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n0.6.6 ... v0.5

Author SHA1 Message Date
Diego Biurrun
f8429ed58c Peg libswscale to the revision corresponding to the moment the branch was cut.
Originally committed as revision 17887 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-03-08 22:13:48 +00:00
Robert Swain
df4763a782 Correct grammar in one sentence and add a note about doc/APIchanges
Originally committed as revision 17805 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-03-04 16:52:37 +00:00
Robert Swain
58af0caf04 Add some release notes for this 0.5 release branch
Originally committed as revision 17787 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-03-03 23:04:47 +00:00
Diego Biurrun
3aafe82485 Output 0.5 as version string.
Originally committed as revision 17754 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-03-03 12:47:47 +00:00
Baptiste Coudurier
07679e680c revert r16717, r16718, r16719, EAGAIN handling, this causes FFserver to hang
Originally committed as revision 17737 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-03-02 20:32:24 +00:00
Diego Biurrun
da835cc8a3 Create 0.5 release branch.
Originally committed as revision 17727 to svn://svn.ffmpeg.org/ffmpeg/branches/0.5
2009-03-02 08:32:29 +00:00
1797 changed files with 66352 additions and 195039 deletions

45
.gitignore vendored
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@@ -1,45 +0,0 @@
.config
.version
*.o
*.d
*.exe
*.ho
*-example
*-test
*_g
config.*
doc/*.1
doc/*.html
doc/*.pod
doxy
ffmpeg
ffplay
ffprobe
ffserver
libavcodec/libavcodec*
libavcore/libavcore*
libavdevice/libavdevice*
libavfilter/libavfilter*
libavformat/libavformat*
libavutil/avconfig.h
libavutil/libavutil*
libpostproc/libpostproc*
libswscale/libswscale*
tests/audiogen
tests/base64
tests/data
tests/rotozoom
tests/seek_test
tests/tiny_psnr
tests/videogen
tests/vsynth1
tests/vsynth2
tools/cws2fws
tools/graph2dot
tools/lavfi-showfiltfmts
tools/pktdumper
tools/probetest
tools/qt-faststart
tools/trasher
tools/trasher*.d
version.h

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@@ -1,674 +0,0 @@
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copy of the Program in return for a fee.
END OF TERMS AND CONDITIONS
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If you develop a new program, and you want it to be of the greatest
possible use to the public, the best way to achieve this is to make it
free software which everyone can redistribute and change under these terms.
To do so, attach the following notices to the program. It is safest
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it under the terms of the GNU General Public License as published by
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along with this program. If not, see <http://www.gnu.org/licenses/>.
Also add information on how to contact you by electronic and paper mail.
If the program does terminal interaction, make it output a short
notice like this when it starts in an interactive mode:
<program> Copyright (C) <year> <name of author>
This program comes with ABSOLUTELY NO WARRANTY; for details type `show w'.
This is free software, and you are welcome to redistribute it
under certain conditions; type `show c' for details.
The hypothetical commands `show w' and `show c' should show the appropriate
parts of the General Public License. Of course, your program's commands
might be different; for a GUI interface, you would use an "about box".
You should also get your employer (if you work as a programmer) or school,
if any, to sign a "copyright disclaimer" for the program, if necessary.
For more information on this, and how to apply and follow the GNU GPL, see
<http://www.gnu.org/licenses/>.
The GNU General Public License does not permit incorporating your program
into proprietary programs. If your program is a subroutine library, you
may consider it more useful to permit linking proprietary applications with
the library. If this is what you want to do, use the GNU Lesser General
Public License instead of this License. But first, please read
<http://www.gnu.org/philosophy/why-not-lgpl.html>.

View File

@@ -1,165 +0,0 @@
GNU LESSER GENERAL PUBLIC LICENSE
Version 3, 29 June 2007
Copyright (C) 2007 Free Software Foundation, Inc. <http://fsf.org/>
Everyone is permitted to copy and distribute verbatim copies
of this license document, but changing it is not allowed.
This version of the GNU Lesser General Public License incorporates
the terms and conditions of version 3 of the GNU General Public
License, supplemented by the additional permissions listed below.
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As used herein, "this License" refers to version 3 of the GNU Lesser
General Public License, and the "GNU GPL" refers to version 3 of the GNU
General Public License.
"The Library" refers to a covered work governed by this License,
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An "Application" is any work that makes use of an interface provided
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A "Combined Work" is a work produced by combining or linking an
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The "Minimal Corresponding Source" for a Combined Work means the
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The "Corresponding Application Code" for a Combined Work means the
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and utility programs needed for reproducing the Combined Work from the
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1. Exception to Section 3 of the GNU GPL.
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is a work based on the Library, and explaining where to find the
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of the GNU Lesser General Public License from time to time. Such new
versions will be similar in spirit to the present version, but may
differ in detail to address new problems or concerns.
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Library as you received it specifies that a certain numbered version
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conditions either of that published version or of any later version
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View File

@@ -7,10 +7,8 @@ Fabrice Bellard
Patrice Bensoussan
Alex Beregszaszi
BERO
Thilo Borgmann
Mario Brito
Ronald Bultje
Alex Converse
Maarten Daniels
Reimar Doeffinger
Tim Ferguson
@@ -39,7 +37,6 @@ Michael Niedermayer
François Revol
Peter Ross
Måns Rullgård
Stefano Sabatini
Roman Shaposhnik
Oded Shimon
Dieter Shirley

278
Changelog
View File

@@ -1,236 +1,8 @@
Entries are sorted chronologically from oldest to youngest within each release,
releases are sorted from youngest to oldest.
version 0.6.6:
- id3v2: fix skipping extended header in id3v2.4
- nsvdec: Fix use of uninitialized streams
- nsvdec: Be more careful with av_malloc()
- nsvdec: Propagate errors
- dv: check stype
- dv: Fix null pointer dereference due to ach=0
- dv: Fix small stack overread related to CVE-2011-3929 and CVE-2011-3936.
- atrac3: Fix crash in tonal component decoding
- mjpegbdec: Fix overflow in SOS
- kgv1dec: Increase offsets array size so it is large enough
- motionpixels: decode only the 111 complete frames for fate
- motionpixels: Clip YUV values after applying a gradient
- celp filters: Do not read earlier than the start of the 'out' vector
- vqavideo: return error if image size is not a multiple of block size (CVE-2012-0947)
- dpcm: ignore extra unpaired bytes in stereo streams (CVE-2011-3951)
- aacsbr: prevent out of bounds memcpy() (CVE-2012-0850)
- h264: Add check for invalid chroma_format_idc (CVE-2012-0851)
- adpcm: ADPCM Electronic Arts has always two channels (CVE-2012-0852)
- shorten: remove VLA and check for buffer overflow
- shorten: Prevent block size from increasing
- shorten: Fix out of bound writes in fix_bitshift()
- shorten: check for realloc failure (CVE-2012-0858)
- shorten: Use separate pointers for the allocated memory for decoded samples (CVE-2012-0858)
- kmvc: Check palsize (CVE-2011-3952)
- qdm2: clip array indices returned by qdm2_get_vlc()
- configure: properly check for mingw-w64 through installed headers
- mingw32: properly check if vfw capture is supported by the system headers
- mingw32: merge checks for mingw-w64 and mingw32-runtime >= 3.15 into one
- dxva2: define required feature selection macros
- png: check bit depth for PAL8/Y400A pixel formats
- ea: check chunk_size for validity
- tqi: Pass errors from the MB decoder
version 0.6.5:
- vorbis: An additional defense in the Vorbis codec. (CVE-2011-3895)
- vorbisdec: Fix decoding bug with channel handling.
- matroskadec: Fix a bug where a pointer was cached to an array that might
later move due to a realloc(). (CVE-2011-3893)
- vorbis: Avoid some out-of-bounds reads. (CVE-2011-3893)
- vp3: fix oob read for negative tokens and memleaks on error, (CVE-2011-3892)
- vp3: fix streams with non-zero last coefficient.
version 0.6.4:
- 4xm: Add a check in decode_i_frame to prevent buffer overreads
- wma: initialize prev_block_len_bits, next_block_len_bits, and block_len_bits.
- swscale: #include "libavutil/mathematics.h"
- vp3dec: Check coefficient index in vp3_dequant() (CVE-2011-4352)
- svq1dec: call avcodec_set_dimensions() after dimensions changed. (CVE-2011-4579)
- commits related to CVE-2011-4353:
- vp6: Fix illegal read.
- vp6: Reset the internal state when aborting key frames header parsing
- vp6: Check for huffman tree build errors
- vp6: partially propagate huffman tree building errors during coeff model parsing and fix misspelling
- commits related to CVE-2011-4351:
- qdm2: check output buffer size before decoding
- Fix qdm2 decoder packet handling to match the api
- Fix out of bound reads in the QDM2 decoder.
- Check for out of bound writes in the QDM2 decoder.
- vmd: fix segfaults on corruped streams (CVE-2011-4364)
- rv34: Check for invalid slice offsets
- rv34: Fix potential overreads
- rv34: Avoid NULL dereference on corrupted bitstream
- rv10: Reject slices that does not have the same type as the first one
- oggdec: fix out of bound write in the ogg demuxer
- smacker: fix a few off by 1 errors
- Check for invalid VLC value in smacker decoder.
- Check and propagate errors when VLC trees cannot be built in smacker decoder.
- Fixed off by one packet size allocation in the smacker demuxer.
- Check for invalid packet size in the smacker demuxer.
- ape demuxer: fix segfault on memory allocation failure.
- Fixed size given to init_get_bits() in xan decoder.
- smacker demuxer: handle possible av_realloc() failure.
- Fixed segfault with wavpack decoder on corrupted decorrelation terms sub-blocks.
- indeo2: fail if input buffer too small
- indeo2: init_get_bits size in bits instead of bytes
- wavpack: Check error codes rather than working around error conditions.
- Fixed invalid writes and reads in wavpack decoder on corrupted bitstreams.
- cpu detection: avoid a signed overflow
- h264: correct implicit weight table computation for long ref pics
- h264: correct the check for invalid long term frame index in MMCO decode
- rv10/20: tell decoder to use edge emulation
- flvenc: use int64_t to store offsets
- VC-1: fix reading of custom PAR.
- h264: notice memory allocation failure
- libx264: do not set pic quality if no frame is output
- mxfdec: Include FF_INPUT_BUFFER_PADDING_SIZE when allocating extradata.
- rv30: return AVERROR(EINVAL) instead of EINVAL
- Do not decode RV30 files if the extradata is too small
- aacps: skip some memcpy() if src and dst would be equal
- mpegts: fix Continuity Counter error detection
- alsa: fallback to buffer_size/4 for period_size.
- mxfenc: fix ignored drop flag in binary timecode representation.
- dca: set AVCodecContext frame_size for DTS audio
- H.264: fix overreads of qscale_table
- postprocess.c: filter name needs to be double 0 terminated
- Replace strncpy() with av_strlcpy() in libpostproc.
- jpegdec: actually search for and parse RSTn
- riff: Add mpgv MPEG-2 fourcc
- Added M701 codec_tag for mpeg2video
version 0.6.3:
- fix compilation with --enable-hardcoded-tables
- mjpeg: Detect overreads in mjpeg_decode_scan() and error out.
- aac: add headers needed for log2f()
- lavf: inspect more frames for fps when container time base is coarse
- AMV: disable DR1 and don't override EMU_EDGE
(addresses http://seclists.org/bugtraq/2011/Apr/257)
- Fix memory (re)allocation in matroskadec.c (MSVR11-011/CVE-2011-3504)
- Fix some crashes with invalid bitstreams in the CAVS decoder
(CVE-2011-3362, CVE-2011-3973, CVE-2011-3974)
- Compilation fixes for gcc-4.6, testsuite now passes again
- Fix a heap corruption issue in the OGG decoder
- Backported the Android VisualOn AAC encoder wrapper from 0.7.2
version 0.6.3:
- AMV: Fix possibly exploitable crash.
- Fix apparently exploitable race condition.
(addresses http://seclists.org/bugtraq/2011/Apr/257)
version 0.6.2:
- fix compilation with --enable-hardcoded-tables
- Fix invalid reads in VC-1 decoding (related to CVE-2011-0723)
- Do not attempt to decode APE file with no frames
(adresses http://packetstorm.linuxsecurity.com/1103-exploits/vlc105-dos.txt)
version 0.6.1:
- fix autodetection of E-AC-3 substream samples
- performance fix for seekable HTTP
- backport AAC-HE v2 from trunk
- add missing VP80 fourcc code for the VP8 codec
- small documentation fixes
- fix several potentially exploitable issues in the FLIC decoder
(addresses CVE-2010-3429)
version 0.6:
- PB-frame decoding for H.263
- deprecated vhook subsystem removed
- deprecated old scaler removed
- VQF demuxer
- alpha channel scaler
- PCX encoder
- RTP packetization of H.263
- RTP packetization of AMR
- RTP depacketization of Vorbis
- CorePNG decoding support
- Cook multichannel decoding support
- introduced avlanguage helpers in libavformat
- 8088flex TMV demuxer and decoder
- per-stream language-tags extraction in asfdec
- V210 decoder and encoder
- remaining GPL parts in AC-3 decoder converted to LGPL
- QCP demuxer
- SoX native format muxer and demuxer
- AMR-NB decoding/encoding, AMR-WB decoding via OpenCORE libraries
- DPX image decoder
- Electronic Arts Madcow decoder
- DivX (XSUB) subtitle encoder
- nonfree libamr support for AMR-NB/WB decoding/encoding removed
- experimental AAC encoder
- RTP depacketization of ASF and RTSP from WMS servers
- RTMP support in libavformat
- noX handling for OPT_BOOL X options
- Wave64 demuxer
- IEC-61937 compatible Muxer
- TwinVQ decoder
- Bluray (PGS) subtitle decoder
- LPCM support in MPEG-TS (HDMV RID as found on Blu-ray disks)
- WMA Pro decoder
- Core Audio Format demuxer
- Atrac1 decoder
- MD STUDIO audio demuxer
- RF64 support in WAV demuxer
- MPEG-4 Audio Lossless Coding (ALS) decoder
- -formats option split into -formats, -codecs, -bsfs, and -protocols
- IV8 demuxer
- CDG demuxer and decoder
- R210 decoder
- Auravision Aura 1 and 2 decoders
- Deluxe Paint Animation playback system
- SIPR decoder
- Adobe Filmstrip muxer and demuxer
- RTP depacketization of H.263
- Bink demuxer and audio/video decoders
- enable symbol versioning by default for linkers that support it
- IFF PBM/ILBM bitmap decoder
- concat protocol
- Indeo 5 decoder
- RTP depacketization of AMR
- WMA Voice decoder
- ffprobe tool
- AMR-NB decoder
- RTSP muxer
- HE-AAC v1 decoder
- Kega Game Video (KGV1) decoder
- VorbisComment writing for FLAC, Ogg FLAC and Ogg Speex files
- RTP depacketization of Theora
- HTTP Digest authentication
- RTMP/RTMPT/RTMPS/RTMPE/RTMPTE protocol support via librtmp
- Psygnosis YOP demuxer and video decoder
- spectral extension support in the E-AC-3 decoder
- unsharp video filter
- RTP hinting in the mov/3gp/mp4 muxer
- Dirac in Ogg demuxing
- seek to keyframes in Ogg
- 4:2:2 and 4:4:4 Theora decoding
- 35% faster VP3/Theora decoding
- faster AAC decoding
- faster H.264 decoding
- WebM support in Matroska de/muxer
- low overhead Ogg muxing
- VP8 de/encoding via libvpx
- CODEC_CAP_EXPERIMENTAL added
version 0.5:
- The "device" muxers and demuxers are now in a new libavdevice library
- DV50 AKA DVCPRO50 encoder, decoder, muxer and demuxer
- DV100 AKA DVCPRO HD decoder and demuxer
- TechSmith Camtasia (TSCC) video decoder
- IBM Ultimotion (ULTI) video decoder
- Sierra Online audio file demuxer and decoder
@@ -250,7 +22,7 @@ version 0.5:
- Fraps FPS1 video decoder
- Snow video encoder/decoder
- Sonic audio encoder/decoder
- Vorbis audio decoder
- Vorbis audio encoder/decoder
- Macromedia ADPCM decoder
- Duck TrueMotion 2 video decoder
- support for decoding FLX and DTA extensions in FLIC files
@@ -261,19 +33,18 @@ version 0.5:
- TrueSpeech audio decoder
- WMA2 audio decoder fixed, now all files should play correctly
- RealAudio 14.4 and 28.8 decoders fixed
- JPEG-LS decoder
- JPEG-LS encoder and decoder
- CamStudio video decoder
- build system improvements
- tabs and trailing whitespace removed from the codebase
- CamStudio video decoder
- AIFF/AIFF-C audio format, encoding and decoding
- ADTS AAC file reading and writing
- Creative VOC file reading and writing
- American Laser Games multimedia (*.mm) playback system
- Zip Motion Blocks Video decoder
- Zip Blocks Motion Video decoder and encoder
- improved Theora/VP3 decoder
- True Audio (TTA) decoder
- AVS demuxer and video decoder
- JPEG-LS encoder
- Smacker demuxer and decoder
- NuppelVideo/MythTV demuxer and RTjpeg decoder
- KMVC decoder
@@ -292,14 +63,12 @@ version 0.5:
- VP6 video decoder
- WavPack lossless audio decoder
- Targa (.TGA) picture decoder
- Vorbis audio encoder
- Delphine Software .cin demuxer/audio and video decoder
- Tiertex .seq demuxer/video decoder
- MTV demuxer
- TIFF picture encoder and decoder
- GIF picture decoder
- Intel Music Coder decoder
- Zip Motion Blocks Video encoder
- Musepack decoder
- Flash Screen Video encoder
- Theora encoding via libtheora
@@ -326,10 +95,10 @@ version 0.5:
- OS/2 support removed and restored again
- AC-3 decoder
- NUT muxer
- additional SPARC (VIS) optimizations
- Matroska muxer
- slice-based parallel H.264 decoding
- Monkey's Audio demuxer and decoder
- additional SPARC (VIS) optimizations
- AMV audio and video decoder
- DNxHD encoder
- H.264 PAFF decoding
@@ -338,7 +107,6 @@ version 0.5:
- libvorbis Vorbis decoding removed in favor of native decoder
- IntraX8 (J-Frame) subdecoder for WMV2 and VC-1
- Ogg (Theora, Vorbis and FLAC) muxer
- The "device" muxers and demuxers are now in a new libavdevice library
- PC Paintbrush PCX decoder
- Sun Rasterfile decoder
- TechnoTrend PVA demuxer
@@ -368,7 +136,6 @@ version 0.5:
- AAC decoder
- floating point PCM encoder/decoder
- MXF muxer
- DV100 AKA DVCPRO HD decoder and demuxer
- E-AC-3 support added to AC-3 decoder
- Nellymoser ASAO encoder
- ASS and SSA demuxer and muxer
@@ -376,9 +143,8 @@ version 0.5:
- SVQ3 watermark decoding support
- Speex decoding via libspeex
- Electronic Arts TGQ decoder
- RV40 decoder
- RV30 and RV40 decoder
- QCELP / PureVoice decoder
- RV30 decoder
- hybrid WavPack support
- R3D REDCODE demuxer
- ALSA support for playback and record
@@ -390,7 +156,6 @@ version 0.5:
- generic metadata API
version 0.4.9-pre1:
- DV encoder, DV muxer
@@ -474,8 +239,6 @@ version 0.4.9-pre1:
- many optimizations and bugfixes
- FunCom ISS audio file demuxer and according ADPCM decoding
version 0.4.8:
- MPEG-2 video encoding (Michael)
@@ -486,8 +249,6 @@ version 0.4.8:
- Interplay MVE playback subsystem (Mike Melanson)
- Duck DK3 and DK4 ADPCM audio decoders (Mike Melanson)
version 0.4.7:
- RealAudio 1.0 (14_4) and 2.0 (28_8) native decoders. Author unknown, code from mplayerhq
@@ -528,8 +289,6 @@ version 0.4.7:
.. And lots more new enhancements and fixes.
version 0.4.6:
- completely new integer only MPEG audio layer 1/2/3 decoder rewritten
@@ -566,8 +325,6 @@ version 0.4.6:
- simple stats output (Juanjo)
- 16-bit and 15-bit RGB/BGR/GBR support (Bisqwit)
version 0.4.5:
- some header fixes (Zdenek Kabelac <kabi at informatics.muni.cz>)
@@ -598,8 +355,6 @@ version 0.4.5:
- added MJPEG raw format (input/ouput)
- added JPEG image format support (input/output)
version 0.4.4:
- fixed some std header definitions (Bjorn Lindgren
@@ -624,8 +379,6 @@ version 0.4.4:
- codecs are no longer searched by read_header() (should fix ffserver
segfault)
version 0.4.3:
- BGR24 patch (initial patch by Jeroen Vreeken <pe1rxq at amsat.org>)
@@ -646,8 +399,6 @@ version 0.4.3:
<lists at reserv.at>, but changed completely since we use a format
instead of a protocol)
version 0.4.2:
- added H.263/MPEG-4/MSMPEG4 decoding support. MPEG-4 decoding support
@@ -664,7 +415,6 @@ version 0.4.2:
- added file overwrite confirmation (can be disabled with -y)
- added custom size picture to H.263 using H.263+ (Juanjo)
version 0.4.1:
- added MSMPEG4 (aka DivX) compatible encoder. Changed default codec
@@ -677,8 +427,6 @@ version 0.4.1:
- improved dct_quantize speed
- factorized some motion estimation code
version 0.4.0:
- removing grab code from ffserver and moved it to ffmpeg. Added
@@ -706,14 +454,10 @@ version 0.4.0:
- changed directories: libav for format handling, libavcodec for
codecs
version 0.3.4:
- added stereo in MPEG audio encoder
version 0.3.3:
- added 'high quality' mode which use motion vectors. It can be used in
@@ -721,21 +465,15 @@ version 0.3.3:
- fixed rounding problems which caused quality problems at high
bitrates and large GOP size
version 0.3.2: small fixes
- ASF fixes
- put_seek bug fix
version 0.3.1: added avi/divx support
- added AVI support
- added MPEG-4 codec compatible with OpenDivX. It is based on the H.263 codec
- added sound for flash format (not tested)
version 0.3: initial public release

View File

@@ -17,7 +17,7 @@
# The PROJECT_NAME tag is a single word (or a sequence of words surrounded
# by quotes) that should identify the project.
PROJECT_NAME = FFmpeg
PROJECT_NAME = ffmpeg
# The PROJECT_NUMBER tag can be used to enter a project or revision number.
# This could be handy for archiving the generated documentation or
@@ -370,7 +370,7 @@ EXCLUDE_SYMLINKS = NO
# EXCLUDE_PATTERNS tag to specify one or more wildcard patterns to exclude
# certain files from those directories.
EXCLUDE_PATTERNS = *.svn *.git
EXCLUDE_PATTERNS =
# The EXAMPLE_PATH tag can be used to specify one or more files or
# directories that contain example code fragments that are included (see
@@ -837,6 +837,7 @@ INCLUDE_FILE_PATTERNS =
PREDEFINED = __attribute__(x)="" "RENAME(x)=x ## _TMPL" "DEF(x)=x ## _TMPL" \
HAVE_AV_CONFIG_H HAVE_MMX HAVE_MMX2 HAVE_AMD3DNOW \
ATTR_ALIGN(x)=""
# 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.

50
LICENSE
View File

@@ -1,50 +0,0 @@
FFmpeg:
-------
Most files in FFmpeg are under the GNU Lesser General Public License version 2.1
or later (LGPL v2.1+). Read the file COPYING.LGPLv2.1 for details. Some other
files have MIT/X11/BSD-style licenses. In combination the LGPL v2.1+ applies to
FFmpeg.
Some optional parts of FFmpeg are licensed under the GNU General Public License
version 2 or later (GPL v2+). See the file COPYING.GPLv2 for details. None of
these parts are used by default, you have to explicitly pass --enable-gpl to
configure to activate them. In this case, FFmpeg's license changes to GPL v2+.
Specifically, the GPL parts of FFmpeg are
- libpostproc
- optional MMX optimizations for YUV to RGB colorspace conversion in
libswscale/x86/yuv2rgb_template.c
- optional x86 optimizations in the files
libavcodec/x86/h264_deblock_sse2.asm
libavcodec/x86/h264_idct_sse2.asm
libavcodec/x86/idct_mmx.c
- the X11 grabber in libavdevice/x11grab.c
There are a handful of files under other licensing terms, namely:
* The files libavcodec/jfdctfst.c, libavcodec/jfdctint.c, libavcodec/jrevdct.c
are taken from libjpeg, see the top of the files for licensing details.
Should you, for whatever reason, prefer to use version 3 of the (L)GPL, then
the configure parameter --enable-version3 will activate this licensing option
for you. Read the file COPYING.LGPLv3 or, if you have enabled GPL parts,
COPYING.GPLv3 to learn the exact legal terms that apply in this case.
external libraries:
-------------------
Some external libraries, e.g. libx264, are under GPL and can be used in
conjunction with FFmpeg. They require --enable-gpl to be passed to configure
as well.
The OpenCORE external libraries are under the Apache License 2.0. That license
is incompatible with the LGPL v2.1 and the GPL v2, but not with version 3 of
those licenses. So to combine the OpenCORE libraries with FFmpeg, the license
version needs to be upgraded by passing --enable-version3 to configure.
The nonfree external library libfaac can be hooked up in FFmpeg. You need to
pass --enable-nonfree to configure to enable it. Employ this option with care
as FFmpeg then becomes nonfree and unredistributable.

View File

@@ -18,6 +18,15 @@ Applications
ffmpeg:
ffmpeg.c Michael Niedermayer
Video Hooks:
vhook
vhook/watermark.c Marcus Engene
vhook/ppm.c
vhook/drawtext.c
vhook/fish.c
vhook/null.c
vhook/imlib2.c
ffplay:
ffplay.c Michael Niedermayer
@@ -28,7 +37,7 @@ Commandline utility code:
cmdutils.c, cmdutils.h Michael Niedermayer
QuickTime faststart:
tools/qt-faststart.c Baptiste Coudurier
qt-faststart.c Mike Melanson
Miscellaneous Areas
@@ -40,8 +49,6 @@ build system (configure,Makefiles) Diego Biurrun, Mans Rullgard
project server Diego Biurrun, Mans Rullgard
mailinglists Michael Niedermayer, Baptiste Coudurier
presets Robert Swain
metadata subsystem Aurelien Jacobs
release management Diego Biurrun, Reinhard Tartler
libavutil
@@ -98,25 +105,22 @@ Generic Parts:
simple_idct.c, simple_idct.h Michael Niedermayer
postprocessing:
libpostproc/* Michael Niedermayer
vdpau:
vdpau* Carl Eugen Hoyos
Codecs:
4xm.c Michael Niedermayer
8bps.c Roberto Togni
8svx.c Jaikrishnan Menon
aasc.c Kostya Shishkov
aac*, sbr.h Alex Converse
aac.[ch], aactab.[ch], aacdectab.h Robert Swain
ac3* Justin Ruggles
alacenc.c Jaikrishnan Menon
alsdec.c Thilo Borgmann
apedec.c Kostya Shishkov
asv* Michael Niedermayer
atrac3* Benjamin Larsson
bgmc.c, bgmc.h Thilo Borgmann
bink.c Kostya Shishkov
binkaudio.c Peter Ross
bmp.c Mans Rullgard, Kostya Shishkov
cavs* Stefan Gehrer
celp_filters.* Vitor Sessak
cinepak.c Roberto Togni
cljr Alex Beregszaszi
cook.c, cookdata.h Benjamin Larsson
@@ -128,7 +132,8 @@ Codecs:
dv.c Roman Shaposhnik
eacmv*, eaidct*, eat* Peter Ross
ffv1.c Michael Niedermayer
flac* Justin Ruggles
flacdec.c Alex Beregszaszi, Justin Ruggles
flacenc.c Justin Ruggles
flashsv* Benjamin Larsson
flicvideo.c Mike Melanson
g726.c Roman Shaposhnik
@@ -140,25 +145,18 @@ Codecs:
idcinvideo.c Mike Melanson
imc* Benjamin Larsson
indeo2* Kostya Shishkov
indeo5* Kostya Shishkov
interplayvideo.c Mike Melanson
ivi* Kostya Shishkov
jpeg_ls.c Kostya Shishkov
kmvc.c Kostya Shishkov
lcl*.c Roberto Togni, Reimar Doeffinger
lcl*.c Roberto Togni
libgsm.c Michel Bardiaux
libdirac* David Conrad
libopenjpeg.c Jaikrishnan Menon
libschroedinger* David Conrad
libspeexdec.c Justin Ruggles
libtheoraenc.c David Conrad
libx264.c Mans Rullgard, Jason Garrett-Glaser
loco.c Kostya Shishkov
lzo.h, lzo.c Reimar Doeffinger
mdec.c Michael Niedermayer
mimic.c Ramiro Polla
mjpeg.c Michael Niedermayer
mlp* Ramiro Polla
mmvideo.c Peter Ross
mpc* Kostya Shishkov
mpeg12.c, mpeg12data.h Michael Niedermayer
@@ -184,7 +182,6 @@ Codecs:
rv3* Kostya Shishkov
rv4* Kostya Shishkov
s3tc* Ivo van Poorten
smacker.c Kostya Shishkov
smc.c Mike Melanson
snow.c Michael Niedermayer, Loren Merritt
sonic.c Alex Beregszaszi
@@ -196,7 +193,7 @@ Codecs:
truemotion2* Kostya Shishkov
truespeech.c Kostya Shishkov
tscc.c Kostya Shishkov
tta.c Alex Beregszaszi, Jaikrishnan Menon
tta.c Alex Beregszaszi
txd.c Ivo van Poorten
ulti* Kostya Shishkov
vb.c Kostya Shishkov
@@ -210,8 +207,6 @@ Codecs:
vp6 Aurelien Jacobs
vqavideo.c Mike Melanson
wavpack.c Kostya Shishkov
wmaprodec.c Sascha Sommer
wmavoice.c Ronald S. Bultje
wmv2.c Michael Niedermayer
wnv1.c Kostya Shishkov
xan.c Mike Melanson
@@ -219,11 +214,6 @@ Codecs:
xvmc.c Ivan Kalvachev
zmbv* Kostya Shishkov
Hardware acceleration:
dxva2* Laurent Aimar
vaapi* Gwenole Beauchesne
vdpau* Carl Eugen Hoyos
libavdevice
===========
@@ -252,12 +242,10 @@ Muxers/Demuxers:
aiff.c Baptiste Coudurier
ape.c Kostya Shishkov
avi* Michael Niedermayer
bink.c Peter Ross
crc.c Michael Niedermayer
daud.c Reimar Doeffinger
dv.c Roman Shaposhnik
dxa.c Kostya Shishkov
electronicarts.c Peter Ross
ffm* Baptiste Coudurier
flac* Justin Ruggles
flic.c Mike Melanson
@@ -273,41 +261,36 @@ Muxers/Demuxers:
libnut.c Oded Shimon
lmlm4.c Ivo van Poorten
matroska.c Aurelien Jacobs
matroskadec.c Aurelien Jacobs
matroskaenc.c David Conrad
metadata* Aurelien Jacobs
mm.c Peter Ross
mov.c Michael Niedermayer, Baptiste Coudurier
movenc.c Michael Niedermayer, Baptiste Coudurier
mpc.c Kostya Shishkov
mpeg.c Michael Niedermayer
mpegenc.c Michael Niedermayer
mpegts* Baptiste Coudurier
mpegts* Mans Rullgard
msnwc_tcp.c Ramiro Polla
mtv.c Reynaldo H. Verdejo Pinochet
mxf* Baptiste Coudurier
nsvdec.c Francois Revol
nut.c Michael Niedermayer
nuv.c Reimar Doeffinger
oggdec.c, oggdec.h David Conrad
oggdec.c, oggdec.h Mans Rullgard
oggenc.c Baptiste Coudurier
oggparse*.c David Conrad
oggparsevorbis.c Mans Rullgard
oggparseogm.c Mans Rullgard
psxstr.c Mike Melanson
pva.c Ivo van Poorten
r3d.c Baptiste Coudurier
raw.c Michael Niedermayer
rdt.c Ronald S. Bultje
rl2.c Sascha Sommer
rmdec.c, rmenc.c Ronald S. Bultje, Kostya Shishkov
rtmp* Kostya Shishkov
rm.c Roberto Togni
rtp.c, rtpenc.c Luca Abeni
rtp_asf.* Ronald S. Bultje
rtp_mpv.*, rtp_aac.* Luca Abeni
rtsp.c Luca Barbato
sdp.c Luca Abeni
segafilm.c Mike Melanson
siff.c Kostya Shishkov
smacker.c Kostya Shishkov
swf.c Baptiste Coudurier
tta.c Alex Beregszaszi
txd.c Ivo van Poorten
@@ -318,7 +301,6 @@ Muxers/Demuxers:
wv.c Kostya Shishkov
Protocols:
http.c Ronald S. Bultje
udp.c Luca Abeni
@@ -327,8 +309,6 @@ Operating systems / CPU architectures
Alpha Mans Rullgard, Falk Hueffner
ARM Mans Rullgard
AVR32 Mans Rullgard
MIPS Mans Rullgard
BeOS Francois Revol
Mac OS X / PowerPC Romain Dolbeau, Guillaume Poirier
Amiga / PowerPC Colin Ward
@@ -343,17 +323,11 @@ x86 Michael Niedermayer
GnuPG Fingerprints of maintainers and others who have svn write access
======================================================================
Attila Kinali 11F0 F9A6 A1D2 11F6 C745 D10C 6520 BCDD F2DF E765
Baptiste Coudurier 8D77 134D 20CC 9220 201F C5DB 0AC9 325C 5C1A BAAA
Benoit Fouet B22A 4F4F 43EF 636B BB66 FCDC 0023 AE1E 2985 49C8
Daniel Verkamp 78A6 07ED 782C 653E C628 B8B9 F0EB 8DD8 2F0E 21C7
Diego Biurrun 8227 1E31 B6D9 4994 7427 E220 9CAE D6CC 4757 FCC5
Jaikrishnan Menon 61A1 F09F 01C9 2D45 78E1 C862 25DC 8831 AF70 D368
Loren Merritt ABD9 08F4 C920 3F65 D8BE 35D7 1540 DAA7 060F 56DE
Michael Niedermayer 9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
Panagiotis Issaris 515C E262 10A8 FDCE 5481 7B9C 3AD7 D9A5 071D B3A9
Peter Ross A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B
Reimar Döffinger C61D 16E5 9E2C D10C 8958 38A4 0899 A2B9 06D4 D9C7
Reinhard Tartler 9300 5DC2 7E87 6C37 ED7B CA9A 9808 3544 9453 48A4
Reynaldo H. Verdejo Pinochet 6E27 CD34 170C C78E 4D4F 5F40 C18E 077F 3114 452A
Sascha Sommer 38A0 F88B 868E 9D3A 97D4 D6A0 E823 706F 1E07 0D3C

437
Makefile
View File

@@ -6,57 +6,49 @@ vpath %.texi $(SRC_PATH_BARE)
PROGS-$(CONFIG_FFMPEG) += ffmpeg
PROGS-$(CONFIG_FFPLAY) += ffplay
PROGS-$(CONFIG_FFPROBE) += ffprobe
PROGS-$(CONFIG_FFSERVER) += ffserver
PROGS := $(addsuffix $(EXESUF), $(PROGS-yes))
PROGS = $(addsuffix $(EXESUF), $(PROGS-yes))
PROGS_G = $(addsuffix _g$(EXESUF), $(PROGS-yes))
OBJS = $(addsuffix .o, $(PROGS-yes)) cmdutils.o
MANPAGES = $(addprefix doc/, $(addsuffix .1, $(PROGS-yes)))
TOOLS = $(addprefix tools/, $(addsuffix $(EXESUF), cws2fws pktdumper probetest qt-faststart trasher))
HOSTPROGS = $(addprefix tests/, audiogen videogen rotozoom tiny_psnr)
BASENAMES = ffmpeg ffplay ffprobe ffserver
BASENAMES = ffmpeg ffplay ffserver
ALLPROGS = $(addsuffix $(EXESUF), $(BASENAMES))
ALLPROGS_G = $(addsuffix _g$(EXESUF), $(BASENAMES))
ALLMANPAGES = $(addsuffix .1, $(BASENAMES))
FFLIBS-$(CONFIG_AVDEVICE) += avdevice
FFLIBS-$(CONFIG_AVFILTER) += avfilter
FFLIBS-$(CONFIG_AVFORMAT) += avformat
FFLIBS-$(CONFIG_AVCODEC) += avcodec
FFLIBS-$(CONFIG_POSTPROC) += postproc
FFLIBS-$(CONFIG_SWSCALE) += swscale
FFLIBS := avutil
FFLIBS := avdevice avformat avcodec avutil
DATA_FILES := $(wildcard $(SRC_DIR)/ffpresets/*.ffpreset)
SKIPHEADERS = cmdutils_common_opts.h
include common.mak
FF_LDFLAGS := $(FFLDFLAGS)
FF_EXTRALIBS := $(FFEXTRALIBS)
FF_DEP_LIBS := $(DEP_LIBS)
ALL_TARGETS-$(CONFIG_DOC) += documentation
ALL_TARGETS-$(CONFIG_VHOOK) += videohook
ALL_TARGETS-$(BUILD_DOC) += documentation
ifdef PROGS
INSTALL_TARGETS-$(CONFIG_VHOOK) += install-vhook
ifneq ($(PROGS),)
INSTALL_TARGETS-yes += install-progs install-data
INSTALL_TARGETS-$(CONFIG_DOC) += install-man
INSTALL_TARGETS-$(BUILD_DOC) += install-man
endif
INSTALL_PROGS_TARGETS-$(CONFIG_SHARED) = install-libs
INSTALL_PROGS_TARGETS-$(BUILD_SHARED) = install-libs
all: $(FF_DEP_LIBS) $(PROGS) $(ALL_TARGETS-yes)
$(PROGS): %$(EXESUF): %_g$(EXESUF)
$(CP) $< $@
cp -p $< $@
$(STRIP) $@
SUBDIR_VARS := OBJS FFLIBS CLEANFILES DIRS TESTPROGS EXAMPLES SKIPHEADERS \
ALTIVEC-OBJS MMX-OBJS NEON-OBJS X86-OBJS YASM-OBJS-FFT YASM-OBJS \
HOSTPROGS BUILT_HEADERS TESTOBJS ARCH_HEADERS
SUBDIR_VARS := OBJS FFLIBS CLEANFILES DIRS TESTS
define RESET
$(1) :=
@@ -75,279 +67,294 @@ ffplay_g$(EXESUF): FF_EXTRALIBS += $(SDL_LIBS)
ffserver_g$(EXESUF): FF_LDFLAGS += $(FFSERVERLDFLAGS)
%_g$(EXESUF): %.o cmdutils.o $(FF_DEP_LIBS)
$(LD) $(FF_LDFLAGS) -o $@ $< cmdutils.o $(FF_EXTRALIBS)
$(CC) $(FF_LDFLAGS) -o $@ $< cmdutils.o $(FF_EXTRALIBS)
tools/%$(EXESUF): tools/%.o
$(LD) $(FF_LDFLAGS) -o $@ $< $(FF_EXTRALIBS)
output_example$(EXESUF): output_example.o $(FF_DEP_LIBS)
$(CC) $(CFLAGS) $(FF_LDFLAGS) -o $@ $< $(FF_EXTRALIBS)
tools/%.o: tools/%.c
$(CC) $(CPPFLAGS) $(CFLAGS) $(CC_O) $<
tools/%$(EXESUF): tools/%.c
$(CC) $(CFLAGS) $(FF_LDFLAGS) -o $@ $< $(FF_EXTRALIBS)
ffplay.o ffplay.d: CFLAGS += $(SDL_CFLAGS)
VERSION_SH = $(SRC_PATH_BARE)/version.sh
GIT_LOG = $(SRC_PATH_BARE)/.git/logs/HEAD
SVN_ENTRIES = $(SRC_PATH_BARE)/.svn/entries
cmdutils.o cmdutils.d: version.h
.version: $(wildcard $(GIT_LOG) $(SVN_ENTRIES)) $(VERSION_SH) config.mak
.version: M=@
alltools: $(addsuffix $(EXESUF),$(addprefix tools/, cws2fws pktdumper qt-faststart trasher))
version.h .version:
$(M)$(VERSION_SH) $(SRC_PATH) version.h $(EXTRA_VERSION)
$(Q)touch .version
VHOOKCFLAGS += $(filter-out -mdynamic-no-pic,$(CFLAGS))
# force version.sh to run whenever version might have changed
-include .version
BASEHOOKS = fish null watermark
ALLHOOKS = $(BASEHOOKS) drawtext imlib2 ppm
ALLHOOKS_SRCS = $(addprefix vhook/, $(addsuffix .c, $(ALLHOOKS)))
alltools: $(TOOLS)
HOOKS-$(HAVE_FORK) += ppm
HOOKS-$(HAVE_IMLIB2) += imlib2
HOOKS-$(HAVE_FREETYPE2) += drawtext
documentation: $(addprefix doc/, developer.html faq.html ffmpeg-doc.html \
ffplay-doc.html ffprobe-doc.html ffserver-doc.html \
general.html libavfilter.html $(ALLMANPAGES))
HOOKS = $(addprefix vhook/, $(addsuffix $(SLIBSUF), $(BASEHOOKS) $(HOOKS-yes)))
VHOOKCFLAGS-$(HAVE_IMLIB2) += `imlib2-config --cflags`
LIBS_imlib2$(SLIBSUF) = `imlib2-config --libs`
VHOOKCFLAGS-$(HAVE_FREETYPE2) += `freetype-config --cflags`
LIBS_drawtext$(SLIBSUF) = `freetype-config --libs`
VHOOKCFLAGS += $(VHOOKCFLAGS-yes)
vhook/%.o vhook/%.d: CFLAGS:=$(VHOOKCFLAGS)
# vhooks compile fine without libav*, but need them nonetheless.
videohook: $(FF_DEP_LIBS) $(HOOKS)
$(eval VHOOKSHFLAGS=$(VHOOKSHFLAGS))
vhook/%$(SLIBSUF): vhook/%.o
$(CC) $(LDFLAGS) -o $@ $(VHOOKSHFLAGS) $< $(VHOOKLIBS) $(LIBS_$(@F))
VHOOK_DEPS = $(HOOKS:$(SLIBSUF)=.d)
depend dep: $(VHOOK_DEPS)
documentation: $(addprefix doc/, ffmpeg-doc.html faq.html ffserver-doc.html \
ffplay-doc.html general.html hooks.html \
$(ALLMANPAGES))
doc/%.html: TAG = HTML
doc/%.html: doc/%.texi
$(M)cd doc && texi2html -monolithic -number $(<:doc/%=%)
texi2html -monolithic -number $<
mv $(@F) $@
doc/%.pod: TAG = POD
doc/%.pod: doc/%-doc.texi
$(M)doc/texi2pod.pl $< $@
doc/texi2pod.pl $< $@
doc/%.1: TAG = MAN
doc/%.1: doc/%.pod
$(M)pod2man --section=1 --center=" " --release=" " $< > $@
pod2man --section=1 --center=" " --release=" " $< > $@
install: $(INSTALL_TARGETS-yes)
install-progs: $(PROGS) $(INSTALL_PROGS_TARGETS-yes)
$(Q)mkdir -p "$(BINDIR)"
$(INSTALL) -c -m 755 $(PROGS) "$(BINDIR)"
install -d "$(BINDIR)"
install -c -m 755 $(PROGS) "$(BINDIR)"
install-data: $(DATA_FILES)
$(Q)mkdir -p "$(DATADIR)"
$(INSTALL) -m 644 $(DATA_FILES) "$(DATADIR)"
install -d "$(DATADIR)"
install -m 644 $(DATA_FILES) "$(DATADIR)"
install-man: $(MANPAGES)
$(Q)mkdir -p "$(MANDIR)/man1"
$(INSTALL) -m 644 $(MANPAGES) "$(MANDIR)/man1"
install -d "$(MANDIR)/man1"
install -m 644 $(MANPAGES) "$(MANDIR)/man1"
uninstall: uninstall-progs uninstall-data uninstall-man
install-vhook: videohook
install -d "$(SHLIBDIR)/vhook"
install -m 755 $(HOOKS) "$(SHLIBDIR)/vhook"
uninstall: uninstall-progs uninstall-data uninstall-man uninstall-vhook
uninstall-progs:
$(RM) $(addprefix "$(BINDIR)/", $(ALLPROGS))
rm -f $(addprefix "$(BINDIR)/", $(ALLPROGS))
uninstall-data:
$(RM) -r "$(DATADIR)"
rm -rf "$(DATADIR)"
uninstall-man:
$(RM) $(addprefix "$(MANDIR)/man1/",$(ALLMANPAGES))
rm -f $(addprefix "$(MANDIR)/man1/",$(ALLMANPAGES))
uninstall-vhook:
rm -f $(addprefix "$(SHLIBDIR)/",$(ALLHOOKS_SRCS:.c=$(SLIBSUF)))
-rmdir "$(SHLIBDIR)/vhook/"
testclean:
$(RM) -r tests/vsynth1 tests/vsynth2 tests/data
$(RM) $(addprefix tests/,$(CLEANSUFFIXES))
$(RM) tests/seek_test$(EXESUF) tests/seek_test.o
$(RM) $(addprefix tests/,$(addsuffix $(HOSTEXESUF),audiogen videogen rotozoom tiny_psnr))
rm -rf tests/vsynth1 tests/vsynth2 tests/data tests/asynth1.sw tests/*~
clean:: testclean
$(RM) $(ALLPROGS) $(ALLPROGS_G)
$(RM) $(CLEANSUFFIXES)
$(RM) doc/*.html doc/*.pod doc/*.1
$(RM) $(TOOLS)
rm -f $(ALLPROGS) $(ALLPROGS_G) output_example$(EXESUF)
rm -f doc/*.html doc/*.pod doc/*.1
rm -f $(addprefix tests/,$(addsuffix $(EXESUF),audiogen videogen rotozoom seek_test tiny_psnr))
rm -f $(addprefix tools/,$(addsuffix $(EXESUF),cws2fws pktdumper qt-faststart trasher))
rm -f vhook/*.o vhook/*~ vhook/*.so vhook/*.dylib vhook/*.dll
distclean::
$(RM) $(DISTCLEANSUFFIXES)
$(RM) version.h config.* libavutil/avconfig.h
config:
$(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION)
rm -f version.h config.* vhook/*.d
# regression tests
check: test checkheaders
fulltest test: codectest lavftest seektest
fulltest test: codectest libavtest seektest
FFMPEG_REFFILE = $(SRC_PATH)/tests/ffmpeg.regression.ref
FFSERVER_REFFILE = $(SRC_PATH)/tests/ffserver.regression.ref
LIBAV_REFFILE = $(SRC_PATH)/tests/libav.regression.ref
ROTOZOOM_REFFILE = $(SRC_PATH)/tests/rotozoom.regression.ref
SEEK_REFFILE = $(SRC_PATH)/tests/seek.regression.ref
ENCDEC = $(and $(CONFIG_$(1)_ENCODER),$(CONFIG_$(1)_DECODER))
MUXDEM = $(and $(CONFIG_$(1)_MUXER),$(CONFIG_$(or $(2),$(1))_DEMUXER))
CODEC_TESTS = $(addprefix regtest-, \
mpeg \
mpeg2 \
mpeg2thread \
msmpeg4v2 \
msmpeg4 \
wmv1 \
wmv2 \
h261 \
h263 \
h263p \
mpeg4 \
huffyuv \
rc \
mpeg4adv \
mpeg4thread \
error \
mpeg4nr \
mpeg1b \
mjpeg \
ljpeg \
jpegls \
rv10 \
rv20 \
asv1 \
asv2 \
flv \
ffv1 \
snow \
snowll \
dv \
dv50 \
svq1 \
flashsv \
mp2 \
ac3 \
g726 \
adpcm_ima_wav \
adpcm_ima_qt \
adpcm_ms \
adpcm_yam \
adpcm_swf \
flac \
wma \
pcm \
)
VCODEC_TESTS =
VCODEC_TESTS-$(call ENCDEC,ASV1) += asv1
VCODEC_TESTS-$(call ENCDEC,ASV2) += asv2
VCODEC_TESTS-$(call ENCDEC,DNXHD) += dnxhd_1080i dnxhd_720p dnxhd_720p_rd
VCODEC_TESTS-$(call ENCDEC,DVVIDEO) += dv dv50
VCODEC_TESTS-$(call ENCDEC,FFV1) += ffv1
VCODEC_TESTS-$(call ENCDEC,FLASHSV) += flashsv
VCODEC_TESTS-$(call ENCDEC,FLV) += flv
VCODEC_TESTS-$(call ENCDEC,H261) += h261
VCODEC_TESTS-$(call ENCDEC,H263) += h263 h263p
VCODEC_TESTS-$(call ENCDEC,HUFFYUV) += huffyuv
VCODEC_TESTS-$(call ENCDEC,JPEGLS) += jpegls
VCODEC_TESTS-$(call ENCDEC,MJPEG) += mjpeg ljpeg
VCODEC_TESTS-$(call ENCDEC,MPEG1VIDEO) += mpeg mpeg1b
VCODEC_TESTS-$(call ENCDEC,MPEG2VIDEO) += mpeg2 mpeg2thread
VCODEC_TESTS-$(call ENCDEC,MPEG4) += mpeg4 mpeg4adv mpeg4nr mpeg4thread error rc
VCODEC_TESTS-$(call ENCDEC,MSMPEG4V1) += msmpeg4
VCODEC_TESTS-$(call ENCDEC,MSMPEG4V2) += msmpeg4v2
VCODEC_TESTS-$(call ENCDEC,ROQ) += roq
VCODEC_TESTS-$(call ENCDEC,RV10) += rv10
VCODEC_TESTS-$(call ENCDEC,RV20) += rv20
VCODEC_TESTS-$(call ENCDEC,SNOW) += snow snowll
VCODEC_TESTS-$(call ENCDEC,SVQ1) += svq1
VCODEC_TESTS-$(call ENCDEC,WMV1) += wmv1
VCODEC_TESTS-$(call ENCDEC,WMV2) += wmv2
LAVF_TESTS = $(addprefix regtest-, \
avi \
asf \
rm \
mpg \
ts \
swf \
ffm \
flv_fmt \
mov \
dv_fmt \
gxf \
nut \
mkv \
pbmpipe \
pgmpipe \
ppmpipe \
gif \
yuv4mpeg \
pgm \
ppm \
bmp \
tga \
tiff \
sgi \
jpg \
wav \
alaw \
mulaw \
au \
mmf \
aiff \
voc \
ogg \
pixfmt \
)
ACODEC_TESTS =
ACODEC_TESTS-$(call ENCDEC,AC3) += ac3
ACODEC_TESTS-$(call ENCDEC,ADPCM_G726) += g726
ACODEC_TESTS-$(call ENCDEC,ADPCM_IMA_QT) += adpcm_ima_qt
ACODEC_TESTS-$(call ENCDEC,ADPCM_IMA_WAV) += adpcm_ima_wav
ACODEC_TESTS-$(call ENCDEC,ADPCM_MS) += adpcm_ms
ACODEC_TESTS-$(call ENCDEC,ADPCM_SWF) += adpcm_swf
ACODEC_TESTS-$(call ENCDEC,ADPCM_YAMAHA) += adpcm_yam
ACODEC_TESTS-$(call ENCDEC,ALAC) += alac
ACODEC_TESTS-$(call ENCDEC,FLAC) += flac
ACODEC_TESTS-$(call ENCDEC,MP2) += mp2
ACODEC_TESTS-$(call ENCDEC,PCM_S16LE) += pcm # fixme
ACODEC_TESTS-$(call ENCDEC,WMAV1) += wmav1
ACODEC_TESTS-$(call ENCDEC,WMAV1) += wmav2
REGFILES = $(addprefix tests/data/,$(addsuffix .$(1),$(2:regtest-%=%)))
LAVF_TESTS =
LAVF_TESTS-$(call MUXDEM,AIFF) += aiff
LAVF_TESTS-$(call MUXDEM,PCM_ALAW) += alaw
LAVF_TESTS-$(call MUXDEM,ASF) += asf
LAVF_TESTS-$(call MUXDEM,AU) += au
LAVF_TESTS-$(call MUXDEM,AVI) += avi
LAVF_TESTS-$(call ENCDEC,BMP) += bmp
LAVF_TESTS-$(call MUXDEM,DV) += dv_fmt
LAVF_TESTS-$(call MUXDEM,FFM) += ffm
LAVF_TESTS-$(call MUXDEM,FLV) += flv_fmt
LAVF_TESTS-$(call ENCDEC,GIF) += gif
LAVF_TESTS-$(call MUXDEM,GXF) += gxf
LAVF_TESTS-$(call ENCDEC,MJPEG) += jpg
LAVF_TESTS-$(call MUXDEM,MATROSKA) += mkv
LAVF_TESTS-$(call MUXDEM,MMF) += mmf
LAVF_TESTS-$(call MUXDEM,MOV) += mov
LAVF_TESTS-$(call MUXDEM,MPEG1SYSTEM,MPEGPS) += mpg
LAVF_TESTS-$(call MUXDEM,PCM_MULAW) += mulaw
LAVF_TESTS-$(call MUXDEM,MXF) += mxf
LAVF_TESTS-$(call MUXDEM,NUT) += nut
LAVF_TESTS-$(call MUXDEM,OGG) += ogg
LAVF_TESTS-$(call ENCDEC,PBM) += pbmpipe
LAVF_TESTS-$(call ENCDEC,PCX) += pcx
LAVF_TESTS-$(call ENCDEC,PGM) += pgm pgmpipe
LAVF_TESTS-$(call MUXDEM,RAWVIDEO) += pixfmt
LAVF_TESTS-$(call ENCDEC,PPM) += ppm ppmpipe
LAVF_TESTS-$(call MUXDEM,RM) += rm
LAVF_TESTS-$(call ENCDEC,SGI) += sgi
LAVF_TESTS-$(call MUXDEM,SWF) += swf
LAVF_TESTS-$(call ENCDEC,TARGA) += tga
LAVF_TESTS-$(call ENCDEC,TIFF) += tiff
LAVF_TESTS-$(call MUXDEM,MPEGTS) += ts
LAVF_TESTS-$(call MUXDEM,VOC) += voc
LAVF_TESTS-$(call MUXDEM,WAV) += wav
LAVF_TESTS-$(call MUXDEM,YUV4MPEGPIPE) += yuv4mpeg
CODEC_ROTOZOOM = $(call REGFILES,rotozoom.regression,$(CODEC_TESTS))
CODEC_VSYNTH = $(call REGFILES,vsynth.regression,$(CODEC_TESTS))
LAVFI_TESTS = \
crop \
crop_scale \
crop_scale_vflip \
crop_vflip \
null \
scale200 \
scale500 \
vflip \
vflip_crop \
vflip_vflip \
LAVF_REGFILES = $(call REGFILES,lavf.regression,$(LAVF_TESTS))
ACODEC_TESTS := $(addprefix regtest-, $(ACODEC_TESTS) $(ACODEC_TESTS-yes))
VCODEC_TESTS := $(addprefix regtest-, $(VCODEC_TESTS) $(VCODEC_TESTS-yes))
LAVF_TESTS := $(addprefix regtest-, $(LAVF_TESTS) $(LAVF_TESTS-yes))
LAVFI_TESTS := $(addprefix regtest-, $(LAVFI_TESTS) $(LAVFI_TESTS-yes))
LAVF_REG = tests/data/lavf.regression
ROTOZOOM_REG = tests/data/rotozoom.regression
VSYNTH_REG = tests/data/vsynth.regression
CODEC_TESTS = $(VCODEC_TESTS) $(ACODEC_TESTS)
ifneq ($(CONFIG_SWSCALE),yes)
servertest codectest $(CODEC_TESTS) libavtest: swscale-error
swscale-error:
@echo
@echo "This regression test requires --enable-swscale."
@echo
@exit 1
endif
codectest: $(CODEC_TESTS)
lavftest: $(LAVF_TESTS)
lavfitest: $(LAVFI_TESTS)
ifneq ($(CONFIG_ZLIB),yes)
regtest-flashsv codectest: zlib-error
endif
zlib-error:
@echo
@echo "This regression test requires zlib."
@echo
@exit 1
$(ACODEC_TESTS): regtest-aref
$(VCODEC_TESTS): regtest-vref
$(LAVF_TESTS) $(LAVFI_TESTS): regtest-ref
codectest: $(VSYNTH_REG) $(ROTOZOOM_REG)
diff -u -w $(FFMPEG_REFFILE) $(VSYNTH_REG)
diff -u -w $(ROTOZOOM_REFFILE) $(ROTOZOOM_REG)
REFFILE = $(SRC_PATH)/tests/ref/$(1)/$(2:regtest-%=%)
RESFILE = tests/data/$(2:regtest-%=%).$(1).regression
libavtest: $(LAVF_REG)
diff -u -w $(LIBAV_REFFILE) $(LAVF_REG)
define CODECTEST_CMD
$(SRC_PATH)/tests/codec-regression.sh $@ vsynth1 tests/vsynth1 "$(TARGET_EXEC)" "$(TARGET_PATH)"
$(SRC_PATH)/tests/codec-regression.sh $@ vsynth2 tests/vsynth2 "$(TARGET_EXEC)" "$(TARGET_PATH)"
endef
$(VSYNTH_REG) $(ROTOZOOM_REG) $(LAVF_REG):
cat $^ > $@
regtest-ref: regtest-aref regtest-vref
$(LAVF_REG): $(LAVF_REGFILES)
$(ROTOZOOM_REG): $(CODEC_ROTOZOOM)
$(VSYNTH_REG): $(CODEC_VSYNTH)
regtest-vref: ffmpeg$(EXESUF) tests/vsynth1/00.pgm tests/vsynth2/00.pgm
$(CODECTEST_CMD)
$(CODEC_VSYNTH) $(CODEC_ROTOZOOM): $(CODEC_TESTS)
regtest-aref: ffmpeg$(EXESUF) tests/data/asynth1.sw
@$(SRC_PATH)/tests/codec-regression.sh $@ acodec tests/acodec "$(TARGET_EXEC)" "$(TARGET_PATH)"
$(LAVF_REGFILES): $(LAVF_TESTS)
$(VCODEC_TESTS): tests/tiny_psnr$(HOSTEXESUF)
@echo "TEST VCODEC $(@:regtest-%=%)"
@$(CODECTEST_CMD)
@diff -u -w $(call REFFILE,vsynth1,$@) $(call RESFILE,vsynth1,$@)
@diff -u -w $(call REFFILE,vsynth2,$@) $(call RESFILE,vsynth2,$@)
$(CODEC_TESTS) $(LAVF_TESTS): regtest-ref
$(ACODEC_TESTS): tests/tiny_psnr$(HOSTEXESUF)
@echo "TEST ACODEC $(@:regtest-%=%)"
@$(SRC_PATH)/tests/codec-regression.sh $@ acodec tests/acodec "$(TARGET_EXEC)" "$(TARGET_PATH)"
@diff -u -w $(call REFFILE,acodec,$@) $(call RESFILE,acodec,$@)
regtest-ref: ffmpeg$(EXESUF) tests/vsynth1/00.pgm tests/vsynth2/00.pgm tests/asynth1.sw
$(CODEC_TESTS) regtest-ref: tests/tiny_psnr$(EXESUF)
$(SRC_PATH)/tests/regression.sh $@ vsynth tests/vsynth1 a "$(TARGET_EXEC)" "$(TARGET_PATH)"
$(SRC_PATH)/tests/regression.sh $@ rotozoom tests/vsynth2 a "$(TARGET_EXEC)" "$(TARGET_PATH)"
$(LAVF_TESTS):
@echo "TEST LAVF $(@:regtest-%=%)"
@$(SRC_PATH)/tests/lavf-regression.sh $@ lavf tests/vsynth1 "$(TARGET_EXEC)" "$(TARGET_PATH)"
@diff -u -w $(call REFFILE,lavf,$@) $(call RESFILE,lavf,$@)
$(SRC_PATH)/tests/regression.sh $@ lavf tests/vsynth1 b "$(TARGET_EXEC)" "$(TARGET_PATH)"
$(LAVFI_TESTS):
@echo "TEST LAVFI $(@:regtest-%=%)"
@$(SRC_PATH)/tests/lavfi-regression.sh $@ lavfi tests/vsynth1 "$(TARGET_EXEC)" "$(TARGET_PATH)"
@diff -u -w $(call REFFILE,lavfi,$@) $(call RESFILE,lavfi,$@)
seektest: codectest libavtest tests/seek_test$(EXESUF)
$(SRC_PATH)/tests/seek_test.sh $(SEEK_REFFILE) "$(TARGET_EXEC)" "$(TARGET_PATH)"
seektest: codectest lavftest tests/seek_test$(EXESUF)
$(SRC_PATH)/tests/seek-regression.sh $(SRC_PATH) "$(TARGET_EXEC)" "$(TARGET_PATH)"
ffservertest: ffserver$(EXESUF) tests/vsynth1/00.pgm tests/data/asynth1.sw
servertest: ffserver$(EXESUF) tests/vsynth1/00.pgm tests/asynth1.sw
@echo
@echo "Unfortunately ffserver is broken and therefore its regression"
@echo "test fails randomly. Treat the results accordingly."
@echo
$(SRC_PATH)/tests/ffserver-regression.sh $(FFSERVER_REFFILE) $(SRC_PATH)/tests/ffserver.conf
$(SRC_PATH)/tests/server-regression.sh $(FFSERVER_REFFILE) $(SRC_PATH)/tests/test.conf
tests/vsynth1/00.pgm: tests/videogen$(HOSTEXESUF)
tests/vsynth1/00.pgm: tests/videogen$(EXESUF)
mkdir -p tests/vsynth1
$(BUILD_ROOT)/$< 'tests/vsynth1/'
tests/vsynth2/00.pgm: tests/rotozoom$(HOSTEXESUF)
tests/vsynth2/00.pgm: tests/rotozoom$(EXESUF)
mkdir -p tests/vsynth2
$(BUILD_ROOT)/$< 'tests/vsynth2/' $(SRC_PATH)/tests/lena.pnm
tests/data/asynth1.sw: tests/audiogen$(HOSTEXESUF)
mkdir -p tests/data
tests/asynth1.sw: tests/audiogen$(EXESUF)
$(BUILD_ROOT)/$< $@
tests/seek_test$(EXESUF): tests/seek_test.o $(FF_DEP_LIBS)
$(LD) $(FF_LDFLAGS) -o $@ $< $(FF_EXTRALIBS)
tests/%$(EXESUF): tests/%.c
$(HOSTCC) $(HOSTCFLAGS) $(HOSTLDFLAGS) -o $@ $< $(HOSTLIBS)
ifdef SAMPLES
include $(SRC_PATH_BARE)/tests/fate.mak
fate: $(FATE_TESTS)
$(FATE_TESTS): ffmpeg$(EXESUF)
@echo "TEST FATE $(@:fate-%=%)"
@$(SRC_PATH)/tests/fate-run.sh $@ "$(SAMPLES)" "$(TARGET_EXEC)" "$(TARGET_PATH)" '$(CMD)'
else
fate:
@echo "SAMPLES not specified, cannot run FATE"
endif
tests/seek_test$(EXESUF): tests/seek_test.c $(FF_DEP_LIBS)
$(CC) $(FF_LDFLAGS) $(CFLAGS) -o $@ $< $(FF_EXTRALIBS)
.PHONY: documentation *test regtest-* zlib-error alltools check config
.PHONY: lib videohook documentation *test regtest-* swscale-error zlib-error alltools check
-include $(VHOOK_DEPS)

30
README
View File

@@ -9,4 +9,32 @@ FFmpeg README
2) Licensing
------------
* See the LICENSE file.
* Read the file COPYING.LGPL. FFmpeg and the associated libraries EXCEPT
for libpostproc are licensed under the GNU Lesser General Public License.
* libpostproc is distributed under the GNU General Public License, see the
file COPYING.GPL for details. Its compilation and use in FFmpeg is optional.
* libswscale contains some optional processor-specific optimizations that are
distributed under the GNU General Public License.
* The files libavcodec/x86/idct_mmx.c, libavcodec/x86/h264_deblock_sse2.asm
and libavcodec/x86/h264_idct_sse2.asm are distributed under the GNU General
Public License. They are strictly optimizations and their use is optional.
* The file libavcodec/ac3dec.c is distributed under the GNU General Public
License. In order for (E-)AC-3 decoding to work you need to enable GPL
components.
* The file libavdevice/x11grab.c is distributed under the GNU General
Public License. X11 grabbing is optional.
* The files libavcodec/jfdctfst.c, libavcodec/jfdctint.c, libavcodec/jrevdct.c
are taken from libjpeg, see the top of the files for licensing details.
* The file libavcodec/fdctref.c is copyrighted by the MPEG Software Simulation
Group with all rights reserved. It is only used to create a DCT test program
and not compiled into libavcodec.
* Some external libraries are under GPL. If you wish to use them with FFmpeg,
you have to configure FFmpeg as GPL as well.

209
RELEASE
View File

@@ -1,18 +1,14 @@
Release Notes
=============
* 0.6 "Works with HTML5" June, 2010
* 0.5 "Bike Shed" March 3, 2009
General notes
-------------
This release focuses on improvements for the new multimedia elements in HTML5.
The H.264 and Theora decoders are now significantly faster, the Vorbis decoder
has seen important updates and this release supports Google's newly released
libvpx library for the VP8 codec and WebM container.
Other important changes are additions of decoders including, but not limited to,
Intel Indeo 5, WMA Pro, WMA Voice and HE-AAC.
It has been so long since the last release that this should be considered the
first FFmpeg release of recent times. Because of the way things have unfolded to
date, the notes for this version cannot be entirely conventional.
See the Changelog file for a list of significant changes.
@@ -23,190 +19,21 @@ version of the development code to check if the issue still exists. If it does,
make your report against the development code following the usual bug reporting
guidelines.
API notes
---------
API and other notable Changes
-----------------------------
In the next release, it is intended to remove a number of deprecated APIs. We
decided to put out a release that includes said APIs for the benefit of third
party software.
Please see the file doc/APIchanges for programmer-centric information.
As such, this release:
- provides a sync point for said APIs
- increases awareness of API changes
- allows the next release to detail how to transition from the old to the new
Notable changes:
- deprecated vhook subsystem removed
- deprecated old scaler removed
- nonfree libamr support for AMR-NB/WB decoding/encoding removed
- RTMP support in libavformat
- -formats option split into -formats, -codecs, -bsfs, and -protocols
- ffprobe tool
- RTMP/RTMPT/RTMPS/RTMPE/RTMPTE protocol support via librtmp
- CODEC_CAP_EXPERIMENTAL added
The deprecated APIs to be removed are:
- imgconvert (to be replaced by libswscale)
- vhook (to be replaced by libavfilter)
Added Codecs:
-------------
- VQF demuxer
- PCX encoder
- CorePNG decoding support
- 8088flex TMV demuxer and decoder
- enable symbol versioning by default for linkers that support it
- V210 decoder and encoder
- QCP demuxer
- SoX native format muxer and demuxer
- AMR-NB decoding/encoding, AMR-WB decoding via OpenCORE libraries
- DPX image decoder
- Electronic Arts Madcow decoder
- DivX (XSUB) subtitle encoder
- experimental AAC encoder
- Wave64 demuxer
- IEC-61937 compatible Muxer
- TwinVQ decoder
- Bluray (PGS) subtitle decoder
- LPCM support in MPEG-TS (HDMV RID as found on Blu-ray disks)
- WMA Pro decoder
- Core Audio Format demuxer
- Atrac1 decoder
- MD STUDIO audio demuxer
- RF64 support in WAV demuxer
- MPEG-4 Audio Lossless Coding (ALS) decoder
- IV8 demuxer
- CDG demuxer and decoder
- R210 decoder
- Auravision Aura 1 and 2 decoders
- Deluxe Paint Animation playback system
- SIPR decoder
- Adobe Filmstrip muxer and demuxer
- RTP packetization and depacketization of H.263 and AMR
- Bink demuxer and audio/video decoders
- IFF PBM/ILBM bitmap decoder
- Indeo 5 decoder
- WMA Voice decoder
- AMR-NB decoder
- RTSP muxer
- HE-AAC v1 decoder
- Kega Game Video (KGV1) decoder
- Psygnosis YOP demuxer and video decoder
- RTP hinting in the mov/3gp/mp4 muxer
- VP8 decoding via libvpx
Notable license related changes
-------------------------------
- remaining GPL parts in AC-3 decoder converted to LGPL
- libswscale can now be compiled in LGPL mode
* 0.6.1
General notes
-------------
This point release includes some updates to make the 0.6 release series usable
for users that need to retain the existing behavior as closely as possible.
The changes follow below:
Bugfixes
--------
- fix autodetection of E-AC-3 substream samples
- performance fix for seekable HTTP
- add missing VP80 fourcc code for the VP8 codec
- small documentation fixes
- fix several potentially exploitable issues in the FLIC decoder
(addresses CVE-2010-3429)
HE-AAC v2 backport
------------------
This release includes a backport of the AAC decoder from trunk, which
enables proper playback of HE-AAC v2 media.
* 0.6.2
General notes
-------------
This is a maintenance-only release that addresses a small number of security
and portability issues. Distributors and system integrators are encouraged
to update and share their patches against this branch.
Security fixes
--------------
Programming errors in container and codec implementations may lead to
denial of service or the execution of arbitrary code if the user is
tricked into opening a malformed media file or stream.
Affected and updated have been the implementations of the following
codecs and container formats:
- VC1 decoder (Change related to CVE-2011-0723)
- APE decoder (cf. http://packetstorm.linuxsecurity.com/1103-exploits/vlc105-dos.txt)
* 0.6.3
General notes
-------------
This is a mostly maintenance-only release that addresses a small number
of bugs such as security and compilation issues. Moreover, this release
has been updated to work with gcc-4.6 and the VisualOn AAC encoder has
been backported from the Libav 0.7.2 release. Distributors and system
integrators are encouraged to update and share their patches against
this branch.
For a full list of changes please see the Changelog file.
* 0.6.4
General notes
-------------
This mostly maintenance-only release that addresses a number a number of
bugs such as security and compilation issues that have been brought to
our attention. Among other (rather minor) fixes, this release features
fixes for the QDM2 decoder (CVE-2011-4351), vp3 decoder (CVE-2011-4352),
DoS in the VP5/VP6 decoders (CVE-2011-4353), a buffer overflow in the
Sierra VMD decoder CVE-2011-4364, and a safety fix in the svq1 decoder
(CVE-2011-4579).
Distributors and system integrators are encouraged
to update and share their patches against this branch. For a full list
of changes please see the Changelog file.
* 0.6.5
General notes
-------------
This mostly maintenance-only release that addresses a number a number of
bugs such as security and compilation issues that have been brought to
our attention. Among other (rather minor) fixes, this release features
fixes for the VP3 decoder (CVE-2011-3892), vorbis decoder, and matroska
demuxer (CVE-2011-3893 and CVE-2011-3895).
Distributors and system integrators are encouraged
to update and share their patches against this branch. For a full list
of changes please see the Changelog file.
* 0.6.6
General notes
-------------
This mostly maintenance-only release that addresses a number a number of
bugs such as security and compilation issues that have been brought to
our attention. Among other fixes, this release features includes
security updates for the, DV decoder (CVE-2011-3929 and CVE-2011-3936),
VQA Video Decoder (CVE-2012-0947), DPCM codecs (CVE-2011-3951), AAC SBR
(CVE-2012-0850), H.264 (CVE-2012-0851), ADPCM (CVE-2012-0852), Shorten
(CVE-2012-0858), and the KMVC decoder (CVE-2011-3952).
Distributors and system integrators are encouraged
to update and share their patches against this branch. For a full list
of changes please see the Changelog file or the git commit history.
If at all possible, do not use the deprecated APIs. All notes on API changes
should appear in doc/APIchanges.

View File

@@ -1 +0,0 @@
0.6.6

View File

@@ -35,24 +35,22 @@
#include "libswscale/swscale.h"
#include "libpostproc/postprocess.h"
#include "libavutil/avstring.h"
#include "libavutil/pixdesc.h"
#include "libavcodec/opt.h"
#include "cmdutils.h"
#include "version.h"
#if CONFIG_NETWORK
#include "libavformat/network.h"
#endif
#if HAVE_SYS_RESOURCE_H
#include <sys/resource.h>
#endif
#undef exit
const char **opt_names;
static int opt_name_count;
AVCodecContext *avcodec_opts[AVMEDIA_TYPE_NB];
AVCodecContext *avctx_opts[CODEC_TYPE_NB];
AVFormatContext *avformat_opts;
struct SwsContext *sws_opts;
const int this_year = 2010;
const int this_year = 2009;
double parse_number_or_die(const char *context, const char *numstr, int type, double min, double max)
{
@@ -127,20 +125,11 @@ void parse_options(int argc, char **argv, const OptionDef *options,
opt = argv[optindex++];
if (handleoptions && opt[0] == '-' && opt[1] != '\0') {
int bool_val = 1;
if (opt[1] == '-' && opt[2] == '\0') {
handleoptions = 0;
continue;
}
opt++;
po= find_option(options, opt);
if (!po->name && opt[0] == 'n' && opt[1] == 'o') {
/* handle 'no' bool option */
po = find_option(options, opt + 2);
if (!(po->name && (po->flags & OPT_BOOL)))
goto unknown_opt;
bool_val = 0;
}
po= find_option(options, opt + 1);
if (!po->name)
po= find_option(options, "default");
if (!po->name) {
@@ -161,18 +150,16 @@ unknown_opt:
str = av_strdup(arg);
*po->u.str_arg = str;
} else if (po->flags & OPT_BOOL) {
*po->u.int_arg = bool_val;
*po->u.int_arg = 1;
} else if (po->flags & OPT_INT) {
*po->u.int_arg = parse_number_or_die(opt, arg, OPT_INT64, INT_MIN, INT_MAX);
*po->u.int_arg = parse_number_or_die(opt+1, arg, OPT_INT64, INT_MIN, INT_MAX);
} else if (po->flags & OPT_INT64) {
*po->u.int64_arg = parse_number_or_die(opt, arg, OPT_INT64, INT64_MIN, INT64_MAX);
*po->u.int64_arg = parse_number_or_die(opt+1, arg, OPT_INT64, INT64_MIN, INT64_MAX);
} else if (po->flags & OPT_FLOAT) {
*po->u.float_arg = parse_number_or_die(opt, arg, OPT_FLOAT, -1.0/0.0, 1.0/0.0);
*po->u.float_arg = parse_number_or_die(opt+1, arg, OPT_FLOAT, -1.0/0.0, 1.0/0.0);
} else if (po->flags & OPT_FUNC2) {
if (po->u.func2_arg(opt, arg) < 0) {
fprintf(stderr, "%s: failed to set value '%s' for option '%s'\n", argv[0], arg, opt);
exit(1);
}
if(po->u.func2_arg(opt+1, arg)<0)
goto unknown_opt;
} else {
po->u.func_arg(arg);
}
@@ -191,91 +178,41 @@ int opt_default(const char *opt, const char *arg){
const AVOption *o= NULL;
int opt_types[]={AV_OPT_FLAG_VIDEO_PARAM, AV_OPT_FLAG_AUDIO_PARAM, 0, AV_OPT_FLAG_SUBTITLE_PARAM, 0};
for(type=0; type<AVMEDIA_TYPE_NB && ret>= 0; type++){
const AVOption *o2 = av_find_opt(avcodec_opts[0], opt, NULL, opt_types[type], opt_types[type]);
for(type=0; type<CODEC_TYPE_NB && ret>= 0; type++){
const AVOption *o2 = av_find_opt(avctx_opts[0], opt, NULL, opt_types[type], opt_types[type]);
if(o2)
ret = av_set_string3(avcodec_opts[type], opt, arg, 1, &o);
ret = av_set_string3(avctx_opts[type], opt, arg, 1, &o);
}
if(!o)
ret = av_set_string3(avformat_opts, opt, arg, 1, &o);
if(!o && sws_opts)
if(!o)
ret = av_set_string3(sws_opts, opt, arg, 1, &o);
if(!o){
if(opt[0] == 'a')
ret = av_set_string3(avcodec_opts[AVMEDIA_TYPE_AUDIO], opt+1, arg, 1, &o);
ret = av_set_string3(avctx_opts[CODEC_TYPE_AUDIO], opt+1, arg, 1, &o);
else if(opt[0] == 'v')
ret = av_set_string3(avcodec_opts[AVMEDIA_TYPE_VIDEO], opt+1, arg, 1, &o);
ret = av_set_string3(avctx_opts[CODEC_TYPE_VIDEO], opt+1, arg, 1, &o);
else if(opt[0] == 's')
ret = av_set_string3(avcodec_opts[AVMEDIA_TYPE_SUBTITLE], opt+1, arg, 1, &o);
ret = av_set_string3(avctx_opts[CODEC_TYPE_SUBTITLE], opt+1, arg, 1, &o);
}
if (o && ret < 0) {
fprintf(stderr, "Invalid value '%s' for option '%s'\n", arg, opt);
exit(1);
}
if (!o) {
fprintf(stderr, "Unrecognized option '%s'\n", opt);
exit(1);
}
if(!o)
return -1;
// av_log(NULL, AV_LOG_ERROR, "%s:%s: %f 0x%0X\n", opt, arg, av_get_double(avcodec_opts, opt, NULL), (int)av_get_int(avcodec_opts, opt, NULL));
// av_log(NULL, AV_LOG_ERROR, "%s:%s: %f 0x%0X\n", opt, arg, av_get_double(avctx_opts, opt, NULL), (int)av_get_int(avctx_opts, opt, NULL));
//FIXME we should always use avcodec_opts, ... for storing options so there will not be any need to keep track of what i set over this
//FIXME we should always use avctx_opts, ... for storing options so there will not be any need to keep track of what i set over this
opt_names= av_realloc(opt_names, sizeof(void*)*(opt_name_count+1));
opt_names[opt_name_count++]= o->name;
if(avcodec_opts[0]->debug || avformat_opts->debug)
if(avctx_opts[0]->debug || avformat_opts->debug)
av_log_set_level(AV_LOG_DEBUG);
return 0;
}
int opt_loglevel(const char *opt, const char *arg)
{
const struct { const char *name; int level; } log_levels[] = {
{ "quiet" , AV_LOG_QUIET },
{ "panic" , AV_LOG_PANIC },
{ "fatal" , AV_LOG_FATAL },
{ "error" , AV_LOG_ERROR },
{ "warning", AV_LOG_WARNING },
{ "info" , AV_LOG_INFO },
{ "verbose", AV_LOG_VERBOSE },
{ "debug" , AV_LOG_DEBUG },
};
char *tail;
int level;
int i;
for (i = 0; i < FF_ARRAY_ELEMS(log_levels); i++) {
if (!strcmp(log_levels[i].name, arg)) {
av_log_set_level(log_levels[i].level);
return 0;
}
}
level = strtol(arg, &tail, 10);
if (*tail) {
fprintf(stderr, "Invalid loglevel \"%s\". "
"Possible levels are numbers or:\n", arg);
for (i = 0; i < FF_ARRAY_ELEMS(log_levels); i++)
fprintf(stderr, "\"%s\"\n", log_levels[i].name);
exit(1);
}
av_log_set_level(level);
return 0;
}
int opt_timelimit(const char *opt, const char *arg)
{
#if HAVE_SETRLIMIT
int lim = parse_number_or_die(opt, arg, OPT_INT64, 0, INT_MAX);
struct rlimit rl = { lim, lim + 1 };
if (setrlimit(RLIMIT_CPU, &rl))
perror("setrlimit");
#else
fprintf(stderr, "Warning: -%s not implemented on this OS\n", opt);
#endif
return 0;
}
void set_context_opts(void *ctx, void *opts_ctx, int flags)
{
int i;
@@ -291,69 +228,78 @@ void set_context_opts(void *ctx, void *opts_ctx, int flags)
void print_error(const char *filename, int err)
{
char errbuf[128];
const char *errbuf_ptr = errbuf;
if (av_strerror(err, errbuf, sizeof(errbuf)) < 0)
errbuf_ptr = strerror(AVUNERROR(err));
fprintf(stderr, "%s: %s\n", filename, errbuf_ptr);
switch(err) {
case AVERROR_NUMEXPECTED:
fprintf(stderr, "%s: Incorrect image filename syntax.\n"
"Use '%%d' to specify the image number:\n"
" for img1.jpg, img2.jpg, ..., use 'img%%d.jpg';\n"
" for img001.jpg, img002.jpg, ..., use 'img%%03d.jpg'.\n",
filename);
break;
case AVERROR_INVALIDDATA:
fprintf(stderr, "%s: Error while parsing header\n", filename);
break;
case AVERROR_NOFMT:
fprintf(stderr, "%s: Unknown format\n", filename);
break;
case AVERROR(EIO):
fprintf(stderr, "%s: I/O error occurred\n"
"Usually that means that input file is truncated and/or corrupted.\n",
filename);
break;
case AVERROR(ENOMEM):
fprintf(stderr, "%s: memory allocation error occurred\n", filename);
break;
case AVERROR(ENOENT):
fprintf(stderr, "%s: no such file or directory\n", filename);
break;
#if CONFIG_NETWORK
case AVERROR(FF_NETERROR(EPROTONOSUPPORT)):
fprintf(stderr, "%s: Unsupported network protocol\n", filename);
break;
#endif
default:
fprintf(stderr, "%s: Error while opening file\n", filename);
break;
}
}
#define PRINT_LIB_VERSION(outstream,libname,LIBNAME,indent) \
if (CONFIG_##LIBNAME) { \
unsigned int version = libname##_version(); \
fprintf(outstream, "%slib%-10s %2d.%2d.%2d / %2d.%2d.%2d\n", \
indent? " " : "", #libname, \
LIB##LIBNAME##_VERSION_MAJOR, \
LIB##LIBNAME##_VERSION_MINOR, \
LIB##LIBNAME##_VERSION_MICRO, \
version >> 16, version >> 8 & 0xff, version & 0xff); \
}
#define PRINT_LIB_VERSION(outstream,libname,LIBNAME,indent) \
version= libname##_version(); \
fprintf(outstream, "%slib%-10s %2d.%2d.%2d / %2d.%2d.%2d\n", indent? " " : "", #libname, \
LIB##LIBNAME##_VERSION_MAJOR, LIB##LIBNAME##_VERSION_MINOR, LIB##LIBNAME##_VERSION_MICRO, \
version >> 16, version >> 8 & 0xff, version & 0xff);
static void print_all_lib_versions(FILE* outstream, int indent)
{
unsigned int version;
PRINT_LIB_VERSION(outstream, avutil, AVUTIL, indent);
PRINT_LIB_VERSION(outstream, avcodec, AVCODEC, indent);
PRINT_LIB_VERSION(outstream, avformat, AVFORMAT, indent);
PRINT_LIB_VERSION(outstream, avdevice, AVDEVICE, indent);
#if CONFIG_AVFILTER
PRINT_LIB_VERSION(outstream, avfilter, AVFILTER, indent);
#endif
#if CONFIG_SWSCALE
PRINT_LIB_VERSION(outstream, swscale, SWSCALE, indent);
#endif
#if CONFIG_POSTPROC
PRINT_LIB_VERSION(outstream, postproc, POSTPROC, indent);
#endif
}
static void maybe_print_config(const char *lib, const char *cfg)
{
static int warned_cfg;
if (strcmp(FFMPEG_CONFIGURATION, cfg)) {
if (!warned_cfg) {
fprintf(stderr, " WARNING: library configuration mismatch\n");
warned_cfg = 1;
}
fprintf(stderr, " %-11s configuration: %s\n", lib, cfg);
}
}
#define PRINT_LIB_CONFIG(lib, tag, cfg) do { \
if (CONFIG_##lib) \
maybe_print_config(tag, cfg); \
} while (0)
void show_banner(void)
{
fprintf(stderr, "%s version " FFMPEG_VERSION ", Copyright (c) %d-%d the FFmpeg developers\n",
fprintf(stderr, "%s version " FFMPEG_VERSION ", Copyright (c) %d-%d Fabrice Bellard, et al.\n",
program_name, program_birth_year, this_year);
fprintf(stderr, " built on %s %s with %s %s\n",
__DATE__, __TIME__, CC_TYPE, CC_VERSION);
fprintf(stderr, " configuration: " FFMPEG_CONFIGURATION "\n");
PRINT_LIB_CONFIG(AVUTIL, "libavutil", avutil_configuration());
PRINT_LIB_CONFIG(AVCODEC, "libavcodec", avcodec_configuration());
PRINT_LIB_CONFIG(AVFORMAT, "libavformat", avformat_configuration());
PRINT_LIB_CONFIG(AVDEVICE, "libavdevice", avdevice_configuration());
PRINT_LIB_CONFIG(AVFILTER, "libavfilter", avfilter_configuration());
PRINT_LIB_CONFIG(SWSCALE, "libswscale", swscale_configuration());
PRINT_LIB_CONFIG(POSTPROC, "libpostproc", postproc_configuration());
print_all_lib_versions(stderr, 1);
fprintf(stderr, " built on " __DATE__ " " __TIME__);
#ifdef __GNUC__
fprintf(stderr, ", gcc: " __VERSION__ "\n");
#else
fprintf(stderr, ", using a non-gcc compiler\n");
#endif
}
void show_version(void) {
@@ -363,26 +309,14 @@ void show_version(void) {
void show_license(void)
{
printf(
#if CONFIG_NONFREE
printf(
"This version of %s has nonfree parts compiled in.\n"
"Therefore it is not legally redistributable.\n",
program_name
#elif CONFIG_GPLV3
"%s is free software; you can redistribute it and/or modify\n"
"it under the terms of the GNU General Public License as published by\n"
"the Free Software Foundation; either version 3 of the License, or\n"
"(at your option) any later version.\n"
"\n"
"%s is distributed in the hope that it will be useful,\n"
"but WITHOUT ANY WARRANTY; without even the implied warranty of\n"
"MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the\n"
"GNU General Public License for more details.\n"
"\n"
"You should have received a copy of the GNU General Public License\n"
"along with %s. If not, see <http://www.gnu.org/licenses/>.\n",
program_name, program_name, program_name
);
#elif CONFIG_GPL
printf(
"%s is free software; you can redistribute it and/or modify\n"
"it under the terms of the GNU General Public License as published by\n"
"the Free Software Foundation; either version 2 of the License, or\n"
@@ -397,21 +331,9 @@ void show_license(void)
"along with %s; if not, write to the Free Software\n"
"Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA\n",
program_name, program_name, program_name
#elif CONFIG_LGPLV3
"%s is free software; you can redistribute it and/or modify\n"
"it under the terms of the GNU Lesser General Public License as published by\n"
"the Free Software Foundation; either version 3 of the License, or\n"
"(at your option) any later version.\n"
"\n"
"%s is distributed in the hope that it will be useful,\n"
"but WITHOUT ANY WARRANTY; without even the implied warranty of\n"
"MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the\n"
"GNU Lesser General Public License for more details.\n"
"\n"
"You should have received a copy of the GNU Lesser General Public License\n"
"along with %s. If not, see <http://www.gnu.org/licenses/>.\n",
program_name, program_name, program_name
);
#else
printf(
"%s is free software; you can redistribute it and/or\n"
"modify it under the terms of the GNU Lesser General Public\n"
"License as published by the Free Software Foundation; either\n"
@@ -426,31 +348,20 @@ void show_license(void)
"License along with %s; if not, write to the Free Software\n"
"Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA\n",
program_name, program_name, program_name
#endif
);
}
void list_fmts(void (*get_fmt_string)(char *buf, int buf_size, int fmt), int nb_fmts)
{
int i;
char fmt_str[128];
for (i=-1; i < nb_fmts; i++) {
get_fmt_string (fmt_str, sizeof(fmt_str), i);
fprintf(stdout, "%s\n", fmt_str);
}
#endif
}
void show_formats(void)
{
AVInputFormat *ifmt=NULL;
AVOutputFormat *ofmt=NULL;
URLProtocol *up=NULL;
AVCodec *p=NULL, *p2;
AVBitStreamFilter *bsf=NULL;
const char *last_name;
printf(
"File formats:\n"
" D. = Demuxing supported\n"
" .E = Muxing supported\n"
" --\n");
printf("File formats:\n");
last_name= "000";
for(;;){
int decode=0;
@@ -487,23 +398,9 @@ void show_formats(void)
name,
long_name ? long_name:" ");
}
}
printf("\n");
void show_codecs(void)
{
AVCodec *p=NULL, *p2;
const char *last_name;
printf(
"Codecs:\n"
" D..... = Decoding supported\n"
" .E.... = Encoding supported\n"
" ..V... = Video codec\n"
" ..A... = Audio codec\n"
" ..S... = Subtitle codec\n"
" ...S.. = Supports draw_horiz_band\n"
" ....D. = Supports direct rendering method 1\n"
" .....T = Supports weird frame truncation\n"
" ------\n");
printf("Codecs:\n");
last_name= "000";
for(;;){
int decode=0;
@@ -529,13 +426,13 @@ void show_codecs(void)
last_name= p2->name;
switch(p2->type) {
case AVMEDIA_TYPE_VIDEO:
case CODEC_TYPE_VIDEO:
type_str = "V";
break;
case AVMEDIA_TYPE_AUDIO:
case CODEC_TYPE_AUDIO:
type_str = "A";
break;
case AVMEDIA_TYPE_SUBTITLE:
case CODEC_TYPE_SUBTITLE:
type_str = "S";
break;
default:
@@ -557,6 +454,19 @@ void show_codecs(void)
printf("\n");
}
printf("\n");
printf("Bitstream filters:\n");
while((bsf = av_bitstream_filter_next(bsf)))
printf(" %s", bsf->name);
printf("\n");
printf("Supported file protocols:\n");
while((up = av_protocol_next(up)))
printf(" %s:", up->name);
printf("\n");
printf("Frame size, frame rate abbreviations:\n ntsc pal qntsc qpal sntsc spal film ntsc-film sqcif qcif cif 4cif\n");
printf("\n");
printf(
"Note, the names of encoders and decoders do not always match, so there are\n"
"several cases where the above table shows encoder only or decoder only entries\n"
@@ -564,101 +474,3 @@ void show_codecs(void)
"decoder corresponds to the h263 and h263p encoders, for file formats it is even\n"
"worse.\n");
}
void show_bsfs(void)
{
AVBitStreamFilter *bsf=NULL;
printf("Bitstream filters:\n");
while((bsf = av_bitstream_filter_next(bsf)))
printf("%s\n", bsf->name);
printf("\n");
}
void show_protocols(void)
{
URLProtocol *up=NULL;
printf("Supported file protocols:\n");
while((up = av_protocol_next(up)))
printf("%s\n", up->name);
}
void show_filters(void)
{
AVFilter av_unused(**filter) = NULL;
printf("Filters:\n");
#if CONFIG_AVFILTER
while ((filter = av_filter_next(filter)) && *filter)
printf("%-16s %s\n", (*filter)->name, (*filter)->description);
#endif
}
void show_pix_fmts(void)
{
enum PixelFormat pix_fmt;
printf(
"Pixel formats:\n"
"I.... = Supported Input format for conversion\n"
".O... = Supported Output format for conversion\n"
"..H.. = Hardware accelerated format\n"
"...P. = Paletted format\n"
"....B = Bitstream format\n"
"FLAGS NAME NB_COMPONENTS BITS_PER_PIXEL\n"
"-----\n");
#if !CONFIG_SWSCALE
# define sws_isSupportedInput(x) 0
# define sws_isSupportedOutput(x) 0
#endif
for (pix_fmt = 0; pix_fmt < PIX_FMT_NB; pix_fmt++) {
const AVPixFmtDescriptor *pix_desc = &av_pix_fmt_descriptors[pix_fmt];
printf("%c%c%c%c%c %-16s %d %2d\n",
sws_isSupportedInput (pix_fmt) ? 'I' : '.',
sws_isSupportedOutput(pix_fmt) ? 'O' : '.',
pix_desc->flags & PIX_FMT_HWACCEL ? 'H' : '.',
pix_desc->flags & PIX_FMT_PAL ? 'P' : '.',
pix_desc->flags & PIX_FMT_BITSTREAM ? 'B' : '.',
pix_desc->name,
pix_desc->nb_components,
av_get_bits_per_pixel(pix_desc));
}
}
int read_yesno(void)
{
int c = getchar();
int yesno = (toupper(c) == 'Y');
while (c != '\n' && c != EOF)
c = getchar();
return yesno;
}
int read_file(const char *filename, char **bufptr, size_t *size)
{
FILE *f = fopen(filename, "rb");
if (!f) {
fprintf(stderr, "Cannot read file '%s': %s\n", filename, strerror(errno));
return AVERROR(errno);
}
fseek(f, 0, SEEK_END);
*size = ftell(f);
fseek(f, 0, SEEK_SET);
*bufptr = av_malloc(*size + 1);
if (!*bufptr) {
fprintf(stderr, "Could not allocate file buffer\n");
fclose(f);
return AVERROR(ENOMEM);
}
fread(*bufptr, 1, *size, f);
(*bufptr)[*size++] = '\0';
fclose(f);
return 0;
}

View File

@@ -40,7 +40,7 @@ extern const int program_birth_year;
extern const int this_year;
extern const char **opt_names;
extern AVCodecContext *avcodec_opts[AVMEDIA_TYPE_NB];
extern AVCodecContext *avctx_opts[CODEC_TYPE_NB];
extern AVFormatContext *avformat_opts;
extern struct SwsContext *sws_opts;
@@ -50,16 +50,6 @@ extern struct SwsContext *sws_opts;
*/
int opt_default(const char *opt, const char *arg);
/**
* Sets the libav* libraries log level.
*/
int opt_loglevel(const char *opt, const char *arg);
/**
* Limit the execution time.
*/
int opt_timelimit(const char *opt, const char *arg);
/**
* Parses a string and returns its corresponding value as a double.
* Exits from the application if the string cannot be correctly
@@ -83,8 +73,8 @@ double parse_number_or_die(const char *context, const char *numstr, int type, do
* @param context the context of the value to be set (e.g. the
* corresponding commandline option name)
* @param timestr the string to be parsed
* @param is_duration a flag which tells how to interpret timestr, if
* not zero timestr is interpreted as a duration, otherwise as a
* @param is_duration a flag which tells how to interpret \p timestr, if
* not zero \p timestr is interpreted as a duration, otherwise as a
* date
*
* @see parse_date()
@@ -134,19 +124,8 @@ void parse_options(int argc, char **argv, const OptionDef *options,
void set_context_opts(void *ctx, void *opts_ctx, int flags);
/**
* Prints an error message to stderr, indicating filename and a human
* readable description of the error code err.
*
* If strerror_r() is not available the use of this function in a
* multithreaded application may be unsafe.
*
* @see av_strerror()
*/
void print_error(const char *filename, int err);
void list_fmts(void (*get_fmt_string)(char *buf, int buf_size, int fmt), int nb_fmts);
/**
* Prints the program banner to stderr. The banner contents depend on the
* current version of the repository and of the libav* libraries used by
@@ -173,51 +152,4 @@ void show_license(void);
*/
void show_formats(void);
/**
* Prints a listing containing all the codecs supported by the
* program.
*/
void show_codecs(void);
/**
* Prints a listing containing all the filters supported by the
* program.
*/
void show_filters(void);
/**
* Prints a listing containing all the bit stream filters supported by the
* program.
*/
void show_bsfs(void);
/**
* Prints a listing containing all the protocols supported by the
* program.
*/
void show_protocols(void);
/**
* Prints a listing containing all the pixel formats supported by the
* program.
*/
void show_pix_fmts(void);
/**
* Returns a positive value if reads from standard input a line
* starting with [yY], otherwise returns 0.
*/
int read_yesno(void);
/**
* Reads the file with name filename, and puts its content in a newly
* allocated 0-terminated buffer.
*
* @param bufptr puts here the pointer to the newly allocated buffer
* @param size puts here the size of the newly allocated buffer
* @return 0 in case of success, a negative value corresponding to an
* AVERROR error code in case of failure.
*/
int read_file(const char *filename, char **bufptr, size_t *size);
#endif /* FFMPEG_CMDUTILS_H */

View File

@@ -1,13 +0,0 @@
{ "L", OPT_EXIT, {(void*)show_license}, "show license" },
{ "h", OPT_EXIT, {(void*)show_help}, "show help" },
{ "?", OPT_EXIT, {(void*)show_help}, "show help" },
{ "help", OPT_EXIT, {(void*)show_help}, "show help" },
{ "-help", OPT_EXIT, {(void*)show_help}, "show help" },
{ "version", OPT_EXIT, {(void*)show_version}, "show version" },
{ "formats" , OPT_EXIT, {(void*)show_formats }, "show available formats" },
{ "codecs" , OPT_EXIT, {(void*)show_codecs }, "show available codecs" },
{ "bsfs" , OPT_EXIT, {(void*)show_bsfs }, "show available bit stream filters" },
{ "protocols", OPT_EXIT, {(void*)show_protocols}, "show available protocols" },
{ "filters", OPT_EXIT, {(void*)show_filters }, "show available filters" },
{ "pix_fmts" , OPT_EXIT, {(void*)show_pix_fmts }, "show available pixel formats" },
{ "loglevel", HAS_ARG | OPT_FUNC2, {(void*)opt_loglevel}, "set libav* logging level", "loglevel" },

View File

@@ -2,15 +2,13 @@
# common bits used by all libraries
#
# first so "all" becomes default target
all: all-yes
all: # make "all" default target
ifndef SUBDIR
vpath %.c $(SRC_DIR)
vpath %.h $(SRC_DIR)
vpath %.S $(SRC_DIR)
vpath %.c $(SRC_DIR)
vpath %.h $(SRC_DIR)
vpath %.S $(SRC_DIR)
vpath %.asm $(SRC_DIR)
vpath %.v $(SRC_DIR)
ifeq ($(SRC_DIR),$(SRC_PATH_BARE))
BUILD_ROOT_REL = .
@@ -18,92 +16,98 @@ else
BUILD_ROOT_REL = ..
endif
ifndef V
Q = @
ECHO = printf "$(1)\t%s\n" $(2)
BRIEF = CC AS YASM AR LD HOSTCC STRIP CP
SILENT = DEPCC YASMDEP RM RANLIB
MSG = $@
M = @$(call ECHO,$(TAG),$@);
$(foreach VAR,$(BRIEF), \
$(eval $(VAR) = @$$(call ECHO,$(VAR),$$(MSG)); $($(VAR))))
$(foreach VAR,$(SILENT),$(eval $(VAR) = @$($(VAR))))
$(eval INSTALL = @$(call ECHO,INSTALL,$$(^:$(SRC_DIR)/%=%)); $(INSTALL))
endif
ALLFFLIBS = avcodec avdevice avfilter avformat avutil postproc swscale
CPPFLAGS := -I$(BUILD_ROOT_REL) -I$(SRC_PATH) $(CPPFLAGS)
CFLAGS += $(ECFLAGS)
CFLAGS := -DHAVE_AV_CONFIG_H -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE \
-I$(BUILD_ROOT_REL) -I$(SRC_PATH) $(OPTFLAGS)
%.o: %.c
$(CCDEP)
$(CC) $(CPPFLAGS) $(CFLAGS) $(CC_DEPFLAGS) -c $(CC_O) $<
$(CC) $(CFLAGS) $(LIBOBJFLAGS) -c -o $@ $<
%.o: %.S
$(ASDEP)
$(AS) $(CPPFLAGS) $(ASFLAGS) $(AS_DEPFLAGS) -c -o $@ $<
$(CC) $(CFLAGS) $(LIBOBJFLAGS) -c -o $@ $<
%.ho: %.h
$(CC) $(CPPFLAGS) $(CFLAGS) -Wno-unused -c -o $@ -x c $<
$(CC) $(CFLAGS) $(LIBOBJFLAGS) -Wno-unused -c -o $@ -x c $<
%.d: %.c
$(DEPEND_CMD) > $@
%.d: %.S
$(DEPEND_CMD) > $@
%.d: %.cpp
$(DEPEND_CMD) > $@
%.o: %.d
%$(EXESUF): %.c
%.ver: %.v
$(Q)sed 's/$$MAJOR/$($(basename $(@F))_VERSION_MAJOR)/' $^ > $@
SVN_ENTRIES = $(SRC_PATH_BARE)/.svn/entries
ifeq ($(wildcard $(SVN_ENTRIES)),$(SVN_ENTRIES))
$(BUILD_ROOT_REL)/version.h: $(SVN_ENTRIES)
endif
%.c %.h: TAG = GEN
$(BUILD_ROOT_REL)/version.h: $(SRC_PATH_BARE)/version.sh
$< $(SRC_PATH) $@ $(EXTRA_VERSION)
install: install-libs install-headers
install-libs: install-libs-yes
uninstall: uninstall-libs uninstall-headers
.PHONY: all depend dep *clean install* uninstall* examples testprogs
# Disable suffix rules. Most of the builtin rules are suffix rules,
# so this saves some time on slow systems.
.SUFFIXES:
# Do not delete intermediate files from chains of implicit rules
$(OBJS):
.PHONY: all depend dep clean distclean install* uninstall* tests
endif
OBJS-$(HAVE_MMX) += $(MMX-OBJS-yes)
CFLAGS += $(CFLAGS-yes)
OBJS += $(OBJS-yes)
FFLIBS := $(FFLIBS-yes) $(FFLIBS)
TESTPROGS += $(TESTPROGS-yes)
CFLAGS += $(CFLAGS-yes)
OBJS += $(OBJS-yes)
FFLIBS := $(FFLIBS-yes) $(FFLIBS)
TESTS += $(TESTS-yes)
FFEXTRALIBS := $(addprefix -l,$(addsuffix $(BUILDSUF),$(FFLIBS))) $(EXTRALIBS)
FFLDFLAGS := $(addprefix -L$(BUILD_ROOT)/lib,$(ALLFFLIBS)) $(LDFLAGS)
FFLDFLAGS := $(addprefix -L$(BUILD_ROOT)/lib,$(FFLIBS)) $(LDFLAGS)
EXAMPLES := $(addprefix $(SUBDIR),$(addsuffix -example$(EXESUF),$(EXAMPLES)))
OBJS := $(addprefix $(SUBDIR),$(sort $(OBJS)))
TESTOBJS := $(addprefix $(SUBDIR),$(TESTOBJS))
TESTPROGS := $(addprefix $(SUBDIR),$(addsuffix -test$(EXESUF),$(TESTPROGS)))
HOSTOBJS := $(addprefix $(SUBDIR),$(addsuffix .o,$(HOSTPROGS)))
HOSTPROGS := $(addprefix $(SUBDIR),$(addsuffix $(HOSTEXESUF),$(HOSTPROGS)))
OBJS := $(addprefix $(SUBDIR),$(OBJS))
TESTS := $(addprefix $(SUBDIR),$(TESTS))
DEP_LIBS := $(foreach NAME,$(FFLIBS),$(BUILD_ROOT_REL)/lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME))
DEP_LIBS:=$(foreach NAME,$(FFLIBS),lib$(NAME)/$($(BUILD_SHARED:yes=S)LIBNAME))
ALLHEADERS := $(subst $(SRC_DIR)/,$(SUBDIR),$(wildcard $(SRC_DIR)/*.h $(SRC_DIR)/$(ARCH)/*.h))
SKIPHEADERS += $(addprefix $(ARCH)/,$(ARCH_HEADERS))
SKIPHEADERS := $(addprefix $(SUBDIR),$(SKIPHEADERS-) $(SKIPHEADERS))
checkheaders: $(filter-out $(SKIPHEADERS:.h=.ho),$(ALLHEADERS:.h=.ho))
$(HOSTOBJS): %.o: %.c
$(HOSTCC) $(HOSTCFLAGS) -c -o $@ $<
$(HOSTPROGS): %$(HOSTEXESUF): %.o
$(HOSTCC) $(HOSTLDFLAGS) -o $@ $< $(HOSTLIBS)
checkheaders: $(filter-out %_template.ho,$(ALLHEADERS:.h=.ho))
DEPS := $(OBJS:.o=.d)
depend dep: $(DEPS)
CLEANSUFFIXES = *.d *.o *~ *.ho *.map *.ver
DISTCLEANSUFFIXES = *.pc
LIBSUFFIXES = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a *.exp
CLEANSUFFIXES = *.o *~ *.ho
LIBSUFFIXES = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a *.exp *.map
DISTCLEANSUFFIXES = *.d *.pc
-include $(wildcard $(DEPS))
define RULES
$(SUBDIR)%$(EXESUF): $(SUBDIR)%.o
$(CC) $(FFLDFLAGS) -o $$@ $$^ $(SUBDIR)$(LIBNAME) $(FFEXTRALIBS)
$(SUBDIR)%-test.o: $(SUBDIR)%.c
$(CC) $(CFLAGS) -DTEST -c -o $$@ $$^
$(SUBDIR)%-test.o: $(SUBDIR)%-test.c
$(CC) $(CFLAGS) -DTEST -c -o $$@ $$^
$(SUBDIR)x86/%.o: $(SUBDIR)x86/%.asm
$(YASM) $(YASMFLAGS) -I $$(<D)/ -o $$@ $$<
$(SUBDIR)x86/%.d: $(SUBDIR)x86/%.asm
$(YASM) $(YASMFLAGS) -I $$(<D)/ -M -o $$(@:%.d=%.o) $$< > $$@
clean::
rm -f $(TESTS) $(addprefix $(SUBDIR),$(CLEANFILES) $(CLEANSUFFIXES) $(LIBSUFFIXES)) \
$(addprefix $(SUBDIR), $(foreach suffix,$(CLEANSUFFIXES),$(addsuffix /$(suffix),$(DIRS))))
distclean:: clean
rm -f $(addprefix $(SUBDIR),$(DISTCLEANSUFFIXES)) \
$(addprefix $(SUBDIR), $(foreach suffix,$(DISTCLEANSUFFIXES),$(addsuffix /$(suffix),$(DIRS))))
endef
$(eval $(RULES))
tests: $(TESTS)
-include $(DEPS)

2439
configure vendored

File diff suppressed because it is too large Load Diff

View File

@@ -1,229 +1,7 @@
Never assume the API of libav* to be stable unless at least 1 week has passed since
the last major version increase.
The last version increases were:
libavcodec: ?
libavdevice: ?
libavfilter: 2009-10-18
libavformat: ?
libpostproc: ?
libswscale: ?
libavutil: 2009-03-08
API changes, most recent first:
2010-06-01 - r31301 - lsws 0.11.0 - convertPalette API
Add sws_convertPalette8ToPacked32 and sws_convertPalette8ToPacked24
2010-05-26 - r23334 - lavc 52.72.0 - CODEC_CAP_EXPERIMENTAL
Add CODEC_CAP_EXPERIMENTAL flag.
2010-05-18 - r23161 - lavf 52.63.0 - AVFMT_FLAG_RTP_HINT
Add AVFMT_FLAG_RTP_HINT as possible value for AVFormatContext.flags
2010-05-01 - r23002 - 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.
2010-04-01 - r22806 - lavf 52.61.0 - metadata API
Add a flag for av_metadata_set2() to disable overwriting of
existing tags.
2010-04-01 - r22753 - lavc 52.66.0
Add avcodec_get_edge_width()
2010-03-31 - r22750 - lavc 52.65.0
Add avcodec_copy_context().
2010-03-31 - r22748 - lavf 52.60.0 - av_match_ext()
Make av_match_ext() public.
2010-03-31 - r22736 - lavu 50.14.0 - AVMediaType
Move AVMediaType enum from libavcodec to libavutil.
2010-03-31 - r22735 - lavc 52.64.0 - AVMediaType
Define AVMediaType enum, and use it instead of enum CodecType, which
is deprecated and will be dropped at the next major bump.
2010-03-25 - r22684 - lavu 50.13.0 - av_strerror()
Implement av_strerror().
2010-03-23 - r22649 - lavc 52.60.0 - av_dct_init()
Support DCT-I and DST-I
2010-03-15 - r22540 - lavf 52.56.0 - AVFormatContext.start_time_realtime
Add AVFormatContext.start_time_realtime field.
2010-03-13 - r22506 - lavfi 1.18.0 - AVFilterPicRef.pos
Add AVFilterPicRef.pos field.
2010-03-13 - r22501 - lavu 50.12.0 - error.h
Move error code definitions from libavcodec/avcodec.h to
the new public header libavutil/error.h.
2010-03-07 - r22291 - lavc 52.56.0 - avfft.h
Add public FFT interface.
2010-03-06 - r22251 - lavu 50.11.0 - av_stristr()
Add av_stristr().
2010-03-03 - r22174 - lavu 50.10.0 - av_tree_enumerate()
Add av_tree_enumerate().
2010-02-07 - r21673 - lavu 50.9.0 - av_compare_ts()
Add av_compare_ts().
2010-02-05 - r30513 - lsws 0.10.0 - sws_getCoefficients()
Add sws_getCoefficients().
2010-02-01 - r21587 - lavf 52.50.0 - metadata API
Add a list of generic tag names, change 'author' -> 'artist',
'year' -> 'date'.
2010-01-30 - r21545 - lavu 50.8.0 - av_get_pix_fmt()
Add av_get_pix_fmt().
2010-01-21 - r30381 - lsws 0.9.0 - sws_scale
Change constness attributes of sws_scale() parameters.
2010-01-10 - r21121 - lavfi 1.15.0 - avfilter_graph_config_links()
Add a log_ctx parameter to avfilter_graph_config_links().
2010-01-07 - r30236 - lsws 0.8.0 - sws_isSupported{In,Out}put
Add sws_isSupportedInput() and sws_isSupportedOutput() functions.
2010-01-06 - r21035 - lavfi 1.14.0 - avfilter_add_colorspace()
Change the avfilter_add_colorspace() signature, make it accept an
(AVFilterFormats **) rather than an (AVFilterFormats *) as before.
2010-01-03 - r21007 - lavfi 1.13.0 - avfilter_add_colorspace()
Add avfilter_add_colorspace().
2010-01-02 - r20998 - lavf 52.46.0 - av_match_ext()
Add av_match_ext(), it should be used in place of match_ext().
2010-01-01 - r20991 - lavf 52.45.0 - av_guess_format()
Add av_guess_format(), it should be used in place of guess_format().
2009-12-13 - r20834 - lavf 52.43.0 - metadata API
Add av_metadata_set2(), AV_METADATA_DONT_STRDUP_KEY and AV_METADATA_DONT_STRDUP_VAL.
2009-12-13 - r20829 - lavu 50.7.0 - avstring.h API
Add av_d2str().
2009-12-13 - r20826 - lavc 52.42.0 - AVStream
Add avg_frame_rate.
2009-12-12 - r20808 - lavu 50.6.0 - av_bmg_next()
Introduce the av_bmg_next() function.
2009-12-05 - r20734 - lavfi 1.12.0 - avfilter_draw_slice()
Add a slice_dir parameter to avfilter_draw_slice().
2009-11-26 - r20611 - lavfi 1.11.0 - AVFilter
Remove the next field from AVFilter, this is not anymore required.
2009-11-25 - r20607 - lavfi 1.10.0 - avfilter_next()
Introduce the avfilter_next() function.
2009-11-25 - r20605 - lavfi 1.9.0 - avfilter_register()
Change the signature of avfilter_register() to make it return an
int. This is required since now the registration operation may fail.
2009-11-25 - r20603 - lavu 50.5.0 - pixdesc.h API
Make the pixdesc.h API public.
2009-10-27 - r20385 - lavfi 1.5.0 - AVFilter.next
Add a next field to AVFilter, this is used for simplifying the
registration and management of the registered filters.
2009-10-23 - r20356 - lavfi 1.4.1 - AVFilter.description
Add a description field to AVFilter.
2009-10-19 - r20302 - lavfi 1.3.0 - avfilter_make_format_list()
Change the interface of avfilter_make_format_list() from
avfilter_make_format_list(int n, ...) to
avfilter_make_format_list(enum PixelFormat *pix_fmts).
2009-10-18 - r20272 - lavfi 1.0.0 - avfilter_get_video_buffer()
Make avfilter_get_video_buffer() recursive and add the w and h
parameters to it.
2009-10-07 - r20189 - lavfi 0.5.1 - AVFilterPic
Add w and h fields to AVFilterPic.
2009-06-22 - r19250 - lavf 52.34.1 - AVFormatContext.packet_size
This is now an unsigned int instead of a signed int.
2009-06-19 - r19222 - lavc 52.32.0 - AVSubtitle.pts
Add a pts field to AVSubtitle which gives the subtitle packet pts
in AV_TIME_BASE. Some subtitle de-/encoders (e.g. XSUB) will
not work right without this.
2009-06-03 - r19078 - lavc 52.30.2 - AV_PKT_FLAG_KEY
PKT_FLAG_KEY has been deprecated and will be dropped at the next
major version. Use AV_PKT_FLAG_KEY instead.
2009-06-01 - r19025 - lavc 52.30.0 - av_lockmgr_register()
av_lockmgr_register() can be used to register a callback function
that lavc (and in the future, libraries that depend on lavc) can use
to implement mutexes. The application should provide a callback function
that implements the AV_LOCK_* operations described in avcodec.h.
When the lock manager is registered, FFmpeg is guaranteed to behave
correctly in a multi-threaded application.
2009-04-30 - r18719 - lavc 52.28.0 - av_free_packet
av_free_packet() is no longer an inline function. It is now exported.
2009-04-11 - r18431 - lavc 52.25.0 - deprecate av_destruct_packet_nofree
Please use NULL instead. This has been supported since r16506
(lavf > 52.23.1, lavc > 52.10.0).
2009-04-07 - r18351 - lavc 52.23.0 - avcodec_decode_video/audio/subtitle
The old decoding functions are deprecated, all new code should use the
new functions avcodec_decode_video2(), avcodec_decode_audio3() and
avcodec_decode_subtitle2(). These new functions take an AVPacket *pkt
argument instead of a const uint8_t *buf / int buf_size pair.
2009-04-03 - r18321 - lavu 50.3.0 - av_fifo_space
Introduce the av_fifo_space() function.
2009-04-02 - r18317 - lavc 52.23.0 - AVPacket
Move AVPacket declaration from libavformat/avformat.h to
libavcodec/avcodec.h.
2009-03-22 - r18163 - lavu 50.2.0 - RGB32 pixel formats
Convert the pixel formats PIX_FMT_ARGB, PIX_FMT_RGBA, PIX_FMT_ABGR,
PIX_FMT_BGRA, which were defined as macros, into enum PixelFormat values.
Conversely PIX_FMT_RGB32, PIX_FMT_RGB32_1, PIX_FMT_BGR32 and
PIX_FMT_BGR32_1 are now macros.
avcodec_get_pix_fmt() now recognizes the "rgb32" and "bgr32" aliases.
Re-sort the enum PixelFormat list accordingly.
This change breaks API/ABI backward compatibility.
2009-03-22 - r18133 - lavu 50.1.0 - PIX_FMT_RGB5X5 endian variants
Add the enum PixelFormat values:
PIX_FMT_RGB565BE, PIX_FMT_RGB565LE, PIX_FMT_RGB555BE, PIX_FMT_RGB555LE,
PIX_FMT_BGR565BE, PIX_FMT_BGR565LE, PIX_FMT_BGR555BE, PIX_FMT_BGR555LE.
2009-03-21 - r18116 - lavu 50.0.0 - av_random*
The Mersenne Twister PRNG implemented through the av_random* functions
was removed. Use the lagged Fibonacci PRNG through the av_lfg* functions
instead.
2009-03-08 - r17869 - lavu 50.0.0 - AVFifoBuffer
av_fifo_init, av_fifo_read, av_fifo_write and av_fifo_realloc were dropped
and replaced by av_fifo_alloc, av_fifo_generic_read, av_fifo_generic_write
and av_fifo_realloc2.
In addition, the order of the function arguments of av_fifo_generic_read
was changed to match av_fifo_generic_write.
The AVFifoBuffer/struct AVFifoBuffer may only be used in an opaque way by
applications, they may not use sizeof() or directly access members.
2009-03-01 - r17682 - lavf 52.31.0 - Generic metadata API
Introduce a new metadata API (see av_metadata_get() and friends).
The old API is now deprecated and should not be used anymore. This especially
includes the following structure fields:
20090301 - r17682 - lavf 52.31.0 - Generic metadata API
This version introduce a new metadata API (see av_metadata_get() and friends).
The old API is now deprecated and shouldn't be used anymore. This especially
include the following structure fields:
- AVFormatContext.title
- AVFormatContext.author
- AVFormatContext.copyright

View File

@@ -81,6 +81,7 @@ unassigned TODO: (unordered)
- add support for using mplayers video filters to ffmpeg
- H264 encoder
- per MB ratecontrol (so VCD and such do work better)
- replace/rewrite libavcodec/fdctref.c
- write a script which iteratively changes all functions between always_inline and noinline and benchmarks the result to find the best set of inlined functions
- convert all the non SIMD asm into small asm vs. C testcases and submit them to the gcc devels so they can improve gcc
- generic audio mixing API

View File

@@ -1,436 +0,0 @@
\input texinfo @c -*- texinfo -*-
@settitle Developer Documentation
@titlepage
@sp 7
@center @titlefont{Developer Documentation}
@sp 3
@end titlepage
@chapter Developers Guide
@section API
@itemize @bullet
@item libavcodec is the library containing the codecs (both encoding and
decoding). Look at @file{libavcodec/apiexample.c} to see how to use it.
@item libavformat is the library containing the file format handling (mux and
demux code for several formats). Look at @file{ffplay.c} to use it in a
player. See @file{libavformat/output-example.c} to use it to generate
audio or video streams.
@end itemize
@section Integrating libavcodec or libavformat in your program
You can integrate all the source code of the libraries to link them
statically to avoid any version problem. All you need is to provide a
'config.mak' and a 'config.h' in the parent directory. See the defines
generated by ./configure to understand what is needed.
You can use libavcodec or libavformat in your commercial program, but
@emph{any patch you make must be published}. The best way to proceed is
to send your patches to the FFmpeg mailing list.
@anchor{Coding Rules}
@section Coding Rules
FFmpeg is programmed in the ISO C90 language with a few additional
features from ISO C99, namely:
@itemize @bullet
@item
the @samp{inline} keyword;
@item
@samp{//} comments;
@item
designated struct initializers (@samp{struct s x = @{ .i = 17 @};})
@item
compound literals (@samp{x = (struct s) @{ 17, 23 @};})
@end itemize
These features are supported by all compilers we care about, so we will not
accept patches to remove their use unless they absolutely do not impair
clarity and performance.
All code must compile with GCC 2.95 and GCC 3.3. Currently, FFmpeg also
compiles with several other compilers, such as the Compaq ccc compiler
or Sun Studio 9, and we would like to keep it that way unless it would
be exceedingly involved. To ensure compatibility, please do not use any
additional C99 features or GCC extensions. Especially watch out for:
@itemize @bullet
@item
mixing statements and declarations;
@item
@samp{long long} (use @samp{int64_t} instead);
@item
@samp{__attribute__} not protected by @samp{#ifdef __GNUC__} or similar;
@item
GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}).
@end itemize
Indent size is 4.
The presentation is one inspired by 'indent -i4 -kr -nut'.
The TAB character is forbidden outside of Makefiles as is any
form of trailing whitespace. Commits containing either will be
rejected by the Subversion repository.
The main priority in FFmpeg is simplicity and small code size in order to
minimize the bug count.
Comments: Use the JavaDoc/Doxygen
format (see examples below) so that code documentation
can be generated automatically. All nontrivial functions should have a comment
above them explaining what the function does, even if it is just one sentence.
All structures and their member variables should be documented, too.
@example
/**
* @@file mpeg.c
* MPEG codec.
* @@author ...
*/
/**
* Summary sentence.
* more text ...
* ...
*/
typedef struct Foobar@{
int var1; /**< var1 description */
int var2; ///< var2 description
/** var3 description */
int var3;
@} Foobar;
/**
* Summary sentence.
* more text ...
* ...
* @@param my_parameter description of my_parameter
* @@return return value description
*/
int myfunc(int my_parameter)
...
@end example
fprintf and printf are forbidden in libavformat and libavcodec,
please use av_log() instead.
Casts should be used only when necessary. Unneeded parentheses
should also be avoided if they don't make the code easier to understand.
@section Development Policy
@enumerate
@item
Contributions should be licensed under the LGPL 2.1, including an
"or any later version" clause, or the MIT license. GPL 2 including
an "or any later version" clause is also acceptable, but LGPL is
preferred.
@item
You must not commit code which breaks FFmpeg! (Meaning unfinished but
enabled code which breaks compilation or compiles but does not work or
breaks the regression tests)
You can commit unfinished stuff (for testing etc), but it must be disabled
(#ifdef etc) by default so it does not interfere with other developers'
work.
@item
You do not have to over-test things. If it works for you, and you think it
should work for others, then commit. If your code has problems
(portability, triggers compiler bugs, unusual environment etc) they will be
reported and eventually fixed.
@item
Do not commit unrelated changes together, split them into self-contained
pieces. Also do not forget that if part B depends on part A, but A does not
depend on B, then A can and should be committed first and separate from B.
Keeping changes well split into self-contained parts makes reviewing and
understanding them on the commit log mailing list easier. This also helps
in case of debugging later on.
Also if you have doubts about splitting or not splitting, do not hesitate to
ask/discuss it on the developer mailing list.
@item
Do not change behavior of the program (renaming options etc) without
first discussing it on the ffmpeg-devel mailing list. Do not remove
functionality from the code. Just improve!
Note: Redundant code can be removed.
@item
Do not commit changes to the build system (Makefiles, configure script)
which change behavior, defaults etc, without asking first. The same
applies to compiler warning fixes, trivial looking fixes and to code
maintained by other developers. We usually have a reason for doing things
the way we do. Send your changes as patches to the ffmpeg-devel mailing
list, and if the code maintainers say OK, you may commit. This does not
apply to files you wrote and/or maintain.
@item
We refuse source indentation and other cosmetic changes if they are mixed
with functional changes, such commits will be rejected and removed. Every
developer has his own indentation style, you should not change it. Of course
if you (re)write something, you can use your own style, even though we would
prefer if the indentation throughout FFmpeg was consistent (Many projects
force a given indentation style - we do not.). If you really need to make
indentation changes (try to avoid this), separate them strictly from real
changes.
NOTE: If you had to put if()@{ .. @} over a large (> 5 lines) chunk of code,
then either do NOT change the indentation of the inner part within (do not
move it to the right)! or do so in a separate commit
@item
Always fill out the commit log message. Describe in a few lines what you
changed and why. You can refer to mailing list postings if you fix a
particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.
@item
If you apply a patch by someone else, include the name and email address in
the log message. Since the ffmpeg-cvslog mailing list is publicly
archived you should add some SPAM protection to the email address. Send an
answer to ffmpeg-devel (or wherever you got the patch from) saying that
you applied the patch.
@item
When applying patches that have been discussed (at length) on the mailing
list, reference the thread in the log message.
@item
Do NOT commit to code actively maintained by others without permission.
Send a patch to ffmpeg-devel instead. If no one answers within a reasonable
timeframe (12h for build failures and security fixes, 3 days small changes,
1 week for big patches) then commit your patch if you think it is OK.
Also note, the maintainer can simply ask for more time to review!
@item
Subscribe to the ffmpeg-cvslog mailing list. The diffs of all commits
are sent there and reviewed by all the other developers. Bugs and possible
improvements or general questions regarding commits are discussed there. We
expect you to react if problems with your code are uncovered.
@item
Update the documentation if you change behavior or add features. If you are
unsure how best to do this, send a patch to ffmpeg-devel, the documentation
maintainer(s) will review and commit your stuff.
@item
Try to keep important discussions and requests (also) on the public
developer mailing list, so that all developers can benefit from them.
@item
Never write to unallocated memory, never write over the end of arrays,
always check values read from some untrusted source before using them
as array index or other risky things.
@item
Remember to check if you need to bump versions for the specific libav
parts (libavutil, libavcodec, libavformat) you are changing. You need
to change the version integer.
Incrementing the first component means no backward compatibility to
previous versions (e.g. removal of a function from the public API).
Incrementing the second component means backward compatible change
(e.g. addition of a function to the public API or extension of an
existing data structure).
Incrementing the third component means a noteworthy binary compatible
change (e.g. encoder bug fix that matters for the decoder).
@item
Compiler warnings indicate potential bugs or code with bad style. If a type of
warning always points to correct and clean code, that warning should
be disabled, not the code changed.
Thus the remaining warnings can either be bugs or correct code.
If it is a bug, the bug has to be fixed. If it is not, the code should
be changed to not generate a warning unless that causes a slowdown
or obfuscates the code.
@item
If you add a new file, give it a proper license header. Do not copy and
paste it from a random place, use an existing file as template.
@end enumerate
We think our rules are not too hard. If you have comments, contact us.
Note, these rules are mostly borrowed from the MPlayer project.
@section Submitting patches
First, (@pxref{Coding Rules}) above if you did not yet.
When you submit your patch, try to send a unified diff (diff '-up'
option). We cannot read other diffs :-)
Also please do not submit a patch which contains several unrelated changes.
Split it into separate, self-contained pieces. This does not mean splitting
file by file. Instead, make the patch as small as possible while still
keeping it as a logical unit that contains an individual change, even
if it spans multiple files. This makes reviewing your patches much easier
for us and greatly increases your chances of getting your patch applied.
Use the patcheck tool of FFmpeg to check your patch.
The tool is located in the tools directory.
Run the regression tests before submitting a patch so that you can
verify that there are no big problems.
Patches should be posted as base64 encoded attachments (or any other
encoding which ensures that the patch will not be trashed during
transmission) to the ffmpeg-devel mailing list, see
@url{http://lists.mplayerhq.hu/mailman/listinfo/ffmpeg-devel}
It also helps quite a bit if you tell us what the patch does (for example
'replaces lrint by lrintf'), and why (for example '*BSD isn't C99 compliant
and has no lrint()')
Also please if you send several patches, send each patch as a separate mail,
do not attach several unrelated patches to the same mail.
Your patch will be reviewed on the mailing list. You will likely be asked
to make some changes and are expected to send in an improved version that
incorporates the requests from the review. This process may go through
several iterations. Once your patch is deemed good enough, some developer
will pick it up and commit it to the official FFmpeg tree.
Give us a few days to react. But if some time passes without reaction,
send a reminder by email. Your patch should eventually be dealt with.
@section New codecs or formats checklist
@enumerate
@item
Did you use av_cold for codec initialization and close functions?
@item
Did you add a long_name under NULL_IF_CONFIG_SMALL to the AVCodec or
AVInputFormat/AVOutputFormat struct?
@item
Did you bump the minor version number in @file{avcodec.h} or
@file{avformat.h}?
@item
Did you register it in @file{allcodecs.c} or @file{allformats.c}?
@item
Did you add the CodecID to @file{avcodec.h}?
@item
If it has a fourcc, did you add it to @file{libavformat/riff.c},
even if it is only a decoder?
@item
Did you add a rule to compile the appropriate files in the Makefile?
Remember to do this even if you're just adding a format to a file that is
already being compiled by some other rule, like a raw demuxer.
@item
Did you add an entry to the table of supported formats or codecs in
@file{doc/general.texi}?
@item
Did you add an entry in the Changelog?
@item
If it depends on a parser or a library, did you add that dependency in
configure?
@item
Did you "svn add" the appropriate files before commiting?
@end enumerate
@section patch submission checklist
@enumerate
@item
Do the regression tests pass with the patch applied?
@item
Does @code{make checkheaders} pass with the patch applied?
@item
Is the patch a unified diff?
@item
Is the patch against latest FFmpeg SVN?
@item
Are you subscribed to ffmpeg-dev?
(the list is subscribers only due to spam)
@item
Have you checked that the changes are minimal, so that the same cannot be
achieved with a smaller patch and/or simpler final code?
@item
If the change is to speed critical code, did you benchmark it?
@item
If you did any benchmarks, did you provide them in the mail?
@item
Have you checked that the patch does not introduce buffer overflows or
other security issues?
@item
Did you test your decoder or demuxer against damaged data? If no, see
tools/trasher and the noise bitstream filter. Your decoder or demuxer
should not crash or end in a (near) infinite loop when fed damaged data.
@item
Is the patch created from the root of the source tree, so it can be
applied with @code{patch -p0}?
@item
Does the patch not mix functional and cosmetic changes?
@item
Did you add tabs or trailing whitespace to the code? Both are forbidden.
@item
Is the patch attached to the email you send?
@item
Is the mime type of the patch correct? It should be text/x-diff or
text/x-patch or at least text/plain and not application/octet-stream.
@item
If the patch fixes a bug, did you provide a verbose analysis of the bug?
@item
If the patch fixes a bug, did you provide enough information, including
a sample, so the bug can be reproduced and the fix can be verified?
Note please do not attach samples >100k to mails but rather provide a
URL, you can upload to ftp://upload.ffmpeg.org
@item
Did you provide a verbose summary about what the patch does change?
@item
Did you provide a verbose explanation why it changes things like it does?
@item
Did you provide a verbose summary of the user visible advantages and
disadvantages if the patch is applied?
@item
Did you provide an example so we can verify the new feature added by the
patch easily?
@item
If you added a new file, did you insert a license header? It should be
taken from FFmpeg, not randomly copied and pasted from somewhere else.
@item
You should maintain alphabetical order in alphabetically ordered lists as
long as doing so does not break API/ABI compatibility.
@item
Lines with similar content should be aligned vertically when doing so
improves readability.
@item
Did you provide a suggestion for a clear commit log message?
@end enumerate
@section Patch review process
All patches posted to ffmpeg-devel will be reviewed, unless they contain a
clear note that the patch is not for SVN.
Reviews and comments will be posted as replies to the patch on the
mailing list. The patch submitter then has to take care of every comment,
that can be by resubmitting a changed patch or by discussion. Resubmitted
patches will themselves be reviewed like any other patch. If at some point
a patch passes review with no comments then it is approved, that can for
simple and small patches happen immediately while large patches will generally
have to be changed and reviewed many times before they are approved.
After a patch is approved it will be committed to the repository.
We will review all submitted patches, but sometimes we are quite busy so
especially for large patches this can take several weeks.
When resubmitting patches, please do not make any significant changes
not related to the comments received during review. Such patches will
be rejected. Instead, submit significant changes or new features as
separate patches.
@section Regression tests
Before submitting a patch (or committing to the repository), you should at least
test that you did not break anything.
The regression tests build a synthetic video stream and a synthetic
audio stream. These are then encoded and decoded with all codecs or
formats. The CRC (or MD5) of each generated file is recorded in a
result file. A 'diff' is launched to compare the reference results and
the result file. The output is checked immediately after each test
has run.
The regression tests then go on to test the FFserver code with a
limited set of streams. It is important that this step runs correctly
as well.
Run 'make test' to test all the codecs and formats. Commands like
'make regtest-mpeg2' can be used to run a single test. By default,
make will abort if any test fails. To run all tests regardless,
use make -k. To get a more verbose output, use 'make V=1 test' or
'make V=2 test'.
Run 'make fulltest' to test all the codecs, formats and FFserver.
[Of course, some patches may change the results of the regression tests. In
this case, the reference results of the regression tests shall be modified
accordingly].
@bye

View File

@@ -138,25 +138,6 @@ Notice that @samp{%d} is replaced by the image number.
@file{img%03d.jpg} means the sequence @file{img001.jpg}, @file{img002.jpg}, etc...
If you have large number of pictures to rename, you can use the
following command to ease the burden. The command, using the bourne
shell syntax, symbolically links all files in the current directory
that match @code{*jpg} to the @file{/tmp} directory in the sequence of
@file{img001.jpg}, @file{img002.jpg} and so on.
@example
x=1; for i in *jpg; do counter=$(printf %03d $x); ln "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done
@end example
If you want to sequence them by oldest modified first, substitute
@code{$(ls -r -t *jpg)} in place of @code{*jpg}.
Then run:
@example
ffmpeg -f image2 -i /tmp/img%03d.jpg /tmp/a.mpg
@end example
The same logic is used for any image format that ffmpeg reads.
@section How do I encode movie to single pictures?
@@ -195,6 +176,14 @@ quite logical that there is a small reduction of quality. This is not a bug.
Use @file{-} as file name.
@section Why does FFmpeg not decode audio in VOB files?
The audio is AC-3 (a.k.a. A/52). AC-3 decoding is an optional component in FFmpeg
as the component that handles AC-3 decoding is currently released under the GPL.
Enable AC-3 decoding with @code{./configure --enable-gpl}. Take care: By
enabling AC-3, you automatically change the license of libavcodec from
LGPL to GPL.
@section Why does the chrominance data seem to be sampled at a different time from the luminance data on bt8x8 captures on Linux?
This is a well-known bug in the bt8x8 driver. For 2.4.26 there is a patch at
@@ -237,7 +226,7 @@ default.
@item non-working stuff
B-frames
@item example command line
ffmpeg -i input -acodec libfaac -ab 128kb -vcodec mpeg4 -b 1200kb -mbd 2 -flags +4mv -trellis 2 -aic 2 -cmp 2 -subcmp 2 -s 320x180 -metadata title=X output.mp4
ffmpeg -i input -acodec libfaac -ab 128kb -vcodec mpeg4 -b 1200kb -mbd 2 -flags +4mv -trellis 2 -aic 2 -cmp 2 -subcmp 2 -s 320x180 -title X output.mp4
@end table
@section How do I encode videos which play on the PSP?
@@ -250,7 +239,7 @@ ffmpeg -i input -acodec libfaac -ab 128kb -vcodec mpeg4 -b 1200kb -mbd 2 -flags
@item non-working stuff
B-frames
@item example command line
ffmpeg -i input -acodec libfaac -ab 128kb -vcodec mpeg4 -b 1200kb -ar 24000 -mbd 2 -flags +4mv -trellis 2 -aic 2 -cmp 2 -subcmp 2 -s 368x192 -r 30000/1001 -metadata title=X -f psp output.mp4
ffmpeg -i input -acodec libfaac -ab 128kb -vcodec mpeg4 -b 1200kb -ar 24000 -mbd 2 -flags +4mv -trellis 2 -aic 2 -cmp 2 -subcmp 2 -s 368x192 -r 30000/1001 -title X -f psp output.mp4
@item needed stuff for H.264
-acodec libfaac -vcodec libx264 width*height<=76800 width%16=0? height%16=0? -ar 48000 -coder 1 -r 30000/1001 or 15000/1001 -f psp
@item working stuff for H.264
@@ -258,7 +247,7 @@ title, loop filter
@item non-working stuff for H.264
CAVLC
@item example command line
ffmpeg -i input -acodec libfaac -ab 128kb -vcodec libx264 -b 1200kb -ar 48000 -mbd 2 -coder 1 -cmp 2 -subcmp 2 -s 368x192 -r 30000/1001 -metadata title=X -f psp -flags loop -trellis 2 -partitions parti4x4+parti8x8+partp4x4+partp8x8+partb8x8 output.mp4
ffmpeg -i input -acodec libfaac -ab 128kb -vcodec libx264 -b 1200kb -ar 48000 -mbd 2 -coder 1 -cmp 2 -subcmp 2 -s 368x192 -r 30000/1001 -title X -f psp -flags loop -trellis 2 -partitions parti4x4+parti8x8+partp4x4+partp8x8+partb8x8 output.mp4
@item higher resolution for newer PSP firmwares, width<=480, height<=272
-vcodec libx264 -level 21 -coder 1 -f psp
@item example command line
@@ -335,10 +324,6 @@ ffmpeg -f mpeg -i - -sameq -vcodec mpeg4 -acodec libmp3lame output.avi
Similarly, the yuv4mpegpipe format, and the raw video, raw audio codecs also
allow concatenation, and the transcoding step is almost lossless.
When using multiple yuv4mpegpipe(s), the first line needs to be discarded
from all but the first stream. This can be accomplished by piping through
@code{tail} as seen below. Note that when piping through @code{tail} you
must use command grouping, @code{@{ ;@}}, to background properly.
For example, let's say we want to join two FLV files into an output.flv file:
@@ -352,7 +337,7 @@ mkfifo all.v
ffmpeg -i input1.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 - > temp1.a < /dev/null &
ffmpeg -i input2.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 - > temp2.a < /dev/null &
ffmpeg -i input1.flv -an -f yuv4mpegpipe - > temp1.v < /dev/null &
@{ ffmpeg -i input2.flv -an -f yuv4mpegpipe - < /dev/null | tail -n +2 > temp2.v ; @} &
ffmpeg -i input2.flv -an -f yuv4mpegpipe - > temp2.v < /dev/null &
cat temp1.a temp2.a > all.a &
cat temp1.v temp2.v > all.v &
ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 44100 -i all.a \

View File

@@ -141,9 +141,8 @@ to get the desired audio language.
NOTE: To see the supported input formats, use @code{ffmpeg -formats}.
* You can extract images from a video, or create a video from many images:
* You can extract images from a video:
For extracting images from a video:
@example
ffmpeg -i foo.avi -r 1 -s WxH -f image2 foo-%03d.jpeg
@end example
@@ -152,20 +151,15 @@ This will extract one video frame per second from the video and will
output them in files named @file{foo-001.jpeg}, @file{foo-002.jpeg},
etc. Images will be rescaled to fit the new WxH values.
If you want to extract just a limited number of frames, you can use the
above command in combination with the -vframes or -t option, or in
combination with -ss to start extracting from a certain point in time.
For creating a video from many images:
@example
ffmpeg -f image2 -i foo-%03d.jpeg -r 12 -s WxH foo.avi
@end example
The syntax @code{foo-%03d.jpeg} specifies to use a decimal number
composed of three digits padded with zeroes to express the sequence
number. It is the same syntax supported by the C printf function, but
only formats accepting a normal integer are suitable.
If you want to extract just a limited number of frames, you can use the
above command in combination with the -vframes or -t option, or in
combination with -ss to start extracting from a certain point in time.
* You can put many streams of the same type in the output:
@example
@@ -222,12 +216,36 @@ specified for the inputs.
@c man end
@c man begin OPTIONS
@include fftools-common-opts.texi
@section Main options
@table @option
@item -L
Show license.
@item -h
Show help.
@item -version
Show version.
@item -formats
Show available formats, codecs, bitstream filters, protocols, and frame size and frame rate abbreviations.
The fields preceding the format and codec names have the following meanings:
@table @samp
@item D
Decoding available
@item E
Encoding available
@item V/A/S
Video/audio/subtitle codec
@item S
Codec supports slices
@item D
Codec supports direct rendering
@item T
Codec can handle input truncated at random locations instead of only at frame boundaries
@end table
@item -f @var{fmt}
Force format.
@@ -258,16 +276,29 @@ The offset is added to the timestamps of the input files.
Specifying a positive offset means that the corresponding
streams are delayed by 'offset' seconds.
@item -title @var{string}
Set the title.
@item -timestamp @var{time}
Set the timestamp.
@item -metadata @var{key}=@var{value}
Set a metadata key/value pair.
@item -author @var{string}
Set the author.
For example, for setting the title in the output file:
@example
ffmpeg -i in.avi -metadata title="my title" out.flv
@end example
@item -copyright @var{string}
Set the copyright.
@item -comment @var{string}
Set the comment.
@item -album @var{string}
Set the album.
@item -track @var{number}
Set the track.
@item -year @var{number}
Set the year.
@item -v @var{number}
Set the logging verbosity level.
@@ -323,8 +354,6 @@ The following abbreviations are recognized:
352x288
@item 4cif
704x576
@item 16cif
1408x1152
@item qqvga
160x120
@item qvga
@@ -447,9 +476,6 @@ stream.
@item -newvideo
Add a new video stream to the current output stream.
@item -vlang @var{code}
Set the ISO 639 language code (3 letters) of the current video stream.
@end table
@section Advanced Video Options
@@ -459,7 +485,7 @@ Set the ISO 639 language code (3 letters) of the current video stream.
Set pixel format. Use 'list' as parameter to show all the supported
pixel formats.
@item -sws_flags @var{flags}
Set SwScaler flags.
Set SwScaler flags (only available when compiled with swscale support).
@item -g @var{gop_size}
Set the group of pictures size.
@item -intra
@@ -634,6 +660,9 @@ Calculate PSNR of compressed frames.
Dump video coding statistics to @file{vstats_HHMMSS.log}.
@item -vstats_file @var{file}
Dump video coding statistics to @var{file}.
@item -vhook @var{module}
Insert video processing @var{module}. @var{module} contains the module
name and its parameters separated by spaces.
@item -top @var{n}
top=1/bottom=0/auto=-1 field first
@item -dc @var{precision}
@@ -658,8 +687,6 @@ Set the number of audio frames to record.
Set the audio sampling frequency (default = 44100 Hz).
@item -ab @var{bitrate}
Set the audio bitrate in bit/s (default = 64k).
@item -aq @var{q}
Set the audio quality (codec-specific, VBR).
@item -ac @var{channels}
Set the number of audio channels (default = 1).
@item -an
@@ -701,8 +728,6 @@ Force subtitle codec ('copy' to copy stream).
Add a new subtitle stream to the current output stream.
@item -slang @var{code}
Set the ISO 639 language code (3 letters) of the current subtitle stream.
@item -sn
Disable subtitle recording.
@item -sbsf @var{bitstream_filter}
Bitstream filters available are "mov2textsub", "text2movsub".
@example
@@ -734,10 +759,7 @@ Set meta data information of @var{outfile} from @var{infile}.
@item -debug
Print specific debug info.
@item -benchmark
Show benchmarking information at the end of an encode.
Shows CPU time used and maximum memory consumption.
Maximum memory consumption is not supported on all systems,
it will usually display as 0 if not supported.
Add timings for benchmarking.
@item -dump
Dump each input packet.
@item -hex
@@ -745,7 +767,7 @@ When dumping packets, also dump the payload.
@item -bitexact
Only use bit exact algorithms (for codec testing).
@item -ps @var{size}
Set RTP payload size in bytes.
Set packet size in bits.
@item -re
Read input at native frame rate. Mainly used to simulate a grab device.
@item -loop_input
@@ -757,15 +779,8 @@ Repeatedly loop output for formats that support looping such as animated GIF
@item -threads @var{count}
Thread count.
@item -vsync @var{parameter}
Video sync method.
0 Each frame is passed with its timestamp from the demuxer to the muxer
1 Frames will be duplicated and dropped to achieve exactly the requested
constant framerate.
2 Frames are passed through with their timestamp or dropped so as to prevent
2 frames from having the same timestamp
-1 Chooses between 1 and 2 depending on muxer capabilities. This is the default method.
With -map you can select from
Video sync method. Video will be stretched/squeezed to match the timestamps,
it is done by duplicating and dropping frames. With -map you can select from
which stream the timestamps should be taken. You can leave either video or
audio unchanged and sync the remaining stream(s) to the unchanged one.
@item -async @var{samples_per_second}
@@ -793,22 +808,18 @@ awkward to specify on the command line. Lines starting with the hash
('#') character are ignored and are used to provide comments. Check
the @file{ffpresets} directory in the FFmpeg source tree for examples.
Preset files are specified with the @code{vpre}, @code{apre},
@code{spre}, and @code{fpre} options. The @code{fpre} option takes the
filename of the preset instead of a preset name as input and can be
used for any kind of codec. For the @code{vpre}, @code{apre}, and
@code{spre} options, the options specified in a preset file are
Preset files are specified with the @code{vpre}, @code{apre} and
@code{spre} options. The options specified in a preset file are
applied to the currently selected codec of the same type as the preset
option.
The argument passed to the @code{vpre}, @code{apre}, and @code{spre}
preset options identifies the preset file to use according to the
following rules:
The argument passed to the preset options identifies the preset file
to use according to the following rules.
First ffmpeg searches for a file named @var{arg}.ffpreset in the
directories @file{$FFMPEG_DATADIR} (if set), and @file{$HOME/.ffmpeg}, and in
the datadir defined at configuration time (usually @file{PREFIX/share/ffmpeg})
in that order. For example, if the argument is @code{libx264-max}, it will
directories @file{$HOME/.ffmpeg}, and in the datadir defined at
configuration time (usually @file{PREFIX/share/ffmpeg}) in that
order. For example, if the argument is @code{libx264-max}, it will
search for the file @file{libx264-max.ffpreset}.
If no such file is found, then ffmpeg will search for a file named
@@ -818,7 +829,12 @@ the preset file options will be applied. For example, if you select
the video codec with @code{-vcodec libx264} and use @code{-vpre max},
then it will search for the file @file{libx264-max.ffpreset}.
@anchor{FFmpeg formula evaluator}
Finally, if the above rules failed and the argument specifies an
absolute pathname, ffmpeg will search for that filename. This way you
can specify the absolute and complete filename of the preset file, for
example @file{./ffpresets/libx264-max.ffpreset}.
@node FFmpeg formula evaluator
@section FFmpeg formula evaluator
When evaluating a rate control string, FFmpeg uses an internal formula
@@ -909,7 +925,7 @@ to standard output.
FFmpeg also handles many protocols specified with an URL syntax.
Use 'ffmpeg -protocols' to see a list of the supported protocols.
Use 'ffmpeg -formats' to see a list of the supported protocols.
The protocol @code{http:} is currently used only to communicate with
FFserver (see the FFserver documentation). When FFmpeg will be a

View File

@@ -26,12 +26,17 @@ ffplay [options] @file{input_file}
@end example
@c man begin OPTIONS
@include fftools-common-opts.texi
@section Main options
@table @option
@item -h
Show help.
@item -version
Show version.
@item -L
Show license.
@item -formats
Show available formats, codecs, protocols, ...
@item -x @var{width}
Force displayed width.
@item -y @var{height}
@@ -45,18 +50,12 @@ Disable audio.
Disable video.
@item -ss @var{pos}
Seek to a given position in seconds.
@item -t @var{duration}
play <duration> seconds of audio/video
@item -bytes
Seek by bytes.
@item -nodisp
Disable graphical display.
@item -f @var{fmt}
Force format.
@item -window_title @var{title}
Set window title (default is the input filename).
@item -loop @var{number}
Loops movie playback <number> times. 0 means forever.
@end table
@section Advanced options

View File

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

View File

@@ -236,12 +236,15 @@ ffserver [options]
@section Options
@c man begin OPTIONS
@include fftools-common-opts.texi
@section Main options
@table @option
@item -version
Show version.
@item -L
Show license.
@item -formats
Show available formats, codecs, protocols, ...
@item -h
Show help.
@item -f @var{configfile}
Use @file{configfile} instead of @file{/etc/ffserver.conf}.
@item -n

View File

@@ -1,72 +0,0 @@
@section Generic options
These options are shared amongst the ff* tools.
@table @option
@item -L
Show license.
@item -h, -?, -help, --help
Show help.
@item -version
Show version.
@item -formats
Show available formats.
The fields preceding the format names have the following meanings:
@table @samp
@item D
Decoding available
@item E
Encoding available
@end table
@item -codecs
Show available codecs.
The fields preceding the codec names have the following meanings:
@table @samp
@item D
Decoding available
@item E
Encoding available
@item V/A/S
Video/audio/subtitle codec
@item S
Codec supports slices
@item D
Codec supports direct rendering
@item T
Codec can handle input truncated at random locations instead of only at frame boundaries
@end table
@item -bsfs
Show available bitstream filters.
@item -protocols
Show available protocols.
@item -filters
Show available libavfilter filters.
@item -pix_fmts
Show available pixel formats.
@item -loglevel @var{loglevel}
Set the logging level used by the library.
@var{loglevel} is a number or a string containing one of the following values:
@table @samp
@item quiet
@item panic
@item fatal
@item error
@item warning
@item info
@item verbose
@item debug
@end table
@end table

View File

@@ -14,25 +14,25 @@ FFmpeg can be hooked up with a number of external libraries to add support
for more formats. None of them are used by default, their use has to be
explicitly requested by passing the appropriate flags to @file{./configure}.
@section OpenCORE AMR
@section AMR
FFmpeg can make use of the OpenCORE libraries for AMR-NB
decoding/encoding and AMR-WB decoding.
AMR comes in two different flavors, wideband and narrowband. FFmpeg can make
use of the AMR wideband (floating-point mode) and the AMR narrowband
(floating-point mode) reference decoders and encoders.
Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the instructions for
installing the libraries. Then pass @code{--enable-libopencore-amrnb} and/or
@code{--enable-libopencore-amrwb} to configure to enable the libraries.
Go to @url{http://www.penguin.cz/~utx/amr} and follow the instructions for
installing the libraries. Then pass @code{--enable-libamr-nb} and/or
@code{--enable-libamr-wb} to configure to enable the libraries.
Note that OpenCORE is under the Apache License 2.0 (see
@url{http://www.apache.org/licenses/LICENSE-2.0} for details), which is
incompatible with the LGPL version 2.1 and GPL version 2. You have to
upgrade FFmpeg's license to LGPL version 3 (or if you have enabled
GPL components, GPL version 3) to use it.
Note that libamr is copyrighted without any sort of license grant. This means
that you can use it if you legally obtained it but you are not allowed to
redistribute it in any way. @strong{Any FFmpeg binaries with libamr support
you create are non-free and unredistributable!}
@chapter Supported File Formats and Codecs
You can use the @code{-formats} and @code{-codecs} options to have an exhaustive list.
You can use the @code{-formats} option to have an exhaustive list.
@section File Formats
@@ -43,8 +43,6 @@ library:
@item Name @tab Encoding @tab Decoding @tab Comments
@item 4xm @tab @tab X
@tab 4X Technologies format, used in some games.
@item 8088flex TMV @tab @tab X
@item Adobe Filmstrip @tab X @tab X
@item Audio IFF (AIFF) @tab X @tab X
@item American Laser Games MM @tab @tab X
@tab Multimedia format used in games like Mad Dog McCree.
@@ -58,25 +56,18 @@ library:
@tab Audio and video format used in some games by Beam Software.
@item Bethesda Softworks VID @tab @tab X
@tab Used in some games from Bethesda Softworks.
@item Bink @tab @tab X
@tab Multimedia format used by many games.
@item Brute Force & Ignorance @tab @tab X
@tab Used in the game Flash Traffic: City of Angels.
@item Interplay C93 @tab @tab X
@tab Used in the game Cyberia from Interplay.
@item Delphine Software International CIN @tab @tab X
@tab Multimedia format used by Delphine Software games.
@item CD+G @tab @tab X
@tab Video format used by CD+G karaoke disks
@item Core Audio Format @tab @tab X
@tab Apple Core Audio Format
@item CRC testing format @tab X @tab
@item Creative Voice @tab X @tab X
@tab Created for the Sound Blaster Pro.
@item CRYO APC @tab @tab X
@tab Audio format used in some games by CRYO Interactive Entertainment.
@item D-Cinema audio @tab X @tab X
@item Deluxe Paint Animation @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
@@ -102,20 +93,16 @@ library:
@item id Quake II CIN video @tab @tab X
@item id RoQ @tab X @tab X
@tab Used in Quake III, Jedi Knight 2, other computer games.
@item IEC61937 encapsulation @tab X @tab
@item IFF @tab @tab X
@tab Interchange File Format
@item Interplay MVE @tab @tab X
@tab Format used in various Interplay computer games.
@item IV8 @tab @tab X
@tab A format generated by IndigoVision 8000 video server.
@item LMLM4 @tab @tab X
@tab Used by Linux Media Labs MPEG-4 PCI boards
@item Matroska @tab X @tab X
@item Matroska audio @tab X @tab
@item MAXIS XA @tab @tab X
@tab Used in Sim City 3000; file extension .xa.
@item MD Studio @tab @tab X
@item Monkey's Audio @tab @tab X
@item Motion Pixels MVI @tab @tab X
@item MOV/QuickTime/MP4 @tab X @tab X
@@ -142,8 +129,6 @@ library:
@tab SMPTE 386M, D-10/IMX Mapping.
@item NC camera feed @tab @tab X
@tab NC (AVIP NC4600) camera streams
@item NTT TwinVQ (VQF) @tab @tab X
@tab Nippon Telegraph and Telephone Corporation TwinVQ.
@item Nullsoft Streaming Video @tab @tab X
@item NuppelVideo @tab @tab X
@item NUT @tab X @tab X
@@ -151,7 +136,6 @@ library:
@item Ogg @tab X @tab X
@item TechnoTrend PVA @tab @tab X
@tab Used by TechnoTrend DVB PCI boards.
@item QCP @tab @tab X
@item raw ADTS (AAC) @tab X @tab X
@item raw AC-3 @tab X @tab X
@item raw Chinese AVS video @tab @tab X
@@ -176,7 +160,6 @@ library:
@item raw video @tab X @tab X
@item raw id RoQ @tab X @tab
@item raw Shorten @tab @tab X
@item raw TrueHD @tab X @tab X
@item raw VC-1 @tab @tab X
@item raw PCM A-law @tab X @tab X
@item raw PCM mu-law @tab X @tab X
@@ -207,10 +190,8 @@ library:
@item RL2 @tab @tab X
@tab Audio and video format used in some games by Entertainment Software Partners.
@item RPL/ARMovie @tab @tab X
@item RTMP @tab X @tab X
@tab Output is performed by publishing stream to RTMP server
@item RTP @tab @tab X
@item RTSP @tab X @tab X
@item RTSP @tab @tab X
@item SDP @tab @tab X
@item Sega FILM/CPK @tab @tab X
@tab Used in many Sega Saturn console games.
@@ -223,8 +204,6 @@ library:
@item Sony OpenMG (OMA) @tab @tab X
@tab Audio format used in Sony Sonic Stage and Sony Vegas.
@item Sony PlayStation STR @tab @tab X
@item Sony Wave64 (W64) @tab @tab X
@item SoX native format @tab X @tab X
@item SUN AU format @tab X @tab X
@item THP @tab @tab X
@tab Used on the Nintendo GameCube.
@@ -234,7 +213,6 @@ library:
@item VC-1 test bitstream @tab X @tab X
@item WAV @tab X @tab X
@item WavPack @tab @tab X
@item WebM @tab X @tab X
@item Wing Commander III movie @tab @tab X
@tab Multimedia format used in Origin's Wing Commander III computer game.
@item Westwood Studios audio @tab @tab X
@@ -242,7 +220,6 @@ library:
@item Westwood Studios VQA @tab @tab X
@tab Multimedia format used in Westwood Studios games.
@item YUV4MPEG pipe @tab X @tab X
@item Psygnosis YOP @tab @tab X
@end multitable
@code{X} means that encoding (resp. decoding) is supported.
@@ -260,8 +237,6 @@ following image formats are supported:
@tab Only uncompressed GIFs are generated.
@item BMP @tab X @tab X
@tab Microsoft BMP image
@item DPX @tab @tab X
@tab Digital Picture Exchange
@item JPEG @tab X @tab X
@tab Progressive JPEG is not supported.
@item JPEG 2000 @tab @tab E
@@ -273,7 +248,7 @@ following image formats are supported:
@tab PAM is a PNM extension with alpha support.
@item PBM @tab X @tab X
@tab Portable BitMap image
@item PCX @tab X @tab X
@item PCX @tab @tab X
@tab PC Paintbrush
@item PGM @tab X @tab X
@tab Portable GrayMap image
@@ -305,7 +280,6 @@ following image formats are supported:
@item Name @tab Encoding @tab Decoding @tab Comments
@item 4X Movie @tab @tab X
@tab Used in certain computer games.
@item 8088flex TMV @tab @tab X
@item 8SVX exponential @tab @tab X
@item 8SVX fibonacci @tab @tab X
@item American Laser Games MM @tab @tab X
@@ -323,8 +297,6 @@ following image formats are supported:
@tab fourcc: VCR1
@item ATI VCR2 @tab @tab X
@tab fourcc: VCR2
@item Auravision Aura @tab @tab X
@item Auravision Aura 2 @tab @tab X
@item Autodesk Animator Flic video @tab @tab X
@item Autodesk RLE @tab @tab X
@tab fourcc: AASC
@@ -333,16 +305,12 @@ following image formats are supported:
@item Beam Software VB @tab @tab X
@item Bethesda VID video @tab @tab X
@tab Used in some games from Bethesda Softworks.
@item Bink Video @tab @tab X
@tab Support for version 'b' is missing.
@item Brute Force & Ignorance @tab @tab X
@tab Used in the game Flash Traffic: City of Angels.
@item C93 video @tab @tab X
@tab Codec used in Cyberia game.
@item CamStudio @tab @tab X
@tab fourcc: CSCD
@item CD+G @tab @tab X
@tab Video codec for CD+G karaoke disks
@item Chinese AVS video @tab @tab X
@tab AVS1-P2, JiZhun profile
@item Delphine Software International CIN video @tab @tab X
@@ -353,7 +321,6 @@ following image formats are supported:
@item Creative YUV (CYUV) @tab @tab X
@item Dirac @tab E @tab E
@tab supported through external libdirac/libschroedinger libraries
@item Deluxe Paint Animation @tab @tab X
@item DNxHD @tab X @tab X
@tab aka SMPTE VC3
@item Duck TrueMotion 1.0 @tab @tab X
@@ -365,12 +332,11 @@ following image formats are supported:
@tab Codec originally used in Feeble Files game.
@item Electronic Arts CMV video @tab @tab X
@tab Used in NHL 95 game.
@item Electronic Arts Madcow video @tab @tab X
@item Electronic Arts TGV video @tab @tab X
@item Electronic Arts TGQ video @tab @tab X
@item Electronic Arts TQI video @tab @tab X
@item Escape 124 @tab @tab X
@item FFmpeg video codec #1 @tab X @tab X
@item FFmpeg codec #1 @tab X @tab X
@tab experimental lossless codec (fourcc: FFV1)
@item Flash Screen Video v1 @tab X @tab X
@tab fourcc: FSV1
@@ -391,22 +357,15 @@ following image formats are supported:
@tab Used in Quake II.
@item id RoQ video @tab X @tab X
@tab Used in Quake III, Jedi Knight 2, other computer games.
@item IFF ILBM @tab @tab X
@tab IFF interlaved bitmap
@item IFF ByteRun1 @tab @tab X
@tab IFF run length encoded bitmap
@item Intel H.263 @tab @tab X
@item Intel Indeo 2 @tab @tab X
@item Intel Indeo 3 @tab @tab X
@item Intel Indeo 5 @tab @tab X
@item Interplay C93 @tab @tab X
@tab Used in the game Cyberia from Interplay.
@item Interplay MVE video @tab @tab X
@tab Used in Interplay .MVE files.
@item Karl Morton's video codec @tab @tab X
@tab Codec used in Worms games.
@item Kega Game Video (KGV1) @tab @tab X
@tab Kega emulator screen capture codec.
@item LCL (LossLess Codec Library) MSZH @tab @tab X
@item LCL (LossLess Codec Library) ZLIB @tab E @tab E
@item LOCO @tab @tab X
@@ -437,8 +396,6 @@ following image formats are supported:
@tab fourcc: VP50
@item On2 VP6 @tab @tab X
@tab fourcc: VP60,VP61,VP62
@item VP8 @tab X @tab X
@tab fourcc: VP80, de/encoding supported through external library libvpx
@item planar RGB @tab @tab X
@tab fourcc: 8BPS
@item Q-team QPEG @tab @tab X
@@ -450,7 +407,6 @@ following image formats are supported:
@tab fourcc: 'smc '
@item QuickTime video (RPZA) @tab @tab X
@tab fourcc: rpza
@item R210 Quicktime Uncompressed RGB 10-bit @tab @tab X
@item Raw Video @tab X @tab X
@item RealVideo 1.0 @tab X @tab X
@item RealVideo 2.0 @tab X @tab X
@@ -481,7 +437,6 @@ following image formats are supported:
@tab encoding supported through external library libtheora
@item Tiertex Limited SEQ video @tab @tab X
@tab Codec used in DOS CD-ROM FlashBack game.
@item V210 Quicktime Uncompressed 4:2:2 10-bit @tab X @tab X
@item VMware Screen Codec / VMware Video @tab @tab X
@tab Codec used in videos captured by VMware.
@item Westwood Studios VQA (Vector Quantized Animation) video @tab @tab X
@@ -494,7 +449,6 @@ following image formats are supported:
@item Winnov WNV1 @tab @tab X
@item WMV7 @tab X @tab X
@item YAMAHA SMAF @tab X @tab X
@item Psygnosis YOP Video @tab @tab X
@item ZLIB @tab X @tab X
@tab part of LCL, encoder experimental
@item Zip Motion Blocks Video @tab X @tab X
@@ -511,7 +465,7 @@ following image formats are supported:
@item Name @tab Encoding @tab Decoding @tab Comments
@item 8SVX audio @tab @tab X
@item AAC @tab E @tab X
@tab encoding supported through external library libfaac and libvo-aacenc
@tab encoding supported through external library libfaac
@item AC-3 @tab IX @tab X
@item ADPCM 4X Movie @tab @tab X
@item ADPCM CDROM XA @tab @tab X
@@ -556,16 +510,13 @@ following image formats are supported:
@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
@item AMR-NB @tab E @tab X
@tab encoding supported through external library libopencore-amrnb
@item AMR-WB @tab @tab E
@tab decoding supported through external library libopencore-amrwb
@item AMR-NB @tab E @tab E
@tab supported through external library libamrnb
@item AMR-WB @tab E @tab E
@tab supported through external library libamrwb
@item Apple lossless audio @tab X @tab X
@tab QuickTime fourcc 'alac'
@item Atrac 1 @tab @tab X
@item Atrac 3 @tab @tab X
@item Bink Audio @tab @tab X
@tab Used in Bink and Smacker files in many games.
@item Delphine Software International CIN audio @tab @tab X
@tab Codec used in Delphine Software International games.
@item COOK @tab @tab X
@@ -579,7 +530,6 @@ following image formats are supported:
@tab Used in Sierra Online game audio files.
@item DPCM Sol @tab @tab X
@item DPCM Xan @tab @tab X
@tab Used in Origin's Wing Commander IV AVI files.
@item DSP Group TrueSpeech @tab @tab X
@item DV audio @tab @tab X
@item Enhanced AC-3 @tab @tab X
@@ -592,15 +542,14 @@ following image formats are supported:
@item IMC (Intel Music Coder) @tab @tab X
@item MACE (Macintosh Audio Compression/Expansion) 3:1 @tab @tab X
@item MACE (Macintosh Audio Compression/Expansion) 6:1 @tab @tab X
@item MLP (Meridian Lossless Packing) @tab @tab X
@tab Used in DVD-Audio discs.
@item MLP(Meridian Lossless Packing)/TrueHD @tab @tab X
@tab Used in DVD-Audio and Blu-Ray discs.
@item Monkey's Audio @tab @tab X
@tab Only versions 3.97-3.99 are supported.
@item MP1 (MPEG audio layer 1) @tab @tab IX
@item MP2 (MPEG audio layer 2) @tab IX @tab IX
@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
@@ -619,7 +568,6 @@ following image formats are supported:
@item PCM signed 24-bit little-endian @tab X @tab X
@item PCM signed 32-bit big-endian @tab X @tab X
@item PCM signed 32-bit little-endian @tab X @tab X
@item PCM signed 16/20/24-bit big-endian in MPEG-TS @tab @tab X
@item PCM unsigned 8-bit @tab X @tab X
@item PCM unsigned 16-bit big-endian @tab X @tab X
@item PCM unsigned 16-bit little-endian @tab X @tab X
@@ -637,7 +585,6 @@ following image formats are supported:
@tab Real 28800 bit/s codec
@item RealAudio 3.0 (dnet) @tab IX @tab X
@tab Real low bitrate AC-3 codec
@item RealAudio SIPR / ACELP.NET @tab @tab X
@item Shorten @tab @tab X
@item Sierra VMD audio @tab @tab X
@tab Used in Sierra VMD files.
@@ -649,17 +596,13 @@ following image formats are supported:
@item Speex @tab @tab E
@tab supported through external library libspeex
@item True Audio (TTA) @tab @tab X
@item TrueHD @tab @tab X
@tab Used in HD-DVD and Blu-Ray discs.
@item TwinVQ (VQF flavor) @tab @tab X
@item Vorbis @tab E @tab X
@tab A native but very primitive encoder exists.
@ A native but very primitive encoder exists.
@item WavPack @tab @tab X
@item Westwood Audio (SND1) @tab @tab X
@item Windows Media Audio 1 @tab X @tab X
@item Windows Media Audio 2 @tab X @tab X
@item Windows Media Audio Pro @tab @tab X
@item Windows Media Audio Voice @tab @tab X
@tab Used in Origin's Wing Commander IV AVI files.
@end multitable
@code{X} means that encoding (resp. decoding) is supported.
@@ -671,20 +614,19 @@ performance on systems without hardware floating point support).
@section Subtitle Formats
@multitable @columnfractions .4 .1 .1 .1 .1
@multitable @columnfractions .4 .1 .1 .1 .1 .1
@item Name @tab Muxing @tab Demuxing @tab Encoding @tab Decoding
@item SSA/ASS @tab X @tab X
@item DVB @tab X @tab X @tab X @tab X
@item DVD @tab X @tab X @tab X @tab X
@item PGS @tab @tab @tab @tab X
@item XSUB @tab @tab @tab X @tab X
@item XSUB @tab @tab @tab @tab X
@end multitable
@code{X} means that the feature is supported.
@section Network Protocols
@multitable @columnfractions .4 .1
@multitable @columnfractions .4 .1 .1 .1 .1 .1
@item Name @tab Support
@item file @tab X
@item Gopher @tab X
@@ -698,90 +640,13 @@ performance on systems without hardware floating point support).
@code{X} means that the protocol is supported.
@section Input/Output Devices
@multitable @columnfractions .4 .1 .1
@item Name @tab Input @tab Output
@item ALSA @tab X @tab X
@item BEOS audio @tab X @tab X
@item BKTR @tab X @tab
@item DV1394 @tab X @tab
@item JACK @tab X @tab
@item LIBDC1394 @tab X @tab
@item OSS @tab X @tab X
@item Video4Linux @tab X @tab
@item Video4Linux2 @tab X @tab
@item VfW capture @tab X @tab
@item X11 grabbing @tab X @tab
@end multitable
@code{X} means that input/output is supported.
@chapter Platform Specific information
@section BeOS
BeOS support is broken in mysterious ways.
@section DOS
Using a cross-compiler is preferred for various reasons.
@subsection DJGPP
FFmpeg cannot be compiled because of broken system headers, add
@code{--extra-cflags=-U__STRICT_ANSI__} to the configure options as a
workaround.
@section OS/2
For information about compiling FFmpeg on OS/2 see
@url{http://www.edm2.com/index.php/FFmpeg}.
@section Unix-like
Some parts of FFmpeg cannot be built with version 2.15 of the GNU
assembler which is still provided by a few AMD64 distributions. To
make sure your compiler really uses the required version of gas
after a binutils upgrade, run:
@example
$(gcc -print-prog-name=as) --version
@end example
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.
@subsection BSD
@section BSD
BSD make will not build FFmpeg, you need to install and use GNU Make
(@file{gmake}).
@subsubsection FreeBSD
FreeBSD will not compile out-of-the-box due to broken system headers.
Passing @code{--extra-cflags=-D__BSD_VISIBLE} to configure will work
around the problem. This may have unexpected sideeffects, so use it at
your own risk. If you care about FreeBSD, please make an attempt at
getting the system headers fixed.
@subsection (Open)Solaris
GNU Make is required to build FFmpeg, so you have to invoke (@file{gmake}),
standard Solaris Make will not work. When building with a non-c99 front-end
(gcc, generic suncc) add either @code{--extra-libs=/usr/lib/values-xpg6.o}
or @code{--extra-libs=/usr/lib/64/values-xpg6.o} to the configure options
since the libc is not c99-compliant by default. The probes performed by
configure may raise an exception leading to the death of configure itself
due to a bug in the system shell. Simply invoke a different shell such as
bash directly to work around this:
@example
bash ./configure
@end example
@section Windows
To get help and instructions for building FFmpeg under Windows, check out
@@ -807,13 +672,13 @@ are listed below:
@item mingw-runtime 3.15
@end itemize
FFmpeg automatically passes @code{-fno-common} to the compiler to work around
You will also need to pass @code{-fno-common} to the compiler to work around
a GCC bug (see @url{http://gcc.gnu.org/bugzilla/show_bug.cgi?id=37216}).
Within the MSYS shell, configure and make with:
@example
./configure --enable-memalign-hack
./configure --enable-memalign-hack --extra-cflags=-fno-common
make
make install
@end example
@@ -826,11 +691,9 @@ Notes:
@itemize
@item Building natively using MSYS can be sped up by disabling implicit rules
in the Makefile by calling @code{make -r} instead of plain @code{make}. This
speed up is close to non-existent for normal one-off builds and is only
noticeable when running make for a second time (for example in
@code{make install}).
@item In order to compile vhooks, you must have a POSIX-compliant libdl in
your MinGW system. Get dlfcn-win32 from
@url{http://code.google.com/p/dlfcn-win32}.
@item In order to compile FFplay, you must have the MinGW development library
of SDL. Get it from @url{http://www.libsdl.org}.
@@ -870,7 +733,7 @@ Application Wizard, uncheck the "Precompiled headers" option.
@item Write the source code for your application, or, for testing, just
copy the code from an existing sample application into the source file
that MSVC++ has already created for you. For example, you can copy
@file{libavformat/output-example.c} from the FFmpeg distribution.
@file{output_example.c} from the FFmpeg distribution.
@item Open the "Project / Properties" dialog box. In the "Configuration"
combo box, select "All Configurations" so that the changes you make will
@@ -986,35 +849,43 @@ Then you can easily test FFmpeg with Wine
@subsection Compilation under Cygwin
Please use Cygwin 1.7.x as the obsolete 1.5.x Cygwin versions lack
llrint() in its C library.
The main issue with the 1.5.x Cygwin versions is that newlib, its C library,
does not contain llrint(). You need to upgrade to the unstable 1.7.x versions,
or leverage the implementation in MinGW (as explained below).
Install your Cygwin with all the "Base" packages, plus the
Just install your Cygwin with all the "Base" packages, plus the
following "Devel" ones:
@example
binutils, gcc4-core, make, subversion, mingw-runtime, texi2html
binutils, gcc-core, make, subversion, mingw-runtime, diffutils
@end example
And the following "Utils" one:
The experimental gcc4 package is still buggy, hence please
use the official gcc 3.4.4 or a 4.2.x compiled from source by yourself.
Install the current binutils-20080624-2 as they work fine (the old
binutils-20060709-1 proved buggy on shared builds).
Then create a small library that just contains llrint():
@example
diffutils
ar x /usr/lib/mingw/libmingwex.a llrint.o
ar cq /usr/local/lib/libllrint.a llrint.o
@end example
Then run
@example
./configure --enable-static --disable-shared
./configure --enable-static --disable-shared --extra-ldflags='-L /usr/local/lib' --extra-libs='-l llrint'
@end example
to make a static build.
The current @code{gcc4-core} package is buggy and needs this flag to build
shared libraries:
to make a static build or
@example
./configure --enable-shared --disable-static --extra-cflags=-fno-reorder-functions
./configure --enable-shared --disable-static --extra-ldflags='-L /usr/local/lib' --extra-libs='-l llrint'
@end example
to build shared libraries.
If you want to build FFmpeg with additional libraries, download Cygwin
"Devel" packages for Ogg and Vorbis from any Cygwin packages repository:
@example
@@ -1058,4 +929,420 @@ and for a build with shared libraries
./configure --target-os=mingw32 --enable-memalign-hack --enable-shared --disable-static --extra-cflags=-mno-cygwin --extra-libs=-mno-cygwin
@end example
@section BeOS
BeOS support is broken in mysterious ways.
@section OS/2
For information about compiling FFmpeg on OS/2 see
@url{http://www.edm2.com/index.php/FFmpeg}.
@chapter Developers Guide
@section API
@itemize @bullet
@item libavcodec is the library containing the codecs (both encoding and
decoding). Look at @file{libavcodec/apiexample.c} to see how to use it.
@item libavformat is the library containing the file format handling (mux and
demux code for several formats). Look at @file{ffplay.c} to use it in a
player. See @file{output_example.c} to use it to generate audio or video
streams.
@end itemize
@section Integrating libavcodec or libavformat in your program
You can integrate all the source code of the libraries to link them
statically to avoid any version problem. All you need is to provide a
'config.mak' and a 'config.h' in the parent directory. See the defines
generated by ./configure to understand what is needed.
You can use libavcodec or libavformat in your commercial program, but
@emph{any patch you make must be published}. The best way to proceed is
to send your patches to the FFmpeg mailing list.
@node Coding Rules
@section Coding Rules
FFmpeg is programmed in the ISO C90 language with a few additional
features from ISO C99, namely:
@itemize @bullet
@item
the @samp{inline} keyword;
@item
@samp{//} comments;
@item
designated struct initializers (@samp{struct s x = @{ .i = 17 @};})
@item
compound literals (@samp{x = (struct s) @{ 17, 23 @};})
@end itemize
These features are supported by all compilers we care about, so we will not
accept patches to remove their use unless they absolutely do not impair
clarity and performance.
All code must compile with GCC 2.95 and GCC 3.3. Currently, FFmpeg also
compiles with several other compilers, such as the Compaq ccc compiler
or Sun Studio 9, and we would like to keep it that way unless it would
be exceedingly involved. To ensure compatibility, please do not use any
additional C99 features or GCC extensions. Especially watch out for:
@itemize @bullet
@item
mixing statements and declarations;
@item
@samp{long long} (use @samp{int64_t} instead);
@item
@samp{__attribute__} not protected by @samp{#ifdef __GNUC__} or similar;
@item
GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}).
@end itemize
Indent size is 4.
The presentation is the one specified by 'indent -i4 -kr -nut'.
The TAB character is forbidden outside of Makefiles as is any
form of trailing whitespace. Commits containing either will be
rejected by the Subversion repository.
The main priority in FFmpeg is simplicity and small code size in order to
minimize the bug count.
Comments: Use the JavaDoc/Doxygen
format (see examples below) so that code documentation
can be generated automatically. All nontrivial functions should have a comment
above them explaining what the function does, even if it is just one sentence.
All structures and their member variables should be documented, too.
@example
/**
* @@file mpeg.c
* MPEG codec.
* @@author ...
*/
/**
* Summary sentence.
* more text ...
* ...
*/
typedef struct Foobar@{
int var1; /**< var1 description */
int var2; ///< var2 description
/** var3 description */
int var3;
@} Foobar;
/**
* Summary sentence.
* more text ...
* ...
* @@param my_parameter description of my_parameter
* @@return return value description
*/
int myfunc(int my_parameter)
...
@end example
fprintf and printf are forbidden in libavformat and libavcodec,
please use av_log() instead.
Casts should be used only when necessary. Unneeded parentheses
should also be avoided if they don't make the code easier to understand.
@section Development Policy
@enumerate
@item
Contributions should be licensed under the LGPL 2.1, including an
"or any later version" clause, or the MIT license. GPL 2 including
an "or any later version" clause is also acceptable, but LGPL is
preferred.
@item
You must not commit code which breaks FFmpeg! (Meaning unfinished but
enabled code which breaks compilation or compiles but does not work or
breaks the regression tests)
You can commit unfinished stuff (for testing etc), but it must be disabled
(#ifdef etc) by default so it does not interfere with other developers'
work.
@item
You do not have to over-test things. If it works for you, and you think it
should work for others, then commit. If your code has problems
(portability, triggers compiler bugs, unusual environment etc) they will be
reported and eventually fixed.
@item
Do not commit unrelated changes together, split them into self-contained
pieces. Also do not forget that if part B depends on part A, but A does not
depend on B, then A can and should be committed first and separate from B.
Keeping changes well split into self-contained parts makes reviewing and
understanding them on the commit log mailing list easier. This also helps
in case of debugging later on.
Also if you have doubts about splitting or not splitting, do not hesitate to
ask/discuss it on the developer mailing list.
@item
Do not change behavior of the program (renaming options etc) without
first discussing it on the ffmpeg-devel mailing list. Do not remove
functionality from the code. Just improve!
Note: Redundant code can be removed.
@item
Do not commit changes to the build system (Makefiles, configure script)
which change behavior, defaults etc, without asking first. The same
applies to compiler warning fixes, trivial looking fixes and to code
maintained by other developers. We usually have a reason for doing things
the way we do. Send your changes as patches to the ffmpeg-devel mailing
list, and if the code maintainers say OK, you may commit. This does not
apply to files you wrote and/or maintain.
@item
We refuse source indentation and other cosmetic changes if they are mixed
with functional changes, such commits will be rejected and removed. Every
developer has his own indentation style, you should not change it. Of course
if you (re)write something, you can use your own style, even though we would
prefer if the indentation throughout FFmpeg was consistent (Many projects
force a given indentation style - we do not.). If you really need to make
indentation changes (try to avoid this), separate them strictly from real
changes.
NOTE: If you had to put if()@{ .. @} over a large (> 5 lines) chunk of code,
then either do NOT change the indentation of the inner part within (do not
move it to the right)! or do so in a separate commit
@item
Always fill out the commit log message. Describe in a few lines what you
changed and why. You can refer to mailing list postings if you fix a
particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.
@item
If you apply a patch by someone else, include the name and email address in
the log message. Since the ffmpeg-cvslog mailing list is publicly
archived you should add some SPAM protection to the email address. Send an
answer to ffmpeg-devel (or wherever you got the patch from) saying that
you applied the patch.
@item
When applying patches that have been discussed (at length) on the mailing
list, reference the thread in the log message.
@item
Do NOT commit to code actively maintained by others without permission.
Send a patch to ffmpeg-devel instead. If no one answers within a reasonable
timeframe (12h for build failures and security fixes, 3 days small changes,
1 week for big patches) then commit your patch if you think it is OK.
Also note, the maintainer can simply ask for more time to review!
@item
Subscribe to the ffmpeg-cvslog mailing list. The diffs of all commits
are sent there and reviewed by all the other developers. Bugs and possible
improvements or general questions regarding commits are discussed there. We
expect you to react if problems with your code are uncovered.
@item
Update the documentation if you change behavior or add features. If you are
unsure how best to do this, send a patch to ffmpeg-devel, the documentation
maintainer(s) will review and commit your stuff.
@item
Try to keep important discussions and requests (also) on the public
developer mailing list, so that all developers can benefit from them.
@item
Never write to unallocated memory, never write over the end of arrays,
always check values read from some untrusted source before using them
as array index or other risky things.
@item
Remember to check if you need to bump versions for the specific libav
parts (libavutil, libavcodec, libavformat) you are changing. You need
to change the version integer.
Incrementing the first component means no backward compatibility to
previous versions (e.g. removal of a function from the public API).
Incrementing the second component means backward compatible change
(e.g. addition of a function to the public API or extension of an
existing data structure).
Incrementing the third component means a noteworthy binary compatible
change (e.g. encoder bug fix that matters for the decoder).
@item
Compiler warnings indicate potential bugs or code with bad style. If a type of
warning always points to correct and clean code, that warning should
be disabled, not the code changed.
Thus the remaining warnings can either be bugs or correct code.
If it is a bug, the bug has to be fixed. If it is not, the code should
be changed to not generate a warning unless that causes a slowdown
or obfuscates the code.
@item
If you add a new file, give it a proper license header. Do not copy and
paste it from a random place, use an existing file as template.
@end enumerate
We think our rules are not too hard. If you have comments, contact us.
Note, these rules are mostly borrowed from the MPlayer project.
@section Submitting patches
First, (@pxref{Coding Rules}) above if you did not yet.
When you submit your patch, try to send a unified diff (diff '-up'
option). We cannot read other diffs :-)
Also please do not submit a patch which contains several unrelated changes.
Split it into separate, self-contained pieces. This does not mean splitting
file by file. Instead, make the patch as small as possible while still
keeping it as a logical unit that contains an individual change, even
if it spans multiple files. This makes reviewing your patches much easier
for us and greatly increases your chances of getting your patch applied.
Run the regression tests before submitting a patch so that you can
verify that there are no big problems.
Patches should be posted as base64 encoded attachments (or any other
encoding which ensures that the patch will not be trashed during
transmission) to the ffmpeg-devel mailing list, see
@url{http://lists.mplayerhq.hu/mailman/listinfo/ffmpeg-devel}
It also helps quite a bit if you tell us what the patch does (for example
'replaces lrint by lrintf'), and why (for example '*BSD isn't C99 compliant
and has no lrint()')
Also please if you send several patches, send each patch as a separate mail,
do not attach several unrelated patches to the same mail.
@section New codecs or formats checklist
@enumerate
@item
Did you use av_cold for codec initialization and close functions?
@item
Did you add a long_name under NULL_IF_CONFIG_SMALL to the AVCodec or
AVInputFormat/AVOutputFormat struct?
@item
Did you bump the minor version number in @file{avcodec.h} or
@file{avformat.h}?
@item
Did you register it in @file{allcodecs.c} or @file{allformats.c}?
@item
Did you add the CodecID to @file{avcodec.h}?
@item
If it has a fourcc, did you add it to @file{libavformat/riff.c},
even if it is only a decoder?
@item
Did you add a rule to compile the appropriate files in the Makefile?
Remember to do this even if you're just adding a format to a file that is
already being compiled by some other rule, like a raw demuxer.
@item
Did you add an entry to the table of supported formats or codecs in the
documentation?
@item
Did you add an entry in the Changelog?
@item
If it depends on a parser or a library, did you add that dependency in
configure?
@item
Did you "svn add" the appropriate files before commiting?
@end enumerate
@section patch submission checklist
@enumerate
@item
Do the regression tests pass with the patch applied?
@item
Does @code{make checkheaders} pass with the patch applied?
@item
Is the patch a unified diff?
@item
Is the patch against latest FFmpeg SVN?
@item
Are you subscribed to ffmpeg-dev?
(the list is subscribers only due to spam)
@item
Have you checked that the changes are minimal, so that the same cannot be
achieved with a smaller patch and/or simpler final code?
@item
If the change is to speed critical code, did you benchmark it?
@item
If you did any benchmarks, did you provide them in the mail?
@item
Have you checked that the patch does not introduce buffer overflows or
other security issues?
@item
Did you test your decoder or demuxer against damaged data? If no, see
tools/trasher and the noise bitstream filter. Your decoder or demuxer
should not crash or end in a (near) infinite loop when fed damaged data.
@item
Is the patch created from the root of the source tree, so it can be
applied with @code{patch -p0}?
@item
Does the patch not mix functional and cosmetic changes?
@item
Did you add tabs or trailing whitespace to the code? Both are forbidden.
@item
Is the patch attached to the email you send?
@item
Is the mime type of the patch correct? It should be text/x-diff or
text/x-patch or at least text/plain and not application/octet-stream.
@item
If the patch fixes a bug, did you provide a verbose analysis of the bug?
@item
If the patch fixes a bug, did you provide enough information, including
a sample, so the bug can be reproduced and the fix can be verified?
Note please do not attach samples >100k to mails but rather provide a
URL, you can upload to ftp://upload.ffmpeg.org
@item
Did you provide a verbose summary about what the patch does change?
@item
Did you provide a verbose explanation why it changes things like it does?
@item
Did you provide a verbose summary of the user visible advantages and
disadvantages if the patch is applied?
@item
Did you provide an example so we can verify the new feature added by the
patch easily?
@item
If you added a new file, did you insert a license header? It should be
taken from FFmpeg, not randomly copied and pasted from somewhere else.
@item
You should maintain alphabetical order in alphabetically ordered lists as
long as doing so does not break API/ABI compatibility.
@item
Lines with similar content should be aligned vertically when doing so
improves readability.
@item
Did you provide a suggestion for a clear commit log message?
@end enumerate
@section Patch review process
All patches posted to ffmpeg-devel will be reviewed, unless they contain a
clear note that the patch is not for SVN.
Reviews and comments will be posted as replies to the patch on the
mailing list. The patch submitter then has to take care of every comment,
that can be by resubmitting a changed patch or by discussion. Resubmitted
patches will themselves be reviewed like any other patch. If at some point
a patch passes review with no comments then it is approved, that can for
simple and small patches happen immediately while large patches will generally
have to be changed and reviewed many times before they are approved.
After a patch is approved it will be committed to the repository.
We will review all submitted patches, but sometimes we are quite busy so
especially for large patches this can take several weeks.
When resubmitting patches, please do not make any significant changes
not related to the comments received during review. Such patches will
be rejected. Instead, submit significant changes or new features as
separate patches.
@section Regression tests
Before submitting a patch (or committing to the repository), you should at least
test that you did not break anything.
The regression tests build a synthetic video stream and a synthetic
audio stream. These are then encoded and decoded with all codecs or
formats. The CRC (or MD5) of each generated file is recorded in a
result file. A 'diff' is launched to compare the reference results and
the result file.
The regression tests then go on to test the FFserver code with a
limited set of streams. It is important that this step runs correctly
as well.
Run 'make test' to test all the codecs and formats.
Run 'make fulltest' to test all the codecs, formats and FFserver.
[Of course, some patches may change the results of the regression tests. In
this case, the reference results of the regression tests shall be modified
accordingly].
@bye

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\input texinfo @c -*- texinfo -*-
@settitle Video Hook Documentation
@titlepage
@sp 7
@center @titlefont{Video Hook Documentation}
@sp 3
@end titlepage
@chapter Introduction
@var{Please be aware that vhook is deprecated, and hence its development is
frozen (bug fixes are still accepted).
The substitute will be 'libavfilter', the result of our 'Video Filter API'
Google Summer of Code project. You may monitor its progress by subscribing to
the ffmpeg-soc mailing list at
@url{http://lists.mplayerhq.hu/mailman/listinfo/ffmpeg-soc}.}
The video hook functionality is designed (mostly) for live video. It allows
the video to be modified or examined between the decoder and the encoder.
Any number of hook modules can be placed inline, and they are run in the
order that they were specified on the ffmpeg command line.
The video hook modules are provided for use as a base for your own modules,
and are described below.
Modules are loaded using the -vhook option to ffmpeg. The value of this parameter
is a space separated list of arguments. The first is the module name, and the rest
are passed as arguments to the Configure function of the module.
The modules are dynamic libraries: They have different suffixes (.so, .dll, .dylib)
depending on your platform. And your platform dictates if they need to be
somewhere in your PATH, or in your LD_LIBRARY_PATH. Otherwise you will need to
specify the full path of the vhook file that you are using.
@section null.c
This does nothing. Actually it converts the input image to RGB24 and then converts
it back again. This is meant as a sample that you can use to test your setup.
@section fish.c
This implements a 'fish detector'. Essentially it converts the image into HSV
space and tests whether more than a certain percentage of the pixels fall into
a specific HSV cuboid. If so, then the image is saved into a file for processing
by other bits of code.
Why use HSV? It turns out that HSV cuboids represent a more compact range of
colors than would an RGB cuboid.
@section imlib2.c
This module implements a text overlay for a video image. Currently it
supports a fixed overlay or reading the text from a file. The string
is passed through strftime() so that it is easy to imprint the date and
time onto the image.
This module depends on the external library imlib2, available on
Sourceforge, among other places, if it is not already installed on
your system.
You may also overlay an image (even semi-transparent) like TV stations do.
You may move either the text or the image around your video to create
scrolling credits, for example.
The font file used is looked for in a FONTPATH environment variable, and
prepended to the point size as a command line option and can be specified
with the full path to the font file, as in:
@example
-F /usr/X11R6/lib/X11/fonts/TTF/VeraBd.ttf/20
@end example
where 20 is the point size.
You can specify the filename to read RGB color names from. If it is not
specified, these defaults are used: @file{/usr/share/X11/rgb.txt} and
@file{/usr/lib/X11/rgb.txt}
Options:
@multitable @columnfractions .2 .8
@item @option{-C <rgb.txt>} @tab The filename to read RGB color names from
@item @option{-c <color>} @tab The color of the text
@item @option{-F <fontname>} @tab The font face and size
@item @option{-t <text>} @tab The text
@item @option{-f <filename>} @tab The filename to read text from
@item @option{-x <expression>}@tab x coordinate of text or image
@item @option{-y <expression>}@tab y coordinate of text or image
@item @option{-i <filename>} @tab The filename to read a image from
@item @option{-R <expression>}@tab Value for R color
@item @option{-G <expression>}@tab Value for G color
@item @option{-B <expression>}@tab Value for B color
@item @option{-A <expression>}@tab Value for Alpha channel
@end multitable
Expressions are functions of these variables:
@multitable @columnfractions .2 .8
@item @var{N} @tab frame number (starting at zero)
@item @var{H} @tab frame height
@item @var{W} @tab frame width
@item @var{h} @tab image height
@item @var{w} @tab image width
@item @var{X} @tab previous x coordinate of text or image
@item @var{Y} @tab previous y coordinate of text or image
@end multitable
You may also use the constants @var{PI}, @var{E}, and the math functions available at the
FFmpeg formula evaluator at (@url{ffmpeg-doc.html#SEC13}), except @var{bits2qp(bits)}
and @var{qp2bits(qp)}.
Usage examples:
@example
# Remember to set the path to your fonts
FONTPATH="/cygdrive/c/WINDOWS/Fonts/"
FONTPATH="$FONTPATH:/usr/share/imlib2/data/fonts/"
FONTPATH="$FONTPATH:/usr/X11R6/lib/X11/fonts/TTF/"
export FONTPATH
# Bulb dancing in a Lissajous pattern
ffmpeg -i input.avi -vhook \
'vhook/imlib2.dll -x W*(0.5+0.25*sin(N/47*PI))-w/2 -y H*(0.5+0.50*cos(N/97*PI))-h/2 -i /usr/share/imlib2/data/images/bulb.png' \
-acodec copy -sameq output.avi
# Text scrolling
ffmpeg -i input.avi -vhook \
'vhook/imlib2.dll -c red -F Vera.ttf/20 -x 150+0.5*N -y 70+0.25*N -t Hello' \
-acodec copy -sameq output.avi
# Date and time stamp, security-camera style:
ffmpeg -r 29.97 -s 320x256 -f video4linux -i /dev/video0 \
-vhook 'vhook/imlib2.so -x 0 -y 0 -i black-260x20.png' \
-vhook 'vhook/imlib2.so -c white -F VeraBd.ttf/12 -x 0 -y 0 -t %A-%D-%T' \
output.avi
In this example the video is captured from the first video capture card as a
320x256 AVI, and a black 260 by 20 pixel PNG image is placed in the upper
left corner, with the day, date and time overlaid on it in Vera Bold 12
point font. A simple black PNG file 260 pixels wide and 20 pixels tall
was created in the GIMP for this purpose.
# Scrolling credits from a text file
ffmpeg -i input.avi -vhook \
'vhook/imlib2.so -c white -F VeraBd.ttf/16 -x 100 -y -1.0*N -f credits.txt' \
-sameq output.avi
In this example, the text is stored in a file, and is positioned 100
pixels from the left hand edge of the video. The text is scrolled from the
bottom up. Making the y factor positive will scroll from the top down.
Increasing the magnitude of the y factor makes the text scroll faster,
decreasing it makes it scroll slower. Hint: Blank lines containing only
a newline are treated as end-of-file. To create blank lines, use lines
that consist of space characters only.
# Scrolling credits with custom color from a text file
ffmpeg -i input.avi -vhook \
'vhook/imlib2.so -C rgb.txt -c CustomColor1 -F VeraBd.ttf/16 -x 100 -y -1.0*N -f credits.txt' \
-sameq output.avi
This example does the same as the one above, but specifies an rgb.txt file
to be used, which has a custom-made color in it.
# Variable colors
ffmpeg -i input.avi -vhook \
'vhook/imlib2.so -t Hello -R abs(255*sin(N/47*PI)) -G abs(255*sin(N/47*PI)) -B abs(255*sin(N/47*PI))' \
-sameq output.avi
In this example, the color for the text goes up and down from black to
white.
# Text fade-out
ffmpeg -i input.avi -vhook \
'vhook/imlib2.so -t Hello -A max(0,255-exp(N/47))' \
-sameq output.avi
In this example, the text fades out in about 10 seconds for a 25 fps input
video file.
# scrolling credits from a graphics file
ffmpeg -sameq -i input.avi \
-vhook 'vhook/imlib2.so -x 0 -y -1.0*N -i credits.png' output.avi
In this example, a transparent PNG file the same width as the video
(e.g. 320 pixels), but very long, (e.g. 3000 pixels), was created, and
text, graphics, brushstrokes, etc, were added to the image. The image
is then scrolled up, from the bottom of the frame.
@end example
@section ppm.c
It's basically a launch point for a PPM pipe, so you can use any
executable (or script) which consumes a PPM on stdin and produces a PPM
on stdout (and flushes each frame). The Netpbm utilities are a series of
such programs.
A list of them is here:
@url{http://netpbm.sourceforge.net/doc/directory.html}
Usage example:
@example
ffmpeg -i input -vhook "/path/to/ppm.so some-ppm-filter args" output
@end example
@section drawtext.c
This module implements a text overlay for a video image. Currently it
supports a fixed overlay or reading the text from a file. The string
is passed through strftime() so that it is easy to imprint the date and
time onto the image.
Features:
@itemize @minus
@item TrueType, Type1 and others via the FreeType2 library
@item Font kerning (better output)
@item Line Wrap (put the text that doesn't fit one line on the next line)
@item Background box (currently in development)
@item Outline
@end itemize
Options:
@multitable @columnfractions .2 .8
@item @option{-c <color>} @tab Foreground color of the text ('internet' way) <#RRGGBB> [default #FFFFFF]
@item @option{-C <color>} @tab Background color of the text ('internet' way) <#RRGGBB> [default #000000]
@item @option{-f <font-filename>} @tab font file to use
@item @option{-t <text>} @tab text to display
@item @option{-T <filename>} @tab file to read text from
@item @option{-x <pos>} @tab x coordinate of the start of the text
@item @option{-y <pos>} @tab y coordinate of the start of the text
@end multitable
Text fonts are being looked for in a FONTPATH environment variable.
If the FONTPATH environment variable is not available, or is not checked by
your target (i.e. Cygwin), then specify the full path to the font file as in:
@example
-f /usr/X11R6/lib/X11/fonts/TTF/VeraBd.ttf
@end example
Usage Example:
@example
# Remember to set the path to your fonts
FONTPATH="/cygdrive/c/WINDOWS/Fonts/"
FONTPATH="$FONTPATH:/usr/share/imlib2/data/fonts/"
FONTPATH="$FONTPATH:/usr/X11R6/lib/X11/fonts/TTF/"
export FONTPATH
# Time and date display
ffmpeg -f video4linux2 -i /dev/video0 \
-vhook 'vhook/drawtext.so -f VeraBd.ttf -t %A-%D-%T' movie.mpg
This example grabs video from the first capture card and outputs it to an
MPEG video, and places "Weekday-dd/mm/yy-hh:mm:ss" at the top left of the
frame, updated every second, using the Vera Bold TrueType Font, which
should exist in: /usr/X11R6/lib/X11/fonts/TTF/
@end example
Check the man page for strftime() for all the various ways you can format
the date and time.
@section watermark.c
Command Line options:
@multitable @columnfractions .2 .8
@item @option{-m [0|1]} @tab Mode (default: 0, see below)
@item @option{-t 000000 - FFFFFF} @tab Threshold, six digit hex number
@item @option{-f <filename>} @tab Watermark image filename, must be specified!
@end multitable
MODE 0:
The watermark picture works like this (assuming color intensities 0..0xFF):
Per color do this:
If mask color is 0x80, no change to the original frame.
If mask color is < 0x80 the absolute difference is subtracted from the
frame. If result < 0, result = 0.
If mask color is > 0x80 the absolute difference is added to the
frame. If result > 0xFF, result = 0xFF.
You can override the 0x80 level with the -t flag. E.g. if threshold is
000000 the color value of watermark is added to the destination.
This way a mask that is visible both in light and dark pictures can be made
(e.g. by using a picture generated by the Gimp and the bump map tool).
An example watermark file is at:
@url{http://engene.se/ffmpeg_watermark.gif}
MODE 1:
Per color do this:
If mask color > threshold color then the watermark pixel is used.
Example usage:
@example
ffmpeg -i infile -vhook '/path/watermark.so -f wm.gif' -an out.mov
ffmpeg -i infile -vhook '/path/watermark.so -f wm.gif -m 1 -t 222222' -an out.mov
@end example
@bye

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@@ -1,307 +0,0 @@
\input texinfo @c -*- texinfo -*-
@settitle Libavfilter Documentation
@titlepage
@sp 7
@center @titlefont{Libavfilter Documentation}
@sp 3
@end titlepage
@chapter Introduction
Libavfilter is the filtering API of FFmpeg. It is the substitute of the
now deprecated 'vhooks' and started as a Google Summer of Code project.
Integrating libavfilter into the main FFmpeg repository is a work in
progress. If you wish to try the unfinished development code of
libavfilter then check it out from the libavfilter repository into
some directory of your choice by:
@example
svn checkout svn://svn.ffmpeg.org/soc/libavfilter
@end example
And then read the README file in the top directory to learn how to
integrate it into ffmpeg and ffplay.
But note that there may still be serious bugs in the code and its API
and ABI should not be considered stable yet!
@chapter Tutorial
In libavfilter, it is possible for filters to have multiple inputs and
multiple outputs.
To illustrate the sorts of things that are possible, we can
use a complex filter graph. For example, the following one:
@example
input --> split --> fifo -----------------------> overlay --> output
| ^
| |
+------> fifo --> crop --> vflip --------+
@end example
splits the stream in two streams, sends one stream through the crop filter
and the vflip filter before merging it back with the other stream by
overlaying it on top. You can use the following command to achieve this:
@example
./ffmpeg -i in.avi -s 240x320 -vfilters "[in] split [T1], fifo, [T2] overlay= 0:240 [out]; [T1] fifo, crop=0:0:-1:240, vflip [T2]
@end example
where input_video.avi has a vertical resolution of 480 pixels. The
result will be that in output the top half of the video is mirrored
onto the bottom half.
Video filters are loaded using the @var{-vfilters} option passed to
ffmpeg or to ffplay. Filters in the same linear chain are separated by
commas. In our example, @var{split, fifo, overlay} are in one linear
chain, and @var{fifo, crop, vflip} are in another. The points where
the linear chains join are labeled by names enclosed in square
brackets. In our example, that is @var{[T1]} and @var{[T2]}. The magic
labels @var{[in]} and @var{[out]} are the points where video is input
and output.
Some filters take in input a list of parameters: they are specified
after the filter name and an equal sign, and are separated each other
by a semicolon.
There exist so-called @var{source filters} that do not have a video
input, and we expect in the future some @var{sink filters} that will
not have video output.
@chapter graph2dot
The @file{graph2dot} program included in the FFmpeg @file{tools}
directory can be used to parse a filter graph description and issue a
corresponding textual representation in the dot language.
Invoke the command:
@example
graph2dot -h
@end example
to see how to use @file{graph2dot}.
You can then pass the dot description to the @file{dot} program (from
the graphviz suite of programs) and obtain a graphical representation
of the filter graph.
For example the sequence of commands:
@example
echo @var{GRAPH_DESCRIPTION} | \
tools/graph2dot -o graph.tmp && \
dot -Tpng graph.tmp -o graph.png && \
display graph.png
@end example
can be used to create and display an image representing the graph
described by the @var{GRAPH_DESCRIPTION} string.
@chapter Available video filters
When you configure your FFmpeg build, you can disable any of the
existing video filters.
The configure output will show the video filters included in your
build.
Below is a description of the currently available video filters.
@section crop
Crop the input video to @var{x}:@var{y}:@var{width}:@var{height}.
@example
./ffmpeg -i in.avi -vfilters "crop=0:0:0:240" out.avi
@end example
@var{x} and @var{y} specify the position of the top-left corner of the
output (non-cropped) area.
The default value of @var{x} and @var{y} is 0.
The @var{width} and @var{height} parameters specify the width and height
of the output (non-cropped) area.
A value of 0 is interpreted as the maximum possible size contained in
the area delimited by the top-left corner at position x:y.
For example the parameters:
@example
"crop=100:100:0:0"
@end example
will delimit the rectangle with the top-left corner placed at position
100:100 and the right-bottom corner corresponding to the right-bottom
corner of the input image.
The default value of @var{width} and @var{height} is 0.
@section format
Convert the input video to one of the specified pixel formats.
Libavfilter will try to pick one that is supported for the input to
the next filter.
The filter accepts a list of pixel format names, separated by ``:'',
for example ``yuv420p:monow:rgb24''.
The following command:
@example
./ffmpeg -i in.avi -vfilters "format=yuv420p" out.avi
@end example
will convert the input video to the format ``yuv420p''.
@section noformat
Force libavfilter not to use any of the specified pixel formats for the
input to the next filter.
The filter accepts a list of pixel format names, separated by ``:'',
for example ``yuv420p:monow:rgb24''.
The following command:
@example
./ffmpeg -i in.avi -vfilters "noformat=yuv420p, vflip" out.avi
@end example
will make libavfilter use a format different from ``yuv420p'' for the
input to the vflip filter.
@section null
Pass the source unchanged to the output.
@section scale
Scale the input video to @var{width}:@var{height} and/or convert the image format.
For example the command:
@example
./ffmpeg -i in.avi -vfilters "scale=200:100" out.avi
@end example
will scale the input video to a size of 200x100.
If the input image format is different from the format requested by
the next filter, the scale filter will convert the input to the
requested format.
If the value for @var{width} or @var{height} is 0, the respective input
size is used for the output.
If the value for @var{width} or @var{height} is -1, the scale filter will
use, for the respective output size, a value that maintains the aspect
ratio of the input image.
The default value of @var{width} and @var{height} is 0.
@section slicify
Pass the images of input video on to next video filter as multiple
slices.
@example
./ffmpeg -i in.avi -vfilters "slicify=32" out.avi
@end example
The filter accepts the slice height as parameter. If the parameter is
not specified it will use the default value of 16.
Adding this in the beginning of filter chains should make filtering
faster due to better use of the memory cache.
@section unsharp
Sharpen or blur the input video. It accepts the following parameters:
@multitable @columnfractions .2 .5 .1 .1 .1
@headitem Name @tab Description @tab Min @tab Max @tab Default
@item @var{luma_msize_x}
@tab Luma matrix horizontal size
@tab 3
@tab 13
@tab 5
@item @var{luma_msize_y}
@tab Luma matrix vertical size
@tab 3
@tab 13
@tab 5
@item @var{luma_amount}
@tab Luma effect strength
@tab -2.0
@tab 5.0
@tab 1.0
@item @var{chroma_msize_x}
@tab Chroma matrix horizontal size
@tab 3
@tab 13
@tab 0
@item @var{chroma_msize_y}
@tab Chroma matrix vertical size
@tab 3
@tab 13
@tab 0
@item @var{chroma_amount}
@tab Chroma effect strength
@tab -2.0
@tab 5.0
@tab 0.0
@end multitable
Negative values for the amount will blur the input video, while positive
values will sharpen. All parameters are optional and default to the
equivalent of the string '5:5:1.0:0:0:0.0'.
@example
# Strong luma sharpen effect parameters
unsharp=7:7:2.5
# Strong blur of both luma and chroma parameters
unsharp=7:7:-2:7:7:-2
# Use the default values with @command{ffmpeg}
./ffmpeg -i in.avi -vfilters "unsharp" out.mp4
@end example
@section vflip
Flip the input video vertically.
@example
./ffmpeg -i in.avi -vfilters "vflip" out.avi
@end example
@chapter Available video sources
Below is a description of the currently available video sources.
@section nullsrc
Null video source, never return images. It is mainly useful as a
template and to be employed in analysis / debugging tools.
It accepts as optional parameter a string of the form
@var{width}:@var{height}, where @var{width} and @var{height} specify the size of
the configured source.
The default values of @var{width} and @var{height} are respectively 352
and 288 (corresponding to the CIF size format).
@chapter Available video sinks
Below is a description of the currently available video sinks.
@section nullsink
Null video sink, do absolutely nothing with the input video. It is
mainly useful as a template and to be employed in analysis / debugging
tools.
@bye

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@@ -1,61 +0,0 @@
A Quick Description Of Rate Distortion Theory.
We want to encode a video, picture or piece of music optimally. What does
"optimally" really mean? It means that we want to get the best quality at a
given filesize OR we want to get the smallest filesize at a given quality
(in practice, these 2 goals are usually the same).
Solving this directly is not practical; trying all byte sequences 1
megabyte in length and selecting the "best looking" sequence will yield
256^1000000 cases to try.
But first, a word about quality, which is also called distortion.
Distortion can be quantified by almost any quality measurement one chooses.
Commonly, the sum of squared differences is used but more complex methods
that consider psychovisual effects can be used as well. It makes no
difference in this discussion.
First step: that rate distortion factor called lambda...
Let's consider the problem of minimizing:
distortion + lambda*rate
rate is the filesize
distortion is the quality
lambda is a fixed value choosen as a tradeoff between quality and filesize
Is this equivalent to finding the best quality for a given max
filesize? The answer is yes. For each filesize limit there is some lambda
factor for which minimizing above will get you the best quality (using your
chosen quality measurement) at the desired (or lower) filesize.
Second step: splitting the problem.
Directly splitting the problem of finding the best quality at a given
filesize is hard because we do not know how many bits from the total
filesize should be allocated to each of the subproblems. But the formula
from above:
distortion + lambda*rate
can be trivially split. Consider:
(distortion0 + distortion1) + lambda*(rate0 + rate1)
This creates a problem made of 2 independent subproblems. The subproblems
might be 2 16x16 macroblocks in a frame of 32x16 size. To minimize:
(distortion0 + distortion1) + lambda*(rate0 + rate1)
we just have to minimize:
distortion0 + lambda*rate0
and
distortion1 + lambda*rate1
I.e, the 2 problems can be solved independently.
Author: Michael Niedermayer
Copyright: LGPL

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@@ -30,7 +30,7 @@ slices, that is, consecutive non-overlapping rectangles of dimension
special converter
These generally are unscaled converters of common
formats, like YUV 4:2:0/4:2:2 -> RGB12/15/16/24/32. Though it could also
formats, like YUV 4:2:0/4:2:2 -> RGB15/16/24/32. Though it could also
in principle contain scalers optimized for specific common cases.
Main path

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@@ -1,64 +0,0 @@
Writing a table generator
This documentation is preliminary.
Parts of the API are not good and should be changed.
Basic concepts
A table generator consists of two files, *_tablegen.c and *_tablegen.h.
The .h file will provide the variable declarations and initialization
code for the tables, the .c calls the initialization code and then prints
the tables as a header file using the tableprint.h helpers.
Both of these files will be compiled for the host system, so to avoid
breakage with cross-compilation neither of them may include, directly
or indirectly, config.h or avconfig.h.
Due to this, the .c file or Makefile may have to provide additional defines
or stubs, though if possible this should be avoided.
In particular, CONFIG_HARDCODED_TABLES should always be defined to 0.
The .c file
This file should include the *_tablegen.h and tableprint.h files and
anything else it needs as long as it does not depend on config.h or
avconfig.h.
In addition to that it must contain a main() function which initializes
all tables by calling the init functions from the .h file and then prints
them.
The printing code typically looks like this:
write_fileheader();
printf("static const uint8_t my_array[100] = {\n");
write_uint8_array(my_array, 100);
printf("};\n");
write_fileheader() adds some minor things like a "this is a generated file"
comment and some standard includes.
tablegen.h defines some write functions for one- and two-dimensional arrays
for standard types - they print only the "core" parts so they are easier
to reuse for multi-dimensional arrays so the outermost {} must be printed
separately.
If there's no standard function for printing the type you need, the
WRITE_1D_FUNC_ARGV macro is a very quick way to create one.
See libavcodec/dv_tablegen.c for an example.
The .h file
This file should contain:
- one or more initialization functions
- the table variable declarations
If CONFIG_HARDCODED_TABLES is set, the initialization functions should
not do anything, and instead of the variable declarations the
generated *_tables.h file should be included.
Since that will be generated in the build directory, the path must be
included, i.e.
#include "libavcodec/example_tables.h"
not
#include "example_tables.h"
Makefile changes
To make the automatic table creation work, you must manually declare the
new dependency.
For this add a line similar to this:
$(SUBDIR)example.o: $(SUBDIR)example_tables.h
under the "ifdef CONFIG_HARDCODED_TABLES" section in the Makefile.

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@@ -1,110 +0,0 @@
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 pathes and scores for each point of our new column is
trivial given we know the previous column best pathes 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

1512
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1280
ffplay.c

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@@ -1,4 +1,3 @@
coder=0
bf=0
flags2=-wpred-dct8x8
wpredp=0

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@@ -3,7 +3,7 @@ flags=+loop
cmp=+chroma
partitions=+parti8x8+parti4x4+partp8x8+partb8x8
me_method=hex
subq=7
subq=6
me_range=16
g=250
keyint_min=25
@@ -14,9 +14,5 @@ qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=3
refs=3
directpred=1
trellis=1
flags2=+mixed_refs+wpred+dct8x8+fastpskip
wpredp=2
flags2=+fastpskip

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@@ -1,23 +0,0 @@
coder=1
flags=+loop
cmp=+chroma
partitions=+parti8x8+parti4x4+partp8x8+partb8x8
me_method=hex
subq=6
me_range=16
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=1
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=3
refs=2
directpred=1
trellis=1
flags2=+bpyramid+mixed_refs+wpred+dct8x8+fastpskip
wpredp=2
rc_lookahead=30

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@@ -1,23 +0,0 @@
coder=1
flags=+loop
cmp=+chroma
partitions=-parti8x8-parti4x4-partp8x8-partb8x8
me_method=dia
subq=2
me_range=16
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=1
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=3
refs=1
directpred=1
trellis=0
flags2=+bpyramid-mixed_refs+wpred-dct8x8+fastpskip
wpredp=2
rc_lookahead=30

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@@ -1,23 +0,0 @@
coder=1
flags=+loop
cmp=+chroma
partitions=+parti8x8+parti4x4+partp8x8+partb8x8
me_method=hex
subq=4
me_range=16
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=1
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=3
refs=2
directpred=1
trellis=1
flags2=+bpyramid-mixed_refs+wpred+dct8x8+fastpskip
wpredp=1
rc_lookahead=20

View File

@@ -1,23 +0,0 @@
coder=1
flags=+loop
cmp=+chroma
partitions=-parti8x8-parti4x4-partp8x8-partb8x8
me_method=dia
subq=2
me_range=16
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=1
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=3
refs=1
directpred=1
trellis=0
flags2=+bpyramid-mixed_refs+wpred-dct8x8+fastpskip
wpredp=1
rc_lookahead=20

View File

@@ -3,7 +3,7 @@ flags=+loop
cmp=+chroma
partitions=-parti8x8-parti4x4-partp8x8-partp4x4-partb8x8
me_method=dia
subq=2
subq=1
me_range=16
g=250
keyint_min=25
@@ -14,9 +14,8 @@ qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=3
bf=4
refs=1
directpred=3
trellis=0
flags2=-bpyramid-wpred-mixed_refs-dct8x8+fastpskip
wpredp=2

View File

@@ -14,9 +14,8 @@ qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=3
bf=4
refs=4
directpred=3
trellis=1
flags2=+wpred+mixed_refs+dct8x8+fastpskip
wpredp=2
flags2=+bpyramid+wpred+mixed_refs+dct8x8+fastpskip

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@@ -4,4 +4,3 @@ flags2=-wpred-dct8x8
level=13
maxrate=768000
bufsize=3000000
wpredp=0

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@@ -5,4 +5,3 @@ flags2=-wpred-dct8x8
level=30
maxrate=10000000
bufsize=10000000
wpredp=0

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@@ -17,4 +17,3 @@ qdiff=4
directpred=1
flags2=+fastpskip
cqp=0
wpredp=0

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@@ -18,4 +18,3 @@ refs=16
directpred=1
flags2=+mixed_refs+dct8x8+fastpskip
cqp=0
wpredp=2

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@@ -17,4 +17,3 @@ qdiff=4
directpred=1
flags2=+fastpskip
cqp=0
wpredp=2

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@@ -18,4 +18,3 @@ refs=2
directpred=1
flags2=+dct8x8+fastpskip
cqp=0
wpredp=2

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@@ -18,4 +18,3 @@ refs=4
directpred=1
flags2=+mixed_refs+dct8x8+fastpskip
cqp=0
wpredp=2

View File

@@ -3,8 +3,8 @@ flags=+loop
cmp=+chroma
partitions=+parti8x8+parti4x4+partp8x8+partp4x4+partb8x8
me_method=tesa
subq=10
me_range=24
subq=9
me_range=32
g=250
keyint_min=25
sc_threshold=40
@@ -14,9 +14,8 @@ qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=3
bf=4
refs=16
directpred=3
trellis=2
flags2=+wpred+mixed_refs+dct8x8-fastpskip
wpredp=2
flags2=+bpyramid+wpred+mixed_refs+dct8x8-fastpskip

View File

@@ -1,22 +0,0 @@
coder=1
flags=+loop
cmp=+chroma
partitions=+parti8x8+parti4x4+partp8x8+partb8x8
me_method=hex
subq=7
me_range=16
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=1
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=3
refs=3
directpred=1
trellis=1
flags2=+bpyramid+mixed_refs+wpred+dct8x8+fastpskip
wpredp=2

View File

@@ -1,22 +0,0 @@
coder=1
flags=+loop
cmp=+chroma
partitions=-parti8x8-parti4x4-partp8x8-partb8x8
me_method=dia
subq=2
me_range=16
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=1
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=3
refs=1
directpred=1
trellis=0
flags2=+bpyramid-mixed_refs+wpred-dct8x8+fastpskip
wpredp=2

View File

@@ -14,9 +14,8 @@ qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=3
bf=4
refs=2
directpred=3
trellis=0
flags2=+wpred+dct8x8+fastpskip
wpredp=2
flags2=+bpyramid+wpred+dct8x8+fastpskip

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@@ -1,23 +0,0 @@
coder=1
flags=+loop
cmp=+chroma
partitions=+parti8x8+parti4x4+partp8x8+partp4x4+partb8x8
me_method=tesa
subq=10
me_range=24
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=2
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=16
refs=16
directpred=3
trellis=2
flags2=+bpyramid+mixed_refs+wpred+dct8x8-fastpskip
wpredp=2
rc_lookahead=60

View File

@@ -1,23 +0,0 @@
coder=1
flags=+loop
cmp=+chroma
partitions=+parti8x8+parti4x4+partp8x8+partp4x4+partb8x8
me_method=tesa
subq=10
me_range=24
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=2
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=16
refs=16
directpred=3
trellis=2
flags2=+bpyramid+mixed_refs+wpred+dct8x8-fastpskip
wpredp=2
rc_lookahead=60

View File

@@ -1,23 +0,0 @@
coder=1
flags=+loop
cmp=+chroma
partitions=+parti8x8+parti4x4+partp8x8+partb8x8
me_method=umh
subq=8
me_range=16
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=2
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=3
refs=5
directpred=3
trellis=1
flags2=+bpyramid+mixed_refs+wpred+dct8x8+fastpskip
wpredp=2
rc_lookahead=50

View File

@@ -1,23 +0,0 @@
coder=1
flags=+loop
cmp=+chroma
partitions=-parti8x8-parti4x4-partp8x8-partb8x8
me_method=dia
subq=2
me_range=16
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=2
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=3
refs=1
directpred=3
trellis=0
flags2=+bpyramid-mixed_refs+wpred-dct8x8+fastpskip
wpredp=2
rc_lookahead=50

View File

@@ -1,23 +0,0 @@
coder=1
flags=+loop
cmp=+chroma
partitions=+parti8x8+parti4x4+partp8x8+partp4x4+partb8x8
me_method=umh
subq=9
me_range=16
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=2
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=3
refs=8
directpred=3
trellis=2
flags2=+bpyramid+mixed_refs+wpred+dct8x8+fastpskip
wpredp=2
rc_lookahead=60

View File

@@ -1,23 +0,0 @@
coder=1
flags=+loop
cmp=+chroma
partitions=-parti8x8-parti4x4-partp8x8-partb8x8
me_method=dia
subq=2
me_range=16
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=2
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=3
refs=1
directpred=3
trellis=0
flags2=+bpyramid-mixed_refs+wpred-dct8x8+fastpskip
wpredp=2
rc_lookahead=60

View File

@@ -14,9 +14,8 @@ qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=3
bf=4
refs=1
directpred=3
trellis=0
flags2=+wpred+dct8x8+fastpskip
wpredp=2
flags2=+bpyramid+wpred+dct8x8+fastpskip

View File

@@ -1,22 +0,0 @@
coder=1
flags=+loop
cmp=+chroma
partitions=+parti8x8+parti4x4-partp8x8-partb8x8
me_method=dia
subq=1
me_range=16
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=1
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=3
refs=1
directpred=1
trellis=0
flags2=+bpyramid-mixed_refs+wpred+dct8x8+fastpskip-mbtree
wpredp=0

View File

@@ -1,22 +0,0 @@
coder=1
flags=+loop
cmp=+chroma
partitions=-parti8x8-parti4x4-partp8x8-partb8x8
me_method=dia
subq=1
me_range=16
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=1
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=3
refs=1
directpred=1
trellis=0
flags2=+bpyramid-mixed_refs+wpred-dct8x8+fastpskip-mbtree
wpredp=0

View File

@@ -1,23 +0,0 @@
coder=0
flags=-loop
cmp=+chroma
partitions=-parti8x8-parti4x4-partp8x8-partb8x8
me_method=dia
subq=0
me_range=16
g=250
keyint_min=25
sc_threshold=0
i_qfactor=0.71
b_strategy=0
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=0
refs=1
directpred=1
trellis=0
flags2=-bpyramid-mixed_refs-wpred-dct8x8+fastpskip-mbtree
wpredp=0
aq_mode=0

View File

@@ -1,23 +0,0 @@
coder=0
flags=-loop
cmp=+chroma
partitions=-parti8x8-parti4x4-partp8x8-partb8x8
me_method=dia
subq=0
me_range=16
g=250
keyint_min=25
sc_threshold=0
i_qfactor=0.71
b_strategy=0
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=0
refs=1
directpred=1
trellis=0
flags2=-bpyramid-mixed_refs-wpred-dct8x8+fastpskip-mbtree
wpredp=0
aq_mode=0

View File

@@ -1,22 +0,0 @@
coder=1
flags=+loop
cmp=+chroma
partitions=+parti8x8+parti4x4+partp8x8+partb8x8
me_method=hex
subq=2
me_range=16
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=1
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=3
refs=1
directpred=1
trellis=0
flags2=+bpyramid-mixed_refs+wpred+dct8x8+fastpskip-mbtree
wpredp=0

View File

@@ -1,22 +0,0 @@
coder=1
flags=+loop
cmp=+chroma
partitions=-parti8x8-parti4x4-partp8x8-partb8x8
me_method=dia
subq=2
me_range=16
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=1
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=3
refs=1
directpred=1
trellis=0
flags2=+bpyramid-mixed_refs+wpred-dct8x8+fastpskip-mbtree
wpredp=0

View File

@@ -1,23 +0,0 @@
coder=1
flags=+loop
cmp=+chroma
partitions=+parti8x8+parti4x4+partp8x8+partp4x4+partb8x8
me_method=umh
subq=10
me_range=24
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=2
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=8
refs=16
directpred=3
trellis=2
flags2=+bpyramid+mixed_refs+wpred+dct8x8+fastpskip
wpredp=2
rc_lookahead=60

View File

@@ -1,23 +0,0 @@
coder=1
flags=+loop
cmp=+chroma
partitions=-parti8x8-parti4x4-partp8x8-partb8x8
me_method=dia
subq=2
me_range=24
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=2
qcomp=0.6
qmin=10
qmax=51
qdiff=4
bf=8
refs=1
directpred=3
trellis=0
flags2=+bpyramid-mixed_refs+wpred-dct8x8+fastpskip
wpredp=2
rc_lookahead=60

367
ffprobe.c
View File

@@ -1,367 +0,0 @@
/*
* FFprobe : Simple Media Prober based on the FFmpeg libraries
* Copyright (c) 2007-2010 Stefano Sabatini
*
* 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 "config.h"
#include "libavformat/avformat.h"
#include "libavcodec/avcodec.h"
#include "libavcodec/opt.h"
#include "libavutil/pixdesc.h"
#include "libavdevice/avdevice.h"
#include "cmdutils.h"
const char program_name[] = "FFprobe";
const int program_birth_year = 2007;
static int do_show_format = 0;
static int do_show_streams = 0;
static int convert_tags = 0;
static int show_value_unit = 0;
static int use_value_prefix = 0;
static int use_byte_value_binary_prefix = 0;
static int use_value_sexagesimal_format = 0;
/* globals */
static const OptionDef options[];
/* FFprobe context */
static const char *input_filename;
static AVInputFormat *iformat = NULL;
static const char *binary_unit_prefixes [] = { "", "Ki", "Mi", "Gi", "Ti", "Pi" };
static const char *decimal_unit_prefixes[] = { "", "K" , "M" , "G" , "T" , "P" };
static const char *unit_second_str = "s" ;
static const char *unit_hertz_str = "Hz" ;
static const char *unit_byte_str = "byte" ;
static const char *unit_bit_per_second_str = "bit/s";
static char *value_string(char *buf, int buf_size, double val, const char *unit)
{
if (unit == unit_second_str && use_value_sexagesimal_format) {
double secs;
int hours, mins;
secs = val;
mins = (int)secs / 60;
secs = secs - mins * 60;
hours = mins / 60;
mins %= 60;
snprintf(buf, buf_size, "%d:%02d:%09.6f", hours, mins, secs);
} else if (use_value_prefix) {
const char *prefix_string;
int index;
if (unit == unit_byte_str && use_byte_value_binary_prefix) {
index = (int) (log(val)/log(2)) / 10;
index = av_clip(index, 0, FF_ARRAY_ELEMS(binary_unit_prefixes) -1);
val /= pow(2, index*10);
prefix_string = binary_unit_prefixes[index];
} else {
index = (int) (log10(val)) / 3;
index = av_clip(index, 0, FF_ARRAY_ELEMS(decimal_unit_prefixes) -1);
val /= pow(10, index*3);
prefix_string = decimal_unit_prefixes[index];
}
snprintf(buf, buf_size, "%.3f %s%s", val, prefix_string, show_value_unit ? unit : "");
} else {
snprintf(buf, buf_size, "%f %s", val, show_value_unit ? unit : "");
}
return buf;
}
static char *time_value_string(char *buf, int buf_size, int64_t val, const AVRational *time_base)
{
if (val == AV_NOPTS_VALUE) {
snprintf(buf, buf_size, "N/A");
} else {
value_string(buf, buf_size, val * av_q2d(*time_base), unit_second_str);
}
return buf;
}
static const char *media_type_string(enum AVMediaType media_type)
{
switch (media_type) {
case AVMEDIA_TYPE_VIDEO: return "video";
case AVMEDIA_TYPE_AUDIO: return "audio";
case AVMEDIA_TYPE_DATA: return "data";
case AVMEDIA_TYPE_SUBTITLE: return "subtitle";
case AVMEDIA_TYPE_ATTACHMENT: return "attachment";
default: return "unknown";
}
}
static void show_stream(AVFormatContext *fmt_ctx, int stream_idx)
{
AVStream *stream = fmt_ctx->streams[stream_idx];
AVCodecContext *dec_ctx;
AVCodec *dec;
char val_str[128];
AVMetadataTag *tag = NULL;
char a, b, c, d;
AVRational display_aspect_ratio;
printf("[STREAM]\n");
printf("index=%d\n", stream->index);
if ((dec_ctx = stream->codec)) {
if ((dec = dec_ctx->codec)) {
printf("codec_name=%s\n", dec->name);
printf("codec_long_name=%s\n", dec->long_name);
} else {
printf("codec_name=unknown\n");
}
printf("codec_type=%s\n", media_type_string(dec_ctx->codec_type));
printf("codec_time_base=%d/%d\n", dec_ctx->time_base.num, dec_ctx->time_base.den);
/* print AVI/FourCC tag */
a = dec_ctx->codec_tag & 0xff;
b = dec_ctx->codec_tag>>8 & 0xff;
c = dec_ctx->codec_tag>>16 & 0xff;
d = dec_ctx->codec_tag>>24 & 0xff;
printf("codec_tag_string=");
if (isprint(a)) printf("%c", a); else printf("[%d]", a);
if (isprint(b)) printf("%c", b); else printf("[%d]", b);
if (isprint(c)) printf("%c", c); else printf("[%d]", c);
if (isprint(d)) printf("%c", d); else printf("[%d]", d);
printf("\ncodec_tag=0x%04x\n", dec_ctx->codec_tag);
switch (dec_ctx->codec_type) {
case AVMEDIA_TYPE_VIDEO:
printf("width=%d\n", dec_ctx->width);
printf("height=%d\n", dec_ctx->height);
printf("has_b_frames=%d\n", dec_ctx->has_b_frames);
if (dec_ctx->sample_aspect_ratio.num) {
printf("sample_aspect_ratio=%d:%d\n", dec_ctx->sample_aspect_ratio.num,
dec_ctx->sample_aspect_ratio.den);
av_reduce(&display_aspect_ratio.num, &display_aspect_ratio.den,
dec_ctx->width * dec_ctx->sample_aspect_ratio.num,
dec_ctx->height * dec_ctx->sample_aspect_ratio.den,
1024*1024);
printf("display_aspect_ratio=%d:%d\n", display_aspect_ratio.num,
display_aspect_ratio.den);
}
printf("pix_fmt=%s\n", dec_ctx->pix_fmt != PIX_FMT_NONE ?
av_pix_fmt_descriptors[dec_ctx->pix_fmt].name : "unknown");
break;
case AVMEDIA_TYPE_AUDIO:
printf("sample_rate=%s\n", value_string(val_str, sizeof(val_str),
dec_ctx->sample_rate,
unit_hertz_str));
printf("channels=%d\n", dec_ctx->channels);
printf("bits_per_sample=%d\n", av_get_bits_per_sample(dec_ctx->codec_id));
break;
}
} else {
printf("codec_type=unknown\n");
}
if (fmt_ctx->iformat->flags & AVFMT_SHOW_IDS)
printf("id=0x%x\n", stream->id);
printf("r_frame_rate=%d/%d\n", stream->r_frame_rate.num, stream->r_frame_rate.den);
printf("avg_frame_rate=%d/%d\n", stream->avg_frame_rate.num, stream->avg_frame_rate.den);
printf("time_base=%d/%d\n", stream->time_base.num, stream->time_base.den);
if (stream->language[0])
printf("language=%s\n", stream->language);
printf("start_time=%s\n", time_value_string(val_str, sizeof(val_str), stream->start_time,
&stream->time_base));
printf("duration=%s\n", time_value_string(val_str, sizeof(val_str), stream->duration,
&stream->time_base));
if (stream->nb_frames)
printf("nb_frames=%"PRId64"\n", stream->nb_frames);
while ((tag = av_metadata_get(stream->metadata, "", tag, AV_METADATA_IGNORE_SUFFIX)))
printf("TAG:%s=%s\n", tag->key, tag->value);
printf("[/STREAM]\n");
}
static void show_format(AVFormatContext *fmt_ctx)
{
AVMetadataTag *tag = NULL;
char val_str[128];
printf("[FORMAT]\n");
printf("filename=%s\n", fmt_ctx->filename);
printf("nb_streams=%d\n", fmt_ctx->nb_streams);
printf("format_name=%s\n", fmt_ctx->iformat->name);
printf("format_long_name=%s\n", fmt_ctx->iformat->long_name);
printf("start_time=%s\n", time_value_string(val_str, sizeof(val_str), fmt_ctx->start_time,
&AV_TIME_BASE_Q));
printf("duration=%s\n", time_value_string(val_str, sizeof(val_str), fmt_ctx->duration,
&AV_TIME_BASE_Q));
printf("size=%s\n", value_string(val_str, sizeof(val_str), fmt_ctx->file_size,
unit_byte_str));
printf("bit_rate=%s\n", value_string(val_str, sizeof(val_str), fmt_ctx->bit_rate,
unit_bit_per_second_str));
if (convert_tags)
av_metadata_conv(fmt_ctx, NULL, fmt_ctx->iformat->metadata_conv);
while ((tag = av_metadata_get(fmt_ctx->metadata, "", tag, AV_METADATA_IGNORE_SUFFIX)))
printf("TAG:%s=%s\n", tag->key, tag->value);
printf("[/FORMAT]\n");
}
static int open_input_file(AVFormatContext **fmt_ctx_ptr, const char *filename)
{
int err, i;
AVFormatContext *fmt_ctx;
fmt_ctx = avformat_alloc_context();
if ((err = av_open_input_file(&fmt_ctx, filename, iformat, 0, NULL)) < 0) {
print_error(filename, err);
return err;
}
/* fill the streams in the format context */
if ((err = av_find_stream_info(fmt_ctx)) < 0) {
print_error(filename, err);
return err;
}
dump_format(fmt_ctx, 0, filename, 0);
/* bind a decoder to each input stream */
for (i = 0; i < fmt_ctx->nb_streams; i++) {
AVStream *stream = fmt_ctx->streams[i];
AVCodec *codec;
if (!(codec = avcodec_find_decoder(stream->codec->codec_id))) {
fprintf(stderr, "Unsupported codec (id=%d) for input stream %d\n",
stream->codec->codec_id, stream->index);
} else if (avcodec_open(stream->codec, codec) < 0) {
fprintf(stderr, "Error while opening codec for input stream %d\n",
stream->index);
}
}
*fmt_ctx_ptr = fmt_ctx;
return 0;
}
static int probe_file(const char *filename)
{
AVFormatContext *fmt_ctx;
int ret, i;
if ((ret = open_input_file(&fmt_ctx, filename)))
return ret;
if (do_show_streams)
for (i = 0; i < fmt_ctx->nb_streams; i++)
show_stream(fmt_ctx, i);
if (do_show_format)
show_format(fmt_ctx);
av_close_input_file(fmt_ctx);
return 0;
}
static void show_usage(void)
{
printf("Simple multimedia streams analyzer\n");
printf("usage: ffprobe [OPTIONS] [INPUT_FILE]\n");
printf("\n");
}
static void opt_format(const char *arg)
{
iformat = av_find_input_format(arg);
if (!iformat) {
fprintf(stderr, "Unknown input format: %s\n", arg);
exit(1);
}
}
static void opt_input_file(const char *arg)
{
if (input_filename) {
fprintf(stderr, "Argument '%s' provided as input filename, but '%s' was already specified.\n",
arg, input_filename);
exit(1);
}
if (!strcmp(arg, "-"))
arg = "pipe:";
input_filename = arg;
}
static void show_help(void)
{
show_usage();
show_help_options(options, "Main options:\n", 0, 0);
printf("\n");
}
static void opt_pretty(void)
{
show_value_unit = 1;
use_value_prefix = 1;
use_byte_value_binary_prefix = 1;
use_value_sexagesimal_format = 1;
}
static const OptionDef options[] = {
#include "cmdutils_common_opts.h"
{ "convert_tags", OPT_BOOL, {(void*)&convert_tags}, "convert tag names to the FFmpeg generic tag names" },
{ "f", HAS_ARG, {(void*)opt_format}, "force format", "format" },
{ "unit", OPT_BOOL, {(void*)&show_value_unit}, "show unit of the displayed values" },
{ "prefix", OPT_BOOL, {(void*)&use_value_prefix}, "use SI prefixes for the displayed values" },
{ "byte_binary_prefix", OPT_BOOL, {(void*)&use_byte_value_binary_prefix},
"use binary prefixes for byte units" },
{ "sexagesimal", OPT_BOOL, {(void*)&use_value_sexagesimal_format},
"use sexagesimal format HOURS:MM:SS.MICROSECONDS for time units" },
{ "pretty", 0, {(void*)&opt_pretty},
"prettify the format of displayed values, make it more human readable" },
{ "show_format", OPT_BOOL, {(void*)&do_show_format} , "show format/container info" },
{ "show_streams", OPT_BOOL, {(void*)&do_show_streams}, "show streams info" },
{ NULL, },
};
int main(int argc, char **argv)
{
av_register_all();
#if CONFIG_AVDEVICE
avdevice_register_all();
#endif
show_banner();
parse_options(argc, argv, options, opt_input_file);
if (!input_filename) {
show_usage();
fprintf(stderr, "You have to specify one input file.\n");
fprintf(stderr, "Use -h to get full help or, even better, run 'man ffprobe'.\n");
exit(1);
}
return probe_file(input_filename);
}

File diff suppressed because it is too large Load Diff

View File

@@ -20,14 +20,14 @@
*/
/**
* @file
* @file libavcodec/4xm.c
* 4XM codec.
*/
#include "libavutil/intreadwrite.h"
#include "avcodec.h"
#include "dsputil.h"
#include "get_bits.h"
#include "bitstream.h"
#include "bytestream.h"
//#undef NDEBUG
@@ -137,8 +137,8 @@ typedef struct FourXContext{
int mv[256];
VLC pre_vlc;
int last_dc;
DECLARE_ALIGNED(16, DCTELEM, block)[6][64];
void *bitstream_buffer;
DECLARE_ALIGNED_8(DCTELEM, block[6][64]);
uint8_t *bitstream_buffer;
unsigned int bitstream_buffer_size;
int version;
CFrameBuffer cfrm[CFRAME_BUFFER_COUNT];
@@ -237,15 +237,12 @@ static void idct(DCTELEM block[64]){
}
static av_cold void init_vlcs(FourXContext *f){
static VLC_TYPE table[8][32][2];
int i;
for(i=0; i<8; i++){
block_type_vlc[0][i].table= table[i];
block_type_vlc[0][i].table_allocated= 32;
init_vlc(&block_type_vlc[0][i], BLOCK_TYPE_VLC_BITS, 7,
&block_type_tab[0][i][0][1], 2, 1,
&block_type_tab[0][i][0][0], 2, 1, INIT_VLC_USE_NEW_STATIC);
&block_type_tab[0][i][0][0], 2, 1, 1);
}
}
@@ -378,10 +375,8 @@ static int decode_p_frame(FourXContext *f, const uint8_t *buf, int length){
return -1;
}
av_fast_malloc(&f->bitstream_buffer, &f->bitstream_buffer_size, bitstream_size + FF_INPUT_BUFFER_PADDING_SIZE);
if (!f->bitstream_buffer)
return AVERROR(ENOMEM);
f->dsp.bswap_buf(f->bitstream_buffer, (const uint32_t*)(buf + extra), bitstream_size/4);
f->bitstream_buffer= av_fast_realloc(f->bitstream_buffer, &f->bitstream_buffer_size, bitstream_size + FF_INPUT_BUFFER_PADDING_SIZE);
f->dsp.bswap_buf((uint32_t*)f->bitstream_buffer, (const uint32_t*)(buf + extra), bitstream_size/4);
init_get_bits(&f->gb, f->bitstream_buffer, 8*bitstream_size);
f->wordstream= (const uint16_t*)(buf + extra + bitstream_size);
@@ -641,18 +636,9 @@ static int decode_i_frame(FourXContext *f, const uint8_t *buf, int length){
uint16_t *dst= (uint16_t*)f->current_picture.data[0];
const int stride= f->current_picture.linesize[0]>>1;
const unsigned int bitstream_size= AV_RL32(buf);
int token_count av_unused;
unsigned int prestream_size;
const uint8_t *prestream;
if (length < bitstream_size + 12) {
av_log(f->avctx, AV_LOG_ERROR, "packet size too small\n");
return AVERROR_INVALIDDATA;
}
token_count = AV_RL32(buf + bitstream_size + 8);
prestream_size = 4 * AV_RL32(buf + bitstream_size + 4);
prestream = buf + bitstream_size + 12;
const int token_count av_unused = AV_RL32(buf + bitstream_size + 8);
unsigned int prestream_size= 4*AV_RL32(buf + bitstream_size + 4);
const uint8_t *prestream= buf + bitstream_size + 12;
if(prestream_size + bitstream_size + 12 != length
|| bitstream_size > (1<<26)
@@ -667,10 +653,8 @@ static int decode_i_frame(FourXContext *f, const uint8_t *buf, int length){
prestream_size= length + buf - prestream;
av_fast_malloc(&f->bitstream_buffer, &f->bitstream_buffer_size, prestream_size + FF_INPUT_BUFFER_PADDING_SIZE);
if (!f->bitstream_buffer)
return AVERROR(ENOMEM);
f->dsp.bswap_buf(f->bitstream_buffer, (const uint32_t*)prestream, prestream_size/4);
f->bitstream_buffer= av_fast_realloc(f->bitstream_buffer, &f->bitstream_buffer_size, prestream_size + FF_INPUT_BUFFER_PADDING_SIZE);
f->dsp.bswap_buf((uint32_t*)f->bitstream_buffer, (const uint32_t*)prestream, prestream_size/4);
init_get_bits(&f->pre_gb, f->bitstream_buffer, 8*prestream_size);
f->last_dc= 0*128*8*8;
@@ -693,10 +677,8 @@ static int decode_i_frame(FourXContext *f, const uint8_t *buf, int length){
static int decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
const uint8_t *buf, int buf_size)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
FourXContext * const f = avctx->priv_data;
AVFrame *picture = data;
AVFrame *p, temp;
@@ -824,7 +806,7 @@ static av_cold int decode_init(AVCodecContext *avctx){
init_vlcs(f);
if(f->version>2) avctx->pix_fmt= PIX_FMT_RGB565;
else avctx->pix_fmt= PIX_FMT_BGR555;
else avctx->pix_fmt= PIX_FMT_RGB555;
return 0;
}
@@ -841,24 +823,20 @@ static av_cold int decode_end(AVCodecContext *avctx){
f->cfrm[i].allocated_size= 0;
}
free_vlc(&f->pre_vlc);
if(f->current_picture.data[0])
avctx->release_buffer(avctx, &f->current_picture);
if(f->last_picture.data[0])
avctx->release_buffer(avctx, &f->last_picture);
return 0;
}
AVCodec fourxm_decoder = {
"4xm",
AVMEDIA_TYPE_VIDEO,
CODEC_TYPE_VIDEO,
CODEC_ID_4XM,
sizeof(FourXContext),
decode_init,
NULL,
decode_end,
decode_frame,
CODEC_CAP_DR1,
/*CODEC_CAP_DR1,*/
.long_name = NULL_IF_CONFIG_SMALL("4X Movie"),
};

View File

@@ -20,7 +20,7 @@
*/
/**
* @file
* @file libavcodec/8bps.c
* QT 8BPS Video Decoder by Roberto Togni
* For more information about the 8BPS format, visit:
* http://www.pcisys.net/~melanson/codecs/
@@ -58,10 +58,8 @@ typedef struct EightBpsContext {
* Decode a frame
*
*/
static int decode_frame(AVCodecContext *avctx, void *data, int *data_size, AVPacket *avpkt)
static int decode_frame(AVCodecContext *avctx, void *data, int *data_size, const uint8_t *buf, int buf_size)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
EightBpsContext * const c = avctx->priv_data;
const unsigned char *encoded = buf;
unsigned char *pixptr, *pixptr_end;
@@ -159,6 +157,10 @@ static av_cold int decode_init(AVCodecContext *avctx)
c->pic.data[0] = NULL;
if (avcodec_check_dimensions(avctx, avctx->width, avctx->height) < 0) {
return 1;
}
switch (avctx->bits_per_coded_sample) {
case 8:
avctx->pix_fmt = PIX_FMT_PAL8;
@@ -179,7 +181,7 @@ static av_cold int decode_init(AVCodecContext *avctx)
case 32:
avctx->pix_fmt = PIX_FMT_RGB32;
c->planes = 4;
#if HAVE_BIGENDIAN
#ifdef WORDS_BIGENDIAN
c->planemap[0] = 1; // 1st plane is red
c->planemap[1] = 2; // 2nd plane is green
c->planemap[2] = 3; // 3rd plane is blue
@@ -221,7 +223,7 @@ static av_cold int decode_end(AVCodecContext *avctx)
AVCodec eightbps_decoder = {
"8bps",
AVMEDIA_TYPE_VIDEO,
CODEC_TYPE_VIDEO,
CODEC_ID_8BPS,
sizeof(EightBpsContext),
decode_init,

View File

@@ -20,7 +20,7 @@
*/
/**
* @file
* @file libavcodec/8svx.c
* 8svx audio decoder
* @author Jaikrishnan Menon
* supports: fibonacci delta encoding
@@ -42,10 +42,8 @@ static const int16_t exponential[16] = { -128<<8, -64<<8, -32<<8, -16<<8, -8<<8,
/** decode a frame */
static int eightsvx_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
AVPacket *avpkt)
const uint8_t *buf, int buf_size)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
EightSvxContext *esc = avctx->priv_data;
int16_t *out_data = data;
int consumed = buf_size;
@@ -94,7 +92,7 @@ static av_cold int eightsvx_decode_init(AVCodecContext *avctx)
AVCodec eightsvx_fib_decoder = {
.name = "8svx_fib",
.type = AVMEDIA_TYPE_AUDIO,
.type = CODEC_TYPE_AUDIO,
.id = CODEC_ID_8SVX_FIB,
.priv_data_size = sizeof (EightSvxContext),
.init = eightsvx_decode_init,
@@ -104,7 +102,7 @@ AVCodec eightsvx_fib_decoder = {
AVCodec eightsvx_exp_decoder = {
.name = "8svx_exp",
.type = AVMEDIA_TYPE_AUDIO,
.type = CODEC_TYPE_AUDIO,
.id = CODEC_ID_8SVX_EXP,
.priv_data_size = sizeof (EightSvxContext),
.init = eightsvx_decode_init,

View File

@@ -3,11 +3,10 @@ include $(SUBDIR)../config.mak
NAME = avcodec
FFLIBS = avutil
HEADERS = avcodec.h avfft.h dxva2.h opt.h vaapi.h vdpau.h xvmc.h
HEADERS = avcodec.h opt.h vdpau.h xvmc.h
OBJS = allcodecs.o \
audioconvert.o \
avpacket.o \
bitstream.o \
bitstream_filter.o \
dsputil.o \
@@ -27,92 +26,56 @@ OBJS = allcodecs.o \
# parts needed for many different codecs
OBJS-$(CONFIG_AANDCT) += aandcttab.o
OBJS-$(CONFIG_ENCODERS) += faandct.o jfdctfst.o jfdctint.o
OBJS-$(CONFIG_DCT) += dct.o
OBJS-$(CONFIG_DWT) += dwt.o
OBJS-$(CONFIG_DXVA2) += dxva2.o
FFT-OBJS-$(CONFIG_HARDCODED_TABLES) += cos_tables.o
OBJS-$(CONFIG_FFT) += avfft.o fft.o $(FFT-OBJS-yes)
OBJS-$(CONFIG_FFT) += fft.o
OBJS-$(CONFIG_GOLOMB) += golomb.o
OBJS-$(CONFIG_H264DSP) += h264dsp.o h264idct.o h264pred.o
OBJS-$(CONFIG_LPC) += lpc.o
OBJS-$(CONFIG_LSP) += lsp.o
OBJS-$(CONFIG_MDCT) += mdct.o
RDFT-OBJS-$(CONFIG_HARDCODED_TABLES) += sin_tables.o
OBJS-$(CONFIG_RDFT) += rdft.o $(RDFT-OBJS-yes)
OBJS-$(CONFIG_VAAPI) += vaapi.o
OBJS-$(CONFIG_VDPAU) += vdpau.o
OBJS-$(CONFIG_OLDSCALER) += imgresample.o
OBJS-$(CONFIG_RDFT) += rdft.o
# decoders/encoders/hardware accelerators
OBJS-$(CONFIG_AAC_DECODER) += aacdec.o aactab.o aacsbr.o aacps.o
OBJS-$(CONFIG_AAC_ENCODER) += aacenc.o aaccoder.o \
aacpsy.o aactab.o \
psymodel.o iirfilter.o \
mpeg4audio.o
# decoders/encoders
OBJS-$(CONFIG_AAC_DECODER) += aac.o aactab.o mpeg4audio.o aac_parser.o aac_ac3_parser.o
OBJS-$(CONFIG_AASC_DECODER) += aasc.o msrledec.o
OBJS-$(CONFIG_AC3_DECODER) += ac3dec.o ac3dec_data.o ac3.o
OBJS-$(CONFIG_AC3_DECODER) += eac3dec.o ac3dec.o ac3tab.o ac3dec_data.o ac3.o
OBJS-$(CONFIG_AC3_ENCODER) += ac3enc.o ac3tab.o ac3.o
OBJS-$(CONFIG_ALAC_DECODER) += alac.o
OBJS-$(CONFIG_ALAC_ENCODER) += alacenc.o
OBJS-$(CONFIG_ALS_DECODER) += alsdec.o bgmc.o mpeg4audio.o
OBJS-$(CONFIG_AMRNB_DECODER) += amrnbdec.o celp_filters.o \
celp_math.o acelp_filters.o \
acelp_vectors.o \
acelp_pitch_delay.o
OBJS-$(CONFIG_ALAC_ENCODER) += alacenc.o lpc.o
OBJS-$(CONFIG_AMV_DECODER) += sp5xdec.o mjpegdec.o mjpeg.o
OBJS-$(CONFIG_ANM_DECODER) += anm.o
OBJS-$(CONFIG_APE_DECODER) += apedec.o
OBJS-$(CONFIG_ASV1_DECODER) += asv1.o mpeg12data.o
OBJS-$(CONFIG_ASV1_ENCODER) += asv1.o mpeg12data.o
OBJS-$(CONFIG_ASV2_DECODER) += asv1.o mpeg12data.o
OBJS-$(CONFIG_ASV2_ENCODER) += asv1.o mpeg12data.o
OBJS-$(CONFIG_ATRAC1_DECODER) += atrac1.o atrac.o
OBJS-$(CONFIG_ATRAC3_DECODER) += atrac3.o atrac.o
OBJS-$(CONFIG_AURA_DECODER) += cyuv.o
OBJS-$(CONFIG_AURA2_DECODER) += aura.o
OBJS-$(CONFIG_ATRAC3_DECODER) += atrac3.o
OBJS-$(CONFIG_AVS_DECODER) += avs.o
OBJS-$(CONFIG_BETHSOFTVID_DECODER) += bethsoftvideo.o
OBJS-$(CONFIG_BFI_DECODER) += bfi.o
OBJS-$(CONFIG_BINK_DECODER) += bink.o binkidct.o
OBJS-$(CONFIG_BINKAUDIO_DCT_DECODER) += binkaudio.o wma.o
OBJS-$(CONFIG_BINKAUDIO_RDFT_DECODER) += binkaudio.o wma.o
OBJS-$(CONFIG_BMP_DECODER) += bmp.o msrledec.o
OBJS-$(CONFIG_BMP_ENCODER) += bmpenc.o
OBJS-$(CONFIG_C93_DECODER) += c93.o
OBJS-$(CONFIG_CAVS_DECODER) += cavs.o cavsdec.o cavsdsp.o \
mpeg12data.o mpegvideo.o
OBJS-$(CONFIG_CDGRAPHICS_DECODER) += cdgraphics.o
OBJS-$(CONFIG_CAVS_DECODER) += cavs.o cavsdec.o cavsdsp.o mpeg12data.o mpegvideo.o
OBJS-$(CONFIG_CINEPAK_DECODER) += cinepak.o
OBJS-$(CONFIG_CLJR_DECODER) += cljr.o
OBJS-$(CONFIG_CLJR_ENCODER) += cljr.o
OBJS-$(CONFIG_COOK_DECODER) += cook.o
OBJS-$(CONFIG_CSCD_DECODER) += cscd.o
OBJS-$(CONFIG_CYUV_DECODER) += cyuv.o
OBJS-$(CONFIG_DCA_DECODER) += dca.o synth_filter.o dcadsp.o
OBJS-$(CONFIG_DCA_DECODER) += dca.o
OBJS-$(CONFIG_DNXHD_DECODER) += dnxhddec.o dnxhddata.o
OBJS-$(CONFIG_DNXHD_ENCODER) += dnxhdenc.o dnxhddata.o \
mpegvideo_enc.o motion_est.o \
ratecontrol.o mpeg12data.o \
mpegvideo.o
OBJS-$(CONFIG_DPX_DECODER) += dpx.o
OBJS-$(CONFIG_DNXHD_ENCODER) += dnxhdenc.o dnxhddata.o mpegvideo_enc.o motion_est.o ratecontrol.o mpeg12data.o mpegvideo.o
OBJS-$(CONFIG_DSICINAUDIO_DECODER) += dsicinav.o
OBJS-$(CONFIG_DSICINVIDEO_DECODER) += dsicinav.o
OBJS-$(CONFIG_DVBSUB_DECODER) += dvbsubdec.o
OBJS-$(CONFIG_DVBSUB_ENCODER) += dvbsub.o
OBJS-$(CONFIG_DVDSUB_DECODER) += dvdsubdec.o
OBJS-$(CONFIG_DVDSUB_ENCODER) += dvdsubenc.o
OBJS-$(CONFIG_DVVIDEO_DECODER) += dv.o dvdata.o
OBJS-$(CONFIG_DVVIDEO_ENCODER) += dv.o dvdata.o
OBJS-$(CONFIG_DVVIDEO_DECODER) += dv.o
OBJS-$(CONFIG_DVVIDEO_ENCODER) += dv.o
OBJS-$(CONFIG_DXA_DECODER) += dxa.o
OBJS-$(CONFIG_EAC3_DECODER) += eac3dec.o eac3dec_data.o
OBJS-$(CONFIG_EAC3_DECODER) += eac3dec.o ac3dec.o ac3tab.o ac3dec_data.o ac3.o
OBJS-$(CONFIG_EACMV_DECODER) += eacmv.o
OBJS-$(CONFIG_EAMAD_DECODER) += eamad.o eaidct.o mpeg12.o \
mpeg12data.o mpegvideo.o \
error_resilience.o
OBJS-$(CONFIG_EATGQ_DECODER) += eatgq.o eaidct.o
OBJS-$(CONFIG_EATGV_DECODER) += eatgv.o
OBJS-$(CONFIG_EATQI_DECODER) += eatqi.o eaidct.o mpeg12.o \
mpeg12data.o mpegvideo.o \
error_resilience.o
OBJS-$(CONFIG_EATQI_DECODER) += eatqi.o eaidct.o mpeg12.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_EIGHTBPS_DECODER) += 8bps.o
OBJS-$(CONFIG_EIGHTSVX_EXP_DECODER) += 8svx.o
OBJS-$(CONFIG_EIGHTSVX_FIB_DECODER) += 8svx.o
@@ -121,155 +84,94 @@ OBJS-$(CONFIG_FFV1_DECODER) += ffv1.o rangecoder.o
OBJS-$(CONFIG_FFV1_ENCODER) += ffv1.o rangecoder.o
OBJS-$(CONFIG_FFVHUFF_DECODER) += huffyuv.o
OBJS-$(CONFIG_FFVHUFF_ENCODER) += huffyuv.o
OBJS-$(CONFIG_FLAC_DECODER) += flacdec.o flacdata.o flac.o
OBJS-$(CONFIG_FLAC_ENCODER) += flacenc.o flacdata.o flac.o
OBJS-$(CONFIG_FLAC_DECODER) += flacdec.o
OBJS-$(CONFIG_FLAC_ENCODER) += flacenc.o lpc.o
OBJS-$(CONFIG_FLASHSV_DECODER) += flashsv.o
OBJS-$(CONFIG_FLASHSV_ENCODER) += flashsvenc.o
OBJS-$(CONFIG_FLIC_DECODER) += flicvideo.o
OBJS-$(CONFIG_FLV_DECODER) += h263dec.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_FLV_ENCODER) += mpegvideo_enc.o motion_est.o ratecontrol.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_FOURXM_DECODER) += 4xm.o
OBJS-$(CONFIG_FRAPS_DECODER) += fraps.o huffman.o
OBJS-$(CONFIG_FRWU_DECODER) += frwu.o
OBJS-$(CONFIG_GIF_DECODER) += gifdec.o lzw.o
OBJS-$(CONFIG_GIF_ENCODER) += gif.o lzwenc.o
OBJS-$(CONFIG_H261_DECODER) += h261dec.o h261.o \
mpegvideo.o error_resilience.o
OBJS-$(CONFIG_H261_ENCODER) += h261enc.o h261.o \
mpegvideo_enc.o motion_est.o \
ratecontrol.o mpeg12data.o \
mpegvideo.o
OBJS-$(CONFIG_H263_DECODER) += h263dec.o h263.o ituh263dec.o \
mpeg4video.o mpeg4videodec.o flvdec.o\
intelh263dec.o mpegvideo.o \
error_resilience.o
OBJS-$(CONFIG_H263_VAAPI_HWACCEL) += vaapi_mpeg4.o
OBJS-$(CONFIG_H263_ENCODER) += mpegvideo_enc.o mpeg4video.o \
mpeg4videoenc.o motion_est.o \
ratecontrol.o h263.o ituh263enc.o \
flvenc.o mpeg12data.o \
mpegvideo.o error_resilience.o
OBJS-$(CONFIG_H264_DECODER) += h264.o \
h264_loopfilter.o h264_direct.o \
cabac.o h264_sei.o h264_ps.o \
h264_refs.o h264_cavlc.o h264_cabac.o\
mpegvideo.o error_resilience.o
OBJS-$(CONFIG_H264_DXVA2_HWACCEL) += dxva2_h264.o
OBJS-$(CONFIG_GIF_ENCODER) += gif.o
OBJS-$(CONFIG_H261_DECODER) += h261dec.o h261.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_H261_ENCODER) += h261enc.o h261.o mpegvideo_enc.o motion_est.o ratecontrol.o mpeg12data.o mpegvideo.o
OBJS-$(CONFIG_H263_DECODER) += h263dec.o h263.o h263_parser.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_H263I_DECODER) += h263dec.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_H263_ENCODER) += mpegvideo_enc.o motion_est.o ratecontrol.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_H263P_ENCODER) += mpegvideo_enc.o motion_est.o ratecontrol.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_H264_DECODER) += h264.o h264idct.o h264pred.o h264_parser.o cabac.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_H264_ENCODER) += h264enc.o h264dspenc.o
OBJS-$(CONFIG_H264_VAAPI_HWACCEL) += vaapi_h264.o
OBJS-$(CONFIG_H264_VDPAU_DECODER) += vdpau.o h264.o h264idct.o h264pred.o h264_parser.o cabac.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_HUFFYUV_DECODER) += huffyuv.o
OBJS-$(CONFIG_HUFFYUV_ENCODER) += huffyuv.o
OBJS-$(CONFIG_IDCIN_DECODER) += idcinvideo.o
OBJS-$(CONFIG_IFF_BYTERUN1_DECODER) += iff.o
OBJS-$(CONFIG_IFF_ILBM_DECODER) += iff.o
OBJS-$(CONFIG_IMC_DECODER) += imc.o
OBJS-$(CONFIG_INDEO2_DECODER) += indeo2.o
OBJS-$(CONFIG_INDEO3_DECODER) += indeo3.o
OBJS-$(CONFIG_INDEO5_DECODER) += indeo5.o ivi_common.o ivi_dsp.o
OBJS-$(CONFIG_INTERPLAY_DPCM_DECODER) += dpcm.o
OBJS-$(CONFIG_INTERPLAY_VIDEO_DECODER) += interplayvideo.o
OBJS-$(CONFIG_JPEGLS_DECODER) += jpeglsdec.o jpegls.o \
mjpegdec.o mjpeg.o
OBJS-$(CONFIG_JPEGLS_DECODER) += jpeglsdec.o jpegls.o mjpegdec.o mjpeg.o
OBJS-$(CONFIG_JPEGLS_ENCODER) += jpeglsenc.o jpegls.o
OBJS-$(CONFIG_KGV1_DECODER) += kgv1dec.o
OBJS-$(CONFIG_KMVC_DECODER) += kmvc.o
OBJS-$(CONFIG_LJPEG_ENCODER) += ljpegenc.o mjpegenc.o mjpeg.o \
mpegvideo_enc.o motion_est.o \
ratecontrol.o mpeg12data.o \
mpegvideo.o
OBJS-$(CONFIG_LJPEG_ENCODER) += ljpegenc.o mjpegenc.o mjpeg.o mpegvideo_enc.o motion_est.o ratecontrol.o mpeg12data.o mpegvideo.o
OBJS-$(CONFIG_LOCO_DECODER) += loco.o
OBJS-$(CONFIG_MACE3_DECODER) += mace.o
OBJS-$(CONFIG_MACE6_DECODER) += mace.o
OBJS-$(CONFIG_MDEC_DECODER) += mdec.o mpeg12.o mpeg12data.o \
mpegvideo.o error_resilience.o
OBJS-$(CONFIG_MDEC_DECODER) += mdec.o mpeg12.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_MIMIC_DECODER) += mimic.o
OBJS-$(CONFIG_MJPEG_DECODER) += mjpegdec.o mjpeg.o
OBJS-$(CONFIG_MJPEG_ENCODER) += mjpegenc.o mjpeg.o \
mpegvideo_enc.o motion_est.o \
ratecontrol.o mpeg12data.o \
mpegvideo.o
OBJS-$(CONFIG_MJPEG_ENCODER) += mjpegenc.o mjpeg.o mpegvideo_enc.o motion_est.o ratecontrol.o mpeg12data.o mpegvideo.o
OBJS-$(CONFIG_MJPEGB_DECODER) += mjpegbdec.o mjpegdec.o mjpeg.o
OBJS-$(CONFIG_MLP_DECODER) += mlpdec.o mlpdsp.o
OBJS-$(CONFIG_MLP_DECODER) += mlpdec.o mlp_parser.o mlp.o
OBJS-$(CONFIG_MMVIDEO_DECODER) += mmvideo.o
OBJS-$(CONFIG_MOTIONPIXELS_DECODER) += motionpixels.o
OBJS-$(CONFIG_MP1_DECODER) += mpegaudiodec.o mpegaudiodecheader.o \
mpegaudio.o mpegaudiodata.o
OBJS-$(CONFIG_MP2_DECODER) += mpegaudiodec.o mpegaudiodecheader.o \
mpegaudio.o mpegaudiodata.o
OBJS-$(CONFIG_MP2_ENCODER) += mpegaudioenc.o mpegaudio.o \
mpegaudiodata.o
OBJS-$(CONFIG_MP3ADU_DECODER) += mpegaudiodec.o mpegaudiodecheader.o \
mpegaudio.o mpegaudiodata.o
OBJS-$(CONFIG_MP3ON4_DECODER) += mpegaudiodec.o mpegaudiodecheader.o \
mpegaudio.o mpegaudiodata.o \
mpeg4audio.o
OBJS-$(CONFIG_MP3_DECODER) += mpegaudiodec.o mpegaudiodecheader.o \
mpegaudio.o mpegaudiodata.o
OBJS-$(CONFIG_MPC7_DECODER) += mpc7.o mpc.o mpegaudiodec.o \
mpegaudiodecheader.o mpegaudio.o \
mpegaudiodata.o
OBJS-$(CONFIG_MPC8_DECODER) += mpc8.o mpc.o mpegaudiodec.o \
mpegaudiodecheader.o mpegaudio.o \
mpegaudiodata.o
OBJS-$(CONFIG_MPEGVIDEO_DECODER) += mpeg12.o mpeg12data.o \
mpegvideo.o error_resilience.o
OBJS-$(CONFIG_MPEG_XVMC_DECODER) += mpegvideo_xvmc.o
OBJS-$(CONFIG_MPEG1VIDEO_DECODER) += mpeg12.o mpeg12data.o \
mpegvideo.o error_resilience.o
OBJS-$(CONFIG_MPEG1VIDEO_ENCODER) += mpeg12enc.o mpegvideo_enc.o \
motion_est.o ratecontrol.o \
mpeg12.o mpeg12data.o \
mpegvideo.o error_resilience.o
OBJS-$(CONFIG_MPEG2_VAAPI_HWACCEL) += vaapi_mpeg2.o
OBJS-$(CONFIG_MPEG2VIDEO_DECODER) += mpeg12.o mpeg12data.o \
mpegvideo.o error_resilience.o
OBJS-$(CONFIG_MPEG2VIDEO_ENCODER) += mpeg12enc.o mpegvideo_enc.o \
motion_est.o ratecontrol.o \
mpeg12.o mpeg12data.o \
mpegvideo.o error_resilience.o
OBJS-$(CONFIG_MPEG4_VAAPI_HWACCEL) += vaapi_mpeg4.o
OBJS-$(CONFIG_MSMPEG4V1_DECODER) += msmpeg4.o msmpeg4data.o
OBJS-$(CONFIG_MSMPEG4V1_ENCODER) += msmpeg4.o msmpeg4data.o h263dec.o \
h263.o ituh263dec.o mpeg4videodec.o
OBJS-$(CONFIG_MSMPEG4V2_DECODER) += msmpeg4.o msmpeg4data.o h263dec.o \
h263.o ituh263dec.o mpeg4videodec.o
OBJS-$(CONFIG_MSMPEG4V2_ENCODER) += msmpeg4.o msmpeg4data.o h263dec.o \
h263.o ituh263dec.o mpeg4videodec.o
OBJS-$(CONFIG_MSMPEG4V3_DECODER) += msmpeg4.o msmpeg4data.o h263dec.o \
h263.o ituh263dec.o mpeg4videodec.o
OBJS-$(CONFIG_MSMPEG4V3_ENCODER) += msmpeg4.o msmpeg4data.o h263dec.o \
h263.o ituh263dec.o mpeg4videodec.o
OBJS-$(CONFIG_MP1_DECODER) += mpegaudiodec.o mpegaudiodecheader.o mpegaudio.o mpegaudiodata.o
OBJS-$(CONFIG_MP2_DECODER) += mpegaudiodec.o mpegaudiodecheader.o mpegaudio.o mpegaudiodata.o
OBJS-$(CONFIG_MP2_ENCODER) += mpegaudioenc.o mpegaudio.o mpegaudiodata.o
OBJS-$(CONFIG_MP3ADU_DECODER) += mpegaudiodec.o mpegaudiodecheader.o mpegaudio.o mpegaudiodata.o
OBJS-$(CONFIG_MP3ON4_DECODER) += mpegaudiodec.o mpegaudiodecheader.o mpegaudio.o mpegaudiodata.o mpeg4audio.o
OBJS-$(CONFIG_MP3_DECODER) += mpegaudiodec.o mpegaudiodecheader.o mpegaudio.o mpegaudiodata.o
OBJS-$(CONFIG_MPC7_DECODER) += mpc7.o mpc.o mpegaudiodec.o mpegaudiodecheader.o mpegaudio.o mpegaudiodata.o
OBJS-$(CONFIG_MPC8_DECODER) += mpc8.o mpc.o mpegaudiodec.o mpegaudiodecheader.o mpegaudio.o mpegaudiodata.o
OBJS-$(CONFIG_MPEG_VDPAU_DECODER) += vdpau.o mpeg12.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_MPEG1_VDPAU_DECODER) += vdpau.o mpeg12.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_MPEG_XVMC_DECODER) += mpegvideo_xvmc.o mpeg12.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_MPEGVIDEO_DECODER) += mpeg12.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_MPEG1VIDEO_DECODER) += mpeg12.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_MPEG1VIDEO_ENCODER) += mpeg12enc.o mpeg12data.o mpegvideo_enc.o motion_est.o ratecontrol.o mpeg12.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_MPEG2VIDEO_DECODER) += mpeg12.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_MPEG2VIDEO_ENCODER) += mpeg12enc.o mpeg12data.o mpegvideo_enc.o motion_est.o ratecontrol.o mpeg12.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_MPEG4_DECODER) += h263dec.o h263.o mpeg4video_parser.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_MPEG4_ENCODER) += mpegvideo_enc.o motion_est.o ratecontrol.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_MSMPEG4V1_DECODER) += msmpeg4.o msmpeg4data.o h263dec.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_MSMPEG4V1_ENCODER) += msmpeg4.o msmpeg4data.o mpegvideo_enc.o motion_est.o ratecontrol.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_MSMPEG4V2_DECODER) += msmpeg4.o msmpeg4data.o h263dec.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_MSMPEG4V2_ENCODER) += msmpeg4.o msmpeg4data.o mpegvideo_enc.o motion_est.o ratecontrol.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_MSMPEG4V3_DECODER) += msmpeg4.o msmpeg4data.o h263dec.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_MSMPEG4V3_ENCODER) += msmpeg4.o msmpeg4data.o mpegvideo_enc.o motion_est.o ratecontrol.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_MSRLE_DECODER) += msrle.o msrledec.o
OBJS-$(CONFIG_MSVIDEO1_DECODER) += msvideo1.o
OBJS-$(CONFIG_MSZH_DECODER) += lcldec.o
OBJS-$(CONFIG_NELLYMOSER_DECODER) += nellymoserdec.o nellymoser.o
OBJS-$(CONFIG_NELLYMOSER_ENCODER) += nellymoserenc.o nellymoser.o
OBJS-$(CONFIG_NUV_DECODER) += nuv.o rtjpeg.o
OBJS-$(CONFIG_PAM_DECODER) += pnmdec.o pnm.o
OBJS-$(CONFIG_PAM_ENCODER) += pamenc.o pnm.o
OBJS-$(CONFIG_PBM_DECODER) += pnmdec.o pnm.o
OBJS-$(CONFIG_PAM_ENCODER) += pnmenc.o pnm.o
OBJS-$(CONFIG_PBM_ENCODER) += pnmenc.o pnm.o
OBJS-$(CONFIG_PCX_DECODER) += pcx.o
OBJS-$(CONFIG_PCX_ENCODER) += pcxenc.o
OBJS-$(CONFIG_PGM_DECODER) += pnmdec.o pnm.o
OBJS-$(CONFIG_PGM_ENCODER) += pnmenc.o pnm.o
OBJS-$(CONFIG_PGMYUV_DECODER) += pnmdec.o pnm.o
OBJS-$(CONFIG_PGMYUV_ENCODER) += pnmenc.o pnm.o
OBJS-$(CONFIG_PGSSUB_DECODER) += pgssubdec.o
OBJS-$(CONFIG_PNG_DECODER) += png.o pngdec.o
OBJS-$(CONFIG_PNG_ENCODER) += png.o pngenc.o
OBJS-$(CONFIG_PPM_DECODER) += pnmdec.o pnm.o
OBJS-$(CONFIG_PPM_ENCODER) += pnmenc.o pnm.o
OBJS-$(CONFIG_PTX_DECODER) += ptx.o
OBJS-$(CONFIG_QCELP_DECODER) += qcelpdec.o celp_math.o \
celp_filters.o acelp_vectors.o \
acelp_filters.o
OBJS-$(CONFIG_QDM2_DECODER) += qdm2.o mpegaudiodec.o \
mpegaudiodecheader.o mpegaudio.o \
mpegaudiodata.o
OBJS-$(CONFIG_QCELP_DECODER) += qcelpdec.o qcelp_lsp.o celp_math.o celp_filters.o
OBJS-$(CONFIG_QDM2_DECODER) += qdm2.o mpegaudiodec.o mpegaudiodecheader.o mpegaudio.o mpegaudiodata.o
OBJS-$(CONFIG_QDRAW_DECODER) += qdrw.o
OBJS-$(CONFIG_QPEG_DECODER) += qpeg.o
OBJS-$(CONFIG_QTRLE_DECODER) += qtrle.o
OBJS-$(CONFIG_QTRLE_ENCODER) += qtrleenc.o
OBJS-$(CONFIG_R210_DECODER) += r210dec.o
OBJS-$(CONFIG_RA_144_DECODER) += ra144.o celp_filters.o
OBJS-$(CONFIG_RA_288_DECODER) += ra288.o celp_math.o celp_filters.o
OBJS-$(CONFIG_RAWVIDEO_DECODER) += rawdec.o
@@ -280,105 +182,70 @@ OBJS-$(CONFIG_ROQ_ENCODER) += roqvideoenc.o roqvideo.o elbg.o
OBJS-$(CONFIG_ROQ_DPCM_DECODER) += dpcm.o
OBJS-$(CONFIG_ROQ_DPCM_ENCODER) += roqaudioenc.o
OBJS-$(CONFIG_RPZA_DECODER) += rpza.o
OBJS-$(CONFIG_RV10_DECODER) += rv10.o
OBJS-$(CONFIG_RV10_ENCODER) += rv10enc.o
OBJS-$(CONFIG_RV20_DECODER) += rv10.o
OBJS-$(CONFIG_RV20_ENCODER) += rv20enc.o
OBJS-$(CONFIG_RV30_DECODER) += rv30.o rv34.o rv30dsp.o \
mpegvideo.o error_resilience.o
OBJS-$(CONFIG_RV40_DECODER) += rv40.o rv34.o rv40dsp.o \
mpegvideo.o error_resilience.o
OBJS-$(CONFIG_RV10_DECODER) += rv10.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_RV10_ENCODER) += rv10.o mpegvideo_enc.o motion_est.o ratecontrol.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_RV20_DECODER) += rv10.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_RV20_ENCODER) += rv10.o mpegvideo_enc.o motion_est.o ratecontrol.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_RV30_DECODER) += rv30.o rv34.o h264pred.o rv30dsp.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_RV40_DECODER) += rv40.o rv34.o h264pred.o rv40dsp.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_SGI_DECODER) += sgidec.o
OBJS-$(CONFIG_SGI_ENCODER) += sgienc.o rle.o
OBJS-$(CONFIG_SHORTEN_DECODER) += shorten.o
OBJS-$(CONFIG_SIPR_DECODER) += sipr.o acelp_pitch_delay.o \
celp_math.o acelp_vectors.o \
acelp_filters.o celp_filters.o \
sipr16k.o
OBJS-$(CONFIG_SMACKAUD_DECODER) += smacker.o
OBJS-$(CONFIG_SMACKER_DECODER) += smacker.o
OBJS-$(CONFIG_SMC_DECODER) += smc.o
OBJS-$(CONFIG_SNOW_DECODER) += snow.o rangecoder.o
OBJS-$(CONFIG_SNOW_ENCODER) += snow.o rangecoder.o motion_est.o \
ratecontrol.o h263.o \
mpegvideo.o error_resilience.o \
ituh263enc.o mpegvideo_enc.o \
mpeg12data.o
OBJS-$(CONFIG_SNOW_ENCODER) += snow.o rangecoder.o motion_est.o ratecontrol.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_SOL_DPCM_DECODER) += dpcm.o
OBJS-$(CONFIG_SONIC_DECODER) += sonic.o
OBJS-$(CONFIG_SONIC_ENCODER) += sonic.o
OBJS-$(CONFIG_SONIC_LS_ENCODER) += sonic.o
OBJS-$(CONFIG_SP5X_DECODER) += sp5xdec.o mjpegdec.o mjpeg.o
OBJS-$(CONFIG_SUNRAST_DECODER) += sunrast.o
OBJS-$(CONFIG_SVQ1_DECODER) += svq1dec.o svq1.o h263.o \
mpegvideo.o error_resilience.o
OBJS-$(CONFIG_SVQ1_ENCODER) += svq1enc.o svq1.o \
motion_est.o h263.o \
mpegvideo.o error_resilience.o \
ituh263enc.o mpegvideo_enc.o \
ratecontrol.o mpeg12data.o
OBJS-$(CONFIG_SVQ3_DECODER) += h264.o svq3.o \
h264_loopfilter.o h264_direct.o \
h264_sei.o h264_ps.o h264_refs.o \
h264_cavlc.o h264_cabac.o cabac.o \
mpegvideo.o error_resilience.o \
svq1dec.o svq1.o h263.o
OBJS-$(CONFIG_SVQ1_DECODER) += svq1dec.o svq1.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_SVQ1_ENCODER) += svq1enc.o svq1.o motion_est.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_SVQ3_DECODER) += h264.o h264idct.o h264pred.o h264_parser.o cabac.o mpegvideo.o error_resilience.o svq1dec.o svq1.o h263.o
OBJS-$(CONFIG_TARGA_DECODER) += targa.o
OBJS-$(CONFIG_TARGA_ENCODER) += targaenc.o rle.o
OBJS-$(CONFIG_THEORA_DECODER) += xiph.o
OBJS-$(CONFIG_THEORA_DECODER) += vp3.o xiph.o vp3dsp.o
OBJS-$(CONFIG_THP_DECODER) += mjpegdec.o mjpeg.o
OBJS-$(CONFIG_TIERTEXSEQVIDEO_DECODER) += tiertexseqv.o
OBJS-$(CONFIG_TIFF_DECODER) += tiff.o lzw.o faxcompr.o
OBJS-$(CONFIG_TIFF_ENCODER) += tiffenc.o rle.o lzwenc.o
OBJS-$(CONFIG_TMV_DECODER) += tmv.o cga_data.o
OBJS-$(CONFIG_TRUEMOTION1_DECODER) += truemotion1.o
OBJS-$(CONFIG_TRUEMOTION2_DECODER) += truemotion2.o
OBJS-$(CONFIG_TRUESPEECH_DECODER) += truespeech.o
OBJS-$(CONFIG_TSCC_DECODER) += tscc.o msrledec.o
OBJS-$(CONFIG_TTA_DECODER) += tta.o
OBJS-$(CONFIG_TWINVQ_DECODER) += twinvq.o celp_math.o
OBJS-$(CONFIG_TXD_DECODER) += txd.o s3tc.o
OBJS-$(CONFIG_ULTI_DECODER) += ulti.o
OBJS-$(CONFIG_V210_DECODER) += v210dec.o
OBJS-$(CONFIG_V210_ENCODER) += v210enc.o
OBJS-$(CONFIG_V210X_DECODER) += v210x.o
OBJS-$(CONFIG_VB_DECODER) += vb.o
OBJS-$(CONFIG_VC1_DECODER) += vc1dec.o vc1.o vc1data.o vc1dsp.o \
msmpeg4.o msmpeg4data.o \
intrax8.o intrax8dsp.o
OBJS-$(CONFIG_VC1_DXVA2_HWACCEL) += dxva2_vc1.o
OBJS-$(CONFIG_VC1_VAAPI_HWACCEL) += vaapi_vc1.o
OBJS-$(CONFIG_VC1_DECODER) += vc1.o vc1data.o vc1dsp.o msmpeg4data.o h263dec.o h263.o intrax8.o intrax8dsp.o error_resilience.o mpegvideo.o msmpeg4.o
OBJS-$(CONFIG_VC1_VDPAU_DECODER) += vdpau.o vc1.o vc1data.o vc1dsp.o msmpeg4data.o h263dec.o h263.o intrax8.o intrax8dsp.o error_resilience.o mpegvideo.o msmpeg4.o
OBJS-$(CONFIG_VCR1_DECODER) += vcr1.o
OBJS-$(CONFIG_VCR1_ENCODER) += vcr1.o
OBJS-$(CONFIG_VMDAUDIO_DECODER) += vmdav.o
OBJS-$(CONFIG_VMDVIDEO_DECODER) += vmdav.o
OBJS-$(CONFIG_VMNC_DECODER) += vmnc.o
OBJS-$(CONFIG_VORBIS_DECODER) += vorbis_dec.o vorbis.o \
vorbis_data.o xiph.o
OBJS-$(CONFIG_VORBIS_ENCODER) += vorbis_enc.o vorbis.o \
vorbis_data.o
OBJS-$(CONFIG_VORBIS_DECODER) += vorbis_dec.o vorbis.o vorbis_data.o xiph.o
OBJS-$(CONFIG_VORBIS_ENCODER) += vorbis_enc.o vorbis.o vorbis_data.o
OBJS-$(CONFIG_VP3_DECODER) += vp3.o vp3dsp.o
OBJS-$(CONFIG_VP5_DECODER) += vp5.o vp56.o vp56data.o vp56dsp.o \
vp3dsp.o
OBJS-$(CONFIG_VP6_DECODER) += vp6.o vp56.o vp56data.o vp56dsp.o \
vp3dsp.o vp6dsp.o huffman.o
OBJS-$(CONFIG_VP5_DECODER) += vp5.o vp56.o vp56data.o vp3dsp.o
OBJS-$(CONFIG_VP6_DECODER) += vp6.o vp56.o vp56data.o vp3dsp.o vp6dsp.o huffman.o
OBJS-$(CONFIG_VP6A_DECODER) += vp6.o vp56.o vp56data.o vp3dsp.o vp6dsp.o huffman.o
OBJS-$(CONFIG_VP6F_DECODER) += vp6.o vp56.o vp56data.o vp3dsp.o vp6dsp.o huffman.o
OBJS-$(CONFIG_VQA_DECODER) += vqavideo.o
OBJS-$(CONFIG_WAVPACK_DECODER) += wavpack.o
OBJS-$(CONFIG_WMAPRO_DECODER) += wmaprodec.o wma.o
OBJS-$(CONFIG_WMAV1_DECODER) += wmadec.o wma.o aactab.o
OBJS-$(CONFIG_WMAV1_ENCODER) += wmaenc.o wma.o aactab.o
OBJS-$(CONFIG_WMAV2_DECODER) += wmadec.o wma.o aactab.o
OBJS-$(CONFIG_WMAV2_ENCODER) += wmaenc.o wma.o aactab.o
OBJS-$(CONFIG_WMAVOICE_DECODER) += wmavoice.o \
celp_math.o celp_filters.o \
acelp_vectors.o acelp_filters.o
OBJS-$(CONFIG_WMV1_DECODER) += msmpeg4.o msmpeg4data.o
OBJS-$(CONFIG_WMV2_DECODER) += wmv2dec.o wmv2.o \
msmpeg4.o msmpeg4data.o \
intrax8.o intrax8dsp.o
OBJS-$(CONFIG_WMV2_ENCODER) += wmv2enc.o wmv2.o \
msmpeg4.o msmpeg4data.o \
mpeg4videodec.o ituh263dec.o h263dec.o
OBJS-$(CONFIG_WMAV1_DECODER) += wmadec.o wma.o
OBJS-$(CONFIG_WMAV1_ENCODER) += wmaenc.o wma.o
OBJS-$(CONFIG_WMAV2_DECODER) += wmadec.o wma.o
OBJS-$(CONFIG_WMAV2_ENCODER) += wmaenc.o wma.o
OBJS-$(CONFIG_WMV1_DECODER) += h263dec.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_WMV1_ENCODER) += mpegvideo_enc.o motion_est.o ratecontrol.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_WMV2_DECODER) += wmv2dec.o wmv2.o msmpeg4.o msmpeg4data.o h263dec.o h263.o intrax8.o intrax8dsp.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_WMV2_ENCODER) += wmv2enc.o wmv2.o msmpeg4.o msmpeg4data.o mpegvideo_enc.o motion_est.o ratecontrol.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_WMV3_DECODER) += vc1.o vc1data.o vc1dsp.o msmpeg4data.o h263dec.o h263.o intrax8.o intrax8dsp.o error_resilience.o mpegvideo.o msmpeg4.o
OBJS-$(CONFIG_WMV3_VDPAU_DECODER) += vdpau.o vc1.o vc1data.o vc1dsp.o msmpeg4data.o h263dec.o h263.o intrax8.o intrax8dsp.o error_resilience.o mpegvideo.o msmpeg4.o
OBJS-$(CONFIG_WNV1_DECODER) += wnv1.o
OBJS-$(CONFIG_WS_SND1_DECODER) += ws-snd1.o
OBJS-$(CONFIG_XAN_DPCM_DECODER) += dpcm.o
@@ -386,8 +253,6 @@ OBJS-$(CONFIG_XAN_WC3_DECODER) += xan.o
OBJS-$(CONFIG_XAN_WC4_DECODER) += xan.o
OBJS-$(CONFIG_XL_DECODER) += xl.o
OBJS-$(CONFIG_XSUB_DECODER) += xsubdec.o
OBJS-$(CONFIG_XSUB_ENCODER) += xsubenc.o
OBJS-$(CONFIG_YOP_DECODER) += yop.o
OBJS-$(CONFIG_ZLIB_DECODER) += lcldec.o
OBJS-$(CONFIG_ZLIB_ENCODER) += lclenc.o
OBJS-$(CONFIG_ZMBV_DECODER) += zmbv.o
@@ -396,7 +261,6 @@ OBJS-$(CONFIG_ZMBV_ENCODER) += zmbvenc.o
# (AD)PCM decoders/encoders
OBJS-$(CONFIG_PCM_ALAW_DECODER) += pcm.o
OBJS-$(CONFIG_PCM_ALAW_ENCODER) += pcm.o
OBJS-$(CONFIG_PCM_BLURAY_DECODER) += pcm-mpeg.o
OBJS-$(CONFIG_PCM_DVD_DECODER) += pcm.o
OBJS-$(CONFIG_PCM_DVD_ENCODER) += pcm.o
OBJS-$(CONFIG_PCM_F32BE_DECODER) += pcm.o
@@ -480,64 +344,40 @@ OBJS-$(CONFIG_ADPCM_YAMAHA_DECODER) += adpcm.o
OBJS-$(CONFIG_ADPCM_YAMAHA_ENCODER) += adpcm.o
# libavformat dependencies
OBJS-$(CONFIG_ADTS_MUXER) += mpeg4audio.o
OBJS-$(CONFIG_CAF_DEMUXER) += mpeg4audio.o mpegaudiodata.o
OBJS-$(CONFIG_DV_DEMUXER) += dvdata.o
OBJS-$(CONFIG_DV_MUXER) += dvdata.o
OBJS-$(CONFIG_FLAC_DEMUXER) += flacdec.o flacdata.o flac.o
OBJS-$(CONFIG_FLAC_MUXER) += flacdec.o flacdata.o flac.o
OBJS-$(CONFIG_FLV_DEMUXER) += mpeg4audio.o
OBJS-$(CONFIG_EAC3_DEMUXER) += ac3_parser.o ac3tab.o aac_ac3_parser.o
OBJS-$(CONFIG_FLAC_MUXER) += flacdec.o
OBJS-$(CONFIG_GXF_DEMUXER) += mpeg12data.o
OBJS-$(CONFIG_IFF_DEMUXER) += iff.o
OBJS-$(CONFIG_MATROSKA_AUDIO_MUXER) += xiph.o mpeg4audio.o \
flacdec.o flacdata.o flac.o
OBJS-$(CONFIG_MATROSKA_AUDIO_MUXER) += xiph.o mpeg4audio.o flacdec.o
OBJS-$(CONFIG_MATROSKA_DEMUXER) += mpeg4audio.o
OBJS-$(CONFIG_MATROSKA_MUXER) += xiph.o mpeg4audio.o \
flacdec.o flacdata.o flac.o
OBJS-$(CONFIG_MATROSKA_MUXER) += xiph.o mpeg4audio.o flacdec.o
OBJS-$(CONFIG_MOV_DEMUXER) += mpeg4audio.o mpegaudiodata.o
OBJS-$(CONFIG_MPEGTS_MUXER) += mpegvideo.o mpeg4audio.o
OBJS-$(CONFIG_MPEGTS_MUXER) += mpegvideo.o
OBJS-$(CONFIG_NUT_MUXER) += mpegaudiodata.o
OBJS-$(CONFIG_OGG_DEMUXER) += flacdec.o flacdata.o flac.o \
dirac.o mpeg12data.o
OBJS-$(CONFIG_OGG_MUXER) += xiph.o flacdec.o flacdata.o flac.o
OBJS-$(CONFIG_OGG_DEMUXER) += flacdec.o
OBJS-$(CONFIG_OGG_MUXER) += xiph.o flacdec.o
OBJS-$(CONFIG_RTP_MUXER) += mpegvideo.o
OBJS-$(CONFIG_WEBM_MUXER) += xiph.o mpeg4audio.o \
flacdec.o flacdata.o flac.o
# external codec libraries
OBJS-$(CONFIG_LIBDIRAC_DECODER) += libdiracdec.o
OBJS-$(CONFIG_LIBDIRAC_ENCODER) += libdiracenc.o libdirac_libschro.o
OBJS-$(CONFIG_LIBFAAC_ENCODER) += libfaac.o
OBJS-$(CONFIG_LIBFAAD_DECODER) += libfaad.o
OBJS-$(CONFIG_LIBGSM_DECODER) += libgsm.o
OBJS-$(CONFIG_LIBGSM_ENCODER) += libgsm.o
OBJS-$(CONFIG_LIBGSM_MS_DECODER) += libgsm.o
OBJS-$(CONFIG_LIBGSM_MS_ENCODER) += libgsm.o
OBJS-$(CONFIG_LIBMP3LAME_ENCODER) += libmp3lame.o
OBJS-$(CONFIG_LIBOPENCORE_AMRNB_DECODER) += libopencore-amr.o
OBJS-$(CONFIG_LIBOPENCORE_AMRNB_ENCODER) += libopencore-amr.o
OBJS-$(CONFIG_LIBOPENCORE_AMRWB_DECODER) += libopencore-amr.o
OBJS-$(CONFIG_LIBOPENJPEG_DECODER) += libopenjpeg.o
OBJS-$(CONFIG_LIBSCHROEDINGER_DECODER) += libschroedingerdec.o \
libschroedinger.o \
libdirac_libschro.o
OBJS-$(CONFIG_LIBSCHROEDINGER_ENCODER) += libschroedingerenc.o \
libschroedinger.o \
libdirac_libschro.o
OBJS-$(CONFIG_LIBSPEEX_DECODER) += libspeexdec.o
OBJS-$(CONFIG_LIBTHEORA_ENCODER) += libtheoraenc.o
OBJS-$(CONFIG_LIBVORBIS_ENCODER) += libvorbis.o
OBJS-$(CONFIG_LIBVO_AACENC_ENCODER) += libvo-aacenc.o mpeg4audio.o
OBJS-$(CONFIG_LIBVPX_DECODER) += libvpxdec.o
OBJS-$(CONFIG_LIBVPX_ENCODER) += libvpxenc.o
OBJS-$(CONFIG_LIBX264_ENCODER) += libx264.o
OBJS-$(CONFIG_LIBXVID_ENCODER) += libxvidff.o libxvid_rc.o
OBJS-$(CONFIG_LIBAMR_NB) += libamr.o
OBJS-$(CONFIG_LIBAMR_WB) += libamr.o
OBJS-$(CONFIG_LIBDIRAC_DECODER) += libdiracdec.o
OBJS-$(CONFIG_LIBDIRAC_ENCODER) += libdiracenc.o libdirac_libschro.o
OBJS-$(CONFIG_LIBFAAC) += libfaac.o
OBJS-$(CONFIG_LIBFAAD) += libfaad.o
OBJS-$(CONFIG_LIBGSM) += libgsm.o
OBJS-$(CONFIG_LIBMP3LAME) += libmp3lame.o
OBJS-$(CONFIG_LIBOPENJPEG) += libopenjpeg.o
OBJS-$(CONFIG_LIBSCHROEDINGER_DECODER) += libschroedingerdec.o libschroedinger.o libdirac_libschro.o
OBJS-$(CONFIG_LIBSCHROEDINGER_ENCODER) += libschroedingerenc.o libschroedinger.o libdirac_libschro.o
OBJS-$(CONFIG_LIBSPEEX) += libspeexdec.o
OBJS-$(CONFIG_LIBTHEORA) += libtheoraenc.o
OBJS-$(CONFIG_LIBVORBIS) += libvorbis.o
OBJS-$(CONFIG_LIBX264) += libx264.o
OBJS-$(CONFIG_LIBXVID) += libxvidff.o libxvid_rc.o
# parsers
OBJS-$(CONFIG_AAC_PARSER) += aac_parser.o aac_ac3_parser.o \
mpeg4audio.o
OBJS-$(CONFIG_AC3_PARSER) += ac3_parser.o ac3tab.o \
aac_ac3_parser.o
OBJS-$(CONFIG_AAC_PARSER) += aac_parser.o aac_ac3_parser.o mpeg4audio.o
OBJS-$(CONFIG_AC3_PARSER) += ac3_parser.o ac3tab.o aac_ac3_parser.o
OBJS-$(CONFIG_CAVSVIDEO_PARSER) += cavs_parser.o
OBJS-$(CONFIG_DCA_PARSER) += dca_parser.o
OBJS-$(CONFIG_DIRAC_PARSER) += dirac_parser.o
@@ -546,39 +386,24 @@ OBJS-$(CONFIG_DVBSUB_PARSER) += dvbsub_parser.o
OBJS-$(CONFIG_DVDSUB_PARSER) += dvdsub_parser.o
OBJS-$(CONFIG_H261_PARSER) += h261_parser.o
OBJS-$(CONFIG_H263_PARSER) += h263_parser.o
OBJS-$(CONFIG_H264_PARSER) += h264_parser.o h264.o \
cabac.o \
h264_refs.o h264_sei.o h264_direct.o \
h264_loopfilter.o h264_cabac.o \
h264_cavlc.o h264_ps.o \
mpegvideo.o error_resilience.o
OBJS-$(CONFIG_H264_PARSER) += h264_parser.o
OBJS-$(CONFIG_MJPEG_PARSER) += mjpeg_parser.o
OBJS-$(CONFIG_MLP_PARSER) += mlp_parser.o mlp.o
OBJS-$(CONFIG_MPEG4VIDEO_PARSER) += mpeg4video_parser.o h263.o \
mpegvideo.o error_resilience.o \
mpeg4videodec.o mpeg4video.o \
ituh263dec.o h263dec.o
OBJS-$(CONFIG_MPEGAUDIO_PARSER) += mpegaudio_parser.o \
mpegaudiodecheader.o mpegaudiodata.o
OBJS-$(CONFIG_MPEGVIDEO_PARSER) += mpegvideo_parser.o \
mpeg12.o mpeg12data.o \
mpegvideo.o error_resilience.o
OBJS-$(CONFIG_MPEG4VIDEO_PARSER) += mpeg4video_parser.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_MPEGAUDIO_PARSER) += mpegaudio_parser.o mpegaudiodecheader.o mpegaudiodata.o
OBJS-$(CONFIG_MPEGVIDEO_PARSER) += mpegvideo_parser.o mpeg12.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_PNM_PARSER) += pnm_parser.o pnm.o
OBJS-$(CONFIG_VC1_PARSER) += vc1_parser.o vc1.o vc1data.o \
msmpeg4.o msmpeg4data.o mpeg4video.o \
h263.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_VC1_PARSER) += vc1_parser.o
OBJS-$(CONFIG_VP3_PARSER) += vp3_parser.o
# bitstream filters
OBJS-$(CONFIG_AAC_ADTSTOASC_BSF) += aac_adtstoasc_bsf.o
OBJS-$(CONFIG_DUMP_EXTRADATA_BSF) += dump_extradata_bsf.o
OBJS-$(CONFIG_H264_MP4TOANNEXB_BSF) += h264_mp4toannexb_bsf.o
OBJS-$(CONFIG_IMX_DUMP_HEADER_BSF) += imx_dump_header_bsf.o
OBJS-$(CONFIG_MJPEGA_DUMP_HEADER_BSF) += mjpega_dump_header_bsf.o
OBJS-$(CONFIG_MOV2TEXTSUB_BSF) += movsub_bsf.o
OBJS-$(CONFIG_MP3_HEADER_COMPRESS_BSF) += mp3_header_compress_bsf.o
OBJS-$(CONFIG_MP3_HEADER_DECOMPRESS_BSF) += mp3_header_decompress_bsf.o \
mpegaudiodata.o
OBJS-$(CONFIG_MP3_HEADER_DECOMPRESS_BSF) += mp3_header_decompress_bsf.o mpegaudiodata.o
OBJS-$(CONFIG_NOISE_BSF) += noise_bsf.o
OBJS-$(CONFIG_REMOVE_EXTRADATA_BSF) += remove_extradata_bsf.o
OBJS-$(CONFIG_TEXT2MOVSUB_BSF) += movsub_bsf.o
@@ -589,70 +414,124 @@ OBJS-$(HAVE_OS2THREADS) += os2thread.o
OBJS-$(HAVE_PTHREADS) += pthread.o
OBJS-$(HAVE_W32THREADS) += w32thread.o
# processor-specific code
YASM-OBJS-FFT-$(HAVE_AMD3DNOW) += x86/fft_3dn.o
YASM-OBJS-FFT-$(HAVE_AMD3DNOWEXT) += x86/fft_3dn2.o
YASM-OBJS-FFT-$(HAVE_SSE) += x86/fft_sse.o
YASM-OBJS-$(CONFIG_FFT) += x86/fft_mmx.o $(YASM-OBJS-FFT-yes)
YASM-OBJS-$(CONFIG_GPL) += x86/h264_deblock_sse2.o \
x86/h264_idct_sse2.o \
MMX-OBJS-$(CONFIG_CAVS_DECODER) += x86/cavsdsp_mmx.o
MMX-OBJS-$(CONFIG_ENCODERS) += x86/dsputilenc_mmx.o
MMX-OBJS-$(CONFIG_FLAC_ENCODER) += x86/flacdsp_mmx.o
MMX-OBJS-$(CONFIG_GPL) += x86/idct_mmx.o
MMX-OBJS-$(CONFIG_SNOW_DECODER) += x86/snowdsp_mmx.o
MMX-OBJS-$(CONFIG_THEORA_DECODER) += x86/vp3dsp_mmx.o x86/vp3dsp_sse2.o
MMX-OBJS-$(CONFIG_VC1_DECODER) += x86/vc1dsp_mmx.o
MMX-OBJS-$(CONFIG_VP3_DECODER) += x86/vp3dsp_mmx.o x86/vp3dsp_sse2.o
MMX-OBJS-$(CONFIG_VP5_DECODER) += x86/vp3dsp_mmx.o x86/vp3dsp_sse2.o
MMX-OBJS-$(CONFIG_VP6_DECODER) += x86/vp3dsp_mmx.o x86/vp3dsp_sse2.o \
x86/vp6dsp_mmx.o x86/vp6dsp_sse2.o
MMX-OBJS-$(CONFIG_VP6A_DECODER) += x86/vp3dsp_mmx.o x86/vp3dsp_sse2.o \
x86/vp6dsp_mmx.o x86/vp6dsp_sse2.o
MMX-OBJS-$(CONFIG_VP6F_DECODER) += x86/vp3dsp_mmx.o x86/vp3dsp_sse2.o \
x86/vp6dsp_mmx.o x86/vp6dsp_sse2.o
MMX-OBJS-$(CONFIG_WMV3_DECODER) += x86/vc1dsp_mmx.o
MMX-OBJS-$(HAVE_YASM) += x86/dsputil_yasm.o \
$(YASM-OBJS-yes)
OBJS-$(HAVE_MMX) += x86/cpuid.o \
x86/dnxhd_mmx.o \
x86/dsputil_mmx.o \
x86/fdct_mmx.o \
x86/idct_mmx_xvid.o \
x86/idct_sse2_xvid.o \
x86/motion_est_mmx.o \
x86/mpegvideo_mmx.o \
x86/simple_idct_mmx.o \
$(MMX-OBJS-yes)
OBJS-$(ARCH_ALPHA) += alpha/dsputil_alpha.o \
alpha/dsputil_alpha_asm.o \
alpha/motion_est_alpha.o \
alpha/motion_est_mvi_asm.o \
alpha/mpegvideo_alpha.o \
alpha/simple_idct_alpha.o \
OBJS-$(ARCH_ARM) += arm/dsputil_arm.o \
arm/dsputil_arm_s.o \
arm/jrevdct_arm.o \
arm/mpegvideo_arm.o \
arm/simple_idct_arm.o \
OBJS-$(HAVE_ARMV5TE) += arm/mpegvideo_armv5te.o \
arm/mpegvideo_armv5te_s.o \
arm/simple_idct_armv5te.o \
OBJS-$(HAVE_ARMV6) += arm/simple_idct_armv6.o \
OBJS-$(HAVE_ARMVFP) += arm/dsputil_vfp.o \
arm/float_arm_vfp.o \
OBJS-$(HAVE_IWMMXT) += arm/dsputil_iwmmxt.o \
arm/mpegvideo_iwmmxt.o \
OBJS-$(HAVE_NEON) += arm/dsputil_neon.o \
arm/dsputil_neon_s.o \
arm/h264dsp_neon.o \
arm/h264idct_neon.o \
arm/simple_idct_neon.o \
OBJS-$(ARCH_BFIN) += bfin/dsputil_bfin.o \
bfin/fdct_bfin.o \
bfin/idct_bfin.o \
bfin/mpegvideo_bfin.o \
bfin/pixels_bfin.o \
bfin/vp3_bfin.o \
bfin/vp3_idct_bfin.o \
OBJS-$(ARCH_PPC) += ppc/dsputil_ppc.o \
ALTIVEC-OBJS-$(CONFIG_H264_DECODER) += ppc/h264_altivec.o
ALTIVEC-OBJS-$(CONFIG_OLDSCALER) += ppc/imgresample_altivec.o
ALTIVEC-OBJS-$(CONFIG_SNOW_DECODER) += ppc/snow_altivec.o
ALTIVEC-OBJS-$(CONFIG_VC1_DECODER) += ppc/vc1dsp_altivec.o
ALTIVEC-OBJS-$(CONFIG_WMV3_DECODER) += ppc/vc1dsp_altivec.o
OBJS-$(HAVE_ALTIVEC) += ppc/check_altivec.o \
ppc/dsputil_altivec.o \
ppc/fdct_altivec.o \
ppc/fft_altivec.o \
ppc/float_altivec.o \
ppc/gmc_altivec.o \
ppc/idct_altivec.o \
ppc/int_altivec.o \
ppc/mpegvideo_altivec.o \
$(ALTIVEC-OBJS-yes)
OBJS-$(ARCH_SH4) += sh4/dsputil_align.o \
sh4/dsputil_sh4.o \
sh4/idct_sh4.o \
OBJS-$(CONFIG_MLIB) += mlib/dsputil_mlib.o \
-include $(SUBDIR)$(ARCH)/Makefile
OBJS-$(HAVE_MMI) += ps2/dsputil_mmi.o \
ps2/idct_mmi.o \
ps2/mpegvideo_mmi.o \
SKIPHEADERS = %_tablegen.h
SKIPHEADERS-$(CONFIG_DXVA2) += dxva2.h dxva2_internal.h
SKIPHEADERS-$(CONFIG_LIBDIRAC) += libdirac.h
SKIPHEADERS-$(CONFIG_LIBSCHROEDINGER) += libschroedinger.h
SKIPHEADERS-$(CONFIG_VAAPI) += vaapi_internal.h
SKIPHEADERS-$(CONFIG_VDPAU) += vdpau.h
SKIPHEADERS-$(CONFIG_XVMC) += xvmc.h
SKIPHEADERS += mpegaudio3.h
OBJS-$(HAVE_VIS) += sparc/dsputil_vis.o \
sparc/simple_idct_vis.o \
EXAMPLES = api
TESTPROGS = cabac dct eval fft h264 iirfilter rangecoder snow
TESTPROGS-$(ARCH_X86) += x86/cpuid
TESTPROGS-$(HAVE_MMX) += motion
TESTOBJS = dctref.o
HOSTPROGS = costablegen
TESTS = $(addsuffix -test$(EXESUF), cabac dct eval fft h264 rangecoder snow)
TESTS-$(CONFIG_OLDSCALER) += imgresample-test$(EXESUF)
TESTS-$(ARCH_X86) += x86/cpuid-test$(EXESUF) motion-test$(EXESUF)
CLEANFILES = apiexample$(EXESUF)
DIRS = alpha arm bfin mlib ppc ps2 sh4 sparc x86
CLEANFILES = sin_tables.c cos_tables.c *_tables.h *_tablegen$(HOSTEXESUF)
include $(SUBDIR)../subdir.mak
$(SUBDIR)dct-test$(EXESUF): $(SUBDIR)dctref.o
$(SUBDIR)cos_tables.c: $(SUBDIR)costablegen$(HOSTEXESUF)
$(M)./$< > $@
$(SUBDIR)sin_tables.c: $(SUBDIR)costablegen$(HOSTEXESUF)
$(M)./$< sin > $@
ifdef CONFIG_MPEGAUDIO_HP
$(SUBDIR)mpegaudio_tablegen$(HOSTEXESUF): HOSTCFLAGS += -DFRAC_BITS=23
$(SUBDIR)mpegaudio_tablegen.ho: CPPFLAGS += -DFRAC_BITS=23
else
$(SUBDIR)mpegaudio_tablegen$(HOSTEXESUF): HOSTCFLAGS += -DFRAC_BITS=15
$(SUBDIR)mpegaudio_tablegen.ho: CPPFLAGS += -DFRAC_BITS=15
endif
ifdef CONFIG_SMALL
$(SUBDIR)%_tablegen$(HOSTEXESUF): HOSTCFLAGS += -DCONFIG_SMALL=1
else
$(SUBDIR)%_tablegen$(HOSTEXESUF): HOSTCFLAGS += -DCONFIG_SMALL=0
endif
$(SUBDIR)%_tablegen$(HOSTEXESUF): $(SUBDIR)%_tablegen.c $(SUBDIR)tableprint.c
$(HOSTCC) $(HOSTCFLAGS) $(HOSTLDFLAGS) -o $@ $^ $(HOSTLIBS)
$(SUBDIR)%_tables.h: $(SUBDIR)%_tablegen$(HOSTEXESUF)
$(M)./$< > $@
ifdef CONFIG_HARDCODED_TABLES
$(SUBDIR)aacdec.o: $(SUBDIR)cbrt_tables.h
$(SUBDIR)aacps.o: $(SUBDIR)aacps_tables.h
$(SUBDIR)aactab.o: $(SUBDIR)aac_tables.h
$(SUBDIR)dv.o: $(SUBDIR)dv_tables.h
$(SUBDIR)mdct.o: $(SUBDIR)mdct_tables.h
$(SUBDIR)mpegaudiodec.o: $(SUBDIR)mpegaudio_tables.h
$(SUBDIR)motionpixels.o: $(SUBDIR)motionpixels_tables.h
$(SUBDIR)pcm.o: $(SUBDIR)pcm_tables.h
$(SUBDIR)qdm2.o: $(SUBDIR)qdm2_tables.h
endif
$(SUBDIR)dct-test$(EXESUF): $(SUBDIR)fdctref.o $(SUBDIR)aandcttab.o
$(SUBDIR)fft-test$(EXESUF): $(SUBDIR)fdctref.o

1686
libavcodec/aac.c Normal file

File diff suppressed because it is too large Load Diff

View File

@@ -21,7 +21,7 @@
*/
/**
* @file
* @file libavcodec/aac.h
* AAC definitions and structures
* @author Oded Shimon ( ods15 ods15 dyndns org )
* @author Maxim Gavrilov ( maxim.gavrilov gmail com )
@@ -30,19 +30,54 @@
#ifndef AVCODEC_AAC_H
#define AVCODEC_AAC_H
#include "libavutil/internal.h"
#include "avcodec.h"
#include "dsputil.h"
#include "fft.h"
#include "mpeg4audio.h"
#include "sbr.h"
#include <stdint.h>
#define AAC_INIT_VLC_STATIC(num, size) \
INIT_VLC_STATIC(&vlc_spectral[num], 6, ff_aac_spectral_sizes[num], \
ff_aac_spectral_bits[num], sizeof( ff_aac_spectral_bits[num][0]), sizeof( ff_aac_spectral_bits[num][0]), \
ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), sizeof(ff_aac_spectral_codes[num][0]), \
size);
#define MAX_CHANNELS 64
#define MAX_ELEM_ID 16
#define TNS_MAX_ORDER 20
enum AudioObjectType {
AOT_NULL,
// Support? Name
AOT_AAC_MAIN, ///< Y Main
AOT_AAC_LC, ///< Y Low Complexity
AOT_AAC_SSR, ///< N (code in SoC repo) Scalable Sample Rate
AOT_AAC_LTP, ///< N (code in SoC repo) Long Term Prediction
AOT_SBR, ///< N (in progress) Spectral Band Replication
AOT_AAC_SCALABLE, ///< N Scalable
AOT_TWINVQ, ///< N Twin Vector Quantizer
AOT_CELP, ///< N Code Excited Linear Prediction
AOT_HVXC, ///< N Harmonic Vector eXcitation Coding
AOT_TTSI = 12, ///< N Text-To-Speech Interface
AOT_MAINSYNTH, ///< N Main Synthesis
AOT_WAVESYNTH, ///< N Wavetable Synthesis
AOT_MIDI, ///< N General MIDI
AOT_SAFX, ///< N Algorithmic Synthesis and Audio Effects
AOT_ER_AAC_LC, ///< N Error Resilient Low Complexity
AOT_ER_AAC_LTP = 19, ///< N Error Resilient Long Term Prediction
AOT_ER_AAC_SCALABLE, ///< N Error Resilient Scalable
AOT_ER_TWINVQ, ///< N Error Resilient Twin Vector Quantizer
AOT_ER_BSAC, ///< N Error Resilient Bit-Sliced Arithmetic Coding
AOT_ER_AAC_LD, ///< N Error Resilient Low Delay
AOT_ER_CELP, ///< N Error Resilient Code Excited Linear Prediction
AOT_ER_HVXC, ///< N Error Resilient Harmonic Vector eXcitation Coding
AOT_ER_HILN, ///< N Error Resilient Harmonic and Individual Lines plus Noise
AOT_ER_PARAM, ///< N Error Resilient Parametric
AOT_SSC, ///< N SinuSoidal Coding
};
enum RawDataBlockType {
TYPE_SCE,
TYPE_CPE,
@@ -98,17 +133,6 @@ enum CouplingPoint {
AFTER_IMDCT = 3,
};
/**
* Output configuration status
*/
enum OCStatus {
OC_NONE, //< Output unconfigured
OC_TRIAL_PCE, //< Output configuration under trial specified by an inband PCE
OC_TRIAL_FRAME, //< Output configuration under trial specified by a frame header
OC_GLOBAL_HDR, //< Output configuration set in a global header but not yet locked
OC_LOCKED, //< Output configuration locked in place
};
/**
* Predictor State
*/
@@ -123,12 +147,6 @@ typedef struct {
#define MAX_PREDICTORS 672
#define SCALE_DIV_512 36 ///< scalefactor difference that corresponds to scale difference in 512 times
#define SCALE_ONE_POS 140 ///< scalefactor index that corresponds to scale=1.0
#define SCALE_MAX_POS 255 ///< scalefactor index maximum value
#define SCALE_MAX_DIFF 60 ///< maximum scalefactor difference allowed by standard
#define SCALE_DIFF_ZERO 60 ///< codebook index corresponding to zero scalefactor indices difference
/**
* Individual Channel Stream
*/
@@ -139,7 +157,6 @@ typedef struct {
int num_window_groups;
uint8_t group_len[8];
const uint16_t *swb_offset; ///< table of offsets to the lowest spectral coefficient of a scalefactor band, sfb, for a particular window
const uint8_t *swb_sizes; ///< table of scalefactor band sizes for a particular window
int num_swb; ///< number of scalefactor window bands
int num_windows;
int tns_max_bands;
@@ -179,7 +196,6 @@ typedef struct {
typedef struct {
int num_pulse;
int start;
int pos[4];
int amp[4];
} Pulse;
@@ -204,15 +220,12 @@ typedef struct {
typedef struct {
IndividualChannelStream ics;
TemporalNoiseShaping tns;
Pulse pulse;
enum BandType band_type[128]; ///< band types
enum BandType band_type[120]; ///< band types
int band_type_run_end[120]; ///< band type run end points
float sf[120]; ///< scalefactors
int sf_idx[128]; ///< scalefactor indices (used by encoder)
uint8_t zeroes[128]; ///< band is not coded (used by encoder)
DECLARE_ALIGNED(16, float, coeffs)[1024]; ///< coefficients for IMDCT
DECLARE_ALIGNED(16, float, saved)[1024]; ///< overlap
DECLARE_ALIGNED(16, float, ret)[2048]; ///< PCM output
DECLARE_ALIGNED_16(float, coeffs[1024]); ///< coefficients for IMDCT
DECLARE_ALIGNED_16(float, saved[512]); ///< overlap
DECLARE_ALIGNED_16(float, ret[1024]); ///< PCM output
PredictorState predictor_state[MAX_PREDICTORS];
} SingleChannelElement;
@@ -221,21 +234,18 @@ typedef struct {
*/
typedef struct {
// CPE specific
int common_window; ///< Set if channels share a common 'IndividualChannelStream' in bitstream.
int ms_mode; ///< Signals mid/side stereo flags coding mode (used by encoder)
uint8_t ms_mask[128]; ///< Set if mid/side stereo is used for each scalefactor window band
uint8_t ms_mask[120]; ///< Set if mid/side stereo is used for each scalefactor window band
// shared
SingleChannelElement ch[2];
// CCE specific
ChannelCoupling coup;
SpectralBandReplication sbr;
} ChannelElement;
/**
* main AAC context
*/
typedef struct {
AVCodecContext *avctx;
AVCodecContext * avccontext;
MPEG4AudioConfig m4ac;
@@ -249,25 +259,22 @@ typedef struct {
enum ChannelPosition che_pos[4][MAX_ELEM_ID]; /**< channel element channel mapping with the
* first index as the first 4 raw data block types
*/
ChannelElement *che[4][MAX_ELEM_ID];
ChannelElement *tag_che_map[4][MAX_ELEM_ID];
uint8_t tags_seen_this_frame[4][MAX_ELEM_ID];
int tags_mapped;
ChannelElement * che[4][MAX_ELEM_ID];
/** @} */
/**
* @defgroup temporary aligned temporary buffers (We do not want to have these on the stack.)
* @{
*/
DECLARE_ALIGNED(16, float, buf_mdct)[1024];
DECLARE_ALIGNED_16(float, buf_mdct[1024]);
/** @} */
/**
* @defgroup tables Computed / set up during initialization.
* @{
*/
FFTContext mdct;
FFTContext mdct_small;
MDCTContext mdct;
MDCTContext mdct_small;
DSPContext dsp;
int random_state;
/** @} */
@@ -282,9 +289,7 @@ typedef struct {
int sf_offset; ///< offset into pow2sf_tab as appropriate for dsp.float_to_int16
/** @} */
DECLARE_ALIGNED(16, float, temp)[128];
enum OCStatus output_configured;
DECLARE_ALIGNED(16, float, temp[128]);
} AACContext;
#endif /* AVCODEC_AAC_H */

View File

@@ -71,32 +71,23 @@ get_next:
*poutbuf_size = buf_size;
/* update codec info */
avctx->sample_rate = s->sample_rate;
if(s->codec_id)
avctx->codec_id = s->codec_id;
/* Due to backwards compatible HE-AAC the sample rate, channel count,
and total number of samples found in an AAC ADTS header are not
reliable. Bit rate is still accurate because the total frame duration in
seconds is still correct (as is the number of bits in the frame). */
if (avctx->codec_id != CODEC_ID_AAC) {
avctx->sample_rate = s->sample_rate;
/* allow downmixing to stereo (or mono for AC-3) */
if(avctx->request_channels > 0 &&
avctx->request_channels < s->channels &&
(avctx->request_channels <= 2 ||
(avctx->request_channels == 1 &&
(avctx->codec_id == CODEC_ID_AC3 ||
avctx->codec_id == CODEC_ID_EAC3)))) {
avctx->channels = avctx->request_channels;
} else {
avctx->channels = s->channels;
avctx->channel_layout = s->channel_layout;
}
avctx->frame_size = s->samples;
/* allow downmixing to stereo (or mono for AC-3) */
if(avctx->request_channels > 0 &&
avctx->request_channels < s->channels &&
(avctx->request_channels <= 2 ||
(avctx->request_channels == 1 &&
(avctx->codec_id == CODEC_ID_AC3 ||
avctx->codec_id == CODEC_ID_EAC3)))) {
avctx->channels = avctx->request_channels;
} else {
avctx->channels = s->channels;
}
avctx->bit_rate = s->bit_rate;
avctx->frame_size = s->samples;
return i;
}

View File

@@ -48,7 +48,6 @@ typedef struct AACAC3ParseContext {
int sample_rate;
int bit_rate;
int samples;
int64_t channel_layout;
int remaining_size;
uint64_t state;

View File

@@ -1,114 +0,0 @@
/*
* MPEG-2/4 AAC ADTS to MPEG-4 Audio Specific Configuration bitstream filter
* Copyright (c) 2009 Alex Converse <alex.converse@gmail.com>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "avcodec.h"
#include "aac_parser.h"
#include "put_bits.h"
#include "get_bits.h"
#include "mpeg4audio.h"
#include "internal.h"
typedef struct AACBSFContext {
int first_frame_done;
} AACBSFContext;
/**
* This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4
* ADTS header and removes the ADTS header.
*/
static int aac_adtstoasc_filter(AVBitStreamFilterContext *bsfc,
AVCodecContext *avctx, const char *args,
uint8_t **poutbuf, int *poutbuf_size,
const uint8_t *buf, int buf_size,
int keyframe)
{
GetBitContext gb;
PutBitContext pb;
AACADTSHeaderInfo hdr;
AACBSFContext *ctx = bsfc->priv_data;
init_get_bits(&gb, buf, AAC_ADTS_HEADER_SIZE*8);
*poutbuf = (uint8_t*) buf;
*poutbuf_size = buf_size;
if (avctx->extradata)
if (show_bits(&gb, 12) != 0xfff)
return 0;
if (ff_aac_parse_header(&gb, &hdr) < 0) {
av_log(avctx, AV_LOG_ERROR, "Error parsing ADTS frame header!\n");
return -1;
}
if (!hdr.crc_absent && hdr.num_aac_frames > 1) {
av_log_missing_feature(avctx, "Multiple RDBs per frame with CRC is", 0);
return -1;
}
buf += AAC_ADTS_HEADER_SIZE + 2*!hdr.crc_absent;
buf_size -= AAC_ADTS_HEADER_SIZE + 2*!hdr.crc_absent;
if (!ctx->first_frame_done) {
int pce_size = 0;
uint8_t pce_data[MAX_PCE_SIZE];
if (!hdr.chan_config) {
init_get_bits(&gb, buf, buf_size);
if (get_bits(&gb, 3) != 5) {
av_log_missing_feature(avctx, "PCE based channel configuration, where the PCE is not the first syntax element is", 0);
return -1;
}
init_put_bits(&pb, pce_data, MAX_PCE_SIZE);
pce_size = ff_copy_pce_data(&pb, &gb)/8;
flush_put_bits(&pb);
buf_size -= get_bits_count(&gb)/8;
buf += get_bits_count(&gb)/8;
}
avctx->extradata_size = 2 + pce_size;
avctx->extradata = av_mallocz(avctx->extradata_size + FF_INPUT_BUFFER_PADDING_SIZE);
init_put_bits(&pb, avctx->extradata, avctx->extradata_size);
put_bits(&pb, 5, hdr.object_type);
put_bits(&pb, 4, hdr.sampling_index);
put_bits(&pb, 4, hdr.chan_config);
put_bits(&pb, 1, 0); //frame length - 1024 samples
put_bits(&pb, 1, 0); //does not depend on core coder
put_bits(&pb, 1, 0); //is not extension
flush_put_bits(&pb);
if (pce_size) {
memcpy(avctx->extradata + 2, pce_data, pce_size);
}
ctx->first_frame_done = 1;
}
*poutbuf = (uint8_t*) buf;
*poutbuf_size = buf_size;
return 0;
}
AVBitStreamFilter aac_adtstoasc_bsf = {
"aac_adtstoasc",
sizeof(AACBSFContext),
aac_adtstoasc_filter,
};

View File

@@ -23,9 +23,11 @@
#include "parser.h"
#include "aac_ac3_parser.h"
#include "aac_parser.h"
#include "get_bits.h"
#include "bitstream.h"
#include "mpeg4audio.h"
#define AAC_HEADER_SIZE 7
int ff_aac_parse_header(GetBitContext *gbc, AACADTSHeaderInfo *hdr)
{
int size, rdb, ch, sr;
@@ -44,6 +46,9 @@ int ff_aac_parse_header(GetBitContext *gbc, AACADTSHeaderInfo *hdr)
skip_bits1(gbc); /* private_bit */
ch = get_bits(gbc, 3); /* channel_configuration */
if(!ff_mpeg4audio_channels[ch])
return AAC_AC3_PARSE_ERROR_CHANNEL_CFG;
skip_bits1(gbc); /* original/copy */
skip_bits1(gbc); /* home */
@@ -51,7 +56,7 @@ int ff_aac_parse_header(GetBitContext *gbc, AACADTSHeaderInfo *hdr)
skip_bits1(gbc); /* copyright_identification_bit */
skip_bits1(gbc); /* copyright_identification_start */
size = get_bits(gbc, 13); /* aac_frame_length */
if(size < AAC_ADTS_HEADER_SIZE)
if(size < AAC_HEADER_SIZE)
return AAC_AC3_PARSE_ERROR_FRAME_SIZE;
skip_bits(gbc, 11); /* adts_buffer_fullness */
@@ -81,7 +86,7 @@ static int aac_sync(uint64_t state, AACAC3ParseContext *hdr_info,
} tmp;
tmp.u64 = be2me_64(state);
init_get_bits(&bits, tmp.u8+8-AAC_ADTS_HEADER_SIZE, AAC_ADTS_HEADER_SIZE * 8);
init_get_bits(&bits, tmp.u8+8-AAC_HEADER_SIZE, AAC_HEADER_SIZE * 8);
if ((size = ff_aac_parse_header(&bits, &hdr)) < 0)
return 0;
@@ -97,7 +102,7 @@ static int aac_sync(uint64_t state, AACAC3ParseContext *hdr_info,
static av_cold int aac_parse_init(AVCodecParserContext *s1)
{
AACAC3ParseContext *s = s1->priv_data;
s->header_size = AAC_ADTS_HEADER_SIZE;
s->header_size = AAC_HEADER_SIZE;
s->sync = aac_sync;
return 0;
}

View File

@@ -25,9 +25,7 @@
#include <stdint.h>
#include "aac_ac3_parser.h"
#include "get_bits.h"
#define AAC_ADTS_HEADER_SIZE 7
#include "bitstream.h"
typedef struct {
uint32_t sample_rate;

View File

@@ -1,39 +0,0 @@
/*
* Generate a header file for hardcoded AAC tables
*
* Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <stdlib.h>
#define CONFIG_HARDCODED_TABLES 0
#include "aac_tablegen.h"
#include "tableprint.h"
int main(void)
{
ff_aac_tableinit();
write_fileheader();
printf("const float ff_aac_pow2sf_tab[428] = {\n");
write_float_array(ff_aac_pow2sf_tab, 428);
printf("};\n");
return 0;
}

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@@ -1,42 +0,0 @@
/*
* Header file for hardcoded AAC tables
*
* Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AAC_TABLEGEN_H
#define AAC_TABLEGEN_H
#include "aac_tablegen_decl.h"
#if CONFIG_HARDCODED_TABLES
#include "libavcodec/aac_tables.h"
#else
#include "../libavutil/mathematics.h"
float ff_aac_pow2sf_tab[428];
void ff_aac_tableinit(void)
{
int i;
for (i = 0; i < 428; i++)
ff_aac_pow2sf_tab[i] = pow(2, (i - 200) / 4.);
}
#endif /* CONFIG_HARDCODED_TABLES */
#endif /* AAC_TABLEGEN_H */

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@@ -1,34 +0,0 @@
/*
* Header file for hardcoded AAC tables
*
* Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AAC_TABLEGEN_INIT_H
#define AAC_TABLEGEN_INIT_H
#if CONFIG_HARDCODED_TABLES
#define ff_aac_tableinit()
extern const float ff_aac_pow2sf_tab[428];
#else
void ff_aac_tableinit(void);
extern float ff_aac_pow2sf_tab[428];
#endif /* CONFIG_HARDCODED_TABLES */
#endif /* AAC_TABLEGEN_INIT_H */

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@@ -1,961 +0,0 @@
/*
* AAC coefficients encoder
* Copyright (C) 2008-2009 Konstantin Shishkov
*
* 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
* AAC coefficients encoder
*/
/***********************************
* TODOs:
* speedup quantizer selection
* add sane pulse detection
***********************************/
#include "avcodec.h"
#include "put_bits.h"
#include "aac.h"
#include "aacenc.h"
#include "aactab.h"
#include "libavutil/libm.h"
/** bits needed to code codebook run value for long windows */
static const uint8_t run_value_bits_long[64] = {
5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 10,
10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10,
10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 15
};
/** bits needed to code codebook run value for short windows */
static const uint8_t run_value_bits_short[16] = {
3, 3, 3, 3, 3, 3, 3, 6, 6, 6, 6, 6, 6, 6, 6, 9
};
static const uint8_t *run_value_bits[2] = {
run_value_bits_long, run_value_bits_short
};
/**
* Quantize one coefficient.
* @return absolute value of the quantized coefficient
* @see 3GPP TS26.403 5.6.2 "Scalefactor determination"
*/
static av_always_inline int quant(float coef, const float Q)
{
float a = coef * Q;
return sqrtf(a * sqrtf(a)) + 0.4054;
}
static void quantize_bands(int (*out)[2], const float *in, const float *scaled,
int size, float Q34, int is_signed, int maxval)
{
int i;
double qc;
for (i = 0; i < size; i++) {
qc = scaled[i] * Q34;
out[i][0] = (int)FFMIN(qc, (double)maxval);
out[i][1] = (int)FFMIN(qc + 0.4054, (double)maxval);
if (is_signed && in[i] < 0.0f) {
out[i][0] = -out[i][0];
out[i][1] = -out[i][1];
}
}
}
static void abs_pow34_v(float *out, const float *in, const int size)
{
#ifndef USE_REALLY_FULL_SEARCH
int i;
for (i = 0; i < size; i++) {
float a = fabsf(in[i]);
out[i] = sqrtf(a * sqrtf(a));
}
#endif /* USE_REALLY_FULL_SEARCH */
}
static const uint8_t aac_cb_range [12] = {0, 3, 3, 3, 3, 9, 9, 8, 8, 13, 13, 17};
static const uint8_t aac_cb_maxval[12] = {0, 1, 1, 2, 2, 4, 4, 7, 7, 12, 12, 16};
/**
* Calculate rate distortion cost for quantizing with given codebook
*
* @return quantization distortion
*/
static float quantize_and_encode_band_cost(struct AACEncContext *s,
PutBitContext *pb, const float *in,
const float *scaled, int size, int scale_idx,
int cb, const float lambda, const float uplim,
int *bits)
{
const float IQ = ff_aac_pow2sf_tab[200 + scale_idx - SCALE_ONE_POS + SCALE_DIV_512];
const float Q = ff_aac_pow2sf_tab[200 - scale_idx + SCALE_ONE_POS - SCALE_DIV_512];
const float CLIPPED_ESCAPE = 165140.0f*IQ;
int i, j, k;
float cost = 0;
const int dim = cb < FIRST_PAIR_BT ? 4 : 2;
int resbits = 0;
#ifndef USE_REALLY_FULL_SEARCH
const float Q34 = sqrtf(Q * sqrtf(Q));
const int range = aac_cb_range[cb];
const int maxval = aac_cb_maxval[cb];
int offs[4];
#endif /* USE_REALLY_FULL_SEARCH */
if (!cb) {
for (i = 0; i < size; i++)
cost += in[i]*in[i];
if (bits)
*bits = 0;
return cost * lambda;
}
#ifndef USE_REALLY_FULL_SEARCH
offs[0] = 1;
for (i = 1; i < dim; i++)
offs[i] = offs[i-1]*range;
if (!scaled) {
abs_pow34_v(s->scoefs, in, size);
scaled = s->scoefs;
}
quantize_bands(s->qcoefs, in, scaled, size, Q34, !IS_CODEBOOK_UNSIGNED(cb), maxval);
#endif /* USE_REALLY_FULL_SEARCH */
for (i = 0; i < size; i += dim) {
float mincost;
int minidx = 0;
int minbits = 0;
const float *vec;
#ifndef USE_REALLY_FULL_SEARCH
int (*quants)[2] = &s->qcoefs[i];
mincost = 0.0f;
for (j = 0; j < dim; j++)
mincost += in[i+j]*in[i+j];
minidx = IS_CODEBOOK_UNSIGNED(cb) ? 0 : 40;
minbits = ff_aac_spectral_bits[cb-1][minidx];
mincost = mincost * lambda + minbits;
for (j = 0; j < (1<<dim); j++) {
float rd = 0.0f;
int curbits;
int curidx = IS_CODEBOOK_UNSIGNED(cb) ? 0 : 40;
int same = 0;
for (k = 0; k < dim; k++) {
if ((j & (1 << k)) && quants[k][0] == quants[k][1]) {
same = 1;
break;
}
}
if (same)
continue;
for (k = 0; k < dim; k++)
curidx += quants[k][!!(j & (1 << k))] * offs[dim - 1 - k];
curbits = ff_aac_spectral_bits[cb-1][curidx];
vec = &ff_aac_codebook_vectors[cb-1][curidx*dim];
#else
mincost = INFINITY;
vec = ff_aac_codebook_vectors[cb-1];
for (j = 0; j < ff_aac_spectral_sizes[cb-1]; j++, vec += dim) {
float rd = 0.0f;
int curbits = ff_aac_spectral_bits[cb-1][j];
int curidx = j;
#endif /* USE_REALLY_FULL_SEARCH */
if (IS_CODEBOOK_UNSIGNED(cb)) {
for (k = 0; k < dim; k++) {
float t = fabsf(in[i+k]);
float di;
if (vec[k] == 64.0f) { //FIXME: slow
//do not code with escape sequence small values
if (t < 39.0f*IQ) {
rd = INFINITY;
break;
}
if (t >= CLIPPED_ESCAPE) {
di = t - CLIPPED_ESCAPE;
curbits += 21;
} else {
int c = av_clip(quant(t, Q), 0, 8191);
di = t - c*cbrtf(c)*IQ;
curbits += av_log2(c)*2 - 4 + 1;
}
} else {
di = t - vec[k]*IQ;
}
if (vec[k] != 0.0f)
curbits++;
rd += di*di;
}
} else {
for (k = 0; k < dim; k++) {
float di = in[i+k] - vec[k]*IQ;
rd += di*di;
}
}
rd = rd * lambda + curbits;
if (rd < mincost) {
mincost = rd;
minidx = curidx;
minbits = curbits;
}
}
cost += mincost;
resbits += minbits;
if (cost >= uplim)
return uplim;
if (pb) {
put_bits(pb, ff_aac_spectral_bits[cb-1][minidx], ff_aac_spectral_codes[cb-1][minidx]);
if (IS_CODEBOOK_UNSIGNED(cb))
for (j = 0; j < dim; j++)
if (ff_aac_codebook_vectors[cb-1][minidx*dim+j] != 0.0f)
put_bits(pb, 1, in[i+j] < 0.0f);
if (cb == ESC_BT) {
for (j = 0; j < 2; j++) {
if (ff_aac_codebook_vectors[cb-1][minidx*2+j] == 64.0f) {
int coef = av_clip(quant(fabsf(in[i+j]), Q), 0, 8191);
int len = av_log2(coef);
put_bits(pb, len - 4 + 1, (1 << (len - 4 + 1)) - 2);
put_bits(pb, len, coef & ((1 << len) - 1));
}
}
}
}
}
if (bits)
*bits = resbits;
return cost;
}
static float quantize_band_cost(struct AACEncContext *s, const float *in,
const float *scaled, int size, int scale_idx,
int cb, const float lambda, const float uplim,
int *bits)
{
return quantize_and_encode_band_cost(s, NULL, in, scaled, size, scale_idx,
cb, lambda, uplim, bits);
}
static void quantize_and_encode_band(struct AACEncContext *s, PutBitContext *pb,
const float *in, int size, int scale_idx,
int cb, const float lambda)
{
quantize_and_encode_band_cost(s, pb, in, NULL, size, scale_idx, cb, lambda,
INFINITY, NULL);
}
/**
* structure used in optimal codebook search
*/
typedef struct BandCodingPath {
int prev_idx; ///< pointer to the previous path point
float cost; ///< path cost
int run;
} BandCodingPath;
/**
* Encode band info for single window group bands.
*/
static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce,
int win, int group_len, const float lambda)
{
BandCodingPath path[120][12];
int w, swb, cb, start, start2, size;
int i, j;
const int max_sfb = sce->ics.max_sfb;
const int run_bits = sce->ics.num_windows == 1 ? 5 : 3;
const int run_esc = (1 << run_bits) - 1;
int idx, ppos, count;
int stackrun[120], stackcb[120], stack_len;
float next_minrd = INFINITY;
int next_mincb = 0;
abs_pow34_v(s->scoefs, sce->coeffs, 1024);
start = win*128;
for (cb = 0; cb < 12; cb++) {
path[0][cb].cost = 0.0f;
path[0][cb].prev_idx = -1;
path[0][cb].run = 0;
}
for (swb = 0; swb < max_sfb; swb++) {
start2 = start;
size = sce->ics.swb_sizes[swb];
if (sce->zeroes[win*16 + swb]) {
for (cb = 0; cb < 12; cb++) {
path[swb+1][cb].prev_idx = cb;
path[swb+1][cb].cost = path[swb][cb].cost;
path[swb+1][cb].run = path[swb][cb].run + 1;
}
} else {
float minrd = next_minrd;
int mincb = next_mincb;
next_minrd = INFINITY;
next_mincb = 0;
for (cb = 0; cb < 12; cb++) {
float cost_stay_here, cost_get_here;
float rd = 0.0f;
for (w = 0; w < group_len; w++) {
FFPsyBand *band = &s->psy.psy_bands[s->cur_channel*PSY_MAX_BANDS+(win+w)*16+swb];
rd += quantize_band_cost(s, sce->coeffs + start + w*128,
s->scoefs + start + w*128, size,
sce->sf_idx[(win+w)*16+swb], cb,
lambda / band->threshold, INFINITY, NULL);
}
cost_stay_here = path[swb][cb].cost + rd;
cost_get_here = minrd + rd + run_bits + 4;
if ( run_value_bits[sce->ics.num_windows == 8][path[swb][cb].run]
!= run_value_bits[sce->ics.num_windows == 8][path[swb][cb].run+1])
cost_stay_here += run_bits;
if (cost_get_here < cost_stay_here) {
path[swb+1][cb].prev_idx = mincb;
path[swb+1][cb].cost = cost_get_here;
path[swb+1][cb].run = 1;
} else {
path[swb+1][cb].prev_idx = cb;
path[swb+1][cb].cost = cost_stay_here;
path[swb+1][cb].run = path[swb][cb].run + 1;
}
if (path[swb+1][cb].cost < next_minrd) {
next_minrd = path[swb+1][cb].cost;
next_mincb = cb;
}
}
}
start += sce->ics.swb_sizes[swb];
}
//convert resulting path from backward-linked list
stack_len = 0;
idx = 0;
for (cb = 1; cb < 12; cb++)
if (path[max_sfb][cb].cost < path[max_sfb][idx].cost)
idx = cb;
ppos = max_sfb;
while (ppos > 0) {
cb = idx;
stackrun[stack_len] = path[ppos][cb].run;
stackcb [stack_len] = cb;
idx = path[ppos-path[ppos][cb].run+1][cb].prev_idx;
ppos -= path[ppos][cb].run;
stack_len++;
}
//perform actual band info encoding
start = 0;
for (i = stack_len - 1; i >= 0; i--) {
put_bits(&s->pb, 4, stackcb[i]);
count = stackrun[i];
memset(sce->zeroes + win*16 + start, !stackcb[i], count);
//XXX: memset when band_type is also uint8_t
for (j = 0; j < count; j++) {
sce->band_type[win*16 + start] = stackcb[i];
start++;
}
while (count >= run_esc) {
put_bits(&s->pb, run_bits, run_esc);
count -= run_esc;
}
put_bits(&s->pb, run_bits, count);
}
}
typedef struct TrellisPath {
float cost;
int prev;
int min_val;
int max_val;
} TrellisPath;
#define TRELLIS_STAGES 121
#define TRELLIS_STATES 256
static void search_for_quantizers_anmr(AVCodecContext *avctx, AACEncContext *s,
SingleChannelElement *sce,
const float lambda)
{
int q, w, w2, g, start = 0;
int i, j;
int idx;
TrellisPath paths[TRELLIS_STAGES][TRELLIS_STATES];
int bandaddr[TRELLIS_STAGES];
int minq;
float mincost;
for (i = 0; i < TRELLIS_STATES; i++) {
paths[0][i].cost = 0.0f;
paths[0][i].prev = -1;
paths[0][i].min_val = i;
paths[0][i].max_val = i;
}
for (j = 1; j < TRELLIS_STAGES; j++) {
for (i = 0; i < TRELLIS_STATES; i++) {
paths[j][i].cost = INFINITY;
paths[j][i].prev = -2;
paths[j][i].min_val = INT_MAX;
paths[j][i].max_val = 0;
}
}
idx = 1;
abs_pow34_v(s->scoefs, sce->coeffs, 1024);
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
start = w*128;
for (g = 0; g < sce->ics.num_swb; g++) {
const float *coefs = sce->coeffs + start;
float qmin, qmax;
int nz = 0;
bandaddr[idx] = w * 16 + g;
qmin = INT_MAX;
qmax = 0.0f;
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
FFPsyBand *band = &s->psy.psy_bands[s->cur_channel*PSY_MAX_BANDS+(w+w2)*16+g];
if (band->energy <= band->threshold || band->threshold == 0.0f) {
sce->zeroes[(w+w2)*16+g] = 1;
continue;
}
sce->zeroes[(w+w2)*16+g] = 0;
nz = 1;
for (i = 0; i < sce->ics.swb_sizes[g]; i++) {
float t = fabsf(coefs[w2*128+i]);
if (t > 0.0f)
qmin = FFMIN(qmin, t);
qmax = FFMAX(qmax, t);
}
}
if (nz) {
int minscale, maxscale;
float minrd = INFINITY;
//minimum scalefactor index is when minimum nonzero coefficient after quantizing is not clipped
minscale = av_clip_uint8(log2(qmin)*4 - 69 + SCALE_ONE_POS - SCALE_DIV_512);
//maximum scalefactor index is when maximum coefficient after quantizing is still not zero
maxscale = av_clip_uint8(log2(qmax)*4 + 6 + SCALE_ONE_POS - SCALE_DIV_512);
for (q = minscale; q < maxscale; q++) {
float dists[12], dist;
memset(dists, 0, sizeof(dists));
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
FFPsyBand *band = &s->psy.psy_bands[s->cur_channel*PSY_MAX_BANDS+(w+w2)*16+g];
int cb;
for (cb = 0; cb <= ESC_BT; cb++)
dists[cb] += quantize_band_cost(s, coefs + w2*128, s->scoefs + start + w2*128, sce->ics.swb_sizes[g],
q, cb, lambda / band->threshold, INFINITY, NULL);
}
dist = dists[0];
for (i = 1; i <= ESC_BT; i++)
dist = FFMIN(dist, dists[i]);
minrd = FFMIN(minrd, dist);
for (i = FFMAX(q - SCALE_MAX_DIFF, 0); i < FFMIN(q + SCALE_MAX_DIFF, TRELLIS_STATES); i++) {
float cost;
int minv, maxv;
if (isinf(paths[idx - 1][i].cost))
continue;
cost = paths[idx - 1][i].cost + dist
+ ff_aac_scalefactor_bits[q - i + SCALE_DIFF_ZERO];
minv = FFMIN(paths[idx - 1][i].min_val, q);
maxv = FFMAX(paths[idx - 1][i].max_val, q);
if (cost < paths[idx][q].cost && maxv-minv < SCALE_MAX_DIFF) {
paths[idx][q].cost = cost;
paths[idx][q].prev = i;
paths[idx][q].min_val = minv;
paths[idx][q].max_val = maxv;
}
}
}
} else {
for (q = 0; q < TRELLIS_STATES; q++) {
if (!isinf(paths[idx - 1][q].cost)) {
paths[idx][q].cost = paths[idx - 1][q].cost + 1;
paths[idx][q].prev = q;
paths[idx][q].min_val = FFMIN(paths[idx - 1][q].min_val, q);
paths[idx][q].max_val = FFMAX(paths[idx - 1][q].max_val, q);
continue;
}
for (i = FFMAX(q - SCALE_MAX_DIFF, 0); i < FFMIN(q + SCALE_MAX_DIFF, TRELLIS_STATES); i++) {
float cost;
int minv, maxv;
if (isinf(paths[idx - 1][i].cost))
continue;
cost = paths[idx - 1][i].cost + ff_aac_scalefactor_bits[q - i + SCALE_DIFF_ZERO];
minv = FFMIN(paths[idx - 1][i].min_val, q);
maxv = FFMAX(paths[idx - 1][i].max_val, q);
if (cost < paths[idx][q].cost && maxv-minv < SCALE_MAX_DIFF) {
paths[idx][q].cost = cost;
paths[idx][q].prev = i;
paths[idx][q].min_val = minv;
paths[idx][q].max_val = maxv;
}
}
}
}
sce->zeroes[w*16+g] = !nz;
start += sce->ics.swb_sizes[g];
idx++;
}
}
idx--;
mincost = paths[idx][0].cost;
minq = 0;
for (i = 1; i < TRELLIS_STATES; i++) {
if (paths[idx][i].cost < mincost) {
mincost = paths[idx][i].cost;
minq = i;
}
}
while (idx) {
sce->sf_idx[bandaddr[idx]] = minq;
minq = paths[idx][minq].prev;
idx--;
}
//set the same quantizers inside window groups
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
for (g = 0; g < sce->ics.num_swb; g++)
for (w2 = 1; w2 < sce->ics.group_len[w]; w2++)
sce->sf_idx[(w+w2)*16+g] = sce->sf_idx[w*16+g];
}
/**
* two-loop quantizers search taken from ISO 13818-7 Appendix C
*/
static void search_for_quantizers_twoloop(AVCodecContext *avctx,
AACEncContext *s,
SingleChannelElement *sce,
const float lambda)
{
int start = 0, i, w, w2, g;
int destbits = avctx->bit_rate * 1024.0 / avctx->sample_rate / avctx->channels;
float dists[128], uplims[128];
int fflag, minscaler;
int its = 0;
int allz = 0;
float minthr = INFINITY;
//XXX: some heuristic to determine initial quantizers will reduce search time
memset(dists, 0, sizeof(dists));
//determine zero bands and upper limits
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
for (g = 0; g < sce->ics.num_swb; g++) {
int nz = 0;
float uplim = 0.0f;
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
FFPsyBand *band = &s->psy.psy_bands[s->cur_channel*PSY_MAX_BANDS+(w+w2)*16+g];
uplim += band->threshold;
if (band->energy <= band->threshold || band->threshold == 0.0f) {
sce->zeroes[(w+w2)*16+g] = 1;
continue;
}
nz = 1;
}
uplims[w*16+g] = uplim *512;
sce->zeroes[w*16+g] = !nz;
if (nz)
minthr = FFMIN(minthr, uplim);
allz = FFMAX(allz, nz);
}
}
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
for (g = 0; g < sce->ics.num_swb; g++) {
if (sce->zeroes[w*16+g]) {
sce->sf_idx[w*16+g] = SCALE_ONE_POS;
continue;
}
sce->sf_idx[w*16+g] = SCALE_ONE_POS + FFMIN(log2(uplims[w*16+g]/minthr)*4,59);
}
}
if (!allz)
return;
abs_pow34_v(s->scoefs, sce->coeffs, 1024);
//perform two-loop search
//outer loop - improve quality
do {
int tbits, qstep;
minscaler = sce->sf_idx[0];
//inner loop - quantize spectrum to fit into given number of bits
qstep = its ? 1 : 32;
do {
int prev = -1;
tbits = 0;
fflag = 0;
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
start = w*128;
for (g = 0; g < sce->ics.num_swb; g++) {
const float *coefs = sce->coeffs + start;
const float *scaled = s->scoefs + start;
int bits = 0;
int cb;
float mindist = INFINITY;
int minbits = 0;
if (sce->zeroes[w*16+g] || sce->sf_idx[w*16+g] >= 218) {
start += sce->ics.swb_sizes[g];
continue;
}
minscaler = FFMIN(minscaler, sce->sf_idx[w*16+g]);
for (cb = 0; cb <= ESC_BT; cb++) {
float dist = 0.0f;
int bb = 0;
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
int b;
dist += quantize_band_cost(s, coefs + w2*128,
scaled + w2*128,
sce->ics.swb_sizes[g],
sce->sf_idx[w*16+g],
cb,
lambda,
INFINITY,
&b);
bb += b;
}
if (dist < mindist) {
mindist = dist;
minbits = bb;
}
}
dists[w*16+g] = (mindist - minbits) / lambda;
bits = minbits;
if (prev != -1) {
bits += ff_aac_scalefactor_bits[sce->sf_idx[w*16+g] - prev + SCALE_DIFF_ZERO];
}
tbits += bits;
start += sce->ics.swb_sizes[g];
prev = sce->sf_idx[w*16+g];
}
}
if (tbits > destbits) {
for (i = 0; i < 128; i++)
if (sce->sf_idx[i] < 218 - qstep)
sce->sf_idx[i] += qstep;
} else {
for (i = 0; i < 128; i++)
if (sce->sf_idx[i] > 60 - qstep)
sce->sf_idx[i] -= qstep;
}
qstep >>= 1;
if (!qstep && tbits > destbits*1.02)
qstep = 1;
if (sce->sf_idx[0] >= 217)
break;
} while (qstep);
fflag = 0;
minscaler = av_clip(minscaler, 60, 255 - SCALE_MAX_DIFF);
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
start = w*128;
for (g = 0; g < sce->ics.num_swb; g++) {
int prevsc = sce->sf_idx[w*16+g];
if (dists[w*16+g] > uplims[w*16+g] && sce->sf_idx[w*16+g] > 60)
sce->sf_idx[w*16+g]--;
sce->sf_idx[w*16+g] = av_clip(sce->sf_idx[w*16+g], minscaler, minscaler + SCALE_MAX_DIFF);
sce->sf_idx[w*16+g] = FFMIN(sce->sf_idx[w*16+g], 219);
if (sce->sf_idx[w*16+g] != prevsc)
fflag = 1;
}
}
its++;
} while (fflag && its < 10);
}
static void search_for_quantizers_faac(AVCodecContext *avctx, AACEncContext *s,
SingleChannelElement *sce,
const float lambda)
{
int start = 0, i, w, w2, g;
float uplim[128], maxq[128];
int minq, maxsf;
float distfact = ((sce->ics.num_windows > 1) ? 85.80 : 147.84) / lambda;
int last = 0, lastband = 0, curband = 0;
float avg_energy = 0.0;
if (sce->ics.num_windows == 1) {
start = 0;
for (i = 0; i < 1024; i++) {
if (i - start >= sce->ics.swb_sizes[curband]) {
start += sce->ics.swb_sizes[curband];
curband++;
}
if (sce->coeffs[i]) {
avg_energy += sce->coeffs[i] * sce->coeffs[i];
last = i;
lastband = curband;
}
}
} else {
for (w = 0; w < 8; w++) {
const float *coeffs = sce->coeffs + w*128;
start = 0;
for (i = 0; i < 128; i++) {
if (i - start >= sce->ics.swb_sizes[curband]) {
start += sce->ics.swb_sizes[curband];
curband++;
}
if (coeffs[i]) {
avg_energy += coeffs[i] * coeffs[i];
last = FFMAX(last, i);
lastband = FFMAX(lastband, curband);
}
}
}
}
last++;
avg_energy /= last;
if (avg_energy == 0.0f) {
for (i = 0; i < FF_ARRAY_ELEMS(sce->sf_idx); i++)
sce->sf_idx[i] = SCALE_ONE_POS;
return;
}
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
start = w*128;
for (g = 0; g < sce->ics.num_swb; g++) {
float *coefs = sce->coeffs + start;
const int size = sce->ics.swb_sizes[g];
int start2 = start, end2 = start + size, peakpos = start;
float maxval = -1, thr = 0.0f, t;
maxq[w*16+g] = 0.0f;
if (g > lastband) {
maxq[w*16+g] = 0.0f;
start += size;
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++)
memset(coefs + w2*128, 0, sizeof(coefs[0])*size);
continue;
}
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
for (i = 0; i < size; i++) {
float t = coefs[w2*128+i]*coefs[w2*128+i];
maxq[w*16+g] = FFMAX(maxq[w*16+g], fabsf(coefs[w2*128 + i]));
thr += t;
if (sce->ics.num_windows == 1 && maxval < t) {
maxval = t;
peakpos = start+i;
}
}
}
if (sce->ics.num_windows == 1) {
start2 = FFMAX(peakpos - 2, start2);
end2 = FFMIN(peakpos + 3, end2);
} else {
start2 -= start;
end2 -= start;
}
start += size;
thr = pow(thr / (avg_energy * (end2 - start2)), 0.3 + 0.1*(lastband - g) / lastband);
t = 1.0 - (1.0 * start2 / last);
uplim[w*16+g] = distfact / (1.4 * thr + t*t*t + 0.075);
}
}
memset(sce->sf_idx, 0, sizeof(sce->sf_idx));
abs_pow34_v(s->scoefs, sce->coeffs, 1024);
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
start = w*128;
for (g = 0; g < sce->ics.num_swb; g++) {
const float *coefs = sce->coeffs + start;
const float *scaled = s->scoefs + start;
const int size = sce->ics.swb_sizes[g];
int scf, prev_scf, step;
int min_scf = -1, max_scf = 256;
float curdiff;
if (maxq[w*16+g] < 21.544) {
sce->zeroes[w*16+g] = 1;
start += size;
continue;
}
sce->zeroes[w*16+g] = 0;
scf = prev_scf = av_clip(SCALE_ONE_POS - SCALE_DIV_512 - log2(1/maxq[w*16+g])*16/3, 60, 218);
step = 16;
for (;;) {
float dist = 0.0f;
int quant_max;
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
int b;
dist += quantize_band_cost(s, coefs + w2*128,
scaled + w2*128,
sce->ics.swb_sizes[g],
scf,
ESC_BT,
lambda,
INFINITY,
&b);
dist -= b;
}
dist *= 1.0f / 512.0f / lambda;
quant_max = quant(maxq[w*16+g], ff_aac_pow2sf_tab[200 - scf + SCALE_ONE_POS - SCALE_DIV_512]);
if (quant_max >= 8191) { // too much, return to the previous quantizer
sce->sf_idx[w*16+g] = prev_scf;
break;
}
prev_scf = scf;
curdiff = fabsf(dist - uplim[w*16+g]);
if (curdiff <= 1.0f)
step = 0;
else
step = log2(curdiff);
if (dist > uplim[w*16+g])
step = -step;
scf += step;
scf = av_clip_uint8(scf);
step = scf - prev_scf;
if (FFABS(step) <= 1 || (step > 0 && scf >= max_scf) || (step < 0 && scf <= min_scf)) {
sce->sf_idx[w*16+g] = av_clip(scf, min_scf, max_scf);
break;
}
if (step > 0)
min_scf = prev_scf;
else
max_scf = prev_scf;
}
start += size;
}
}
minq = sce->sf_idx[0] ? sce->sf_idx[0] : INT_MAX;
for (i = 1; i < 128; i++) {
if (!sce->sf_idx[i])
sce->sf_idx[i] = sce->sf_idx[i-1];
else
minq = FFMIN(minq, sce->sf_idx[i]);
}
if (minq == INT_MAX)
minq = 0;
minq = FFMIN(minq, SCALE_MAX_POS);
maxsf = FFMIN(minq + SCALE_MAX_DIFF, SCALE_MAX_POS);
for (i = 126; i >= 0; i--) {
if (!sce->sf_idx[i])
sce->sf_idx[i] = sce->sf_idx[i+1];
sce->sf_idx[i] = av_clip(sce->sf_idx[i], minq, maxsf);
}
}
static void search_for_quantizers_fast(AVCodecContext *avctx, AACEncContext *s,
SingleChannelElement *sce,
const float lambda)
{
int start = 0, i, w, w2, g;
int minq = 255;
memset(sce->sf_idx, 0, sizeof(sce->sf_idx));
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
start = w*128;
for (g = 0; g < sce->ics.num_swb; g++) {
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
FFPsyBand *band = &s->psy.psy_bands[s->cur_channel*PSY_MAX_BANDS+(w+w2)*16+g];
if (band->energy <= band->threshold) {
sce->sf_idx[(w+w2)*16+g] = 218;
sce->zeroes[(w+w2)*16+g] = 1;
} else {
sce->sf_idx[(w+w2)*16+g] = av_clip(SCALE_ONE_POS - SCALE_DIV_512 + log2(band->threshold), 80, 218);
sce->zeroes[(w+w2)*16+g] = 0;
}
minq = FFMIN(minq, sce->sf_idx[(w+w2)*16+g]);
}
}
}
for (i = 0; i < 128; i++) {
sce->sf_idx[i] = 140;
//av_clip(sce->sf_idx[i], minq, minq + SCALE_MAX_DIFF - 1);
}
//set the same quantizers inside window groups
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
for (g = 0; g < sce->ics.num_swb; g++)
for (w2 = 1; w2 < sce->ics.group_len[w]; w2++)
sce->sf_idx[(w+w2)*16+g] = sce->sf_idx[w*16+g];
}
static void search_for_ms(AACEncContext *s, ChannelElement *cpe,
const float lambda)
{
int start = 0, i, w, w2, g;
float M[128], S[128];
float *L34 = s->scoefs, *R34 = s->scoefs + 128, *M34 = s->scoefs + 128*2, *S34 = s->scoefs + 128*3;
SingleChannelElement *sce0 = &cpe->ch[0];
SingleChannelElement *sce1 = &cpe->ch[1];
if (!cpe->common_window)
return;
for (w = 0; w < sce0->ics.num_windows; w += sce0->ics.group_len[w]) {
for (g = 0; g < sce0->ics.num_swb; g++) {
if (!cpe->ch[0].zeroes[w*16+g] && !cpe->ch[1].zeroes[w*16+g]) {
float dist1 = 0.0f, dist2 = 0.0f;
for (w2 = 0; w2 < sce0->ics.group_len[w]; w2++) {
FFPsyBand *band0 = &s->psy.psy_bands[(s->cur_channel+0)*PSY_MAX_BANDS+(w+w2)*16+g];
FFPsyBand *band1 = &s->psy.psy_bands[(s->cur_channel+1)*PSY_MAX_BANDS+(w+w2)*16+g];
float minthr = FFMIN(band0->threshold, band1->threshold);
float maxthr = FFMAX(band0->threshold, band1->threshold);
for (i = 0; i < sce0->ics.swb_sizes[g]; i++) {
M[i] = (sce0->coeffs[start+w2*128+i]
+ sce1->coeffs[start+w2*128+i]) * 0.5;
S[i] = sce0->coeffs[start+w2*128+i]
- sce1->coeffs[start+w2*128+i];
}
abs_pow34_v(L34, sce0->coeffs+start+w2*128, sce0->ics.swb_sizes[g]);
abs_pow34_v(R34, sce1->coeffs+start+w2*128, sce0->ics.swb_sizes[g]);
abs_pow34_v(M34, M, sce0->ics.swb_sizes[g]);
abs_pow34_v(S34, S, sce0->ics.swb_sizes[g]);
dist1 += quantize_band_cost(s, sce0->coeffs + start + w2*128,
L34,
sce0->ics.swb_sizes[g],
sce0->sf_idx[(w+w2)*16+g],
sce0->band_type[(w+w2)*16+g],
lambda / band0->threshold, INFINITY, NULL);
dist1 += quantize_band_cost(s, sce1->coeffs + start + w2*128,
R34,
sce1->ics.swb_sizes[g],
sce1->sf_idx[(w+w2)*16+g],
sce1->band_type[(w+w2)*16+g],
lambda / band1->threshold, INFINITY, NULL);
dist2 += quantize_band_cost(s, M,
M34,
sce0->ics.swb_sizes[g],
sce0->sf_idx[(w+w2)*16+g],
sce0->band_type[(w+w2)*16+g],
lambda / maxthr, INFINITY, NULL);
dist2 += quantize_band_cost(s, S,
S34,
sce1->ics.swb_sizes[g],
sce1->sf_idx[(w+w2)*16+g],
sce1->band_type[(w+w2)*16+g],
lambda / minthr, INFINITY, NULL);
}
cpe->ms_mask[w*16+g] = dist2 < dist1;
}
start += sce0->ics.swb_sizes[g];
}
}
}
AACCoefficientsEncoder ff_aac_coders[] = {
{
search_for_quantizers_faac,
encode_window_bands_info,
quantize_and_encode_band,
search_for_ms,
},
{
search_for_quantizers_anmr,
encode_window_bands_info,
quantize_and_encode_band,
search_for_ms,
},
{
search_for_quantizers_twoloop,
encode_window_bands_info,
quantize_and_encode_band,
search_for_ms,
},
{
search_for_quantizers_fast,
encode_window_bands_info,
quantize_and_encode_band,
search_for_ms,
},
};

File diff suppressed because it is too large Load Diff

View File

@@ -21,7 +21,7 @@
*/
/**
* @file
* @file libavcodec/aacdectab.h
* AAC decoder data
* @author Oded Shimon ( ods15 ods15 dyndns org )
* @author Maxim Gavrilov ( maxim.gavrilov gmail com )
@@ -34,6 +34,145 @@
#include <stdint.h>
/* @name swb_offsets
* Sample offset into the window indicating the beginning of a scalefactor
* window band
*
* scalefactor window band - term for scalefactor bands within a window,
* given in Table 4.110 to Table 4.128.
*
* scalefactor band - a set of spectral coefficients which are scaled by one
* scalefactor. In case of EIGHT_SHORT_SEQUENCE and grouping a scalefactor band
* may contain several scalefactor window bands of corresponding frequency. For
* all other window_sequences scalefactor bands and scalefactor window bands are
* identical.
* @{
*/
static const uint16_t swb_offset_1024_96[] = {
0, 4, 8, 12, 16, 20, 24, 28,
32, 36, 40, 44, 48, 52, 56, 64,
72, 80, 88, 96, 108, 120, 132, 144,
156, 172, 188, 212, 240, 276, 320, 384,
448, 512, 576, 640, 704, 768, 832, 896,
960, 1024
};
static const uint16_t swb_offset_128_96[] = {
0, 4, 8, 12, 16, 20, 24, 32, 40, 48, 64, 92, 128
};
static const uint16_t swb_offset_1024_64[] = {
0, 4, 8, 12, 16, 20, 24, 28,
32, 36, 40, 44, 48, 52, 56, 64,
72, 80, 88, 100, 112, 124, 140, 156,
172, 192, 216, 240, 268, 304, 344, 384,
424, 464, 504, 544, 584, 624, 664, 704,
744, 784, 824, 864, 904, 944, 984, 1024
};
static const uint16_t swb_offset_1024_48[] = {
0, 4, 8, 12, 16, 20, 24, 28,
32, 36, 40, 48, 56, 64, 72, 80,
88, 96, 108, 120, 132, 144, 160, 176,
196, 216, 240, 264, 292, 320, 352, 384,
416, 448, 480, 512, 544, 576, 608, 640,
672, 704, 736, 768, 800, 832, 864, 896,
928, 1024
};
static const uint16_t swb_offset_128_48[] = {
0, 4, 8, 12, 16, 20, 28, 36,
44, 56, 68, 80, 96, 112, 128
};
static const uint16_t swb_offset_1024_32[] = {
0, 4, 8, 12, 16, 20, 24, 28,
32, 36, 40, 48, 56, 64, 72, 80,
88, 96, 108, 120, 132, 144, 160, 176,
196, 216, 240, 264, 292, 320, 352, 384,
416, 448, 480, 512, 544, 576, 608, 640,
672, 704, 736, 768, 800, 832, 864, 896,
928, 960, 992, 1024
};
static const uint16_t swb_offset_1024_24[] = {
0, 4, 8, 12, 16, 20, 24, 28,
32, 36, 40, 44, 52, 60, 68, 76,
84, 92, 100, 108, 116, 124, 136, 148,
160, 172, 188, 204, 220, 240, 260, 284,
308, 336, 364, 396, 432, 468, 508, 552,
600, 652, 704, 768, 832, 896, 960, 1024
};
static const uint16_t swb_offset_128_24[] = {
0, 4, 8, 12, 16, 20, 24, 28,
36, 44, 52, 64, 76, 92, 108, 128
};
static const uint16_t swb_offset_1024_16[] = {
0, 8, 16, 24, 32, 40, 48, 56,
64, 72, 80, 88, 100, 112, 124, 136,
148, 160, 172, 184, 196, 212, 228, 244,
260, 280, 300, 320, 344, 368, 396, 424,
456, 492, 532, 572, 616, 664, 716, 772,
832, 896, 960, 1024
};
static const uint16_t swb_offset_128_16[] = {
0, 4, 8, 12, 16, 20, 24, 28,
32, 40, 48, 60, 72, 88, 108, 128
};
static const uint16_t swb_offset_1024_8[] = {
0, 12, 24, 36, 48, 60, 72, 84,
96, 108, 120, 132, 144, 156, 172, 188,
204, 220, 236, 252, 268, 288, 308, 328,
348, 372, 396, 420, 448, 476, 508, 544,
580, 620, 664, 712, 764, 820, 880, 944,
1024
};
static const uint16_t swb_offset_128_8[] = {
0, 4, 8, 12, 16, 20, 24, 28,
36, 44, 52, 60, 72, 88, 108, 128
};
static const uint16_t *swb_offset_1024[] = {
swb_offset_1024_96, swb_offset_1024_96, swb_offset_1024_64,
swb_offset_1024_48, swb_offset_1024_48, swb_offset_1024_32,
swb_offset_1024_24, swb_offset_1024_24, swb_offset_1024_16,
swb_offset_1024_16, swb_offset_1024_16, swb_offset_1024_8,
swb_offset_1024_8
};
static const uint16_t *swb_offset_128[] = {
/* The last entry on the following row is swb_offset_128_64 but is a
duplicate of swb_offset_128_96. */
swb_offset_128_96, swb_offset_128_96, swb_offset_128_96,
swb_offset_128_48, swb_offset_128_48, swb_offset_128_48,
swb_offset_128_24, swb_offset_128_24, swb_offset_128_16,
swb_offset_128_16, swb_offset_128_16, swb_offset_128_8,
swb_offset_128_8
};
// @}
/* @name tns_max_bands
* The maximum number of scalefactor bands on which TNS can operate for the long
* and short transforms respectively. The index to these tables is related to
* the sample rate of the audio.
* @{
*/
static const uint8_t tns_max_bands_1024[] = {
31, 31, 34, 40, 42, 51, 46, 46, 42, 42, 42, 39, 39
};
static const uint8_t tns_max_bands_128[] = {
9, 9, 10, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14
};
// @}
/* @name tns_tmp2_map
* Tables of the tmp2[] arrays of LPC coefficients used for TNS.
* The suffix _M_N[] indicate the values of coef_compress and coef_res
@@ -69,27 +208,4 @@ static const float * const tns_tmp2_map[4] = {
};
// @}
static const int8_t tags_per_config[16] = { 0, 1, 1, 2, 3, 3, 4, 5, 0, 0, 0, 0, 0, 0, 0, 0 };
static const uint8_t aac_channel_layout_map[7][5][2] = {
{ { TYPE_SCE, 0 }, },
{ { TYPE_CPE, 0 }, },
{ { TYPE_CPE, 0 }, { TYPE_SCE, 0 }, },
{ { TYPE_CPE, 0 }, { TYPE_SCE, 0 }, { TYPE_SCE, 1 }, },
{ { TYPE_CPE, 0 }, { TYPE_SCE, 0 }, { TYPE_CPE, 1 }, },
{ { TYPE_CPE, 0 }, { TYPE_SCE, 0 }, { TYPE_LFE, 0 }, { TYPE_CPE, 1 }, },
{ { TYPE_CPE, 0 }, { TYPE_SCE, 0 }, { TYPE_LFE, 0 }, { TYPE_CPE, 2 }, { TYPE_CPE, 1 }, },
};
static const int64_t aac_channel_layout[8] = {
CH_LAYOUT_MONO,
CH_LAYOUT_STEREO,
CH_LAYOUT_SURROUND,
CH_LAYOUT_4POINT0,
CH_LAYOUT_5POINT0_BACK,
CH_LAYOUT_5POINT1_BACK,
CH_LAYOUT_7POINT1_WIDE,
0,
};
#endif /* AVCODEC_AACDECTAB_H */

View File

@@ -20,26 +20,25 @@
*/
/**
* @file
* @file libavcodec/aacenc.c
* AAC encoder
*/
/***********************************
* TODOs:
* psy model selection with some option
* add sane pulse detection
* add temporal noise shaping
***********************************/
#include "avcodec.h"
#include "put_bits.h"
#include "bitstream.h"
#include "dsputil.h"
#include "mpeg4audio.h"
#include "aacpsy.h"
#include "aac.h"
#include "aactab.h"
#include "aacenc.h"
#include "psymodel.h"
static const uint8_t swb_size_1024_96[] = {
4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8,
@@ -84,7 +83,7 @@ static const uint8_t swb_size_1024_8[] = {
32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80
};
static const uint8_t *swb_size_1024[] = {
static const uint8_t * const swb_size_1024[] = {
swb_size_1024_96, swb_size_1024_96, swb_size_1024_64,
swb_size_1024_48, swb_size_1024_48, swb_size_1024_32,
swb_size_1024_24, swb_size_1024_24, swb_size_1024_16,
@@ -111,7 +110,7 @@ static const uint8_t swb_size_128_8[] = {
4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20
};
static const uint8_t *swb_size_128[] = {
static const uint8_t * const swb_size_128[] = {
/* the last entry on the following row is swb_size_128_64 but is a
duplicate of swb_size_128_96 */
swb_size_128_96, swb_size_128_96, swb_size_128_96,
@@ -120,6 +119,23 @@ static const uint8_t *swb_size_128[] = {
swb_size_128_16, swb_size_128_16, swb_size_128_8
};
/** bits needed to code codebook run value for long windows */
static const uint8_t run_value_bits_long[64] = {
5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 10,
10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10,
10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 15
};
/** bits needed to code codebook run value for short windows */
static const uint8_t run_value_bits_short[16] = {
3, 3, 3, 3, 3, 3, 3, 6, 6, 6, 6, 6, 6, 6, 6, 9
};
static const uint8_t* const run_value_bits[2] = {
run_value_bits_long, run_value_bits_short
};
/** default channel configurations */
static const uint8_t aac_chan_configs[6][5] = {
{1, TYPE_SCE}, // 1 channel - single channel element
@@ -130,6 +146,33 @@ static const uint8_t aac_chan_configs[6][5] = {
{4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE
};
/**
* structure used in optimal codebook search
*/
typedef struct BandCodingPath {
int prev_idx; ///< pointer to the previous path point
int codebook; ///< codebook for coding band run
int bits; ///< number of bit needed to code given number of bands
} BandCodingPath;
/**
* AAC encoder context
*/
typedef struct {
PutBitContext pb;
MDCTContext mdct1024; ///< long (1024 samples) frame transform context
MDCTContext mdct128; ///< short (128 samples) frame transform context
DSPContext dsp;
DECLARE_ALIGNED_16(FFTSample, output[2048]); ///< temporary buffer for MDCT input coefficients
int16_t* samples; ///< saved preprocessed input
int samplerate_index; ///< MPEG-4 samplerate index
ChannelElement *cpe; ///< channel elements
AACPsyContext psy; ///< psychoacoustic model context
int last_frame;
} AACEncContext;
/**
* Make AAC audio config object.
* @see 1.6.2.1 "Syntax - AudioSpecificConfig"
@@ -154,243 +197,82 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
{
AACEncContext *s = avctx->priv_data;
int i;
const uint8_t *sizes[2];
int lengths[2];
avctx->frame_size = 1024;
for (i = 0; i < 16; i++)
if (avctx->sample_rate == ff_mpeg4audio_sample_rates[i])
for(i = 0; i < 16; i++)
if(avctx->sample_rate == ff_mpeg4audio_sample_rates[i])
break;
if (i == 16) {
if(i == 16){
av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n", avctx->sample_rate);
return -1;
}
if (avctx->channels > 6) {
if(avctx->channels > 6){
av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n", avctx->channels);
return -1;
}
if (avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW) {
av_log(avctx, AV_LOG_ERROR, "Unsupported profile %d\n", avctx->profile);
return -1;
}
if (1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * avctx->channels) {
av_log(avctx, AV_LOG_ERROR, "Too many bits per frame requested\n");
return -1;
}
s->samplerate_index = i;
dsputil_init(&s->dsp, avctx);
ff_mdct_init(&s->mdct1024, 11, 0, 1.0);
ff_mdct_init(&s->mdct128, 8, 0, 1.0);
ff_mdct_init(&s->mdct1024, 11, 0);
ff_mdct_init(&s->mdct128, 8, 0);
// window init
ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
ff_init_ff_sine_windows(10);
ff_init_ff_sine_windows(7);
ff_sine_window_init(ff_sine_1024, 1024);
ff_sine_window_init(ff_sine_128, 128);
s->samples = av_malloc(2 * 1024 * avctx->channels * sizeof(s->samples[0]));
s->cpe = av_mallocz(sizeof(ChannelElement) * aac_chan_configs[avctx->channels-1][0]);
avctx->extradata = av_malloc(2);
s->samples = av_malloc(2 * 1024 * avctx->channels * sizeof(s->samples[0]));
s->cpe = av_mallocz(sizeof(ChannelElement) * aac_chan_configs[avctx->channels-1][0]);
if(ff_aac_psy_init(&s->psy, avctx, AAC_PSY_3GPP,
aac_chan_configs[avctx->channels-1][0], 0,
swb_size_1024[i], ff_aac_num_swb_1024[i], swb_size_128[i], ff_aac_num_swb_128[i]) < 0){
av_log(avctx, AV_LOG_ERROR, "Cannot initialize selected model.\n");
return -1;
}
avctx->extradata = av_malloc(2);
avctx->extradata_size = 2;
put_audio_specific_config(avctx);
sizes[0] = swb_size_1024[i];
sizes[1] = swb_size_128[i];
lengths[0] = ff_aac_num_swb_1024[i];
lengths[1] = ff_aac_num_swb_128[i];
ff_psy_init(&s->psy, avctx, 2, sizes, lengths);
s->psypp = ff_psy_preprocess_init(avctx);
s->coder = &ff_aac_coders[2];
s->lambda = avctx->global_quality ? avctx->global_quality : 120;
ff_aac_tableinit();
if (avctx->channels > 5)
av_log(avctx, AV_LOG_ERROR, "This encoder does not yet enforce the restrictions on LFEs. "
"The output will most likely be an illegal bitstream.\n");
return 0;
}
static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s,
SingleChannelElement *sce, short *audio, int channel)
{
int i, j, k;
const float * lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
const float * swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
const float * pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
memcpy(s->output, sce->saved, sizeof(float)*1024);
if (sce->ics.window_sequence[0] == LONG_STOP_SEQUENCE) {
memset(s->output, 0, sizeof(s->output[0]) * 448);
for (i = 448; i < 576; i++)
s->output[i] = sce->saved[i] * pwindow[i - 448];
for (i = 576; i < 704; i++)
s->output[i] = sce->saved[i];
}
if (sce->ics.window_sequence[0] != LONG_START_SEQUENCE) {
for (i = 0, j = channel; i < 1024; i++, j += avctx->channels) {
s->output[i+1024] = audio[j] * lwindow[1024 - i - 1];
sce->saved[i] = audio[j] * lwindow[i];
}
} else {
for (i = 0, j = channel; i < 448; i++, j += avctx->channels)
s->output[i+1024] = audio[j];
for (; i < 576; i++, j += avctx->channels)
s->output[i+1024] = audio[j] * swindow[576 - i - 1];
memset(s->output+1024+576, 0, sizeof(s->output[0]) * 448);
for (i = 0, j = channel; i < 1024; i++, j += avctx->channels)
sce->saved[i] = audio[j];
}
ff_mdct_calc(&s->mdct1024, sce->coeffs, s->output);
} else {
for (k = 0; k < 1024; k += 128) {
for (i = 448 + k; i < 448 + k + 256; i++)
s->output[i - 448 - k] = (i < 1024)
? sce->saved[i]
: audio[channel + (i-1024)*avctx->channels];
s->dsp.vector_fmul (s->output, k ? swindow : pwindow, 128);
s->dsp.vector_fmul_reverse(s->output+128, s->output+128, swindow, 128);
ff_mdct_calc(&s->mdct128, sce->coeffs + k, s->output);
}
for (i = 0, j = channel; i < 1024; i++, j += avctx->channels)
sce->saved[i] = audio[j];
}
}
/**
* Encode ics_info element.
* @see Table 4.6 (syntax of ics_info)
*/
static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
{
int w;
int i;
put_bits(&s->pb, 1, 0); // ics_reserved bit
put_bits(&s->pb, 2, info->window_sequence[0]);
put_bits(&s->pb, 1, info->use_kb_window[0]);
if (info->window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
if(info->window_sequence[0] != EIGHT_SHORT_SEQUENCE){
put_bits(&s->pb, 6, info->max_sfb);
put_bits(&s->pb, 1, 0); // no prediction
} else {
}else{
put_bits(&s->pb, 4, info->max_sfb);
for (w = 1; w < 8; w++)
put_bits(&s->pb, 1, !info->group_len[w]);
for(i = 1; i < info->num_windows; i++)
put_bits(&s->pb, 1, info->group_len[i]);
}
}
/**
* Encode MS data.
* @see 4.6.8.1 "Joint Coding - M/S Stereo"
* Calculate the number of bits needed to code all coefficient signs in current band.
*/
static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)
static int calculate_band_sign_bits(AACEncContext *s, SingleChannelElement *sce,
int group_len, int start, int size)
{
int bits = 0;
int i, w;
put_bits(pb, 2, cpe->ms_mode);
if (cpe->ms_mode == 1)
for (w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w])
for (i = 0; i < cpe->ch[0].ics.max_sfb; i++)
put_bits(pb, 1, cpe->ms_mask[w*16 + i]);
}
/**
* Produce integer coefficients from scalefactors provided by the model.
*/
static void adjust_frame_information(AACEncContext *apc, ChannelElement *cpe, int chans)
{
int i, w, w2, g, ch;
int start, sum, maxsfb, cmaxsfb;
for (ch = 0; ch < chans; ch++) {
IndividualChannelStream *ics = &cpe->ch[ch].ics;
start = 0;
maxsfb = 0;
cpe->ch[ch].pulse.num_pulse = 0;
for (w = 0; w < ics->num_windows*16; w += 16) {
for (g = 0; g < ics->num_swb; g++) {
sum = 0;
//apply M/S
if (!ch && cpe->ms_mask[w + g]) {
for (i = 0; i < ics->swb_sizes[g]; i++) {
cpe->ch[0].coeffs[start+i] = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) / 2.0;
cpe->ch[1].coeffs[start+i] = cpe->ch[0].coeffs[start+i] - cpe->ch[1].coeffs[start+i];
}
}
start += ics->swb_sizes[g];
}
for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w+cmaxsfb-1]; cmaxsfb--)
;
maxsfb = FFMAX(maxsfb, cmaxsfb);
}
ics->max_sfb = maxsfb;
//adjust zero bands for window groups
for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
for (g = 0; g < ics->max_sfb; g++) {
i = 1;
for (w2 = w; w2 < w + ics->group_len[w]; w2++) {
if (!cpe->ch[ch].zeroes[w2*16 + g]) {
i = 0;
break;
}
}
cpe->ch[ch].zeroes[w*16 + g] = i;
}
}
}
if (chans > 1 && cpe->common_window) {
IndividualChannelStream *ics0 = &cpe->ch[0].ics;
IndividualChannelStream *ics1 = &cpe->ch[1].ics;
int msc = 0;
ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb);
ics1->max_sfb = ics0->max_sfb;
for (w = 0; w < ics0->num_windows*16; w += 16)
for (i = 0; i < ics0->max_sfb; i++)
if (cpe->ms_mask[w+i])
msc++;
if (msc == 0 || ics0->max_sfb == 0)
cpe->ms_mode = 0;
else
cpe->ms_mode = msc < ics0->max_sfb ? 1 : 2;
}
}
/**
* Encode scalefactor band coding type.
*/
static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
{
int w;
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);
}
/**
* Encode scalefactors.
*/
static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s,
SingleChannelElement *sce)
{
int off = sce->sf_idx[0], diff;
int i, w;
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
for (i = 0; i < sce->ics.max_sfb; i++) {
if (!sce->zeroes[w*16 + i]) {
diff = sce->sf_idx[w*16 + i] - off + SCALE_DIFF_ZERO;
if (diff < 0 || diff > 120)
av_log(avctx, AV_LOG_ERROR, "Scalefactor difference is too big to be coded\n");
off = sce->sf_idx[w*16 + i];
put_bits(&s->pb, ff_aac_scalefactor_bits[diff], ff_aac_scalefactor_code[diff]);
}
for(w = 0; w < group_len; w++){
for(i = 0; i < size; i++){
if(sce->icoefs[start + i])
bits++;
}
start += 128;
}
return bits;
}
/**
@@ -401,12 +283,11 @@ static void encode_pulses(AACEncContext *s, Pulse *pulse)
int i;
put_bits(&s->pb, 1, !!pulse->num_pulse);
if (!pulse->num_pulse)
return;
if(!pulse->num_pulse) return;
put_bits(&s->pb, 2, pulse->num_pulse - 1);
put_bits(&s->pb, 6, pulse->start);
for (i = 0; i < pulse->num_pulse; i++) {
for(i = 0; i < pulse->num_pulse; i++){
put_bits(&s->pb, 5, pulse->pos[i]);
put_bits(&s->pb, 4, pulse->amp[i]);
}
@@ -417,214 +298,54 @@ static void encode_pulses(AACEncContext *s, Pulse *pulse)
*/
static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
{
int start, i, w, w2;
int start, i, w, w2, wg;
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
w = 0;
for(wg = 0; wg < sce->ics.num_window_groups; wg++){
start = 0;
for (i = 0; i < sce->ics.max_sfb; i++) {
if (sce->zeroes[w*16 + i]) {
for(i = 0; i < sce->ics.max_sfb; i++){
if(sce->zeroes[w*16 + i]){
start += sce->ics.swb_sizes[i];
continue;
}
for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++)
s->coder->quantize_and_encode_band(s, &s->pb, sce->coeffs + start + w2*128,
sce->ics.swb_sizes[i],
sce->sf_idx[w*16 + i],
sce->band_type[w*16 + i],
s->lambda);
for(w2 = w; w2 < w + sce->ics.group_len[wg]; w2++){
encode_band_coeffs(s, sce, start + w2*128,
sce->ics.swb_sizes[i],
sce->band_type[w*16 + i]);
}
start += sce->ics.swb_sizes[i];
}
w += sce->ics.group_len[wg];
}
}
/**
* Encode one channel of audio data.
*/
static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s,
SingleChannelElement *sce,
int common_window)
{
put_bits(&s->pb, 8, sce->sf_idx[0]);
if (!common_window)
put_ics_info(s, &sce->ics);
encode_band_info(s, sce);
encode_scale_factors(avctx, s, sce);
encode_pulses(s, &sce->pulse);
put_bits(&s->pb, 1, 0); //tns
put_bits(&s->pb, 1, 0); //ssr
encode_spectral_coeffs(s, sce);
return 0;
}
/**
* Write some auxiliary information about the created AAC file.
*/
static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s,
const char *name)
static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s, const char *name)
{
int i, namelen, padbits;
namelen = strlen(name) + 2;
put_bits(&s->pb, 3, TYPE_FIL);
put_bits(&s->pb, 4, FFMIN(namelen, 15));
if (namelen >= 15)
if(namelen >= 15)
put_bits(&s->pb, 8, namelen - 16);
put_bits(&s->pb, 4, 0); //extension type - filler
padbits = 8 - (put_bits_count(&s->pb) & 7);
align_put_bits(&s->pb);
for (i = 0; i < namelen - 2; i++)
for(i = 0; i < namelen - 2; i++)
put_bits(&s->pb, 8, name[i]);
put_bits(&s->pb, 12 - padbits, 0);
}
static int aac_encode_frame(AVCodecContext *avctx,
uint8_t *frame, int buf_size, void *data)
{
AACEncContext *s = avctx->priv_data;
int16_t *samples = s->samples, *samples2, *la;
ChannelElement *cpe;
int i, j, chans, tag, start_ch;
const uint8_t *chan_map = aac_chan_configs[avctx->channels-1];
int chan_el_counter[4];
FFPsyWindowInfo windows[avctx->channels];
if (s->last_frame)
return 0;
if (data) {
if (!s->psypp) {
memcpy(s->samples + 1024 * avctx->channels, data,
1024 * avctx->channels * sizeof(s->samples[0]));
} else {
start_ch = 0;
samples2 = s->samples + 1024 * avctx->channels;
for (i = 0; i < chan_map[0]; i++) {
tag = chan_map[i+1];
chans = tag == TYPE_CPE ? 2 : 1;
ff_psy_preprocess(s->psypp, (uint16_t*)data + start_ch,
samples2 + start_ch, start_ch, chans);
start_ch += chans;
}
}
}
if (!avctx->frame_number) {
memcpy(s->samples, s->samples + 1024 * avctx->channels,
1024 * avctx->channels * sizeof(s->samples[0]));
return 0;
}
start_ch = 0;
for (i = 0; i < chan_map[0]; i++) {
FFPsyWindowInfo* wi = windows + start_ch;
tag = chan_map[i+1];
chans = tag == TYPE_CPE ? 2 : 1;
cpe = &s->cpe[i];
samples2 = samples + start_ch;
la = samples2 + 1024 * avctx->channels + start_ch;
if (!data)
la = NULL;
for (j = 0; j < chans; j++) {
IndividualChannelStream *ics = &cpe->ch[j].ics;
int k;
wi[j] = ff_psy_suggest_window(&s->psy, samples2, la, start_ch + j, ics->window_sequence[0]);
ics->window_sequence[1] = ics->window_sequence[0];
ics->window_sequence[0] = wi[j].window_type[0];
ics->use_kb_window[1] = ics->use_kb_window[0];
ics->use_kb_window[0] = wi[j].window_shape;
ics->num_windows = wi[j].num_windows;
ics->swb_sizes = s->psy.bands [ics->num_windows == 8];
ics->num_swb = s->psy.num_bands[ics->num_windows == 8];
for (k = 0; k < ics->num_windows; k++)
ics->group_len[k] = wi[j].grouping[k];
s->cur_channel = start_ch + j;
apply_window_and_mdct(avctx, s, &cpe->ch[j], samples2, j);
}
start_ch += chans;
}
do {
int frame_bits;
init_put_bits(&s->pb, frame, buf_size*8);
if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT))
put_bitstream_info(avctx, s, LIBAVCODEC_IDENT);
start_ch = 0;
memset(chan_el_counter, 0, sizeof(chan_el_counter));
for (i = 0; i < chan_map[0]; i++) {
FFPsyWindowInfo* wi = windows + start_ch;
tag = chan_map[i+1];
chans = tag == TYPE_CPE ? 2 : 1;
cpe = &s->cpe[i];
for (j = 0; j < chans; j++) {
s->cur_channel = start_ch + j;
ff_psy_set_band_info(&s->psy, s->cur_channel, cpe->ch[j].coeffs, &wi[j]);
s->coder->search_for_quantizers(avctx, s, &cpe->ch[j], s->lambda);
}
cpe->common_window = 0;
if (chans > 1
&& wi[0].window_type[0] == wi[1].window_type[0]
&& wi[0].window_shape == wi[1].window_shape) {
cpe->common_window = 1;
for (j = 0; j < wi[0].num_windows; j++) {
if (wi[0].grouping[j] != wi[1].grouping[j]) {
cpe->common_window = 0;
break;
}
}
}
s->cur_channel = start_ch;
if (cpe->common_window && s->coder->search_for_ms)
s->coder->search_for_ms(s, cpe, s->lambda);
adjust_frame_information(s, cpe, chans);
put_bits(&s->pb, 3, tag);
put_bits(&s->pb, 4, chan_el_counter[tag]++);
if (chans == 2) {
put_bits(&s->pb, 1, cpe->common_window);
if (cpe->common_window) {
put_ics_info(s, &cpe->ch[0].ics);
encode_ms_info(&s->pb, cpe);
}
}
for (j = 0; j < chans; j++) {
s->cur_channel = start_ch + j;
encode_individual_channel(avctx, s, &cpe->ch[j], cpe->common_window);
}
start_ch += chans;
}
frame_bits = put_bits_count(&s->pb);
if (frame_bits <= 6144 * avctx->channels - 3)
break;
s->lambda *= avctx->bit_rate * 1024.0f / avctx->sample_rate / frame_bits;
} while (1);
put_bits(&s->pb, 3, TYPE_END);
flush_put_bits(&s->pb);
avctx->frame_bits = put_bits_count(&s->pb);
// rate control stuff
if (!(avctx->flags & CODEC_FLAG_QSCALE)) {
float ratio = avctx->bit_rate * 1024.0f / avctx->sample_rate / avctx->frame_bits;
s->lambda *= ratio;
s->lambda = FFMIN(s->lambda, 65536.f);
}
if (!data)
s->last_frame = 1;
memcpy(s->samples, s->samples + 1024 * avctx->channels,
1024 * avctx->channels * sizeof(s->samples[0]));
return put_bits_count(&s->pb)>>3;
}
static av_cold int aac_encode_end(AVCodecContext *avctx)
{
AACEncContext *s = avctx->priv_data;
ff_mdct_end(&s->mdct1024);
ff_mdct_end(&s->mdct128);
ff_psy_end(&s->psy);
ff_psy_preprocess_end(s->psypp);
ff_aac_psy_end(&s->psy);
av_freep(&s->samples);
av_freep(&s->cpe);
return 0;
@@ -632,13 +353,13 @@ static av_cold int aac_encode_end(AVCodecContext *avctx)
AVCodec aac_encoder = {
"aac",
AVMEDIA_TYPE_AUDIO,
CODEC_TYPE_AUDIO,
CODEC_ID_AAC,
sizeof(AACEncContext),
aac_encode_init,
aac_encode_frame,
aac_encode_end,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY | CODEC_CAP_EXPERIMENTAL,
.sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
.capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
.sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
};

View File

@@ -1,71 +0,0 @@
/*
* AAC encoder
* Copyright (C) 2008 Konstantin Shishkov
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_AACENC_H
#define AVCODEC_AACENC_H
#include "avcodec.h"
#include "put_bits.h"
#include "dsputil.h"
#include "aac.h"
#include "psymodel.h"
struct AACEncContext;
typedef struct AACCoefficientsEncoder {
void (*search_for_quantizers)(AVCodecContext *avctx, struct AACEncContext *s,
SingleChannelElement *sce, const float lambda);
void (*encode_window_bands_info)(struct AACEncContext *s, SingleChannelElement *sce,
int win, int group_len, const float lambda);
void (*quantize_and_encode_band)(struct AACEncContext *s, PutBitContext *pb, const float *in, int size,
int scale_idx, int cb, const float lambda);
void (*search_for_ms)(struct AACEncContext *s, ChannelElement *cpe, const float lambda);
} AACCoefficientsEncoder;
extern AACCoefficientsEncoder ff_aac_coders[];
/**
* AAC encoder context
*/
typedef struct AACEncContext {
PutBitContext pb;
FFTContext mdct1024; ///< long (1024 samples) frame transform context
FFTContext mdct128; ///< short (128 samples) frame transform context
DSPContext dsp;
DECLARE_ALIGNED(16, FFTSample, output)[2048]; ///< temporary buffer for MDCT input coefficients
int16_t* samples; ///< saved preprocessed input
int samplerate_index; ///< MPEG-4 samplerate index
ChannelElement *cpe; ///< channel elements
FFPsyContext psy;
struct FFPsyPreprocessContext* psypp;
AACCoefficientsEncoder *coder;
int cur_channel;
int last_frame;
float lambda;
DECLARE_ALIGNED(16, int, qcoefs)[96]; ///< quantized coefficients
DECLARE_ALIGNED(16, float, scoefs)[1024]; ///< scaled coefficients
} AACEncContext;
#endif /* AVCODEC_AACENC_H */

File diff suppressed because it is too large Load Diff

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@@ -1,82 +0,0 @@
/*
* MPEG-4 Parametric Stereo definitions and declarations
* Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_PS_H
#define AVCODEC_PS_H
#include <stdint.h>
#include "avcodec.h"
#include "get_bits.h"
#define PS_MAX_NUM_ENV 5
#define PS_MAX_NR_IIDICC 34
#define PS_MAX_NR_IPDOPD 17
#define PS_MAX_SSB 91
#define PS_MAX_AP_BANDS 50
#define PS_QMF_TIME_SLOTS 32
#define PS_MAX_DELAY 14
#define PS_AP_LINKS 3
#define PS_MAX_AP_DELAY 5
typedef struct {
int start;
int enable_iid;
int iid_quant;
int nr_iid_par;
int nr_ipdopd_par;
int enable_icc;
int icc_mode;
int nr_icc_par;
int enable_ext;
int frame_class;
int num_env_old;
int num_env;
int enable_ipdopd;
int border_position[PS_MAX_NUM_ENV+1];
int8_t iid_par[PS_MAX_NUM_ENV][PS_MAX_NR_IIDICC]; //<Inter-channel Intensity Difference Parameters
int8_t icc_par[PS_MAX_NUM_ENV][PS_MAX_NR_IIDICC]; //<Inter-Channel Coherence Parameters
/* ipd/opd is iid/icc sized so that the same functions can handle both */
int8_t ipd_par[PS_MAX_NUM_ENV][PS_MAX_NR_IIDICC]; //<Inter-channel Phase Difference Parameters
int8_t opd_par[PS_MAX_NUM_ENV][PS_MAX_NR_IIDICC]; //<Overall Phase Difference Parameters
int is34bands;
int is34bands_old;
float in_buf[5][44][2];
float delay[PS_MAX_SSB][PS_QMF_TIME_SLOTS + PS_MAX_DELAY][2];
float ap_delay[PS_MAX_AP_BANDS][PS_AP_LINKS][PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2];
float peak_decay_nrg[34];
float power_smooth[34];
float peak_decay_diff_smooth[34];
float H11[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
float H12[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
float H21[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
float H22[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
int8_t opd_hist[PS_MAX_NR_IIDICC];
int8_t ipd_hist[PS_MAX_NR_IIDICC];
} PSContext;
void ff_ps_init(void);
void ff_ps_ctx_init(PSContext *ps);
int ff_ps_read_data(AVCodecContext *avctx, GetBitContext *gb, PSContext *ps, int bits_left);
int ff_ps_apply(AVCodecContext *avctx, PSContext *ps, float L[2][38][64], float R[2][38][64], int top);
#endif /* AVCODEC_PS_H */

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@@ -1,93 +0,0 @@
/*
* Generate a header file for hardcoded Parametric Stereo tables
*
* Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <stdlib.h>
#define CONFIG_HARDCODED_TABLES 0
#include "aacps_tablegen.h"
#include "tableprint.h"
void write_float_3d_array (const void *p, int b, int c, int d)
{
int i;
const float *f = p;
for (i = 0; i < b; i++) {
printf("{\n");
write_float_2d_array(f, c, d);
printf("},\n");
f += c * d;
}
}
void write_float_4d_array (const void *p, int a, int b, int c, int d)
{
int i;
const float *f = p;
for (i = 0; i < a; i++) {
printf("{\n");
write_float_3d_array(f, b, c, d);
printf("},\n");
f += b * c * d;
}
}
int main(void)
{
ps_tableinit();
write_fileheader();
printf("static const float pd_re_smooth[8*8*8] = {\n");
write_float_array(pd_re_smooth, 8*8*8);
printf("};\n");
printf("static const float pd_im_smooth[8*8*8] = {\n");
write_float_array(pd_im_smooth, 8*8*8);
printf("};\n");
printf("static const float HA[46][8][4] = {\n");
write_float_3d_array(HA, 46, 8, 4);
printf("};\n");
printf("static const float HB[46][8][4] = {\n");
write_float_3d_array(HB, 46, 8, 4);
printf("};\n");
printf("static const float f20_0_8[8][7][2] = {\n");
write_float_3d_array(f20_0_8, 8, 7, 2);
printf("};\n");
printf("static const float f34_0_12[12][7][2] = {\n");
write_float_3d_array(f34_0_12, 12, 7, 2);
printf("};\n");
printf("static const float f34_1_8[8][7][2] = {\n");
write_float_3d_array(f34_1_8, 8, 7, 2);
printf("};\n");
printf("static const float f34_2_4[4][7][2] = {\n");
write_float_3d_array(f34_2_4, 4, 7, 2);
printf("};\n");
printf("static const float Q_fract_allpass[2][50][3][2] = {\n");
write_float_4d_array(Q_fract_allpass, 2, 50, 3, 2);
printf("};\n");
printf("static const float phi_fract[2][50][2] = {\n");
write_float_3d_array(phi_fract, 2, 50, 2);
printf("};\n");
return 0;
}

View File

@@ -1,212 +0,0 @@
/*
* Header file for hardcoded Parametric Stereo tables
*
* Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AACPS_TABLEGEN_H
#define AACPS_TABLEGEN_H
#include <stdint.h>
#if CONFIG_HARDCODED_TABLES
#define ps_tableinit()
#include "libavcodec/aacps_tables.h"
#else
#include "../libavutil/common.h"
#include "../libavutil/mathematics.h"
#define NR_ALLPASS_BANDS20 30
#define NR_ALLPASS_BANDS34 50
#define PS_AP_LINKS 3
static float pd_re_smooth[8*8*8];
static float pd_im_smooth[8*8*8];
static float HA[46][8][4];
static float HB[46][8][4];
static float f20_0_8 [ 8][7][2];
static float f34_0_12[12][7][2];
static float f34_1_8 [ 8][7][2];
static float f34_2_4 [ 4][7][2];
static float Q_fract_allpass[2][50][3][2];
static float phi_fract[2][50][2];
static const float g0_Q8[] = {
0.00746082949812f, 0.02270420949825f, 0.04546865930473f, 0.07266113929591f,
0.09885108575264f, 0.11793710567217f, 0.125f
};
static const float g0_Q12[] = {
0.04081179924692f, 0.03812810994926f, 0.05144908135699f, 0.06399831151592f,
0.07428313801106f, 0.08100347892914f, 0.08333333333333f
};
static const float g1_Q8[] = {
0.01565675600122f, 0.03752716391991f, 0.05417891378782f, 0.08417044116767f,
0.10307344158036f, 0.12222452249753f, 0.125f
};
static const float g2_Q4[] = {
-0.05908211155639f, -0.04871498374946f, 0.0f, 0.07778723915851f,
0.16486303567403f, 0.23279856662996f, 0.25f
};
static void make_filters_from_proto(float (*filter)[7][2], const float *proto, int bands)
{
int q, n;
for (q = 0; q < bands; q++) {
for (n = 0; n < 7; n++) {
double theta = 2 * M_PI * (q + 0.5) * (n - 6) / bands;
filter[q][n][0] = proto[n] * cos(theta);
filter[q][n][1] = proto[n] * -sin(theta);
}
}
}
static void ps_tableinit(void)
{
static const float ipdopd_sin[] = { 0, M_SQRT1_2, 1, M_SQRT1_2, 0, -M_SQRT1_2, -1, -M_SQRT1_2 };
static const float ipdopd_cos[] = { 1, M_SQRT1_2, 0, -M_SQRT1_2, -1, -M_SQRT1_2, 0, M_SQRT1_2 };
int pd0, pd1, pd2;
static const float iid_par_dequant[] = {
//iid_par_dequant_default
0.05623413251903, 0.12589254117942, 0.19952623149689, 0.31622776601684,
0.44668359215096, 0.63095734448019, 0.79432823472428, 1,
1.25892541179417, 1.58489319246111, 2.23872113856834, 3.16227766016838,
5.01187233627272, 7.94328234724282, 17.7827941003892,
//iid_par_dequant_fine
0.00316227766017, 0.00562341325190, 0.01, 0.01778279410039,
0.03162277660168, 0.05623413251903, 0.07943282347243, 0.11220184543020,
0.15848931924611, 0.22387211385683, 0.31622776601684, 0.39810717055350,
0.50118723362727, 0.63095734448019, 0.79432823472428, 1,
1.25892541179417, 1.58489319246111, 1.99526231496888, 2.51188643150958,
3.16227766016838, 4.46683592150963, 6.30957344480193, 8.91250938133745,
12.5892541179417, 17.7827941003892, 31.6227766016838, 56.2341325190349,
100, 177.827941003892, 316.227766016837,
};
static const float icc_invq[] = {
1, 0.937, 0.84118, 0.60092, 0.36764, 0, -0.589, -1
};
static const float acos_icc_invq[] = {
0, 0.35685527, 0.57133466, 0.92614472, 1.1943263, M_PI/2, 2.2006171, M_PI
};
int iid, icc;
int k, m;
static const int8_t f_center_20[] = {
-3, -1, 1, 3, 5, 7, 10, 14, 18, 22,
};
static const int8_t f_center_34[] = {
2, 6, 10, 14, 18, 22, 26, 30,
34,-10, -6, -2, 51, 57, 15, 21,
27, 33, 39, 45, 54, 66, 78, 42,
102, 66, 78, 90,102,114,126, 90,
};
static const float fractional_delay_links[] = { 0.43f, 0.75f, 0.347f };
const float fractional_delay_gain = 0.39f;
for (pd0 = 0; pd0 < 8; pd0++) {
float pd0_re = ipdopd_cos[pd0];
float pd0_im = ipdopd_sin[pd0];
for (pd1 = 0; pd1 < 8; pd1++) {
float pd1_re = ipdopd_cos[pd1];
float pd1_im = ipdopd_sin[pd1];
for (pd2 = 0; pd2 < 8; pd2++) {
float pd2_re = ipdopd_cos[pd2];
float pd2_im = ipdopd_sin[pd2];
float re_smooth = 0.25f * pd0_re + 0.5f * pd1_re + pd2_re;
float im_smooth = 0.25f * pd0_im + 0.5f * pd1_im + pd2_im;
float pd_mag = 1 / sqrt(im_smooth * im_smooth + re_smooth * re_smooth);
pd_re_smooth[pd0*64+pd1*8+pd2] = re_smooth * pd_mag;
pd_im_smooth[pd0*64+pd1*8+pd2] = im_smooth * pd_mag;
}
}
}
for (iid = 0; iid < 46; iid++) {
float c = iid_par_dequant[iid]; //<Linear Inter-channel Intensity Difference
float c1 = (float)M_SQRT2 / sqrtf(1.0f + c*c);
float c2 = c * c1;
for (icc = 0; icc < 8; icc++) {
/*if (PS_BASELINE || ps->icc_mode < 3)*/ {
float alpha = 0.5f * acos_icc_invq[icc];
float beta = alpha * (c1 - c2) * (float)M_SQRT1_2;
HA[iid][icc][0] = c2 * cosf(beta + alpha);
HA[iid][icc][1] = c1 * cosf(beta - alpha);
HA[iid][icc][2] = c2 * sinf(beta + alpha);
HA[iid][icc][3] = c1 * sinf(beta - alpha);
} /* else */ {
float alpha, gamma, mu, rho;
float alpha_c, alpha_s, gamma_c, gamma_s;
rho = FFMAX(icc_invq[icc], 0.05f);
alpha = 0.5f * atan2f(2.0f * c * rho, c*c - 1.0f);
mu = c + 1.0f / c;
mu = sqrtf(1 + (4 * rho * rho - 4)/(mu * mu));
gamma = atanf(sqrtf((1.0f - mu)/(1.0f + mu)));
if (alpha < 0) alpha += M_PI/2;
alpha_c = cosf(alpha);
alpha_s = sinf(alpha);
gamma_c = cosf(gamma);
gamma_s = sinf(gamma);
HB[iid][icc][0] = M_SQRT2 * alpha_c * gamma_c;
HB[iid][icc][1] = M_SQRT2 * alpha_s * gamma_c;
HB[iid][icc][2] = -M_SQRT2 * alpha_s * gamma_s;
HB[iid][icc][3] = M_SQRT2 * alpha_c * gamma_s;
}
}
}
for (k = 0; k < NR_ALLPASS_BANDS20; k++) {
double f_center, theta;
if (k < FF_ARRAY_ELEMS(f_center_20))
f_center = f_center_20[k] * 0.125;
else
f_center = k - 6.5f;
for (m = 0; m < PS_AP_LINKS; m++) {
theta = -M_PI * fractional_delay_links[m] * f_center;
Q_fract_allpass[0][k][m][0] = cos(theta);
Q_fract_allpass[0][k][m][1] = sin(theta);
}
theta = -M_PI*fractional_delay_gain*f_center;
phi_fract[0][k][0] = cos(theta);
phi_fract[0][k][1] = sin(theta);
}
for (k = 0; k < NR_ALLPASS_BANDS34; k++) {
double f_center, theta;
if (k < FF_ARRAY_ELEMS(f_center_34))
f_center = f_center_34[k] / 24.;
else
f_center = k - 26.5f;
for (m = 0; m < PS_AP_LINKS; m++) {
theta = -M_PI * fractional_delay_links[m] * f_center;
Q_fract_allpass[1][k][m][0] = cos(theta);
Q_fract_allpass[1][k][m][1] = sin(theta);
}
theta = -M_PI*fractional_delay_gain*f_center;
phi_fract[1][k][0] = cos(theta);
phi_fract[1][k][1] = sin(theta);
}
make_filters_from_proto(f20_0_8, g0_Q8, 8);
make_filters_from_proto(f34_0_12, g0_Q12, 12);
make_filters_from_proto(f34_1_8, g1_Q8, 8);
make_filters_from_proto(f34_2_4, g2_Q4, 4);
}
#endif /* CONFIG_HARDCODED_TABLES */
#endif /* AACPS_TABLEGEN_H */

View File

@@ -1,163 +0,0 @@
/*
* MPEG-4 Parametric Stereo data tables
* Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
static const uint8_t huff_iid_df1_bits[] = {
18, 18, 18, 18, 18, 18, 18, 18, 18, 17, 18, 17, 17, 16, 16, 15, 14, 14,
13, 12, 12, 11, 10, 10, 8, 7, 6, 5, 4, 3, 1, 3, 4, 5, 6, 7,
8, 9, 10, 11, 11, 12, 13, 14, 14, 15, 16, 16, 17, 17, 18, 17, 18, 18,
18, 18, 18, 18, 18, 18, 18,
};
static const uint32_t huff_iid_df1_codes[] = {
0x01FEB4, 0x01FEB5, 0x01FD76, 0x01FD77, 0x01FD74, 0x01FD75, 0x01FE8A,
0x01FE8B, 0x01FE88, 0x00FE80, 0x01FEB6, 0x00FE82, 0x00FEB8, 0x007F42,
0x007FAE, 0x003FAF, 0x001FD1, 0x001FE9, 0x000FE9, 0x0007EA, 0x0007FB,
0x0003FB, 0x0001FB, 0x0001FF, 0x00007C, 0x00003C, 0x00001C, 0x00000C,
0x000000, 0x000001, 0x000001, 0x000002, 0x000001, 0x00000D, 0x00001D,
0x00003D, 0x00007D, 0x0000FC, 0x0001FC, 0x0003FC, 0x0003F4, 0x0007EB,
0x000FEA, 0x001FEA, 0x001FD6, 0x003FD0, 0x007FAF, 0x007F43, 0x00FEB9,
0x00FE83, 0x01FEB7, 0x00FE81, 0x01FE89, 0x01FE8E, 0x01FE8F, 0x01FE8C,
0x01FE8D, 0x01FEB2, 0x01FEB3, 0x01FEB0, 0x01FEB1,
};
static const uint8_t huff_iid_dt1_bits[] = {
16, 16, 16, 16, 16, 16, 16, 16, 16, 15, 15, 15, 15, 15, 15, 14, 14, 13,
13, 13, 12, 12, 11, 10, 9, 9, 7, 6, 5, 3, 1, 2, 5, 6, 7, 8,
9, 10, 11, 11, 12, 12, 13, 13, 14, 14, 15, 15, 15, 15, 16, 16, 16, 16,
16, 16, 16, 16, 16, 16, 16,
};
static const uint16_t huff_iid_dt1_codes[] = {
0x004ED4, 0x004ED5, 0x004ECE, 0x004ECF, 0x004ECC, 0x004ED6, 0x004ED8,
0x004F46, 0x004F60, 0x002718, 0x002719, 0x002764, 0x002765, 0x00276D,
0x0027B1, 0x0013B7, 0x0013D6, 0x0009C7, 0x0009E9, 0x0009ED, 0x0004EE,
0x0004F7, 0x000278, 0x000139, 0x00009A, 0x00009F, 0x000020, 0x000011,
0x00000A, 0x000003, 0x000001, 0x000000, 0x00000B, 0x000012, 0x000021,
0x00004C, 0x00009B, 0x00013A, 0x000279, 0x000270, 0x0004EF, 0x0004E2,
0x0009EA, 0x0009D8, 0x0013D7, 0x0013D0, 0x0027B2, 0x0027A2, 0x00271A,
0x00271B, 0x004F66, 0x004F67, 0x004F61, 0x004F47, 0x004ED9, 0x004ED7,
0x004ECD, 0x004ED2, 0x004ED3, 0x004ED0, 0x004ED1,
};
static const uint8_t huff_iid_df0_bits[] = {
17, 17, 17, 17, 16, 15, 13, 10, 9, 7, 6, 5, 4, 3, 1, 3, 4, 5,
6, 6, 8, 11, 13, 14, 14, 15, 17, 18, 18,
};
static const uint32_t huff_iid_df0_codes[] = {
0x01FFFB, 0x01FFFC, 0x01FFFD, 0x01FFFA, 0x00FFFC, 0x007FFC, 0x001FFD,
0x0003FE, 0x0001FE, 0x00007E, 0x00003C, 0x00001D, 0x00000D, 0x000005,
0x000000, 0x000004, 0x00000C, 0x00001C, 0x00003D, 0x00003E, 0x0000FE,
0x0007FE, 0x001FFC, 0x003FFC, 0x003FFD, 0x007FFD, 0x01FFFE, 0x03FFFE,
0x03FFFF,
};
static const uint8_t huff_iid_dt0_bits[] = {
19, 19, 19, 20, 20, 20, 17, 15, 12, 10, 8, 6, 4, 2, 1, 3, 5, 7,
9, 11, 13, 14, 17, 19, 20, 20, 20, 20, 20,
};
static const uint32_t huff_iid_dt0_codes[] = {
0x07FFF9, 0x07FFFA, 0x07FFFB, 0x0FFFF8, 0x0FFFF9, 0x0FFFFA, 0x01FFFD,
0x007FFE, 0x000FFE, 0x0003FE, 0x0000FE, 0x00003E, 0x00000E, 0x000002,
0x000000, 0x000006, 0x00001E, 0x00007E, 0x0001FE, 0x0007FE, 0x001FFE,
0x003FFE, 0x01FFFC, 0x07FFF8, 0x0FFFFB, 0x0FFFFC, 0x0FFFFD, 0x0FFFFE,
0x0FFFFF,
};
static const uint8_t huff_icc_df_bits[] = {
14, 14, 12, 10, 7, 5, 3, 1, 2, 4, 6, 8, 9, 11, 13,
};
static const uint16_t huff_icc_df_codes[] = {
0x3FFF, 0x3FFE, 0x0FFE, 0x03FE, 0x007E, 0x001E, 0x0006, 0x0000,
0x0002, 0x000E, 0x003E, 0x00FE, 0x01FE, 0x07FE, 0x1FFE,
};
static const uint8_t huff_icc_dt_bits[] = {
14, 13, 11, 9, 7, 5, 3, 1, 2, 4, 6, 8, 10, 12, 14,
};
static const uint16_t huff_icc_dt_codes[] = {
0x3FFE, 0x1FFE, 0x07FE, 0x01FE, 0x007E, 0x001E, 0x0006, 0x0000,
0x0002, 0x000E, 0x003E, 0x00FE, 0x03FE, 0x0FFE, 0x3FFF,
};
static const uint8_t huff_ipd_df_bits[] = {
1, 3, 4, 4, 4, 4, 4, 4,
};
static const uint8_t huff_ipd_df_codes[] = {
0x01, 0x00, 0x06, 0x04, 0x02, 0x03, 0x05, 0x07,
};
static const uint8_t huff_ipd_dt_bits[] = {
1, 3, 4, 5, 5, 4, 4, 3,
};
static const uint8_t huff_ipd_dt_codes[] = {
0x01, 0x02, 0x02, 0x03, 0x02, 0x00, 0x03, 0x03,
};
static const uint8_t huff_opd_df_bits[] = {
1, 3, 4, 4, 5, 5, 4, 3,
};
static const uint8_t huff_opd_df_codes[] = {
0x01, 0x01, 0x06, 0x04, 0x0F, 0x0E, 0x05, 0x00,
};
static const uint8_t huff_opd_dt_bits[] = {
1, 3, 4, 5, 5, 4, 4, 3,
};
static const uint8_t huff_opd_dt_codes[] = {
0x01, 0x02, 0x01, 0x07, 0x06, 0x00, 0x02, 0x03,
};
static const int8_t huff_offset[] = {
30, 30,
14, 14,
7, 7,
0, 0,
0, 0,
};
///Table 8.48
static const int8_t k_to_i_20[] = {
1, 0, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 14, 15,
15, 15, 16, 16, 16, 16, 17, 17, 17, 17, 17, 18, 18, 18, 18, 18, 18, 18, 18,
18, 18, 18, 18, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19,
19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19
};
///Table 8.49
static const int8_t k_to_i_34[] = {
0, 1, 2, 3, 4, 5, 6, 6, 7, 2, 1, 0, 10, 10, 4, 5, 6, 7, 8,
9, 10, 11, 12, 9, 14, 11, 12, 13, 14, 15, 16, 13, 16, 17, 18, 19, 20, 21,
22, 22, 23, 23, 24, 24, 25, 25, 26, 26, 27, 27, 27, 28, 28, 28, 29, 29, 29,
30, 30, 30, 31, 31, 31, 31, 32, 32, 32, 32, 33, 33, 33, 33, 33, 33, 33, 33,
33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33
};
static const float g1_Q2[] = {
0.0f, 0.01899487526049f, 0.0f, -0.07293139167538f,
0.0f, 0.30596630545168f, 0.5f
};

View File

@@ -20,30 +20,59 @@
*/
/**
* @file
* @file libavcodec/aacpsy.c
* AAC encoder psychoacoustic model
*/
#include "avcodec.h"
#include "aacpsy.h"
#include "aactab.h"
#include "psymodel.h"
/***********************************
* TODOs:
* General:
* better audio preprocessing (add DC highpass filter?)
* more psy models
* maybe improve coefficient quantization function in some way
*
* 3GPP-based psy model:
* thresholds linearization after their modifications for attaining given bitrate
* try other bitrate controlling mechanism (maybe use ratecontrol.c?)
* control quality for quality-based output
**********************************/
/**
* Quantize one coefficient.
* @return absolute value of the quantized coefficient
* @see 3GPP TS26.403 5.6.2 "Scalefactor determination"
*/
static av_always_inline int quant(float coef, const float Q)
{
return av_clip((int)(pow(fabsf(coef) * Q, 0.75) + 0.4054), 0, 8191);
}
static inline float get_approximate_quant_error(float *c, int size, int scale_idx)
{
int i;
int q;
float coef, unquant, sum = 0.0f;
const float Q = ff_aac_pow2sf_tab[200 - scale_idx + SCALE_ONE_POS - SCALE_DIV_512];
const float IQ = ff_aac_pow2sf_tab[200 + scale_idx - SCALE_ONE_POS + SCALE_DIV_512];
for(i = 0; i < size; i++){
coef = fabs(c[i]);
q = quant(c[i], Q);
unquant = (q * cbrt(q)) * IQ;
sum += (coef - unquant) * (coef - unquant);
}
return sum;
}
/**
* constants for 3GPP AAC psychoacoustic model
* @{
*/
#define PSY_3GPP_SPREAD_LOW 1.5f // spreading factor for ascending threshold spreading (15 dB/Bark)
#define PSY_3GPP_SPREAD_HI 3.0f // spreading factor for descending threshold spreading (30 dB/Bark)
#define PSY_3GPP_RPEMIN 0.01f
#define PSY_3GPP_RPELEV 2.0f
/**
* @}
*/
@@ -54,24 +83,8 @@
typedef struct Psy3gppBand{
float energy; ///< band energy
float ffac; ///< form factor
float thr; ///< energy threshold
float min_snr; ///< minimal SNR
float thr_quiet; ///< threshold in quiet
}Psy3gppBand;
/**
* single/pair channel context for psychoacoustic model
*/
typedef struct Psy3gppChannel{
Psy3gppBand band[128]; ///< bands information
Psy3gppBand prev_band[128]; ///< bands information from the previous frame
float win_energy; ///< sliding average of channel energy
float iir_state[2]; ///< hi-pass IIR filter state
uint8_t next_grouping; ///< stored grouping scheme for the next frame (in case of 8 short window sequence)
enum WindowSequence next_window_seq; ///< window sequence to be used in the next frame
}Psy3gppChannel;
/**
* psychoacoustic model frame type-dependent coefficients
*/
@@ -82,237 +95,10 @@ typedef struct Psy3gppCoeffs{
float spread_hi [64]; ///< spreading factor for high-to-low threshold spreading in long frame
}Psy3gppCoeffs;
/**
* 3GPP TS26.403-inspired psychoacoustic model specific data
*/
typedef struct Psy3gppContext{
Psy3gppCoeffs psy_coef[2];
Psy3gppChannel *ch;
}Psy3gppContext;
/**
* Calculate Bark value for given line.
*/
static av_cold float calc_bark(float f)
static inline float calc_bark(float f)
{
return 13.3f * atanf(0.00076f * f) + 3.5f * atanf((f / 7500.0f) * (f / 7500.0f));
}
#define ATH_ADD 4
/**
* Calculate ATH value for given frequency.
* Borrowed from Lame.
*/
static av_cold float ath(float f, float add)
{
f /= 1000.0f;
return 3.64 * pow(f, -0.8)
- 6.8 * exp(-0.6 * (f - 3.4) * (f - 3.4))
+ 6.0 * exp(-0.15 * (f - 8.7) * (f - 8.7))
+ (0.6 + 0.04 * add) * 0.001 * f * f * f * f;
}
static av_cold int psy_3gpp_init(FFPsyContext *ctx) {
Psy3gppContext *pctx;
float barks[1024];
int i, j, g, start;
float prev, minscale, minath;
ctx->model_priv_data = av_mallocz(sizeof(Psy3gppContext));
pctx = (Psy3gppContext*) ctx->model_priv_data;
for (i = 0; i < 1024; i++)
barks[i] = calc_bark(i * ctx->avctx->sample_rate / 2048.0);
minath = ath(3410, ATH_ADD);
for (j = 0; j < 2; j++) {
Psy3gppCoeffs *coeffs = &pctx->psy_coef[j];
i = 0;
prev = 0.0;
for (g = 0; g < ctx->num_bands[j]; g++) {
i += ctx->bands[j][g];
coeffs->barks[g] = (barks[i - 1] + prev) / 2.0;
prev = barks[i - 1];
}
for (g = 0; g < ctx->num_bands[j] - 1; g++) {
coeffs->spread_low[g] = pow(10.0, -(coeffs->barks[g+1] - coeffs->barks[g]) * PSY_3GPP_SPREAD_LOW);
coeffs->spread_hi [g] = pow(10.0, -(coeffs->barks[g+1] - coeffs->barks[g]) * PSY_3GPP_SPREAD_HI);
}
start = 0;
for (g = 0; g < ctx->num_bands[j]; g++) {
minscale = ath(ctx->avctx->sample_rate * start / 1024.0, ATH_ADD);
for (i = 1; i < ctx->bands[j][g]; i++)
minscale = FFMIN(minscale, ath(ctx->avctx->sample_rate * (start + i) / 1024.0 / 2.0, ATH_ADD));
coeffs->ath[g] = minscale - minath;
start += ctx->bands[j][g];
}
}
pctx->ch = av_mallocz(sizeof(Psy3gppChannel) * ctx->avctx->channels);
return 0;
}
/**
* IIR filter used in block switching decision
*/
static float iir_filter(int in, float state[2])
{
float ret;
ret = 0.7548f * (in - state[0]) + 0.5095f * state[1];
state[0] = in;
state[1] = ret;
return ret;
}
/**
* window grouping information stored as bits (0 - new group, 1 - group continues)
*/
static const uint8_t window_grouping[9] = {
0xB6, 0x6C, 0xD8, 0xB2, 0x66, 0xC6, 0x96, 0x36, 0x36
};
/**
* Tell encoder which window types to use.
* @see 3GPP TS26.403 5.4.1 "Blockswitching"
*/
static FFPsyWindowInfo psy_3gpp_window(FFPsyContext *ctx,
const int16_t *audio, const int16_t *la,
int channel, int prev_type)
{
int i, j;
int br = ctx->avctx->bit_rate / ctx->avctx->channels;
int attack_ratio = br <= 16000 ? 18 : 10;
Psy3gppContext *pctx = (Psy3gppContext*) ctx->model_priv_data;
Psy3gppChannel *pch = &pctx->ch[channel];
uint8_t grouping = 0;
FFPsyWindowInfo wi;
memset(&wi, 0, sizeof(wi));
if (la) {
float s[8], v;
int switch_to_eight = 0;
float sum = 0.0, sum2 = 0.0;
int attack_n = 0;
for (i = 0; i < 8; i++) {
for (j = 0; j < 128; j++) {
v = iir_filter(audio[(i*128+j)*ctx->avctx->channels], pch->iir_state);
sum += v*v;
}
s[i] = sum;
sum2 += sum;
}
for (i = 0; i < 8; i++) {
if (s[i] > pch->win_energy * attack_ratio) {
attack_n = i + 1;
switch_to_eight = 1;
break;
}
}
pch->win_energy = pch->win_energy*7/8 + sum2/64;
wi.window_type[1] = prev_type;
switch (prev_type) {
case ONLY_LONG_SEQUENCE:
wi.window_type[0] = switch_to_eight ? LONG_START_SEQUENCE : ONLY_LONG_SEQUENCE;
break;
case LONG_START_SEQUENCE:
wi.window_type[0] = EIGHT_SHORT_SEQUENCE;
grouping = pch->next_grouping;
break;
case LONG_STOP_SEQUENCE:
wi.window_type[0] = ONLY_LONG_SEQUENCE;
break;
case EIGHT_SHORT_SEQUENCE:
wi.window_type[0] = switch_to_eight ? EIGHT_SHORT_SEQUENCE : LONG_STOP_SEQUENCE;
grouping = switch_to_eight ? pch->next_grouping : 0;
break;
}
pch->next_grouping = window_grouping[attack_n];
} else {
for (i = 0; i < 3; i++)
wi.window_type[i] = prev_type;
grouping = (prev_type == EIGHT_SHORT_SEQUENCE) ? window_grouping[0] : 0;
}
wi.window_shape = 1;
if (wi.window_type[0] != EIGHT_SHORT_SEQUENCE) {
wi.num_windows = 1;
wi.grouping[0] = 1;
} else {
int lastgrp = 0;
wi.num_windows = 8;
for (i = 0; i < 8; i++) {
if (!((grouping >> i) & 1))
lastgrp = i;
wi.grouping[lastgrp]++;
}
}
return wi;
}
/**
* Calculate band thresholds as suggested in 3GPP TS26.403
*/
static void psy_3gpp_analyze(FFPsyContext *ctx, int channel,
const float *coefs, FFPsyWindowInfo *wi)
{
Psy3gppContext *pctx = (Psy3gppContext*) ctx->model_priv_data;
Psy3gppChannel *pch = &pctx->ch[channel];
int start = 0;
int i, w, g;
const int num_bands = ctx->num_bands[wi->num_windows == 8];
const uint8_t* band_sizes = ctx->bands[wi->num_windows == 8];
Psy3gppCoeffs *coeffs = &pctx->psy_coef[wi->num_windows == 8];
//calculate energies, initial thresholds and related values - 5.4.2 "Threshold Calculation"
for (w = 0; w < wi->num_windows*16; w += 16) {
for (g = 0; g < num_bands; g++) {
Psy3gppBand *band = &pch->band[w+g];
band->energy = 0.0f;
for (i = 0; i < band_sizes[g]; i++)
band->energy += coefs[start+i] * coefs[start+i];
band->energy *= 1.0f / (512*512);
band->thr = band->energy * 0.001258925f;
start += band_sizes[g];
ctx->psy_bands[channel*PSY_MAX_BANDS+w+g].energy = band->energy;
}
}
//modify thresholds - spread, threshold in quiet - 5.4.3 "Spreaded Energy Calculation"
for (w = 0; w < wi->num_windows*16; w += 16) {
Psy3gppBand *band = &pch->band[w];
for (g = 1; g < num_bands; g++)
band[g].thr = FFMAX(band[g].thr, band[g-1].thr * coeffs->spread_low[g-1]);
for (g = num_bands - 2; g >= 0; g--)
band[g].thr = FFMAX(band[g].thr, band[g+1].thr * coeffs->spread_hi [g]);
for (g = 0; g < num_bands; g++) {
band[g].thr_quiet = FFMAX(band[g].thr, coeffs->ath[g]);
if (wi->num_windows != 8 && wi->window_type[1] != EIGHT_SHORT_SEQUENCE)
band[g].thr_quiet = FFMAX(PSY_3GPP_RPEMIN*band[g].thr_quiet,
FFMIN(band[g].thr_quiet,
PSY_3GPP_RPELEV*pch->prev_band[w+g].thr_quiet));
band[g].thr = FFMAX(band[g].thr, band[g].thr_quiet * 0.25);
ctx->psy_bands[channel*PSY_MAX_BANDS+w+g].threshold = band[g].thr;
}
}
memcpy(pch->prev_band, pch->band, sizeof(pch->band));
}
static av_cold void psy_3gpp_end(FFPsyContext *apc)
{
Psy3gppContext *pctx = (Psy3gppContext*) apc->model_priv_data;
av_freep(&pctx->ch);
av_freep(&apc->model_priv_data);
}
const FFPsyModel ff_aac_psy_model =
{
.name = "3GPP TS 26.403-inspired model",
.init = psy_3gpp_init,
.window = psy_3gpp_window,
.analyze = psy_3gpp_analyze,
.end = psy_3gpp_end,
};

File diff suppressed because it is too large Load Diff

View File

@@ -1,49 +0,0 @@
/*
* AAC Spectral Band Replication function declarations
* Copyright (c) 2008-2009 Robert Swain ( rob opendot cl )
* Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* AAC Spectral Band Replication function declarations
* @author Robert Swain ( rob opendot cl )
*/
#ifndef AVCODEC_AACSBR_H
#define AVCODEC_AACSBR_H
#include "get_bits.h"
#include "aac.h"
#include "sbr.h"
/** Initialize SBR. */
av_cold void ff_aac_sbr_init(void);
/** Initialize one SBR context. */
av_cold void ff_aac_sbr_ctx_init(SpectralBandReplication *sbr);
/** Close one SBR context. */
av_cold void ff_aac_sbr_ctx_close(SpectralBandReplication *sbr);
/** Decode one SBR element. */
int ff_decode_sbr_extension(AACContext *ac, SpectralBandReplication *sbr,
GetBitContext *gb, int crc, int cnt, int id_aac);
/** Apply one SBR element to one AAC element. */
void ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
float* L, float *R);
#endif /* AVCODEC_AACSBR_H */

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