->codec->codec->capabilities &
                CODEC_CAP_DELAY))
        return 0;
-    av_log(NULL, AV_LOG_INFO, "Flushing stream #%u encoder\n", stream_index);
+    while (1) {
-    return encode_write_frame(stream_index, 1);
+        av_log(NULL, AV_LOG_INFO, "Flushing stream #%u encoder\n", stream_index);
        ret = encode_write_frame(NULL, stream_index, &got_frame);
        if (ret < 0)
            break;
        if (!got_frame)
            return 0;
    }
    return ret;
 }
 int main(int argc, char **argv)
 {
    int ret;
-    AVPacket *packet = NULL;
+    AVPacket packet = { .data = NULL, .size = 0 };
    AVFrame *frame = NULL;
    enum AVMediaType type;
    unsigned int stream_index;
    unsigned int i;
    int got_frame;
    int (*dec_func)(AVCodecContext *, AVFrame *, int *, const AVPacket *);
    if (argc != 3) {
        av_log(NULL, AV_LOG_ERROR, "Usage: %s <input file> <output file>\n", argv[0]);
        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)
        goto end;
    if ((ret = init_filters()) < 0)
        goto end;
    if (!(packet = av_packet_alloc()))
        goto end;
    /* read all packets */
    while (1) {
-        if ((ret = av_read_frame(ifmt_ctx, packet)) < 0)
+        if ((ret = av_read_frame(ifmt_ctx, &packet)) < 0)
            break;
-        stream_index = packet->stream_index;
+        stream_index = packet.stream_index;
        type = ifmt_ctx->streams[packet.stream_index]->codec->codec_type;
        av_log(NULL, AV_LOG_DEBUG, "Demuxer gave frame of stream_index %u\n",
                stream_index);
        if (filter_ctx[stream_index].filter_graph) {
            StreamContext *stream = &stream_ctx[stream_index];
            av_log(NULL, AV_LOG_DEBUG, "Going to reencode&filter the frame\n");
-
+            frame = av_frame_alloc();
-            av_packet_rescale_ts(packet,
+            if (!frame) {
-                                 ifmt_ctx->streams[stream_index]->time_base,
+                ret = AVERROR(ENOMEM);
-                                 stream->dec_ctx->time_base);
+                break;
-            ret = avcodec_send_packet(stream->dec_ctx, packet);
+            }
            packet.dts = av_rescale_q_rnd(packet.dts,
                    ifmt_ctx->streams[stream_index]->time_base,
                    ifmt_ctx->streams[stream_index]->codec->time_base,
                    AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
            packet.pts = av_rescale_q_rnd(packet.pts,
                    ifmt_ctx->streams[stream_index]->time_base,
                    ifmt_ctx->streams[stream_index]->codec->time_base,
                    AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
            dec_func = (type == AVMEDIA_TYPE_VIDEO) ? avcodec_decode_video2 :
                avcodec_decode_audio4;
            ret = dec_func(ifmt_ctx->streams[stream_index]->codec, frame,
                    &got_frame, &packet);
            if (ret < 0) {
                av_frame_free(&frame);
                av_log(NULL, AV_LOG_ERROR, "Decoding failed\n");
                break;
            }
-            while (ret >= 0) {
+            if (got_frame) {
-                ret = avcodec_receive_frame(stream->dec_ctx, stream->dec_frame);
+                frame->pts = av_frame_get_best_effort_timestamp(frame);
-                if (ret == AVERROR_EOF || ret == AVERROR(EAGAIN))
+                ret = filter_encode_write_frame(frame, stream_index);
-                    break;
+                av_frame_free(&frame);
                else if (ret < 0)
                    goto end;
                stream->dec_frame->pts = stream->dec_frame->best_effort_timestamp;
                ret = filter_encode_write_frame(stream->dec_frame, stream_index);
                if (ret < 0)
                    goto end;
            } else {
                av_frame_free(&frame);
            }
        } else {
            /* remux this frame without reencoding */
-            av_packet_rescale_ts(packet,
+            packet.dts = av_rescale_q_rnd(packet.dts,
-                                 ifmt_ctx->streams[stream_index]->time_base,
+                    ifmt_ctx->streams[stream_index]->time_base,
-                                 ofmt_ctx->streams[stream_index]->time_base);
+                    ofmt_ctx->streams[stream_index]->time_base,
                    AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
            packet.pts = av_rescale_q_rnd(packet.pts,
                    ifmt_ctx->streams[stream_index]->time_base,
                    ofmt_ctx->streams[stream_index]->time_base,
                    AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
-            ret = av_interleaved_write_frame(ofmt_ctx, packet);
+            ret = av_interleaved_write_frame(ofmt_ctx, &packet);
            if (ret < 0)
                goto end;
        }
-        av_packet_unref(packet);
+        av_free_packet(&packet);
    }
    /* flush filters and encoders */
@@ -597,24 +575,19 @@ int main(int argc, char **argv)
    av_write_trailer(ofmt_ctx);
 end:
-    av_packet_free(&packet);
+    av_free_packet(&packet);
    av_frame_free(&frame);
    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
-        avcodec_free_context(&stream_ctx[i].dec_ctx);
+        avcodec_close(ifmt_ctx->streams[i]->codec);
-        if (ofmt_ctx && ofmt_ctx->nb_streams > i && ofmt_ctx->streams[i] && stream_ctx[i].enc_ctx)
+        if (ofmt_ctx && ofmt_ctx->nb_streams > i && ofmt_ctx->streams[i] && ofmt_ctx->streams[i]->codec)
-            avcodec_free_context(&stream_ctx[i].enc_ctx);
+            avcodec_close(ofmt_ctx->streams[i]->codec);
-        if (filter_ctx && filter_ctx[i].filter_graph) {
+        if (filter_ctx && filter_ctx[i].filter_graph)
            avfilter_graph_free(&filter_ctx[i].filter_graph);
            av_packet_free(&filter_ctx[i].enc_pkt);
            av_frame_free(&filter_ctx[i].filtered_frame);
        }
        av_frame_free(&stream_ctx[i].dec_frame);
    }
    av_free(filter_ctx);
    av_free(stream_ctx);
    avformat_close_input(&ifmt_ctx);
    if (ofmt_ctx && !(ofmt_ctx->oformat->flags & AVFMT_NOFILE))
-        avio_closep(&ofmt_ctx->pb);
+        avio_close(ofmt_ctx->pb);
    avformat_free_context(ofmt_ctx);
    if (ret < 0)
--- a/doc/examples/vaapi_encode.c
+++ b/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;
    if (!(enc_pkt = av_packet_alloc()))
        return AVERROR(ENOMEM);
    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:
    av_packet_free(&enc_pkt);
    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;
    const 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, 0)) < 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;
 }
--- a/doc/examples/vaapi_transcode.c
+++ b/doc/examples/vaapi_transcode.c
@@ -1,308 +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;
    const AVCodec *decoder = NULL;
    AVStream *video = NULL;
    if ((ret = avformat_open_input(&ifmt_ctx, filename, NULL, NULL)) < 0) {
        fprintf(stderr, "Cannot open input file '%s', Error code: %s\n",
                filename, av_err2str(ret));
        return ret;
    }
    if ((ret = avformat_find_stream_info(ifmt_ctx, NULL)) < 0) {
        fprintf(stderr, "Cannot find input stream information. Error code: %s\n",
                av_err2str(ret));
        return ret;
    }
    ret = av_find_best_stream(ifmt_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, &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(AVPacket *enc_pkt, AVFrame *frame)
 {
    int ret = 0;
    av_packet_unref(enc_pkt);
    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, const 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(pkt, frame)) < 0)
            fprintf(stderr, "Error during encoding and writing.\n");
 fail:
        av_frame_free(&frame);
        if (ret < 0)
            return ret;
    }
    return 0;
 }
 int main(int argc, char **argv)
 {
    const AVCodec *enc_codec;
    int ret = 0;
    AVPacket *dec_pkt;
    if (argc != 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;
    }
    dec_pkt = av_packet_alloc();
    if (!dec_pkt) {
        fprintf(stderr, "Failed to allocate decode packet\n");
        goto end;
    }
    if ((ret = open_input_file(argv[1])) < 0)
        goto end;
    if (!(enc_codec = avcodec_find_encoder_by_name(argv[2]))) {
        fprintf(stderr, "Could not find encoder '%s'\n", argv[2]);
        ret = -1;
        goto end;
    }
    if ((ret = (avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, argv[3]))) < 0) {
        fprintf(stderr, "Failed to deduce output format from file extension. Error code: "
                "%s\n", av_err2str(ret));
        goto end;
    }
    if (!(encoder_ctx = avcodec_alloc_context3(enc_codec))) {
        ret = AVERROR(ENOMEM);
        goto end;
    }
    ret = avio_open(&ofmt_ctx->pb, argv[3], AVIO_FLAG_WRITE);
    if (ret < 0) {
        fprintf(stderr, "Cannot open output file. "
                "Error code: %s\n", av_err2str(ret));
        goto end;
    }
    /* read all packets and only transcoding video */
    while (ret >= 0) {
        if ((ret = av_read_frame(ifmt_ctx, dec_pkt)) < 0)
            break;
        if (video_stream == dec_pkt->stream_index)
            ret = dec_enc(dec_pkt, enc_codec);
        av_packet_unref(dec_pkt);
    }
    /* flush decoder */
    av_packet_unref(dec_pkt);
    ret = dec_enc(dec_pkt, enc_codec);
    /* flush encoder */
    ret = encode_write(dec_pkt, 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);
    av_packet_free(&dec_pkt);
    return ret;
 }
--- a/doc/faq.texi
+++ b/doc/faq.texi
@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg FAQ
@titlepage
@@ -76,7 +75,7 @@ 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...
@@ -91,63 +90,13 @@ To build FFmpeg, you need to install the development package. It is usually
 called @file{libfoo-dev} or @file{libfoo-devel}. You can remove it after the
 build is finished, but be sure to keep the main package.
@section How do I make @command{pkg-config} find my libraries?
 Somewhere along with your libraries, there is a @file{.pc} file (or several)
 in a @file{pkgconfig} directory. You need to set environment variables to
 point @command{pkg-config} to these files.
 If you need to @emph{add} directories to @command{pkg-config}'s search list
 (typical use case: library installed separately), add it to
@code{$PKG_CONFIG_PATH}:
@example
 export PKG_CONFIG_PATH=/opt/x264/lib/pkgconfig:/opt/opus/lib/pkgconfig
@end example
 If you need to @emph{replace} @command{pkg-config}'s search list
 (typical use case: cross-compiling), set it in
@code{$PKG_CONFIG_LIBDIR}:
@example
 export PKG_CONFIG_LIBDIR=/home/me/cross/usr/lib/pkgconfig:/home/me/cross/usr/local/lib/pkgconfig
@end example
 If you need to know the library's internal dependencies (typical use: static
 linking), add the @code{--static} option to @command{pkg-config}:
@example
 ./configure --pkg-config-flags=--static
@end example
@section How do I use @command{pkg-config} when cross-compiling?
 The best way is to install @command{pkg-config} in your cross-compilation
 environment. It will automatically use the cross-compilation libraries.
 You can also use @command{pkg-config} from the host environment by
 specifying explicitly @code{--pkg-config=pkg-config} to @command{configure}.
 In that case, you must point @command{pkg-config} to the correct directories
 using the @code{PKG_CONFIG_LIBDIR}, as explained in the previous entry.
 As an intermediate solution, you can place in your cross-compilation
 environment a script that calls the host @command{pkg-config} with
@code{PKG_CONFIG_LIBDIR} set. That script can look like that:
@example
 #!/bin/sh
 PKG_CONFIG_LIBDIR=/path/to/cross/lib/pkgconfig
 export PKG_CONFIG_LIBDIR
 exec /usr/bin/pkg-config "$@@"
@end example
@chapter Usage
@section ffmpeg does not work; what is wrong?
 Try a @code{make distclean} in the ffmpeg source directory before the build.
 If this does not help see
-(@url{https://ffmpeg.org/bugreports.html}).
+(@url{http://ffmpeg.org/bugreports.html}).
@section How do I encode single pictures into movies?
@@ -257,13 +206,13 @@ 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?
@@ -311,18 +260,18 @@ invoking ffmpeg with several @option{-i} options.
 For audio, to put all channels together in a single stream (example: two
 mono streams into one stereo stream): this is sometimes called to
@emph{merge} them, and can be done using the
-@url{ffmpeg-filters.html#amerge, @code{amerge}} filter.
+@url{http://ffmpeg.org/ffmpeg-filters.html#amerge, @code{amerge}} filter.
@item
 For audio, to play one on top of the other: this is called to @emph{mix}
 them, and can be done by first merging them into a single stream and then
-using the @url{ffmpeg-filters.html#pan, @code{pan}} filter to mix
+using the @url{http://ffmpeg.org/ffmpeg-filters.html#pan, @code{pan}} filter to mix
 the channels at will.
@item
 For video, to display both together, side by side or one on top of a part of
 the other; it can be done using the
-@url{ffmpeg-filters.html#overlay, @code{overlay}} video filter.
+@url{http://ffmpeg.org/ffmpeg-filters.html#overlay, @code{overlay}} video filter.
@end itemize
@@ -333,23 +282,23 @@ There are several solutions, depending on the exact circumstances.
@subsection Concatenating using the concat @emph{filter}
-FFmpeg has a @url{ffmpeg-filters.html#concat,
+FFmpeg has a @url{http://ffmpeg.org/ffmpeg-filters.html#concat,
@code{concat}} filter designed specifically for that, with examples in the
 documentation. This operation is recommended if you need to re-encode.
@subsection Concatenating using the concat @emph{demuxer}
-FFmpeg has a @url{ffmpeg-formats.html#concat,
+FFmpeg has a @url{http://www.ffmpeg.org/ffmpeg-formats.html#concat,
@code{concat}} demuxer which you can use when you want to avoid a re-encode and
 your format doesn't support file level concatenation.
@subsection Concatenating using the concat @emph{protocol} (file level)
-FFmpeg has a @url{ffmpeg-protocols.html#concat,
+FFmpeg has a @url{http://ffmpeg.org/ffmpeg-protocols.html#concat,
@code{concat}} protocol designed specifically for that, with examples in the
 documentation.
-A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow one to concatenate
+A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow to concatenate
 video by merely concatenating the files containing them.
 Hence you may concatenate your multimedia files by first transcoding them to
@@ -385,7 +334,7 @@ mkfifo intermediate2.mpg
 ffmpeg -i input1.avi -qscale:v 1 -y intermediate1.mpg < /dev/null &
 ffmpeg -i input2.avi -qscale:v 1 -y intermediate2.mpg < /dev/null &
 cat intermediate1.mpg intermediate2.mpg |\
-ffmpeg -f mpeg -i - -c:v mpeg4 -c:a libmp3lame output.avi
+ffmpeg -f mpeg -i - -c:v mpeg4 -acodec libmp3lame output.avi
@end example
@subsection Concatenating using raw audio and video
@@ -407,13 +356,13 @@ mkfifo temp2.a
 mkfifo temp2.v
 mkfifo all.a
 mkfifo all.v
-ffmpeg -i input1.flv -vn -f u16le -c:a pcm_s16le -ac 2 -ar 44100 - > temp1.a < /dev/null &
+ffmpeg -i input1.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 - > temp1.a < /dev/null &
-ffmpeg -i input2.flv -vn -f u16le -c:a pcm_s16le -ac 2 -ar 44100 - > temp2.a < /dev/null &
+ffmpeg -i input2.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 - > temp2.a < /dev/null &
 ffmpeg -i input1.flv -an -f yuv4mpegpipe - > temp1.v < /dev/null &
@{ ffmpeg -i input2.flv -an -f yuv4mpegpipe - < /dev/null | tail -n +2 > temp2.v ; @} &
 cat temp1.a temp2.a > all.a &
 cat temp1.v temp2.v > all.v &
-ffmpeg -f u16le -c:a pcm_s16le -ac 2 -ar 44100 -i all.a \
+ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 44100 -i all.a \
       -f yuv4mpegpipe -i all.v \
       -y output.flv
 rm temp[12].[av] all.[av]
@@ -467,105 +416,6 @@ point acceptable for your tastes. The most common options to do that are
@option{-qscale} and @option{-qmax}, but you should peruse the documentation
 of the encoder you chose.
@section I have a stretched video, why does scaling does not fix it?
 A lot of video codecs and formats can store the @emph{aspect ratio} of the
 video: this is the ratio between the width and the height of either the full
 image (DAR, display aspect ratio) or individual pixels (SAR, sample aspect
 ratio). For example, EGA screens at resolution 640×350 had 4:3 DAR and 35:48
 SAR.
 Most still image processing work with square pixels, i.e. 1:1 SAR, but a lot
 of video standards, especially from the analogic-numeric transition era, use
 non-square pixels.
 Most processing filters in FFmpeg handle the aspect ratio to avoid
 stretching the image: cropping adjusts the DAR to keep the SAR constant,
 scaling adjusts the SAR to keep the DAR constant.
 If you want to stretch, or “unstretch”, the image, you need to override the
 information with the
@url{ffmpeg-filters.html#setdar_002c-setsar, @code{setdar or setsar filters}}.
 Do not forget to examine carefully the original video to check whether the
 stretching comes from the image or from the aspect ratio information.
 For example, to fix a badly encoded EGA capture, use the following commands,
 either the first one to upscale to square pixels or the second one to set
 the correct aspect ratio or the third one to avoid transcoding (may not work
 depending on the format / codec / player / phase of the moon):
@example
 ffmpeg -i ega_screen.nut -vf scale=640:480,setsar=1 ega_screen_scaled.nut
 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 +451,7 @@ 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?
@@ -654,7 +504,7 @@ see @file{libavformat/aviobuf.c} in FFmpeg and @file{libmpdemux/demux_lavf.c} in
@section Where is the documentation about ffv1, msmpeg4, asv1, 4xm?
-see @url{https://www.ffmpeg.org/~michael/}
+see @url{http://www.ffmpeg.org/~michael/}
@section How do I feed H.263-RTP (and other codecs in RTP) to libavcodec?
--- a/doc/fate.texi
+++ b/doc/fate.texi
@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Automated Testing Environment
@titlepage
@@ -13,36 +12,36 @@
@chapter Introduction
-FATE is an extended regression suite on the client-side and a means
+  FATE is an extended regression suite on the client-side and a means
 for results aggregation and presentation on the server-side.
-The first part of this document explains how you can use FATE from
+  The first part of this document explains how you can use FATE from
 your FFmpeg source directory to test your ffmpeg binary. The second
 part describes how you can run FATE to submit the results to FFmpeg's
 FATE server.
-In any way you can have a look at the publicly viewable FATE results
+  In any way you can have a look at the publicly viewable FATE results
 by visiting this website:
-@url{http://fate.ffmpeg.org/}
+  @url{http://fate.ffmpeg.org/}
-This is especially recommended for all people contributing source
+  This is especially recommended for all people contributing source
 code to FFmpeg, as it can be seen if some test on some platform broke
 with their recent contribution. This usually happens on the platforms
 the developers could not test on.
-The second part of this document describes how you can run FATE to
+  The second part of this document describes how you can run FATE to
 submit your results to FFmpeg's FATE server. If you want to submit your
 results be sure to check that your combination of CPU, OS and compiler
 is not already listed on the above mentioned website.
-In the third part you can find a comprehensive listing of FATE makefile
+  In the third part you can find a comprehensive listing of FATE makefile
 targets and variables.
@chapter Using FATE from your FFmpeg source directory
-If you want to run FATE on your machine you need to have the samples
+  If you want to run FATE on your machine you need to have the samples
 in place. You can get the samples via the build target fate-rsync.
 Use this command from the top-level source directory:
@@ -51,11 +50,11 @@ make fate-rsync SAMPLES=fate-suite/
 make fate       SAMPLES=fate-suite/
@end example
-The above commands set the samples location by passing a makefile
+  The above commands set the samples location by passing a makefile
 variable via command line. It is also possible to set the samples
 location at source configuration time by invoking configure with
-@option{--samples=<path to the samples directory>}. Afterwards you can
+`--samples=<path to the samples directory>'. Afterwards you can
-invoke the makefile targets without setting the @var{SAMPLES} makefile
+invoke the makefile targets without setting the SAMPLES makefile
 variable. This is illustrated by the following commands:
@example
@@ -64,7 +63,7 @@ make fate-rsync
 make fate
@end example
-Yet another way to tell FATE about the location of the sample
+  Yet another way to tell FATE about the location of the sample
 directory is by making sure the environment variable FATE_SAMPLES
 contains the path to your samples directory. This can be achieved
 by e.g. putting that variable in your shell profile or by setting
@@ -79,28 +78,13 @@ Do not put a '~' character in the samples path to indicate a home
 directory. Because of shell nuances, this will cause FATE to fail.
@end float
 To get the complete list of tests, run the command:
@example
 make fate-list
@end example
 You can specify a subset of tests to run by specifying the
 corresponding elements from the list with the @code{fate-} prefix,
 e.g. as in:
@example
 make fate-ffprobe_compact fate-ffprobe_xml
@end example
 This makes it easier to run a few tests in case of failure without
 running the complete test suite.
 To use a custom wrapper to run the test, pass @option{--target-exec} to
@command{configure} or set the @var{TARGET_EXEC} Make variable.
@chapter Submitting the results to the FFmpeg result aggregation server
-To submit your results to the server you should run fate through the
+  To submit your results to the server you should run fate through the
 shell script @file{tests/fate.sh} from the FFmpeg sources. This script needs
 to be invoked with a configuration file as its first argument.
@@ -108,23 +92,23 @@ to be invoked with a configuration file as its first argument.
 tests/fate.sh /path/to/fate_config
@end example
-A configuration file template with comments describing the individual
+  A configuration file template with comments describing the individual
 configuration variables can be found at @file{doc/fate_config.sh.template}.
@ifhtml
-The mentioned configuration template is also available here:
+  The mentioned configuration template is also available here:
@verbatiminclude fate_config.sh.template
@end ifhtml
-Create a configuration that suits your needs, based on the configuration
+  Create a configuration that suits your needs, based on the configuration
-template. The @env{slot} configuration variable can be any string that is not
+template. The `slot' configuration variable can be any string that is not
 yet used, but it is suggested that you name it adhering to the following
-pattern @samp{@var{arch}-@var{os}-@var{compiler}-@var{compiler version}}. The
+pattern <arch>-<os>-<compiler>-<compiler version>. The configuration file
-configuration file itself will be sourced in a shell script, therefore all
+itself will be sourced in a shell script, therefore all shell features may
-shell features may be used. This enables you to setup the environment as you
+be used. This enables you to setup the environment as you need it for your
-need it for your build.
+build.
-For your first test runs the @env{fate_recv} variable should be empty or
+  For your first test runs the `fate_recv' variable should be empty or
 commented out. This will run everything as normal except that it will omit
 the submission of the results to the server. The following files should be
 present in $workdir as specified in the configuration file:
@@ -137,57 +121,31 @@ present in $workdir as specified in the configuration file:
    @item version
@end itemize
-When you have everything working properly you can create an SSH key pair
+  When you have everything working properly you can create an SSH key pair
 and send the public key to the FATE server administrator who can be contacted
 at the email address @email{fate-admin@@ffmpeg.org}.
-Configure your SSH client to use public key authentication with that key
+  Configure your SSH client to use public key authentication with that key
 when connecting to the FATE server. Also do not forget to check the identity
 of the server and to accept its host key. This can usually be achieved by
 running your SSH client manually and killing it after you accepted the key.
 The FATE server's fingerprint is:
-@table @samp
+@table @option
@item RSA
   d3:f1:83:97:a4:75:2b:a6:fb:d6:e8:aa:81:93:97:51
@item ECDSA
   76:9f:68:32:04:1e:d5:d4:ec:47:3f:dc:fc:18:17:86
@end table
-If you have problems connecting to the FATE server, it may help to try out
+  If you have problems connecting to the FATE server, it may help to try out
 the @command{ssh} command with one or more @option{-v} options. You should
 get detailed output concerning your SSH configuration and the authentication
 process.
-The only thing left is to automate the execution of the fate.sh script and
+  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
 If you need a sample uploaded send a mail to samples-request.
 This is for developers who have an account on the fate suite server.
 If you upload new samples, please make sure they are as small as possible,
 space on each client, network bandwidth and so on benefit from smaller test cases.
 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.
 If you need an account for frequently uploading samples or you wish to help
 others by doing that send a mail to ffmpeg-devel.
@example
 #First update your local samples copy:
 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
@@ -206,7 +164,7 @@ Run the FATE test suite (requires the fate-suite dataset).
@section Makefile variables
-@table @env
+@table @option
@item V
 Verbosity level, can be set to 0, 1 or 2.
    @itemize
@@ -224,30 +182,20 @@ Specify how many threads to use while running regression tests, it is
 quite useful to detect thread-related regressions.
@item THREAD_TYPE
-Specify which threading strategy test, either @samp{slice} or @samp{frame},
+Specify which threading strategy test, either @var{slice} or @var{frame},
-by default @samp{slice+frame}
+by default @var{slice+frame}
@item CPUFLAGS
 Specify CPU flags.
@item TARGET_EXEC
 Specify or override the wrapper used to run the tests.
-The @env{TARGET_EXEC} option provides a way to run FATE wrapped in
+The @var{TARGET_EXEC} option provides a way to run FATE wrapped in
@command{valgrind}, @command{qemu-user} or @command{wine} or on remote targets
 through @command{ssh}.
@item GEN
-Set to @samp{1} to generate the missing or mismatched references.
+Set to @var{1} to generate the missing or mismatched references.
@item HWACCEL
 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
--- a/doc/fate_config.sh.template
+++ b/doc/fate_config.sh.template
@@ -1,12 +1,10 @@
 slot=                                    # some unique identifier
 repo=git://source.ffmpeg.org/ffmpeg.git  # the source repository
 #branch=release/2.6                       # the branch to test
 samples=                                 # path to samples directory
 workdir=                                 # directory in which to do all the work
 #fate_recv="ssh -T fate@fate.ffmpeg.org" # command to submit report
 comment=                                 # optional description
 build_only=     # set to "yes" for a compile-only instance that skips tests
 ignore_tests=
 # the following are optional and map to configure options
 arch=
@@ -27,7 +25,5 @@ extra_conf=     # extra configure options not covered above
 #make=          # name of GNU make if not 'make'
 makeopts=       # extra options passed to 'make'
 #makeopts_fate= # extra options passed to 'make' when running tests,
                # defaulting to makeopts above if this is not set
 #tar=           # command to create a tar archive from its arguments on stdout,
                # defaults to 'tar c'
--- a/doc/ffmpeg-bitstream-filters.texi
+++ b/doc/ffmpeg-bitstream-filters.texi
@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Bitstream Filters Documentation
@titlepage
@@ -26,12 +25,12 @@ 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
--- a/Show More
+++ b/Show More
		`@@ -1,2 +0,0 @@`
			`*.pnm -diff -text`
			`tests/ref/fate/sub-scc eol=crlf`
		`@@ -0,0 +1,2 @@`
							`#!/bin/sh`
							`exec awk "BEGIN { for (i = 2; i < ARGC; i++) printf \"$1\", ARGV[i] }" "$@"`