update for 2.1

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

.gitattributes

@@ -1 +0,0 @@
-*.pnm -diff -text

.gitignore

@@ -15,7 +15,6 @@
 *.pdb
 *.so
 *.so.*
-*.swp
 *.ver
 *-example
 *-test
@@ -28,6 +27,7 @@
 /ffserver
 /config.*
 /coverage.info
+/version.h
 /doc/*.1
 /doc/*.3
 /doc/*.html
@@ -35,34 +35,26 @@
 /doc/config.texi
 /doc/avoptions_codec.texi
 /doc/avoptions_format.texi
-/doc/doxy/html/
-/doc/examples/avio_reading
 /doc/examples/decoding_encoding
-/doc/examples/demuxing_decoding
-/doc/examples/extract_mvs
-/doc/examples/filter_audio
+/doc/examples/demuxing
 /doc/examples/filtering_audio
 /doc/examples/filtering_video
 /doc/examples/metadata
 /doc/examples/muxing
 /doc/examples/pc-uninstalled
-/doc/examples/remuxing
 /doc/examples/resampling_audio
 /doc/examples/scaling_video
-/doc/examples/transcode_aac
-/doc/examples/transcoding
 /doc/fate.txt
+/doc/doxy/html/
 /doc/print_options
 /lcov/
 /libavcodec/*_tablegen
 /libavcodec/*_tables.c
 /libavcodec/*_tables.h
 /libavutil/avconfig.h
-/libavutil/ffversion.h
 /tests/audiogen
 /tests/base64
 /tests/data/
-/tests/pixfmts.mak
 /tests/rotozoom
 /tests/tiny_psnr
 /tests/tiny_ssim
@@ -84,5 +76,4 @@
 /tools/qt-faststart
 /tools/trasher
 /tools/seek_print
-/tools/uncoded_frame
 /tools/zmqsend

Changelog

@@ -1,282 +1,7 @@
 Entries are sorted chronologically from oldest to youngest within each release,
 releases are sorted from youngest to oldest.
 
-version <next>:
-
-version 2.4.7:
-- avcodec/flac_parser: fix handling EOF if no headers are found
-- avfilter/vf_framepack: Check and update frame_rate
-- avcodec/hevc: Fix handling of skipped_bytes() reallocation failures
-- qpeg: avoid pointless invalid memcpy()
-- avcodec/arm/videodsp_armv5te: Fix linking failure with "g++ -shared -D__STDC_CONSTANT_MACROS -o test.so ... libavcodec.a"
-- avcodec/mjpegdec: Skip blocks which are outside the visible area
-- lavc/aarch64: Do not use the neon horizontal chroma loop filter for H.264 4:2:2. (cherry picked from commit 4faea46bd906b3897018736208123aa36c3f45d5)
-- avcodec/h264_slice: assert that reinit does not occur after the first slice
-- avcodec/h264_slice: ignore SAR changes in slices after the first
-- avcodec/h264_slice: Check picture structure before setting the related fields
-- avcodec/h264_slice: Do not change frame_num after the first slice
-- avutil/opt: Fix type used to access AV_OPT_TYPE_SAMPLE_FMT
-- avutil/opt: Fix types used to access AV_OPT_TYPE_PIXEL_FMT
-- avcodec/h264: Be more strict on rejecting pps/sps changes
-- avcodec/h264: Be more strict on rejecting pps_id changes
-- avcodec/h264_ps: More completely check the bit depths
-- avformat/thp: Check av_get_packet() for failure not only for partial output
-- swscale/utils: Limit filter shifting so as not to read from prior the array
-- avcodec/mpegvideo_motion: Fix gmc chroma dimensions
-- avcodec/mjpegdec: Check number of components for JPEG-LS
-- avcodec/mjpegdec: Check escape sequence validity
-- avformat/mpc8: Use uint64_t in *_get_v() to avoid undefined behavior
-- avformat/mpc8: fix broken pointer math
-- avformat/mpc8: fix hang with fuzzed file
-- avformat/tta: fix crash with corrupted files
-
-version 2.4.6:
-- doc/examples: fix lib math dep for decoding_encoding
-- avformat/movenc: workaround bug in "PathScale EKOPath(tm) Compiler Suite Version 4.0.12.1"
-- vp9: fix parser return values in error case
-- ffmpeg: Clear error message array at init.
-- avcodec/dvdsubdec: fix accessing dangling pointers
-- avcodec/dvdsubdec: error on bitmaps with size 0
-- avformat/mov: Fix mixed declaration and statement warning
-- cmdutils: Use 64bit for file size/offset related variable in cmdutils_read_file()
-- avformat/utils: Clear pointer in ff_alloc_extradata() to avoid leaving a stale pointer in memory
-- avformat/matroskadec: Use av_freep() to avoid leaving stale pointers in memory
-- lavfi: check av_strdup() return value
-- mov: Fix negative size calculation in mov_read_default().
-- avformat/mov: fix integer overflow in mov_read_udta_string()
-- mov: Avoid overflow with mov_metadata_raw()
-- avcodec/dvdsubdec: fix out of bounds accesses
-- avfilter/vf_sab: fix filtering tiny images
-- avformat/flvdec: Increase string array size
-- avformat/flvdec: do not inject dts=0 metadata packets which failed to be parsed into a new data stream
-- avformat/cdxl: Fix integer overflow of image_size
-- avformat/segment: Use av_freep() avoid leaving stale pointers in memory
-- avformat/mov: Fix memleaks for duplicate STCO/CO64/STSC atoms
-- mov: avoid a memleak when multiple stss boxes are presen
-
-version 2.4.5:
-- lavu/frame: fix malloc error path in av_frame_copy_props()
-- avformat/utils: Do not update programs streams from program-less streams in update_wrap_reference()
-- avformat/aviobuf: Check that avio_seek() target is non negative
-- swresample/soxr_resample: fix error handling
-- avformat/flvdec: fix potential use of uninitialized variables
-- avformat/matroskadec: fix handling of recursive SeekHead elements
-- doc/examples/transcoding: check encoder before using it
-- swscale/x86/rgb2rgb_template: fix crash with tiny size and nv12 output
-- avformat/rmdec: Check codec_data_size
-- avformat/aviobuf: Fix infinite loop in ff_get_line()
-- vc1: Do not assume seek happens after decoding
-- mmvideo: check frame dimensions
-- jvdec: check frame dimensions
-- avcodec/indeo3: ensure offsets are non negative
-- avcodec/h264: Check *log2_weight_denom
-- avcodec/hevc_ps: Check diff_cu_qp_delta_depth
-- avcodec/h264: Clear delayed_pic on deallocation
-- avcodec/hevc: clear filter_slice_edges() on allocation
-- avcodec/dcadec: Check that the added xch channel isnt already there
-- avcodec/indeo3: use signed variables to avoid underflow
-- swscale: increase yuv2rgb table headroom
-- avformat/mov: fix integer overflow of size
-- avformat/mov: check atom nesting depth
-- avcodec/utvideodec: Fix handling of slice_height=0
-- avcodec/vmdvideo: Check len before using it in method 3
-- avformat/flvdec: Use av_freep() avoid leaving stale pointers in memory
-- avformat/hdsenc: Use av_freep() avoid leaving stale pointers in memory
-- configure: create the tests directory like the doc directory
-- v4l2: Make use of the VIDIOC_ENUM_FRAMESIZES ioctl on OpenBSD
-- avcodec/motion_est: use 2x8x8 for interlaced qpel
-- Treat all '*.pnm' files as non-text file
-
-version 2.4.4:
-- avformat: replace some odd 30-60 rates by higher less odd ones in  get_std_framerate()
-- swscale: fix yuv2yuvX_8 assembly on x86
-- avcodec/hevc_ps: Check num_long_term_ref_pics_sps
-- avcodec/mjpegdec: Fix integer overflow in shift
-- avcodec/hevc_ps: Check return code from pps_range_extensions()
-- avcodec/rawdec: Check the return code of avpicture_get_size()
-- avcodec/pngdec: Check IHDR/IDAT order
-- avcodec/flacdec: Call ff_flacdsp_init() unconditionally
-- avcodec/utils: Check that the data is complete in avpriv_bprint_to_extradata()
-- avcodec/mjpegdec: Fix context fields becoming inconsistent
-- avcodec/mjpegdec: Check for pixfmtid 0x42111100 || 0x24111100 with more than 8 bits
-- swscale/x86/rgb2rgb_template: handle the first 2 lines with C in rgb24toyv12_*()
-- doc/APIchanges: Fix some wrong versions
-- avformat/hlsenc: Free context after hls_append_segment
-- avcodec/mpeg4video_parser: fix spurious extradata parse warnings
-- lavu/opt: fix av_opt_get function
-- avcodec/wmaprodec: Fix integer overflow in sfb_offsets initialization
-- avcodec/utvideodec: fix assumtation that slice_height >= 1
-- avcodec/options_table fix min of audio channels and sample rate
-- libavutil/thread.h: Support OS/2 threads
-- fix Makefile objects for pulseaudio support
-- opusdec: make sure all substreams have the same number of coded samples
-- lavu: add wrappers for the pthreads mutex API
-- avformat/avidec: fix handling dv in avi
-- avfilter/vf_lut: gammaval709()
-- cinedec: report white balance gain coefficients using metadata
-- swscale/utils: support bayer input + scaling, and bayer input + any supported output
-- swscale: support internal scaler cascades
-- avformat/dtsdec: dts_probe: check reserved bit, check lfe, check sr_code similarity
-- avformat/segment: export inner muxer timebase
-- Remove fminf() emulation, fix build issues
-- avcodec/mpegaudio_parser: fix off by 1 error in bitrate calculation
-- Use -fno-optimize-sibling-calls on parisc also for gcc 4.9.
-- ffmpeg_opt: store canvas size in decoder context
-- avcodec/mpeg12dec: do not trust AVCodecContext input dimensions
-
-version 2.4.3:
-- avcodec/svq1dec: zero terminate embedded message before printing
-- avcodec/cook: check that the subpacket sizes fit in block_align
-- avcodec/g2meet: check tile dimensions to avoid integer overflow
-- avcodec/utils: Align dimensions by at least their chroma sub-sampling factors.
-- avcodec/dnxhddec: treat pix_fmt like width/height
-- avcodec/dxa: check dimensions
-- avcodec/dirac_arith: fix integer overflow
-- avcodec/diracdec: Tighter checks on CODEBLOCKS_X/Y
-- avcodec/diracdec: Use 64bit in calculation of codeblock coordinates
-- avcodec/sgidec: fix count check
-- avcodec/sgidec: fix linesize for 16bit
-- avcodec/hevc_ps: Check default display window bitstream and skip if invalid
-- avcodec/tiffenc: properly compute packet size
-- lavd: export all symbols with av_ prefix
-- avformat/mxfdec: Fix termination of mxf_data_essence_container_uls
-- postproc: fix qp count
-- postproc/postprocess: fix quant store for fq mode
-- vf_drawtext: add missing clear of pointers after av_expr_free()
-- utvideoenc: properly set slice height/last line
-- swresample: fix sample drop loop end condition
-- resample: Avoid off-by-1 errors in PTS calcs.
-- imc: fix order of operations in coefficients read
-- hevc_mvs: make sure to always initialize the temporal MV fully
-- hevc_mvs: initialize the temporal MV in case of missing reference
-
-version 2.4.2:
-- avcodec/on2avc: Check number of channels
-- avcodec/hevc: fix chroma transform_add size
-- avcodec/h264: Check mode before considering mixed mode intra prediction
-- avformat/mpegts: use a padded buffer in read_sl_header()
-- avformat/mpegts: Check desc_len / get8() return code
-- avcodec/vorbisdec: Fix off by 1 error in ptns_to_read
-- sdp: add support for H.261
-- avcodec/svq3: Do not memcpy AVFrame
-- avcodec/smc: fix off by 1 error
-- avcodec/qpeg: fix off by 1 error in MV bounds check
-- avcodec/gifdec: factorize interleave end handling out
-- avcodec/cinepak: fix integer underflow
-- avcodec/pngdec: Check bits per pixel before setting monoblack pixel format
-- avcodec/pngdec: Calculate MPNG bytewidth more defensively
-- avcodec/tiff: more completely check bpp/bppcount
-- avcodec/mmvideo: Bounds check 2nd line of HHV Intra blocks
-- avcodec/h263dec: Fix decoding messenger.h263
-- avcodec/utils: Add case for jv to avcodec_align_dimensions2()
-- avcodec/mjpegdec: check bits per pixel for changes similar to dimensions
-- avcodec/jpeglsdec: Check run value more completely in ls_decode_line()
-- avformat/hlsenc: export inner muxer timebase
-- configure: add noexecstack to linker options if supported.
-- avcodec/ac3enc_template: fix out of array read
-- avutil/x86/cpu: fix cpuid sub-leaf selection
-- avformat/img2dec: enable generic seeking for image pipes
-- avformat/img2dec: initialize pkt->pos for image pipes
-- avformat/img2dec: pass error code and signal EOF
-- avformat/img2dec: fix error code at EOF for pipes
-- libavutil/opt: fix av_opt_set_channel_layout() to access correct memory address
-- tests/fate-run.sh: Cat .err file in case of error with V>0
-- avformat/riffenc: Filter out "BottomUp" in ff_put_bmp_header()
-- avcodec/webp: fix default palette color 0xff000000 -> 0x00000000
-- avcodec/asvenc: fix AAN scaling
-- Fix compile error on arm4/arm5 platform
-
-
-version 2.4.1:
-- swscale: Allow chroma samples to be above and to the left of luma samples
-- avcodec/libilbc: support for latest git of libilbc
-- avcodec/webp: treat out-of-bound palette index as translucent black
-- vf_deshake: rename Transform.vector to Transform.vec to avoid compiler confusion
-- apetag: Fix APE tag size check
-- tools/crypto_bench: fix build when AV_READ_TIME is unavailable
-
-
-version 2.4:
-- Icecast protocol
-- ported lenscorrection filter from frei0r filter
-- large optimizations in dctdnoiz to make it usable
-- ICY metadata are now requested by default with the HTTP protocol
-- support for using metadata in stream specifiers in fftools
-- LZMA compression support in TIFF decoder
-- support for H.261 RTP payload format (RFC 4587)
-- HEVC/H.265 RTP payload format (draft v6) depacketizer
-- added codecview filter to visualize information exported by some codecs
-- Matroska 3D support thorugh side data
-- HTML generation using texi2html is deprecated in favor of makeinfo/texi2any
-- silenceremove filter
-
-
-version 2.3:
-- AC3 fixed-point decoding
-- shuffleplanes filter
-- subfile protocol
-- Phantom Cine demuxer
-- replaygain data export
-- VP7 video decoder
-- Alias PIX image encoder and decoder
-- Improvements to the BRender PIX image decoder
-- Improvements to the XBM decoder
-- QTKit input device
-- improvements to OpenEXR image decoder
-- support decoding 16-bit RLE SGI images
-- GDI screen grabbing for Windows
-- alternative rendition support for HTTP Live Streaming
-- AVFoundation input device
-- Direct Stream Digital (DSD) decoder
-- Magic Lantern Video (MLV) demuxer
-- On2 AVC (Audio for Video) decoder
-- support for decoding through DXVA2 in ffmpeg
-- libbs2b-based stereo-to-binaural audio filter
-- libx264 reference frames count limiting depending on level
-- native Opus decoder
-- display matrix export and rotation API
-- WebVTT encoder
-- showcqt multimedia filter
-- zoompan filter
-- signalstats filter
-- hqx filter (hq2x, hq3x, hq4x)
-- flanger filter
-- Image format auto-detection
-- LRC demuxer and muxer
-- Samba protocol (via libsmbclient)
-- WebM DASH Manifest muxer
-- libfribidi support in drawtext
-
-
-version 2.2:
-
-- HNM version 4 demuxer and video decoder
-- Live HDS muxer
-- setsar/setdar filters now support variables in ratio expressions
-- elbg filter
-- string validation in ffprobe
-- support for decoding through VDPAU in ffmpeg (the -hwaccel option)
-- complete Voxware MetaSound decoder
-- remove mp3_header_compress bitstream filter
-- Windows resource files for shared libraries
-- aeval filter
-- stereoscopic 3d metadata handling
-- WebP encoding via libwebp
-- ATRAC3+ decoder
-- VP8 in Ogg demuxing
-- side & metadata support in NUT
-- framepack filter
-- XYZ12 rawvideo support in NUT
-- Exif metadata support in WebP decoder
-- OpenGL device
-- Use metadata_header_padding to control padding in ID3 tags (currently used in
-  MP3, AIFF, and OMA files), FLAC header, and the AVI "junk" block.
-- Mirillis FIC video decoder
-- Support DNx444
-- libx265 encoder
-- dejudder filter
-- Autodetect VDA like all other hardware accelerations
-- aliases and defaults for Ogg subtypes (opus, spx)
+version <next>
 
 
 version 2.1:
@@ -321,8 +46,7 @@ version 2.1:
 - 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
+- HEVC decoder, raw HEVC demuxer, HEVC demuxing in TS, Matroska and MP4
 - mergeplanes filter
 
 
@@ -338,7 +62,7 @@ version 2.0:
 - 10% faster aac encoding on x86 and MIPS
 - sine audio filter source
 - WebP demuxing and decoding support
-- ffmpeg options -filter_script and -filter_complex_script, which allow a
+- new ffmpeg options -filter_script and -filter_complex_script, which allow a
   filtergraph description to be read from a file
 - OpenCL support
 - audio phaser filter
@@ -346,7 +70,7 @@ version 2.0:
 - libquvi demuxer
 - uniform options syntax across all filters
 - telecine filter
-- interlace filter
+- new interlace filter
 - smptehdbars source
 - inverse telecine filters (fieldmatch and decimate)
 - colorbalance filter
@@ -464,7 +188,7 @@ version 1.1:
 - JSON captions for TED talks decoding support
 - SOX Resampler support in libswresample
 - aselect filter
-- SGI RLE 8-bit / Silicon Graphics RLE 8-bit video decoder
+- SGI RLE 8-bit decoder
 - Silicon Graphics Motion Video Compressor 1 & 2 decoder
 - Silicon Graphics Movie demuxer
 - apad filter
@@ -508,9 +232,7 @@ version 1.0:
 - RTMPE protocol support
 - RTMPTE protocol support
 - showwaves and showspectrum filter
-- LucasArts SMUSH SANM playback support
-- LucasArts SMUSH VIMA audio decoder (ADPCM)
-- LucasArts SMUSH demuxer
+- LucasArts SMUSH playback support
 - SAMI, RealText and SubViewer demuxers and decoders
 - Heart Of Darkness PAF playback support
 - iec61883 device
@@ -634,7 +356,6 @@ version 0.10:
 - ffwavesynth decoder
 - aviocat tool
 - ffeval tool
-- support encoding and decoding 4-channel SGI images
 
 
 version 0.9:

INSTALL

@@ -0,0 +1,15 @@
+
+1) Type './configure' to create the configuration. A list of configure
+options is printed by running 'configure --help'.
+
+'configure' can be launched from a directory different from the FFmpeg
+sources to build the objects out of tree. To do this, use an absolute
+path when launching 'configure', e.g. '/ffmpegdir/ffmpeg/configure'.
+
+2) Then type 'make' to build FFmpeg. GNU Make 3.81 or later is required.
+
+3) Type 'make install' to install all binaries and libraries you built.
+
+NOTICE
+
+ - Non system dependencies (e.g. libx264, libvpx) are disabled by default.

INSTALL.md

@@ -1,17 +0,0 @@
-#Installing FFmpeg:
-
-1. Type `./configure` to create the configuration. A list of configure
-options is printed by running `configure --help`.
-
-    `configure` can be launched from a directory different from the FFmpeg
-sources to build the objects out of tree. To do this, use an absolute
-path when launching `configure`, e.g. `/ffmpegdir/ffmpeg/configure`.
-
-2. Then type `make` to build FFmpeg. GNU Make 3.81 or later is required.
-
-3. Type `make install` to install all binaries and libraries you built.
-
-NOTICE
-------
-
- - Non system dependencies (e.g. libx264, libvpx) are disabled by default.

LICENSE

@@ -1,4 +1,4 @@
-#FFmpeg:
+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
@@ -10,12 +10,11 @@ version 2 or later (GPL v2+). See the file COPYING.GPLv2 for details. None of
 these parts are used by default, you have to explicitly pass --enable-gpl to
 configure to activate them. In this case, FFmpeg's license changes to GPL v2+.
 
-Specifically, the GPL parts of FFmpeg are:
+Specifically, the GPL parts of FFmpeg are
 
 - libpostproc
 - libmpcodecs
 - optional x86 optimizations in the files
-  libavcodec/x86/flac_dsp_gpl.asm
   libavcodec/x86/idct_mmx.c
 - libutvideo encoding/decoding wrappers in
   libavcodec/libutvideo*.cpp
@@ -34,7 +33,6 @@ Specifically, the GPL parts of FFmpeg are:
     - vf_geq.c
     - vf_histeq.c
     - vf_hqdn3d.c
-    - vf_interlace.c
     - vf_kerndeint.c
     - vf_mcdeint.c
     - vf_mp.c
@@ -49,6 +47,7 @@ Specifically, the GPL parts of FFmpeg are:
     - vf_stereo3d.c
     - vf_super2xsai.c
     - vf_tinterlace.c
+    - vf_yadif.c
     - vsrc_mptestsrc.c
 
 Should you, for whatever reason, prefer to use version 3 of the (L)GPL, then
@@ -81,7 +80,6 @@ The following libraries are under GPL:
     - libutvideo
     - libvidstab
     - libx264
-    - libx265
     - libxavs
     - libxvid
 When combining them with FFmpeg, FFmpeg needs to be licensed as GPL as well by

MAINTAINERS

@@ -31,7 +31,7 @@ ffprobe:
   ffprobe.c                             Stefano Sabatini
 
 ffserver:
-  ffserver.c                            Reynaldo H. Verdejo Pinochet
+  ffserver.c, ffserver.h                Baptiste Coudurier
 
 Commandline utility code:
   cmdutils.c, cmdutils.h                Michael Niedermayer
@@ -44,8 +44,8 @@ Miscellaneous Areas
 ===================
 
 documentation                           Stefano Sabatini, Mike Melanson, Timothy Gu
-build system (configure, makefiles)     Diego Biurrun, Mans Rullgard
-project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser
+build system (configure,Makefiles)      Diego Biurrun, Mans Rullgard
+project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Döffinger, Alexander Strasser
 presets                                 Robert Swain
 metadata subsystem                      Aurelien Jacobs
 release management                      Michael Niedermayer
@@ -54,10 +54,8 @@ release management                      Michael Niedermayer
 Communication
 =============
 
-website                                 Deby Barbara Lepage
-fate.ffmpeg.org                         Timothy Gu
-Trac bug tracker                        Alexander Strasser, Michael Niedermayer, Carl Eugen Hoyos, Lou Logan
-mailing lists                           Michael Niedermayer, Baptiste Coudurier, Lou Logan
+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
@@ -75,7 +73,6 @@ Other:
   bprint                                Nicolas George
   bswap.h
   des                                   Reimar Doeffinger
-  dynarray.h                            Nicolas George
   eval.c, eval.h                        Michael Niedermayer
   float_dsp                             Loren Merritt
   hash                                  Reimar Doeffinger
@@ -132,7 +129,6 @@ Generic Parts:
     tableprint.c, tableprint.h          Reimar Doeffinger
   fixed point FFT:
     fft*                                Zeljko Lukac
-  Text Subtitles                        Clément Bœsch
 
 Codecs:
   4xm.c                                 Michael Niedermayer
@@ -146,7 +142,6 @@ Codecs:
   ass*                                  Aurelien Jacobs
   asv*                                  Michael Niedermayer
   atrac3*                               Benjamin Larsson
-  atrac3plus*                           Maxim Poliakovski
   bgmc.c, bgmc.h                        Thilo Borgmann
   bink.c                                Kostya Shishkov
   binkaudio.c                           Peter Ross
@@ -155,7 +150,6 @@ Codecs:
   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
@@ -166,13 +160,11 @@ Codecs:
   dnxhd*                                Baptiste Coudurier
   dpcm.c                                Mike Melanson
   dv.c                                  Roman Shaposhnik
-  dvbsubdec.c                           Anshul Maheshwari
   dxa.c                                 Kostya Shishkov
   eacmv*, eaidct*, eat*                 Peter Ross
   exif.c, exif.h                        Thilo Borgmann
-  ffv1*                                 Michael Niedermayer
+  ffv1.c                                Michael Niedermayer
   ffwavesynth.c                         Nicolas George
-  fic.c                                 Derek Buitenhuis
   flac*                                 Justin Ruggles
   flashsv*                              Benjamin Larsson
   flicvideo.c                           Mike Melanson
@@ -182,7 +174,7 @@ Codecs:
   h261*                                 Michael Niedermayer
   h263*                                 Michael Niedermayer
   h264*                                 Loren Merritt, Michael Niedermayer
-  huffyuv*                              Michael Niedermayer, Christophe Gisquet
+  huffyuv.c                             Michael Niedermayer
   idcinvideo.c                          Mike Melanson
   imc*                                  Benjamin Larsson
   indeo2*                               Kostya Shishkov
@@ -205,11 +197,8 @@ Codecs:
   libtheoraenc.c                        David Conrad
   libutvideo*                           Derek Buitenhuis
   libvorbis.c                           David Conrad
-  libvpx*                               James Zern
   libx264.c                             Mans Rullgard, Jason Garrett-Glaser
-  libx265.c                             Derek Buitenhuis
   libxavs.c                             Stefan Gehrer
-  libzvbi-teletextdec.c                 Marton Balint
   loco.c                                Kostya Shishkov
   lzo.h, lzo.c                          Reimar Doeffinger
   mdec.c                                Michael Niedermayer
@@ -242,12 +231,12 @@ Codecs:
   rtjpeg.c, rtjpeg.h                    Reimar Doeffinger
   rv10.c                                Michael Niedermayer
   rv3*                                  Kostya Shishkov
-  rv4*                                  Kostya Shishkov, Christophe Gisquet
+  rv4*                                  Kostya Shishkov
   s3tc*                                 Ivo van Poorten
   smacker.c                             Kostya Shishkov
   smc.c                                 Mike Melanson
   smvjpegdec.c                          Ash Hughes
-  snow*                                 Michael Niedermayer, Loren Merritt
+  snow.c                                Michael Niedermayer, Loren Merritt
   sonic.c                               Alex Beregszaszi
   srt*                                  Aurelien Jacobs
   sunrast.c                             Ivo van Poorten
@@ -266,18 +255,17 @@ Codecs:
   v410*.c                               Derek Buitenhuis
   vb.c                                  Kostya Shishkov
   vble.c                                Derek Buitenhuis
-  vc1*                                  Kostya Shishkov, Christophe Gisquet
+  vc1*                                  Kostya Shishkov
   vcr1.c                                Michael Niedermayer
   vda_h264_dec.c                        Xidorn Quan
   vima.c                                Paul B Mahol
   vmnc.c                                Kostya Shishkov
-  vorbisdec.c                           Denes Balatoni, David Conrad
-  vorbisenc.c                           Oded Shimon
+  vorbis_dec.c                          Denes Balatoni, David Conrad
+  vorbis_enc.c                          Oded Shimon
   vp3*                                  Mike Melanson
   vp5                                   Aurelien Jacobs
   vp6                                   Aurelien Jacobs
   vp8                                   David Conrad, Jason Garrett-Glaser, Ronald Bultje
-  vp9                                   Ronald Bultje, Clément Bœsch
   vqavideo.c                            Mike Melanson
   wavpack.c                             Kostya Shishkov
   wmaprodec.c                           Sascha Sommer
@@ -308,20 +296,15 @@ libavdevice
     libavdevice/avdevice.h
 
 
-  avfoundation.m                        Thilo Borgmann
-  dshow.c                               Roger Pack (CC rogerdpack@gmail.com)
+  dshow.c                               Roger Pack
   fbdev_enc.c                           Lukasz Marek
-  gdigrab.c                             Roger Pack (CC rogerdpack@gmail.com)
   iec61883.c                            Georg Lippitsch
   lavfi                                 Stefano Sabatini
   libdc1394.c                           Roman Shaposhnik
-  opengl_enc.c                          Lukasz Marek
   pulse_audio_enc.c                     Lukasz Marek
-  qtkit.m                               Thilo Borgmann
   sdl                                   Stefano Sabatini
-  v4l2.c                                Giorgio Vazzana
+  v4l2.c                                Luca Abeni
   vfwcap.c                              Ramiro Polla
-  xv.c                                  Lukasz Marek
 
 libavfilter
 ===========
@@ -343,20 +326,14 @@ Filters:
   af_compand.c                          Paul B Mahol
   af_ladspa.c                           Paul B Mahol
   af_pan.c                              Nicolas George
-  af_silenceremove.c                    Paul B Mahol
   avf_avectorscope.c                    Paul B Mahol
-  avf_showcqt.c                         Muhammad Faiz
   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_hqx.c                              Clément Bœsch
-  vf_idet.c                             Pascal Massimino
   vf_il.c                               Paul B Mahol
-  vf_lenscorrection.c                   Daniel Oberhoff
   vf_mergeplanes.c                      Paul B Mahol
   vf_psnr.c                             Paul B Mahol
   vf_scale.c                            Michael Niedermayer
@@ -408,7 +385,6 @@ Muxers/Demuxers:
   flvdec.c, flvenc.c                    Michael Niedermayer
   gxf.c                                 Reimar Doeffinger
   gxfenc.c                              Baptiste Coudurier
-  hls.c                                 Anssi Hannula
   idcin.c                               Mike Melanson
   idroqdec.c                            Mike Melanson
   iff.c                                 Jaikrishnan Menon
@@ -426,7 +402,6 @@ Muxers/Demuxers:
   matroska.c                            Aurelien Jacobs
   matroskadec.c                         Aurelien Jacobs
   matroskaenc.c                         David Conrad
-  matroska subtitles (matroskaenc.c)    John Peebles
   metadata*                             Aurelien Jacobs
   mgsts.c                               Paul B Mahol
   microdvd*                             Aurelien Jacobs
@@ -436,15 +411,14 @@ Muxers/Demuxers:
   mpc.c                                 Kostya Shishkov
   mpeg.c                                Michael Niedermayer
   mpegenc.c                             Michael Niedermayer
-  mpegts.c                              Marton Balint
-  mpegtsenc.c                           Baptiste Coudurier
+  mpegts*                               Baptiste Coudurier
   msnwc_tcp.c                           Ramiro Polla
   mtv.c                                 Reynaldo H. Verdejo Pinochet
   mxf*                                  Baptiste Coudurier
   mxfdec.c                              Tomas Härdin
   nistspheredec.c                       Paul B Mahol
   nsvdec.c                              Francois Revol
-  nut*                                  Michael Niedermayer
+  nut.c                                 Michael Niedermayer
   nuv.c                                 Reimar Doeffinger
   oggdec.c, oggdec.h                    David Conrad
   oggenc.c                              Baptiste Coudurier
@@ -461,19 +435,15 @@ Muxers/Demuxers:
   rmdec.c, rmenc.c                      Ronald S. Bultje, Kostya Shishkov
   rtmp*                                 Kostya Shishkov
   rtp.c, rtpenc.c                       Martin Storsjo
-  rtpdec_h261.*, rtpenc_h261.*          Thomas Volkert
-  rtpdec_hevc.*                         Thomas Volkert
   rtpdec_asf.*                          Ronald S. Bultje
   rtpenc_mpv.*, rtpenc_aac.*            Martin Storsjo
   rtsp.c                                Luca Barbato
   sbgdec.c                              Nicolas George
   sdp.c                                 Martin Storsjo
   segafilm.c                            Mike Melanson
-  segment.c                             Stefano Sabatini
   siff.c                                Kostya Shishkov
   smacker.c                             Kostya Shishkov
   smjpeg*                               Paul B Mahol
-  spdif*                                Anssi Hannula
   srtdec.c                              Aurelien Jacobs
   swf.c                                 Baptiste Coudurier
   takdec.c                              Paul B Mahol
@@ -482,7 +452,6 @@ Muxers/Demuxers:
   voc.c                                 Aurelien Jacobs
   wav.c                                 Michael Niedermayer
   wc3movie.c                            Mike Melanson
-  webm dash (matroskaenc.c)             Vignesh Venkatasubramanian
   webvtt*                               Matthew J Heaney
   westwood.c                            Mike Melanson
   wtv.c                                 Peter Ross
@@ -524,8 +493,6 @@ Amiga / PowerPC                         Colin Ward
 Linux / PowerPC                         Luca Barbato
 Windows MinGW                           Alex Beregszaszi, Ramiro Polla
 Windows Cygwin                          Victor Paesa
-Windows MSVC                            Matthew Oliver
-Windows ICL                             Matthew Oliver
 ADI/Blackfin DSP                        Marc Hoffman
 Sparc                                   Roman Shaposhnik
 x86                                     Michael Niedermayer
@@ -534,9 +501,8 @@ x86                                     Michael Niedermayer
 Releases
 ========
 
-2.4                                     Michael Niedermayer
-2.2                                     Michael Niedermayer
-1.2                                     Michael Niedermayer
+2.1                                     Michael Niedermayer
+2.0                                     Michael Niedermayer
 
 If you want to maintain an older release, please contact us
 
@@ -552,7 +518,7 @@ Attila Kinali                 11F0 F9A6 A1D2 11F6 C745 D10C 6520 BCDD F2DF E765
 Baptiste Coudurier            8D77 134D 20CC 9220 201F C5DB 0AC9 325C 5C1A BAAA
 Ben Littler                   3EE3 3723 E560 3214 A8CD 4DEB 2CDB FCE7 768C 8D2C
 Benoit Fouet                  B22A 4F4F 43EF 636B BB66 FCDC 0023 AE1E 2985 49C8
-Clément Bœsch                 52D0 3A82 D445 F194 DB8B 2B16 87EE 2CB8 F4B8 FCF9
+Bœsch Clément                 52D0 3A82 D445 F194 DB8B 2B16 87EE 2CB8 F4B8 FCF9
 Daniel Verkamp                78A6 07ED 782C 653E C628 B8B9 F0EB 8DD8 2F0E 21C7
 Diego Biurrun                 8227 1E31 B6D9 4994 7427 E220 9CAE D6CC 4757 FCC5
 FFmpeg release signing key    FCF9 86EA 15E6 E293 A564 4F10 B432 2F04 D676 58D8
@@ -567,14 +533,12 @@ Michael Niedermayer           9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
 Nicolas George                24CE 01CE 9ACC 5CEB 74D8 8D9D B063 D997 36E5 4C93
 Panagiotis Issaris            6571 13A3 33D9 3726 F728 AA98 F643 B12E ECF3 E029
 Peter Ross                    A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B
-Reimar Doeffinger             C61D 16E5 9E2C D10C 8958 38A4 0899 A2B9 06D4 D9C7
+Reimar Döffinger              C61D 16E5 9E2C D10C 8958 38A4 0899 A2B9 06D4 D9C7
 Reinhard Tartler              9300 5DC2 7E87 6C37 ED7B CA9A 9808 3544 9453 48A4
 Reynaldo H. Verdejo Pinochet  6E27 CD34 170C C78E 4D4F 5F40 C18E 077F 3114 452A
 Robert Swain                  EE7A 56EA 4A81 A7B5 2001 A521 67FA 362D A2FC 3E71
 Sascha Sommer                 38A0 F88B 868E 9D3A 97D4 D6A0 E823 706F 1E07 0D3C
 Stefano Sabatini              0D0B AD6B 5330 BBAD D3D6 6A0C 719C 2839 FC43 2D5F
 Stephan Hilb                  4F38 0B3A 5F39 B99B F505 E562 8D5C 5554 4E17 8863
-Tiancheng "Timothy" Gu        9456 AFC0 814A 8139 E994 8351 7FE6 B095 B582 B0D4
-Tim Nicholson                 38CF DB09 3ED0 F607 8B67 6CED 0C0B FC44 8B0B FC83
 Tomas Härdin                  A79D 4E3D F38F 763F 91F5 8B33 A01E 8AE0 41BB 2551
 Wei Gao                       4269 7741 857A 0E60 9EC5 08D2 4744 4EFA 62C1 87B9

Makefile

@@ -4,49 +4,39 @@ include config.mak
 vpath %.c    $(SRC_PATH)
 vpath %.cpp  $(SRC_PATH)
 vpath %.h    $(SRC_PATH)
-vpath %.m    $(SRC_PATH)
 vpath %.S    $(SRC_PATH)
 vpath %.asm  $(SRC_PATH)
-vpath %.rc   $(SRC_PATH)
 vpath %.v    $(SRC_PATH)
 vpath %.texi $(SRC_PATH)
 vpath %/fate_config.sh.template $(SRC_PATH)
 
-AVPROGS-$(CONFIG_FFMPEG)   += ffmpeg
-AVPROGS-$(CONFIG_FFPLAY)   += ffplay
-AVPROGS-$(CONFIG_FFPROBE)  += ffprobe
-AVPROGS-$(CONFIG_FFSERVER) += ffserver
+PROGS-$(CONFIG_FFMPEG)   += ffmpeg
+PROGS-$(CONFIG_FFPLAY)   += ffplay
+PROGS-$(CONFIG_FFPROBE)  += ffprobe
+PROGS-$(CONFIG_FFSERVER) += ffserver
 
-AVPROGS    := $(AVPROGS-yes:%=%$(PROGSSUF)$(EXESUF))
-INSTPROGS   = $(AVPROGS-yes:%=%$(PROGSSUF)$(EXESUF))
-PROGS      += $(AVPROGS)
-
-AVBASENAMES  = ffmpeg ffplay ffprobe ffserver
-ALLAVPROGS   = $(AVBASENAMES:%=%$(PROGSSUF)$(EXESUF))
-ALLAVPROGS_G = $(AVBASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
-
-$(foreach prog,$(AVBASENAMES),$(eval OBJS-$(prog) += cmdutils.o))
-$(foreach prog,$(AVBASENAMES),$(eval OBJS-$(prog)-$(CONFIG_OPENCL) += cmdutils_opencl.o))
-
-OBJS-ffmpeg                   += ffmpeg_opt.o ffmpeg_filter.o
-OBJS-ffmpeg-$(HAVE_VDPAU_X11) += ffmpeg_vdpau.o
-OBJS-ffmpeg-$(HAVE_DXVA2_LIB) += ffmpeg_dxva2.o
-OBJS-ffmpeg-$(CONFIG_VDA)     += ffmpeg_vda.o
+PROGS      := $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))
+INSTPROGS   = $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))
 
+OBJS        = cmdutils.o $(EXEOBJS)
+OBJS-ffmpeg = ffmpeg_opt.o ffmpeg_filter.o
 TESTTOOLS   = audiogen videogen rotozoom tiny_psnr tiny_ssim base64
 HOSTPROGS  := $(TESTTOOLS:%=tests/%) doc/print_options
-TOOLS       = qt-faststart trasher uncoded_frame
+TOOLS       = qt-faststart trasher
 TOOLS-$(CONFIG_ZLIB) += cws2fws
 
-# $(FFLIBS-yes) needs to be in linking order
-FFLIBS-$(CONFIG_AVDEVICE)   += avdevice
-FFLIBS-$(CONFIG_AVFILTER)   += avfilter
-FFLIBS-$(CONFIG_AVFORMAT)   += avformat
-FFLIBS-$(CONFIG_AVCODEC)    += avcodec
+BASENAMES   = ffmpeg ffplay ffprobe ffserver
+ALLPROGS    = $(BASENAMES:%=%$(PROGSSUF)$(EXESUF))
+ALLPROGS_G  = $(BASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
+
+FFLIBS-$(CONFIG_AVDEVICE) += avdevice
+FFLIBS-$(CONFIG_AVFILTER) += avfilter
+FFLIBS-$(CONFIG_AVFORMAT) += avformat
 FFLIBS-$(CONFIG_AVRESAMPLE) += avresample
-FFLIBS-$(CONFIG_POSTPROC)   += postproc
-FFLIBS-$(CONFIG_SWRESAMPLE) += swresample
-FFLIBS-$(CONFIG_SWSCALE)    += swscale
+FFLIBS-$(CONFIG_AVCODEC)  += avcodec
+FFLIBS-$(CONFIG_POSTPROC) += postproc
+FFLIBS-$(CONFIG_SWRESAMPLE)+= swresample
+FFLIBS-$(CONFIG_SWSCALE)  += swscale
 
 FFLIBS := avutil
 
@@ -60,14 +50,16 @@ include $(SRC_PATH)/common.mak
 FF_EXTRALIBS := $(FFEXTRALIBS)
 FF_DEP_LIBS  := $(DEP_LIBS)
 
-all: $(AVPROGS)
+all: $(PROGS)
+
+$(PROGS): %$(EXESUF): %_g$(EXESUF)
+	$(CP) $< $@
+	$(STRIP) $@
 
 $(TOOLS): %$(EXESUF): %.o $(EXEOBJS)
-	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS)
+	$(LD) $(LDFLAGS) $(LD_O) $^ $(ELIBS)
 
 tools/cws2fws$(EXESUF): ELIBS = $(ZLIB)
-tools/uncoded_frame$(EXESUF): $(FF_DEP_LIBS)
-tools/uncoded_frame$(EXESUF): ELIBS = $(FF_EXTRALIBS)
 
 config.h: .config
 .config: $(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c))
@@ -77,10 +69,11 @@ config.h: .config
 
 SUBDIR_VARS := CLEANFILES EXAMPLES FFLIBS HOSTPROGS TESTPROGS TOOLS      \
                HEADERS ARCH_HEADERS BUILT_HEADERS SKIPHEADERS            \
-               ARMV5TE-OBJS ARMV6-OBJS ARMV8-OBJS VFP-OBJS NEON-OBJS     \
-               ALTIVEC-OBJS MMX-OBJS YASM-OBJS                           \
+               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
+               OBJS HOSTOBJS TESTOBJS
 
 define RESET
 $(1) :=
@@ -92,16 +85,13 @@ $(foreach V,$(SUBDIR_VARS),$(eval $(call RESET,$(V))))
 SUBDIR := $(1)/
 include $(SRC_PATH)/$(1)/Makefile
 -include $(SRC_PATH)/$(1)/$(ARCH)/Makefile
--include $(SRC_PATH)/$(1)/$(INTRINSICS)/Makefile
 include $(SRC_PATH)/library.mak
 endef
 
 $(foreach D,$(FFLIBS),$(eval $(call DOSUBDIR,lib$(D))))
 
-include $(SRC_PATH)/doc/Makefile
-
 define DOPROG
-OBJS-$(1) += $(1).o $(EXEOBJS) $(OBJS-$(1)-yes)
+OBJS-$(1) += $(1).o cmdutils.o $(EXEOBJS)
 $(1)$(PROGSSUF)_g$(EXESUF): $$(OBJS-$(1))
 $$(OBJS-$(1)): CFLAGS  += $(CFLAGS-$(1))
 $(1)$(PROGSSUF)_g$(EXESUF): LDFLAGS += $(LDFLAGS-$(1))
@@ -109,16 +99,10 @@ $(1)$(PROGSSUF)_g$(EXESUF): FF_EXTRALIBS += $(LIBS-$(1))
 -include $$(OBJS-$(1):.o=.d)
 endef
 
-$(foreach P,$(PROGS),$(eval $(call DOPROG,$(P:$(PROGSSUF)$(EXESUF)=))))
-
-ffprobe.o cmdutils.o libavcodec/utils.o libavformat/utils.o libavdevice/avdevice.o libavfilter/avfilter.o libavutil/utils.o libpostproc/postprocess.o libswresample/swresample.o libswscale/utils.o : libavutil/ffversion.h
-
-$(PROGS): %$(PROGSSUF)$(EXESUF): %$(PROGSSUF)_g$(EXESUF)
-	$(CP) $< $@
-	$(STRIP) $@
+$(foreach P,$(PROGS-yes),$(eval $(call DOPROG,$(P))))
 
 %$(PROGSSUF)_g$(EXESUF): %.o $(FF_DEP_LIBS)
-	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
+	$(LD) $(LDFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
 
 OBJDIRS += tools
 
@@ -130,14 +114,14 @@ GIT_LOG     = $(SRC_PATH)/.git/logs/HEAD
 .version: $(wildcard $(GIT_LOG)) $(VERSION_SH) config.mak
 .version: M=@
 
-libavutil/ffversion.h .version:
-	$(M)$(VERSION_SH) $(SRC_PATH) libavutil/ffversion.h $(EXTRA_VERSION)
+version.h .version:
+	$(M)$(VERSION_SH) $(SRC_PATH) version.h $(EXTRA_VERSION)
 	$(Q)touch .version
 
 # force version.sh to run whenever version might have changed
 -include .version
 
-ifdef AVPROGS
+ifdef PROGS
 install: install-progs install-data
 endif
 
@@ -148,7 +132,7 @@ install-libs: install-libs-yes
 install-progs-yes:
 install-progs-$(CONFIG_SHARED): install-libs
 
-install-progs: install-progs-yes $(AVPROGS)
+install-progs: install-progs-yes $(PROGS)
 	$(Q)mkdir -p "$(BINDIR)"
 	$(INSTALL) -c -m 755 $(INSTPROGS) "$(BINDIR)"
 
@@ -160,13 +144,13 @@ install-data: $(DATA_FILES) $(EXAMPLES_FILES)
 uninstall: uninstall-libs uninstall-headers uninstall-progs uninstall-data
 
 uninstall-progs:
-	$(RM) $(addprefix "$(BINDIR)/", $(ALLAVPROGS))
+	$(RM) $(addprefix "$(BINDIR)/", $(ALLPROGS))
 
 uninstall-data:
 	$(RM) -r "$(DATADIR)"
 
 clean::
-	$(RM) $(ALLAVPROGS) $(ALLAVPROGS_G)
+	$(RM) $(ALLPROGS) $(ALLPROGS_G)
 	$(RM) $(CLEANSUFFIXES)
 	$(RM) $(CLEANSUFFIXES:%=tools/%)
 	$(RM) -r coverage-html
@@ -174,13 +158,14 @@ clean::
 
 distclean::
 	$(RM) $(DISTCLEANSUFFIXES)
-	$(RM) config.* .config libavutil/avconfig.h .version version.h libavutil/ffversion.h libavcodec/codec_names.h
+	$(RM) config.* .config libavutil/avconfig.h .version version.h libavcodec/codec_names.h
 
 config:
 	$(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION)
 
 check: all alltools examples testprogs fate
 
+include $(SRC_PATH)/doc/Makefile
 include $(SRC_PATH)/tests/Makefile
 
 $(sort $(OBJDIRS)):

README

@@ -0,0 +1,18 @@
+FFmpeg README
+-------------
+
+1) Documentation
+----------------
+
+* Read the documentation in the doc/ directory in git.
+  You can also view it online at http://ffmpeg.org/documentation.html
+
+2) Licensing
+------------
+
+* See the LICENSE file.
+
+3) Build and Install
+--------------------
+
+* See the INSTALL file.

README.md

@@ -1,40 +0,0 @@
-FFmpeg README
-=============
-
-FFmpeg is a collection of libraries and tools to process multimedia content
-such as audio, video, subtitles and related metadata.
-
-## Libraries
-
-* `libavcodec` provides implementation of a wider range of codecs.
-* `libavformat` implements streaming protocols, container formats and basic I/O access.
-* `libavutil` includes hashers, decompressors and miscellaneous utility functions.
-* `libavfilter` provides a mean to alter decoded Audio and Video through chain of filters.
-* `libavdevice` provides an abstraction to access capture and playback devices.
-* `libswresample` implements audio mixing and resampling routines.
-* `libswscale` implements color conversion and scaling routines.
-
-## Tools
-
-* [ffmpeg](http://ffmpeg.org/ffmpeg.html) is a command line toolbox to
-  manipulate, convert and stream multimedia content.
-* [ffplay](http://ffmpeg.org/ffplay.html) is a minimalistic multimedia player.
-* [ffprobe](http://ffmpeg.org/ffprobe.html) is a simple analisys tool to inspect
-  multimedia content.
-* Additional small tools such as `aviocat`, `ismindex` and `qt-faststart`.
-
-## Documentation
-
-The offline documentation is available in the **doc/** directory.
-
-The online documentation is available in the main [website](http://ffmpeg.org)
-and in the [wiki](http://trac.ffmpeg.org).
-
-### Examples
-
-Conding examples are available in the **doc/example** directory.
-
-## License
-
-FFmpeg codebase is mainly LGPL-licensed with optional components licensed under
-GPL. Please refer to the LICENSE file for detailed information.

RELEASE

@@ -1 +1 @@
-2.4.7
+2.1

RELEASE_NOTES

@@ -1,83 +0,0 @@
- ┌────────────────────────────────────────┐
- │ RELEASE NOTES for FFmpeg 2.4 "Fresnel" │
- └────────────────────────────────────────┘
-
-   The FFmpeg Project proudly presents FFmpeg 2.4 "Fresnel", just 2 months
-   after the release of 2.3. Since this wasn't a long time ago, the Changelog
-   is a bit short this time.
-
-   The most important thing in this release is the major version bump of the
-   libraries.  This means that this release is neither ABI-compatible nor
-   fully API-compatible. But on the other hand it is aligned with the Libav
-   11 release series, and will as a result probably end up being maintained for
-   a long time.
-
-   As usual, if you have any question on this release or any FFmpeg related
-   topic, feel free to join us on the #ffmpeg IRC channel (on
-   irc.freenode.net).
-
-   ┌────────────────────────────┐
-   │ 🔨  API Information         │
-   └────────────────────────────┘
-
-     FFmpeg 2.4 includes the following library versions:
-
-       • libavutil      54.7.100
-       • libavcodec     56.1.100
-       • libavformat    56.4.101
-       • libavdevice    56.0.100
-       • libavfilter     5.1.100
-       • libswscale      3.0.100
-       • libswresample   1.1.100
-       • libpostproc    53.0.100
-
-     Important API changes since 2.3:
-
-       • The new field mime_type was added to AVProbeData, which can
-         cause crashes, if it is not initialized.
-       • Some deprecated functions were removed.
-       • The avfilter_graph_parse function was made compatible with Libav.
-       • The Matroska demuxer now outputs verbatim ASS packets.
-
-     Please refer to the doc/APIchanges file for more information.
-
- ┌────────────────────────────┐
- │ ★  List of New Features    │
- └────────────────────────────┘
-
-   ┌────────────────────────────┐
-   │ libavformat                │
-   └────────────────────────────┘
-
-    • Icecast protocol.
-    • API for live metadata updates through event flags.
-    • UTF-16 support in text subtitles formats.
-    • The ASS muxer now reorders the Dialogue events properly.
-    • support for H.261 RTP payload format (RFC 4587)
-    • HEVC/H.265 RTP payload format (draft v6) depacketizer
-
-   ┌────────────────────────────┐
-   │ libavfilter                │
-   └────────────────────────────┘
-
-    • Ported lenscorrection filter from frei0r filter.
-    • Large optimizations in dctdnoiz to make it usable.
-    • Added codecview filter to visualize information exported by some codecs.
-    • Added silenceremove filter.
-
-   ┌────────────────────────────┐
-   │ libavutil                  │
-   └────────────────────────────┘
-
-    • Added clip() function in eval.
-
- ┌────────────────────────────┐
- │ ⚠  Behaviour changes       │
- └────────────────────────────┘
-
-  • dctdnoiz filter now uses a block size of 8x8 instead of 16x16 by default
-  • -vismv option is deprecated in favor of the codecview filter
-  • libmodplug is now detected through pkg-config
-  • HTML documentation generation through texi2html is deprecated in
-    favor of makeinfo/texi2any
-  • ICY metadata are now requested by default with the HTTP protocol

VERSION

@@ -0,0 +1 @@
+2.1

arch.mak

@@ -1,6 +1,5 @@
 OBJS-$(HAVE_ARMV5TE) += $(ARMV5TE-OBJS) $(ARMV5TE-OBJS-yes)
 OBJS-$(HAVE_ARMV6)   += $(ARMV6-OBJS)   $(ARMV6-OBJS-yes)
-OBJS-$(HAVE_ARMV8)   += $(ARMV8-OBJS)   $(ARMV8-OBJS-yes)
 OBJS-$(HAVE_VFP)     += $(VFP-OBJS)     $(VFP-OBJS-yes)
 OBJS-$(HAVE_NEON)    += $(NEON-OBJS)    $(NEON-OBJS-yes)
 
@@ -11,5 +10,7 @@ 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.c

@@ -20,7 +20,6 @@
  */
 
 #include <string.h>
-#include <stdint.h>
 #include <stdlib.h>
 #include <errno.h>
 #include <math.h>
@@ -49,8 +48,8 @@
 #include "libavutil/dict.h"
 #include "libavutil/opt.h"
 #include "libavutil/cpu.h"
-#include "libavutil/ffversion.h"
 #include "cmdutils.h"
+#include "version.h"
 #if CONFIG_NETWORK
 #include "libavformat/network.h"
 #endif
@@ -58,6 +57,10 @@
 #include <sys/time.h>
 #include <sys/resource.h>
 #endif
+#if CONFIG_OPENCL
+#include "libavutil/opencl.h"
+#endif
+
 
 static int init_report(const char *env);
 
@@ -65,9 +68,9 @@ struct SwsContext *sws_opts;
 AVDictionary *swr_opts;
 AVDictionary *format_opts, *codec_opts, *resample_opts;
 
+const int this_year = 2013;
+
 static FILE *report_file;
-static int report_file_level = AV_LOG_DEBUG;
-int hide_banner = 0;
 
 void init_opts(void)
 {
@@ -105,10 +108,8 @@ static void log_callback_report(void *ptr, int level, const char *fmt, va_list v
     av_log_default_callback(ptr, level, fmt, vl);
     av_log_format_line(ptr, level, fmt, vl2, line, sizeof(line), &print_prefix);
     va_end(vl2);
-    if (report_file_level >= level) {
-        fputs(line, report_file);
-        fflush(report_file);
-    }
+    fputs(line, report_file);
+    fflush(report_file);
 }
 
 static void (*program_exit)(int ret);
@@ -166,7 +167,7 @@ void show_help_options(const OptionDef *options, const char *msg, int req_flags,
     int first;
 
     first = 1;
-    for (po = options; po->name; po++) {
+    for (po = options; po->name != NULL; po++) {
         char buf[64];
 
         if (((po->flags & req_flags) != req_flags) ||
@@ -205,7 +206,7 @@ static const OptionDef *find_option(const OptionDef *po, const char *name)
     const char *p = strchr(name, ':');
     int len = p ? p - name : strlen(name);
 
-    while (po->name) {
+    while (po->name != NULL) {
         if (!strncmp(name, po->name, len) && strlen(po->name) == len)
             break;
         po++;
@@ -254,7 +255,7 @@ static void prepare_app_arguments(int *argc_ptr, char ***argv_ptr)
 
     win32_argv_utf8 = av_mallocz(sizeof(char *) * (win32_argc + 1) + buffsize);
     argstr_flat     = (char *)win32_argv_utf8 + sizeof(char *) * (win32_argc + 1);
-    if (!win32_argv_utf8) {
+    if (win32_argv_utf8 == NULL) {
         LocalFree(argv_w);
         return;
     }
@@ -444,7 +445,7 @@ int locate_option(int argc, char **argv, const OptionDef *options,
              (po->name && !strcmp(optname, po->name)))
             return i;
 
-        if (!po->name || po->flags & HAS_ARG)
+        if (po->flags & HAS_ARG)
             i++;
     }
     return 0;
@@ -495,9 +496,6 @@ void parse_loglevel(int argc, char **argv, const OptionDef *options)
             fflush(report_file);
         }
     }
-    idx = locate_option(argc, argv, options, "hide_banner");
-    if (idx)
-        hide_banner = 1;
 }
 
 static const AVOption *opt_find(void *obj, const char *name, const char *unit,
@@ -555,11 +553,6 @@ int opt_default(void *optctx, const char *opt, const char *arg)
         }
         consumed = 1;
     }
-#else
-    if (!consumed && !strcmp(opt, "sws_flags")) {
-        av_log(NULL, AV_LOG_WARNING, "Ignoring %s %s, due to disabled swscale\n", opt, arg);
-        consumed = 1;
-    }
 #endif
 #if CONFIG_SWRESAMPLE
     swr_class = swr_get_class();
@@ -670,7 +663,7 @@ static void init_parse_context(OptionParseContext *octx,
     memset(octx, 0, sizeof(*octx));
 
     octx->nb_groups = nb_groups;
-    octx->groups    = av_mallocz_array(octx->nb_groups, sizeof(*octx->groups));
+    octx->groups    = av_mallocz(sizeof(*octx->groups) * octx->nb_groups);
     if (!octx->groups)
         exit_program(1);
 
@@ -842,17 +835,10 @@ int opt_loglevel(void *optctx, const char *opt, const char *arg)
     };
     char *tail;
     int level;
-    int flags;
     int i;
 
-    flags = av_log_get_flags();
     tail = strstr(arg, "repeat");
-    if (tail)
-        flags &= ~AV_LOG_SKIP_REPEATED;
-    else
-        flags |= AV_LOG_SKIP_REPEATED;
-
-    av_log_set_flags(flags);
+    av_log_set_flags(tail ? 0 : AV_LOG_SKIP_REPEATED);
     if (tail == arg)
         arg += 6 + (arg[6]=='+');
     if(tail && !*arg)
@@ -934,13 +920,6 @@ static int init_report(const char *env)
             av_free(filename_template);
             filename_template = val;
             val = NULL;
-        } else if (!strcmp(key, "level")) {
-            char *tail;
-            report_file_level = strtol(val, &tail, 10);
-            if (*tail) {
-                av_log(NULL, AV_LOG_FATAL, "Invalid report file level\n");
-                exit_program(1);
-            }
         } else {
             av_log(NULL, AV_LOG_ERROR, "Unknown key '%s' in FFREPORT\n", key);
         }
@@ -1007,6 +986,26 @@ int opt_timelimit(void *optctx, const char *opt, const char *arg)
     return 0;
 }
 
+#if CONFIG_OPENCL
+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;
+}
+#endif
+
 void print_error(const char *filename, int err)
 {
     char errbuf[128];
@@ -1072,7 +1071,7 @@ static void print_program_info(int flags, int level)
     av_log(NULL, level, "%s version " FFMPEG_VERSION, program_name);
     if (flags & SHOW_COPYRIGHT)
         av_log(NULL, level, " Copyright (c) %d-%d the FFmpeg developers",
-               program_birth_year, CONFIG_THIS_YEAR);
+               program_birth_year, this_year);
     av_log(NULL, level, "\n");
     av_log(NULL, level, "%sbuilt on %s %s with %s\n",
            indent, __DATE__, __TIME__, CC_IDENT);
@@ -1080,36 +1079,10 @@ static void print_program_info(int flags, int level)
     av_log(NULL, level, "%sconfiguration: " FFMPEG_CONFIGURATION "\n", indent);
 }
 
-static void print_buildconf(int flags, int level)
-{
-    const char *indent = flags & INDENT ? "  " : "";
-    char str[] = { FFMPEG_CONFIGURATION };
-    char *conflist, *remove_tilde, *splitconf;
-
-    // Change all the ' --' strings to '~--' so that
-    // they can be identified as tokens.
-    while ((conflist = strstr(str, " --")) != NULL) {
-        strncpy(conflist, "~--", 3);
-    }
-
-    // Compensate for the weirdness this would cause
-    // when passing 'pkg-config --static'.
-    while ((remove_tilde = strstr(str, "pkg-config~")) != NULL) {
-        strncpy(remove_tilde, "pkg-config ", 11);
-    }
-
-    splitconf = strtok(str, "~");
-    av_log(NULL, level, "\n%sconfiguration:\n", indent);
-    while (splitconf != NULL) {
-        av_log(NULL, level, "%s%s%s\n", indent, indent, splitconf);
-        splitconf = strtok(NULL, "~");
-    }
-}
-
 void show_banner(int argc, char **argv, const OptionDef *options)
 {
     int idx = locate_option(argc, argv, options, "version");
-    if (hide_banner || idx)
+    if (idx)
         return;
 
     print_program_info (INDENT|SHOW_COPYRIGHT, AV_LOG_INFO);
@@ -1120,20 +1093,12 @@ void show_banner(int argc, char **argv, const OptionDef *options)
 int show_version(void *optctx, const char *opt, const char *arg)
 {
     av_log_set_callback(log_callback_help);
-    print_program_info (SHOW_COPYRIGHT, AV_LOG_INFO);
+    print_program_info (0           , AV_LOG_INFO);
     print_all_libs_info(SHOW_VERSION, AV_LOG_INFO);
 
     return 0;
 }
 
-int show_buildconf(void *optctx, const char *opt, const char *arg)
-{
-    av_log_set_callback(log_callback_help);
-    print_buildconf      (INDENT|0, AV_LOG_INFO);
-
-    return 0;
-}
-
 int show_license(void *optctx, const char *opt, const char *arg)
 {
 #if CONFIG_NONFREE
@@ -1208,29 +1173,16 @@ int show_license(void *optctx, const char *opt, const char *arg)
     return 0;
 }
 
-static int is_device(const AVClass *avclass)
-{
-    if (!avclass)
-        return 0;
-    return avclass->category == AV_CLASS_CATEGORY_DEVICE_VIDEO_OUTPUT ||
-           avclass->category == AV_CLASS_CATEGORY_DEVICE_VIDEO_INPUT ||
-           avclass->category == AV_CLASS_CATEGORY_DEVICE_AUDIO_OUTPUT ||
-           avclass->category == AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT ||
-           avclass->category == AV_CLASS_CATEGORY_DEVICE_OUTPUT ||
-           avclass->category == AV_CLASS_CATEGORY_DEVICE_INPUT;
-}
-
-static int show_formats_devices(void *optctx, const char *opt, const char *arg, int device_only)
+int show_formats(void *optctx, const char *opt, const char *arg)
 {
     AVInputFormat *ifmt  = NULL;
     AVOutputFormat *ofmt = NULL;
     const char *last_name;
-    int is_dev;
 
-    printf("%s\n"
+    printf("File formats:\n"
            " D. = Demuxing supported\n"
            " .E = Muxing supported\n"
-           " --\n", device_only ? "Devices:" : "File formats:");
+           " --\n");
     last_name = "000";
     for (;;) {
         int decode = 0;
@@ -1239,10 +1191,7 @@ static int show_formats_devices(void *optctx, const char *opt, const char *arg,
         const char *long_name = NULL;
 
         while ((ofmt = av_oformat_next(ofmt))) {
-            is_dev = is_device(ofmt->priv_class);
-            if (!is_dev && device_only)
-                continue;
-            if ((!name || strcmp(ofmt->name, name) < 0) &&
+            if ((name == NULL || strcmp(ofmt->name, name) < 0) &&
                 strcmp(ofmt->name, last_name) > 0) {
                 name      = ofmt->name;
                 long_name = ofmt->long_name;
@@ -1250,10 +1199,7 @@ static int show_formats_devices(void *optctx, const char *opt, const char *arg,
             }
         }
         while ((ifmt = av_iformat_next(ifmt))) {
-            is_dev = is_device(ifmt->priv_class);
-            if (!is_dev && device_only)
-                continue;
-            if ((!name || strcmp(ifmt->name, name) < 0) &&
+            if ((name == NULL || strcmp(ifmt->name, name) < 0) &&
                 strcmp(ifmt->name, last_name) > 0) {
                 name      = ifmt->name;
                 long_name = ifmt->long_name;
@@ -1262,7 +1208,7 @@ static int show_formats_devices(void *optctx, const char *opt, const char *arg,
             if (name && strcmp(ifmt->name, name) == 0)
                 decode = 1;
         }
-        if (!name)
+        if (name == NULL)
             break;
         last_name = name;
 
@@ -1275,16 +1221,6 @@ static int show_formats_devices(void *optctx, const char *opt, const char *arg,
     return 0;
 }
 
-int show_formats(void *optctx, const char *opt, const char *arg)
-{
-    return show_formats_devices(optctx, opt, arg, 0);
-}
-
-int show_devices(void *optctx, const char *opt, const char *arg)
-{
-    return show_formats_devices(optctx, opt, arg, 1);
-}
-
 #define PRINT_CODEC_SUPPORTED(codec, field, type, list_name, term, get_name) \
     if (codec->field) {                                                      \
         const type *p = codec->field;                                        \
@@ -1429,9 +1365,6 @@ int show_codecs(void *optctx, const char *opt, const char *arg)
         const AVCodecDescriptor *desc = codecs[i];
         const AVCodec *codec = NULL;
 
-        if (strstr(desc->name, "_deprecated"))
-            continue;
-
         printf(" ");
         printf(avcodec_find_decoder(desc->id) ? "D" : ".");
         printf(avcodec_find_encoder(desc->id) ? "E" : ".");
@@ -1585,7 +1518,7 @@ int show_filters(void *optctx, const char *opt, const char *arg)
     return 0;
 }
 
-int show_colors(void *optctx, const char *opt, const char *arg)
+void show_colors(void *optctx, const char *opt, const char *arg)
 {
     const char *name;
     const uint8_t *rgb;
@@ -1595,8 +1528,6 @@ int show_colors(void *optctx, const char *opt, const char *arg)
 
     for (i = 0; name = av_get_known_color_name(i, &rgb); i++)
         printf("%-32s #%02x%02x%02x\n", name, rgb[0], rgb[1], rgb[2]);
-
-    return 0;
 }
 
 int show_pix_fmts(void *optctx, const char *opt, const char *arg)
@@ -1639,19 +1570,19 @@ int show_layouts(void *optctx, const char *opt, const char *arg)
     const char *name, *descr;
 
     printf("Individual channels:\n"
-           "NAME           DESCRIPTION\n");
+           "NAME        DESCRIPTION\n");
     for (i = 0; i < 63; i++) {
         name = av_get_channel_name((uint64_t)1 << i);
         if (!name)
             continue;
         descr = av_get_channel_description((uint64_t)1 << i);
-        printf("%-14s %s\n", name, descr);
+        printf("%-12s%s\n", name, descr);
     }
     printf("\nStandard channel layouts:\n"
-           "NAME           DECOMPOSITION\n");
+           "NAME        DECOMPOSITION\n");
     for (i = 0; !av_get_standard_channel_layout(i, &layout, &name); i++) {
         if (name) {
-            printf("%-14s ", name);
+            printf("%-12s", name);
             for (j = 1; j; j <<= 1)
                 if ((layout & j))
                     printf("%s%s", (layout & (j - 1)) ? "+" : "", av_get_channel_name(j));
@@ -1857,39 +1788,27 @@ int read_yesno(void)
 
 int cmdutils_read_file(const char *filename, char **bufptr, size_t *size)
 {
-    int64_t ret;
-    FILE *f = av_fopen_utf8(filename, "rb");
+    int ret;
+    FILE *f = fopen(filename, "rb");
 
     if (!f) {
         av_log(NULL, AV_LOG_ERROR, "Cannot read file '%s': %s\n", filename,
                strerror(errno));
         return AVERROR(errno);
     }
-
-    ret = fseek(f, 0, SEEK_END);
-    if (ret == -1) {
-        ret = AVERROR(errno);
-        goto out;
+    fseek(f, 0, SEEK_END);
+    *size = ftell(f);
+    fseek(f, 0, SEEK_SET);
+    if (*size == (size_t)-1) {
+        av_log(NULL, AV_LOG_ERROR, "IO error: %s\n", strerror(errno));
+        fclose(f);
+        return AVERROR(errno);
     }
-
-    ret = ftell(f);
-    if (ret < 0) {
-        ret = AVERROR(errno);
-        goto out;
-    }
-    *size = ret;
-
-    ret = fseek(f, 0, SEEK_SET);
-    if (ret == -1) {
-        ret = AVERROR(errno);
-        goto out;
-    }
-
     *bufptr = av_malloc(*size + 1);
     if (!*bufptr) {
         av_log(NULL, AV_LOG_ERROR, "Could not allocate file buffer\n");
-        ret = AVERROR(ENOMEM);
-        goto out;
+        fclose(f);
+        return AVERROR(ENOMEM);
     }
     ret = fread(*bufptr, 1, *size, f);
     if (ret < *size) {
@@ -1905,8 +1824,6 @@ int cmdutils_read_file(const char *filename, char **bufptr, size_t *size)
         (*bufptr)[(*size)++] = '\0';
     }
 
-out:
-    av_log(NULL, AV_LOG_ERROR, "IO error: %s\n", av_err2str(ret));
     fclose(f);
     return ret;
 }
@@ -2010,8 +1927,7 @@ AVDictionary *filter_codec_opts(AVDictionary *opts, enum AVCodecID codec_id,
             }
 
         if (av_opt_find(&cc, t->key, NULL, flags, AV_OPT_SEARCH_FAKE_OBJ) ||
-            !codec ||
-            (codec->priv_class &&
+            (codec && codec->priv_class &&
              av_opt_find(&codec->priv_class, t->key, NULL, flags,
                          AV_OPT_SEARCH_FAKE_OBJ)))
             av_dict_set(&ret, t->key, t->value, 0);
@@ -2034,7 +1950,7 @@ AVDictionary **setup_find_stream_info_opts(AVFormatContext *s,
 
     if (!s->nb_streams)
         return NULL;
-    opts = av_mallocz_array(s->nb_streams, sizeof(*opts));
+    opts = av_mallocz(s->nb_streams * sizeof(*opts));
     if (!opts) {
         av_log(NULL, AV_LOG_ERROR,
                "Could not alloc memory for stream options.\n");

cmdutils.h

@@ -24,13 +24,12 @@
 
 #include <stdint.h>
 
-#include "config.h"
 #include "libavcodec/avcodec.h"
 #include "libavfilter/avfilter.h"
 #include "libavformat/avformat.h"
 #include "libswscale/swscale.h"
 
-#ifdef _WIN32
+#ifdef __MINGW32__
 #undef main /* We don't want SDL to override our main() */
 #endif
 
@@ -44,12 +43,16 @@ extern const char program_name[];
  */
 extern const int program_birth_year;
 
+/**
+ * this year, defined by the program for show_banner()
+ */
+extern const int this_year;
+
 extern AVCodecContext *avcodec_opts[AVMEDIA_TYPE_NB];
 extern AVFormatContext *avformat_opts;
 extern struct SwsContext *sws_opts;
 extern AVDictionary *swr_opts;
 extern AVDictionary *format_opts, *codec_opts, *resample_opts;
-extern int hide_banner;
 
 /**
  * Register a program-specific cleanup routine.
@@ -59,7 +62,7 @@ void register_exit(void (*cb)(int ret));
 /**
  * Wraps exit with a program-specific cleanup routine.
  */
-void exit_program(int ret) av_noreturn;
+void exit_program(int ret);
 
 /**
  * Initialize the cmdutils option system, in particular
@@ -100,12 +103,8 @@ int opt_max_alloc(void *optctx, const char *opt, const char *arg);
 
 int opt_codec_debug(void *optctx, const char *opt, const char *arg);
 
-#if CONFIG_OPENCL
 int opt_opencl(void *optctx, const char *opt, const char *arg);
 
-int opt_opencl_bench(void *optctx, const char *opt, const char *arg);
-#endif
-
 /**
  * Limit the execution time.
  */
@@ -415,13 +414,6 @@ void show_banner(int argc, char **argv, const OptionDef *options);
  */
 int show_version(void *optctx, const char *opt, const char *arg);
 
-/**
- * Print the build configuration of the program to stdout. The contents
- * depend on the definition of FFMPEG_CONFIGURATION.
- * This option processing function does not utilize the arguments.
- */
-int show_buildconf(void *optctx, const char *opt, const char *arg);
-
 /**
  * Print the license of the program to stdout. The license depends on
  * the license of the libraries compiled into the program.
@@ -431,17 +423,10 @@ int show_license(void *optctx, const char *opt, const char *arg);
 
 /**
  * Print a listing containing all the formats supported by the
- * program (including devices).
- * This option processing function does not utilize the arguments.
- */
-int show_formats(void *optctx, const char *opt, const char *arg);
-
-/**
- * Print a listing containing all the devices supported by the
  * program.
  * This option processing function does not utilize the arguments.
  */
-int show_devices(void *optctx, const char *opt, const char *arg);
+int show_formats(void *optctx, const char *opt, const char *arg);
 
 /**
  * Print a listing containing all the codecs supported by the
@@ -507,7 +492,7 @@ int show_sample_fmts(void *optctx, const char *opt, const char *arg);
  * Print a listing containing all the color names and values recognized
  * by the program.
  */
-int show_colors(void *optctx, const char *opt, const char *arg);
+void show_colors(void *optctx, const char *opt, const char *arg);
 
 /**
  * Return a positive value if a line read from standard input

cmdutils_common_opts.h

@@ -4,9 +4,7 @@
     { "help"       , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
     { "-help"      , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
     { "version"    , OPT_EXIT, {.func_arg = show_version},      "show version" },
-    { "buildconf"  , OPT_EXIT, {.func_arg = show_buildconf},    "show build configuration" },
     { "formats"    , OPT_EXIT, {.func_arg = show_formats  },    "show available formats" },
-    { "devices"    , OPT_EXIT, {.func_arg = show_devices  },    "show available devices" },
     { "codecs"     , OPT_EXIT, {.func_arg = show_codecs   },    "show available codecs" },
     { "decoders"   , OPT_EXIT, {.func_arg = show_decoders },    "show available decoders" },
     { "encoders"   , OPT_EXIT, {.func_arg = show_encoders },    "show available encoders" },
@@ -22,8 +20,6 @@
     { "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

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_relative();
-    for (i = 0; i < OPENCL_NB_ITER; i++)
-        OCLCHECK(clEnqueueNDRangeKernel, ext_opencl_env->command_queue, kernel, 2, NULL,
-                 global_work_size_2d, local_work_size_2d, 0, NULL, NULL);
-    clFinish(ext_opencl_env->command_queue);
-    ret = (av_gettime_relative() - start)/OPENCL_NB_ITER;
-end:
-    if (kernel)
-        clReleaseKernel(kernel);
-    if (program)
-        clReleaseProgram(program);
-    if (cl_inbuf)
-        clReleaseMemObject(cl_inbuf);
-    if (cl_outbuf)
-        clReleaseMemObject(cl_outbuf);
-    if (cl_mask)
-        clReleaseMemObject(cl_mask);
-    av_free(inbuf);
-    av_free(mask);
-    return ret;
-}
-
-static int compare_ocl_device_desc(const void *a, const void *b)
-{
-    return ((OpenCLDeviceBenchmark*)a)->runtime - ((OpenCLDeviceBenchmark*)b)->runtime;
-}
-
-int opt_opencl_bench(void *optctx, const char *opt, const char *arg)
-{
-    int i, j, nb_devices = 0, count = 0;
-    int64_t score = 0;
-    AVOpenCLDeviceList *device_list;
-    AVOpenCLDeviceNode *device_node = NULL;
-    OpenCLDeviceBenchmark *devices = NULL;
-    cl_platform_id platform;
-
-    av_opencl_get_device_list(&device_list);
-    for (i = 0; i < device_list->platform_num; i++)
-        nb_devices += device_list->platform_node[i]->device_num;
-    if (!nb_devices) {
-        av_log(NULL, AV_LOG_ERROR, "No OpenCL device detected!\n");
-        return AVERROR(EINVAL);
-    }
-    if (!(devices = av_malloc_array(nb_devices, sizeof(OpenCLDeviceBenchmark)))) {
-        av_log(NULL, AV_LOG_ERROR, "Could not allocate buffer\n");
-        return AVERROR(ENOMEM);
-    }
-
-    for (i = 0; i < device_list->platform_num; i++) {
-        for (j = 0; j < device_list->platform_node[i]->device_num; j++) {
-            device_node = device_list->platform_node[i]->device_node[j];
-            platform = device_list->platform_node[i]->platform_id;
-            score = av_opencl_benchmark(device_node, platform, run_opencl_bench);
-            if (score > 0) {
-                devices[count].platform_idx = i;
-                devices[count].device_idx = j;
-                devices[count].runtime = score;
-                strcpy(devices[count].device_name, device_node->device_name);
-                count++;
-            }
-        }
-    }
-    qsort(devices, count, sizeof(OpenCLDeviceBenchmark), compare_ocl_device_desc);
-    fprintf(stderr, "platform_idx\tdevice_idx\tdevice_name\truntime\n");
-    for (i = 0; i < count; i++)
-        fprintf(stdout, "%d\t%d\t%s\t%"PRId64"\n",
-                devices[i].platform_idx, devices[i].device_idx,
-                devices[i].device_name, devices[i].runtime);
-
-    av_opencl_free_device_list(&device_list);
-    av_free(devices);
-    return 0;
-}
-
-int opt_opencl(void *optctx, const char *opt, const char *arg)
-{
-    char *key, *value;
-    const char *opts = arg;
-    int ret = 0;
-    while (*opts) {
-        ret = av_opt_get_key_value(&opts, "=", ":", 0, &key, &value);
-        if (ret < 0)
-            return ret;
-        ret = av_opencl_set_option(key, value);
-        if (ret < 0)
-            return ret;
-        if (*opts)
-            opts++;
-    }
-    return ret;
-}

common.mak

@@ -10,7 +10,7 @@ ifndef SUBDIR
 ifndef V
 Q      = @
 ECHO   = printf "$(1)\t%s\n" $(2)
-BRIEF  = CC CXX HOSTCC HOSTLD AS YASM AR LD STRIP CP WINDRES
+BRIEF  = CC CXX HOSTCC HOSTLD AS YASM AR LD STRIP CP
 SILENT = DEPCC DEPHOSTCC DEPAS DEPYASM RANLIB RM
 
 MSG    = $@
@@ -43,7 +43,6 @@ endef
 COMPILE_C = $(call COMPILE,CC)
 COMPILE_CXX = $(call COMPILE,CXX)
 COMPILE_S = $(call COMPILE,AS)
-COMPILE_HOSTC = $(call COMPILE,HOSTCC)
 
 %.o: %.c
 	$(COMPILE_C)
@@ -51,21 +50,12 @@ COMPILE_HOSTC = $(call COMPILE,HOSTCC)
 %.o: %.cpp
 	$(COMPILE_CXX)
 
-%.o: %.m
-	$(COMPILE_C)
-
 %.s: %.c
 	$(CC) $(CPPFLAGS) $(CFLAGS) -S -o $@ $<
 
 %.o: %.S
 	$(COMPILE_S)
 
-%_host.o: %.c
-	$(COMPILE_HOSTC)
-
-%.o: %.rc
-	$(WINDRES) $(IFLAGS) --preprocessor "$(DEPWINDRES) -E -xc-header -DRC_INVOKED $(CC_DEPFLAGS)" -o $@ $<
-
 %.i: %.c
 	$(CC) $(CCFLAGS) $(CC_E) $<
 
@@ -92,15 +82,14 @@ endif
 include $(SRC_PATH)/arch.mak
 
 OBJS      += $(OBJS-yes)
-SLIBOBJS  += $(SLIBOBJS-yes)
-FFLIBS    := $($(NAME)_FFLIBS) $(FFLIBS-yes) $(FFLIBS)
+FFLIBS    := $(FFLIBS-yes) $(FFLIBS)
 TESTPROGS += $(TESTPROGS-yes)
 
 LDLIBS       = $(FFLIBS:%=%$(BUILDSUF))
 FFEXTRALIBS := $(LDLIBS:%=$(LD_LIB)) $(EXTRALIBS)
 
+EXAMPLES  := $(EXAMPLES:%=$(SUBDIR)%-example$(EXESUF))
 OBJS      := $(sort $(OBJS:%=$(SUBDIR)%))
-SLIBOBJS  := $(sort $(SLIBOBJS:%=$(SUBDIR)%))
 TESTOBJS  := $(TESTOBJS:%=$(SUBDIR)%) $(TESTPROGS:%=$(SUBDIR)%-test.o)
 TESTPROGS := $(TESTPROGS:%=$(SUBDIR)%-test$(EXESUF))
 HOSTOBJS  := $(HOSTPROGS:%=$(SUBDIR)%.o)
@@ -124,19 +113,18 @@ checkheaders: $(HOBJS)
 alltools: $(TOOLS)
 
 $(HOSTOBJS): %.o: %.c
-	$(COMPILE_HOSTC)
+	$(call COMPILE,HOSTCC)
 
 $(HOSTPROGS): %$(HOSTEXESUF): %.o
-	$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $^ $(HOSTLIBS)
+	$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $< $(HOSTLIBS)
 
 $(OBJS):     | $(sort $(dir $(OBJS)))
 $(HOBJS):    | $(sort $(dir $(HOBJS)))
 $(HOSTOBJS): | $(sort $(dir $(HOSTOBJS)))
-$(SLIBOBJS): | $(sort $(dir $(SLIBOBJS)))
 $(TESTOBJS): | $(sort $(dir $(TESTOBJS)))
 $(TOOLOBJS): | tools
 
-OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(SLIBOBJS) $(TESTOBJS))
+OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(TESTOBJS))
 
 CLEANSUFFIXES     = *.d *.o *~ *.h.c *.map *.ver *.ho *.gcno *.gcda
 DISTCLEANSUFFIXES = *.pc
@@ -151,4 +139,4 @@ endef
 
 $(eval $(RULES))
 
--include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d) $(SLIBOBJS:.o=.d))
+-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d))

compat/avisynth/avisynth_c.h

@@ -13,8 +13,7 @@
 //
 // You should have received a copy of the GNU General Public License
 // along with this program; if not, write to the Free Software
-// Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston,
-// MA 02110-1301 USA, or visit
+// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
 // http://www.gnu.org/copyleft/gpl.html .
 //
 // As a special exception, I give you permission to link to the
@@ -805,7 +804,7 @@ struct AVS_Library {
 
 AVSC_INLINE AVS_Library * avs_load_library() {
   AVS_Library *library = (AVS_Library *)malloc(sizeof(AVS_Library));
-  if (!library)
+  if (library == NULL)
     return NULL;
   library->handle = LoadLibrary("avisynth");
   if (library->handle == NULL)
@@ -870,7 +869,7 @@ fail:
 }
 
 AVSC_INLINE void avs_free_library(AVS_Library *library) {
-  if (!library)
+  if (library == NULL)
     return;
   FreeLibrary(library->handle);
   free(library);

compat/avisynth/avxsynth_c.h

@@ -13,8 +13,7 @@
 //
 // You should have received a copy of the GNU General Public License
 // along with this program; if not, write to the Free Software
-// Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston,
-// MA 02110-1301 USA, or visit
+// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
 // http://www.gnu.org/copyleft/gpl.html .
 //
 // As a special exception, I give you permission to link to the

compat/float/float.h

@@ -1,35 +0,0 @@
-/*
- * Work around broken floating point limits on some systems.
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include_next <float.h>
-
-#ifdef FLT_MAX
-#undef  FLT_MAX
-#define FLT_MAX 3.40282346638528859812e+38F
-
-#undef  FLT_MIN
-#define FLT_MIN 1.17549435082228750797e-38F
-
-#undef  DBL_MAX
-#define DBL_MAX ((double)1.79769313486231570815e+308L)
-
-#undef  DBL_MIN
-#define DBL_MIN ((double)2.22507385850720138309e-308L)
-#endif

compat/float/limits.h

@@ -1,22 +0,0 @@
-/*
- * Work around broken floating point limits on some systems.
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include_next <limits.h>
-#include <float.h>

compat/getopt.c

@@ -38,6 +38,8 @@ static int optind = 1;
 static int optopt;
 static char *optarg;
 
+#undef fprintf
+
 static int getopt(int argc, char *argv[], char *opts)
 {
     static int sp = 1;
@@ -54,7 +56,7 @@ static int getopt(int argc, char *argv[], char *opts)
         }
     }
     optopt = c = argv[optind][sp];
-    if (c == ':' || !(cp = strchr(opts, c))) {
+    if (c == ':' || (cp = strchr(opts, c)) == NULL) {
         fprintf(stderr, ": illegal option -- %c\n", c);
         if (argv[optind][++sp] == '\0') {
             optind++;

compat/os2threads.h

@@ -32,8 +32,6 @@
 #undef __STRICT_ANSI__          /* for _beginthread() */
 #include <stdlib.h>
 
-#include "libavutil/mem.h"
-
 typedef TID  pthread_t;
 typedef void pthread_attr_t;
 

compat/va_copy.h

@@ -24,6 +24,3 @@
 #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

@@ -39,7 +39,6 @@
 #include <windows.h>
 #include <process.h>
 
-#include "libavutil/attributes.h"
 #include "libavutil/common.h"
 #include "libavutil/internal.h"
 #include "libavutil/mem.h"
@@ -74,29 +73,17 @@ static BOOL (WINAPI *cond_wait)(pthread_cond_t *cond, pthread_mutex_t *mutex,
 #define cond_broadcast WakeAllConditionVariable
 #define cond_signal    WakeConditionVariable
 #define cond_wait      SleepConditionVariableCS
-
-#define CreateEvent(a, reset, init, name)                   \
-    CreateEventEx(a, name,                                  \
-                  (reset ? CREATE_EVENT_MANUAL_RESET : 0) | \
-                  (init ? CREATE_EVENT_INITIAL_SET : 0),    \
-                  EVENT_ALL_ACCESS)
-// CreateSemaphoreExA seems to be desktop-only, but as long as we don't
-// use named semaphores, it doesn't matter if we use the W version.
-#define CreateSemaphore(a, b, c, d) \
-    CreateSemaphoreExW(a, b, c, d, 0, SEMAPHORE_ALL_ACCESS)
-#define InitializeCriticalSection(x) InitializeCriticalSectionEx(x, 0, 0)
-#define WaitForSingleObject(a, b) WaitForSingleObjectEx(a, b, FALSE)
 #endif
 
-static av_unused unsigned __stdcall attribute_align_arg win32thread_worker(void *arg)
+static unsigned __stdcall attribute_align_arg win32thread_worker(void *arg)
 {
     pthread_t *h = arg;
     h->ret = h->func(h->arg);
     return 0;
 }
 
-static av_unused int pthread_create(pthread_t *thread, const void *unused_attr,
-                                    void *(*start_routine)(void*), void *arg)
+static int pthread_create(pthread_t *thread, const void *unused_attr,
+                          void *(*start_routine)(void*), void *arg)
 {
     thread->func   = start_routine;
     thread->arg    = arg;
@@ -105,7 +92,7 @@ static av_unused int pthread_create(pthread_t *thread, const void *unused_attr,
     return !thread->handle;
 }
 
-static av_unused void pthread_join(pthread_t thread, void **value_ptr)
+static void pthread_join(pthread_t thread, void **value_ptr)
 {
     DWORD ret = WaitForSingleObject(thread.handle, INFINITE);
     if (ret != WAIT_OBJECT_0)
@@ -147,32 +134,31 @@ typedef struct  win32_cond_t {
     volatile int is_broadcast;
 } win32_cond_t;
 
-static av_unused int pthread_cond_init(pthread_cond_t *cond, const void *unused_attr)
+static void pthread_cond_init(pthread_cond_t *cond, const void *unused_attr)
 {
     win32_cond_t *win32_cond = NULL;
     if (cond_init) {
         cond_init(cond);
-        return 0;
+        return;
     }
 
     /* non native condition variables */
     win32_cond = av_mallocz(sizeof(win32_cond_t));
     if (!win32_cond)
-        return ENOMEM;
+        return;
     cond->ptr = win32_cond;
     win32_cond->semaphore = CreateSemaphore(NULL, 0, 0x7fffffff, NULL);
     if (!win32_cond->semaphore)
-        return ENOMEM;
+        return;
     win32_cond->waiters_done = CreateEvent(NULL, TRUE, FALSE, NULL);
     if (!win32_cond->waiters_done)
-        return ENOMEM;
+        return;
 
     pthread_mutex_init(&win32_cond->mtx_waiter_count, NULL);
     pthread_mutex_init(&win32_cond->mtx_broadcast, NULL);
-    return 0;
 }
 
-static av_unused void pthread_cond_destroy(pthread_cond_t *cond)
+static void pthread_cond_destroy(pthread_cond_t *cond)
 {
     win32_cond_t *win32_cond = cond->ptr;
     /* native condition variables do not destroy */
@@ -188,7 +174,7 @@ static av_unused void pthread_cond_destroy(pthread_cond_t *cond)
     cond->ptr = NULL;
 }
 
-static av_unused void pthread_cond_broadcast(pthread_cond_t *cond)
+static void pthread_cond_broadcast(pthread_cond_t *cond)
 {
     win32_cond_t *win32_cond = cond->ptr;
     int have_waiter;
@@ -219,7 +205,7 @@ static av_unused void pthread_cond_broadcast(pthread_cond_t *cond)
     pthread_mutex_unlock(&win32_cond->mtx_broadcast);
 }
 
-static av_unused int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
+static int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
 {
     win32_cond_t *win32_cond = cond->ptr;
     int last_waiter;
@@ -251,7 +237,7 @@ static av_unused int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mu
     return pthread_mutex_lock(mutex);
 }
 
-static av_unused void pthread_cond_signal(pthread_cond_t *cond)
+static void pthread_cond_signal(pthread_cond_t *cond)
 {
     win32_cond_t *win32_cond = cond->ptr;
     int have_waiter;
@@ -276,7 +262,7 @@ static av_unused void pthread_cond_signal(pthread_cond_t *cond)
     pthread_mutex_unlock(&win32_cond->mtx_broadcast);
 }
 
-static av_unused void w32thread_init(void)
+static void w32thread_init(void)
 {
 #if _WIN32_WINNT < 0x0600
     HANDLE kernel_dll = GetModuleHandle(TEXT("kernel32.dll"));

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/APIchanges

@@ -2,478 +2,44 @@ Never assume the API of libav* to be stable unless at least 1 month has passed
 since the last major version increase or the API was added.
 
 The last version increases were:
-libavcodec:    2014-08-09
-libavdevice:   2014-08-09
-libavfilter:   2014-08-09
-libavformat:   2014-08-09
-libavresample: 2014-08-09
-libpostproc:   2014-08-09
-libswresample: 2014-08-09
-libswscale:    2014-08-09
-libavutil:     2014-08-09
+libavcodec:    2013-03-xx
+libavdevice:   2013-03-xx
+libavfilter:   2012-06-22
+libavformat:   2013-03-xx
+libavresample: 2012-10-05
+libpostproc:   2011-04-18
+libswresample: 2011-09-19
+libswscale:    2011-06-20
+libavutil:     2012-10-22
 
 
 API changes, most recent first:
 
-
--------- 8< --------- FFmpeg 2.4 was cut here -------- 8< ---------
-
-2014-08-28 - f30a815 / 9301486 - lavc 56.1.100 / 56.1.0 - avcodec.h
-  Add AV_PKT_DATA_STEREO3D to export container-level stereo3d information.
-
-2014-08-25 - 215db29 / b263f8f - lavf 56.3.100 / 56.3.0 - avformat.h
-  Add AVFormatContext.max_ts_probe.
-
-2014-08-23 - 8fc9bd0 - lavu 54.7.100 - dict.h
-  AV_DICT_DONT_STRDUP_KEY and AV_DICT_DONT_STRDUP_VAL arguments are now
-  freed even on error. This is consistent with the behaviour all users
-  of it we could find expect.
-
-2014-08-21 - 980a5b0 - lavu 54.6.100 - frame.h motion_vector.h
-  Add AV_FRAME_DATA_MOTION_VECTORS side data and AVMotionVector structure
-
-2014-08-16 - b7d5e01 - lswr 1.1.100 - swresample.h
-  Add AVFrame based API
-
-2014-08-16 - c2829dc - lavu 54.4.100 - dict.h
-  Add av_dict_set_int helper function.
-
-2014-08-13 - c8571c6 / 8ddc326 - lavu 54.3.100 / 54.3.0 - mem.h
-  Add av_strndup().
-
-2014-08-13 - 2ba4577 / a8c104a - lavu 54.2.100 / 54.2.0 - opt.h
-  Add av_opt_get_dict_val/set_dict_val with AV_OPT_TYPE_DICT to support
-  dictionary types being set as options.
-
-2014-08-13 - afbd4b8 - lavf 56.01.0 - avformat.h
-  Add AVFormatContext.event_flags and AVStream.event_flags for signaling to
-  the user when events happen in the file/stream.
-
-2014-08-10 - 78eaaa8 / fb1ddcd - lavr 2.1.0 - avresample.h
-  Add avresample_convert_frame() and avresample_config().
-
-2014-08-10 - 78eaaa8 / fb1ddcd - lavu 54.1.100 / 54.1.0 - error.h
-  Add AVERROR_INPUT_CHANGED and AVERROR_OUTPUT_CHANGED.
-
-2014-08-08 - 3841f2a / d35b94f - lavc 55.73.102 / 55.57.4 - avcodec.h
-  Deprecate FF_IDCT_XVIDMMX define and xvidmmx idct option.
-  Replaced by FF_IDCT_XVID and xvid respectively.
-
-2014-08-08 - 5c3c671 - lavf 55.53.100 - avio.h
-  Add avio_feof() and deprecate url_feof().
-
-2014-08-07 - bb78903 - lsws 2.1.3 - swscale.h
-  sws_getContext is not going to be removed in the future.
-
-2014-08-07 - a561662 / ad1ee5f - lavc 55.73.101 / 55.57.3 - avcodec.h
-  reordered_opaque is not going to be removed in the future.
-
-2014-08-02 - 28a2107 - lavu 52.98.100 - pixelutils.h
-  Add pixelutils API with SAD functions
-
-2014-08-04 - 6017c98 / e9abafc - lavu 52.97.100 / 53.22.0 - pixfmt.h
-  Add AV_PIX_FMT_YA16 pixel format for 16 bit packed gray with alpha.
-
-2014-08-04 - 4c8bc6f / e96c3b8 - lavu 52.96.101 / 53.21.1 - avstring.h
-  Rename AV_PIX_FMT_Y400A to AV_PIX_FMT_YA8 to better identify the format.
-  An alias pixel format and color space name are provided for compatibility.
-
-2014-08-04 - 073c074 / d2962e9 - lavu 52.96.100 / 53.21.0 - pixdesc.h
-  Support name aliases for pixel formats.
-
-2014-08-03 - 71d008e / 1ef9e83 - lavc 55.72.101 / 55.57.2 - avcodec.h
-2014-08-03 - 71d008e / 1ef9e83 - lavu 52.95.100 / 53.20.0 - frame.h
-  Deprecate AVCodecContext.dtg_active_format and use side-data instead.
-
-2014-08-03 - e680c73 - lavc 55.72.100 - avcodec.h
-  Add get_pixels() to AVDCT
-
-2014-08-03 - 9400603 / 9f17685 - lavc 55.71.101 / 55.57.1 - avcodec.h
-  Deprecate unused FF_IDCT_IPP define and ipp avcodec option.
-  Deprecate unused FF_DEBUG_PTS define and pts avcodec option.
-  Deprecate unused FF_CODER_TYPE_DEFLATE define and deflate avcodec option.
-  Deprecate unused FF_DCT_INT define and int avcodec option.
-  Deprecate unused avcodec option scenechange_factor.
-
-2014-07-30 - ba3e331 - lavu 52.94.100 - frame.h
-  Add av_frame_side_data_name()
-
-2014-07-29 - 80a3a66 / 3a19405 - lavf 56.01.100 / 56.01.0 - avformat.h
-  Add mime_type field to AVProbeData, which now MUST be initialized in
-  order to avoid uninitialized reads of the mime_type pointer, likely
-  leading to crashes.
-  Typically, this means you will do 'AVProbeData pd = { 0 };' instead of
-  'AVProbeData pd;'.
-
-2014-07-29 - 31e0b5d / 69e7336 - lavu 52.92.100 / 53.19.0 - avstring.h
-  Make name matching function from lavf public as av_match_name().
-
-2014-07-28 - 2e5c8b0 / c5fca01 - lavc 55.71.100 / 55.57.0 - avcodec.h
-  Add AV_CODEC_PROP_REORDER to mark codecs supporting frame reordering.
-
-2014-07-27 - ff9a154 - lavf 55.50.100 - avformat.h
-  New field int64_t probesize2 instead of deprecated
-  field int probesize.
-
-2014-07-27 - 932ff70 - lavc 55.70.100 - avdct.h
-  Add AVDCT / avcodec_dct_alloc() / avcodec_dct_init().
-
-2014-07-23 - 8a4c086 - lavf 55.49.100 - avio.h
-  Add avio_read_to_bprint()
-
-
--------- 8< --------- FFmpeg 2.3 was cut here -------- 8< ---------
-
-2014-07-14 - 62227a7 - lavf 55.47.100 - avformat.h
-  Add av_stream_get_parser()
-
-2014-07-09 - c67690f / a54f03b - lavu 52.92.100 / 53.18.0 - display.h
-  Add av_display_matrix_flip() to flip the transformation matrix.
-
-2014-07-09 - 1b58f13 / f6ee61f - lavc 55.69.100 / 55.56.0 - dv_profile.h
-  Add a public API for DV profile handling.
-
-2014-06-20 - 0dceefc / 9e500ef - lavu 52.90.100 / 53.17.0 - imgutils.h
-  Add av_image_check_sar().
-
-2014-06-20 - 4a99333 / 874390e - lavc 55.68.100 / 55.55.0 - avcodec.h
-  Add av_packet_rescale_ts() to simplify timestamp conversion.
-
-2014-06-18 - ac293b6 / 194be1f - lavf 55.44.100 / 55.20.0 - avformat.h
-  The proper way for providing a hint about the desired timebase to the muxers
-  is now setting AVStream.time_base, instead of AVStream.codec.time_base as was
-  done previously. The old method is now deprecated.
-
-2014-06-11 - 67d29da - lavc 55.66.101 - avcodec.h
-  Increase FF_INPUT_BUFFER_PADDING_SIZE to 32 due to some corner cases needing
-  it
-
-2014-06-10 - 5482780 - lavf 55.43.100 - avformat.h
-  New field int64_t max_analyze_duration2 instead of deprecated
-  int max_analyze_duration.
-
-2014-05-30 - 00759d7 - lavu 52.89.100 - opt.h
-  Add av_opt_copy()
-
-2014-06-01 - 03bb99a / 0957b27 - lavc 55.66.100 / 55.54.0 - avcodec.h
-  Add AVCodecContext.side_data_only_packets to allow encoders to output packets
-  with only side data. This option may become mandatory in the future, so all
-  users are recommended to update their code and enable this option.
-
-2014-06-01 - 6e8e9f1 / 8c02adc - lavu 52.88.100 / 53.16.0 - frame.h, pixfmt.h
-  Move all color-related enums (AVColorPrimaries, AVColorSpace, AVColorRange,
-  AVColorTransferCharacteristic, and AVChromaLocation) inside lavu.
-  And add AVFrame fields for them.
-
-2014-05-29 - bdb2e80 / b2d4565 - lavr 1.3.0 - avresample.h
-  Add avresample_max_output_samples
-
-2014-05-28 - d858ee7 / 6d21259 - lavf 55.42.100 / 55.19.0 - avformat.h
-  Add strict_std_compliance and related AVOptions to support experimental
-  muxing.
-
-2014-05-26 - 55cc60c - lavu 52.87.100 - threadmessage.h
-  Add thread message queue API.
-
-2014-05-26 - c37d179 - lavf 55.41.100 - avformat.h
-  Add format_probesize to AVFormatContext.
-
-2014-05-20 - 7d25af1 / c23c96b - lavf 55.39.100 / 55.18.0 - avformat.h
-  Add av_stream_get_side_data() to access stream-level side data
-  in the same way as av_packet_get_side_data().
-
-2014-05-20 - 7336e39 - lavu 52.86.100 - fifo.h
-  Add av_fifo_alloc_array() function.
-
-2014-05-19 - ef1d4ee / bddd8cb - lavu 52.85.100 / 53.15.0 - frame.h, display.h
-  Add AV_FRAME_DATA_DISPLAYMATRIX for exporting frame-level
-  spatial rendering on video frames for proper display.
-
-2014-05-19 - ef1d4ee / bddd8cb - lavc 55.64.100 / 55.53.0 - avcodec.h
-  Add AV_PKT_DATA_DISPLAYMATRIX for exporting packet-level
-  spatial rendering on video frames for proper display.
-
-2014-05-19 - 999a99c / a312f71 - lavf 55.38.101 / 55.17.1 - avformat.h
-  Deprecate AVStream.pts and the AVFrac struct, which was its only use case.
-  See use av_stream_get_end_pts()
-
-2014-05-18 - 68c0518 / fd05602 - lavc 55.63.100 / 55.52.0 - avcodec.h
-  Add avcodec_free_context(). From now on it should be used for freeing
-  AVCodecContext.
-
-2014-05-17 - 0eec06e - lavu 52.84.100 - time.h
-  Add av_gettime_relative() av_gettime_relative_is_monotonic()
-
-2014-05-15 - eacf7d6 / 0c1959b - lavf 55.38.100 / 55.17.0 - avformat.h
-  Add AVFMT_FLAG_BITEXACT flag. Muxers now use it instead of checking
-  CODEC_FLAG_BITEXACT on the first stream.
-
-2014-05-15 - 96cb4c8 - lswr 0.19.100 - swresample.h
-  Add swr_close()
-
-2014-05-11 - 14aef38 / 66e6c8a - lavu 52.83.100 / 53.14.0 - pixfmt.h
-  Add AV_PIX_FMT_VDA for new-style VDA acceleration.
-
-2014-05-07 - 351f611 - lavu 52.82.100 - fifo.h
-  Add av_fifo_freep() function.
-
-2014-05-02 - ba52fb11 - lavu 52.81.100 - opt.h
-  Add av_opt_set_dict2() function.
-
-2014-05-01 - e77b985 / a2941c8 - lavc 55.60.103 / 55.50.3 - avcodec.h
-  Deprecate CODEC_FLAG_MV0. It is replaced by the flag "mv0" in the
-  "mpv_flags" private option of the mpegvideo encoders.
-
-2014-05-01 - e40ae8c / 6484149 - lavc 55.60.102 / 55.50.2 - avcodec.h
-  Deprecate CODEC_FLAG_GMC. It is replaced by the "gmc" private option of the
-  libxvid encoder.
-
-2014-05-01 - 1851643 / b2c3171 - lavc 55.60.101 / 55.50.1 - avcodec.h
-  Deprecate CODEC_FLAG_NORMALIZE_AQP. It is replaced by the flag "naq" in the
-  "mpv_flags" private option of the mpegvideo encoders.
-
-2014-05-01 - cac07d0 / 5fcceda - avcodec.h
-  Deprecate CODEC_FLAG_INPUT_PRESERVED. Its functionality is replaced by passing
-  reference-counted frames to encoders.
-
-2014-04-30 - 617e866 - lavu 52.81.100 - pixdesc.h
-  Add av_find_best_pix_fmt_of_2(), av_get_pix_fmt_loss()
-  Deprecate avcodec_get_pix_fmt_loss(), avcodec_find_best_pix_fmt_of_2()
-
-2014-04-29 - 1bf6396 - lavc 55.60.100 - avcodec.h
-  Add AVCodecDescriptor.mime_types field.
-
-2014-04-29 - b804eb4 - lavu 52.80.100 - hash.h
-  Add av_hash_final_bin(), av_hash_final_hex() and av_hash_final_b64().
-
-2014-03-07 - 8b2a130 - lavc 55.50.0 / 55.53.100 - dxva2.h
-  Add FF_DXVA2_WORKAROUND_INTEL_CLEARVIDEO for old Intel GPUs.
-
-2014-04-22 - 502512e /dac7e8a - lavu 53.13.0 / 52.78.100 - avutil.h
-  Add av_get_time_base_q().
-
-2014-04-17 - a8d01a7 / 0983d48 - lavu 53.12.0 / 52.77.100 - crc.h
-  Add AV_CRC_16_ANSI_LE crc variant.
-
-2014-04-15 - ef818d8 - lavf 55.37.101 - avformat.h
-  Add av_format_inject_global_side_data()
-
-2014-04-12 - 4f698be - lavu 52.76.100 - log.h
-  Add av_log_get_flags()
-
-2014-04-11 - 6db42a2b - lavd 55.12.100 - avdevice.h
-  Add avdevice_capabilities_create() function.
-  Add avdevice_capabilities_free() function.
-
-2014-04-07 - 0a1cc04 / 8b17243 - lavu 52.75.100 / 53.11.0 - pixfmt.h
-  Add AV_PIX_FMT_YVYU422 pixel format.
-
-2014-04-04 - c1d0536 / 8542f9c - lavu 52.74.100 / 53.10.0 - replaygain.h
-  Full scale for peak values is now 100000 (instead of UINT32_MAX) and values
-  may overflow.
-
-2014-04-03 - c16e006 / 7763118 - lavu 52.73.100 / 53.9.0 - log.h
-  Add AV_LOG(c) macro to have 256 color debug messages.
-
-2014-04-03 - eaed4da9 - lavu 52.72.100 - opt.h
-  Add AV_OPT_MULTI_COMPONENT_RANGE define to allow return
-  multi-component option ranges.
-
-2014-03-29 - cd50a44b - lavu 52.70.100 - mem.h
-  Add av_dynarray_add_nofree() function.
-
-2014-02-24 - 3e1f241 / d161ae0 - lavu 52.69.100 / 53.8.0 - frame.h
-  Add av_frame_remove_side_data() for removing a single side data
-  instance from a frame.
-
-2014-03-24 - 83e8978 / 5a7e35d - lavu 52.68.100 / 53.7.0 - frame.h, replaygain.h
-  Add AV_FRAME_DATA_REPLAYGAIN for exporting replaygain tags.
-  Add a new header replaygain.h with the AVReplayGain struct.
-
-2014-03-24 - 83e8978 / 5a7e35d - lavc 55.54.100 / 55.36.0 - avcodec.h
-  Add AV_PKT_DATA_REPLAYGAIN for exporting replaygain tags.
-
-2014-03-24 - 595ba3b / 25b3258 - lavf 55.35.100 / 55.13.0 - avformat.h
-  Add AVStream.side_data and AVStream.nb_side_data for exporting stream-global
-  side data (e.g. replaygain tags, video rotation)
-
-2014-03-24 - bd34e26 / 0e2c3ee - lavc 55.53.100 / 55.35.0 - avcodec.h
-  Give the name AVPacketSideData to the previously anonymous struct used for
-  AVPacket.side_data.
-
-
--------- 8< --------- FFmpeg 2.2 was cut here -------- 8< ---------
-
-2014-03-18 - 37c07d4 - lsws 2.5.102
-  Make gray16 full-scale.
-
-2014-03-16 - 6b1ca17 / 1481d24 - lavu 52.67.100 / 53.6.0 - pixfmt.h
-  Add RGBA64_LIBAV pixel format and variants for compatibility
-
-2014-03-11 - 3f3229c - lavf 55.34.101 - avformat.h
-  Set AVFormatContext.start_time_realtime when demuxing.
-
-2014-03-03 - 06fed440 - lavd 55.11.100 - avdevice.h
-  Add av_input_audio_device_next().
-  Add av_input_video_device_next().
-  Add av_output_audio_device_next().
-  Add av_output_video_device_next().
-
-2014-02-24 - fff5262 / 1155fd0 - lavu 52.66.100 / 53.5.0 - frame.h
-  Add av_frame_copy() for copying the frame data.
-
-2014-02-24 - a66be60 - lswr 0.18.100 - swresample.h
-  Add swr_is_initialized() for checking whether a resample context is initialized.
-
-2014-02-22 - 5367c0b / 7e86c27 - lavr 1.2.0 - avresample.h
-  Add avresample_is_open() for checking whether a resample context is open.
-
-2014-02-19 - 6a24d77 / c3ecd96 - lavu 52.65.100 / 53.4.0  - opt.h
-  Add AV_OPT_FLAG_EXPORT and AV_OPT_FLAG_READONLY to mark options meant (only)
-  for reading.
-
-2014-02-19 - f4c8d00 / 6bb8720 - lavu 52.64.101 / 53.3.1 - opt.h
-  Deprecate unused AV_OPT_FLAG_METADATA.
-
-2014-02-16 - 81c3f81 - lavd 55.10.100 - avdevice.h
-  Add avdevice_list_devices() and avdevice_free_list_devices()
-
-2014-02-16 - db3c970 - lavf 55.33.100 - avio.h
-  Add avio_find_protocol_name() to find out the name of the protocol that would
-  be selected for a given URL.
-
-2014-02-15 - a2bc6c1 / c98f316 - lavu 52.64.100 / 53.3.0 - frame.h
-  Add AV_FRAME_DATA_DOWNMIX_INFO value to the AVFrameSideDataType enum and
-  downmix_info.h API, which identify downmix-related metadata.
-
-2014-02-11 - 1b05ac2 - lavf 55.32.100 - avformat.h
-  Add av_write_uncoded_frame() and av_interleaved_write_uncoded_frame().
-
-2014-02-04 - 3adb5f8 / d9ae103 - lavf 55.30.100 / 55.11.0 - avformat.h
-  Add AVFormatContext.max_interleave_delta for controlling amount of buffering
-  when interleaving.
-
-2014-02-02 - 5871ee5 - lavf 55.29.100 - avformat.h
-  Add output_ts_offset muxing option to AVFormatContext.
-
-2014-01-27 - 102bd64 - lavd 55.7.100 - avdevice.h
-                       lavf 55.28.100 - avformat.h
-  Add avdevice_dev_to_app_control_message() function.
-
-2014-01-27 - 7151411 - lavd 55.6.100 - avdevice.h
-                       lavf 55.27.100 - avformat.h
-  Add avdevice_app_to_dev_control_message() function.
-
-2014-01-24 - 86bee79 - lavf 55.26.100 - avformat.h
-  Add AVFormatContext option metadata_header_padding to allow control over the
-  amount of padding added.
-
-2014-01-20 - eef74b2 / 93c553c - lavc 55.48.102 / 55.32.1 - avcodec.h
-  Edges are not required anymore on video buffers allocated by get_buffer2()
-  (i.e. as if the CODEC_FLAG_EMU_EDGE flag was always on). Deprecate
-  CODEC_FLAG_EMU_EDGE and avcodec_get_edge_width().
-
-2014-01-19 - 1a193c4 - lavf 55.25.100 - avformat.h
-  Add avformat_get_mov_video_tags() and avformat_get_mov_audio_tags().
-
-2014-01-19 - 3532dd5 - lavu 52.63.100 - rational.h
-  Add av_make_q() function.
-
-2014-01-05 - 4cf4da9 / 5b4797a - lavu 52.62.100 / 53.2.0 - frame.h
-  Add AV_FRAME_DATA_MATRIXENCODING value to the AVFrameSideDataType enum, which
-  identifies AVMatrixEncoding data.
-
-2014-01-05 - 751385f / 5c437fb - lavu 52.61.100 / 53.1.0 - channel_layout.h
-  Add values for various Dolby flags to the AVMatrixEncoding enum.
-
-2014-01-04 - b317f94 - lavu 52.60.100 - mathematics.h
-  Add av_add_stable() function.
-
-2013-12-22 - 911676c - lavu 52.59.100 - avstring.h
-  Add av_strnlen() function.
-
-2013-12-09 - 64f73ac - lavu 52.57.100 - opencl.h
-  Add av_opencl_benchmark() function.
-
-2013-11-30 - 82b2e9c - lavu 52.56.100 - ffversion.h
-  Moves version.h to libavutil/ffversion.h.
-  Install ffversion.h and make it public.
-
-2013-12-11 - 29c83d2 / b9fb59d,409a143 / 9431356,44967ab / d7b3ee9 - lavc 55.45.101 / 55.28.1 - avcodec.h
-  av_frame_alloc(), av_frame_unref() and av_frame_free() now can and should be
-  used instead of avcodec_alloc_frame(), avcodec_get_frame_defaults() and
-  avcodec_free_frame() respectively. The latter three functions are deprecated.
-
-2013-12-09 - 7a60348 / 7e244c6- - lavu 52.58.100 / 52.20.0 - frame.h
-  Add AV_FRAME_DATA_STEREO3D value to the AVFrameSideDataType enum and
-  stereo3d.h API, that identify codec-independent stereo3d information.
-
-2013-11-26 - 625b290 / 1eaac1d- - lavu 52.55.100 / 52.19.0 - frame.h
-  Add AV_FRAME_DATA_A53_CC value to the AVFrameSideDataType enum, which
-  identifies ATSC A53 Part 4 Closed Captions data.
-
-2013-11-22 - 6859065 - lavu 52.54.100 - avstring.h
-  Add av_utf8_decode() function.
-
-2013-11-22 - fb7d70c - lavc 55.44.100 - avcodec.h
-  Add HEVC profiles
-
-2013-11-20 - c28b61c - lavc 55.44.100 - avcodec.h
-  Add av_packet_{un,}pack_dictionary()
-  Add AV_PKT_METADATA_UPDATE side data type, used to transmit key/value
-  strings between a stream and the application.
-
-2013-11-14 - 7c888ae / cce3e0a - lavu 52.53.100 / 52.18.0 - mem.h
-  Move av_fast_malloc() and av_fast_realloc() for libavcodec to libavutil.
-
-2013-11-14 - b71e4d8 / 8941971 - lavc 55.43.100 / 55.27.0 - avcodec.h
-  Deprecate AVCodecContext.error_rate, it is replaced by the 'error_rate'
-  private option of the mpegvideo encoder family.
-
-2013-11-14 - 31c09b7 / 728c465 - lavc 55.42.100 / 55.26.0 - vdpau.h
-  Add av_vdpau_get_profile().
-  Add av_vdpau_alloc_context(). This function must from now on be
-  used for allocating AVVDPAUContext.
-
-2013-11-04 - be41f21 / cd8f772 - lavc 55.41.100 / 55.25.0 - avcodec.h
-                       lavu 52.51.100 - frame.h
-  Add ITU-R BT.2020 and other not yet included values to color primaries,
-  transfer characteristics and colorspaces.
-
-2013-11-04 - 85cabf1 - lavu 52.50.100 - avutil.h
-  Add av_fopen_utf8()
-
-2013-10-31 - 78265fc / 28096e0 - lavu 52.49.100 / 52.17.0 - frame.h
-  Add AVFrame.flags and AV_FRAME_FLAG_CORRUPT.
-
-
--------- 8< --------- FFmpeg 2.1 was cut here -------- 8< ---------
-
-2013-10-27 - dbe6f9f - lavc 55.39.100 - avcodec.h
+2013-10-27 - xxxxxxx - lavc 55.39.100 - avcodec.h
   Add CODEC_CAP_DELAY support to avcodec_decode_subtitle2.
 
-2013-10-27 - d61617a - lavu 52.48.100 - parseutils.h
+2013-10-27 - xxxxxxx - lavu 52.48.100 - parseutils.h
   Add av_get_known_color_name().
 
-2013-10-17 - 8696e51 - lavu 52.47.100 - opt.h
+2013-10-17 - xxxxxxx - lavu 52.47.100 - opt.h
   Add AV_OPT_TYPE_CHANNEL_LAYOUT and channel layout option handlers
   av_opt_get_channel_layout() and av_opt_set_channel_layout().
 
-2013-10-06 - ccf96f8 -libswscale 2.5.101 - options.c
+2013-10-xx - xxxxxxx -libswscale 2.5.101 - options.c
   Change default scaler to bicubic
 
-2013-10-03 - e57dba0 - lavc 55.34.100 - avcodec.h
+2013-10-03 - xxxxxxx - lavc 55.34.100 - avcodec.h
   Add av_codec_get_max_lowres()
 
-2013-10-02 - 5082fcc - lavf 55.19.100 - avformat.h
+2013-10-02 - xxxxxxx - lavf 55.19.100 - avformat.h
   Add audio/video/subtitle AVCodec fields to AVFormatContext to force specific
   decoders
 
-2013-09-28 - 7381d31 / 0767bfd - lavfi 3.88.100 / 3.11.0 - avfilter.h
+2013-08-xx - xxxxxxx - lavfi 3.11.0 - avfilter.h
   Add AVFilterGraph.execute and AVFilterGraph.opaque for custom slice threading
   implementations.
 
-2013-09-21 - 85f8a3c / e208e6d - lavu 52.46.100 / 52.16.0 - pixfmt.h
+2013-09-21 - xxxxxxx - lavu 52.16.0 - pixfmt.h
   Add interleaved 4:2:2 8/10-bit formats AV_PIX_FMT_NV16 and
   AV_PIX_FMT_NV20.
 
@@ -483,7 +49,7 @@ API changes, most recent first:
 2013-09-04 - 3e1f507 - lavc 55.31.101 - avcodec.h
   avcodec_close() argument can be NULL.
 
-2013-09-04 - 36cd017a - lavf 55.16.101 - avformat.h
+2013-09-04 - 36cd017 - lavf 55.16.101 - avformat.h
   avformat_close_input() argument can be NULL and point on NULL.
 
 2013-08-29 - e31db62 - lavf 55.15.100 - avformat.h
@@ -492,10 +58,10 @@ API changes, most recent first:
 2013-08-15 - 1e0e193 - lsws 2.5.100 -
   Add a sws_dither AVOption, allowing to set the dither algorithm used
 
-2013-08-11 - d404fe35 - lavc 55.27.100 - vdpau.h
+2013-08-xx - xxxxxxx - lavc 55.27.100 - vdpau.h
   Add a render2 alternative to the render callback function.
 
-2013-08-11 - af05edc - lavc 55.26.100 - vdpau.h
+2013-08-xx - xxxxxxx - lavc 55.26.100 - vdpau.h
   Add allocation function for AVVDPAUContext, allowing
   to extend it in the future without breaking ABI/API.
 
@@ -505,7 +71,7 @@ API changes, most recent first:
 
 2013-08-05 - 9547e3e / f824535 - lavc 55.22.100 / 55.13.0 - avcodec.h
   Deprecate the bitstream-related members from struct AVVDPAUContext.
-  The bitstream buffers no longer need to be explicitly freed.
+  The bistream buffers no longer need to be explicitly freed.
 
 2013-08-05 - 3b805dc / 549294f - lavc 55.21.100 / 55.12.0 - avcodec.h
   Deprecate the CODEC_CAP_HWACCEL_VDPAU codec capability. Use CODEC_CAP_HWACCEL
@@ -521,9 +87,6 @@ API changes, most recent first:
   Add avcodec_chroma_pos_to_enum()
   Add avcodec_enum_to_chroma_pos()
 
-
--------- 8< --------- FFmpeg 2.0 was cut here -------- 8< ---------
-
 2013-07-03 - 838bd73 - lavfi 3.78.100 - avfilter.h
   Deprecate avfilter_graph_parse() in favor of the equivalent
   avfilter_graph_parse_ptr().
@@ -596,9 +159,6 @@ API changes, most recent first:
 2013-03-17 - 7aa9af5 - lavu 52.20.100 - opt.h
   Add AV_OPT_TYPE_VIDEO_RATE value to AVOptionType enum.
 
-
--------- 8< --------- FFmpeg 1.2 was cut here -------- 8< ---------
-
 2013-03-07 - 9767ec6 - lavu 52.18.100 - avstring.h,bprint.h
   Add av_escape() and av_bprint_escape() API.
 
@@ -611,9 +171,6 @@ API changes, most recent first:
 2013-01-01 - 2eb2e17 - lavfi 3.34.100
   Add avfilter_get_audio_buffer_ref_from_arrays_channels.
 
-
--------- 8< --------- FFmpeg 1.1 was cut here -------- 8< ---------
-
 2012-12-20 - 34de47aa - lavfi 3.29.100 - avfilter.h
   Add AVFilterLink.channels, avfilter_link_get_channels()
   and avfilter_ref_get_channels().
@@ -659,9 +216,6 @@ API changes, most recent first:
   Add LIBSWRESAMPLE_VERSION, LIBSWRESAMPLE_BUILD
   and LIBSWRESAMPLE_IDENT symbols.
 
-
--------- 8< --------- FFmpeg 1.0 was cut here -------- 8< ---------
-
 2012-09-06 - 29e972f - lavu 51.72.100 - parseutils.h
   Add av_small_strptime() time parsing function.
 
@@ -1080,9 +634,6 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
 2012-01-12 - b18e17e / 3167dc9 - lavfi 2.59.100 / 2.15.0
   Add a new installed header -- libavfilter/version.h -- with version macros.
 
-
--------- 8< --------- FFmpeg 0.9 was cut here -------- 8< ---------
-
 2011-12-08 - a502939 - lavfi 2.52.0
   Add av_buffersink_poll_frame() to buffersink.h.
 
@@ -1111,9 +662,6 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
   Add avformat_close_input().
   Deprecate av_close_input_file() and av_close_input_stream().
 
-2011-12-09 - c59b80c / b2890f5 - lavu 51.32.0 / 51.20.0 - audioconvert.h
-  Expand the channel layout list.
-
 2011-12-02 - e4de716 / 0eea212 - lavc 53.40.0 / 53.25.0
   Add nb_samples and extended_data fields to AVFrame.
   Deprecate AVCODEC_MAX_AUDIO_FRAME_SIZE.
@@ -1127,10 +675,6 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
   Change AVCodecContext.error[4] to [8] at next major bump.
   Add AV_NUM_DATA_POINTERS to simplify the bump transition.
 
-2011-11-24 - lavu 51.29.0 / 51.19.0
-  92afb43 / bd97b2e - add planar RGB pixel formats
-  92afb43 / 6b0768e - add PIX_FMT_PLANAR and PIX_FMT_RGB pixel descriptions
-
 2011-11-23 - 8e576d5 / bbb46f3 - lavu 51.27.0 / 51.18.0
   Add av_samples_get_buffer_size(), av_samples_fill_arrays(), and
   av_samples_alloc(), to samplefmt.h.
@@ -1292,13 +836,6 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
 2011-06-28 - 5129336 - lavu 51.11.0 - avutil.h
   Define the AV_PICTURE_TYPE_NONE value in AVPictureType enum.
 
-
--------- 8< --------- FFmpeg 0.7 was cut here -------- 8< ---------
-
-
-
--------- 8< --------- FFmpeg 0.8 was cut here -------- 8< ---------
-
 2011-06-19 - fd2c0a5 - lavfi 2.23.0 - avfilter.h
   Add layout negotiation fields and helper functions.
 
@@ -1976,9 +1513,6 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
 2010-06-02 - 7e566bb - lavc 52.73.0 - av_get_codec_tag_string()
   Add av_get_codec_tag_string().
 
-
--------- 8< --------- FFmpeg 0.6 was cut here -------- 8< ---------
-
 2010-06-01 - 2b99142 - lsws 0.11.0 - convertPalette API
   Add sws_convertPalette8ToPacked32() and sws_convertPalette8ToPacked24().
 
@@ -1996,6 +1530,10 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
 2010-05-09 - b6bc205 - lavfi 1.20.0 - AVFilterPicRef
   Add interlaced and top_field_first fields to AVFilterPicRef.
 
+------------------------------8<-------------------------------------
+                   0.6 branch was cut here
+----------------------------->8--------------------------------------
+
 2010-05-01 - 8e2ee18 - lavf 52.62.0 - probe function
   Add av_probe_input_format2 to API, it allows ignoring probe
   results below given score and returns the actual probe score.

doc/Doxyfile

@@ -31,7 +31,7 @@ PROJECT_NAME           = FFmpeg
 # This could be handy for archiving the generated documentation or
 # if some version control system is used.
 
-PROJECT_NUMBER         = 2.4.7
+PROJECT_NUMBER         = 2.1
 
 # With the PROJECT_LOGO tag one can specify a logo or icon that is included
 # in the documentation. The maximum height of the logo should not exceed 55
@@ -759,7 +759,7 @@ ALPHABETICAL_INDEX     = YES
 # the COLS_IN_ALPHA_INDEX tag can be used to specify the number of columns
 # in which this list will be split (can be a number in the range [1..20])
 
-COLS_IN_ALPHA_INDEX    = 5
+COLS_IN_ALPHA_INDEX    = 2
 
 # In case all classes in a project start with a common prefix, all
 # classes will be put under the same header in the alphabetical index.
@@ -793,13 +793,13 @@ HTML_FILE_EXTENSION    = .html
 # each generated HTML page. If it is left blank doxygen will generate a
 # standard header.
 
-HTML_HEADER            =
+#HTML_HEADER            = doc/doxy/header.html
 
 # The HTML_FOOTER tag can be used to specify a personal HTML footer for
 # each generated HTML page. If it is left blank doxygen will generate a
 # standard footer.
 
-HTML_FOOTER            =
+#HTML_FOOTER            = doc/doxy/footer.html
 
 # The HTML_STYLESHEET tag can be used to specify a user-defined cascading
 # style sheet that is used by each HTML page. It can be used to
@@ -808,7 +808,7 @@ HTML_FOOTER            =
 # the style sheet file to the HTML output directory, so don't put your own
 # stylesheet in the HTML output directory as well, or it will be erased!
 
-HTML_STYLESHEET        =
+#HTML_STYLESHEET        = doc/doxy/doxy_stylesheet.css
 
 # The HTML_COLORSTYLE_HUE tag controls the color of the HTML output.
 # Doxygen will adjust the colors in the stylesheet and background images
@@ -1056,7 +1056,7 @@ FORMULA_TRANSPARENT    = YES
 # typically be disabled. For large projects the javascript based search engine
 # can be slow, then enabling SERVER_BASED_SEARCH may provide a better solution.
 
-SEARCHENGINE           = YES
+SEARCHENGINE           = NO
 
 # When the SERVER_BASED_SEARCH tag is enabled the search engine will be
 # implemented using a PHP enabled web server instead of at the web client
@@ -1359,8 +1359,6 @@ PREDEFINED             = "__attribute__(x)=" \
                          "DECLARE_ALIGNED(a,t,n)=t n" \
                          "offsetof(x,y)=0x42" \
                          av_alloc_size \
-                         AV_GCC_VERSION_AT_LEAST(x,y)=1 \
-                         __GNUC__=1 \
 
 # If the MACRO_EXPANSION and EXPAND_ONLY_PREDEF tags are set to YES then
 # this tag can be used to specify a list of macro names that should be expanded.

doc/Makefile

@@ -14,11 +14,11 @@ COMPONENTS-$(CONFIG_AVFORMAT)   += ffmpeg-formats ffmpeg-protocols
 COMPONENTS-$(CONFIG_AVDEVICE)   += ffmpeg-devices
 COMPONENTS-$(CONFIG_AVFILTER)   += ffmpeg-filters
 
-MANPAGES1   = $(AVPROGS-yes:%=doc/%.1)    $(AVPROGS-yes:%=doc/%-all.1)    $(COMPONENTS-yes:%=doc/%.1)
+MANPAGES1   = $(PROGS-yes:%=doc/%.1)    $(PROGS-yes:%=doc/%-all.1)    $(COMPONENTS-yes:%=doc/%.1)
 MANPAGES3   = $(LIBRARIES-yes:%=doc/%.3)
 MANPAGES    = $(MANPAGES1) $(MANPAGES3)
-PODPAGES    = $(AVPROGS-yes:%=doc/%.pod)  $(AVPROGS-yes:%=doc/%-all.pod)  $(COMPONENTS-yes:%=doc/%.pod)  $(LIBRARIES-yes:%=doc/%.pod)
-HTMLPAGES   = $(AVPROGS-yes:%=doc/%.html) $(AVPROGS-yes:%=doc/%-all.html) $(COMPONENTS-yes:%=doc/%.html) $(LIBRARIES-yes:%=doc/%.html) \
+PODPAGES    = $(PROGS-yes:%=doc/%.pod)  $(PROGS-yes:%=doc/%-all.pod)  $(COMPONENTS-yes:%=doc/%.pod)  $(LIBRARIES-yes:%=doc/%.pod)
+HTMLPAGES   = $(PROGS-yes:%=doc/%.html) $(PROGS-yes:%=doc/%-all.html) $(COMPONENTS-yes:%=doc/%.html) $(LIBRARIES-yes:%=doc/%.html) \
               doc/developer.html                                        \
               doc/faq.html                                              \
               doc/fate.html                                             \
@@ -36,28 +36,6 @@ DOCS-$(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-$(CONFIG_DECODING_ENCODING_EXAMPLE) += decoding_encoding
-DOC_EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding
-DOC_EXAMPLES-$(CONFIG_EXTRACT_MVS_EXAMPLE)       += extract_mvs
-DOC_EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE)      += filter_audio
-DOC_EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE)   += filtering_audio
-DOC_EXAMPLES-$(CONFIG_FILTERING_VIDEO_EXAMPLE)   += filtering_video
-DOC_EXAMPLES-$(CONFIG_METADATA_EXAMPLE)          += metadata
-DOC_EXAMPLES-$(CONFIG_MUXING_EXAMPLE)            += muxing
-DOC_EXAMPLES-$(CONFIG_REMUXING_EXAMPLE)          += remuxing
-DOC_EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE)  += resampling_audio
-DOC_EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE)     += scaling_video
-DOC_EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE)     += transcode_aac
-DOC_EXAMPLES-$(CONFIG_TRANSCODING_EXAMPLE)       += transcoding
-ALL_DOC_EXAMPLES_LIST = $(DOC_EXAMPLES-) $(DOC_EXAMPLES-yes)
-
-DOC_EXAMPLES       := $(DOC_EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)$(EXESUF))
-ALL_DOC_EXAMPLES   := $(ALL_DOC_EXAMPLES_LIST:%=doc/examples/%$(PROGSSUF)$(EXESUF))
-ALL_DOC_EXAMPLES_G := $(ALL_DOC_EXAMPLES_LIST:%=doc/examples/%$(PROGSSUF)_g$(EXESUF))
-PROGS              += $(DOC_EXAMPLES)
-
 all-$(CONFIG_DOC): doc
 
 doc: documentation
@@ -65,9 +43,7 @@ doc: documentation
 apidoc: doc/doxy/html
 documentation: $(DOCS)
 
-examples: $(DOC_EXAMPLES)
-
-TEXIDEP = perl $(SRC_PATH)/doc/texidep.pl $(SRC_PATH) $< $@ >$(@:%=%.d)
+TEXIDEP = awk '/^@(verbatim)?include/ { printf "$@: $(@D)/%s\n", $$2 }' <$< >$(@:%=%.d)
 
 doc/%.txt: TAG = TXT
 doc/%.txt: doc/%.texi
@@ -82,25 +58,14 @@ $(GENTEXI): doc/avoptions_%.texi: doc/print_options$(HOSTEXESUF)
 	$(M)doc/print_options $* > $@
 
 doc/%.html: TAG = HTML
-doc/%-all.html: TAG = HTML
-
-ifdef HAVE_MAKEINFO_HTML
-doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.pm $(GENTEXI)
-	$(Q)$(TEXIDEP)
-	$(M)makeinfo --html -I doc --no-split -D config-not-all --init-file=$(SRC_PATH)/doc/t2h.pm --output $@ $<
-
-doc/%-all.html: doc/%.texi $(SRC_PATH)/doc/t2h.pm $(GENTEXI)
-	$(Q)$(TEXIDEP)
-	$(M)makeinfo --html -I doc --no-split -D config-all --init-file=$(SRC_PATH)/doc/t2h.pm --output $@ $<
-else
 doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
 	$(Q)$(TEXIDEP)
 	$(M)texi2html -I doc -monolithic --D=config-not-all --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
 
+doc/%-all.html: TAG = HTML
 doc/%-all.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
 	$(Q)$(TEXIDEP)
 	$(M)texi2html -I doc -monolithic --D=config-all --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
-endif
 
 doc/%.pod: TAG = POD
 doc/%.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI)
@@ -119,14 +84,9 @@ 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: TAG = DOXY
-doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(SRC_PATH)/doc/doxy-wrapper.sh $(DOXY_INPUT)
-	$(M)$(SRC_PATH)/doc/doxy-wrapper.sh $(SRC_PATH) $< $(DOXYGEN) $(DOXY_INPUT)
+doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(INSTHEADERS)
+	$(M)$(SRC_PATH)/doc/doxy-wrapper.sh $(SRC_PATH) $^
 
 install-doc: install-html install-man
 
@@ -160,7 +120,7 @@ uninstall-html:
 	$(RM) -r "$(DOCDIR)"
 
 uninstall-man:
-	$(RM) $(addprefix "$(MANDIR)/man1/",$(AVPROGS-yes:%=%.1) $(AVPROGS-yes:%=%-all.1) $(COMPONENTS-yes:%=%.1))
+	$(RM) $(addprefix "$(MANDIR)/man1/",$(PROGS-yes:%=%.1) $(PROGS-yes:%=%-all.1) $(COMPONENTS-yes:%=%.1))
 	$(RM) $(addprefix "$(MANDIR)/man3/",$(LIBRARIES-yes:%=%.3))
 
 clean:: docclean
@@ -168,13 +128,8 @@ clean:: docclean
 distclean:: docclean
 	$(RM) doc/config.texi
 
-examplesclean:
-	$(RM) $(ALL_DOC_EXAMPLES) $(ALL_DOC_EXAMPLES_G)
-	$(RM) $(CLEANSUFFIXES:%=doc/examples/%)
-
-docclean: examplesclean
-	$(RM) $(CLEANSUFFIXES:%=doc/%)
-	$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 doc/*.3 doc/avoptions_*.texi
+docclean:
+	$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 doc/*.3 $(CLEANSUFFIXES:%=doc/%) doc/avoptions_*.texi
 	$(RM) -r doc/doxy/html
 
 -include $(wildcard $(DOCS:%=%.d))

doc/RELEASE_NOTES

@@ -0,0 +1,16 @@
+Release Notes
+=============
+
+* 2.1 "Fourier"  October, 2013
+
+
+General notes
+-------------
+See the Changelog file for a list of significant changes. Note, there
+are many more new features and bugfixes than whats listed there.
+
+Bugreports against FFmpeg git master or the most recent FFmpeg release are
+accepted. If you are experiencing issues with any formally released version of
+FFmpeg, please try git master to check if the issue still exists. If it does,
+make your report against the development code following the usual bug reporting
+guidelines.

doc/avutil.txt

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

doc/bitstream_filters.texi

@@ -74,18 +74,7 @@ format with @command{ffmpeg}, you can use the command:
 ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts
 @end example
 
-@section imxdump
-
-Modifies the bitstream to fit in MOV and to be usable by the Final Cut
-Pro decoder. This filter only applies to the mpeg2video codec, and is
-likely not needed for Final Cut Pro 7 and newer with the appropriate
-@option{-tag:v}.
-
-For example, to remux 30 MB/sec NTSC IMX to MOV:
-
-@example
-ffmpeg -i input.mxf -c copy -bsf:v imxdump -tag:v mx3n output.mov
-@end example
+@section imx_dump_header
 
 @section mjpeg2jpeg
 
@@ -128,17 +117,12 @@ ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
 
 @section movsub
 
+@section mp3_header_compress
+
 @section mp3_header_decompress
 
 @section noise
 
-Damages the contents of packets without damaging the container. Can be
-used for fuzzing or testing error resilience/concealment.
-
-@example
-ffmpeg -i INPUT -c copy -bsf noise output.mkv
-@end example
-
 @section remove_extra
 
 @c man end BITSTREAM FILTERS

doc/build_system.txt

@@ -25,9 +25,6 @@ fate-list
 install
     Install headers, libraries and programs.
 
-examples
-    Build all examples located in doc/examples.
-
 libavformat/output-example
     Build the libavformat basic example.
 
@@ -37,9 +34,6 @@ libavcodec/api-example
 libswscale/swscale-test
     Build the swscale self-test (useful also as example).
 
-config
-    Reconfigure the project with current configuration.
-
 
 Useful standard make commands:
 make -t <target>

doc/codecs.texi

@@ -172,13 +172,7 @@ Set max video quantizer scale (VBR). Must be included between -1 and
 Set max difference between the quantizer scale (VBR).
 
 @item bf @var{integer} (@emph{encoding,video})
-Set max number of B frames between non-B-frames.
-
-Must be an integer between -1 and 16. 0 means that B-frames are
-disabled. If a value of -1 is used, it will choose an automatic value
-depending on the encoder.
-
-Default value is 0.
+Set max number of B frames.
 
 @item b_qfactor @var{float} (@emph{encoding,video})
 Set qp factor between P and B frames.
@@ -285,11 +279,6 @@ detect bitstream specification deviations
 detect improper bitstream length
 @item explode
 abort decoding on minor error detection
-@item ignore_err
-ignore decoding errors, and continue decoding.
-This is useful if you want to analyze the content of a video and thus want
-everything to be decoded no matter what. This option will not result in a video
-that is pleasing to watch in case of errors.
 @item careful
 consider things that violate the spec and have not been seen in the wild as errors
 @item compliant
@@ -394,9 +383,6 @@ Possible values:
 
 @item simplemmx
 
-@item simpleauto
-Automatically pick a IDCT compatible with the simple one
-
 @item arm
 
 @item altivec
@@ -432,8 +418,6 @@ Possible values:
 iterative motion vector (MV) search (slow)
 @item deblock
 use strong deblock filter for damaged MBs
-@item favor_inter
-favor predicting from the previous frame instead of the current
 @end table
 
 @item bits_per_coded_sample @var{integer}
@@ -498,8 +482,6 @@ threading operations
 @item vismv @var{integer} (@emph{decoding,video})
 Visualize motion vectors (MVs).
 
-This option is deprecated, see the codecview filter instead.
-
 Possible values:
 @table @samp
 @item pf
@@ -799,9 +781,6 @@ Frame data might be split into multiple chunks.
 Show all frames before the first keyframe.
 @item skiprd
 Deprecated, use mpegvideo private options instead.
-@item export_mvs
-Export motion vectors into frame side-data (see @code{AV_FRAME_DATA_MOTION_VECTORS})
-for codecs that support it. See also @file{doc/examples/export_mvs.c}.
 @end table
 
 @item error @var{integer} (@emph{encoding,video})
@@ -892,9 +871,6 @@ Set frame skip factor.
 
 @item skip_exp @var{integer} (@emph{encoding,video})
 Set frame skip exponent.
-Negative values behave identical to the corresponding positive ones, except
-that the score is normalized.
-Positive values exist primarily for compatibility reasons and are not so useful.
 
 @item skipcmp @var{integer} (@emph{encoding,video})
 Set frame skip compare function.
@@ -1040,26 +1016,15 @@ Set the log level offset.
 Number of slices, used in parallelized encoding.
 
 @item thread_type @var{flags} (@emph{decoding/encoding,video})
-Select which multithreading methods to use.
-
-Use of @samp{frame} will increase decoding delay by one frame per
-thread, so clients which cannot provide future frames should not use
-it.
+Select multithreading type.
 
 Possible values:
 @table @samp
 @item slice
-Decode more than one part of a single frame at once.
-
-Multithreading using slices works only when the video was encoded with
-slices.
 
 @item frame
-Decode more than one frame at once.
+
 @end table
-
-Default value is @samp{slice+frame}.
-
 @item audio_service_type @var{integer} (@emph{encoding,audio})
 Set audio service type.
 
@@ -1118,9 +1083,5 @@ instead of alpha. Default is 0.
 
 @c man end CODEC OPTIONS
 
-@ifclear config-writeonly
 @include decoders.texi
-@end ifclear
-@ifclear config-readonly
 @include encoders.texi
-@end ifclear

doc/decoders.texi

@@ -14,7 +14,7 @@ You can disable all the decoders with the configure option
 with the options @code{--enable-decoder=@var{DECODER}} /
 @code{--disable-decoder=@var{DECODER}}.
 
-The option @code{-decoders} of the ff* tools will display the list of
+The option @code{-codecs} of the ff* tools will display the list of
 enabled decoders.
 
 @c man end DECODERS
@@ -52,37 +52,6 @@ top-field-first is assumed
 @chapter Audio Decoders
 @c man begin AUDIO DECODERS
 
-A description of some of the currently available audio decoders
-follows.
-
-@section ac3
-
-AC-3 audio decoder.
-
-This decoder implements part of ATSC A/52:2010 and ETSI TS 102 366, as well as
-the undocumented RealAudio 3 (a.k.a. dnet).
-
-@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.
@@ -163,9 +132,6 @@ Requires the presence of the libopus headers and library during
 configuration. You need to explicitly configure the build with
 @code{--enable-libopus}.
 
-An FFmpeg native decoder for Opus exists, so users can decode Opus
-without this library.
-
 @c man end AUDIO DECODERS
 
 @chapter Subtitles Decoders

doc/default.css

@@ -17,8 +17,8 @@ a:visited {
 }
 
 #banner img {
-    margin-bottom: 1px;
-    margin-top: 5px;
+    padding-bottom: 1px;
+    padding-top: 5px;
 }
 
 #body {

doc/demuxers.texi

@@ -74,7 +74,7 @@ following directive is recognized:
 Path to a file to read; special characters and spaces must be escaped with
 backslash or single quotes.
 
-All subsequent file-related directives apply to that file.
+All subsequent directives apply to that file.
 
 @item @code{ffconcat version 1.0}
 Identify the script type and version. It also sets the @option{safe} option
@@ -92,22 +92,6 @@ file is not available or accurate.
 If the duration is set for all files, then it is possible to seek in the
 whole concatenated video.
 
-@item @code{stream}
-Introduce a stream in the virtual file.
-All subsequent stream-related directives apply to the last introduced
-stream.
-Some streams properties must be set in order to allow identifying the
-matching streams in the subfiles.
-If no streams are defined in the script, the streams from the first file are
-copied.
-
-@item @code{exact_stream_id @var{id}}
-Set the id of the stream.
-If this directive is given, the string with the corresponding id in the
-subfiles will be used.
-This is especially useful for MPEG-PS (VOB) files, where the order of the
-streams is not reliable.
-
 @end table
 
 @subsection Options
@@ -128,14 +112,6 @@ If set to 0, any file name is accepted.
 The default is -1, it is equivalent to 1 if the format was automatically
 probed and 0 otherwise.
 
-@item auto_convert
-If set to 1, try to perform automatic conversions on packet data to make the
-streams concatenable.
-
-Currently, the only conversion is adding the h264_mp4toannexb bitstream
-filter to H.264 streams in MP4 format. This is necessary in particular if
-there are resolution changes.
-
 @end table
 
 @section flv
@@ -174,40 +150,6 @@ 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 gif
-
-Animated GIF demuxer.
-
-It accepts the following options:
-
-@table @option
-@item min_delay
-Set the minimum valid delay between frames in hundredths of seconds.
-Range is 0 to 6000. Default value is 2.
-
-@item default_delay
-Set the default delay between frames in hundredths of seconds.
-Range is 0 to 6000. Default value is 10.
-
-@item ignore_loop
-GIF files can contain information to loop a certain number of times (or
-infinitely). If @option{ignore_loop} is set to 1, then the loop setting
-from the input will be ignored and looping will not occur. If set to 0,
-then looping will occur and will cycle the number of times according to
-the GIF. Default value is 1.
-@end table
-
-For example, with the overlay filter, place an infinitely looping GIF
-over another video:
-@example
-ffmpeg -i input.mp4 -ignore_loop 0 -i input.gif -filter_complex overlay=shortest=1 out.mkv
-@end example
-
-Note that in the above example the shortest option for overlay filter is
-used to end the output video at the length of the shortest input file,
-which in this case is @file{input.mp4} as the GIF in this example loops
-infinitely.
-
 @section image2
 
 Image file demuxer.
@@ -307,8 +249,6 @@ is 5.
 If set to 1, will set frame timestamp to modification time of image file. Note
 that monotonity of timestamps is not provided: images go in the same order as
 without this option. Default value is 0.
-If set to 2, will set frame timestamp to the modification time of the image file in
-nanosecond precision.
 @item video_size
 Set the video size of the images to read. If not specified the video
 size is guessed from the first image file in the sequence.
@@ -356,7 +296,7 @@ teletext packet PTS and DTS values untouched.
 
 Raw video demuxer.
 
-This demuxer allows one to read raw video data. Since there is no header
+This demuxer allows to read raw video data. Since there is no header
 specifying the assumed video parameters, the user must specify them
 in order to be able to decode the data correctly.
 

doc/developer.texi

@@ -92,7 +92,7 @@ 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 */
@@ -248,7 +248,7 @@ 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://www.isc.org/software/license/, ISC} or
 @uref{http://mit-license.org/, MIT} license.
 @uref{http://www.gnu.org/licenses/gpl-2.0.html, GPL 2} including
 an "or any later version" clause is also acceptable, but LGPL is
@@ -323,12 +323,9 @@ Always fill out the commit log message. Describe in a few lines what you
 changed and why. You can refer to mailing list postings if you fix a
 particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.
 Recommended format:
-
-@example
 area changed: Short 1 line description
 
 details describing what and why and giving references.
-@end example
 
 @item
 Make sure the author of the commit is set correctly. (see git commit --author)

doc/devices.texi

@@ -17,9 +17,5 @@ for programmatic use.
 
 @c man end DEVICE OPTIONS
 
-@ifclear config-writeonly
 @include indevs.texi
-@end ifclear
-@ifclear config-readonly
 @include outdevs.texi
-@end ifclear

doc/doxy-wrapper.sh

@@ -2,12 +2,13 @@
 
 SRC_PATH="${1}"
 DOXYFILE="${2}"
-DOXYGEN="${3}"
 
-shift 3
+shift 2
 
-$DOXYGEN - <<EOF
+doxygen - <<EOF
 @INCLUDE        = ${DOXYFILE}
 INPUT           = $@
-EXAMPLE_PATH    = ${SRC_PATH}/doc/examples
+HTML_HEADER     = ${SRC_PATH}/doc/doxy/header.html
+HTML_FOOTER     = ${SRC_PATH}/doc/doxy/footer.html
+HTML_STYLESHEET = ${SRC_PATH}/doc/doxy/doxy_stylesheet.css
 EOF

doc/doxy/footer.html

@@ -0,0 +1,9 @@
+
+ <footer class="footer pagination-right">
+   <span class="label label-info">
+    Generated on $datetime for $projectname by&#160;<a href="http://www.doxygen.org/index.html">doxygen</a> $doxygenversion
+   </span>
+ </footer>
+</div>
+</body>
+</html>

doc/doxy/header.html

@@ -0,0 +1,16 @@
+<!DOCTYPE html>
+<html>
+<head>
+<meta http-equiv="Content-Type" content="text/html; charset=UTF-8"/>
+<meta http-equiv="X-UA-Compatible" content="IE=9"/>
+<!--BEGIN PROJECT_NAME--><title>$projectname: $title</title><!--END PROJECT_NAME-->
+<!--BEGIN !PROJECT_NAME--><title>$title</title><!--END !PROJECT_NAME-->
+<link href="$relpath$doxy_stylesheet.css" rel="stylesheet" type="text/css" />
+<!--Header replace -->
+
+</head>
+
+<div class="container">
+
+<!--Header replace -->
+<div class="menu">

doc/encoders.texi

@@ -14,7 +14,7 @@ You can disable all the encoders with the configure option
 with the options @code{--enable-encoder=@var{ENCODER}} /
 @code{--disable-encoder=@var{ENCODER}}.
 
-The option @code{-encoders} of the ff* tools will display the list of
+The option @code{-codecs} of the ff* tools will display the list of
 enabled encoders.
 
 @c man end ENCODERS
@@ -38,8 +38,8 @@ As this encoder is experimental, unexpected behavior may exist from time to
 time. For a more stable AAC encoder, see @ref{libvo-aacenc}. However, be warned
 that it has a worse quality reported by some users.
 
-@c todo @ref{libaacplus}
-See also @ref{libfdk-aac-enc,,libfdk_aac} and @ref{libfaac}.
+@c Comment this out until somebody writes the respective documentation.
+@c See also @ref{libfaac}, @ref{libaacplus}, and @ref{libfdk-aac-enc}.
 
 @subsection Options
 
@@ -80,7 +80,7 @@ thresholds with quantizer steps to find the appropriate quantization with
 distortion below threshold band by band.
 
 The quality of this method is comparable to the two loop searching method
-described below, but somewhat a little better and slower.
+descibed below, but somewhat a little better and slower.
 
 @item anmr
 Average noise to mask ratio (ANMR) trellis-based solution.
@@ -494,285 +494,6 @@ Selected by Encoder (default)
 
 @end table
 
-@anchor{libfaac}
-@section libfaac
-
-libfaac AAC (Advanced Audio Coding) encoder wrapper.
-
-Requires the presence of the libfaac headers and library during
-configuration. You need to explicitly configure the build with
-@code{--enable-libfaac --enable-nonfree}.
-
-This encoder is considered to be of higher quality with respect to the
-@ref{aacenc,,the native experimental FFmpeg AAC encoder}.
-
-For more information see the libfaac project at
-@url{http://www.audiocoding.com/faac.html/}.
-
-@subsection Options
-
-The following shared FFmpeg codec options are recognized.
-
-The following options are supported by the libfaac wrapper. The
-@command{faac}-equivalent of the options are listed in parentheses.
-
-@table @option
-@item b (@emph{-b})
-Set bit rate in bits/s for ABR (Average Bit Rate) mode. If the bit rate
-is not explicitly specified, it is automatically set to a suitable
-value depending on the selected profile. @command{faac} bitrate is
-expressed in kilobits/s.
-
-Note that libfaac does not support CBR (Constant Bit Rate) but only
-ABR (Average Bit Rate).
-
-If VBR mode is enabled this option is ignored.
-
-@item ar (@emph{-R})
-Set audio sampling rate (in Hz).
-
-@item ac (@emph{-c})
-Set the number of audio channels.
-
-@item cutoff (@emph{-C})
-Set cutoff frequency. If not specified (or explicitly set to 0) it
-will use a value automatically computed by the library. Default value
-is 0.
-
-@item profile
-Set audio profile.
-
-The following profiles are recognized:
-@table @samp
-@item aac_main
-Main AAC (Main)
-
-@item aac_low
-Low Complexity AAC (LC)
-
-@item aac_ssr
-Scalable Sample Rate (SSR)
-
-@item aac_ltp
-Long Term Prediction (LTP)
-@end table
-
-If not specified it is set to @samp{aac_low}.
-
-@item flags +qscale
-Set constant quality VBR (Variable Bit Rate) mode.
-
-@item global_quality
-Set quality in VBR mode as an integer number of lambda units.
-
-Only relevant when VBR mode is enabled with @code{flags +qscale}.  The
-value is converted to QP units by dividing it by @code{FF_QP2LAMBDA},
-and used to set the quality value used by libfaac. A reasonable range
-for the option value in QP units is [10-500], the higher the value the
-higher the quality.
-
-@item q (@emph{-q})
-Enable VBR mode when set to a non-negative value, and set constant
-quality value as a double floating point value in QP units.
-
-The value sets the quality value used by libfaac. A reasonable range
-for the option value is [10-500], the higher the value the higher the
-quality.
-
-This option is valid only using the @command{ffmpeg} command-line
-tool. For library interface users, use @option{global_quality}.
-@end table
-
-@subsection Examples
-
-@itemize
-@item
-Use @command{ffmpeg} to convert an audio file to ABR 128 kbps AAC in an M4A (MP4)
-container:
-@example
-ffmpeg -i input.wav -codec:a libfaac -b:a 128k -output.m4a
-@end example
-
-@item
-Use @command{ffmpeg} to convert an audio file to VBR AAC, using the
-LTP AAC profile:
-@example
-ffmpeg -i input.wav -c:a libfaac -profile:a aac_ltp -q:a 100 output.m4a
-@end example
-@end itemize
-
-@anchor{libfdk-aac-enc}
-@section libfdk_aac
-
-libfdk-aac AAC (Advanced Audio Coding) encoder wrapper.
-
-The libfdk-aac library is based on the Fraunhofer FDK AAC code from
-the Android project.
-
-Requires the presence of the libfdk-aac headers and library during
-configuration. You need to explicitly configure the build with
-@code{--enable-libfdk-aac}. The library is also incompatible with GPL,
-so if you allow the use of GPL, you should configure with
-@code{--enable-gpl --enable-nonfree --enable-libfdk-aac}.
-
-This encoder is considered to be of higher quality with respect to
-both @ref{aacenc,,the native experimental FFmpeg AAC encoder} and
-@ref{libfaac}.
-
-VBR encoding, enabled through the @option{vbr} or @option{flags
-+qscale} options, is experimental and only works with some
-combinations of parameters.
-
-Support for encoding 7.1 audio is only available with libfdk-aac 0.1.3 or
-higher.
-
-For more information see the fdk-aac project at
-@url{http://sourceforge.net/p/opencore-amr/fdk-aac/}.
-
-@subsection Options
-
-The following options are mapped on the shared FFmpeg codec options.
-
-@table @option
-@item b
-Set bit rate in bits/s. If the bitrate is not explicitly specified, it
-is automatically set to a suitable value depending on the selected
-profile.
-
-In case VBR mode is enabled the option is ignored.
-
-@item ar
-Set audio sampling rate (in Hz).
-
-@item channels
-Set the number of audio channels.
-
-@item flags +qscale
-Enable fixed quality, VBR (Variable Bit Rate) mode.
-Note that VBR is implicitly enabled when the @option{vbr} value is
-positive.
-
-@item cutoff
-Set cutoff frequency. If not specified (or explicitly set to 0) it
-will use a value automatically computed by the library. Default value
-is 0.
-
-@item profile
-Set audio profile.
-
-The following profiles are recognized:
-@table @samp
-@item aac_low
-Low Complexity AAC (LC)
-
-@item aac_he
-High Efficiency AAC (HE-AAC)
-
-@item aac_he_v2
-High Efficiency AAC version 2 (HE-AACv2)
-
-@item aac_ld
-Low Delay AAC (LD)
-
-@item aac_eld
-Enhanced Low Delay AAC (ELD)
-@end table
-
-If not specified it is set to @samp{aac_low}.
-@end table
-
-The following are private options of the libfdk_aac encoder.
-
-@table @option
-@item afterburner
-Enable afterburner feature if set to 1, disabled if set to 0. This
-improves the quality but also the required processing power.
-
-Default value is 1.
-
-@item eld_sbr
-Enable SBR (Spectral Band Replication) for ELD if set to 1, disabled
-if set to 0.
-
-Default value is 0.
-
-@item signaling
-Set SBR/PS signaling style.
-
-It can assume one of the following values:
-@table @samp
-@item default
-choose signaling implicitly (explicit hierarchical by default,
-implicit if global header is disabled)
-
-@item implicit
-implicit backwards compatible signaling
-
-@item explicit_sbr
-explicit SBR, implicit PS signaling
-
-@item explicit_hierarchical
-explicit hierarchical signaling
-@end table
-
-Default value is @samp{default}.
-
-@item latm
-Output LATM/LOAS encapsulated data if set to 1, disabled if set to 0.
-
-Default value is 0.
-
-@item header_period
-Set StreamMuxConfig and PCE repetition period (in frames) for sending
-in-band configuration buffers within LATM/LOAS transport layer.
-
-Must be a 16-bits non-negative integer.
-
-Default value is 0.
-
-@item vbr
-Set VBR mode, from 1 to 5. 1 is lowest quality (though still pretty
-good) and 5 is highest quality. A value of 0 will disable VBR, and CBR
-(Constant Bit Rate) is enabled.
-
-Currently only the @samp{aac_low} profile supports VBR encoding.
-
-VBR modes 1-5 correspond to roughly the following average bit rates:
-
-@table @samp
-@item 1
-32 kbps/channel
-@item 2
-40 kbps/channel
-@item 3
-48-56 kbps/channel
-@item 4
-64 kbps/channel
-@item 5
-about 80-96 kbps/channel
-@end table
-
-Default value is 0.
-@end table
-
-@subsection Examples
-
-@itemize
-@item
-Use @command{ffmpeg} to convert an audio file to VBR AAC in an M4A (MP4)
-container:
-@example
-ffmpeg -i input.wav -codec:a libfdk_aac -vbr 3 output.m4a
-@end example
-
-@item
-Use @command{ffmpeg} to convert an audio file to CBR 64k kbps AAC, using the
-High-Efficiency AAC profile:
-@example
-ffmpeg -i input.wav -c:a libfdk_aac -profile:a aac_he -b:a 64k output.m4a
-@end example
-@end itemize
-
 @anchor{libmp3lame}
 @section libmp3lame
 
@@ -792,7 +513,7 @@ The following options are supported by the libmp3lame wrapper. The
 
 @table @option
 @item b (@emph{-b})
-Set bitrate expressed in bits/s for CBR or ABR. LAME @code{bitrate} is
+Set bitrate expressed in bits/s for CBR. LAME @code{bitrate} is
 expressed in kilobits/s.
 
 @item q (@emph{-V})
@@ -807,18 +528,13 @@ while producing the worst quality.
 
 @item reservoir
 Enable use of bit reservoir when set to 1. Default value is 1. LAME
-has this enabled by default, but can be overridden by use
+has this enabled by default, but can be overriden by use
 @option{--nores} option.
 
 @item joint_stereo (@emph{-m j})
 Enable the encoder to use (on a frame by frame basis) either L/R
 stereo or mid/side stereo. Default value is 1.
 
-@item abr (@emph{--abr})
-Enable the encoder to use ABR when set to 1. The @command{lame}
-@option{--abr} sets the target bitrate, while this options only
-tells FFmpeg to use ABR still relies on @option{b} to set bitrate.
-
 @end table
 
 @section libopencore-amrnb
@@ -1032,7 +748,7 @@ configuration. You need to explicitly configure the build with
 
 @subsection Option Mapping
 
-Most libopus options are modelled after the @command{opusenc} utility from
+Most libopus options are modeled after the @command{opusenc} utility from
 opus-tools. The following is an option mapping chart describing options
 supported by the libopus wrapper, and their @command{opusenc}-equivalent
 in parentheses.
@@ -1070,7 +786,7 @@ Set maximum frame size, or duration of a frame in milliseconds. The
 argument must be exactly the following: 2.5, 5, 10, 20, 40, 60. Smaller
 frame sizes achieve lower latency but less quality at a given bitrate.
 Sizes greater than 20ms are only interesting at fairly low bitrates.
-The default is 20ms.
+The default of FFmpeg is 10ms, but is 20ms in @command{opusenc}.
 
 @item packet_loss (@emph{expect-loss})
 Set expected packet loss percentage. The default is 0.
@@ -1147,111 +863,32 @@ transient response is a higher bitrate.
 
 @end table
 
-@anchor{libwavpack}
 @section libwavpack
 
 A wrapper providing WavPack encoding through libwavpack.
 
 Only lossless mode using 32-bit integer samples is supported currently.
-
-Requires the presence of the libwavpack headers and library during
-configuration. You need to explicitly configure the build with
-@code{--enable-libwavpack}.
-
-Note that a libavcodec-native encoder for the WavPack codec exists so users can
-encode audios with this codec without using this encoder. See @ref{wavpackenc}.
-
-@subsection Options
-
-@command{wavpack} command line utility's corresponding options are listed in
-parentheses, if any.
+The @option{compression_level} option can be used to control speed vs.
+compression tradeoff, with the values mapped to libwavpack as follows:
 
 @table @option
-@item frame_size (@emph{--blocksize})
-Default is 32768.
 
-@item compression_level
-Set speed vs. compression tradeoff. Acceptable arguments are listed below:
-
-@table @samp
-@item 0 (@emph{-f})
-Fast mode.
+@item 0
+Fast mode - corresponding to the wavpack @option{-f} option.
 
 @item 1
 Normal (default) settings.
 
-@item 2 (@emph{-h})
-High quality.
+@item 2
+High quality - corresponding to the wavpack @option{-h} option.
 
-@item 3 (@emph{-hh})
-Very high quality.
+@item 3
+Very high quality - corresponding to the wavpack @option{-hh} option.
 
-@item 4-8 (@emph{-hh -x}@var{EXTRAPROC})
-Same as @samp{3}, but with extra processing enabled.
-
-@samp{4} is the same as @option{-x2} and @samp{8} is the same as @option{-x6}.
-
-@end table
-@end table
-
-@anchor{wavpackenc}
-@section wavpack
-
-WavPack lossless audio encoder.
-
-This is a libavcodec-native WavPack encoder. There is also an encoder based on
-libwavpack, but there is virtually no reason to use that encoder.
-
-See also @ref{libwavpack}.
-
-@subsection Options
-
-The equivalent options for @command{wavpack} command line utility are listed in
-parentheses.
-
-@subsubsection Shared options
-
-The following shared options are effective for this encoder. Only special notes
-about this particular encoder will be documented here. For the general meaning
-of the options, see @ref{codec-options,,the Codec Options chapter}.
-
-@table @option
-@item frame_size (@emph{--blocksize})
-For this encoder, the range for this option is between 128 and 131072. Default
-is automatically decided based on sample rate and number of channel.
-
-For the complete formula of calculating default, see
-@file{libavcodec/wavpackenc.c}.
-
-@item compression_level (@emph{-f}, @emph{-h}, @emph{-hh}, and @emph{-x})
-This option's syntax is consistent with @ref{libwavpack}'s.
-@end table
-
-@subsubsection Private options
-
-@table @option
-@item joint_stereo (@emph{-j})
-Set whether to enable joint stereo. Valid values are:
-
-@table @samp
-@item on (@emph{1})
-Force mid/side audio encoding.
-@item off (@emph{0})
-Force left/right audio encoding.
-@item auto
-Let the encoder decide automatically.
-@end table
-
-@item optimize_mono
-Set whether to enable optimization for mono. This option is only effective for
-non-mono streams. Available values:
-
-@table @samp
-@item on
-enabled
-@item off
-disabled
-@end table
+@item 4-8
+Same as 3, but with extra processing enabled - corresponding to the wavpack
+@option{-x} option. I.e. 4 is the same as @option{-x2} and 8 is the same as
+@option{-x6}.
 
 @end table
 
@@ -1265,15 +902,12 @@ follows.
 
 @section libtheora
 
-libtheora Theora encoder wrapper.
+Theora format supported through libtheora.
 
 Requires the presence of the libtheora headers and library during
 configuration. You need to explicitly configure the build with
 @code{--enable-libtheora}.
 
-For more information about the libtheora project see
-@url{http://www.theora.org/}.
-
 @subsection Options
 
 The following global options are mapped to internal libtheora options
@@ -1281,11 +915,11 @@ which affect the quality and the bitrate of the encoded stream.
 
 @table @option
 @item b
-Set the video bitrate in bit/s for CBR (Constant Bit Rate) mode.  In
-case VBR (Variable Bit Rate) mode is enabled this option is ignored.
+Set the video bitrate, only works if the @code{qscale} flag in
+@option{flags} is not enabled.
 
 @item flags
-Used to enable constant quality mode (VBR) encoding through the
+Used to enable constant quality mode encoding through the
 @option{qscale} flag, and to enable the @code{pass1} and @code{pass2}
 modes.
 
@@ -1293,44 +927,22 @@ modes.
 Set the GOP size.
 
 @item global_quality
-Set the global quality as an integer in lambda units.
+Set the global quality in lambda units, only works if the
+@code{qscale} flag in @option{flags} is enabled. The value is clipped
+in the [0 - 10*@code{FF_QP2LAMBDA}] range, and then multiplied for 6.3
+to get a value in the native libtheora range [0-63]. A higher value
+corresponds to a higher quality.
 
-Only relevant when VBR mode is enabled with @code{flags +qscale}. The
-value is converted to QP units by dividing it by @code{FF_QP2LAMBDA},
-clipped in the [0 - 10] range, and then multiplied by 6.3 to get a
-value in the native libtheora range [0-63]. A higher value corresponds
-to a higher quality.
-
-@item q
-Enable VBR mode when set to a non-negative value, and set constant
-quality value as a double floating point value in QP units.
-
-The value is clipped in the [0-10] range, and then multiplied by 6.3
-to get a value in the native libtheora range [0-63].
-
-This option is valid only using the @command{ffmpeg} command-line
-tool. For library interface users, use @option{global_quality}.
+For example, to set maximum constant quality encoding with
+@command{ffmpeg}:
+@example
+ffmpeg -i INPUT -flags:v qscale -global_quality:v "10*QP2LAMBDA" -codec:v libtheora OUTPUT.ogg
+@end example
 @end table
 
-@subsection Examples
-
-@itemize
-@item
-Set maximum constant quality (VBR) encoding with @command{ffmpeg}:
-@example
-ffmpeg -i INPUT -codec:v libtheora -q:v 10 OUTPUT.ogg
-@end example
-
-@item
-Use @command{ffmpeg} to convert a CBR 1000 kbps Theora video stream:
-@example
-ffmpeg -i INPUT -codec:v libtheora -b:v 1000k OUTPUT.ogg
-@end example
-@end itemize
-
 @section libvpx
 
-VP8/VP9 format supported through libvpx.
+VP8 format supported through libvpx.
 
 Requires the presence of the libvpx headers and library during configuration.
 You need to explicitly configure the build with @code{--enable-libvpx}.
@@ -1442,77 +1054,12 @@ g_lag_in_frames
 @item vp8flags error_resilient
 g_error_resilient
 
-@item aq_mode
-@code{VP9E_SET_AQ_MODE}
-
 @end table
 
 For more information about libvpx see:
 @url{http://www.webmproject.org/}
 
-
-@section libwebp
-
-libwebp WebP Image encoder wrapper
-
-libwebp is Google's official encoder for WebP images. It can encode in either
-lossy or lossless mode. Lossy images are essentially a wrapper around a VP8
-frame. Lossless images are a separate codec developed by Google.
-
-@subsection Pixel Format
-
-Currently, libwebp only supports YUV420 for lossy and RGB for lossless due
-to limitations of the format and libwebp. Alpha is supported for either mode.
-Because of API limitations, if RGB is passed in when encoding lossy or YUV is
-passed in for encoding lossless, the pixel format will automatically be
-converted using functions from libwebp. This is not ideal and is done only for
-convenience.
-
-@subsection Options
-
-@table @option
-
-@item -lossless @var{boolean}
-Enables/Disables use of lossless mode. Default is 0.
-
-@item -compression_level @var{integer}
-For lossy, this is a quality/speed tradeoff. Higher values give better quality
-for a given size at the cost of increased encoding time. For lossless, this is
-a size/speed tradeoff. Higher values give smaller size at the cost of increased
-encoding time. More specifically, it controls the number of extra algorithms
-and compression tools used, and varies the combination of these tools. This
-maps to the @var{method} option in libwebp. The valid range is 0 to 6.
-Default is 4.
-
-@item -qscale @var{float}
-For lossy encoding, this controls image quality, 0 to 100. For lossless
-encoding, this controls the effort and time spent at compressing more. The
-default value is 75. Note that for usage via libavcodec, this option is called
-@var{global_quality} and must be multiplied by @var{FF_QP2LAMBDA}.
-
-@item -preset @var{type}
-Configuration preset. This does some automatic settings based on the general
-type of the image.
-@table @option
-@item none
-Do not use a preset.
-@item default
-Use the encoder default.
-@item picture
-Digital picture, like portrait, inner shot
-@item photo
-Outdoor photograph, with natural lighting
-@item drawing
-Hand or line drawing, with high-contrast details
-@item icon
-Small-sized colorful images
-@item text
-Text-like
-@end table
-
-@end table
-
-@section libx264, libx264rgb
+@section libx264
 
 x264 H.264/MPEG-4 AVC encoder wrapper.
 
@@ -1528,22 +1075,12 @@ for detail retention (adaptive quantization, psy-RD, psy-trellis).
 Many libx264 encoder options are mapped to FFmpeg global codec
 options, while unique encoder options are provided through private
 options. Additionally the @option{x264opts} and @option{x264-params}
-private options allows one to pass a list of key=value tuples as accepted
+private options allows to pass a list of key=value tuples as accepted
 by the libx264 @code{x264_param_parse} function.
 
 The x264 project website is at
 @url{http://www.videolan.org/developers/x264.html}.
 
-The libx264rgb encoder is the same as libx264, except it accepts packed RGB
-pixel formats as input instead of YUV.
-
-@subsection Supported Pixel Formats
-
-x264 supports 8- to 10-bit color spaces. The exact bit depth is controlled at
-x264's configure time. FFmpeg only supports one bit depth in one particular
-build. In other words, it is not possible to build one FFmpeg with multiple
-versions of x264 with different bit depths.
-
 @subsection Options
 
 The following options are supported by the libx264 wrapper. The
@@ -1569,34 +1106,25 @@ kilobits/s.
 
 @item g (@emph{keyint})
 
-@item qmin (@emph{qpmin})
-Minimum quantizer scale.
-
 @item qmax (@emph{qpmax})
-Maximum quantizer scale.
+
+@item qmin (@emph{qpmin})
 
 @item qdiff (@emph{qpstep})
-Maximum difference between quantizer scales.
 
 @item qblur (@emph{qblur})
-Quantizer curve blur
 
 @item qcomp (@emph{qcomp})
-Quantizer curve compression factor
 
 @item refs (@emph{ref})
-Number of reference frames each P-frame can use. The range is from @var{0-16}.
 
 @item sc_threshold (@emph{scenecut})
-Sets the threshold for the scene change detection.
 
 @item trellis (@emph{trellis})
-Performs Trellis quantization to increase efficiency. Enabled by default.
 
 @item nr  (@emph{nr})
 
 @item me_range (@emph{merange})
-Maximum range of the motion search in pixels.
 
 @item me_method (@emph{me})
 Set motion estimation method. Possible values in the decreasing order
@@ -1618,13 +1146,10 @@ Hadamard exhaustive search (slowest).
 @end table
 
 @item subq (@emph{subme})
-Sub-pixel motion estimation method.
 
 @item b_strategy (@emph{b-adapt})
-Adaptive B-frame placement decision algorithm. Use only on first-pass.
 
 @item keyint_min (@emph{min-keyint})
-Minimum GOP size.
 
 @item coder
 Set entropy encoder. Possible values:
@@ -1651,7 +1176,6 @@ Ignore chroma in motion estimation. It generates the same effect as
 @end table
 
 @item threads (@emph{threads})
-Number of encoding threads.
 
 @item thread_type
 Set multithreading technique. Possible values:
@@ -1869,7 +1393,7 @@ Override the x264 configuration using a :-separated list of key=value
 parameters.
 
 This option is functionally the same as the @option{x264opts}, but is
-duplicated for compatibility with the Libav fork.
+duplicated for compability with the Libav fork.
 
 For example to specify libx264 encoding options with @command{ffmpeg}:
 @example
@@ -2045,30 +1569,6 @@ fastest.
 
 @end table
 
-@section mpeg2
-
-MPEG-2 video encoder.
-
-@subsection Options
-
-@table @option
-@item seq_disp_ext @var{integer}
-Specifies if the encoder should write a sequence_display_extension to the
-output.
-@table @option
-@item -1
-@itemx auto
-Decide automatically to write it or not (this is the default) by checking if
-the data to be written is different from the default or unspecified values.
-@item 0
-@itemx never
-Never write it.
-@item 1
-@itemx always
-Always write it.
-@end table
-@end table
-
 @section png
 
 PNG image encoder.
@@ -2087,7 +1587,7 @@ Set physical density of pixels, in dots per meter, unset by default
 Apple ProRes encoder.
 
 FFmpeg contains 2 ProRes encoders, the prores-aw and prores-ks encoder.
-The used encoder can be chosen with the @code{-vcodec} option.
+The used encoder can be choosen with the @code{-vcodec} option.
 
 @subsection Private Options for prores-ks
 
@@ -2150,27 +1650,3 @@ For the fastest encoding speed set the @option{qscale} parameter (4 is the
 recommended value) and do not set a size constraint.
 
 @c man end VIDEO ENCODERS
-
-@chapter Subtitles Encoders
-@c man begin SUBTITLES ENCODERS
-
-@section dvdsub
-
-This codec encodes the bitmap subtitle format that is used in DVDs.
-Typically they are stored in VOBSUB file pairs (*.idx + *.sub),
-and they can also be used in Matroska files.
-
-@subsection Options
-
-@table @option
-@item even_rows_fix
-When set to 1, enable a work-around that makes the number of pixel rows
-even in all subtitles.  This fixes a problem with some players that
-cut off the bottom row if the number is odd.  The work-around just adds
-a fully transparent row if needed.  The overhead is low, typically
-one byte per subtitle on average.
-
-By default, this work-around is disabled.
-@end table
-
-@c man end SUBTITLES ENCODERS

doc/examples/Makefile

@@ -11,24 +11,18 @@ CFLAGS += -Wall -g
 CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
 LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
 
-EXAMPLES=       avio_reading                       \
-                decoding_encoding                  \
-                demuxing_decoding                  \
-                extract_mvs                        \
+EXAMPLES=       decoding_encoding                  \
+                demuxing                           \
                 filtering_video                    \
                 filtering_audio                    \
                 metadata                           \
                 muxing                             \
-                remuxing                           \
                 resampling_audio                   \
                 scaling_video                      \
-                transcode_aac                      \
-                transcoding                        \
 
 OBJS=$(addsuffix .o,$(EXAMPLES))
 
 # the following examples make explicit use of the math library
-avcodec:           LDLIBS += -lm
 decoding_encoding: LDLIBS += -lm
 muxing:            LDLIBS += -lm
 resampling_audio:  LDLIBS += -lm

doc/examples/README

@@ -5,19 +5,14 @@ Both following use cases rely on pkg-config and make, thus make sure
 that you have them installed and working on your system.
 
 
-Method 1: build the installed examples in a generic read/write user directory
+1) Build the installed examples in a generic read/write user directory
 
 Copy to a read/write user directory and just use "make", it will link
 to the libraries on your system, assuming the PKG_CONFIG_PATH is
 correctly configured.
 
-Method 2: build the examples in-tree
+2) Build the examples in-tree
 
 Assuming you are in the source FFmpeg checkout directory, you need to build
-FFmpeg (no need to make install in any prefix). Then just run "make examples".
-This will build the examples using the FFmpeg build system. You can clean those
-examples using "make examplesclean"
-
-If you want to try the dedicated Makefile examples (to emulate the first
-method), go into doc/examples and run a command such as
-PKG_CONFIG_PATH=pc-uninstalled make.
+FFmpeg (no need to make install in any prefix). Then you can go into
+doc/examples and run a command such as PKG_CONFIG_PATH=pc-uninstalled make.

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/decoding_encoding.c

@@ -24,10 +24,10 @@
  * @file
  * libavcodec API use example.
  *
- * @example decoding_encoding.c
  * Note that libavcodec only handles codecs (mpeg, mpeg4, etc...),
  * not file formats (avi, vob, mp4, mov, mkv, mxf, flv, mpegts, mpegps, etc...). See library 'libavformat' for the
  * format handling
+ * @example doc/examples/decoding_encoding.c
  */
 
 #include <math.h>
@@ -156,7 +156,7 @@ static void audio_encode_example(const char *filename)
     }
 
     /* frame containing input raw audio */
-    frame = av_frame_alloc();
+    frame = avcodec_alloc_frame();
     if (!frame) {
         fprintf(stderr, "Could not allocate audio frame\n");
         exit(1);
@@ -170,10 +170,6 @@ static void audio_encode_example(const char *filename)
      * we calculate the size of the samples buffer in bytes */
     buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size,
                                              c->sample_fmt, 0);
-    if (buffer_size < 0) {
-        fprintf(stderr, "Could not get sample buffer size\n");
-        exit(1);
-    }
     samples = av_malloc(buffer_size);
     if (!samples) {
         fprintf(stderr, "Could not allocate %d bytes for samples buffer\n",
@@ -191,7 +187,7 @@ static void audio_encode_example(const char *filename)
     /* encode a single tone sound */
     t = 0;
     tincr = 2 * M_PI * 440.0 / c->sample_rate;
-    for (i = 0; i < 200; i++) {
+    for(i=0;i<200;i++) {
         av_init_packet(&pkt);
         pkt.data = NULL; // packet data will be allocated by the encoder
         pkt.size = 0;
@@ -231,7 +227,7 @@ static void audio_encode_example(const char *filename)
     fclose(f);
 
     av_freep(&samples);
-    av_frame_free(&frame);
+    avcodec_free_frame(&frame);
     avcodec_close(c);
     av_free(c);
 }
@@ -291,11 +287,12 @@ static void audio_decode_example(const char *outfilename, const char *filename)
         int got_frame = 0;
 
         if (!decoded_frame) {
-            if (!(decoded_frame = av_frame_alloc())) {
+            if (!(decoded_frame = avcodec_alloc_frame())) {
                 fprintf(stderr, "Could not allocate audio frame\n");
                 exit(1);
             }
-        }
+        } else
+            avcodec_get_frame_defaults(decoded_frame);
 
         len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
         if (len < 0) {
@@ -307,11 +304,6 @@ static void audio_decode_example(const char *outfilename, const char *filename)
             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;
@@ -337,7 +329,7 @@ static void audio_decode_example(const char *outfilename, const char *filename)
 
     avcodec_close(c);
     av_free(c);
-    av_frame_free(&decoded_frame);
+    avcodec_free_frame(&decoded_frame);
 }
 
 /*
@@ -374,18 +366,12 @@ static void video_encode_example(const char *filename, int codec_id)
     c->width = 352;
     c->height = 288;
     /* frames per second */
-    c->time_base = (AVRational){1,25};
-    /* emit one intra frame every ten frames
-     * check frame pict_type before passing frame
-     * to encoder, if frame->pict_type is AV_PICTURE_TYPE_I
-     * then gop_size is ignored and the output of encoder
-     * will always be I frame irrespective to gop_size
-     */
-    c->gop_size = 10;
-    c->max_b_frames = 1;
+    c->time_base= (AVRational){1,25};
+    c->gop_size = 10; /* emit one intra frame every ten frames */
+    c->max_b_frames=1;
     c->pix_fmt = AV_PIX_FMT_YUV420P;
 
-    if (codec_id == AV_CODEC_ID_H264)
+    if(codec_id == AV_CODEC_ID_H264)
         av_opt_set(c->priv_data, "preset", "slow", 0);
 
     /* open it */
@@ -400,7 +386,7 @@ static void video_encode_example(const char *filename, int codec_id)
         exit(1);
     }
 
-    frame = av_frame_alloc();
+    frame = avcodec_alloc_frame();
     if (!frame) {
         fprintf(stderr, "Could not allocate video frame\n");
         exit(1);
@@ -419,7 +405,7 @@ static void video_encode_example(const char *filename, int codec_id)
     }
 
     /* encode 1 second of video */
-    for (i = 0; i < 25; i++) {
+    for(i=0;i<25;i++) {
         av_init_packet(&pkt);
         pkt.data = NULL;    // packet data will be allocated by the encoder
         pkt.size = 0;
@@ -427,15 +413,15 @@ static void video_encode_example(const char *filename, int codec_id)
         fflush(stdout);
         /* prepare a dummy image */
         /* Y */
-        for (y = 0; y < c->height; y++) {
-            for (x = 0; x < c->width; x++) {
+        for(y=0;y<c->height;y++) {
+            for(x=0;x<c->width;x++) {
                 frame->data[0][y * frame->linesize[0] + x] = x + y + i * 3;
             }
         }
 
         /* Cb and Cr */
-        for (y = 0; y < c->height/2; y++) {
-            for (x = 0; x < c->width/2; x++) {
+        for(y=0;y<c->height/2;y++) {
+            for(x=0;x<c->width/2;x++) {
                 frame->data[1][y * frame->linesize[1] + x] = 128 + y + i * 2;
                 frame->data[2][y * frame->linesize[2] + x] = 64 + x + i * 5;
             }
@@ -481,7 +467,7 @@ static void video_encode_example(const char *filename, int codec_id)
     avcodec_close(c);
     av_free(c);
     av_freep(&frame->data[0]);
-    av_frame_free(&frame);
+    avcodec_free_frame(&frame);
     printf("\n");
 }
 
@@ -495,10 +481,10 @@ static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
     FILE *f;
     int i;
 
-    f = fopen(filename,"w");
-    fprintf(f, "P5\n%d %d\n%d\n", xsize, ysize, 255);
-    for (i = 0; i < ysize; i++)
-        fwrite(buf + i * wrap, 1, xsize, f);
+    f=fopen(filename,"w");
+    fprintf(f,"P5\n%d %d\n%d\n",xsize,ysize,255);
+    for(i=0;i<ysize;i++)
+        fwrite(buf + i * wrap,1,xsize,f);
     fclose(f);
 }
 
@@ -579,14 +565,14 @@ static void video_decode_example(const char *outfilename, const char *filename)
         exit(1);
     }
 
-    frame = av_frame_alloc();
+    frame = avcodec_alloc_frame();
     if (!frame) {
         fprintf(stderr, "Could not allocate video frame\n");
         exit(1);
     }
 
     frame_count = 0;
-    for (;;) {
+    for(;;) {
         avpkt.size = fread(inbuf, 1, INBUF_SIZE, f);
         if (avpkt.size == 0)
             break;
@@ -623,7 +609,7 @@ static void video_decode_example(const char *outfilename, const char *filename)
 
     avcodec_close(c);
     av_free(c);
-    av_frame_free(&frame);
+    avcodec_free_frame(&frame);
     printf("\n");
 }
 
@@ -640,7 +626,7 @@ int main(int argc, char **argv)
                "This program generates a synthetic stream and encodes it to a file\n"
                "named test.h264, test.mp2 or test.mpg depending on output_type.\n"
                "The encoded stream is then decoded and written to a raw data output.\n"
-               "output_type must be chosen between 'h264', 'mp2', 'mpg'.\n",
+               "output_type must be choosen between 'h264', 'mp2', 'mpg'.\n",
                argv[0]);
         return 1;
     }

doc/examples/demuxing.c

@@ -22,11 +22,11 @@
 
 /**
  * @file
- * Demuxing and decoding example.
+ * libavformat demuxing API use example.
  *
  * Show how to use the libavformat and libavcodec API to demux and
  * decode audio and video data.
- * @example demuxing_decoding.c
+ * @example doc/examples/demuxing.c
  */
 
 #include <libavutil/imgutils.h>
@@ -53,30 +53,16 @@ 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));
+            fprintf(stderr, "Error decoding video frame\n");
             return ret;
         }
 
@@ -99,7 +85,7 @@ static int decode_packet(int *got_frame, int cached)
         /* decode audio frame */
         ret = avcodec_decode_audio4(audio_dec_ctx, frame, got_frame, &pkt);
         if (ret < 0) {
-            fprintf(stderr, "Error decoding audio frame (%s)\n", av_err2str(ret));
+            fprintf(stderr, "Error decoding audio frame\n");
             return ret;
         }
         /* Some audio decoders decode only part of the packet, and have to be
@@ -127,11 +113,6 @@ static int decode_packet(int *got_frame, int cached)
         }
     }
 
-    /* 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;
 }
 
@@ -142,7 +123,6 @@ static int open_codec_context(int *stream_idx,
     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) {
@@ -159,13 +139,10 @@ static int open_codec_context(int *stream_idx,
         if (!dec) {
             fprintf(stderr, "Failed to find %s codec\n",
                     av_get_media_type_string(type));
-            return AVERROR(EINVAL);
+            return ret;
         }
 
-        /* Init the decoders, with or without reference counting */
-        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) {
+        if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
             fprintf(stderr, "Failed to open %s codec\n",
                     av_get_media_type_string(type));
             return ret;
@@ -208,31 +185,15 @@ 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"
+    if (argc != 4) {
+        fprintf(stderr, "usage: %s input_file video_output_file audio_output_file\n"
                 "API example program to show how to read frames from an input file.\n"
                 "This program reads frames from a file, decodes them, and writes decoded\n"
                 "video frames to a rawvideo file named video_output_file, and decoded\n"
-                "audio frames to a rawaudio file named audio_output_file.\n\n"
-                "If the -refcount option is specified, the program use the\n"
-                "reference counting frame system which allows keeping a copy of\n"
-                "the data for longer than one decode call. If unset, it's using\n"
-                "the classic old method.\n"
+                "audio frames to a rawaudio file named audio_output_file.\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];
@@ -279,7 +240,7 @@ int main (int argc, char **argv)
         audio_dec_ctx = audio_stream->codec;
         audio_dst_file = fopen(audio_dst_filename, "wb");
         if (!audio_dst_file) {
-            fprintf(stderr, "Could not open destination file %s\n", audio_dst_filename);
+            fprintf(stderr, "Could not open destination file %s\n", video_dst_filename);
             ret = 1;
             goto end;
         }
@@ -294,12 +255,7 @@ int main (int argc, char **argv)
         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();
+    frame = avcodec_alloc_frame();
     if (!frame) {
         fprintf(stderr, "Could not allocate frame\n");
         ret = AVERROR(ENOMEM);
@@ -369,17 +325,16 @@ int main (int argc, char **argv)
     }
 
 end:
-    avcodec_close(video_dec_ctx);
-    avcodec_close(audio_dec_ctx);
+    if (video_dec_ctx)
+        avcodec_close(video_dec_ctx);
+    if (audio_dec_ctx)
+        avcodec_close(audio_dec_ctx);
     avformat_close_input(&fmt_ctx);
     if (video_dst_file)
         fclose(video_dst_file);
     if (audio_dst_file)
         fclose(audio_dst_file);
-    if (api_mode == API_MODE_OLD)
-        avcodec_free_frame(&frame);
-    else
-        av_frame_free(&frame);
+    av_free(frame);
     av_free(video_dst_data[0]);
 
     return ret < 0;

doc/examples/extract_mvs.c

@@ -1,185 +0,0 @@
-/*
- * Copyright (c) 2012 Stefano Sabatini
- * Copyright (c) 2014 Clément Bœsch
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to deal
- * in the Software without restriction, including without limitation the rights
- * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- * copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
- * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
- * THE SOFTWARE.
- */
-
-#include <libavutil/motion_vector.h>
-#include <libavformat/avformat.h>
-
-static AVFormatContext *fmt_ctx = NULL;
-static AVCodecContext *video_dec_ctx = NULL;
-static AVStream *video_stream = NULL;
-static const char *src_filename = NULL;
-
-static int video_stream_idx = -1;
-static AVFrame *frame = NULL;
-static AVPacket pkt;
-static int video_frame_count = 0;
-
-static int decode_packet(int *got_frame, int cached)
-{
-    int decoded = pkt.size;
-
-    *got_frame = 0;
-
-    if (pkt.stream_index == video_stream_idx) {
-        int ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
-        if (ret < 0) {
-            fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret));
-            return ret;
-        }
-
-        if (*got_frame) {
-            int i;
-            AVFrameSideData *sd;
-
-            video_frame_count++;
-            sd = av_frame_get_side_data(frame, AV_FRAME_DATA_MOTION_VECTORS);
-            if (sd) {
-                const AVMotionVector *mvs = (const AVMotionVector *)sd->data;
-                for (i = 0; i < sd->size / sizeof(*mvs); i++) {
-                    const AVMotionVector *mv = &mvs[i];
-                    printf("%d,%2d,%2d,%2d,%4d,%4d,%4d,%4d,0x%"PRIx64"\n",
-                           video_frame_count, mv->source,
-                           mv->w, mv->h, mv->src_x, mv->src_y,
-                           mv->dst_x, mv->dst_y, mv->flags);
-                }
-            }
-        }
-    }
-
-    return decoded;
-}
-
-static int open_codec_context(int *stream_idx,
-                              AVFormatContext *fmt_ctx, enum AVMediaType type)
-{
-    int ret;
-    AVStream *st;
-    AVCodecContext *dec_ctx = NULL;
-    AVCodec *dec = NULL;
-    AVDictionary *opts = NULL;
-
-    ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
-    if (ret < 0) {
-        fprintf(stderr, "Could not find %s stream in input file '%s'\n",
-                av_get_media_type_string(type), src_filename);
-        return ret;
-    } else {
-        *stream_idx = ret;
-        st = fmt_ctx->streams[*stream_idx];
-
-        /* find decoder for the stream */
-        dec_ctx = st->codec;
-        dec = avcodec_find_decoder(dec_ctx->codec_id);
-        if (!dec) {
-            fprintf(stderr, "Failed to find %s codec\n",
-                    av_get_media_type_string(type));
-            return AVERROR(EINVAL);
-        }
-
-        /* Init the video decoder */
-        av_dict_set(&opts, "flags2", "+export_mvs", 0);
-        if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) {
-            fprintf(stderr, "Failed to open %s codec\n",
-                    av_get_media_type_string(type));
-            return ret;
-        }
-    }
-
-    return 0;
-}
-
-int main(int argc, char **argv)
-{
-    int ret = 0, got_frame;
-
-    if (argc != 2) {
-        fprintf(stderr, "Usage: %s <video>\n", argv[0]);
-        exit(1);
-    }
-    src_filename = argv[1];
-
-    av_register_all();
-
-    if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
-        fprintf(stderr, "Could not open source file %s\n", src_filename);
-        exit(1);
-    }
-
-    if (avformat_find_stream_info(fmt_ctx, NULL) < 0) {
-        fprintf(stderr, "Could not find stream information\n");
-        exit(1);
-    }
-
-    if (open_codec_context(&video_stream_idx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) {
-        video_stream = fmt_ctx->streams[video_stream_idx];
-        video_dec_ctx = video_stream->codec;
-    }
-
-    av_dump_format(fmt_ctx, 0, src_filename, 0);
-
-    if (!video_stream) {
-        fprintf(stderr, "Could not find video stream in the input, aborting\n");
-        ret = 1;
-        goto end;
-    }
-
-    frame = av_frame_alloc();
-    if (!frame) {
-        fprintf(stderr, "Could not allocate frame\n");
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-
-    printf("framenum,source,blockw,blockh,srcx,srcy,dstx,dsty,flags\n");
-
-    /* initialize packet, set data to NULL, let the demuxer fill it */
-    av_init_packet(&pkt);
-    pkt.data = NULL;
-    pkt.size = 0;
-
-    /* read frames from the file */
-    while (av_read_frame(fmt_ctx, &pkt) >= 0) {
-        AVPacket orig_pkt = pkt;
-        do {
-            ret = decode_packet(&got_frame, 0);
-            if (ret < 0)
-                break;
-            pkt.data += ret;
-            pkt.size -= ret;
-        } while (pkt.size > 0);
-        av_free_packet(&orig_pkt);
-    }
-
-    /* flush cached frames */
-    pkt.data = NULL;
-    pkt.size = 0;
-    do {
-        decode_packet(&got_frame, 1);
-    } while (got_frame);
-
-end:
-    avcodec_close(video_dec_ctx);
-    avformat_close_input(&fmt_ctx);
-    av_frame_free(&frame);
-    return ret < 0;
-}

doc/examples/filter_audio.c

@@ -1,365 +0,0 @@
-/*
- * copyright (c) 2013 Andrew Kelley
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file
- * libavfilter API usage example.
- *
- * @example filter_audio.c
- * This example will generate a sine wave audio,
- * pass it through a simple filter chain, and then compute the MD5 checksum of
- * the output data.
- *
- * The filter chain it uses is:
- * (input) -> abuffer -> volume -> aformat -> abuffersink -> (output)
- *
- * abuffer: This provides the endpoint where you can feed the decoded samples.
- * volume: In this example we hardcode it to 0.90.
- * aformat: This converts the samples to the samplefreq, channel layout,
- *          and sample format required by the audio device.
- * abuffersink: This provides the endpoint where you can read the samples after
- *              they have passed through the filter chain.
- */
-
-#include <inttypes.h>
-#include <math.h>
-#include <stdio.h>
-#include <stdlib.h>
-
-#include "libavutil/channel_layout.h"
-#include "libavutil/md5.h"
-#include "libavutil/mem.h"
-#include "libavutil/opt.h"
-#include "libavutil/samplefmt.h"
-
-#include "libavfilter/avfilter.h"
-#include "libavfilter/buffersink.h"
-#include "libavfilter/buffersrc.h"
-
-#define INPUT_SAMPLERATE     48000
-#define INPUT_FORMAT         AV_SAMPLE_FMT_FLTP
-#define INPUT_CHANNEL_LAYOUT AV_CH_LAYOUT_5POINT0
-
-#define VOLUME_VAL 0.90
-
-static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
-                             AVFilterContext **sink)
-{
-    AVFilterGraph *filter_graph;
-    AVFilterContext *abuffer_ctx;
-    AVFilter        *abuffer;
-    AVFilterContext *volume_ctx;
-    AVFilter        *volume;
-    AVFilterContext *aformat_ctx;
-    AVFilter        *aformat;
-    AVFilterContext *abuffersink_ctx;
-    AVFilter        *abuffersink;
-
-    AVDictionary *options_dict = NULL;
-    uint8_t options_str[1024];
-    uint8_t ch_layout[64];
-
-    int err;
-
-    /* Create a new filtergraph, which will contain all the filters. */
-    filter_graph = avfilter_graph_alloc();
-    if (!filter_graph) {
-        fprintf(stderr, "Unable to create filter graph.\n");
-        return AVERROR(ENOMEM);
-    }
-
-    /* Create the abuffer filter;
-     * it will be used for feeding the data into the graph. */
-    abuffer = avfilter_get_by_name("abuffer");
-    if (!abuffer) {
-        fprintf(stderr, "Could not find the abuffer filter.\n");
-        return AVERROR_FILTER_NOT_FOUND;
-    }
-
-    abuffer_ctx = avfilter_graph_alloc_filter(filter_graph, abuffer, "src");
-    if (!abuffer_ctx) {
-        fprintf(stderr, "Could not allocate the abuffer instance.\n");
-        return AVERROR(ENOMEM);
-    }
-
-    /* Set the filter options through the AVOptions API. */
-    av_get_channel_layout_string(ch_layout, sizeof(ch_layout), 0, INPUT_CHANNEL_LAYOUT);
-    av_opt_set    (abuffer_ctx, "channel_layout", ch_layout,                            AV_OPT_SEARCH_CHILDREN);
-    av_opt_set    (abuffer_ctx, "sample_fmt",     av_get_sample_fmt_name(INPUT_FORMAT), AV_OPT_SEARCH_CHILDREN);
-    av_opt_set_q  (abuffer_ctx, "time_base",      (AVRational){ 1, INPUT_SAMPLERATE },  AV_OPT_SEARCH_CHILDREN);
-    av_opt_set_int(abuffer_ctx, "sample_rate",    INPUT_SAMPLERATE,                     AV_OPT_SEARCH_CHILDREN);
-
-    /* Now initialize the filter; we pass NULL options, since we have already
-     * set all the options above. */
-    err = avfilter_init_str(abuffer_ctx, NULL);
-    if (err < 0) {
-        fprintf(stderr, "Could not initialize the abuffer filter.\n");
-        return err;
-    }
-
-    /* Create volume filter. */
-    volume = avfilter_get_by_name("volume");
-    if (!volume) {
-        fprintf(stderr, "Could not find the volume filter.\n");
-        return AVERROR_FILTER_NOT_FOUND;
-    }
-
-    volume_ctx = avfilter_graph_alloc_filter(filter_graph, volume, "volume");
-    if (!volume_ctx) {
-        fprintf(stderr, "Could not allocate the volume instance.\n");
-        return AVERROR(ENOMEM);
-    }
-
-    /* A different way of passing the options is as key/value pairs in a
-     * dictionary. */
-    av_dict_set(&options_dict, "volume", AV_STRINGIFY(VOLUME_VAL), 0);
-    err = avfilter_init_dict(volume_ctx, &options_dict);
-    av_dict_free(&options_dict);
-    if (err < 0) {
-        fprintf(stderr, "Could not initialize the volume filter.\n");
-        return err;
-    }
-
-    /* Create the aformat filter;
-     * it ensures that the output is of the format we want. */
-    aformat = avfilter_get_by_name("aformat");
-    if (!aformat) {
-        fprintf(stderr, "Could not find the aformat filter.\n");
-        return AVERROR_FILTER_NOT_FOUND;
-    }
-
-    aformat_ctx = avfilter_graph_alloc_filter(filter_graph, aformat, "aformat");
-    if (!aformat_ctx) {
-        fprintf(stderr, "Could not allocate the aformat instance.\n");
-        return AVERROR(ENOMEM);
-    }
-
-    /* A third way of passing the options is in a string of the form
-     * key1=value1:key2=value2.... */
-    snprintf(options_str, sizeof(options_str),
-             "sample_fmts=%s:sample_rates=%d:channel_layouts=0x%"PRIx64,
-             av_get_sample_fmt_name(AV_SAMPLE_FMT_S16), 44100,
-             (uint64_t)AV_CH_LAYOUT_STEREO);
-    err = avfilter_init_str(aformat_ctx, options_str);
-    if (err < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Could not initialize the aformat filter.\n");
-        return err;
-    }
-
-    /* Finally create the abuffersink filter;
-     * it will be used to get the filtered data out of the graph. */
-    abuffersink = avfilter_get_by_name("abuffersink");
-    if (!abuffersink) {
-        fprintf(stderr, "Could not find the abuffersink filter.\n");
-        return AVERROR_FILTER_NOT_FOUND;
-    }
-
-    abuffersink_ctx = avfilter_graph_alloc_filter(filter_graph, abuffersink, "sink");
-    if (!abuffersink_ctx) {
-        fprintf(stderr, "Could not allocate the abuffersink instance.\n");
-        return AVERROR(ENOMEM);
-    }
-
-    /* This filter takes no options. */
-    err = avfilter_init_str(abuffersink_ctx, NULL);
-    if (err < 0) {
-        fprintf(stderr, "Could not initialize the abuffersink instance.\n");
-        return err;
-    }
-
-    /* Connect the filters;
-     * in this simple case the filters just form a linear chain. */
-    err = avfilter_link(abuffer_ctx, 0, volume_ctx, 0);
-    if (err >= 0)
-        err = avfilter_link(volume_ctx, 0, aformat_ctx, 0);
-    if (err >= 0)
-        err = avfilter_link(aformat_ctx, 0, abuffersink_ctx, 0);
-    if (err < 0) {
-        fprintf(stderr, "Error connecting filters\n");
-        return err;
-    }
-
-    /* Configure the graph. */
-    err = avfilter_graph_config(filter_graph, NULL);
-    if (err < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Error configuring the filter graph\n");
-        return err;
-    }
-
-    *graph = filter_graph;
-    *src   = abuffer_ctx;
-    *sink  = abuffersink_ctx;
-
-    return 0;
-}
-
-/* Do something useful with the filtered data: this simple
- * example just prints the MD5 checksum of each plane to stdout. */
-static int process_output(struct AVMD5 *md5, AVFrame *frame)
-{
-    int planar     = av_sample_fmt_is_planar(frame->format);
-    int channels   = av_get_channel_layout_nb_channels(frame->channel_layout);
-    int planes     = planar ? channels : 1;
-    int bps        = av_get_bytes_per_sample(frame->format);
-    int plane_size = bps * frame->nb_samples * (planar ? 1 : channels);
-    int i, j;
-
-    for (i = 0; i < planes; i++) {
-        uint8_t checksum[16];
-
-        av_md5_init(md5);
-        av_md5_sum(checksum, frame->extended_data[i], plane_size);
-
-        fprintf(stdout, "plane %d: 0x", i);
-        for (j = 0; j < sizeof(checksum); j++)
-            fprintf(stdout, "%02X", checksum[j]);
-        fprintf(stdout, "\n");
-    }
-    fprintf(stdout, "\n");
-
-    return 0;
-}
-
-/* Construct a frame of audio data to be filtered;
- * this simple example just synthesizes a sine wave. */
-static int get_input(AVFrame *frame, int frame_num)
-{
-    int err, i, j;
-
-#define FRAME_SIZE 1024
-
-    /* Set up the frame properties and allocate the buffer for the data. */
-    frame->sample_rate    = INPUT_SAMPLERATE;
-    frame->format         = INPUT_FORMAT;
-    frame->channel_layout = INPUT_CHANNEL_LAYOUT;
-    frame->nb_samples     = FRAME_SIZE;
-    frame->pts            = frame_num * FRAME_SIZE;
-
-    err = av_frame_get_buffer(frame, 0);
-    if (err < 0)
-        return err;
-
-    /* Fill the data for each channel. */
-    for (i = 0; i < 5; i++) {
-        float *data = (float*)frame->extended_data[i];
-
-        for (j = 0; j < frame->nb_samples; j++)
-            data[j] = sin(2 * M_PI * (frame_num + j) * (i + 1) / FRAME_SIZE);
-    }
-
-    return 0;
-}
-
-int main(int argc, char *argv[])
-{
-    struct AVMD5 *md5;
-    AVFilterGraph *graph;
-    AVFilterContext *src, *sink;
-    AVFrame *frame;
-    uint8_t errstr[1024];
-    float duration;
-    int err, nb_frames, i;
-
-    if (argc < 2) {
-        fprintf(stderr, "Usage: %s <duration>\n", argv[0]);
-        return 1;
-    }
-
-    duration  = atof(argv[1]);
-    nb_frames = duration * INPUT_SAMPLERATE / FRAME_SIZE;
-    if (nb_frames <= 0) {
-        fprintf(stderr, "Invalid duration: %s\n", argv[1]);
-        return 1;
-    }
-
-    avfilter_register_all();
-
-    /* Allocate the frame we will be using to store the data. */
-    frame  = av_frame_alloc();
-    if (!frame) {
-        fprintf(stderr, "Error allocating the frame\n");
-        return 1;
-    }
-
-    md5 = av_md5_alloc();
-    if (!md5) {
-        fprintf(stderr, "Error allocating the MD5 context\n");
-        return 1;
-    }
-
-    /* Set up the filtergraph. */
-    err = init_filter_graph(&graph, &src, &sink);
-    if (err < 0) {
-        fprintf(stderr, "Unable to init filter graph:");
-        goto fail;
-    }
-
-    /* the main filtering loop */
-    for (i = 0; i < nb_frames; i++) {
-        /* get an input frame to be filtered */
-        err = get_input(frame, i);
-        if (err < 0) {
-            fprintf(stderr, "Error generating input frame:");
-            goto fail;
-        }
-
-        /* Send the frame to the input of the filtergraph. */
-        err = av_buffersrc_add_frame(src, frame);
-        if (err < 0) {
-            av_frame_unref(frame);
-            fprintf(stderr, "Error submitting the frame to the filtergraph:");
-            goto fail;
-        }
-
-        /* Get all the filtered output that is available. */
-        while ((err = av_buffersink_get_frame(sink, frame)) >= 0) {
-            /* now do something with our filtered frame */
-            err = process_output(md5, frame);
-            if (err < 0) {
-                fprintf(stderr, "Error processing the filtered frame:");
-                goto fail;
-            }
-            av_frame_unref(frame);
-        }
-
-        if (err == AVERROR(EAGAIN)) {
-            /* Need to feed more frames in. */
-            continue;
-        } else if (err == AVERROR_EOF) {
-            /* Nothing more to do, finish. */
-            break;
-        } else if (err < 0) {
-            /* An error occurred. */
-            fprintf(stderr, "Error filtering the data:");
-            goto fail;
-        }
-    }
-
-    avfilter_graph_free(&graph);
-    av_frame_free(&frame);
-    av_freep(&md5);
-
-    return 0;
-
-fail:
-    av_strerror(err, errstr, sizeof(errstr));
-    fprintf(stderr, "%s\n", errstr);
-    return 1;
-}

doc/examples/filtering_audio.c

@@ -25,7 +25,7 @@
 /**
  * @file
  * API example for audio decoding and filtering
- * @example filtering_audio.c
+ * @example doc/examples/filtering_audio.c
  */
 
 #include <unistd.h>
@@ -85,7 +85,7 @@ static int open_input_file(const char *filename)
 static int init_filters(const char *filters_descr)
 {
     char args[512];
-    int ret = 0;
+    int ret;
     AVFilter *abuffersrc  = avfilter_get_by_name("abuffer");
     AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
     AVFilterInOut *outputs = avfilter_inout_alloc();
@@ -97,10 +97,6 @@ static int init_filters(const char *filters_descr)
     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)
@@ -113,7 +109,7 @@ static int init_filters(const char *filters_descr)
                                        args, NULL, filter_graph);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
-        goto end;
+        return ret;
     }
 
     /* buffer audio sink: to terminate the filter chain. */
@@ -121,28 +117,28 @@ static int init_filters(const char *filters_descr)
                                        NULL, NULL, filter_graph);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
-        goto end;
+        return ret;
     }
 
     ret = av_opt_set_int_list(buffersink_ctx, "sample_fmts", out_sample_fmts, -1,
                               AV_OPT_SEARCH_CHILDREN);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
-        goto end;
+        return ret;
     }
 
     ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts", out_channel_layouts, -1,
                               AV_OPT_SEARCH_CHILDREN);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
-        goto end;
+        return ret;
     }
 
     ret = av_opt_set_int_list(buffersink_ctx, "sample_rates", out_sample_rates, -1,
                               AV_OPT_SEARCH_CHILDREN);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
-        goto end;
+        return ret;
     }
 
     /* Endpoints for the filter graph. */
@@ -157,11 +153,11 @@ static int init_filters(const char *filters_descr)
     inputs->next       = NULL;
 
     if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
-                                        &inputs, &outputs, NULL)) < 0)
-        goto end;
+                                    &inputs, &outputs, NULL)) < 0)
+        return ret;
 
     if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
-        goto end;
+        return ret;
 
     /* Print summary of the sink buffer
      * Note: args buffer is reused to store channel layout string */
@@ -172,11 +168,7 @@ static int init_filters(const char *filters_descr)
            (char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
            args);
 
-end:
-    avfilter_inout_free(&inputs);
-    avfilter_inout_free(&outputs);
-
-    return ret;
+    return 0;
 }
 
 static void print_frame(const AVFrame *frame)
@@ -196,7 +188,7 @@ static void print_frame(const AVFrame *frame)
 int main(int argc, char **argv)
 {
     int ret;
-    AVPacket packet0, packet;
+    AVPacket packet;
     AVFrame *frame = av_frame_alloc();
     AVFrame *filt_frame = av_frame_alloc();
     int got_frame;
@@ -210,6 +202,7 @@ int main(int argc, char **argv)
         exit(1);
     }
 
+    avcodec_register_all();
     av_register_all();
     avfilter_register_all();
 
@@ -219,24 +212,18 @@ int main(int argc, char **argv)
         goto end;
 
     /* read all packets */
-    packet0.data = NULL;
-    packet.data = NULL;
     while (1) {
-        if (!packet0.data) {
-            if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
-                break;
-            packet0 = packet;
-        }
+        if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
+            break;
 
         if (packet.stream_index == audio_stream_index) {
+            avcodec_get_frame_defaults(frame);
             got_frame = 0;
             ret = avcodec_decode_audio4(dec_ctx, frame, &got_frame, &packet);
             if (ret < 0) {
                 av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n");
                 continue;
             }
-            packet.size -= ret;
-            packet.data += ret;
 
             if (got_frame) {
                 /* push the audio data from decoded frame into the filtergraph */
@@ -248,31 +235,29 @@ int main(int argc, char **argv)
                 /* pull filtered audio from the filtergraph */
                 while (1) {
                     ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
-                    if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
+                    if(ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
                         break;
-                    if (ret < 0)
+                    if(ret < 0)
                         goto end;
                     print_frame(filt_frame);
                     av_frame_unref(filt_frame);
                 }
             }
-
-            if (packet.size <= 0)
-                av_free_packet(&packet0);
-        } else {
-            /* discard non-wanted packets */
-            av_free_packet(&packet0);
         }
+        av_free_packet(&packet);
     }
 end:
     avfilter_graph_free(&filter_graph);
-    avcodec_close(dec_ctx);
+    if (dec_ctx)
+        avcodec_close(dec_ctx);
     avformat_close_input(&fmt_ctx);
     av_frame_free(&frame);
     av_frame_free(&filt_frame);
 
     if (ret < 0 && ret != AVERROR_EOF) {
-        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
+        char buf[1024];
+        av_strerror(ret, buf, sizeof(buf));
+        fprintf(stderr, "Error occurred: %s\n", buf);
         exit(1);
     }
 

doc/examples/filtering_video.c

@@ -24,7 +24,7 @@
 /**
  * @file
  * API example for decoding and filtering
- * @example filtering_video.c
+ * @example doc/examples/filtering_video.c
  */
 
 #define _XOPEN_SOURCE 600 /* for usleep */
@@ -36,7 +36,6 @@
 #include <libavfilter/avcodec.h>
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
-#include <libavutil/opt.h>
 
 const char *filter_descr = "scale=78:24";
 
@@ -71,7 +70,6 @@ static int open_input_file(const char *filename)
     }
     video_stream_index = ret;
     dec_ctx = fmt_ctx->streams[video_stream_index]->codec;
-    av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
 
     /* init the video decoder */
     if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
@@ -85,18 +83,15 @@ static int open_input_file(const char *filename)
 static int init_filters(const char *filters_descr)
 {
     char args[512];
-    int ret = 0;
+    int ret;
     AVFilter *buffersrc  = avfilter_get_by_name("buffer");
     AVFilter *buffersink = avfilter_get_by_name("buffersink");
     AVFilterInOut *outputs = avfilter_inout_alloc();
     AVFilterInOut *inputs  = avfilter_inout_alloc();
     enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE };
+    AVBufferSinkParams *buffersink_params;
 
     filter_graph = avfilter_graph_alloc();
-    if (!outputs || !inputs || !filter_graph) {
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
 
     /* buffer video source: the decoded frames from the decoder will be inserted here. */
     snprintf(args, sizeof(args),
@@ -109,22 +104,18 @@ static int init_filters(const char *filters_descr)
                                        args, NULL, filter_graph);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n");
-        goto end;
+        return ret;
     }
 
     /* buffer video sink: to terminate the filter chain. */
+    buffersink_params = av_buffersink_params_alloc();
+    buffersink_params->pixel_fmts = pix_fmts;
     ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
-                                       NULL, NULL, filter_graph);
+                                       NULL, buffersink_params, filter_graph);
+    av_free(buffersink_params);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
-        goto end;
-    }
-
-    ret = av_opt_set_int_list(buffersink_ctx, "pix_fmts", pix_fmts,
-                              AV_PIX_FMT_NONE, AV_OPT_SEARCH_CHILDREN);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n");
-        goto end;
+        return ret;
     }
 
     /* Endpoints for the filter graph. */
@@ -140,16 +131,11 @@ static int init_filters(const char *filters_descr)
 
     if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
                                     &inputs, &outputs, NULL)) < 0)
-        goto end;
+        return ret;
 
     if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
-        goto end;
-
-end:
-    avfilter_inout_free(&inputs);
-    avfilter_inout_free(&outputs);
-
-    return ret;
+        return ret;
+    return 0;
 }
 
 static void display_frame(const AVFrame *frame, AVRational time_base)
@@ -200,6 +186,7 @@ int main(int argc, char **argv)
         exit(1);
     }
 
+    avcodec_register_all();
     av_register_all();
     avfilter_register_all();
 
@@ -214,6 +201,7 @@ int main(int argc, char **argv)
             break;
 
         if (packet.stream_index == video_stream_index) {
+            avcodec_get_frame_defaults(frame);
             got_frame = 0;
             ret = avcodec_decode_video2(dec_ctx, frame, &got_frame, &packet);
             if (ret < 0) {
@@ -240,20 +228,22 @@ int main(int argc, char **argv)
                     display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base);
                     av_frame_unref(filt_frame);
                 }
-                av_frame_unref(frame);
             }
         }
         av_free_packet(&packet);
     }
 end:
     avfilter_graph_free(&filter_graph);
-    avcodec_close(dec_ctx);
+    if (dec_ctx)
+        avcodec_close(dec_ctx);
     avformat_close_input(&fmt_ctx);
     av_frame_free(&frame);
     av_frame_free(&filt_frame);
 
     if (ret < 0 && ret != AVERROR_EOF) {
-        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
+        char buf[1024];
+        av_strerror(ret, buf, sizeof(buf));
+        fprintf(stderr, "Error occurred: %s\n", buf);
         exit(1);
     }
 

doc/examples/metadata.c

@@ -23,7 +23,7 @@
 /**
  * @file
  * Shows how the metadata API can be used in application programs.
- * @example metadata.c
+ * @example doc/examples/metadata.c
  */
 
 #include <stdio.h>

doc/examples/muxing.c

@@ -24,9 +24,9 @@
  * @file
  * libavformat API example.
  *
- * Output a media file in any supported libavformat format. The default
- * codecs are used.
- * @example muxing.c
+ * Output a media file in any supported libavformat format.
+ * The default codecs are used.
+ * @example doc/examples/muxing.c
  */
 
 #include <stdlib.h>
@@ -34,67 +34,26 @@
 #include <string.h>
 #include <math.h>
 
-#include <libavutil/avassert.h>
-#include <libavutil/channel_layout.h>
 #include <libavutil/opt.h>
 #include <libavutil/mathematics.h>
-#include <libavutil/timestamp.h>
 #include <libavformat/avformat.h>
 #include <libswscale/swscale.h>
 #include <libswresample/swresample.h>
 
-#define STREAM_DURATION   10.0
+/* 5 seconds stream duration */
+#define STREAM_DURATION   200.0
 #define STREAM_FRAME_RATE 25 /* 25 images/s */
+#define STREAM_NB_FRAMES  ((int)(STREAM_DURATION * STREAM_FRAME_RATE))
 #define STREAM_PIX_FMT    AV_PIX_FMT_YUV420P /* default pix_fmt */
 
-#define SCALE_FLAGS SWS_BICUBIC
-
-// a wrapper around a single output AVStream
-typedef struct OutputStream {
-    AVStream *st;
-
-    /* pts of the next frame that will be generated */
-    int64_t next_pts;
-    int samples_count;
-
-    AVFrame *frame;
-    AVFrame *tmp_frame;
-
-    float t, tincr, tincr2;
-
-    struct SwsContext *sws_ctx;
-    struct SwrContext *swr_ctx;
-} OutputStream;
-
-static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt)
-{
-    AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base;
-
-    printf("pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n",
-           av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base),
-           av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base),
-           av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base),
-           pkt->stream_index);
-}
-
-static int write_frame(AVFormatContext *fmt_ctx, const AVRational *time_base, AVStream *st, AVPacket *pkt)
-{
-    /* rescale output packet timestamp values from codec to stream timebase */
-    av_packet_rescale_ts(pkt, *time_base, st->time_base);
-    pkt->stream_index = st->index;
-
-    /* Write the compressed frame to the media file. */
-    log_packet(fmt_ctx, pkt);
-    return av_interleaved_write_frame(fmt_ctx, pkt);
-}
+static int sws_flags = SWS_BICUBIC;
 
 /* Add an output stream. */
-static void add_stream(OutputStream *ost, AVFormatContext *oc,
-                       AVCodec **codec,
-                       enum AVCodecID codec_id)
+static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
+                            enum AVCodecID codec_id)
 {
     AVCodecContext *c;
-    int i;
+    AVStream *st;
 
     /* find the encoder */
     *codec = avcodec_find_encoder(codec_id);
@@ -104,38 +63,20 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
         exit(1);
     }
 
-    ost->st = avformat_new_stream(oc, *codec);
-    if (!ost->st) {
+    st = avformat_new_stream(oc, *codec);
+    if (!st) {
         fprintf(stderr, "Could not allocate stream\n");
         exit(1);
     }
-    ost->st->id = oc->nb_streams-1;
-    c = ost->st->codec;
+    st->id = oc->nb_streams-1;
+    c = st->codec;
 
     switch ((*codec)->type) {
     case AVMEDIA_TYPE_AUDIO:
-        c->sample_fmt  = (*codec)->sample_fmts ?
-            (*codec)->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
+        c->sample_fmt  = AV_SAMPLE_FMT_FLTP;
         c->bit_rate    = 64000;
         c->sample_rate = 44100;
-        if ((*codec)->supported_samplerates) {
-            c->sample_rate = (*codec)->supported_samplerates[0];
-            for (i = 0; (*codec)->supported_samplerates[i]; i++) {
-                if ((*codec)->supported_samplerates[i] == 44100)
-                    c->sample_rate = 44100;
-            }
-        }
-        c->channels        = av_get_channel_layout_nb_channels(c->channel_layout);
-        c->channel_layout = AV_CH_LAYOUT_STEREO;
-        if ((*codec)->channel_layouts) {
-            c->channel_layout = (*codec)->channel_layouts[0];
-            for (i = 0; (*codec)->channel_layouts[i]; i++) {
-                if ((*codec)->channel_layouts[i] == AV_CH_LAYOUT_STEREO)
-                    c->channel_layout = AV_CH_LAYOUT_STEREO;
-            }
-        }
-        c->channels        = av_get_channel_layout_nb_channels(c->channel_layout);
-        ost->st->time_base = (AVRational){ 1, c->sample_rate };
+        c->channels    = 2;
         break;
 
     case AVMEDIA_TYPE_VIDEO:
@@ -149,9 +90,8 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
          * of which frame timestamps are represented. For fixed-fps content,
          * timebase should be 1/framerate and timestamp increments should be
          * identical to 1. */
-        ost->st->time_base = (AVRational){ 1, STREAM_FRAME_RATE };
-        c->time_base       = ost->st->time_base;
-
+        c->time_base.den = STREAM_FRAME_RATE;
+        c->time_base.num = 1;
         c->gop_size      = 12; /* emit one intra frame every twelve frames at most */
         c->pix_fmt       = STREAM_PIX_FMT;
         if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) {
@@ -173,262 +113,237 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
     /* 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 AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt,
-                                  uint64_t channel_layout,
-                                  int sample_rate, int nb_samples)
-{
-    AVFrame *frame = av_frame_alloc();
-    int ret;
+static float t, tincr, tincr2;
 
-    if (!frame) {
-        fprintf(stderr, "Error allocating an audio frame\n");
-        exit(1);
-    }
+static uint8_t **src_samples_data;
+static int       src_samples_linesize;
+static int       src_nb_samples;
 
-    frame->format = sample_fmt;
-    frame->channel_layout = channel_layout;
-    frame->sample_rate = sample_rate;
-    frame->nb_samples = nb_samples;
+static int max_dst_nb_samples;
+uint8_t **dst_samples_data;
+int       dst_samples_linesize;
+int       dst_samples_size;
 
-    if (nb_samples) {
-        ret = av_frame_get_buffer(frame, 0);
-        if (ret < 0) {
-            fprintf(stderr, "Error allocating an audio buffer\n");
-            exit(1);
-        }
-    }
+struct SwrContext *swr_ctx = NULL;
 
-    return frame;
-}
-
-static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
+static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
 {
     AVCodecContext *c;
-    int nb_samples;
     int ret;
-    AVDictionary *opt = NULL;
 
-    c = ost->st->codec;
+    c = st->codec;
 
     /* open it */
-    av_dict_copy(&opt, opt_arg, 0);
-    ret = avcodec_open2(c, codec, &opt);
-    av_dict_free(&opt);
+    ret = avcodec_open2(c, codec, NULL);
     if (ret < 0) {
         fprintf(stderr, "Could not open audio codec: %s\n", av_err2str(ret));
         exit(1);
     }
 
     /* init signal generator */
-    ost->t     = 0;
-    ost->tincr = 2 * M_PI * 110.0 / c->sample_rate;
+    t     = 0;
+    tincr = 2 * M_PI * 110.0 / c->sample_rate;
     /* increment frequency by 110 Hz per second */
-    ost->tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
+    tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
 
-    if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
-        nb_samples = 10000;
-    else
-        nb_samples = c->frame_size;
+    src_nb_samples = c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE ?
+        10000 : c->frame_size;
 
-    ost->frame     = alloc_audio_frame(c->sample_fmt, c->channel_layout,
-                                       c->sample_rate, nb_samples);
-    ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, c->channel_layout,
-                                       c->sample_rate, nb_samples);
+    ret = av_samples_alloc_array_and_samples(&src_samples_data, &src_samples_linesize, c->channels,
+                                             src_nb_samples, c->sample_fmt, 0);
+    if (ret < 0) {
+        fprintf(stderr, "Could not allocate source samples\n");
+        exit(1);
+    }
 
     /* create resampler context */
-        ost->swr_ctx = swr_alloc();
-        if (!ost->swr_ctx) {
+    if (c->sample_fmt != AV_SAMPLE_FMT_S16) {
+        swr_ctx = swr_alloc();
+        if (!swr_ctx) {
             fprintf(stderr, "Could not allocate resampler context\n");
             exit(1);
         }
 
         /* set options */
-        av_opt_set_int       (ost->swr_ctx, "in_channel_count",   c->channels,       0);
-        av_opt_set_int       (ost->swr_ctx, "in_sample_rate",     c->sample_rate,    0);
-        av_opt_set_sample_fmt(ost->swr_ctx, "in_sample_fmt",      AV_SAMPLE_FMT_S16, 0);
-        av_opt_set_int       (ost->swr_ctx, "out_channel_count",  c->channels,       0);
-        av_opt_set_int       (ost->swr_ctx, "out_sample_rate",    c->sample_rate,    0);
-        av_opt_set_sample_fmt(ost->swr_ctx, "out_sample_fmt",     c->sample_fmt,     0);
+        av_opt_set_int       (swr_ctx, "in_channel_count",   c->channels,       0);
+        av_opt_set_int       (swr_ctx, "in_sample_rate",     c->sample_rate,    0);
+        av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt",      AV_SAMPLE_FMT_S16, 0);
+        av_opt_set_int       (swr_ctx, "out_channel_count",  c->channels,       0);
+        av_opt_set_int       (swr_ctx, "out_sample_rate",    c->sample_rate,    0);
+        av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt",     c->sample_fmt,     0);
 
         /* initialize the resampling context */
-        if ((ret = swr_init(ost->swr_ctx)) < 0) {
+        if ((ret = swr_init(swr_ctx)) < 0) {
             fprintf(stderr, "Failed to initialize the resampling context\n");
             exit(1);
         }
+    }
+
+    /* compute the number of converted samples: buffering is avoided
+     * ensuring that the output buffer will contain at least all the
+     * converted input samples */
+    max_dst_nb_samples = src_nb_samples;
+    ret = av_samples_alloc_array_and_samples(&dst_samples_data, &dst_samples_linesize, c->channels,
+                                             max_dst_nb_samples, c->sample_fmt, 0);
+    if (ret < 0) {
+        fprintf(stderr, "Could not allocate destination samples\n");
+        exit(1);
+    }
+    dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, max_dst_nb_samples,
+                                                  c->sample_fmt, 0);
 }
 
 /* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
  * 'nb_channels' channels. */
-static AVFrame *get_audio_frame(OutputStream *ost)
+static void get_audio_frame(int16_t *samples, int frame_size, int nb_channels)
 {
-    AVFrame *frame = ost->tmp_frame;
     int j, i, v;
-    int16_t *q = (int16_t*)frame->data[0];
+    int16_t *q;
 
-    /* check if we want to generate more frames */
-    if (av_compare_ts(ost->next_pts, ost->st->codec->time_base,
-                      STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
-        return NULL;
-
-    for (j = 0; j <frame->nb_samples; j++) {
-        v = (int)(sin(ost->t) * 10000);
-        for (i = 0; i < ost->st->codec->channels; i++)
+    q = samples;
+    for (j = 0; j < frame_size; j++) {
+        v = (int)(sin(t) * 10000);
+        for (i = 0; i < nb_channels; i++)
             *q++ = v;
-        ost->t     += ost->tincr;
-        ost->tincr += ost->tincr2;
+        t     += tincr;
+        tincr += tincr2;
     }
-
-    frame->pts = ost->next_pts;
-    ost->next_pts  += frame->nb_samples;
-
-    return frame;
 }
 
-/*
- * encode one audio frame and send it to the muxer
- * return 1 when encoding is finished, 0 otherwise
- */
-static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
+static void write_audio_frame(AVFormatContext *oc, AVStream *st)
 {
     AVCodecContext *c;
     AVPacket pkt = { 0 }; // data and size must be 0;
-    AVFrame *frame;
-    int ret;
-    int got_packet;
-    int dst_nb_samples;
+    AVFrame *frame = avcodec_alloc_frame();
+    int got_packet, ret, dst_nb_samples;
 
     av_init_packet(&pkt);
-    c = ost->st->codec;
+    c = st->codec;
 
-    frame = get_audio_frame(ost);
+    get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels);
 
-    if (frame) {
-        /* convert samples from native format to destination codec format, using the resampler */
-            /* compute destination number of samples */
-            dst_nb_samples = av_rescale_rnd(swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples,
-                                            c->sample_rate, c->sample_rate, AV_ROUND_UP);
-            av_assert0(dst_nb_samples == frame->nb_samples);
-
-        /* when we pass a frame to the encoder, it may keep a reference to it
-         * internally;
-         * make sure we do not overwrite it here
-         */
-        ret = av_frame_make_writable(ost->frame);
-        if (ret < 0)
-            exit(1);
-
-            /* convert to destination format */
-            ret = swr_convert(ost->swr_ctx,
-                              ost->frame->data, dst_nb_samples,
-                              (const uint8_t **)frame->data, frame->nb_samples);
-            if (ret < 0) {
-                fprintf(stderr, "Error while converting\n");
+    /* convert samples from native format to destination codec format, using the resampler */
+    if (swr_ctx) {
+        /* compute destination number of samples */
+        dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + src_nb_samples,
+                                        c->sample_rate, c->sample_rate, AV_ROUND_UP);
+        if (dst_nb_samples > max_dst_nb_samples) {
+            av_free(dst_samples_data[0]);
+            ret = av_samples_alloc(dst_samples_data, &dst_samples_linesize, c->channels,
+                                   dst_nb_samples, c->sample_fmt, 0);
+            if (ret < 0)
                 exit(1);
-            }
-            frame = ost->frame;
+            max_dst_nb_samples = dst_nb_samples;
+            dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, dst_nb_samples,
+                                                          c->sample_fmt, 0);
+        }
 
-        frame->pts = av_rescale_q(ost->samples_count, (AVRational){1, c->sample_rate}, c->time_base);
-        ost->samples_count += dst_nb_samples;
+        /* convert to destination format */
+        ret = swr_convert(swr_ctx,
+                          dst_samples_data, dst_nb_samples,
+                          (const uint8_t **)src_samples_data, src_nb_samples);
+        if (ret < 0) {
+            fprintf(stderr, "Error while converting\n");
+            exit(1);
+        }
+    } else {
+        dst_samples_data[0] = src_samples_data[0];
+        dst_nb_samples = src_nb_samples;
     }
 
+    frame->nb_samples = dst_nb_samples;
+    avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
+                             dst_samples_data[0], dst_samples_size, 0);
+
     ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
     if (ret < 0) {
         fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
         exit(1);
     }
 
-    if (got_packet) {
-        ret = write_frame(oc, &c->time_base, ost->st, &pkt);
-        if (ret < 0) {
-            fprintf(stderr, "Error while writing audio frame: %s\n",
-                    av_err2str(ret));
-            exit(1);
-        }
-    }
+    if (!got_packet)
+        return;
 
-    return (frame || got_packet) ? 0 : 1;
+    pkt.stream_index = st->index;
+
+    /* Write the compressed frame to the media file. */
+    ret = av_interleaved_write_frame(oc, &pkt);
+    if (ret != 0) {
+        fprintf(stderr, "Error while writing audio frame: %s\n",
+                av_err2str(ret));
+        exit(1);
+    }
+    avcodec_free_frame(&frame);
+}
+
+static void close_audio(AVFormatContext *oc, AVStream *st)
+{
+    avcodec_close(st->codec);
+    av_free(src_samples_data[0]);
+    av_free(dst_samples_data[0]);
 }
 
 /**************************************************************/
 /* video output */
 
-static AVFrame *alloc_picture(enum AVPixelFormat pix_fmt, int width, int height)
-{
-    AVFrame *picture;
-    int ret;
+static AVFrame *frame;
+static AVPicture src_picture, dst_picture;
+static int frame_count;
 
-    picture = av_frame_alloc();
-    if (!picture)
-        return NULL;
-
-    picture->format = pix_fmt;
-    picture->width  = width;
-    picture->height = height;
-
-    /* allocate the buffers for the frame data */
-    ret = av_frame_get_buffer(picture, 32);
-    if (ret < 0) {
-        fprintf(stderr, "Could not allocate frame data.\n");
-        exit(1);
-    }
-
-    return picture;
-}
-
-static void open_video(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
+static void open_video(AVFormatContext *oc, AVCodec *codec, AVStream *st)
 {
     int ret;
-    AVCodecContext *c = ost->st->codec;
-    AVDictionary *opt = NULL;
-
-    av_dict_copy(&opt, opt_arg, 0);
+    AVCodecContext *c = st->codec;
 
     /* open the codec */
-    ret = avcodec_open2(c, codec, &opt);
-    av_dict_free(&opt);
+    ret = avcodec_open2(c, codec, NULL);
     if (ret < 0) {
         fprintf(stderr, "Could not open video codec: %s\n", av_err2str(ret));
         exit(1);
     }
 
     /* allocate and init a re-usable frame */
-    ost->frame = alloc_picture(c->pix_fmt, c->width, c->height);
-    if (!ost->frame) {
+    frame = avcodec_alloc_frame();
+    if (!frame) {
         fprintf(stderr, "Could not allocate video frame\n");
         exit(1);
     }
 
+    /* Allocate the encoded raw picture. */
+    ret = avpicture_alloc(&dst_picture, c->pix_fmt, c->width, c->height);
+    if (ret < 0) {
+        fprintf(stderr, "Could not allocate picture: %s\n", av_err2str(ret));
+        exit(1);
+    }
+
     /* If the output format is not YUV420P, then a temporary YUV420P
      * picture is needed too. It is then converted to the required
      * output format. */
-    ost->tmp_frame = NULL;
     if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
-        ost->tmp_frame = alloc_picture(AV_PIX_FMT_YUV420P, c->width, c->height);
-        if (!ost->tmp_frame) {
-            fprintf(stderr, "Could not allocate temporary picture\n");
+        ret = avpicture_alloc(&src_picture, AV_PIX_FMT_YUV420P, c->width, c->height);
+        if (ret < 0) {
+            fprintf(stderr, "Could not allocate temporary picture: %s\n",
+                    av_err2str(ret));
             exit(1);
         }
     }
+
+    /* copy data and linesize picture pointers to frame */
+    *((AVPicture *)frame) = dst_picture;
 }
 
 /* Prepare a dummy image. */
-static void fill_yuv_image(AVFrame *pict, int frame_index,
+static void fill_yuv_image(AVPicture *pict, int frame_index,
                            int width, int height)
 {
-    int x, y, i, ret;
-
-    /* when we pass a frame to the encoder, it may keep a reference to it
-     * internally;
-     * make sure we do not overwrite it here
-     */
-    ret = av_frame_make_writable(pict);
-    if (ret < 0)
-        exit(1);
+    int x, y, i;
 
     i = frame_index;
 
@@ -446,77 +361,53 @@ static void fill_yuv_image(AVFrame *pict, int frame_index,
     }
 }
 
-static AVFrame *get_video_frame(OutputStream *ost)
-{
-    AVCodecContext *c = ost->st->codec;
-
-    /* check if we want to generate more frames */
-    if (av_compare_ts(ost->next_pts, ost->st->codec->time_base,
-                      STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
-        return NULL;
-
-    if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
-        /* as we only generate a YUV420P picture, we must convert it
-         * to the codec pixel format if needed */
-        if (!ost->sws_ctx) {
-            ost->sws_ctx = sws_getContext(c->width, c->height,
-                                          AV_PIX_FMT_YUV420P,
-                                          c->width, c->height,
-                                          c->pix_fmt,
-                                          SCALE_FLAGS, NULL, NULL, NULL);
-            if (!ost->sws_ctx) {
-                fprintf(stderr,
-                        "Could not initialize the conversion context\n");
-                exit(1);
-            }
-        }
-        fill_yuv_image(ost->tmp_frame, ost->next_pts, c->width, c->height);
-        sws_scale(ost->sws_ctx,
-                  (const uint8_t * const *)ost->tmp_frame->data, ost->tmp_frame->linesize,
-                  0, c->height, ost->frame->data, ost->frame->linesize);
-    } else {
-        fill_yuv_image(ost->frame, ost->next_pts, c->width, c->height);
-    }
-
-    ost->frame->pts = ost->next_pts++;
-
-    return ost->frame;
-}
-
-/*
- * encode one video frame and send it to the muxer
- * return 1 when encoding is finished, 0 otherwise
- */
-static int write_video_frame(AVFormatContext *oc, OutputStream *ost)
+static void write_video_frame(AVFormatContext *oc, AVStream *st)
 {
     int ret;
-    AVCodecContext *c;
-    AVFrame *frame;
-    int got_packet = 0;
+    static struct SwsContext *sws_ctx;
+    AVCodecContext *c = st->codec;
 
-    c = ost->st->codec;
-
-    frame = get_video_frame(ost);
+    if (frame_count >= STREAM_NB_FRAMES) {
+        /* No more frames to compress. The codec has a latency of a few
+         * frames if using B-frames, so we get the last frames by
+         * passing the same picture again. */
+    } else {
+        if (c->pix_fmt != 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) {
-        /* a hack to avoid data copy with some raw video muxers */
+        /* Raw video case - directly store the picture in the packet */
         AVPacket pkt;
         av_init_packet(&pkt);
 
-        if (!frame)
-            return 1;
-
         pkt.flags        |= AV_PKT_FLAG_KEY;
-        pkt.stream_index  = ost->st->index;
-        pkt.data          = (uint8_t *)frame;
+        pkt.stream_index  = st->index;
+        pkt.data          = dst_picture.data[0];
         pkt.size          = sizeof(AVPicture);
 
-        pkt.pts = pkt.dts = frame->pts;
-        av_packet_rescale_ts(&pkt, c->time_base, ost->st->time_base);
-
         ret = av_interleaved_write_frame(oc, &pkt);
     } else {
         AVPacket pkt = { 0 };
+        int got_packet;
         av_init_packet(&pkt);
 
         /* encode the image */
@@ -525,29 +416,30 @@ static int write_video_frame(AVFormatContext *oc, OutputStream *ost)
             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, ost->st, &pkt);
+        if (!ret && got_packet && pkt.size) {
+            pkt.stream_index = st->index;
+
+            /* Write the compressed frame to the media file. */
+            ret = av_interleaved_write_frame(oc, &pkt);
         } else {
             ret = 0;
         }
     }
-
-    if (ret < 0) {
+    if (ret != 0) {
         fprintf(stderr, "Error while writing video frame: %s\n", av_err2str(ret));
         exit(1);
     }
-
-    return (frame || got_packet) ? 0 : 1;
+    frame_count++;
 }
 
-static void close_stream(AVFormatContext *oc, OutputStream *ost)
+static void close_video(AVFormatContext *oc, AVStream *st)
 {
-    avcodec_close(ost->st->codec);
-    av_frame_free(&ost->frame);
-    av_frame_free(&ost->tmp_frame);
-    sws_freeContext(ost->sws_ctx);
-    swr_free(&ost->swr_ctx);
+    avcodec_close(st->codec);
+    av_free(src_picture.data[0]);
+    av_free(dst_picture.data[0]);
+    av_free(frame);
 }
 
 /**************************************************************/
@@ -555,20 +447,18 @@ static void close_stream(AVFormatContext *oc, OutputStream *ost)
 
 int main(int argc, char **argv)
 {
-    OutputStream video_st = { 0 }, audio_st = { 0 };
     const char *filename;
     AVOutputFormat *fmt;
     AVFormatContext *oc;
+    AVStream *audio_st, *video_st;
     AVCodec *audio_codec, *video_codec;
+    double audio_time, video_time;
     int ret;
-    int have_video = 0, have_audio = 0;
-    int encode_video = 0, encode_audio = 0;
-    AVDictionary *opt = NULL;
 
     /* Initialize libavcodec, and register all codecs and formats. */
     av_register_all();
 
-    if (argc < 2) {
+    if (argc != 2) {
         printf("usage: %s output_file\n"
                "API example program to output a media file with libavformat.\n"
                "This program generates a synthetic audio and video stream, encodes and\n"
@@ -580,9 +470,6 @@ int main(int argc, char **argv)
     }
 
     filename = argv[1];
-    if (argc > 3 && !strcmp(argv[2], "-flags")) {
-        av_dict_set(&opt, argv[2]+1, argv[3], 0);
-    }
 
     /* allocate the output media context */
     avformat_alloc_output_context2(&oc, NULL, NULL, filename);
@@ -590,31 +477,29 @@ int main(int argc, char **argv)
         printf("Could not deduce output format from file extension: using MPEG.\n");
         avformat_alloc_output_context2(&oc, NULL, "mpeg", filename);
     }
-    if (!oc)
+    if (!oc) {
         return 1;
-
+    }
     fmt = oc->oformat;
 
     /* Add the audio and video streams using the default format codecs
      * and initialize the codecs. */
+    video_st = NULL;
+    audio_st = NULL;
+
     if (fmt->video_codec != AV_CODEC_ID_NONE) {
-        add_stream(&video_st, oc, &video_codec, fmt->video_codec);
-        have_video = 1;
-        encode_video = 1;
+        video_st = add_stream(oc, &video_codec, fmt->video_codec);
     }
     if (fmt->audio_codec != AV_CODEC_ID_NONE) {
-        add_stream(&audio_st, oc, &audio_codec, fmt->audio_codec);
-        have_audio = 1;
-        encode_audio = 1;
+        audio_st = add_stream(oc, &audio_codec, fmt->audio_codec);
     }
 
     /* Now that all the parameters are set, we can open the audio and
      * video codecs and allocate the necessary encode buffers. */
-    if (have_video)
-        open_video(oc, video_codec, &video_st, opt);
-
-    if (have_audio)
-        open_audio(oc, audio_codec, &audio_st, opt);
+    if (video_st)
+        open_video(oc, video_codec, video_st);
+    if (audio_st)
+        open_audio(oc, audio_codec, audio_st);
 
     av_dump_format(oc, 0, filename, 1);
 
@@ -629,21 +514,30 @@ int main(int argc, char **argv)
     }
 
     /* Write the stream header, if any. */
-    ret = avformat_write_header(oc, &opt);
+    ret = avformat_write_header(oc, NULL);
     if (ret < 0) {
         fprintf(stderr, "Error occurred when opening output file: %s\n",
                 av_err2str(ret));
         return 1;
     }
 
-    while (encode_video || encode_audio) {
-        /* select the stream to encode */
-        if (encode_video &&
-            (!encode_audio || av_compare_ts(video_st.next_pts, video_st.st->codec->time_base,
-                                            audio_st.next_pts, audio_st.st->codec->time_base) <= 0)) {
-            encode_video = !write_video_frame(oc, &video_st);
+    if (frame)
+        frame->pts = 0;
+    for (;;) {
+        /* Compute current audio and video time. */
+        audio_time = audio_st ? audio_st->pts.val * av_q2d(audio_st->time_base) : 0.0;
+        video_time = video_st ? video_st->pts.val * av_q2d(video_st->time_base) : 0.0;
+
+        if ((!audio_st || audio_time >= STREAM_DURATION) &&
+            (!video_st || video_time >= STREAM_DURATION))
+            break;
+
+        /* write interleaved audio and video frames */
+        if (!video_st || (video_st && audio_st && audio_time < video_time)) {
+            write_audio_frame(oc, audio_st);
         } else {
-            encode_audio = !write_audio_frame(oc, &audio_st);
+            write_video_frame(oc, video_st);
+            frame->pts += av_rescale_q(1, video_st->codec->time_base, video_st->time_base);
         }
     }
 
@@ -654,10 +548,10 @@ int main(int argc, char **argv)
     av_write_trailer(oc);
 
     /* Close each codec. */
-    if (have_video)
-        close_stream(oc, &video_st);
-    if (have_audio)
-        close_stream(oc, &audio_st);
+    if (video_st)
+        close_video(oc, video_st);
+    if (audio_st)
+        close_audio(oc, audio_st);
 
     if (!(fmt->flags & AVFMT_NOFILE))
         /* Close the output file. */

doc/examples/remuxing.c

@@ -1,165 +0,0 @@
-/*
- * Copyright (c) 2013 Stefano Sabatini
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to deal
- * in the Software without restriction, including without limitation the rights
- * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- * copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
- * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
- * THE SOFTWARE.
- */
-
-/**
- * @file
- * libavformat/libavcodec demuxing and muxing API example.
- *
- * Remux streams from one container format to another.
- * @example remuxing.c
- */
-
-#include <libavutil/timestamp.h>
-#include <libavformat/avformat.h>
-
-static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt, const char *tag)
-{
-    AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base;
-
-    printf("%s: pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n",
-           tag,
-           av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base),
-           av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base),
-           av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base),
-           pkt->stream_index);
-}
-
-int main(int argc, char **argv)
-{
-    AVOutputFormat *ofmt = NULL;
-    AVFormatContext *ifmt_ctx = NULL, *ofmt_ctx = NULL;
-    AVPacket pkt;
-    const char *in_filename, *out_filename;
-    int ret, i;
-
-    if (argc < 3) {
-        printf("usage: %s input output\n"
-               "API example program to remux a media file with libavformat and libavcodec.\n"
-               "The output format is guessed according to the file extension.\n"
-               "\n", argv[0]);
-        return 1;
-    }
-
-    in_filename  = argv[1];
-    out_filename = argv[2];
-
-    av_register_all();
-
-    if ((ret = avformat_open_input(&ifmt_ctx, in_filename, 0, 0)) < 0) {
-        fprintf(stderr, "Could not open input file '%s'", in_filename);
-        goto end;
-    }
-
-    if ((ret = avformat_find_stream_info(ifmt_ctx, 0)) < 0) {
-        fprintf(stderr, "Failed to retrieve input stream information");
-        goto end;
-    }
-
-    av_dump_format(ifmt_ctx, 0, in_filename, 0);
-
-    avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, out_filename);
-    if (!ofmt_ctx) {
-        fprintf(stderr, "Could not create output context\n");
-        ret = AVERROR_UNKNOWN;
-        goto end;
-    }
-
-    ofmt = ofmt_ctx->oformat;
-
-    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
-        AVStream *in_stream = ifmt_ctx->streams[i];
-        AVStream *out_stream = avformat_new_stream(ofmt_ctx, in_stream->codec->codec);
-        if (!out_stream) {
-            fprintf(stderr, "Failed allocating output stream\n");
-            ret = AVERROR_UNKNOWN;
-            goto end;
-        }
-
-        ret = avcodec_copy_context(out_stream->codec, in_stream->codec);
-        if (ret < 0) {
-            fprintf(stderr, "Failed to copy context from input to output stream codec context\n");
-            goto end;
-        }
-        out_stream->codec->codec_tag = 0;
-        if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
-            out_stream->codec->flags |= 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

@@ -21,7 +21,7 @@
  */
 
 /**
- * @example resampling_audio.c
+ * @example doc/examples/resampling_audio.c
  * libswresample API use example.
  */
 
@@ -62,7 +62,7 @@ static int get_format_from_sample_fmt(const char **fmt,
 /**
  * Fill dst buffer with nb_samples, generated starting from t.
  */
-static void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t)
+void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t)
 {
     int i, j;
     double tincr = 1.0 / sample_rate, *dstp = dst;
@@ -168,7 +168,7 @@ int main(int argc, char **argv)
         dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) +
                                         src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
         if (dst_nb_samples > max_dst_nb_samples) {
-            av_freep(&dst_data[0]);
+            av_free(dst_data[0]);
             ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
                                    dst_nb_samples, dst_sample_fmt, 1);
             if (ret < 0)
@@ -184,10 +184,6 @@ int main(int argc, char **argv)
         }
         dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
                                                  ret, dst_sample_fmt, 1);
-        if (dst_bufsize < 0) {
-            fprintf(stderr, "Could not get sample buffer size\n");
-            goto end;
-        }
         printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret);
         fwrite(dst_data[0], 1, dst_bufsize, dst_file);
     } while (t < 10);
@@ -199,7 +195,8 @@ int main(int argc, char **argv)
             fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);
 
 end:
-    fclose(dst_file);
+    if (dst_file)
+        fclose(dst_file);
 
     if (src_data)
         av_freep(&src_data[0]);

doc/examples/scaling_video.c

@@ -23,7 +23,7 @@
 /**
  * @file
  * libswscale API use example.
- * @example scaling_video.c
+ * @example doc/examples/scaling_video.c
  */
 
 #include <libavutil/imgutils.h>
@@ -132,7 +132,8 @@ int main(int argc, char **argv)
            av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h, dst_filename);
 
 end:
-    fclose(dst_file);
+    if (dst_file)
+        fclose(dst_file);
     av_freep(&src_data[0]);
     av_freep(&dst_data[0]);
     sws_freeContext(sws_ctx);

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 const char *get_error_text(const int error)
-{
-    static char error_buffer[255];
-    av_strerror(error, error_buffer, sizeof(error_buffer));
-    return error_buffer;
-}
-
-/** Open an input file and the required decoder. */
-static int open_input_file(const char *filename,
-                           AVFormatContext **input_format_context,
-                           AVCodecContext **input_codec_context)
-{
-    AVCodec *input_codec;
-    int error;
-
-    /** Open the input file to read from it. */
-    if ((error = avformat_open_input(input_format_context, filename, NULL,
-                                     NULL)) < 0) {
-        fprintf(stderr, "Could not open input file '%s' (error '%s')\n",
-                filename, get_error_text(error));
-        *input_format_context = NULL;
-        return error;
-    }
-
-    /** Get information on the input file (number of streams etc.). */
-    if ((error = avformat_find_stream_info(*input_format_context, NULL)) < 0) {
-        fprintf(stderr, "Could not open find stream info (error '%s')\n",
-                get_error_text(error));
-        avformat_close_input(input_format_context);
-        return error;
-    }
-
-    /** Make sure that there is only one stream in the input file. */
-    if ((*input_format_context)->nb_streams != 1) {
-        fprintf(stderr, "Expected one audio input stream, but found %d\n",
-                (*input_format_context)->nb_streams);
-        avformat_close_input(input_format_context);
-        return AVERROR_EXIT;
-    }
-
-    /** Find a decoder for the audio stream. */
-    if (!(input_codec = avcodec_find_decoder((*input_format_context)->streams[0]->codec->codec_id))) {
-        fprintf(stderr, "Could not find input codec\n");
-        avformat_close_input(input_format_context);
-        return AVERROR_EXIT;
-    }
-
-    /** Open the decoder for the audio stream to use it later. */
-    if ((error = avcodec_open2((*input_format_context)->streams[0]->codec,
-                               input_codec, NULL)) < 0) {
-        fprintf(stderr, "Could not open input codec (error '%s')\n",
-                get_error_text(error));
-        avformat_close_input(input_format_context);
-        return error;
-    }
-
-    /** Save the decoder context for easier access later. */
-    *input_codec_context = (*input_format_context)->streams[0]->codec;
-
-    return 0;
-}
-
-/**
- * Open an output file and the required encoder.
- * Also set some basic encoder parameters.
- * Some of these parameters are based on the input file's parameters.
- */
-static int open_output_file(const char *filename,
-                            AVCodecContext *input_codec_context,
-                            AVFormatContext **output_format_context,
-                            AVCodecContext **output_codec_context)
-{
-    AVIOContext *output_io_context = NULL;
-    AVStream *stream               = NULL;
-    AVCodec *output_codec          = NULL;
-    int error;
-
-    /** Open the output file to write to it. */
-    if ((error = avio_open(&output_io_context, filename,
-                           AVIO_FLAG_WRITE)) < 0) {
-        fprintf(stderr, "Could not open output file '%s' (error '%s')\n",
-                filename, get_error_text(error));
-        return error;
-    }
-
-    /** Create a new format context for the output container format. */
-    if (!(*output_format_context = avformat_alloc_context())) {
-        fprintf(stderr, "Could not allocate output format context\n");
-        return AVERROR(ENOMEM);
-    }
-
-    /** Associate the output file (pointer) with the container format context. */
-    (*output_format_context)->pb = output_io_context;
-
-    /** Guess the desired container format based on the file extension. */
-    if (!((*output_format_context)->oformat = av_guess_format(NULL, filename,
-                                                              NULL))) {
-        fprintf(stderr, "Could not find output file format\n");
-        goto cleanup;
-    }
-
-    av_strlcpy((*output_format_context)->filename, filename,
-               sizeof((*output_format_context)->filename));
-
-    /** Find the encoder to be used by its name. */
-    if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC))) {
-        fprintf(stderr, "Could not find an AAC encoder.\n");
-        goto cleanup;
-    }
-
-    /** Create a new audio stream in the output file container. */
-    if (!(stream = avformat_new_stream(*output_format_context, output_codec))) {
-        fprintf(stderr, "Could not create new stream\n");
-        error = AVERROR(ENOMEM);
-        goto cleanup;
-    }
-
-    /** Save the encoder context for 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/examples/transcoding.c

@@ -1,601 +0,0 @@
-/*
- * Copyright (c) 2010 Nicolas George
- * Copyright (c) 2011 Stefano Sabatini
- * Copyright (c) 2014 Andrey Utkin
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to deal
- * in the Software without restriction, including without limitation the rights
- * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- * copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
- * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
- * THE SOFTWARE.
- */
-
-/**
- * @file
- * API example for demuxing, decoding, filtering, encoding and muxing
- * @example transcoding.c
- */
-
-#include <libavcodec/avcodec.h>
-#include <libavformat/avformat.h>
-#include <libavfilter/avfiltergraph.h>
-#include <libavfilter/avcodec.h>
-#include <libavfilter/buffersink.h>
-#include <libavfilter/buffersrc.h>
-#include <libavutil/opt.h>
-#include <libavutil/pixdesc.h>
-
-static AVFormatContext *ifmt_ctx;
-static AVFormatContext *ofmt_ctx;
-typedef struct FilteringContext {
-    AVFilterContext *buffersink_ctx;
-    AVFilterContext *buffersrc_ctx;
-    AVFilterGraph *filter_graph;
-} FilteringContext;
-static FilteringContext *filter_ctx;
-
-static int open_input_file(const char *filename)
-{
-    int ret;
-    unsigned int i;
-
-    ifmt_ctx = NULL;
-    if ((ret = avformat_open_input(&ifmt_ctx, filename, NULL, NULL)) < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
-        return ret;
-    }
-
-    if ((ret = avformat_find_stream_info(ifmt_ctx, NULL)) < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
-        return ret;
-    }
-
-    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
-        AVStream *stream;
-        AVCodecContext *codec_ctx;
-        stream = ifmt_ctx->streams[i];
-        codec_ctx = stream->codec;
-        /* Reencode video & audio and remux subtitles etc. */
-        if (codec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
-                || codec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
-            /* Open decoder */
-            ret = avcodec_open2(codec_ctx,
-                    avcodec_find_decoder(codec_ctx->codec_id), NULL);
-            if (ret < 0) {
-                av_log(NULL, AV_LOG_ERROR, "Failed to open decoder for stream #%u\n", i);
-                return ret;
-            }
-        }
-    }
-
-    av_dump_format(ifmt_ctx, 0, filename, 0);
-    return 0;
-}
-
-static int open_output_file(const char *filename)
-{
-    AVStream *out_stream;
-    AVStream *in_stream;
-    AVCodecContext *dec_ctx, *enc_ctx;
-    AVCodec *encoder;
-    int ret;
-    unsigned int i;
-
-    ofmt_ctx = NULL;
-    avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, filename);
-    if (!ofmt_ctx) {
-        av_log(NULL, AV_LOG_ERROR, "Could not create output context\n");
-        return AVERROR_UNKNOWN;
-    }
-
-
-    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
-        out_stream = avformat_new_stream(ofmt_ctx, NULL);
-        if (!out_stream) {
-            av_log(NULL, AV_LOG_ERROR, "Failed allocating output stream\n");
-            return AVERROR_UNKNOWN;
-        }
-
-        in_stream = ifmt_ctx->streams[i];
-        dec_ctx = in_stream->codec;
-        enc_ctx = out_stream->codec;
-
-        if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
-                || dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
-            /* in this example, we choose transcoding to same codec */
-            encoder = avcodec_find_encoder(dec_ctx->codec_id);
-            if (!encoder) {
-                av_log(NULL, AV_LOG_FATAL, "Neccessary encoder not found\n");
-                return AVERROR_INVALIDDATA;
-            }
-
-            /* In this example, we transcode to same properties (picture size,
-             * sample rate etc.). These properties can be changed for output
-             * streams easily using filters */
-            if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) {
-                enc_ctx->height = dec_ctx->height;
-                enc_ctx->width = dec_ctx->width;
-                enc_ctx->sample_aspect_ratio = dec_ctx->sample_aspect_ratio;
-                /* take first format from list of supported formats */
-                enc_ctx->pix_fmt = encoder->pix_fmts[0];
-                /* video time_base can be set to whatever is handy and supported by encoder */
-                enc_ctx->time_base = dec_ctx->time_base;
-            } else {
-                enc_ctx->sample_rate = dec_ctx->sample_rate;
-                enc_ctx->channel_layout = dec_ctx->channel_layout;
-                enc_ctx->channels = av_get_channel_layout_nb_channels(enc_ctx->channel_layout);
-                /* take first format from list of supported formats */
-                enc_ctx->sample_fmt = encoder->sample_fmts[0];
-                enc_ctx->time_base = (AVRational){1, enc_ctx->sample_rate};
-            }
-
-            /* Third parameter can be used to pass settings to encoder */
-            ret = avcodec_open2(enc_ctx, encoder, NULL);
-            if (ret < 0) {
-                av_log(NULL, AV_LOG_ERROR, "Cannot open video encoder for stream #%u\n", i);
-                return ret;
-            }
-        } else if (dec_ctx->codec_type == AVMEDIA_TYPE_UNKNOWN) {
-            av_log(NULL, AV_LOG_FATAL, "Elementary stream #%d is of unknown type, cannot proceed\n", i);
-            return AVERROR_INVALIDDATA;
-        } else {
-            /* if this stream must be remuxed */
-            ret = avcodec_copy_context(ofmt_ctx->streams[i]->codec,
-                    ifmt_ctx->streams[i]->codec);
-            if (ret < 0) {
-                av_log(NULL, AV_LOG_ERROR, "Copying stream context failed\n");
-                return ret;
-            }
-        }
-
-        if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
-            enc_ctx->flags |= CODEC_FLAG_GLOBAL_HEADER;
-
-    }
-    av_dump_format(ofmt_ctx, 0, filename, 1);
-
-    if (!(ofmt_ctx->oformat->flags & AVFMT_NOFILE)) {
-        ret = avio_open(&ofmt_ctx->pb, filename, AVIO_FLAG_WRITE);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Could not open output file '%s'", filename);
-            return ret;
-        }
-    }
-
-    /* init muxer, write output file header */
-    ret = avformat_write_header(ofmt_ctx, NULL);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Error occurred when opening output file\n");
-        return ret;
-    }
-
-    return 0;
-}
-
-static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
-        AVCodecContext *enc_ctx, const char *filter_spec)
-{
-    char args[512];
-    int ret = 0;
-    AVFilter *buffersrc = NULL;
-    AVFilter *buffersink = NULL;
-    AVFilterContext *buffersrc_ctx = NULL;
-    AVFilterContext *buffersink_ctx = NULL;
-    AVFilterInOut *outputs = avfilter_inout_alloc();
-    AVFilterInOut *inputs  = avfilter_inout_alloc();
-    AVFilterGraph *filter_graph = avfilter_graph_alloc();
-
-    if (!outputs || !inputs || !filter_graph) {
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-
-    if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) {
-        buffersrc = avfilter_get_by_name("buffer");
-        buffersink = avfilter_get_by_name("buffersink");
-        if (!buffersrc || !buffersink) {
-            av_log(NULL, AV_LOG_ERROR, "filtering source or sink element not found\n");
-            ret = AVERROR_UNKNOWN;
-            goto end;
-        }
-
-        snprintf(args, sizeof(args),
-                "video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
-                dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
-                dec_ctx->time_base.num, dec_ctx->time_base.den,
-                dec_ctx->sample_aspect_ratio.num,
-                dec_ctx->sample_aspect_ratio.den);
-
-        ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
-                args, NULL, filter_graph);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n");
-            goto end;
-        }
-
-        ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
-                NULL, NULL, filter_graph);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
-            goto end;
-        }
-
-        ret = av_opt_set_bin(buffersink_ctx, "pix_fmts",
-                (uint8_t*)&enc_ctx->pix_fmt, sizeof(enc_ctx->pix_fmt),
-                AV_OPT_SEARCH_CHILDREN);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n");
-            goto end;
-        }
-    } else if (dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
-        buffersrc = avfilter_get_by_name("abuffer");
-        buffersink = avfilter_get_by_name("abuffersink");
-        if (!buffersrc || !buffersink) {
-            av_log(NULL, AV_LOG_ERROR, "filtering source or sink element not found\n");
-            ret = AVERROR_UNKNOWN;
-            goto end;
-        }
-
-        if (!dec_ctx->channel_layout)
-            dec_ctx->channel_layout =
-                av_get_default_channel_layout(dec_ctx->channels);
-        snprintf(args, sizeof(args),
-                "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
-                dec_ctx->time_base.num, dec_ctx->time_base.den, dec_ctx->sample_rate,
-                av_get_sample_fmt_name(dec_ctx->sample_fmt),
-                dec_ctx->channel_layout);
-        ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
-                args, NULL, filter_graph);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
-            goto end;
-        }
-
-        ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
-                NULL, NULL, filter_graph);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
-            goto end;
-        }
-
-        ret = av_opt_set_bin(buffersink_ctx, "sample_fmts",
-                (uint8_t*)&enc_ctx->sample_fmt, sizeof(enc_ctx->sample_fmt),
-                AV_OPT_SEARCH_CHILDREN);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
-            goto end;
-        }
-
-        ret = av_opt_set_bin(buffersink_ctx, "channel_layouts",
-                (uint8_t*)&enc_ctx->channel_layout,
-                sizeof(enc_ctx->channel_layout), AV_OPT_SEARCH_CHILDREN);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
-            goto end;
-        }
-
-        ret = av_opt_set_bin(buffersink_ctx, "sample_rates",
-                (uint8_t*)&enc_ctx->sample_rate, sizeof(enc_ctx->sample_rate),
-                AV_OPT_SEARCH_CHILDREN);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
-            goto end;
-        }
-    } else {
-        ret = AVERROR_UNKNOWN;
-        goto end;
-    }
-
-    /* Endpoints for the filter graph. */
-    outputs->name       = av_strdup("in");
-    outputs->filter_ctx = buffersrc_ctx;
-    outputs->pad_idx    = 0;
-    outputs->next       = NULL;
-
-    inputs->name       = av_strdup("out");
-    inputs->filter_ctx = buffersink_ctx;
-    inputs->pad_idx    = 0;
-    inputs->next       = NULL;
-
-    if (!outputs->name || !inputs->name) {
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-
-    if ((ret = avfilter_graph_parse_ptr(filter_graph, filter_spec,
-                    &inputs, &outputs, NULL)) < 0)
-        goto end;
-
-    if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
-        goto end;
-
-    /* Fill FilteringContext */
-    fctx->buffersrc_ctx = buffersrc_ctx;
-    fctx->buffersink_ctx = buffersink_ctx;
-    fctx->filter_graph = filter_graph;
-
-end:
-    avfilter_inout_free(&inputs);
-    avfilter_inout_free(&outputs);
-
-    return ret;
-}
-
-static int init_filters(void)
-{
-    const char *filter_spec;
-    unsigned int i;
-    int ret;
-    filter_ctx = av_malloc_array(ifmt_ctx->nb_streams, sizeof(*filter_ctx));
-    if (!filter_ctx)
-        return AVERROR(ENOMEM);
-
-    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
-        filter_ctx[i].buffersrc_ctx  = NULL;
-        filter_ctx[i].buffersink_ctx = NULL;
-        filter_ctx[i].filter_graph   = NULL;
-        if (!(ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO
-                || ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO))
-            continue;
-
-
-        if (ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO)
-            filter_spec = "null"; /* passthrough (dummy) filter for video */
-        else
-            filter_spec = "anull"; /* passthrough (dummy) filter for audio */
-        ret = init_filter(&filter_ctx[i], ifmt_ctx->streams[i]->codec,
-                ofmt_ctx->streams[i]->codec, filter_spec);
-        if (ret)
-            return ret;
-    }
-    return 0;
-}
-
-static int encode_write_frame(AVFrame *filt_frame, unsigned int stream_index, int *got_frame) {
-    int ret;
-    int got_frame_local;
-    AVPacket enc_pkt;
-    int (*enc_func)(AVCodecContext *, AVPacket *, const AVFrame *, int *) =
-        (ifmt_ctx->streams[stream_index]->codec->codec_type ==
-         AVMEDIA_TYPE_VIDEO) ? avcodec_encode_video2 : avcodec_encode_audio2;
-
-    if (!got_frame)
-        got_frame = &got_frame_local;
-
-    av_log(NULL, AV_LOG_INFO, "Encoding frame\n");
-    /* encode filtered frame */
-    enc_pkt.data = NULL;
-    enc_pkt.size = 0;
-    av_init_packet(&enc_pkt);
-    ret = enc_func(ofmt_ctx->streams[stream_index]->codec, &enc_pkt,
-            filt_frame, got_frame);
-    av_frame_free(&filt_frame);
-    if (ret < 0)
-        return ret;
-    if (!(*got_frame))
-        return 0;
-
-    /* prepare packet for muxing */
-    enc_pkt.stream_index = stream_index;
-    enc_pkt.dts = av_rescale_q_rnd(enc_pkt.dts,
-            ofmt_ctx->streams[stream_index]->codec->time_base,
-            ofmt_ctx->streams[stream_index]->time_base,
-            AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
-    enc_pkt.pts = av_rescale_q_rnd(enc_pkt.pts,
-            ofmt_ctx->streams[stream_index]->codec->time_base,
-            ofmt_ctx->streams[stream_index]->time_base,
-            AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
-    enc_pkt.duration = av_rescale_q(enc_pkt.duration,
-            ofmt_ctx->streams[stream_index]->codec->time_base,
-            ofmt_ctx->streams[stream_index]->time_base);
-
-    av_log(NULL, AV_LOG_DEBUG, "Muxing frame\n");
-    /* mux encoded frame */
-    ret = av_interleaved_write_frame(ofmt_ctx, &enc_pkt);
-    return ret;
-}
-
-static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index)
-{
-    int ret;
-    AVFrame *filt_frame;
-
-    av_log(NULL, AV_LOG_INFO, "Pushing decoded frame to filters\n");
-    /* push the decoded frame into the filtergraph */
-    ret = av_buffersrc_add_frame_flags(filter_ctx[stream_index].buffersrc_ctx,
-            frame, 0);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
-        return ret;
-    }
-
-    /* pull filtered frames from the filtergraph */
-    while (1) {
-        filt_frame = av_frame_alloc();
-        if (!filt_frame) {
-            ret = AVERROR(ENOMEM);
-            break;
-        }
-        av_log(NULL, AV_LOG_INFO, "Pulling filtered frame from filters\n");
-        ret = av_buffersink_get_frame(filter_ctx[stream_index].buffersink_ctx,
-                filt_frame);
-        if (ret < 0) {
-            /* if no more frames for output - returns AVERROR(EAGAIN)
-             * if flushed and no more frames for output - returns AVERROR_EOF
-             * rewrite retcode to 0 to show it as normal procedure completion
-             */
-            if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
-                ret = 0;
-            av_frame_free(&filt_frame);
-            break;
-        }
-
-        filt_frame->pict_type = AV_PICTURE_TYPE_NONE;
-        ret = encode_write_frame(filt_frame, stream_index, NULL);
-        if (ret < 0)
-            break;
-    }
-
-    return ret;
-}
-
-static int flush_encoder(unsigned int stream_index)
-{
-    int ret;
-    int got_frame;
-
-    if (!(ofmt_ctx->streams[stream_index]->codec->codec->capabilities &
-                CODEC_CAP_DELAY))
-        return 0;
-
-    while (1) {
-        av_log(NULL, AV_LOG_INFO, "Flushing stream #%u encoder\n", stream_index);
-        ret = encode_write_frame(NULL, stream_index, &got_frame);
-        if (ret < 0)
-            break;
-        if (!got_frame)
-            return 0;
-    }
-    return ret;
-}
-
-int main(int argc, char **argv)
-{
-    int ret;
-    AVPacket packet = { .data = NULL, .size = 0 };
-    AVFrame *frame = NULL;
-    enum AVMediaType type;
-    unsigned int stream_index;
-    unsigned int i;
-    int got_frame;
-    int (*dec_func)(AVCodecContext *, AVFrame *, int *, const AVPacket *);
-
-    if (argc != 3) {
-        av_log(NULL, AV_LOG_ERROR, "Usage: %s <input file> <output file>\n", argv[0]);
-        return 1;
-    }
-
-    av_register_all();
-    avfilter_register_all();
-
-    if ((ret = open_input_file(argv[1])) < 0)
-        goto end;
-    if ((ret = open_output_file(argv[2])) < 0)
-        goto end;
-    if ((ret = init_filters()) < 0)
-        goto end;
-
-    /* read all packets */
-    while (1) {
-        if ((ret = av_read_frame(ifmt_ctx, &packet)) < 0)
-            break;
-        stream_index = packet.stream_index;
-        type = ifmt_ctx->streams[packet.stream_index]->codec->codec_type;
-        av_log(NULL, AV_LOG_DEBUG, "Demuxer gave frame of stream_index %u\n",
-                stream_index);
-
-        if (filter_ctx[stream_index].filter_graph) {
-            av_log(NULL, AV_LOG_DEBUG, "Going to reencode&filter the frame\n");
-            frame = av_frame_alloc();
-            if (!frame) {
-                ret = AVERROR(ENOMEM);
-                break;
-            }
-            packet.dts = av_rescale_q_rnd(packet.dts,
-                    ifmt_ctx->streams[stream_index]->time_base,
-                    ifmt_ctx->streams[stream_index]->codec->time_base,
-                    AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
-            packet.pts = av_rescale_q_rnd(packet.pts,
-                    ifmt_ctx->streams[stream_index]->time_base,
-                    ifmt_ctx->streams[stream_index]->codec->time_base,
-                    AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
-            dec_func = (type == AVMEDIA_TYPE_VIDEO) ? avcodec_decode_video2 :
-                avcodec_decode_audio4;
-            ret = dec_func(ifmt_ctx->streams[stream_index]->codec, frame,
-                    &got_frame, &packet);
-            if (ret < 0) {
-                av_frame_free(&frame);
-                av_log(NULL, AV_LOG_ERROR, "Decoding failed\n");
-                break;
-            }
-
-            if (got_frame) {
-                frame->pts = av_frame_get_best_effort_timestamp(frame);
-                ret = filter_encode_write_frame(frame, stream_index);
-                av_frame_free(&frame);
-                if (ret < 0)
-                    goto end;
-            } else {
-                av_frame_free(&frame);
-            }
-        } else {
-            /* remux this frame without reencoding */
-            packet.dts = av_rescale_q_rnd(packet.dts,
-                    ifmt_ctx->streams[stream_index]->time_base,
-                    ofmt_ctx->streams[stream_index]->time_base,
-                    AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
-            packet.pts = av_rescale_q_rnd(packet.pts,
-                    ifmt_ctx->streams[stream_index]->time_base,
-                    ofmt_ctx->streams[stream_index]->time_base,
-                    AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
-
-            ret = av_interleaved_write_frame(ofmt_ctx, &packet);
-            if (ret < 0)
-                goto end;
-        }
-        av_free_packet(&packet);
-    }
-
-    /* flush filters and encoders */
-    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
-        /* flush filter */
-        if (!filter_ctx[i].filter_graph)
-            continue;
-        ret = filter_encode_write_frame(NULL, i);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Flushing filter failed\n");
-            goto end;
-        }
-
-        /* flush encoder */
-        ret = flush_encoder(i);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Flushing encoder failed\n");
-            goto end;
-        }
-    }
-
-    av_write_trailer(ofmt_ctx);
-end:
-    av_free_packet(&packet);
-    av_frame_free(&frame);
-    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
-        avcodec_close(ifmt_ctx->streams[i]->codec);
-        if (ofmt_ctx && ofmt_ctx->nb_streams > i && ofmt_ctx->streams[i] && ofmt_ctx->streams[i]->codec)
-            avcodec_close(ofmt_ctx->streams[i]->codec);
-        if (filter_ctx && filter_ctx[i].filter_graph)
-            avfilter_graph_free(&filter_ctx[i].filter_graph);
-    }
-    av_free(filter_ctx);
-    avformat_close_input(&ifmt_ctx);
-    if (ofmt_ctx && !(ofmt_ctx->oformat->flags & AVFMT_NOFILE))
-        avio_close(ofmt_ctx->pb);
-    avformat_free_context(ofmt_ctx);
-
-    if (ret < 0)
-        av_log(NULL, AV_LOG_ERROR, "Error occurred: %s\n", av_err2str(ret));
-
-    return ret ? 1 : 0;
-}

doc/faq.texi

@@ -368,6 +368,26 @@ ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 44100 -i all.a \
 rm temp[12].[av] all.[av]
 @end example
 
+@section -profile option fails when encoding H.264 video with AAC audio
+
+@command{ffmpeg} prints an error like
+
+@example
+Undefined constant or missing '(' in 'baseline'
+Unable to parse option value "baseline"
+Error setting option profile to value baseline.
+@end example
+
+Short answer: write @option{-profile:v} instead of @option{-profile}.
+
+Long answer: this happens because the @option{-profile} option can apply to both
+video and audio.  Specifically the AAC encoder also defines some profiles, none
+of which are named @var{baseline}.
+
+The solution is to apply the @option{-profile} option to the video stream only
+by using @url{http://ffmpeg.org/ffmpeg.html#Stream-specifiers-1, Stream specifiers}.
+Appending @code{:v} to it will do exactly that.
+
 @section Using @option{-f lavfi}, audio becomes mono for no apparent reason.
 
 Use @option{-dumpgraph -} to find out exactly where the channel layout is
@@ -392,7 +412,7 @@ VOB and a few other formats do not have a global header that describes
 everything present in the file. Instead, applications are supposed to scan
 the file to see what it contains. Since VOB files are frequently large, only
 the beginning is scanned. If the subtitles happen only later in the file,
-they will not be initially detected.
+they will not be initally detected.
 
 Some applications, including the @code{ffmpeg} command-line tool, can only
 work with streams that were detected during the initial scan; streams that

doc/ffmpeg-resampler.texi

@@ -14,7 +14,7 @@
 
 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,
+allows to perform audio resampling, audio channel layout rematrixing,
 and convert audio format and packing layout.
 
 @c man end DESCRIPTION

doc/ffmpeg-scaler.texi

@@ -13,7 +13,7 @@
 @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
+library image conversion utilities. In particular it allows to perform
 image rescaling and pixel format conversion.
 
 @c man end DESCRIPTION

doc/ffmpeg.texi

@@ -80,23 +80,11 @@ The transcoding process in @command{ffmpeg} for each output can be described by
 the following diagram:
 
 @example
- _______              ______________
-|       |            |              |
-| input |  demuxer   | encoded data |   decoder
-| file  | ---------> | packets      | -----+
-|_______|            |______________|      |
-                                           v
-                                       _________
-                                      |         |
-                                      | decoded |
-                                      | frames  |
-                                      |_________|
- ________             ______________       |
-|        |           |              |      |
-| output | <-------- | encoded data | <----+
-| file   |   muxer   | packets      |   encoder
-|________|           |______________|
-
+ _______              ______________               _________              ______________            ________
+|       |            |              |             |         |            |              |          |        |
+| input |  demuxer   | encoded data |   decoder   | decoded |  encoder   | encoded data |  muxer   | output |
+| file  | ---------> | packets      |  ---------> | frames  | ---------> | packets      | -------> | file   |
+|_______|            |______________|             |_________|            |______________|          |________|
 
 @end example
 
@@ -124,16 +112,11 @@ the same type. In the above diagram they can be represented by simply inserting
 an additional step between decoding and encoding:
 
 @example
- _________                        ______________
-|         |                      |              |
-| decoded |                      | encoded data |
-| frames  |\                   _ | packets      |
-|_________| \                  /||______________|
-             \   __________   /
-  simple     _\||          | /  encoder
-  filtergraph   | filtered |/
-                | frames   |
-                |__________|
+ _________                        __________              ______________
+|         |                      |          |            |              |
+| decoded |  simple filtergraph  | filtered |  encoder   | encoded data |
+| frames  | -------------------> | frames   | ---------> | packets      |
+|_________|                      |__________|            |______________|
 
 @end example
 
@@ -142,10 +125,10 @@ Simple filtergraphs are configured with the per-stream @option{-filter} option
 A simple filtergraph for video can look for example like this:
 
 @example
- _______        _____________        _______        ________
-|       |      |             |      |       |      |        |
-| input | ---> | deinterlace | ---> | scale | ---> | output |
-|_______|      |_____________|      |_______|      |________|
+ _______        _____________        _______        _____        ________
+|       |      |             |      |       |      |     |      |        |
+| input | ---> | deinterlace | ---> | scale | ---> | fps | ---> | output |
+|_______|      |_____________|      |_______|      |_____|      |________|
 
 @end example
 
@@ -272,13 +255,8 @@ ffmpeg -i INPUT -map 0 -c copy -c:v:1 libx264 -c:a:137 libvorbis OUTPUT
 will copy all the streams except the second video, which will be encoded with
 libx264, and the 138th audio, which will be encoded with libvorbis.
 
-@item -t @var{duration} (@emph{input/output})
-When used as an input option (before @code{-i}), limit the @var{duration} of
-data read from the input file.
-
-When used as an output option (before an output filename), stop writing the
-output after its duration reaches @var{duration}.
-
+@item -t @var{duration} (@emph{output})
+Stop writing the output after its duration reaches @var{duration}.
 @var{duration} may be a number in seconds, or in @code{hh:mm:ss[.xxx]} form.
 
 -to and -t are mutually exclusive and -t has priority.
@@ -307,20 +285,23 @@ input until the timestamps reach @var{position}.
 @var{position} may be either in seconds or in @code{hh:mm:ss[.xxx]} form.
 
 @item -itsoffset @var{offset} (@emph{input})
-Set the input time offset.
+Set the input time offset in seconds.
+@code{[-]hh:mm:ss[.xxx]} syntax is also supported.
+The offset is added to the timestamps of the input files.
+Specifying a positive offset means that the corresponding
+streams are delayed by @var{offset} seconds.
 
-@var{offset} must be a time duration specification,
-see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
-
-The offset is added to the timestamps of the input files. Specifying
-a positive offset means that the corresponding streams are delayed by
-the time duration specified in @var{offset}.
-
-@item -timestamp @var{date} (@emph{output})
+@item -timestamp @var{time} (@emph{output})
 Set the recording timestamp in the container.
-
-@var{date} must be a time duration specification,
-see @ref{date syntax,,the Date section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
+The syntax for @var{time} is:
+@example
+now|([(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH:MM:SS[.m...])|(HHMMSS[.m...]))[Z|z])
+@end example
+If the value is "now" it takes the current time.
+Time is local time unless 'Z' or 'z' is appended, in which case it is
+interpreted as UTC.
+If the year-month-day part is not specified it takes the current
+year-month-day.
 
 @item -metadata[:metadata_specifier] @var{key}=@var{value} (@emph{output,per-metadata})
 Set a metadata key/value pair.
@@ -339,7 +320,7 @@ ffmpeg -i in.avi -metadata title="my title" out.flv
 
 To set the language of the first audio stream:
 @example
-ffmpeg -i INPUT -metadata:s:a:0 language=eng OUTPUT
+ffmpeg -i INPUT -metadata:s:a:1 language=eng OUTPUT
 @end example
 
 @item -target @var{type} (@emph{output})
@@ -367,13 +348,8 @@ Stop writing to the stream after @var{framecount} frames.
 
 @item -q[:@var{stream_specifier}] @var{q} (@emph{output,per-stream})
 @itemx -qscale[:@var{stream_specifier}] @var{q} (@emph{output,per-stream})
-Use fixed quality scale (VBR). The meaning of @var{q}/@var{qscale} is
+Use fixed quality scale (VBR). The meaning of @var{q} is
 codec-dependent.
-If @var{qscale} is used without a @var{stream_specifier} then it applies only
-to the video stream, this is to maintain compatibility with previous behavior
-and as specifying the same codec specific value to 2 different codecs that is
-audio and video generally is not what is intended when no stream_specifier is
-used.
 
 @anchor{filter_option}
 @item -filter[:@var{stream_specifier}] @var{filtergraph} (@emph{output,per-stream})
@@ -473,9 +449,6 @@ Set frame rate (Hz value, fraction or abbreviation).
 
 As an input option, ignore any timestamps stored in the file and instead
 generate timestamps assuming constant frame rate @var{fps}.
-This is not the same as the @option{-framerate} option used for some input formats
-like image2 or v4l2 (it used to be the same in older versions of FFmpeg).
-If in doubt use @option{-framerate} instead of the input option @option{-r}.
 
 As an output option, duplicate or drop input frames to achieve constant output
 frame rate @var{fps}.
@@ -530,6 +503,9 @@ prefix is ``ffmpeg2pass''. The complete file name will be
 @file{PREFIX-N.log}, where N is a number specific to the output
 stream
 
+@item -vlang @var{code}
+Set the ISO 639 language code (3 letters) of the current video stream.
+
 @item -vf @var{filtergraph} (@emph{output})
 Create the filtergraph specified by @var{filtergraph} and use it to
 filter the stream.
@@ -537,7 +513,7 @@ filter the stream.
 This is an alias for @code{-filter:v}, see the @ref{filter_option,,-filter option}.
 @end table
 
-@section Advanced Video options
+@section Advanced Video Options
 
 @table @option
 @item -pix_fmt[:@var{stream_specifier}] @var{format} (@emph{input/output,per-stream})
@@ -640,52 +616,6 @@ would be more efficient.
 @item -copyinkf[:@var{stream_specifier}] (@emph{output,per-stream})
 When doing stream copy, copy also non-key frames found at the
 beginning.
-
-@item -hwaccel[:@var{stream_specifier}] @var{hwaccel} (@emph{input,per-stream})
-Use hardware acceleration to decode the matching stream(s). The allowed values
-of @var{hwaccel} are:
-@table @option
-@item none
-Do not use any hardware acceleration (the default).
-
-@item auto
-Automatically select the hardware acceleration method.
-
-@item vda
-Use Apple VDA hardware acceleration.
-
-@item vdpau
-Use VDPAU (Video Decode and Presentation API for Unix) hardware acceleration.
-
-@item dxva2
-Use DXVA2 (DirectX Video Acceleration) hardware acceleration.
-@end table
-
-This option has no effect if the selected hwaccel is not available or not
-supported by the chosen decoder.
-
-Note that most acceleration methods are intended for playback and will not be
-faster than software decoding on modern CPUs. Additionally, @command{ffmpeg}
-will usually need to copy the decoded frames from the GPU memory into the system
-memory, resulting in further performance loss. This option is thus mainly
-useful for testing.
-
-@item -hwaccel_device[:@var{stream_specifier}] @var{hwaccel_device} (@emph{input,per-stream})
-Select a device to use for hardware acceleration.
-
-This option only makes sense when the @option{-hwaccel} option is also
-specified. Its exact meaning depends on the specific hardware acceleration
-method chosen.
-
-@table @option
-@item vdpau
-For VDPAU, this option specifies the X11 display/screen to use. If this option
-is not specified, the value of the @var{DISPLAY} environment variable is used
-
-@item dxva2
-For DXVA2, this option should contain the number of the display adapter to use.
-If this option is not specified, the default adapter is used.
-@end table
 @end table
 
 @section Audio Options
@@ -720,7 +650,7 @@ filter the stream.
 This is an alias for @code{-filter:a}, see the @ref{filter_option,,-filter option}.
 @end table
 
-@section Advanced Audio options
+@section Advanced Audio options:
 
 @table @option
 @item -atag @var{fourcc/tag} (@emph{output})
@@ -735,9 +665,11 @@ stereo but not 6 channels as 5.1. The default is to always try to guess. Use
 0 to disable all guessing.
 @end table
 
-@section Subtitle options
+@section Subtitle options:
 
 @table @option
+@item -slang @var{code}
+Set the ISO 639 language code (3 letters) of the current subtitle stream.
 @item -scodec @var{codec} (@emph{input/output})
 Set the subtitle codec. This is an alias for @code{-codec:s}.
 @item -sn (@emph{output})
@@ -746,7 +678,7 @@ Disable subtitle recording.
 Deprecated, see -bsf
 @end table
 
-@section Advanced Subtitle options
+@section Advanced Subtitle options:
 
 @table @option
 
@@ -824,11 +756,6 @@ To map all the streams except the second audio, use negative mappings
 ffmpeg -i INPUT -map 0 -map -0:a:1 OUTPUT
 @end example
 
-To pick the English audio stream:
-@example
-ffmpeg -i INPUT -map 0:m:language:eng OUTPUT
-@end example
-
 Note that using this option disables the default mappings for this output file.
 
 @item -map_channel [@var{input_file_id}.@var{stream_specifier}.@var{channel_id}|-1][:@var{output_file_id}.@var{stream_specifier}]
@@ -1071,7 +998,7 @@ ffmpeg -i h264.mp4 -c:v copy -bsf:v h264_mp4toannexb -an out.h264
 ffmpeg -i file.mov -an -vn -bsf:s mov2textsub -c:s copy -f rawvideo sub.txt
 @end example
 
-@item -tag[:@var{stream_specifier}] @var{codec_tag} (@emph{input/output,per-stream})
+@item -tag[:@var{stream_specifier}] @var{codec_tag} (@emph{per-stream})
 Force a tag/fourcc for matching streams.
 
 @item -timecode @var{hh}:@var{mm}:@var{ss}SEP@var{ff}
@@ -1145,38 +1072,12 @@ transcoding. Use @option{-noaccurate_seek} to disable it, which may be useful
 e.g. when copying some streams and transcoding the others.
 
 @item -override_ffserver (@emph{global})
-Overrides the input specifications from @command{ffserver}. Using this
-option you can map any input stream to @command{ffserver} and control
-many aspects of the encoding from @command{ffmpeg}. Without this
-option @command{ffmpeg} will transmit to @command{ffserver} what is
-requested by @command{ffserver}.
-
+Overrides the input specifications from ffserver. Using this option you can
+map any input stream to ffserver and control many aspects of the encoding from
+ffmpeg. Without this option ffmpeg will transmit to ffserver what is requested by
+ffserver.
 The option is intended for cases where features are needed that cannot be
-specified to @command{ffserver} but can be to @command{ffmpeg}.
-
-@item -discard (@emph{input})
-Allows discarding specific streams or frames of streams at the demuxer.
-Not all demuxers support this.
-
-@table @option
-@item none
-Discard no frame.
-
-@item default
-Default, which discards no frames.
-
-@item noref
-Discard all non-reference frames.
-
-@item bidir
-Discard all bidirectional frames.
-
-@item nokey
-Discard all frames excepts keyframes.
-
-@item all
-Discard all frames.
-@end table
+specified to ffserver but can be to ffmpeg.
 
 @end table
 
@@ -1458,11 +1359,11 @@ ffmpeg -f image2 -pattern_type glob -i 'foo-*.jpeg' -r 12 -s WxH foo.avi
 You can put many streams of the same type in the output:
 
 @example
-ffmpeg -i test1.avi -i test2.avi -map 1:1 -map 1:0 -map 0:1 -map 0:0 -c copy -y test12.nut
+ffmpeg -i test1.avi -i test2.avi -map 0:3 -map 0:2 -map 0:1 -map 0:0 -c copy test12.nut
 @end example
 
-The resulting output file @file{test12.nut} will contain the first four streams
-from the input files in reverse order.
+The resulting output file @file{test12.avi} will contain first four streams from
+the input file in reverse order.
 
 @item
 To force CBR video output:

doc/ffplay.texi

@@ -84,9 +84,6 @@ output. In the filtergraph, the input is associated to the label
 ffmpeg-filters manual for more information about the filtergraph
 syntax.
 
-You can specify this parameter multiple times and cycle through the specified
-filtergraphs along with the show modes by pressing the key @key{w}.
-
 @item -af @var{filtergraph}
 @var{filtergraph} is a description of the filtergraph to apply to
 the input audio.
@@ -162,10 +159,6 @@ Force a specific video decoder.
 
 @item -scodec @var{codec_name}
 Force a specific subtitle decoder.
-
-@item -autorotate
-Automatically rotate the video according to presentation metadata. Set by
-default, use -noautorotate to disable.
 @end table
 
 @section While playing
@@ -181,7 +174,7 @@ Toggle full screen.
 Pause.
 
 @item a
-Cycle audio channel in the current program.
+Cycle audio channel in the curret program.
 
 @item v
 Cycle video channel.
@@ -193,13 +186,7 @@ Cycle subtitle channel in the current program.
 Cycle program.
 
 @item w
-Cycle video filters or show modes.
-
-@item s
-Step to the next frame.
-
-Pause if the stream is not already paused, step to the next video
-frame, and pause.
+Show audio waves.
 
 @item left/right
 Seek backward/forward 10 seconds.
@@ -208,8 +195,6 @@ Seek backward/forward 10 seconds.
 Seek backward/forward 1 minute.
 
 @item page down/page up
-Seek to the previous/next chapter.
-or if there are no chapters
 Seek backward/forward 10 minutes.
 
 @item mouse click
@@ -221,7 +206,6 @@ Seek to percentage in file corresponding to fraction of width.
 
 @include config.texi
 @ifset config-all
-@set config-readonly
 @ifset config-avutil
 @include utils.texi
 @end ifset

doc/ffprobe.texi

@@ -119,10 +119,6 @@ Show payload data, as a hexadecimal and ASCII dump. Coupled with
 
 The dump is printed as the "data" field. It may contain newlines.
 
-@item -show_data_hash @var{algorithm}
-Show a hash of payload data, for packets with @option{-show_packets} and for
-codec extradata with @option{-show_streams}.
-
 @item -show_error
 Show information about the error found when trying to probe the input.
 
@@ -184,7 +180,7 @@ format : stream=codec_type
 
 To show all the tags in the stream and format sections:
 @example
-stream_tags : format_tags
+format_tags : format_tags
 @end example
 
 To show only the @code{title} tag (if available) in the stream
@@ -201,11 +197,11 @@ The information for each single packet is printed within a dedicated
 section with name "PACKET".
 
 @item -show_frames
-Show information about each frame and subtitle contained in the input
-multimedia stream.
+Show information about each frame contained in the input multimedia
+stream.
 
 The information for each single frame is printed within a dedicated
-section with name "FRAME" or "SUBTITLE".
+section with name "FRAME".
 
 @item -show_streams
 Show information about each media stream contained in the input
@@ -341,39 +337,6 @@ A writer may accept one or more arguments, which specify the options
 to adopt. The options are specified as a list of @var{key}=@var{value}
 pairs, separated by ":".
 
-All writers support the following options:
-
-@table @option
-@item string_validation, sv
-Set string validation mode.
-
-The following values are accepted.
-@table @samp
-@item fail
-The writer will fail immediately in case an invalid string (UTF-8)
-sequence or code point is found in the input. This is especially
-useful to validate input metadata.
-
-@item ignore
-Any validation error will be ignored. This will result in possibly
-broken output, especially with the json or xml writer.
-
-@item replace
-The writer will substitute invalid UTF-8 sequences or code points with
-the string specified with the @option{string_validation_replacement}.
-@end table
-
-Default value is @samp{replace}.
-
-@item string_validation_replacement, svr
-Set replacement string to use in case @option{string_validation} is
-set to @samp{replace}.
-
-In case the option is not specified, the writer will assume the empty
-string, that is it will remove the invalid sequences from the input
-strings.
-@end table
-
 A description of the currently available writers follows.
 
 @section default
@@ -603,7 +566,6 @@ DV, GXF and AVI timecodes are available in format metadata
 
 @include config.texi
 @ifset config-all
-@set config-readonly
 @ifset config-avutil
 @include utils.texi
 @end ifset

doc/ffprobe.xsd

@@ -8,15 +8,15 @@
 
     <xsd:complexType name="ffprobeType">
         <xsd:sequence>
-            <xsd:element name="program_version"  type="ffprobe:programVersionType"  minOccurs="0" maxOccurs="1" />
-            <xsd:element name="library_versions" type="ffprobe:libraryVersionsType" minOccurs="0" maxOccurs="1" />
             <xsd:element name="packets"  type="ffprobe:packetsType" minOccurs="0" maxOccurs="1" />
             <xsd:element name="frames"   type="ffprobe:framesType"  minOccurs="0" maxOccurs="1" />
-            <xsd:element name="programs" type="ffprobe:programsType" minOccurs="0" maxOccurs="1" />
             <xsd:element name="streams"  type="ffprobe:streamsType" minOccurs="0" maxOccurs="1" />
+            <xsd:element name="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>
 
@@ -28,10 +28,7 @@
 
     <xsd:complexType name="framesType">
         <xsd:sequence>
-            <xsd:choice minOccurs="0" maxOccurs="unbounded">
-                <xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
-                <xsd:element name="subtitle" type="ffprobe:subtitleType" minOccurs="0" maxOccurs="unbounded"/>
-            </xsd:choice>
+            <xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
         </xsd:sequence>
     </xsd:complexType>
 
@@ -50,15 +47,9 @@
       <xsd:attribute name="pos"           type="xsd:long"  />
       <xsd:attribute name="flags"         type="xsd:string" use="required" />
       <xsd:attribute name="data"          type="xsd:string" />
-      <xsd:attribute name="data_hash"     type="xsd:string" />
     </xsd:complexType>
 
     <xsd:complexType name="frameType">
-      <xsd:sequence>
-            <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
-            <xsd:element name="side_data_list" type="ffprobe:frameSideDataListType"   minOccurs="0" maxOccurs="1" />
-      </xsd:sequence>
-
       <xsd:attribute name="media_type"    type="xsd:string" use="required"/>
       <xsd:attribute name="key_frame"     type="xsd:int"    use="required"/>
       <xsd:attribute name="pts"           type="xsd:long" />
@@ -67,8 +58,6 @@
       <xsd:attribute name="pkt_pts_time"  type="xsd:float"/>
       <xsd:attribute name="pkt_dts"       type="xsd:long" />
       <xsd:attribute name="pkt_dts_time"  type="xsd:float"/>
-      <xsd:attribute name="best_effort_timestamp"      type="xsd:long" />
-      <xsd:attribute name="best_effort_timestamp_time" type="xsd:float" />
       <xsd:attribute name="pkt_duration"  type="xsd:long" />
       <xsd:attribute name="pkt_duration_time" type="xsd:float"/>
       <xsd:attribute name="pkt_pos"       type="xsd:long" />
@@ -93,26 +82,6 @@
       <xsd:attribute name="repeat_pict"            type="xsd:int"   />
     </xsd:complexType>
 
-    <xsd:complexType name="frameSideDataListType">
-        <xsd:sequence>
-            <xsd:element name="side_data" type="ffprobe:frameSideDataType" minOccurs="1" maxOccurs="unbounded"/>
-        </xsd:sequence>
-    </xsd:complexType>
-    <xsd:complexType name="frameSideDataType">
-        <xsd:attribute name="side_data_type"              type="xsd:string"/>
-        <xsd:attribute name="side_data_size"              type="xsd:int"   />
-    </xsd:complexType>
-
-    <xsd:complexType name="subtitleType">
-      <xsd:attribute name="media_type"         type="xsd:string" fixed="subtitle" use="required"/>
-      <xsd:attribute name="pts"                type="xsd:long" />
-      <xsd:attribute name="pts_time"           type="xsd:float"/>
-      <xsd:attribute name="format"             type="xsd:int"  />
-      <xsd:attribute name="start_display_time" type="xsd:int"  />
-      <xsd:attribute name="end_display_time"   type="xsd:int"  />
-      <xsd:attribute name="num_rects"          type="xsd:int"  />
-    </xsd:complexType>
-
     <xsd:complexType name="streamsType">
         <xsd:sequence>
             <xsd:element name="stream" type="ffprobe:streamType" minOccurs="0" maxOccurs="unbounded"/>
@@ -154,7 +123,6 @@
       <xsd:attribute name="codec_tag"        type="xsd:string" use="required"/>
       <xsd:attribute name="codec_tag_string" type="xsd:string" use="required"/>
       <xsd:attribute name="extradata"        type="xsd:string" />
-      <xsd:attribute name="extradata_hash"   type="xsd:string" />
 
       <!-- video attributes -->
       <xsd:attribute name="width"                type="xsd:int"/>
@@ -164,8 +132,6 @@
       <xsd:attribute name="display_aspect_ratio" type="xsd:string"/>
       <xsd:attribute name="pix_fmt"              type="xsd:string"/>
       <xsd:attribute name="level"                type="xsd:int"/>
-      <xsd:attribute name="color_range"          type="xsd:string"/>
-      <xsd:attribute name="color_space"          type="xsd:string"/>
       <xsd:attribute name="timecode"             type="xsd:string"/>
 
       <!-- audio attributes -->
@@ -184,8 +150,6 @@
       <xsd:attribute name="duration_ts"      type="xsd:long"/>
       <xsd:attribute name="duration"         type="xsd:float"/>
       <xsd:attribute name="bit_rate"         type="xsd:int"/>
-      <xsd:attribute name="max_bit_rate"     type="xsd:int"/>
-      <xsd:attribute name="bits_per_raw_sample" type="xsd:int"/>
       <xsd:attribute name="nb_frames"        type="xsd:int"/>
       <xsd:attribute name="nb_read_frames"   type="xsd:int"/>
       <xsd:attribute name="nb_read_packets"  type="xsd:int"/>
@@ -240,7 +204,8 @@
       <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_ident"   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>
 

doc/ffserver.conf

@@ -1,11 +1,11 @@
 # Port on which the server is listening. You must select a different
 # port from your standard HTTP web server if it is running on the same
 # computer.
-HTTPPort 8090
+Port 8090
 
 # Address on which the server is bound. Only useful if you have
 # several network interfaces.
-HTTPBindAddress 0.0.0.0
+BindAddress 0.0.0.0
 
 # Number of simultaneous HTTP connections that can be handled. It has
 # to be defined *before* the MaxClients parameter, since it defines the
@@ -235,7 +235,7 @@ StartSendOnKey
 
 #<Stream test.ogg>
 #Feed feed1.ffm
-#Metadata title "Stream title"
+#Title "Stream title"
 #AudioBitRate 64
 #AudioChannels 2
 #AudioSampleRate 44100
@@ -280,10 +280,10 @@ StartSendOnKey
 #<Stream file.asf>
 #File "/usr/local/httpd/htdocs/test.asf"
 #NoAudio
-#Metadata author "Me"
-#Metadata copyright "Super MegaCorp"
-#Metadata title "Test stream from disk"
-#Metadata comment "Test comment"
+#Author "Me"
+#Copyright "Super MegaCorp"
+#Title "Test stream from disk"
+#Comment "Test comment"
 #</Stream>
 
 

doc/ffserver.texi

@@ -16,14 +16,11 @@ ffserver [@var{options}]
 @chapter Description
 @c man begin DESCRIPTION
 
-@command{ffserver} is a streaming server for both audio and video.
-It supports several live feeds, streaming from files and time shifting
-on live feeds. You can seek to positions in the past on each live
-feed, provided you specify a big enough feed storage.
-
-@command{ffserver} is configured through a configuration file, which
-is read at startup. If not explicitly specified, it will read from
-@file{/etc/ffserver.conf}.
+@command{ffserver} is a streaming server for both audio and video. It
+supports several live feeds, streaming from files and time shifting on
+live feeds (you can seek to positions in the past on each live feed,
+provided you specify a big enough feed storage in
+@file{ffserver.conf}).
 
 @command{ffserver} receives prerecorded files or FFM streams from some
 @command{ffmpeg} instance as input, then streams them over
@@ -42,126 +39,10 @@ For each feed you can have different output streams in various
 formats, each one specified by a @code{<Stream>} section in the
 configuration file.
 
-@chapter Detailed description
-
-@command{ffserver} works by forwarding streams encoded by
-@command{ffmpeg}, or pre-recorded streams which are read from disk.
-
-Precisely, @command{ffserver} acts as an HTTP server, accepting POST
-requests from @command{ffmpeg} to acquire the stream to publish, and
-serving RTSP clients or HTTP clients GET requests with the stream
-media content.
-
-A feed is an @ref{FFM} stream created by @command{ffmpeg}, and sent to
-a port where @command{ffserver} is listening.
-
-Each feed is identified by a unique name, corresponding to the name
-of the resource published on @command{ffserver}, and is configured by
-a dedicated @code{Feed} section in the configuration file.
-
-The feed publish URL is given by:
-@example
-http://@var{ffserver_ip_address}:@var{http_port}/@var{feed_name}
-@end example
-
-where @var{ffserver_ip_address} is the IP address of the machine where
-@command{ffserver} is installed, @var{http_port} is the port number of
-the HTTP server (configured through the @option{HTTPPort} option), and
-@var{feed_name} is the name of the corresponding feed defined in the
-configuration file.
-
-Each feed is associated to a file which is stored on disk. This stored
-file is used to 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{HTTPPort} and
-@var{RTSPPort} respectively.
-
-In case the stream is associated to a feed, the encoding parameters
-must be configured in the stream configuration. They are sent to
-@command{ffmpeg} when setting up the encoding. This allows
-@command{ffserver} to define the encoding parameters used by
-the @command{ffmpeg} encoders.
-
-The @command{ffmpeg} @option{override_ffserver} commandline option
-allows one to override the encoding parameters set by the server.
-
-Multiple streams can be connected to the same feed.
-
-For example, you can have a situation described by the following
-graph:
-@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.
+ffserver supports an HTTP interface which exposes the current status
+of the server.
 
 Simply point your browser to the address of the special status stream
 specified in the configuration file.
@@ -180,8 +61,27 @@ ACL allow 192.168.0.0 192.168.255.255
 then the server will post a page with the status information when
 the special stream @file{status.html} is requested.
 
+@section What can this do?
+
+When properly configured and running, you can capture video and audio in real
+time from a suitable capture card, and stream it out over the Internet to
+either Windows Media Player or RealAudio player (with some restrictions).
+
+It can also stream from files, though that is currently broken. Very often, a
+web server can be used to serve up the files just as well.
+
+It can stream prerecorded video from .ffm files, though it is somewhat tricky
+to make it work correctly.
+
 @section How do I make it work?
 
+First, build the kit. It *really* helps to have installed LAME first. Then when
+you run the ffserver ./configure, make sure that you have the
+@code{--enable-libmp3lame} flag turned on.
+
+LAME is important as it allows for streaming audio to Windows Media Player.
+Don't ask why the other audio types do not work.
+
 As a simple test, just run the following two command lines where INPUTFILE
 is some file which you can decode with ffmpeg:
 
@@ -203,9 +103,40 @@ WARNING: trying to stream test1.mpg doesn't work with WMP as it tries to
 transfer the entire file before starting to play.
 The same is true of AVI files.
 
-You should edit the @file{ffserver.conf} file to suit your needs (in
-terms of frame rates etc). Then install @command{ffserver} and
-@command{ffmpeg}, write a script to start them up, and off you go.
+@section What happens next?
+
+You should edit the ffserver.conf file to suit your needs (in terms of
+frame rates etc). Then install ffserver and ffmpeg, write a script to start
+them up, and off you go.
+
+@section Troubleshooting
+
+@subsection I don't hear any audio, but video is fine.
+
+Maybe you didn't install LAME, or got your ./configure statement wrong. Check
+the ffmpeg output to see if a line referring to MP3 is present. If not, then
+your configuration was incorrect. If it is, then maybe your wiring is not
+set up correctly. Maybe the sound card is not getting data from the right
+input source. Maybe you have a really awful audio interface (like I do)
+that only captures in stereo and also requires that one channel be flipped.
+If you are one of these people, then export 'AUDIO_FLIP_LEFT=1' before
+starting ffmpeg.
+
+@subsection The audio and video lose sync after a while.
+
+Yes, they do.
+
+@subsection After a long while, the video update rate goes way down in WMP.
+
+Yes, it does. Who knows why?
+
+@subsection WMP 6.4 behaves differently to WMP 7.
+
+Yes, it does. Any thoughts on this would be gratefully received. These
+differences extend to embedding WMP into a web page. [There are two
+object IDs that you can use: The old one, which does not play well, and
+the new one, which does (both tested on the same system). However,
+I suspect that the new one is not available unless you have installed WMP 7].
 
 @section What else can it do?
 
@@ -246,6 +177,9 @@ specify a time. In addition, ffserver will skip frames until a key_frame
 is found. This further reduces the startup delay by not transferring data
 that will be discarded.
 
+* You may want to adjust the MaxBandwidth in the ffserver.conf to limit
+the amount of bandwidth consumed by live streams.
+
 @section Why does the ?buffer / Preroll stop working after a time?
 
 It turns out that (on my machine at least) the number of frames successfully
@@ -279,6 +213,19 @@ You use this by adding the ?date= to the end of the URL for the stream.
 For example:   @samp{http://localhost:8080/test.asf?date=2002-07-26T23:05:00}.
 @c man end
 
+@section What is FFM, FFM2
+
+FFM and FFM2 are formats used by ffserver. They allow storing a wide variety of
+video and audio streams and encoding options, and can store a moving time segment
+of an infinite movie or a whole movie.
+
+FFM is version specific, and there is limited compatibility of FFM files
+generated by one version of ffmpeg/ffserver and another version of
+ffmpeg/ffserver. It may work but it is not guaranteed to work.
+
+FFM2 is extensible while maintaining compatibility and should work between
+differing versions of tools. FFM2 is the default.
+
 @chapter Options
 @c man begin OPTIONS
 
@@ -288,549 +235,15 @@ For example:   @samp{http://localhost:8080/test.asf?date=2002-07-26T23:05:00}.
 
 @table @option
 @item -f @var{configfile}
-Read configuration file @file{configfile}. If not specified it will
-read by default from @file{/etc/ffserver.conf}.
-
+Use @file{configfile} instead of @file{/etc/ffserver.conf}.
 @item -n
-Enable no-launch mode. This option disables all the @code{Launch}
-directives within the various @code{<Feed>} sections. Since
-@command{ffserver} will not launch any @command{ffmpeg} instances, you
-will have to launch them manually.
-
+Enable no-launch mode. This option disables all the Launch directives
+within the various <Stream> sections. Since ffserver will not launch
+any ffmpeg instances, you will have to launch them manually.
 @item -d
-Enable debug mode. This option increases log verbosity, and directs
-log messages to stdout. When specified, the @option{CustomLog} option
-is ignored.
+Enable debug mode. This option increases log verbosity, directs log
+messages to stdout.
 @end table
-
-@chapter Configuration file syntax
-
-@command{ffserver} reads a configuration file containing global
-options and settings for each stream and feed.
-
-The configuration file consists of global options and dedicated
-sections, which must be introduced by "<@var{SECTION_NAME}
-@var{ARGS}>" on a separate line and must be terminated by a line in
-the form "</@var{SECTION_NAME}>". @var{ARGS} is optional.
-
-Currently the following sections are recognized: @samp{Feed},
-@samp{Stream}, @samp{Redirect}.
-
-A line starting with @code{#} is ignored and treated as a comment.
-
-Name of options and sections are case-insensitive.
-
-@section ACL syntax
-An ACL (Access Control List) specifies the address which are allowed
-to access a given stream, or to write a given feed.
-
-It accepts the folling forms
-@itemize
-@item
-Allow/deny access to @var{address}.
-@example
-ACL ALLOW <address>
-ACL DENY <address>
-@end example
-
-@item
-Allow/deny access to ranges of addresses from @var{first_address} to
-@var{last_address}.
-@example
-ACL ALLOW <first_address> <last_address>
-ACL DENY <first_address> <last_address>
-@end example
-@end itemize
-
-You can repeat the ACL allow/deny as often as you like. It is on a per
-stream basis. The first match defines the action. If there are no matches,
-then the default is the inverse of the last ACL statement.
-
-Thus 'ACL allow localhost' only allows access from localhost.
-'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and
-allow everybody else.
-
-@section Global options
-@table @option
-@item HTTPPort @var{port_number}
-@item Port @var{port_number}
-@item RTSPPort @var{port_number}
-
-@var{HTTPPort} sets the HTTP server listening TCP port number,
-@var{RTSPPort} sets the RTSP server listening TCP port number.
-
-@var{Port} is the equivalent of @var{HTTPPort} and is deprecated.
-
-You must select a different port from your standard HTTP web server if
-it is running on the same computer.
-
-If not specified, no corresponding server will be created.
-
-@item HTTPBindAddress @var{ip_address}
-@item BindAddress @var{ip_address}
-@item RTSPBindAddress @var{ip_address}
-Set address on which the HTTP/RTSP server is bound. Only useful if you
-have several network interfaces.
-
-@var{BindAddress} is the equivalent of @var{HTTPBindAddress} and is
-deprecated.
-
-@item MaxHTTPConnections @var{n}
-Set number of simultaneous HTTP connections that can be handled. It
-has to be defined @emph{before} the @option{MaxClients} parameter,
-since it defines the @option{MaxClients} maximum limit.
-
-Default value is 2000.
-
-@item MaxClients @var{n}
-Set number of simultaneous requests that can be handled. Since
-@command{ffserver} is very fast, it is more likely that you will want
-to leave this high and use @option{MaxBandwidth}.
-
-Default value is 5.
-
-@item MaxBandwidth @var{kbps}
-Set the maximum amount of kbit/sec that you are prepared to consume
-when streaming to clients.
-
-Default value is 1000.
-
-@item CustomLog @var{filename}
-Set access log file (uses standard Apache log file format). '-' is the
-standard output.
-
-If not specified @command{ffserver} will produce no log.
-
-In case the commandline option @option{-d} is specified this option is
-ignored, and the log is written to standard output.
-
-@item NoDaemon
-Set no-daemon mode. This option is currently ignored since now
-@command{ffserver} will always work in no-daemon mode, and is
-deprecated.
-@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

doc/fftools-common-opts.texi

@@ -3,7 +3,7 @@ representing a number as input, which may be followed by one of the SI
 unit prefixes, for example: 'K', 'M', or 'G'.
 
 If 'i' is appended to the SI unit prefix, the complete prefix will be
-interpreted as a unit prefix for binary multiples, which are based on
+interpreted as a unit prefix for binary multiplies, which are based on
 powers of 1024 instead of powers of 1000. Appending 'B' to the SI unit
 prefix multiplies the value by 8. This allows using, for example:
 'KB', 'MiB', 'G' and 'B' as number suffixes.
@@ -44,15 +44,8 @@ streams of this type.
 If @var{stream_index} is given, then it matches the stream with number @var{stream_index}
 in the program with the id @var{program_id}. Otherwise, it matches all streams in the
 program.
-@item #@var{stream_id} or i:@var{stream_id}
-Match the stream by stream id (e.g. PID in MPEG-TS container).
-@item m:@var{key}[:@var{value}]
-Matches streams with the metadata tag @var{key} having the specified value. If
-@var{value} is not given, matches streams that contain the given tag with any
-value.
-
-Note that in @command{ffmpeg}, matching by metadata will only work properly for
-input files.
+@item #@var{stream_id}
+Matches the stream by a format-specific ID.
 @end table
 
 @section Generic options
@@ -196,20 +189,11 @@ following option is recognized:
 set the file name to use for the report; @code{%p} is expanded to the name
 of the program, @code{%t} is expanded to a timestamp, @code{%%} is expanded
 to a plain @code{%}
-@item level
-set the log level
 @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.
@@ -266,10 +250,6 @@ Possible flags for this option are:
 @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}.

doc/formats.texi

@@ -125,26 +125,18 @@ Consider things that a sane encoder should not do as an error.
 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
+Shift timestamps to make them non-negative. A value of 1 enables shifting,
+a value of 0 disables it, the default value of -1 enables shifting
+when required by the target format.
 
 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.
 
+Also note that this affects only leading negative timestamps, and not
+non-monotonic negative timestamps.
+
 @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.
@@ -156,18 +148,6 @@ Correct single timestamp overflows if set to 1. Default is 1.
 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
@@ -203,10 +183,6 @@ 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/general.texi

@@ -122,20 +122,6 @@ x264 is under the GNU Public License Version 2 or later
 details), you must upgrade FFmpeg's license to GPL in order to use it.
 @end float
 
-@section x265
-
-FFmpeg can make use of the x265 library for HEVC encoding.
-
-Go to @url{http://x265.org/developers.html} and follow the instructions
-for installing the library. Then pass @code{--enable-libx265} to configure
-to enable it.
-
-@float NOTE
-x265 is under the GNU Public License Version 2 or later
-(see @url{http://www.gnu.org/licenses/old-licenses/gpl-2.0.html} for
-details), you must upgrade FFmpeg's license to GPL in order to use it.
-@end float
-
 @section libilbc
 
 iLBC is a narrowband speech codec that has been made freely available
@@ -162,27 +148,6 @@ libzvbi is licensed under the GNU General Public License Version 2 or later
 you must upgrade FFmpeg's license to GPL in order to use it.
 @end float
 
-@section AviSynth
-
-FFmpeg can read AviSynth scripts as input. To enable support, pass
-@code{--enable-avisynth} to configure.  The correct headers are
-included in compat/avisynth/, which allows the user to enable support
-without needing to search for these headers themselves.
-
-For Windows, supported AviSynth variants are
-@url{http://avisynth.nl, AviSynth 2.5 or 2.6} for 32-bit builds and
-@url{http://avs-plus.net, AviSynth+ 0.1} for 32-bit and 64-bit builds.
-
-For Linux and OS X, the supported AviSynth variant is
-@url{https://github.com/avxsynth/avxsynth, AvxSynth}.
-
-@float NOTE
-AviSynth and AvxSynth are loaded dynamically.  Distributors can build FFmpeg
-with @code{--enable-avisynth}, and the binaries will work regardless of the
-end user having AviSynth or AvxSynth installed - they'll only need to be
-installed to use AviSynth scripts (obviously).
-@end float
-
 
 @chapter Supported File Formats, Codecs or Features
 
@@ -205,7 +170,7 @@ library:
 @item American Laser Games MM   @tab   @tab X
     @tab Multimedia format used in games like Mad Dog McCree.
 @item 3GPP AMR                  @tab X @tab X
-@item Amazing Studio Packed Animation File  @tab   @tab X
+@item Amazing Studio Packed Animation File        @tab   @tab X
     @tab Multimedia format used in game Heart Of Darkness.
 @item Apple HTTP Live Streaming @tab   @tab X
 @item Artworx Data Format       @tab   @tab X
@@ -245,7 +210,6 @@ library:
     @tab Multimedia format used by Delphine Software games.
 @item CD+G                      @tab   @tab X
     @tab Video format used by CD+G karaoke disks
-@item Phantom Cine              @tab   @tab X
 @item Commodore CDXL            @tab   @tab X
     @tab Amiga CD video format
 @item Core Audio Format         @tab X @tab X
@@ -259,7 +223,6 @@ library:
 @item Deluxe Paint Animation    @tab   @tab X
 @item DFA                       @tab   @tab X
     @tab This format is used in Chronomaster game
-@item DSD Stream File (DSF)     @tab   @tab X
 @item DV video                  @tab X @tab X
 @item DXA                       @tab   @tab X
     @tab This format is used in the non-Windows version of the Feeble Files
@@ -286,8 +249,6 @@ library:
 @item GXF                       @tab X @tab X
     @tab General eXchange Format SMPTE 360M, used by Thomson Grass Valley
          playout servers.
-@item HNM @tab   @tab X
-    @tab Only version 4 supported, used in some games from Cryo Interactive
 @item iCEDraw File              @tab   @tab X
 @item ICO                       @tab X @tab X
     @tab Microsoft Windows ICO
@@ -310,11 +271,9 @@ library:
     @tab Used by Linux Media Labs MPEG-4 PCI boards
 @item LOAS                      @tab   @tab X
     @tab contains LATM multiplexed AAC audio
-@item LRC                       @tab X @tab X
 @item LVF                       @tab   @tab X
 @item LXF                       @tab   @tab X
     @tab VR native stream format, used by Leitch/Harris' video servers.
-@item Magic Lantern Video (MLV) @tab   @tab X
 @item Matroska                  @tab X @tab X
 @item Matroska audio            @tab X @tab
 @item FFmpeg metadata           @tab X @tab X
@@ -340,8 +299,6 @@ library:
     @tab also known as DVB Transport Stream
 @item MPEG-4                    @tab X @tab X
     @tab MPEG-4 is a variant of QuickTime.
-@item Mirillis FIC video        @tab   @tab X
-    @tab No cursor rendering.
 @item MIME multipart JPEG       @tab X @tab
 @item MSN TCP webcam            @tab   @tab X
     @tab Used by MSN Messenger webcam streams.
@@ -381,7 +338,6 @@ library:
 @item raw H.261                 @tab X @tab X
 @item raw H.263                 @tab X @tab X
 @item raw H.264                 @tab X @tab X
-@item raw HEVC                  @tab X @tab X
 @item raw Ingenient MJPEG       @tab   @tab X
 @item raw MJPEG                 @tab X @tab X
 @item raw MLP                   @tab   @tab X
@@ -492,13 +448,11 @@ following image formats are supported:
 @item Name @tab Encoding @tab Decoding @tab Comments
 @item .Y.U.V       @tab X @tab X
     @tab one raw file per component
-@item Alias PIX    @tab X @tab X
-    @tab Alias/Wavefront PIX image format
 @item animated GIF @tab X @tab X
 @item BMP          @tab X @tab X
     @tab Microsoft BMP image
-@item BRender PIX  @tab   @tab X
-    @tab Argonaut BRender 3D engine image format.
+@item PIX          @tab   @tab X
+    @tab PIX is an image format used in the Argonaut BRender engine.
 @item DPX          @tab X @tab X
     @tab Digital Picture Exchange
 @item EXR          @tab   @tab X
@@ -534,8 +488,8 @@ following image formats are supported:
     @tab YUV, JPEG and some extension is not supported yet.
 @item Truevision Targa  @tab X @tab X
     @tab Targa (.TGA) image format
-@item WebP         @tab E @tab X
-    @tab WebP image format, encoding supported through external library libwebp
+@item WebP         @tab @tab X
+    @tab WebP image format
 @item XBM  @tab X @tab X
     @tab X BitMap image format
 @item XFace @tab X @tab X
@@ -653,9 +607,6 @@ following image formats are supported:
 @item H.263+ / H.263-1998 / H.263 version 2  @tab  X  @tab  X
 @item H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10  @tab  E  @tab  X
     @tab encoding supported through external library libx264
-@item HEVC                   @tab  X  @tab  X
-    @tab encoding supported through the external library libx265
-@item HNM version 4          @tab     @tab  X
 @item HuffYUV                @tab  X  @tab  X
 @item HuffYUV FFmpeg variant @tab  X  @tab  X
 @item IBM Ultimotion         @tab     @tab  X
@@ -686,8 +637,8 @@ following image formats are supported:
 @item LCL (LossLess Codec Library) MSZH  @tab     @tab  X
 @item LCL (LossLess Codec Library) ZLIB  @tab  E  @tab  E
 @item LOCO                   @tab     @tab  X
-@item LucasArts SANM/Smush   @tab     @tab  X
-    @tab Used in LucasArts games / SMUSH animations.
+@item LucasArts Smush        @tab     @tab  X
+    @tab Used in LucasArts games.
 @item lossless MJPEG         @tab  X  @tab  X
 @item Microsoft ATC Screen   @tab     @tab  X
     @tab Also known as Microsoft Screen 3.
@@ -707,6 +658,7 @@ following image formats are supported:
 @item Mobotix MxPEG video    @tab     @tab  X
 @item Motion Pixels video    @tab     @tab  X
 @item MPEG-1 video           @tab  X  @tab  X
+@item MPEG-1/2 video XvMC (X-Video Motion Compensation)  @tab     @tab  X
 @item MPEG-2 video           @tab  X  @tab  X
 @item MPEG-4 part 2          @tab  X  @tab  X
     @tab libxvidcore can be used alternatively for encoding.
@@ -722,8 +674,6 @@ following image formats are supported:
     @tab fourcc: VP50
 @item On2 VP6                @tab     @tab  X
     @tab fourcc: VP60,VP61,VP62
-@item On2 VP7                @tab     @tab  X
-    @tab fourcc: VP70,VP71
 @item VP8                    @tab  E  @tab  X
     @tab fourcc: VP80, encoding supported through external library libvpx
 @item VP9                    @tab  E  @tab  X
@@ -753,11 +703,11 @@ following image formats are supported:
     @tab Texture dictionaries used by the Renderware Engine.
 @item RL2 video              @tab     @tab  X
     @tab used in some games by Entertainment Software Partners
+@item SGI RLE 8-bit          @tab     @tab  X
 @item Sierra VMD video       @tab     @tab  X
     @tab Used in Sierra VMD files.
 @item Silicon Graphics Motion Video Compressor 1 (MVC1)  @tab     @tab  X
 @item Silicon Graphics Motion Video Compressor 2 (MVC2)  @tab     @tab  X
-@item Silicon Graphics RLE 8-bit video  @tab     @tab  X
 @item Smacker video          @tab     @tab  X
     @tab Video encoding used in Smacker.
 @item SMPTE VC-1             @tab     @tab  X
@@ -823,7 +773,7 @@ following image formats are supported:
     @tab encoding supported through external library libaacplus
 @item AAC                    @tab  E  @tab  X
     @tab encoding supported through external library libfaac and libvo-aacenc
-@item AC-3                   @tab IX  @tab  IX
+@item AC-3                   @tab IX  @tab  X
 @item ADPCM 4X Movie         @tab     @tab  X
 @item ADPCM CDROM XA         @tab     @tab  X
 @item ADPCM Creative Technology @tab     @tab  X
@@ -867,8 +817,6 @@ following image formats are supported:
 @item ADPCM Sound Blaster Pro 2-bit  @tab     @tab  X
 @item ADPCM Sound Blaster Pro 2.6-bit  @tab     @tab  X
 @item ADPCM Sound Blaster Pro 4-bit  @tab     @tab  X
-@item ADPCM VIMA
-    @tab Used in LucasArts SMUSH animations.
 @item ADPCM Westwood Studios IMA @tab     @tab  X
     @tab Used in Westwood Studios games like Command and Conquer.
 @item ADPCM Yamaha           @tab  X  @tab  X
@@ -881,7 +829,6 @@ following image formats are supported:
     @tab QuickTime fourcc 'alac'
 @item ATRAC1                 @tab     @tab  X
 @item ATRAC3                 @tab     @tab  X
-@item ATRAC3+                @tab     @tab  X
 @item Bink Audio             @tab     @tab  X
     @tab Used in Bink and Smacker files in many games.
 @item CELT                   @tab     @tab  E
@@ -901,10 +848,6 @@ following image formats are supported:
 @item DPCM Sol               @tab     @tab  X
 @item DPCM Xan               @tab     @tab  X
     @tab Used in Origin's Wing Commander IV AVI files.
-@item DSD (Direct Stream Digitial), least significant bit first  @tab  @tab  X
-@item DSD (Direct Stream Digitial), most significant bit first   @tab  @tab  X
-@item DSD (Direct Stream Digitial), least significant bit first, planar  @tab  @tab  X
-@item DSD (Direct Stream Digitial), most significant bit first, planar   @tab  @tab  X
 @item DSP Group TrueSpeech   @tab     @tab  X
 @item DV audio               @tab     @tab  X
 @item Enhanced AC-3          @tab  X  @tab  X
@@ -927,14 +870,13 @@ following image formats are supported:
 @item Monkey's Audio         @tab     @tab  X
 @item MP1 (MPEG audio layer 1)  @tab     @tab IX
 @item MP2 (MPEG audio layer 2)  @tab IX  @tab IX
-    @tab encoding supported also through external library TwoLAME
+    @tab libtwolame can be used alternatively for encoding.
 @item MP3 (MPEG audio layer 3)  @tab  E  @tab IX
     @tab encoding supported through external library LAME, ADU MP3 and MP3onMP4 also supported
 @item MPEG-4 Audio Lossless Coding (ALS)  @tab     @tab  X
 @item Musepack SV7           @tab     @tab  X
 @item Musepack SV8           @tab     @tab  X
 @item Nellymoser Asao        @tab  X  @tab  X
-@item On2 AVC (Audio for Video Codec) @tab     @tab  X
 @item Opus                   @tab  E  @tab  E
     @tab supported through external library libopus
 @item PCM A-law              @tab  X  @tab  X
@@ -997,6 +939,7 @@ following image formats are supported:
 @item Vorbis                 @tab  E  @tab  X
     @tab A native but very primitive encoder exists.
 @item Voxware MetaSound      @tab     @tab  X
+    @tab imperfect and incomplete support
 @item WavPack                @tab  X  @tab  X
 @item Westwood Audio (SND1)  @tab     @tab  X
 @item Windows Media Audio 1  @tab  X  @tab  X
@@ -1037,7 +980,7 @@ performance on systems without hardware floating point support).
 @item TED Talks captions @tab @tab X @tab   @tab X
 @item VobSub (IDX+SUB) @tab   @tab X @tab   @tab X
 @item VPlayer          @tab   @tab X @tab   @tab X
-@item WebVTT           @tab X @tab X @tab X @tab X
+@item WebVTT           @tab X @tab X @tab   @tab X
 @item XSUB             @tab   @tab   @tab X @tab X
 @end multitable
 
@@ -1050,12 +993,10 @@ performance on systems without hardware floating point support).
 @multitable @columnfractions .4 .1
 @item Name         @tab Support
 @item file         @tab X
-@item FTP          @tab X
 @item Gopher       @tab X
 @item HLS          @tab X
 @item HTTP         @tab X
 @item HTTPS        @tab X
-@item Icecast      @tab X
 @item MMSH         @tab X
 @item MMST         @tab X
 @item pipe         @tab X
@@ -1066,9 +1007,7 @@ performance on systems without hardware floating point support).
 @item RTMPTE       @tab X
 @item RTMPTS       @tab X
 @item RTP          @tab X
-@item SAMBA        @tab E
 @item SCTP         @tab X
-@item SFTP         @tab E
 @item TCP          @tab X
 @item TLS          @tab X
 @item UDP          @tab X
@@ -1088,19 +1027,17 @@ performance on systems without hardware floating point support).
 @item caca              @tab        @tab X
 @item DV1394            @tab X      @tab
 @item Lavfi virtual device @tab X   @tab
-@item Linux framebuffer @tab X      @tab X
+@item Linux framebuffer @tab X      @tab
 @item JACK              @tab X      @tab
 @item LIBCDIO           @tab X
 @item LIBDC1394         @tab X      @tab
 @item OpenAL            @tab X
-@item OpenGL            @tab        @tab X
 @item OSS               @tab X      @tab X
-@item PulseAudio        @tab X      @tab X
+@item Pulseaudio        @tab X      @tab
 @item SDL               @tab        @tab X
 @item Video4Linux2      @tab X      @tab X
 @item VfW capture       @tab X      @tab
 @item X11 grabbing      @tab X      @tab
-@item Win32 grabbing    @tab X      @tab
 @end multitable
 
 @code{X} means that input/output is supported.

doc/git-howto.texi

@@ -299,7 +299,7 @@ the current branch history.
 git commit --amend
 @end example
 
-allows one to amend the last commit details quickly.
+allows to amend the last commit details quickly.
 
 @example
 git rebase -i origin/master

doc/git-howto.txt

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

doc/indevs.texi

@@ -13,8 +13,8 @@ You can disable all the input devices using the configure option
 option "--enable-indev=@var{INDEV}", or you can disable a particular
 input device using the option "--disable-indev=@var{INDEV}".
 
-The option "-devices" of the ff* tools will display the list of
-supported input devices.
+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.
 
@@ -51,41 +51,6 @@ ffmpeg -f alsa -i hw:0 alsaout.wav
 For more information see:
 @url{http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html}
 
-@section avfoundation
-
-AVFoundation input device.
-
-AVFoundation is the currently recommended framework by Apple for streamgrabbing on OSX >= 10.7 as well as on iOS.
-The older QTKit framework has been marked deprecated since OSX version 10.7.
-
-The filename passed as input is parsed to contain either a device name or index.
-The device index can also be given by using -video_device_index.
-A given device index will override any given device name.
-If the desired device consists of numbers only, use -video_device_index to identify it.
-The default device will be chosen if an empty string  or the device name "default" is given.
-The available devices can be enumerated by using -list_devices.
-The pixel format can be set using -pixel_format.
-Available formats:
- monob, rgb555be, rgb555le, rgb565be, rgb565le, rgb24, bgr24, 0rgb, bgr0, 0bgr, rgb0,
- bgr48be, uyvy422, yuva444p, yuva444p16le, yuv444p, yuv422p16, yuv422p10, yuv444p10,
- yuv420p, nv12, yuyv422, gray
-
-@example
-ffmpeg -f avfoundation -i "0" out.mpg
-@end example
-
-@example
-ffmpeg -f avfoundation -video_device_index 0 -i "" out.mpg
-@end example
-
-@example
-ffmpeg -f avfoundation -pixel_format bgr0 -i "default" out.mpg
-@end example
-
-@example
-ffmpeg -f avfoundation -list_devices true -i ""
-@end example
-
 @section bktr
 
 BSD video input device.
@@ -227,81 +192,6 @@ ffmpeg -f fbdev -frames:v 1 -r 1 -i /dev/fb0 screenshot.jpeg
 
 See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1).
 
-@section gdigrab
-
-Win32 GDI-based screen capture device.
-
-This device allows you to capture a region of the display on Windows.
-
-There are two options for the input filename:
-@example
-desktop
-@end example
-or
-@example
-title=@var{window_title}
-@end example
-
-The first option will capture the entire desktop, or a fixed region of the
-desktop. The second option will instead capture the contents of a single
-window, regardless of its position on the screen.
-
-For example, to grab the entire desktop using @command{ffmpeg}:
-@example
-ffmpeg -f gdigrab -framerate 6 -i desktop out.mpg
-@end example
-
-Grab a 640x480 region at position @code{10,20}:
-@example
-ffmpeg -f gdigrab -framerate 6 -offset_x 10 -offset_y 20 -video_size vga -i desktop out.mpg
-@end example
-
-Grab the contents of the window named "Calculator"
-@example
-ffmpeg -f gdigrab -framerate 6 -i title=Calculator out.mpg
-@end example
-
-@subsection Options
-
-@table @option
-@item draw_mouse
-Specify whether to draw the mouse pointer. Use the value @code{0} to
-not draw the pointer. Default value is @code{1}.
-
-@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.
-
-Note that @var{show_region} is incompatible with grabbing the contents
-of a single window.
-
-For example:
-@example
-ffmpeg -f gdigrab -show_region 1 -framerate 6 -video_size cif -offset_x 10 -offset_y 20 -i desktop out.mpg
-@end example
-
-@item video_size
-Set the video frame size. The default is to capture the full screen if @file{desktop} is selected, or the full window size if @file{title=@var{window_title}} is selected.
-
-@item offset_x
-When capturing a region with @var{video_size}, set the distance from the left edge of the screen or desktop.
-
-Note that the offset calculation is from the top left corner of the primary monitor on Windows. If you have a monitor positioned to the left of your primary monitor, you will need to use a negative @var{offset_x} value to move the region to that monitor.
-
-@item offset_y
-When capturing a region with @var{video_size}, set the distance from the top edge of the screen or desktop.
-
-Note that the offset calculation is from the top left corner of the primary monitor on Windows. If you have a monitor positioned above your primary monitor, you will need to use a negative @var{offset_y} value to move the region to that monitor.
-
-@end table
-
 @section iec61883
 
 FireWire DV/HDV input device using libiec61883.
@@ -483,28 +373,10 @@ ffplay -f lavfi "movie=test.avi[out0];amovie=test.wav[out1]"
 
 @end itemize
 
-@section libcdio
-
-Audio-CD input device based on cdio.
-
-To enable this input device during configuration you need libcdio
-installed on your system. Requires the configure option
-@code{--enable-libcdio}.
-
-This device allows playing and grabbing from an Audio-CD.
-
-For example to copy with @command{ffmpeg} the entire Audio-CD in /dev/sr0,
-you may run the command:
-@example
-ffmpeg -f libcdio -i /dev/sr0 cd.wav
-@end example
-
 @section libdc1394
 
 IIDC1394 input device, based on libdc1394 and libraw1394.
 
-Requires the configure option @code{--enable-libdc1394}.
-
 @section openal
 
 The OpenAL input device provides audio capture on all systems with a
@@ -537,7 +409,7 @@ 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
+This device allows to capture from an audio input device handled
 through OpenAL.
 
 You need to specify the name of the device to capture in the provided
@@ -659,33 +531,6 @@ Record a stream from default device:
 ffmpeg -f pulse -i default /tmp/pulse.wav
 @end example
 
-@section qtkit
-
-QTKit input device.
-
-The filename passed as input is parsed to contain either a device name or index.
-The device index can also be given by using -video_device_index.
-A given device index will override any given device name.
-If the desired device consists of numbers only, use -video_device_index to identify it.
-The default device will be chosen if an empty string  or the device name "default" is given.
-The available devices can be enumerated by using -list_devices.
-
-@example
-ffmpeg -f qtkit -i "0" out.mpg
-@end example
-
-@example
-ffmpeg -f qtkit -video_device_index 0 -i "" out.mpg
-@end example
-
-@example
-ffmpeg -f qtkit -i "default" out.mpg
-@end example
-
-@example
-ffmpeg -f qtkit -list_devices true -i ""
-@end example
-
 @section sndio
 
 sndio input device.
@@ -772,7 +617,7 @@ 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
+This option allows to select the input format, when several are
 available.
 
 @item framerate
@@ -833,10 +678,7 @@ other filename will be interpreted as device number 0.
 
 X11 video input device.
 
-Depends on X11, Xext, and Xfixes. Requires the configure option
-@code{--enable-x11grab}.
-
-This device allows one to capture a region of an X11 display.
+This device allows to capture a region of an X11 display.
 
 The filename passed as input has the syntax:
 @example
@@ -916,10 +758,6 @@ ffmpeg -f x11grab -follow_mouse centered -show_region 1 -framerate 25 -video_siz
 
 @item video_size
 Set the video frame size. Default value is @code{vga}.
-
-@item use_shm
-Use the MIT-SHM extension for shared memory. Default value is @code{1}.
-It may be necessary to disable it for remote displays.
 @end table
 
 @c man end INPUT DEVICES

doc/issue_tracker.txt

@@ -22,7 +22,7 @@ a mail for every change to every issue.
 (the above does all work already after light testing)
 
 The subscription URL for the ffmpeg-trac list is:
-http(s)://lists.ffmpeg.org/mailman/listinfo/ffmpeg-trac
+http(s)://ffmpeg.org/mailman/listinfo/ffmpeg-trac
 The URL of the webinterface of the tracker is:
 http(s)://trac.ffmpeg.org
 

doc/libavutil.texi

@@ -16,25 +16,7 @@ The libavutil library is a utility library to aid portable
 multimedia programming. It contains safe portable string functions,
 random number generators, data structures, additional mathematics
 functions, cryptography and multimedia related functionality (like
-enumerations for pixel and sample formats). It is not a library for
-code needed by both libavcodec and libavformat.
-
-The goals for this library is to be:
-
-@table @strong
-@item Modular
-It should have few interdependencies and the possibility of disabling individual
-parts during @command{./configure}.
-
-@item Small
-Both sources and objects should be small.
-
-@item Efficient
-It should have low CPU and memory usage.
-
-@item Useful
-It should avoid useless features that almost no one needs.
-@end table
+enumerations for pixel and sample formats).
 
 @c man end DESCRIPTION
 

doc/muxers.texi

@@ -23,8 +23,6 @@ A description of some of the currently available muxers follows.
 
 Audio Interchange File Format muxer.
 
-@subsection Options
-
 It accepts the following options:
 
 @table @option
@@ -51,10 +49,6 @@ The output of the muxer consists of a single line of the form:
 CRC=0x@var{CRC}, where @var{CRC} is a hexadecimal number 0-padded to
 8 digits containing the CRC for all the decoded input frames.
 
-See also the @ref{framecrc} muxer.
-
-@subsection Examples
-
 For example to compute the CRC of the input, and store it in the file
 @file{out.crc}:
 @example
@@ -74,6 +68,8 @@ and the input video converted to MPEG-2 video, use the command:
 ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f crc -
 @end example
 
+See also the @ref{framecrc} muxer.
+
 @anchor{framecrc}
 @section framecrc
 
@@ -93,8 +89,6 @@ packet of the form:
 @var{CRC} is a hexadecimal number 0-padded to 8 digits containing the
 CRC of the packet.
 
-@subsection Examples
-
 For example to compute the CRC of the audio and video frames in
 @file{INPUT}, converted to raw audio and video packets, and store it
 in the file @file{out.crc}:
@@ -138,8 +132,6 @@ packet of the form:
 @var{MD5} is a hexadecimal number representing the computed MD5 hash
 for the packet.
 
-@subsection Examples
-
 For example to compute the MD5 of the audio and video frames in
 @file{INPUT}, converted to raw audio and video packets, and store it
 in the file @file{out.md5}:
@@ -154,93 +146,30 @@ ffmpeg -i INPUT -f framemd5 -
 
 See also the @ref{md5} muxer.
 
-@anchor{gif}
-@section gif
-
-Animated GIF muxer.
-
-It accepts the following options:
-
-@table @option
-@item loop
-Set the number of times to loop the output. Use @code{-1} for no loop, @code{0}
-for looping indefinitely (default).
-
-@item final_delay
-Force the delay (expressed in centiseconds) after the last frame. Each frame
-ends with a delay until the next frame. The default is @code{-1}, which is a
-special value to tell the muxer to re-use the previous delay. In case of a
-loop, you might want to customize this value to mark a pause for instance.
-@end table
-
-For example, to encode a gif looping 10 times, with a 5 seconds delay between
-the loops:
-@example
-ffmpeg -i INPUT -loop 10 -final_delay 500 out.gif
-@end example
-
-Note 1: if you wish to extract the frames in separate GIF files, you need to
-force the @ref{image2} muxer:
-@example
-ffmpeg -i INPUT -c:v gif -f image2 "out%d.gif"
-@end example
-
-Note 2: the GIF format has a very small time base: the delay between two frames
-can not be smaller than one centi second.
-
 @anchor{hls}
 @section hls
 
 Apple HTTP Live Streaming muxer that segments MPEG-TS according to
-the HTTP Live Streaming (HLS) specification.
+the HTTP Live Streaming specification.
 
 It creates a playlist file and numbered segment files. The output
 filename specifies the playlist filename; the segment filenames
 receive the same basename as the playlist, a sequential number and
 a .ts extension.
 
-For example, to convert an input file with @command{ffmpeg}:
 @example
 ffmpeg -i in.nut out.m3u8
 @end example
 
-See also the @ref{segment} muxer, which provides a more generic and
-flexible implementation of a segmenter, and can be used to perform HLS
-segmentation.
-
-@subsection Options
-
-This muxer supports the following options:
-
 @table @option
-@item hls_time @var{seconds}
-Set the segment length in seconds. Default value is 2.
-
-@item hls_list_size @var{size}
-Set the maximum number of playlist entries. If set to 0 the list file
-will contain all the segments. Default value is 5.
-
-@item hls_wrap @var{wrap}
-Set the number after which the segment filename number (the number
-specified in each segment file) wraps. If set to 0 the number will be
-never wrapped. Default value is 0.
-
-This option is useful to avoid to fill the disk with many segment
-files, and limits the maximum number of segment files written to disk
-to @var{wrap}.
-
-@item start_number @var{number}
-Start the playlist sequence number from @var{number}. Default value is
-0.
-
-@item hls_base_url @var{baseurl}
-Append @var{baseurl} to every entry in the playlist.
-Useful to generate playlists with absolute paths.
-
-Note that the playlist sequence number must be unique for each segment
-and it is not to be confused with the segment filename sequence number
-which can be cyclic, for example if the @option{wrap} option is
-specified.
+@item -hls_time @var{seconds}
+Set the segment length in seconds.
+@item -hls_list_size @var{size}
+Set the maximum number of playlist entries.
+@item -hls_wrap @var{wrap}
+Set the number after which index wraps.
+@item -start_number @var{number}
+Start the sequence from @var{number}.
 @end table
 
 @anchor{ico}
@@ -306,8 +235,6 @@ The pattern "img%%-%d.jpg" will specify a sequence of filenames of the
 form @file{img%-1.jpg}, @file{img%-2.jpg}, ..., @file{img%-10.jpg},
 etc.
 
-@subsection Examples
-
 The following example shows how to use @command{ffmpeg} for creating a
 sequence of files @file{img-001.jpeg}, @file{img-002.jpeg}, ...,
 taking one image every second from the input video:
@@ -330,32 +257,16 @@ Note also that the pattern must not necessarily contain "%d" or
 ffmpeg -i in.avi -f image2 -frames:v 1 img.jpeg
 @end example
 
-The @option{strftime} option allows you to expand the filename with
-date and time information. Check the documentation of
-the @code{strftime()} function for the syntax.
-
-For example to generate image files from the @code{strftime()}
-"%Y-%m-%d_%H-%M-%S" pattern, the following @command{ffmpeg} command
-can be used:
-@example
-ffmpeg -f v4l2 -r 1 -i /dev/video0 -f image2 -strftime 1 "%Y-%m-%d_%H-%M-%S.jpg"
-@end example
-
-@subsection Options
-
 @table @option
-@item start_number
-Start the sequence from the specified number. Default value is 1. Must
-be a non-negative number.
+@item start_number @var{number}
+Start the sequence from @var{number}. Default value is 1. Must be a
+non-negative number.
 
-@item update
-If set to 1, the filename will always be interpreted as just a
-filename, not a pattern, and the corresponding file will be continuously
-overwritten with new images. Default value is 0.
+@item -update @var{number}
+If @var{number} is nonzero, the filename will always be interpreted as just a
+filename, not a pattern, and this file will be continuously overwritten with new
+images.
 
-@item strftime
-If set to 1, expand the filename with date and time information from
-@code{strftime()}. Default value is 0.
 @end table
 
 The image muxer supports the .Y.U.V image file format. This format is
@@ -370,27 +281,25 @@ Matroska container muxer.
 
 This muxer implements the matroska and webm container specs.
 
-@subsection Metadata
-
 The recognized metadata settings in this muxer are:
 
 @table @option
-@item title
-Set title name provided to a single track.
 
-@item language
-Specify the language of the track in the Matroska languages form.
+@item title=@var{title name}
+Name provided to a single track
+@end table
 
-The language can be either the 3 letters bibliographic ISO-639-2 (ISO
-639-2/B) form (like "fre" for French), or a language code mixed with a
-country code for specialities in languages (like "fre-ca" for Canadian
-French).
+@table @option
 
-@item stereo_mode
-Set stereo 3D video layout of two views in a single video track.
+@item language=@var{language name}
+Specifies the language of the track in the Matroska languages form
+@end table
 
-The following values are recognized:
-@table @samp
+@table @option
+
+@item stereo_mode=@var{mode}
+Stereo 3D video layout of two views in a single video track
+@table @option
 @item mono
 video is not stereo
 @item left_right
@@ -429,11 +338,10 @@ For example a 3D WebM clip can be created using the following command line:
 ffmpeg -i sample_left_right_clip.mpg -an -c:v libvpx -metadata stereo_mode=left_right -y stereo_clip.webm
 @end example
 
-@subsection Options
-
 This muxer supports the following options:
 
 @table @option
+
 @item reserve_index_space
 By default, this muxer writes the index for seeking (called cues in Matroska
 terms) at the end of the file, because it cannot know in advance how much space
@@ -448,6 +356,7 @@ for most use cases should be about 50kB per hour of video.
 
 Note that cues are only written if the output is seekable and this option will
 have no effect if it is not.
+
 @end table
 
 @anchor{md5}
@@ -477,9 +386,7 @@ ffmpeg -i INPUT -f md5 -
 
 See also the @ref{framemd5} muxer.
 
-@section mov, mp4, ismv
-
-MOV/MP4/ISMV (Smooth Streaming) muxer.
+@section MOV/MP4/ISMV
 
 The mov/mp4/ismv muxer supports fragmentation. Normally, a MOV/MP4
 file has all the metadata about all packets stored in one location
@@ -495,8 +402,6 @@ very long files (since writing normal MOV/MP4 files stores info about
 every single packet in memory until the file is closed). The downside
 is that it is less compatible with other applications.
 
-@subsection Options
-
 Fragmentation is enabled by setting one of the AVOptions that define
 how to cut the file into fragments:
 
@@ -551,16 +456,8 @@ This operation can take a while, and will not work in various situations such
 as fragmented output, thus it is not enabled by default.
 @item -movflags rtphint
 Add RTP hinting tracks to the output file.
-@item -movflags disable_chpl
-Disable Nero chapter markers (chpl atom).  Normally, both Nero chapters
-and a QuickTime chapter track are written to the file. With this option
-set, only the QuickTime chapter track will be written. Nero chapters can
-cause failures when the file is reprocessed with certain tagging programs, like
-mp3Tag 2.61a and iTunes 11.3, most likely other versions are affected as well.
 @end table
 
-@subsection Example
-
 Smooth Streaming content can be pushed in real time to a publishing
 point on IIS with this muxer. Example:
 @example
@@ -571,15 +468,12 @@ ffmpeg -re @var{<normal input/transcoding options>} -movflags isml+frag_keyframe
 
 The MP3 muxer writes a raw MP3 stream with an ID3v2 header at the beginning and
 optionally an ID3v1 tag at the end. ID3v2.3 and ID3v2.4 are supported, the
-@code{id3v2_version} option controls which one is used. Setting
-@code{id3v2_version} to 0 will disable the ID3v2 header completely. The legacy
-ID3v1 tag is not written by default, but may be enabled with the
-@code{write_id3v1} option.
+@code{id3v2_version} option controls which one is used. The legacy ID3v1 tag is
+not written by default, but may be enabled with the @code{write_id3v1} option.
 
-The muxer may also write a Xing frame at the beginning, which contains the
-number of frames in the file. It is useful for computing duration of VBR files.
-The Xing frame is written if the output stream is seekable and if the
-@code{write_xing} option is set to 1 (the default).
+For seekable output the muxer also writes a Xing frame at the beginning, which
+contains the number of frames in the file. It is useful for computing duration
+of VBR files.
 
 The muxer supports writing ID3v2 attached pictures (APIC frames). The pictures
 are supplied to the muxer in form of a video stream with a single packet. There
@@ -606,24 +500,12 @@ ffmpeg -i input.mp3 -i cover.png -c copy -map 0 -map 1
 -metadata:s:v title="Album cover" -metadata:s:v comment="Cover (Front)" out.mp3
 @end example
 
-Write a "clean" MP3 without any extra features:
-@example
-ffmpeg -i input.wav -write_xing 0 -id3v2_version 0 out.mp3
-@end example
-
 @section mpegts
 
 MPEG transport stream muxer.
 
 This muxer implements ISO 13818-1 and part of ETSI EN 300 468.
 
-The recognized metadata settings in mpegts muxer are @code{service_provider}
-and @code{service_name}. If they are not set the default for
-@code{service_provider} is "FFmpeg" and the default for
-@code{service_name} is "Service01".
-
-@subsection Options
-
 The muxer options are:
 
 @table @option
@@ -643,10 +525,7 @@ Set the first PID for data packets (default 0x0100, max 0x0f00).
 @item -mpegts_m2ts_mode @var{number}
 Enable m2ts mode if set to 1. Default value is -1 which disables m2ts mode.
 @item -muxrate @var{number}
-Set a constant muxrate (default VBR).
-@item -pcr_period @var{numer}
-Override the default PCR retransmission time (default 20ms), ignored
-if variable muxrate is selected.
+Set muxrate.
 @item -pes_payload_size @var{number}
 Set minimum PES packet payload in bytes.
 @item -mpegts_flags @var{flags}
@@ -679,7 +558,10 @@ Reemit PAT/PMT before writing the next packet.
 Use LATM packetization for AAC.
 @end table
 
-@subsection Example
+The recognized metadata settings in mpegts muxer are @code{service_provider}
+and @code{service_name}. If they are not set the default for
+@code{service_provider} is "FFmpeg" and the default for
+@code{service_name} is "Service01".
 
 @example
 ffmpeg -i file.mpg -c copy \
@@ -715,30 +597,6 @@ Alternatively you can write the command as:
 ffmpeg -benchmark -i INPUT -f null -
 @end example
 
-@section nut
-
-@table @option
-@item -syncpoints @var{flags}
-Change the syncpoint usage in nut:
-@table @option
-@item @var{default} use the normal low-overhead seeking aids.
-@item @var{none} do not use the syncpoints at all, reducing the overhead but making the stream non-seekable;
-    Use of this option is not recommended, as the resulting files are very damage
-    sensitive and seeking is not possible. Also in general the overhead from
-    syncpoints is negligible. Note, -@code{write_index} 0 can be used to disable
-    all growing data tables, allowing to mux endless streams with limited memory
-    and wihout these disadvantages.
-@item @var{timestamped} extend the syncpoint with a wallclock field.
-@end table
-The @var{none} and @var{timestamped} flags are experimental.
-@item -write_index @var{bool}
-Write index at the end, the default is to write an index.
-@end table
-
-@example
-ffmpeg -i INPUT -f_strict experimental -syncpoints none - | processor
-@end example
-
 @section ogg
 
 Ogg container muxer.
@@ -754,12 +612,11 @@ situations, giving a small seek granularity at the cost of additional container
 overhead.
 @end table
 
-@anchor{segment}
 @section segment, stream_segment, ssegment
 
 Basic stream segmenter.
 
-This muxer outputs streams to a number of separate files of nearly
+The segmenter muxer outputs streams to a number of separate files of nearly
 fixed duration. Output filename pattern can be set in a fashion similar to
 @ref{image2}.
 
@@ -781,21 +638,14 @@ The segment muxer works best with a single constant frame rate video.
 
 Optionally it can generate a list of the created segments, by setting
 the option @var{segment_list}. The list type is specified by the
-@var{segment_list_type} option. The entry filenames in the segment
-list are set by default to the basename of the corresponding segment
-files.
-
-See also the @ref{hls} muxer, which provides a more specific
-implementation for HLS segmentation.
-
-@subsection Options
+@var{segment_list_type} option.
 
 The segment muxer supports the following options:
 
 @table @option
 @item reference_stream @var{specifier}
 Set the reference stream, as specified by the string @var{specifier}.
-If @var{specifier} is set to @code{auto}, the reference is chosen
+If @var{specifier} is set to @code{auto}, the reference is choosen
 automatically. Otherwise it must be a stream specifier (see the ``Stream
 specifiers'' chapter in the ffmpeg manual) which specifies the
 reference stream. The default value is @code{auto}.
@@ -804,11 +654,6 @@ reference stream. The default value is @code{auto}.
 Override the inner container format, by default it is guessed by the filename
 extension.
 
-@item segment_format_options @var{options_list}
-Set output format options using a :-separated list of key=value
-parameters. Values containing the @code{:} special character must be
-escaped.
-
 @item segment_list @var{name}
 Generate also a listfile named @var{name}. If not specified no
 listfile is generated.
@@ -825,21 +670,15 @@ Allow caching (only affects M3U8 list files).
 Allow live-friendly file generation.
 @end table
 
-@item segment_list_type @var{type}
-Select the listing format.
-@table @option
-@item @var{flat} use a simple flat list of entries.
-@item @var{hls} use a m3u8-like structure.
-@end table
+Default value is @code{samp}.
 
 @item segment_list_size @var{size}
-Update the list file so that it contains at most @var{size}
+Update the list file so that it contains at most the last @var{size}
 segments. If 0 the list file will contain all the segments. Default
 value is 0.
 
-@item segment_list_entry_prefix @var{prefix}
-Prepend @var{prefix} to each entry. Useful to generate absolute paths.
-By default no prefix is applied.
+@item segment_list_type @var{type}
+Specify the format for the segment list file.
 
 The following values are recognized:
 @table @samp
@@ -890,16 +729,6 @@ Note that splitting may not be accurate, unless you force the
 reference stream key-frames at the given time. See the introductory
 notice and the examples below.
 
-@item segment_atclocktime @var{1|0}
-If set to "1" split at regular clock time intervals starting from 00:00
-o'clock. The @var{time} value specified in @option{segment_time} is
-used for setting the length of the splitting interval.
-
-For example with @option{segment_time} set to "900" this makes it possible
-to create files at 12:00 o'clock, 12:15, 12:30, etc.
-
-Default value is "0".
-
 @item segment_time_delta @var{delta}
 Specify the accuracy time when selecting the start time for a
 segment, expressed as a duration specification. Default value is "0".
@@ -919,7 +748,7 @@ In particular may be used in combination with the @file{ffmpeg} option
 @var{force_key_frames} may not be set accurately because of rounding
 issues, with the consequence that a key frame time may result set just
 before the specified time. For constant frame rate videos a value of
-1/(2*@var{frame_rate}) should address the worst case mismatch between
+1/2*@var{frame_rate} should address the worst case mismatch between
 the specified time and the time set by @var{force_key_frames}.
 
 @item segment_times @var{times}
@@ -956,7 +785,7 @@ argument must be a time duration specification, and defaults to 0.
 
 @itemize
 @item
-Remux the content of file @file{in.mkv} to a list of segments
+To remux the content of file @file{in.mkv} to a list of segments
 @file{out-000.nut}, @file{out-001.nut}, etc., and write the list of
 generated segments to @file{out.list}:
 @example
@@ -964,20 +793,14 @@ ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.list out%03d.nu
 @end example
 
 @item
-Segment input and set output format options for the output segments:
-@example
-ffmpeg -i in.mkv -f segment -segment_time 10 -segment_format_options movflags=+faststart out%03d.mp4
-@end example
-
-@item
-Segment the input file according to the split points specified by the
-@var{segment_times} option:
+As the example above, but segment the input file according to the split
+points specified by the @var{segment_times} option:
 @example
 ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_times 1,2,3,5,8,13,21 out%03d.nut
 @end example
 
 @item
-Use the @command{ffmpeg} @option{force_key_frames}
+As the example above, but use the @command{ffmpeg} @option{force_key_frames}
 option to force key frames in the input at the specified location, together
 with the segment option @option{segment_time_delta} to account for
 possible roundings operated when setting key frame times.
@@ -996,7 +819,7 @@ ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_fr
 @end example
 
 @item
-Convert the @file{in.mkv} to TS segments using the @code{libx264}
+To convert the @file{in.mkv} to TS segments using the @code{libx264}
 and @code{libfaac} encoders:
 @example
 ffmpeg -i in.mkv -map 0 -codec:v libx264 -codec:a libfaac -f ssegment -segment_list out.list out%03d.ts
@@ -1011,28 +834,6 @@ ffmpeg -re -i in.mkv -codec copy -map 0 -f segment -segment_list playlist.m3u8 \
 @end example
 @end itemize
 
-@section smoothstreaming
-
-Smooth Streaming muxer generates a set of files (Manifest, chunks) suitable for serving with conventional web server.
-
-@table @option
-@item window_size
-Specify the number of fragments kept in the manifest. Default 0 (keep all).
-
-@item extra_window_size
-Specify the number of fragments kept outside of the manifest before removing from disk. Default 5.
-
-@item lookahead_count
-Specify the number of lookahead fragments. Default 2.
-
-@item min_frag_duration
-Specify the minimum fragment duration (in microseconds). Default 5000000.
-
-@item remove_at_exit
-Specify whether to remove all fragments when finished. Default 0 (do not remove).
-
-@end table
-
 @section tee
 
 The tee muxer can be used to write the same data to several files or any
@@ -1048,8 +849,8 @@ to feed the same packets to several muxers directly.
 The slave outputs are specified in the file name given to the muxer,
 separated by '|'. If any of the slave name contains the '|' separator,
 leading or trailing spaces or any special character, it must be
-escaped (see @ref{quoting_and_escaping,,the "Quoting and escaping"
-section in the ffmpeg-utils(1) manual,ffmpeg-utils}).
+escaped (see the ``Quoting and escaping'' section in the ffmpeg-utils
+manual).
 
 Muxer options can be specified for each slave by prepending them as a list of
 @var{key}=@var{value} pairs separated by ':', between square brackets. If
@@ -1064,13 +865,10 @@ output name suffix.
 
 @item bsfs[/@var{spec}]
 Specify a list of bitstream filters to apply to the specified
-output.
-
-It is possible to specify to which streams a given bitstream filter
-applies, by appending a stream specifier to the option separated by
-@code{/}. @var{spec} must be a stream specifier (see @ref{Format
-stream specifiers}).  If the stream specifier is not specified, the
-bitstream filters will be applied to all streams in the output.
+output. It is possible to specify to which streams a given bitstream
+filter applies, by appending a stream specifier to the option
+separated by @code{/}. If the stream specifier is not specified, the
+bistream filters will be applied to all streams in the output.
 
 Several bitstream filters can be specified, separated by ",".
 
@@ -1080,8 +878,7 @@ specified by a stream specifier. If not specified, this defaults to
 all the input streams.
 @end table
 
-@subsection Examples
-
+Some examples follow.
 @itemize
 @item
 Encode something and both archive it in a WebM file and stream it
@@ -1102,49 +899,10 @@ audio packets.
 ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac -strict experimental
        -f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=a]out.aac"
 @end example
-
-@item
-As below, but select only stream @code{a:1} for the audio output. Note
-that a second level escaping must be performed, as ":" is a special
-character used to separate options.
-@example
-ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac -strict experimental
-       -f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=\'a:1\']out.aac"
-@end example
 @end itemize
 
 Note: some codecs may need different options depending on the output format;
 the auto-detection of this can not work with the tee muxer. The main example
 is the @option{global_header} flag.
 
-@section webm_dash_manifest
-
-WebM DASH Manifest muxer.
-
-This muxer implements the WebM DASH Manifest specification to generate the DASH manifest XML.
-
-@subsection Options
-
-This muxer supports the following options:
-
-@table @option
-@item adaptation_sets
-This option has the following syntax: "id=x,streams=a,b,c id=y,streams=d,e" where x and y are the
-unique identifiers of the adaptation sets and a,b,c,d and e are the indices of the corresponding
-audio and video streams. Any number of adaptation sets can be added using this option.
-@end table
-
-@subsection Example
-@example
-ffmpeg -f webm_dash_manifest -i video1.webm \
-       -f webm_dash_manifest -i video2.webm \
-       -f webm_dash_manifest -i audio1.webm \
-       -f webm_dash_manifest -i audio2.webm \
-       -map 0 -map 1 -map 2 -map 3 \
-       -c copy \
-       -f webm_dash_manifest \
-       -adaptation_sets "id=0,streams=0,1 id=1,streams=2,3" \
-       manifest.xml
-@end example
-
 @c man end MUXERS

doc/nut.texi

@@ -21,27 +21,6 @@ The official nut specification is at svn://svn.mplayerhq.hu/nut
 In case of any differences between this text and the official specification,
 the official specification shall prevail.
 
-@chapter Modes
-NUT has some variants signaled by using the flags field in its main header.
-
-@multitable @columnfractions .4 .4
-@item BROADCAST   @tab Extend the syncpoint to report the sender wallclock
-@item PIPE        @tab Omit completely the syncpoint
-@end multitable
-
-@section BROADCAST
-
-The BROADCAST variant provides a secondary time reference to facilitate
-detecting endpoint latency and network delays.
-It assumes all the endpoint clocks are syncronized.
-To be used in real-time scenarios.
-
-@section PIPE
-
-The PIPE variant assumes NUT is used as non-seekable intermediate container,
-by not using syncpoint removes unneeded overhead and reduces the overall
-memory usage.
-
 @chapter Container-specific codec tags
 
 @section Generic raw YUVA formats

doc/optimization.txt

@@ -79,6 +79,9 @@ qpel{8,16}_mc??_old_c / *pixels{8,16}_l4
     Just used to work around a bug in an old libavcodec encoder version.
     Don't optimize them.
 
+tpel_mc_func {put,avg}_tpel_pixels_tab
+    Used only for SVQ3, so only optimize them if you need fast SVQ3 decoding.
+
 add_bytes/diff_bytes
     For huffyuv only, optimize if you want a faster ffhuffyuv codec.
 
@@ -136,6 +139,9 @@ dct_unquantize_mpeg2
 dct_unquantize_h263
     Used in MPEG-4/H.263 en/decoding.
 
+FIXME remaining functions?
+BTW, most of these functions are in dsputil.c/.h, some are in mpegvideo.c/.h.
+
 
 
 Alignment:
@@ -262,6 +268,17 @@ CELL/SPU:
 http://www-01.ibm.com/chips/techlib/techlib.nsf/techdocs/30B3520C93F437AB87257060006FFE5E/$file/Language_Extensions_for_CBEA_2.4.pdf
 http://www-01.ibm.com/chips/techlib/techlib.nsf/techdocs/9F820A5FFA3ECE8C8725716A0062585F/$file/CBE_Handbook_v1.1_24APR2007_pub.pdf
 
+SPARC-specific:
+---------------
+SPARC Joint Programming Specification (JPS1): Commonality
+http://www.fujitsu.com/downloads/PRMPWR/JPS1-R1.0.4-Common-pub.pdf
+
+UltraSPARC III Processor User's Manual (contains instruction timings)
+http://www.sun.com/processors/manuals/USIIIv2.pdf
+
+VIS Whitepaper (contains optimization guidelines)
+http://www.sun.com/processors/vis/download/vis/vis_whitepaper.pdf
+
 GCC asm links:
 --------------
 official doc but quite ugly

doc/outdevs.texi

@@ -13,8 +13,8 @@ You can disable all the output devices using the configure option
 option "--enable-outdev=@var{OUTDEV}", or you can disable a particular
 input device using the option "--disable-outdev=@var{OUTDEV}".
 
-The option "-devices" of the ff* tools will display the list of
-enabled output devices.
+The option "-formats" of the ff* tools will display the list of
+enabled output devices (amongst the muxers).
 
 A description of the currently available output devices follows.
 
@@ -22,27 +22,11 @@ A description of the currently available output devices follows.
 
 ALSA (Advanced Linux Sound Architecture) output device.
 
-@subsection Examples
-
-@itemize
-@item
-Play a file on default ALSA device:
-@example
-ffmpeg -i INPUT -f alsa default
-@end example
-
-@item
-Play a file on soundcard 1, audio device 7:
-@example
-ffmpeg -i INPUT -f alsa hw:1,7
-@end example
-@end itemize
-
 @section caca
 
 CACA output device.
 
-This output device allows one to show a video stream in CACA window.
+This output device allows to show a video stream in CACA window.
 Only one CACA window is allowed per application, so you can
 have only one instance of this output device in an application.
 
@@ -120,68 +104,6 @@ ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_dither colors -
 @end example
 @end itemize
 
-@section decklink
-
-The decklink output device provides playback capabilities for Blackmagic
-DeckLink devices.
-
-To enable this output device, you need the Blackmagic DeckLink SDK and you
-need to configure with the appropriate @code{--extra-cflags}
-and @code{--extra-ldflags}.
-On Windows, you need to run the IDL files through @command{widl}.
-
-DeckLink is very picky about the formats it supports. Pixel format is always
-uyvy422, framerate and video size must be determined for your device with
-@command{-list_formats 1}. Audio sample rate is always 48 kHz.
-
-@subsection Options
-
-@table @option
-
-@item list_devices
-If set to @option{true}, print a list of devices and exit.
-Defaults to @option{false}.
-
-@item list_formats
-If set to @option{true}, print a list of supported formats and exit.
-Defaults to @option{false}.
-
-@item preroll
-Amount of time to preroll video in seconds.
-Defaults to @option{0.5}.
-
-@end table
-
-@subsection Examples
-
-@itemize
-
-@item
-List output devices:
-@example
-ffmpeg -i test.avi -f decklink -list_devices 1 dummy
-@end example
-
-@item
-List supported formats:
-@example
-ffmpeg -i test.avi -f decklink -list_formats 1 'DeckLink Mini Monitor'
-@end example
-
-@item
-Play video clip:
-@example
-ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 'DeckLink Mini Monitor'
-@end example
-
-@item
-Play video clip with non-standard framerate or video size:
-@example
-ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 -s 720x486 -r 24000/1001 'DeckLink Mini Monitor'
-@end example
-
-@end itemize
-
 @section fbdev
 
 Linux framebuffer output device.
@@ -211,45 +133,6 @@ ffmpeg -re -i INPUT -vcodec rawvideo -pix_fmt bgra -f fbdev /dev/fb0
 
 See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1).
 
-@section opengl
-OpenGL output device.
-
-To enable this output device you need to configure FFmpeg with @code{--enable-opengl}.
-
-This output device allows one to render to OpenGL context.
-Context may be provided by application or default SDL window is created.
-
-When device renders to external context, application must implement handlers for following messages:
-@code{AV_DEV_TO_APP_CREATE_WINDOW_BUFFER} - create OpenGL context on current thread.
-@code{AV_DEV_TO_APP_PREPARE_WINDOW_BUFFER} - make OpenGL context current.
-@code{AV_DEV_TO_APP_DISPLAY_WINDOW_BUFFER} - swap buffers.
-@code{AV_DEV_TO_APP_DESTROY_WINDOW_BUFFER} - destroy OpenGL context.
-Application is also required to inform a device about current resolution by sending @code{AV_APP_TO_DEV_WINDOW_SIZE} message.
-
-@subsection Options
-@table @option
-
-@item background
-Set background color. Black is a default.
-@item no_window
-Disables default SDL window when set to non-zero value.
-Application must provide OpenGL context and both @code{window_size_cb} and @code{window_swap_buffers_cb} callbacks when set.
-@item window_title
-Set the SDL window title, if not specified default to the filename specified for the output device.
-Ignored when @option{no_window} is set.
-@item window_size
-Set preferred window size, can be a string of the form widthxheight or a video size abbreviation.
-If not specified it defaults to the size of the input video, downscaled according to the aspect ratio.
-Mostly usable when @option{no_window} is not set.
-
-@end table
-
-@subsection Examples
-Play a file on SDL window using OpenGL rendering:
-@example
-ffmpeg  -i INPUT -f opengl "window title"
-@end example
-
 @section oss
 
 OSS (Open Sound System) output device.
@@ -281,33 +164,6 @@ by default it is set to the specified output name.
 Specify the device to use. Default device is used when not provided.
 List of output devices can be obtained with command @command{pactl list sinks}.
 
-@item buffer_size
-@item buffer_duration
-Control the size and duration of the PulseAudio buffer. A small buffer
-gives more control, but requires more frequent updates.
-
-@option{buffer_size} specifies size in bytes while
-@option{buffer_duration} specifies duration in milliseconds.
-
-When both options are provided then the highest value is used
-(duration is recalculated to bytes using stream parameters). If they
-are set to 0 (which is default), the device will use the default
-PulseAudio duration value. By default PulseAudio set buffer duration
-to around 2 seconds.
-
-@item prebuf
-Specify pre-buffering size in bytes. The server does not start with
-playback before at least @option{prebuf} bytes are available in the
-buffer. By default this option is initialized to the same value as
-@option{buffer_size} or @option{buffer_duration} (whichever is bigger).
-
-@item minreq
-Specify minimum request size in bytes. The server does not request less
-than @option{minreq} bytes from the client, instead waits until the buffer
-is free enough to request more bytes at once. It is recommended to not set
-this option, which will initialize this to a value that is deemed sensible
-by the server.
-
 @end table
 
 @subsection Examples
@@ -320,7 +176,7 @@ ffmpeg  -i INPUT -f pulse "stream name"
 
 SDL (Simple DirectMedia Layer) output device.
 
-This output device allows one to show a video stream in an SDL
+This output device allows to show a video stream in an SDL
 window. Only one SDL window is allowed per application, so you can
 have only one instance of this output device in an application.
 
@@ -350,17 +206,7 @@ downscaled according to the aspect ratio.
 
 @item window_fullscreen
 Set fullscreen mode when non-zero value is provided.
-Default value is zero.
-@end table
-
-@subsection Interactive commands
-
-The window created by the device can be controlled through the
-following interactive commands.
-
-@table @key
-@item q, ESC
-Quit the device immediately.
+Zero is a default.
 @end table
 
 @subsection Examples
@@ -379,7 +225,7 @@ sndio audio output device.
 
 XV (XVideo) output device.
 
-This output device allows one to show a video stream in a X Window System
+This output device allows to show a video stream in a X Window System
 window.
 
 @subsection Options
@@ -406,26 +252,19 @@ For example, @code{dual-headed:0.1} would specify screen 1 of display
 Check the X11 specification for more detailed information about the
 display name format.
 
-@item window_id
-When set to non-zero value then device doesn't create new window,
-but uses existing one with provided @var{window_id}. By default
-this options is set to zero and device creates its own window.
-
 @item window_size
 Set the created window size, can be a string of the form
 @var{width}x@var{height} or a video size abbreviation. If not
 specified it defaults to the size of the input video.
-Ignored when @var{window_id} is set.
 
 @item window_x
 @item window_y
 Set the X and Y window offsets for the created window. They are both
 set to 0 by default. The values may be ignored by the window manager.
-Ignored when @var{window_id} is set.
 
 @item window_title
 Set the window title, if not specified default to the filename
-specified for the output device. Ignored when @var{window_id} is set.
+specified for the output device.
 @end table
 
 For more information about XVideo see @url{http://www.x.org/}.

doc/platform.texi

@@ -24,20 +24,6 @@ If not, then you should install a different compiler that has no
 hard-coded path to gas. In the worst case pass @code{--disable-asm}
 to configure.
 
-@section Advanced linking configuration
-
-If you compiled FFmpeg libraries statically and you want to use them to
-build your own shared library, you may need to force PIC support (with
-@code{--enable-pic} during FFmpeg configure) and add the following option
-to your project LDFLAGS:
-
-@example
--Wl,-Bsymbolic
-@end example
-
-If your target platform requires position independent binaries, you should
-pass the correct linking flag (e.g. @code{-pie}) to @code{--extra-ldexeflags}.
-
 @section BSD
 
 BSD make will not build FFmpeg, you need to install and use GNU Make
@@ -65,15 +51,14 @@ The toolchain provided with Xcode is sufficient to build the basic
 unacelerated code.
 
 Mac OS X on PowerPC or ARM (iPhone) requires a preprocessor from
-@url{https://github.com/FFmpeg/gas-preprocessor} or
-@url{https://github.com/yuvi/gas-preprocessor}(currently outdated) to build the optimized
-assembly functions. Put the Perl script somewhere
+@url{http://github.com/yuvi/gas-preprocessor} to build the optimized
+assembler functions. Just download the Perl script and put it somewhere
 in your PATH, FFmpeg's configure will pick it up automatically.
 
 Mac OS X on amd64 and x86 requires @command{yasm} to build most of the
-optimized assembly functions. @uref{http://www.finkproject.org/, Fink},
+optimized assembler functions. @uref{http://www.finkproject.org/, Fink},
 @uref{http://www.gentoo.org/proj/en/gentoo-alt/prefix/bootstrap-macos.xml, Gentoo Prefix},
-@uref{https://mxcl.github.com/homebrew/, Homebrew}
+@uref{http://mxcl.github.com/homebrew/, Homebrew}
 or @uref{http://www.macports.org, MacPorts} can easily provide it.
 
 
@@ -123,16 +108,14 @@ libavformat) as DLLs.
 
 @section Microsoft Visual C++ or Intel C++ Compiler for Windows
 
-FFmpeg can be built with MSVC 2012 or earlier using a C99-to-C89 conversion utility
-and wrapper, or with MSVC 2013 and ICL natively.
+FFmpeg can be built with MSVC or ICL using a C99-to-C89 conversion utility and
+wrapper. For ICL, only the wrapper is used, since ICL supports C99.
 
 You will need the following prerequisites:
 
 @itemize
-@item @uref{https://github.com/libav/c99-to-c89/, C99-to-C89 Converter & Wrapper}
-(if using MSVC 2012 or earlier)
+@item @uref{http://download.videolan.org/pub/contrib/c99-to-c89/, C99-to-C89 Converter & Wrapper}
 @item @uref{http://code.google.com/p/msinttypes/, msinttypes}
-(if using MSVC 2012 or earlier)
 @item @uref{http://www.mingw.org/, MSYS}
 @item @uref{http://yasm.tortall.net/, YASM}
 @item @uref{http://gnuwin32.sourceforge.net/packages/bc.htm, bc for Windows} if
@@ -142,16 +125,14 @@ you want to run @uref{fate.html, FATE}.
 To set up a proper environment in MSYS, you need to run @code{msys.bat} from
 the Visual Studio or Intel Compiler command prompt.
 
-Place @code{yasm.exe} somewhere in your @code{PATH}. If using MSVC 2012 or
-earlier, place @code{c99wrap.exe} and @code{c99conv.exe} somewhere in your
-@code{PATH} as well.
+Place @code{makedef}, @code{c99wrap.exe}, @code{c99conv.exe}, and @code{yasm.exe}
+somewhere in your @code{PATH}.
 
-Next, make sure any other headers and libs you want to use, such as zlib, are
-located in a spot that the compiler can see. Do so by modifying the @code{LIB}
-and @code{INCLUDE} environment variables to include the @strong{Windows-style}
-paths to these directories. Alternatively, you can try and use the
-@code{--extra-cflags}/@code{--extra-ldflags} configure options. If using MSVC
-2012 or earlier, place @code{inttypes.h} somewhere the compiler can see too.
+Next, make sure @code{inttypes.h} and any other headers and libs you want to use
+are located in a spot that the compiler can see. Do so by modifying the @code{LIB}
+and @code{INCLUDE} environment variables to include the @strong{Windows} paths to
+these directories. Alternatively, you can try and use the
+@code{--extra-cflags}/@code{--extra-ldflags} configure options.
 
 Finally, run:
 
@@ -201,9 +182,7 @@ can see.
 @itemize
 @item Visual Studio 2010 Pro and Express
 @item Visual Studio 2012 Pro and Express
-@item Visual Studio 2013 Pro and Express
 @item Intel Composer XE 2013
-@item Intel Composer XE 2013 SP1
 @end itemize
 Anything else is not officially supported.
 
@@ -278,7 +257,7 @@ llrint() in its C library.
 Install your Cygwin with all the "Base" packages, plus the
 following "Devel" ones:
 @example
-binutils, gcc4-core, make, git, mingw-runtime, texinfo
+binutils, gcc4-core, make, git, mingw-runtime, texi2html
 @end example
 
 In order to run FATE you will also need the following "Utils" packages:

doc/protocols.texi

@@ -117,19 +117,7 @@ ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP/////////////
 
 File access protocol.
 
-Allow to read from or write to a file.
-
-A file URL can have the form:
-@example
-file:@var{filename}
-@end example
-
-where @var{filename} is the path of the file to read.
-
-An URL that does not have a protocol prefix will be assumed to be a
-file URL. Depending on the build, an URL that looks like a Windows
-path with the drive letter at the beginning will also be assumed to be
-a file URL (usually not the case in builds for unix-like systems).
+Allow to read from or read to a file.
 
 For example to read from a file @file{input.mpeg} with @command{ffmpeg}
 use the command:
@@ -137,6 +125,10 @@ use the command:
 ffmpeg -i file:input.mpeg output.mpeg
 @end example
 
+The ff* tools default to the file protocol, that is a resource
+specified with the name "FILE.mpeg" is interpreted as the URL
+"file:FILE.mpeg".
+
 This protocol accepts the following options:
 
 @table @option
@@ -166,7 +158,7 @@ This protocol accepts the following options.
 
 @table @option
 @item timeout
-Set timeout in microseconds of socket I/O operations used by the underlying low level
+Set timeout of socket I/O operations used by the underlying low level
 operation. By default it is set to -1, which means that the timeout is
 not specified.
 
@@ -213,7 +205,7 @@ m3u8 files.
 
 HTTP (Hyper Text Transfer Protocol).
 
-This protocol accepts the following options:
+This protocol accepts the following options.
 
 @table @option
 @item seekable
@@ -223,60 +215,51 @@ if set to -1 it will try to autodetect if it is seekable. Default
 value is -1.
 
 @item chunked_post
-If set to 1 use chunked Transfer-Encoding for posts, default is 1.
-
-@item content_type
-Set a specific content type for the POST messages.
+If set to 1 use chunked transfer-encoding for posts, default is 1.
 
 @item headers
 Set custom HTTP headers, can override built in default headers. The
 value must be a string encoding the headers.
 
+@item content_type
+Force a content type.
+
+@item user-agent
+Override User-Agent header. If not specified the protocol will use a
+string describing the libavformat build.
+
 @item multiple_requests
-Use persistent connections if set to 1, default is 0.
+Use persistent connections if set to 1. By default it is 0.
 
 @item post_data
 Set custom HTTP post data.
 
-@item user-agent
-@item user_agent
-Override the User-Agent header. If not specified the protocol will use a
-string describing the libavformat build. ("Lavf/<version>")
-
 @item timeout
-Set timeout in microseconds of socket I/O operations used by the underlying low level
+Set timeout of socket I/O operations used by the underlying low level
 operation. By default it is set to -1, which means that the timeout is
 not specified.
 
 @item mime_type
-Export the MIME type.
+Set MIME type.
 
 @item icy
 If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
 supports this, the metadata has to be retrieved by the application by reading
 the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
-The default is 1.
+The default is 0.
 
 @item icy_metadata_headers
-If the server supports ICY metadata, this contains the ICY-specific HTTP reply
-headers, separated by newline characters.
+If the server supports ICY metadata, this contains the ICY specific HTTP reply
+headers, separated with newline characters.
 
 @item icy_metadata_packet
 If the server supports ICY metadata, and @option{icy} was set to 1, this
-contains the last non-empty metadata packet sent by the server. It should be
-polled in regular intervals by applications interested in mid-stream metadata
-updates.
+contains the last non-empty metadata packet sent by the server.
 
 @item cookies
 Set the cookies to be sent in future requests. The format of each cookie is the
 same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
 delimited by a newline character.
-
-@item offset
-Set initial byte offset.
-
-@item end_offset
-Try to limit the request to bytes preceding this offset.
 @end table
 
 @subsection HTTP Cookies
@@ -293,50 +276,6 @@ The required syntax to play a stream specifying a cookie is:
 ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
 @end example
 
-@section Icecast
-
-Icecast protocol (stream to Icecast servers)
-
-This protocol accepts the following options:
-
-@table @option
-@item ice_genre
-Set the stream genre.
-
-@item ice_name
-Set the stream name.
-
-@item ice_description
-Set the stream description.
-
-@item ice_url
-Set the stream website URL.
-
-@item ice_public
-Set if the stream should be public.
-The default is 0 (not public).
-
-@item user_agent
-Override the User-Agent header. If not specified a string of the form
-"Lavf/<version>" will be used.
-
-@item password
-Set the Icecast mountpoint password.
-
-@item content_type
-Set the stream content type. This must be set if it is different from
-audio/mpeg.
-
-@item legacy_icecast
-This enables support for Icecast versions < 2.4.0, that do not support the
-HTTP PUT method but the SOURCE method.
-
-@end table
-
-@example
-icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint}
-@end example
-
 @section mmst
 
 MMS (Microsoft Media Server) protocol over TCP.
@@ -581,35 +520,6 @@ The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
 for streaming multimedia content within HTTPS requests to traverse
 firewalls.
 
-@section libsmbclient
-
-libsmbclient permits one to manipulate CIFS/SMB network resources.
-
-Following syntax is required.
-
-@example
-smb://[[domain:]user[:password@@]]server[/share[/path[/file]]]
-@end example
-
-This protocol accepts the following options.
-
-@table @option
-@item timeout
-Set timeout in miliseconds of socket I/O operations used by the underlying
-low level operation. By default it is set to -1, which means that the timeout
-is not specified.
-
-@item truncate
-Truncate existing files on write, if set to 1. A value of 0 prevents
-truncating. Default value is 1.
-
-@item workgroup
-Set the workgroup used for making connections. By default workgroup is not specified.
-
-@end table
-
-For more information see: @url{http://www.samba.org/}.
-
 @section libssh
 
 Secure File Transfer Protocol via libssh
@@ -634,10 +544,6 @@ is not specified.
 Truncate existing files on write, if set to 1. A value of 0 prevents
 truncating. Default value is 1.
 
-@item private_key
-Specify the path of the file containing private key to use during authorization.
-By default libssh searches for keys in the @file{~/.ssh/} directory.
-
 @end table
 
 Example: Play a file stored on remote server.
@@ -755,8 +661,6 @@ set to the the local RTP port value plus 1.
 
 @section rtsp
 
-Real-Time Streaming Protocol.
-
 RTSP is not technically a protocol handler in libavformat, it is a demuxer
 and muxer. The demuxer supports both normal RTSP (with data transferred
 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
@@ -764,29 +668,21 @@ data transferred over RDT).
 
 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
-@uref{https://github.com/revmischa/rtsp-server, RTSP server}).
+@uref{http://github.com/revmischa/rtsp-server, RTSP server}).
 
 The required syntax for a RTSP url is:
 @example
 rtsp://@var{hostname}[:@var{port}]/@var{path}
 @end example
 
-Options can be set on the @command{ffmpeg}/@command{ffplay} command
-line, or set in code via @code{AVOption}s or in
-@code{avformat_open_input}.
+The following options (set on the @command{ffmpeg}/@command{ffplay} command
+line, or set in code via @code{AVOption}s or in @code{avformat_open_input}),
+are supported:
 
-The following options are supported.
+Flags for @code{rtsp_transport}:
 
 @table @option
-@item initial_pause
-Do not start playing the stream immediately if set to 1. Default value
-is 0.
 
-@item rtsp_transport
-Set RTSP transport protocols.
-
-It accepts the following values:
-@table @samp
 @item udp
 Use UDP as lower transport protocol.
 
@@ -804,56 +700,15 @@ passing proxies.
 
 Multiple lower transport protocols may be specified, in that case they are
 tried one at a time (if the setup of one fails, the next one is tried).
-For the muxer, only the @samp{tcp} and @samp{udp} options are supported.
+For the muxer, only the @code{tcp} and @code{udp} options are supported.
 
-@item rtsp_flags
-Set RTSP flags.
+Flags for @code{rtsp_flags}:
 
-The following values are accepted:
-@table @samp
+@table @option
 @item filter_src
 Accept packets only from negotiated peer address and port.
 @item listen
 Act as a server, listening for an incoming connection.
-@item prefer_tcp
-Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
-@end table
-
-Default value is @samp{none}.
-
-@item allowed_media_types
-Set media types to accept from the server.
-
-The following flags are accepted:
-@table @samp
-@item video
-@item audio
-@item data
-@end table
-
-By default it accepts all media types.
-
-@item min_port
-Set minimum local UDP port. Default value is 5000.
-
-@item max_port
-Set maximum local UDP port. Default value is 65000.
-
-@item timeout
-Set maximum timeout (in seconds) to wait for incoming connections.
-
-A value of -1 means infinite (default). This option implies the
-@option{rtsp_flags} set to @samp{listen}.
-
-@item reorder_queue_size
-Set number of packets to buffer for handling of reordered packets.
-
-@item stimeout
-Set socket TCP I/O timeout in microseconds.
-
-@item user-agent
-Override User-Agent header. If not specified, it defaults to the
-libavformat identifier string.
 @end table
 
 When receiving data over UDP, the demuxer tries to reorder received packets
@@ -866,36 +721,36 @@ streams to display can be chosen with @code{-vst} @var{n} and
 @code{-ast} @var{n} for video and audio respectively, and can be switched
 on the fly by pressing @code{v} and @code{a}.
 
-@subsection Examples
+Example command lines:
 
-The following examples all make use of the @command{ffplay} and
-@command{ffmpeg} tools.
+To watch a stream over UDP, with a max reordering delay of 0.5 seconds:
 
-@itemize
-@item
-Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
 @example
 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
 @end example
 
-@item
-Watch a stream tunneled over HTTP:
+To watch a stream tunneled over HTTP:
+
 @example
 ffplay -rtsp_transport http rtsp://server/video.mp4
 @end example
 
-@item
-Send a stream in realtime to a RTSP server, for others to watch:
+To send a stream in realtime to a RTSP server, for others to watch:
+
 @example
 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
 @end example
 
-@item
-Receive a stream in realtime:
+To receive a stream in realtime:
+
 @example
 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
 @end example
-@end itemize
+
+@table @option
+@item stimeout
+Socket IO timeout in micro seconds.
+@end table
 
 @section sap
 
@@ -1033,65 +888,32 @@ this binary block are used as master key, the following 14 bytes are
 used as master salt.
 @end table
 
-@section subfile
-
-Virtually extract a segment of a file or another stream.
-The underlying stream must be seekable.
-
-Accepted options:
-@table @option
-@item start
-Start offset of the extracted segment, in bytes.
-@item end
-End offset of the extracted segment, in bytes.
-@end table
-
-Examples:
-
-Extract a chapter from a DVD VOB file (start and end sectors obtained
-externally and multiplied by 2048):
-@example
-subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
-@end example
-
-Play an AVI file directly from a TAR archive:
-subfile,,start,183241728,end,366490624,,:archive.tar
-
 @section tcp
 
-Transmission Control Protocol.
+Trasmission Control Protocol.
 
 The required syntax for a TCP url is:
 @example
 tcp://@var{hostname}:@var{port}[?@var{options}]
 @end example
 
-@var{options} contains a list of &-separated options of the form
-@var{key}=@var{val}.
-
-The list of supported options follows.
-
 @table @option
-@item listen=@var{1|0}
-Listen for an incoming connection. Default value is 0.
+
+@item listen
+Listen for an incoming connection
 
 @item timeout=@var{microseconds}
-Set raise error timeout, expressed in microseconds.
+In read mode: if no data arrived in more than this time interval, raise error.
+In write mode: if socket cannot be written in more than this time interval, raise error.
+This also sets timeout on TCP connection establishing.
 
-This option is only relevant in read mode: if no data arrived in more
-than this time interval, raise error.
-
-@item listen_timeout=@var{microseconds}
-Set listen timeout, expressed in microseconds.
-@end table
-
-The following example shows how to setup a listening TCP connection
-with @command{ffmpeg}, which is then accessed with @command{ffplay}:
 @example
 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
 ffplay tcp://@var{hostname}:@var{port}
 @end example
 
+@end table
+
 @section tls
 
 Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
@@ -1157,7 +979,7 @@ ffplay tls://@var{hostname}:@var{port}
 
 User Datagram Protocol.
 
-The required syntax for an UDP URL is:
+The required syntax for a UDP url is:
 @example
 udp://@var{hostname}:@var{port}[?@var{options}]
 @end example
@@ -1165,17 +987,17 @@ udp://@var{hostname}:@var{port}[?@var{options}]
 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
 
 In case threading is enabled on the system, a circular buffer is used
-to store the incoming data, which allows one to reduce loss of data due to
+to store the incoming data, which allows to reduce loss of data due to
 UDP socket buffer overruns. The @var{fifo_size} and
 @var{overrun_nonfatal} options are related to this buffer.
 
 The list of supported options follows.
 
 @table @option
+
 @item buffer_size=@var{size}
-Set the UDP maximum socket buffer size in bytes. This is used to set either
-the receive or send buffer size, depending on what the socket is used for.
-Default is 64KB.  See also @var{fifo_size}.
+Set the UDP socket buffer size in bytes. This is used both for the
+receiving and the sending buffer size.
 
 @item localport=@var{port}
 Override the local UDP port to bind with.
@@ -1222,40 +1044,25 @@ Survive in case of UDP receiving circular buffer overrun. Default
 value is 0.
 
 @item timeout=@var{microseconds}
-Set raise error timeout, expressed in microseconds.
-
-This option is only relevant in read mode: if no data arrived in more
-than this time interval, raise error.
-
-@item broadcast=@var{1|0}
-Explicitly allow or disallow UDP broadcasting.
-
-Note that broadcasting may not work properly on networks having
-a broadcast storm protection.
+In read mode: if no data arrived in more than this time interval, raise error.
 @end table
 
-@subsection Examples
+Some usage examples of the UDP protocol with @command{ffmpeg} follow.
 
-@itemize
-@item
-Use @command{ffmpeg} to stream over UDP to a remote endpoint:
+To stream over UDP to a remote endpoint:
 @example
 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
 @end example
 
-@item
-Use @command{ffmpeg} to stream in mpegts format over UDP using 188
-sized UDP packets, using a large input buffer:
+To stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer:
 @example
 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
 @end example
 
-@item
-Use @command{ffmpeg} to receive over UDP from a remote endpoint:
+To receive over UDP from a remote endpoint:
 @example
-ffmpeg -i udp://[@var{multicast-address}]:@var{port} ...
+ffmpeg -i udp://[@var{multicast-address}]:@var{port}
 @end example
-@end itemize
 
 @section unix
 

doc/scaler.texi

@@ -35,7 +35,7 @@ Select nearest neighbor rescaling algorithm.
 @item area
 Select averaging area rescaling algorithm.
 
-@item bicublin
+@item bicubiclin
 Select bicubic scaling algorithm for the luma component, bilinear for
 chroma components.
 
@@ -112,14 +112,6 @@ bayer dither
 
 @item ed
 error diffusion dither
-
-@item a_dither
-arithmetic dither, based using addition
-
-@item x_dither
-arithmetic dither, based using xor (more random/less apparent patterning that
-a_dither).
-
 @end table
 
 @end table

doc/snow.txt

@@ -618,6 +618,7 @@ flip wavelet?
 try to use the wavelet transformed predicted image (motion compensated image) as context for coding the residual coefficients
 try the MV length as context for coding the residual coefficients
 use extradata for stuff which is in the keyframes now?
+the MV median predictor is patented IIRC
 implement per picture halfpel interpolation
 try different range coder state transition tables for different contexts
 

doc/soc.txt

@@ -0,0 +1,24 @@
+Google Summer of Code and similar project guidelines
+
+Summer of Code is a project by Google in which students are paid to implement
+some nice new features for various participating open source projects ...
+
+This text is a collection of things to take care of for the next soc as
+it's a little late for this year's soc (2006).
+
+The Goal:
+Our goal in respect to soc is and must be of course exactly one thing and
+that is to improve FFmpeg, to reach this goal, code must
+* conform to the development policy and patch submission guidelines
+* must improve FFmpeg somehow (faster, smaller, "better",
+  more codecs supported, fewer bugs, cleaner, ...)
+
+for mentors and other developers to help students to reach that goal it is
+essential that changes to their codebase are publicly visible, clean and
+easy reviewable that again leads us to:
+* use of a revision control system like git
+* separation of cosmetic from non-cosmetic changes (this is almost entirely
+  ignored by mentors and students in soc 2006 which might lead to a surprise
+  when the code will be reviewed at the end before a possible inclusion in
+  FFmpeg, individual changes were generally not reviewable due to cosmetics).
+* frequent commits, so that comments can be provided early

doc/t2h.init

@@ -1,35 +1,26 @@
-# Init file for texi2html.
-
-# This is deprecated, and the makeinfo/texi2any version is doc/t2h.pm
-
 # no horiz rules between sections
 $end_section = \&FFmpeg_end_section;
 sub FFmpeg_end_section($$)
 {
 }
 
-my $TEMPLATE_HEADER1 = $ENV{"FFMPEG_HEADER1"} || <<EOT;
-<!DOCTYPE html>
-<html lang="en">
-  <head>
-    <meta charset="utf-8" />
-    <meta http-equiv="X-UA-Compatible" content="IE=edge" />
-    <title>FFmpeg documentation</title>
-    <link rel="stylesheet" href="bootstrap.min.css" />
-    <link rel="stylesheet" href="style.min.css" />
+$EXTRA_HEAD =
+'<link rel="icon" href="favicon.png" type="image/png" />
+';
+
+$CSS_LINES = $ENV{"FFMPEG_CSS"} || <<EOT;
+<link rel="stylesheet" type="text/css" href="default.css" />
 EOT
 
-my $TEMPLATE_HEADER2 = $ENV{"FFMPEG_HEADER2"} || <<EOT;
-  </head>
-  <body>
-    <div style="width: 95%; margin: auto">
+my $TEMPLATE_HEADER = $ENV{"FFMPEG_HEADER"} || <<EOT;
+<link rel="icon" href="favicon.png" type="image/png" />
+</head>
+<body>
+<div id="container">
+<div id="body">
 EOT
 
-my $TEMPLATE_FOOTER = $ENV{"FFMPEG_FOOTER"} || <<EOT;
-    </div>
-  </body>
-</html>
-EOT
+$PRE_BODY_CLOSE = '</div></div>';
 
 $SMALL_RULE = '';
 $BODYTEXT = '';
@@ -91,25 +82,21 @@ sub FFmpeg_print_page_head($$)
     $longtitle = "FFmpeg documentation : " . $longtitle;
 
     print $fh <<EOT;
-$TEMPLATE_HEADER1
-$description
-<meta name="keywords" content="$longtitle">
-<meta name="Generator" content="$Texi2HTML::THISDOC{program}">
+<!DOCTYPE html>
+<html>
 $Texi2HTML::THISDOC{'copying'}<!-- Created on $Texi2HTML::THISDOC{today} by $Texi2HTML::THISDOC{program} -->
 <!--
 $Texi2HTML::THISDOC{program_authors}
 -->
-$encoding
-$TEMPLATE_HEADER2
-EOT
-}
+<head>
+<title>$longtitle</title>
 
-$print_page_foot = \&FFmpeg_print_page_foot;
-sub FFmpeg_print_page_foot($$)
-{
-    my $fh = shift;
-    print $fh <<EOT;
-$TEMPLATE_FOOTER
+$description
+<meta name="keywords" content="$longtitle">
+<meta name="Generator" content="$Texi2HTML::THISDOC{program}">
+$encoding
+$CSS_LINES
+$TEMPLATE_HEADER
 EOT
 }
 

doc/t2h.pm

@@ -1,221 +0,0 @@
-# makeinfo HTML output init file
-#
-# Copyright (c) 2011, 2012 Free Software Foundation, Inc.
-# Copyright (c) 2014 Andreas Cadhalpun
-# Copyright (c) 2014 Tiancheng "Timothy" Gu
-#
-# This file is part of FFmpeg.
-#
-# FFmpeg is free software; you can redistribute it and/or modify
-# it under the terms of the GNU General Public License as published by
-# the Free Software Foundation; either version 3 of the License, or
-# (at your option) any later version.
-#
-# 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
-# General Public License for more details.
-#
-# You should have received a copy of the GNU 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
-
-# no navigation elements
-set_from_init_file('HEADERS', 0);
-
-# TOC and Chapter headings link
-set_from_init_file('TOC_LINKS', 1);
-
-# print the TOC where @contents is used
-set_from_init_file('INLINE_CONTENTS', 1);
-
-# make chapters <h2>
-set_from_init_file('CHAPTER_HEADER_LEVEL', 2);
-
-# Do not add <hr>
-set_from_init_file('DEFAULT_RULE', '');
-set_from_init_file('BIG_RULE', '');
-
-# Customized file beginning
-sub ffmpeg_begin_file($$$)
-{
-    my $self = shift;
-    my $filename = shift;
-    my $element = shift;
-
-    my $command;
-    if ($element and $self->get_conf('SPLIT')) {
-        $command = $self->element_command($element);
-    }
-
-    my ($title, $description, $encoding, $date, $css_lines,
-        $doctype, $bodytext, $copying_comment, $after_body_open,
-        $extra_head, $program_and_version, $program_homepage,
-        $program, $generator) = $self->_file_header_informations($command);
-
-    my $links = $self->_get_links ($filename, $element);
-
-    my $head1 = $ENV{"FFMPEG_HEADER1"} || <<EOT;
-<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
-<html>
-<!-- Created by $program_and_version, $program_homepage -->
-  <head>
-    <meta charset="utf-8">
-    <title>
-EOT
-    my $head_title = <<EOT;
-      $title
-EOT
-
-    my $head2 = $ENV{"FFMPEG_HEADER2"} || <<EOT;
-    </title>
-    <link rel="stylesheet" type="text/css" href="bootstrap.min.css">
-    <link rel="stylesheet" type="text/css" href="style.min.css">
-  </head>
-  <body>
-    <div style="width: 95%; margin: auto">
-      <h1>
-EOT
-
-    my $head3 = $ENV{"FFMPEG_HEADER3"} || <<EOT;
-      </h1>
-EOT
-
-    return $head1 . $head_title . $head2 . $head_title . $head3;
-}
-texinfo_register_formatting_function('begin_file', \&ffmpeg_begin_file);
-
-# Customized file ending
-sub ffmpeg_end_file($)
-{
-    my $self = shift;
-    my $program_string = &{$self->{'format_program_string'}}($self);
-    my $program_text = <<EOT;
-      <p style="font-size: small;">
-        $program_string
-      </p>
-EOT
-    my $footer = $ENV{FFMPEG_FOOTER} || <<EOT;
-    </div>
-  </body>
-</html>
-EOT
-    return $program_text . $footer;
-}
-texinfo_register_formatting_function('end_file', \&ffmpeg_end_file);
-
-# Dummy title command
-# Ignore title. Title is handled through ffmpeg_begin_file().
-set_from_init_file('USE_TITLEPAGE_FOR_TITLE', 1);
-sub ffmpeg_title($$$$)
-{
-    return '';
-}
-
-texinfo_register_command_formatting('titlefont',
-                                    \&ffmpeg_title);
-
-# Customized float command. Part of code borrowed from GNU Texinfo.
-sub ffmpeg_float($$$$$)
-{
-    my $self = shift;
-    my $cmdname = shift;
-    my $command = shift;
-    my $args = shift;
-    my $content = shift;
-
-    my ($caption, $prepended) = Texinfo::Common::float_name_caption($self,
-                                                                $command);
-    my $caption_text = '';
-    my $prepended_text;
-    my $prepended_save = '';
-
-    if ($self->in_string()) {
-        if ($prepended) {
-            $prepended_text = $self->convert_tree_new_formatting_context(
-                $prepended, 'float prepended');
-        } else {
-            $prepended_text = '';
-        }
-        if ($caption) {
-            $caption_text = $self->convert_tree_new_formatting_context(
-                {'contents' => $caption->{'args'}->[0]->{'contents'}},
-                'float caption');
-        }
-        return $prepended.$content.$caption_text;
-    }
-
-    my $id = $self->command_id($command);
-    my $label;
-    if (defined($id) and $id ne '') {
-        $label = "<a name=\"$id\"></a>";
-    } else {
-        $label = '';
-    }
-
-    if ($prepended) {
-        if ($caption) {
-            # prepend the prepended tree to the first paragraph
-            my @caption_original_contents = @{$caption->{'args'}->[0]->{'contents'}};
-            my @caption_contents;
-            my $new_paragraph;
-            while (@caption_original_contents) {
-                my $content = shift @caption_original_contents;
-                if ($content->{'type'} and $content->{'type'} eq 'paragraph') {
-                    %{$new_paragraph} = %{$content};
-                    $new_paragraph->{'contents'} = [@{$content->{'contents'}}];
-                    unshift (@{$new_paragraph->{'contents'}}, {'cmdname' => 'strong',
-                             'args' => [{'type' => 'brace_command_arg',
-                                                    'contents' => [$prepended]}]});
-                    push @caption_contents, $new_paragraph;
-                    last;
-                } else {
-                    push @caption_contents, $content;
-                }
-            }
-            push @caption_contents, @caption_original_contents;
-            if ($new_paragraph) {
-                $caption_text = $self->convert_tree_new_formatting_context(
-                 {'contents' => \@caption_contents}, 'float caption');
-                $prepended_text = '';
-            }
-        }
-        if ($caption_text eq '') {
-            $prepended_text = $self->convert_tree_new_formatting_context(
-                $prepended, 'float prepended');
-            if ($prepended_text ne '') {
-                $prepended_save = $prepended_text;
-                $prepended_text = '<p><strong>'.$prepended_text.'</strong></p>';
-            }
-        }
-    } else {
-        $prepended_text = '';
-    }
-
-    if ($caption and $caption_text eq '') {
-        $caption_text = $self->convert_tree_new_formatting_context(
-            $caption->{'args'}->[0], 'float caption');
-    }
-    if ($prepended_text.$caption_text ne '') {
-        $prepended_text = $self->_attribute_class('div','float-caption'). '>'
-                . $prepended_text;
-        $caption_text .= '</div>';
-    }
-    my $html_class = '';
-    if ($prepended_save =~ /NOTE/) {
-        $html_class = 'info';
-        $prepended_text = '';
-        $caption_text   = '';
-    } elsif ($prepended_save =~ /IMPORTANT/) {
-        $html_class = 'warning';
-        $prepended_text = '';
-        $caption_text   = '';
-    }
-    return $self->_attribute_class('div', $html_class). '>' . "\n" .
-        $prepended_text . $caption_text . $content . '</div>';
-}
-
-texinfo_register_command_formatting('float',
-                                    \&ffmpeg_float);
-
-1;

doc/texi2pod.pl

@@ -1,4 +1,4 @@
-#!/usr/bin/env perl
+#! /usr/bin/perl
 
 #   Copyright (C) 1999, 2000, 2001 Free Software Foundation, Inc.
 
@@ -282,14 +282,6 @@ INF: while(<$inf>) {
         $_ = "\n=over 4\n";
     };
 
-    /^\@(multitable)\s+{.*/ and do {
-        push @endwstack, $endw;
-        push @icstack, $ic;
-        $endw = $1;
-        $ic = "";
-        $_ = "\n=over 4\n";
-    };
-
     /^\@((?:small)?example|display)/ and do {
         push @endwstack, $endw;
         $endw = $1;
@@ -306,10 +298,10 @@ INF: while(<$inf>) {
 
     /^\@tab\s+(.*\S)\s*$/ and $endw eq "multitable" and do {
         my $columns = $1;
-        $columns =~ s/\@tab//;
+        $columns =~ s/\@tab/ : /;
 
-        $_ = $columns;
-        $chapter =~ s/$//;
+        $_ = " : ". $columns;
+        $chapter =~ s/\n+\s+$//;
     };
 
     /^\@itemx?\s*(.+)?$/ and do {
@@ -332,9 +324,6 @@ $inf = pop @instack;
 
 die "No filename or title\n" unless defined $fn && defined $tl;
 
-# always use utf8
-print "=encoding utf8\n\n";
-
 $chapters{NAME} = "$fn \- $tl\n";
 $chapters{FOOTNOTES} .= "=back\n" if exists $chapters{FOOTNOTES};
 
@@ -388,8 +377,8 @@ sub postprocess
     s/\(?\@xref\{(?:[^\}]*)\}(?:[^.<]|(?:<[^<>]*>))*\.\)?//g;
     s/\s+\(\@pxref\{(?:[^\}]*)\}\)//g;
     s/;\s+\@pxref\{(?:[^\}]*)\}//g;
-    s/\@ref\{(?:[^,\}]*,)(?:[^,\}]*,)([^,\}]*).*\}/B<$1>/g;
-    s/\@ref\{([^\}]*)\}/B<$1>/g;
+    s/\@ref\{(?:[^,\}]*,)(?:[^,\}]*,)([^,\}]*).*\}/$1/g;
+    s/\@ref\{([^\}]*)\}/$1/g;
     s/\@noindent\s*//g;
     s/\@refill//g;
     s/\@gol//g;

doc/texidep.pl

@@ -1,32 +0,0 @@
-#! /usr/bin/env perl
-
-# This script will print the dependency of a Texinfo file to stdout.
-# texidep.pl <src-path> <input.texi> <output.ext>
-
-use warnings;
-use strict;
-
-die unless @ARGV == 3;
-
-my ($src_path, $root, $target) = @ARGV;
-
-sub print_deps {
-    my ($file, $deps) = @_;
-    $deps->{$file} = 1;
-
-    open(my $fh, "<", "$file") or die "Cannot open file '$file': $!";
-    while (<$fh>) {
-        if (my ($i) = /^\@(?:verbatim)?include\s+(\S+)/) {
-            die "Circular dependency found in file $root\n" if exists $deps->{"doc/$1"};
-            print "$target: doc/$1\n";
-
-            # skip looking for config.texi dependencies, since it has
-            # none, and is not located in the source tree
-            if ("$1" ne "config.texi") {
-                print_deps("$src_path/doc/$1", {%$deps});
-            }
-        }
-    }
-}
-
-print_deps($root, {});

doc/utils.texi

@@ -782,9 +782,6 @@ large numbers (usually 2^53 and larger).
 Round the value of expression @var{expr} upwards to the nearest
 integer. For example, "ceil(1.5)" is "2.0".
 
-@item clip(x, min, max)
-Return the value of @var{x} clipped between @var{min} and @var{max}.
-
 @item cos(x)
 Compute cosine of @var{x}.
 
@@ -1034,7 +1031,7 @@ indication of the corresponding powers of 10 and of 2.
 10^24 / 2^70
 @end table
 
-@c man end EXPRESSION EVALUATION
+@c man end
 
 @chapter OpenCL Options
 @c man begin OPENCL OPTIONS
@@ -1054,13 +1051,13 @@ See reference "OpenCL Specification Version: 1.2 chapter 5.6.4".
 Select the index of the platform to run OpenCL code.
 
 The specified index must be one of the indexes in the device list
-which can be obtained with @code{ffmpeg -opencl_bench} or @code{av_opencl_get_device_list()}.
+which can be obtained with @code{av_opencl_get_device_list()}.
 
 @item device_idx
 Select the index of the device used to run OpenCL code.
 
-The specified index must be one of the indexes in the device list which
-can be obtained with @code{ffmpeg -opencl_bench} or @code{av_opencl_get_device_list()}.
+The specifed index must be one of the indexes in the device list which
+can be obtained with @code{av_opencl_get_device_list()}.
 
 @end table
 

doc/viterbi.txt

@@ -0,0 +1,109 @@
+This is a quick description of the viterbi aka dynamic programing
+algorthm.
+
+Its reason for existence is that wikipedia has become very poor on
+describing algorithms in a way that makes it useable for understanding
+them or anything else actually. It tends now to describe the very same
+algorithm under 50 different names and pages with few understandable
+by even people who fully understand the algorithm and the theory behind.
+
+Problem description: (that is what it can solve)
+assume we have a 2d table, or you could call it a graph or matrix if you
+prefer
+
+    O   O   O   O   O   O   O
+
+    O   O   O   O   O   O   O
+
+    O   O   O   O   O   O   O
+
+    O   O   O   O   O   O   O
+
+
+That table has edges connecting points from each column to the next column
+and each edge has a score like: (only some edge and scores shown to keep it
+readable)
+
+
+    O--5--O-----O-----O-----O-----O
+     2   / 7   / \   / \   / \   /
+      \ /   \ /   \ /   \ /   \ /
+    O7-/--O--/--O--/--O--/--O--/--O
+     \/ \/ 1/ \/ \/ \/ \/ \/ \/ \/
+     /\ /\ 2\ /\ /\ /\ /\ /\ /\ /\
+    O3-/--O--/--O--/--O--/--O--/--O
+      / \   / \   / \   / \   / \
+     1   \ 9   \ /   \ /   \ /   \
+    O--2--O--1--O--5--O--3--O--8--O
+
+
+
+Our goal is to find a path from left to right through it which
+minimizes the sum of the score of all edges.
+(and of course left/right is just a convention here it could be top down too)
+Similarly the minimum could be the maximum by just fliping the sign,
+Example of a path with scores:
+
+    O   O   O   O   O   O   O
+
+>---O.  O   O  .O-2-O   O   O
+      5.     .7      .
+    O   O-1-O   O   O 8 O   O
+                       .
+    O   O   O   O   O   O-1-O---> (sum here is 24)
+
+
+The viterbi algorthm now solves this simply column by column
+For the previous column each point has a best path and a associated
+score:
+
+    O-----5     O
+     \
+      \
+    O  \  1     O
+        \/
+        /\
+    O  /  2     O
+      /
+     /
+    O-----2     O
+
+
+To move one column forward we just need to find the best path and associated
+scores for the next column
+here are some edges we could choose from:
+
+
+    O-----5--3--O
+     \      \8
+      \       \
+    O  \  1--9--O
+        \/  \3
+        /\     \
+    O  /  2--1--O
+      /     \2
+     /        \
+    O-----2--4--O
+
+Finding the new best paths and scores for each point of our new column is
+trivial given we know the previous column best paths and scores:
+
+    O-----0-----8
+     \
+      \
+    O  \  0----10
+        \/
+        /\
+    O  /  0-----3
+      /     \
+     /        \
+    O     0     4
+
+
+the viterbi algorthm continues exactly like this column for column until the
+end and then just picks the path with the best score (above that would be the
+one with score 3)
+
+
+Author: Michael niedermayer
+Copyright LGPL

doc/writing_filters.txt

@@ -1,424 +0,0 @@
-This document is a tutorial/initiation for writing simple filters in
-libavfilter.
-
-Foreword: just like everything else in FFmpeg, libavfilter is monolithic, which
-means that it is highly recommended that you submit your filters to the FFmpeg
-development mailing-list and make sure it is applied. Otherwise, your filter is
-likely to have a very short lifetime due to more a less regular internal API
-changes, and a limited distribution, review, and testing.
-
-Bootstrap
-=========
-
-Let's say you want to write a new simple video filter called "foobar" which
-takes one frame in input, changes the pixels in whatever fashion you fancy, and
-outputs the modified frame. The most simple way of doing this is to take a
-similar filter.  We'll pick edgedetect, but any other should do. You can look
-for others using the `./ffmpeg -v 0 -filters|grep ' V->V '` command.
-
- - cp libavfilter/vf_{edgedetect,foobar}.c
- - sed -i s/edgedetect/foobar/g -i libavfilter/vf_foobar.c
- - sed -i s/EdgeDetect/Foobar/g -i libavfilter/vf_foobar.c
- - edit libavfilter/Makefile, and add an entry for "foobar" following the
-   pattern of the other filters.
- - edit libavfilter/allfilters.c, and add an entry for "foobar" following the
-   pattern of the other filters.
- - ./configure ...
- - make -j<whatever> ffmpeg
- - ./ffmpeg -i tests/lena.pnm -vf foobar foobar.png
-
-If everything went right, you should get a foobar.png with Lena edge-detected.
-
-That's it, your new playground is ready.
-
-Some little details about what's going on:
-libavfilter/allfilters.c:avfilter_register_all() is called at runtime to create
-a list of the available filters, but it's important to know that this file is
-also parsed by the configure script, which in turn will define variables for
-the build system and the C:
-
-    --- after running configure ---
-
-    $ grep FOOBAR config.mak
-    CONFIG_FOOBAR_FILTER=yes
-    $ grep FOOBAR config.h
-    #define CONFIG_FOOBAR_FILTER 1
-
-CONFIG_FOOBAR_FILTER=yes from the config.mak is later used to enable the filter in
-libavfilter/Makefile and CONFIG_FOOBAR_FILTER=1 from the config.h will be used
-for registering the filter in libavfilter/allfilters.c.
-
-Filter code layout
-==================
-
-You now need some theory about the general code layout of a filter. Open your
-libavfilter/vf_foobar.c. This section will detail the important parts of the
-code you need to understand before messing with it.
-
-Copyright
----------
-
-First chunk is the copyright. Most filters are LGPL, and we are assuming
-vf_foobar is as well. We are also assuming vf_foobar is not an edge detector
-filter, so you can update the boilerplate with your credits.
-
-Doxy
-----
-
-Next chunk is the Doxygen about the file. See http://ffmpeg.org/doxygen/trunk/.
-Detail here what the filter is, does, and add some references if you feel like
-it.
-
-Context
--------
-
-Skip the headers and scroll down to the definition of FoobarContext. This is
-your local state context. It is already filled with 0 when you get it so do not
-worry about uninitialized read into this context. This is where you put every
-"global" information you need, typically the variable storing the user options.
-You'll notice the first field "const AVClass *class"; it's the only field you
-need to keep assuming you have a context. There are some magic you don't care
-about around this field, just let it be (in first position) for now.
-
-Options
--------
-
-Then comes the options array. This is what will define the user accessible
-options. For example, -vf foobar=mode=colormix:high=0.4:low=0.1. Most options
-have the following pattern:
-  name, description, offset, type, default value, minimum value, maximum value, flags
-
- - name is the option name, keep it simple, lowercase
- - description are short, in lowercase, without period, and describe what they
-   do, for example "set the foo of the bar"
- - offset is the offset of the field in your local context, see the OFFSET()
-   macro; the option parser will use that information to fill the fields
-   according to the user input
- - type is any of AV_OPT_TYPE_* defined in libavutil/opt.h
- - default value is an union where you pick the appropriate type; "{.dbl=0.3}",
-   "{.i64=0x234}", "{.str=NULL}", ...
- - min and max values define the range of available values, inclusive
- - flags are AVOption generic flags. See AV_OPT_FLAG_* definitions
-
-In doubt, just look at the other AVOption definitions all around the codebase,
-there are tons of examples.
-
-Class
------
-
-AVFILTER_DEFINE_CLASS(foobar) will define a unique foobar_class with some kind
-of signature referencing the options, etc. which will be referenced in the
-definition of the AVFilter.
-
-Filter definition
------------------
-
-At the end of the file, you will find foobar_inputs, foobar_outputs and
-the AVFilter ff_vf_foobar. Don't forget to update the AVFilter.description with
-a description of what the filter does, starting with a capitalized letter and
-ending with a period. You'd better drop the AVFilter.flags entry for now, and
-re-add them later depending on the capabilities of your filter.
-
-Callbacks
----------
-
-Let's now study the common callbacks. Before going into details, note that all
-these callbacks are explained in details in libavfilter/avfilter.h, so in
-doubt, refer to the doxy in that file.
-
-init()
-~~~~~~
-
-First one to be called is init(). It's flagged as cold because not called
-often. Look for "cold" on
-http://gcc.gnu.org/onlinedocs/gcc/Function-Attributes.html for more
-information.
-
-As the name suggests, init() is where you eventually initialize and allocate
-your buffers, pre-compute your data, etc. Note that at this point, your local
-context already has the user options initialized, but you still haven't any
-clue about the kind of data input you will get, so this function is often
-mainly used to sanitize the user options.
-
-Some init()s will also define the number of inputs or outputs dynamically
-according to the user options. A good example of this is the split filter, but
-we won't cover this here since vf_foobar is just a simple 1:1 filter.
-
-uninit()
-~~~~~~~~
-
-Similarly, there is the uninit() callback, doing what the name suggest. Free
-everything you allocated here.
-
-query_formats()
-~~~~~~~~~~~~~~~
-
-This is following the init() and is used for the format negotiation, basically
-where you say what pixel format(s) (gray, rgb 32, yuv 4:2:0, ...) you accept
-for your inputs, and what you can output. All pixel formats are defined in
-libavutil/pixfmt.h. If you don't change the pixel format between the input and
-the output, you just have to define a pixel formats array and call
-ff_set_common_formats(). For more complex negotiation, you can refer to other
-filters such as vf_scale.
-
-config_props()
-~~~~~~~~~~~~~~
-
-This callback is not necessary, but you will probably have one or more
-config_props() anyway. It's not a callback for the filter itself but for its
-inputs or outputs (they're called "pads" - AVFilterPad - in libavfilter's
-lexicon).
-
-Inside the input config_props(), you are at a point where you know which pixel
-format has been picked after query_formats(), and more information such as the
-video width and height (inlink->{w,h}). So if you need to update your internal
-context state depending on your input you can do it here. In edgedetect you can
-see that this callback is used to allocate buffers depending on these
-information. They will be destroyed in uninit().
-
-Inside the output config_props(), you can define what you want to change in the
-output. Typically, if your filter is going to double the size of the video, you
-will update outlink->w and outlink->h.
-
-filter_frame()
-~~~~~~~~~~~~~~
-
-This is the callback you are waiting from the beginning: it is where you
-process the received frames. Along with the frame, you get the input link from
-where the frame comes from.
-
-    static int filter_frame(AVFilterLink *inlink, AVFrame *in) { ... }
-
-You can get the filter context through that input link:
-
-    AVFilterContext *ctx = inlink->dst;
-
-Then access your internal state context:
-
-    FoobarContext *foobar = ctx->priv;
-
-And also the output link where you will send your frame when you are done:
-
-    AVFilterLink *outlink = ctx->outputs[0];
-
-Here, we are picking the first output. You can have several, but in our case we
-only have one since we are in a 1:1 input-output situation.
-
-If you want to define a simple pass-through filter, you can just do:
-
-    return ff_filter_frame(outlink, in);
-
-But of course, you probably want to change the data of that frame.
-
-This can be done by accessing frame->data[] and frame->linesize[].  Important
-note here: the width does NOT match the linesize. The linesize is always
-greater or equal to the width. The padding created should not be changed or
-even read. Typically, keep in mind that a previous filter in your chain might
-have altered the frame dimension but not the linesize. Imagine a crop filter
-that halves the video size: the linesizes won't be changed, just the width.
-
-    <-------------- linesize ------------------------>
-    +-------------------------------+----------------+ ^
-    |                               |                | |
-    |                               |                | |
-    |           picture             |    padding     | | height
-    |                               |                | |
-    |                               |                | |
-    +-------------------------------+----------------+ v
-    <----------- width ------------->
-
-Before modifying the "in" frame, you have to make sure it is writable, or get a
-new one. Multiple scenarios are possible here depending on the kind of
-processing you are doing.
-
-Let's say you want to change one pixel depending on multiple pixels (typically
-the surrounding ones) of the input. In that case, you can't do an in-place
-processing of the input so you will need to allocate a new frame, with the same
-properties as the input one, and send that new frame to the next filter:
-
-    AVFrame *out = ff_get_video_buffer(outlink, outlink->w, outlink->h);
-    if (!out) {
-        av_frame_free(&in);
-        return AVERROR(ENOMEM);
-    }
-    av_frame_copy_props(out, in);
-
-    // out->data[...] = foobar(in->data[...])
-
-    av_frame_free(&in);
-    return ff_filter_frame(outlink, out);
-
-In-place processing
-~~~~~~~~~~~~~~~~~~~
-
-If you can just alter the input frame, you probably just want to do that
-instead:
-
-    av_frame_make_writable(in);
-    // in->data[...] = foobar(in->data[...])
-    return ff_filter_frame(outlink, in);
-
-You may wonder why a frame might not be writable. The answer is that for
-example a previous filter might still own the frame data: imagine a filter
-prior to yours in the filtergraph that needs to cache the frame. You must not
-alter that frame, otherwise it will make that previous filter buggy. This is
-where av_frame_make_writable() helps (it won't have any effect if the frame
-already is writable).
-
-The problem with using av_frame_make_writable() is that in the worst case it
-will copy the whole input frame before you change it all over again with your
-filter: if the frame is not writable, av_frame_make_writable() will allocate
-new buffers, and copy the input frame data. You don't want that, and you can
-avoid it by just allocating a new buffer if necessary, and process from in to
-out in your filter, saving the memcpy. Generally, this is done following this
-scheme:
-
-    int direct = 0;
-    AVFrame *out;
-
-    if (av_frame_is_writable(in)) {
-        direct = 1;
-        out = in;
-    } else {
-        out = ff_get_video_buffer(outlink, outlink->w, outlink->h);
-        if (!out) {
-            av_frame_free(&in);
-            return AVERROR(ENOMEM);
-        }
-        av_frame_copy_props(out, in);
-    }
-
-    // out->data[...] = foobar(in->data[...])
-
-    if (!direct)
-        av_frame_free(&in);
-    return ff_filter_frame(outlink, out);
-
-Of course, this will only work if you can do in-place processing. To test if
-your filter handles well the permissions, you can use the perms filter. For
-example with:
-
-    -vf perms=random,foobar
-
-Make sure no automatic pixel conversion is inserted between perms and foobar,
-otherwise the frames permissions might change again and the test will be
-meaningless: add av_log(0,0,"direct=%d\n",direct) in your code to check that.
-You can avoid the issue with something like:
-
-    -vf format=rgb24,perms=random,foobar
-
-...assuming your filter accepts rgb24 of course. This will make sure the
-necessary conversion is inserted before the perms filter.
-
-Timeline
-~~~~~~~~
-
-Adding timeline support
-(http://ffmpeg.org/ffmpeg-filters.html#Timeline-editing) is often an easy
-feature to add. In the most simple case, you just have to add
-AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC to the AVFilter.flags. You can typically
-do this when your filter does not need to save the previous context frames, or
-basically if your filter just alter whatever goes in and doesn't need
-previous/future information. See for instance commit 86cb986ce that adds
-timeline support to the fieldorder filter.
-
-In some cases, you might need to reset your context somehow. This is handled by
-the AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL flag which is used if the filter
-must not process the frames but still wants to keep track of the frames going
-through (to keep them in cache for when it's enabled again). See for example
-commit 69d72140a that adds timeline support to the phase filter.
-
-Threading
-~~~~~~~~~
-
-libavfilter does not yet support frame threading, but you can add slice
-threading to your filters.
-
-Let's say the foobar filter has the following frame processing function:
-
-    dst = out->data[0];
-    src = in ->data[0];
-
-    for (y = 0; y < inlink->h; y++) {
-        for (x = 0; x < inlink->w; x++)
-            dst[x] = foobar(src[x]);
-        dst += out->linesize[0];
-        src += in ->linesize[0];
-    }
-
-The first thing is to make this function work into slices. The new code will
-look like this:
-
-    for (y = slice_start; y < slice_end; y++) {
-        for (x = 0; x < inlink->w; x++)
-            dst[x] = foobar(src[x]);
-        dst += out->linesize[0];
-        src += in ->linesize[0];
-    }
-
-The source and destination pointers, and slice_start/slice_end will be defined
-according to the number of jobs. Generally, it looks like this:
-
-    const int slice_start = (in->height *  jobnr   ) / nb_jobs;
-    const int slice_end   = (in->height * (jobnr+1)) / nb_jobs;
-    uint8_t       *dst = out->data[0] + slice_start * out->linesize[0];
-    const uint8_t *src =  in->data[0] + slice_start *  in->linesize[0];
-
-This new code will be isolated in a new filter_slice():
-
-    static int filter_slice(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) { ... }
-
-Note that we need our input and output frame to define slice_{start,end} and
-dst/src, which are not available in that callback. They will be transmitted
-through the opaque void *arg. You have to define a structure which contains
-everything you need:
-
-    typedef struct ThreadData {
-        AVFrame *in, *out;
-    } ThreadData;
-
-If you need some more information from your local context, put them here.
-
-In you filter_slice function, you access it like that:
-
-    const ThreadData *td = arg;
-
-Then in your filter_frame() callback, you need to call the threading
-distributor with something like this:
-
-    ThreadData td;
-
-    // ...
-
-    td.in  = in;
-    td.out = out;
-    ctx->internal->execute(ctx, filter_slice, &td, NULL, FFMIN(outlink->h, ctx->graph->nb_threads));
-
-    // ...
-
-    return ff_filter_frame(outlink, out);
-
-Last step is to add AVFILTER_FLAG_SLICE_THREADS flag to AVFilter.flags.
-
-For more example of slice threading additions, you can try to run git log -p
---grep 'slice threading' libavfilter/
-
-Finalization
-~~~~~~~~~~~~
-
-When your awesome filter is finished, you have a few more steps before you're
-done:
-
- - write its documentation in doc/filters.texi, and test the output with make
-   doc/ffmpeg-filters.html.
- - add a FATE test, generally by adding an entry in
-   tests/fate/filter-video.mak, add running make fate-filter-foobar GEN=1 to
-   generate the data.
- - add an entry in the Changelog
- - edit libavfilter/version.h and increase LIBAVFILTER_VERSION_MINOR by one
-   (and reset LIBAVFILTER_VERSION_MICRO to 100)
- - git add ... && git commit -m "avfilter: add foobar filter." && git format-patch -1
-
-When all of this is done, you can submit your patch to the ffmpeg-devel
-mailing-list for review.  If you need any help, feel free to come on our IRC
-channel, #ffmpeg-devel on irc.freenode.net.