update for 2.0.4

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
avcodec/utvideoenc: fix slice_bits size
2025-12-07 15:30:00 +01:00 · 2014-03-04 01:20:37 +01:00 · 2014-03-01 20:28:49 +01:00 · 2014-02-27 00:33:00 +01:00 · 2014-02-18 05:10:13 +01:00 · 2014-02-17 20:56:24 +01:00
4752 changed files with 173583 additions and 600794 deletions
--- a/.gitattributes
+++ b/.gitattributes
@@ -1 +0,0 @@
-*.pnm -diff -text
--- a/.gitignore
+++ b/.gitignore
@@ -1,6 +1,5 @@
 *.a
 *.o
-*.o.*
 *.d
 *.def
 *.dll
@@ -16,13 +15,10 @@
 *.pdb
 *.so
 *.so.*
-*.swp
 *.ver
 *-example
 *-test
 *_g
-\#*
-.\#*
 /.config
 /.version
 /ffmpeg
@@ -31,7 +27,7 @@
 /ffserver
 /config.*
 /coverage.info
-/avversion.h
+/version.h
 /doc/*.1
 /doc/*.3
 /doc/*.html
@@ -39,39 +35,27 @@
 /doc/config.texi
 /doc/avoptions_codec.texi
 /doc/avoptions_format.texi
-/doc/doxy/html/
-/doc/examples/avio_dir_cmd
-/doc/examples/avio_reading
 /doc/examples/decoding_encoding
-/doc/examples/demuxing_decoding
-/doc/examples/extract_mvs
-/doc/examples/filter_audio
+/doc/examples/demuxing
 /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/doxy/html/
 /doc/print_options
 /lcov/
 /libavcodec/*_tablegen
 /libavcodec/*_tables.c
 /libavcodec/*_tables.h
 /libavutil/avconfig.h
-/libavutil/ffversion.h
-/src
 /tests/audiogen
 /tests/base64
-/tests/checkasm/checkasm
 /tests/data/
-/tests/pixfmts.mak
 /tests/rotozoom
-/tests/test_copy.ffmeta
 /tests/tiny_psnr
 /tests/tiny_ssim
 /tests/videogen
@@ -79,7 +63,6 @@
 /tools/aviocat
 /tools/ffbisect
 /tools/bisect.need
-/tools/crypto_bench
 /tools/cws2fws
 /tools/fourcc2pixfmt
 /tools/ffescape
@@ -90,8 +73,6 @@
 /tools/pktdumper
 /tools/probetest
 /tools/qt-faststart
-/tools/sidxindex
 /tools/trasher
 /tools/seek_print
-/tools/uncoded_frame
 /tools/zmqsend
--- a/.travis.yml
+++ b/.travis.yml
@@ -1,26 +0,0 @@
-language: c
-sudo: false
-os:
-  - linux
-  - osx
-addons:
-  apt:
-    packages:
-      - yasm
-      - diffutils
-compiler:
-  - clang
-  - gcc
-cache:
-  directories:
-    - ffmpeg-samples
-before_install:
-  - if [ "$TRAVIS_OS_NAME" == "osx" ]; then brew update --all; fi
-install:
-  - if [ "$TRAVIS_OS_NAME" == "osx" ]; then brew install yasm; fi
-script:
-  - mkdir -p ffmpeg-samples
-  - ./configure --samples=ffmpeg-samples --cc=$CC
-  - make -j 8
-  - make fate-rsync
-  - make check -j 8
--- a/346
+++ b/346
@@ -1,333 +1,6 @@
 Entries are sorted chronologically from oldest to youngest within each release,
 releases are sorted from youngest to oldest.

-version <next>:
-
-
-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
- shuffleplanes filter
- subfile protocol
- Phantom Cine demuxer
- replaygain data export
- VP7 video decoder
- Alias PIX image encoder and decoder
- Improvements to the BRender PIX image decoder
- Improvements to the XBM decoder
- QTKit input device
- improvements to OpenEXR image decoder
- support decoding 16-bit RLE SGI images
- GDI screen grabbing for Windows
- alternative rendition support for HTTP Live Streaming
- AVFoundation input device
- Direct Stream Digital (DSD) decoder
- Magic Lantern Video (MLV) demuxer
- On2 AVC (Audio for Video) decoder
- support for decoding through DXVA2 in ffmpeg
- libbs2b-based stereo-to-binaural audio filter
- libx264 reference frames count limiting depending on level
- native Opus decoder
- display matrix export and rotation API
- WebVTT encoder
- showcqt multimedia filter
- zoompan filter
- signalstats filter
- hqx filter (hq2x, hq3x, hq4x)
- flanger filter
- Image format auto-detection
- LRC demuxer and muxer
- Samba protocol (via libsmbclient)
- WebM DASH Manifest muxer
- libfribidi support in drawtext
-
-
-version 2.2:
-
- HNM version 4 demuxer and video decoder
- Live HDS muxer
- setsar/setdar filters now support variables in ratio expressions
- elbg filter
- string validation in ffprobe
- support for decoding through VDPAU in ffmpeg (the -hwaccel option)
- complete Voxware MetaSound decoder
- remove mp3_header_compress bitstream filter
- Windows resource files for shared libraries
- aeval filter
- stereoscopic 3d metadata handling
- WebP encoding via libwebp
- ATRAC3+ decoder
- VP8 in Ogg demuxing
- side & metadata support in NUT
- framepack filter
- XYZ12 rawvideo support in NUT
- Exif metadata support in WebP decoder
- OpenGL device
- Use metadata_header_padding to control padding in ID3 tags (currently used in
-  MP3, AIFF, and OMA files), FLAC header, and the AVI "junk" block.
- Mirillis FIC video decoder
- Support DNx444
- libx265 encoder
- dejudder filter
- Autodetect VDA like all other hardware accelerations
- aliases and defaults for Ogg subtypes (opus, spx)
-
-
-version 2.1:
-
- aecho filter
- perspective filter ported from libmpcodecs
- ffprobe -show_programs option
- compand filter
- RTMP seek support
- when transcoding with ffmpeg (i.e. not streamcopying), -ss is now accurate
-  even when used as an input option. Previous behavior can be restored with
-  the -noaccurate_seek option.
- ffmpeg -t option can now be used for inputs, to limit the duration of
-  data read from an input file
- incomplete Voxware MetaSound decoder
- read EXIF metadata from JPEG
- DVB teletext decoder
- phase filter ported from libmpcodecs
- w3fdif filter
- Opus support in Matroska
- FFV1 version 1.3 is stable and no longer experimental
- FFV1: YUVA(444,422,420) 9, 10 and 16 bit support
- changed DTS stream id in lavf mpeg ps muxer from 0x8a to 0x88, to be
-  more consistent with other muxers.
- adelay filter
- pullup filter ported from libmpcodecs
- ffprobe -read_intervals option
- Lossless and alpha support for WebP decoder
- Error Resilient AAC syntax (ER AAC LC) decoding
- Low Delay AAC (ER AAC LD) decoding
- mux chapters in ASF files
- SFTP protocol (via libssh)
- libx264: add ability to encode in YUVJ422P and YUVJ444P
- Fraps: use BT.709 colorspace by default for yuv, as reference fraps decoder does
- make decoding alpha optional for prores, ffv1 and vp6 by setting
-  the skip_alpha flag.
- ladspa wrapper filter
- native VP9 decoder
- dpx parser
- max_error_rate parameter in ffmpeg
- PulseAudio output device
- ReplayGain scanner
- Enhanced Low Delay AAC (ER AAC ELD) decoding (no LD SBR support)
- Linux framebuffer output device
- HEVC decoder
- raw HEVC, HEVC in MOV/MP4, HEVC in Matroska, HEVC in MPEG-TS demuxing
- mergeplanes filter
-
-
 version 2.0:

 - curves filter
@@ -340,7 +13,7 @@ version 2.0:
 - 10% faster aac encoding on x86 and MIPS
 - sine audio filter source
 - WebP demuxing and decoding support
- ffmpeg options -filter_script and -filter_complex_script, which allow a
+- new ffmpeg options -filter_script and -filter_complex_script, which allow a
  filtergraph description to be read from a file
 - OpenCL support
 - audio phaser filter
@@ -348,7 +21,7 @@ version 2.0:
 - libquvi demuxer
 - uniform options syntax across all filters
 - telecine filter
- interlace filter
+- new interlace filter
 - smptehdbars source
 - inverse telecine filters (fieldmatch and decimate)
 - colorbalance filter
@@ -466,7 +139,7 @@ version 1.1:
 - JSON captions for TED talks decoding support
 - SOX Resampler support in libswresample
 - aselect filter
- SGI RLE 8-bit / Silicon Graphics RLE 8-bit video decoder
+- SGI RLE 8-bit decoder
 - Silicon Graphics Motion Video Compressor 1 & 2 decoder
 - Silicon Graphics Movie demuxer
 - apad filter
@@ -510,9 +183,7 @@ version 1.0:
 - RTMPE protocol support
 - RTMPTE protocol support
 - showwaves and showspectrum filter
- LucasArts SMUSH SANM playback support
- LucasArts SMUSH VIMA audio decoder (ADPCM)
- LucasArts SMUSH demuxer
+- LucasArts SMUSH playback support
 - SAMI, RealText and SubViewer demuxers and decoders
 - Heart Of Darkness PAF playback support
 - iec61883 device
@@ -636,7 +307,6 @@ version 0.10:
 - ffwavesynth decoder
 - aviocat tool
 - ffeval tool
- support encoding and decoding 4-channel SGI images


 version 0.9:
@@ -685,7 +355,7 @@ 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
@@ -937,7 +607,7 @@ version 0.6:
 - LPCM support in MPEG-TS (HDMV RID as found on Blu-ray disks)
 - WMA Pro decoder
 - Core Audio Format demuxer
- ATRAC1 decoder
+- Atrac1 decoder
 - MD STUDIO audio demuxer
 - RF64 support in WAV demuxer
 - MPEG-4 Audio Lossless Coding (ALS) decoder
@@ -1037,7 +707,7 @@ version 0.5:
 - MXF demuxer
 - VC-1/WMV3/WMV9 video decoder
 - MacIntel support
- AviSynth support
+- AVISynth support
 - VMware video decoder
 - VP5 video decoder
 - VP6 video decoder
@@ -1065,7 +735,7 @@ version 0.5:
 - Interplay C93 demuxer and video decoder
 - Bethsoft VID demuxer and video decoder
 - CRYO APC demuxer
- ATRAC3 decoder
+- Atrac3 decoder
 - V.Flash PTX decoder
 - RoQ muxer, RoQ audio encoder
 - Renderware TXD demuxer and decoder
--- a/15
+++ b/15
@@ -0,0 +1,15 @@
+
+1) Type './configure' to create the configuration. A list of configure
+options is printed by running 'configure --help'.
+
+'configure' can be launched from a directory different from the FFmpeg
+sources to build the objects out of tree. To do this, use an absolute
+path when launching 'configure', e.g. '/ffmpegdir/ffmpeg/configure'.
+
+2) Then type 'make' to build FFmpeg. GNU Make 3.81 or later is required.
+
+3) Type 'make install' to install all binaries and libraries you built.
+
+NOTICE
+
+ - Non system dependencies (e.g. libx264, libvpx) are disabled by default.
--- a/INSTALL.md
+++ b/INSTALL.md
@@ -1,17 +0,0 @@
-#Installing FFmpeg:
-
-1. Type `./configure` to create the configuration. A list of configure
-options is printed by running `configure --help`.
-
-    `configure` can be launched from a directory different from the FFmpeg
-sources to build the objects out of tree. To do this, use an absolute
-path when launching `configure`, e.g. `/ffmpegdir/ffmpeg/configure`.
-
-2. Then type `make` to build FFmpeg. GNU Make 3.81 or later is required.
-
-3. Type `make install` to install all binaries and libraries you built.
-
-NOTICE
------
-
- - Non system dependencies (e.g. libx264, libvpx) are disabled by default.
--- a/102
+++ b/102
@@ -0,0 +1,102 @@
+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
+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
+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
+- libmpcodecs
+- optional x86 optimizations in the files
+  libavcodec/x86/idct_mmx.c
+- libutvideo encoding/decoding wrappers in
+  libavcodec/libutvideo*.cpp
+- the X11 grabber in libavdevice/x11grab.c
+- the swresample test app in
+  libswresample/swresample-test.c
+- the texi2pod.pl tool
+- the following filters in libavfilter:
+    - f_ebur128.c
+    - vf_blackframe.c
+    - vf_boxblur.c
+    - vf_colormatrix.c
+    - vf_cropdetect.c
+    - vf_decimate.c
+    - vf_delogo.c
+    - vf_geq.c
+    - vf_histeq.c
+    - vf_hqdn3d.c
+    - vf_hue.c
+    - vf_kerndeint.c
+    - vf_mcdeint.c
+    - vf_mp.c
+    - vf_noise.c
+    - vf_owdenoise.c
+    - vf_pp.c
+    - vf_sab.c
+    - vf_smartblur.c
+    - vf_spp.c
+    - vf_stereo3d.c
+    - vf_super2xsai.c
+    - vf_tinterlace.c
+    - vf_yadif.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
+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.
+
+There are a handful of files under other licensing terms, namely:
+
+* The files libavcodec/jfdctfst.c, libavcodec/jfdctint_template.c and
+  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.
+
+
+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
+--------------------
+
+The following libraries are under GPL:
+    - frei0r
+    - libcdio
+    - libutvideo
+    - libvidstab
+    - libx264
+    - libxavs
+    - libxvid
+When combining them with FFmpeg, FFmpeg needs to be licensed as GPL as well by
+passing --enable-gpl to configure.
+
+The OpenCORE and VisualOn libraries are under the Apache License 2.0. That
+license is incompatible with the LGPL v2.1 and the GPL v2, but not with
+version 3 of those licenses. So to combine these libraries with FFmpeg, the
+license version needs to be upgraded by passing --enable-version3 to configure.
+
+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
+licenses are compatible with the LGPL.
+If you wish to enable these libraries, pass --enable-nonfree to configure.
+But note that if you enable any of these libraries the resulting binary will
+be under a complex license mix that is more restrictive than the LGPL and that
+may result in additional obligations. It is possible that these
+restrictions cause the resulting binary to be unredistributeable.
--- a/LICENSE.md
+++ b/LICENSE.md
@@ -1,114 +0,0 @@
-#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
-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
-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 optimizations in the files
-  - `libavcodec/x86/flac_dsp_gpl.asm`
-  - `libavcodec/x86/idct_mmx.c`
-  - `libavfilter/x86/vf_removegrain.asm`
- libutvideo encoding/decoding wrappers in
-  `libavcodec/libutvideo*.cpp`
- the X11 grabber in `libavdevice/x11grab.c`
- the swresample test app in
-  `libswresample/swresample-test.c`
- the `texi2pod.pl` tool
- the following filters in libavfilter:
-    - `f_ebur128.c`
-    - `vf_blackframe.c`
-    - `vf_boxblur.c`
-    - `vf_colormatrix.c`
-    - `vf_cover_rect.c`
-    - `vf_cropdetect.c`
-    - `vf_delogo.c`
-    - `vf_eq.c`
-    - `vf_find_rect.c`
-    - `vf_fspp.c`
-    - `vf_geq.c`
-    - `vf_histeq.c`
-    - `vf_hqdn3d.c`
-    - `vf_interlace.c`
-    - `vf_kerndeint.c`
-    - `vf_mcdeint.c`
-    - `vf_mpdecimate.c`
-    - `vf_owdenoise.c`
-    - `vf_perspective.c`
-    - `vf_phase.c`
-    - `vf_pp.c`
-    - `vf_pp7.c`
-    - `vf_pullup.c`
-    - `vf_sab.c`
-    - `vf_smartblur.c`
-    - `vf_repeatfields.c`
-    - `vf_spp.c`
-    - `vf_stereo3d.c`
-    - `vf_super2xsai.c`
-    - `vf_tinterlace.c`
-    - `vf_uspp.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
-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.
-
-There are a handful of files under other licensing terms, namely:
-
-* The files `libavcodec/jfdctfst.c`, `libavcodec/jfdctint_template.c` and
-  `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
-==================
-
-FFmpeg can be combined with a number of external libraries, which sometimes
-affect the licensing of binaries resulting from the combination.
-
-compatible libraries
--------------------
-
-The following libraries are under GPL:
- frei0r
- libcdio
- librubberband
- 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.
-
-The OpenCORE and VisualOn libraries are under the Apache License 2.0. That
-license is incompatible with the LGPL v2.1 and the GPL v2, but not with
-version 3 of those licenses. So to combine these libraries with FFmpeg, the
-license version needs to be upgraded by passing `--enable-version3` to configure.
-
-incompatible libraries
----------------------
-
-The Fraunhofer AAC library and FAAC are under licenses which
-are incompatible with the GPLv2 and v3. We do not know for certain if their
-licenses are compatible with the LGPL.
-If you wish to enable these libraries, pass `--enable-nonfree` to configure.
-But note that if you enable any of these libraries the resulting binary will
-be under a complex license mix that is more restrictive than the LGPL and that
-may result in additional obligations. It is possible that these
-restrictions cause the resulting binary to be unredistributeable.
--- a/177
+++ b/177
@@ -14,6 +14,7 @@ patches and related discussions.
 Project Leader
 ==============

+Michael Niedermayer
  final design decisions


@@ -30,7 +31,7 @@ ffprobe:
  ffprobe.c                             Stefano Sabatini

 ffserver:
-  ffserver.c                            Reynaldo H. Verdejo Pinochet
+  ffserver.c, ffserver.h                Baptiste Coudurier

 Commandline utility code:
  cmdutils.c, cmdutils.h                Michael Niedermayer
@@ -42,26 +43,16 @@ QuickTime faststart:
 Miscellaneous Areas
 ===================

-documentation                           Stefano Sabatini, Mike Melanson, Timothy Gu, Lou Logan
-build system (configure, makefiles)     Diego Biurrun, Mans Rullgard
-project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser, Lou Logan
+documentation                           Mike Melanson
+website                                 Robert Swain, Lou Logan
+build system (configure,Makefiles)      Diego Biurrun, Mans Rullgard
+project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Döffinger, Alexander Strasser
+mailinglists                            Michael Niedermayer, Baptiste Coudurier, Lou Logan
 presets                                 Robert Swain
 metadata subsystem                      Aurelien Jacobs
 release management                      Michael Niedermayer


-Communication
-=============
-
-website                                 Deby Barbara Lepage
-fate.ffmpeg.org                         Timothy Gu
-Trac bug tracker                        Alexander Strasser, Michael Niedermayer, Carl Eugen Hoyos, Lou Logan
-mailing lists                           Michael Niedermayer, Baptiste Coudurier, Lou Logan
-Google+                                 Paul B Mahol, Michael Niedermayer, Alexander Strasser
-Twitter                                 Lou Logan, Reynaldo H. Verdejo Pinochet
-Launchpad                               Timothy Gu
-
-
 libavutil
 =========

@@ -71,21 +62,16 @@ Internal Interfaces:
  libavutil/common.h                    Michael Niedermayer

 Other:
-  aes_ctr.c, aes_ctr.h                  Eran Kornblau
  bprint                                Nicolas George
  bswap.h
  des                                   Reimar Doeffinger
-  dynarray.h                            Nicolas George
-  eval.c, eval.h                        Michael Niedermayer
  float_dsp                             Loren Merritt
  hash                                  Reimar Doeffinger
  intfloat*                             Michael Niedermayer
  integer.c, integer.h                  Michael Niedermayer
  lzo                                   Reimar Doeffinger
  mathematics.c, mathematics.h          Michael Niedermayer
-  mem.c, mem.h                          Michael Niedermayer
  opencl.c, opencl.h                    Wei Gao
-  opt.c, opt.h                          Michael Niedermayer
  rational.c, rational.h                Michael Niedermayer
  rc4                                   Reimar Doeffinger
  ripemd.c, ripemd.h                    James Almer
@@ -100,6 +86,10 @@ Generic Parts:
    avcodec.h                           Michael Niedermayer
  utility code:
    utils.c                             Michael Niedermayer
+    mem.c                               Michael Niedermayer
+    opt.c, opt.h                        Michael Niedermayer
+  arithmetic expression evaluator:
+    eval.c                              Michael Niedermayer
  audio and video frame extraction:
    parser.c                            Michael Niedermayer
  bitstream reading:
@@ -130,15 +120,11 @@ Generic Parts:
    libpostproc/*                       Michael Niedermayer
  table generation:
    tableprint.c, tableprint.h          Reimar Doeffinger
-  fixed point FFT:
-    fft*                                Zeljko Lukac
-  Text Subtitles                        Clément Bœsch

 Codecs:
  4xm.c                                 Michael Niedermayer
  8bps.c                                Roberto Togni
  8svx.c                                Jaikrishnan Menon
-  aacenc*, aaccoder.c                   Rostislav Pehlivanov
  aasc.c                                Kostya Shishkov
  ac3*                                  Justin Ruggles
  alacenc.c                             Jaikrishnan Menon
@@ -147,7 +133,6 @@ Codecs:
  ass*                                  Aurelien Jacobs
  asv*                                  Michael Niedermayer
  atrac3*                               Benjamin Larsson
-  atrac3plus*                           Maxim Poliakovski
  bgmc.c, bgmc.h                        Thilo Borgmann
  bink.c                                Kostya Shishkov
  binkaudio.c                           Peter Ross
@@ -156,8 +141,6 @@ Codecs:
  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
  cljr                                  Alex Beregszaszi
  cllc.c                                Derek Buitenhuis
  cook.c, cookdata.h                    Benjamin Larsson
@@ -165,19 +148,13 @@ Codecs:
  crystalhd.c                           Philip Langdale
  cscd.c                                Reimar Doeffinger
  dca.c                                 Kostya Shishkov, Benjamin Larsson
-  dirac*                                Rostislav Pehlivanov
  dnxhd*                                Baptiste Coudurier
  dpcm.c                                Mike Melanson
-  dss_sp.c                              Oleksij Rempel, Michael Niedermayer
  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
+  ffv1.c                                Michael Niedermayer
  ffwavesynth.c                         Nicolas George
-  fic.c                                 Derek Buitenhuis
  flac*                                 Justin Ruggles
  flashsv*                              Benjamin Larsson
  flicvideo.c                           Mike Melanson
@@ -187,8 +164,7 @@ Codecs:
  h261*                                 Michael Niedermayer
  h263*                                 Michael Niedermayer
  h264*                                 Loren Merritt, Michael Niedermayer
-  hap*                                  Tom Butterworth
-  huffyuv*                              Michael Niedermayer, Christophe Gisquet
+  huffyuv.c                             Michael Niedermayer
  idcinvideo.c                          Mike Melanson
  imc*                                  Benjamin Larsson
  indeo2*                               Kostya Shishkov
@@ -204,19 +180,15 @@ Codecs:
  libcelt_dec.c                         Nicolas George
  libdirac*                             David Conrad
  libgsm.c                              Michel Bardiaux
-  libkvazaar.c                          Arttu Ylä-Outinen
  libopenjpeg.c                         Jaikrishnan Menon
  libopenjpegenc.c                      Michael Bradshaw
  libschroedinger*                      David Conrad
  libspeexdec.c                         Justin Ruggles
  libtheoraenc.c                        David Conrad
-  libutvideo*                           Carl Eugen Hoyos
+  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
-  libzvbi-teletextdec.c                 Marton Balint
  loco.c                                Kostya Shishkov
  lzo.h, lzo.c                          Reimar Doeffinger
  mdec.c                                Michael Niedermayer
@@ -233,7 +205,6 @@ Codecs:
  msvideo1.c                            Mike Melanson
  nellymoserdec.c                       Benjamin Larsson
  nuv.c                                 Reimar Doeffinger
-  nvenc.c                               Timo Rothenpieler
  paf.*                                 Paul B Mahol
  pcx.c                                 Ivo van Poorten
  pgssubdec.c                           Reimar Doeffinger
@@ -242,7 +213,6 @@ Codecs:
  qdm2.c, qdm2data.h                    Roberto Togni, Benjamin Larsson
  qdrw.c                                Kostya Shishkov
  qpeg.c                                Kostya Shishkov
-  qsv*                                  Ivan Uskov
  qtrle.c                               Mike Melanson
  ra144.c, ra144.h, ra288.c, ra288.h    Roberto Togni
  resample2.c                           Michael Niedermayer
@@ -251,12 +221,12 @@ Codecs:
  rtjpeg.c, rtjpeg.h                    Reimar Doeffinger
  rv10.c                                Michael Niedermayer
  rv3*                                  Kostya Shishkov
-  rv4*                                  Kostya Shishkov, Christophe Gisquet
+  rv4*                                  Kostya Shishkov
  s3tc*                                 Ivo van Poorten
  smacker.c                             Kostya Shishkov
  smc.c                                 Mike Melanson
  smvjpegdec.c                          Ash Hughes
-  snow*                                 Michael Niedermayer, Loren Merritt
+  snow.c                                Michael Niedermayer, Loren Merritt
  sonic.c                               Alex Beregszaszi
  srt*                                  Aurelien Jacobs
  sunrast.c                             Ivo van Poorten
@@ -275,19 +245,17 @@ Codecs:
  v410*.c                               Derek Buitenhuis
  vb.c                                  Kostya Shishkov
  vble.c                                Derek Buitenhuis
-  vc1*                                  Kostya Shishkov, Christophe Gisquet
-  vc2*                                  Rostislav Pehlivanov
+  vc1*                                  Kostya Shishkov
  vcr1.c                                Michael Niedermayer
  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
+  vorbis_dec.c                          Denes Balatoni, David Conrad
+  vorbis_enc.c                          Oded Shimon
  vp3*                                  Mike Melanson
  vp5                                   Aurelien Jacobs
  vp6                                   Aurelien Jacobs
  vp8                                   David Conrad, Jason Garrett-Glaser, Ronald Bultje
-  vp9                                   Ronald Bultje, Clément Bœsch
  vqavideo.c                            Mike Melanson
  wavpack.c                             Kostya Shishkov
  wmaprodec.c                           Sascha Sommer
@@ -296,7 +264,6 @@ Codecs:
  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
@@ -305,11 +272,11 @@ Codecs:

 Hardware acceleration:
  crystalhd.c                           Philip Langdale
-  dxva2*                                Hendrik Leppkes, Laurent Aimar
+  dxva2*                                Laurent Aimar
+  libstagefright.cpp                    Mohamed Naufal
  vaapi*                                Gwenole Beauchesne
  vda*                                  Sebastien Zwickert
-  vdpau*                                Philip Langdale, Carl Eugen Hoyos
-  videotoolbox*                         Sebastien Zwickert
+  vdpau*                                Carl Eugen Hoyos


 libavdevice
@@ -318,21 +285,11 @@ libavdevice
    libavdevice/avdevice.h


-  avfoundation.m                        Thilo Borgmann
-  decklink*                             Deti Fliegl
-  dshow.c                               Roger Pack (CC rogerdpack@gmail.com)
-  fbdev_enc.c                           Lukasz Marek
-  gdigrab.c                             Roger Pack (CC rogerdpack@gmail.com)
+  dshow.c                               Roger Pack
  iec61883.c                            Georg Lippitsch
-  lavfi                                 Stefano Sabatini
  libdc1394.c                           Roman Shaposhnik
-  opengl_enc.c                          Lukasz Marek
-  pulse_audio_enc.c                     Lukasz Marek
-  qtkit.m                               Thilo Borgmann
-  sdl                                   Stefano Sabatini
-  v4l2.c                                Giorgio Vazzana
+  v4l2.c                                Luca Abeni
  vfwcap.c                              Ramiro Polla
-  xv.c                                  Lukasz Marek

 libavfilter
 ===========
@@ -341,52 +298,15 @@ Generic parts:
  graphdump.c                           Nicolas George

 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
  af_amerge.c                           Nicolas George
-  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_ladspa.c                           Paul B Mahol
  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_chromakey.c                        Timo Rothenpieler
-  vf_colorchannelmixer.c                Paul B Mahol
-  vf_colorbalance.c                     Paul B Mahol
-  vf_colorkey.c                         Timo Rothenpieler
-  vf_colorlevels.c                      Paul B Mahol
-  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_lenscorrection.c                   Daniel Oberhoff
-  vf_mergeplanes.c                      Paul B Mahol
-  vf_neighbor.c                         Paul B Mahol
-  vf_psnr.c                             Paul B Mahol
-  vf_random.c                           Paul B Mahol
  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_yadif.c                            Michael Niedermayer
-  vf_zoompan.c                          Paul B Mahol

 Sources:
  vsrc_mandelbrot.c                     Michael Niedermayer
@@ -403,13 +323,11 @@ Generic parts:

 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
  ape.c                                 Kostya Shishkov
-  apngdec.c                             Benoit Fouet
  ass*                                  Aurelien Jacobs
  astdec.c                              Paul B Mahol
  astenc.c                              James Almer
@@ -422,7 +340,6 @@ Muxers/Demuxers:
  cdxl.c                                Paul B Mahol
  crc.c                                 Michael Niedermayer
  daud.c                                Reimar Doeffinger
-  dss.c                                 Oleksij Rempel, Michael Niedermayer
  dtshddec.c                            Paul B Mahol
  dv.c                                  Roman Shaposhnik
  dxa.c                                 Kostya Shishkov
@@ -434,8 +351,6 @@ Muxers/Demuxers:
  flvdec.c, flvenc.c                    Michael Niedermayer
  gxf.c                                 Reimar Doeffinger
  gxfenc.c                              Baptiste Coudurier
-  hls.c                                 Anssi Hannula
-  hls encryption (hlsenc.c)             Christian Suloway
  idcin.c                               Mike Melanson
  idroqdec.c                            Mike Melanson
  iff.c                                 Jaikrishnan Menon
@@ -453,31 +368,27 @@ Muxers/Demuxers:
  matroska.c                            Aurelien Jacobs
  matroskadec.c                         Aurelien Jacobs
  matroskaenc.c                         David Conrad
-  matroska subtitles (matroskaenc.c)    John Peebles
  metadata*                             Aurelien Jacobs
  mgsts.c                               Paul B Mahol
  microdvd*                             Aurelien Jacobs
  mm.c                                  Peter Ross
  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
+  mpegts*                               Baptiste Coudurier
  msnwc_tcp.c                           Ramiro Polla
  mtv.c                                 Reynaldo H. Verdejo Pinochet
  mxf*                                  Baptiste Coudurier
  mxfdec.c                              Tomas Härdin
  nistspheredec.c                       Paul B Mahol
  nsvdec.c                              Francois Revol
-  nut*                                  Michael Niedermayer
+  nut.c                                 Michael Niedermayer
  nuv.c                                 Reimar Doeffinger
  oggdec.c, oggdec.h                    David Conrad
  oggenc.c                              Baptiste Coudurier
  oggparse*.c                           David Conrad
-  oggparsedaala*                        Rostislav Pehlivanov
  oma.c                                 Maxim Poliakovski
  paf.c                                 Paul B Mahol
  psxstr.c                              Mike Melanson
@@ -490,23 +401,15 @@ Muxers/Demuxers:
  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_vp9.c                          Thomas Volkert
  rtpenc_mpv.*, rtpenc_aac.*            Martin Storsjo
  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
  swf.c                                 Baptiste Coudurier
  takdec.c                              Paul B Mahol
@@ -515,7 +418,6 @@ Muxers/Demuxers:
  voc.c                                 Aurelien Jacobs
  wav.c                                 Michael Niedermayer
  wc3movie.c                            Mike Melanson
-  webm dash (matroskaenc.c)             Vignesh Venkatasubramanian
  webvtt*                               Matthew J Heaney
  westwood.c                            Mike Melanson
  wtv.c                                 Peter Ross
@@ -523,14 +425,11 @@ Muxers/Demuxers:
  wvenc.c                               Paul B Mahol

 Protocols:
-  async.c                               Zhang Rui
  bluray.c                              Petri Hintukainen
  ftp.c                                 Lukasz Marek
  http.c                                Ronald S. Bultje
-  libssh.c                              Lukasz Marek
  mms*.c                                Ronald S. Bultje
  udp.c                                 Luca Abeni
-  icecast.c                             Marvin Scholz


 libswresample
@@ -553,14 +452,12 @@ Operating systems / CPU architectures
 Alpha                                   Mans Rullgard, Falk Hueffner
 ARM                                     Mans Rullgard
 AVR32                                   Mans Rullgard
-MIPS                                    Mans Rullgard, Nedeljko Babic
+MIPS                                    Mans Rullgard
 Mac OS X / PowerPC                      Romain Dolbeau, Guillaume Poirier
 Amiga / PowerPC                         Colin Ward
 Linux / PowerPC                         Luca Barbato
 Windows MinGW                           Alex Beregszaszi, Ramiro Polla
 Windows Cygwin                          Victor Paesa
-Windows MSVC                            Matthew Oliver, Hendrik Leppkes
-Windows ICL                             Matthew Oliver
 ADI/Blackfin DSP                        Marc Hoffman
 Sparc                                   Roman Shaposhnik
 x86                                     Michael Niedermayer
@@ -569,11 +466,8 @@ x86                                     Michael Niedermayer
 Releases
 ========

-2.8                                     Michael Niedermayer
-2.7                                     Michael Niedermayer
-2.6                                     Michael Niedermayer
-2.5                                     Michael Niedermayer
-2.4                                     Michael Niedermayer
+2.0                                     Michael Niedermayer
+1.2                                     Michael Niedermayer

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

@@ -581,7 +475,6 @@ If you want to maintain an older release, please contact us
 GnuPG Fingerprints of maintainers and contributors
 ==================================================

-Alexander Strasser            1C96 78B7 83CB 8AA7 9AF5 D1EB A7D8 A57B A876 E58F
 Anssi Hannula                 1A92 FF42 2DD9 8D2E 8AF7 65A9 4278 C520 513D F3CB
 Anton Khirnov                 6D0C 6625 56F8 65D1 E5F5 814B B50A 1241 C067 07AB
 Ash Hughes                    694D 43D2 D180 C7C7 6421 ABD3 A641 D0B7 623D 6029
@@ -589,11 +482,10 @@ Attila Kinali                 11F0 F9A6 A1D2 11F6 C745 D10C 6520 BCDD F2DF E765
 Baptiste Coudurier            8D77 134D 20CC 9220 201F C5DB 0AC9 325C 5C1A BAAA
 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
+Bœsch Clément                 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
 Jaikrishnan Menon             61A1 F09F 01C9 2D45 78E1 C862 25DC 8831 AF70 D368
 Jean Delvare                  7CA6 9F44 60F1 BDC4 1FD2 C858 A552 6B9B B3CD 4E6A
@@ -605,15 +497,12 @@ Michael Niedermayer           9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
 Nicolas George                24CE 01CE 9ACC 5CEB 74D8 8D9D B063 D997 36E5 4C93
 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
-Reimar Doeffinger             C61D 16E5 9E2C D10C 8958 38A4 0899 A2B9 06D4 D9C7
+Reimar Döffinger              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
 Stephan Hilb                  4F38 0B3A 5F39 B99B F505 E562 8D5C 5554 4E17 8863
-Tiancheng "Timothy" Gu        9456 AFC0 814A 8139 E994 8351 7FE6 B095 B582 B0D4
-Tim Nicholson                 38CF DB09 3ED0 F607 8B67 6CED 0C0B FC44 8B0B FC83
 Tomas Härdin                  A79D 4E3D F38F 763F 91F5 8B33 A01E 8AE0 41BB 2551
 Wei Gao                       4269 7741 857A 0E60 9EC5 08D2 4744 4EFA 62C1 87B9
--- a/112
+++ b/112
@@ -4,55 +4,39 @@ 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 %/fate_config.sh.template $(SRC_PATH)

-AVPROGS-$(CONFIG_FFMPEG)   += ffmpeg
-AVPROGS-$(CONFIG_FFPLAY)   += ffplay
-AVPROGS-$(CONFIG_FFPROBE)  += ffprobe
-AVPROGS-$(CONFIG_FFSERVER) += ffserver
+PROGS-$(CONFIG_FFMPEG)   += ffmpeg
+PROGS-$(CONFIG_FFPLAY)   += ffplay
+PROGS-$(CONFIG_FFPROBE)  += ffprobe
+PROGS-$(CONFIG_FFSERVER) += ffserver

-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
-ifndef CONFIG_VIDEOTOOLBOX
-OBJS-ffmpeg-$(CONFIG_VDA)     += ffmpeg_videotoolbox.o
-endif
-OBJS-ffmpeg-$(CONFIG_VIDEOTOOLBOX) += ffmpeg_videotoolbox.o
-OBJS-ffmpeg-$(CONFIG_LIBMFX)  += ffmpeg_qsv.o
-OBJS-ffserver                 += ffserver_config.o
+PROGS      := $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))
+INSTPROGS   = $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))

+OBJS        = cmdutils.o $(EXEOBJS)
+OBJS-ffmpeg = ffmpeg_opt.o ffmpeg_filter.o
 TESTTOOLS   = audiogen videogen rotozoom tiny_psnr tiny_ssim base64
 HOSTPROGS  := $(TESTTOOLS:%=tests/%) doc/print_options
-TOOLS       = qt-faststart trasher uncoded_frame
+TOOLS       = qt-faststart trasher
 TOOLS-$(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
+BASENAMES   = ffmpeg ffplay ffprobe ffserver
+ALLPROGS    = $(BASENAMES:%=%$(PROGSSUF)$(EXESUF))
+ALLPROGS_G  = $(BASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
+
+FFLIBS-$(CONFIG_AVDEVICE) += avdevice
+FFLIBS-$(CONFIG_AVFILTER) += avfilter
+FFLIBS-$(CONFIG_AVFORMAT) += avformat
 FFLIBS-$(CONFIG_AVRESAMPLE) += avresample
-FFLIBS-$(CONFIG_POSTPROC)   += postproc
-FFLIBS-$(CONFIG_SWRESAMPLE) += swresample
-FFLIBS-$(CONFIG_SWSCALE)    += swscale
+FFLIBS-$(CONFIG_AVCODEC)  += avcodec
+FFLIBS-$(CONFIG_POSTPROC) += postproc
+FFLIBS-$(CONFIG_SWRESAMPLE)+= swresample
+FFLIBS-$(CONFIG_SWSCALE)  += swscale

 FFLIBS := avutil

@@ -65,16 +49,17 @@ include $(SRC_PATH)/common.mak

 FF_EXTRALIBS := $(FFEXTRALIBS)
 FF_DEP_LIBS  := $(DEP_LIBS)
-FF_STATIC_DEP_LIBS := $(STATIC_DEP_LIBS)

-all: $(AVPROGS)
+all: $(PROGS)
+
+$(PROGS): %$(EXESUF): %_g$(EXESUF)
+	$(CP) $< $@
+	$(STRIP) $@

 $(TOOLS): %$(EXESUF): %.o $(EXEOBJS)
-	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS)
+	$(LD) $(LDFLAGS) $(LD_O) $^ $(ELIBS)

 tools/cws2fws$(EXESUF): ELIBS = $(ZLIB)
-tools/uncoded_frame$(EXESUF): $(FF_DEP_LIBS)
-tools/uncoded_frame$(EXESUF): ELIBS = $(FF_EXTRALIBS)

 config.h: .config
 .config: $(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c))
@@ -84,10 +69,11 @@ config.h: .config

 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 MMX-OBJS YASM-OBJS                           \
-               MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSP-OBJS MSA-OBJS         \
-               MMI-OBJS OBJS SLIBOBJS HOSTOBJS TESTOBJS
+               ARMV5TE-OBJS ARMV6-OBJS VFP-OBJS NEON-OBJS                \
+               ALTIVEC-OBJS VIS-OBJS                                     \
+               MMX-OBJS YASM-OBJS                                        \
+               MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSPR1-OBJS MIPS32R2-OBJS  \
+               OBJS HOSTOBJS TESTOBJS

 define RESET
 $(1) :=
@@ -99,16 +85,13 @@ $(foreach V,$(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
 endef

 $(foreach D,$(FFLIBS),$(eval $(call DOSUBDIR,lib$(D))))

-include $(SRC_PATH)/doc/Makefile
-
 define DOPROG
-OBJS-$(1) += $(1).o $(EXEOBJS) $(OBJS-$(1)-yes)
+OBJS-$(1) += $(1).o cmdutils.o $(EXEOBJS)
 $(1)$(PROGSSUF)_g$(EXESUF): $$(OBJS-$(1))
 $$(OBJS-$(1)): CFLAGS  += $(CFLAGS-$(1))
 $(1)$(PROGSSUF)_g$(EXESUF): LDFLAGS += $(LDFLAGS-$(1))
@@ -116,16 +99,10 @@ $(1)$(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 libavcodec/utils.o libavformat/utils.o libavdevice/avdevice.o libavfilter/avfilter.o libavutil/utils.o libpostproc/postprocess.o libswresample/swresample.o libswscale/utils.o : libavutil/ffversion.h
-
-$(PROGS): %$(PROGSSUF)$(EXESUF): %$(PROGSSUF)_g$(EXESUF)
-	$(CP) $< $@
-	$(STRIP) $@
+$(foreach P,$(PROGS-yes),$(eval $(call DOPROG,$(P))))

 %$(PROGSSUF)_g$(EXESUF): %.o $(FF_DEP_LIBS)
-	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
+	$(LD) $(LDFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)

 OBJDIRS += tools

@@ -137,14 +114,14 @@ GIT_LOG     = $(SRC_PATH)/.git/logs/HEAD
 .version: $(wildcard $(GIT_LOG)) $(VERSION_SH) config.mak
 .version: M=@

-libavutil/ffversion.h .version:
-	$(M)$(VERSION_SH) $(SRC_PATH) libavutil/ffversion.h $(EXTRA_VERSION)
+version.h .version:
+	$(M)$(VERSION_SH) $(SRC_PATH) version.h $(EXTRA_VERSION)
 	$(Q)touch .version

 # force version.sh to run whenever version might have changed
 -include .version

-ifdef AVPROGS
+ifdef PROGS
 install: install-progs install-data
 endif

@@ -155,7 +132,7 @@ install-libs: install-libs-yes
 install-progs-yes:
 install-progs-$(CONFIG_SHARED): install-libs

-install-progs: install-progs-yes $(AVPROGS)
+install-progs: install-progs-yes $(PROGS)
 	$(Q)mkdir -p "$(BINDIR)"
 	$(INSTALL) -c -m 755 $(INSTPROGS) "$(BINDIR)"

@@ -167,31 +144,28 @@ install-data: $(DATA_FILES) $(EXAMPLES_FILES)
 uninstall: uninstall-libs uninstall-headers uninstall-progs uninstall-data

 uninstall-progs:
-	$(RM) $(addprefix "$(BINDIR)/", $(ALLAVPROGS))
+	$(RM) $(addprefix "$(BINDIR)/", $(ALLPROGS))

 uninstall-data:
 	$(RM) -r "$(DATADIR)"

 clean::
-	$(RM) $(ALLAVPROGS) $(ALLAVPROGS_G)
+	$(RM) $(ALLPROGS) $(ALLPROGS_G)
 	$(RM) $(CLEANSUFFIXES)
 	$(RM) $(CLEANSUFFIXES:%=tools/%)
 	$(RM) -r coverage-html
-	$(RM) -rf coverage.info coverage.info.in lcov
+	$(RM) -rf coverage.info lcov

 distclean::
 	$(RM) $(DISTCLEANSUFFIXES)
-	$(RM) config.* .config libavutil/avconfig.h .version avversion.h version.h libavutil/ffversion.h libavcodec/codec_names.h
-ifeq ($(SRC_LINK),src)
-	$(RM) src
-endif
-	$(RM) -rf doc/examples/pc-uninstalled
+	$(RM) config.* .config libavutil/avconfig.h .version version.h libavcodec/codec_names.h

 config:
 	$(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION)

 check: all alltools examples testprogs fate

+include $(SRC_PATH)/doc/Makefile
 include $(SRC_PATH)/tests/Makefile

 $(sort $(OBJDIRS)):
--- a/18
+++ b/18
@@ -0,0 +1,18 @@
+FFmpeg README
+-------------
+
+1) Documentation
+----------------
+
+* Read the documentation in the doc/ directory in git.
+  You can also view it online at http://ffmpeg.org/documentation.html
+
+2) Licensing
+------------
+
+* See the LICENSE file.
+
+3) Build and Install
+--------------------
+
+* See the INSTALL file.
--- a/README.md
+++ b/README.md
@@ -1,49 +0,0 @@
-FFmpeg README
-=============
-
-FFmpeg is a collection of libraries and tools to process multimedia content
-such as audio, video, subtitles and related metadata.
-
-## Libraries
-
-* `libavcodec` provides implementation of a wider range of codecs.
-* `libavformat` implements streaming protocols, container formats and basic I/O access.
-* `libavutil` includes hashers, decompressors and miscellaneous utility functions.
-* `libavfilter` provides a mean to alter decoded Audio and Video through chain of filters.
-* `libavdevice` provides an abstraction to access capture and playback devices.
-* `libswresample` implements audio mixing and resampling routines.
-* `libswscale` implements color conversion and scaling routines.
-
-## Tools
-
-* [ffmpeg](https://ffmpeg.org/ffmpeg.html) is a command line toolbox to
-  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.
-* [ffserver](https://ffmpeg.org/ffserver.html) is a multimedia streaming server
-  for live broadcasts.
-* Additional small tools such as `aviocat`, `ismindex` and `qt-faststart`.
-
-## Documentation
-
-The offline documentation is available in the **doc/** directory.
-
-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. Few developers
-follow pull requests so they will likely be ignored.
--- a/2
+++ b/2
@@ -1 +1 @@
-2.8.git
+2.0.4
--- a/1
+++ b/1
@@ -0,0 +1 @@
+2.0.4
--- a/arch.mak
+++ b/arch.mak
@@ -1,17 +1,16 @@
 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_MIPSDSP)   += $(MIPSDSP-OBJS)    $(MIPSDSP-OBJS-yes)
+OBJS-$(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_MSA)       += $(MSA-OBJS)        $(MSA-OBJS-yes)
-OBJS-$(HAVE_MMI)   += $(MMI-OBJS)   $(MMI-OBJS-yes)

 OBJS-$(HAVE_ALTIVEC) += $(ALTIVEC-OBJS) $(ALTIVEC-OBJS-yes)
-OBJS-$(HAVE_VSX)     += $(VSX-OBJS) $(VSX-OBJS-yes)
+
+OBJS-$(HAVE_VIS)     += $(VIS-OBJS)     $(VIS-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
@@ -19,18 +19,17 @@
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

-#ifndef CMDUTILS_H
-#define CMDUTILS_H
+#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
+#ifdef __MINGW32__
 #undef main /* We don't want SDL to override our main() */
 #endif

@@ -44,12 +43,16 @@ extern const char program_name[];
 */
 extern const int program_birth_year;

+/**
+ * this year, defined by the program for show_banner()
+ */
+extern const int this_year;
+
 extern AVCodecContext *avcodec_opts[AVMEDIA_TYPE_NB];
 extern AVFormatContext *avformat_opts;
-extern AVDictionary *sws_dict;
+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.
@@ -59,7 +62,7 @@ void register_exit(void (*cb)(int ret));
 /**
 * Wraps exit with a program-specific cleanup routine.
 */
-void exit_program(int ret) av_noreturn;
+void exit_program(int ret);

 /**
 * Initialize the cmdutils option system, in particular
@@ -78,11 +81,6 @@ void uninit_opts(void);
 */
 void log_callback_help(void* ptr, int level, const char* fmt, va_list vl);

-/**
- * Override the cpuflags.
- */
-int opt_cpuflags(void *optctx, const char *opt, const char *arg);
-
 /**
 * Fallback for options that are not explicitly handled, these will be
 * parsed through AVOptions.
@@ -98,14 +96,12 @@ int opt_report(const char *opt);

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

+int opt_cpuflags(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.
 */
@@ -277,7 +273,7 @@ typedef struct OptionGroup {
    AVDictionary *codec_opts;
    AVDictionary *format_opts;
    AVDictionary *resample_opts;
-    AVDictionary *sws_dict;
+    struct SwsContext *sws_opts;
    AVDictionary *swr_opts;
 } OptionGroup;

@@ -415,13 +411,6 @@ void show_banner(int argc, char **argv, const OptionDef *options);
 */
 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.
@@ -431,31 +420,10 @@ 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);
-
-#if CONFIG_AVDEVICE
-/**
- * Print a listing containing audodetected sinks of the output device.
- * Device name with options may be passed as an argument to limit results.
- */
-int show_sinks(void *optctx, const char *opt, const char *arg);
-
-/**
- * Print a listing containing audodetected sources of the input device.
- * Device name with options may be passed as an argument to limit results.
- */
-int show_sources(void *optctx, const char *opt, const char *arg);
-#endif
+int show_formats(void *optctx, const char *opt, const char *arg);

 /**
 * Print a listing containing all the codecs supported by the
@@ -517,18 +485,24 @@ int show_layouts(void *optctx, const char *opt, const char *arg);
 */
 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.
 *
@@ -585,6 +559,4 @@ void *grow_array(void *array, int elem_size, int *size, int new_size);
    char name[128];\
    av_get_channel_layout_string(name, sizeof(name), 0, ch_layout);

-double get_rotation(AVStream *st);
-
 #endif /* CMDUTILS_H */
--- a/cmdutils_common_opts.h
+++ b/cmdutils_common_opts.h
@@ -4,9 +4,7 @@
    { "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" },
@@ -16,20 +14,11 @@
    { "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" },
+    { "cpuflags"   , HAS_ARG | OPT_EXPERT, {.func_arg = opt_cpuflags}, "force specific cpu flags", "flags" },
 #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
-#if CONFIG_AVDEVICE
-    { "sources"    , OPT_EXIT | HAS_ARG, { .func_arg = show_sources },
-      "list sources of the input device", "device" },
-    { "sinks"      , OPT_EXIT | HAS_ARG, { .func_arg = show_sinks },
-      "list sinks of the output device", "device" },
-#endif
--- a/cmdutils_opencl.c
+++ b/cmdutils_opencl.c
@@ -1,278 +0,0 @@
-/*
- * 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 "libavutil/avstring.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)
-{
-    const OpenCLDeviceBenchmark* va = (const OpenCLDeviceBenchmark*)a;
-    const OpenCLDeviceBenchmark* vb = (const OpenCLDeviceBenchmark*)b;
-    return FFDIFFSIGN(va->runtime , vb->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;
-                av_strlcpy(devices[count].device_name, device_node->device_name,
-                           sizeof(devices[count].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
@@ -5,20 +5,12 @@
 # first so "all" becomes default target
 all: all-yes

-DEFAULT_YASMD=.dbg
-
-ifeq ($(DBG),1)
-YASMD=$(DEFAULT_YASMD)
-else
-YASMD=
-endif
-
 ifndef SUBDIR

 ifndef V
 Q      = @
 ECHO   = printf "$(1)\t%s\n" $(2)
-BRIEF  = CC CXX OBJCC HOSTCC HOSTLD AS YASM AR LD STRIP CP WINDRES
+BRIEF  = CC CXX HOSTCC HOSTLD AS YASM AR LD STRIP CP
 SILENT = DEPCC DEPHOSTCC DEPAS DEPYASM RANLIB RM

 MSG    = $@
@@ -32,12 +24,10 @@ endif
 ALLFFLIBS = avcodec avdevice avfilter avformat avresample avutil postproc swscale swresample

 # NASM requires -I path terminated with /
-IFLAGS     := -I. -I$(SRC_LINK)/
+IFLAGS     := -I. -I$(SRC_PATH)/
 CPPFLAGS   := $(IFLAGS) $(CPPFLAGS)
 CFLAGS     += $(ECFLAGS)
 CCFLAGS     = $(CPPFLAGS) $(CFLAGS)
-OBJCFLAGS  += $(EOBJCFLAGS)
-OBJCCFLAGS  = $(CPPFLAGS) $(CFLAGS) $(OBJCFLAGS)
 ASFLAGS    := $(CPPFLAGS) $(ASFLAGS)
 CXXFLAGS   += $(CPPFLAGS) $(CFLAGS)
 YASMFLAGS  += $(IFLAGS:%=%/) -Pconfig.asm
@@ -47,14 +37,12 @@ LDFLAGS    := $(ALLFFLIBS:%=$(LD_PATH)lib%) $(LDFLAGS)

 define COMPILE
       $(call $(1)DEP,$(1))
-       $($(1)) $($(1)FLAGS) $($(1)_DEPFLAGS) $($(1)_C) $($(1)_O) $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<)
+       $($(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_M = $(call COMPILE,OBJCC)
-COMPILE_HOSTC = $(call COMPILE,HOSTCC)

 %.o: %.c
 	$(COMPILE_C)
@@ -62,21 +50,12 @@ COMPILE_HOSTC = $(call COMPILE,HOSTCC)
 %.o: %.cpp
 	$(COMPILE_CXX)

-%.o: %.m
-	$(COMPILE_M)
-
 %.s: %.c
-	$(CC) $(CCFLAGS) -S -o $@ $<
+	$(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) $<

@@ -84,9 +63,7 @@ COMPILE_HOSTC = $(call COMPILE,HOSTCC)
 	$(Q)echo '#include "$*.h"' >$@

 %.ver: %.v
-	$(Q)sed 's/$$MAJOR/$($(basename $(@F))_VERSION_MAJOR)/' $^ | sed -e 's/:/:\
-/' -e 's/; /;\
-/g' > $@
+	$(Q)sed 's/$$MAJOR/$($(basename $(@F))_VERSION_MAJOR)/' $^ > $@

 %.c %.h: TAG = GEN

@@ -105,15 +82,14 @@ endif
 include $(SRC_PATH)/arch.mak

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

 LDLIBS       = $(FFLIBS:%=%$(BUILDSUF))
 FFEXTRALIBS := $(LDLIBS:%=$(LD_LIB)) $(EXTRALIBS)

+EXAMPLES  := $(EXAMPLES:%=$(SUBDIR)%-example$(EXESUF))
 OBJS      := $(sort $(OBJS:%=$(SUBDIR)%))
-SLIBOBJS  := $(sort $(SLIBOBJS:%=$(SUBDIR)%))
 TESTOBJS  := $(TESTOBJS:%=$(SUBDIR)%) $(TESTPROGS:%=$(SUBDIR)%-test.o)
 TESTPROGS := $(TESTPROGS:%=$(SUBDIR)%-test$(EXESUF))
 HOSTOBJS  := $(HOSTPROGS:%=$(SUBDIR)%.o)
@@ -123,9 +99,7 @@ TOOLOBJS  := $(TOOLS:%=tools/%.o)
 TOOLS     := $(TOOLS:%=tools/%$(EXESUF))
 HEADERS   += $(HEADERS-yes)

-PATH_LIBNAME = $(foreach NAME,$(1),lib$(NAME)/$($(2)LIBNAME))
-DEP_LIBS := $(foreach lib,$(FFLIBS),$(call PATH_LIBNAME,$(lib),$(CONFIG_SHARED:yes=S)))
-STATIC_DEP_LIBS := $(foreach lib,$(FFLIBS),$(call PATH_LIBNAME,$(lib)))
+DEP_LIBS := $(foreach NAME,$(FFLIBS),lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME))

 SRC_DIR    := $(SRC_PATH)/lib$(NAME)
 ALLHEADERS := $(subst $(SRC_DIR)/,$(SUBDIR),$(wildcard $(SRC_DIR)/*.h $(SRC_DIR)/$(ARCH)/*.h))
@@ -138,31 +112,30 @@ checkheaders: $(HOBJS)
 alltools: $(TOOLS)

 $(HOSTOBJS): %.o: %.c
-	$(COMPILE_HOSTC)
+	$(call COMPILE,HOSTCC)

 $(HOSTPROGS): %$(HOSTEXESUF): %.o
-	$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $^ $(HOSTLIBS)
+	$(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))
+OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(TESTOBJS))

-CLEANSUFFIXES     = *.d *.o *~ *.h.c *.map *.ver *.ver-sol2 *.ho *.gcno *.gcda *$(DEFAULT_YASMD).asm
+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) $(OBJS:.o=$(DEFAULT_YASMD).d)
+	$(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)) $(OBJS:.o=$(DEFAULT_YASMD).d)
+-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d))
--- a/compat/aix/math.h
+++ b/compat/aix/math.h
@@ -1,22 +1,5 @@
 /*
- * Work around the class() function in AIX math.h clashing with
- * identifiers named "class".
- *
- * 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
+ * Workaround aix-specific class() function clashing with ffmpeg class usage
 */

 #ifndef COMPAT_AIX_MATH_H
--- a/compat/avisynth/avisynth_c.h
+++ b/compat/avisynth/avisynth_c.h
@@ -13,8 +13,7 @@
 //
 // 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
+// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
 // http://www.gnu.org/copyleft/gpl.html .
 //
 // As a special exception, I give you permission to link to the
@@ -38,9 +37,40 @@
 #ifndef __AVISYNTH_C__
 #define __AVISYNTH_C__

-#include "avs/config.h"
-#include "avs/capi.h"
-#include "avs/types.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
+
+typedef unsigned char BYTE;
+#ifdef __GNUC__
+typedef long long int INT64;
+#else
+typedef __int64 INT64;
+#endif


 /////////////////////////////////////////////////////////////////////
@@ -48,8 +78,8 @@
 // Constants
 //

-#ifndef __AVISYNTH_6_H__
-enum { AVISYNTH_INTERFACE_VERSION = 6 };
+#ifndef __AVISYNTH_H__
+enum { AVISYNTH_INTERFACE_VERSION = 4 };
 #endif

 enum {AVS_SAMPLE_INT8  = 1<<0,
@@ -81,8 +111,8 @@ enum {AVS_CS_BGR = 1<<28,
      AVS_CS_PLANAR = 1<<31,

      AVS_CS_SHIFT_SUB_WIDTH   = 0,
-      AVS_CS_SHIFT_SUB_HEIGHT  = 8,
-      AVS_CS_SHIFT_SAMPLE_BITS = 16,
+      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
@@ -149,66 +179,15 @@ enum {  //SUBTYPES
  AVS_FILTER_OUTPUT_TYPE_DIFFERENT=4};

 enum {
-  // New 2.6 explicitly defined cache hints.
-  AVS_CACHE_NOTHING=10, // Do not cache video.
-  AVS_CACHE_WINDOW=11, // Hard protect upto X frames within a range of X from the current frame N.
-  AVS_CACHE_GENERIC=12, // LRU cache upto X frames.
-  AVS_CACHE_FORCE_GENERIC=13, // LRU cache upto X frames, override any previous CACHE_WINDOW.
-
-  AVS_CACHE_GET_POLICY=30, // Get the current policy.
-  AVS_CACHE_GET_WINDOW=31, // Get the current window h_span.
-  AVS_CACHE_GET_RANGE=32, // Get the current generic frame range.
-
-  AVS_CACHE_AUDIO=50, // Explicitly do cache audio, X byte cache.
-  AVS_CACHE_AUDIO_NOTHING=51, // Explicitly do not cache audio.
-  AVS_CACHE_AUDIO_NONE=52, // Audio cache off (auto mode), X byte intial cache.
-  AVS_CACHE_AUDIO_AUTO=53, // Audio cache on (auto mode), X byte intial cache.
-
-  AVS_CACHE_GET_AUDIO_POLICY=70, // Get the current audio policy.
-  AVS_CACHE_GET_AUDIO_SIZE=71, // Get the current audio cache size.
-
-  AVS_CACHE_PREFETCH_FRAME=100, // Queue request to prefetch frame N.
-  AVS_CACHE_PREFETCH_GO=101, // Action video prefetches.
-
-  AVS_CACHE_PREFETCH_AUDIO_BEGIN=120, // Begin queue request transaction to prefetch audio (take critical section).
-  AVS_CACHE_PREFETCH_AUDIO_STARTLO=121, // Set low 32 bits of start.
-  AVS_CACHE_PREFETCH_AUDIO_STARTHI=122, // Set high 32 bits of start.
-  AVS_CACHE_PREFETCH_AUDIO_COUNT=123, // Set low 32 bits of length.
-  AVS_CACHE_PREFETCH_AUDIO_COMMIT=124, // Enqueue request transaction to prefetch audio (release critical section).
-  AVS_CACHE_PREFETCH_AUDIO_GO=125, // Action audio prefetches.
-
-  AVS_CACHE_GETCHILD_CACHE_MODE=200, // Cache ask Child for desired video cache mode.
-  AVS_CACHE_GETCHILD_CACHE_SIZE=201, // Cache ask Child for desired video cache size.
-  AVS_CACHE_GETCHILD_AUDIO_MODE=202, // Cache ask Child for desired audio cache mode.
-  AVS_CACHE_GETCHILD_AUDIO_SIZE=203, // Cache ask Child for desired audio cache size.
-
-  AVS_CACHE_GETCHILD_COST=220, // Cache ask Child for estimated processing cost.
-    AVS_CACHE_COST_ZERO=221, // Child response of zero cost (ptr arithmetic only).
-    AVS_CACHE_COST_UNIT=222, // Child response of unit cost (less than or equal 1 full frame blit).
-    AVS_CACHE_COST_LOW=223, // Child response of light cost. (Fast)
-    AVS_CACHE_COST_MED=224, // Child response of medium cost. (Real time)
-    AVS_CACHE_COST_HI=225, // Child response of heavy cost. (Slow)
-
-  AVS_CACHE_GETCHILD_THREAD_MODE=240, // Cache ask Child for thread safetyness.
-    AVS_CACHE_THREAD_UNSAFE=241, // Only 1 thread allowed for all instances. 2.5 filters default!
-    AVS_CACHE_THREAD_CLASS=242, // Only 1 thread allowed for each instance. 2.6 filters default!
-    AVS_CACHE_THREAD_SAFE=243, //  Allow all threads in any instance.
-    AVS_CACHE_THREAD_OWN=244, // Safe but limit to 1 thread, internally threaded.
-
-  AVS_CACHE_GETCHILD_ACCESS_COST=260, // Cache ask Child for preferred access pattern.
-    AVS_CACHE_ACCESS_RAND=261, // Filter is access order agnostic.
-    AVS_CACHE_ACCESS_SEQ0=262, // Filter prefers sequential access (low cost)
-    AVS_CACHE_ACCESS_SEQ1=263, // Filter needs sequential access (high cost)
+  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
  };

-#ifdef BUILDING_AVSCORE
-struct AVS_ScriptEnvironment {
-        IScriptEnvironment * env;
-        const char * error;
-        AVS_ScriptEnvironment(IScriptEnvironment * e = 0)
-                : env(e), error(0) {}
-};
-#endif
+#define AVS_FRAME_ALIGN 16

 typedef struct AVS_Clip AVS_Clip;
 typedef struct AVS_ScriptEnvironment AVS_ScriptEnvironment;
@@ -258,23 +237,29 @@ AVSC_INLINE int avs_is_yuv(const AVS_VideoInfo * p)
 AVSC_INLINE int avs_is_yuy2(const AVS_VideoInfo * p)
        { return (p->pixel_type & AVS_CS_YUY2) == AVS_CS_YUY2; }

-AVSC_API(int, avs_is_yv24)(const AVS_VideoInfo * p);
+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_API(int, avs_is_yv16)(const AVS_VideoInfo * p);
+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_API(int, avs_is_yv12)(const AVS_VideoInfo * p) ;
+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_API(int, avs_is_yv411)(const AVS_VideoInfo * p);
+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_API(int, avs_is_y8)(const AVS_VideoInfo * p);
+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->image_type & property)==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_API(int, avs_is_color_space)(const AVS_VideoInfo * p, int c_space);
+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); }
@@ -288,18 +273,25 @@ AVSC_INLINE int avs_is_bff(const AVS_VideoInfo * p)
 AVSC_INLINE int avs_is_tff(const AVS_VideoInfo * p)
        { return !!(p->image_type & AVS_IT_TFF); }

-AVSC_API(int, avs_get_plane_width_subsampling)(const AVS_VideoInfo * p, int plane);
+AVSC_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_API(int, avs_get_plane_height_subsampling)(const AVS_VideoInfo * p, int plane);
+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_API(int, avs_bits_per_pixel)(const AVS_VideoInfo * p);
-
-AVSC_API(int, avs_bytes_from_pixels)(const AVS_VideoInfo * p, int pixels);
-
-AVSC_API(int, avs_row_size)(const AVS_VideoInfo * p, int plane);
-
-AVSC_API(int, avs_bmp_size)(const AVS_VideoInfo * vi);
+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; }
@@ -357,13 +349,11 @@ AVSC_INLINE void avs_set_fps(AVS_VideoInfo * p, unsigned numerator, unsigned den
    p->fps_denominator = denominator/x;
 }

-#ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
 AVSC_INLINE int avs_is_same_colorspace(AVS_VideoInfo * x, AVS_VideoInfo * y)
 {
        return (x->pixel_type == y->pixel_type)
                || (avs_is_yv12(x) && avs_is_yv12(y));
 }
-#endif

 /////////////////////////////////////////////////////////////////////
 //
@@ -400,38 +390,89 @@ typedef struct AVS_VideoFrame {
 } AVS_VideoFrame;

 // Access functions for AVS_VideoFrame
-AVSC_API(int, avs_get_pitch_p)(const AVS_VideoFrame * p, int plane);
-
-#ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
 AVSC_INLINE int avs_get_pitch(const AVS_VideoFrame * p) {
-        return avs_get_pitch_p(p, 0);}
-#endif
+        return p->pitch;}

-AVSC_API(int, avs_get_row_size_p)(const AVS_VideoFrame * p, int plane);
+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_API(int, avs_get_height_p)(const AVS_VideoFrame * p, int plane);
+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_API(const BYTE *, avs_get_read_ptr_p)(const AVS_VideoFrame * p, int plane);
+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;}

-#ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
 AVSC_INLINE const BYTE* avs_get_read_ptr(const AVS_VideoFrame * p) {
-        return avs_get_read_ptr_p(p, 0);}
-#endif
+        return p->vfb->data + p->offset;}

-AVSC_API(int, avs_is_writable)(const AVS_VideoFrame * p);
+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_API(BYTE *, avs_get_write_ptr_p)(const AVS_VideoFrame * p, int plane);
+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;
+                }
+        }
+}

-#ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
-AVSC_INLINE BYTE* avs_get_write_ptr(const AVS_VideoFrame * p) {
-        return avs_get_write_ptr_p(p, 0);}
-#endif

 AVSC_API(void, avs_release_video_frame)(AVS_VideoFrame *);
 // makes a shallow copy of a video frame
@@ -616,16 +657,12 @@ enum {
  AVS_CPUF_SSSE3      = 0x200,   //  Core 2
  AVS_CPUF_SSE4       = 0x400,   //  Penryn, Wolfdale, Yorkfield
  AVS_CPUF_SSE4_1     = 0x400,
-//AVS_CPUF_AVX        = 0x800,   //  Sandy Bridge, Bulldozer
-  AVS_CPUF_SSE4_2    = 0x1000,   //  Nehalem
-//AVS_CPUF_AVX2      = 0x2000,   //  Haswell
-//AVS_CPUF_AVX512    = 0x4000,   //  Knights Landing
+  AVS_CPUF_SSE4_2     = 0x800,   //  Nehalem
 };

-
 AVSC_API(const char *, avs_get_error)(AVS_ScriptEnvironment *); // return 0 if no error

-AVSC_API(int, avs_get_cpu_flags)(AVS_ScriptEnvironment *);
+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);
@@ -662,12 +699,12 @@ AVSC_API(AVS_VideoFrame *, avs_new_video_frame_a)(AVS_ScriptEnvironment *,
 AVSC_INLINE
 AVS_VideoFrame * avs_new_video_frame(AVS_ScriptEnvironment * env,
                                     const AVS_VideoInfo * vi)
-  {return avs_new_video_frame_a(env,vi,FRAME_ALIGN);}
+  {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,FRAME_ALIGN);}
+  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
 #endif


@@ -735,6 +772,7 @@ struct AVS_Library {
  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);
@@ -759,27 +797,6 @@ struct AVS_Library {
  AVSC_DECLARE_FUNC(avs_subframe_planar);
  AVSC_DECLARE_FUNC(avs_take_clip);
  AVSC_DECLARE_FUNC(avs_vsprintf);
-
-  AVSC_DECLARE_FUNC(avs_get_error);
-  AVSC_DECLARE_FUNC(avs_is_yv24);
-  AVSC_DECLARE_FUNC(avs_is_yv16);
-  AVSC_DECLARE_FUNC(avs_is_yv12);
-  AVSC_DECLARE_FUNC(avs_is_yv411);
-  AVSC_DECLARE_FUNC(avs_is_y8);
-  AVSC_DECLARE_FUNC(avs_is_color_space);
-
-  AVSC_DECLARE_FUNC(avs_get_plane_width_subsampling);
-  AVSC_DECLARE_FUNC(avs_get_plane_height_subsampling);
-  AVSC_DECLARE_FUNC(avs_bits_per_pixel);
-  AVSC_DECLARE_FUNC(avs_bytes_from_pixels);
-  AVSC_DECLARE_FUNC(avs_row_size);
-  AVSC_DECLARE_FUNC(avs_bmp_size);
-  AVSC_DECLARE_FUNC(avs_get_pitch_p);
-  AVSC_DECLARE_FUNC(avs_get_row_size_p);
-  AVSC_DECLARE_FUNC(avs_get_height_p);
-  AVSC_DECLARE_FUNC(avs_get_read_ptr_p);
-  AVSC_DECLARE_FUNC(avs_is_writable);
-  AVSC_DECLARE_FUNC(avs_get_write_ptr_p);
 };

 #undef AVSC_DECLARE_FUNC
@@ -814,6 +831,7 @@ AVSC_INLINE AVS_Library * avs_load_library() {
  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);
@@ -839,27 +857,6 @@ AVSC_INLINE AVS_Library * avs_load_library() {
  AVSC_LOAD_FUNC(avs_take_clip);
  AVSC_LOAD_FUNC(avs_vsprintf);

-  AVSC_LOAD_FUNC(avs_get_error);
-  AVSC_LOAD_FUNC(avs_is_yv24);
-  AVSC_LOAD_FUNC(avs_is_yv16);
-  AVSC_LOAD_FUNC(avs_is_yv12);
-  AVSC_LOAD_FUNC(avs_is_yv411);
-  AVSC_LOAD_FUNC(avs_is_y8);
-  AVSC_LOAD_FUNC(avs_is_color_space);
-
-  AVSC_LOAD_FUNC(avs_get_plane_width_subsampling);
-  AVSC_LOAD_FUNC(avs_get_plane_height_subsampling);
-  AVSC_LOAD_FUNC(avs_bits_per_pixel);
-  AVSC_LOAD_FUNC(avs_bytes_from_pixels);
-  AVSC_LOAD_FUNC(avs_row_size);
-  AVSC_LOAD_FUNC(avs_bmp_size);
-  AVSC_LOAD_FUNC(avs_get_pitch_p);
-  AVSC_LOAD_FUNC(avs_get_row_size_p);
-  AVSC_LOAD_FUNC(avs_get_height_p);
-  AVSC_LOAD_FUNC(avs_get_read_ptr_p);
-  AVSC_LOAD_FUNC(avs_is_writable);
-  AVSC_LOAD_FUNC(avs_get_write_ptr_p);
-
 #undef __AVSC_STRINGIFY
 #undef AVSC_STRINGIFY
 #undef AVSC_LOAD_FUNC
--- 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/avs/capi.h
+++ b/compat/avisynth/avs/capi.h
@@ -1,62 +0,0 @@
-// Avisynth C Interface Version 0.20
-// Copyright 2003 Kevin Atkinson
-
-// This program is free software; you can redistribute it and/or modify
-// it under the terms of the GNU General Public License as published by
-// the Free Software Foundation; either version 2 of the License, or
-// (at your option) any later version.
-//
-// This program is distributed in the hope that it will be useful,
-// but WITHOUT ANY WARRANTY; without even the implied warranty of
-// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-// GNU General Public License for more details.
-//
-// You should have received a copy of the GNU General Public License
-// along with this program; if not, write to the Free Software
-// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
-// http://www.gnu.org/copyleft/gpl.html .
-//
-// As a special exception, I give you permission to link to the
-// Avisynth C interface with independent modules that communicate with
-// the Avisynth C interface solely through the interfaces defined in
-// avisynth_c.h, regardless of the license terms of these independent
-// modules, and to copy and distribute the resulting combined work
-// under terms of your choice, provided that every copy of the
-// combined work is accompanied by a complete copy of the source code
-// of the Avisynth C interface and Avisynth itself (with the version
-// used to produce the combined work), being distributed under the
-// terms of the GNU General Public License plus this exception.  An
-// independent module is a module which is not derived from or based
-// on Avisynth C Interface, such as 3rd-party filters, import and
-// export plugins, or graphical user interfaces.
-
-#ifndef AVS_CAPI_H
-#define AVS_CAPI_H
-
-#ifdef __cplusplus
-#  define EXTERN_C extern "C"
-#else
-#  define EXTERN_C
-#endif
-
-#ifndef AVSC_USE_STDCALL
-#  define AVSC_CC __cdecl
-#else
-#  define AVSC_CC __stdcall
-#endif
-
-#define AVSC_INLINE static __inline
-
-#ifdef BUILDING_AVSCORE
-#  define AVSC_EXPORT EXTERN_C
-#  define AVSC_API(ret, name) EXTERN_C __declspec(dllexport) ret AVSC_CC name
-#else
-#  define AVSC_EXPORT EXTERN_C __declspec(dllexport)
-#  ifndef AVSC_NO_DECLSPEC
-#    define AVSC_API(ret, name) EXTERN_C __declspec(dllimport) ret AVSC_CC name
-#  else
-#    define AVSC_API(ret, name) typedef ret (AVSC_CC *name##_func)
-#  endif
-#endif
-
-#endif //AVS_CAPI_H
--- a/compat/avisynth/avs/config.h
+++ b/compat/avisynth/avs/config.h
@@ -1,55 +0,0 @@
-// Avisynth C Interface Version 0.20
-// Copyright 2003 Kevin Atkinson
-
-// This program is free software; you can redistribute it and/or modify
-// it under the terms of the GNU General Public License as published by
-// the Free Software Foundation; either version 2 of the License, or
-// (at your option) any later version.
-//
-// This program is distributed in the hope that it will be useful,
-// but WITHOUT ANY WARRANTY; without even the implied warranty of
-// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-// GNU General Public License for more details.
-//
-// You should have received a copy of the GNU General Public License
-// along with this program; if not, write to the Free Software
-// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
-// http://www.gnu.org/copyleft/gpl.html .
-//
-// As a special exception, I give you permission to link to the
-// Avisynth C interface with independent modules that communicate with
-// the Avisynth C interface solely through the interfaces defined in
-// avisynth_c.h, regardless of the license terms of these independent
-// modules, and to copy and distribute the resulting combined work
-// under terms of your choice, provided that every copy of the
-// combined work is accompanied by a complete copy of the source code
-// of the Avisynth C interface and Avisynth itself (with the version
-// used to produce the combined work), being distributed under the
-// terms of the GNU General Public License plus this exception.  An
-// independent module is a module which is not derived from or based
-// on Avisynth C Interface, such as 3rd-party filters, import and
-// export plugins, or graphical user interfaces.
-
-#ifndef AVS_CONFIG_H
-#define AVS_CONFIG_H
-
-// Undefine this to get cdecl calling convention
-#define AVSC_USE_STDCALL 1
-
-// NOTE TO PLUGIN AUTHORS:
-// Because FRAME_ALIGN can be substantially higher than the alignment
-// a plugin actually needs, plugins should not use FRAME_ALIGN to check for
-// alignment. They should always request the exact alignment value they need.
-// This is to make sure that plugins work over the widest range of AviSynth
-// builds possible.
-#define FRAME_ALIGN 32
-
-#if   defined(_M_AMD64) || defined(__x86_64)
-#   define X86_64
-#elif defined(_M_IX86) || defined(__i386__)
-#   define X86_32
-#else
-#   error Unsupported CPU architecture.
-#endif
-
-#endif //AVS_CONFIG_H
--- a/compat/avisynth/avs/types.h
+++ b/compat/avisynth/avs/types.h
@@ -1,51 +0,0 @@
-// Avisynth C Interface Version 0.20
-// Copyright 2003 Kevin Atkinson
-
-// This program is free software; you can redistribute it and/or modify
-// it under the terms of the GNU General Public License as published by
-// the Free Software Foundation; either version 2 of the License, or
-// (at your option) any later version.
-//
-// This program is distributed in the hope that it will be useful,
-// but WITHOUT ANY WARRANTY; without even the implied warranty of
-// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-// GNU General Public License for more details.
-//
-// You should have received a copy of the GNU General Public License
-// along with this program; if not, write to the Free Software
-// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
-// http://www.gnu.org/copyleft/gpl.html .
-//
-// As a special exception, I give you permission to link to the
-// Avisynth C interface with independent modules that communicate with
-// the Avisynth C interface solely through the interfaces defined in
-// avisynth_c.h, regardless of the license terms of these independent
-// modules, and to copy and distribute the resulting combined work
-// under terms of your choice, provided that every copy of the
-// combined work is accompanied by a complete copy of the source code
-// of the Avisynth C interface and Avisynth itself (with the version
-// used to produce the combined work), being distributed under the
-// terms of the GNU General Public License plus this exception.  An
-// independent module is a module which is not derived from or based
-// on Avisynth C Interface, such as 3rd-party filters, import and
-// export plugins, or graphical user interfaces.
-
-#ifndef AVS_TYPES_H
-#define AVS_TYPES_H
-
-// Define all types necessary for interfacing with avisynth.dll
-
-// Raster types used by VirtualDub & Avisynth
-typedef unsigned int    Pixel32;
-typedef unsigned char   BYTE;
-
-// Audio Sample information
-typedef float SFLOAT;
-
-#ifdef __GNUC__
-typedef long long int INT64;
-#else
-typedef __int64 INT64;
-#endif
-
-#endif //AVS_TYPES_H
--- a/compat/avisynth/avxsynth_c.h
+++ b/compat/avisynth/avxsynth_c.h
@@ -13,8 +13,7 @@
 //
 // 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
+// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
 // http://www.gnu.org/copyleft/gpl.html .
 //
 // As a special exception, I give you permission to link to the
@@ -513,21 +512,21 @@ AVSC_INLINE AVS_Value avs_array_elt(AVS_Value v, int index)
 // only use these functions on am AVS_Value that does not already have
 // an active value.  Remember, treat AVS_Value as a fat pointer.
 AVSC_INLINE AVS_Value avs_new_value_bool(int v0)
-        { AVS_Value v = {0}; v.type = 'b'; v.d.boolean = v0 == 0 ? 0 : 1; return v; }
+        { 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 = {0}; v.type = 'i'; v.d.integer = v0; return v; }
+        { 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 = {0}; v.type = 's'; v.d.string = v0; return v; }
+        { 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 = {0}; v.type = 'f'; v.d.floating_pt = v0; return v;}
+        { 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 = {0}; v.type = 'e'; v.d.string = v0; return v; }
+        { 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 = {0}; avs_set_to_clip(&v, v0); return v; }
+        { 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 = {0}; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; }
+        { AVS_Value v; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; }

 /////////////////////////////////////////////////////////////////////
 //
--- a/compat/avisynth/windowsPorts/windows2linux.h
+++ b/compat/avisynth/windowsPorts/windows2linux.h
@@ -52,8 +52,8 @@ namespace avxsynth {
 //
 // Functions
 //
-#define MAKEDWORD(a,b,c,d) (((a) << 24) | ((b) << 16) | ((c) << 8) | (d))
-#define MAKEWORD(a,b) (((a) << 8) | (b))
+#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
--- a/compat/float/float.h
+++ b/compat/float/float.h
@@ -1,35 +0,0 @@
-/*
- * Work around broken floating point limits on some systems.
- *
- * 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_next <float.h>
-
-#ifdef FLT_MAX
-#undef  FLT_MAX
-#define FLT_MAX 3.40282346638528859812e+38F
-
-#undef  FLT_MIN
-#define FLT_MIN 1.17549435082228750797e-38F
-
-#undef  DBL_MAX
-#define DBL_MAX ((double)1.79769313486231570815e+308L)
-
-#undef  DBL_MIN
-#define DBL_MIN ((double)2.22507385850720138309e-308L)
-#endif
--- a/compat/float/limits.h
+++ b/compat/float/limits.h
@@ -1,22 +0,0 @@
-/*
- * Work around broken floating point limits on some systems.
- *
- * 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_next <limits.h>
-#include <float.h>
--- a/compat/getopt.c
+++ b/compat/getopt.c
@@ -38,6 +38,8 @@ static int optind = 1;
 static int optopt;
 static char *optarg;

+#undef fprintf
+
 static int getopt(int argc, char *argv[], char *opts)
 {
    static int sp = 1;
@@ -54,7 +56,7 @@ 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.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
-#define COMPAT_MSVCRT_SNPRINTF_H
+#ifndef COMPAT_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
@@ -23,8 +23,8 @@
 * os2threads to pthreads wrapper
 */

-#ifndef COMPAT_OS2THREADS_H
-#define COMPAT_OS2THREADS_H
+#ifndef AVCODEC_OS2PTHREADS_H
+#define AVCODEC_OS2PTHREADS_H

 #define INCL_DOS
 #include <os2.h>
@@ -32,71 +32,55 @@
 #undef __STRICT_ANSI__          /* for _beginthread() */
 #include <stdlib.h>

-#include <sys/builtin.h>
-#include <sys/fmutex.h>
-
-#include "libavutil/attributes.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 HMTX pthread_mutex_t;
 typedef void pthread_mutexattr_t;

 typedef struct {
-    HEV event_sem;
-    HEV ack_sem;
-    volatile unsigned  wait_count;
+    HEV  event_sem;
+    int  wait_count;
 } pthread_cond_t;

 typedef void pthread_condattr_t;

-typedef struct {
-    volatile int done;
-    _fmutex mtx;
-} pthread_once_t;
-
-#define PTHREAD_ONCE_INIT {0, _FMUTEX_INITIALIZER}
+struct thread_arg {
+    void *(*start_routine)(void *);
+    void *arg;
+};

 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,
-                                           const pthread_attr_t *attr,
-                                           void *(*start_routine)(void*),
-                                           void *arg)
+static av_always_inline int pthread_create(pthread_t *thread, const pthread_attr_t *attr, void *(*start_routine)(void*), void *arg)
 {
-    thread->start_routine = start_routine;
-    thread->arg = arg;
-    thread->result = NULL;
+    struct thread_arg *thread_arg;

-    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);
-
-    if (value_ptr)
-        *value_ptr = thread.result;
+    DosWaitThread((PTID)&thread, DCWW_WAIT);

    return 0;
 }

-static av_always_inline int pthread_mutex_init(pthread_mutex_t *mutex,
-                                               const pthread_mutexattr_t *attr)
+static av_always_inline int pthread_mutex_init(pthread_mutex_t *mutex, const pthread_mutexattr_t *attr)
 {
    DosCreateMutexSem(NULL, (PHMTX)mutex, 0, FALSE);

@@ -124,11 +108,9 @@ static av_always_inline int pthread_mutex_unlock(pthread_mutex_t *mutex)
    return 0;
 }

-static av_always_inline int pthread_cond_init(pthread_cond_t *cond,
-                                              const pthread_condattr_t *attr)
+static av_always_inline int pthread_cond_init(pthread_cond_t *cond, 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;

@@ -138,16 +120,16 @@ 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;
@@ -155,47 +137,26 @@ 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))
-        pthread_cond_signal(cond);
+    while (cond->wait_count > 0) {
+        DosPostEventSem(cond->event_sem);
+
+        cond->wait_count--;
+    }

    return 0;
 }

-static av_always_inline int pthread_cond_wait(pthread_cond_t *cond,
-                                              pthread_mutex_t *mutex)
+static av_always_inline int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
 {
-    __atomic_increment(&cond->wait_count);
+    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,
-                                         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 */
+#endif /* AVCODEC_OS2PTHREADS_H */
--- 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/tms470/math.h
+++ b/compat/tms470/math.h
@@ -1,24 +1,3 @@
-/*
- * 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_TMS470_MATH_H
-#define COMPAT_TMS470_MATH_H
-
 #include_next <math.h>

 #undef INFINITY
@@ -26,5 +5,3 @@

 #define INFINITY (*(const float*)((const unsigned []){ 0x7f800000 }))
 #define NAN      (*(const float*)((const unsigned []){ 0x7fc00000 }))
-
-#endif /* COMPAT_TMS470_MATH_H */
--- a/compat/va_copy.h
+++ b/compat/va_copy.h
@@ -19,16 +19,8 @@
 * 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)
 #define va_copy(dst, src) ((dst) = (src))
 #endif
-#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/w32pthreads.h
+++ b/compat/w32pthreads.h
@@ -26,8 +26,8 @@
 * w32threads to pthreads wrapper
 */

-#ifndef COMPAT_W32PTHREADS_H
-#define COMPAT_W32PTHREADS_H
+#ifndef LIBAV_W32PTHREADS_H
+#define LIBAV_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
@@ -39,12 +39,6 @@
 #include <windows.h>
 #include <process.h>

-#if _WIN32_WINNT < 0x0600 && defined(__MINGW32__)
-#undef MemoryBarrier
-#define MemoryBarrier __sync_synchronize
-#endif
-
-#include "libavutil/attributes.h"
 #include "libavutil/common.h"
 #include "libavutil/internal.h"
 #include "libavutil/mem.h"
@@ -60,56 +54,45 @@ typedef struct pthread_t {
 * not mutexes */
 typedef CRITICAL_SECTION pthread_mutex_t;

-/* This is the CONDITION_VARIABLE typedef for using Windows' native
- * conditional variables on kernels 6.0+. */
-#if HAVE_CONDITION_VARIABLE_PTR
-typedef CONDITION_VARIABLE pthread_cond_t;
-#else
+/* This is the CONDITIONAL_VARIABLE typedef for using Window's native
+ * conditional variables on kernels 6.0+.
+ * MinGW does not currently have this typedef. */
 typedef struct pthread_cond_t {
-    void *Ptr;
+    void *ptr;
 } pthread_cond_t;
-#endif

-#if _WIN32_WINNT >= 0x0600
-#define InitializeCriticalSection(x) InitializeCriticalSectionEx(x, 0, 0)
-#define WaitForSingleObject(a, b) WaitForSingleObjectEx(a, b, FALSE)
-#endif
+/* function pointers to conditional variable API on windows 6.0+ kernels */
+static void (WINAPI *cond_broadcast)(pthread_cond_t *cond);
+static void (WINAPI *cond_init)(pthread_cond_t *cond);
+static void (WINAPI *cond_signal)(pthread_cond_t *cond);
+static BOOL (WINAPI *cond_wait)(pthread_cond_t *cond, pthread_mutex_t *mutex,
+                                DWORD milliseconds);

-static av_unused unsigned __stdcall attribute_align_arg win32thread_worker(void *arg)
+static unsigned __stdcall attribute_align_arg win32thread_worker(void *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,
-                                    void *(*start_routine)(void*), void *arg)
+static int pthread_create(pthread_t *thread, const void *unused_attr,
+                          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_ABANDONED)
-            return EINVAL;
-        else
-            return EDEADLK;
-    }
+    if (ret != WAIT_OBJECT_0)
+        return;
    if (value_ptr)
        *value_ptr = thread.ret;
    CloseHandle(thread.handle);
-    return 0;
 }

 static inline int pthread_mutex_init(pthread_mutex_t *m, void* attr)
@@ -133,115 +116,8 @@ static inline int pthread_mutex_unlock(pthread_mutex_t *m)
    return 0;
 }

-#if _WIN32_WINNT >= 0x0600
-typedef INIT_ONCE pthread_once_t;
-#define PTHREAD_ONCE_INIT INIT_ONCE_STATIC_INIT
-
-static av_unused int pthread_once(pthread_once_t *once_control, void (*init_routine)(void))
-{
-    BOOL pending = FALSE;
-    InitOnceBeginInitialize(once_control, 0, &pending, NULL);
-    if (pending)
-        init_routine();
-    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)
-{
-    SleepConditionVariableCS(cond, mutex, INFINITE);
-    return 0;
-}
-
-static inline int pthread_cond_signal(pthread_cond_t *cond)
-{
-    WakeConditionVariable(cond);
-    return 0;
-}
-
-#else // _WIN32_WINNT < 0x0600
-
-/* atomic init state of dynamically loaded functions */
-static LONG w32thread_init_state = 0;
-static av_unused void w32thread_init(void);
-
-/* for pre-Windows 6.0 platforms, define INIT_ONCE struct,
- * compatible to the one used in the native API */
-
-typedef union pthread_once_t  {
-    void * Ptr;    ///< For the Windows 6.0+ native functions
-    LONG state;    ///< For the pre-Windows 6.0 compat code
-} pthread_once_t;
-
-#define PTHREAD_ONCE_INIT {0}
-
-/* function pointers to init once API on windows 6.0+ kernels */
-static BOOL (WINAPI *initonce_begin)(pthread_once_t *lpInitOnce, DWORD dwFlags, BOOL *fPending, void **lpContext);
-static BOOL (WINAPI *initonce_complete)(pthread_once_t *lpInitOnce, DWORD dwFlags, void *lpContext);
-
-/* pre-Windows 6.0 compat using a spin-lock */
-static inline void w32thread_once_fallback(LONG volatile *state, void (*init_routine)(void))
-{
-    switch (InterlockedCompareExchange(state, 1, 0)) {
-    /* Initial run */
-    case 0:
-        init_routine();
-        InterlockedExchange(state, 2);
-        break;
-    /* Another thread is running init */
-    case 1:
-        while (1) {
-            MemoryBarrier();
-            if (*state == 2)
-                break;
-            Sleep(0);
-        }
-        break;
-    /* Initialization complete */
-    case 2:
-        break;
-    }
-}
-
-static av_unused int pthread_once(pthread_once_t *once_control, void (*init_routine)(void))
-{
-    w32thread_once_fallback(&w32thread_init_state, w32thread_init);
-
-    /* Use native functions on Windows 6.0+ */
-    if (initonce_begin && initonce_complete) {
-        BOOL pending = FALSE;
-        initonce_begin(once_control, 0, &pending, NULL);
-        if (pending)
-            init_routine();
-        initonce_complete(once_control, 0, NULL);
-        return 0;
-    }
-
-    w32thread_once_fallback(&once_control->state, init_routine);
-    return 0;
-}
-
 /* for pre-Windows 6.0 platforms we need to define and use our own condition
 * variable and api */
-
 typedef struct  win32_cond_t {
    pthread_mutex_t mtx_broadcast;
    pthread_mutex_t mtx_waiter_count;
@@ -251,47 +127,36 @@ typedef struct  win32_cond_t {
    volatile int is_broadcast;
 } win32_cond_t;

-/* function pointers to conditional variable API on windows 6.0+ kernels */
-static void (WINAPI *cond_broadcast)(pthread_cond_t *cond);
-static void (WINAPI *cond_init)(pthread_cond_t *cond);
-static void (WINAPI *cond_signal)(pthread_cond_t *cond);
-static BOOL (WINAPI *cond_wait)(pthread_cond_t *cond, pthread_mutex_t *mutex,
-                                DWORD milliseconds);
-
-static av_unused int pthread_cond_init(pthread_cond_t *cond, const void *unused_attr)
+static void pthread_cond_init(pthread_cond_t *cond, const void *unused_attr)
 {
    win32_cond_t *win32_cond = NULL;
-
-    w32thread_once_fallback(&w32thread_init_state, w32thread_init);
-
    if (cond_init) {
        cond_init(cond);
-        return 0;
+        return;
    }

    /* non native condition variables */
    win32_cond = av_mallocz(sizeof(win32_cond_t));
    if (!win32_cond)
-        return ENOMEM;
-    cond->Ptr = win32_cond;
+        return;
+    cond->ptr = win32_cond;
    win32_cond->semaphore = CreateSemaphore(NULL, 0, 0x7fffffff, NULL);
    if (!win32_cond->semaphore)
-        return ENOMEM;
+        return;
    win32_cond->waiters_done = CreateEvent(NULL, TRUE, FALSE, NULL);
    if (!win32_cond->waiters_done)
-        return ENOMEM;
+        return;

    pthread_mutex_init(&win32_cond->mtx_waiter_count, NULL);
    pthread_mutex_init(&win32_cond->mtx_broadcast, NULL);
-    return 0;
 }

-static av_unused int pthread_cond_destroy(pthread_cond_t *cond)
+static void pthread_cond_destroy(pthread_cond_t *cond)
 {
-    win32_cond_t *win32_cond = cond->Ptr;
+    win32_cond_t *win32_cond = cond->ptr;
    /* native condition variables do not destroy */
    if (cond_init)
-        return 0;
+        return;

    /* non native condition variables */
    CloseHandle(win32_cond->semaphore);
@@ -299,18 +164,17 @@ static av_unused int pthread_cond_destroy(pthread_cond_t *cond)
    pthread_mutex_destroy(&win32_cond->mtx_waiter_count);
    pthread_mutex_destroy(&win32_cond->mtx_broadcast);
    av_freep(&win32_cond);
-    cond->Ptr = NULL;
-    return 0;
+    cond->ptr = NULL;
 }

-static av_unused int pthread_cond_broadcast(pthread_cond_t *cond)
+static void pthread_cond_broadcast(pthread_cond_t *cond)
 {
-    win32_cond_t *win32_cond = cond->Ptr;
+    win32_cond_t *win32_cond = cond->ptr;
    int have_waiter;

    if (cond_broadcast) {
        cond_broadcast(cond);
-        return 0;
+        return;
    }

    /* non native condition variables */
@@ -332,12 +196,11 @@ static av_unused int pthread_cond_broadcast(pthread_cond_t *cond)
    } else
        pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
    pthread_mutex_unlock(&win32_cond->mtx_broadcast);
-    return 0;
 }

-static av_unused int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
+static int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
 {
-    win32_cond_t *win32_cond = cond->Ptr;
+    win32_cond_t *win32_cond = cond->ptr;
    int last_waiter;
    if (cond_wait) {
        cond_wait(cond, mutex, INFINITE);
@@ -367,13 +230,13 @@ static av_unused int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mu
    return pthread_mutex_lock(mutex);
 }

-static av_unused int pthread_cond_signal(pthread_cond_t *cond)
+static void pthread_cond_signal(pthread_cond_t *cond)
 {
-    win32_cond_t *win32_cond = cond->Ptr;
+    win32_cond_t *win32_cond = cond->ptr;
    int have_waiter;
    if (cond_signal) {
        cond_signal(cond);
-        return 0;
+        return;
    }

    pthread_mutex_lock(&win32_cond->mtx_broadcast);
@@ -390,11 +253,9 @@ static av_unused int pthread_cond_signal(pthread_cond_t *cond)
    }

    pthread_mutex_unlock(&win32_cond->mtx_broadcast);
-    return 0;
 }
-#endif

-static av_unused void w32thread_init(void)
+static void w32thread_init(void)
 {
 #if _WIN32_WINNT < 0x0600
    HANDLE kernel_dll = GetModuleHandle(TEXT("kernel32.dll"));
@@ -407,12 +268,13 @@ static av_unused void w32thread_init(void)
        (void*)GetProcAddress(kernel_dll, "WakeConditionVariable");
    cond_wait      =
        (void*)GetProcAddress(kernel_dll, "SleepConditionVariableCS");
-    initonce_begin =
-        (void*)GetProcAddress(kernel_dll, "InitOnceBeginInitialize");
-    initonce_complete =
-        (void*)GetProcAddress(kernel_dll, "InitOnceComplete");
+#else
+    cond_init      = InitializeConditionVariable;
+    cond_broadcast = WakeAllConditionVariable;
+    cond_signal    = WakeConditionVariable;
+    cond_wait      = SleepConditionVariableCS;
 #endif

 }

-#endif /* COMPAT_W32PTHREADS_H */
+#endif /* LIBAV_W32PTHREADS_H */
--- a/compat/windows/makedef
+++ b/compat/windows/makedef
@@ -1,132 +0,0 @@
-#!/bin/sh
-
-# Copyright (c) 2013, Derek Buitenhuis
-#
-# Permission to use, copy, modify, and/or distribute this software for any
-# purpose with or without fee is hereby granted, provided that the above
-# copyright notice and this permission notice appear in all copies.
-#
-# THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
-# WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
-# MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
-# ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
-# WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
-# ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
-# OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
-
-# mktemp isn't POSIX, so supply an implementation
-mktemp() {
-    echo "${2%%XXX*}.${HOSTNAME}.${UID}.$$"
-}
-
-if [ $# -lt 2 ]; then
-    echo "Usage: makedef <version_script> <objects>" >&2
-    exit 0
-fi
-
-vscript=$1
-shift
-
-if [ ! -f "$vscript" ]; then
-    echo "Version script does not exist" >&2
-    exit 1
-fi
-
-for object in "$@"; do
-    if [ ! -f "$object" ]; then
-        echo "Object does not exist: ${object}" >&2
-        exit 1
-    fi
-done
-
-# Create a lib temporarily to dump symbols from.
-# It's just much easier to do it this way
-libname=$(mktemp -u "library").lib
-
-trap 'rm -f -- $libname' EXIT
-
-lib -out:${libname} $@ >/dev/null
-if [ $? != 0 ]; then
-    echo "Could not create temporary library." >&2
-    exit 1
-fi
-
-IFS='
-'
-
-# Determine if we're building for x86 or x86_64 and
-# set the symbol prefix accordingly.
-prefix=""
-arch=$(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"
-
-for line in $(cat ${vscript} | tr '\t' ' '); do
-    # We only care about global symbols
-    echo "${line}" | grep -q '^ \+global:'
-    if [ $? = 0 ]; then
-        started=1
-        line=$(echo "${line}" | sed -e 's/^ \{1,\}global: *//')
-    else
-        echo "${line}" | grep -q '^ \+local:'
-        if [ $? = 0 ]; then
-            started=0
-        fi
-    fi
-
-    if [ ${started} = 0 ]; then
-        continue
-    fi
-
-    # Handle multiple symbols on one line
-    IFS=';'
-
-    # Work around stupid expansion to filenames
-    line=$(echo "${line}" | sed -e 's/\*/.\\+/g')
-    for exp in ${line}; do
-        # Remove leading and trailing whitespace
-        exp=$(echo "${exp}" | sed -e 's/^ *//' -e 's/ *$//')
-
-        if [ "${regex}" = "none" ]; then
-            regex="${exp}"
-        else
-            regex="${regex};${exp}"
-        fi
-    done
-
-    IFS='
-'
-done
-
-dump=$(dumpbin -linkermember:1 ${libname})
-
-rm ${libname}
-
-IFS=';'
-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
-
-echo "EXPORTS"
-echo "${list}" | sort | uniq | tail -n +2
--- a/compat/windows/mslink
+++ b/compat/windows/mslink
@@ -1,9 +0,0 @@
-#!/bin/sh
-
-LINK_EXE_PATH=$(dirname "$(command -v cl)")/link
-if [ -x "$LINK_EXE_PATH" ]; then
-    "$LINK_EXE_PATH" $@
-else
-    link $@
-fi
-exit $?
--- a/3207
+++ b/3207
--- a/doc/APIchanges
+++ b/doc/APIchanges
--- a/doc/Doxyfile
+++ b/doc/Doxyfile
@@ -31,9 +31,9 @@ PROJECT_NAME           = FFmpeg
 # This could be handy for archiving the generated documentation or
 # if some version control system is used.

-PROJECT_NUMBER         =
+PROJECT_NUMBER         = 2.0.4

-# With the PROJECT_LOGO tag one can specify a logo or icon that is included
+# With the PROJECT_LOGO tag one can specify an logo or icon that is included
 # in the documentation. The maximum height of the logo should not exceed 55
 # pixels and the maximum width should not exceed 200 pixels. Doxygen will
 # copy the logo to the output directory.
@@ -759,7 +759,7 @@ ALPHABETICAL_INDEX     = YES
 # the COLS_IN_ALPHA_INDEX tag can be used to specify the number of columns
 # in which this list will be split (can be a number in the range [1..20])

-COLS_IN_ALPHA_INDEX    = 5
+COLS_IN_ALPHA_INDEX    = 2

 # In case all classes in a project start with a common prefix, all
 # classes will be put under the same header in the alphabetical index.
@@ -793,13 +793,13 @@ HTML_FILE_EXTENSION    = .html
 # each generated HTML page. If it is left blank doxygen will generate a
 # standard header.

-HTML_HEADER            =
+#HTML_HEADER            = doc/doxy/header.html

 # The HTML_FOOTER tag can be used to specify a personal HTML footer for
 # each generated HTML page. If it is left blank doxygen will generate a
 # standard footer.

-HTML_FOOTER            =
+#HTML_FOOTER            = doc/doxy/footer.html

 # The HTML_STYLESHEET tag can be used to specify a user-defined cascading
 # style sheet that is used by each HTML page. It can be used to
@@ -808,7 +808,7 @@ HTML_FOOTER            =
 # the style sheet file to the HTML output directory, so don't put your own
 # stylesheet in the HTML output directory as well, or it will be erased!

-HTML_STYLESHEET        =
+#HTML_STYLESHEET        = doc/doxy/doxy_stylesheet.css

 # The HTML_COLORSTYLE_HUE tag controls the color of the HTML output.
 # Doxygen will adjust the colors in the stylesheet and background images
@@ -1056,7 +1056,7 @@ FORMULA_TRANSPARENT    = YES
 # typically be disabled. For large projects the javascript based search engine
 # can be slow, then enabling SERVER_BASED_SEARCH may provide a better solution.

-SEARCHENGINE           = YES
+SEARCHENGINE           = NO

 # When the SERVER_BASED_SEARCH tag is enabled the search engine will be
 # implemented using a PHP enabled web server instead of at the web client
@@ -1359,9 +1359,6 @@ PREDEFINED             = "__attribute__(x)=" \
                         "DECLARE_ALIGNED(a,t,n)=t n" \
                         "offsetof(x,y)=0x42" \
                         av_alloc_size \
-                         AV_GCC_VERSION_AT_LEAST(x,y)=1 \
-                         AV_GCC_VERSION_AT_MOST(x,y)=0 \
-                         __GNUC__=1 \

 # If the MACRO_EXPANSION and EXPAND_ONLY_PREDEF tags are set to YES then
 # this tag can be used to specify a list of macro names that should be expanded.
--- a/doc/Makefile
+++ b/doc/Makefile
@@ -14,11 +14,11 @@ COMPONENTS-$(CONFIG_AVFORMAT)   += ffmpeg-formats ffmpeg-protocols
 COMPONENTS-$(CONFIG_AVDEVICE)   += ffmpeg-devices
 COMPONENTS-$(CONFIG_AVFILTER)   += ffmpeg-filters

-MANPAGES1   = $(AVPROGS-yes:%=doc/%.1)    $(AVPROGS-yes:%=doc/%-all.1)    $(COMPONENTS-yes:%=doc/%.1)
+MANPAGES1   = $(PROGS-yes:%=doc/%.1)    $(PROGS-yes:%=doc/%-all.1)    $(COMPONENTS-yes:%=doc/%.1)
 MANPAGES3   = $(LIBRARIES-yes:%=doc/%.3)
 MANPAGES    = $(MANPAGES1) $(MANPAGES3)
-PODPAGES    = $(AVPROGS-yes:%=doc/%.pod)  $(AVPROGS-yes:%=doc/%-all.pod)  $(COMPONENTS-yes:%=doc/%.pod)  $(LIBRARIES-yes:%=doc/%.pod)
-HTMLPAGES   = $(AVPROGS-yes:%=doc/%.html) $(AVPROGS-yes:%=doc/%-all.html) $(COMPONENTS-yes:%=doc/%.html) $(LIBRARIES-yes:%=doc/%.html) \
+PODPAGES    = $(PROGS-yes:%=doc/%.pod)  $(PROGS-yes:%=doc/%-all.pod)  $(COMPONENTS-yes:%=doc/%.pod)  $(LIBRARIES-yes:%=doc/%.pod)
+HTMLPAGES   = $(PROGS-yes:%=doc/%.html) $(PROGS-yes:%=doc/%-all.html) $(COMPONENTS-yes:%=doc/%.html) $(LIBRARIES-yes:%=doc/%.html) \
              doc/developer.html                                        \
              doc/faq.html                                              \
              doc/fate.html                                             \
@@ -36,30 +36,6 @@ DOCS-$(CONFIG_MANPAGES)  += $(MANPAGES)
 DOCS-$(CONFIG_TXTPAGES)  += $(TXTPAGES)
 DOCS = $(DOCS-yes)

-DOC_EXAMPLES-$(CONFIG_AVIO_DIR_CMD_EXAMPLE)      += avio_dir_cmd
-DOC_EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE)      += avio_reading
-DOC_EXAMPLES-$(CONFIG_AVCODEC_EXAMPLE)           += avcodec
-DOC_EXAMPLES-$(CONFIG_DECODING_ENCODING_EXAMPLE) += decoding_encoding
-DOC_EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding
-DOC_EXAMPLES-$(CONFIG_EXTRACT_MVS_EXAMPLE)       += extract_mvs
-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_QSVDEC_EXAMPLE)            += qsvdec
-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
@@ -67,9 +43,7 @@ doc: documentation
 apidoc: doc/doxy/html
 documentation: $(DOCS)

-examples: $(DOC_EXAMPLES)
-
-TEXIDEP = perl $(SRC_PATH)/doc/texidep.pl $(SRC_PATH) $< $@ >$(@:%=%.d)
+TEXIDEP = awk '/^@(verbatim)?include/ { printf "$@: $(@D)/%s\n", $$2 }' <$< >$(@:%=%.d)

 doc/%.txt: TAG = TXT
 doc/%.txt: doc/%.texi
@@ -84,25 +58,14 @@ $(GENTEXI): doc/avoptions_%.texi: doc/print_options$(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)
@@ -116,35 +79,17 @@ 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) $(DOC_EXAMPLES:%$(EXESUF)=%.c) $(LIB_EXAMPLES:%$(EXESUF)=%.c)
-DOXY_INPUT_DEPS = $(addprefix $(SRC_PATH)/, $(DOXY_INPUT))
-
-doc/doxy/html: TAG = DOXY
-doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(SRC_PATH)/doc/doxy-wrapper.sh $(DOXY_INPUT_DEPS)
-	$(M)OUT_DIR=$$PWD/doc/doxy; cd $(SRC_PATH); ./doc/doxy-wrapper.sh $$OUT_DIR $< $(DOXYGEN) $(DOXY_INPUT);
-
-install-doc: install-html install-man
-
-install-html:
+doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(INSTHEADERS)
+	$(M)$(SRC_PATH)/doc/doxy-wrapper.sh $(SRC_PATH) $^

 install-man:

-ifdef CONFIG_HTMLPAGES
-install-progs-$(CONFIG_DOC): install-html
-
-install-html: $(HTMLPAGES)
-	$(Q)mkdir -p "$(DOCDIR)"
-	$(INSTALL) -m 644 $(HTMLPAGES) "$(DOCDIR)"
-endif
-
 ifdef CONFIG_MANPAGES
 install-progs-$(CONFIG_DOC): install-man

@@ -155,15 +100,10 @@ install-man: $(MANPAGES)
 	$(INSTALL) -m 644 $(MANPAGES3) "$(MANDIR)/man3"
 endif

-uninstall: uninstall-doc
-
-uninstall-doc: uninstall-html uninstall-man
-
-uninstall-html:
-	$(RM) -r "$(DOCDIR)"
+uninstall: uninstall-man

 uninstall-man:
-	$(RM) $(addprefix "$(MANDIR)/man1/",$(AVPROGS-yes:%=%.1) $(AVPROGS-yes:%=%-all.1) $(COMPONENTS-yes:%=%.1))
+	$(RM) $(addprefix "$(MANDIR)/man1/",$(PROGS-yes:%=%.1) $(PROGS-yes:%=%-all.1) $(COMPONENTS-yes:%=%.1))
 	$(RM) $(addprefix "$(MANDIR)/man3/",$(LIBRARIES-yes:%=%.3))

 clean:: docclean
@@ -171,13 +111,8 @@ clean:: docclean
 distclean:: docclean
 	$(RM) doc/config.texi

-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
+docclean:
+	$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 doc/*.3 $(CLEANSUFFIXES:%=doc/%) doc/avoptions_*.texi
 	$(RM) -r doc/doxy/html

 -include $(wildcard $(DOCS:%=%.d))
--- a/doc/RELEASE_NOTES
+++ b/doc/RELEASE_NOTES
@@ -0,0 +1,16 @@
+Release Notes
+=============
+
+* 2.0 "Nameless"  July, 2013
+
+
+General notes
+-------------
+See the Changelog file for a list of significant changes. Note, there
+are many more new features and bugfixes than whats listed there.
+
+Bugreports against FFmpeg git master or the most recent FFmpeg release are
+accepted. If you are experiencing issues with any formally released version of
+FFmpeg, please try git master to check if the issue still exists. If it does,
+make your report against the development code following the usual bug reporting
+guidelines.
--- a/doc/avtools-common-opts.texi
+++ b/doc/avtools-common-opts.texi
@@ -3,7 +3,7 @@ representing a number as input, which may be followed by one of the SI
 unit prefixes, for example: 'K', 'M', or 'G'.

 If 'i' is appended to the SI unit prefix, the complete prefix will be
-interpreted as a unit prefix for binary multiples, which are based on
+interpreted as a unit prefix for binary multiplies, which are based on
 powers of 1024 instead of powers of 1000. Appending 'B' to the SI unit
 prefix multiplies the value by 8. This allows using, for example:
 'KB', 'MiB', 'G' and 'B' as number suffixes.
@@ -36,28 +36,16 @@ Possible forms of stream specifiers are:
 Matches the stream with this index. E.g. @code{-threads:1 4} would set the
 thread count for the second stream to 4.
@item @var{stream_type}[:@var{stream_index}]
-@var{stream_type} is one of following: 'v' or 'V' for video, 'a' for audio, 's'
-for subtitle, 'd' for data, and 't' for attachments. 'v' matches all video
-streams, 'V' only matches video streams which are not attached pictures, video
-thumbnails or cover arts.  If @var{stream_index} is given, then it matches
+@var{stream_type} is one of following: 'v' for video, 'a' for audio, 's' for subtitle,
+'d' for data, and 't' for attachments. If @var{stream_index} is given, then it matches
 stream number @var{stream_index} of this type. Otherwise, it matches all
 streams of this type.
@item p:@var{program_id}[:@var{stream_index}]
 If @var{stream_index} is given, then it matches the stream with number @var{stream_index}
 in the program with the id @var{program_id}. Otherwise, it matches all streams in the
 program.
-@item #@var{stream_id} or i:@var{stream_id}
-Match the stream by stream id (e.g. PID in MPEG-TS container).
-@item m:@var{key}[:@var{value}]
-Matches streams with the metadata tag @var{key} having the specified value. If
-@var{value} is not given, matches streams that contain the given tag with any
-value.
-@item u
-Matches streams with usable configuration, the codec must be defined and the
-essential information such as video dimension or audio sample rate must be present.
-
-Note that in @command{ffmpeg}, matching by metadata will only work properly for
-input files.
+@item #@var{stream_id}
+Matches the stream by a format-specific ID.
@end table

@section Generic options
@@ -71,18 +59,10 @@ Show license.

@item -h, -?, -help, --help [@var{arg}]
 Show help. An optional parameter may be specified to print help about a specific
-item. If no argument is specified, only basic (non advanced) tool
-options are shown.
+item.

 Possible values of @var{arg} are:
@table @option
-@item long
-Print advanced tool options in addition to the basic tool options.
-
-@item full
-Print complete list of options, including shared and private options
-for encoders, decoders, demuxers, muxers, filters, etc.
-
@item decoder=@var{decoder_name}
 Print detailed information about the decoder named @var{decoder_name}. Use the
@option{-decoders} option to get a list of all decoders.
@@ -102,16 +82,14 @@ Print detailed information about the muxer named @var{muxer_name}. Use the
@item filter=@var{filter_name}
 Print detailed information about the filter name @var{filter_name}. Use the
@option{-filters} option to get a list of all filters.
+
@end table

@item -version
 Show version.

@item -formats
-Show available formats (including devices).
-
-@item -devices
-Show available devices.
+Show available formats.

@item -codecs
 Show all codecs known to libavcodec.
@@ -143,25 +121,6 @@ Show available sample formats.
@item -layouts
 Show channel names and standard channel layouts.

-@item -colors
-Show recognized color names.
-
-@item -sources @var{device}[,@var{opt1}=@var{val1}[,@var{opt2}=@var{val2}]...]
-Show autodetected sources of the intput device.
-Some devices may provide system-dependent source names that cannot be autodetected.
-The returned list cannot be assumed to be always complete.
-@example
-ffmpeg -sources pulse,server=192.168.0.4
-@end example
-
-@item -sinks @var{device}[,@var{opt1}=@var{val1}[,@var{opt2}=@var{val2}]...]
-Show autodetected sinks of the output device.
-Some devices may provide system-dependent sink names that cannot be autodetected.
-The returned list cannot be assumed to be always complete.
-@example
-ffmpeg -sinks pulse,server=192.168.0.4
-@end example
-
@item -loglevel [repeat+]@var{loglevel} | -v [repeat+]@var{loglevel}
 Set the logging level used by the library.
 Adding "repeat+" indicates that repeated log output should not be compressed
@@ -170,29 +129,28 @@ omitted. "repeat" can also be used alone.
 If "repeat" is used alone, and with no prior loglevel set, the default
 loglevel will be used. If multiple loglevel parameters are given, using
 'repeat' will not change the loglevel.
-@var{loglevel} is a string or a number containing one of the following values:
+@var{loglevel} is a number or a string containing one of the following values:
@table @samp
-@item quiet, -8
+@item quiet
 Show nothing at all; be silent.
-@item panic, 0
+@item panic
 Only show fatal errors which could lead the process to crash, such as
 and assert failure. This is not currently used for anything.
-@item fatal, 8
+@item fatal
 Only show fatal errors. These are errors after which the process absolutely
 cannot continue after.
-@item error, 16
+@item error
 Show all errors, including ones which can be recovered from.
-@item warning, 24
+@item warning
 Show all warnings and errors. Any message related to possibly
 incorrect or unexpected events will be shown.
-@item info, 32
+@item info
 Show informative messages during processing. This is in addition to
 warnings and errors. This is the default value.
-@item verbose, 40
+@item verbose
 Same as @code{info}, except more verbose.
-@item debug, 48
+@item debug
 Show everything, including debugging information.
-@item trace, 56
@end table

 By default the program logs to stderr, if coloring is supported by the
@@ -210,39 +168,22 @@ directory.
 This file can be useful for bug reports.
 It also implies @code{-loglevel verbose}.

-Setting the environment variable @env{FFREPORT} to any value has the
+Setting the environment variable @code{FFREPORT} to any value has the
 same effect. If the value is a ':'-separated key=value sequence, these
-options will affect the report; option values must be escaped if they
+options will affect the report; options values must be escaped if they
 contain special characters or the options delimiter ':' (see the
-``Quoting and escaping'' section in the ffmpeg-utils manual).
-
-The following options are recognized:
+``Quoting and escaping'' section in the ffmpeg-utils manual). The
+following option is recognized:
@table @option
@item file
 set the file name to use for the report; @code{%p} is expanded to the name
 of the program, @code{%t} is expanded to a timestamp, @code{%%} is expanded
 to a plain @code{%}
-@item level
-set the log verbosity level using a numerical value (see @code{-loglevel}).
@end table

-For example, to output a report to a file named @file{ffreport.log}
-using a log level of @code{32} (alias for log level @code{info}):
-
-@example
-FFREPORT=file=ffreport.log:level=32 ffmpeg -i input output
-@end example
-
 Errors in parsing the environment variable are not fatal, and will not
 appear in the report.

-@item -hide_banner
-Suppress printing banner.
-
-All FFmpeg tools will normally show a copyright notice, build options
-and library versions. This option can be used to suppress printing
-this information.
-
@item -cpuflags flags (@emph{global})
 Allows setting and clearing cpu flags. This option is intended
 for testing. Do not use it unless you know what you're doing.
@@ -267,14 +208,10 @@ Possible flags for this option are:
@item sse4.1
@item sse4.2
@item avx
-@item avx2
@item xop
-@item fma3
@item fma4
@item 3dnow
@item 3dnowext
-@item bmi1
-@item bmi2
@item cmov
@end table
@item ARM
@@ -285,13 +222,6 @@ Possible flags for this option are:
@item vfp
@item vfpv3
@item neon
-@item setend
-@end table
-@item AArch64
-@table @samp
-@item armv8
-@item vfp
-@item neon
@end table
@item PowerPC
@table @samp
@@ -310,43 +240,6 @@ Possible flags for this option are:
@end table
@end table

-@item -opencl_bench
-This option is used to benchmark all available OpenCL devices and print the
-results. This option is only available when FFmpeg has been compiled with
-@code{--enable-opencl}.
-
-When FFmpeg is configured with @code{--enable-opencl}, the options for the
-global OpenCL context are set via @option{-opencl_options}. See the
-"OpenCL Options" section in the ffmpeg-utils manual for the complete list of
-supported options. Amongst others, these options include the ability to select
-a specific platform and device to run the OpenCL code on. By default, FFmpeg
-will run on the first device of the first platform. While the options for the
-global OpenCL context provide flexibility to the user in selecting the OpenCL
-device of their choice, most users would probably want to select the fastest
-OpenCL device for their system.
-
-This option assists the selection of the most efficient configuration by
-identifying the appropriate device for the user's system. The built-in
-benchmark is run on all the OpenCL devices and the performance is measured for
-each device. The devices in the results list are sorted based on their
-performance with the fastest device listed first. The user can subsequently
-invoke @command{ffmpeg} using the device deemed most appropriate via
-@option{-opencl_options} to obtain the best performance for the OpenCL
-accelerated code.
-
-Typical usage to use the fastest OpenCL device involve the following steps.
-
-Run the command:
-@example
-ffmpeg -opencl_bench
-@end example
-Note down the platform ID (@var{pidx}) and device ID (@var{didx}) of the first
-i.e. fastest device in the list.
-Select the platform and device using the command:
-@example
-ffmpeg -opencl_options platform_idx=@var{pidx}:device_idx=@var{didx} ...
-@end example
-
@item -opencl_options options (@emph{global})
 Set OpenCL environment options. This option is only available when
 FFmpeg has been compiled with @code{--enable-opencl}.
@@ -378,12 +271,11 @@ muxer:
 ffmpeg -i input.flac -id3v2_version 3 out.mp3
@end example

-All codec AVOptions are per-stream, and thus a stream specifier
-should be attached to them.
+All codec AVOptions are obviously per-stream, so the chapter on stream
+specifiers applies to them

-Note: the @option{-nooption} syntax cannot be used for boolean
-AVOptions, use @option{-option 0}/@option{-option 1}.
+Note @option{-nooption} syntax cannot be used for boolean AVOptions,
+use @option{-option 0}/@option{-option 1}.

-Note: the old undocumented way of specifying per-stream AVOptions by
-prepending v/a/s to the options name is now obsolete and will be
-removed soon.
+Note2 old undocumented way of specifying per-stream AVOptions by prepending
+v/a/s to the options name is now obsolete and will be removed soon.
--- a/doc/avutil.txt
+++ b/doc/avutil.txt
@@ -0,0 +1,36 @@
+AVUtil
+======
+libavutil is a small lightweight library of generally useful functions.
+It is not a library for code needed by both libavcodec and libavformat.
+
+
+Overview:
+=========
+adler32.c               adler32 checksum
+aes.c                   AES encryption and decryption
+fifo.c                  resizeable first in first out buffer
+intfloat_readwrite.c    portable reading and writing of floating point values
+log.c                   "printf" with context and level
+md5.c                   MD5 Message-Digest Algorithm
+rational.c              code to perform exact calculations with rational numbers
+tree.c                  generic AVL tree
+crc.c                   generic CRC checksumming code
+integer.c               128bit integer math
+lls.c
+mathematics.c           greatest common divisor, integer sqrt, integer log2, ...
+mem.c                   memory allocation routines with guaranteed alignment
+
+Headers:
+bswap.h                 big/little/native-endian conversion code
+x86_cpu.h               a few useful macros for unifying x86-64 and x86-32 code
+avutil.h
+common.h
+intreadwrite.h          reading and writing of unaligned big/little/native-endian integers
+
+
+Goals:
+======
+* Modular (few interdependencies and the possibility of disabling individual parts during ./configure)
+* Small (source and object)
+* Efficient (low CPU and memory usage)
+* Useful (avoid useless features almost no one needs)
--- a/doc/bitstream_filters.texi
+++ b/doc/bitstream_filters.texi
@@ -13,16 +13,7 @@ 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
-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.
+Below is a description of the currently available bitstream filters.

@section aac_adtstoasc

@@ -39,33 +30,7 @@ ADTS AAC container to a FLV or a MOV/MP4 file.

 Remove zero padding at the end of a packet.

-@section dump_extra
-
-Add extradata to the beginning of the filtered packets.
-
-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
-add extradata to all key packets
-
-@item e
-add extradata to all packets
-@end table
-
-If not specified it is assumed @samp{k}.
-
-For example the following @command{ffmpeg} command forces a global
-header (thus disabling individual packet headers) in the H.264 packets
-generated by the @code{libx264} encoder, but corrects them by adding
-the header stored in extradata to the key packets:
-@example
-ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts
-@end example
+@section dump_extradata

@section h264_mp4toannexb

@@ -83,18 +48,7 @@ format with @command{ffmpeg}, you can use the command:
 ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts
@end example

-@section imxdump
-
-Modifies the bitstream to fit in MOV and to be usable by the Final Cut
-Pro decoder. This filter only applies to the mpeg2video codec, and is
-likely not needed for Final Cut Pro 7 and newer with the appropriate
-@option{-tag:v}.
-
-For example, to remux 30 MB/sec NTSC IMX to MOV:
-
-@example
-ffmpeg -i input.mxf -c copy -bsf:v imxdump -tag:v mx3n output.mov
-@end example
+@section imx_dump_header

@section mjpeg2jpeg

@@ -137,44 +91,12 @@ ffmpeg -i frame_%d.jpg -c:v copy rotated.avi

@section movsub

+@section mp3_header_compress
+
@section mp3_header_decompress

-@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 without damaging the container. Can be
-used for fuzzing or testing error resilience/concealment.
-
-Parameters:
-A numeral string, whose value is related to how often output bytes will
-be modified. Therefore, values below or equal to 0 are forbidden, and
-the lower the more frequent bytes will be modified, with 1 meaning
-every byte is modified.
-
-@example
-ffmpeg -i INPUT -c copy -bsf noise[=1] output.mkv
-@end example
-applies the modification to every byte.
-
-@section remove_extra
+@section remove_extradata

@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
@@ -7,35 +7,24 @@ 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.

-examples
-    Build all examples located in doc/examples.
-
 libavformat/output-example
    Build the libavformat basic example.

@@ -43,22 +32,19 @@ libavcodec/api-example
    Build the libavcodec basic example.

 libswscale/swscale-test
-    Build the swscale self-test (useful also as an example).
-
-config
-    Reconfigure the project with the current configuration.
+    Build the swscale self-test (useful also as example).


 Useful standard make commands:
 make -t <target>
-    Touch all files that otherwise would be built, this is useful to reduce
-    unneeded rebuilding when changing headers, but note that you must force rebuilds
+    Touch all files that otherwise would be build, this is useful to reduce
+    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
-    sometimes but note that it will still not run all reg tests.
+    continue build in case of errors, this is useful for the regression 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.

@@ -71,9 +71,7 @@ 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).
-This ensures that file and data checksums are reproducible and match between
-platforms. Its primary use is for regression testing.
+Use only bitexact stuff (except (I)DCT).
@item aic
 Apply H263 advanced intra coding / mpeg4 ac prediction.
@item cbp
@@ -86,6 +84,9 @@ Apply interlaced motion estimation.
 Use closed gop.
@end table

+@item sub_id @var{integer}
+Deprecated, currently unused.
+
@item me_method @var{integer} (@emph{encoding,video})
 Set motion estimation method.

@@ -129,7 +130,7 @@ 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).
@@ -174,13 +175,7 @@ Set max video quantizer scale (VBR). Must be included between -1 and
 Set max difference between the quantizer scale (VBR).

@item bf @var{integer} (@emph{encoding,video})
-Set max number of B frames between non-B-frames.
-
-Must be an integer between -1 and 16. 0 means that B-frames are
-disabled. If a value of -1 is used, it will choose an automatic value
-depending on the encoder.
-
-Default value is 0.
+Set max number of B frames.

@item b_qfactor @var{float} (@emph{encoding,video})
 Set qp factor between P and B frames.
@@ -287,11 +282,6 @@ detect bitstream specification deviations
 detect improper bitstream length
@item explode
 abort decoding on minor error detection
-@item ignore_err
-ignore decoding errors, and continue decoding.
-This is useful if you want to analyze the content of a video and thus want
-everything to be decoded no matter what. This option will not result in a video
-that is pleasing to watch in case of errors.
@item careful
 consider things that violate the spec and have not been seen in the wild as errors
@item compliant
@@ -396,8 +386,9 @@ Possible values:

@item simplemmx

-@item simpleauto
-Automatically pick a IDCT compatible with the simple one
+@item libmpeg2mmx
+
+@item mmi

@item arm

@@ -415,6 +406,10 @@ Automatically pick a IDCT compatible with the simple one

@item simplealpha

+@item h264
+
+@item vp3
+
@item ipp

@item xvidmmx
@@ -434,8 +429,6 @@ Possible values:
 iterative motion vector (MV) search (slow)
@item deblock
 use strong deblock filter for damaged MBs
-@item favor_inter
-favor predicting from the previous frame instead of the current
@end table

@item bits_per_coded_sample @var{integer}
@@ -475,9 +468,6 @@ per-block quantization parameter (QP)
 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
@@ -498,15 +488,11 @@ visualize block types
 picture buffer allocations
@item thread_ops
 threading operations
-@item nomc
-skip motion compensation
@end table

@item vismv @var{integer} (@emph{decoding,video})
 Visualize motion vectors (MVs).

-This option is deprecated, see the codecview filter instead.
-
 Possible values:
@table @samp
@item pf
@@ -786,29 +772,26 @@ Set noise reduction.
 Set number of bits which should be loaded into the rc buffer before
 decoding starts.

+@item inter_threshold @var{integer} (@emph{encoding,video})
+
@item flags2 @var{flags} (@emph{decoding/encoding,audio,video})

 Possible values:
@table @samp
@item fast
-Allow non spec compliant speedup tricks.
+allow non spec compliant speedup tricks
@item sgop
-Deprecated, use mpegvideo private options instead.
+Deprecated, use mpegvideo private options instead
@item noout
-Skip bitstream encoding.
-@item ignorecrop
-Ignore cropping information from sps.
+skip bitstream encoding
@item local_header
-Place global headers at every keyframe instead of in extradata.
+place global headers at every keyframe instead of in extradata
@item chunks
-Frame data might be split into multiple chunks.
+Frame data might be split into multiple chunks
@item showall
-Show all frames before the first keyframe.
+Show all frames before the first keyframe
@item skiprd
-Deprecated, use mpegvideo private options instead.
-@item export_mvs
-Export motion vectors into frame side-data (see @code{AV_FRAME_DATA_MOTION_VECTORS})
-for codecs that support it. See also @file{doc/examples/export_mvs.c}.
+Deprecated, use mpegvideo private options instead
@end table

@item error @var{integer} (@emph{encoding,video})
@@ -817,17 +800,13 @@ for codecs that support it. See also @file{doc/examples/export_mvs.c}.
 Deprecated, use mpegvideo private options instead.

@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
-automatically select the number of threads to set
+@item auto
+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.

@@ -868,18 +847,6 @@ Possible values:

@item aac_eld

-@item mpeg2_aac_low
-
-@item mpeg2_aac_he
-
-@item mpeg4_sp
-
-@item mpeg4_core
-
-@item mpeg4_main
-
-@item mpeg4_asp
-
@item dts

@item dts_es
@@ -911,9 +878,6 @@ 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.
@@ -1049,11 +1013,7 @@ Possible values:
@item color_primaries @var{integer} (@emph{decoding/encoding,video})
@item color_trc @var{integer} (@emph{decoding/encoding,video})
@item colorspace @var{integer} (@emph{decoding/encoding,video})
-
@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.
-
@item chroma_sample_location @var{integer} (@emph{decoding/encoding,video})

@item log_level_offset @var{integer}
@@ -1063,26 +1023,15 @@ Set the log level offset.
 Number of slices, used in parallelized encoding.

@item thread_type @var{flags} (@emph{decoding/encoding,video})
-Select which multithreading methods to use.
-
-Use of @samp{frame} will increase decoding delay by one frame per
-thread, so clients which cannot provide future frames should not use
-it.
+Select multithreading type.

 Possible values:
@table @samp
@item slice
-Decode more than one part of a single frame at once.
-
-Multithreading using slices works only when the video was encoded with
-slices.

@item frame
-Decode more than one frame at once.
+
@end table
-
-Default value is @samp{slice+frame}.
-
@item audio_service_type @var{integer} (@emph{encoding,audio})
 Set audio service type.

@@ -1116,47 +1065,9 @@ Set sample format audio decoders should prefer. Default value is

@item sub_charenc @var{encoding} (@emph{decoding,subtitles})
 Set the input subtitles character encoding.
-
-@item field_order  @var{field_order} (@emph{video})
-Set/override the field order of the video.
-Possible values:
-@table @samp
-@item progressive
-Progressive video
-@item tt
-Interlaced video, top field coded and displayed first
-@item bb
-Interlaced video, bottom field coded and displayed first
-@item tb
-Interlaced video, top coded first, bottom displayed first
-@item bt
-Interlaced video, bottom coded first, top displayed first
-@end table
-
-@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 indention:
-@example
-ffprobe -dump_separator "
-                          "  -i ~/videos/matrixbench_mpeg2.mpg
-@end example
-
@end table

@c man end CODEC OPTIONS

-@ifclear config-writeonly
@include decoders.texi
-@end ifclear
-@ifclear config-readonly
@include encoders.texi
-@end ifclear
--- a/doc/decoders.texi
+++ b/doc/decoders.texi
@@ -14,7 +14,7 @@ You can disable all the decoders with the configure option
 with the options @code{--enable-decoder=@var{DECODER}} /
@code{--disable-decoder=@var{DECODER}}.

-The option @code{-decoders} of the ff* tools will display the list of
+The option @code{-codecs} of the ff* tools will display the list of
 enabled decoders.

@c man end DECODERS
@@ -25,13 +25,6 @@ enabled decoders.
 A description of some of the currently available video decoders
 follows.

-@section hevc
-
-HEVC / H.265 decoder.
-
-Note: the @option{skip_loop_filter} option has effect only at level
-@code{all}.
-
@section rawvideo

 Raw video decoder.
@@ -59,54 +52,6 @@ top-field-first is assumed
@chapter Audio Decoders
@c man begin AUDIO DECODERS

-A description of some of the currently available audio decoders
-follows.
-
-@section ac3
-
-AC-3 audio decoder.
-
-This decoder implements part of ATSC A/52:2010 and ETSI TS 102 366, as well as
-the undocumented RealAudio 3 (a.k.a. dnet).
-
-@subsection AC-3 Decoder Options
-
-@table @option
-
-@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.
-There are 3 notable scale factor ranges:
-@table @option
-@item drc_scale == 0
-DRC disabled. Produces full range audio.
-@item 0 < drc_scale <= 1
-DRC enabled.  Applies a fraction of the stream DRC value.
-Audio reproduction is between full range and full compression.
-@item drc_scale > 1
-DRC enabled. Applies drc_scale asymmetrically.
-Loud sounds are fully compressed.  Soft sounds are enhanced.
-@end table
-
-@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 synthetizer.
@@ -187,33 +132,11 @@ 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 dvbsub
-
-@subsection Options
-
-@table @option
-@item compute_clut
-@table @option
-@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
@@ -233,62 +156,6 @@ The format for this option is a string containing 16 24-bits hexadecimal
 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
-
-Libzvbi allows libavcodec to decode DVB teletext pages and DVB teletext
-subtitles. Requires the presence of the libzvbi headers and library during
-configuration. You need to explicitly configure the build with
-@code{--enable-libzvbi}.
-
-@subsection Options
-
-@table @option
-@item txt_page
-List of teletext page numbers to decode. You may use the special * string to
-match all pages. Pages that do not match the specified list are dropped.
-Default value is *.
-@item txt_chop_top
-Discards the top teletext line. Default value is 1.
-@item txt_format
-Specifies the format of the decoded subtitles. The teletext decoder is capable
-of decoding the teletext pages to bitmaps or to simple text, you should use
-"bitmap" for teletext pages, because certain graphics and colors cannot be
-expressed in simple text. You might use "text" for teletext based subtitles if
-your application can handle simple text based subtitles. Default value is
-bitmap.
-@item txt_left
-X offset of generated bitmaps, default is 0.
-@item txt_top
-Y offset of generated bitmaps, default is 0.
-@item txt_chop_spaces
-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 charactes.
-Default value is 1.
-@item txt_duration
-Sets the display duration of the decoded teletext pages or subtitles in
-miliseconds. Default value is 30000 which is 30 seconds.
-@item txt_transparent
-Force transparent background of the generated teletext bitmaps. Default value
-is 0 which means an opaque background.
-@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/default.css
+++ b/doc/default.css
@@ -1,7 +1,3 @@
-a.summary-letter {
-    text-decoration: none;
-}
-
 a {
    color: #2D6198;
 }
@@ -17,8 +13,8 @@ a:visited {
 }

 #banner img {
-    margin-bottom: 1px;
-    margin-top: 5px;
+    padding-bottom: 1px;
+    padding-top: 5px;
 }

 #body {
@@ -49,16 +45,11 @@ body {
    text-align: center;
 }

-h1 a, h2 a, h3 a, h4 a {
-    text-decoration: inherit;
-    color: inherit;
-}
-
-h1, h2, h3, h4 {
+h1, h2, h3 {
    padding-left: 0.4em;
    border-radius: 4px;
-    padding-bottom: 0.25em;
-    padding-top: 0.25em;
+    padding-bottom: 0.2em;
+    padding-top: 0.2em;
    border: 1px solid #6A996A;
 }

@@ -72,22 +63,15 @@ h1 {

 h2 {
    color: #313131;
-    font-size: 1.0em;
+    font-size: 0.9em;
    background-color: #ABE3AB;
 }

 h3 {
-    color: #313131;
-    font-size: 0.9em;
-    margin-bottom: -6px;
-    background-color: #BBF3BB;
-}
-
-h4 {
    color: #313131;
    font-size: 0.8em;
    margin-bottom: -8px;
-    background-color: #D1FDD1;
+    background-color: #BBF3BB;
 }

 img {
--- a/doc/demuxers.texi
+++ b/doc/demuxers.texi
@@ -1,7 +1,7 @@
@chapter Demuxers
@c man begin DEMUXERS

-Demuxers are configured elements in FFmpeg that can read the
+Demuxers are configured elements in FFmpeg which allow to read the
 multimedia streams from a particular type of file.

 When you configure your FFmpeg build, all the supported demuxers
@@ -18,12 +18,6 @@ enabled demuxers.

 The description of some of the currently available demuxers follows.

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

 Apple HTTP Live Streaming demuxer.
@@ -35,37 +29,6 @@ the caller can decide which variant streams to actually receive.
 The total bitrate of the variant that the stream belongs to is
 available in a metadata key named "variant_bitrate".

-@section apng
-
-Animated Portable Network Graphics demuxer.
-
-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.
-@item -max_fps @var{int}
-Maximum framerate in frames per second (0 for no limit).
-@item -default_fps @var{int}
-Default framerate in frames per second when none is specified in the file
-(0 meaning as fast as possible).
-@end table
-
-@section asf
-
-Advanced Systems Format demuxer.
-
-This demuxer is used to demux ASF files and MMS network streams.
-
-@table @option
-@item -no_resync_search @var{bool}
-Do not try to resynchronize by looking for a certain optional start code.
-@end table
-
@anchor{concat}
@section concat

@@ -100,7 +63,7 @@ following directive is recognized:
 Path to a file to read; special characters and spaces must be escaped with
 backslash or single quotes.

-All subsequent file-related directives apply to that file.
+All subsequent directives apply to that file.

@item @code{ffconcat version 1.0}
 Identify the script type and version. It also sets the @option{safe} option
@@ -118,63 +81,6 @@ 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.
-
-@item @code{stream}
-Introduce a stream in the virtual file.
-All subsequent stream-related directives apply to the last introduced
-stream.
-Some streams properties must be set in order to allow identifying the
-matching streams in the subfiles.
-If no streams are defined in the script, the streams from the first file are
-copied.
-
-@item @code{exact_stream_id @var{id}}
-Set the id of the stream.
-If this directive is given, the string with the corresponding id in the
-subfiles will be used.
-This is especially useful for MPEG-PS (VOB) files, where the order of the
-streams is not reliable.
-
@end table

@subsection Options
@@ -195,61 +101,6 @@ If set to 0, any file name is accepted.
 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
-
-@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 flv
-
-Adobe Flash Video Format demuxer.
-
-This demuxer is used to demux FLV files and RTMP network streams.
-
-@table @option
-@item -flv_metadata @var{bool}
-Allocate the streams according to the onMetaData array content.
@end table

@section libgme
@@ -265,44 +116,17 @@ track. Track indexes start at 0. The demuxer exports the number of tracks as

 For very large files, the @option{max_size} option may have to be adjusted.

-@section gif
+@section libquvi

-Animated GIF demuxer.
+Play media from Internet services using the quvi project.

-It accepts the following options:
+The demuxer accepts a @option{format} option to request a specific quality. It
+is by default set to @var{best}.

-@table @option
-@item min_delay
-Set the minimum valid delay between frames in hundredths of seconds.
-Range is 0 to 6000. Default value is 2.
+See @url{http://quvi.sourceforge.net/} for more information.

-@item max_gif_delay
-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
-Set the default delay between frames in hundredths of seconds.
-Range is 0 to 6000. Default value is 10.
-
-@item ignore_loop
-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
-over another video:
-@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
-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.
+FFmpeg needs to be built with @code{--enable-libquvi} for this demuxer to be
+enabled.

@section image2

@@ -329,10 +153,6 @@ 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.
@@ -407,8 +227,6 @@ is 5.
 If set to 1, will set frame timestamp to modification time of image file. Note
 that monotonity of timestamps is not provided: images go in the same order as
 without this option. Default value is 0.
-If set to 2, will set frame timestamp to the modification time of the image file in
-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.
@@ -422,86 +240,28 @@ Use @command{ffmpeg} for creating a video from the images in the file
 sequence @file{img-001.jpeg}, @file{img-002.jpeg}, ..., assuming an
 input frame rate of 10 frames per second:
@example
-ffmpeg -framerate 10 -i 'img-%03d.jpeg' out.mkv
+ffmpeg -i 'img-%03d.jpeg' -r 10 out.mkv
@end example

@item
 As above, but start by reading from a file with index 100 in the sequence:
@example
-ffmpeg -framerate 10 -start_number 100 -i 'img-%03d.jpeg' out.mkv
+ffmpeg -start_number 100 -i 'img-%03d.jpeg' -r 10 out.mkv
@end example

@item
 Read images matching the "*.png" glob pattern , that is all the files
 terminating with the ".png" suffix:
@example
-ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv
+ffmpeg -pattern_type glob -i "*.png" -r 10 out.mkv
@end example
@end itemize

-@section mov/mp4/3gp/Quicktme
-
-Quicktime / MP4 demuxer.
-
-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.
-
-@end table
-
-@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 fix_teletext_pts
-Override teletext packet PTS and DTS values with the timestamps calculated
-from the PCR of the first program which the teletext stream is part of and is
-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.
-@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

 Raw video demuxer.

-This demuxer allows one to read raw video data. Since there is no header
+This demuxer allows to read raw video data. Since there is no header
 specifying the assumed video parameters, the user must specify them
 in order to be able to decode the data correctly.

--- a/doc/developer.texi
+++ b/doc/developer.texi
@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8

@settitle Developer Documentation
@titlepage
@@ -12,30 +11,36 @@

@chapter Developers Guide

-@section Notes for external developers
+@section API
+@itemize @bullet
+@item libavcodec is the library containing the codecs (both encoding and
+decoding). Look at @file{doc/examples/decoding_encoding.c} to see how to use
+it.

-This document is mostly useful for internal FFmpeg developers.
-External developers who need to use the API in their application should
-refer to the API doxygen documentation in the public headers, and
-check the examples in @file{doc/examples} and in the source code to
-see how the public API is employed.
+@item libavformat is the library containing the file format handling (mux and
+demux code for several formats). Look at @file{ffplay.c} to use it in a
+player. See @file{doc/examples/muxing.c} to use it to generate audio or video
+streams.

-You can use the FFmpeg libraries in your commercial program, but you
-are encouraged to @emph{publish any patch you make}. In this case the
-best way to proceed is to send your patches to the ffmpeg-devel
-mailing list following the guidelines illustrated in the remainder of
-this document.
+@end itemize

-For more detailed legal information about the use of FFmpeg in
-external programs read the @file{LICENSE} file in the source tree and
-consult @url{https://ffmpeg.org/legal.html}.
+@section Integrating libavcodec or libavformat in your program
+
+You can integrate all the source code of the libraries to link them
+statically to avoid any version problem. All you need is to provide a
+'config.mak' and a 'config.h' in the parent directory. See the defines
+generated by ./configure to understand what is needed.
+
+You can use libavcodec or libavformat in your commercial program, but
+@emph{any patch you make must be published}. The best way to proceed is
+to send your patches to the FFmpeg mailing list.

@section Contributing

-There are 3 ways by which code gets into FFmpeg.
+There are 3 ways by which code gets into ffmpeg.
@itemize @bullet
-@item Submitting patches to the main developer mailing list.
-      See @ref{Submitting patches} for details.
+@item Submitting Patches to the main developer mailing list
+      see @ref{Submitting patches} for details.
@item Directly committing changes to the main tree.
@item Committing changes to a git clone, for example on github.com or
      gitorious.org. And asking us to merge these changes.
@@ -52,22 +57,16 @@ and should try to fix issues their commit causes.
@subsection Code formatting conventions

 There are the following guidelines regarding the indentation in files:
-
@itemize @bullet
@item
 Indent size is 4.
-
@item
 The TAB character is forbidden outside of Makefiles as is any
 form of trailing whitespace. Commits containing either will be
 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'.

@@ -96,7 +95,7 @@ for markup commands, i.e. use @code{@@param} and not @code{\param}.
 * more text ...
 * ...
 */
-typedef struct Foobar @{
+typedef struct Foobar@{
    int var1; /**< var1 description */
    int var2; ///< var2 description
    /** var3 description */
@@ -118,19 +117,15 @@ int myfunc(int my_parameter)

 FFmpeg is programmed in the ISO C90 language with a few additional
 features from ISO C99, namely:
-
@itemize @bullet
@item
 the @samp{inline} keyword;
-
@item
@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 @};})
@end itemize

 These features are supported by all compilers we care about, so we will not
@@ -140,17 +135,13 @@ clarity and performance.
 All code must compile with recent versions of GCC and a number of other
 currently supported compilers. To ensure compatibility, please do not use
 additional C99 features or GCC extensions. Especially watch out for:
-
@itemize @bullet
@item
 mixing statements and declarations;
-
@item
@samp{long long} (use @samp{int64_t} instead);
-
@item
@samp{__attribute__} not protected by @samp{#ifdef __GNUC__} or similar;
-
@item
 GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}).
@end itemize
@@ -159,28 +150,23 @@ GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}).
 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:
-
@itemize @bullet
@item
 For local variables no prefix is required.
-
@item
 For file-scope variables and functions declared as @code{static}, no prefix
 is required.
-
@item
 For variables and functions visible outside of file scope, but only used
 internally by a library, an @code{ff_} prefix should be used,
 e.g. @samp{ff_w64_demuxer}.
-
@item
 For variables and functions visible outside of file scope, used internally
 across multiple libraries, use @code{avpriv_} as prefix, for example,
@samp{avpriv_aac_parse_header}.
-
@item
 Each library has its own prefix for public symbols, in addition to the
 commonly used @code{av_} (@code{avformat_} for libavformat,
@@ -200,12 +186,10 @@ 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.

@subsection Miscellaneous conventions
-
@itemize @bullet
@item
 fprintf and printf are forbidden in libavformat and libavcodec,
 please use av_log() instead.
-
@item
 Casts should be used only when necessary. Unneeded parentheses
 should also be avoided if they don't make the code easier to understand.
@@ -231,7 +215,7 @@ 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)
@@ -242,163 +226,137 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
               )
             )
 (setq c-default-style "ffmpeg")
-@end lisp
+@end example

@section Development Policy

@enumerate
@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
-a gift-style license, the
-@uref{http://opensource.org/licenses/isc-license.txt, ISC} or
-@uref{http://mit-license.org/, MIT} license.
-@uref{http://www.gnu.org/licenses/gpl-2.0.html, GPL 2} including
-an "or any later version" clause is also acceptable, but LGPL is
-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.
-
+   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
+   a gift-style license, the
+   @uref{http://www.isc.org/software/license/, ISC} or
+   @uref{http://mit-license.org/, MIT} license.
+   @uref{http://www.gnu.org/licenses/gpl-2.0.html, GPL 2} including
+   an "or any later version" clause is also acceptable, but LGPL is
+   preferred.
@item
-You must not commit code which breaks FFmpeg! (Meaning unfinished but
-enabled code which breaks compilation or compiles but does not work or
-breaks the regression tests)
-You can commit unfinished stuff (for testing etc), but it must be disabled
-(#ifdef etc) by default so it does not interfere with other developers'
-work.
-
+   You must not commit code which breaks FFmpeg! (Meaning unfinished but
+   enabled code which breaks compilation or compiles but does not work or
+   breaks the regression tests)
+   You can commit unfinished stuff (for testing etc), but it must be disabled
+   (#ifdef etc) by default so it does not interfere with other developers'
+   work.
@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.
-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.
-
+   The commit message should have a short first line in the form of
+   a @samp{topic: short description} as a header, separated by a newline
+   from the body consisting of an explanation of why the change is necessary.
+   If the commit fixes a known bug on the bug tracker, the commit message
+   should include its bug ID. Referring to the issue on the bug tracker does
+   not exempt you from writing an excerpt of the bug in the commit message.
@item
-You do not have to over-test things. If it works for you, and you think it
-should work for others, then commit. If your code has problems
-(portability, triggers compiler bugs, unusual environment etc) they will be
-reported and eventually fixed.
-
+   You do not have to over-test things. If it works for you, and you think it
+   should work for others, then commit. If your code has problems
+   (portability, triggers compiler bugs, unusual environment etc) they will be
+   reported and eventually fixed.
@item
-Do not commit unrelated changes together, split them into self-contained
-pieces. Also do not forget that if part B depends on part A, but A does not
-depend on B, then A can and should be committed first and separate from B.
-Keeping changes well split into self-contained parts makes reviewing and
-understanding them on the commit log mailing list easier. This also helps
-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.
-
+   Do not commit unrelated changes together, split them into self-contained
+   pieces. Also do not forget that if part B depends on part A, but A does not
+   depend on B, then A can and should be committed first and separate from B.
+   Keeping changes well split into self-contained parts makes reviewing and
+   understanding them on the commit log mailing list easier. This also helps
+   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.
@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.
+   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
-maintained by other developers. We usually have a reason for doing things
-the way we do. Send your changes as patches to the ffmpeg-devel mailing
-list, and if the code maintainers say OK, you may commit. This does not
-apply to files you wrote and/or maintain.
-
+   Do not commit changes to the build system (Makefiles, configure script)
+   which change behavior, defaults etc, without asking first. The same
+   applies to compiler warning fixes, trivial looking fixes and to code
+   maintained by other developers. We usually have a reason for doing things
+   the way we do. Send your changes as patches to the ffmpeg-devel mailing
+   list, and if the code maintainers say OK, you may commit. This does not
+   apply to files you wrote and/or maintain.
@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
-if you (re)write something, you can use your own style, even though we would
-prefer if the indentation throughout FFmpeg was consistent (Many projects
-force a given indentation style - we do not.). If you really need to make
-indentation changes (try to avoid this), separate them strictly from real
-changes.
-
-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
+   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
+   if you (re)write something, you can use your own style, even though we would
+   prefer if the indentation throughout FFmpeg was consistent (Many projects
+   force a given indentation style - we do not.). If you really need to make
+   indentation changes (try to avoid this), separate them strictly from real
+   changes.

+   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
@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
+   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:
+   area changed: Short 1 line description

+   details describing what and why and giving references.
@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.
-
+   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.
@item
-When applying patches that have been discussed (at length) on the mailing
-list, reference the thread in the log message.
-
+   When applying patches that have been discussed (at length) on the mailing
+   list, reference the thread in the log message.
@item
-Do NOT commit to code actively maintained by others without permission.
-Send a patch to ffmpeg-devel instead. If no one answers within a reasonable
-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!
-
+    Do NOT commit to code actively maintained by others without permission.
+    Send a patch to ffmpeg-devel instead. If no one answers within a reasonable
+    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!
@item
-Subscribe to the ffmpeg-cvslog mailing list. The diffs of all commits
-are sent there and reviewed by all the other developers. Bugs and possible
-improvements or general questions regarding commits are discussed there. We
-expect you to react if problems with your code are uncovered.
-
+    Subscribe to the ffmpeg-cvslog mailing list. The diffs of all commits
+    are sent there and reviewed by all the other developers. Bugs and possible
+    improvements or general questions regarding commits are discussed there. We
+    expect you to react if problems with your code are uncovered.
@item
-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.
-
+    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.
-
+    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.
-
+    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
-(e.g. addition of a function to the public API or extension of an
-existing data structure).
-Incrementing the third component means a noteworthy binary compatible
-change (e.g. encoder bug fix that matters for the decoder). The third
-component always starts at 100 to distinguish FFmpeg from Libav.
-
+    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
+    (e.g. addition of a function to the public API or extension of an
+    existing data structure).
+    Incrementing the third component means a noteworthy binary compatible
+    change (e.g. encoder bug fix that matters for the decoder). The third
+    component always starts at 100 to distinguish FFmpeg from Libav.
@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.
-Thus the remaining warnings can either be bugs or correct code.
-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.
-
+    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.
+    Thus the remaining warnings can either be bugs or correct code.
+    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.
@item
-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
-finding a new maintainer and also don't forget to update the @file{MAINTAINERS} file.
+    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.
@end enumerate

 We think our rules are not too hard. If you have comments, contact us.
@@ -410,7 +368,7 @@ 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
@@ -433,7 +391,7 @@ 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
@@ -453,51 +411,40 @@ send a reminder by email. Your patch should eventually be dealt with.

@enumerate
@item
-Did you use av_cold for codec initialization and close functions?
-
+    Did you use av_cold for codec initialization and close functions?
@item
-Did you add a long_name under NULL_IF_CONFIG_SMALL to the AVCodec or
-AVInputFormat/AVOutputFormat struct?
-
+    Did you add a long_name under NULL_IF_CONFIG_SMALL to the AVCodec or
+    AVInputFormat/AVOutputFormat struct?
@item
-Did you bump the minor version number (and reset the micro version
-number) in @file{libavcodec/version.h} or @file{libavformat/version.h}?
-
+    Did you bump the minor version number (and reset the micro version
+    number) in @file{libavcodec/version.h} or @file{libavformat/version.h}?
@item
-Did you register it in @file{allcodecs.c} or @file{allformats.c}?
-
+    Did you register it in @file{allcodecs.c} or @file{allformats.c}?
@item
-Did you add the AVCodecID to @file{avcodec.h}?
-When adding new codec IDs, also add an entry to the codec descriptor
-list in @file{libavcodec/codec_desc.c}.
-
+    Did you add the AVCodecID to @file{avcodec.h}?
+    When adding new codec IDs, also add an entry to the codec descriptor
+    list in @file{libavcodec/codec_desc.c}.
@item
-If it has a FourCC, did you add it to @file{libavformat/riff.c},
-even if it is only a decoder?
-
+    If it has a FourCC, did you add it to @file{libavformat/riff.c},
+    even if it is only a decoder?
@item
-Did you add a rule to compile the appropriate files in the Makefile?
-Remember to do this even if you're just adding a format to a file that is
-already being compiled by some other rule, like a raw demuxer.
-
+    Did you add a rule to compile the appropriate files in the Makefile?
+    Remember to do this even if you're just adding a format to a file that is
+    already being compiled by some other rule, like a raw demuxer.
@item
-Did you add an entry to the table of supported formats or codecs in
-@file{doc/general.texi}?
-
+    Did you add an entry to the table of supported formats or codecs in
+    @file{doc/general.texi}?
@item
-Did you add an entry in the Changelog?
-
+    Did you add an entry in the Changelog?
@item
-If it depends on a parser or a library, did you add that dependency in
-configure?
-
+    If it depends on a parser or a library, did you add that dependency in
+    configure?
@item
-Did you @code{git add} the appropriate files before committing?
-
+    Did you @code{git add} the appropriate files before committing?
@item
-Did you make sure it compiles standalone, i.e. with
-@code{configure --disable-everything --enable-decoder=foo}
-(or @code{--enable-demuxer} or whatever your component is)?
+    Did you make sure it compiles standalone, i.e. with
+    @code{configure --disable-everything --enable-decoder=foo}
+    (or @code{--enable-demuxer} or whatever your component is)?
@end enumerate


@@ -505,113 +452,82 @@ Did you make sure it compiles standalone, i.e. with

@enumerate
@item
-Does @code{make fate} pass with the patch applied?
-
+    Does @code{make fate} pass with the patch applied?
@item
-Was the patch generated with git format-patch or send-email?
-
+    Was the patch generated with git format-patch or send-email?
@item
-Did you sign off your patch? (git commit -s)
-See @url{http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob_plain;f=Documentation/SubmittingPatches} for the meaning
-of sign off.
-
+    Did you sign off your patch? (git commit -s)
+    See @url{http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob_plain;f=Documentation/SubmittingPatches} for the meaning
+    of sign off.
@item
-Did you provide a clear git commit log message?
-
+    Did you provide a clear git commit log message?
@item
-Is the patch against latest FFmpeg git master branch?
-
+    Is the patch against latest FFmpeg git master branch?
@item
-Are you subscribed to ffmpeg-devel?
-(the list is subscribers only due to spam)
-
+    Are you subscribed to ffmpeg-devel?
+    (the list is subscribers only due to spam)
@item
-Have you checked that the changes are minimal, so that the same cannot be
-achieved with a smaller patch and/or simpler final code?
-
+    Have you checked that the changes are minimal, so that the same cannot be
+    achieved with a smaller patch and/or simpler final code?
@item
-If the change is to speed critical code, did you benchmark it?
-
+    If the change is to speed critical code, did you benchmark it?
@item
-If you did any benchmarks, did you provide them in the mail?
-
+    If you did any benchmarks, did you provide them in the mail?
@item
-Have you checked that the patch does not introduce buffer overflows or
-other security issues?
-
+    Have you checked that the patch does not introduce buffer overflows or
+    other security issues?
@item
-Did you test your decoder or demuxer against damaged data? If no, see
-tools/trasher, the noise bitstream filter, and
-@uref{http://caca.zoy.org/wiki/zzuf, zzuf}. Your decoder or demuxer
-should not crash, end in a (near) infinite loop, or allocate ridiculous
-amounts of memory when fed damaged data.
-
+    Did you test your decoder or demuxer against damaged data? If no, see
+    tools/trasher, the noise bitstream filter, and
+    @uref{http://caca.zoy.org/wiki/zzuf, zzuf}. Your decoder or demuxer
+    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}.
-
+    Does the patch not mix functional and cosmetic changes?
@item
-Does the patch not mix functional and cosmetic changes?
-
+    Did you add tabs or trailing whitespace to the code? Both are forbidden.
@item
-Did you add tabs or trailing whitespace to the code? Both are forbidden.
-
+    Is the patch attached to the email you send?
@item
-Is the patch attached to the email you send?
-
+    Is the mime type of the patch correct? It should be text/x-diff or
+    text/x-patch or at least text/plain and not application/octet-stream.
@item
-Is the mime type of the patch correct? It should be text/x-diff or
-text/x-patch or at least text/plain and not application/octet-stream.
-
+    If the patch fixes a bug, did you provide a verbose analysis of the bug?
@item
-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 ftp://upload.ffmpeg.org
@item
-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 ftp://upload.ffmpeg.org.
-
+    Did you provide a verbose summary about what the patch does change?
@item
-Did you provide a verbose summary about what the patch does change?
-
+    Did you provide a verbose explanation why it changes things like it does?
@item
-Did you provide a verbose explanation why it changes things like it does?
-
+    Did you provide a verbose summary of the user visible advantages and
+    disadvantages if the patch is applied?
@item
-Did you provide a verbose summary of the user visible advantages and
-disadvantages if the patch is applied?
-
+    Did you provide an example so we can verify the new feature added by the
+    patch easily?
@item
-Did you provide an example so we can verify the new feature added by the
-patch easily?
-
+    If you added a new file, did you insert a license header? It should be
+    taken from FFmpeg, not randomly copied and pasted from somewhere else.
@item
-If you added a new file, did you insert a license header? It should be
-taken from FFmpeg, not randomly copied and pasted from somewhere else.
-
+    You should maintain alphabetical order in alphabetically ordered lists as
+    long as doing so does not break API/ABI compatibility.
@item
-You should maintain alphabetical order in alphabetically ordered lists as
-long as doing so does not break API/ABI compatibility.
-
+    Lines with similar content should be aligned vertically when doing so
+    improves readability.
@item
-Lines with similar content should be aligned vertically when doing so
-improves readability.
-
+    Consider to add a regression test for your code.
@item
-Consider adding a regression test for your code.
-
+    If you added YASM code please check that things still work with --disable-yasm
@item
-If you added YASM code please check that things still work with --disable-yasm.
-
+    Make sure you check the return values of function and return appropriate
+    error codes. Especially memory allocation functions like @code{av_malloc()}
+    are notoriously left unchecked, which is a serious problem.
@item
-Make sure you check the return values of function and return appropriate
-error codes. Especially memory allocation functions like @code{av_malloc()}
-are notoriously left unchecked, which is a serious problem.
-
-@item
-Test your code with valgrind and or Address Sanitizer to ensure it's free
-of leaks, out of array accesses, etc.
+    Test your code with valgrind and or Address Sanitizer to ensure it's free
+    of leaks, out of array accesses, etc.
@end enumerate

@section Patch review process
@@ -640,10 +556,6 @@ 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}
@section Regression tests

@@ -659,14 +571,15 @@ accordingly].
@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.

+
@subsection Visualizing Test Coverage

 The FFmpeg build system allows visualizing the test coverage in an easy
@@ -677,15 +590,12 @@ the following steps:
@item
    Configure to compile with instrumentation enabled:
    @code{configure --toolchain=gcov}.
-
@item
    Run your test case, either manually or via FATE. This can be either
    the full FATE regression suite, or any arbitrary invocation of any
    front-end tool provided by FFmpeg, in any combination.
-
@item
    Run @code{make lcov} to generate coverage data in HTML format.
-
@item
    View @code{lcov/index.html} in your preferred HTML viewer.
@end enumerate
@@ -714,19 +624,18 @@ 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:

@enumerate
@item
-@strong{Major releases} always include the latest and greatest
-features and functionality.
-
+    @strong{Major releases} always include the latest and greatest
+    features and functionality.
@item
-@strong{Point releases} are cut from @strong{release} branches,
-which are named @code{release/X}, with @code{X} being the release
-version number.
+    @strong{Point releases} are cut from @strong{release} branches,
+    which are named @code{release/X}, with @code{X} being the release
+    version number.
@end enumerate

 Note that we promise to our users that shared libraries from any FFmpeg
@@ -747,18 +656,15 @@ inclusion into a point release:

@enumerate
@item
-Fixes a security issue, preferably identified by a @strong{CVE
-number} issued by @url{http://cve.mitre.org/}.
-
+    Fixes a security issue, preferably identified by a @strong{CVE
+    number} issued by @url{http://cve.mitre.org/}.
@item
-Fixes a documented bug in @url{https://trac.ffmpeg.org}.
-
+    Fixes a documented bug in @url{https://trac.ffmpeg.org}.
@item
-Improves the included documentation.
-
+    Improves the included documentation.
@item
-Retains both source code and binary compatibility with previous
-point releases of the same release branch.
+    Retains both source code and binary compatibility with previous
+    point releases of the same release branch.
@end enumerate

 The order for checking the rules is (1 OR 2 OR 3) AND 4.
@@ -770,42 +676,33 @@ The release process involves the following steps:

@enumerate
@item
-Ensure that the @file{RELEASE} file contains the version number for
-the upcoming release.
-
+    Ensure that the @file{RELEASE} file contains the version number for
+    the upcoming release.
@item
-Add the release at @url{https://trac.ffmpeg.org/admin/ticket/versions}.
-
+    Add the release at @url{https://trac.ffmpeg.org/admin/ticket/versions}.
@item
-Announce the intent to do a release to the mailing list.
-
+    Announce the intent to do a release to the mailing list.
@item
-Make sure all relevant security fixes have been backported. See
-@url{https://ffmpeg.org/security.html}.
-
+    Make sure all relevant security fixes have been backported. See
+    @url{https://ffmpeg.org/security.html}.
@item
-Ensure that the FATE regression suite still passes in the release
-branch on at least @strong{i386} and @strong{amd64}
-(cf. @ref{Regression tests}).
-
+    Ensure that the FATE regression suite still passes in the release
+    branch on at least @strong{i386} and @strong{amd64}
+    (cf. @ref{Regression tests}).
@item
-Prepare the release tarballs in @code{bz2} and @code{gz} formats, and
-supplementing files that contain @code{gpg} signatures
-
+    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
-push an annotated tag in the form @code{nX}, with @code{X}
-containing the version number.
-
+    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.
@item
-Propose and send a patch to the @strong{ffmpeg-devel} mailing list
-with a news entry for the website.
-
+    Propose and send a patch to the @strong{ffmpeg-devel} mailing list
+    with a news entry for the website.
@item
-Publish the news entry.
-
+    Publish the news entry.
@item
-Send an announcement to the mailing list.
+    Send announcement to the mailing list.
@end enumerate

@bye
--- a/doc/devices.texi
+++ b/doc/devices.texi
@@ -17,9 +17,5 @@ for programmatic use.

@c man end DEVICE OPTIONS

-@ifclear config-writeonly
@include indevs.texi
-@end ifclear
-@ifclear config-readonly
@include outdevs.texi
-@end ifclear
--- a/doc/doxy-wrapper.sh
+++ b/doc/doxy-wrapper.sh
@@ -1,21 +1,14 @@
 #!/bin/sh

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

-shift 3
+shift 2

-if [ -e "VERSION" ]; then
-    VERSION=`cat "VERSION"`
-else
-    VERSION=`git describe`
-fi
-
-$DOXYGEN - <<EOF
+doxygen - <<EOF
@INCLUDE        = ${DOXYFILE}
 INPUT           = $@
-HTML_TIMESTAMP  = NO
-PROJECT_NUMBER  = $VERSION
-OUTPUT_DIRECTORY = $OUT_DIR
+HTML_HEADER     = ${SRC_PATH}/doc/doxy/header.html
+HTML_FOOTER     = ${SRC_PATH}/doc/doxy/footer.html
+HTML_STYLESHEET = ${SRC_PATH}/doc/doxy/doxy_stylesheet.css
 EOF
--- a/doc/doxy/doxy_stylesheet.css
+++ b/doc/doxy/doxy_stylesheet.css
--- a/doc/doxy/footer.html
+++ b/doc/doxy/footer.html
@@ -0,0 +1,9 @@
+
+ <footer class="footer pagination-right">
+   <span class="label label-info">
+    Generated on $datetime for $projectname by&#160;<a href="http://www.doxygen.org/index.html">doxygen</a> $doxygenversion
+   </span>
+ </footer>
+</div>
+</body>
+</html>
--- a/doc/doxy/header.html
+++ b/doc/doxy/header.html
@@ -0,0 +1,16 @@
+<!DOCTYPE html>
+<html>
+<head>
+<meta http-equiv="Content-Type" content="text/html; charset=UTF-8"/>
+<meta http-equiv="X-UA-Compatible" content="IE=9"/>
+<!--BEGIN PROJECT_NAME--><title>$projectname: $title</title><!--END PROJECT_NAME-->
+<!--BEGIN !PROJECT_NAME--><title>$title</title><!--END !PROJECT_NAME-->
+<link href="$relpath$doxy_stylesheet.css" rel="stylesheet" type="text/css" />
+<!--Header replace -->
+
+</head>
+
+<div class="container">
+
+<!--Header replace -->
+<div class="menu">
--- 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/Makefile
+++ b/doc/examples/Makefile
@@ -11,29 +11,20 @@ CFLAGS += -Wall -g
 CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
 LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)

-EXAMPLES=       avio_dir_cmd                       \
-                avio_reading                       \
-                decoding_encoding                  \
-                demuxing_decoding                  \
-                extract_mvs                        \
+EXAMPLES=       decoding_encoding                  \
+                demuxing                           \
                filtering_video                    \
                filtering_audio                    \
-                http_multiclient                   \
                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
 decoding_encoding: LDLIBS += -lm
 muxing:            LDLIBS += -lm
-resampling_audio:  LDLIBS += -lm

 .phony: all clean-test clean

--- a/doc/examples/README
+++ b/doc/examples/README
@@ -5,19 +5,14 @@ Both following use cases rely on pkg-config and make, thus make sure
 that you have them installed and working on your system.


-Method 1: build the installed examples in a generic read/write user directory
+1) Build the installed examples in a generic read/write user directory

 Copy to a read/write user directory and just use "make", it will link
 to the libraries on your system, assuming the PKG_CONFIG_PATH is
 correctly configured.

-Method 2: build the examples in-tree
+2) Build the examples in-tree

 Assuming you are in the source FFmpeg checkout directory, you need to build
-FFmpeg (no need to make install in any prefix). Then just run "make examples".
-This will build the examples using the FFmpeg build system. You can clean those
-examples using "make examplesclean"
-
-If you want to try the dedicated Makefile examples (to emulate the first
-method), go into doc/examples and run a command such as
-PKG_CONFIG_PATH=pc-uninstalled make.
+FFmpeg (no need to make install in any prefix). Then you can go into the
+doc/examples and run a command such as PKG_CONFIG_PATH=pc-uninstalled make.
--- a/doc/examples/avio_dir_cmd.c
+++ b/doc/examples/avio_dir_cmd.c
@@ -1,180 +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 int del_op(const char *url)
-{
-    int ret = avpriv_io_delete(url);
-    if (ret < 0)
-        av_log(NULL, AV_LOG_ERROR, "Cannot delete '%s': %s.\n", url, av_err2str(ret));
-    return ret;
-}
-
-static int move_op(const char *src, const char *dst)
-{
-    int ret = avpriv_io_move(src, dst);
-    if (ret < 0)
-        av_log(NULL, AV_LOG_ERROR, "Cannot move '%s' into '%s': %s.\n", src, dst, av_err2str(ret));
-    return ret;
-}
-
-
-static void usage(const char *program_name)
-{
-    fprintf(stderr, "usage: %s OPERATION entry1 [entry2]\n"
-            "API example program to show how to manipulate resources "
-            "accessed through AVIOContext.\n"
-            "OPERATIONS:\n"
-            "list      list content of the directory\n"
-            "move      rename content in directory\n"
-            "del       delete content in directory\n",
-            program_name);
-}
-
-int main(int argc, char *argv[])
-{
-    const char *op = NULL;
-    int ret;
-
-    av_log_set_level(AV_LOG_DEBUG);
-
-    if (argc < 2) {
-        usage(argv[0]);
-        return 1;
-    }
-
-    /* register codecs and formats and other lavf/lavc components*/
-    av_register_all();
-    avformat_network_init();
-
-    op = argv[1];
-    if (strcmp(op, "list") == 0) {
-        if (argc < 3) {
-            av_log(NULL, AV_LOG_INFO, "Missing argument for list operation.\n");
-            ret = AVERROR(EINVAL);
-        } else {
-            ret = list_op(argv[2]);
-        }
-    } else if (strcmp(op, "del") == 0) {
-        if (argc < 3) {
-            av_log(NULL, AV_LOG_INFO, "Missing argument for del operation.\n");
-            ret = AVERROR(EINVAL);
-        } else {
-            ret = del_op(argv[2]);
-        }
-    } else if (strcmp(op, "move") == 0) {
-        if (argc < 4) {
-            av_log(NULL, AV_LOG_INFO, "Missing argument for move operation.\n");
-            ret = AVERROR(EINVAL);
-        } else {
-            ret = move_op(argv[2], argv[3]);
-        }
-    } else {
-        av_log(NULL, AV_LOG_INFO, "Invalid operation %s\n", op);
-        ret = AVERROR(EINVAL);
-    }
-
-    avformat_network_deinit();
-
-    return ret < 0 ? 1 : 0;
-}
--- a/doc/examples/avio_reading.c
+++ b/doc/examples/avio_reading.c
@@ -1,134 +0,0 @@
-/*
- * Copyright (c) 2014 Stefano Sabatini
- *
- * 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 AVIOContext API example.
- *
- * Make libavformat demuxer access media content through a custom
- * AVIOContext read callback.
- * @example avio_reading.c
- */
-
-#include <libavcodec/avcodec.h>
-#include <libavformat/avformat.h>
-#include <libavformat/avio.h>
-#include <libavutil/file.h>
-
-struct buffer_data {
-    uint8_t *ptr;
-    size_t size; ///< size left in the buffer
-};
-
-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);
-
-    printf("ptr:%p size:%zu\n", bd->ptr, bd->size);
-
-    /* copy internal buffer data to buf */
-    memcpy(buf, bd->ptr, buf_size);
-    bd->ptr  += buf_size;
-    bd->size -= buf_size;
-
-    return buf_size;
-}
-
-int main(int argc, char *argv[])
-{
-    AVFormatContext *fmt_ctx = NULL;
-    AVIOContext *avio_ctx = NULL;
-    uint8_t *buffer = NULL, *avio_ctx_buffer = NULL;
-    size_t buffer_size, avio_ctx_buffer_size = 4096;
-    char *input_filename = NULL;
-    int ret = 0;
-    struct buffer_data bd = { 0 };
-
-    if (argc != 2) {
-        fprintf(stderr, "usage: %s input_file\n"
-                "API example program to show how to read from a custom buffer "
-                "accessed through AVIOContext.\n", argv[0]);
-        return 1;
-    }
-    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)
-        goto end;
-
-    /* fill opaque structure used by the AVIOContext read callback */
-    bd.ptr  = buffer;
-    bd.size = buffer_size;
-
-    if (!(fmt_ctx = avformat_alloc_context())) {
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-
-    avio_ctx_buffer = av_malloc(avio_ctx_buffer_size);
-    if (!avio_ctx_buffer) {
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-    avio_ctx = avio_alloc_context(avio_ctx_buffer, avio_ctx_buffer_size,
-                                  0, &bd, &read_packet, NULL, NULL);
-    if (!avio_ctx) {
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-    fmt_ctx->pb = avio_ctx;
-
-    ret = avformat_open_input(&fmt_ctx, NULL, NULL, NULL);
-    if (ret < 0) {
-        fprintf(stderr, "Could not open input\n");
-        goto end;
-    }
-
-    ret = avformat_find_stream_info(fmt_ctx, NULL);
-    if (ret < 0) {
-        fprintf(stderr, "Could not find stream information\n");
-        goto end;
-    }
-
-    av_dump_format(fmt_ctx, 0, input_filename, 0);
-
-end:
-    avformat_close_input(&fmt_ctx);
-    /* note: the internal buffer could have changed, and be != avio_ctx_buffer */
-    if (avio_ctx) {
-        av_freep(&avio_ctx->buffer);
-        av_freep(&avio_ctx);
-    }
-    av_file_unmap(buffer, buffer_size);
-
-    if (ret < 0) {
-        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
-        return 1;
-    }
-
-    return 0;
-}
--- a/doc/examples/decoding_encoding.c
+++ b/doc/examples/decoding_encoding.c
@@ -24,10 +24,10 @@
 * @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
+ * @example doc/examples/decoding_encoding.c
 */

 #include <math.h>
@@ -156,7 +156,7 @@ static void audio_encode_example(const char *filename)
    }

    /* frame containing input raw audio */
-    frame = av_frame_alloc();
+    frame = avcodec_alloc_frame();
    if (!frame) {
        fprintf(stderr, "Could not allocate audio frame\n");
        exit(1);
@@ -170,10 +170,6 @@ static void audio_encode_example(const char *filename)
     * 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",
@@ -191,7 +187,7 @@ static void audio_encode_example(const char *filename)
    /* encode a single tone sound */
    t = 0;
    tincr = 2 * M_PI * 440.0 / c->sample_rate;
-    for (i = 0; i < 200; i++) {
+    for(i=0;i<200;i++) {
        av_init_packet(&pkt);
        pkt.data = NULL; // packet data will be allocated by the encoder
        pkt.size = 0;
@@ -211,7 +207,7 @@ static void audio_encode_example(const char *filename)
        }
        if (got_output) {
            fwrite(pkt.data, 1, pkt.size, f);
-            av_packet_unref(&pkt);
+            av_free_packet(&pkt);
        }
    }

@@ -225,13 +221,13 @@ static void audio_encode_example(const char *filename)

        if (got_output) {
            fwrite(pkt.data, 1, pkt.size, f);
-            av_packet_unref(&pkt);
+            av_free_packet(&pkt);
        }
    }
    fclose(f);

    av_freep(&samples);
-    av_frame_free(&frame);
+    avcodec_free_frame(&frame);
    avcodec_close(c);
    av_free(c);
 }
@@ -245,7 +241,7 @@ static void audio_decode_example(const char *outfilename, const char *filename)
    AVCodecContext *c= NULL;
    int len;
    FILE *f, *outfile;
-    uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
+    uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
    AVPacket avpkt;
    AVFrame *decoded_frame = NULL;

@@ -288,15 +284,15 @@ static void audio_decode_example(const char *outfilename, const char *filename)
    avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);

    while (avpkt.size > 0) {
-        int i, ch;
        int got_frame = 0;

        if (!decoded_frame) {
-            if (!(decoded_frame = av_frame_alloc())) {
+            if (!(decoded_frame = avcodec_alloc_frame())) {
                fprintf(stderr, "Could not allocate audio frame\n");
                exit(1);
            }
-        }
+        } else
+            avcodec_get_frame_defaults(decoded_frame);

        len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
        if (len < 0) {
@@ -305,15 +301,10 @@ static void audio_decode_example(const char *outfilename, const char *filename)
        }
        if (got_frame) {
            /* if a frame has been decoded, output it */
-            int data_size = av_get_bytes_per_sample(c->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<decoded_frame->nb_samples; i++)
-                for (ch=0; ch<c->channels; ch++)
-                    fwrite(decoded_frame->data[ch] + data_size*i, 1, data_size, outfile);
+            int data_size = av_samples_get_buffer_size(NULL, c->channels,
+                                                       decoded_frame->nb_samples,
+                                                       c->sample_fmt, 1);
+            fwrite(decoded_frame->data[0], 1, data_size, outfile);
        }
        avpkt.size -= len;
        avpkt.data += len;
@@ -338,7 +329,7 @@ static void audio_decode_example(const char *outfilename, const char *filename)

    avcodec_close(c);
    av_free(c);
-    av_frame_free(&decoded_frame);
+    avcodec_free_frame(&decoded_frame);
 }

 /*
@@ -375,18 +366,12 @@ static void video_encode_example(const char *filename, int codec_id)
    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->time_base= (AVRational){1,25};
+    c->gop_size = 10; /* emit one intra frame every ten frames */
+    c->max_b_frames=1;
    c->pix_fmt = AV_PIX_FMT_YUV420P;

-    if (codec_id == AV_CODEC_ID_H264)
+    if(codec_id == AV_CODEC_ID_H264)
        av_opt_set(c->priv_data, "preset", "slow", 0);

    /* open it */
@@ -401,7 +386,7 @@ static void video_encode_example(const char *filename, int codec_id)
        exit(1);
    }

-    frame = av_frame_alloc();
+    frame = avcodec_alloc_frame();
    if (!frame) {
        fprintf(stderr, "Could not allocate video frame\n");
        exit(1);
@@ -420,7 +405,7 @@ static void video_encode_example(const char *filename, int codec_id)
    }

    /* encode 1 second of video */
-    for (i = 0; i < 25; i++) {
+    for(i=0;i<25;i++) {
        av_init_packet(&pkt);
        pkt.data = NULL;    // packet data will be allocated by the encoder
        pkt.size = 0;
@@ -428,15 +413,15 @@ static void video_encode_example(const char *filename, int codec_id)
        fflush(stdout);
        /* prepare a dummy image */
        /* Y */
-        for (y = 0; y < c->height; y++) {
-            for (x = 0; x < c->width; x++) {
+        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++) {
+        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;
            }
@@ -454,7 +439,7 @@ static void video_encode_example(const char *filename, int codec_id)
        if (got_output) {
            printf("Write frame %3d (size=%5d)\n", i, pkt.size);
            fwrite(pkt.data, 1, pkt.size, f);
-            av_packet_unref(&pkt);
+            av_free_packet(&pkt);
        }
    }

@@ -471,7 +456,7 @@ static void video_encode_example(const char *filename, int codec_id)
        if (got_output) {
            printf("Write frame %3d (size=%5d)\n", i, pkt.size);
            fwrite(pkt.data, 1, pkt.size, f);
-            av_packet_unref(&pkt);
+            av_free_packet(&pkt);
        }
    }

@@ -482,7 +467,7 @@ static void video_encode_example(const char *filename, int codec_id)
    avcodec_close(c);
    av_free(c);
    av_freep(&frame->data[0]);
-    av_frame_free(&frame);
+    avcodec_free_frame(&frame);
    printf("\n");
 }

@@ -496,10 +481,10 @@ static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
    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);
+    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);
 }

@@ -521,7 +506,7 @@ static int decode_write_frame(const char *outfilename, AVCodecContext *avctx,
        /* 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],
-                 frame->width, frame->height, buf);
+                 avctx->width, avctx->height, buf);
        (*frame_count)++;
    }
    if (pkt->data) {
@@ -538,13 +523,13 @@ static void video_decode_example(const char *outfilename, const char *filename)
    int frame_count;
    FILE *f;
    AVFrame *frame;
-    uint8_t inbuf[INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
+    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, AV_INPUT_BUFFER_PADDING_SIZE);
+    memset(inbuf + INBUF_SIZE, 0, FF_INPUT_BUFFER_PADDING_SIZE);

    printf("Decode video file %s to %s\n", filename, outfilename);

@@ -561,8 +546,8 @@ static void video_decode_example(const char *outfilename, const char *filename)
        exit(1);
    }

-    if (codec->capabilities & AV_CODEC_CAP_TRUNCATED)
-        c->flags |= AV_CODEC_FLAG_TRUNCATED; // we do not send complete frames
+    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
@@ -580,14 +565,14 @@ static void video_decode_example(const char *outfilename, const char *filename)
        exit(1);
    }

-    frame = av_frame_alloc();
+    frame = avcodec_alloc_frame();
    if (!frame) {
        fprintf(stderr, "Could not allocate video frame\n");
        exit(1);
    }

    frame_count = 0;
-    for (;;) {
+    for(;;) {
        avpkt.size = fread(inbuf, 1, INBUF_SIZE, f);
        if (avpkt.size == 0)
            break;
@@ -624,7 +609,7 @@ static void video_decode_example(const char *outfilename, const char *filename)

    avcodec_close(c);
    av_free(c);
-    av_frame_free(&frame);
+    avcodec_free_frame(&frame);
    printf("\n");
 }

@@ -641,7 +626,7 @@ int main(int argc, char **argv)
               "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",
+               "output_type must be choosen between 'h264', 'mp2', 'mpg'.\n",
               argv[0]);
        return 1;
    }
@@ -651,7 +636,7 @@ int main(int argc, char **argv)
        video_encode_example("test.h264", AV_CODEC_ID_H264);
    } else if (!strcmp(output_type, "mp2")) {
        audio_encode_example("test.mp2");
-        audio_decode_example("test.pcm", "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");
--- a/doc/examples/demuxing_decoding.c
+++ b/doc/examples/demuxing_decoding.c
@@ -22,11 +22,11 @@

 /**
 * @file
- * Demuxing and decoding example.
+ * libavformat demuxing API use example.
 *
 * Show how to use the libavformat and libavcodec API to demux and
 * decode audio and video data.
- * @example demuxing_decoding.c
+ * @example doc/examples/demuxing.c
 */

 #include <libavutil/imgutils.h>
@@ -36,8 +36,6 @@

 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;
@@ -49,50 +47,29 @@ static uint8_t *video_dst_data[4] = {NULL};
 static int      video_dst_linesize[4];
 static int video_dst_bufsize;

+static uint8_t **audio_dst_data = NULL;
+static int       audio_dst_linesize;
+static int audio_dst_bufsize;
+
 static int video_stream_idx = -1, audio_stream_idx = -1;
 static AVFrame *frame = NULL;
 static AVPacket pkt;
 static int video_frame_count = 0;
 static int audio_frame_count = 0;

-/* Enable or disable frame reference counting. You are not supposed to support
- * both paths in your application but pick the one most appropriate to your
- * needs. Look for the use of refcount in this example to see what are the
- * differences of API usage between them. */
-static int refcount = 0;
-
 static int decode_packet(int *got_frame, int cached)
 {
    int ret = 0;
-    int decoded = pkt.size;
-
-    *got_frame = 0;

    if (pkt.stream_index == video_stream_idx) {
        /* decode video frame */
        ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
        if (ret < 0) {
-            fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret));
+            fprintf(stderr, "Error decoding video frame\n");
            return ret;
        }

        if (*got_frame) {
-
-            if (frame->width != width || frame->height != height ||
-                frame->format != pix_fmt) {
-                /* To handle this change, one could call av_image_alloc again and
-                 * decode the following frames into another rawvideo file. */
-                fprintf(stderr, "Error: Width, height and pixel format have to be "
-                        "constant in a rawvideo file, but the width, height or "
-                        "pixel format of the input video changed:\n"
-                        "old: width = %d, height = %d, format = %s\n"
-                        "new: width = %d, height = %d, format = %s\n",
-                        width, height, av_get_pix_fmt_name(pix_fmt),
-                        frame->width, frame->height,
-                        av_get_pix_fmt_name(frame->format));
-                return -1;
-            }
-
            printf("video_frame%s n:%d coded_n:%d pts:%s\n",
                   cached ? "(cached)" : "",
                   video_frame_count++, frame->coded_picture_number,
@@ -102,7 +79,7 @@ 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);
+                          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);
@@ -111,50 +88,49 @@ static int decode_packet(int *got_frame, int cached)
        /* 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));
+            fprintf(stderr, "Error decoding audio frame\n");
            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);
+            ret = av_samples_alloc(audio_dst_data, &audio_dst_linesize, av_frame_get_channels(frame),
+                                   frame->nb_samples, frame->format, 1);
+            if (ret < 0) {
+                fprintf(stderr, "Could not allocate audio buffer\n");
+                return AVERROR(ENOMEM);
+            }
+
+            /* TODO: extend return code of the av_samples_* functions so that this call is not needed */
+            audio_dst_bufsize =
+                av_samples_get_buffer_size(NULL, av_frame_get_channels(frame),
+                                           frame->nb_samples, frame->format, 1);
+
+            /* copy audio data to destination buffer:
+             * this is required since rawaudio expects non aligned data */
+            av_samples_copy(audio_dst_data, frame->data, 0, 0,
+                            frame->nb_samples, av_frame_get_channels(frame), frame->format);
+
+            /* write to rawaudio file */
+            fwrite(audio_dst_data[0], 1, audio_dst_bufsize, audio_dst_file);
+            av_freep(&audio_dst_data[0]);
        }
    }

-    /* If we use frame reference counting, we own the data and need
-     * to de-reference it when we don't use it anymore */
-    if (*got_frame && refcount)
-        av_frame_unref(frame);
-
-    return decoded;
+    return ret;
 }

 static int open_codec_context(int *stream_idx,
                              AVFormatContext *fmt_ctx, enum AVMediaType type)
 {
-    int ret, stream_index;
+    int ret;
    AVStream *st;
    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) {
@@ -162,8 +138,8 @@ static int open_codec_context(int *stream_idx,
                av_get_media_type_string(type), src_filename);
        return ret;
    } else {
-        stream_index = ret;
-        st = fmt_ctx->streams[stream_index];
+        *stream_idx = ret;
+        st = fmt_ctx->streams[*stream_idx];

        /* find decoder for the stream */
        dec_ctx = st->codec;
@@ -171,17 +147,14 @@ static int open_codec_context(int *stream_idx,
        if (!dec) {
            fprintf(stderr, "Failed to find %s codec\n",
                    av_get_media_type_string(type));
-            return AVERROR(EINVAL);
+            return ret;
        }

-        /* Init the decoders, with or without reference counting */
-        av_dict_set(&opts, "refcounted_frames", refcount ? "1" : "0", 0);
-        if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) {
+        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;
@@ -220,22 +193,15 @@ int main (int argc, char **argv)
 {
    int ret = 0, got_frame;

-    if (argc != 4 && argc != 5) {
-        fprintf(stderr, "usage: %s [-refcount] input_file video_output_file audio_output_file\n"
+    if (argc != 4) {
+        fprintf(stderr, "usage: %s 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\n"
-                "If the -refcount option is specified, the program use the\n"
-                "reference counting frame system which allows keeping a copy of\n"
-                "the data for longer than one decode call.\n"
+                "audio frames to a rawaudio file named audio_output_file.\n"
                "\n", argv[0]);
        exit(1);
    }
-    if (argc == 5 && !strcmp(argv[1], "-refcount")) {
-        refcount = 1;
-        argv++;
-    }
    src_filename = argv[1];
    video_dst_filename = argv[2];
    audio_dst_filename = argv[3];
@@ -267,11 +233,9 @@ 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;
@@ -280,14 +244,25 @@ int main (int argc, char **argv)
    }

    if (open_codec_context(&audio_stream_idx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) {
+        int nb_planes;
+
        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);
+            fprintf(stderr, "Could not open destination file %s\n", video_dst_filename);
            ret = 1;
            goto end;
        }
+
+        nb_planes = av_sample_fmt_is_planar(audio_dec_ctx->sample_fmt) ?
+            audio_dec_ctx->channels : 1;
+        audio_dst_data = av_mallocz(sizeof(uint8_t *) * nb_planes);
+        if (!audio_dst_data) {
+            fprintf(stderr, "Could not allocate audio data buffers\n");
+            ret = AVERROR(ENOMEM);
+            goto end;
+        }
    }

    /* dump input information to stderr */
@@ -299,7 +274,7 @@ int main (int argc, char **argv)
        goto end;
    }

-    frame = av_frame_alloc();
+    frame = avcodec_alloc_frame();
    if (!frame) {
        fprintf(stderr, "Could not allocate frame\n");
        ret = AVERROR(ENOMEM);
@@ -318,15 +293,8 @@ int main (int argc, char **argv)

    /* read frames from the file */
    while (av_read_frame(fmt_ctx, &pkt) >= 0) {
-        AVPacket orig_pkt = pkt;
-        do {
-            ret = decode_packet(&got_frame, 0);
-            if (ret < 0)
-                break;
-            pkt.data += ret;
-            pkt.size -= ret;
-        } while (pkt.size > 0);
-        av_packet_unref(&orig_pkt);
+        decode_packet(&got_frame, 0);
+        av_free_packet(&pkt);
    }

    /* flush cached frames */
@@ -341,43 +309,34 @@ int main (int argc, char **argv)
    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->channels;
        const char *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 = 1;
-        }
-
-        if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
+        if ((ret = get_format_from_sample_fmt(&fmt, audio_dec_ctx->sample_fmt)) < 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, audio_dec_ctx->sample_rate,
+               fmt, audio_dec_ctx->channels, audio_dec_ctx->sample_rate,
               audio_dst_filename);
    }

 end:
-    avcodec_close(video_dec_ctx);
-    avcodec_close(audio_dec_ctx);
+    if (video_dec_ctx)
+        avcodec_close(video_dec_ctx);
+    if (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_frame_free(&frame);
+    av_free(frame);
    av_free(video_dst_data[0]);
+    av_free(audio_dst_data);

    return ret < 0;
 }
--- a/doc/examples/extract_mvs.c
+++ b/doc/examples/extract_mvs.c
@@ -1,185 +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 <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 AVPacket pkt;
-static int video_frame_count = 0;
-
-static int decode_packet(int *got_frame, int cached)
-{
-    int decoded = pkt.size;
-
-    *got_frame = 0;
-
-    if (pkt.stream_index == video_stream_idx) {
-        int ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
-        if (ret < 0) {
-            fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret));
-            return ret;
-        }
-
-        if (*got_frame) {
-            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);
-                }
-            }
-        }
-    }
-
-    return decoded;
-}
-
-static int open_codec_context(int *stream_idx,
-                              AVFormatContext *fmt_ctx, enum AVMediaType type)
-{
-    int ret;
-    AVStream *st;
-    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) {
-        fprintf(stderr, "Could not find %s stream in input file '%s'\n",
-                av_get_media_type_string(type), src_filename);
-        return ret;
-    } else {
-        *stream_idx = ret;
-        st = fmt_ctx->streams[*stream_idx];
-
-        /* find decoder for the stream */
-        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);
-        }
-
-        /* Init the video decoder */
-        av_dict_set(&opts, "flags2", "+export_mvs", 0);
-        if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) {
-            fprintf(stderr, "Failed to open %s codec\n",
-                    av_get_media_type_string(type));
-            return ret;
-        }
-    }
-
-    return 0;
-}
-
-int main(int argc, char **argv)
-{
-    int ret = 0, got_frame;
-
-    if (argc != 2) {
-        fprintf(stderr, "Usage: %s <video>\n", argv[0]);
-        exit(1);
-    }
-    src_filename = argv[1];
-
-    av_register_all();
-
-    if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
-        fprintf(stderr, "Could not open source file %s\n", src_filename);
-        exit(1);
-    }
-
-    if (avformat_find_stream_info(fmt_ctx, NULL) < 0) {
-        fprintf(stderr, "Could not find stream information\n");
-        exit(1);
-    }
-
-    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;
-    }
-
-    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;
-    }
-
-    printf("framenum,source,blockw,blockh,srcx,srcy,dstx,dsty,flags\n");
-
-    /* initialize packet, set data to NULL, let the demuxer fill it */
-    av_init_packet(&pkt);
-    pkt.data = NULL;
-    pkt.size = 0;
-
-    /* read frames from the file */
-    while (av_read_frame(fmt_ctx, &pkt) >= 0) {
-        AVPacket orig_pkt = pkt;
-        do {
-            ret = decode_packet(&got_frame, 0);
-            if (ret < 0)
-                break;
-            pkt.data += ret;
-            pkt.size -= ret;
-        } while (pkt.size > 0);
-        av_packet_unref(&orig_pkt);
-    }
-
-    /* flush cached frames */
-    pkt.data = NULL;
-    pkt.size = 0;
-    do {
-        decode_packet(&got_frame, 1);
-    } while (got_frame);
-
-end:
-    avcodec_close(video_dec_ctx);
-    avformat_close_input(&fmt_ctx);
-    av_frame_free(&frame);
-    return ret < 0;
-}
--- a/doc/examples/filter_audio.c
+++ b/doc/examples/filter_audio.c
@@ -1,365 +0,0 @@
-/*
- * copyright (c) 2013 Andrew Kelley
- *
- * 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
- */
-
-/**
- * @file
- * libavfilter API usage example.
- *
- * @example filter_audio.c
- * This example will generate a sine wave audio,
- * pass it through a simple filter chain, and then compute the MD5 checksum of
- * the output data.
- *
- * The filter chain it uses is:
- * (input) -> abuffer -> volume -> aformat -> abuffersink -> (output)
- *
- * abuffer: This provides the endpoint where you can feed the decoded samples.
- * volume: In this example we hardcode it to 0.90.
- * aformat: This converts the samples to the samplefreq, channel layout,
- *          and sample format required by the audio device.
- * abuffersink: This provides the endpoint where you can read the samples after
- *              they have passed through the filter chain.
- */
-
-#include <inttypes.h>
-#include <math.h>
-#include <stdio.h>
-#include <stdlib.h>
-
-#include "libavutil/channel_layout.h"
-#include "libavutil/md5.h"
-#include "libavutil/mem.h"
-#include "libavutil/opt.h"
-#include "libavutil/samplefmt.h"
-
-#include "libavfilter/avfilter.h"
-#include "libavfilter/buffersink.h"
-#include "libavfilter/buffersrc.h"
-
-#define INPUT_SAMPLERATE     48000
-#define INPUT_FORMAT         AV_SAMPLE_FMT_FLTP
-#define INPUT_CHANNEL_LAYOUT AV_CH_LAYOUT_5POINT0
-
-#define VOLUME_VAL 0.90
-
-static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
-                             AVFilterContext **sink)
-{
-    AVFilterGraph *filter_graph;
-    AVFilterContext *abuffer_ctx;
-    AVFilter        *abuffer;
-    AVFilterContext *volume_ctx;
-    AVFilter        *volume;
-    AVFilterContext *aformat_ctx;
-    AVFilter        *aformat;
-    AVFilterContext *abuffersink_ctx;
-    AVFilter        *abuffersink;
-
-    AVDictionary *options_dict = NULL;
-    uint8_t options_str[1024];
-    uint8_t ch_layout[64];
-
-    int err;
-
-    /* Create a new filtergraph, which will contain all the filters. */
-    filter_graph = avfilter_graph_alloc();
-    if (!filter_graph) {
-        fprintf(stderr, "Unable to create filter graph.\n");
-        return AVERROR(ENOMEM);
-    }
-
-    /* Create the abuffer filter;
-     * it will be used for feeding the data into the graph. */
-    abuffer = avfilter_get_by_name("abuffer");
-    if (!abuffer) {
-        fprintf(stderr, "Could not find the abuffer filter.\n");
-        return AVERROR_FILTER_NOT_FOUND;
-    }
-
-    abuffer_ctx = avfilter_graph_alloc_filter(filter_graph, abuffer, "src");
-    if (!abuffer_ctx) {
-        fprintf(stderr, "Could not allocate the abuffer instance.\n");
-        return AVERROR(ENOMEM);
-    }
-
-    /* Set the filter options through the AVOptions API. */
-    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);
-    av_opt_set_int(abuffer_ctx, "sample_rate",    INPUT_SAMPLERATE,                     AV_OPT_SEARCH_CHILDREN);
-
-    /* Now initialize the filter; we pass NULL options, since we have already
-     * set all the options above. */
-    err = avfilter_init_str(abuffer_ctx, NULL);
-    if (err < 0) {
-        fprintf(stderr, "Could not initialize the abuffer filter.\n");
-        return err;
-    }
-
-    /* Create volume filter. */
-    volume = avfilter_get_by_name("volume");
-    if (!volume) {
-        fprintf(stderr, "Could not find the volume filter.\n");
-        return AVERROR_FILTER_NOT_FOUND;
-    }
-
-    volume_ctx = avfilter_graph_alloc_filter(filter_graph, volume, "volume");
-    if (!volume_ctx) {
-        fprintf(stderr, "Could not allocate the volume instance.\n");
-        return AVERROR(ENOMEM);
-    }
-
-    /* A different way of passing the options is as key/value pairs in a
-     * dictionary. */
-    av_dict_set(&options_dict, "volume", AV_STRINGIFY(VOLUME_VAL), 0);
-    err = avfilter_init_dict(volume_ctx, &options_dict);
-    av_dict_free(&options_dict);
-    if (err < 0) {
-        fprintf(stderr, "Could not initialize the volume filter.\n");
-        return err;
-    }
-
-    /* Create the aformat filter;
-     * it ensures that the output is of the format we want. */
-    aformat = avfilter_get_by_name("aformat");
-    if (!aformat) {
-        fprintf(stderr, "Could not find the aformat filter.\n");
-        return AVERROR_FILTER_NOT_FOUND;
-    }
-
-    aformat_ctx = avfilter_graph_alloc_filter(filter_graph, aformat, "aformat");
-    if (!aformat_ctx) {
-        fprintf(stderr, "Could not allocate the aformat instance.\n");
-        return AVERROR(ENOMEM);
-    }
-
-    /* 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=0x%"PRIx64,
-             av_get_sample_fmt_name(AV_SAMPLE_FMT_S16), 44100,
-             (uint64_t)AV_CH_LAYOUT_STEREO);
-    err = avfilter_init_str(aformat_ctx, options_str);
-    if (err < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Could not initialize the aformat filter.\n");
-        return err;
-    }
-
-    /* Finally create the abuffersink filter;
-     * it will be used to get the filtered data out of the graph. */
-    abuffersink = avfilter_get_by_name("abuffersink");
-    if (!abuffersink) {
-        fprintf(stderr, "Could not find the abuffersink filter.\n");
-        return AVERROR_FILTER_NOT_FOUND;
-    }
-
-    abuffersink_ctx = avfilter_graph_alloc_filter(filter_graph, abuffersink, "sink");
-    if (!abuffersink_ctx) {
-        fprintf(stderr, "Could not allocate the abuffersink instance.\n");
-        return AVERROR(ENOMEM);
-    }
-
-    /* This filter takes no options. */
-    err = avfilter_init_str(abuffersink_ctx, NULL);
-    if (err < 0) {
-        fprintf(stderr, "Could not initialize the abuffersink instance.\n");
-        return err;
-    }
-
-    /* Connect the filters;
-     * in this simple case the filters just form a linear chain. */
-    err = avfilter_link(abuffer_ctx, 0, volume_ctx, 0);
-    if (err >= 0)
-        err = avfilter_link(volume_ctx, 0, aformat_ctx, 0);
-    if (err >= 0)
-        err = avfilter_link(aformat_ctx, 0, abuffersink_ctx, 0);
-    if (err < 0) {
-        fprintf(stderr, "Error connecting filters\n");
-        return err;
-    }
-
-    /* Configure the graph. */
-    err = avfilter_graph_config(filter_graph, NULL);
-    if (err < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Error configuring the filter graph\n");
-        return err;
-    }
-
-    *graph = filter_graph;
-    *src   = abuffer_ctx;
-    *sink  = abuffersink_ctx;
-
-    return 0;
-}
-
-/* Do something useful with the filtered data: this simple
- * example just prints the MD5 checksum of each plane to stdout. */
-static int process_output(struct AVMD5 *md5, AVFrame *frame)
-{
-    int planar     = av_sample_fmt_is_planar(frame->format);
-    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);
-    int i, j;
-
-    for (i = 0; i < planes; i++) {
-        uint8_t checksum[16];
-
-        av_md5_init(md5);
-        av_md5_sum(checksum, frame->extended_data[i], plane_size);
-
-        fprintf(stdout, "plane %d: 0x", i);
-        for (j = 0; j < sizeof(checksum); j++)
-            fprintf(stdout, "%02X", checksum[j]);
-        fprintf(stdout, "\n");
-    }
-    fprintf(stdout, "\n");
-
-    return 0;
-}
-
-/* Construct a frame of audio data to be filtered;
- * this simple example just synthesizes a sine wave. */
-static int get_input(AVFrame *frame, int frame_num)
-{
-    int err, i, j;
-
-#define FRAME_SIZE 1024
-
-    /* Set up the frame properties and allocate the buffer for the data. */
-    frame->sample_rate    = INPUT_SAMPLERATE;
-    frame->format         = INPUT_FORMAT;
-    frame->channel_layout = INPUT_CHANNEL_LAYOUT;
-    frame->nb_samples     = FRAME_SIZE;
-    frame->pts            = frame_num * FRAME_SIZE;
-
-    err = av_frame_get_buffer(frame, 0);
-    if (err < 0)
-        return err;
-
-    /* Fill the data for each channel. */
-    for (i = 0; i < 5; i++) {
-        float *data = (float*)frame->extended_data[i];
-
-        for (j = 0; j < frame->nb_samples; j++)
-            data[j] = sin(2 * M_PI * (frame_num + j) * (i + 1) / FRAME_SIZE);
-    }
-
-    return 0;
-}
-
-int main(int argc, char *argv[])
-{
-    struct AVMD5 *md5;
-    AVFilterGraph *graph;
-    AVFilterContext *src, *sink;
-    AVFrame *frame;
-    uint8_t errstr[1024];
-    float duration;
-    int err, nb_frames, i;
-
-    if (argc < 2) {
-        fprintf(stderr, "Usage: %s <duration>\n", argv[0]);
-        return 1;
-    }
-
-    duration  = atof(argv[1]);
-    nb_frames = duration * INPUT_SAMPLERATE / FRAME_SIZE;
-    if (nb_frames <= 0) {
-        fprintf(stderr, "Invalid duration: %s\n", argv[1]);
-        return 1;
-    }
-
-    avfilter_register_all();
-
-    /* Allocate the frame we will be using to store the data. */
-    frame  = av_frame_alloc();
-    if (!frame) {
-        fprintf(stderr, "Error allocating the frame\n");
-        return 1;
-    }
-
-    md5 = av_md5_alloc();
-    if (!md5) {
-        fprintf(stderr, "Error allocating the MD5 context\n");
-        return 1;
-    }
-
-    /* Set up the filtergraph. */
-    err = init_filter_graph(&graph, &src, &sink);
-    if (err < 0) {
-        fprintf(stderr, "Unable to init filter graph:");
-        goto fail;
-    }
-
-    /* the main filtering loop */
-    for (i = 0; i < nb_frames; i++) {
-        /* get an input frame to be filtered */
-        err = get_input(frame, i);
-        if (err < 0) {
-            fprintf(stderr, "Error generating input frame:");
-            goto fail;
-        }
-
-        /* Send the frame to the input of the filtergraph. */
-        err = av_buffersrc_add_frame(src, frame);
-        if (err < 0) {
-            av_frame_unref(frame);
-            fprintf(stderr, "Error submitting the frame to the filtergraph:");
-            goto fail;
-        }
-
-        /* Get all the filtered output that is available. */
-        while ((err = av_buffersink_get_frame(sink, frame)) >= 0) {
-            /* now do something with our filtered frame */
-            err = process_output(md5, frame);
-            if (err < 0) {
-                fprintf(stderr, "Error processing the filtered frame:");
-                goto fail;
-            }
-            av_frame_unref(frame);
-        }
-
-        if (err == AVERROR(EAGAIN)) {
-            /* Need to feed more frames in. */
-            continue;
-        } else if (err == AVERROR_EOF) {
-            /* Nothing more to do, finish. */
-            break;
-        } else if (err < 0) {
-            /* An error occurred. */
-            fprintf(stderr, "Error filtering the data:");
-            goto fail;
-        }
-    }
-
-    avfilter_graph_free(&graph);
-    av_frame_free(&frame);
-    av_freep(&md5);
-
-    return 0;
-
-fail:
-    av_strerror(err, errstr, sizeof(errstr));
-    fprintf(stderr, "%s\n", errstr);
-    return 1;
-}
--- a/doc/examples/filtering_audio.c
+++ b/doc/examples/filtering_audio.c
@@ -25,7 +25,7 @@
 /**
 * @file
 * API example for audio decoding and filtering
- * @example filtering_audio.c
+ * @example doc/examples/filtering_audio.c
 */

 #include <unistd.h>
@@ -33,12 +33,13 @@
 #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>

-static const char *filter_descr = "aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono";
-static const char *player       = "ffplay -f s16le -ar 8000 -ac 1 -";
+const char *filter_descr = "aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono";
+const char *player       = "ffplay -f s16le -ar 8000 -ac 1 -";

 static AVFormatContext *fmt_ctx;
 static AVCodecContext *dec_ctx;
@@ -84,22 +85,18 @@ static int open_input_file(const char *filename)
 static int init_filters(const char *filters_descr)
 {
    char args[512];
-    int ret = 0;
+    int ret;
    AVFilter *abuffersrc  = avfilter_get_by_name("abuffer");
    AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
    AVFilterInOut *outputs = avfilter_inout_alloc();
    AVFilterInOut *inputs  = avfilter_inout_alloc();
-    static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
-    static const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 };
-    static const int out_sample_rates[] = { 8000, -1 };
+    const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
+    const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 };
+    const int out_sample_rates[] = { 8000, -1 };
    const AVFilterLink *outlink;
    AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;

    filter_graph = avfilter_graph_alloc();
-    if (!outputs || !inputs || !filter_graph) {
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }

    /* buffer audio source: the decoded frames from the decoder will be inserted here. */
    if (!dec_ctx->channel_layout)
@@ -112,7 +109,7 @@ static int init_filters(const char *filters_descr)
                                       args, NULL, filter_graph);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
-        goto end;
+        return ret;
    }

    /* buffer audio sink: to terminate the filter chain. */
@@ -120,63 +117,47 @@ static int init_filters(const char *filters_descr)
                                       NULL, NULL, filter_graph);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
-        goto end;
+        return ret;
    }

    ret = av_opt_set_int_list(buffersink_ctx, "sample_fmts", out_sample_fmts, -1,
                              AV_OPT_SEARCH_CHILDREN);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
-        goto end;
+        return ret;
    }

    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");
-        goto end;
+        return ret;
    }

    ret = av_opt_set_int_list(buffersink_ctx, "sample_rates", out_sample_rates, -1,
                              AV_OPT_SEARCH_CHILDREN);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
-        goto end;
+        return ret;
    }

-    /*
-     * 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.
-     */
+    /* Endpoints for the filter graph. */
    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;
    inputs->next       = NULL;

    if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
-                                        &inputs, &outputs, NULL)) < 0)
-        goto end;
+                                    &inputs, &outputs, NULL)) < 0)
+        return ret;

    if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
-        goto end;
+        return ret;

    /* Print summary of the sink buffer
     * Note: args buffer is reused to store channel layout string */
@@ -187,11 +168,7 @@ static int init_filters(const char *filters_descr)
           (char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
           args);

-end:
-    avfilter_inout_free(&inputs);
-    avfilter_inout_free(&outputs);
-
-    return ret;
+    return 0;
 }

 static void print_frame(const AVFrame *frame)
@@ -211,7 +188,7 @@ static void print_frame(const AVFrame *frame)
 int main(int argc, char **argv)
 {
    int ret;
-    AVPacket packet0, packet;
+    AVPacket packet;
    AVFrame *frame = av_frame_alloc();
    AVFrame *filt_frame = av_frame_alloc();
    int got_frame;
@@ -225,6 +202,7 @@ int main(int argc, char **argv)
        exit(1);
    }

+    avcodec_register_all();
    av_register_all();
    avfilter_register_all();

@@ -234,24 +212,18 @@ int main(int argc, char **argv)
        goto end;

    /* read all packets */
-    packet0.data = NULL;
-    packet.data = NULL;
    while (1) {
-        if (!packet0.data) {
-            if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
-                break;
-            packet0 = packet;
-        }
+        if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
+            break;

        if (packet.stream_index == audio_stream_index) {
+            avcodec_get_frame_defaults(frame);
            got_frame = 0;
            ret = avcodec_decode_audio4(dec_ctx, frame, &got_frame, &packet);
            if (ret < 0) {
                av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n");
                continue;
            }
-            packet.size -= ret;
-            packet.data += ret;

            if (got_frame) {
                /* push the audio data from decoded frame into the filtergraph */
@@ -263,31 +235,29 @@ int main(int argc, char **argv)
                /* pull filtered audio from the filtergraph */
                while (1) {
                    ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
-                    if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
+                    if(ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
                        break;
-                    if (ret < 0)
+                    if(ret < 0)
                        goto end;
                    print_frame(filt_frame);
                    av_frame_unref(filt_frame);
                }
            }
-
-            if (packet.size <= 0)
-                av_packet_unref(&packet0);
-        } else {
-            /* discard non-wanted packets */
-            av_packet_unref(&packet0);
        }
+        av_free_packet(&packet);
    }
 end:
    avfilter_graph_free(&filter_graph);
-    avcodec_close(dec_ctx);
+    if (dec_ctx)
+        avcodec_close(dec_ctx);
    avformat_close_input(&fmt_ctx);
    av_frame_free(&frame);
    av_frame_free(&filt_frame);

    if (ret < 0 && ret != AVERROR_EOF) {
-        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
+        char buf[1024];
+        av_strerror(ret, buf, sizeof(buf));
+        fprintf(stderr, "Error occurred: %s\n", buf);
        exit(1);
    }

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

 #define _XOPEN_SOURCE 600 /* for usleep */
@@ -33,14 +33,11 @@
 #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";
-/* other way:
-   scale=78:24 [scl]; [scl] transpose=cclock // assumes "[in]" and "[out]" to be input output pads respectively
- */
+const char *filter_descr = "scale=78:24";

 static AVFormatContext *fmt_ctx;
 static AVCodecContext *dec_ctx;
@@ -73,7 +70,6 @@ static int open_input_file(const char *filename)
    }
    video_stream_index = ret;
    dec_ctx = fmt_ctx->streams[video_stream_index]->codec;
-    av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);

    /* init the video decoder */
    if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
@@ -87,71 +83,47 @@ static int open_input_file(const char *filename)
 static int init_filters(const char *filters_descr)
 {
    char args[512];
-    int ret = 0;
+    int ret;
    AVFilter *buffersrc  = avfilter_get_by_name("buffer");
    AVFilter *buffersink = avfilter_get_by_name("buffersink");
    AVFilterInOut *outputs = avfilter_inout_alloc();
    AVFilterInOut *inputs  = avfilter_inout_alloc();
-    AVRational time_base = fmt_ctx->streams[video_stream_index]->time_base;
    enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE };
+    AVBufferSinkParams *buffersink_params;

    filter_graph = avfilter_graph_alloc();
-    if (!outputs || !inputs || !filter_graph) {
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }

    /* buffer video source: the decoded frames from the decoder will be inserted here. */
    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",
                                       args, NULL, filter_graph);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n");
-        goto end;
+        return ret;
    }

    /* buffer video sink: to terminate the filter chain. */
+    buffersink_params = av_buffersink_params_alloc();
+    buffersink_params->pixel_fmts = pix_fmts;
    ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
-                                       NULL, NULL, filter_graph);
+                                       NULL, buffersink_params, filter_graph);
+    av_free(buffersink_params);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
-        goto end;
+        return ret;
    }

-    ret = av_opt_set_int_list(buffersink_ctx, "pix_fmts", pix_fmts,
-                              AV_PIX_FMT_NONE, AV_OPT_SEARCH_CHILDREN);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n");
-        goto end;
-    }
-
-    /*
-     * 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.
-     */
+    /* Endpoints for the filter graph. */
    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;
@@ -159,16 +131,11 @@ static int init_filters(const char *filters_descr)

    if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
                                    &inputs, &outputs, NULL)) < 0)
-        goto end;
+        return ret;

    if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
-        goto end;
-
-end:
-    avfilter_inout_free(&inputs);
-    avfilter_inout_free(&outputs);
-
-    return ret;
+        return ret;
+    return 0;
 }

 static void display_frame(const AVFrame *frame, AVRational time_base)
@@ -219,6 +186,7 @@ int main(int argc, char **argv)
        exit(1);
    }

+    avcodec_register_all();
    av_register_all();
    avfilter_register_all();

@@ -233,6 +201,7 @@ int main(int argc, char **argv)
            break;

        if (packet.stream_index == video_stream_index) {
+            avcodec_get_frame_defaults(frame);
            got_frame = 0;
            ret = avcodec_decode_video2(dec_ctx, frame, &got_frame, &packet);
            if (ret < 0) {
@@ -259,20 +228,22 @@ int main(int argc, char **argv)
                    display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base);
                    av_frame_unref(filt_frame);
                }
-                av_frame_unref(frame);
            }
        }
-        av_packet_unref(&packet);
+        av_free_packet(&packet);
    }
 end:
    avfilter_graph_free(&filter_graph);
-    avcodec_close(dec_ctx);
+    if (dec_ctx)
+        avcodec_close(dec_ctx);
    avformat_close_input(&fmt_ctx);
    av_frame_free(&frame);
    av_frame_free(&filt_frame);

    if (ret < 0 && ret != AVERROR_EOF) {
-        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
+        char buf[1024];
+        av_strerror(ret, buf, sizeof(buf));
+        fprintf(stderr, "Error occurred: %s\n", buf);
        exit(1);
    }

--- a/doc/examples/http_multiclient.c
+++ b/doc/examples/http_multiclient.c
@@ -1,155 +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>
-
-void process_client(AVIOContext *client, const char *in_uri)
-{
-    AVIOContext *input = NULL;
-    uint8_t buf[1024];
-    int ret, n, reply_code;
-    char *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;
-    }
-    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);
-}
-
-int main(int argc, char **argv)
-{
-    av_log_set_level(AV_LOG_TRACE);
-    AVDictionary *options = NULL;
-    AVIOContext *client = NULL, *server = NULL;
-    const char *in_uri, *out_uri;
-    int ret, pid;
-    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];
-
-    av_register_all();
-    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/metadata.c
+++ b/doc/examples/metadata.c
@@ -23,7 +23,7 @@
 /**
 * @file
 * Shows how the metadata API can be used in application programs.
- * @example metadata.c
+ * @example doc/examples/metadata.c
 */

 #include <stdio.h>
--- a/doc/examples/muxing.c
+++ b/doc/examples/muxing.c
@@ -24,9 +24,9 @@
 * @file
 * libavformat API example.
 *
- * Output a media file in any supported libavformat format. The default
- * codecs are used.
- * @example muxing.c
+ * Output a media file in any supported libavformat format.
+ * The default codecs are used.
+ * @example doc/examples/muxing.c
 */

 #include <stdlib.h>
@@ -34,67 +34,26 @@
 #include <string.h>
 #include <math.h>

-#include <libavutil/avassert.h>
-#include <libavutil/channel_layout.h>
 #include <libavutil/opt.h>
 #include <libavutil/mathematics.h>
-#include <libavutil/timestamp.h>
 #include <libavformat/avformat.h>
 #include <libswscale/swscale.h>
 #include <libswresample/swresample.h>

-#define STREAM_DURATION   10.0
+/* 5 seconds stream duration */
+#define STREAM_DURATION   200.0
 #define STREAM_FRAME_RATE 25 /* 25 images/s */
+#define STREAM_NB_FRAMES  ((int)(STREAM_DURATION * STREAM_FRAME_RATE))
 #define STREAM_PIX_FMT    AV_PIX_FMT_YUV420P /* default pix_fmt */

-#define SCALE_FLAGS SWS_BICUBIC
-
-// a wrapper around a single output AVStream
-typedef struct OutputStream {
-    AVStream *st;
-
-    /* pts of the next frame that will be generated */
-    int64_t next_pts;
-    int samples_count;
-
-    AVFrame *frame;
-    AVFrame *tmp_frame;
-
-    float t, tincr, tincr2;
-
-    struct SwsContext *sws_ctx;
-    struct SwrContext *swr_ctx;
-} OutputStream;
-
-static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt)
-{
-    AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base;
-
-    printf("pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n",
-           av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base),
-           av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base),
-           av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base),
-           pkt->stream_index);
-}
-
-static int write_frame(AVFormatContext *fmt_ctx, const AVRational *time_base, AVStream *st, AVPacket *pkt)
-{
-    /* 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;
-
-    /* Write the compressed frame to the media file. */
-    log_packet(fmt_ctx, pkt);
-    return av_interleaved_write_frame(fmt_ctx, pkt);
-}
+static int sws_flags = SWS_BICUBIC;

 /* Add an output stream. */
-static void add_stream(OutputStream *ost, AVFormatContext *oc,
-                       AVCodec **codec,
-                       enum AVCodecID codec_id)
+static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
+                            enum AVCodecID codec_id)
 {
    AVCodecContext *c;
-    int i;
+    AVStream *st;

    /* find the encoder */
    *codec = avcodec_find_encoder(codec_id);
@@ -104,38 +63,20 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
        exit(1);
    }

-    ost->st = avformat_new_stream(oc, *codec);
-    if (!ost->st) {
+    st = avformat_new_stream(oc, *codec);
+    if (!st) {
        fprintf(stderr, "Could not allocate stream\n");
        exit(1);
    }
-    ost->st->id = oc->nb_streams-1;
-    c = ost->st->codec;
+    st->id = oc->nb_streams-1;
+    c = st->codec;

    switch ((*codec)->type) {
    case AVMEDIA_TYPE_AUDIO:
-        c->sample_fmt  = (*codec)->sample_fmts ?
-            (*codec)->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
+        c->sample_fmt  = AV_SAMPLE_FMT_FLTP;
        c->bit_rate    = 64000;
        c->sample_rate = 44100;
-        if ((*codec)->supported_samplerates) {
-            c->sample_rate = (*codec)->supported_samplerates[0];
-            for (i = 0; (*codec)->supported_samplerates[i]; i++) {
-                if ((*codec)->supported_samplerates[i] == 44100)
-                    c->sample_rate = 44100;
-            }
-        }
-        c->channels        = av_get_channel_layout_nb_channels(c->channel_layout);
-        c->channel_layout = AV_CH_LAYOUT_STEREO;
-        if ((*codec)->channel_layouts) {
-            c->channel_layout = (*codec)->channel_layouts[0];
-            for (i = 0; (*codec)->channel_layouts[i]; i++) {
-                if ((*codec)->channel_layouts[i] == AV_CH_LAYOUT_STEREO)
-                    c->channel_layout = AV_CH_LAYOUT_STEREO;
-            }
-        }
-        c->channels        = av_get_channel_layout_nb_channels(c->channel_layout);
-        ost->st->time_base = (AVRational){ 1, c->sample_rate };
+        c->channels    = 2;
        break;

    case AVMEDIA_TYPE_VIDEO:
@@ -149,9 +90,8 @@ 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       = ost->st->time_base;
-
+        c->time_base.den = STREAM_FRAME_RATE;
+        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) {
@@ -172,263 +112,238 @@ 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;
+
+    return st;
 }

 /**************************************************************/
 /* audio output */

-static AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt,
-                                  uint64_t channel_layout,
-                                  int sample_rate, int nb_samples)
-{
-    AVFrame *frame = av_frame_alloc();
-    int ret;
+static float t, tincr, tincr2;

-    if (!frame) {
-        fprintf(stderr, "Error allocating an audio frame\n");
-        exit(1);
-    }
+static uint8_t **src_samples_data;
+static int       src_samples_linesize;
+static int       src_nb_samples;

-    frame->format = sample_fmt;
-    frame->channel_layout = channel_layout;
-    frame->sample_rate = sample_rate;
-    frame->nb_samples = nb_samples;
+static int max_dst_nb_samples;
+uint8_t **dst_samples_data;
+int       dst_samples_linesize;
+int       dst_samples_size;

-    if (nb_samples) {
-        ret = av_frame_get_buffer(frame, 0);
-        if (ret < 0) {
-            fprintf(stderr, "Error allocating an audio buffer\n");
-            exit(1);
-        }
-    }
+struct SwrContext *swr_ctx = NULL;

-    return frame;
-}
-
-static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
+static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
 {
    AVCodecContext *c;
-    int nb_samples;
    int ret;
-    AVDictionary *opt = NULL;

-    c = ost->st->codec;
+    c = st->codec;

    /* open it */
-    av_dict_copy(&opt, opt_arg, 0);
-    ret = avcodec_open2(c, codec, &opt);
-    av_dict_free(&opt);
+    ret = avcodec_open2(c, codec, NULL);
    if (ret < 0) {
        fprintf(stderr, "Could not open audio codec: %s\n", av_err2str(ret));
        exit(1);
    }

    /* init signal generator */
-    ost->t     = 0;
-    ost->tincr = 2 * M_PI * 110.0 / c->sample_rate;
+    t     = 0;
+    tincr = 2 * M_PI * 110.0 / c->sample_rate;
    /* increment frequency by 110 Hz per second */
-    ost->tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
+    tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;

-    if (c->codec->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE)
-        nb_samples = 10000;
-    else
-        nb_samples = c->frame_size;
+    src_nb_samples = c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE ?
+        10000 : c->frame_size;

-    ost->frame     = alloc_audio_frame(c->sample_fmt, c->channel_layout,
-                                       c->sample_rate, nb_samples);
-    ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, c->channel_layout,
-                                       c->sample_rate, nb_samples);
+    ret = av_samples_alloc_array_and_samples(&src_samples_data, &src_samples_linesize, c->channels,
+                                             src_nb_samples, c->sample_fmt, 0);
+    if (ret < 0) {
+        fprintf(stderr, "Could not allocate source samples\n");
+        exit(1);
+    }

    /* create resampler context */
-        ost->swr_ctx = swr_alloc();
-        if (!ost->swr_ctx) {
+    if (c->sample_fmt != AV_SAMPLE_FMT_S16) {
+        swr_ctx = swr_alloc();
+        if (!swr_ctx) {
            fprintf(stderr, "Could not allocate resampler context\n");
            exit(1);
        }

        /* set options */
-        av_opt_set_int       (ost->swr_ctx, "in_channel_count",   c->channels,       0);
-        av_opt_set_int       (ost->swr_ctx, "in_sample_rate",     c->sample_rate,    0);
-        av_opt_set_sample_fmt(ost->swr_ctx, "in_sample_fmt",      AV_SAMPLE_FMT_S16, 0);
-        av_opt_set_int       (ost->swr_ctx, "out_channel_count",  c->channels,       0);
-        av_opt_set_int       (ost->swr_ctx, "out_sample_rate",    c->sample_rate,    0);
-        av_opt_set_sample_fmt(ost->swr_ctx, "out_sample_fmt",     c->sample_fmt,     0);
+        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(ost->swr_ctx)) < 0) {
+        if ((ret = swr_init(swr_ctx)) < 0) {
            fprintf(stderr, "Failed to initialize the resampling context\n");
            exit(1);
        }
+    }
+
+    /* compute the number of converted samples: buffering is avoided
+     * ensuring that the output buffer will contain at least all the
+     * converted input samples */
+    max_dst_nb_samples = src_nb_samples;
+    ret = av_samples_alloc_array_and_samples(&dst_samples_data, &dst_samples_linesize, c->channels,
+                                             max_dst_nb_samples, c->sample_fmt, 0);
+    if (ret < 0) {
+        fprintf(stderr, "Could not allocate destination samples\n");
+        exit(1);
+    }
+    dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, max_dst_nb_samples,
+                                                  c->sample_fmt, 0);
 }

 /* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
 * 'nb_channels' channels. */
-static AVFrame *get_audio_frame(OutputStream *ost)
+static void get_audio_frame(int16_t *samples, int frame_size, int nb_channels)
 {
-    AVFrame *frame = ost->tmp_frame;
    int j, i, v;
-    int16_t *q = (int16_t*)frame->data[0];
+    int16_t *q;

-    /* check if we want to generate more frames */
-    if (av_compare_ts(ost->next_pts, ost->st->codec->time_base,
-                      STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
-        return NULL;
-
-    for (j = 0; j <frame->nb_samples; j++) {
-        v = (int)(sin(ost->t) * 10000);
-        for (i = 0; i < ost->st->codec->channels; i++)
+    q = samples;
+    for (j = 0; j < frame_size; j++) {
+        v = (int)(sin(t) * 10000);
+        for (i = 0; i < nb_channels; i++)
            *q++ = v;
-        ost->t     += ost->tincr;
-        ost->tincr += ost->tincr2;
+        t     += tincr;
+        tincr += tincr2;
    }
-
-    frame->pts = ost->next_pts;
-    ost->next_pts  += frame->nb_samples;
-
-    return frame;
 }

-/*
- * encode one audio frame and send it to the muxer
- * return 1 when encoding is finished, 0 otherwise
- */
-static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
+static void write_audio_frame(AVFormatContext *oc, AVStream *st)
 {
    AVCodecContext *c;
    AVPacket pkt = { 0 }; // data and size must be 0;
-    AVFrame *frame;
-    int ret;
-    int got_packet;
-    int dst_nb_samples;
+    AVFrame *frame = avcodec_alloc_frame();
+    int got_packet, ret, dst_nb_samples;

    av_init_packet(&pkt);
-    c = ost->st->codec;
+    c = st->codec;

-    frame = get_audio_frame(ost);
+    get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels);

-    if (frame) {
-        /* convert samples from native format to destination codec format, using the resampler */
-            /* compute destination number of samples */
-            dst_nb_samples = av_rescale_rnd(swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples,
-                                            c->sample_rate, c->sample_rate, AV_ROUND_UP);
-            av_assert0(dst_nb_samples == frame->nb_samples);
-
-        /* 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);
-
-            /* convert to destination format */
-            ret = swr_convert(ost->swr_ctx,
-                              ost->frame->data, dst_nb_samples,
-                              (const uint8_t **)frame->data, frame->nb_samples);
-            if (ret < 0) {
-                fprintf(stderr, "Error while converting\n");
+    /* convert samples from native format to destination codec format, using the resampler */
+    if (swr_ctx) {
+        /* compute destination number of samples */
+        dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + src_nb_samples,
+                                        c->sample_rate, c->sample_rate, AV_ROUND_UP);
+        if (dst_nb_samples > max_dst_nb_samples) {
+            av_free(dst_samples_data[0]);
+            ret = av_samples_alloc(dst_samples_data, &dst_samples_linesize, c->channels,
+                                   dst_nb_samples, c->sample_fmt, 0);
+            if (ret < 0)
                exit(1);
-            }
-            frame = ost->frame;
+            max_dst_nb_samples = dst_nb_samples;
+            dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, dst_nb_samples,
+                                                          c->sample_fmt, 0);
+        }

-        frame->pts = av_rescale_q(ost->samples_count, (AVRational){1, c->sample_rate}, c->time_base);
-        ost->samples_count += dst_nb_samples;
+        /* convert to destination format */
+        ret = swr_convert(swr_ctx,
+                          dst_samples_data, dst_nb_samples,
+                          (const uint8_t **)src_samples_data, src_nb_samples);
+        if (ret < 0) {
+            fprintf(stderr, "Error while converting\n");
+            exit(1);
+        }
+    } else {
+        dst_samples_data[0] = src_samples_data[0];
+        dst_nb_samples = src_nb_samples;
    }

+    frame->nb_samples = dst_nb_samples;
+    avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
+                             dst_samples_data[0], dst_samples_size, 0);
+
    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);
-        }
-    }
+    if (!got_packet)
+        return;

-    return (frame || got_packet) ? 0 : 1;
+    pkt.stream_index = st->index;
+
+    /* Write the compressed frame to the media file. */
+    ret = av_interleaved_write_frame(oc, &pkt);
+    if (ret != 0) {
+        fprintf(stderr, "Error while writing audio frame: %s\n",
+                av_err2str(ret));
+        exit(1);
+    }
+    avcodec_free_frame(&frame);
+}
+
+static void close_audio(AVFormatContext *oc, AVStream *st)
+{
+    avcodec_close(st->codec);
+    av_free(src_samples_data[0]);
+    av_free(dst_samples_data[0]);
 }

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

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

-    picture = av_frame_alloc();
-    if (!picture)
-        return NULL;
-
-    picture->format = pix_fmt;
-    picture->width  = width;
-    picture->height = height;
-
-    /* allocate the buffers for the frame data */
-    ret = av_frame_get_buffer(picture, 32);
-    if (ret < 0) {
-        fprintf(stderr, "Could not allocate frame data.\n");
-        exit(1);
-    }
-
-    return picture;
-}
-
-static void open_video(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
+static void open_video(AVFormatContext *oc, AVCodec *codec, AVStream *st)
 {
    int ret;
-    AVCodecContext *c = ost->st->codec;
-    AVDictionary *opt = NULL;
-
-    av_dict_copy(&opt, opt_arg, 0);
+    AVCodecContext *c = st->codec;

    /* open the codec */
-    ret = avcodec_open2(c, codec, &opt);
-    av_dict_free(&opt);
+    ret = avcodec_open2(c, codec, NULL);
    if (ret < 0) {
        fprintf(stderr, "Could not open video codec: %s\n", av_err2str(ret));
        exit(1);
    }

    /* allocate and init a re-usable frame */
-    ost->frame = alloc_picture(c->pix_fmt, c->width, c->height);
-    if (!ost->frame) {
+    frame = avcodec_alloc_frame();
+    if (!frame) {
        fprintf(stderr, "Could not allocate video frame\n");
        exit(1);
    }

+    /* Allocate the encoded raw picture. */
+    ret = avpicture_alloc(&dst_picture, c->pix_fmt, c->width, c->height);
+    if (ret < 0) {
+        fprintf(stderr, "Could not allocate picture: %s\n", av_err2str(ret));
+        exit(1);
+    }
+
    /* If the output format is not YUV420P, then a temporary YUV420P
     * picture is needed too. It is then converted to the required
     * output format. */
-    ost->tmp_frame = NULL;
    if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
-        ost->tmp_frame = alloc_picture(AV_PIX_FMT_YUV420P, c->width, c->height);
-        if (!ost->tmp_frame) {
-            fprintf(stderr, "Could not allocate temporary picture\n");
+        ret = avpicture_alloc(&src_picture, AV_PIX_FMT_YUV420P, c->width, c->height);
+        if (ret < 0) {
+            fprintf(stderr, "Could not allocate temporary picture: %s\n",
+                    av_err2str(ret));
            exit(1);
        }
    }
+
+    /* copy data and linesize picture pointers to frame */
+    *((AVPicture *)frame) = dst_picture;
 }

 /* Prepare a dummy image. */
-static void fill_yuv_image(AVFrame *pict, int frame_index,
+static void fill_yuv_image(AVPicture *pict, int frame_index,
                           int width, int height)
 {
-    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);
+    int x, y, i;

    i = frame_index;

@@ -446,89 +361,85 @@ static void fill_yuv_image(AVFrame *pict, int frame_index,
    }
 }

-static AVFrame *get_video_frame(OutputStream *ost)
-{
-    AVCodecContext *c = ost->st->codec;
-
-    /* check if we want to generate more frames */
-    if (av_compare_ts(ost->next_pts, ost->st->codec->time_base,
-                      STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
-        return NULL;
-
-    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 */
-        if (!ost->sws_ctx) {
-            ost->sws_ctx = sws_getContext(c->width, c->height,
-                                          AV_PIX_FMT_YUV420P,
-                                          c->width, c->height,
-                                          c->pix_fmt,
-                                          SCALE_FLAGS, NULL, NULL, NULL);
-            if (!ost->sws_ctx) {
-                fprintf(stderr,
-                        "Could not initialize the conversion context\n");
-                exit(1);
-            }
-        }
-        fill_yuv_image(ost->tmp_frame, ost->next_pts, c->width, c->height);
-        sws_scale(ost->sws_ctx,
-                  (const uint8_t * const *)ost->tmp_frame->data, ost->tmp_frame->linesize,
-                  0, c->height, ost->frame->data, ost->frame->linesize);
-    } else {
-        fill_yuv_image(ost->frame, ost->next_pts, c->width, c->height);
-    }
-
-    ost->frame->pts = ost->next_pts++;
-
-    return ost->frame;
-}
-
-/*
- * encode one video frame and send it to the muxer
- * return 1 when encoding is finished, 0 otherwise
- */
-static int write_video_frame(AVFormatContext *oc, OutputStream *ost)
+static void write_video_frame(AVFormatContext *oc, AVStream *st)
 {
    int ret;
-    AVCodecContext *c;
-    AVFrame *frame;
-    int got_packet = 0;
-    AVPacket pkt = { 0 };
+    static struct SwsContext *sws_ctx;
+    AVCodecContext *c = st->codec;

-    c = ost->st->codec;
-
-    frame = get_video_frame(ost);
-
-    av_init_packet(&pkt);
-
-    /* encode the image */
-    ret = avcodec_encode_video2(c, &pkt, frame, &got_packet);
-    if (ret < 0) {
-        fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret));
-        exit(1);
-    }
-
-    if (got_packet) {
-        ret = write_frame(oc, &c->time_base, ost->st, &pkt);
+    if (frame_count >= STREAM_NB_FRAMES) {
+        /* No more frames to compress. The codec has a latency of a few
+         * frames if using B-frames, so we get the last frames by
+         * passing the same picture again. */
    } else {
-        ret = 0;
+        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 */
+            if (!sws_ctx) {
+                sws_ctx = sws_getContext(c->width, c->height, AV_PIX_FMT_YUV420P,
+                                         c->width, c->height, c->pix_fmt,
+                                         sws_flags, NULL, NULL, NULL);
+                if (!sws_ctx) {
+                    fprintf(stderr,
+                            "Could not initialize the conversion context\n");
+                    exit(1);
+                }
+            }
+            fill_yuv_image(&src_picture, frame_count, c->width, c->height);
+            sws_scale(sws_ctx,
+                      (const uint8_t * const *)src_picture.data, src_picture.linesize,
+                      0, c->height, dst_picture.data, dst_picture.linesize);
+        } else {
+            fill_yuv_image(&dst_picture, frame_count, c->width, c->height);
+        }
    }

-    if (ret < 0) {
+    if (oc->oformat->flags & AVFMT_RAWPICTURE) {
+        /* Raw video case - directly store the picture in the packet */
+        AVPacket pkt;
+        av_init_packet(&pkt);
+
+        pkt.flags        |= AV_PKT_FLAG_KEY;
+        pkt.stream_index  = st->index;
+        pkt.data          = dst_picture.data[0];
+        pkt.size          = sizeof(AVPicture);
+
+        ret = av_interleaved_write_frame(oc, &pkt);
+    } else {
+        AVPacket pkt = { 0 };
+        int got_packet;
+        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 size is zero, it means the image was buffered. */
+
+        if (!ret && got_packet && pkt.size) {
+            pkt.stream_index = st->index;
+
+            /* Write the compressed frame to the media file. */
+            ret = av_interleaved_write_frame(oc, &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;
+    frame_count++;
 }

-static void close_stream(AVFormatContext *oc, OutputStream *ost)
+static void close_video(AVFormatContext *oc, AVStream *st)
 {
-    avcodec_close(ost->st->codec);
-    av_frame_free(&ost->frame);
-    av_frame_free(&ost->tmp_frame);
-    sws_freeContext(ost->sws_ctx);
-    swr_free(&ost->swr_ctx);
+    avcodec_close(st->codec);
+    av_free(src_picture.data[0]);
+    av_free(dst_picture.data[0]);
+    av_free(frame);
 }

 /**************************************************************/
@@ -536,20 +447,18 @@ static void close_stream(AVFormatContext *oc, OutputStream *ost)

 int main(int argc, char **argv)
 {
-    OutputStream video_st = { 0 }, audio_st = { 0 };
    const char *filename;
    AVOutputFormat *fmt;
    AVFormatContext *oc;
+    AVStream *audio_st, *video_st;
    AVCodec *audio_codec, *video_codec;
+    double audio_time, video_time;
    int ret;
-    int have_video = 0, have_audio = 0;
-    int encode_video = 0, encode_audio = 0;
-    AVDictionary *opt = NULL;

    /* Initialize libavcodec, and register all codecs and formats. */
    av_register_all();

-    if (argc < 2) {
+    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"
@@ -561,9 +470,6 @@ int main(int argc, char **argv)
    }

    filename = argv[1];
-    if (argc > 3 && !strcmp(argv[2], "-flags")) {
-        av_dict_set(&opt, argv[2]+1, argv[3], 0);
-    }

    /* allocate the output media context */
    avformat_alloc_output_context2(&oc, NULL, NULL, filename);
@@ -571,31 +477,29 @@ int main(int argc, char **argv)
        printf("Could not deduce output format from file extension: using MPEG.\n");
        avformat_alloc_output_context2(&oc, NULL, "mpeg", filename);
    }
-    if (!oc)
+    if (!oc) {
        return 1;
-
+    }
    fmt = oc->oformat;

    /* Add the audio and video streams using the default format codecs
     * and initialize the codecs. */
+    video_st = NULL;
+    audio_st = NULL;
+
    if (fmt->video_codec != AV_CODEC_ID_NONE) {
-        add_stream(&video_st, oc, &video_codec, fmt->video_codec);
-        have_video = 1;
-        encode_video = 1;
+        video_st = add_stream(oc, &video_codec, fmt->video_codec);
    }
    if (fmt->audio_codec != AV_CODEC_ID_NONE) {
-        add_stream(&audio_st, oc, &audio_codec, fmt->audio_codec);
-        have_audio = 1;
-        encode_audio = 1;
+        audio_st = add_stream(oc, &audio_codec, fmt->audio_codec);
    }

    /* 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);
-
-    if (have_audio)
-        open_audio(oc, audio_codec, &audio_st, opt);
+    if (video_st)
+        open_video(oc, video_codec, video_st);
+    if (audio_st)
+        open_audio(oc, audio_codec, audio_st);

    av_dump_format(oc, 0, filename, 1);

@@ -610,21 +514,30 @@ 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));
        return 1;
    }

-    while (encode_video || encode_audio) {
-        /* select the stream to encode */
-        if (encode_video &&
-            (!encode_audio || av_compare_ts(video_st.next_pts, video_st.st->codec->time_base,
-                                            audio_st.next_pts, audio_st.st->codec->time_base) <= 0)) {
-            encode_video = !write_video_frame(oc, &video_st);
+    if (frame)
+        frame->pts = 0;
+    for (;;) {
+        /* Compute current audio and video time. */
+        audio_time = audio_st ? audio_st->pts.val * av_q2d(audio_st->time_base) : 0.0;
+        video_time = video_st ? video_st->pts.val * av_q2d(video_st->time_base) : 0.0;
+
+        if ((!audio_st || audio_time >= STREAM_DURATION) &&
+            (!video_st || video_time >= STREAM_DURATION))
+            break;
+
+        /* write interleaved audio and video frames */
+        if (!video_st || (video_st && audio_st && audio_time < video_time)) {
+            write_audio_frame(oc, audio_st);
        } else {
-            encode_audio = !write_audio_frame(oc, &audio_st);
+            write_video_frame(oc, video_st);
+            frame->pts += av_rescale_q(1, video_st->codec->time_base, video_st->time_base);
        }
    }

@@ -635,14 +548,14 @@ int main(int argc, char **argv)
    av_write_trailer(oc);

    /* Close each codec. */
-    if (have_video)
-        close_stream(oc, &video_st);
-    if (have_audio)
-        close_stream(oc, &audio_st);
+    if (video_st)
+        close_video(oc, video_st);
+    if (audio_st)
+        close_audio(oc, audio_st);

    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
@@ -1,487 +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 VA-API video surfaces.
- */
-
-#include "config.h"
-
-#include <stdio.h>
-
-#include <mfx/mfxvideo.h>
-
-#include <va/va.h>
-#include <va/va_x11.h>
-#include <X11/Xlib.h>
-
-#include "libavformat/avformat.h"
-#include "libavformat/avio.h"
-
-#include "libavcodec/avcodec.h"
-#include "libavcodec/qsv.h"
-
-#include "libavutil/error.h"
-#include "libavutil/mem.h"
-
-typedef struct DecodeContext {
-    mfxSession mfx_session;
-    VADisplay va_dpy;
-
-    VASurfaceID *surfaces;
-    mfxMemId    *surface_ids;
-    int         *surface_used;
-    int       nb_surfaces;
-
-    mfxFrameInfo frame_info;
-} DecodeContext;
-
-static mfxStatus frame_alloc(mfxHDL pthis, mfxFrameAllocRequest *req,
-                             mfxFrameAllocResponse *resp)
-{
-    DecodeContext *decode = pthis;
-    int err, i;
-
-    if (decode->surfaces) {
-        fprintf(stderr, "Multiple allocation requests.\n");
-        return MFX_ERR_MEMORY_ALLOC;
-    }
-    if (!(req->Type & MFX_MEMTYPE_VIDEO_MEMORY_DECODER_TARGET)) {
-        fprintf(stderr, "Unsupported surface type: %d\n", req->Type);
-        return MFX_ERR_UNSUPPORTED;
-    }
-    if (req->Info.BitDepthLuma != 8 || req->Info.BitDepthChroma != 8 ||
-        req->Info.Shift || req->Info.FourCC != MFX_FOURCC_NV12 ||
-        req->Info.ChromaFormat != MFX_CHROMAFORMAT_YUV420) {
-        fprintf(stderr, "Unsupported surface properties.\n");
-        return MFX_ERR_UNSUPPORTED;
-    }
-
-    decode->surfaces     = av_malloc_array (req->NumFrameSuggested, sizeof(*decode->surfaces));
-    decode->surface_ids  = av_malloc_array (req->NumFrameSuggested, sizeof(*decode->surface_ids));
-    decode->surface_used = av_mallocz_array(req->NumFrameSuggested, sizeof(*decode->surface_used));
-    if (!decode->surfaces || !decode->surface_ids || !decode->surface_used)
-        goto fail;
-
-    err = vaCreateSurfaces(decode->va_dpy, VA_RT_FORMAT_YUV420,
-                           req->Info.Width, req->Info.Height,
-                           decode->surfaces, req->NumFrameSuggested,
-                           NULL, 0);
-    if (err != VA_STATUS_SUCCESS) {
-        fprintf(stderr, "Error allocating VA surfaces\n");
-        goto fail;
-    }
-    decode->nb_surfaces = req->NumFrameSuggested;
-
-    for (i = 0; i < decode->nb_surfaces; i++)
-        decode->surface_ids[i] = &decode->surfaces[i];
-
-    resp->mids           = decode->surface_ids;
-    resp->NumFrameActual = decode->nb_surfaces;
-
-    decode->frame_info = req->Info;
-
-    return MFX_ERR_NONE;
-fail:
-    av_freep(&decode->surfaces);
-    av_freep(&decode->surface_ids);
-    av_freep(&decode->surface_used);
-
-    return MFX_ERR_MEMORY_ALLOC;
-}
-
-static mfxStatus frame_free(mfxHDL pthis, mfxFrameAllocResponse *resp)
-{
-    return MFX_ERR_NONE;
-}
-
-static mfxStatus frame_lock(mfxHDL pthis, mfxMemId mid, mfxFrameData *ptr)
-{
-    return MFX_ERR_UNSUPPORTED;
-}
-
-static mfxStatus frame_unlock(mfxHDL pthis, mfxMemId mid, mfxFrameData *ptr)
-{
-    return MFX_ERR_UNSUPPORTED;
-}
-
-static mfxStatus frame_get_hdl(mfxHDL pthis, mfxMemId mid, mfxHDL *hdl)
-{
-    *hdl = mid;
-    return MFX_ERR_NONE;
-}
-
-static void free_surfaces(DecodeContext *decode)
-{
-    if (decode->surfaces)
-        vaDestroySurfaces(decode->va_dpy, decode->surfaces, decode->nb_surfaces);
-    av_freep(&decode->surfaces);
-    av_freep(&decode->surface_ids);
-    av_freep(&decode->surface_used);
-    decode->nb_surfaces = 0;
-}
-
-static void free_buffer(void *opaque, uint8_t *data)
-{
-    int *used = opaque;
-    *used = 0;
-    av_freep(&data);
-}
-
-static int get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
-{
-    DecodeContext *decode = avctx->opaque;
-
-    mfxFrameSurface1 *surf;
-    AVBufferRef *surf_buf;
-    int idx;
-
-    for (idx = 0; idx < decode->nb_surfaces; idx++) {
-        if (!decode->surface_used[idx])
-            break;
-    }
-    if (idx == decode->nb_surfaces) {
-        fprintf(stderr, "No free surfaces\n");
-        return AVERROR(ENOMEM);
-    }
-
-    surf = av_mallocz(sizeof(*surf));
-    if (!surf)
-        return AVERROR(ENOMEM);
-    surf_buf = av_buffer_create((uint8_t*)surf, sizeof(*surf), free_buffer,
-                                &decode->surface_used[idx], AV_BUFFER_FLAG_READONLY);
-    if (!surf_buf) {
-        av_freep(&surf);
-        return AVERROR(ENOMEM);
-    }
-
-    surf->Info       = decode->frame_info;
-    surf->Data.MemId = &decode->surfaces[idx];
-
-    frame->buf[0]  = surf_buf;
-    frame->data[3] = (uint8_t*)surf;
-
-    decode->surface_used[idx] = 1;
-
-    return 0;
-}
-
-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) {
-            if (!avctx->hwaccel_context) {
-                DecodeContext *decode = avctx->opaque;
-                AVQSVContext *qsv = av_qsv_alloc_context();
-                if (!qsv)
-                    return AV_PIX_FMT_NONE;
-
-                qsv->session   = decode->mfx_session;
-                qsv->iopattern = MFX_IOPATTERN_OUT_VIDEO_MEMORY;
-
-                avctx->hwaccel_context = 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(DecodeContext *decode, AVCodecContext *decoder_ctx,
-                         AVFrame *frame, AVPacket *pkt,
-                         AVIOContext *output_ctx)
-{
-    int ret = 0;
-    int got_frame = 1;
-
-    while (pkt->size > 0 || (!pkt->data && got_frame)) {
-        ret = avcodec_decode_video2(decoder_ctx, frame, &got_frame, pkt);
-        if (ret < 0) {
-            fprintf(stderr, "Error during decoding\n");
-            return ret;
-        }
-
-        pkt->data += ret;
-        pkt->size -= 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. */
-        if (got_frame) {
-            mfxFrameSurface1 *surf = (mfxFrameSurface1*)frame->data[3];
-            VASurfaceID    surface = *(VASurfaceID*)surf->Data.MemId;
-
-            VAImageFormat img_fmt = {
-                .fourcc         = VA_FOURCC_NV12,
-                .byte_order     = VA_LSB_FIRST,
-                .bits_per_pixel = 8,
-                .depth          = 8,
-            };
-
-            VAImage img;
-
-            VAStatus err;
-            uint8_t *data;
-            int i, j;
-
-            img.buf      = VA_INVALID_ID;
-            img.image_id = VA_INVALID_ID;
-
-            err = vaCreateImage(decode->va_dpy, &img_fmt,
-                                frame->width, frame->height, &img);
-            if (err != VA_STATUS_SUCCESS) {
-                fprintf(stderr, "Error creating an image: %s\n",
-                        vaErrorStr(err));
-                ret = AVERROR_UNKNOWN;
-                goto fail;
-            }
-
-            err = vaGetImage(decode->va_dpy, surface, 0, 0,
-                             frame->width, frame->height,
-                             img.image_id);
-            if (err != VA_STATUS_SUCCESS) {
-                fprintf(stderr, "Error getting an image: %s\n",
-                        vaErrorStr(err));
-                ret = AVERROR_UNKNOWN;
-                goto fail;
-            }
-
-            err = vaMapBuffer(decode->va_dpy, img.buf, (void**)&data);
-            if (err != VA_STATUS_SUCCESS) {
-                fprintf(stderr, "Error mapping the image buffer: %s\n",
-                        vaErrorStr(err));
-                ret = AVERROR_UNKNOWN;
-                goto fail;
-            }
-
-            for (i = 0; i < img.num_planes; i++)
-                for (j = 0; j < (img.height >> (i > 0)); j++)
-                    avio_write(output_ctx, data + img.offsets[i] + j * img.pitches[i], img.width);
-
-fail:
-            if (img.buf != VA_INVALID_ID)
-                vaUnmapBuffer(decode->va_dpy, img.buf);
-            if (img.image_id != VA_INVALID_ID)
-                vaDestroyImage(decode->va_dpy, img.image_id);
-            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 = { 0 };
-    AVFrame *frame = NULL;
-
-    DecodeContext decode = { NULL };
-
-    Display *dpy = NULL;
-    int va_ver_major, va_ver_minor;
-
-    mfxIMPL mfx_impl = MFX_IMPL_AUTO_ANY;
-    mfxVersion mfx_ver = { { 1, 1 } };
-
-    mfxFrameAllocator frame_allocator = {
-        .pthis = &decode,
-        .Alloc = frame_alloc,
-        .Lock  = frame_lock,
-        .Unlock = frame_unlock,
-        .GetHDL = frame_get_hdl,
-        .Free   = frame_free,
-    };
-
-    AVIOContext *output_ctx = NULL;
-
-    int ret, i, err;
-
-    av_register_all();
-
-    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->codec->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;
-    }
-
-    /* initialize VA-API */
-    dpy = XOpenDisplay(NULL);
-    if (!dpy) {
-        fprintf(stderr, "Cannot open the X display\n");
-        goto finish;
-    }
-    decode.va_dpy = vaGetDisplay(dpy);
-    if (!decode.va_dpy) {
-        fprintf(stderr, "Cannot open the VA display\n");
-        goto finish;
-    }
-
-    err = vaInitialize(decode.va_dpy, &va_ver_major, &va_ver_minor);
-    if (err != VA_STATUS_SUCCESS) {
-        fprintf(stderr, "Cannot initialize VA: %s\n", vaErrorStr(err));
-        goto finish;
-    }
-    fprintf(stderr, "Initialized VA v%d.%d\n", va_ver_major, va_ver_minor);
-
-    /* initialize an MFX session */
-    err = MFXInit(mfx_impl, &mfx_ver, &decode.mfx_session);
-    if (err != MFX_ERR_NONE) {
-        fprintf(stderr, "Error initializing an MFX session\n");
-        goto finish;
-    }
-
-    MFXVideoCORE_SetHandle(decode.mfx_session, MFX_HANDLE_VA_DISPLAY, decode.va_dpy);
-    MFXVideoCORE_SetFrameAllocator(decode.mfx_session, &frame_allocator);
-
-    /* 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->codec->extradata_size) {
-        decoder_ctx->extradata = av_mallocz(video_st->codec->extradata_size +
-                                            AV_INPUT_BUFFER_PADDING_SIZE);
-        if (!decoder_ctx->extradata) {
-            ret = AVERROR(ENOMEM);
-            goto finish;
-        }
-        memcpy(decoder_ctx->extradata, video_st->codec->extradata,
-               video_st->codec->extradata_size);
-        decoder_ctx->extradata_size = video_st->codec->extradata_size;
-    }
-    decoder_ctx->refcounted_frames = 1;
-
-    decoder_ctx->opaque      = &decode;
-    decoder_ctx->get_buffer2 = get_buffer;
-    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();
-    if (!frame) {
-        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(&decode, decoder_ctx, frame, &pkt, output_ctx);
-
-        av_packet_unref(&pkt);
-    }
-
-    /* flush the decoder */
-    pkt.data = NULL;
-    pkt.size = 0;
-    ret = decode_packet(&decode, decoder_ctx, frame, &pkt, 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);
-
-    if (decoder_ctx)
-        av_freep(&decoder_ctx->hwaccel_context);
-    avcodec_free_context(&decoder_ctx);
-
-    free_surfaces(&decode);
-
-    if (decode.mfx_session)
-        MFXClose(decode.mfx_session);
-    if (decode.va_dpy)
-        vaTerminate(decode.va_dpy);
-    if (dpy)
-        XCloseDisplay(dpy);
-
-    avio_close(output_ctx);
-
-    return ret;
-}
--- a/doc/examples/remuxing.c
+++ b/doc/examples/remuxing.c
@@ -1,165 +0,0 @@
-/*
- * Copyright (c) 2013 Stefano Sabatini
- *
- * 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/libavcodec demuxing and muxing API example.
- *
- * Remux streams from one container format to another.
- * @example remuxing.c
- */
-
-#include <libavutil/timestamp.h>
-#include <libavformat/avformat.h>
-
-static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt, const char *tag)
-{
-    AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base;
-
-    printf("%s: pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n",
-           tag,
-           av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base),
-           av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base),
-           av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base),
-           pkt->stream_index);
-}
-
-int main(int argc, char **argv)
-{
-    AVOutputFormat *ofmt = NULL;
-    AVFormatContext *ifmt_ctx = NULL, *ofmt_ctx = NULL;
-    AVPacket pkt;
-    const char *in_filename, *out_filename;
-    int ret, i;
-
-    if (argc < 3) {
-        printf("usage: %s input output\n"
-               "API example program to remux a media file with libavformat and libavcodec.\n"
-               "The output format is guessed according to the file extension.\n"
-               "\n", argv[0]);
-        return 1;
-    }
-
-    in_filename  = argv[1];
-    out_filename = argv[2];
-
-    av_register_all();
-
-    if ((ret = avformat_open_input(&ifmt_ctx, in_filename, 0, 0)) < 0) {
-        fprintf(stderr, "Could not open input file '%s'", in_filename);
-        goto end;
-    }
-
-    if ((ret = avformat_find_stream_info(ifmt_ctx, 0)) < 0) {
-        fprintf(stderr, "Failed to retrieve input stream information");
-        goto end;
-    }
-
-    av_dump_format(ifmt_ctx, 0, in_filename, 0);
-
-    avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, out_filename);
-    if (!ofmt_ctx) {
-        fprintf(stderr, "Could not create output context\n");
-        ret = AVERROR_UNKNOWN;
-        goto end;
-    }
-
-    ofmt = ofmt_ctx->oformat;
-
-    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
-        AVStream *in_stream = ifmt_ctx->streams[i];
-        AVStream *out_stream = avformat_new_stream(ofmt_ctx, in_stream->codec->codec);
-        if (!out_stream) {
-            fprintf(stderr, "Failed allocating output stream\n");
-            ret = AVERROR_UNKNOWN;
-            goto end;
-        }
-
-        ret = avcodec_copy_context(out_stream->codec, in_stream->codec);
-        if (ret < 0) {
-            fprintf(stderr, "Failed to copy context from input to output stream codec context\n");
-            goto end;
-        }
-        out_stream->codec->codec_tag = 0;
-        if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
-            out_stream->codec->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
-    }
-    av_dump_format(ofmt_ctx, 0, out_filename, 1);
-
-    if (!(ofmt->flags & AVFMT_NOFILE)) {
-        ret = avio_open(&ofmt_ctx->pb, out_filename, AVIO_FLAG_WRITE);
-        if (ret < 0) {
-            fprintf(stderr, "Could not open output file '%s'", out_filename);
-            goto end;
-        }
-    }
-
-    ret = avformat_write_header(ofmt_ctx, NULL);
-    if (ret < 0) {
-        fprintf(stderr, "Error occurred when opening output file\n");
-        goto end;
-    }
-
-    while (1) {
-        AVStream *in_stream, *out_stream;
-
-        ret = av_read_frame(ifmt_ctx, &pkt);
-        if (ret < 0)
-            break;
-
-        in_stream  = ifmt_ctx->streams[pkt.stream_index];
-        out_stream = ofmt_ctx->streams[pkt.stream_index];
-
-        log_packet(ifmt_ctx, &pkt, "in");
-
-        /* copy packet */
-        pkt.pts = av_rescale_q_rnd(pkt.pts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
-        pkt.dts = av_rescale_q_rnd(pkt.dts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
-        pkt.duration = av_rescale_q(pkt.duration, in_stream->time_base, out_stream->time_base);
-        pkt.pos = -1;
-        log_packet(ofmt_ctx, &pkt, "out");
-
-        ret = av_interleaved_write_frame(ofmt_ctx, &pkt);
-        if (ret < 0) {
-            fprintf(stderr, "Error muxing packet\n");
-            break;
-        }
-        av_packet_unref(&pkt);
-    }
-
-    av_write_trailer(ofmt_ctx);
-end:
-
-    avformat_close_input(&ifmt_ctx);
-
-    /* close output */
-    if (ofmt_ctx && !(ofmt->flags & AVFMT_NOFILE))
-        avio_closep(&ofmt_ctx->pb);
-    avformat_free_context(ofmt_ctx);
-
-    if (ret < 0 && ret != AVERROR_EOF) {
-        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
-        return 1;
-    }
-
-    return 0;
-}
--- a/doc/examples/resampling_audio.c
+++ b/doc/examples/resampling_audio.c
@@ -21,7 +21,7 @@
 */

 /**
- * @example resampling_audio.c
+ * @example doc/examples/resampling_audio.c
 * libswresample API use example.
 */

@@ -62,7 +62,7 @@ static int get_format_from_sample_fmt(const char **fmt,
 /**
 * Fill dst buffer with nb_samples, generated starting from t.
 */
-static void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t)
+void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t)
 {
    int i, j;
    double tincr = 1.0 / sample_rate, *dstp = dst;
@@ -168,7 +168,7 @@ int main(int argc, char **argv)
        dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) +
                                        src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
        if (dst_nb_samples > max_dst_nb_samples) {
-            av_freep(&dst_data[0]);
+            av_free(dst_data[0]);
            ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
                                   dst_nb_samples, dst_sample_fmt, 1);
            if (ret < 0)
@@ -184,10 +184,6 @@ int main(int argc, char **argv)
        }
        dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
                                                 ret, dst_sample_fmt, 1);
-        if (dst_bufsize < 0) {
-            fprintf(stderr, "Could not get sample buffer size\n");
-            goto end;
-        }
        printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret);
        fwrite(dst_data[0], 1, dst_bufsize, dst_file);
    } while (t < 10);
@@ -199,7 +195,8 @@ int main(int argc, char **argv)
            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
@@ -23,7 +23,7 @@
 /**
 * @file
 * libswscale API use example.
- * @example scaling_video.c
+ * @example doc/examples/scaling_video.c
 */

 #include <libavutil/imgutils.h>
@@ -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
@@ -1,770 +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
- */
-
-/**
- * @file
- * simple audio converter
- *
- * @example transcode_aac.c
- * Convert an input audio file to AAC in an MP4 container using FFmpeg.
- * @author Andreas Unterweger (dustsigns@gmail.com)
- */
-
-#include <stdio.h>
-
-#include "libavformat/avformat.h"
-#include "libavformat/avio.h"
-
-#include "libavcodec/avcodec.h"
-
-#include "libavutil/audio_fifo.h"
-#include "libavutil/avassert.h"
-#include "libavutil/avstring.h"
-#include "libavutil/frame.h"
-#include "libavutil/opt.h"
-
-#include "libswresample/swresample.h"
-
-/** The output bit rate in kbit/s */
-#define OUTPUT_BIT_RATE 96000
-/** The number of output channels */
-#define OUTPUT_CHANNELS 2
-
-/**
- * Convert an error code into a text message.
- * @param error Error code to be converted
- * @return Corresponding error text (not thread-safe)
- */
-static const char *get_error_text(const int error)
-{
-    static char error_buffer[255];
-    av_strerror(error, error_buffer, sizeof(error_buffer));
-    return error_buffer;
-}
-
-/** Open an input file and the required decoder. */
-static int open_input_file(const char *filename,
-                           AVFormatContext **input_format_context,
-                           AVCodecContext **input_codec_context)
-{
-    AVCodec *input_codec;
-    int error;
-
-    /** Open the input file to read from it. */
-    if ((error = avformat_open_input(input_format_context, filename, NULL,
-                                     NULL)) < 0) {
-        fprintf(stderr, "Could not open input file '%s' (error '%s')\n",
-                filename, get_error_text(error));
-        *input_format_context = NULL;
-        return error;
-    }
-
-    /** Get information on the input file (number of streams etc.). */
-    if ((error = avformat_find_stream_info(*input_format_context, NULL)) < 0) {
-        fprintf(stderr, "Could not open find stream info (error '%s')\n",
-                get_error_text(error));
-        avformat_close_input(input_format_context);
-        return error;
-    }
-
-    /** Make sure that there is only one stream in the input file. */
-    if ((*input_format_context)->nb_streams != 1) {
-        fprintf(stderr, "Expected one audio input stream, but found %d\n",
-                (*input_format_context)->nb_streams);
-        avformat_close_input(input_format_context);
-        return AVERROR_EXIT;
-    }
-
-    /** Find a decoder for the audio stream. */
-    if (!(input_codec = avcodec_find_decoder((*input_format_context)->streams[0]->codec->codec_id))) {
-        fprintf(stderr, "Could not find input codec\n");
-        avformat_close_input(input_format_context);
-        return AVERROR_EXIT;
-    }
-
-    /** Open the decoder for the audio stream to use it later. */
-    if ((error = avcodec_open2((*input_format_context)->streams[0]->codec,
-                               input_codec, NULL)) < 0) {
-        fprintf(stderr, "Could not open input codec (error '%s')\n",
-                get_error_text(error));
-        avformat_close_input(input_format_context);
-        return error;
-    }
-
-    /** Save the decoder context for easier access later. */
-    *input_codec_context = (*input_format_context)->streams[0]->codec;
-
-    return 0;
-}
-
-/**
- * Open an output file and the required encoder.
- * Also set some basic encoder parameters.
- * Some of these parameters are based on the input file's parameters.
- */
-static int open_output_file(const char *filename,
-                            AVCodecContext *input_codec_context,
-                            AVFormatContext **output_format_context,
-                            AVCodecContext **output_codec_context)
-{
-    AVIOContext *output_io_context = NULL;
-    AVStream *stream               = NULL;
-    AVCodec *output_codec          = NULL;
-    int error;
-
-    /** Open the output file to write to it. */
-    if ((error = avio_open(&output_io_context, filename,
-                           AVIO_FLAG_WRITE)) < 0) {
-        fprintf(stderr, "Could not open output file '%s' (error '%s')\n",
-                filename, get_error_text(error));
-        return error;
-    }
-
-    /** Create a new format context for the output container format. */
-    if (!(*output_format_context = avformat_alloc_context())) {
-        fprintf(stderr, "Could not allocate output format context\n");
-        return AVERROR(ENOMEM);
-    }
-
-    /** Associate the output file (pointer) with the container format context. */
-    (*output_format_context)->pb = output_io_context;
-
-    /** Guess the desired container format based on the file extension. */
-    if (!((*output_format_context)->oformat = av_guess_format(NULL, filename,
-                                                              NULL))) {
-        fprintf(stderr, "Could not find output file format\n");
-        goto cleanup;
-    }
-
-    av_strlcpy((*output_format_context)->filename, filename,
-               sizeof((*output_format_context)->filename));
-
-    /** Find the encoder to be used by its name. */
-    if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC))) {
-        fprintf(stderr, "Could not find an AAC encoder.\n");
-        goto cleanup;
-    }
-
-    /** Create a new audio stream in the output file container. */
-    if (!(stream = avformat_new_stream(*output_format_context, output_codec))) {
-        fprintf(stderr, "Could not create new stream\n");
-        error = AVERROR(ENOMEM);
-        goto cleanup;
-    }
-
-    /** Save the encoder context for easier access later. */
-    *output_codec_context = stream->codec;
-
-    /**
-     * Set the basic encoder parameters.
-     * The input file's sample rate is used to avoid a sample rate conversion.
-     */
-    (*output_codec_context)->channels       = OUTPUT_CHANNELS;
-    (*output_codec_context)->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
-    (*output_codec_context)->sample_rate    = input_codec_context->sample_rate;
-    (*output_codec_context)->sample_fmt     = output_codec->sample_fmts[0];
-    (*output_codec_context)->bit_rate       = OUTPUT_BIT_RATE;
-
-    /** Allow the use of the experimental AAC encoder */
-    (*output_codec_context)->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
-
-    /** Set the sample rate for the container. */
-    stream->time_base.den = input_codec_context->sample_rate;
-    stream->time_base.num = 1;
-
-    /**
-     * Some container formats (like MP4) require global headers to be present
-     * Mark the encoder so that it behaves accordingly.
-     */
-    if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
-        (*output_codec_context)->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
-
-    /** Open the encoder for the audio stream to use it later. */
-    if ((error = avcodec_open2(*output_codec_context, output_codec, NULL)) < 0) {
-        fprintf(stderr, "Could not open output codec (error '%s')\n",
-                get_error_text(error));
-        goto cleanup;
-    }
-
-    return 0;
-
-cleanup:
-    avio_closep(&(*output_format_context)->pb);
-    avformat_free_context(*output_format_context);
-    *output_format_context = NULL;
-    return error < 0 ? error : AVERROR_EXIT;
-}
-
-/** Initialize one data packet for reading or writing. */
-static void init_packet(AVPacket *packet)
-{
-    av_init_packet(packet);
-    /** Set the packet data and size so that it is recognized as being empty. */
-    packet->data = NULL;
-    packet->size = 0;
-}
-
-/** Initialize one audio frame for reading from the input file */
-static int init_input_frame(AVFrame **frame)
-{
-    if (!(*frame = av_frame_alloc())) {
-        fprintf(stderr, "Could not allocate input frame\n");
-        return AVERROR(ENOMEM);
-    }
-    return 0;
-}
-
-/**
- * Initialize the audio resampler based on the input and output codec settings.
- * If the input and output sample formats differ, a conversion is required
- * libswresample takes care of this, but requires initialization.
- */
-static int init_resampler(AVCodecContext *input_codec_context,
-                          AVCodecContext *output_codec_context,
-                          SwrContext **resample_context)
-{
-        int error;
-
-        /**
-         * Create a resampler context for the conversion.
-         * Set the conversion parameters.
-         * Default channel layouts based on the number of channels
-         * are assumed for simplicity (they are sometimes not detected
-         * properly by the demuxer and/or decoder).
-         */
-        *resample_context = swr_alloc_set_opts(NULL,
-                                              av_get_default_channel_layout(output_codec_context->channels),
-                                              output_codec_context->sample_fmt,
-                                              output_codec_context->sample_rate,
-                                              av_get_default_channel_layout(input_codec_context->channels),
-                                              input_codec_context->sample_fmt,
-                                              input_codec_context->sample_rate,
-                                              0, NULL);
-        if (!*resample_context) {
-            fprintf(stderr, "Could not allocate resample context\n");
-            return AVERROR(ENOMEM);
-        }
-        /**
-        * Perform a sanity check so that the number of converted samples is
-        * not greater than the number of samples to be converted.
-        * If the sample rates differ, this case has to be handled differently
-        */
-        av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);
-
-        /** Open the resampler with the specified parameters. */
-        if ((error = swr_init(*resample_context)) < 0) {
-            fprintf(stderr, "Could not open resample context\n");
-            swr_free(resample_context);
-            return error;
-        }
-    return 0;
-}
-
-/** Initialize a FIFO buffer for the audio samples to be encoded. */
-static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
-{
-    /** Create the FIFO buffer based on the specified output sample format. */
-    if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
-                                      output_codec_context->channels, 1))) {
-        fprintf(stderr, "Could not allocate FIFO\n");
-        return AVERROR(ENOMEM);
-    }
-    return 0;
-}
-
-/** Write the header of the output file container. */
-static int write_output_file_header(AVFormatContext *output_format_context)
-{
-    int error;
-    if ((error = avformat_write_header(output_format_context, NULL)) < 0) {
-        fprintf(stderr, "Could not write output file header (error '%s')\n",
-                get_error_text(error));
-        return error;
-    }
-    return 0;
-}
-
-/** Decode one audio frame from the input file. */
-static int decode_audio_frame(AVFrame *frame,
-                              AVFormatContext *input_format_context,
-                              AVCodecContext *input_codec_context,
-                              int *data_present, int *finished)
-{
-    /** Packet used for temporary storage. */
-    AVPacket input_packet;
-    int error;
-    init_packet(&input_packet);
-
-    /** Read one audio frame from the input file into a temporary packet. */
-    if ((error = av_read_frame(input_format_context, &input_packet)) < 0) {
-        /** If we are at the end of the file, flush the decoder below. */
-        if (error == AVERROR_EOF)
-            *finished = 1;
-        else {
-            fprintf(stderr, "Could not read frame (error '%s')\n",
-                    get_error_text(error));
-            return error;
-        }
-    }
-
-    /**
-     * Decode the audio frame stored in the temporary packet.
-     * The input audio stream decoder is used to do this.
-     * If we are at the end of the file, pass an empty packet to the decoder
-     * to flush it.
-     */
-    if ((error = avcodec_decode_audio4(input_codec_context, frame,
-                                       data_present, &input_packet)) < 0) {
-        fprintf(stderr, "Could not decode frame (error '%s')\n",
-                get_error_text(error));
-        av_packet_unref(&input_packet);
-        return error;
-    }
-
-    /**
-     * If the decoder has not been flushed completely, we are not finished,
-     * so that this function has to be called again.
-     */
-    if (*finished && *data_present)
-        *finished = 0;
-    av_packet_unref(&input_packet);
-    return 0;
-}
-
-/**
- * Initialize a temporary storage for the specified number of audio samples.
- * The conversion requires temporary storage due to the different format.
- * The number of audio samples to be allocated is specified in frame_size.
- */
-static int init_converted_samples(uint8_t ***converted_input_samples,
-                                  AVCodecContext *output_codec_context,
-                                  int frame_size)
-{
-    int error;
-
-    /**
-     * Allocate as many pointers as there are audio channels.
-     * Each pointer will later point to the audio samples of the corresponding
-     * channels (although it may be NULL for interleaved formats).
-     */
-    if (!(*converted_input_samples = calloc(output_codec_context->channels,
-                                            sizeof(**converted_input_samples)))) {
-        fprintf(stderr, "Could not allocate converted input sample pointers\n");
-        return AVERROR(ENOMEM);
-    }
-
-    /**
-     * Allocate memory for the samples of all channels in one consecutive
-     * block for convenience.
-     */
-    if ((error = av_samples_alloc(*converted_input_samples, NULL,
-                                  output_codec_context->channels,
-                                  frame_size,
-                                  output_codec_context->sample_fmt, 0)) < 0) {
-        fprintf(stderr,
-                "Could not allocate converted input samples (error '%s')\n",
-                get_error_text(error));
-        av_freep(&(*converted_input_samples)[0]);
-        free(*converted_input_samples);
-        return error;
-    }
-    return 0;
-}
-
-/**
- * Convert the input audio samples into the output sample format.
- * The conversion happens on a per-frame basis, the size of which is specified
- * by frame_size.
- */
-static int convert_samples(const uint8_t **input_data,
-                           uint8_t **converted_data, const int frame_size,
-                           SwrContext *resample_context)
-{
-    int error;
-
-    /** Convert the samples using the resampler. */
-    if ((error = swr_convert(resample_context,
-                             converted_data, frame_size,
-                             input_data    , frame_size)) < 0) {
-        fprintf(stderr, "Could not convert input samples (error '%s')\n",
-                get_error_text(error));
-        return error;
-    }
-
-    return 0;
-}
-
-/** Add converted input audio samples to the FIFO buffer for later processing. */
-static int add_samples_to_fifo(AVAudioFifo *fifo,
-                               uint8_t **converted_input_samples,
-                               const int frame_size)
-{
-    int error;
-
-    /**
-     * Make the FIFO as large as it needs to be to hold both,
-     * the old and the new samples.
-     */
-    if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
-        fprintf(stderr, "Could not reallocate FIFO\n");
-        return error;
-    }
-
-    /** Store the new samples in the FIFO buffer. */
-    if (av_audio_fifo_write(fifo, (void **)converted_input_samples,
-                            frame_size) < frame_size) {
-        fprintf(stderr, "Could not write data to FIFO\n");
-        return AVERROR_EXIT;
-    }
-    return 0;
-}
-
-/**
- * Read one audio frame from the input file, decodes, converts and stores
- * it in the FIFO buffer.
- */
-static int read_decode_convert_and_store(AVAudioFifo *fifo,
-                                         AVFormatContext *input_format_context,
-                                         AVCodecContext *input_codec_context,
-                                         AVCodecContext *output_codec_context,
-                                         SwrContext *resampler_context,
-                                         int *finished)
-{
-    /** Temporary storage of the input samples of the frame read from the file. */
-    AVFrame *input_frame = NULL;
-    /** Temporary storage for the converted input samples. */
-    uint8_t **converted_input_samples = NULL;
-    int data_present;
-    int ret = AVERROR_EXIT;
-
-    /** Initialize temporary storage for one input frame. */
-    if (init_input_frame(&input_frame))
-        goto cleanup;
-    /** Decode one frame worth of audio samples. */
-    if (decode_audio_frame(input_frame, input_format_context,
-                           input_codec_context, &data_present, finished))
-        goto cleanup;
-    /**
-     * If we are at the end of the file and there are no more samples
-     * in the decoder which are delayed, we are actually finished.
-     * This must not be treated as an error.
-     */
-    if (*finished && !data_present) {
-        ret = 0;
-        goto cleanup;
-    }
-    /** If there is decoded data, convert and store it */
-    if (data_present) {
-        /** Initialize the temporary storage for the converted input samples. */
-        if (init_converted_samples(&converted_input_samples, output_codec_context,
-                                   input_frame->nb_samples))
-            goto cleanup;
-
-        /**
-         * Convert the input samples to the desired output sample format.
-         * This requires a temporary storage provided by converted_input_samples.
-         */
-        if (convert_samples((const uint8_t**)input_frame->extended_data, converted_input_samples,
-                            input_frame->nb_samples, resampler_context))
-            goto cleanup;
-
-        /** Add the converted input samples to the FIFO buffer for later processing. */
-        if (add_samples_to_fifo(fifo, converted_input_samples,
-                                input_frame->nb_samples))
-            goto cleanup;
-        ret = 0;
-    }
-    ret = 0;
-
-cleanup:
-    if (converted_input_samples) {
-        av_freep(&converted_input_samples[0]);
-        free(converted_input_samples);
-    }
-    av_frame_free(&input_frame);
-
-    return ret;
-}
-
-/**
- * Initialize one input frame for writing to the output file.
- * The frame will be exactly frame_size samples large.
- */
-static int init_output_frame(AVFrame **frame,
-                             AVCodecContext *output_codec_context,
-                             int frame_size)
-{
-    int error;
-
-    /** Create a new frame to store the audio samples. */
-    if (!(*frame = av_frame_alloc())) {
-        fprintf(stderr, "Could not allocate output frame\n");
-        return AVERROR_EXIT;
-    }
-
-    /**
-     * Set the frame's parameters, especially its size and format.
-     * av_frame_get_buffer needs this to allocate memory for the
-     * audio samples of the frame.
-     * Default channel layouts based on the number of channels
-     * are assumed for simplicity.
-     */
-    (*frame)->nb_samples     = frame_size;
-    (*frame)->channel_layout = output_codec_context->channel_layout;
-    (*frame)->format         = output_codec_context->sample_fmt;
-    (*frame)->sample_rate    = output_codec_context->sample_rate;
-
-    /**
-     * Allocate the samples of the created frame. This call will make
-     * sure that the audio frame can hold as many samples as specified.
-     */
-    if ((error = av_frame_get_buffer(*frame, 0)) < 0) {
-        fprintf(stderr, "Could allocate output frame samples (error '%s')\n",
-                get_error_text(error));
-        av_frame_free(frame);
-        return error;
-    }
-
-    return 0;
-}
-
-/** Global timestamp for the audio frames */
-static int64_t pts = 0;
-
-/** Encode one frame worth of audio to the output file. */
-static int encode_audio_frame(AVFrame *frame,
-                              AVFormatContext *output_format_context,
-                              AVCodecContext *output_codec_context,
-                              int *data_present)
-{
-    /** Packet used for temporary storage. */
-    AVPacket output_packet;
-    int error;
-    init_packet(&output_packet);
-
-    /** Set a timestamp based on the sample rate for the container. */
-    if (frame) {
-        frame->pts = pts;
-        pts += frame->nb_samples;
-    }
-
-    /**
-     * Encode the audio frame and store it in the temporary packet.
-     * The output audio stream encoder is used to do this.
-     */
-    if ((error = avcodec_encode_audio2(output_codec_context, &output_packet,
-                                       frame, data_present)) < 0) {
-        fprintf(stderr, "Could not encode frame (error '%s')\n",
-                get_error_text(error));
-        av_packet_unref(&output_packet);
-        return error;
-    }
-
-    /** Write one audio frame from the temporary packet to the output file. */
-    if (*data_present) {
-        if ((error = av_write_frame(output_format_context, &output_packet)) < 0) {
-            fprintf(stderr, "Could not write frame (error '%s')\n",
-                    get_error_text(error));
-            av_packet_unref(&output_packet);
-            return error;
-        }
-
-        av_packet_unref(&output_packet);
-    }
-
-    return 0;
-}
-
-/**
- * Load one audio frame from the FIFO buffer, encode and write it to the
- * output file.
- */
-static int load_encode_and_write(AVAudioFifo *fifo,
-                                 AVFormatContext *output_format_context,
-                                 AVCodecContext *output_codec_context)
-{
-    /** Temporary storage of the output samples of the frame written to the file. */
-    AVFrame *output_frame;
-    /**
-     * Use the maximum number of possible samples per frame.
-     * If there is less than the maximum possible frame size in the FIFO
-     * buffer use this number. Otherwise, use the maximum possible frame size
-     */
-    const int frame_size = FFMIN(av_audio_fifo_size(fifo),
-                                 output_codec_context->frame_size);
-    int data_written;
-
-    /** Initialize temporary storage for one output frame. */
-    if (init_output_frame(&output_frame, output_codec_context, frame_size))
-        return AVERROR_EXIT;
-
-    /**
-     * Read as many samples from the FIFO buffer as required to fill the frame.
-     * The samples are stored in the frame temporarily.
-     */
-    if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) {
-        fprintf(stderr, "Could not read data from FIFO\n");
-        av_frame_free(&output_frame);
-        return AVERROR_EXIT;
-    }
-
-    /** Encode one frame worth of audio samples. */
-    if (encode_audio_frame(output_frame, output_format_context,
-                           output_codec_context, &data_written)) {
-        av_frame_free(&output_frame);
-        return AVERROR_EXIT;
-    }
-    av_frame_free(&output_frame);
-    return 0;
-}
-
-/** Write the trailer of the output file container. */
-static int write_output_file_trailer(AVFormatContext *output_format_context)
-{
-    int error;
-    if ((error = av_write_trailer(output_format_context)) < 0) {
-        fprintf(stderr, "Could not write output file trailer (error '%s')\n",
-                get_error_text(error));
-        return error;
-    }
-    return 0;
-}
-
-/** Convert an audio file to an AAC file in an MP4 container. */
-int main(int argc, char **argv)
-{
-    AVFormatContext *input_format_context = NULL, *output_format_context = NULL;
-    AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL;
-    SwrContext *resample_context = NULL;
-    AVAudioFifo *fifo = NULL;
-    int ret = AVERROR_EXIT;
-
-    if (argc < 3) {
-        fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
-        exit(1);
-    }
-
-    /** Register all codecs and formats so that they can be used. */
-    av_register_all();
-    /** Open the input file for reading. */
-    if (open_input_file(argv[1], &input_format_context,
-                        &input_codec_context))
-        goto cleanup;
-    /** Open the output file for writing. */
-    if (open_output_file(argv[2], input_codec_context,
-                         &output_format_context, &output_codec_context))
-        goto cleanup;
-    /** Initialize the resampler to be able to convert audio sample formats. */
-    if (init_resampler(input_codec_context, output_codec_context,
-                       &resample_context))
-        goto cleanup;
-    /** Initialize the FIFO buffer to store audio samples to be encoded. */
-    if (init_fifo(&fifo, output_codec_context))
-        goto cleanup;
-    /** Write the header of the output file container. */
-    if (write_output_file_header(output_format_context))
-        goto cleanup;
-
-    /**
-     * Loop as long as we have input samples to read or output samples
-     * to write; abort as soon as we have neither.
-     */
-    while (1) {
-        /** Use the encoder's desired frame size for processing. */
-        const int output_frame_size = output_codec_context->frame_size;
-        int finished                = 0;
-
-        /**
-         * Make sure that there is one frame worth of samples in the FIFO
-         * buffer so that the encoder can do its work.
-         * Since the decoder's and the encoder's frame size may differ, we
-         * need to FIFO buffer to store as many frames worth of input samples
-         * that they make up at least one frame worth of output samples.
-         */
-        while (av_audio_fifo_size(fifo) < output_frame_size) {
-            /**
-             * Decode one frame worth of audio samples, convert it to the
-             * output sample format and put it into the FIFO buffer.
-             */
-            if (read_decode_convert_and_store(fifo, input_format_context,
-                                              input_codec_context,
-                                              output_codec_context,
-                                              resample_context, &finished))
-                goto cleanup;
-
-            /**
-             * If we are at the end of the input file, we continue
-             * encoding the remaining audio samples to the output file.
-             */
-            if (finished)
-                break;
-        }
-
-        /**
-         * If we have enough samples for the encoder, we encode them.
-         * At the end of the file, we pass the remaining samples to
-         * the encoder.
-         */
-        while (av_audio_fifo_size(fifo) >= output_frame_size ||
-               (finished && av_audio_fifo_size(fifo) > 0))
-            /**
-             * Take one frame worth of audio samples from the FIFO buffer,
-             * encode it and write it to the output file.
-             */
-            if (load_encode_and_write(fifo, output_format_context,
-                                      output_codec_context))
-                goto cleanup;
-
-        /**
-         * If we are at the end of the input file and have encoded
-         * all remaining samples, we can exit this loop and finish.
-         */
-        if (finished) {
-            int data_written;
-            /** Flush the encoder as it may have delayed frames. */
-            do {
-                if (encode_audio_frame(NULL, output_format_context,
-                                       output_codec_context, &data_written))
-                    goto cleanup;
-            } while (data_written);
-            break;
-        }
-    }
-
-    /** Write the trailer of the output file container. */
-    if (write_output_file_trailer(output_format_context))
-        goto cleanup;
-    ret = 0;
-
-cleanup:
-    if (fifo)
-        av_audio_fifo_free(fifo);
-    swr_free(&resample_context);
-    if (output_codec_context)
-        avcodec_close(output_codec_context);
-    if (output_format_context) {
-        avio_closep(&output_format_context->pb);
-        avformat_free_context(output_format_context);
-    }
-    if (input_codec_context)
-        avcodec_close(input_codec_context);
-    if (input_format_context)
-        avformat_close_input(&input_format_context);
-
-    return ret;
-}
--- a/doc/examples/transcoding.c
+++ b/doc/examples/transcoding.c
@@ -1,582 +0,0 @@
-/*
- * Copyright (c) 2010 Nicolas George
- * Copyright (c) 2011 Stefano Sabatini
- * Copyright (c) 2014 Andrey Utkin
- *
- * 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
- * API example for demuxing, decoding, filtering, encoding and muxing
- * @example transcoding.c
- */
-
-#include <libavcodec/avcodec.h>
-#include <libavformat/avformat.h>
-#include <libavfilter/avfiltergraph.h>
-#include <libavfilter/buffersink.h>
-#include <libavfilter/buffersrc.h>
-#include <libavutil/opt.h>
-#include <libavutil/pixdesc.h>
-
-static AVFormatContext *ifmt_ctx;
-static AVFormatContext *ofmt_ctx;
-typedef struct FilteringContext {
-    AVFilterContext *buffersink_ctx;
-    AVFilterContext *buffersrc_ctx;
-    AVFilterGraph *filter_graph;
-} FilteringContext;
-static FilteringContext *filter_ctx;
-
-static int open_input_file(const char *filename)
-{
-    int ret;
-    unsigned int i;
-
-    ifmt_ctx = NULL;
-    if ((ret = avformat_open_input(&ifmt_ctx, filename, NULL, NULL)) < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
-        return ret;
-    }
-
-    if ((ret = avformat_find_stream_info(ifmt_ctx, NULL)) < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
-        return ret;
-    }
-
-    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
-        AVStream *stream;
-        AVCodecContext *codec_ctx;
-        stream = ifmt_ctx->streams[i];
-        codec_ctx = stream->codec;
-        /* Reencode video & audio and remux subtitles etc. */
-        if (codec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
-                || codec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
-            /* Open decoder */
-            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;
-            }
-        }
-    }
-
-    av_dump_format(ifmt_ctx, 0, filename, 0);
-    return 0;
-}
-
-static int open_output_file(const char *filename)
-{
-    AVStream *out_stream;
-    AVStream *in_stream;
-    AVCodecContext *dec_ctx, *enc_ctx;
-    AVCodec *encoder;
-    int ret;
-    unsigned int i;
-
-    ofmt_ctx = NULL;
-    avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, filename);
-    if (!ofmt_ctx) {
-        av_log(NULL, AV_LOG_ERROR, "Could not create output context\n");
-        return AVERROR_UNKNOWN;
-    }
-
-
-    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
-        out_stream = avformat_new_stream(ofmt_ctx, NULL);
-        if (!out_stream) {
-            av_log(NULL, AV_LOG_ERROR, "Failed allocating output stream\n");
-            return AVERROR_UNKNOWN;
-        }
-
-        in_stream = ifmt_ctx->streams[i];
-        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;
-            }
-
-            /* In this example, we transcode to same properties (picture size,
-             * sample rate etc.). These properties can be changed for output
-             * streams easily using filters */
-            if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) {
-                enc_ctx->height = dec_ctx->height;
-                enc_ctx->width = dec_ctx->width;
-                enc_ctx->sample_aspect_ratio = dec_ctx->sample_aspect_ratio;
-                /* take first format from list of supported formats */
-                enc_ctx->pix_fmt = encoder->pix_fmts[0];
-                /* video time_base can be set to whatever is handy and supported by encoder */
-                enc_ctx->time_base = dec_ctx->time_base;
-            } else {
-                enc_ctx->sample_rate = dec_ctx->sample_rate;
-                enc_ctx->channel_layout = dec_ctx->channel_layout;
-                enc_ctx->channels = av_get_channel_layout_nb_channels(enc_ctx->channel_layout);
-                /* take first format from list of supported formats */
-                enc_ctx->sample_fmt = encoder->sample_fmts[0];
-                enc_ctx->time_base = (AVRational){1, enc_ctx->sample_rate};
-            }
-
-            /* 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;
-            }
-        } 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_copy_context(ofmt_ctx->streams[i]->codec,
-                    ifmt_ctx->streams[i]->codec);
-            if (ret < 0) {
-                av_log(NULL, AV_LOG_ERROR, "Copying stream context failed\n");
-                return ret;
-            }
-        }
-
-        if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
-            enc_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
-
-    }
-    av_dump_format(ofmt_ctx, 0, filename, 1);
-
-    if (!(ofmt_ctx->oformat->flags & AVFMT_NOFILE)) {
-        ret = avio_open(&ofmt_ctx->pb, filename, AVIO_FLAG_WRITE);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Could not open output file '%s'", filename);
-            return ret;
-        }
-    }
-
-    /* init muxer, write output file header */
-    ret = avformat_write_header(ofmt_ctx, NULL);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Error occurred when opening output file\n");
-        return ret;
-    }
-
-    return 0;
-}
-
-static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
-        AVCodecContext *enc_ctx, const char *filter_spec)
-{
-    char args[512];
-    int ret = 0;
-    AVFilter *buffersrc = NULL;
-    AVFilter *buffersink = NULL;
-    AVFilterContext *buffersrc_ctx = NULL;
-    AVFilterContext *buffersink_ctx = NULL;
-    AVFilterInOut *outputs = avfilter_inout_alloc();
-    AVFilterInOut *inputs  = avfilter_inout_alloc();
-    AVFilterGraph *filter_graph = avfilter_graph_alloc();
-
-    if (!outputs || !inputs || !filter_graph) {
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-
-    if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) {
-        buffersrc = avfilter_get_by_name("buffer");
-        buffersink = avfilter_get_by_name("buffersink");
-        if (!buffersrc || !buffersink) {
-            av_log(NULL, AV_LOG_ERROR, "filtering source or sink element not found\n");
-            ret = AVERROR_UNKNOWN;
-            goto end;
-        }
-
-        snprintf(args, sizeof(args),
-                "video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
-                dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
-                dec_ctx->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",
-                args, NULL, filter_graph);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n");
-            goto end;
-        }
-
-        ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
-                NULL, NULL, filter_graph);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
-            goto end;
-        }
-
-        ret = av_opt_set_bin(buffersink_ctx, "pix_fmts",
-                (uint8_t*)&enc_ctx->pix_fmt, sizeof(enc_ctx->pix_fmt),
-                AV_OPT_SEARCH_CHILDREN);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n");
-            goto end;
-        }
-    } else if (dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
-        buffersrc = avfilter_get_by_name("abuffer");
-        buffersink = avfilter_get_by_name("abuffersink");
-        if (!buffersrc || !buffersink) {
-            av_log(NULL, AV_LOG_ERROR, "filtering source or sink element not found\n");
-            ret = AVERROR_UNKNOWN;
-            goto end;
-        }
-
-        if (!dec_ctx->channel_layout)
-            dec_ctx->channel_layout =
-                av_get_default_channel_layout(dec_ctx->channels);
-        snprintf(args, sizeof(args),
-                "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
-                dec_ctx->time_base.num, dec_ctx->time_base.den, dec_ctx->sample_rate,
-                av_get_sample_fmt_name(dec_ctx->sample_fmt),
-                dec_ctx->channel_layout);
-        ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
-                args, NULL, filter_graph);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
-            goto end;
-        }
-
-        ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
-                NULL, NULL, filter_graph);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
-            goto end;
-        }
-
-        ret = av_opt_set_bin(buffersink_ctx, "sample_fmts",
-                (uint8_t*)&enc_ctx->sample_fmt, sizeof(enc_ctx->sample_fmt),
-                AV_OPT_SEARCH_CHILDREN);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
-            goto end;
-        }
-
-        ret = av_opt_set_bin(buffersink_ctx, "channel_layouts",
-                (uint8_t*)&enc_ctx->channel_layout,
-                sizeof(enc_ctx->channel_layout), AV_OPT_SEARCH_CHILDREN);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
-            goto end;
-        }
-
-        ret = av_opt_set_bin(buffersink_ctx, "sample_rates",
-                (uint8_t*)&enc_ctx->sample_rate, sizeof(enc_ctx->sample_rate),
-                AV_OPT_SEARCH_CHILDREN);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
-            goto end;
-        }
-    } else {
-        ret = AVERROR_UNKNOWN;
-        goto end;
-    }
-
-    /* Endpoints for the filter graph. */
-    outputs->name       = av_strdup("in");
-    outputs->filter_ctx = buffersrc_ctx;
-    outputs->pad_idx    = 0;
-    outputs->next       = NULL;
-
-    inputs->name       = av_strdup("out");
-    inputs->filter_ctx = buffersink_ctx;
-    inputs->pad_idx    = 0;
-    inputs->next       = NULL;
-
-    if (!outputs->name || !inputs->name) {
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-
-    if ((ret = avfilter_graph_parse_ptr(filter_graph, filter_spec,
-                    &inputs, &outputs, NULL)) < 0)
-        goto end;
-
-    if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
-        goto end;
-
-    /* Fill FilteringContext */
-    fctx->buffersrc_ctx = buffersrc_ctx;
-    fctx->buffersink_ctx = buffersink_ctx;
-    fctx->filter_graph = filter_graph;
-
-end:
-    avfilter_inout_free(&inputs);
-    avfilter_inout_free(&outputs);
-
-    return ret;
-}
-
-static int init_filters(void)
-{
-    const char *filter_spec;
-    unsigned int i;
-    int ret;
-    filter_ctx = av_malloc_array(ifmt_ctx->nb_streams, sizeof(*filter_ctx));
-    if (!filter_ctx)
-        return AVERROR(ENOMEM);
-
-    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
-        filter_ctx[i].buffersrc_ctx  = NULL;
-        filter_ctx[i].buffersink_ctx = NULL;
-        filter_ctx[i].filter_graph   = NULL;
-        if (!(ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO
-                || ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO))
-            continue;
-
-
-        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], ifmt_ctx->streams[i]->codec,
-                ofmt_ctx->streams[i]->codec, filter_spec);
-        if (ret)
-            return ret;
-    }
-    return 0;
-}
-
-static int encode_write_frame(AVFrame *filt_frame, unsigned int stream_index, int *got_frame) {
-    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 */
-    enc_pkt.data = NULL;
-    enc_pkt.size = 0;
-    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;
-
-    /* prepare packet for muxing */
-    enc_pkt.stream_index = stream_index;
-    av_packet_rescale_ts(&enc_pkt,
-                         ofmt_ctx->streams[stream_index]->codec->time_base,
-                         ofmt_ctx->streams[stream_index]->time_base);
-
-    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)
-{
-    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_ctx[stream_index].buffersrc_ctx,
-            frame, 0);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
-        return ret;
-    }
-
-    /* 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_ctx[stream_index].buffersink_ctx,
-                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
-             * rewrite retcode to 0 to show it as normal procedure completion
-             */
-            if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
-                ret = 0;
-            av_frame_free(&filt_frame);
-            break;
-        }
-
-        filt_frame->pict_type = AV_PICTURE_TYPE_NONE;
-        ret = encode_write_frame(filt_frame, stream_index, NULL);
-        if (ret < 0)
-            break;
-    }
-
-    return ret;
-}
-
-static int flush_encoder(unsigned int stream_index)
-{
-    int ret;
-    int got_frame;
-
-    if (!(ofmt_ctx->streams[stream_index]->codec->codec->capabilities &
-                AV_CODEC_CAP_DELAY))
-        return 0;
-
-    while (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 = { .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;
-
-    /* read all packets */
-    while (1) {
-        if ((ret = av_read_frame(ifmt_ctx, &packet)) < 0)
-            break;
-        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) {
-            av_log(NULL, AV_LOG_DEBUG, "Going to reencode&filter the frame\n");
-            frame = av_frame_alloc();
-            if (!frame) {
-                ret = AVERROR(ENOMEM);
-                break;
-            }
-            av_packet_rescale_ts(&packet,
-                                 ifmt_ctx->streams[stream_index]->time_base,
-                                 ifmt_ctx->streams[stream_index]->codec->time_base);
-            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;
-            }
-
-            if (got_frame) {
-                frame->pts = av_frame_get_best_effort_timestamp(frame);
-                ret = filter_encode_write_frame(frame, stream_index);
-                av_frame_free(&frame);
-                if (ret < 0)
-                    goto end;
-            } else {
-                av_frame_free(&frame);
-            }
-        } else {
-            /* remux this frame without reencoding */
-            av_packet_rescale_ts(&packet,
-                                 ifmt_ctx->streams[stream_index]->time_base,
-                                 ofmt_ctx->streams[stream_index]->time_base);
-
-            ret = av_interleaved_write_frame(ofmt_ctx, &packet);
-            if (ret < 0)
-                goto end;
-        }
-        av_packet_unref(&packet);
-    }
-
-    /* flush filters and encoders */
-    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
-        /* flush filter */
-        if (!filter_ctx[i].filter_graph)
-            continue;
-        ret = filter_encode_write_frame(NULL, i);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Flushing filter failed\n");
-            goto end;
-        }
-
-        /* flush encoder */
-        ret = flush_encoder(i);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Flushing encoder failed\n");
-            goto end;
-        }
-    }
-
-    av_write_trailer(ofmt_ctx);
-end:
-    av_packet_unref(&packet);
-    av_frame_free(&frame);
-    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
-        avcodec_close(ifmt_ctx->streams[i]->codec);
-        if (ofmt_ctx && ofmt_ctx->nb_streams > i && ofmt_ctx->streams[i] && ofmt_ctx->streams[i]->codec)
-            avcodec_close(ofmt_ctx->streams[i]->codec);
-        if (filter_ctx && filter_ctx[i].filter_graph)
-            avfilter_graph_free(&filter_ctx[i].filter_graph);
-    }
-    av_free(filter_ctx);
-    avformat_close_input(&ifmt_ctx);
-    if (ofmt_ctx && !(ofmt_ctx->oformat->flags & AVFMT_NOFILE))
-        avio_closep(&ofmt_ctx->pb);
-    avformat_free_context(ofmt_ctx);
-
-    if (ret < 0)
-        av_log(NULL, AV_LOG_ERROR, "Error occurred: %s\n", av_err2str(ret));
-
-    return ret ? 1 : 0;
-}
--- a/doc/faq.texi
+++ b/doc/faq.texi
@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8

@settitle FFmpeg FAQ
@titlepage
@@ -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?

@@ -156,7 +105,7 @@ For example, img1.jpg, img2.jpg, img3.jpg,...
 Then you may run:

@example
-ffmpeg -f image2 -i img%d.jpg /tmp/a.mpg
+  ffmpeg -f image2 -i img%d.jpg /tmp/a.mpg
@end example

 Notice that @samp{%d} is replaced by the image number.
@@ -169,7 +118,7 @@ the sequence. This is useful if your sequence does not start with
 example will start with @file{img100.jpg}:

@example
-ffmpeg -f image2 -start_number 100 -i img%d.jpg /tmp/a.mpg
+  ffmpeg -f image2 -start_number 100 -i img%d.jpg /tmp/a.mpg
@end example

 If you have large number of pictures to rename, you can use the
@@ -179,7 +128,7 @@ that match @code{*jpg} to the @file{/tmp} directory in the sequence of
@file{img001.jpg}, @file{img002.jpg} and so on.

@example
-x=1; for i in *jpg; do counter=$(printf %03d $x); ln -s "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done
+  x=1; for i in *jpg; do counter=$(printf %03d $x); ln -s "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done
@end example

 If you want to sequence them by oldest modified first, substitute
@@ -188,7 +137,7 @@ If you want to sequence them by oldest modified first, substitute
 Then run:

@example
-ffmpeg -f image2 -i /tmp/img%03d.jpg /tmp/a.mpg
+  ffmpeg -f image2 -i /tmp/img%03d.jpg /tmp/a.mpg
@end example

 The same logic is used for any image format that ffmpeg reads.
@@ -196,7 +145,7 @@ The same logic is used for any image format that ffmpeg reads.
 You can also use @command{cat} to pipe images to ffmpeg:

@example
-cat *.jpg | ffmpeg -f image2pipe -c:v mjpeg -i - output.mpg
+  cat *.jpg | ffmpeg -f image2pipe -c:v mjpeg -i - output.mpg
@end example

@section How do I encode movie to single pictures?
@@ -204,7 +153,7 @@ cat *.jpg | ffmpeg -f image2pipe -c:v mjpeg -i - output.mpg
 Use:

@example
-ffmpeg -i movie.mpg movie%d.jpg
+  ffmpeg -i movie.mpg movie%d.jpg
@end example

 The @file{movie.mpg} used as input will be converted to
@@ -220,7 +169,7 @@ to force the encoding.

 Applying that to the previous example:
@example
-ffmpeg -i movie.mpg -f image2 -c:v mjpeg menu%d.jpg
+  ffmpeg -i movie.mpg -f image2 -c:v mjpeg menu%d.jpg
@end example

 Beware that there is no "jpeg" codec. Use "mjpeg" instead.
@@ -278,15 +227,15 @@ then you may use any file that DirectShow can read as input.

 Just create an "input.avs" text file with this single line ...
@example
-DirectShowSource("C:\path to your file\yourfile.asf")
+  DirectShowSource("C:\path to your file\yourfile.asf")
@end example
 ... and then feed that text file to ffmpeg:
@example
-ffmpeg -i input.avs
+  ffmpeg -i input.avs
@end example

-For ANY other help on AviSynth, please visit the
-@uref{http://www.avisynth.org/, AviSynth homepage}.
+For ANY other help on Avisynth, please visit the
+@uref{http://www.avisynth.org/, Avisynth homepage}.

@section How can I join video files?

@@ -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{https://ffmpeg.org/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{https://ffmpeg.org/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{https://ffmpeg.org/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{https://ffmpeg.org/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{https://www.ffmpeg.org/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{https://ffmpeg.org/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
@@ -419,6 +368,26 @@ ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 44100 -i all.a \
 rm temp[12].[av] all.[av]
@end example

+@section -profile option fails when encoding H.264 video with AAC audio
+
+@command{ffmpeg} prints an error like
+
+@example
+Undefined constant or missing '(' in 'baseline'
+Unable to parse option value "baseline"
+Error setting option profile to value baseline.
+@end example
+
+Short answer: write @option{-profile:v} instead of @option{-profile}.
+
+Long answer: this happens because the @option{-profile} option can apply to both
+video and audio.  Specifically the AAC encoder also defines some profiles, none
+of which are named @var{baseline}.
+
+The solution is to apply the @option{-profile} option to the video stream only
+by using @url{http://ffmpeg.org/ffmpeg.html#Stream-specifiers-1, Stream specifiers}.
+Appending @code{:v} to it will do exactly that.
+
@section Using @option{-f lavfi}, audio becomes mono for no apparent reason.

 Use @option{-dumpgraph -} to find out exactly where the channel layout is
@@ -443,7 +412,7 @@ VOB and a few other formats do not have a global header that describes
 everything present in the file. Instead, applications are supposed to scan
 the file to see what it contains. Since VOB files are frequently large, only
 the beginning is scanned. If the subtitles happen only later in the file,
-they will not be initially detected.
+they will not be initally detected.

 Some applications, including the @code{ffmpeg} command-line tool, can only
 work with streams that were detected during the initial scan; streams that
@@ -467,40 +436,6 @@ 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{https://ffmpeg.org/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
-
@chapter Development

@section Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat?
@@ -540,10 +475,9 @@ read @uref{http://www.tux.org/lkml/#s15, "Programming Religion"}.

@section Why are the ffmpeg programs devoid of debugging symbols?

-The build process creates @command{ffmpeg_g}, @command{ffplay_g}, etc. which
-contain full debug information. Those binaries are stripped to create
-@command{ffmpeg}, @command{ffplay}, etc. If you need the debug information, use
-the *_g versions.
+The build process creates ffmpeg_g, ffplay_g, etc. which contain full debug
+information. Those binaries are stripped to create ffmpeg, ffplay, etc. If
+you need the debug information, use the *_g versions.

@section I do not like the LGPL, can I contribute code under the GPL instead?

@@ -563,7 +497,7 @@ An easy way to get the full list of required libraries in dependency order
 is to use @code{pkg-config}.

@example
-c99 -o program program.c $(pkg-config --cflags --libs libavformat libavcodec)
+  c99 -o program program.c $(pkg-config --cflags --libs libavformat libavcodec)
@end example

 See @file{doc/example/Makefile} and @file{doc/example/pc-uninstalled} for
@@ -587,9 +521,13 @@ to use them you have to append -D__STDC_CONSTANT_MACROS to your CXXFLAGS
 You have to create a custom AVIOContext using @code{avio_alloc_context},
 see @file{libavformat/aviobuf.c} in FFmpeg and @file{libmpdemux/demux_lavf.c} in MPlayer or MPlayer2 sources.

+@section Where can I find libav* headers for Pascal/Delphi?
+
+see @url{http://www.iversenit.dk/dev/ffmpeg-headers/}
+
@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?

@@ -599,12 +537,11 @@ In this specific case please look at RFC 4629 to see how it should be done.

@section AVStream.r_frame_rate is wrong, it is much larger than the frame rate.

-@code{r_frame_rate} is NOT the average frame rate, it is the smallest frame rate
+r_frame_rate is NOT the average frame rate, it is the smallest frame rate
 that can accurately represent all timestamps. So no, it is not
 wrong if it is larger than the average!
-For example, if you have mixed 25 and 30 fps content, then @code{r_frame_rate}
-will be 150 (it is the least common multiple).
-If you are looking for the average frame rate, see @code{AVStream.avg_frame_rate}.
+For example, if you have mixed 25 and 30 fps content, then r_frame_rate
+will be 150.

@section Why is @code{make fate} not running all tests?

--- 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
-invoke the makefile targets without setting the @var{SAMPLES} makefile
+`--samples=<path to the samples directory>'. Afterwards you can
+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
@@ -85,7 +84,7 @@ To use a custom wrapper to run the test, pass @option{--target-exec} to

@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.

@@ -93,23 +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
-template. The @env{slot} configuration variable can be any string that is not
+  Create a configuration that suits your needs, based on the configuration
+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
-configuration file itself will be sourced in a shell script, therefore all
-shell features may be used. This enables you to setup the environment as you
-need it for your build.
+pattern <arch>-<os>-<compiler>-<compiler version>. The configuration file
+itself will be sourced in a shell script, therefore all shell features may
+be used. This enables you to setup the environment as you need it for your
+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:
@@ -122,29 +121,29 @@ 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.


@@ -154,20 +153,20 @@ the synchronisation of the samples directory.

@table @option
@item fate-rsync
-Download/synchronize sample files to the configured samples directory.
+    Download/synchronize sample files to the configured samples directory.

@item fate-list
-Will list all fate/regression test targets.
+    Will list all fate/regression test targets.

@item fate
-Run the FATE test suite (requires the fate-suite dataset).
+    Run the FATE test suite (requires the fate-suite dataset).
@end table

@section Makefile variables

-@table @env
+@table @option
@item V
-Verbosity level, can be set to 0, 1 or 2.
+    Verbosity level, can be set to 0, 1 or 2.
    @itemize
        @item 0: show just the test arguments
        @item 1: show just the command used in the test
@@ -175,28 +174,24 @@ Verbosity level, can be set to 0, 1 or 2.
    @end itemize

@item SAMPLES
-Specify or override the path to the FATE samples at make time, it has a
-meaning only while running the regression tests.
+    Specify or override the path to the FATE samples at make time, it has a
+    meaning only while running the regression tests.

@item THREADS
-Specify how many threads to use while running regression tests, it is
-quite useful to detect thread-related regressions.
-
+    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},
-by default @samp{slice+frame}
-
+    Specify which threading strategy test, either @var{slice} or @var{frame},
+    by default @var{slice+frame}
@item CPUFLAGS
-Specify CPU flags.
-
+    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
-@command{valgrind}, @command{qemu-user} or @command{wine} or on remote targets
-through @command{ssh}.
-
+    Specify or override the wrapper used to run the tests.
+    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.
@end table

@section Examples
--- a/doc/fate_config.sh.template
+++ b/doc/fate_config.sh.template
@@ -1,6 +1,5 @@
 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
--- 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
--- a/doc/ffmpeg-codecs.texi
+++ b/doc/ffmpeg-codecs.texi
@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8

@settitle FFmpeg Codecs Documentation
@titlepage
--- a/doc/ffmpeg-devices.texi
+++ b/doc/ffmpeg-devices.texi
@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8

@settitle FFmpeg Devices Documentation
@titlepage
--- a/doc/ffmpeg-filters.texi
+++ b/doc/ffmpeg-filters.texi
@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8

@settitle FFmpeg Filters Documentation
@titlepage
--- a/doc/ffmpeg-formats.texi
+++ b/doc/ffmpeg-formats.texi
@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8

@settitle FFmpeg Formats Documentation
@titlepage
--- a/doc/ffmpeg-protocols.texi
+++ b/doc/ffmpeg-protocols.texi
@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8

@settitle FFmpeg Protocols Documentation
@titlepage
--- a/doc/ffmpeg-resampler.texi
+++ b/doc/ffmpeg-resampler.texi
@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8

@settitle FFmpeg Resampler Documentation
@titlepage
@@ -13,9 +12,9 @@
@chapter Description
@c man begin DESCRIPTION

-The FFmpeg resampler provides a high-level interface to the
+The FFmpeg resampler provides an high-level interface to the
 libswresample library audio resampling utilities. In particular it
-allows one to perform audio resampling, audio channel layout rematrixing,
+allows to perform audio resampling, audio channel layout rematrixing,
 and convert audio format and packing layout.

@c man end DESCRIPTION
--- a/doc/ffmpeg-scaler.texi
+++ b/doc/ffmpeg-scaler.texi
@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8

@settitle FFmpeg Scaler Documentation
@titlepage
@@ -13,8 +12,8 @@
@chapter Description
@c man begin DESCRIPTION

-The FFmpeg rescaler provides a high-level interface to the libswscale
-library image conversion utilities. In particular it allows one to perform
+The FFmpeg rescaler provides an high-level interface to the libswscale
+library image conversion utilities. In particular it allows to perform
 image rescaling and pixel format conversion.

@c man end DESCRIPTION
--- a/doc/ffmpeg-utils.texi
+++ b/doc/ffmpeg-utils.texi
@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8

@settitle FFmpeg Utilities Documentation
@titlepage
--- a/doc/ffmpeg.texi
+++ b/doc/ffmpeg.texi
@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8

@settitle ffmpeg Documentation
@titlepage
@@ -80,26 +79,14 @@ The format option may be needed for raw input files.
 The transcoding process in @command{ffmpeg} for each output can be described by
 the following diagram:

-@verbatim
- _______              ______________
-|       |            |              |
-| input |  demuxer   | encoded data |   decoder
-| file  | ---------> | packets      | -----+
-|_______|            |______________|      |
-                                           v
-                                       _________
-                                      |         |
-                                      | decoded |
-                                      | frames  |
-                                      |_________|
- ________             ______________       |
-|        |           |              |      |
-| output | <-------- | encoded data | <----+
-| file   |   muxer   | packets      |   encoder
-|________|           |______________|
+@example
+ _______              ______________               _________              ______________            ________
+|       |            |              |             |         |            |              |          |        |
+| input |  demuxer   | encoded data |   decoder   | decoded |  encoder   | encoded data |  muxer   | output |
+| file  | ---------> | packets      |  ---------> | frames  | ---------> | packets      | -------> | file   |
+|_______|            |______________|             |_________|            |______________|          |________|

-
-@end verbatim
+@end example

@command{ffmpeg} calls the libavformat library (containing demuxers) to read
 input files and get packets containing encoded data from them. When there are
@@ -124,31 +111,26 @@ Simple filtergraphs are those that have exactly one input and output, both of
 the same type. In the above diagram they can be represented by simply inserting
 an additional step between decoding and encoding:

-@verbatim
- _________                        ______________
-|         |                      |              |
-| decoded |                      | encoded data |
-| frames  |\                   _ | packets      |
-|_________| \                  /||______________|
-             \   __________   /
-  simple     _\||          | /  encoder
-  filtergraph   | filtered |/
-                | frames   |
-                |__________|
+@example
+ _________                        __________              ______________
+|         |                      |          |            |              |
+| decoded |  simple filtergraph  | filtered |  encoder   | encoded data |
+| frames  | -------------------> | frames   | ---------> | packets      |
+|_________|                      |__________|            |______________|

-@end verbatim
+@end example

 Simple filtergraphs are configured with the per-stream @option{-filter} option
 (with @option{-vf} and @option{-af} aliases for video and audio respectively).
 A simple filtergraph for video can look for example like this:

-@verbatim
- _______        _____________        _______        ________
-|       |      |             |      |       |      |        |
-| input | ---> | deinterlace | ---> | scale | ---> | output |
-|_______|      |_____________|      |_______|      |________|
+@example
+ _______        _____________        _______        _____        ________
+|       |      |             |      |       |      |     |      |        |
+| input | ---> | deinterlace | ---> | scale | ---> | fps | ---> | output |
+|_______|      |_____________|      |_______|      |_____|      |________|

-@end verbatim
+@end example

 Note that some filters change frame properties but not frame contents. E.g. the
@code{fps} filter in the example above changes number of frames, but does not
@@ -161,7 +143,7 @@ processing chain applied to one stream. This is the case, for example, when the
 more than one input and/or output, or when output stream type is different from
 input. They can be represented with the following diagram:

-@verbatim
+@example
 _________
 |         |
 | input 0 |\                    __________
@@ -179,7 +161,7 @@ input. They can be represented with the following diagram:
 | input 2 |/
 |_________|

-@end verbatim
+@end example

 Complex filtergraphs are configured with the @option{-filter_complex} option.
 Note that this option is global, since a complex filtergraph, by its nature,
@@ -198,14 +180,14 @@ step for the specified stream, so it does only demuxing and muxing. It is useful
 for changing the container format or modifying container-level metadata. The
 diagram above will, in this case, simplify to this:

-@verbatim
+@example
 _______              ______________            ________
 |       |            |              |          |        |
 | input |  demuxer   | encoded data |  muxer   | output |
 | file  | ---------> | packets      | -------> | file   |
 |_______|            |______________|          |________|

-@end verbatim
+@end example

 Since there is no decoding or encoding, it is very fast and there is no quality
 loss. However, it might not work in some cases because of many factors. Applying
@@ -232,7 +214,7 @@ described.
@chapter Options
@c man begin OPTIONS

-@include fftools-common-opts.texi
+@include avtools-common-opts.texi

@section Main options

@@ -253,10 +235,6 @@ Overwrite output files without asking.
 Do not overwrite output files, and exit immediately if a specified
 output file already exists.

-@item -stream_loop @var{number} (@emph{input})
-Set number of times input stream shall be looped. Loop 0 means no loop,
-loop -1 means infinite loop.
-
@item -c[:@var{stream_specifier}] @var{codec} (@emph{input/output,per-stream})
@itemx -codec[:@var{stream_specifier}] @var{codec} (@emph{input/output,per-stream})
 Select an encoder (when used before an output file) or a decoder (when used
@@ -277,72 +255,54 @@ ffmpeg -i INPUT -map 0 -c copy -c:v:1 libx264 -c:a:137 libvorbis OUTPUT
 will copy all the streams except the second video, which will be encoded with
 libx264, and the 138th audio, which will be encoded with libvorbis.

-@item -t @var{duration} (@emph{input/output})
-When used as an input option (before @code{-i}), limit the @var{duration} of
-data read from the input file.
-
-When used as an output option (before an output filename), stop writing the
-output after its duration reaches @var{duration}.
-
-@var{duration} must be a time duration specification,
-see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
+@item -t @var{duration} (@emph{output})
+Stop writing the output after its duration reaches @var{duration}.
+@var{duration} may be a number in seconds, or in @code{hh:mm:ss[.xxx]} form.

 -to and -t are mutually exclusive and -t has priority.

@item -to @var{position} (@emph{output})
 Stop writing the output at @var{position}.
-@var{position} must be a time duration specification,
-see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
+@var{position} may be a number in seconds, or in @code{hh:mm:ss[.xxx]} form.

 -to and -t are mutually exclusive and -t has priority.

@item -fs @var{limit_size} (@emph{output})
-Set the file size limit, expressed in bytes. No further chunk of bytes is written
-after the limit is exceeded. The size of the output file is slightly more than the
-requested file size.
+Set the file size limit, expressed in bytes.

@item -ss @var{position} (@emph{input/output})
 When used as an input option (before @code{-i}), seeks in this input file to
-@var{position}. Note that in most formats it is not possible to seek exactly,
-so @command{ffmpeg} will seek to the closest seek point before @var{position}.
-When transcoding and @option{-accurate_seek} is enabled (the default), this
-extra segment between the seek point and @var{position} will be decoded and
-discarded. When doing stream copy or when @option{-noaccurate_seek} is used, it
-will be preserved.
+@var{position}. When used as an output option (before an output filename),
+decodes but discards input until the timestamps reach @var{position}. This is
+slower, but more accurate.

-When used as an output option (before an output filename), decodes but discards
-input until the timestamps reach @var{position}.
-
-@var{position} must be a time duration specification,
-see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
-
-@item -sseof @var{position} (@emph{input/output})
-
-Like the @code{-ss} option but relative to the "end of file". That is negative
-values are earlier in the file, 0 is at EOF.
+@var{position} may be either in seconds or in @code{hh:mm:ss[.xxx]} form.

@item -itsoffset @var{offset} (@emph{input})
-Set the input time offset.
+Set the input time offset in seconds.
+@code{[-]hh:mm:ss[.xxx]} syntax is also supported.
+The offset is added to the timestamps of the input files.
+Specifying a positive offset means that the corresponding
+streams are delayed by @var{offset} seconds.

-@var{offset} must be a time duration specification,
-see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
-
-The offset is added to the timestamps of the input files. Specifying
-a positive offset means that the corresponding streams are delayed by
-the time duration specified in @var{offset}.
-
-@item -timestamp @var{date} (@emph{output})
+@item -timestamp @var{time} (@emph{output})
 Set the recording timestamp in the container.
-
-@var{date} must be a date specification,
-see @ref{date syntax,,the Date section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
+The syntax for @var{time} is:
+@example
+now|([(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH:MM:SS[.m...])|(HHMMSS[.m...]))[Z|z])
+@end example
+If the value is "now" it takes the current time.
+Time is local time unless 'Z' or 'z' is appended, in which case it is
+interpreted as UTC.
+If the year-month-day part is not specified it takes the current
+year-month-day.

@item -metadata[:metadata_specifier] @var{key}=@var{value} (@emph{output,per-metadata})
 Set a metadata key/value pair.

 An optional @var{metadata_specifier} may be given to set metadata
-on streams, chapters or programs. See @code{-map_metadata}
-documentation for details.
+on streams or chapters. See @code{-map_metadata} documentation for
+details.

 This option overrides metadata set with @code{-map_metadata}. It is
 also possible to delete metadata by using an empty value.
@@ -354,14 +314,9 @@ ffmpeg -i in.avi -metadata title="my title" out.flv

 To set the language of the first audio stream:
@example
-ffmpeg -i INPUT -metadata:s:a:0 language=eng OUTPUT
+ffmpeg -i INPUT -metadata:s:a:1 language=eng OUTPUT
@end example

-@item -program [title=@var{title}:][program_num=@var{program_num}:]st=@var{stream}[:st=@var{stream}...] (@emph{output})
-
-Creates a program with the specified @var{title}, @var{program_num} and adds the specified
-@var{stream}(s) to it.
-
@item -target @var{type} (@emph{output})
 Specify target file type (@code{vcd}, @code{svcd}, @code{dvd}, @code{dv},
@code{dv50}). @var{type} may be prefixed with @code{pal-}, @code{ntsc-} or
@@ -380,20 +335,15 @@ ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg
@end example

@item -dframes @var{number} (@emph{output})
-Set the number of data frames to output. This is an alias for @code{-frames:d}.
+Set the number of data frames to record. This is an alias for @code{-frames:d}.

@item -frames[:@var{stream_specifier}] @var{framecount} (@emph{output,per-stream})
 Stop writing to the stream after @var{framecount} frames.

@item -q[:@var{stream_specifier}] @var{q} (@emph{output,per-stream})
@itemx -qscale[:@var{stream_specifier}] @var{q} (@emph{output,per-stream})
-Use fixed quality scale (VBR). The meaning of @var{q}/@var{qscale} is
+Use fixed quality scale (VBR). The meaning of @var{q} is
 codec-dependent.
-If @var{qscale} is used without a @var{stream_specifier} then it applies only
-to the video stream, this is to maintain compatibility with previous behavior
-and as specifying the same codec specific value to 2 different codecs that is
-audio and video generally is not what is intended when no stream_specifier is
-used.

@anchor{filter_option}
@item -filter[:@var{stream_specifier}] @var{filtergraph} (@emph{output,per-stream})
@@ -481,24 +431,18 @@ Technical note -- attachments are implemented as codec extradata, so this
 option can actually be used to extract extradata from any stream, not just
 attachments.

-@item -noautorotate
-Disable automatically rotating video based on file metadata.
-
@end table

@section Video Options

@table @option
@item -vframes @var{number} (@emph{output})
-Set the number of video frames to output. This is an alias for @code{-frames:v}.
+Set the number of video frames to record. This is an alias for @code{-frames:v}.
@item -r[:@var{stream_specifier}] @var{fps} (@emph{input/output,per-stream})
 Set frame rate (Hz value, fraction or abbreviation).

 As an input option, ignore any timestamps stored in the file and instead
 generate timestamps assuming constant frame rate @var{fps}.
-This is not the same as the @option{-framerate} option used for some input formats
-like image2 or v4l2 (it used to be the same in older versions of FFmpeg).
-If in doubt use @option{-framerate} instead of the input option @option{-r}.

 As an output option, duplicate or drop input frames to achieve constant output
 frame rate @var{fps}.
@@ -553,6 +497,9 @@ prefix is ``ffmpeg2pass''. The complete file name will be
@file{PREFIX-N.log}, where N is a number specific to the output
 stream

+@item -vlang @var{code}
+Set the ISO 639 language code (3 letters) of the current video stream.
+
@item -vf @var{filtergraph} (@emph{output})
 Create the filtergraph specified by @var{filtergraph} and use it to
 filter the stream.
@@ -560,7 +507,7 @@ filter the stream.
 This is an alias for @code{-filter:v}, see the @ref{filter_option,,-filter option}.
@end table

-@section Advanced Video options
+@section Advanced Video Options

@table @option
@item -pix_fmt[:@var{stream_specifier}] @var{format} (@emph{input/output,per-stream})
@@ -663,87 +610,13 @@ would be more efficient.
@item -copyinkf[:@var{stream_specifier}] (@emph{output,per-stream})
 When doing stream copy, copy also non-key frames found at the
 beginning.
-
-@item -hwaccel[:@var{stream_specifier}] @var{hwaccel} (@emph{input,per-stream})
-Use hardware acceleration to decode the matching stream(s). The allowed values
-of @var{hwaccel} are:
-@table @option
-@item none
-Do not use any hardware acceleration (the default).
-
-@item auto
-Automatically select the hardware acceleration method.
-
-@item vda
-Use Apple VDA hardware acceleration.
-
-@item vdpau
-Use VDPAU (Video Decode and Presentation API for Unix) hardware acceleration.
-
-@item dxva2
-Use DXVA2 (DirectX Video Acceleration) hardware acceleration.
-
-@item qsv
-Use the Intel QuickSync Video acceleration for video transcoding.
-
-Unlike most other values, this option does not enable accelerated decoding (that
-is used automatically whenever a qsv decoder is selected), but accelerated
-transcoding, without copying the frames into the system memory.
-
-For it to work, both the decoder and the encoder must support QSV acceleration
-and no filters must be used.
-@end table
-
-This option has no effect if the selected hwaccel is not available or not
-supported by the chosen decoder.
-
-Note that most acceleration methods are intended for playback and will not be
-faster than software decoding on modern CPUs. Additionally, @command{ffmpeg}
-will usually need to copy the decoded frames from the GPU memory into the system
-memory, resulting in further performance loss. This option is thus mainly
-useful for testing.
-
-@item -hwaccel_device[:@var{stream_specifier}] @var{hwaccel_device} (@emph{input,per-stream})
-Select a device to use for hardware acceleration.
-
-This option only makes sense when the @option{-hwaccel} option is also
-specified. Its exact meaning depends on the specific hardware acceleration
-method chosen.
-
-@table @option
-@item vdpau
-For VDPAU, this option specifies the X11 display/screen to use. If this option
-is not specified, the value of the @var{DISPLAY} environment variable is used
-
-@item dxva2
-For DXVA2, this option should contain the number of the display adapter to use.
-If this option is not specified, the default adapter is used.
-
-@item qsv
-For QSV, this option corresponds to the valus of MFX_IMPL_* . Allowed values
-are:
-@table @option
-@item auto
-@item sw
-@item hw
-@item auto_any
-@item hw_any
-@item hw2
-@item hw3
-@item hw4
-@end table
-@end table
-
-@item -hwaccels
-List all hardware acceleration methods supported in this build of ffmpeg.
-
@end table

@section Audio Options

@table @option
@item -aframes @var{number} (@emph{output})
-Set the number of audio frames to output. This is an alias for @code{-frames:a}.
+Set the number of audio frames to record. This is an alias for @code{-frames:a}.
@item -ar[:@var{stream_specifier}] @var{freq} (@emph{input/output,per-stream})
 Set the audio sampling frequency. For output streams it is set by
 default to the frequency of the corresponding input stream. For input
@@ -771,7 +644,7 @@ filter the stream.
 This is an alias for @code{-filter:a}, see the @ref{filter_option,,-filter option}.
@end table

-@section Advanced Audio options
+@section Advanced Audio options:

@table @option
@item -atag @var{fourcc/tag} (@emph{output})
@@ -786,9 +659,11 @@ stereo but not 6 channels as 5.1. The default is to always try to guess. Use
 0 to disable all guessing.
@end table

-@section Subtitle options
+@section Subtitle options:

@table @option
+@item -slang @var{code}
+Set the ISO 639 language code (3 letters) of the current subtitle stream.
@item -scodec @var{codec} (@emph{input/output})
 Set the subtitle codec. This is an alias for @code{-codec:s}.
@item -sn (@emph{output})
@@ -797,7 +672,7 @@ Disable subtitle recording.
 Deprecated, see -bsf
@end table

-@section Advanced Subtitle options
+@section Advanced Subtitle options:

@table @option

@@ -875,21 +750,8 @@ To map all the streams except the second audio, use negative mappings
 ffmpeg -i INPUT -map 0 -map -0:a:1 OUTPUT
@end example

-To pick the English audio stream:
-@example
-ffmpeg -i INPUT -map 0:m:language:eng OUTPUT
-@end example
-
 Note that using this option disables the default mappings for this output file.

-@item -ignore_unknown
-Ignore input streams with unknown type instead of failing if copying
-such streams is attempted.
-
-@item -copy_unknown
-Allow input streams with unknown type to be copied instead of failing if copying
-such streams is attempted.
-
@item -map_channel [@var{input_file_id}.@var{stream_specifier}.@var{channel_id}|-1][:@var{output_file_id}.@var{stream_specifier}]
 Map an audio channel from a given input to an output. If
@var{output_file_id}.@var{stream_specifier} is not set, the audio channel will
@@ -1009,12 +871,13 @@ Dump each input packet to stderr.
 When dumping packets, also dump the payload.
@item -re (@emph{input})
 Read input at native frame rate. Mainly used to simulate a grab device.
-or live input stream (e.g. when reading from a file). Should not be used
-with actual grab devices or live input streams (where it can cause packet
-loss).
 By default @command{ffmpeg} attempts to read the input(s) as fast as possible.
 This option will slow down the reading of the input(s) to the native frame rate
-of the input(s). It is useful for real-time output (e.g. live streaming).
+of the input(s). It is useful for real-time output (e.g. live streaming). If
+your input(s) is coming from some other live streaming source (through HTTP or
+UDP for example) the server might already be in real-time, thus the option will
+likely not be required. On the other hand, this is meaningful if your input(s)
+is a file you are trying to push in real-time.
@item -loop_input
 Loop over the input stream. Currently it works only for image
 streams. This option is used for automatic FFserver testing.
@@ -1053,13 +916,6 @@ With -map you can select from which stream the timestamps should be
 taken. You can leave either video or audio unchanged and sync the
 remaining stream(s) to the unchanged one.

-@item -frame_drop_threshold @var{parameter}
-Frame drop threshold, which specifies how much behind video frames can
-be before they are dropped. In frame rate units, so 1.0 is one frame.
-The default is -1.1. One possible usecase is to avoid framedrops in case
-of noisy timestamps or to increase frame drop precision in case of exact
-timestamps.
-
@item -async @var{samples_per_second}
 Audio sync method. "Stretches/squeezes" the audio stream to match the timestamps,
 the parameter is the maximum samples per second by which the audio is changed.
@@ -1082,12 +938,6 @@ processing (e.g. in case the format option @option{avoid_negative_ts}
 is enabled) the output timestamps may mismatch with the input
 timestamps even when this option is selected.

-@item -start_at_zero
-When used with @option{copyts}, shift input timestamps so they start at zero.
-
-This means that using e.g. @code{-ss 50} will make output timestamps start at
-50 seconds, regardless of what timestamp the input file started at.
-
@item -copytb @var{mode}
 Specify how to set the encoder timebase when stream copying.  @var{mode} is an
 integer numeric value, and can assume one of the following values:
@@ -1143,7 +993,7 @@ ffmpeg -i h264.mp4 -c:v copy -bsf:v h264_mp4toannexb -an out.h264
 ffmpeg -i file.mov -an -vn -bsf:s mov2textsub -c:s copy -f rawvideo sub.txt
@end example

-@item -tag[:@var{stream_specifier}] @var{codec_tag} (@emph{input/output,per-stream})
+@item -tag[:@var{stream_specifier}] @var{codec_tag} (@emph{per-stream})
 Force a tag/fourcc for matching streams.

@item -timecode @var{hh}:@var{mm}:@var{ss}SEP@var{ff}
@@ -1210,74 +1060,13 @@ This option is similar to @option{-filter_complex}, the only difference is that
 its argument is the name of the file from which a complex filtergraph
 description is to be read.

-@item -accurate_seek (@emph{input})
-This option enables or disables accurate seeking in input files with the
-@option{-ss} option. It is enabled by default, so seeking is accurate when
-transcoding. Use @option{-noaccurate_seek} to disable it, which may be useful
-e.g. when copying some streams and transcoding the others.
-
-@item -seek_timestamp (@emph{input})
-This option enables or disables seeking by timestamp in input files with the
-@option{-ss} option. It is disabled by default. If enabled, the argument
-to the @option{-ss} option is considered an actual timestamp, and is not
-offset by the start time of the file. This matters only for files which do
-not start from timestamp 0, such as transport streams.
-
-@item -thread_queue_size @var{size} (@emph{input})
-This option sets the maximum number of queued packets when reading from the
-file or device. With low latency / high rate live streams, packets may be
-discarded if they are not read in a timely manner; raising this value can
-avoid it.
-
@item -override_ffserver (@emph{global})
-Overrides the input specifications from @command{ffserver}. Using this
-option you can map any input stream to @command{ffserver} and control
-many aspects of the encoding from @command{ffmpeg}. Without this
-option @command{ffmpeg} will transmit to @command{ffserver} what is
-requested by @command{ffserver}.
-
+Overrides the input specifications from ffserver. Using this option you can
+map any input stream to ffserver and control many aspects of the encoding from
+ffmpeg. Without this option ffmpeg will transmit to ffserver what is requested by
+ffserver.
 The option is intended for cases where features are needed that cannot be
-specified to @command{ffserver} but can be to @command{ffmpeg}.
-
-@item -sdp_file @var{file} (@emph{global})
-Print sdp information for an output stream to @var{file}.
-This allows dumping sdp information when at least one output isn't an
-rtp stream. (Requires at least one of the output formats to be rtp).
-
-@item -discard (@emph{input})
-Allows discarding specific streams or frames of streams at the demuxer.
-Not all demuxers support this.
-
-@table @option
-@item none
-Discard no frame.
-
-@item default
-Default, which discards no frames.
-
-@item noref
-Discard all non-reference frames.
-
-@item bidir
-Discard all bidirectional frames.
-
-@item nokey
-Discard all frames excepts keyframes.
-
-@item all
-Discard all frames.
-@end table
-
-@item -abort_on @var{flags} (@emph{global})
-Stop and abort on various conditions. The following flags are available:
-
-@table @option
-@item empty_output
-No packets were passed to the muxer, the output is empty.
-@end table
-
-@item -xerror (@emph{global})
-Stop and exit on error
+specified to ffserver but can be to ffmpeg.

@end table

@@ -1304,10 +1093,7 @@ awkward to specify on the command line. Lines starting with the hash
 ('#') character are ignored and are used to provide comments. Check
 the @file{presets} directory in the FFmpeg source tree for examples.

-There are two types of preset files: ffpreset and avpreset files.
-
-@subsection ffpreset files
-ffpreset files are specified with the @code{vpre}, @code{apre},
+Preset files are specified with the @code{vpre}, @code{apre},
@code{spre}, and @code{fpre} options. The @code{fpre} option takes the
 filename of the preset instead of a preset name as input and can be
 used for any kind of codec. For the @code{vpre}, @code{apre}, and
@@ -1332,31 +1118,67 @@ directories, where @var{codec_name} is the name of the codec to which
 the preset file options will be applied. For example, if you select
 the video codec with @code{-vcodec libvpx} and use @code{-vpre 1080p},
 then it will search for the file @file{libvpx-1080p.ffpreset}.
-
-@subsection avpreset files
-avpreset files are specified with the @code{pre} option. They work similar to
-ffpreset files, but they only allow encoder- specific options. Therefore, an
-@var{option}=@var{value} pair specifying an encoder cannot be used.
-
-When the @code{pre} option is specified, ffmpeg will look for files with the
-suffix .avpreset in the directories @file{$AVCONV_DATADIR} (if set), and
-@file{$HOME/.avconv}, and in the datadir defined at configuration time (usually
-@file{PREFIX/share/ffmpeg}), in that order.
-
-First ffmpeg searches for a file named @var{codec_name}-@var{arg}.avpreset in
-the above-mentioned directories, where @var{codec_name} is the name of the codec
-to which the preset file options will be applied. For example, if you select the
-video codec with @code{-vcodec libvpx} and use @code{-pre 1080p}, then it will
-search for the file @file{libvpx-1080p.avpreset}.
-
-If no such file is found, then ffmpeg will search for a file named
-@var{arg}.avpreset in the same directories.
-
@c man end OPTIONS

+@chapter Tips
+@c man begin TIPS
+
+@itemize
+@item
+For streaming at very low bitrate application, use a low frame rate
+and a small GOP size. This is especially true for RealVideo where
+the Linux player does not seem to be very fast, so it can miss
+frames. An example is:
+
+@example
+ffmpeg -g 3 -r 3 -t 10 -b:v 50k -s qcif -f rv10 /tmp/b.rm
+@end example
+
+@item
+The parameter 'q' which is displayed while encoding is the current
+quantizer. The value 1 indicates that a very good quality could
+be achieved. The value 31 indicates the worst quality. If q=31 appears
+too often, it means that the encoder cannot compress enough to meet
+your bitrate. You must either increase the bitrate, decrease the
+frame rate or decrease the frame size.
+
+@item
+If your computer is not fast enough, you can speed up the
+compression at the expense of the compression ratio. You can use
+'-me zero' to speed up motion estimation, and '-g 0' to disable
+motion estimation completely (you have only I-frames, which means it
+is about as good as JPEG compression).
+
+@item
+To have very low audio bitrates, reduce the sampling frequency
+(down to 22050 Hz for MPEG audio, 22050 or 11025 for AC-3).
+
+@item
+To have a constant quality (but a variable bitrate), use the option
+'-qscale n' when 'n' is between 1 (excellent quality) and 31 (worst
+quality).
+
+@end itemize
+@c man end TIPS
+
@chapter Examples
@c man begin EXAMPLES

+@section Preset files
+
+A preset file contains a sequence of @var{option=value} pairs, one for
+each line, specifying a sequence of options which can be specified also on
+the command line. Lines starting with the hash ('#') character are ignored and
+are used to provide comments. Empty lines are also ignored. Check the
+@file{presets} directory in the FFmpeg source tree for examples.
+
+Preset files are specified with the @code{pre} option, this option takes a
+preset name as input.  FFmpeg searches for a file named @var{preset_name}.avpreset in
+the directories @file{$AVCONV_DATADIR} (if set), and @file{$HOME/.ffmpeg}, and in
+the data directory defined at configuration time (usually @file{$PREFIX/share/ffmpeg})
+in that order.  For example, if the argument is @code{libx264-max}, it will
+search for the file @file{libx264-max.avpreset}.
+
@section Video and Audio grabbing

 If you specify the input format and device then ffmpeg can grab video
@@ -1382,14 +1204,14 @@ standard mixer.
 Grab the X11 display with ffmpeg via

@example
-ffmpeg -f x11grab -video_size cif -framerate 25 -i :0.0 /tmp/out.mpg
+ffmpeg -f x11grab -s cif -r 25 -i :0.0 /tmp/out.mpg
@end example

 0.0 is display.screen number of your X11 server, same as
 the DISPLAY environment variable.

@example
-ffmpeg -f x11grab -video_size cif -framerate 25 -i :0.0+10,20 /tmp/out.mpg
+ffmpeg -f x11grab -s cif -r 25 -i :0.0+10,20 /tmp/out.mpg
@end example

 0.0 is display.screen number of your X11 server, same as the DISPLAY environment
@@ -1504,7 +1326,7 @@ combination with -ss to start extracting from a certain point in time.

 For creating a video from many images:
@example
-ffmpeg -f image2 -framerate 12 -i foo-%03d.jpeg -s WxH foo.avi
+ffmpeg -f image2 -i foo-%03d.jpeg -r 12 -s WxH foo.avi
@end example

 The syntax @code{foo-%03d.jpeg} specifies to use a decimal number
@@ -1519,18 +1341,18 @@ image2-specific @code{-pattern_type glob} option.
 For example, for creating a video from filenames matching the glob pattern
@code{foo-*.jpeg}:
@example
-ffmpeg -f image2 -pattern_type glob -framerate 12 -i 'foo-*.jpeg' -s WxH foo.avi
+ffmpeg -f image2 -pattern_type glob -i 'foo-*.jpeg' -r 12 -s WxH foo.avi
@end example

@item
 You can put many streams of the same type in the output:

@example
-ffmpeg -i test1.avi -i test2.avi -map 1:1 -map 1:0 -map 0:1 -map 0:0 -c copy -y test12.nut
+ffmpeg -i test1.avi -i test2.avi -map 0:3 -map 0:2 -map 0:1 -map 0:0 -c copy test12.nut
@end example

-The resulting output file @file{test12.nut} will contain the first four streams
-from the input files in reverse order.
+The resulting output file @file{test12.avi} will contain first four streams from
+the input file in reverse order.

@item
 To force CBR video output:
--- a/doc/ffplay.texi
+++ b/doc/ffplay.texi
@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8

@settitle ffplay Documentation
@titlepage
@@ -25,7 +24,7 @@ various FFmpeg APIs.
@chapter Options
@c man begin OPTIONS

-@include fftools-common-opts.texi
+@include avtools-common-opts.texi

@section Main options

@@ -38,26 +37,14 @@ Force displayed height.
 Set frame size (WxH or abbreviation), needed for videos which do
 not contain a header with the frame size like raw YUV.  This option
 has been deprecated in favor of private options, try -video_size.
-@item -fs
-Start in fullscreen mode.
@item -an
 Disable audio.
@item -vn
 Disable video.
-@item -sn
-Disable subtitles.
@item -ss @var{pos}
-Seek to @var{pos}. Note that in most formats it is not possible to seek
-exactly, so @command{ffplay} will seek to the nearest seek point to
-@var{pos}.
-
-@var{pos} must be a time duration specification,
-see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
+Seek to a given position in seconds.
@item -t @var{duration}
-Play @var{duration} seconds of audio/video.
-
-@var{duration} must be a time duration specification,
-see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
+play <duration> seconds of audio/video
@item -bytes
 Seek by bytes.
@item -nodisp
@@ -97,9 +84,6 @@ output. In the filtergraph, the input is associated to the label
 ffmpeg-filters manual for more information about the filtergraph
 syntax.

-You can specify this parameter multiple times and cycle through the specified
-filtergraphs along with the show modes by pressing the key @key{w}.
-
@item -af @var{filtergraph}
@var{filtergraph} is a description of the filtergraph to apply to
 the input audio.
@@ -122,10 +106,15 @@ duration, the codec parameters, the current position in the stream and
 the audio/video synchronisation drift. It is on by default, to
 explicitly disable it you need to specify @code{-nostats}.

+@item -bug
+Work around bugs.
@item -fast
 Non-spec-compliant optimizations.
@item -genpts
 Generate pts.
+@item -rtp_tcp
+Force RTP/TCP protocol usage instead of RTP/UDP. It is only meaningful
+if you are streaming with the RTSP protocol.
@item -sync @var{type}
 Set the master clock to audio (@code{type=audio}), video
 (@code{type=video}) or external (@code{type=ext}). Default is audio. The
@@ -133,20 +122,23 @@ master clock is used to control audio-video synchronization. Most media
 players use audio as master clock, but in some cases (streaming or high
 quality broadcast) it is necessary to change that. This option is mainly
 used for debugging purposes.
-@item -ast @var{audio_stream_specifier}
-Select the desired audio stream using the given stream specifier. The stream
-specifiers are described in the @ref{Stream specifiers} chapter. If this option
-is not specified, the "best" audio stream is selected in the program of the
-already selected video stream.
-@item -vst @var{video_stream_specifier}
-Select the desired video stream using the given stream specifier. The stream
-specifiers are described in the @ref{Stream specifiers} chapter. If this option
-is not specified, the "best" video stream is selected.
-@item -sst @var{subtitle_stream_specifier}
-Select the desired subtitle stream using the given stream specifier. The stream
-specifiers are described in the @ref{Stream specifiers} chapter. If this option
-is not specified, the "best" subtitle stream is selected in the program of the
-already selected video or audio stream.
+@item -threads @var{count}
+Set the thread count.
+@item -ast @var{audio_stream_number}
+Select the desired audio stream number, counting from 0. The number
+refers to the list of all the input audio streams. If it is greater
+than the number of audio streams minus one, then the last one is
+selected, if it is negative the audio playback is disabled.
+@item -vst @var{video_stream_number}
+Select the desired video stream number, counting from 0. The number
+refers to the list of all the input video streams. If it is greater
+than the number of video streams minus one, then the last one is
+selected, if it is negative the video playback is disabled.
+@item -sst @var{subtitle_stream_number}
+Select the desired subtitle stream number, counting from 0. The number
+refers to the list of all the input subtitle streams. If it is greater
+than the number of subtitle streams minus one, then the last one is
+selected, if it is negative the subtitle rendering is disabled.
@item -autoexit
 Exit when video is done playing.
@item -exitonkeydown
@@ -167,22 +159,6 @@ Force a specific video decoder.

@item -scodec @var{codec_name}
 Force a specific subtitle decoder.
-
-@item -autorotate
-Automatically rotate the video according to file metadata. Enabled by
-default, use @option{-noautorotate} to disable it.
-
-@item -framedrop
-Drop video frames if video is out of sync. Enabled by default if the master
-clock is not set to video. Use this option to enable frame dropping for all
-master clock sources, use @option{-noframedrop} to disable it.
-
-@item -infbuf
-Do not limit the input buffer size, read as much data as possible from the
-input as soon as possible. Enabled by default for realtime streams, where data
-may be dropped if not read in time. Use this option to enable infinite buffers
-for all inputs, use @option{-noinfbuf} to disable it.
-
@end table

@section While playing
@@ -197,35 +173,17 @@ Toggle full screen.
@item p, SPC
 Pause.

-@item m
-Toggle mute.
-
-@item 9, 0
-Decrease and increase volume respectively.
-
-@item /, *
-Decrease and increase volume respectively.
-
@item a
-Cycle audio channel in the current program.
+Cycle audio channel.

@item v
 Cycle video channel.

@item t
-Cycle subtitle channel in the current program.
-
-@item c
-Cycle program.
+Cycle subtitle channel.

@item w
-Cycle video filters or show modes.
-
-@item s
-Step to the next frame.
-
-Pause if the stream is not already paused, step to the next video
-frame, and pause.
+Show audio waves.

@item left/right
 Seek backward/forward 10 seconds.
@@ -234,23 +192,17 @@ Seek backward/forward 10 seconds.
 Seek backward/forward 1 minute.

@item page down/page up
-Seek to the previous/next chapter.
-or if there are no chapters
 Seek backward/forward 10 minutes.

-@item right mouse click
+@item mouse click
 Seek to percentage in file corresponding to fraction of width.

-@item left mouse double-click
-Toggle full screen.
-
@end table

@c man end

@include config.texi
@ifset config-all
-@set config-readonly
@ifset config-avutil
@include utils.texi
@end ifset
--- a/doc/ffprobe.texi
+++ b/doc/ffprobe.texi
@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8

@settitle ffprobe Documentation
@titlepage
@@ -45,15 +44,14 @@ name (which may be shared by other sections), and an unique
 name. See the output of @option{sections}.

 Metadata tags stored in the container or in the streams are recognized
-and printed in the corresponding "FORMAT", "STREAM" or "PROGRAM_STREAM"
-section.
+and printed in the corresponding "FORMAT" or "STREAM" section.

@c man end

@chapter Options
@c man begin OPTIONS

-@include fftools-common-opts.texi
+@include avtools-common-opts.texi

@section Main options

@@ -114,16 +112,12 @@ ffprobe -show_packets -select_streams v:1 INPUT
@end example

@item -show_data
-Show payload data, as a hexadecimal and ASCII dump. Coupled with
+Show payload data, as an hexadecimal and ASCII dump. Coupled with
@option{-show_packets}, it will dump the packets' data. Coupled with
@option{-show_streams}, it will dump the codec extradata.

 The dump is printed as the "data" field. It may contain newlines.

-@item -show_data_hash @var{algorithm}
-Show a hash of payload data, for packets with @option{-show_packets} and for
-codec extradata with @option{-show_streams}.
-
@item -show_error
 Show information about the error found when trying to probe the input.

@@ -185,7 +179,7 @@ format : stream=codec_type

 To show all the tags in the stream and format sections:
@example
-stream_tags : format_tags
+format_tags : format_tags
@end example

 To show only the @code{title} tag (if available) in the stream
@@ -202,11 +196,11 @@ The information for each single packet is printed within a dedicated
 section with name "PACKET".

@item -show_frames
-Show information about each frame and subtitle contained in the input
-multimedia stream.
+Show information about each frame contained in the input multimedia
+stream.

 The information for each single frame is printed within a dedicated
-section with name "FRAME" or "SUBTITLE".
+section with name "FRAME".

@item -show_streams
 Show information about each media stream contained in the input
@@ -215,13 +209,6 @@ multimedia stream.
 Each media stream information is printed within a dedicated section
 with name "STREAM".

-@item -show_programs
-Show information about programs and their streams contained in the input
-multimedia stream.
-
-Each media stream information is printed within a dedicated section
-with name "PROGRAM_STREAM".
-
@item -show_chapters
 Show information about chapters stored in the format.

@@ -235,70 +222,6 @@ corresponding stream section.
 Count the number of packets per stream and report it in the
 corresponding stream section.

-@item -read_intervals @var{read_intervals}
-
-Read only the specified intervals. @var{read_intervals} must be a
-sequence of interval specifications separated by ",".
-@command{ffprobe} will seek to the interval starting point, and will
-continue reading from that.
-
-Each interval is specified by two optional parts, separated by "%".
-
-The first part specifies the interval start position. It is
-interpreted as an abolute position, or as a relative offset from the
-current position if it is preceded by the "+" character. If this first
-part is not specified, no seeking will be performed when reading this
-interval.
-
-The second part specifies the interval end position. It is interpreted
-as an absolute position, or as a relative offset from the current
-position if it is preceded by the "+" character. If the offset
-specification starts with "#", it is interpreted as the number of
-packets to read (not including the flushing packets) from the interval
-start. If no second part is specified, the program will read until the
-end of the input.
-
-Note that seeking is not accurate, thus the actual interval start
-point may be different from the specified position. Also, when an
-interval duration is specified, the absolute end time will be computed
-by adding the duration to the interval start point found by seeking
-the file, rather than to the specified start value.
-
-The formal syntax is given by:
-@example
-@var{INTERVAL}  ::= [@var{START}|+@var{START_OFFSET}][%[@var{END}|+@var{END_OFFSET}]]
-@var{INTERVALS} ::= @var{INTERVAL}[,@var{INTERVALS}]
-@end example
-
-A few examples follow.
-@itemize
-@item
-Seek to time 10, read packets until 20 seconds after the found seek
-point, then seek to position @code{01:30} (1 minute and thirty
-seconds) and read packets until position @code{01:45}.
-@example
-10%+20,01:30%01:45
-@end example
-
-@item
-Read only 42 packets after seeking to position @code{01:23}:
-@example
-01:23%+#42
-@end example
-
-@item
-Read only the first 20 seconds from the start:
-@example
-%+20
-@end example
-
-@item
-Read from the start until position @code{02:30}:
-@example
-%02:30
-@end example
-@end itemize
-
@item -show_private_data, -private
 Show private data, that is data depending on the format of the
 particular shown element.
@@ -322,12 +245,6 @@ Show information related to program and library versions. This is the
 equivalent of setting both @option{-show_program_version} and
@option{-show_library_versions} options.

-@item -show_pixel_formats
-Show information about all pixel formats supported by FFmpeg.
-
-Pixel format information for each format is printed within a section
-with name "PIXEL_FORMAT".
-
@item -bitexact
 Force bitexact output, useful to produce output which is not dependent
 on the specific build.
@@ -348,39 +265,6 @@ A writer may accept one or more arguments, which specify the options
 to adopt. The options are specified as a list of @var{key}=@var{value}
 pairs, separated by ":".

-All writers support the following options:
-
-@table @option
-@item string_validation, sv
-Set string validation mode.
-
-The following values are accepted.
-@table @samp
-@item fail
-The writer will fail immediately in case an invalid string (UTF-8)
-sequence or code point is found in the input. This is especially
-useful to validate input metadata.
-
-@item ignore
-Any validation error will be ignored. This will result in possibly
-broken output, especially with the json or xml writer.
-
-@item replace
-The writer will substitute invalid UTF-8 sequences or code points with
-the string specified with the @option{string_validation_replacement}.
-@end table
-
-Default value is @samp{replace}.
-
-@item string_validation_replacement, svr
-Set replacement string to use in case @option{string_validation} is
-set to @samp{replace}.
-
-In case the option is not specified, the writer will assume the empty
-string, that is it will remove the invalid sequences from the input
-strings.
-@end table
-
 A description of the currently available writers follows.

@section default
@@ -395,8 +279,8 @@ keyN=valN
 [/SECTION]
@end example

-Metadata tags are printed as a line in the corresponding FORMAT, STREAM or
-PROGRAM_STREAM section, and are prefixed by the string "TAG:".
+Metadata tags are printed as a line in the corresponding FORMAT or
+STREAM section, and are prefixed by the string "TAG:".

 A description of the accepted options follows.

@@ -447,17 +331,17 @@ writer).
 It can assume one of the following values:
@table @option
@item c
-Perform C-like escaping. Strings containing a newline (@samp{\n}), carriage
-return (@samp{\r}), a tab (@samp{\t}), a form feed (@samp{\f}), the escaping
-character (@samp{\}) or the item separator character @var{SEP} are escaped
-using C-like fashioned escaping, so that a newline is converted to the
-sequence @samp{\n}, a carriage return to @samp{\r}, @samp{\} to @samp{\\} and
-the separator @var{SEP} is converted to @samp{\@var{SEP}}.
+Perform C-like escaping. Strings containing a newline ('\n'), carriage
+return ('\r'), a tab ('\t'), a form feed ('\f'), the escaping
+character ('\') or the item separator character @var{SEP} are escaped using C-like fashioned
+escaping, so that a newline is converted to the sequence "\n", a
+carriage return to "\r", '\' to "\\" and the separator @var{SEP} is
+converted to "\@var{SEP}".

@item csv
 Perform CSV-like escaping, as described in RFC4180.  Strings
-containing a newline (@samp{\n}), a carriage return (@samp{\r}), a double quote
-(@samp{"}), or @var{SEP} are enclosed in double-quotes.
+containing a newline ('\n'), a carriage return ('\r'), a double quote
+('"'), or @var{SEP} are enclosed in double-quotes.

@item none
 Perform no escaping.
@@ -485,7 +369,7 @@ The description of the accepted options follows.
 Separator character used to separate the chapter, the section name, IDs and
 potential tags in the printed field key.

-Default value is @samp{.}.
+Default value is '.'.

@item hierarchical, h
 Specify if the section name specification should be hierarchical. If
@@ -507,22 +391,21 @@ The following conventions are adopted:
@item
 all key and values are UTF-8
@item
-@samp{.} is the subgroup separator
+'.' is the subgroup separator
@item
-newline, @samp{\t}, @samp{\f}, @samp{\b} and the following characters are
-escaped
+newline, '\t', '\f', '\b' and the following characters are escaped
@item
-@samp{\} is the escape character
+'\' is the escape character
@item
-@samp{#} is the comment indicator
+'#' is the comment indicator
@item
-@samp{=} is the key/value separator
+'=' is the key/value separator
@item
-@samp{:} is not used but usually parsed as key/value separator
+':' is not used but usually parsed as key/value separator
@end itemize

 This writer accepts options as a list of @var{key}=@var{value} pairs,
-separated by @samp{:}.
+separated by ":".

 The description of the accepted options follows.

@@ -611,7 +494,6 @@ DV, GXF and AVI timecodes are available in format metadata

@include config.texi
@ifset config-all
-@set config-readonly
@ifset config-avutil
@include utils.texi
@end ifset
--- a/doc/ffprobe.xsd
+++ b/doc/ffprobe.xsd
@@ -8,17 +8,14 @@

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

@@ -30,29 +27,11 @@

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

    <xsd:complexType name="packetType">
-      <xsd:sequence>
-        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
-        <xsd:element name="side_data_list" type="ffprobe:packetSideDataListType"   minOccurs="0" maxOccurs="1" />
-      </xsd:sequence>
-
      <xsd:attribute name="codec_type"    type="xsd:string" use="required" />
      <xsd:attribute name="stream_index"  type="xsd:int" use="required" />
      <xsd:attribute name="pts"           type="xsd:long"  />
@@ -67,27 +46,10 @@
      <xsd:attribute name="pos"           type="xsd:long"  />
      <xsd:attribute name="flags"         type="xsd:string" use="required" />
      <xsd:attribute name="data"          type="xsd:string" />
-      <xsd:attribute name="data_hash"     type="xsd:string" />
-    </xsd:complexType>
-
-    <xsd:complexType name="packetSideDataListType">
-        <xsd:sequence>
-            <xsd:element name="side_data" type="ffprobe:packetSideDataType" minOccurs="1" maxOccurs="unbounded"/>
-        </xsd:sequence>
-    </xsd:complexType>
-    <xsd:complexType name="packetSideDataType">
-        <xsd:attribute name="side_data_type"              type="xsd:string"/>
-        <xsd:attribute name="side_data_size"              type="xsd:int"   />
    </xsd:complexType>

    <xsd:complexType name="frameType">
-      <xsd:sequence>
-            <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
-            <xsd:element name="side_data_list" type="ffprobe:frameSideDataListType"   minOccurs="0" maxOccurs="1" />
-      </xsd:sequence>
-
      <xsd:attribute name="media_type"    type="xsd:string" use="required"/>
-      <xsd:attribute name="stream_index"  type="xsd:int"  />
      <xsd:attribute name="key_frame"     type="xsd:int"    use="required"/>
      <xsd:attribute name="pts"           type="xsd:long" />
      <xsd:attribute name="pts_time"      type="xsd:float"/>
@@ -95,8 +57,6 @@
      <xsd:attribute name="pkt_pts_time"  type="xsd:float"/>
      <xsd:attribute name="pkt_dts"       type="xsd:long" />
      <xsd:attribute name="pkt_dts_time"  type="xsd:float"/>
-      <xsd:attribute name="best_effort_timestamp"      type="xsd:long" />
-      <xsd:attribute name="best_effort_timestamp_time" type="xsd:float" />
      <xsd:attribute name="pkt_duration"  type="xsd:long" />
      <xsd:attribute name="pkt_duration_time" type="xsd:float"/>
      <xsd:attribute name="pkt_pos"       type="xsd:long" />
@@ -121,38 +81,12 @@
      <xsd:attribute name="repeat_pict"            type="xsd:int"   />
    </xsd:complexType>

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

-    <xsd:complexType name="programsType">
-        <xsd:sequence>
-            <xsd:element name="program" type="ffprobe:programType" minOccurs="0" maxOccurs="unbounded"/>
-        </xsd:sequence>
-    </xsd:complexType>
-
    <xsd:complexType name="streamDispositionType">
      <xsd:attribute name="default"          type="xsd:int" use="required" />
      <xsd:attribute name="dub"              type="xsd:int" use="required" />
@@ -169,9 +103,8 @@

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

      <xsd:attribute name="index"            type="xsd:int" use="required"/>
@@ -183,31 +116,21 @@
      <xsd:attribute name="codec_tag"        type="xsd:string" use="required"/>
      <xsd:attribute name="codec_tag_string" type="xsd:string" use="required"/>
      <xsd:attribute name="extradata"        type="xsd:string" />
-      <xsd:attribute name="extradata_hash"   type="xsd:string" />

      <!-- video attributes -->
      <xsd:attribute name="width"                type="xsd:int"/>
      <xsd:attribute name="height"               type="xsd:int"/>
-      <xsd:attribute name="coded_width"          type="xsd:int"/>
-      <xsd:attribute name="coded_height"         type="xsd:int"/>
      <xsd:attribute name="has_b_frames"         type="xsd:int"/>
      <xsd:attribute name="sample_aspect_ratio"  type="xsd:string"/>
      <xsd:attribute name="display_aspect_ratio" type="xsd:string"/>
      <xsd:attribute name="pix_fmt"              type="xsd:string"/>
      <xsd:attribute name="level"                type="xsd:int"/>
-      <xsd:attribute name="color_range"          type="xsd:string"/>
-      <xsd:attribute name="color_space"          type="xsd:string"/>
-      <xsd:attribute name="color_transfer"       type="xsd:string"/>
-      <xsd:attribute name="color_primaries"      type="xsd:string"/>
-      <xsd:attribute name="chroma_location"      type="xsd:string"/>
      <xsd:attribute name="timecode"             type="xsd:string"/>
-      <xsd:attribute name="refs"                 type="xsd:int"/>

      <!-- audio attributes -->
      <xsd:attribute name="sample_fmt"       type="xsd:string"/>
      <xsd:attribute name="sample_rate"      type="xsd:int"/>
      <xsd:attribute name="channels"         type="xsd:int"/>
-      <xsd:attribute name="channel_layout"   type="xsd:string"/>
      <xsd:attribute name="bits_per_sample"  type="xsd:int"/>

      <xsd:attribute name="id"               type="xsd:string"/>
@@ -219,30 +142,11 @@
      <xsd:attribute name="duration_ts"      type="xsd:long"/>
      <xsd:attribute name="duration"         type="xsd:float"/>
      <xsd:attribute name="bit_rate"         type="xsd:int"/>
-      <xsd:attribute name="max_bit_rate"     type="xsd:int"/>
-      <xsd:attribute name="bits_per_raw_sample" type="xsd:int"/>
      <xsd:attribute name="nb_frames"        type="xsd:int"/>
      <xsd:attribute name="nb_read_frames"   type="xsd:int"/>
      <xsd:attribute name="nb_read_packets"  type="xsd:int"/>
    </xsd:complexType>

-    <xsd:complexType name="programType">
-      <xsd:sequence>
-        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
-        <xsd:element name="streams" type="ffprobe:streamsType" minOccurs="0" maxOccurs="1"/>
-      </xsd:sequence>
-
-      <xsd:attribute name="program_id"           type="xsd:int"    use="required"/>
-      <xsd:attribute name="program_num"          type="xsd:int"    use="required"/>
-      <xsd:attribute name="nb_streams"           type="xsd:int"    use="required"/>
-      <xsd:attribute name="start_time"           type="xsd:float"/>
-      <xsd:attribute name="start_pts"            type="xsd:long"/>
-      <xsd:attribute name="end_time"             type="xsd:float"/>
-      <xsd:attribute name="end_pts"              type="xsd:long"/>
-      <xsd:attribute name="pmt_pid"              type="xsd:int"    use="required"/>
-      <xsd:attribute name="pcr_pid"              type="xsd:int"    use="required"/>
-    </xsd:complexType>
-
    <xsd:complexType name="formatType">
      <xsd:sequence>
        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
@@ -250,14 +154,12 @@

      <xsd:attribute name="filename"         type="xsd:string" use="required"/>
      <xsd:attribute name="nb_streams"       type="xsd:int"    use="required"/>
-      <xsd:attribute name="nb_programs"      type="xsd:int"    use="required"/>
      <xsd:attribute name="format_name"      type="xsd:string" use="required"/>
      <xsd:attribute name="format_long_name" type="xsd:string"/>
      <xsd:attribute name="start_time"       type="xsd:float"/>
      <xsd:attribute name="duration"         type="xsd:float"/>
      <xsd:attribute name="size"             type="xsd:long"/>
      <xsd:attribute name="bit_rate"         type="xsd:long"/>
-      <xsd:attribute name="probe_score"      type="xsd:int"/>
    </xsd:complexType>

    <xsd:complexType name="tagType">
@@ -273,9 +175,10 @@
    <xsd:complexType name="programVersionType">
      <xsd:attribute name="version"          type="xsd:string" use="required"/>
      <xsd:attribute name="copyright"        type="xsd:string" use="required"/>
-      <xsd:attribute name="build_date"       type="xsd:string"/>
-      <xsd:attribute name="build_time"       type="xsd:string"/>
-      <xsd:attribute name="compiler_ident"   type="xsd:string" use="required"/>
+      <xsd:attribute name="build_date"       type="xsd:string" use="required"/>
+      <xsd:attribute name="build_time"       type="xsd:string" use="required"/>
+      <xsd:attribute name="compiler_type"    type="xsd:string" use="required"/>
+      <xsd:attribute name="compiler_version" type="xsd:string" use="required"/>
      <xsd:attribute name="configuration"    type="xsd:string" use="required"/>
    </xsd:complexType>

@@ -312,45 +215,4 @@
          <xsd:element name="library_version" type="ffprobe:libraryVersionType" minOccurs="0" maxOccurs="unbounded"/>
        </xsd:sequence>
    </xsd:complexType>
-
-    <xsd:complexType name="pixelFormatFlagsType">
-      <xsd:attribute name="big_endian" type="xsd:int" use="required"/>
-      <xsd:attribute name="palette"    type="xsd:int" use="required"/>
-      <xsd:attribute name="bitstream"  type="xsd:int" use="required"/>
-      <xsd:attribute name="hwaccel"    type="xsd:int" use="required"/>
-      <xsd:attribute name="planar"     type="xsd:int" use="required"/>
-      <xsd:attribute name="rgb"        type="xsd:int" use="required"/>
-      <xsd:attribute name="pseudopal"  type="xsd:int" use="required"/>
-      <xsd:attribute name="alpha"      type="xsd:int" use="required"/>
-    </xsd:complexType>
-
-    <xsd:complexType name="pixelFormatComponentType">
-      <xsd:attribute name="index"      type="xsd:int" use="required"/>
-      <xsd:attribute name="bit_depth"  type="xsd:int" use="required"/>
-    </xsd:complexType>
-
-    <xsd:complexType name="pixelFormatComponentsType">
-      <xsd:sequence>
-        <xsd:element name="component" type="ffprobe:pixelFormatComponentType" minOccurs="0" maxOccurs="unbounded"/>
-      </xsd:sequence>
-    </xsd:complexType>
-
-    <xsd:complexType name="pixelFormatType">
-      <xsd:sequence>
-        <xsd:element name="flags" type="ffprobe:pixelFormatFlagsType" minOccurs="0" maxOccurs="1"/>
-        <xsd:element name="components" type="ffprobe:pixelFormatComponentsType" minOccurs="0" maxOccurs="1"/>
-      </xsd:sequence>
-
-      <xsd:attribute name="name"                type="xsd:string" use="required"/>
-      <xsd:attribute name="nb_components"       type="xsd:int"    use="required"/>
-      <xsd:attribute name="log2_chroma_w"       type="xsd:int"/>
-      <xsd:attribute name="log2_chroma_h"       type="xsd:int"/>
-      <xsd:attribute name="bits_per_pixel"      type="xsd:int"/>
-    </xsd:complexType>
-
-    <xsd:complexType name="pixelFormatsType">
-      <xsd:sequence>
-        <xsd:element name="pixel_format" type="ffprobe:pixelFormatType" minOccurs="0" maxOccurs="unbounded"/>
-      </xsd:sequence>
-    </xsd:complexType>
 </xsd:schema>
--- a/doc/ffserver.conf
+++ b/doc/ffserver.conf
@@ -1,11 +1,11 @@
 # Port on which the server is listening. You must select a different
 # port from your standard HTTP web server if it is running on the same
 # computer.
-HTTPPort 8090
+Port 8090

 # Address on which the server is bound. Only useful if you have
 # several network interfaces.
-HTTPBindAddress 0.0.0.0
+BindAddress 0.0.0.0

 # Number of simultaneous HTTP connections that can be handled. It has
 # to be defined *before* the MaxClients parameter, since it defines the
@@ -82,7 +82,6 @@ Feed feed1.ffm
 # ra         : RealNetworks-compatible stream. Audio only.
 # mpjpeg     : Multipart JPEG (works with Netscape without any plugin)
 # jpeg       : Generate a single JPEG image.
-# mjpeg      : Generate a M-JPEG stream.
 # asf        : ASF compatible streaming (Windows Media Player format).
 # swf        : Macromedia Flash compatible stream
 # avi        : AVI format (MPEG-4 video, MPEG audio sound)
@@ -236,7 +235,7 @@ StartSendOnKey

 #<Stream test.ogg>
 #Feed feed1.ffm
-#Metadata title "Stream title"
+#Title "Stream title"
 #AudioBitRate 64
 #AudioChannels 2
 #AudioSampleRate 44100
@@ -281,10 +280,10 @@ StartSendOnKey
 #<Stream file.asf>
 #File "/usr/local/httpd/htdocs/test.asf"
 #NoAudio
-#Metadata author "Me"
-#Metadata copyright "Super MegaCorp"
-#Metadata title "Test stream from disk"
-#Metadata comment "Test comment"
+#Author "Me"
+#Copyright "Super MegaCorp"
+#Title "Test stream from disk"
+#Comment "Test comment"
 #</Stream>


--- a/doc/ffserver.texi
+++ b/doc/ffserver.texi
@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8

@settitle ffserver Documentation
@titlepage
@@ -17,14 +16,11 @@ ffserver [@var{options}]
@chapter Description
@c man begin DESCRIPTION

-@command{ffserver} is a streaming server for both audio and video.
-It supports several live feeds, streaming from files and time shifting
-on live feeds. You can seek to positions in the past on each live
-feed, provided you specify a big enough feed storage.
-
-@command{ffserver} is configured through a configuration file, which
-is read at startup. If not explicitly specified, it will read from
-@file{/etc/ffserver.conf}.
+@command{ffserver} is a streaming server for both audio and video. It
+supports several live feeds, streaming from files and time shifting on
+live feeds (you can seek to positions in the past on each live feed,
+provided you specify a big enough feed storage in
+@file{ffserver.conf}).

@command{ffserver} receives prerecorded files or FFM streams from some
@command{ffmpeg} instance as input, then streams them over
@@ -43,128 +39,10 @@ For each feed you can have different output streams in various
 formats, each one specified by a @code{<Stream>} section in the
 configuration file.

-@chapter Detailed description
-
-@command{ffserver} works by forwarding streams encoded by
-@command{ffmpeg}, or pre-recorded streams which are read from disk.
-
-Precisely, @command{ffserver} acts as an HTTP server, accepting POST
-requests from @command{ffmpeg} to acquire the stream to publish, and
-serving RTSP clients or HTTP clients GET requests with the stream
-media content.
-
-A feed is an @ref{FFM} stream created by @command{ffmpeg}, and sent to
-a port where @command{ffserver} is listening.
-
-Each feed is identified by a unique name, corresponding to the name
-of the resource published on @command{ffserver}, and is configured by
-a dedicated @code{Feed} section in the configuration file.
-
-The feed publish URL is given by:
-@example
-http://@var{ffserver_ip_address}:@var{http_port}/@var{feed_name}
-@end example
-
-where @var{ffserver_ip_address} is the IP address of the machine where
-@command{ffserver} is installed, @var{http_port} is the port number of
-the HTTP server (configured through the @option{HTTPPort} option), and
-@var{feed_name} is the name of the corresponding feed defined in the
-configuration file.
-
-Each feed is associated to a file which is stored on disk. This stored
-file is used to send pre-recorded data to a player as fast as
-possible when new content is added in real-time to the stream.
-
-A "live-stream" or "stream" is a resource published by
-@command{ffserver}, and made accessible through the HTTP protocol to
-clients.
-
-A stream can be connected to a feed, or to a file. In the first case,
-the published stream is forwarded from the corresponding feed
-generated by a running instance of @command{ffmpeg}, in the second
-case the stream is read from a pre-recorded file.
-
-Each stream is identified by a unique name, corresponding to the name
-of the resource served by @command{ffserver}, and is configured by
-a dedicated @code{Stream} section in the configuration file.
-
-The stream access HTTP URL is given by:
-@example
-http://@var{ffserver_ip_address}:@var{http_port}/@var{stream_name}[@var{options}]
-@end example
-
-The stream access RTSP URL is given by:
-@example
-http://@var{ffserver_ip_address}:@var{rtsp_port}/@var{stream_name}[@var{options}]
-@end example
-
-@var{stream_name} is the name of the corresponding stream defined in
-the configuration file. @var{options} is a list of options specified
-after the URL which affects how the stream is served by
-@command{ffserver}. @var{http_port} and @var{rtsp_port} are the HTTP
-and RTSP ports configured with the options @var{HTTPPort} and
-@var{RTSPPort} respectively.
-
-In case the stream is associated to a feed, the encoding parameters
-must be configured in the stream configuration. They are sent to
-@command{ffmpeg} when setting up the encoding. This allows
-@command{ffserver} to define the encoding parameters used by
-the @command{ffmpeg} encoders.
-
-The @command{ffmpeg} @option{override_ffserver} commandline option
-allows one to override the encoding parameters set by the server.
-
-Multiple streams can be connected to the same feed.
-
-For example, you can have a situation described by the following
-graph:
-
-@verbatim
-               _________       __________
-              |         |     |          |
-ffmpeg 1 -----| feed 1  |-----| stream 1 |
-    \         |_________|\    |__________|
-     \                    \
-      \                    \   __________
-       \                    \ |          |
-        \                    \| stream 2 |
-         \                    |__________|
-          \
-           \   _________       __________
-            \ |         |     |          |
-             \| feed 2  |-----| stream 3 |
-              |_________|     |__________|
-
-               _________       __________
-              |         |     |          |
-ffmpeg 2 -----| feed 3  |-----| stream 4 |
-              |_________|     |__________|
-
-               _________       __________
-              |         |     |          |
-              | file 1  |-----| stream 5 |
-              |_________|     |__________|
-
-@end verbatim
-
-@anchor{FFM}
-@section FFM, FFM2 formats
-
-FFM and FFM2 are formats used by ffserver. They allow storing a wide variety of
-video and audio streams and encoding options, and can store a moving time segment
-of an infinite movie or a whole movie.
-
-FFM is version specific, and there is limited compatibility of FFM files
-generated by one version of ffmpeg/ffserver and another version of
-ffmpeg/ffserver. It may work but it is not guaranteed to work.
-
-FFM2 is extensible while maintaining compatibility and should work between
-differing versions of tools. FFM2 is the default.
-
@section Status stream

-@command{ffserver} supports an HTTP interface which exposes the
-current status of the server.
+ffserver supports an HTTP interface which exposes the current status
+of the server.

 Simply point your browser to the address of the special status stream
 specified in the configuration file.
@@ -183,8 +61,27 @@ ACL allow 192.168.0.0 192.168.255.255
 then the server will post a page with the status information when
 the special stream @file{status.html} is requested.

+@section What can this do?
+
+When properly configured and running, you can capture video and audio in real
+time from a suitable capture card, and stream it out over the Internet to
+either Windows Media Player or RealAudio player (with some restrictions).
+
+It can also stream from files, though that is currently broken. Very often, a
+web server can be used to serve up the files just as well.
+
+It can stream prerecorded video from .ffm files, though it is somewhat tricky
+to make it work correctly.
+
@section How do I make it work?

+First, build the kit. It *really* helps to have installed LAME first. Then when
+you run the ffserver ./configure, make sure that you have the
+@code{--enable-libmp3lame} flag turned on.
+
+LAME is important as it allows for streaming audio to Windows Media Player.
+Don't ask why the other audio types do not work.
+
 As a simple test, just run the following two command lines where INPUTFILE
 is some file which you can decode with ffmpeg:

@@ -206,9 +103,40 @@ WARNING: trying to stream test1.mpg doesn't work with WMP as it tries to
 transfer the entire file before starting to play.
 The same is true of AVI files.

-You should edit the @file{ffserver.conf} file to suit your needs (in
-terms of frame rates etc). Then install @command{ffserver} and
-@command{ffmpeg}, write a script to start them up, and off you go.
+@section What happens next?
+
+You should edit the ffserver.conf file to suit your needs (in terms of
+frame rates etc). Then install ffserver and ffmpeg, write a script to start
+them up, and off you go.
+
+@section Troubleshooting
+
+@subsection I don't hear any audio, but video is fine.
+
+Maybe you didn't install LAME, or got your ./configure statement wrong. Check
+the ffmpeg output to see if a line referring to MP3 is present. If not, then
+your configuration was incorrect. If it is, then maybe your wiring is not
+set up correctly. Maybe the sound card is not getting data from the right
+input source. Maybe you have a really awful audio interface (like I do)
+that only captures in stereo and also requires that one channel be flipped.
+If you are one of these people, then export 'AUDIO_FLIP_LEFT=1' before
+starting ffmpeg.
+
+@subsection The audio and video lose sync after a while.
+
+Yes, they do.
+
+@subsection After a long while, the video update rate goes way down in WMP.
+
+Yes, it does. Who knows why?
+
+@subsection WMP 6.4 behaves differently to WMP 7.
+
+Yes, it does. Any thoughts on this would be gratefully received. These
+differences extend to embedding WMP into a web page. [There are two
+object IDs that you can use: The old one, which does not play well, and
+the new one, which does (both tested on the same system). However,
+I suspect that the new one is not available unless you have installed WMP 7].

@section What else can it do?

@@ -249,6 +177,9 @@ specify a time. In addition, ffserver will skip frames until a key_frame
 is found. This further reduces the startup delay by not transferring data
 that will be discarded.

+* You may want to adjust the MaxBandwidth in the ffserver.conf to limit
+the amount of bandwidth consumed by live streams.
+
@section Why does the ?buffer / Preroll stop working after a time?

 It turns out that (on my machine at least) the number of frames successfully
@@ -282,571 +213,37 @@ You use this by adding the ?date= to the end of the URL for the stream.
 For example:   @samp{http://localhost:8080/test.asf?date=2002-07-26T23:05:00}.
@c man end

+@section What is FFM, FFM2
+
+FFM and FFM2 are formats used by ffserver. They allow storing a wide variety of
+video and audio streams and encoding options, and can store a moving time segment
+of an infinite movie or a whole movie.
+
+FFM is version specific, and there is limited compatibility of FFM files
+generated by one version of ffmpeg/ffserver and another version of
+ffmpeg/ffserver. It may work but it is not guaranteed to work.
+
+FFM2 is extensible while maintaining compatibility and should work between
+differing versions of tools. FFM2 is the default.
+
@chapter Options
@c man begin OPTIONS

-@include fftools-common-opts.texi
+@include avtools-common-opts.texi

@section Main options

@table @option
@item -f @var{configfile}
-Read configuration file @file{configfile}. If not specified it will
-read by default from @file{/etc/ffserver.conf}.
-
+Use @file{configfile} instead of @file{/etc/ffserver.conf}.
@item -n
-Enable no-launch mode. This option disables all the @code{Launch}
-directives within the various @code{<Feed>} sections. Since
-@command{ffserver} will not launch any @command{ffmpeg} instances, you
-will have to launch them manually.
-
+Enable no-launch mode. This option disables all the Launch directives
+within the various <Stream> sections. Since ffserver will not launch
+any ffmpeg instances, you will have to launch them manually.
@item -d
-Enable debug mode. This option increases log verbosity, and directs
-log messages to stdout. When specified, the @option{CustomLog} option
-is ignored.
+Enable debug mode. This option increases log verbosity, directs log
+messages to stdout.
@end table
-
-@chapter Configuration file syntax
-
-@command{ffserver} reads a configuration file containing global
-options and settings for each stream and feed.
-
-The configuration file consists of global options and dedicated
-sections, which must be introduced by "<@var{SECTION_NAME}
-@var{ARGS}>" on a separate line and must be terminated by a line in
-the form "</@var{SECTION_NAME}>". @var{ARGS} is optional.
-
-Currently the following sections are recognized: @samp{Feed},
-@samp{Stream}, @samp{Redirect}.
-
-A line starting with @code{#} is ignored and treated as a comment.
-
-Name of options and sections are case-insensitive.
-
-@section ACL syntax
-An ACL (Access Control List) specifies the address which are allowed
-to access a given stream, or to write a given feed.
-
-It accepts the folling forms
-@itemize
-@item
-Allow/deny access to @var{address}.
-@example
-ACL ALLOW <address>
-ACL DENY <address>
-@end example
-
-@item
-Allow/deny access to ranges of addresses from @var{first_address} to
-@var{last_address}.
-@example
-ACL ALLOW <first_address> <last_address>
-ACL DENY <first_address> <last_address>
-@end example
-@end itemize
-
-You can repeat the ACL allow/deny as often as you like. It is on a per
-stream basis. The first match defines the action. If there are no matches,
-then the default is the inverse of the last ACL statement.
-
-Thus 'ACL allow localhost' only allows access from localhost.
-'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and
-allow everybody else.
-
-@section Global options
-@table @option
-@item HTTPPort @var{port_number}
-@item Port @var{port_number}
-@item RTSPPort @var{port_number}
-
-@var{HTTPPort} sets the HTTP server listening TCP port number,
-@var{RTSPPort} sets the RTSP server listening TCP port number.
-
-@var{Port} is the equivalent of @var{HTTPPort} and is deprecated.
-
-You must select a different port from your standard HTTP web server if
-it is running on the same computer.
-
-If not specified, no corresponding server will be created.
-
-@item HTTPBindAddress @var{ip_address}
-@item BindAddress @var{ip_address}
-@item RTSPBindAddress @var{ip_address}
-Set address on which the HTTP/RTSP server is bound. Only useful if you
-have several network interfaces.
-
-@var{BindAddress} is the equivalent of @var{HTTPBindAddress} and is
-deprecated.
-
-@item MaxHTTPConnections @var{n}
-Set number of simultaneous HTTP connections that can be handled. It
-has to be defined @emph{before} the @option{MaxClients} parameter,
-since it defines the @option{MaxClients} maximum limit.
-
-Default value is 2000.
-
-@item MaxClients @var{n}
-Set number of simultaneous requests that can be handled. Since
-@command{ffserver} is very fast, it is more likely that you will want
-to leave this high and use @option{MaxBandwidth}.
-
-Default value is 5.
-
-@item MaxBandwidth @var{kbps}
-Set the maximum amount of kbit/sec that you are prepared to consume
-when streaming to clients.
-
-Default value is 1000.
-
-@item CustomLog @var{filename}
-Set access log file (uses standard Apache log file format). '-' is the
-standard output.
-
-If not specified @command{ffserver} will produce no log.
-
-In case the commandline option @option{-d} is specified this option is
-ignored, and the log is written to standard output.
-
-@item NoDaemon
-Set no-daemon mode. This option is currently ignored since now
-@command{ffserver} will always work in no-daemon mode, and is
-deprecated.
-
-@item UseDefaults
-@item NoDefaults
-Control whether default codec options are used for the all streams or not.
-Each stream may overwrite this setting for its own. Default is @var{UseDefaults}.
-The lastest occurrence overrides previous if multiple definitions.
-@end table
-
-@section Feed section
-
-A Feed section defines a feed provided to @command{ffserver}.
-
-Each live feed contains one video and/or audio sequence coming from an
-@command{ffmpeg} encoder or another @command{ffserver}. This sequence
-may be encoded simultaneously with several codecs at several
-resolutions.
-
-A feed instance specification is introduced by a line in the form:
-@example
-<Feed FEED_FILENAME>
-@end example
-
-where @var{FEED_FILENAME} specifies the unique name of the FFM stream.
-
-The following options are recognized within a Feed section.
-
-@table @option
-@item File @var{filename}
-@item ReadOnlyFile @var{filename}
-Set the path where the feed file is stored on disk.
-
-If not specified, the @file{/tmp/FEED.ffm} is assumed, where
-@var{FEED} is the feed name.
-
-If @option{ReadOnlyFile} is used the file is marked as read-only and
-it will not be deleted or updated.
-
-@item Truncate
-Truncate the feed file, rather than appending to it. By default
-@command{ffserver} will append data to the file, until the maximum
-file size value is reached (see @option{FileMaxSize} option).
-
-@item FileMaxSize @var{size}
-Set maximum size of the feed file in bytes. 0 means unlimited. The
-postfixes @code{K} (2^10), @code{M} (2^20), and @code{G} (2^30) are
-recognized.
-
-Default value is 5M.
-
-@item Launch @var{args}
-Launch an @command{ffmpeg} command when creating @command{ffserver}.
-
-@var{args} must be a sequence of arguments to be provided to an
-@command{ffmpeg} instance. The first provided argument is ignored, and
-it is replaced by a path with the same dirname of the @command{ffserver}
-instance, followed by the remaining argument and terminated with a
-path corresponding to the feed.
-
-When the launched process exits, @command{ffserver} will launch
-another program instance.
-
-In case you need a more complex @command{ffmpeg} configuration,
-e.g. if you need to generate multiple FFM feeds with a single
-@command{ffmpeg} instance, you should launch @command{ffmpeg} by hand.
-
-This option is ignored in case the commandline option @option{-n} is
-specified.
-
-@item ACL @var{spec}
-Specify the list of IP address which are allowed or denied to write
-the feed. Multiple ACL options can be specified.
-@end table
-
-@section Stream section
-
-A Stream section defines a stream provided by @command{ffserver}, and
-identified by a single name.
-
-The stream is sent when answering a request containing the stream
-name.
-
-A stream section must be introduced by the line:
-@example
-<Stream STREAM_NAME>
-@end example
-
-where @var{STREAM_NAME} specifies the unique name of the stream.
-
-The following options are recognized within a Stream section.
-
-Encoding options are marked with the @emph{encoding} tag, and they are
-used to set the encoding parameters, and are mapped to libavcodec
-encoding options. Not all encoding options are supported, in
-particular it is not possible to set encoder private options. In order
-to override the encoding options specified by @command{ffserver}, you
-can use the @command{ffmpeg} @option{override_ffserver} commandline
-option.
-
-Only one of the @option{Feed} and @option{File} options should be set.
-
-@table @option
-@item Feed @var{feed_name}
-Set the input feed. @var{feed_name} must correspond to an existing
-feed defined in a @code{Feed} section.
-
-When this option is set, encoding options are used to setup the
-encoding operated by the remote @command{ffmpeg} process.
-
-@item File @var{filename}
-Set the filename of the pre-recorded input file to stream.
-
-When this option is set, encoding options are ignored and the input
-file content is re-streamed as is.
-
-@item Format @var{format_name}
-Set the format of the output stream.
-
-Must be the name of a format recognized by FFmpeg. If set to
-@samp{status}, it is treated as a status stream.
-
-@item InputFormat @var{format_name}
-Set input format. If not specified, it is automatically guessed.
-
-@item Preroll @var{n}
-Set this to the number of seconds backwards in time to start. Note that
-most players will buffer 5-10 seconds of video, and also you need to allow
-for a keyframe to appear in the data stream.
-
-Default value is 0.
-
-@item StartSendOnKey
-Do not send stream until it gets the first key frame. By default
-@command{ffserver} will send data immediately.
-
-@item MaxTime @var{n}
-Set the number of seconds to run. This value set the maximum duration
-of the stream a client will be able to receive.
-
-A value of 0 means that no limit is set on the stream duration.
-
-@item ACL @var{spec}
-Set ACL for the stream.
-
-@item DynamicACL @var{spec}
-
-@item RTSPOption @var{option}
-
-@item MulticastAddress @var{address}
-
-@item MulticastPort @var{port}
-
-@item MulticastTTL @var{integer}
-
-@item NoLoop
-
-@item FaviconURL @var{url}
-Set favicon (favourite icon) for the server status page. It is ignored
-for regular streams.
-
-@item Author @var{value}
-@item Comment @var{value}
-@item Copyright @var{value}
-@item Title @var{value}
-Set metadata corresponding to the option. All these options are
-deprecated in favor of @option{Metadata}.
-
-@item Metadata @var{key} @var{value}
-Set metadata value on the output stream.
-
-@item UseDefaults
-@item NoDefaults
-Control whether default codec options are used for the stream or not.
-Default is @var{UseDefaults} unless disabled globally.
-
-@item NoAudio
-@item NoVideo
-Suppress audio/video.
-
-@item AudioCodec @var{codec_name} (@emph{encoding,audio})
-Set audio codec.
-
-@item AudioBitRate @var{rate} (@emph{encoding,audio})
-Set bitrate for the audio stream in kbits per second.
-
-@item AudioChannels @var{n} (@emph{encoding,audio})
-Set number of audio channels.
-
-@item AudioSampleRate @var{n} (@emph{encoding,audio})
-Set sampling frequency for audio. When using low bitrates, you should
-lower this frequency to 22050 or 11025. The supported frequencies
-depend on the selected audio codec.
-
-@item AVOptionAudio [@var{codec}:]@var{option} @var{value} (@emph{encoding,audio})
-Set generic or private option for audio stream.
-Private option must be prefixed with codec name or codec must be defined before.
-
-@item AVPresetAudio @var{preset} (@emph{encoding,audio})
-Set preset for audio stream.
-
-@item VideoCodec @var{codec_name} (@emph{encoding,video})
-Set video codec.
-
-@item VideoBitRate @var{n} (@emph{encoding,video})
-Set bitrate for the video stream in kbits per second.
-
-@item VideoBitRateRange @var{range} (@emph{encoding,video})
-Set video bitrate range.
-
-A range must be specified in the form @var{minrate}-@var{maxrate}, and
-specifies the @option{minrate} and @option{maxrate} encoding options
-expressed in kbits per second.
-
-@item VideoBitRateRangeTolerance @var{n} (@emph{encoding,video})
-Set video bitrate tolerance in kbits per second.
-
-@item PixelFormat @var{pixel_format} (@emph{encoding,video})
-Set video pixel format.
-
-@item Debug @var{integer} (@emph{encoding,video})
-Set video @option{debug} encoding option.
-
-@item Strict @var{integer} (@emph{encoding,video})
-Set video @option{strict} encoding option.
-
-@item VideoBufferSize @var{n} (@emph{encoding,video})
-Set ratecontrol buffer size, expressed in KB.
-
-@item VideoFrameRate @var{n} (@emph{encoding,video})
-Set number of video frames per second.
-
-@item VideoSize (@emph{encoding,video})
-Set size of the video frame, must be an abbreviation or in the form
-@var{W}x@var{H}.  See @ref{video size syntax,,the Video size section
-in the ffmpeg-utils(1) manual,ffmpeg-utils}.
-
-Default value is @code{160x128}.
-
-@item VideoIntraOnly (@emph{encoding,video})
-Transmit only intra frames (useful for low bitrates, but kills frame rate).
-
-@item VideoGopSize @var{n} (@emph{encoding,video})
-If non-intra only, an intra frame is transmitted every VideoGopSize
-frames. Video synchronization can only begin at an intra frame.
-
-@item VideoTag @var{tag} (@emph{encoding,video})
-Set video tag.
-
-@item VideoHighQuality (@emph{encoding,video})
-@item Video4MotionVector (@emph{encoding,video})
-
-@item BitExact (@emph{encoding,video})
-Set bitexact encoding flag.
-
-@item IdctSimple (@emph{encoding,video})
-Set simple IDCT algorithm.
-
-@item Qscale @var{n} (@emph{encoding,video})
-Enable constant quality encoding, and set video qscale (quantization
-scale) value, expressed in @var{n} QP units.
-
-@item VideoQMin @var{n} (@emph{encoding,video})
-@item VideoQMax @var{n} (@emph{encoding,video})
-Set video qmin/qmax.
-
-@item VideoQDiff @var{integer} (@emph{encoding,video})
-Set video @option{qdiff} encoding option.
-
-@item LumiMask @var{float} (@emph{encoding,video})
-@item DarkMask @var{float} (@emph{encoding,video})
-Set @option{lumi_mask}/@option{dark_mask} encoding options.
-
-@item AVOptionVideo [@var{codec}:]@var{option} @var{value} (@emph{encoding,video})
-Set generic or private option for video stream.
-Private option must be prefixed with codec name or codec must be defined before.
-
-@item AVPresetVideo @var{preset} (@emph{encoding,video})
-Set preset for video stream.
-
-@var{preset} must be the path of a preset file.
-@end table
-
-@subsection Server status stream
-
-A server status stream is a special stream which is used to show
-statistics about the @command{ffserver} operations.
-
-It must be specified setting the option @option{Format} to
-@samp{status}.
-
-@section Redirect section
-
-A redirect section specifies where to redirect the requested URL to
-another page.
-
-A redirect section must be introduced by the line:
-@example
-<Redirect NAME>
-@end example
-
-where @var{NAME} is the name of the page which should be redirected.
-
-It only accepts the option @option{URL}, which specify the redirection
-URL.
-
-@chapter Stream examples
-
-@itemize
-@item
-Multipart JPEG
-@example
-<Stream test.mjpg>
-Feed feed1.ffm
-Format mpjpeg
-VideoFrameRate 2
-VideoIntraOnly
-NoAudio
-Strict -1
-</Stream>
-@end example
-
-@item
-Single JPEG
-@example
-<Stream test.jpg>
-Feed feed1.ffm
-Format jpeg
-VideoFrameRate 2
-VideoIntraOnly
-VideoSize 352x240
-NoAudio
-Strict -1
-</Stream>
-@end example
-
-@item
-Flash
-@example
-<Stream test.swf>
-Feed feed1.ffm
-Format swf
-VideoFrameRate 2
-VideoIntraOnly
-NoAudio
-</Stream>
-@end example
-
-@item
-ASF compatible
-@example
-<Stream test.asf>
-Feed feed1.ffm
-Format asf
-VideoFrameRate 15
-VideoSize 352x240
-VideoBitRate 256
-VideoBufferSize 40
-VideoGopSize 30
-AudioBitRate 64
-StartSendOnKey
-</Stream>
-@end example
-
-@item
-MP3 audio
-@example
-<Stream test.mp3>
-Feed feed1.ffm
-Format mp2
-AudioCodec mp3
-AudioBitRate 64
-AudioChannels 1
-AudioSampleRate 44100
-NoVideo
-</Stream>
-@end example
-
-@item
-Ogg Vorbis audio
-@example
-<Stream test.ogg>
-Feed feed1.ffm
-Metadata title "Stream title"
-AudioBitRate 64
-AudioChannels 2
-AudioSampleRate 44100
-NoVideo
-</Stream>
-@end example
-
-@item
-Real with audio only at 32 kbits
-@example
-<Stream test.ra>
-Feed feed1.ffm
-Format rm
-AudioBitRate 32
-NoVideo
-</Stream>
-@end example
-
-@item
-Real with audio and video at 64 kbits
-@example
-<Stream test.rm>
-Feed feed1.ffm
-Format rm
-AudioBitRate 32
-VideoBitRate 128
-VideoFrameRate 25
-VideoGopSize 25
-</Stream>
-@end example
-
-@item
-For stream coming from a file: you only need to set the input filename
-and optionally a new format.
-
-@example
-<Stream file.rm>
-File "/usr/local/httpd/htdocs/tlive.rm"
-NoAudio
-</Stream>
-@end example
-
-@example
-<Stream file.asf>
-File "/usr/local/httpd/htdocs/test.asf"
-NoAudio
-Metadata author "Me"
-Metadata copyright "Super MegaCorp"
-Metadata title "Test stream from disk"
-Metadata comment "Test comment"
-</Stream>
-@end example
-@end itemize
-
@c man end

@include config.texi
--- a/doc/filter_design.txt
+++ b/doc/filter_design.txt
@@ -98,7 +98,7 @@ Buffer references ownership and permissions
    The AVFilterLink structure has a few AVFilterBufferRef fields. The
    cur_buf and out_buf were used with the deprecated
    start_frame/draw_slice/end_frame API and should no longer be used.
-    src_buf and partial_buf are used by libavfilter internally
+    src_buf, cur_buf_copy and partial_buf are used by libavfilter internally
    and must not be accessed by filters.

  Reference permissions
@@ -166,7 +166,7 @@ Buffer references ownership and permissions
      WRITE permission.

    * Filters that read their input to produce a new frame on output (like
-      scale) need the READ permission on input and must request a buffer
+      scale) need the READ permission on input and and must request a buffer
      with the WRITE permission.

    * Filters that intend to keep a reference after the filtering process
@@ -232,8 +232,7 @@ Frame scheduling
    one of its inputs, repeatedly until at least one frame has been pushed.

    Return values:
-    if request_frame could produce a frame, or at least make progress
-    towards producing a frame, it should return 0;
+    if request_frame could produce a frame, it should return 0;
    if it could not for temporary reasons, it should return AVERROR(EAGAIN);
    if it could not because there are no more frames, it should return
    AVERROR_EOF.
@@ -245,18 +244,20 @@ Frame scheduling
            push_one_frame();
            return 0;
        }
-        input = input_where_a_frame_is_most_needed();
-        ret = ff_request_frame(input);
-        if (ret == AVERROR_EOF) {
-            process_eof_on_input();
-        } else if (ret < 0) {
-            return ret;
+        while (!frame_pushed) {
+            input = input_where_a_frame_is_most_needed();
+            ret = ff_request_frame(input);
+            if (ret == AVERROR_EOF) {
+                process_eof_on_input();
+            } else if (ret < 0) {
+                return ret;
+            }
        }
        return 0;

    Note that, except for filters that can have queued frames, request_frame
    does not push frames: it requests them to its input, and as a reaction,
-    the filter_frame method possibly will be called and do the work.
+    the filter_frame method will be called and do the work.

 Legacy API
 ==========
--- a/doc/filters.texi
+++ b/doc/filters.texi
--- a/Show More
+++ b/Show More
@@ -1 +1 @@
 .8.git
 .0.4