update for 0.11.3

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
update the current year
2013-03-22 04:05:35 +01:00 · 2013-03-22 04:04:36 +01:00 · 2013-03-22 04:04:28 +01:00 · 2013-03-22 04:04:11 +01:00 · 2013-03-20 23:11:46 +01:00 · 2013-03-20 22:10:40 +01:00
3841 changed files with 169977 additions and 435647 deletions
--- a/.gitignore
+++ b/.gitignore
@@ -1,83 +1,64 @@
+.config
+.version
 *.a
 *.o
 *.d
 *.def
 *.dll
-*.dylib
 *.exe
-*.exp
-*.gcda
-*.gcno
-*.h.c
-*.ilk
+*.ho
 *.lib
 *.pc
-*.pdb
 *.so
 *.so.*
 *.ver
 *-example
 *-test
 *_g
-/.config
-/.version
-/ffmpeg
-/ffplay
-/ffprobe
-/ffserver
-/config.*
-/coverage.info
-/doc/*.1
-/doc/*.3
-/doc/*.html
-/doc/*.pod
-/doc/config.texi
-/doc/avoptions_codec.texi
-/doc/avoptions_format.texi
-/doc/doxy/html/
-/doc/examples/avio_reading
-/doc/examples/avcodec
-/doc/examples/demuxing_decoding
-/doc/examples/filter_audio
-/doc/examples/filtering_audio
-/doc/examples/filtering_video
-/doc/examples/metadata
-/doc/examples/muxing
-/doc/examples/pc-uninstalled
-/doc/examples/remuxing
-/doc/examples/resampling_audio
-/doc/examples/scaling_video
-/doc/examples/transcode_aac
-/doc/fate.txt
-/doc/print_options
-/lcov/
-/libavcodec/*_tablegen
-/libavcodec/*_tables.c
-/libavcodec/*_tables.h
-/libavutil/avconfig.h
-/libavutil/ffversion.h
-/tests/audiogen
-/tests/base64
-/tests/data/
-/tests/rotozoom
-/tests/tiny_psnr
-/tests/tiny_ssim
-/tests/videogen
-/tests/vsynth1/
-/tools/aviocat
-/tools/ffbisect
-/tools/bisect.need
-/tools/crypto_bench
-/tools/cws2fws
-/tools/fourcc2pixfmt
-/tools/ffescape
-/tools/ffeval
-/tools/ffhash
-/tools/graph2dot
-/tools/ismindex
-/tools/pktdumper
-/tools/probetest
-/tools/qt-faststart
-/tools/trasher
-/tools/seek_print
-/tools/zmqsend
+*.def
+*.dll
+*.lib
+*.exp
+config.*
+doc/*.1
+doc/*.html
+doc/*.pod
+doc/fate.txt
+doxy
+ffmpeg
+ffplay
+ffprobe
+ffserver
+avconv
+doc/avoptions_codec.texi
+doc/avoptions_format.texi
+doc/print_options
+doc/examples/decoding_encoding
+doc/examples/filtering_audio
+doc/examples/filtering_video
+doc/examples/metadata
+doc/examples/muxing
+libavcodec/*_tablegen
+libavcodec/*_tables.c
+libavcodec/*_tables.h
+libavcodec/codec_names.h
+libavutil/avconfig.h
+tests/audiogen
+tests/base64
+tests/data
+tests/rotozoom
+tests/tiny_psnr
+tests/videogen
+tests/vsynth1
+tests/vsynth2
+tools/aviocat
+tools/cws2fws
+tools/ffeval
+tools/graph2dot
+tools/ismindex
+tools/lavfi-showfiltfmts
+tools/pktdumper
+tools/probetest
+tools/qt-faststart
+tools/trasher
+version.h
--- a/59
+++ b/59
@@ -1,6 +1,55 @@
-See the Git history of the project (git://source.ffmpeg.org/ffmpeg) to
-get the names of people who have contributed to FFmpeg.
+This file contains the names of some of the people who have contributed to
+FFmpeg. The names are sorted alphabetically by last name.  As this file is
+currently quite outdated and git serves as a much better tool for determining
+authorship, it remains here for historical reasons only.

-To check the log, you can type the command "git log" in the FFmpeg
-source directory, or browse the online repository at
-http://source.ffmpeg.org.
+Dénes Balatoni
+Michel Bardiaux
+Fabrice Bellard
+Patrice Bensoussan
+Alex Beregszaszi
+BERO
+Thilo Borgmann
+Mario Brito
+Ronald Bultje
+Alex Converse
+Maarten Daniels
+Reimar Doeffinger
+Tim Ferguson
+Brian Foley
+Arpad Gereoffy
+Philip Gladstone
+Vladimir Gneushev
+Roine Gustafsson
+David Hammerton
+Wolfgang Hesseler
+Marc Hoffman
+Falk Hueffner
+Aurélien Jacobs
+Steven Johnson
+Zdenek Kabelac
+Robin Kay
+Todd Kirby
+Nick Kurshev
+Benjamin Larsson
+Loïc Le Loarer
+Daniel Maas
+Mike Melanson
+Loren Merritt
+Jeff Muizelaar
+Michael Niedermayer
+François Revol
+Peter Ross
+Måns Rullgård
+Stefano Sabatini
+Roman Shaposhnik
+Oded Shimon
+Dieter Shirley
+Konstantin Shishkov
+Juan J. Sierralta
+Ewald Snel
+Sascha Sommer
+Leon van Stuivenberg
+Roberto Togni
+Lionel Ulmer
+Reynaldo Verdejo
--- a/326
+++ b/326
@@ -1,318 +1,21 @@
 Entries are sorted chronologically from oldest to youngest within each release,
 releases are sorted from youngest to oldest.

-version <next>
+version next:

+version 0.11.2:

-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
-
-
-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
- reference-counting for AVFrame and AVPacket data
- ffmpeg now fails when input options are used for output file
-  or vice versa
- support for Monkey's Audio versions from 3.93
- perms and aperms filters
- audio filtering support in ffplay
- 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
-  filtergraph description to be read from a file
- OpenCL support
- audio phaser filter
- separatefields filter
- libquvi demuxer
- uniform options syntax across all filters
- telecine filter
- interlace filter
- smptehdbars source
- inverse telecine filters (fieldmatch and decimate)
- colorbalance filter
- colorchannelmixer filter
- The matroska demuxer can now output proper verbatim ASS packets. It will
-  become the default at the next libavformat major bump.
- decent native animated GIF encoding
- asetrate filter
- interleave filter
- timeline editing with filters
- vidstabdetect and vidstabtransform filters for video stabilization using
-  the vid.stab library
- astats filter
- trim and atrim filters
- ffmpeg -t and -ss (output-only) options are now sample-accurate when
-  transcoding audio
- Matroska muxer can now put the index at the beginning of the file.
- extractplanes filter
- avectorscope filter
- ADPCM DTK decoder
- ADP demuxer
- RSD demuxer
- RedSpark demuxer
- ADPCM IMA Radical decoder
- zmq filters
- DCT denoiser filter (dctdnoiz)
- Wavelet denoiser filter ported from libmpcodecs as owdenoise (formerly "ow")
- Apple Intermediate Codec decoder
- Escape 130 video decoder
- FTP protocol support
- V4L2 output device
- 3D LUT filter (lut3d)
- SMPTE 302M audio encoder
- support for slice multithreading in libavfilter
- Hald CLUT support (generation and filtering)
- VC-1 interlaced B-frame support
- support for WavPack muxing (raw and in Matroska)
- XVideo output device
- vignette filter
- True Audio (TTA) encoder
- Go2Webinar decoder
- mcdeint filter ported from libmpcodecs
- sab filter ported from libmpcodecs
- ffprobe -show_chapters option
- WavPack encoding through libwavpack
- rotate filter
- spp filter ported from libmpcodecs
- libgme support
- psnr filter
-
-
-version 1.2:
-
- VDPAU hardware acceleration through normal hwaccel
- SRTP support
- Error diffusion dither in Swscale
- Chained Ogg support
- Theora Midstream reconfiguration support
- EVRC decoder
- audio fade filter
- filtering audio with unknown channel layout
- allpass, bass, bandpass, bandreject, biquad, equalizer, highpass, lowpass
-  and treble audio filter
- improved showspectrum filter, with multichannel support and sox-like colors
- histogram filter
- tee muxer
- il filter ported from libmpcodecs
- support ID3v2 tags in ASF files
- encrypted TTA stream decoding support
- RF64 support in WAV muxer
- noise filter ported from libmpcodecs
- Subtitles character encoding conversion
- blend filter
- stereo3d filter ported from libmpcodecs
-
-
-version 1.1:
-
- stream disposition information printing in ffprobe
- filter for loudness analysis following EBU R128
- Opus encoder using libopus
- ffprobe -select_streams option
- Pinnacle TARGA CineWave YUV16 decoder
- TAK demuxer, decoder and parser
- DTS-HD demuxer
- remove -same_quant, it hasn't worked for years
- FFM2 support
- X-Face image encoder and decoder
- 24-bit FLAC encoding
- multi-channel ALAC encoding up to 7.1
- metadata (INFO tag) support in WAV muxer
- subtitles raw text decoder
- support for building DLLs using MSVC
- LVF demuxer
- ffescape tool
- metadata (info chunk) support in CAF muxer
- field filter ported from libmpcodecs
- AVR demuxer
- geq filter ported from libmpcodecs
- remove ffserver daemon mode
- AST muxer/demuxer
- new expansion syntax for drawtext
- BRender PIX image decoder
- ffprobe -show_entries option
- ffprobe -sections option
- ADPCM IMA Dialogic decoder
- BRSTM demuxer
- animated GIF decoder and demuxer
- PVF demuxer
- subtitles filter
- IRCAM muxer/demuxer
- Paris Audio File demuxer
- Virtual concatenation demuxer
- VobSub demuxer
- JSON captions for TED talks decoding support
- SOX Resampler support in libswresample
- aselect filter
- SGI RLE 8-bit decoder
- Silicon Graphics Motion Video Compressor 1 & 2 decoder
- Silicon Graphics Movie demuxer
- apad filter
- Resolution & pixel format change support with multithreading for H.264
- documentation split into per-component manuals
- pp (postproc) filter ported from MPlayer
- NIST Sphere demuxer
- MPL2, VPlayer, MPlayer, AQTitle, PJS and SubViewer v1 subtitles demuxers and decoders
- Sony Wave64 muxer
- adobe and limelight publisher authentication in RTMP
- data: URI scheme
- support building on the Plan 9 operating system
- kerndeint filter ported from MPlayer
- histeq filter ported from VirtualDub
- Megalux Frame demuxer
- 012v decoder
- Improved AVC Intra decoding support
-
-
-version 1.0:
-
- INI and flat output in ffprobe
- Scene detection in libavfilter
- Indeo Audio decoder
- channelsplit audio filter
- setnsamples audio filter
- atempo filter
- ffprobe -show_data option
- RTMPT protocol support
- iLBC encoding/decoding via libilbc
- Microsoft Screen 1 decoder
- join audio filter
- audio channel mapping filter
- Microsoft ATC Screen decoder
- RTSP listen mode
- TechSmith Screen Codec 2 decoder
- AAC encoding via libfdk-aac
- Microsoft Expression Encoder Screen decoder
- RTMPS protocol support
- RTMPTS protocol support
- RTMPE protocol support
- RTMPTE protocol support
- showwaves and showspectrum filter
- LucasArts SMUSH playback support
- SAMI, RealText and SubViewer demuxers and decoders
- Heart Of Darkness PAF playback support
- iec61883 device
- asettb filter
- new option: -progress
- 3GPP Timed Text encoder/decoder
- GeoTIFF decoder support
- ffmpeg -(no)stdin option
- Opus decoder using libopus
- caca output device using libcaca
- alphaextract and alphamerge filters
- concat filter
- flite filter
- Canopus Lossless Codec decoder
- bitmap subtitles in filters (experimental and temporary)
- MP2 encoding via TwoLAME
- bmp parser
- smptebars source
- asetpts filter
- hue filter
- ICO muxer
- SubRip encoder and decoder without embedded timing
- edge detection filter
- framestep filter
- ffmpeg -shortest option is now per-output file
-  -pass and -passlogfile are now per-output stream
- volume measurement filter
- Ut Video encoder
- Microsoft Screen 2 decoder
- smartblur filter ported from MPlayer
- CPiA decoder
- decimate filter ported from MPlayer
- RTP depacketization of JPEG
- Smooth Streaming live segmenter muxer
- F4V muxer
- sendcmd and asendcmd filters
- WebVTT demuxer and decoder (simple tags supported)
- RTP packetization of JPEG
- faststart option in the MOV/MP4 muxer
- support for building with MSVC
+- Several bugs and crashes have been fixed as well as build problems
+  with recent mingw64


 version 0.11:
-
- Fixes: CVE-2012-2772, CVE-2012-2774, CVE-2012-2775, CVE-2012-2776, CVE-2012-2777,
-         CVE-2012-2779, CVE-2012-2782, CVE-2012-2783, CVE-2012-2784, CVE-2012-2785,
-         CVE-2012-2786, CVE-2012-2787, CVE-2012-2788, CVE-2012-2789, CVE-2012-2790,
-         CVE-2012-2791, CVE-2012-2792, CVE-2012-2793, CVE-2012-2794, CVE-2012-2795,
-         CVE-2012-2796, CVE-2012-2797, CVE-2012-2798, CVE-2012-2799, CVE-2012-2800,
-         CVE-2012-2801, CVE-2012-2802, CVE-2012-2803, CVE-2012-2804,
+Fixes:CVE-2012-2772, CVE-2012-2774, CVE-2012-2775, CVE-2012-2776, CVE-2012-2777,
+      CVE-2012-2779, CVE-2012-2782, CVE-2012-2783, CVE-2012-2784, CVE-2012-2785,
+      CVE-2012-2786, CVE-2012-2787, CVE-2012-2788, CVE-2012-2789, CVE-2012-2790,
+      CVE-2012-2791, CVE-2012-2792, CVE-2012-2793, CVE-2012-2794, CVE-2012-2795,
+      CVE-2012-2796, CVE-2012-2797, CVE-2012-2798, CVE-2012-2799, CVE-2012-2800,
+      CVE-2012-2801, CVE-2012-2802, CVE-2012-2803, CVE-2012-2804,
 - v408 Quicktime and Microsoft AYUV Uncompressed 4:4:4:4 encoder and decoder
 - setfield filter
 - CDXL demuxer and decoder
@@ -343,14 +46,13 @@ version 0.11:
 - accept + prefix to -pix_fmt option to disable automatic conversions.
 - complete audio filtering in libavfilter and ffmpeg
 - add fps filter
+- audio split filter
 - vorbis parser
 - png parser
 - audio mix filter
- ffv1: support (draft) version 1.3


 version 0.10:
-
 - Fixes: CVE-2011-3929, CVE-2011-3934, CVE-2011-3935, CVE-2011-3936,
         CVE-2011-3937, CVE-2011-3940, CVE-2011-3941, CVE-2011-3944,
         CVE-2011-3945, CVE-2011-3946, CVE-2011-3947, CVE-2011-3949,
@@ -686,7 +388,7 @@ 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
@@ -786,7 +488,7 @@ 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
@@ -814,7 +516,7 @@ 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/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         = 0.11.3

-# 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.
@@ -44,7 +44,7 @@ PROJECT_LOGO           =
 # If a relative path is entered, it will be relative to the location
 # where doxygen was started. If left blank the current directory will be used.

-OUTPUT_DIRECTORY       = doc/doxy
+OUTPUT_DIRECTORY       = doxy

 # If the CREATE_SUBDIRS tag is set to YES, then doxygen will create
 # 4096 sub-directories (in 2 levels) under the output directory of each output
@@ -277,7 +277,7 @@ SUBGROUPING            = YES
 # be useful for C code in case the coding convention dictates that all compound
 # types are typedef'ed and only the typedef is referenced, never the tag name.

-TYPEDEF_HIDES_STRUCT   = YES
+TYPEDEF_HIDES_STRUCT   = NO

 # The SYMBOL_CACHE_SIZE determines the size of the internal cache use to
 # determine which symbols to keep in memory and which to flush to disk.
@@ -288,7 +288,7 @@ TYPEDEF_HIDES_STRUCT   = YES
 # causing a significant performance penality.
 # If the system has enough physical memory increasing the cache will improve the
 # performance by keeping more symbols in memory. Note that the value works on
-# a logarithmic scale so increasing the size by one will roughly double the
+# a logarithmic scale so increasing the size by one will rougly double the
 # memory usage. The cache size is given by this formula:
 # 2^(16+SYMBOL_CACHE_SIZE). The valid range is 0..9, the default is 0,
 # corresponding to a cache size of 2^16 = 65536 symbols
@@ -409,7 +409,7 @@ INLINE_INFO            = YES
 # alphabetically by member name. If set to NO the members will appear in
 # declaration order.

-SORT_MEMBER_DOCS       = NO
+SORT_MEMBER_DOCS       = YES

 # If the SORT_BRIEF_DOCS tag is set to YES then doxygen will sort the
 # brief documentation of file, namespace and class members alphabetically
@@ -489,6 +489,12 @@ MAX_INITIALIZER_LINES  = 30

 SHOW_USED_FILES        = YES

+# If the sources in your project are distributed over multiple directories
+# then setting the SHOW_DIRECTORIES tag to YES will show the directory hierarchy
+# in the documentation. The default is NO.
+
+SHOW_DIRECTORIES       = NO
+
 # Set the SHOW_FILES tag to NO to disable the generation of the Files page.
 # This will remove the Files entry from the Quick Index and from the
 # Folder Tree View (if specified). The default is YES.
@@ -639,14 +645,15 @@ EXCLUDE_SYMBOLS        =
 # directories that contain example code fragments that are included (see
 # the \include command).

-EXAMPLE_PATH           = doc/examples/
+EXAMPLE_PATH           = libavcodec/ \
+                         libavformat/

 # If the value of the EXAMPLE_PATH tag contains directories, you can use the
 # EXAMPLE_PATTERNS tag to specify one or more wildcard pattern (like *.cpp
 # and *.h) to filter out the source-files in the directories. If left
 # blank all files are included.

-EXAMPLE_PATTERNS       = *.c
+EXAMPLE_PATTERNS       = *-example.c

 # If the EXAMPLE_RECURSIVE tag is set to YES then subdirectories will be
 # searched for input files to be used with the \include or \dontinclude
@@ -709,7 +716,7 @@ INLINE_SOURCES         = NO
 # doxygen to hide any special comment blocks from generated source code
 # fragments. Normal C and C++ comments will always remain visible.

-STRIP_CODE_COMMENTS    = NO
+STRIP_CODE_COMMENTS    = YES

 # If the REFERENCED_BY_RELATION tag is set to YES
 # then for each documented function all documented
@@ -793,13 +800,13 @@ 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 +815,7 @@ 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
@@ -818,7 +825,7 @@ HTML_STYLESHEET        =
 # 180 is cyan, 240 is blue, 300 purple, and 360 is red again.
 # The allowed range is 0 to 359.

-#HTML_COLORSTYLE_HUE    = 120
+HTML_COLORSTYLE_HUE    = 120

 # The HTML_COLORSTYLE_SAT tag controls the purity (or saturation) of
 # the colors in the HTML output. For a value of 0 the output will use
@@ -841,6 +848,12 @@ HTML_COLORSTYLE_GAMMA  = 80

 HTML_TIMESTAMP         = YES

+# If the HTML_ALIGN_MEMBERS tag is set to YES, the members of classes,
+# files or namespaces will be aligned in HTML using tables. If set to
+# NO a bullet list will be used.
+
+HTML_ALIGN_MEMBERS     = YES
+
 # If the HTML_DYNAMIC_SECTIONS tag is set to YES then the generated HTML
 # documentation will contain sections that can be hidden and shown after the
 # page has loaded. For this to work a browser that supports
@@ -1021,6 +1034,11 @@ ENUM_VALUES_PER_LINE   = 4

 GENERATE_TREEVIEW      = NO

+# By enabling USE_INLINE_TREES, doxygen will generate the Groups, Directories,
+# and Class Hierarchy pages using a tree view instead of an ordered list.
+
+USE_INLINE_TREES       = NO
+
 # If the treeview is enabled (see GENERATE_TREEVIEW) then this tag can be
 # used to set the initial width (in pixels) of the frame in which the tree
 # is shown.
@@ -1356,9 +1374,14 @@ INCLUDE_FILE_PATTERNS  =
 # instead of the = operator.

 PREDEFINED             = "__attribute__(x)=" \
+                         "RENAME(x)=x ## _TMPL" \
+                         "DEF(x)=x ## _TMPL" \
+                         HAVE_AV_CONFIG_H \
+                         HAVE_MMX \
+                         HAVE_MMX2 \
+                         HAVE_AMD3DNOW \
                         "DECLARE_ALIGNED(a,t,n)=t n" \
-                         "offsetof(x,y)=0x42" \
-                         av_alloc_size \
+                         "offsetof(x,y)=0x42"

 # 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/87
+++ b/87
@@ -1,4 +1,5 @@
 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
@@ -13,46 +14,9 @@ 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_kerndeint.c
-    - vf_mcdeint.c
-    - vf_mp.c
-    - vf_owdenoise.c
-    - vf_perspective.c
-    - vf_phase.c
-    - vf_pp.c
-    - vf_pullup.c
-    - vf_sab.c
-    - vf_smartblur.c
-    - vf_spp.c
-    - vf_stereo3d.c
-    - vf_super2xsai.c
-    - vf_tinterlace.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:

@@ -63,41 +27,24 @@ There are a handful of files under other licensing terms, namely:
  You must also indicate any changes including additions and deletions to
  those three files in the documentation.

+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.

-external libraries
-==================

-FFmpeg can be combined with a number of external libraries, which sometimes
-affect the licensing of binaries resulting from the combination.
+external libraries:
+-------------------

-compatible libraries
--------------------
+Some external libraries, e.g. libx264, are under GPL and can be used in
+conjunction with FFmpeg. They require --enable-gpl to be passed to configure
+as well.

-The following libraries are under GPL:
-    - frei0r
-    - libcdio
-    - libutvideo
-    - libvidstab
-    - libx264
-    - libx265
-    - libxavs
-    - libxvid
-When combining them with FFmpeg, FFmpeg needs to be licensed as GPL as well by
-passing --enable-gpl to configure.
+The OpenCORE external 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 the OpenCORE libraries with FFmpeg, the license
+version needs to be upgraded by passing --enable-version3 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.
+The nonfree external libraries libfaac and libaacplus can be hooked up in FFmpeg.
+You need to pass --enable-nonfree to configure to enable it. Employ this option
+with care as FFmpeg then becomes nonfree and unredistributable.
--- a/180
+++ b/180
@@ -7,8 +7,8 @@ FFmpeg code.
 Please try to keep entries where you are the maintainer up to date!

 Names in () mean that the maintainer currently has no time to maintain the code.
-A (CC <address>) after the name means that the maintainer prefers to be CC-ed on
-patches and related discussions.
+A CC after the name means that the maintainer prefers to be CC-ed on patches
+and related discussions.


 Project Leader
@@ -31,7 +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
@@ -43,24 +43,16 @@ QuickTime faststart:
 Miscellaneous Areas
 ===================

-documentation                           Stefano Sabatini, Mike Melanson, Timothy Gu
+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
+project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Döffinger
+mailinglists                            Michael Niedermayer, Baptiste Coudurier, Lou Logan
 presets                                 Robert Swain
 metadata subsystem                      Aurelien Jacobs
 release management                      Michael Niedermayer


-Communication
-=============
-
-website                                 Robert Swain, Lou Logan
-mailinglists                            Michael Niedermayer, Baptiste Coudurier, Lou Logan
-Google+                                 Paul B Mahol, Michael Niedermayer, Alexander Strasser
-Twitter                                 Lou Logan
-Launchpad                               Timothy Gu
-
-
 libavutil
 =========

@@ -70,23 +62,11 @@ Internal Interfaces:
  libavutil/common.h                    Michael Niedermayer

 Other:
-  bprint                                Nicolas George
-  bswap.h
-  des                                   Reimar Doeffinger
-  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
-  timecode                              Clément Bœsch
+  mathematics.c, mathematics.h          Michael Niedermayer
+  integer.c, integer.h                  Michael Niedermayer
+  bswap.h


 libavcodec
@@ -97,6 +77,10 @@ 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:
@@ -127,8 +111,6 @@ Generic Parts:
    libpostproc/*                       Michael Niedermayer
  table generation:
    tableprint.c, tableprint.h          Reimar Doeffinger
-  fixed point FFT:
-    fft*                                Zeljko Lukac

 Codecs:
  4xm.c                                 Michael Niedermayer
@@ -142,29 +124,23 @@ 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
  bmp.c                                 Mans Rullgard, Kostya Shishkov
  cavs*                                 Stefan Gehrer
-  cdxl.c                                Paul B Mahol
  celp_filters.*                        Vitor Sessak
  cinepak.c                             Roberto Togni
-  cinepakenc.c                          Rl / Aetey G.T. AB
  cljr                                  Alex Beregszaszi
-  cllc.c                                Derek Buitenhuis
  cook.c, cookdata.h                    Benjamin Larsson
-  cpia.c                                Stephan Hilb
  crystalhd.c                           Philip Langdale
  cscd.c                                Reimar Doeffinger
  dca.c                                 Kostya Shishkov, Benjamin Larsson
  dnxhd*                                Baptiste Coudurier
  dpcm.c                                Mike Melanson
-  dv.c                                  Roman Shaposhnik
  dxa.c                                 Kostya Shishkov
+  dv.c                                  Roman Shaposhnik
  eacmv*, eaidct*, eat*                 Peter Ross
-  exif.c, exif.h                        Thilo Borgmann
  ffv1.c                                Michael Niedermayer
  ffwavesynth.c                         Nicolas George
  flac*                                 Justin Ruggles
@@ -173,9 +149,9 @@ Codecs:
  g722.c                                Martin Storsjo
  g726.c                                Roman Shaposhnik
  gifdec.c                              Baptiste Coudurier
+  h264*                                 Loren Merritt, Michael Niedermayer
  h261*                                 Michael Niedermayer
  h263*                                 Michael Niedermayer
-  h264*                                 Loren Merritt, Michael Niedermayer
  huffyuv.c                             Michael Niedermayer
  idcinvideo.c                          Mike Melanson
  imc*                                  Benjamin Larsson
@@ -184,14 +160,13 @@ Codecs:
  interplayvideo.c                      Mike Melanson
  ivi*                                  Kostya Shishkov
  jacosub*                              Clément Bœsch
-  jpeg2000*                             Nicolas Bertrand
  jpeg_ls.c                             Kostya Shishkov
  jvdec.c                               Peter Ross
  kmvc.c                                Kostya Shishkov
  lcl*.c                                Roberto Togni, Reimar Doeffinger
  libcelt_dec.c                         Nicolas George
-  libdirac*                             David Conrad
  libgsm.c                              Michel Bardiaux
+  libdirac*                             David Conrad
  libopenjpeg.c                         Jaikrishnan Menon
  libopenjpegenc.c                      Michael Bradshaw
  libschroedinger*                      David Conrad
@@ -199,28 +174,23 @@ Codecs:
  libtheoraenc.c                        David Conrad
  libutvideo*                           Derek Buitenhuis
  libvorbis.c                           David Conrad
-  libvpx*                               James Zern
-  libx264.c                             Mans Rullgard, Jason Garrett-Glaser
-  libx265.c                             Derek Buitenhuis
  libxavs.c                             Stefan Gehrer
-  libzvbi-teletextdec.c                 Marton Balint
+  libx264.c                             Mans Rullgard, Jason Garrett-Glaser
  loco.c                                Kostya Shishkov
  lzo.h, lzo.c                          Reimar Doeffinger
  mdec.c                                Michael Niedermayer
  mimic.c                               Ramiro Polla
-  mjpeg*.c                              Michael Niedermayer
+  mjpeg.c                               Michael Niedermayer
  mlp*                                  Ramiro Polla
  mmvideo.c                             Peter Ross
  mpc*                                  Kostya Shishkov
  mpeg12.c, mpeg12data.h                Michael Niedermayer
  mpegvideo.c, mpegvideo.h              Michael Niedermayer
-  mqc*                                  Nicolas Bertrand
  msmpeg4.c, msmpeg4data.h              Michael Niedermayer
  msrle.c                               Mike Melanson
  msvideo1.c                            Mike Melanson
  nellymoserdec.c                       Benjamin Larsson
  nuv.c                                 Reimar Doeffinger
-  paf.*                                 Paul B Mahol
  pcx.c                                 Ivo van Poorten
  pgssubdec.c                           Reimar Doeffinger
  ptx.c                                 Ivo van Poorten
@@ -240,13 +210,11 @@ Codecs:
  s3tc*                                 Ivo van Poorten
  smacker.c                             Kostya Shishkov
  smc.c                                 Mike Melanson
-  smvjpegdec.c                          Ash Hughes
  snow.c                                Michael Niedermayer, Loren Merritt
  sonic.c                               Alex Beregszaszi
  srt*                                  Aurelien Jacobs
  sunrast.c                             Ivo van Poorten
  svq3.c                                Michael Niedermayer
-  tak*                                  Paul B Mahol
  targa.c                               Kostya Shishkov
  tiff.c                                Kostya Shishkov
  truemotion1*                          Mike Melanson
@@ -254,7 +222,6 @@ Codecs:
  truespeech.c                          Kostya Shishkov
  tscc.c                                Kostya Shishkov
  tta.c                                 Alex Beregszaszi, Jaikrishnan Menon
-  ttaenc.c                              Paul B Mahol
  txd.c                                 Ivo van Poorten
  ulti*                                 Kostya Shishkov
  v410*.c                               Derek Buitenhuis
@@ -262,16 +229,13 @@ Codecs:
  vble.c                                Derek Buitenhuis
  vc1*                                  Kostya Shishkov
  vcr1.c                                Michael Niedermayer
-  vda_h264_dec.c                        Xidorn Quan
-  vima.c                                Paul B Mahol
  vmnc.c                                Kostya Shishkov
-  vorbis_dec.c                          Denes Balatoni, David Conrad
  vorbis_enc.c                          Oded Shimon
+  vorbis_dec.c                          Denes Balatoni, David Conrad
  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
@@ -279,11 +243,8 @@ Codecs:
  wmv2.c                                Michael Niedermayer
  wnv1.c                                Kostya Shishkov
  xan.c                                 Mike Melanson
-  xbm*                                  Paul B Mahol
-  xface                                 Stefano Sabatini
  xl.c                                  Kostya Shishkov
  xvmc.c                                Ivan Kalvachev
-  xwd*                                  Paul B Mahol
  zerocodec.c                           Derek Buitenhuis
  zmbv*                                 Kostya Shishkov

@@ -302,56 +263,22 @@ libavdevice
    libavdevice/avdevice.h


-  dshow.c                               Roger Pack
-  fbdev_enc.c                           Lukasz Marek
-  iec61883.c                            Georg Lippitsch
-  lavfi                                 Stefano Sabatini
  libdc1394.c                           Roman Shaposhnik
-  opengl_enc.c                          Lukasz Marek
-  pulse_audio_enc.c                     Lukasz Marek
-  sdl                                   Stefano Sabatini
  v4l2.c                                Luca Abeni
  vfwcap.c                              Ramiro Polla

+
 libavfilter
 ===========

-Generic parts:
+Video filters:
  graphdump.c                           Nicolas George
-
-Filters:
-  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_compand.c                          Paul B Mahol
-  af_ladspa.c                           Paul B Mahol
  af_pan.c                              Nicolas George
-  avf_avectorscope.c                    Paul B Mahol
-  vf_blend.c                            Paul B Mahol
-  vf_colorbalance.c                     Paul B Mahol
-  vf_dejudder.c                         Nicholas Robbins
-  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_il.c                               Paul B Mahol
-  vf_mergeplanes.c                      Paul B Mahol
-  vf_psnr.c                             Paul B Mahol
-  vf_scale.c                            Michael Niedermayer
-  vf_separatefields.c                   Paul B Mahol
-  vf_stereo3d.c                         Paul B Mahol
-  vf_telecine.c                         Paul B Mahol
+  vsrc_mandelbrot.c                     Michael Niedermayer
  vf_yadif.c                            Michael Niedermayer

-Sources:
-  vsrc_mandelbrot.c                     Michael Niedermayer

 libavformat
 ===========
@@ -366,57 +293,43 @@ Generic parts:
 Muxers/Demuxers:
  4xm.c                                 Mike Melanson
  adtsenc.c                             Robert Swain
-  afc.c                                 Paul B Mahol
-  aiffdec.c                             Baptiste Coudurier, Matthieu Bouron
-  aiffenc.c                             Baptiste Coudurier, Matthieu Bouron
+  aiff.c                                Baptiste Coudurier
  ape.c                                 Kostya Shishkov
  ass*                                  Aurelien Jacobs
-  astdec.c                              Paul B Mahol
-  astenc.c                              James Almer
  avi*                                  Michael Niedermayer
-  avisynth.c                            AvxSynth Team (avxsynth.testing at gmail dot com)
-  avr.c                                 Paul B Mahol
  bink.c                                Peter Ross
-  brstm.c                               Paul B Mahol
  caf*                                  Peter Ross
-  cdxl.c                                Paul B Mahol
  crc.c                                 Michael Niedermayer
  daud.c                                Reimar Doeffinger
-  dtshddec.c                            Paul B Mahol
  dv.c                                  Roman Shaposhnik
  dxa.c                                 Kostya Shishkov
  electronicarts.c                      Peter Ross
-  epafdec.c                             Paul B Mahol
  ffm*                                  Baptiste Coudurier
  flac*                                 Justin Ruggles
  flic.c                                Mike Melanson
  flvdec.c, flvenc.c                    Michael Niedermayer
  gxf.c                                 Reimar Doeffinger
  gxfenc.c                              Baptiste Coudurier
-  hls.c                                 Anssi Hannula
  idcin.c                               Mike Melanson
  idroqdec.c                            Mike Melanson
  iff.c                                 Jaikrishnan Menon
-  img2*.c                               Michael Niedermayer
  ipmovie.c                             Mike Melanson
-  ircam*                                Paul B Mahol
+  img2.c                                Michael Niedermayer
  iss.c                                 Stefan Gehrer
  jacosub*                              Clément Bœsch
  jvdec.c                               Peter Ross
  libmodplug.c                          Clément Bœsch
  libnut.c                              Oded Shimon
  lmlm4.c                               Ivo van Poorten
-  lvfdec.c                              Paul B Mahol
  lxfdec.c                              Tomas Härdin
  matroska.c                            Aurelien Jacobs
  matroskadec.c                         Aurelien Jacobs
  matroskaenc.c                         David Conrad
  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
+  movenc.c                              Michael Niedermayer, Baptiste Coudurier
  mpc.c                                 Kostya Shishkov
  mpeg.c                                Michael Niedermayer
  mpegenc.c                             Michael Niedermayer
@@ -425,7 +338,6 @@ Muxers/Demuxers:
  mtv.c                                 Reynaldo H. Verdejo Pinochet
  mxf*                                  Baptiste Coudurier
  mxfdec.c                              Tomas Härdin
-  nistspheredec.c                       Paul B Mahol
  nsvdec.c                              Francois Revol
  nut.c                                 Michael Niedermayer
  nuv.c                                 Reimar Doeffinger
@@ -433,10 +345,8 @@ Muxers/Demuxers:
  oggenc.c                              Baptiste Coudurier
  oggparse*.c                           David Conrad
  oma.c                                 Maxim Poliakovski
-  paf.c                                 Paul B Mahol
  psxstr.c                              Mike Melanson
  pva.c                                 Ivo van Poorten
-  pvfdec.c                              Paul B Mahol
  r3d.c                                 Baptiste Coudurier
  raw.c                                 Michael Niedermayer
  rdt.c                                 Ronald S. Bultje
@@ -452,52 +362,31 @@ Muxers/Demuxers:
  segafilm.c                            Mike Melanson
  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
  tta.c                                 Alex Beregszaszi
  txd.c                                 Ivo van Poorten
  voc.c                                 Aurelien Jacobs
  wav.c                                 Michael Niedermayer
  wc3movie.c                            Mike Melanson
-  webvtt*                               Matthew J Heaney
  westwood.c                            Mike Melanson
  wtv.c                                 Peter Ross
  wv.c                                  Kostya Shishkov
-  wvenc.c                               Paul B Mahol

 Protocols:
  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


-libswresample
-=============
-
-Generic parts:
-  audioconvert.c                        Michael Niedermayer
-  dither.c                              Michael Niedermayer
-  rematrix*.c                           Michael Niedermayer
-  swresample*.c                         Michael Niedermayer
-
-Resamplers:
-  resample*.c                           Michael Niedermayer
-  soxr_resample.c                       Rob Sykes
-
-
 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
@@ -511,9 +400,9 @@ x86                                     Michael Niedermayer
 Releases
 ========

-2.2                                     Michael Niedermayer
-2.1                                     Michael Niedermayer
 1.2                                     Michael Niedermayer
+1.1                                     Michael Niedermayer
+1.0                                     Michael Niedermayer

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

@@ -521,21 +410,16 @@ 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
 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
-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
 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
 Justin Ruggles                3136 ECC0 C10D 6C04 5F43 CA29 FCBE CD2A 3787 1EBF
 Loren Merritt                 ABD9 08F4 C920 3F65 D8BE 35D7 1540 DAA7 060F 56DE
 Lou Logan                     7D68 DC73 CBEF EABB 671A B6CF 621C 2E28 82F8 DC3A
@@ -549,7 +433,5 @@ 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
-Tomas Härdin                  A79D 4E3D F38F 763F 91F5 8B33 A01E 8AE0 41BB 2551
-Wei Gao                       4269 7741 857A 0E60 9EC5 08D2 4744 4EFA 62C1 87B9
+Stefano Sabatini              9A43 10F8 D32C D33C 48E7 C52C 5DF2 8E4D B2EE 066B
+Tomas Härdin                  D133 29CA 4EEC 9DB4 7076 F697 B04B 7403 3313 41FD
--- a/116
+++ b/116
@@ -6,34 +6,28 @@ vpath %.cpp  $(SRC_PATH)
 vpath %.h    $(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
-TESTTOOLS   = audiogen videogen rotozoom tiny_psnr tiny_ssim base64
+PROGS      := $(PROGS-yes:%=%$(EXESUF))
+INSTPROGS   = $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))
+OBJS        = $(PROGS-yes:%=%.o) cmdutils.o
+TESTTOOLS   = audiogen videogen rotozoom tiny_psnr base64
 HOSTPROGS  := $(TESTTOOLS:%=tests/%) doc/print_options
-TOOLS       = qt-faststart trasher uncoded_frame
+TOOLS       = qt-faststart trasher
 TOOLS-$(CONFIG_ZLIB) += cws2fws

+BASENAMES   = ffmpeg ffplay ffprobe ffserver
+ALLPROGS    = $(BASENAMES:%=%$(PROGSSUF)$(EXESUF))
+ALLPROGS_G  = $(BASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
+ALLMANPAGES = $(BASENAMES:%=%.1)
+
 FFLIBS-$(CONFIG_AVDEVICE) += avdevice
 FFLIBS-$(CONFIG_AVFILTER) += avfilter
 FFLIBS-$(CONFIG_AVFORMAT) += avformat
@@ -46,23 +40,25 @@ FFLIBS-$(CONFIG_SWSCALE)  += swscale
 FFLIBS := avutil

 DATA_FILES := $(wildcard $(SRC_PATH)/presets/*.ffpreset) $(SRC_PATH)/doc/ffprobe.xsd
-EXAMPLES_FILES := $(wildcard $(SRC_PATH)/doc/examples/*.c) $(SRC_PATH)/doc/examples/Makefile $(SRC_PATH)/doc/examples/README
+EXAMPLES_FILES := $(wildcard $(SRC_PATH)/doc/examples/*.c) $(SRC_PATH)/doc/examples/Makefile

-SKIPHEADERS = cmdutils_common_opts.h compat/w32pthreads.h
+SKIPHEADERS = cmdutils_common_opts.h

 include $(SRC_PATH)/common.mak

 FF_EXTRALIBS := $(FFEXTRALIBS)
 FF_DEP_LIBS  := $(DEP_LIBS)

-all: $(AVPROGS)
+all: $(PROGS)

-$(TOOLS): %$(EXESUF): %.o $(EXEOBJS)
-	$(LD) $(LDFLAGS) $(LD_O) $^ $(ELIBS)
+$(PROGS): %$(EXESUF): %$(PROGSSUF)_g$(EXESUF)
+	$(CP) $< $@$(PROGSSUF)
+	$(STRIP) $@$(PROGSSUF)

-tools/cws2fws$(EXESUF): ELIBS = $(ZLIB)
-tools/uncoded_frame$(EXESUF): $(FF_DEP_LIBS)
-tools/uncoded_frame$(EXESUF): ELIBS = $(FF_EXTRALIBS)
+$(TOOLS): %$(EXESUF): %.o
+	$(LD) $(LDFLAGS) -o $@ $< $(ELIBS)
+
+tools/cws2fws$(EXESUF): ELIBS = -lz

 config.h: .config
 .config: $(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c))
@@ -71,12 +67,10 @@ config.h: .config
 	@-tput sgr0 2>/dev/null

 SUBDIR_VARS := CLEANFILES EXAMPLES FFLIBS HOSTPROGS TESTPROGS TOOLS      \
-               HEADERS ARCH_HEADERS BUILT_HEADERS SKIPHEADERS            \
-               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 SLIBOBJS HOSTOBJS TESTOBJS
+               ARCH_HEADERS BUILT_HEADERS SKIPHEADERS                    \
+               ALTIVEC-OBJS ARMV5TE-OBJS ARMV6-OBJS ARMVFP-OBJS MMI-OBJS \
+               MMX-OBJS NEON-OBJS VIS-OBJS YASM-OBJS                     \
+               OBJS TESTOBJS

 define RESET
 $(1) :=
@@ -93,27 +87,12 @@ endef

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

-include $(SRC_PATH)/doc/Makefile
+ffplay.o: CFLAGS += $(SDL_CFLAGS)
+ffplay_g$(EXESUF): FF_EXTRALIBS += $(SDL_LIBS)
+ffserver_g$(EXESUF): LDFLAGS += $(FFSERVERLDFLAGS)

-define DOPROG
-OBJS-$(1) += $(1).o $(EXEOBJS) $(OBJS-$(1)-yes)
-$(1)$(PROGSSUF)_g$(EXESUF): $$(OBJS-$(1))
-$$(OBJS-$(1)): CFLAGS  += $(CFLAGS-$(1))
-$(1)$(PROGSSUF)_g$(EXESUF): LDFLAGS += $(LDFLAGS-$(1))
-$(1)$(PROGSSUF)_g$(EXESUF): FF_EXTRALIBS += $(LIBS-$(1))
-include $$(OBJS-$(1):.o=.d)
-endef
-
-$(foreach P,$(PROGS),$(eval $(call DOPROG,$(P:$(PROGSSUF)$(EXESUF)=))))
-
-ffprobe.o cmdutils.o : libavutil/ffversion.h
-
-$(PROGS): %$(PROGSSUF)$(EXESUF): %$(PROGSSUF)_g$(EXESUF)
-	$(CP) $< $@
-	$(STRIP) $@
-
-%$(PROGSSUF)_g$(EXESUF): %.o $(FF_DEP_LIBS)
-	$(LD) $(LDFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
+%$(PROGSSUF)_g$(EXESUF): %.o cmdutils.o $(FF_DEP_LIBS)
+	$(LD) $(LDFLAGS) -o $@ $< cmdutils.o $(FF_EXTRALIBS)

 OBJDIRS += tools

@@ -125,14 +104,14 @@ 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

@@ -143,7 +122,7 @@ 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)"

@@ -155,27 +134,38 @@ 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) $(TOOLS)
 	$(RM) $(CLEANSUFFIXES:%=tools/%)
+	$(RM) coverage.info
 	$(RM) -r coverage-html
-	$(RM) -rf coverage.info lcov

 distclean::
 	$(RM) $(DISTCLEANSUFFIXES)
-	$(RM) config.* .config libavutil/avconfig.h .version version.h libavutil/ffversion.h libavcodec/codec_names.h
+	$(RM) config.* .version version.h libavutil/avconfig.h

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

-check: all alltools examples testprogs fate
+# Without the sed genthml thinks "libavutil" and "./libavutil" are two different things
+coverage.info: $(wildcard *.gcda *.gcno */*.gcda */*.gcno */*/*.gcda */*/*.gcno)
+	$(Q)lcov -c -d . -b . | sed -e 's#/./#/#g' > $@

+coverage-html: coverage.info
+	$(Q)mkdir -p $@
+	$(Q)genhtml -o $@ $<
+	$(Q)touch $@
+
+check: all alltools checkheaders examples testprogs fate
+
+include $(SRC_PATH)/doc/Makefile
 include $(SRC_PATH)/tests/Makefile

 $(sort $(OBJDIRS)):
--- a/2
+++ b/2
@@ -1 +1 @@
-2.1.git
+0.11.3
--- a/1
+++ b/1
@@ -0,0 +1 @@
+0.11.3
--- a/arch.mak
+++ b/arch.mak
@@ -1,12 +1,9 @@
 OBJS-$(HAVE_ARMV5TE) += $(ARMV5TE-OBJS) $(ARMV5TE-OBJS-yes)
 OBJS-$(HAVE_ARMV6)   += $(ARMV6-OBJS)   $(ARMV6-OBJS-yes)
-OBJS-$(HAVE_VFP)     += $(VFP-OBJS)     $(VFP-OBJS-yes)
+OBJS-$(HAVE_ARMVFP)  += $(ARMVFP-OBJS)  $(ARMVFP-OBJS-yes)
 OBJS-$(HAVE_NEON)    += $(NEON-OBJS)    $(NEON-OBJS-yes)

-OBJS-$(HAVE_MIPSFPU)   += $(MIPSFPU-OBJS)    $(MIPSFPU-OBJS-yes)
-OBJS-$(HAVE_MIPS32R2)  += $(MIPS32R2-OBJS)   $(MIPS32R2-OBJS-yes)
-OBJS-$(HAVE_MIPSDSPR1) += $(MIPSDSPR1-OBJS)  $(MIPSDSPR1-OBJS-yes)
-OBJS-$(HAVE_MIPSDSPR2) += $(MIPSDSPR2-OBJS)  $(MIPSDSPR2-OBJS-yes)
+OBJS-$(HAVE_MMI)     += $(MMI-OBJS)     $(MMI-OBJS-yes)

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

--- a/cmdutils.c
+++ b/cmdutils.c
--- a/cmdutils.h
+++ b/cmdutils.h
@@ -43,22 +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 struct SwsContext *sws_opts;
-extern AVDictionary *swr_opts;
-extern AVDictionary *format_opts, *codec_opts, *resample_opts;
-extern int hide_banner;
-
-/**
- * Register a program-specific cleanup routine.
- */
-void register_exit(void (*cb)(int ret));
-
-/**
- * Wraps exit with a program-specific cleanup routine.
- */
-void exit_program(int ret);
+extern struct SwrContext *swr_opts;
+extern AVDictionary *format_opts, *codec_opts;

 /**
 * Initialize the cmdutils option system, in particular
@@ -77,38 +71,29 @@ void uninit_opts(void);
 */
 void log_callback_help(void* ptr, int level, const char* fmt, va_list vl);

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

 /**
 * Set the libav* libraries log level.
 */
-int opt_loglevel(void *optctx, const char *opt, const char *arg);
+int opt_loglevel(const char *opt, const char *arg);

 int opt_report(const char *opt);

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

-int opt_codec_debug(void *optctx, const char *opt, const char *arg);
+int opt_cpuflags(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
+int opt_codec_debug(const char *opt, const char *arg);

 /**
 * Limit the execution time.
 */
-int opt_timelimit(void *optctx, const char *opt, const char *arg);
+int opt_timelimit(const char *opt, const char *arg);

 /**
 * Parse a string and return its corresponding value as a double.
@@ -138,7 +123,7 @@ double parse_number_or_die(const char *context, const char *numstr, int type,
 * not zero timestr is interpreted as a duration, otherwise as a
 * date
 *
- * @see av_parse_time()
+ * @see parse_date()
 */
 int64_t parse_time_or_die(const char *context, const char *timestr,
                          int is_duration);
@@ -154,7 +139,7 @@ typedef struct SpecifierOpt {
    } u;
 } SpecifierOpt;

-typedef struct OptionDef {
+typedef struct {
    const char *name;
    int flags;
 #define HAS_ARG    0x0001
@@ -163,42 +148,32 @@ typedef struct OptionDef {
 #define OPT_STRING 0x0008
 #define OPT_VIDEO  0x0010
 #define OPT_AUDIO  0x0020
+#define OPT_GRAB   0x0040
 #define OPT_INT    0x0080
 #define OPT_FLOAT  0x0100
 #define OPT_SUBTITLE 0x0200
 #define OPT_INT64  0x0400
 #define OPT_EXIT   0x0800
 #define OPT_DATA   0x1000
-#define OPT_PERFILE  0x2000     /* the option is per-file (currently ffmpeg-only).
-                                   implied by OPT_OFFSET or OPT_SPEC */
+#define OPT_FUNC2  0x2000
 #define OPT_OFFSET 0x4000       /* option is specified as an offset in a passed optctx */
 #define OPT_SPEC   0x8000       /* option is to be stored in an array of SpecifierOpt.
                                   Implies OPT_OFFSET. Next element after the offset is
                                   an int containing element count in the array. */
 #define OPT_TIME  0x10000
 #define OPT_DOUBLE 0x20000
-#define OPT_INPUT  0x40000
-#define OPT_OUTPUT 0x80000
     union {
        void *dst_ptr;
-        int (*func_arg)(void *, const char *, const char *);
+        int (*func_arg)(const char *, const char *);
+        int (*func2_arg)(void *, const char *, const char *);
        size_t off;
    } u;
    const char *help;
    const char *argname;
 } OptionDef;

-/**
- * Print help for all options matching specified flags.
- *
- * @param options a list of options
- * @param msg title of this group. Only printed if at least one option matches.
- * @param req_flags print only options which have all those flags set.
- * @param rej_flags don't print options which have any of those flags set.
- * @param alt_flags print only options that have at least one of those flags set
- */
-void show_help_options(const OptionDef *options, const char *msg, int req_flags,
-                       int rej_flags, int alt_flags);
+void show_help_options(const OptionDef *options, const char *msg, int mask,
+                       int value);

 /**
 * Show help for all options with given flags in class and all its
@@ -206,23 +181,10 @@ void show_help_options(const OptionDef *options, const char *msg, int req_flags,
 */
 void show_help_children(const AVClass *class, int flags);

-/**
- * Per-fftool specific help handler. Implemented in each
- * fftool, called by show_help().
- */
-void show_help_default(const char *opt, const char *arg);
-
-/**
- * Generic -h handler common to all fftools.
- */
-int show_help(void *optctx, const char *opt, const char *arg);
-
 /**
 * Parse the command line arguments.
 *
 * @param optctx an opaque options context
- * @param argc   number of command line arguments
- * @param argv   values of command line arguments
 * @param options Array with the definitions required to interpret every
 * option of the form: -option_name [argument]
 * @param parse_arg_function Name of the function called to process every
@@ -240,101 +202,6 @@ void parse_options(void *optctx, int argc, char **argv, const OptionDef *options
 int parse_option(void *optctx, const char *opt, const char *arg,
                 const OptionDef *options);

-/**
- * An option extracted from the commandline.
- * Cannot use AVDictionary because of options like -map which can be
- * used multiple times.
- */
-typedef struct Option {
-    const OptionDef  *opt;
-    const char       *key;
-    const char       *val;
-} Option;
-
-typedef struct OptionGroupDef {
-    /**< group name */
-    const char *name;
-    /**
-     * Option to be used as group separator. Can be NULL for groups which
-     * are terminated by a non-option argument (e.g. ffmpeg output files)
-     */
-    const char *sep;
-    /**
-     * Option flags that must be set on each option that is
-     * applied to this group
-     */
-    int flags;
-} OptionGroupDef;
-
-typedef struct OptionGroup {
-    const OptionGroupDef *group_def;
-    const char *arg;
-
-    Option *opts;
-    int  nb_opts;
-
-    AVDictionary *codec_opts;
-    AVDictionary *format_opts;
-    AVDictionary *resample_opts;
-    struct SwsContext *sws_opts;
-    AVDictionary *swr_opts;
-} OptionGroup;
-
-/**
- * A list of option groups that all have the same group type
- * (e.g. input files or output files)
- */
-typedef struct OptionGroupList {
-    const OptionGroupDef *group_def;
-
-    OptionGroup *groups;
-    int       nb_groups;
-} OptionGroupList;
-
-typedef struct OptionParseContext {
-    OptionGroup global_opts;
-
-    OptionGroupList *groups;
-    int           nb_groups;
-
-    /* parsing state */
-    OptionGroup cur_group;
-} OptionParseContext;
-
-/**
- * Parse an options group and write results into optctx.
- *
- * @param optctx an app-specific options context. NULL for global options group
- */
-int parse_optgroup(void *optctx, OptionGroup *g);
-
-/**
- * Split the commandline into an intermediate form convenient for further
- * processing.
- *
- * The commandline is assumed to be composed of options which either belong to a
- * group (those with OPT_SPEC, OPT_OFFSET or OPT_PERFILE) or are global
- * (everything else).
- *
- * A group (defined by an OptionGroupDef struct) is a sequence of options
- * terminated by either a group separator option (e.g. -i) or a parameter that
- * is not an option (doesn't start with -). A group without a separator option
- * must always be first in the supplied groups list.
- *
- * All options within the same group are stored in one OptionGroup struct in an
- * OptionGroupList, all groups with the same group definition are stored in one
- * OptionGroupList in OptionParseContext.groups. The order of group lists is the
- * same as the order of group definitions.
- */
-int split_commandline(OptionParseContext *octx, int argc, char *argv[],
-                      const OptionDef *options,
-                      const OptionGroupDef *groups, int nb_groups);
-
-/**
- * Free all allocated memory in an OptionParseContext.
- */
-void uninit_parse_context(OptionParseContext *octx);
-
 /**
 * Find the '-loglevel' option in the command line args and apply it.
 */
@@ -363,16 +230,12 @@ int check_stream_specifier(AVFormatContext *s, AVStream *st, const char *spec);
 * Create a new options dictionary containing only the options from
 * opts which apply to the codec with ID codec_id.
 *
- * @param opts     dictionary to place options in
- * @param codec_id ID of the codec that should be filtered for
 * @param s Corresponding format context.
 * @param st A stream from s for which the options should be filtered.
- * @param codec The particular codec for which the options should be filtered.
- *              If null, the default one is looked up according to the codec id.
 * @return a pointer to the created dictionary
 */
-AVDictionary *filter_codec_opts(AVDictionary *opts, enum AVCodecID codec_id,
-                                AVFormatContext *s, AVStream *st, AVCodec *codec);
+AVDictionary *filter_codec_opts(AVDictionary *opts, AVCodec *codec,
+                                AVFormatContext *s, AVStream *st);

 /**
 * Setup AVCodecContext options for avformat_find_stream_info().
@@ -412,94 +275,62 @@ void show_banner(int argc, char **argv, const OptionDef *options);
 * libraries.
 * This option processing function does not utilize the arguments.
 */
-int show_version(void *optctx, const char *opt, const char *arg);
-
-/**
- * Print the build configuration of the program to stdout. The contents
- * depend on the definition of FFMPEG_CONFIGURATION.
- * This option processing function does not utilize the arguments.
- */
-int show_buildconf(void *optctx, const char *opt, const char *arg);
+int opt_version(const char *opt, const char *arg);

 /**
 * Print the license of the program to stdout. The license depends on
 * the license of the libraries compiled into the program.
 * This option processing function does not utilize the arguments.
 */
-int show_license(void *optctx, const char *opt, const char *arg);
+int opt_license(const char *opt, const char *arg);

 /**
 * Print a listing containing all the formats supported by the
 * program.
 * This option processing function does not utilize the arguments.
 */
-int show_formats(void *optctx, const char *opt, const char *arg);
+int opt_formats(const char *opt, const char *arg);

 /**
 * Print a listing containing all the codecs supported by the
 * program.
 * This option processing function does not utilize the arguments.
 */
-int show_codecs(void *optctx, const char *opt, const char *arg);
-
-/**
- * Print a listing containing all the decoders supported by the
- * program.
- */
-int show_decoders(void *optctx, const char *opt, const char *arg);
-
-/**
- * Print a listing containing all the encoders supported by the
- * program.
- */
-int show_encoders(void *optctx, const char *opt, const char *arg);
+int opt_codecs(const char *opt, const char *arg);

 /**
 * Print a listing containing all the filters supported by the
 * program.
 * This option processing function does not utilize the arguments.
 */
-int show_filters(void *optctx, const char *opt, const char *arg);
+int opt_filters(const char *opt, const char *arg);

 /**
 * Print a listing containing all the bit stream filters supported by the
 * program.
 * This option processing function does not utilize the arguments.
 */
-int show_bsfs(void *optctx, const char *opt, const char *arg);
+int opt_bsfs(const char *opt, const char *arg);

 /**
 * Print a listing containing all the protocols supported by the
 * program.
 * This option processing function does not utilize the arguments.
 */
-int show_protocols(void *optctx, const char *opt, const char *arg);
+int opt_protocols(const char *opt, const char *arg);

 /**
 * Print a listing containing all the pixel formats supported by the
 * program.
 * This option processing function does not utilize the arguments.
 */
-int show_pix_fmts(void *optctx, const char *opt, const char *arg);
-
-/**
- * Print a listing containing all the standard channel layouts supported by
- * the program.
- * This option processing function does not utilize the arguments.
- */
-int show_layouts(void *optctx, const char *opt, const char *arg);
+int opt_pix_fmts(const char *opt, const char *arg);

 /**
 * Print a listing containing all the sample formats supported by the
 * program.
 */
-int show_sample_fmts(void *optctx, const char *opt, const char *arg);
-
-/**
- * Print a listing containing all the color names and values recognized
- * by the program.
- */
-int show_colors(void *optctx, const char *opt, const char *arg);
+int show_sample_fmts(const char *opt, const char *arg);

 /**
 * Return a positive value if a line read from standard input
@@ -511,10 +342,9 @@ 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
+ * @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);
@@ -540,39 +370,20 @@ int cmdutils_read_file(const char *filename, char **bufptr, size_t *size);
 FILE *get_preset_file(char *filename, size_t filename_size,
                      const char *preset_name, int is_path, const char *codec_name);

+/**
+ * Do all the necessary cleanup and abort.
+ * This function is implemented in the avtools, not cmdutils.
+ */
+void exit_program(int ret);
+
 /**
 * Realloc array to hold new_size elements of elem_size.
- * Calls exit() on failure.
+ * Calls exit_program() on failure.
 *
- * @param array array to reallocate
 * @param elem_size size in bytes of each element
 * @param size new element count will be written here
- * @param new_size number of elements to place in reallocated array
 * @return reallocated array
 */
 void *grow_array(void *array, int elem_size, int *size, int new_size);

-#define media_type_string av_get_media_type_string
-
-#define GROW_ARRAY(array, nb_elems)\
-    array = grow_array(array, sizeof(*array), &nb_elems, nb_elems + 1)
-
-#define GET_PIX_FMT_NAME(pix_fmt)\
-    const char *name = av_get_pix_fmt_name(pix_fmt);
-
-#define GET_SAMPLE_FMT_NAME(sample_fmt)\
-    const char *name = av_get_sample_fmt_name(sample_fmt)
-
-#define GET_SAMPLE_RATE_NAME(rate)\
-    char name[16];\
-    snprintf(name, sizeof(name), "%d", rate);
-
-#define GET_CH_LAYOUT_NAME(ch_layout)\
-    char name[16];\
-    snprintf(name, sizeof(name), "0x%"PRIx64, ch_layout);
-
-#define GET_CH_LAYOUT_DESC(ch_layout)\
-    char name[128];\
-    av_get_channel_layout_string(name, sizeof(name), 0, ch_layout);
-
 #endif /* CMDUTILS_H */
--- a/cmdutils_common_opts.h
+++ b/cmdutils_common_opts.h
@@ -1,28 +1,20 @@
-    { "L"          , OPT_EXIT, {.func_arg = show_license},      "show license" },
-    { "h"          , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
-    { "?"          , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
-    { "help"       , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
-    { "-help"      , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
-    { "version"    , OPT_EXIT, {.func_arg = show_version},      "show version" },
-    { "buildconf"  , OPT_EXIT, {.func_arg = show_buildconf},    "show build configuration" },
-    { "formats"    , OPT_EXIT, {.func_arg = show_formats  },    "show available formats" },
-    { "codecs"     , OPT_EXIT, {.func_arg = show_codecs   },    "show available codecs" },
-    { "decoders"   , OPT_EXIT, {.func_arg = show_decoders },    "show available decoders" },
-    { "encoders"   , OPT_EXIT, {.func_arg = show_encoders },    "show available encoders" },
-    { "bsfs"       , OPT_EXIT, {.func_arg = show_bsfs     },    "show available bit stream filters" },
-    { "protocols"  , OPT_EXIT, {.func_arg = show_protocols},    "show available protocols" },
-    { "filters"    , OPT_EXIT, {.func_arg = show_filters  },    "show available filters" },
-    { "pix_fmts"   , OPT_EXIT, {.func_arg = show_pix_fmts },    "show available pixel formats" },
-    { "layouts"    , OPT_EXIT, {.func_arg = show_layouts  },    "show standard channel layouts" },
+    { "L", OPT_EXIT, {(void*)opt_license}, "show license" },
+    { "h", OPT_EXIT, {(void*)opt_help}, "show help" },
+    { "?", OPT_EXIT, {(void*)opt_help}, "show help" },
+    { "help", OPT_EXIT, {(void*)opt_help}, "show help" },
+    { "-help", OPT_EXIT, {(void*)opt_help}, "show help" },
+    { "version", OPT_EXIT, {(void*)opt_version}, "show version" },
+    { "formats"  , OPT_EXIT, {(void*)opt_formats  }, "show available formats" },
+    { "codecs"   , OPT_EXIT, {(void*)opt_codecs   }, "show available codecs" },
+    { "bsfs"     , OPT_EXIT, {(void*)opt_bsfs     }, "show available bit stream filters" },
+    { "protocols", OPT_EXIT, {(void*)opt_protocols}, "show available protocols" },
+    { "filters",   OPT_EXIT, {(void*)opt_filters  }, "show available filters" },
+    { "pix_fmts" , OPT_EXIT, {(void*)opt_pix_fmts }, "show available pixel formats" },
    { "sample_fmts", OPT_EXIT, {.func_arg = show_sample_fmts }, "show available audio sample formats" },
-    { "colors"     , OPT_EXIT, {.func_arg = show_colors },      "show available color names" },
-    { "loglevel"   , HAS_ARG,  {.func_arg = opt_loglevel},      "set logging level", "loglevel" },
-    { "v",           HAS_ARG,  {.func_arg = opt_loglevel},      "set logging level", "loglevel" },
-    { "report"     , 0,        {(void*)opt_report}, "generate a report" },
-    { "max_alloc"  , HAS_ARG,  {.func_arg = opt_max_alloc},     "set maximum size of a single allocated block", "bytes" },
-    { "cpuflags"   , HAS_ARG | OPT_EXPERT, { .func_arg = opt_cpuflags }, "force specific cpu flags", "flags" },
-    { "hide_banner", OPT_BOOL | OPT_EXPERT, {&hide_banner},     "do not show program banner", "hide_banner" },
-#if CONFIG_OPENCL
-    { "opencl_bench", OPT_EXIT, {.func_arg = opt_opencl_bench}, "run benchmark on all OpenCL devices and show results" },
-    { "opencl_options", HAS_ARG, {.func_arg = opt_opencl},      "set OpenCL environment options" },
-#endif
+    { "loglevel", HAS_ARG, {(void*)opt_loglevel}, "set libav* logging level", "loglevel" },
+    { "v", HAS_ARG, {(void*)opt_loglevel}, "set libav* logging level", "loglevel" },
+    { "debug", HAS_ARG, {(void*)opt_codec_debug}, "set debug flags", "flags" },
+    { "fdebug", HAS_ARG, {(void*)opt_codec_debug}, "set debug flags", "flags" },
+    { "report", 0, {(void*)opt_report}, "generate a report" },
+    { "max_alloc", HAS_ARG, {(void*)opt_max_alloc}, "set maximum size of a single allocated block", "bytes" },
+    { "cpuflags", HAS_ARG | OPT_EXPERT, {(void*)opt_cpuflags}, "force specific cpu flags", "flags" },
--- a/cmdutils_opencl.c
+++ b/cmdutils_opencl.c
@@ -1,274 +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 "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();
-    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() - start)/OPENCL_NB_ITER;
-end:
-    if (kernel)
-        clReleaseKernel(kernel);
-    if (program)
-        clReleaseProgram(program);
-    if (cl_inbuf)
-        clReleaseMemObject(cl_inbuf);
-    if (cl_outbuf)
-        clReleaseMemObject(cl_outbuf);
-    if (cl_mask)
-        clReleaseMemObject(cl_mask);
-    av_free(inbuf);
-    av_free(mask);
-    return ret;
-}
-
-static int compare_ocl_device_desc(const void *a, const void *b)
-{
-    return ((OpenCLDeviceBenchmark*)a)->runtime - ((OpenCLDeviceBenchmark*)b)->runtime;
-}
-
-int opt_opencl_bench(void *optctx, const char *opt, const char *arg)
-{
-    int i, j, nb_devices = 0, count = 0;
-    int64_t score = 0;
-    AVOpenCLDeviceList *device_list;
-    AVOpenCLDeviceNode *device_node = NULL;
-    OpenCLDeviceBenchmark *devices = NULL;
-    cl_platform_id platform;
-
-    av_opencl_get_device_list(&device_list);
-    for (i = 0; i < device_list->platform_num; i++)
-        nb_devices += device_list->platform_node[i]->device_num;
-    if (!nb_devices) {
-        av_log(NULL, AV_LOG_ERROR, "No OpenCL device detected!\n");
-        return AVERROR(EINVAL);
-    }
-    if (!(devices = av_malloc(sizeof(OpenCLDeviceBenchmark) * nb_devices))) {
-        av_log(NULL, AV_LOG_ERROR, "Could not allocate buffer\n");
-        return AVERROR(ENOMEM);
-    }
-
-    for (i = 0; i < device_list->platform_num; i++) {
-        for (j = 0; j < device_list->platform_node[i]->device_num; j++) {
-            device_node = device_list->platform_node[i]->device_node[j];
-            platform = device_list->platform_node[i]->platform_id;
-            score = av_opencl_benchmark(device_node, platform, run_opencl_bench);
-            if (score > 0) {
-                devices[count].platform_idx = i;
-                devices[count].device_idx = j;
-                devices[count].runtime = score;
-                strcpy(devices[count].device_name, device_node->device_name);
-                count++;
-            }
-        }
-    }
-    qsort(devices, count, sizeof(OpenCLDeviceBenchmark), compare_ocl_device_desc);
-    fprintf(stderr, "platform_idx\tdevice_idx\tdevice_name\truntime\n");
-    for (i = 0; i < count; i++)
-        fprintf(stdout, "%d\t%d\t%s\t%"PRId64"\n",
-                devices[i].platform_idx, devices[i].device_idx,
-                devices[i].device_name, devices[i].runtime);
-
-    av_opencl_free_device_list(&device_list);
-    av_free(devices);
-    return 0;
-}
-
-int opt_opencl(void *optctx, const char *opt, const char *arg)
-{
-    char *key, *value;
-    const char *opts = arg;
-    int ret = 0;
-    while (*opts) {
-        ret = av_opt_get_key_value(&opts, "=", ":", 0, &key, &value);
-        if (ret < 0)
-            return ret;
-        ret = av_opencl_set_option(key, value);
-        if (ret < 0)
-            return ret;
-        if (*opts)
-            opts++;
-    }
-    return ret;
-}
--- a/common.mak
+++ b/common.mak
@@ -10,9 +10,8 @@ ifndef SUBDIR
 ifndef V
 Q      = @
 ECHO   = printf "$(1)\t%s\n" $(2)
-BRIEF  = CC CXX HOSTCC HOSTLD AS YASM AR LD STRIP CP WINDRES
-SILENT = DEPCC DEPHOSTCC DEPAS DEPYASM RANLIB RM
-
+BRIEF  = CC CXX AS YASM AR LD HOSTCC STRIP CP
+SILENT = DEPCC YASMDEP RM RANLIB
 MSG    = $@
 M      = @$(call ECHO,$(TAG),$@);
 $(foreach VAR,$(BRIEF), \
@@ -27,23 +26,20 @@ ALLFFLIBS = avcodec avdevice avfilter avformat avresample avutil postproc swscal
 IFLAGS     := -I. -I$(SRC_PATH)/
 CPPFLAGS   := $(IFLAGS) $(CPPFLAGS)
 CFLAGS     += $(ECFLAGS)
-CCFLAGS     = $(CPPFLAGS) $(CFLAGS)
-ASFLAGS    := $(CPPFLAGS) $(ASFLAGS)
-CXXFLAGS   += $(CPPFLAGS) $(CFLAGS)
-YASMFLAGS  += $(IFLAGS:%=%/) -Pconfig.asm
-
-HOSTCCFLAGS = $(IFLAGS) $(HOSTCPPFLAGS) $(HOSTCFLAGS)
-LDFLAGS    := $(ALLFFLIBS:%=$(LD_PATH)lib%) $(LDFLAGS)
+CCFLAGS     = $(CFLAGS)
+CXXFLAGS   := $(CFLAGS) $(CXXFLAGS)
+YASMFLAGS  += $(IFLAGS) -I$(SRC_PATH)/libavutil/x86/ -Pconfig.asm
+HOSTCFLAGS += $(IFLAGS)
+LDFLAGS    := $(ALLFFLIBS:%=-Llib%) $(LDFLAGS)

 define COMPILE
-       $(call $(1)DEP,$(1))
-       $($(1)) $($(1)FLAGS) $($(1)_DEPFLAGS) $($(1)_C) $($(1)_O) $<
+       $($(1)DEP)
+       $($(1)) $(CPPFLAGS) $($(1)FLAGS) $($(1)_DEPFLAGS) -c $($(1)_O) $<
 endef

 COMPILE_C = $(call COMPILE,CC)
 COMPILE_CXX = $(call COMPILE,CXX)
 COMPILE_S = $(call COMPILE,AS)
-COMPILE_HOSTC = $(call COMPILE,HOSTCC)

 %.o: %.c
 	$(COMPILE_C)
@@ -57,17 +53,8 @@ COMPILE_HOSTC = $(call COMPILE,HOSTCC)
 %.o: %.S
 	$(COMPILE_S)

-%_host.o: %.c
-	$(COMPILE_HOSTC)
-
-%.o: %.rc
-	$(WINDRES) $(IFLAGS) --preprocessor "$(DEPWINDRES) -E -xc-header -DRC_INVOKED $(CC_DEPFLAGS)" -o $@ $<
-
-%.i: %.c
-	$(CC) $(CCFLAGS) $(CC_E) $<
-
-%.h.c:
-	$(Q)echo '#include "$*.h"' >$@
+%.ho: %.h
+	$(CC) $(CPPFLAGS) $(CFLAGS) -Wno-unused -c -o $@ -x c $<

 %.ver: %.v
 	$(Q)sed 's/$$MAJOR/$($(basename $(@F))_VERSION_MAJOR)/' $^ > $@
@@ -89,15 +76,13 @@ endif
 include $(SRC_PATH)/arch.mak

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

-LDLIBS       = $(FFLIBS:%=%$(BUILDSUF))
-FFEXTRALIBS := $(LDLIBS:%=$(LD_LIB)) $(EXTRALIBS)
+FFEXTRALIBS := $(FFLIBS:%=-l%$(BUILDSUF)) $(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)
@@ -105,47 +90,31 @@ HOSTPROGS := $(HOSTPROGS:%=$(SUBDIR)%$(HOSTEXESUF))
 TOOLS     += $(TOOLS-yes)
 TOOLOBJS  := $(TOOLS:%=tools/%.o)
 TOOLS     := $(TOOLS:%=tools/%$(EXESUF))
-HEADERS   += $(HEADERS-yes)

-PATH_LIBNAME = $(foreach NAME,$(1),lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME))
-DEP_LIBS := $(foreach lib,$(FFLIBS),$(call PATH_LIBNAME,$(lib)))
+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))
 SKIPHEADERS += $(ARCH_HEADERS:%=$(ARCH)/%) $(SKIPHEADERS-)
 SKIPHEADERS := $(SKIPHEADERS:%=$(SUBDIR)%)
-HOBJS        = $(filter-out $(SKIPHEADERS:.h=.h.o),$(ALLHEADERS:.h=.h.o))
-checkheaders: $(HOBJS)
-.SECONDARY:   $(HOBJS:.o=.c)
+checkheaders: $(filter-out $(SKIPHEADERS:.h=.ho),$(ALLHEADERS:.h=.ho))

 alltools: $(TOOLS)

 $(HOSTOBJS): %.o: %.c
-	$(COMPILE_HOSTC)
+	$(HOSTCC) $(HOSTCFLAGS) -c -o $@ $<

 $(HOSTPROGS): %$(HOSTEXESUF): %.o
-	$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $^ $(HOSTLIBS)
+	$(HOSTCC) $(HOSTLDFLAGS) -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) $(HOSTOBJS) $(TESTOBJS))

-CLEANSUFFIXES     = *.d *.o *~ *.h.c *.map *.ver *.ho *.gcno *.gcda
+CLEANSUFFIXES     = *.d *.o *~ *.ho *.map *.ver *.gcno *.gcda
 DISTCLEANSUFFIXES = *.pc
 LIBSUFFIXES       = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a

-define RULES
-clean::
-	$(RM) $(OBJS) $(OBJS:.o=.d)
-	$(RM) $(HOSTPROGS)
-	$(RM) $(TOOLS)
-endef
-
-$(eval $(RULES))
-
-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d) $(SLIBOBJS:.o=.d))
+-include $(wildcard $(OBJS:.o=.d) $(TESTOBJS:.o=.d))
--- a/compat/aix/math.h
+++ b/compat/aix/math.h
@@ -1,31 +0,0 @@
-/*
- * 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
- */
-
-#ifndef FFMPEG_COMPAT_AIX_MATH_H
-#define FFMPEG_COMPAT_AIX_MATH_H
-
-#define class class_in_math_h_causes_problems
-
-#include_next <math.h>
-
-#undef class
-
-#endif /* FFMPEG_COMPAT_AIX_MATH_H */
--- a/compat/avisynth/avisynth_c.h
+++ b/compat/avisynth/avisynth_c.h
@@ -1,879 +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.
-
-// NOTE: this is a partial update of the Avisynth C interface to recognize
-// new color spaces added in Avisynth 2.60. By no means is this document
-// completely Avisynth 2.60 compliant.
-
-#ifndef __AVISYNTH_C__
-#define __AVISYNTH_C__
-
-#ifdef __cplusplus
-#  define EXTERN_C extern "C"
-#else
-#  define EXTERN_C
-#endif
-
-#define AVSC_USE_STDCALL 1
-
-#ifndef AVSC_USE_STDCALL
-#  define AVSC_CC __cdecl
-#else
-#  define AVSC_CC __stdcall
-#endif
-
-#define AVSC_INLINE static __inline
-
-#ifdef AVISYNTH_C_EXPORTS
-#  define AVSC_EXPORT EXTERN_C
-#  define AVSC_API(ret, name) EXTERN_C __declspec(dllexport) ret AVSC_CC name
-#else
-#  define AVSC_EXPORT EXTERN_C __declspec(dllexport)
-#  ifndef AVSC_NO_DECLSPEC
-#    define AVSC_API(ret, name) EXTERN_C __declspec(dllimport) ret AVSC_CC name
-#  else
-#    define AVSC_API(ret, name) typedef ret (AVSC_CC *name##_func)
-#  endif
-#endif
-
-typedef unsigned char BYTE;
-#ifdef __GNUC__
-typedef long long int INT64;
-#else
-typedef __int64 INT64;
-#endif
-
-
-/////////////////////////////////////////////////////////////////////
-//
-// Constants
-//
-
-#ifndef __AVISYNTH_H__
-enum { AVISYNTH_INTERFACE_VERSION = 4 };
-#endif
-
-enum {AVS_SAMPLE_INT8  = 1<<0,
-      AVS_SAMPLE_INT16 = 1<<1,
-      AVS_SAMPLE_INT24 = 1<<2,
-      AVS_SAMPLE_INT32 = 1<<3,
-      AVS_SAMPLE_FLOAT = 1<<4};
-
-enum {AVS_PLANAR_Y=1<<0,
-      AVS_PLANAR_U=1<<1,
-      AVS_PLANAR_V=1<<2,
-      AVS_PLANAR_ALIGNED=1<<3,
-      AVS_PLANAR_Y_ALIGNED=AVS_PLANAR_Y|AVS_PLANAR_ALIGNED,
-      AVS_PLANAR_U_ALIGNED=AVS_PLANAR_U|AVS_PLANAR_ALIGNED,
-      AVS_PLANAR_V_ALIGNED=AVS_PLANAR_V|AVS_PLANAR_ALIGNED,
-      AVS_PLANAR_A=1<<4,
-      AVS_PLANAR_R=1<<5,
-      AVS_PLANAR_G=1<<6,
-      AVS_PLANAR_B=1<<7,
-      AVS_PLANAR_A_ALIGNED=AVS_PLANAR_A|AVS_PLANAR_ALIGNED,
-      AVS_PLANAR_R_ALIGNED=AVS_PLANAR_R|AVS_PLANAR_ALIGNED,
-      AVS_PLANAR_G_ALIGNED=AVS_PLANAR_G|AVS_PLANAR_ALIGNED,
-      AVS_PLANAR_B_ALIGNED=AVS_PLANAR_B|AVS_PLANAR_ALIGNED};
-
-  // Colorspace properties.
-enum {AVS_CS_BGR = 1<<28,
-      AVS_CS_YUV = 1<<29,
-      AVS_CS_INTERLEAVED = 1<<30,
-      AVS_CS_PLANAR = 1<<31,
-
-      AVS_CS_SHIFT_SUB_WIDTH   = 0,
-      AVS_CS_SHIFT_SUB_HEIGHT  = 1 << 3,
-      AVS_CS_SHIFT_SAMPLE_BITS = 1 << 4,
-
-      AVS_CS_SUB_WIDTH_MASK    = 7 << AVS_CS_SHIFT_SUB_WIDTH,
-      AVS_CS_SUB_WIDTH_1       = 3 << AVS_CS_SHIFT_SUB_WIDTH, // YV24
-      AVS_CS_SUB_WIDTH_2       = 0 << AVS_CS_SHIFT_SUB_WIDTH, // YV12, I420, YV16
-      AVS_CS_SUB_WIDTH_4       = 1 << AVS_CS_SHIFT_SUB_WIDTH, // YUV9, YV411
-
-      AVS_CS_VPLANEFIRST       = 1 << 3, // YV12, YV16, YV24, YV411, YUV9
-      AVS_CS_UPLANEFIRST       = 1 << 4, // I420
-
-      AVS_CS_SUB_HEIGHT_MASK   = 7 << AVS_CS_SHIFT_SUB_HEIGHT,
-      AVS_CS_SUB_HEIGHT_1      = 3 << AVS_CS_SHIFT_SUB_HEIGHT, // YV16, YV24, YV411
-      AVS_CS_SUB_HEIGHT_2      = 0 << AVS_CS_SHIFT_SUB_HEIGHT, // YV12, I420
-      AVS_CS_SUB_HEIGHT_4      = 1 << AVS_CS_SHIFT_SUB_HEIGHT, // YUV9
-
-      AVS_CS_SAMPLE_BITS_MASK  = 7 << AVS_CS_SHIFT_SAMPLE_BITS,
-      AVS_CS_SAMPLE_BITS_8     = 0 << AVS_CS_SHIFT_SAMPLE_BITS,
-      AVS_CS_SAMPLE_BITS_16    = 1 << AVS_CS_SHIFT_SAMPLE_BITS,
-      AVS_CS_SAMPLE_BITS_32    = 2 << AVS_CS_SHIFT_SAMPLE_BITS,
-
-      AVS_CS_PLANAR_MASK       = AVS_CS_PLANAR | AVS_CS_INTERLEAVED | AVS_CS_YUV | AVS_CS_BGR | AVS_CS_SAMPLE_BITS_MASK | AVS_CS_SUB_HEIGHT_MASK | AVS_CS_SUB_WIDTH_MASK,
-      AVS_CS_PLANAR_FILTER     = ~( AVS_CS_VPLANEFIRST | AVS_CS_UPLANEFIRST )};
-
-  // Specific colorformats
-enum {
-  AVS_CS_UNKNOWN = 0,
-  AVS_CS_BGR24 = 1<<0 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
-  AVS_CS_BGR32 = 1<<1 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
-  AVS_CS_YUY2 = 1<<2 | AVS_CS_YUV | AVS_CS_INTERLEAVED,
-  //  AVS_CS_YV12  = 1<<3  Reserved
-  //  AVS_CS_I420  = 1<<4  Reserved
-  AVS_CS_RAW32 = 1<<5 | AVS_CS_INTERLEAVED,
-
-  AVS_CS_YV24  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_1,  // YVU 4:4:4 planar
-  AVS_CS_YV16  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_2,  // YVU 4:2:2 planar
-  AVS_CS_YV12  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_2 | AVS_CS_SUB_WIDTH_2,  // YVU 4:2:0 planar
-  AVS_CS_I420  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_UPLANEFIRST | AVS_CS_SUB_HEIGHT_2 | AVS_CS_SUB_WIDTH_2,  // YUV 4:2:0 planar
-  AVS_CS_IYUV  = AVS_CS_I420,
-  AVS_CS_YV411 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_4,  // YVU 4:1:1 planar
-  AVS_CS_YUV9  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_4 | AVS_CS_SUB_WIDTH_4,  // YVU 4:1:0 planar
-  AVS_CS_Y8    = AVS_CS_PLANAR | AVS_CS_INTERLEAVED | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8                                              // Y   4:0:0 planar
-};
-
-enum {
-  AVS_IT_BFF = 1<<0,
-  AVS_IT_TFF = 1<<1,
-  AVS_IT_FIELDBASED = 1<<2};
-
-enum {
-  AVS_FILTER_TYPE=1,
-  AVS_FILTER_INPUT_COLORSPACE=2,
-  AVS_FILTER_OUTPUT_TYPE=9,
-  AVS_FILTER_NAME=4,
-  AVS_FILTER_AUTHOR=5,
-  AVS_FILTER_VERSION=6,
-  AVS_FILTER_ARGS=7,
-  AVS_FILTER_ARGS_INFO=8,
-  AVS_FILTER_ARGS_DESCRIPTION=10,
-  AVS_FILTER_DESCRIPTION=11};
-
-enum {  //SUBTYPES
-  AVS_FILTER_TYPE_AUDIO=1,
-  AVS_FILTER_TYPE_VIDEO=2,
-  AVS_FILTER_OUTPUT_TYPE_SAME=3,
-  AVS_FILTER_OUTPUT_TYPE_DIFFERENT=4};
-
-enum {
-  AVS_CACHE_NOTHING=0,
-  AVS_CACHE_RANGE=1,
-  AVS_CACHE_ALL=2,
-  AVS_CACHE_AUDIO=3,
-  AVS_CACHE_AUDIO_NONE=4,
-  AVS_CACHE_AUDIO_AUTO=5
-  };
-
-#define AVS_FRAME_ALIGN 16
-
-typedef struct AVS_Clip AVS_Clip;
-typedef struct AVS_ScriptEnvironment AVS_ScriptEnvironment;
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_VideoInfo
-//
-
-// AVS_VideoInfo is layed out identicly to VideoInfo
-typedef struct AVS_VideoInfo {
-  int width, height;    // width=0 means no video
-  unsigned fps_numerator, fps_denominator;
-  int num_frames;
-
-  int pixel_type;
-
-  int audio_samples_per_second;   // 0 means no audio
-  int sample_type;
-  INT64 num_audio_samples;
-  int nchannels;
-
-  // Imagetype properties
-
-  int image_type;
-} AVS_VideoInfo;
-
-// useful functions of the above
-AVSC_INLINE int avs_has_video(const AVS_VideoInfo * p)
-        { return (p->width!=0); }
-
-AVSC_INLINE int avs_has_audio(const AVS_VideoInfo * p)
-        { return (p->audio_samples_per_second!=0); }
-
-AVSC_INLINE int avs_is_rgb(const AVS_VideoInfo * p)
-        { return !!(p->pixel_type&AVS_CS_BGR); }
-
-AVSC_INLINE int avs_is_rgb24(const AVS_VideoInfo * p)
-        { return (p->pixel_type&AVS_CS_BGR24)==AVS_CS_BGR24; } // Clear out additional properties
-
-AVSC_INLINE int avs_is_rgb32(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_BGR32) == AVS_CS_BGR32 ; }
-
-AVSC_INLINE int avs_is_yuv(const AVS_VideoInfo * p)
-        { return !!(p->pixel_type&AVS_CS_YUV ); }
-
-AVSC_INLINE int avs_is_yuy2(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_YUY2) == AVS_CS_YUY2; }
-
-AVSC_INLINE int avs_is_yv24(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV24  & AVS_CS_PLANAR_FILTER); }
-
-AVSC_INLINE int avs_is_yv16(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV16  & AVS_CS_PLANAR_FILTER); }
-
-AVSC_INLINE int avs_is_yv12(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV12  & AVS_CS_PLANAR_FILTER); }
-
-AVSC_INLINE int avs_is_yv411(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV411 & AVS_CS_PLANAR_FILTER); }
-
-AVSC_INLINE int avs_is_y8(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_Y8    & AVS_CS_PLANAR_FILTER); }
-
-AVSC_INLINE int avs_is_property(const AVS_VideoInfo * p, int property)
-        { return ((p->pixel_type & property)==property ); }
-
-AVSC_INLINE int avs_is_planar(const AVS_VideoInfo * p)
-        { return !!(p->pixel_type & AVS_CS_PLANAR); }
-
-AVSC_INLINE int avs_is_color_space(const AVS_VideoInfo * p, int c_space)
-        { return avs_is_planar(p) ? ((p->pixel_type & AVS_CS_PLANAR_MASK) == (c_space & AVS_CS_PLANAR_FILTER)) : ((p->pixel_type & c_space) == c_space); }
-
-AVSC_INLINE int avs_is_field_based(const AVS_VideoInfo * p)
-        { return !!(p->image_type & AVS_IT_FIELDBASED); }
-
-AVSC_INLINE int avs_is_parity_known(const AVS_VideoInfo * p)
-        { return ((p->image_type & AVS_IT_FIELDBASED)&&(p->image_type & (AVS_IT_BFF | AVS_IT_TFF))); }
-
-AVSC_INLINE int avs_is_bff(const AVS_VideoInfo * p)
-        { return !!(p->image_type & AVS_IT_BFF); }
-
-AVSC_INLINE int avs_is_tff(const AVS_VideoInfo * p)
-        { return !!(p->image_type & AVS_IT_TFF); }
-
-AVSC_INLINE int avs_bits_per_pixel(const AVS_VideoInfo * p)
-{
-  switch (p->pixel_type) {
-      case AVS_CS_BGR24: return 24;
-      case AVS_CS_BGR32: return 32;
-      case AVS_CS_YUY2:  return 16;
-      case AVS_CS_YV12:
-      case AVS_CS_I420:  return 12;
-      default:           return 0;
-    }
-}
-AVSC_INLINE int avs_bytes_from_pixels(const AVS_VideoInfo * p, int pixels)
-        { return pixels * (avs_bits_per_pixel(p)>>3); }   // Will work on planar images, but will return only luma planes
-
-AVSC_INLINE int avs_row_size(const AVS_VideoInfo * p)
-        { return avs_bytes_from_pixels(p,p->width); }  // Also only returns first plane on planar images
-
-AVSC_INLINE int avs_bmp_size(const AVS_VideoInfo * vi)
-        { if (avs_is_planar(vi)) {int p = vi->height * ((avs_row_size(vi)+3) & ~3); p+=p>>1; return p;  } return vi->height * ((avs_row_size(vi)+3) & ~3); }
-
-AVSC_INLINE int avs_samples_per_second(const AVS_VideoInfo * p)
-        { return p->audio_samples_per_second; }
-
-
-AVSC_INLINE int avs_bytes_per_channel_sample(const AVS_VideoInfo * p)
-{
-    switch (p->sample_type) {
-      case AVS_SAMPLE_INT8:  return sizeof(signed char);
-      case AVS_SAMPLE_INT16: return sizeof(signed short);
-      case AVS_SAMPLE_INT24: return 3;
-      case AVS_SAMPLE_INT32: return sizeof(signed int);
-      case AVS_SAMPLE_FLOAT: return sizeof(float);
-      default: return 0;
-    }
-}
-AVSC_INLINE int avs_bytes_per_audio_sample(const AVS_VideoInfo * p)
-        { return p->nchannels*avs_bytes_per_channel_sample(p);}
-
-AVSC_INLINE INT64 avs_audio_samples_from_frames(const AVS_VideoInfo * p, INT64 frames)
-        { return ((INT64)(frames) * p->audio_samples_per_second * p->fps_denominator / p->fps_numerator); }
-
-AVSC_INLINE int avs_frames_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
-        { return (int)(samples * (INT64)p->fps_numerator / (INT64)p->fps_denominator / (INT64)p->audio_samples_per_second); }
-
-AVSC_INLINE INT64 avs_audio_samples_from_bytes(const AVS_VideoInfo * p, INT64 bytes)
-        { return bytes / avs_bytes_per_audio_sample(p); }
-
-AVSC_INLINE INT64 avs_bytes_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
-        { return samples * avs_bytes_per_audio_sample(p); }
-
-AVSC_INLINE int avs_audio_channels(const AVS_VideoInfo * p)
-        { return p->nchannels; }
-
-AVSC_INLINE int avs_sample_type(const AVS_VideoInfo * p)
-        { return p->sample_type;}
-
-// useful mutator
-AVSC_INLINE void avs_set_property(AVS_VideoInfo * p, int property)
-        { p->image_type|=property; }
-
-AVSC_INLINE void avs_clear_property(AVS_VideoInfo * p, int property)
-        { p->image_type&=~property; }
-
-AVSC_INLINE void avs_set_field_based(AVS_VideoInfo * p, int isfieldbased)
-        { if (isfieldbased) p->image_type|=AVS_IT_FIELDBASED; else p->image_type&=~AVS_IT_FIELDBASED; }
-
-AVSC_INLINE void avs_set_fps(AVS_VideoInfo * p, unsigned numerator, unsigned denominator)
-{
-    unsigned x=numerator, y=denominator;
-    while (y) {   // find gcd
-      unsigned t = x%y; x = y; y = t;
-    }
-    p->fps_numerator = numerator/x;
-    p->fps_denominator = denominator/x;
-}
-
-AVSC_INLINE int avs_is_same_colorspace(AVS_VideoInfo * x, AVS_VideoInfo * y)
-{
-        return (x->pixel_type == y->pixel_type)
-                || (avs_is_yv12(x) && avs_is_yv12(y));
-}
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_VideoFrame
-//
-
-// VideoFrameBuffer holds information about a memory block which is used
-// for video data.  For efficiency, instances of this class are not deleted
-// when the refcount reaches zero; instead they're stored in a linked list
-// to be reused.  The instances are deleted when the corresponding AVS
-// file is closed.
-
-// AVS_VideoFrameBuffer is layed out identicly to VideoFrameBuffer
-// DO NOT USE THIS STRUCTURE DIRECTLY
-typedef struct AVS_VideoFrameBuffer {
-  BYTE * data;
-  int data_size;
-  // sequence_number is incremented every time the buffer is changed, so
-  // that stale views can tell they're no longer valid.
-  volatile long sequence_number;
-
-  volatile long refcount;
-} AVS_VideoFrameBuffer;
-
-// VideoFrame holds a "window" into a VideoFrameBuffer.
-
-// AVS_VideoFrame is layed out identicly to IVideoFrame
-// DO NOT USE THIS STRUCTURE DIRECTLY
-typedef struct AVS_VideoFrame {
-  volatile long refcount;
-  AVS_VideoFrameBuffer * vfb;
-  int offset, pitch, row_size, height, offsetU, offsetV, pitchUV;  // U&V offsets are from top of picture.
-  int row_sizeUV, heightUV;
-} AVS_VideoFrame;
-
-// Access functions for AVS_VideoFrame
-AVSC_INLINE int avs_get_pitch(const AVS_VideoFrame * p) {
-        return p->pitch;}
-
-AVSC_INLINE int avs_get_pitch_p(const AVS_VideoFrame * p, int plane) {
-  switch (plane) {
-  case AVS_PLANAR_U: case AVS_PLANAR_V: return p->pitchUV;}
-  return p->pitch;}
-
-AVSC_INLINE int avs_get_row_size(const AVS_VideoFrame * p) {
-        return p->row_size; }
-
-AVSC_INLINE int avs_get_row_size_p(const AVS_VideoFrame * p, int plane) {
-        int r;
-    switch (plane) {
-    case AVS_PLANAR_U: case AVS_PLANAR_V:
-                if (p->pitchUV) return p->row_sizeUV;
-                else            return 0;
-    case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED:
-                if (p->pitchUV) {
-                        r = (p->row_sizeUV+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
-                        if (r < p->pitchUV)
-                                return r;
-                        return p->row_sizeUV;
-                } else return 0;
-    case AVS_PLANAR_Y_ALIGNED:
-                r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
-                if (r <= p->pitch)
-                        return r;
-                return p->row_size;
-    }
-    return p->row_size;
-}
-
-AVSC_INLINE int avs_get_height(const AVS_VideoFrame * p) {
-        return p->height;}
-
-AVSC_INLINE int avs_get_height_p(const AVS_VideoFrame * p, int plane) {
-        switch (plane) {
-                case AVS_PLANAR_U: case AVS_PLANAR_V:
-                        if (p->pitchUV) return p->heightUV;
-                        return 0;
-        }
-        return p->height;}
-
-AVSC_INLINE const BYTE* avs_get_read_ptr(const AVS_VideoFrame * p) {
-        return p->vfb->data + p->offset;}
-
-AVSC_INLINE const BYTE* avs_get_read_ptr_p(const AVS_VideoFrame * p, int plane)
-{
-        switch (plane) {
-                case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
-                case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
-                default:           return p->vfb->data + p->offset;}
-}
-
-AVSC_INLINE int avs_is_writable(const AVS_VideoFrame * p) {
-        return (p->refcount == 1 && p->vfb->refcount == 1);}
-
-AVSC_INLINE BYTE* avs_get_write_ptr(const AVS_VideoFrame * p)
-{
-        if (avs_is_writable(p)) {
-                ++p->vfb->sequence_number;
-                return p->vfb->data + p->offset;
-        } else
-                return 0;
-}
-
-AVSC_INLINE BYTE* avs_get_write_ptr_p(const AVS_VideoFrame * p, int plane)
-{
-        if (plane==AVS_PLANAR_Y && avs_is_writable(p)) {
-                ++p->vfb->sequence_number;
-                return p->vfb->data + p->offset;
-        } else if (plane==AVS_PLANAR_Y) {
-                return 0;
-        } else {
-                switch (plane) {
-                        case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
-                        case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
-                        default:       return p->vfb->data + p->offset;
-                }
-        }
-}
-
-
-AVSC_API(void, avs_release_video_frame)(AVS_VideoFrame *);
-// makes a shallow copy of a video frame
-AVSC_API(AVS_VideoFrame *, avs_copy_video_frame)(AVS_VideoFrame *);
-
-#ifndef AVSC_NO_DECLSPEC
-AVSC_INLINE void avs_release_frame(AVS_VideoFrame * f)
-  {avs_release_video_frame(f);}
-AVSC_INLINE AVS_VideoFrame * avs_copy_frame(AVS_VideoFrame * f)
-  {return avs_copy_video_frame(f);}
-#endif
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_Value
-//
-
-// Treat AVS_Value as a fat pointer.  That is use avs_copy_value
-// and avs_release_value appropiaty as you would if AVS_Value was
-// a pointer.
-
-// To maintain source code compatibility with future versions of the
-// avisynth_c API don't use the AVS_Value directly.  Use the helper
-// functions below.
-
-// AVS_Value is layed out identicly to AVSValue
-typedef struct AVS_Value AVS_Value;
-struct AVS_Value {
-  short type;  // 'a'rray, 'c'lip, 'b'ool, 'i'nt, 'f'loat, 's'tring, 'v'oid, or 'l'ong
-               // for some function e'rror
-  short array_size;
-  union {
-    void * clip; // do not use directly, use avs_take_clip
-    char boolean;
-    int integer;
-    float floating_pt;
-    const char * string;
-    const AVS_Value * array;
-  } d;
-};
-
-// AVS_Value should be initilized with avs_void.
-// Should also set to avs_void after the value is released
-// with avs_copy_value.  Consider it the equalvent of setting
-// a pointer to NULL
-static const AVS_Value avs_void = {'v'};
-
-AVSC_API(void, avs_copy_value)(AVS_Value * dest, AVS_Value src);
-AVSC_API(void, avs_release_value)(AVS_Value);
-
-AVSC_INLINE int avs_defined(AVS_Value v) { return v.type != 'v'; }
-AVSC_INLINE int avs_is_clip(AVS_Value v) { return v.type == 'c'; }
-AVSC_INLINE int avs_is_bool(AVS_Value v) { return v.type == 'b'; }
-AVSC_INLINE int avs_is_int(AVS_Value v) { return v.type == 'i'; }
-AVSC_INLINE int avs_is_float(AVS_Value v) { return v.type == 'f' || v.type == 'i'; }
-AVSC_INLINE int avs_is_string(AVS_Value v) { return v.type == 's'; }
-AVSC_INLINE int avs_is_array(AVS_Value v) { return v.type == 'a'; }
-AVSC_INLINE int avs_is_error(AVS_Value v) { return v.type == 'e'; }
-
-AVSC_API(AVS_Clip *, avs_take_clip)(AVS_Value, AVS_ScriptEnvironment *);
-AVSC_API(void, avs_set_to_clip)(AVS_Value *, AVS_Clip *);
-
-AVSC_INLINE int avs_as_bool(AVS_Value v)
-        { return v.d.boolean; }
-AVSC_INLINE int avs_as_int(AVS_Value v)
-        { return v.d.integer; }
-AVSC_INLINE const char * avs_as_string(AVS_Value v)
-        { return avs_is_error(v) || avs_is_string(v) ? v.d.string : 0; }
-AVSC_INLINE double avs_as_float(AVS_Value v)
-        { return avs_is_int(v) ? v.d.integer : v.d.floating_pt; }
-AVSC_INLINE const char * avs_as_error(AVS_Value v)
-        { return avs_is_error(v) ? v.d.string : 0; }
-AVSC_INLINE const AVS_Value * avs_as_array(AVS_Value v)
-        { return v.d.array; }
-AVSC_INLINE int avs_array_size(AVS_Value v)
-        { return avs_is_array(v) ? v.array_size : 1; }
-AVSC_INLINE AVS_Value avs_array_elt(AVS_Value v, int index)
-        { return avs_is_array(v) ? v.d.array[index] : v; }
-
-// only use these functions on an AVS_Value that does not already have
-// an active value.  Remember, treat AVS_Value as a fat pointer.
-AVSC_INLINE AVS_Value avs_new_value_bool(int v0)
-        { AVS_Value v; v.type = 'b'; v.d.boolean = v0 == 0 ? 0 : 1; return v; }
-AVSC_INLINE AVS_Value avs_new_value_int(int v0)
-        { AVS_Value v; v.type = 'i'; v.d.integer = v0; return v; }
-AVSC_INLINE AVS_Value avs_new_value_string(const char * v0)
-        { AVS_Value v; v.type = 's'; v.d.string = v0; return v; }
-AVSC_INLINE AVS_Value avs_new_value_float(float v0)
-        { AVS_Value v; v.type = 'f'; v.d.floating_pt = v0; return v;}
-AVSC_INLINE AVS_Value avs_new_value_error(const char * v0)
-        { AVS_Value v; v.type = 'e'; v.d.string = v0; return v; }
-#ifndef AVSC_NO_DECLSPEC
-AVSC_INLINE AVS_Value avs_new_value_clip(AVS_Clip * v0)
-        { AVS_Value v; avs_set_to_clip(&v, v0); return v; }
-#endif
-AVSC_INLINE AVS_Value avs_new_value_array(AVS_Value * v0, int size)
-        { AVS_Value v; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; }
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_Clip
-//
-
-AVSC_API(void, avs_release_clip)(AVS_Clip *);
-AVSC_API(AVS_Clip *, avs_copy_clip)(AVS_Clip *);
-
-AVSC_API(const char *, avs_clip_get_error)(AVS_Clip *); // return 0 if no error
-
-AVSC_API(const AVS_VideoInfo *, avs_get_video_info)(AVS_Clip *);
-
-AVSC_API(int, avs_get_version)(AVS_Clip *);
-
-AVSC_API(AVS_VideoFrame *, avs_get_frame)(AVS_Clip *, int n);
-// The returned video frame must be released with avs_release_video_frame
-
-AVSC_API(int, avs_get_parity)(AVS_Clip *, int n);
-// return field parity if field_based, else parity of first field in frame
-
-AVSC_API(int, avs_get_audio)(AVS_Clip *, void * buf,
-                             INT64 start, INT64 count);
-// start and count are in samples
-
-AVSC_API(int, avs_set_cache_hints)(AVS_Clip *,
-                                   int cachehints, int frame_range);
-
-// This is the callback type used by avs_add_function
-typedef AVS_Value (AVSC_CC * AVS_ApplyFunc)
-                        (AVS_ScriptEnvironment *, AVS_Value args, void * user_data);
-
-typedef struct AVS_FilterInfo AVS_FilterInfo;
-struct AVS_FilterInfo
-{
-  // these members should not be modified outside of the AVS_ApplyFunc callback
-  AVS_Clip * child;
-  AVS_VideoInfo vi;
-  AVS_ScriptEnvironment * env;
-  AVS_VideoFrame * (AVSC_CC * get_frame)(AVS_FilterInfo *, int n);
-  int (AVSC_CC * get_parity)(AVS_FilterInfo *, int n);
-  int (AVSC_CC * get_audio)(AVS_FilterInfo *, void * buf,
-                                  INT64 start, INT64 count);
-  int (AVSC_CC * set_cache_hints)(AVS_FilterInfo *, int cachehints,
-                                        int frame_range);
-  void (AVSC_CC * free_filter)(AVS_FilterInfo *);
-
-  // Should be set when ever there is an error to report.
-  // It is cleared before any of the above methods are called
-  const char * error;
-  // this is to store whatever and may be modified at will
-  void * user_data;
-};
-
-// Create a new filter
-// fi is set to point to the AVS_FilterInfo so that you can
-//   modify it once it is initilized.
-// store_child should generally be set to true.  If it is not
-//    set than ALL methods (the function pointers) must be defined
-// If it is set than you do not need to worry about freeing the child
-//    clip.
-AVSC_API(AVS_Clip *, avs_new_c_filter)(AVS_ScriptEnvironment * e,
-                                       AVS_FilterInfo * * fi,
-                                       AVS_Value child, int store_child);
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_ScriptEnvironment
-//
-
-// For GetCPUFlags.  These are backwards-compatible with those in VirtualDub.
-enum {
-                                /* slowest CPU to support extension */
-  AVS_CPU_FORCE        = 0x01,   // N/A
-  AVS_CPU_FPU          = 0x02,   // 386/486DX
-  AVS_CPU_MMX          = 0x04,   // P55C, K6, PII
-  AVS_CPU_INTEGER_SSE  = 0x08,   // PIII, Athlon
-  AVS_CPU_SSE          = 0x10,   // PIII, Athlon XP/MP
-  AVS_CPU_SSE2         = 0x20,   // PIV, Hammer
-  AVS_CPU_3DNOW        = 0x40,   // K6-2
-  AVS_CPU_3DNOW_EXT    = 0x80,   // Athlon
-  AVS_CPU_X86_64       = 0xA0,   // Hammer (note: equiv. to 3DNow + SSE2,
-                                 // which only Hammer will have anyway)
-  AVS_CPUF_SSE3       = 0x100,   //  PIV+, K8 Venice
-  AVS_CPUF_SSSE3      = 0x200,   //  Core 2
-  AVS_CPUF_SSE4       = 0x400,   //  Penryn, Wolfdale, Yorkfield
-  AVS_CPUF_SSE4_1     = 0x400,
-  AVS_CPUF_SSE4_2     = 0x800,   //  Nehalem
-};
-
-AVSC_API(const char *, avs_get_error)(AVS_ScriptEnvironment *); // return 0 if no error
-
-AVSC_API(long, avs_get_cpu_flags)(AVS_ScriptEnvironment *);
-AVSC_API(int, avs_check_version)(AVS_ScriptEnvironment *, int version);
-
-AVSC_API(char *, avs_save_string)(AVS_ScriptEnvironment *, const char* s, int length);
-AVSC_API(char *, avs_sprintf)(AVS_ScriptEnvironment *, const char * fmt, ...);
-
-AVSC_API(char *, avs_vsprintf)(AVS_ScriptEnvironment *, const char * fmt, void* val);
- // note: val is really a va_list; I hope everyone typedefs va_list to a pointer
-
-AVSC_API(int, avs_add_function)(AVS_ScriptEnvironment *,
-                                const char * name, const char * params,
-                                AVS_ApplyFunc apply, void * user_data);
-
-AVSC_API(int, avs_function_exists)(AVS_ScriptEnvironment *, const char * name);
-
-AVSC_API(AVS_Value, avs_invoke)(AVS_ScriptEnvironment *, const char * name,
-                               AVS_Value args, const char** arg_names);
-// The returned value must be be released with avs_release_value
-
-AVSC_API(AVS_Value, avs_get_var)(AVS_ScriptEnvironment *, const char* name);
-// The returned value must be be released with avs_release_value
-
-AVSC_API(int, avs_set_var)(AVS_ScriptEnvironment *, const char* name, AVS_Value val);
-
-AVSC_API(int, avs_set_global_var)(AVS_ScriptEnvironment *, const char* name, const AVS_Value val);
-
-//void avs_push_context(AVS_ScriptEnvironment *, int level=0);
-//void avs_pop_context(AVS_ScriptEnvironment *);
-
-AVSC_API(AVS_VideoFrame *, avs_new_video_frame_a)(AVS_ScriptEnvironment *,
-                                          const AVS_VideoInfo * vi, int align);
-// align should be at least 16
-
-#ifndef AVSC_NO_DECLSPEC
-AVSC_INLINE
-AVS_VideoFrame * avs_new_video_frame(AVS_ScriptEnvironment * env,
-                                     const AVS_VideoInfo * vi)
-  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
-
-AVSC_INLINE
-AVS_VideoFrame * avs_new_frame(AVS_ScriptEnvironment * env,
-                               const AVS_VideoInfo * vi)
-  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
-#endif
-
-
-AVSC_API(int, avs_make_writable)(AVS_ScriptEnvironment *, AVS_VideoFrame * * pvf);
-
-AVSC_API(void, avs_bit_blt)(AVS_ScriptEnvironment *, BYTE* dstp, int dst_pitch, const BYTE* srcp, int src_pitch, int row_size, int height);
-
-typedef void (AVSC_CC *AVS_ShutdownFunc)(void* user_data, AVS_ScriptEnvironment * env);
-AVSC_API(void, avs_at_exit)(AVS_ScriptEnvironment *, AVS_ShutdownFunc function, void * user_data);
-
-AVSC_API(AVS_VideoFrame *, avs_subframe)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height);
-// The returned video frame must be be released
-
-AVSC_API(int, avs_set_memory_max)(AVS_ScriptEnvironment *, int mem);
-
-AVSC_API(int, avs_set_working_dir)(AVS_ScriptEnvironment *, const char * newdir);
-
-// avisynth.dll exports this; it's a way to use it as a library, without
-// writing an AVS script or without going through AVIFile.
-AVSC_API(AVS_ScriptEnvironment *, avs_create_script_environment)(int version);
-
-// this symbol is the entry point for the plugin and must
-// be defined
-AVSC_EXPORT
-const char * AVSC_CC avisynth_c_plugin_init(AVS_ScriptEnvironment* env);
-
-
-AVSC_API(void, avs_delete_script_environment)(AVS_ScriptEnvironment *);
-
-
-AVSC_API(AVS_VideoFrame *, avs_subframe_planar)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height, int rel_offsetU, int rel_offsetV, int new_pitchUV);
-// The returned video frame must be be released
-
-#ifdef AVSC_NO_DECLSPEC
-// use LoadLibrary and related functions to dynamically load Avisynth instead of declspec(dllimport)
-/*
-  The following functions needs to have been declared, probably from windows.h
-
-  void* malloc(size_t)
-  void free(void*);
-
-  HMODULE LoadLibrary(const char*);
-  void* GetProcAddress(HMODULE, const char*);
-  FreeLibrary(HMODULE);
-*/
-
-
-typedef struct AVS_Library AVS_Library;
-
-#define AVSC_DECLARE_FUNC(name) name##_func name
-
-struct AVS_Library {
-  HMODULE handle;
-
-  AVSC_DECLARE_FUNC(avs_add_function);
-  AVSC_DECLARE_FUNC(avs_at_exit);
-  AVSC_DECLARE_FUNC(avs_bit_blt);
-  AVSC_DECLARE_FUNC(avs_check_version);
-  AVSC_DECLARE_FUNC(avs_clip_get_error);
-  AVSC_DECLARE_FUNC(avs_copy_clip);
-  AVSC_DECLARE_FUNC(avs_copy_value);
-  AVSC_DECLARE_FUNC(avs_copy_video_frame);
-  AVSC_DECLARE_FUNC(avs_create_script_environment);
-  AVSC_DECLARE_FUNC(avs_delete_script_environment);
-  AVSC_DECLARE_FUNC(avs_function_exists);
-  AVSC_DECLARE_FUNC(avs_get_audio);
-  AVSC_DECLARE_FUNC(avs_get_cpu_flags);
-  AVSC_DECLARE_FUNC(avs_get_error);
-  AVSC_DECLARE_FUNC(avs_get_frame);
-  AVSC_DECLARE_FUNC(avs_get_parity);
-  AVSC_DECLARE_FUNC(avs_get_var);
-  AVSC_DECLARE_FUNC(avs_get_version);
-  AVSC_DECLARE_FUNC(avs_get_video_info);
-  AVSC_DECLARE_FUNC(avs_invoke);
-  AVSC_DECLARE_FUNC(avs_make_writable);
-  AVSC_DECLARE_FUNC(avs_new_c_filter);
-  AVSC_DECLARE_FUNC(avs_new_video_frame_a);
-  AVSC_DECLARE_FUNC(avs_release_clip);
-  AVSC_DECLARE_FUNC(avs_release_value);
-  AVSC_DECLARE_FUNC(avs_release_video_frame);
-  AVSC_DECLARE_FUNC(avs_save_string);
-  AVSC_DECLARE_FUNC(avs_set_cache_hints);
-  AVSC_DECLARE_FUNC(avs_set_global_var);
-  AVSC_DECLARE_FUNC(avs_set_memory_max);
-  AVSC_DECLARE_FUNC(avs_set_to_clip);
-  AVSC_DECLARE_FUNC(avs_set_var);
-  AVSC_DECLARE_FUNC(avs_set_working_dir);
-  AVSC_DECLARE_FUNC(avs_sprintf);
-  AVSC_DECLARE_FUNC(avs_subframe);
-  AVSC_DECLARE_FUNC(avs_subframe_planar);
-  AVSC_DECLARE_FUNC(avs_take_clip);
-  AVSC_DECLARE_FUNC(avs_vsprintf);
-};
-
-#undef AVSC_DECLARE_FUNC
-
-
-AVSC_INLINE AVS_Library * avs_load_library() {
-  AVS_Library *library = (AVS_Library *)malloc(sizeof(AVS_Library));
-  if (library == NULL)
-    return NULL;
-  library->handle = LoadLibrary("avisynth");
-  if (library->handle == NULL)
-    goto fail;
-
-#define __AVSC_STRINGIFY(x) #x
-#define AVSC_STRINGIFY(x) __AVSC_STRINGIFY(x)
-#define AVSC_LOAD_FUNC(name) {\
-  library->name = (name##_func) GetProcAddress(library->handle, AVSC_STRINGIFY(name));\
-  if (library->name == NULL)\
-    goto fail;\
-}
-
-  AVSC_LOAD_FUNC(avs_add_function);
-  AVSC_LOAD_FUNC(avs_at_exit);
-  AVSC_LOAD_FUNC(avs_bit_blt);
-  AVSC_LOAD_FUNC(avs_check_version);
-  AVSC_LOAD_FUNC(avs_clip_get_error);
-  AVSC_LOAD_FUNC(avs_copy_clip);
-  AVSC_LOAD_FUNC(avs_copy_value);
-  AVSC_LOAD_FUNC(avs_copy_video_frame);
-  AVSC_LOAD_FUNC(avs_create_script_environment);
-  AVSC_LOAD_FUNC(avs_delete_script_environment);
-  AVSC_LOAD_FUNC(avs_function_exists);
-  AVSC_LOAD_FUNC(avs_get_audio);
-  AVSC_LOAD_FUNC(avs_get_cpu_flags);
-  AVSC_LOAD_FUNC(avs_get_error);
-  AVSC_LOAD_FUNC(avs_get_frame);
-  AVSC_LOAD_FUNC(avs_get_parity);
-  AVSC_LOAD_FUNC(avs_get_var);
-  AVSC_LOAD_FUNC(avs_get_version);
-  AVSC_LOAD_FUNC(avs_get_video_info);
-  AVSC_LOAD_FUNC(avs_invoke);
-  AVSC_LOAD_FUNC(avs_make_writable);
-  AVSC_LOAD_FUNC(avs_new_c_filter);
-  AVSC_LOAD_FUNC(avs_new_video_frame_a);
-  AVSC_LOAD_FUNC(avs_release_clip);
-  AVSC_LOAD_FUNC(avs_release_value);
-  AVSC_LOAD_FUNC(avs_release_video_frame);
-  AVSC_LOAD_FUNC(avs_save_string);
-  AVSC_LOAD_FUNC(avs_set_cache_hints);
-  AVSC_LOAD_FUNC(avs_set_global_var);
-  AVSC_LOAD_FUNC(avs_set_memory_max);
-  AVSC_LOAD_FUNC(avs_set_to_clip);
-  AVSC_LOAD_FUNC(avs_set_var);
-  AVSC_LOAD_FUNC(avs_set_working_dir);
-  AVSC_LOAD_FUNC(avs_sprintf);
-  AVSC_LOAD_FUNC(avs_subframe);
-  AVSC_LOAD_FUNC(avs_subframe_planar);
-  AVSC_LOAD_FUNC(avs_take_clip);
-  AVSC_LOAD_FUNC(avs_vsprintf);
-
-#undef __AVSC_STRINGIFY
-#undef AVSC_STRINGIFY
-#undef AVSC_LOAD_FUNC
-
-  return library;
-
-fail:
-  free(library);
-  return NULL;
-}
-
-AVSC_INLINE void avs_free_library(AVS_Library *library) {
-  if (library == NULL)
-    return;
-  FreeLibrary(library->handle);
-  free(library);
-}
-#endif
-
-#endif
--- a/compat/avisynth/avisynth_c_25.h
+++ b/compat/avisynth/avisynth_c_25.h
@@ -1,68 +0,0 @@
-//  Copyright (c) 2011 FFmpegSource Project
-//
-//  Permission is hereby granted, free of charge, to any person obtaining a copy
-//  of this software and associated documentation files (the "Software"), to deal
-//  in the Software without restriction, including without limitation the rights
-//  to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
-//  copies of the Software, and to permit persons to whom the Software is
-//  furnished to do so, subject to the following conditions:
-//
-//  The above copyright notice and this permission notice shall be included in
-//  all copies or substantial portions of the Software.
-//
-//  THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
-//  IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
-//  FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
-//  AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
-//  LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
-//  OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
-//  THE SOFTWARE.
-
-/* these are defines/functions that are used and were changed in the switch to 2.6
- * and are needed to maintain full compatility with 2.5 */
-
-enum {
-  AVS_CS_YV12_25 = 1<<3 | AVS_CS_YUV | AVS_CS_PLANAR,  // y-v-u, planar
-  AVS_CS_I420_25 = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR,  // y-u-v, planar
-};
-
-AVSC_INLINE int avs_get_height_p_25(const AVS_VideoFrame * p, int plane) {
-    switch (plane)
-    {
-        case AVS_PLANAR_U: case AVS_PLANAR_V:
-            if (p->pitchUV)
-                return p->height>>1;
-            return 0;
-    }
-    return p->height;}
-
-AVSC_INLINE int avs_get_row_size_p_25(const AVS_VideoFrame * p, int plane) {
-    int r;
-    switch (plane)
-    {
-    case AVS_PLANAR_U: case AVS_PLANAR_V:
-        if (p->pitchUV)
-            return p->row_size>>1;
-        else
-            return 0;
-    case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED:
-        if (p->pitchUV)
-        {
-            r = ((p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)) )>>1; // Aligned rowsize
-            if (r < p->pitchUV)
-                return r;
-            return p->row_size>>1;
-        }
-        else
-            return 0;
-    case AVS_PLANAR_Y_ALIGNED:
-        r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
-        if (r <= p->pitch)
-            return r;
-        return p->row_size;
-    }
-    return p->row_size;
-}
-
-AVSC_INLINE int avs_is_yv12_25(const AVS_VideoInfo * p)
-    { return ((p->pixel_type & AVS_CS_YV12_25) == AVS_CS_YV12_25)||((p->pixel_type & AVS_CS_I420_25) == AVS_CS_I420_25); }
--- a/compat/avisynth/avxsynth_c.h
+++ b/compat/avisynth/avxsynth_c.h
@@ -1,727 +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 __AVXSYNTH_C__
-#define __AVXSYNTH_C__
-
-#include "windowsPorts/windows2linux.h"
-#include <stdarg.h>
-
-#ifdef __cplusplus
-#  define EXTERN_C extern "C"
-#else
-#  define EXTERN_C
-#endif
-
-#define AVSC_USE_STDCALL 1
-
-#ifndef AVSC_USE_STDCALL
-#  define AVSC_CC __cdecl
-#else
-#  define AVSC_CC __stdcall
-#endif
-
-#define AVSC_INLINE static __inline
-
-#ifdef AVISYNTH_C_EXPORTS
-#  define AVSC_EXPORT EXTERN_C
-#  define AVSC_API(ret, name) EXTERN_C __declspec(dllexport) ret AVSC_CC name
-#else
-#  define AVSC_EXPORT EXTERN_C __declspec(dllexport)
-#  ifndef AVSC_NO_DECLSPEC
-#    define AVSC_API(ret, name) EXTERN_C __declspec(dllimport) ret AVSC_CC name
-#  else
-#    define AVSC_API(ret, name) typedef ret (AVSC_CC *name##_func)
-#  endif
-#endif
-
-#ifdef __GNUC__
-typedef long long int INT64;
-#else
-typedef __int64 INT64;
-#endif
-
-
-/////////////////////////////////////////////////////////////////////
-//
-// Constants
-//
-
-#ifndef __AVXSYNTH_H__
-enum { AVISYNTH_INTERFACE_VERSION = 3 };
-#endif
-
-enum {AVS_SAMPLE_INT8  = 1<<0,
-      AVS_SAMPLE_INT16 = 1<<1,
-      AVS_SAMPLE_INT24 = 1<<2,
-      AVS_SAMPLE_INT32 = 1<<3,
-      AVS_SAMPLE_FLOAT = 1<<4};
-
-enum {AVS_PLANAR_Y=1<<0,
-      AVS_PLANAR_U=1<<1,
-      AVS_PLANAR_V=1<<2,
-      AVS_PLANAR_ALIGNED=1<<3,
-      AVS_PLANAR_Y_ALIGNED=AVS_PLANAR_Y|AVS_PLANAR_ALIGNED,
-      AVS_PLANAR_U_ALIGNED=AVS_PLANAR_U|AVS_PLANAR_ALIGNED,
-      AVS_PLANAR_V_ALIGNED=AVS_PLANAR_V|AVS_PLANAR_ALIGNED};
-
-  // Colorspace properties.
-enum {AVS_CS_BGR = 1<<28,
-      AVS_CS_YUV = 1<<29,
-      AVS_CS_INTERLEAVED = 1<<30,
-      AVS_CS_PLANAR = 1<<31};
-
-  // Specific colorformats
-enum {
-  AVS_CS_UNKNOWN = 0,
-  AVS_CS_BGR24 = 1<<0 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
-  AVS_CS_BGR32 = 1<<1 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
-  AVS_CS_YUY2 = 1<<2 | AVS_CS_YUV | AVS_CS_INTERLEAVED,
-  AVS_CS_YV12 = 1<<3 | AVS_CS_YUV | AVS_CS_PLANAR,  // y-v-u, planar
-  AVS_CS_I420 = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR,  // y-u-v, planar
-  AVS_CS_IYUV = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR  // same as above
-};
-
-enum {
-  AVS_IT_BFF = 1<<0,
-  AVS_IT_TFF = 1<<1,
-  AVS_IT_FIELDBASED = 1<<2};
-
-enum {
-  AVS_FILTER_TYPE=1,
-  AVS_FILTER_INPUT_COLORSPACE=2,
-  AVS_FILTER_OUTPUT_TYPE=9,
-  AVS_FILTER_NAME=4,
-  AVS_FILTER_AUTHOR=5,
-  AVS_FILTER_VERSION=6,
-  AVS_FILTER_ARGS=7,
-  AVS_FILTER_ARGS_INFO=8,
-  AVS_FILTER_ARGS_DESCRIPTION=10,
-  AVS_FILTER_DESCRIPTION=11};
-
-enum {  //SUBTYPES
-  AVS_FILTER_TYPE_AUDIO=1,
-  AVS_FILTER_TYPE_VIDEO=2,
-  AVS_FILTER_OUTPUT_TYPE_SAME=3,
-  AVS_FILTER_OUTPUT_TYPE_DIFFERENT=4};
-
-enum {
-  AVS_CACHE_NOTHING=0,
-  AVS_CACHE_RANGE=1,
-  AVS_CACHE_ALL=2,
-  AVS_CACHE_AUDIO=3,
-  AVS_CACHE_AUDIO_NONE=4,
-  AVS_CACHE_AUDIO_AUTO=5
-};
-
-#define AVS_FRAME_ALIGN 16
-
-typedef struct AVS_Clip AVS_Clip;
-typedef struct AVS_ScriptEnvironment AVS_ScriptEnvironment;
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_VideoInfo
-//
-
-// AVS_VideoInfo is layed out identicly to VideoInfo
-typedef struct AVS_VideoInfo {
-  int width, height;    // width=0 means no video
-  unsigned fps_numerator, fps_denominator;
-  int num_frames;
-
-  int pixel_type;
-
-  int audio_samples_per_second;   // 0 means no audio
-  int sample_type;
-  INT64 num_audio_samples;
-  int nchannels;
-
-  // Imagetype properties
-
-  int image_type;
-} AVS_VideoInfo;
-
-// useful functions of the above
-AVSC_INLINE int avs_has_video(const AVS_VideoInfo * p)
-        { return (p->width!=0); }
-
-AVSC_INLINE int avs_has_audio(const AVS_VideoInfo * p)
-        { return (p->audio_samples_per_second!=0); }
-
-AVSC_INLINE int avs_is_rgb(const AVS_VideoInfo * p)
-        { return !!(p->pixel_type&AVS_CS_BGR); }
-
-AVSC_INLINE int avs_is_rgb24(const AVS_VideoInfo * p)
-        { return (p->pixel_type&AVS_CS_BGR24)==AVS_CS_BGR24; } // Clear out additional properties
-
-AVSC_INLINE int avs_is_rgb32(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_BGR32) == AVS_CS_BGR32 ; }
-
-AVSC_INLINE int avs_is_yuv(const AVS_VideoInfo * p)
-        { return !!(p->pixel_type&AVS_CS_YUV ); }
-
-AVSC_INLINE int avs_is_yuy2(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_YUY2) == AVS_CS_YUY2; }
-
-AVSC_INLINE int avs_is_yv12(const AVS_VideoInfo * p)
-        { return ((p->pixel_type & AVS_CS_YV12) == AVS_CS_YV12)||((p->pixel_type & AVS_CS_I420) == AVS_CS_I420); }
-
-AVSC_INLINE int avs_is_color_space(const AVS_VideoInfo * p, int c_space)
-        { return ((p->pixel_type & c_space) == c_space); }
-
-AVSC_INLINE int avs_is_property(const AVS_VideoInfo * p, int property)
-        { return ((p->pixel_type & property)==property ); }
-
-AVSC_INLINE int avs_is_planar(const AVS_VideoInfo * p)
-        { return !!(p->pixel_type & AVS_CS_PLANAR); }
-
-AVSC_INLINE int avs_is_field_based(const AVS_VideoInfo * p)
-        { return !!(p->image_type & AVS_IT_FIELDBASED); }
-
-AVSC_INLINE int avs_is_parity_known(const AVS_VideoInfo * p)
-        { return ((p->image_type & AVS_IT_FIELDBASED)&&(p->image_type & (AVS_IT_BFF | AVS_IT_TFF))); }
-
-AVSC_INLINE int avs_is_bff(const AVS_VideoInfo * p)
-        { return !!(p->image_type & AVS_IT_BFF); }
-
-AVSC_INLINE int avs_is_tff(const AVS_VideoInfo * p)
-        { return !!(p->image_type & AVS_IT_TFF); }
-
-AVSC_INLINE int avs_bits_per_pixel(const AVS_VideoInfo * p)
-{
-  switch (p->pixel_type) {
-      case AVS_CS_BGR24: return 24;
-      case AVS_CS_BGR32: return 32;
-      case AVS_CS_YUY2:  return 16;
-      case AVS_CS_YV12:
-      case AVS_CS_I420:  return 12;
-      default:           return 0;
-    }
-}
-AVSC_INLINE int avs_bytes_from_pixels(const AVS_VideoInfo * p, int pixels)
-        { return pixels * (avs_bits_per_pixel(p)>>3); }   // Will work on planar images, but will return only luma planes
-
-AVSC_INLINE int avs_row_size(const AVS_VideoInfo * p)
-        { return avs_bytes_from_pixels(p,p->width); }  // Also only returns first plane on planar images
-
-AVSC_INLINE int avs_bmp_size(const AVS_VideoInfo * vi)
-        { if (avs_is_planar(vi)) {int p = vi->height * ((avs_row_size(vi)+3) & ~3); p+=p>>1; return p;  } return vi->height * ((avs_row_size(vi)+3) & ~3); }
-
-AVSC_INLINE int avs_samples_per_second(const AVS_VideoInfo * p)
-        { return p->audio_samples_per_second; }
-
-
-AVSC_INLINE int avs_bytes_per_channel_sample(const AVS_VideoInfo * p)
-{
-    switch (p->sample_type) {
-      case AVS_SAMPLE_INT8:  return sizeof(signed char);
-      case AVS_SAMPLE_INT16: return sizeof(signed short);
-      case AVS_SAMPLE_INT24: return 3;
-      case AVS_SAMPLE_INT32: return sizeof(signed int);
-      case AVS_SAMPLE_FLOAT: return sizeof(float);
-      default: return 0;
-    }
-}
-AVSC_INLINE int avs_bytes_per_audio_sample(const AVS_VideoInfo * p)
-        { return p->nchannels*avs_bytes_per_channel_sample(p);}
-
-AVSC_INLINE INT64 avs_audio_samples_from_frames(const AVS_VideoInfo * p, INT64 frames)
-        { return ((INT64)(frames) * p->audio_samples_per_second * p->fps_denominator / p->fps_numerator); }
-
-AVSC_INLINE int avs_frames_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
-        { return (int)(samples * (INT64)p->fps_numerator / (INT64)p->fps_denominator / (INT64)p->audio_samples_per_second); }
-
-AVSC_INLINE INT64 avs_audio_samples_from_bytes(const AVS_VideoInfo * p, INT64 bytes)
-        { return bytes / avs_bytes_per_audio_sample(p); }
-
-AVSC_INLINE INT64 avs_bytes_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
-        { return samples * avs_bytes_per_audio_sample(p); }
-
-AVSC_INLINE int avs_audio_channels(const AVS_VideoInfo * p)
-        { return p->nchannels; }
-
-AVSC_INLINE int avs_sample_type(const AVS_VideoInfo * p)
-        { return p->sample_type;}
-
-// useful mutator
-AVSC_INLINE void avs_set_property(AVS_VideoInfo * p, int property)
-        { p->image_type|=property; }
-
-AVSC_INLINE void avs_clear_property(AVS_VideoInfo * p, int property)
-        { p->image_type&=~property; }
-
-AVSC_INLINE void avs_set_field_based(AVS_VideoInfo * p, int isfieldbased)
-        { if (isfieldbased) p->image_type|=AVS_IT_FIELDBASED; else p->image_type&=~AVS_IT_FIELDBASED; }
-
-AVSC_INLINE void avs_set_fps(AVS_VideoInfo * p, unsigned numerator, unsigned denominator)
-{
-    unsigned x=numerator, y=denominator;
-    while (y) {   // find gcd
-      unsigned t = x%y; x = y; y = t;
-    }
-    p->fps_numerator = numerator/x;
-    p->fps_denominator = denominator/x;
-}
-
-AVSC_INLINE int avs_is_same_colorspace(AVS_VideoInfo * x, AVS_VideoInfo * y)
-{
-        return (x->pixel_type == y->pixel_type)
-                || (avs_is_yv12(x) && avs_is_yv12(y));
-}
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_VideoFrame
-//
-
-// VideoFrameBuffer holds information about a memory block which is used
-// for video data.  For efficiency, instances of this class are not deleted
-// when the refcount reaches zero; instead they're stored in a linked list
-// to be reused.  The instances are deleted when the corresponding AVS
-// file is closed.
-
-// AVS_VideoFrameBuffer is layed out identicly to VideoFrameBuffer
-// DO NOT USE THIS STRUCTURE DIRECTLY
-typedef struct AVS_VideoFrameBuffer {
-  unsigned char * data;
-  int data_size;
-  // sequence_number is incremented every time the buffer is changed, so
-  // that stale views can tell they're no longer valid.
-  long sequence_number;
-
-  long refcount;
-} AVS_VideoFrameBuffer;
-
-// VideoFrame holds a "window" into a VideoFrameBuffer.
-
-// AVS_VideoFrame is layed out identicly to IVideoFrame
-// DO NOT USE THIS STRUCTURE DIRECTLY
-typedef struct AVS_VideoFrame {
-  int refcount;
-  AVS_VideoFrameBuffer * vfb;
-  int offset, pitch, row_size, height, offsetU, offsetV, pitchUV;  // U&V offsets are from top of picture.
-} AVS_VideoFrame;
-
-// Access functions for AVS_VideoFrame
-AVSC_INLINE int avs_get_pitch(const AVS_VideoFrame * p) {
-        return p->pitch;}
-
-AVSC_INLINE int avs_get_pitch_p(const AVS_VideoFrame * p, int plane) {
-  switch (plane) {
-  case AVS_PLANAR_U: case AVS_PLANAR_V: return p->pitchUV;}
-  return p->pitch;}
-
-AVSC_INLINE int avs_get_row_size(const AVS_VideoFrame * p) {
-        return p->row_size; }
-
-AVSC_INLINE int avs_get_row_size_p(const AVS_VideoFrame * p, int plane) {
-        int r;
-    switch (plane) {
-    case AVS_PLANAR_U: case AVS_PLANAR_V:
-                if (p->pitchUV) return p->row_size>>1;
-                else            return 0;
-    case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED:
-                if (p->pitchUV) {
-                        r = ((p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)) )>>1; // Aligned rowsize
-                        if (r < p->pitchUV)
-                                return r;
-                        return p->row_size>>1;
-                } else return 0;
-    case AVS_PLANAR_Y_ALIGNED:
-                r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
-                if (r <= p->pitch)
-                        return r;
-                return p->row_size;
-    }
-    return p->row_size;
-}
-
-AVSC_INLINE int avs_get_height(const AVS_VideoFrame * p) {
-        return p->height;}
-
-AVSC_INLINE int avs_get_height_p(const AVS_VideoFrame * p, int plane) {
-        switch (plane) {
-                case AVS_PLANAR_U: case AVS_PLANAR_V:
-                        if (p->pitchUV) return p->height>>1;
-                        return 0;
-        }
-        return p->height;}
-
-AVSC_INLINE const unsigned char* avs_get_read_ptr(const AVS_VideoFrame * p) {
-        return p->vfb->data + p->offset;}
-
-AVSC_INLINE const unsigned char* avs_get_read_ptr_p(const AVS_VideoFrame * p, int plane)
-{
-        switch (plane) {
-                case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
-                case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
-                default:           return p->vfb->data + p->offset;}
-}
-
-AVSC_INLINE int avs_is_writable(const AVS_VideoFrame * p) {
-        return (p->refcount == 1 && p->vfb->refcount == 1);}
-
-AVSC_INLINE unsigned char* avs_get_write_ptr(const AVS_VideoFrame * p)
-{
-        if (avs_is_writable(p)) {
-                ++p->vfb->sequence_number;
-                return p->vfb->data + p->offset;
-        } else
-                return 0;
-}
-
-AVSC_INLINE unsigned char* avs_get_write_ptr_p(const AVS_VideoFrame * p, int plane)
-{
-        if (plane==AVS_PLANAR_Y && avs_is_writable(p)) {
-                ++p->vfb->sequence_number;
-                return p->vfb->data + p->offset;
-        } else if (plane==AVS_PLANAR_Y) {
-                return 0;
-        } else {
-                switch (plane) {
-                        case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
-                        case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
-                        default:       return p->vfb->data + p->offset;
-                }
-        }
-}
-
-#if defined __cplusplus
-extern "C"
-{
-#endif // __cplusplus
-AVSC_API(void, avs_release_video_frame)(AVS_VideoFrame *);
-// makes a shallow copy of a video frame
-AVSC_API(AVS_VideoFrame *, avs_copy_video_frame)(AVS_VideoFrame *);
-#if defined __cplusplus
-}
-#endif // __cplusplus
-
-#ifndef AVSC_NO_DECLSPEC
-AVSC_INLINE void avs_release_frame(AVS_VideoFrame * f)
-  {avs_release_video_frame(f);}
-AVSC_INLINE AVS_VideoFrame * avs_copy_frame(AVS_VideoFrame * f)
-  {return avs_copy_video_frame(f);}
-#endif
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_Value
-//
-
-// Treat AVS_Value as a fat pointer.  That is use avs_copy_value
-// and avs_release_value appropiaty as you would if AVS_Value was
-// a pointer.
-
-// To maintain source code compatibility with future versions of the
-// avisynth_c API don't use the AVS_Value directly.  Use the helper
-// functions below.
-
-// AVS_Value is layed out identicly to AVSValue
-typedef struct AVS_Value AVS_Value;
-struct AVS_Value {
-  short type;  // 'a'rray, 'c'lip, 'b'ool, 'i'nt, 'f'loat, 's'tring, 'v'oid, or 'l'ong
-               // for some function e'rror
-  short array_size;
-  union {
-    void * clip; // do not use directly, use avs_take_clip
-    char boolean;
-    int integer;
-    INT64 integer64; // match addition of __int64 to avxplugin.h
-    float floating_pt;
-    const char * string;
-    const AVS_Value * array;
-  } d;
-};
-
-// AVS_Value should be initilized with avs_void.
-// Should also set to avs_void after the value is released
-// with avs_copy_value.  Consider it the equalvent of setting
-// a pointer to NULL
-static const AVS_Value avs_void = {'v'};
-
-AVSC_API(void, avs_copy_value)(AVS_Value * dest, AVS_Value src);
-AVSC_API(void, avs_release_value)(AVS_Value);
-
-AVSC_INLINE int avs_defined(AVS_Value v) { return v.type != 'v'; }
-AVSC_INLINE int avs_is_clip(AVS_Value v) { return v.type == 'c'; }
-AVSC_INLINE int avs_is_bool(AVS_Value v) { return v.type == 'b'; }
-AVSC_INLINE int avs_is_int(AVS_Value v) { return v.type == 'i'; }
-AVSC_INLINE int avs_is_float(AVS_Value v) { return v.type == 'f' || v.type == 'i'; }
-AVSC_INLINE int avs_is_string(AVS_Value v) { return v.type == 's'; }
-AVSC_INLINE int avs_is_array(AVS_Value v) { return v.type == 'a'; }
-AVSC_INLINE int avs_is_error(AVS_Value v) { return v.type == 'e'; }
-
-#if defined __cplusplus
-extern "C"
-{
-#endif // __cplusplus
-AVSC_API(AVS_Clip *, avs_take_clip)(AVS_Value, AVS_ScriptEnvironment *);
-AVSC_API(void, avs_set_to_clip)(AVS_Value *, AVS_Clip *);
-#if defined __cplusplus
-}
-#endif // __cplusplus
-
-AVSC_INLINE int avs_as_bool(AVS_Value v)
-        { return v.d.boolean; }
-AVSC_INLINE int avs_as_int(AVS_Value v)
-        { return v.d.integer; }
-AVSC_INLINE const char * avs_as_string(AVS_Value v)
-        { return avs_is_error(v) || avs_is_string(v) ? v.d.string : 0; }
-AVSC_INLINE double avs_as_float(AVS_Value v)
-        { return avs_is_int(v) ? v.d.integer : v.d.floating_pt; }
-AVSC_INLINE const char * avs_as_error(AVS_Value v)
-        { return avs_is_error(v) ? v.d.string : 0; }
-AVSC_INLINE const AVS_Value * avs_as_array(AVS_Value v)
-        { return v.d.array; }
-AVSC_INLINE int avs_array_size(AVS_Value v)
-        { return avs_is_array(v) ? v.array_size : 1; }
-AVSC_INLINE AVS_Value avs_array_elt(AVS_Value v, int index)
-        { return avs_is_array(v) ? v.d.array[index] : v; }
-
-// only use these functions on am AVS_Value that does not already have
-// an active value.  Remember, treat AVS_Value as a fat pointer.
-AVSC_INLINE AVS_Value avs_new_value_bool(int v0)
-        { AVS_Value v; v.type = 'b'; v.d.boolean = v0 == 0 ? 0 : 1; return v; }
-AVSC_INLINE AVS_Value avs_new_value_int(int v0)
-        { AVS_Value v; v.type = 'i'; v.d.integer = v0; return v; }
-AVSC_INLINE AVS_Value avs_new_value_string(const char * v0)
-        { AVS_Value v; v.type = 's'; v.d.string = v0; return v; }
-AVSC_INLINE AVS_Value avs_new_value_float(float v0)
-        { AVS_Value v; v.type = 'f'; v.d.floating_pt = v0; return v;}
-AVSC_INLINE AVS_Value avs_new_value_error(const char * v0)
-        { AVS_Value v; v.type = 'e'; v.d.string = v0; return v; }
-#ifndef AVSC_NO_DECLSPEC
-AVSC_INLINE AVS_Value avs_new_value_clip(AVS_Clip * v0)
-        { AVS_Value v; avs_set_to_clip(&v, v0); return v; }
-#endif
-AVSC_INLINE AVS_Value avs_new_value_array(AVS_Value * v0, int size)
-        { AVS_Value v; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; }
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_Clip
-//
-#if defined __cplusplus
-extern "C"
-{
-#endif // __cplusplus
-AVSC_API(void, avs_release_clip)(AVS_Clip *);
-AVSC_API(AVS_Clip *, avs_copy_clip)(AVS_Clip *);
-
-AVSC_API(const char *, avs_clip_get_error)(AVS_Clip *); // return 0 if no error
-
-AVSC_API(const AVS_VideoInfo *, avs_get_video_info)(AVS_Clip *);
-
-AVSC_API(int, avs_get_version)(AVS_Clip *);
-
-AVSC_API(AVS_VideoFrame *, avs_get_frame)(AVS_Clip *, int n);
-// The returned video frame must be released with avs_release_video_frame
-
-AVSC_API(int, avs_get_parity)(AVS_Clip *, int n);
-// return field parity if field_based, else parity of first field in frame
-
-AVSC_API(int, avs_get_audio)(AVS_Clip *, void * buf,
-                                  INT64 start, INT64 count);
-// start and count are in samples
-
-AVSC_API(int, avs_set_cache_hints)(AVS_Clip *,
-                                        int cachehints, size_t frame_range);
-#if defined __cplusplus
-}
-#endif // __cplusplus
-
-// This is the callback type used by avs_add_function
-typedef AVS_Value (AVSC_CC * AVS_ApplyFunc)
-                        (AVS_ScriptEnvironment *, AVS_Value args, void * user_data);
-
-typedef struct AVS_FilterInfo AVS_FilterInfo;
-struct AVS_FilterInfo
-{
-  // these members should not be modified outside of the AVS_ApplyFunc callback
-  AVS_Clip * child;
-  AVS_VideoInfo vi;
-  AVS_ScriptEnvironment * env;
-  AVS_VideoFrame * (AVSC_CC * get_frame)(AVS_FilterInfo *, int n);
-  int (AVSC_CC * get_parity)(AVS_FilterInfo *, int n);
-  int (AVSC_CC * get_audio)(AVS_FilterInfo *, void * buf,
-                                  INT64 start, INT64 count);
-  int (AVSC_CC * set_cache_hints)(AVS_FilterInfo *, int cachehints,
-                                        int frame_range);
-  void (AVSC_CC * free_filter)(AVS_FilterInfo *);
-
-  // Should be set when ever there is an error to report.
-  // It is cleared before any of the above methods are called
-  const char * error;
-  // this is to store whatever and may be modified at will
-  void * user_data;
-};
-
-// Create a new filter
-// fi is set to point to the AVS_FilterInfo so that you can
-//   modify it once it is initilized.
-// store_child should generally be set to true.  If it is not
-//    set than ALL methods (the function pointers) must be defined
-// If it is set than you do not need to worry about freeing the child
-//    clip.
-#if defined __cplusplus
-extern "C"
-{
-#endif // __cplusplus
-AVSC_API(AVS_Clip *, avs_new_c_filter)(AVS_ScriptEnvironment * e,
-                                      AVS_FilterInfo * * fi,
-                                      AVS_Value child, int store_child);
-#if defined __cplusplus
-}
-#endif // __cplusplus
-
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_ScriptEnvironment
-//
-
-// For GetCPUFlags.  These are backwards-compatible with those in VirtualDub.
-enum {
-                                /* slowest CPU to support extension */
-  AVS_CPU_FORCE        = 0x01,   // N/A
-  AVS_CPU_FPU          = 0x02,   // 386/486DX
-  AVS_CPU_MMX          = 0x04,   // P55C, K6, PII
-  AVS_CPU_INTEGER_SSE  = 0x08,   // PIII, Athlon
-  AVS_CPU_SSE          = 0x10,   // PIII, Athlon XP/MP
-  AVS_CPU_SSE2         = 0x20,   // PIV, Hammer
-  AVS_CPU_3DNOW        = 0x40,   // K6-2
-  AVS_CPU_3DNOW_EXT    = 0x80,   // Athlon
-  AVS_CPU_X86_64       = 0xA0,   // Hammer (note: equiv. to 3DNow + SSE2,
-                                 // which only Hammer will have anyway)
-};
-
-#if defined __cplusplus
-extern "C"
-{
-#endif // __cplusplus
-AVSC_API(const char *, avs_get_error)(AVS_ScriptEnvironment *); // return 0 if no error
-
-AVSC_API(long, avs_get_cpu_flags)(AVS_ScriptEnvironment *);
-AVSC_API(int, avs_check_version)(AVS_ScriptEnvironment *, int version);
-
-AVSC_API(char *, avs_save_string)(AVS_ScriptEnvironment *, const char* s, int length);
-AVSC_API(char *, avs_sprintf)(AVS_ScriptEnvironment *, const char * fmt, ...);
-
-AVSC_API(char *, avs_vsprintf)(AVS_ScriptEnvironment *, const char * fmt, va_list val);
- // note: val is really a va_list; I hope everyone typedefs va_list to a pointer
-
-AVSC_API(int, avs_add_function)(AVS_ScriptEnvironment *,
-                                     const char * name, const char * params,
-                                     AVS_ApplyFunc apply, void * user_data);
-
-AVSC_API(int, avs_function_exists)(AVS_ScriptEnvironment *, const char * name);
-
-AVSC_API(AVS_Value, avs_invoke)(AVS_ScriptEnvironment *, const char * name,
-                               AVS_Value args, const char** arg_names);
-// The returned value must be be released with avs_release_value
-
-AVSC_API(AVS_Value, avs_get_var)(AVS_ScriptEnvironment *, const char* name);
-// The returned value must be be released with avs_release_value
-
-AVSC_API(int, avs_set_var)(AVS_ScriptEnvironment *, const char* name, AVS_Value val);
-
-AVSC_API(int, avs_set_global_var)(AVS_ScriptEnvironment *, const char* name, const AVS_Value val);
-
-//void avs_push_context(AVS_ScriptEnvironment *, int level=0);
-//void avs_pop_context(AVS_ScriptEnvironment *);
-
-AVSC_API(AVS_VideoFrame *, avs_new_video_frame_a)(AVS_ScriptEnvironment *,
-                                          const AVS_VideoInfo * vi, int align);
-// align should be at least 16
-#if defined __cplusplus
-}
-#endif // __cplusplus
-
-#ifndef AVSC_NO_DECLSPEC
-AVSC_INLINE
-AVS_VideoFrame * avs_new_video_frame(AVS_ScriptEnvironment * env,
-                                     const AVS_VideoInfo * vi)
-  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
-
-AVSC_INLINE
-AVS_VideoFrame * avs_new_frame(AVS_ScriptEnvironment * env,
-                               const AVS_VideoInfo * vi)
-  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
-#endif
-
-#if defined __cplusplus
-extern "C"
-{
-#endif // __cplusplus
-AVSC_API(int, avs_make_writable)(AVS_ScriptEnvironment *, AVS_VideoFrame * * pvf);
-
-AVSC_API(void, avs_bit_blt)(AVS_ScriptEnvironment *, unsigned char* dstp, int dst_pitch, const unsigned char* srcp, int src_pitch, int row_size, int height);
-
-typedef void (AVSC_CC *AVS_ShutdownFunc)(void* user_data, AVS_ScriptEnvironment * env);
-AVSC_API(void, avs_at_exit)(AVS_ScriptEnvironment *, AVS_ShutdownFunc function, void * user_data);
-
-AVSC_API(AVS_VideoFrame *, avs_subframe)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height);
-// The returned video frame must be be released
-
-AVSC_API(int, avs_set_memory_max)(AVS_ScriptEnvironment *, int mem);
-
-AVSC_API(int, avs_set_working_dir)(AVS_ScriptEnvironment *, const char * newdir);
-
-// avisynth.dll exports this; it's a way to use it as a library, without
-// writing an AVS script or without going through AVIFile.
-AVSC_API(AVS_ScriptEnvironment *, avs_create_script_environment)(int version);
-#if defined __cplusplus
-}
-#endif // __cplusplus
-
-// this symbol is the entry point for the plugin and must
-// be defined
-AVSC_EXPORT
-const char * AVSC_CC avisynth_c_plugin_init(AVS_ScriptEnvironment* env);
-
-
-#if defined __cplusplus
-extern "C"
-{
-#endif // __cplusplus
-AVSC_API(void, avs_delete_script_environment)(AVS_ScriptEnvironment *);
-
-
-AVSC_API(AVS_VideoFrame *, avs_subframe_planar)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height, int rel_offsetU, int rel_offsetV, int new_pitchUV);
-// The returned video frame must be be released
-#if defined __cplusplus
-}
-#endif // __cplusplus
-
-#endif //__AVXSYNTH_C__
--- a/compat/avisynth/windowsPorts/basicDataTypeConversions.h
+++ b/compat/avisynth/windowsPorts/basicDataTypeConversions.h
@@ -1,85 +0,0 @@
-#ifndef __DATA_TYPE_CONVERSIONS_H__
-#define __DATA_TYPE_CONVERSIONS_H__
-
-#include <stdint.h>
-#include <wchar.h>
-
-#ifdef __cplusplus
-namespace avxsynth {
-#endif // __cplusplus
-
-typedef int64_t __int64;
-typedef int32_t __int32;
-#ifdef __cplusplus
-typedef bool BOOL;
-#else
-typedef uint32_t BOOL;
-#endif // __cplusplus
-typedef void* HMODULE;
-typedef void* LPVOID;
-typedef void* PVOID;
-typedef PVOID HANDLE;
-typedef HANDLE HWND;
-typedef HANDLE HINSTANCE;
-typedef void* HDC;
-typedef void* HBITMAP;
-typedef void* HICON;
-typedef void* HFONT;
-typedef void* HGDIOBJ;
-typedef void* HBRUSH;
-typedef void* HMMIO;
-typedef void* HACMSTREAM;
-typedef void* HACMDRIVER;
-typedef void* HIC;
-typedef void* HACMOBJ;
-typedef HACMSTREAM* LPHACMSTREAM;
-typedef void* HACMDRIVERID;
-typedef void* LPHACMDRIVER;
-typedef unsigned char BYTE;
-typedef BYTE* LPBYTE;
-typedef char TCHAR;
-typedef TCHAR* LPTSTR;
-typedef const TCHAR* LPCTSTR;
-typedef char* LPSTR;
-typedef LPSTR LPOLESTR;
-typedef const char* LPCSTR;
-typedef LPCSTR LPCOLESTR;
-typedef wchar_t WCHAR;
-typedef unsigned short WORD;
-typedef unsigned int UINT;
-typedef UINT MMRESULT;
-typedef uint32_t DWORD;
-typedef DWORD COLORREF;
-typedef DWORD FOURCC;
-typedef DWORD HRESULT;
-typedef DWORD* LPDWORD;
-typedef DWORD* DWORD_PTR;
-typedef int32_t LONG;
-typedef int32_t* LONG_PTR;
-typedef LONG_PTR LRESULT;
-typedef uint32_t ULONG;
-typedef uint32_t* ULONG_PTR;
-//typedef __int64_t intptr_t;
-typedef uint64_t _fsize_t;
-
-
-//
-// Structures
-//
-
-typedef struct _GUID {
-  DWORD Data1;
-  WORD  Data2;
-  WORD  Data3;
-  BYTE  Data4[8];
-} GUID;
-
-typedef GUID REFIID;
-typedef GUID CLSID;
-typedef CLSID* LPCLSID;
-typedef GUID IID;
-
-#ifdef __cplusplus
-}; // namespace avxsynth
-#endif // __cplusplus
-#endif //  __DATA_TYPE_CONVERSIONS_H__
--- a/compat/avisynth/windowsPorts/windows2linux.h
+++ b/compat/avisynth/windowsPorts/windows2linux.h
@@ -1,77 +0,0 @@
-#ifndef __WINDOWS2LINUX_H__
-#define __WINDOWS2LINUX_H__
-
-/*
- * LINUX SPECIFIC DEFINITIONS
-*/
-//
-// Data types conversions
-//
-#include <stdlib.h>
-#include <string.h>
-#include "basicDataTypeConversions.h"
-
-#ifdef __cplusplus
-namespace avxsynth {
-#endif // __cplusplus
-//
-// purposefully define the following MSFT definitions
-// to mean nothing (as they do not mean anything on Linux)
-//
-#define __stdcall
-#define __cdecl
-#define noreturn
-#define __declspec(x)
-#define STDAPI       extern "C" HRESULT
-#define STDMETHODIMP HRESULT __stdcall
-#define STDMETHODIMP_(x) x __stdcall
-
-#define STDMETHOD(x)    virtual HRESULT x
-#define STDMETHOD_(a, x) virtual a x
-
-#ifndef TRUE
-#define TRUE  true
-#endif
-
-#ifndef FALSE
-#define FALSE false
-#endif
-
-#define S_OK                (0x00000000)
-#define S_FALSE             (0x00000001)
-#define E_NOINTERFACE       (0X80004002)
-#define E_POINTER           (0x80004003)
-#define E_FAIL              (0x80004005)
-#define E_OUTOFMEMORY       (0x8007000E)
-
-#define INVALID_HANDLE_VALUE    ((HANDLE)((LONG_PTR)-1))
-#define FAILED(hr)              ((hr) & 0x80000000)
-#define SUCCEEDED(hr)           (!FAILED(hr))
-
-
-//
-// Functions
-//
-#define MAKEDWORD(a,b,c,d) ((a << 24) | (b << 16) | (c << 8) | (d))
-#define MAKEWORD(a,b) ((a << 8) | (b))
-
-#define lstrlen                             strlen
-#define lstrcpy                             strcpy
-#define lstrcmpi                            strcasecmp
-#define _stricmp                            strcasecmp
-#define InterlockedIncrement(x)             __sync_fetch_and_add((x), 1)
-#define InterlockedDecrement(x)             __sync_fetch_and_sub((x), 1)
-// Windows uses (new, old) ordering but GCC has (old, new)
-#define InterlockedCompareExchange(x,y,z)   __sync_val_compare_and_swap(x,z,y)
-
-#define UInt32x32To64(a, b)                 ( (uint64_t) ( ((uint64_t)((uint32_t)(a))) * ((uint32_t)(b))  ) )
-#define Int64ShrlMod32(a, b)                ( (uint64_t) ( (uint64_t)(a) >> (b) ) )
-#define Int32x32To64(a, b)                  ((__int64)(((__int64)((long)(a))) * ((long)(b))))
-
-#define MulDiv(nNumber, nNumerator, nDenominator)   (int32_t) (((int64_t) (nNumber) * (int64_t) (nNumerator) + (int64_t) ((nDenominator)/2)) / (int64_t) (nDenominator))
-
-#ifdef __cplusplus
-}; // namespace avxsynth
-#endif // __cplusplus
-
-#endif //  __WINDOWS2LINUX_H__
--- a/compat/getopt.c
+++ b/compat/getopt.c
@@ -1,84 +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
- */
-
-/*
- * This file was copied from the following newsgroup posting:
- *
- * Newsgroups: mod.std.unix
- * Subject: public domain AT&T getopt source
- * Date: 3 Nov 85 19:34:15 GMT
- *
- * Here's something you've all been waiting for:  the AT&T public domain
- * source for getopt(3).  It is the code which was given out at the 1985
- * UNIFORUM conference in Dallas.  I obtained it by electronic mail
- * directly from AT&T.  The people there assure me that it is indeed
- * in the public domain.
- */
-
-#include <stdio.h>
-#include <string.h>
-
-static int opterr = 1;
-static int optind = 1;
-static int optopt;
-static char *optarg;
-
-static int getopt(int argc, char *argv[], char *opts)
-{
-    static int sp = 1;
-    int c;
-    char *cp;
-
-    if (sp == 1) {
-        if (optind >= argc ||
-            argv[optind][0] != '-' || argv[optind][1] == '\0')
-            return EOF;
-        else if (!strcmp(argv[optind], "--")) {
-            optind++;
-            return EOF;
-        }
-    }
-    optopt = c = argv[optind][sp];
-    if (c == ':' || (cp = strchr(opts, c)) == NULL) {
-        fprintf(stderr, ": illegal option -- %c\n", c);
-        if (argv[optind][++sp] == '\0') {
-            optind++;
-            sp = 1;
-        }
-        return '?';
-    }
-    if (*++cp == ':') {
-        if (argv[optind][sp+1] != '\0')
-            optarg = &argv[optind++][sp+1];
-        else if(++optind >= argc) {
-            fprintf(stderr, ": option requires an argument -- %c\n", c);
-            sp = 1;
-            return '?';
-        } else
-            optarg = argv[optind++];
-        sp = 1;
-    } else {
-        if (argv[optind][++sp] == '\0') {
-            sp = 1;
-            optind++;
-        }
-        optarg = NULL;
-    }
-
-    return c;
-}
--- a/compat/msvcrt/snprintf.c
+++ b/compat/msvcrt/snprintf.c
@@ -1,71 +0,0 @@
-/*
- * C99-compatible snprintf() and vsnprintf() implementations
- * Copyright (c) 2012 Ronald S. Bultje <rsbultje@gmail.com>
- *
- * 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 <stdio.h>
-#include <stdarg.h>
-#include <limits.h>
-#include <string.h>
-
-#include "compat/va_copy.h"
-#include "libavutil/error.h"
-
-#if defined(__MINGW32__)
-#define EOVERFLOW EFBIG
-#endif
-
-int avpriv_snprintf(char *s, size_t n, const char *fmt, ...)
-{
-    va_list ap;
-    int ret;
-
-    va_start(ap, fmt);
-    ret = avpriv_vsnprintf(s, n, fmt, ap);
-    va_end(ap);
-
-    return ret;
-}
-
-int avpriv_vsnprintf(char *s, size_t n, const char *fmt,
-                     va_list ap)
-{
-    int ret;
-    va_list ap_copy;
-
-    if (n == 0)
-        return _vscprintf(fmt, ap);
-    else if (n > INT_MAX)
-        return AVERROR(EOVERFLOW);
-
-    /* we use n - 1 here because if the buffer is not big enough, the MS
-     * runtime libraries don't add a terminating zero at the end. MSDN
-     * recommends to provide _snprintf/_vsnprintf() a buffer size that
-     * is one less than the actual buffer, and zero it before calling
-     * _snprintf/_vsnprintf() to workaround this problem.
-     * See http://msdn.microsoft.com/en-us/library/1kt27hek(v=vs.80).aspx */
-    memset(s, 0, n);
-    va_copy(ap_copy, ap);
-    ret = _vsnprintf(s, n - 1, fmt, ap_copy);
-    va_end(ap_copy);
-    if (ret == -1)
-        ret = _vscprintf(fmt, ap);
-
-    return ret;
-}
--- a/compat/msvcrt/snprintf.h
+++ b/compat/msvcrt/snprintf.h
@@ -1,38 +0,0 @@
-/*
- * C99-compatible snprintf() and vsnprintf() implementations
- * Copyright (c) 2012 Ronald S. Bultje <rsbultje@gmail.com>
- *
- * 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_SNPRINTF_H
-#define COMPAT_SNPRINTF_H
-
-#include <stdarg.h>
-#include <stdio.h>
-
-int avpriv_snprintf(char *s, size_t n, const char *fmt, ...);
-int avpriv_vsnprintf(char *s, size_t n, const char *fmt, va_list ap);
-
-#undef snprintf
-#undef _snprintf
-#undef vsnprintf
-#define snprintf avpriv_snprintf
-#define _snprintf avpriv_snprintf
-#define vsnprintf avpriv_vsnprintf
-
-#endif /* COMPAT_SNPRINTF_H */
--- a/compat/plan9/head
+++ b/compat/plan9/head
@@ -1,10 +0,0 @@
-#!/bin/sh
-
-n=10
-
-case "$1" in
-    -n)  n=$2;      shift 2 ;;
-    -n*) n=${1#-n}; shift   ;;
-esac
-
-exec sed ${n}q "$@"
--- a/compat/plan9/main.c
+++ b/compat/plan9/main.c
@@ -1,34 +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
- */
-
-int plan9_main(int argc, char **argv);
-
-#undef main
-int main(int argc, char **argv)
-{
-    /* The setfcr() function in lib9 is broken, must use asm. */
-#ifdef __i386
-    short fcr;
-    __asm__ volatile ("fstcw        %0 \n"
-                      "or      $63, %0 \n"
-                      "fldcw        %0 \n"
-                      : "=m"(fcr));
-#endif
-
-    return plan9_main(argc, argv);
-}
--- a/compat/plan9/printf
+++ b/compat/plan9/printf
@@ -1,2 +0,0 @@
-#!/bin/sh
-exec awk "BEGIN { for (i = 2; i < ARGC; i++) printf \"$1\", ARGV[i] }" "$@"
--- a/compat/strtod.c
+++ b/compat/strtod.c
@@ -1,93 +0,0 @@
-/*
- * C99-compatible strtod() implementation
- * Copyright (c) 2012 Ronald S. Bultje <rsbultje@gmail.com>
- *
- * 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 <limits.h>
-#include <stdlib.h>
-
-#include "libavutil/avstring.h"
-#include "libavutil/mathematics.h"
-
-static char *check_nan_suffix(char *s)
-{
-    char *start = s;
-
-    if (*s++ != '(')
-        return start;
-
-    while ((*s >= 'a' && *s <= 'z') || (*s >= 'A' && *s <= 'Z') ||
-           (*s >= '0' && *s <= '9') ||  *s == '_')
-        s++;
-
-    return *s == ')' ? s + 1 : start;
-}
-
-#undef strtod
-double strtod(const char *, char **);
-
-double avpriv_strtod(const char *nptr, char **endptr)
-{
-    char *end;
-    double res;
-
-    /* Skip leading spaces */
-    while (av_isspace(*nptr))
-        nptr++;
-
-    if (!av_strncasecmp(nptr, "infinity", 8)) {
-        end = nptr + 8;
-        res = INFINITY;
-    } else if (!av_strncasecmp(nptr, "inf", 3)) {
-        end = nptr + 3;
-        res = INFINITY;
-    } else if (!av_strncasecmp(nptr, "+infinity", 9)) {
-        end = nptr + 9;
-        res = INFINITY;
-    } else if (!av_strncasecmp(nptr, "+inf", 4)) {
-        end = nptr + 4;
-        res = INFINITY;
-    } else if (!av_strncasecmp(nptr, "-infinity", 9)) {
-        end = nptr + 9;
-        res = -INFINITY;
-    } else if (!av_strncasecmp(nptr, "-inf", 4)) {
-        end = nptr + 4;
-        res = -INFINITY;
-    } else if (!av_strncasecmp(nptr, "nan", 3)) {
-        end = check_nan_suffix(nptr + 3);
-        res = NAN;
-    } else if (!av_strncasecmp(nptr, "+nan", 4) ||
-               !av_strncasecmp(nptr, "-nan", 4)) {
-        end = check_nan_suffix(nptr + 4);
-        res = NAN;
-    } else if (!av_strncasecmp(nptr, "0x", 2) ||
-               !av_strncasecmp(nptr, "-0x", 3) ||
-               !av_strncasecmp(nptr, "+0x", 3)) {
-        /* FIXME this doesn't handle exponents, non-integers (float/double)
-         * and numbers too large for long long */
-        res = strtoll(nptr, &end, 16);
-    } else {
-        res = strtod(nptr, &end);
-    }
-
-    if (endptr)
-        *endptr = end;
-
-    return res;
-}
--- a/compat/tms470/math.h
+++ b/compat/tms470/math.h
@@ -1,30 +0,0 @@
-/*
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#ifndef FFMPEG_COMPAT_TMS470_MATH_H
-#define FFMPEG_COMPAT_TMS470_MATH_H
-
-#include_next <math.h>
-
-#undef INFINITY
-#undef NAN
-
-#define INFINITY (*(const float*)((const unsigned []){ 0x7f800000 }))
-#define NAN      (*(const float*)((const unsigned []){ 0x7fc00000 }))
-
-#endif /* FFMPEG_COMPAT_TMS470_MATH_H */
--- a/compat/va_copy.h
+++ b/compat/va_copy.h
@@ -1,29 +0,0 @@
-/*
- * MSVC Compatible va_copy macro
- * Copyright (c) 2012 Derek Buitenhuis
- *
- * 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 <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
--- 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/3893
+++ b/3893
--- a/doc/APIchanges
+++ b/doc/APIchanges
@@ -1,622 +1,36 @@
 Never assume the API of libav* to be stable unless at least 1 month has passed
-since the last major version increase or the API was added.
+since the last major version increase.

 The last version increases were:
-libavcodec:    2013-03-xx
-libavdevice:   2013-03-xx
-libavfilter:   2013-12-xx
-libavformat:   2013-03-xx
-libavresample: 2012-10-05
+libavcodec:    2012-01-27
+libavdevice:   2011-04-18
+libavfilter:   2011-04-18
+libavformat:   2012-01-27
+libavresample: 2012-04-24
 libpostproc:   2011-04-18
 libswresample: 2011-09-19
 libswscale:    2011-06-20
-libavutil:     2012-10-22
+libavutil:     2011-04-18


 API changes, most recent first:

-2014-xx-xx - xxxxxxx - lavu 53.05.0 - frame.h
-  Add av_frame_copy() for copying the frame data.
-
-2014-02-xx - xxxxxxx - lswr 0.18.100 - swresample.h
-  Add swr_is_initialized() for checking whether a resample context is initialized.
-
-2014-02-xx - xxxxxxx - lavr 1.2.0 - avresample.h
-  Add avresample_is_open() for checking whether a resample context is open.
-
-2014-xx-xx - xxxxxxx - lavu 53.04.0  - opt.h
-  Add AV_OPT_FLAG_EXPORT and AV_OPT_FLAG_READONLY to mark options meant (only)
-  for reading.
-
-2014-xx-xx - xxxxxxx - lavu 53.03.01 - opt.h
-  Deprecate unused AV_OPT_FLAG_METADATA.
-
-2014-02-xx - xxxxxxx - lavd 55.10.100 - avdevice.h
-  Add avdevice_list_devices() and avdevice_free_list_devices()
-
-2014-02-16 - db3c970 - lavf 55.33.100 - avio.h
-  Add avio_find_protocol_name() to find out the name of the protocol that would
-  be selected for a given URL.
-
-2014-02-xx - xxxxxxx - lavu 53.3.0 - frame.h
-  Add AV_FRAME_DATA_DOWNMIX_INFO value to the AVFrameSideDataType enum and
-  downmix_info.h API, which identify downmix-related metadata.
-
-2014-02-11 - 1b05ac2 - lavf 55.32.100 - avformat.h
-  Add av_write_uncoded_frame() and av_interleaved_write_uncoded_frame().
-
-2014-02-04 - 3adb5f8 / d9ae103 - lavf 55.30.100 / 55.11.0 - avformat.h
-  Add AVFormatContext.max_interleave_delta for controlling amount of buffering
-  when interleaving.
-
-2014-02-02 - xxxxxxx - lavf 55.29.100 - avformat.h
-  Add output_ts_offset muxing option to AVFormatContext.
-
-2014-01-27 - 102bd64 - lavd 55.7.100 - avdevice.h
-                       lavf 55.28.100 - avformat.h
-  Add avdevice_dev_to_app_control_message() function.
-
-2014-01-27 - 7151411 - lavd 55.6.100 - avdevice.h
-                       lavf 55.27.100 - avformat.h
-  Add avdevice_app_to_dev_control_message() function.
-
-2014-01-24 - 86bee79 - lavf 55.26.100 - avformat.h
-  Add AVFormatContext option metadata_header_padding to allow control over the
-  amount of padding added.
-
-2014-01-20 - eef74b2 / 93c553c - lavc 55.48.102 / 55.32.1 - avcodec.h
-  Edges are not required anymore on video buffers allocated by get_buffer2()
-  (i.e. as if the CODEC_FLAG_EMU_EDGE flag was always on). Deprecate
-  CODEC_FLAG_EMU_EDGE and avcodec_get_edge_width().
-
-2014-01-19 - xxxxxxx - lavf 55.25.100 - avformat.h
-  Add avformat_get_mov_video_tags() and avformat_get_mov_audio_tags().
-
-2014-01-19 - xxxxxxx - lavu 52.63.100 - rational.h
-  Add av_make_q() function.
-
-2014-01-05 - 4cf4da9 / 5b4797a - lavu 52.62.100 / 53.2.0 - frame.h
-  Add AV_FRAME_DATA_MATRIXENCODING value to the AVFrameSideDataType enum, which
-  identifies AVMatrixEncoding data.
-
-2014-01-05 - 751385f / 5c437fb - lavu 52.61.100 / 53.1.0 - channel_layout.h
-  Add values for various Dolby flags to the AVMatrixEncoding enum.
-
-2014-01-04 - xxxxxxx - lavu 52.60.100 - mathematics.h
-  Add av_add_stable() function.
-
-2013-12-22 - xxxxxxx - lavu 52.59.100 - avstring.h
-  Add av_strnlen() function.
-
-2013-12-xx - xxxxxxx - lavu 52.57.100 - opencl.h
-  Add av_opencl_benchmark() function.
-
-2013-11-xx - xxxxxxx - lavu 52.56.100 - ffversion.h
-  Moves version.h to libavutil/ffversion.h.
-  Install ffversion.h and make it public.
-
-2013-12-11 - 29c83d2 / b9fb59d,409a143 / 9431356,44967ab / d7b3ee9 - lavc 55.45.101 / 55.28.1 - avcodec.h
-  av_frame_alloc(), av_frame_unref() and av_frame_free() now can and should be
-  used instead of avcodec_alloc_frame(), avcodec_get_frame_defaults() and
-  avcodec_free_frame() respectively. The latter three functions are deprecated.
-
-2013-12-09 - 7a60348 / 7e244c6- - lavu 52.58.100 / 52.20.0 - frame.h
-  Add AV_FRAME_DATA_STEREO3D value to the AVFrameSideDataType enum and
-  stereo3d.h API, that identify codec-independent stereo3d information.
-
-2013-11-26 - 625b290 / 1eaac1d- - lavu 52.55.100 / 52.19.0 - frame.h
-  Add AV_FRAME_DATA_A53_CC value to the AVFrameSideDataType enum, which
-  identifies ATSC A53 Part 4 Closed Captions data.
-
-2013-11-XX - xxxxxxx - lavu 52.54.100 - avstring.h
-  Add av_utf8_decode() function.
-
-2013-11-22 - fb7d70c - lavc 55.44.100 - avcodec.h
-  Add HEVC profiles
-
-2013-11-xx - xxxxxxx - lavc 55.44.100 - avcodec.h
-  Add av_packet_{un,}pack_dictionary()
-  Add AV_PKT_METADATA_UPDATE side data type, used to transmit key/value
-  strings between a stream and the application.
-
-2013-11-14 - 7c888ae / cce3e0a - lavu 52.53.100 / 52.18.0 - mem.h
-  Move av_fast_malloc() and av_fast_realloc() for libavcodec to libavutil.
-
-2013-11-14 - b71e4d8 / 8941971 - lavc 55.43.100 / 55.27.0 - avcodec.h
-  Deprecate AVCodecContext.error_rate, it is replaced by the 'error_rate'
-  private option of the mpegvideo encoder family.
-
-2013-11-14 - 31c09b7 / 728c465 - lavc 55.26.0 - vdpau.h
-  Add av_vdpau_get_profile().
-  Add av_vdpau_alloc_context(). This function must from now on be
-  used for allocating AVVDPAUContext.
-
-2013-11-04 - be41f21 / cd8f772 - lavc 55.41.100 / 55.25.0 - avcodec.h
-                       lavu 52.51.100 - frame.h
-  Add ITU-R BT.2020 and other not yet included values to color primaries,
-  transfer characteristics and colorspaces.
-
-2013-11-04 - xxxxxxx - lavu 52.50.100 - avutil.h
-  Add av_fopen_utf8()
-
-2013-10-31 - 78265fc / 28096e0 - lavu 52.49.100 / 52.17.0 - frame.h
-  Add AVFrame.flags and AV_FRAME_FLAG_CORRUPT.
-
-2013-10-27 - xxxxxxx - lavc 55.39.100 - avcodec.h
-  Add CODEC_CAP_DELAY support to avcodec_decode_subtitle2.
-
-2013-10-27 - xxxxxxx - lavu 52.48.100 - parseutils.h
-  Add av_get_known_color_name().
-
-2013-10-17 - xxxxxxx - lavu 52.47.100 - opt.h
-  Add AV_OPT_TYPE_CHANNEL_LAYOUT and channel layout option handlers
-  av_opt_get_channel_layout() and av_opt_set_channel_layout().
-
-2013-10-xx - xxxxxxx -libswscale 2.5.101 - options.c
-  Change default scaler to bicubic
-
-2013-10-03 - xxxxxxx - lavc 55.34.100 - avcodec.h
-  Add av_codec_get_max_lowres()
-
-2013-10-02 - xxxxxxx - lavf 55.19.100 - avformat.h
-  Add audio/video/subtitle AVCodec fields to AVFormatContext to force specific
-  decoders
-
-2013-09-28 - 7381d31 / 0767bfd - lavfi 3.88.100 / 3.11.0 - avfilter.h
-  Add AVFilterGraph.execute and AVFilterGraph.opaque for custom slice threading
-  implementations.
-
-2013-09-21 - 85f8a3c / e208e6d - lavu 52.46.100 / 52.16.0 - pixfmt.h
-  Add interleaved 4:2:2 8/10-bit formats AV_PIX_FMT_NV16 and
-  AV_PIX_FMT_NV20.
-
-2013-09-16 - c74c3fb / 3feb3d6 - lavu 52.44.100 / 52.15.0 - mem.h
-  Add av_reallocp.
-
-2013-09-04 - 3e1f507 - lavc 55.31.101 - avcodec.h
-  avcodec_close() argument can be NULL.
-
-2013-09-04 - 36cd017 - lavf 55.16.101 - avformat.h
-  avformat_close_input() argument can be NULL and point on NULL.
-
-2013-08-29 - e31db62 - lavf 55.15.100 - avformat.h
-  Add av_format_get_probe_score().
-
-2013-08-15 - 1e0e193 - lsws 2.5.100 -
-  Add a sws_dither AVOption, allowing to set the dither algorithm used
-
-2013-08-xx - xxxxxxx - lavc 55.27.100 - vdpau.h
-  Add a render2 alternative to the render callback function.
-
-2013-08-xx - xxxxxxx - lavc 55.26.100 - vdpau.h
-  Add allocation function for AVVDPAUContext, allowing
-  to extend it in the future without breaking ABI/API.
-
-2013-08-10 - 67a580f / 5a9a9d4 - lavc 55.25.100 / 55.16.0 - avcodec.h
-  Extend AVPacket API with av_packet_unref, av_packet_ref,
-  av_packet_move_ref, av_packet_copy_props, av_packet_free_side_data.
-
-2013-08-05 - 9547e3e / f824535 - lavc 55.22.100 / 55.13.0 - avcodec.h
-  Deprecate the bitstream-related members from struct AVVDPAUContext.
-  The bistream buffers no longer need to be explicitly freed.
-
-2013-08-05 - 3b805dc / 549294f - lavc 55.21.100 / 55.12.0 - avcodec.h
-  Deprecate the CODEC_CAP_HWACCEL_VDPAU codec capability. Use CODEC_CAP_HWACCEL
-  and select the AV_PIX_FMT_VDPAU format with get_format() instead.
-
-2013-08-05 - 4ee0984 / a0ad5d0 - lavu 52.41.100 / 52.14.0 - pixfmt.h
-  Deprecate AV_PIX_FMT_VDPAU_*. Use AV_PIX_FMT_VDPAU instead.
-
-2013-08-02 - 82fdfe8 / a8b1927 - lavc 55.20.100 / 55.11.0 - avcodec.h
-  Add output_picture_number to AVCodecParserContext.
-
-2013-07-23 - abc8110 - lavc 55.19.100 - avcodec.h
-  Add avcodec_chroma_pos_to_enum()
-  Add avcodec_enum_to_chroma_pos()
-
-2013-07-03 - 838bd73 - lavfi 3.78.100 - avfilter.h
-  Deprecate avfilter_graph_parse() in favor of the equivalent
-  avfilter_graph_parse_ptr().
-
-2013-06-24 - af5f9c0 / 95d5246 - lavc 55.17.100 / 55.10.0 - avcodec.h
-  Add MPEG-2 AAC profiles
-
-2013-06-25 - af5f9c0 / 95d5246 - lavf 55.10.100 - avformat.h
-  Add AV_DISPOSITION_* flags to indicate text track kind.
-
-2013-06-15 - 99b8cd0 - lavu 52.36.100
-  Add AVRIPEMD:
-   av_ripemd_alloc()
-   av_ripemd_init()
-   av_ripemd_update()
-   av_ripemd_final()
-
-2013-06-04 - 30b491f / fc962d4 - lavu 52.35.100 / 52.13.0 - mem.h
-  Add av_realloc_array and av_reallocp_array
-
-2013-05-30 - 682b227 - lavu 52.35.100
-  Add AVSHA512:
-   av_sha512_alloc()
-   av_sha512_init()
-   av_sha512_update()
-   av_sha512_final()
-
-2013-05-24 - 8d4e969 / 129bb23 - lavfi 3.10.0 / 3.70.100 - avfilter.h
-  Add support for slice multithreading to lavfi. Filters supporting threading
-  are marked with AVFILTER_FLAG_SLICE_THREADS.
-  New fields AVFilterContext.thread_type, AVFilterGraph.thread_type and
-  AVFilterGraph.nb_threads (accessible directly or through AVOptions) may be
-  used to configure multithreading.
-
-2013-05-24 - fe40a9f / 2a6eaea - lavu 52.12.0 / 52.34.100 - cpu.h
-  Add av_cpu_count() function for getting the number of logical CPUs.
-
-2013-05-24 - 0c25c39 / b493847 - lavc 55.7.0 / 55.12.100 - avcodec.h
-  Add picture_structure to AVCodecParserContext.
-
-2013-05-17 - 3a751ea - lavu 52.33.100 - opt.h
-  Add AV_OPT_TYPE_COLOR value to AVOptionType enum.
-
-2013-05-13 - e398416 - lavu 52.31.100 - mem.h
-  Add av_dynarray2_add().
-
-2013-05-12 - 1776177 - lavfi 3.65.100
-  Add AVFILTER_FLAG_SUPPORT_TIMELINE* filter flags.
-
-2013-04-19 - 380cfce - lavc 55.4.100
-  Add AV_CODEC_PROP_TEXT_SUB property for text based subtitles codec.
-
-2013-04-18 - 7c1a002 - lavf 55.3.100
-  The matroska demuxer can now output proper verbatim ASS packets. It will
-  become the default starting lavf 56.0.100.
-
-2013-04-10 - af0d270 - lavu 25.26.100 - avutil.h,opt.h
-  Add av_int_list_length()
-  and av_opt_set_int_list().
-
-2013-03-30 - 5c73645 - lavu 52.24.100 - samplefmt.h
-  Add av_samples_alloc_array_and_samples().
-
-2013-03-29 - ef7b6b4 - lavf 55.1.100 - avformat.h
-  Add av_guess_frame_rate()
-
-2013-03-20 - 8d928a9 - lavu 52.22.100 - opt.h
-  Add AV_OPT_TYPE_DURATION value to AVOptionType enum.
-
-2013-03-17 - 7aa9af5 - lavu 52.20.100 - opt.h
-  Add AV_OPT_TYPE_VIDEO_RATE value to AVOptionType enum.
-
-2013-03-07 - 9767ec6 - lavu 52.18.100 - avstring.h,bprint.h
-  Add av_escape() and av_bprint_escape() API.
-
-2013-02-24 - b59cd08 - lavfi 3.41.100 - buffersink.h
-  Add sample_rates field to AVABufferSinkParams.
-
-2013-01-17 - a1a707f - lavf 54.61.100
-  Add av_codec_get_tag2().
-
-2013-01-01 - 2eb2e17 - lavfi 3.34.100
-  Add avfilter_get_audio_buffer_ref_from_arrays_channels.
-
-2012-12-20 - 34de47aa - lavfi 3.29.100 - avfilter.h
-  Add AVFilterLink.channels, avfilter_link_get_channels()
-  and avfilter_ref_get_channels().
-
-2012-12-15 - 96d815fc - lavc 54.80.100 - avcodec.h
-  Add pkt_size field to AVFrame.
-
-2012-11-25 - c70ec631 - lavu 52.9.100 - opt.h
-  Add the following convenience functions to opt.h:
-   av_opt_get_image_size
-   av_opt_get_pixel_fmt
-   av_opt_get_sample_fmt
-   av_opt_set_image_size
-   av_opt_set_pixel_fmt
-   av_opt_set_sample_fmt
-
-2012-11-17 - 4cd74c81 - lavu 52.8.100 - bprint.h
-  Add av_bprint_strftime().
-
-2012-11-15 - 92648107 - lavu 52.7.100 - opt.h
-  Add av_opt_get_key_value().
-
-2012-11-13 - 79456652 - lavfi 3.23.100 - avfilter.h
-  Add channels field to AVFilterBufferRefAudioProps.
-
-2012-11-03 - 481fdeee - lavu 52.3.100 - opt.h
-  Add AV_OPT_TYPE_SAMPLE_FMT value to AVOptionType enum.
-
-2012-10-21 - 6fb2fd8 - lavc  54.68.100 - avcodec.h
-                       lavfi  3.20.100 - avfilter.h
-  Add AV_PKT_DATA_STRINGS_METADATA side data type, used to transmit key/value
-  strings between AVPacket and AVFrame, and add metadata field to
-  AVCodecContext (which shall not be accessed by users; see AVFrame metadata
-  instead).
-
-2012-09-27 - a70b493 - lavd 54.3.100 - version.h
-  Add LIBAVDEVICE_IDENT symbol.
-
-2012-09-27 - a70b493 - lavfi 3.18.100 - version.h
-  Add LIBAVFILTER_IDENT symbol.
-
-2012-09-27 - a70b493 - libswr 0.16.100 - version.h
-  Add LIBSWRESAMPLE_VERSION, LIBSWRESAMPLE_BUILD
-  and LIBSWRESAMPLE_IDENT symbols.
-
-2012-09-06 - 29e972f - lavu 51.72.100 - parseutils.h
-  Add av_small_strptime() time parsing function.
-
-  Can be used as a stripped-down replacement for strptime(), on
-  systems which do not support it.
-
-2012-08-25 - 2626cc4 - lavf 54.28.100
-  Matroska demuxer now identifies SRT subtitles as AV_CODEC_ID_SUBRIP instead
-  of AV_CODEC_ID_TEXT.
-
-2012-08-13 - 5c0d8bc - lavfi 3.8.100 - avfilter.h
-  Add avfilter_get_class() function, and priv_class field to AVFilter
-  struct.
-
-2012-08-12 - a25346e - lavu 51.69.100 - opt.h
-  Add AV_OPT_FLAG_FILTERING_PARAM symbol in opt.h.
-
-2012-07-31 - 23fc4dd - lavc 54.46.100
-  Add channels field to AVFrame.
-
-2012-07-30 - f893904 - lavu 51.66.100
-  Add av_get_channel_description()
-  and av_get_standard_channel_layout() functions.
-
-2012-07-21 - 016a472 - lavc 54.43.100
-  Add decode_error_flags field to AVFrame.
-
-2012-07-20 - b062936 - lavf 54.18.100
-  Add avformat_match_stream_specifier() function.
-
-2012-07-14 - f49ec1b - lavc 54.38.100 - avcodec.h
-  Add metadata to AVFrame, and the accessor functions
-  av_frame_get_metadata() and av_frame_set_metadata().
-
-2012-07-10 - 0e003d8 - lavc 54.33.100
-  Add av_fast_padded_mallocz().
-
-2012-07-10 - 21d5609 - lavfi 3.2.0 - avfilter.h
-  Add init_opaque() callback to AVFilter struct.
-
-2012-06-26 - e6674e4 - lavu 51.63.100 - imgutils.h
-  Add functions to libavutil/imgutils.h:
-  av_image_get_buffer_size()
-  av_image_fill_arrays()
-  av_image_copy_to_buffer()
-
-2012-06-24 - c41899a - lavu 51.62.100 - version.h
-  version moved from avutil.h to version.h
-
-2012-04-11 - 359abb1 - lavu 51.58.100 - error.h
-  Add av_make_error_string() and av_err2str() utilities to
-  libavutil/error.h.
-
-2012-06-05 - 62b39d4 - lavc 54.24.100
-  Add pkt_duration field to AVFrame.
-
-2012-05-24 - f2ee065 - lavu 51.54.100
+2012-05-24 - xxxxxxx - lavu 51.54.100
  Move AVPALETTE_SIZE and AVPALETTE_COUNT macros from
  libavcodec/avcodec.h to libavutil/pixfmt.h.

-2012-05-14 - 94a9ac1 - lavf 54.5.100
+2012-05-07 - xxxxxxx - lavf 54.5.100
  Add av_guess_sample_aspect_ratio() function.

-2012-04-20 - 65fa7bc - lavfi 2.70.100
+2012-04-20 - xxxxxxx - lavfi 2.70.100
  Add avfilter_unref_bufferp() to avfilter.h.

-2012-04-13 - 162e400 - lavfi 2.68.100
+2012-04-12 - xxxxxxx - lavfi 2.68.100
  Install libavfilter/asrc_abuffer.h public header.

 2012-03-26 - a67d9cf - lavfi 2.66.100
  Add avfilter_fill_frame_from_{audio_,}buffer_ref() functions.

-2013-05-15 - ff46809 / e6c4ac7 - lavu 52.32.100 / 52.11.0 - pixdesc.h
-  Replace PIX_FMT_* flags with AV_PIX_FMT_FLAG_*.
-
-2013-04-03 - 6fc58a8 / 507b1e4 - lavc 55.7.100 / 55.4.0 - avcodec.h
-  Add field_order to AVCodecParserContext.
-
-2013-04-19 - f4b05cd / 5e83d9a - lavc 55.5.100 / 55.2.0 - avcodec.h
-  Add CODEC_FLAG_UNALIGNED to allow decoders to produce unaligned output.
-
-2013-04-11 - lavfi 3.53.100 / 3.8.0
-  231fd44 / 38f0c07 - Move all content from avfiltergraph.h to avfilter.h. Deprecate
-            avfilterhraph.h, user applications should include just avfilter.h
-  86070b8 / bc1a985 - Add avfilter_graph_alloc_filter(), deprecate avfilter_open() and
-            avfilter_graph_add_filter().
-  4fde705 / 1113672 - Add AVFilterContext.graph pointing to the AVFilterGraph that contains the
-            filter.
-  710b0aa / 48a5ada - Add avfilter_init_str(), deprecate avfilter_init_filter().
-  46de9ba / 1ba95a9 - Add avfilter_init_dict().
-  16fc24b / 7cdd737 - Add AVFilter.flags field and AVFILTER_FLAG_DYNAMIC_{INPUTS,OUTPUTS} flags.
-  f4db6bf / 7e8fe4b - Add avfilter_pad_count() for counting filter inputs/outputs.
-  835cc0f / fa2a34c - Add avfilter_next(), deprecate av_filter_next().
-            Deprecate avfilter_uninit().
-
-2013-04-09 - lavfi 3.51.100 / 3.7.0 - avfilter.h
-  0594ef0 / b439c99 - Add AVFilter.priv_class for exporting filter options through the
-            AVOptions API in the similar way private options work in lavc and lavf.
-  44d4488 / 8114c10 - Add avfilter_get_class().
-  Switch all filters to use AVOptions.
-
-2013-03-19 - 17ebef2 / 2c328a9 - lavu 52.20.100 / 52.9.0 - pixdesc.h
-  Add av_pix_fmt_count_planes() function for counting planes in a pixel format.
-
-2013-03-16 - ecade98 / 42c7c61 - lavfi 3.47.100 / 3.6.0
-  Add AVFilterGraph.nb_filters, deprecate AVFilterGraph.filter_count.
-
-2013-03-08 - Reference counted buffers - lavu 52.8.0, lavc 55.0.100 / 55.0.0, lavf 55.0.100 / 55.0.0,
-lavd 54.4.100 / 54.0.0, lavfi 3.5.0
-  36099df / 8e401db, 532f31a / 1cec062 - add a new API for reference counted buffers and buffer
-                     pools (new header libavutil/buffer.h).
-  2653e12 / 1afddbe - add AVPacket.buf to allow reference counting for the AVPacket data.
-            Add av_packet_from_data() function for constructing packets from
-            av_malloc()ed data.
-  c4e8821 / 7ecc2d4 - move AVFrame from lavc to lavu (new header libavutil/frame.h), add
-            AVFrame.buf/extended_buf to allow reference counting for the AVFrame
-            data. Add new API for working with reference-counted AVFrames.
-  80e9e63 / 759001c - add the refcounted_frames field to AVCodecContext to make audio and
-            video decoders return reference-counted frames. Add get_buffer2()
-            callback to AVCodecContext which allocates reference-counted frames.
-            Add avcodec_default_get_buffer2() as the default get_buffer2()
-            implementation.
-            Deprecate AVCodecContext.get_buffer() / release_buffer() /
-            reget_buffer(), avcodec_default_get_buffer(),
-            avcodec_default_reget_buffer(), avcodec_default_release_buffer().
-            Remove avcodec_default_free_buffers(), which should not have ever
-            been called from outside of lavc.
-            Deprecate the following AVFrame fields:
-                * base -- is now stored in AVBufferRef
-                * reference, type, buffer_hints -- are unnecessary in the new API
-                * hwaccel_picture_private, owner, thread_opaque -- should not
-                  have been acessed from outside of lavc
-                * qscale_table, qstride, qscale_type, mbskip_table, motion_val,
-                  mb_type, dct_coeff, ref_index -- mpegvideo-specific tables,
-                  which are not exported anymore.
-  a05a44e / 7e35037 - switch libavfilter to use AVFrame instead of AVFilterBufferRef. Add
-            av_buffersrc_add_frame(), deprecate av_buffersrc_buffer().
-            Add av_buffersink_get_frame() and av_buffersink_get_samples(),
-            deprecate av_buffersink_read() and av_buffersink_read_samples().
-            Deprecate AVFilterBufferRef and all functions for working with it.
-
-2013-03-17 - 6c17ff8 / 12c5c1d - lavu 52.19.100 / 52.8.0 - avstring.h
-  Add av_isdigit, av_isgraph, av_isspace, av_isxdigit.
-
-2013-02-23 - 71cf094 / 9f12235 - lavfi 3.40.100 / 3.4.0 - avfiltergraph.h
-  Add resample_lavr_opts to AVFilterGraph for setting libavresample options
-  for auto-inserted resample filters.
-
-2013-01-25 - e7e14bc / 38c1466 - lavu 52.17.100 / 52.7.0 - dict.h
-  Add av_dict_parse_string() to set multiple key/value pairs at once from a
-  string.
-
-2013-01-25 - 25be630 / b85a5e8 - lavu 52.16.100 / 52.6.0 - avstring.h
-  Add av_strnstr()
-
-2013-01-15 - e7e0186 / 8ee288d - lavu 52.15.100 / 52.5.0 - hmac.h
-  Add AVHMAC.
-
-2013-01-13 - 8ee7b38 / 44e065d - lavc 54.87.100 / 54.36.0 - vdpau.h
-  Add AVVDPAUContext struct for VDPAU hardware-accelerated decoding.
-
-2013-01-12 - dae382b / 169fb94 - lavu 52.14.100 / 52.4.0 - pixdesc.h
-  Add AV_PIX_FMT_VDPAU flag.
-
-2013-01-07 - 249fca3 / 074a00d - lavr 1.1.0
-  Add avresample_set_channel_mapping() for input channel reordering,
-  duplication, and silencing.
-
-2012-12-29 - 2ce43b3 / d8fd06c - lavu 52.13.100 / 52.3.0 - avstring.h
-  Add av_basename() and av_dirname().
-
-2012-11-11 - 03b0787 / 5980f5d - lavu 52.6.100 / 52.2.0 - audioconvert.h
-  Rename audioconvert.h to channel_layout.h. audioconvert.h is now deprecated.
-
-2012-11-05 - 7d26be6 / dfde8a3 - lavu 52.5.100 / 52.1.0 - intmath.h
-  Add av_ctz() for trailing zero bit count
-
-2012-10-21 - e3a91c5 / a893655 - lavu 51.77.100 / 51.45.0 - error.h
-  Add AVERROR_EXPERIMENTAL
-
-2012-10-12 - a33ed6b / d2fcb35 - lavu 51.76.100 / 51.44.0 - pixdesc.h
-  Add functions for accessing pixel format descriptors.
-  Accessing the av_pix_fmt_descriptors array directly is now
-  deprecated.
-
-2012-10-11 - f391e40 / 9a92aea - lavu 51.75.100 / 51.43.0 - aes.h, md5.h, sha.h, tree.h
-  Add functions for allocating the opaque contexts for the algorithms,
-
-2012-10-10 - de31814 / b522000 - lavf 54.32.100 / 54.18.0 - avio.h
-  Add avio_closep to complement avio_close.
-
-2012-10-08 - ae77266 / 78071a1 - lavu 51.74.100 / 51.42.0 - pixfmt.h
-  Rename PixelFormat to AVPixelFormat and all PIX_FMT_* to AV_PIX_FMT_*.
-  To provide backwards compatibility, PixelFormat is now #defined as
-  AVPixelFormat.
-  Note that this can break user code that includes pixfmt.h and uses the
-  'PixelFormat' identifier. Such code should either #undef PixelFormat
-  or stop using the PixelFormat name.
-
-2012-10-05 - 55c49af / e7ba5b1 - lavr 1.0.0 - avresample.h
-  Data planes parameters to avresample_convert() and
-  avresample_read() are now uint8_t** instead of void**.
-  Libavresample is now stable.
-
-2012-09-24 - 46a3595 / a42aada - lavc 54.59.100 / 54.28.0 - avcodec.h
-  Add avcodec_free_frame(). This function must now
-  be used for freeing an AVFrame.
-
-2012-09-12 - e3e09f2 / 8919fee - lavu 51.73.100 / 51.41.0 - audioconvert.h
-  Added AV_CH_LOW_FREQUENCY_2 channel mask value.
-
-2012-09-04 - b21b5b0 / 686a329 - lavu 51.71.100 / 51.40.0 - opt.h
-  Reordered the fields in default_val in AVOption, changed which
-  default_val field is used for which AVOptionType.
-
-2012-08-30 - 98298eb / a231832 - lavc 54.54.101 / 54.26.1 - avcodec.h
-  Add codec descriptor properties AV_CODEC_PROP_LOSSY and
-  AV_CODEC_PROP_LOSSLESS.
-
-2012-08-18 - lavc 54.26 - avcodec.h
-  Add codec descriptors for accessing codec properties without having
-  to refer to a specific decoder or encoder.
-
-  f5f3684 / c223d79 - Add an AVCodecDescriptor struct and functions
-            avcodec_descriptor_get() and avcodec_descriptor_next().
-  f5f3684 / 51efed1 - Add AVCodecDescriptor.props and AV_CODEC_PROP_INTRA_ONLY.
-  6c180b3 / 91e59fe - Add avcodec_descriptor_get_by_name().
-
-2012-08-08 - f5f3684 / 987170c - lavu 51.68.100 / 51.38.0 - dict.h
-  Add av_dict_count().
-
-2012-08-07 - 7a72695 / 104e10f - lavc 54.51.100 / 54.25.0 - avcodec.h
-  Rename CodecID to AVCodecID and all CODEC_ID_* to AV_CODEC_ID_*.
-  To provide backwards compatibility, CodecID is now #defined as AVCodecID.
-  Note that this can break user code that includes avcodec.h and uses the
-  'CodecID' identifier. Such code should either #undef CodecID or stop using the
-  CodecID name.
-
-2012-08-03 - e776ee8 / 239fdf1 - lavu 51.66.101 / 51.37.1 - cpu.h
-                       lsws 2.1.1   - swscale.h
-  Rename AV_CPU_FLAG_MMX2  ---> AV_CPU_FLAG_MMXEXT.
-  Rename SWS_CPU_CAPS_MMX2 ---> SWS_CPU_CAPS_MMXEXT.
-
-2012-07-29 - 7c26761 / 681ed00 - lavf 54.22.100 / 54.13.0 - avformat.h
-  Add AVFMT_FLAG_NOBUFFER for low latency use cases.
-
-2012-07-10 - fbe0245 / f3e5e6f - lavu 51.65.100 / 51.37.0
-  Add av_malloc_array() and av_mallocz_array()
-
-2012-06-22 - e847f41 / d3d3a32 - lavu 51.61.100 / 51.34.0
-  Add av_usleep()
-
-2012-06-20 - 4da42eb / ae0a301 - lavu 51.60.100 / 51.33.0
-  Move av_gettime() to libavutil, add libavutil/time.h
-
-2012-06-09 - 82edf67 / 3971be0 - lavr 0.0.3
-  Add a parameter to avresample_build_matrix() for Dolby/DPLII downmixing.
-
-2012-06-12 - c7b9eab / 9baeff9 - lavfi 2.79.100 / 2.23.0 - avfilter.h
-  Add AVFilterContext.nb_inputs/outputs. Deprecate
-  AVFilterContext.input/output_count.
-
-2012-06-12 - c7b9eab / 84b9fbe - lavfi 2.79.100 / 2.22.0 - avfilter.h
-  Add avfilter_pad_get_type() and avfilter_pad_get_name(). Those
-  should now be used instead of accessing AVFilterPad members
-  directly.
-
-2012-06-12 - 3630a07 / b0f0dfc - lavu 51.57.100 / 51.32.0 - audioconvert.h
-  Add av_get_channel_layout_channel_index(), av_get_channel_name()
-  and av_channel_layout_extract_channel().
-
-2012-05-25 - 53ce990 / 154486f - lavu 51.55.100 / 51.31.0 - opt.h
-  Add av_opt_set_bin()
-
 2012-05-15 - lavfi 2.74.100 / 2.17.0
  Add support for audio filters
  61930bd / ac71230, 1cbf7fb / a2cd9be - add video/audio buffer sink in a new installed
@@ -692,10 +106,6 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
 2012-01-24 - 0c3577b - lavfi 2.60.100
  Add avfilter_graph_dump.

-2012-03-20 - 0ebd836 / 3c90cc2 - lavfo 54.2.0
-  Deprecate av_read_packet(), use av_read_frame() with
-  AVFMT_FLAG_NOPARSE | AVFMT_FLAG_NOFILLIN in AVFormatContext.flags
-
 2012-03-05 - lavc 54.10.100 / 54.8.0
  f095391 / 6699d07 Add av_get_exact_bits_per_sample()
  f095391 / 9524cf7 Add av_get_audio_frame_duration()
--- a/doc/Makefile
+++ b/doc/Makefile
@@ -1,76 +1,36 @@
-LIBRARIES-$(CONFIG_AVUTIL)     += libavutil
-LIBRARIES-$(CONFIG_SWSCALE)    += libswscale
-LIBRARIES-$(CONFIG_SWRESAMPLE) += libswresample
-LIBRARIES-$(CONFIG_AVCODEC)    += libavcodec
-LIBRARIES-$(CONFIG_AVFORMAT)   += libavformat
-LIBRARIES-$(CONFIG_AVDEVICE)   += libavdevice
-LIBRARIES-$(CONFIG_AVFILTER)   += libavfilter
-
-COMPONENTS-$(CONFIG_AVUTIL)     += ffmpeg-utils
-COMPONENTS-$(CONFIG_SWSCALE)    += ffmpeg-scaler
-COMPONENTS-$(CONFIG_SWRESAMPLE) += ffmpeg-resampler
-COMPONENTS-$(CONFIG_AVCODEC)    += ffmpeg-codecs ffmpeg-bitstream-filters
-COMPONENTS-$(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)
-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) \
+MANPAGES    = $(PROGS-yes:%=doc/%.1)
+PODPAGES    = $(PROGS-yes:%=doc/%.pod)
+HTMLPAGES   = $(PROGS-yes:%=doc/%.html)                                 \
              doc/developer.html                                        \
              doc/faq.html                                              \
              doc/fate.html                                             \
              doc/general.html                                          \
              doc/git-howto.html                                        \
-              doc/nut.html                                              \
+              doc/libavfilter.html                                      \
              doc/platform.html                                         \
+              doc/syntax.html                                           \

 TXTPAGES    = doc/fate.txt                                              \


-DOCS-$(CONFIG_HTMLPAGES) += $(HTMLPAGES)
-DOCS-$(CONFIG_PODPAGES)  += $(PODPAGES)
-DOCS-$(CONFIG_MANPAGES)  += $(MANPAGES)
-DOCS-$(CONFIG_TXTPAGES)  += $(TXTPAGES)
-DOCS = $(DOCS-yes)
+DOCS = $(HTMLPAGES) $(MANPAGES) $(PODPAGES)
+ifdef HAVE_MAKEINFO
+DOCS += $(TXTPAGES)
+endif

-DOC_EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE)      += avio_reading
-DOC_EXAMPLES-$(CONFIG_AVCODEC_EXAMPLE)           += avcodec
-DOC_EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding
-DOC_EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE)      += filter_audio
-DOC_EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE)   += filtering_audio
-DOC_EXAMPLES-$(CONFIG_FILTERING_VIDEO_EXAMPLE)   += filtering_video
-DOC_EXAMPLES-$(CONFIG_METADATA_EXAMPLE)          += metadata
-DOC_EXAMPLES-$(CONFIG_MUXING_EXAMPLE)            += muxing
-DOC_EXAMPLES-$(CONFIG_REMUXING_EXAMPLE)          += remuxing
-DOC_EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE)  += resampling_audio
-DOC_EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE)     += scaling_video
-DOC_EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE)     += transcode_aac
-ALL_DOC_EXAMPLES_LIST = $(DOC_EXAMPLES-) $(DOC_EXAMPLES-yes)
+all-$(CONFIG_DOC): documentation

-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
-
-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
 	$(Q)$(TEXIDEP)
 	$(M)makeinfo --force --no-headers -o $@ $< 2>/dev/null

+doc/print_options.o: libavformat/options_table.h libavcodec/options_table.h
+
 GENTEXI  = format codec
 GENTEXI := $(GENTEXI:%=doc/avoptions_%.texi)

@@ -81,87 +41,33 @@ $(GENTEXI): doc/avoptions_%.texi: doc/print_options$(HOSTEXESUF)
 doc/%.html: TAG = HTML
 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 $@ $<
+	$(M)texi2html -I doc -monolithic --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<

 doc/%.pod: TAG = POD
-doc/%.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI)
+doc/%.pod: doc/%.texi $(GENTEXI)
 	$(Q)$(TEXIDEP)
-	$(M)perl $(SRC_PATH)/doc/texi2pod.pl -Dconfig-not-all=yes -Idoc $< $@
+	$(M)$(SRC_PATH)/doc/texi2pod.pl -Idoc $< $@

-doc/%-all.pod: TAG = POD
-doc/%-all.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI)
-	$(Q)$(TEXIDEP)
-	$(M)perl $(SRC_PATH)/doc/texi2pod.pl -Dconfig-all=yes -Idoc $< $@
-
-doc/%.1 doc/%.3: TAG = MAN
+doc/%.1: TAG = MAN
 doc/%.1: doc/%.pod $(GENTEXI)
 	$(M)pod2man --section=1 --center=" " --release=" " $< > $@
-doc/%.3: doc/%.pod $(GENTEXI)
-	$(M)pod2man --section=3 --center=" " --release=" " $< > $@

-$(DOCS) doc/doxy/html: | doc/
-$(DOC_EXAMPLES:%$(EXESUF)=%.o): | doc/examples
-OBJDIRS += doc/examples
+$(DOCS): | doc/

-DOXY_INPUT      = $(addprefix $(SRC_PATH)/, $(INSTHEADERS) $(DOC_EXAMPLES:%$(EXESUF)=%.c) $(LIB_EXAMPLES:%$(EXESUF)=%.c))
-
-doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(DOXY_INPUT)
-	$(M)$(SRC_PATH)/doc/doxy-wrapper.sh $(SRC_PATH) $< $(DOXY_INPUT)
-
-install-doc: install-html install-man
-
-install-html:
-
-install-man:
-
-ifdef CONFIG_HTMLPAGES
-install-progs-$(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

 install-man: $(MANPAGES)
 	$(Q)mkdir -p "$(MANDIR)/man1"
-	$(INSTALL) -m 644 $(MANPAGES1) "$(MANDIR)/man1"
-	$(Q)mkdir -p "$(MANDIR)/man3"
-	$(INSTALL) -m 644 $(MANPAGES3) "$(MANDIR)/man3"
-endif
+	$(INSTALL) -m 644 $(MANPAGES) "$(MANDIR)/man1"

-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)/man3/",$(LIBRARIES-yes:%=%.3))
+	$(RM) $(addprefix "$(MANDIR)/man1/",$(ALLMANPAGES))

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

 -include $(wildcard $(DOCS:%=%.d))

-.PHONY: apidoc doc documentation
+.PHONY: documentation
--- a/doc/RELEASE_NOTES
+++ b/doc/RELEASE_NOTES
@@ -1,7 +1,7 @@
 Release Notes
 =============

-* 2.2 "Muybridge"  March, 2014
+* 0.11 "Happiness"  May, 2012


 General notes
--- a/doc/authors.texi
+++ b/doc/authors.texi
@@ -1,11 +0,0 @@
-@chapter Authors
-
-The FFmpeg developers.
-
-For details about the authorship, see the Git history of the project
-(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
-@command{git log} in the FFmpeg source directory, or browsing the
-online repository at @url{http://source.ffmpeg.org}.
-
-Maintainers for the specific components are listed in the file
-@file{MAINTAINERS} in the source code tree.
--- a/doc/avtools-common-opts.texi
+++ b/doc/avtools-common-opts.texi
@@ -0,0 +1,182 @@
+All the numerical options, if not specified otherwise, accept in input
+a string representing a number, which may contain one of the
+International System number postfixes, for example 'K', 'M', 'G'.
+If 'i' is appended after the postfix, powers of 2 are used instead of
+powers of 10. The 'B' postfix multiplies the value for 8, and can be
+appended after another postfix or used alone. This allows using for
+example 'KB', 'MiB', 'G' and 'B' as postfix.
+
+Options which do not take arguments are boolean options, and set the
+corresponding value to true. They can be set to false by prefixing
+with "no" the option name, for example using "-nofoo" in the
+command line will set to false the boolean option with name "foo".
+
+@anchor{Stream specifiers}
+@section Stream specifiers
+Some options are applied per-stream, e.g. bitrate or codec. Stream specifiers
+are used to precisely specify which stream(s) does a given option belong to.
+
+A stream specifier is a string generally appended to the option name and
+separated from it by a colon. E.g. @code{-codec:a:1 ac3} option contains
+@code{a:1} stream specifer, which matches the second audio stream. Therefore it
+would select the ac3 codec for the second audio stream.
+
+A stream specifier can match several stream, the option is then applied to all
+of them. E.g. the stream specifier in @code{-b:a 128k} matches all audio
+streams.
+
+An empty stream specifier matches all streams, for example @code{-codec copy}
+or @code{-codec: copy} would copy all the streams without reencoding.
+
+Possible forms of stream specifiers are:
+@table @option
+@item @var{stream_index}
+Matches the stream with this index. E.g. @code{-threads:1 4} would set the
+thread count for the second stream to 4.
+@item @var{stream_type}[:@var{stream_index}]
+@var{stream_type} is one of: 'v' for video, 'a' for audio, 's' for subtitle,
+'d' for data and 't' for attachments. If @var{stream_index} is given, then
+matches stream number @var{stream_index} of this type. Otherwise matches all
+streams of this type.
+@item p:@var{program_id}[:@var{stream_index}]
+If @var{stream_index} is given, then matches stream number @var{stream_index} in
+program with id @var{program_id}. Otherwise matches all streams in this program.
+@item #@var{stream_id}
+Matches the stream by format-specific ID.
+@end table
+@section Generic options
+
+These options are shared amongst the av* tools.
+
+@table @option
+
+@item -L
+Show license.
+
+@item -h, -?, -help, --help
+Show help.
+
+@item -version
+Show version.
+
+@item -formats
+Show available formats.
+
+The fields preceding the format names have the following meanings:
+@table @samp
+@item D
+Decoding available
+@item E
+Encoding available
+@end table
+
+@item -codecs
+Show available codecs.
+
+The fields preceding the codec names have the following meanings:
+@table @samp
+@item D
+Decoding available
+@item E
+Encoding available
+@item V/A/S
+Video/audio/subtitle codec
+@item S
+Codec supports slices
+@item D
+Codec supports direct rendering
+@item T
+Codec can handle input truncated at random locations instead of only at frame boundaries
+@end table
+
+@item -bsfs
+Show available bitstream filters.
+
+@item -protocols
+Show available protocols.
+
+@item -filters
+Show available libavfilter filters.
+
+@item -pix_fmts
+Show available pixel formats.
+
+@item -sample_fmts
+Show available sample formats.
+
+@item -loglevel @var{loglevel} | -v @var{loglevel}
+Set the logging level used by the library.
+@var{loglevel} is a number or a string containing one of the following values:
+@table @samp
+@item quiet
+@item panic
+@item fatal
+@item error
+@item warning
+@item info
+@item verbose
+@item debug
+@end table
+
+By default the program logs to stderr, if coloring is supported by the
+terminal, colors are used to mark errors and warnings. Log coloring
+can be disabled setting the environment variable
+@env{AV_LOG_FORCE_NOCOLOR} or @env{NO_COLOR}, or can be forced setting
+the environment variable @env{AV_LOG_FORCE_COLOR}.
+The use of the environment variable @env{NO_COLOR} is deprecated and
+will be dropped in a following FFmpeg version.
+
+@item -report
+Dump full command line and console output to a file named
+@code{@var{program}-@var{YYYYMMDD}-@var{HHMMSS}.log} in the current
+directory.
+This file can be useful for bug reports.
+It also implies @code{-loglevel verbose}.
+
+Note: setting the environment variable @code{FFREPORT} to any value has the
+same effect.
+
+@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.
+@example
+ffmpeg -cpuflags -sse+mmx ...
+ffmpeg -cpuflags mmx ...
+ffmpeg -cpuflags 0 ...
+@end example
+
+@end table
+
+@section AVOptions
+
+These options are provided directly by the libavformat, libavdevice and
+libavcodec libraries. To see the list of available AVOptions, use the
+@option{-help} option. They are separated into two categories:
+@table @option
+@item generic
+These options can be set for any container, codec or device. Generic options
+are listed under AVFormatContext options for containers/devices and under
+AVCodecContext options for codecs.
+@item private
+These options are specific to the given container, device or codec. Private
+options are listed under their corresponding containers/devices/codecs.
+@end table
+
+For example to write an ID3v2.3 header instead of a default ID3v2.4 to
+an MP3 file, use the @option{id3v2_version} private option of the MP3
+muxer:
+@example
+ffmpeg -i input.flac -id3v2_version 3 out.mp3
+@end example
+
+All codec AVOptions are obviously per-stream, so the chapter on stream
+specifiers applies to them
+
+Note @option{-nooption} syntax cannot be used for boolean AVOptions,
+use @option{-option 0}/@option{-option 1}.
+
+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.
+
+@include avoptions_codec.texi
+@include avoptions_format.texi
--- 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
@@ -17,46 +17,9 @@ Below is a description of the currently available bitstream filters.

@section aac_adtstoasc

-Convert MPEG-2/4 AAC ADTS to MPEG-4 Audio Specific Configuration
-bitstream filter.
-
-This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4
-ADTS header and removes the ADTS header.
-
-This is required for example when copying an AAC stream from a raw
-ADTS AAC container to a FLV or a MOV/MP4 file.
-
@section chomp

-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

@@ -117,10 +80,12 @@ ffmpeg -i frame_%d.jpg -c:v copy rotated.avi

@section movsub

+@section mp3_header_compress
+
@section mp3_header_decompress

@section noise

-@section remove_extra
+@section remove_extradata

@c man end BITSTREAM FILTERS
--- a/doc/codecs.texi
+++ b/doc/codecs.texi
--- 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
@@ -52,37 +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).
-
-@subsubsection 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 ffwavesynth

 Internal wave synthetizer.
@@ -91,143 +60,4 @@ This decoder generates wave patterns according to predefined sequences. Its
 use is purely internal and the format of the data it accepts is not publicly
 documented.

-@section libcelt
-
-libcelt decoder wrapper.
-
-libcelt allows libavcodec to decode the Xiph CELT ultra-low delay audio codec.
-Requires the presence of the libcelt headers and library during configuration.
-You need to explicitly configure the build with @code{--enable-libcelt}.
-
-@section libgsm
-
-libgsm decoder wrapper.
-
-libgsm allows libavcodec to decode the GSM full rate audio codec. Requires
-the presence of the libgsm headers and library during configuration. You need
-to explicitly configure the build with @code{--enable-libgsm}.
-
-This decoder supports both the ordinary GSM and the Microsoft variant.
-
-@section libilbc
-
-libilbc decoder wrapper.
-
-libilbc allows libavcodec to decode the Internet Low Bitrate Codec (iLBC)
-audio codec. Requires the presence of the libilbc headers and library during
-configuration. You need to explicitly configure the build with
-@code{--enable-libilbc}.
-
-@subsection Options
-
-The following option is supported by the libilbc wrapper.
-
-@table @option
-@item enhance
-
-Enable the enhancement of the decoded audio when set to 1. The default
-value is 0 (disabled).
-
-@end table
-
-@section libopencore-amrnb
-
-libopencore-amrnb decoder wrapper.
-
-libopencore-amrnb allows libavcodec to decode the Adaptive Multi-Rate
-Narrowband audio codec. Using it requires the presence of the
-libopencore-amrnb headers and library during configuration. You need to
-explicitly configure the build with @code{--enable-libopencore-amrnb}.
-
-An FFmpeg native decoder for AMR-NB exists, so users can decode AMR-NB
-without this library.
-
-@section libopencore-amrwb
-
-libopencore-amrwb decoder wrapper.
-
-libopencore-amrwb allows libavcodec to decode the Adaptive Multi-Rate
-Wideband audio codec. Using it requires the presence of the
-libopencore-amrwb headers and library during configuration. You need to
-explicitly configure the build with @code{--enable-libopencore-amrwb}.
-
-An FFmpeg native decoder for AMR-WB exists, so users can decode AMR-WB
-without this library.
-
-@section libopus
-
-libopus decoder wrapper.
-
-libopus allows libavcodec to decode the Opus Interactive Audio Codec.
-Requires the presence of the libopus headers and library during
-configuration. You need to explicitly configure the build with
-@code{--enable-libopus}.
-
@c man end AUDIO DECODERS
-
-@chapter Subtitles Decoders
-@c man begin SUBTILES DECODERS
-
-@section dvdsub
-
-This codec decodes the bitmap subtitles used in DVDs; the same subtitles can
-also be found in VobSub file pairs and in some Matroska files.
-
-@subsection Options
-
-@table @option
-@item palette
-Specify the global palette used by the bitmaps. When stored in VobSub, the
-palette is normally specified in the index file; in Matroska, the palette is
-stored in the codec extra-data in the same format as in VobSub. In DVDs, the
-palette is stored in the IFO file, and therefore not available when reading
-from dumped VOB files.
-
-The format for this option is a string containing 16 24-bits hexadecimal
-numbers (without 0x prefix) separated by comas, for example @code{0d00ee,
-ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1,
-7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b}.
-@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 (black) background.
-@end table
-
-@c man end SUBTILES DECODERS
--- a/doc/default.css
+++ b/doc/default.css
@@ -1,165 +0,0 @@
-a.summary-letter {
-    text-decoration: none;
-}
-
-a {
-    color: #2D6198;
-}
-
-a:visited {
-    color: #884488;
-}
-
-#banner {
-    background-color: white;
-    position: relative;
-    text-align: center;
-}
-
-#banner img {
-    margin-bottom: 1px;
-    margin-top: 5px;
-}
-
-#body {
-    margin-left: 1em;
-    margin-right: 1em;
-}
-
-body {
-    background-color: #313131;
-    margin: 0;
-    text-align: justify;
-}
-
-.center {
-    margin-left: auto;
-    margin-right: auto;
-    text-align: center;
-}
-
-#container {
-    background-color: white;
-    color: #202020;
-    margin-left: 1em;
-    margin-right: 1em;
-}
-
-#footer {
-    text-align: center;
-}
-
-h1 a, h2 a, h3 a, h4 a {
-    text-decoration: inherit;
-    color: inherit;
-}
-
-h1, h2, h3, h4 {
-    padding-left: 0.4em;
-    border-radius: 4px;
-    padding-bottom: 0.25em;
-    padding-top: 0.25em;
-    border: 1px solid #6A996A;
-}
-
-h1 {
-    background-color: #7BB37B;
-    color: #151515;
-    font-size: 1.2em;
-    padding-bottom: 0.3em;
-    padding-top: 0.3em;
-}
-
-h2 {
-    color: #313131;
-    font-size: 1.0em;
-    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;
-}
-
-img {
-    border: 0;
-}
-
-#navbar {
-    background-color: #738073;
-    border-bottom: 1px solid #5C665C;
-    border-top: 1px solid #5C665C;
-    margin-top: 12px;
-    padding: 0.3em;
-    position: relative;
-    text-align: center;
-}
-
-#navbar a, #navbar_secondary a {
-    color: white;
-    padding: 0.3em;
-    text-decoration: none;
-}
-
-#navbar a:hover, #navbar_secondary a:hover {
-    background-color: #313131;
-    color: white;
-    text-decoration: none;
-}
-
-#navbar_secondary {
-    background-color: #738073;
-    border-bottom: 1px solid #5C665C;
-    border-left: 1px solid #5C665C;
-    border-right: 1px solid #5C665C;
-    padding: 0.3em;
-    position: relative;
-    text-align: center;
-}
-
-p {
-    margin-left: 1em;
-    margin-right: 1em;
-}
-
-pre {
-    margin-left: 3em;
-    margin-right: 3em;
-    padding: 0.3em;
-    border: 1px solid #bbb;
-    background-color: #f7f7f7;
-}
-
-dl dt {
-    font-weight: bold;
-}
-
-#proj_desc {
-    font-size: 1.2em;
-}
-
-#repos {
-    margin-left: 1em;
-    margin-right: 1em;
-    border-collapse: collapse;
-    border: solid 1px #6A996A;
-}
-
-#repos th {
-    background-color: #7BB37B;
-    border: solid 1px #6A996A;
-}
-
-#repos td {
-    padding: 0.2em;
-    border: solid 1px #6A996A;
-}
--- a/doc/demuxers.texi
+++ b/doc/demuxers.texi
@@ -1,23 +1,69 @@
@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
 are enabled by default. You can list all available ones using the
-configure option @code{--list-demuxers}.
+configure option "--list-demuxers".

 You can disable all the demuxers using the configure option
-@code{--disable-demuxers}, and selectively enable a single demuxer with
-the option @code{--enable-demuxer=@var{DEMUXER}}, or disable it
-with the option @code{--disable-demuxer=@var{DEMUXER}}.
+"--disable-demuxers", and selectively enable a single demuxer with
+the option "--enable-demuxer=@var{DEMUXER}", or disable it
+with the option "--disable-demuxer=@var{DEMUXER}".

-The option @code{-formats} of the ff* tools will display the list of
+The option "-formats" of the ff* tools will display the list of
 enabled demuxers.

 The description of some of the currently available demuxers follows.

+@section image2
+
+Image file demuxer.
+
+This demuxer reads from a list of image files specified by a pattern.
+
+The pattern may contain the string "%d" or "%0@var{N}d", which
+specifies the position of the characters representing a sequential
+number in each filename matched by the pattern. If the form
+"%d0@var{N}d" is used, the string representing the number in each
+filename is 0-padded and @var{N} is the total number of 0-padded
+digits representing the number. The literal character '%' can be
+specified in the pattern with the string "%%".
+
+If the pattern contains "%d" or "%0@var{N}d", the first filename of
+the file list specified by the pattern must contain a number
+inclusively contained between 0 and 4, all the following numbers must
+be sequential. This limitation may be hopefully fixed.
+
+The pattern may contain a suffix which is used to automatically
+determine the format of the images contained in the files.
+
+For example the pattern "img-%03d.bmp" will match a sequence of
+filenames of the form @file{img-001.bmp}, @file{img-002.bmp}, ...,
+@file{img-010.bmp}, etc.; the pattern "i%%m%%g-%d.jpg" will match a
+sequence of filenames of the form @file{i%m%g-1.jpg},
+@file{i%m%g-2.jpg}, ..., @file{i%m%g-10.jpg}, etc.
+
+The size, the pixel format, and the format of each image must be the
+same for all the files in the sequence.
+
+The following example shows how to 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 -i 'img-%03d.jpeg' -r 10 out.mkv
+@end example
+
+Note that the pattern must not necessarily contain "%d" or
+"%0@var{N}d", for example to convert a single image file
+@file{img.jpeg} you can employ the command:
+@example
+ffmpeg -i img.jpeg img.png
+@end example
+
@section applehttp

 Apple HTTP Live Streaming demuxer.
@@ -29,298 +75,6 @@ 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 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
-
-Virtual concatenation script demuxer.
-
-This demuxer reads a list of files and other directives from a text file and
-demuxes them one after the other, as if all their packet had been muxed
-together.
-
-The timestamps in the files are adjusted so that the first file starts at 0
-and each next file starts where the previous one finishes. Note that it is
-done globally and may cause gaps if all streams do not have exactly the same
-length.
-
-All files must have the same streams (same codecs, same time base, etc.).
-
-The duration of each file is used to adjust the timestamps of the next file:
-if the duration is incorrect (because it was computed using the bit-rate or
-because the file is truncated, for example), it can cause artifacts. The
-@code{duration} directive can be used to override the duration stored in
-each file.
-
-@subsection Syntax
-
-The script is a text file in extended-ASCII, with one directive per line.
-Empty lines, leading spaces and lines starting with '#' are ignored. The
-following directive is recognized:
-
-@table @option
-
-@item @code{file @var{path}}
-Path to a file to read; special characters and spaces must be escaped with
-backslash or single quotes.
-
-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
-to 1 if it was to its default -1.
-
-To make FFmpeg recognize the format automatically, this directive must
-appears exactly as is (no extra space or byte-order-mark) on the very first
-line of the script.
-
-@item @code{duration @var{dur}}
-Duration of the file. This information can be specified from the file;
-specifying it here may be more efficient or help if the information from the
-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.
-
-@end table
-
-@subsection Options
-
-This demuxer accepts the following option:
-
-@table @option
-
-@item safe
-If set to 1, reject unsafe file paths. A file path is considered safe if it
-does not contain a protocol specification and is relative and all components
-only contain characters from the portable character set (letters, digits,
-period, underscore and hyphen) and have no period at the beginning of a
-component.
-
-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.
-
-@end table
-
-@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
-
-The Game Music Emu library is a collection of video game music file emulators.
-
-See @url{http://code.google.com/p/game-music-emu/} for more information.
-
-Some files have multiple tracks. The demuxer will pick the first track by
-default. The @option{track_index} option can be used to select a different
-track. Track indexes start at 0. The demuxer exports the number of tracks as
-@var{tracks} meta data entry.
-
-For very large files, the @option{max_size} option may have to be adjusted.
-
-@section libquvi
-
-Play media from Internet services using the quvi project.
-
-The demuxer accepts a @option{format} option to request a specific quality. It
-is by default set to @var{best}.
-
-See @url{http://quvi.sourceforge.net/} for more information.
-
-FFmpeg needs to be built with @code{--enable-libquvi} for this demuxer to be
-enabled.
-
-@section image2
-
-Image file demuxer.
-
-This demuxer reads from a list of image files specified by a pattern.
-The syntax and meaning of the pattern is specified by the
-option @var{pattern_type}.
-
-The pattern may contain a suffix which is used to automatically
-determine the format of the images contained in the files.
-
-The size, the pixel format, and the format of each image must be the
-same for all the files in the sequence.
-
-This demuxer accepts the following options:
-@table @option
-@item framerate
-Set the frame rate for the video stream. It defaults to 25.
-@item loop
-If set to 1, loop over the input. Default value is 0.
-@item pattern_type
-Select the pattern type used to interpret the provided filename.
-
-@var{pattern_type} accepts one of the following values.
-@table @option
-@item sequence
-Select a sequence pattern type, used to specify a sequence of files
-indexed by sequential numbers.
-
-A sequence pattern may contain the string "%d" or "%0@var{N}d", which
-specifies the position of the characters representing a sequential
-number in each filename matched by the pattern. If the form
-"%d0@var{N}d" is used, the string representing the number in each
-filename is 0-padded and @var{N} is the total number of 0-padded
-digits representing the number. The literal character '%' can be
-specified in the pattern with the string "%%".
-
-If the sequence pattern contains "%d" or "%0@var{N}d", the first filename of
-the file list specified by the pattern must contain a number
-inclusively contained between @var{start_number} and
-@var{start_number}+@var{start_number_range}-1, and all the following
-numbers must be sequential.
-
-For example the pattern "img-%03d.bmp" will match a sequence of
-filenames of the form @file{img-001.bmp}, @file{img-002.bmp}, ...,
-@file{img-010.bmp}, etc.; the pattern "i%%m%%g-%d.jpg" will match a
-sequence of filenames of the form @file{i%m%g-1.jpg},
-@file{i%m%g-2.jpg}, ..., @file{i%m%g-10.jpg}, etc.
-
-Note that the pattern must not necessarily contain "%d" or
-"%0@var{N}d", for example to convert a single image file
-@file{img.jpeg} you can employ the command:
-@example
-ffmpeg -i img.jpeg img.png
-@end example
-
-@item glob
-Select a glob wildcard pattern type.
-
-The pattern is interpreted like a @code{glob()} pattern. This is only
-selectable if libavformat was compiled with globbing support.
-
-@item glob_sequence @emph{(deprecated, will be removed)}
-Select a mixed glob wildcard/sequence pattern.
-
-If your version of libavformat was compiled with globbing support, and
-the provided pattern contains at least one glob meta character among
-@code{%*?[]@{@}} that is preceded by an unescaped "%", the pattern is
-interpreted like a @code{glob()} pattern, otherwise it is interpreted
-like a sequence pattern.
-
-All glob special characters @code{%*?[]@{@}} must be prefixed
-with "%". To escape a literal "%" you shall use "%%".
-
-For example the pattern @code{foo-%*.jpeg} will match all the
-filenames prefixed by "foo-" and terminating with ".jpeg", and
-@code{foo-%?%?%?.jpeg} will match all the filenames prefixed with
-"foo-", followed by a sequence of three characters, and terminating
-with ".jpeg".
-
-This pattern type is deprecated in favor of @var{glob} and
-@var{sequence}.
-@end table
-
-Default value is @var{glob_sequence}.
-@item pixel_format
-Set the pixel format of the images to read. If not specified the pixel
-format is guessed from the first image file in the sequence.
-@item start_number
-Set the index of the file matched by the image file pattern to start
-to read from. Default value is 0.
-@item start_number_range
-Set the index interval range to check when looking for the first image
-file in the sequence, starting from @var{start_number}. Default value
-is 5.
-@item ts_from_file
-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.
-@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.
-@end table
-
-@subsection Examples
-
-@itemize
-@item
-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
-@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
-@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
-@end example
-@end itemize
-
-@section mpegts
-
-MPEG-2 transport stream demuxer.
-
-@table @option
-
-@item fix_teletext_pts
-Overrides teletext packet PTS and DTS values with the timestamps calculated
-from the PCR of the first program which the teletext stream is part of and is
-not discarded. Default value is 1, set this option to 0 if you want your
-teletext packet PTS and DTS values untouched.
-@end table
-
-@section rawvideo
-
-Raw video demuxer.
-
-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.
-
-This demuxer accepts the following options:
-@table @option
-
-@item framerate
-Set input video frame rate. Default value is 25.
-
-@item pixel_format
-Set the input video pixel format. Default value is @code{yuv420p}.
-
-@item video_size
-Set the input video size. This value must be specified explicitly.
-@end table
-
-For example to read a rawvideo file @file{input.raw} with
-@command{ffplay}, assuming a pixel format of @code{rgb24}, a video
-size of @code{320x240}, and a frame rate of 10 images per second, use
-the command:
-@example
-ffplay -f rawvideo -pixel_format rgb24 -video_size 320x240 -framerate 10 input.raw
-@end example
-
@section sbg

 SBaGen script demuxer.
@@ -351,25 +105,4 @@ the script is directly played, the actual times will match the absolute
 timestamps up to the sound controller's clock accuracy, but if the user
 somehow pauses the playback or seeks, all times will be shifted accordingly.

-@section tedcaptions
-
-JSON captions used for @url{http://www.ted.com/, TED Talks}.
-
-TED does not provide links to the captions, but they can be guessed from the
-page. The file @file{tools/bookmarklets.html} from the FFmpeg source tree
-contains a bookmarklet to expose them.
-
-This demuxer accepts the following option:
-@table @option
-@item start_time
-Set the start time of the TED talk, in milliseconds. The default is 15000
-(15s). It is used to sync the captions with the downloadable videos, because
-they include a 15s intro.
-@end table
-
-Example: convert the captions to a format most players understand:
-@example
-ffmpeg -i http://www.ted.com/talks/subtitles/id/1/lang/en talk1-en.srt
-@end example
-
-@c man end DEMUXERS
+@c man end INPUT DEVICES
--- a/doc/developer.texi
+++ b/doc/developer.texi
@@ -11,23 +11,29 @@

@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{http://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

@@ -51,16 +57,13 @@ 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.
@@ -92,7 +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 */
@@ -114,17 +117,13 @@ 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 @};})
-
@item
 compound literals (@samp{x = (struct s) @{ 17, 23 @};})
@end itemize
@@ -136,72 +135,46 @@ 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

@subsection Naming conventions
-All names should be composed with underscores (_), not CamelCase. For example,
-@samp{avfilter_get_video_buffer} is an acceptable function name and
-@samp{AVFilterGetVideo} is not. The exception from this are type names, like
+All names are using underscores (_), not CamelCase. For example, @samp{avfilter_get_video_buffer} is
+a valid 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 the CamelCase

-There are the following conventions for naming variables and functions:

+There are 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.
-
+For variables and functions declared as @code{static} no prefixes are 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}.
-
+For variables and functions used internally by the library, @code{ff_} prefix
+should be used.
+For example, @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}.
-
+For variables and functions used internally across multiple libraries, use
+@code{avpriv_}. 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,
-@code{avcodec_} for libavcodec, @code{swr_} for libswresample, etc).
-Check the existing code and choose names accordingly.
-Note that some symbols without these prefixes are also exported for
-retro-compatibility reasons. These exceptions are declared in the
-@code{lib<name>/lib<name>.v} files.
+For exported names, each library has its own prefixes. Just check the existing
+code and name accordingly.
@end itemize

-Furthermore, name space reserved for the system should not be invaded.
-Identifiers ending in @code{_t} are reserved by
-@url{http://pubs.opengroup.org/onlinepubs/007904975/functions/xsh_chap02_02.html#tag_02_02_02, POSIX}.
-Also avoid names starting with @code{__} or @code{_} followed by an uppercase
-letter as they are reserved by the C standard. Names starting with @code{_}
-are reserved at the file level and may not be used for externally visible
-symbols. If in doubt, just avoid names starting with @code{_} altogether.
-
-@subsection Miscellaneous conventions
-
+@subsection Miscellanous 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.
@@ -217,8 +190,8 @@ set shiftwidth=4
 set softtabstop=4
 set cindent
 set cinoptions=(0
-" Allow tabs in Makefiles.
-autocmd FileType make,automake set noexpandtab shiftwidth=8 softtabstop=8
+" allow tabs in Makefiles
+autocmd FileType make set noexpandtab shiftwidth=8 softtabstop=8
 " Trailing whitespace and tabs are forbidden, so highlight them.
 highlight ForbiddenWhitespace ctermbg=red guibg=red
 match ForbiddenWhitespace /\s\+$\|\t/
@@ -231,8 +204,8 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
 (c-add-style "ffmpeg"
             '("k&r"
               (c-basic-offset . 4)
-               (indent-tabs-mode . nil)
-               (show-trailing-whitespace . t)
+               (indent-tabs-mode nil)
+               (show-trailing-whitespace t)
               (c-offsets-alist
                (statement-cont . (c-lineup-assignments +)))
               )
@@ -244,158 +217,124 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:

@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 LGPL 2.1, including an
+   "or any later version" clause, or the MIT license.  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.
-
+   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
-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.
-
+   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 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 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 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 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
-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.
+   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
-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
+   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
-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.
-
+   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
-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.
-
+   When applying patches that have been discussed (at length) on the mailing
+   list, reference the thread in the log message.
@item
-When applying patches that have been discussed (at length) on the mailing
-list, reference the thread in the log message.
-
+    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
-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!
-
+    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
-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.
-
+    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
-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.
-
+    Try to keep important discussions and requests (also) on the public
+    developer mailing list, so that all developers can benefit from them.
@item
-Try to keep important discussions and requests (also) on the public
-developer mailing list, so that all developers can benefit from them.
-
+    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
-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.
-
+    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).
@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.
-
+    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
-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
-finding a new maintainer and also don't forget updating 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.

+Note, these rules are mostly borrowed from the MPlayer project.
+
@anchor{Submitting patches}
@section Submitting patches

@@ -418,6 +357,11 @@ The tool is located in the tools directory.
 Run the @ref{Regression tests} before submitting a patch in order to verify
 it does not cause unexpected problems.

+Patches should be posted as base64 encoded attachments (or any other
+encoding which ensures that the patch will not be trashed during
+transmission) to the ffmpeg-devel mailing list, see
+@url{http://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel}
+
 It also helps quite a bit if you tell us what the patch does (for example
 'replaces lrint by lrintf'), and why (for example '*BSD isn't C99 compliant
 and has no lrint()')
@@ -425,13 +369,6 @@ and has no lrint()')
 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{http://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel, ffmpeg-devel}
-mailing list. Use @code{git send-email} when possible since it will properly
-send patches without requiring extra care. If you cannot, then send patches
-as base64-encoded attachments, so your patch is not trashed during
-transmission.
-
 Your patch will be reviewed on the mailing list. You will likely be asked
 to make some changes and are expected to send in an improved version that
 incorporates the requests from the review. This process may go through
@@ -446,51 +383,38 @@ 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 CodecID to @file{avcodec.h}?
@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


@@ -498,109 +422,73 @@ 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://kerneltrap.org/files/Jeremy/DCO.txt} 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 and the noise bitstream filter. Your decoder or demuxer
+    should not crash or end in a (near) infinite loop when fed damaged data.
@item
-Does the patch not mix functional and cosmetic changes?
-
+    Does the patch not mix functional and cosmetic changes?
@item
-Did you add tabs or trailing whitespace to the code? Both are forbidden.
-
+    Did you add tabs or trailing whitespace to the code? Both are forbidden.
@item
-Is the patch attached to the email you send?
-
+    Is the patch attached to the email you send?
@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.
-
+    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
-If the patch fixes a bug, did you provide a verbose analysis of the bug?
-
+    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 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
-
+    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
-Did you provide a verbose summary about what the patch does change?
-
+    Did you provide a verbose summary about what the patch does change?
@item
-Did you provide a verbose explanation why it changes things like it does?
-
+    Did you provide a verbose explanation why it changes things like it does?
@item
-Did you provide a verbose summary of the user visible advantages and
-disadvantages if the patch is applied?
-
+    Did you provide a verbose summary of the user visible advantages and
+    disadvantages if the patch is applied?
@item
-Did you provide an example so we can verify the new feature added by the
-patch easily?
-
+    Did you provide an example so we can verify the new feature added by the
+    patch easily?
@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.
-
+    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
-You should maintain alphabetical order in alphabetically ordered lists as
-long as doing so does not break API/ABI compatibility.
-
+    You should maintain alphabetical order in alphabetically ordered lists as
+    long as doing so does not break API/ABI compatibility.
@item
-Lines with similar content should be aligned vertically when doing so
-improves readability.
-
+    Lines with similar content should be aligned vertically when doing so
+    improves readability.
@item
-Consider to add a regression test for your code.
-
+    Consider to add a regression test for your code.
@item
-If you added YASM code please check that things still work with --disable-yasm
-
-@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.
+    If you added YASM code please check that things still work with --disable-yasm
@end enumerate

@section Patch review process
@@ -641,157 +529,4 @@ Running 'make fate' accomplishes this, please see @url{fate.html} for details.
 this case, the reference results of the regression tests shall be modified
 accordingly].

-@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 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 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
-manner with the coverage tools @code{gcov}/@code{lcov}.  This involves
-the following steps:
-
-@enumerate
-@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
-
-You can use the command @code{make lcov-reset} to reset the coverage
-measurements. You will need to rerun @code{make lcov} after running a
-new test.
-
-@subsection Using Valgrind
-
-The configure script provides a shortcut for using valgrind to spot bugs
-related to memory handling. Just add the option
-@code{--toolchain=valgrind-memcheck} or @code{--toolchain=valgrind-massif}
-to your configure line, and reasonable defaults will be set for running
-FATE under the supervision of either the @strong{memcheck} or the
-@strong{massif} tool of the valgrind suite.
-
-In case you need finer control over how valgrind is invoked, use the
-@code{--target-exec='valgrind <your_custom_valgrind_options>} option in
-your configure line instead.
-
-@anchor{Release process}
-@section Release process
-
-FFmpeg maintains a set of @strong{release branches}, which are the
-recommended deliverable for system integrators and distributors (such as
-Linux distributions, etc.). At regular times, a @strong{release
-manager} prepares, tests and publishes tarballs on the
-@url{http://ffmpeg.org} website.
-
-There are two kinds of releases:
-
-@enumerate
-@item
-@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.
-@end enumerate
-
-Note that we promise to our users that shared libraries from any FFmpeg
-release never break programs that have been @strong{compiled} against
-previous versions of @strong{the same release series} in any case!
-
-However, from time to time, we do make API changes that require adaptations
-in applications. Such changes are only allowed in (new) major releases and
-require further steps such as bumping library version numbers and/or
-adjustments to the symbol versioning file. Please discuss such changes
-on the @strong{ffmpeg-devel} mailing list in time to allow forward planning.
-
-@anchor{Criteria for Point Releases}
-@subsection Criteria for Point Releases
-
-Changes that match the following criteria are valid candidates for
-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/}.
-
-@item
-Fixes a documented bug in @url{https://trac.ffmpeg.org}.
-
-@item
-Improves the included documentation.
-
-@item
-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.
-
-
-@subsection Release Checklist
-
-The release process involves the following steps:
-
-@enumerate
-@item
-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}.
-
-@item
-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}.
-
-@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}).
-
-@item
-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{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.
-
-@item
-Publish the news entry.
-
-@item
-Send announcement to the mailing list.
-@end enumerate
-
@bye
--- a/doc/devices.texi
+++ b/doc/devices.texi
@@ -1,25 +0,0 @@
-@chapter Device Options
-@c man begin DEVICE OPTIONS
-
-The libavdevice library provides the same interface as
-libavformat. Namely, an input device is considered like a demuxer, and
-an output device like a muxer, and the interface and generic device
-options are the same provided by libavformat (see the ffmpeg-formats
-manual).
-
-In addition each input or output device may support so-called private
-options, which are specific for that component.
-
-Options may be set by specifying -@var{option} @var{value} in the
-FFmpeg tools, or by setting the value explicitly in the device
-@code{AVFormatContext} options or using the @file{libavutil/opt.h} API
-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
@@ -1,12 +0,0 @@
-#!/bin/sh
-
-SRC_PATH="${1}"
-DOXYFILE="${2}"
-
-shift 2
-
-doxygen - <<EOF
-@INCLUDE        = ${DOXYFILE}
-INPUT           = $@
-EXAMPLE_PATH    = ${SRC_PATH}/doc/examples
-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 +1,10 @@
+ </div>
+
+ <div id="footer">
+   Generated on $datetime for $projectname by&#160;<a href="http://www.doxygen.org/index.html">doxygen</a> $doxygenversion
+ </div>
+
+</div>
+
+</body>
+</html>
--- a/doc/doxy/header.html
+++ b/doc/doxy/header.html
@@ -0,0 +1,14 @@
+<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN" "http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd">
+<html xmlns="http://www.w3.org/1999/xhtml">
+<head>
+<meta http-equiv="Content-Type" content="text/xhtml;charset=UTF-8"/>
+<meta http-equiv="X-UA-Compatible" content="IE=9"/>
+<!--BEGIN PROJECT_NAME--><title>$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" />
+</head>
+
+<div id="container">
+
+<div id="body">
+ <div>
--- a/doc/encoders.texi
+++ b/doc/encoders.texi
--- a/doc/eval.texi
+++ b/doc/eval.texi
@@ -0,0 +1,204 @@
+@chapter Expression Evaluation
+@c man begin EXPRESSION EVALUATION
+
+When evaluating an arithmetic expression, FFmpeg uses an internal
+formula evaluator, implemented through the @file{libavutil/eval.h}
+interface.
+
+An expression may contain unary, binary operators, constants, and
+functions.
+
+Two expressions @var{expr1} and @var{expr2} can be combined to form
+another expression "@var{expr1};@var{expr2}".
+@var{expr1} and @var{expr2} are evaluated in turn, and the new
+expression evaluates to the value of @var{expr2}.
+
+The following binary operators are available: @code{+}, @code{-},
+@code{*}, @code{/}, @code{^}.
+
+The following unary operators are available: @code{+}, @code{-}.
+
+The following functions are available:
+@table @option
+@item sinh(x)
+@item cosh(x)
+@item tanh(x)
+@item sin(x)
+@item cos(x)
+@item tan(x)
+@item atan(x)
+@item asin(x)
+@item acos(x)
+@item exp(x)
+@item log(x)
+@item abs(x)
+@item squish(x)
+@item gauss(x)
+@item isnan(x)
+Return 1.0 if @var{x} is NAN, 0.0 otherwise.
+
+@item mod(x, y)
+@item max(x, y)
+@item min(x, y)
+@item eq(x, y)
+@item gte(x, y)
+@item gt(x, y)
+@item lte(x, y)
+@item lt(x, y)
+@item st(var, expr)
+Allow to store the value of the expression @var{expr} in an internal
+variable. @var{var} specifies the number of the variable where to
+store the value, and it is a value ranging from 0 to 9. The function
+returns the value stored in the internal variable.
+Note, Variables are currently not shared between expressions.
+
+@item ld(var)
+Allow to load the value of the internal variable with number
+@var{var}, which was previously stored with st(@var{var}, @var{expr}).
+The function returns the loaded value.
+
+@item while(cond, expr)
+Evaluate expression @var{expr} while the expression @var{cond} is
+non-zero, and returns the value of the last @var{expr} evaluation, or
+NAN if @var{cond} was always false.
+
+@item ceil(expr)
+Round the value of expression @var{expr} upwards to the nearest
+integer. For example, "ceil(1.5)" is "2.0".
+
+@item floor(expr)
+Round the value of expression @var{expr} downwards to the nearest
+integer. For example, "floor(-1.5)" is "-2.0".
+
+@item trunc(expr)
+Round the value of expression @var{expr} towards zero to the nearest
+integer. For example, "trunc(-1.5)" is "-1.0".
+
+@item sqrt(expr)
+Compute the square root of @var{expr}. This is equivalent to
+"(@var{expr})^.5".
+
+@item not(expr)
+Return 1.0 if @var{expr} is zero, 0.0 otherwise.
+
+@item pow(x, y)
+Compute the power of @var{x} elevated @var{y}, it is equivalent to
+"(@var{x})^(@var{y})".
+
+@item random(x)
+Return a pseudo random value between 0.0 and 1.0. @var{x} is the index of the
+internal variable which will be used to save the seed/state.
+
+@item hypot(x, y)
+This function is similar to the C function with the same name; it returns
+"sqrt(@var{x}*@var{x} + @var{y}*@var{y})", the length of the hypotenuse of a
+right triangle with sides of length @var{x} and @var{y}, or the distance of the
+point (@var{x}, @var{y}) from the origin.
+
+@item gcd(x, y)
+Return the greatest common divisor of @var{x} and @var{y}. If both @var{x} and
+@var{y} are 0 or either or both are less than zero then behavior is undefined.
+
+@item if(x, y)
+Evaluate @var{x}, and if the result is non-zero return the result of
+the evaluation of @var{y}, return 0 otherwise.
+
+@item ifnot(x, y)
+Evaluate @var{x}, and if the result is zero return the result of the
+evaluation of @var{y}, return 0 otherwise.
+
+@item taylor(expr, x) taylor(expr, x, id)
+Evaluate a taylor series at x.
+expr represents the LD(id)-th derivates of f(x) at 0. If id is not specified
+then 0 is assumed.
+note, when you have the derivatives at y instead of 0
+taylor(expr, x-y) can be used
+When the series does not converge the results are undefined.
+
+@item root(expr, max)
+Finds x where f(x)=0 in the interval 0..max.
+f() must be continuous or the result is undefined.
+@end table
+
+The following constants are available:
+@table @option
+@item PI
+area of the unit disc, approximately 3.14
+@item E
+exp(1) (Euler's number), approximately 2.718
+@item PHI
+golden ratio (1+sqrt(5))/2, approximately 1.618
+@end table
+
+Assuming that an expression is considered "true" if it has a non-zero
+value, note that:
+
+@code{*} works like AND
+
+@code{+} works like OR
+
+and the construct:
+@example
+if A then B else C
+@end example
+is equivalent to
+@example
+if(A,B) + ifnot(A,C)
+@end example
+
+In your C code, you can extend the list of unary and binary functions,
+and define recognized constants, so that they are available for your
+expressions.
+
+The evaluator also recognizes the International System number
+postfixes. If 'i' is appended after the postfix, powers of 2 are used
+instead of powers of 10. The 'B' postfix multiplies the value for 8,
+and can be appended after another postfix or used alone. This allows
+using for example 'KB', 'MiB', 'G' and 'B' as postfix.
+
+Follows the list of available International System postfixes, with
+indication of the corresponding powers of 10 and of 2.
+@table @option
+@item y
+-24 / -80
+@item z
+-21 / -70
+@item a
+-18 / -60
+@item f
+-15 / -50
+@item p
+-12 / -40
+@item n
+-9 / -30
+@item u
+-6 / -20
+@item m
+-3 / -10
+@item c
+-2
+@item d
+-1
+@item h
+2
+@item k
+3 / 10
+@item K
+3 / 10
+@item M
+6 / 20
+@item G
+9 / 30
+@item T
+12 / 40
+@item P
+15 / 40
+@item E
+18 / 50
+@item Z
+21 / 60
+@item Y
+24 / 70
+@end table
+
+@c man end
--- a/doc/examples/Makefile
+++ b/doc/examples/Makefile
@@ -3,39 +3,30 @@ FFMPEG_LIBS=    libavdevice                        \
                libavformat                        \
                libavfilter                        \
                libavcodec                         \
+                libavresample                      \
                libswresample                      \
                libswscale                         \
                libavutil                          \

-CFLAGS += -Wall -g
-CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
-LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
+CFLAGS += -Wall -O2 -g
+CFLAGS += $(shell pkg-config --cflags $(FFMPEG_LIBS))
+LDLIBS += $(shell pkg-config --libs $(FFMPEG_LIBS))

-EXAMPLES=       avio_reading                       \
-                avcodec                            \
-                demuxing_decoding                  \
+EXAMPLES=       decoding_encoding                  \
                filtering_video                    \
                filtering_audio                    \
                metadata                           \
                muxing                             \
-                remuxing                           \
-                resampling_audio                   \
-                scaling_video                      \
-                transcode_aac                      \

 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
+.phony: all clean

 all: $(OBJS) $(EXAMPLES)

-clean-test:
-	$(RM) test*.pgm test.h264 test.mp2 test.sw test.mpg
-
-clean: clean-test
-	$(RM) $(EXAMPLES) $(OBJS)
+clean:
+	rm -rf $(EXAMPLES) $(OBJS)
--- a/doc/examples/README
+++ b/doc/examples/README
@@ -1,23 +0,0 @@
-FFmpeg examples README
----------------------
-
-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
-
-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
-
-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.
--- a/doc/examples/avcodec.c
+++ b/doc/examples/avcodec.c
@@ -1,658 +0,0 @@
-/*
- * Copyright (c) 2001 Fabrice Bellard
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to deal
- * in the Software without restriction, including without limitation the rights
- * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- * copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
- * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
- * THE SOFTWARE.
- */
-
-/**
- * @file
- * libavcodec API use example.
- *
- * @example avcodec.c
- * Note that libavcodec only handles codecs (mpeg, mpeg4, etc...),
- * not file formats (avi, vob, mp4, mov, mkv, mxf, flv, mpegts, mpegps, etc...). See library 'libavformat' for the
- * format handling
- */
-
-#include <math.h>
-
-#include <libavutil/opt.h>
-#include <libavcodec/avcodec.h>
-#include <libavutil/channel_layout.h>
-#include <libavutil/common.h>
-#include <libavutil/imgutils.h>
-#include <libavutil/mathematics.h>
-#include <libavutil/samplefmt.h>
-
-#define INBUF_SIZE 4096
-#define AUDIO_INBUF_SIZE 20480
-#define AUDIO_REFILL_THRESH 4096
-
-/* check that a given sample format is supported by the encoder */
-static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt)
-{
-    const enum AVSampleFormat *p = codec->sample_fmts;
-
-    while (*p != AV_SAMPLE_FMT_NONE) {
-        if (*p == sample_fmt)
-            return 1;
-        p++;
-    }
-    return 0;
-}
-
-/* just pick the highest supported samplerate */
-static int select_sample_rate(AVCodec *codec)
-{
-    const int *p;
-    int best_samplerate = 0;
-
-    if (!codec->supported_samplerates)
-        return 44100;
-
-    p = codec->supported_samplerates;
-    while (*p) {
-        best_samplerate = FFMAX(*p, best_samplerate);
-        p++;
-    }
-    return best_samplerate;
-}
-
-/* select layout with the highest channel count */
-static int select_channel_layout(AVCodec *codec)
-{
-    const uint64_t *p;
-    uint64_t best_ch_layout = 0;
-    int best_nb_channels   = 0;
-
-    if (!codec->channel_layouts)
-        return AV_CH_LAYOUT_STEREO;
-
-    p = codec->channel_layouts;
-    while (*p) {
-        int nb_channels = av_get_channel_layout_nb_channels(*p);
-
-        if (nb_channels > best_nb_channels) {
-            best_ch_layout    = *p;
-            best_nb_channels = nb_channels;
-        }
-        p++;
-    }
-    return best_ch_layout;
-}
-
-/*
- * Audio encoding example
- */
-static void audio_encode_example(const char *filename)
-{
-    AVCodec *codec;
-    AVCodecContext *c= NULL;
-    AVFrame *frame;
-    AVPacket pkt;
-    int i, j, k, ret, got_output;
-    int buffer_size;
-    FILE *f;
-    uint16_t *samples;
-    float t, tincr;
-
-    printf("Encode audio file %s\n", filename);
-
-    /* find the MP2 encoder */
-    codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
-    if (!codec) {
-        fprintf(stderr, "Codec not found\n");
-        exit(1);
-    }
-
-    c = avcodec_alloc_context3(codec);
-    if (!c) {
-        fprintf(stderr, "Could not allocate audio codec context\n");
-        exit(1);
-    }
-
-    /* put sample parameters */
-    c->bit_rate = 64000;
-
-    /* check that the encoder supports s16 pcm input */
-    c->sample_fmt = AV_SAMPLE_FMT_S16;
-    if (!check_sample_fmt(codec, c->sample_fmt)) {
-        fprintf(stderr, "Encoder does not support sample format %s",
-                av_get_sample_fmt_name(c->sample_fmt));
-        exit(1);
-    }
-
-    /* select other audio parameters supported by the encoder */
-    c->sample_rate    = select_sample_rate(codec);
-    c->channel_layout = select_channel_layout(codec);
-    c->channels       = av_get_channel_layout_nb_channels(c->channel_layout);
-
-    /* open it */
-    if (avcodec_open2(c, codec, NULL) < 0) {
-        fprintf(stderr, "Could not open codec\n");
-        exit(1);
-    }
-
-    f = fopen(filename, "wb");
-    if (!f) {
-        fprintf(stderr, "Could not open %s\n", filename);
-        exit(1);
-    }
-
-    /* frame containing input raw audio */
-    frame = av_frame_alloc();
-    if (!frame) {
-        fprintf(stderr, "Could not allocate audio frame\n");
-        exit(1);
-    }
-
-    frame->nb_samples     = c->frame_size;
-    frame->format         = c->sample_fmt;
-    frame->channel_layout = c->channel_layout;
-
-    /* the codec gives us the frame size, in samples,
-     * we calculate the size of the samples buffer in bytes */
-    buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size,
-                                             c->sample_fmt, 0);
-    if (buffer_size < 0) {
-        fprintf(stderr, "Could not get sample buffer size\n");
-        exit(1);
-    }
-    samples = av_malloc(buffer_size);
-    if (!samples) {
-        fprintf(stderr, "Could not allocate %d bytes for samples buffer\n",
-                buffer_size);
-        exit(1);
-    }
-    /* setup the data pointers in the AVFrame */
-    ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
-                                   (const uint8_t*)samples, buffer_size, 0);
-    if (ret < 0) {
-        fprintf(stderr, "Could not setup audio frame\n");
-        exit(1);
-    }
-
-    /* encode a single tone sound */
-    t = 0;
-    tincr = 2 * M_PI * 440.0 / c->sample_rate;
-    for (i = 0; i < 200; i++) {
-        av_init_packet(&pkt);
-        pkt.data = NULL; // packet data will be allocated by the encoder
-        pkt.size = 0;
-
-        for (j = 0; j < c->frame_size; j++) {
-            samples[2*j] = (int)(sin(t) * 10000);
-
-            for (k = 1; k < c->channels; k++)
-                samples[2*j + k] = samples[2*j];
-            t += tincr;
-        }
-        /* encode the samples */
-        ret = avcodec_encode_audio2(c, &pkt, frame, &got_output);
-        if (ret < 0) {
-            fprintf(stderr, "Error encoding audio frame\n");
-            exit(1);
-        }
-        if (got_output) {
-            fwrite(pkt.data, 1, pkt.size, f);
-            av_free_packet(&pkt);
-        }
-    }
-
-    /* get the delayed frames */
-    for (got_output = 1; got_output; i++) {
-        ret = avcodec_encode_audio2(c, &pkt, NULL, &got_output);
-        if (ret < 0) {
-            fprintf(stderr, "Error encoding frame\n");
-            exit(1);
-        }
-
-        if (got_output) {
-            fwrite(pkt.data, 1, pkt.size, f);
-            av_free_packet(&pkt);
-        }
-    }
-    fclose(f);
-
-    av_freep(&samples);
-    av_frame_free(&frame);
-    avcodec_close(c);
-    av_free(c);
-}
-
-/*
- * Audio decoding.
- */
-static void audio_decode_example(const char *outfilename, const char *filename)
-{
-    AVCodec *codec;
-    AVCodecContext *c= NULL;
-    int len;
-    FILE *f, *outfile;
-    uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
-    AVPacket avpkt;
-    AVFrame *decoded_frame = NULL;
-
-    av_init_packet(&avpkt);
-
-    printf("Decode audio file %s to %s\n", filename, outfilename);
-
-    /* find the mpeg audio decoder */
-    codec = avcodec_find_decoder(AV_CODEC_ID_MP2);
-    if (!codec) {
-        fprintf(stderr, "Codec not found\n");
-        exit(1);
-    }
-
-    c = avcodec_alloc_context3(codec);
-    if (!c) {
-        fprintf(stderr, "Could not allocate audio codec context\n");
-        exit(1);
-    }
-
-    /* open it */
-    if (avcodec_open2(c, codec, NULL) < 0) {
-        fprintf(stderr, "Could not open codec\n");
-        exit(1);
-    }
-
-    f = fopen(filename, "rb");
-    if (!f) {
-        fprintf(stderr, "Could not open %s\n", filename);
-        exit(1);
-    }
-    outfile = fopen(outfilename, "wb");
-    if (!outfile) {
-        av_free(c);
-        exit(1);
-    }
-
-    /* decode until eof */
-    avpkt.data = inbuf;
-    avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
-
-    while (avpkt.size > 0) {
-        int got_frame = 0;
-
-        if (!decoded_frame) {
-            if (!(decoded_frame = av_frame_alloc())) {
-                fprintf(stderr, "Could not allocate audio frame\n");
-                exit(1);
-            }
-        }
-
-        len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
-        if (len < 0) {
-            fprintf(stderr, "Error while decoding\n");
-            exit(1);
-        }
-        if (got_frame) {
-            /* if a frame has been decoded, output it */
-            int data_size = av_samples_get_buffer_size(NULL, c->channels,
-                                                       decoded_frame->nb_samples,
-                                                       c->sample_fmt, 1);
-            if (data_size < 0) {
-                /* This should not occur, checking just for paranoia */
-                fprintf(stderr, "Failed to calculate data size\n");
-                exit(1);
-            }
-            fwrite(decoded_frame->data[0], 1, data_size, outfile);
-        }
-        avpkt.size -= len;
-        avpkt.data += len;
-        avpkt.dts =
-        avpkt.pts = AV_NOPTS_VALUE;
-        if (avpkt.size < AUDIO_REFILL_THRESH) {
-            /* Refill the input buffer, to avoid trying to decode
-             * incomplete frames. Instead of this, one could also use
-             * a parser, or use a proper container format through
-             * libavformat. */
-            memmove(inbuf, avpkt.data, avpkt.size);
-            avpkt.data = inbuf;
-            len = fread(avpkt.data + avpkt.size, 1,
-                        AUDIO_INBUF_SIZE - avpkt.size, f);
-            if (len > 0)
-                avpkt.size += len;
-        }
-    }
-
-    fclose(outfile);
-    fclose(f);
-
-    avcodec_close(c);
-    av_free(c);
-    av_frame_free(&decoded_frame);
-}
-
-/*
- * Video encoding example
- */
-static void video_encode_example(const char *filename, int codec_id)
-{
-    AVCodec *codec;
-    AVCodecContext *c= NULL;
-    int i, ret, x, y, got_output;
-    FILE *f;
-    AVFrame *frame;
-    AVPacket pkt;
-    uint8_t endcode[] = { 0, 0, 1, 0xb7 };
-
-    printf("Encode video file %s\n", filename);
-
-    /* find the mpeg1 video encoder */
-    codec = avcodec_find_encoder(codec_id);
-    if (!codec) {
-        fprintf(stderr, "Codec not found\n");
-        exit(1);
-    }
-
-    c = avcodec_alloc_context3(codec);
-    if (!c) {
-        fprintf(stderr, "Could not allocate video codec context\n");
-        exit(1);
-    }
-
-    /* put sample parameters */
-    c->bit_rate = 400000;
-    /* resolution must be a multiple of two */
-    c->width = 352;
-    c->height = 288;
-    /* frames per second */
-    c->time_base = (AVRational){1,25};
-    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)
-        av_opt_set(c->priv_data, "preset", "slow", 0);
-
-    /* open it */
-    if (avcodec_open2(c, codec, NULL) < 0) {
-        fprintf(stderr, "Could not open codec\n");
-        exit(1);
-    }
-
-    f = fopen(filename, "wb");
-    if (!f) {
-        fprintf(stderr, "Could not open %s\n", filename);
-        exit(1);
-    }
-
-    frame = av_frame_alloc();
-    if (!frame) {
-        fprintf(stderr, "Could not allocate video frame\n");
-        exit(1);
-    }
-    frame->format = c->pix_fmt;
-    frame->width  = c->width;
-    frame->height = c->height;
-
-    /* the image can be allocated by any means and av_image_alloc() is
-     * just the most convenient way if av_malloc() is to be used */
-    ret = av_image_alloc(frame->data, frame->linesize, c->width, c->height,
-                         c->pix_fmt, 32);
-    if (ret < 0) {
-        fprintf(stderr, "Could not allocate raw picture buffer\n");
-        exit(1);
-    }
-
-    /* encode 1 second of video */
-    for (i = 0; i < 25; i++) {
-        av_init_packet(&pkt);
-        pkt.data = NULL;    // packet data will be allocated by the encoder
-        pkt.size = 0;
-
-        fflush(stdout);
-        /* prepare a dummy image */
-        /* Y */
-        for (y = 0; y < c->height; y++) {
-            for (x = 0; x < c->width; x++) {
-                frame->data[0][y * frame->linesize[0] + x] = x + y + i * 3;
-            }
-        }
-
-        /* Cb and Cr */
-        for (y = 0; y < c->height/2; y++) {
-            for (x = 0; x < c->width/2; x++) {
-                frame->data[1][y * frame->linesize[1] + x] = 128 + y + i * 2;
-                frame->data[2][y * frame->linesize[2] + x] = 64 + x + i * 5;
-            }
-        }
-
-        frame->pts = i;
-
-        /* encode the image */
-        ret = avcodec_encode_video2(c, &pkt, frame, &got_output);
-        if (ret < 0) {
-            fprintf(stderr, "Error encoding frame\n");
-            exit(1);
-        }
-
-        if (got_output) {
-            printf("Write frame %3d (size=%5d)\n", i, pkt.size);
-            fwrite(pkt.data, 1, pkt.size, f);
-            av_free_packet(&pkt);
-        }
-    }
-
-    /* get the delayed frames */
-    for (got_output = 1; got_output; i++) {
-        fflush(stdout);
-
-        ret = avcodec_encode_video2(c, &pkt, NULL, &got_output);
-        if (ret < 0) {
-            fprintf(stderr, "Error encoding frame\n");
-            exit(1);
-        }
-
-        if (got_output) {
-            printf("Write frame %3d (size=%5d)\n", i, pkt.size);
-            fwrite(pkt.data, 1, pkt.size, f);
-            av_free_packet(&pkt);
-        }
-    }
-
-    /* add sequence end code to have a real mpeg file */
-    fwrite(endcode, 1, sizeof(endcode), f);
-    fclose(f);
-
-    avcodec_close(c);
-    av_free(c);
-    av_freep(&frame->data[0]);
-    av_frame_free(&frame);
-    printf("\n");
-}
-
-/*
- * Video decoding example
- */
-
-static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
-                     char *filename)
-{
-    FILE *f;
-    int i;
-
-    f = fopen(filename,"w");
-    fprintf(f, "P5\n%d %d\n%d\n", xsize, ysize, 255);
-    for (i = 0; i < ysize; i++)
-        fwrite(buf + i * wrap, 1, xsize, f);
-    fclose(f);
-}
-
-static int decode_write_frame(const char *outfilename, AVCodecContext *avctx,
-                              AVFrame *frame, int *frame_count, AVPacket *pkt, int last)
-{
-    int len, got_frame;
-    char buf[1024];
-
-    len = avcodec_decode_video2(avctx, frame, &got_frame, pkt);
-    if (len < 0) {
-        fprintf(stderr, "Error while decoding frame %d\n", *frame_count);
-        return len;
-    }
-    if (got_frame) {
-        printf("Saving %sframe %3d\n", last ? "last " : "", *frame_count);
-        fflush(stdout);
-
-        /* the picture is allocated by the decoder, no need to free it */
-        snprintf(buf, sizeof(buf), outfilename, *frame_count);
-        pgm_save(frame->data[0], frame->linesize[0],
-                 avctx->width, avctx->height, buf);
-        (*frame_count)++;
-    }
-    if (pkt->data) {
-        pkt->size -= len;
-        pkt->data += len;
-    }
-    return 0;
-}
-
-static void video_decode_example(const char *outfilename, const char *filename)
-{
-    AVCodec *codec;
-    AVCodecContext *c= NULL;
-    int frame_count;
-    FILE *f;
-    AVFrame *frame;
-    uint8_t inbuf[INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
-    AVPacket avpkt;
-
-    av_init_packet(&avpkt);
-
-    /* set end of buffer to 0 (this ensures that no overreading happens for damaged mpeg streams) */
-    memset(inbuf + INBUF_SIZE, 0, FF_INPUT_BUFFER_PADDING_SIZE);
-
-    printf("Decode video file %s to %s\n", filename, outfilename);
-
-    /* find the mpeg1 video decoder */
-    codec = avcodec_find_decoder(AV_CODEC_ID_MPEG1VIDEO);
-    if (!codec) {
-        fprintf(stderr, "Codec not found\n");
-        exit(1);
-    }
-
-    c = avcodec_alloc_context3(codec);
-    if (!c) {
-        fprintf(stderr, "Could not allocate video codec context\n");
-        exit(1);
-    }
-
-    if(codec->capabilities&CODEC_CAP_TRUNCATED)
-        c->flags|= CODEC_FLAG_TRUNCATED; /* we do not send complete frames */
-
-    /* For some codecs, such as msmpeg4 and mpeg4, width and height
-       MUST be initialized there because this information is not
-       available in the bitstream. */
-
-    /* open it */
-    if (avcodec_open2(c, codec, NULL) < 0) {
-        fprintf(stderr, "Could not open codec\n");
-        exit(1);
-    }
-
-    f = fopen(filename, "rb");
-    if (!f) {
-        fprintf(stderr, "Could not open %s\n", filename);
-        exit(1);
-    }
-
-    frame = av_frame_alloc();
-    if (!frame) {
-        fprintf(stderr, "Could not allocate video frame\n");
-        exit(1);
-    }
-
-    frame_count = 0;
-    for (;;) {
-        avpkt.size = fread(inbuf, 1, INBUF_SIZE, f);
-        if (avpkt.size == 0)
-            break;
-
-        /* NOTE1: some codecs are stream based (mpegvideo, mpegaudio)
-           and this is the only method to use them because you cannot
-           know the compressed data size before analysing it.
-
-           BUT some other codecs (msmpeg4, mpeg4) are inherently frame
-           based, so you must call them with all the data for one
-           frame exactly. You must also initialize 'width' and
-           'height' before initializing them. */
-
-        /* NOTE2: some codecs allow the raw parameters (frame size,
-           sample rate) to be changed at any frame. We handle this, so
-           you should also take care of it */
-
-        /* here, we use a stream based decoder (mpeg1video), so we
-           feed decoder and see if it could decode a frame */
-        avpkt.data = inbuf;
-        while (avpkt.size > 0)
-            if (decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 0) < 0)
-                exit(1);
-    }
-
-    /* some codecs, such as MPEG, transmit the I and P frame with a
-       latency of one frame. You must do the following to have a
-       chance to get the last frame of the video */
-    avpkt.data = NULL;
-    avpkt.size = 0;
-    decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 1);
-
-    fclose(f);
-
-    avcodec_close(c);
-    av_free(c);
-    av_frame_free(&frame);
-    printf("\n");
-}
-
-int main(int argc, char **argv)
-{
-    const char *output_type;
-
-    /* register all the codecs */
-    avcodec_register_all();
-
-    if (argc < 2) {
-        printf("usage: %s output_type\n"
-               "API example program to decode/encode a media stream with libavcodec.\n"
-               "This program generates a synthetic stream and encodes it to a file\n"
-               "named test.h264, test.mp2 or test.mpg depending on output_type.\n"
-               "The encoded stream is then decoded and written to a raw data output.\n"
-               "output_type must be choosen between 'h264', 'mp2', 'mpg'.\n",
-               argv[0]);
-        return 1;
-    }
-    output_type = argv[1];
-
-    if (!strcmp(output_type, "h264")) {
-        video_encode_example("test.h264", AV_CODEC_ID_H264);
-    } else if (!strcmp(output_type, "mp2")) {
-        audio_encode_example("test.mp2");
-        audio_decode_example("test.sw", "test.mp2");
-    } else if (!strcmp(output_type, "mpg")) {
-        video_encode_example("test.mpg", AV_CODEC_ID_MPEG1VIDEO);
-        video_decode_example("test%02d.pgm", "test.mpg");
-    } else {
-        fprintf(stderr, "Invalid output type '%s', choose between 'h264', 'mp2', or 'mpg'\n",
-                output_type);
-        return 1;
-    }
-
-    return 0;
-}
--- a/doc/examples/avio_reading.c
+++ b/doc/examples/avio_reading.c
@@ -1,132 +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 */
-    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
@@ -0,0 +1,479 @@
+/*
+ * Copyright (c) 2001 Fabrice Bellard
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * libavcodec API use example.
+ *
+ * Note that libavcodec only handles codecs (mpeg, mpeg4, etc...),
+ * not file formats (avi, vob, mp4, mov, mkv, mxf, flv, mpegts, mpegps, etc...). See library 'libavformat' for the
+ * format handling
+ */
+
+#include <math.h>
+
+#include <libavutil/imgutils.h>
+#include <libavutil/opt.h>
+#include <libavcodec/avcodec.h>
+#include <libavutil/mathematics.h>
+#include <libavutil/samplefmt.h>
+
+#define INBUF_SIZE 4096
+#define AUDIO_INBUF_SIZE 20480
+#define AUDIO_REFILL_THRESH 4096
+
+/*
+ * Audio encoding example
+ */
+static void audio_encode_example(const char *filename)
+{
+    AVCodec *codec;
+    AVCodecContext *c= NULL;
+    int frame_size, i, j, out_size, outbuf_size;
+    FILE *f;
+    short *samples;
+    float t, tincr;
+    uint8_t *outbuf;
+
+    printf("Encode audio file %s\n", filename);
+
+    /* find the MP2 encoder */
+    codec = avcodec_find_encoder(CODEC_ID_MP2);
+    if (!codec) {
+        fprintf(stderr, "codec not found\n");
+        exit(1);
+    }
+
+    c = avcodec_alloc_context3(codec);
+
+    /* put sample parameters */
+    c->bit_rate = 64000;
+    c->sample_rate = 44100;
+    c->channels = 2;
+    c->sample_fmt = AV_SAMPLE_FMT_S16;
+
+    /* open it */
+    if (avcodec_open2(c, codec, NULL) < 0) {
+        fprintf(stderr, "could not open codec\n");
+        exit(1);
+    }
+
+    /* the codec gives us the frame size, in samples */
+    frame_size = c->frame_size;
+    samples = malloc(frame_size * 2 * c->channels);
+    outbuf_size = 10000;
+    outbuf = malloc(outbuf_size);
+
+    f = fopen(filename, "wb");
+    if (!f) {
+        fprintf(stderr, "could not open %s\n", filename);
+        exit(1);
+    }
+
+    /* encode a single tone sound */
+    t = 0;
+    tincr = 2 * M_PI * 440.0 / c->sample_rate;
+    for(i=0;i<200;i++) {
+        for(j=0;j<frame_size;j++) {
+            samples[2*j] = (int)(sin(t) * 10000);
+            samples[2*j+1] = samples[2*j];
+            t += tincr;
+        }
+        /* encode the samples */
+        out_size = avcodec_encode_audio(c, outbuf, outbuf_size, samples);
+        fwrite(outbuf, 1, out_size, f);
+    }
+    fclose(f);
+    free(outbuf);
+    free(samples);
+
+    avcodec_close(c);
+    av_free(c);
+}
+
+/*
+ * Audio decoding.
+ */
+static void audio_decode_example(const char *outfilename, const char *filename)
+{
+    AVCodec *codec;
+    AVCodecContext *c= NULL;
+    int len;
+    FILE *f, *outfile;
+    uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
+    AVPacket avpkt;
+    AVFrame *decoded_frame = NULL;
+
+    av_init_packet(&avpkt);
+
+    printf("Decode audio file %s\n", filename);
+
+    /* find the mpeg audio decoder */
+    codec = avcodec_find_decoder(CODEC_ID_MP2);
+    if (!codec) {
+        fprintf(stderr, "codec not found\n");
+        exit(1);
+    }
+
+    c = avcodec_alloc_context3(codec);
+
+    /* open it */
+    if (avcodec_open2(c, codec, NULL) < 0) {
+        fprintf(stderr, "could not open codec\n");
+        exit(1);
+    }
+
+    f = fopen(filename, "rb");
+    if (!f) {
+        fprintf(stderr, "could not open %s\n", filename);
+        exit(1);
+    }
+    outfile = fopen(outfilename, "wb");
+    if (!outfile) {
+        av_free(c);
+        exit(1);
+    }
+
+    /* decode until eof */
+    avpkt.data = inbuf;
+    avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
+
+    while (avpkt.size > 0) {
+        int got_frame = 0;
+
+        if (!decoded_frame) {
+            if (!(decoded_frame = avcodec_alloc_frame())) {
+                fprintf(stderr, "out of memory\n");
+                exit(1);
+            }
+        } else
+            avcodec_get_frame_defaults(decoded_frame);
+
+        len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
+        if (len < 0) {
+            fprintf(stderr, "Error while decoding\n");
+            exit(1);
+        }
+        if (got_frame) {
+            /* if a frame has been decoded, output it */
+            int data_size = av_samples_get_buffer_size(NULL, c->channels,
+                                                       decoded_frame->nb_samples,
+                                                       c->sample_fmt, 1);
+            fwrite(decoded_frame->data[0], 1, data_size, outfile);
+        }
+        avpkt.size -= len;
+        avpkt.data += len;
+        avpkt.dts =
+        avpkt.pts = AV_NOPTS_VALUE;
+        if (avpkt.size < AUDIO_REFILL_THRESH) {
+            /* Refill the input buffer, to avoid trying to decode
+             * incomplete frames. Instead of this, one could also use
+             * a parser, or use a proper container format through
+             * libavformat. */
+            memmove(inbuf, avpkt.data, avpkt.size);
+            avpkt.data = inbuf;
+            len = fread(avpkt.data + avpkt.size, 1,
+                        AUDIO_INBUF_SIZE - avpkt.size, f);
+            if (len > 0)
+                avpkt.size += len;
+        }
+    }
+
+    fclose(outfile);
+    fclose(f);
+
+    avcodec_close(c);
+    av_free(c);
+    av_free(decoded_frame);
+}
+
+/*
+ * Video encoding example
+ */
+static void video_encode_example(const char *filename, int codec_id)
+{
+    AVCodec *codec;
+    AVCodecContext *c= NULL;
+    int i, out_size, x, y, outbuf_size;
+    FILE *f;
+    AVFrame *picture;
+    uint8_t *outbuf;
+    int had_output=0;
+
+    printf("Encode video file %s\n", filename);
+
+    /* find the mpeg1 video encoder */
+    codec = avcodec_find_encoder(codec_id);
+    if (!codec) {
+        fprintf(stderr, "codec not found\n");
+        exit(1);
+    }
+
+    c = avcodec_alloc_context3(codec);
+    picture= avcodec_alloc_frame();
+
+    /* put sample parameters */
+    c->bit_rate = 400000;
+    /* resolution must be a multiple of two */
+    c->width = 352;
+    c->height = 288;
+    /* frames per second */
+    c->time_base= (AVRational){1,25};
+    c->gop_size = 10; /* emit one intra frame every ten frames */
+    c->max_b_frames=1;
+    c->pix_fmt = PIX_FMT_YUV420P;
+
+    if(codec_id == CODEC_ID_H264)
+        av_opt_set(c->priv_data, "preset", "slow", 0);
+
+    /* open it */
+    if (avcodec_open2(c, codec, NULL) < 0) {
+        fprintf(stderr, "could not open codec\n");
+        exit(1);
+    }
+
+    f = fopen(filename, "wb");
+    if (!f) {
+        fprintf(stderr, "could not open %s\n", filename);
+        exit(1);
+    }
+
+    /* alloc image and output buffer */
+    outbuf_size = 100000 + 12*c->width*c->height;
+    outbuf = malloc(outbuf_size);
+
+    /* the image can be allocated by any means and av_image_alloc() is
+     * just the most convenient way if av_malloc() is to be used */
+    av_image_alloc(picture->data, picture->linesize,
+                   c->width, c->height, c->pix_fmt, 1);
+
+    /* encode 1 second of video */
+    for(i=0;i<25;i++) {
+        fflush(stdout);
+        /* prepare a dummy image */
+        /* Y */
+        for(y=0;y<c->height;y++) {
+            for(x=0;x<c->width;x++) {
+                picture->data[0][y * picture->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++) {
+                picture->data[1][y * picture->linesize[1] + x] = 128 + y + i * 2;
+                picture->data[2][y * picture->linesize[2] + x] = 64 + x + i * 5;
+            }
+        }
+
+        /* encode the image */
+        out_size = avcodec_encode_video(c, outbuf, outbuf_size, picture);
+        had_output |= out_size;
+        printf("encoding frame %3d (size=%5d)\n", i, out_size);
+        fwrite(outbuf, 1, out_size, f);
+    }
+
+    /* get the delayed frames */
+    for(; out_size || !had_output; i++) {
+        fflush(stdout);
+
+        out_size = avcodec_encode_video(c, outbuf, outbuf_size, NULL);
+        had_output |= out_size;
+        printf("write frame %3d (size=%5d)\n", i, out_size);
+        fwrite(outbuf, 1, out_size, f);
+    }
+
+    /* add sequence end code to have a real mpeg file */
+    outbuf[0] = 0x00;
+    outbuf[1] = 0x00;
+    outbuf[2] = 0x01;
+    outbuf[3] = 0xb7;
+    fwrite(outbuf, 1, 4, f);
+    fclose(f);
+    free(outbuf);
+
+    avcodec_close(c);
+    av_free(c);
+    av_free(picture->data[0]);
+    av_free(picture);
+    printf("\n");
+}
+
+/*
+ * Video decoding example
+ */
+
+static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
+                     char *filename)
+{
+    FILE *f;
+    int i;
+
+    f=fopen(filename,"w");
+    fprintf(f,"P5\n%d %d\n%d\n",xsize,ysize,255);
+    for(i=0;i<ysize;i++)
+        fwrite(buf + i * wrap,1,xsize,f);
+    fclose(f);
+}
+
+static void video_decode_example(const char *outfilename, const char *filename)
+{
+    AVCodec *codec;
+    AVCodecContext *c= NULL;
+    int frame, got_picture, len;
+    FILE *f;
+    AVFrame *picture;
+    uint8_t inbuf[INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
+    char buf[1024];
+    AVPacket avpkt;
+
+    av_init_packet(&avpkt);
+
+    /* set end of buffer to 0 (this ensures that no overreading happens for damaged mpeg streams) */
+    memset(inbuf + INBUF_SIZE, 0, FF_INPUT_BUFFER_PADDING_SIZE);
+
+    printf("Decode video file %s\n", filename);
+
+    /* find the mpeg1 video decoder */
+    codec = avcodec_find_decoder(CODEC_ID_MPEG1VIDEO);
+    if (!codec) {
+        fprintf(stderr, "codec not found\n");
+        exit(1);
+    }
+
+    c = avcodec_alloc_context3(codec);
+    picture= avcodec_alloc_frame();
+
+    if(codec->capabilities&CODEC_CAP_TRUNCATED)
+        c->flags|= CODEC_FLAG_TRUNCATED; /* we do not send complete frames */
+
+    /* For some codecs, such as msmpeg4 and mpeg4, width and height
+       MUST be initialized there because this information is not
+       available in the bitstream. */
+
+    /* open it */
+    if (avcodec_open2(c, codec, NULL) < 0) {
+        fprintf(stderr, "could not open codec\n");
+        exit(1);
+    }
+
+    /* the codec gives us the frame size, in samples */
+
+    f = fopen(filename, "rb");
+    if (!f) {
+        fprintf(stderr, "could not open %s\n", filename);
+        exit(1);
+    }
+
+    frame = 0;
+    for(;;) {
+        avpkt.size = fread(inbuf, 1, INBUF_SIZE, f);
+        if (avpkt.size == 0)
+            break;
+
+        /* NOTE1: some codecs are stream based (mpegvideo, mpegaudio)
+           and this is the only method to use them because you cannot
+           know the compressed data size before analysing it.
+
+           BUT some other codecs (msmpeg4, mpeg4) are inherently frame
+           based, so you must call them with all the data for one
+           frame exactly. You must also initialize 'width' and
+           'height' before initializing them. */
+
+        /* NOTE2: some codecs allow the raw parameters (frame size,
+           sample rate) to be changed at any frame. We handle this, so
+           you should also take care of it */
+
+        /* here, we use a stream based decoder (mpeg1video), so we
+           feed decoder and see if it could decode a frame */
+        avpkt.data = inbuf;
+        while (avpkt.size > 0) {
+            len = avcodec_decode_video2(c, picture, &got_picture, &avpkt);
+            if (len < 0) {
+                fprintf(stderr, "Error while decoding frame %d\n", frame);
+                exit(1);
+            }
+            if (got_picture) {
+                printf("saving frame %3d\n", frame);
+                fflush(stdout);
+
+                /* the picture is allocated by the decoder. no need to
+                   free it */
+                snprintf(buf, sizeof(buf), outfilename, frame);
+                pgm_save(picture->data[0], picture->linesize[0],
+                         c->width, c->height, buf);
+                frame++;
+            }
+            avpkt.size -= len;
+            avpkt.data += len;
+        }
+    }
+
+    /* some codecs, such as MPEG, transmit the I and P frame with a
+       latency of one frame. You must do the following to have a
+       chance to get the last frame of the video */
+    avpkt.data = NULL;
+    avpkt.size = 0;
+    len = avcodec_decode_video2(c, picture, &got_picture, &avpkt);
+    if (got_picture) {
+        printf("saving last frame %3d\n", frame);
+        fflush(stdout);
+
+        /* the picture is allocated by the decoder. no need to
+           free it */
+        snprintf(buf, sizeof(buf), outfilename, frame);
+        pgm_save(picture->data[0], picture->linesize[0],
+                 c->width, c->height, buf);
+        frame++;
+    }
+
+    fclose(f);
+
+    avcodec_close(c);
+    av_free(c);
+    av_free(picture);
+    printf("\n");
+}
+
+int main(int argc, char **argv)
+{
+    const char *filename;
+
+    /* register all the codecs */
+    avcodec_register_all();
+
+    if (argc <= 1) {
+        audio_encode_example("/tmp/test.mp2");
+        audio_decode_example("/tmp/test.sw", "/tmp/test.mp2");
+
+        video_encode_example("/tmp/test.h264", CODEC_ID_H264);
+        video_encode_example("/tmp/test.mpg", CODEC_ID_MPEG1VIDEO);
+        filename = "/tmp/test.mpg";
+    } else {
+        filename = argv[1];
+    }
+
+    //    audio_decode_example("/tmp/test.sw", filename);
+    video_decode_example("/tmp/test%d.pgm", filename);
+
+    return 0;
+}
--- a/doc/examples/demuxing_decoding.c
+++ b/doc/examples/demuxing_decoding.c
@@ -1,386 +0,0 @@
-/*
- * Copyright (c) 2012 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
- * Demuxing and decoding example.
- *
- * Show how to use the libavformat and libavcodec API to demux and
- * decode audio and video data.
- * @example demuxing_decoding.c
- */
-
-#include <libavutil/imgutils.h>
-#include <libavutil/samplefmt.h>
-#include <libavutil/timestamp.h>
-#include <libavformat/avformat.h>
-
-static AVFormatContext *fmt_ctx = NULL;
-static AVCodecContext *video_dec_ctx = NULL, *audio_dec_ctx;
-static AVStream *video_stream = NULL, *audio_stream = NULL;
-static const char *src_filename = NULL;
-static const char *video_dst_filename = NULL;
-static const char *audio_dst_filename = NULL;
-static FILE *video_dst_file = NULL;
-static FILE *audio_dst_file = NULL;
-
-static uint8_t *video_dst_data[4] = {NULL};
-static int      video_dst_linesize[4];
-static int video_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;
-
-/* The different ways of decoding and managing data memory. You are not
- * supposed to support all the modes in your application but pick the one most
- * appropriate to your needs. Look for the use of api_mode in this example to
- * see what are the differences of API usage between them */
-enum {
-    API_MODE_OLD                  = 0, /* old method, deprecated */
-    API_MODE_NEW_API_REF_COUNT    = 1, /* new method, using the frame reference counting */
-    API_MODE_NEW_API_NO_REF_COUNT = 2, /* new method, without reference counting */
-};
-
-static int api_mode = API_MODE_OLD;
-
-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));
-            return ret;
-        }
-
-        if (*got_frame) {
-            printf("video_frame%s n:%d coded_n:%d pts:%s\n",
-                   cached ? "(cached)" : "",
-                   video_frame_count++, frame->coded_picture_number,
-                   av_ts2timestr(frame->pts, &video_dec_ctx->time_base));
-
-            /* copy decoded frame to destination buffer:
-             * this is required since rawvideo expects non aligned data */
-            av_image_copy(video_dst_data, video_dst_linesize,
-                          (const uint8_t **)(frame->data), frame->linesize,
-                          video_dec_ctx->pix_fmt, video_dec_ctx->width, video_dec_ctx->height);
-
-            /* write to rawvideo file */
-            fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file);
-        }
-    } else if (pkt.stream_index == audio_stream_idx) {
-        /* decode audio frame */
-        ret = avcodec_decode_audio4(audio_dec_ctx, frame, got_frame, &pkt);
-        if (ret < 0) {
-            fprintf(stderr, "Error decoding audio frame (%s)\n", av_err2str(ret));
-            return ret;
-        }
-        /* Some audio decoders decode only part of the packet, and have to be
-         * called again with the remainder of the packet data.
-         * Sample: fate-suite/lossless-audio/luckynight-partial.shn
-         * Also, some decoders might over-read the packet. */
-        decoded = FFMIN(ret, pkt.size);
-
-        if (*got_frame) {
-            size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format);
-            printf("audio_frame%s n:%d nb_samples:%d pts:%s\n",
-                   cached ? "(cached)" : "",
-                   audio_frame_count++, frame->nb_samples,
-                   av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));
-
-            /* Write the raw audio data samples of the first plane. This works
-             * fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However,
-             * most audio decoders output planar audio, which uses a separate
-             * plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P).
-             * In other words, this code will write only the first audio channel
-             * in these cases.
-             * You should use libswresample or libavfilter to convert the frame
-             * to packed data. */
-            fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file);
-        }
-    }
-
-    /* If we use the new API with reference counting, we own the data and need
-     * to de-reference it when we don't use it anymore */
-    if (*got_frame && api_mode == API_MODE_NEW_API_REF_COUNT)
-        av_frame_unref(frame);
-
-    return decoded;
-}
-
-static int open_codec_context(int *stream_idx,
-                              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 decoders, with or without reference counting */
-        if (api_mode == API_MODE_NEW_API_REF_COUNT)
-            av_dict_set(&opts, "refcounted_frames", "1", 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;
-}
-
-static int get_format_from_sample_fmt(const char **fmt,
-                                      enum AVSampleFormat sample_fmt)
-{
-    int i;
-    struct sample_fmt_entry {
-        enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
-    } sample_fmt_entries[] = {
-        { AV_SAMPLE_FMT_U8,  "u8",    "u8"    },
-        { AV_SAMPLE_FMT_S16, "s16be", "s16le" },
-        { AV_SAMPLE_FMT_S32, "s32be", "s32le" },
-        { AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
-        { AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
-    };
-    *fmt = NULL;
-
-    for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
-        struct sample_fmt_entry *entry = &sample_fmt_entries[i];
-        if (sample_fmt == entry->sample_fmt) {
-            *fmt = AV_NE(entry->fmt_be, entry->fmt_le);
-            return 0;
-        }
-    }
-
-    fprintf(stderr,
-            "sample format %s is not supported as output format\n",
-            av_get_sample_fmt_name(sample_fmt));
-    return -1;
-}
-
-int main (int argc, char **argv)
-{
-    int ret = 0, got_frame;
-
-    if (argc != 4 && argc != 5) {
-        fprintf(stderr, "usage: %s [-refcount=<old|new_norefcount|new_refcount>] "
-                "input_file video_output_file audio_output_file\n"
-                "API example program to show how to read frames from an input file.\n"
-                "This program reads frames from a file, decodes them, and writes decoded\n"
-                "video frames to a rawvideo file named video_output_file, and decoded\n"
-                "audio frames to a rawaudio file named audio_output_file.\n\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. If unset, it's using\n"
-                "the classic old method.\n"
-                "\n", argv[0]);
-        exit(1);
-    }
-    if (argc == 5) {
-        const char *mode = argv[1] + strlen("-refcount=");
-        if      (!strcmp(mode, "old"))            api_mode = API_MODE_OLD;
-        else if (!strcmp(mode, "new_norefcount")) api_mode = API_MODE_NEW_API_NO_REF_COUNT;
-        else if (!strcmp(mode, "new_refcount"))   api_mode = API_MODE_NEW_API_REF_COUNT;
-        else {
-            fprintf(stderr, "unknow mode '%s'\n", mode);
-            exit(1);
-        }
-        argv++;
-    }
-    src_filename = argv[1];
-    video_dst_filename = argv[2];
-    audio_dst_filename = argv[3];
-
-    /* register all formats and codecs */
-    av_register_all();
-
-    /* open input file, and allocate format context */
-    if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
-        fprintf(stderr, "Could not open source file %s\n", src_filename);
-        exit(1);
-    }
-
-    /* retrieve stream information */
-    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;
-
-        video_dst_file = fopen(video_dst_filename, "wb");
-        if (!video_dst_file) {
-            fprintf(stderr, "Could not open destination file %s\n", video_dst_filename);
-            ret = 1;
-            goto end;
-        }
-
-        /* allocate image where the decoded image will be put */
-        ret = av_image_alloc(video_dst_data, video_dst_linesize,
-                             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;
-        }
-        video_dst_bufsize = ret;
-    }
-
-    if (open_codec_context(&audio_stream_idx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) {
-        audio_stream = fmt_ctx->streams[audio_stream_idx];
-        audio_dec_ctx = audio_stream->codec;
-        audio_dst_file = fopen(audio_dst_filename, "wb");
-        if (!audio_dst_file) {
-            fprintf(stderr, "Could not open destination file %s\n", video_dst_filename);
-            ret = 1;
-            goto end;
-        }
-    }
-
-    /* dump input information to stderr */
-    av_dump_format(fmt_ctx, 0, src_filename, 0);
-
-    if (!audio_stream && !video_stream) {
-        fprintf(stderr, "Could not find audio or video stream in the input, aborting\n");
-        ret = 1;
-        goto end;
-    }
-
-    /* When using the new API, you need to use the libavutil/frame.h API, while
-     * the classic frame management is available in libavcodec */
-    if (api_mode == API_MODE_OLD)
-        frame = avcodec_alloc_frame();
-    else
-        frame = av_frame_alloc();
-    if (!frame) {
-        fprintf(stderr, "Could not allocate frame\n");
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-
-    /* initialize packet, set data to NULL, let the demuxer fill it */
-    av_init_packet(&pkt);
-    pkt.data = NULL;
-    pkt.size = 0;
-
-    if (video_stream)
-        printf("Demuxing video from file '%s' into '%s'\n", src_filename, video_dst_filename);
-    if (audio_stream)
-        printf("Demuxing audio from file '%s' into '%s'\n", src_filename, audio_dst_filename);
-
-    /* read frames from the file */
-    while (av_read_frame(fmt_ctx, &pkt) >= 0) {
-        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_free_packet(&orig_pkt);
-    }
-
-    /* flush cached frames */
-    pkt.data = NULL;
-    pkt.size = 0;
-    do {
-        decode_packet(&got_frame, 1);
-    } while (got_frame);
-
-    printf("Demuxing succeeded.\n");
-
-    if (video_stream) {
-        printf("Play the output video file with the command:\n"
-               "ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
-               av_get_pix_fmt_name(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)
-            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,
-               audio_dst_filename);
-    }
-
-end:
-    avcodec_close(video_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);
-    if (api_mode == API_MODE_OLD)
-        avcodec_free_frame(&frame);
-    else
-        av_frame_free(&frame);
-    av_free(video_dst_data[0]);
-
-    return ret < 0;
-}
--- a/doc/examples/filter_audio.c
+++ b/doc/examples/filter_audio.c
@@ -1,364 +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/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 +25,6 @@
 /**
 * @file
 * API example for audio decoding and filtering
- * @example filtering_audio.c
 */

 #include <unistd.h>
@@ -36,10 +35,9 @@
 #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,aconvert=s16:mono";
+const char *player       = "ffplay -f s16le -ar 8000 -ac 1 -";

 static AVFormatContext *fmt_ctx;
 static AVCodecContext *dec_ctx;
@@ -71,7 +69,6 @@ static int open_input_file(const char *filename)
    }
    audio_stream_index = ret;
    dec_ctx = fmt_ctx->streams[audio_stream_index]->codec;
-    av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);

    /* init the audio decoder */
    if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
@@ -85,64 +82,40 @@ 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 sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
+    const int64_t *chlayouts                = avfilter_all_channel_layouts;
+    AVABufferSinkParams *abuffersink_params;
    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)
        dec_ctx->channel_layout = av_get_default_channel_layout(dec_ctx->channels);
-    snprintf(args, sizeof(args),
-            "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
-             time_base.num, time_base.den, dec_ctx->sample_rate,
-             av_get_sample_fmt_name(dec_ctx->sample_fmt), dec_ctx->channel_layout);
+    snprintf(args, sizeof(args), "%d:%d:0x%"PRIx64,
+             dec_ctx->sample_rate, dec_ctx->sample_fmt, dec_ctx->channel_layout);
    ret = avfilter_graph_create_filter(&buffersrc_ctx, abuffersrc, "in",
                                       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. */
+    abuffersink_params = av_abuffersink_params_alloc();
+    abuffersink_params->sample_fmts     = sample_fmts;
+    abuffersink_params->channel_layouts = chlayouts;
    ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out",
-                                       NULL, NULL, filter_graph);
+                                       NULL, abuffersink_params, filter_graph);
+    av_free(abuffersink_params);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
-        goto end;
-    }
-
-    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;
-    }
-
-    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;
-    }
-
-    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;
    }

    /* Endpoints for the filter graph. */
@@ -156,12 +129,12 @@ static int init_filters(const char *filters_descr)
    inputs->pad_idx    = 0;
    inputs->next       = NULL;

-    if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
-                                        &inputs, &outputs, NULL)) < 0)
-        goto end;
+    if ((ret = avfilter_graph_parse(filter_graph, filters_descr,
+                                    &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 */
@@ -172,17 +145,14 @@ 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)
+static void print_samplesref(AVFilterBufferRef *samplesref)
 {
-    const int n = frame->nb_samples * av_get_channel_layout_nb_channels(av_frame_get_channel_layout(frame));
-    const uint16_t *p     = (uint16_t*)frame->data[0];
+    const AVFilterBufferRefAudioProps *props = samplesref->audio;
+    const int n = props->nb_samples * av_get_channel_layout_nb_channels(props->channel_layout);
+    const uint16_t *p     = (uint16_t*)samplesref->data[0];
    const uint16_t *p_end = p + n;

    while (p < p_end) {
@@ -196,20 +166,16 @@ static void print_frame(const AVFrame *frame)
 int main(int argc, char **argv)
 {
    int ret;
-    AVPacket packet0, packet;
-    AVFrame *frame = av_frame_alloc();
-    AVFrame *filt_frame = av_frame_alloc();
+    AVPacket packet;
+    AVFrame frame;
    int got_frame;

-    if (!frame || !filt_frame) {
-        perror("Could not allocate frame");
-        exit(1);
-    }
    if (argc != 2) {
        fprintf(stderr, "Usage: %s file | %s\n", argv[0], player);
        exit(1);
    }

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

@@ -219,60 +185,49 @@ 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;
-        }
+        AVFilterBufferRef *samplesref;
+        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);
+            ret = avcodec_decode_audio4(dec_ctx, &frame, &got_frame, &packet);
+            av_free_packet(&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 */
-                if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, 0) < 0) {
+                if (av_buffersrc_add_frame(buffersrc_ctx, &frame, 0) < 0) {
                    av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
                    break;
                }

                /* pull filtered audio from the filtergraph */
-                while (1) {
-                    ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
-                    if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
-                        break;
-                    if (ret < 0)
-                        goto end;
-                    print_frame(filt_frame);
-                    av_frame_unref(filt_frame);
+                while (avfilter_poll_frame(buffersink_ctx->inputs[0])) {
+                    av_buffersink_get_buffer_ref(buffersink_ctx, &samplesref, 0);
+                    if (samplesref) {
+                        print_samplesref(samplesref);
+                        avfilter_unref_buffer(samplesref);
+                    }
                }
            }
-
-            if (packet.size <= 0)
-                av_free_packet(&packet0);
-        } else {
-            /* discard non-wanted packets */
-            av_free_packet(&packet0);
        }
    }
 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 +24,6 @@
 /**
 * @file
 * API example for decoding and filtering
- * @example filtering_video.c
 */

 #define _XOPEN_SOURCE 600 /* for usleep */
@@ -35,8 +34,6 @@
 #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";

@@ -71,7 +68,6 @@ 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) {
@@ -85,46 +81,32 @@ 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();
-    enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE };
-
+    enum PixelFormat pix_fmts[] = { PIX_FMT_GRAY8, PIX_FMT_NONE };
    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,
-            dec_ctx->time_base.num, dec_ctx->time_base.den,
-            dec_ctx->sample_aspect_ratio.num, dec_ctx->sample_aspect_ratio.den);
-
+    snprintf(args, sizeof(args), "%d:%d:%d:%d:%d:%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;
+        return ret;
    }

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

    /* Endpoints for the filter graph. */
@@ -138,47 +120,42 @@ static int init_filters(const char *filters_descr)
    inputs->pad_idx    = 0;
    inputs->next       = NULL;

-    if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
+    if ((ret = avfilter_graph_parse(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)
+static void display_picref(AVFilterBufferRef *picref, AVRational time_base)
 {
    int x, y;
    uint8_t *p0, *p;
    int64_t delay;

-    if (frame->pts != AV_NOPTS_VALUE) {
+    if (picref->pts != AV_NOPTS_VALUE) {
        if (last_pts != AV_NOPTS_VALUE) {
            /* sleep roughly the right amount of time;
             * usleep is in microseconds, just like AV_TIME_BASE. */
-            delay = av_rescale_q(frame->pts - last_pts,
+            delay = av_rescale_q(picref->pts - last_pts,
                                 time_base, AV_TIME_BASE_Q);
            if (delay > 0 && delay < 1000000)
                usleep(delay);
        }
-        last_pts = frame->pts;
+        last_pts = picref->pts;
    }

    /* Trivial ASCII grayscale display. */
-    p0 = frame->data[0];
+    p0 = picref->data[0];
    puts("\033c");
-    for (y = 0; y < frame->height; y++) {
+    for (y = 0; y < picref->video->h; y++) {
        p = p0;
-        for (x = 0; x < frame->width; x++)
+        for (x = 0; x < picref->video->w; x++)
            putchar(" .-+#"[*(p++) / 52]);
        putchar('\n');
-        p0 += frame->linesize[0];
+        p0 += picref->linesize[0];
    }
    fflush(stdout);
 }
@@ -187,19 +164,15 @@ int main(int argc, char **argv)
 {
    int ret;
    AVPacket packet;
-    AVFrame *frame = av_frame_alloc();
-    AVFrame *filt_frame = av_frame_alloc();
+    AVFrame frame;
    int got_frame;

-    if (!frame || !filt_frame) {
-        perror("Could not allocate frame");
-        exit(1);
-    }
    if (argc != 2) {
        fprintf(stderr, "Usage: %s file\n", argv[0]);
        exit(1);
    }

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

@@ -210,50 +183,47 @@ int main(int argc, char **argv)

    /* read all packets */
    while (1) {
+        AVFilterBufferRef *picref;
        if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
            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);
+            ret = avcodec_decode_video2(dec_ctx, &frame, &got_frame, &packet);
+            av_free_packet(&packet);
            if (ret < 0) {
                av_log(NULL, AV_LOG_ERROR, "Error decoding video\n");
                break;
            }

            if (got_frame) {
-                frame->pts = av_frame_get_best_effort_timestamp(frame);
+                frame.pts = av_frame_get_best_effort_timestamp(&frame);

                /* push the decoded frame into the filtergraph */
-                if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
-                    av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
-                    break;
-                }
+                av_vsrc_buffer_add_frame(buffersrc_ctx, &frame, 0);

-                /* pull filtered frames from the filtergraph */
-                while (1) {
-                    ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
-                    if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
-                        break;
-                    if (ret < 0)
-                        goto end;
-                    display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base);
-                    av_frame_unref(filt_frame);
+                /* pull filtered pictures from the filtergraph */
+                while (avfilter_poll_frame(buffersink_ctx->inputs[0])) {
+                    av_buffersink_get_buffer_ref(buffersink_ctx, &picref, 0);
+                    if (picref) {
+                        display_picref(picref, buffersink_ctx->inputs[0]->time_base);
+                        avfilter_unref_buffer(picref);
+                    }
                }
-                av_frame_unref(frame);
            }
        }
-        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/metadata.c
+++ b/doc/examples/metadata.c
@@ -23,7 +23,6 @@
 /**
 * @file
 * Shows how the metadata API can be used in application programs.
- * @example metadata.c
 */

 #include <stdio.h>
--- a/doc/examples/muxing.c
+++ b/doc/examples/muxing.c
@@ -24,9 +24,8 @@
 * @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.
 */

 #include <stdlib.h>
@@ -34,151 +33,74 @@
 #include <string.h>
 #include <math.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>

-static int audio_is_eof, video_is_eof;
+#undef exit

-#define STREAM_DURATION   10.0
+/* 5 seconds stream duration */
+#define STREAM_DURATION   200.0
 #define STREAM_FRAME_RATE 25 /* 25 images/s */
-#define STREAM_PIX_FMT    AV_PIX_FMT_YUV420P /* default pix_fmt */
+#define STREAM_NB_FRAMES  ((int)(STREAM_DURATION * STREAM_FRAME_RATE))
+#define STREAM_PIX_FMT    PIX_FMT_YUV420P /* default pix_fmt */

 static int sws_flags = SWS_BICUBIC;

-static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt)
-{
-    AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base;
+/**************************************************************/
+/* audio output */

-    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 float t, tincr, tincr2;
+static int16_t *samples;
+static int audio_input_frame_size;

-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 */
-    pkt->pts = av_rescale_q_rnd(pkt->pts, *time_base, st->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
-    pkt->dts = av_rescale_q_rnd(pkt->dts, *time_base, st->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
-    pkt->duration = av_rescale_q(pkt->duration, *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);
-}
-
-/* Add an output stream. */
-static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
-                            enum AVCodecID codec_id)
+/*
+ * add an audio output stream
+ */
+static AVStream *add_audio_stream(AVFormatContext *oc, enum CodecID codec_id)
 {
    AVCodecContext *c;
    AVStream *st;
+    AVCodec *codec;

-    /* find the encoder */
-    *codec = avcodec_find_encoder(codec_id);
-    if (!(*codec)) {
-        fprintf(stderr, "Could not find encoder for '%s'\n",
-                avcodec_get_name(codec_id));
+    /* find the audio encoder */
+    codec = avcodec_find_encoder(codec_id);
+    if (!codec) {
+        fprintf(stderr, "codec not found\n");
        exit(1);
    }

-    st = avformat_new_stream(oc, *codec);
+    st = avformat_new_stream(oc, codec);
    if (!st) {
-        fprintf(stderr, "Could not allocate stream\n");
+        fprintf(stderr, "Could not alloc stream\n");
        exit(1);
    }
-    st->id = oc->nb_streams-1;
+    st->id = 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->bit_rate    = 64000;
-        c->sample_rate = 44100;
-        c->channels    = 2;
-        break;
+    /* put sample parameters */
+    c->sample_fmt  = AV_SAMPLE_FMT_S16;
+    c->bit_rate    = 64000;
+    c->sample_rate = 44100;
+    c->channels    = 2;

-    case AVMEDIA_TYPE_VIDEO:
-        c->codec_id = codec_id;
-
-        c->bit_rate = 400000;
-        /* Resolution must be a multiple of two. */
-        c->width    = 352;
-        c->height   = 288;
-        /* timebase: This is the fundamental unit of time (in seconds) in terms
-         * of which frame timestamps are represented. For fixed-fps content,
-         * timebase should be 1/framerate and timestamp increments should be
-         * identical to 1. */
-        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) {
-            /* just for testing, we also add B frames */
-            c->max_b_frames = 2;
-        }
-        if (c->codec_id == AV_CODEC_ID_MPEG1VIDEO) {
-            /* Needed to avoid using macroblocks in which some coeffs overflow.
-             * This does not happen with normal video, it just happens here as
-             * the motion of the chroma plane does not match the luma plane. */
-            c->mb_decision = 2;
-        }
-    break;
-
-    default:
-        break;
-    }
-
-    /* Some formats want stream headers to be separate. */
+    // some formats want stream headers to be separate
    if (oc->oformat->flags & AVFMT_GLOBALHEADER)
        c->flags |= CODEC_FLAG_GLOBAL_HEADER;

    return st;
 }

-/**************************************************************/
-/* audio output */
-
-static float t, tincr, tincr2;
-
-AVFrame *audio_frame;
-static uint8_t **src_samples_data;
-static int       src_samples_linesize;
-static int       src_nb_samples;
-
-static int max_dst_nb_samples;
-uint8_t **dst_samples_data;
-int       dst_samples_linesize;
-int       dst_samples_size;
-int samples_count;
-
-struct SwrContext *swr_ctx = NULL;
-
-static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
+static void open_audio(AVFormatContext *oc, AVStream *st)
 {
    AVCodecContext *c;
-    int ret;

    c = st->codec;

-    /* allocate and init a re-usable frame */
-    audio_frame = av_frame_alloc();
-    if (!audio_frame) {
-        fprintf(stderr, "Could not allocate audio frame\n");
-        exit(1);
-    }
-
    /* open it */
-    ret = avcodec_open2(c, codec, NULL);
-    if (ret < 0) {
-        fprintf(stderr, "Could not open audio codec: %s\n", av_err2str(ret));
+    if (avcodec_open2(c, NULL, NULL) < 0) {
+        fprintf(stderr, "could not open codec\n");
        exit(1);
    }

@@ -188,54 +110,13 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
    /* increment frequency by 110 Hz per second */
    tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;

-    src_nb_samples = c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE ?
-        10000 : c->frame_size;
-
-    ret = av_samples_alloc_array_and_samples(&src_samples_data, &src_samples_linesize, c->channels,
-                                             src_nb_samples, AV_SAMPLE_FMT_S16, 0);
-    if (ret < 0) {
-        fprintf(stderr, "Could not allocate source samples\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;
-
-    /* create resampler context */
-    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       (swr_ctx, "in_channel_count",   c->channels,       0);
-        av_opt_set_int       (swr_ctx, "in_sample_rate",     c->sample_rate,    0);
-        av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt",      AV_SAMPLE_FMT_S16, 0);
-        av_opt_set_int       (swr_ctx, "out_channel_count",  c->channels,       0);
-        av_opt_set_int       (swr_ctx, "out_sample_rate",    c->sample_rate,    0);
-        av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt",     c->sample_fmt,     0);
-
-        /* initialize the resampling context */
-        if ((ret = swr_init(swr_ctx)) < 0) {
-            fprintf(stderr, "Failed to initialize the resampling context\n");
-            exit(1);
-        }
-
-        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);
-        }
-    } else {
-        dst_samples_data = src_samples_data;
-    }
-    dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, max_dst_nb_samples,
-                                                  c->sample_fmt, 0);
+    if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
+        audio_input_frame_size = 10000;
+    else
+        audio_input_frame_size = c->frame_size;
+    samples = av_malloc(audio_input_frame_size *
+                        av_get_bytes_per_sample(c->sample_fmt) *
+                        c->channels);
 }

 /* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
@@ -255,69 +136,33 @@ static void get_audio_frame(int16_t *samples, int frame_size, int nb_channels)
    }
 }

-static void write_audio_frame(AVFormatContext *oc, AVStream *st, int flush)
+static void write_audio_frame(AVFormatContext *oc, AVStream *st)
 {
    AVCodecContext *c;
    AVPacket pkt = { 0 }; // data and size must be 0;
-    int got_packet, ret, dst_nb_samples;
+    AVFrame *frame = avcodec_alloc_frame();
+    int got_packet;

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

-    if (!flush) {
-        get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels);
+    get_audio_frame(samples, audio_input_frame_size, c->channels);
+    frame->nb_samples = audio_input_frame_size;
+    avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
+                             (uint8_t *)samples,
+                             audio_input_frame_size *
+                             av_get_bytes_per_sample(c->sample_fmt) *
+                             c->channels, 1);

-        /* 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);
-                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);
-            }
-
-            /* 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_nb_samples = src_nb_samples;
-        }
-
-        audio_frame->nb_samples = dst_nb_samples;
-        audio_frame->pts = av_rescale_q(samples_count, (AVRational){1, c->sample_rate}, c->time_base);
-        avcodec_fill_audio_frame(audio_frame, c->channels, c->sample_fmt,
-                                 dst_samples_data[0], dst_samples_size, 0);
-        samples_count += dst_nb_samples;
-    }
-
-    ret = avcodec_encode_audio2(c, &pkt, flush ? NULL : audio_frame, &got_packet);
-    if (ret < 0) {
-        fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
-        exit(1);
-    }
-
-    if (!got_packet) {
-        if (flush)
-            audio_is_eof = 1;
+    avcodec_encode_audio2(c, &pkt, frame, &got_packet);
+    if (!got_packet)
        return;
-    }

-    ret = write_frame(oc, &c->time_base, st, &pkt);
-    if (ret < 0) {
-        fprintf(stderr, "Error while writing audio frame: %s\n",
-                av_err2str(ret));
+    pkt.stream_index = st->index;
+
+    /* Write the compressed frame to the media file. */
+    if (av_interleaved_write_frame(oc, &pkt) != 0) {
+        fprintf(stderr, "Error while writing audio frame\n");
        exit(1);
    }
 }
@@ -325,69 +170,145 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st, int flush)
 static void close_audio(AVFormatContext *oc, AVStream *st)
 {
    avcodec_close(st->codec);
-    if (dst_samples_data != src_samples_data) {
-        av_free(dst_samples_data[0]);
-        av_free(dst_samples_data);
-    }
-    av_free(src_samples_data[0]);
-    av_free(src_samples_data);
-    av_frame_free(&audio_frame);
+
+    av_free(samples);
 }

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

-static AVFrame *frame;
-static AVPicture src_picture, dst_picture;
-static int frame_count;
+static AVFrame *picture, *tmp_picture;
+static uint8_t *video_outbuf;
+static int frame_count, video_outbuf_size;

-static void open_video(AVFormatContext *oc, AVCodec *codec, AVStream *st)
+/* Add a video output stream. */
+static AVStream *add_video_stream(AVFormatContext *oc, enum CodecID codec_id)
 {
-    int ret;
-    AVCodecContext *c = st->codec;
+    AVCodecContext *c;
+    AVStream *st;
+    AVCodec *codec;
+
+    /* find the video encoder */
+    codec = avcodec_find_encoder(codec_id);
+    if (!codec) {
+        fprintf(stderr, "codec not found\n");
+        exit(1);
+    }
+
+    st = avformat_new_stream(oc, codec);
+    if (!st) {
+        fprintf(stderr, "Could not alloc stream\n");
+        exit(1);
+    }
+
+    c = st->codec;
+
+    /* find the video encoder */
+    codec = avcodec_find_encoder(codec_id);
+    if (!codec) {
+        fprintf(stderr, "codec not found\n");
+        exit(1);
+    }
+    avcodec_get_context_defaults3(c, codec);
+
+    c->codec_id = codec_id;
+
+    /* Put sample parameters. */
+    c->bit_rate = 400000;
+    /* Resolution must be a multiple of two. */
+    c->width    = 352;
+    c->height   = 288;
+    /* timebase: This is the fundamental unit of time (in seconds) in terms
+     * of which frame timestamps are represented. For fixed-fps content,
+     * timebase should be 1/framerate and timestamp increments should be
+     * identical to 1. */
+    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 == CODEC_ID_MPEG2VIDEO) {
+        /* just for testing, we also add B frames */
+        c->max_b_frames = 2;
+    }
+    if (c->codec_id == CODEC_ID_MPEG1VIDEO) {
+        /* Needed to avoid using macroblocks in which some coeffs overflow.
+         * This does not happen with normal video, it just happens here as
+         * the motion of the chroma plane does not match the luma plane. */
+        c->mb_decision = 2;
+    }
+    /* Some formats want stream headers to be separate. */
+    if (oc->oformat->flags & AVFMT_GLOBALHEADER)
+        c->flags |= CODEC_FLAG_GLOBAL_HEADER;
+
+    return st;
+}
+
+static AVFrame *alloc_picture(enum PixelFormat pix_fmt, int width, int height)
+{
+    AVFrame *picture;
+    uint8_t *picture_buf;
+    int size;
+
+    picture = avcodec_alloc_frame();
+    if (!picture)
+        return NULL;
+    size        = avpicture_get_size(pix_fmt, width, height);
+    picture_buf = av_malloc(size);
+    if (!picture_buf) {
+        av_free(picture);
+        return NULL;
+    }
+    avpicture_fill((AVPicture *)picture, picture_buf,
+                   pix_fmt, width, height);
+    return picture;
+}
+
+static void open_video(AVFormatContext *oc, AVStream *st)
+{
+    AVCodecContext *c;
+
+    c = st->codec;

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

-    /* allocate and init a re-usable frame */
-    frame = av_frame_alloc();
-    if (!frame) {
-        fprintf(stderr, "Could not allocate video frame\n");
-        exit(1);
+    video_outbuf = NULL;
+    if (!(oc->oformat->flags & AVFMT_RAWPICTURE)) {
+        /* Allocate output buffer. */
+        /* XXX: API change will be done. */
+        /* Buffers passed into lav* can be allocated any way you prefer,
+         * as long as they're aligned enough for the architecture, and
+         * they're freed appropriately (such as using av_free for buffers
+         * allocated with av_malloc). */
+        video_outbuf_size = 200000;
+        video_outbuf      = av_malloc(video_outbuf_size);
    }
-    frame->format = c->pix_fmt;
-    frame->width = c->width;
-    frame->height = c->height;

    /* 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));
+    picture = alloc_picture(c->pix_fmt, c->width, c->height);
+    if (!picture) {
+        fprintf(stderr, "Could not allocate picture\n");
        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. */
-    if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
-        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));
+    tmp_picture = NULL;
+    if (c->pix_fmt != PIX_FMT_YUV420P) {
+        tmp_picture = alloc_picture(PIX_FMT_YUV420P, c->width, c->height);
+        if (!tmp_picture) {
+            fprintf(stderr, "Could not allocate temporary picture\n");
            exit(1);
        }
    }
-
-    /* copy data and linesize picture pointers to frame */
-    *((AVPicture *)frame) = dst_picture;
 }

 /* Prepare a dummy image. */
-static void fill_yuv_image(AVPicture *pict, int frame_index,
+static void fill_yuv_image(AVFrame *pict, int frame_index,
                           int width, int height)
 {
    int x, y, i;
@@ -408,71 +329,81 @@ static void fill_yuv_image(AVPicture *pict, int frame_index,
    }
 }

-static void write_video_frame(AVFormatContext *oc, AVStream *st, int flush)
+static void write_video_frame(AVFormatContext *oc, AVStream *st)
 {
-    int ret;
-    static struct SwsContext *sws_ctx;
-    AVCodecContext *c = st->codec;
+    int out_size, ret;
+    AVCodecContext *c;
+    static struct SwsContext *img_convert_ctx;

-    if (!flush) {
-        if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
+    c = st->codec;
+
+    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 {
+        if (c->pix_fmt != 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) {
+            if (img_convert_ctx == NULL) {
+                img_convert_ctx = sws_getContext(c->width, c->height,
+                                                 PIX_FMT_YUV420P,
+                                                 c->width, c->height,
+                                                 c->pix_fmt,
+                                                 sws_flags, NULL, NULL, NULL);
+                if (img_convert_ctx == NULL) {
                    fprintf(stderr,
-                            "Could not initialize the conversion context\n");
+                            "Cannot 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);
+            fill_yuv_image(tmp_picture, frame_count, c->width, c->height);
+            sws_scale(img_convert_ctx, tmp_picture->data, tmp_picture->linesize,
+                      0, c->height, picture->data, picture->linesize);
        } else {
-            fill_yuv_image(&dst_picture, frame_count, c->width, c->height);
+            fill_yuv_image(picture, frame_count, c->width, c->height);
        }
    }

-    if (oc->oformat->flags & AVFMT_RAWPICTURE && !flush) {
-        /* Raw video case - directly store the picture in the packet */
+    if (oc->oformat->flags & AVFMT_RAWPICTURE) {
+        /* Raw video case - the API will change slightly in the near
+         * future for that. */
        AVPacket pkt;
        av_init_packet(&pkt);

        pkt.flags        |= AV_PKT_FLAG_KEY;
        pkt.stream_index  = st->index;
-        pkt.data          = dst_picture.data[0];
+        pkt.data          = (uint8_t *)picture;
        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 */
-        frame->pts = frame_count;
-        ret = avcodec_encode_video2(c, &pkt, flush ? NULL : frame, &got_packet);
-        if (ret < 0) {
-            fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret));
-            exit(1);
-        }
+        out_size = avcodec_encode_video(c, video_outbuf,
+                                        video_outbuf_size, picture);
        /* If size is zero, it means the image was buffered. */
+        if (out_size > 0) {
+            AVPacket pkt;
+            av_init_packet(&pkt);

-        if (got_packet) {
-            ret = write_frame(oc, &c->time_base, st, &pkt);
+            if (c->coded_frame->pts != AV_NOPTS_VALUE)
+                pkt.pts = av_rescale_q(c->coded_frame->pts,
+                                       c->time_base, st->time_base);
+            if (c->coded_frame->key_frame)
+                pkt.flags |= AV_PKT_FLAG_KEY;
+
+            pkt.stream_index = st->index;
+            pkt.data         = video_outbuf;
+            pkt.size         = out_size;
+
+            /* Write the compressed frame to the media file. */
+            ret = av_interleaved_write_frame(oc, &pkt);
        } else {
-            if (flush)
-                video_is_eof = 1;
            ret = 0;
        }
    }
-
-    if (ret < 0) {
-        fprintf(stderr, "Error while writing video frame: %s\n", av_err2str(ret));
+    if (ret != 0) {
+        fprintf(stderr, "Error while writing video frame\n");
        exit(1);
    }
    frame_count++;
@@ -481,9 +412,13 @@ static void write_video_frame(AVFormatContext *oc, AVStream *st, int flush)
 static void close_video(AVFormatContext *oc, AVStream *st)
 {
    avcodec_close(st->codec);
-    av_free(src_picture.data[0]);
-    av_free(dst_picture.data[0]);
-    av_frame_free(&frame);
+    av_free(picture->data[0]);
+    av_free(picture);
+    if (tmp_picture) {
+        av_free(tmp_picture->data[0]);
+        av_free(tmp_picture);
+    }
+    av_free(video_outbuf);
 }

 /**************************************************************/
@@ -495,9 +430,8 @@ int main(int argc, char **argv)
    AVOutputFormat *fmt;
    AVFormatContext *oc;
    AVStream *audio_st, *video_st;
-    AVCodec *audio_codec, *video_codec;
-    double audio_time, video_time;
-    int flush, ret;
+    double audio_pts, video_pts;
+    int i;

    /* Initialize libavcodec, and register all codecs and formats. */
    av_register_all();
@@ -505,10 +439,8 @@ int main(int argc, char **argv)
    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"
-               "muxes them into a file named output_file.\n"
               "The output format is automatically guessed according to the file extension.\n"
-               "Raw images can also be output by using '%%d' in the filename.\n"
+               "Raw images can also be output by using '%%d' in the filename\n"
               "\n", argv[0]);
        return 1;
    }
@@ -521,65 +453,66 @@ 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)
-        video_st = add_stream(oc, &video_codec, fmt->video_codec);
-    if (fmt->audio_codec != AV_CODEC_ID_NONE)
-        audio_st = add_stream(oc, &audio_codec, fmt->audio_codec);
+    if (fmt->video_codec != CODEC_ID_NONE) {
+        video_st = add_video_stream(oc, fmt->video_codec);
+    }
+    if (fmt->audio_codec != CODEC_ID_NONE) {
+        audio_st = add_audio_stream(oc, 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 (video_st)
-        open_video(oc, video_codec, video_st);
+        open_video(oc, video_st);
    if (audio_st)
-        open_audio(oc, audio_codec, audio_st);
+        open_audio(oc, audio_st);

    av_dump_format(oc, 0, filename, 1);

    /* open the output file, if needed */
    if (!(fmt->flags & AVFMT_NOFILE)) {
-        ret = avio_open(&oc->pb, filename, AVIO_FLAG_WRITE);
-        if (ret < 0) {
-            fprintf(stderr, "Could not open '%s': %s\n", filename,
-                    av_err2str(ret));
+        if (avio_open(&oc->pb, filename, AVIO_FLAG_WRITE) < 0) {
+            fprintf(stderr, "Could not open '%s'\n", filename);
            return 1;
        }
    }

    /* Write the stream header, if any. */
-    ret = avformat_write_header(oc, NULL);
-    if (ret < 0) {
-        fprintf(stderr, "Error occurred when opening output file: %s\n",
-                av_err2str(ret));
-        return 1;
-    }
+    avformat_write_header(oc, NULL);

-    flush = 0;
-    while ((video_st && !video_is_eof) || (audio_st && !audio_is_eof)) {
+    picture->pts = 0;
+    for (;;) {
        /* Compute current audio and video time. */
-        audio_time = (audio_st && !audio_is_eof) ? audio_st->pts.val * av_q2d(audio_st->time_base) : INFINITY;
-        video_time = (video_st && !video_is_eof) ? video_st->pts.val * av_q2d(video_st->time_base) : INFINITY;
+        if (audio_st)
+            audio_pts = (double)audio_st->pts.val * audio_st->time_base.num / audio_st->time_base.den;
+        else
+            audio_pts = 0.0;

-        if (!flush &&
-            (!audio_st || audio_time >= STREAM_DURATION) &&
-            (!video_st || video_time >= STREAM_DURATION)) {
-            flush = 1;
-        }
+        if (video_st)
+            video_pts = (double)video_st->pts.val * video_st->time_base.num /
+                        video_st->time_base.den;
+        else
+            video_pts = 0.0;
+
+        if ((!audio_st || audio_pts >= STREAM_DURATION) &&
+            (!video_st || video_pts >= STREAM_DURATION))
+            break;

        /* write interleaved audio and video frames */
-        if (audio_st && !audio_is_eof && audio_time <= video_time) {
-            write_audio_frame(oc, audio_st, flush);
-        } else if (video_st && !video_is_eof && video_time < audio_time) {
-            write_video_frame(oc, video_st, flush);
+        if (!video_st || (video_st && audio_st && audio_pts < video_pts)) {
+            write_audio_frame(oc, audio_st);
+        } else {
+            write_video_frame(oc, video_st);
+            picture->pts++;
        }
    }

@@ -595,12 +528,18 @@ int main(int argc, char **argv)
    if (audio_st)
        close_audio(oc, audio_st);

+    /* Free the streams. */
+    for (i = 0; i < oc->nb_streams; i++) {
+        av_freep(&oc->streams[i]->codec);
+        av_freep(&oc->streams[i]);
+    }
+
    if (!(fmt->flags & AVFMT_NOFILE))
        /* Close the output file. */
        avio_close(oc->pb);

    /* free the stream */
-    avformat_free_context(oc);
+    av_free(oc);

    return 0;
 }
--- a/doc/examples/remuxing.c
+++ b/doc/examples/remuxing.c
@@ -1,164 +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;
-        }
-        if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
-            out_stream->codec->flags |= 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_free_packet(&pkt);
-    }
-
-    av_write_trailer(ofmt_ctx);
-end:
-
-    avformat_close_input(&ifmt_ctx);
-
-    /* close output */
-    if (ofmt_ctx && !(ofmt->flags & AVFMT_NOFILE))
-        avio_close(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
@@ -1,215 +0,0 @@
-/*
- * Copyright (c) 2012 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.
- */
-
-/**
- * @example resampling_audio.c
- * libswresample API use example.
- */
-
-#include <libavutil/opt.h>
-#include <libavutil/channel_layout.h>
-#include <libavutil/samplefmt.h>
-#include <libswresample/swresample.h>
-
-static int get_format_from_sample_fmt(const char **fmt,
-                                      enum AVSampleFormat sample_fmt)
-{
-    int i;
-    struct sample_fmt_entry {
-        enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
-    } sample_fmt_entries[] = {
-        { AV_SAMPLE_FMT_U8,  "u8",    "u8"    },
-        { AV_SAMPLE_FMT_S16, "s16be", "s16le" },
-        { AV_SAMPLE_FMT_S32, "s32be", "s32le" },
-        { AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
-        { AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
-    };
-    *fmt = NULL;
-
-    for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
-        struct sample_fmt_entry *entry = &sample_fmt_entries[i];
-        if (sample_fmt == entry->sample_fmt) {
-            *fmt = AV_NE(entry->fmt_be, entry->fmt_le);
-            return 0;
-        }
-    }
-
-    fprintf(stderr,
-            "Sample format %s not supported as output format\n",
-            av_get_sample_fmt_name(sample_fmt));
-    return AVERROR(EINVAL);
-}
-
-/**
- * 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)
-{
-    int i, j;
-    double tincr = 1.0 / sample_rate, *dstp = dst;
-    const double c = 2 * M_PI * 440.0;
-
-    /* generate sin tone with 440Hz frequency and duplicated channels */
-    for (i = 0; i < nb_samples; i++) {
-        *dstp = sin(c * *t);
-        for (j = 1; j < nb_channels; j++)
-            dstp[j] = dstp[0];
-        dstp += nb_channels;
-        *t += tincr;
-    }
-}
-
-int main(int argc, char **argv)
-{
-    int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND;
-    int src_rate = 48000, dst_rate = 44100;
-    uint8_t **src_data = NULL, **dst_data = NULL;
-    int src_nb_channels = 0, dst_nb_channels = 0;
-    int src_linesize, dst_linesize;
-    int src_nb_samples = 1024, dst_nb_samples, max_dst_nb_samples;
-    enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_DBL, dst_sample_fmt = AV_SAMPLE_FMT_S16;
-    const char *dst_filename = NULL;
-    FILE *dst_file;
-    int dst_bufsize;
-    const char *fmt;
-    struct SwrContext *swr_ctx;
-    double t;
-    int ret;
-
-    if (argc != 2) {
-        fprintf(stderr, "Usage: %s output_file\n"
-                "API example program to show how to resample an audio stream with libswresample.\n"
-                "This program generates a series of audio frames, resamples them to a specified "
-                "output format and rate and saves them to an output file named output_file.\n",
-            argv[0]);
-        exit(1);
-    }
-    dst_filename = argv[1];
-
-    dst_file = fopen(dst_filename, "wb");
-    if (!dst_file) {
-        fprintf(stderr, "Could not open destination file %s\n", dst_filename);
-        exit(1);
-    }
-
-    /* create resampler context */
-    swr_ctx = swr_alloc();
-    if (!swr_ctx) {
-        fprintf(stderr, "Could not allocate resampler context\n");
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-
-    /* set options */
-    av_opt_set_int(swr_ctx, "in_channel_layout",    src_ch_layout, 0);
-    av_opt_set_int(swr_ctx, "in_sample_rate",       src_rate, 0);
-    av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);
-
-    av_opt_set_int(swr_ctx, "out_channel_layout",    dst_ch_layout, 0);
-    av_opt_set_int(swr_ctx, "out_sample_rate",       dst_rate, 0);
-    av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);
-
-    /* initialize the resampling context */
-    if ((ret = swr_init(swr_ctx)) < 0) {
-        fprintf(stderr, "Failed to initialize the resampling context\n");
-        goto end;
-    }
-
-    /* allocate source and destination samples buffers */
-
-    src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
-    ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_nb_channels,
-                                             src_nb_samples, src_sample_fmt, 0);
-    if (ret < 0) {
-        fprintf(stderr, "Could not allocate source samples\n");
-        goto end;
-    }
-
-    /* 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 = dst_nb_samples =
-        av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
-
-    /* buffer is going to be directly written to a rawaudio file, no alignment */
-    dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
-    ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels,
-                                             dst_nb_samples, dst_sample_fmt, 0);
-    if (ret < 0) {
-        fprintf(stderr, "Could not allocate destination samples\n");
-        goto end;
-    }
-
-    t = 0;
-    do {
-        /* generate synthetic audio */
-        fill_samples((double *)src_data[0], src_nb_samples, src_nb_channels, src_rate, &t);
-
-        /* compute destination number of samples */
-        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_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)
-                break;
-            max_dst_nb_samples = dst_nb_samples;
-        }
-
-        /* convert to destination format */
-        ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)src_data, src_nb_samples);
-        if (ret < 0) {
-            fprintf(stderr, "Error while converting\n");
-            goto end;
-        }
-        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);
-
-    if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0)
-        goto end;
-    fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"
-            "ffplay -f %s -channel_layout %"PRId64" -channels %d -ar %d %s\n",
-            fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);
-
-end:
-    if (dst_file)
-        fclose(dst_file);
-
-    if (src_data)
-        av_freep(&src_data[0]);
-    av_freep(&src_data);
-
-    if (dst_data)
-        av_freep(&dst_data[0]);
-    av_freep(&dst_data);
-
-    swr_free(&swr_ctx);
-    return ret < 0;
-}
--- a/doc/examples/scaling_video.c
+++ b/doc/examples/scaling_video.c
@@ -1,141 +0,0 @@
-/*
- * Copyright (c) 2012 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
- * libswscale API use example.
- * @example scaling_video.c
- */
-
-#include <libavutil/imgutils.h>
-#include <libavutil/parseutils.h>
-#include <libswscale/swscale.h>
-
-static void fill_yuv_image(uint8_t *data[4], int linesize[4],
-                           int width, int height, int frame_index)
-{
-    int x, y;
-
-    /* Y */
-    for (y = 0; y < height; y++)
-        for (x = 0; x < width; x++)
-            data[0][y * linesize[0] + x] = x + y + frame_index * 3;
-
-    /* Cb and Cr */
-    for (y = 0; y < height / 2; y++) {
-        for (x = 0; x < width / 2; x++) {
-            data[1][y * linesize[1] + x] = 128 + y + frame_index * 2;
-            data[2][y * linesize[2] + x] = 64 + x + frame_index * 5;
-        }
-    }
-}
-
-int main(int argc, char **argv)
-{
-    uint8_t *src_data[4], *dst_data[4];
-    int src_linesize[4], dst_linesize[4];
-    int src_w = 320, src_h = 240, dst_w, dst_h;
-    enum AVPixelFormat src_pix_fmt = AV_PIX_FMT_YUV420P, dst_pix_fmt = AV_PIX_FMT_RGB24;
-    const char *dst_size = NULL;
-    const char *dst_filename = NULL;
-    FILE *dst_file;
-    int dst_bufsize;
-    struct SwsContext *sws_ctx;
-    int i, ret;
-
-    if (argc != 3) {
-        fprintf(stderr, "Usage: %s output_file output_size\n"
-                "API example program to show how to scale an image with libswscale.\n"
-                "This program generates a series of pictures, rescales them to the given "
-                "output_size and saves them to an output file named output_file\n."
-                "\n", argv[0]);
-        exit(1);
-    }
-    dst_filename = argv[1];
-    dst_size     = argv[2];
-
-    if (av_parse_video_size(&dst_w, &dst_h, dst_size) < 0) {
-        fprintf(stderr,
-                "Invalid size '%s', must be in the form WxH or a valid size abbreviation\n",
-                dst_size);
-        exit(1);
-    }
-
-    dst_file = fopen(dst_filename, "wb");
-    if (!dst_file) {
-        fprintf(stderr, "Could not open destination file %s\n", dst_filename);
-        exit(1);
-    }
-
-    /* create scaling context */
-    sws_ctx = sws_getContext(src_w, src_h, src_pix_fmt,
-                             dst_w, dst_h, dst_pix_fmt,
-                             SWS_BILINEAR, NULL, NULL, NULL);
-    if (!sws_ctx) {
-        fprintf(stderr,
-                "Impossible to create scale context for the conversion "
-                "fmt:%s s:%dx%d -> fmt:%s s:%dx%d\n",
-                av_get_pix_fmt_name(src_pix_fmt), src_w, src_h,
-                av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h);
-        ret = AVERROR(EINVAL);
-        goto end;
-    }
-
-    /* allocate source and destination image buffers */
-    if ((ret = av_image_alloc(src_data, src_linesize,
-                              src_w, src_h, src_pix_fmt, 16)) < 0) {
-        fprintf(stderr, "Could not allocate source image\n");
-        goto end;
-    }
-
-    /* buffer is going to be written to rawvideo file, no alignment */
-    if ((ret = av_image_alloc(dst_data, dst_linesize,
-                              dst_w, dst_h, dst_pix_fmt, 1)) < 0) {
-        fprintf(stderr, "Could not allocate destination image\n");
-        goto end;
-    }
-    dst_bufsize = ret;
-
-    for (i = 0; i < 100; i++) {
-        /* generate synthetic video */
-        fill_yuv_image(src_data, src_linesize, src_w, src_h, i);
-
-        /* convert to destination format */
-        sws_scale(sws_ctx, (const uint8_t * const*)src_data,
-                  src_linesize, 0, src_h, dst_data, dst_linesize);
-
-        /* write scaled image to file */
-        fwrite(dst_data[0], 1, dst_bufsize, dst_file);
-    }
-
-    fprintf(stderr, "Scaling succeeded. Play the output file with the command:\n"
-           "ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
-           av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h, dst_filename);
-
-end:
-    if (dst_file)
-        fclose(dst_file);
-    av_freep(&src_data[0]);
-    av_freep(&dst_data[0]);
-    sws_freeContext(sws_ctx);
-    return ret < 0;
-}
--- a/doc/examples/transcode_aac.c
+++ b/doc/examples/transcode_aac.c
@@ -1,755 +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 48000
-/** The number of output channels */
-#define OUTPUT_CHANNELS 2
-/** The audio sample output format */
-#define OUTPUT_SAMPLE_FORMAT AV_SAMPLE_FMT_S16
-
-/**
- * Convert an error code into a text message.
- * @param error Error code to be converted
- * @return Corresponding error text (not thread-safe)
- */
-static char *const 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 easiert 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     = AV_SAMPLE_FMT_S16;
-    (*output_codec_context)->bit_rate       = OUTPUT_BIT_RATE;
-
-    /**
-     * 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 |= 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_close((*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)
-{
-    /** Create the FIFO buffer based on the specified output sample format. */
-    if (!(*fifo = av_audio_fifo_alloc(OUTPUT_SAMPLE_FORMAT, OUTPUT_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 the 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_free_packet(&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_free_packet(&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;
-}
-
-/** 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);
-
-    /**
-     * 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_free_packet(&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_free_packet(&output_packet);
-            return error;
-        }
-
-        av_free_packet(&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))
-        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_close(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/faq.texi
+++ b/doc/faq.texi
@@ -79,17 +79,6 @@ not a bug they should fix:
 Then again, some of them do not know the difference between an undecidable
 problem and an NP-hard problem...

-@section I have installed this library with my distro's package manager. Why does @command{configure} not see it?
-
-Distributions usually split libraries in several packages. The main package
-contains the files necessary to run programs using the library. The
-development package contains the files necessary to build programs using the
-library. Sometimes, docs and/or data are in a separate package too.
-
-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.
-
@chapter Usage

@section ffmpeg does not work; what is wrong?
@@ -105,21 +94,12 @@ 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.

-@file{img%03d.jpg} means the sequence @file{img001.jpg}, @file{img002.jpg}, etc.
-
-Use the @option{-start_number} option to declare a starting number for
-the sequence. This is useful if your sequence does not start with
-@file{img001.jpg} but is still in a numerical order. The following
-example will start with @file{img100.jpg}:
-
-@example
-ffmpeg -f image2 -start_number 100 -i img%d.jpg /tmp/a.mpg
-@end example
+@file{img%03d.jpg} means the sequence @file{img001.jpg}, @file{img002.jpg}, etc...

 If you have large number of pictures to rename, you can use the
 following command to ease the burden. The command, using the bourne
@@ -128,7 +108,7 @@ 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
@@ -137,23 +117,17 @@ 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.

-You can also use @command{cat} to pipe images to ffmpeg:
-
-@example
-cat *.jpg | ffmpeg -f image2pipe -c:v mjpeg -i - output.mpg
-@end example
-
@section How do I encode movie to single pictures?

 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
@@ -169,7 +143,7 @@ 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.
@@ -227,79 +201,20 @@ 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?

-To "join" video files is quite ambiguous. The following list explains the
-different kinds of "joining" and points out how those are addressed in
-FFmpeg. To join video files may mean:
-
-@itemize
-
-@item
-To put them one after the other: this is called to @emph{concatenate} them
-(in short: concat) and is addressed
-@ref{How can I concatenate video files, in this very faq}.
-
-@item
-To put them together in the same file, to let the user choose between the
-different versions (example: different audio languages): this is called to
-@emph{multiplex} them together (in short: mux), and is done by simply
-invoking ffmpeg with several @option{-i} options.
-
-@item
-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{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{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{http://ffmpeg.org/ffmpeg-filters.html#overlay, @code{overlay}} video filter.
-
-@end itemize
-
-@anchor{How can I concatenate video files}
-@section How can I concatenate video files?
-
-There are several solutions, depending on the exact circumstances.
-
-@subsection Concatenating using the concat @emph{filter}
-
-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{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{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 to concatenate
-video by merely concatenating the files containing them.
+A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow to join video files by
+merely concatenating them.

 Hence you may concatenate your multimedia files by first transcoding them to
 these privileged formats, then using the humble @code{cat} command (or the
@@ -307,38 +222,28 @@ equally humble @code{copy} under Windows), and finally transcoding back to your
 format of choice.

@example
-ffmpeg -i input1.avi -qscale:v 1 intermediate1.mpg
-ffmpeg -i input2.avi -qscale:v 1 intermediate2.mpg
+ffmpeg -i input1.avi -same_quant intermediate1.mpg
+ffmpeg -i input2.avi -same_quant intermediate2.mpg
 cat intermediate1.mpg intermediate2.mpg > intermediate_all.mpg
-ffmpeg -i intermediate_all.mpg -qscale:v 2 output.avi
+ffmpeg -i intermediate_all.mpg -same_quant output.avi
@end example

-Additionally, you can use the @code{concat} protocol instead of @code{cat} or
-@code{copy} which will avoid creation of a potentially huge intermediate file.
+Notice that you should either use @code{-same_quant} or set a reasonably high
+bitrate for your intermediate and output files, if you want to preserve
+video quality.

-@example
-ffmpeg -i input1.avi -qscale:v 1 intermediate1.mpg
-ffmpeg -i input2.avi -qscale:v 1 intermediate2.mpg
-ffmpeg -i concat:"intermediate1.mpg|intermediate2.mpg" -c copy intermediate_all.mpg
-ffmpeg -i intermediate_all.mpg -qscale:v 2 output.avi
-@end example
-
-Note that you may need to escape the character "|" which is special for many
-shells.
-
-Another option is usage of named pipes, should your platform support it:
+Also notice that you may avoid the huge intermediate files by taking advantage
+of named pipes, should your platform support it:

@example
 mkfifo intermediate1.mpg
 mkfifo intermediate2.mpg
-ffmpeg -i input1.avi -qscale:v 1 -y intermediate1.mpg < /dev/null &
-ffmpeg -i input2.avi -qscale:v 1 -y intermediate2.mpg < /dev/null &
+ffmpeg -i input1.avi -same_quant -y intermediate1.mpg < /dev/null &
+ffmpeg -i input2.avi -same_quant -y intermediate2.mpg < /dev/null &
 cat intermediate1.mpg intermediate2.mpg |\
-ffmpeg -f mpeg -i - -c:v mpeg4 -acodec libmp3lame output.avi
+ffmpeg -f mpeg -i - -same_quant -c:v mpeg4 -acodec libmp3lame output.avi
@end example

-@subsection Concatenating using raw audio and video
-
 Similarly, the yuv4mpegpipe format, and the raw video, raw audio codecs also
 allow concatenation, and the transcoding step is almost lossless.
 When using multiple yuv4mpegpipe(s), the first line needs to be discarded
@@ -346,8 +251,7 @@ from all but the first stream. This can be accomplished by piping through
@code{tail} as seen below. Note that when piping through @code{tail} you
 must use command grouping, @code{@{  ;@}}, to background properly.

-For example, let's say we want to concatenate two FLV files into an
-output.flv file:
+For example, let's say we want to join two FLV files into an output.flv file:

@example
 mkfifo temp1.a
@@ -364,70 +268,53 @@ cat temp1.a temp2.a > all.a &
 cat temp1.v temp2.v > all.v &
 ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 44100 -i all.a \
       -f yuv4mpegpipe -i all.v \
-       -y output.flv
+       -same_quant -y output.flv
 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
 lost.

-Most likely, it is through @code{auto-inserted aresample}. Try to understand
+Most likely, it is through @code{auto-inserted aconvert}. Try to understand
 why the converting filter was needed at that place.

 Just before the output is a likely place, as @option{-f lavfi} currently
 only support packed S16.

-Then insert the correct @code{aformat} explicitly in the filtergraph,
+Then insert the correct @code{aconvert} explicitly in the filter graph,
 specifying the exact format.

@example
-aformat=sample_fmts=s16:channel_layouts=stereo
+aconvert=s16:stereo:packed
@end example

-@section Why does FFmpeg not see the subtitles in my VOB file?
-
-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 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
-are detected later are ignored.
-
-The size of the initial scan is controlled by two options: @code{probesize}
-(default ~5 Mo) and @code{analyzeduration} (default 5,000,000 µs = 5 s). For
-the subtitle stream to be detected, both values must be large enough.
-
-@section Why was the @command{ffmpeg} @option{-sameq} option removed? What to use instead?
-
-The @option{-sameq} option meant "same quantizer", and made sense only in a
-very limited set of cases. Unfortunately, a lot of people mistook it for
-"same quality" and used it in places where it did not make sense: it had
-roughly the expected visible effect, but achieved it in a very inefficient
-way.
-
-Each encoder has its own set of options to set the quality-vs-size balance,
-use the options for the encoder you are using to set the quality level to a
-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.
-
@chapter Development

@section Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat?

-Yes. Check the @file{doc/examples} directory in the source
-repository, also available online at:
-@url{https://github.com/FFmpeg/FFmpeg/tree/master/doc/examples}.
-
-Examples are also installed by default, usually in
-@code{$PREFIX/share/ffmpeg/examples}.
-
-Also you may read the Developers Guide of the FFmpeg documentation. Alternatively,
+Yes. Read the Developers Guide of the FFmpeg documentation. Alternatively,
 examine the source code for one of the many open source projects that
 already incorporate FFmpeg at (@url{projects.html}).

@@ -439,8 +326,31 @@ with @code{#ifdef}s related to the compiler.

@section Is Microsoft Visual C++ supported?

-Yes. Please see the @uref{platform.html, Microsoft Visual C++}
-section in the FFmpeg documentation.
+No. Microsoft Visual C++ is not compliant to the C99 standard and does
+not - among other things - support the inline assembly used in FFmpeg.
+If you wish to use MSVC++ for your
+project then you can link the MSVC++ code with libav* as long as
+you compile the latter with a working C compiler. For more information, see
+the @emph{Microsoft Visual C++ compatibility} section in the FFmpeg
+documentation.
+
+There have been efforts to make FFmpeg compatible with MSVC++ in the
+past. However, they have all been rejected as too intrusive, especially
+since MinGW does the job adequately. None of the core developers
+work with MSVC++ and thus this item is low priority. Should you find
+the silver bullet that solves this problem, feel free to shoot it at us.
+
+We strongly recommend you to move over from MSVC++ to MinGW tools.
+
+@section Can I use FFmpeg or libavcodec under Windows?
+
+Yes, but the Cygwin or MinGW tools @emph{must} be used to compile FFmpeg.
+Read the @emph{Windows} section in the FFmpeg documentation to find more
+information.
+
+To get help and instructions for building FFmpeg under Windows, check out
+the FFmpeg Windows Help Forum at
+@url{http://ffmpeg.arrozcru.org/}.

@section Can you add automake, libtool or autoconf support?

@@ -455,10 +365,9 @@ 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?

@@ -466,24 +375,6 @@ Yes, as long as the code is optional and can easily and cleanly be placed
 under #if CONFIG_GPL without breaking anything. So, for example, a new codec
 or filter would be OK under GPL while a bug fix to LGPL code would not.

-@section I'm using FFmpeg from within my C application but the linker complains about missing symbols from the libraries themselves.
-
-FFmpeg builds static libraries by default. In static libraries, dependencies
-are not handled. That has two consequences. First, you must specify the
-libraries in dependency order: @code{-lavdevice} must come before
-@code{-lavformat}, @code{-lavutil} must come after everything else, etc.
-Second, external libraries that are used in FFmpeg have to be specified too.
-
-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)
-@end example
-
-See @file{doc/example/Makefile} and @file{doc/example/pc-uninstalled} for
-more details.
-
@section I'm using FFmpeg from within my C++ application but the linker complains about missing symbols which seem to be available.

 FFmpeg is a pure C project, so to use the libraries within your C++ application
@@ -499,8 +390,12 @@ to use them you have to append -D__STDC_CONSTANT_MACROS to your CXXFLAGS

@section I have a file in memory / a API different from *open/*read/ libc how do I use it with libavformat?

-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.
+You have to implement a URLProtocol, see @file{libavformat/file.c} in
+FFmpeg and @file{libmpdemux/demux_lavf.c} in MPlayer 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?

@@ -514,12 +409,11 @@ 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
@@ -1,8 +1,8 @@
 \input texinfo @c -*- texinfo -*-

-@settitle FFmpeg Automated Testing Environment
+@settitle FATE Automated Testing Environment
@titlepage
-@center @titlefont{FFmpeg Automated Testing Environment}
+@center @titlefont{FATE Automated Testing Environment}
@end titlepage

@node Top
@@ -27,7 +27,7 @@ by visiting this website:

  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
+with there 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
@@ -78,14 +78,11 @@ Do not put a '~' character in the samples path to indicate a home
 directory. Because of shell nuances, this will cause FATE to fail.
@end float

-To use a custom wrapper to run the test, pass @option{--target-exec} to
-@command{configure} or set the @var{TARGET_EXEC} Make variable.
-

@chapter Submitting the results to the FFmpeg result aggregation server

  To submit your results to the server you should run fate through the
-shell script @file{tests/fate.sh} from the FFmpeg sources. This script needs
+shell script tests/fate.sh from the FFmpeg sources. This script needs
 to be invoked with a configuration file as its first argument.

@example
@@ -93,11 +90,11 @@ tests/fate.sh /path/to/fate_config
@end example

  A configuration file template with comments describing the individual
-configuration variables can be found at @file{doc/fate_config.sh.template}.
+configuration variables can be found at @file{tests/fate_config.sh.template}.

@ifhtml
  The mentioned configuration template is also available here:
-@verbatiminclude fate_config.sh.template
+@verbatiminclude ../tests/fate_config.sh.template
@end ifhtml

  Create a configuration that suits your needs, based on the configuration
@@ -121,9 +118,8 @@ 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
-and send the public key to the FATE server administrator who can be contacted
-at the email address @email{fate-admin@@ffmpeg.org}.
+  When you have everything working properly you can create an SSH key and
+send its public part to the FATE server administrator.

  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
@@ -131,17 +127,7 @@ 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 @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
-the @command{ssh} command with one or more @option{-v} options. You should
-get detailed output concerning your SSH configuration and the authentication
-process.
+  b1:31:c8:79:3f:04:1d:f8:f2:23:26:5a:fd:55:fa:92

  The only thing left is to automate the execution of the fate.sh script and
 the synchronisation of the samples directory.
@@ -153,20 +139,20 @@ 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 @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
@@ -174,32 +160,17 @@ 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.
-
-@item THREAD_TYPE
-Specify which threading strategy test, either @var{slice} or @var{frame},
-by default @var{slice+frame}
-
+    Specify how many threads to use while running regression tests, it is
+    quite useful to detect thread-related regressions.
@item CPUFLAGS
-Specify CPU flags.
-
-@item TARGET_EXEC
-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 @var{1} to generate the missing or mismatched references.
+    Specify CPU flags.
@end table

-@section Examples
-
+Example:
@example
 make V=1 SAMPLES=/var/fate/samples THREADS=2 CPUFLAGS=mmx fate
@end example
--- a/doc/ffmpeg-bitstream-filters.texi
+++ b/doc/ffmpeg-bitstream-filters.texi
@@ -1,45 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle FFmpeg Bitstream Filters Documentation
-@titlepage
-@center @titlefont{FFmpeg Bitstream Filters Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-This document describes the bitstream filters provided by the
-libavcodec library.
-
-A bitstream filter operates on the encoded stream data, and performs
-bitstream level modifications without performing decoding.
-
-@c man end DESCRIPTION
-
-@include bitstream_filters.texi
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{libavcodec.html,libavcodec}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavcodec(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename ffmpeg-bitstream-filters
-@settitle FFmpeg bitstream filters
-
-@end ignore
-
-@bye
--- a/doc/ffmpeg-codecs.texi
+++ b/doc/ffmpeg-codecs.texi
@@ -1,42 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle FFmpeg Codecs Documentation
-@titlepage
-@center @titlefont{FFmpeg Codecs Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-This document describes the codecs (decoders and encoders) provided by
-the libavcodec library.
-
-@c man end DESCRIPTION
-
-@include codecs.texi
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{libavcodec.html,libavcodec}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavcodec(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename ffmpeg-codecs
-@settitle FFmpeg codecs
-
-@end ignore
-
-@bye
--- a/doc/ffmpeg-devices.texi
+++ b/doc/ffmpeg-devices.texi
@@ -1,42 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle FFmpeg Devices Documentation
-@titlepage
-@center @titlefont{FFmpeg Devices Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-This document describes the input and output devices provided by the
-libavdevice library.
-
-@c man end DESCRIPTION
-
-@include devices.texi
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{libavdevice.html,libavdevice}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavdevice(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename ffmpeg-devices
-@settitle FFmpeg devices
-
-@end ignore
-
-@bye
--- a/doc/ffmpeg-filters.texi
+++ b/doc/ffmpeg-filters.texi
@@ -1,42 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle FFmpeg Filters Documentation
-@titlepage
-@center @titlefont{FFmpeg Filters Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-This document describes filters, sources, and sinks provided by the
-libavfilter library.
-
-@c man end DESCRIPTION
-
-@include filters.texi
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{libavfilter.html,libavfilter}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavfilter(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename ffmpeg-filters
-@settitle FFmpeg filters
-
-@end ignore
-
-@bye
--- a/doc/ffmpeg-formats.texi
+++ b/doc/ffmpeg-formats.texi
@@ -1,42 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle FFmpeg Formats Documentation
-@titlepage
-@center @titlefont{FFmpeg Formats Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-This document describes the supported formats (muxers and demuxers)
-provided by the libavformat library.
-
-@c man end DESCRIPTION
-
-@include formats.texi
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{libavformat.html,libavformat}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavformat(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename ffmpeg-formats
-@settitle FFmpeg formats
-
-@end ignore
-
-@bye
--- a/doc/ffmpeg-protocols.texi
+++ b/doc/ffmpeg-protocols.texi
@@ -1,42 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle FFmpeg Protocols Documentation
-@titlepage
-@center @titlefont{FFmpeg Protocols Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-This document describes the input and output protocols provided by the
-libavformat library.
-
-@c man end DESCRIPTION
-
-@include protocols.texi
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{libavformat.html,libavformat}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavformat(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename ffmpeg-protocols
-@settitle FFmpeg protocols
-
-@end ignore
-
-@bye
--- a/doc/ffmpeg-resampler.texi
+++ b/doc/ffmpeg-resampler.texi
@@ -1,44 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle FFmpeg Resampler Documentation
-@titlepage
-@center @titlefont{FFmpeg Resampler Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-The FFmpeg resampler provides a high-level interface to the
-libswresample library audio resampling utilities. In particular it
-allows to perform audio resampling, audio channel layout rematrixing,
-and convert audio format and packing layout.
-
-@c man end DESCRIPTION
-
-@include resampler.texi
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{libswresample.html,libswresample}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libswresample(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename ffmpeg-resampler
-@settitle FFmpeg Resampler
-
-@end ignore
-
-@bye
--- a/doc/ffmpeg-scaler.texi
+++ b/doc/ffmpeg-scaler.texi
@@ -1,43 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle FFmpeg Scaler Documentation
-@titlepage
-@center @titlefont{FFmpeg Scaler Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@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 to perform
-image rescaling and pixel format conversion.
-
-@c man end DESCRIPTION
-
-@include scaler.texi
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{libswscale.html,libswscale}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libswscale(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename ffmpeg-scaler
-@settitle FFmpeg video scaling and pixel format converter
-
-@end ignore
-
-@bye
--- a/doc/ffmpeg-utils.texi
+++ b/doc/ffmpeg-utils.texi
@@ -1,42 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle FFmpeg Utilities Documentation
-@titlepage
-@center @titlefont{FFmpeg Utilities Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-This document describes some generic features and utilities provided
-by the libavutil library.
-
-@c man end DESCRIPTION
-
-@include utils.texi
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{libavutil.html,libavutil}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavutil(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename ffmpeg-utils
-@settitle FFmpeg utilities
-
-@end ignore
-
-@bye
--- a/doc/ffmpeg.texi
+++ b/doc/ffmpeg.texi
--- a/doc/ffplay.texi
+++ b/doc/ffplay.texi
@@ -11,7 +11,11 @@

@chapter Synopsis

-ffplay [@var{options}] [@file{input_file}]
+@example
+@c man begin SYNOPSIS
+ffplay [options] [@file{input_file}]
+@c man end
+@end example

@chapter Description
@c man begin DESCRIPTION
@@ -24,7 +28,7 @@ various FFmpeg APIs.
@chapter Options
@c man begin OPTIONS

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

@section Main options

@@ -73,22 +77,11 @@ Default value is "video", if video is not present or cannot be played
 You can interactively cycle through the available show modes by
 pressing the key @key{w}.

-@item -vf @var{filtergraph}
-Create the filtergraph specified by @var{filtergraph} and use it to
-filter the video stream.
-
-@var{filtergraph} is a description of the filtergraph to apply to
-the stream, and must have a single video input and a single video
-output. In the filtergraph, the input is associated to the label
-@code{in}, and the output to the label @code{out}. See the
-ffmpeg-filters manual for more information about the filtergraph
-syntax.
-
-@item -af @var{filtergraph}
-@var{filtergraph} is a description of the filtergraph to apply to
-the input audio.
+@item -vf @var{filter_graph}
+@var{filter_graph} is a description of the filter graph to apply to
+the input video.
 Use the option "-filters" to show all the available filters (including
-sources and sinks).
+also sources and sinks).

@item -i @var{input_file}
 Read @var{input_file}.
@@ -99,13 +92,9 @@ Read @var{input_file}.
@item -pix_fmt @var{format}
 Set pixel format.
 This option has been deprecated in favor of private options, try -pixel_format.
-
@item -stats
-Print several playback statistics, in particular show the stream
-duration, the codec parameters, the current position in the stream and
-the audio/video synchronisation drift. It is on by default, to
-explicitly disable it you need to specify @code{-nostats}.
-
+Show the stream duration, the codec parameters, the current position in
+the stream and the audio/video synchronisation drift.
@item -bug
 Work around bugs.
@item -fast
@@ -145,20 +134,8 @@ Exit when video is done playing.
 Exit if any key is pressed.
@item -exitonmousedown
 Exit if any mouse button is pressed.
-
-@item -codec:@var{media_specifier} @var{codec_name}
-Force a specific decoder implementation for the stream identified by
-@var{media_specifier}, which can assume the values @code{a} (audio),
-@code{v} (video), and @code{s} subtitle.
-
-@item -acodec @var{codec_name}
-Force a specific audio decoder.
-
-@item -vcodec @var{codec_name}
-Force a specific video decoder.
-
-@item -scodec @var{codec_name}
-Force a specific subtitle decoder.
+@item -codec:@var{stream_type}
+Force a specific decoder implementation
@end table

@section While playing
@@ -174,26 +151,17 @@ Toggle full screen.
 Pause.

@item a
-Cycle audio channel in the curret 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
 Show audio waves.

-@item s
-Step to the next frame.
-
-Pause if the stream is not already paused, step to the next video
-frame, and pause.
-
@item left/right
 Seek backward/forward 10 seconds.

@@ -201,8 +169,6 @@ 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 mouse click
@@ -212,75 +178,29 @@ Seek to percentage in file corresponding to fraction of width.

@c man end

-@include config.texi
-@ifset config-all
-@set config-readonly
-@ifset config-avutil
-@include utils.texi
-@end ifset
-@ifset config-avcodec
-@include codecs.texi
-@include bitstream_filters.texi
-@end ifset
-@ifset config-avformat
-@include formats.texi
+@include syntax.texi
+@include eval.texi
+@include decoders.texi
+@include demuxers.texi
+@include muxers.texi
+@include indevs.texi
+@include outdevs.texi
@include protocols.texi
-@end ifset
-@ifset config-avdevice
-@include devices.texi
-@end ifset
-@ifset config-swresample
-@include resampler.texi
-@end ifset
-@ifset config-swscale
-@include scaler.texi
-@end ifset
-@ifset config-avfilter
@include filters.texi
-@end ifset
-@end ifset
-
-@chapter See Also
-
-@ifhtml
-@ifset config-all
-@url{ffplay.html,ffplay},
-@end ifset
-@ifset config-not-all
-@url{ffplay-all.html,ffmpeg-all},
-@end ifset
-@url{ffmpeg.html,ffmpeg}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{ffmpeg-utils.html,ffmpeg-utils},
-@url{ffmpeg-scaler.html,ffmpeg-scaler},
-@url{ffmpeg-resampler.html,ffmpeg-resampler},
-@url{ffmpeg-codecs.html,ffmpeg-codecs},
-@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters},
-@url{ffmpeg-formats.html,ffmpeg-formats},
-@url{ffmpeg-devices.html,ffmpeg-devices},
-@url{ffmpeg-protocols.html,ffmpeg-protocols},
-@url{ffmpeg-filters.html,ffmpeg-filters}
-@end ifhtml
-
-@ifnothtml
-@ifset config-all
-ffplay(1),
-@end ifset
-@ifset config-not-all
-ffplay-all(1),
-@end ifset
-ffmpeg(1), ffprobe(1), ffserver(1),
-ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
-ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
-ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)
-@end ifnothtml
-
-@include authors.texi

@ignore

@setfilename ffplay
@settitle FFplay media player

+@c man begin SEEALSO
+ffmpeg(1), ffprobe(1), ffserver(1) and the FFmpeg HTML documentation
+@c man end
+
+@c man begin AUTHORS
+The FFmpeg developers
+@c man end
+
@end ignore

@bye
--- a/doc/ffprobe.texi
+++ b/doc/ffprobe.texi
@@ -11,7 +11,13 @@

@chapter Synopsis

-ffprobe [@var{options}] [@file{input_file}]
+The generic syntax is:
+
+@example
+@c man begin SYNOPSIS
+ffprobe [options] [@file{input_file}]
+@c man end
+@end example

@chapter Description
@c man begin DESCRIPTION
@@ -39,20 +45,15 @@ ffprobe output is designed to be easily parsable by a textual filter,
 and consists of one or more sections of a form defined by the selected
 writer, which is specified by the @option{print_format} option.

-Sections may contain other nested sections, and are identified by a
-name (which may be shared by other sections), and an unique
-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

@@ -79,7 +80,7 @@ Use sexagesimal format HH:MM:SS.MICROSECONDS for time values.
 Prettify the format of the displayed values, it corresponds to the
 options "-unit -prefix -byte_binary_prefix -sexagesimal".

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

@var{writer_name} specifies the name of the writer, and
@@ -93,32 +94,6 @@ For example for printing the output in JSON format, specify:
 For more details on the available output printing formats, see the
 Writers section below.

-@item -sections
-Print sections structure and section information, and exit. The output
-is not meant to be parsed by a machine.
-
-@item -select_streams @var{stream_specifier}
-Select only the streams specified by @var{stream_specifier}. This
-option affects only the options related to streams
-(e.g. @code{show_streams}, @code{show_packets}, etc.).
-
-For example to show only audio streams, you can use the command:
-@example
-ffprobe -show_streams -select_streams a INPUT
-@end example
-
-To show only video packets belonging to the video stream with index 1:
-@example
-ffprobe -show_packets -select_streams v:1 INPUT
-@end example
-
-@item -show_data
-Show payload data, as a 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_error
 Show information about the error found when trying to probe the input.

@@ -136,59 +111,6 @@ Like @option{-show_format}, but only prints the specified entry of the
 container format information, rather than all. This option may be given more
 than once, then all specified entries will be shown.

-This option is deprecated, use @code{show_entries} instead.
-
-@item -show_entries @var{section_entries}
-Set list of entries to show.
-
-Entries are specified according to the following
-syntax. @var{section_entries} contains a list of section entries
-separated by @code{:}. Each section entry is composed by a section
-name (or unique name), optionally followed by a list of entries local
-to that section, separated by @code{,}.
-
-If section name is specified but is followed by no @code{=}, all
-entries are printed to output, together with all the contained
-sections. Otherwise only the entries specified in the local section
-entries list are printed. In particular, if @code{=} is specified but
-the list of local entries is empty, then no entries will be shown for
-that section.
-
-Note that the order of specification of the local section entries is
-not honored in the output, and the usual display order will be
-retained.
-
-The formal syntax is given by:
-@example
-@var{LOCAL_SECTION_ENTRIES} ::= @var{SECTION_ENTRY_NAME}[,@var{LOCAL_SECTION_ENTRIES}]
-@var{SECTION_ENTRY}         ::= @var{SECTION_NAME}[=[@var{LOCAL_SECTION_ENTRIES}]]
-@var{SECTION_ENTRIES}       ::= @var{SECTION_ENTRY}[:@var{SECTION_ENTRIES}]
-@end example
-
-For example, to show only the index and type of each stream, and the PTS
-time, duration time, and stream index of the packets, you can specify
-the argument:
-@example
-packet=pts_time,duration_time,stream_index : stream=index,codec_type
-@end example
-
-To show all the entries in the section "format", but only the codec
-type in the section "stream", specify the argument:
-@example
-format : stream=codec_type
-@end example
-
-To show all the tags in the stream and format sections:
-@example
-format_tags : format_tags
-@end example
-
-To show only the @code{title} tag (if available) in the stream
-sections:
-@example
-stream_tags=title
-@end example
-
@item -show_packets
 Show information about each packet contained in the input multimedia
 stream.
@@ -197,11 +119,11 @@ 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
@@ -210,18 +132,6 @@ 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.
-
-Each chapter is printed within a dedicated section with name "CHAPTER".
-
@item -count_frames
 Count the number of frames per stream and report it in the
 corresponding stream section.
@@ -230,70 +140,6 @@ 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.
@@ -317,10 +163,6 @@ 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 -bitexact
-Force bitexact output, useful to produce output which is not dependent
-on the specific build.
-
@item -i @var{input_file}
 Read @var{input_file}.

@@ -333,42 +175,8 @@ Read @var{input_file}.
 A writer defines the output format adopted by @command{ffprobe}, and will be
 used for printing all the parts of the output.

-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 writer may accept one or more arguments, which specify the options to
+adopt.

 A description of the currently available writers follows.

@@ -384,8 +192,11 @@ 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:".
+
+This writer accepts options as a list of @var{key}=@var{value} pairs,
+separated by ":".

 A description of the accepted options follows.

@@ -400,11 +211,8 @@ If set to 1 specify not to print the section header and footer.
 Default value is 0.
@end table

-@section compact, csv
-Compact and CSV format.
-
-The @code{csv} writer is equivalent to @code{compact}, but supports
-different defaults.
+@section compact
+Compact format.

 Each section is printed on a single line.
 If no option is specifid, the output has the form:
@@ -416,29 +224,30 @@ Metadata tags are printed in the corresponding "format" or "stream"
 section. A metadata tag key, if printed, is prefixed by the string
 "tag:".

+This writer accepts options as a list of @var{key}=@var{value} pairs,
+separated by ":".
+
 The description of the accepted options follows.

@table @option

@item item_sep, s
 Specify the character to use for separating fields in the output line.
-It must be a single printable character, it is "|" by default ("," for
-the @code{csv} writer).
+It must be a single printable character, it is "|" by default.

@item nokey, nk
 If set to 1 specify not to print the key of each field. Its default
-value is 0 (1 for the @code{csv} writer).
+value is 0.

@item escape, e
-Set the escape mode to use, default to "c" ("csv" for the @code{csv}
-writer).
+Set the escape mode to use, default to "c".

 It can assume one of the following values:
@table @option
@item c
-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
+Perform C-like escaping. Strings containing a newline ('\n') or
+carriage return ('\r'), 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}".
@@ -452,83 +261,22 @@ containing a newline ('\n'), a carriage return ('\r'), a double quote
 Perform no escaping.
@end table

-@item print_section, p
-Print the section name at the begin of each line if the value is
-@code{1}, disable it with value set to @code{0}. Default value is
-@code{1}.
-
@end table

-@section flat
-Flat format.
+@section csv
+CSV format.

-A free-form output where each line contains an explicit key=value, such as
-"streams.stream.3.tags.foo=bar". The output is shell escaped, so it can be
-directly embedded in sh scripts as long as the separator character is an
-alphanumeric character or an underscore (see @var{sep_char} option).
-
-The description of the accepted options follows.
-
-@table @option
-@item sep_char, s
-Separator character used to separate the chapter, the section name, IDs and
-potential tags in the printed field key.
-
-Default value is '.'.
-
-@item hierarchical, h
-Specify if the section name specification should be hierarchical. If
-set to 1, and if there is more than one section in the current
-chapter, the section name will be prefixed by the name of the
-chapter. A value of 0 will disable this behavior.
-
-Default value is 1.
-@end table
-
-@section ini
-INI format output.
-
-Print output in an INI based format.
-
-The following conventions are adopted:
-
-@itemize
-@item
-all key and values are UTF-8
-@item
-'.' is the subgroup separator
-@item
-newline, '\t', '\f', '\b' and the following characters are escaped
-@item
-'\' is the escape character
-@item
-'#' is the comment indicator
-@item
-'=' is the key/value separator
-@item
-':' 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 ":".
-
-The description of the accepted options follows.
-
-@table @option
-@item hierarchical, h
-Specify if the section name specification should be hierarchical. If
-set to 1, and if there is more than one section in the current
-chapter, the section name will be prefixed by the name of the
-chapter. A value of 0 will disable this behavior.
-
-Default value is 1.
-@end table
+This writer is equivalent to
+@code{compact=item_sep=,:nokey=1:escape=csv}.

@section json
 JSON based format.

 Each section is printed using JSON notation.

+This writer accepts options as a list of @var{key}=@var{value} pairs,
+separated by ":".
+
 The description of the accepted options follows.

@table @option
@@ -555,6 +303,9 @@ Note that the output issued will be compliant to the
 (@option{unit}, @option{prefix}, @option{byte_binary_prefix},
@option{sexagesimal} etc.) are specified.

+This writer accepts options as a list of @var{key}=@var{value} pairs,
+separated by ":".
+
 The description of the accepted options follows.

@table @option
@@ -591,81 +342,31 @@ MOV timecode is extracted from tmcd track, so is available in the tmcd
 stream metadata (@option{-show_streams}, see @var{TAG:timecode}).

@item
-DV, GXF and AVI timecodes are available in format metadata
+DV and GXF timecodes are available in format metadata
 (@option{-show_format}, see @var{TAG:timecode}).

@end itemize
@c man end TIMECODE

-@include config.texi
-@ifset config-all
-@set config-readonly
-@ifset config-avutil
-@include utils.texi
-@end ifset
-@ifset config-avcodec
-@include codecs.texi
-@include bitstream_filters.texi
-@end ifset
-@ifset config-avformat
-@include formats.texi
+@include syntax.texi
+@include decoders.texi
+@include demuxers.texi
@include protocols.texi
-@end ifset
-@ifset config-avdevice
-@include devices.texi
-@end ifset
-@ifset config-swresample
-@include resampler.texi
-@end ifset
-@ifset config-swscale
-@include scaler.texi
-@end ifset
-@ifset config-avfilter
-@include filters.texi
-@end ifset
-@end ifset
-
-@chapter See Also
-
-@ifhtml
-@ifset config-all
-@url{ffprobe.html,ffprobe},
-@end ifset
-@ifset config-not-all
-@url{ffprobe-all.html,ffprobe-all},
-@end ifset
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffserver.html,ffserver},
-@url{ffmpeg-utils.html,ffmpeg-utils},
-@url{ffmpeg-scaler.html,ffmpeg-scaler},
-@url{ffmpeg-resampler.html,ffmpeg-resampler},
-@url{ffmpeg-codecs.html,ffmpeg-codecs},
-@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters},
-@url{ffmpeg-formats.html,ffmpeg-formats},
-@url{ffmpeg-devices.html,ffmpeg-devices},
-@url{ffmpeg-protocols.html,ffmpeg-protocols},
-@url{ffmpeg-filters.html,ffmpeg-filters}
-@end ifhtml
-
-@ifnothtml
-@ifset config-all
-ffprobe(1),
-@end ifset
-@ifset config-not-all
-ffprobe-all(1),
-@end ifset
-ffmpeg(1), ffplay(1), ffserver(1),
-ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
-ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
-ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)
-@end ifnothtml
-
-@include authors.texi
+@include indevs.texi

@ignore

@setfilename ffprobe
@settitle ffprobe media prober

+@c man begin SEEALSO
+ffmpeg(1), ffplay(1), ffserver(1) and the FFmpeg HTML documentation
+@c man end
+
+@c man begin AUTHORS
+The FFmpeg developers
+@c man end
+
@end ignore

@bye
--- a/doc/ffprobe.xsd
+++ b/doc/ffprobe.xsd
@@ -11,8 +11,6 @@
            <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="streams"  type="ffprobe:streamsType" minOccurs="0" maxOccurs="1" />
-            <xsd:element name="programs" type="ffprobe:programsType" 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" />
@@ -28,10 +26,7 @@

    <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:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
        </xsd:sequence>
    </xsd:complexType>

@@ -44,12 +39,9 @@
      <xsd:attribute name="dts_time"      type="xsd:float" />
      <xsd:attribute name="duration"      type="xsd:long"  />
      <xsd:attribute name="duration_time" type="xsd:float" />
-      <xsd:attribute name="convergence_duration"      type="xsd:long"  />
-      <xsd:attribute name="convergence_duration_time" type="xsd:float" />
      <xsd:attribute name="size"          type="xsd:long" use="required" />
      <xsd:attribute name="pos"           type="xsd:long"  />
      <xsd:attribute name="flags"         type="xsd:string" use="required" />
-      <xsd:attribute name="data"          type="xsd:string" />
    </xsd:complexType>

    <xsd:complexType name="frameType">
@@ -61,18 +53,11 @@
      <xsd:attribute name="pkt_pts_time"  type="xsd:float"/>
      <xsd:attribute name="pkt_dts"       type="xsd:long" />
      <xsd:attribute name="pkt_dts_time"  type="xsd:float"/>
-      <xsd:attribute name="best_effort_timestamp"      type="xsd:long" />
-      <xsd:attribute name="best_effort_timestamp_time" type="xsd:float" />
-      <xsd:attribute name="pkt_duration"  type="xsd:long" />
-      <xsd:attribute name="pkt_duration_time" type="xsd:float"/>
      <xsd:attribute name="pkt_pos"       type="xsd:long" />
-      <xsd:attribute name="pkt_size"      type="xsd:int" />

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

      <!-- video attributes -->
      <xsd:attribute name="width"                  type="xsd:long"  />
@@ -85,16 +70,7 @@
      <xsd:attribute name="interlaced_frame"       type="xsd:int"   />
      <xsd:attribute name="top_field_first"        type="xsd:int"   />
      <xsd:attribute name="repeat_pict"            type="xsd:int"   />
-    </xsd: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:attribute name="reference"              type="xsd:int"   />
    </xsd:complexType>

    <xsd:complexType name="streamsType">
@@ -103,41 +79,14 @@
        </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" />
-      <xsd:attribute name="original"         type="xsd:int" use="required" />
-      <xsd:attribute name="comment"          type="xsd:int" use="required" />
-      <xsd:attribute name="lyrics"           type="xsd:int" use="required" />
-      <xsd:attribute name="karaoke"          type="xsd:int" use="required" />
-      <xsd:attribute name="forced"           type="xsd:int" use="required" />
-      <xsd:attribute name="hearing_impaired" type="xsd:int" use="required" />
-      <xsd:attribute name="visual_impaired"  type="xsd:int" use="required" />
-      <xsd:attribute name="clean_effects"    type="xsd:int" use="required" />
-      <xsd:attribute name="attached_pic"     type="xsd:int" use="required" />
-    </xsd:complexType>
-
    <xsd:complexType name="streamType">
-      <xsd:sequence>
-        <xsd:element name="disposition" type="ffprobe:streamDispositionType" minOccurs="0" maxOccurs="1"/>
-        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
-      </xsd:sequence>
-
      <xsd:attribute name="index"            type="xsd:int" use="required"/>
      <xsd:attribute name="codec_name"       type="xsd:string" />
      <xsd:attribute name="codec_long_name"  type="xsd:string" />
-      <xsd:attribute name="profile"          type="xsd:string" />
      <xsd:attribute name="codec_type"       type="xsd:string" />
      <xsd:attribute name="codec_time_base"  type="xsd:string" use="required"/>
      <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" />

      <!-- video attributes -->
      <xsd:attribute name="width"                type="xsd:int"/>
@@ -153,16 +102,13 @@
      <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"/>
      <xsd:attribute name="r_frame_rate"     type="xsd:string" use="required"/>
      <xsd:attribute name="avg_frame_rate"   type="xsd:string" use="required"/>
      <xsd:attribute name="time_base"        type="xsd:string" use="required"/>
-      <xsd:attribute name="start_pts"        type="xsd:long"/>
      <xsd:attribute name="start_time"       type="xsd:float"/>
-      <xsd:attribute name="duration_ts"      type="xsd:long"/>
      <xsd:attribute name="duration"         type="xsd:float"/>
      <xsd:attribute name="bit_rate"         type="xsd:int"/>
      <xsd:attribute name="nb_frames"        type="xsd:int"/>
@@ -170,23 +116,6 @@
      <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"/>
@@ -194,14 +123,12 @@

      <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="format_long_name" type="xsd:string" use="required"/>
      <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">
@@ -224,32 +151,12 @@
      <xsd:attribute name="configuration"    type="xsd:string" use="required"/>
    </xsd:complexType>

-    <xsd:complexType name="chaptersType">
-      <xsd:sequence>
-        <xsd:element name="chapter" type="ffprobe:chapterType" minOccurs="0" maxOccurs="unbounded"/>
-      </xsd:sequence>
-    </xsd:complexType>
-
-    <xsd:complexType name="chapterType">
-      <xsd:sequence>
-        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
-      </xsd:sequence>
-
-      <xsd:attribute name="id"         type="xsd:int" use="required"/>
-      <xsd:attribute name="time_base"  type="xsd:string" use="required"/>
-      <xsd:attribute name="start"      type="xsd:int" use="required"/>
-      <xsd:attribute name="start_time" type="xsd:float"/>
-      <xsd:attribute name="end"        type="xsd:int" use="required"/>
-      <xsd:attribute name="end_time"   type="xsd:float" use="required"/>
-    </xsd:complexType>
-
    <xsd:complexType name="libraryVersionType">
      <xsd:attribute name="name"        type="xsd:string" use="required"/>
      <xsd:attribute name="major"       type="xsd:int"    use="required"/>
      <xsd:attribute name="minor"       type="xsd:int"    use="required"/>
      <xsd:attribute name="micro"       type="xsd:int"    use="required"/>
      <xsd:attribute name="version"     type="xsd:int"    use="required"/>
-      <xsd:attribute name="ident"       type="xsd:string" use="required"/>
    </xsd:complexType>

    <xsd:complexType name="libraryVersionsType">
--- a/doc/ffserver.conf
+++ b/doc/ffserver.conf
@@ -25,6 +25,10 @@ MaxBandwidth 1000
 # '-' is the standard output.
 CustomLog -

+# Suppress that if you want to launch ffserver as a daemon.
+NoDaemon
+
+
 ##################################################################
 # Definition of the live feeds. Each live feed contains one video
 # and/or audio sequence coming from an ffmpeg encoder or another
@@ -235,7 +239,7 @@ StartSendOnKey

 #<Stream test.ogg>
 #Feed feed1.ffm
-#Metadata title "Stream title"
+#Title "Stream title"
 #AudioBitRate 64
 #AudioChannels 2
 #AudioSampleRate 44100
@@ -280,10 +284,10 @@ 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
@@ -9,159 +9,57 @@

@contents

-@chapter Synopsis
+@chapter Synopsys

-ffserver [@var{options}]
+The generic syntax is:
+
+@example
+@c man begin SYNOPSIS
+ffserver [options]
+@c man end
+@end example

@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.
+ffserver is a streaming server for both audio and video. It supports

-@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}.
+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 ffserver.conf).

-@command{ffserver} receives prerecorded files or FFM streams from some
-@command{ffmpeg} instance as input, then streams them over
-RTP/RTSP/HTTP.
+ffserver runs in daemon mode by default; that is, it puts itself in
+the background and detaches from its TTY, unless it is launched in
+debug mode or a NoDaemon option is specified in the configuration
+file.

-An @command{ffserver} instance will listen on some port as specified
-in the configuration file. You can launch one or more instances of
-@command{ffmpeg} and send one or more FFM streams to the port where
-ffserver is expecting to receive them. Alternately, you can make
-@command{ffserver} launch such @command{ffmpeg} instances at startup.
+This documentation covers only the streaming aspects of ffserver /
+ffmpeg. All questions about parameters for ffmpeg, codec questions,
+etc. are not covered here. Read @file{ffmpeg.html} for more
+information.

-Input streams are called feeds, and each one is specified by a
-@code{<Feed>} section in the configuration file.
+@section How does it work?
+
+ffserver receives prerecorded files or FFM streams from some ffmpeg
+instance as input, then streams them over RTP/RTSP/HTTP.
+
+An ffserver instance will listen on some port as specified in the
+configuration file. You can launch one or more instances of ffmpeg and
+send one or more FFM streams to the port where ffserver is expecting
+to receive them. Alternately, you can make ffserver launch such ffmpeg
+instances at startup.
+
+Input streams are called feeds, and each one is specified by a <Feed>
+section in the configuration file.

 For each feed you can have different output streams in various
-formats, each one specified by a @code{<Stream>} section in the
-configuration file.
-
-@chapter Detailed description
-
-@command{ffserver} works by forwarding streams encoded by
-@command{ffmpeg}, or pre-recorded streams which are read from disk.
-
-Precisely, @command{ffserver} acts as an HTTP server, accepting POST
-requests from @command{ffmpeg} to acquire the stream to publish, and
-serving RTSP clients or HTTP clients GET requests with the stream
-media content.
-
-A feed is an @ref{FFM} stream created by @command{ffmpeg}, and sent to
-a port where @command{ffserver} is listening.
-
-Each feed is identified by a unique name, corresponding to the name
-of the resource published on @command{ffserver}, and is configured by
-a dedicated @code{Feed} section in the configuration file.
-
-The feed publish URL is given by:
-@example
-http://@var{ffserver_ip_address}:@var{http_port}/@var{feed_name}
-@end example
-
-where @var{ffserver_ip_address} is the IP address of the machine where
-@command{ffserver} is installed, @var{http_port} is the port number of
-the HTTP server (configured through the @option{Port} 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 allow 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{Port} 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 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:
-@example
-               _________       __________
-              |         |     |          |
-ffmpeg 1 -----| feed 1  |-----| stream 1 |
-    \         |_________|\    |__________|
-     \                    \
-      \                    \   __________
-       \                    \ |          |
-        \                    \| stream 2 |
-         \                    |__________|
-          \
-           \   _________       __________
-            \ |         |     |          |
-             \| feed 2  |-----| stream 3 |
-              |_________|     |__________|
-
-               _________       __________
-              |         |     |          |
-ffmpeg 2 -----| feed 3  |-----| stream 4 |
-              |_________|     |__________|
-
-               _________       __________
-              |         |     |          |
-              | file 1  |-----| stream 5 |
-              |_________|     |__________|
-@end example
-
-@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.
+formats, each one specified by a <Stream> section in the configuration
+file.

@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.
@@ -180,8 +78,35 @@ 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 What do I need?
+
+I use Linux on a 900 MHz Duron with a cheapo Bt848 based TV capture card. I'm
+using stock Linux 2.4.17 with the stock drivers. [Actually that isn't true,
+I needed some special drivers for my motherboard-based sound card.]
+
+I understand that FreeBSD systems work just fine as well.
+
@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:

@@ -209,6 +134,35 @@ 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?

 You can replay video from .ffm files that was recorded earlier.
@@ -248,6 +202,9 @@ 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
@@ -284,617 +241,39 @@ For example:   @samp{http://localhost:8080/test.asf?date=2002-07-26T23:05:00}.
@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 and causes ffserver to run in the foreground
+rather than as a daemon.
@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 Port @var{port_number}
-@item RTSPPort @var{port_number}
-
-Set TCP port number on which the HTTP/RTSP server is listening. 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 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.
-
-@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.
-@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 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{option} @var{value} (@emph{encoding,audio})
-Set generic option for audio stream.
-
-@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{option} @var{value} (@emph{encoding,video})
-Set generic option for video stream.
-
-@item AVPresetVideo @var{preset} (@emph{encoding,video})
-Set preset for video stream.
-
-@var{preset} must be the path of a preset file.
-@end table
-
-@subsection Server status stream
-
-A server status stream is a special stream which is used to show
-statistics about the @command{ffserver} operations.
-
-It must be specified setting the option @option{Format} to
-@samp{status}.
-
-@section Redirect section
-
-A redirect section specifies where to redirect the requested URL to
-another page.
-
-A redirect section must be introduced by the line:
-@example
-<Redirect NAME>
-@end example
-
-where @var{NAME} is the name of the page which should be redirected.
-
-It only accepts the option @option{URL}, which specify the redirection
-URL.
-
-@chapter Stream examples
-
-@itemize
-@item
-Multipart JPEG
-@example
-<Stream test.mjpg>
-Feed feed1.ffm
-Format mpjpeg
-VideoFrameRate 2
-VideoIntraOnly
-NoAudio
-Strict -1
-</Stream>
-@end example
-
-@item
-Single JPEG
-@example
-<Stream test.jpg>
-Feed feed1.ffm
-Format jpeg
-VideoFrameRate 2
-VideoIntraOnly
-VideoSize 352x240
-NoAudio
-Strict -1
-</Stream>
-@end example
-
-@item
-Flash
-@example
-<Stream test.swf>
-Feed feed1.ffm
-Format swf
-VideoFrameRate 2
-VideoIntraOnly
-NoAudio
-</Stream>
-@end example
-
-@item
-ASF compatible
-@example
-<Stream test.asf>
-Feed feed1.ffm
-Format asf
-VideoFrameRate 15
-VideoSize 352x240
-VideoBitRate 256
-VideoBufferSize 40
-VideoGopSize 30
-AudioBitRate 64
-StartSendOnKey
-</Stream>
-@end example
-
-@item
-MP3 audio
-@example
-<Stream test.mp3>
-Feed feed1.ffm
-Format mp2
-AudioCodec mp3
-AudioBitRate 64
-AudioChannels 1
-AudioSampleRate 44100
-NoVideo
-</Stream>
-@end example
-
-@item
-Ogg Vorbis audio
-@example
-<Stream test.ogg>
-Feed feed1.ffm
-Metadata title "Stream title"
-AudioBitRate 64
-AudioChannels 2
-AudioSampleRate 44100
-NoVideo
-</Stream>
-@end example
-
-@item
-Real with audio only at 32 kbits
-@example
-<Stream test.ra>
-Feed feed1.ffm
-Format rm
-AudioBitRate 32
-NoVideo
-</Stream>
-@end example
-
-@item
-Real with audio and video at 64 kbits
-@example
-<Stream test.rm>
-Feed feed1.ffm
-Format rm
-AudioBitRate 32
-VideoBitRate 128
-VideoFrameRate 25
-VideoGopSize 25
-</Stream>
-@end example
-
-@item
-For stream coming from a file: you only need to set the input filename
-and optionally a new format.
-
-@example
-<Stream file.rm>
-File "/usr/local/httpd/htdocs/tlive.rm"
-NoAudio
-</Stream>
-@end example
-
-@example
-<Stream file.asf>
-File "/usr/local/httpd/htdocs/test.asf"
-NoAudio
-Metadata author "Me"
-Metadata copyright "Super MegaCorp"
-Metadata title "Test stream from disk"
-Metadata comment "Test comment"
-</Stream>
-@end example
-@end itemize
-
@c man end

-@include config.texi
-@ifset config-all
-@ifset config-avutil
-@include utils.texi
-@end ifset
-@ifset config-avcodec
-@include codecs.texi
-@include bitstream_filters.texi
-@end ifset
-@ifset config-avformat
-@include formats.texi
-@include protocols.texi
-@end ifset
-@ifset config-avdevice
-@include devices.texi
-@end ifset
-@ifset config-swresample
-@include resampler.texi
-@end ifset
-@ifset config-swscale
-@include scaler.texi
-@end ifset
-@ifset config-avfilter
-@include filters.texi
-@end ifset
-@end ifset
-
-@chapter See Also
-
-@ifhtml
-@ifset config-all
-@url{ffserver.html,ffserver},
-@end ifset
-@ifset config-not-all
-@url{ffserver-all.html,ffserver-all},
-@end ifset
-the @file{doc/ffserver.conf} example,
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
-@url{ffmpeg-utils.html,ffmpeg-utils},
-@url{ffmpeg-scaler.html,ffmpeg-scaler},
-@url{ffmpeg-resampler.html,ffmpeg-resampler},
-@url{ffmpeg-codecs.html,ffmpeg-codecs},
-@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters},
-@url{ffmpeg-formats.html,ffmpeg-formats},
-@url{ffmpeg-devices.html,ffmpeg-devices},
-@url{ffmpeg-protocols.html,ffmpeg-protocols},
-@url{ffmpeg-filters.html,ffmpeg-filters}
-@end ifhtml
-
-@ifnothtml
-@ifset config-all
-ffserver(1),
-@end ifset
-@ifset config-not-all
-ffserver-all(1),
-@end ifset
-the @file{doc/ffserver.conf} example, ffmpeg(1), ffplay(1), ffprobe(1),
-ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
-ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
-ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)
-@end ifnothtml
-
-@include authors.texi
-
@ignore

@setfilename ffserver
@settitle ffserver video server

+@c man begin SEEALSO
+
+ffmpeg(1), ffplay(1), ffprobe(1), the @file{ffserver.conf}
+example and the FFmpeg HTML documentation
+@c man end
+
+@c man begin AUTHORS
+The FFmpeg developers
+@c man end
+
@end ignore

@bye
--- a/doc/fftools-common-opts.texi
+++ b/doc/fftools-common-opts.texi
@@ -1,303 +0,0 @@
-All the numerical options, if not specified otherwise, accept a string
-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 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.
-
-Options which do not take arguments are boolean options, and set the
-corresponding value to true. They can be set to false by prefixing
-the option name with "no". For example using "-nofoo"
-will set the boolean option with name "foo" to false.
-
-@anchor{Stream specifiers}
-@section Stream specifiers
-Some options are applied per-stream, e.g. bitrate or codec. Stream specifiers
-are used to precisely specify which stream(s) a given option belongs to.
-
-A stream specifier is a string generally appended to the option name and
-separated from it by a colon. E.g. @code{-codec:a:1 ac3} contains the
-@code{a:1} stream specifier, which matches the second audio stream. Therefore, it
-would select the ac3 codec for the second audio stream.
-
-A stream specifier can match several streams, so that the option is applied to all
-of them. E.g. the stream specifier in @code{-b:a 128k} matches all audio
-streams.
-
-An empty stream specifier matches all streams. For example, @code{-codec copy}
-or @code{-codec: copy} would copy all the streams without reencoding.
-
-Possible forms of stream specifiers are:
-@table @option
-@item @var{stream_index}
-Matches the stream with this index. E.g. @code{-threads:1 4} would set the
-thread count for the second stream to 4.
-@item @var{stream_type}[:@var{stream_index}]
-@var{stream_type} is one of following: 'v' for video, 'a' for audio, 's' for subtitle,
-'d' for data, and 't' for attachments. If @var{stream_index} is given, then it matches
-stream number @var{stream_index} of this type. Otherwise, it matches all
-streams of this type.
-@item p:@var{program_id}[:@var{stream_index}]
-If @var{stream_index} is given, then it matches the stream with number @var{stream_index}
-in the program with the id @var{program_id}. Otherwise, it matches all streams in the
-program.
-@item #@var{stream_id}
-Matches the stream by a format-specific ID.
-@end table
-
-@section Generic options
-
-These options are shared amongst the ff* tools.
-
-@table @option
-
-@item -L
-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.
-
-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.
-
-@item encoder=@var{encoder_name}
-Print detailed information about the encoder named @var{encoder_name}. Use the
-@option{-encoders} option to get a list of all encoders.
-
-@item demuxer=@var{demuxer_name}
-Print detailed information about the demuxer named @var{demuxer_name}. Use the
-@option{-formats} option to get a list of all demuxers and muxers.
-
-@item muxer=@var{muxer_name}
-Print detailed information about the muxer named @var{muxer_name}. Use the
-@option{-formats} option to get a list of all muxers and demuxers.
-
-@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.
-
-@item -codecs
-Show all codecs known to libavcodec.
-
-Note that the term 'codec' is used throughout this documentation as a shortcut
-for what is more correctly called a media bitstream format.
-
-@item -decoders
-Show available decoders.
-
-@item -encoders
-Show all available encoders.
-
-@item -bsfs
-Show available bitstream filters.
-
-@item -protocols
-Show available protocols.
-
-@item -filters
-Show available libavfilter filters.
-
-@item -pix_fmts
-Show available pixel formats.
-
-@item -sample_fmts
-Show available sample formats.
-
-@item -layouts
-Show channel names and standard channel layouts.
-
-@item -colors
-Show recognized color names.
-
-@item -loglevel [repeat+]@var{loglevel} | -v [repeat+]@var{loglevel}
-Set the logging level used by the library.
-Adding "repeat+" indicates that repeated log output should not be compressed
-to the first line and the "Last message repeated n times" line will be
-omitted. "repeat" can also be used alone.
-If "repeat" is used alone, and with no prior loglevel set, the default
-loglevel will be used. If multiple loglevel parameters are given, using
-'repeat' will not change the loglevel.
-@var{loglevel} is a number or a string containing one of the following values:
-@table @samp
-@item quiet
-Show nothing at all; be silent.
-@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
-Only show fatal errors. These are errors after which the process absolutely
-cannot continue after.
-@item error
-Show all errors, including ones which can be recovered from.
-@item warning
-Show all warnings and errors. Any message related to possibly
-incorrect or unexpected events will be shown.
-@item info
-Show informative messages during processing. This is in addition to
-warnings and errors. This is the default value.
-@item verbose
-Same as @code{info}, except more verbose.
-@item debug
-Show everything, including debugging information.
-@end table
-
-By default the program logs to stderr, if coloring is supported by the
-terminal, colors are used to mark errors and warnings. Log coloring
-can be disabled setting the environment variable
-@env{AV_LOG_FORCE_NOCOLOR} or @env{NO_COLOR}, or can be forced setting
-the environment variable @env{AV_LOG_FORCE_COLOR}.
-The use of the environment variable @env{NO_COLOR} is deprecated and
-will be dropped in a following FFmpeg version.
-
-@item -report
-Dump full command line and console output to a file named
-@code{@var{program}-@var{YYYYMMDD}-@var{HHMMSS}.log} in the current
-directory.
-This file can be useful for bug reports.
-It also implies @code{-loglevel verbose}.
-
-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; 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 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{%}
-@end table
-
-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.
-@example
-ffmpeg -cpuflags -sse+mmx ...
-ffmpeg -cpuflags mmx ...
-ffmpeg -cpuflags 0 ...
-@end example
-Possible flags for this option are:
-@table @samp
-@item x86
-@table @samp
-@item mmx
-@item mmxext
-@item sse
-@item sse2
-@item sse2slow
-@item sse3
-@item sse3slow
-@item ssse3
-@item atom
-@item sse4.1
-@item sse4.2
-@item avx
-@item xop
-@item fma4
-@item 3dnow
-@item 3dnowext
-@item cmov
-@end table
-@item ARM
-@table @samp
-@item armv5te
-@item armv6
-@item armv6t2
-@item vfp
-@item vfpv3
-@item neon
-@end table
-@item PowerPC
-@table @samp
-@item altivec
-@end table
-@item Specific Processors
-@table @samp
-@item pentium2
-@item pentium3
-@item pentium4
-@item k6
-@item k62
-@item athlon
-@item athlonxp
-@item k8
-@end table
-@end table
-
-@item -opencl_bench
-Benchmark all available OpenCL devices and show the results. This option
-is only available when FFmpeg has been compiled with @code{--enable-opencl}.
-
-@item -opencl_options options (@emph{global})
-Set OpenCL environment options. This option is only available when
-FFmpeg has been compiled with @code{--enable-opencl}.
-
-@var{options} must be a list of @var{key}=@var{value} option pairs
-separated by ':'. See the ``OpenCL Options'' section in the
-ffmpeg-utils manual for the list of supported options.
-@end table
-
-@section AVOptions
-
-These options are provided directly by the libavformat, libavdevice and
-libavcodec libraries. To see the list of available AVOptions, use the
-@option{-help} option. They are separated into two categories:
-@table @option
-@item generic
-These options can be set for any container, codec or device. Generic options
-are listed under AVFormatContext options for containers/devices and under
-AVCodecContext options for codecs.
-@item private
-These options are specific to the given container, device or codec. Private
-options are listed under their corresponding containers/devices/codecs.
-@end table
-
-For example to write an ID3v2.3 header instead of a default ID3v2.4 to
-an MP3 file, use the @option{id3v2_version} private option of the MP3
-muxer:
-@example
-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.
-
-Note: the @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.
--- a/doc/filter_design.txt
+++ b/doc/filter_design.txt
@@ -12,172 +12,28 @@ Format negotiation
 ==================

  The query_formats method should set, for each input and each output links,
-  the list of supported formats.
+  the list supported formats.

  For video links, that means pixel format. For audio links, that means
-  channel layout, sample format (the sample packing is implied by the sample
-  format) and sample rate.
+  channel layout, and sample format (the sample packing is implied by the
+  sample format).

  The lists are not just lists, they are references to shared objects. When
  the negotiation mechanism computes the intersection of the formats
-  supported at each end of a link, all references to both lists are replaced
-  with a reference to the intersection. And when a single format is
+  supported at each ends of a link, all references to both lists are
+  replaced with a reference to the intersection. And when a single format is
  eventually chosen for a link amongst the remaining list, again, all
  references to the list are updated.

  That means that if a filter requires that its input and output have the
-  same format amongst a supported list, all it has to do is use a reference
+  same format amongst a supported list, all it have to do is use a reference
  to the same list of formats.

-  query_formats can leave some formats unset and return AVERROR(EAGAIN) to
-  cause the negotiation mechanism to try again later. That can be used by
-  filters with complex requirements to use the format negotiated on one link
-  to set the formats supported on another.
-

 Buffer references ownership and permissions
 ===========================================

-  Principle
-  ---------
-
-    Audio and video data are voluminous; the buffer and buffer reference
-    mechanism is intended to avoid, as much as possible, expensive copies of
-    that data while still allowing the filters to produce correct results.
-
-    The data is stored in buffers represented by AVFilterBuffer structures.
-    They must not be accessed directly, but through references stored in
-    AVFilterBufferRef structures. Several references can point to the
-    same buffer; the buffer is automatically deallocated once all
-    corresponding references have been destroyed.
-
-    The characteristics of the data (resolution, sample rate, etc.) are
-    stored in the reference; different references for the same buffer can
-    show different characteristics. In particular, a video reference can
-    point to only a part of a video buffer.
-
-    A reference is usually obtained as input to the start_frame or
-    filter_frame method or requested using the ff_get_video_buffer or
-    ff_get_audio_buffer functions. A new reference on an existing buffer can
-    be created with the avfilter_ref_buffer. A reference is destroyed using
-    the avfilter_unref_bufferp function.
-
-  Reference ownership
-  -------------------
-
-    At any time, a reference “belongs” to a particular piece of code,
-    usually a filter. With a few caveats that will be explained below, only
-    that piece of code is allowed to access it. It is also responsible for
-    destroying it, although this is sometimes done automatically (see the
-    section on link reference fields).
-
-    Here are the (fairly obvious) rules for reference ownership:
-
-    * A reference received by the filter_frame method (or its start_frame
-      deprecated version) belongs to the corresponding filter.
-
-      Special exception: for video references: the reference may be used
-      internally for automatic copying and must not be destroyed before
-      end_frame; it can be given away to ff_start_frame.
-
-    * A reference passed to ff_filter_frame (or the deprecated
-      ff_start_frame) is given away and must no longer be used.
-
-    * A reference created with avfilter_ref_buffer belongs to the code that
-      created it.
-
-    * A reference obtained with ff_get_video_buffer or ff_get_audio_buffer
-      belongs to the code that requested it.
-
-    * A reference given as return value by the get_video_buffer or
-      get_audio_buffer method is given away and must no longer be used.
-
-  Link reference fields
-  ---------------------
-
-    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, cur_buf_copy and partial_buf are used by libavfilter internally
-    and must not be accessed by filters.
-
-  Reference permissions
-  ---------------------
-
-    The AVFilterBufferRef structure has a perms field that describes what
-    the code that owns the reference is allowed to do to the buffer data.
-    Different references for the same buffer can have different permissions.
-
-    For video filters that implement the deprecated
-    start_frame/draw_slice/end_frame API, the permissions only apply to the
-    parts of the buffer that have already been covered by the draw_slice
-    method.
-
-    The value is a binary OR of the following constants:
-
-    * AV_PERM_READ: the owner can read the buffer data; this is essentially
-      always true and is there for self-documentation.
-
-    * AV_PERM_WRITE: the owner can modify the buffer data.
-
-    * AV_PERM_PRESERVE: the owner can rely on the fact that the buffer data
-      will not be modified by previous filters.
-
-    * AV_PERM_REUSE: the owner can output the buffer several times, without
-      modifying the data in between.
-
-    * AV_PERM_REUSE2: the owner can output the buffer several times and
-      modify the data in between (useless without the WRITE permissions).
-
-    * AV_PERM_ALIGN: the owner can access the data using fast operations
-      that require data alignment.
-
-    The READ, WRITE and PRESERVE permissions are about sharing the same
-    buffer between several filters to avoid expensive copies without them
-    doing conflicting changes on the data.
-
-    The REUSE and REUSE2 permissions are about special memory for direct
-    rendering. For example a buffer directly allocated in video memory must
-    not modified once it is displayed on screen, or it will cause tearing;
-    it will therefore not have the REUSE2 permission.
-
-    The ALIGN permission is about extracting part of the buffer, for
-    copy-less padding or cropping for example.
-
-
-    References received on input pads are guaranteed to have all the
-    permissions stated in the min_perms field and none of the permissions
-    stated in the rej_perms.
-
-    References obtained by ff_get_video_buffer and ff_get_audio_buffer are
-    guaranteed to have at least all the permissions requested as argument.
-
-    References created by avfilter_ref_buffer have the same permissions as
-    the original reference minus the ones explicitly masked; the mask is
-    usually ~0 to keep the same permissions.
-
-    Filters should remove permissions on reference they give to output
-    whenever necessary. It can be automatically done by setting the
-    rej_perms field on the output pad.
-
-    Here are a few guidelines corresponding to common situations:
-
-    * Filters that modify and forward their frame (like drawtext) need the
-      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
-      with the WRITE permission.
-
-    * Filters that intend to keep a reference after the filtering process
-      is finished (after filter_frame returns) must have the PRESERVE
-      permission on it and remove the WRITE permission if they create a new
-      reference to give it away.
-
-    * Filters that intend to modify a reference they have kept after the end
-      of the filtering process need the REUSE2 permission and must remove
-      the PRESERVE permission if they create a new reference to give it
-      away.
+  TODO


 Frame scheduling
@@ -189,11 +45,11 @@ Frame scheduling
  Simple filters that output one frame for each input frame should not have
  to worry about it.

-  filter_frame
-  ------------
+  start_frame / filter_samples
+  ----------------------------

-    This method is called when a frame is pushed to the filter's input. It
-    can be called at any time except in a reentrant way.
+    These methods are called when a frame is pushed to the filter's input.
+    They can be called at any time except in a reentrant way.

    If the input frame is enough to produce output, then the filter should
    push the output frames on the output link immediately.
@@ -204,7 +60,7 @@ Frame scheduling
    filter; these buffered frames must be flushed immediately if a new input
    produces new output.

-    (Example: frame rate-doubling filter: filter_frame must (1) flush the
+    (Example: framerate-doubling filter: start_frame must (1) flush the
    second copy of the previous frame, if it is still there, (2) push the
    first copy of the incoming frame, (3) keep the second copy for later.)

@@ -214,7 +70,7 @@ Frame scheduling
    request_frame method or the application.

    If a filter has several inputs, the filter must be ready for frames
-    arriving randomly on any input. Therefore, any filter with several inputs
+    arriving randomly on any input. Therefore, any filter with several input
    will most likely require some kind of queuing mechanism. It is perfectly
    acceptable to have a limited queue and to drop frames when the inputs
    are too unbalanced.
@@ -224,12 +80,12 @@ Frame scheduling

    This method is called when a frame is wanted on an output.

-    For an input, it should directly call filter_frame on the corresponding
-    output.
+    For an input, it should directly call start_frame or filter_samples on
+    the corresponding output.

    For a filter, if there are queued frames already ready, one of these
    frames should be pushed. If not, the filter should request a frame on
-    one of its inputs, repeatedly until at least one frame has been pushed.
+    one of its input, repeatedly until at least one frame has been pushed.

    Return values:
    if request_frame could produce a frame, it should return 0;
@@ -246,7 +102,7 @@ Frame scheduling
        }
        while (!frame_pushed) {
            input = input_where_a_frame_is_most_needed();
-            ret = ff_request_frame(input);
+            ret = avfilter_request_frame(input);
            if (ret == AVERROR_EOF) {
                process_eof_on_input();
            } else if (ret < 0) {
@@ -257,14 +113,4 @@ Frame scheduling

    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 will be called and do the work.
-
-Legacy API
-==========
-
-  Until libavfilter 3.23, the filter_frame method was split:
-
-  - for video filters, it was made of start_frame, draw_slice (that could be
-    called several times on distinct parts of the frame) and end_frame;
-
-  - for audio filters, it was called filter_samples.
+    the start_frame / filter_samples method will be called and do the work.
--- a/doc/filters.texi
+++ b/doc/filters.texi
--- a/doc/formats.texi
+++ b/doc/formats.texi
@@ -1,212 +0,0 @@
-@chapter Format Options
-@c man begin FORMAT OPTIONS
-
-The libavformat library provides some generic global options, which
-can be set on all the muxers and demuxers. In addition each muxer or
-demuxer may support so-called private options, which are specific for
-that component.
-
-Options may be set by specifying -@var{option} @var{value} in the
-FFmpeg tools, or by setting the value explicitly in the
-@code{AVFormatContext} options or using the @file{libavutil/opt.h} API
-for programmatic use.
-
-The list of supported options follows:
-
-@table @option
-@item avioflags @var{flags} (@emph{input/output})
-Possible values:
-@table @samp
-@item direct
-Reduce buffering.
-@end table
-
-@item probesize @var{integer} (@emph{input})
-Set probing size in bytes, i.e. the size of the data to analyze to get
-stream information. A higher value will allow to detect more
-information in case it is dispersed into the stream, but will increase
-latency. Must be an integer not lesser than 32. It is 5000000 by default.
-
-@item packetsize @var{integer} (@emph{output})
-Set packet size.
-
-@item fflags @var{flags} (@emph{input/output})
-Set format flags.
-
-Possible values:
-@table @samp
-@item ignidx
-Ignore index.
-@item genpts
-Generate PTS.
-@item nofillin
-Do not fill in missing values that can be exactly calculated.
-@item noparse
-Disable AVParsers, this needs @code{+nofillin} too.
-@item igndts
-Ignore DTS.
-@item discardcorrupt
-Discard corrupted frames.
-@item sortdts
-Try to interleave output packets by DTS.
-@item keepside
-Do not merge side data.
-@item latm
-Enable RTP MP4A-LATM payload.
-@item nobuffer
-Reduce the latency introduced by optional buffering
-@end table
-
-@item seek2any @var{integer} (@emph{input})
-Allow seeking to non-keyframes on demuxer level when supported if set to 1.
-Default is 0.
-
-@item analyzeduration @var{integer} (@emph{input})
-Specify how many microseconds are analyzed to probe the input. A
-higher value will allow to detect more accurate information, but will
-increase latency. It defaults to 5,000,000 microseconds = 5 seconds.
-
-@item cryptokey @var{hexadecimal string} (@emph{input})
-Set decryption key.
-
-@item indexmem @var{integer} (@emph{input})
-Set max memory used for timestamp index (per stream).
-
-@item rtbufsize @var{integer} (@emph{input})
-Set max memory used for buffering real-time frames.
-
-@item fdebug @var{flags} (@emph{input/output})
-Print specific debug info.
-
-Possible values:
-@table @samp
-@item ts
-@end table
-
-@item max_delay @var{integer} (@emph{input/output})
-Set maximum muxing or demuxing delay in microseconds.
-
-@item fpsprobesize @var{integer} (@emph{input})
-Set number of frames used to probe fps.
-
-@item audio_preload @var{integer} (@emph{output})
-Set microseconds by which audio packets should be interleaved earlier.
-
-@item chunk_duration @var{integer} (@emph{output})
-Set microseconds for each chunk.
-
-@item chunk_size @var{integer} (@emph{output})
-Set size in bytes for each chunk.
-
-@item err_detect, f_err_detect @var{flags} (@emph{input})
-Set error detection flags. @code{f_err_detect} is deprecated and
-should be used only via the @command{ffmpeg} tool.
-
-Possible values:
-@table @samp
-@item crccheck
-Verify embedded CRCs.
-@item bitstream
-Detect bitstream specification deviations.
-@item buffer
-Detect improper bitstream length.
-@item explode
-Abort decoding on minor error detection.
-@item careful
-Consider things that violate the spec and have not been seen in the
-wild as errors.
-@item compliant
-Consider all spec non compliancies as errors.
-@item aggressive
-Consider things that a sane encoder should not do as an error.
-@end table
-
-@item use_wallclock_as_timestamps @var{integer} (@emph{input})
-Use wallclock as timestamps.
-
-@item avoid_negative_ts @var{integer} (@emph{output})
-
-Possible values:
-@table @samp
-@item make_non_negative
-Shift timestamps to make them non-negative.
-Also note that this affects only leading negative timestamps, and not
-non-monotonic negative timestamps.
-@item make_zero
-Shift timestamps so that the first timestamp is 0.
-@item auto (default)
-Enables shifting when required by the target format.
-@item disabled
-Disables shifting of timestamp.
-@end table
-
-When shifting is enabled, all output timestamps are shifted by the
-same amount. Audio, video, and subtitles desynching and relative
-timestamp differences are preserved compared to how they would have
-been without shifting.
-
-@item skip_initial_bytes @var{integer} (@emph{input})
-Set number of bytes to skip before reading header and frames if set to 1.
-Default is 0.
-
-@item correct_ts_overflow @var{integer} (@emph{input})
-Correct single timestamp overflows if set to 1. Default is 1.
-
-@item flush_packets @var{integer} (@emph{output})
-Flush the underlying I/O stream after each packet. Default 1 enables it, and
-has the effect of reducing the latency; 0 disables it and may slightly
-increase performance in some cases.
-
-@item output_ts_offset @var{offset} (@emph{output})
-Set the output time offset.
-
-@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 by the muxer to the output timestamps.
-
-Specifying a positive offset means that the corresponding streams are
-delayed bt the time duration specified in @var{offset}. Default value
-is @code{0} (meaning that no offset is applied).
-@end table
-
-@c man end FORMAT OPTIONS
-
-@anchor{Format stream specifiers}
-@section Format stream specifiers
-
-Format stream specifiers allow selection of one or more streams that
-match specific properties.
-
-Possible forms of stream specifiers are:
-@table @option
-@item @var{stream_index}
-Matches the stream with this index.
-
-@item @var{stream_type}[:@var{stream_index}]
-@var{stream_type} is one of following: 'v' for video, 'a' for audio,
-'s' for subtitle, 'd' for data, and 't' for attachments. If
-@var{stream_index} is given, then it matches the stream number
-@var{stream_index} of this type. Otherwise, it matches all streams of
-this type.
-
-@item p:@var{program_id}[:@var{stream_index}]
-If @var{stream_index} is given, then it matches the stream with number
-@var{stream_index} in the program with the id
-@var{program_id}. Otherwise, it matches all streams in the program.
-
-@item #@var{stream_id}
-Matches the stream by a format-specific ID.
-@end table
-
-The exact semantics of stream specifiers is defined by the
-@code{avformat_match_stream_specifier()} function declared in the
-@file{libavformat/avformat.h} header.
-
-@ifclear config-writeonly
-@include demuxers.texi
-@end ifclear
-@ifclear config-readonly
-@include muxers.texi
-@end ifclear
-@include metadata.texi
--- a/doc/general.texi
+++ b/doc/general.texi
@@ -24,22 +24,17 @@ instructions.  To enable using OpenJPEG in FFmpeg, pass @code{--enable-libopenjp
@file{./configure}.


-@section OpenCORE, VisualOn, and Fraunhofer libraries
+@section OpenCORE and VisualOn libraries

-Spun off Google Android sources, OpenCore, VisualOn and Fraunhofer
-libraries provide encoders for a number of audio codecs.
+Spun off Google Android sources, OpenCore and VisualOn libraries provide
+encoders for a number of audio codecs.

@float NOTE
 OpenCORE and VisualOn libraries are under the Apache License 2.0
 (see @url{http://www.apache.org/licenses/LICENSE-2.0} for details), which is
-incompatible to the LGPL version 2.1 and GPL version 2. You have to
+incompatible with the LGPL version 2.1 and GPL version 2. You have to
 upgrade FFmpeg's license to LGPL version 3 (or if you have enabled
-GPL components, GPL version 3) by passing @code{--enable-version3} to configure in
-order to use it.
-
-The Fraunhofer AAC library is licensed under a license incompatible to the GPL
-and is not known to be compatible to the LGPL. Therefore, you have to pass
-@code{--enable-nonfree} to configure to use it.
+GPL components, GPL version 3) to use it.
@end float

@subsection OpenCORE AMR
@@ -68,14 +63,6 @@ Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the
 instructions for installing the library.
 Then pass @code{--enable-libvo-amrwbenc} to configure to enable it.

-@subsection Fraunhofer AAC library
-
-FFmpeg can make use of the Fraunhofer AAC library for AAC encoding.
-
-Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the
-instructions for installing the library.
-Then pass @code{--enable-libfdk-aac} to configure to enable it.
-
@section LAME

 FFmpeg can make use of the LAME library for MP3 encoding.
@@ -84,30 +71,14 @@ Go to @url{http://lame.sourceforge.net/} and follow the
 instructions for installing the library.
 Then pass @code{--enable-libmp3lame} to configure to enable it.

-@section TwoLAME
-
-FFmpeg can make use of the TwoLAME library for MP2 encoding.
-
-Go to @url{http://www.twolame.org/} and follow the
-instructions for installing the library.
-Then pass @code{--enable-libtwolame} to configure to enable it.
-
@section libvpx

-FFmpeg can make use of the libvpx library for VP8/VP9 encoding.
+FFmpeg can make use of the libvpx library for VP8 encoding.

 Go to @url{http://www.webmproject.org/} and follow the instructions for
 installing the library. Then pass @code{--enable-libvpx} to configure to
 enable it.

-@section libwavpack
-
-FFmpeg can make use of the libwavpack library for WavPack encoding.
-
-Go to @url{http://www.wavpack.com/} and follow the instructions for
-installing the library. Then pass @code{--enable-libwavpack} to configure to
-enable it.
-
@section x264

 FFmpeg can make use of the x264 library for H.264 encoding.
@@ -122,45 +93,6 @@ x264 is under the GNU Public License Version 2 or later
 details), you must upgrade FFmpeg's license to GPL in order to use it.
@end float

-@section x265
-
-FFmpeg can make use of the x265 library for HEVC encoding.
-
-Go to @url{http://x265.org/developers.html} and follow the instructions
-for installing the library. Then pass @code{--enable-libx265} to configure
-to enable it.
-
-@float note
-x265 is under the GNU Public License Version 2 or later
-(see @url{http://www.gnu.org/licenses/old-licenses/gpl-2.0.html} for
-details), you must upgrade FFmpeg's license to GPL in order to use it.
-@end float
-
-@section libilbc
-
-iLBC is a narrowband speech codec that has been made freely available
-by Google as part of the WebRTC project. libilbc is a packaging friendly
-copy of the iLBC codec. FFmpeg can make use of the libilbc library for
-iLBC encoding and decoding.
-
-Go to @url{https://github.com/dekkers/libilbc} and follow the instructions for
-installing the library. Then pass @code{--enable-libilbc} to configure to
-enable it.
-
-@section libzvbi
-
-libzvbi is a VBI decoding library which can be used by FFmpeg to decode DVB
-teletext pages and DVB teletext subtitles.
-
-Go to @url{http://sourceforge.net/projects/zapping/} and follow the instructions for
-installing the library. Then pass @code{--enable-libzvbi} to configure to
-enable it.
-
-@float NOTE
-libzvbi is licensed under the GNU General Public License Version 2 or later
-(see @url{http://www.gnu.org/licenses/old-licenses/gpl-2.0.html} for details),
-you must upgrade FFmpeg's license to GPL in order to use it.
-@end float


@chapter Supported File Formats, Codecs or Features
@@ -184,21 +116,11 @@ library:
@item American Laser Games MM   @tab   @tab X
    @tab Multimedia format used in games like Mad Dog McCree.
@item 3GPP AMR                  @tab X @tab X
-@item Amazing Studio Packed Animation File        @tab   @tab X
-    @tab Multimedia format used in game Heart Of Darkness.
@item Apple HTTP Live Streaming @tab   @tab X
@item Artworx Data Format       @tab   @tab X
-@item ADP                       @tab   @tab X
-    @tab Audio format used on the Nintendo Gamecube.
-@item AFC                       @tab   @tab X
-    @tab Audio format used on the Nintendo Gamecube.
@item ASF                       @tab X @tab X
-@item AST                       @tab X @tab X
-    @tab Audio format used on the Nintendo Wii.
@item AVI                       @tab X @tab X
-@item AviSynth                  @tab   @tab X
-@item AVR                       @tab   @tab X
-    @tab Audio format used on Mac.
+@item AVISynth                  @tab   @tab X
@item AVS                       @tab   @tab X
    @tab Multimedia format used by the Creature Shock game.
@item Beam Software SIFF        @tab   @tab X
@@ -212,8 +134,6 @@ library:
    @tab Used in Z and Z95 games.
@item Brute Force & Ignorance   @tab   @tab X
    @tab Used in the game Flash Traffic: City of Angels.
-@item BRSTM                     @tab   @tab X
-    @tab Audio format used on the Nintendo Wii.
@item BWF                       @tab X @tab X
@item CRI ADX                   @tab X @tab X
    @tab Audio-only format used in console video games.
@@ -244,7 +164,6 @@ library:
@item Electronic Arts cdata  @tab    @tab X
@item Electronic Arts Multimedia  @tab    @tab X
    @tab Used in various EA games; files have extensions like WVE and UV2.
-@item Ensoniq Paris Audio File  @tab   @tab X
@item FFM (FFserver live feed)  @tab X @tab X
@item Flash (SWF)               @tab X @tab X
@item Flash 9 (AVM2)            @tab X @tab X
@@ -259,14 +178,12 @@ library:
@item G.723.1                   @tab X @tab X
@item G.729 BIT                 @tab X @tab X
@item G.729 raw                 @tab   @tab X
-@item GIF Animation             @tab X @tab X
+@item GIF Animation             @tab X @tab
@item GXF                       @tab X @tab X
    @tab General eXchange Format SMPTE 360M, used by Thomson Grass Valley
         playout servers.
-@item HNM @tab   @tab X
-    @tab Only version 4 supported, used in some games from Cryo Interactive
@item iCEDraw File              @tab   @tab X
-@item ICO                       @tab X @tab X
+@item ICO                       @tab   @tab X
    @tab Microsoft Windows ICO
@item id Quake II CIN video     @tab   @tab X
@item id RoQ                    @tab X @tab X
@@ -274,20 +191,17 @@ library:
@item IEC61937 encapsulation @tab X @tab X
@item IFF                       @tab   @tab X
    @tab Interchange File Format
-@item iLBC                      @tab X @tab X
@item Interplay MVE             @tab   @tab X
    @tab Format used in various Interplay computer games.
@item IV8                       @tab   @tab X
    @tab A format generated by IndigoVision 8000 video server.
@item IVF (On2)                 @tab X @tab X
    @tab A format used by libvpx
-@item IRCAM                     @tab X @tab X
@item LATM                      @tab X @tab X
@item LMLM4                     @tab   @tab X
    @tab Used by Linux Media Labs MPEG-4 PCI boards
@item LOAS                      @tab   @tab X
    @tab contains LATM multiplexed AAC audio
-@item LVF                       @tab   @tab X
@item LXF                       @tab   @tab X
    @tab VR native stream format, used by Leitch/Harris' video servers.
@item Matroska                  @tab X @tab X
@@ -298,8 +212,6 @@ library:
    @tab Used in Sim City 3000; file extension .xa.
@item MD Studio                 @tab   @tab X
@item Metal Gear Solid: The Twin Snakes @tab @tab X
-@item Megalux Frame             @tab   @tab X
-    @tab Used by Megalux Ultimate Paint
@item Mobotix .mxg              @tab   @tab X
@item Monkey's Audio            @tab   @tab X
@item Motion Pixels MVI         @tab   @tab X
@@ -315,8 +227,6 @@ library:
    @tab also known as DVB Transport Stream
@item MPEG-4                    @tab X @tab X
    @tab MPEG-4 is a variant of QuickTime.
-@item Mirillis FIC video        @tab   @tab X
-    @tab No cursor rendering.
@item MIME multipart JPEG       @tab X @tab
@item MSN TCP webcam            @tab   @tab X
    @tab Used by MSN Messenger webcam streams.
@@ -329,7 +239,6 @@ library:
    @tab SMPTE 386M, D-10/IMX Mapping.
@item NC camera feed            @tab   @tab X
    @tab NC (AVIP NC4600) camera streams
-@item NIST SPeech HEader REsources @tab   @tab X
@item NTT TwinVQ (VQF)          @tab   @tab X
    @tab Nippon Telegraph and Telephone Corporation TwinVQ.
@item Nullsoft Streaming Video  @tab   @tab X
@@ -338,7 +247,6 @@ library:
    @tab NUT Open Container Format
@item Ogg                       @tab X @tab X
@item Playstation Portable PMP  @tab   @tab X
-@item Portable Voice Format     @tab   @tab X
@item TechnoTrend PVA           @tab   @tab X
    @tab Used by TechnoTrend DVB PCI boards.
@item QCP                       @tab   @tab X
@@ -349,14 +257,12 @@ library:
@item raw Dirac                 @tab X @tab X
@item raw DNxHD                 @tab X @tab X
@item raw DTS                   @tab X @tab X
-@item raw DTS-HD                @tab   @tab X
@item raw E-AC-3                @tab X @tab X
@item raw FLAC                  @tab X @tab X
@item raw GSM                   @tab   @tab X
@item raw H.261                 @tab X @tab X
@item raw H.263                 @tab X @tab X
@item raw H.264                 @tab X @tab X
-@item raw HEVC                  @tab X @tab X
@item raw Ingenient MJPEG       @tab   @tab X
@item raw MJPEG                 @tab X @tab X
@item raw MLP                   @tab   @tab X
@@ -368,9 +274,8 @@ library:
@item raw video                 @tab X @tab X
@item raw id RoQ                @tab X @tab
@item raw Shorten               @tab   @tab X
-@item raw TAK                   @tab   @tab X
@item raw TrueHD                @tab X @tab X
-@item raw VC-1                  @tab X @tab X
+@item raw VC-1                  @tab   @tab X
@item raw PCM A-law             @tab X @tab X
@item raw PCM mu-law            @tab X @tab X
@item raw PCM signed 8 bit      @tab X @tab X
@@ -396,13 +301,11 @@ library:
    @tab File format used by RED Digital cameras, contains JPEG 2000 frames and PCM audio.
@item RealMedia                 @tab X @tab X
@item Redirector                @tab   @tab X
-@item RedSpark                  @tab   @tab X
@item Renderware TeXture Dictionary @tab   @tab X
@item RL2                       @tab   @tab X
    @tab Audio and video format used in some games by Entertainment Software Partners.
@item RPL/ARMovie               @tab   @tab X
@item Lego Mindstorms RSO       @tab X @tab X
-@item RSD                       @tab   @tab X
@item RTMP                      @tab X @tab X
    @tab Output is performed by publishing stream to RTMP server
@item RTP                       @tab X @tab X
@@ -412,7 +315,6 @@ library:
@item SDP                       @tab   @tab X
@item Sega FILM/CPK             @tab   @tab X
    @tab Used in many Sega Saturn console games.
-@item Silicon Graphics Movie    @tab   @tab X
@item Sierra SOL                @tab   @tab X
    @tab .sol files used in Sierra Online games.
@item Sierra VMD                @tab   @tab X
@@ -421,12 +323,10 @@ library:
    @tab Multimedia format used by many games.
@item SMJPEG                    @tab X @tab X
    @tab Used in certain Loki game ports.
-@item Smush                     @tab   @tab X
-    @tab Multimedia format used in some LucasArts games.
@item Sony OpenMG (OMA)         @tab X @tab X
    @tab Audio format used in Sony Sonic Stage and Sony Vegas.
@item Sony PlayStation STR      @tab   @tab X
-@item Sony Wave64 (W64)         @tab X @tab X
+@item Sony Wave64 (W64)         @tab   @tab X
@item SoX native format         @tab X @tab X
@item SUN AU format             @tab X @tab X
@item Text files                @tab   @tab X
@@ -436,9 +336,8 @@ library:
    @tab Tiertex .seq files used in the DOS CD-ROM version of the game Flashback.
@item True Audio                @tab   @tab X
@item VC-1 test bitstream       @tab X @tab X
-@item Vivo                      @tab   @tab X
@item WAV                       @tab X @tab X
-@item WavPack                   @tab X @tab X
+@item WavPack                   @tab   @tab X
@item WebM                      @tab X @tab X
@item Windows Televison (WTV)   @tab X @tab X
@item Wing Commander III movie  @tab   @tab X
@@ -454,6 +353,7 @@ library:
@item eXtended BINary text (XBIN) @tab @tab X
@item YUV4MPEG pipe             @tab X @tab X
@item Psygnosis YOP             @tab   @tab X
+@item ZeroCodec Lossless Video  @tab   @tab X
@end multitable

@code{X} means that encoding (resp. decoding) is supported.
@@ -468,10 +368,9 @@ following image formats are supported:
@item .Y.U.V       @tab X @tab X
    @tab one raw file per component
@item animated GIF @tab X @tab X
+    @tab Only uncompressed GIFs are generated.
@item BMP          @tab X @tab X
    @tab Microsoft BMP image
-@item PIX          @tab   @tab X
-    @tab PIX is an image format used in the Argonaut BRender engine.
@item DPX          @tab X @tab X
    @tab Digital Picture Exchange
@item EXR          @tab   @tab X
@@ -507,12 +406,8 @@ following image formats are supported:
    @tab YUV, JPEG and some extension is not supported yet.
@item Truevision Targa  @tab X @tab X
    @tab Targa (.TGA) image format
-@item WebP         @tab E @tab X
-    @tab WebP image format, encoding supported through external library libwebp
@item XBM  @tab X @tab X
    @tab X BitMap image format
-@item XFace @tab X @tab X
-    @tab X-Face image format
@item XWD  @tab X @tab X
    @tab X Window Dump image format
@end multitable
@@ -528,15 +423,15 @@ following image formats are supported:
@item 4X Movie               @tab     @tab  X
    @tab Used in certain computer games.
@item 8088flex TMV           @tab     @tab  X
+@item 8SVX exponential       @tab     @tab  X
+@item 8SVX fibonacci         @tab     @tab  X
@item A64 multicolor         @tab  X  @tab
    @tab Creates video suitable to be played on a commodore 64 (multicolor mode).
-@item Amazing Studio PAF Video @tab     @tab  X
@item American Laser Games MM  @tab    @tab X
    @tab Used in games like Mad Dog McCree.
@item AMV Video              @tab  X  @tab  X
    @tab Used in Chinese MP3 players.
@item ANSI/ASCII art         @tab     @tab  X
-@item Apple Intermediate Codec @tab     @tab  X
@item Apple MJPEG-B          @tab     @tab  X
@item Apple ProRes           @tab  X  @tab  X
@item Apple QuickDraw        @tab     @tab  X
@@ -581,11 +476,9 @@ following image formats are supported:
@item Delphine Software International CIN video  @tab     @tab  X
    @tab Codec used in Delphine Software International games.
@item Discworld II BMV Video @tab     @tab  X
-@item Canopus Lossless Codec @tab     @tab  X
@item Cinepak                @tab     @tab  X
@item Cirrus Logic AccuPak   @tab  X  @tab  X
    @tab fourcc: CLJR
-@item CPiA Video Format      @tab     @tab  X
@item Creative YUV (CYUV)    @tab     @tab  X
@item DFA                    @tab     @tab  X
    @tab Codec used in Chronomaster game.
@@ -619,16 +512,12 @@ following image formats are supported:
    @tab Sorenson H.263 used in Flash
@item Forward Uncompressed   @tab     @tab  X
@item Fraps                  @tab     @tab  X
-@item Go2Webinar             @tab     @tab  X
-    @tab fourcc: G2M4
@item H.261                  @tab  X  @tab  X
@item H.263 / H.263-1996     @tab  X  @tab  X
@item H.263+ / H.263-1998 / H.263 version 2  @tab  X  @tab  X
@item H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10  @tab  E  @tab  X
    @tab encoding supported through external library libx264
-@item HEVC                   @tab  X  @tab  X
-    @tab encoding supported through the external library libx265
-@item HNM version 4          @tab     @tab  X
+@item H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 (VDPAU acceleration)  @tab  E  @tab  X
@item HuffYUV                @tab  X  @tab  X
@item HuffYUV FFmpeg variant @tab  X  @tab  X
@item IBM Ultimotion         @tab     @tab  X
@@ -659,18 +548,8 @@ following image formats are supported:
@item LCL (LossLess Codec Library) MSZH  @tab     @tab  X
@item LCL (LossLess Codec Library) ZLIB  @tab  E  @tab  E
@item LOCO                   @tab     @tab  X
-@item LucasArts Smush        @tab     @tab  X
-    @tab Used in LucasArts games.
@item lossless MJPEG         @tab  X  @tab  X
-@item Microsoft ATC Screen   @tab     @tab  X
-    @tab Also known as Microsoft Screen 3.
-@item Microsoft Expression Encoder Screen  @tab     @tab  X
-    @tab Also known as Microsoft Titanium Screen 2.
@item Microsoft RLE          @tab     @tab  X
-@item Microsoft Screen 1     @tab     @tab  X
-    @tab Also known as Windows Media Video V7 Screen.
-@item Microsoft Screen 2     @tab     @tab  X
-    @tab Also known as Windows Media Video V9 Screen.
@item Microsoft Video 1      @tab     @tab  X
@item Mimic                  @tab     @tab  X
    @tab Used in MSN Messenger Webcam streams.
@@ -680,6 +559,8 @@ following image formats are supported:
@item Mobotix MxPEG video    @tab     @tab  X
@item Motion Pixels video    @tab     @tab  X
@item MPEG-1 video           @tab  X  @tab  X
+@item MPEG-1/2 video XvMC (X-Video Motion Compensation)  @tab     @tab  X
+@item MPEG-1/2 video (VDPAU acceleration)  @tab     @tab  X
@item MPEG-2 video           @tab  X  @tab  X
@item MPEG-4 part 2          @tab  X  @tab  X
    @tab libxvidcore can be used alternatively for encoding.
@@ -697,10 +578,8 @@ following image formats are supported:
    @tab fourcc: VP60,VP61,VP62
@item VP8                    @tab  E  @tab  X
    @tab fourcc: VP80, encoding supported through external library libvpx
-@item VP9                    @tab  E  @tab  X
-    @tab encoding supported through external library libvpx
-@item Pinnacle TARGA CineWave YUV16 @tab     @tab  X
-    @tab fourcc: Y216
+@item planar RGB             @tab     @tab  X
+    @tab fourcc: 8BPS
@item Prores                 @tab     @tab  X
    @tab fourcc: apch,apcn,apcs,apco
@item Q-team QPEG            @tab     @tab  X
@@ -724,11 +603,8 @@ following image formats are supported:
    @tab Texture dictionaries used by the Renderware Engine.
@item RL2 video              @tab     @tab  X
    @tab used in some games by Entertainment Software Partners
-@item SGI RLE 8-bit          @tab     @tab  X
@item Sierra VMD video       @tab     @tab  X
    @tab Used in Sierra VMD files.
-@item Silicon Graphics Motion Video Compressor 1 (MVC1)  @tab     @tab  X
-@item Silicon Graphics Motion Video Compressor 2 (MVC2)  @tab     @tab  X
@item Smacker video          @tab     @tab  X
    @tab Video encoding used in Smacker.
@item SMPTE VC-1             @tab     @tab  X
@@ -743,13 +619,11 @@ following image formats are supported:
    @tab fourcc: SP5X
@item TechSmith Screen Capture Codec  @tab     @tab  X
    @tab fourcc: TSCC
-@item TechSmith Screen Capture Codec 2  @tab     @tab  X
-    @tab fourcc: TSC2
@item Theora                 @tab  E  @tab  X
    @tab encoding supported through external library libtheora
@item Tiertex Limited SEQ video  @tab     @tab  X
    @tab Codec used in DOS CD-ROM FlashBack game.
-@item Ut Video               @tab  X  @tab  X
+@item Ut Video               @tab     @tab  X
@item v210 QuickTime uncompressed 4:2:2 10-bit     @tab  X  @tab  X
@item v308 QuickTime uncompressed 4:4:4            @tab  X  @tab  X
@item v408 QuickTime uncompressed 4:4:4:4          @tab  X  @tab  X
@@ -773,7 +647,6 @@ following image formats are supported:
@item Psygnosis YOP Video    @tab     @tab  X
@item yuv4                   @tab  X  @tab  X
    @tab libquicktime uncompressed packed 4:2:0
-@item ZeroCodec Lossless Video @tab     @tab  X
@item ZLIB                   @tab  X  @tab  X
    @tab part of LCL, encoder experimental
@item Zip Motion Blocks Video  @tab   X @tab  X
@@ -788,8 +661,7 @@ following image formats are supported:

@multitable @columnfractions .4 .1 .1 .4
@item Name @tab Encoding @tab Decoding @tab Comments
-@item 8SVX exponential       @tab     @tab  X
-@item 8SVX fibonacci         @tab     @tab  X
+@item 8SVX audio             @tab     @tab  X
@item AAC+                   @tab  E  @tab  X
    @tab encoding supported through external library libaacplus
@item AAC                    @tab  E  @tab  X
@@ -820,21 +692,19 @@ following image formats are supported:
@item ADPCM IMA Westwood     @tab     @tab  X
@item ADPCM ISS IMA          @tab     @tab  X
    @tab Used in FunCom games.
-@item ADPCM IMA Dialogic     @tab     @tab  X
@item ADPCM IMA Duck DK3     @tab     @tab  X
    @tab Used in some Sega Saturn console games.
@item ADPCM IMA Duck DK4     @tab     @tab  X
    @tab Used in some Sega Saturn console games.
-@item ADPCM IMA Radical      @tab     @tab  X
@item ADPCM Microsoft        @tab  X  @tab  X
@item ADPCM MS IMA           @tab  X  @tab  X
-@item ADPCM Nintendo Gamecube AFC  @tab     @tab  X
-@item ADPCM Nintendo Gamecube DTK  @tab     @tab  X
@item ADPCM Nintendo Gamecube THP  @tab     @tab  X
@item ADPCM QT IMA           @tab  X  @tab  X
@item ADPCM SEGA CRI ADX     @tab  X  @tab  X
    @tab Used in Sega Dreamcast games.
@item ADPCM Shockwave Flash  @tab  X  @tab  X
+@item ADPCM SMJPEG IMA       @tab     @tab  X
+    @tab Used in certain Loki game ports.
@item ADPCM Sound Blaster Pro 2-bit  @tab     @tab  X
@item ADPCM Sound Blaster Pro 2.6-bit  @tab     @tab  X
@item ADPCM Sound Blaster Pro 4-bit  @tab     @tab  X
@@ -845,12 +715,10 @@ following image formats are supported:
    @tab encoding supported through external library libopencore-amrnb
@item AMR-WB                 @tab  E  @tab  X
    @tab encoding supported through external library libvo-amrwbenc
-@item Amazing Studio PAF Audio @tab     @tab  X
@item Apple lossless audio   @tab  X  @tab  X
    @tab QuickTime fourcc 'alac'
-@item ATRAC1                 @tab     @tab  X
-@item ATRAC3                 @tab     @tab  X
-@item ATRAC3+                @tab     @tab  X
+@item Atrac 1                @tab     @tab  X
+@item Atrac 3                @tab     @tab  X
@item Bink Audio             @tab     @tab  X
    @tab Used in Bink and Smacker files in many games.
@item CELT                   @tab     @tab  E
@@ -873,7 +741,6 @@ following image formats are supported:
@item DSP Group TrueSpeech   @tab     @tab  X
@item DV audio               @tab     @tab  X
@item Enhanced AC-3          @tab  X  @tab  X
-@item EVRC (Enhanced Variable Rate Codec) @tab     @tab  X
@item FLAC (Free Lossless Audio Codec)  @tab  X  @tab  IX
@item G.723.1                @tab X @tab X
@item G.729                  @tab     @tab  X
@@ -881,33 +748,24 @@ following image formats are supported:
    @tab encoding supported through external library libgsm
@item GSM Microsoft variant  @tab  E  @tab  X
    @tab encoding supported through external library libgsm
-@item IAC (Indeo Audio Coder)  @tab     @tab  X
-@item iLBC (Internet Low Bitrate Codec) @tab  E  @tab  E
-    @tab encoding and decoding supported through external library libilbc
@item IMC (Intel Music Coder)  @tab     @tab  X
@item MACE (Macintosh Audio Compression/Expansion) 3:1  @tab     @tab  X
@item MACE (Macintosh Audio Compression/Expansion) 6:1  @tab     @tab  X
@item MLP (Meridian Lossless Packing)  @tab     @tab  X
    @tab Used in DVD-Audio discs.
@item Monkey's Audio         @tab     @tab  X
+    @tab Only versions 3.97-3.99 are supported.
@item MP1 (MPEG audio layer 1)  @tab     @tab IX
@item MP2 (MPEG audio layer 2)  @tab IX  @tab IX
-    @tab libtwolame can be used alternatively for encoding.
@item MP3 (MPEG audio layer 3)  @tab  E  @tab IX
    @tab encoding supported through external library LAME, ADU MP3 and MP3onMP4 also supported
@item MPEG-4 Audio Lossless Coding (ALS)  @tab     @tab  X
@item Musepack SV7           @tab     @tab  X
@item Musepack SV8           @tab     @tab  X
@item Nellymoser Asao        @tab  X  @tab  X
-@item Opus                   @tab  E  @tab  E
-    @tab supported through external library libopus
@item PCM A-law              @tab  X  @tab  X
@item PCM mu-law             @tab  X  @tab  X
-@item PCM signed 8-bit planar  @tab  X  @tab  X
-@item PCM signed 16-bit big-endian planar  @tab  X  @tab  X
-@item PCM signed 16-bit little-endian planar  @tab  X  @tab  X
-@item PCM signed 24-bit little-endian planar  @tab  X  @tab  X
-@item PCM signed 32-bit little-endian planar  @tab  X  @tab  X
+@item PCM 16-bit little-endian planar  @tab     @tab  X
@item PCM 32-bit floating point big-endian  @tab  X  @tab  X
@item PCM 32-bit floating point little-endian  @tab  X  @tab  X
@item PCM 64-bit floating point big-endian  @tab  X  @tab  X
@@ -944,24 +802,20 @@ following image formats are supported:
@item Sierra VMD audio       @tab     @tab  X
    @tab Used in Sierra VMD files.
@item Smacker audio          @tab     @tab  X
-@item SMPTE 302M AES3 audio  @tab  X  @tab  X
+@item SMPTE 302M AES3 audio  @tab     @tab  X
@item Sonic                  @tab  X  @tab  X
    @tab experimental codec
@item Sonic lossless         @tab  X  @tab  X
    @tab experimental codec
@item Speex                  @tab  E  @tab  E
    @tab supported through external library libspeex
-@item TAK (Tom's lossless Audio Kompressor)  @tab     @tab  X
-@item True Audio (TTA)       @tab  X  @tab  X
+@item True Audio (TTA)       @tab     @tab  X
@item TrueHD                 @tab     @tab  X
    @tab Used in HD-DVD and Blu-Ray discs.
@item TwinVQ (VQF flavor)    @tab     @tab  X
-@item VIMA                   @tab     @tab  X
-    @tab Used in LucasArts SMUSH animations.
@item Vorbis                 @tab  E  @tab  X
    @tab A native but very primitive encoder exists.
-@item Voxware MetaSound      @tab     @tab  X
-@item WavPack                @tab  X  @tab  X
+@item WavPack                @tab     @tab  X
@item Westwood Audio (SND1)  @tab     @tab  X
@item Windows Media Audio 1  @tab  X  @tab  X
@item Windows Media Audio 2  @tab  X  @tab  X
@@ -981,65 +835,35 @@ performance on systems without hardware floating point support).

@multitable @columnfractions .4 .1 .1 .1 .1
@item Name @tab Muxing @tab Demuxing @tab Encoding @tab Decoding
-@item 3GPP Timed Text  @tab   @tab   @tab X @tab X
-@item AQTitle          @tab   @tab X @tab   @tab X
-@item DVB              @tab X @tab X @tab X @tab X
-@item DVB teletext     @tab   @tab X @tab   @tab E
-@item DVD              @tab X @tab X @tab X @tab X
-@item JACOsub          @tab X @tab X @tab   @tab X
-@item MicroDVD         @tab X @tab X @tab   @tab X
-@item MPL2             @tab   @tab X @tab   @tab X
-@item MPsub (MPlayer)  @tab   @tab X @tab   @tab X
-@item PGS              @tab   @tab   @tab   @tab X
-@item PJS (Phoenix)    @tab   @tab X @tab   @tab X
-@item RealText         @tab   @tab X @tab   @tab X
-@item SAMI             @tab   @tab X @tab   @tab X
-@item SSA/ASS          @tab X @tab X @tab X @tab X
-@item SubRip (SRT)     @tab X @tab X @tab X @tab X
-@item SubViewer v1     @tab   @tab X @tab   @tab X
-@item SubViewer        @tab   @tab X @tab   @tab X
-@item TED Talks captions @tab @tab X @tab   @tab X
-@item VobSub (IDX+SUB) @tab   @tab X @tab   @tab X
-@item VPlayer          @tab   @tab X @tab   @tab X
-@item WebVTT           @tab X @tab X @tab   @tab X
-@item XSUB             @tab   @tab   @tab X @tab X
+@item SSA/ASS      @tab X @tab X @tab X @tab X
+@item DVB          @tab X @tab X @tab X @tab X
+@item DVD          @tab X @tab X @tab X @tab X
+@item JACOsub      @tab X @tab X @tab   @tab X
+@item MicroDVD     @tab X @tab X @tab   @tab X
+@item PGS          @tab   @tab   @tab   @tab X
+@item SubRip (SRT) @tab X @tab X @tab X @tab X
+@item XSUB         @tab   @tab   @tab X @tab X
@end multitable

@code{X} means that the feature is supported.

-@code{E} means that support is provided through an external library.
-
@section Network Protocols

@multitable @columnfractions .4 .1
@item Name         @tab Support
+@item Apple HTTP Live Streaming @tab X
@item file         @tab X
-@item FTP          @tab X
@item Gopher       @tab X
-@item HLS          @tab X
@item HTTP         @tab X
-@item HTTPS        @tab X
-@item MMSH         @tab X
-@item MMST         @tab X
+@item MMS          @tab X
@item pipe         @tab X
-@item RTMP         @tab X
-@item RTMPE        @tab X
-@item RTMPS        @tab X
-@item RTMPT        @tab X
-@item RTMPTE       @tab X
-@item RTMPTS       @tab X
@item RTP          @tab X
-@item SCTP         @tab X
-@item SFTP         @tab E
@item TCP          @tab X
-@item TLS          @tab X
@item UDP          @tab X
@end multitable

@code{X} means that the protocol is supported.

-@code{E} means that support is provided through an external library.
-

@section Input/Output Devices

@@ -1047,19 +871,14 @@ performance on systems without hardware floating point support).
@item Name              @tab Input  @tab Output
@item ALSA              @tab X      @tab X
@item BKTR              @tab X      @tab
-@item caca              @tab        @tab X
@item DV1394            @tab X      @tab
-@item Lavfi virtual device @tab X   @tab
-@item Linux framebuffer @tab X      @tab X
+@item Linux framebuffer @tab X      @tab
@item JACK              @tab X      @tab
@item LIBCDIO           @tab X
@item LIBDC1394         @tab X      @tab
-@item OpenAL            @tab X
-@item OpenGL            @tab        @tab X
@item OSS               @tab X      @tab X
-@item PulseAudio        @tab X      @tab X
-@item SDL               @tab        @tab X
-@item Video4Linux2      @tab X      @tab X
+@item Pulseaudio        @tab X      @tab
+@item Video4Linux2      @tab X      @tab
@item VfW capture       @tab X      @tab
@item X11 grabbing      @tab X      @tab
@end multitable
@@ -1070,10 +889,9 @@ performance on systems without hardware floating point support).

@multitable @columnfractions .4 .1 .1
@item Codec/format      @tab Read   @tab Write
-@item AVI               @tab X      @tab X
@item DV                @tab X      @tab X
@item GXF               @tab X      @tab X
-@item MOV               @tab X      @tab X
+@item MOV               @tab X      @tab
@item MPEG1/2           @tab X      @tab X
@item MXF               @tab X      @tab X
@end multitable
--- a/doc/git-howto.texi
+++ b/doc/git-howto.texi
@@ -258,32 +258,6 @@ git commit
@end example


-@chapter Git configuration
-
-In order to simplify a few workflows, it is advisable to configure both
-your personal Git installation and your local FFmpeg repository.
-
-@section Personal Git installation
-
-Add the following to your @file{~/.gitconfig} to help @command{git send-email}
-and @command{git format-patch} detect renames:
-
-@example
-[diff]
-        renames = copy
-@end example
-
-@section Repository configuration
-
-In order to have @command{git send-email} automatically send patches
-to the ffmpeg-devel mailing list, add the following stanza
-to @file{/path/to/ffmpeg/repository/.git/config}:
-
-@example
-[sendemail]
-        to = ffmpeg-devel@@ffmpeg.org
-@end example
-
@chapter FFmpeg specific

@section Reverting broken commits
@@ -372,43 +346,6 @@ git checkout -b svn_23456 $SHA1

 where @var{$SHA1} is the commit hash from the @command{git log} output.

-
-@chapter pre-push checklist
-
-Once you have a set of commits that you feel are ready for pushing,
-work through the following checklist to doublecheck everything is in
-proper order. This list tries to be exhaustive. In case you are just
-pushing a typo in a comment, some of the steps may be unnecessary.
-Apply your common sense, but if in doubt, err on the side of caution.
-
-First, make sure that the commits and branches you are going to push
-match what you want pushed and that nothing is missing, extraneous or
-wrong. You can see what will be pushed by running the git push command
-with --dry-run first. And then inspecting the commits listed with
-@command{git log -p 1234567..987654}. The @command{git status} command
-may help in finding local changes that have been forgotten to be added.
-
-Next let the code pass through a full run of our testsuite.
-
-@itemize
-@item @command{make distclean}
-@item @command{/path/to/ffmpeg/configure}
-@item @command{make check}
-@item if fate fails due to missing samples run @command{make fate-rsync} and retry
-@end itemize
-
-Make sure all your changes have been checked before pushing them, the
-testsuite only checks against regressions and that only to some extend. It does
-obviously not check newly added features/code to be working unless you have
-added a test for that (which is recommended).
-
-Also note that every single commit should pass the test suite, not just
-the result of a series of patches.
-
-Once everything passed, push the changes to your public ffmpeg clone and post a
-merge request to ffmpeg-devel. You can also push them directly but this is not
-recommended.
-
@chapter Server Issues

 Contact the project admins @email{root@@ffmpeg.org} if you have technical
--- a/doc/git-howto.txt
+++ b/doc/git-howto.txt
@@ -0,0 +1,273 @@
+
+About Git write access:
+~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+
+Before everything else, you should know how to use GIT properly.
+Luckily Git comes with excellent documentation.
+
+  git --help
+  man git
+
+shows you the available subcommands,
+
+  git <command> --help
+  man git-<command>
+
+shows information about the subcommand <command>.
+
+The most comprehensive manual is the website Git Reference
+
+http://gitref.org/
+
+For more information about the Git project, visit
+
+http://git-scm.com/
+
+Consult these resources whenever you have problems, they are quite exhaustive.
+
+You do not need a special username or password.
+All you need is to provide a ssh public key to the Git server admin.
+
+What follows now is a basic introduction to Git and some FFmpeg-specific
+guidelines. Read it at least once, if you are granted commit privileges to the
+FFmpeg project you are expected to be familiar with these rules.
+
+
+
+I. BASICS:
+==========
+
+0. Get GIT:
+
+  Most distributions have a git package, if not
+  You can get git from http://git-scm.com/
+
+
+1. Cloning the source tree:
+
+    git clone git://source.ffmpeg.org/ffmpeg <target>
+
+  This will put the FFmpeg sources into the directory <target>.
+
+    git clone git@source.ffmpeg.org:ffmpeg <target>
+
+  This will put the FFmpeg sources into the directory <target> and let
+  you push back your changes to the remote repository.
+
+
+2. Updating the source tree to the latest revision:
+
+    git pull (--ff-only)
+
+  pulls in the latest changes from the tracked branch. The tracked branch
+  can be remote. By default the master branch tracks the branch master in
+  the remote origin.
+  Caveat: Since merge commits are forbidden at least for the initial
+          months of git --ff-only or --rebase (see below) are recommended.
+          --ff-only will fail and not create merge commits if your branch
+          has diverged (has a different history) from the tracked branch.
+
+2.a Rebasing your local branches:
+
+    git pull --rebase
+
+  fetches the changes from the main repository and replays your local commits
+  over it. This is required to keep all your local changes at the top of
+  FFmpeg's master tree. The master tree will reject pushes with merge commits.
+
+
+3. Adding/removing files/directories:
+
+    git add [-A] <filename/dirname>
+    git rm [-r] <filename/dirname>
+
+  GIT needs to get notified of all changes you make to your working
+  directory that makes files appear or disappear.
+  Line moves across files are automatically tracked.
+
+
+4. Showing modifications:
+
+    git diff <filename(s)>
+
+  will show all local modifications in your working directory as unified diff.
+
+
+5. Inspecting the changelog:
+
+    git log <filename(s)>
+
+  You may also use the graphical tools like gitview or gitk or the web
+  interface available at http://source.ffmpeg.org
+
+6. Checking source tree status:
+
+    git status
+
+  detects all the changes you made and lists what actions will be taken in case
+  of a commit (additions, modifications, deletions, etc.).
+
+
+7. Committing:
+
+    git diff --check
+
+  to double check your changes before committing them to avoid trouble later
+  on. All experienced developers do this on each and every commit, no matter
+  how small.
+  Every one of them has been saved from looking like a fool by this many times.
+  It's very easy for stray debug output or cosmetic modifications to slip in,
+  please avoid problems through this extra level of scrutiny.
+
+  For cosmetics-only commits you should get (almost) empty output from
+
+    git diff -w -b <filename(s)>
+
+  Also check the output of
+
+    git status
+
+  to make sure you don't have untracked files or deletions.
+
+    git add [-i|-p|-A] <filenames/dirnames>
+
+  Make sure you have told git your name and email address, e.g. by running
+    git config --global user.name "My Name"
+    git config --global user.email my@email.invalid
+  (--global to set the global configuration for all your git checkouts).
+
+  Git will select the changes to the files for commit. Optionally you can use
+  the interactive or the patch mode to select hunk by hunk what should be
+  added to the commit.
+
+    git commit
+
+  Git will commit the selected changes to your current local branch.
+
+  You will be prompted for a log message in an editor, which is either
+  set in your personal configuration file through
+
+    git config core.editor
+
+  or set by one of the following environment variables:
+  GIT_EDITOR, VISUAL or EDITOR.
+
+  Log messages should be concise but descriptive. Explain why you made a change,
+  what you did will be obvious from the changes themselves most of the time.
+  Saying just "bug fix" or "10l" is bad. Remember that people of varying skill
+  levels look at and educate themselves while reading through your code. Don't
+  include filenames in log messages, Git provides that information.
+
+  Possibly make the commit message have a terse, descriptive first line, an
+  empty line and then a full description. The first line will be used to name
+  the patch by git format-patch.
+
+
+8. Renaming/moving/copying files or contents of files:
+
+  Git automatically tracks such changes, making those normal commits.
+
+    mv/cp path/file otherpath/otherfile
+
+    git add [-A] .
+
+    git commit
+
+  Do not move, rename or copy files of which you are not the maintainer without
+  discussing it on the mailing list first!
+
+9. Reverting broken commits
+
+    git revert <commit>
+
+  git revert will generate a revert commit. This will not make the faulty
+  commit disappear from the history.
+
+    git reset <commit>
+
+  git reset will uncommit the changes till <commit> rewriting the current
+  branch history.
+
+    git commit --amend
+
+  allows to amend the last commit details quickly.
+
+    git rebase -i origin/master
+
+  will replay local commits over the main repository allowing to edit,
+  merge or remove some of them in the process.
+
+  Note that the reset, commit --amend and rebase rewrite history, so you
+  should use them ONLY on your local or topic branches.
+
+  The main repository will reject those changes.
+
+10. Preparing a patchset.
+
+    git format-patch <commit> [-o directory]
+
+  will generate a set of patches for each commit between <commit> and
+  current HEAD. E.g.
+
+    git format-patch origin/master
+
+  will generate patches for all commits on current branch which are not
+  present in upstream.
+  A useful shortcut is also
+
+    git format-patch -n
+
+  which will generate patches from last n commits.
+  By default the patches are created in the current directory.
+
+11. Sending patches for review
+
+    git send-email <commit list|directory>
+
+  will send the patches created by git format-patch or directly generates
+  them. All the email fields can be configured in the global/local
+  configuration or overridden by command line.
+  Note that this tool must often be installed separately (e.g. git-email
+  package on Debian-based distros).
+
+12. Pushing changes to remote trees
+
+    git push
+
+  Will push the changes to the default remote (origin).
+  Git will prevent you from pushing changes if the local and remote trees are
+  out of sync. Refer to 2 and 2.a to sync the local tree.
+
+    git remote add <name> <url>
+
+  Will add additional remote with a name reference, it is useful if you want
+  to push your local branch for review on a remote host.
+
+    git push <remote> <refspec>
+
+  Will push the changes to the remote repository. Omitting refspec makes git
+  push update all the remote branches matching the local ones.
+
+13. Finding a specific svn revision
+
+  Since version 1.7.1 git supports ':/foo' syntax for specifying commits
+  based on a regular expression. see man gitrevisions
+
+    git show :/'as revision 23456'
+
+  will show the svn changeset r23456. With older git versions searching in
+  the git log output is the easiest option (especially if a pager with
+  search capabilities is used).
+  This commit can be checked out with
+
+    git checkout -b svn_23456 :/'as revision 23456'
+
+  or for git < 1.7.1 with
+
+    git checkout -b svn_23456 $SHA1
+
+  where $SHA1 is the commit SHA1 from the 'git log' output.
+
+
+Contact the project admins <root at ffmpeg dot org> if you have technical
+problems with the GIT server.
--- a/doc/indevs.texi
+++ b/doc/indevs.texi
@@ -86,7 +86,7 @@ fail to open.
 Set the video size in the captured video.

@item framerate
-Set the frame rate in the captured video.
+Set the framerate in the captured video.

@item sample_rate
 Set the sample rate (in Hz) of the captured audio.
@@ -112,19 +112,6 @@ defaults to 0).
 Set audio device number for devices with same name (starts at 0,
 defaults to 0).

-@item pixel_format
-Select pixel format to be used by DirectShow. This may only be set when
-the video codec is not set or set to rawvideo.
-
-@item audio_buffer_size
-Set audio device buffer size in milliseconds (which can directly
-impact latency, depending on the device).
-Defaults to using the audio device's
-default buffer size (typically some multiple of 500ms).
-Setting this value too low can degrade performance.
-See also
-@url{http://msdn.microsoft.com/en-us/library/windows/desktop/dd377582(v=vs.85).aspx}
-
@end table

@subsection Examples
@@ -192,66 +179,6 @@ ffmpeg -f fbdev -frames:v 1 -r 1 -i /dev/fb0 screenshot.jpeg

 See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1).

-@section iec61883
-
-FireWire DV/HDV input device using libiec61883.
-
-To enable this input device, you need libiec61883, libraw1394 and
-libavc1394 installed on your system. Use the configure option
-@code{--enable-libiec61883} to compile with the device enabled.
-
-The iec61883 capture device supports capturing from a video device
-connected via IEEE1394 (FireWire), using libiec61883 and the new Linux
-FireWire stack (juju). This is the default DV/HDV input method in Linux
-Kernel 2.6.37 and later, since the old FireWire stack was removed.
-
-Specify the FireWire port to be used as input file, or "auto"
-to choose the first port connected.
-
-@subsection Options
-
-@table @option
-
-@item dvtype
-Override autodetection of DV/HDV. This should only be used if auto
-detection does not work, or if usage of a different device type
-should be prohibited. Treating a DV device as HDV (or vice versa) will
-not work and result in undefined behavior.
-The values @option{auto}, @option{dv} and @option{hdv} are supported.
-
-@item dvbuffer
-Set maxiumum size of buffer for incoming data, in frames. For DV, this
-is an exact value. For HDV, it is not frame exact, since HDV does
-not have a fixed frame size.
-
-@item dvguid
-Select the capture device by specifying it's GUID. Capturing will only
-be performed from the specified device and fails if no device with the
-given GUID is found. This is useful to select the input if multiple
-devices are connected at the same time.
-Look at /sys/bus/firewire/devices to find out the GUIDs.
-
-@end table
-
-@subsection Examples
-
-@itemize
-
-@item
-Grab and show the input of a FireWire DV/HDV device.
-@example
-ffplay -f iec61883 -i auto
-@end example
-
-@item
-Grab and record the input of a FireWire DV/HDV device,
-using a packet buffer of 100000 packets if the source is HDV.
-@example
-ffmpeg -f iec61883 -i auto -hdvbuffer 100000 out.mpg
-@end example
-
-@end itemize
-
@section jack

 JACK input device.
@@ -327,12 +254,6 @@ label, but all the others need to be specified explicitly.

 If not specified defaults to the filename specified for the input
 device.
-
-@item graph_file
-Set the filename of the filtergraph to be read and sent to the other
-filters. Syntax of the filtergraph is the same as the one specified by
-the option @var{graph}.
-
@end table

@subsection Examples
@@ -341,14 +262,14 @@ the option @var{graph}.
@item
 Create a color video stream and play it back with @command{ffplay}:
@example
-ffplay -f lavfi -graph "color=c=pink [out0]" dummy
+ffplay -f lavfi -graph "color=pink [out0]" dummy
@end example

@item
 As the previous example, but use filename for specifying the graph
 description, and omit the "out0" label:
@example
-ffplay -f lavfi color=c=pink
+ffplay -f lavfi color=pink
@end example

@item
@@ -485,52 +406,87 @@ For more information about OSS see:

@section pulse

-PulseAudio input device.
+pulseaudio input device.

-To enable this output device you need to configure FFmpeg with @code{--enable-libpulse}.
+To enable this input device during configuration you need libpulse-simple
+installed in your system.

 The filename to provide to the input device is a source device or the
 string "default"

-To list the PulseAudio source devices and their properties you can invoke
+To list the pulse source devices and their properties you can invoke
 the command @command{pactl list sources}.

-More information about PulseAudio can be found on @url{http://www.pulseaudio.org}.
-
-@subsection Options
-@table @option
-@item server
-Connect to a specific PulseAudio server, specified by an IP address.
-Default server is used when not provided.
-
-@item name
-Specify the application name PulseAudio will use when showing active clients,
-by default it is the @code{LIBAVFORMAT_IDENT} string.
-
-@item stream_name
-Specify the stream name PulseAudio will use when showing active streams,
-by default it is "record".
-
-@item sample_rate
-Specify the samplerate in Hz, by default 48kHz is used.
-
-@item channels
-Specify the channels in use, by default 2 (stereo) is set.
-
-@item frame_size
-Specify the number of bytes per frame, by default it is set to 1024.
-
-@item fragment_size
-Specify the minimal buffering fragment in PulseAudio, it will affect the
-audio latency. By default it is unset.
-@end table
-
-@subsection Examples
-Record a stream from default device:
@example
 ffmpeg -f pulse -i default /tmp/pulse.wav
@end example

+@subsection @var{server} AVOption
+
+The syntax is:
+@example
+-server @var{server name}
+@end example
+
+Connects to a specific server.
+
+@subsection @var{name} AVOption
+
+The syntax is:
+@example
+-name @var{application name}
+@end example
+
+Specify the application name pulse will use when showing active clients,
+by default it is the LIBAVFORMAT_IDENT string
+
+@subsection @var{stream_name} AVOption
+
+The syntax is:
+@example
+-stream_name @var{stream name}
+@end example
+
+Specify the stream name pulse will use when showing active streams,
+by default it is "record"
+
+@subsection @var{sample_rate} AVOption
+
+The syntax is:
+@example
+-sample_rate @var{samplerate}
+@end example
+
+Specify the samplerate in Hz, by default 48kHz is used.
+
+@subsection @var{channels} AVOption
+
+The syntax is:
+@example
+-channels @var{N}
+@end example
+
+Specify the channels in use, by default 2 (stereo) is set.
+
+@subsection @var{frame_size} AVOption
+
+The syntax is:
+@example
+-frame_size @var{bytes}
+@end example
+
+Specify the number of byte per frame, by default it is set to 1024.
+
+@subsection @var{fragment_size} AVOption
+
+The syntax is:
+@example
+-fragment_size @var{bytes}
+@end example
+
+Specify the minimal buffering fragment in pulseaudio, it will affect the
+audio latency. By default it is unset.
+
@section sndio

 sndio input device.
@@ -548,16 +504,10 @@ command:
 ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav
@end example

-@section video4linux2, v4l2
+@section video4linux2

 Video4Linux2 input video device.

-"v4l2" can be used as alias for "video4linux2".
-
-If FFmpeg is built with v4l-utils support (by using the
-@code{--enable-libv4l2} configure option), it is possible to use it with the
-@code{-use_libv4l2} input device option.
-
 The name of the device to grab is a file device node, usually Linux
 systems tend to automatically create such nodes when the device
 (e.g. an USB webcam) is plugged into the system, and has a name of the
@@ -565,10 +515,10 @@ kind @file{/dev/video@var{N}}, where @var{N} is a number associated to
 the device.

 Video4Linux2 devices usually support a limited set of
-@var{width}x@var{height} sizes and frame rates. You can check which are
+@var{width}x@var{height} sizes and framerates. You can check which are
 supported using @command{-list_formats all} for Video4Linux2 devices.
-Some devices, like TV cards, support one or more standards. It is possible
-to list all the supported standards using @command{-list_standards all}.
+
+Some usage examples of the video4linux2 devices with ffmpeg and ffplay:

 The time base for the timestamps is 1 microsecond. Depending on the kernel
 version and configuration, the timestamps may be derived from the real time
@@ -577,94 +527,19 @@ boot time, unaffected by NTP or manual changes to the clock). The
@option{-timestamps abs} or @option{-ts abs} option can be used to force
 conversion into the real time clock.

-Some usage examples of the video4linux2 device with @command{ffmpeg}
-and @command{ffplay}:
-@itemize
-@item
-Grab and show the input of a video4linux2 device:
+Note that if FFmpeg is build with v4l-utils support ("--enable-libv4l2"
+option), it will always be used.
@example
+# Grab and show the input of a video4linux2 device.
 ffplay -f video4linux2 -framerate 30 -video_size hd720 /dev/video0
-@end example

-@item
-Grab and record the input of a video4linux2 device, leave the
-frame rate and size as previously set:
-@example
+# Grab and record the input of a video4linux2 device, leave the
+framerate and size as previously set.
 ffmpeg -f video4linux2 -input_format mjpeg -i /dev/video0 out.mpeg
@end example
-@end itemize

-For more information about Video4Linux, check @url{http://linuxtv.org/}.
-
-@subsection Options
-
-@table @option
-@item standard
-Set the standard. Must be the name of a supported standard. To get a
-list of the supported standards, use the @option{list_standards}
-option.
-
-@item channel
-Set the input channel number. Default to -1, which means using the
-previously selected channel.
-
-@item video_size
-Set the video frame size. The argument must be a string in the form
-@var{WIDTH}x@var{HEIGHT} or a valid size abbreviation.
-
-@item pixel_format
-Select the pixel format (only valid for raw video input).
-
-@item input_format
-Set the preferred pixel format (for raw video) or a codec name.
-This option allows to select the input format, when several are
-available.
-
-@item framerate
-Set the preferred video frame rate.
-
-@item list_formats
-List available formats (supported pixel formats, codecs, and frame
-sizes) and exit.
-
-Available values are:
-@table @samp
-@item all
-Show all available (compressed and non-compressed) formats.
-
-@item raw
-Show only raw video (non-compressed) formats.
-
-@item compressed
-Show only compressed formats.
-@end table
-
-@item list_standards
-List supported standards and exit.
-
-Available values are:
-@table @samp
-@item all
-Show all supported standards.
-@end table
-
-@item timestamps, ts
-Set type of timestamps for grabbed frames.
-
-Available values are:
-@table @samp
-@item default
-Use timestamps from the kernel.
-
-@item abs
-Use absolute timestamps (wall clock).
-
-@item mono2abs
-Force conversion from monotonic to absolute timestamps.
-@end table
-
-Default value is @code{default}.
-@end table
+"v4l" and "v4l2" can be used as aliases for the respective "video4linux" and
+"video4linux2".

@section vfwcap

@@ -701,25 +576,19 @@ properties of your X11 display (e.g. grep for "name" or "dimensions").

 For example to grab from @file{:0.0} using @command{ffmpeg}:
@example
-ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0 out.mpg
+ffmpeg -f x11grab -r 25 -s cif -i :0.0 out.mpg
+
+# Grab at position 10,20.
+ffmpeg -f x11grab -r 25 -s cif -i :0.0+10,20 out.mpg
@end example

-Grab at position @code{10,20}:
+@subsection @var{follow_mouse} AVOption
+
+The syntax is:
@example
-ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0+10,20 out.mpg
+-follow_mouse centered|@var{PIXELS}
@end example

-@subsection Options
-
-@table @option
-@item draw_mouse
-Specify whether to draw the mouse pointer. A value of @code{0} specify
-not to draw the pointer. Default value is @code{1}.
-
-@item follow_mouse
-Make the grabbed area follow the mouse. The argument can be
-@code{centered} or a number of pixels @var{PIXELS}.
-
 When it is specified with "centered", the grabbing region follows the mouse
 pointer and keeps the pointer at the center of region; otherwise, the region
 follows only when the mouse pointer reaches within @var{PIXELS} (greater than
@@ -727,37 +596,29 @@ zero) to the edge of region.

 For example:
@example
-ffmpeg -f x11grab -follow_mouse centered -framerate 25 -video_size cif -i :0.0 out.mpg
+ffmpeg -f x11grab -follow_mouse centered -r 25 -s cif -i :0.0 out.mpg
+
+# Follows only when the mouse pointer reaches within 100 pixels to edge
+ffmpeg -f x11grab -follow_mouse 100 -r 25 -s cif -i :0.0 out.mpg
@end example

-To follow only when the mouse pointer reaches within 100 pixels to edge:
+@subsection @var{show_region} AVOption
+
+The syntax is:
@example
-ffmpeg -f x11grab -follow_mouse 100 -framerate 25 -video_size cif -i :0.0 out.mpg
+-show_region 1
@end example

-@item framerate
-Set the grabbing frame rate. Default value is @code{ntsc},
-corresponding to a frame rate of @code{30000/1001}.
-
-@item show_region
-Show grabbed region on screen.
-
-If @var{show_region} is specified with @code{1}, then the grabbing
-region will be indicated on screen. With this option, it is easy to
-know what is being grabbed if only a portion of the screen is grabbed.
+If @var{show_region} AVOption is specified with @var{1}, then the grabbing
+region will be indicated on screen. With this option, it's easy to know what is
+being grabbed if only a portion of the screen is grabbed.

 For example:
@example
-ffmpeg -f x11grab -show_region 1 -framerate 25 -video_size cif -i :0.0+10,20 out.mpg
-@end example
+ffmpeg -f x11grab -show_region 1 -r 25 -s cif -i :0.0+10,20 out.mpg

-With @var{follow_mouse}:
-@example
-ffmpeg -f x11grab -follow_mouse centered -show_region 1 -framerate 25 -video_size cif -i :0.0 out.mpg
+# With follow_mouse
+ffmpeg -f x11grab -follow_mouse centered -show_region 1  -r 25 -s cif -i :0.0 out.mpg
@end example

-@item video_size
-Set the video frame size. Default value is @code{vga}.
-@end table
-
@c man end INPUT DEVICES
--- a/doc/issue_tracker.txt
+++ b/doc/issue_tracker.txt
@@ -1,4 +1,4 @@
-FFmpeg's bug/feature request tracker manual
+FFmpeg's bug/patch/feature request tracker manual
 =================================================

 NOTE: This is a draft.
@@ -11,7 +11,7 @@ existing issues can be done through a web interface.

 Issues can be different kinds of things we want to keep track of
 but that do not belong into the source tree itself. This includes
-bug reports, feature requests and license violations. We
+bug reports, patches, feature requests and license violations. We
 might add more items to this list in the future, so feel free to
 propose a new `type of issue' on the ffmpeg-devel mailing list if
 you feel it is worth tracking.
@@ -24,13 +24,10 @@ a mail for every change to every issue.
 The subscription URL for the ffmpeg-trac list is:
 http(s)://ffmpeg.org/mailman/listinfo/ffmpeg-trac
 The URL of the webinterface of the tracker is:
-http(s)://trac.ffmpeg.org
+http(s)://ffmpeg.org/trac/ffmpeg

 Type:
 -----
-art
-    Artwork such as photos, music, banners, and logos.
-
 bug / defect
    An error, flaw, mistake, failure, or fault in FFmpeg or libav* that
    prevents it from behaving as intended.
@@ -44,18 +41,20 @@ feature request / enhancement
 license violation
    ticket to keep track of (L)GPL violations of ffmpeg by others

-sponsoring request
-    Developer requests for hardware, software, specifications, money,
-    refunds, etc.
+patch
+    A patch as generated by diff which conforms to the patch submission and
+    development policy.
+

 Priority:
 ---------
 critical
-    Bugs about data loss and security issues.
+    Bugs and patches which deal with data loss and security issues.
    No feature request can be critical.

 important
-    Bugs which make FFmpeg unusable for a significant number of users.
+    Bugs which make FFmpeg unusable for a significant number of users, and
+    patches fixing them.
    Examples here might be completely broken MPEG-4 decoding or a build issue
    on Linux.
    While broken 4xm decoding or a broken OS/2 build would not be important,
@@ -69,7 +68,7 @@ normal


 minor
-    Bugs about things like spelling errors, "mp2" instead of
+    Bugs and patches about things like spelling errors, "mp2" instead of
    "mp3" being shown and such.
    Feature requests about things few people want or which do not make a big
    difference.
@@ -104,13 +103,13 @@ This state implicates that the bug either has been reproduced or that
 reproduction is not needed as the bug is already understood.


-Type/Status:
+Type/Status/Substatus:
 ----------
-*/new
-    Initial state of new bugs and feature requests submitted by
+*/new/new
+    Initial state of new bugs, patches and feature requests submitted by
    users.

-*/open
+*/open/open
    Issues which have been briefly looked at and which did not look outright
    invalid.
    This implicates that no real more detailed state applies yet. Conversely,
@@ -118,7 +117,9 @@ Type/Status:
    looked at.

 */closed/duplicate
-    Bugs or feature requests which are duplicates.
+    Bugs, patches or feature requests which are duplicates.
+    Note that patches dealing with the same thing in a different way are not
+    duplicates.
    Note, if you mark something as duplicate, do not forget setting the
    superseder so bug reports are properly linked.

@@ -133,7 +134,7 @@ Type/Status:
 bug/closed/fixed
    Bugs which have to the best of our knowledge been fixed.

-bug/closed/wontfix
+bug/closed/wont_fix
    Bugs which we will not fix. Possible reasons include legality, high
    complexity for the sake of supporting obscure corner cases, speed loss
    for similarly esoteric purposes, et cetera.
@@ -147,15 +148,33 @@ bug/closed/works_for_me
    reproduction failed - that is the code seems to work correctly to the
    best of our knowledge.

-feature_request/closed/fixed
+patch/open/approved
+    Patches which have been reviewed and approved by a developer.
+    Such patches can be applied anytime by any other developer after some
+    reasonable testing (compile + regression tests + does the patch do
+    what the author claimed).
+
+patch/open/needs_changes
+    Patches which have been reviewed and need changes to be accepted.
+
+patch/closed/applied
+    Patches which have been applied.
+
+patch/closed/rejected
+    Patches which have been rejected.
+
+feature_request/closed/implemented
    Feature requests which have been implemented.

-feature_request/closed/wontfix
+feature_request/closed/wont_implement
    Feature requests which will not be implemented. The reasons here could
    be legal, philosophical or others.

+Note, please do not use type-status-substatus combinations other than the
+above without asking on ffmpeg-dev first!
+
 Note2, if you provide the requested info do not forget to remove the
-needs_more_info resolution.
+needs_more_info substatus.

 Component:
 ----------
--- a/doc/libavcodec.texi
+++ b/doc/libavcodec.texi
@@ -1,48 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle Libavcodec Documentation
-@titlepage
-@center @titlefont{Libavcodec Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-The libavcodec library provides a generic encoding/decoding framework
-and contains multiple decoders and encoders for audio, video and
-subtitle streams, and several bitstream filters.
-
-The shared architecture provides various services ranging from bit
-stream I/O to DSP optimizations, and makes it suitable for
-implementing robust and fast codecs as well as for experimentation.
-
-@c man end DESCRIPTION
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{ffmpeg-codecs.html,ffmpeg-codecs}, @url{ffmpeg-bitstream-filters.html,bitstream-filters},
-@url{libavutil.html,libavutil}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
-ffmpeg-codecs(1), ffmpeg-bitstream-filters(1),
-libavutil(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename libavcodec
-@settitle media streams decoding and encoding library
-
-@end ignore
-
-@bye
--- a/doc/libavdevice.texi
+++ b/doc/libavdevice.texi
@@ -1,45 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle Libavdevice Documentation
-@titlepage
-@center @titlefont{Libavdevice Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-The libavdevice library provides a generic framework for grabbing from
-and rendering to many common multimedia input/output devices, and
-supports several input and output devices, including Video4Linux2,
-VfW, DShow, and ALSA.
-
-@c man end DESCRIPTION
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{ffmpeg-devices.html,ffmpeg-devices},
-@url{libavutil.html,libavutil}, @url{libavcodec.html,libavcodec}, @url{libavformat.html,libavformat}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
-ffmpeg-devices(1),
-libavutil(3), libavcodec(3), libavformat(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename libavdevice
-@settitle multimedia device handling library
-
-@end ignore
-
-@bye
--- a/doc/libavfilter.texi
+++ b/doc/libavfilter.texi
@@ -9,36 +9,84 @@

@contents

-@chapter Description
-@c man begin DESCRIPTION
+@chapter Introduction

-The libavfilter library provides a generic audio/video filtering
-framework containing several filters, sources and sinks.
+Libavfilter is the filtering API of FFmpeg. It is the substitute of the
+now deprecated 'vhooks' and started as a Google Summer of Code project.

-@c man end DESCRIPTION
+Audio filtering integration into the main FFmpeg repository is a work in
+progress, so audio API and ABI should not be considered stable yet.

-@chapter See Also
+@chapter Tutorial

-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{ffmpeg-filters.html,ffmpeg-filters},
-@url{libavutil.html,libavutil}, @url{libswscale.html,libswscale}, @url{libswresample.html,libswresample},
-@url{libavcodec.html,libavcodec}, @url{libavformat.html,libavformat}, @url{libavdevice.html,libavdevice}
-@end ifhtml
+In libavfilter, it is possible for filters to have multiple inputs and
+multiple outputs.
+To illustrate the sorts of things that are possible, we can
+use a complex filter graph. For example, the following one:

-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
-ffmpeg-filters(1),
-libavutil(3), libswscale(3), libswresample(3), libavcodec(3), libavformat(3), libavdevice(3)
-@end ifnothtml
+@example
+input --> split --> fifo -----------------------> overlay --> output
+            |                                        ^
+            |                                        |
+            +------> fifo --> crop --> vflip --------+
+@end example

-@include authors.texi
+splits the stream in two streams, sends one stream through the crop filter
+and the vflip filter before merging it back with the other stream by
+overlaying it on top. You can use the following command to achieve this:

-@ignore
+@example
+ffmpeg -i input -vf "[in] split [T1], fifo, [T2] overlay=0:H/2 [out]; [T1] fifo, crop=iw:ih/2:0:ih/2, vflip [T2]" output
+@end example

-@setfilename libavfilter
-@settitle multimedia filtering library
+The result will be that in output the top half of the video is mirrored
+onto the bottom half.

-@end ignore
+Video filters are loaded using the @var{-vf} option passed to
+@command{ffmpeg} or to @command{ffplay}. Filters in the same linear
+chain are separated by commas. In our example, @var{split, fifo,
+overlay} are in one linear chain, and @var{fifo, crop, vflip} are in
+another. The points where the linear chains join are labeled by names
+enclosed in square brackets. In our example, that is @var{[T1]} and
+@var{[T2]}. The magic labels @var{[in]} and @var{[out]} are the points
+where video is input and output.
+
+Some filters take in input a list of parameters: they are specified
+after the filter name and an equal sign, and are separated each other
+by a semicolon.
+
+There exist so-called @var{source filters} that do not have a video
+input, and we expect in the future some @var{sink filters} that will
+not have video output.
+
+@chapter graph2dot
+
+The @file{graph2dot} program included in the FFmpeg @file{tools}
+directory can be used to parse a filter graph description and issue a
+corresponding textual representation in the dot language.
+
+Invoke the command:
+@example
+graph2dot -h
+@end example
+
+to see how to use @file{graph2dot}.
+
+You can then pass the dot description to the @file{dot} program (from
+the graphviz suite of programs) and obtain a graphical representation
+of the filter graph.
+
+For example the sequence of commands:
+@example
+echo @var{GRAPH_DESCRIPTION} | \
+tools/graph2dot -o graph.tmp && \
+dot -Tpng graph.tmp -o graph.png && \
+display graph.png
+@end example
+
+can be used to create and display an image representing the graph
+described by the @var{GRAPH_DESCRIPTION} string.
+
+@include filters.texi

@bye
--- a/doc/libavformat.texi
+++ b/doc/libavformat.texi
@@ -1,48 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle Libavformat Documentation
-@titlepage
-@center @titlefont{Libavformat Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-The libavformat library provides a generic framework for multiplexing
-and demultiplexing (muxing and demuxing) audio, video and subtitle
-streams. It encompasses multiple muxers and demuxers for multimedia
-container formats.
-
-It also supports several input and output protocols to access a media
-resource.
-
-@c man end DESCRIPTION
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{ffmpeg-formats.html,ffmpeg-formats}, @url{ffmpeg-protocols.html,ffmpeg-protocols},
-@url{libavutil.html,libavutil}, @url{libavcodec.html,libavcodec}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
-ffmpeg-formats(1), ffmpeg-protocols(1),
-libavutil(3), libavcodec(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename libavformat
-@settitle multimedia muxing and demuxing library
-
-@end ignore
-
-@bye
--- a/doc/libavutil.texi
+++ b/doc/libavutil.texi
@@ -1,62 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle Libavutil Documentation
-@titlepage
-@center @titlefont{Libavutil Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-The libavutil library is a utility library to aid portable
-multimedia programming. It contains safe portable string functions,
-random number generators, data structures, additional mathematics
-functions, cryptography and multimedia related functionality (like
-enumerations for pixel and sample formats). It is not a library for
-code needed by both libavcodec and libavformat.
-
-The goals for this library is to be:
-
-@table @strong
-@item Modular
-It should have few interdependencies and the possibility of disabling individual
-parts during @command{./configure}.
-
-@item Small
-Both sources and objects should be small.
-
-@item Efficient
-It should have low CPU and memory usage.
-
-@item Useful
-It should avoid useless features that almost no one needs.
-@end table
-
-@c man end DESCRIPTION
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{ffmpeg-utils.html,ffmpeg-utils}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
-ffmpeg-utils(1)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename libavutil
-@settitle multimedia-biased utility library
-
-@end ignore
-
-@bye
--- a/doc/libswresample.texi
+++ b/doc/libswresample.texi
@@ -1,70 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle Libswresample Documentation
-@titlepage
-@center @titlefont{Libswresample Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-The libswresample library performs highly optimized audio resampling,
-rematrixing and sample format conversion operations.
-
-Specifically, this library performs the following conversions:
-
-@itemize
-@item
-@emph{Resampling}: is the process of changing the audio rate, for
-example from a high sample rate of 44100Hz to 8000Hz. Audio
-conversion from high to low sample rate is a lossy process. Several
-resampling options and algorithms are available.
-
-@item
-@emph{Format conversion}: is the process of converting the type of
-samples, for example from 16-bit signed samples to unsigned 8-bit or
-float samples. It also handles packing conversion, when passing from
-packed layout (all samples belonging to distinct channels interleaved
-in the same buffer), to planar layout (all samples belonging to the
-same channel stored in a dedicated buffer or "plane").
-
-@item
-@emph{Rematrixing}: is the process of changing the channel layout, for
-example from stereo to mono. When the input channels cannot be mapped
-to the output streams, the process is lossy, since it involves
-different gain factors and mixing.
-@end itemize
-
-Various other audio conversions (e.g. stretching and padding) are
-enabled through dedicated options.
-
-@c man end DESCRIPTION
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{ffmpeg-resampler.html,ffmpeg-resampler},
-@url{libavutil.html,libavutil}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
-ffmpeg-resampler(1),
-libavutil(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename libswresample
-@settitle audio resampling library
-
-@end ignore
-
-@bye
--- a/doc/libswscale.texi
+++ b/doc/libswscale.texi
@@ -1,63 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle Libswscale Documentation
-@titlepage
-@center @titlefont{Libswscale Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Description
-@c man begin DESCRIPTION
-
-The libswscale library performs highly optimized image scaling and
-colorspace and pixel format conversion operations.
-
-Specifically, this library performs the following conversions:
-
-@itemize
-@item
-@emph{Rescaling}: is the process of changing the video size. Several
-rescaling options and algorithms are available. This is usually a
-lossy process.
-
-@item
-@emph{Pixel format conversion}: is the process of converting the image
-format and colorspace of the image, for example from planar YUV420P to
-RGB24 packed. It also handles packing conversion, that is converts
-from packed layout (all pixels belonging to distinct planes
-interleaved in the same buffer), to planar layout (all samples
-belonging to the same plane stored in a dedicated buffer or "plane").
-
-This is usually a lossy process in case the source and destination
-colorspaces differ.
-@end itemize
-
-@c man end DESCRIPTION
-
-@chapter See Also
-
-@ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
-@url{ffmpeg-scaler.html,ffmpeg-scaler},
-@url{libavutil.html,libavutil}
-@end ifhtml
-
-@ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
-ffmpeg-scaler(1),
-libavutil(3)
-@end ifnothtml
-
-@include authors.texi
-
-@ignore
-
-@setfilename libswscale
-@settitle video scaling and pixel format conversion library
-
-@end ignore
-
-@bye
--- a/doc/metadata.texi
+++ b/doc/metadata.texi
@@ -65,20 +65,4 @@ title=chapter \#1
 title=multi\
 line
@end example
-
-By using the ffmetadata muxer and demuxer it is possible to extract
-metadata from an input file to an ffmetadata file, and then transcode
-the file into an output file with the edited ffmetadata file.
-
-Extracting an ffmetadata file with @file{ffmpeg} goes as follows:
-@example
-ffmpeg -i INPUT -f ffmetadata FFMETADATAFILE
-@end example
-
-Reinserting edited metadata information from the FFMETADATAFILE file can
-be done as:
-@example
-ffmpeg -i INPUT -i FFMETADATAFILE -map_metadata 1 -codec copy OUTPUT
-@end example
-
@c man end METADATA
--- a/Show More
+++ b/Show More
@@ -1 +1 @@
 .1.git
 .11.3