Compare commits

..

5 Commits
n3.2 ... n2.3

Author SHA1 Message Date
Timothy Gu
bef4d9bf87 RELEASE_NOTES: update
Signed-off-by: Timothy Gu <timothygu99@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-07-16 02:42:16 +02:00
Michael Niedermayer
bc4f6ae88e Changelog: change "version <next>" to "version 2.3"
Found-by: jamrial
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit faafd1e4f1)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-07-16 02:09:50 +02:00
Michael Niedermayer
2678b25099 doc/APIchanges: update
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 13a72d9b08)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-07-16 02:07:02 +02:00
Michael Niedermayer
e322496054 RELEASE_NOTES: update version numbers
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-07-16 02:01:14 +02:00
Michael Niedermayer
7fa72ff19c update for FFmpeg 2.3
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-07-16 01:45:34 +02:00
5367 changed files with 142453 additions and 516757 deletions

1
.gitattributes vendored
View File

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

63
.gitignore vendored
View File

@@ -1,6 +1,5 @@
*.a
*.o
*.o.*
*.d
*.def
*.dll
@@ -18,9 +17,9 @@
*.so.*
*.swp
*.ver
*-example
*-test
*_g
\#*
.\#*
/.config
/.version
/ffmpeg
@@ -29,7 +28,59 @@
/ffserver
/config.*
/coverage.info
/avversion.h
/doc/*.1
/doc/*.3
/doc/*.html
/doc/*.pod
/doc/config.texi
/doc/avoptions_codec.texi
/doc/avoptions_format.texi
/doc/doxy/html/
/doc/examples/avio_reading
/doc/examples/decoding_encoding
/doc/examples/demuxing_decoding
/doc/examples/filter_audio
/doc/examples/filtering_audio
/doc/examples/filtering_video
/doc/examples/metadata
/doc/examples/muxing
/doc/examples/pc-uninstalled
/doc/examples/remuxing
/doc/examples/resampling_audio
/doc/examples/scaling_video
/doc/examples/transcode_aac
/doc/examples/transcoding
/doc/fate.txt
/doc/print_options
/lcov/
/src
/mapfile
/libavcodec/*_tablegen
/libavcodec/*_tables.c
/libavcodec/*_tables.h
/libavutil/avconfig.h
/libavutil/ffversion.h
/tests/audiogen
/tests/base64
/tests/data/
/tests/rotozoom
/tests/tiny_psnr
/tests/tiny_ssim
/tests/videogen
/tests/vsynth1/
/tools/aviocat
/tools/ffbisect
/tools/bisect.need
/tools/crypto_bench
/tools/cws2fws
/tools/fourcc2pixfmt
/tools/ffescape
/tools/ffeval
/tools/ffhash
/tools/graph2dot
/tools/ismindex
/tools/pktdumper
/tools/probetest
/tools/qt-faststart
/tools/trasher
/tools/seek_print
/tools/uncoded_frame
/tools/zmqsend

View File

@@ -1,26 +0,0 @@
language: c
sudo: false
os:
- linux
- osx
addons:
apt:
packages:
- yasm
- diffutils
compiler:
- clang
- gcc
cache:
directories:
- ffmpeg-samples
before_install:
- if [ "$TRAVIS_OS_NAME" == "osx" ]; then brew update --all; fi
install:
- if [ "$TRAVIS_OS_NAME" == "osx" ]; then brew install yasm; fi
script:
- mkdir -p ffmpeg-samples
- ./configure --samples=ffmpeg-samples --cc=$CC
- make -j 8
- make fate-rsync
- make check -j 8

View File

@@ -1,4 +0,0 @@
# Note to Github users
Patches should be submitted to the [ffmpeg-devel mailing list](https://ffmpeg.org/mailman/listinfo/ffmpeg-devel) using `git format-patch` or `git send-email`. Github pull requests should be avoided because they are not part of our review process and **will be ignored**.
See [https://ffmpeg.org/developer.html#Contributing](https://ffmpeg.org/developer.html#Contributing) for more information.

306
Changelog
View File

@@ -1,301 +1,6 @@
Entries are sorted chronologically from oldest to youngest within each release,
releases are sorted from youngest to oldest.
version 3.2:
- libopenmpt demuxer
- tee protocol
- Changed metadata print option to accept general urls
- Alias muxer for Ogg Video (.ogv)
- VP8 in Ogg muxing
- curves filter doesn't automatically insert points at x=0 and x=1 anymore
- 16-bit support in curves filter and selectivecolor filter
- OpenH264 decoder wrapper
- MediaCodec H.264/HEVC/MPEG-4/VP8/VP9 hwaccel
- True Audio (TTA) muxer
- crystalizer audio filter
- acrusher audio filter
- bitplanenoise video filter
- floating point support in als decoder
- fifo muxer
- maskedclamp filter
- hysteresis filter
- lut2 filter
- yuvtestsrc filter
- CUDA CUVID H.263/VP8/VP9/10 bit HEVC (Dithered) Decoding
- vaguedenoiser filter
- added threads option per filter instance
- weave filter
- gblur filter
- avgblur filter
- sobel and prewitt filter
- MediaCodec HEVC/MPEG-4/VP8/VP9 decoding
- Meridian Lossless Packing (MLP) / TrueHD encoder
- Non-Local Means (nlmeans) denoising filter
- sdl2 output device and ffplay support
- sdl1 output device and sdl1 support removed
- extended mov edit list support
- libfaac encoder removed
- Matroska muxer now writes CRC32 elements by default in all Level 1 elements
- sidedata video and asidedata audio filter
- Changed mapping of rtp MIME type G726 to codec g726le.
version 3.1:
- DXVA2-accelerated HEVC Main10 decoding
- fieldhint filter
- loop video filter and aloop audio filter
- Bob Weaver deinterlacing filter
- firequalizer filter
- datascope filter
- bench and abench filters
- ciescope filter
- protocol blacklisting API
- MediaCodec H264 decoding
- VC-2 HQ RTP payload format (draft v1) depacketizer and packetizer
- VP9 RTP payload format (draft v2) packetizer
- AudioToolbox audio decoders
- AudioToolbox audio encoders
- coreimage filter (GPU based image filtering on OSX)
- libdcadec removed
- bitstream filter for extracting DTS core
- ADPCM IMA DAT4 decoder
- musx demuxer
- aix demuxer
- remap filter
- hash and framehash muxers
- colorspace filter
- hdcd filter
- readvitc filter
- VAAPI-accelerated format conversion and scaling
- libnpp/CUDA-accelerated format conversion and scaling
- Duck TrueMotion 2.0 Real Time decoder
- Wideband Single-bit Data (WSD) demuxer
- VAAPI-accelerated H.264/HEVC/MJPEG encoding
- DTS Express (LBR) decoder
- Generic OpenMAX IL encoder with support for Raspberry Pi
- IFF ANIM demuxer & decoder
- Direct Stream Transfer (DST) decoder
- loudnorm filter
- MTAF demuxer and decoder
- MagicYUV decoder
- OpenExr improvements (tile data and B44/B44A support)
- BitJazz SheerVideo decoder
- CUDA CUVID H264/HEVC decoder
- 10-bit depth support in native utvideo decoder
- libutvideo wrapper removed
- YUY2 Lossless Codec decoder
- VideoToolbox H.264 encoder
version 3.0:
- Common Encryption (CENC) MP4 encoding and decoding support
- DXV decoding
- extrastereo filter
- ocr filter
- alimiter filter
- stereowiden filter
- stereotools filter
- rubberband filter
- tremolo filter
- agate filter
- chromakey filter
- maskedmerge filter
- Screenpresso SPV1 decoding
- chromaprint fingerprinting muxer
- ffplay dynamic volume control
- displace filter
- selectivecolor filter
- extensive native AAC encoder improvements and removal of experimental flag
- ADPCM PSX decoder
- 3dostr, dcstr, fsb, genh, vag, xvag, ads, msf, svag & vpk demuxer
- zscale filter
- wve demuxer
- zero-copy Intel QSV transcoding in ffmpeg
- shuffleframes filter
- SDX2 DPCM decoder
- vibrato filter
- innoHeim/Rsupport Screen Capture Codec decoder
- ADPCM AICA decoder
- Interplay ACM demuxer and audio decoder
- XMA1 & XMA2 decoder
- realtime filter
- anoisesrc audio filter source
- IVR demuxer
- compensationdelay filter
- acompressor filter
- support encoding 16-bit RLE SGI images
- apulsator filter
- sidechaingate audio filter
- mipsdspr1 option has been renamed to mipsdsp
- aemphasis filter
- mips32r5 option has been removed
- mips64r6 option has been removed
- DXVA2-accelerated VP9 decoding
- SOFAlizer: virtual binaural acoustics filter
- VAAPI VP9 hwaccel
- audio high-order multiband parametric equalizer
- automatic bitstream filtering
- showspectrumpic filter
- libstagefright support removed
- spectrumsynth filter
- ahistogram filter
- only seek with the right mouse button in ffplay
- toggle full screen when double-clicking with the left mouse button in ffplay
- afftfilt filter
- convolution filter
- libquvi support removed
- support for dvaudio in wav and avi
- libaacplus and libvo-aacenc support removed
- Cineform HD decoder
- new DCA decoder with full support for DTS-HD extensions
- significant performance improvements in Windows Television (WTV) demuxer
- nnedi deinterlacer
- streamselect video and astreamselect audio filter
- swaprect filter
- metadata video and ametadata audio filter
- SMPTE VC-2 HQ profile support for the Dirac decoder
- SMPTE VC-2 native encoder supporting the HQ profile
version 2.8:
- colorkey video filter
- BFSTM/BCSTM demuxer
- little-endian ADPCM_THP decoder
- Hap decoder and encoder
- DirectDraw Surface image/texture decoder
- ssim filter
- optional new ASF demuxer
- showvolume filter
- Many improvements to the JPEG 2000 decoder
- Go2Meeting decoding support
- adrawgraph audio and drawgraph video filter
- removegrain video filter
- Intel QSV-accelerated MPEG-2 video and HEVC encoding
- Intel QSV-accelerated MPEG-2 video and HEVC decoding
- Intel QSV-accelerated VC-1 video decoding
- libkvazaar HEVC encoder
- erosion, dilation, deflate and inflate video filters
- Dynamic Audio Normalizer as dynaudnorm filter
- Reverse video and areverse audio filter
- Random filter
- deband filter
- AAC fixed-point decoding
- sidechaincompress audio filter
- bitstream filter for converting HEVC from MP4 to Annex B
- acrossfade audio filter
- allyuv and allrgb video sources
- atadenoise video filter
- OS X VideoToolbox support
- aphasemeter filter
- showfreqs filter
- vectorscope filter
- waveform filter
- hstack and vstack filter
- Support DNx100 (1440x1080@8)
- VAAPI hevc hwaccel
- VDPAU hevc hwaccel
- framerate filter
- Switched default encoders for webm to VP9 and Opus
- Removed experimental flag from the JPEG 2000 encoder
version 2.7:
- FFT video filter
- TDSC decoder
- DTS lossless extension (XLL) decoding (not lossless, disabled by default)
- showwavespic filter
- DTS decoding through libdcadec
- Drop support for nvenc API before 5.0
- nvenc HEVC encoder
- Detelecine filter
- Intel QSV-accelerated H.264 encoding
- MMAL-accelerated H.264 decoding
- basic APNG encoder and muxer with default extension "apng"
- unpack DivX-style packed B-frames in MPEG-4 bitstream filter
- WebM Live Chunk Muxer
- nvenc level and tier options
- chorus filter
- Canopus HQ/HQA decoder
- Automatically rotate videos based on metadata in ffmpeg
- improved Quickdraw compatibility
- VP9 high bit-depth and extended colorspaces decoding support
- WebPAnimEncoder API when available for encoding and muxing WebP
- Direct3D11-accelerated decoding
- Support Secure Transport
- Multipart JPEG demuxer
version 2.6:
- nvenc encoder
- 10bit spp filter
- colorlevels filter
- RIFX format for *.wav files
- RTP/mpegts muxer
- non continuous cache protocol support
- tblend filter
- cropdetect support for non 8bpp, absolute (if limit >= 1) and relative (if limit < 1.0) threshold
- Camellia symmetric block cipher
- OpenH264 encoder wrapper
- VOC seeking support
- Closed caption Decoder
- fspp, uspp, pp7 MPlayer postprocessing filters ported to native filters
- showpalette filter
- Twofish symmetric block cipher
- Support DNx100 (960x720@8)
- eq2 filter ported from libmpcodecs as eq filter
- removed libmpcodecs
- Changed default DNxHD colour range in QuickTime .mov derivatives to mpeg range
- ported softpulldown filter from libmpcodecs as repeatfields filter
- dcshift filter
- RTP depacketizer for loss tolerant payload format for MP3 audio (RFC 5219)
- RTP depacketizer for AC3 payload format (RFC 4184)
- palettegen and paletteuse filters
- VP9 RTP payload format (draft 0) experimental depacketizer
- RTP depacketizer for DV (RFC 6469)
- DXVA2-accelerated HEVC decoding
- AAC ELD 480 decoding
- Intel QSV-accelerated H.264 decoding
- DSS SP decoder and DSS demuxer
- Fix stsd atom corruption in DNxHD QuickTimes
- Canopus HQX decoder
- RTP depacketization of T.140 text (RFC 4103)
- Port MIPS optimizations to 64-bit
version 2.5:
- HEVC/H.265 RTP payload format (draft v6) packetizer
- SUP/PGS subtitle demuxer
- ffprobe -show_pixel_formats option
- CAST128 symmetric block cipher, ECB mode
- STL subtitle demuxer and decoder
- libutvideo YUV 4:2:2 10bit support
- XCB-based screen-grabber
- UDP-Lite support (RFC 3828)
- xBR scaling filter
- AVFoundation screen capturing support
- ffserver supports codec private options
- creating DASH compatible fragmented MP4, MPEG-DASH segmenting muxer
- WebP muxer with animated WebP support
- zygoaudio decoding support
- APNG demuxer
- postproc visualization support
version 2.4:
- Icecast protocol
- ported lenscorrection filter from frei0r filter
- large optimizations in dctdnoiz to make it usable
- ICY metadata are now requested by default with the HTTP protocol
- support for using metadata in stream specifiers in fftools
- LZMA compression support in TIFF decoder
- H.261 RTP payload format (RFC 4587) depacketizer and experimental packetizer
- HEVC/H.265 RTP payload format (draft v6) depacketizer
- added codecview filter to visualize information exported by some codecs
- Matroska 3D support thorugh side data
- HTML generation using texi2html is deprecated in favor of makeinfo/texi2any
- silenceremove filter
version 2.3:
- AC3 fixed-point decoding
- shuffleplanes filter
@@ -319,7 +24,7 @@ version 2.3:
- libbs2b-based stereo-to-binaural audio filter
- libx264 reference frames count limiting depending on level
- native Opus decoder
- display matrix export and rotation API
- display matrix export and rotation api
- WebVTT encoder
- showcqt multimedia filter
- zoompan filter
@@ -361,7 +66,6 @@ version 2.2:
- libx265 encoder
- dejudder filter
- Autodetect VDA like all other hardware accelerations
- aliases and defaults for Ogg subtypes (opus, spx)
version 2.1:
@@ -768,7 +472,7 @@ easier to use. The changes are:
all the stream in the first input file, except for the second audio
stream'.
* There is a new option -c (or -codec) for choosing the decoder/encoder to
use, which makes it possible to precisely specify target stream(s) consistently with
use, which allows to precisely specify target stream(s) consistently with
other options. E.g. -c:v lib264 sets the codec for all video streams, -c:a:0
libvorbis sets the codec for the first audio stream and -c copy copies all
the streams without reencoding. Old -vcodec/-acodec/-scodec options are now
@@ -957,8 +661,8 @@ version 0.8:
- showinfo filter added
- SMPTE 302M AES3 audio decoder
- Apple Core Audio Format muxer
- 9 bits and 10 bits per sample support in the H.264 decoder
- 9 bits and 10 bits FFV1 encoding / decoding
- 9bit and 10bit per sample support in the H.264 decoder
- 9bit and 10bit FFV1 encoding / decoding
- split filter added
- select filter added
- sdl output device added
@@ -1251,7 +955,7 @@ version 0.4.9-pre1:
- rate distorted optimal lambda->qp support
- AAC encoding with libfaac
- Sunplus JPEG codec (SP5X) support
- use Lagrange multiplier instead of QP for ratecontrol
- use Lagrange multipler instead of QP for ratecontrol
- Theora/VP3 decoding support
- XA and ADX ADPCM codecs
- export MPEG-2 active display area / pan scan

View File

@@ -1,122 +1,103 @@
# License
#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
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
version 2 or later (GPL v2+). See the file COPYING.GPLv2 for details. None of
these parts are used by default, you have to explicitly pass --enable-gpl to
configure to activate them. In this case, FFmpeg's license changes to GPL v2+.
Specifically, the GPL parts of FFmpeg are:
- libpostproc
- optional x86 optimization in the files
- `libavcodec/x86/flac_dsp_gpl.asm`
- `libavcodec/x86/idct_mmx.c`
- `libavfilter/x86/vf_removegrain.asm`
- the X11 grabber in `libavdevice/x11grab.c`
- the following building and testing tools
- `compat/solaris/make_sunver.pl`
- `doc/t2h.pm`
- `doc/texi2pod.pl`
- `libswresample/swresample-test.c`
- `tests/checkasm/*`
- `tests/tiny_ssim.c`
- libmpcodecs
- optional x86 optimizations in the files
libavcodec/x86/idct_mmx.c
- libutvideo encoding/decoding wrappers in
libavcodec/libutvideo*.cpp
- the X11 grabber in libavdevice/x11grab.c
- the swresample test app in
libswresample/swresample-test.c
- the texi2pod.pl tool
- the following filters in libavfilter:
- `f_ebur128.c`
- `vf_blackframe.c`
- `vf_boxblur.c`
- `vf_colormatrix.c`
- `vf_cover_rect.c`
- `vf_cropdetect.c`
- `vf_delogo.c`
- `vf_eq.c`
- `vf_find_rect.c`
- `vf_fspp.c`
- `vf_geq.c`
- `vf_histeq.c`
- `vf_hqdn3d.c`
- `vf_interlace.c`
- `vf_kerndeint.c`
- `vf_mcdeint.c`
- `vf_mpdecimate.c`
- `vf_owdenoise.c`
- `vf_perspective.c`
- `vf_phase.c`
- `vf_pp.c`
- `vf_pp7.c`
- `vf_pullup.c`
- `vf_repeatfields.c`
- `vf_sab.c`
- `vf_smartblur.c`
- `vf_spp.c`
- `vf_stereo3d.c`
- `vf_super2xsai.c`
- `vf_tinterlace.c`
- `vf_uspp.c`
- `vsrc_mptestsrc.c`
- f_ebur128.c
- vf_blackframe.c
- vf_boxblur.c
- vf_colormatrix.c
- vf_cropdetect.c
- vf_decimate.c
- vf_delogo.c
- vf_geq.c
- vf_histeq.c
- vf_hqdn3d.c
- vf_kerndeint.c
- vf_mcdeint.c
- vf_mp.c
- vf_owdenoise.c
- vf_perspective.c
- vf_phase.c
- vf_pp.c
- vf_pullup.c
- vf_sab.c
- vf_smartblur.c
- vf_spp.c
- vf_stereo3d.c
- vf_super2xsai.c
- vf_tinterlace.c
- vsrc_mptestsrc.c
Should you, for whatever reason, prefer to use version 3 of the (L)GPL, then
the configure parameter `--enable-version3` will activate this licensing option
for you. Read the file `COPYING.LGPLv3` or, if you have enabled GPL parts,
`COPYING.GPLv3` to learn the exact legal terms that apply in this case.
the configure parameter --enable-version3 will activate this licensing option
for you. Read the file COPYING.LGPLv3 or, if you have enabled GPL parts,
COPYING.GPLv3 to learn the exact legal terms that apply in this case.
There are a handful of files under other licensing terms, namely:
* The files `libavcodec/jfdctfst.c`, `libavcodec/jfdctint_template.c` and
`libavcodec/jrevdct.c` are taken from libjpeg, see the top of the files for
* The files libavcodec/jfdctfst.c, libavcodec/jfdctint_template.c and
libavcodec/jrevdct.c are taken from libjpeg, see the top of the files for
licensing details. Specifically note that you must credit the IJG in the
documentation accompanying your program if you only distribute executables.
You must also indicate any changes including additions and deletions to
those three files in the documentation.
* `tests/reference.pnm` is under the expat license.
## External libraries
external libraries
==================
FFmpeg can be combined with a number of external libraries, which sometimes
affect the licensing of binaries resulting from the combination.
### Compatible libraries
compatible libraries
--------------------
The following libraries are under GPL:
- frei0r
- libcdio
- librubberband
- libvidstab
- libx264
- libx265
- libxavs
- libxvid
- frei0r
- libcdio
- libutvideo
- libvidstab
- libx264
- libx265
- libxavs
- libxvid
When combining them with FFmpeg, FFmpeg needs to be licensed as GPL as well by
passing `--enable-gpl` to configure.
passing --enable-gpl to configure.
The OpenCORE and VisualOn libraries are under the Apache License 2.0. That
license is incompatible with the LGPL v2.1 and the GPL v2, but not with
version 3 of those licenses. So to combine these libraries with FFmpeg, the
license version needs to be upgraded by passing `--enable-version3` to configure.
license version needs to be upgraded by passing --enable-version3 to configure.
### Incompatible libraries
incompatible libraries
----------------------
There are certain libraries you can combine with FFmpeg whose licenses are not
compatible with the GPL and/or the LGPL. If you wish to enable these
libraries, even in circumstances that their license may be incompatible, pass
`--enable-nonfree` to configure. But note that if you enable any of these
libraries the resulting binary will be under a complex license mix that is
more restrictive than the LGPL and that may result in additional obligations.
It is possible that these restrictions cause the resulting binary to be
unredistributable.
The Fraunhofer FDK AAC and OpenSSL libraries are under licenses which are
incompatible with the GPLv2 and v3. To the best of our knowledge, they are
compatible with the LGPL.
The NVENC library, while its header file is licensed under the compatible MIT
license, requires a proprietary binary blob at run time, and is deemed to be
incompatible with the GPL. We are not certain if it is compatible with the
LGPL, but we require `--enable-nonfree` even with LGPL configurations in case
it is not.
The Fraunhofer AAC library, FAAC and aacplus are under licenses which
are incompatible with the GPLv2 and v3. We do not know for certain if their
licenses are compatible with the LGPL.
If you wish to enable these libraries, pass --enable-nonfree to configure.
But note that if you enable any of these libraries the resulting binary will
be under a complex license mix that is more restrictive than the LGPL and that
may result in additional obligations. It is possible that these
restrictions cause the resulting binary to be unredistributeable.

View File

@@ -14,6 +14,7 @@ patches and related discussions.
Project Leader
==============
Michael Niedermayer
final design decisions
@@ -42,8 +43,9 @@ QuickTime faststart:
Miscellaneous Areas
===================
documentation Stefano Sabatini, Mike Melanson, Timothy Gu, Lou Logan
project server Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser, Nikolay Aleksandrov
documentation Stefano Sabatini, Mike Melanson, Timothy Gu
build system (configure, makefiles) Diego Biurrun, Mans Rullgard
project server Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser
presets Robert Swain
metadata subsystem Aurelien Jacobs
release management Michael Niedermayer
@@ -52,12 +54,12 @@ release management Michael Niedermayer
Communication
=============
website Deby Barbara Lepage
website Robert Swain
fate.ffmpeg.org Timothy Gu
Trac bug tracker Alexander Strasser, Michael Niedermayer, Carl Eugen Hoyos, Lou Logan
mailing lists Baptiste Coudurier, Lou Logan
mailing lists Michael Niedermayer, Baptiste Coudurier, Lou Logan
Google+ Paul B Mahol, Michael Niedermayer, Alexander Strasser
Twitter Lou Logan, Reynaldo H. Verdejo Pinochet
Twitter Lou Logan
Launchpad Timothy Gu
@@ -70,7 +72,6 @@ Internal Interfaces:
libavutil/common.h Michael Niedermayer
Other:
aes_ctr.c, aes_ctr.h Eran Kornblau
bprint Nicolas George
bswap.h
des Reimar Doeffinger
@@ -78,7 +79,6 @@ Other:
eval.c, eval.h Michael Niedermayer
float_dsp Loren Merritt
hash Reimar Doeffinger
hwcontext_cuda* Timo Rothenpieler
intfloat* Michael Niedermayer
integer.c, integer.h Michael Niedermayer
lzo Reimar Doeffinger
@@ -89,6 +89,7 @@ Other:
rational.c, rational.h Michael Niedermayer
rc4 Reimar Doeffinger
ripemd.c, ripemd.h James Almer
timecode Clément Bœsch
libavcodec
@@ -116,6 +117,8 @@ Generic Parts:
faandct.c, faandct.h Michael Niedermayer
Golomb coding:
golomb.c, golomb.h Michael Niedermayer
LPC:
lpc.c, lpc.h Justin Ruggles
motion estimation:
motion* Michael Niedermayer
rate control:
@@ -135,38 +138,43 @@ Codecs:
4xm.c Michael Niedermayer
8bps.c Roberto Togni
8svx.c Jaikrishnan Menon
aacenc*, aaccoder.c Rostislav Pehlivanov
aasc.c Kostya Shishkov
ac3* Justin Ruggles
alacenc.c Jaikrishnan Menon
alsdec.c Thilo Borgmann, Umair Khan
alsdec.c Thilo Borgmann
apedec.c Kostya Shishkov
ass* Aurelien Jacobs
asv* Michael Niedermayer
atrac3* Benjamin Larsson
atrac3plus* Maxim Poliakovski
audiotoolbox* Rodger Combs
bgmc.c, bgmc.h Thilo Borgmann
bink.c Kostya Shishkov
binkaudio.c Peter Ross
bmp.c Mans Rullgard, Kostya Shishkov
cavs* Stefan Gehrer
cdxl.c Paul B Mahol
celp_filters.* Vitor Sessak
cinepak.c Roberto Togni
cinepakenc.c Rl / Aetey G.T. AB
ccaption_dec.c Anshul Maheshwari, Aman Gupta
cljr Alex Beregszaszi
cllc.c Derek Buitenhuis
cook.c, cookdata.h Benjamin Larsson
cpia.c Stephan Hilb
crystalhd.c Philip Langdale
cscd.c Reimar Doeffinger
cuvid.c Timo Rothenpieler
dirac* Rostislav Pehlivanov
dca.c Kostya Shishkov, Benjamin Larsson
dnxhd* Baptiste Coudurier
dpcm.c Mike Melanson
dss_sp.c Oleksij Rempel
dv.c Roman Shaposhnik
dvbsubdec.c Anshul Maheshwari
dxa.c Kostya Shishkov
eacmv*, eaidct*, eat* Peter Ross
evrc* Paul B Mahol
exif.c, exif.h Thilo Borgmann
ffv1* Michael Niedermayer
ffwavesynth.c Nicolas George
fifo.c Jan Sebechlebsky
fic.c Derek Buitenhuis
flac* Justin Ruggles
flashsv* Benjamin Larsson
flicvideo.c Mike Melanson
g722.c Martin Storsjo
g726.c Roman Shaposhnik
@@ -174,47 +182,58 @@ Codecs:
h261* Michael Niedermayer
h263* Michael Niedermayer
h264* Loren Merritt, Michael Niedermayer
hap* Tom Butterworth
huffyuv* Michael Niedermayer, Christophe Gisquet
idcinvideo.c Mike Melanson
imc* Benjamin Larsson
indeo2* Kostya Shishkov
indeo5* Kostya Shishkov
interplayvideo.c Mike Melanson
jni*, ffjni* Matthieu Bouron
ivi* Kostya Shishkov
jacosub* Clément Bœsch
jpeg2000* Nicolas Bertrand
jpeg_ls.c Kostya Shishkov
jvdec.c Peter Ross
kmvc.c Kostya Shishkov
lcl*.c Roberto Togni, Reimar Doeffinger
libcelt_dec.c Nicolas George
libdirac* David Conrad
libgsm.c Michel Bardiaux
libkvazaar.c Arttu Ylä-Outinen
libopenjpeg.c Jaikrishnan Menon
libopenjpegenc.c Michael Bradshaw
libschroedinger* David Conrad
libspeexdec.c Justin Ruggles
libtheoraenc.c David Conrad
libutvideo* Derek Buitenhuis
libvorbis.c David Conrad
libvpx* James Zern
libx264.c Mans Rullgard, Jason Garrett-Glaser
libx265.c Derek Buitenhuis
libxavs.c Stefan Gehrer
libzvbi-teletextdec.c Marton Balint
loco.c Kostya Shishkov
lzo.h, lzo.c Reimar Doeffinger
mdec.c Michael Niedermayer
mimic.c Ramiro Polla
mjpeg*.c Michael Niedermayer
mlp* Ramiro Polla, Jai Luthra
mlp* Ramiro Polla
mmvideo.c Peter Ross
mpc* Kostya Shishkov
mpeg12.c, mpeg12data.h Michael Niedermayer
mpegvideo.c, mpegvideo.h Michael Niedermayer
mqc* Nicolas Bertrand
msmpeg4.c, msmpeg4data.h Michael Niedermayer
msrle.c Mike Melanson
msvideo1.c Mike Melanson
nellymoserdec.c Benjamin Larsson
nuv.c Reimar Doeffinger
nvenc* Timo Rothenpieler
paf.* Paul B Mahol
pcx.c Ivo van Poorten
pgssubdec.c Reimar Doeffinger
ptx.c Ivo van Poorten
qcelp* Reynaldo H. Verdejo Pinochet
qdm2.c, qdm2data.h Roberto Togni
qsv* Ivan Uskov
qdm2.c, qdm2data.h Roberto Togni, Benjamin Larsson
qdrw.c Kostya Shishkov
qpeg.c Kostya Shishkov
qtrle.c Mike Melanson
ra144.c, ra144.h, ra288.c, ra288.h Roberto Togni
resample2.c Michael Niedermayer
@@ -222,8 +241,10 @@ Codecs:
rpza.c Roberto Togni
rtjpeg.c, rtjpeg.h Reimar Doeffinger
rv10.c Michael Niedermayer
rv4* Christophe Gisquet
rv3* Kostya Shishkov
rv4* Kostya Shishkov, Christophe Gisquet
s3tc* Ivo van Poorten
smacker.c Kostya Shishkov
smc.c Mike Melanson
smvjpegdec.c Ash Hughes
snow* Michael Niedermayer, Loren Merritt
@@ -232,41 +253,53 @@ Codecs:
sunrast.c Ivo van Poorten
svq3.c Michael Niedermayer
tak* Paul B Mahol
targa.c Kostya Shishkov
tiff.c Kostya Shishkov
truemotion1* Mike Melanson
truemotion2* Kostya Shishkov
truespeech.c Kostya Shishkov
tscc.c Kostya Shishkov
tta.c Alex Beregszaszi, Jaikrishnan Menon
ttaenc.c Paul B Mahol
txd.c Ivo van Poorten
vc1* Christophe Gisquet
vc2* Rostislav Pehlivanov
ulti* Kostya Shishkov
v410*.c Derek Buitenhuis
vb.c Kostya Shishkov
vble.c Derek Buitenhuis
vc1* Kostya Shishkov, Christophe Gisquet
vcr1.c Michael Niedermayer
vda_h264_dec.c Xidorn Quan
videotoolboxenc.c Rick Kern
vima.c Paul B Mahol
vmnc.c Kostya Shishkov
vorbisdec.c Denes Balatoni, David Conrad
vorbisenc.c Oded Shimon
vp3* Mike Melanson
vp5 Aurelien Jacobs
vp6 Aurelien Jacobs
vp8 David Conrad, Ronald Bultje
vp9 Ronald Bultje
vp8 David Conrad, Jason Garrett-Glaser, Ronald Bultje
vp9 Ronald Bultje, Clément Bœsch
vqavideo.c Mike Melanson
wavpack.c Kostya Shishkov
wmaprodec.c Sascha Sommer
wmavoice.c Ronald S. Bultje
wmv2.c Michael Niedermayer
wnv1.c Kostya Shishkov
xan.c Mike Melanson
xbm* Paul B Mahol
xface Stefano Sabatini
xl.c Kostya Shishkov
xvmc.c Ivan Kalvachev
xwd* Paul B Mahol
zerocodec.c Derek Buitenhuis
zmbv* Kostya Shishkov
Hardware acceleration:
crystalhd.c Philip Langdale
dxva2* Hendrik Leppkes, Laurent Aimar
mediacodec* Matthieu Bouron
dxva2* Laurent Aimar
libstagefright.cpp Mohamed Naufal
vaapi* Gwenole Beauchesne
vaapi_encode* Mark Thompson
vdpau* Philip Langdale, Carl Eugen Hoyos
videotoolbox* Rick Kern
vda* Sebastien Zwickert
vdpau* Carl Eugen Hoyos
libavdevice
@@ -276,7 +309,6 @@ libavdevice
avfoundation.m Thilo Borgmann
decklink* Deti Fliegl
dshow.c Roger Pack (CC rogerdpack@gmail.com)
fbdev_enc.c Lukasz Marek
gdigrab.c Roger Pack (CC rogerdpack@gmail.com)
@@ -287,8 +319,7 @@ libavdevice
pulse_audio_enc.c Lukasz Marek
qtkit.m Thilo Borgmann
sdl Stefano Sabatini
sdl2.c Josh de Kock
v4l2.c Giorgio Vazzana
v4l2.c Luca Abeni
vfwcap.c Ramiro Polla
xv.c Lukasz Marek
@@ -298,10 +329,7 @@ libavfilter
Generic parts:
graphdump.c Nicolas George
motion_estimation.c Davinder Singh
Filters:
f_drawgraph.c Paul B Mahol
af_adelay.c Paul B Mahol
af_aecho.c Paul B Mahol
af_afade.c Paul B Mahol
@@ -309,51 +337,30 @@ Filters:
af_aphaser.c Paul B Mahol
af_aresample.c Michael Niedermayer
af_astats.c Paul B Mahol
af_astreamsync.c Nicolas George
af_atempo.c Pavel Koshevoy
af_biquads.c Paul B Mahol
af_chorus.c Paul B Mahol
af_compand.c Paul B Mahol
af_firequalizer.c Muhammad Faiz
af_hdcd.c Burt P.
af_ladspa.c Paul B Mahol
af_loudnorm.c Kyle Swanson
af_pan.c Nicolas George
af_sidechaincompress.c Paul B Mahol
af_silenceremove.c Paul B Mahol
avf_aphasemeter.c Paul B Mahol
avf_avectorscope.c Paul B Mahol
avf_showcqt.c Muhammad Faiz
vf_blend.c Paul B Mahol
vf_chromakey.c Timo Rothenpieler
vf_colorchannelmixer.c Paul B Mahol
vf_colorbalance.c Paul B Mahol
vf_colorkey.c Timo Rothenpieler
vf_colorlevels.c Paul B Mahol
vf_coreimage.m Thilo Borgmann
vf_deband.c Paul B Mahol
vf_dejudder.c Nicholas Robbins
vf_delogo.c Jean Delvare (CC <jdelvare@suse.com>)
vf_delogo.c Jean Delvare (CC <khali@linux-fr.org>)
vf_drawbox.c/drawgrid Andrey Utkin
vf_extractplanes.c Paul B Mahol
vf_histogram.c Paul B Mahol
vf_hqx.c Clément Bœsch
vf_idet.c Pascal Massimino
vf_il.c Paul B Mahol
vf_lenscorrection.c Daniel Oberhoff
vf_mergeplanes.c Paul B Mahol
vf_mestimate.c Davinder Singh
vf_minterpolate.c Davinder Singh
vf_neighbor.c Paul B Mahol
vf_psnr.c Paul B Mahol
vf_random.c Paul B Mahol
vf_readvitc.c Tobias Rapp (CC t.rapp at noa-archive dot com)
vf_scale.c Michael Niedermayer
vf_separatefields.c Paul B Mahol
vf_ssim.c Paul B Mahol
vf_stereo3d.c Paul B Mahol
vf_telecine.c Paul B Mahol
vf_yadif.c Michael Niedermayer
vf_zoompan.c Paul B Mahol
Sources:
vsrc_mandelbrot.c Michael Niedermayer
@@ -366,22 +373,20 @@ Generic parts:
libavformat/avformat.h Michael Niedermayer
Utility Code:
libavformat/utils.c Michael Niedermayer
Text Subtitles Clément Bœsch
Muxers/Demuxers:
4xm.c Mike Melanson
aadec.c Vesselin Bontchev (vesselin.bontchev at yandex dot com)
adtsenc.c Robert Swain
afc.c Paul B Mahol
aiffdec.c Baptiste Coudurier, Matthieu Bouron
aiffenc.c Baptiste Coudurier, Matthieu Bouron
apngdec.c Benoit Fouet
ape.c Kostya Shishkov
ass* Aurelien Jacobs
astdec.c Paul B Mahol
astenc.c James Almer
avi* Michael Niedermayer
avisynth.c Stephen Hutchinson
avisynth.c AvxSynth Team (avxsynth.testing at gmail dot com)
avr.c Paul B Mahol
bink.c Peter Ross
brstm.c Paul B Mahol
@@ -389,18 +394,18 @@ Muxers/Demuxers:
cdxl.c Paul B Mahol
crc.c Michael Niedermayer
daud.c Reimar Doeffinger
dss.c Oleksij Rempel
dtshddec.c Paul B Mahol
dv.c Roman Shaposhnik
dxa.c Kostya Shishkov
electronicarts.c Peter Ross
epafdec.c Paul B Mahol
ffm* Baptiste Coudurier
flac* Justin Ruggles
flic.c Mike Melanson
flvdec.c, flvenc.c Michael Niedermayer
gxf.c Reimar Doeffinger
gxfenc.c Baptiste Coudurier
hls.c Anssi Hannula
hls encryption (hlsenc.c) Christian Suloway, Steven Liu
idcin.c Mike Melanson
idroqdec.c Mike Melanson
iff.c Jaikrishnan Menon
@@ -408,10 +413,10 @@ Muxers/Demuxers:
ipmovie.c Mike Melanson
ircam* Paul B Mahol
iss.c Stefan Gehrer
jacosub* Clément Bœsch
jvdec.c Peter Ross
libmodplug.c Clément Bœsch
libnut.c Oded Shimon
libopenmpt.c Josh de Kock
lmlm4.c Ivo van Poorten
lvfdec.c Paul B Mahol
lxfdec.c Tomas Härdin
@@ -423,9 +428,9 @@ Muxers/Demuxers:
mgsts.c Paul B Mahol
microdvd* Aurelien Jacobs
mm.c Peter Ross
mov.c Baptiste Coudurier
mov.c Michael Niedermayer, Baptiste Coudurier
movenc.c Baptiste Coudurier, Matthieu Bouron
movenccenc.c Eran Kornblau
mpc.c Kostya Shishkov
mpeg.c Michael Niedermayer
mpegenc.c Michael Niedermayer
mpegts.c Marton Balint
@@ -441,7 +446,6 @@ Muxers/Demuxers:
oggdec.c, oggdec.h David Conrad
oggenc.c Baptiste Coudurier
oggparse*.c David Conrad
oggparsedaala* Rostislav Pehlivanov
oma.c Maxim Poliakovski
paf.c Paul B Mahol
psxstr.c Mike Melanson
@@ -451,21 +455,17 @@ Muxers/Demuxers:
raw.c Michael Niedermayer
rdt.c Ronald S. Bultje
rl2.c Sascha Sommer
rmdec.c, rmenc.c Ronald S. Bultje
rmdec.c, rmenc.c Ronald S. Bultje, Kostya Shishkov
rtmp* Kostya Shishkov
rtp.c, rtpenc.c Martin Storsjo
rtpdec_ac3.* Gilles Chanteperdrix
rtpdec_dv.* Thomas Volkert
rtpdec_h261.*, rtpenc_h261.* Thomas Volkert
rtpdec_hevc.*, rtpenc_hevc.* Thomas Volkert
rtpdec_mpa_robust.* Gilles Chanteperdrix
rtpdec_asf.* Ronald S. Bultje
rtpdec_vc2hq.*, rtpenc_vc2hq.* Thomas Volkert
rtpdec_vp9.c Thomas Volkert
rtpenc_mpv.*, rtpenc_aac.* Martin Storsjo
rtsp.c Luca Barbato
sbgdec.c Nicolas George
sdp.c Martin Storsjo
segafilm.c Mike Melanson
segment.c Stefano Sabatini
siff.c Kostya Shishkov
smacker.c Kostya Shishkov
smjpeg* Paul B Mahol
spdif* Anssi Hannula
srtdec.c Aurelien Jacobs
@@ -480,17 +480,16 @@ Muxers/Demuxers:
webvtt* Matthew J Heaney
westwood.c Mike Melanson
wtv.c Peter Ross
wv.c Kostya Shishkov
wvenc.c Paul B Mahol
Protocols:
async.c Zhang Rui
bluray.c Petri Hintukainen
ftp.c Lukasz Marek
http.c Ronald S. Bultje
libssh.c Lukasz Marek
mms*.c Ronald S. Bultje
udp.c Luca Abeni
icecast.c Marvin Scholz
libswresample
@@ -510,26 +509,28 @@ Resamplers:
Operating systems / CPU architectures
=====================================
Alpha Falk Hueffner
MIPS Nedeljko Babic
Alpha Mans Rullgard, Falk Hueffner
ARM Mans Rullgard
AVR32 Mans Rullgard
MIPS Mans Rullgard, Nedeljko Babic
Mac OS X / PowerPC Romain Dolbeau, Guillaume Poirier
Amiga / PowerPC Colin Ward
Linux / PowerPC Luca Barbato
Windows MinGW Alex Beregszaszi, Ramiro Polla
Windows Cygwin Victor Paesa
Windows MSVC Matthew Oliver, Hendrik Leppkes
Windows MSVC Matthew Oliver
Windows ICL Matthew Oliver
ADI/Blackfin DSP Marc Hoffman
Sparc Roman Shaposhnik
OS/2 KO Myung-Hun
x86 Michael Niedermayer
Releases
========
2.8 Michael Niedermayer
2.7 Michael Niedermayer
2.6 Michael Niedermayer
2.5 Michael Niedermayer
2.2 Michael Niedermayer
2.1 Michael Niedermayer
1.2 Michael Niedermayer
If you want to maintain an older release, please contact us
@@ -539,6 +540,7 @@ GnuPG Fingerprints of maintainers and contributors
Alexander Strasser 1C96 78B7 83CB 8AA7 9AF5 D1EB A7D8 A57B A876 E58F
Anssi Hannula 1A92 FF42 2DD9 8D2E 8AF7 65A9 4278 C520 513D F3CB
Anton Khirnov 6D0C 6625 56F8 65D1 E5F5 814B B50A 1241 C067 07AB
Ash Hughes 694D 43D2 D180 C7C7 6421 ABD3 A641 D0B7 623D 6029
Attila Kinali 11F0 F9A6 A1D2 11F6 C745 D10C 6520 BCDD F2DF E765
Baptiste Coudurier 8D77 134D 20CC 9220 201F C5DB 0AC9 325C 5C1A BAAA
@@ -546,19 +548,19 @@ Ben Littler 3EE3 3723 E560 3214 A8CD 4DEB 2CDB FCE7 768C 8D2C
Benoit Fouet B22A 4F4F 43EF 636B BB66 FCDC 0023 AE1E 2985 49C8
Clément Bœsch 52D0 3A82 D445 F194 DB8B 2B16 87EE 2CB8 F4B8 FCF9
Daniel Verkamp 78A6 07ED 782C 653E C628 B8B9 F0EB 8DD8 2F0E 21C7
Diego Biurrun 8227 1E31 B6D9 4994 7427 E220 9CAE D6CC 4757 FCC5
FFmpeg release signing key FCF9 86EA 15E6 E293 A564 4F10 B432 2F04 D676 58D8
Ganesh Ajjanagadde C96A 848E 97C3 CEA2 AB72 5CE4 45F9 6A2D 3C36 FB1B
Gwenole Beauchesne 2E63 B3A6 3E44 37E2 017D 2704 53C7 6266 B153 99C4
Jaikrishnan Menon 61A1 F09F 01C9 2D45 78E1 C862 25DC 8831 AF70 D368
Jean Delvare 7CA6 9F44 60F1 BDC4 1FD2 C858 A552 6B9B B3CD 4E6A
Justin Ruggles 3136 ECC0 C10D 6C04 5F43 CA29 FCBE CD2A 3787 1EBF
Loren Merritt ABD9 08F4 C920 3F65 D8BE 35D7 1540 DAA7 060F 56DE
Lou Logan 7D68 DC73 CBEF EABB 671A B6CF 621C 2E28 82F8 DC3A
Luca Barbato 6677 4209 213C 8843 5B67 29E7 E84C 78C2 84E9 0E34
Michael Niedermayer 9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
Nicolas George 24CE 01CE 9ACC 5CEB 74D8 8D9D B063 D997 36E5 4C93
Nikolay Aleksandrov 8978 1D8C FB71 588E 4B27 EAA8 C4F0 B5FC E011 13B1
Panagiotis Issaris 6571 13A3 33D9 3726 F728 AA98 F643 B12E ECF3 E029
Peter Ross A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B
Philip Langdale 5DC5 8D66 5FBA 3A43 18EC 045E F8D6 B194 6A75 682E
Reimar Doeffinger C61D 16E5 9E2C D10C 8958 38A4 0899 A2B9 06D4 D9C7
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

View File

@@ -4,7 +4,6 @@ include config.mak
vpath %.c $(SRC_PATH)
vpath %.cpp $(SRC_PATH)
vpath %.h $(SRC_PATH)
vpath %.inc $(SRC_PATH)
vpath %.m $(SRC_PATH)
vpath %.S $(SRC_PATH)
vpath %.asm $(SRC_PATH)
@@ -30,18 +29,11 @@ $(foreach prog,$(AVBASENAMES),$(eval OBJS-$(prog) += cmdutils.o))
$(foreach prog,$(AVBASENAMES),$(eval OBJS-$(prog)-$(CONFIG_OPENCL) += cmdutils_opencl.o))
OBJS-ffmpeg += ffmpeg_opt.o ffmpeg_filter.o
OBJS-ffmpeg-$(CONFIG_VIDEOTOOLBOX) += ffmpeg_videotoolbox.o
OBJS-ffmpeg-$(CONFIG_LIBMFX) += ffmpeg_qsv.o
OBJS-ffmpeg-$(CONFIG_VAAPI) += ffmpeg_vaapi.o
ifndef CONFIG_VIDEOTOOLBOX
OBJS-ffmpeg-$(CONFIG_VDA) += ffmpeg_videotoolbox.o
endif
OBJS-ffmpeg-$(CONFIG_CUVID) += ffmpeg_cuvid.o
OBJS-ffmpeg-$(HAVE_DXVA2_LIB) += ffmpeg_dxva2.o
OBJS-ffmpeg-$(HAVE_VDPAU_X11) += ffmpeg_vdpau.o
OBJS-ffserver += ffserver_config.o
OBJS-ffmpeg-$(HAVE_DXVA2_LIB) += ffmpeg_dxva2.o
OBJS-ffmpeg-$(CONFIG_VDA) += ffmpeg_vda.o
TESTTOOLS = audiogen videogen rotozoom tiny_psnr tiny_ssim base64 audiomatch
TESTTOOLS = audiogen videogen rotozoom tiny_psnr tiny_ssim base64
HOSTPROGS := $(TESTTOOLS:%=tests/%) doc/print_options
TOOLS = qt-faststart trasher uncoded_frame
TOOLS-$(CONFIG_ZLIB) += cws2fws
@@ -61,19 +53,17 @@ FFLIBS := avutil
DATA_FILES := $(wildcard $(SRC_PATH)/presets/*.ffpreset) $(SRC_PATH)/doc/ffprobe.xsd
EXAMPLES_FILES := $(wildcard $(SRC_PATH)/doc/examples/*.c) $(SRC_PATH)/doc/examples/Makefile $(SRC_PATH)/doc/examples/README
SKIPHEADERS = cmdutils_common_opts.h \
compat/w32pthreads.h
SKIPHEADERS = cmdutils_common_opts.h compat/w32pthreads.h
include $(SRC_PATH)/common.mak
FF_EXTRALIBS := $(FFEXTRALIBS)
FF_DEP_LIBS := $(DEP_LIBS)
FF_STATIC_DEP_LIBS := $(STATIC_DEP_LIBS)
all: $(AVPROGS)
$(TOOLS): %$(EXESUF): %.o $(EXEOBJS)
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS)
$(LD) $(LDFLAGS) $(LD_O) $^ $(ELIBS)
tools/cws2fws$(EXESUF): ELIBS = $(ZLIB)
tools/uncoded_frame$(EXESUF): $(FF_DEP_LIBS)
@@ -89,8 +79,8 @@ SUBDIR_VARS := CLEANFILES EXAMPLES FFLIBS HOSTPROGS TESTPROGS TOOLS \
HEADERS ARCH_HEADERS BUILT_HEADERS SKIPHEADERS \
ARMV5TE-OBJS ARMV6-OBJS ARMV8-OBJS VFP-OBJS NEON-OBJS \
ALTIVEC-OBJS MMX-OBJS YASM-OBJS \
MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSP-OBJS MSA-OBJS \
MMI-OBJS OBJS SLIBOBJS HOSTOBJS TESTOBJS
MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSPR1-OBJS MIPS32R2-OBJS \
OBJS SLIBOBJS HOSTOBJS TESTOBJS
define RESET
$(1) :=
@@ -102,7 +92,6 @@ $(foreach V,$(SUBDIR_VARS),$(eval $(call RESET,$(V))))
SUBDIR := $(1)/
include $(SRC_PATH)/$(1)/Makefile
-include $(SRC_PATH)/$(1)/$(ARCH)/Makefile
-include $(SRC_PATH)/$(1)/$(INTRINSICS)/Makefile
include $(SRC_PATH)/library.mak
endef
@@ -121,14 +110,14 @@ endef
$(foreach P,$(PROGS),$(eval $(call DOPROG,$(P:$(PROGSSUF)$(EXESUF)=))))
ffprobe.o cmdutils.o libavcodec/utils.o libavformat/utils.o libavdevice/avdevice.o libavfilter/avfilter.o libavutil/utils.o libpostproc/postprocess.o libswresample/swresample.o libswscale/utils.o : libavutil/ffversion.h
ffprobe.o cmdutils.o : libavutil/ffversion.h
$(PROGS): %$(PROGSSUF)$(EXESUF): %$(PROGSSUF)_g$(EXESUF)
$(CP) $< $@
$(STRIP) $@
%$(PROGSSUF)_g$(EXESUF): %.o $(FF_DEP_LIBS)
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
$(LD) $(LDFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
OBJDIRS += tools
@@ -180,15 +169,11 @@ clean::
$(RM) $(CLEANSUFFIXES)
$(RM) $(CLEANSUFFIXES:%=tools/%)
$(RM) -r coverage-html
$(RM) -rf coverage.info coverage.info.in lcov
$(RM) -rf coverage.info lcov
distclean::
$(RM) $(DISTCLEANSUFFIXES)
$(RM) config.* .config libavutil/avconfig.h .version mapfile avversion.h version.h libavutil/ffversion.h libavcodec/codec_names.h libavcodec/bsf_list.c libavformat/protocol_list.c
ifeq ($(SRC_LINK),src)
$(RM) src
endif
$(RM) -rf doc/examples/pc-uninstalled
$(RM) config.* .config libavutil/avconfig.h .version version.h libavutil/ffversion.h libavcodec/codec_names.h
config:
$(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION)

View File

@@ -1,48 +1,19 @@
FFmpeg README
=============
FFmpeg is a collection of libraries and tools to process multimedia content
such as audio, video, subtitles and related metadata.
1) Documentation
----------------
## Libraries
* Read the documentation in the doc/ directory in git.
* `libavcodec` provides implementation of a wider range of codecs.
* `libavformat` implements streaming protocols, container formats and basic I/O access.
* `libavutil` includes hashers, decompressors and miscellaneous utility functions.
* `libavfilter` provides a mean to alter decoded Audio and Video through chain of filters.
* `libavdevice` provides an abstraction to access capture and playback devices.
* `libswresample` implements audio mixing and resampling routines.
* `libswscale` implements color conversion and scaling routines.
You can also view it online at http://ffmpeg.org/documentation.html
## Tools
2) Licensing
------------
* [ffmpeg](https://ffmpeg.org/ffmpeg.html) is a command line toolbox to
manipulate, convert and stream multimedia content.
* [ffplay](https://ffmpeg.org/ffplay.html) is a minimalistic multimedia player.
* [ffprobe](https://ffmpeg.org/ffprobe.html) is a simple analysis tool to inspect
multimedia content.
* [ffserver](https://ffmpeg.org/ffserver.html) is a multimedia streaming server
for live broadcasts.
* Additional small tools such as `aviocat`, `ismindex` and `qt-faststart`.
* See the LICENSE file.
## Documentation
3) Build and Install
--------------------
The offline documentation is available in the **doc/** directory.
The online documentation is available in the main [website](https://ffmpeg.org)
and in the [wiki](https://trac.ffmpeg.org).
### Examples
Coding examples are available in the **doc/examples** directory.
## License
FFmpeg codebase is mainly LGPL-licensed with optional components licensed under
GPL. Please refer to the LICENSE file for detailed information.
## Contributing
Patches should be submitted to the ffmpeg-devel mailing list using
`git format-patch` or `git send-email`. Github pull requests should be
avoided because they are not part of our review process and will be ignored.
* See the INSTALL file.

View File

@@ -1 +1 @@
3.2
2.3

View File

@@ -1,15 +1,177 @@
┌───────────────────────────────────────────┐
│ RELEASE NOTES for FFmpeg 2.3 "Mandelbrot" │
└───────────────────────────────────────────┘
┌────────────────────────────────────────┐
│ RELEASE NOTES for FFmpeg 3.2 "Hypatia" │
└────────────────────────────────────────┘
The FFmpeg Project proudly presents FFmpeg 2.3 "Mandelbrot", a major
release with all the great features committed during the three-month period
since the release of FFmpeg 2.2.
The FFmpeg Project proudly presents FFmpeg 3.2 "Hypatia", about 4
months after the release of FFmpeg 3.1.
In this release, there are lots of internal overhauls that make FFmpeg a
more accessible project for new developers. Many important new
optimizations and features like QTKit and AVFoundation input devices are
committed. Contributions done by Libav such as a new native Opus decoder
are also merged.
A complete Changelog is available at the root of the project, and the
complete Git history on http://source.ffmpeg.org.
Because of the increasing difficulty to maintain and lack of maintainers,
we are very sorry to say that we have removed all Blackfin and SPARC
architecture assembly optimizations with the cleanups done. If you are
interested in maintaining optimization for these two architecture, feel
free to contact us and we will restore the code!
We hope you will like this release as much as we enjoyed working on it, and
as usual, if you have any questions about it, or any FFmpeg related topic,
feel free to join us on the #ffmpeg IRC channel (on irc.freenode.net) or ask
on the mailing-lists.
Oh, and since this release, this modern-looking release note is provided in
addition to the old-style Changelog file, to make it easier for you to
focus on the most important features in this release.
Enjoy!
┌────────────────────────────┐
│ * API Information │
└────────────────────────────┘
FFmpeg 2.3 is completely source-compatible to the FFmpeg 2.2 series. There
are however some API deprecations that you need to take care of. Use `git
diff n2.2 n2.3 doc/APIchanges` to show the list of added and deprecated
APIs. FFmpeg 2.3 includes the following library versions:
• libavutil 52.92.100
• libavcodec 55.69.100
• libavformat 55.48.100
• libavdevice 55.13.102
• libavfilter 4.11.100
• libswscale 2. 6.100
• libswresample 0.19.100
• libpostproc 52. 3.100
Please refer to the doc/APIChanges file for more information.
┌────────────────────────────┐
│ New Optimization │
└────────────────────────────┘
We are excited to announce that we have committed new x86 assembly
optimization for HEVC, and FFmpeg's audio resampler libswresample. ARM
users will get a boost in MLP/TrueHD decoding thanks to new optimization.
Decoding Huffyuv also got a major boost from optimization on the C code.
Of special interest for Microsoft Visual Studio users, we have also
converted some preexisting x86 assembly to NASM/Yasm format compatible
with MSVC setup, especially in the area of audio resampling.
Another major feature in this release is the introduction of AArch64
(ARMv8) assembly optimization. AArch64 is another name for the first
64-bit ARM architecture, used by Apple A7 SoC inside iPhone 5S. Some
32-bit ARM assembly has already been ported to AArch64, but more work is
underway.
┌────────────────────────────┐
│ Native Opus decoder │
└────────────────────────────┘
Opus is an open audio format jointly developed by Xiph.Org, Mozilla,
Skype/Microsoft, and Broadcom. It combines the features of the Skype Cilk
speech codec and the Xiph.Org CELT music codec into one low-latency
codec. Decoding Opus is already possible since FFmpeg 1.0 using the
libopus library, but the new Opus native decoder brings a higher level of
stability and speed.
┌────────────────────────────┐
│ QTKit and AVFoundation │
└────────────────────────────┘
For OS X users, the new QTKit and AVFoundation devices allow you to use
the integrated camera on Macs. AVFoundation is a newer API only available
on OS X 10.7 "Lion" or newer. For users with older OS X systems, the
QTKit device using the older OS X API is for you.
┌────────────────────────────┐
│ API Additions │
└────────────────────────────┘
In this release, stream side data are introduced as AVStream.side_data as
a way to store miscellaneous stream-wide information. The format is
similar to the previously anonymous structure AVPacket.side_data (now
named as AVPacketSideData). With this change, audio ReplayGain
information and video rotation matrix are now exported through this API,
if available in the demuxer.
We also have improved libswresample's Doxygen API documentation, so new
developers wishing to use FFmpeg's excellent libraries can get started
more easily and faster.
┌────────────────────────────┐
│ Last But Not Least │
└────────────────────────────┘
Other interesting new features including hqx video filter, a pixel art
scaling filter; a fixed-point AC-3 decoder contributed by Imagination
Technologies; an On2 TrueMotion VP7 video decoder; an HTML5 WebVTT
subtitle decoder that allows creation of WebVTT from any text-based
subtitles; and an 1-bit Direct Stream Digital audio decoder.
┌────────────────────────────┐
│ ★ List of New Features │
└────────────────────────────┘
┌────────────────────────────┐
│ Command line tools │
└────────────────────────────┘
• Support for decoding through DXVA2 in ffmpeg
┌────────────────────────────┐
│ libavcodec │
└────────────────────────────┘
• AC3 fixed-point decoding
• VP7 video decoder
• Alias PIX image encoder and decoder
• Improvements to the BRender PIX image decoder
• Improvements to the XBM decoder
• Improvements to OpenEXR image decoder
• Support decoding 16-bit RLE SGI images
• Direct Stream Digital (DSD) decoder
• On2 AVC (Audio for Video) decoder
• Native Opus decoder
• WebVTT encoder
┌────────────────────────────┐
│ libavdevice │
└────────────────────────────┘
• QTKit input device
• GDI screen grabbing for Windows
• AVFoundation input device
┌────────────────────────────┐
│ libavformat │
└────────────────────────────┘
• subfile protocol
• Phantom Cine demuxer
• Alternative rendition support for HTTP Live Streaming
• Magic Lantern Video (MLV) demuxer
• Image format auto-detection
• LRC lyric file demuxer and muxer
• Samba protocol (via libsmbclient)
• WebM DASH Manifest muxer
┌────────────────────────────┐
│ libavfilter │
└────────────────────────────┘
• shuffleplanes filter
• libbs2b-based stereo-to-binaural audio filter
• showcqt multimedia filter
• zoompan filter
• signalstats filter
• hqx filter (hq2x, hq3x, hq4x)
• flanger filter
• libfribidi support in drawtext
┌────────────────────────────┐
│ ⚠ Behaviour changes │
└────────────────────────────┘
• libx264 reference frames count is now limited depending on level chosen
• Because of the new image format auto-detection feature, you don't need to
specify image format when decoding an image with no extension.

1
VERSION Normal file
View File

@@ -0,0 +1 @@
2.3

View File

@@ -5,13 +5,11 @@ OBJS-$(HAVE_VFP) += $(VFP-OBJS) $(VFP-OBJS-yes)
OBJS-$(HAVE_NEON) += $(NEON-OBJS) $(NEON-OBJS-yes)
OBJS-$(HAVE_MIPSFPU) += $(MIPSFPU-OBJS) $(MIPSFPU-OBJS-yes)
OBJS-$(HAVE_MIPSDSP) += $(MIPSDSP-OBJS) $(MIPSDSP-OBJS-yes)
OBJS-$(HAVE_MIPS32R2) += $(MIPS32R2-OBJS) $(MIPS32R2-OBJS-yes)
OBJS-$(HAVE_MIPSDSPR1) += $(MIPSDSPR1-OBJS) $(MIPSDSPR1-OBJS-yes)
OBJS-$(HAVE_MIPSDSPR2) += $(MIPSDSPR2-OBJS) $(MIPSDSPR2-OBJS-yes)
OBJS-$(HAVE_MSA) += $(MSA-OBJS) $(MSA-OBJS-yes)
OBJS-$(HAVE_MMI) += $(MMI-OBJS) $(MMI-OBJS-yes)
OBJS-$(HAVE_ALTIVEC) += $(ALTIVEC-OBJS) $(ALTIVEC-OBJS-yes)
OBJS-$(HAVE_VSX) += $(VSX-OBJS) $(VSX-OBJS-yes)
OBJS-$(HAVE_MMX) += $(MMX-OBJS) $(MMX-OBJS-yes)
OBJS-$(HAVE_YASM) += $(YASM-OBJS) $(YASM-OBJS-yes)

View File

@@ -41,10 +41,8 @@
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/bprint.h"
#include "libavutil/display.h"
#include "libavutil/mathematics.h"
#include "libavutil/imgutils.h"
#include "libavutil/libm.h"
#include "libavutil/parseutils.h"
#include "libavutil/pixdesc.h"
#include "libavutil/eval.h"
@@ -52,7 +50,6 @@
#include "libavutil/opt.h"
#include "libavutil/cpu.h"
#include "libavutil/ffversion.h"
#include "libavutil/version.h"
#include "cmdutils.h"
#if CONFIG_NETWORK
#include "libavformat/network.h"
@@ -61,13 +58,10 @@
#include <sys/time.h>
#include <sys/resource.h>
#endif
#ifdef _WIN32
#include <windows.h>
#endif
static int init_report(const char *env);
AVDictionary *sws_dict;
struct SwsContext *sws_opts;
AVDictionary *swr_opts;
AVDictionary *format_opts, *codec_opts, *resample_opts;
@@ -77,13 +71,20 @@ int hide_banner = 0;
void init_opts(void)
{
av_dict_set(&sws_dict, "flags", "bicubic", 0);
if(CONFIG_SWSCALE)
sws_opts = sws_getContext(16, 16, 0, 16, 16, 0, SWS_BICUBIC,
NULL, NULL, NULL);
}
void uninit_opts(void)
{
#if CONFIG_SWSCALE
sws_freeContext(sws_opts);
sws_opts = NULL;
#endif
av_dict_free(&swr_opts);
av_dict_free(&sws_dict);
av_dict_free(&format_opts);
av_dict_free(&codec_opts);
av_dict_free(&resample_opts);
@@ -110,15 +111,6 @@ static void log_callback_report(void *ptr, int level, const char *fmt, va_list v
}
}
void init_dynload(void)
{
#ifdef _WIN32
/* Calling SetDllDirectory with the empty string (but not NULL) removes the
* current working directory from the DLL search path as a security pre-caution. */
SetDllDirectory("");
#endif
}
static void (*program_exit)(int ret);
void register_exit(void (*cb)(int ret))
@@ -174,7 +166,7 @@ void show_help_options(const OptionDef *options, const char *msg, int req_flags,
int first;
first = 1;
for (po = options; po->name; po++) {
for (po = options; po->name != NULL; po++) {
char buf[64];
if (((po->flags & req_flags) != req_flags) ||
@@ -213,7 +205,7 @@ static const OptionDef *find_option(const OptionDef *po, const char *name)
const char *p = strchr(name, ':');
int len = p ? p - name : strlen(name);
while (po->name) {
while (po->name != NULL) {
if (!strncmp(name, po->name, len) && strlen(po->name) == len)
break;
po++;
@@ -262,7 +254,7 @@ static void prepare_app_arguments(int *argc_ptr, char ***argv_ptr)
win32_argv_utf8 = av_mallocz(sizeof(char *) * (win32_argc + 1) + buffsize);
argstr_flat = (char *)win32_argv_utf8 + sizeof(char *) * (win32_argc + 1);
if (!win32_argv_utf8) {
if (win32_argv_utf8 == NULL) {
LocalFree(argv_w);
return;
}
@@ -298,14 +290,10 @@ static int write_option(void *optctx, const OptionDef *po, const char *opt,
if (po->flags & OPT_SPEC) {
SpecifierOpt **so = dst;
char *p = strchr(opt, ':');
char *str;
dstcount = (int *)(so + 1);
*so = grow_array(*so, sizeof(**so), dstcount, *dstcount + 1);
str = av_strdup(p ? p + 1 : "");
if (!str)
return AVERROR(ENOMEM);
(*so)[*dstcount - 1].specifier = str;
(*so)[*dstcount - 1].specifier = av_strdup(p ? p + 1 : "");
dst = &(*so)[*dstcount - 1].u;
}
@@ -313,8 +301,6 @@ static int write_option(void *optctx, const OptionDef *po, const char *opt,
char *str;
str = av_strdup(arg);
av_freep(dst);
if (!str)
return AVERROR(ENOMEM);
*(char **)dst = str;
} else if (po->flags & OPT_BOOL || po->flags & OPT_INT) {
*(int *)dst = parse_number_or_die(opt, arg, OPT_INT64, INT_MIN, INT_MAX);
@@ -458,7 +444,7 @@ int locate_option(int argc, char **argv, const OptionDef *options,
(po->name && !strcmp(optname, po->name)))
return i;
if (!po->name || po->flags & HAS_ARG)
if (po->flags & HAS_ARG)
i++;
}
return 0;
@@ -488,22 +474,10 @@ static void dump_argument(const char *a)
fputc('"', report_file);
}
static void check_options(const OptionDef *po)
{
while (po->name) {
if (po->flags & OPT_PERFILE)
av_assert0(po->flags & (OPT_INPUT | OPT_OUTPUT));
po++;
}
}
void parse_loglevel(int argc, char **argv, const OptionDef *options)
{
int idx = locate_option(argc, argv, options, "loglevel");
const char *env;
check_options(options);
if (!idx)
idx = locate_option(argc, argv, options, "v");
if (idx && argv[idx + 1])
@@ -535,7 +509,7 @@ static const AVOption *opt_find(void *obj, const char *name, const char *unit,
return o;
}
#define FLAGS (o->type == AV_OPT_TYPE_FLAGS && (arg[0]=='-' || arg[0]=='+')) ? AV_DICT_APPEND : 0
#define FLAGS (o->type == AV_OPT_TYPE_FLAGS) ? AV_DICT_APPEND : 0
int opt_default(void *optctx, const char *opt, const char *arg)
{
const AVOption *o;
@@ -546,12 +520,7 @@ int opt_default(void *optctx, const char *opt, const char *arg)
#if CONFIG_AVRESAMPLE
const AVClass *rc = avresample_get_class();
#endif
#if CONFIG_SWSCALE
const AVClass *sc = sws_get_class();
#endif
#if CONFIG_SWRESAMPLE
const AVClass *swr_class = swr_get_class();
#endif
const AVClass *sc, *swr_class;
if (!strcmp(opt, "debug") || !strcmp(opt, "fdebug"))
av_log_set_level(AV_LOG_DEBUG);
@@ -575,24 +544,15 @@ int opt_default(void *optctx, const char *opt, const char *arg)
consumed = 1;
}
#if CONFIG_SWSCALE
if (!consumed && (o = opt_find(&sc, opt, NULL, 0,
AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) {
struct SwsContext *sws = sws_alloc_context();
int ret = av_opt_set(sws, opt, arg, 0);
sws_freeContext(sws);
if (!strcmp(opt, "srcw") || !strcmp(opt, "srch") ||
!strcmp(opt, "dstw") || !strcmp(opt, "dsth") ||
!strcmp(opt, "src_format") || !strcmp(opt, "dst_format")) {
av_log(NULL, AV_LOG_ERROR, "Directly using swscale dimensions/format options is not supported, please use the -s or -pix_fmt options\n");
return AVERROR(EINVAL);
}
sc = sws_get_class();
if (!consumed && opt_find(&sc, opt, NULL, 0,
AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ)) {
// XXX we only support sws_flags, not arbitrary sws options
int ret = av_opt_set(sws_opts, opt, arg, 0);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error setting option %s.\n", opt);
return ret;
}
av_dict_set(&sws_dict, opt, arg, FLAGS);
consumed = 1;
}
#else
@@ -602,6 +562,7 @@ int opt_default(void *optctx, const char *opt, const char *arg)
}
#endif
#if CONFIG_SWRESAMPLE
swr_class = swr_get_class();
if (!consumed && (o=opt_find(&swr_class, opt, NULL, 0,
AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) {
struct SwrContext *swr = swr_alloc();
@@ -665,7 +626,9 @@ static void finish_group(OptionParseContext *octx, int group_idx,
*g = octx->cur_group;
g->arg = arg;
g->group_def = l->group_def;
g->sws_dict = sws_dict;
#if CONFIG_SWSCALE
g->sws_opts = sws_opts;
#endif
g->swr_opts = swr_opts;
g->codec_opts = codec_opts;
g->format_opts = format_opts;
@@ -674,7 +637,9 @@ static void finish_group(OptionParseContext *octx, int group_idx,
codec_opts = NULL;
format_opts = NULL;
resample_opts = NULL;
sws_dict = NULL;
#if CONFIG_SWSCALE
sws_opts = NULL;
#endif
swr_opts = NULL;
init_opts();
@@ -730,8 +695,9 @@ void uninit_parse_context(OptionParseContext *octx)
av_dict_free(&l->groups[j].codec_opts);
av_dict_free(&l->groups[j].format_opts);
av_dict_free(&l->groups[j].resample_opts);
av_dict_free(&l->groups[j].sws_dict);
#if CONFIG_SWSCALE
sws_freeContext(l->groups[j].sws_opts);
#endif
av_dict_free(&l->groups[j].swr_opts);
}
av_freep(&l->groups);
@@ -873,7 +839,6 @@ int opt_loglevel(void *optctx, const char *opt, const char *arg)
{ "info" , AV_LOG_INFO },
{ "verbose", AV_LOG_VERBOSE },
{ "debug" , AV_LOG_DEBUG },
{ "trace" , AV_LOG_TRACE },
};
char *tail;
int level;
@@ -994,10 +959,9 @@ static int init_report(const char *env)
report_file = fopen(filename.str, "w");
if (!report_file) {
int ret = AVERROR(errno);
av_log(NULL, AV_LOG_ERROR, "Failed to open report \"%s\": %s\n",
filename.str, strerror(errno));
return ret;
return AVERROR(errno);
}
av_log_set_callback(log_callback_report);
av_log(NULL, AV_LOG_INFO,
@@ -1071,8 +1035,7 @@ static int warned_cfg = 0;
LIB##LIBNAME##_VERSION_MAJOR, \
LIB##LIBNAME##_VERSION_MINOR, \
LIB##LIBNAME##_VERSION_MICRO, \
AV_VERSION_MAJOR(version), AV_VERSION_MINOR(version),\
AV_VERSION_MICRO(version)); \
version >> 16, version >> 8 & 0xff, version & 0xff); \
} \
if (flags & SHOW_CONFIG) { \
const char *cfg = libname##_configuration(); \
@@ -1091,15 +1054,15 @@ static int warned_cfg = 0;
static void print_all_libs_info(int flags, int level)
{
PRINT_LIB_INFO(avutil, AVUTIL, flags, level);
PRINT_LIB_INFO(avcodec, AVCODEC, flags, level);
PRINT_LIB_INFO(avformat, AVFORMAT, flags, level);
PRINT_LIB_INFO(avdevice, AVDEVICE, flags, level);
PRINT_LIB_INFO(avfilter, AVFILTER, flags, level);
PRINT_LIB_INFO(avutil, AVUTIL, flags, level);
PRINT_LIB_INFO(avcodec, AVCODEC, flags, level);
PRINT_LIB_INFO(avformat, AVFORMAT, flags, level);
PRINT_LIB_INFO(avdevice, AVDEVICE, flags, level);
PRINT_LIB_INFO(avfilter, AVFILTER, flags, level);
PRINT_LIB_INFO(avresample, AVRESAMPLE, flags, level);
PRINT_LIB_INFO(swscale, SWSCALE, flags, level);
PRINT_LIB_INFO(swresample, SWRESAMPLE, flags, level);
PRINT_LIB_INFO(postproc, POSTPROC, flags, level);
PRINT_LIB_INFO(swscale, SWSCALE, flags, level);
PRINT_LIB_INFO(swresample,SWRESAMPLE, flags, level);
PRINT_LIB_INFO(postproc, POSTPROC, flags, level);
}
static void print_program_info(int flags, int level)
@@ -1111,7 +1074,8 @@ static void print_program_info(int flags, int level)
av_log(NULL, level, " Copyright (c) %d-%d the FFmpeg developers",
program_birth_year, CONFIG_THIS_YEAR);
av_log(NULL, level, "\n");
av_log(NULL, level, "%sbuilt with %s\n", indent, CC_IDENT);
av_log(NULL, level, "%sbuilt on %s %s with %s\n",
indent, __DATE__, __TIME__, CC_IDENT);
av_log(NULL, level, "%sconfiguration: " FFMPEG_CONFIGURATION "\n", indent);
}
@@ -1248,7 +1212,12 @@ static int is_device(const AVClass *avclass)
{
if (!avclass)
return 0;
return AV_IS_INPUT_DEVICE(avclass->category) || AV_IS_OUTPUT_DEVICE(avclass->category);
return avclass->category == AV_CLASS_CATEGORY_DEVICE_VIDEO_OUTPUT ||
avclass->category == AV_CLASS_CATEGORY_DEVICE_VIDEO_INPUT ||
avclass->category == AV_CLASS_CATEGORY_DEVICE_AUDIO_OUTPUT ||
avclass->category == AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT ||
avclass->category == AV_CLASS_CATEGORY_DEVICE_OUTPUT ||
avclass->category == AV_CLASS_CATEGORY_DEVICE_INPUT;
}
static int show_formats_devices(void *optctx, const char *opt, const char *arg, int device_only)
@@ -1273,7 +1242,7 @@ static int show_formats_devices(void *optctx, const char *opt, const char *arg,
is_dev = is_device(ofmt->priv_class);
if (!is_dev && device_only)
continue;
if ((!name || strcmp(ofmt->name, name) < 0) &&
if ((name == NULL || strcmp(ofmt->name, name) < 0) &&
strcmp(ofmt->name, last_name) > 0) {
name = ofmt->name;
long_name = ofmt->long_name;
@@ -1284,7 +1253,7 @@ static int show_formats_devices(void *optctx, const char *opt, const char *arg,
is_dev = is_device(ifmt->priv_class);
if (!is_dev && device_only)
continue;
if ((!name || strcmp(ifmt->name, name) < 0) &&
if ((name == NULL || strcmp(ifmt->name, name) < 0) &&
strcmp(ifmt->name, last_name) > 0) {
name = ifmt->name;
long_name = ifmt->long_name;
@@ -1293,7 +1262,7 @@ static int show_formats_devices(void *optctx, const char *opt, const char *arg,
if (name && strcmp(ifmt->name, name) == 0)
decode = 1;
}
if (!name)
if (name == NULL)
break;
last_name = name;
@@ -1336,47 +1305,16 @@ static void print_codec(const AVCodec *c)
printf("%s %s [%s]:\n", encoder ? "Encoder" : "Decoder", c->name,
c->long_name ? c->long_name : "");
printf(" General capabilities: ");
if (c->capabilities & AV_CODEC_CAP_DRAW_HORIZ_BAND)
printf("horizband ");
if (c->capabilities & AV_CODEC_CAP_DR1)
printf("dr1 ");
if (c->capabilities & AV_CODEC_CAP_TRUNCATED)
printf("trunc ");
if (c->capabilities & AV_CODEC_CAP_DELAY)
printf("delay ");
if (c->capabilities & AV_CODEC_CAP_SMALL_LAST_FRAME)
printf("small ");
if (c->capabilities & AV_CODEC_CAP_SUBFRAMES)
printf("subframes ");
if (c->capabilities & AV_CODEC_CAP_EXPERIMENTAL)
printf("exp ");
if (c->capabilities & AV_CODEC_CAP_CHANNEL_CONF)
printf("chconf ");
if (c->capabilities & AV_CODEC_CAP_PARAM_CHANGE)
printf("paramchange ");
if (c->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE)
printf("variable ");
if (c->capabilities & (AV_CODEC_CAP_FRAME_THREADS |
AV_CODEC_CAP_SLICE_THREADS |
AV_CODEC_CAP_AUTO_THREADS))
printf("threads ");
if (!c->capabilities)
printf("none");
printf("\n");
if (c->type == AVMEDIA_TYPE_VIDEO ||
c->type == AVMEDIA_TYPE_AUDIO) {
printf(" Threading capabilities: ");
switch (c->capabilities & (AV_CODEC_CAP_FRAME_THREADS |
AV_CODEC_CAP_SLICE_THREADS |
AV_CODEC_CAP_AUTO_THREADS)) {
case AV_CODEC_CAP_FRAME_THREADS |
AV_CODEC_CAP_SLICE_THREADS: printf("frame and slice"); break;
case AV_CODEC_CAP_FRAME_THREADS: printf("frame"); break;
case AV_CODEC_CAP_SLICE_THREADS: printf("slice"); break;
case AV_CODEC_CAP_AUTO_THREADS : printf("auto"); break;
default: printf("none"); break;
switch (c->capabilities & (CODEC_CAP_FRAME_THREADS |
CODEC_CAP_SLICE_THREADS)) {
case CODEC_CAP_FRAME_THREADS |
CODEC_CAP_SLICE_THREADS: printf("frame and slice"); break;
case CODEC_CAP_FRAME_THREADS: printf("frame"); break;
case CODEC_CAP_SLICE_THREADS: printf("slice"); break;
default: printf("no"); break;
}
printf("\n");
}
@@ -1435,7 +1373,7 @@ static int compare_codec_desc(const void *a, const void *b)
const AVCodecDescriptor * const *da = a;
const AVCodecDescriptor * const *db = b;
return (*da)->type != (*db)->type ? FFDIFFSIGN((*da)->type, (*db)->type) :
return (*da)->type != (*db)->type ? (*da)->type - (*db)->type :
strcmp((*da)->name, (*db)->name);
}
@@ -1549,11 +1487,11 @@ static void print_codecs(int encoder)
while ((codec = next_codec_for_id(desc->id, codec, encoder))) {
printf(" %c", get_media_type_char(desc->type));
printf((codec->capabilities & AV_CODEC_CAP_FRAME_THREADS) ? "F" : ".");
printf((codec->capabilities & AV_CODEC_CAP_SLICE_THREADS) ? "S" : ".");
printf((codec->capabilities & AV_CODEC_CAP_EXPERIMENTAL) ? "X" : ".");
printf((codec->capabilities & AV_CODEC_CAP_DRAW_HORIZ_BAND)?"B" : ".");
printf((codec->capabilities & AV_CODEC_CAP_DR1) ? "D" : ".");
printf((codec->capabilities & CODEC_CAP_FRAME_THREADS) ? "F" : ".");
printf((codec->capabilities & CODEC_CAP_SLICE_THREADS) ? "S" : ".");
printf((codec->capabilities & CODEC_CAP_EXPERIMENTAL) ? "X" : ".");
printf((codec->capabilities & CODEC_CAP_DRAW_HORIZ_BAND)?"B" : ".");
printf((codec->capabilities & CODEC_CAP_DR1) ? "D" : ".");
printf(" %-20s %s", codec->name, codec->long_name ? codec->long_name : "");
if (strcmp(codec->name, desc->name))
@@ -1579,11 +1517,10 @@ int show_encoders(void *optctx, const char *opt, const char *arg)
int show_bsfs(void *optctx, const char *opt, const char *arg)
{
const AVBitStreamFilter *bsf = NULL;
void *opaque = NULL;
AVBitStreamFilter *bsf = NULL;
printf("Bitstream filters:\n");
while ((bsf = av_bsf_next(&opaque)))
while ((bsf = av_bitstream_filter_next(bsf)))
printf("%s\n", bsf->name);
printf("\n");
return 0;
@@ -1597,17 +1534,16 @@ int show_protocols(void *optctx, const char *opt, const char *arg)
printf("Supported file protocols:\n"
"Input:\n");
while ((name = avio_enum_protocols(&opaque, 0)))
printf(" %s\n", name);
printf("%s\n", name);
printf("Output:\n");
while ((name = avio_enum_protocols(&opaque, 1)))
printf(" %s\n", name);
printf("%s\n", name);
return 0;
}
int show_filters(void *optctx, const char *opt, const char *arg)
{
#if CONFIG_AVFILTER
const AVFilter *filter = NULL;
const AVFilter av_unused(*filter) = NULL;
char descr[64], *descr_cur;
int i, j;
const AVFilterPad *pad;
@@ -1615,11 +1551,12 @@ int show_filters(void *optctx, const char *opt, const char *arg)
printf("Filters:\n"
" T.. = Timeline support\n"
" .S. = Slice threading\n"
" ..C = Command support\n"
" ..C = Commmand support\n"
" A = Audio input/output\n"
" V = Video input/output\n"
" N = Dynamic number and/or type of input/output\n"
" | = Source or sink filter\n");
#if CONFIG_AVFILTER
while ((filter = avfilter_next(filter))) {
descr_cur = descr;
for (i = 0; i < 2; i++) {
@@ -1628,24 +1565,22 @@ int show_filters(void *optctx, const char *opt, const char *arg)
*(descr_cur++) = '>';
}
pad = i ? filter->outputs : filter->inputs;
for (j = 0; pad && avfilter_pad_get_name(pad, j); j++) {
for (j = 0; pad && pad[j].name; j++) {
if (descr_cur >= descr + sizeof(descr) - 4)
break;
*(descr_cur++) = get_media_type_char(avfilter_pad_get_type(pad, j));
*(descr_cur++) = get_media_type_char(pad[j].type);
}
if (!j)
*(descr_cur++) = ((!i && (filter->flags & AVFILTER_FLAG_DYNAMIC_INPUTS)) ||
( i && (filter->flags & AVFILTER_FLAG_DYNAMIC_OUTPUTS))) ? 'N' : '|';
}
*descr_cur = 0;
printf(" %c%c%c %-17s %-10s %s\n",
printf(" %c%c%c %-16s %-10s %s\n",
filter->flags & AVFILTER_FLAG_SUPPORT_TIMELINE ? 'T' : '.',
filter->flags & AVFILTER_FLAG_SLICE_THREADS ? 'S' : '.',
filter->process_command ? 'C' : '.',
filter->name, descr, filter->description);
}
#else
printf("No filters available: libavfilter disabled\n");
#endif
return 0;
}
@@ -1704,19 +1639,19 @@ int show_layouts(void *optctx, const char *opt, const char *arg)
const char *name, *descr;
printf("Individual channels:\n"
"NAME DESCRIPTION\n");
"NAME DESCRIPTION\n");
for (i = 0; i < 63; i++) {
name = av_get_channel_name((uint64_t)1 << i);
if (!name)
continue;
descr = av_get_channel_description((uint64_t)1 << i);
printf("%-14s %s\n", name, descr);
printf("%-12s%s\n", name, descr);
}
printf("\nStandard channel layouts:\n"
"NAME DECOMPOSITION\n");
"NAME DECOMPOSITION\n");
for (i = 0; !av_get_standard_channel_layout(i, &layout, &name); i++) {
if (name) {
printf("%-14s ", name);
printf("%-12s", name);
for (j = 1; j; j <<= 1)
if ((layout & j))
printf("%s%s", (layout & (j - 1)) ? "+" : "", av_get_channel_name(j));
@@ -1883,8 +1818,6 @@ int show_help(void *optctx, const char *opt, const char *arg)
av_log_set_callback(log_callback_help);
topic = av_strdup(arg ? arg : "");
if (!topic)
return AVERROR(ENOMEM);
par = strchr(topic, '=');
if (par)
*par++ = 0;
@@ -1922,6 +1855,48 @@ int read_yesno(void)
return yesno;
}
int cmdutils_read_file(const char *filename, char **bufptr, size_t *size)
{
int ret;
FILE *f = av_fopen_utf8(filename, "rb");
if (!f) {
av_log(NULL, AV_LOG_ERROR, "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);
if (*size == (size_t)-1) {
av_log(NULL, AV_LOG_ERROR, "IO error: %s\n", strerror(errno));
fclose(f);
return AVERROR(errno);
}
*bufptr = av_malloc(*size + 1);
if (!*bufptr) {
av_log(NULL, AV_LOG_ERROR, "Could not allocate file buffer\n");
fclose(f);
return AVERROR(ENOMEM);
}
ret = fread(*bufptr, 1, *size, f);
if (ret < *size) {
av_free(*bufptr);
if (ferror(f)) {
av_log(NULL, AV_LOG_ERROR, "Error while reading file '%s': %s\n",
filename, strerror(errno));
ret = AVERROR(errno);
} else
ret = AVERROR_EOF;
} else {
ret = 0;
(*bufptr)[(*size)++] = '\0';
}
fclose(f);
return ret;
}
FILE *get_preset_file(char *filename, size_t filename_size,
const char *preset_name, int is_path,
const char *codec_name)
@@ -1994,7 +1969,7 @@ AVDictionary *filter_codec_opts(AVDictionary *opts, enum AVCodecID codec_id,
codec = s->oformat ? avcodec_find_encoder(codec_id)
: avcodec_find_decoder(codec_id);
switch (st->codecpar->codec_type) {
switch (st->codec->codec_type) {
case AVMEDIA_TYPE_VIDEO:
prefix = 'v';
flags |= AV_OPT_FLAG_VIDEO_PARAM;
@@ -2017,7 +1992,7 @@ AVDictionary *filter_codec_opts(AVDictionary *opts, enum AVCodecID codec_id,
switch (check_stream_specifier(s, st, p + 1)) {
case 1: *p = 0; break;
case 0: continue;
default: exit_program(1);
default: return NULL;
}
if (av_opt_find(&cc, t->key, NULL, flags, AV_OPT_SEARCH_FAKE_OBJ) ||
@@ -2052,7 +2027,7 @@ AVDictionary **setup_find_stream_info_opts(AVFormatContext *s,
return NULL;
}
for (i = 0; i < s->nb_streams; i++)
opts[i] = filter_codec_opts(codec_opts, s->streams[i]->codecpar->codec_id,
opts[i] = filter_codec_opts(codec_opts, s->streams[i]->codec->codec_id,
s, s->streams[i], NULL);
return opts;
}
@@ -2064,7 +2039,7 @@ void *grow_array(void *array, int elem_size, int *size, int new_size)
exit_program(1);
}
if (*size < new_size) {
uint8_t *tmp = av_realloc_array(array, new_size, elem_size);
uint8_t *tmp = av_realloc(array, new_size*elem_size);
if (!tmp) {
av_log(NULL, AV_LOG_ERROR, "Could not alloc buffer.\n");
exit_program(1);
@@ -2075,189 +2050,3 @@ void *grow_array(void *array, int elem_size, int *size, int new_size)
}
return array;
}
double get_rotation(AVStream *st)
{
AVDictionaryEntry *rotate_tag = av_dict_get(st->metadata, "rotate", NULL, 0);
uint8_t* displaymatrix = av_stream_get_side_data(st,
AV_PKT_DATA_DISPLAYMATRIX, NULL);
double theta = 0;
if (rotate_tag && *rotate_tag->value && strcmp(rotate_tag->value, "0")) {
char *tail;
theta = av_strtod(rotate_tag->value, &tail);
if (*tail)
theta = 0;
}
if (displaymatrix && !theta)
theta = -av_display_rotation_get((int32_t*) displaymatrix);
theta -= 360*floor(theta/360 + 0.9/360);
if (fabs(theta - 90*round(theta/90)) > 2)
av_log(NULL, AV_LOG_WARNING, "Odd rotation angle.\n"
"If you want to help, upload a sample "
"of this file to ftp://upload.ffmpeg.org/incoming/ "
"and contact the ffmpeg-devel mailing list. (ffmpeg-devel@ffmpeg.org)");
return theta;
}
#if CONFIG_AVDEVICE
static int print_device_sources(AVInputFormat *fmt, AVDictionary *opts)
{
int ret, i;
AVDeviceInfoList *device_list = NULL;
if (!fmt || !fmt->priv_class || !AV_IS_INPUT_DEVICE(fmt->priv_class->category))
return AVERROR(EINVAL);
printf("Auto-detected sources for %s:\n", fmt->name);
if (!fmt->get_device_list) {
ret = AVERROR(ENOSYS);
printf("Cannot list sources. Not implemented.\n");
goto fail;
}
if ((ret = avdevice_list_input_sources(fmt, NULL, opts, &device_list)) < 0) {
printf("Cannot list sources.\n");
goto fail;
}
for (i = 0; i < device_list->nb_devices; i++) {
printf("%s %s [%s]\n", device_list->default_device == i ? "*" : " ",
device_list->devices[i]->device_name, device_list->devices[i]->device_description);
}
fail:
avdevice_free_list_devices(&device_list);
return ret;
}
static int print_device_sinks(AVOutputFormat *fmt, AVDictionary *opts)
{
int ret, i;
AVDeviceInfoList *device_list = NULL;
if (!fmt || !fmt->priv_class || !AV_IS_OUTPUT_DEVICE(fmt->priv_class->category))
return AVERROR(EINVAL);
printf("Auto-detected sinks for %s:\n", fmt->name);
if (!fmt->get_device_list) {
ret = AVERROR(ENOSYS);
printf("Cannot list sinks. Not implemented.\n");
goto fail;
}
if ((ret = avdevice_list_output_sinks(fmt, NULL, opts, &device_list)) < 0) {
printf("Cannot list sinks.\n");
goto fail;
}
for (i = 0; i < device_list->nb_devices; i++) {
printf("%s %s [%s]\n", device_list->default_device == i ? "*" : " ",
device_list->devices[i]->device_name, device_list->devices[i]->device_description);
}
fail:
avdevice_free_list_devices(&device_list);
return ret;
}
static int show_sinks_sources_parse_arg(const char *arg, char **dev, AVDictionary **opts)
{
int ret;
if (arg) {
char *opts_str = NULL;
av_assert0(dev && opts);
*dev = av_strdup(arg);
if (!*dev)
return AVERROR(ENOMEM);
if ((opts_str = strchr(*dev, ','))) {
*(opts_str++) = '\0';
if (opts_str[0] && ((ret = av_dict_parse_string(opts, opts_str, "=", ":", 0)) < 0)) {
av_freep(dev);
return ret;
}
}
} else
printf("\nDevice name is not provided.\n"
"You can pass devicename[,opt1=val1[,opt2=val2...]] as an argument.\n\n");
return 0;
}
int show_sources(void *optctx, const char *opt, const char *arg)
{
AVInputFormat *fmt = NULL;
char *dev = NULL;
AVDictionary *opts = NULL;
int ret = 0;
int error_level = av_log_get_level();
av_log_set_level(AV_LOG_ERROR);
if ((ret = show_sinks_sources_parse_arg(arg, &dev, &opts)) < 0)
goto fail;
do {
fmt = av_input_audio_device_next(fmt);
if (fmt) {
if (!strcmp(fmt->name, "lavfi"))
continue; //it's pointless to probe lavfi
if (dev && !av_match_name(dev, fmt->name))
continue;
print_device_sources(fmt, opts);
}
} while (fmt);
do {
fmt = av_input_video_device_next(fmt);
if (fmt) {
if (dev && !av_match_name(dev, fmt->name))
continue;
print_device_sources(fmt, opts);
}
} while (fmt);
fail:
av_dict_free(&opts);
av_free(dev);
av_log_set_level(error_level);
return ret;
}
int show_sinks(void *optctx, const char *opt, const char *arg)
{
AVOutputFormat *fmt = NULL;
char *dev = NULL;
AVDictionary *opts = NULL;
int ret = 0;
int error_level = av_log_get_level();
av_log_set_level(AV_LOG_ERROR);
if ((ret = show_sinks_sources_parse_arg(arg, &dev, &opts)) < 0)
goto fail;
do {
fmt = av_output_audio_device_next(fmt);
if (fmt) {
if (dev && !av_match_name(dev, fmt->name))
continue;
print_device_sinks(fmt, opts);
}
} while (fmt);
do {
fmt = av_output_video_device_next(fmt);
if (fmt) {
if (dev && !av_match_name(dev, fmt->name))
continue;
print_device_sinks(fmt, opts);
}
} while (fmt);
fail:
av_dict_free(&opts);
av_free(dev);
av_log_set_level(error_level);
return ret;
}
#endif

View File

@@ -19,8 +19,8 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef CMDUTILS_H
#define CMDUTILS_H
#ifndef FFMPEG_CMDUTILS_H
#define FFMPEG_CMDUTILS_H
#include <stdint.h>
@@ -46,7 +46,7 @@ extern const int program_birth_year;
extern AVCodecContext *avcodec_opts[AVMEDIA_TYPE_NB];
extern AVFormatContext *avformat_opts;
extern AVDictionary *sws_dict;
extern struct SwsContext *sws_opts;
extern AVDictionary *swr_opts;
extern AVDictionary *format_opts, *codec_opts, *resample_opts;
extern int hide_banner;
@@ -61,11 +61,6 @@ void register_exit(void (*cb)(int ret));
*/
void exit_program(int ret) av_noreturn;
/**
* Initialize dynamic library loading
*/
void init_dynload(void);
/**
* Initialize the cmdutils option system, in particular
* allocate the *_opts contexts.
@@ -282,7 +277,7 @@ typedef struct OptionGroup {
AVDictionary *codec_opts;
AVDictionary *format_opts;
AVDictionary *resample_opts;
AVDictionary *sws_dict;
struct SwsContext *sws_opts;
AVDictionary *swr_opts;
} OptionGroup;
@@ -448,20 +443,6 @@ int show_formats(void *optctx, const char *opt, const char *arg);
*/
int show_devices(void *optctx, const char *opt, const char *arg);
#if CONFIG_AVDEVICE
/**
* Print a listing containing autodetected sinks of the output device.
* Device name with options may be passed as an argument to limit results.
*/
int show_sinks(void *optctx, const char *opt, const char *arg);
/**
* Print a listing containing autodetected sources of the input device.
* Device name with options may be passed as an argument to limit results.
*/
int show_sources(void *optctx, const char *opt, const char *arg);
#endif
/**
* Print a listing containing all the codecs supported by the
* program.
@@ -534,6 +515,18 @@ int show_colors(void *optctx, const char *opt, const char *arg);
*/
int read_yesno(void);
/**
* Read the file with name filename, and put its content in a newly
* allocated 0-terminated buffer.
*
* @param filename file to read from
* @param bufptr location where pointer to buffer is returned
* @param size location where size of buffer is returned
* @return >= 0 in case of success, a negative value corresponding to an
* AVERROR error code in case of failure.
*/
int cmdutils_read_file(const char *filename, char **bufptr, size_t *size);
/**
* Get a file corresponding to a preset file.
*
@@ -590,6 +583,4 @@ void *grow_array(void *array, int elem_size, int *size, int new_size);
char name[128];\
av_get_channel_layout_string(name, sizeof(name), 0, ch_layout);
double get_rotation(AVStream *st);
#endif /* CMDUTILS_H */

View File

@@ -27,9 +27,3 @@
{ "opencl_bench", OPT_EXIT, {.func_arg = opt_opencl_bench}, "run benchmark on all OpenCL devices and show results" },
{ "opencl_options", HAS_ARG, {.func_arg = opt_opencl}, "set OpenCL environment options" },
#endif
#if CONFIG_AVDEVICE
{ "sources" , OPT_EXIT | HAS_ARG, { .func_arg = show_sources },
"list sources of the input device", "device" },
{ "sinks" , OPT_EXIT | HAS_ARG, { .func_arg = show_sinks },
"list sinks of the output device", "device" },
#endif

View File

@@ -22,7 +22,6 @@
#include "libavutil/time.h"
#include "libavutil/log.h"
#include "libavutil/opencl.h"
#include "libavutil/avstring.h"
#include "cmdutils.h"
typedef struct {
@@ -206,9 +205,7 @@ end:
static int compare_ocl_device_desc(const void *a, const void *b)
{
const OpenCLDeviceBenchmark* va = (const OpenCLDeviceBenchmark*)a;
const OpenCLDeviceBenchmark* vb = (const OpenCLDeviceBenchmark*)b;
return FFDIFFSIGN(va->runtime , vb->runtime);
return ((OpenCLDeviceBenchmark*)a)->runtime - ((OpenCLDeviceBenchmark*)b)->runtime;
}
int opt_opencl_bench(void *optctx, const char *opt, const char *arg)
@@ -241,8 +238,7 @@ int opt_opencl_bench(void *optctx, const char *opt, const char *arg)
devices[count].platform_idx = i;
devices[count].device_idx = j;
devices[count].runtime = score;
av_strlcpy(devices[count].device_name, device_node->device_name,
sizeof(devices[count].device_name));
strcpy(devices[count].device_name, device_node->device_name);
count++;
}
}

View File

@@ -5,20 +5,12 @@
# first so "all" becomes default target
all: all-yes
DEFAULT_YASMD=.dbg
ifeq ($(DBG),1)
YASMD=$(DEFAULT_YASMD)
else
YASMD=
endif
ifndef SUBDIR
ifndef V
Q = @
ECHO = printf "$(1)\t%s\n" $(2)
BRIEF = CC CXX OBJCC HOSTCC HOSTLD AS YASM AR LD STRIP CP WINDRES
BRIEF = CC CXX HOSTCC HOSTLD AS YASM AR LD STRIP CP WINDRES
SILENT = DEPCC DEPHOSTCC DEPAS DEPYASM RANLIB RM
MSG = $@
@@ -32,14 +24,12 @@ endif
ALLFFLIBS = avcodec avdevice avfilter avformat avresample avutil postproc swscale swresample
# NASM requires -I path terminated with /
IFLAGS := -I. -I$(SRC_LINK)/
IFLAGS := -I. -I$(SRC_PATH)/
CPPFLAGS := $(IFLAGS) $(CPPFLAGS)
CFLAGS += $(ECFLAGS)
CCFLAGS = $(CPPFLAGS) $(CFLAGS)
OBJCFLAGS += $(EOBJCFLAGS)
OBJCCFLAGS = $(CPPFLAGS) $(CFLAGS) $(OBJCFLAGS)
ASFLAGS := $(CPPFLAGS) $(ASFLAGS)
CXXFLAGS := $(CPPFLAGS) $(CFLAGS) $(CXXFLAGS)
CXXFLAGS += $(CPPFLAGS) $(CFLAGS)
YASMFLAGS += $(IFLAGS:%=%/) -Pconfig.asm
HOSTCCFLAGS = $(IFLAGS) $(HOSTCPPFLAGS) $(HOSTCFLAGS)
@@ -47,13 +37,12 @@ LDFLAGS := $(ALLFFLIBS:%=$(LD_PATH)lib%) $(LDFLAGS)
define COMPILE
$(call $(1)DEP,$(1))
$($(1)) $($(1)FLAGS) $($(1)_DEPFLAGS) $($(1)_C) $($(1)_O) $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<)
$($(1)) $($(1)FLAGS) $($(1)_DEPFLAGS) $($(1)_C) $($(1)_O) $<
endef
COMPILE_C = $(call COMPILE,CC)
COMPILE_CXX = $(call COMPILE,CXX)
COMPILE_S = $(call COMPILE,AS)
COMPILE_M = $(call COMPILE,OBJCC)
COMPILE_HOSTC = $(call COMPILE,HOSTCC)
%.o: %.c
@@ -63,10 +52,10 @@ COMPILE_HOSTC = $(call COMPILE,HOSTCC)
$(COMPILE_CXX)
%.o: %.m
$(COMPILE_M)
$(COMPILE_C)
%.s: %.c
$(CC) $(CCFLAGS) -S -o $@ $<
$(CC) $(CPPFLAGS) $(CFLAGS) -S -o $@ $<
%.o: %.S
$(COMPILE_S)
@@ -83,7 +72,10 @@ COMPILE_HOSTC = $(call COMPILE,HOSTCC)
%.h.c:
$(Q)echo '#include "$*.h"' >$@
%.c %.h %.ver: TAG = GEN
%.ver: %.v
$(Q)sed 's/$$MAJOR/$($(basename $(@F))_VERSION_MAJOR)/' $^ > $@
%.c %.h: TAG = GEN
# Dummy rule to stop make trying to rebuild removed or renamed headers
%.h:
@@ -109,8 +101,8 @@ FFEXTRALIBS := $(LDLIBS:%=$(LD_LIB)) $(EXTRALIBS)
OBJS := $(sort $(OBJS:%=$(SUBDIR)%))
SLIBOBJS := $(sort $(SLIBOBJS:%=$(SUBDIR)%))
TESTOBJS := $(TESTOBJS:%=$(SUBDIR)tests/%) $(TESTPROGS:%=$(SUBDIR)tests/%.o)
TESTPROGS := $(TESTPROGS:%=$(SUBDIR)tests/%$(EXESUF))
TESTOBJS := $(TESTOBJS:%=$(SUBDIR)%) $(TESTPROGS:%=$(SUBDIR)%-test.o)
TESTPROGS := $(TESTPROGS:%=$(SUBDIR)%-test$(EXESUF))
HOSTOBJS := $(HOSTPROGS:%=$(SUBDIR)%.o)
HOSTPROGS := $(HOSTPROGS:%=$(SUBDIR)%$(HOSTEXESUF))
TOOLS += $(TOOLS-yes)
@@ -118,9 +110,8 @@ TOOLOBJS := $(TOOLS:%=tools/%.o)
TOOLS := $(TOOLS:%=tools/%$(EXESUF))
HEADERS += $(HEADERS-yes)
PATH_LIBNAME = $(foreach NAME,$(1),lib$(NAME)/$($(2)LIBNAME))
DEP_LIBS := $(foreach lib,$(FFLIBS),$(call PATH_LIBNAME,$(lib),$(CONFIG_SHARED:yes=S)))
STATIC_DEP_LIBS := $(foreach lib,$(FFLIBS),$(call PATH_LIBNAME,$(lib)))
PATH_LIBNAME = $(foreach NAME,$(1),lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME))
DEP_LIBS := $(foreach lib,$(FFLIBS),$(call PATH_LIBNAME,$(lib)))
SRC_DIR := $(SRC_PATH)/lib$(NAME)
ALLHEADERS := $(subst $(SRC_DIR)/,$(SUBDIR),$(wildcard $(SRC_DIR)/*.h $(SRC_DIR)/$(ARCH)/*.h))
@@ -147,15 +138,17 @@ $(TOOLOBJS): | tools
OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(SLIBOBJS) $(TESTOBJS))
CLEANSUFFIXES = *.d *.o *~ *.h.c *.gcda *.gcno *.map *.ver *.ho *$(DEFAULT_YASMD).asm
CLEANSUFFIXES = *.d *.o *~ *.h.c *.map *.ver *.ho *.gcno *.gcda
DISTCLEANSUFFIXES = *.pc
LIBSUFFIXES = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a
define RULES
clean::
$(RM) $(HOSTPROGS) $(TESTPROGS) $(TOOLS)
$(RM) $(OBJS) $(OBJS:.o=.d)
$(RM) $(HOSTPROGS)
$(RM) $(TOOLS)
endef
$(eval $(RULES))
-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d) $(SLIBOBJS:.o=.d)) $(OBJS:.o=$(DEFAULT_YASMD).d)
-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d) $(SLIBOBJS:.o=.d))

View File

@@ -19,8 +19,8 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef COMPAT_AIX_MATH_H
#define COMPAT_AIX_MATH_H
#ifndef FFMPEG_COMPAT_AIX_MATH_H
#define FFMPEG_COMPAT_AIX_MATH_H
#define class class_in_math_h_causes_problems
@@ -28,4 +28,4 @@
#undef class
#endif /* COMPAT_AIX_MATH_H */
#endif /* FFMPEG_COMPAT_AIX_MATH_H */

View File

@@ -38,9 +38,40 @@
#ifndef __AVISYNTH_C__
#define __AVISYNTH_C__
#include "avs/config.h"
#include "avs/capi.h"
#include "avs/types.h"
#ifdef __cplusplus
# define EXTERN_C extern "C"
#else
# define EXTERN_C
#endif
#define AVSC_USE_STDCALL 1
#ifndef AVSC_USE_STDCALL
# define AVSC_CC __cdecl
#else
# define AVSC_CC __stdcall
#endif
#define AVSC_INLINE static __inline
#ifdef AVISYNTH_C_EXPORTS
# define AVSC_EXPORT EXTERN_C
# define AVSC_API(ret, name) EXTERN_C __declspec(dllexport) ret AVSC_CC name
#else
# define AVSC_EXPORT EXTERN_C __declspec(dllexport)
# ifndef AVSC_NO_DECLSPEC
# define AVSC_API(ret, name) EXTERN_C __declspec(dllimport) ret AVSC_CC name
# else
# define AVSC_API(ret, name) typedef ret (AVSC_CC *name##_func)
# endif
#endif
typedef unsigned char BYTE;
#ifdef __GNUC__
typedef long long int INT64;
#else
typedef __int64 INT64;
#endif
/////////////////////////////////////////////////////////////////////
@@ -48,8 +79,8 @@
// Constants
//
#ifndef __AVISYNTH_6_H__
enum { AVISYNTH_INTERFACE_VERSION = 6 };
#ifndef __AVISYNTH_H__
enum { AVISYNTH_INTERFACE_VERSION = 4 };
#endif
enum {AVS_SAMPLE_INT8 = 1<<0,
@@ -75,149 +106,54 @@ enum {AVS_PLANAR_Y=1<<0,
AVS_PLANAR_B_ALIGNED=AVS_PLANAR_B|AVS_PLANAR_ALIGNED};
// Colorspace properties.
enum {
AVS_CS_YUVA = 1 << 27,
AVS_CS_BGR = 1 << 28,
AVS_CS_YUV = 1 << 29,
AVS_CS_INTERLEAVED = 1 << 30,
AVS_CS_PLANAR = 1 << 31,
enum {AVS_CS_BGR = 1<<28,
AVS_CS_YUV = 1<<29,
AVS_CS_INTERLEAVED = 1<<30,
AVS_CS_PLANAR = 1<<31,
AVS_CS_SHIFT_SUB_WIDTH = 0,
AVS_CS_SHIFT_SUB_HEIGHT = 8,
AVS_CS_SHIFT_SAMPLE_BITS = 16,
AVS_CS_SHIFT_SUB_WIDTH = 0,
AVS_CS_SHIFT_SUB_HEIGHT = 1 << 3,
AVS_CS_SHIFT_SAMPLE_BITS = 1 << 4,
AVS_CS_SUB_WIDTH_MASK = 7 << AVS_CS_SHIFT_SUB_WIDTH,
AVS_CS_SUB_WIDTH_1 = 3 << AVS_CS_SHIFT_SUB_WIDTH, // YV24
AVS_CS_SUB_WIDTH_2 = 0 << AVS_CS_SHIFT_SUB_WIDTH, // YV12, I420, YV16
AVS_CS_SUB_WIDTH_4 = 1 << AVS_CS_SHIFT_SUB_WIDTH, // YUV9, YV411
AVS_CS_SUB_WIDTH_MASK = 7 << AVS_CS_SHIFT_SUB_WIDTH,
AVS_CS_SUB_WIDTH_1 = 3 << AVS_CS_SHIFT_SUB_WIDTH, // YV24
AVS_CS_SUB_WIDTH_2 = 0 << AVS_CS_SHIFT_SUB_WIDTH, // YV12, I420, YV16
AVS_CS_SUB_WIDTH_4 = 1 << AVS_CS_SHIFT_SUB_WIDTH, // YUV9, YV411
AVS_CS_VPLANEFIRST = 1 << 3, // YV12, YV16, YV24, YV411, YUV9
AVS_CS_UPLANEFIRST = 1 << 4, // I420
AVS_CS_VPLANEFIRST = 1 << 3, // YV12, YV16, YV24, YV411, YUV9
AVS_CS_UPLANEFIRST = 1 << 4, // I420
AVS_CS_SUB_HEIGHT_MASK = 7 << AVS_CS_SHIFT_SUB_HEIGHT,
AVS_CS_SUB_HEIGHT_1 = 3 << AVS_CS_SHIFT_SUB_HEIGHT, // YV16, YV24, YV411
AVS_CS_SUB_HEIGHT_2 = 0 << AVS_CS_SHIFT_SUB_HEIGHT, // YV12, I420
AVS_CS_SUB_HEIGHT_4 = 1 << AVS_CS_SHIFT_SUB_HEIGHT, // YUV9
AVS_CS_SUB_HEIGHT_MASK = 7 << AVS_CS_SHIFT_SUB_HEIGHT,
AVS_CS_SUB_HEIGHT_1 = 3 << AVS_CS_SHIFT_SUB_HEIGHT, // YV16, YV24, YV411
AVS_CS_SUB_HEIGHT_2 = 0 << AVS_CS_SHIFT_SUB_HEIGHT, // YV12, I420
AVS_CS_SUB_HEIGHT_4 = 1 << AVS_CS_SHIFT_SUB_HEIGHT, // YUV9
AVS_CS_SAMPLE_BITS_MASK = 7 << AVS_CS_SHIFT_SAMPLE_BITS,
AVS_CS_SAMPLE_BITS_8 = 0 << AVS_CS_SHIFT_SAMPLE_BITS,
AVS_CS_SAMPLE_BITS_10 = 5 << AVS_CS_SHIFT_SAMPLE_BITS,
AVS_CS_SAMPLE_BITS_12 = 6 << AVS_CS_SHIFT_SAMPLE_BITS,
AVS_CS_SAMPLE_BITS_14 = 7 << AVS_CS_SHIFT_SAMPLE_BITS,
AVS_CS_SAMPLE_BITS_16 = 1 << AVS_CS_SHIFT_SAMPLE_BITS,
AVS_CS_SAMPLE_BITS_32 = 2 << AVS_CS_SHIFT_SAMPLE_BITS,
AVS_CS_PLANAR_MASK = AVS_CS_PLANAR | AVS_CS_INTERLEAVED | AVS_CS_YUV | AVS_CS_BGR | AVS_CS_YUVA | AVS_CS_SAMPLE_BITS_MASK | AVS_CS_SUB_HEIGHT_MASK | AVS_CS_SUB_WIDTH_MASK,
AVS_CS_PLANAR_FILTER = ~(AVS_CS_VPLANEFIRST | AVS_CS_UPLANEFIRST),
AVS_CS_RGB_TYPE = 1 << 0,
AVS_CS_RGBA_TYPE = 1 << 1,
AVS_CS_GENERIC_YUV420 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_2 | AVS_CS_SUB_WIDTH_2, // 4:2:0 planar
AVS_CS_GENERIC_YUV422 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_2, // 4:2:2 planar
AVS_CS_GENERIC_YUV444 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_1, // 4:4:4 planar
AVS_CS_GENERIC_Y = AVS_CS_PLANAR | AVS_CS_INTERLEAVED | AVS_CS_YUV, // Y only (4:0:0)
AVS_CS_GENERIC_RGBP = AVS_CS_PLANAR | AVS_CS_BGR | AVS_CS_RGB_TYPE, // planar RGB
AVS_CS_GENERIC_RGBAP = AVS_CS_PLANAR | AVS_CS_BGR | AVS_CS_RGBA_TYPE, // planar RGBA
AVS_CS_GENERIC_YUVA420 = AVS_CS_PLANAR | AVS_CS_YUVA | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_2 | AVS_CS_SUB_WIDTH_2, // 4:2:0:A planar
AVS_CS_GENERIC_YUVA422 = AVS_CS_PLANAR | AVS_CS_YUVA | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_2, // 4:2:2:A planar
AVS_CS_GENERIC_YUVA444 = AVS_CS_PLANAR | AVS_CS_YUVA | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_1 }; // 4:4:4:A planar
AVS_CS_SAMPLE_BITS_MASK = 7 << AVS_CS_SHIFT_SAMPLE_BITS,
AVS_CS_SAMPLE_BITS_8 = 0 << AVS_CS_SHIFT_SAMPLE_BITS,
AVS_CS_SAMPLE_BITS_16 = 1 << AVS_CS_SHIFT_SAMPLE_BITS,
AVS_CS_SAMPLE_BITS_32 = 2 << AVS_CS_SHIFT_SAMPLE_BITS,
AVS_CS_PLANAR_MASK = AVS_CS_PLANAR | AVS_CS_INTERLEAVED | AVS_CS_YUV | AVS_CS_BGR | AVS_CS_SAMPLE_BITS_MASK | AVS_CS_SUB_HEIGHT_MASK | AVS_CS_SUB_WIDTH_MASK,
AVS_CS_PLANAR_FILTER = ~( AVS_CS_VPLANEFIRST | AVS_CS_UPLANEFIRST )};
// Specific colorformats
enum {
AVS_CS_UNKNOWN = 0,
AVS_CS_BGR24 = AVS_CS_RGB_TYPE | AVS_CS_BGR | AVS_CS_INTERLEAVED,
AVS_CS_BGR32 = AVS_CS_RGBA_TYPE | AVS_CS_BGR | AVS_CS_INTERLEAVED,
AVS_CS_BGR24 = 1<<0 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
AVS_CS_BGR32 = 1<<1 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
AVS_CS_YUY2 = 1<<2 | AVS_CS_YUV | AVS_CS_INTERLEAVED,
// AVS_CS_YV12 = 1<<3 Reserved
// AVS_CS_I420 = 1<<4 Reserved
AVS_CS_RAW32 = 1<<5 | AVS_CS_INTERLEAVED,
AVS_CS_YV24 = AVS_CS_GENERIC_YUV444 | AVS_CS_SAMPLE_BITS_8, // YVU 4:4:4 planar
AVS_CS_YV16 = AVS_CS_GENERIC_YUV422 | AVS_CS_SAMPLE_BITS_8, // YVU 4:2:2 planar
AVS_CS_YV12 = AVS_CS_GENERIC_YUV420 | AVS_CS_SAMPLE_BITS_8, // YVU 4:2:0 planar
AVS_CS_YV24 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_1, // YVU 4:4:4 planar
AVS_CS_YV16 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_2, // YVU 4:2:2 planar
AVS_CS_YV12 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_2 | AVS_CS_SUB_WIDTH_2, // YVU 4:2:0 planar
AVS_CS_I420 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_UPLANEFIRST | AVS_CS_SUB_HEIGHT_2 | AVS_CS_SUB_WIDTH_2, // YUV 4:2:0 planar
AVS_CS_IYUV = AVS_CS_I420,
AVS_CS_YV411 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_4, // YVU 4:1:1 planar
AVS_CS_YUV9 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_4 | AVS_CS_SUB_WIDTH_4, // YVU 4:1:0 planar
AVS_CS_Y8 = AVS_CS_GENERIC_Y | AVS_CS_SAMPLE_BITS_8, // Y 4:0:0 planar
//-------------------------
// AVS16: new planar constants go live! Experimental PF 160613
// 10-12-14 bit + planar RGB + BRG48/64 160725
AVS_CS_YUV444P10 = AVS_CS_GENERIC_YUV444 | AVS_CS_SAMPLE_BITS_10, // YUV 4:4:4 10bit samples
AVS_CS_YUV422P10 = AVS_CS_GENERIC_YUV422 | AVS_CS_SAMPLE_BITS_10, // YUV 4:2:2 10bit samples
AVS_CS_YUV420P10 = AVS_CS_GENERIC_YUV420 | AVS_CS_SAMPLE_BITS_10, // YUV 4:2:0 10bit samples
AVS_CS_Y10 = AVS_CS_GENERIC_Y | AVS_CS_SAMPLE_BITS_10, // Y 4:0:0 10bit samples
AVS_CS_YUV444P12 = AVS_CS_GENERIC_YUV444 | AVS_CS_SAMPLE_BITS_12, // YUV 4:4:4 12bit samples
AVS_CS_YUV422P12 = AVS_CS_GENERIC_YUV422 | AVS_CS_SAMPLE_BITS_12, // YUV 4:2:2 12bit samples
AVS_CS_YUV420P12 = AVS_CS_GENERIC_YUV420 | AVS_CS_SAMPLE_BITS_12, // YUV 4:2:0 12bit samples
AVS_CS_Y12 = AVS_CS_GENERIC_Y | AVS_CS_SAMPLE_BITS_12, // Y 4:0:0 12bit samples
AVS_CS_YUV444P14 = AVS_CS_GENERIC_YUV444 | AVS_CS_SAMPLE_BITS_14, // YUV 4:4:4 14bit samples
AVS_CS_YUV422P14 = AVS_CS_GENERIC_YUV422 | AVS_CS_SAMPLE_BITS_14, // YUV 4:2:2 14bit samples
AVS_CS_YUV420P14 = AVS_CS_GENERIC_YUV420 | AVS_CS_SAMPLE_BITS_14, // YUV 4:2:0 14bit samples
AVS_CS_Y14 = AVS_CS_GENERIC_Y | AVS_CS_SAMPLE_BITS_14, // Y 4:0:0 14bit samples
AVS_CS_YUV444P16 = AVS_CS_GENERIC_YUV444 | AVS_CS_SAMPLE_BITS_16, // YUV 4:4:4 16bit samples
AVS_CS_YUV422P16 = AVS_CS_GENERIC_YUV422 | AVS_CS_SAMPLE_BITS_16, // YUV 4:2:2 16bit samples
AVS_CS_YUV420P16 = AVS_CS_GENERIC_YUV420 | AVS_CS_SAMPLE_BITS_16, // YUV 4:2:0 16bit samples
AVS_CS_Y16 = AVS_CS_GENERIC_Y | AVS_CS_SAMPLE_BITS_16, // Y 4:0:0 16bit samples
// 32 bit samples (float)
AVS_CS_YUV444PS = AVS_CS_GENERIC_YUV444 | AVS_CS_SAMPLE_BITS_32, // YUV 4:4:4 32bit samples
AVS_CS_YUV422PS = AVS_CS_GENERIC_YUV422 | AVS_CS_SAMPLE_BITS_32, // YUV 4:2:2 32bit samples
AVS_CS_YUV420PS = AVS_CS_GENERIC_YUV420 | AVS_CS_SAMPLE_BITS_32, // YUV 4:2:0 32bit samples
AVS_CS_Y32 = AVS_CS_GENERIC_Y | AVS_CS_SAMPLE_BITS_32, // Y 4:0:0 32bit samples
// RGB packed
AVS_CS_BGR48 = AVS_CS_RGB_TYPE | AVS_CS_BGR | AVS_CS_INTERLEAVED | AVS_CS_SAMPLE_BITS_16, // BGR 3x16 bit
AVS_CS_BGR64 = AVS_CS_RGBA_TYPE | AVS_CS_BGR | AVS_CS_INTERLEAVED | AVS_CS_SAMPLE_BITS_16, // BGR 4x16 bit
// no packed 32 bit (float) support for these legacy types
// RGB planar
AVS_CS_RGBP = AVS_CS_GENERIC_RGBP | AVS_CS_SAMPLE_BITS_8, // Planar RGB 8 bit samples
AVS_CS_RGBP10 = AVS_CS_GENERIC_RGBP | AVS_CS_SAMPLE_BITS_10, // Planar RGB 10bit samples
AVS_CS_RGBP12 = AVS_CS_GENERIC_RGBP | AVS_CS_SAMPLE_BITS_12, // Planar RGB 12bit samples
AVS_CS_RGBP14 = AVS_CS_GENERIC_RGBP | AVS_CS_SAMPLE_BITS_14, // Planar RGB 14bit samples
AVS_CS_RGBP16 = AVS_CS_GENERIC_RGBP | AVS_CS_SAMPLE_BITS_16, // Planar RGB 16bit samples
AVS_CS_RGBPS = AVS_CS_GENERIC_RGBP | AVS_CS_SAMPLE_BITS_32, // Planar RGB 32bit samples
// RGBA planar
AVS_CS_RGBAP = AVS_CS_GENERIC_RGBAP | AVS_CS_SAMPLE_BITS_8, // Planar RGBA 8 bit samples
AVS_CS_RGBAP10 = AVS_CS_GENERIC_RGBAP | AVS_CS_SAMPLE_BITS_10, // Planar RGBA 10bit samples
AVS_CS_RGBAP12 = AVS_CS_GENERIC_RGBAP | AVS_CS_SAMPLE_BITS_12, // Planar RGBA 12bit samples
AVS_CS_RGBAP14 = AVS_CS_GENERIC_RGBAP | AVS_CS_SAMPLE_BITS_14, // Planar RGBA 14bit samples
AVS_CS_RGBAP16 = AVS_CS_GENERIC_RGBAP | AVS_CS_SAMPLE_BITS_16, // Planar RGBA 16bit samples
AVS_CS_RGBAPS = AVS_CS_GENERIC_RGBAP | AVS_CS_SAMPLE_BITS_32, // Planar RGBA 32bit samples
// Planar YUVA
AVS_CS_YUVA444 = AVS_CS_GENERIC_YUVA444 | AVS_CS_SAMPLE_BITS_8, // YUVA 4:4:4 8bit samples
AVS_CS_YUVA422 = AVS_CS_GENERIC_YUVA422 | AVS_CS_SAMPLE_BITS_8, // YUVA 4:2:2 8bit samples
AVS_CS_YUVA420 = AVS_CS_GENERIC_YUVA420 | AVS_CS_SAMPLE_BITS_8, // YUVA 4:2:0 8bit samples
AVS_CS_YUVA444P10 = AVS_CS_GENERIC_YUVA444 | AVS_CS_SAMPLE_BITS_10, // YUVA 4:4:4 10bit samples
AVS_CS_YUVA422P10 = AVS_CS_GENERIC_YUVA422 | AVS_CS_SAMPLE_BITS_10, // YUVA 4:2:2 10bit samples
AVS_CS_YUVA420P10 = AVS_CS_GENERIC_YUVA420 | AVS_CS_SAMPLE_BITS_10, // YUVA 4:2:0 10bit samples
AVS_CS_YUVA444P12 = AVS_CS_GENERIC_YUVA444 | AVS_CS_SAMPLE_BITS_12, // YUVA 4:4:4 12bit samples
AVS_CS_YUVA422P12 = AVS_CS_GENERIC_YUVA422 | AVS_CS_SAMPLE_BITS_12, // YUVA 4:2:2 12bit samples
AVS_CS_YUVA420P12 = AVS_CS_GENERIC_YUVA420 | AVS_CS_SAMPLE_BITS_12, // YUVA 4:2:0 12bit samples
AVS_CS_YUVA444P14 = AVS_CS_GENERIC_YUVA444 | AVS_CS_SAMPLE_BITS_14, // YUVA 4:4:4 14bit samples
AVS_CS_YUVA422P14 = AVS_CS_GENERIC_YUVA422 | AVS_CS_SAMPLE_BITS_14, // YUVA 4:2:2 14bit samples
AVS_CS_YUVA420P14 = AVS_CS_GENERIC_YUVA420 | AVS_CS_SAMPLE_BITS_14, // YUVA 4:2:0 14bit samples
AVS_CS_YUVA444P16 = AVS_CS_GENERIC_YUVA444 | AVS_CS_SAMPLE_BITS_16, // YUVA 4:4:4 16bit samples
AVS_CS_YUVA422P16 = AVS_CS_GENERIC_YUVA422 | AVS_CS_SAMPLE_BITS_16, // YUVA 4:2:2 16bit samples
AVS_CS_YUVA420P16 = AVS_CS_GENERIC_YUVA420 | AVS_CS_SAMPLE_BITS_16, // YUVA 4:2:0 16bit samples
AVS_CS_YUVA444PS = AVS_CS_GENERIC_YUVA444 | AVS_CS_SAMPLE_BITS_32, // YUVA 4:4:4 32bit samples
AVS_CS_YUVA422PS = AVS_CS_GENERIC_YUVA422 | AVS_CS_SAMPLE_BITS_32, // YUVA 4:2:2 32bit samples
AVS_CS_YUVA420PS = AVS_CS_GENERIC_YUVA420 | AVS_CS_SAMPLE_BITS_32, // YUVA 4:2:0 32bit samples
AVS_CS_Y8 = AVS_CS_PLANAR | AVS_CS_INTERLEAVED | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 // Y 4:0:0 planar
};
enum {
@@ -244,66 +180,15 @@ enum { //SUBTYPES
AVS_FILTER_OUTPUT_TYPE_DIFFERENT=4};
enum {
// New 2.6 explicitly defined cache hints.
AVS_CACHE_NOTHING=10, // Do not cache video.
AVS_CACHE_WINDOW=11, // Hard protect upto X frames within a range of X from the current frame N.
AVS_CACHE_GENERIC=12, // LRU cache upto X frames.
AVS_CACHE_FORCE_GENERIC=13, // LRU cache upto X frames, override any previous CACHE_WINDOW.
AVS_CACHE_GET_POLICY=30, // Get the current policy.
AVS_CACHE_GET_WINDOW=31, // Get the current window h_span.
AVS_CACHE_GET_RANGE=32, // Get the current generic frame range.
AVS_CACHE_AUDIO=50, // Explicitly do cache audio, X byte cache.
AVS_CACHE_AUDIO_NOTHING=51, // Explicitly do not cache audio.
AVS_CACHE_AUDIO_NONE=52, // Audio cache off (auto mode), X byte intial cache.
AVS_CACHE_AUDIO_AUTO=53, // Audio cache on (auto mode), X byte intial cache.
AVS_CACHE_GET_AUDIO_POLICY=70, // Get the current audio policy.
AVS_CACHE_GET_AUDIO_SIZE=71, // Get the current audio cache size.
AVS_CACHE_PREFETCH_FRAME=100, // Queue request to prefetch frame N.
AVS_CACHE_PREFETCH_GO=101, // Action video prefetches.
AVS_CACHE_PREFETCH_AUDIO_BEGIN=120, // Begin queue request transaction to prefetch audio (take critical section).
AVS_CACHE_PREFETCH_AUDIO_STARTLO=121, // Set low 32 bits of start.
AVS_CACHE_PREFETCH_AUDIO_STARTHI=122, // Set high 32 bits of start.
AVS_CACHE_PREFETCH_AUDIO_COUNT=123, // Set low 32 bits of length.
AVS_CACHE_PREFETCH_AUDIO_COMMIT=124, // Enqueue request transaction to prefetch audio (release critical section).
AVS_CACHE_PREFETCH_AUDIO_GO=125, // Action audio prefetches.
AVS_CACHE_GETCHILD_CACHE_MODE=200, // Cache ask Child for desired video cache mode.
AVS_CACHE_GETCHILD_CACHE_SIZE=201, // Cache ask Child for desired video cache size.
AVS_CACHE_GETCHILD_AUDIO_MODE=202, // Cache ask Child for desired audio cache mode.
AVS_CACHE_GETCHILD_AUDIO_SIZE=203, // Cache ask Child for desired audio cache size.
AVS_CACHE_GETCHILD_COST=220, // Cache ask Child for estimated processing cost.
AVS_CACHE_COST_ZERO=221, // Child response of zero cost (ptr arithmetic only).
AVS_CACHE_COST_UNIT=222, // Child response of unit cost (less than or equal 1 full frame blit).
AVS_CACHE_COST_LOW=223, // Child response of light cost. (Fast)
AVS_CACHE_COST_MED=224, // Child response of medium cost. (Real time)
AVS_CACHE_COST_HI=225, // Child response of heavy cost. (Slow)
AVS_CACHE_GETCHILD_THREAD_MODE=240, // Cache ask Child for thread safetyness.
AVS_CACHE_THREAD_UNSAFE=241, // Only 1 thread allowed for all instances. 2.5 filters default!
AVS_CACHE_THREAD_CLASS=242, // Only 1 thread allowed for each instance. 2.6 filters default!
AVS_CACHE_THREAD_SAFE=243, // Allow all threads in any instance.
AVS_CACHE_THREAD_OWN=244, // Safe but limit to 1 thread, internally threaded.
AVS_CACHE_GETCHILD_ACCESS_COST=260, // Cache ask Child for preferred access pattern.
AVS_CACHE_ACCESS_RAND=261, // Filter is access order agnostic.
AVS_CACHE_ACCESS_SEQ0=262, // Filter prefers sequential access (low cost)
AVS_CACHE_ACCESS_SEQ1=263, // Filter needs sequential access (high cost)
AVS_CACHE_NOTHING=0,
AVS_CACHE_RANGE=1,
AVS_CACHE_ALL=2,
AVS_CACHE_AUDIO=3,
AVS_CACHE_AUDIO_NONE=4,
AVS_CACHE_AUDIO_AUTO=5
};
#ifdef BUILDING_AVSCORE
struct AVS_ScriptEnvironment {
IScriptEnvironment * env;
const char * error;
AVS_ScriptEnvironment(IScriptEnvironment * e = 0)
: env(e), error(0) {}
};
#endif
#define AVS_FRAME_ALIGN 16
typedef struct AVS_Clip AVS_Clip;
typedef struct AVS_ScriptEnvironment AVS_ScriptEnvironment;
@@ -342,10 +227,10 @@ AVSC_INLINE int avs_is_rgb(const AVS_VideoInfo * p)
{ return !!(p->pixel_type&AVS_CS_BGR); }
AVSC_INLINE int avs_is_rgb24(const AVS_VideoInfo * p)
{ return ((p->pixel_type&AVS_CS_BGR24)==AVS_CS_BGR24) && ((p->pixel_type & AVS_CS_SAMPLE_BITS_MASK) == AVS_CS_SAMPLE_BITS_8); }
{ return (p->pixel_type&AVS_CS_BGR24)==AVS_CS_BGR24; } // Clear out additional properties
AVSC_INLINE int avs_is_rgb32(const AVS_VideoInfo * p)
{ return ((p->pixel_type&AVS_CS_BGR32)==AVS_CS_BGR32) && ((p->pixel_type & AVS_CS_SAMPLE_BITS_MASK) == AVS_CS_SAMPLE_BITS_8); }
{ return (p->pixel_type & AVS_CS_BGR32) == AVS_CS_BGR32 ; }
AVSC_INLINE int avs_is_yuv(const AVS_VideoInfo * p)
{ return !!(p->pixel_type&AVS_CS_YUV ); }
@@ -353,59 +238,29 @@ AVSC_INLINE int avs_is_yuv(const AVS_VideoInfo * p)
AVSC_INLINE int avs_is_yuy2(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_YUY2) == AVS_CS_YUY2; }
AVSC_API(int, avs_is_rgb48)(const AVS_VideoInfo * p);
AVSC_INLINE int avs_is_yv24(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV24 & AVS_CS_PLANAR_FILTER); }
AVSC_API(int, avs_is_rgb64)(const AVS_VideoInfo * p);
AVSC_INLINE int avs_is_yv16(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV16 & AVS_CS_PLANAR_FILTER); }
AVSC_API(int, avs_is_yv24)(const AVS_VideoInfo * p);
AVSC_API(int, avs_is_yv16)(const AVS_VideoInfo * p);
AVSC_API(int, avs_is_yv12)(const AVS_VideoInfo * p) ;
AVSC_API(int, avs_is_yv411)(const AVS_VideoInfo * p);
AVSC_API(int, avs_is_y8)(const AVS_VideoInfo * p);
AVSC_API(int, avs_is_yuv444p16)(const AVS_VideoInfo * p);
AVSC_API(int, avs_is_yuv422p16)(const AVS_VideoInfo * p);
AVSC_API(int, avs_is_yuv420p16)(const AVS_VideoInfo * p);
AVSC_API(int, avs_is_y16)(const AVS_VideoInfo * p);
AVSC_API(int, avs_is_yuv444ps)(const AVS_VideoInfo * p);
AVSC_API(int, avs_is_yuv422ps)(const AVS_VideoInfo * p);
AVSC_API(int, avs_is_yuv420ps)(const AVS_VideoInfo * p);
AVSC_API(int, avs_is_y32)(const AVS_VideoInfo * p);
AVSC_API(int, avs_is_444)(const AVS_VideoInfo * p);
AVSC_API(int, avs_is_422)(const AVS_VideoInfo * p);
AVSC_API(int, avs_is_420)(const AVS_VideoInfo * p);
AVSC_API(int, avs_is_y)(const AVS_VideoInfo * p);
AVSC_API(int, avs_is_yuva)(const AVS_VideoInfo * p);
AVSC_API(int, avs_is_planar_rgb)(const AVS_VideoInfo * p);
AVSC_API(int, avs_is_planar_rgba)(const AVS_VideoInfo * p);
AVSC_INLINE int avs_is_yv12(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV12 & AVS_CS_PLANAR_FILTER); }
AVSC_INLINE int avs_is_yv411(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV411 & AVS_CS_PLANAR_FILTER); }
AVSC_INLINE int avs_is_y8(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_Y8 & AVS_CS_PLANAR_FILTER); }
AVSC_INLINE int avs_is_property(const AVS_VideoInfo * p, int property)
{ return ((p->image_type & property)==property ); }
{ return ((p->pixel_type & property)==property ); }
AVSC_INLINE int avs_is_planar(const AVS_VideoInfo * p)
{ return !!(p->pixel_type & AVS_CS_PLANAR); }
AVSC_API(int, avs_is_color_space)(const AVS_VideoInfo * p, int c_space);
AVSC_INLINE int avs_is_color_space(const AVS_VideoInfo * p, int c_space)
{ return avs_is_planar(p) ? ((p->pixel_type & AVS_CS_PLANAR_MASK) == (c_space & AVS_CS_PLANAR_FILTER)) : ((p->pixel_type & c_space) == c_space); }
AVSC_INLINE int avs_is_field_based(const AVS_VideoInfo * p)
{ return !!(p->image_type & AVS_IT_FIELDBASED); }
@@ -419,18 +274,25 @@ AVSC_INLINE int avs_is_bff(const AVS_VideoInfo * p)
AVSC_INLINE int avs_is_tff(const AVS_VideoInfo * p)
{ return !!(p->image_type & AVS_IT_TFF); }
AVSC_API(int, avs_get_plane_width_subsampling)(const AVS_VideoInfo * p, int plane);
AVSC_INLINE int avs_bits_per_pixel(const AVS_VideoInfo * p)
{
switch (p->pixel_type) {
case AVS_CS_BGR24: return 24;
case AVS_CS_BGR32: return 32;
case AVS_CS_YUY2: return 16;
case AVS_CS_YV12:
case AVS_CS_I420: return 12;
default: return 0;
}
}
AVSC_INLINE int avs_bytes_from_pixels(const AVS_VideoInfo * p, int pixels)
{ return pixels * (avs_bits_per_pixel(p)>>3); } // Will work on planar images, but will return only luma planes
AVSC_API(int, avs_get_plane_height_subsampling)(const AVS_VideoInfo * p, int plane);
AVSC_INLINE int avs_row_size(const AVS_VideoInfo * p)
{ return avs_bytes_from_pixels(p,p->width); } // Also only returns first plane on planar images
AVSC_API(int, avs_bits_per_pixel)(const AVS_VideoInfo * p);
AVSC_API(int, avs_bytes_from_pixels)(const AVS_VideoInfo * p, int pixels);
AVSC_API(int, avs_row_size)(const AVS_VideoInfo * p, int plane);
AVSC_API(int, avs_bmp_size)(const AVS_VideoInfo * vi);
AVSC_INLINE int avs_bmp_size(const AVS_VideoInfo * vi)
{ if (avs_is_planar(vi)) {int p = vi->height * ((avs_row_size(vi)+3) & ~3); p+=p>>1; return p; } return vi->height * ((avs_row_size(vi)+3) & ~3); }
AVSC_INLINE int avs_samples_per_second(const AVS_VideoInfo * p)
{ return p->audio_samples_per_second; }
@@ -488,19 +350,11 @@ AVSC_INLINE void avs_set_fps(AVS_VideoInfo * p, unsigned numerator, unsigned den
p->fps_denominator = denominator/x;
}
#ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
AVSC_INLINE int avs_is_same_colorspace(AVS_VideoInfo * x, AVS_VideoInfo * y)
{
return (x->pixel_type == y->pixel_type)
|| (avs_is_yv12(x) && avs_is_yv12(y));
}
#endif
AVSC_API(int, avs_num_components)(const AVS_VideoInfo * p);
AVSC_API(int, avs_component_size)(const AVS_VideoInfo * p);
AVSC_API(int, avs_bits_per_component)(const AVS_VideoInfo * p);
/////////////////////////////////////////////////////////////////////
//
@@ -537,38 +391,89 @@ typedef struct AVS_VideoFrame {
} AVS_VideoFrame;
// Access functions for AVS_VideoFrame
AVSC_API(int, avs_get_pitch_p)(const AVS_VideoFrame * p, int plane);
#ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
AVSC_INLINE int avs_get_pitch(const AVS_VideoFrame * p) {
return avs_get_pitch_p(p, 0);}
#endif
return p->pitch;}
AVSC_API(int, avs_get_row_size_p)(const AVS_VideoFrame * p, int plane);
AVSC_INLINE int avs_get_pitch_p(const AVS_VideoFrame * p, int plane) {
switch (plane) {
case AVS_PLANAR_U: case AVS_PLANAR_V: return p->pitchUV;}
return p->pitch;}
AVSC_INLINE int avs_get_row_size(const AVS_VideoFrame * p) {
return p->row_size; }
AVSC_API(int, avs_get_height_p)(const AVS_VideoFrame * p, int plane);
AVSC_INLINE int avs_get_row_size_p(const AVS_VideoFrame * p, int plane) {
int r;
switch (plane) {
case AVS_PLANAR_U: case AVS_PLANAR_V:
if (p->pitchUV) return p->row_sizeUV;
else return 0;
case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED:
if (p->pitchUV) {
r = (p->row_sizeUV+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
if (r < p->pitchUV)
return r;
return p->row_sizeUV;
} else return 0;
case AVS_PLANAR_Y_ALIGNED:
r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
if (r <= p->pitch)
return r;
return p->row_size;
}
return p->row_size;
}
AVSC_INLINE int avs_get_height(const AVS_VideoFrame * p) {
return p->height;}
AVSC_API(const BYTE *, avs_get_read_ptr_p)(const AVS_VideoFrame * p, int plane);
AVSC_INLINE int avs_get_height_p(const AVS_VideoFrame * p, int plane) {
switch (plane) {
case AVS_PLANAR_U: case AVS_PLANAR_V:
if (p->pitchUV) return p->heightUV;
return 0;
}
return p->height;}
#ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
AVSC_INLINE const BYTE* avs_get_read_ptr(const AVS_VideoFrame * p) {
return avs_get_read_ptr_p(p, 0);}
#endif
return p->vfb->data + p->offset;}
AVSC_API(int, avs_is_writable)(const AVS_VideoFrame * p);
AVSC_INLINE const BYTE* avs_get_read_ptr_p(const AVS_VideoFrame * p, int plane)
{
switch (plane) {
case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
default: return p->vfb->data + p->offset;}
}
AVSC_API(BYTE *, avs_get_write_ptr_p)(const AVS_VideoFrame * p, int plane);
AVSC_INLINE int avs_is_writable(const AVS_VideoFrame * p) {
return (p->refcount == 1 && p->vfb->refcount == 1);}
AVSC_INLINE BYTE* avs_get_write_ptr(const AVS_VideoFrame * p)
{
if (avs_is_writable(p)) {
++p->vfb->sequence_number;
return p->vfb->data + p->offset;
} else
return 0;
}
AVSC_INLINE BYTE* avs_get_write_ptr_p(const AVS_VideoFrame * p, int plane)
{
if (plane==AVS_PLANAR_Y && avs_is_writable(p)) {
++p->vfb->sequence_number;
return p->vfb->data + p->offset;
} else if (plane==AVS_PLANAR_Y) {
return 0;
} else {
switch (plane) {
case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
default: return p->vfb->data + p->offset;
}
}
}
#ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
AVSC_INLINE BYTE* avs_get_write_ptr(const AVS_VideoFrame * p) {
return avs_get_write_ptr_p(p, 0);}
#endif
AVSC_API(void, avs_release_video_frame)(AVS_VideoFrame *);
// makes a shallow copy of a video frame
@@ -665,7 +570,7 @@ AVSC_INLINE AVS_Value avs_new_value_clip(AVS_Clip * v0)
{ AVS_Value v; avs_set_to_clip(&v, v0); return v; }
#endif
AVSC_INLINE AVS_Value avs_new_value_array(AVS_Value * v0, int size)
{ AVS_Value v; v.type = 'a'; v.d.array = v0; v.array_size = (short)size; return v; }
{ AVS_Value v; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; }
/////////////////////////////////////////////////////////////////////
//
@@ -753,16 +658,12 @@ enum {
AVS_CPUF_SSSE3 = 0x200, // Core 2
AVS_CPUF_SSE4 = 0x400, // Penryn, Wolfdale, Yorkfield
AVS_CPUF_SSE4_1 = 0x400,
//AVS_CPUF_AVX = 0x800, // Sandy Bridge, Bulldozer
AVS_CPUF_SSE4_2 = 0x1000, // Nehalem
//AVS_CPUF_AVX2 = 0x2000, // Haswell
//AVS_CPUF_AVX512 = 0x4000, // Knights Landing
AVS_CPUF_SSE4_2 = 0x800, // Nehalem
};
AVSC_API(const char *, avs_get_error)(AVS_ScriptEnvironment *); // return 0 if no error
AVSC_API(int, avs_get_cpu_flags)(AVS_ScriptEnvironment *);
AVSC_API(long, avs_get_cpu_flags)(AVS_ScriptEnvironment *);
AVSC_API(int, avs_check_version)(AVS_ScriptEnvironment *, int version);
AVSC_API(char *, avs_save_string)(AVS_ScriptEnvironment *, const char* s, int length);
@@ -799,12 +700,12 @@ AVSC_API(AVS_VideoFrame *, avs_new_video_frame_a)(AVS_ScriptEnvironment *,
AVSC_INLINE
AVS_VideoFrame * avs_new_video_frame(AVS_ScriptEnvironment * env,
const AVS_VideoInfo * vi)
{return avs_new_video_frame_a(env,vi,FRAME_ALIGN);}
{return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
AVSC_INLINE
AVS_VideoFrame * avs_new_frame(AVS_ScriptEnvironment * env,
const AVS_VideoInfo * vi)
{return avs_new_video_frame_a(env,vi,FRAME_ALIGN);}
{return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
#endif
@@ -872,6 +773,7 @@ struct AVS_Library {
AVSC_DECLARE_FUNC(avs_function_exists);
AVSC_DECLARE_FUNC(avs_get_audio);
AVSC_DECLARE_FUNC(avs_get_cpu_flags);
AVSC_DECLARE_FUNC(avs_get_error);
AVSC_DECLARE_FUNC(avs_get_frame);
AVSC_DECLARE_FUNC(avs_get_parity);
AVSC_DECLARE_FUNC(avs_get_var);
@@ -896,49 +798,6 @@ struct AVS_Library {
AVSC_DECLARE_FUNC(avs_subframe_planar);
AVSC_DECLARE_FUNC(avs_take_clip);
AVSC_DECLARE_FUNC(avs_vsprintf);
AVSC_DECLARE_FUNC(avs_get_error);
AVSC_DECLARE_FUNC(avs_is_rgb48);
AVSC_DECLARE_FUNC(avs_is_rgb64);
AVSC_DECLARE_FUNC(avs_is_yv24);
AVSC_DECLARE_FUNC(avs_is_yv16);
AVSC_DECLARE_FUNC(avs_is_yv12);
AVSC_DECLARE_FUNC(avs_is_yv411);
AVSC_DECLARE_FUNC(avs_is_y8);
AVSC_DECLARE_FUNC(avs_is_yuv444p16);
AVSC_DECLARE_FUNC(avs_is_yuv422p16);
AVSC_DECLARE_FUNC(avs_is_yuv420p16);
AVSC_DECLARE_FUNC(avs_is_y16);
AVSC_DECLARE_FUNC(avs_is_yuv444ps);
AVSC_DECLARE_FUNC(avs_is_yuv422ps);
AVSC_DECLARE_FUNC(avs_is_yuv420ps);
AVSC_DECLARE_FUNC(avs_is_y32);
AVSC_DECLARE_FUNC(avs_is_444);
AVSC_DECLARE_FUNC(avs_is_422);
AVSC_DECLARE_FUNC(avs_is_420);
AVSC_DECLARE_FUNC(avs_is_y);
AVSC_DECLARE_FUNC(avs_is_yuva);
AVSC_DECLARE_FUNC(avs_is_planar_rgb);
AVSC_DECLARE_FUNC(avs_is_planar_rgba);
AVSC_DECLARE_FUNC(avs_is_color_space);
AVSC_DECLARE_FUNC(avs_get_plane_width_subsampling);
AVSC_DECLARE_FUNC(avs_get_plane_height_subsampling);
AVSC_DECLARE_FUNC(avs_bits_per_pixel);
AVSC_DECLARE_FUNC(avs_bytes_from_pixels);
AVSC_DECLARE_FUNC(avs_row_size);
AVSC_DECLARE_FUNC(avs_bmp_size);
AVSC_DECLARE_FUNC(avs_get_pitch_p);
AVSC_DECLARE_FUNC(avs_get_row_size_p);
AVSC_DECLARE_FUNC(avs_get_height_p);
AVSC_DECLARE_FUNC(avs_get_read_ptr_p);
AVSC_DECLARE_FUNC(avs_is_writable);
AVSC_DECLARE_FUNC(avs_get_write_ptr_p);
AVSC_DECLARE_FUNC(avs_num_components);
AVSC_DECLARE_FUNC(avs_component_size);
AVSC_DECLARE_FUNC(avs_bits_per_component);
};
#undef AVSC_DECLARE_FUNC
@@ -973,6 +832,7 @@ AVSC_INLINE AVS_Library * avs_load_library() {
AVSC_LOAD_FUNC(avs_function_exists);
AVSC_LOAD_FUNC(avs_get_audio);
AVSC_LOAD_FUNC(avs_get_cpu_flags);
AVSC_LOAD_FUNC(avs_get_error);
AVSC_LOAD_FUNC(avs_get_frame);
AVSC_LOAD_FUNC(avs_get_parity);
AVSC_LOAD_FUNC(avs_get_var);
@@ -998,50 +858,6 @@ AVSC_INLINE AVS_Library * avs_load_library() {
AVSC_LOAD_FUNC(avs_take_clip);
AVSC_LOAD_FUNC(avs_vsprintf);
AVSC_LOAD_FUNC(avs_get_error);
AVSC_LOAD_FUNC(avs_is_rgb48);
AVSC_LOAD_FUNC(avs_is_rgb64);
AVSC_LOAD_FUNC(avs_is_yv24);
AVSC_LOAD_FUNC(avs_is_yv16);
AVSC_LOAD_FUNC(avs_is_yv12);
AVSC_LOAD_FUNC(avs_is_yv411);
AVSC_LOAD_FUNC(avs_is_y8);
AVSC_LOAD_FUNC(avs_is_yuv444p16);
AVSC_LOAD_FUNC(avs_is_yuv422p16);
AVSC_LOAD_FUNC(avs_is_yuv420p16);
AVSC_LOAD_FUNC(avs_is_y16);
AVSC_LOAD_FUNC(avs_is_yuv444ps);
AVSC_LOAD_FUNC(avs_is_yuv422ps);
AVSC_LOAD_FUNC(avs_is_yuv420ps);
AVSC_LOAD_FUNC(avs_is_y32);
AVSC_LOAD_FUNC(avs_is_444);
AVSC_LOAD_FUNC(avs_is_422);
AVSC_LOAD_FUNC(avs_is_420);
AVSC_LOAD_FUNC(avs_is_y);
AVSC_LOAD_FUNC(avs_is_yuva);
AVSC_LOAD_FUNC(avs_is_planar_rgb);
AVSC_LOAD_FUNC(avs_is_planar_rgba);
AVSC_LOAD_FUNC(avs_is_color_space);
AVSC_LOAD_FUNC(avs_get_plane_width_subsampling);
AVSC_LOAD_FUNC(avs_get_plane_height_subsampling);
AVSC_LOAD_FUNC(avs_bits_per_pixel);
AVSC_LOAD_FUNC(avs_bytes_from_pixels);
AVSC_LOAD_FUNC(avs_row_size);
AVSC_LOAD_FUNC(avs_bmp_size);
AVSC_LOAD_FUNC(avs_get_pitch_p);
AVSC_LOAD_FUNC(avs_get_row_size_p);
AVSC_LOAD_FUNC(avs_get_height_p);
AVSC_LOAD_FUNC(avs_get_read_ptr_p);
AVSC_LOAD_FUNC(avs_is_writable);
AVSC_LOAD_FUNC(avs_get_write_ptr_p);
AVSC_LOAD_FUNC(avs_num_components);
AVSC_LOAD_FUNC(avs_component_size);
AVSC_LOAD_FUNC(avs_bits_per_component);
#undef __AVSC_STRINGIFY
#undef AVSC_STRINGIFY
#undef AVSC_LOAD_FUNC

View File

@@ -0,0 +1,68 @@
// Copyright (c) 2011 FFmpegSource Project
//
// Permission is hereby granted, free of charge, to any person obtaining a copy
// of this software and associated documentation files (the "Software"), to deal
// in the Software without restriction, including without limitation the rights
// to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
// copies of the Software, and to permit persons to whom the Software is
// furnished to do so, subject to the following conditions:
//
// The above copyright notice and this permission notice shall be included in
// all copies or substantial portions of the Software.
//
// THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
// IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
// FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
// AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
// LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
// OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
// THE SOFTWARE.
/* these are defines/functions that are used and were changed in the switch to 2.6
* and are needed to maintain full compatility with 2.5 */
enum {
AVS_CS_YV12_25 = 1<<3 | AVS_CS_YUV | AVS_CS_PLANAR, // y-v-u, planar
AVS_CS_I420_25 = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR, // y-u-v, planar
};
AVSC_INLINE int avs_get_height_p_25(const AVS_VideoFrame * p, int plane) {
switch (plane)
{
case AVS_PLANAR_U: case AVS_PLANAR_V:
if (p->pitchUV)
return p->height>>1;
return 0;
}
return p->height;}
AVSC_INLINE int avs_get_row_size_p_25(const AVS_VideoFrame * p, int plane) {
int r;
switch (plane)
{
case AVS_PLANAR_U: case AVS_PLANAR_V:
if (p->pitchUV)
return p->row_size>>1;
else
return 0;
case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED:
if (p->pitchUV)
{
r = ((p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)) )>>1; // Aligned rowsize
if (r < p->pitchUV)
return r;
return p->row_size>>1;
}
else
return 0;
case AVS_PLANAR_Y_ALIGNED:
r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
if (r <= p->pitch)
return r;
return p->row_size;
}
return p->row_size;
}
AVSC_INLINE int avs_is_yv12_25(const AVS_VideoInfo * p)
{ return ((p->pixel_type & AVS_CS_YV12_25) == AVS_CS_YV12_25)||((p->pixel_type & AVS_CS_I420_25) == AVS_CS_I420_25); }

View File

@@ -1,62 +0,0 @@
// Avisynth C Interface Version 0.20
// Copyright 2003 Kevin Atkinson
// This program is free software; you can redistribute it and/or modify
// it under the terms of the GNU General Public License as published by
// the Free Software Foundation; either version 2 of the License, or
// (at your option) any later version.
//
// This program is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
// GNU General Public License for more details.
//
// You should have received a copy of the GNU General Public License
// along with this program; if not, write to the Free Software
// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
// http://www.gnu.org/copyleft/gpl.html .
//
// As a special exception, I give you permission to link to the
// Avisynth C interface with independent modules that communicate with
// the Avisynth C interface solely through the interfaces defined in
// avisynth_c.h, regardless of the license terms of these independent
// modules, and to copy and distribute the resulting combined work
// under terms of your choice, provided that every copy of the
// combined work is accompanied by a complete copy of the source code
// of the Avisynth C interface and Avisynth itself (with the version
// used to produce the combined work), being distributed under the
// terms of the GNU General Public License plus this exception. An
// independent module is a module which is not derived from or based
// on Avisynth C Interface, such as 3rd-party filters, import and
// export plugins, or graphical user interfaces.
#ifndef AVS_CAPI_H
#define AVS_CAPI_H
#ifdef __cplusplus
# define EXTERN_C extern "C"
#else
# define EXTERN_C
#endif
#ifndef AVSC_USE_STDCALL
# define AVSC_CC __cdecl
#else
# define AVSC_CC __stdcall
#endif
#define AVSC_INLINE static __inline
#ifdef BUILDING_AVSCORE
# define AVSC_EXPORT EXTERN_C
# define AVSC_API(ret, name) EXTERN_C __declspec(dllexport) ret AVSC_CC name
#else
# define AVSC_EXPORT EXTERN_C __declspec(dllexport)
# ifndef AVSC_NO_DECLSPEC
# define AVSC_API(ret, name) EXTERN_C __declspec(dllimport) ret AVSC_CC name
# else
# define AVSC_API(ret, name) typedef ret (AVSC_CC *name##_func)
# endif
#endif
#endif //AVS_CAPI_H

View File

@@ -1,55 +0,0 @@
// Avisynth C Interface Version 0.20
// Copyright 2003 Kevin Atkinson
// This program is free software; you can redistribute it and/or modify
// it under the terms of the GNU General Public License as published by
// the Free Software Foundation; either version 2 of the License, or
// (at your option) any later version.
//
// This program is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
// GNU General Public License for more details.
//
// You should have received a copy of the GNU General Public License
// along with this program; if not, write to the Free Software
// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
// http://www.gnu.org/copyleft/gpl.html .
//
// As a special exception, I give you permission to link to the
// Avisynth C interface with independent modules that communicate with
// the Avisynth C interface solely through the interfaces defined in
// avisynth_c.h, regardless of the license terms of these independent
// modules, and to copy and distribute the resulting combined work
// under terms of your choice, provided that every copy of the
// combined work is accompanied by a complete copy of the source code
// of the Avisynth C interface and Avisynth itself (with the version
// used to produce the combined work), being distributed under the
// terms of the GNU General Public License plus this exception. An
// independent module is a module which is not derived from or based
// on Avisynth C Interface, such as 3rd-party filters, import and
// export plugins, or graphical user interfaces.
#ifndef AVS_CONFIG_H
#define AVS_CONFIG_H
// Undefine this to get cdecl calling convention
#define AVSC_USE_STDCALL 1
// NOTE TO PLUGIN AUTHORS:
// Because FRAME_ALIGN can be substantially higher than the alignment
// a plugin actually needs, plugins should not use FRAME_ALIGN to check for
// alignment. They should always request the exact alignment value they need.
// This is to make sure that plugins work over the widest range of AviSynth
// builds possible.
#define FRAME_ALIGN 32
#if defined(_M_AMD64) || defined(__x86_64)
# define X86_64
#elif defined(_M_IX86) || defined(__i386__)
# define X86_32
#else
# error Unsupported CPU architecture.
#endif
#endif //AVS_CONFIG_H

View File

@@ -1,51 +0,0 @@
// Avisynth C Interface Version 0.20
// Copyright 2003 Kevin Atkinson
// This program is free software; you can redistribute it and/or modify
// it under the terms of the GNU General Public License as published by
// the Free Software Foundation; either version 2 of the License, or
// (at your option) any later version.
//
// This program is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
// GNU General Public License for more details.
//
// You should have received a copy of the GNU General Public License
// along with this program; if not, write to the Free Software
// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
// http://www.gnu.org/copyleft/gpl.html .
//
// As a special exception, I give you permission to link to the
// Avisynth C interface with independent modules that communicate with
// the Avisynth C interface solely through the interfaces defined in
// avisynth_c.h, regardless of the license terms of these independent
// modules, and to copy and distribute the resulting combined work
// under terms of your choice, provided that every copy of the
// combined work is accompanied by a complete copy of the source code
// of the Avisynth C interface and Avisynth itself (with the version
// used to produce the combined work), being distributed under the
// terms of the GNU General Public License plus this exception. An
// independent module is a module which is not derived from or based
// on Avisynth C Interface, such as 3rd-party filters, import and
// export plugins, or graphical user interfaces.
#ifndef AVS_TYPES_H
#define AVS_TYPES_H
// Define all types necessary for interfacing with avisynth.dll
// Raster types used by VirtualDub & Avisynth
typedef unsigned int Pixel32;
typedef unsigned char BYTE;
// Audio Sample information
typedef float SFLOAT;
#ifdef __GNUC__
typedef long long int INT64;
#else
typedef __int64 INT64;
#endif
#endif //AVS_TYPES_H

View File

@@ -513,21 +513,21 @@ AVSC_INLINE AVS_Value avs_array_elt(AVS_Value v, int index)
// only use these functions on am AVS_Value that does not already have
// an active value. Remember, treat AVS_Value as a fat pointer.
AVSC_INLINE AVS_Value avs_new_value_bool(int v0)
{ AVS_Value v = {0}; v.type = 'b'; v.d.boolean = v0 == 0 ? 0 : 1; return v; }
{ AVS_Value v; v.type = 'b'; v.d.boolean = v0 == 0 ? 0 : 1; return v; }
AVSC_INLINE AVS_Value avs_new_value_int(int v0)
{ AVS_Value v = {0}; v.type = 'i'; v.d.integer = v0; return v; }
{ AVS_Value v; v.type = 'i'; v.d.integer = v0; return v; }
AVSC_INLINE AVS_Value avs_new_value_string(const char * v0)
{ AVS_Value v = {0}; v.type = 's'; v.d.string = v0; return v; }
{ AVS_Value v; v.type = 's'; v.d.string = v0; return v; }
AVSC_INLINE AVS_Value avs_new_value_float(float v0)
{ AVS_Value v = {0}; v.type = 'f'; v.d.floating_pt = v0; return v;}
{ AVS_Value v; v.type = 'f'; v.d.floating_pt = v0; return v;}
AVSC_INLINE AVS_Value avs_new_value_error(const char * v0)
{ AVS_Value v = {0}; v.type = 'e'; v.d.string = v0; return v; }
{ AVS_Value v; v.type = 'e'; v.d.string = v0; return v; }
#ifndef AVSC_NO_DECLSPEC
AVSC_INLINE AVS_Value avs_new_value_clip(AVS_Clip * v0)
{ AVS_Value v = {0}; avs_set_to_clip(&v, v0); return v; }
{ AVS_Value v; avs_set_to_clip(&v, v0); return v; }
#endif
AVSC_INLINE AVS_Value avs_new_value_array(AVS_Value * v0, int size)
{ AVS_Value v = {0}; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; }
{ AVS_Value v; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; }
/////////////////////////////////////////////////////////////////////
//

View File

@@ -52,8 +52,8 @@ namespace avxsynth {
//
// Functions
//
#define MAKEDWORD(a,b,c,d) (((a) << 24) | ((b) << 16) | ((c) << 8) | (d))
#define MAKEWORD(a,b) (((a) << 8) | (b))
#define MAKEDWORD(a,b,c,d) ((a << 24) | (b << 16) | (c << 8) | (d))
#define MAKEWORD(a,b) ((a << 8) | (b))
#define lstrlen strlen
#define lstrcpy strcpy

View File

@@ -1,827 +0,0 @@
/*
* This copyright notice applies to this header file only:
*
* Copyright (c) 2010-2016 NVIDIA Corporation
*
* Permission is hereby granted, free of charge, to any person
* obtaining a copy of this software and associated documentation
* files (the "Software"), to deal in the Software without
* restriction, including without limitation the rights to use,
* copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the software, and to permit persons to whom the
* software is furnished to do so, subject to the following
* conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES
* OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND
* NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT
* HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY,
* WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR
* OTHER DEALINGS IN THE SOFTWARE.
*/
/**
* \file cuviddec.h
* NvCuvid API provides Video Decoding interface to NVIDIA GPU devices.
* \date 2015-2016
* This file contains constants, structure definitions and function prototypes used for decoding.
*/
#if !defined(__CUDA_VIDEO_H__)
#define __CUDA_VIDEO_H__
#ifndef __cuda_cuda_h__
#include <cuda.h>
#endif // __cuda_cuda_h__
#if defined(__x86_64) || defined(AMD64) || defined(_M_AMD64)
#if (CUDA_VERSION >= 3020) && (!defined(CUDA_FORCE_API_VERSION) || (CUDA_FORCE_API_VERSION >= 3020))
#define __CUVID_DEVPTR64
#endif
#endif
#if defined(__cplusplus)
extern "C" {
#endif /* __cplusplus */
typedef void *CUvideodecoder;
typedef struct _CUcontextlock_st *CUvideoctxlock;
/**
* \addtogroup VIDEO_DECODER Video Decoder
* @{
*/
/*!
* \enum cudaVideoCodec
* Video Codec Enums
*/
typedef enum cudaVideoCodec_enum {
cudaVideoCodec_MPEG1=0, /**< MPEG1 */
cudaVideoCodec_MPEG2, /**< MPEG2 */
cudaVideoCodec_MPEG4, /**< MPEG4 */
cudaVideoCodec_VC1, /**< VC1 */
cudaVideoCodec_H264, /**< H264 */
cudaVideoCodec_JPEG, /**< JPEG */
cudaVideoCodec_H264_SVC, /**< H264-SVC */
cudaVideoCodec_H264_MVC, /**< H264-MVC */
cudaVideoCodec_HEVC, /**< HEVC */
cudaVideoCodec_VP8, /**< VP8 */
cudaVideoCodec_VP9, /**< VP9 */
cudaVideoCodec_NumCodecs, /**< Max COdecs */
// Uncompressed YUV
cudaVideoCodec_YUV420 = (('I'<<24)|('Y'<<16)|('U'<<8)|('V')), /**< Y,U,V (4:2:0) */
cudaVideoCodec_YV12 = (('Y'<<24)|('V'<<16)|('1'<<8)|('2')), /**< Y,V,U (4:2:0) */
cudaVideoCodec_NV12 = (('N'<<24)|('V'<<16)|('1'<<8)|('2')), /**< Y,UV (4:2:0) */
cudaVideoCodec_YUYV = (('Y'<<24)|('U'<<16)|('Y'<<8)|('V')), /**< YUYV/YUY2 (4:2:2) */
cudaVideoCodec_UYVY = (('U'<<24)|('Y'<<16)|('V'<<8)|('Y')) /**< UYVY (4:2:2) */
} cudaVideoCodec;
/*!
* \enum cudaVideoSurfaceFormat
* Video Surface Formats Enums
*/
typedef enum cudaVideoSurfaceFormat_enum {
cudaVideoSurfaceFormat_NV12=0 /**< NV12 (currently the only supported output format) */
} cudaVideoSurfaceFormat;
/*!
* \enum cudaVideoDeinterlaceMode
* Deinterlacing Modes Enums
*/
typedef enum cudaVideoDeinterlaceMode_enum {
cudaVideoDeinterlaceMode_Weave=0, /**< Weave both fields (no deinterlacing) */
cudaVideoDeinterlaceMode_Bob, /**< Drop one field */
cudaVideoDeinterlaceMode_Adaptive /**< Adaptive deinterlacing */
} cudaVideoDeinterlaceMode;
/*!
* \enum cudaVideoChromaFormat
* Chroma Formats Enums
*/
typedef enum cudaVideoChromaFormat_enum {
cudaVideoChromaFormat_Monochrome=0, /**< MonoChrome */
cudaVideoChromaFormat_420, /**< 4:2:0 */
cudaVideoChromaFormat_422, /**< 4:2:2 */
cudaVideoChromaFormat_444 /**< 4:4:4 */
} cudaVideoChromaFormat;
/*!
* \enum cudaVideoCreateFlags
* Decoder Flags Enums
*/
typedef enum cudaVideoCreateFlags_enum {
cudaVideoCreate_Default = 0x00, /**< Default operation mode: use dedicated video engines */
cudaVideoCreate_PreferCUDA = 0x01, /**< Use a CUDA-based decoder if faster than dedicated engines (requires a valid vidLock object for multi-threading) */
cudaVideoCreate_PreferDXVA = 0x02, /**< Go through DXVA internally if possible (requires D3D9 interop) */
cudaVideoCreate_PreferCUVID = 0x04 /**< Use dedicated video engines directly */
} cudaVideoCreateFlags;
/*!
* \struct CUVIDDECODECREATEINFO
* Struct used in create decoder
*/
typedef struct _CUVIDDECODECREATEINFO
{
unsigned long ulWidth; /**< Coded Sequence Width */
unsigned long ulHeight; /**< Coded Sequence Height */
unsigned long ulNumDecodeSurfaces; /**< Maximum number of internal decode surfaces */
cudaVideoCodec CodecType; /**< cudaVideoCodec_XXX */
cudaVideoChromaFormat ChromaFormat; /**< cudaVideoChromaFormat_XXX (only 4:2:0 is currently supported) */
unsigned long ulCreationFlags; /**< Decoder creation flags (cudaVideoCreateFlags_XXX) */
unsigned long bitDepthMinus8;
unsigned long Reserved1[4]; /**< Reserved for future use - set to zero */
/**
* area of the frame that should be displayed
*/
struct {
short left;
short top;
short right;
short bottom;
} display_area;
cudaVideoSurfaceFormat OutputFormat; /**< cudaVideoSurfaceFormat_XXX */
cudaVideoDeinterlaceMode DeinterlaceMode; /**< cudaVideoDeinterlaceMode_XXX */
unsigned long ulTargetWidth; /**< Post-processed Output Width (Should be aligned to 2) */
unsigned long ulTargetHeight; /**< Post-processed Output Height (Should be aligbed to 2) */
unsigned long ulNumOutputSurfaces; /**< Maximum number of output surfaces simultaneously mapped */
CUvideoctxlock vidLock; /**< If non-NULL, context lock used for synchronizing ownership of the cuda context */
/**
* target rectangle in the output frame (for aspect ratio conversion)
* if a null rectangle is specified, {0,0,ulTargetWidth,ulTargetHeight} will be used
*/
struct {
short left;
short top;
short right;
short bottom;
} target_rect;
unsigned long Reserved2[5]; /**< Reserved for future use - set to zero */
} CUVIDDECODECREATEINFO;
/*!
* \struct CUVIDH264DPBENTRY
* H.264 DPB Entry
*/
typedef struct _CUVIDH264DPBENTRY
{
int PicIdx; /**< picture index of reference frame */
int FrameIdx; /**< frame_num(short-term) or LongTermFrameIdx(long-term) */
int is_long_term; /**< 0=short term reference, 1=long term reference */
int not_existing; /**< non-existing reference frame (corresponding PicIdx should be set to -1) */
int used_for_reference; /**< 0=unused, 1=top_field, 2=bottom_field, 3=both_fields */
int FieldOrderCnt[2]; /**< field order count of top and bottom fields */
} CUVIDH264DPBENTRY;
/*!
* \struct CUVIDH264MVCEXT
* H.264 MVC Picture Parameters Ext
*/
typedef struct _CUVIDH264MVCEXT
{
int num_views_minus1;
int view_id;
unsigned char inter_view_flag;
unsigned char num_inter_view_refs_l0;
unsigned char num_inter_view_refs_l1;
unsigned char MVCReserved8Bits;
int InterViewRefsL0[16];
int InterViewRefsL1[16];
} CUVIDH264MVCEXT;
/*!
* \struct CUVIDH264SVCEXT
* H.264 SVC Picture Parameters Ext
*/
typedef struct _CUVIDH264SVCEXT
{
unsigned char profile_idc;
unsigned char level_idc;
unsigned char DQId;
unsigned char DQIdMax;
unsigned char disable_inter_layer_deblocking_filter_idc;
unsigned char ref_layer_chroma_phase_y_plus1;
signed char inter_layer_slice_alpha_c0_offset_div2;
signed char inter_layer_slice_beta_offset_div2;
unsigned short DPBEntryValidFlag;
unsigned char inter_layer_deblocking_filter_control_present_flag;
unsigned char extended_spatial_scalability_idc;
unsigned char adaptive_tcoeff_level_prediction_flag;
unsigned char slice_header_restriction_flag;
unsigned char chroma_phase_x_plus1_flag;
unsigned char chroma_phase_y_plus1;
unsigned char tcoeff_level_prediction_flag;
unsigned char constrained_intra_resampling_flag;
unsigned char ref_layer_chroma_phase_x_plus1_flag;
unsigned char store_ref_base_pic_flag;
unsigned char Reserved8BitsA;
unsigned char Reserved8BitsB;
// For the 4 scaled_ref_layer_XX fields below,
// if (extended_spatial_scalability_idc == 1), SPS field, G.7.3.2.1.4, add prefix "seq_"
// if (extended_spatial_scalability_idc == 2), SLH field, G.7.3.3.4,
short scaled_ref_layer_left_offset;
short scaled_ref_layer_top_offset;
short scaled_ref_layer_right_offset;
short scaled_ref_layer_bottom_offset;
unsigned short Reserved16Bits;
struct _CUVIDPICPARAMS *pNextLayer; /**< Points to the picparams for the next layer to be decoded. Linked list ends at the target layer. */
int bRefBaseLayer; /**< whether to store ref base pic */
} CUVIDH264SVCEXT;
/*!
* \struct CUVIDH264PICPARAMS
* H.264 Picture Parameters
*/
typedef struct _CUVIDH264PICPARAMS
{
// SPS
int log2_max_frame_num_minus4;
int pic_order_cnt_type;
int log2_max_pic_order_cnt_lsb_minus4;
int delta_pic_order_always_zero_flag;
int frame_mbs_only_flag;
int direct_8x8_inference_flag;
int num_ref_frames; // NOTE: shall meet level 4.1 restrictions
unsigned char residual_colour_transform_flag;
unsigned char bit_depth_luma_minus8; // Must be 0 (only 8-bit supported)
unsigned char bit_depth_chroma_minus8; // Must be 0 (only 8-bit supported)
unsigned char qpprime_y_zero_transform_bypass_flag;
// PPS
int entropy_coding_mode_flag;
int pic_order_present_flag;
int num_ref_idx_l0_active_minus1;
int num_ref_idx_l1_active_minus1;
int weighted_pred_flag;
int weighted_bipred_idc;
int pic_init_qp_minus26;
int deblocking_filter_control_present_flag;
int redundant_pic_cnt_present_flag;
int transform_8x8_mode_flag;
int MbaffFrameFlag;
int constrained_intra_pred_flag;
int chroma_qp_index_offset;
int second_chroma_qp_index_offset;
int ref_pic_flag;
int frame_num;
int CurrFieldOrderCnt[2];
// DPB
CUVIDH264DPBENTRY dpb[16]; // List of reference frames within the DPB
// Quantization Matrices (raster-order)
unsigned char WeightScale4x4[6][16];
unsigned char WeightScale8x8[2][64];
// FMO/ASO
unsigned char fmo_aso_enable;
unsigned char num_slice_groups_minus1;
unsigned char slice_group_map_type;
signed char pic_init_qs_minus26;
unsigned int slice_group_change_rate_minus1;
union
{
unsigned long long slice_group_map_addr;
const unsigned char *pMb2SliceGroupMap;
} fmo;
unsigned int Reserved[12];
// SVC/MVC
union
{
CUVIDH264MVCEXT mvcext;
CUVIDH264SVCEXT svcext;
};
} CUVIDH264PICPARAMS;
/*!
* \struct CUVIDMPEG2PICPARAMS
* MPEG-2 Picture Parameters
*/
typedef struct _CUVIDMPEG2PICPARAMS
{
int ForwardRefIdx; // Picture index of forward reference (P/B-frames)
int BackwardRefIdx; // Picture index of backward reference (B-frames)
int picture_coding_type;
int full_pel_forward_vector;
int full_pel_backward_vector;
int f_code[2][2];
int intra_dc_precision;
int frame_pred_frame_dct;
int concealment_motion_vectors;
int q_scale_type;
int intra_vlc_format;
int alternate_scan;
int top_field_first;
// Quantization matrices (raster order)
unsigned char QuantMatrixIntra[64];
unsigned char QuantMatrixInter[64];
} CUVIDMPEG2PICPARAMS;
////////////////////////////////////////////////////////////////////////////////////////////////
//
// MPEG-4 Picture Parameters
//
// MPEG-4 has VOP types instead of Picture types
#define I_VOP 0
#define P_VOP 1
#define B_VOP 2
#define S_VOP 3
/*!
* \struct CUVIDMPEG4PICPARAMS
* MPEG-4 Picture Parameters
*/
typedef struct _CUVIDMPEG4PICPARAMS
{
int ForwardRefIdx; // Picture index of forward reference (P/B-frames)
int BackwardRefIdx; // Picture index of backward reference (B-frames)
// VOL
int video_object_layer_width;
int video_object_layer_height;
int vop_time_increment_bitcount;
int top_field_first;
int resync_marker_disable;
int quant_type;
int quarter_sample;
int short_video_header;
int divx_flags;
// VOP
int vop_coding_type;
int vop_coded;
int vop_rounding_type;
int alternate_vertical_scan_flag;
int interlaced;
int vop_fcode_forward;
int vop_fcode_backward;
int trd[2];
int trb[2];
// Quantization matrices (raster order)
unsigned char QuantMatrixIntra[64];
unsigned char QuantMatrixInter[64];
int gmc_enabled;
} CUVIDMPEG4PICPARAMS;
/*!
* \struct CUVIDVC1PICPARAMS
* VC1 Picture Parameters
*/
typedef struct _CUVIDVC1PICPARAMS
{
int ForwardRefIdx; /**< Picture index of forward reference (P/B-frames) */
int BackwardRefIdx; /**< Picture index of backward reference (B-frames) */
int FrameWidth; /**< Actual frame width */
int FrameHeight; /**< Actual frame height */
// PICTURE
int intra_pic_flag; /**< Set to 1 for I,BI frames */
int ref_pic_flag; /**< Set to 1 for I,P frames */
int progressive_fcm; /**< Progressive frame */
// SEQUENCE
int profile;
int postprocflag;
int pulldown;
int interlace;
int tfcntrflag;
int finterpflag;
int psf;
int multires;
int syncmarker;
int rangered;
int maxbframes;
// ENTRYPOINT
int panscan_flag;
int refdist_flag;
int extended_mv;
int dquant;
int vstransform;
int loopfilter;
int fastuvmc;
int overlap;
int quantizer;
int extended_dmv;
int range_mapy_flag;
int range_mapy;
int range_mapuv_flag;
int range_mapuv;
int rangeredfrm; // range reduction state
} CUVIDVC1PICPARAMS;
/*!
* \struct CUVIDJPEGPICPARAMS
* JPEG Picture Parameters
*/
typedef struct _CUVIDJPEGPICPARAMS
{
int Reserved;
} CUVIDJPEGPICPARAMS;
/*!
* \struct CUVIDHEVCPICPARAMS
* HEVC Picture Parameters
*/
typedef struct _CUVIDHEVCPICPARAMS
{
// sps
int pic_width_in_luma_samples;
int pic_height_in_luma_samples;
unsigned char log2_min_luma_coding_block_size_minus3;
unsigned char log2_diff_max_min_luma_coding_block_size;
unsigned char log2_min_transform_block_size_minus2;
unsigned char log2_diff_max_min_transform_block_size;
unsigned char pcm_enabled_flag;
unsigned char log2_min_pcm_luma_coding_block_size_minus3;
unsigned char log2_diff_max_min_pcm_luma_coding_block_size;
unsigned char pcm_sample_bit_depth_luma_minus1;
unsigned char pcm_sample_bit_depth_chroma_minus1;
unsigned char pcm_loop_filter_disabled_flag;
unsigned char strong_intra_smoothing_enabled_flag;
unsigned char max_transform_hierarchy_depth_intra;
unsigned char max_transform_hierarchy_depth_inter;
unsigned char amp_enabled_flag;
unsigned char separate_colour_plane_flag;
unsigned char log2_max_pic_order_cnt_lsb_minus4;
unsigned char num_short_term_ref_pic_sets;
unsigned char long_term_ref_pics_present_flag;
unsigned char num_long_term_ref_pics_sps;
unsigned char sps_temporal_mvp_enabled_flag;
unsigned char sample_adaptive_offset_enabled_flag;
unsigned char scaling_list_enable_flag;
unsigned char IrapPicFlag;
unsigned char IdrPicFlag;
unsigned char bit_depth_luma_minus8;
unsigned char bit_depth_chroma_minus8;
unsigned char reserved1[14];
// pps
unsigned char dependent_slice_segments_enabled_flag;
unsigned char slice_segment_header_extension_present_flag;
unsigned char sign_data_hiding_enabled_flag;
unsigned char cu_qp_delta_enabled_flag;
unsigned char diff_cu_qp_delta_depth;
signed char init_qp_minus26;
signed char pps_cb_qp_offset;
signed char pps_cr_qp_offset;
unsigned char constrained_intra_pred_flag;
unsigned char weighted_pred_flag;
unsigned char weighted_bipred_flag;
unsigned char transform_skip_enabled_flag;
unsigned char transquant_bypass_enabled_flag;
unsigned char entropy_coding_sync_enabled_flag;
unsigned char log2_parallel_merge_level_minus2;
unsigned char num_extra_slice_header_bits;
unsigned char loop_filter_across_tiles_enabled_flag;
unsigned char loop_filter_across_slices_enabled_flag;
unsigned char output_flag_present_flag;
unsigned char num_ref_idx_l0_default_active_minus1;
unsigned char num_ref_idx_l1_default_active_minus1;
unsigned char lists_modification_present_flag;
unsigned char cabac_init_present_flag;
unsigned char pps_slice_chroma_qp_offsets_present_flag;
unsigned char deblocking_filter_override_enabled_flag;
unsigned char pps_deblocking_filter_disabled_flag;
signed char pps_beta_offset_div2;
signed char pps_tc_offset_div2;
unsigned char tiles_enabled_flag;
unsigned char uniform_spacing_flag;
unsigned char num_tile_columns_minus1;
unsigned char num_tile_rows_minus1;
unsigned short column_width_minus1[21];
unsigned short row_height_minus1[21];
unsigned int reserved3[15];
// RefPicSets
int NumBitsForShortTermRPSInSlice;
int NumDeltaPocsOfRefRpsIdx;
int NumPocTotalCurr;
int NumPocStCurrBefore;
int NumPocStCurrAfter;
int NumPocLtCurr;
int CurrPicOrderCntVal;
int RefPicIdx[16]; // [refpic] Indices of valid reference pictures (-1 if unused for reference)
int PicOrderCntVal[16]; // [refpic]
unsigned char IsLongTerm[16]; // [refpic] 0=not a long-term reference, 1=long-term reference
unsigned char RefPicSetStCurrBefore[8]; // [0..NumPocStCurrBefore-1] -> refpic (0..15)
unsigned char RefPicSetStCurrAfter[8]; // [0..NumPocStCurrAfter-1] -> refpic (0..15)
unsigned char RefPicSetLtCurr[8]; // [0..NumPocLtCurr-1] -> refpic (0..15)
unsigned char RefPicSetInterLayer0[8];
unsigned char RefPicSetInterLayer1[8];
unsigned int reserved4[12];
// scaling lists (diag order)
unsigned char ScalingList4x4[6][16]; // [matrixId][i]
unsigned char ScalingList8x8[6][64]; // [matrixId][i]
unsigned char ScalingList16x16[6][64]; // [matrixId][i]
unsigned char ScalingList32x32[2][64]; // [matrixId][i]
unsigned char ScalingListDCCoeff16x16[6]; // [matrixId]
unsigned char ScalingListDCCoeff32x32[2]; // [matrixId]
} CUVIDHEVCPICPARAMS;
/*!
* \struct CUVIDVP8PICPARAMS
* VP8 Picture Parameters
*/
typedef struct _CUVIDVP8PICPARAMS
{
int width;
int height;
unsigned int first_partition_size;
//Frame Indexes
unsigned char LastRefIdx;
unsigned char GoldenRefIdx;
unsigned char AltRefIdx;
union {
struct {
unsigned char frame_type : 1; /**< 0 = KEYFRAME, 1 = INTERFRAME */
unsigned char version : 3;
unsigned char show_frame : 1;
unsigned char update_mb_segmentation_data : 1; /**< Must be 0 if segmentation is not enabled */
unsigned char Reserved2Bits : 2;
};
unsigned char wFrameTagFlags;
};
unsigned char Reserved1[4];
unsigned int Reserved2[3];
} CUVIDVP8PICPARAMS;
/*!
* \struct CUVIDVP9PICPARAMS
* VP9 Picture Parameters
*/
typedef struct _CUVIDVP9PICPARAMS
{
unsigned int width;
unsigned int height;
//Frame Indices
unsigned char LastRefIdx;
unsigned char GoldenRefIdx;
unsigned char AltRefIdx;
unsigned char colorSpace;
unsigned short profile : 3;
unsigned short frameContextIdx : 2;
unsigned short frameType : 1;
unsigned short showFrame : 1;
unsigned short errorResilient : 1;
unsigned short frameParallelDecoding : 1;
unsigned short subSamplingX : 1;
unsigned short subSamplingY : 1;
unsigned short intraOnly : 1;
unsigned short allow_high_precision_mv : 1;
unsigned short refreshEntropyProbs : 1;
unsigned short reserved2Bits : 2;
unsigned short reserved16Bits;
unsigned char refFrameSignBias[4];
unsigned char bitDepthMinus8Luma;
unsigned char bitDepthMinus8Chroma;
unsigned char loopFilterLevel;
unsigned char loopFilterSharpness;
unsigned char modeRefLfEnabled;
unsigned char log2_tile_columns;
unsigned char log2_tile_rows;
unsigned char segmentEnabled : 1;
unsigned char segmentMapUpdate : 1;
unsigned char segmentMapTemporalUpdate : 1;
unsigned char segmentFeatureMode : 1;
unsigned char reserved4Bits : 4;
unsigned char segmentFeatureEnable[8][4];
short segmentFeatureData[8][4];
unsigned char mb_segment_tree_probs[7];
unsigned char segment_pred_probs[3];
unsigned char reservedSegment16Bits[2];
int qpYAc;
int qpYDc;
int qpChDc;
int qpChAc;
unsigned int activeRefIdx[3];
unsigned int resetFrameContext;
unsigned int mcomp_filter_type;
unsigned int mbRefLfDelta[4];
unsigned int mbModeLfDelta[2];
unsigned int frameTagSize;
unsigned int offsetToDctParts;
unsigned int reserved128Bits[4];
} CUVIDVP9PICPARAMS;
/*!
* \struct CUVIDPICPARAMS
* Picture Parameters for Decoding
*/
typedef struct _CUVIDPICPARAMS
{
int PicWidthInMbs; /**< Coded Frame Size */
int FrameHeightInMbs; /**< Coded Frame Height */
int CurrPicIdx; /**< Output index of the current picture */
int field_pic_flag; /**< 0=frame picture, 1=field picture */
int bottom_field_flag; /**< 0=top field, 1=bottom field (ignored if field_pic_flag=0) */
int second_field; /**< Second field of a complementary field pair */
// Bitstream data
unsigned int nBitstreamDataLen; /**< Number of bytes in bitstream data buffer */
const unsigned char *pBitstreamData; /**< Ptr to bitstream data for this picture (slice-layer) */
unsigned int nNumSlices; /**< Number of slices in this picture */
const unsigned int *pSliceDataOffsets; /**< nNumSlices entries, contains offset of each slice within the bitstream data buffer */
int ref_pic_flag; /**< This picture is a reference picture */
int intra_pic_flag; /**< This picture is entirely intra coded */
unsigned int Reserved[30]; /**< Reserved for future use */
// Codec-specific data
union {
CUVIDMPEG2PICPARAMS mpeg2; /**< Also used for MPEG-1 */
CUVIDH264PICPARAMS h264;
CUVIDVC1PICPARAMS vc1;
CUVIDMPEG4PICPARAMS mpeg4;
CUVIDJPEGPICPARAMS jpeg;
CUVIDHEVCPICPARAMS hevc;
CUVIDVP8PICPARAMS vp8;
CUVIDVP9PICPARAMS vp9;
unsigned int CodecReserved[1024];
} CodecSpecific;
} CUVIDPICPARAMS;
/*!
* \struct CUVIDPROCPARAMS
* Picture Parameters for Postprocessing
*/
typedef struct _CUVIDPROCPARAMS
{
int progressive_frame; /**< Input is progressive (deinterlace_mode will be ignored) */
int second_field; /**< Output the second field (ignored if deinterlace mode is Weave) */
int top_field_first; /**< Input frame is top field first (1st field is top, 2nd field is bottom) */
int unpaired_field; /**< Input only contains one field (2nd field is invalid) */
// The fields below are used for raw YUV input
unsigned int reserved_flags; /**< Reserved for future use (set to zero) */
unsigned int reserved_zero; /**< Reserved (set to zero) */
unsigned long long raw_input_dptr; /**< Input CUdeviceptr for raw YUV extensions */
unsigned int raw_input_pitch; /**< pitch in bytes of raw YUV input (should be aligned appropriately) */
unsigned int raw_input_format; /**< Reserved for future use (set to zero) */
unsigned long long raw_output_dptr; /**< Reserved for future use (set to zero) */
unsigned int raw_output_pitch; /**< Reserved for future use (set to zero) */
unsigned int Reserved[48];
void *Reserved3[3];
} CUVIDPROCPARAMS;
/**
*
* In order to minimize decode latencies, there should be always at least 2 pictures in the decode
* queue at any time, in order to make sure that all decode engines are always busy.
*
* Overall data flow:
* - cuvidCreateDecoder(...)
* For each picture:
* - cuvidDecodePicture(N)
* - cuvidMapVideoFrame(N-4)
* - do some processing in cuda
* - cuvidUnmapVideoFrame(N-4)
* - cuvidDecodePicture(N+1)
* - cuvidMapVideoFrame(N-3)
* ...
* - cuvidDestroyDecoder(...)
*
* NOTE:
* - When the cuda context is created from a D3D device, the D3D device must also be created
* with the D3DCREATE_MULTITHREADED flag.
* - There is a limit to how many pictures can be mapped simultaneously (ulNumOutputSurfaces)
* - cuVidDecodePicture may block the calling thread if there are too many pictures pending
* in the decode queue
*/
/**
* \fn CUresult CUDAAPI cuvidCreateDecoder(CUvideodecoder *phDecoder, CUVIDDECODECREATEINFO *pdci)
* Create the decoder object
*/
CUresult CUDAAPI cuvidCreateDecoder(CUvideodecoder *phDecoder, CUVIDDECODECREATEINFO *pdci);
/**
* \fn CUresult CUDAAPI cuvidDestroyDecoder(CUvideodecoder hDecoder)
* Destroy the decoder object
*/
CUresult CUDAAPI cuvidDestroyDecoder(CUvideodecoder hDecoder);
/**
* \fn CUresult CUDAAPI cuvidDecodePicture(CUvideodecoder hDecoder, CUVIDPICPARAMS *pPicParams)
* Decode a single picture (field or frame)
*/
CUresult CUDAAPI cuvidDecodePicture(CUvideodecoder hDecoder, CUVIDPICPARAMS *pPicParams);
#if !defined(__CUVID_DEVPTR64) || defined(__CUVID_INTERNAL)
/**
* \fn CUresult CUDAAPI cuvidMapVideoFrame(CUvideodecoder hDecoder, int nPicIdx, unsigned int *pDevPtr, unsigned int *pPitch, CUVIDPROCPARAMS *pVPP);
* Post-process and map a video frame for use in cuda
*/
CUresult CUDAAPI cuvidMapVideoFrame(CUvideodecoder hDecoder, int nPicIdx,
unsigned int *pDevPtr, unsigned int *pPitch,
CUVIDPROCPARAMS *pVPP);
/**
* \fn CUresult CUDAAPI cuvidUnmapVideoFrame(CUvideodecoder hDecoder, unsigned int DevPtr)
* Unmap a previously mapped video frame
*/
CUresult CUDAAPI cuvidUnmapVideoFrame(CUvideodecoder hDecoder, unsigned int DevPtr);
#endif
#if defined(WIN64) || defined(_WIN64) || defined(__x86_64) || defined(AMD64) || defined(_M_AMD64)
/**
* \fn CUresult CUDAAPI cuvidMapVideoFrame64(CUvideodecoder hDecoder, int nPicIdx, unsigned long long *pDevPtr, unsigned int *pPitch, CUVIDPROCPARAMS *pVPP);
* map a video frame
*/
CUresult CUDAAPI cuvidMapVideoFrame64(CUvideodecoder hDecoder, int nPicIdx, unsigned long long *pDevPtr,
unsigned int *pPitch, CUVIDPROCPARAMS *pVPP);
/**
* \fn CUresult CUDAAPI cuvidUnmapVideoFrame64(CUvideodecoder hDecoder, unsigned long long DevPtr);
* Unmap a previously mapped video frame
*/
CUresult CUDAAPI cuvidUnmapVideoFrame64(CUvideodecoder hDecoder, unsigned long long DevPtr);
#if defined(__CUVID_DEVPTR64) && !defined(__CUVID_INTERNAL)
#define cuvidMapVideoFrame cuvidMapVideoFrame64
#define cuvidUnmapVideoFrame cuvidUnmapVideoFrame64
#endif
#endif
/**
*
* Context-locking: to facilitate multi-threaded implementations, the following 4 functions
* provide a simple mutex-style host synchronization. If a non-NULL context is specified
* in CUVIDDECODECREATEINFO, the codec library will acquire the mutex associated with the given
* context before making any cuda calls.
* A multi-threaded application could create a lock associated with a context handle so that
* multiple threads can safely share the same cuda context:
* - use cuCtxPopCurrent immediately after context creation in order to create a 'floating' context
* that can be passed to cuvidCtxLockCreate.
* - When using a floating context, all cuda calls should only be made within a cuvidCtxLock/cuvidCtxUnlock section.
*
* NOTE: This is a safer alternative to cuCtxPushCurrent and cuCtxPopCurrent, and is not related to video
* decoder in any way (implemented as a critical section associated with cuCtx{Push|Pop}Current calls).
*/
/**
* \fn CUresult CUDAAPI cuvidCtxLockCreate(CUvideoctxlock *pLock, CUcontext ctx)
*/
CUresult CUDAAPI cuvidCtxLockCreate(CUvideoctxlock *pLock, CUcontext ctx);
/**
* \fn CUresult CUDAAPI cuvidCtxLockDestroy(CUvideoctxlock lck)
*/
CUresult CUDAAPI cuvidCtxLockDestroy(CUvideoctxlock lck);
/**
* \fn CUresult CUDAAPI cuvidCtxLock(CUvideoctxlock lck, unsigned int reserved_flags)
*/
CUresult CUDAAPI cuvidCtxLock(CUvideoctxlock lck, unsigned int reserved_flags);
/**
* \fn CUresult CUDAAPI cuvidCtxUnlock(CUvideoctxlock lck, unsigned int reserved_flags)
*/
CUresult CUDAAPI cuvidCtxUnlock(CUvideoctxlock lck, unsigned int reserved_flags);
/** @} */ /* End VIDEO_DECODER */
////////////////////////////////////////////////////////////////////////////////////////////////
#if defined(__cplusplus)
// Auto-lock helper for C++ applications
class CCtxAutoLock
{
private:
CUvideoctxlock m_ctx;
public:
CCtxAutoLock(CUvideoctxlock ctx);
~CCtxAutoLock();
};
}
#endif /* __cplusplus */
#endif // __CUDA_VIDEO_H__

View File

@@ -1,321 +0,0 @@
/*
* This copyright notice applies to this header file only:
*
* Copyright (c) 2010-2016 NVIDIA Corporation
*
* Permission is hereby granted, free of charge, to any person
* obtaining a copy of this software and associated documentation
* files (the "Software"), to deal in the Software without
* restriction, including without limitation the rights to use,
* copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the software, and to permit persons to whom the
* software is furnished to do so, subject to the following
* conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES
* OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND
* NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT
* HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY,
* WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR
* OTHER DEALINGS IN THE SOFTWARE.
*/
/**
* \file nvcuvid.h
* NvCuvid API provides Video Decoding interface to NVIDIA GPU devices.
* \date 2015-2015
* This file contains the interface constants, structure definitions and function prototypes.
*/
#if !defined(__NVCUVID_H__)
#define __NVCUVID_H__
#include "compat/cuda/cuviddec.h"
#if defined(__cplusplus)
extern "C" {
#endif /* __cplusplus */
/*********************************
** Initialization
*********************************/
CUresult CUDAAPI cuvidInit(unsigned int Flags);
////////////////////////////////////////////////////////////////////////////////////////////////
//
// High-level helper APIs for video sources
//
typedef void *CUvideosource;
typedef void *CUvideoparser;
typedef long long CUvideotimestamp;
/**
* \addtogroup VIDEO_PARSER Video Parser
* @{
*/
/*!
* \enum cudaVideoState
* Video Source State
*/
typedef enum {
cudaVideoState_Error = -1, /**< Error state (invalid source) */
cudaVideoState_Stopped = 0, /**< Source is stopped (or reached end-of-stream) */
cudaVideoState_Started = 1 /**< Source is running and delivering data */
} cudaVideoState;
/*!
* \enum cudaAudioCodec
* Audio compression
*/
typedef enum {
cudaAudioCodec_MPEG1=0, /**< MPEG-1 Audio */
cudaAudioCodec_MPEG2, /**< MPEG-2 Audio */
cudaAudioCodec_MP3, /**< MPEG-1 Layer III Audio */
cudaAudioCodec_AC3, /**< Dolby Digital (AC3) Audio */
cudaAudioCodec_LPCM /**< PCM Audio */
} cudaAudioCodec;
/*!
* \struct CUVIDEOFORMAT
* Video format
*/
typedef struct
{
cudaVideoCodec codec; /**< Compression format */
/**
* frame rate = numerator / denominator (for example: 30000/1001)
*/
struct {
unsigned int numerator; /**< frame rate numerator (0 = unspecified or variable frame rate) */
unsigned int denominator; /**< frame rate denominator (0 = unspecified or variable frame rate) */
} frame_rate;
unsigned char progressive_sequence; /**< 0=interlaced, 1=progressive */
unsigned char bit_depth_luma_minus8; /**< high bit depth Luma */
unsigned char bit_depth_chroma_minus8; /**< high bit depth Chroma */
unsigned char reserved1; /**< Reserved for future use */
unsigned int coded_width; /**< coded frame width */
unsigned int coded_height; /**< coded frame height */
/**
* area of the frame that should be displayed
* typical example:
* coded_width = 1920, coded_height = 1088
* display_area = { 0,0,1920,1080 }
*/
struct {
int left; /**< left position of display rect */
int top; /**< top position of display rect */
int right; /**< right position of display rect */
int bottom; /**< bottom position of display rect */
} display_area;
cudaVideoChromaFormat chroma_format; /**< Chroma format */
unsigned int bitrate; /**< video bitrate (bps, 0=unknown) */
/**
* Display Aspect Ratio = x:y (4:3, 16:9, etc)
*/
struct {
int x;
int y;
} display_aspect_ratio;
/**
* Video Signal Description
*/
struct {
unsigned char video_format : 3;
unsigned char video_full_range_flag : 1;
unsigned char reserved_zero_bits : 4;
unsigned char color_primaries;
unsigned char transfer_characteristics;
unsigned char matrix_coefficients;
} video_signal_description;
unsigned int seqhdr_data_length; /**< Additional bytes following (CUVIDEOFORMATEX) */
} CUVIDEOFORMAT;
/*!
* \struct CUVIDEOFORMATEX
* Video format including raw sequence header information
*/
typedef struct
{
CUVIDEOFORMAT format;
unsigned char raw_seqhdr_data[1024];
} CUVIDEOFORMATEX;
/*!
* \struct CUAUDIOFORMAT
* Audio Formats
*/
typedef struct
{
cudaAudioCodec codec; /**< Compression format */
unsigned int channels; /**< number of audio channels */
unsigned int samplespersec; /**< sampling frequency */
unsigned int bitrate; /**< For uncompressed, can also be used to determine bits per sample */
unsigned int reserved1; /**< Reserved for future use */
unsigned int reserved2; /**< Reserved for future use */
} CUAUDIOFORMAT;
/*!
* \enum CUvideopacketflags
* Data packet flags
*/
typedef enum {
CUVID_PKT_ENDOFSTREAM = 0x01, /**< Set when this is the last packet for this stream */
CUVID_PKT_TIMESTAMP = 0x02, /**< Timestamp is valid */
CUVID_PKT_DISCONTINUITY = 0x04 /**< Set when a discontinuity has to be signalled */
} CUvideopacketflags;
/*!
* \struct CUVIDSOURCEDATAPACKET
* Data Packet
*/
typedef struct _CUVIDSOURCEDATAPACKET
{
unsigned long flags; /**< Combination of CUVID_PKT_XXX flags */
unsigned long payload_size; /**< number of bytes in the payload (may be zero if EOS flag is set) */
const unsigned char *payload; /**< Pointer to packet payload data (may be NULL if EOS flag is set) */
CUvideotimestamp timestamp; /**< Presentation timestamp (10MHz clock), only valid if CUVID_PKT_TIMESTAMP flag is set */
} CUVIDSOURCEDATAPACKET;
// Callback for packet delivery
typedef int (CUDAAPI *PFNVIDSOURCECALLBACK)(void *, CUVIDSOURCEDATAPACKET *);
/*!
* \struct CUVIDSOURCEPARAMS
* Source Params
*/
typedef struct _CUVIDSOURCEPARAMS
{
unsigned int ulClockRate; /**< Timestamp units in Hz (0=default=10000000Hz) */
unsigned int uReserved1[7]; /**< Reserved for future use - set to zero */
void *pUserData; /**< Parameter passed in to the data handlers */
PFNVIDSOURCECALLBACK pfnVideoDataHandler; /**< Called to deliver audio packets */
PFNVIDSOURCECALLBACK pfnAudioDataHandler; /**< Called to deliver video packets */
void *pvReserved2[8]; /**< Reserved for future use - set to NULL */
} CUVIDSOURCEPARAMS;
/*!
* \enum CUvideosourceformat_flags
* CUvideosourceformat_flags
*/
typedef enum {
CUVID_FMT_EXTFORMATINFO = 0x100 /**< Return extended format structure (CUVIDEOFORMATEX) */
} CUvideosourceformat_flags;
#if !defined(__APPLE__)
/**
* \fn CUresult CUDAAPI cuvidCreateVideoSource(CUvideosource *pObj, const char *pszFileName, CUVIDSOURCEPARAMS *pParams)
* Create Video Source
*/
CUresult CUDAAPI cuvidCreateVideoSource(CUvideosource *pObj, const char *pszFileName, CUVIDSOURCEPARAMS *pParams);
/**
* \fn CUresult CUDAAPI cuvidCreateVideoSourceW(CUvideosource *pObj, const wchar_t *pwszFileName, CUVIDSOURCEPARAMS *pParams)
* Create Video Source
*/
CUresult CUDAAPI cuvidCreateVideoSourceW(CUvideosource *pObj, const wchar_t *pwszFileName, CUVIDSOURCEPARAMS *pParams);
/**
* \fn CUresult CUDAAPI cuvidDestroyVideoSource(CUvideosource obj)
* Destroy Video Source
*/
CUresult CUDAAPI cuvidDestroyVideoSource(CUvideosource obj);
/**
* \fn CUresult CUDAAPI cuvidSetVideoSourceState(CUvideosource obj, cudaVideoState state)
* Set Video Source state
*/
CUresult CUDAAPI cuvidSetVideoSourceState(CUvideosource obj, cudaVideoState state);
/**
* \fn cudaVideoState CUDAAPI cuvidGetVideoSourceState(CUvideosource obj)
* Get Video Source state
*/
cudaVideoState CUDAAPI cuvidGetVideoSourceState(CUvideosource obj);
/**
* \fn CUresult CUDAAPI cuvidGetSourceVideoFormat(CUvideosource obj, CUVIDEOFORMAT *pvidfmt, unsigned int flags)
* Get Video Source Format
*/
CUresult CUDAAPI cuvidGetSourceVideoFormat(CUvideosource obj, CUVIDEOFORMAT *pvidfmt, unsigned int flags);
/**
* \fn CUresult CUDAAPI cuvidGetSourceAudioFormat(CUvideosource obj, CUAUDIOFORMAT *paudfmt, unsigned int flags)
* Set Video Source state
*/
CUresult CUDAAPI cuvidGetSourceAudioFormat(CUvideosource obj, CUAUDIOFORMAT *paudfmt, unsigned int flags);
#endif
/**
* \struct CUVIDPARSERDISPINFO
*/
typedef struct _CUVIDPARSERDISPINFO
{
int picture_index; /**< */
int progressive_frame; /**< */
int top_field_first; /**< */
int repeat_first_field; /**< Number of additional fields (1=ivtc, 2=frame doubling, 4=frame tripling, -1=unpaired field) */
CUvideotimestamp timestamp; /**< */
} CUVIDPARSERDISPINFO;
//
// Parser callbacks
// The parser will call these synchronously from within cuvidParseVideoData(), whenever a picture is ready to
// be decoded and/or displayed.
//
typedef int (CUDAAPI *PFNVIDSEQUENCECALLBACK)(void *, CUVIDEOFORMAT *);
typedef int (CUDAAPI *PFNVIDDECODECALLBACK)(void *, CUVIDPICPARAMS *);
typedef int (CUDAAPI *PFNVIDDISPLAYCALLBACK)(void *, CUVIDPARSERDISPINFO *);
/**
* \struct CUVIDPARSERPARAMS
*/
typedef struct _CUVIDPARSERPARAMS
{
cudaVideoCodec CodecType; /**< cudaVideoCodec_XXX */
unsigned int ulMaxNumDecodeSurfaces; /**< Max # of decode surfaces (parser will cycle through these) */
unsigned int ulClockRate; /**< Timestamp units in Hz (0=default=10000000Hz) */
unsigned int ulErrorThreshold; /**< % Error threshold (0-100) for calling pfnDecodePicture (100=always call pfnDecodePicture even if picture bitstream is fully corrupted) */
unsigned int ulMaxDisplayDelay; /**< Max display queue delay (improves pipelining of decode with display) - 0=no delay (recommended values: 2..4) */
unsigned int uReserved1[5]; /**< Reserved for future use - set to 0 */
void *pUserData; /**< User data for callbacks */
PFNVIDSEQUENCECALLBACK pfnSequenceCallback; /**< Called before decoding frames and/or whenever there is a format change */
PFNVIDDECODECALLBACK pfnDecodePicture; /**< Called when a picture is ready to be decoded (decode order) */
PFNVIDDISPLAYCALLBACK pfnDisplayPicture; /**< Called whenever a picture is ready to be displayed (display order) */
void *pvReserved2[7]; /**< Reserved for future use - set to NULL */
CUVIDEOFORMATEX *pExtVideoInfo; /**< [Optional] sequence header data from system layer */
} CUVIDPARSERPARAMS;
/**
* \fn CUresult CUDAAPI cuvidCreateVideoParser(CUvideoparser *pObj, CUVIDPARSERPARAMS *pParams)
*/
CUresult CUDAAPI cuvidCreateVideoParser(CUvideoparser *pObj, CUVIDPARSERPARAMS *pParams);
/**
* \fn CUresult CUDAAPI cuvidParseVideoData(CUvideoparser obj, CUVIDSOURCEDATAPACKET *pPacket)
*/
CUresult CUDAAPI cuvidParseVideoData(CUvideoparser obj, CUVIDSOURCEDATAPACKET *pPacket);
/**
* \fn CUresult CUDAAPI cuvidDestroyVideoParser(CUvideoparser obj)
*/
CUresult CUDAAPI cuvidDestroyVideoParser(CUvideoparser obj);
/** @} */ /* END VIDEO_PARSER */
////////////////////////////////////////////////////////////////////////////////////////////////
#if defined(__cplusplus)
}
#endif /* __cplusplus */
#endif // __NVCUVID_H__

View File

@@ -1,42 +0,0 @@
/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef COMPAT_DISPATCH_SEMAPHORE_SEMAPHORE_H
#define COMPAT_DISPATCH_SEMAPHORE_SEMAPHORE_H
#include <dispatch/dispatch.h>
#include <errno.h>
#define sem_t dispatch_semaphore_t
#define sem_post(psem) dispatch_semaphore_signal(*psem)
#define sem_wait(psem) dispatch_semaphore_wait(*psem, DISPATCH_TIME_FOREVER)
#define sem_timedwait(psem, val) dispatch_semaphore_wait(*psem, dispatch_walltime(val, 0))
#define sem_destroy(psem) dispatch_release(*psem)
static inline int compat_sem_init(dispatch_semaphore_t *psem,
int unused, int val)
{
int ret = !!(*psem = dispatch_semaphore_create(val)) - 1;
if (ret < 0)
errno = ENOMEM;
return ret;
}
#define sem_init compat_sem_init
#endif /* COMPAT_DISPATCH_SEMAPHORE_SEMAPHORE_H */

View File

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

View File

@@ -19,8 +19,8 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef COMPAT_MSVCRT_SNPRINTF_H
#define COMPAT_MSVCRT_SNPRINTF_H
#ifndef COMPAT_SNPRINTF_H
#define COMPAT_SNPRINTF_H
#include <stdarg.h>
#include <stdio.h>
@@ -35,4 +35,4 @@ int avpriv_vsnprintf(char *s, size_t n, const char *fmt, va_list ap);
#define _snprintf avpriv_snprintf
#define vsnprintf avpriv_vsnprintf
#endif /* COMPAT_MSVCRT_SNPRINTF_H */
#endif /* COMPAT_SNPRINTF_H */

File diff suppressed because it is too large Load Diff

View File

@@ -23,8 +23,8 @@
* os2threads to pthreads wrapper
*/
#ifndef COMPAT_OS2THREADS_H
#define COMPAT_OS2THREADS_H
#ifndef AVCODEC_OS2PTHREADS_H
#define AVCODEC_OS2PTHREADS_H
#define INCL_DOS
#include <os2.h>
@@ -32,71 +32,57 @@
#undef __STRICT_ANSI__ /* for _beginthread() */
#include <stdlib.h>
#include <sys/builtin.h>
#include <sys/fmutex.h>
#include "libavutil/attributes.h"
typedef struct {
TID tid;
void *(*start_routine)(void *);
void *arg;
void *result;
} pthread_t;
#include "libavutil/mem.h"
typedef TID pthread_t;
typedef void pthread_attr_t;
typedef HMTX pthread_mutex_t;
typedef void pthread_mutexattr_t;
typedef struct {
HEV event_sem;
HEV ack_sem;
volatile unsigned wait_count;
HEV event_sem;
int wait_count;
} pthread_cond_t;
typedef void pthread_condattr_t;
typedef struct {
volatile int done;
_fmutex mtx;
} pthread_once_t;
#define PTHREAD_ONCE_INIT {0, _FMUTEX_INITIALIZER}
struct thread_arg {
void *(*start_routine)(void *);
void *arg;
};
static void thread_entry(void *arg)
{
pthread_t *thread = arg;
struct thread_arg *thread_arg = arg;
thread->result = thread->start_routine(thread->arg);
thread_arg->start_routine(thread_arg->arg);
av_free(thread_arg);
}
static av_always_inline int pthread_create(pthread_t *thread,
const pthread_attr_t *attr,
void *(*start_routine)(void*),
void *arg)
static av_always_inline int pthread_create(pthread_t *thread, const pthread_attr_t *attr, void *(*start_routine)(void*), void *arg)
{
thread->start_routine = start_routine;
thread->arg = arg;
thread->result = NULL;
struct thread_arg *thread_arg;
thread->tid = _beginthread(thread_entry, NULL, 1024 * 1024, thread);
thread_arg = av_mallocz(sizeof(struct thread_arg));
thread_arg->start_routine = start_routine;
thread_arg->arg = arg;
*thread = _beginthread(thread_entry, NULL, 256 * 1024, thread_arg);
return 0;
}
static av_always_inline int pthread_join(pthread_t thread, void **value_ptr)
{
DosWaitThread(&thread.tid, DCWW_WAIT);
if (value_ptr)
*value_ptr = thread.result;
DosWaitThread((PTID)&thread, DCWW_WAIT);
return 0;
}
static av_always_inline int pthread_mutex_init(pthread_mutex_t *mutex,
const pthread_mutexattr_t *attr)
static av_always_inline int pthread_mutex_init(pthread_mutex_t *mutex, const pthread_mutexattr_t *attr)
{
DosCreateMutexSem(NULL, (PHMTX)mutex, 0, FALSE);
@@ -124,11 +110,9 @@ static av_always_inline int pthread_mutex_unlock(pthread_mutex_t *mutex)
return 0;
}
static av_always_inline int pthread_cond_init(pthread_cond_t *cond,
const pthread_condattr_t *attr)
static av_always_inline int pthread_cond_init(pthread_cond_t *cond, const pthread_condattr_t *attr)
{
DosCreateEventSem(NULL, &cond->event_sem, DCE_POSTONE, FALSE);
DosCreateEventSem(NULL, &cond->ack_sem, DCE_POSTONE, FALSE);
cond->wait_count = 0;
@@ -138,16 +122,16 @@ static av_always_inline int pthread_cond_init(pthread_cond_t *cond,
static av_always_inline int pthread_cond_destroy(pthread_cond_t *cond)
{
DosCloseEventSem(cond->event_sem);
DosCloseEventSem(cond->ack_sem);
return 0;
}
static av_always_inline int pthread_cond_signal(pthread_cond_t *cond)
{
if (!__atomic_cmpxchg32(&cond->wait_count, 0, 0)) {
if (cond->wait_count > 0) {
DosPostEventSem(cond->event_sem);
DosWaitEventSem(cond->ack_sem, SEM_INDEFINITE_WAIT);
cond->wait_count--;
}
return 0;
@@ -155,47 +139,26 @@ static av_always_inline int pthread_cond_signal(pthread_cond_t *cond)
static av_always_inline int pthread_cond_broadcast(pthread_cond_t *cond)
{
while (!__atomic_cmpxchg32(&cond->wait_count, 0, 0))
pthread_cond_signal(cond);
while (cond->wait_count > 0) {
DosPostEventSem(cond->event_sem);
cond->wait_count--;
}
return 0;
}
static av_always_inline int pthread_cond_wait(pthread_cond_t *cond,
pthread_mutex_t *mutex)
static av_always_inline int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
{
__atomic_increment(&cond->wait_count);
cond->wait_count++;
pthread_mutex_unlock(mutex);
DosWaitEventSem(cond->event_sem, SEM_INDEFINITE_WAIT);
__atomic_decrement(&cond->wait_count);
DosPostEventSem(cond->ack_sem);
pthread_mutex_lock(mutex);
return 0;
}
static av_always_inline int pthread_once(pthread_once_t *once_control,
void (*init_routine)(void))
{
if (!once_control->done)
{
_fmutex_request(&once_control->mtx, 0);
if (!once_control->done)
{
init_routine();
once_control->done = 1;
}
_fmutex_release(&once_control->mtx);
}
return 0;
}
#endif /* COMPAT_OS2THREADS_H */
#endif /* AVCODEC_OS2PTHREADS_H */

View File

@@ -1,352 +0,0 @@
#!/usr/bin/env perl
# make_sunver.pl
#
# Copyright (C) 2010, 2011, 2012, 2013
# Free Software Foundation, Inc.
#
# This file is free software; you can redistribute it and/or modify it
# under the terms of the GNU General Public License as published by
# the Free Software Foundation; either version 3 of the License, or
# (at your option) any later version.
#
# This program is distributed in the hope that it will be useful, but
# WITHOUT ANY WARRANTY; without even the implied warranty of
# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
# General Public License for more details.
#
# You should have received a copy of the GNU General Public License
# along with this program; see the file COPYING.GPLv3. If not see
# <http://www.gnu.org/licenses/>.
# This script takes at least two arguments, a GNU style version script and
# a list of object and archive files, and generates a corresponding Sun
# style version script as follows:
#
# Each glob pattern, C++ mangled pattern or literal in the input script is
# matched against all global symbols in the input objects, emitting those
# that matched (or nothing if no match was found).
# A comment with the original pattern and its type is left in the output
# file to make it easy to understand the matches.
#
# It uses elfdump when present (native), GNU readelf otherwise.
# It depends on the GNU version of c++filt, since it must understand the
# GNU mangling style.
use FileHandle;
use IPC::Open2;
# Enforce C locale.
$ENV{'LC_ALL'} = "C";
$ENV{'LANG'} = "C";
# Input version script, GNU style.
my $symvers = shift;
##########
# Get all the symbols from the library, match them, and add them to a hash.
my %sym_hash = ();
# List of objects and archives to process.
my @OBJECTS = ();
# List of shared objects to omit from processing.
my @SHAREDOBJS = ();
# Filter out those input archives that have corresponding shared objects to
# avoid adding all symbols matched in the archive to the output map.
foreach $file (@ARGV) {
if (($so = $file) =~ s/\.a$/.so/ && -e $so) {
printf STDERR "omitted $file -> $so\n";
push (@SHAREDOBJS, $so);
} else {
push (@OBJECTS, $file);
}
}
# We need to detect and ignore hidden symbols. Solaris nm can only detect
# this in the harder to parse default output format, and GNU nm not at all,
# so use elfdump -s in the native case and GNU readelf -s otherwise.
# GNU objdump -t cannot be used since it produces a variable number of
# columns.
# The path to elfdump.
my $elfdump = "/usr/ccs/bin/elfdump";
if (-f $elfdump) {
open ELFDUMP,$elfdump.' -s '.(join ' ',@OBJECTS).'|' or die $!;
my $skip_arsym = 0;
while (<ELFDUMP>) {
chomp;
# Ignore empty lines.
if (/^$/) {
# End of archive symbol table, stop skipping.
$skip_arsym = 0 if $skip_arsym;
next;
}
# Keep skipping until end of archive symbol table.
next if ($skip_arsym);
# Ignore object name header for individual objects and archives.
next if (/:$/);
# Ignore table header lines.
next if (/^Symbol Table Section:/);
next if (/index.*value.*size/);
# Start of archive symbol table: start skipping.
if (/^Symbol Table: \(archive/) {
$skip_arsym = 1;
next;
}
# Split table.
(undef, undef, undef, undef, $bind, $oth, undef, $shndx, $name) = split;
# Error out for unknown input.
die "unknown input line:\n$_" unless defined($bind);
# Ignore local symbols.
next if ($bind eq "LOCL");
# Ignore hidden symbols.
next if ($oth eq "H");
# Ignore undefined symbols.
next if ($shndx eq "UNDEF");
# Error out for unhandled cases.
if ($bind !~ /^(GLOB|WEAK)/ or $oth ne "D") {
die "unhandled symbol:\n$_";
}
# Remember symbol.
$sym_hash{$name}++;
}
close ELFDUMP or die "$elfdump error";
} else {
open READELF, 'readelf -s -W '.(join ' ',@OBJECTS).'|' or die $!;
# Process each symbol.
while (<READELF>) {
chomp;
# Ignore empty lines.
next if (/^$/);
# Ignore object name header.
next if (/^File: .*$/);
# Ignore table header lines.
next if (/^Symbol table.*contains.*:/);
next if (/Num:.*Value.*Size/);
# Split table.
(undef, undef, undef, undef, $bind, $vis, $ndx, $name) = split;
# Error out for unknown input.
die "unknown input line:\n$_" unless defined($bind);
# Ignore local symbols.
next if ($bind eq "LOCAL");
# Ignore hidden symbols.
next if ($vis eq "HIDDEN");
# Ignore undefined symbols.
next if ($ndx eq "UND");
# Error out for unhandled cases.
if ($bind !~ /^(GLOBAL|WEAK)/ or $vis ne "DEFAULT") {
die "unhandled symbol:\n$_";
}
# Remember symbol.
$sym_hash{$name}++;
}
close READELF or die "readelf error";
}
##########
# The various types of glob patterns.
#
# A glob pattern that is to be applied to the demangled name: 'cxx'.
# A glob patterns that applies directly to the name in the .o files: 'glob'.
# This pattern is ignored; used for local variables (usually just '*'): 'ign'.
# The type of the current pattern.
my $glob = 'glob';
# We're currently inside `extern "C++"', which Sun ld doesn't understand.
my $in_extern = 0;
# The c++filt command to use. This *must* be GNU c++filt; the Sun Studio
# c++filt doesn't handle the GNU mangling style.
my $cxxfilt = $ENV{'CXXFILT'} || "c++filt";
# The current version name.
my $current_version = "";
# Was there any attempt to match a symbol to this version?
my $matches_attempted;
# The number of versions which matched this symbol.
my $matched_symbols;
open F,$symvers or die $!;
# Print information about generating this file
print "# This file was generated by make_sunver.pl. DO NOT EDIT!\n";
print "# It was generated by:\n";
printf "# %s %s %s\n", $0, $symvers, (join ' ',@ARGV);
printf "# Omitted archives with corresponding shared libraries: %s\n",
(join ' ', @SHAREDOBJS) if $#SHAREDOBJS >= 0;
print "#\n\n";
print "\$mapfile_version 2\n";
while (<F>) {
# Lines of the form '};'
if (/^([ \t]*)(\}[ \t]*;[ \t]*)$/) {
$glob = 'glob';
if ($in_extern) {
$in_extern--;
print "$1##$2\n";
} else {
print;
}
next;
}
# Lines of the form '} SOME_VERSION_NAME_1.0;'
if (/^[ \t]*\}[ \tA-Z0-9_.a-z]+;[ \t]*$/) {
$glob = 'glob';
# We tried to match symbols agains this version, but none matched.
# Emit dummy hidden symbol to avoid marking this version WEAK.
if ($matches_attempted && $matched_symbols == 0) {
print " hidden:\n";
print " .force_WEAK_off_$current_version = DATA S0x0 V0x0;\n";
}
print; next;
}
# Comment and blank lines
if (/^[ \t]*\#/) { print; next; }
if (/^[ \t]*$/) { print; next; }
# Lines of the form '{'
if (/^([ \t]*){$/) {
if ($in_extern) {
print "$1##{\n";
} else {
print;
}
next;
}
# Lines of the form 'SOME_VERSION_NAME_1.1 {'
if (/^([A-Z0-9_.]+)[ \t]+{$/) {
# Record version name.
$current_version = $1;
# Reset match attempts, #matched symbols for this version.
$matches_attempted = 0;
$matched_symbols = 0;
print "SYMBOL_VERSION $1 {\n";
next;
}
# Ignore 'global:'
if (/^[ \t]*global:$/) { print; next; }
# After 'local:', globs should be ignored, they won't be exported.
if (/^[ \t]*local:$/) {
$glob = 'ign';
print;
next;
}
# After 'extern "C++"', globs are C++ patterns
if (/^([ \t]*)(extern \"C\+\+\"[ \t]*)$/) {
$in_extern++;
$glob = 'cxx';
# Need to comment, Sun ld cannot handle this.
print "$1##$2\n"; next;
}
# Chomp newline now we're done with passing through the input file.
chomp;
# Catch globs. Note that '{}' is not allowed in globs by this script,
# so only '*' and '[]' are available.
if (/^([ \t]*)([^ \t;{}#]+);?[ \t]*$/) {
my $ws = $1;
my $ptn = $2;
# Turn the glob into a regex by replacing '*' with '.*', '?' with '.'.
# Keep $ptn so we can still print the original form.
($pattern = $ptn) =~ s/\*/\.\*/g;
$pattern =~ s/\?/\./g;
if ($glob eq 'ign') {
# We're in a local: * section; just continue.
print "$_\n";
next;
}
# Print the glob commented for human readers.
print "$ws##$ptn ($glob)\n";
# We tried to match a symbol to this version.
$matches_attempted++;
if ($glob eq 'glob') {
my %ptn_syms = ();
# Match ptn against symbols in %sym_hash.
foreach my $sym (keys %sym_hash) {
# Maybe it matches one of the patterns based on the symbol in
# the .o file.
$ptn_syms{$sym}++ if ($sym =~ /^$pattern$/);
}
foreach my $sym (sort keys(%ptn_syms)) {
$matched_symbols++;
print "$ws$sym;\n";
}
} elsif ($glob eq 'cxx') {
my %dem_syms = ();
# Verify that we're actually using GNU c++filt. Other versions
# most likely cannot handle GNU style symbol mangling.
my $cxxout = `$cxxfilt --version 2>&1`;
$cxxout =~ m/GNU/ or die "$0 requires GNU c++filt to function";
# Talk to c++filt through a pair of file descriptors.
# Need to start a fresh instance per pattern, otherwise the
# process grows to 500+ MB.
my $pid = open2(*FILTIN, *FILTOUT, $cxxfilt) or die $!;
# Match ptn against symbols in %sym_hash.
foreach my $sym (keys %sym_hash) {
# No? Well, maybe its demangled form matches one of those
# patterns.
printf FILTOUT "%s\n",$sym;
my $dem = <FILTIN>;
chomp $dem;
$dem_syms{$sym}++ if ($dem =~ /^$pattern$/);
}
close FILTOUT or die "c++filt error";
close FILTIN or die "c++filt error";
# Need to wait for the c++filt process to avoid lots of zombies.
waitpid $pid, 0;
foreach my $sym (sort keys(%dem_syms)) {
$matched_symbols++;
print "$ws$sym;\n";
}
} else {
# No? Well, then ignore it.
}
next;
}
# Important sanity check. This script can't handle lots of formats
# that GNU ld can, so be sure to error out if one is seen!
die "strange line `$_'";
}
close F;

View File

@@ -16,8 +16,8 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef COMPAT_TMS470_MATH_H
#define COMPAT_TMS470_MATH_H
#ifndef FFMPEG_COMPAT_TMS470_MATH_H
#define FFMPEG_COMPAT_TMS470_MATH_H
#include_next <math.h>
@@ -27,4 +27,4 @@
#define INFINITY (*(const float*)((const unsigned []){ 0x7f800000 }))
#define NAN (*(const float*)((const unsigned []){ 0x7fc00000 }))
#endif /* COMPAT_TMS470_MATH_H */
#endif /* FFMPEG_COMPAT_TMS470_MATH_H */

View File

@@ -19,9 +19,6 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef COMPAT_VA_COPY_H
#define COMPAT_VA_COPY_H
#include <stdarg.h>
#if !defined(va_copy) && defined(_MSC_VER)
@@ -30,5 +27,3 @@
#if !defined(va_copy) && defined(__GNUC__) && __GNUC__ < 3
#define va_copy(dst, src) __va_copy(dst, src)
#endif
#endif /* COMPAT_VA_COPY_H */

View File

@@ -26,8 +26,8 @@
* w32threads to pthreads wrapper
*/
#ifndef COMPAT_W32PTHREADS_H
#define COMPAT_W32PTHREADS_H
#ifndef FFMPEG_COMPAT_W32PTHREADS_H
#define FFMPEG_COMPAT_W32PTHREADS_H
/* Build up a pthread-like API using underlying Windows API. Have only static
* methods so as to not conflict with a potentially linked in pthread-win32
@@ -39,12 +39,6 @@
#include <windows.h>
#include <process.h>
#if _WIN32_WINNT < 0x0600 && defined(__MINGW32__)
#undef MemoryBarrier
#define MemoryBarrier __sync_synchronize
#endif
#include "libavutil/attributes.h"
#include "libavutil/common.h"
#include "libavutil/internal.h"
#include "libavutil/mem.h"
@@ -60,56 +54,52 @@ typedef struct pthread_t {
* not mutexes */
typedef CRITICAL_SECTION pthread_mutex_t;
/* This is the CONDITION_VARIABLE typedef for using Windows' native
* conditional variables on kernels 6.0+. */
#if HAVE_CONDITION_VARIABLE_PTR
typedef CONDITION_VARIABLE pthread_cond_t;
#else
/* This is the CONDITIONAL_VARIABLE typedef for using Window's native
* conditional variables on kernels 6.0+.
* MinGW does not currently have this typedef. */
typedef struct pthread_cond_t {
void *Ptr;
void *ptr;
} pthread_cond_t;
/* function pointers to conditional variable API on windows 6.0+ kernels */
#if _WIN32_WINNT < 0x0600
static void (WINAPI *cond_broadcast)(pthread_cond_t *cond);
static void (WINAPI *cond_init)(pthread_cond_t *cond);
static void (WINAPI *cond_signal)(pthread_cond_t *cond);
static BOOL (WINAPI *cond_wait)(pthread_cond_t *cond, pthread_mutex_t *mutex,
DWORD milliseconds);
#else
#define cond_init InitializeConditionVariable
#define cond_broadcast WakeAllConditionVariable
#define cond_signal WakeConditionVariable
#define cond_wait SleepConditionVariableCS
#endif
#if _WIN32_WINNT >= 0x0600
#define InitializeCriticalSection(x) InitializeCriticalSectionEx(x, 0, 0)
#define WaitForSingleObject(a, b) WaitForSingleObjectEx(a, b, FALSE)
#endif
static av_unused unsigned __stdcall attribute_align_arg win32thread_worker(void *arg)
static unsigned __stdcall attribute_align_arg win32thread_worker(void *arg)
{
pthread_t *h = arg;
h->ret = h->func(h->arg);
return 0;
}
static av_unused int pthread_create(pthread_t *thread, const void *unused_attr,
void *(*start_routine)(void*), void *arg)
static int pthread_create(pthread_t *thread, const void *unused_attr,
void *(*start_routine)(void*), void *arg)
{
thread->func = start_routine;
thread->arg = arg;
#if HAVE_WINRT
thread->handle = (void*)CreateThread(NULL, 0, win32thread_worker, thread,
0, NULL);
#else
thread->handle = (void*)_beginthreadex(NULL, 0, win32thread_worker, thread,
0, NULL);
#endif
return !thread->handle;
}
static av_unused int pthread_join(pthread_t thread, void **value_ptr)
static void pthread_join(pthread_t thread, void **value_ptr)
{
DWORD ret = WaitForSingleObject(thread.handle, INFINITE);
if (ret != WAIT_OBJECT_0) {
if (ret == WAIT_ABANDONED)
return EINVAL;
else
return EDEADLK;
}
if (ret != WAIT_OBJECT_0)
return;
if (value_ptr)
*value_ptr = thread.ret;
CloseHandle(thread.handle);
return 0;
}
static inline int pthread_mutex_init(pthread_mutex_t *m, void* attr)
@@ -133,115 +123,8 @@ static inline int pthread_mutex_unlock(pthread_mutex_t *m)
return 0;
}
#if _WIN32_WINNT >= 0x0600
typedef INIT_ONCE pthread_once_t;
#define PTHREAD_ONCE_INIT INIT_ONCE_STATIC_INIT
static av_unused int pthread_once(pthread_once_t *once_control, void (*init_routine)(void))
{
BOOL pending = FALSE;
InitOnceBeginInitialize(once_control, 0, &pending, NULL);
if (pending)
init_routine();
InitOnceComplete(once_control, 0, NULL);
return 0;
}
static inline int pthread_cond_init(pthread_cond_t *cond, const void *unused_attr)
{
InitializeConditionVariable(cond);
return 0;
}
/* native condition variables do not destroy */
static inline int pthread_cond_destroy(pthread_cond_t *cond)
{
return 0;
}
static inline int pthread_cond_broadcast(pthread_cond_t *cond)
{
WakeAllConditionVariable(cond);
return 0;
}
static inline int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
{
SleepConditionVariableCS(cond, mutex, INFINITE);
return 0;
}
static inline int pthread_cond_signal(pthread_cond_t *cond)
{
WakeConditionVariable(cond);
return 0;
}
#else // _WIN32_WINNT < 0x0600
/* atomic init state of dynamically loaded functions */
static LONG w32thread_init_state = 0;
static av_unused void w32thread_init(void);
/* for pre-Windows 6.0 platforms, define INIT_ONCE struct,
* compatible to the one used in the native API */
typedef union pthread_once_t {
void * Ptr; ///< For the Windows 6.0+ native functions
LONG state; ///< For the pre-Windows 6.0 compat code
} pthread_once_t;
#define PTHREAD_ONCE_INIT {0}
/* function pointers to init once API on windows 6.0+ kernels */
static BOOL (WINAPI *initonce_begin)(pthread_once_t *lpInitOnce, DWORD dwFlags, BOOL *fPending, void **lpContext);
static BOOL (WINAPI *initonce_complete)(pthread_once_t *lpInitOnce, DWORD dwFlags, void *lpContext);
/* pre-Windows 6.0 compat using a spin-lock */
static inline void w32thread_once_fallback(LONG volatile *state, void (*init_routine)(void))
{
switch (InterlockedCompareExchange(state, 1, 0)) {
/* Initial run */
case 0:
init_routine();
InterlockedExchange(state, 2);
break;
/* Another thread is running init */
case 1:
while (1) {
MemoryBarrier();
if (*state == 2)
break;
Sleep(0);
}
break;
/* Initialization complete */
case 2:
break;
}
}
static av_unused int pthread_once(pthread_once_t *once_control, void (*init_routine)(void))
{
w32thread_once_fallback(&w32thread_init_state, w32thread_init);
/* Use native functions on Windows 6.0+ */
if (initonce_begin && initonce_complete) {
BOOL pending = FALSE;
initonce_begin(once_control, 0, &pending, NULL);
if (pending)
init_routine();
initonce_complete(once_control, 0, NULL);
return 0;
}
w32thread_once_fallback(&once_control->state, init_routine);
return 0;
}
/* for pre-Windows 6.0 platforms we need to define and use our own condition
* variable and api */
typedef struct win32_cond_t {
pthread_mutex_t mtx_broadcast;
pthread_mutex_t mtx_waiter_count;
@@ -251,19 +134,9 @@ typedef struct win32_cond_t {
volatile int is_broadcast;
} win32_cond_t;
/* function pointers to conditional variable API on windows 6.0+ kernels */
static void (WINAPI *cond_broadcast)(pthread_cond_t *cond);
static void (WINAPI *cond_init)(pthread_cond_t *cond);
static void (WINAPI *cond_signal)(pthread_cond_t *cond);
static BOOL (WINAPI *cond_wait)(pthread_cond_t *cond, pthread_mutex_t *mutex,
DWORD milliseconds);
static av_unused int pthread_cond_init(pthread_cond_t *cond, const void *unused_attr)
static int pthread_cond_init(pthread_cond_t *cond, const void *unused_attr)
{
win32_cond_t *win32_cond = NULL;
w32thread_once_fallback(&w32thread_init_state, w32thread_init);
if (cond_init) {
cond_init(cond);
return 0;
@@ -273,7 +146,7 @@ static av_unused int pthread_cond_init(pthread_cond_t *cond, const void *unused_
win32_cond = av_mallocz(sizeof(win32_cond_t));
if (!win32_cond)
return ENOMEM;
cond->Ptr = win32_cond;
cond->ptr = win32_cond;
win32_cond->semaphore = CreateSemaphore(NULL, 0, 0x7fffffff, NULL);
if (!win32_cond->semaphore)
return ENOMEM;
@@ -286,12 +159,12 @@ static av_unused int pthread_cond_init(pthread_cond_t *cond, const void *unused_
return 0;
}
static av_unused int pthread_cond_destroy(pthread_cond_t *cond)
static void pthread_cond_destroy(pthread_cond_t *cond)
{
win32_cond_t *win32_cond = cond->Ptr;
win32_cond_t *win32_cond = cond->ptr;
/* native condition variables do not destroy */
if (cond_init)
return 0;
return;
/* non native condition variables */
CloseHandle(win32_cond->semaphore);
@@ -299,18 +172,17 @@ static av_unused int pthread_cond_destroy(pthread_cond_t *cond)
pthread_mutex_destroy(&win32_cond->mtx_waiter_count);
pthread_mutex_destroy(&win32_cond->mtx_broadcast);
av_freep(&win32_cond);
cond->Ptr = NULL;
return 0;
cond->ptr = NULL;
}
static av_unused int pthread_cond_broadcast(pthread_cond_t *cond)
static void pthread_cond_broadcast(pthread_cond_t *cond)
{
win32_cond_t *win32_cond = cond->Ptr;
win32_cond_t *win32_cond = cond->ptr;
int have_waiter;
if (cond_broadcast) {
cond_broadcast(cond);
return 0;
return;
}
/* non native condition variables */
@@ -332,12 +204,11 @@ static av_unused int pthread_cond_broadcast(pthread_cond_t *cond)
} else
pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
pthread_mutex_unlock(&win32_cond->mtx_broadcast);
return 0;
}
static av_unused int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
static int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
{
win32_cond_t *win32_cond = cond->Ptr;
win32_cond_t *win32_cond = cond->ptr;
int last_waiter;
if (cond_wait) {
cond_wait(cond, mutex, INFINITE);
@@ -367,13 +238,13 @@ static av_unused int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mu
return pthread_mutex_lock(mutex);
}
static av_unused int pthread_cond_signal(pthread_cond_t *cond)
static void pthread_cond_signal(pthread_cond_t *cond)
{
win32_cond_t *win32_cond = cond->Ptr;
win32_cond_t *win32_cond = cond->ptr;
int have_waiter;
if (cond_signal) {
cond_signal(cond);
return 0;
return;
}
pthread_mutex_lock(&win32_cond->mtx_broadcast);
@@ -390,11 +261,9 @@ static av_unused int pthread_cond_signal(pthread_cond_t *cond)
}
pthread_mutex_unlock(&win32_cond->mtx_broadcast);
return 0;
}
#endif
static av_unused void w32thread_init(void)
static void w32thread_init(void)
{
#if _WIN32_WINNT < 0x0600
HANDLE kernel_dll = GetModuleHandle(TEXT("kernel32.dll"));
@@ -407,12 +276,8 @@ static av_unused void w32thread_init(void)
(void*)GetProcAddress(kernel_dll, "WakeConditionVariable");
cond_wait =
(void*)GetProcAddress(kernel_dll, "SleepConditionVariableCS");
initonce_begin =
(void*)GetProcAddress(kernel_dll, "InitOnceBeginInitialize");
initonce_complete =
(void*)GetProcAddress(kernel_dll, "InitOnceComplete");
#endif
}
#endif /* COMPAT_W32PTHREADS_H */
#endif /* FFMPEG_COMPAT_W32PTHREADS_H */

View File

@@ -1,9 +0,0 @@
#!/bin/sh
LINK_EXE_PATH=$(dirname "$(command -v cl)")/link
if [ -x "$LINK_EXE_PATH" ]; then
"$LINK_EXE_PATH" $@
else
link $@
fi
exit $?

2285
configure vendored

File diff suppressed because it is too large Load Diff

9
doc/.gitignore vendored
View File

@@ -1,9 +0,0 @@
/*.1
/*.3
/*.html
/*.pod
/config.texi
/avoptions_codec.texi
/avoptions_format.texi
/fate.txt
/print_options

View File

@@ -2,598 +2,19 @@ Never assume the API of libav* to be stable unless at least 1 month has passed
since the last major version increase or the API was added.
The last version increases were:
libavcodec: 2015-08-28
libavdevice: 2015-08-28
libavfilter: 2015-08-28
libavformat: 2015-08-28
libavresample: 2015-08-28
libpostproc: 2015-08-28
libswresample: 2015-08-28
libswscale: 2015-08-28
libavutil: 2015-08-28
libavcodec: 2013-03-xx
libavdevice: 2013-03-xx
libavfilter: 2013-12-xx
libavformat: 2013-03-xx
libavresample: 2012-10-05
libpostproc: 2011-04-18
libswresample: 2011-09-19
libswscale: 2011-06-20
libavutil: 2012-10-22
API changes, most recent first:
-------- 8< --------- FFmpeg 3.2 was cut here -------- 8< ---------
2016-10-24 - 73ead47 - lavf 57.55.100 - avformat.h
Add AV_DISPOSITION_TIMED_THUMBNAILS
2016-10-24 - a246fef - lavf 57.54.100 - avformat.h
Add avformat_init_output() and AVSTREAM_INIT_IN_ macros
2016-10-22 - f5495c9 - lavu 55.33.100 - avassert.h
Add av_assert0_fpu() / av_assert2_fpu()
2016-10-07 - 3f9137c / 32c8359 - lavc 57.61.100 / 57.24.0 - avcodec.h
Decoders now export the frame timestamp as AVFrame.pts. It was
previously exported as AVFrame.pkt_pts, which is now deprecated.
Note: When decoding, AVFrame.pts uses the stream/packet timebase,
and not the codec timebase.
2016-09-28 - eba0414 - lavu 55.32.100 / 55.16.0 - hwcontext.h hwcontext_qsv.h
Add AV_HWDEVICE_TYPE_QSV and a new installed header with QSV-specific
hwcontext definitions.
2016-09-26 - 32c25f0 - lavc 57.59.100 / 57.23.0 - avcodec.h
AVCodecContext.hw_frames_ctx now may be used by decoders.
2016-09-27 - f0b6f72 - lavf 57.51.100 - avformat.h
Add av_stream_get_codec_timebase()
2016-09-27 - 23c0779 - lswr 2.2.100 - swresample.h
Add swr_build_matrix().
2016-09-23 - 30d3e36 - lavc 57.58.100 - avcodec.h
Add AV_CODEC_CAP_AVOID_PROBING codec capability flag.
2016-09-14 - ae1dd0c - lavf 57.49.100 - avformat.h
Add avformat_transfer_internal_stream_timing_info helper to help with stream
copy.
2016-08-29 - 4493390 - lavfi 6.58.100 - avfilter.h
Add AVFilterContext.nb_threads.
2016-08-15 - c3c4c72 - lavc 57.53.100 - avcodec.h
Add trailing_padding to AVCodecContext to match the corresponding
field in AVCodecParameters.
2016-08-15 - b746ed7 - lavc 57.52.100 - avcodec.h
Add a new API for chained BSF filters and passthrough (null) BSF --
av_bsf_list_alloc(), av_bsf_list_free(), av_bsf_list_append(),
av_bsf_list_append2(), av_bsf_list_finalize(), av_bsf_list_parse_str()
and av_bsf_get_null_filter().
2016-08-04 - 82a33c8 - lavf 57.46.100 - avformat.h
Add av_get_frame_filename2()
2016-07-09 - 775389f / 90f469a - lavc 57.50.100 / 57.20.0 - avcodec.h
Add FF_PROFILE_H264_MULTIVIEW_HIGH and FF_PROFILE_H264_STEREO_HIGH.
2016-06-30 - c1c7e0ab - lavf 57.41.100 - avformat.h
Moved codecpar field from AVStream to the end of the struct, so that
the following private fields are in the same location as in FFmpeg 3.0 (lavf 57.25.100).
2016-06-30 - 042fb69d - lavu 55.28.100 - frame.h
Moved hw_frames_ctx field from AVFrame to the end of the struct, so that
the following private fields are in the same location as in FFmpeg 3.0 (lavu 55.17.103).
2016-06-29 - 1a751455 - lavfi 6.47.100 - avfilter.h
Fix accidental ABI breakage in AVFilterContext.
ABI was broken in 8688d3a, lavfi 6.42.100 and released as ffmpeg 3.1.
Because of this, ffmpeg and ffplay built against lavfi>=6.42.100 will not be
compatible with lavfi>=6.47.100. Potentially also affects other users of
libavfilter if they are using one of the affected fields.
-------- 8< --------- FFmpeg 3.1 was cut here -------- 8< ---------
2016-06-26 - 481f320 / 1c9e861 - lavu 55.27.100 / 55.13.0 - hwcontext.h
Add av_hwdevice_ctx_create().
2016-06-26 - b95534b / e47b8bb - lavc 57.48.101 / 57.19.1 - avcodec.h
Adjust values for JPEG 2000 profiles.
2016-06-23 - 5d75e46 / db7968b - lavf 57.40.100 / 57.7.0 - avio.h
Add AVIODataMarkerType, write_data_type, ignore_boundary_point and
avio_write_marker.
2016-06-23 - abb3cc4 / 0c4468d - lavu 55.26.100 / 55.12.0 - opt.h
Add av_stereo3d_type_name() and av_stereo3d_from_name().
2016-06-22 - 3689efe / c46db38 - lavu 55.25.100 / 55.11.0 - hwcontext_dxva2.h
Add new installed header with DXVA2-specific hwcontext definitions.
2016-04-27 - fb91871 - lavu 55.23.100 - log.h
Add a new function av_log_format_line2() which returns number of bytes
written to the target buffer.
2016-04-21 - 7fc329e - lavc 57.37.100 - avcodec.h
Add a new audio/video encoding and decoding API with decoupled input
and output -- avcodec_send_packet(), avcodec_receive_frame(),
avcodec_send_frame() and avcodec_receive_packet().
2016-04-17 - af9cac1 / 33d1898 - lavc 57.35.100 / 57.15.0 - avcodec.h
Add a new bitstream filtering API working with AVPackets.
Deprecate the old bitstream filtering API.
2016-04-14 - 8688d3a / 07a844f - lavfi 6.42.100 / 6.3.0 - avfilter.h
Add AVFilterContext.hw_device_ctx.
2016-04-14 - 28abb21 / 551c677 - lavu 55.22.100 / 55.9.0 - hwcontext_vaapi.h
Add new installed header with VAAPI-specific hwcontext definitions.
2016-04-14 - afccfaf / b1f01e8 - lavu 55.21.100 / 55.7.0 - hwcontext.h
Add AVHWFramesConstraints and associated API.
2016-04-11 - 6f69f7a / 9200514 - lavf 57.33.100 / 57.5.0 - avformat.h
Add AVStream.codecpar, deprecate AVStream.codec.
2016-04-02 - e8a9b64 - lavu 55.20.100 - base64.h
Add AV_BASE64_DECODE_SIZE(x) macro.
2016-xx-xx - lavc 57.33.100 / 57.14.0 - avcodec.h
f9b1cf1 / 998e1b8 - Add AVCodecParameters and its related API.
e6053b3 / a806834 - Add av_get_audio_frame_duration2().
2016-03-11 - 6d8ab35 - lavf/lavc 57.28.101
Add requirement to bitstream filtering API that returned packets with
size == 0 and side_data_elems == 0 are to be skipped by the caller.
2016-03-04 - 9362973 - lavf 57.28.100
Add protocol blacklisting API
2016-02-28 - 4dd4d53 - lavc 57.27.101
Validate AVFrame returned by get_buffer2 to have required
planes not NULL and unused planes set to NULL as crashes
and buffer overflow are possible with certain streams if
that is not the case.
2016-02-26 - 30e7685 - lavc 57.27.100 - avcodec.h
"flags2" decoding option now allows the flag "ass_ro_flush_noop" preventing
the reset of the ASS ReadOrder field on flush. This affects the content of
AVSubtitles.rects[N]->ass when "sub_text_format" is set to "ass" (see
previous entry).
2016-02-26 - 2941282 - lavc 57.26.100 - avcodec.h
Add a "sub_text_format" subtitles decoding option allowing the values "ass"
(recommended) and "ass_with_timings" (not recommended, deprecated, default).
The default value for this option will change to "ass" at the next major
libavcodec version bump.
The current default is "ass_with_timings" for compatibility. This means that
all subtitles text decoders currently still output ASS with timings printed
as strings in the AVSubtitles.rects[N]->ass fields.
Setting "sub_text_format" to "ass" allows a better timing accuracy (ASS
timing is limited to a 1/100 time base, so this is relevant for any subtitles
format needing a bigger one), ease timing adjustments, and prevents the need
of removing the timing from the decoded string yourself. This form is also
known as "the Matroska form". The timing information (start time, duration)
can be found in the AVSubtitles fields.
2016-02-24 - 7e49cdd / 7b3214d0 - lavc 57.25.100 / 57.13.0 - avcodec.h
Add AVCodecContext.hw_frames_ctx.
2016-02-24 - 1042402 / b3dd30d - lavfi 6.36.100 / 6.2.0 - avfilter.h
avfilter.h - Add AVFilterLink.hw_frames_ctx.
buffersrc.h - Add AVBufferSrcParameters and functions for handling it.
2016-02-23 - 14f7a3d - lavc 57.25.100
Add AV_PKT_DATA_MPEGTS_STREAM_ID for exporting the MPEGTS stream ID.
2016-02-18 - 08acab8 - lavu 55.18.100 - audio_fifo.h
Add av_audio_fifo_peek_at().
2016-xx-xx - lavu 55.18.100 / 55.6.0
26abd51 / 721a4ef buffer.h - Add av_buffer_pool_init2().
1a70878 / 89923e4 hwcontext.h - Add a new installed header hwcontext.h with a new API
for handling hwaccel frames.
6992276 / ad884d1 hwcontext_cuda.h - Add a new installed header hwcontext_cuda.h with
CUDA-specific hwcontext definitions.
d779d8d / a001ce3 hwcontext_vdpau.h - Add a new installed header hwcontext_vdpau.h with
VDPAU-specific hwcontext definitions.
63c3e35 / 7bc780c pixfmt.h - Add AV_PIX_FMT_CUDA.
-------- 8< --------- FFmpeg 3.0 was cut here -------- 8< ---------
2016-02-10 - bc9a596 / 9f61abc - lavf 57.25.100 / 57.3.0 - avformat.h
Add AVFormatContext.opaque, io_open and io_close, allowing custom IO
2016-02-01 - 1dba837 - lavf 57.24.100 - avformat.h, avio.h
Add protocol_whitelist to AVFormatContext, AVIOContext
2016-01-31 - 66e9d2f - lavu 55.17.100 - frame.h
Add AV_FRAME_DATA_GOP_TIMECODE for exporting MPEG1/2 GOP timecodes.
2016-01-01 - 5e8b053 / 2c68113 - lavc 57.21.100 / 57.12.0 - avcodec.h
Add AVCodecDescriptor.profiles and avcodec_profile_name().
2015-12-28 - 1f9139b - lavf 57.21.100 - avformat.h
Add automatic bitstream filtering; add av_apply_bitstream_filters()
2015-12-22 - 39a09e9 - lavfi 6.21.101 - avfilter.h
Deprecate avfilter_link_set_closed().
Applications are not supposed to mess with links,
they should close the sinks.
2015-12-17 - lavc 57.18.100 / 57.11.0 - avcodec.h dirac.h
xxxxxxx - Add av_packet_add_side_data().
xxxxxxx - Add AVCodecContext.coded_side_data.
xxxxxxx - Add AVCPBProperties API.
xxxxxxx - Add a new public header dirac.h containing
av_dirac_parse_sequence_header()
2015-12-11 - 676a93f - lavf 57.20.100 - avformat.h
Add av_program_add_stream_index()
2015-11-29 - 93fb4a4 - lavc 57.16.101 - avcodec.h
Deprecate rtp_callback without replacement, i.e. it won't be possible to
get image slices before the full frame is encoded any more. The libavformat
rtpenc muxer can still be used for RFC-2190 packetization.
2015-11-22 - fe20e34 - lavc 57.16.100 - avcodec.h
Add AV_PKT_DATA_FALLBACK_TRACK for making fallback associations between
streams.
2015-11-22 - ad317c9 - lavf 57.19.100 - avformat.h
Add av_stream_new_side_data().
2015-11-22 - e12f403 - lavu 55.8.100 - xtea.h
Add av_xtea_le_init and av_xtea_le_crypt
2015-11-18 - lavu 55.7.100 - mem.h
Add av_fast_mallocz()
2015-10-29 - lavc 57.12.100 / 57.8.0 - avcodec.h
xxxxxx - Deprecate av_free_packet(). Use av_packet_unref() as replacement,
it resets the packet in a more consistent way.
xxxxxx - Deprecate av_dup_packet(), it is a no-op for most cases.
Use av_packet_ref() to make a non-refcounted AVPacket refcounted.
xxxxxx - Add av_packet_alloc(), av_packet_clone(), av_packet_free().
They match the AVFrame functions with the same name.
2015-10-27 - 1e477a9 - lavu 55.5.100 - cpu.h
Add AV_CPU_FLAG_AESNI.
2015-10-22 - ee573b4 / a17a766 - lavc 57.9.100 / 57.5.0 - avcodec.h
Add data and linesize array to AVSubtitleRect, to be used instead of
the ones from the embedded AVPicture.
2015-10-22 - 866a417 / dc923bc - lavc 57.8.100 / 57.0.0 - qsv.h
Add an API for allocating opaque surfaces.
2015-10-15 - 2c2d162 - lavf 57.4.100
Remove the latm demuxer that was a duplicate of the loas demuxer.
2015-10-14 - b994788 / 11c5f43 - lavu 55.4.100 / 55.2.0 - dict.h
Change return type of av_dict_copy() from void to int, so that a proper
error code can be reported.
2015-09-29 - b01891a / 948f3c1 - lavc 57.3.100 / 57.2.0 - avcodec.h
Change type of AVPacket.duration from int to int64_t.
2015-09-17 - 7c46f24 / e3d4784 - lavc 57.3.100 / 57.2.0 - d3d11va.h
Add av_d3d11va_alloc_context(). This function must from now on be used for
allocating AVD3D11VAContext.
2015-09-15 - lavf 57.2.100 - avformat.h
probesize and max_analyze_duration switched to 64bit, both
are only accessible through AVOptions
2015-09-15 - lavf 57.1.100 - avformat.h
bit_rate was changed to 64bit, make sure you update any
printf() or other type sensitive code
2015-09-15 - lavc 57.2.100 - avcodec.h
bit_rate/rc_max_rate/rc_min_rate were changed to 64bit, make sure you update
any printf() or other type sensitive code
2015-09-07 - lavu 55.0.100 / 55.0.0
c734b34 / b8b5d82 - Change type of AVPixFmtDescriptor.flags from uint8_t to uint64_t.
f53569a / 6b3ef7f - Change type of AVComponentDescriptor fields from uint16_t to int
and drop bit packing.
151aa2e / 2268db2 - Add step, offset, and depth to AVComponentDescriptor to replace
the deprecated step_minus1, offset_plus1, and depth_minus1.
-------- 8< --------- FFmpeg 2.8 was cut here -------- 8< ---------
2015-08-27 - 1dd854e1 - lavc 56.58.100 - vaapi.h
Deprecate old VA-API context (vaapi_context) fields that were only
set and used by libavcodec. They are all managed internally now.
2015-08-19 - 9f8e57ef - lavu 54.31.100 - pixfmt.h
Add a unique pixel format for VA-API (AV_PIX_FMT_VAAPI) that
indicates the nature of the underlying storage: a VA surface. This
yields the same value as AV_PIX_FMT_VAAPI_VLD.
Deprecate old VA-API related pixel formats: AV_PIX_FMT_VAAPI_MOCO,
AV_PIX_FMT_VAAPI_IDCT, AV_PIX_FMT_VAAPI_VLD.
2015-08-02 - lavu 54.30.100 / 54.17.0
9ed59f1 / 7a7df34c - Add av_blowfish_alloc().
a130ec9 / ae365453 - Add av_rc4_alloc().
9ca1997 / 5d8bea3b - Add av_xtea_alloc().
3cf08e9 / d9e8b47e - Add av_des_alloc().
2015-07-27 - lavc 56.56.100 / 56.35.0 - avcodec.h
94d68a4 / 7c6eb0a1 - Rename CODEC_FLAG* defines to AV_CODEC_FLAG*.
444e987 / def97856 - Rename CODEC_CAP_* defines to AV_CODEC_CAP_*.
29d147c / 059a9348 - Rename FF_INPUT_BUFFER_PADDING_SIZE and FF_MIN_BUFFER_SIZE
to AV_INPUT_BUFFER_PADDING_SIZE and AV_INPUT_BUFFER_MIN_SIZE.
2015-07-22 - c40ecff - lavc 56.51.100 - avcodec.h
Add AV_PKT_DATA_QUALITY_STATS to export the quality value, PSNR, and pict_type
of an AVPacket.
2015-07-16 - 8dad213 - lavc 56.49.100
Add av_codec_get_codec_properties(), FF_CODEC_PROPERTY_LOSSLESS
and FF_CODEC_PROPERTY_CLOSED_CAPTIONS
2015-07-03 - d563e13 / 83212943 - lavu 54.28.100 / 56.15.0
Add av_version_info().
-------- 8< --------- FFmpeg 2.7 was cut here -------- 8< ---------
2015-06-04 - cc17b43 - lswr 1.2.100
Add swr_get_out_samples()
2015-05-27 - c312bfa - lavu 54.26.100 - cpu.h
Add AV_CPU_FLAG_AVXSLOW.
2015-05-26 - 1fb9b2a - lavu 54.25.100 - rational.h
Add av_q2intfloat().
2015-05-13 - cc48409 / e7c5e17 - lavc 56.39.100 / 56.23.0
Add av_vda_default_init2.
2015-05-11 - 541d75f - lavf 56.33.100 - avformat.h
Add AVOpenCallback AVFormatContext.open_cb
2015-05-07 - a7dd933 - 56.38.100 - avcodec.h
Add av_packet_side_data_name().
2015-05-07 - 01e59d4 - 56.37.102 - avcodec.h
Add FF_PROFILE_VP9_2 and FF_PROFILE_VP9_3.
2015-05-04 - 079b7f6 - 56.37.100 - avcodec.h
Add FF_PROFILE_VP9_0 and FF_PROFILE_VP9_1.
2015-04-22 - 748d481 - lavf 56.31.100 - avformat.h
Add AVFMT_FLAG_FAST_SEEK flag. Some formats (initially mp3) use it to enable
fast, but inaccurate seeking.
2015-04-20 - 8e8219e / c253340 - lavu 54.23.100 / 54.12.0 - log.h
Add AV_LOG_TRACE for extremely verbose debugging.
2015-04-02 - 26e0e393 - lavf 56.29.100 - avio.h
Add AVIODirEntryType.AVIO_ENTRY_SERVER.
Add AVIODirEntryType.AVIO_ENTRY_SHARE.
Add AVIODirEntryType.AVIO_ENTRY_WORKGROUP.
2015-03-31 - 3188696 - lavu 54.22.100 - avstring.h
Add av_append_path_component()
2015-03-27 - 184084c - lavf 56.27.100 - avio.h url.h
New directory listing API.
Add AVIODirEntryType enum.
Add AVIODirEntry, AVIODirContext structures.
Add avio_open_dir(), avio_read_dir(), avio_close_dir(), avio_free_directory_entry().
Add ff_alloc_dir_entry().
Extend URLProtocol with url_open_dir(), url_read_dir(), url_close_dir().
2015-03-29 - 268ff17 / c484561 - lavu 54.21.100 / 54.10.0 - pixfmt.h
Add AV_PIX_FMT_MMAL for MMAL hardware acceleration.
2015-03-19 - 11fe56c - 56.29.100 / lavc 56.22.0
Add FF_PROFILE_DTS_EXPRESS.
-------- 8< --------- FFmpeg 2.6 was cut here -------- 8< ---------
2015-03-04 - cca4476 - lavf 56.25.100
Add avformat_flush()
2015-03-03 - 81a9126 - lavf 56.24.100
Add avio_put_str16be()
2015-02-19 - 560eb71 / 31d2039 - lavc 56.23.100 / 56.13.0
Add width, height, coded_width, coded_height and format to
AVCodecParserContext.
2015-02-19 - e375511 / 5b1d9ce - lavu 54.19.100 / 54.9.0
Add AV_PIX_FMT_QSV for QSV hardware acceleration.
2015-02-14 - ba22295 - lavc 56.21.102
Deprecate VIMA decoder.
2015-01-27 - 62a82c6 / 728685f - lavc 56.21.100 / 56.12.0, lavu 54.18.100 / 54.8.0 - avcodec.h, frame.h
Add AV_PKT_DATA_AUDIO_SERVICE_TYPE and AV_FRAME_DATA_AUDIO_SERVICE_TYPE for
storing the audio service type as side data.
2015-01-16 - a47c933 - lavf 56.19.100 - avformat.h
Add data_codec and data_codec_id for storing codec of data stream
2015-01-11 - 007c33d - lavd 56.4.100 - avdevice.h
Add avdevice_list_input_sources().
Add avdevice_list_output_sinks().
2014-12-25 - d7aaeea / c220a60 - lavc 56.19.100 / 56.10.0 - vdpau.h
Add av_vdpau_get_surface_parameters().
2014-12-25 - ddb9a24 / 6c99c92 - lavc 56.18.100 / 56.9.0 - avcodec.h
Add AV_HWACCEL_FLAG_ALLOW_HIGH_DEPTH flag to av_vdpau_bind_context().
2014-12-25 - d16079a / 57b6704 - lavc 56.17.100 / 56.8.0 - avcodec.h
Add AVCodecContext.sw_pix_fmt.
2014-12-04 - 6e9ac02 - lavc 56.14.100 - dv_profile.h
Add av_dv_codec_profile2().
-------- 8< --------- FFmpeg 2.5 was cut here -------- 8< ---------
2014-11-21 - ab922f9 - lavu 54.15.100 - dict.h
Add av_dict_get_string().
2014-11-18 - a54a51c - lavu 54.14.100 - float_dsp.h
Add avpriv_float_dsp_alloc().
2014-11-16 - 6690d4c3 - lavf 56.13.100 - avformat.h
Add AVStream.recommended_encoder_configuration with accessors.
2014-11-16 - bee5844d - lavu 54.13.100 - opt.h
Add av_opt_serialize().
2014-11-16 - eec69332 - lavu 54.12.100 - opt.h
Add av_opt_is_set_to_default().
2014-11-06 - 44fa267 / 5e80fb7 - lavc 56.11.100 / 56.6.0 - vorbis_parser.h
Add a public API for parsing vorbis packets.
2014-10-15 - 17085a0 / 7ea1b34 - lavc 56.7.100 / 56.5.0 - avcodec.h
Replace AVCodecContext.time_base used for decoding
with AVCodecContext.framerate.
2014-10-15 - 51c810e / d565fef1 - lavc 56.6.100 / 56.4.0 - avcodec.h
Add AV_HWACCEL_FLAG_IGNORE_LEVEL flag to av_vdpau_bind_context().
2014-10-13 - da21895 / 2df0c32e - lavc 56.5.100 / 56.3.0 - avcodec.h
Add AVCodecContext.initial_padding. Deprecate the use of AVCodecContext.delay
for audio encoding.
2014-10-08 - bb44f7d / 5a419b2 - lavu 54.10.100 / 54.4.0 - pixdesc.h
Add API to return the name of frame and context color properties.
2014-10-06 - a61899a / e3e158e - lavc 56.3.100 / 56.2.0 - vdpau.h
Add av_vdpau_bind_context(). This function should now be used for creating
(or resetting) a AVVDPAUContext instead of av_vdpau_alloc_context().
2014-10-02 - cdd6f05 - lavc 56.2.100 - avcodec.h
2014-10-02 - cdd6f05 - lavu 54.9.100 - frame.h
Add AV_FRAME_DATA_SKIP_SAMPLES. Add lavc CODEC_FLAG2_SKIP_MANUAL and
AVOption "skip_manual", which makes lavc export skip information via
AV_FRAME_DATA_SKIP_SAMPLES AVFrame side data, instead of skipping and
discarding samples automatically.
2014-10-02 - 0d92b0d - lavu 54.8.100 - avstring.h
Add av_match_list()
2014-09-24 - ac68295 - libpostproc 53.1.100
Add visualization support
2014-09-19 - 6edd6a4 - lavc 56.1.101 - dv_profile.h
deprecate avpriv_dv_frame_profile2(), which was made public by accident.
-------- 8< --------- FFmpeg 2.4 was cut here -------- 8< ---------
2014-08-25 - 215db29 / b263f8f - lavf 56.3.100 / 56.3.0 - avformat.h
Add AVFormatContext.max_ts_probe.
2014-08-28 - f30a815 / 9301486 - lavc 56.1.100 / 56.1.0 - avcodec.h
Add AV_PKT_DATA_STEREO3D to export container-level stereo3d information.
2014-08-23 - 8fc9bd0 - lavu 54.7.100 - dict.h
AV_DICT_DONT_STRDUP_KEY and AV_DICT_DONT_STRDUP_VAL arguments are now
freed even on error. This is consistent with the behaviour all users
of it we could find expect.
2014-08-21 - 980a5b0 - lavu 54.6.100 - frame.h motion_vector.h
Add AV_FRAME_DATA_MOTION_VECTORS side data and AVMotionVector structure
2014-08-16 - b7d5e01 - lswr 1.1.100 - swresample.h
Add AVFrame based API
2014-08-16 - c2829dc - lavu 54.4.100 - dict.h
Add av_dict_set_int helper function.
2014-08-13 - c8571c6 / 8ddc326 - lavu 54.3.100 / 54.3.0 - mem.h
Add av_strndup().
2014-08-13 - 2ba4577 / a8c104a - lavu 54.2.100 / 54.2.0 - opt.h
Add av_opt_get_dict_val/set_dict_val with AV_OPT_TYPE_DICT to support
dictionary types being set as options.
2014-08-13 - afbd4b8 - lavf 56.01.0 - avformat.h
Add AVFormatContext.event_flags and AVStream.event_flags for signaling to
the user when events happen in the file/stream.
2014-08-10 - 78eaaa8 / fb1ddcd - lavr 2.1.0 - avresample.h
Add avresample_convert_frame() and avresample_config().
2014-08-10 - 78eaaa8 / fb1ddcd - lavu 54.1.100 / 54.1.0 - error.h
Add AVERROR_INPUT_CHANGED and AVERROR_OUTPUT_CHANGED.
2014-08-08 - 3841f2a / d35b94f - lavc 55.73.102 / 55.57.4 - avcodec.h
Deprecate FF_IDCT_XVIDMMX define and xvidmmx idct option.
Replaced by FF_IDCT_XVID and xvid respectively.
2014-08-08 - 5c3c671 - lavf 55.53.100 - avio.h
Add avio_feof() and deprecate url_feof().
2014-08-07 - bb78903 - lsws 2.1.3 - swscale.h
sws_getContext is not going to be removed in the future.
2014-08-07 - a561662 / ad1ee5f - lavc 55.73.101 / 55.57.3 - avcodec.h
reordered_opaque is not going to be removed in the future.
2014-08-02 - 28a2107 - lavu 52.98.100 - pixelutils.h
Add pixelutils API with SAD functions
2014-08-04 - 6017c98 / e9abafc - lavu 52.97.100 / 53.22.0 - pixfmt.h
Add AV_PIX_FMT_YA16 pixel format for 16 bit packed gray with alpha.
2014-08-04 - 4c8bc6f / e96c3b8 - lavu 52.96.101 / 53.21.1 - avstring.h
Rename AV_PIX_FMT_Y400A to AV_PIX_FMT_YA8 to better identify the format.
An alias pixel format and color space name are provided for compatibility.
2014-08-04 - 073c074 / d2962e9 - lavu 52.96.100 / 53.21.0 - pixdesc.h
Support name aliases for pixel formats.
2014-08-03 - 71d008e / 1ef9e83 - lavc 55.72.101 / 55.57.2 - avcodec.h
2014-08-03 - 71d008e / 1ef9e83 - lavu 52.95.100 / 53.20.0 - frame.h
Deprecate AVCodecContext.dtg_active_format and use side-data instead.
2014-08-03 - e680c73 - lavc 55.72.100 - avcodec.h
Add get_pixels() to AVDCT
2014-08-03 - 9400603 / 9f17685 - lavc 55.71.101 / 55.57.1 - avcodec.h
Deprecate unused FF_IDCT_IPP define and ipp avcodec option.
Deprecate unused FF_DEBUG_PTS define and pts avcodec option.
Deprecate unused FF_CODER_TYPE_DEFLATE define and deflate avcodec option.
Deprecate unused FF_DCT_INT define and int avcodec option.
Deprecate unused avcodec option scenechange_factor.
2014-07-30 - ba3e331 - lavu 52.94.100 - frame.h
Add av_frame_side_data_name()
2014-07-29 - 80a3a66 / 3a19405 - lavf 56.01.100 / 56.01.0 - avformat.h
Add mime_type field to AVProbeData, which now MUST be initialized in
order to avoid uninitialized reads of the mime_type pointer, likely
leading to crashes.
Typically, this means you will do 'AVProbeData pd = { 0 };' instead of
'AVProbeData pd;'.
2014-07-29 - 31e0b5d / 69e7336 - lavu 52.92.100 / 53.19.0 - avstring.h
Make name matching function from lavf public as av_match_name().
2014-07-28 - 2e5c8b0 / c5fca01 - lavc 55.71.100 / 55.57.0 - avcodec.h
Add AV_CODEC_PROP_REORDER to mark codecs supporting frame reordering.
2014-07-27 - ff9a154 - lavf 55.50.100 - avformat.h
New field int64_t probesize2 instead of deprecated
field int probesize.
2014-07-27 - 932ff70 - lavc 55.70.100 - avdct.h
Add AVDCT / avcodec_dct_alloc() / avcodec_dct_init().
2014-07-23 - 8a4c086 - lavf 55.49.100 - avio.h
Add avio_read_to_bprint()
-------- 8< --------- FFmpeg 2.3 was cut here -------- 8< ---------
2014-07-14 - 62227a7 - lavf 55.47.100 - avformat.h
Add av_stream_get_parser()
@@ -614,11 +35,7 @@ API changes, most recent first:
is now setting AVStream.time_base, instead of AVStream.codec.time_base as was
done previously. The old method is now deprecated.
2014-06-11 - 67d29da - lavc 55.66.101 - avcodec.h
Increase FF_INPUT_BUFFER_PADDING_SIZE to 32 due to some corner cases needing
it
2014-06-10 - 5482780 - lavf 55.43.100 - avformat.h
2014-06-10 - xxxxxxx - lavf 55.43.100 - avformat.h
New field int64_t max_analyze_duration2 instead of deprecated
int max_analyze_duration.
@@ -638,28 +55,28 @@ API changes, most recent first:
2014-05-29 - bdb2e80 / b2d4565 - lavr 1.3.0 - avresample.h
Add avresample_max_output_samples
2014-05-28 - d858ee7 / 6d21259 - lavf 55.42.100 / 55.19.0 - avformat.h
2014-05-24 - d858ee7 / 6d21259 - lavf 55.42.100 / 55.19.0 - avformat.h
Add strict_std_compliance and related AVOptions to support experimental
muxing.
2014-05-26 - 55cc60c - lavu 52.87.100 - threadmessage.h
2014-05-26 - xxxxxxx - lavu 52.87.100 - threadmessage.h
Add thread message queue API.
2014-05-26 - c37d179 - lavf 55.41.100 - avformat.h
Add format_probesize to AVFormatContext.
2014-05-20 - 7d25af1 / c23c96b - lavf 55.39.100 / 55.18.0 - avformat.h
2014-05-19 - 7d25af1 / c23c96b - lavf 55.39.100 / 55.18.0 - avformat.h
Add av_stream_get_side_data() to access stream-level side data
in the same way as av_packet_get_side_data().
2014-05-20 - 7336e39 - lavu 52.86.100 - fifo.h
2014-05-xx - xxxxxxx - lavu 52.86.100 - fifo.h
Add av_fifo_alloc_array() function.
2014-05-19 - ef1d4ee / bddd8cb - lavu 52.85.100 / 53.15.0 - frame.h, display.h
Add AV_FRAME_DATA_DISPLAYMATRIX for exporting frame-level
spatial rendering on video frames for proper display.
2014-05-19 - ef1d4ee / bddd8cb - lavc 55.64.100 / 55.53.0 - avcodec.h
2014-05-xx - xxxxxxx - lavc 55.64.100 / 55.53.0 - avcodec.h
Add AV_PKT_DATA_DISPLAYMATRIX for exporting packet-level
spatial rendering on video frames for proper display.
@@ -671,11 +88,11 @@ API changes, most recent first:
Add avcodec_free_context(). From now on it should be used for freeing
AVCodecContext.
2014-05-17 - 0eec06e / 1bd0bdc - lavu 52.84.100 / 54.5.0 - time.h
2014-05-17 - 0eec06e - lavu 52.84.100 - time.h
Add av_gettime_relative() av_gettime_relative_is_monotonic()
2014-05-15 - eacf7d6 / 0c1959b - lavf 55.38.100 / 55.17.0 - avformat.h
Add AVFMT_FLAG_BITEXACT flag. Muxers now use it instead of checking
Add AVMFT_FLAG_BITEXACT flag. Muxers now use it instead of checking
CODEC_FLAG_BITEXACT on the first stream.
2014-05-15 - 96cb4c8 - lswr 0.19.100 - swresample.h
@@ -684,10 +101,10 @@ API changes, most recent first:
2014-05-11 - 14aef38 / 66e6c8a - lavu 52.83.100 / 53.14.0 - pixfmt.h
Add AV_PIX_FMT_VDA for new-style VDA acceleration.
2014-05-07 - 351f611 - lavu 52.82.100 - fifo.h
2014-05-xx - xxxxxxx - lavu 52.82.0 - fifo.h
Add av_fifo_freep() function.
2014-05-02 - ba52fb11 - lavu 52.81.100 - opt.h
2014-05-02 - ba52fb11 - lavu 52.81.0 - opt.h
Add av_opt_set_dict2() function.
2014-05-01 - e77b985 / a2941c8 - lavc 55.60.103 / 55.50.3 - avcodec.h
@@ -706,14 +123,10 @@ API changes, most recent first:
Deprecate CODEC_FLAG_INPUT_PRESERVED. Its functionality is replaced by passing
reference-counted frames to encoders.
2014-04-30 - 617e866 - lavu 52.81.100 - pixdesc.h
Add av_find_best_pix_fmt_of_2(), av_get_pix_fmt_loss()
Deprecate avcodec_get_pix_fmt_loss(), avcodec_find_best_pix_fmt_of_2()
2014-04-29 - 1bf6396 - lavc 55.60.100 - avcodec.h
Add AVCodecDescriptor.mime_types field.
2014-04-29 - b804eb4 - lavu 52.80.100 - hash.h
2014-04-29 - xxxxxxx - lavu 52.80.0 - hash.h
Add av_hash_final_bin(), av_hash_final_hex() and av_hash_final_b64().
2014-03-07 - 8b2a130 - lavc 55.50.0 / 55.53.100 - dxva2.h
@@ -725,7 +138,7 @@ API changes, most recent first:
2014-04-17 - a8d01a7 / 0983d48 - lavu 53.12.0 / 52.77.100 - crc.h
Add AV_CRC_16_ANSI_LE crc variant.
2014-04-15 - ef818d8 - lavf 55.37.101 - avformat.h
2014-04-XX - xxxxxxx - lavf xx.xx.1xx - avformat.h
Add av_format_inject_global_side_data()
2014-04-12 - 4f698be - lavu 52.76.100 - log.h
@@ -771,9 +184,6 @@ API changes, most recent first:
Give the name AVPacketSideData to the previously anonymous struct used for
AVPacket.side_data.
-------- 8< --------- FFmpeg 2.2 was cut here -------- 8< ---------
2014-03-18 - 37c07d4 - lsws 2.5.102
Make gray16 full-scale.
@@ -805,7 +215,7 @@ API changes, most recent first:
2014-02-19 - f4c8d00 / 6bb8720 - lavu 52.64.101 / 53.3.1 - opt.h
Deprecate unused AV_OPT_FLAG_METADATA.
2014-02-16 - 81c3f81 - lavd 55.10.100 - avdevice.h
2014-02-xx - xxxxxxx - lavd 55.10.100 - avdevice.h
Add avdevice_list_devices() and avdevice_free_list_devices()
2014-02-16 - db3c970 - lavf 55.33.100 - avio.h
@@ -846,7 +256,7 @@ API changes, most recent first:
2014-01-19 - 1a193c4 - lavf 55.25.100 - avformat.h
Add avformat_get_mov_video_tags() and avformat_get_mov_audio_tags().
2014-01-19 - 3532dd5 - lavu 52.63.100 - rational.h
2014-01-19 - xxxxxxx - lavu 52.63.100 - rational.h
Add av_make_q() function.
2014-01-05 - 4cf4da9 / 5b4797a - lavu 52.62.100 / 53.2.0 - frame.h
@@ -916,9 +326,6 @@ API changes, most recent first:
2013-10-31 - 78265fc / 28096e0 - lavu 52.49.100 / 52.17.0 - frame.h
Add AVFrame.flags and AV_FRAME_FLAG_CORRUPT.
-------- 8< --------- FFmpeg 2.1 was cut here -------- 8< ---------
2013-10-27 - dbe6f9f - lavc 55.39.100 - avcodec.h
Add CODEC_CAP_DELAY support to avcodec_decode_subtitle2.
@@ -991,9 +398,6 @@ API changes, most recent first:
Add avcodec_chroma_pos_to_enum()
Add avcodec_enum_to_chroma_pos()
-------- 8< --------- FFmpeg 2.0 was cut here -------- 8< ---------
2013-07-03 - 838bd73 - lavfi 3.78.100 - avfilter.h
Deprecate avfilter_graph_parse() in favor of the equivalent
avfilter_graph_parse_ptr().
@@ -1011,9 +415,6 @@ API changes, most recent first:
av_ripemd_update()
av_ripemd_final()
2013-06-10 - 82ef670 - lavu 52.35.101 - hmac.h
Add AV_HMAC_SHA224, AV_HMAC_SHA256, AV_HMAC_SHA384, AV_HMAC_SHA512
2013-06-04 - 30b491f / fc962d4 - lavu 52.35.100 / 52.13.0 - mem.h
Add av_realloc_array and av_reallocp_array
@@ -1069,9 +470,6 @@ API changes, most recent first:
2013-03-17 - 7aa9af5 - lavu 52.20.100 - opt.h
Add AV_OPT_TYPE_VIDEO_RATE value to AVOptionType enum.
-------- 8< --------- FFmpeg 1.2 was cut here -------- 8< ---------
2013-03-07 - 9767ec6 - lavu 52.18.100 - avstring.h,bprint.h
Add av_escape() and av_bprint_escape() API.
@@ -1084,9 +482,6 @@ API changes, most recent first:
2013-01-01 - 2eb2e17 - lavfi 3.34.100
Add avfilter_get_audio_buffer_ref_from_arrays_channels.
-------- 8< --------- FFmpeg 1.1 was cut here -------- 8< ---------
2012-12-20 - 34de47aa - lavfi 3.29.100 - avfilter.h
Add AVFilterLink.channels, avfilter_link_get_channels()
and avfilter_ref_get_channels().
@@ -1132,9 +527,6 @@ API changes, most recent first:
Add LIBSWRESAMPLE_VERSION, LIBSWRESAMPLE_BUILD
and LIBSWRESAMPLE_IDENT symbols.
-------- 8< --------- FFmpeg 1.0 was cut here -------- 8< ---------
2012-09-06 - 29e972f - lavu 51.72.100 - parseutils.h
Add av_small_strptime() time parsing function.
@@ -1266,7 +658,7 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
* base -- is now stored in AVBufferRef
* reference, type, buffer_hints -- are unnecessary in the new API
* hwaccel_picture_private, owner, thread_opaque -- should not
have been accessed from outside of lavc
have been acessed from outside of lavc
* qscale_table, qstride, qscale_type, mbskip_table, motion_val,
mb_type, dct_coeff, ref_index -- mpegvideo-specific tables,
which are not exported anymore.
@@ -1303,14 +695,15 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
Add avresample_set_channel_mapping() for input channel reordering,
duplication, and silencing.
2012-12-29 - lavu 52.13.100 / 52.3.0 - avstring.h
2ce43b3 / d8fd06c - Add av_basename() and av_dirname().
e13d5e9 / c1a02e8 - Add av_pix_fmt_get_chroma_sub_sample and deprecate
avcodec_get_chroma_sub_sample.
2012-12-29 - 2ce43b3 / d8fd06c - lavu 52.13.100 / 52.3.0 - avstring.h
Add av_basename() and av_dirname().
2012-11-11 - 03b0787 / 5980f5d - lavu 52.6.100 / 52.2.0 - audioconvert.h
Rename audioconvert.h to channel_layout.h. audioconvert.h is now deprecated.
2012-11-05 - 7d26be6 / dfde8a3 - lavu 52.5.100 / 52.1.0 - intmath.h
Add av_ctz() for trailing zero bit count
2012-10-21 - e3a91c5 / a893655 - lavu 51.77.100 / 51.45.0 - error.h
Add AVERROR_EXPERIMENTAL
@@ -1338,9 +731,6 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
avresample_read() are now uint8_t** instead of void**.
Libavresample is now stable.
2012-09-26 - 3ba0dab7 / 1384df64 - lavf 54.29.101 / 56.06.3 - avformat.h
Add AVFormatContext.avoid_negative_ts.
2012-09-24 - 46a3595 / a42aada - lavc 54.59.100 / 54.28.0 - avcodec.h
Add avcodec_free_frame(). This function must now
be used for freeing an AVFrame.
@@ -1555,9 +945,6 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
2012-01-12 - b18e17e / 3167dc9 - lavfi 2.59.100 / 2.15.0
Add a new installed header -- libavfilter/version.h -- with version macros.
-------- 8< --------- FFmpeg 0.9 was cut here -------- 8< ---------
2011-12-08 - a502939 - lavfi 2.52.0
Add av_buffersink_poll_frame() to buffersink.h.
@@ -1767,13 +1154,6 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
2011-06-28 - 5129336 - lavu 51.11.0 - avutil.h
Define the AV_PICTURE_TYPE_NONE value in AVPictureType enum.
-------- 8< --------- FFmpeg 0.7 was cut here -------- 8< ---------
-------- 8< --------- FFmpeg 0.8 was cut here -------- 8< ---------
2011-06-19 - fd2c0a5 - lavfi 2.23.0 - avfilter.h
Add layout negotiation fields and helper functions.
@@ -2451,9 +1831,6 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
2010-06-02 - 7e566bb - lavc 52.73.0 - av_get_codec_tag_string()
Add av_get_codec_tag_string().
-------- 8< --------- FFmpeg 0.6 was cut here -------- 8< ---------
2010-06-01 - 2b99142 - lsws 0.11.0 - convertPalette API
Add sws_convertPalette8ToPacked32() and sws_convertPalette8ToPacked24().
@@ -2471,6 +1848,10 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
2010-05-09 - b6bc205 - lavfi 1.20.0 - AVFilterPicRef
Add interlaced and top_field_first fields to AVFilterPicRef.
------------------------------8<-------------------------------------
0.6 branch was cut here
----------------------------->8--------------------------------------
2010-05-01 - 8e2ee18 - lavf 52.62.0 - probe function
Add av_probe_input_format2 to API, it allows ignoring probe
results below given score and returns the actual probe score.

File diff suppressed because it is too large Load Diff

View File

@@ -36,18 +36,14 @@ DOCS-$(CONFIG_MANPAGES) += $(MANPAGES)
DOCS-$(CONFIG_TXTPAGES) += $(TXTPAGES)
DOCS = $(DOCS-yes)
DOC_EXAMPLES-$(CONFIG_AVIO_DIR_CMD_EXAMPLE) += avio_dir_cmd
DOC_EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE) += avio_reading
DOC_EXAMPLES-$(CONFIG_DECODING_ENCODING_EXAMPLE) += decoding_encoding
DOC_EXAMPLES-$(CONFIG_AVCODEC_EXAMPLE) += avcodec
DOC_EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding
DOC_EXAMPLES-$(CONFIG_EXTRACT_MVS_EXAMPLE) += extract_mvs
DOC_EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE) += filter_audio
DOC_EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE) += filtering_audio
DOC_EXAMPLES-$(CONFIG_FILTERING_VIDEO_EXAMPLE) += filtering_video
DOC_EXAMPLES-$(CONFIG_HTTP_MULTICLIENT_EXAMPLE) += http_multiclient
DOC_EXAMPLES-$(CONFIG_METADATA_EXAMPLE) += metadata
DOC_EXAMPLES-$(CONFIG_MUXING_EXAMPLE) += muxing
DOC_EXAMPLES-$(CONFIG_QSVDEC_EXAMPLE) += qsvdec
DOC_EXAMPLES-$(CONFIG_REMUXING_EXAMPLE) += remuxing
DOC_EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE) += resampling_audio
DOC_EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE) += scaling_video
@@ -84,25 +80,14 @@ $(GENTEXI): doc/avoptions_%.texi: doc/print_options$(HOSTEXESUF)
$(M)doc/print_options $* > $@
doc/%.html: TAG = HTML
doc/%-all.html: TAG = HTML
ifdef HAVE_MAKEINFO_HTML
doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.pm $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)makeinfo --html -I doc --no-split -D config-not-all --init-file=$(SRC_PATH)/doc/t2h.pm --output $@ $<
doc/%-all.html: doc/%.texi $(SRC_PATH)/doc/t2h.pm $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)makeinfo --html -I doc --no-split -D config-all --init-file=$(SRC_PATH)/doc/t2h.pm --output $@ $<
else
doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)texi2html -I doc -monolithic --D=config-not-all --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
doc/%-all.html: TAG = HTML
doc/%-all.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)texi2html -I doc -monolithic --D=config-all --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
endif
doc/%.pod: TAG = POD
doc/%.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI)
@@ -116,20 +101,19 @@ doc/%-all.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI)
doc/%.1 doc/%.3: TAG = MAN
doc/%.1: doc/%.pod $(GENTEXI)
$(M)pod2man --section=1 --center=" " --release=" " --date=" " $< > $@
$(M)pod2man --section=1 --center=" " --release=" " $< > $@
doc/%.3: doc/%.pod $(GENTEXI)
$(M)pod2man --section=3 --center=" " --release=" " --date=" " $< > $@
$(M)pod2man --section=3 --center=" " --release=" " $< > $@
$(DOCS) doc/doxy/html: | doc/
$(DOC_EXAMPLES:%$(EXESUF)=%.o): | doc/examples
OBJDIRS += doc/examples
DOXY_INPUT = $(INSTHEADERS) $(DOC_EXAMPLES:%$(EXESUF)=%.c) $(LIB_EXAMPLES:%$(EXESUF)=%.c)
DOXY_INPUT_DEPS = $(addprefix $(SRC_PATH)/, $(DOXY_INPUT)) config.mak
DOXY_INPUT = $(addprefix $(SRC_PATH)/, $(INSTHEADERS) $(DOC_EXAMPLES:%$(EXESUF)=%.c) $(LIB_EXAMPLES:%$(EXESUF)=%.c))
doc/doxy/html: TAG = DOXY
doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(SRC_PATH)/doc/doxy-wrapper.sh $(DOXY_INPUT_DEPS)
$(M)OUT_DIR=$$PWD/doc/doxy; cd $(SRC_PATH); ./doc/doxy-wrapper.sh $$OUT_DIR $< $(DOXYGEN) $(DOXY_INPUT);
doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(SRC_PATH)/doc/doxy-wrapper.sh $(DOXY_INPUT)
$(M)$(SRC_PATH)/doc/doxy-wrapper.sh $(SRC_PATH) $< $(DOXYGEN) $(DOXY_INPUT)
install-doc: install-html install-man

View File

@@ -13,16 +13,7 @@ bitstream filter using the option @code{--disable-bsf=BSF}.
The option @code{-bsfs} of the ff* tools will display the list of
all the supported bitstream filters included in your build.
The ff* tools have a -bsf option applied per stream, taking a
comma-separated list of filters, whose parameters follow the filter
name after a '='.
@example
ffmpeg -i INPUT -c:v copy -bsf:v filter1[=opt1=str1:opt2=str2][,filter2] OUTPUT
@end example
Below is a description of the currently available bitstream filters,
with their parameters, if any.
Below is a description of the currently available bitstream filters.
@section aac_adtstoasc
@@ -67,10 +58,6 @@ the header stored in extradata to the key packets:
ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts
@end example
@section dca_core
Extract DCA core from DTS-HD streams.
@section h264_mp4toannexb
Convert an H.264 bitstream from length prefixed mode to start code
@@ -143,41 +130,14 @@ ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
@section mp3_header_decompress
@section mpeg4_unpack_bframes
Unpack DivX-style packed B-frames.
DivX-style packed B-frames are not valid MPEG-4 and were only a
workaround for the broken Video for Windows subsystem.
They use more space, can cause minor AV sync issues, require more
CPU power to decode (unless the player has some decoded picture queue
to compensate the 2,0,2,0 frame per packet style) and cause
trouble if copied into a standard container like mp4 or mpeg-ps/ts,
because MPEG-4 decoders may not be able to decode them, since they are
not valid MPEG-4.
For example to fix an AVI file containing an MPEG-4 stream with
DivX-style packed B-frames using @command{ffmpeg}, you can use the command:
@example
ffmpeg -i INPUT.avi -codec copy -bsf:v mpeg4_unpack_bframes OUTPUT.avi
@end example
@section noise
Damages the contents of packets without damaging the container. Can be
used for fuzzing or testing error resilience/concealment.
Parameters:
A numeral string, whose value is related to how often output bytes will
be modified. Therefore, values below or equal to 0 are forbidden, and
the lower the more frequent bytes will be modified, with 1 meaning
every byte is modified.
@example
ffmpeg -i INPUT -c copy -bsf noise[=1] output.mkv
ffmpeg -i INPUT -c copy -bsf noise output.mkv
@end example
applies the modification to every byte.
@section remove_extra

File diff suppressed because one or more lines are too long

View File

@@ -7,28 +7,20 @@ V
Disable the default terse mode, the full command issued by make and its
output will be shown on the screen.
DBG
Preprocess x86 external assembler files to a .dbg.asm file in the object
directory, which then gets compiled. Helps in developing those assembler
files.
DESTDIR
Destination directory for the install targets, useful to prepare packages
or install FFmpeg in cross-environments.
GEN
Set to 1 to generate the missing or mismatched references.
Makefile targets:
all
Default target, builds all the libraries and the executables.
fate
Run the fate test suite, note that you must have installed it.
Run the fate test suite, note you must have installed it
fate-list
List all fate/regression test targets.
Will list all fate/regression test targets
install
Install headers, libraries and programs.
@@ -39,23 +31,26 @@ examples
libavformat/output-example
Build the libavformat basic example.
libavcodec/api-example
Build the libavcodec basic example.
libswscale/swscale-test
Build the swscale self-test (useful also as an example).
Build the swscale self-test (useful also as example).
config
Reconfigure the project with the current configuration.
Reconfigure the project with current configuration.
Useful standard make commands:
make -t <target>
Touch all files that otherwise would be built, this is useful to reduce
unneeded rebuilding when changing headers, but note that you must force rebuilds
Touch all files that otherwise would be build, this is useful to reduce
unneeded rebuilding when changing headers, but note you must force rebuilds
of files that actually need it by hand then.
make -j<num>
Rebuild with multiple jobs at the same time. Faster on multi processor systems.
rebuild with multiple jobs at the same time. Faster on multi processor systems
make -k
Continue build in case of errors, this is useful for the regression tests
sometimes but note that it will still not run all reg tests.
continue build in case of errors, this is useful for the regression tests
sometimes but note it will still not run all reg tests.

View File

@@ -7,7 +7,7 @@ all the encoders and decoders. In addition each codec may support
so-called private options, which are specific for a given codec.
Sometimes, a global option may only affect a specific kind of codec,
and may be nonsensical or ignored by another, so you need to be aware
and may be unsensical or ignored by another, so you need to be aware
of the meaning of the specified options. Also some options are
meant only for decoding or encoding.
@@ -71,9 +71,7 @@ Force low delay.
@item global_header
Place global headers in extradata instead of every keyframe.
@item bitexact
Only write platform-, build- and time-independent data. (except (I)DCT).
This ensures that file and data checksums are reproducible and match between
platforms. Its primary use is for regression testing.
Use only bitexact stuff (except (I)DCT).
@item aic
Apply H263 advanced intra coding / mpeg4 ac prediction.
@item cbp
@@ -129,7 +127,7 @@ should be @code{1 / frame_rate} and timestamp increments should be
identically 1.
@item g @var{integer} (@emph{encoding,video})
Set the group of picture (GOP) size. Default value is 12.
Set the group of picture size. Default value is 12.
@item ar @var{integer} (@emph{decoding/encoding,audio})
Set audio sampling rate (in Hz).
@@ -257,7 +255,7 @@ Specify how strictly to follow the standards.
Possible values:
@table @samp
@item very
strictly conform to an older more strict version of the spec or reference software
strictly conform to a older more strict version of the spec or reference software
@item strict
strictly conform to all the things in the spec no matter what consequences
@item normal
@@ -456,9 +454,6 @@ Possible values:
@item aspect @var{rational number} (@emph{encoding,video})
Set sample aspect ratio.
@item sar @var{rational number} (@emph{encoding,video})
Set sample aspect ratio. Alias to @var{aspect}.
@item debug @var{flags} (@emph{decoding/encoding,audio,video,subtitles})
Print specific debug info.
@@ -478,9 +473,6 @@ per-block quantization parameter (QP)
motion vector
@item dct_coeff
@item green_metadata
display complexity metadata for the upcoming frame, GoP or for a given duration.
@item skip
@item startcode
@@ -501,15 +493,11 @@ visualize block types
picture buffer allocations
@item thread_ops
threading operations
@item nomc
skip motion compensation
@end table
@item vismv @var{integer} (@emph{decoding,video})
Visualize motion vectors (MVs).
This option is deprecated, see the codecview filter instead.
Possible values:
@table @samp
@item pf
@@ -809,9 +797,6 @@ Frame data might be split into multiple chunks.
Show all frames before the first keyframe.
@item skiprd
Deprecated, use mpegvideo private options instead.
@item export_mvs
Export motion vectors into frame side-data (see @code{AV_FRAME_DATA_MOTION_VECTORS})
for codecs that support it. See also @file{doc/examples/export_mvs.c}.
@end table
@item error @var{integer} (@emph{encoding,video})
@@ -820,17 +805,13 @@ for codecs that support it. See also @file{doc/examples/export_mvs.c}.
Deprecated, use mpegvideo private options instead.
@item threads @var{integer} (@emph{decoding/encoding,video})
Set the number of threads to be used, in case the selected codec
implementation supports multi-threading.
Possible values:
@table @samp
@item auto, 0
automatically select the number of threads to set
@item auto
detect a good number of threads
@end table
Default value is @samp{auto}.
@item me_threshold @var{integer} (@emph{encoding,video})
Set motion estimation threshold.
@@ -875,14 +856,6 @@ Possible values:
@item mpeg2_aac_he
@item mpeg4_sp
@item mpeg4_core
@item mpeg4_main
@item mpeg4_asp
@item dts
@item dts_es
@@ -1049,98 +1022,10 @@ Possible values:
@item rc_max_vbv_use @var{float} (@emph{encoding,video})
@item rc_min_vbv_use @var{float} (@emph{encoding,video})
@item ticks_per_frame @var{integer} (@emph{decoding/encoding,audio,video})
@item color_primaries @var{integer} (@emph{decoding/encoding,video})
Possible values:
@table @samp
@item bt709
BT.709
@item bt470m
BT.470 M
@item bt470bg
BT.470 BG
@item smpte170m
SMPTE 170 M
@item smpte240m
SMPTE 240 M
@item film
Film
@item bt2020
BT.2020
@item smpte428_1
SMPTE ST 428-1
@item smpte431
SMPTE 431-2
@item smpte432
SMPTE 432-1
@end table
@item color_trc @var{integer} (@emph{decoding/encoding,video})
Possible values:
@table @samp
@item bt709
BT.709
@item gamma22
BT.470 M
@item gamma28
BT.470 BG
@item smpte170m
SMPTE 170 M
@item smpte240m
SMPTE 240 M
@item linear
Linear
@item log
Log
@item log_sqrt
Log square root
@item iec61966_2_4
IEC 61966-2-4
@item bt1361
BT.1361
@item iec61966_2_1
IEC 61966-2-1
@item bt2020_10bit
BT.2020 - 10 bit
@item bt2020_12bit
BT.2020 - 12 bit
@item smpte2084
SMPTE ST 2084
@item smpte428_1
SMPTE ST 428-1
@item arib-std-b67
ARIB STD-B67
@end table
@item colorspace @var{integer} (@emph{decoding/encoding,video})
Possible values:
@table @samp
@item rgb
RGB
@item bt709
BT.709
@item fcc
FCC
@item bt470bg
BT.470 BG
@item smpte170m
SMPTE 170 M
@item smpte240m
SMPTE 240 M
@item ycocg
YCOCG
@item bt2020_ncl
BT.2020 NCL
@item bt2020_cl
BT.2020 CL
@item smpte2085
SMPTE 2085
@end table
@item color_range @var{integer} (@emph{decoding/encoding,video})
If used as input parameter, it serves as a hint to the decoder, which
color_range the input has.
@item chroma_sample_location @var{integer} (@emph{decoding/encoding,video})
@item log_level_offset @var{integer}
@@ -1224,19 +1109,6 @@ Interlaced video, bottom coded first, top displayed first
Set to 1 to disable processing alpha (transparency). This works like the
@samp{gray} flag in the @option{flags} option which skips chroma information
instead of alpha. Default is 0.
@item codec_whitelist @var{list} (@emph{input})
"," separated list of allowed decoders. By default all are allowed.
@item dump_separator @var{string} (@emph{input})
Separator used to separate the fields printed on the command line about the
Stream parameters.
For example to separate the fields with newlines and indention:
@example
ffprobe -dump_separator "
" -i ~/videos/matrixbench_mpeg2.mpg
@end example
@end table
@c man end CODEC OPTIONS

View File

@@ -25,13 +25,6 @@ enabled decoders.
A description of some of the currently available video decoders
follows.
@section hevc
HEVC / H.265 decoder.
Note: the @option{skip_loop_filter} option has effect only at level
@code{all}.
@section rawvideo
Raw video decoder.
@@ -90,23 +83,6 @@ Loud sounds are fully compressed. Soft sounds are enhanced.
@end table
@section flac
FLAC audio decoder.
This decoder aims to implement the complete FLAC specification from Xiph.
@subsection FLAC Decoder options
@table @option
@item -use_buggy_lpc
The lavc FLAC encoder used to produce buggy streams with high lpc values
(like the default value). This option makes it possible to decode such streams
correctly by using lavc's old buggy lpc logic for decoding.
@end table
@section ffwavesynth
Internal wave synthetizer.
@@ -187,33 +163,11 @@ Requires the presence of the libopus headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libopus}.
An FFmpeg native decoder for Opus exists, so users can decode Opus
without this library.
@c man end AUDIO DECODERS
@chapter Subtitles Decoders
@c man begin SUBTILES DECODERS
@section dvbsub
@subsection Options
@table @option
@item compute_clut
@table @option
@item -1
Compute clut if no matching CLUT is in the stream.
@item 0
Never compute CLUT
@item 1
Always compute CLUT and override the one provided in the stream.
@end table
@item dvb_substream
Selects the dvb substream, or all substreams if -1 which is default.
@end table
@section dvdsub
This codec decodes the bitmap subtitles used in DVDs; the same subtitles can
@@ -233,15 +187,6 @@ The format for this option is a string containing 16 24-bits hexadecimal
numbers (without 0x prefix) separated by comas, for example @code{0d00ee,
ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1,
7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b}.
@item ifo_palette
Specify the IFO file from which the global palette is obtained.
(experimental)
@item forced_subs_only
Only decode subtitle entries marked as forced. Some titles have forced
and non-forced subtitles in the same track. Setting this flag to @code{1}
will only keep the forced subtitles. Default value is @code{0}.
@end table
@section libzvbi-teletext
@@ -279,16 +224,10 @@ present between the subtitle lines because of double-sized teletext charactes.
Default value is 1.
@item txt_duration
Sets the display duration of the decoded teletext pages or subtitles in
milliseconds. Default value is 30000 which is 30 seconds.
miliseconds. Default value is 30000 which is 30 seconds.
@item txt_transparent
Force transparent background of the generated teletext bitmaps. Default value
is 0 which means an opaque background.
@item txt_opacity
Sets the opacity (0-255) of the teletext background. If
@option{txt_transparent} is not set, it only affects characters between a start
box and an end box, typically subtitles. Default value is 0 if
@option{txt_transparent} is set, 255 otherwise.
is 0 which means an opaque (black) background.
@end table
@c man end SUBTILES DECODERS

View File

@@ -18,12 +18,6 @@ enabled demuxers.
The description of some of the currently available demuxers follows.
@section aa
Audible Format 2, 3, and 4 demuxer.
This demuxer is used to demux Audible Format 2, 3, and 4 (.aa) files.
@section applehttp
Apple HTTP Live Streaming demuxer.
@@ -35,26 +29,6 @@ the caller can decide which variant streams to actually receive.
The total bitrate of the variant that the stream belongs to is
available in a metadata key named "variant_bitrate".
@section apng
Animated Portable Network Graphics demuxer.
This demuxer is used to demux APNG files.
All headers, but the PNG signature, up to (but not including) the first
fcTL chunk are transmitted as extradata.
Frames are then split as being all the chunks between two fcTL ones, or
between the last fcTL and IEND chunks.
@table @option
@item -ignore_loop @var{bool}
Ignore the loop variable in the file if set.
@item -max_fps @var{int}
Maximum framerate in frames per second (0 for no limit).
@item -default_fps @var{int}
Default framerate in frames per second when none is specified in the file
(0 meaning as fast as possible).
@end table
@section asf
Advanced Systems Format demuxer.
@@ -72,7 +46,7 @@ Do not try to resynchronize by looking for a certain optional start code.
Virtual concatenation script demuxer.
This demuxer reads a list of files and other directives from a text file and
demuxes them one after the other, as if all their packets had been muxed
demuxes them one after the other, as if all their packet had been muxed
together.
The timestamps in the files are adjusted so that the first file starts at 0
@@ -104,10 +78,10 @@ All subsequent file-related directives apply to that file.
@item @code{ffconcat version 1.0}
Identify the script type and version. It also sets the @option{safe} option
to 1 if it was -1.
to 1 if it was to its default -1.
To make FFmpeg recognize the format automatically, this directive must
appear exactly as is (no extra space or byte-order-mark) on the very first
appears exactly as is (no extra space or byte-order-mark) on the very first
line of the script.
@item @code{duration @var{dur}}
@@ -118,47 +92,6 @@ file is not available or accurate.
If the duration is set for all files, then it is possible to seek in the
whole concatenated video.
@item @code{inpoint @var{timestamp}}
In point of the file. When the demuxer opens the file it instantly seeks to the
specified timestamp. Seeking is done so that all streams can be presented
successfully at In point.
This directive works best with intra frame codecs, because for non-intra frame
ones you will usually get extra packets before the actual In point and the
decoded content will most likely contain frames before In point too.
For each file, packets before the file In point will have timestamps less than
the calculated start timestamp of the file (negative in case of the first
file), and the duration of the files (if not specified by the @code{duration}
directive) will be reduced based on their specified In point.
Because of potential packets before the specified In point, packet timestamps
may overlap between two concatenated files.
@item @code{outpoint @var{timestamp}}
Out point of the file. When the demuxer reaches the specified decoding
timestamp in any of the streams, it handles it as an end of file condition and
skips the current and all the remaining packets from all streams.
Out point is exclusive, which means that the demuxer will not output packets
with a decoding timestamp greater or equal to Out point.
This directive works best with intra frame codecs and formats where all streams
are tightly interleaved. For non-intra frame codecs you will usually get
additional packets with presentation timestamp after Out point therefore the
decoded content will most likely contain frames after Out point too. If your
streams are not tightly interleaved you may not get all the packets from all
streams before Out point and you may only will be able to decode the earliest
stream until Out point.
The duration of the files (if not specified by the @code{duration}
directive) will be reduced based on their specified Out point.
@item @code{file_packet_metadata @var{key=value}}
Metadata of the packets of the file. The specified metadata will be set for
each file packet. You can specify this directive multiple times to add multiple
metadata entries.
@item @code{stream}
Introduce a stream in the virtual file.
All subsequent stream-related directives apply to the last introduced
@@ -192,57 +125,19 @@ component.
If set to 0, any file name is accepted.
The default is 1.
-1 is equivalent to 1 if the format was automatically
The default is -1, it is equivalent to 1 if the format was automatically
probed and 0 otherwise.
@item auto_convert
If set to 1, try to perform automatic conversions on packet data to make the
streams concatenable.
The default is 1.
Currently, the only conversion is adding the h264_mp4toannexb bitstream
filter to H.264 streams in MP4 format. This is necessary in particular if
there are resolution changes.
@item segment_time_metadata
If set to 1, every packet will contain the @var{lavf.concat.start_time} and the
@var{lavf.concat.duration} packet metadata values which are the start_time and
the duration of the respective file segments in the concatenated output
expressed in microseconds. The duration metadata is only set if it is known
based on the concat file.
The default is 0.
@end table
@subsection Examples
@itemize
@item
Use absolute filenames and include some comments:
@example
# my first filename
file /mnt/share/file-1.wav
# my second filename including whitespace
file '/mnt/share/file 2.wav'
# my third filename including whitespace plus single quote
file '/mnt/share/file 3'\''.wav'
@end example
@item
Allow for input format auto-probing, use safe filenames and set the duration of
the first file:
@example
ffconcat version 1.0
file file-1.wav
duration 20.0
file subdir/file-2.wav
@end example
@end itemize
@section flv
Adobe Flash Video Format demuxer.
@@ -254,44 +149,30 @@ This demuxer is used to demux FLV files and RTMP network streams.
Allocate the streams according to the onMetaData array content.
@end table
@section gif
@section libgme
Animated GIF demuxer.
The Game Music Emu library is a collection of video game music file emulators.
It accepts the following options:
See @url{http://code.google.com/p/game-music-emu/} for more information.
@table @option
@item min_delay
Set the minimum valid delay between frames in hundredths of seconds.
Range is 0 to 6000. Default value is 2.
Some files have multiple tracks. The demuxer will pick the first track by
default. The @option{track_index} option can be used to select a different
track. Track indexes start at 0. The demuxer exports the number of tracks as
@var{tracks} meta data entry.
@item max_gif_delay
Set the maximum valid delay between frames in hundredth of seconds.
Range is 0 to 65535. Default value is 65535 (nearly eleven minutes),
the maximum value allowed by the specification.
For very large files, the @option{max_size} option may have to be adjusted.
@item default_delay
Set the default delay between frames in hundredths of seconds.
Range is 0 to 6000. Default value is 10.
@section libquvi
@item ignore_loop
GIF files can contain information to loop a certain number of times (or
infinitely). If @option{ignore_loop} is set to 1, then the loop setting
from the input will be ignored and looping will not occur. If set to 0,
then looping will occur and will cycle the number of times according to
the GIF. Default value is 1.
@end table
Play media from Internet services using the quvi project.
For example, with the overlay filter, place an infinitely looping GIF
over another video:
@example
ffmpeg -i input.mp4 -ignore_loop 0 -i input.gif -filter_complex overlay=shortest=1 out.mkv
@end example
The demuxer accepts a @option{format} option to request a specific quality. It
is by default set to @var{best}.
Note that in the above example the shortest option for overlay filter is
used to end the output video at the length of the shortest input file,
which in this case is @file{input.mp4} as the GIF in this example loops
infinitely.
See @url{http://quvi.sourceforge.net/} for more information.
FFmpeg needs to be built with @code{--enable-libquvi} for this demuxer to be
enabled.
@section image2
@@ -318,10 +199,6 @@ Select the pattern type used to interpret the provided filename.
@var{pattern_type} accepts one of the following values.
@table @option
@item none
Disable pattern matching, therefore the video will only contain the specified
image. You should use this option if you do not want to create sequences from
multiple images and your filenames may contain special pattern characters.
@item sequence
Select a sequence pattern type, used to specify a sequence of files
indexed by sequential numbers.
@@ -428,102 +305,17 @@ ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv
@end example
@end itemize
@section libgme
The Game Music Emu library is a collection of video game music file emulators.
See @url{http://code.google.com/p/game-music-emu/} for more information.
Some files have multiple tracks. The demuxer will pick the first track by
default. The @option{track_index} option can be used to select a different
track. Track indexes start at 0. The demuxer exports the number of tracks as
@var{tracks} meta data entry.
For very large files, the @option{max_size} option may have to be adjusted.
@section libopenmpt
libopenmpt based module demuxer
See @url{https://lib.openmpt.org/libopenmpt/} for more information.
Some files have multiple subsongs (tracks) this can be set with the @option{subsong}
option.
It accepts the following options:
@table @option
@item subsong
Set the subsong index. This can be either 'all', 'auto', or the index of the
subsong. Subsong indexes start at 0. The default is 'auto'.
The default value is to let libopenmpt choose.
@item layout
Set the channel layout. Valid values are 1, 2, and 4 channel layouts.
The default value is STEREO.
@item sample_rate
Set the sample rate for libopenmpt to output.
Range is from 1000 to INT_MAX. The value default is 48000.
@end table
@section mov/mp4/3gp/QuickTime
QuickTime / MP4 demuxer.
This demuxer accepts the following options:
@table @option
@item enable_drefs
Enable loading of external tracks, disabled by default.
Enabling this can theoretically leak information in some use cases.
@item use_absolute_path
Allows loading of external tracks via absolute paths, disabled by default.
Enabling this poses a security risk. It should only be enabled if the source
is known to be non malicious.
@end table
@section mpegts
MPEG-2 transport stream demuxer.
This demuxer accepts the following options:
@table @option
@item resync_size
Set size limit for looking up a new synchronization. Default value is
65536.
@item fix_teletext_pts
Override teletext packet PTS and DTS values with the timestamps calculated
Overrides teletext packet PTS and DTS values with the timestamps calculated
from the PCR of the first program which the teletext stream is part of and is
not discarded. Default value is 1, set this option to 0 if you want your
teletext packet PTS and DTS values untouched.
@item ts_packetsize
Output option carrying the raw packet size in bytes.
Show the detected raw packet size, cannot be set by the user.
@item scan_all_pmts
Scan and combine all PMTs. The value is an integer with value from -1
to 1 (-1 means automatic setting, 1 means enabled, 0 means
disabled). Default value is -1.
@end table
@section mpjpeg
MJPEG encapsulated in multi-part MIME demuxer.
This demuxer allows reading of MJPEG, where each frame is represented as a part of
multipart/x-mixed-replace stream.
@table @option
@item strict_mime_boundary
Default implementation applies a relaxed standard to multi-part MIME boundary detection,
to prevent regression with numerous existing endpoints not generating a proper MIME
MJPEG stream. Turning this option on by setting it to 1 will result in a stricter check
of the boundary value.
@end table
@section rawvideo

View File

@@ -1,5 +1,4 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle Developer Documentation
@titlepage
@@ -28,14 +27,14 @@ this document.
For more detailed legal information about the use of FFmpeg in
external programs read the @file{LICENSE} file in the source tree and
consult @url{https://ffmpeg.org/legal.html}.
consult @url{http://ffmpeg.org/legal.html}.
@section Contributing
There are 3 ways by which code gets into FFmpeg.
There are 3 ways by which code gets into ffmpeg.
@itemize @bullet
@item Submitting patches to the main developer mailing list.
See @ref{Submitting patches} for details.
@item Submitting Patches to the main developer mailing list
see @ref{Submitting patches} for details.
@item Directly committing changes to the main tree.
@item Committing changes to a git clone, for example on github.com or
gitorious.org. And asking us to merge these changes.
@@ -65,9 +64,6 @@ rejected by the git repository.
@item
You should try to limit your code lines to 80 characters; however, do so if
and only if this improves readability.
@item
K&R coding style is used.
@end itemize
The presentation is one inspired by 'indent -i4 -kr -nut'.
@@ -127,10 +123,10 @@ the @samp{inline} keyword;
@samp{//} comments;
@item
designated struct initializers (@samp{struct s x = @{ .i = 17 @};});
designated struct initializers (@samp{struct s x = @{ .i = 17 @};})
@item
compound literals (@samp{x = (struct s) @{ 17, 23 @};}).
compound literals (@samp{x = (struct s) @{ 17, 23 @};})
@end itemize
These features are supported by all compilers we care about, so we will not
@@ -159,7 +155,7 @@ GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}).
All names should be composed with underscores (_), not CamelCase. For example,
@samp{avfilter_get_video_buffer} is an acceptable function name and
@samp{AVFilterGetVideo} is not. The exception from this are type names, like
for example structs and enums; they should always be in CamelCase.
for example structs and enums; they should always be in the CamelCase
There are the following conventions for naming variables and functions:
@@ -231,7 +227,7 @@ autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@@<!$/
@end example
For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
@lisp
@example
(c-add-style "ffmpeg"
'("k&r"
(c-basic-offset . 4)
@@ -242,12 +238,12 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
)
)
(setq c-default-style "ffmpeg")
@end lisp
@end example
@section Development Policy
@subsection Patches/Committing
@subheading Licenses for patches must be compatible with FFmpeg.
@enumerate
@item
Contributions should be licensed under the
@uref{http://www.gnu.org/licenses/lgpl-2.1.html, LGPL 2.1},
including an "or any later version" clause, or, if you prefer
@@ -260,15 +256,15 @@ preferred.
If you add a new file, give it a proper license header. Do not copy and
paste it from a random place, use an existing file as template.
@subheading You must not commit code which breaks FFmpeg!
This means unfinished code which is enabled and breaks compilation,
or compiles but does not work/breaks the regression tests. Code which
is unfinished but disabled may be permitted under-circumstances, like
missing samples or an implementation with a small subset of features.
Always check the mailing list for any reviewers with issues and test
FATE before you push.
@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.
@subheading Keep the main commit message short with an extended description below.
@item
The commit message should have a short first line in the form of
a @samp{topic: short description} as a header, separated by a newline
from the body consisting of an explanation of why the change is necessary.
@@ -276,24 +272,30 @@ If the commit fixes a known bug on the bug tracker, the commit message
should include its bug ID. Referring to the issue on the bug tracker does
not exempt you from writing an excerpt of the bug in the commit message.
@subheading Testing must be adequate but not excessive.
If it works for you, others, and passes FATE then it should be OK to commit
it, provided it fits the other committing criteria. You should not worry about
over-testing things. If your code has problems (portability, triggers
compiler bugs, unusual environment etc) they will be reported and eventually
fixed.
@item
You do not have to over-test things. If it works for you, and you think it
should work for others, then commit. If your code has problems
(portability, triggers compiler bugs, unusual environment etc) they will be
reported and eventually fixed.
@subheading Do not commit unrelated changes together.
They should be 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.
@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.
@subheading Ask before you change the build system (configure, etc).
@item
Do not change behavior of the programs (renaming options etc) or public
API or ABI without first discussing it on the ffmpeg-devel mailing list.
Do not remove functionality from the code. Just improve!
Note: Redundant code can be removed.
@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
@@ -302,7 +304,7 @@ 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.
@subheading Cosmetic changes should be kept in separate patches.
@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
@@ -316,43 +318,56 @@ 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
@subheading Commit messages should always be filled out properly.
@item
Always fill out the commit log message. Describe in a few lines what you
changed and why. You can refer to mailing list postings if you fix a
particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.
Recommended format:
@example
area changed: Short 1 line description
details describing what and why and giving references.
@end example
@subheading Credit the author of the patch.
@item
Make sure the author of the commit is set correctly. (see git commit --author)
If you apply a patch, send an
answer to ffmpeg-devel (or wherever you got the patch from) saying that
you applied the patch.
@subheading Complex patches should refer to discussion surrounding them.
@item
When applying patches that have been discussed (at length) on the mailing
list, reference the thread in the log message.
@subheading Always wait long enough before pushing changes
@item
Do NOT commit to code actively maintained by others without permission.
Send a patch to ffmpeg-devel. If no one answers within a reasonable
time-frame (12h for build failures and security fixes, 3 days small changes,
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!
@subsection Code
@subheading API/ABI changes should be discussed before they are made.
Do not change behavior of the programs (renaming options etc) or public
API or ABI without first discussing it on the ffmpeg-devel mailing list.
Do not remove widely used functionality or features (redundant code can be removed).
@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.
@subheading Remember to check if you need to bump versions for libav*.
Depending on the change, you may need to change the version integer.
@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
@@ -362,7 +377,7 @@ Incrementing the third component means a noteworthy binary compatible
change (e.g. encoder bug fix that matters for the decoder). The third
component always starts at 100 to distinguish FFmpeg from Libav.
@subheading Warnings for correct code may be disabled if there is no other option.
@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.
@@ -371,65 +386,16 @@ 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.
@subheading Check untrusted input properly.
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.
@subsection Documentation/Other
@subheading Subscribe to the ffmpeg-cvslog mailing list.
It is important to do this as 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.
@subheading Keep the documentation up to date.
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.
@subheading Important discussions should be accessible to all.
Try to keep important discussions and requests (also) on the public
developer mailing list, so that all developers can benefit from them.
@subheading Check your entries in MAINTAINERS.
@item
Make sure that no parts of the codebase that you maintain are missing from the
@file{MAINTAINERS} file. If something that you want to maintain is missing add it with
your name after it.
If at some point you no longer want to maintain some code, then please help in
finding a new maintainer and also don't forget to update the @file{MAINTAINERS} file.
If at some point you no longer want to maintain some code, then please help
finding a new maintainer and also don't forget updating the @file{MAINTAINERS} file.
@end enumerate
We think our rules are not too hard. If you have comments, contact us.
@section Code of conduct
Be friendly and respectful towards others and third parties.
Treat others the way you yourself want to be treated.
Be considerate. Not everyone shares the same viewpoint and priorities as you do.
Different opinions and interpretations help the project.
Looking at issues from a different perspective assists development.
Do not assume malice for things that can be attributed to incompetence. Even if
it is malice, it's rarely good to start with that as initial assumption.
Stay friendly even if someone acts contrarily. Everyone has a bad day
once in a while.
If you yourself have a bad day or are angry then try to take a break and reply
once you are calm and without anger if you have to.
Try to help other team members and cooperate if you can.
The goal of software development is to create technical excellence, not for any
individual to be better and "win" against the others. Large software projects
are only possible and successful through teamwork.
If someone struggles do not put them down. Give them a helping hand
instead and point them in the right direction.
Finally, keep in mind the immortal words of Bill and Ted,
"Be excellent to each other."
@anchor{Submitting patches}
@section Submitting patches
@@ -437,7 +403,7 @@ First, read the @ref{Coding Rules} above if you did not yet, in particular
the rules regarding patch submission.
When you submit your patch, please use @code{git format-patch} or
@code{git send-email}. We cannot read other diffs :-).
@code{git send-email}. We cannot read other diffs :-)
Also please do not submit a patch which contains several unrelated changes.
Split it into separate, self-contained pieces. This does not mean splitting
@@ -460,15 +426,11 @@ Also please if you send several patches, send each patch as a separate mail,
do not attach several unrelated patches to the same mail.
Patches should be posted to the
@uref{https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel, ffmpeg-devel}
@uref{http://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel, ffmpeg-devel}
mailing list. Use @code{git send-email} when possible since it will properly
send patches without requiring extra care. If you cannot, then send patches
as base64-encoded attachments, so your patch is not trashed during
transmission. Also ensure the correct mime type is used
(text/x-diff or text/x-patch or at least text/plain) and that only one
patch is inline or attached per mail.
You can check @url{https://patchwork.ffmpeg.org}, if your patch does not show up, its mime type
likely was wrong.
transmission.
Your patch will be reviewed on the mailing list. You will likely be asked
to make some changes and are expected to send in an improved version that
@@ -577,10 +539,6 @@ tools/trasher, the noise bitstream filter, and
should not crash, end in a (near) infinite loop, or allocate ridiculous
amounts of memory when fed damaged data.
@item
Did you test your decoder or demuxer against sample files?
Samples may be obtained at @url{https://samples.ffmpeg.org}.
@item
Does the patch not mix functional and cosmetic changes?
@@ -601,7 +559,7 @@ If the patch fixes a bug, did you provide a verbose analysis of the bug?
If the patch fixes a bug, did you provide enough information, including
a sample, so the bug can be reproduced and the fix can be verified?
Note please do not attach samples >100k to mails but rather provide a
URL, you can upload to ftp://upload.ffmpeg.org.
URL, you can upload to ftp://upload.ffmpeg.org
@item
Did you provide a verbose summary about what the patch does change?
@@ -630,10 +588,10 @@ Lines with similar content should be aligned vertically when doing so
improves readability.
@item
Consider adding a regression test for your code.
Consider to add a regression test for your code.
@item
If you added YASM code please check that things still work with --disable-yasm.
If you added YASM code please check that things still work with --disable-yasm
@item
Make sure you check the return values of function and return appropriate
@@ -671,10 +629,6 @@ not related to the comments received during review. Such patches will
be rejected. Instead, submit significant changes or new features as
separate patches.
Everyone is welcome to review patches. Also if you are waiting for your patch
to be reviewed, please consider helping to review other patches, that is a great
way to get everyone's patches reviewed sooner.
@anchor{Regression tests}
@section Regression tests
@@ -690,14 +644,15 @@ accordingly].
@subsection Adding files to the fate-suite dataset
When there is no muxer or encoder available to generate test media for a
specific test then the media has to be included in the fate-suite.
specific test then the media has to be inlcuded in the fate-suite.
First please make sure that the sample file is as small as possible to test the
respective decoder or demuxer sufficiently. Large files increase network
bandwidth and disk space requirements.
Once you have a working fate test and fate sample, provide in the commit
message or introductory message for the patch series that you post to
message or introductionary message for the patch series that you post to
the ffmpeg-devel mailing list, a direct link to download the sample media.
@subsection Visualizing Test Coverage
The FFmpeg build system allows visualizing the test coverage in an easy
@@ -745,7 +700,7 @@ FFmpeg maintains a set of @strong{release branches}, which are the
recommended deliverable for system integrators and distributors (such as
Linux distributions, etc.). At regular times, a @strong{release
manager} prepares, tests and publishes tarballs on the
@url{https://ffmpeg.org} website.
@url{http://ffmpeg.org} website.
There are two kinds of releases:
@@ -824,7 +779,7 @@ Prepare the release tarballs in @code{bz2} and @code{gz} formats, and
supplementing files that contain @code{gpg} signatures
@item
Publish the tarballs at @url{https://ffmpeg.org/releases}. Create and
Publish the tarballs at @url{http://ffmpeg.org/releases}. Create and
push an annotated tag in the form @code{nX}, with @code{X}
containing the version number.
@@ -836,7 +791,7 @@ with a news entry for the website.
Publish the news entry.
@item
Send an announcement to the mailing list.
Send announcement to the mailing list.
@end enumerate
@bye

View File

@@ -1,21 +1,13 @@
#!/bin/sh
OUT_DIR="${1}"
SRC_PATH="${1}"
DOXYFILE="${2}"
DOXYGEN="${3}"
shift 3
if [ -e "VERSION" ]; then
VERSION=`cat "VERSION"`
else
VERSION=`git describe`
fi
$DOXYGEN - <<EOF
@INCLUDE = ${DOXYFILE}
INPUT = $@
HTML_TIMESTAMP = NO
PROJECT_NUMBER = $VERSION
OUTPUT_DIRECTORY = $OUT_DIR
EXAMPLE_PATH = ${SRC_PATH}/doc/examples
EOF

1
doc/doxy/.gitignore vendored
View File

@@ -1 +0,0 @@
/html/

View File

@@ -30,120 +30,81 @@ follows.
Advanced Audio Coding (AAC) encoder.
This encoder is the default AAC encoder, natively implemented into FFmpeg. Its
quality is on par or better than libfdk_aac at the default bitrate of 128kbps.
This encoder also implements more options, profiles and samplerates than
other encoders (with only the AAC-HE profile pending to be implemented) so this
encoder has become the default and is the recommended choice.
This encoder is an experimental FFmpeg-native AAC encoder. Currently only the
low complexity (AAC-LC) profile is supported. To use this encoder, you must set
@option{strict} option to @samp{experimental} or lower.
As this encoder is experimental, unexpected behavior may exist from time to
time. For a more stable AAC encoder, see @ref{libvo-aacenc}. However, be warned
that it has a worse quality reported by some users.
@c todo @ref{libaacplus}
See also @ref{libfdk-aac-enc,,libfdk_aac} and @ref{libfaac}.
@subsection Options
@table @option
@item b
Set bit rate in bits/s. Setting this automatically activates constant bit rate
(CBR) mode. If this option is unspecified it is set to 128kbps.
(CBR) mode.
@item q
Set quality for variable bit rate (VBR) mode. This option is valid only using
the @command{ffmpeg} command-line tool. For library interface users, use
@option{global_quality}.
@item cutoff
Set cutoff frequency. If unspecified will allow the encoder to dynamically
adjust the cutoff to improve clarity on low bitrates.
@item stereo_mode
Set stereo encoding mode. Possible values:
@table @samp
@item auto
Automatically selected by the encoder.
@item ms_off
Disable middle/side encoding. This is the default.
@item ms_force
Force middle/side encoding.
@end table
@item aac_coder
Set AAC encoder coding method. Possible values:
@table @samp
@item faac
FAAC-inspired method.
This method is a simplified reimplementation of the method used in FAAC, which
sets thresholds proportional to the band energies, and then decreases all the
thresholds with quantizer steps to find the appropriate quantization with
distortion below threshold band by band.
The quality of this method is comparable to the two loop searching method
described below, but somewhat a little better and slower.
@item anmr
Average noise to mask ratio (ANMR) trellis-based solution.
This has a theoretic best quality out of all the coding methods, but at the
cost of the slowest speed.
@item twoloop
Two loop searching (TLS) method.
This method first sets quantizers depending on band thresholds and then tries
to find an optimal combination by adding or subtracting a specific value from
all quantizers and adjusting some individual quantizer a little. Will tune
itself based on whether @option{aac_is}, @option{aac_ms} and @option{aac_pns}
are enabled.
This is the default choice for a coder.
all quantizers and adjusting some individual quantizer a little.
@item anmr
Average noise to mask ratio (ANMR) trellis-based solution.
This is an experimental coder which currently produces a lower quality, is more
unstable and is slower than the default twoloop coder but has potential.
Currently has no support for the @option{aac_is} or @option{aac_pns} options.
Not currently recommended.
This method produces similar quality with the FAAC method and is the default.
@item fast
Constant quantizer method.
This method sets a constant quantizer for all bands. This is the fastest of all
the methods and has no rate control or support for @option{aac_is} or
@option{aac_pns}.
Not recommended.
the methods, yet produces the worst quality.
@end table
@item aac_ms
Sets mid/side coding mode. The default value of "auto" will automatically use
M/S with bands which will benefit from such coding. Can be forced for all bands
using the value "enable", which is mainly useful for debugging or disabled using
"disable".
@item aac_is
Sets intensity stereo coding tool usage. By default, it's enabled and will
automatically toggle IS for similar pairs of stereo bands if it's benefitial.
Can be disabled for debugging by setting the value to "disable".
@item aac_pns
Uses perceptual noise substitution to replace low entropy high frequency bands
with imperceivable white noise during the decoding process. By default, it's
enabled, but can be disabled for debugging purposes by using "disable".
@item aac_tns
Enables the use of a multitap FIR filter which spans through the high frequency
bands to hide quantization noise during the encoding process and is reverted
by the decoder. As well as decreasing unpleasant artifacts in the high range
this also reduces the entropy in the high bands and allows for more bits to
be used by the mid-low bands. By default it's enabled but can be disabled for
debugging by setting the option to "disable".
@item aac_ltp
Enables the use of the long term prediction extension which increases coding
efficiency in very low bandwidth situations such as encoding of voice or
solo piano music by extending constant harmonic peaks in bands throughout
frames. This option is implied by profile:a aac_low and is incompatible with
aac_pred. Use in conjunction with @option{-ar} to decrease the samplerate.
@item aac_pred
Enables the use of a more traditional style of prediction where the spectral
coefficients transmitted are replaced by the difference of the current
coefficients minus the previous "predicted" coefficients. In theory and sometimes
in practice this can improve quality for low to mid bitrate audio.
This option implies the aac_main profile and is incompatible with aac_ltp.
@item profile
Sets the encoding profile, possible values:
@table @samp
@item aac_low
The default, AAC "Low-complexity" profile. Is the most compatible and produces
decent quality.
@item mpeg2_aac_low
Equivalent to @code{-profile:a aac_low -aac_pns 0}. PNS was introduced with the
MPEG4 specifications.
@item aac_ltp
Long term prediction profile, is enabled by and will enable the @option{aac_ltp}
option. Introduced in MPEG4.
@item aac_main
Main-type prediction profile, is enabled by and will enable the @option{aac_pred}
option. Introduced in MPEG2.
@end table
If this option is unspecified it is set to @samp{aac_low}.
@end table
@section ac3 and ac3_fixed
@@ -533,84 +494,112 @@ Selected by Encoder (default)
@end table
@anchor{flac}
@section flac
@anchor{libfaac}
@section libfaac
FLAC (Free Lossless Audio Codec) Encoder
libfaac AAC (Advanced Audio Coding) encoder wrapper.
Requires the presence of the libfaac headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libfaac --enable-nonfree}.
This encoder is considered to be of higher quality with respect to the
@ref{aacenc,,the native experimental FFmpeg AAC encoder}.
For more information see the libfaac project at
@url{http://www.audiocoding.com/faac.html/}.
@subsection Options
The following options are supported by FFmpeg's flac encoder.
The following shared FFmpeg codec options are recognized.
The following options are supported by the libfaac wrapper. The
@command{faac}-equivalent of the options are listed in parentheses.
@table @option
@item compression_level
Sets the compression level, which chooses defaults for many other options
if they are not set explicitly.
@item b (@emph{-b})
Set bit rate in bits/s for ABR (Average Bit Rate) mode. If the bit rate
is not explicitly specified, it is automatically set to a suitable
value depending on the selected profile. @command{faac} bitrate is
expressed in kilobits/s.
@item frame_size
Sets the size of the frames in samples per channel.
Note that libfaac does not support CBR (Constant Bit Rate) but only
ABR (Average Bit Rate).
@item lpc_coeff_precision
Sets the LPC coefficient precision, valid values are from 1 to 15, 15 is the
default.
If VBR mode is enabled this option is ignored.
@item lpc_type
Sets the first stage LPC algorithm
@item ar (@emph{-R})
Set audio sampling rate (in Hz).
@item ac (@emph{-c})
Set the number of audio channels.
@item cutoff (@emph{-C})
Set cutoff frequency. If not specified (or explicitly set to 0) it
will use a value automatically computed by the library. Default value
is 0.
@item profile
Set audio profile.
The following profiles are recognized:
@table @samp
@item none
LPC is not used
@item aac_main
Main AAC (Main)
@item fixed
fixed LPC coefficients
@item aac_low
Low Complexity AAC (LC)
@item levinson
@item aac_ssr
Scalable Sample Rate (SSR)
@item cholesky
@item aac_ltp
Long Term Prediction (LTP)
@end table
@item lpc_passes
Number of passes to use for Cholesky factorization during LPC analysis
If not specified it is set to @samp{aac_low}.
@item min_partition_order
The minimum partition order
@item flags +qscale
Set constant quality VBR (Variable Bit Rate) mode.
@item max_partition_order
The maximum partition order
@item global_quality
Set quality in VBR mode as an integer number of lambda units.
@item prediction_order_method
@table @samp
@item estimation
@item 2level
@item 4level
@item 8level
@item search
Bruteforce search
@item log
Only relevant when VBR mode is enabled with @code{flags +qscale}. The
value is converted to QP units by dividing it by @code{FF_QP2LAMBDA},
and used to set the quality value used by libfaac. A reasonable range
for the option value in QP units is [10-500], the higher the value the
higher the quality.
@item q (@emph{-q})
Enable VBR mode when set to a non-negative value, and set constant
quality value as a double floating point value in QP units.
The value sets the quality value used by libfaac. A reasonable range
for the option value is [10-500], the higher the value the higher the
quality.
This option is valid only using the @command{ffmpeg} command-line
tool. For library interface users, use @option{global_quality}.
@end table
@item ch_mode
Channel mode
@table @samp
@item auto
The mode is chosen automatically for each frame
@item indep
Chanels are independently coded
@item left_side
@item right_side
@item mid_side
@end table
@subsection Examples
@item exact_rice_parameters
Chooses if rice parameters are calculated exactly or approximately.
if set to 1 then they are chosen exactly, which slows the code down slightly and
improves compression slightly.
@itemize
@item
Use @command{ffmpeg} to convert an audio file to ABR 128 kbps AAC in an M4A (MP4)
container:
@example
ffmpeg -i input.wav -codec:a libfaac -b:a 128k -output.m4a
@end example
@item multi_dim_quant
Multi Dimensional Quantization. If set to 1 then a 2nd stage LPC algorithm is
applied after the first stage to finetune the coefficients. This is quite slow
and slightly improves compression.
@end table
@item
Use @command{ffmpeg} to convert an audio file to VBR AAC, using the
LTP AAC profile:
@example
ffmpeg -i input.wav -c:a libfaac -profile:a aac_ltp -q:a 100 output.m4a
@end example
@end itemize
@anchor{libfdk-aac-enc}
@section libfdk_aac
@@ -626,10 +615,9 @@ configuration. You need to explicitly configure the build with
so if you allow the use of GPL, you should configure with
@code{--enable-gpl --enable-nonfree --enable-libfdk-aac}.
This encoder is considered to produce output on par or worse at 128kbps to the
@ref{aacenc,,the native FFmpeg AAC encoder} but can often produce better
sounding audio at identical or lower bitrates and has support for the
AAC-HE profiles.
This encoder is considered to be of higher quality with respect to
both @ref{aacenc,,the native experimental FFmpeg AAC encoder} and
@ref{libfaac}.
VBR encoding, enabled through the @option{vbr} or @option{flags
+qscale} options, is experimental and only works with some
@@ -971,6 +959,31 @@ Set MPEG audio original flag when set to 1. The default value is 0
@end table
@anchor{libvo-aacenc}
@section libvo-aacenc
VisualOn AAC encoder.
Requires the presence of the libvo-aacenc headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libvo-aacenc --enable-version3}.
This encoder is considered to be worse than the
@ref{aacenc,,native experimental FFmpeg AAC encoder}, according to
multiple sources.
@subsection Options
The VisualOn AAC encoder only support encoding AAC-LC and up to 2
channels. It is also CBR-only.
@table @option
@item b
Set bit rate in bits/s.
@end table
@section libvo-amrwbenc
VisualOn Adaptive Multi-Rate Wideband encoder.
@@ -1019,7 +1032,7 @@ configuration. You need to explicitly configure the build with
@subsection Option Mapping
Most libopus options are modelled after the @command{opusenc} utility from
Most libopus options are modeled after the @command{opusenc} utility from
opus-tools. The following is an option mapping chart describing options
supported by the libopus wrapper, and their @command{opusenc}-equivalent
in parentheses.
@@ -1033,7 +1046,7 @@ kilobits/s.
@item vbr (@emph{vbr}, @emph{hard-cbr}, and @emph{cvbr})
Set VBR mode. The FFmpeg @option{vbr} option has the following
valid arguments, with the @command{opusenc} equivalent options
valid arguments, with the their @command{opusenc} equivalent options
in parentheses:
@table @samp
@@ -1080,17 +1093,6 @@ following: 4000, 6000, 8000, 12000, or 20000, corresponding to
narrowband, mediumband, wideband, super wideband, and fullband
respectively. The default is 0 (cutoff disabled).
@item mapping_family (@emph{mapping_family})
Set channel mapping family to be used by the encoder. The default value of -1
uses mapping family 0 for mono and stereo inputs, and mapping family 1
otherwise. The default also disables the surround masking and LFE bandwidth
optimzations in libopus, and requires that the input contains 8 channels or
fewer.
Other values include 0 for mono and stereo, 1 for surround sound with masking
and LFE bandwidth optimizations, and 255 for independent streams with an
unspecified channel layout.
@end table
@section libvorbis
@@ -1261,96 +1263,6 @@ disabled
A description of some of the currently available video encoders
follows.
@section libopenh264
Cisco libopenh264 H.264/MPEG-4 AVC encoder wrapper.
This encoder requires the presence of the libopenh264 headers and
library during configuration. You need to explicitly configure the
build with @code{--enable-libopenh264}. The library is detected using
@command{pkg-config}.
For more information about the library see
@url{http://www.openh264.org}.
@subsection Options
The following FFmpeg global options affect the configurations of the
libopenh264 encoder.
@table @option
@item b
Set the bitrate (as a number of bits per second).
@item g
Set the GOP size.
@item maxrate
Set the max bitrate (as a number of bits per second).
@item flags +global_header
Set global header in the bitstream.
@item slices
Set the number of slices, used in parallelized encoding. Default value
is 0. This is only used when @option{slice_mode} is set to
@samp{fixed}.
@item slice_mode
Set slice mode. Can assume one of the following possible values:
@table @samp
@item fixed
a fixed number of slices
@item rowmb
one slice per row of macroblocks
@item auto
automatic number of slices according to number of threads
@item dyn
dynamic slicing
@end table
Default value is @samp{auto}.
@item loopfilter
Enable loop filter, if set to 1 (automatically enabled). To disable
set a value of 0.
@item profile
Set profile restrictions. If set to the value of @samp{main} enable
CABAC (set the @code{SEncParamExt.iEntropyCodingModeFlag} flag to 1).
@item max_nal_size
Set maximum NAL size in bytes.
@item allow_skip_frames
Allow skipping frames to hit the target bitrate if set to 1.
@end table
@section jpeg2000
The native jpeg 2000 encoder is lossy by default, the @code{-q:v}
option can be used to set the encoding quality. Lossless encoding
can be selected with @code{-pred 1}.
@subsection Options
@table @option
@item format
Can be set to either @code{j2k} or @code{jp2} (the default) that
makes it possible to store non-rgb pix_fmts.
@end table
@section snow
@subsection Options
@table @option
@item iterative_dia_size
dia size for the iterative motion estimation
@end table
@section libtheora
libtheora Theora encoder wrapper.
@@ -1418,166 +1330,117 @@ ffmpeg -i INPUT -codec:v libtheora -b:v 1000k OUTPUT.ogg
@section libvpx
VP8/VP9 format supported through libvpx.
VP8 format supported through libvpx.
Requires the presence of the libvpx headers and library during configuration.
You need to explicitly configure the build with @code{--enable-libvpx}.
@subsection Options
The following options are supported by the libvpx wrapper. The
@command{vpxenc}-equivalent options or values are listed in parentheses
for easy migration.
To reduce the duplication of documentation, only the private options
and some others requiring special attention are documented here. For
the documentation of the undocumented generic options, see
@ref{codec-options,,the Codec Options chapter}.
To get more documentation of the libvpx options, invoke the command
@command{ffmpeg -h encoder=libvpx}, @command{ffmpeg -h encoder=libvpx-vp9} or
@command{vpxenc --help}. Further information is available in the libvpx API
documentation.
Mapping from FFmpeg to libvpx options with conversion notes in parentheses.
@table @option
@item b (@emph{target-bitrate})
Set bitrate in bits/s. Note that FFmpeg's @option{b} option is
expressed in bits/s, while @command{vpxenc}'s @option{target-bitrate} is in
kilobits/s.
@item threads
g_threads
@item g (@emph{kf-max-dist})
@item profile
g_profile
@item keyint_min (@emph{kf-min-dist})
@item vb
rc_target_bitrate
@item qmin (@emph{min-q})
@item g
kf_max_dist
@item qmax (@emph{max-q})
@item keyint_min
kf_min_dist
@item bufsize (@emph{buf-sz}, @emph{buf-optimal-sz})
Set ratecontrol buffer size (in bits). Note @command{vpxenc}'s options are
specified in milliseconds, the libvpx wrapper converts this value as follows:
@code{buf-sz = bufsize * 1000 / bitrate},
@code{buf-optimal-sz = bufsize * 1000 / bitrate * 5 / 6}.
@item qmin
rc_min_quantizer
@item rc_init_occupancy (@emph{buf-initial-sz})
Set number of bits which should be loaded into the rc buffer before decoding
starts. Note @command{vpxenc}'s option is specified in milliseconds, the libvpx
wrapper converts this value as follows:
@code{rc_init_occupancy * 1000 / bitrate}.
@item qmax
rc_max_quantizer
@item undershoot-pct
Set datarate undershoot (min) percentage of the target bitrate.
@item bufsize, vb
rc_buf_sz
@code{(bufsize * 1000 / vb)}
@item overshoot-pct
Set datarate overshoot (max) percentage of the target bitrate.
rc_buf_optimal_sz
@code{(bufsize * 1000 / vb * 5 / 6)}
@item skip_threshold (@emph{drop-frame})
@item rc_init_occupancy, vb
rc_buf_initial_sz
@code{(rc_init_occupancy * 1000 / vb)}
@item qcomp (@emph{bias-pct})
@item rc_buffer_aggressivity
rc_undershoot_pct
@item maxrate (@emph{maxsection-pct})
Set GOP max bitrate in bits/s. Note @command{vpxenc}'s option is specified as a
percentage of the target bitrate, the libvpx wrapper converts this value as
follows: @code{(maxrate * 100 / bitrate)}.
@item skip_threshold
rc_dropframe_thresh
@item minrate (@emph{minsection-pct})
Set GOP min bitrate in bits/s. Note @command{vpxenc}'s option is specified as a
percentage of the target bitrate, the libvpx wrapper converts this value as
follows: @code{(minrate * 100 / bitrate)}.
@item qcomp
rc_2pass_vbr_bias_pct
@item minrate, maxrate, b @emph{end-usage=cbr}
@code{(minrate == maxrate == bitrate)}.
@item maxrate, vb
rc_2pass_vbr_maxsection_pct
@code{(maxrate * 100 / vb)}
@item crf (@emph{end-usage=cq}, @emph{cq-level})
@item minrate, vb
rc_2pass_vbr_minsection_pct
@code{(minrate * 100 / vb)}
@item tune (@emph{tune})
@table @samp
@item psnr (@emph{psnr})
@item ssim (@emph{ssim})
@item minrate, maxrate, vb
@code{VPX_CBR}
@code{(minrate == maxrate == vb)}
@item crf
@code{VPX_CQ}, @code{VP8E_SET_CQ_LEVEL}
@item quality
@table @option
@item @var{best}
@code{VPX_DL_BEST_QUALITY}
@item @var{good}
@code{VPX_DL_GOOD_QUALITY}
@item @var{realtime}
@code{VPX_DL_REALTIME}
@end table
@item quality, deadline (@emph{deadline})
@table @samp
@item best
Use best quality deadline. Poorly named and quite slow, this option should be
avoided as it may give worse quality output than good.
@item good
Use good quality deadline. This is a good trade-off between speed and quality
when used with the @option{cpu-used} option.
@item realtime
Use realtime quality deadline.
@end table
@item speed
@code{VP8E_SET_CPUUSED}
@item speed, cpu-used (@emph{cpu-used})
Set quality/speed ratio modifier. Higher values speed up the encode at the cost
of quality.
@item nr
@code{VP8E_SET_NOISE_SENSITIVITY}
@item nr (@emph{noise-sensitivity})
@item mb_threshold
@code{VP8E_SET_STATIC_THRESHOLD}
@item static-thresh
Set a change threshold on blocks below which they will be skipped by the
encoder.
@item slices (@emph{token-parts})
Note that FFmpeg's @option{slices} option gives the total number of partitions,
while @command{vpxenc}'s @option{token-parts} is given as
@code{log2(partitions)}.
@item slices
@code{VP8E_SET_TOKEN_PARTITIONS}
@item max-intra-rate
Set maximum I-frame bitrate as a percentage of the target bitrate. A value of 0
means unlimited.
@code{VP8E_SET_MAX_INTRA_BITRATE_PCT}
@item force_key_frames
@code{VPX_EFLAG_FORCE_KF}
@item Alternate reference frame related
@table @option
@item auto-alt-ref
Enable use of alternate reference frames (2-pass only).
@item arnr-max-frames
Set altref noise reduction max frame count.
@item arnr-type
Set altref noise reduction filter type: backward, forward, centered.
@item arnr-strength
Set altref noise reduction filter strength.
@item rc-lookahead, lag-in-frames (@emph{lag-in-frames})
Set number of frames to look ahead for frametype and ratecontrol.
@item vp8flags altref
@code{VP8E_SET_ENABLEAUTOALTREF}
@item @var{arnr_max_frames}
@code{VP8E_SET_ARNR_MAXFRAMES}
@item @var{arnr_type}
@code{VP8E_SET_ARNR_TYPE}
@item @var{arnr_strength}
@code{VP8E_SET_ARNR_STRENGTH}
@item @var{rc_lookahead}
g_lag_in_frames
@end table
@item error-resilient
Enable error resiliency features.
@item VP9-specific options
@table @option
@item lossless
Enable lossless mode.
@item tile-columns
Set number of tile columns to use. Note this is given as
@code{log2(tile_columns)}. For example, 8 tile columns would be requested by
setting the @option{tile-columns} option to 3.
@item tile-rows
Set number of tile rows to use. Note this is given as @code{log2(tile_rows)}.
For example, 4 tile rows would be requested by setting the @option{tile-rows}
option to 2.
@item frame-parallel
Enable frame parallel decodability features.
@item aq-mode
Set adaptive quantization mode (0: off (default), 1: variance 2: complexity, 3:
cyclic refresh).
@item colorspace @emph{color-space}
Set input color space. The VP9 bitstream supports signaling the following
colorspaces:
@table @option
@item @samp{rgb} @emph{sRGB}
@item @samp{bt709} @emph{bt709}
@item @samp{unspecified} @emph{unknown}
@item @samp{bt470bg} @emph{bt601}
@item @samp{smpte170m} @emph{smpte170}
@item @samp{smpte240m} @emph{smpte240}
@item @samp{bt2020_ncl} @emph{bt2020}
@end table
@end table
@item vp8flags error_resilient
g_error_resilient
@end table
@@ -1773,7 +1636,7 @@ Enable CAVLC and disable CABAC. It generates the same effect as
@end table
@item cmp
Set full pixel motion estimation comparison algorithm. Possible values:
Set full pixel motion estimation comparation algorithm. Possible values:
@table @samp
@item chroma
@@ -1879,10 +1742,6 @@ Enable calculation and printing SSIM stats after the encoding.
Enable the use of Periodic Intra Refresh instead of IDR frames when set
to 1.
@item avcintra-class (@emph{class})
Configure the encoder to generate AVC-Intra.
Valid values are 50,100 and 200
@item bluray-compat (@emph{bluray-compat})
Configure the encoder to be compatible with the bluray standard.
It is a shorthand for setting "bluray-compat=1 force-cfr=1".
@@ -2002,10 +1861,6 @@ For example to specify libx264 encoding options with @command{ffmpeg}:
ffmpeg -i foo.mpg -vcodec libx264 -x264opts keyint=123:min-keyint=20 -an out.mkv
@end example
@item a53cc @var{boolean}
Import closed captions (which must be ATSC compatible format) into output.
Only the mpeg2 and h264 decoders provide these. Default is 1 (on).
@item x264-params (N.A.)
Override the x264 configuration using a :-separated list of key=value
parameters.
@@ -2024,34 +1879,6 @@ no-fast-pskip=1:subq=6:8x8dct=0:trellis=0 OUTPUT
Encoding ffpresets for common usages are provided so they can be used with the
general presets system (e.g. passing the @option{pre} option).
@section libx265
x265 H.265/HEVC encoder wrapper.
This encoder requires the presence of the libx265 headers and library
during configuration. You need to explicitly configure the build with
@option{--enable-libx265}.
@subsection Options
@table @option
@item preset
Set the x265 preset.
@item tune
Set the x265 tune parameter.
@item x265-params
Set x265 options using a list of @var{key}=@var{value} couples separated
by ":". See @command{x265 --help} for a list of options.
For example to specify libx265 encoding options with @option{-x265-params}:
@example
ffmpeg -i input -c:v libx265 -x265-params crf=26:psy-rd=1 output.mp4
@end example
@end table
@section libxvid
Xvid MPEG-4 Part 2 encoder wrapper.
@@ -2309,7 +2136,7 @@ Use @var{0} to disable alpha plane coding.
@subsection Speed considerations
In the default mode of operation the encoder has to honor frame constraints
(i.e. not produce frames with size bigger than requested) while still making
(i.e. not produc frames with size bigger than requested) while still making
output picture as good as possible.
A frame containing a lot of small details is harder to compress and the encoder
would spend more time searching for appropriate quantizers for each slice.
@@ -2319,180 +2146,6 @@ Setting a higher @option{bits_per_mb} limit will improve the speed.
For the fastest encoding speed set the @option{qscale} parameter (4 is the
recommended value) and do not set a size constraint.
@section libkvazaar
Kvazaar H.265/HEVC encoder.
Requires the presence of the libkvazaar headers and library during
configuration. You need to explicitly configure the build with
@option{--enable-libkvazaar}.
@subsection Options
@table @option
@item b
Set target video bitrate in bit/s and enable rate control.
@item kvazaar-params
Set kvazaar parameters as a list of @var{name}=@var{value} pairs separated
by commas (,). See kvazaar documentation for a list of options.
@end table
@section QSV encoders
The family of Intel QuickSync Video encoders (MPEG-2, H.264 and HEVC)
The ratecontrol method is selected as follows:
@itemize @bullet
@item
When @option{global_quality} is specified, a quality-based mode is used.
Specifically this means either
@itemize @minus
@item
@var{CQP} - constant quantizer scale, when the @option{qscale} codec flag is
also set (the @option{-qscale} ffmpeg option).
@item
@var{LA_ICQ} - intelligent constant quality with lookahead, when the
@option{look_ahead} option is also set.
@item
@var{ICQ} -- intelligent constant quality otherwise.
@end itemize
@item
Otherwise, a bitrate-based mode is used. For all of those, you should specify at
least the desired average bitrate with the @option{b} option.
@itemize @minus
@item
@var{LA} - VBR with lookahead, when the @option{look_ahead} option is specified.
@item
@var{VCM} - video conferencing mode, when the @option{vcm} option is set.
@item
@var{CBR} - constant bitrate, when @option{maxrate} is specified and equal to
the average bitrate.
@item
@var{VBR} - variable bitrate, when @option{maxrate} is specified, but is higher
than the average bitrate.
@item
@var{AVBR} - average VBR mode, when @option{maxrate} is not specified. This mode
is further configured by the @option{avbr_accuracy} and
@option{avbr_convergence} options.
@end itemize
@end itemize
Note that depending on your system, a different mode than the one you specified
may be selected by the encoder. Set the verbosity level to @var{verbose} or
higher to see the actual settings used by the QSV runtime.
Additional libavcodec global options are mapped to MSDK options as follows:
@itemize
@item
@option{g/gop_size} -> @option{GopPicSize}
@item
@option{bf/max_b_frames}+1 -> @option{GopRefDist}
@item
@option{rc_init_occupancy/rc_initial_buffer_occupancy} ->
@option{InitialDelayInKB}
@item
@option{slices} -> @option{NumSlice}
@item
@option{refs} -> @option{NumRefFrame}
@item
@option{b_strategy/b_frame_strategy} -> @option{BRefType}
@item
@option{cgop/CLOSED_GOP} codec flag -> @option{GopOptFlag}
@item
For the @var{CQP} mode, the @option{i_qfactor/i_qoffset} and
@option{b_qfactor/b_qoffset} set the difference between @var{QPP} and @var{QPI},
and @var{QPP} and @var{QPB} respectively.
@item
Setting the @option{coder} option to the value @var{vlc} will make the H.264
encoder use CAVLC instead of CABAC.
@end itemize
@section vc2
SMPTE VC-2 (previously BBC Dirac Pro). This codec was primarily aimed at
professional broadcasting but since it supports yuv420, yuv422 and yuv444 at
8 (limited range or full range), 10 or 12 bits, this makes it suitable for
other tasks which require low overhead and low compression (like screen
recording).
@subsection Options
@table @option
@item b
Sets target video bitrate. Usually that's around 1:6 of the uncompressed
video bitrate (e.g. for 1920x1080 50fps yuv422p10 that's around 400Mbps). Higher
values (close to the uncompressed bitrate) turn on lossless compression mode.
@item field_order
Enables field coding when set (e.g. to tt - top field first) for interlaced
inputs. Should increase compression with interlaced content as it splits the
fields and encodes each separately.
@item wavelet_depth
Sets the total amount of wavelet transforms to apply, between 1 and 5 (default).
Lower values reduce compression and quality. Less capable decoders may not be
able to handle values of @option{wavelet_depth} over 3.
@item wavelet_type
Sets the transform type. Currently only @var{5_3} (LeGall) and @var{9_7}
(Deslauriers-Dubuc)
are implemented, with 9_7 being the one with better compression and thus
is the default.
@item slice_width
@item slice_height
Sets the slice size for each slice. Larger values result in better compression.
For compatibility with other more limited decoders use @option{slice_width} of
32 and @option{slice_height} of 8.
@item tolerance
Sets the undershoot tolerance of the rate control system in percent. This is
to prevent an expensive search from being run.
@item qm
Sets the quantization matrix preset to use by default or when @option{wavelet_depth}
is set to 5
@itemize @minus
@item
@var{default}
Uses the default quantization matrix from the specifications, extended with
values for the fifth level. This provides a good balance between keeping detail
and omitting artifacts.
@item
@var{flat}
Use a completely zeroed out quantization matrix. This increases PSNR but might
reduce perception. Use in bogus benchmarks.
@item
@var{color}
Reduces detail but attempts to preserve color at extremely low bitrates.
@end itemize
@end table
@c man end VIDEO ENCODERS
@chapter Subtitles Encoders

View File

@@ -76,7 +76,7 @@ EMFILE POSIX ++++++ Too many open files
EMLINK POSIX ++++++ Too many links
EMSGSIZE POSIX +++..+ Message too long
EMULTIHOP POSIX ++4... Multihop attempted
ENAMETOOLONG POSIX - ++++++ File name too long
ENAMETOOLONG POSIX - ++++++ Filen ame too long
ENAVAIL +..... No XENIX semaphores available
ENEEDAUTH .++... Need authenticator
ENETDOWN POSIX +++..+ Network is down

View File

@@ -1,18 +0,0 @@
/avio_dir_cmd
/avio_reading
/decoding_encoding
/demuxing_decoding
/extract_mvs
/filter_audio
/filtering_audio
/filtering_video
/http_multiclient
/metadata
/muxing
/pc-uninstalled
/qsvdec
/remuxing
/resampling_audio
/scaling_video
/transcode_aac
/transcoding

View File

@@ -11,14 +11,11 @@ CFLAGS += -Wall -g
CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
EXAMPLES= avio_dir_cmd \
avio_reading \
EXAMPLES= avio_reading \
decoding_encoding \
demuxing_decoding \
extract_mvs \
filtering_video \
filtering_audio \
http_multiclient \
metadata \
muxing \
remuxing \
@@ -31,7 +28,6 @@ OBJS=$(addsuffix .o,$(EXAMPLES))
# the following examples make explicit use of the math library
avcodec: LDLIBS += -lm
decoding_encoding: LDLIBS += -lm
muxing: LDLIBS += -lm
resampling_audio: LDLIBS += -lm

View File

@@ -1,180 +0,0 @@
/*
* Copyright (c) 2014 Lukasz Marek
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavformat/avio.h>
static const char *type_string(int type)
{
switch (type) {
case AVIO_ENTRY_DIRECTORY:
return "<DIR>";
case AVIO_ENTRY_FILE:
return "<FILE>";
case AVIO_ENTRY_BLOCK_DEVICE:
return "<BLOCK DEVICE>";
case AVIO_ENTRY_CHARACTER_DEVICE:
return "<CHARACTER DEVICE>";
case AVIO_ENTRY_NAMED_PIPE:
return "<PIPE>";
case AVIO_ENTRY_SYMBOLIC_LINK:
return "<LINK>";
case AVIO_ENTRY_SOCKET:
return "<SOCKET>";
case AVIO_ENTRY_SERVER:
return "<SERVER>";
case AVIO_ENTRY_SHARE:
return "<SHARE>";
case AVIO_ENTRY_WORKGROUP:
return "<WORKGROUP>";
case AVIO_ENTRY_UNKNOWN:
default:
break;
}
return "<UNKNOWN>";
}
static int list_op(const char *input_dir)
{
AVIODirEntry *entry = NULL;
AVIODirContext *ctx = NULL;
int cnt, ret;
char filemode[4], uid_and_gid[20];
if ((ret = avio_open_dir(&ctx, input_dir, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open directory: %s.\n", av_err2str(ret));
goto fail;
}
cnt = 0;
for (;;) {
if ((ret = avio_read_dir(ctx, &entry)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot list directory: %s.\n", av_err2str(ret));
goto fail;
}
if (!entry)
break;
if (entry->filemode == -1) {
snprintf(filemode, 4, "???");
} else {
snprintf(filemode, 4, "%3"PRIo64, entry->filemode);
}
snprintf(uid_and_gid, 20, "%"PRId64"(%"PRId64")", entry->user_id, entry->group_id);
if (cnt == 0)
av_log(NULL, AV_LOG_INFO, "%-9s %12s %30s %10s %s %16s %16s %16s\n",
"TYPE", "SIZE", "NAME", "UID(GID)", "UGO", "MODIFIED",
"ACCESSED", "STATUS_CHANGED");
av_log(NULL, AV_LOG_INFO, "%-9s %12"PRId64" %30s %10s %s %16"PRId64" %16"PRId64" %16"PRId64"\n",
type_string(entry->type),
entry->size,
entry->name,
uid_and_gid,
filemode,
entry->modification_timestamp,
entry->access_timestamp,
entry->status_change_timestamp);
avio_free_directory_entry(&entry);
cnt++;
};
fail:
avio_close_dir(&ctx);
return ret;
}
static int del_op(const char *url)
{
int ret = avpriv_io_delete(url);
if (ret < 0)
av_log(NULL, AV_LOG_ERROR, "Cannot delete '%s': %s.\n", url, av_err2str(ret));
return ret;
}
static int move_op(const char *src, const char *dst)
{
int ret = avpriv_io_move(src, dst);
if (ret < 0)
av_log(NULL, AV_LOG_ERROR, "Cannot move '%s' into '%s': %s.\n", src, dst, av_err2str(ret));
return ret;
}
static void usage(const char *program_name)
{
fprintf(stderr, "usage: %s OPERATION entry1 [entry2]\n"
"API example program to show how to manipulate resources "
"accessed through AVIOContext.\n"
"OPERATIONS:\n"
"list list content of the directory\n"
"move rename content in directory\n"
"del delete content in directory\n",
program_name);
}
int main(int argc, char *argv[])
{
const char *op = NULL;
int ret;
av_log_set_level(AV_LOG_DEBUG);
if (argc < 2) {
usage(argv[0]);
return 1;
}
/* register codecs and formats and other lavf/lavc components*/
av_register_all();
avformat_network_init();
op = argv[1];
if (strcmp(op, "list") == 0) {
if (argc < 3) {
av_log(NULL, AV_LOG_INFO, "Missing argument for list operation.\n");
ret = AVERROR(EINVAL);
} else {
ret = list_op(argv[2]);
}
} else if (strcmp(op, "del") == 0) {
if (argc < 3) {
av_log(NULL, AV_LOG_INFO, "Missing argument for del operation.\n");
ret = AVERROR(EINVAL);
} else {
ret = del_op(argv[2]);
}
} else if (strcmp(op, "move") == 0) {
if (argc < 4) {
av_log(NULL, AV_LOG_INFO, "Missing argument for move operation.\n");
ret = AVERROR(EINVAL);
} else {
ret = move_op(argv[2], argv[3]);
}
} else {
av_log(NULL, AV_LOG_INFO, "Invalid operation %s\n", op);
ret = AVERROR(EINVAL);
}
avformat_network_deinit();
return ret < 0 ? 1 : 0;
}

View File

@@ -25,9 +25,9 @@
* libavcodec API use example.
*
* @example decoding_encoding.c
* Note that libavcodec only handles codecs (MPEG, MPEG-4, etc...),
* not file formats (AVI, VOB, MP4, MOV, MKV, MXF, FLV, MPEG-TS, MPEG-PS, etc...).
* See library 'libavformat' for the format handling
* Note that libavcodec only handles codecs (mpeg, mpeg4, etc...),
* not file formats (avi, vob, mp4, mov, mkv, mxf, flv, mpegts, mpegps, etc...). See library 'libavformat' for the
* format handling
*/
#include <math.h>
@@ -211,7 +211,7 @@ static void audio_encode_example(const char *filename)
}
if (got_output) {
fwrite(pkt.data, 1, pkt.size, f);
av_packet_unref(&pkt);
av_free_packet(&pkt);
}
}
@@ -225,7 +225,7 @@ static void audio_encode_example(const char *filename)
if (got_output) {
fwrite(pkt.data, 1, pkt.size, f);
av_packet_unref(&pkt);
av_free_packet(&pkt);
}
}
fclose(f);
@@ -245,7 +245,7 @@ static void audio_decode_example(const char *outfilename, const char *filename)
AVCodecContext *c= NULL;
int len;
FILE *f, *outfile;
uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
AVPacket avpkt;
AVFrame *decoded_frame = NULL;
@@ -253,7 +253,7 @@ static void audio_decode_example(const char *outfilename, const char *filename)
printf("Decode audio file %s to %s\n", filename, outfilename);
/* find the MPEG audio decoder */
/* find the mpeg audio decoder */
codec = avcodec_find_decoder(AV_CODEC_ID_MP2);
if (!codec) {
fprintf(stderr, "Codec not found\n");
@@ -288,7 +288,6 @@ static void audio_decode_example(const char *outfilename, const char *filename)
avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
while (avpkt.size > 0) {
int i, ch;
int got_frame = 0;
if (!decoded_frame) {
@@ -305,15 +304,15 @@ static void audio_decode_example(const char *outfilename, const char *filename)
}
if (got_frame) {
/* if a frame has been decoded, output it */
int data_size = av_get_bytes_per_sample(c->sample_fmt);
int data_size = av_samples_get_buffer_size(NULL, c->channels,
decoded_frame->nb_samples,
c->sample_fmt, 1);
if (data_size < 0) {
/* This should not occur, checking just for paranoia */
fprintf(stderr, "Failed to calculate data size\n");
exit(1);
}
for (i=0; i<decoded_frame->nb_samples; i++)
for (ch=0; ch<c->channels; ch++)
fwrite(decoded_frame->data[ch] + data_size*i, 1, data_size, outfile);
fwrite(decoded_frame->data[0], 1, data_size, outfile);
}
avpkt.size -= len;
avpkt.data += len;
@@ -356,7 +355,7 @@ static void video_encode_example(const char *filename, int codec_id)
printf("Encode video file %s\n", filename);
/* find the video encoder */
/* find the mpeg1 video encoder */
codec = avcodec_find_encoder(codec_id);
if (!codec) {
fprintf(stderr, "Codec not found\n");
@@ -454,7 +453,7 @@ static void video_encode_example(const char *filename, int codec_id)
if (got_output) {
printf("Write frame %3d (size=%5d)\n", i, pkt.size);
fwrite(pkt.data, 1, pkt.size, f);
av_packet_unref(&pkt);
av_free_packet(&pkt);
}
}
@@ -471,11 +470,11 @@ static void video_encode_example(const char *filename, int codec_id)
if (got_output) {
printf("Write frame %3d (size=%5d)\n", i, pkt.size);
fwrite(pkt.data, 1, pkt.size, f);
av_packet_unref(&pkt);
av_free_packet(&pkt);
}
}
/* add sequence end code to have a real MPEG file */
/* add sequence end code to have a real mpeg file */
fwrite(endcode, 1, sizeof(endcode), f);
fclose(f);
@@ -521,7 +520,7 @@ static int decode_write_frame(const char *outfilename, AVCodecContext *avctx,
/* the picture is allocated by the decoder, no need to free it */
snprintf(buf, sizeof(buf), outfilename, *frame_count);
pgm_save(frame->data[0], frame->linesize[0],
frame->width, frame->height, buf);
avctx->width, avctx->height, buf);
(*frame_count)++;
}
if (pkt->data) {
@@ -538,17 +537,17 @@ static void video_decode_example(const char *outfilename, const char *filename)
int frame_count;
FILE *f;
AVFrame *frame;
uint8_t inbuf[INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
uint8_t inbuf[INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
AVPacket avpkt;
av_init_packet(&avpkt);
/* set end of buffer to 0 (this ensures that no overreading happens for damaged MPEG streams) */
memset(inbuf + INBUF_SIZE, 0, AV_INPUT_BUFFER_PADDING_SIZE);
/* set end of buffer to 0 (this ensures that no overreading happens for damaged mpeg streams) */
memset(inbuf + INBUF_SIZE, 0, FF_INPUT_BUFFER_PADDING_SIZE);
printf("Decode video file %s to %s\n", filename, outfilename);
/* find the MPEG-1 video decoder */
/* find the mpeg1 video decoder */
codec = avcodec_find_decoder(AV_CODEC_ID_MPEG1VIDEO);
if (!codec) {
fprintf(stderr, "Codec not found\n");
@@ -561,8 +560,8 @@ static void video_decode_example(const char *outfilename, const char *filename)
exit(1);
}
if (codec->capabilities & AV_CODEC_CAP_TRUNCATED)
c->flags |= AV_CODEC_FLAG_TRUNCATED; // we do not send complete frames
if(codec->capabilities&CODEC_CAP_TRUNCATED)
c->flags|= CODEC_FLAG_TRUNCATED; /* we do not send complete frames */
/* For some codecs, such as msmpeg4 and mpeg4, width and height
MUST be initialized there because this information is not
@@ -613,9 +612,9 @@ static void video_decode_example(const char *outfilename, const char *filename)
exit(1);
}
/* Some codecs, such as MPEG, transmit the I- and P-frame with a
/* some codecs, such as MPEG, transmit the I and P frame with a
latency of one frame. You must do the following to have a
chance to get the last frame of the video. */
chance to get the last frame of the video */
avpkt.data = NULL;
avpkt.size = 0;
decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 1);
@@ -651,7 +650,7 @@ int main(int argc, char **argv)
video_encode_example("test.h264", AV_CODEC_ID_H264);
} else if (!strcmp(output_type, "mp2")) {
audio_encode_example("test.mp2");
audio_decode_example("test.pcm", "test.mp2");
audio_decode_example("test.sw", "test.mp2");
} else if (!strcmp(output_type, "mpg")) {
video_encode_example("test.mpg", AV_CODEC_ID_MPEG1VIDEO);
video_decode_example("test%02d.pgm", "test.mpg");

View File

@@ -36,8 +36,6 @@
static AVFormatContext *fmt_ctx = NULL;
static AVCodecContext *video_dec_ctx = NULL, *audio_dec_ctx;
static int width, height;
static enum AVPixelFormat pix_fmt;
static AVStream *video_stream = NULL, *audio_stream = NULL;
static const char *src_filename = NULL;
static const char *video_dst_filename = NULL;
@@ -55,11 +53,17 @@ static AVPacket pkt;
static int video_frame_count = 0;
static int audio_frame_count = 0;
/* Enable or disable frame reference counting. You are not supposed to support
* both paths in your application but pick the one most appropriate to your
* needs. Look for the use of refcount in this example to see what are the
* differences of API usage between them. */
static int refcount = 0;
/* The different ways of decoding and managing data memory. You are not
* supposed to support all the modes in your application but pick the one most
* appropriate to your needs. Look for the use of api_mode in this example to
* see what are the differences of API usage between them */
enum {
API_MODE_OLD = 0, /* old method, deprecated */
API_MODE_NEW_API_REF_COUNT = 1, /* new method, using the frame reference counting */
API_MODE_NEW_API_NO_REF_COUNT = 2, /* new method, without reference counting */
};
static int api_mode = API_MODE_OLD;
static int decode_packet(int *got_frame, int cached)
{
@@ -77,31 +81,16 @@ static int decode_packet(int *got_frame, int cached)
}
if (*got_frame) {
if (frame->width != width || frame->height != height ||
frame->format != pix_fmt) {
/* To handle this change, one could call av_image_alloc again and
* decode the following frames into another rawvideo file. */
fprintf(stderr, "Error: Width, height and pixel format have to be "
"constant in a rawvideo file, but the width, height or "
"pixel format of the input video changed:\n"
"old: width = %d, height = %d, format = %s\n"
"new: width = %d, height = %d, format = %s\n",
width, height, av_get_pix_fmt_name(pix_fmt),
frame->width, frame->height,
av_get_pix_fmt_name(frame->format));
return -1;
}
printf("video_frame%s n:%d coded_n:%d\n",
printf("video_frame%s n:%d coded_n:%d pts:%s\n",
cached ? "(cached)" : "",
video_frame_count++, frame->coded_picture_number);
video_frame_count++, frame->coded_picture_number,
av_ts2timestr(frame->pts, &video_dec_ctx->time_base));
/* copy decoded frame to destination buffer:
* this is required since rawvideo expects non aligned data */
av_image_copy(video_dst_data, video_dst_linesize,
(const uint8_t **)(frame->data), frame->linesize,
pix_fmt, width, height);
video_dec_ctx->pix_fmt, video_dec_ctx->width, video_dec_ctx->height);
/* write to rawvideo file */
fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file);
@@ -138,19 +127,20 @@ static int decode_packet(int *got_frame, int cached)
}
}
/* If we use frame reference counting, we own the data and need
/* If we use the new API with reference counting, we own the data and need
* to de-reference it when we don't use it anymore */
if (*got_frame && refcount)
if (*got_frame && api_mode == API_MODE_NEW_API_REF_COUNT)
av_frame_unref(frame);
return decoded;
}
static int open_codec_context(int *stream_idx,
AVCodecContext **dec_ctx, AVFormatContext *fmt_ctx, enum AVMediaType type)
AVFormatContext *fmt_ctx, enum AVMediaType type)
{
int ret, stream_index;
int ret;
AVStream *st;
AVCodecContext *dec_ctx = NULL;
AVCodec *dec = NULL;
AVDictionary *opts = NULL;
@@ -160,40 +150,26 @@ static int open_codec_context(int *stream_idx,
av_get_media_type_string(type), src_filename);
return ret;
} else {
stream_index = ret;
st = fmt_ctx->streams[stream_index];
*stream_idx = ret;
st = fmt_ctx->streams[*stream_idx];
/* find decoder for the stream */
dec = avcodec_find_decoder(st->codecpar->codec_id);
dec_ctx = st->codec;
dec = avcodec_find_decoder(dec_ctx->codec_id);
if (!dec) {
fprintf(stderr, "Failed to find %s codec\n",
av_get_media_type_string(type));
return AVERROR(EINVAL);
}
/* Allocate a codec context for the decoder */
*dec_ctx = avcodec_alloc_context3(dec);
if (!*dec_ctx) {
fprintf(stderr, "Failed to allocate the %s codec context\n",
av_get_media_type_string(type));
return AVERROR(ENOMEM);
}
/* Copy codec parameters from input stream to output codec context */
if ((ret = avcodec_parameters_to_context(*dec_ctx, st->codecpar)) < 0) {
fprintf(stderr, "Failed to copy %s codec parameters to decoder context\n",
av_get_media_type_string(type));
return ret;
}
/* Init the decoders, with or without reference counting */
av_dict_set(&opts, "refcounted_frames", refcount ? "1" : "0", 0);
if ((ret = avcodec_open2(*dec_ctx, dec, &opts)) < 0) {
if (api_mode == API_MODE_NEW_API_REF_COUNT)
av_dict_set(&opts, "refcounted_frames", "1", 0);
if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) {
fprintf(stderr, "Failed to open %s codec\n",
av_get_media_type_string(type));
return ret;
}
*stream_idx = stream_index;
}
return 0;
@@ -233,19 +209,28 @@ int main (int argc, char **argv)
int ret = 0, got_frame;
if (argc != 4 && argc != 5) {
fprintf(stderr, "usage: %s [-refcount] input_file video_output_file audio_output_file\n"
fprintf(stderr, "usage: %s [-refcount=<old|new_norefcount|new_refcount>] "
"input_file video_output_file audio_output_file\n"
"API example program to show how to read frames from an input file.\n"
"This program reads frames from a file, decodes them, and writes decoded\n"
"video frames to a rawvideo file named video_output_file, and decoded\n"
"audio frames to a rawaudio file named audio_output_file.\n\n"
"If the -refcount option is specified, the program use the\n"
"reference counting frame system which allows keeping a copy of\n"
"the data for longer than one decode call.\n"
"the data for longer than one decode call. If unset, it's using\n"
"the classic old method.\n"
"\n", argv[0]);
exit(1);
}
if (argc == 5 && !strcmp(argv[1], "-refcount")) {
refcount = 1;
if (argc == 5) {
const char *mode = argv[1] + strlen("-refcount=");
if (!strcmp(mode, "old")) api_mode = API_MODE_OLD;
else if (!strcmp(mode, "new_norefcount")) api_mode = API_MODE_NEW_API_NO_REF_COUNT;
else if (!strcmp(mode, "new_refcount")) api_mode = API_MODE_NEW_API_REF_COUNT;
else {
fprintf(stderr, "unknow mode '%s'\n", mode);
exit(1);
}
argv++;
}
src_filename = argv[1];
@@ -267,8 +252,9 @@ int main (int argc, char **argv)
exit(1);
}
if (open_codec_context(&video_stream_idx, &video_dec_ctx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) {
if (open_codec_context(&video_stream_idx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) {
video_stream = fmt_ctx->streams[video_stream_idx];
video_dec_ctx = video_stream->codec;
video_dst_file = fopen(video_dst_filename, "wb");
if (!video_dst_file) {
@@ -278,11 +264,9 @@ int main (int argc, char **argv)
}
/* allocate image where the decoded image will be put */
width = video_dec_ctx->width;
height = video_dec_ctx->height;
pix_fmt = video_dec_ctx->pix_fmt;
ret = av_image_alloc(video_dst_data, video_dst_linesize,
width, height, pix_fmt, 1);
video_dec_ctx->width, video_dec_ctx->height,
video_dec_ctx->pix_fmt, 1);
if (ret < 0) {
fprintf(stderr, "Could not allocate raw video buffer\n");
goto end;
@@ -290,8 +274,9 @@ int main (int argc, char **argv)
video_dst_bufsize = ret;
}
if (open_codec_context(&audio_stream_idx, &audio_dec_ctx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) {
if (open_codec_context(&audio_stream_idx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) {
audio_stream = fmt_ctx->streams[audio_stream_idx];
audio_dec_ctx = audio_stream->codec;
audio_dst_file = fopen(audio_dst_filename, "wb");
if (!audio_dst_file) {
fprintf(stderr, "Could not open destination file %s\n", audio_dst_filename);
@@ -309,7 +294,12 @@ int main (int argc, char **argv)
goto end;
}
frame = av_frame_alloc();
/* When using the new API, you need to use the libavutil/frame.h API, while
* the classic frame management is available in libavcodec */
if (api_mode == API_MODE_OLD)
frame = avcodec_alloc_frame();
else
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate frame\n");
ret = AVERROR(ENOMEM);
@@ -336,7 +326,7 @@ int main (int argc, char **argv)
pkt.data += ret;
pkt.size -= ret;
} while (pkt.size > 0);
av_packet_unref(&orig_pkt);
av_free_packet(&orig_pkt);
}
/* flush cached frames */
@@ -351,7 +341,7 @@ int main (int argc, char **argv)
if (video_stream) {
printf("Play the output video file with the command:\n"
"ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
av_get_pix_fmt_name(pix_fmt), width, height,
av_get_pix_fmt_name(video_dec_ctx->pix_fmt), video_dec_ctx->width, video_dec_ctx->height,
video_dst_filename);
}
@@ -379,14 +369,17 @@ int main (int argc, char **argv)
}
end:
avcodec_free_context(&video_dec_ctx);
avcodec_free_context(&audio_dec_ctx);
avcodec_close(video_dec_ctx);
avcodec_close(audio_dec_ctx);
avformat_close_input(&fmt_ctx);
if (video_dst_file)
fclose(video_dst_file);
if (audio_dst_file)
fclose(audio_dst_file);
av_frame_free(&frame);
if (api_mode == API_MODE_OLD)
avcodec_free_frame(&frame);
else
av_frame_free(&frame);
av_free(video_dst_data[0]);
return ret < 0;

View File

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

View File

@@ -45,7 +45,6 @@
#include "libavutil/channel_layout.h"
#include "libavutil/md5.h"
#include "libavutil/mem.h"
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"

View File

@@ -33,6 +33,7 @@
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/avcodec.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
@@ -65,7 +66,7 @@ static int open_input_file(const char *filename)
/* select the audio stream */
ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find an audio stream in the input file\n");
av_log(NULL, AV_LOG_ERROR, "Cannot find a audio stream in the input file\n");
return ret;
}
audio_stream_index = ret;
@@ -144,28 +145,12 @@ static int init_filters(const char *filters_descr)
goto end;
}
/*
* Set the endpoints for the filter graph. The filter_graph will
* be linked to the graph described by filters_descr.
*/
/*
* The buffer source output must be connected to the input pad of
* the first filter described by filters_descr; since the first
* filter input label is not specified, it is set to "in" by
* default.
*/
/* Endpoints for the filter graph. */
outputs->name = av_strdup("in");
outputs->filter_ctx = buffersrc_ctx;
outputs->pad_idx = 0;
outputs->next = NULL;
/*
* The buffer sink input must be connected to the output pad of
* the last filter described by filters_descr; since the last
* filter output label is not specified, it is set to "out" by
* default.
*/
inputs->name = av_strdup("out");
inputs->filter_ctx = buffersink_ctx;
inputs->pad_idx = 0;
@@ -273,10 +258,10 @@ int main(int argc, char **argv)
}
if (packet.size <= 0)
av_packet_unref(&packet0);
av_free_packet(&packet0);
} else {
/* discard non-wanted packets */
av_packet_unref(&packet0);
av_free_packet(&packet0);
}
}
end:

View File

@@ -33,14 +33,12 @@
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/avcodec.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
const char *filter_descr = "scale=78:24,transpose=cclock";
/* other way:
scale=78:24 [scl]; [scl] transpose=cclock // assumes "[in]" and "[out]" to be input output pads respectively
*/
const char *filter_descr = "scale=78:24";
static AVFormatContext *fmt_ctx;
static AVCodecContext *dec_ctx;
@@ -92,7 +90,6 @@ static int init_filters(const char *filters_descr)
AVFilter *buffersink = avfilter_get_by_name("buffersink");
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
AVRational time_base = fmt_ctx->streams[video_stream_index]->time_base;
enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE };
filter_graph = avfilter_graph_alloc();
@@ -105,7 +102,7 @@ static int init_filters(const char *filters_descr)
snprintf(args, sizeof(args),
"video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
time_base.num, time_base.den,
dec_ctx->time_base.num, dec_ctx->time_base.den,
dec_ctx->sample_aspect_ratio.num, dec_ctx->sample_aspect_ratio.den);
ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
@@ -130,28 +127,12 @@ static int init_filters(const char *filters_descr)
goto end;
}
/*
* Set the endpoints for the filter graph. The filter_graph will
* be linked to the graph described by filters_descr.
*/
/*
* The buffer source output must be connected to the input pad of
* the first filter described by filters_descr; since the first
* filter input label is not specified, it is set to "in" by
* default.
*/
/* Endpoints for the filter graph. */
outputs->name = av_strdup("in");
outputs->filter_ctx = buffersrc_ctx;
outputs->pad_idx = 0;
outputs->next = NULL;
/*
* The buffer sink input must be connected to the output pad of
* the last filter described by filters_descr; since the last
* filter output label is not specified, it is set to "out" by
* default.
*/
inputs->name = av_strdup("out");
inputs->filter_ctx = buffersink_ctx;
inputs->pad_idx = 0;
@@ -262,7 +243,7 @@ int main(int argc, char **argv)
av_frame_unref(frame);
}
}
av_packet_unref(&packet);
av_free_packet(&packet);
}
end:
avfilter_graph_free(&filter_graph);

View File

@@ -1,155 +0,0 @@
/*
* Copyright (c) 2015 Stephan Holljes
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* libavformat multi-client network API usage example.
*
* @example http_multiclient.c
* This example will serve a file without decoding or demuxing it over http.
* Multiple clients can connect and will receive the same file.
*/
#include <libavformat/avformat.h>
#include <libavutil/opt.h>
#include <unistd.h>
static void process_client(AVIOContext *client, const char *in_uri)
{
AVIOContext *input = NULL;
uint8_t buf[1024];
int ret, n, reply_code;
uint8_t *resource = NULL;
while ((ret = avio_handshake(client)) > 0) {
av_opt_get(client, "resource", AV_OPT_SEARCH_CHILDREN, &resource);
// check for strlen(resource) is necessary, because av_opt_get()
// may return empty string.
if (resource && strlen(resource))
break;
}
if (ret < 0)
goto end;
av_log(client, AV_LOG_TRACE, "resource=%p\n", resource);
if (resource && resource[0] == '/' && !strcmp((resource + 1), in_uri)) {
reply_code = 200;
} else {
reply_code = AVERROR_HTTP_NOT_FOUND;
}
if ((ret = av_opt_set_int(client, "reply_code", reply_code, AV_OPT_SEARCH_CHILDREN)) < 0) {
av_log(client, AV_LOG_ERROR, "Failed to set reply_code: %s.\n", av_err2str(ret));
goto end;
}
av_log(client, AV_LOG_TRACE, "Set reply code to %d\n", reply_code);
while ((ret = avio_handshake(client)) > 0);
if (ret < 0)
goto end;
fprintf(stderr, "Handshake performed.\n");
if (reply_code != 200)
goto end;
fprintf(stderr, "Opening input file.\n");
if ((ret = avio_open2(&input, in_uri, AVIO_FLAG_READ, NULL, NULL)) < 0) {
av_log(input, AV_LOG_ERROR, "Failed to open input: %s: %s.\n", in_uri,
av_err2str(ret));
goto end;
}
for(;;) {
n = avio_read(input, buf, sizeof(buf));
if (n < 0) {
if (n == AVERROR_EOF)
break;
av_log(input, AV_LOG_ERROR, "Error reading from input: %s.\n",
av_err2str(n));
break;
}
avio_write(client, buf, n);
avio_flush(client);
}
end:
fprintf(stderr, "Flushing client\n");
avio_flush(client);
fprintf(stderr, "Closing client\n");
avio_close(client);
fprintf(stderr, "Closing input\n");
avio_close(input);
}
int main(int argc, char **argv)
{
AVDictionary *options = NULL;
AVIOContext *client = NULL, *server = NULL;
const char *in_uri, *out_uri;
int ret, pid;
av_log_set_level(AV_LOG_TRACE);
if (argc < 3) {
printf("usage: %s input http://hostname[:port]\n"
"API example program to serve http to multiple clients.\n"
"\n", argv[0]);
return 1;
}
in_uri = argv[1];
out_uri = argv[2];
av_register_all();
avformat_network_init();
if ((ret = av_dict_set(&options, "listen", "2", 0)) < 0) {
fprintf(stderr, "Failed to set listen mode for server: %s\n", av_err2str(ret));
return ret;
}
if ((ret = avio_open2(&server, out_uri, AVIO_FLAG_WRITE, NULL, &options)) < 0) {
fprintf(stderr, "Failed to open server: %s\n", av_err2str(ret));
return ret;
}
fprintf(stderr, "Entering main loop.\n");
for(;;) {
if ((ret = avio_accept(server, &client)) < 0)
goto end;
fprintf(stderr, "Accepted client, forking process.\n");
// XXX: Since we don't reap our children and don't ignore signals
// this produces zombie processes.
pid = fork();
if (pid < 0) {
perror("Fork failed");
ret = AVERROR(errno);
goto end;
}
if (pid == 0) {
fprintf(stderr, "In child.\n");
process_client(client, in_uri);
avio_close(server);
exit(0);
}
if (pid > 0)
avio_close(client);
}
end:
avio_close(server);
if (ret < 0 && ret != AVERROR_EOF) {
fprintf(stderr, "Some errors occurred: %s\n", av_err2str(ret));
return 1;
}
return 0;
}

View File

@@ -52,11 +52,9 @@
// a wrapper around a single output AVStream
typedef struct OutputStream {
AVStream *st;
AVCodecContext *enc;
/* pts of the next frame that will be generated */
int64_t next_pts;
int samples_count;
AVFrame *frame;
AVFrame *tmp_frame;
@@ -64,7 +62,6 @@ typedef struct OutputStream {
float t, tincr, tincr2;
struct SwsContext *sws_ctx;
struct SwrContext *swr_ctx;
} OutputStream;
static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt)
@@ -95,7 +92,6 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
enum AVCodecID codec_id)
{
AVCodecContext *c;
int i;
/* find the encoder */
*codec = avcodec_find_encoder(codec_id);
@@ -105,18 +101,13 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
exit(1);
}
ost->st = avformat_new_stream(oc, NULL);
ost->st = avformat_new_stream(oc, *codec);
if (!ost->st) {
fprintf(stderr, "Could not allocate stream\n");
exit(1);
}
ost->st->id = oc->nb_streams-1;
c = avcodec_alloc_context3(*codec);
if (!c) {
fprintf(stderr, "Could not alloc an encoding context\n");
exit(1);
}
ost->enc = c;
c = ost->st->codec;
switch ((*codec)->type) {
case AVMEDIA_TYPE_AUDIO:
@@ -124,24 +115,8 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
(*codec)->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
c->bit_rate = 64000;
c->sample_rate = 44100;
if ((*codec)->supported_samplerates) {
c->sample_rate = (*codec)->supported_samplerates[0];
for (i = 0; (*codec)->supported_samplerates[i]; i++) {
if ((*codec)->supported_samplerates[i] == 44100)
c->sample_rate = 44100;
}
}
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
c->channels = 2;
c->channel_layout = AV_CH_LAYOUT_STEREO;
if ((*codec)->channel_layouts) {
c->channel_layout = (*codec)->channel_layouts[0];
for (i = 0; (*codec)->channel_layouts[i]; i++) {
if ((*codec)->channel_layouts[i] == AV_CH_LAYOUT_STEREO)
c->channel_layout = AV_CH_LAYOUT_STEREO;
}
}
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
ost->st->time_base = (AVRational){ 1, c->sample_rate };
break;
case AVMEDIA_TYPE_VIDEO:
@@ -155,13 +130,12 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
* of which frame timestamps are represented. For fixed-fps content,
* timebase should be 1/framerate and timestamp increments should be
* identical to 1. */
ost->st->time_base = (AVRational){ 1, STREAM_FRAME_RATE };
c->time_base = ost->st->time_base;
c->time_base.den = STREAM_FRAME_RATE;
c->time_base.num = 1;
c->gop_size = 12; /* emit one intra frame every twelve frames at most */
c->pix_fmt = STREAM_PIX_FMT;
if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) {
/* just for testing, we also add B-frames */
/* just for testing, we also add B frames */
c->max_b_frames = 2;
}
if (c->codec_id == AV_CODEC_ID_MPEG1VIDEO) {
@@ -178,53 +152,25 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
/* Some formats want stream headers to be separate. */
if (oc->oformat->flags & AVFMT_GLOBALHEADER)
c->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
c->flags |= CODEC_FLAG_GLOBAL_HEADER;
}
/**************************************************************/
/* audio output */
static AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt,
uint64_t channel_layout,
int sample_rate, int nb_samples)
{
AVFrame *frame = av_frame_alloc();
int ret;
int samples_count;
if (!frame) {
fprintf(stderr, "Error allocating an audio frame\n");
exit(1);
}
struct SwrContext *swr_ctx = NULL;
frame->format = sample_fmt;
frame->channel_layout = channel_layout;
frame->sample_rate = sample_rate;
frame->nb_samples = nb_samples;
if (nb_samples) {
ret = av_frame_get_buffer(frame, 0);
if (ret < 0) {
fprintf(stderr, "Error allocating an audio buffer\n");
exit(1);
}
}
return frame;
}
static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost)
{
AVCodecContext *c;
int nb_samples;
int ret;
AVDictionary *opt = NULL;
c = ost->enc;
c = ost->st->codec;
/* open it */
av_dict_copy(&opt, opt_arg, 0);
ret = avcodec_open2(c, codec, &opt);
av_dict_free(&opt);
ret = avcodec_open2(c, codec, NULL);
if (ret < 0) {
fprintf(stderr, "Could not open audio codec: %s\n", av_err2str(ret));
exit(1);
@@ -236,70 +182,95 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, A
/* increment frequency by 110 Hz per second */
ost->tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
if (c->codec->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE)
nb_samples = 10000;
else
nb_samples = c->frame_size;
ost->frame = alloc_audio_frame(c->sample_fmt, c->channel_layout,
c->sample_rate, nb_samples);
ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, c->channel_layout,
c->sample_rate, nb_samples);
/* copy the stream parameters to the muxer */
ret = avcodec_parameters_from_context(ost->st->codecpar, c);
if (ret < 0) {
fprintf(stderr, "Could not copy the stream parameters\n");
ost->frame = av_frame_alloc();
if (!ost->frame)
exit(1);
}
ost->frame->sample_rate = c->sample_rate;
ost->frame->format = AV_SAMPLE_FMT_S16;
ost->frame->channel_layout = c->channel_layout;
if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
ost->frame->nb_samples = 10000;
else
ost->frame->nb_samples = c->frame_size;
ost->tmp_frame = av_frame_alloc();
if (!ost->frame)
exit(1);
ost->tmp_frame->sample_rate = c->sample_rate;
ost->tmp_frame->format = c->sample_fmt;
ost->tmp_frame->channel_layout = c->channel_layout;
ost->tmp_frame->nb_samples = ost->frame->nb_samples;
/* create resampler context */
ost->swr_ctx = swr_alloc();
if (!ost->swr_ctx) {
if (c->sample_fmt != AV_SAMPLE_FMT_S16) {
swr_ctx = swr_alloc();
if (!swr_ctx) {
fprintf(stderr, "Could not allocate resampler context\n");
exit(1);
}
/* set options */
av_opt_set_int (ost->swr_ctx, "in_channel_count", c->channels, 0);
av_opt_set_int (ost->swr_ctx, "in_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(ost->swr_ctx, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
av_opt_set_int (ost->swr_ctx, "out_channel_count", c->channels, 0);
av_opt_set_int (ost->swr_ctx, "out_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(ost->swr_ctx, "out_sample_fmt", c->sample_fmt, 0);
av_opt_set_int (swr_ctx, "in_channel_count", c->channels, 0);
av_opt_set_int (swr_ctx, "in_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
av_opt_set_int (swr_ctx, "out_channel_count", c->channels, 0);
av_opt_set_int (swr_ctx, "out_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", c->sample_fmt, 0);
/* initialize the resampling context */
if ((ret = swr_init(ost->swr_ctx)) < 0) {
if ((ret = swr_init(swr_ctx)) < 0) {
fprintf(stderr, "Failed to initialize the resampling context\n");
exit(1);
}
}
ret = av_frame_get_buffer(ost->frame, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate an audio frame.\n");
exit(1);
}
ret = av_frame_get_buffer(ost->tmp_frame, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate an audio frame.\n");
exit(1);
}
}
/* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
* 'nb_channels' channels. */
static AVFrame *get_audio_frame(OutputStream *ost)
{
AVFrame *frame = ost->tmp_frame;
int j, i, v;
int16_t *q = (int16_t*)frame->data[0];
int j, i, v, ret;
int16_t *q = (int16_t*)ost->frame->data[0];
/* check if we want to generate more frames */
if (av_compare_ts(ost->next_pts, ost->enc->time_base,
if (av_compare_ts(ost->next_pts, ost->st->codec->time_base,
STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
return NULL;
for (j = 0; j <frame->nb_samples; j++) {
/* when we pass a frame to the encoder, it may keep a reference to it
* internally;
* make sure we do not overwrite it here
*/
ret = av_frame_make_writable(ost->frame);
if (ret < 0)
exit(1);
for (j = 0; j < ost->frame->nb_samples; j++) {
v = (int)(sin(ost->t) * 10000);
for (i = 0; i < ost->enc->channels; i++)
for (i = 0; i < ost->st->codec->channels; i++)
*q++ = v;
ost->t += ost->tincr;
ost->tincr += ost->tincr2;
}
frame->pts = ost->next_pts;
ost->next_pts += frame->nb_samples;
ost->frame->pts = ost->next_pts;
ost->next_pts += ost->frame->nb_samples;
return frame;
return ost->frame;
}
/*
@@ -316,37 +287,33 @@ static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
int dst_nb_samples;
av_init_packet(&pkt);
c = ost->enc;
c = ost->st->codec;
frame = get_audio_frame(ost);
if (frame) {
/* convert samples from native format to destination codec format, using the resampler */
if (swr_ctx) {
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples,
dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + frame->nb_samples,
c->sample_rate, c->sample_rate, AV_ROUND_UP);
av_assert0(dst_nb_samples == frame->nb_samples);
/* when we pass a frame to the encoder, it may keep a reference to it
* internally;
* make sure we do not overwrite it here
*/
ret = av_frame_make_writable(ost->frame);
if (ret < 0)
exit(1);
/* convert to destination format */
ret = swr_convert(ost->swr_ctx,
ost->frame->data, dst_nb_samples,
ret = swr_convert(swr_ctx,
ost->tmp_frame->data, dst_nb_samples,
(const uint8_t **)frame->data, frame->nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
exit(1);
}
frame = ost->frame;
frame = ost->tmp_frame;
} else {
dst_nb_samples = frame->nb_samples;
}
frame->pts = av_rescale_q(ost->samples_count, (AVRational){1, c->sample_rate}, c->time_base);
ost->samples_count += dst_nb_samples;
frame->pts = av_rescale_q(samples_count, (AVRational){1, c->sample_rate}, c->time_base);
samples_count += dst_nb_samples;
}
ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
@@ -393,17 +360,13 @@ static AVFrame *alloc_picture(enum AVPixelFormat pix_fmt, int width, int height)
return picture;
}
static void open_video(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
static void open_video(AVFormatContext *oc, AVCodec *codec, OutputStream *ost)
{
int ret;
AVCodecContext *c = ost->enc;
AVDictionary *opt = NULL;
av_dict_copy(&opt, opt_arg, 0);
AVCodecContext *c = ost->st->codec;
/* open the codec */
ret = avcodec_open2(c, codec, &opt);
av_dict_free(&opt);
ret = avcodec_open2(c, codec, NULL);
if (ret < 0) {
fprintf(stderr, "Could not open video codec: %s\n", av_err2str(ret));
exit(1);
@@ -427,13 +390,6 @@ static void open_video(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, A
exit(1);
}
}
/* copy the stream parameters to the muxer */
ret = avcodec_parameters_from_context(ost->st->codecpar, c);
if (ret < 0) {
fprintf(stderr, "Could not copy the stream parameters\n");
exit(1);
}
}
/* Prepare a dummy image. */
@@ -468,10 +424,10 @@ static void fill_yuv_image(AVFrame *pict, int frame_index,
static AVFrame *get_video_frame(OutputStream *ost)
{
AVCodecContext *c = ost->enc;
AVCodecContext *c = ost->st->codec;
/* check if we want to generate more frames */
if (av_compare_ts(ost->next_pts, c->time_base,
if (av_compare_ts(ost->next_pts, ost->st->codec->time_base,
STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
return NULL;
@@ -513,25 +469,44 @@ static int write_video_frame(AVFormatContext *oc, OutputStream *ost)
AVCodecContext *c;
AVFrame *frame;
int got_packet = 0;
AVPacket pkt = { 0 };
c = ost->enc;
c = ost->st->codec;
frame = get_video_frame(ost);
av_init_packet(&pkt);
if (oc->oformat->flags & AVFMT_RAWPICTURE) {
/* a hack to avoid data copy with some raw video muxers */
AVPacket pkt;
av_init_packet(&pkt);
/* encode the image */
ret = avcodec_encode_video2(c, &pkt, frame, &got_packet);
if (ret < 0) {
fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret));
exit(1);
}
if (!frame)
return 1;
if (got_packet) {
ret = write_frame(oc, &c->time_base, ost->st, &pkt);
pkt.flags |= AV_PKT_FLAG_KEY;
pkt.stream_index = ost->st->index;
pkt.data = (uint8_t *)frame;
pkt.size = sizeof(AVPicture);
pkt.pts = pkt.dts = frame->pts;
av_packet_rescale_ts(&pkt, c->time_base, ost->st->time_base);
ret = av_interleaved_write_frame(oc, &pkt);
} else {
ret = 0;
AVPacket pkt = { 0 };
av_init_packet(&pkt);
/* encode the image */
ret = avcodec_encode_video2(c, &pkt, frame, &got_packet);
if (ret < 0) {
fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret));
exit(1);
}
if (got_packet) {
ret = write_frame(oc, &c->time_base, ost->st, &pkt);
} else {
ret = 0;
}
}
if (ret < 0) {
@@ -544,11 +519,10 @@ static int write_video_frame(AVFormatContext *oc, OutputStream *ost)
static void close_stream(AVFormatContext *oc, OutputStream *ost)
{
avcodec_free_context(&ost->enc);
avcodec_close(ost->st->codec);
av_frame_free(&ost->frame);
av_frame_free(&ost->tmp_frame);
sws_freeContext(ost->sws_ctx);
swr_free(&ost->swr_ctx);
}
/**************************************************************/
@@ -564,13 +538,11 @@ int main(int argc, char **argv)
int ret;
int have_video = 0, have_audio = 0;
int encode_video = 0, encode_audio = 0;
AVDictionary *opt = NULL;
int i;
/* Initialize libavcodec, and register all codecs and formats. */
av_register_all();
if (argc < 2) {
if (argc != 2) {
printf("usage: %s output_file\n"
"API example program to output a media file with libavformat.\n"
"This program generates a synthetic audio and video stream, encodes and\n"
@@ -582,10 +554,6 @@ int main(int argc, char **argv)
}
filename = argv[1];
for (i = 2; i+1 < argc; i+=2) {
if (!strcmp(argv[i], "-flags") || !strcmp(argv[i], "-fflags"))
av_dict_set(&opt, argv[i]+1, argv[i+1], 0);
}
/* allocate the output media context */
avformat_alloc_output_context2(&oc, NULL, NULL, filename);
@@ -614,10 +582,10 @@ int main(int argc, char **argv)
/* Now that all the parameters are set, we can open the audio and
* video codecs and allocate the necessary encode buffers. */
if (have_video)
open_video(oc, video_codec, &video_st, opt);
open_video(oc, video_codec, &video_st);
if (have_audio)
open_audio(oc, audio_codec, &audio_st, opt);
open_audio(oc, audio_codec, &audio_st);
av_dump_format(oc, 0, filename, 1);
@@ -632,7 +600,7 @@ int main(int argc, char **argv)
}
/* Write the stream header, if any. */
ret = avformat_write_header(oc, &opt);
ret = avformat_write_header(oc, NULL);
if (ret < 0) {
fprintf(stderr, "Error occurred when opening output file: %s\n",
av_err2str(ret));
@@ -642,8 +610,8 @@ int main(int argc, char **argv)
while (encode_video || encode_audio) {
/* select the stream to encode */
if (encode_video &&
(!encode_audio || av_compare_ts(video_st.next_pts, video_st.enc->time_base,
audio_st.next_pts, audio_st.enc->time_base) <= 0)) {
(!encode_audio || av_compare_ts(video_st.next_pts, video_st.st->codec->time_base,
audio_st.next_pts, audio_st.st->codec->time_base) <= 0)) {
encode_video = !write_video_frame(oc, &video_st);
} else {
encode_audio = !write_audio_frame(oc, &audio_st);
@@ -664,7 +632,7 @@ int main(int argc, char **argv)
if (!(fmt->flags & AVFMT_NOFILE))
/* Close the output file. */
avio_closep(&oc->pb);
avio_close(oc->pb);
/* free the stream */
avformat_free_context(oc);

View File

@@ -1,487 +0,0 @@
/*
* Copyright (c) 2015 Anton Khirnov
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* Intel QSV-accelerated H.264 decoding example.
*
* @example qsvdec.c
* This example shows how to do QSV-accelerated H.264 decoding with output
* frames in the VA-API video surfaces.
*/
#include "config.h"
#include <stdio.h>
#include <mfx/mfxvideo.h>
#include <va/va.h>
#include <va/va_x11.h>
#include <X11/Xlib.h>
#include "libavformat/avformat.h"
#include "libavformat/avio.h"
#include "libavcodec/avcodec.h"
#include "libavcodec/qsv.h"
#include "libavutil/error.h"
#include "libavutil/mem.h"
typedef struct DecodeContext {
mfxSession mfx_session;
VADisplay va_dpy;
VASurfaceID *surfaces;
mfxMemId *surface_ids;
int *surface_used;
int nb_surfaces;
mfxFrameInfo frame_info;
} DecodeContext;
static mfxStatus frame_alloc(mfxHDL pthis, mfxFrameAllocRequest *req,
mfxFrameAllocResponse *resp)
{
DecodeContext *decode = pthis;
int err, i;
if (decode->surfaces) {
fprintf(stderr, "Multiple allocation requests.\n");
return MFX_ERR_MEMORY_ALLOC;
}
if (!(req->Type & MFX_MEMTYPE_VIDEO_MEMORY_DECODER_TARGET)) {
fprintf(stderr, "Unsupported surface type: %d\n", req->Type);
return MFX_ERR_UNSUPPORTED;
}
if (req->Info.BitDepthLuma != 8 || req->Info.BitDepthChroma != 8 ||
req->Info.Shift || req->Info.FourCC != MFX_FOURCC_NV12 ||
req->Info.ChromaFormat != MFX_CHROMAFORMAT_YUV420) {
fprintf(stderr, "Unsupported surface properties.\n");
return MFX_ERR_UNSUPPORTED;
}
decode->surfaces = av_malloc_array (req->NumFrameSuggested, sizeof(*decode->surfaces));
decode->surface_ids = av_malloc_array (req->NumFrameSuggested, sizeof(*decode->surface_ids));
decode->surface_used = av_mallocz_array(req->NumFrameSuggested, sizeof(*decode->surface_used));
if (!decode->surfaces || !decode->surface_ids || !decode->surface_used)
goto fail;
err = vaCreateSurfaces(decode->va_dpy, VA_RT_FORMAT_YUV420,
req->Info.Width, req->Info.Height,
decode->surfaces, req->NumFrameSuggested,
NULL, 0);
if (err != VA_STATUS_SUCCESS) {
fprintf(stderr, "Error allocating VA surfaces\n");
goto fail;
}
decode->nb_surfaces = req->NumFrameSuggested;
for (i = 0; i < decode->nb_surfaces; i++)
decode->surface_ids[i] = &decode->surfaces[i];
resp->mids = decode->surface_ids;
resp->NumFrameActual = decode->nb_surfaces;
decode->frame_info = req->Info;
return MFX_ERR_NONE;
fail:
av_freep(&decode->surfaces);
av_freep(&decode->surface_ids);
av_freep(&decode->surface_used);
return MFX_ERR_MEMORY_ALLOC;
}
static mfxStatus frame_free(mfxHDL pthis, mfxFrameAllocResponse *resp)
{
return MFX_ERR_NONE;
}
static mfxStatus frame_lock(mfxHDL pthis, mfxMemId mid, mfxFrameData *ptr)
{
return MFX_ERR_UNSUPPORTED;
}
static mfxStatus frame_unlock(mfxHDL pthis, mfxMemId mid, mfxFrameData *ptr)
{
return MFX_ERR_UNSUPPORTED;
}
static mfxStatus frame_get_hdl(mfxHDL pthis, mfxMemId mid, mfxHDL *hdl)
{
*hdl = mid;
return MFX_ERR_NONE;
}
static void free_surfaces(DecodeContext *decode)
{
if (decode->surfaces)
vaDestroySurfaces(decode->va_dpy, decode->surfaces, decode->nb_surfaces);
av_freep(&decode->surfaces);
av_freep(&decode->surface_ids);
av_freep(&decode->surface_used);
decode->nb_surfaces = 0;
}
static void free_buffer(void *opaque, uint8_t *data)
{
int *used = opaque;
*used = 0;
av_freep(&data);
}
static int get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
{
DecodeContext *decode = avctx->opaque;
mfxFrameSurface1 *surf;
AVBufferRef *surf_buf;
int idx;
for (idx = 0; idx < decode->nb_surfaces; idx++) {
if (!decode->surface_used[idx])
break;
}
if (idx == decode->nb_surfaces) {
fprintf(stderr, "No free surfaces\n");
return AVERROR(ENOMEM);
}
surf = av_mallocz(sizeof(*surf));
if (!surf)
return AVERROR(ENOMEM);
surf_buf = av_buffer_create((uint8_t*)surf, sizeof(*surf), free_buffer,
&decode->surface_used[idx], AV_BUFFER_FLAG_READONLY);
if (!surf_buf) {
av_freep(&surf);
return AVERROR(ENOMEM);
}
surf->Info = decode->frame_info;
surf->Data.MemId = &decode->surfaces[idx];
frame->buf[0] = surf_buf;
frame->data[3] = (uint8_t*)surf;
decode->surface_used[idx] = 1;
return 0;
}
static int get_format(AVCodecContext *avctx, const enum AVPixelFormat *pix_fmts)
{
while (*pix_fmts != AV_PIX_FMT_NONE) {
if (*pix_fmts == AV_PIX_FMT_QSV) {
if (!avctx->hwaccel_context) {
DecodeContext *decode = avctx->opaque;
AVQSVContext *qsv = av_qsv_alloc_context();
if (!qsv)
return AV_PIX_FMT_NONE;
qsv->session = decode->mfx_session;
qsv->iopattern = MFX_IOPATTERN_OUT_VIDEO_MEMORY;
avctx->hwaccel_context = qsv;
}
return AV_PIX_FMT_QSV;
}
pix_fmts++;
}
fprintf(stderr, "The QSV pixel format not offered in get_format()\n");
return AV_PIX_FMT_NONE;
}
static int decode_packet(DecodeContext *decode, AVCodecContext *decoder_ctx,
AVFrame *frame, AVPacket *pkt,
AVIOContext *output_ctx)
{
int ret = 0;
int got_frame = 1;
while (pkt->size > 0 || (!pkt->data && got_frame)) {
ret = avcodec_decode_video2(decoder_ctx, frame, &got_frame, pkt);
if (ret < 0) {
fprintf(stderr, "Error during decoding\n");
return ret;
}
pkt->data += ret;
pkt->size -= ret;
/* A real program would do something useful with the decoded frame here.
* We just retrieve the raw data and write it to a file, which is rather
* useless but pedagogic. */
if (got_frame) {
mfxFrameSurface1 *surf = (mfxFrameSurface1*)frame->data[3];
VASurfaceID surface = *(VASurfaceID*)surf->Data.MemId;
VAImageFormat img_fmt = {
.fourcc = VA_FOURCC_NV12,
.byte_order = VA_LSB_FIRST,
.bits_per_pixel = 8,
.depth = 8,
};
VAImage img;
VAStatus err;
uint8_t *data;
int i, j;
img.buf = VA_INVALID_ID;
img.image_id = VA_INVALID_ID;
err = vaCreateImage(decode->va_dpy, &img_fmt,
frame->width, frame->height, &img);
if (err != VA_STATUS_SUCCESS) {
fprintf(stderr, "Error creating an image: %s\n",
vaErrorStr(err));
ret = AVERROR_UNKNOWN;
goto fail;
}
err = vaGetImage(decode->va_dpy, surface, 0, 0,
frame->width, frame->height,
img.image_id);
if (err != VA_STATUS_SUCCESS) {
fprintf(stderr, "Error getting an image: %s\n",
vaErrorStr(err));
ret = AVERROR_UNKNOWN;
goto fail;
}
err = vaMapBuffer(decode->va_dpy, img.buf, (void**)&data);
if (err != VA_STATUS_SUCCESS) {
fprintf(stderr, "Error mapping the image buffer: %s\n",
vaErrorStr(err));
ret = AVERROR_UNKNOWN;
goto fail;
}
for (i = 0; i < img.num_planes; i++)
for (j = 0; j < (img.height >> (i > 0)); j++)
avio_write(output_ctx, data + img.offsets[i] + j * img.pitches[i], img.width);
fail:
if (img.buf != VA_INVALID_ID)
vaUnmapBuffer(decode->va_dpy, img.buf);
if (img.image_id != VA_INVALID_ID)
vaDestroyImage(decode->va_dpy, img.image_id);
av_frame_unref(frame);
if (ret < 0)
return ret;
}
}
return 0;
}
int main(int argc, char **argv)
{
AVFormatContext *input_ctx = NULL;
AVStream *video_st = NULL;
AVCodecContext *decoder_ctx = NULL;
const AVCodec *decoder;
AVPacket pkt = { 0 };
AVFrame *frame = NULL;
DecodeContext decode = { NULL };
Display *dpy = NULL;
int va_ver_major, va_ver_minor;
mfxIMPL mfx_impl = MFX_IMPL_AUTO_ANY;
mfxVersion mfx_ver = { { 1, 1 } };
mfxFrameAllocator frame_allocator = {
.pthis = &decode,
.Alloc = frame_alloc,
.Lock = frame_lock,
.Unlock = frame_unlock,
.GetHDL = frame_get_hdl,
.Free = frame_free,
};
AVIOContext *output_ctx = NULL;
int ret, i, err;
av_register_all();
if (argc < 3) {
fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
return 1;
}
/* open the input file */
ret = avformat_open_input(&input_ctx, argv[1], NULL, NULL);
if (ret < 0) {
fprintf(stderr, "Cannot open input file '%s': ", argv[1]);
goto finish;
}
/* find the first H.264 video stream */
for (i = 0; i < input_ctx->nb_streams; i++) {
AVStream *st = input_ctx->streams[i];
if (st->codecpar->codec_id == AV_CODEC_ID_H264 && !video_st)
video_st = st;
else
st->discard = AVDISCARD_ALL;
}
if (!video_st) {
fprintf(stderr, "No H.264 video stream in the input file\n");
goto finish;
}
/* initialize VA-API */
dpy = XOpenDisplay(NULL);
if (!dpy) {
fprintf(stderr, "Cannot open the X display\n");
goto finish;
}
decode.va_dpy = vaGetDisplay(dpy);
if (!decode.va_dpy) {
fprintf(stderr, "Cannot open the VA display\n");
goto finish;
}
err = vaInitialize(decode.va_dpy, &va_ver_major, &va_ver_minor);
if (err != VA_STATUS_SUCCESS) {
fprintf(stderr, "Cannot initialize VA: %s\n", vaErrorStr(err));
goto finish;
}
fprintf(stderr, "Initialized VA v%d.%d\n", va_ver_major, va_ver_minor);
/* initialize an MFX session */
err = MFXInit(mfx_impl, &mfx_ver, &decode.mfx_session);
if (err != MFX_ERR_NONE) {
fprintf(stderr, "Error initializing an MFX session\n");
goto finish;
}
MFXVideoCORE_SetHandle(decode.mfx_session, MFX_HANDLE_VA_DISPLAY, decode.va_dpy);
MFXVideoCORE_SetFrameAllocator(decode.mfx_session, &frame_allocator);
/* initialize the decoder */
decoder = avcodec_find_decoder_by_name("h264_qsv");
if (!decoder) {
fprintf(stderr, "The QSV decoder is not present in libavcodec\n");
goto finish;
}
decoder_ctx = avcodec_alloc_context3(decoder);
if (!decoder_ctx) {
ret = AVERROR(ENOMEM);
goto finish;
}
decoder_ctx->codec_id = AV_CODEC_ID_H264;
if (video_st->codecpar->extradata_size) {
decoder_ctx->extradata = av_mallocz(video_st->codecpar->extradata_size +
AV_INPUT_BUFFER_PADDING_SIZE);
if (!decoder_ctx->extradata) {
ret = AVERROR(ENOMEM);
goto finish;
}
memcpy(decoder_ctx->extradata, video_st->codecpar->extradata,
video_st->codecpar->extradata_size);
decoder_ctx->extradata_size = video_st->codecpar->extradata_size;
}
decoder_ctx->refcounted_frames = 1;
decoder_ctx->opaque = &decode;
decoder_ctx->get_buffer2 = get_buffer;
decoder_ctx->get_format = get_format;
ret = avcodec_open2(decoder_ctx, NULL, NULL);
if (ret < 0) {
fprintf(stderr, "Error opening the decoder: ");
goto finish;
}
/* open the output stream */
ret = avio_open(&output_ctx, argv[2], AVIO_FLAG_WRITE);
if (ret < 0) {
fprintf(stderr, "Error opening the output context: ");
goto finish;
}
frame = av_frame_alloc();
if (!frame) {
ret = AVERROR(ENOMEM);
goto finish;
}
/* actual decoding */
while (ret >= 0) {
ret = av_read_frame(input_ctx, &pkt);
if (ret < 0)
break;
if (pkt.stream_index == video_st->index)
ret = decode_packet(&decode, decoder_ctx, frame, &pkt, output_ctx);
av_packet_unref(&pkt);
}
/* flush the decoder */
pkt.data = NULL;
pkt.size = 0;
ret = decode_packet(&decode, decoder_ctx, frame, &pkt, output_ctx);
finish:
if (ret < 0) {
char buf[1024];
av_strerror(ret, buf, sizeof(buf));
fprintf(stderr, "%s\n", buf);
}
avformat_close_input(&input_ctx);
av_frame_free(&frame);
if (decoder_ctx)
av_freep(&decoder_ctx->hwaccel_context);
avcodec_free_context(&decoder_ctx);
free_surfaces(&decode);
if (decode.mfx_session)
MFXClose(decode.mfx_session);
if (decode.va_dpy)
vaTerminate(decode.va_dpy);
if (dpy)
XCloseDisplay(dpy);
avio_close(output_ctx);
return ret;
}

View File

@@ -101,7 +101,7 @@ int main(int argc, char **argv)
}
out_stream->codec->codec_tag = 0;
if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
out_stream->codec->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
out_stream->codec->flags |= CODEC_FLAG_GLOBAL_HEADER;
}
av_dump_format(ofmt_ctx, 0, out_filename, 1);
@@ -143,7 +143,7 @@ int main(int argc, char **argv)
fprintf(stderr, "Error muxing packet\n");
break;
}
av_packet_unref(&pkt);
av_free_packet(&pkt);
}
av_write_trailer(ofmt_ctx);
@@ -153,7 +153,7 @@ end:
/* close output */
if (ofmt_ctx && !(ofmt->flags & AVFMT_NOFILE))
avio_closep(&ofmt_ctx->pb);
avio_close(ofmt_ctx->pb);
avformat_free_context(ofmt_ctx);
if (ret < 0 && ret != AVERROR_EOF) {

View File

@@ -199,7 +199,8 @@ int main(int argc, char **argv)
fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);
end:
fclose(dst_file);
if (dst_file)
fclose(dst_file);
if (src_data)
av_freep(&src_data[0]);

View File

@@ -132,7 +132,8 @@ int main(int argc, char **argv)
av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h, dst_filename);
end:
fclose(dst_file);
if (dst_file)
fclose(dst_file);
av_freep(&src_data[0]);
av_freep(&dst_data[0]);
sws_freeContext(sws_ctx);

View File

@@ -41,16 +41,18 @@
#include "libswresample/swresample.h"
/** The output bit rate in kbit/s */
#define OUTPUT_BIT_RATE 96000
#define OUTPUT_BIT_RATE 48000
/** The number of output channels */
#define OUTPUT_CHANNELS 2
/** The audio sample output format */
#define OUTPUT_SAMPLE_FORMAT AV_SAMPLE_FMT_S16
/**
* Convert an error code into a text message.
* @param error Error code to be converted
* @return Corresponding error text (not thread-safe)
*/
static const char *get_error_text(const int error)
static char *const get_error_text(const int error)
{
static char error_buffer[255];
av_strerror(error, error_buffer, sizeof(error_buffer));
@@ -62,7 +64,6 @@ static int open_input_file(const char *filename,
AVFormatContext **input_format_context,
AVCodecContext **input_codec_context)
{
AVCodecContext *avctx;
AVCodec *input_codec;
int error;
@@ -92,39 +93,23 @@ static int open_input_file(const char *filename,
}
/** Find a decoder for the audio stream. */
if (!(input_codec = avcodec_find_decoder((*input_format_context)->streams[0]->codecpar->codec_id))) {
if (!(input_codec = avcodec_find_decoder((*input_format_context)->streams[0]->codec->codec_id))) {
fprintf(stderr, "Could not find input codec\n");
avformat_close_input(input_format_context);
return AVERROR_EXIT;
}
/** allocate a new decoding context */
avctx = avcodec_alloc_context3(input_codec);
if (!avctx) {
fprintf(stderr, "Could not allocate a decoding context\n");
avformat_close_input(input_format_context);
return AVERROR(ENOMEM);
}
/** initialize the stream parameters with demuxer information */
error = avcodec_parameters_to_context(avctx, (*input_format_context)->streams[0]->codecpar);
if (error < 0) {
avformat_close_input(input_format_context);
avcodec_free_context(&avctx);
return error;
}
/** Open the decoder for the audio stream to use it later. */
if ((error = avcodec_open2(avctx, input_codec, NULL)) < 0) {
if ((error = avcodec_open2((*input_format_context)->streams[0]->codec,
input_codec, NULL)) < 0) {
fprintf(stderr, "Could not open input codec (error '%s')\n",
get_error_text(error));
avcodec_free_context(&avctx);
avformat_close_input(input_format_context);
return error;
}
/** Save the decoder context for easier access later. */
*input_codec_context = avctx;
*input_codec_context = (*input_format_context)->streams[0]->codec;
return 0;
}
@@ -139,7 +124,6 @@ static int open_output_file(const char *filename,
AVFormatContext **output_format_context,
AVCodecContext **output_codec_context)
{
AVCodecContext *avctx = NULL;
AVIOContext *output_io_context = NULL;
AVStream *stream = NULL;
AVCodec *output_codec = NULL;
@@ -179,64 +163,43 @@ static int open_output_file(const char *filename,
}
/** Create a new audio stream in the output file container. */
if (!(stream = avformat_new_stream(*output_format_context, NULL))) {
if (!(stream = avformat_new_stream(*output_format_context, output_codec))) {
fprintf(stderr, "Could not create new stream\n");
error = AVERROR(ENOMEM);
goto cleanup;
}
avctx = avcodec_alloc_context3(output_codec);
if (!avctx) {
fprintf(stderr, "Could not allocate an encoding context\n");
error = AVERROR(ENOMEM);
goto cleanup;
}
/** Save the encoder context for easiert access later. */
*output_codec_context = stream->codec;
/**
* Set the basic encoder parameters.
* The input file's sample rate is used to avoid a sample rate conversion.
*/
avctx->channels = OUTPUT_CHANNELS;
avctx->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
avctx->sample_rate = input_codec_context->sample_rate;
avctx->sample_fmt = output_codec->sample_fmts[0];
avctx->bit_rate = OUTPUT_BIT_RATE;
/** Allow the use of the experimental AAC encoder */
avctx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
/** Set the sample rate for the container. */
stream->time_base.den = input_codec_context->sample_rate;
stream->time_base.num = 1;
(*output_codec_context)->channels = OUTPUT_CHANNELS;
(*output_codec_context)->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
(*output_codec_context)->sample_rate = input_codec_context->sample_rate;
(*output_codec_context)->sample_fmt = AV_SAMPLE_FMT_S16;
(*output_codec_context)->bit_rate = OUTPUT_BIT_RATE;
/**
* Some container formats (like MP4) require global headers to be present
* Mark the encoder so that it behaves accordingly.
*/
if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
avctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
(*output_codec_context)->flags |= CODEC_FLAG_GLOBAL_HEADER;
/** Open the encoder for the audio stream to use it later. */
if ((error = avcodec_open2(avctx, output_codec, NULL)) < 0) {
if ((error = avcodec_open2(*output_codec_context, output_codec, NULL)) < 0) {
fprintf(stderr, "Could not open output codec (error '%s')\n",
get_error_text(error));
goto cleanup;
}
error = avcodec_parameters_from_context(stream->codecpar, avctx);
if (error < 0) {
fprintf(stderr, "Could not initialize stream parameters\n");
goto cleanup;
}
/** Save the encoder context for easier access later. */
*output_codec_context = avctx;
return 0;
cleanup:
avcodec_free_context(&avctx);
avio_closep(&(*output_format_context)->pb);
avio_close((*output_format_context)->pb);
avformat_free_context(*output_format_context);
*output_format_context = NULL;
return error < 0 ? error : AVERROR_EXIT;
@@ -308,11 +271,10 @@ static int init_resampler(AVCodecContext *input_codec_context,
}
/** Initialize a FIFO buffer for the audio samples to be encoded. */
static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
static int init_fifo(AVAudioFifo **fifo)
{
/** Create the FIFO buffer based on the specified output sample format. */
if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
output_codec_context->channels, 1))) {
if (!(*fifo = av_audio_fifo_alloc(OUTPUT_SAMPLE_FORMAT, OUTPUT_CHANNELS, 1))) {
fprintf(stderr, "Could not allocate FIFO\n");
return AVERROR(ENOMEM);
}
@@ -344,7 +306,7 @@ static int decode_audio_frame(AVFrame *frame,
/** Read one audio frame from the input file into a temporary packet. */
if ((error = av_read_frame(input_format_context, &input_packet)) < 0) {
/** If we are at the end of the file, flush the decoder below. */
/** If we are the the end of the file, flush the decoder below. */
if (error == AVERROR_EOF)
*finished = 1;
else {
@@ -364,7 +326,7 @@ static int decode_audio_frame(AVFrame *frame,
data_present, &input_packet)) < 0) {
fprintf(stderr, "Could not decode frame (error '%s')\n",
get_error_text(error));
av_packet_unref(&input_packet);
av_free_packet(&input_packet);
return error;
}
@@ -374,7 +336,7 @@ static int decode_audio_frame(AVFrame *frame,
*/
if (*finished && *data_present)
*finished = 0;
av_packet_unref(&input_packet);
av_free_packet(&input_packet);
return 0;
}
@@ -575,9 +537,6 @@ static int init_output_frame(AVFrame **frame,
return 0;
}
/** Global timestamp for the audio frames */
static int64_t pts = 0;
/** Encode one frame worth of audio to the output file. */
static int encode_audio_frame(AVFrame *frame,
AVFormatContext *output_format_context,
@@ -589,12 +548,6 @@ static int encode_audio_frame(AVFrame *frame,
int error;
init_packet(&output_packet);
/** Set a timestamp based on the sample rate for the container. */
if (frame) {
frame->pts = pts;
pts += frame->nb_samples;
}
/**
* Encode the audio frame and store it in the temporary packet.
* The output audio stream encoder is used to do this.
@@ -603,7 +556,7 @@ static int encode_audio_frame(AVFrame *frame,
frame, data_present)) < 0) {
fprintf(stderr, "Could not encode frame (error '%s')\n",
get_error_text(error));
av_packet_unref(&output_packet);
av_free_packet(&output_packet);
return error;
}
@@ -612,11 +565,11 @@ static int encode_audio_frame(AVFrame *frame,
if ((error = av_write_frame(output_format_context, &output_packet)) < 0) {
fprintf(stderr, "Could not write frame (error '%s')\n",
get_error_text(error));
av_packet_unref(&output_packet);
av_free_packet(&output_packet);
return error;
}
av_packet_unref(&output_packet);
av_free_packet(&output_packet);
}
return 0;
@@ -706,7 +659,7 @@ int main(int argc, char **argv)
&resample_context))
goto cleanup;
/** Initialize the FIFO buffer to store audio samples to be encoded. */
if (init_fifo(&fifo, output_codec_context))
if (init_fifo(&fifo))
goto cleanup;
/** Write the header of the output file container. */
if (write_output_file_header(output_format_context))
@@ -788,13 +741,13 @@ cleanup:
av_audio_fifo_free(fifo);
swr_free(&resample_context);
if (output_codec_context)
avcodec_free_context(&output_codec_context);
avcodec_close(output_codec_context);
if (output_format_context) {
avio_closep(&output_format_context->pb);
avio_close(output_format_context->pb);
avformat_free_context(output_format_context);
}
if (input_codec_context)
avcodec_free_context(&input_codec_context);
avcodec_close(input_codec_context);
if (input_format_context)
avformat_close_input(&input_format_context);

View File

@@ -31,6 +31,7 @@
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/avcodec.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
@@ -115,10 +116,6 @@ static int open_output_file(const char *filename)
|| dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
/* in this example, we choose transcoding to same codec */
encoder = avcodec_find_encoder(dec_ctx->codec_id);
if (!encoder) {
av_log(NULL, AV_LOG_FATAL, "Necessary encoder not found\n");
return AVERROR_INVALIDDATA;
}
/* In this example, we transcode to same properties (picture size,
* sample rate etc.). These properties can be changed for output
@@ -128,10 +125,7 @@ static int open_output_file(const char *filename)
enc_ctx->width = dec_ctx->width;
enc_ctx->sample_aspect_ratio = dec_ctx->sample_aspect_ratio;
/* take first format from list of supported formats */
if (encoder->pix_fmts)
enc_ctx->pix_fmt = encoder->pix_fmts[0];
else
enc_ctx->pix_fmt = dec_ctx->pix_fmt;
enc_ctx->pix_fmt = encoder->pix_fmts[0];
/* video time_base can be set to whatever is handy and supported by encoder */
enc_ctx->time_base = dec_ctx->time_base;
} else {
@@ -163,7 +157,7 @@ static int open_output_file(const char *filename)
}
if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
enc_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
enc_ctx->flags |= CODEC_FLAG_GLOBAL_HEADER;
}
av_dump_format(ofmt_ctx, 0, filename, 1);
@@ -391,9 +385,17 @@ static int encode_write_frame(AVFrame *filt_frame, unsigned int stream_index, in
/* prepare packet for muxing */
enc_pkt.stream_index = stream_index;
av_packet_rescale_ts(&enc_pkt,
ofmt_ctx->streams[stream_index]->codec->time_base,
ofmt_ctx->streams[stream_index]->time_base);
enc_pkt.dts = av_rescale_q_rnd(enc_pkt.dts,
ofmt_ctx->streams[stream_index]->codec->time_base,
ofmt_ctx->streams[stream_index]->time_base,
AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
enc_pkt.pts = av_rescale_q_rnd(enc_pkt.pts,
ofmt_ctx->streams[stream_index]->codec->time_base,
ofmt_ctx->streams[stream_index]->time_base,
AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
enc_pkt.duration = av_rescale_q(enc_pkt.duration,
ofmt_ctx->streams[stream_index]->codec->time_base,
ofmt_ctx->streams[stream_index]->time_base);
av_log(NULL, AV_LOG_DEBUG, "Muxing frame\n");
/* mux encoded frame */
@@ -451,7 +453,7 @@ static int flush_encoder(unsigned int stream_index)
int got_frame;
if (!(ofmt_ctx->streams[stream_index]->codec->codec->capabilities &
AV_CODEC_CAP_DELAY))
CODEC_CAP_DELAY))
return 0;
while (1) {
@@ -507,9 +509,14 @@ int main(int argc, char **argv)
ret = AVERROR(ENOMEM);
break;
}
av_packet_rescale_ts(&packet,
ifmt_ctx->streams[stream_index]->time_base,
ifmt_ctx->streams[stream_index]->codec->time_base);
packet.dts = av_rescale_q_rnd(packet.dts,
ifmt_ctx->streams[stream_index]->time_base,
ifmt_ctx->streams[stream_index]->codec->time_base,
AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
packet.pts = av_rescale_q_rnd(packet.pts,
ifmt_ctx->streams[stream_index]->time_base,
ifmt_ctx->streams[stream_index]->codec->time_base,
AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
dec_func = (type == AVMEDIA_TYPE_VIDEO) ? avcodec_decode_video2 :
avcodec_decode_audio4;
ret = dec_func(ifmt_ctx->streams[stream_index]->codec, frame,
@@ -531,15 +538,20 @@ int main(int argc, char **argv)
}
} else {
/* remux this frame without reencoding */
av_packet_rescale_ts(&packet,
ifmt_ctx->streams[stream_index]->time_base,
ofmt_ctx->streams[stream_index]->time_base);
packet.dts = av_rescale_q_rnd(packet.dts,
ifmt_ctx->streams[stream_index]->time_base,
ofmt_ctx->streams[stream_index]->time_base,
AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
packet.pts = av_rescale_q_rnd(packet.pts,
ifmt_ctx->streams[stream_index]->time_base,
ofmt_ctx->streams[stream_index]->time_base,
AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
ret = av_interleaved_write_frame(ofmt_ctx, &packet);
if (ret < 0)
goto end;
}
av_packet_unref(&packet);
av_free_packet(&packet);
}
/* flush filters and encoders */
@@ -563,7 +575,7 @@ int main(int argc, char **argv)
av_write_trailer(ofmt_ctx);
end:
av_packet_unref(&packet);
av_free_packet(&packet);
av_frame_free(&frame);
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
avcodec_close(ifmt_ctx->streams[i]->codec);
@@ -575,7 +587,7 @@ end:
av_free(filter_ctx);
avformat_close_input(&ifmt_ctx);
if (ofmt_ctx && !(ofmt_ctx->oformat->flags & AVFMT_NOFILE))
avio_closep(&ofmt_ctx->pb);
avio_close(ofmt_ctx->pb);
avformat_free_context(ofmt_ctx);
if (ret < 0)

View File

@@ -1,5 +1,4 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg FAQ
@titlepage
@@ -91,63 +90,13 @@ To build FFmpeg, you need to install the development package. It is usually
called @file{libfoo-dev} or @file{libfoo-devel}. You can remove it after the
build is finished, but be sure to keep the main package.
@section How do I make @command{pkg-config} find my libraries?
Somewhere along with your libraries, there is a @file{.pc} file (or several)
in a @file{pkgconfig} directory. You need to set environment variables to
point @command{pkg-config} to these files.
If you need to @emph{add} directories to @command{pkg-config}'s search list
(typical use case: library installed separately), add it to
@code{$PKG_CONFIG_PATH}:
@example
export PKG_CONFIG_PATH=/opt/x264/lib/pkgconfig:/opt/opus/lib/pkgconfig
@end example
If you need to @emph{replace} @command{pkg-config}'s search list
(typical use case: cross-compiling), set it in
@code{$PKG_CONFIG_LIBDIR}:
@example
export PKG_CONFIG_LIBDIR=/home/me/cross/usr/lib/pkgconfig:/home/me/cross/usr/local/lib/pkgconfig
@end example
If you need to know the library's internal dependencies (typical use: static
linking), add the @code{--static} option to @command{pkg-config}:
@example
./configure --pkg-config-flags=--static
@end example
@section How do I use @command{pkg-config} when cross-compiling?
The best way is to install @command{pkg-config} in your cross-compilation
environment. It will automatically use the cross-compilation libraries.
You can also use @command{pkg-config} from the host environment by
specifying explicitly @code{--pkg-config=pkg-config} to @command{configure}.
In that case, you must point @command{pkg-config} to the correct directories
using the @code{PKG_CONFIG_LIBDIR}, as explained in the previous entry.
As an intermediate solution, you can place in your cross-compilation
environment a script that calls the host @command{pkg-config} with
@code{PKG_CONFIG_LIBDIR} set. That script can look like that:
@example
#!/bin/sh
PKG_CONFIG_LIBDIR=/path/to/cross/lib/pkgconfig
export PKG_CONFIG_LIBDIR
exec /usr/bin/pkg-config "$@@"
@end example
@chapter Usage
@section ffmpeg does not work; what is wrong?
Try a @code{make distclean} in the ffmpeg source directory before the build.
If this does not help see
(@url{https://ffmpeg.org/bugreports.html}).
(@url{http://ffmpeg.org/bugreports.html}).
@section How do I encode single pictures into movies?
@@ -311,18 +260,18 @@ invoking ffmpeg with several @option{-i} options.
For audio, to put all channels together in a single stream (example: two
mono streams into one stereo stream): this is sometimes called to
@emph{merge} them, and can be done using the
@url{ffmpeg-filters.html#amerge, @code{amerge}} filter.
@url{http://ffmpeg.org/ffmpeg-filters.html#amerge, @code{amerge}} filter.
@item
For audio, to play one on top of the other: this is called to @emph{mix}
them, and can be done by first merging them into a single stream and then
using the @url{ffmpeg-filters.html#pan, @code{pan}} filter to mix
using the @url{http://ffmpeg.org/ffmpeg-filters.html#pan, @code{pan}} filter to mix
the channels at will.
@item
For video, to display both together, side by side or one on top of a part of
the other; it can be done using the
@url{ffmpeg-filters.html#overlay, @code{overlay}} video filter.
@url{http://ffmpeg.org/ffmpeg-filters.html#overlay, @code{overlay}} video filter.
@end itemize
@@ -333,23 +282,23 @@ There are several solutions, depending on the exact circumstances.
@subsection Concatenating using the concat @emph{filter}
FFmpeg has a @url{ffmpeg-filters.html#concat,
FFmpeg has a @url{http://ffmpeg.org/ffmpeg-filters.html#concat,
@code{concat}} filter designed specifically for that, with examples in the
documentation. This operation is recommended if you need to re-encode.
@subsection Concatenating using the concat @emph{demuxer}
FFmpeg has a @url{ffmpeg-formats.html#concat,
FFmpeg has a @url{http://www.ffmpeg.org/ffmpeg-formats.html#concat,
@code{concat}} demuxer which you can use when you want to avoid a re-encode and
your format doesn't support file level concatenation.
@subsection Concatenating using the concat @emph{protocol} (file level)
FFmpeg has a @url{ffmpeg-protocols.html#concat,
FFmpeg has a @url{http://ffmpeg.org/ffmpeg-protocols.html#concat,
@code{concat}} protocol designed specifically for that, with examples in the
documentation.
A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow one to concatenate
A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow to concatenate
video by merely concatenating the files containing them.
Hence you may concatenate your multimedia files by first transcoding them to
@@ -467,40 +416,6 @@ point acceptable for your tastes. The most common options to do that are
@option{-qscale} and @option{-qmax}, but you should peruse the documentation
of the encoder you chose.
@section I have a stretched video, why does scaling does not fix it?
A lot of video codecs and formats can store the @emph{aspect ratio} of the
video: this is the ratio between the width and the height of either the full
image (DAR, display aspect ratio) or individual pixels (SAR, sample aspect
ratio). For example, EGA screens at resolution 640×350 had 4:3 DAR and 35:48
SAR.
Most still image processing work with square pixels, i.e. 1:1 SAR, but a lot
of video standards, especially from the analogic-numeric transition era, use
non-square pixels.
Most processing filters in FFmpeg handle the aspect ratio to avoid
stretching the image: cropping adjusts the DAR to keep the SAR constant,
scaling adjusts the SAR to keep the DAR constant.
If you want to stretch, or “unstretch”, the image, you need to override the
information with the
@url{ffmpeg-filters.html#setdar_002c-setsar, @code{setdar or setsar filters}}.
Do not forget to examine carefully the original video to check whether the
stretching comes from the image or from the aspect ratio information.
For example, to fix a badly encoded EGA capture, use the following commands,
either the first one to upscale to square pixels or the second one to set
the correct aspect ratio or the third one to avoid transcoding (may not work
depending on the format / codec / player / phase of the moon):
@example
ffmpeg -i ega_screen.nut -vf scale=640:480,setsar=1 ega_screen_scaled.nut
ffmpeg -i ega_screen.nut -vf setdar=4/3 ega_screen_anamorphic.nut
ffmpeg -i ega_screen.nut -aspect 4/3 -c copy ega_screen_overridden.nut
@end example
@chapter Development
@section Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat?
@@ -589,7 +504,7 @@ see @file{libavformat/aviobuf.c} in FFmpeg and @file{libmpdemux/demux_lavf.c} in
@section Where is the documentation about ffv1, msmpeg4, asv1, 4xm?
see @url{https://www.ffmpeg.org/~michael/}
see @url{http://www.ffmpeg.org/~michael/}
@section How do I feed H.263-RTP (and other codecs in RTP) to libavcodec?

View File

@@ -1,5 +1,4 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Automated Testing Environment
@titlepage
@@ -13,36 +12,36 @@
@chapter Introduction
FATE is an extended regression suite on the client-side and a means
FATE is an extended regression suite on the client-side and a means
for results aggregation and presentation on the server-side.
The first part of this document explains how you can use FATE from
The first part of this document explains how you can use FATE from
your FFmpeg source directory to test your ffmpeg binary. The second
part describes how you can run FATE to submit the results to FFmpeg's
FATE server.
In any way you can have a look at the publicly viewable FATE results
In any way you can have a look at the publicly viewable FATE results
by visiting this website:
@url{http://fate.ffmpeg.org/}
@url{http://fate.ffmpeg.org/}
This is especially recommended for all people contributing source
This is especially recommended for all people contributing source
code to FFmpeg, as it can be seen if some test on some platform broke
with their recent contribution. This usually happens on the platforms
the developers could not test on.
The second part of this document describes how you can run FATE to
The second part of this document describes how you can run FATE to
submit your results to FFmpeg's FATE server. If you want to submit your
results be sure to check that your combination of CPU, OS and compiler
is not already listed on the above mentioned website.
In the third part you can find a comprehensive listing of FATE makefile
In the third part you can find a comprehensive listing of FATE makefile
targets and variables.
@chapter Using FATE from your FFmpeg source directory
If you want to run FATE on your machine you need to have the samples
If you want to run FATE on your machine you need to have the samples
in place. You can get the samples via the build target fate-rsync.
Use this command from the top-level source directory:
@@ -51,11 +50,11 @@ make fate-rsync SAMPLES=fate-suite/
make fate SAMPLES=fate-suite/
@end example
The above commands set the samples location by passing a makefile
The above commands set the samples location by passing a makefile
variable via command line. It is also possible to set the samples
location at source configuration time by invoking configure with
@option{--samples=<path to the samples directory>}. Afterwards you can
invoke the makefile targets without setting the @var{SAMPLES} makefile
`--samples=<path to the samples directory>'. Afterwards you can
invoke the makefile targets without setting the SAMPLES makefile
variable. This is illustrated by the following commands:
@example
@@ -64,7 +63,7 @@ make fate-rsync
make fate
@end example
Yet another way to tell FATE about the location of the sample
Yet another way to tell FATE about the location of the sample
directory is by making sure the environment variable FATE_SAMPLES
contains the path to your samples directory. This can be achieved
by e.g. putting that variable in your shell profile or by setting
@@ -85,7 +84,7 @@ To use a custom wrapper to run the test, pass @option{--target-exec} to
@chapter Submitting the results to the FFmpeg result aggregation server
To submit your results to the server you should run fate through the
To submit your results to the server you should run fate through the
shell script @file{tests/fate.sh} from the FFmpeg sources. This script needs
to be invoked with a configuration file as its first argument.
@@ -93,23 +92,23 @@ to be invoked with a configuration file as its first argument.
tests/fate.sh /path/to/fate_config
@end example
A configuration file template with comments describing the individual
A configuration file template with comments describing the individual
configuration variables can be found at @file{doc/fate_config.sh.template}.
@ifhtml
The mentioned configuration template is also available here:
The mentioned configuration template is also available here:
@verbatiminclude fate_config.sh.template
@end ifhtml
Create a configuration that suits your needs, based on the configuration
template. The @env{slot} configuration variable can be any string that is not
Create a configuration that suits your needs, based on the configuration
template. The `slot' configuration variable can be any string that is not
yet used, but it is suggested that you name it adhering to the following
pattern @samp{@var{arch}-@var{os}-@var{compiler}-@var{compiler version}}. The
configuration file itself will be sourced in a shell script, therefore all
shell features may be used. This enables you to setup the environment as you
need it for your build.
pattern <arch>-<os>-<compiler>-<compiler version>. The configuration file
itself will be sourced in a shell script, therefore all shell features may
be used. This enables you to setup the environment as you need it for your
build.
For your first test runs the @env{fate_recv} variable should be empty or
For your first test runs the `fate_recv' variable should be empty or
commented out. This will run everything as normal except that it will omit
the submission of the results to the server. The following files should be
present in $workdir as specified in the configuration file:
@@ -122,29 +121,29 @@ present in $workdir as specified in the configuration file:
@item version
@end itemize
When you have everything working properly you can create an SSH key pair
When you have everything working properly you can create an SSH key pair
and send the public key to the FATE server administrator who can be contacted
at the email address @email{fate-admin@@ffmpeg.org}.
Configure your SSH client to use public key authentication with that key
Configure your SSH client to use public key authentication with that key
when connecting to the FATE server. Also do not forget to check the identity
of the server and to accept its host key. This can usually be achieved by
running your SSH client manually and killing it after you accepted the key.
The FATE server's fingerprint is:
@table @samp
@table @option
@item RSA
d3:f1:83:97:a4:75:2b:a6:fb:d6:e8:aa:81:93:97:51
@item ECDSA
76:9f:68:32:04:1e:d5:d4:ec:47:3f:dc:fc:18:17:86
@end table
If you have problems connecting to the FATE server, it may help to try out
If you have problems connecting to the FATE server, it may help to try out
the @command{ssh} command with one or more @option{-v} options. You should
get detailed output concerning your SSH configuration and the authentication
process.
The only thing left is to automate the execution of the fate.sh script and
The only thing left is to automate the execution of the fate.sh script and
the synchronisation of the samples directory.
@@ -165,7 +164,7 @@ Run the FATE test suite (requires the fate-suite dataset).
@section Makefile variables
@table @env
@table @option
@item V
Verbosity level, can be set to 0, 1 or 2.
@itemize
@@ -183,20 +182,20 @@ Specify how many threads to use while running regression tests, it is
quite useful to detect thread-related regressions.
@item THREAD_TYPE
Specify which threading strategy test, either @samp{slice} or @samp{frame},
by default @samp{slice+frame}
Specify which threading strategy test, either @var{slice} or @var{frame},
by default @var{slice+frame}
@item CPUFLAGS
Specify CPU flags.
@item TARGET_EXEC
Specify or override the wrapper used to run the tests.
The @env{TARGET_EXEC} option provides a way to run FATE wrapped in
The @var{TARGET_EXEC} option provides a way to run FATE wrapped in
@command{valgrind}, @command{qemu-user} or @command{wine} or on remote targets
through @command{ssh}.
@item GEN
Set to @samp{1} to generate the missing or mismatched references.
Set to @var{1} to generate the missing or mismatched references.
@end table
@section Examples

View File

@@ -1,6 +1,5 @@
slot= # some unique identifier
repo=git://source.ffmpeg.org/ffmpeg.git # the source repository
#branch=release/2.6 # the branch to test
samples= # path to samples directory
workdir= # directory in which to do all the work
#fate_recv="ssh -T fate@fate.ffmpeg.org" # command to submit report

View File

@@ -1,5 +1,4 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Bitstream Filters Documentation
@titlepage

View File

@@ -1,5 +1,4 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Codecs Documentation
@titlepage

View File

@@ -1,5 +1,4 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Devices Documentation
@titlepage

View File

@@ -1,5 +1,4 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Filters Documentation
@titlepage

View File

@@ -1,5 +1,4 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Formats Documentation
@titlepage

View File

@@ -1,5 +1,4 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Protocols Documentation
@titlepage

View File

@@ -1,5 +1,4 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Resampler Documentation
@titlepage

View File

@@ -1,5 +1,4 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Scaler Documentation
@titlepage

View File

@@ -1,5 +1,4 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Utilities Documentation
@titlepage

View File

@@ -1,5 +1,4 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle ffmpeg Documentation
@titlepage
@@ -80,7 +79,7 @@ The format option may be needed for raw input files.
The transcoding process in @command{ffmpeg} for each output can be described by
the following diagram:
@verbatim
@example
_______ ______________
| | | |
| input | demuxer | encoded data | decoder
@@ -99,7 +98,7 @@ the following diagram:
|________| |______________|
@end verbatim
@end example
@command{ffmpeg} calls the libavformat library (containing demuxers) to read
input files and get packets containing encoded data from them. When there are
@@ -124,7 +123,7 @@ Simple filtergraphs are those that have exactly one input and output, both of
the same type. In the above diagram they can be represented by simply inserting
an additional step between decoding and encoding:
@verbatim
@example
_________ ______________
| | | |
| decoded | | encoded data |
@@ -136,19 +135,19 @@ an additional step between decoding and encoding:
| frames |
|__________|
@end verbatim
@end example
Simple filtergraphs are configured with the per-stream @option{-filter} option
(with @option{-vf} and @option{-af} aliases for video and audio respectively).
A simple filtergraph for video can look for example like this:
@verbatim
@example
_______ _____________ _______ ________
| | | | | | | |
| input | ---> | deinterlace | ---> | scale | ---> | output |
|_______| |_____________| |_______| |________|
@end verbatim
@end example
Note that some filters change frame properties but not frame contents. E.g. the
@code{fps} filter in the example above changes number of frames, but does not
@@ -161,7 +160,7 @@ processing chain applied to one stream. This is the case, for example, when the
more than one input and/or output, or when output stream type is different from
input. They can be represented with the following diagram:
@verbatim
@example
_________
| |
| input 0 |\ __________
@@ -179,7 +178,7 @@ input. They can be represented with the following diagram:
| input 2 |/
|_________|
@end verbatim
@end example
Complex filtergraphs are configured with the @option{-filter_complex} option.
Note that this option is global, since a complex filtergraph, by its nature,
@@ -198,14 +197,14 @@ step for the specified stream, so it does only demuxing and muxing. It is useful
for changing the container format or modifying container-level metadata. The
diagram above will, in this case, simplify to this:
@verbatim
@example
_______ ______________ ________
| | | | | |
| input | demuxer | encoded data | muxer | output |
| file | ---------> | packets | -------> | file |
|_______| |______________| |________|
@end verbatim
@end example
Since there is no decoding or encoding, it is very fast and there is no quality
loss. However, it might not work in some cases because of many factors. Applying
@@ -223,7 +222,7 @@ with the highest resolution, for audio, it is the stream with the most channels,
subtitles, it is the first subtitle stream. In the case where several streams of
the same type rate equally, the stream with the lowest index is chosen.
You can disable some of those defaults by using the @code{-vn/-an/-sn/-dn} options. For
You can disable some of those defaults by using the @code{-vn/-an/-sn} options. For
full manual control, use the @code{-map} option, which disables the defaults just
described.
@@ -253,10 +252,6 @@ Overwrite output files without asking.
Do not overwrite output files, and exit immediately if a specified
output file already exists.
@item -stream_loop @var{number} (@emph{input})
Set number of times input stream shall be looped. Loop 0 means no loop,
loop -1 means infinite loop.
@item -c[:@var{stream_specifier}] @var{codec} (@emph{input/output,per-stream})
@itemx -codec[:@var{stream_specifier}] @var{codec} (@emph{input/output,per-stream})
Select an encoder (when used before an output file) or a decoder (when used
@@ -284,27 +279,23 @@ data read from the input file.
When used as an output option (before an output filename), stop writing the
output after its duration reaches @var{duration}.
@var{duration} must be a time duration specification,
see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
@var{duration} may be a number in seconds, or in @code{hh:mm:ss[.xxx]} form.
-to and -t are mutually exclusive and -t has priority.
@item -to @var{position} (@emph{output})
Stop writing the output at @var{position}.
@var{position} must be a time duration specification,
see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
@var{position} may be a number in seconds, or in @code{hh:mm:ss[.xxx]} form.
-to and -t are mutually exclusive and -t has priority.
@item -fs @var{limit_size} (@emph{output})
Set the file size limit, expressed in bytes. No further chunk of bytes is written
after the limit is exceeded. The size of the output file is slightly more than the
requested file size.
Set the file size limit, expressed in bytes.
@item -ss @var{position} (@emph{input/output})
When used as an input option (before @code{-i}), seeks in this input file to
@var{position}. Note that in most formats it is not possible to seek exactly,
so @command{ffmpeg} will seek to the closest seek point before @var{position}.
@var{position}. Note the in most formats it is not possible to seek exactly, so
@command{ffmpeg} will seek to the closest seek point before @var{position}.
When transcoding and @option{-accurate_seek} is enabled (the default), this
extra segment between the seek point and @var{position} will be decoded and
discarded. When doing stream copy or when @option{-noaccurate_seek} is used, it
@@ -313,13 +304,7 @@ will be preserved.
When used as an output option (before an output filename), decodes but discards
input until the timestamps reach @var{position}.
@var{position} must be a time duration specification,
see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
@item -sseof @var{position} (@emph{input/output})
Like the @code{-ss} option but relative to the "end of file". That is negative
values are earlier in the file, 0 is at EOF.
@var{position} may be either in seconds or in @code{hh:mm:ss[.xxx]} form.
@item -itsoffset @var{offset} (@emph{input})
Set the input time offset.
@@ -334,15 +319,15 @@ the time duration specified in @var{offset}.
@item -timestamp @var{date} (@emph{output})
Set the recording timestamp in the container.
@var{date} must be a date specification,
@var{date} must be a time duration specification,
see @ref{date syntax,,the Date section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
@item -metadata[:metadata_specifier] @var{key}=@var{value} (@emph{output,per-metadata})
Set a metadata key/value pair.
An optional @var{metadata_specifier} may be given to set metadata
on streams, chapters or programs. See @code{-map_metadata}
documentation for details.
on streams or chapters. See @code{-map_metadata} documentation for
details.
This option overrides metadata set with @code{-map_metadata}. It is
also possible to delete metadata by using an empty value.
@@ -354,14 +339,9 @@ ffmpeg -i in.avi -metadata title="my title" out.flv
To set the language of the first audio stream:
@example
ffmpeg -i INPUT -metadata:s:a:0 language=eng OUTPUT
ffmpeg -i INPUT -metadata:s:a:1 language=eng OUTPUT
@end example
@item -program [title=@var{title}:][program_num=@var{program_num}:]st=@var{stream}[:st=@var{stream}...] (@emph{output})
Creates a program with the specified @var{title}, @var{program_num} and adds the specified
@var{stream}(s) to it.
@item -target @var{type} (@emph{output})
Specify target file type (@code{vcd}, @code{svcd}, @code{dvd}, @code{dv},
@code{dv50}). @var{type} may be prefixed with @code{pal-}, @code{ntsc-} or
@@ -380,7 +360,7 @@ ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg
@end example
@item -dframes @var{number} (@emph{output})
Set the number of data frames to output. This is an alias for @code{-frames:d}.
Set the number of data frames to record. This is an alias for @code{-frames:d}.
@item -frames[:@var{stream_specifier}] @var{framecount} (@emph{output,per-stream})
Stop writing to the stream after @var{framecount} frames.
@@ -481,24 +461,18 @@ Technical note -- attachments are implemented as codec extradata, so this
option can actually be used to extract extradata from any stream, not just
attachments.
@item -noautorotate
Disable automatically rotating video based on file metadata.
@end table
@section Video Options
@table @option
@item -vframes @var{number} (@emph{output})
Set the number of video frames to output. This is an alias for @code{-frames:v}.
Set the number of video frames to record. This is an alias for @code{-frames:v}.
@item -r[:@var{stream_specifier}] @var{fps} (@emph{input/output,per-stream})
Set frame rate (Hz value, fraction or abbreviation).
As an input option, ignore any timestamps stored in the file and instead
generate timestamps assuming constant frame rate @var{fps}.
This is not the same as the @option{-framerate} option used for some input formats
like image2 or v4l2 (it used to be the same in older versions of FFmpeg).
If in doubt use @option{-framerate} instead of the input option @option{-r}.
As an output option, duplicate or drop input frames to achieve constant output
frame rate @var{fps}.
@@ -682,16 +656,6 @@ Use VDPAU (Video Decode and Presentation API for Unix) hardware acceleration.
@item dxva2
Use DXVA2 (DirectX Video Acceleration) hardware acceleration.
@item qsv
Use the Intel QuickSync Video acceleration for video transcoding.
Unlike most other values, this option does not enable accelerated decoding (that
is used automatically whenever a qsv decoder is selected), but accelerated
transcoding, without copying the frames into the system memory.
For it to work, both the decoder and the encoder must support QSV acceleration
and no filters must be used.
@end table
This option has no effect if the selected hwaccel is not available or not
@@ -718,32 +682,14 @@ is not specified, the value of the @var{DISPLAY} environment variable is used
@item dxva2
For DXVA2, this option should contain the number of the display adapter to use.
If this option is not specified, the default adapter is used.
@item qsv
For QSV, this option corresponds to the values of MFX_IMPL_* . Allowed values
are:
@table @option
@item auto
@item sw
@item hw
@item auto_any
@item hw_any
@item hw2
@item hw3
@item hw4
@end table
@end table
@item -hwaccels
List all hardware acceleration methods supported in this build of ffmpeg.
@end table
@section Audio Options
@table @option
@item -aframes @var{number} (@emph{output})
Set the number of audio frames to output. This is an alias for @code{-frames:a}.
Set the number of audio frames to record. This is an alias for @code{-frames:a}.
@item -ar[:@var{stream_specifier}] @var{freq} (@emph{input/output,per-stream})
Set the audio sampling frequency. For output streams it is set by
default to the frequency of the corresponding input stream. For input
@@ -875,21 +821,8 @@ To map all the streams except the second audio, use negative mappings
ffmpeg -i INPUT -map 0 -map -0:a:1 OUTPUT
@end example
To pick the English audio stream:
@example
ffmpeg -i INPUT -map 0:m:language:eng OUTPUT
@end example
Note that using this option disables the default mappings for this output file.
@item -ignore_unknown
Ignore input streams with unknown type instead of failing if copying
such streams is attempted.
@item -copy_unknown
Allow input streams with unknown type to be copied instead of failing if copying
such streams is attempted.
@item -map_channel [@var{input_file_id}.@var{stream_specifier}.@var{channel_id}|-1][:@var{output_file_id}.@var{stream_specifier}]
Map an audio channel from a given input to an output. If
@var{output_file_id}.@var{stream_specifier} is not set, the audio channel will
@@ -1008,7 +941,7 @@ Dump each input packet to stderr.
@item -hex (@emph{global})
When dumping packets, also dump the payload.
@item -re (@emph{input})
Read input at native frame rate. Mainly used to simulate a grab device,
Read input at native frame rate. Mainly used to simulate a grab device.
or live input stream (e.g. when reading from a file). Should not be used
with actual grab devices or live input streams (where it can cause packet
loss).
@@ -1053,13 +986,6 @@ With -map you can select from which stream the timestamps should be
taken. You can leave either video or audio unchanged and sync the
remaining stream(s) to the unchanged one.
@item -frame_drop_threshold @var{parameter}
Frame drop threshold, which specifies how much behind video frames can
be before they are dropped. In frame rate units, so 1.0 is one frame.
The default is -1.1. One possible usecase is to avoid framedrops in case
of noisy timestamps or to increase frame drop precision in case of exact
timestamps.
@item -async @var{samples_per_second}
Audio sync method. "Stretches/squeezes" the audio stream to match the timestamps,
the parameter is the maximum samples per second by which the audio is changed.
@@ -1082,12 +1008,6 @@ processing (e.g. in case the format option @option{avoid_negative_ts}
is enabled) the output timestamps may mismatch with the input
timestamps even when this option is selected.
@item -start_at_zero
When used with @option{copyts}, shift input timestamps so they start at zero.
This means that using e.g. @code{-ss 50} will make output timestamps start at
50 seconds, regardless of what timestamp the input file started at.
@item -copytb @var{mode}
Specify how to set the encoder timebase when stream copying. @var{mode} is an
integer numeric value, and can assume one of the following values:
@@ -1216,19 +1136,6 @@ This option enables or disables accurate seeking in input files with the
transcoding. Use @option{-noaccurate_seek} to disable it, which may be useful
e.g. when copying some streams and transcoding the others.
@item -seek_timestamp (@emph{input})
This option enables or disables seeking by timestamp in input files with the
@option{-ss} option. It is disabled by default. If enabled, the argument
to the @option{-ss} option is considered an actual timestamp, and is not
offset by the start time of the file. This matters only for files which do
not start from timestamp 0, such as transport streams.
@item -thread_queue_size @var{size} (@emph{input})
This option sets the maximum number of queued packets when reading from the
file or device. With low latency / high rate live streams, packets may be
discarded if they are not read in a timely manner; raising this value can
avoid it.
@item -override_ffserver (@emph{global})
Overrides the input specifications from @command{ffserver}. Using this
option you can map any input stream to @command{ffserver} and control
@@ -1239,11 +1146,6 @@ requested by @command{ffserver}.
The option is intended for cases where features are needed that cannot be
specified to @command{ffserver} but can be to @command{ffmpeg}.
@item -sdp_file @var{file} (@emph{global})
Print sdp information for an output stream to @var{file}.
This allows dumping sdp information when at least one output isn't an
rtp stream. (Requires at least one of the output formats to be rtp).
@item -discard (@emph{input})
Allows discarding specific streams or frames of streams at the demuxer.
Not all demuxers support this.
@@ -1268,26 +1170,6 @@ Discard all frames excepts keyframes.
Discard all frames.
@end table
@item -abort_on @var{flags} (@emph{global})
Stop and abort on various conditions. The following flags are available:
@table @option
@item empty_output
No packets were passed to the muxer, the output is empty.
@end table
@item -xerror (@emph{global})
Stop and exit on error
@item -max_muxing_queue_size @var{packets} (@emph{output,per-stream})
When transcoding audio and/or video streams, ffmpeg will not begin writing into
the output until it has one packet for each such stream. While waiting for that
to happen, packets for other streams are buffered. This option sets the size of
this buffer, in packets, for the matching output stream.
The default value of this option should be high enough for most uses, so only
touch this option if you are sure that you need it.
@end table
As a special exception, you can use a bitmap subtitle stream as input: it
@@ -1313,10 +1195,7 @@ awkward to specify on the command line. Lines starting with the hash
('#') character are ignored and are used to provide comments. Check
the @file{presets} directory in the FFmpeg source tree for examples.
There are two types of preset files: ffpreset and avpreset files.
@subsection ffpreset files
ffpreset files are specified with the @code{vpre}, @code{apre},
Preset files are specified with the @code{vpre}, @code{apre},
@code{spre}, and @code{fpre} options. The @code{fpre} option takes the
filename of the preset instead of a preset name as input and can be
used for any kind of codec. For the @code{vpre}, @code{apre}, and
@@ -1341,31 +1220,67 @@ directories, where @var{codec_name} is the name of the codec to which
the preset file options will be applied. For example, if you select
the video codec with @code{-vcodec libvpx} and use @code{-vpre 1080p},
then it will search for the file @file{libvpx-1080p.ffpreset}.
@subsection avpreset files
avpreset files are specified with the @code{pre} option. They work similar to
ffpreset files, but they only allow encoder- specific options. Therefore, an
@var{option}=@var{value} pair specifying an encoder cannot be used.
When the @code{pre} option is specified, ffmpeg will look for files with the
suffix .avpreset in the directories @file{$AVCONV_DATADIR} (if set), and
@file{$HOME/.avconv}, and in the datadir defined at configuration time (usually
@file{PREFIX/share/ffmpeg}), in that order.
First ffmpeg searches for a file named @var{codec_name}-@var{arg}.avpreset in
the above-mentioned directories, where @var{codec_name} is the name of the codec
to which the preset file options will be applied. For example, if you select the
video codec with @code{-vcodec libvpx} and use @code{-pre 1080p}, then it will
search for the file @file{libvpx-1080p.avpreset}.
If no such file is found, then ffmpeg will search for a file named
@var{arg}.avpreset in the same directories.
@c man end OPTIONS
@chapter Tips
@c man begin TIPS
@itemize
@item
For streaming at very low bitrates, use a low frame rate
and a small GOP size. This is especially true for RealVideo where
the Linux player does not seem to be very fast, so it can miss
frames. An example is:
@example
ffmpeg -g 3 -r 3 -t 10 -b:v 50k -s qcif -f rv10 /tmp/b.rm
@end example
@item
The parameter 'q' which is displayed while encoding is the current
quantizer. The value 1 indicates that a very good quality could
be achieved. The value 31 indicates the worst quality. If q=31 appears
too often, it means that the encoder cannot compress enough to meet
your bitrate. You must either increase the bitrate, decrease the
frame rate or decrease the frame size.
@item
If your computer is not fast enough, you can speed up the
compression at the expense of the compression ratio. You can use
'-me zero' to speed up motion estimation, and '-g 0' to disable
motion estimation completely (you have only I-frames, which means it
is about as good as JPEG compression).
@item
To have very low audio bitrates, reduce the sampling frequency
(down to 22050 Hz for MPEG audio, 22050 or 11025 for AC-3).
@item
To have a constant quality (but a variable bitrate), use the option
'-qscale n' when 'n' is between 1 (excellent quality) and 31 (worst
quality).
@end itemize
@c man end TIPS
@chapter Examples
@c man begin EXAMPLES
@section Preset files
A preset file contains a sequence of @var{option=value} pairs, one for
each line, specifying a sequence of options which can be specified also on
the command line. Lines starting with the hash ('#') character are ignored and
are used to provide comments. Empty lines are also ignored. Check the
@file{presets} directory in the FFmpeg source tree for examples.
Preset files are specified with the @code{pre} option, this option takes a
preset name as input. FFmpeg searches for a file named @var{preset_name}.avpreset in
the directories @file{$AVCONV_DATADIR} (if set), and @file{$HOME/.ffmpeg}, and in
the data directory defined at configuration time (usually @file{$PREFIX/share/ffmpeg})
in that order. For example, if the argument is @code{libx264-max}, it will
search for the file @file{libx264-max.avpreset}.
@section Video and Audio grabbing
If you specify the input format and device then ffmpeg can grab video
@@ -1513,7 +1428,7 @@ combination with -ss to start extracting from a certain point in time.
For creating a video from many images:
@example
ffmpeg -f image2 -framerate 12 -i foo-%03d.jpeg -s WxH foo.avi
ffmpeg -f image2 -i foo-%03d.jpeg -r 12 -s WxH foo.avi
@end example
The syntax @code{foo-%03d.jpeg} specifies to use a decimal number
@@ -1528,18 +1443,18 @@ image2-specific @code{-pattern_type glob} option.
For example, for creating a video from filenames matching the glob pattern
@code{foo-*.jpeg}:
@example
ffmpeg -f image2 -pattern_type glob -framerate 12 -i 'foo-*.jpeg' -s WxH foo.avi
ffmpeg -f image2 -pattern_type glob -i 'foo-*.jpeg' -r 12 -s WxH foo.avi
@end example
@item
You can put many streams of the same type in the output:
@example
ffmpeg -i test1.avi -i test2.avi -map 1:1 -map 1:0 -map 0:1 -map 0:0 -c copy -y test12.nut
ffmpeg -i test1.avi -i test2.avi -map 0:3 -map 0:2 -map 0:1 -map 0:0 -c copy test12.nut
@end example
The resulting output file @file{test12.nut} will contain the first four streams
from the input files in reverse order.
The resulting output file @file{test12.avi} will contain first four streams from
the input file in reverse order.
@item
To force CBR video output:

View File

@@ -1,5 +1,4 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle ffplay Documentation
@titlepage
@@ -38,26 +37,14 @@ Force displayed height.
Set frame size (WxH or abbreviation), needed for videos which do
not contain a header with the frame size like raw YUV. This option
has been deprecated in favor of private options, try -video_size.
@item -fs
Start in fullscreen mode.
@item -an
Disable audio.
@item -vn
Disable video.
@item -sn
Disable subtitles.
@item -ss @var{pos}
Seek to @var{pos}. Note that in most formats it is not possible to seek
exactly, so @command{ffplay} will seek to the nearest seek point to
@var{pos}.
@var{pos} must be a time duration specification,
see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
Seek to a given position in seconds.
@item -t @var{duration}
Play @var{duration} seconds of audio/video.
@var{duration} must be a time duration specification,
see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
play <duration> seconds of audio/video
@item -bytes
Seek by bytes.
@item -nodisp
@@ -122,10 +109,15 @@ duration, the codec parameters, the current position in the stream and
the audio/video synchronisation drift. It is on by default, to
explicitly disable it you need to specify @code{-nostats}.
@item -bug
Work around bugs.
@item -fast
Non-spec-compliant optimizations.
@item -genpts
Generate pts.
@item -rtp_tcp
Force RTP/TCP protocol usage instead of RTP/UDP. It is only meaningful
if you are streaming with the RTSP protocol.
@item -sync @var{type}
Set the master clock to audio (@code{type=audio}), video
(@code{type=video}) or external (@code{type=ext}). Default is audio. The
@@ -133,20 +125,23 @@ master clock is used to control audio-video synchronization. Most media
players use audio as master clock, but in some cases (streaming or high
quality broadcast) it is necessary to change that. This option is mainly
used for debugging purposes.
@item -ast @var{audio_stream_specifier}
Select the desired audio stream using the given stream specifier. The stream
specifiers are described in the @ref{Stream specifiers} chapter. If this option
is not specified, the "best" audio stream is selected in the program of the
already selected video stream.
@item -vst @var{video_stream_specifier}
Select the desired video stream using the given stream specifier. The stream
specifiers are described in the @ref{Stream specifiers} chapter. If this option
is not specified, the "best" video stream is selected.
@item -sst @var{subtitle_stream_specifier}
Select the desired subtitle stream using the given stream specifier. The stream
specifiers are described in the @ref{Stream specifiers} chapter. If this option
is not specified, the "best" subtitle stream is selected in the program of the
already selected video or audio stream.
@item -threads @var{count}
Set the thread count.
@item -ast @var{audio_stream_number}
Select the desired audio stream number, counting from 0. The number
refers to the list of all the input audio streams. If it is greater
than the number of audio streams minus one, then the last one is
selected, if it is negative the audio playback is disabled.
@item -vst @var{video_stream_number}
Select the desired video stream number, counting from 0. The number
refers to the list of all the input video streams. If it is greater
than the number of video streams minus one, then the last one is
selected, if it is negative the video playback is disabled.
@item -sst @var{subtitle_stream_number}
Select the desired subtitle stream number, counting from 0. The number
refers to the list of all the input subtitle streams. If it is greater
than the number of subtitle streams minus one, then the last one is
selected, if it is negative the subtitle rendering is disabled.
@item -autoexit
Exit when video is done playing.
@item -exitonkeydown
@@ -169,20 +164,8 @@ Force a specific video decoder.
Force a specific subtitle decoder.
@item -autorotate
Automatically rotate the video according to file metadata. Enabled by
default, use @option{-noautorotate} to disable it.
@item -framedrop
Drop video frames if video is out of sync. Enabled by default if the master
clock is not set to video. Use this option to enable frame dropping for all
master clock sources, use @option{-noframedrop} to disable it.
@item -infbuf
Do not limit the input buffer size, read as much data as possible from the
input as soon as possible. Enabled by default for realtime streams, where data
may be dropped if not read in time. Use this option to enable infinite buffers
for all inputs, use @option{-noinfbuf} to disable it.
Automatically rotate the video according to presentation metadata. Set by
default, use -noautorotate to disable.
@end table
@section While playing
@@ -197,15 +180,6 @@ Toggle full screen.
@item p, SPC
Pause.
@item m
Toggle mute.
@item 9, 0
Decrease and increase volume respectively.
@item /, *
Decrease and increase volume respectively.
@item a
Cycle audio channel in the current program.
@@ -238,12 +212,9 @@ Seek to the previous/next chapter.
or if there are no chapters
Seek backward/forward 10 minutes.
@item right mouse click
@item mouse click
Seek to percentage in file corresponding to fraction of width.
@item left mouse double-click
Toggle full screen.
@end table
@c man end

View File

@@ -1,5 +1,4 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle ffprobe Documentation
@titlepage
@@ -120,10 +119,6 @@ Show payload data, as a hexadecimal and ASCII dump. Coupled with
The dump is printed as the "data" field. It may contain newlines.
@item -show_data_hash @var{algorithm}
Show a hash of payload data, for packets with @option{-show_packets} and for
codec extradata with @option{-show_streams}.
@item -show_error
Show information about the error found when trying to probe the input.
@@ -185,7 +180,7 @@ format : stream=codec_type
To show all the tags in the stream and format sections:
@example
stream_tags : format_tags
format_tags : format_tags
@end example
To show only the @code{title} tag (if available) in the stream
@@ -245,7 +240,7 @@ continue reading from that.
Each interval is specified by two optional parts, separated by "%".
The first part specifies the interval start position. It is
interpreted as an absolute position, or as a relative offset from the
interpreted as an abolute position, or as a relative offset from the
current position if it is preceded by the "+" character. If this first
part is not specified, no seeking will be performed when reading this
interval.
@@ -322,12 +317,6 @@ Show information related to program and library versions. This is the
equivalent of setting both @option{-show_program_version} and
@option{-show_library_versions} options.
@item -show_pixel_formats
Show information about all pixel formats supported by FFmpeg.
Pixel format information for each format is printed within a section
with name "PIXEL_FORMAT".
@item -bitexact
Force bitexact output, useful to produce output which is not dependent
on the specific build.
@@ -447,17 +436,17 @@ writer).
It can assume one of the following values:
@table @option
@item c
Perform C-like escaping. Strings containing a newline (@samp{\n}), carriage
return (@samp{\r}), a tab (@samp{\t}), a form feed (@samp{\f}), the escaping
character (@samp{\}) or the item separator character @var{SEP} are escaped
using C-like fashioned escaping, so that a newline is converted to the
sequence @samp{\n}, a carriage return to @samp{\r}, @samp{\} to @samp{\\} and
the separator @var{SEP} is converted to @samp{\@var{SEP}}.
Perform C-like escaping. Strings containing a newline ('\n'), carriage
return ('\r'), a tab ('\t'), a form feed ('\f'), the escaping
character ('\') or the item separator character @var{SEP} are escaped using C-like fashioned
escaping, so that a newline is converted to the sequence "\n", a
carriage return to "\r", '\' to "\\" and the separator @var{SEP} is
converted to "\@var{SEP}".
@item csv
Perform CSV-like escaping, as described in RFC4180. Strings
containing a newline (@samp{\n}), a carriage return (@samp{\r}), a double quote
(@samp{"}), or @var{SEP} are enclosed in double-quotes.
containing a newline ('\n'), a carriage return ('\r'), a double quote
('"'), or @var{SEP} are enclosed in double-quotes.
@item none
Perform no escaping.
@@ -485,7 +474,7 @@ The description of the accepted options follows.
Separator character used to separate the chapter, the section name, IDs and
potential tags in the printed field key.
Default value is @samp{.}.
Default value is '.'.
@item hierarchical, h
Specify if the section name specification should be hierarchical. If
@@ -507,22 +496,21 @@ The following conventions are adopted:
@item
all key and values are UTF-8
@item
@samp{.} is the subgroup separator
'.' is the subgroup separator
@item
newline, @samp{\t}, @samp{\f}, @samp{\b} and the following characters are
escaped
newline, '\t', '\f', '\b' and the following characters are escaped
@item
@samp{\} is the escape character
'\' is the escape character
@item
@samp{#} is the comment indicator
'#' is the comment indicator
@item
@samp{=} is the key/value separator
'=' is the key/value separator
@item
@samp{:} is not used but usually parsed as key/value separator
':' is not used but usually parsed as key/value separator
@end itemize
This writer accepts options as a list of @var{key}=@var{value} pairs,
separated by @samp{:}.
separated by ":".
The description of the accepted options follows.

View File

@@ -10,10 +10,8 @@
<xsd:sequence>
<xsd:element name="program_version" type="ffprobe:programVersionType" minOccurs="0" maxOccurs="1" />
<xsd:element name="library_versions" type="ffprobe:libraryVersionsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="pixel_formats" type="ffprobe:pixelFormatsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="packets" type="ffprobe:packetsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="frames" type="ffprobe:framesType" minOccurs="0" maxOccurs="1" />
<xsd:element name="packets_and_frames" type="ffprobe:packetsAndFramesType" minOccurs="0" maxOccurs="1" />
<xsd:element name="programs" type="ffprobe:programsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="streams" type="ffprobe:streamsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="chapters" type="ffprobe:chaptersType" minOccurs="0" maxOccurs="1" />
@@ -37,22 +35,7 @@
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="packetsAndFramesType">
<xsd:sequence>
<xsd:choice minOccurs="0" maxOccurs="unbounded">
<xsd:element name="packet" type="ffprobe:packetType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="subtitle" type="ffprobe:subtitleType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:choice>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="packetType">
<xsd:sequence>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="side_data_list" type="ffprobe:packetSideDataListType" minOccurs="0" maxOccurs="1" />
</xsd:sequence>
<xsd:attribute name="codec_type" type="xsd:string" use="required" />
<xsd:attribute name="stream_index" type="xsd:int" use="required" />
<xsd:attribute name="pts" type="xsd:long" />
@@ -67,27 +50,14 @@
<xsd:attribute name="pos" type="xsd:long" />
<xsd:attribute name="flags" type="xsd:string" use="required" />
<xsd:attribute name="data" type="xsd:string" />
<xsd:attribute name="data_hash" type="xsd:string" />
</xsd:complexType>
<xsd:complexType name="packetSideDataListType">
<xsd:sequence>
<xsd:element name="side_data" type="ffprobe:packetSideDataType" minOccurs="1" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="packetSideDataType">
<xsd:attribute name="side_data_type" type="xsd:string"/>
<xsd:attribute name="side_data_size" type="xsd:int" />
</xsd:complexType>
<xsd:complexType name="frameType">
<xsd:sequence>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="side_data_list" type="ffprobe:frameSideDataListType" minOccurs="0" maxOccurs="1" />
</xsd:sequence>
<xsd:attribute name="media_type" type="xsd:string" use="required"/>
<xsd:attribute name="stream_index" type="xsd:int" />
<xsd:attribute name="key_frame" type="xsd:int" use="required"/>
<xsd:attribute name="pts" type="xsd:long" />
<xsd:attribute name="pts_time" type="xsd:float"/>
@@ -121,17 +91,6 @@
<xsd:attribute name="repeat_pict" type="xsd:int" />
</xsd:complexType>
<xsd:complexType name="frameSideDataListType">
<xsd:sequence>
<xsd:element name="side_data" type="ffprobe:frameSideDataType" minOccurs="1" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="frameSideDataType">
<xsd:attribute name="side_data_type" type="xsd:string"/>
<xsd:attribute name="side_data_size" type="xsd:int" />
<xsd:attribute name="timecode" type="xsd:string"/>
</xsd:complexType>
<xsd:complexType name="subtitleType">
<xsd:attribute name="media_type" type="xsd:string" fixed="subtitle" use="required"/>
<xsd:attribute name="pts" type="xsd:long" />
@@ -166,14 +125,12 @@
<xsd:attribute name="visual_impaired" type="xsd:int" use="required" />
<xsd:attribute name="clean_effects" type="xsd:int" use="required" />
<xsd:attribute name="attached_pic" type="xsd:int" use="required" />
<xsd:attribute name="timed_thumbnails" type="xsd:int" use="required" />
</xsd:complexType>
<xsd:complexType name="streamType">
<xsd:sequence>
<xsd:element name="disposition" type="ffprobe:streamDispositionType" minOccurs="0" maxOccurs="1"/>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="side_data_list" type="ffprobe:packetSideDataListType" minOccurs="0" maxOccurs="1" />
</xsd:sequence>
<xsd:attribute name="index" type="xsd:int" use="required"/>
@@ -185,13 +142,10 @@
<xsd:attribute name="codec_tag" type="xsd:string" use="required"/>
<xsd:attribute name="codec_tag_string" type="xsd:string" use="required"/>
<xsd:attribute name="extradata" type="xsd:string" />
<xsd:attribute name="extradata_hash" type="xsd:string" />
<!-- video attributes -->
<xsd:attribute name="width" type="xsd:int"/>
<xsd:attribute name="height" type="xsd:int"/>
<xsd:attribute name="coded_width" type="xsd:int"/>
<xsd:attribute name="coded_height" type="xsd:int"/>
<xsd:attribute name="has_b_frames" type="xsd:int"/>
<xsd:attribute name="sample_aspect_ratio" type="xsd:string"/>
<xsd:attribute name="display_aspect_ratio" type="xsd:string"/>
@@ -199,12 +153,7 @@
<xsd:attribute name="level" type="xsd:int"/>
<xsd:attribute name="color_range" type="xsd:string"/>
<xsd:attribute name="color_space" type="xsd:string"/>
<xsd:attribute name="color_transfer" type="xsd:string"/>
<xsd:attribute name="color_primaries" type="xsd:string"/>
<xsd:attribute name="chroma_location" type="xsd:string"/>
<xsd:attribute name="field_order" type="xsd:string"/>
<xsd:attribute name="timecode" type="xsd:string"/>
<xsd:attribute name="refs" type="xsd:int"/>
<!-- audio attributes -->
<xsd:attribute name="sample_fmt" type="xsd:string"/>
@@ -223,7 +172,6 @@
<xsd:attribute name="duration" type="xsd:float"/>
<xsd:attribute name="bit_rate" type="xsd:int"/>
<xsd:attribute name="max_bit_rate" type="xsd:int"/>
<xsd:attribute name="bits_per_raw_sample" type="xsd:int"/>
<xsd:attribute name="nb_frames" type="xsd:int"/>
<xsd:attribute name="nb_read_frames" type="xsd:int"/>
<xsd:attribute name="nb_read_packets" type="xsd:int"/>
@@ -276,8 +224,8 @@
<xsd:complexType name="programVersionType">
<xsd:attribute name="version" type="xsd:string" use="required"/>
<xsd:attribute name="copyright" type="xsd:string" use="required"/>
<xsd:attribute name="build_date" type="xsd:string"/>
<xsd:attribute name="build_time" type="xsd:string"/>
<xsd:attribute name="build_date" type="xsd:string" use="required"/>
<xsd:attribute name="build_time" type="xsd:string" use="required"/>
<xsd:attribute name="compiler_ident" type="xsd:string" use="required"/>
<xsd:attribute name="configuration" type="xsd:string" use="required"/>
</xsd:complexType>
@@ -315,45 +263,4 @@
<xsd:element name="library_version" type="ffprobe:libraryVersionType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="pixelFormatFlagsType">
<xsd:attribute name="big_endian" type="xsd:int" use="required"/>
<xsd:attribute name="palette" type="xsd:int" use="required"/>
<xsd:attribute name="bitstream" type="xsd:int" use="required"/>
<xsd:attribute name="hwaccel" type="xsd:int" use="required"/>
<xsd:attribute name="planar" type="xsd:int" use="required"/>
<xsd:attribute name="rgb" type="xsd:int" use="required"/>
<xsd:attribute name="pseudopal" type="xsd:int" use="required"/>
<xsd:attribute name="alpha" type="xsd:int" use="required"/>
</xsd:complexType>
<xsd:complexType name="pixelFormatComponentType">
<xsd:attribute name="index" type="xsd:int" use="required"/>
<xsd:attribute name="bit_depth" type="xsd:int" use="required"/>
</xsd:complexType>
<xsd:complexType name="pixelFormatComponentsType">
<xsd:sequence>
<xsd:element name="component" type="ffprobe:pixelFormatComponentType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="pixelFormatType">
<xsd:sequence>
<xsd:element name="flags" type="ffprobe:pixelFormatFlagsType" minOccurs="0" maxOccurs="1"/>
<xsd:element name="components" type="ffprobe:pixelFormatComponentsType" minOccurs="0" maxOccurs="1"/>
</xsd:sequence>
<xsd:attribute name="name" type="xsd:string" use="required"/>
<xsd:attribute name="nb_components" type="xsd:int" use="required"/>
<xsd:attribute name="log2_chroma_w" type="xsd:int"/>
<xsd:attribute name="log2_chroma_h" type="xsd:int"/>
<xsd:attribute name="bits_per_pixel" type="xsd:int"/>
</xsd:complexType>
<xsd:complexType name="pixelFormatsType">
<xsd:sequence>
<xsd:element name="pixel_format" type="ffprobe:pixelFormatType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
</xsd:schema>

View File

@@ -1,11 +1,11 @@
# Port on which the server is listening. You must select a different
# port from your standard HTTP web server if it is running on the same
# computer.
HTTPPort 8090
Port 8090
# Address on which the server is bound. Only useful if you have
# several network interfaces.
HTTPBindAddress 0.0.0.0
BindAddress 0.0.0.0
# Number of simultaneous HTTP connections that can be handled. It has
# to be defined *before* the MaxClients parameter, since it defines the
@@ -82,7 +82,6 @@ Feed feed1.ffm
# ra : RealNetworks-compatible stream. Audio only.
# mpjpeg : Multipart JPEG (works with Netscape without any plugin)
# jpeg : Generate a single JPEG image.
# mjpeg : Generate a M-JPEG stream.
# asf : ASF compatible streaming (Windows Media Player format).
# swf : Macromedia Flash compatible stream
# avi : AVI format (MPEG-4 video, MPEG audio sound)
@@ -317,7 +316,7 @@ StartSendOnKey
#AVPresetVideo baseline
#AVOptionVideo flags +global_header
#
#AudioCodec aac
#AudioCodec libfaac
#AudioBitRate 32
#AudioChannels 2
#AudioSampleRate 22050

View File

@@ -1,5 +1,4 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle ffserver Documentation
@titlepage
@@ -67,12 +66,12 @@ http://@var{ffserver_ip_address}:@var{http_port}/@var{feed_name}
where @var{ffserver_ip_address} is the IP address of the machine where
@command{ffserver} is installed, @var{http_port} is the port number of
the HTTP server (configured through the @option{HTTPPort} option), and
the HTTP server (configured through the @option{Port} option), and
@var{feed_name} is the name of the corresponding feed defined in the
configuration file.
Each feed is associated to a file which is stored on disk. This stored
file is used to send pre-recorded data to a player as fast as
file is used to allow to send pre-recorded data to a player as fast as
possible when new content is added in real-time to the stream.
A "live-stream" or "stream" is a resource published by
@@ -102,7 +101,7 @@ http://@var{ffserver_ip_address}:@var{rtsp_port}/@var{stream_name}[@var{options}
the configuration file. @var{options} is a list of options specified
after the URL which affects how the stream is served by
@command{ffserver}. @var{http_port} and @var{rtsp_port} are the HTTP
and RTSP ports configured with the options @var{HTTPPort} and
and RTSP ports configured with the options @var{Port} and
@var{RTSPPort} respectively.
In case the stream is associated to a feed, the encoding parameters
@@ -118,8 +117,7 @@ Multiple streams can be connected to the same feed.
For example, you can have a situation described by the following
graph:
@verbatim
@example
_________ __________
| | | |
ffmpeg 1 -----| feed 1 |-----| stream 1 |
@@ -144,8 +142,7 @@ ffmpeg 2 -----| feed 3 |-----| stream 4 |
| | | |
| file 1 |-----| stream 5 |
|_________| |__________|
@end verbatim
@end example
@anchor{FFM}
@section FFM, FFM2 formats
@@ -206,9 +203,11 @@ WARNING: trying to stream test1.mpg doesn't work with WMP as it tries to
transfer the entire file before starting to play.
The same is true of AVI files.
You should edit the @file{ffserver.conf} file to suit your needs (in
terms of frame rates etc). Then install @command{ffserver} and
@command{ffmpeg}, write a script to start them up, and off you go.
@section What happens next?
You should edit the ffserver.conf file to suit your needs (in terms of
frame rates etc). Then install ffserver and ffmpeg, write a script to start
them up, and off you go.
@section What else can it do?
@@ -355,29 +354,20 @@ allow everybody else.
@section Global options
@table @option
@item HTTPPort @var{port_number}
@item Port @var{port_number}
@item RTSPPort @var{port_number}
@var{HTTPPort} sets the HTTP server listening TCP port number,
@var{RTSPPort} sets the RTSP server listening TCP port number.
@var{Port} is the equivalent of @var{HTTPPort} and is deprecated.
You must select a different port from your standard HTTP web server if
it is running on the same computer.
Set TCP port number on which the HTTP/RTSP server is listening. You
must select a different port from your standard HTTP web server if it
is running on the same computer.
If not specified, no corresponding server will be created.
@item HTTPBindAddress @var{ip_address}
@item BindAddress @var{ip_address}
@item RTSPBindAddress @var{ip_address}
Set address on which the HTTP/RTSP server is bound. Only useful if you
have several network interfaces.
@var{BindAddress} is the equivalent of @var{HTTPBindAddress} and is
deprecated.
@item MaxHTTPConnections @var{n}
Set number of simultaneous HTTP connections that can be handled. It
has to be defined @emph{before} the @option{MaxClients} parameter,
@@ -411,12 +401,6 @@ ignored, and the log is written to standard output.
Set no-daemon mode. This option is currently ignored since now
@command{ffserver} will always work in no-daemon mode, and is
deprecated.
@item UseDefaults
@item NoDefaults
Control whether default codec options are used for the all streams or not.
Each stream may overwrite this setting for its own. Default is @var{UseDefaults}.
The lastest occurrence overrides previous if multiple definitions.
@end table
@section Feed section
@@ -580,11 +564,6 @@ deprecated in favor of @option{Metadata}.
@item Metadata @var{key} @var{value}
Set metadata value on the output stream.
@item UseDefaults
@item NoDefaults
Control whether default codec options are used for the stream or not.
Default is @var{UseDefaults} unless disabled globally.
@item NoAudio
@item NoVideo
Suppress audio/video.
@@ -603,9 +582,8 @@ Set sampling frequency for audio. When using low bitrates, you should
lower this frequency to 22050 or 11025. The supported frequencies
depend on the selected audio codec.
@item AVOptionAudio [@var{codec}:]@var{option} @var{value} (@emph{encoding,audio})
Set generic or private option for audio stream.
Private option must be prefixed with codec name or codec must be defined before.
@item AVOptionAudio @var{option} @var{value} (@emph{encoding,audio})
Set generic option for audio stream.
@item AVPresetAudio @var{preset} (@emph{encoding,audio})
Set preset for audio stream.
@@ -682,9 +660,8 @@ Set video @option{qdiff} encoding option.
@item DarkMask @var{float} (@emph{encoding,video})
Set @option{lumi_mask}/@option{dark_mask} encoding options.
@item AVOptionVideo [@var{codec}:]@var{option} @var{value} (@emph{encoding,video})
Set generic or private option for video stream.
Private option must be prefixed with codec name or codec must be defined before.
@item AVOptionVideo @var{option} @var{value} (@emph{encoding,video})
Set generic option for video stream.
@item AVPresetVideo @var{preset} (@emph{encoding,video})
Set preset for video stream.

View File

@@ -3,7 +3,7 @@ representing a number as input, which may be followed by one of the SI
unit prefixes, for example: 'K', 'M', or 'G'.
If 'i' is appended to the SI unit prefix, the complete prefix will be
interpreted as a unit prefix for binary multiples, which are based on
interpreted as a unit prefix for binary multiplies, which are based on
powers of 1024 instead of powers of 1000. Appending 'B' to the SI unit
prefix multiplies the value by 8. This allows using, for example:
'KB', 'MiB', 'G' and 'B' as number suffixes.
@@ -36,10 +36,8 @@ Possible forms of stream specifiers are:
Matches the stream with this index. E.g. @code{-threads:1 4} would set the
thread count for the second stream to 4.
@item @var{stream_type}[:@var{stream_index}]
@var{stream_type} is one of following: 'v' or 'V' for video, 'a' for audio, 's'
for subtitle, 'd' for data, and 't' for attachments. 'v' matches all video
streams, 'V' only matches video streams which are not attached pictures, video
thumbnails or cover arts. If @var{stream_index} is given, then it matches
@var{stream_type} is one of following: 'v' for video, 'a' for audio, 's' for subtitle,
'd' for data, and 't' for attachments. If @var{stream_index} is given, then it matches
stream number @var{stream_index} of this type. Otherwise, it matches all
streams of this type.
@item p:@var{program_id}[:@var{stream_index}]
@@ -48,16 +46,6 @@ in the program with the id @var{program_id}. Otherwise, it matches all streams i
program.
@item #@var{stream_id} or i:@var{stream_id}
Match the stream by stream id (e.g. PID in MPEG-TS container).
@item m:@var{key}[:@var{value}]
Matches streams with the metadata tag @var{key} having the specified value. If
@var{value} is not given, matches streams that contain the given tag with any
value.
@item u
Matches streams with usable configuration, the codec must be defined and the
essential information such as video dimension or audio sample rate must be present.
Note that in @command{ffmpeg}, matching by metadata will only work properly for
input files.
@end table
@section Generic options
@@ -108,10 +96,7 @@ Print detailed information about the filter name @var{filter_name}. Use the
Show version.
@item -formats
Show available formats (including devices).
@item -devices
Show available devices.
Show available formats.
@item -codecs
Show all codecs known to libavcodec.
@@ -146,22 +131,6 @@ Show channel names and standard channel layouts.
@item -colors
Show recognized color names.
@item -sources @var{device}[,@var{opt1}=@var{val1}[,@var{opt2}=@var{val2}]...]
Show autodetected sources of the intput device.
Some devices may provide system-dependent source names that cannot be autodetected.
The returned list cannot be assumed to be always complete.
@example
ffmpeg -sources pulse,server=192.168.0.4
@end example
@item -sinks @var{device}[,@var{opt1}=@var{val1}[,@var{opt2}=@var{val2}]...]
Show autodetected sinks of the output device.
Some devices may provide system-dependent sink names that cannot be autodetected.
The returned list cannot be assumed to be always complete.
@example
ffmpeg -sinks pulse,server=192.168.0.4
@end example
@item -loglevel [repeat+]@var{loglevel} | -v [repeat+]@var{loglevel}
Set the logging level used by the library.
Adding "repeat+" indicates that repeated log output should not be compressed
@@ -170,38 +139,37 @@ omitted. "repeat" can also be used alone.
If "repeat" is used alone, and with no prior loglevel set, the default
loglevel will be used. If multiple loglevel parameters are given, using
'repeat' will not change the loglevel.
@var{loglevel} is a string or a number containing one of the following values:
@var{loglevel} is a number or a string containing one of the following values:
@table @samp
@item quiet, -8
@item quiet
Show nothing at all; be silent.
@item panic, 0
@item panic
Only show fatal errors which could lead the process to crash, such as
an assertion failure. This is not currently used for anything.
@item fatal, 8
and assert failure. This is not currently used for anything.
@item fatal
Only show fatal errors. These are errors after which the process absolutely
cannot continue.
@item error, 16
cannot continue after.
@item error
Show all errors, including ones which can be recovered from.
@item warning, 24
@item warning
Show all warnings and errors. Any message related to possibly
incorrect or unexpected events will be shown.
@item info, 32
@item info
Show informative messages during processing. This is in addition to
warnings and errors. This is the default value.
@item verbose, 40
@item verbose
Same as @code{info}, except more verbose.
@item debug, 48
@item debug
Show everything, including debugging information.
@item trace, 56
@end table
By default the program logs to stderr. If coloring is supported by the
By default the program logs to stderr, if coloring is supported by the
terminal, colors are used to mark errors and warnings. Log coloring
can be disabled setting the environment variable
@env{AV_LOG_FORCE_NOCOLOR} or @env{NO_COLOR}, or can be forced setting
the environment variable @env{AV_LOG_FORCE_COLOR}.
The use of the environment variable @env{NO_COLOR} is deprecated and
will be dropped in a future FFmpeg version.
will be dropped in a following FFmpeg version.
@item -report
Dump full command line and console output to a file named
@@ -210,29 +178,21 @@ directory.
This file can be useful for bug reports.
It also implies @code{-loglevel verbose}.
Setting the environment variable @env{FFREPORT} to any value has the
Setting the environment variable @code{FFREPORT} to any value has the
same effect. If the value is a ':'-separated key=value sequence, these
options will affect the report; option values must be escaped if they
options will affect the report; options values must be escaped if they
contain special characters or the options delimiter ':' (see the
``Quoting and escaping'' section in the ffmpeg-utils manual).
The following options are recognized:
``Quoting and escaping'' section in the ffmpeg-utils manual). The
following option is recognized:
@table @option
@item file
set the file name to use for the report; @code{%p} is expanded to the name
of the program, @code{%t} is expanded to a timestamp, @code{%%} is expanded
to a plain @code{%}
@item level
set the log verbosity level using a numerical value (see @code{-loglevel}).
set the log level
@end table
For example, to output a report to a file named @file{ffreport.log}
using a log level of @code{32} (alias for log level @code{info}):
@example
FFREPORT=file=ffreport.log:level=32 ffmpeg -i input output
@end example
Errors in parsing the environment variable are not fatal, and will not
appear in the report.
@@ -267,14 +227,10 @@ Possible flags for this option are:
@item sse4.1
@item sse4.2
@item avx
@item avx2
@item xop
@item fma3
@item fma4
@item 3dnow
@item 3dnowext
@item bmi1
@item bmi2
@item cmov
@end table
@item ARM
@@ -285,13 +241,6 @@ Possible flags for this option are:
@item vfp
@item vfpv3
@item neon
@item setend
@end table
@item AArch64
@table @samp
@item armv8
@item vfp
@item neon
@end table
@item PowerPC
@table @samp
@@ -311,41 +260,8 @@ Possible flags for this option are:
@end table
@item -opencl_bench
This option is used to benchmark all available OpenCL devices and print the
results. This option is only available when FFmpeg has been compiled with
@code{--enable-opencl}.
When FFmpeg is configured with @code{--enable-opencl}, the options for the
global OpenCL context are set via @option{-opencl_options}. See the
"OpenCL Options" section in the ffmpeg-utils manual for the complete list of
supported options. Amongst others, these options include the ability to select
a specific platform and device to run the OpenCL code on. By default, FFmpeg
will run on the first device of the first platform. While the options for the
global OpenCL context provide flexibility to the user in selecting the OpenCL
device of their choice, most users would probably want to select the fastest
OpenCL device for their system.
This option assists the selection of the most efficient configuration by
identifying the appropriate device for the user's system. The built-in
benchmark is run on all the OpenCL devices and the performance is measured for
each device. The devices in the results list are sorted based on their
performance with the fastest device listed first. The user can subsequently
invoke @command{ffmpeg} using the device deemed most appropriate via
@option{-opencl_options} to obtain the best performance for the OpenCL
accelerated code.
Typical usage to use the fastest OpenCL device involve the following steps.
Run the command:
@example
ffmpeg -opencl_bench
@end example
Note down the platform ID (@var{pidx}) and device ID (@var{didx}) of the first
i.e. fastest device in the list.
Select the platform and device using the command:
@example
ffmpeg -opencl_options platform_idx=@var{pidx}:device_idx=@var{didx} ...
@end example
Benchmark all available OpenCL devices and show the results. This option
is only available when FFmpeg has been compiled with @code{--enable-opencl}.
@item -opencl_options options (@emph{global})
Set OpenCL environment options. This option is only available when

View File

@@ -98,7 +98,7 @@ Buffer references ownership and permissions
The AVFilterLink structure has a few AVFilterBufferRef fields. The
cur_buf and out_buf were used with the deprecated
start_frame/draw_slice/end_frame API and should no longer be used.
src_buf and partial_buf are used by libavfilter internally
src_buf, cur_buf_copy and partial_buf are used by libavfilter internally
and must not be accessed by filters.
Reference permissions
@@ -232,8 +232,7 @@ Frame scheduling
one of its inputs, repeatedly until at least one frame has been pushed.
Return values:
if request_frame could produce a frame, or at least make progress
towards producing a frame, it should return 0;
if request_frame could produce a frame, it should return 0;
if it could not for temporary reasons, it should return AVERROR(EAGAIN);
if it could not because there are no more frames, it should return
AVERROR_EOF.
@@ -245,18 +244,20 @@ Frame scheduling
push_one_frame();
return 0;
}
input = input_where_a_frame_is_most_needed();
ret = ff_request_frame(input);
if (ret == AVERROR_EOF) {
process_eof_on_input();
} else if (ret < 0) {
return ret;
while (!frame_pushed) {
input = input_where_a_frame_is_most_needed();
ret = ff_request_frame(input);
if (ret == AVERROR_EOF) {
process_eof_on_input();
} else if (ret < 0) {
return ret;
}
}
return 0;
Note that, except for filters that can have queued frames, request_frame
does not push frames: it requests them to its input, and as a reaction,
the filter_frame method possibly will be called and do the work.
the filter_frame method will be called and do the work.
Legacy API
==========

File diff suppressed because it is too large Load Diff

View File

@@ -23,7 +23,7 @@ Reduce buffering.
@item probesize @var{integer} (@emph{input})
Set probing size in bytes, i.e. the size of the data to analyze to get
stream information. A higher value will enable detecting more
stream information. A higher value will allow to detect more
information in case it is dispersed into the stream, but will increase
latency. Must be an integer not lesser than 32. It is 5000000 by default.
@@ -37,8 +37,6 @@ Possible values:
@table @samp
@item ignidx
Ignore index.
@item fastseek
Enable fast, but inaccurate seeks for some formats.
@item genpts
Generate PTS.
@item nofillin
@@ -57,14 +55,6 @@ Do not merge side data.
Enable RTP MP4A-LATM payload.
@item nobuffer
Reduce the latency introduced by optional buffering
@item bitexact
Only write platform-, build- and time-independent data.
This ensures that file and data checksums are reproducible and match between
platforms. Its primary use is for regression testing.
@item shortest
Stop muxing at the end of the shortest stream.
It may be needed to increase max_interleave_delta to avoid flusing the longer
streams before EOF.
@end table
@item seek2any @var{integer} (@emph{input})
@@ -73,7 +63,7 @@ Default is 0.
@item analyzeduration @var{integer} (@emph{input})
Specify how many microseconds are analyzed to probe the input. A
higher value will enable detecting more accurate information, but will
higher value will allow to detect more accurate information, but will
increase latency. It defaults to 5,000,000 microseconds = 5 seconds.
@item cryptokey @var{hexadecimal string} (@emph{input})
@@ -131,27 +121,8 @@ Consider all spec non compliancies as errors.
Consider things that a sane encoder should not do as an error.
@end table
@item max_interleave_delta @var{integer} (@emph{output})
Set maximum buffering duration for interleaving. The duration is
expressed in microseconds, and defaults to 1000000 (1 second).
To ensure all the streams are interleaved correctly, libavformat will
wait until it has at least one packet for each stream before actually
writing any packets to the output file. When some streams are
"sparse" (i.e. there are large gaps between successive packets), this
can result in excessive buffering.
This field specifies the maximum difference between the timestamps of the
first and the last packet in the muxing queue, above which libavformat
will output a packet regardless of whether it has queued a packet for all
the streams.
If set to 0, libavformat will continue buffering packets until it has
a packet for each stream, regardless of the maximum timestamp
difference between the buffered packets.
@item use_wallclock_as_timestamps @var{integer} (@emph{input})
Use wallclock as timestamps if set to 1. Default is 0.
Use wallclock as timestamps.
@item avoid_negative_ts @var{integer} (@emph{output})
@@ -197,18 +168,6 @@ The offset is added by the muxer to the output timestamps.
Specifying a positive offset means that the corresponding streams are
delayed bt the time duration specified in @var{offset}. Default value
is @code{0} (meaning that no offset is applied).
@item format_whitelist @var{list} (@emph{input})
"," separated list of allowed demuxers. By default all are allowed.
@item dump_separator @var{string} (@emph{input})
Separator used to separate the fields printed on the command line about the
Stream parameters.
For example to separate the fields with newlines and indention:
@example
ffprobe -dump_separator "
" -i ~/videos/matrixbench_mpeg2.mpg
@end example
@end table
@c man end FORMAT OPTIONS

View File

@@ -1,5 +1,4 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle General Documentation
@titlepage
@@ -53,6 +52,14 @@ instructions for installing the libraries.
Then pass @code{--enable-libopencore-amrnb} and/or
@code{--enable-libopencore-amrwb} to configure to enable them.
@subsection VisualOn AAC encoder library
FFmpeg can make use of the VisualOn AACenc library for AAC encoding.
Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the
instructions for installing the library.
Then pass @code{--enable-libvo-aacenc} to configure to enable it.
@subsection VisualOn AMR-WB encoder library
FFmpeg can make use of the VisualOn AMR-WBenc library for AMR-WB encoding.
@@ -101,21 +108,6 @@ Go to @url{http://www.wavpack.com/} and follow the instructions for
installing the library. Then pass @code{--enable-libwavpack} to configure to
enable it.
@section OpenH264
FFmpeg can make use of the OpenH264 library for H.264 encoding and decoding.
Go to @url{http://www.openh264.org/} and follow the instructions for
installing the library. Then pass @code{--enable-libopenh264} to configure to
enable it.
For decoding, this library is much more limited than the built-in decoder
in libavcodec; currently, this library lacks support for decoding B-frames
and some other main/high profile features. (It currently only supports
constrained baseline profile and CABAC.) Using it is mostly useful for
testing and for taking advantage of Cisco's patent portfolio license
(@url{http://www.openh264.org/BINARY_LICENSE.txt}).
@section x264
FFmpeg can make use of the x264 library for H.264 encoding.
@@ -138,20 +130,12 @@ Go to @url{http://x265.org/developers.html} and follow the instructions
for installing the library. Then pass @code{--enable-libx265} to configure
to enable it.
@float NOTE
@float note
x265 is under the GNU Public License Version 2 or later
(see @url{http://www.gnu.org/licenses/old-licenses/gpl-2.0.html} for
details), you must upgrade FFmpeg's license to GPL in order to use it.
@end float
@section kvazaar
FFmpeg can make use of the kvazaar library for HEVC encoding.
Go to @url{https://github.com/ultravideo/kvazaar} and follow the
instructions for installing the library. Then pass
@code{--enable-libkvazaar} to configure to enable it.
@section libilbc
iLBC is a narrowband speech codec that has been made freely available
@@ -159,7 +143,7 @@ by Google as part of the WebRTC project. libilbc is a packaging friendly
copy of the iLBC codec. FFmpeg can make use of the libilbc library for
iLBC encoding and decoding.
Go to @url{https://github.com/TimothyGu/libilbc} and follow the instructions for
Go to @url{https://github.com/dekkers/libilbc} and follow the instructions for
installing the library. Then pass @code{--enable-libilbc} to configure to
enable it.
@@ -172,6 +156,12 @@ Go to @url{http://sourceforge.net/projects/zapping/} and follow the instructions
installing the library. Then pass @code{--enable-libzvbi} to configure to
enable it.
@float NOTE
libzvbi is licensed under the GNU General Public License Version 2 or later
(see @url{http://www.gnu.org/licenses/old-licenses/gpl-2.0.html} for details),
you must upgrade FFmpeg's license to GPL in order to use it.
@end float
@section AviSynth
FFmpeg can read AviSynth scripts as input. To enable support, pass
@@ -180,8 +170,8 @@ included in compat/avisynth/, which allows the user to enable support
without needing to search for these headers themselves.
For Windows, supported AviSynth variants are
@url{http://avisynth.nl, AviSynth 2.6 RC1 or higher} for 32-bit builds and
@url{http://avs-plus.net, AviSynth+ r1718 or higher} for 32-bit and 64-bit builds.
@url{http://avisynth.nl, AviSynth 2.5 or 2.6} for 32-bit builds and
@url{http://avs-plus.net, AviSynth+ 0.1} for 32-bit and 64-bit builds.
For Linux and OS X, the supported AviSynth variant is
@url{https://github.com/avxsynth/avxsynth, AvxSynth}.
@@ -193,17 +183,6 @@ end user having AviSynth or AvxSynth installed - they'll only need to be
installed to use AviSynth scripts (obviously).
@end float
@section Intel QuickSync Video
FFmpeg can use Intel QuickSync Video (QSV) for accelerated encoding and decoding
of multiple codecs. To use QSV, FFmpeg must be linked against the @code{libmfx}
dispatcher, which loads the actual decoding libraries.
The dispatcher is open source and can be downloaded from
@url{https://github.com/lu-zero/mfx_dispatch.git}. FFmpeg needs to be configured
with the @code{--enable-libmfx} option and @code{pkg-config} needs to be able to
locate the dispatcher's @code{.pc} files.
@chapter Supported File Formats, Codecs or Features
@@ -216,14 +195,9 @@ library:
@multitable @columnfractions .4 .1 .1 .4
@item Name @tab Encoding @tab Decoding @tab Comments
@item 3dostr @tab @tab X
@item 4xm @tab @tab X
@tab 4X Technologies format, used in some games.
@item 8088flex TMV @tab @tab X
@item AAX @tab @tab X
@tab Audible Enhanced Audio format, used in audiobooks.
@item AA @tab @tab X
@tab Audible Format 2, 3, and 4, used in audiobooks.
@item ACT Voice @tab @tab X
@tab contains G.729 audio
@item Adobe Filmstrip @tab X @tab X
@@ -235,15 +209,10 @@ library:
@tab Multimedia format used in game Heart Of Darkness.
@item Apple HTTP Live Streaming @tab @tab X
@item Artworx Data Format @tab @tab X
@item Interplay ACM @tab @tab X
@tab Audio only format used in some Interplay games.
@item ADP @tab @tab X
@tab Audio format used on the Nintendo Gamecube.
@item AFC @tab @tab X
@tab Audio format used on the Nintendo Gamecube.
@item ADS/SS2 @tab @tab X
@tab Audio format used on the PS2.
@item APNG @tab X @tab X
@item ASF @tab X @tab X
@item AST @tab X @tab X
@tab Audio format used on the Nintendo Wii.
@@ -264,8 +233,6 @@ library:
@tab Used in Z and Z95 games.
@item Brute Force & Ignorance @tab @tab X
@tab Used in the game Flash Traffic: City of Angels.
@item BFSTM @tab @tab X
@tab Audio format used on the Nintendo WiiU (based on BRSTM).
@item BRSTM @tab @tab X
@tab Audio format used on the Nintendo Wii.
@item BWF @tab X @tab X
@@ -276,14 +243,9 @@ library:
@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 Digital Speech Standard (DSS) @tab @tab X
@item Canopus HQ @tab @tab X
@item Canopus HQA @tab @tab X
@item Canopus HQX @tab @tab X
@item CD+G @tab @tab X
@tab Video format used by CD+G karaoke disks
@item Phantom Cine @tab @tab X
@item Cineform HD @tab @tab X
@item Commodore CDXL @tab @tab X
@tab Amiga CD video format
@item Core Audio Format @tab X @tab X
@@ -295,10 +257,8 @@ library:
@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 DCSTR @tab @tab X
@item DFA @tab @tab X
@tab This format is used in Chronomaster game
@item DirectDraw Surface @tab @tab X
@item DSD Stream File (DSF) @tab @tab X
@item DV video @tab X @tab X
@item DXA @tab @tab X
@@ -322,8 +282,6 @@ library:
@item G.723.1 @tab X @tab X
@item G.729 BIT @tab X @tab X
@item G.729 raw @tab @tab X
@item GENH @tab @tab X
@tab Audio format for various games.
@item GIF Animation @tab X @tab X
@item GXF @tab X @tab X
@tab General eXchange Format SMPTE 360M, used by Thomson Grass Valley
@@ -346,7 +304,6 @@ library:
@tab A format generated by IndigoVision 8000 video server.
@item IVF (On2) @tab X @tab X
@tab A format used by libvpx
@item Internet Video Recording @tab @tab X
@item IRCAM @tab X @tab X
@item LATM @tab X @tab X
@item LMLM4 @tab @tab X
@@ -383,8 +340,6 @@ library:
@tab also known as DVB Transport Stream
@item MPEG-4 @tab X @tab X
@tab MPEG-4 is a variant of QuickTime.
@item MSF @tab @tab X
@tab Audio format used on the PS3.
@item Mirillis FIC video @tab @tab X
@tab No cursor rendering.
@item MIME multipart JPEG @tab X @tab
@@ -468,7 +423,6 @@ library:
@item Redirector @tab @tab X
@item RedSpark @tab @tab X
@item Renderware TeXture Dictionary @tab @tab X
@item Resolume DXV @tab @tab X
@item RL2 @tab @tab X
@tab Audio and video format used in some games by Entertainment Software Partners.
@item RPL/ARMovie @tab @tab X
@@ -500,23 +454,14 @@ library:
@item Sony Wave64 (W64) @tab X @tab X
@item SoX native format @tab X @tab X
@item SUN AU format @tab X @tab X
@item SUP raw PGS subtitles @tab @tab X
@item SVAG @tab @tab X
@tab Audio format used in Konami PS2 games.
@item TDSC @tab @tab X
@item Text files @tab @tab X
@item THP @tab @tab X
@tab Used on the Nintendo GameCube.
@item Tiertex Limited SEQ @tab @tab X
@tab Tiertex .seq files used in the DOS CD-ROM version of the game Flashback.
@item True Audio @tab X @tab X
@item VAG @tab @tab X
@tab Audio format used in many Sony PS2 games.
@item True Audio @tab @tab X
@item VC-1 test bitstream @tab X @tab X
@item Vidvox Hap @tab X @tab X
@item Vivo @tab @tab X
@item VPK @tab @tab X
@tab Audio format used in Sony PS games.
@item WAV @tab X @tab X
@item WavPack @tab X @tab X
@item WebM @tab X @tab X
@@ -527,12 +472,8 @@ library:
@tab Multimedia format used in Westwood Studios games.
@item Westwood Studios VQA @tab @tab X
@tab Multimedia format used in Westwood Studios games.
@item Wideband Single-bit Data (WSD) @tab @tab X
@item WVE @tab @tab X
@item XMV @tab @tab X
@tab Microsoft video container used in Xbox games.
@item XVAG @tab @tab X
@tab Audio format used on the PS3.
@item xWMA @tab @tab X
@tab Microsoft audio container used by XAudio 2.
@item eXtended BINary text (XBIN) @tab @tab X
@@ -554,7 +495,6 @@ following image formats are supported:
@item Alias PIX @tab X @tab X
@tab Alias/Wavefront PIX image format
@item animated GIF @tab X @tab X
@item APNG @tab X @tab X
@item BMP @tab X @tab X
@tab Microsoft BMP image
@item BRender PIX @tab @tab X
@@ -651,7 +591,6 @@ following image formats are supported:
@item Bethesda VID video @tab @tab X
@tab Used in some games from Bethesda Softworks.
@item Bink Video @tab @tab X
@item BitJazz SheerVideo @tab @tab X
@item Bitmap Brothers JV video @tab @tab X
@item y41p Brooktree uncompressed 4:1:1 12-bit @tab X @tab X
@item Brute Force & Ignorance @tab @tab X
@@ -686,8 +625,6 @@ following image formats are supported:
@tab fourcc: DUCK
@item Duck TrueMotion 2.0 @tab @tab X
@tab fourcc: TM20
@item Duck TrueMotion 2.0 RT @tab @tab X
@tab fourcc: TR20
@item DV (Digital Video) @tab X @tab X
@item Dxtory capture format @tab @tab X
@item Feeble Files/ScummVM DXA @tab @tab X
@@ -709,17 +646,15 @@ following image formats are supported:
@tab Sorenson H.263 used in Flash
@item Forward Uncompressed @tab @tab X
@item Fraps @tab @tab X
@item Go2Meeting @tab @tab X
@tab fourcc: G2M2, G2M3
@item Go2Webinar @tab @tab X
@tab fourcc: G2M4
@item H.261 @tab X @tab X
@item H.263 / H.263-1996 @tab X @tab X
@item H.263+ / H.263-1998 / H.263 version 2 @tab X @tab X
@item H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 @tab E @tab X
@tab encoding supported through external library libx264 and OpenH264
@tab encoding supported through external library libx264
@item HEVC @tab X @tab X
@tab encoding supported through external library libx265 and libkvazaar
@tab encoding supported through the external library libx265
@item HNM version 4 @tab @tab X
@item HuffYUV @tab X @tab X
@item HuffYUV FFmpeg variant @tab X @tab X
@@ -754,7 +689,6 @@ following image formats are supported:
@item LucasArts SANM/Smush @tab @tab X
@tab Used in LucasArts games / SMUSH animations.
@item lossless MJPEG @tab X @tab X
@item MagicYUV Video @tab @tab X
@item Microsoft ATC Screen @tab @tab X
@tab Also known as Microsoft Screen 3.
@item Microsoft Expression Encoder Screen @tab @tab X
@@ -819,7 +753,6 @@ following image formats are supported:
@tab Texture dictionaries used by the Renderware Engine.
@item RL2 video @tab @tab X
@tab used in some games by Entertainment Software Partners
@item Screenpresso @tab @tab X
@item Sierra VMD video @tab @tab X
@tab Used in Sierra VMD files.
@item Silicon Graphics Motion Video Compressor 1 (MVC1) @tab @tab X
@@ -886,13 +819,12 @@ following image formats are supported:
@item Name @tab Encoding @tab Decoding @tab Comments
@item 8SVX exponential @tab @tab X
@item 8SVX fibonacci @tab @tab X
@item AAC @tab EX @tab X
@tab encoding supported through internal encoder and external library libfdk-aac
@item AAC+ @tab E @tab IX
@tab encoding supported through external library libfdk-aac
@item AAC+ @tab E @tab X
@tab encoding supported through external library libaacplus
@item AAC @tab E @tab X
@tab encoding supported through external library libfaac and libvo-aacenc
@item AC-3 @tab IX @tab IX
@item ADPCM 4X Movie @tab @tab X
@item APDCM Yamaha AICA @tab @tab X
@item ADPCM CDROM XA @tab @tab X
@item ADPCM Creative Technology @tab @tab X
@tab 16 -> 4, 8 -> 4, 8 -> 3, 8 -> 2
@@ -927,8 +859,7 @@ following image formats are supported:
@item ADPCM MS IMA @tab X @tab X
@item ADPCM Nintendo Gamecube AFC @tab @tab X
@item ADPCM Nintendo Gamecube DTK @tab @tab X
@item ADPCM Nintendo THP @tab @tab X
@item APDCM Playstation @tab @tab X
@item ADPCM Nintendo Gamecube THP @tab @tab X
@item ADPCM QT IMA @tab X @tab X
@item ADPCM SEGA CRI ADX @tab X @tab X
@tab Used in Sega Dreamcast games.
@@ -936,7 +867,7 @@ following image formats are supported:
@item ADPCM Sound Blaster Pro 2-bit @tab @tab X
@item ADPCM Sound Blaster Pro 2.6-bit @tab @tab X
@item ADPCM Sound Blaster Pro 4-bit @tab @tab X
@item ADPCM VIMA @tab @tab X
@item ADPCM VIMA
@tab Used in LucasArts SMUSH animations.
@item ADPCM Westwood Studios IMA @tab @tab X
@tab Used in Westwood Studios games like Command and Conquer.
@@ -957,18 +888,14 @@ following image formats are supported:
@tab decoding supported through external library libcelt
@item Delphine Software International CIN audio @tab @tab X
@tab Codec used in Delphine Software International games.
@item Digital Speech Standard - Standard Play mode (DSS SP) @tab @tab X
@item Discworld II BMV Audio @tab @tab X
@item COOK @tab @tab X
@tab All versions except 5.1 are supported.
@item DCA (DTS Coherent Acoustics) @tab X @tab X
@tab supported extensions: XCh, XXCH, X96, XBR, XLL, LBR (partially)
@item DPCM id RoQ @tab X @tab X
@tab Used in Quake III, Jedi Knight 2 and other computer games.
@item DPCM Interplay @tab @tab X
@tab Used in various Interplay computer games.
@item DPCM Squareroot-Delta-Exact @tab @tab X
@tab Used in various games.
@item DPCM Sierra Online @tab @tab X
@tab Used in Sierra Online game audio files.
@item DPCM Sol @tab @tab X
@@ -979,12 +906,11 @@ following image formats are supported:
@item DSD (Direct Stream Digitial), least significant bit first, planar @tab @tab X
@item DSD (Direct Stream Digitial), most significant bit first, planar @tab @tab X
@item DSP Group TrueSpeech @tab @tab X
@item DST (Direct Stream Transfer) @tab @tab X
@item DV audio @tab @tab X
@item Enhanced AC-3 @tab X @tab X
@item EVRC (Enhanced Variable Rate Codec) @tab @tab X
@item FLAC (Free Lossless Audio Codec) @tab X @tab IX
@item G.723.1 @tab X @tab X
@item G.723.1 @tab X @tab X
@item G.729 @tab @tab X
@item GSM @tab E @tab X
@tab encoding supported through external library libgsm
@@ -994,10 +920,9 @@ following image formats are supported:
@item iLBC (Internet Low Bitrate Codec) @tab E @tab E
@tab encoding and decoding supported through external library libilbc
@item IMC (Intel Music Coder) @tab @tab X
@item Interplay ACM @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 X @tab X
@item MLP (Meridian Lossless Packing) @tab @tab X
@tab Used in DVD-Audio discs.
@item Monkey's Audio @tab @tab X
@item MP1 (MPEG audio layer 1) @tab @tab IX
@@ -1010,8 +935,8 @@ following image formats are supported:
@item Musepack SV8 @tab @tab X
@item Nellymoser Asao @tab X @tab X
@item On2 AVC (Audio for Video Codec) @tab @tab X
@item Opus @tab E @tab X
@tab encoding supported through external library libopus
@item Opus @tab E @tab E
@tab supported through external library libopus
@item PCM A-law @tab X @tab X
@item PCM mu-law @tab X @tab X
@item PCM signed 8-bit planar @tab X @tab X
@@ -1064,7 +989,7 @@ following image formats are supported:
@tab supported through external library libspeex
@item TAK (Tom's lossless Audio Kompressor) @tab @tab X
@item True Audio (TTA) @tab X @tab X
@item TrueHD @tab X @tab X
@item TrueHD @tab @tab X
@tab Used in HD-DVD and Blu-Ray discs.
@item TwinVQ (VQF flavor) @tab @tab X
@item VIMA @tab @tab X
@@ -1079,8 +1004,6 @@ following image formats are supported:
@item Windows Media Audio Lossless @tab @tab X
@item Windows Media Audio Pro @tab @tab X
@item Windows Media Audio Voice @tab @tab X
@item Xbox Media Audio 1 @tab @tab X
@item Xbox Media Audio 2 @tab @tab X
@end multitable
@code{X} means that encoding (resp. decoding) is supported.
@@ -1107,7 +1030,6 @@ performance on systems without hardware floating point support).
@item PJS (Phoenix) @tab @tab X @tab @tab X
@item RealText @tab @tab X @tab @tab X
@item SAMI @tab @tab X @tab @tab X
@item Spruce format (STL) @tab @tab X @tab @tab X
@item SSA/ASS @tab X @tab X @tab X @tab X
@item SubRip (SRT) @tab X @tab X @tab X @tab X
@item SubViewer v1 @tab @tab X @tab @tab X
@@ -1133,7 +1055,6 @@ performance on systems without hardware floating point support).
@item HLS @tab X
@item HTTP @tab X
@item HTTPS @tab X
@item Icecast @tab X
@item MMSH @tab X
@item MMST @tab X
@item pipe @tab X

View File

@@ -1,10 +1,9 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle Using Git to develop FFmpeg
@settitle Using git to develop FFmpeg
@titlepage
@center @titlefont{Using Git to develop FFmpeg}
@center @titlefont{Using git to develop FFmpeg}
@end titlepage
@top
@@ -13,9 +12,9 @@
@chapter Introduction
This document aims in giving some quick references on a set of useful Git
This document aims in giving some quick references on a set of useful git
commands. You should always use the extensive and detailed documentation
provided directly by Git:
provided directly by git:
@example
git --help
@@ -32,21 +31,22 @@ man git-<command>
shows information about the subcommand <command>.
Additional information could be found on the
@url{http://gitref.org, Git Reference} website.
@url{http://gitref.org, Git Reference} website
For more information about the Git project, visit the
@url{http://git-scm.com/, Git website}.
@url{http://git-scm.com/, Git website}
Consult these resources whenever you have problems, they are quite exhaustive.
What follows now is a basic introduction to Git and some FFmpeg-specific
guidelines to ease the contribution to the project.
guidelines to ease the contribution to the project
@chapter Basics Usage
@section Get Git
@section Get GIT
You can get Git from @url{http://git-scm.com/}
You can get git from @url{http://git-scm.com/}
Most distribution and operating system provide a package for it.
@@ -65,21 +65,6 @@ git clone git@@source.ffmpeg.org:ffmpeg <target>
This will put the FFmpeg sources into the directory @var{<target>} and let
you push back your changes to the remote repository.
@example
git clone gil@@ffmpeg.org:ffmpeg-web <target>
@end example
This will put the source of the FFmpeg website into the directory
@var{<target>} and let you push back your changes to the remote repository.
(Note that @var{gil} stands for GItoLite and is not a typo of @var{git}.)
If you don't have write-access to the ffmpeg-web repository, you can
create patches after making a read-only ffmpeg-web clone:
@example
git clone git://ffmpeg.org/ffmpeg-web <target>
@end example
Make sure that you do not have Windows line endings in your checkouts,
otherwise you may experience spurious compilation failures. One way to
achieve this is to run
@@ -89,7 +74,6 @@ git config --global core.autocrlf false
@end example
@anchor{Updating the source tree to the latest revision}
@section Updating the source tree to the latest revision
@example
@@ -122,7 +106,7 @@ git add [-A] <filename/dirname>
git rm [-r] <filename/dirname>
@end example
Git needs to get notified of all changes you make to your working
GIT needs to get notified of all changes you make to your working
directory that makes files appear or disappear.
Line moves across files are automatically tracked.
@@ -142,8 +126,8 @@ will show all local modifications in your working directory as unified diff.
git log <filename(s)>
@end example
You may also use the graphical tools like @command{gitview} or @command{gitk}
or the web interface available at @url{http://source.ffmpeg.org/}.
You may also use the graphical tools like gitview or gitk or the web
interface available at http://source.ffmpeg.org/
@section Checking source tree status
@@ -164,7 +148,6 @@ git diff --check
to double check your changes before committing them to avoid trouble later
on. All experienced developers do this on each and every commit, no matter
how small.
Every one of them has been saved from looking like a fool by this many times.
It's very easy for stray debug output or cosmetic modifications to slip in,
please avoid problems through this extra level of scrutiny.
@@ -187,14 +170,14 @@ to make sure you don't have untracked files or deletions.
git add [-i|-p|-A] <filenames/dirnames>
@end example
Make sure you have told Git your name and email address
Make sure you have told git your name and email address
@example
git config --global user.name "My Name"
git config --global user.email my@@email.invalid
@end example
Use @option{--global} to set the global configuration for all your Git checkouts.
Use @var{--global} to set the global configuration for all your git checkouts.
Git will select the changes to the files for commit. Optionally you can use
the interactive or the patch mode to select hunk by hunk what should be
@@ -225,7 +208,7 @@ include filenames in log messages, Git provides that information.
Possibly make the commit message have a terse, descriptive first line, an
empty line and then a full description. The first line will be used to name
the patch by @command{git format-patch}.
the patch by git format-patch.
@section Preparing a patchset
@@ -341,14 +324,12 @@ faulty commit disappear from the history.
@section Pushing changes to remote trees
@example
git push origin master --dry-run
git push
@end example
Will simulate a push of the local master branch to the default remote
(@var{origin}). And list which branches and ranges or commits would have been
pushed.
Will push the changes to the default remote (@var{origin}).
Git will prevent you from pushing changes if the local and remote trees are
out of sync. Refer to @ref{Updating the source tree to the latest revision}.
out of sync. Refer to and to sync the local tree.
@example
git remote add <name> <url>
@@ -367,24 +348,23 @@ branches matching the local ones.
@section Finding a specific svn revision
Since version 1.7.1 Git supports @samp{:/foo} syntax for specifying commits
Since version 1.7.1 git supports @var{:/foo} syntax for specifying commits
based on a regular expression. see man gitrevisions
@example
git show :/'as revision 23456'
@end example
will show the svn changeset @samp{r23456}. With older Git versions searching in
will show the svn changeset @var{r23456}. With older git versions searching in
the @command{git log} output is the easiest option (especially if a pager with
search capabilities is used).
This commit can be checked out with
@example
git checkout -b svn_23456 :/'as revision 23456'
@end example
or for Git < 1.7.1 with
or for git < 1.7.1 with
@example
git checkout -b svn_23456 $SHA1
@@ -393,7 +373,7 @@ git checkout -b svn_23456 $SHA1
where @var{$SHA1} is the commit hash from the @command{git log} output.
@chapter Pre-push checklist
@chapter pre-push checklist
Once you have a set of commits that you feel are ready for pushing,
work through the following checklist to doublecheck everything is in
@@ -404,21 +384,21 @@ Apply your common sense, but if in doubt, err on the side of caution.
First, make sure that the commits and branches you are going to push
match what you want pushed and that nothing is missing, extraneous or
wrong. You can see what will be pushed by running the git push command
with @option{--dry-run} first. And then inspecting the commits listed with
with --dry-run first. And then inspecting the commits listed with
@command{git log -p 1234567..987654}. The @command{git status} command
may help in finding local changes that have been forgotten to be added.
Next let the code pass through a full run of our test suite.
Next let the code pass through a full run of our testsuite.
@itemize
@item @command{make distclean}
@item @command{/path/to/ffmpeg/configure}
@item @command{make fate}
@item @command{make check}
@item if fate fails due to missing samples run @command{make fate-rsync} and retry
@end itemize
Make sure all your changes have been checked before pushing them, the
test suite only checks against regressions and that only to some extend. It does
testsuite only checks against regressions and that only to some extend. It does
obviously not check newly added features/code to be working unless you have
added a test for that (which is recommended).
@@ -431,5 +411,5 @@ recommended.
@chapter Server Issues
Contact the project admins at @email{root@@ffmpeg.org} if you have technical
problems with the Git server.
Contact the project admins @email{root@@ffmpeg.org} if you have technical
problems with the GIT server.

View File

@@ -1,7 +1,7 @@
@chapter Input Devices
@c man begin INPUT DEVICES
Input devices are configured elements in FFmpeg which enable accessing
Input devices are configured elements in FFmpeg which allow to access
the data coming from a multimedia device attached to your system.
When you configure your FFmpeg build, all the supported input devices
@@ -51,18 +51,6 @@ ffmpeg -f alsa -i hw:0 alsaout.wav
For more information see:
@url{http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html}
@subsection Options
@table @option
@item sample_rate
Set the sample rate in Hz. Default is 48000.
@item channels
Set the number of channels. Default is 2.
@end table
@section avfoundation
AVFoundation input device.
@@ -70,251 +58,38 @@ AVFoundation input device.
AVFoundation is the currently recommended framework by Apple for streamgrabbing on OSX >= 10.7 as well as on iOS.
The older QTKit framework has been marked deprecated since OSX version 10.7.
The input filename has to be given in the following syntax:
@example
-i "[[VIDEO]:[AUDIO]]"
@end example
The first entry selects the video input while the latter selects the audio input.
The stream has to be specified by the device name or the device index as shown by the device list.
Alternatively, the video and/or audio input device can be chosen by index using the
@option{
-video_device_index <INDEX>
}
and/or
@option{
-audio_device_index <INDEX>
}
, overriding any
device name or index given in the input filename.
All available devices can be enumerated by using @option{-list_devices true}, listing
all device names and corresponding indices.
There are two device name aliases:
@table @code
@item default
Select the AVFoundation default device of the corresponding type.
@item none
Do not record the corresponding media type.
This is equivalent to specifying an empty device name or index.
@end table
@subsection Options
AVFoundation supports the following options:
@table @option
@item -list_devices <TRUE|FALSE>
If set to true, a list of all available input devices is given showing all
device names and indices.
@item -video_device_index <INDEX>
Specify the video device by its index. Overrides anything given in the input filename.
@item -audio_device_index <INDEX>
Specify the audio device by its index. Overrides anything given in the input filename.
@item -pixel_format <FORMAT>
Request the video device to use a specific pixel format.
If the specified format is not supported, a list of available formats is given
and the first one in this list is used instead. Available pixel formats are:
@code{monob, rgb555be, rgb555le, rgb565be, rgb565le, rgb24, bgr24, 0rgb, bgr0, 0bgr, rgb0,
The filename passed as input is parsed to contain either a device name or index.
The device index can also be given by using -video_device_index.
A given device index will override any given device name.
If the desired device consists of numbers only, use -video_device_index to identify it.
The default device will be chosen if an empty string or the device name "default" is given.
The available devices can be enumerated by using -list_devices.
The pixel format can be set using -pixel_format.
Available formats:
monob, rgb555be, rgb555le, rgb565be, rgb565le, rgb24, bgr24, 0rgb, bgr0, 0bgr, rgb0,
bgr48be, uyvy422, yuva444p, yuva444p16le, yuv444p, yuv422p16, yuv422p10, yuv444p10,
yuv420p, nv12, yuyv422, gray}
yuv420p, nv12, yuyv422, gray
@item -framerate
Set the grabbing frame rate. Default is @code{ntsc}, corresponding to a
frame rate of @code{30000/1001}.
@item -video_size
Set the video frame size.
@item -capture_cursor
Capture the mouse pointer. Default is 0.
@item -capture_mouse_clicks
Capture the screen mouse clicks. Default is 0.
@end table
@subsection Examples
@itemize
@item
Print the list of AVFoundation supported devices and exit:
@example
$ ffmpeg -f avfoundation -list_devices true -i ""
ffmpeg -f avfoundation -i "0" out.mpg
@end example
@item
Record video from video device 0 and audio from audio device 0 into out.avi:
@example
$ ffmpeg -f avfoundation -i "0:0" out.avi
ffmpeg -f avfoundation -video_device_index 0 -i "" out.mpg
@end example
@item
Record video from video device 2 and audio from audio device 1 into out.avi:
@example
$ ffmpeg -f avfoundation -video_device_index 2 -i ":1" out.avi
ffmpeg -f avfoundation -pixel_format bgr0 -i "default" out.mpg
@end example
@item
Record video from the system default video device using the pixel format bgr0 and do not record any audio into out.avi:
@example
$ ffmpeg -f avfoundation -pixel_format bgr0 -i "default:none" out.avi
ffmpeg -f avfoundation -list_devices true -i ""
@end example
@end itemize
@section bktr
BSD video input device.
@subsection Options
@table @option
@item framerate
Set the frame rate.
@item video_size
Set the video frame size. Default is @code{vga}.
@item standard
Available values are:
@table @samp
@item pal
@item ntsc
@item secam
@item paln
@item palm
@item ntscj
@end table
@end table
@section decklink
The decklink input device provides capture capabilities for Blackmagic
DeckLink devices.
To enable this input device, you need the Blackmagic DeckLink SDK and you
need to configure with the appropriate @code{--extra-cflags}
and @code{--extra-ldflags}.
On Windows, you need to run the IDL files through @command{widl}.
DeckLink is very picky about the formats it supports. Pixel format is
uyvy422 or v210, framerate and video size must be determined for your device with
@command{-list_formats 1}. Audio sample rate is always 48 kHz and the number
of channels can be 2, 8 or 16. Note that all audio channels are bundled in one single
audio track.
@subsection Options
@table @option
@item list_devices
If set to @option{true}, print a list of devices and exit.
Defaults to @option{false}.
@item list_formats
If set to @option{true}, print a list of supported formats and exit.
Defaults to @option{false}.
@item bm_v210
If set to @samp{1}, video is captured in 10 bit v210 instead
of uyvy422. Not all Blackmagic devices support this option.
@item teletext_lines
If set to nonzero, an additional teletext stream will be captured from the
vertical ancillary data. This option is a bitmask of the VBI lines checked,
specifically lines 6 to 22, and lines 318 to 335. Line 6 is the LSB in the mask.
Selected lines which do not contain teletext information will be ignored. You
can use the special @option{all} constant to select all possible lines, or
@option{standard} to skip lines 6, 318 and 319, which are not compatible with all
receivers. Capturing teletext only works for SD PAL sources in 8 bit mode.
To use this option, ffmpeg needs to be compiled with @code{--enable-libzvbi}.
@item channels
Defines number of audio channels to capture. Must be @samp{2}, @samp{8} or @samp{16}.
Defaults to @samp{2}.
@item duplex_mode
Sets the decklink device duplex mode. Must be @samp{unset}, @samp{half} or @samp{full}.
Defaults to @samp{unset}.
@item video_input
Sets the video input source. Must be @samp{unset}, @samp{sdi}, @samp{hdmi},
@samp{optical_sdi}, @samp{component}, @samp{composite} or @samp{s_video}.
Defaults to @samp{unset}.
@item audio_input
Sets the audio input source. Must be @samp{unset}, @samp{embedded},
@samp{aes_ebu}, @samp{analog}, @samp{analog_xlr}, @samp{analog_rca} or
@samp{microphone}. Defaults to @samp{unset}.
@item video_pts
Sets the video packet timestamp source. Must be @samp{video}, @samp{audio},
@samp{reference} or @samp{wallclock}. Defaults to @samp{video}.
@item audio_pts
Sets the audio packet timestamp source. Must be @samp{video}, @samp{audio},
@samp{reference} or @samp{wallclock}. Defaults to @samp{audio}.
@item draw_bars
If set to @samp{true}, color bars are drawn in the event of a signal loss.
Defaults to @samp{true}.
@end table
@subsection Examples
@itemize
@item
List input devices:
@example
ffmpeg -f decklink -list_devices 1 -i dummy
@end example
@item
List supported formats:
@example
ffmpeg -f decklink -list_formats 1 -i 'Intensity Pro'
@end example
@item
Capture video clip at 1080i50 (format 11):
@example
ffmpeg -f decklink -i 'Intensity Pro@@11' -acodec copy -vcodec copy output.avi
@end example
@item
Capture video clip at 1080i50 10 bit:
@example
ffmpeg -bm_v210 1 -f decklink -i 'UltraStudio Mini Recorder@@11' -acodec copy -vcodec copy output.avi
@end example
@item
Capture video clip at 1080i50 with 16 audio channels:
@example
ffmpeg -channels 16 -f decklink -i 'UltraStudio Mini Recorder@@11' -acodec copy -vcodec copy output.avi
@end example
@end itemize
@section dshow
Windows DirectShow input device.
@@ -332,7 +107,7 @@ The input name should be in the format:
@end example
where @var{TYPE} can be either @var{audio} or @var{video},
and @var{NAME} is the device's name or alternative name..
and @var{NAME} is the device's name.
@subsection Options
@@ -365,11 +140,11 @@ If set to @option{true}, print a list of selected device's options
and exit.
@item video_device_number
Set video device number for devices with the same name (starts at 0,
Set video device number for devices with same name (starts at 0,
defaults to 0).
@item audio_device_number
Set audio device number for devices with the same name (starts at 0,
Set audio device number for devices with same name (starts at 0,
defaults to 0).
@item pixel_format
@@ -385,85 +160,6 @@ Setting this value too low can degrade performance.
See also
@url{http://msdn.microsoft.com/en-us/library/windows/desktop/dd377582(v=vs.85).aspx}
@item video_pin_name
Select video capture pin to use by name or alternative name.
@item audio_pin_name
Select audio capture pin to use by name or alternative name.
@item crossbar_video_input_pin_number
Select video input pin number for crossbar device. This will be
routed to the crossbar device's Video Decoder output pin.
Note that changing this value can affect future invocations
(sets a new default) until system reboot occurs.
@item crossbar_audio_input_pin_number
Select audio input pin number for crossbar device. This will be
routed to the crossbar device's Audio Decoder output pin.
Note that changing this value can affect future invocations
(sets a new default) until system reboot occurs.
@item show_video_device_dialog
If set to @option{true}, before capture starts, popup a display dialog
to the end user, allowing them to change video filter properties
and configurations manually.
Note that for crossbar devices, adjusting values in this dialog
may be needed at times to toggle between PAL (25 fps) and NTSC (29.97)
input frame rates, sizes, interlacing, etc. Changing these values can
enable different scan rates/frame rates and avoiding green bars at
the bottom, flickering scan lines, etc.
Note that with some devices, changing these properties can also affect future
invocations (sets new defaults) until system reboot occurs.
@item show_audio_device_dialog
If set to @option{true}, before capture starts, popup a display dialog
to the end user, allowing them to change audio filter properties
and configurations manually.
@item show_video_crossbar_connection_dialog
If set to @option{true}, before capture starts, popup a display
dialog to the end user, allowing them to manually
modify crossbar pin routings, when it opens a video device.
@item show_audio_crossbar_connection_dialog
If set to @option{true}, before capture starts, popup a display
dialog to the end user, allowing them to manually
modify crossbar pin routings, when it opens an audio device.
@item show_analog_tv_tuner_dialog
If set to @option{true}, before capture starts, popup a display
dialog to the end user, allowing them to manually
modify TV channels and frequencies.
@item show_analog_tv_tuner_audio_dialog
If set to @option{true}, before capture starts, popup a display
dialog to the end user, allowing them to manually
modify TV audio (like mono vs. stereo, Language A,B or C).
@item audio_device_load
Load an audio capture filter device from file instead of searching
it by name. It may load additional parameters too, if the filter
supports the serialization of its properties to.
To use this an audio capture source has to be specified, but it can
be anything even fake one.
@item audio_device_save
Save the currently used audio capture filter device and its
parameters (if the filter supports it) to a file.
If a file with the same name exists it will be overwritten.
@item video_device_load
Load a video capture filter device from file instead of searching
it by name. It may load additional parameters too, if the filter
supports the serialization of its properties to.
To use this a video capture source has to be specified, but it can
be anything even fake one.
@item video_device_save
Save the currently used video capture filter device and its
parameters (if the filter supports it) to a file.
If a file with the same name exists it will be overwritten.
@end table
@subsection Examples
@@ -500,46 +196,12 @@ Print the list of supported options in selected device and exit:
$ ffmpeg -list_options true -f dshow -i video="Camera"
@end example
@item
Specify pin names to capture by name or alternative name, specify alternative device name:
@example
$ ffmpeg -f dshow -audio_pin_name "Audio Out" -video_pin_name 2 -i video=video="@@device_pnp_\\?\pci#ven_1a0a&dev_6200&subsys_62021461&rev_01#4&e2c7dd6&0&00e1#@{65e8773d-8f56-11d0-a3b9-00a0c9223196@}\@{ca465100-deb0-4d59-818f-8c477184adf6@}":audio="Microphone"
@end example
@item
Configure a crossbar device, specifying crossbar pins, allow user to adjust video capture properties at startup:
@example
$ ffmpeg -f dshow -show_video_device_dialog true -crossbar_video_input_pin_number 0
-crossbar_audio_input_pin_number 3 -i video="AVerMedia BDA Analog Capture":audio="AVerMedia BDA Analog Capture"
@end example
@end itemize
@section dv1394
Linux DV 1394 input device.
@subsection Options
@table @option
@item framerate
Set the frame rate. Default is 25.
@item standard
Available values are:
@table @samp
@item pal
@item ntsc
@end table
Default value is @code{ntsc}.
@end table
@section fbdev
Linux framebuffer input device.
@@ -552,27 +214,18 @@ console. It is accessed through a file device node, usually
For more detailed information read the file
Documentation/fb/framebuffer.txt included in the Linux source tree.
See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1).
To record from the framebuffer device @file{/dev/fb0} with
@command{ffmpeg}:
@example
ffmpeg -f fbdev -framerate 10 -i /dev/fb0 out.avi
ffmpeg -f fbdev -r 10 -i /dev/fb0 out.avi
@end example
You can take a single screenshot image with the command:
@example
ffmpeg -f fbdev -framerate 1 -i /dev/fb0 -frames:v 1 screenshot.jpeg
ffmpeg -f fbdev -frames:v 1 -r 1 -i /dev/fb0 screenshot.jpeg
@end example
@subsection Options
@table @option
@item framerate
Set the frame rate. Default is 25.
@end table
See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1).
@section gdigrab
@@ -677,12 +330,12 @@ not work and result in undefined behavior.
The values @option{auto}, @option{dv} and @option{hdv} are supported.
@item dvbuffer
Set maximum size of buffer for incoming data, in frames. For DV, this
Set maxiumum size of buffer for incoming data, in frames. For DV, this
is an exact value. For HDV, it is not frame exact, since HDV does
not have a fixed frame size.
@item dvguid
Select the capture device by specifying its GUID. Capturing will only
Select the capture device by specifying it's GUID. Capturing will only
be performed from the specified device and fails if no device with the
given GUID is found. This is useful to select the input if multiple
devices are connected at the same time.
@@ -758,15 +411,6 @@ $ jack_connect metro:120_bpm ffmpeg:input_1
For more information read:
@url{http://jackaudio.org/}
@subsection Options
@table @option
@item channels
Set the number of channels. Default is 2.
@end table
@section lavfi
Libavfilter input virtual device.
@@ -791,14 +435,6 @@ generated by the device.
The first unlabelled output is automatically assigned to the "out0"
label, but all the others need to be specified explicitly.
The suffix "+subcc" can be appended to the output label to create an extra
stream with the closed captions packets attached to that output
(experimental; only for EIA-608 / CEA-708 for now).
The subcc streams are created after all the normal streams, in the order of
the corresponding stream.
For example, if there is "out19+subcc", "out7+subcc" and up to "out42", the
stream #43 is subcc for stream #7 and stream #44 is subcc for stream #19.
If not specified defaults to the filename specified for the input
device.
@@ -807,9 +443,6 @@ Set the filename of the filtergraph to be read and sent to the other
filters. Syntax of the filtergraph is the same as the one specified by
the option @var{graph}.
@item dumpgraph
Dump graph to stderr.
@end table
@subsection Examples
@@ -848,63 +481,12 @@ Read an audio stream and a video stream and play it back with
ffplay -f lavfi "movie=test.avi[out0];amovie=test.wav[out1]"
@end example
@item
Dump decoded frames to images and closed captions to a file (experimental):
@example
ffmpeg -f lavfi -i "movie=test.ts[out0+subcc]" -map v frame%08d.png -map s -c copy -f rawvideo subcc.bin
@end example
@end itemize
@section libcdio
Audio-CD input device based on libcdio.
To enable this input device during configuration you need libcdio
installed on your system. It requires the configure option
@code{--enable-libcdio}.
This device allows playing and grabbing from an Audio-CD.
For example to copy with @command{ffmpeg} the entire Audio-CD in @file{/dev/sr0},
you may run the command:
@example
ffmpeg -f libcdio -i /dev/sr0 cd.wav
@end example
@subsection Options
@table @option
@item speed
Set drive reading speed. Default value is 0.
The speed is specified CD-ROM speed units. The speed is set through
the libcdio @code{cdio_cddap_speed_set} function. On many CD-ROM
drives, specifying a value too large will result in using the fastest
speed.
@item paranoia_mode
Set paranoia recovery mode flags. It accepts one of the following values:
@table @samp
@item disable
@item verify
@item overlap
@item neverskip
@item full
@end table
Default value is @samp{disable}.
For more information about the available recovery modes, consult the
paranoia project documentation.
@end table
@section libdc1394
IIDC1394 input device, based on libdc1394 and libraw1394.
Requires the configure option @code{--enable-libdc1394}.
@section openal
The OpenAL input device provides audio capture on all systems with a
@@ -1011,19 +593,6 @@ ffmpeg -f oss -i /dev/dsp /tmp/oss.wav
For more information about OSS see:
@url{http://manuals.opensound.com/usersguide/dsp.html}
@subsection Options
@table @option
@item sample_rate
Set the sample rate in Hz. Default is 48000.
@item channels
Set the number of channels. Default is 2.
@end table
@section pulse
PulseAudio input device.
@@ -1064,10 +633,6 @@ Specify the number of bytes per frame, by default it is set to 1024.
@item fragment_size
Specify the minimal buffering fragment in PulseAudio, it will affect the
audio latency. By default it is unset.
@item wallclock
Set the initial PTS using the current time. Default is 1.
@end table
@subsection Examples
@@ -1103,22 +668,6 @@ ffmpeg -f qtkit -i "default" out.mpg
ffmpeg -f qtkit -list_devices true -i ""
@end example
@subsection Options
@table @option
@item frame_rate
Set frame rate. Default is 30.
@item list_devices
If set to @code{true}, print a list of devices and exit. Default is
@code{false}.
@item video_device_index
Select the video device by index for devices with the same name (starts at 0).
@end table
@section sndio
sndio input device.
@@ -1136,18 +685,6 @@ command:
ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav
@end example
@subsection Options
@table @option
@item sample_rate
Set the sample rate in Hz. Default is 48000.
@item channels
Set the number of channels. Default is 2.
@end table
@section video4linux2, v4l2
Video4Linux2 input video device.
@@ -1180,12 +717,6 @@ conversion into the real time clock.
Some usage examples of the video4linux2 device with @command{ffmpeg}
and @command{ffplay}:
@itemize
@item
List supported formats for a video4linux2 device:
@example
ffplay -f video4linux2 -list_formats all /dev/video0
@end example
@item
Grab and show the input of a video4linux2 device:
@example
@@ -1270,10 +801,6 @@ Force conversion from monotonic to absolute timestamps.
@end table
Default value is @code{default}.
@item use_libv4l2
Use libv4l2 (v4l-utils) conversion functions. Default is 0.
@end table
@section vfwcap
@@ -1284,30 +811,10 @@ The filename passed as input is the capture driver number, ranging from
0 to 9. You may use "list" as filename to print a list of drivers. Any
other filename will be interpreted as device number 0.
@subsection Options
@table @option
@item video_size
Set the video frame size.
@item framerate
Set the grabbing frame rate. Default value is @code{ntsc},
corresponding to a frame rate of @code{30000/1001}.
@end table
@section x11grab
X11 video input device.
To enable this input device during configuration you need libxcb
installed on your system. It will be automatically detected during
configuration.
Alternatively, the configure option @option{--enable-x11grab} exists
for legacy Xlib users.
This device allows one to capture a region of an X11 display.
The filename passed as input has the syntax:
@@ -1324,12 +831,10 @@ omitted, and defaults to "localhost". The environment variable
area with respect to the top-left border of the X11 screen. They
default to 0.
Check the X11 documentation (e.g. @command{man X}) for more detailed
information.
Check the X11 documentation (e.g. man X) for more detailed information.
Use the @command{xdpyinfo} program for getting basic information about
the properties of your X11 display (e.g. grep for "name" or
"dimensions").
Use the @command{dpyinfo} program for getting basic information about the
properties of your X11 display (e.g. grep for "name" or "dimensions").
For example to grab from @file{:0.0} using @command{ffmpeg}:
@example
@@ -1345,7 +850,7 @@ ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0+10,20 out.mpg
@table @option
@item draw_mouse
Specify whether to draw the mouse pointer. A value of @code{0} specifies
Specify whether to draw the mouse pointer. A value of @code{0} specify
not to draw the pointer. Default value is @code{1}.
@item follow_mouse
@@ -1378,10 +883,6 @@ If @var{show_region} is specified with @code{1}, then the grabbing
region will be indicated on screen. With this option, it is easy to
know what is being grabbed if only a portion of the screen is grabbed.
@item region_border
Set the region border thickness if @option{-show_region 1} is used.
Range is 1 to 128 and default is 3 (XCB-based x11grab only).
For example:
@example
ffmpeg -f x11grab -show_region 1 -framerate 25 -video_size cif -i :0.0+10,20 out.mpg
@@ -1394,18 +895,6 @@ ffmpeg -f x11grab -follow_mouse centered -show_region 1 -framerate 25 -video_siz
@item video_size
Set the video frame size. Default value is @code{vga}.
@item use_shm
Use the MIT-SHM extension for shared memory. Default value is @code{1}.
It may be necessary to disable it for remote displays (legacy x11grab
only).
@item grab_x
@item grab_y
Set the grabbing region coordinates. They are expressed as offset from
the top left corner of the X11 window and correspond to the
@var{x_offset} and @var{y_offset} parameters in the device name. The
default value for both options is 0.
@end table
@c man end INPUT DEVICES

View File

@@ -1,6 +1,8 @@
FFmpeg's bug/feature request tracker manual
=================================================
NOTE: This is a draft.
Overview:
---------
@@ -20,9 +22,9 @@ a mail for every change to every issue.
(the above does all work already after light testing)
The subscription URL for the ffmpeg-trac list is:
https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-trac
http(s)://ffmpeg.org/mailman/listinfo/ffmpeg-trac
The URL of the webinterface of the tracker is:
https://trac.ffmpeg.org
http(s)://trac.ffmpeg.org
Type:
-----
@@ -40,16 +42,12 @@ feature request / enhancement
where the current implementation cannot be considered wrong.
license violation
Ticket to keep track of (L)GPL violations of ffmpeg by others.
ticket to keep track of (L)GPL violations of ffmpeg by others
sponsoring request
Developer requests for hardware, software, specifications, money,
refunds, etc.
task
A task/reminder such as setting up a FATE client, adding filters to
Trac, etc.
Priority:
---------
critical
@@ -68,8 +66,7 @@ important
don't exist in a past revision or another branch.
normal
Default setting. Use this if the bug does not match the other
priorities or if you are unsure of what priority to choose.
minor
Bugs about things like spelling errors, "mp2" instead of
@@ -166,23 +163,14 @@ Component:
avcodec
issues in libavcodec/*
avdevice
issues in libavdevice/*
avfilter
issues in libavfilter/*
avformat
issues in libavformat/*
avutil
issues in libavutil/*
build system
issues in or related to configure/Makefile
documentation
issues in or related to doc/*
regression test
issues in tests/*
ffmpeg
issues in or related to ffmpeg.c
@@ -196,23 +184,11 @@ ffprobe
ffserver
issues in or related to ffserver.c
postproc
issues in libpostproc/*
build system
issues in or related to configure/Makefile
swresample
issues in libswresample/*
swscale
issues in libswscale/*
regression
bugs which were not present in a past revision
trac
issues related to our issue tracker
undetermined
default component; choose this if unsure
website
issues related to the website
wiki
issues related to the wiki

View File

@@ -1,98 +0,0 @@
CONTEXT
=======
The FFmpeg project merges all the changes from the Libav project
(https://libav.org) since the origin of the fork (around 2011).
With the exceptions of some commits due to technical/political disagreements or
issues, the changes are merged on a more or less regular schedule (daily for
years thanks to Michael, but more sparse nowadays).
WHY
===
The majority of the active developers believe the project needs to keep this
policy for various reasons.
The most important one is that we don't want our users to have to choose
between two distributors of libraries of the exact same name in order to have a
different set of features and bugfixes. By taking the responsibility of
unifying the two codebases, we allow users to benefit from the changes from the
two teams.
Today, FFmpeg has a much larger user database (we are distributed by every
major distribution), so we consider this mission a priority.
A different approach to the merge could have been to pick the changes we are
interested in and drop most of the cosmetics and other less important changes.
Unfortunately, this makes the following picks much harder, especially since the
Libav project is involved in various deep API changes. As a result, we decide
to virtually take everything done there.
Any Libav developer is of course welcome anytime to contribute directly to the
FFmpeg tree. Of course, we fully understand and are forced to accept that very
few Libav developers are interested in doing so, but we still want to recognize
their work. This leads us to create merge commits for every single one from
Libav. The original commit appears totally unchanged with full authorship in
our history (and the conflict are solved in the merge one). That way, not a
single thing from Libav will be lost in the future in case some reunification
happens, or that project disappears one way or another.
DOWNSIDES
=========
Of course, there are many downsides to this approach.
- It causes a non negligible merge commits pollution. We make sure there are
not several level of merges entangled (we do a 1:1 merge/commit), but it's
still a non-linear history.
- Many duplicated work. For instance, we added libavresample in our tree to
keep compatibility with Libav when our libswresample was already covering the
exact same purpose. The same thing happened for various elements such as the
ProRes support (but differences in features, bugs, licenses, ...). There are
many work to do to unify them, and any help is very much welcome.
- So much manpower from both FFmpeg and Libav is lost because of this mess. We
know it, and we don't know how to fix it. It takes incredible time to do
these merges, so we have even less time to work on things we personally care
about. The bad vibes also do not help with keeping our developers motivated.
- There is a growing technical risk factor with the merges due to the codebase
differing more and more.
MERGE GUIDELINES
================
The following gives developer guidelines on how to proceed when merging Libav commits.
Before starting, you can reduce the risk of errors on merge conflicts by using
a different merge conflict style:
$ git config --global merge.conflictstyle diff3
tools/libav-merge-next-commit is a script to help merging the next commit in
the queue. It assumes a remote named libav. It has two modes: merge, and noop.
The noop mode creates a merge with no change to the HEAD. You can pass a hash
as extra argument to reference a justification (it is common that we already
have the change done in FFmpeg).
Also see tools/murge, you can copy and paste a 3 way conflict into its stdin
and it will display colored diffs. Any arguments to murge (like ones to suppress
whitespace differences) are passed into colordiff.
TODO/FIXME/UNMERGED
===================
Stuff that didn't reach the codebase:
-------------------------------------
- HEVC DSP and x86 MC SIMD improvements from Libav (see https://ffmpeg.org/pipermail/ffmpeg-devel/2015-December/184777.html)
- QSV HWContext integration (04b17ff and 130e1f1), QSV scaling filter (62c58c5)
Collateral damage that needs work locally:
------------------------------------------
- Merge proresdec2.c and proresdec_lgpl.c
- Merge proresenc_anatoliy.c and proresenc_kostya.c
- Remove ADVANCED_PARSER in libavcodec/hevc_parser.c

View File

@@ -1,5 +1,4 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle Libavcodec Documentation
@titlepage

View File

@@ -1,5 +1,4 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle Libavdevice Documentation
@titlepage

Some files were not shown because too many files have changed in this diff Show More