update for 0.6.7

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

.gitignore

@@ -1,83 +1,45 @@
-*.a
+.config
+.version
 *.o
 *.d
-*.def
-*.dll
-*.dylib
 *.exe
-*.exp
-*.gcda
-*.gcno
-*.h.c
-*.ilk
-*.lib
-*.pc
-*.pdb
-*.so
-*.so.*
-*.ver
+*.ho
 *-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
+config.*
+doc/*.1
+doc/*.html
+doc/*.pod
+doxy
+ffmpeg
+ffplay
+ffprobe
+ffserver
+libavcodec/libavcodec*
+libavcore/libavcore*
+libavdevice/libavdevice*
+libavfilter/libavfilter*
+libavformat/libavformat*
+libavutil/avconfig.h
+libavutil/libavutil*
+libpostproc/libpostproc*
+libswscale/libswscale*
+tests/audiogen
+tests/base64
+tests/data
+tests/rotozoom
+tests/seek_test
+tests/tiny_psnr
+tests/videogen
+tests/vsynth1
+tests/vsynth2
+tools/cws2fws
+tools/graph2dot
+tools/lavfi-showfiltfmts
+tools/pktdumper
+tools/probetest
+tools/qt-faststart
+tools/trasher
+tools/trasher*.d
+version.h

COPYING.LGPLv2.1

@@ -500,3 +500,5 @@ necessary.  Here is a sample; alter the names:
   Ty Coon, President of Vice
 
 That's all there is to it!
+
+

CREDITS

@@ -1,6 +1,53 @@
-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 name of the people who have contributed to
+FFmpeg. The names are sorted alphabetically by last name.
 
-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

Changelog

@@ -1,667 +1,149 @@
 Entries are sorted chronologically from oldest to youngest within each release,
 releases are sorted from youngest to oldest.
 
-version 2.2.7
-- snow: fix null pointer dereference
-- iff: fix out of array access
-- svq1dec: fix input data corruption
-- proresenc_ks: check buffer size
+version 0.6.6:
+- id3v2: fix skipping extended header in id3v2.4
+- nsvdec: Fix use of uninitialized streams
+- nsvdec: Be more careful with av_malloc()
+- nsvdec: Propagate errors
+- dv: check stype
+- dv: Fix null pointer dereference due to ach=0
+- dv: Fix small stack overread related to CVE-2011-3929 and CVE-2011-3936.
+- atrac3: Fix crash in tonal component decoding
+- mjpegbdec: Fix overflow in SOS
+- kgv1dec: Increase offsets array size so it is large enough
+- motionpixels: decode only the 111 complete frames for fate
+- motionpixels: Clip YUV values after applying a gradient
+- celp filters: Do not read earlier than the start of the 'out' vector
+- vqavideo: return error if image size is not a multiple of block size (CVE-2012-0947)
+- dpcm: ignore extra unpaired bytes in stereo streams (CVE-2011-3951)
+- aacsbr: prevent out of bounds memcpy() (CVE-2012-0850)
+- h264: Add check for invalid chroma_format_idc (CVE-2012-0851)
+- adpcm: ADPCM Electronic Arts has always two channels (CVE-2012-0852)
+- shorten: remove VLA and check for buffer overflow
+- shorten: Prevent block size from increasing
+- shorten: Fix out of bound writes in fix_bitshift()
+- shorten: check for realloc failure (CVE-2012-0858)
+- shorten: Use separate pointers for the allocated memory for decoded samples (CVE-2012-0858)
+- kmvc: Check palsize (CVE-2011-3952)
+- qdm2: clip array indices returned by qdm2_get_vlc()
+- configure: properly check for mingw-w64 through installed headers
+- mingw32: properly check if vfw capture is supported by the system headers
+- mingw32: merge checks for mingw-w64 and mingw32-runtime >= 3.15 into one
+- dxva2: define required feature selection macros
+- png: check bit depth for PAL8/Y400A pixel formats
+- ea: check chunk_size for validity
+- tqi: Pass errors from the MB decoder
 
 
-version 2.2.6
-- fix infinite loop in dvbsub parser
-- fix some interlaced MPEG-2 videos
-- fix decoding issues in dv (Ticket2340, 2341)
-- fix v4l2 and v4l2enc crashes
-- fix theoretical librtmp crash
-- fix theoretical eamad crash
-- support dimension change in g2meet
+version 0.6.5:
+- vorbis: An additional defense in the Vorbis codec. (CVE-2011-3895)
+- vorbisdec: Fix decoding bug with channel handling.
+- matroskadec: Fix a bug where a pointer was cached to an array that might
+  later move due to a realloc(). (CVE-2011-3893)
+- vorbis: Avoid some out-of-bounds reads. (CVE-2011-3893)
+- vp3: fix oob read for negative tokens and memleaks on error, (CVE-2011-3892)
+- vp3: fix streams with non-zero last coefficient.
 
 
-version 2.2:
-- HNM version 4 demuxer and video decoder
-- Live HDS muxer
-- setsar/setdar filters now support variables in ratio expressions
-- elbg filter
-- string validation in ffprobe
-- support for decoding through VDPAU in ffmpeg (the -hwaccel option)
-- complete Voxware MetaSound decoder
-- remove mp3_header_compress bitstream filter
-- Windows resource files for shared libraries
-- aeval filter
-- stereoscopic 3d metadata handling
-- WebP encoding via libwebp
-- ATRAC3+ decoder
-- VP8 in Ogg demuxing
-- side & metadata support in NUT
-- framepack filter
-- XYZ12 rawvideo support in NUT
-- Exif metadata support in WebP decoder
-- OpenGL device
-- Use metadata_header_padding to control padding in ID3 tags (currently used in
-  MP3, AIFF, and OMA files), FLAC header, and the AVI "junk" block.
-- Mirillis FIC video decoder
-- Support DNx444
-- libx265 encoder
-- dejudder filter
-- Autodetect VDA like all other hardware accelerations
+version 0.6.4:
+- 4xm: Add a check in decode_i_frame to prevent buffer overreads
+- wma: initialize prev_block_len_bits, next_block_len_bits, and block_len_bits.
+- swscale: #include "libavutil/mathematics.h"
+- vp3dec: Check coefficient index in vp3_dequant() (CVE-2011-4352)
+- svq1dec: call avcodec_set_dimensions() after dimensions changed. (CVE-2011-4579)
+- commits related to CVE-2011-4353:
+  - vp6: Fix illegal read.
+  - vp6: Reset the internal state when aborting key frames header parsing
+  - vp6: Check for huffman tree build errors
+  - vp6: partially propagate huffman tree building errors during coeff model parsing and fix misspelling
+- commits related to CVE-2011-4351:
+  - qdm2: check output buffer size before decoding
+  - Fix qdm2 decoder packet handling to match the api
+  - Fix out of bound reads in the QDM2 decoder.
+  - Check for out of bound writes in the QDM2 decoder.
+- vmd: fix segfaults on corruped streams (CVE-2011-4364)
+- rv34: Check for invalid slice offsets
+- rv34: Fix potential overreads
+- rv34: Avoid NULL dereference on corrupted bitstream
+- rv10: Reject slices that does not have the same type as the first one
+- oggdec: fix out of bound write in the ogg demuxer
+- smacker: fix a few off by 1 errors
+- Check for invalid VLC value in smacker decoder.
+- Check and propagate errors when VLC trees cannot be built in smacker decoder.
+- Fixed off by one packet size allocation in the smacker demuxer.
+- Check for invalid packet size in the smacker demuxer.
+- ape demuxer: fix segfault on memory allocation failure.
+- Fixed size given to init_get_bits() in xan decoder.
+- smacker demuxer: handle possible av_realloc() failure.
+- Fixed segfault with wavpack decoder on corrupted decorrelation terms sub-blocks.
+- indeo2: fail if input buffer too small
+- indeo2: init_get_bits size in bits instead of bytes
+- wavpack: Check error codes rather than working around error conditions.
+- Fixed invalid writes and reads in wavpack decoder on corrupted bitstreams.
+- cpu detection: avoid a signed overflow
+- h264: correct implicit weight table computation for long ref pics
+- h264: correct the check for invalid long term frame index in MMCO decode
+- rv10/20: tell decoder to use edge emulation
+- flvenc: use int64_t to store offsets
+- VC-1: fix reading of custom PAR.
+- h264: notice memory allocation failure
+- libx264: do not set pic quality if no frame is output
+- mxfdec: Include FF_INPUT_BUFFER_PADDING_SIZE when allocating extradata.
+- rv30: return AVERROR(EINVAL) instead of EINVAL
+- Do not decode RV30 files if the extradata is too small
+- aacps: skip some memcpy() if src and dst would be equal
+- mpegts: fix Continuity Counter error detection
+- alsa: fallback to buffer_size/4 for period_size.
+- mxfenc: fix ignored drop flag in binary timecode representation.
+- dca: set AVCodecContext frame_size for DTS audio
+- H.264: fix overreads of qscale_table
+- postprocess.c: filter name needs to be double 0 terminated
+- Replace strncpy() with av_strlcpy() in libpostproc.
+- jpegdec: actually search for and parse RSTn
+- riff: Add mpgv MPEG-2 fourcc
+- Added M701 codec_tag for mpeg2video
 
 
-version 2.1:
+version 0.6.3:
 
-- 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
+- fix compilation with --enable-hardcoded-tables
+- mjpeg: Detect overreads in mjpeg_decode_scan() and error out.
+- aac: add headers needed for log2f()
+- lavf: inspect more frames for fps when container time base is coarse
+- AMV: disable DR1 and don't override EMU_EDGE
+  (addresses http://seclists.org/bugtraq/2011/Apr/257)
+- Fix memory (re)allocation in matroskadec.c (MSVR11-011/CVE-2011-3504)
+- Fix some crashes with invalid bitstreams in the CAVS decoder
+  (CVE-2011-3362, CVE-2011-3973, CVE-2011-3974)
+- Compilation fixes for gcc-4.6, testsuite now passes again
+- Fix a heap corruption issue in the OGG decoder
+- Backported the Android VisualOn AAC encoder wrapper from 0.7.2
 
 
-version 2.0:
+version 0.6.3:
 
-- 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
+- AMV: Fix possibly exploitable crash.
+- Fix apparently exploitable race condition.
+  (addresses http://seclists.org/bugtraq/2011/Apr/257)
+
+version 0.6.2:
+
+- fix compilation with --enable-hardcoded-tables
+- Fix invalid reads in VC-1 decoding (related to CVE-2011-0723)
+- Do not attempt to decode APE file with no frames
+  (adresses http://packetstorm.linuxsecurity.com/1103-exploits/vlc105-dos.txt)
 
 
-version 1.2:
+version 0.6.1:
 
-- 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
-
-
-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,
-- v408 Quicktime and Microsoft AYUV Uncompressed 4:4:4:4 encoder and decoder
-- setfield filter
-- CDXL demuxer and decoder
-- Apple ProRes encoder
-- ffprobe -count_packets and -count_frames options
-- Sun Rasterfile Encoder
-- ID3v2 attached pictures reading and writing
-- WMA Lossless decoder
-- bluray protocol
-- blackdetect filter
-- libutvideo encoder wrapper (--enable-libutvideo)
-- swapuv filter
-- bbox filter
-- XBM encoder and decoder
-- RealAudio Lossless decoder
-- ZeroCodec decoder
-- tile video filter
-- Metal Gear Solid: The Twin Snakes demuxer
-- OpenEXR image decoder
-- removelogo filter
-- drop support for ffmpeg without libavfilter
-- drawtext video filter: fontconfig support
-- ffmpeg -benchmark_all option
-- super2xsai filter ported from libmpcodecs
-- add libavresample audio conversion library for compatibility
-- MicroDVD decoder
-- Avid Meridien (AVUI) encoder and decoder
-- accept + prefix to -pix_fmt option to disable automatic conversions.
-- complete audio filtering in libavfilter and ffmpeg
-- add fps 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,
-         CVE-2011-3950, CVE-2011-3951, CVE-2011-3952
-- v410 Quicktime Uncompressed 4:4:4 10-bit encoder and decoder
-- SBaGen (SBG) binaural beats script demuxer
-- OpenMG Audio muxer
-- Timecode extraction in DV and MOV
-- thumbnail video filter
-- XML output in ffprobe
-- asplit audio filter
-- tinterlace video filter
-- astreamsync audio filter
-- amerge audio filter
-- ISMV (Smooth Streaming) muxer
-- GSM audio parser
-- SMJPEG muxer
-- XWD encoder and decoder
-- Automatic thread count based on detection number of (available) CPU cores
-- y41p Brooktree Uncompressed 4:1:1 12-bit encoder and decoder
-- ffprobe -show_error option
-- Avid 1:1 10-bit RGB Packer codec
-- v308 Quicktime Uncompressed 4:4:4 encoder and decoder
-- yuv4 libquicktime packed 4:2:0 encoder and decoder
-- ffprobe -show_frames option
-- silencedetect audio filter
-- ffprobe -show_program_version, -show_library_versions, -show_versions options
-- rv34: frame-level multi-threading
-- optimized iMDCT transform on x86 using SSE for for mpegaudiodec
-- Improved PGS subtitle decoder
-- dumpgraph option to lavfi device
-- r210 and r10k encoders
-- ffwavesynth decoder
-- aviocat tool
-- ffeval tool
-
-
-version 0.9:
-
-- openal input device added
-- boxblur filter added
-- BWF muxer
-- Flash Screen Video 2 decoder
-- lavfi input device added
-- added avconv, which is almost the same for now, except
-for a few incompatible changes in the options, which will hopefully make them
-easier to use. The changes are:
-    * The options placement is now strictly enforced! While in theory the
-      options for ffmpeg should be given in [input options] -i INPUT [output
-      options] OUTPUT order, in practice it was possible to give output options
-      before the -i and it mostly worked. Except when it didn't - the behavior was
-      a bit inconsistent. In avconv, it is not possible to mix input and output
-      options. All non-global options are reset after an input or output filename.
-    * All per-file options are now truly per-file - they apply only to the next
-      input or output file and specifying different values for different files
-      will now work properly (notably -ss and -t options).
-    * All per-stream options are now truly per-stream - it is possible to
-      specify which stream(s) should a given option apply to. See the Stream
-      specifiers section in the avconv manual for details.
-    * In ffmpeg some options (like -newvideo/-newaudio/...) are irregular in the
-      sense that they're specified after the output filename instead of before,
-      like all other options. In avconv this irregularity is removed, all options
-      apply to the next input or output file.
-    * -newvideo/-newaudio/-newsubtitle options were removed. Not only were they
-      irregular and highly confusing, they were also redundant. In avconv the -map
-      option will create new streams in the output file and map input streams to
-      them. E.g. avconv -i INPUT -map 0 OUTPUT will create an output stream for
-      each stream in the first input file.
-    * The -map option now has slightly different and more powerful syntax:
-        + Colons (':') are used to separate file index/stream type/stream index
-          instead of dots. Comma (',') is used to separate the sync stream instead
-          of colon.. This is done for consistency with other options.
-        + It's possible to specify stream type. E.g. -map 0:a:2 creates an
-          output stream from the third input audio stream.
-        + Omitting the stream index now maps all the streams of the given type,
-          not just the first. E.g. -map 0:s creates output streams for all the
-          subtitle streams in the first input file.
-        + Since -map can now match multiple streams, negative mappings were
-          introduced. Negative mappings disable some streams from an already
-          defined map. E.g. '-map 0 -map -0:a:1' means 'create output streams for
-          all the stream in the first input file, except for the second audio
-          stream'.
-    * There is a new option -c (or -codec) for choosing the decoder/encoder to
-      use, which allows to precisely specify target stream(s) consistently with
-      other options. E.g. -c:v lib264 sets the codec for all video streams, -c:a:0
-      libvorbis sets the codec for the first audio stream and -c copy copies all
-      the streams without reencoding. Old -vcodec/-acodec/-scodec options are now
-      aliases to -c:v/a/s
-    * It is now possible to precisely specify which stream should an AVOption
-      apply to. E.g. -b:v:0 2M sets the bitrate for the first video stream, while
-      -b:a 128k sets the bitrate for all audio streams. Note that the old -ab 128k
-      syntax is deprecated and will stop working soon.
-    * -map_chapters now takes only an input file index and applies to the next
-      output file. This is consistent with how all the other options work.
-    * -map_metadata now takes only an input metadata specifier and applies to
-      the next output file. Output metadata specifier is now part of the option
-      name, similarly to the AVOptions/map/codec feature above.
-    * -metadata can now be used to set metadata on streams and chapters, e.g.
-      -metadata:s:1 language=eng sets the language of the first stream to 'eng'.
-      This made -vlang/-alang/-slang options redundant, so they were removed.
-    * -qscale option now uses stream specifiers and applies to all streams, not
-      just video. I.e. plain -qscale number would now apply to all streams. To get
-      the old behavior, use -qscale:v. Also there is now a shortcut -q for -qscale
-      and -aq is now an alias for -q:a.
-    * -vbsf/-absf/-sbsf options were removed and replaced by a -bsf option which
-      uses stream specifiers. Use -bsf:v/a/s instead of the old options.
-    * -itsscale option now uses stream specifiers, so its argument is only the
-      scale parameter.
-    * -intra option was removed, use -g 0 for the same effect.
-    * -psnr option was removed, use -flags +psnr for the same effect.
-    * -vf option is now an alias to the new -filter option, which uses stream specifiers.
-    * -vframes/-aframes/-dframes options are now aliases to the new -frames option.
-    * -vtag/-atag/-stag options are now aliases to the new -tag option.
-- XMV demuxer
-- LOAS demuxer
-- ashowinfo filter added
-- Windows Media Image decoder
-- amovie source added
-- LATM muxer/demuxer
-- Speex encoder via libspeex
-- JSON output in ffprobe
-- WTV muxer
-- Optional C++ Support (needed for libstagefright)
-- H.264 Decoding on Android via Stagefright
-- Prores decoder
-- BIN/XBIN/ADF/IDF text file decoder
-- aconvert audio filter added
-- audio support to lavfi input device added
-- libcdio-paranoia input device for audio CD grabbing
-- Apple ProRes decoder
-- CELT in Ogg demuxing
-- G.723.1 demuxer and decoder
-- libmodplug support (--enable-libmodplug)
-- VC-1 interlaced decoding
-- libutvideo wrapper (--enable-libutvideo)
-- aevalsrc audio source added
-- Ut Video decoder
-- Speex encoding via libspeex
-- 4:2:2 H.264 decoding support
-- 4:2:2 and 4:4:4 H.264 encoding with libx264
-- Pulseaudio input device
-- Prores encoder
-- Video Decoder Acceleration (VDA) HWAccel module.
-- replacement Indeo 3 decoder
-- new ffmpeg option: -map_channel
-- volume audio filter added
-- earwax audio filter added
-- libv4l2 support (--enable-libv4l2)
-- TLS/SSL and HTTPS protocol support
-- AVOptions API rewritten and documented
-- most of CODEC_FLAG2_*, some CODEC_FLAG_* and many codec-specific fields in
-  AVCodecContext deprecated. Codec private options should be used instead.
-- Properly working defaults in libx264 wrapper, support for native presets.
-- Encrypted OMA files support
-- Discworld II BMV decoding support
-- VBLE Decoder
-- OS X Video Decoder Acceleration (VDA) support
-- compact and csv output in ffprobe
-- pan audio filter
-- IFF Amiga Continuous Bitmap (ACBM) decoder
-- ass filter
-- CRI ADX audio format muxer and demuxer
-- Playstation Portable PMP format demuxer
-- Microsoft Windows ICO demuxer
-- life source
-- PCM format support in OMA demuxer
-- CLJR encoder
-- new option: -report
-- Dxtory capture format decoder
-- cellauto source
-- Simple segmenting muxer
-- Indeo 4 decoder
-- SMJPEG demuxer
-
-
-version 0.8:
-
-- many many things we forgot because we rather write code than changelogs
-- WebM support in Matroska de/muxer
-- low overhead Ogg muxing
-- MMS-TCP support
-- VP8 de/encoding via libvpx
-- Demuxer for On2's IVF format
-- Pictor/PC Paint decoder
-- HE-AAC v2 decoder
-- HE-AAC v2 encoding with libaacplus
-- libfaad2 wrapper removed
-- DTS-ES extension (XCh) decoding support
-- native VP8 decoder
-- RTSP tunneling over HTTP
-- RTP depacketization of SVQ3
-- -strict inofficial replaced by -strict unofficial
-- ffplay -exitonkeydown and -exitonmousedown options added
-- native GSM / GSM MS decoder
-- RTP depacketization of QDM2
-- ANSI/ASCII art playback system
-- Lego Mindstorms RSO de/muxer
-- libavcore added (and subsequently removed)
-- SubRip subtitle file muxer and demuxer
-- Chinese AVS encoding via libxavs
-- ffprobe -show_packets option added
-- RTP packetization of Theora and Vorbis
-- RTP depacketization of MP4A-LATM
-- RTP packetization and depacketization of VP8
-- hflip filter
-- Apple HTTP Live Streaming demuxer
-- a64 codec
-- MMS-HTTP support
-- G.722 ADPCM audio encoder/decoder
-- R10k video decoder
-- ocv_smooth filter
-- frei0r wrapper filter
-- change crop filter syntax to width:height:x:y
-- make the crop filter accept parametric expressions
-- make ffprobe accept AVFormatContext options
-- yadif filter
-- blackframe filter
-- Demuxer for Leitch/Harris' VR native stream format (LXF)
-- RTP depacketization of the X-QT QuickTime format
-- SAP (Session Announcement Protocol, RFC 2974) muxer and demuxer
-- cropdetect filter
-- ffmpeg -crop* options removed
-- transpose filter added
-- ffmpeg -force_key_frames option added
-- demuxer for receiving raw rtp:// URLs without an SDP description
-- single stream LATM/LOAS decoder
-- setpts filter added
-- Win64 support for optimized x86 assembly functions
-- MJPEG/AVI1 to JPEG/JFIF bitstream filter
-- ASS subtitle encoder and decoder
-- IEC 61937 encapsulation for E-AC-3, TrueHD, DTS-HD (for HDMI passthrough)
-- overlay filter added
-- rename aspect filter to setdar, and pixelaspect to setsar
-- IEC 61937 demuxer
-- Mobotix .mxg demuxer
-- frei0r source added
-- hqdn3d filter added
-- RTP depacketization of QCELP
-- FLAC parser added
-- gradfun filter added
-- AMR-WB decoder
-- replace the ocv_smooth filter with a more generic ocv filter
-- Windows Televison (WTV) demuxer
-- FFmpeg metadata format muxer and demuxer
-- SubRip (srt) subtitle encoder and decoder
-- floating-point AC-3 encoder added
-- Lagarith decoder
-- ffmpeg -copytb option added
-- IVF muxer added
-- Wing Commander IV movies decoder added
-- movie source added
-- Bink version 'b' audio and video decoder
-- Bitmap Brothers JV playback system
-- Apple HTTP Live Streaming protocol handler
-- sndio support for playback and record
-- Linux framebuffer input device added
-- Chronomaster DFA decoder
-- DPX image encoder
-- MicroDVD subtitle file muxer and demuxer
-- Playstation Portable PMP format demuxer
-- fieldorder video filter added
-- AAC encoding via libvo-aacenc
-- AMR-WB encoding via libvo-amrwbenc
-- xWMA demuxer
-- Mobotix MxPEG decoder
-- VP8 frame-multithreading
-- NEON optimizations for VP8
-- Lots of deprecated API cruft removed
-- fft and imdct optimizations for AVX (Sandy Bridge) processors
-- showinfo filter added
-- SMPTE 302M AES3 audio decoder
-- Apple Core Audio Format muxer
-- 9bit and 10bit per sample support in the H.264 decoder
-- 9bit and 10bit FFV1 encoding / decoding
-- split filter added
-- select filter added
-- sdl output device added
-- libmpcodecs video filter support (3 times as many filters than before)
-- mpeg2 aspect ratio dection fixed
-- libxvid aspect pickiness fixed
-- Frame multithreaded decoding
-- E-AC-3 audio encoder
-- ac3enc: add channel coupling support
-- floating-point sample format support to the ac3, eac3, dca, aac, and vorbis decoders.
-- H264/MPEG frame-level multi-threading
-- All av_metadata_* functions renamed to av_dict_* and moved to libavutil
-- 4:4:4 H.264 decoding support
-- 10-bit H.264 optimizations for x86
-- lut, lutrgb, and lutyuv filters added
-- buffersink libavfilter sink added
-- Bump libswscale for recently reported ABI break
-- New J2K encoder (via OpenJPEG)
-
-
-version 0.7:
-
-- all the changes for 0.8, but keeping API/ABI compatibility with the 0.6 release
+- fix autodetection of E-AC-3 substream samples
+- performance fix for seekable HTTP
+- backport AAC-HE v2 from trunk
+- add missing VP80 fourcc code for the VP8 codec
+- small documentation fixes
+- fix several potentially exploitable issues in the FLIC decoder
+  (addresses CVE-2010-3429)
 
 
 version 0.6:
@@ -670,7 +152,7 @@ version 0.6:
 - deprecated vhook subsystem removed
 - deprecated old scaler removed
 - VQF demuxer
-- Alpha channel scaler
+- alpha channel scaler
 - PCX encoder
 - RTP packetization of H.263
 - RTP packetization of AMR
@@ -700,7 +182,7 @@ 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
@@ -739,7 +221,11 @@ version 0.6:
 - 35% faster VP3/Theora decoding
 - faster AAC decoding
 - faster H.264 decoding
-- RealAudio 1.0 (14.4K) encoder
+- WebM support in Matroska de/muxer
+- low overhead Ogg muxing
+- VP8 de/encoding via libvpx
+- CODEC_CAP_EXPERIMENTAL added
+
 
 
 version 0.5:
@@ -800,7 +286,7 @@ 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
@@ -828,7 +314,7 @@ 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
@@ -902,7 +388,7 @@ version 0.5:
 - Gopher client support
 - MXF D-10 muxer
 - generic metadata API
-- flash ScreenVideo2 encoder
+
 
 
 version 0.4.9-pre1:
@@ -989,6 +475,7 @@ version 0.4.9-pre1:
 - FunCom ISS audio file demuxer and according ADPCM decoding
 
 
+
 version 0.4.8:
 
 - MPEG-2 video encoding (Michael)
@@ -1000,6 +487,7 @@ version 0.4.8:
 - Duck DK3 and DK4 ADPCM audio decoders (Mike Melanson)
 
 
+
 version 0.4.7:
 
 - RealAudio 1.0 (14_4) and 2.0 (28_8) native decoders. Author unknown, code from mplayerhq
@@ -1041,6 +529,7 @@ version 0.4.7:
 .. And lots more new enhancements and fixes.
 
 
+
 version 0.4.6:
 
 - completely new integer only MPEG audio layer 1/2/3 decoder rewritten
@@ -1078,6 +567,7 @@ version 0.4.6:
 - 16-bit and 15-bit RGB/BGR/GBR support (Bisqwit)
 
 
+
 version 0.4.5:
 
 - some header fixes (Zdenek Kabelac <kabi at informatics.muni.cz>)
@@ -1105,10 +595,11 @@ version 0.4.5:
 - MPEG-4 vol header fixes (Jonathan Marsden <snmjbm at pacbell.net>)
 - ARM optimizations (Lionel Ulmer <lionel.ulmer at free.fr>).
 - Windows porting of file converter
-- added MJPEG raw format (input/output)
+- added MJPEG raw format (input/ouput)
 - added JPEG image format support (input/output)
 
 
+
 version 0.4.4:
 
 - fixed some std header definitions (Bjorn Lindgren
@@ -1134,6 +625,7 @@ version 0.4.4:
   segfault)
 
 
+
 version 0.4.3:
 
 - BGR24 patch (initial patch by Jeroen Vreeken <pe1rxq at amsat.org>)
@@ -1155,6 +647,7 @@ version 0.4.3:
   instead of a protocol)
 
 
+
 version 0.4.2:
 
 - added H.263/MPEG-4/MSMPEG4 decoding support. MPEG-4 decoding support
@@ -1185,6 +678,7 @@ version 0.4.1:
 - factorized some motion estimation code
 
 
+
 version 0.4.0:
 
 - removing grab code from ffserver and moved it to ffmpeg. Added
@@ -1213,11 +707,13 @@ version 0.4.0:
   codecs
 
 
+
 version 0.3.4:
 
 - added stereo in MPEG audio encoder
 
 
+
 version 0.3.3:
 
 - added 'high quality' mode which use motion vectors. It can be used in
@@ -1226,12 +722,14 @@ version 0.3.3:
   bitrates and large GOP size
 
 
+
 version 0.3.2: small fixes
 
 - ASF fixes
 - put_seek bug fix
 
 
+
 version 0.3.1: added avi/divx support
 
 - added AVI support
@@ -1239,4 +737,5 @@ version 0.3.1: added avi/divx support
 - added sound for flash format (not tested)
 
 
+
 version 0.3: initial public release

INSTALL

@@ -9,7 +9,3 @@ path when launching 'configure', e.g. '/ffmpegdir/ffmpeg/configure'.
 2) Then type 'make' to build FFmpeg. GNU Make 3.81 or later is required.
 
 3) Type 'make install' to install all binaries and libraries you built.
-
-NOTICE
-
- - Non system dependencies (e.g. libx264, libvpx) are disabled by default.

LICENSE

@@ -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
@@ -13,92 +14,37 @@ configure to activate them. In this case, FFmpeg's license changes to GPL v2+.
 Specifically, the GPL parts of FFmpeg are
 
 - libpostproc
-- libmpcodecs
+- optional MMX optimizations for YUV to RGB colorspace conversion in
+  libswscale/x86/yuv2rgb_template.c
 - optional x86 optimizations in the files
+  libavcodec/x86/h264_deblock_sse2.asm
+  libavcodec/x86/h264_idct_sse2.asm
   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_interlace.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
+
+There are a handful of files under other licensing terms, namely:
+
+* The files libavcodec/jfdctfst.c, libavcodec/jfdctint.c, libavcodec/jrevdct.c
+  are taken from libjpeg, see the top of the files for licensing details.
 
 Should you, for whatever reason, prefer to use version 3 of the (L)GPL, then
 the configure parameter --enable-version3 will activate this licensing option
 for you. Read the file COPYING.LGPLv3 or, if you have enabled GPL parts,
 COPYING.GPLv3 to learn the exact legal terms that apply in this case.
 
-There are a handful of files under other licensing terms, namely:
 
-* The files libavcodec/jfdctfst.c, libavcodec/jfdctint_template.c and
-  libavcodec/jrevdct.c are taken from libjpeg, see the top of the files for
-  licensing details. Specifically note that you must credit the IJG in the
-  documentation accompanying your program if you only distribute executables.
-  You must also indicate any changes including additions and deletions to
-  those three files in the documentation.
+external libraries:
+-------------------
 
+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.
 
-external libraries
-==================
+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.
 
-FFmpeg can be combined with a number of external libraries, which sometimes
-affect the licensing of binaries resulting from the combination.
-
-compatible libraries
---------------------
-
-The following libraries are under GPL:
-    - frei0r
-    - libcdio
-    - libutvideo
-    - libvidstab
-    - libx264
-    - libx265
-    - libxavs
-    - libxvid
-When combining them with FFmpeg, FFmpeg needs to be licensed as GPL as well by
-passing --enable-gpl to configure.
-
-The OpenCORE and VisualOn libraries are under the Apache License 2.0. That
-license is incompatible with the LGPL v2.1 and the GPL v2, but not with
-version 3 of those licenses. So to combine these libraries with FFmpeg, the
-license version needs to be upgraded by passing --enable-version3 to configure.
-
-incompatible libraries
-----------------------
-
-The Fraunhofer AAC library, 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 library libfaac 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.

MAINTAINERS

@@ -4,12 +4,6 @@ FFmpeg maintainers
 Below is a list of the people maintaining different parts of the
 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.
-
 
 Project Leader
 ==============
@@ -25,13 +19,10 @@ ffmpeg:
   ffmpeg.c                              Michael Niedermayer
 
 ffplay:
-  ffplay.c                              Marton Balint
-
-ffprobe:
-  ffprobe.c                             Stefano Sabatini
+  ffplay.c                              Michael Niedermayer
 
 ffserver:
-  ffserver.c                            Reynaldo H. Verdejo Pinochet
+  ffserver.c, ffserver.h                Baptiste Coudurier
 
 Commandline utility code:
   cmdutils.c, cmdutils.h                Michael Niedermayer
@@ -43,22 +34,14 @@ QuickTime faststart:
 Miscellaneous Areas
 ===================
 
-documentation                           Stefano Sabatini, Mike Melanson, Timothy Gu
+documentation                           Mike Melanson, Diego Biurrun
+website                                 Robert Swain
 build system (configure,Makefiles)      Diego Biurrun, Mans Rullgard
-project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Döffinger, Alexander Strasser
+project server                          Diego Biurrun, Mans Rullgard
+mailinglists                            Michael Niedermayer, Baptiste Coudurier
 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
+release management                      Diego Biurrun, Reinhard Tartler
 
 
 libavutil
@@ -70,23 +53,11 @@ 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
@@ -97,14 +68,16 @@ 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:
     bitstream.c, bitstream.h            Michael Niedermayer
   CABAC:
     cabac.h, cabac.c                    Michael Niedermayer
-  codec names:
-    codec_names.sh                      Nicolas George
   DSP utilities:
     dsputils.c, dsputils.h              Michael Niedermayer
   entropy coding:
@@ -125,57 +98,44 @@ Generic Parts:
     simple_idct.c, simple_idct.h        Michael Niedermayer
   postprocessing:
     libpostproc/*                       Michael Niedermayer
-  table generation:
-    tableprint.c, tableprint.h          Reimar Doeffinger
-  fixed point FFT:
-    fft*                                Zeljko Lukac
 
 Codecs:
   4xm.c                                 Michael Niedermayer
   8bps.c                                Roberto Togni
   8svx.c                                Jaikrishnan Menon
   aasc.c                                Kostya Shishkov
+  aac*, sbr.h                           Alex Converse
   ac3*                                  Justin Ruggles
   alacenc.c                             Jaikrishnan Menon
   alsdec.c                              Thilo Borgmann
   apedec.c                              Kostya Shishkov
-  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
   flashsv*                              Benjamin Larsson
   flicvideo.c                           Mike Melanson
-  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
@@ -183,46 +143,32 @@ Codecs:
   indeo5*                               Kostya Shishkov
   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
   libspeexdec.c                         Justin Ruggles
   libtheoraenc.c                        David Conrad
-  libutvideo*                           Derek Buitenhuis
-  libvorbis.c                           David Conrad
-  libvpx*                               James Zern
   libx264.c                             Mans Rullgard, Jason Garrett-Glaser
-  libx265.c                             Derek Buitenhuis
-  libxavs.c                             Stefan Gehrer
-  libzvbi-teletextdec.c                 Marton Balint
   loco.c                                Kostya Shishkov
   lzo.h, lzo.c                          Reimar Doeffinger
   mdec.c                                Michael Niedermayer
   mimic.c                               Ramiro Polla
-  mjpeg*.c                              Michael Niedermayer
+  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
   qcelp*                                Reynaldo H. Verdejo Pinochet
   qdm2.c, qdm2data.h                    Roberto Togni, Benjamin Larsson
@@ -240,13 +186,10 @@ 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
@@ -254,24 +197,17 @@ 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
   vb.c                                  Kostya Shishkov
-  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
   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
@@ -279,20 +215,13 @@ 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
 
 Hardware acceleration:
-  crystalhd.c                           Philip Langdale
   dxva2*                                Laurent Aimar
-  libstagefright.cpp                    Mohamed Naufal
   vaapi*                                Gwenole Beauchesne
-  vda*                                  Sebastien Zwickert
   vdpau*                                Carl Eugen Hoyos
 
 
@@ -302,56 +231,10 @@ 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:
-  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
-  vf_yadif.c                            Michael Niedermayer
-
-Sources:
-  vsrc_mandelbrot.c                     Michael Niedermayer
 
 libavformat
 ===========
@@ -366,57 +249,36 @@ 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
+  idroq.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
@@ -424,80 +286,50 @@ Muxers/Demuxers:
   msnwc_tcp.c                           Ramiro Polla
   mtv.c                                 Reynaldo H. Verdejo Pinochet
   mxf*                                  Baptiste Coudurier
-  mxfdec.c                              Tomas Härdin
-  nistspheredec.c                       Paul B Mahol
   nsvdec.c                              Francois Revol
   nut.c                                 Michael Niedermayer
   nuv.c                                 Reimar Doeffinger
   oggdec.c, oggdec.h                    David Conrad
   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
   rl2.c                                 Sascha Sommer
   rmdec.c, rmenc.c                      Ronald S. Bultje, Kostya Shishkov
   rtmp*                                 Kostya Shishkov
-  rtp.c, rtpenc.c                       Martin Storsjo
-  rtpdec_asf.*                          Ronald S. Bultje
-  rtpenc_mpv.*, rtpenc_aac.*            Martin Storsjo
+  rtp.c, rtpenc.c                       Luca Abeni
+  rtp_asf.*                             Ronald S. Bultje
+  rtp_mpv.*, rtp_aac.*                  Luca Abeni
   rtsp.c                                Luca Barbato
-  sbgdec.c                              Nicolas George
-  sdp.c                                 Martin Storsjo
+  sdp.c                                 Luca Abeni
   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
+BeOS                                    Francois Revol
 Mac OS X / PowerPC                      Romain Dolbeau, Guillaume Poirier
 Amiga / PowerPC                         Colin Ward
 Linux / PowerPC                         Luca Barbato
@@ -508,48 +340,20 @@ Sparc                                   Roman Shaposhnik
 x86                                     Michael Niedermayer
 
 
-Releases
-========
+GnuPG Fingerprints of maintainers and others who have svn write access
+======================================================================
 
-2.2                                     Michael Niedermayer
-2.1                                     Michael Niedermayer
-1.2                                     Michael Niedermayer
-
-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
-Luca Barbato                  6677 4209 213C 8843 5B67 29E7 E84C 78C2 84E9 0E34
 Michael Niedermayer           9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
-Nicolas George                24CE 01CE 9ACC 5CEB 74D8 8D9D B063 D997 36E5 4C93
-Panagiotis Issaris            6571 13A3 33D9 3726 F728 AA98 F643 B12E ECF3 E029
+Panagiotis Issaris            515C E262 10A8 FDCE 5481 7B9C 3AD7 D9A5 071D B3A9
 Peter Ross                    A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B
 Reimar Döffinger              C61D 16E5 9E2C D10C 8958 38A4 0899 A2B9 06D4 D9C7
 Reinhard Tartler              9300 5DC2 7E87 6C37 ED7B CA9A 9808 3544 9453 48A4
 Reynaldo H. Verdejo Pinochet  6E27 CD34 170C C78E 4D4F 5F40 C18E 077F 3114 452A
-Robert Swain                  EE7A 56EA 4A81 A7B5 2001 A521 67FA 362D A2FC 3E71
 Sascha Sommer                 38A0 F88B 868E 9D3A 97D4 D6A0 E823 706F 1E07 0D3C
-Stefano Sabatini              0D0B AD6B 5330 BBAD D3D6 6A0C 719C 2839 FC43 2D5F
-Stephan Hilb                  4F38 0B3A 5F39 B99B F505 E562 8D5C 5554 4E17 8863
-Tomas Härdin                  A79D 4E3D F38F 763F 91F5 8B33 A01E 8AE0 41BB 2551
-Wei Gao                       4269 7741 857A 0E60 9EC5 08D2 4744 4EFA 62C1 87B9

Makefile

@@ -1,82 +1,62 @@
-MAIN_MAKEFILE=1
 include config.mak
 
-vpath %.c    $(SRC_PATH)
-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)
+SRC_DIR = $(SRC_PATH_BARE)
 
-AVPROGS-$(CONFIG_FFMPEG)   += ffmpeg
-AVPROGS-$(CONFIG_FFPLAY)   += ffplay
-AVPROGS-$(CONFIG_FFPROBE)  += ffprobe
-AVPROGS-$(CONFIG_FFSERVER) += ffserver
+vpath %.texi $(SRC_PATH_BARE)
 
-AVPROGS    := $(AVPROGS-yes:%=%$(PROGSSUF)$(EXESUF))
-INSTPROGS   = $(AVPROGS-yes:%=%$(PROGSSUF)$(EXESUF))
-PROGS      += $(AVPROGS)
+PROGS-$(CONFIG_FFMPEG)   += ffmpeg
+PROGS-$(CONFIG_FFPLAY)   += ffplay
+PROGS-$(CONFIG_FFPROBE)  += ffprobe
+PROGS-$(CONFIG_FFSERVER) += ffserver
 
-AVBASENAMES  = ffmpeg ffplay ffprobe ffserver
-ALLAVPROGS   = $(AVBASENAMES:%=%$(PROGSSUF)$(EXESUF))
-ALLAVPROGS_G = $(AVBASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
+PROGS      := $(addsuffix   $(EXESUF), $(PROGS-yes))
+PROGS_G     = $(addsuffix _g$(EXESUF), $(PROGS-yes))
+OBJS        = $(addsuffix .o,          $(PROGS-yes)) cmdutils.o
+MANPAGES    = $(addprefix doc/, $(addsuffix .1, $(PROGS-yes)))
+TOOLS       = $(addprefix tools/, $(addsuffix $(EXESUF), cws2fws pktdumper probetest qt-faststart trasher))
+HOSTPROGS   = $(addprefix tests/, audiogen videogen rotozoom tiny_psnr)
 
-$(foreach prog,$(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
-HOSTPROGS  := $(TESTTOOLS:%=tests/%) doc/print_options
-TOOLS       = qt-faststart trasher uncoded_frame
-TOOLS-$(CONFIG_ZLIB) += cws2fws
+BASENAMES   = ffmpeg ffplay ffprobe ffserver
+ALLPROGS    = $(addsuffix   $(EXESUF), $(BASENAMES))
+ALLPROGS_G  = $(addsuffix _g$(EXESUF), $(BASENAMES))
+ALLMANPAGES = $(addsuffix .1, $(BASENAMES))
 
 FFLIBS-$(CONFIG_AVDEVICE) += avdevice
 FFLIBS-$(CONFIG_AVFILTER) += avfilter
 FFLIBS-$(CONFIG_AVFORMAT) += avformat
-FFLIBS-$(CONFIG_AVRESAMPLE) += avresample
 FFLIBS-$(CONFIG_AVCODEC)  += avcodec
 FFLIBS-$(CONFIG_POSTPROC) += postproc
-FFLIBS-$(CONFIG_SWRESAMPLE)+= swresample
 FFLIBS-$(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
+DATA_FILES := $(wildcard $(SRC_DIR)/ffpresets/*.ffpreset)
 
-SKIPHEADERS = cmdutils_common_opts.h compat/w32pthreads.h
+SKIPHEADERS = cmdutils_common_opts.h
 
-include $(SRC_PATH)/common.mak
+include common.mak
 
+FF_LDFLAGS   := $(FFLDFLAGS)
 FF_EXTRALIBS := $(FFEXTRALIBS)
 FF_DEP_LIBS  := $(DEP_LIBS)
 
-all: $(AVPROGS)
+ALL_TARGETS-$(CONFIG_DOC)       += documentation
 
-$(TOOLS): %$(EXESUF): %.o $(EXEOBJS)
-	$(LD) $(LDFLAGS) $(LD_O) $^ $(ELIBS)
+ifdef PROGS
+INSTALL_TARGETS-yes             += install-progs install-data
+INSTALL_TARGETS-$(CONFIG_DOC)   += install-man
+endif
+INSTALL_PROGS_TARGETS-$(CONFIG_SHARED) = install-libs
 
-tools/cws2fws$(EXESUF): ELIBS = $(ZLIB)
-tools/uncoded_frame$(EXESUF): $(FF_DEP_LIBS)
-tools/uncoded_frame$(EXESUF): ELIBS = $(FF_EXTRALIBS)
+all: $(FF_DEP_LIBS) $(PROGS) $(ALL_TARGETS-yes)
 
-config.h: .config
-.config: $(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c))
-	@-tput bold 2>/dev/null
-	@-printf '\nWARNING: $(?F) newer than config.h, rerun configure\n\n'
-	@-tput sgr0 2>/dev/null
+$(PROGS): %$(EXESUF): %_g$(EXESUF)
+	$(CP) $< $@
+	$(STRIP) $@
 
-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
+SUBDIR_VARS := OBJS FFLIBS CLEANFILES DIRS TESTPROGS EXAMPLES SKIPHEADERS \
+               ALTIVEC-OBJS MMX-OBJS NEON-OBJS X86-OBJS YASM-OBJS-FFT YASM-OBJS \
+               HOSTPROGS BUILT_HEADERS TESTOBJS ARCH_HEADERS
 
 define RESET
 $(1) :=
@@ -86,108 +66,288 @@ endef
 define DOSUBDIR
 $(foreach V,$(SUBDIR_VARS),$(eval $(call RESET,$(V))))
 SUBDIR := $(1)/
-include $(SRC_PATH)/$(1)/Makefile
--include $(SRC_PATH)/$(1)/$(ARCH)/Makefile
-include $(SRC_PATH)/library.mak
+include $(1)/Makefile
 endef
 
 $(foreach D,$(FFLIBS),$(eval $(call DOSUBDIR,lib$(D))))
 
-include $(SRC_PATH)/doc/Makefile
+ffplay_g$(EXESUF): FF_EXTRALIBS += $(SDL_LIBS)
+ffserver_g$(EXESUF): FF_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-$(1):.o=.d)
-endef
+%_g$(EXESUF): %.o cmdutils.o $(FF_DEP_LIBS)
+	$(LD) $(FF_LDFLAGS) -o $@ $< cmdutils.o $(FF_EXTRALIBS)
 
-$(foreach P,$(PROGS),$(eval $(call DOPROG,$(P:$(PROGSSUF)$(EXESUF)=))))
+tools/%$(EXESUF): tools/%.o
+	$(LD) $(FF_LDFLAGS) -o $@ $< $(FF_EXTRALIBS)
 
-ffprobe.o cmdutils.o : libavutil/ffversion.h
+tools/%.o: tools/%.c
+	$(CC) $(CPPFLAGS) $(CFLAGS) $(CC_O) $<
 
-$(PROGS): %$(PROGSSUF)$(EXESUF): %$(PROGSSUF)_g$(EXESUF)
-	$(CP) $< $@
-	$(STRIP) $@
+ffplay.o ffplay.d: CFLAGS += $(SDL_CFLAGS)
 
-%$(PROGSSUF)_g$(EXESUF): %.o $(FF_DEP_LIBS)
-	$(LD) $(LDFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
+VERSION_SH  = $(SRC_PATH_BARE)/version.sh
+GIT_LOG     = $(SRC_PATH_BARE)/.git/logs/HEAD
+SVN_ENTRIES = $(SRC_PATH_BARE)/.svn/entries
 
-OBJDIRS += tools
-
--include $(wildcard tools/*.d)
-
-VERSION_SH  = $(SRC_PATH)/version.sh
-GIT_LOG     = $(SRC_PATH)/.git/logs/HEAD
-
-.version: $(wildcard $(GIT_LOG)) $(VERSION_SH) config.mak
+.version: $(wildcard $(GIT_LOG) $(SVN_ENTRIES)) $(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
-install: install-progs install-data
-endif
+alltools: $(TOOLS)
 
-install: install-libs install-headers
+documentation: $(addprefix doc/, developer.html faq.html ffmpeg-doc.html \
+                                 ffplay-doc.html ffprobe-doc.html ffserver-doc.html \
+                                 general.html libavfilter.html $(ALLMANPAGES))
 
-install-libs: install-libs-yes
+doc/%.html: TAG = HTML
+doc/%.html: doc/%.texi
+	$(M)cd doc && texi2html -monolithic -number $(<:doc/%=%)
 
-install-progs-yes:
-install-progs-$(CONFIG_SHARED): install-libs
+doc/%.pod: TAG = POD
+doc/%.pod: doc/%-doc.texi
+	$(M)doc/texi2pod.pl $< $@
 
-install-progs: install-progs-yes $(AVPROGS)
+doc/%.1: TAG = MAN
+doc/%.1: doc/%.pod
+	$(M)pod2man --section=1 --center=" " --release=" " $< > $@
+
+install: $(INSTALL_TARGETS-yes)
+
+install-progs: $(PROGS) $(INSTALL_PROGS_TARGETS-yes)
 	$(Q)mkdir -p "$(BINDIR)"
-	$(INSTALL) -c -m 755 $(INSTPROGS) "$(BINDIR)"
+	$(INSTALL) -c -m 755 $(PROGS) "$(BINDIR)"
 
-install-data: $(DATA_FILES) $(EXAMPLES_FILES)
-	$(Q)mkdir -p "$(DATADIR)/examples"
+install-data: $(DATA_FILES)
+	$(Q)mkdir -p "$(DATADIR)"
 	$(INSTALL) -m 644 $(DATA_FILES) "$(DATADIR)"
-	$(INSTALL) -m 644 $(EXAMPLES_FILES) "$(DATADIR)/examples"
 
-uninstall: uninstall-libs uninstall-headers uninstall-progs uninstall-data
+install-man: $(MANPAGES)
+	$(Q)mkdir -p "$(MANDIR)/man1"
+	$(INSTALL) -m 644 $(MANPAGES) "$(MANDIR)/man1"
+
+uninstall: uninstall-progs uninstall-data uninstall-man
 
 uninstall-progs:
-	$(RM) $(addprefix "$(BINDIR)/", $(ALLAVPROGS))
+	$(RM) $(addprefix "$(BINDIR)/", $(ALLPROGS))
 
 uninstall-data:
 	$(RM) -r "$(DATADIR)"
 
-clean::
-	$(RM) $(ALLAVPROGS) $(ALLAVPROGS_G)
+uninstall-man:
+	$(RM) $(addprefix "$(MANDIR)/man1/",$(ALLMANPAGES))
+
+testclean:
+	$(RM) -r tests/vsynth1 tests/vsynth2 tests/data
+	$(RM) $(addprefix tests/,$(CLEANSUFFIXES))
+	$(RM) tests/seek_test$(EXESUF) tests/seek_test.o
+	$(RM) $(addprefix tests/,$(addsuffix $(HOSTEXESUF),audiogen videogen rotozoom tiny_psnr))
+
+clean:: testclean
+	$(RM) $(ALLPROGS) $(ALLPROGS_G)
 	$(RM) $(CLEANSUFFIXES)
-	$(RM) $(CLEANSUFFIXES:%=tools/%)
-	$(RM) -r coverage-html
-	$(RM) -rf coverage.info lcov
+	$(RM) doc/*.html doc/*.pod doc/*.1
+	$(RM) $(TOOLS)
 
 distclean::
 	$(RM) $(DISTCLEANSUFFIXES)
-	$(RM) config.* .config libavutil/avconfig.h .version version.h libavutil/ffversion.h libavcodec/codec_names.h
+	$(RM) version.h config.* libavutil/avconfig.h
 
 config:
 	$(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION)
 
-check: all alltools examples testprogs fate
+# regression tests
 
-include $(SRC_PATH)/tests/Makefile
+check: test checkheaders
 
-$(sort $(OBJDIRS)):
-	$(Q)mkdir -p $@
+fulltest test: codectest lavftest seektest
 
-# Dummy rule to stop make trying to rebuild removed or renamed headers
-%.h:
-	@:
+FFSERVER_REFFILE = $(SRC_PATH)/tests/ffserver.regression.ref
+SEEK_REFFILE     = $(SRC_PATH)/tests/seek.regression.ref
 
-# Disable suffix rules.  Most of the builtin rules are suffix rules,
-# so this saves some time on slow systems.
-.SUFFIXES:
+ENCDEC = $(and $(CONFIG_$(1)_ENCODER),$(CONFIG_$(1)_DECODER))
+MUXDEM = $(and $(CONFIG_$(1)_MUXER),$(CONFIG_$(or $(2),$(1))_DEMUXER))
 
-.PHONY: all all-yes alltools check *clean config install*
-.PHONY: testprogs uninstall*
+VCODEC_TESTS =
+VCODEC_TESTS-$(call ENCDEC,ASV1)             += asv1
+VCODEC_TESTS-$(call ENCDEC,ASV2)             += asv2
+VCODEC_TESTS-$(call ENCDEC,DNXHD)            += dnxhd_1080i dnxhd_720p dnxhd_720p_rd
+VCODEC_TESTS-$(call ENCDEC,DVVIDEO)          += dv dv50
+VCODEC_TESTS-$(call ENCDEC,FFV1)             += ffv1
+VCODEC_TESTS-$(call ENCDEC,FLASHSV)          += flashsv
+VCODEC_TESTS-$(call ENCDEC,FLV)              += flv
+VCODEC_TESTS-$(call ENCDEC,H261)             += h261
+VCODEC_TESTS-$(call ENCDEC,H263)             += h263 h263p
+VCODEC_TESTS-$(call ENCDEC,HUFFYUV)          += huffyuv
+VCODEC_TESTS-$(call ENCDEC,JPEGLS)           += jpegls
+VCODEC_TESTS-$(call ENCDEC,MJPEG)            += mjpeg ljpeg
+VCODEC_TESTS-$(call ENCDEC,MPEG1VIDEO)       += mpeg mpeg1b
+VCODEC_TESTS-$(call ENCDEC,MPEG2VIDEO)       += mpeg2 mpeg2thread
+VCODEC_TESTS-$(call ENCDEC,MPEG4)            += mpeg4 mpeg4adv mpeg4nr mpeg4thread error rc
+VCODEC_TESTS-$(call ENCDEC,MSMPEG4V1)        += msmpeg4
+VCODEC_TESTS-$(call ENCDEC,MSMPEG4V2)        += msmpeg4v2
+VCODEC_TESTS-$(call ENCDEC,ROQ)              += roq
+VCODEC_TESTS-$(call ENCDEC,RV10)             += rv10
+VCODEC_TESTS-$(call ENCDEC,RV20)             += rv20
+VCODEC_TESTS-$(call ENCDEC,SNOW)             += snow snowll
+VCODEC_TESTS-$(call ENCDEC,SVQ1)             += svq1
+VCODEC_TESTS-$(call ENCDEC,WMV1)             += wmv1
+VCODEC_TESTS-$(call ENCDEC,WMV2)             += wmv2
+
+ACODEC_TESTS =
+ACODEC_TESTS-$(call ENCDEC,AC3)              += ac3
+ACODEC_TESTS-$(call ENCDEC,ADPCM_G726)       += g726
+ACODEC_TESTS-$(call ENCDEC,ADPCM_IMA_QT)     += adpcm_ima_qt
+ACODEC_TESTS-$(call ENCDEC,ADPCM_IMA_WAV)    += adpcm_ima_wav
+ACODEC_TESTS-$(call ENCDEC,ADPCM_MS)         += adpcm_ms
+ACODEC_TESTS-$(call ENCDEC,ADPCM_SWF)        += adpcm_swf
+ACODEC_TESTS-$(call ENCDEC,ADPCM_YAMAHA)     += adpcm_yam
+ACODEC_TESTS-$(call ENCDEC,ALAC)             += alac
+ACODEC_TESTS-$(call ENCDEC,FLAC)             += flac
+ACODEC_TESTS-$(call ENCDEC,MP2)              += mp2
+ACODEC_TESTS-$(call ENCDEC,PCM_S16LE)        += pcm         # fixme
+ACODEC_TESTS-$(call ENCDEC,WMAV1)            += wmav1
+ACODEC_TESTS-$(call ENCDEC,WMAV1)            += wmav2
+
+LAVF_TESTS =
+LAVF_TESTS-$(call MUXDEM,AIFF)               += aiff
+LAVF_TESTS-$(call MUXDEM,PCM_ALAW)           += alaw
+LAVF_TESTS-$(call MUXDEM,ASF)                += asf
+LAVF_TESTS-$(call MUXDEM,AU)                 += au
+LAVF_TESTS-$(call MUXDEM,AVI)                += avi
+LAVF_TESTS-$(call ENCDEC,BMP)                += bmp
+LAVF_TESTS-$(call MUXDEM,DV)                 += dv_fmt
+LAVF_TESTS-$(call MUXDEM,FFM)                += ffm
+LAVF_TESTS-$(call MUXDEM,FLV)                += flv_fmt
+LAVF_TESTS-$(call ENCDEC,GIF)                += gif
+LAVF_TESTS-$(call MUXDEM,GXF)                += gxf
+LAVF_TESTS-$(call ENCDEC,MJPEG)              += jpg
+LAVF_TESTS-$(call MUXDEM,MATROSKA)           += mkv
+LAVF_TESTS-$(call MUXDEM,MMF)                += mmf
+LAVF_TESTS-$(call MUXDEM,MOV)                += mov
+LAVF_TESTS-$(call MUXDEM,MPEG1SYSTEM,MPEGPS) += mpg
+LAVF_TESTS-$(call MUXDEM,PCM_MULAW)          += mulaw
+LAVF_TESTS-$(call MUXDEM,MXF)                += mxf
+LAVF_TESTS-$(call MUXDEM,NUT)                += nut
+LAVF_TESTS-$(call MUXDEM,OGG)                += ogg
+LAVF_TESTS-$(call ENCDEC,PBM)                += pbmpipe
+LAVF_TESTS-$(call ENCDEC,PCX)                += pcx
+LAVF_TESTS-$(call ENCDEC,PGM)                += pgm pgmpipe
+LAVF_TESTS-$(call MUXDEM,RAWVIDEO)           += pixfmt
+LAVF_TESTS-$(call ENCDEC,PPM)                += ppm ppmpipe
+LAVF_TESTS-$(call MUXDEM,RM)                 += rm
+LAVF_TESTS-$(call ENCDEC,SGI)                += sgi
+LAVF_TESTS-$(call MUXDEM,SWF)                += swf
+LAVF_TESTS-$(call ENCDEC,TARGA)              += tga
+LAVF_TESTS-$(call ENCDEC,TIFF)               += tiff
+LAVF_TESTS-$(call MUXDEM,MPEGTS)             += ts
+LAVF_TESTS-$(call MUXDEM,VOC)                += voc
+LAVF_TESTS-$(call MUXDEM,WAV)                += wav
+LAVF_TESTS-$(call MUXDEM,YUV4MPEGPIPE)       += yuv4mpeg
+
+LAVFI_TESTS =           \
+    crop                \
+    crop_scale          \
+    crop_scale_vflip    \
+    crop_vflip          \
+    null                \
+    scale200            \
+    scale500            \
+    vflip               \
+    vflip_crop          \
+    vflip_vflip         \
+
+ACODEC_TESTS := $(addprefix regtest-, $(ACODEC_TESTS) $(ACODEC_TESTS-yes))
+VCODEC_TESTS := $(addprefix regtest-, $(VCODEC_TESTS) $(VCODEC_TESTS-yes))
+LAVF_TESTS  := $(addprefix regtest-, $(LAVF_TESTS)  $(LAVF_TESTS-yes))
+LAVFI_TESTS := $(addprefix regtest-, $(LAVFI_TESTS) $(LAVFI_TESTS-yes))
+
+CODEC_TESTS = $(VCODEC_TESTS) $(ACODEC_TESTS)
+
+codectest: $(CODEC_TESTS)
+lavftest:  $(LAVF_TESTS)
+lavfitest: $(LAVFI_TESTS)
+
+$(ACODEC_TESTS): regtest-aref
+$(VCODEC_TESTS): regtest-vref
+$(LAVF_TESTS) $(LAVFI_TESTS): regtest-ref
+
+REFFILE = $(SRC_PATH)/tests/ref/$(1)/$(2:regtest-%=%)
+RESFILE = tests/data/$(2:regtest-%=%).$(1).regression
+
+define CODECTEST_CMD
+	$(SRC_PATH)/tests/codec-regression.sh $@ vsynth1 tests/vsynth1 "$(TARGET_EXEC)" "$(TARGET_PATH)"
+	$(SRC_PATH)/tests/codec-regression.sh $@ vsynth2 tests/vsynth2 "$(TARGET_EXEC)" "$(TARGET_PATH)"
+endef
+
+regtest-ref: regtest-aref regtest-vref
+
+regtest-vref: ffmpeg$(EXESUF) tests/vsynth1/00.pgm tests/vsynth2/00.pgm
+	$(CODECTEST_CMD)
+
+regtest-aref: ffmpeg$(EXESUF) tests/data/asynth1.sw
+	@$(SRC_PATH)/tests/codec-regression.sh $@ acodec tests/acodec "$(TARGET_EXEC)" "$(TARGET_PATH)"
+
+$(VCODEC_TESTS): tests/tiny_psnr$(HOSTEXESUF)
+	@echo "TEST VCODEC $(@:regtest-%=%)"
+	@$(CODECTEST_CMD)
+	@diff -u -w $(call REFFILE,vsynth1,$@) $(call RESFILE,vsynth1,$@)
+	@diff -u -w $(call REFFILE,vsynth2,$@) $(call RESFILE,vsynth2,$@)
+
+$(ACODEC_TESTS): tests/tiny_psnr$(HOSTEXESUF)
+	@echo "TEST ACODEC $(@:regtest-%=%)"
+	@$(SRC_PATH)/tests/codec-regression.sh $@ acodec tests/acodec "$(TARGET_EXEC)" "$(TARGET_PATH)"
+	@diff -u -w $(call REFFILE,acodec,$@) $(call RESFILE,acodec,$@)
+
+$(LAVF_TESTS):
+	@echo "TEST LAVF  $(@:regtest-%=%)"
+	@$(SRC_PATH)/tests/lavf-regression.sh $@ lavf tests/vsynth1 "$(TARGET_EXEC)" "$(TARGET_PATH)"
+	@diff -u -w $(call REFFILE,lavf,$@) $(call RESFILE,lavf,$@)
+
+$(LAVFI_TESTS):
+	@echo "TEST LAVFI $(@:regtest-%=%)"
+	@$(SRC_PATH)/tests/lavfi-regression.sh $@ lavfi tests/vsynth1 "$(TARGET_EXEC)" "$(TARGET_PATH)"
+	@diff -u -w $(call REFFILE,lavfi,$@) $(call RESFILE,lavfi,$@)
+
+seektest: codectest lavftest tests/seek_test$(EXESUF)
+	$(SRC_PATH)/tests/seek-regression.sh $(SRC_PATH) "$(TARGET_EXEC)" "$(TARGET_PATH)"
+
+ffservertest: ffserver$(EXESUF) tests/vsynth1/00.pgm tests/data/asynth1.sw
+	@echo
+	@echo "Unfortunately ffserver is broken and therefore its regression"
+	@echo "test fails randomly. Treat the results accordingly."
+	@echo
+	$(SRC_PATH)/tests/ffserver-regression.sh $(FFSERVER_REFFILE) $(SRC_PATH)/tests/ffserver.conf
+
+tests/vsynth1/00.pgm: tests/videogen$(HOSTEXESUF)
+	mkdir -p tests/vsynth1
+	$(BUILD_ROOT)/$< 'tests/vsynth1/'
+
+tests/vsynth2/00.pgm: tests/rotozoom$(HOSTEXESUF)
+	mkdir -p tests/vsynth2
+	$(BUILD_ROOT)/$< 'tests/vsynth2/' $(SRC_PATH)/tests/lena.pnm
+
+tests/data/asynth1.sw: tests/audiogen$(HOSTEXESUF)
+	mkdir -p tests/data
+	$(BUILD_ROOT)/$< $@
+
+tests/seek_test$(EXESUF): tests/seek_test.o $(FF_DEP_LIBS)
+	$(LD) $(FF_LDFLAGS) -o $@ $< $(FF_EXTRALIBS)
+
+ifdef SAMPLES
+include $(SRC_PATH_BARE)/tests/fate.mak
+fate: $(FATE_TESTS)
+$(FATE_TESTS): ffmpeg$(EXESUF)
+	@echo "TEST FATE   $(@:fate-%=%)"
+	@$(SRC_PATH)/tests/fate-run.sh $@ "$(SAMPLES)" "$(TARGET_EXEC)" "$(TARGET_PATH)" '$(CMD)'
+else
+fate:
+	@echo "SAMPLES not specified, cannot run FATE"
+endif
+
+.PHONY: documentation *test regtest-* zlib-error alltools check config

README

@@ -4,15 +4,9 @@ FFmpeg README
 1) Documentation
 ----------------
 
-* Read the documentation in the doc/ directory in git.
-  You can also view it online at http://ffmpeg.org/documentation.html
+* Read the documentation in the doc/ directory.
 
 2) Licensing
 ------------
 
 * See the LICENSE file.
-
-3) Build and Install
---------------------
-
-* See the INSTALL file.

RELEASE

@@ -1 +1,212 @@
-2.2.8
+Release Notes
+=============
+
+* 0.6 "Works with HTML5" June, 2010
+
+General notes
+-------------
+
+This release focuses on improvements for the new multimedia elements in HTML5.
+The H.264 and Theora decoders are now significantly faster, the Vorbis decoder
+has seen important updates and this release supports Google's newly released
+libvpx library for the VP8 codec and WebM container.
+
+Other important changes are additions of decoders including, but not limited to,
+Intel Indeo 5, WMA Pro, WMA Voice and HE-AAC.
+
+See the Changelog file for a list of significant changes.
+
+Please note that our policy on bug reports has not changed. We still only accept
+bug reports against HEAD of the FFmpeg trunk repository. If you are experiencing
+any issues with any formally released version of FFmpeg, please try a current
+version of the development code to check if the issue still exists. If it does,
+make your report against the development code following the usual bug reporting
+guidelines.
+
+
+API and other notable Changes
+-----------------------------
+
+Please see the file doc/APIchanges for programmer-centric information.
+
+Notable changes:
+- deprecated vhook subsystem removed
+- deprecated old scaler removed
+- nonfree libamr support for AMR-NB/WB decoding/encoding removed
+- RTMP support in libavformat
+- -formats option split into -formats, -codecs, -bsfs, and -protocols
+- ffprobe tool
+- RTMP/RTMPT/RTMPS/RTMPE/RTMPTE protocol support via librtmp
+- CODEC_CAP_EXPERIMENTAL added
+
+
+Added Codecs:
+-------------
+
+- VQF demuxer
+- PCX encoder
+- CorePNG decoding support
+- 8088flex TMV demuxer and decoder
+- enable symbol versioning by default for linkers that support it
+- V210 decoder and encoder
+- QCP demuxer
+- SoX native format muxer and demuxer
+- AMR-NB decoding/encoding, AMR-WB decoding via OpenCORE libraries
+- DPX image decoder
+- Electronic Arts Madcow decoder
+- DivX (XSUB) subtitle encoder
+- experimental AAC encoder
+- Wave64 demuxer
+- IEC-61937 compatible Muxer
+- TwinVQ decoder
+- Bluray (PGS) subtitle decoder
+- LPCM support in MPEG-TS (HDMV RID as found on Blu-ray disks)
+- WMA Pro decoder
+- Core Audio Format demuxer
+- Atrac1 decoder
+- MD STUDIO audio demuxer
+- RF64 support in WAV demuxer
+- MPEG-4 Audio Lossless Coding (ALS) decoder
+- IV8 demuxer
+- CDG demuxer and decoder
+- R210 decoder
+- Auravision Aura 1 and 2 decoders
+- Deluxe Paint Animation playback system
+- SIPR decoder
+- Adobe Filmstrip muxer and demuxer
+- RTP packetization and depacketization of H.263 and AMR
+- Bink demuxer and audio/video decoders
+- IFF PBM/ILBM bitmap decoder
+- Indeo 5 decoder
+- WMA Voice decoder
+- AMR-NB decoder
+- RTSP muxer
+- HE-AAC v1 decoder
+- Kega Game Video (KGV1) decoder
+- Psygnosis YOP demuxer and video decoder
+- RTP hinting in the mov/3gp/mp4 muxer
+- VP8 decoding via libvpx
+
+
+Notable license related changes
+-------------------------------
+
+- remaining GPL parts in AC-3 decoder converted to LGPL
+- libswscale can now be compiled in LGPL mode
+
+
+
+* 0.6.1
+
+General notes
+-------------
+
+This point release includes some updates to make the 0.6 release series usable
+for users that need to retain the existing behavior as closely as possible.
+The changes follow below:
+
+Bugfixes
+--------
+
+- fix autodetection of E-AC-3 substream samples
+- performance fix for seekable HTTP
+- add missing VP80 fourcc code for the VP8 codec
+- small documentation fixes
+- fix several potentially exploitable issues in the FLIC decoder
+  (addresses CVE-2010-3429)
+
+
+HE-AAC v2 backport
+------------------
+
+This release includes a backport of the AAC decoder from trunk, which
+enables proper playback of HE-AAC v2 media.
+
+
+* 0.6.2
+
+General notes
+-------------
+
+This is a maintenance-only release that addresses a small number of security
+and portability issues. Distributors and system integrators are encouraged
+to update and share their patches against this branch.
+
+Security fixes
+--------------
+
+Programming errors in container and codec implementations may lead to
+denial of service or the execution of arbitrary code if the user is
+tricked into opening a malformed media file or stream.
+
+Affected and updated have been the implementations of the following
+codecs and container formats:
+
+ - VC1 decoder (Change related to CVE-2011-0723)
+ - APE decoder (cf. http://packetstorm.linuxsecurity.com/1103-exploits/vlc105-dos.txt)
+
+
+* 0.6.3
+
+General notes
+-------------
+
+This is a mostly maintenance-only release that addresses a small number
+of bugs such as security and compilation issues. Moreover, this release
+has been updated to work with gcc-4.6 and the VisualOn AAC encoder has
+been backported from the Libav 0.7.2 release. Distributors and system
+integrators are encouraged to update and share their patches against
+this branch.
+For a full list of changes please see the Changelog file.
+
+
+* 0.6.4
+
+General notes
+-------------
+
+This mostly maintenance-only release that addresses a number a number of
+bugs such as security and compilation issues that have been brought to
+our attention. Among other (rather minor) fixes, this release features
+fixes for the QDM2 decoder (CVE-2011-4351), vp3 decoder (CVE-2011-4352),
+DoS in the VP5/VP6 decoders (CVE-2011-4353), a buffer overflow in the
+Sierra VMD decoder CVE-2011-4364, and a safety fix in the svq1 decoder
+(CVE-2011-4579).
+
+Distributors and system integrators are encouraged
+to update and share their patches against this branch.  For a full list
+of changes please see the Changelog file.
+
+
+* 0.6.5
+
+General notes
+-------------
+
+This mostly maintenance-only release that addresses a number a number of
+bugs such as security and compilation issues that have been brought to
+our attention. Among other (rather minor) fixes, this release features
+fixes for the VP3 decoder (CVE-2011-3892), vorbis decoder, and matroska
+demuxer (CVE-2011-3893 and CVE-2011-3895).
+
+Distributors and system integrators are encouraged
+to update and share their patches against this branch.  For a full list
+of changes please see the Changelog file.
+
+
+* 0.6.6
+
+General notes
+-------------
+
+This mostly maintenance-only release that addresses a number a number of
+bugs such as security and compilation issues that have been brought to
+our attention. Among other fixes, this release features includes
+security updates for the, DV decoder (CVE-2011-3929 and CVE-2011-3936),
+VQA Video Decoder (CVE-2012-0947), DPCM codecs (CVE-2011-3951), AAC SBR
+(CVE-2012-0850), H.264 (CVE-2012-0851), ADPCM (CVE-2012-0852), Shorten
+(CVE-2012-0858), and the KMVC decoder (CVE-2011-3952).
+
+Distributors and system integrators are encouraged
+to update and share their patches against this branch.  For a full list
+of changes please see the Changelog file or the git commit history.

VERSION

@@ -0,0 +1 @@
+0.6.7

arch.mak

@@ -1,16 +0,0 @@
-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_NEON)    += $(NEON-OBJS)    $(NEON-OBJS-yes)
-
-OBJS-$(HAVE_MIPSFPU)   += $(MIPSFPU-OBJS)    $(MIPSFPU-OBJS-yes)
-OBJS-$(HAVE_MIPS32R2)  += $(MIPS32R2-OBJS)   $(MIPS32R2-OBJS-yes)
-OBJS-$(HAVE_MIPSDSPR1) += $(MIPSDSPR1-OBJS)  $(MIPSDSPR1-OBJS-yes)
-OBJS-$(HAVE_MIPSDSPR2) += $(MIPSDSPR2-OBJS)  $(MIPSDSPR2-OBJS-yes)
-
-OBJS-$(HAVE_ALTIVEC) += $(ALTIVEC-OBJS) $(ALTIVEC-OBJS-yes)
-
-OBJS-$(HAVE_VIS)     += $(VIS-OBJS)     $(VIS-OBJS-yes)
-
-OBJS-$(HAVE_MMX)     += $(MMX-OBJS)     $(MMX-OBJS-yes)
-OBJS-$(HAVE_YASM)    += $(YASM-OBJS)    $(YASM-OBJS-yes)

cmdutils.h

@@ -22,17 +22,11 @@
 #ifndef FFMPEG_CMDUTILS_H
 #define FFMPEG_CMDUTILS_H
 
-#include <stdint.h>
-
+#include <inttypes.h>
 #include "libavcodec/avcodec.h"
-#include "libavfilter/avfilter.h"
 #include "libavformat/avformat.h"
 #include "libswscale/swscale.h"
 
-#ifdef __MINGW32__
-#undef main /* We don't want SDL to override our main() */
-#endif
-
 /**
  * program name, defined by the program for show_version().
  */
@@ -43,118 +37,61 @@ extern const char program_name[];
  */
 extern const int program_birth_year;
 
+extern const int this_year;
+
+extern const char **opt_names;
 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);
-
-/**
- * Initialize the cmdutils option system, in particular
- * allocate the *_opts contexts.
- */
-void init_opts(void);
-/**
- * Uninitialize the cmdutils option system, in particular
- * free the *_opts contexts and their contents.
- */
-void uninit_opts(void);
-
-/**
- * Trivial log callback.
- * Only suitable for opt_help and similar since it lacks prefix handling.
- */
-void log_callback_help(void* ptr, int level, const char* fmt, va_list vl);
-
-/**
- * Override the cpuflags.
- */
-int opt_cpuflags(void *optctx, const char *opt, const char *arg);
 
 /**
  * Fallback for options that are not explicitly handled, these will be
  * parsed through AVOptions.
  */
-int opt_default(void *optctx, const char *opt, const char *arg);
+int opt_default(const char *opt, const char *arg);
 
 /**
- * Set the libav* libraries log level.
+ * Sets the libav* libraries log level.
  */
-int opt_loglevel(void *optctx, const char *opt, const char *arg);
-
-int opt_report(const char *opt);
-
-int opt_max_alloc(void *optctx, const char *opt, const char *arg);
-
-int opt_codec_debug(void *optctx, const char *opt, const char *arg);
-
-#if CONFIG_OPENCL
-int opt_opencl(void *optctx, const char *opt, const char *arg);
-
-int opt_opencl_bench(void *optctx, const char *opt, const char *arg);
-#endif
+int opt_loglevel(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.
- * Exit from the application if the string cannot be correctly
+ * Parses a string and returns its corresponding value as a double.
+ * Exits from the application if the string cannot be correctly
  * parsed or the corresponding value is invalid.
  *
  * @param context the context of the value to be set (e.g. the
- * corresponding command line option name)
+ * corresponding commandline option name)
  * @param numstr the string to be parsed
  * @param type the type (OPT_INT64 or OPT_FLOAT) as which the
  * string should be parsed
  * @param min the minimum valid accepted value
  * @param max the maximum valid accepted value
  */
-double parse_number_or_die(const char *context, const char *numstr, int type,
-                           double min, double max);
+double parse_number_or_die(const char *context, const char *numstr, int type, double min, double max);
 
 /**
- * Parse a string specifying a time and return its corresponding
- * value as a number of microseconds. Exit from the application if
+ * Parses a string specifying a time and returns its corresponding
+ * value as a number of microseconds. Exits from the application if
  * the string cannot be correctly parsed.
  *
  * @param context the context of the value to be set (e.g. the
- * corresponding command line option name)
+ * corresponding commandline option name)
  * @param timestr the string to be parsed
  * @param is_duration a flag which tells how to interpret timestr, if
  * not zero timestr is interpreted as a duration, otherwise as a
  * date
  *
- * @see av_parse_time()
+ * @see parse_date()
  */
-int64_t parse_time_or_die(const char *context, const char *timestr,
-                          int is_duration);
+int64_t parse_time_or_die(const char *context, const char *timestr, int is_duration);
 
-typedef struct SpecifierOpt {
-    char *specifier;    /**< stream/chapter/program/... specifier */
-    union {
-        uint8_t *str;
-        int        i;
-        int64_t  i64;
-        float      f;
-        double   dbl;
-    } u;
-} SpecifierOpt;
-
-typedef struct OptionDef {
+typedef struct {
     const char *name;
     int flags;
 #define HAS_ARG    0x0001
@@ -163,233 +100,42 @@ 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_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
+#define OPT_FUNC2  0x0400
+#define OPT_INT64  0x0800
+#define OPT_EXIT   0x1000
      union {
-        void *dst_ptr;
-        int (*func_arg)(void *, const char *, const char *);
-        size_t off;
+        void (*func_arg)(const char *); //FIXME passing error code as int return would be nicer then exit() in the func
+        int *int_arg;
+        char **str_arg;
+        float *float_arg;
+        int (*func2_arg)(const char *, const char *);
+        int64_t *int64_arg;
     } 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
- * children.
- */
-void show_help_children(const AVClass *class, int flags);
-
-/**
- * Per-fftool specific help handler. Implemented in each
- * fftool, called by show_help().
- */
-void show_help_default(const char *opt, const char *arg);
-
-/**
- * Generic -h handler common to all fftools.
- */
-int show_help(void *optctx, const char *opt, const char *arg);
-
-/**
- * Parse the command line arguments.
- *
- * @param optctx an opaque options context
- * @param argc   number of command line arguments
- * @param argv   values of command line arguments
+ * Parses the command line arguments.
  * @param options Array with the definitions required to interpret every
- * option of the form: -option_name [argument]
+ * option of the form: -<option_name> [<argument>]
  * @param parse_arg_function Name of the function called to process every
  * argument without a leading option name flag. NULL if such arguments do
  * not have to be processed.
  */
-void parse_options(void *optctx, int argc, char **argv, const OptionDef *options,
-                   void (* parse_arg_function)(void *optctx, const char*));
+void parse_options(int argc, char **argv, const OptionDef *options,
+                   void (* parse_arg_function)(const char*));
+
+void set_context_opts(void *ctx, void *opts_ctx, int flags);
 
 /**
- * Parse one given option.
- *
- * @return on success 1 if arg was consumed, 0 otherwise; negative number on error
- */
-int parse_option(void *optctx, const char *opt, const char *arg,
-                 const OptionDef *options);
-
-/**
- * An option extracted from the commandline.
- * Cannot use AVDictionary because of options like -map which can be
- * used multiple times.
- */
-typedef struct Option {
-    const OptionDef  *opt;
-    const char       *key;
-    const char       *val;
-} Option;
-
-typedef struct OptionGroupDef {
-    /**< group name */
-    const char *name;
-    /**
-     * Option to be used as group separator. Can be NULL for groups which
-     * are terminated by a non-option argument (e.g. ffmpeg output files)
-     */
-    const char *sep;
-    /**
-     * Option flags that must be set on each option that is
-     * applied to this group
-     */
-    int flags;
-} OptionGroupDef;
-
-typedef struct OptionGroup {
-    const OptionGroupDef *group_def;
-    const char *arg;
-
-    Option *opts;
-    int  nb_opts;
-
-    AVDictionary *codec_opts;
-    AVDictionary *format_opts;
-    AVDictionary *resample_opts;
-    struct SwsContext *sws_opts;
-    AVDictionary *swr_opts;
-} OptionGroup;
-
-/**
- * A list of option groups that all have the same group type
- * (e.g. input files or output files)
- */
-typedef struct OptionGroupList {
-    const OptionGroupDef *group_def;
-
-    OptionGroup *groups;
-    int       nb_groups;
-} OptionGroupList;
-
-typedef struct OptionParseContext {
-    OptionGroup global_opts;
-
-    OptionGroupList *groups;
-    int           nb_groups;
-
-    /* parsing state */
-    OptionGroup cur_group;
-} OptionParseContext;
-
-/**
- * Parse an options group and write results into optctx.
- *
- * @param optctx an app-specific options context. NULL for global options group
- */
-int parse_optgroup(void *optctx, OptionGroup *g);
-
-/**
- * Split the commandline into an intermediate form convenient for further
- * processing.
- *
- * The commandline is assumed to be composed of options which either belong to a
- * group (those with OPT_SPEC, OPT_OFFSET or OPT_PERFILE) or are global
- * (everything else).
- *
- * A group (defined by an OptionGroupDef struct) is a sequence of options
- * terminated by either a group separator option (e.g. -i) or a parameter that
- * is not an option (doesn't start with -). A group without a separator option
- * must always be first in the supplied groups list.
- *
- * All options within the same group are stored in one OptionGroup struct in an
- * OptionGroupList, all groups with the same group definition are stored in one
- * OptionGroupList in OptionParseContext.groups. The order of group lists is the
- * same as the order of group definitions.
- */
-int split_commandline(OptionParseContext *octx, int argc, char *argv[],
-                      const OptionDef *options,
-                      const OptionGroupDef *groups, int nb_groups);
-
-/**
- * Free all allocated memory in an OptionParseContext.
- */
-void uninit_parse_context(OptionParseContext *octx);
-
-/**
- * Find the '-loglevel' option in the command line args and apply it.
- */
-void parse_loglevel(int argc, char **argv, const OptionDef *options);
-
-/**
- * Return index of option opt in argv or 0 if not found.
- */
-int locate_option(int argc, char **argv, const OptionDef *options,
-                  const char *optname);
-
-/**
- * Check if the given stream matches a stream specifier.
- *
- * @param s  Corresponding format context.
- * @param st Stream from s to be checked.
- * @param spec A stream specifier of the [v|a|s|d]:[\<stream index\>] form.
- *
- * @return 1 if the stream matches, 0 if it doesn't, <0 on error
- */
-int check_stream_specifier(AVFormatContext *s, AVStream *st, const char *spec);
-
-/**
- * Filter out options for given codec.
- *
- * Create a new options dictionary containing only the options from
- * opts which apply to the codec with ID codec_id.
- *
- * @param opts     dictionary to place options in
- * @param codec_id ID of the codec that should be filtered for
- * @param s Corresponding format context.
- * @param st A stream from s for which the options should be filtered.
- * @param codec The particular codec for which the options should be filtered.
- *              If null, the default one is looked up according to the codec id.
- * @return a pointer to the created dictionary
- */
-AVDictionary *filter_codec_opts(AVDictionary *opts, enum AVCodecID codec_id,
-                                AVFormatContext *s, AVStream *st, AVCodec *codec);
-
-/**
- * Setup AVCodecContext options for avformat_find_stream_info().
- *
- * Create an array of dictionaries, one dictionary for each stream
- * contained in s.
- * Each dictionary will contain the options from codec_opts which can
- * be applied to the corresponding stream codec context.
- *
- * @return pointer to the created array of dictionaries, NULL if it
- * cannot be created
- */
-AVDictionary **setup_find_stream_info_opts(AVFormatContext *s,
-                                           AVDictionary *codec_opts);
-
-/**
- * Print an error message to stderr, indicating filename and a human
+ * Prints an error message to stderr, indicating filename and a human
  * readable description of the error code err.
  *
  * If strerror_r() is not available the use of this function in a
@@ -399,180 +145,79 @@ AVDictionary **setup_find_stream_info_opts(AVFormatContext *s,
  */
 void print_error(const char *filename, int err);
 
+void list_fmts(void (*get_fmt_string)(char *buf, int buf_size, int fmt), int nb_fmts);
+
 /**
- * Print the program banner to stderr. The banner contents depend on the
+ * Prints the program banner to stderr. The banner contents depend on the
  * current version of the repository and of the libav* libraries used by
  * the program.
  */
-void show_banner(int argc, char **argv, const OptionDef *options);
+void show_banner(void);
 
 /**
- * Print the version of the program to stdout. The version message
+ * Prints the version of the program to stdout. The version message
  * depends on the current versions of the repository and of the libav*
  * libraries.
- * This option processing function does not utilize the arguments.
  */
-int show_version(void *optctx, const char *opt, const char *arg);
+void show_version(void);
 
 /**
- * Print the build configuration of the program to stdout. The contents
- * depend on the definition of FFMPEG_CONFIGURATION.
- * This option processing function does not utilize the arguments.
- */
-int show_buildconf(void *optctx, const char *opt, const char *arg);
-
-/**
- * Print the license of the program to stdout. The license depends on
+ * Prints 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);
+void show_license(void);
 
 /**
- * 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);
-
-/**
- * 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
+ * Prints a listing containing all the formats supported by the
  * program.
  */
-int show_decoders(void *optctx, const char *opt, const char *arg);
+void show_formats(void);
 
 /**
- * Print a listing containing all the encoders supported by the
+ * Prints a listing containing all the codecs supported by the
  * program.
  */
-int show_encoders(void *optctx, const char *opt, const char *arg);
+void show_codecs(void);
 
 /**
- * Print a listing containing all the filters supported by the
- * program.
- * This option processing function does not utilize the arguments.
- */
-int show_filters(void *optctx, const char *opt, const char *arg);
-
-/**
- * Print a listing containing all the bit stream filters supported by the
- * program.
- * This option processing function does not utilize the arguments.
- */
-int show_bsfs(void *optctx, const char *opt, const char *arg);
-
-/**
- * Print a listing containing all the protocols supported by the
- * program.
- * This option processing function does not utilize the arguments.
- */
-int show_protocols(void *optctx, const char *opt, const char *arg);
-
-/**
- * Print a listing containing all the pixel formats supported by the
- * program.
- * This option processing function does not utilize the arguments.
- */
-int show_pix_fmts(void *optctx, const char *opt, const char *arg);
-
-/**
- * Print a listing containing all the standard channel layouts supported by
- * the program.
- * This option processing function does not utilize the arguments.
- */
-int show_layouts(void *optctx, const char *opt, const char *arg);
-
-/**
- * Print a listing containing all the sample formats supported by the
+ * Prints a listing containing all the filters supported by the
  * program.
  */
-int show_sample_fmts(void *optctx, const char *opt, const char *arg);
+void show_filters(void);
 
 /**
- * Print a listing containing all the color names and values recognized
- * by the program.
+ * Prints a listing containing all the bit stream filters supported by the
+ * program.
  */
-int show_colors(void *optctx, const char *opt, const char *arg);
+void show_bsfs(void);
 
 /**
- * Return a positive value if a line read from standard input
- * starts with [yY], otherwise return 0.
+ * Prints a listing containing all the protocols supported by the
+ * program.
+ */
+void show_protocols(void);
+
+/**
+ * Prints a listing containing all the pixel formats supported by the
+ * program.
+ */
+void show_pix_fmts(void);
+
+/**
+ * Returns a positive value if reads from standard input a line
+ * starting with [yY], otherwise returns 0.
  */
 int read_yesno(void);
 
 /**
- * Read the file with name filename, and put its content in a newly
+ * Reads the file with name filename, and puts 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
+ * @param bufptr puts here the pointer to the newly allocated buffer
+ * @param size puts here the size of the newly allocated buffer
+ * @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);
+int read_file(const char *filename, char **bufptr, size_t *size);
 
-/**
- * Get a file corresponding to a preset file.
- *
- * If is_path is non-zero, look for the file in the path preset_name.
- * Otherwise search for a file named arg.ffpreset in the directories
- * $FFMPEG_DATADIR (if set), $HOME/.ffmpeg, and in the datadir defined
- * at configuration time or in a "ffpresets" folder along the executable
- * on win32, in that order. If no such file is found and
- * codec_name is defined, then search for a file named
- * codec_name-preset_name.avpreset in the above-mentioned directories.
- *
- * @param filename buffer where the name of the found filename is written
- * @param filename_size size in bytes of the filename buffer
- * @param preset_name name of the preset to search
- * @param is_path tell if preset_name is a filename path
- * @param codec_name name of the codec for which to look for the
- * preset, may be NULL
- */
-FILE *get_preset_file(char *filename, size_t filename_size,
-                      const char *preset_name, int is_path, const char *codec_name);
-
-/**
- * Realloc array to hold new_size elements of elem_size.
- * Calls exit() on failure.
- *
- * @param array array to reallocate
- * @param elem_size size in bytes of each element
- * @param size new element count will be written here
- * @param new_size number of elements to place in reallocated array
- * @return reallocated array
- */
-void *grow_array(void *array, int elem_size, int *size, int new_size);
-
-#define media_type_string av_get_media_type_string
-
-#define GROW_ARRAY(array, nb_elems)\
-    array = grow_array(array, sizeof(*array), &nb_elems, nb_elems + 1)
-
-#define GET_PIX_FMT_NAME(pix_fmt)\
-    const char *name = av_get_pix_fmt_name(pix_fmt);
-
-#define GET_SAMPLE_FMT_NAME(sample_fmt)\
-    const char *name = av_get_sample_fmt_name(sample_fmt)
-
-#define GET_SAMPLE_RATE_NAME(rate)\
-    char name[16];\
-    snprintf(name, sizeof(name), "%d", rate);
-
-#define GET_CH_LAYOUT_NAME(ch_layout)\
-    char name[16];\
-    snprintf(name, sizeof(name), "0x%"PRIx64, ch_layout);
-
-#define GET_CH_LAYOUT_DESC(ch_layout)\
-    char name[128];\
-    av_get_channel_layout_string(name, sizeof(name), 0, ch_layout);
-
-#endif /* CMDUTILS_H */
+#endif /* FFMPEG_CMDUTILS_H */

cmdutils_common_opts.h

@@ -1,28 +1,13 @@
-    { "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" },
-    { "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
+    { "L", OPT_EXIT, {(void*)show_license}, "show license" },
+    { "h", OPT_EXIT, {(void*)show_help}, "show help" },
+    { "?", OPT_EXIT, {(void*)show_help}, "show help" },
+    { "help", OPT_EXIT, {(void*)show_help}, "show help" },
+    { "-help", OPT_EXIT, {(void*)show_help}, "show help" },
+    { "version", OPT_EXIT, {(void*)show_version}, "show version" },
+    { "formats"  , OPT_EXIT, {(void*)show_formats  }, "show available formats" },
+    { "codecs"   , OPT_EXIT, {(void*)show_codecs   }, "show available codecs" },
+    { "bsfs"     , OPT_EXIT, {(void*)show_bsfs     }, "show available bit stream filters" },
+    { "protocols", OPT_EXIT, {(void*)show_protocols}, "show available protocols" },
+    { "filters",   OPT_EXIT, {(void*)show_filters  }, "show available filters" },
+    { "pix_fmts" , OPT_EXIT, {(void*)show_pix_fmts }, "show available pixel formats" },
+    { "loglevel", HAS_ARG | OPT_FUNC2, {(void*)opt_loglevel}, "set libav* logging level", "loglevel" },

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_array(nb_devices, sizeof(OpenCLDeviceBenchmark)))) {
-        av_log(NULL, AV_LOG_ERROR, "Could not allocate buffer\n");
-        return AVERROR(ENOMEM);
-    }
-
-    for (i = 0; i < device_list->platform_num; i++) {
-        for (j = 0; j < device_list->platform_node[i]->device_num; j++) {
-            device_node = device_list->platform_node[i]->device_node[j];
-            platform = device_list->platform_node[i]->platform_id;
-            score = av_opencl_benchmark(device_node, platform, run_opencl_bench);
-            if (score > 0) {
-                devices[count].platform_idx = i;
-                devices[count].device_idx = j;
-                devices[count].runtime = score;
-                strcpy(devices[count].device_name, device_node->device_name);
-                count++;
-            }
-        }
-    }
-    qsort(devices, count, sizeof(OpenCLDeviceBenchmark), compare_ocl_device_desc);
-    fprintf(stderr, "platform_idx\tdevice_idx\tdevice_name\truntime\n");
-    for (i = 0; i < count; i++)
-        fprintf(stdout, "%d\t%d\t%s\t%"PRId64"\n",
-                devices[i].platform_idx, devices[i].device_idx,
-                devices[i].device_name, devices[i].runtime);
-
-    av_opencl_free_device_list(&device_list);
-    av_free(devices);
-    return 0;
-}
-
-int opt_opencl(void *optctx, const char *opt, const char *arg)
-{
-    char *key, *value;
-    const char *opts = arg;
-    int ret = 0;
-    while (*opts) {
-        ret = av_opt_get_key_value(&opts, "=", ":", 0, &key, &value);
-        if (ret < 0)
-            return ret;
-        ret = av_opencl_set_option(key, value);
-        if (ret < 0)
-            return ret;
-        if (*opts)
-            opts++;
-    }
-    return ret;
-}

common.mak

@@ -6,77 +6,60 @@
 all: all-yes
 
 ifndef SUBDIR
+vpath %.c   $(SRC_DIR)
+vpath %.h   $(SRC_DIR)
+vpath %.S   $(SRC_DIR)
+vpath %.asm $(SRC_DIR)
+vpath %.v   $(SRC_DIR)
+
+ifeq ($(SRC_DIR),$(SRC_PATH_BARE))
+BUILD_ROOT_REL = .
+else
+BUILD_ROOT_REL = ..
+endif
 
 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 AS YASM AR LD HOSTCC STRIP CP
+SILENT = DEPCC YASMDEP RM RANLIB
 MSG    = $@
 M      = @$(call ECHO,$(TAG),$@);
 $(foreach VAR,$(BRIEF), \
-    $(eval override $(VAR) = @$$(call ECHO,$(VAR),$$(MSG)); $($(VAR))))
-$(foreach VAR,$(SILENT),$(eval override $(VAR) = @$($(VAR))))
+    $(eval $(VAR) = @$$(call ECHO,$(VAR),$$(MSG)); $($(VAR))))
+$(foreach VAR,$(SILENT),$(eval $(VAR) = @$($(VAR))))
 $(eval INSTALL = @$(call ECHO,INSTALL,$$(^:$(SRC_DIR)/%=%)); $(INSTALL))
 endif
 
-ALLFFLIBS = avcodec avdevice avfilter avformat avresample avutil postproc swscale swresample
+ALLFFLIBS = avcodec avdevice avfilter avformat avutil postproc swscale
 
-# NASM requires -I path terminated with /
-IFLAGS     := -I. -I$(SRC_PATH)/
-CPPFLAGS   := $(IFLAGS) $(CPPFLAGS)
-CFLAGS     += $(ECFLAGS)
-CCFLAGS     = $(CPPFLAGS) $(CFLAGS)
-ASFLAGS    := $(CPPFLAGS) $(ASFLAGS)
-CXXFLAGS   += $(CPPFLAGS) $(CFLAGS)
-YASMFLAGS  += $(IFLAGS:%=%/) -Pconfig.asm
-
-HOSTCCFLAGS = $(IFLAGS) $(HOSTCPPFLAGS) $(HOSTCFLAGS)
-LDFLAGS    := $(ALLFFLIBS:%=$(LD_PATH)lib%) $(LDFLAGS)
-
-define COMPILE
-       $(call $(1)DEP,$(1))
-       $($(1)) $($(1)FLAGS) $($(1)_DEPFLAGS) $($(1)_C) $($(1)_O) $<
-endef
-
-COMPILE_C = $(call COMPILE,CC)
-COMPILE_CXX = $(call COMPILE,CXX)
-COMPILE_S = $(call COMPILE,AS)
-COMPILE_HOSTC = $(call COMPILE,HOSTCC)
+CPPFLAGS := -I$(BUILD_ROOT_REL) -I$(SRC_PATH) $(CPPFLAGS)
+CFLAGS   += $(ECFLAGS)
 
 %.o: %.c
-	$(COMPILE_C)
-
-%.o: %.cpp
-	$(COMPILE_CXX)
-
-%.s: %.c
-	$(CC) $(CPPFLAGS) $(CFLAGS) -S -o $@ $<
+	$(CCDEP)
+	$(CC) $(CPPFLAGS) $(CFLAGS) $(CC_DEPFLAGS) -c $(CC_O) $<
 
 %.o: %.S
-	$(COMPILE_S)
+	$(ASDEP)
+	$(AS) $(CPPFLAGS) $(ASFLAGS) $(AS_DEPFLAGS) -c -o $@ $<
 
-%_host.o: %.c
-	$(COMPILE_HOSTC)
+%.ho: %.h
+	$(CC) $(CPPFLAGS) $(CFLAGS) -Wno-unused -c -o $@ -x c $<
 
-%.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"' >$@
+%$(EXESUF): %.c
 
 %.ver: %.v
 	$(Q)sed 's/$$MAJOR/$($(basename $(@F))_VERSION_MAJOR)/' $^ > $@
 
 %.c %.h: TAG = GEN
 
-# Dummy rule to stop make trying to rebuild removed or renamed headers
-%.h:
-	@:
+install: install-libs install-headers
+install-libs: install-libs-yes
+
+uninstall: uninstall-libs uninstall-headers
+
+.PHONY: all depend dep *clean install* uninstall* examples testprogs
 
 # Disable suffix rules.  Most of the builtin rules are suffix rules,
 # so this saves some time on slow systems.
@@ -86,66 +69,41 @@ COMPILE_HOSTC = $(call COMPILE,HOSTCC)
 $(OBJS):
 endif
 
-include $(SRC_PATH)/arch.mak
+OBJS-$(HAVE_MMX) +=  $(MMX-OBJS-yes)
 
+CFLAGS    += $(CFLAGS-yes)
 OBJS      += $(OBJS-yes)
-SLIBOBJS  += $(SLIBOBJS-yes)
 FFLIBS    := $(FFLIBS-yes) $(FFLIBS)
 TESTPROGS += $(TESTPROGS-yes)
 
-LDLIBS       = $(FFLIBS:%=%$(BUILDSUF))
-FFEXTRALIBS := $(LDLIBS:%=$(LD_LIB)) $(EXTRALIBS)
+FFEXTRALIBS := $(addprefix -l,$(addsuffix $(BUILDSUF),$(FFLIBS))) $(EXTRALIBS)
+FFLDFLAGS   := $(addprefix -L$(BUILD_ROOT)/lib,$(ALLFFLIBS)) $(LDFLAGS)
 
-OBJS      := $(sort $(OBJS:%=$(SUBDIR)%))
-SLIBOBJS  := $(sort $(SLIBOBJS:%=$(SUBDIR)%))
-TESTOBJS  := $(TESTOBJS:%=$(SUBDIR)%) $(TESTPROGS:%=$(SUBDIR)%-test.o)
-TESTPROGS := $(TESTPROGS:%=$(SUBDIR)%-test$(EXESUF))
-HOSTOBJS  := $(HOSTPROGS:%=$(SUBDIR)%.o)
-HOSTPROGS := $(HOSTPROGS:%=$(SUBDIR)%$(HOSTEXESUF))
-TOOLS     += $(TOOLS-yes)
-TOOLOBJS  := $(TOOLS:%=tools/%.o)
-TOOLS     := $(TOOLS:%=tools/%$(EXESUF))
-HEADERS   += $(HEADERS-yes)
+EXAMPLES  := $(addprefix $(SUBDIR),$(addsuffix -example$(EXESUF),$(EXAMPLES)))
+OBJS      := $(addprefix $(SUBDIR),$(sort $(OBJS)))
+TESTOBJS  := $(addprefix $(SUBDIR),$(TESTOBJS))
+TESTPROGS := $(addprefix $(SUBDIR),$(addsuffix -test$(EXESUF),$(TESTPROGS)))
+HOSTOBJS  := $(addprefix $(SUBDIR),$(addsuffix .o,$(HOSTPROGS)))
+HOSTPROGS := $(addprefix $(SUBDIR),$(addsuffix $(HOSTEXESUF),$(HOSTPROGS)))
 
-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),$(BUILD_ROOT_REL)/lib$(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)
-
-alltools: $(TOOLS)
+SKIPHEADERS += $(addprefix $(ARCH)/,$(ARCH_HEADERS))
+SKIPHEADERS := $(addprefix $(SUBDIR),$(SKIPHEADERS-) $(SKIPHEADERS))
+checkheaders: $(filter-out $(SKIPHEADERS:.h=.ho),$(ALLHEADERS:.h=.ho))
 
 $(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
+DEPS := $(OBJS:.o=.d)
+depend dep: $(DEPS)
 
-OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(SLIBOBJS) $(TESTOBJS))
-
-CLEANSUFFIXES     = *.d *.o *~ *.h.c *.map *.ver *.ho *.gcno *.gcda
+CLEANSUFFIXES     = *.d *.o *~ *.ho *.map *.ver
 DISTCLEANSUFFIXES = *.pc
-LIBSUFFIXES       = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a
+LIBSUFFIXES       = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a *.exp
 
-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 $(DEPS))

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 */

compat/avisynth/avisynth_c.h

@@ -1,880 +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., 51 Franklin Street, Fifth Floor, Boston,
-// MA 02110-1301 USA, or visit
-// http://www.gnu.org/copyleft/gpl.html .
-//
-// As a special exception, I give you permission to link to the
-// Avisynth C interface with independent modules that communicate with
-// the Avisynth C interface solely through the interfaces defined in
-// avisynth_c.h, regardless of the license terms of these independent
-// modules, and to copy and distribute the resulting combined work
-// under terms of your choice, provided that every copy of the
-// combined work is accompanied by a complete copy of the source code
-// of the Avisynth C interface and Avisynth itself (with the version
-// used to produce the combined work), being distributed under the
-// terms of the GNU General Public License plus this exception.  An
-// independent module is a module which is not derived from or based
-// on Avisynth C Interface, such as 3rd-party filters, import and
-// export plugins, or graphical user interfaces.
-
-// NOTE: this is a partial update of the Avisynth C interface to recognize
-// new color spaces added in Avisynth 2.60. By no means is this document
-// completely Avisynth 2.60 compliant.
-
-#ifndef __AVISYNTH_C__
-#define __AVISYNTH_C__
-
-#ifdef __cplusplus
-#  define EXTERN_C extern "C"
-#else
-#  define EXTERN_C
-#endif
-
-#define AVSC_USE_STDCALL 1
-
-#ifndef AVSC_USE_STDCALL
-#  define AVSC_CC __cdecl
-#else
-#  define AVSC_CC __stdcall
-#endif
-
-#define AVSC_INLINE static __inline
-
-#ifdef AVISYNTH_C_EXPORTS
-#  define AVSC_EXPORT EXTERN_C
-#  define AVSC_API(ret, name) EXTERN_C __declspec(dllexport) ret AVSC_CC name
-#else
-#  define AVSC_EXPORT EXTERN_C __declspec(dllexport)
-#  ifndef AVSC_NO_DECLSPEC
-#    define AVSC_API(ret, name) EXTERN_C __declspec(dllimport) ret AVSC_CC name
-#  else
-#    define AVSC_API(ret, name) typedef ret (AVSC_CC *name##_func)
-#  endif
-#endif
-
-typedef unsigned char BYTE;
-#ifdef __GNUC__
-typedef long long int INT64;
-#else
-typedef __int64 INT64;
-#endif
-
-
-/////////////////////////////////////////////////////////////////////
-//
-// Constants
-//
-
-#ifndef __AVISYNTH_H__
-enum { AVISYNTH_INTERFACE_VERSION = 4 };
-#endif
-
-enum {AVS_SAMPLE_INT8  = 1<<0,
-      AVS_SAMPLE_INT16 = 1<<1,
-      AVS_SAMPLE_INT24 = 1<<2,
-      AVS_SAMPLE_INT32 = 1<<3,
-      AVS_SAMPLE_FLOAT = 1<<4};
-
-enum {AVS_PLANAR_Y=1<<0,
-      AVS_PLANAR_U=1<<1,
-      AVS_PLANAR_V=1<<2,
-      AVS_PLANAR_ALIGNED=1<<3,
-      AVS_PLANAR_Y_ALIGNED=AVS_PLANAR_Y|AVS_PLANAR_ALIGNED,
-      AVS_PLANAR_U_ALIGNED=AVS_PLANAR_U|AVS_PLANAR_ALIGNED,
-      AVS_PLANAR_V_ALIGNED=AVS_PLANAR_V|AVS_PLANAR_ALIGNED,
-      AVS_PLANAR_A=1<<4,
-      AVS_PLANAR_R=1<<5,
-      AVS_PLANAR_G=1<<6,
-      AVS_PLANAR_B=1<<7,
-      AVS_PLANAR_A_ALIGNED=AVS_PLANAR_A|AVS_PLANAR_ALIGNED,
-      AVS_PLANAR_R_ALIGNED=AVS_PLANAR_R|AVS_PLANAR_ALIGNED,
-      AVS_PLANAR_G_ALIGNED=AVS_PLANAR_G|AVS_PLANAR_ALIGNED,
-      AVS_PLANAR_B_ALIGNED=AVS_PLANAR_B|AVS_PLANAR_ALIGNED};
-
-  // Colorspace properties.
-enum {AVS_CS_BGR = 1<<28,
-      AVS_CS_YUV = 1<<29,
-      AVS_CS_INTERLEAVED = 1<<30,
-      AVS_CS_PLANAR = 1<<31,
-
-      AVS_CS_SHIFT_SUB_WIDTH   = 0,
-      AVS_CS_SHIFT_SUB_HEIGHT  = 1 << 3,
-      AVS_CS_SHIFT_SAMPLE_BITS = 1 << 4,
-
-      AVS_CS_SUB_WIDTH_MASK    = 7 << AVS_CS_SHIFT_SUB_WIDTH,
-      AVS_CS_SUB_WIDTH_1       = 3 << AVS_CS_SHIFT_SUB_WIDTH, // YV24
-      AVS_CS_SUB_WIDTH_2       = 0 << AVS_CS_SHIFT_SUB_WIDTH, // YV12, I420, YV16
-      AVS_CS_SUB_WIDTH_4       = 1 << AVS_CS_SHIFT_SUB_WIDTH, // YUV9, YV411
-
-      AVS_CS_VPLANEFIRST       = 1 << 3, // YV12, YV16, YV24, YV411, YUV9
-      AVS_CS_UPLANEFIRST       = 1 << 4, // I420
-
-      AVS_CS_SUB_HEIGHT_MASK   = 7 << AVS_CS_SHIFT_SUB_HEIGHT,
-      AVS_CS_SUB_HEIGHT_1      = 3 << AVS_CS_SHIFT_SUB_HEIGHT, // YV16, YV24, YV411
-      AVS_CS_SUB_HEIGHT_2      = 0 << AVS_CS_SHIFT_SUB_HEIGHT, // YV12, I420
-      AVS_CS_SUB_HEIGHT_4      = 1 << AVS_CS_SHIFT_SUB_HEIGHT, // YUV9
-
-      AVS_CS_SAMPLE_BITS_MASK  = 7 << AVS_CS_SHIFT_SAMPLE_BITS,
-      AVS_CS_SAMPLE_BITS_8     = 0 << AVS_CS_SHIFT_SAMPLE_BITS,
-      AVS_CS_SAMPLE_BITS_16    = 1 << AVS_CS_SHIFT_SAMPLE_BITS,
-      AVS_CS_SAMPLE_BITS_32    = 2 << AVS_CS_SHIFT_SAMPLE_BITS,
-
-      AVS_CS_PLANAR_MASK       = AVS_CS_PLANAR | AVS_CS_INTERLEAVED | AVS_CS_YUV | AVS_CS_BGR | AVS_CS_SAMPLE_BITS_MASK | AVS_CS_SUB_HEIGHT_MASK | AVS_CS_SUB_WIDTH_MASK,
-      AVS_CS_PLANAR_FILTER     = ~( AVS_CS_VPLANEFIRST | AVS_CS_UPLANEFIRST )};
-
-  // Specific colorformats
-enum {
-  AVS_CS_UNKNOWN = 0,
-  AVS_CS_BGR24 = 1<<0 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
-  AVS_CS_BGR32 = 1<<1 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
-  AVS_CS_YUY2 = 1<<2 | AVS_CS_YUV | AVS_CS_INTERLEAVED,
-  //  AVS_CS_YV12  = 1<<3  Reserved
-  //  AVS_CS_I420  = 1<<4  Reserved
-  AVS_CS_RAW32 = 1<<5 | AVS_CS_INTERLEAVED,
-
-  AVS_CS_YV24  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_1,  // YVU 4:4:4 planar
-  AVS_CS_YV16  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_2,  // YVU 4:2:2 planar
-  AVS_CS_YV12  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_2 | AVS_CS_SUB_WIDTH_2,  // YVU 4:2:0 planar
-  AVS_CS_I420  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_UPLANEFIRST | AVS_CS_SUB_HEIGHT_2 | AVS_CS_SUB_WIDTH_2,  // YUV 4:2:0 planar
-  AVS_CS_IYUV  = AVS_CS_I420,
-  AVS_CS_YV411 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_4,  // YVU 4:1:1 planar
-  AVS_CS_YUV9  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_4 | AVS_CS_SUB_WIDTH_4,  // YVU 4:1:0 planar
-  AVS_CS_Y8    = AVS_CS_PLANAR | AVS_CS_INTERLEAVED | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8                                              // Y   4:0:0 planar
-};
-
-enum {
-  AVS_IT_BFF = 1<<0,
-  AVS_IT_TFF = 1<<1,
-  AVS_IT_FIELDBASED = 1<<2};
-
-enum {
-  AVS_FILTER_TYPE=1,
-  AVS_FILTER_INPUT_COLORSPACE=2,
-  AVS_FILTER_OUTPUT_TYPE=9,
-  AVS_FILTER_NAME=4,
-  AVS_FILTER_AUTHOR=5,
-  AVS_FILTER_VERSION=6,
-  AVS_FILTER_ARGS=7,
-  AVS_FILTER_ARGS_INFO=8,
-  AVS_FILTER_ARGS_DESCRIPTION=10,
-  AVS_FILTER_DESCRIPTION=11};
-
-enum {  //SUBTYPES
-  AVS_FILTER_TYPE_AUDIO=1,
-  AVS_FILTER_TYPE_VIDEO=2,
-  AVS_FILTER_OUTPUT_TYPE_SAME=3,
-  AVS_FILTER_OUTPUT_TYPE_DIFFERENT=4};
-
-enum {
-  AVS_CACHE_NOTHING=0,
-  AVS_CACHE_RANGE=1,
-  AVS_CACHE_ALL=2,
-  AVS_CACHE_AUDIO=3,
-  AVS_CACHE_AUDIO_NONE=4,
-  AVS_CACHE_AUDIO_AUTO=5
-  };
-
-#define AVS_FRAME_ALIGN 16
-
-typedef struct AVS_Clip AVS_Clip;
-typedef struct AVS_ScriptEnvironment AVS_ScriptEnvironment;
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_VideoInfo
-//
-
-// AVS_VideoInfo is layed out identicly to VideoInfo
-typedef struct AVS_VideoInfo {
-  int width, height;    // width=0 means no video
-  unsigned fps_numerator, fps_denominator;
-  int num_frames;
-
-  int pixel_type;
-
-  int audio_samples_per_second;   // 0 means no audio
-  int sample_type;
-  INT64 num_audio_samples;
-  int nchannels;
-
-  // Imagetype properties
-
-  int image_type;
-} AVS_VideoInfo;
-
-// useful functions of the above
-AVSC_INLINE int avs_has_video(const AVS_VideoInfo * p)
-        { return (p->width!=0); }
-
-AVSC_INLINE int avs_has_audio(const AVS_VideoInfo * p)
-        { return (p->audio_samples_per_second!=0); }
-
-AVSC_INLINE int avs_is_rgb(const AVS_VideoInfo * p)
-        { return !!(p->pixel_type&AVS_CS_BGR); }
-
-AVSC_INLINE int avs_is_rgb24(const AVS_VideoInfo * p)
-        { return (p->pixel_type&AVS_CS_BGR24)==AVS_CS_BGR24; } // Clear out additional properties
-
-AVSC_INLINE int avs_is_rgb32(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_BGR32) == AVS_CS_BGR32 ; }
-
-AVSC_INLINE int avs_is_yuv(const AVS_VideoInfo * p)
-        { return !!(p->pixel_type&AVS_CS_YUV ); }
-
-AVSC_INLINE int avs_is_yuy2(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_YUY2) == AVS_CS_YUY2; }
-
-AVSC_INLINE int avs_is_yv24(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV24  & AVS_CS_PLANAR_FILTER); }
-
-AVSC_INLINE int avs_is_yv16(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV16  & AVS_CS_PLANAR_FILTER); }
-
-AVSC_INLINE int avs_is_yv12(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV12  & AVS_CS_PLANAR_FILTER); }
-
-AVSC_INLINE int avs_is_yv411(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV411 & AVS_CS_PLANAR_FILTER); }
-
-AVSC_INLINE int avs_is_y8(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_Y8    & AVS_CS_PLANAR_FILTER); }
-
-AVSC_INLINE int avs_is_property(const AVS_VideoInfo * p, int property)
-        { return ((p->pixel_type & property)==property ); }
-
-AVSC_INLINE int avs_is_planar(const AVS_VideoInfo * p)
-        { return !!(p->pixel_type & AVS_CS_PLANAR); }
-
-AVSC_INLINE int avs_is_color_space(const AVS_VideoInfo * p, int c_space)
-        { return avs_is_planar(p) ? ((p->pixel_type & AVS_CS_PLANAR_MASK) == (c_space & AVS_CS_PLANAR_FILTER)) : ((p->pixel_type & c_space) == c_space); }
-
-AVSC_INLINE int avs_is_field_based(const AVS_VideoInfo * p)
-        { return !!(p->image_type & AVS_IT_FIELDBASED); }
-
-AVSC_INLINE int avs_is_parity_known(const AVS_VideoInfo * p)
-        { return ((p->image_type & AVS_IT_FIELDBASED)&&(p->image_type & (AVS_IT_BFF | AVS_IT_TFF))); }
-
-AVSC_INLINE int avs_is_bff(const AVS_VideoInfo * p)
-        { return !!(p->image_type & AVS_IT_BFF); }
-
-AVSC_INLINE int avs_is_tff(const AVS_VideoInfo * p)
-        { return !!(p->image_type & AVS_IT_TFF); }
-
-AVSC_INLINE int avs_bits_per_pixel(const AVS_VideoInfo * p)
-{
-  switch (p->pixel_type) {
-      case AVS_CS_BGR24: return 24;
-      case AVS_CS_BGR32: return 32;
-      case AVS_CS_YUY2:  return 16;
-      case AVS_CS_YV12:
-      case AVS_CS_I420:  return 12;
-      default:           return 0;
-    }
-}
-AVSC_INLINE int avs_bytes_from_pixels(const AVS_VideoInfo * p, int pixels)
-        { return pixels * (avs_bits_per_pixel(p)>>3); }   // Will work on planar images, but will return only luma planes
-
-AVSC_INLINE int avs_row_size(const AVS_VideoInfo * p)
-        { return avs_bytes_from_pixels(p,p->width); }  // Also only returns first plane on planar images
-
-AVSC_INLINE int avs_bmp_size(const AVS_VideoInfo * vi)
-        { if (avs_is_planar(vi)) {int p = vi->height * ((avs_row_size(vi)+3) & ~3); p+=p>>1; return p;  } return vi->height * ((avs_row_size(vi)+3) & ~3); }
-
-AVSC_INLINE int avs_samples_per_second(const AVS_VideoInfo * p)
-        { return p->audio_samples_per_second; }
-
-
-AVSC_INLINE int avs_bytes_per_channel_sample(const AVS_VideoInfo * p)
-{
-    switch (p->sample_type) {
-      case AVS_SAMPLE_INT8:  return sizeof(signed char);
-      case AVS_SAMPLE_INT16: return sizeof(signed short);
-      case AVS_SAMPLE_INT24: return 3;
-      case AVS_SAMPLE_INT32: return sizeof(signed int);
-      case AVS_SAMPLE_FLOAT: return sizeof(float);
-      default: return 0;
-    }
-}
-AVSC_INLINE int avs_bytes_per_audio_sample(const AVS_VideoInfo * p)
-        { return p->nchannels*avs_bytes_per_channel_sample(p);}
-
-AVSC_INLINE INT64 avs_audio_samples_from_frames(const AVS_VideoInfo * p, INT64 frames)
-        { return ((INT64)(frames) * p->audio_samples_per_second * p->fps_denominator / p->fps_numerator); }
-
-AVSC_INLINE int avs_frames_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
-        { return (int)(samples * (INT64)p->fps_numerator / (INT64)p->fps_denominator / (INT64)p->audio_samples_per_second); }
-
-AVSC_INLINE INT64 avs_audio_samples_from_bytes(const AVS_VideoInfo * p, INT64 bytes)
-        { return bytes / avs_bytes_per_audio_sample(p); }
-
-AVSC_INLINE INT64 avs_bytes_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
-        { return samples * avs_bytes_per_audio_sample(p); }
-
-AVSC_INLINE int avs_audio_channels(const AVS_VideoInfo * p)
-        { return p->nchannels; }
-
-AVSC_INLINE int avs_sample_type(const AVS_VideoInfo * p)
-        { return p->sample_type;}
-
-// useful mutator
-AVSC_INLINE void avs_set_property(AVS_VideoInfo * p, int property)
-        { p->image_type|=property; }
-
-AVSC_INLINE void avs_clear_property(AVS_VideoInfo * p, int property)
-        { p->image_type&=~property; }
-
-AVSC_INLINE void avs_set_field_based(AVS_VideoInfo * p, int isfieldbased)
-        { if (isfieldbased) p->image_type|=AVS_IT_FIELDBASED; else p->image_type&=~AVS_IT_FIELDBASED; }
-
-AVSC_INLINE void avs_set_fps(AVS_VideoInfo * p, unsigned numerator, unsigned denominator)
-{
-    unsigned x=numerator, y=denominator;
-    while (y) {   // find gcd
-      unsigned t = x%y; x = y; y = t;
-    }
-    p->fps_numerator = numerator/x;
-    p->fps_denominator = denominator/x;
-}
-
-AVSC_INLINE int avs_is_same_colorspace(AVS_VideoInfo * x, AVS_VideoInfo * y)
-{
-        return (x->pixel_type == y->pixel_type)
-                || (avs_is_yv12(x) && avs_is_yv12(y));
-}
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_VideoFrame
-//
-
-// VideoFrameBuffer holds information about a memory block which is used
-// for video data.  For efficiency, instances of this class are not deleted
-// when the refcount reaches zero; instead they're stored in a linked list
-// to be reused.  The instances are deleted when the corresponding AVS
-// file is closed.
-
-// AVS_VideoFrameBuffer is layed out identicly to VideoFrameBuffer
-// DO NOT USE THIS STRUCTURE DIRECTLY
-typedef struct AVS_VideoFrameBuffer {
-  BYTE * data;
-  int data_size;
-  // sequence_number is incremented every time the buffer is changed, so
-  // that stale views can tell they're no longer valid.
-  volatile long sequence_number;
-
-  volatile long refcount;
-} AVS_VideoFrameBuffer;
-
-// VideoFrame holds a "window" into a VideoFrameBuffer.
-
-// AVS_VideoFrame is layed out identicly to IVideoFrame
-// DO NOT USE THIS STRUCTURE DIRECTLY
-typedef struct AVS_VideoFrame {
-  volatile long refcount;
-  AVS_VideoFrameBuffer * vfb;
-  int offset, pitch, row_size, height, offsetU, offsetV, pitchUV;  // U&V offsets are from top of picture.
-  int row_sizeUV, heightUV;
-} AVS_VideoFrame;
-
-// Access functions for AVS_VideoFrame
-AVSC_INLINE int avs_get_pitch(const AVS_VideoFrame * p) {
-        return p->pitch;}
-
-AVSC_INLINE int avs_get_pitch_p(const AVS_VideoFrame * p, int plane) {
-  switch (plane) {
-  case AVS_PLANAR_U: case AVS_PLANAR_V: return p->pitchUV;}
-  return p->pitch;}
-
-AVSC_INLINE int avs_get_row_size(const AVS_VideoFrame * p) {
-        return p->row_size; }
-
-AVSC_INLINE int avs_get_row_size_p(const AVS_VideoFrame * p, int plane) {
-        int r;
-    switch (plane) {
-    case AVS_PLANAR_U: case AVS_PLANAR_V:
-                if (p->pitchUV) return p->row_sizeUV;
-                else            return 0;
-    case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED:
-                if (p->pitchUV) {
-                        r = (p->row_sizeUV+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
-                        if (r < p->pitchUV)
-                                return r;
-                        return p->row_sizeUV;
-                } else return 0;
-    case AVS_PLANAR_Y_ALIGNED:
-                r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
-                if (r <= p->pitch)
-                        return r;
-                return p->row_size;
-    }
-    return p->row_size;
-}
-
-AVSC_INLINE int avs_get_height(const AVS_VideoFrame * p) {
-        return p->height;}
-
-AVSC_INLINE int avs_get_height_p(const AVS_VideoFrame * p, int plane) {
-        switch (plane) {
-                case AVS_PLANAR_U: case AVS_PLANAR_V:
-                        if (p->pitchUV) return p->heightUV;
-                        return 0;
-        }
-        return p->height;}
-
-AVSC_INLINE const BYTE* avs_get_read_ptr(const AVS_VideoFrame * p) {
-        return p->vfb->data + p->offset;}
-
-AVSC_INLINE const BYTE* avs_get_read_ptr_p(const AVS_VideoFrame * p, int plane)
-{
-        switch (plane) {
-                case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
-                case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
-                default:           return p->vfb->data + p->offset;}
-}
-
-AVSC_INLINE int avs_is_writable(const AVS_VideoFrame * p) {
-        return (p->refcount == 1 && p->vfb->refcount == 1);}
-
-AVSC_INLINE BYTE* avs_get_write_ptr(const AVS_VideoFrame * p)
-{
-        if (avs_is_writable(p)) {
-                ++p->vfb->sequence_number;
-                return p->vfb->data + p->offset;
-        } else
-                return 0;
-}
-
-AVSC_INLINE BYTE* avs_get_write_ptr_p(const AVS_VideoFrame * p, int plane)
-{
-        if (plane==AVS_PLANAR_Y && avs_is_writable(p)) {
-                ++p->vfb->sequence_number;
-                return p->vfb->data + p->offset;
-        } else if (plane==AVS_PLANAR_Y) {
-                return 0;
-        } else {
-                switch (plane) {
-                        case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
-                        case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
-                        default:       return p->vfb->data + p->offset;
-                }
-        }
-}
-
-
-AVSC_API(void, avs_release_video_frame)(AVS_VideoFrame *);
-// makes a shallow copy of a video frame
-AVSC_API(AVS_VideoFrame *, avs_copy_video_frame)(AVS_VideoFrame *);
-
-#ifndef AVSC_NO_DECLSPEC
-AVSC_INLINE void avs_release_frame(AVS_VideoFrame * f)
-  {avs_release_video_frame(f);}
-AVSC_INLINE AVS_VideoFrame * avs_copy_frame(AVS_VideoFrame * f)
-  {return avs_copy_video_frame(f);}
-#endif
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_Value
-//
-
-// Treat AVS_Value as a fat pointer.  That is use avs_copy_value
-// and avs_release_value appropiaty as you would if AVS_Value was
-// a pointer.
-
-// To maintain source code compatibility with future versions of the
-// avisynth_c API don't use the AVS_Value directly.  Use the helper
-// functions below.
-
-// AVS_Value is layed out identicly to AVSValue
-typedef struct AVS_Value AVS_Value;
-struct AVS_Value {
-  short type;  // 'a'rray, 'c'lip, 'b'ool, 'i'nt, 'f'loat, 's'tring, 'v'oid, or 'l'ong
-               // for some function e'rror
-  short array_size;
-  union {
-    void * clip; // do not use directly, use avs_take_clip
-    char boolean;
-    int integer;
-    float floating_pt;
-    const char * string;
-    const AVS_Value * array;
-  } d;
-};
-
-// AVS_Value should be initilized with avs_void.
-// Should also set to avs_void after the value is released
-// with avs_copy_value.  Consider it the equalvent of setting
-// a pointer to NULL
-static const AVS_Value avs_void = {'v'};
-
-AVSC_API(void, avs_copy_value)(AVS_Value * dest, AVS_Value src);
-AVSC_API(void, avs_release_value)(AVS_Value);
-
-AVSC_INLINE int avs_defined(AVS_Value v) { return v.type != 'v'; }
-AVSC_INLINE int avs_is_clip(AVS_Value v) { return v.type == 'c'; }
-AVSC_INLINE int avs_is_bool(AVS_Value v) { return v.type == 'b'; }
-AVSC_INLINE int avs_is_int(AVS_Value v) { return v.type == 'i'; }
-AVSC_INLINE int avs_is_float(AVS_Value v) { return v.type == 'f' || v.type == 'i'; }
-AVSC_INLINE int avs_is_string(AVS_Value v) { return v.type == 's'; }
-AVSC_INLINE int avs_is_array(AVS_Value v) { return v.type == 'a'; }
-AVSC_INLINE int avs_is_error(AVS_Value v) { return v.type == 'e'; }
-
-AVSC_API(AVS_Clip *, avs_take_clip)(AVS_Value, AVS_ScriptEnvironment *);
-AVSC_API(void, avs_set_to_clip)(AVS_Value *, AVS_Clip *);
-
-AVSC_INLINE int avs_as_bool(AVS_Value v)
-        { return v.d.boolean; }
-AVSC_INLINE int avs_as_int(AVS_Value v)
-        { return v.d.integer; }
-AVSC_INLINE const char * avs_as_string(AVS_Value v)
-        { return avs_is_error(v) || avs_is_string(v) ? v.d.string : 0; }
-AVSC_INLINE double avs_as_float(AVS_Value v)
-        { return avs_is_int(v) ? v.d.integer : v.d.floating_pt; }
-AVSC_INLINE const char * avs_as_error(AVS_Value v)
-        { return avs_is_error(v) ? v.d.string : 0; }
-AVSC_INLINE const AVS_Value * avs_as_array(AVS_Value v)
-        { return v.d.array; }
-AVSC_INLINE int avs_array_size(AVS_Value v)
-        { return avs_is_array(v) ? v.array_size : 1; }
-AVSC_INLINE AVS_Value avs_array_elt(AVS_Value v, int index)
-        { return avs_is_array(v) ? v.d.array[index] : v; }
-
-// only use these functions on an AVS_Value that does not already have
-// an active value.  Remember, treat AVS_Value as a fat pointer.
-AVSC_INLINE AVS_Value avs_new_value_bool(int v0)
-        { AVS_Value v; v.type = 'b'; v.d.boolean = v0 == 0 ? 0 : 1; return v; }
-AVSC_INLINE AVS_Value avs_new_value_int(int v0)
-        { AVS_Value v; v.type = 'i'; v.d.integer = v0; return v; }
-AVSC_INLINE AVS_Value avs_new_value_string(const char * v0)
-        { AVS_Value v; v.type = 's'; v.d.string = v0; return v; }
-AVSC_INLINE AVS_Value avs_new_value_float(float v0)
-        { AVS_Value v; v.type = 'f'; v.d.floating_pt = v0; return v;}
-AVSC_INLINE AVS_Value avs_new_value_error(const char * v0)
-        { AVS_Value v; v.type = 'e'; v.d.string = v0; return v; }
-#ifndef AVSC_NO_DECLSPEC
-AVSC_INLINE AVS_Value avs_new_value_clip(AVS_Clip * v0)
-        { AVS_Value v; avs_set_to_clip(&v, v0); return v; }
-#endif
-AVSC_INLINE AVS_Value avs_new_value_array(AVS_Value * v0, int size)
-        { AVS_Value v; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; }
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_Clip
-//
-
-AVSC_API(void, avs_release_clip)(AVS_Clip *);
-AVSC_API(AVS_Clip *, avs_copy_clip)(AVS_Clip *);
-
-AVSC_API(const char *, avs_clip_get_error)(AVS_Clip *); // return 0 if no error
-
-AVSC_API(const AVS_VideoInfo *, avs_get_video_info)(AVS_Clip *);
-
-AVSC_API(int, avs_get_version)(AVS_Clip *);
-
-AVSC_API(AVS_VideoFrame *, avs_get_frame)(AVS_Clip *, int n);
-// The returned video frame must be released with avs_release_video_frame
-
-AVSC_API(int, avs_get_parity)(AVS_Clip *, int n);
-// return field parity if field_based, else parity of first field in frame
-
-AVSC_API(int, avs_get_audio)(AVS_Clip *, void * buf,
-                             INT64 start, INT64 count);
-// start and count are in samples
-
-AVSC_API(int, avs_set_cache_hints)(AVS_Clip *,
-                                   int cachehints, int frame_range);
-
-// This is the callback type used by avs_add_function
-typedef AVS_Value (AVSC_CC * AVS_ApplyFunc)
-                        (AVS_ScriptEnvironment *, AVS_Value args, void * user_data);
-
-typedef struct AVS_FilterInfo AVS_FilterInfo;
-struct AVS_FilterInfo
-{
-  // these members should not be modified outside of the AVS_ApplyFunc callback
-  AVS_Clip * child;
-  AVS_VideoInfo vi;
-  AVS_ScriptEnvironment * env;
-  AVS_VideoFrame * (AVSC_CC * get_frame)(AVS_FilterInfo *, int n);
-  int (AVSC_CC * get_parity)(AVS_FilterInfo *, int n);
-  int (AVSC_CC * get_audio)(AVS_FilterInfo *, void * buf,
-                                  INT64 start, INT64 count);
-  int (AVSC_CC * set_cache_hints)(AVS_FilterInfo *, int cachehints,
-                                        int frame_range);
-  void (AVSC_CC * free_filter)(AVS_FilterInfo *);
-
-  // Should be set when ever there is an error to report.
-  // It is cleared before any of the above methods are called
-  const char * error;
-  // this is to store whatever and may be modified at will
-  void * user_data;
-};
-
-// Create a new filter
-// fi is set to point to the AVS_FilterInfo so that you can
-//   modify it once it is initilized.
-// store_child should generally be set to true.  If it is not
-//    set than ALL methods (the function pointers) must be defined
-// If it is set than you do not need to worry about freeing the child
-//    clip.
-AVSC_API(AVS_Clip *, avs_new_c_filter)(AVS_ScriptEnvironment * e,
-                                       AVS_FilterInfo * * fi,
-                                       AVS_Value child, int store_child);
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_ScriptEnvironment
-//
-
-// For GetCPUFlags.  These are backwards-compatible with those in VirtualDub.
-enum {
-                                /* slowest CPU to support extension */
-  AVS_CPU_FORCE        = 0x01,   // N/A
-  AVS_CPU_FPU          = 0x02,   // 386/486DX
-  AVS_CPU_MMX          = 0x04,   // P55C, K6, PII
-  AVS_CPU_INTEGER_SSE  = 0x08,   // PIII, Athlon
-  AVS_CPU_SSE          = 0x10,   // PIII, Athlon XP/MP
-  AVS_CPU_SSE2         = 0x20,   // PIV, Hammer
-  AVS_CPU_3DNOW        = 0x40,   // K6-2
-  AVS_CPU_3DNOW_EXT    = 0x80,   // Athlon
-  AVS_CPU_X86_64       = 0xA0,   // Hammer (note: equiv. to 3DNow + SSE2,
-                                 // which only Hammer will have anyway)
-  AVS_CPUF_SSE3       = 0x100,   //  PIV+, K8 Venice
-  AVS_CPUF_SSSE3      = 0x200,   //  Core 2
-  AVS_CPUF_SSE4       = 0x400,   //  Penryn, Wolfdale, Yorkfield
-  AVS_CPUF_SSE4_1     = 0x400,
-  AVS_CPUF_SSE4_2     = 0x800,   //  Nehalem
-};
-
-AVSC_API(const char *, avs_get_error)(AVS_ScriptEnvironment *); // return 0 if no error
-
-AVSC_API(long, avs_get_cpu_flags)(AVS_ScriptEnvironment *);
-AVSC_API(int, avs_check_version)(AVS_ScriptEnvironment *, int version);
-
-AVSC_API(char *, avs_save_string)(AVS_ScriptEnvironment *, const char* s, int length);
-AVSC_API(char *, avs_sprintf)(AVS_ScriptEnvironment *, const char * fmt, ...);
-
-AVSC_API(char *, avs_vsprintf)(AVS_ScriptEnvironment *, const char * fmt, void* val);
- // note: val is really a va_list; I hope everyone typedefs va_list to a pointer
-
-AVSC_API(int, avs_add_function)(AVS_ScriptEnvironment *,
-                                const char * name, const char * params,
-                                AVS_ApplyFunc apply, void * user_data);
-
-AVSC_API(int, avs_function_exists)(AVS_ScriptEnvironment *, const char * name);
-
-AVSC_API(AVS_Value, avs_invoke)(AVS_ScriptEnvironment *, const char * name,
-                               AVS_Value args, const char** arg_names);
-// The returned value must be be released with avs_release_value
-
-AVSC_API(AVS_Value, avs_get_var)(AVS_ScriptEnvironment *, const char* name);
-// The returned value must be be released with avs_release_value
-
-AVSC_API(int, avs_set_var)(AVS_ScriptEnvironment *, const char* name, AVS_Value val);
-
-AVSC_API(int, avs_set_global_var)(AVS_ScriptEnvironment *, const char* name, const AVS_Value val);
-
-//void avs_push_context(AVS_ScriptEnvironment *, int level=0);
-//void avs_pop_context(AVS_ScriptEnvironment *);
-
-AVSC_API(AVS_VideoFrame *, avs_new_video_frame_a)(AVS_ScriptEnvironment *,
-                                          const AVS_VideoInfo * vi, int align);
-// align should be at least 16
-
-#ifndef AVSC_NO_DECLSPEC
-AVSC_INLINE
-AVS_VideoFrame * avs_new_video_frame(AVS_ScriptEnvironment * env,
-                                     const AVS_VideoInfo * vi)
-  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
-
-AVSC_INLINE
-AVS_VideoFrame * avs_new_frame(AVS_ScriptEnvironment * env,
-                               const AVS_VideoInfo * vi)
-  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
-#endif
-
-
-AVSC_API(int, avs_make_writable)(AVS_ScriptEnvironment *, AVS_VideoFrame * * pvf);
-
-AVSC_API(void, avs_bit_blt)(AVS_ScriptEnvironment *, BYTE* dstp, int dst_pitch, const BYTE* srcp, int src_pitch, int row_size, int height);
-
-typedef void (AVSC_CC *AVS_ShutdownFunc)(void* user_data, AVS_ScriptEnvironment * env);
-AVSC_API(void, avs_at_exit)(AVS_ScriptEnvironment *, AVS_ShutdownFunc function, void * user_data);
-
-AVSC_API(AVS_VideoFrame *, avs_subframe)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height);
-// The returned video frame must be be released
-
-AVSC_API(int, avs_set_memory_max)(AVS_ScriptEnvironment *, int mem);
-
-AVSC_API(int, avs_set_working_dir)(AVS_ScriptEnvironment *, const char * newdir);
-
-// avisynth.dll exports this; it's a way to use it as a library, without
-// writing an AVS script or without going through AVIFile.
-AVSC_API(AVS_ScriptEnvironment *, avs_create_script_environment)(int version);
-
-// this symbol is the entry point for the plugin and must
-// be defined
-AVSC_EXPORT
-const char * AVSC_CC avisynth_c_plugin_init(AVS_ScriptEnvironment* env);
-
-
-AVSC_API(void, avs_delete_script_environment)(AVS_ScriptEnvironment *);
-
-
-AVSC_API(AVS_VideoFrame *, avs_subframe_planar)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height, int rel_offsetU, int rel_offsetV, int new_pitchUV);
-// The returned video frame must be be released
-
-#ifdef AVSC_NO_DECLSPEC
-// use LoadLibrary and related functions to dynamically load Avisynth instead of declspec(dllimport)
-/*
-  The following functions needs to have been declared, probably from windows.h
-
-  void* malloc(size_t)
-  void free(void*);
-
-  HMODULE LoadLibrary(const char*);
-  void* GetProcAddress(HMODULE, const char*);
-  FreeLibrary(HMODULE);
-*/
-
-
-typedef struct AVS_Library AVS_Library;
-
-#define AVSC_DECLARE_FUNC(name) name##_func name
-
-struct AVS_Library {
-  HMODULE handle;
-
-  AVSC_DECLARE_FUNC(avs_add_function);
-  AVSC_DECLARE_FUNC(avs_at_exit);
-  AVSC_DECLARE_FUNC(avs_bit_blt);
-  AVSC_DECLARE_FUNC(avs_check_version);
-  AVSC_DECLARE_FUNC(avs_clip_get_error);
-  AVSC_DECLARE_FUNC(avs_copy_clip);
-  AVSC_DECLARE_FUNC(avs_copy_value);
-  AVSC_DECLARE_FUNC(avs_copy_video_frame);
-  AVSC_DECLARE_FUNC(avs_create_script_environment);
-  AVSC_DECLARE_FUNC(avs_delete_script_environment);
-  AVSC_DECLARE_FUNC(avs_function_exists);
-  AVSC_DECLARE_FUNC(avs_get_audio);
-  AVSC_DECLARE_FUNC(avs_get_cpu_flags);
-  AVSC_DECLARE_FUNC(avs_get_error);
-  AVSC_DECLARE_FUNC(avs_get_frame);
-  AVSC_DECLARE_FUNC(avs_get_parity);
-  AVSC_DECLARE_FUNC(avs_get_var);
-  AVSC_DECLARE_FUNC(avs_get_version);
-  AVSC_DECLARE_FUNC(avs_get_video_info);
-  AVSC_DECLARE_FUNC(avs_invoke);
-  AVSC_DECLARE_FUNC(avs_make_writable);
-  AVSC_DECLARE_FUNC(avs_new_c_filter);
-  AVSC_DECLARE_FUNC(avs_new_video_frame_a);
-  AVSC_DECLARE_FUNC(avs_release_clip);
-  AVSC_DECLARE_FUNC(avs_release_value);
-  AVSC_DECLARE_FUNC(avs_release_video_frame);
-  AVSC_DECLARE_FUNC(avs_save_string);
-  AVSC_DECLARE_FUNC(avs_set_cache_hints);
-  AVSC_DECLARE_FUNC(avs_set_global_var);
-  AVSC_DECLARE_FUNC(avs_set_memory_max);
-  AVSC_DECLARE_FUNC(avs_set_to_clip);
-  AVSC_DECLARE_FUNC(avs_set_var);
-  AVSC_DECLARE_FUNC(avs_set_working_dir);
-  AVSC_DECLARE_FUNC(avs_sprintf);
-  AVSC_DECLARE_FUNC(avs_subframe);
-  AVSC_DECLARE_FUNC(avs_subframe_planar);
-  AVSC_DECLARE_FUNC(avs_take_clip);
-  AVSC_DECLARE_FUNC(avs_vsprintf);
-};
-
-#undef AVSC_DECLARE_FUNC
-
-
-AVSC_INLINE AVS_Library * avs_load_library() {
-  AVS_Library *library = (AVS_Library *)malloc(sizeof(AVS_Library));
-  if (library == NULL)
-    return NULL;
-  library->handle = LoadLibrary("avisynth");
-  if (library->handle == NULL)
-    goto fail;
-
-#define __AVSC_STRINGIFY(x) #x
-#define AVSC_STRINGIFY(x) __AVSC_STRINGIFY(x)
-#define AVSC_LOAD_FUNC(name) {\
-  library->name = (name##_func) GetProcAddress(library->handle, AVSC_STRINGIFY(name));\
-  if (library->name == NULL)\
-    goto fail;\
-}
-
-  AVSC_LOAD_FUNC(avs_add_function);
-  AVSC_LOAD_FUNC(avs_at_exit);
-  AVSC_LOAD_FUNC(avs_bit_blt);
-  AVSC_LOAD_FUNC(avs_check_version);
-  AVSC_LOAD_FUNC(avs_clip_get_error);
-  AVSC_LOAD_FUNC(avs_copy_clip);
-  AVSC_LOAD_FUNC(avs_copy_value);
-  AVSC_LOAD_FUNC(avs_copy_video_frame);
-  AVSC_LOAD_FUNC(avs_create_script_environment);
-  AVSC_LOAD_FUNC(avs_delete_script_environment);
-  AVSC_LOAD_FUNC(avs_function_exists);
-  AVSC_LOAD_FUNC(avs_get_audio);
-  AVSC_LOAD_FUNC(avs_get_cpu_flags);
-  AVSC_LOAD_FUNC(avs_get_error);
-  AVSC_LOAD_FUNC(avs_get_frame);
-  AVSC_LOAD_FUNC(avs_get_parity);
-  AVSC_LOAD_FUNC(avs_get_var);
-  AVSC_LOAD_FUNC(avs_get_version);
-  AVSC_LOAD_FUNC(avs_get_video_info);
-  AVSC_LOAD_FUNC(avs_invoke);
-  AVSC_LOAD_FUNC(avs_make_writable);
-  AVSC_LOAD_FUNC(avs_new_c_filter);
-  AVSC_LOAD_FUNC(avs_new_video_frame_a);
-  AVSC_LOAD_FUNC(avs_release_clip);
-  AVSC_LOAD_FUNC(avs_release_value);
-  AVSC_LOAD_FUNC(avs_release_video_frame);
-  AVSC_LOAD_FUNC(avs_save_string);
-  AVSC_LOAD_FUNC(avs_set_cache_hints);
-  AVSC_LOAD_FUNC(avs_set_global_var);
-  AVSC_LOAD_FUNC(avs_set_memory_max);
-  AVSC_LOAD_FUNC(avs_set_to_clip);
-  AVSC_LOAD_FUNC(avs_set_var);
-  AVSC_LOAD_FUNC(avs_set_working_dir);
-  AVSC_LOAD_FUNC(avs_sprintf);
-  AVSC_LOAD_FUNC(avs_subframe);
-  AVSC_LOAD_FUNC(avs_subframe_planar);
-  AVSC_LOAD_FUNC(avs_take_clip);
-  AVSC_LOAD_FUNC(avs_vsprintf);
-
-#undef __AVSC_STRINGIFY
-#undef AVSC_STRINGIFY
-#undef AVSC_LOAD_FUNC
-
-  return library;
-
-fail:
-  free(library);
-  return NULL;
-}
-
-AVSC_INLINE void avs_free_library(AVS_Library *library) {
-  if (library == NULL)
-    return;
-  FreeLibrary(library->handle);
-  free(library);
-}
-#endif
-
-#endif

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); }

compat/avisynth/avxsynth_c.h

@@ -1,728 +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., 51 Franklin Street, Fifth Floor, Boston,
-// MA 02110-1301 USA, or visit
-// http://www.gnu.org/copyleft/gpl.html .
-//
-// As a special exception, I give you permission to link to the
-// Avisynth C interface with independent modules that communicate with
-// the Avisynth C interface solely through the interfaces defined in
-// avisynth_c.h, regardless of the license terms of these independent
-// modules, and to copy and distribute the resulting combined work
-// under terms of your choice, provided that every copy of the
-// combined work is accompanied by a complete copy of the source code
-// of the Avisynth C interface and Avisynth itself (with the version
-// used to produce the combined work), being distributed under the
-// terms of the GNU General Public License plus this exception.  An
-// independent module is a module which is not derived from or based
-// on Avisynth C Interface, such as 3rd-party filters, import and
-// export plugins, or graphical user interfaces.
-
-#ifndef __AVXSYNTH_C__
-#define __AVXSYNTH_C__
-
-#include "windowsPorts/windows2linux.h"
-#include <stdarg.h>
-
-#ifdef __cplusplus
-#  define EXTERN_C extern "C"
-#else
-#  define EXTERN_C
-#endif
-
-#define AVSC_USE_STDCALL 1
-
-#ifndef AVSC_USE_STDCALL
-#  define AVSC_CC __cdecl
-#else
-#  define AVSC_CC __stdcall
-#endif
-
-#define AVSC_INLINE static __inline
-
-#ifdef AVISYNTH_C_EXPORTS
-#  define AVSC_EXPORT EXTERN_C
-#  define AVSC_API(ret, name) EXTERN_C __declspec(dllexport) ret AVSC_CC name
-#else
-#  define AVSC_EXPORT EXTERN_C __declspec(dllexport)
-#  ifndef AVSC_NO_DECLSPEC
-#    define AVSC_API(ret, name) EXTERN_C __declspec(dllimport) ret AVSC_CC name
-#  else
-#    define AVSC_API(ret, name) typedef ret (AVSC_CC *name##_func)
-#  endif
-#endif
-
-#ifdef __GNUC__
-typedef long long int INT64;
-#else
-typedef __int64 INT64;
-#endif
-
-
-/////////////////////////////////////////////////////////////////////
-//
-// Constants
-//
-
-#ifndef __AVXSYNTH_H__
-enum { AVISYNTH_INTERFACE_VERSION = 3 };
-#endif
-
-enum {AVS_SAMPLE_INT8  = 1<<0,
-      AVS_SAMPLE_INT16 = 1<<1,
-      AVS_SAMPLE_INT24 = 1<<2,
-      AVS_SAMPLE_INT32 = 1<<3,
-      AVS_SAMPLE_FLOAT = 1<<4};
-
-enum {AVS_PLANAR_Y=1<<0,
-      AVS_PLANAR_U=1<<1,
-      AVS_PLANAR_V=1<<2,
-      AVS_PLANAR_ALIGNED=1<<3,
-      AVS_PLANAR_Y_ALIGNED=AVS_PLANAR_Y|AVS_PLANAR_ALIGNED,
-      AVS_PLANAR_U_ALIGNED=AVS_PLANAR_U|AVS_PLANAR_ALIGNED,
-      AVS_PLANAR_V_ALIGNED=AVS_PLANAR_V|AVS_PLANAR_ALIGNED};
-
-  // Colorspace properties.
-enum {AVS_CS_BGR = 1<<28,
-      AVS_CS_YUV = 1<<29,
-      AVS_CS_INTERLEAVED = 1<<30,
-      AVS_CS_PLANAR = 1<<31};
-
-  // Specific colorformats
-enum {
-  AVS_CS_UNKNOWN = 0,
-  AVS_CS_BGR24 = 1<<0 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
-  AVS_CS_BGR32 = 1<<1 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
-  AVS_CS_YUY2 = 1<<2 | AVS_CS_YUV | AVS_CS_INTERLEAVED,
-  AVS_CS_YV12 = 1<<3 | AVS_CS_YUV | AVS_CS_PLANAR,  // y-v-u, planar
-  AVS_CS_I420 = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR,  // y-u-v, planar
-  AVS_CS_IYUV = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR  // same as above
-};
-
-enum {
-  AVS_IT_BFF = 1<<0,
-  AVS_IT_TFF = 1<<1,
-  AVS_IT_FIELDBASED = 1<<2};
-
-enum {
-  AVS_FILTER_TYPE=1,
-  AVS_FILTER_INPUT_COLORSPACE=2,
-  AVS_FILTER_OUTPUT_TYPE=9,
-  AVS_FILTER_NAME=4,
-  AVS_FILTER_AUTHOR=5,
-  AVS_FILTER_VERSION=6,
-  AVS_FILTER_ARGS=7,
-  AVS_FILTER_ARGS_INFO=8,
-  AVS_FILTER_ARGS_DESCRIPTION=10,
-  AVS_FILTER_DESCRIPTION=11};
-
-enum {  //SUBTYPES
-  AVS_FILTER_TYPE_AUDIO=1,
-  AVS_FILTER_TYPE_VIDEO=2,
-  AVS_FILTER_OUTPUT_TYPE_SAME=3,
-  AVS_FILTER_OUTPUT_TYPE_DIFFERENT=4};
-
-enum {
-  AVS_CACHE_NOTHING=0,
-  AVS_CACHE_RANGE=1,
-  AVS_CACHE_ALL=2,
-  AVS_CACHE_AUDIO=3,
-  AVS_CACHE_AUDIO_NONE=4,
-  AVS_CACHE_AUDIO_AUTO=5
-};
-
-#define AVS_FRAME_ALIGN 16
-
-typedef struct AVS_Clip AVS_Clip;
-typedef struct AVS_ScriptEnvironment AVS_ScriptEnvironment;
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_VideoInfo
-//
-
-// AVS_VideoInfo is layed out identicly to VideoInfo
-typedef struct AVS_VideoInfo {
-  int width, height;    // width=0 means no video
-  unsigned fps_numerator, fps_denominator;
-  int num_frames;
-
-  int pixel_type;
-
-  int audio_samples_per_second;   // 0 means no audio
-  int sample_type;
-  INT64 num_audio_samples;
-  int nchannels;
-
-  // Imagetype properties
-
-  int image_type;
-} AVS_VideoInfo;
-
-// useful functions of the above
-AVSC_INLINE int avs_has_video(const AVS_VideoInfo * p)
-        { return (p->width!=0); }
-
-AVSC_INLINE int avs_has_audio(const AVS_VideoInfo * p)
-        { return (p->audio_samples_per_second!=0); }
-
-AVSC_INLINE int avs_is_rgb(const AVS_VideoInfo * p)
-        { return !!(p->pixel_type&AVS_CS_BGR); }
-
-AVSC_INLINE int avs_is_rgb24(const AVS_VideoInfo * p)
-        { return (p->pixel_type&AVS_CS_BGR24)==AVS_CS_BGR24; } // Clear out additional properties
-
-AVSC_INLINE int avs_is_rgb32(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_BGR32) == AVS_CS_BGR32 ; }
-
-AVSC_INLINE int avs_is_yuv(const AVS_VideoInfo * p)
-        { return !!(p->pixel_type&AVS_CS_YUV ); }
-
-AVSC_INLINE int avs_is_yuy2(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_YUY2) == AVS_CS_YUY2; }
-
-AVSC_INLINE int avs_is_yv12(const AVS_VideoInfo * p)
-        { return ((p->pixel_type & AVS_CS_YV12) == AVS_CS_YV12)||((p->pixel_type & AVS_CS_I420) == AVS_CS_I420); }
-
-AVSC_INLINE int avs_is_color_space(const AVS_VideoInfo * p, int c_space)
-        { return ((p->pixel_type & c_space) == c_space); }
-
-AVSC_INLINE int avs_is_property(const AVS_VideoInfo * p, int property)
-        { return ((p->pixel_type & property)==property ); }
-
-AVSC_INLINE int avs_is_planar(const AVS_VideoInfo * p)
-        { return !!(p->pixel_type & AVS_CS_PLANAR); }
-
-AVSC_INLINE int avs_is_field_based(const AVS_VideoInfo * p)
-        { return !!(p->image_type & AVS_IT_FIELDBASED); }
-
-AVSC_INLINE int avs_is_parity_known(const AVS_VideoInfo * p)
-        { return ((p->image_type & AVS_IT_FIELDBASED)&&(p->image_type & (AVS_IT_BFF | AVS_IT_TFF))); }
-
-AVSC_INLINE int avs_is_bff(const AVS_VideoInfo * p)
-        { return !!(p->image_type & AVS_IT_BFF); }
-
-AVSC_INLINE int avs_is_tff(const AVS_VideoInfo * p)
-        { return !!(p->image_type & AVS_IT_TFF); }
-
-AVSC_INLINE int avs_bits_per_pixel(const AVS_VideoInfo * p)
-{
-  switch (p->pixel_type) {
-      case AVS_CS_BGR24: return 24;
-      case AVS_CS_BGR32: return 32;
-      case AVS_CS_YUY2:  return 16;
-      case AVS_CS_YV12:
-      case AVS_CS_I420:  return 12;
-      default:           return 0;
-    }
-}
-AVSC_INLINE int avs_bytes_from_pixels(const AVS_VideoInfo * p, int pixels)
-        { return pixels * (avs_bits_per_pixel(p)>>3); }   // Will work on planar images, but will return only luma planes
-
-AVSC_INLINE int avs_row_size(const AVS_VideoInfo * p)
-        { return avs_bytes_from_pixels(p,p->width); }  // Also only returns first plane on planar images
-
-AVSC_INLINE int avs_bmp_size(const AVS_VideoInfo * vi)
-        { if (avs_is_planar(vi)) {int p = vi->height * ((avs_row_size(vi)+3) & ~3); p+=p>>1; return p;  } return vi->height * ((avs_row_size(vi)+3) & ~3); }
-
-AVSC_INLINE int avs_samples_per_second(const AVS_VideoInfo * p)
-        { return p->audio_samples_per_second; }
-
-
-AVSC_INLINE int avs_bytes_per_channel_sample(const AVS_VideoInfo * p)
-{
-    switch (p->sample_type) {
-      case AVS_SAMPLE_INT8:  return sizeof(signed char);
-      case AVS_SAMPLE_INT16: return sizeof(signed short);
-      case AVS_SAMPLE_INT24: return 3;
-      case AVS_SAMPLE_INT32: return sizeof(signed int);
-      case AVS_SAMPLE_FLOAT: return sizeof(float);
-      default: return 0;
-    }
-}
-AVSC_INLINE int avs_bytes_per_audio_sample(const AVS_VideoInfo * p)
-        { return p->nchannels*avs_bytes_per_channel_sample(p);}
-
-AVSC_INLINE INT64 avs_audio_samples_from_frames(const AVS_VideoInfo * p, INT64 frames)
-        { return ((INT64)(frames) * p->audio_samples_per_second * p->fps_denominator / p->fps_numerator); }
-
-AVSC_INLINE int avs_frames_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
-        { return (int)(samples * (INT64)p->fps_numerator / (INT64)p->fps_denominator / (INT64)p->audio_samples_per_second); }
-
-AVSC_INLINE INT64 avs_audio_samples_from_bytes(const AVS_VideoInfo * p, INT64 bytes)
-        { return bytes / avs_bytes_per_audio_sample(p); }
-
-AVSC_INLINE INT64 avs_bytes_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
-        { return samples * avs_bytes_per_audio_sample(p); }
-
-AVSC_INLINE int avs_audio_channels(const AVS_VideoInfo * p)
-        { return p->nchannels; }
-
-AVSC_INLINE int avs_sample_type(const AVS_VideoInfo * p)
-        { return p->sample_type;}
-
-// useful mutator
-AVSC_INLINE void avs_set_property(AVS_VideoInfo * p, int property)
-        { p->image_type|=property; }
-
-AVSC_INLINE void avs_clear_property(AVS_VideoInfo * p, int property)
-        { p->image_type&=~property; }
-
-AVSC_INLINE void avs_set_field_based(AVS_VideoInfo * p, int isfieldbased)
-        { if (isfieldbased) p->image_type|=AVS_IT_FIELDBASED; else p->image_type&=~AVS_IT_FIELDBASED; }
-
-AVSC_INLINE void avs_set_fps(AVS_VideoInfo * p, unsigned numerator, unsigned denominator)
-{
-    unsigned x=numerator, y=denominator;
-    while (y) {   // find gcd
-      unsigned t = x%y; x = y; y = t;
-    }
-    p->fps_numerator = numerator/x;
-    p->fps_denominator = denominator/x;
-}
-
-AVSC_INLINE int avs_is_same_colorspace(AVS_VideoInfo * x, AVS_VideoInfo * y)
-{
-        return (x->pixel_type == y->pixel_type)
-                || (avs_is_yv12(x) && avs_is_yv12(y));
-}
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_VideoFrame
-//
-
-// VideoFrameBuffer holds information about a memory block which is used
-// for video data.  For efficiency, instances of this class are not deleted
-// when the refcount reaches zero; instead they're stored in a linked list
-// to be reused.  The instances are deleted when the corresponding AVS
-// file is closed.
-
-// AVS_VideoFrameBuffer is layed out identicly to VideoFrameBuffer
-// DO NOT USE THIS STRUCTURE DIRECTLY
-typedef struct AVS_VideoFrameBuffer {
-  unsigned char * data;
-  int data_size;
-  // sequence_number is incremented every time the buffer is changed, so
-  // that stale views can tell they're no longer valid.
-  long sequence_number;
-
-  long refcount;
-} AVS_VideoFrameBuffer;
-
-// VideoFrame holds a "window" into a VideoFrameBuffer.
-
-// AVS_VideoFrame is layed out identicly to IVideoFrame
-// DO NOT USE THIS STRUCTURE DIRECTLY
-typedef struct AVS_VideoFrame {
-  int refcount;
-  AVS_VideoFrameBuffer * vfb;
-  int offset, pitch, row_size, height, offsetU, offsetV, pitchUV;  // U&V offsets are from top of picture.
-} AVS_VideoFrame;
-
-// Access functions for AVS_VideoFrame
-AVSC_INLINE int avs_get_pitch(const AVS_VideoFrame * p) {
-        return p->pitch;}
-
-AVSC_INLINE int avs_get_pitch_p(const AVS_VideoFrame * p, int plane) {
-  switch (plane) {
-  case AVS_PLANAR_U: case AVS_PLANAR_V: return p->pitchUV;}
-  return p->pitch;}
-
-AVSC_INLINE int avs_get_row_size(const AVS_VideoFrame * p) {
-        return p->row_size; }
-
-AVSC_INLINE int avs_get_row_size_p(const AVS_VideoFrame * p, int plane) {
-        int r;
-    switch (plane) {
-    case AVS_PLANAR_U: case AVS_PLANAR_V:
-                if (p->pitchUV) return p->row_size>>1;
-                else            return 0;
-    case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED:
-                if (p->pitchUV) {
-                        r = ((p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)) )>>1; // Aligned rowsize
-                        if (r < p->pitchUV)
-                                return r;
-                        return p->row_size>>1;
-                } else return 0;
-    case AVS_PLANAR_Y_ALIGNED:
-                r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
-                if (r <= p->pitch)
-                        return r;
-                return p->row_size;
-    }
-    return p->row_size;
-}
-
-AVSC_INLINE int avs_get_height(const AVS_VideoFrame * p) {
-        return p->height;}
-
-AVSC_INLINE int avs_get_height_p(const AVS_VideoFrame * p, int plane) {
-        switch (plane) {
-                case AVS_PLANAR_U: case AVS_PLANAR_V:
-                        if (p->pitchUV) return p->height>>1;
-                        return 0;
-        }
-        return p->height;}
-
-AVSC_INLINE const unsigned char* avs_get_read_ptr(const AVS_VideoFrame * p) {
-        return p->vfb->data + p->offset;}
-
-AVSC_INLINE const unsigned char* avs_get_read_ptr_p(const AVS_VideoFrame * p, int plane)
-{
-        switch (plane) {
-                case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
-                case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
-                default:           return p->vfb->data + p->offset;}
-}
-
-AVSC_INLINE int avs_is_writable(const AVS_VideoFrame * p) {
-        return (p->refcount == 1 && p->vfb->refcount == 1);}
-
-AVSC_INLINE unsigned char* avs_get_write_ptr(const AVS_VideoFrame * p)
-{
-        if (avs_is_writable(p)) {
-                ++p->vfb->sequence_number;
-                return p->vfb->data + p->offset;
-        } else
-                return 0;
-}
-
-AVSC_INLINE unsigned char* avs_get_write_ptr_p(const AVS_VideoFrame * p, int plane)
-{
-        if (plane==AVS_PLANAR_Y && avs_is_writable(p)) {
-                ++p->vfb->sequence_number;
-                return p->vfb->data + p->offset;
-        } else if (plane==AVS_PLANAR_Y) {
-                return 0;
-        } else {
-                switch (plane) {
-                        case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
-                        case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
-                        default:       return p->vfb->data + p->offset;
-                }
-        }
-}
-
-#if defined __cplusplus
-extern "C"
-{
-#endif // __cplusplus
-AVSC_API(void, avs_release_video_frame)(AVS_VideoFrame *);
-// makes a shallow copy of a video frame
-AVSC_API(AVS_VideoFrame *, avs_copy_video_frame)(AVS_VideoFrame *);
-#if defined __cplusplus
-}
-#endif // __cplusplus
-
-#ifndef AVSC_NO_DECLSPEC
-AVSC_INLINE void avs_release_frame(AVS_VideoFrame * f)
-  {avs_release_video_frame(f);}
-AVSC_INLINE AVS_VideoFrame * avs_copy_frame(AVS_VideoFrame * f)
-  {return avs_copy_video_frame(f);}
-#endif
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_Value
-//
-
-// Treat AVS_Value as a fat pointer.  That is use avs_copy_value
-// and avs_release_value appropiaty as you would if AVS_Value was
-// a pointer.
-
-// To maintain source code compatibility with future versions of the
-// avisynth_c API don't use the AVS_Value directly.  Use the helper
-// functions below.
-
-// AVS_Value is layed out identicly to AVSValue
-typedef struct AVS_Value AVS_Value;
-struct AVS_Value {
-  short type;  // 'a'rray, 'c'lip, 'b'ool, 'i'nt, 'f'loat, 's'tring, 'v'oid, or 'l'ong
-               // for some function e'rror
-  short array_size;
-  union {
-    void * clip; // do not use directly, use avs_take_clip
-    char boolean;
-    int integer;
-    INT64 integer64; // match addition of __int64 to avxplugin.h
-    float floating_pt;
-    const char * string;
-    const AVS_Value * array;
-  } d;
-};
-
-// AVS_Value should be initilized with avs_void.
-// Should also set to avs_void after the value is released
-// with avs_copy_value.  Consider it the equalvent of setting
-// a pointer to NULL
-static const AVS_Value avs_void = {'v'};
-
-AVSC_API(void, avs_copy_value)(AVS_Value * dest, AVS_Value src);
-AVSC_API(void, avs_release_value)(AVS_Value);
-
-AVSC_INLINE int avs_defined(AVS_Value v) { return v.type != 'v'; }
-AVSC_INLINE int avs_is_clip(AVS_Value v) { return v.type == 'c'; }
-AVSC_INLINE int avs_is_bool(AVS_Value v) { return v.type == 'b'; }
-AVSC_INLINE int avs_is_int(AVS_Value v) { return v.type == 'i'; }
-AVSC_INLINE int avs_is_float(AVS_Value v) { return v.type == 'f' || v.type == 'i'; }
-AVSC_INLINE int avs_is_string(AVS_Value v) { return v.type == 's'; }
-AVSC_INLINE int avs_is_array(AVS_Value v) { return v.type == 'a'; }
-AVSC_INLINE int avs_is_error(AVS_Value v) { return v.type == 'e'; }
-
-#if defined __cplusplus
-extern "C"
-{
-#endif // __cplusplus
-AVSC_API(AVS_Clip *, avs_take_clip)(AVS_Value, AVS_ScriptEnvironment *);
-AVSC_API(void, avs_set_to_clip)(AVS_Value *, AVS_Clip *);
-#if defined __cplusplus
-}
-#endif // __cplusplus
-
-AVSC_INLINE int avs_as_bool(AVS_Value v)
-        { return v.d.boolean; }
-AVSC_INLINE int avs_as_int(AVS_Value v)
-        { return v.d.integer; }
-AVSC_INLINE const char * avs_as_string(AVS_Value v)
-        { return avs_is_error(v) || avs_is_string(v) ? v.d.string : 0; }
-AVSC_INLINE double avs_as_float(AVS_Value v)
-        { return avs_is_int(v) ? v.d.integer : v.d.floating_pt; }
-AVSC_INLINE const char * avs_as_error(AVS_Value v)
-        { return avs_is_error(v) ? v.d.string : 0; }
-AVSC_INLINE const AVS_Value * avs_as_array(AVS_Value v)
-        { return v.d.array; }
-AVSC_INLINE int avs_array_size(AVS_Value v)
-        { return avs_is_array(v) ? v.array_size : 1; }
-AVSC_INLINE AVS_Value avs_array_elt(AVS_Value v, int index)
-        { return avs_is_array(v) ? v.d.array[index] : v; }
-
-// only use these functions on am AVS_Value that does not already have
-// an active value.  Remember, treat AVS_Value as a fat pointer.
-AVSC_INLINE AVS_Value avs_new_value_bool(int v0)
-        { AVS_Value v; v.type = 'b'; v.d.boolean = v0 == 0 ? 0 : 1; return v; }
-AVSC_INLINE AVS_Value avs_new_value_int(int v0)
-        { AVS_Value v; v.type = 'i'; v.d.integer = v0; return v; }
-AVSC_INLINE AVS_Value avs_new_value_string(const char * v0)
-        { AVS_Value v; v.type = 's'; v.d.string = v0; return v; }
-AVSC_INLINE AVS_Value avs_new_value_float(float v0)
-        { AVS_Value v; v.type = 'f'; v.d.floating_pt = v0; return v;}
-AVSC_INLINE AVS_Value avs_new_value_error(const char * v0)
-        { AVS_Value v; v.type = 'e'; v.d.string = v0; return v; }
-#ifndef AVSC_NO_DECLSPEC
-AVSC_INLINE AVS_Value avs_new_value_clip(AVS_Clip * v0)
-        { AVS_Value v; avs_set_to_clip(&v, v0); return v; }
-#endif
-AVSC_INLINE AVS_Value avs_new_value_array(AVS_Value * v0, int size)
-        { AVS_Value v; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; }
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_Clip
-//
-#if defined __cplusplus
-extern "C"
-{
-#endif // __cplusplus
-AVSC_API(void, avs_release_clip)(AVS_Clip *);
-AVSC_API(AVS_Clip *, avs_copy_clip)(AVS_Clip *);
-
-AVSC_API(const char *, avs_clip_get_error)(AVS_Clip *); // return 0 if no error
-
-AVSC_API(const AVS_VideoInfo *, avs_get_video_info)(AVS_Clip *);
-
-AVSC_API(int, avs_get_version)(AVS_Clip *);
-
-AVSC_API(AVS_VideoFrame *, avs_get_frame)(AVS_Clip *, int n);
-// The returned video frame must be released with avs_release_video_frame
-
-AVSC_API(int, avs_get_parity)(AVS_Clip *, int n);
-// return field parity if field_based, else parity of first field in frame
-
-AVSC_API(int, avs_get_audio)(AVS_Clip *, void * buf,
-                                  INT64 start, INT64 count);
-// start and count are in samples
-
-AVSC_API(int, avs_set_cache_hints)(AVS_Clip *,
-                                        int cachehints, size_t frame_range);
-#if defined __cplusplus
-}
-#endif // __cplusplus
-
-// This is the callback type used by avs_add_function
-typedef AVS_Value (AVSC_CC * AVS_ApplyFunc)
-                        (AVS_ScriptEnvironment *, AVS_Value args, void * user_data);
-
-typedef struct AVS_FilterInfo AVS_FilterInfo;
-struct AVS_FilterInfo
-{
-  // these members should not be modified outside of the AVS_ApplyFunc callback
-  AVS_Clip * child;
-  AVS_VideoInfo vi;
-  AVS_ScriptEnvironment * env;
-  AVS_VideoFrame * (AVSC_CC * get_frame)(AVS_FilterInfo *, int n);
-  int (AVSC_CC * get_parity)(AVS_FilterInfo *, int n);
-  int (AVSC_CC * get_audio)(AVS_FilterInfo *, void * buf,
-                                  INT64 start, INT64 count);
-  int (AVSC_CC * set_cache_hints)(AVS_FilterInfo *, int cachehints,
-                                        int frame_range);
-  void (AVSC_CC * free_filter)(AVS_FilterInfo *);
-
-  // Should be set when ever there is an error to report.
-  // It is cleared before any of the above methods are called
-  const char * error;
-  // this is to store whatever and may be modified at will
-  void * user_data;
-};
-
-// Create a new filter
-// fi is set to point to the AVS_FilterInfo so that you can
-//   modify it once it is initilized.
-// store_child should generally be set to true.  If it is not
-//    set than ALL methods (the function pointers) must be defined
-// If it is set than you do not need to worry about freeing the child
-//    clip.
-#if defined __cplusplus
-extern "C"
-{
-#endif // __cplusplus
-AVSC_API(AVS_Clip *, avs_new_c_filter)(AVS_ScriptEnvironment * e,
-                                      AVS_FilterInfo * * fi,
-                                      AVS_Value child, int store_child);
-#if defined __cplusplus
-}
-#endif // __cplusplus
-
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_ScriptEnvironment
-//
-
-// For GetCPUFlags.  These are backwards-compatible with those in VirtualDub.
-enum {
-                                /* slowest CPU to support extension */
-  AVS_CPU_FORCE        = 0x01,   // N/A
-  AVS_CPU_FPU          = 0x02,   // 386/486DX
-  AVS_CPU_MMX          = 0x04,   // P55C, K6, PII
-  AVS_CPU_INTEGER_SSE  = 0x08,   // PIII, Athlon
-  AVS_CPU_SSE          = 0x10,   // PIII, Athlon XP/MP
-  AVS_CPU_SSE2         = 0x20,   // PIV, Hammer
-  AVS_CPU_3DNOW        = 0x40,   // K6-2
-  AVS_CPU_3DNOW_EXT    = 0x80,   // Athlon
-  AVS_CPU_X86_64       = 0xA0,   // Hammer (note: equiv. to 3DNow + SSE2,
-                                 // which only Hammer will have anyway)
-};
-
-#if defined __cplusplus
-extern "C"
-{
-#endif // __cplusplus
-AVSC_API(const char *, avs_get_error)(AVS_ScriptEnvironment *); // return 0 if no error
-
-AVSC_API(long, avs_get_cpu_flags)(AVS_ScriptEnvironment *);
-AVSC_API(int, avs_check_version)(AVS_ScriptEnvironment *, int version);
-
-AVSC_API(char *, avs_save_string)(AVS_ScriptEnvironment *, const char* s, int length);
-AVSC_API(char *, avs_sprintf)(AVS_ScriptEnvironment *, const char * fmt, ...);
-
-AVSC_API(char *, avs_vsprintf)(AVS_ScriptEnvironment *, const char * fmt, va_list val);
- // note: val is really a va_list; I hope everyone typedefs va_list to a pointer
-
-AVSC_API(int, avs_add_function)(AVS_ScriptEnvironment *,
-                                     const char * name, const char * params,
-                                     AVS_ApplyFunc apply, void * user_data);
-
-AVSC_API(int, avs_function_exists)(AVS_ScriptEnvironment *, const char * name);
-
-AVSC_API(AVS_Value, avs_invoke)(AVS_ScriptEnvironment *, const char * name,
-                               AVS_Value args, const char** arg_names);
-// The returned value must be be released with avs_release_value
-
-AVSC_API(AVS_Value, avs_get_var)(AVS_ScriptEnvironment *, const char* name);
-// The returned value must be be released with avs_release_value
-
-AVSC_API(int, avs_set_var)(AVS_ScriptEnvironment *, const char* name, AVS_Value val);
-
-AVSC_API(int, avs_set_global_var)(AVS_ScriptEnvironment *, const char* name, const AVS_Value val);
-
-//void avs_push_context(AVS_ScriptEnvironment *, int level=0);
-//void avs_pop_context(AVS_ScriptEnvironment *);
-
-AVSC_API(AVS_VideoFrame *, avs_new_video_frame_a)(AVS_ScriptEnvironment *,
-                                          const AVS_VideoInfo * vi, int align);
-// align should be at least 16
-#if defined __cplusplus
-}
-#endif // __cplusplus
-
-#ifndef AVSC_NO_DECLSPEC
-AVSC_INLINE
-AVS_VideoFrame * avs_new_video_frame(AVS_ScriptEnvironment * env,
-                                     const AVS_VideoInfo * vi)
-  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
-
-AVSC_INLINE
-AVS_VideoFrame * avs_new_frame(AVS_ScriptEnvironment * env,
-                               const AVS_VideoInfo * vi)
-  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
-#endif
-
-#if defined __cplusplus
-extern "C"
-{
-#endif // __cplusplus
-AVSC_API(int, avs_make_writable)(AVS_ScriptEnvironment *, AVS_VideoFrame * * pvf);
-
-AVSC_API(void, avs_bit_blt)(AVS_ScriptEnvironment *, unsigned char* dstp, int dst_pitch, const unsigned char* srcp, int src_pitch, int row_size, int height);
-
-typedef void (AVSC_CC *AVS_ShutdownFunc)(void* user_data, AVS_ScriptEnvironment * env);
-AVSC_API(void, avs_at_exit)(AVS_ScriptEnvironment *, AVS_ShutdownFunc function, void * user_data);
-
-AVSC_API(AVS_VideoFrame *, avs_subframe)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height);
-// The returned video frame must be be released
-
-AVSC_API(int, avs_set_memory_max)(AVS_ScriptEnvironment *, int mem);
-
-AVSC_API(int, avs_set_working_dir)(AVS_ScriptEnvironment *, const char * newdir);
-
-// avisynth.dll exports this; it's a way to use it as a library, without
-// writing an AVS script or without going through AVIFile.
-AVSC_API(AVS_ScriptEnvironment *, avs_create_script_environment)(int version);
-#if defined __cplusplus
-}
-#endif // __cplusplus
-
-// this symbol is the entry point for the plugin and must
-// be defined
-AVSC_EXPORT
-const char * AVSC_CC avisynth_c_plugin_init(AVS_ScriptEnvironment* env);
-
-
-#if defined __cplusplus
-extern "C"
-{
-#endif // __cplusplus
-AVSC_API(void, avs_delete_script_environment)(AVS_ScriptEnvironment *);
-
-
-AVSC_API(AVS_VideoFrame *, avs_subframe_planar)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height, int rel_offsetU, int rel_offsetV, int new_pitchUV);
-// The returned video frame must be be released
-#if defined __cplusplus
-}
-#endif // __cplusplus
-
-#endif //__AVXSYNTH_C__

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__

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__

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;
-}

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;
-}

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 */

compat/os2threads.h

@@ -1,164 +0,0 @@
-/*
- * Copyright (c) 2011 KO Myung-Hun <komh@chollian.net>
- *
- * 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
- * os2threads to pthreads wrapper
- */
-
-#ifndef AVCODEC_OS2PTHREADS_H
-#define AVCODEC_OS2PTHREADS_H
-
-#define INCL_DOS
-#include <os2.h>
-
-#undef __STRICT_ANSI__          /* for _beginthread() */
-#include <stdlib.h>
-
-#include "libavutil/mem.h"
-
-typedef TID  pthread_t;
-typedef void pthread_attr_t;
-
-typedef HMTX pthread_mutex_t;
-typedef void pthread_mutexattr_t;
-
-typedef struct {
-    HEV  event_sem;
-    int  wait_count;
-} pthread_cond_t;
-
-typedef void pthread_condattr_t;
-
-struct thread_arg {
-    void *(*start_routine)(void *);
-    void *arg;
-};
-
-static void thread_entry(void *arg)
-{
-    struct thread_arg *thread_arg = arg;
-
-    thread_arg->start_routine(thread_arg->arg);
-
-    av_free(thread_arg);
-}
-
-static av_always_inline int pthread_create(pthread_t *thread, const pthread_attr_t *attr, void *(*start_routine)(void*), void *arg)
-{
-    struct thread_arg *thread_arg;
-
-    thread_arg = av_mallocz(sizeof(struct thread_arg));
-
-    thread_arg->start_routine = start_routine;
-    thread_arg->arg = arg;
-
-    *thread = _beginthread(thread_entry, NULL, 256 * 1024, thread_arg);
-
-    return 0;
-}
-
-static av_always_inline int pthread_join(pthread_t thread, void **value_ptr)
-{
-    DosWaitThread((PTID)&thread, DCWW_WAIT);
-
-    return 0;
-}
-
-static av_always_inline int pthread_mutex_init(pthread_mutex_t *mutex, const pthread_mutexattr_t *attr)
-{
-    DosCreateMutexSem(NULL, (PHMTX)mutex, 0, FALSE);
-
-    return 0;
-}
-
-static av_always_inline int pthread_mutex_destroy(pthread_mutex_t *mutex)
-{
-    DosCloseMutexSem(*(PHMTX)mutex);
-
-    return 0;
-}
-
-static av_always_inline int pthread_mutex_lock(pthread_mutex_t *mutex)
-{
-    DosRequestMutexSem(*(PHMTX)mutex, SEM_INDEFINITE_WAIT);
-
-    return 0;
-}
-
-static av_always_inline int pthread_mutex_unlock(pthread_mutex_t *mutex)
-{
-    DosReleaseMutexSem(*(PHMTX)mutex);
-
-    return 0;
-}
-
-static av_always_inline int pthread_cond_init(pthread_cond_t *cond, const pthread_condattr_t *attr)
-{
-    DosCreateEventSem(NULL, &cond->event_sem, DCE_POSTONE, FALSE);
-
-    cond->wait_count = 0;
-
-    return 0;
-}
-
-static av_always_inline int pthread_cond_destroy(pthread_cond_t *cond)
-{
-    DosCloseEventSem(cond->event_sem);
-
-    return 0;
-}
-
-static av_always_inline int pthread_cond_signal(pthread_cond_t *cond)
-{
-    if (cond->wait_count > 0) {
-        DosPostEventSem(cond->event_sem);
-
-        cond->wait_count--;
-    }
-
-    return 0;
-}
-
-static av_always_inline int pthread_cond_broadcast(pthread_cond_t *cond)
-{
-    while (cond->wait_count > 0) {
-        DosPostEventSem(cond->event_sem);
-
-        cond->wait_count--;
-    }
-
-    return 0;
-}
-
-static av_always_inline int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
-{
-    cond->wait_count++;
-
-    pthread_mutex_unlock(mutex);
-
-    DosWaitEventSem(cond->event_sem, SEM_INDEFINITE_WAIT);
-
-    pthread_mutex_lock(mutex);
-
-    return 0;
-}
-
-#endif /* AVCODEC_OS2PTHREADS_H */

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 ${n}q "$@"

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);
-}

compat/plan9/printf

@@ -1,2 +0,0 @@
-#!/bin/sh
-exec awk "BEGIN { for (i = 2; i < ARGC; i++) printf \"$1\", ARGV[i] }" "$@"

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;
-}

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 */

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

compat/w32pthreads.h

@@ -1,282 +0,0 @@
-/*
- * Copyright (C) 2010-2011 x264 project
- *
- * Authors: Steven Walters <kemuri9@gmail.com>
- *          Pegasys Inc. <http://www.pegasys-inc.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
- */
-
-/**
- * @file
- * w32threads to pthreads wrapper
- */
-
-#ifndef FFMPEG_COMPAT_W32PTHREADS_H
-#define FFMPEG_COMPAT_W32PTHREADS_H
-
-/* Build up a pthread-like API using underlying Windows API. Have only static
- * methods so as to not conflict with a potentially linked in pthread-win32
- * library.
- * As most functions here are used without checking return values,
- * only implement return values as necessary. */
-
-#define WIN32_LEAN_AND_MEAN
-#include <windows.h>
-#include <process.h>
-
-#include "libavutil/common.h"
-#include "libavutil/internal.h"
-#include "libavutil/mem.h"
-
-typedef struct pthread_t {
-    void *handle;
-    void *(*func)(void* arg);
-    void *arg;
-    void *ret;
-} pthread_t;
-
-/* the conditional variable api for windows 6.0+ uses critical sections and
- * not mutexes */
-typedef CRITICAL_SECTION pthread_mutex_t;
-
-/* This is the CONDITIONAL_VARIABLE typedef for using Window's native
- * conditional variables on kernels 6.0+.
- * MinGW does not currently have this typedef. */
-typedef struct pthread_cond_t {
-    void *ptr;
-} pthread_cond_t;
-
-/* function pointers to conditional variable API on windows 6.0+ kernels */
-#if _WIN32_WINNT < 0x0600
-static void (WINAPI *cond_broadcast)(pthread_cond_t *cond);
-static void (WINAPI *cond_init)(pthread_cond_t *cond);
-static void (WINAPI *cond_signal)(pthread_cond_t *cond);
-static BOOL (WINAPI *cond_wait)(pthread_cond_t *cond, pthread_mutex_t *mutex,
-                                DWORD milliseconds);
-#else
-#define cond_init      InitializeConditionVariable
-#define cond_broadcast WakeAllConditionVariable
-#define cond_signal    WakeConditionVariable
-#define cond_wait      SleepConditionVariableCS
-#endif
-
-static unsigned __stdcall attribute_align_arg win32thread_worker(void *arg)
-{
-    pthread_t *h = arg;
-    h->ret = h->func(h->arg);
-    return 0;
-}
-
-static int pthread_create(pthread_t *thread, const void *unused_attr,
-                          void *(*start_routine)(void*), void *arg)
-{
-    thread->func   = start_routine;
-    thread->arg    = arg;
-    thread->handle = (void*)_beginthreadex(NULL, 0, win32thread_worker, thread,
-                                           0, NULL);
-    return !thread->handle;
-}
-
-static void pthread_join(pthread_t thread, void **value_ptr)
-{
-    DWORD ret = WaitForSingleObject(thread.handle, INFINITE);
-    if (ret != WAIT_OBJECT_0)
-        return;
-    if (value_ptr)
-        *value_ptr = thread.ret;
-    CloseHandle(thread.handle);
-}
-
-static inline int pthread_mutex_init(pthread_mutex_t *m, void* attr)
-{
-    InitializeCriticalSection(m);
-    return 0;
-}
-static inline int pthread_mutex_destroy(pthread_mutex_t *m)
-{
-    DeleteCriticalSection(m);
-    return 0;
-}
-static inline int pthread_mutex_lock(pthread_mutex_t *m)
-{
-    EnterCriticalSection(m);
-    return 0;
-}
-static inline int pthread_mutex_unlock(pthread_mutex_t *m)
-{
-    LeaveCriticalSection(m);
-    return 0;
-}
-
-/* for pre-Windows 6.0 platforms we need to define and use our own condition
- * variable and api */
-typedef struct  win32_cond_t {
-    pthread_mutex_t mtx_broadcast;
-    pthread_mutex_t mtx_waiter_count;
-    volatile int waiter_count;
-    HANDLE semaphore;
-    HANDLE waiters_done;
-    volatile int is_broadcast;
-} win32_cond_t;
-
-static void pthread_cond_init(pthread_cond_t *cond, const void *unused_attr)
-{
-    win32_cond_t *win32_cond = NULL;
-    if (cond_init) {
-        cond_init(cond);
-        return;
-    }
-
-    /* non native condition variables */
-    win32_cond = av_mallocz(sizeof(win32_cond_t));
-    if (!win32_cond)
-        return;
-    cond->ptr = win32_cond;
-    win32_cond->semaphore = CreateSemaphore(NULL, 0, 0x7fffffff, NULL);
-    if (!win32_cond->semaphore)
-        return;
-    win32_cond->waiters_done = CreateEvent(NULL, TRUE, FALSE, NULL);
-    if (!win32_cond->waiters_done)
-        return;
-
-    pthread_mutex_init(&win32_cond->mtx_waiter_count, NULL);
-    pthread_mutex_init(&win32_cond->mtx_broadcast, NULL);
-}
-
-static void pthread_cond_destroy(pthread_cond_t *cond)
-{
-    win32_cond_t *win32_cond = cond->ptr;
-    /* native condition variables do not destroy */
-    if (cond_init)
-        return;
-
-    /* non native condition variables */
-    CloseHandle(win32_cond->semaphore);
-    CloseHandle(win32_cond->waiters_done);
-    pthread_mutex_destroy(&win32_cond->mtx_waiter_count);
-    pthread_mutex_destroy(&win32_cond->mtx_broadcast);
-    av_freep(&win32_cond);
-    cond->ptr = NULL;
-}
-
-static void pthread_cond_broadcast(pthread_cond_t *cond)
-{
-    win32_cond_t *win32_cond = cond->ptr;
-    int have_waiter;
-
-    if (cond_broadcast) {
-        cond_broadcast(cond);
-        return;
-    }
-
-    /* non native condition variables */
-    pthread_mutex_lock(&win32_cond->mtx_broadcast);
-    pthread_mutex_lock(&win32_cond->mtx_waiter_count);
-    have_waiter = 0;
-
-    if (win32_cond->waiter_count) {
-        win32_cond->is_broadcast = 1;
-        have_waiter = 1;
-    }
-
-    if (have_waiter) {
-        ReleaseSemaphore(win32_cond->semaphore, win32_cond->waiter_count, NULL);
-        pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
-        WaitForSingleObject(win32_cond->waiters_done, INFINITE);
-        ResetEvent(win32_cond->waiters_done);
-        win32_cond->is_broadcast = 0;
-    } else
-        pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
-    pthread_mutex_unlock(&win32_cond->mtx_broadcast);
-}
-
-static int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
-{
-    win32_cond_t *win32_cond = cond->ptr;
-    int last_waiter;
-    if (cond_wait) {
-        cond_wait(cond, mutex, INFINITE);
-        return 0;
-    }
-
-    /* non native condition variables */
-    pthread_mutex_lock(&win32_cond->mtx_broadcast);
-    pthread_mutex_lock(&win32_cond->mtx_waiter_count);
-    win32_cond->waiter_count++;
-    pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
-    pthread_mutex_unlock(&win32_cond->mtx_broadcast);
-
-    // unlock the external mutex
-    pthread_mutex_unlock(mutex);
-    WaitForSingleObject(win32_cond->semaphore, INFINITE);
-
-    pthread_mutex_lock(&win32_cond->mtx_waiter_count);
-    win32_cond->waiter_count--;
-    last_waiter = !win32_cond->waiter_count || !win32_cond->is_broadcast;
-    pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
-
-    if (last_waiter)
-        SetEvent(win32_cond->waiters_done);
-
-    // lock the external mutex
-    return pthread_mutex_lock(mutex);
-}
-
-static void pthread_cond_signal(pthread_cond_t *cond)
-{
-    win32_cond_t *win32_cond = cond->ptr;
-    int have_waiter;
-    if (cond_signal) {
-        cond_signal(cond);
-        return;
-    }
-
-    pthread_mutex_lock(&win32_cond->mtx_broadcast);
-
-    /* non-native condition variables */
-    pthread_mutex_lock(&win32_cond->mtx_waiter_count);
-    have_waiter = win32_cond->waiter_count;
-    pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
-
-    if (have_waiter) {
-        ReleaseSemaphore(win32_cond->semaphore, 1, NULL);
-        WaitForSingleObject(win32_cond->waiters_done, INFINITE);
-        ResetEvent(win32_cond->waiters_done);
-    }
-
-    pthread_mutex_unlock(&win32_cond->mtx_broadcast);
-}
-
-static void w32thread_init(void)
-{
-#if _WIN32_WINNT < 0x0600
-    HANDLE kernel_dll = GetModuleHandle(TEXT("kernel32.dll"));
-    /* if one is available, then they should all be available */
-    cond_init      =
-        (void*)GetProcAddress(kernel_dll, "InitializeConditionVariable");
-    cond_broadcast =
-        (void*)GetProcAddress(kernel_dll, "WakeAllConditionVariable");
-    cond_signal    =
-        (void*)GetProcAddress(kernel_dll, "WakeConditionVariable");
-    cond_wait      =
-        (void*)GetProcAddress(kernel_dll, "SleepConditionVariableCS");
-#endif
-
-}
-
-#endif /* FFMPEG_COMPAT_W32PTHREADS_H */

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

doc/Makefile

@@ -1,167 +0,0 @@
-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) $(LIBRARIES-yes:%=doc/%.html) \
-              doc/developer.html                                        \
-              doc/faq.html                                              \
-              doc/fate.html                                             \
-              doc/general.html                                          \
-              doc/git-howto.html                                        \
-              doc/nut.html                                              \
-              doc/platform.html                                         \
-
-TXTPAGES    = doc/fate.txt                                              \
-
-
-DOCS-$(CONFIG_HTMLPAGES) += $(HTMLPAGES)
-DOCS-$(CONFIG_PODPAGES)  += $(PODPAGES)
-DOCS-$(CONFIG_MANPAGES)  += $(MANPAGES)
-DOCS-$(CONFIG_TXTPAGES)  += $(TXTPAGES)
-DOCS = $(DOCS-yes)
-
-DOC_EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE)      += avio_reading
-DOC_EXAMPLES-$(CONFIG_AVCODEC_EXAMPLE)           += avcodec
-DOC_EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding
-DOC_EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE)      += filter_audio
-DOC_EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE)   += filtering_audio
-DOC_EXAMPLES-$(CONFIG_FILTERING_VIDEO_EXAMPLE)   += filtering_video
-DOC_EXAMPLES-$(CONFIG_METADATA_EXAMPLE)          += metadata
-DOC_EXAMPLES-$(CONFIG_MUXING_EXAMPLE)            += muxing
-DOC_EXAMPLES-$(CONFIG_REMUXING_EXAMPLE)          += remuxing
-DOC_EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE)  += resampling_audio
-DOC_EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE)     += scaling_video
-DOC_EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE)     += transcode_aac
-ALL_DOC_EXAMPLES_LIST = $(DOC_EXAMPLES-) $(DOC_EXAMPLES-yes)
-
-DOC_EXAMPLES       := $(DOC_EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)$(EXESUF))
-ALL_DOC_EXAMPLES   := $(ALL_DOC_EXAMPLES_LIST:%=doc/examples/%$(PROGSSUF)$(EXESUF))
-ALL_DOC_EXAMPLES_G := $(ALL_DOC_EXAMPLES_LIST:%=doc/examples/%$(PROGSSUF)_g$(EXESUF))
-PROGS              += $(DOC_EXAMPLES)
-
-all-$(CONFIG_DOC): doc
-
-doc: documentation
-
-apidoc: doc/doxy/html
-documentation: $(DOCS)
-
-examples: $(DOC_EXAMPLES)
-
-TEXIDEP = perl $(SRC_PATH)/doc/texidep.pl $(SRC_PATH) $< $@ >$(@:%=%.d)
-
-doc/%.txt: TAG = TXT
-doc/%.txt: doc/%.texi
-	$(Q)$(TEXIDEP)
-	$(M)makeinfo --force --no-headers -o $@ $< 2>/dev/null
-
-GENTEXI  = format codec
-GENTEXI := $(GENTEXI:%=doc/avoptions_%.texi)
-
-$(GENTEXI): TAG = GENTEXI
-$(GENTEXI): doc/avoptions_%.texi: doc/print_options$(HOSTEXESUF)
-	$(M)doc/print_options $* > $@
-
-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 $@ $<
-
-doc/%.pod: TAG = POD
-doc/%.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI)
-	$(Q)$(TEXIDEP)
-	$(M)perl $(SRC_PATH)/doc/texi2pod.pl -Dconfig-not-all=yes -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: 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
-
-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
-
-uninstall: uninstall-doc
-
-uninstall-doc: uninstall-html uninstall-man
-
-uninstall-html:
-	$(RM) -r "$(DOCDIR)"
-
-uninstall-man:
-	$(RM) $(addprefix "$(MANDIR)/man1/",$(AVPROGS-yes:%=%.1) $(AVPROGS-yes:%=%-all.1) $(COMPONENTS-yes:%=%.1))
-	$(RM) $(addprefix "$(MANDIR)/man3/",$(LIBRARIES-yes:%=%.3))
-
-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
-
--include $(wildcard $(DOCS:%=%.d))
-
-.PHONY: apidoc doc documentation

doc/RELEASE_NOTES

@@ -1,16 +0,0 @@
-Release Notes
-=============
-
-* 2.2 "Muybridge"  March, 2014
-
-
-General notes
--------------
-See the Changelog file for a list of significant changes. Note, there
-are many more new features and bugfixes than whats listed there.
-
-Bugreports against FFmpeg git master or the most recent FFmpeg release are
-accepted. If you are experiencing issues with any formally released version of
-FFmpeg, please try git master to check if the issue still exists. If it does,
-make your report against the development code following the usual bug reporting
-guidelines.

doc/TODO

@@ -0,0 +1,90 @@
+ffmpeg TODO list:
+----------------
+
+Fabrice's TODO list: (unordered)
+-------------------
+Short term:
+
+- use AVFMTCTX_DISCARD_PKT in ffplay so that DV has a chance to work
+- add RTSP regression test (both client and server)
+- make ffserver allocate AVFormatContext
+- clean up (incompatible change, for 0.5.0):
+    * AVStream -> AVComponent
+    * AVFormatContext -> AVInputStream/AVOutputStream
+    * suppress rate_emu from AVCodecContext
+- add new float/integer audio filterting and conversion : suppress
+  CODEC_ID_PCM_xxc and use CODEC_ID_RAWAUDIO.
+- fix telecine and frame rate conversion
+
+Long term (ask me if you want to help):
+
+- commit new imgconvert API and new PIX_FMT_xxx alpha formats
+- commit new LGPL'ed float and integer-only AC3 decoder
+- add WMA integer-only decoder
+- add new MPEG4-AAC audio decoder (both integer-only and float version)
+
+Michael's TODO list: (unordered) (if anyone wanna help with sth, just ask)
+-------------------
+- optimize H264 CABAC
+- more optimizations
+- simper rate control
+
+Francois' TODO list: (unordered, without any timeframe)
+-------------------
+- test MACE decoder against the openquicktime one as suggested by A'rpi
+- BeOS audio input grabbing backend
+- BeOS video input grabbing backend
+- publish my BeOS libposix on BeBits so I can officially support ffserver :)
+- check the whole code for thread-safety (global and init stuff)
+
+Philip'a TODO list: (alphabetically ordered) (please help)
+------------------
+- Add a multi-ffm filetype so that feeds can be recorded into multiple files rather
+  than one big file.
+- Authenticated users support -- where the authentication is in the URL
+- Change ASF files so that the embedded timestamp in the frames is right rather
+  than being an offset from the start of the stream
+- Make ffm files more resilient to changes in the codec structures so that you
+  can play old ffm files.
+
+Baptiste's TODO list:
+-----------------
+- mov edit list support (AVEditList)
+- YUV 10 bit per component support "2vuy"
+- mxf muxer
+- mpeg2 non linear quantizer
+
+unassigned TODO: (unordered)
+---------------
+- use AVFrame for audio codecs too
+- rework aviobuf.c buffering strategy and fix url_fskip
+- generate optimal huffman tables for mjpeg encoding
+- fix ffserver regression tests
+- support xvids motion estimation
+- support x264s motion estimation
+- support x264s rate control
+- SNOW: non translational motion compensation
+- SNOW: more optimal quantization
+- SNOW: 4x4 block support
+- SNOW: 1/8 pel motion compensation support
+- SNOW: iterative motion estimation based on subsampled images
+- SNOW: try B frames and MCTF and see how their PSNR/bitrate/complexity behaves
+- SNOW: try to use the wavelet transformed MC-ed reference frame as context for the entropy coder
+- SNOW: think about/analyize how to make snow use multiple cpus/threads
+- SNOW: finish spec
+- FLAC: lossy encoding (viterbi and naive scalar quantization)
+- libavfilter
+- JPEG2000 decoder & encoder
+- MPEG4 GMC encoding support
+- macroblock based pixel format (better cache locality, somewhat complex, one paper claimed it faster for high res)
+- regression tests for codecs which do not have an encoder (I+P-frame bitstream in svn)
+- add support for using mplayers video filters to ffmpeg
+- H264 encoder
+- per MB ratecontrol (so VCD and such do work better)
+- write a script which iteratively changes all functions between always_inline and noinline and benchmarks the result to find the best set of inlined functions
+- convert all the non SIMD asm into small asm vs. C testcases and submit them to the gcc devels so they can improve gcc
+- generic audio mixing API
+- extract PES packetizer from PS muxer and use it for new TS muxer
+- implement automatic AVBistreamFilter activation
+- make cabac encoder use bytestream (see http://trac.videolan.org/x264/changeset/?format=diff&new=651)
+- merge imdct and windowing, the current code does considerable amounts of redundant work

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.

doc/avutil.txt

@@ -0,0 +1,37 @@
+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
+softfloat.c
+
+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)

doc/bitstream_filters.texi

@@ -1,126 +0,0 @@
-@chapter Bitstream Filters
-@c man begin BITSTREAM FILTERS
-
-When you configure your FFmpeg build, all the supported bitstream
-filters are enabled by default. You can list all available ones using
-the configure option @code{--list-bsfs}.
-
-You can disable all the bitstream filters using the configure option
-@code{--disable-bsfs}, and selectively enable any bitstream filter using
-the option @code{--enable-bsf=BSF}, or you can disable a particular
-bitstream filter using the option @code{--disable-bsf=BSF}.
-
-The option @code{-bsfs} of the ff* tools will display the list of
-all the supported bitstream filters included in your build.
-
-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 h264_mp4toannexb
-
-Convert an H.264 bitstream from length prefixed mode to start code
-prefixed mode (as defined in the Annex B of the ITU-T H.264
-specification).
-
-This is required by some streaming formats, typically the MPEG-2
-transport stream format ("mpegts").
-
-For example to remux an MP4 file containing an H.264 stream to mpegts
-format with @command{ffmpeg}, you can use the command:
-
-@example
-ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts
-@end example
-
-@section imx_dump_header
-
-@section mjpeg2jpeg
-
-Convert MJPEG/AVI1 packets to full JPEG/JFIF packets.
-
-MJPEG is a video codec wherein each video frame is essentially a
-JPEG image. The individual frames can be extracted without loss,
-e.g. by
-
-@example
-ffmpeg -i ../some_mjpeg.avi -c:v copy frames_%d.jpg
-@end example
-
-Unfortunately, these chunks are incomplete JPEG images, because
-they lack the DHT segment required for decoding. Quoting from
-@url{http://www.digitalpreservation.gov/formats/fdd/fdd000063.shtml}:
-
-Avery Lee, writing in the rec.video.desktop newsgroup in 2001,
-commented that "MJPEG, or at least the MJPEG in AVIs having the
-MJPG fourcc, is restricted JPEG with a fixed -- and *omitted* --
-Huffman table. The JPEG must be YCbCr colorspace, it must be 4:2:2,
-and it must use basic Huffman encoding, not arithmetic or
-progressive. . . . You can indeed extract the MJPEG frames and
-decode them with a regular JPEG decoder, but you have to prepend
-the DHT segment to them, or else the decoder won't have any idea
-how to decompress the data. The exact table necessary is given in
-the OpenDML spec."
-
-This bitstream filter patches the header of frames extracted from an MJPEG
-stream (carrying the AVI1 header ID and lacking a DHT segment) to
-produce fully qualified JPEG images.
-
-@example
-ffmpeg -i mjpeg-movie.avi -c:v copy -bsf:v mjpeg2jpeg frame_%d.jpg
-exiftran -i -9 frame*.jpg
-ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
-@end example
-
-@section mjpega_dump_header
-
-@section movsub
-
-@section mp3_header_decompress
-
-@section noise
-
-@section remove_extra
-
-@c man end BITSTREAM FILTERS

doc/build_system.txt

@@ -1,50 +0,0 @@
-FFmpeg currently uses a custom build system, this text attempts to document
-some of its obscure features and options.
-
-Makefile variables:
-
-V
-    Disable the default terse mode, the full command issued by make and its
-    output will be shown on the screen.
-
-DESTDIR
-    Destination directory for the install targets, useful to prepare packages
-    or install FFmpeg in cross-environments.
-
-Makefile targets:
-
-all
-    Default target, builds all the libraries and the executables.
-
-fate
-    Run the fate test suite, note you must have installed it
-
-fate-list
-    Will list all fate/regression test targets
-
-install
-    Install headers, libraries and programs.
-
-libavformat/output-example
-    Build the libavformat basic example.
-
-libavcodec/api-example
-    Build the libavcodec basic example.
-
-libswscale/swscale-test
-    Build the swscale self-test (useful also as example).
-
-
-Useful standard make commands:
-make -t <target>
-    Touch all files that otherwise would be build, this is useful to reduce
-    unneeded rebuilding when changing headers, but note you must force rebuilds
-    of files that actually need it by hand then.
-
-make -j<num>
-    rebuild with multiple jobs at the same time. Faster on multi processor systems
-
-make -k
-    continue build in case of errors, this is useful for the regression tests
-    sometimes but note it will still not run all reg tests.
-

doc/decoders.texi

@@ -1,233 +0,0 @@
-@chapter Decoders
-@c man begin DECODERS
-
-Decoders are configured elements in FFmpeg which allow the decoding of
-multimedia streams.
-
-When you configure your FFmpeg build, all the supported native decoders
-are enabled by default. Decoders requiring an external library must be enabled
-manually via the corresponding @code{--enable-lib} option. You can list all
-available decoders using the configure option @code{--list-decoders}.
-
-You can disable all the decoders with the configure option
-@code{--disable-decoders} and selectively enable / disable single decoders
-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
-enabled decoders.
-
-@c man end DECODERS
-
-@chapter Video Decoders
-@c man begin VIDEO DECODERS
-
-A description of some of the currently available video decoders
-follows.
-
-@section rawvideo
-
-Raw video decoder.
-
-This decoder decodes rawvideo streams.
-
-@subsection Options
-
-@table @option
-@item top @var{top_field_first}
-Specify the assumed field type of the input video.
-@table @option
-@item -1
-the video is assumed to be progressive (default)
-@item 0
-bottom-field-first is assumed
-@item 1
-top-field-first is assumed
-@end table
-
-@end table
-
-@c man end VIDEO DECODERS
-
-@chapter Audio Decoders
-@c man begin AUDIO DECODERS
-
-A description of some of the currently available audio decoders
-follows.
-
-@section ac3
-
-AC-3 audio decoder.
-
-This decoder implements part of ATSC A/52:2010 and ETSI TS 102 366, as well as
-the undocumented RealAudio 3 (a.k.a. dnet).
-
-@subsection AC-3 Decoder Options
-
-@table @option
-
-@item -drc_scale @var{value}
-Dynamic Range Scale Factor. The factor to apply to dynamic range values
-from the AC-3 stream. This factor is applied exponentially.
-There are 3 notable scale factor ranges:
-@table @option
-@item drc_scale == 0
-DRC disabled. Produces full range audio.
-@item 0 < drc_scale <= 1
-DRC enabled.  Applies a fraction of the stream DRC value.
-Audio reproduction is between full range and full compression.
-@item drc_scale > 1
-DRC enabled. Applies drc_scale asymmetrically.
-Loud sounds are fully compressed.  Soft sounds are enhanced.
-@end table
-
-@end table
-
-@section ffwavesynth
-
-Internal wave synthetizer.
-
-This decoder generates wave patterns according to predefined sequences. Its
-use is purely internal and the format of the data it accepts is not publicly
-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

doc/default.css

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doc/demuxers.texi

@@ -1,375 +0,0 @@
-@chapter Demuxers
-@c man begin DEMUXERS
-
-Demuxers are configured elements in FFmpeg that can 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}.
-
-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}}.
-
-The option @code{-formats} of the ff* tools will display the list of
-enabled demuxers.
-
-The description of some of the currently available demuxers follows.
-
-@section applehttp
-
-Apple HTTP Live Streaming demuxer.
-
-This demuxer presents all AVStreams from all variant streams.
-The id field is set to the bitrate variant index number. By setting
-the discard flags on AVStreams (by pressing 'a' or 'v' in ffplay),
-the caller can decide which variant streams to actually receive.
-The total bitrate of the variant that the stream belongs to is
-available in a metadata key named "variant_bitrate".
-
-@section 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 one 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.
-
-This demuxer reads the script language used by SBaGen
-@url{http://uazu.net/sbagen/} to generate binaural beats sessions. A SBG
-script looks like that:
-@example
--SE
-a: 300-2.5/3 440+4.5/0
-b: 300-2.5/0 440+4.5/3
-off: -
-NOW      == a
-+0:07:00 == b
-+0:14:00 == a
-+0:21:00 == b
-+0:30:00    off
-@end example
-
-A SBG script can mix absolute and relative timestamps. If the script uses
-either only absolute timestamps (including the script start time) or only
-relative ones, then its layout is fixed, and the conversion is
-straightforward. On the other hand, if the script mixes both kind of
-timestamps, then the @var{NOW} reference for relative timestamps will be
-taken from the current time of day at the time the script is read, and the
-script layout will be frozen according to that reference. That means that if
-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

doc/developer.texi

@@ -2,87 +2,90 @@
 
 @settitle Developer Documentation
 @titlepage
+@sp 7
 @center @titlefont{Developer Documentation}
+@sp 3
 @end titlepage
 
-@top
-
-@contents
 
 @chapter Developers Guide
 
-@section Notes for external developers
-
-This document is mostly useful for internal FFmpeg developers.
-External developers who need to use the API in their application should
-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.
-
-You can use the FFmpeg libraries in your commercial program, but you
-are encouraged to @emph{publish any patch you make}. In this case the
-best way to proceed is to send your patches to the ffmpeg-devel
-mailing list following the guidelines illustrated in the remainder of
-this document.
-
-For more detailed legal information about the use of FFmpeg in
-external programs read the @file{LICENSE} file in the source tree and
-consult @url{http://ffmpeg.org/legal.html}.
-
-@section Contributing
-
-There are 3 ways by which code gets into ffmpeg.
+@section API
 @itemize @bullet
-@item Submitting Patches to the main developer mailing list
-      see @ref{Submitting patches} for details.
-@item Directly committing changes to the main tree.
-@item Committing changes to a git clone, for example on github.com or
-      gitorious.org. And asking us to merge these changes.
+@item libavcodec is the library containing the codecs (both encoding and
+decoding). Look at @file{libavcodec/apiexample.c} to see how to use it.
+
+@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{libavformat/output-example.c} to use it to generate
+audio or video streams.
+
 @end itemize
 
-Whichever way, changes should be reviewed by the maintainer of the code
-before they are committed. And they should follow the @ref{Coding Rules}.
-The developer making the commit and the author are responsible for their changes
-and should try to fix issues their commit causes.
+@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.
 
 @anchor{Coding Rules}
 @section Coding Rules
 
-@subsection Code formatting conventions
-
-There are the following guidelines regarding the indentation in files:
-
+FFmpeg is programmed in the ISO C90 language with a few additional
+features from ISO C99, namely:
 @itemize @bullet
 @item
-Indent size is 4.
-
+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
+
+These features are supported by all compilers we care about, so we will not
+accept patches to remove their use unless they absolutely do not impair
+clarity and performance.
+
+All code must compile with GCC 2.95 and GCC 3.3. Currently, FFmpeg also
+compiles with several other compilers, such as the Compaq ccc compiler
+or Sun Studio 9, and we would like to keep it that way unless it would
+be exceedingly involved. To ensure compatibility, please do not use any
+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
+
+Indent size is 4.
+The presentation is one inspired by 'indent -i4 -kr -nut'.
 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.
-@end itemize
-The presentation is one inspired by 'indent -i4 -kr -nut'.
+rejected by the Subversion repository.
 
 The main priority in FFmpeg is simplicity and small code size in order to
 minimize the bug count.
 
-@subsection Comments
-Use the JavaDoc/Doxygen  format (see examples below) so that code documentation
+Comments: Use the JavaDoc/Doxygen
+format (see examples below) so that code documentation
 can be generated automatically. All nontrivial functions should have a comment
 above them explaining what the function does, even if it is just one sentence.
 All structures and their member variables should be documented, too.
-
-Avoid Qt-style and similar Doxygen syntax with @code{!} in it, i.e. replace
-@code{//!} with @code{///} and similar.  Also @@ syntax should be employed
-for markup commands, i.e. use @code{@@param} and not @code{\param}.
-
 @example
 /**
- * @@file
+ * @@file mpeg.c
  * MPEG codec.
  * @@author ...
  */
@@ -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 */
@@ -110,300 +113,137 @@ int myfunc(int my_parameter)
 ...
 @end example
 
-@subsection C language features
-
-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
-
-These features are supported by all compilers we care about, so we will not
-accept patches to remove their use unless they absolutely do not impair
-clarity and performance.
-
-All code must compile with recent versions of GCC and a number of other
-currently supported compilers. To ensure compatibility, please do not use
-additional C99 features or GCC extensions. Especially watch out for:
-
-@itemize @bullet
-@item
-mixing statements and declarations;
-
-@item
-@samp{long long} (use @samp{int64_t} instead);
-
-@item
-@samp{__attribute__} not protected by @samp{#ifdef __GNUC__} or similar;
-
-@item
-GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}).
-@end itemize
-
-@subsection Naming conventions
-All names should be composed with underscores (_), not CamelCase. For example,
-@samp{avfilter_get_video_buffer} is an acceptable function name and
-@samp{AVFilterGetVideo} is not. The exception from this are type names, like
-for example structs and enums; they should always be in the CamelCase
-
-There are the following conventions for naming variables and functions:
-
-@itemize @bullet
-@item
-For local variables no prefix is required.
-
-@item
-For file-scope variables and functions declared as @code{static}, no prefix
-is required.
-
-@item
-For variables and functions visible outside of file scope, but only used
-internally by a library, an @code{ff_} prefix should be used,
-e.g. @samp{ff_w64_demuxer}.
-
-@item
-For variables and functions visible outside of file scope, used internally
-across multiple libraries, use @code{avpriv_} as prefix, for example,
-@samp{avpriv_aac_parse_header}.
-
-@item
-Each library has its own prefix for public symbols, in addition to the
-commonly used @code{av_} (@code{avformat_} for libavformat,
-@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.
-@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
-
-@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.
-@end itemize
-
-@subsection Editor configuration
-In order to configure Vim to follow FFmpeg formatting conventions, paste
-the following snippet into your @file{.vimrc}:
-@example
-" indentation rules for FFmpeg: 4 spaces, no tabs
-set expandtab
-set shiftwidth=4
-set softtabstop=4
-set cindent
-set cinoptions=(0
-" Allow tabs in Makefiles.
-autocmd FileType make,automake set noexpandtab shiftwidth=8 softtabstop=8
-" Trailing whitespace and tabs are forbidden, so highlight them.
-highlight ForbiddenWhitespace ctermbg=red guibg=red
-match ForbiddenWhitespace /\s\+$\|\t/
-" Do not highlight spaces at the end of line while typing on that line.
-autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@@<!$/
-@end example
-
-For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
-@example
-(c-add-style "ffmpeg"
-             '("k&r"
-               (c-basic-offset . 4)
-               (indent-tabs-mode . nil)
-               (show-trailing-whitespace . t)
-               (c-offsets-alist
-                (statement-cont . (c-lineup-assignments +)))
-               )
-             )
-(setq c-default-style "ffmpeg")
-@end example
 
 @section Development Policy
 
 @enumerate
 @item
-Contributions should be licensed under the
-@uref{http://www.gnu.org/licenses/lgpl-2.1.html, LGPL 2.1},
-including an "or any later version" clause, or, if you prefer
-a gift-style license, the
-@uref{http://opensource.org/licenses/isc-license.txt, ISC} or
-@uref{http://mit-license.org/, MIT} license.
-@uref{http://www.gnu.org/licenses/gpl-2.0.html, GPL 2} including
-an "or any later version" clause is also acceptable, but LGPL is
-preferred.
-If you add a new file, give it a proper license header. Do not copy and
-paste it from a random place, use an existing file as template.
-
+   Contributions should be licensed under the 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 program (renaming options etc) 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.
 @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.
-
+   If you apply a patch by someone else, include the name and email address in
+   the log message. Since the ffmpeg-cvslog mailing list is publicly
+   archived you should add some SPAM protection to the email address. 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.
 
-@anchor{Submitting patches}
+Note, these rules are mostly borrowed from the MPlayer project.
+
 @section Submitting patches
 
-First, read the @ref{Coding Rules} above if you did not yet, in particular
-the rules regarding patch submission.
+First, (@pxref{Coding Rules}) above if you did not yet.
 
-When you submit your patch, please use @code{git format-patch} or
-@code{git send-email}. We cannot read other diffs :-)
+When you submit your patch, try to send a unified diff (diff '-up'
+option). We cannot read other diffs :-)
 
 Also please do not submit a patch which contains several unrelated changes.
 Split it into separate, self-contained pieces. This does not mean splitting
@@ -415,8 +255,13 @@ for us and greatly increases your chances of getting your patch applied.
 Use the patcheck tool of FFmpeg to check your patch.
 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.
+Run the regression tests before submitting a patch so that you can
+verify that there are no big 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.mplayerhq.hu/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
@@ -425,13 +270,6 @@ 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
@@ -446,167 +284,110 @@ 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 in @file{avcodec.h} or
+    @file{avformat.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?
-
-@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 "svn add" the appropriate files before commiting?
 @end enumerate
 
-
 @section patch submission checklist
 
 @enumerate
 @item
-Does @code{make fate} pass with the patch applied?
-
+    Do the regression tests pass with the patch applied?
 @item
-Was the patch generated with git format-patch or send-email?
-
+    Does @code{make checkheaders} pass with the patch applied?
 @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.
-
+    Is the patch a unified diff?
 @item
-Did you provide a clear git commit log message?
-
+    Is the patch against latest FFmpeg SVN?
 @item
-Is the patch against latest FFmpeg git master branch?
-
+    Are you subscribed to ffmpeg-dev?
+    (the list is subscribers only due to spam)
 @item
-Are you subscribed to ffmpeg-devel?
-(the list is subscribers only due to spam)
-
+    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
-Have you checked that the changes are minimal, so that the same cannot be
-achieved with a smaller patch and/or simpler final code?
-
+    If the change is to speed critical code, did you benchmark it?
 @item
-If the change is to speed critical code, did you benchmark it?
-
+    If you did any benchmarks, did you provide them in the mail?
 @item
-If you did any benchmarks, did you provide them in the mail?
-
+    Have you checked that the patch does not introduce buffer overflows or
+    other security issues?
 @item
-Have you checked that the patch does not introduce buffer overflows or
-other security issues?
-
+    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
-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.
-
+    Is the patch created from the root of the source tree, so it can be
+    applied with @code{patch -p0}?
 @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.
-
-@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.
+    Did you provide a suggestion for a clear commit log message?
 @end enumerate
 
 @section Patch review process
 
 All patches posted to ffmpeg-devel will be reviewed, unless they contain a
-clear note that the patch is not for the git master branch.
+clear note that the patch is not for SVN.
 Reviews and comments will be posted as replies to the patch on the
 mailing list. The patch submitter then has to take care of every comment,
 that can be by resubmitting a changed patch or by discussion. Resubmitted
@@ -619,179 +400,37 @@ After a patch is approved it will be committed to the repository.
 We will review all submitted patches, but sometimes we are quite busy so
 especially for large patches this can take several weeks.
 
-If you feel that the review process is too slow and you are willing to try to
-take over maintainership of the area of code you change then just clone
-git master and maintain the area of code there. We will merge each area from
-where its best maintained.
-
 When resubmitting patches, please do not make any significant changes
 not related to the comments received during review. Such patches will
-be rejected. Instead, submit significant changes or new features as
+be rejected. Instead, submit  significant changes or new features as
 separate patches.
 
-@anchor{Regression tests}
 @section Regression tests
 
 Before submitting a patch (or committing to the repository), you should at least
 test that you did not break anything.
 
-Running 'make fate' accomplishes this, please see @url{fate.html} for details.
+The regression tests build a synthetic video stream and a synthetic
+audio stream. These are then encoded and decoded with all codecs or
+formats. The CRC (or MD5) of each generated file is recorded in a
+result file. A 'diff' is launched to compare the reference results and
+the result file. The output is checked immediately after each test
+has run.
+
+The regression tests then go on to test the FFserver code with a
+limited set of streams. It is important that this step runs correctly
+as well.
+
+Run 'make test' to test all the codecs and formats. Commands like
+'make regtest-mpeg2' can be used to run a single test. By default,
+make will abort if any test fails. To run all tests regardless,
+use make -k. To get a more verbose output, use 'make V=1 test' or
+'make V=2 test'.
+
+Run 'make fulltest' to test all the codecs, formats and FFserver.
 
 [Of course, some patches may change the results of the regression tests. In
 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

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

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

doc/errno.txt

@@ -1,174 +0,0 @@
-The following table lists most error codes found in various operating
-systems supported by FFmpeg.
-
-                             OS
-Code             Std    F  LBMWwb Text (YMMV)
-
-E2BIG            POSIX     ++++++  Argument list too long
-EACCES           POSIX     ++++++  Permission denied
-EADDRINUSE       POSIX     +++..+  Address in use
-EADDRNOTAVAIL    POSIX     +++..+  Cannot assign requested address
-EADV                       +.....  Advertise error
-EAFNOSUPPORT     POSIX     +++..+  Address family not supported
-EAGAIN           POSIX  +  ++++++  Resource temporarily unavailable
-EALREADY         POSIX     +++..+  Operation already in progress
-EAUTH                      .++...  Authentication error
-EBADARCH                   ..+...  Bad CPU type in executable
-EBADE                      +.....  Invalid exchange
-EBADEXEC                   ..+...  Bad executable
-EBADF            POSIX     ++++++  Bad file descriptor
-EBADFD                     +.....  File descriptor in bad state
-EBADMACHO                  ..+...  Malformed Macho file
-EBADMSG          POSIX     ++4...  Bad message
-EBADR                      +.....  Invalid request descriptor
-EBADRPC                    .++...  RPC struct is bad
-EBADRQC                    +.....  Invalid request code
-EBADSLT                    +.....  Invalid slot
-EBFONT                     +.....  Bad font file format
-EBUSY            POSIX  -  ++++++  Device or resource busy
-ECANCELED        POSIX     +++...  Operation canceled
-ECHILD           POSIX     ++++++  No child processes
-ECHRNG                     +.....  Channel number out of range
-ECOMM                      +.....  Communication error on send
-ECONNABORTED     POSIX     +++..+  Software caused connection abort
-ECONNREFUSED     POSIX  -  +++ss+  Connection refused
-ECONNRESET       POSIX     +++..+  Connection reset
-EDEADLK          POSIX     ++++++  Resource deadlock avoided
-EDEADLOCK                  +..++.  File locking deadlock error
-EDESTADDRREQ     POSIX     +++...  Destination address required
-EDEVERR                    ..+...  Device error
-EDOM             C89    -  ++++++  Numerical argument out of domain
-EDOOFUS                    .F....  Programming error
-EDOTDOT                    +.....  RFS specific error
-EDQUOT           POSIX     +++...  Disc quota exceeded
-EEXIST           POSIX     ++++++  File exists
-EFAULT           POSIX  -  ++++++  Bad address
-EFBIG            POSIX  -  ++++++  File too large
-EFTYPE                     .++...  Inappropriate file type or format
-EHOSTDOWN                  +++...  Host is down
-EHOSTUNREACH     POSIX     +++..+  No route to host
-EHWPOISON                  +.....  Memory page has hardware error
-EIDRM            POSIX     +++...  Identifier removed
-EILSEQ           C99       ++++++  Illegal byte sequence
-EINPROGRESS      POSIX  -  +++ss+  Operation in progress
-EINTR            POSIX  -  ++++++  Interrupted system call
-EINVAL           POSIX  +  ++++++  Invalid argument
-EIO              POSIX  +  ++++++  I/O error
-EISCONN          POSIX     +++..+  Socket is already connected
-EISDIR           POSIX     ++++++  Is a directory
-EISNAM                     +.....  Is a named type file
-EKEYEXPIRED                +.....  Key has expired
-EKEYREJECTED               +.....  Key was rejected by service
-EKEYREVOKED                +.....  Key has been revoked
-EL2HLT                     +.....  Level 2 halted
-EL2NSYNC                   +.....  Level 2 not synchronized
-EL3HLT                     +.....  Level 3 halted
-EL3RST                     +.....  Level 3 reset
-ELIBACC                    +.....  Can not access a needed shared library
-ELIBBAD                    +.....  Accessing a corrupted shared library
-ELIBEXEC                   +.....  Cannot exec a shared library directly
-ELIBMAX                    +.....  Too many shared libraries
-ELIBSCN                    +.....  .lib section in a.out corrupted
-ELNRNG                     +.....  Link number out of range
-ELOOP            POSIX     +++..+  Too many levels of symbolic links
-EMEDIUMTYPE                +.....  Wrong medium type
-EMFILE           POSIX     ++++++  Too many open files
-EMLINK           POSIX     ++++++  Too many links
-EMSGSIZE         POSIX     +++..+  Message too long
-EMULTIHOP        POSIX     ++4...  Multihop attempted
-ENAMETOOLONG     POSIX  -  ++++++  Filen ame too long
-ENAVAIL                    +.....  No XENIX semaphores available
-ENEEDAUTH                  .++...  Need authenticator
-ENETDOWN         POSIX     +++..+  Network is down
-ENETRESET        SUSv3     +++..+  Network dropped connection on reset
-ENETUNREACH      POSIX     +++..+  Network unreachable
-ENFILE           POSIX     ++++++  Too many open files in system
-ENOANO                     +.....  No anode
-ENOATTR                    .++...  Attribute not found
-ENOBUFS          POSIX  -  +++..+  No buffer space available
-ENOCSI                     +.....  No CSI structure available
-ENODATA          XSR       +N4...  No message available
-ENODEV           POSIX  -  ++++++  No such device
-ENOENT           POSIX  -  ++++++  No such file or directory
-ENOEXEC          POSIX     ++++++  Exec format error
-ENOFILE                    ...++.  No such file or directory
-ENOKEY                     +.....  Required key not available
-ENOLCK           POSIX     ++++++  No locks available
-ENOLINK          POSIX     ++4...  Link has been severed
-ENOMEDIUM                  +.....  No medium found
-ENOMEM           POSIX     ++++++  Not enough space
-ENOMSG           POSIX     +++..+  No message of desired type
-ENONET                     +.....  Machine is not on the network
-ENOPKG                     +.....  Package not installed
-ENOPROTOOPT      POSIX     +++..+  Protocol not available
-ENOSPC           POSIX     ++++++  No space left on device
-ENOSR            XSR       +N4...  No STREAM resources
-ENOSTR           XSR       +N4...  Not a STREAM
-ENOSYS           POSIX  +  ++++++  Function not implemented
-ENOTBLK                    +++...  Block device required
-ENOTCONN         POSIX     +++..+  Socket is not connected
-ENOTDIR          POSIX     ++++++  Not a directory
-ENOTEMPTY        POSIX     ++++++  Directory not empty
-ENOTNAM                    +.....  Not a XENIX named type file
-ENOTRECOVERABLE  SUSv4  -  +.....  State not recoverable
-ENOTSOCK         POSIX     +++..+  Socket operation on non-socket
-ENOTSUP          POSIX     +++...  Operation not supported
-ENOTTY           POSIX     ++++++  Inappropriate I/O control operation
-ENOTUNIQ                   +.....  Name not unique on network
-ENXIO            POSIX     ++++++  No such device or address
-EOPNOTSUPP       POSIX     +++..+  Operation not supported (on socket)
-EOVERFLOW        POSIX     +++..+  Value too large to be stored in data type
-EOWNERDEAD       SUSv4     +.....  Owner died
-EPERM            POSIX  -  ++++++  Operation not permitted
-EPFNOSUPPORT               +++..+  Protocol family not supported
-EPIPE            POSIX  -  ++++++  Broken pipe
-EPROCLIM                   .++...  Too many processes
-EPROCUNAVAIL               .++...  Bad procedure for program
-EPROGMISMATCH              .++...  Program version wrong
-EPROGUNAVAIL               .++...  RPC prog. not avail
-EPROTO           POSIX     ++4...  Protocol error
-EPROTONOSUPPORT  POSIX  -  +++ss+  Protocol not supported
-EPROTOTYPE       POSIX     +++..+  Protocol wrong type for socket
-EPWROFF                    ..+...  Device power is off
-ERANGE           C89    -  ++++++  Result too large
-EREMCHG                    +.....  Remote address changed
-EREMOTE                    +++...  Object is remote
-EREMOTEIO                  +.....  Remote I/O error
-ERESTART                   +.....  Interrupted system call should be restarted
-ERFKILL                    +.....  Operation not possible due to RF-kill
-EROFS            POSIX     ++++++  Read-only file system
-ERPCMISMATCH               .++...  RPC version wrong
-ESHLIBVERS                 ..+...  Shared library version mismatch
-ESHUTDOWN                  +++..+  Cannot send after socket shutdown
-ESOCKTNOSUPPORT            +++...  Socket type not supported
-ESPIPE           POSIX     ++++++  Illegal seek
-ESRCH            POSIX     ++++++  No such process
-ESRMNT                     +.....  Srmount error
-ESTALE           POSIX     +++..+  Stale NFS file handle
-ESTRPIPE                   +.....  Streams pipe error
-ETIME            XSR       +N4...  Stream ioctl timeout
-ETIMEDOUT        POSIX  -  +++ss+  Connection timed out
-ETOOMANYREFS               +++...  Too many references: cannot splice
-ETXTBSY          POSIX     +++...  Text file busy
-EUCLEAN                    +.....  Structure needs cleaning
-EUNATCH                    +.....  Protocol driver not attached
-EUSERS                     +++...  Too many users
-EWOULDBLOCK      POSIX     +++..+  Operation would block
-EXDEV            POSIX     ++++++  Cross-device link
-EXFULL                     +.....  Exchange full
-
-Notations:
-
-F: used in FFmpeg (-: a few times, +: a lot)
-
-SUSv3: Single Unix Specification, version 3
-SUSv4: Single Unix Specification, version 4
-XSR: XSI STREAMS (obsolete)
-
-OS: availability on some supported operating systems
-L: GNU/Linux
-B: BSD (F: FreeBSD, N: NetBSD)
-M: MacOS X
-W: Microsoft Windows (s: emulated with winsock, see libavformat/network.h)
-w: Mingw32 (3.17) and Mingw64 (2.0.1)
-b: BeOS

doc/examples/Makefile

@@ -1,41 +0,0 @@
-# use pkg-config for getting CFLAGS and LDLIBS
-FFMPEG_LIBS=    libavdevice                        \
-                libavformat                        \
-                libavfilter                        \
-                libavcodec                         \
-                libswresample                      \
-                libswscale                         \
-                libavutil                          \
-
-CFLAGS += -Wall -g
-CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
-LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
-
-EXAMPLES=       avio_reading                       \
-                avcodec                            \
-                demuxing_decoding                  \
-                filtering_video                    \
-                filtering_audio                    \
-                metadata                           \
-                muxing                             \
-                remuxing                           \
-                resampling_audio                   \
-                scaling_video                      \
-                transcode_aac                      \
-
-OBJS=$(addsuffix .o,$(EXAMPLES))
-
-# the following examples make explicit use of the math library
-avcodec:           LDLIBS += -lm
-muxing:            LDLIBS += -lm
-resampling_audio:  LDLIBS += -lm
-
-.phony: all clean-test clean
-
-all: $(OBJS) $(EXAMPLES)
-
-clean-test:
-	$(RM) test*.pgm test.h264 test.mp2 test.sw test.mpg
-
-clean: clean-test
-	$(RM) $(EXAMPLES) $(OBJS)

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.

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;
-}

doc/examples/avio_reading.c

@@ -1,134 +0,0 @@
-/*
- * Copyright (c) 2014 Stefano Sabatini
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to deal
- * in the Software without restriction, including without limitation the rights
- * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- * copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
- * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
- * THE SOFTWARE.
- */
-
-/**
- * @file
- * libavformat AVIOContext API example.
- *
- * Make libavformat demuxer access media content through a custom
- * AVIOContext read callback.
- * @example avio_reading.c
- */
-
-#include <libavcodec/avcodec.h>
-#include <libavformat/avformat.h>
-#include <libavformat/avio.h>
-#include <libavutil/file.h>
-
-struct buffer_data {
-    uint8_t *ptr;
-    size_t size; ///< size left in the buffer
-};
-
-static int read_packet(void *opaque, uint8_t *buf, int buf_size)
-{
-    struct buffer_data *bd = (struct buffer_data *)opaque;
-    buf_size = FFMIN(buf_size, bd->size);
-
-    printf("ptr:%p size:%zu\n", bd->ptr, bd->size);
-
-    /* copy internal buffer data to buf */
-    memcpy(buf, bd->ptr, buf_size);
-    bd->ptr  += buf_size;
-    bd->size -= buf_size;
-
-    return buf_size;
-}
-
-int main(int argc, char *argv[])
-{
-    AVFormatContext *fmt_ctx = NULL;
-    AVIOContext *avio_ctx = NULL;
-    uint8_t *buffer = NULL, *avio_ctx_buffer = NULL;
-    size_t buffer_size, avio_ctx_buffer_size = 4096;
-    char *input_filename = NULL;
-    int ret = 0;
-    struct buffer_data bd = { 0 };
-
-    if (argc != 2) {
-        fprintf(stderr, "usage: %s input_file\n"
-                "API example program to show how to read from a custom buffer "
-                "accessed through AVIOContext.\n", argv[0]);
-        return 1;
-    }
-    input_filename = argv[1];
-
-    /* register codecs and formats and other lavf/lavc components*/
-    av_register_all();
-
-    /* slurp file content into buffer */
-    ret = av_file_map(input_filename, &buffer, &buffer_size, 0, NULL);
-    if (ret < 0)
-        goto end;
-
-    /* fill opaque structure used by the AVIOContext read callback */
-    bd.ptr  = buffer;
-    bd.size = buffer_size;
-
-    if (!(fmt_ctx = avformat_alloc_context())) {
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-
-    avio_ctx_buffer = av_malloc(avio_ctx_buffer_size);
-    if (!avio_ctx_buffer) {
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-    avio_ctx = avio_alloc_context(avio_ctx_buffer, avio_ctx_buffer_size,
-                                  0, &bd, &read_packet, NULL, NULL);
-    if (!avio_ctx) {
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-    fmt_ctx->pb = avio_ctx;
-
-    ret = avformat_open_input(&fmt_ctx, NULL, NULL, NULL);
-    if (ret < 0) {
-        fprintf(stderr, "Could not open input\n");
-        goto end;
-    }
-
-    ret = avformat_find_stream_info(fmt_ctx, NULL);
-    if (ret < 0) {
-        fprintf(stderr, "Could not find stream information\n");
-        goto end;
-    }
-
-    av_dump_format(fmt_ctx, 0, input_filename, 0);
-
-end:
-    avformat_close_input(&fmt_ctx);
-    /* note: the internal buffer could have changed, and be != avio_ctx_buffer */
-    if (avio_ctx) {
-        av_freep(&avio_ctx->buffer);
-        av_freep(&avio_ctx);
-    }
-    av_file_unmap(buffer, buffer_size);
-
-    if (ret < 0) {
-        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
-        return 1;
-    }
-
-    return 0;
-}

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;
-}

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;
-}

doc/examples/filtering_audio.c

@@ -1,280 +0,0 @@
-/*
- * Copyright (c) 2010 Nicolas George
- * Copyright (c) 2011 Stefano Sabatini
- * Copyright (c) 2012 Clément Bœsch
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to deal
- * in the Software without restriction, including without limitation the rights
- * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- * copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
- * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
- * THE SOFTWARE.
- */
-
-/**
- * @file
- * API example for audio decoding and filtering
- * @example filtering_audio.c
- */
-
-#include <unistd.h>
-
-#include <libavcodec/avcodec.h>
-#include <libavformat/avformat.h>
-#include <libavfilter/avfiltergraph.h>
-#include <libavfilter/avcodec.h>
-#include <libavfilter/buffersink.h>
-#include <libavfilter/buffersrc.h>
-#include <libavutil/opt.h>
-
-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 -";
-
-static AVFormatContext *fmt_ctx;
-static AVCodecContext *dec_ctx;
-AVFilterContext *buffersink_ctx;
-AVFilterContext *buffersrc_ctx;
-AVFilterGraph *filter_graph;
-static int audio_stream_index = -1;
-
-static int open_input_file(const char *filename)
-{
-    int ret;
-    AVCodec *dec;
-
-    if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
-        return ret;
-    }
-
-    if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
-        return ret;
-    }
-
-    /* select the audio stream */
-    ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot find a audio stream in the input file\n");
-        return ret;
-    }
-    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) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n");
-        return ret;
-    }
-
-    return 0;
-}
-
-static int init_filters(const char *filters_descr)
-{
-    char args[512];
-    int ret = 0;
-    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 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);
-    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;
-    }
-
-    /* buffer audio sink: to terminate the filter chain. */
-    ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out",
-                                       NULL, NULL, filter_graph);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
-        goto end;
-    }
-
-    ret = av_opt_set_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;
-    }
-
-    /* Endpoints for the filter graph. */
-    outputs->name       = av_strdup("in");
-    outputs->filter_ctx = buffersrc_ctx;
-    outputs->pad_idx    = 0;
-    outputs->next       = NULL;
-
-    inputs->name       = av_strdup("out");
-    inputs->filter_ctx = buffersink_ctx;
-    inputs->pad_idx    = 0;
-    inputs->next       = NULL;
-
-    if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
-                                        &inputs, &outputs, NULL)) < 0)
-        goto end;
-
-    if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
-        goto end;
-
-    /* Print summary of the sink buffer
-     * Note: args buffer is reused to store channel layout string */
-    outlink = buffersink_ctx->inputs[0];
-    av_get_channel_layout_string(args, sizeof(args), -1, outlink->channel_layout);
-    av_log(NULL, AV_LOG_INFO, "Output: srate:%dHz fmt:%s chlayout:%s\n",
-           (int)outlink->sample_rate,
-           (char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
-           args);
-
-end:
-    avfilter_inout_free(&inputs);
-    avfilter_inout_free(&outputs);
-
-    return ret;
-}
-
-static void print_frame(const AVFrame *frame)
-{
-    const int n = frame->nb_samples * av_get_channel_layout_nb_channels(av_frame_get_channel_layout(frame));
-    const uint16_t *p     = (uint16_t*)frame->data[0];
-    const uint16_t *p_end = p + n;
-
-    while (p < p_end) {
-        fputc(*p    & 0xff, stdout);
-        fputc(*p>>8 & 0xff, stdout);
-        p++;
-    }
-    fflush(stdout);
-}
-
-int main(int argc, char **argv)
-{
-    int ret;
-    AVPacket packet0, packet;
-    AVFrame *frame = av_frame_alloc();
-    AVFrame *filt_frame = av_frame_alloc();
-    int got_frame;
-
-    if (!frame || !filt_frame) {
-        perror("Could not allocate frame");
-        exit(1);
-    }
-    if (argc != 2) {
-        fprintf(stderr, "Usage: %s file | %s\n", argv[0], player);
-        exit(1);
-    }
-
-    av_register_all();
-    avfilter_register_all();
-
-    if ((ret = open_input_file(argv[1])) < 0)
-        goto end;
-    if ((ret = init_filters(filter_descr)) < 0)
-        goto end;
-
-    /* read all packets */
-    packet0.data = NULL;
-    packet.data = NULL;
-    while (1) {
-        if (!packet0.data) {
-            if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
-                break;
-            packet0 = packet;
-        }
-
-        if (packet.stream_index == audio_stream_index) {
-            got_frame = 0;
-            ret = avcodec_decode_audio4(dec_ctx, frame, &got_frame, &packet);
-            if (ret < 0) {
-                av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n");
-                continue;
-            }
-            packet.size -= ret;
-            packet.data += ret;
-
-            if (got_frame) {
-                /* push the audio data from decoded frame into the filtergraph */
-                if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, 0) < 0) {
-                    av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
-                    break;
-                }
-
-                /* pull filtered audio from the filtergraph */
-                while (1) {
-                    ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
-                    if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
-                        break;
-                    if (ret < 0)
-                        goto end;
-                    print_frame(filt_frame);
-                    av_frame_unref(filt_frame);
-                }
-            }
-
-            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);
-    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));
-        exit(1);
-    }
-
-    exit(0);
-}

doc/examples/filtering_video.c

@@ -1,261 +0,0 @@
-/*
- * Copyright (c) 2010 Nicolas George
- * Copyright (c) 2011 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
- * API example for decoding and filtering
- * @example filtering_video.c
- */
-
-#define _XOPEN_SOURCE 600 /* for usleep */
-#include <unistd.h>
-
-#include <libavcodec/avcodec.h>
-#include <libavformat/avformat.h>
-#include <libavfilter/avfiltergraph.h>
-#include <libavfilter/avcodec.h>
-#include <libavfilter/buffersink.h>
-#include <libavfilter/buffersrc.h>
-#include <libavutil/opt.h>
-
-const char *filter_descr = "scale=78:24";
-
-static AVFormatContext *fmt_ctx;
-static AVCodecContext *dec_ctx;
-AVFilterContext *buffersink_ctx;
-AVFilterContext *buffersrc_ctx;
-AVFilterGraph *filter_graph;
-static int video_stream_index = -1;
-static int64_t last_pts = AV_NOPTS_VALUE;
-
-static int open_input_file(const char *filename)
-{
-    int ret;
-    AVCodec *dec;
-
-    if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
-        return ret;
-    }
-
-    if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
-        return ret;
-    }
-
-    /* select the video stream */
-    ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, &dec, 0);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot find a video stream in the input file\n");
-        return ret;
-    }
-    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) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot open video decoder\n");
-        return ret;
-    }
-
-    return 0;
-}
-
-static int init_filters(const char *filters_descr)
-{
-    char args[512];
-    int ret = 0;
-    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 };
-
-    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);
-
-    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;
-    }
-
-    /* buffer video sink: to terminate the filter chain. */
-    ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
-                                       NULL, NULL, filter_graph);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
-        goto end;
-    }
-
-    ret = av_opt_set_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;
-    }
-
-    /* Endpoints for the filter graph. */
-    outputs->name       = av_strdup("in");
-    outputs->filter_ctx = buffersrc_ctx;
-    outputs->pad_idx    = 0;
-    outputs->next       = NULL;
-
-    inputs->name       = av_strdup("out");
-    inputs->filter_ctx = buffersink_ctx;
-    inputs->pad_idx    = 0;
-    inputs->next       = NULL;
-
-    if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
-                                    &inputs, &outputs, NULL)) < 0)
-        goto end;
-
-    if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
-        goto end;
-
-end:
-    avfilter_inout_free(&inputs);
-    avfilter_inout_free(&outputs);
-
-    return ret;
-}
-
-static void display_frame(const AVFrame *frame, AVRational time_base)
-{
-    int x, y;
-    uint8_t *p0, *p;
-    int64_t delay;
-
-    if (frame->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,
-                                 time_base, AV_TIME_BASE_Q);
-            if (delay > 0 && delay < 1000000)
-                usleep(delay);
-        }
-        last_pts = frame->pts;
-    }
-
-    /* Trivial ASCII grayscale display. */
-    p0 = frame->data[0];
-    puts("\033c");
-    for (y = 0; y < frame->height; y++) {
-        p = p0;
-        for (x = 0; x < frame->width; x++)
-            putchar(" .-+#"[*(p++) / 52]);
-        putchar('\n');
-        p0 += frame->linesize[0];
-    }
-    fflush(stdout);
-}
-
-int main(int argc, char **argv)
-{
-    int ret;
-    AVPacket packet;
-    AVFrame *frame = av_frame_alloc();
-    AVFrame *filt_frame = av_frame_alloc();
-    int got_frame;
-
-    if (!frame || !filt_frame) {
-        perror("Could not allocate frame");
-        exit(1);
-    }
-    if (argc != 2) {
-        fprintf(stderr, "Usage: %s file\n", argv[0]);
-        exit(1);
-    }
-
-    av_register_all();
-    avfilter_register_all();
-
-    if ((ret = open_input_file(argv[1])) < 0)
-        goto end;
-    if ((ret = init_filters(filter_descr)) < 0)
-        goto end;
-
-    /* read all packets */
-    while (1) {
-        if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
-            break;
-
-        if (packet.stream_index == video_stream_index) {
-            got_frame = 0;
-            ret = avcodec_decode_video2(dec_ctx, frame, &got_frame, &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);
-
-                /* push the decoded frame into the filtergraph */
-                if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
-                    av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
-                    break;
-                }
-
-                /* pull filtered frames from the filtergraph */
-                while (1) {
-                    ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
-                    if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
-                        break;
-                    if (ret < 0)
-                        goto end;
-                    display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base);
-                    av_frame_unref(filt_frame);
-                }
-                av_frame_unref(frame);
-            }
-        }
-        av_free_packet(&packet);
-    }
-end:
-    avfilter_graph_free(&filter_graph);
-    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));
-        exit(1);
-    }
-
-    exit(0);
-}

doc/examples/metadata.c

@@ -1,56 +0,0 @@
-/*
- * Copyright (c) 2011 Reinhard Tartler
- *
- * 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
- * Shows how the metadata API can be used in application programs.
- * @example metadata.c
- */
-
-#include <stdio.h>
-
-#include <libavformat/avformat.h>
-#include <libavutil/dict.h>
-
-int main (int argc, char **argv)
-{
-    AVFormatContext *fmt_ctx = NULL;
-    AVDictionaryEntry *tag = NULL;
-    int ret;
-
-    if (argc != 2) {
-        printf("usage: %s <input_file>\n"
-               "example program to demonstrate the use of the libavformat metadata API.\n"
-               "\n", argv[0]);
-        return 1;
-    }
-
-    av_register_all();
-    if ((ret = avformat_open_input(&fmt_ctx, argv[1], NULL, NULL)))
-        return ret;
-
-    while ((tag = av_dict_get(fmt_ctx->metadata, "", tag, AV_DICT_IGNORE_SUFFIX)))
-        printf("%s=%s\n", tag->key, tag->value);
-
-    avformat_close_input(&fmt_ctx);
-    return 0;
-}

doc/examples/muxing.c

@@ -1,606 +0,0 @@
-/*
- * Copyright (c) 2003 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
- * libavformat API example.
- *
- * Output a media file in any supported libavformat format. The default
- * codecs are used.
- * @example muxing.c
- */
-
-#include <stdlib.h>
-#include <stdio.h>
-#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;
-
-#define STREAM_DURATION   10.0
-#define STREAM_FRAME_RATE 25 /* 25 images/s */
-#define STREAM_PIX_FMT    AV_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;
-
-    printf("pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n",
-           av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base),
-           av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base),
-           av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base),
-           pkt->stream_index);
-}
-
-static int write_frame(AVFormatContext *fmt_ctx, const AVRational *time_base, AVStream *st, AVPacket *pkt)
-{
-    /* rescale output packet timestamp values from codec to stream timebase */
-    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)
-{
-    AVCodecContext *c;
-    AVStream *st;
-
-    /* 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));
-        exit(1);
-    }
-
-    st = avformat_new_stream(oc, *codec);
-    if (!st) {
-        fprintf(stderr, "Could not allocate stream\n");
-        exit(1);
-    }
-    st->id = oc->nb_streams-1;
-    c = st->codec;
-
-    switch ((*codec)->type) {
-    case AVMEDIA_TYPE_AUDIO:
-        c->sample_fmt  = (*codec)->sample_fmts ?
-            (*codec)->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
-        c->bit_rate    = 64000;
-        c->sample_rate = 44100;
-        c->channels    = 2;
-        break;
-
-    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. */
-    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)
-{
-    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));
-        exit(1);
-    }
-
-    /* init signal generator */
-    t     = 0;
-    tincr = 2 * M_PI * 110.0 / c->sample_rate;
-    /* 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);
-}
-
-/* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
- * 'nb_channels' channels. */
-static void get_audio_frame(int16_t *samples, int frame_size, int nb_channels)
-{
-    int j, i, v;
-    int16_t *q;
-
-    q = samples;
-    for (j = 0; j < frame_size; j++) {
-        v = (int)(sin(t) * 10000);
-        for (i = 0; i < nb_channels; i++)
-            *q++ = v;
-        t     += tincr;
-        tincr += tincr2;
-    }
-}
-
-static void write_audio_frame(AVFormatContext *oc, AVStream *st, int flush)
-{
-    AVCodecContext *c;
-    AVPacket pkt = { 0 }; // data and size must be 0;
-    int got_packet, ret, dst_nb_samples;
-
-    av_init_packet(&pkt);
-    c = st->codec;
-
-    if (!flush) {
-        get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels);
-
-        /* 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;
-        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));
-        exit(1);
-    }
-}
-
-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);
-}
-
-/**************************************************************/
-/* video output */
-
-static AVFrame *frame;
-static AVPicture src_picture, dst_picture;
-static int frame_count;
-
-static void open_video(AVFormatContext *oc, AVCodec *codec, AVStream *st)
-{
-    int ret;
-    AVCodecContext *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));
-        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);
-    }
-    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));
-        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));
-            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,
-                           int width, int height)
-{
-    int x, y, i;
-
-    i = frame_index;
-
-    /* Y */
-    for (y = 0; y < height; y++)
-        for (x = 0; x < width; x++)
-            pict->data[0][y * pict->linesize[0] + x] = x + y + i * 3;
-
-    /* Cb and Cr */
-    for (y = 0; y < height / 2; y++) {
-        for (x = 0; x < width / 2; x++) {
-            pict->data[1][y * pict->linesize[1] + x] = 128 + y + i * 2;
-            pict->data[2][y * pict->linesize[2] + x] = 64 + x + i * 5;
-        }
-    }
-}
-
-static void write_video_frame(AVFormatContext *oc, AVStream *st, int flush)
-{
-    int ret;
-    static struct SwsContext *sws_ctx;
-    AVCodecContext *c = st->codec;
-
-    if (!flush) {
-        if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
-            /* as we only generate a YUV420P picture, we must convert it
-             * to the codec pixel format if needed */
-            if (!sws_ctx) {
-                sws_ctx = sws_getContext(c->width, c->height, AV_PIX_FMT_YUV420P,
-                                         c->width, c->height, c->pix_fmt,
-                                         sws_flags, NULL, NULL, NULL);
-                if (!sws_ctx) {
-                    fprintf(stderr,
-                            "Could not initialize the conversion context\n");
-                    exit(1);
-                }
-            }
-            fill_yuv_image(&src_picture, frame_count, c->width, c->height);
-            sws_scale(sws_ctx,
-                      (const uint8_t * const *)src_picture.data, src_picture.linesize,
-                      0, c->height, dst_picture.data, dst_picture.linesize);
-        } else {
-            fill_yuv_image(&dst_picture, frame_count, c->width, c->height);
-        }
-    }
-
-    if (oc->oformat->flags & AVFMT_RAWPICTURE && !flush) {
-        /* Raw video case - directly store the picture in the packet */
-        AVPacket pkt;
-        av_init_packet(&pkt);
-
-        pkt.flags        |= AV_PKT_FLAG_KEY;
-        pkt.stream_index  = st->index;
-        pkt.data          = dst_picture.data[0];
-        pkt.size          = sizeof(AVPicture);
-
-        ret = av_interleaved_write_frame(oc, &pkt);
-    } else {
-        AVPacket pkt = { 0 };
-        int got_packet;
-        av_init_packet(&pkt);
-
-        /* encode the image */
-        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);
-        }
-        /* If size is zero, it means the image was buffered. */
-
-        if (got_packet) {
-            ret = write_frame(oc, &c->time_base, st, &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));
-        exit(1);
-    }
-    frame_count++;
-}
-
-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);
-}
-
-/**************************************************************/
-/* media file output */
-
-int main(int argc, char **argv)
-{
-    const char *filename;
-    AVOutputFormat *fmt;
-    AVFormatContext *oc;
-    AVStream *audio_st, *video_st;
-    AVCodec *audio_codec, *video_codec;
-    double audio_time, video_time;
-    int flush, ret;
-
-    /* Initialize libavcodec, and register all codecs and formats. */
-    av_register_all();
-
-    if (argc != 2) {
-        printf("usage: %s output_file\n"
-               "API example program to output a media file with libavformat.\n"
-               "This program generates a synthetic audio and video stream, encodes and\n"
-               "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"
-               "\n", argv[0]);
-        return 1;
-    }
-
-    filename = argv[1];
-
-    /* allocate the output media context */
-    avformat_alloc_output_context2(&oc, NULL, NULL, filename);
-    if (!oc) {
-        printf("Could not deduce output format from file extension: using MPEG.\n");
-        avformat_alloc_output_context2(&oc, NULL, "mpeg", filename);
-    }
-    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);
-
-    /* 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);
-    if (audio_st)
-        open_audio(oc, audio_codec, 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));
-            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;
-    }
-
-    flush = 0;
-    while ((video_st && !video_is_eof) || (audio_st && !audio_is_eof)) {
-        /* 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 (!flush &&
-            (!audio_st || audio_time >= STREAM_DURATION) &&
-            (!video_st || video_time >= STREAM_DURATION)) {
-            flush = 1;
-        }
-
-        /* 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);
-        }
-    }
-
-    /* Write the trailer, if any. The trailer must be written before you
-     * close the CodecContexts open when you wrote the header; otherwise
-     * av_write_trailer() may try to use memory that was freed on
-     * av_codec_close(). */
-    av_write_trailer(oc);
-
-    /* Close each codec. */
-    if (video_st)
-        close_video(oc, video_st);
-    if (audio_st)
-        close_audio(oc, audio_st);
-
-    if (!(fmt->flags & AVFMT_NOFILE))
-        /* Close the output file. */
-        avio_close(oc->pb);
-
-    /* free the stream */
-    avformat_free_context(oc);
-
-    return 0;
-}

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;
-}

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;
-}

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;
-}

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;
-}

doc/faq.texi

@@ -2,15 +2,29 @@
 
 @settitle FFmpeg FAQ
 @titlepage
+@sp 7
 @center @titlefont{FFmpeg FAQ}
+@sp 3
 @end titlepage
 
-@top
-
-@contents
 
 @chapter General Questions
 
+@section When will the next FFmpeg version be released? / Why are FFmpeg releases so few and far between?
+
+Like most open source projects FFmpeg suffers from a certain lack of
+manpower. For this reason the developers have to prioritize the work
+they do and putting out releases is not at the top of the list, fixing
+bugs and reviewing patches takes precedence. Please don't complain or
+request more timely and/or frequent releases unless you are willing to
+help out creating them.
+
+@section I have a problem with an old version of FFmpeg; where should I report it?
+Nowhere. Upgrade to the latest release or if there is no recent release upgrade
+to Subversion HEAD. You could also try to report it. Maybe you will get lucky and
+become the first person in history to get an answer different from "upgrade
+to Subversion HEAD".
+
 @section Why doesn't FFmpeg support feature [xyz]?
 
 Because no one has taken on that task yet. FFmpeg development is
@@ -24,6 +38,30 @@ No. Windows DLLs are not portable, bloated and often slow.
 Moreover FFmpeg strives to support all codecs natively.
 A DLL loader is not conducive to that goal.
 
+@section My bug report/mail to ffmpeg-devel/user has not received any replies.
+
+Likely reasons
+@itemize
+@item We are busy and haven't had time yet to read your report or
+investigate the issue.
+@item You didn't follow bugreports.html.
+@item You didn't use Subversion HEAD.
+@item You reported a segmentation fault without gdb output.
+@item You describe a problem but not how to reproduce it.
+@item It's unclear if you use ffmpeg as command line tool or use
+libav* from another application.
+@item You speak about a video having problems on playback but
+not what you use to play it.
+@item We have no faint clue what you are talking about besides
+that it is related to FFmpeg.
+@end itemize
+
+@section Is there a forum for FFmpeg? I do not like mailing lists.
+
+You may view our mailing lists with a more forum-alike look here:
+@url{http://dir.gmane.org/gmane.comp.video.ffmpeg.user},
+but, if you post, please remember that our mailing list rules still apply there.
+
 @section I cannot read this file although this format seems to be supported by ffmpeg.
 
 Even if ffmpeg can read the container format, it may not support all its
@@ -47,7 +85,7 @@ The following list of video codecs should work on most Windows systems:
 .asf only
 @item mpeg4
 Only if you have some MPEG-4 codec like ffdshow or Xvid installed.
-@item mpeg1video
+@item mpeg1
 .mpg only
 @end table
 Note, ASF files often have .wmv or .wma extensions in Windows. It should also
@@ -59,9 +97,9 @@ The following list of audio codecs should work on most Windows systems:
 @table @option
 @item adpcm_ima_wav
 @item adpcm_ms
-@item pcm_s16le
+@item pcm
 always
-@item libmp3lame
+@item mp3
 If some MP3 codec like LAME is installed.
 @end table
 
@@ -79,23 +117,11 @@ 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?
 
-Try a @code{make distclean} in the ffmpeg source directory before the build.
-If this does not help see
+Try a @code{make distclean} in the ffmpeg source directory before the build. If this does not help see
 (@url{http://ffmpeg.org/bugreports.html}).
 
 @section How do I encode single pictures into movies?
@@ -105,21 +131,12 @@ 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 +145,7 @@ 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 $x); ln "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done
 @end example
 
 If you want to sequence them by oldest modified first, substitute
@@ -137,23 +154,17 @@ 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
@@ -161,15 +172,15 @@ The @file{movie.mpg} used as input will be converted to
 
 Instead of relying on file format self-recognition, you may also use
 @table @option
-@item -c:v ppm
-@item -c:v png
-@item -c:v mjpeg
+@item -vcodec ppm
+@item -vcodec png
+@item -vcodec mjpeg
 @end table
 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 -vcodec mjpeg menu%d.jpg
 @end example
 
 Beware that there is no "jpeg" codec. Use "mjpeg" instead.
@@ -184,28 +195,79 @@ quite logical that there is a small reduction of quality. This is not a bug.
 
 Use @file{-} as file name.
 
+@section Why does the chrominance data seem to be sampled at a different time from the luminance data on bt8x8 captures on Linux?
+
+This is a well-known bug in the bt8x8 driver. For 2.4.26 there is a patch at
+(@url{http://svn.ffmpeg.org/michael/trunk/patches/bttv-420-2.4.26.patch?view=co}). This may also
+apply cleanly to other 2.4-series kernels.
+
+@section How do I avoid the ugly aliasing artifacts in bt8x8 captures on Linux?
+
+Pass 'combfilter=1 lumafilter=1' to the bttv driver. Note though that 'combfilter=1'
+will cause somewhat too strong filtering. A fix is to apply (@url{http://svn.ffmpeg.org/michael/trunk/patches/bttv-comb-2.4.26.patch?view=co})
+or (@url{http://svn.ffmpeg.org/michael/trunk/patches/bttv-comb-2.6.6.patch?view=co})
+and pass 'combfilter=2'.
+
 @section -f jpeg doesn't work.
 
 Try '-f image2 test%d.jpg'.
 
-@section Why can I not change the frame rate?
+@section Why can I not change the framerate?
 
-Some codecs, like MPEG-1/2, only allow a small number of fixed frame rates.
-Choose a different codec with the -c:v command line option.
+Some codecs, like MPEG-1/2, only allow a small number of fixed framerates.
+Choose a different codec with the -vcodec command line option.
 
 @section How do I encode Xvid or DivX video with ffmpeg?
 
 Both Xvid and DivX (version 4+) are implementations of the ISO MPEG-4
 standard (note that there are many other coding formats that use this
-same standard). Thus, use '-c:v mpeg4' to encode in these formats. The
+same standard). Thus, use '-vcodec mpeg4' to encode in these formats. The
 default fourcc stored in an MPEG-4-coded file will be 'FMP4'. If you want
 a different fourcc, use the '-vtag' option. E.g., '-vtag xvid' will
 force the fourcc 'xvid' to be stored as the video fourcc rather than the
 default.
 
+@section How do I encode videos which play on the iPod?
+
+@table @option
+@item needed stuff
+-acodec libfaac -vcodec mpeg4 width<=320 height<=240
+@item working stuff
+4mv, title
+@item non-working stuff
+B-frames
+@item example command line
+ffmpeg -i input -acodec libfaac -ab 128kb -vcodec mpeg4 -b 1200kb -mbd 2 -flags +4mv -trellis 2 -aic 2 -cmp 2 -subcmp 2 -s 320x180 -metadata title=X output.mp4
+@end table
+
+@section How do I encode videos which play on the PSP?
+
+@table @option
+@item needed stuff
+-acodec libfaac -vcodec mpeg4 width*height<=76800 width%16=0 height%16=0 -ar 24000 -r 30000/1001 or 15000/1001 -f psp
+@item working stuff
+4mv, title
+@item non-working stuff
+B-frames
+@item example command line
+ffmpeg -i input -acodec libfaac -ab 128kb -vcodec mpeg4 -b 1200kb -ar 24000 -mbd 2 -flags +4mv -trellis 2 -aic 2 -cmp 2 -subcmp 2 -s 368x192 -r 30000/1001 -metadata title=X -f psp output.mp4
+@item needed stuff for H.264
+-acodec libfaac -vcodec libx264 width*height<=76800 width%16=0? height%16=0? -ar 48000 -coder 1 -r 30000/1001 or 15000/1001 -f psp
+@item working stuff for H.264
+title, loop filter
+@item non-working stuff for H.264
+CAVLC
+@item example command line
+ffmpeg -i input -acodec libfaac -ab 128kb -vcodec libx264 -b 1200kb -ar 48000 -mbd 2 -coder 1 -cmp 2 -subcmp 2 -s 368x192 -r 30000/1001 -metadata title=X -f psp -flags loop -trellis 2 -partitions parti4x4+parti8x8+partp4x4+partp8x8+partb8x8 output.mp4
+@item higher resolution for newer PSP firmwares, width<=480, height<=272
+-vcodec libx264 -level 21 -coder 1 -f psp
+@item example command line
+ffmpeg -i input -acodec libfaac -ab 128kb -ac 2 -ar 48000 -vcodec libx264 -level 21 -b 640kb -coder 1 -f psp -flags +loop -trellis 2 -partitions +parti4x4+parti8x8+partp4x4+partp8x8+partb8x8 -g 250 -s 480x272 output.mp4
+@end table
+
 @section Which are good parameters for encoding high quality MPEG-4?
 
-'-mbd rd -flags +mv4+aic -trellis 2 -cmp 2 -subcmp 2 -g 300 -pass 1/2',
+'-mbd rd -flags +4mv+aic -trellis 2 -cmp 2 -subcmp 2 -g 300 -pass 1/2',
 things to try: '-bf 2', '-flags qprd', '-flags mv0', '-flags skiprd'.
 
 @section Which are good parameters for encoding high quality MPEG-1/MPEG-2?
@@ -224,82 +286,24 @@ material, and try '-top 0/1' if the result looks really messed-up.
 If you have built FFmpeg with @code{./configure --enable-avisynth}
 (only possible on MinGW/Cygwin platforms),
 then you may use any file that DirectShow can read as input.
+(Be aware that this feature has been recently added,
+so you will need to help yourself in case of problems.)
 
 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:
+... 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 @url{http://www.avisynth.org/}.
 
 @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 +311,28 @@ 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 -sameq intermediate1.mpg
+ffmpeg -i input2.avi -sameq 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 -sameq 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{-sameq} 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 -sameq -y intermediate1.mpg < /dev/null &
+ffmpeg -i input2.avi -sameq -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 - -sameq -vcodec 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 +340,7 @@ 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 +357,33 @@ 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
+       -sameq -y output.flv
 rm temp[12].[av] all.[av]
 @end example
 
-@section Using @option{-f lavfi}, audio becomes mono for no apparent reason.
+@section FFmpeg does not adhere to the -maxrate setting, some frames are bigger than maxrate/fps.
 
-Use @option{-dumpgraph -} to find out exactly where the channel layout is
-lost.
+Read the MPEG spec about video buffer verifier.
 
-Most likely, it is through @code{auto-inserted aresample}. Try to understand
-why the converting filter was needed at that place.
+@section I want CBR, but no matter what I do frame sizes differ.
 
-Just before the output is a likely place, as @option{-f lavfi} currently
-only support packed S16.
+You do not understand what CBR is, please read the MPEG spec.
+Read about video buffer verifier and constant bitrate.
+The one sentence summary is that there is a buffer and the input rate is
+constant, the output can vary as needed.
 
-Then insert the correct @code{aformat} explicitly in the filtergraph,
-specifying the exact format.
+@section How do I check if a stream is CBR?
 
-@example
-aformat=sample_fmts=s16:channel_layouts=stereo
-@end example
+To quote the MPEG-2 spec:
+"There is no way to tell that a bitstream is constant bitrate without
+examining all of the vbv_delay values and making complicated computations."
 
-@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}).
 
@@ -439,52 +395,64 @@ 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?
 
 No. These tools are too bloated and they complicate the build.
 
-@section Why not rewrite FFmpeg in object-oriented C++?
+@section Why not rewrite ffmpeg in object-oriented C++?
 
 FFmpeg is already organized in a highly modular manner and does not need to
 be rewritten in a formal object language. Further, many of the developers
 favor straight C; it works for them. For more arguments on this matter,
-read @uref{http://www.tux.org/lkml/#s15, "Programming Religion"}.
+read "Programming Religion" at (@url{http://www.tux.org/lkml/#s15}).
 
 @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, used the *_g versions.
 
 @section I do not like the LGPL, can I contribute code under the GPL instead?
 
 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
+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.
+@section I want to compile xyz.c alone but my compiler produced many errors.
 
-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.
+Common code is in its own files in libav* and is used by the individual
+codecs. They will not work without the common parts, you have to compile
+the whole libav*. If you wish, disable some parts with configure switches.
+You can also try to hack it and remove more, but if you had problems fixing
+the compilation failure then you are probably not qualified for this.
 
-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.
+@section I'm using libavcodec 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
 you need to explicitly state that you are using a C library. You can do this by
@@ -492,19 +460,31 @@ encompassing your FFmpeg includes using @code{extern "C"}.
 
 See @url{http://www.parashift.com/c++-faq-lite/mixing-c-and-cpp.html#faq-32.3}
 
-@section I'm using libavutil from within my C++ application but the compiler complains about 'UINT64_C' was not declared in this scope
-
-FFmpeg is a pure C project using C99 math features, in order to enable C++
-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 libavformat/file.c in FFmpeg
+and libmpdemux/demux_lavf.c in MPlayer sources.
+
+@section I get "No compatible shell script interpreter found." in MSys.
+
+The standard MSys bash (2.04) is broken. You need to install 2.05 or later.
+
+@section I get "./configure: line <xxx>: pr: command not found" in MSys.
+
+The standard MSys install doesn't come with pr. You need to get it from the coreutils package.
+
+@section I tried to pass RTP packets into a decoder, but it doesn't work.
+
+RTP is a container format like any other, you must first depacketize the
+codec frames/samples stored in RTP and then feed to the decoder.
+
+@section Where can I find libav* headers for Pascal/Delphi?
+
+see @url{http://www.iversenit.dk/dev/ffmpeg-headers/}
 
 @section Where is the documentation about ffv1, msmpeg4, asv1, 4xm?
 
-see @url{http://www.ffmpeg.org/~michael/}
+see @url{http://svn.ffmpeg.org/michael/trunk/docs/}
 
 @section How do I feed H.263-RTP (and other codecs in RTP) to libavcodec?
 
@@ -512,25 +492,12 @@ Even if peculiar since it is network oriented, RTP is a container like any
 other. You have to @emph{demux} RTP before feeding the payload to libavcodec.
 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.
+@section AVStream.r_frame_rate is wrong, it is much larger than the framerate.
 
-@code{r_frame_rate} is NOT the average frame rate, it is the smallest frame rate
+r_frame_rate is NOT the average framerate, it is the smallest framerate
 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}.
-
-@section Why is @code{make fate} not running all tests?
-
-Make sure you have the fate-suite samples and the @code{SAMPLES} Make variable
-or @code{FATE_SAMPLES} environment variable or the @code{--samples}
-@command{configure} option is set to the right path.
-
-@section Why is @code{make fate} not finding the samples?
-
-Do you happen to have a @code{~} character in the samples path to indicate a
-home directory? The value is used in ways where the shell cannot expand it,
-causing FATE to not find files. Just replace @code{~} by the full path.
+For example, if you have mixed 25 and 30 fps content, then r_frame_rate
+will be 150.
 
 @bye

doc/fate.texi

@@ -1,205 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle FFmpeg Automated Testing Environment
-@titlepage
-@center @titlefont{FFmpeg Automated Testing Environment}
-@end titlepage
-
-@node Top
-@top
-
-@contents
-
-@chapter Introduction
-
-  FATE is an extended regression suite on the client-side and a means
-for results aggregation and presentation on the server-side.
-
-  The first part of this document explains how you can use FATE from
-your FFmpeg source directory to test your ffmpeg binary. The second
-part describes how you can run FATE to submit the results to FFmpeg's
-FATE server.
-
-  In any way you can have a look at the publicly viewable FATE results
-by visiting this website:
-
-  @url{http://fate.ffmpeg.org/}
-
-  This is especially recommended for all people contributing source
-code to FFmpeg, as it can be seen if some test on some platform broke
-with their recent contribution. This usually happens on the platforms
-the developers could not test on.
-
-  The second part of this document describes how you can run FATE to
-submit your results to FFmpeg's FATE server. If you want to submit your
-results be sure to check that your combination of CPU, OS and compiler
-is not already listed on the above mentioned website.
-
-  In the third part you can find a comprehensive listing of FATE makefile
-targets and variables.
-
-
-@chapter Using FATE from your FFmpeg source directory
-
-  If you want to run FATE on your machine you need to have the samples
-in place. You can get the samples via the build target fate-rsync.
-Use this command from the top-level source directory:
-
-@example
-make fate-rsync SAMPLES=fate-suite/
-make fate       SAMPLES=fate-suite/
-@end example
-
-  The above commands set the samples location by passing a makefile
-variable via command line. It is also possible to set the samples
-location at source configuration time by invoking configure with
-`--samples=<path to the samples directory>'. Afterwards you can
-invoke the makefile targets without setting the SAMPLES makefile
-variable. This is illustrated by the following commands:
-
-@example
-./configure --samples=fate-suite/
-make fate-rsync
-make fate
-@end example
-
-  Yet another way to tell FATE about the location of the sample
-directory is by making sure the environment variable FATE_SAMPLES
-contains the path to your samples directory. This can be achieved
-by e.g. putting that variable in your shell profile or by setting
-it in your interactive session.
-
-@example
-FATE_SAMPLES=fate-suite/ make fate
-@end example
-
-@float NOTE
-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
-to be invoked with a configuration file as its first argument.
-
-@example
-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}.
-
-@ifhtml
-  The mentioned configuration template is also available here:
-@verbatiminclude fate_config.sh.template
-@end ifhtml
-
-  Create a configuration that suits your needs, based on the configuration
-template. The `slot' configuration variable can be any string that is not
-yet used, but it is suggested that you name it adhering to the following
-pattern <arch>-<os>-<compiler>-<compiler version>. The configuration file
-itself will be sourced in a shell script, therefore all shell features may
-be used. This enables you to setup the environment as you need it for your
-build.
-
-  For your first test runs the `fate_recv' variable should be empty or
-commented out. This will run everything as normal except that it will omit
-the submission of the results to the server. The following files should be
-present in $workdir as specified in the configuration file:
-
-@itemize
-    @item configure.log
-    @item compile.log
-    @item test.log
-    @item report
-    @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}.
-
-  Configure your SSH client to use public key authentication with that key
-when connecting to the FATE server. Also do not forget to check the identity
-of the server and to accept its host key. This can usually be achieved by
-running your SSH client manually and killing it after you accepted the key.
-The FATE server's fingerprint is:
-
-@table @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.
-
-  The only thing left is to automate the execution of the fate.sh script and
-the synchronisation of the samples directory.
-
-
-@chapter FATE makefile targets and variables
-
-@section Makefile targets
-
-@table @option
-@item fate-rsync
-Download/synchronize sample files to the configured samples directory.
-
-@item fate-list
-Will list all fate/regression test targets.
-
-@item fate
-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.
-    @itemize
-        @item 0: show just the test arguments
-        @item 1: show just the command used in the test
-        @item 2: show everything
-    @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.
-
-@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}
-
-@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.
-@end table
-
-@section Examples
-
-@example
-make V=1 SAMPLES=/var/fate/samples THREADS=2 CPUFLAGS=mmx fate
-@end example

doc/fate_config.sh.template

@@ -1,29 +0,0 @@
-slot=                                    # some unique identifier
-repo=git://source.ffmpeg.org/ffmpeg.git  # the source repository
-samples=                                 # path to samples directory
-workdir=                                 # directory in which to do all the work
-#fate_recv="ssh -T fate@fate.ffmpeg.org" # command to submit report
-comment=                                 # optional description
-build_only=     # set to "yes" for a compile-only instance that skips tests
-
-# the following are optional and map to configure options
-arch=
-cpu=
-cross_prefix=
-as=
-cc=
-ld=
-target_os=
-sysroot=
-target_exec=
-target_path=
-target_samples=
-extra_cflags=
-extra_ldflags=
-extra_libs=
-extra_conf=     # extra configure options not covered above
-
-#make=          # name of GNU make if not 'make'
-makeopts=       # extra options passed to 'make'
-#tar=           # command to create a tar archive from its arguments on stdout,
-                # defaults to 'tar c'

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

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

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

doc/ffmpeg-doc.texi

@@ -0,0 +1,956 @@
+\input texinfo @c -*- texinfo -*-
+
+@settitle FFmpeg Documentation
+@titlepage
+@sp 7
+@center @titlefont{FFmpeg Documentation}
+@sp 3
+@end titlepage
+
+
+@chapter Introduction
+
+FFmpeg is a very fast video and audio converter. It can also grab from
+a live audio/video source.
+
+The command line interface is designed to be intuitive, in the sense
+that FFmpeg tries to figure out all parameters that can possibly be
+derived automatically. You usually only have to specify the target
+bitrate you want.
+
+FFmpeg can also convert from any sample rate to any other, and resize
+video on the fly with a high quality polyphase filter.
+
+@chapter Quick Start
+
+@c man begin EXAMPLES
+@section Video and Audio grabbing
+
+FFmpeg can grab video and audio from devices given that you specify the input
+format and device.
+
+@example
+ffmpeg -f oss -i /dev/dsp -f video4linux2 -i /dev/video0 /tmp/out.mpg
+@end example
+
+Note that you must activate the right video source and channel before
+launching FFmpeg with any TV viewer such as xawtv
+(@url{http://linux.bytesex.org/xawtv/}) by Gerd Knorr. You also
+have to set the audio recording levels correctly with a
+standard mixer.
+
+@section X11 grabbing
+
+FFmpeg can grab the X11 display.
+
+@example
+ffmpeg -f x11grab -s cif -i :0.0 /tmp/out.mpg
+@end example
+
+0.0 is display.screen number of your X11 server, same as
+the DISPLAY environment variable.
+
+@example
+ffmpeg -f x11grab -s cif -i :0.0+10,20 /tmp/out.mpg
+@end example
+
+0.0 is display.screen number of your X11 server, same as the DISPLAY environment
+variable. 10 is the x-offset and 20 the y-offset for the grabbing.
+
+@section Video and Audio file format conversion
+
+* FFmpeg can use any supported file format and protocol as input:
+
+Examples:
+
+* You can use YUV files as input:
+
+@example
+ffmpeg -i /tmp/test%d.Y /tmp/out.mpg
+@end example
+
+It will use the files:
+@example
+/tmp/test0.Y, /tmp/test0.U, /tmp/test0.V,
+/tmp/test1.Y, /tmp/test1.U, /tmp/test1.V, etc...
+@end example
+
+The Y files use twice the resolution of the U and V files. They are
+raw files, without header. They can be generated by all decent video
+decoders. You must specify the size of the image with the @option{-s} option
+if FFmpeg cannot guess it.
+
+* You can input from a raw YUV420P file:
+
+@example
+ffmpeg -i /tmp/test.yuv /tmp/out.avi
+@end example
+
+test.yuv is a file containing raw YUV planar data. Each frame is composed
+of the Y plane followed by the U and V planes at half vertical and
+horizontal resolution.
+
+* You can output to a raw YUV420P file:
+
+@example
+ffmpeg -i mydivx.avi hugefile.yuv
+@end example
+
+* You can set several input files and output files:
+
+@example
+ffmpeg -i /tmp/a.wav -s 640x480 -i /tmp/a.yuv /tmp/a.mpg
+@end example
+
+Converts the audio file a.wav and the raw YUV video file a.yuv
+to MPEG file a.mpg.
+
+* You can also do audio and video conversions at the same time:
+
+@example
+ffmpeg -i /tmp/a.wav -ar 22050 /tmp/a.mp2
+@end example
+
+Converts a.wav to MPEG audio at 22050 Hz sample rate.
+
+* You can encode to several formats at the same time and define a
+mapping from input stream to output streams:
+
+@example
+ffmpeg -i /tmp/a.wav -ab 64k /tmp/a.mp2 -ab 128k /tmp/b.mp2 -map 0:0 -map 0:0
+@end example
+
+Converts a.wav to a.mp2 at 64 kbits and to b.mp2 at 128 kbits. '-map
+file:index' specifies which input stream is used for each output
+stream, in the order of the definition of output streams.
+
+* You can transcode decrypted VOBs:
+
+@example
+ffmpeg -i snatch_1.vob -f avi -vcodec mpeg4 -b 800k -g 300 -bf 2 -acodec libmp3lame -ab 128k snatch.avi
+@end example
+
+This is a typical DVD ripping example; the input is a VOB file, the
+output an AVI file with MPEG-4 video and MP3 audio. Note that in this
+command we use B-frames so the MPEG-4 stream is DivX5 compatible, and
+GOP size is 300 which means one intra frame every 10 seconds for 29.97fps
+input video. Furthermore, the audio stream is MP3-encoded so you need
+to enable LAME support by passing @code{--enable-libmp3lame} to configure.
+The mapping is particularly useful for DVD transcoding
+to get the desired audio language.
+
+NOTE: To see the supported input formats, use @code{ffmpeg -formats}.
+
+* You can extract images from a video, or create a video from many images:
+
+For extracting images from a video:
+@example
+ffmpeg -i foo.avi -r 1 -s WxH -f image2 foo-%03d.jpeg
+@end example
+
+This will extract one video frame per second from the video and will
+output them in files named @file{foo-001.jpeg}, @file{foo-002.jpeg},
+etc. Images will be rescaled to fit the new WxH values.
+
+If you want to extract just a limited number of frames, you can use the
+above command in combination with the -vframes or -t option, or in
+combination with -ss to start extracting from a certain point in time.
+
+For creating a video from many images:
+@example
+ffmpeg -f image2 -i foo-%03d.jpeg -r 12 -s WxH foo.avi
+@end example
+
+The syntax @code{foo-%03d.jpeg} specifies to use a decimal number
+composed of three digits padded with zeroes to express the sequence
+number. It is the same syntax supported by the C printf function, but
+only formats accepting a normal integer are suitable.
+
+* You can put many streams of the same type in the output:
+
+@example
+ffmpeg -i test1.avi -i test2.avi -vcodec copy -acodec copy -vcodec copy -acodec copy test12.avi -newvideo -newaudio
+@end example
+
+In addition to the first video and audio streams, the resulting
+output file @file{test12.avi} will contain the second video
+and the second audio stream found in the input streams list.
+
+The @code{-newvideo}, @code{-newaudio} and @code{-newsubtitle}
+options have to be specified immediately after the name of the output
+file to which you want to add them.
+@c man end
+
+@chapter Invocation
+
+@section Syntax
+
+The generic syntax is:
+
+@example
+@c man begin SYNOPSIS
+ffmpeg [[infile options][@option{-i} @var{infile}]]... @{[outfile options] @var{outfile}@}...
+@c man end
+@end example
+@c man begin DESCRIPTION
+As a general rule, options are applied to the next specified
+file. Therefore, order is important, and you can have the same
+option on the command line multiple times. Each occurrence is
+then applied to the next input or output file.
+
+* To set the video bitrate of the output file to 64kbit/s:
+@example
+ffmpeg -i input.avi -b 64k output.avi
+@end example
+
+* To force the frame rate of the output file to 24 fps:
+@example
+ffmpeg -i input.avi -r 24 output.avi
+@end example
+
+* To force the frame rate of the input file (valid for raw formats only)
+to 1 fps and the frame rate of the output file to 24 fps:
+@example
+ffmpeg -r 1 -i input.m2v -r 24 output.avi
+@end example
+
+The format option may be needed for raw input files.
+
+By default, FFmpeg tries to convert as losslessly as possible: It
+uses the same audio and video parameters for the outputs as the one
+specified for the inputs.
+@c man end
+
+@c man begin OPTIONS
+
+@include fftools-common-opts.texi
+
+@section Main options
+
+@table @option
+
+@item -f @var{fmt}
+Force format.
+
+@item -i @var{filename}
+input file name
+
+@item -y
+Overwrite output files.
+
+@item -t @var{duration}
+Restrict the transcoded/captured video sequence
+to the duration specified in seconds.
+@code{hh:mm:ss[.xxx]} syntax is also supported.
+
+@item -fs @var{limit_size}
+Set the file size limit.
+
+@item -ss @var{position}
+Seek to given time position in seconds.
+@code{hh:mm:ss[.xxx]} syntax is also supported.
+
+@item -itsoffset @var{offset}
+Set the input time offset in seconds.
+@code{[-]hh:mm:ss[.xxx]} syntax is also supported.
+This option affects all the input files that follow it.
+The offset is added to the timestamps of the input files.
+Specifying a positive offset means that the corresponding
+streams are delayed by 'offset' seconds.
+
+@item -timestamp @var{time}
+Set the timestamp.
+
+@item -metadata @var{key}=@var{value}
+Set a metadata key/value pair.
+
+For example, for setting the title in the output file:
+@example
+ffmpeg -i in.avi -metadata title="my title" out.flv
+@end example
+
+@item -v @var{number}
+Set the logging verbosity level.
+
+@item -target @var{type}
+Specify target file type ("vcd", "svcd", "dvd", "dv", "dv50", "pal-vcd",
+"ntsc-svcd", ... ). All the format options (bitrate, codecs,
+buffer sizes) are then set automatically. You can just type:
+
+@example
+ffmpeg -i myfile.avi -target vcd /tmp/vcd.mpg
+@end example
+
+Nevertheless you can specify additional options as long as you know
+they do not conflict with the standard, as in:
+
+@example
+ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg
+@end example
+
+@item -dframes @var{number}
+Set the number of data frames to record.
+
+@item -scodec @var{codec}
+Force subtitle codec ('copy' to copy stream).
+
+@item -newsubtitle
+Add a new subtitle stream to the current output stream.
+
+@item -slang @var{code}
+Set the ISO 639 language code (3 letters) of the current subtitle stream.
+
+@end table
+
+@section Video Options
+
+@table @option
+@item -b @var{bitrate}
+Set the video bitrate in bit/s (default = 200 kb/s).
+@item -vframes @var{number}
+Set the number of video frames to record.
+@item -r @var{fps}
+Set frame rate (Hz value, fraction or abbreviation), (default = 25).
+@item -s @var{size}
+Set frame size. The format is @samp{wxh} (ffserver default = 160x128, ffmpeg default = same as source).
+The following abbreviations are recognized:
+@table @samp
+@item sqcif
+128x96
+@item qcif
+176x144
+@item cif
+352x288
+@item 4cif
+704x576
+@item 16cif
+1408x1152
+@item qqvga
+160x120
+@item qvga
+320x240
+@item vga
+640x480
+@item svga
+800x600
+@item xga
+1024x768
+@item uxga
+1600x1200
+@item qxga
+2048x1536
+@item sxga
+1280x1024
+@item qsxga
+2560x2048
+@item hsxga
+5120x4096
+@item wvga
+852x480
+@item wxga
+1366x768
+@item wsxga
+1600x1024
+@item wuxga
+1920x1200
+@item woxga
+2560x1600
+@item wqsxga
+3200x2048
+@item wquxga
+3840x2400
+@item whsxga
+6400x4096
+@item whuxga
+7680x4800
+@item cga
+320x200
+@item ega
+640x350
+@item hd480
+852x480
+@item hd720
+1280x720
+@item hd1080
+1920x1080
+@end table
+
+@item -aspect @var{aspect}
+Set aspect ratio (4:3, 16:9 or 1.3333, 1.7777).
+@item -croptop @var{size}
+Set top crop band size (in pixels).
+@item -cropbottom @var{size}
+Set bottom crop band size (in pixels).
+@item -cropleft @var{size}
+Set left crop band size (in pixels).
+@item -cropright @var{size}
+Set right crop band size (in pixels).
+@item -padtop @var{size}
+Set top pad band size (in pixels).
+@item -padbottom @var{size}
+Set bottom pad band size (in pixels).
+@item -padleft @var{size}
+Set left pad band size (in pixels).
+@item -padright @var{size}
+Set right pad band size (in pixels).
+@item -padcolor @var{hex_color}
+Set color of padded bands. The value for padcolor is expressed
+as a six digit hexadecimal number where the first two digits
+represent red, the middle two digits green and last two digits
+blue (default = 000000 (black)).
+@item -vn
+Disable video recording.
+@item -bt @var{tolerance}
+Set video bitrate tolerance (in bits, default 4000k).
+Has a minimum value of: (target_bitrate/target_framerate).
+In 1-pass mode, bitrate tolerance specifies how far ratecontrol is
+willing to deviate from the target average bitrate value. This is
+not related to min/max bitrate. Lowering tolerance too much has
+an adverse effect on quality.
+@item -maxrate @var{bitrate}
+Set max video bitrate (in bit/s).
+Requires -bufsize to be set.
+@item -minrate @var{bitrate}
+Set min video bitrate (in bit/s).
+Most useful in setting up a CBR encode:
+@example
+ffmpeg -i myfile.avi -b 4000k -minrate 4000k -maxrate 4000k -bufsize 1835k out.m2v
+@end example
+It is of little use elsewise.
+@item -bufsize @var{size}
+Set video buffer verifier buffer size (in bits).
+@item -vcodec @var{codec}
+Force video codec to @var{codec}. Use the @code{copy} special value to
+tell that the raw codec data must be copied as is.
+@item -sameq
+Use same video quality as source (implies VBR).
+
+@item -pass @var{n}
+Select the pass number (1 or 2). It is used to do two-pass
+video encoding. The statistics of the video are recorded in the first
+pass into a log file (see also the option -passlogfile),
+and in the second pass that log file is used to generate the video
+at the exact requested bitrate.
+On pass 1, you may just deactivate audio and set output to null,
+examples for Windows and Unix:
+@example
+ffmpeg -i foo.mov -vcodec libxvid -pass 1 -an -f rawvideo -y NUL
+ffmpeg -i foo.mov -vcodec libxvid -pass 1 -an -f rawvideo -y /dev/null
+@end example
+
+@item -passlogfile @var{prefix}
+Set two-pass log file name prefix to @var{prefix}, the default file name
+prefix is ``ffmpeg2pass''. The complete file name will be
+@file{PREFIX-N.log}, where N is a number specific to the output
+stream.
+
+@item -newvideo
+Add a new video stream to the current output stream.
+
+@item -vlang @var{code}
+Set the ISO 639 language code (3 letters) of the current video stream.
+
+@end table
+
+@section Advanced Video Options
+
+@table @option
+@item -pix_fmt @var{format}
+Set pixel format. Use 'list' as parameter to show all the supported
+pixel formats.
+@item -sws_flags @var{flags}
+Set SwScaler flags.
+@item -g @var{gop_size}
+Set the group of pictures size.
+@item -intra
+Use only intra frames.
+@item -vdt @var{n}
+Discard threshold.
+@item -qscale @var{q}
+Use fixed video quantizer scale (VBR).
+@item -qmin @var{q}
+minimum video quantizer scale (VBR)
+@item -qmax @var{q}
+maximum video quantizer scale (VBR)
+@item -qdiff @var{q}
+maximum difference between the quantizer scales (VBR)
+@item -qblur @var{blur}
+video quantizer scale blur (VBR) (range 0.0 - 1.0)
+@item -qcomp @var{compression}
+video quantizer scale compression (VBR) (default 0.5).
+Constant of ratecontrol equation. Recommended range for default rc_eq: 0.0-1.0
+
+@item -lmin @var{lambda}
+minimum video lagrange factor (VBR)
+@item -lmax @var{lambda}
+max video lagrange factor (VBR)
+@item -mblmin @var{lambda}
+minimum macroblock quantizer scale (VBR)
+@item -mblmax @var{lambda}
+maximum macroblock quantizer scale (VBR)
+
+These four options (lmin, lmax, mblmin, mblmax) use 'lambda' units,
+but you may use the QP2LAMBDA constant to easily convert from 'q' units:
+@example
+ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
+@end example
+
+@item -rc_init_cplx @var{complexity}
+initial complexity for single pass encoding
+@item -b_qfactor @var{factor}
+qp factor between P- and B-frames
+@item -i_qfactor @var{factor}
+qp factor between P- and I-frames
+@item -b_qoffset @var{offset}
+qp offset between P- and B-frames
+@item -i_qoffset @var{offset}
+qp offset between P- and I-frames
+@item -rc_eq @var{equation}
+Set rate control equation (@pxref{FFmpeg formula
+evaluator}) (default = @code{tex^qComp}).
+@item -rc_override @var{override}
+rate control override for specific intervals
+@item -me_method @var{method}
+Set motion estimation method to @var{method}.
+Available methods are (from lowest to best quality):
+@table @samp
+@item zero
+Try just the (0, 0) vector.
+@item phods
+@item log
+@item x1
+@item hex
+@item umh
+@item epzs
+(default method)
+@item full
+exhaustive search (slow and marginally better than epzs)
+@end table
+
+@item -dct_algo @var{algo}
+Set DCT algorithm to @var{algo}. Available values are:
+@table @samp
+@item 0
+FF_DCT_AUTO (default)
+@item 1
+FF_DCT_FASTINT
+@item 2
+FF_DCT_INT
+@item 3
+FF_DCT_MMX
+@item 4
+FF_DCT_MLIB
+@item 5
+FF_DCT_ALTIVEC
+@end table
+
+@item -idct_algo @var{algo}
+Set IDCT algorithm to @var{algo}. Available values are:
+@table @samp
+@item 0
+FF_IDCT_AUTO (default)
+@item 1
+FF_IDCT_INT
+@item 2
+FF_IDCT_SIMPLE
+@item 3
+FF_IDCT_SIMPLEMMX
+@item 4
+FF_IDCT_LIBMPEG2MMX
+@item 5
+FF_IDCT_PS2
+@item 6
+FF_IDCT_MLIB
+@item 7
+FF_IDCT_ARM
+@item 8
+FF_IDCT_ALTIVEC
+@item 9
+FF_IDCT_SH4
+@item 10
+FF_IDCT_SIMPLEARM
+@end table
+
+@item -er @var{n}
+Set error resilience to @var{n}.
+@table @samp
+@item 1
+FF_ER_CAREFUL (default)
+@item 2
+FF_ER_COMPLIANT
+@item 3
+FF_ER_AGGRESSIVE
+@item 4
+FF_ER_VERY_AGGRESSIVE
+@end table
+
+@item -ec @var{bit_mask}
+Set error concealment to @var{bit_mask}. @var{bit_mask} is a bit mask of
+the following values:
+@table @samp
+@item 1
+FF_EC_GUESS_MVS (default = enabled)
+@item 2
+FF_EC_DEBLOCK (default = enabled)
+@end table
+
+@item -bf @var{frames}
+Use 'frames' B-frames (supported for MPEG-1, MPEG-2 and MPEG-4).
+@item -mbd @var{mode}
+macroblock decision
+@table @samp
+@item 0
+FF_MB_DECISION_SIMPLE: Use mb_cmp (cannot change it yet in FFmpeg).
+@item 1
+FF_MB_DECISION_BITS: Choose the one which needs the fewest bits.
+@item 2
+FF_MB_DECISION_RD: rate distortion
+@end table
+
+@item -4mv
+Use four motion vector by macroblock (MPEG-4 only).
+@item -part
+Use data partitioning (MPEG-4 only).
+@item -bug @var{param}
+Work around encoder bugs that are not auto-detected.
+@item -strict @var{strictness}
+How strictly to follow the standards.
+@item -aic
+Enable Advanced intra coding (h263+).
+@item -umv
+Enable Unlimited Motion Vector (h263+)
+
+@item -deinterlace
+Deinterlace pictures.
+@item -ilme
+Force interlacing support in encoder (MPEG-2 and MPEG-4 only).
+Use this option if your input file is interlaced and you want
+to keep the interlaced format for minimum losses.
+The alternative is to deinterlace the input stream with
+@option{-deinterlace}, but deinterlacing introduces losses.
+@item -psnr
+Calculate PSNR of compressed frames.
+@item -vstats
+Dump video coding statistics to @file{vstats_HHMMSS.log}.
+@item -vstats_file @var{file}
+Dump video coding statistics to @var{file}.
+@item -top @var{n}
+top=1/bottom=0/auto=-1 field first
+@item -dc @var{precision}
+Intra_dc_precision.
+@item -vtag @var{fourcc/tag}
+Force video tag/fourcc.
+@item -qphist
+Show QP histogram.
+@item -vbsf @var{bitstream_filter}
+Bitstream filters available are "dump_extra", "remove_extra", "noise", "h264_mp4toannexb", "imxdump", "mjpegadump".
+@example
+ffmpeg -i h264.mp4 -vcodec copy -vbsf h264_mp4toannexb -an out.h264
+@end example
+@end table
+
+@section Audio Options
+
+@table @option
+@item -aframes @var{number}
+Set the number of audio frames to record.
+@item -ar @var{freq}
+Set the audio sampling frequency (default = 44100 Hz).
+@item -ab @var{bitrate}
+Set the audio bitrate in bit/s (default = 64k).
+@item -aq @var{q}
+Set the audio quality (codec-specific, VBR).
+@item -ac @var{channels}
+Set the number of audio channels (default = 1).
+@item -an
+Disable audio recording.
+@item -acodec @var{codec}
+Force audio codec to @var{codec}. Use the @code{copy} special value to
+specify that the raw codec data must be copied as is.
+@item -newaudio
+Add a new audio track to the output file. If you want to specify parameters,
+do so before @code{-newaudio} (@code{-acodec}, @code{-ab}, etc..).
+
+Mapping will be done automatically, if the number of output streams is equal to
+the number of input streams, else it will pick the first one that matches. You
+can override the mapping using @code{-map} as usual.
+
+Example:
+@example
+ffmpeg -i file.mpg -vcodec copy -acodec ac3 -ab 384k test.mpg -acodec mp2 -ab 192k -newaudio
+@end example
+@item -alang @var{code}
+Set the ISO 639 language code (3 letters) of the current audio stream.
+@end table
+
+@section Advanced Audio options:
+
+@table @option
+@item -atag @var{fourcc/tag}
+Force audio tag/fourcc.
+@item -absf @var{bitstream_filter}
+Bitstream filters available are "dump_extra", "remove_extra", "noise", "mp3comp", "mp3decomp".
+@end table
+
+@section Subtitle options:
+
+@table @option
+@item -scodec @var{codec}
+Force subtitle codec ('copy' to copy stream).
+@item -newsubtitle
+Add a new subtitle stream to the current output stream.
+@item -slang @var{code}
+Set the ISO 639 language code (3 letters) of the current subtitle stream.
+@item -sn
+Disable subtitle recording.
+@item -sbsf @var{bitstream_filter}
+Bitstream filters available are "mov2textsub", "text2movsub".
+@example
+ffmpeg -i file.mov -an -vn -sbsf mov2textsub -scodec copy -f rawvideo sub.txt
+@end example
+@end table
+
+@section Audio/Video grab options
+
+@table @option
+@item -vc @var{channel}
+Set video grab channel (DV1394 only).
+@item -tvstd @var{standard}
+Set television standard (NTSC, PAL (SECAM)).
+@item -isync
+Synchronize read on input.
+@end table
+
+@section Advanced options
+
+@table @option
+@item -map @var{input_stream_id}[:@var{sync_stream_id}]
+Set stream mapping from input streams to output streams.
+Just enumerate the input streams in the order you want them in the output.
+@var{sync_stream_id} if specified sets the input stream to sync
+against.
+@item -map_meta_data @var{outfile}:@var{infile}
+Set meta data information of @var{outfile} from @var{infile}.
+@item -debug
+Print specific debug info.
+@item -benchmark
+Show benchmarking information at the end of an encode.
+Shows CPU time used and maximum memory consumption.
+Maximum memory consumption is not supported on all systems,
+it will usually display as 0 if not supported.
+@item -dump
+Dump each input packet.
+@item -hex
+When dumping packets, also dump the payload.
+@item -bitexact
+Only use bit exact algorithms (for codec testing).
+@item -ps @var{size}
+Set RTP payload size in bytes.
+@item -re
+Read input at native frame rate. Mainly used to simulate a grab device.
+@item -loop_input
+Loop over the input stream. Currently it works only for image
+streams. This option is used for automatic FFserver testing.
+@item -loop_output @var{number_of_times}
+Repeatedly loop output for formats that support looping such as animated GIF
+(0 will loop the output infinitely).
+@item -threads @var{count}
+Thread count.
+@item -vsync @var{parameter}
+Video sync method.
+0   Each frame is passed with its timestamp from the demuxer to the muxer
+1   Frames will be duplicated and dropped to achieve exactly the requested
+    constant framerate.
+2   Frames are passed through with their timestamp or dropped so as to prevent
+    2 frames from having the same timestamp
+-1  Chooses between 1 and 2 depending on muxer capabilities. This is the default method.
+
+With -map you can select from
+which stream the timestamps should be taken. You can leave either video or
+audio unchanged and sync the remaining stream(s) to the unchanged one.
+@item -async @var{samples_per_second}
+Audio sync method. "Stretches/squeezes" the audio stream to match the timestamps,
+the parameter is the maximum samples per second by which the audio is changed.
+-async 1 is a special case where only the start of the audio stream is corrected
+without any later correction.
+@item -copyts
+Copy timestamps from input to output.
+@item -shortest
+Finish encoding when the shortest input stream ends.
+@item -dts_delta_threshold
+Timestamp discontinuity delta threshold.
+@item -muxdelay @var{seconds}
+Set the maximum demux-decode delay.
+@item -muxpreload @var{seconds}
+Set the initial demux-decode delay.
+@end table
+
+@section Preset files
+
+A preset file contains a sequence of @var{option}=@var{value} pairs,
+one for each line, specifying a sequence of options which would be
+awkward to specify on the command line. Lines starting with the hash
+('#') character are ignored and are used to provide comments. Check
+the @file{ffpresets} directory in the FFmpeg source tree for examples.
+
+Preset files are specified with the @code{vpre}, @code{apre},
+@code{spre}, and @code{fpre} options. The @code{fpre} option takes the
+filename of the preset instead of a preset name as input and can be
+used for any kind of codec. For the @code{vpre}, @code{apre}, and
+@code{spre} options, the options specified in a preset file are
+applied to the currently selected codec of the same type as the preset
+option.
+
+The argument passed to the @code{vpre}, @code{apre}, and @code{spre}
+preset options identifies the preset file to use according to the
+following rules:
+
+First ffmpeg searches for a file named @var{arg}.ffpreset in the
+directories @file{$FFMPEG_DATADIR} (if set), and @file{$HOME/.ffmpeg}, and in
+the datadir defined at configuration time (usually @file{PREFIX/share/ffmpeg})
+in that order. For example, if the argument is @code{libx264-max}, it will
+search for the file @file{libx264-max.ffpreset}.
+
+If no such file is found, then ffmpeg will search for a file named
+@var{codec_name}-@var{arg}.ffpreset in the above-mentioned
+directories, where @var{codec_name} is the name of the codec to which
+the preset file options will be applied. For example, if you select
+the video codec with @code{-vcodec libx264} and use @code{-vpre max},
+then it will search for the file @file{libx264-max.ffpreset}.
+
+@anchor{FFmpeg formula evaluator}
+@section FFmpeg formula evaluator
+
+When evaluating a rate control string, FFmpeg uses an internal formula
+evaluator.
+
+The following binary operators are available: @code{+}, @code{-},
+@code{*}, @code{/}, @code{^}.
+
+The following unary operators are available: @code{+}, @code{-},
+@code{(...)}.
+
+The following statements are available: @code{ld}, @code{st},
+@code{while}.
+
+The following functions are available:
+@table @var
+@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 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 bits2qp(bits)
+@item qp2bits(qp)
+@end table
+
+The following constants are available:
+@table @var
+@item PI
+@item E
+@item iTex
+@item pTex
+@item tex
+@item mv
+@item fCode
+@item iCount
+@item mcVar
+@item var
+@item isI
+@item isP
+@item isB
+@item avgQP
+@item qComp
+@item avgIITex
+@item avgPITex
+@item avgPPTex
+@item avgBPTex
+@item avgTex
+@end table
+
+@c man end
+
+@ignore
+
+@setfilename ffmpeg
+@settitle FFmpeg video converter
+
+@c man begin SEEALSO
+ffserver(1), ffplay(1) and the HTML documentation of @file{ffmpeg}.
+@c man end
+
+@c man begin AUTHOR
+Fabrice Bellard
+@c man end
+
+@end ignore
+
+@section Protocols
+
+The file name can be @file{-} to read from standard input or to write
+to standard output.
+
+FFmpeg also handles many protocols specified with an URL syntax.
+
+Use 'ffmpeg -protocols' to see a list of the supported protocols.
+
+The protocol @code{http:} is currently used only to communicate with
+FFserver (see the FFserver documentation). When FFmpeg will be a
+video player it will also be used for streaming :-)
+
+@chapter Tips
+
+@itemize
+@item For streaming at very low bitrate application, use a low frame rate
+and a small GOP size. This is especially true for RealVideo where
+the Linux player does not seem to be very fast, so it can miss
+frames. An example is:
+
+@example
+ffmpeg -g 3 -r 3 -t 10 -b 50k -s qcif -f rv10 /tmp/b.rm
+@end example
+
+@item  The parameter 'q' which is displayed while encoding is the current
+quantizer. The value 1 indicates that a very good quality could
+be achieved. The value 31 indicates the worst quality. If q=31 appears
+too often, it means that the encoder cannot compress enough to meet
+your bitrate. You must either increase the bitrate, decrease the
+frame rate or decrease the frame size.
+
+@item If your computer is not fast enough, you can speed up the
+compression at the expense of the compression ratio. You can use
+'-me zero' to speed up motion estimation, and '-intra' to disable
+motion estimation completely (you have only I-frames, which means it
+is about as good as JPEG compression).
+
+@item To have very low audio bitrates, reduce the sampling frequency
+(down to 22050 Hz for MPEG audio, 22050 or 11025 for AC-3).
+
+@item To have a constant quality (but a variable bitrate), use the option
+'-qscale n' when 'n' is between 1 (excellent quality) and 31 (worst
+quality).
+
+@item When converting video files, you can use the '-sameq' option which
+uses the same quality factor in the encoder as in the decoder.
+It allows almost lossless encoding.
+
+@end itemize
+
+@bye

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

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

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

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 one 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

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 one 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

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

doc/ffmpeg.txt

@@ -1,47 +0,0 @@
-                                                                                                                                       :
-                                             ffmpeg.c                                                                                  :       libav*
-                                             ========                                                                                  :       ======
-                                                                                                                                       :
-                                                                                                                                       :
-                                                                                                       --------------------------------:---> AVStream...
-                                                                    InputStream input_streams[]      /                                 :
-                                                                                                    /                                  :
-                    InputFile input_files[]                         +==========================+   /   ^                               :
-                                                          ------> 0 |      : st ---:-----------:--/    :                               :
-                 ^  +------+-----------+-----+          /           +--------------------------+       :                               :
-                 :  |      :ist_index--:-----:---------/          1 |      : st    :           |       :                               :
-                 :  +------+-----------+-----+                      +==========================+       :                               :
- nb_input_files  :  |      :ist_index--:-----:------------------> 2 |      : st    :           |       :                               :
-                 :  +------+-----------+-----+                      +--------------------------+       :  nb_input_streams             :
-                 :  |      :ist_index  :     |                    3 |            ...           |       :                               :
-                 v  +------+-----------+-----+                      +--------------------------+       :                               :
-                                                              --> 4 |                          |       :                               :
-                                                             |      +--------------------------+       :                               :
-                                                             |    5 |                          |       :                               :
-                                                             |      +==========================+       v                               :
-                                                             |                                                                         :
-                                                             |                                                                         :
-                                                             |                                                                         :
-                                                             |                                                                         :
-                                                              ---------                                --------------------------------:---> AVStream...
-                                                                        \                            /                                 :
-                                                                    OutputStream output_streams[]   /                                  :
-                                                                          \                        /                                   :
-                                                                    +======\======================/======+      ^                      :
-                                                          ------> 0 |   : source_index  : st-:---        |      :                      :
-                    OutputFile output_files[]           /           +------------------------------------+      :                      :
-                                                       /          1 |   :               :    :           |      :                      :
-                 ^  +------+------------+-----+       /             +------------------------------------+      :                      :
-                 :  |      : ost_index -:-----:------/            2 |   :               :    :           |      :                      :
- nb_output_files :  +------+------------+-----+                     +====================================+      :                      :
-                 :  |      : ost_index -:-----|-----------------> 3 |   :               :    :           |      :                      :
-                 :  +------+------------+-----+                     +------------------------------------+      : nb_output_streams    :
-                 :  |      :            :     |                   4 |                                    |      :                      :
-                 :  +------+------------+-----+                     +------------------------------------+      :                      :
-                 :  |      :            :     |                   5 |                                    |      :                      :
-                 v  +------+------------+-----+                     +------------------------------------+      :                      :
-                                                                  6 |                                    |      :                      :
-                                                                    +------------------------------------+      :                      :
-                                                                  7 |                                    |      :                      :
-                                                                    +====================================+      v                      :
-                                                                                                                                       :

doc/ffmpeg_powerpc_performance_evaluation_howto.txt

@@ -0,0 +1,172 @@
+FFmpeg & evaluating performance on the PowerPC Architecture HOWTO
+
+(c) 2003-2004 Romain Dolbeau <romain@dolbeau.org>
+
+
+
+I - Introduction
+
+The PowerPC architecture and its SIMD extension AltiVec offer some
+interesting tools to evaluate performance and improve the code.
+This document tries to explain how to use those tools with FFmpeg.
+
+The architecture itself offers two ways to evaluate the performance of
+a given piece of code:
+
+1) The Time Base Registers (TBL)
+2) The Performance Monitor Counter Registers (PMC)
+
+The first ones are always available, always active, but they're not very
+accurate: the registers increment by one every four *bus* cycles. On
+my 667 Mhz tiBook (ppc7450), this means once every twenty *processor*
+cycles. So we won't use that.
+
+The PMC are much more useful: not only can they report cycle-accurate
+timing, but they can also be used to monitor many other parameters,
+such as the number of AltiVec stalls for every kind of instruction,
+or instruction cache misses. The downside is that not all processors
+support the PMC (all G3, all G4 and the 970 do support them), and
+they're inactive by default - you need to activate them with a
+dedicated tool. Also, the number of available PMC depends on the
+procesor: the various 604 have 2, the various 75x (aka. G3) have 4,
+and the various 74xx (aka G4) have 6.
+
+*WARNING*: The PowerPC 970 is not very well documented, and its PMC
+registers are 64 bits wide. To properly notify the code, you *must*
+tune for the 970 (using --tune=970), or the code will assume 32 bit
+registers.
+
+
+II - Enabling FFmpeg PowerPC performance support
+
+This needs to be done by hand. First, you need to configure FFmpeg as
+usual, but add the "--powerpc-perf-enable" option. For instance:
+
+#####
+./configure --prefix=/usr/local/ffmpeg-svn --cc=gcc-3.3 --tune=7450 --powerpc-perf-enable
+#####
+
+This will configure FFmpeg to install inside /usr/local/ffmpeg-svn,
+compiling with gcc-3.3 (you should try to use this one or a newer
+gcc), and tuning for the PowerPC 7450 (i.e. the newer G4; as a rule of
+thumb, those at 550Mhz and more). It will also enable the PMC.
+
+You may also edit the file "config.h" to enable the following line:
+
+#####
+// #define ALTIVEC_USE_REFERENCE_C_CODE 1
+#####
+
+If you enable this line, then the code will not make use of AltiVec,
+but will use the reference C code instead. This is useful to compare
+performance between two versions of the code.
+
+Also, the number of enabled PMC is defined in "libavcodec/ppc/dsputil_ppc.h":
+
+#####
+#define POWERPC_NUM_PMC_ENABLED 4
+#####
+
+If you have a G4 CPU, you can enable all 6 PMC. DO NOT enable more
+PMC than available on your CPU!
+
+Then, simply compile FFmpeg as usual (make && make install).
+
+
+
+III - Using FFmpeg PowerPC performance support
+
+This FFmeg can be used exactly as usual. But before exiting, FFmpeg
+will dump a per-function report that looks like this:
+
+#####
+PowerPC performance report
+ Values are from the PMC registers, and represent whatever the
+ registers are set to record.
+ Function "gmc1_altivec" (pmc1):
+        min: 231
+        max: 1339867
+        avg: 558.25 (255302)
+ Function "gmc1_altivec" (pmc2):
+        min: 93
+        max: 2164
+        avg: 267.31 (255302)
+ Function "gmc1_altivec" (pmc3):
+        min: 72
+        max: 1987
+        avg: 276.20 (255302)
+(...)
+#####
+
+In this example, PMC1 was set to record CPU cycles, PMC2 was set to
+record AltiVec Permute Stall Cycles, and PMC3 was set to record AltiVec
+Issue Stalls.
+
+The function "gmc1_altivec" was monitored 255302 times, and the
+minimum execution time was 231 processor cycles. The max and average
+aren't much use, as it's very likely the OS interrupted execution for
+reasons of its own :-(
+
+With the exact same settings and source file, but using the reference C
+code we get:
+
+#####
+PowerPC performance report
+ Values are from the PMC registers, and represent whatever the
+ registers are set to record.
+ Function "gmc1_altivec" (pmc1):
+        min: 592
+        max: 2532235
+        avg: 962.88 (255302)
+ Function "gmc1_altivec" (pmc2):
+        min: 0
+        max: 33
+        avg: 0.00 (255302)
+ Function "gmc1_altivec" (pmc3):
+        min: 0
+        max: 350
+        avg: 0.03 (255302)
+(...)
+#####
+
+592 cycles, so the fastest AltiVec execution is about 2.5x faster than
+the fastest C execution in this example. It's not perfect but it's not
+bad (well I wrote this function so I can't say otherwise :-).
+
+Once you have that kind of report, you can try to improve things by
+finding what goes wrong and fixing it; in the example above, one
+should try to diminish the number of AltiVec stalls, as this *may*
+improve performance.
+
+
+
+IV) Enabling the PMC in Mac OS X
+
+This is easy. Use "Monster" and "monster". Those tools come from
+Apple's CHUD package, and can be found hidden in the developer web
+site & FTP site. "MONster" is the graphical application, use it to
+generate a config file specifying what each register should
+monitor. Then use the command-line application "monster" to use that
+config file, and enjoy the results.
+
+Note that "MONster" can be used for many other things, but it's
+documented by Apple, it's not my subject.
+
+If you are using CHUD 4.4.2 or later, you'll notice that MONster is
+no longer available. It's been superseeded by Shark, where
+configuration of PMCs is available as a plugin.
+
+
+
+V) Enabling the PMC on Linux
+
+On linux you may use oprofile from http://oprofile.sf.net, depending on the
+version and the cpu you may need to apply a patch[1] to access a set of the
+possibile counters from the userspace application. You can always define them
+using the kernel interface /dev/oprofile/* .
+
+[1] http://dev.gentoo.org/~lu_zero/development/oprofile-g4-20060423.patch
+
+--
+Romain Dolbeau <romain@dolbeau.org>
+Luca Barbato <lu_zero@gentoo.org>

doc/ffplay-doc.texi

@@ -0,0 +1,160 @@
+\input texinfo @c -*- texinfo -*-
+
+@settitle FFplay Documentation
+@titlepage
+@sp 7
+@center @titlefont{FFplay Documentation}
+@sp 3
+@end titlepage
+
+
+@chapter Introduction
+
+@c man begin DESCRIPTION
+FFplay is a very simple and portable media player using the FFmpeg
+libraries and the SDL library. It is mostly used as a testbed for the
+various FFmpeg APIs.
+@c man end
+
+@chapter Invocation
+
+@section Syntax
+@example
+@c man begin SYNOPSIS
+ffplay [options] @file{input_file}
+@c man end
+@end example
+
+@c man begin OPTIONS
+
+@include fftools-common-opts.texi
+
+@section Main options
+
+@table @option
+@item -x @var{width}
+Force displayed width.
+@item -y @var{height}
+Force displayed height.
+@item -s @var{size}
+Set frame size (WxH or abbreviation), needed for videos which don't
+contain a header with the frame size like raw YUV.
+@item -an
+Disable audio.
+@item -vn
+Disable video.
+@item -ss @var{pos}
+Seek to a given position in seconds.
+@item -t @var{duration}
+play <duration> seconds of audio/video
+@item -bytes
+Seek by bytes.
+@item -nodisp
+Disable graphical display.
+@item -f @var{fmt}
+Force format.
+@item -window_title @var{title}
+Set window title (default is the input filename).
+@item -loop @var{number}
+Loops movie playback <number> times. 0 means forever.
+@end table
+
+@section Advanced options
+@table @option
+@item -pix_fmt @var{format}
+Set pixel format.
+@item -stats
+Show the stream duration, the codec parameters, the current position in
+the stream and the audio/video synchronisation drift.
+@item -debug
+Print specific debug info.
+@item -bug
+Work around bugs.
+@item -vismv
+Visualize motion vectors.
+@item -fast
+Non-spec-compliant optimizations.
+@item -genpts
+Generate pts.
+@item -rtp_tcp
+Force RTP/TCP protocol usage instead of RTP/UDP. It is only meaningful
+if you are streaming with the RTSP protocol.
+@item -sync @var{type}
+Set the master clock to audio (@code{type=audio}), video
+(@code{type=video}) or external (@code{type=ext}). Default is audio. The
+master clock is used to control audio-video synchronization. Most media
+players use audio as master clock, but in some cases (streaming or high
+quality broadcast) it is necessary to change that. This option is mainly
+used for debugging purposes.
+@item -threads @var{count}
+Set the thread count.
+@item -ast @var{audio_stream_number}
+Select the desired audio stream number, counting from 0. The number
+refers to the list of all the input audio streams. If it is greater
+than the number of audio streams minus one, then the last one is
+selected, if it is negative the audio playback is disabled.
+@item -vst @var{video_stream_number}
+Select the desired video stream number, counting from 0. The number
+refers to the list of all the input video streams. If it is greater
+than the number of video streams minus one, then the last one is
+selected, if it is negative the video playback is disabled.
+@item -sst @var{subtitle_stream_number}
+Select the desired subtitle stream number, counting from 0. The number
+refers to the list of all the input subtitle streams. If it is greater
+than the number of subtitle streams minus one, then the last one is
+selected, if it is negative the subtitle rendering is disabled.
+@end table
+
+@section While playing
+
+@table @key
+@item q, ESC
+Quit.
+
+@item f
+Toggle full screen.
+
+@item p, SPC
+Pause.
+
+@item a
+Cycle audio channel.
+
+@item v
+Cycle video channel.
+
+@item t
+Cycle subtitle channel.
+
+@item w
+Show audio waves.
+
+@item left/right
+Seek backward/forward 10 seconds.
+
+@item down/up
+Seek backward/forward 1 minute.
+
+@item mouse click
+Seek to percentage in file corresponding to fraction of width.
+
+@end table
+
+@c man end
+
+@ignore
+
+@setfilename ffplay
+@settitle FFplay media player
+
+@c man begin SEEALSO
+ffmpeg(1), ffserver(1) and the HTML documentation of @file{ffmpeg}.
+@c man end
+
+@c man begin AUTHOR
+Fabrice Bellard
+@c man end
+
+@end ignore
+
+@bye

doc/ffplay.texi

@@ -1,286 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle ffplay Documentation
-@titlepage
-@center @titlefont{ffplay Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Synopsis
-
-ffplay [@var{options}] [@file{input_file}]
-
-@chapter Description
-@c man begin DESCRIPTION
-
-FFplay is a very simple and portable media player using the FFmpeg
-libraries and the SDL library. It is mostly used as a testbed for the
-various FFmpeg APIs.
-@c man end
-
-@chapter Options
-@c man begin OPTIONS
-
-@include fftools-common-opts.texi
-
-@section Main options
-
-@table @option
-@item -x @var{width}
-Force displayed width.
-@item -y @var{height}
-Force displayed height.
-@item -s @var{size}
-Set frame size (WxH or abbreviation), needed for videos which do
-not contain a header with the frame size like raw YUV.  This option
-has been deprecated in favor of private options, try -video_size.
-@item -an
-Disable audio.
-@item -vn
-Disable video.
-@item -ss @var{pos}
-Seek to a given position in seconds.
-@item -t @var{duration}
-play <duration> seconds of audio/video
-@item -bytes
-Seek by bytes.
-@item -nodisp
-Disable graphical display.
-@item -f @var{fmt}
-Force format.
-@item -window_title @var{title}
-Set window title (default is the input filename).
-@item -loop @var{number}
-Loops movie playback <number> times. 0 means forever.
-@item -showmode @var{mode}
-Set the show mode to use.
-Available values for @var{mode} are:
-@table @samp
-@item 0, video
-show video
-@item 1, waves
-show audio waves
-@item 2, rdft
-show audio frequency band using RDFT ((Inverse) Real Discrete Fourier Transform)
-@end table
-
-Default value is "video", if video is not present or cannot be played
-"rdft" is automatically selected.
-
-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.
-Use the option "-filters" to show all the available filters (including
-sources and sinks).
-
-@item -i @var{input_file}
-Read @var{input_file}.
-@end table
-
-@section Advanced options
-@table @option
-@item -pix_fmt @var{format}
-Set pixel format.
-This option has been deprecated in favor of private options, try -pixel_format.
-
-@item -stats
-Print several playback statistics, in particular show the stream
-duration, the codec parameters, the current position in the stream and
-the audio/video synchronisation drift. It is on by default, to
-explicitly disable it you need to specify @code{-nostats}.
-
-@item -bug
-Work around bugs.
-@item -fast
-Non-spec-compliant optimizations.
-@item -genpts
-Generate pts.
-@item -rtp_tcp
-Force RTP/TCP protocol usage instead of RTP/UDP. It is only meaningful
-if you are streaming with the RTSP protocol.
-@item -sync @var{type}
-Set the master clock to audio (@code{type=audio}), video
-(@code{type=video}) or external (@code{type=ext}). Default is audio. The
-master clock is used to control audio-video synchronization. Most media
-players use audio as master clock, but in some cases (streaming or high
-quality broadcast) it is necessary to change that. This option is mainly
-used for debugging purposes.
-@item -threads @var{count}
-Set the thread count.
-@item -ast @var{audio_stream_number}
-Select the desired audio stream number, counting from 0. The number
-refers to the list of all the input audio streams. If it is greater
-than the number of audio streams minus one, then the last one is
-selected, if it is negative the audio playback is disabled.
-@item -vst @var{video_stream_number}
-Select the desired video stream number, counting from 0. The number
-refers to the list of all the input video streams. If it is greater
-than the number of video streams minus one, then the last one is
-selected, if it is negative the video playback is disabled.
-@item -sst @var{subtitle_stream_number}
-Select the desired subtitle stream number, counting from 0. The number
-refers to the list of all the input subtitle streams. If it is greater
-than the number of subtitle streams minus one, then the last one is
-selected, if it is negative the subtitle rendering is disabled.
-@item -autoexit
-Exit when video is done playing.
-@item -exitonkeydown
-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.
-@end table
-
-@section While playing
-
-@table @key
-@item q, ESC
-Quit.
-
-@item f
-Toggle full screen.
-
-@item p, SPC
-Pause.
-
-@item a
-Cycle audio channel in the curret program.
-
-@item v
-Cycle video channel.
-
-@item t
-Cycle subtitle channel in the current program.
-
-@item c
-Cycle program.
-
-@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.
-
-@item down/up
-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
-Seek to percentage in file corresponding to fraction of width.
-
-@end table
-
-@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 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
-
-@end ignore
-
-@bye

doc/ffprobe-doc.texi

@@ -0,0 +1,121 @@
+\input texinfo @c -*- texinfo -*-
+
+@settitle FFprobe Documentation
+@titlepage
+@sp 7
+@center @titlefont{FFprobe Documentation}
+@sp 3
+@end titlepage
+
+@chapter Introduction
+
+@c man begin DESCRIPTION
+
+FFprobe gathers information from multimedia streams and prints it in
+human- and machine-readable fashion.
+
+For example it can be used to check the format of the container used
+by a multimedia stream and the format and type of each media stream
+contained in it.
+
+If a filename is specified in input, ffprobe will try to open and
+probe the file content. If the file cannot be opened or recognized as
+a multimedia file, a positive exit code is returned.
+
+FFprobe may be employed both as a standalone application or in
+combination with a textual filter, which may perform more
+sophisticated processing, e.g. statistical processing or plotting.
+
+Options are used to list some of the formats supported by ffprobe or
+for specifying which information to display, and for setting how
+ffprobe will show it.
+
+FFprobe output is designed to be easily parsable by a textual filter,
+and consists of one or more sections of the form:
+@example
+[SECTION]
+key1=val1
+...
+keyN=valN
+[/SECTION]
+@end example
+
+Metadata tags stored in the container or in the streams are recognized
+and printed in the corresponding ``FORMAT'' or ``STREAM'' section, and
+are prefixed by the string ``TAG:''.
+
+@c man end
+
+@chapter Invocation
+
+@section Syntax
+
+The generic syntax is:
+
+@example
+@c man begin SYNOPSIS
+ffprobe [options] [@file{input_file}]
+@c man end
+@end example
+
+@c man begin OPTIONS
+
+@include fftools-common-opts.texi
+
+@section Main options
+
+@table @option
+
+@item -convert_tags
+Convert the tag names in the format container to the generic FFmpeg tag names.
+
+@item -f @var{format}
+Force format to use.
+
+@item -unit
+Show the unit of the displayed values.
+
+@item -prefix
+Show a SI prefixes of the displayed values.
+Unless ``-byte_binary_prefix'' option is used all the prefix
+are decimal.
+
+@item -byte_binary_prefix
+Force the use of binary prefixes for byte values.
+
+@item -sexagesimal
+Use sexagesimal format HH:MM:SS.MICROSECONDS for time values.
+
+@item -pretty
+Prettify the format of the displayed values, it corresponds to the
+options ``-unit -prefix -byte_binary_prefix -sexagesimal''.
+
+@item -show_format
+Show information about the container format of the input multimedia
+stream.
+
+All the container format information is printed within a section with
+name ``FORMAT''.
+
+@item -show_streams
+Show information about each media stream contained in the input
+multimedia stream.
+
+Each media stream information is printed within a dedicated section
+with name ``STREAM''.
+
+@end table
+@c man end
+
+@ignore
+
+@setfilename ffprobe
+@settitle FFprobe media prober
+
+@c man begin SEEALSO
+ffmpeg(1), ffplay(1), ffserver(1)
+@c man end
+
+@end ignore
+
+@bye

doc/ffprobe.texi

@@ -1,671 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle ffprobe Documentation
-@titlepage
-@center @titlefont{ffprobe Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Synopsis
-
-ffprobe [@var{options}] [@file{input_file}]
-
-@chapter Description
-@c man begin DESCRIPTION
-
-ffprobe gathers information from multimedia streams and prints it in
-human- and machine-readable fashion.
-
-For example it can be used to check the format of the container used
-by a multimedia stream and the format and type of each media stream
-contained in it.
-
-If a filename is specified in input, ffprobe will try to open and
-probe the file content. If the file cannot be opened or recognized as
-a multimedia file, a positive exit code is returned.
-
-ffprobe may be employed both as a standalone application or in
-combination with a textual filter, which may perform more
-sophisticated processing, e.g. statistical processing or plotting.
-
-Options are used to list some of the formats supported by ffprobe or
-for specifying which information to display, and for setting how
-ffprobe will show it.
-
-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.
-
-@c man end
-
-@chapter Options
-@c man begin OPTIONS
-
-@include fftools-common-opts.texi
-
-@section Main options
-
-@table @option
-
-@item -f @var{format}
-Force format to use.
-
-@item -unit
-Show the unit of the displayed values.
-
-@item -prefix
-Use SI prefixes for the displayed values.
-Unless the "-byte_binary_prefix" option is used all the prefixes
-are decimal.
-
-@item -byte_binary_prefix
-Force the use of binary prefixes for byte values.
-
-@item -sexagesimal
-Use sexagesimal format HH:MM:SS.MICROSECONDS for time values.
-
-@item -pretty
-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}]
-Set the output printing format.
-
-@var{writer_name} specifies the name of the writer, and
-@var{writer_options} specifies the options to be passed to the writer.
-
-For example for printing the output in JSON format, specify:
-@example
--print_format json
-@end example
-
-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.
-
-The error information is printed within a section with name "ERROR".
-
-@item -show_format
-Show information about the container format of the input multimedia
-stream.
-
-All the container format information is printed within a section with
-name "FORMAT".
-
-@item -show_format_entry @var{name}
-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.
-
-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.
-
-The information for each single frame is printed within a dedicated
-section with name "FRAME" or "SUBTITLE".
-
-@item -show_streams
-Show information about each media stream contained in the input
-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.
-
-@item -count_packets
-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.
-This option is enabled by default, but you may need to disable it
-for specific uses, for example when creating XSD-compliant XML output.
-
-@item -show_program_version
-Show information related to program version.
-
-Version information is printed within a section with name
-"PROGRAM_VERSION".
-
-@item -show_library_versions
-Show information related to library versions.
-
-Version information for each library is printed within a section with
-name "LIBRARY_VERSION".
-
-@item -show_versions
-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}.
-
-@end table
-@c man end
-
-@chapter Writers
-@c man begin WRITERS
-
-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 description of the currently available writers follows.
-
-@section default
-Default format.
-
-Print each section in the form:
-@example
-[SECTION]
-key1=val1
-...
-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:".
-
-A description of the accepted options follows.
-
-@table @option
-
-@item nokey, nk
-If set to 1 specify not to print the key of each field. Default value
-is 0.
-
-@item noprint_wrappers, nw
-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.
-
-Each section is printed on a single line.
-If no option is specifid, the output has the form:
-@example
-section|key1=val1| ... |keyN=valN
-@end example
-
-Metadata tags are printed in the corresponding "format" or "stream"
-section. A metadata tag key, if printed, is prefixed by the string
-"tag:".
-
-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).
-
-@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).
-
-@item escape, e
-Set the escape mode to use, default to "c" ("csv" for the @code{csv}
-writer).
-
-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
-escaping, so that a newline is converted to the sequence "\n", a
-carriage return to "\r", '\' to "\\" and the separator @var{SEP} is
-converted to "\@var{SEP}".
-
-@item csv
-Perform CSV-like escaping, as described in RFC4180.  Strings
-containing a newline ('\n'), a carriage return ('\r'), a double quote
-('"'), or @var{SEP} are enclosed in double-quotes.
-
-@item none
-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.
-
-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
-
-@section json
-JSON based format.
-
-Each section is printed using JSON notation.
-
-The description of the accepted options follows.
-
-@table @option
-
-@item compact, c
-If set to 1 enable compact output, that is each section will be
-printed on a single line. Default value is 0.
-@end table
-
-For more information about JSON, see @url{http://www.json.org/}.
-
-@section xml
-XML based format.
-
-The XML output is described in the XML schema description file
-@file{ffprobe.xsd} installed in the FFmpeg datadir.
-
-An updated version of the schema can be retrieved at the url
-@url{http://www.ffmpeg.org/schema/ffprobe.xsd}, which redirects to the
-latest schema committed into the FFmpeg development source code tree.
-
-Note that the output issued will be compliant to the
-@file{ffprobe.xsd} schema only when no special global output options
-(@option{unit}, @option{prefix}, @option{byte_binary_prefix},
-@option{sexagesimal} etc.) are specified.
-
-The description of the accepted options follows.
-
-@table @option
-
-@item fully_qualified, q
-If set to 1 specify if the output should be fully qualified. Default
-value is 0.
-This is required for generating an XML file which can be validated
-through an XSD file.
-
-@item xsd_compliant, x
-If set to 1 perform more checks for ensuring that the output is XSD
-compliant. Default value is 0.
-This option automatically sets @option{fully_qualified} to 1.
-@end table
-
-For more information about the XML format, see
-@url{http://www.w3.org/XML/}.
-@c man end WRITERS
-
-@chapter Timecode
-@c man begin TIMECODE
-
-@command{ffprobe} supports Timecode extraction:
-
-@itemize
-
-@item
-MPEG1/2 timecode is extracted from the GOP, and is available in the video
-stream details (@option{-show_streams}, see @var{timecode}).
-
-@item
-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
-(@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 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
-
-@ignore
-
-@setfilename ffprobe
-@settitle ffprobe media prober
-
-@end ignore
-
-@bye

doc/ffprobe.xsd

@@ -1,260 +0,0 @@
-<?xml version="1.0" encoding="UTF-8"?>
-
-<xsd:schema xmlns:xsd="http://www.w3.org/2001/XMLSchema"
-    targetNamespace="http://www.ffmpeg.org/schema/ffprobe"
-    xmlns:ffprobe="http://www.ffmpeg.org/schema/ffprobe">
-
-    <xsd:element name="ffprobe" type="ffprobe:ffprobeType"/>
-
-    <xsd:complexType name="ffprobeType">
-        <xsd:sequence>
-            <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" />
-            <xsd:element name="library_versions" type="ffprobe:libraryVersionsType" minOccurs="0" maxOccurs="1" />
-        </xsd:sequence>
-    </xsd:complexType>
-
-    <xsd:complexType name="packetsType">
-        <xsd:sequence>
-            <xsd:element name="packet" type="ffprobe:packetType" minOccurs="0" maxOccurs="unbounded"/>
-        </xsd:sequence>
-    </xsd:complexType>
-
-    <xsd:complexType name="framesType">
-        <xsd:sequence>
-            <xsd:choice minOccurs="0" maxOccurs="unbounded">
-                <xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
-                <xsd:element name="subtitle" type="ffprobe:subtitleType" minOccurs="0" maxOccurs="unbounded"/>
-            </xsd:choice>
-        </xsd:sequence>
-    </xsd:complexType>
-
-    <xsd:complexType name="packetType">
-      <xsd:attribute name="codec_type"    type="xsd:string" use="required" />
-      <xsd:attribute name="stream_index"  type="xsd:int" use="required" />
-      <xsd:attribute name="pts"           type="xsd:long"  />
-      <xsd:attribute name="pts_time"      type="xsd:float" />
-      <xsd:attribute name="dts"           type="xsd:long"  />
-      <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">
-      <xsd:attribute name="media_type"    type="xsd:string" use="required"/>
-      <xsd:attribute name="key_frame"     type="xsd:int"    use="required"/>
-      <xsd:attribute name="pts"           type="xsd:long" />
-      <xsd:attribute name="pts_time"      type="xsd:float"/>
-      <xsd:attribute name="pkt_pts"       type="xsd:long" />
-      <xsd:attribute name="pkt_pts_time"  type="xsd:float"/>
-      <xsd:attribute name="pkt_dts"       type="xsd:long" />
-      <xsd:attribute name="pkt_dts_time"  type="xsd:float"/>
-      <xsd:attribute name="best_effort_timestamp"      type="xsd:long" />
-      <xsd:attribute name="best_effort_timestamp_time" type="xsd:float" />
-      <xsd:attribute name="pkt_duration"  type="xsd:long" />
-      <xsd:attribute name="pkt_duration_time" type="xsd:float"/>
-      <xsd:attribute name="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"  />
-      <xsd:attribute name="height"                 type="xsd:long"  />
-      <xsd:attribute name="pix_fmt"                type="xsd:string"/>
-      <xsd:attribute name="sample_aspect_ratio"    type="xsd:string"/>
-      <xsd:attribute name="pict_type"              type="xsd:string"/>
-      <xsd:attribute name="coded_picture_number"   type="xsd:long"  />
-      <xsd:attribute name="display_picture_number" type="xsd:long"  />
-      <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:complexType>
-
-    <xsd:complexType name="streamsType">
-        <xsd:sequence>
-            <xsd:element name="stream" type="ffprobe:streamType" minOccurs="0" maxOccurs="unbounded"/>
-        </xsd:sequence>
-    </xsd:complexType>
-
-    <xsd:complexType name="programsType">
-        <xsd:sequence>
-            <xsd:element name="program" type="ffprobe:programType" minOccurs="0" maxOccurs="unbounded"/>
-        </xsd:sequence>
-    </xsd:complexType>
-
-    <xsd:complexType name="streamDispositionType">
-      <xsd:attribute name="default"          type="xsd:int" use="required" />
-      <xsd:attribute name="dub"              type="xsd:int" use="required" />
-      <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"/>
-      <xsd:attribute name="height"               type="xsd:int"/>
-      <xsd:attribute name="has_b_frames"         type="xsd:int"/>
-      <xsd:attribute name="sample_aspect_ratio"  type="xsd:string"/>
-      <xsd:attribute name="display_aspect_ratio" type="xsd:string"/>
-      <xsd:attribute name="pix_fmt"              type="xsd:string"/>
-      <xsd:attribute name="level"                type="xsd:int"/>
-      <xsd:attribute name="timecode"             type="xsd:string"/>
-
-      <!-- audio attributes -->
-      <xsd:attribute name="sample_fmt"       type="xsd:string"/>
-      <xsd:attribute name="sample_rate"      type="xsd:int"/>
-      <xsd:attribute name="channels"         type="xsd:int"/>
-      <xsd:attribute name="channel_layout"   type="xsd:string"/>
-      <xsd:attribute name="bits_per_sample"  type="xsd:int"/>
-
-      <xsd:attribute name="id"               type="xsd:string"/>
-      <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"/>
-      <xsd:attribute name="nb_read_frames"   type="xsd:int"/>
-      <xsd:attribute name="nb_read_packets"  type="xsd:int"/>
-    </xsd:complexType>
-
-    <xsd:complexType name="programType">
-      <xsd:sequence>
-        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
-        <xsd:element name="streams" type="ffprobe:streamsType" minOccurs="0" maxOccurs="1"/>
-      </xsd:sequence>
-
-      <xsd:attribute name="program_id"           type="xsd:int"    use="required"/>
-      <xsd:attribute name="program_num"          type="xsd:int"    use="required"/>
-      <xsd:attribute name="nb_streams"           type="xsd:int"    use="required"/>
-      <xsd:attribute name="start_time"           type="xsd:float"/>
-      <xsd:attribute name="start_pts"            type="xsd:long"/>
-      <xsd:attribute name="end_time"             type="xsd:float"/>
-      <xsd:attribute name="end_pts"              type="xsd:long"/>
-      <xsd:attribute name="pmt_pid"              type="xsd:int"    use="required"/>
-      <xsd:attribute name="pcr_pid"              type="xsd:int"    use="required"/>
-    </xsd:complexType>
-
-    <xsd:complexType name="formatType">
-      <xsd:sequence>
-        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
-      </xsd:sequence>
-
-      <xsd:attribute name="filename"         type="xsd:string" use="required"/>
-      <xsd:attribute name="nb_streams"       type="xsd:int"    use="required"/>
-      <xsd:attribute name="nb_programs"      type="xsd:int"    use="required"/>
-      <xsd:attribute name="format_name"      type="xsd:string" use="required"/>
-      <xsd:attribute name="format_long_name" type="xsd:string"/>
-      <xsd:attribute name="start_time"       type="xsd:float"/>
-      <xsd:attribute name="duration"         type="xsd:float"/>
-      <xsd:attribute name="size"             type="xsd:long"/>
-      <xsd:attribute name="bit_rate"         type="xsd:long"/>
-      <xsd:attribute name="probe_score"      type="xsd:int"/>
-    </xsd:complexType>
-
-    <xsd:complexType name="tagType">
-      <xsd:attribute name="key"   type="xsd:string" use="required"/>
-      <xsd:attribute name="value" type="xsd:string" use="required"/>
-    </xsd:complexType>
-
-    <xsd:complexType name="errorType">
-      <xsd:attribute name="code"   type="xsd:int"    use="required"/>
-      <xsd:attribute name="string" type="xsd:string" use="required"/>
-    </xsd:complexType>
-
-    <xsd:complexType name="programVersionType">
-      <xsd:attribute name="version"          type="xsd:string" use="required"/>
-      <xsd:attribute name="copyright"        type="xsd:string" use="required"/>
-      <xsd:attribute name="build_date"       type="xsd:string" use="required"/>
-      <xsd:attribute name="build_time"       type="xsd:string" use="required"/>
-      <xsd:attribute name="compiler_type"    type="xsd:string" use="required"/>
-      <xsd:attribute name="compiler_version" type="xsd:string" use="required"/>
-      <xsd:attribute name="configuration"    type="xsd:string" use="required"/>
-    </xsd:complexType>
-
-    <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">
-        <xsd:sequence>
-          <xsd:element name="library_version" type="ffprobe:libraryVersionType" minOccurs="0" maxOccurs="unbounded"/>
-        </xsd:sequence>
-    </xsd:complexType>
-</xsd:schema>

doc/ffserver-doc.texi

@@ -0,0 +1,274 @@
+\input texinfo @c -*- texinfo -*-
+
+@settitle FFserver Documentation
+@titlepage
+@sp 7
+@center @titlefont{FFserver Documentation}
+@sp 3
+@end titlepage
+
+
+@chapter Introduction
+
+@c man begin DESCRIPTION
+FFserver is a streaming server for both audio and video. It supports
+several live feeds, streaming from files and time shifting on live feeds
+(you can seek to positions in the past on each live feed, provided you
+specify a big enough feed storage in ffserver.conf).
+
+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.
+
+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-doc.html} for more
+information.
+
+@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 <Stream> section in the configuration
+file.
+
+@section Status stream
+
+FFserver supports an HTTP interface which exposes the current status
+of the server.
+
+Simply point your browser to the address of the special status stream
+specified in the configuration file.
+
+For example if you have:
+@example
+<Stream status.html>
+Format status
+
+# Only allow local people to get the status
+ACL allow localhost
+ACL allow 192.168.0.0 192.168.255.255
+</Stream>
+@end example
+
+then the server will post a page with the status information when
+the special stream @file{status.html} is requested.
+
+@section 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:
+
+@example
+./ffserver -f doc/ffserver.conf &
+./ffmpeg -i INPUTFILE http://localhost:8090/feed1.ffm
+@end example
+
+At this point you should be able to go to your Windows machine and fire up
+Windows Media Player (WMP). Go to Open URL and enter
+
+@example
+    http://<linuxbox>:8090/test.asf
+@end example
+
+You should (after a short delay) see video and hear audio.
+
+WARNING: trying to stream test1.mpg doesn't work with WMP as it tries to
+transfer the entire file before starting to play.
+The same is true of AVI files.
+
+@section What happens next?
+
+You should edit the ffserver.conf file to suit your needs (in terms of
+frame rates etc). Then install ffserver and ffmpeg, write a script to start
+them up, and off you go.
+
+@section Troubleshooting
+
+@subsection I don't hear any audio, but video is fine.
+
+Maybe you didn't install LAME, or got your ./configure statement wrong. Check
+the ffmpeg output to see if a line referring to MP3 is present. If not, then
+your configuration was incorrect. If it is, then maybe your wiring is not
+set up correctly. Maybe the sound card is not getting data from the right
+input source. Maybe you have a really awful audio interface (like I do)
+that only captures in stereo and also requires that one channel be flipped.
+If you are one of these people, then export 'AUDIO_FLIP_LEFT=1' before
+starting ffmpeg.
+
+@subsection The audio and video loose 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.
+However, there are a number of caveats, including the fact that the
+ffserver parameters must match the original parameters used to record the
+file. If they do not, then ffserver deletes the file before recording into it.
+(Now that I write this, it seems broken).
+
+You can fiddle with many of the codec choices and encoding parameters, and
+there are a bunch more parameters that you cannot control. Post a message
+to the mailing list if there are some 'must have' parameters. Look in
+ffserver.conf for a list of the currently available controls.
+
+It will automatically generate the ASX or RAM files that are often used
+in browsers. These files are actually redirections to the underlying ASF
+or RM file. The reason for this is that the browser often fetches the
+entire file before starting up the external viewer. The redirection files
+are very small and can be transferred quickly. [The stream itself is
+often 'infinite' and thus the browser tries to download it and never
+finishes.]
+
+@section Tips
+
+* When you connect to a live stream, most players (WMP, RA, etc) want to
+buffer a certain number of seconds of material so that they can display the
+signal continuously. However, ffserver (by default) starts sending data
+in realtime. This means that there is a pause of a few seconds while the
+buffering is being done by the player. The good news is that this can be
+cured by adding a '?buffer=5' to the end of the URL. This means that the
+stream should start 5 seconds in the past -- and so the first 5 seconds
+of the stream are sent as fast as the network will allow. It will then
+slow down to real time. This noticeably improves the startup experience.
+
+You can also add a 'Preroll 15' statement into the ffserver.conf that will
+add the 15 second prebuffering on all requests that do not otherwise
+specify a time. In addition, ffserver will skip frames until a key_frame
+is found. This further reduces the startup delay by not transferring data
+that will be discarded.
+
+* 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
+grabbed is marginally less than the number that ought to be grabbed. This
+means that the timestamp in the encoded data stream gets behind realtime.
+This means that if you say 'Preroll 10', then when the stream gets 10
+or more seconds behind, there is no Preroll left.
+
+Fixing this requires a change in the internals of how timestamps are
+handled.
+
+@section Does the @code{?date=} stuff work.
+
+Yes (subject to the limitation outlined above). Also note that whenever you
+start ffserver, it deletes the ffm file (if any parameters have changed),
+thus wiping out what you had recorded before.
+
+The format of the @code{?date=xxxxxx} is fairly flexible. You should use one
+of the following formats (the 'T' is literal):
+
+@example
+* YYYY-MM-DDTHH:MM:SS     (localtime)
+* YYYY-MM-DDTHH:MM:SSZ    (UTC)
+@end example
+
+You can omit the YYYY-MM-DD, and then it refers to the current day. However
+note that @samp{?date=16:00:00} refers to 16:00 on the current day -- this
+may be in the future and so is unlikely to be useful.
+
+You use this by adding the ?date= to the end of the URL for the stream.
+For example:   @samp{http://localhost:8080/test.asf?date=2002-07-26T23:05:00}.
+@c man end
+
+@chapter Invocation
+@section Syntax
+@example
+@c man begin SYNOPSIS
+ffserver [options]
+@c man end
+@end example
+
+@section Options
+@c man begin OPTIONS
+
+@include fftools-common-opts.texi
+
+@section Main options
+
+@table @option
+@item -f @var{configfile}
+Use @file{configfile} instead of @file{/etc/ffserver.conf}.
+@item -n
+Enable no-launch mode. This option disables all the Launch directives
+within the various <Stream> sections. FFserver will not launch any
+ffmpeg instance, so you will have to launch them manually.
+@item -d
+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
+@c man end
+
+@ignore
+
+@setfilename ffserver
+@settitle FFserver video server
+
+@c man begin SEEALSO
+ffmpeg(1), ffplay(1), the @file{ffmpeg/doc/ffserver.conf} example and
+the HTML documentation of @file{ffmpeg}.
+@c man end
+
+@c man begin AUTHOR
+Fabrice Bellard
+@c man end
+
+@end ignore
+
+@bye

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>
 
 
@@ -302,27 +306,6 @@ StartSendOnKey
 #</Stream>
 
 
-# Transcode an incoming live feed to another live feed,
-# using libx264 and video presets
-
-#<Stream live.h264>
-#Format rtp
-#Feed feed1.ffm
-#VideoCodec libx264
-#VideoFrameRate 24
-#VideoBitRate 100
-#VideoSize 480x272
-#AVPresetVideo default
-#AVPresetVideo baseline
-#AVOptionVideo flags +global_header
-#
-#AudioCodec libfaac
-#AudioBitRate 32
-#AudioChannels 2
-#AudioSampleRate 22050
-#AVOptionAudio flags +global_header
-#</Stream>
-
 ##################################################################
 # SDP/multicast examples
 #
@@ -369,3 +352,5 @@ ACL allow 192.168.0.0 192.168.255.255
 <Redirect index.html>
 URL http://www.ffmpeg.org/
 </Redirect>
+
+

doc/ffserver.texi

@@ -1,900 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle ffserver Documentation
-@titlepage
-@center @titlefont{ffserver Documentation}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Synopsis
-
-ffserver [@var{options}]
-
-@chapter Description
-@c man begin DESCRIPTION
-
-@command{ffserver} is a streaming server for both audio and video.
-It supports several live feeds, streaming from files and time shifting
-on live feeds. You can seek to positions in the past on each live
-feed, provided you specify a big enough feed storage.
-
-@command{ffserver} is configured through a configuration file, which
-is read at startup. If not explicitly specified, it will read from
-@file{/etc/ffserver.conf}.
-
-@command{ffserver} receives prerecorded files or FFM streams from some
-@command{ffmpeg} instance as input, then streams them over
-RTP/RTSP/HTTP.
-
-An @command{ffserver} instance will listen on some port as specified
-in the configuration file. You can launch one or more instances of
-@command{ffmpeg} and send one or more FFM streams to the port where
-ffserver is expecting to receive them. Alternately, you can make
-@command{ffserver} launch such @command{ffmpeg} instances at startup.
-
-Input streams are called feeds, and each one is specified by a
-@code{<Feed>} section in the configuration file.
-
-For each feed you can have different output streams in various
-formats, each one specified by a @code{<Stream>} section in the
-configuration file.
-
-@chapter Detailed description
-
-@command{ffserver} works by forwarding streams encoded by
-@command{ffmpeg}, or pre-recorded streams which are read from disk.
-
-Precisely, @command{ffserver} acts as an HTTP server, accepting POST
-requests from @command{ffmpeg} to acquire the stream to publish, and
-serving RTSP clients or HTTP clients GET requests with the stream
-media content.
-
-A feed is an @ref{FFM} stream created by @command{ffmpeg}, and sent to
-a port where @command{ffserver} is listening.
-
-Each feed is identified by a unique name, corresponding to the name
-of the resource published on @command{ffserver}, and is configured by
-a dedicated @code{Feed} section in the configuration file.
-
-The feed publish URL is given by:
-@example
-http://@var{ffserver_ip_address}:@var{http_port}/@var{feed_name}
-@end example
-
-where @var{ffserver_ip_address} is the IP address of the machine where
-@command{ffserver} is installed, @var{http_port} is the port number of
-the HTTP server (configured through the @option{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 one to override the encoding parameters set by the server.
-
-Multiple streams can be connected to the same feed.
-
-For example, you can have a situation described by the following
-graph:
-@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.
-
-@section Status stream
-
-@command{ffserver} supports an HTTP interface which exposes the
-current status of the server.
-
-Simply point your browser to the address of the special status stream
-specified in the configuration file.
-
-For example if you have:
-@example
-<Stream status.html>
-Format status
-
-# Only allow local people to get the status
-ACL allow localhost
-ACL allow 192.168.0.0 192.168.255.255
-</Stream>
-@end example
-
-then the server will post a page with the status information when
-the special stream @file{status.html} is requested.
-
-@section How do I make it work?
-
-As a simple test, just run the following two command lines where INPUTFILE
-is some file which you can decode with ffmpeg:
-
-@example
-ffserver -f doc/ffserver.conf &
-ffmpeg -i INPUTFILE http://localhost:8090/feed1.ffm
-@end example
-
-At this point you should be able to go to your Windows machine and fire up
-Windows Media Player (WMP). Go to Open URL and enter
-
-@example
-    http://<linuxbox>:8090/test.asf
-@end example
-
-You should (after a short delay) see video and hear audio.
-
-WARNING: trying to stream test1.mpg doesn't work with WMP as it tries to
-transfer the entire file before starting to play.
-The same is true of AVI files.
-
-@section What happens next?
-
-You should edit the ffserver.conf file to suit your needs (in terms of
-frame rates etc). Then install ffserver and ffmpeg, write a script to start
-them up, and off you go.
-
-@section What else can it do?
-
-You can replay video from .ffm files that was recorded earlier.
-However, there are a number of caveats, including the fact that the
-ffserver parameters must match the original parameters used to record the
-file. If they do not, then ffserver deletes the file before recording into it.
-(Now that I write this, it seems broken).
-
-You can fiddle with many of the codec choices and encoding parameters, and
-there are a bunch more parameters that you cannot control. Post a message
-to the mailing list if there are some 'must have' parameters. Look in
-ffserver.conf for a list of the currently available controls.
-
-It will automatically generate the ASX or RAM files that are often used
-in browsers. These files are actually redirections to the underlying ASF
-or RM file. The reason for this is that the browser often fetches the
-entire file before starting up the external viewer. The redirection files
-are very small and can be transferred quickly. [The stream itself is
-often 'infinite' and thus the browser tries to download it and never
-finishes.]
-
-@section Tips
-
-* When you connect to a live stream, most players (WMP, RA, etc) want to
-buffer a certain number of seconds of material so that they can display the
-signal continuously. However, ffserver (by default) starts sending data
-in realtime. This means that there is a pause of a few seconds while the
-buffering is being done by the player. The good news is that this can be
-cured by adding a '?buffer=5' to the end of the URL. This means that the
-stream should start 5 seconds in the past -- and so the first 5 seconds
-of the stream are sent as fast as the network will allow. It will then
-slow down to real time. This noticeably improves the startup experience.
-
-You can also add a 'Preroll 15' statement into the ffserver.conf that will
-add the 15 second prebuffering on all requests that do not otherwise
-specify a time. In addition, ffserver will skip frames until a key_frame
-is found. This further reduces the startup delay by not transferring data
-that will be discarded.
-
-@section Why does the ?buffer / Preroll stop working after a time?
-
-It turns out that (on my machine at least) the number of frames successfully
-grabbed is marginally less than the number that ought to be grabbed. This
-means that the timestamp in the encoded data stream gets behind realtime.
-This means that if you say 'Preroll 10', then when the stream gets 10
-or more seconds behind, there is no Preroll left.
-
-Fixing this requires a change in the internals of how timestamps are
-handled.
-
-@section Does the @code{?date=} stuff work.
-
-Yes (subject to the limitation outlined above). Also note that whenever you
-start ffserver, it deletes the ffm file (if any parameters have changed),
-thus wiping out what you had recorded before.
-
-The format of the @code{?date=xxxxxx} is fairly flexible. You should use one
-of the following formats (the 'T' is literal):
-
-@example
-* YYYY-MM-DDTHH:MM:SS     (localtime)
-* YYYY-MM-DDTHH:MM:SSZ    (UTC)
-@end example
-
-You can omit the YYYY-MM-DD, and then it refers to the current day. However
-note that @samp{?date=16:00:00} refers to 16:00 on the current day -- this
-may be in the future and so is unlikely to be useful.
-
-You use this by adding the ?date= to the end of the URL for the stream.
-For example:   @samp{http://localhost:8080/test.asf?date=2002-07-26T23:05:00}.
-@c man end
-
-@chapter Options
-@c man begin OPTIONS
-
-@include fftools-common-opts.texi
-
-@section Main options
-
-@table @option
-@item -f @var{configfile}
-Read configuration file @file{configfile}. If not specified it will
-read by default from @file{/etc/ffserver.conf}.
-
-@item -n
-Enable no-launch mode. This option disables all the @code{Launch}
-directives within the various @code{<Feed>} sections. Since
-@command{ffserver} will not launch any @command{ffmpeg} instances, you
-will have to launch them manually.
-
-@item -d
-Enable debug mode. This option increases log verbosity, and directs
-log messages to stdout. When specified, the @option{CustomLog} option
-is ignored.
-@end table
-
-@chapter Configuration file syntax
-
-@command{ffserver} reads a configuration file containing global
-options and settings for each stream and feed.
-
-The configuration file consists of global options and dedicated
-sections, which must be introduced by "<@var{SECTION_NAME}
-@var{ARGS}>" on a separate line and must be terminated by a line in
-the form "</@var{SECTION_NAME}>". @var{ARGS} is optional.
-
-Currently the following sections are recognized: @samp{Feed},
-@samp{Stream}, @samp{Redirect}.
-
-A line starting with @code{#} is ignored and treated as a comment.
-
-Name of options and sections are case-insensitive.
-
-@section ACL syntax
-An ACL (Access Control List) specifies the address which are allowed
-to access a given stream, or to write a given feed.
-
-It accepts the 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
-
-@end ignore
-
-@bye

doc/fftools-common-opts.texi

@@ -1,53 +1,3 @@
-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.
@@ -57,40 +7,8 @@ These options are shared amongst the ff* tools.
 @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 -h, -?, -help, --help
+Show help.
 
 @item -version
 Show version.
@@ -98,17 +16,32 @@ 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 all codecs known to libavcodec.
+Show available codecs.
 
-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.
+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.
@@ -122,182 +55,18 @@ 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}
+@item -loglevel @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.

doc/filter_design.txt

@@ -1,270 +0,0 @@
-Filter design
-=============
-
-This document explains guidelines that should be observed (or ignored with
-good reason) when writing filters for libavfilter.
-
-In this document, the word “frame” indicates either a video frame or a group
-of audio samples, as stored in an AVFilterBuffer structure.
-
-
-Format negotiation
-==================
-
-  The query_formats method should set, for each input and each output links,
-  the list of 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.
-
-  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
-  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
-  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.
-
-
-Frame scheduling
-================
-
-  The purpose of these rules is to ensure that frames flow in the filter
-  graph without getting stuck and accumulating somewhere.
-
-  Simple filters that output one frame for each input frame should not have
-  to worry about it.
-
-  filter_frame
-  ------------
-
-    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.
-
-    If the input frame is enough to produce output, then the filter should
-    push the output frames on the output link immediately.
-
-    As an exception to the previous rule, if the input frame is enough to
-    produce several output frames, then the filter needs output only at
-    least one per link. The additional frames can be left buffered in the
-    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
-    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.)
-
-    If the input frame is not enough to produce output, the filter must not
-    call request_frame to get more. It must just process the frame or queue
-    it. The task of requesting more frames is left to the filter's
-    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
-    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.
-
-  request_frame
-  -------------
-
-    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 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.
-
-    Return values:
-    if request_frame could produce a frame, it should return 0;
-    if it could not for temporary reasons, it should return AVERROR(EAGAIN);
-    if it could not because there are no more frames, it should return
-    AVERROR_EOF.
-
-    The typical implementation of request_frame for a filter with several
-    inputs will look like that:
-
-        if (frames_queued) {
-            push_one_frame();
-            return 0;
-        }
-        while (!frame_pushed) {
-            input = input_where_a_frame_is_most_needed();
-            ret = ff_request_frame(input);
-            if (ret == AVERROR_EOF) {
-                process_eof_on_input();
-            } else if (ret < 0) {
-                return ret;
-            }
-        }
-        return 0;
-
-    Note that, except for filters that can have queued frames, request_frame
-    does not push frames: it requests them to its input, and as a reaction,
-    the filter_frame method 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.

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

doc/git-howto.texi

@@ -1,415 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle Using git to develop FFmpeg
-
-@titlepage
-@center @titlefont{Using git to develop FFmpeg}
-@end titlepage
-
-@top
-
-@contents
-
-@chapter Introduction
-
-This document aims in giving some quick references on a set of useful git
-commands. You should always use the extensive and detailed documentation
-provided directly by git:
-
-@example
-git --help
-man git
-@end example
-
-shows you the available subcommands,
-
-@example
-git <command> --help
-man git-<command>
-@end example
-
-shows information about the subcommand <command>.
-
-Additional information could be found on the
-@url{http://gitref.org, Git Reference} website
-
-For more information about the Git project, visit the
-
-@url{http://git-scm.com/, Git website}
-
-Consult these resources whenever you have problems, they are quite exhaustive.
-
-What follows now is a basic introduction to Git and some FFmpeg-specific
-guidelines to ease the contribution to the project
-
-@chapter Basics Usage
-
-@section Get GIT
-
-You can get git from @url{http://git-scm.com/}
-Most distribution and operating system provide a package for it.
-
-
-@section Cloning the source tree
-
-@example
-git clone git://source.ffmpeg.org/ffmpeg <target>
-@end example
-
-This will put the FFmpeg sources into the directory @var{<target>}.
-
-@example
-git clone git@@source.ffmpeg.org:ffmpeg <target>
-@end example
-
-This will put the FFmpeg sources into the directory @var{<target>} and let
-you push back your changes to the remote repository.
-
-Make sure that you do not have Windows line endings in your checkouts,
-otherwise you may experience spurious compilation failures. One way to
-achieve this is to run
-
-@example
-git config --global core.autocrlf false
-@end example
-
-
-@section Updating the source tree to the latest revision
-
-@example
-git pull (--rebase)
-@end example
-
-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.
-
-@float IMPORTANT
-@command{--rebase} (see below) is recommended.
-@end float
-
-@section Rebasing your local branches
-
-@example
-git pull --rebase
-@end example
-
-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.
-
-
-@section Adding/removing files/directories
-
-@example
-git add [-A] <filename/dirname>
-git rm [-r] <filename/dirname>
-@end example
-
-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.
-
-
-@section Showing modifications
-
-@example
-git diff <filename(s)>
-@end example
-
-will show all local modifications in your working directory as unified diff.
-
-
-@section Inspecting the changelog
-
-@example
-git log <filename(s)>
-@end example
-
-You may also use the graphical tools like gitview or gitk or the web
-interface available at http://source.ffmpeg.org/
-
-@section Checking source tree status
-
-@example
-git status
-@end example
-
-detects all the changes you made and lists what actions will be taken in case
-of a commit (additions, modifications, deletions, etc.).
-
-
-@section Committing
-
-@example
-git diff --check
-@end example
-
-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
-
-@example
-git diff -w -b <filename(s)>
-@end example
-
-Also check the output of
-
-@example
-git status
-@end example
-
-to make sure you don't have untracked files or deletions.
-
-@example
-git add [-i|-p|-A] <filenames/dirnames>
-@end example
-
-Make sure you have told git your name and email address
-
-@example
-git config --global user.name "My Name"
-git config --global user.email my@@email.invalid
-@end example
-
-Use @var{--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.
-
-
-@example
-git commit
-@end example
-
-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
-
-@example
-git config --global core.editor
-@end example
-
-or set by one of the following environment variables:
-@var{GIT_EDITOR}, @var{VISUAL} or @var{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.
-
-@section Preparing a patchset
-
-@example
-git format-patch <commit> [-o directory]
-@end example
-
-will generate a set of patches for each commit between @var{<commit>} and
-current @var{HEAD}. E.g.
-
-@example
-git format-patch origin/master
-@end example
-
-will generate patches for all commits on current branch which are not
-present in upstream.
-A useful shortcut is also
-
-@example
-git format-patch -n
-@end example
-
-which will generate patches from last @var{n} commits.
-By default the patches are created in the current directory.
-
-@section Sending patches for review
-
-@example
-git send-email <commit list|directory>
-@end example
-
-will send the patches created by @command{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. @var{git-email}
-package on Debian-based distros).
-
-
-@section Renaming/moving/copying files or contents of files
-
-Git automatically tracks such changes, making those normal commits.
-
-@example
-mv/cp path/file otherpath/otherfile
-git add [-A] .
-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
-
-@example
-git reset <commit>
-@end example
-
-@command{git reset} will uncommit the changes till @var{<commit>} rewriting
-the current branch history.
-
-@example
-git commit --amend
-@end example
-
-allows one to amend the last commit details quickly.
-
-@example
-git rebase -i origin/master
-@end example
-
-will replay local commits over the main repository allowing to edit, merge
-or remove some of them in the process.
-
-@float NOTE
-@command{git reset}, @command{git commit --amend} and @command{git rebase}
-rewrite history, so you should use them ONLY on your local or topic branches.
-The main repository will reject those changes.
-@end float
-
-@example
-git revert <commit>
-@end example
-
-@command{git revert} will generate a revert commit. This will not make the
-faulty commit disappear from the history.
-
-@section Pushing changes to remote trees
-
-@example
-git push
-@end example
-
-Will push the changes to the default remote (@var{origin}).
-Git will prevent you from pushing changes if the local and remote trees are
-out of sync. Refer to and to sync the local tree.
-
-@example
-git remote add <name> <url>
-@end example
-
-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.
-
-@example
-git push <remote> <refspec>
-@end example
-
-Will push the changes to the @var{<remote>} repository.
-Omitting @var{<refspec>} makes @command{git push} update all the remote
-branches matching the local ones.
-
-@section Finding a specific svn revision
-
-Since version 1.7.1 git supports @var{:/foo} syntax for specifying commits
-based on a regular expression. see man gitrevisions
-
-@example
-git show :/'as revision 23456'
-@end example
-
-will show the svn changeset @var{r23456}. With older git versions searching in
-the @command{git log} output is the easiest option (especially if a pager with
-search capabilities is used).
-This commit can be checked out with
-
-@example
-git checkout -b svn_23456 :/'as revision 23456'
-@end example
-
-or for git < 1.7.1 with
-
-@example
-git checkout -b svn_23456 $SHA1
-@end example
-
-where @var{$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
-problems with the GIT server.

doc/indevs.texi

@@ -1,763 +0,0 @@
-@chapter Input Devices
-@c man begin INPUT DEVICES
-
-Input devices are configured elements in FFmpeg which allow to access
-the data coming from a multimedia device attached to your system.
-
-When you configure your FFmpeg build, all the supported input devices
-are enabled by default. You can list all available ones using the
-configure option "--list-indevs".
-
-You can disable all the input devices using the configure option
-"--disable-indevs", and selectively enable an input device using the
-option "--enable-indev=@var{INDEV}", or you can disable a particular
-input device using the option "--disable-indev=@var{INDEV}".
-
-The option "-formats" of the ff* tools will display the list of
-supported input devices (amongst the demuxers).
-
-A description of the currently available input devices follows.
-
-@section alsa
-
-ALSA (Advanced Linux Sound Architecture) input device.
-
-To enable this input device during configuration you need libasound
-installed on your system.
-
-This device allows capturing from an ALSA device. The name of the
-device to capture has to be an ALSA card identifier.
-
-An ALSA identifier has the syntax:
-@example
-hw:@var{CARD}[,@var{DEV}[,@var{SUBDEV}]]
-@end example
-
-where the @var{DEV} and @var{SUBDEV} components are optional.
-
-The three arguments (in order: @var{CARD},@var{DEV},@var{SUBDEV})
-specify card number or identifier, device number and subdevice number
-(-1 means any).
-
-To see the list of cards currently recognized by your system check the
-files @file{/proc/asound/cards} and @file{/proc/asound/devices}.
-
-For example to capture with @command{ffmpeg} from an ALSA device with
-card id 0, you may run the command:
-@example
-ffmpeg -f alsa -i hw:0 alsaout.wav
-@end example
-
-For more information see:
-@url{http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html}
-
-@section bktr
-
-BSD video input device.
-
-@section dshow
-
-Windows DirectShow input device.
-
-DirectShow support is enabled when FFmpeg is built with the mingw-w64 project.
-Currently only audio and video devices are supported.
-
-Multiple devices may be opened as separate inputs, but they may also be
-opened on the same input, which should improve synchronism between them.
-
-The input name should be in the format:
-
-@example
-@var{TYPE}=@var{NAME}[:@var{TYPE}=@var{NAME}]
-@end example
-
-where @var{TYPE} can be either @var{audio} or @var{video},
-and @var{NAME} is the device's name.
-
-@subsection Options
-
-If no options are specified, the device's defaults are used.
-If the device does not support the requested options, it will
-fail to open.
-
-@table @option
-
-@item video_size
-Set the video size in the captured video.
-
-@item framerate
-Set the frame rate in the captured video.
-
-@item sample_rate
-Set the sample rate (in Hz) of the captured audio.
-
-@item sample_size
-Set the sample size (in bits) of the captured audio.
-
-@item channels
-Set the number of channels in the captured audio.
-
-@item list_devices
-If set to @option{true}, print a list of devices and exit.
-
-@item list_options
-If set to @option{true}, print a list of selected device's options
-and exit.
-
-@item video_device_number
-Set video device number for devices with same name (starts at 0,
-defaults to 0).
-
-@item audio_device_number
-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
-
-@itemize
-
-@item
-Print the list of DirectShow supported devices and exit:
-@example
-$ ffmpeg -list_devices true -f dshow -i dummy
-@end example
-
-@item
-Open video device @var{Camera}:
-@example
-$ ffmpeg -f dshow -i video="Camera"
-@end example
-
-@item
-Open second video device with name @var{Camera}:
-@example
-$ ffmpeg -f dshow -video_device_number 1 -i video="Camera"
-@end example
-
-@item
-Open video device @var{Camera} and audio device @var{Microphone}:
-@example
-$ ffmpeg -f dshow -i video="Camera":audio="Microphone"
-@end example
-
-@item
-Print the list of supported options in selected device and exit:
-@example
-$ ffmpeg -list_options true -f dshow -i video="Camera"
-@end example
-
-@end itemize
-
-@section dv1394
-
-Linux DV 1394 input device.
-
-@section fbdev
-
-Linux framebuffer input device.
-
-The Linux framebuffer is a graphic hardware-independent abstraction
-layer to show graphics on a computer monitor, typically on the
-console. It is accessed through a file device node, usually
-@file{/dev/fb0}.
-
-For more detailed information read the file
-Documentation/fb/framebuffer.txt included in the Linux source tree.
-
-To record from the framebuffer device @file{/dev/fb0} with
-@command{ffmpeg}:
-@example
-ffmpeg -f fbdev -r 10 -i /dev/fb0 out.avi
-@end example
-
-You can take a single screenshot image with the command:
-@example
-ffmpeg -f fbdev -frames:v 1 -r 1 -i /dev/fb0 screenshot.jpeg
-@end example
-
-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.
-
-To enable this input device during configuration you need libjack
-installed on your system.
-
-A JACK input device creates one or more JACK writable clients, one for
-each audio channel, with name @var{client_name}:input_@var{N}, where
-@var{client_name} is the name provided by the application, and @var{N}
-is a number which identifies the channel.
-Each writable client will send the acquired data to the FFmpeg input
-device.
-
-Once you have created one or more JACK readable clients, you need to
-connect them to one or more JACK writable clients.
-
-To connect or disconnect JACK clients you can use the @command{jack_connect}
-and @command{jack_disconnect} programs, or do it through a graphical interface,
-for example with @command{qjackctl}.
-
-To list the JACK clients and their properties you can invoke the command
-@command{jack_lsp}.
-
-Follows an example which shows how to capture a JACK readable client
-with @command{ffmpeg}.
-@example
-# Create a JACK writable client with name "ffmpeg".
-$ ffmpeg -f jack -i ffmpeg -y out.wav
-
-# Start the sample jack_metro readable client.
-$ jack_metro -b 120 -d 0.2 -f 4000
-
-# List the current JACK clients.
-$ jack_lsp -c
-system:capture_1
-system:capture_2
-system:playback_1
-system:playback_2
-ffmpeg:input_1
-metro:120_bpm
-
-# Connect metro to the ffmpeg writable client.
-$ jack_connect metro:120_bpm ffmpeg:input_1
-@end example
-
-For more information read:
-@url{http://jackaudio.org/}
-
-@section lavfi
-
-Libavfilter input virtual device.
-
-This input device reads data from the open output pads of a libavfilter
-filtergraph.
-
-For each filtergraph open output, the input device will create a
-corresponding stream which is mapped to the generated output. Currently
-only video data is supported. The filtergraph is specified through the
-option @option{graph}.
-
-@subsection Options
-
-@table @option
-
-@item graph
-Specify the filtergraph to use as input. Each video open output must be
-labelled by a unique string of the form "out@var{N}", where @var{N} is a
-number starting from 0 corresponding to the mapped input stream
-generated by the device.
-The first unlabelled output is automatically assigned to the "out0"
-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
-
-@itemize
-@item
-Create a color video stream and play it back with @command{ffplay}:
-@example
-ffplay -f lavfi -graph "color=c=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
-@end example
-
-@item
-Create three different video test filtered sources and play them:
-@example
-ffplay -f lavfi -graph "testsrc [out0]; testsrc,hflip [out1]; testsrc,negate [out2]" test3
-@end example
-
-@item
-Read an audio stream from a file using the amovie source and play it
-back with @command{ffplay}:
-@example
-ffplay -f lavfi "amovie=test.wav"
-@end example
-
-@item
-Read an audio stream and a video stream and play it back with
-@command{ffplay}:
-@example
-ffplay -f lavfi "movie=test.avi[out0];amovie=test.wav[out1]"
-@end example
-
-@end itemize
-
-@section libdc1394
-
-IIDC1394 input device, based on libdc1394 and libraw1394.
-
-@section openal
-
-The OpenAL input device provides audio capture on all systems with a
-working OpenAL 1.1 implementation.
-
-To enable this input device during configuration, you need OpenAL
-headers and libraries installed on your system, and need to configure
-FFmpeg with @code{--enable-openal}.
-
-OpenAL headers and libraries should be provided as part of your OpenAL
-implementation, or as an additional download (an SDK). Depending on your
-installation you may need to specify additional flags via the
-@code{--extra-cflags} and @code{--extra-ldflags} for allowing the build
-system to locate the OpenAL headers and libraries.
-
-An incomplete list of OpenAL implementations follows:
-
-@table @strong
-@item Creative
-The official Windows implementation, providing hardware acceleration
-with supported devices and software fallback.
-See @url{http://openal.org/}.
-@item OpenAL Soft
-Portable, open source (LGPL) software implementation. Includes
-backends for the most common sound APIs on the Windows, Linux,
-Solaris, and BSD operating systems.
-See @url{http://kcat.strangesoft.net/openal.html}.
-@item Apple
-OpenAL is part of Core Audio, the official Mac OS X Audio interface.
-See @url{http://developer.apple.com/technologies/mac/audio-and-video.html}
-@end table
-
-This device allows one to capture from an audio input device handled
-through OpenAL.
-
-You need to specify the name of the device to capture in the provided
-filename. If the empty string is provided, the device will
-automatically select the default device. You can get the list of the
-supported devices by using the option @var{list_devices}.
-
-@subsection Options
-
-@table @option
-
-@item channels
-Set the number of channels in the captured audio. Only the values
-@option{1} (monaural) and @option{2} (stereo) are currently supported.
-Defaults to @option{2}.
-
-@item sample_size
-Set the sample size (in bits) of the captured audio. Only the values
-@option{8} and @option{16} are currently supported. Defaults to
-@option{16}.
-
-@item sample_rate
-Set the sample rate (in Hz) of the captured audio.
-Defaults to @option{44.1k}.
-
-@item list_devices
-If set to @option{true}, print a list of devices and exit.
-Defaults to @option{false}.
-
-@end table
-
-@subsection Examples
-
-Print the list of OpenAL supported devices and exit:
-@example
-$ ffmpeg -list_devices true -f openal -i dummy out.ogg
-@end example
-
-Capture from the OpenAL device @file{DR-BT101 via PulseAudio}:
-@example
-$ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out.ogg
-@end example
-
-Capture from the default device (note the empty string '' as filename):
-@example
-$ ffmpeg -f openal -i '' out.ogg
-@end example
-
-Capture from two devices simultaneously, writing to two different files,
-within the same @command{ffmpeg} command:
-@example
-$ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out1.ogg -f openal -i 'ALSA Default' out2.ogg
-@end example
-Note: not all OpenAL implementations support multiple simultaneous capture -
-try the latest OpenAL Soft if the above does not work.
-
-@section oss
-
-Open Sound System input device.
-
-The filename to provide to the input device is the device node
-representing the OSS input device, and is usually set to
-@file{/dev/dsp}.
-
-For example to grab from @file{/dev/dsp} using @command{ffmpeg} use the
-command:
-@example
-ffmpeg -f oss -i /dev/dsp /tmp/oss.wav
-@end example
-
-For more information about OSS see:
-@url{http://manuals.opensound.com/usersguide/dsp.html}
-
-@section pulse
-
-PulseAudio input device.
-
-To enable this output device you need to configure FFmpeg with @code{--enable-libpulse}.
-
-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
-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
-
-@section sndio
-
-sndio input device.
-
-To enable this input device during configuration you need libsndio
-installed on your system.
-
-The filename to provide to the input device is the device node
-representing the sndio input device, and is usually set to
-@file{/dev/audio0}.
-
-For example to grab from @file{/dev/audio0} using @command{ffmpeg} use the
-command:
-@example
-ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav
-@end example
-
-@section video4linux2, v4l2
-
-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
-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
-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}.
-
-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
-clock (origin at the Unix Epoch) or the monotonic clock (origin usually at
-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:
-@example
-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
-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 one 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
-
-@section vfwcap
-
-VfW (Video for Windows) capture input device.
-
-The filename passed as input is the capture driver number, ranging from
-0 to 9. You may use "list" as filename to print a list of drivers. Any
-other filename will be interpreted as device number 0.
-
-@section x11grab
-
-X11 video input device.
-
-This device allows one to capture a region of an X11 display.
-
-The filename passed as input has the syntax:
-@example
-[@var{hostname}]:@var{display_number}.@var{screen_number}[+@var{x_offset},@var{y_offset}]
-@end example
-
-@var{hostname}:@var{display_number}.@var{screen_number} specifies the
-X11 display name of the screen to grab from. @var{hostname} can be
-omitted, and defaults to "localhost". The environment variable
-@env{DISPLAY} contains the default display name.
-
-@var{x_offset} and @var{y_offset} specify the offsets of the grabbed
-area with respect to the top-left border of the X11 screen. They
-default to 0.
-
-Check the X11 documentation (e.g. man X) for more detailed information.
-
-Use the @command{dpyinfo} program for getting basic information about the
-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
-@end example
-
-Grab at position @code{10,20}:
-@example
-ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0+10,20 out.mpg
-@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
-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
-@end example
-
-To follow only when the mouse pointer reaches within 100 pixels to edge:
-@example
-ffmpeg -f x11grab -follow_mouse 100 -framerate 25 -video_size cif -i :0.0 out.mpg
-@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.
-
-For example:
-@example
-ffmpeg -f x11grab -show_region 1 -framerate 25 -video_size cif -i :0.0+10,20 out.mpg
-@end example
-
-With @var{follow_mouse}:
-@example
-ffmpeg -f x11grab -follow_mouse centered -show_region 1 -framerate 25 -video_size 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