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6 Commits
n4.1.7 ... n3.0

Author SHA1 Message Date
Michael Niedermayer
c40983a6f6 fate/source-check.sh: Use "git show" instead of git --version to test for git
This fixes fate with non git source trees

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 295de3efc5)

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-02-15 03:51:39 +01:00
Michael Niedermayer
bd0497b28b avcodec/cfhd: Temporary disable frame threading until related bugs have been fixed
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-02-15 03:12:13 +01:00
James Almer
4d95207938 x86: use the new helper macros where useful
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit 70d685a77f)
2016-02-14 20:04:19 -03:00
James Almer
1e8a75fae4 x86: add some more helper macros to check for slow cpuflags
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit 73a4589d4b)
2016-02-14 20:04:06 -03:00
Rostislav Pehlivanov
380980e0d2 Changelog: add entries for the SMPTE VC-2 decoder and encoder
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
2016-02-14 22:12:06 +00:00
Michael Niedermayer
1fd8eb4d4f Basic updates for 3.0
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-02-14 23:03:41 +01:00
5224 changed files with 146800 additions and 459001 deletions

1
.gitattributes vendored
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@@ -1,2 +1 @@
*.pnm -diff -text
tests/ref/fate/sub-scc eol=crlf

71
.gitignore vendored
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@@ -18,9 +18,8 @@
*.so.*
*.swp
*.ver
*.version
*.ptx
*.ptx.c
*-example
*-test
*_g
\#*
.\#*
@@ -29,10 +28,70 @@
/ffmpeg
/ffplay
/ffprobe
/config.asm
/config.h
/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_dir_cmd
/doc/examples/avio_reading
/doc/examples/decoding_encoding
/doc/examples/demuxing_decoding
/doc/examples/extract_mvs
/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/
/libavcodec/*_tablegen
/libavcodec/*_tables.c
/libavcodec/*_tables.h
/libavutil/avconfig.h
/libavutil/ffversion.h
/src
/mapfile
/tests/audiogen
/tests/base64
/tests/checkasm/checkasm
/tests/data/
/tests/pixfmts.mak
/tests/rotozoom
/tests/test_copy.ffmeta
/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/sidxindex
/tools/trasher
/tools/seek_print
/tools/uncoded_frame
/tools/zmqsend

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@@ -6,22 +6,18 @@ os:
addons:
apt:
packages:
- nasm
- yasm
- diffutils
compiler:
- clang
- gcc
matrix:
exclude:
- os: osx
compiler: 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 nasm; fi
- if [ "$TRAVIS_OS_NAME" == "osx" ]; then brew install yasm; fi
script:
- mkdir -p ffmpeg-samples
- ./configure --samples=ffmpeg-samples --cc=$CC

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@@ -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.

1411
Changelog

File diff suppressed because it is too large Load Diff

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@@ -1,4 +1,4 @@
# 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
@@ -13,18 +13,18 @@ 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 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`
- optional x86 optimizations in the files
- `libavcodec/x86/flac_dsp_gpl.asm`
- `libavcodec/x86/idct_mmx.c`
- `libavfilter/x86/vf_removegrain.asm`
- 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`
@@ -47,9 +47,9 @@ Specifically, the GPL parts of FFmpeg are:
- `vf_pp.c`
- `vf_pp7.c`
- `vf_pullup.c`
- `vf_repeatfields.c`
- `vf_sab.c`
- `vf_smartblur.c`
- `vf_repeatfields.c`
- `vf_spp.c`
- `vf_stereo3d.c`
- `vf_super2xsai.c`
@@ -73,17 +73,20 @@ There are a handful of files under other licensing terms, namely:
* `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
- libutvideo
- libvidstab
- libx264
- libx265
@@ -98,23 +101,14 @@ license is incompatible with the LGPL v2.1 and the GPL v2, but not with
version 3 of those licenses. So to combine these libraries with FFmpeg, the
license version needs to be upgraded by passing `--enable-version3` to configure.
### Incompatible libraries
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 and FAAC 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.

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@@ -29,6 +29,9 @@ ffplay:
ffprobe:
ffprobe.c Stefano Sabatini
ffserver:
ffserver.c Reynaldo H. Verdejo Pinochet
Commandline utility code:
cmdutils.c, cmdutils.h Michael Niedermayer
@@ -39,12 +42,12 @@ QuickTime faststart:
Miscellaneous Areas
===================
documentation Stefano Sabatini, Mike Melanson, Timothy Gu, Lou Logan, Gyan Doshi
project server Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser, Nikolay Aleksandrov
documentation Stefano Sabatini, Mike Melanson, Timothy Gu, Lou Logan
build system (configure, makefiles) Diego Biurrun, Mans Rullgard
project server Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser, Lou Logan
presets Robert Swain
metadata subsystem Aurelien Jacobs
release management Michael Niedermayer
API tests Ludmila Glinskih
Communication
@@ -53,11 +56,10 @@ Communication
website Deby Barbara Lepage
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
Launchpad Timothy Gu
ffmpeg-security Andreas Cadhalpun, Carl Eugen Hoyos, Clément Bœsch, Michael Niedermayer, Reimar Doeffinger, Rodger Combs, wm4
libavutil
@@ -77,7 +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
@@ -88,6 +89,7 @@ Other:
rational.c, rational.h Michael Niedermayer
rc4 Reimar Doeffinger
ripemd.c, ripemd.h James Almer
timecode Clément Bœsch
libavcodec
@@ -113,14 +115,15 @@ Generic Parts:
lzw.* Michael Niedermayer
floating point AAN DCT:
faandct.c, faandct.h Michael Niedermayer
Non-power-of-two MDCT:
mdct15.c, mdct15.h Rostislav Pehlivanov
Golomb coding:
golomb.c, golomb.h Michael Niedermayer
LPC:
lpc.c, lpc.h Justin Ruggles
motion estimation:
motion* Michael Niedermayer
rate control:
ratecontrol.c Michael Niedermayer
libxvid_rc.c Michael Niedermayer
simple IDCT:
simple_idct.c, simple_idct.h Michael Niedermayer
postprocessing:
@@ -136,41 +139,47 @@ Codecs:
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
aptx.c Aurelien Jacobs
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
ccaption_dec.c Anshul Maheshwari
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
cuviddec.c Timo Rothenpieler
dca* foo86
dca.c Kostya Shishkov, Benjamin Larsson
dirac* Rostislav Pehlivanov
dnxhd* Baptiste Coudurier
dolby_e* foo86
dpcm.c Mike Melanson
dss_sp.c Oleksij Rempel
dss_sp.c Oleksij Rempel, Michael Niedermayer
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
exr.c Martin Vignali
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
@@ -179,47 +188,61 @@ Codecs:
h263* Michael Niedermayer
h264* Loren Merritt, Michael Niedermayer
hap* Tom Butterworth
huffyuv* Michael Niedermayer
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
libcodec2.c Tomas Härdin
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* Carl Eugen Hoyos
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
nvdec*, nvenc* Timo Rothenpieler
opus* Rostislav Pehlivanov
nvenc.c 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* Mark Thompson, Zhong Li
qdm2.c, qdm2data.h Roberto Togni, Benjamin Larsson
qdrw.c Kostya Shishkov
qpeg.c Kostya Shishkov
qsv* Ivan Uskov
qtrle.c Mike Melanson
ra144.c, ra144.h, ra288.c, ra288.h Roberto Togni
resample2.c Michael Niedermayer
@@ -227,51 +250,66 @@ Codecs:
rpza.c Roberto Togni
rtjpeg.c, rtjpeg.h Reimar Doeffinger
rv10.c Michael Niedermayer
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
sonic.c Alex Beregszaszi
speedhq.c Steinar H. Gunderson
srt* Aurelien Jacobs
sunrast.c Ivo van Poorten
svq3.c Michael Niedermayer
tak* Paul B Mahol
targa.c Kostya Shishkov
tiff.c Kostya Shishkov
truemotion1* Mike Melanson
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
v4l2_* Jorge Ramirez-Ortiz
ulti* Kostya Shishkov
v410*.c Derek Buitenhuis
vb.c Kostya Shishkov
vble.c Derek Buitenhuis
vc1* Kostya Shishkov, Christophe Gisquet
vc2* Rostislav Pehlivanov
vcr1.c Michael Niedermayer
videotoolboxenc.c Rick Kern, Aman Gupta
vda_h264_dec.c Xidorn Quan
vima.c Paul B Mahol
vmnc.c Kostya Shishkov
vorbisdec.c Denes Balatoni, David Conrad
vorbisenc.c Oded Shimon
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, Steve Lhomme
d3d11va* Steve Lhomme
mediacodec* Matthieu Bouron, Aman Gupta
dxva2* Hendrik Leppkes, Laurent Aimar
vaapi* Gwenole Beauchesne
vaapi_encode* Mark Thompson
vda* Sebastien Zwickert
vdpau* Philip Langdale, Carl Eugen Hoyos
videotoolbox* Rick Kern, Aman Gupta
videotoolbox* Sebastien Zwickert
libavdevice
@@ -281,8 +319,7 @@ libavdevice
avfoundation.m Thilo Borgmann
android_camera.c Felix Matouschek
decklink* Marton Balint
decklink* Deti Fliegl
dshow.c Roger Pack (CC rogerdpack@gmail.com)
fbdev_enc.c Lukasz Marek
gdigrab.c Roger Pack (CC rogerdpack@gmail.com)
@@ -291,8 +328,8 @@ libavdevice
libdc1394.c Roman Shaposhnik
opengl_enc.c Lukasz Marek
pulse_audio_enc.c Lukasz Marek
qtkit.m Thilo Borgmann
sdl Stefano Sabatini
sdl2.c Josh de Kock
v4l2.c Giorgio Vazzana
vfwcap.c Ramiro Polla
xv.c Lukasz Marek
@@ -303,8 +340,6 @@ 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
@@ -318,10 +353,7 @@ Filters:
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
@@ -329,14 +361,11 @@ Filters:
avf_avectorscope.c Paul B Mahol
avf_showcqt.c Muhammad Faiz
vf_blend.c Paul B Mahol
vf_bwdif Thomas Mundt (CC <thomas.mundt@hr.de>)
vf_chromakey.c Timo Rothenpieler
vf_colorchannelmixer.c Paul B Mahol
vf_colorconstancy.c Mina Sami (CC <minas.gorgy@gmail.com>)
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>)
@@ -346,15 +375,11 @@ Filters:
vf_hqx.c Clément Bœsch
vf_idet.c Pascal Massimino
vf_il.c Paul B Mahol
vf_(t)interlace Thomas Mundt (CC <thomas.mundt@hr.de>)
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
@@ -374,7 +399,6 @@ Generic parts:
libavformat/avformat.h Michael Niedermayer
Utility Code:
libavformat/utils.c Michael Niedermayer
Text Subtitles Clément Bœsch
Muxers/Demuxers:
@@ -384,36 +408,34 @@ Muxers/Demuxers:
afc.c Paul B Mahol
aiffdec.c Baptiste Coudurier, Matthieu Bouron
aiffenc.c Baptiste Coudurier, Matthieu Bouron
ape.c Kostya Shishkov
apngdec.c Benoit Fouet
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
boadec.c Michael Niedermayer
brstm.c Paul B Mahol
caf* Peter Ross
cdxl.c Paul B Mahol
codec2.c Tomas Härdin
crc.c Michael Niedermayer
dashdec.c Steven Liu
dashenc.c Karthick Jeyapal
daud.c Reimar Doeffinger
dss.c Oleksij Rempel
dtsdec.c foo86
dss.c Oleksij Rempel, Michael Niedermayer
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 Michael Niedermayer
flvenc.c Michael Niedermayer, Steven Liu
flvdec.c, flvenc.c Michael Niedermayer
gxf.c Reimar Doeffinger
gxfenc.c Baptiste Coudurier
hlsenc.c Christian Suloway, Steven Liu
hls.c Anssi Hannula
hls encryption (hlsenc.c) Christian Suloway
idcin.c Mike Melanson
idroqdec.c Mike Melanson
iff.c Jaikrishnan Menon
@@ -421,9 +443,10 @@ 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
libopenmpt.c Josh de Kock
libnut.c Oded Shimon
lmlm4.c Ivo van Poorten
lvfdec.c Paul B Mahol
lxfdec.c Tomas Härdin
@@ -435,9 +458,10 @@ 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
@@ -445,6 +469,7 @@ Muxers/Demuxers:
msnwc_tcp.c Ramiro Polla
mtv.c Reynaldo H. Verdejo Pinochet
mxf* Baptiste Coudurier
mxfdec.c Tomas Härdin
nistspheredec.c Paul B Mahol
nsvdec.c Francois Revol
nut* Michael Niedermayer
@@ -462,7 +487,8 @@ 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
@@ -470,14 +496,15 @@ Muxers/Demuxers:
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
s337m.c foo86
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
@@ -492,6 +519,7 @@ Muxers/Demuxers:
webvtt* Matthew J Heaney
westwood.c Mike Melanson
wtv.c Peter Ross
wv.c Kostya Shishkov
wvenc.c Paul B Mahol
Protocols:
@@ -522,45 +550,20 @@ Resamplers:
Operating systems / CPU architectures
=====================================
Alpha Falk Hueffner
MIPS Manojkumar Bhosale, Shiyou Yin
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 ICL Matthew Oliver
ADI/Blackfin DSP Marc Hoffman
Sparc Roman Shaposhnik
OS/2 KO Myung-Hun
Developers with git write access who are currently not maintaining any specific part
====================================================================================
Alex Converse
Andreas Cadhalpun
Anuradha Suraparaju
Ben Littler
Benjamin Larsson
Bobby Bingham
Daniel Verkamp
Derek Buitenhuis
Ganesh Ajjanagadde
Henrik Gramner
Ivan Uskov
James Darnley
Jan Ekström
Joakim Plate
Jun Zhao
Kieran Kunhya
Kirill Gavrilov
Martin Storsjö
Panagiotis Issaris
Pedro Arthur
Sebastien Zwickert
Vittorio Giovara
wm4
(this list is incomplete)
x86 Michael Niedermayer
Releases
@@ -570,18 +573,17 @@ Releases
2.7 Michael Niedermayer
2.6 Michael Niedermayer
2.5 Michael Niedermayer
2.4 Michael Niedermayer
If you want to maintain an older release, please contact us
GnuPG Fingerprints and IRC nicknames of maintainers and contributors
====================================================================
IRC nicknames are in parentheses. These apply
to the IRC channels listed on the website.
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
@@ -589,17 +591,18 @@ 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
James Almer 7751 2E8C FD94 A169 57E6 9A7A 1463 01AD 7376 59E0
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 (llogan) 7D68 DC73 CBEF EABB 671A B6CF 621C 2E28 82F8 DC3A
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
@@ -609,9 +612,8 @@ Reynaldo H. Verdejo Pinochet 6E27 CD34 170C C78E 4D4F 5F40 C18E 077F 3114 452A
Robert Swain EE7A 56EA 4A81 A7B5 2001 A521 67FA 362D A2FC 3E71
Sascha Sommer 38A0 F88B 868E 9D3A 97D4 D6A0 E823 706F 1E07 0D3C
Stefano Sabatini 0D0B AD6B 5330 BBAD D3D6 6A0C 719C 2839 FC43 2D5F
Steinar H. Gunderson C2E9 004F F028 C18E 4EAD DB83 7F61 7561 7797 8F76
Stephan Hilb 4F38 0B3A 5F39 B99B F505 E562 8D5C 5554 4E17 8863
Tiancheng "Timothy" Gu 9456 AFC0 814A 8139 E994 8351 7FE6 B095 B582 B0D4
Tim Nicholson 38CF DB09 3ED0 F607 8B67 6CED 0C0B FC44 8B0B FC83
Tomas Härdin (thardin) A79D 4E3D F38F 763F 91F5 8B33 A01E 8AE0 41BB 2551
Tomas Härdin A79D 4E3D F38F 763F 91F5 8B33 A01E 8AE0 41BB 2551
Wei Gao 4269 7741 857A 0E60 9EC5 08D2 4744 4EFA 62C1 87B9

141
Makefile
View File

@@ -1,5 +1,5 @@
MAIN_MAKEFILE=1
include ffbuild/config.mak
include config.mak
vpath %.c $(SRC_PATH)
vpath %.cpp $(SRC_PATH)
@@ -11,12 +11,38 @@ vpath %.asm $(SRC_PATH)
vpath %.rc $(SRC_PATH)
vpath %.v $(SRC_PATH)
vpath %.texi $(SRC_PATH)
vpath %.cu $(SRC_PATH)
vpath %.ptx $(SRC_PATH)
vpath %/fate_config.sh.template $(SRC_PATH)
TESTTOOLS = audiogen videogen rotozoom tiny_psnr tiny_ssim base64 audiomatch
AVPROGS-$(CONFIG_FFMPEG) += ffmpeg
AVPROGS-$(CONFIG_FFPLAY) += ffplay
AVPROGS-$(CONFIG_FFPROBE) += ffprobe
AVPROGS-$(CONFIG_FFSERVER) += ffserver
AVPROGS := $(AVPROGS-yes:%=%$(PROGSSUF)$(EXESUF))
INSTPROGS = $(AVPROGS-yes:%=%$(PROGSSUF)$(EXESUF))
PROGS += $(AVPROGS)
AVBASENAMES = ffmpeg ffplay ffprobe ffserver
ALLAVPROGS = $(AVBASENAMES:%=%$(PROGSSUF)$(EXESUF))
ALLAVPROGS_G = $(AVBASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
$(foreach prog,$(AVBASENAMES),$(eval OBJS-$(prog) += cmdutils.o))
$(foreach prog,$(AVBASENAMES),$(eval OBJS-$(prog)-$(CONFIG_OPENCL) += cmdutils_opencl.o))
OBJS-ffmpeg += ffmpeg_opt.o ffmpeg_filter.o
OBJS-ffmpeg-$(HAVE_VDPAU_X11) += ffmpeg_vdpau.o
OBJS-ffmpeg-$(HAVE_DXVA2_LIB) += ffmpeg_dxva2.o
ifndef CONFIG_VIDEOTOOLBOX
OBJS-ffmpeg-$(CONFIG_VDA) += ffmpeg_videotoolbox.o
endif
OBJS-ffmpeg-$(CONFIG_VIDEOTOOLBOX) += ffmpeg_videotoolbox.o
OBJS-ffmpeg-$(CONFIG_LIBMFX) += ffmpeg_qsv.o
OBJS-ffserver += ffserver_config.o
TESTTOOLS = audiogen videogen rotozoom tiny_psnr tiny_ssim base64
HOSTPROGS := $(TESTTOOLS:%=tests/%) doc/print_options
TOOLS = qt-faststart trasher uncoded_frame
TOOLS-$(CONFIG_ZLIB) += cws2fws
# $(FFLIBS-yes) needs to be in linking order
FFLIBS-$(CONFIG_AVDEVICE) += avdevice
@@ -31,46 +57,35 @@ FFLIBS-$(CONFIG_SWSCALE) += swscale
FFLIBS := avutil
DATA_FILES := $(wildcard $(SRC_PATH)/presets/*.ffpreset) $(SRC_PATH)/doc/ffprobe.xsd
EXAMPLES_FILES := $(wildcard $(SRC_PATH)/doc/examples/*.c) $(SRC_PATH)/doc/examples/Makefile $(SRC_PATH)/doc/examples/README
SKIPHEADERS = compat/w32pthreads.h
SKIPHEADERS = cmdutils_common_opts.h compat/w32pthreads.h
# first so "all" becomes default target
all: all-yes
include $(SRC_PATH)/tools/Makefile
include $(SRC_PATH)/ffbuild/common.mak
include $(SRC_PATH)/common.mak
FF_EXTRALIBS := $(FFEXTRALIBS)
FF_DEP_LIBS := $(DEP_LIBS)
FF_STATIC_DEP_LIBS := $(STATIC_DEP_LIBS)
$(TOOLS): %$(EXESUF): %.o
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(EXTRALIBS-$(*F)) $(EXTRALIBS) $(ELIBS)
all: $(AVPROGS)
target_dec_%_fuzzer$(EXESUF): target_dec_%_fuzzer.o $(FF_DEP_LIBS)
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
$(TOOLS): %$(EXESUF): %.o $(EXEOBJS)
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS)
tools/sofa2wavs$(EXESUF): ELIBS = $(FF_EXTRALIBS)
tools/cws2fws$(EXESUF): ELIBS = $(ZLIB)
tools/uncoded_frame$(EXESUF): $(FF_DEP_LIBS)
tools/uncoded_frame$(EXESUF): ELIBS = $(FF_EXTRALIBS)
tools/target_dec_%_fuzzer$(EXESUF): $(FF_DEP_LIBS)
CONFIGURABLE_COMPONENTS = \
$(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c)) \
$(SRC_PATH)/libavcodec/bitstream_filters.c \
$(SRC_PATH)/libavcodec/parsers.c \
$(SRC_PATH)/libavformat/protocols.c \
config.h: ffbuild/.config
ffbuild/.config: $(CONFIGURABLE_COMPONENTS)
config.h: .config
.config: $(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c))
@-tput bold 2>/dev/null
@-printf '\nWARNING: $(?) newer than config.h, rerun configure\n\n'
@-printf '\nWARNING: $(?F) newer than config.h, rerun configure\n\n'
@-tput sgr0 2>/dev/null
SUBDIR_VARS := CLEANFILES FFLIBS HOSTPROGS TESTPROGS TOOLS \
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 VSX-OBJS MMX-OBJS X86ASM-OBJS \
ALTIVEC-OBJS MMX-OBJS YASM-OBJS \
MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSP-OBJS MSA-OBJS \
MMI-OBJS OBJS SLIBOBJS HOSTOBJS TESTOBJS
@@ -85,32 +100,41 @@ SUBDIR := $(1)/
include $(SRC_PATH)/$(1)/Makefile
-include $(SRC_PATH)/$(1)/$(ARCH)/Makefile
-include $(SRC_PATH)/$(1)/$(INTRINSICS)/Makefile
include $(SRC_PATH)/ffbuild/library.mak
include $(SRC_PATH)/library.mak
endef
$(foreach D,$(FFLIBS),$(eval $(call DOSUBDIR,lib$(D))))
include $(SRC_PATH)/fftools/Makefile
include $(SRC_PATH)/doc/Makefile
include $(SRC_PATH)/doc/examples/Makefile
libavcodec/utils.o libavformat/utils.o libavdevice/avdevice.o libavfilter/avfilter.o libavutil/utils.o libpostproc/postprocess.o libswresample/swresample.o libswscale/utils.o : libavutil/ffversion.h
define DOPROG
OBJS-$(1) += $(1).o $(EXEOBJS) $(OBJS-$(1)-yes)
$(1)$(PROGSSUF)_g$(EXESUF): $$(OBJS-$(1))
$$(OBJS-$(1)): CFLAGS += $(CFLAGS-$(1))
$(1)$(PROGSSUF)_g$(EXESUF): LDFLAGS += $(LDFLAGS-$(1))
$(1)$(PROGSSUF)_g$(EXESUF): FF_EXTRALIBS += $(LIBS-$(1))
-include $$(OBJS-$(1):.o=.d)
endef
$(foreach P,$(PROGS),$(eval $(call DOPROG,$(P:$(PROGSSUF)$(EXESUF)=))))
ffprobe.o cmdutils.o libavcodec/utils.o libavformat/utils.o libavdevice/avdevice.o libavfilter/avfilter.o libavutil/utils.o libpostproc/postprocess.o libswresample/swresample.o libswscale/utils.o : libavutil/ffversion.h
$(PROGS): %$(PROGSSUF)$(EXESUF): %$(PROGSSUF)_g$(EXESUF)
ifeq ($(STRIPTYPE),direct)
$(STRIP) -o $@ $<
else
$(CP) $< $@
$(STRIP) $@
endif
%$(PROGSSUF)_g$(EXESUF): $(FF_DEP_LIBS)
%$(PROGSSUF)_g$(EXESUF): %.o $(FF_DEP_LIBS)
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
VERSION_SH = $(SRC_PATH)/ffbuild/version.sh
OBJDIRS += tools
-include $(wildcard tools/*.d)
VERSION_SH = $(SRC_PATH)/version.sh
GIT_LOG = $(SRC_PATH)/.git/logs/HEAD
.version: $(wildcard $(GIT_LOG)) $(VERSION_SH) ffbuild/config.mak
.version: $(wildcard $(GIT_LOG)) $(VERSION_SH) config.mak
.version: M=@
libavutil/ffversion.h .version:
@@ -120,32 +144,44 @@ libavutil/ffversion.h .version:
# force version.sh to run whenever version might have changed
-include .version
ifdef AVPROGS
install: install-progs install-data
endif
install: install-libs install-headers
install-libs: install-libs-yes
install-data: $(DATA_FILES)
$(Q)mkdir -p "$(DATADIR)"
$(INSTALL) -m 644 $(DATA_FILES) "$(DATADIR)"
install-progs-yes:
install-progs-$(CONFIG_SHARED): install-libs
uninstall: uninstall-data uninstall-headers uninstall-libs uninstall-pkgconfig
install-progs: install-progs-yes $(AVPROGS)
$(Q)mkdir -p "$(BINDIR)"
$(INSTALL) -c -m 755 $(INSTPROGS) "$(BINDIR)"
install-data: $(DATA_FILES) $(EXAMPLES_FILES)
$(Q)mkdir -p "$(DATADIR)/examples"
$(INSTALL) -m 644 $(DATA_FILES) "$(DATADIR)"
$(INSTALL) -m 644 $(EXAMPLES_FILES) "$(DATADIR)/examples"
uninstall: uninstall-libs uninstall-headers uninstall-progs uninstall-data
uninstall-progs:
$(RM) $(addprefix "$(BINDIR)/", $(ALLAVPROGS))
uninstall-data:
$(RM) -r "$(DATADIR)"
clean::
$(RM) $(ALLAVPROGS) $(ALLAVPROGS_G)
$(RM) $(CLEANSUFFIXES)
$(RM) $(addprefix compat/,$(CLEANSUFFIXES)) $(addprefix compat/*/,$(CLEANSUFFIXES))
$(RM) $(CLEANSUFFIXES:%=tools/%)
$(RM) -r coverage-html
$(RM) -rf coverage.info coverage.info.in lcov
distclean:: clean
$(RM) .version avversion.h config.asm config.h mapfile \
ffbuild/.config ffbuild/config.* libavutil/avconfig.h \
version.h libavutil/ffversion.h libavcodec/codec_names.h \
libavcodec/bsf_list.c libavformat/protocol_list.c \
libavcodec/codec_list.c libavcodec/parser_list.c \
libavformat/muxer_list.c libavformat/demuxer_list.c
distclean::
$(RM) $(DISTCLEANSUFFIXES)
$(RM) config.* .config libavutil/avconfig.h .version avversion.h version.h libavutil/ffversion.h libavcodec/codec_names.h
ifeq ($(SRC_LINK),src)
$(RM) src
endif
@@ -154,7 +190,6 @@ endif
config:
$(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION)
build: all alltools examples testprogs
check: all alltools examples testprogs fate
include $(SRC_PATH)/tests/Makefile
@@ -170,5 +205,5 @@ $(sort $(OBJDIRS)):
# so this saves some time on slow systems.
.SUFFIXES:
.PHONY: all all-yes alltools build check config testprogs
.PHONY: *clean install* uninstall*
.PHONY: all all-yes alltools check *clean config install*
.PHONY: testprogs uninstall*

View File

@@ -21,6 +21,8 @@ such as audio, video, subtitles and related metadata.
* [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`.
## Documentation
@@ -43,4 +45,5 @@ GPL. Please refer to the LICENSE file for detailed information.
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.
avoided because they are not part of our review process. Few developers
follow pull requests so they will likely be ignored.

View File

@@ -1 +1 @@
4.1.7
3.0

View File

@@ -1,15 +1,15 @@
┌─────────────────────────────────────────────
│ RELEASE NOTES for FFmpeg 4.1 "al-Khwarizmi" │
└─────────────────────────────────────────────
┌─────────────────────────────────────────┐
│ RELEASE NOTES for FFmpeg 3.0 "Einstein" │
└─────────────────────────────────────────┘
The FFmpeg Project proudly presents FFmpeg 4.1 "al-Khwarizmi", about 6
months after the release of FFmpeg 4.0.
The FFmpeg Project proudly presents FFmpeg 3.0 "Einstein", about 5
months after the release of FFmpeg 2.8.
A complete Changelog is available at the root of the project, and the
complete Git history on https://git.ffmpeg.org/gitweb/ffmpeg.git
complete Git history on http://source.ffmpeg.org.
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.libera.chat) or ask
feel free to join us on the #ffmpeg IRC channel (on irc.freenode.net) or ask
on the mailing-lists.

View File

@@ -14,4 +14,4 @@ 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_X86ASM) += $(X86ASM-OBJS) $(X86ASM-OBJS-yes)
OBJS-$(HAVE_YASM) += $(YASM-OBJS) $(YASM-OBJS-yes)

View File

@@ -38,7 +38,6 @@
#include "libswscale/swscale.h"
#include "libswresample/swresample.h"
#include "libpostproc/postprocess.h"
#include "libavutil/attributes.h"
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/bprint.h"
@@ -62,9 +61,6 @@
#include <sys/time.h>
#include <sys/resource.h>
#endif
#ifdef _WIN32
#include <windows.h>
#endif
static int init_report(const char *env);
@@ -76,12 +72,6 @@ static FILE *report_file;
static int report_file_level = AV_LOG_DEBUG;
int hide_banner = 0;
enum show_muxdemuxers {
SHOW_DEFAULT,
SHOW_DEMUXERS,
SHOW_MUXERS,
};
void init_opts(void)
{
av_dict_set(&sws_dict, "flags", "bicubic", 0);
@@ -117,15 +107,6 @@ 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))
@@ -232,6 +213,7 @@ static const OptionDef *find_option(const OptionDef *po, const char *name)
* by default. HAVE_COMMANDLINETOARGVW is true on cygwin, while
* it doesn't provide the actual command line via GetCommandLineW(). */
#if HAVE_COMMANDLINETOARGVW && defined(_WIN32)
#include <windows.h>
#include <shellapi.h>
/* Will be leaked on exit */
static char** win32_argv_utf8 = NULL;
@@ -881,54 +863,28 @@ int opt_loglevel(void *optctx, const char *opt, const char *arg)
{ "debug" , AV_LOG_DEBUG },
{ "trace" , AV_LOG_TRACE },
};
const char *token;
char *tail;
int flags = av_log_get_flags();
int level = av_log_get_level();
int cmd, i = 0;
int level;
int flags;
int i;
av_assert0(arg);
while (*arg) {
token = arg;
if (*token == '+' || *token == '-') {
cmd = *token++;
} else {
cmd = 0;
}
if (!i && !cmd) {
flags = 0; /* missing relative prefix, build absolute value */
}
if (!strncmp(token, "repeat", 6)) {
if (cmd == '-') {
flags |= AV_LOG_SKIP_REPEATED;
} else {
flags &= ~AV_LOG_SKIP_REPEATED;
}
arg = token + 6;
} else if (!strncmp(token, "level", 5)) {
if (cmd == '-') {
flags &= ~AV_LOG_PRINT_LEVEL;
} else {
flags |= AV_LOG_PRINT_LEVEL;
}
arg = token + 5;
} else {
break;
}
i++;
}
if (!*arg) {
goto end;
} else if (*arg == '+') {
arg++;
} else if (!i) {
flags = av_log_get_flags(); /* level value without prefix, reset flags */
}
flags = av_log_get_flags();
tail = strstr(arg, "repeat");
if (tail)
flags &= ~AV_LOG_SKIP_REPEATED;
else
flags |= AV_LOG_SKIP_REPEATED;
av_log_set_flags(flags);
if (tail == arg)
arg += 6 + (arg[6]=='+');
if(tail && !*arg)
return 0;
for (i = 0; i < FF_ARRAY_ELEMS(log_levels); i++) {
if (!strcmp(log_levels[i].name, arg)) {
level = log_levels[i].level;
goto end;
av_log_set_level(log_levels[i].level);
return 0;
}
}
@@ -940,9 +896,6 @@ int opt_loglevel(void *optctx, const char *opt, const char *arg)
av_log(NULL, AV_LOG_FATAL, "\"%s\"\n", log_levels[i].name);
exit_program(1);
}
end:
av_log_set_flags(flags);
av_log_set_level(level);
return 0;
}
@@ -1018,7 +971,7 @@ static int init_report(const char *env)
av_free(key);
}
av_bprint_init(&filename, 0, AV_BPRINT_SIZE_AUTOMATIC);
av_bprint_init(&filename, 0, 1);
expand_filename_template(&filename,
av_x_if_null(filename_template, "%p-%t.log"), tm);
av_free(filename_template);
@@ -1286,12 +1239,10 @@ static int is_device(const AVClass *avclass)
return AV_IS_INPUT_DEVICE(avclass->category) || AV_IS_OUTPUT_DEVICE(avclass->category);
}
static int show_formats_devices(void *optctx, const char *opt, const char *arg, int device_only, int muxdemuxers)
static int show_formats_devices(void *optctx, const char *opt, const char *arg, int device_only)
{
void *ifmt_opaque = NULL;
const AVInputFormat *ifmt = NULL;
void *ofmt_opaque = NULL;
const AVOutputFormat *ofmt = NULL;
AVInputFormat *ifmt = NULL;
AVOutputFormat *ofmt = NULL;
const char *last_name;
int is_dev;
@@ -1306,35 +1257,29 @@ static int show_formats_devices(void *optctx, const char *opt, const char *arg,
const char *name = NULL;
const char *long_name = NULL;
if (muxdemuxers !=SHOW_DEMUXERS) {
ofmt_opaque = NULL;
while ((ofmt = av_muxer_iterate(&ofmt_opaque))) {
is_dev = is_device(ofmt->priv_class);
if (!is_dev && device_only)
continue;
if ((!name || strcmp(ofmt->name, name) < 0) &&
strcmp(ofmt->name, last_name) > 0) {
name = ofmt->name;
long_name = ofmt->long_name;
encode = 1;
}
while ((ofmt = av_oformat_next(ofmt))) {
is_dev = is_device(ofmt->priv_class);
if (!is_dev && device_only)
continue;
if ((!name || strcmp(ofmt->name, name) < 0) &&
strcmp(ofmt->name, last_name) > 0) {
name = ofmt->name;
long_name = ofmt->long_name;
encode = 1;
}
}
if (muxdemuxers != SHOW_MUXERS) {
ifmt_opaque = NULL;
while ((ifmt = av_demuxer_iterate(&ifmt_opaque))) {
is_dev = is_device(ifmt->priv_class);
if (!is_dev && device_only)
continue;
if ((!name || strcmp(ifmt->name, name) < 0) &&
strcmp(ifmt->name, last_name) > 0) {
name = ifmt->name;
long_name = ifmt->long_name;
encode = 0;
}
if (name && strcmp(ifmt->name, name) == 0)
decode = 1;
while ((ifmt = av_iformat_next(ifmt))) {
is_dev = is_device(ifmt->priv_class);
if (!is_dev && device_only)
continue;
if ((!name || strcmp(ifmt->name, name) < 0) &&
strcmp(ifmt->name, last_name) > 0) {
name = ifmt->name;
long_name = ifmt->long_name;
encode = 0;
}
if (name && strcmp(ifmt->name, name) == 0)
decode = 1;
}
if (!name)
break;
@@ -1351,22 +1296,12 @@ static int show_formats_devices(void *optctx, const char *opt, const char *arg,
int show_formats(void *optctx, const char *opt, const char *arg)
{
return show_formats_devices(optctx, opt, arg, 0, SHOW_DEFAULT);
}
int show_muxers(void *optctx, const char *opt, const char *arg)
{
return show_formats_devices(optctx, opt, arg, 0, SHOW_MUXERS);
}
int show_demuxers(void *optctx, const char *opt, const char *arg)
{
return show_formats_devices(optctx, opt, arg, 0, SHOW_DEMUXERS);
return show_formats_devices(optctx, opt, arg, 0);
}
int show_devices(void *optctx, const char *opt, const char *arg)
{
return show_formats_devices(optctx, opt, arg, 1, SHOW_DEFAULT);
return show_formats_devices(optctx, opt, arg, 1);
}
#define PRINT_CODEC_SUPPORTED(codec, field, type, list_name, term, get_name) \
@@ -1414,16 +1349,6 @@ static void print_codec(const AVCodec *c)
AV_CODEC_CAP_SLICE_THREADS |
AV_CODEC_CAP_AUTO_THREADS))
printf("threads ");
if (c->capabilities & AV_CODEC_CAP_AVOID_PROBING)
printf("avoidprobe ");
if (c->capabilities & AV_CODEC_CAP_INTRA_ONLY)
printf("intraonly ");
if (c->capabilities & AV_CODEC_CAP_LOSSLESS)
printf("lossless ");
if (c->capabilities & AV_CODEC_CAP_HARDWARE)
printf("hardware ");
if (c->capabilities & AV_CODEC_CAP_HYBRID)
printf("hybrid ");
if (!c->capabilities)
printf("none");
printf("\n");
@@ -1444,17 +1369,6 @@ static void print_codec(const AVCodec *c)
printf("\n");
}
if (avcodec_get_hw_config(c, 0)) {
printf(" Supported hardware devices: ");
for (int i = 0;; i++) {
const AVCodecHWConfig *config = avcodec_get_hw_config(c, i);
if (!config)
break;
printf("%s ", av_hwdevice_get_type_name(config->device_type));
}
printf("\n");
}
if (c->supported_framerates) {
const AVRational *fps = c->supported_framerates;
@@ -1653,11 +1567,10 @@ 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_iterate(&opaque)))
while ((bsf = av_bitstream_filter_next(bsf)))
printf("%s\n", bsf->name);
printf("\n");
return 0;
@@ -1683,7 +1596,6 @@ int show_filters(void *optctx, const char *opt, const char *arg)
#if CONFIG_AVFILTER
const AVFilter *filter = NULL;
char descr[64], *descr_cur;
void *opaque = NULL;
int i, j;
const AVFilterPad *pad;
@@ -1695,7 +1607,7 @@ int show_filters(void *optctx, const char *opt, const char *arg)
" V = Video input/output\n"
" N = Dynamic number and/or type of input/output\n"
" | = Source or sink filter\n");
while ((filter = av_filter_iterate(&opaque))) {
while ((filter = avfilter_next(filter))) {
descr_cur = descr;
for (i = 0; i < 2; i++) {
if (i) {
@@ -1758,7 +1670,7 @@ int show_pix_fmts(void *optctx, const char *opt, const char *arg)
#endif
while ((pix_desc = av_pix_fmt_desc_next(pix_desc))) {
enum AVPixelFormat av_unused pix_fmt = av_pix_fmt_desc_get_id(pix_desc);
enum AVPixelFormat pix_fmt = av_pix_fmt_desc_get_id(pix_desc);
printf("%c%c%c%c%c %-16s %d %2d\n",
sws_isSupportedInput (pix_fmt) ? 'I' : '.',
sws_isSupportedOutput(pix_fmt) ? 'O' : '.',
@@ -1952,25 +1864,6 @@ static void show_help_filter(const char *name)
}
#endif
static void show_help_bsf(const char *name)
{
const AVBitStreamFilter *bsf = av_bsf_get_by_name(name);
if (!name) {
av_log(NULL, AV_LOG_ERROR, "No bitstream filter name specified.\n");
return;
} else if (!bsf) {
av_log(NULL, AV_LOG_ERROR, "Unknown bit stream filter '%s'.\n", name);
return;
}
printf("Bit stream filter %s\n", bsf->name);
PRINT_CODEC_SUPPORTED(bsf, codec_ids, enum AVCodecID, "codecs",
AV_CODEC_ID_NONE, GET_CODEC_NAME);
if (bsf->priv_class)
show_help_children(bsf->priv_class, AV_OPT_FLAG_BSF_PARAM);
}
int show_help(void *optctx, const char *opt, const char *arg)
{
char *topic, *par;
@@ -1997,8 +1890,6 @@ int show_help(void *optctx, const char *opt, const char *arg)
} else if (!strcmp(topic, "filter")) {
show_help_filter(par);
#endif
} else if (!strcmp(topic, "bsf")) {
show_help_bsf(par);
} else {
show_help_default(topic, par);
}
@@ -2090,7 +1981,7 @@ 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;
@@ -2148,7 +2039,7 @@ 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;
}
@@ -2174,10 +2065,18 @@ void *grow_array(void *array, int elem_size, int *size, int new_size)
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 (displaymatrix)
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);
@@ -2200,7 +2099,7 @@ static int print_device_sources(AVInputFormat *fmt, AVDictionary *opts)
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);
printf("Audo-detected sources for %s:\n", fmt->name);
if (!fmt->get_device_list) {
ret = AVERROR(ENOSYS);
printf("Cannot list sources. Not implemented.\n");
@@ -2230,7 +2129,7 @@ static int print_device_sinks(AVOutputFormat *fmt, AVDictionary *opts)
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);
printf("Audo-detected sinks for %s:\n", fmt->name);
if (!fmt->get_device_list) {
ret = AVERROR(ENOSYS);
printf("Cannot list sinks. Not implemented.\n");

View File

@@ -19,8 +19,8 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef FFTOOLS_CMDUTILS_H
#define FFTOOLS_CMDUTILS_H
#ifndef CMDUTILS_H
#define CMDUTILS_H
#include <stdint.h>
@@ -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.
@@ -105,6 +100,12 @@ int opt_max_alloc(void *optctx, const char *opt, const char *arg);
int opt_codec_debug(void *optctx, const char *opt, const char *arg);
#if CONFIG_OPENCL
int opt_opencl(void *optctx, const char *opt, const char *arg);
int opt_opencl_bench(void *optctx, const char *opt, const char *arg);
#endif
/**
* Limit the execution time.
*/
@@ -149,7 +150,6 @@ typedef struct SpecifierOpt {
uint8_t *str;
int i;
int64_t i64;
uint64_t ui64;
float f;
double dbl;
} u;
@@ -201,47 +201,6 @@ typedef struct OptionDef {
void show_help_options(const OptionDef *options, const char *msg, int req_flags,
int rej_flags, int alt_flags);
#if CONFIG_AVDEVICE
#define CMDUTILS_COMMON_OPTIONS_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" }, \
#else
#define CMDUTILS_COMMON_OPTIONS_AVDEVICE
#endif
#define CMDUTILS_COMMON_OPTIONS \
{ "L", OPT_EXIT, { .func_arg = show_license }, "show license" }, \
{ "h", OPT_EXIT, { .func_arg = show_help }, "show help", "topic" }, \
{ "?", OPT_EXIT, { .func_arg = show_help }, "show help", "topic" }, \
{ "help", OPT_EXIT, { .func_arg = show_help }, "show help", "topic" }, \
{ "-help", OPT_EXIT, { .func_arg = show_help }, "show help", "topic" }, \
{ "version", OPT_EXIT, { .func_arg = show_version }, "show version" }, \
{ "buildconf", OPT_EXIT, { .func_arg = show_buildconf }, "show build configuration" }, \
{ "formats", OPT_EXIT, { .func_arg = show_formats }, "show available formats" }, \
{ "muxers", OPT_EXIT, { .func_arg = show_muxers }, "show available muxers" }, \
{ "demuxers", OPT_EXIT, { .func_arg = show_demuxers }, "show available demuxers" }, \
{ "devices", OPT_EXIT, { .func_arg = show_devices }, "show available devices" }, \
{ "codecs", OPT_EXIT, { .func_arg = show_codecs }, "show available codecs" }, \
{ "decoders", OPT_EXIT, { .func_arg = show_decoders }, "show available decoders" }, \
{ "encoders", OPT_EXIT, { .func_arg = show_encoders }, "show available encoders" }, \
{ "bsfs", OPT_EXIT, { .func_arg = show_bsfs }, "show available bit stream filters" }, \
{ "protocols", OPT_EXIT, { .func_arg = show_protocols }, "show available protocols" }, \
{ "filters", OPT_EXIT, { .func_arg = show_filters }, "show available filters" }, \
{ "pix_fmts", OPT_EXIT, { .func_arg = show_pix_fmts }, "show available pixel formats" }, \
{ "layouts", OPT_EXIT, { .func_arg = show_layouts }, "show standard channel layouts" }, \
{ "sample_fmts", OPT_EXIT, { .func_arg = show_sample_fmts }, "show available audio sample formats" }, \
{ "colors", OPT_EXIT, { .func_arg = show_colors }, "show available color names" }, \
{ "loglevel", HAS_ARG, { .func_arg = opt_loglevel }, "set logging level", "loglevel" }, \
{ "v", HAS_ARG, { .func_arg = opt_loglevel }, "set logging level", "loglevel" }, \
{ "report", 0, { (void*)opt_report }, "generate a report" }, \
{ "max_alloc", HAS_ARG, { .func_arg = opt_max_alloc }, "set maximum size of a single allocated block", "bytes" }, \
{ "cpuflags", HAS_ARG | OPT_EXPERT, { .func_arg = opt_cpuflags }, "force specific cpu flags", "flags" }, \
{ "hide_banner", OPT_BOOL | OPT_EXPERT, {&hide_banner}, "do not show program banner", "hide_banner" }, \
CMDUTILS_COMMON_OPTIONS_AVDEVICE \
/**
* Show help for all options with given flags in class and all its
* children.
@@ -477,20 +436,6 @@ int show_license(void *optctx, const char *opt, const char *arg);
*/
int show_formats(void *optctx, const char *opt, const char *arg);
/**
* Print a listing containing all the muxers supported by the
* program (including devices).
* This option processing function does not utilize the arguments.
*/
int show_muxers(void *optctx, const char *opt, const char *arg);
/**
* Print a listing containing all the demuxer supported by the
* program (including devices).
* This option processing function does not utilize the arguments.
*/
int show_demuxers(void *optctx, const char *opt, const char *arg);
/**
* Print a listing containing all the devices supported by the
* program.
@@ -500,13 +445,13 @@ int show_devices(void *optctx, const char *opt, const char *arg);
#if CONFIG_AVDEVICE
/**
* Print a listing containing autodetected sinks of the output device.
* Print a listing containing audodetected 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.
* Print a listing containing audodetected 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);
@@ -625,9 +570,6 @@ void *grow_array(void *array, int elem_size, int *size, int new_size);
#define GET_PIX_FMT_NAME(pix_fmt)\
const char *name = av_get_pix_fmt_name(pix_fmt);
#define GET_CODEC_NAME(id)\
const char *name = avcodec_descriptor_get(id)->name;
#define GET_SAMPLE_FMT_NAME(sample_fmt)\
const char *name = av_get_sample_fmt_name(sample_fmt)
@@ -645,4 +587,4 @@ void *grow_array(void *array, int elem_size, int *size, int new_size);
double get_rotation(AVStream *st);
#endif /* FFTOOLS_CMDUTILS_H */
#endif /* CMDUTILS_H */

35
cmdutils_common_opts.h Normal file
View File

@@ -0,0 +1,35 @@
{ "L" , OPT_EXIT, {.func_arg = show_license}, "show license" },
{ "h" , OPT_EXIT, {.func_arg = show_help}, "show help", "topic" },
{ "?" , OPT_EXIT, {.func_arg = show_help}, "show help", "topic" },
{ "help" , OPT_EXIT, {.func_arg = show_help}, "show help", "topic" },
{ "-help" , OPT_EXIT, {.func_arg = show_help}, "show help", "topic" },
{ "version" , OPT_EXIT, {.func_arg = show_version}, "show version" },
{ "buildconf" , OPT_EXIT, {.func_arg = show_buildconf}, "show build configuration" },
{ "formats" , OPT_EXIT, {.func_arg = show_formats }, "show available formats" },
{ "devices" , OPT_EXIT, {.func_arg = show_devices }, "show available devices" },
{ "codecs" , OPT_EXIT, {.func_arg = show_codecs }, "show available codecs" },
{ "decoders" , OPT_EXIT, {.func_arg = show_decoders }, "show available decoders" },
{ "encoders" , OPT_EXIT, {.func_arg = show_encoders }, "show available encoders" },
{ "bsfs" , OPT_EXIT, {.func_arg = show_bsfs }, "show available bit stream filters" },
{ "protocols" , OPT_EXIT, {.func_arg = show_protocols}, "show available protocols" },
{ "filters" , OPT_EXIT, {.func_arg = show_filters }, "show available filters" },
{ "pix_fmts" , OPT_EXIT, {.func_arg = show_pix_fmts }, "show available pixel formats" },
{ "layouts" , OPT_EXIT, {.func_arg = show_layouts }, "show standard channel layouts" },
{ "sample_fmts", OPT_EXIT, {.func_arg = show_sample_fmts }, "show available audio sample formats" },
{ "colors" , OPT_EXIT, {.func_arg = show_colors }, "show available color names" },
{ "loglevel" , HAS_ARG, {.func_arg = opt_loglevel}, "set logging level", "loglevel" },
{ "v", HAS_ARG, {.func_arg = opt_loglevel}, "set logging level", "loglevel" },
{ "report" , 0, {(void*)opt_report}, "generate a report" },
{ "max_alloc" , HAS_ARG, {.func_arg = opt_max_alloc}, "set maximum size of a single allocated block", "bytes" },
{ "cpuflags" , HAS_ARG | OPT_EXPERT, { .func_arg = opt_cpuflags }, "force specific cpu flags", "flags" },
{ "hide_banner", OPT_BOOL | OPT_EXPERT, {&hide_banner}, "do not show program banner", "hide_banner" },
#if CONFIG_OPENCL
{ "opencl_bench", OPT_EXIT, {.func_arg = opt_opencl_bench}, "run benchmark on all OpenCL devices and show results" },
{ "opencl_options", HAS_ARG, {.func_arg = opt_opencl}, "set OpenCL environment options" },
#endif
#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

278
cmdutils_opencl.c Normal file
View File

@@ -0,0 +1,278 @@
/*
* Copyright (C) 2013 Lenny Wang
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/opt.h"
#include "libavutil/time.h"
#include "libavutil/log.h"
#include "libavutil/opencl.h"
#include "libavutil/avstring.h"
#include "cmdutils.h"
typedef struct {
int platform_idx;
int device_idx;
char device_name[64];
int64_t runtime;
} OpenCLDeviceBenchmark;
const char *ocl_bench_source = AV_OPENCL_KERNEL(
inline unsigned char clip_uint8(int a)
{
if (a & (~0xFF))
return (-a)>>31;
else
return a;
}
kernel void unsharp_bench(
global unsigned char *src,
global unsigned char *dst,
global int *mask,
int width,
int height)
{
int i, j, local_idx, lc_idx, sum = 0;
int2 thread_idx, block_idx, global_idx, lm_idx;
thread_idx.x = get_local_id(0);
thread_idx.y = get_local_id(1);
block_idx.x = get_group_id(0);
block_idx.y = get_group_id(1);
global_idx.x = get_global_id(0);
global_idx.y = get_global_id(1);
local uchar data[32][32];
local int lc[128];
for (i = 0; i <= 1; i++) {
lm_idx.y = -8 + (block_idx.y + i) * 16 + thread_idx.y;
lm_idx.y = lm_idx.y < 0 ? 0 : lm_idx.y;
lm_idx.y = lm_idx.y >= height ? height - 1: lm_idx.y;
for (j = 0; j <= 1; j++) {
lm_idx.x = -8 + (block_idx.x + j) * 16 + thread_idx.x;
lm_idx.x = lm_idx.x < 0 ? 0 : lm_idx.x;
lm_idx.x = lm_idx.x >= width ? width - 1: lm_idx.x;
data[i*16 + thread_idx.y][j*16 + thread_idx.x] = src[lm_idx.y*width + lm_idx.x];
}
}
local_idx = thread_idx.y*16 + thread_idx.x;
if (local_idx < 128)
lc[local_idx] = mask[local_idx];
barrier(CLK_LOCAL_MEM_FENCE);
\n#pragma unroll\n
for (i = -4; i <= 4; i++) {
lm_idx.y = 8 + i + thread_idx.y;
\n#pragma unroll\n
for (j = -4; j <= 4; j++) {
lm_idx.x = 8 + j + thread_idx.x;
lc_idx = (i + 4)*8 + j + 4;
sum += (int)data[lm_idx.y][lm_idx.x] * lc[lc_idx];
}
}
int temp = (int)data[thread_idx.y + 8][thread_idx.x + 8];
int res = temp + (((temp - (int)((sum + 1<<15) >> 16))) >> 16);
if (global_idx.x < width && global_idx.y < height)
dst[global_idx.x + global_idx.y*width] = clip_uint8(res);
}
);
#define OCLCHECK(method, ... ) \
do { \
status = method(__VA_ARGS__); \
if (status != CL_SUCCESS) { \
av_log(NULL, AV_LOG_ERROR, # method " error '%s'\n", \
av_opencl_errstr(status)); \
ret = AVERROR_EXTERNAL; \
goto end; \
} \
} while (0)
#define CREATEBUF(out, flags, size) \
do { \
out = clCreateBuffer(ext_opencl_env->context, flags, size, NULL, &status); \
if (status != CL_SUCCESS) { \
av_log(NULL, AV_LOG_ERROR, "Could not create OpenCL buffer\n"); \
ret = AVERROR_EXTERNAL; \
goto end; \
} \
} while (0)
static void fill_rand_int(int *data, int n)
{
int i;
srand(av_gettime());
for (i = 0; i < n; i++)
data[i] = rand();
}
#define OPENCL_NB_ITER 5
static int64_t run_opencl_bench(AVOpenCLExternalEnv *ext_opencl_env)
{
int i, arg = 0, width = 1920, height = 1088;
int64_t start, ret = 0;
cl_int status;
size_t kernel_len;
char *inbuf;
int *mask;
int buf_size = width * height * sizeof(char);
int mask_size = sizeof(uint32_t) * 128;
cl_mem cl_mask, cl_inbuf, cl_outbuf;
cl_kernel kernel = NULL;
cl_program program = NULL;
size_t local_work_size_2d[2] = {16, 16};
size_t global_work_size_2d[2] = {(size_t)width, (size_t)height};
if (!(inbuf = av_malloc(buf_size)) || !(mask = av_malloc(mask_size))) {
av_log(NULL, AV_LOG_ERROR, "Out of memory\n");
ret = AVERROR(ENOMEM);
goto end;
}
fill_rand_int((int*)inbuf, buf_size/4);
fill_rand_int(mask, mask_size/4);
CREATEBUF(cl_mask, CL_MEM_READ_ONLY, mask_size);
CREATEBUF(cl_inbuf, CL_MEM_READ_ONLY, buf_size);
CREATEBUF(cl_outbuf, CL_MEM_READ_WRITE, buf_size);
kernel_len = strlen(ocl_bench_source);
program = clCreateProgramWithSource(ext_opencl_env->context, 1, &ocl_bench_source,
&kernel_len, &status);
if (status != CL_SUCCESS || !program) {
av_log(NULL, AV_LOG_ERROR, "OpenCL unable to create benchmark program\n");
ret = AVERROR_EXTERNAL;
goto end;
}
status = clBuildProgram(program, 1, &(ext_opencl_env->device_id), NULL, NULL, NULL);
if (status != CL_SUCCESS) {
av_log(NULL, AV_LOG_ERROR, "OpenCL unable to build benchmark program\n");
ret = AVERROR_EXTERNAL;
goto end;
}
kernel = clCreateKernel(program, "unsharp_bench", &status);
if (status != CL_SUCCESS) {
av_log(NULL, AV_LOG_ERROR, "OpenCL unable to create benchmark kernel\n");
ret = AVERROR_EXTERNAL;
goto end;
}
OCLCHECK(clEnqueueWriteBuffer, ext_opencl_env->command_queue, cl_inbuf, CL_TRUE, 0,
buf_size, inbuf, 0, NULL, NULL);
OCLCHECK(clEnqueueWriteBuffer, ext_opencl_env->command_queue, cl_mask, CL_TRUE, 0,
mask_size, mask, 0, NULL, NULL);
OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_mem), &cl_inbuf);
OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_mem), &cl_outbuf);
OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_mem), &cl_mask);
OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_int), &width);
OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_int), &height);
start = av_gettime_relative();
for (i = 0; i < OPENCL_NB_ITER; i++)
OCLCHECK(clEnqueueNDRangeKernel, ext_opencl_env->command_queue, kernel, 2, NULL,
global_work_size_2d, local_work_size_2d, 0, NULL, NULL);
clFinish(ext_opencl_env->command_queue);
ret = (av_gettime_relative() - start)/OPENCL_NB_ITER;
end:
if (kernel)
clReleaseKernel(kernel);
if (program)
clReleaseProgram(program);
if (cl_inbuf)
clReleaseMemObject(cl_inbuf);
if (cl_outbuf)
clReleaseMemObject(cl_outbuf);
if (cl_mask)
clReleaseMemObject(cl_mask);
av_free(inbuf);
av_free(mask);
return ret;
}
static int compare_ocl_device_desc(const void *a, const void *b)
{
const OpenCLDeviceBenchmark* va = (const OpenCLDeviceBenchmark*)a;
const OpenCLDeviceBenchmark* vb = (const OpenCLDeviceBenchmark*)b;
return FFDIFFSIGN(va->runtime , vb->runtime);
}
int opt_opencl_bench(void *optctx, const char *opt, const char *arg)
{
int i, j, nb_devices = 0, count = 0;
int64_t score = 0;
AVOpenCLDeviceList *device_list;
AVOpenCLDeviceNode *device_node = NULL;
OpenCLDeviceBenchmark *devices = NULL;
cl_platform_id platform;
av_opencl_get_device_list(&device_list);
for (i = 0; i < device_list->platform_num; i++)
nb_devices += device_list->platform_node[i]->device_num;
if (!nb_devices) {
av_log(NULL, AV_LOG_ERROR, "No OpenCL device detected!\n");
return AVERROR(EINVAL);
}
if (!(devices = av_malloc_array(nb_devices, sizeof(OpenCLDeviceBenchmark)))) {
av_log(NULL, AV_LOG_ERROR, "Could not allocate buffer\n");
return AVERROR(ENOMEM);
}
for (i = 0; i < device_list->platform_num; i++) {
for (j = 0; j < device_list->platform_node[i]->device_num; j++) {
device_node = device_list->platform_node[i]->device_node[j];
platform = device_list->platform_node[i]->platform_id;
score = av_opencl_benchmark(device_node, platform, run_opencl_bench);
if (score > 0) {
devices[count].platform_idx = i;
devices[count].device_idx = j;
devices[count].runtime = score;
av_strlcpy(devices[count].device_name, device_node->device_name,
sizeof(devices[count].device_name));
count++;
}
}
}
qsort(devices, count, sizeof(OpenCLDeviceBenchmark), compare_ocl_device_desc);
fprintf(stderr, "platform_idx\tdevice_idx\tdevice_name\truntime\n");
for (i = 0; i < count; i++)
fprintf(stdout, "%d\t%d\t%s\t%"PRId64"\n",
devices[i].platform_idx, devices[i].device_idx,
devices[i].device_name, devices[i].runtime);
av_opencl_free_device_list(&device_list);
av_free(devices);
return 0;
}
int opt_opencl(void *optctx, const char *opt, const char *arg)
{
char *key, *value;
const char *opts = arg;
int ret = 0;
while (*opts) {
ret = av_opt_get_key_value(&opts, "=", ":", 0, &key, &value);
if (ret < 0)
return ret;
ret = av_opencl_set_option(key, value);
if (ret < 0)
return ret;
if (*opts)
opts++;
}
return ret;
}

View File

@@ -2,12 +2,15 @@
# common bits used by all libraries
#
DEFAULT_X86ASMD=.dbg
# first so "all" becomes default target
all: all-yes
DEFAULT_YASMD=.dbg
ifeq ($(DBG),1)
X86ASMD=$(DEFAULT_X86ASMD)
YASMD=$(DEFAULT_YASMD)
else
X86ASMD=
YASMD=
endif
ifndef SUBDIR
@@ -15,8 +18,8 @@ ifndef SUBDIR
ifndef V
Q = @
ECHO = printf "$(1)\t%s\n" $(2)
BRIEF = CC CXX OBJCC HOSTCC HOSTLD AS X86ASM AR LD STRIP CP WINDRES NVCC
SILENT = DEPCC DEPHOSTCC DEPAS DEPX86ASM RANLIB RM
BRIEF = CC CXX OBJCC HOSTCC HOSTLD AS YASM AR LD STRIP CP WINDRES
SILENT = DEPCC DEPHOSTCC DEPAS DEPYASM RANLIB RM
MSG = $@
M = @$(call ECHO,$(TAG),$@);
@@ -36,9 +39,8 @@ CCFLAGS = $(CPPFLAGS) $(CFLAGS)
OBJCFLAGS += $(EOBJCFLAGS)
OBJCCFLAGS = $(CPPFLAGS) $(CFLAGS) $(OBJCFLAGS)
ASFLAGS := $(CPPFLAGS) $(ASFLAGS)
CXXFLAGS := $(CPPFLAGS) $(CFLAGS) $(CXXFLAGS)
X86ASMFLAGS += $(IFLAGS:%=%/) -I$(<D)/ -Pconfig.asm
NVCCFLAGS += -ptx
CXXFLAGS += $(CPPFLAGS) $(CFLAGS)
YASMFLAGS += $(IFLAGS:%=%/) -Pconfig.asm
HOSTCCFLAGS = $(IFLAGS) $(HOSTCPPFLAGS) $(HOSTCFLAGS)
LDFLAGS := $(ALLFFLIBS:%=$(LD_PATH)lib%) $(LDFLAGS)
@@ -52,9 +54,7 @@ COMPILE_C = $(call COMPILE,CC)
COMPILE_CXX = $(call COMPILE,CXX)
COMPILE_S = $(call COMPILE,AS)
COMPILE_M = $(call COMPILE,OBJCC)
COMPILE_X86ASM = $(call COMPILE,X86ASM)
COMPILE_HOSTC = $(call COMPILE,HOSTCC)
COMPILE_NVCC = $(call COMPILE,NVCC)
%.o: %.c
$(COMPILE_C)
@@ -74,14 +74,6 @@ COMPILE_NVCC = $(call COMPILE,NVCC)
%_host.o: %.c
$(COMPILE_HOSTC)
%$(DEFAULT_X86ASMD).asm: %.asm
$(DEPX86ASM) $(X86ASMFLAGS) -M -o $@ $< > $(@:.asm=.d)
$(X86ASM) $(X86ASMFLAGS) -e $< | sed '/^%/d;/^$$/d;' > $@
%.o: %.asm
$(COMPILE_X86ASM)
-$(if $(ASMSTRIPFLAGS), $(STRIP) $(ASMSTRIPFLAGS) $@)
%.o: %.rc
$(WINDRES) $(IFLAGS) --preprocessor "$(DEPWINDRES) -E -xc-header -DRC_INVOKED $(CC_DEPFLAGS)" -o $@ $<
@@ -91,13 +83,12 @@ COMPILE_NVCC = $(call COMPILE,NVCC)
%.h.c:
$(Q)echo '#include "$*.h"' >$@
%.ptx: %.cu
$(COMPILE_NVCC)
%.ver: %.v
$(Q)sed 's/$$MAJOR/$($(basename $(@F))_VERSION_MAJOR)/' $^ | sed -e 's/:/:\
/' -e 's/; /;\
/g' > $@
%.ptx.c: %.ptx
$(Q)sh $(SRC_PATH)/compat/cuda/ptx2c.sh $@ $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<)
%.c %.h %.pc %.ver %.version: TAG = GEN
%.c %.h: TAG = GEN
# Dummy rule to stop make trying to rebuild removed or renamed headers
%.h:
@@ -111,7 +102,7 @@ COMPILE_NVCC = $(call COMPILE,NVCC)
$(OBJS):
endif
include $(SRC_PATH)/ffbuild/arch.mak
include $(SRC_PATH)/arch.mak
OBJS += $(OBJS-yes)
SLIBOBJS += $(SLIBOBJS-yes)
@@ -119,12 +110,12 @@ FFLIBS := $($(NAME)_FFLIBS) $(FFLIBS-yes) $(FFLIBS)
TESTPROGS += $(TESTPROGS-yes)
LDLIBS = $(FFLIBS:%=%$(BUILDSUF))
FFEXTRALIBS := $(LDLIBS:%=$(LD_LIB)) $(foreach lib,EXTRALIBS-$(NAME) $(FFLIBS:%=EXTRALIBS-%),$($(lib))) $(EXTRALIBS)
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)
@@ -141,10 +132,8 @@ ALLHEADERS := $(subst $(SRC_DIR)/,$(SUBDIR),$(wildcard $(SRC_DIR)/*.h $(SRC_DIR)
SKIPHEADERS += $(ARCH_HEADERS:%=$(ARCH)/%) $(SKIPHEADERS-)
SKIPHEADERS := $(SKIPHEADERS:%=$(SUBDIR)%)
HOBJS = $(filter-out $(SKIPHEADERS:.h=.h.o),$(ALLHEADERS:.h=.h.o))
PTXOBJS = $(filter %.ptx.o,$(OBJS))
$(HOBJS): CCFLAGS += $(CFLAGS_HEADERS)
checkheaders: $(HOBJS)
.SECONDARY: $(HOBJS:.o=.c) $(PTXOBJS:.o=.c) $(PTXOBJS:.o=)
.SECONDARY: $(HOBJS:.o=.c)
alltools: $(TOOLS)
@@ -152,7 +141,7 @@ $(HOSTOBJS): %.o: %.c
$(COMPILE_HOSTC)
$(HOSTPROGS): %$(HOSTEXESUF): %.o
$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $^ $(HOSTEXTRALIBS)
$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $^ $(HOSTLIBS)
$(OBJS): | $(sort $(dir $(OBJS)))
$(HOBJS): | $(sort $(dir $(HOBJS)))
@@ -163,14 +152,17 @@ $(TOOLOBJS): | tools
OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(SLIBOBJS) $(TESTOBJS))
CLEANSUFFIXES = *.d *.gcda *.gcno *.h.c *.ho *.map *.o *.pc *.ptx *.ptx.c *.ver *.version *$(DEFAULT_X86ASMD).asm *~
CLEANSUFFIXES = *.d *.o *~ *.h.c *.map *.ver *.ver-sol2 *.ho *.gcno *.gcda *$(DEFAULT_YASMD).asm
DISTCLEANSUFFIXES = *.pc
LIBSUFFIXES = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a
define RULES
clean::
$(RM) $(HOSTPROGS) $(TESTPROGS) $(TOOLS)
$(RM) $(OBJS) $(OBJS:.o=.d) $(OBJS:.o=$(DEFAULT_YASMD).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_X86ASMD).d)
-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d) $(SLIBOBJS:.o=.d)) $(OBJS:.o=$(DEFAULT_YASMD).d)

View File

@@ -1,176 +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
*/
/*
* based on vlc_atomic.h from VLC
* Copyright (C) 2010 Rémi Denis-Courmont
*/
#ifndef COMPAT_ATOMICS_DUMMY_STDATOMIC_H
#define COMPAT_ATOMICS_DUMMY_STDATOMIC_H
#include <stdint.h>
#define ATOMIC_FLAG_INIT 0
#define ATOMIC_VAR_INIT(value) (value)
#define atomic_init(obj, value) \
do { \
*(obj) = (value); \
} while(0)
#define kill_dependency(y) ((void)0)
#define atomic_thread_fence(order) \
((void)0)
#define atomic_signal_fence(order) \
((void)0)
#define atomic_is_lock_free(obj) 0
typedef intptr_t atomic_flag;
typedef intptr_t atomic_bool;
typedef intptr_t atomic_char;
typedef intptr_t atomic_schar;
typedef intptr_t atomic_uchar;
typedef intptr_t atomic_short;
typedef intptr_t atomic_ushort;
typedef intptr_t atomic_int;
typedef intptr_t atomic_uint;
typedef intptr_t atomic_long;
typedef intptr_t atomic_ulong;
typedef intptr_t atomic_llong;
typedef intptr_t atomic_ullong;
typedef intptr_t atomic_wchar_t;
typedef intptr_t atomic_int_least8_t;
typedef intptr_t atomic_uint_least8_t;
typedef intptr_t atomic_int_least16_t;
typedef intptr_t atomic_uint_least16_t;
typedef intptr_t atomic_int_least32_t;
typedef intptr_t atomic_uint_least32_t;
typedef intptr_t atomic_int_least64_t;
typedef intptr_t atomic_uint_least64_t;
typedef intptr_t atomic_int_fast8_t;
typedef intptr_t atomic_uint_fast8_t;
typedef intptr_t atomic_int_fast16_t;
typedef intptr_t atomic_uint_fast16_t;
typedef intptr_t atomic_int_fast32_t;
typedef intptr_t atomic_uint_fast32_t;
typedef intptr_t atomic_int_fast64_t;
typedef intptr_t atomic_uint_fast64_t;
typedef intptr_t atomic_intptr_t;
typedef intptr_t atomic_uintptr_t;
typedef intptr_t atomic_size_t;
typedef intptr_t atomic_ptrdiff_t;
typedef intptr_t atomic_intmax_t;
typedef intptr_t atomic_uintmax_t;
#define atomic_store(object, desired) \
do { \
*(object) = (desired); \
} while (0)
#define atomic_store_explicit(object, desired, order) \
atomic_store(object, desired)
#define atomic_load(object) \
(*(object))
#define atomic_load_explicit(object, order) \
atomic_load(object)
static inline intptr_t atomic_exchange(intptr_t *object, intptr_t desired)
{
intptr_t ret = *object;
*object = desired;
return ret;
}
#define atomic_exchange_explicit(object, desired, order) \
atomic_exchange(object, desired)
static inline int atomic_compare_exchange_strong(intptr_t *object, intptr_t *expected,
intptr_t desired)
{
int ret;
if (*object == *expected) {
*object = desired;
ret = 1;
} else {
*expected = *object;
ret = 0;
}
return ret;
}
#define atomic_compare_exchange_strong_explicit(object, expected, desired, success, failure) \
atomic_compare_exchange_strong(object, expected, desired)
#define atomic_compare_exchange_weak(object, expected, desired) \
atomic_compare_exchange_strong(object, expected, desired)
#define atomic_compare_exchange_weak_explicit(object, expected, desired, success, failure) \
atomic_compare_exchange_weak(object, expected, desired)
#define FETCH_MODIFY(opname, op) \
static inline intptr_t atomic_fetch_ ## opname(intptr_t *object, intptr_t operand) \
{ \
intptr_t ret; \
ret = *object; \
*object = *object op operand; \
return ret; \
}
FETCH_MODIFY(add, +)
FETCH_MODIFY(sub, -)
FETCH_MODIFY(or, |)
FETCH_MODIFY(xor, ^)
FETCH_MODIFY(and, &)
#undef FETCH_MODIFY
#define atomic_fetch_add_explicit(object, operand, order) \
atomic_fetch_add(object, operand)
#define atomic_fetch_sub_explicit(object, operand, order) \
atomic_fetch_sub(object, operand)
#define atomic_fetch_or_explicit(object, operand, order) \
atomic_fetch_or(object, operand)
#define atomic_fetch_xor_explicit(object, operand, order) \
atomic_fetch_xor(object, operand)
#define atomic_fetch_and_explicit(object, operand, order) \
atomic_fetch_and(object, operand)
#define atomic_flag_test_and_set(object) \
atomic_exchange(object, 1)
#define atomic_flag_test_and_set_explicit(object, order) \
atomic_flag_test_and_set(object)
#define atomic_flag_clear(object) \
atomic_store(object, 0)
#define atomic_flag_clear_explicit(object, order) \
atomic_flag_clear(object)
#endif /* COMPAT_ATOMICS_DUMMY_STDATOMIC_H */

View File

@@ -1,173 +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
*/
/*
* based on vlc_atomic.h from VLC
* Copyright (C) 2010 Rémi Denis-Courmont
*/
#ifndef COMPAT_ATOMICS_GCC_STDATOMIC_H
#define COMPAT_ATOMICS_GCC_STDATOMIC_H
#include <stddef.h>
#include <stdint.h>
#define ATOMIC_FLAG_INIT 0
#define ATOMIC_VAR_INIT(value) (value)
#define atomic_init(obj, value) \
do { \
*(obj) = (value); \
} while(0)
#define kill_dependency(y) ((void)0)
#define atomic_thread_fence(order) \
__sync_synchronize()
#define atomic_signal_fence(order) \
((void)0)
#define atomic_is_lock_free(obj) 0
typedef _Bool atomic_flag;
typedef _Bool atomic_bool;
typedef char atomic_char;
typedef signed char atomic_schar;
typedef unsigned char atomic_uchar;
typedef short atomic_short;
typedef unsigned short atomic_ushort;
typedef int atomic_int;
typedef unsigned int atomic_uint;
typedef long atomic_long;
typedef unsigned long atomic_ulong;
typedef long long atomic_llong;
typedef unsigned long long atomic_ullong;
typedef wchar_t atomic_wchar_t;
typedef int_least8_t atomic_int_least8_t;
typedef uint_least8_t atomic_uint_least8_t;
typedef int_least16_t atomic_int_least16_t;
typedef uint_least16_t atomic_uint_least16_t;
typedef int_least32_t atomic_int_least32_t;
typedef uint_least32_t atomic_uint_least32_t;
typedef int_least64_t atomic_int_least64_t;
typedef uint_least64_t atomic_uint_least64_t;
typedef int_fast8_t atomic_int_fast8_t;
typedef uint_fast8_t atomic_uint_fast8_t;
typedef int_fast16_t atomic_int_fast16_t;
typedef uint_fast16_t atomic_uint_fast16_t;
typedef int_fast32_t atomic_int_fast32_t;
typedef uint_fast32_t atomic_uint_fast32_t;
typedef int_fast64_t atomic_int_fast64_t;
typedef uint_fast64_t atomic_uint_fast64_t;
typedef intptr_t atomic_intptr_t;
typedef uintptr_t atomic_uintptr_t;
typedef size_t atomic_size_t;
typedef ptrdiff_t atomic_ptrdiff_t;
typedef intmax_t atomic_intmax_t;
typedef uintmax_t atomic_uintmax_t;
#define atomic_store(object, desired) \
do { \
*(object) = (desired); \
__sync_synchronize(); \
} while (0)
#define atomic_store_explicit(object, desired, order) \
atomic_store(object, desired)
#define atomic_load(object) \
(__sync_synchronize(), *(object))
#define atomic_load_explicit(object, order) \
atomic_load(object)
#define atomic_exchange(object, desired) \
({ \
__typeof__(object) _obj = (object); \
__typeof__(*object) _old; \
do \
_old = atomic_load(_obj); \
while (!__sync_bool_compare_and_swap(_obj, _old, (desired))); \
_old; \
})
#define atomic_exchange_explicit(object, desired, order) \
atomic_exchange(object, desired)
#define atomic_compare_exchange_strong(object, expected, desired) \
({ \
__typeof__(object) _exp = (expected); \
__typeof__(*object) _old = *_exp; \
*_exp = __sync_val_compare_and_swap((object), _old, (desired)); \
*_exp == _old; \
})
#define atomic_compare_exchange_strong_explicit(object, expected, desired, success, failure) \
atomic_compare_exchange_strong(object, expected, desired)
#define atomic_compare_exchange_weak(object, expected, desired) \
atomic_compare_exchange_strong(object, expected, desired)
#define atomic_compare_exchange_weak_explicit(object, expected, desired, success, failure) \
atomic_compare_exchange_weak(object, expected, desired)
#define atomic_fetch_add(object, operand) \
__sync_fetch_and_add(object, operand)
#define atomic_fetch_add_explicit(object, operand, order) \
atomic_fetch_add(object, operand)
#define atomic_fetch_sub(object, operand) \
__sync_fetch_and_sub(object, operand)
#define atomic_fetch_sub_explicit(object, operand, order) \
atomic_fetch_sub(object, operand)
#define atomic_fetch_or(object, operand) \
__sync_fetch_and_or(object, operand)
#define atomic_fetch_or_explicit(object, operand, order) \
atomic_fetch_or(object, operand)
#define atomic_fetch_xor(object, operand) \
__sync_fetch_and_xor(object, operand)
#define atomic_fetch_xor_explicit(object, operand, order) \
atomic_fetch_xor(object, operand)
#define atomic_fetch_and(object, operand) \
__sync_fetch_and_and(object, operand)
#define atomic_fetch_and_explicit(object, operand, order) \
atomic_fetch_and(object, operand)
#define atomic_flag_test_and_set(object) \
atomic_exchange(object, 1)
#define atomic_flag_test_and_set_explicit(object, order) \
atomic_flag_test_and_set(object)
#define atomic_flag_clear(object) \
atomic_store(object, 0)
#define atomic_flag_clear_explicit(object, order) \
atomic_flag_clear(object)
#endif /* COMPAT_ATOMICS_GCC_STDATOMIC_H */

View File

@@ -1,39 +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
*/
/*
* based on vlc_atomic.h from VLC
* Copyright (C) 2010 Rémi Denis-Courmont
*/
#include <pthread.h>
#include <stdint.h>
#include "stdatomic.h"
static pthread_mutex_t atomic_lock = PTHREAD_MUTEX_INITIALIZER;
void avpriv_atomic_lock(void)
{
pthread_mutex_lock(&atomic_lock);
}
void avpriv_atomic_unlock(void)
{
pthread_mutex_unlock(&atomic_lock);
}

View File

@@ -1,197 +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
*/
/*
* based on vlc_atomic.h from VLC
* Copyright (C) 2010 Rémi Denis-Courmont
*/
#ifndef COMPAT_ATOMICS_PTHREAD_STDATOMIC_H
#define COMPAT_ATOMICS_PTHREAD_STDATOMIC_H
#include <stdint.h>
#define ATOMIC_FLAG_INIT 0
#define ATOMIC_VAR_INIT(value) (value)
#define atomic_init(obj, value) \
do { \
*(obj) = (value); \
} while(0)
#define kill_dependency(y) ((void)0)
#define atomic_signal_fence(order) \
((void)0)
#define atomic_is_lock_free(obj) 0
typedef intptr_t atomic_flag;
typedef intptr_t atomic_bool;
typedef intptr_t atomic_char;
typedef intptr_t atomic_schar;
typedef intptr_t atomic_uchar;
typedef intptr_t atomic_short;
typedef intptr_t atomic_ushort;
typedef intptr_t atomic_int;
typedef intptr_t atomic_uint;
typedef intptr_t atomic_long;
typedef intptr_t atomic_ulong;
typedef intptr_t atomic_llong;
typedef intptr_t atomic_ullong;
typedef intptr_t atomic_wchar_t;
typedef intptr_t atomic_int_least8_t;
typedef intptr_t atomic_uint_least8_t;
typedef intptr_t atomic_int_least16_t;
typedef intptr_t atomic_uint_least16_t;
typedef intptr_t atomic_int_least32_t;
typedef intptr_t atomic_uint_least32_t;
typedef intptr_t atomic_int_least64_t;
typedef intptr_t atomic_uint_least64_t;
typedef intptr_t atomic_int_fast8_t;
typedef intptr_t atomic_uint_fast8_t;
typedef intptr_t atomic_int_fast16_t;
typedef intptr_t atomic_uint_fast16_t;
typedef intptr_t atomic_int_fast32_t;
typedef intptr_t atomic_uint_fast32_t;
typedef intptr_t atomic_int_fast64_t;
typedef intptr_t atomic_uint_fast64_t;
typedef intptr_t atomic_intptr_t;
typedef intptr_t atomic_uintptr_t;
typedef intptr_t atomic_size_t;
typedef intptr_t atomic_ptrdiff_t;
typedef intptr_t atomic_intmax_t;
typedef intptr_t atomic_uintmax_t;
void avpriv_atomic_lock(void);
void avpriv_atomic_unlock(void);
static inline void atomic_thread_fence(int order)
{
avpriv_atomic_lock();
avpriv_atomic_unlock();
}
static inline void atomic_store(intptr_t *object, intptr_t desired)
{
avpriv_atomic_lock();
*object = desired;
avpriv_atomic_unlock();
}
#define atomic_store_explicit(object, desired, order) \
atomic_store(object, desired)
static inline intptr_t atomic_load(intptr_t *object)
{
intptr_t ret;
avpriv_atomic_lock();
ret = *object;
avpriv_atomic_unlock();
return ret;
}
#define atomic_load_explicit(object, order) \
atomic_load(object)
static inline intptr_t atomic_exchange(intptr_t *object, intptr_t desired)
{
intptr_t ret;
avpriv_atomic_lock();
ret = *object;
*object = desired;
avpriv_atomic_unlock();
return ret;
}
#define atomic_exchange_explicit(object, desired, order) \
atomic_exchange(object, desired)
static inline int atomic_compare_exchange_strong(intptr_t *object, intptr_t *expected,
intptr_t desired)
{
int ret;
avpriv_atomic_lock();
if (*object == *expected) {
ret = 1;
*object = desired;
} else {
ret = 0;
*expected = *object;
}
avpriv_atomic_unlock();
return ret;
}
#define atomic_compare_exchange_strong_explicit(object, expected, desired, success, failure) \
atomic_compare_exchange_strong(object, expected, desired)
#define atomic_compare_exchange_weak(object, expected, desired) \
atomic_compare_exchange_strong(object, expected, desired)
#define atomic_compare_exchange_weak_explicit(object, expected, desired, success, failure) \
atomic_compare_exchange_weak(object, expected, desired)
#define FETCH_MODIFY(opname, op) \
static inline intptr_t atomic_fetch_ ## opname(intptr_t *object, intptr_t operand) \
{ \
intptr_t ret; \
avpriv_atomic_lock(); \
ret = *object; \
*object = *object op operand; \
avpriv_atomic_unlock(); \
return ret; \
}
FETCH_MODIFY(add, +)
FETCH_MODIFY(sub, -)
FETCH_MODIFY(or, |)
FETCH_MODIFY(xor, ^)
FETCH_MODIFY(and, &)
#undef FETCH_MODIFY
#define atomic_fetch_add_explicit(object, operand, order) \
atomic_fetch_add(object, operand)
#define atomic_fetch_sub_explicit(object, operand, order) \
atomic_fetch_sub(object, operand)
#define atomic_fetch_or_explicit(object, operand, order) \
atomic_fetch_or(object, operand)
#define atomic_fetch_xor_explicit(object, operand, order) \
atomic_fetch_xor(object, operand)
#define atomic_fetch_and_explicit(object, operand, order) \
atomic_fetch_and(object, operand)
#define atomic_flag_test_and_set(object) \
atomic_exchange(object, 1)
#define atomic_flag_test_and_set_explicit(object, order) \
atomic_flag_test_and_set(object)
#define atomic_flag_clear(object) \
atomic_store(object, 0)
#define atomic_flag_clear_explicit(object, order) \
atomic_flag_clear(object)
#endif /* COMPAT_ATOMICS_PTHREAD_STDATOMIC_H */

View File

@@ -1,186 +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_ATOMICS_SUNCC_STDATOMIC_H
#define COMPAT_ATOMICS_SUNCC_STDATOMIC_H
#include <atomic.h>
#include <mbarrier.h>
#include <stddef.h>
#include <stdint.h>
#define ATOMIC_FLAG_INIT 0
#define ATOMIC_VAR_INIT(value) (value)
#define atomic_init(obj, value) \
do { \
*(obj) = (value); \
} while(0)
#define kill_dependency(y) ((void)0)
#define atomic_thread_fence(order) \
__machine_rw_barrier();
#define atomic_signal_fence(order) \
((void)0)
#define atomic_is_lock_free(obj) 0
typedef intptr_t atomic_flag;
typedef intptr_t atomic_bool;
typedef intptr_t atomic_char;
typedef intptr_t atomic_schar;
typedef intptr_t atomic_uchar;
typedef intptr_t atomic_short;
typedef intptr_t atomic_ushort;
typedef intptr_t atomic_int;
typedef intptr_t atomic_uint;
typedef intptr_t atomic_long;
typedef intptr_t atomic_ulong;
typedef intptr_t atomic_llong;
typedef intptr_t atomic_ullong;
typedef intptr_t atomic_wchar_t;
typedef intptr_t atomic_int_least8_t;
typedef intptr_t atomic_uint_least8_t;
typedef intptr_t atomic_int_least16_t;
typedef intptr_t atomic_uint_least16_t;
typedef intptr_t atomic_int_least32_t;
typedef intptr_t atomic_uint_least32_t;
typedef intptr_t atomic_int_least64_t;
typedef intptr_t atomic_uint_least64_t;
typedef intptr_t atomic_int_fast8_t;
typedef intptr_t atomic_uint_fast8_t;
typedef intptr_t atomic_int_fast16_t;
typedef intptr_t atomic_uint_fast16_t;
typedef intptr_t atomic_int_fast32_t;
typedef intptr_t atomic_uint_fast32_t;
typedef intptr_t atomic_int_fast64_t;
typedef intptr_t atomic_uint_fast64_t;
typedef intptr_t atomic_intptr_t;
typedef intptr_t atomic_uintptr_t;
typedef intptr_t atomic_size_t;
typedef intptr_t atomic_ptrdiff_t;
typedef intptr_t atomic_intmax_t;
typedef intptr_t atomic_uintmax_t;
static inline void atomic_store(intptr_t *object, intptr_t desired)
{
*object = desired;
__machine_rw_barrier();
}
#define atomic_store_explicit(object, desired, order) \
atomic_store(object, desired)
static inline intptr_t atomic_load(intptr_t *object)
{
__machine_rw_barrier();
return *object;
}
#define atomic_load_explicit(object, order) \
atomic_load(object)
#define atomic_exchange(object, desired) \
atomic_swap_ptr(object, desired)
#define atomic_exchange_explicit(object, desired, order) \
atomic_exchange(object, desired)
static inline int atomic_compare_exchange_strong(intptr_t *object, intptr_t *expected,
intptr_t desired)
{
intptr_t old = *expected;
*expected = (intptr_t)atomic_cas_ptr(object, (void *)old, (void *)desired);
return *expected == old;
}
#define atomic_compare_exchange_strong_explicit(object, expected, desired, success, failure) \
atomic_compare_exchange_strong(object, expected, desired)
#define atomic_compare_exchange_weak(object, expected, desired) \
atomic_compare_exchange_strong(object, expected, desired)
#define atomic_compare_exchange_weak_explicit(object, expected, desired, success, failure) \
atomic_compare_exchange_weak(object, expected, desired)
static inline intptr_t atomic_fetch_add(intptr_t *object, intptr_t operand)
{
return atomic_add_ptr_nv(object, operand) - operand;
}
#define atomic_fetch_sub(object, operand) \
atomic_fetch_add(object, -(operand))
static inline intptr_t atomic_fetch_or(intptr_t *object, intptr_t operand)
{
intptr_t old;
do {
old = atomic_load(object);
} while (!atomic_compare_exchange_strong(object, old, old | operand));
return old;
}
static inline intptr_t atomic_fetch_xor(intptr_t *object, intptr_t operand)
{
intptr_t old;
do {
old = atomic_load(object);
} while (!atomic_compare_exchange_strong(object, old, old ^ operand));
return old;
}
static inline intptr_t atomic_fetch_and(intptr_t *object, intptr_t operand)
{
intptr_t old;
do {
old = atomic_load(object);
} while (!atomic_compare_exchange_strong(object, old, old & operand));
return old;
}
#define atomic_fetch_add_explicit(object, operand, order) \
atomic_fetch_add(object, operand)
#define atomic_fetch_sub_explicit(object, operand, order) \
atomic_fetch_sub(object, operand)
#define atomic_fetch_or_explicit(object, operand, order) \
atomic_fetch_or(object, operand)
#define atomic_fetch_xor_explicit(object, operand, order) \
atomic_fetch_xor(object, operand)
#define atomic_fetch_and_explicit(object, operand, order) \
atomic_fetch_and(object, operand)
#define atomic_flag_test_and_set(object) \
atomic_exchange(object, 1)
#define atomic_flag_test_and_set_explicit(object, order) \
atomic_flag_test_and_set(object)
#define atomic_flag_clear(object) \
atomic_store(object, 0)
#define atomic_flag_clear_explicit(object, order) \
atomic_flag_clear(object)
#endif /* COMPAT_ATOMICS_SUNCC_STDATOMIC_H */

View File

@@ -1,181 +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_ATOMICS_WIN32_STDATOMIC_H
#define COMPAT_ATOMICS_WIN32_STDATOMIC_H
#define WIN32_LEAN_AND_MEAN
#include <stddef.h>
#include <stdint.h>
#include <windows.h>
#define ATOMIC_FLAG_INIT 0
#define ATOMIC_VAR_INIT(value) (value)
#define atomic_init(obj, value) \
do { \
*(obj) = (value); \
} while(0)
#define kill_dependency(y) ((void)0)
#define atomic_thread_fence(order) \
MemoryBarrier();
#define atomic_signal_fence(order) \
((void)0)
#define atomic_is_lock_free(obj) 0
typedef intptr_t atomic_flag;
typedef intptr_t atomic_bool;
typedef intptr_t atomic_char;
typedef intptr_t atomic_schar;
typedef intptr_t atomic_uchar;
typedef intptr_t atomic_short;
typedef intptr_t atomic_ushort;
typedef intptr_t atomic_int;
typedef intptr_t atomic_uint;
typedef intptr_t atomic_long;
typedef intptr_t atomic_ulong;
typedef intptr_t atomic_llong;
typedef intptr_t atomic_ullong;
typedef intptr_t atomic_wchar_t;
typedef intptr_t atomic_int_least8_t;
typedef intptr_t atomic_uint_least8_t;
typedef intptr_t atomic_int_least16_t;
typedef intptr_t atomic_uint_least16_t;
typedef intptr_t atomic_int_least32_t;
typedef intptr_t atomic_uint_least32_t;
typedef intptr_t atomic_int_least64_t;
typedef intptr_t atomic_uint_least64_t;
typedef intptr_t atomic_int_fast8_t;
typedef intptr_t atomic_uint_fast8_t;
typedef intptr_t atomic_int_fast16_t;
typedef intptr_t atomic_uint_fast16_t;
typedef intptr_t atomic_int_fast32_t;
typedef intptr_t atomic_uint_fast32_t;
typedef intptr_t atomic_int_fast64_t;
typedef intptr_t atomic_uint_fast64_t;
typedef intptr_t atomic_intptr_t;
typedef intptr_t atomic_uintptr_t;
typedef intptr_t atomic_size_t;
typedef intptr_t atomic_ptrdiff_t;
typedef intptr_t atomic_intmax_t;
typedef intptr_t atomic_uintmax_t;
#define atomic_store(object, desired) \
do { \
*(object) = (desired); \
MemoryBarrier(); \
} while (0)
#define atomic_store_explicit(object, desired, order) \
atomic_store(object, desired)
#define atomic_load(object) \
(MemoryBarrier(), *(object))
#define atomic_load_explicit(object, order) \
atomic_load(object)
#define atomic_exchange(object, desired) \
InterlockedExchangePointer(object, desired);
#define atomic_exchange_explicit(object, desired, order) \
atomic_exchange(object, desired)
static inline int atomic_compare_exchange_strong(intptr_t *object, intptr_t *expected,
intptr_t desired)
{
intptr_t old = *expected;
*expected = (intptr_t)InterlockedCompareExchangePointer(
(PVOID *)object, (PVOID)desired, (PVOID)old);
return *expected == old;
}
#define atomic_compare_exchange_strong_explicit(object, expected, desired, success, failure) \
atomic_compare_exchange_strong(object, expected, desired)
#define atomic_compare_exchange_weak(object, expected, desired) \
atomic_compare_exchange_strong(object, expected, desired)
#define atomic_compare_exchange_weak_explicit(object, expected, desired, success, failure) \
atomic_compare_exchange_weak(object, expected, desired)
#ifdef _WIN64
#define atomic_fetch_add(object, operand) \
InterlockedExchangeAdd64(object, operand)
#define atomic_fetch_sub(object, operand) \
InterlockedExchangeAdd64(object, -(operand))
#define atomic_fetch_or(object, operand) \
InterlockedOr64(object, operand)
#define atomic_fetch_xor(object, operand) \
InterlockedXor64(object, operand)
#define atomic_fetch_and(object, operand) \
InterlockedAnd64(object, operand)
#else
#define atomic_fetch_add(object, operand) \
InterlockedExchangeAdd(object, operand)
#define atomic_fetch_sub(object, operand) \
InterlockedExchangeAdd(object, -(operand))
#define atomic_fetch_or(object, operand) \
InterlockedOr(object, operand)
#define atomic_fetch_xor(object, operand) \
InterlockedXor(object, operand)
#define atomic_fetch_and(object, operand) \
InterlockedAnd(object, operand)
#endif /* _WIN64 */
#define atomic_fetch_add_explicit(object, operand, order) \
atomic_fetch_add(object, operand)
#define atomic_fetch_sub_explicit(object, operand, order) \
atomic_fetch_sub(object, operand)
#define atomic_fetch_or_explicit(object, operand, order) \
atomic_fetch_or(object, operand)
#define atomic_fetch_xor_explicit(object, operand, order) \
atomic_fetch_xor(object, operand)
#define atomic_fetch_and_explicit(object, operand, order) \
atomic_fetch_and(object, operand)
#define atomic_flag_test_and_set(object) \
atomic_exchange(object, 1)
#define atomic_flag_test_and_set_explicit(object, order) \
atomic_flag_test_and_set(object)
#define atomic_flag_clear(object) \
atomic_store(object, 0)
#define atomic_flag_clear_explicit(object, order) \
atomic_flag_clear(object)
#endif /* COMPAT_ATOMICS_WIN32_STDATOMIC_H */

View File

@@ -75,149 +75,54 @@ 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 = 8,
AVS_CS_SHIFT_SAMPLE_BITS = 16,
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 {
@@ -342,10 +247,10 @@ 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 ); }
@@ -353,10 +258,6 @@ 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_API(int, avs_is_rgb64)(const AVS_VideoInfo * p);
AVSC_API(int, avs_is_yv24)(const AVS_VideoInfo * p);
AVSC_API(int, avs_is_yv16)(const AVS_VideoInfo * p);
@@ -367,38 +268,6 @@ 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_property(const AVS_VideoInfo * p, int property)
{ return ((p->image_type & property)==property ); }
@@ -496,12 +365,6 @@ AVSC_INLINE int avs_is_same_colorspace(AVS_VideoInfo * x, AVS_VideoInfo * 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);
/////////////////////////////////////////////////////////////////////
//
// AVS_VideoFrame
@@ -665,7 +528,7 @@ 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; }
/////////////////////////////////////////////////////////////////////
//
@@ -898,28 +761,11 @@ struct AVS_Library {
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);
@@ -934,11 +780,6 @@ struct AVS_Library {
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
@@ -999,28 +840,11 @@ AVSC_INLINE AVS_Library * avs_load_library() {
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);
@@ -1036,12 +860,6 @@ AVSC_INLINE AVS_Library * avs_load_library() {
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

@@ -1,33 +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 AV_COMPAT_CUDA_DYNLINK_LOADER_H
#define AV_COMPAT_CUDA_DYNLINK_LOADER_H
#include "libavutil/log.h"
#include "compat/w32dlfcn.h"
#define FFNV_LOAD_FUNC(path) dlopen((path), RTLD_LAZY)
#define FFNV_SYM_FUNC(lib, sym) dlsym((lib), (sym))
#define FFNV_FREE_FUNC(lib) dlclose(lib)
#define FFNV_LOG_FUNC(logctx, msg, ...) av_log(logctx, AV_LOG_ERROR, msg, __VA_ARGS__)
#define FFNV_DEBUG_LOG_FUNC(logctx, msg, ...) av_log(logctx, AV_LOG_DEBUG, msg, __VA_ARGS__)
#include <ffnvcodec/dynlink_loader.h>
#endif

View File

@@ -1,36 +0,0 @@
#!/bin/sh
# Copyright (c) 2017, NVIDIA CORPORATION. All rights reserved.
#
# 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.
set -e
OUT="$1"
IN="$2"
NAME="$(basename "$IN" | sed 's/\..*//')"
printf "const char %s_ptx[] = \\" "$NAME" > "$OUT"
while read LINE
do
printf "\n\t\"%s\\\n\"" "$(printf "%s" "$LINE" | sed -e 's/\r//g' -e 's/["\\]/\\&/g')" >> "$OUT"
done < "$IN"
printf ";\n" >> "$OUT"
exit 0

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

@@ -1,5 +1,5 @@
/*
* Copyright (c) 2011-2017 KO Myung-Hun <komh@chollian.net>
* Copyright (c) 2011 KO Myung-Hun <komh@chollian.net>
*
* This file is part of FFmpeg.
*
@@ -46,11 +46,9 @@ typedef struct {
typedef void pthread_attr_t;
typedef _fmutex pthread_mutex_t;
typedef HMTX pthread_mutex_t;
typedef void pthread_mutexattr_t;
#define PTHREAD_MUTEX_INITIALIZER _FMUTEX_INITIALIZER
typedef struct {
HEV event_sem;
HEV ack_sem;
@@ -100,28 +98,28 @@ static av_always_inline int pthread_join(pthread_t thread, void **value_ptr)
static av_always_inline int pthread_mutex_init(pthread_mutex_t *mutex,
const pthread_mutexattr_t *attr)
{
_fmutex_create(mutex, 0);
DosCreateMutexSem(NULL, (PHMTX)mutex, 0, FALSE);
return 0;
}
static av_always_inline int pthread_mutex_destroy(pthread_mutex_t *mutex)
{
_fmutex_close(mutex);
DosCloseMutexSem(*(PHMTX)mutex);
return 0;
}
static av_always_inline int pthread_mutex_lock(pthread_mutex_t *mutex)
{
_fmutex_request(mutex, 0);
DosRequestMutexSem(*(PHMTX)mutex, SEM_INDEFINITE_WAIT);
return 0;
}
static av_always_inline int pthread_mutex_unlock(pthread_mutex_t *mutex)
{
_fmutex_release(mutex);
DosReleaseMutexSem(*(PHMTX)mutex);
return 0;
}

10
compat/plan9/head Executable file
View File

@@ -0,0 +1,10 @@
#!/bin/sh
n=10
case "$1" in
-n) n=$2; shift 2 ;;
-n*) n=${1#-n}; shift ;;
esac
exec sed ${n}q "$@"

View File

@@ -1,7 +1,4 @@
/*
* NewTek NDI common code
* Copyright (c) 2017 Maksym Veremeyenko
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
@@ -19,12 +16,19 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVDEVICE_LIBNDI_NEWTEK_COMMON_H
#define AVDEVICE_LIBNDI_NEWTEK_COMMON_H
#include <Processing.NDI.Lib.h>
#define NDI_TIME_BASE 10000000
#define NDI_TIME_BASE_Q (AVRational){1, NDI_TIME_BASE}
int plan9_main(int argc, char **argv);
#undef main
int main(int argc, char **argv)
{
/* The setfcr() function in lib9 is broken, must use asm. */
#ifdef __i386
short fcr;
__asm__ volatile ("fstcw %0 \n"
"or $63, %0 \n"
"fldcw %0 \n"
: "=m"(fcr));
#endif
return plan9_main(argc, argv);
}

2
compat/plan9/printf Executable file
View File

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

View File

@@ -25,9 +25,9 @@
#include "libavutil/avstring.h"
#include "libavutil/mathematics.h"
static const char *check_nan_suffix(const char *s)
static char *check_nan_suffix(char *s)
{
const char *start = s;
char *start = s;
if (*s++ != '(')
return start;
@@ -44,7 +44,7 @@ double strtod(const char *, char **);
double avpriv_strtod(const char *nptr, char **endptr)
{
const char *end;
char *end;
double res;
/* Skip leading spaces */
@@ -81,13 +81,13 @@ double avpriv_strtod(const char *nptr, char **endptr)
!av_strncasecmp(nptr, "+0x", 3)) {
/* FIXME this doesn't handle exponents, non-integers (float/double)
* and numbers too large for long long */
res = strtoll(nptr, (char **)&end, 16);
res = strtoll(nptr, &end, 16);
} else {
res = strtod(nptr, (char **)&end);
res = strtod(nptr, &end);
}
if (endptr)
*endptr = (char *)end;
*endptr = end;
return res;
}

View File

@@ -1,6 +1,4 @@
/*
* Copyright (c) 2017 Google Inc.
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
@@ -18,12 +16,15 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_ARM_VP9DSP_INIT_H
#define AVCODEC_ARM_VP9DSP_INIT_H
#ifndef COMPAT_TMS470_MATH_H
#define COMPAT_TMS470_MATH_H
#include "libavcodec/vp9dsp.h"
#include_next <math.h>
void ff_vp9dsp_init_10bpp_arm(VP9DSPContext *dsp);
void ff_vp9dsp_init_12bpp_arm(VP9DSPContext *dsp);
#undef INFINITY
#undef NAN
#endif /* AVCODEC_ARM_VP9DSP_INIT_H */
#define INFINITY (*(const float*)((const unsigned []){ 0x7f800000 }))
#define NAN (*(const float*)((const unsigned []){ 0x7fc00000 }))
#endif /* COMPAT_TMS470_MATH_H */

View File

@@ -1,94 +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_W32DLFCN_H
#define COMPAT_W32DLFCN_H
#ifdef _WIN32
#include <windows.h>
#include "config.h"
#if (_WIN32_WINNT < 0x0602) || HAVE_WINRT
#include "libavutil/wchar_filename.h"
#endif
/**
* Safe function used to open dynamic libs. This attempts to improve program security
* by removing the current directory from the dll search path. Only dll's found in the
* executable or system directory are allowed to be loaded.
* @param name The dynamic lib name.
* @return A handle to the opened lib.
*/
static inline HMODULE win32_dlopen(const char *name)
{
#if _WIN32_WINNT < 0x0602
// Need to check if KB2533623 is available
if (!GetProcAddress(GetModuleHandleW(L"kernel32.dll"), "SetDefaultDllDirectories")) {
HMODULE module = NULL;
wchar_t *path = NULL, *name_w = NULL;
DWORD pathlen;
if (utf8towchar(name, &name_w))
goto exit;
path = (wchar_t *)av_mallocz_array(MAX_PATH, sizeof(wchar_t));
// Try local directory first
pathlen = GetModuleFileNameW(NULL, path, MAX_PATH);
pathlen = wcsrchr(path, '\\') - path;
if (pathlen == 0 || pathlen + wcslen(name_w) + 2 > MAX_PATH)
goto exit;
path[pathlen] = '\\';
wcscpy(path + pathlen + 1, name_w);
module = LoadLibraryExW(path, NULL, LOAD_WITH_ALTERED_SEARCH_PATH);
if (module == NULL) {
// Next try System32 directory
pathlen = GetSystemDirectoryW(path, MAX_PATH);
if (pathlen == 0 || pathlen + wcslen(name_w) + 2 > MAX_PATH)
goto exit;
path[pathlen] = '\\';
wcscpy(path + pathlen + 1, name_w);
module = LoadLibraryExW(path, NULL, LOAD_WITH_ALTERED_SEARCH_PATH);
}
exit:
av_free(path);
av_free(name_w);
return module;
}
#endif
#ifndef LOAD_LIBRARY_SEARCH_APPLICATION_DIR
# define LOAD_LIBRARY_SEARCH_APPLICATION_DIR 0x00000200
#endif
#ifndef LOAD_LIBRARY_SEARCH_SYSTEM32
# define LOAD_LIBRARY_SEARCH_SYSTEM32 0x00000800
#endif
#if HAVE_WINRT
wchar_t *name_w = NULL;
int ret;
if (utf8towchar(name, &name_w))
return NULL;
ret = LoadPackagedLibrary(name_w, 0);
av_free(name_w);
return ret;
#else
return LoadLibraryExA(name, NULL, LOAD_LIBRARY_SEARCH_APPLICATION_DIR | LOAD_LIBRARY_SEARCH_SYSTEM32);
#endif
}
#define dlopen(name, flags) win32_dlopen(name)
#define dlclose FreeLibrary
#define dlsym GetProcAddress
#else
#include <dlfcn.h>
#endif
#endif /* COMPAT_W32DLFCN_H */

View File

@@ -39,6 +39,11 @@
#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"
@@ -51,19 +56,28 @@ typedef struct pthread_t {
void *ret;
} pthread_t;
/* use light weight mutex/condition variable API for Windows Vista and later */
typedef SRWLOCK pthread_mutex_t;
/* the conditional variable api for windows 6.0+ uses critical sections and
* 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
typedef struct pthread_cond_t {
void *Ptr;
} pthread_cond_t;
#endif
#define PTHREAD_MUTEX_INITIALIZER SRWLOCK_INIT
#define PTHREAD_COND_INITIALIZER CONDITION_VARIABLE_INIT
#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)
{
pthread_t *h = (pthread_t*)arg;
pthread_t *h = arg;
h->ret = h->func(h->arg);
return 0;
}
@@ -100,25 +114,26 @@ static av_unused int pthread_join(pthread_t thread, void **value_ptr)
static inline int pthread_mutex_init(pthread_mutex_t *m, void* attr)
{
InitializeSRWLock(m);
InitializeCriticalSection(m);
return 0;
}
static inline int pthread_mutex_destroy(pthread_mutex_t *m)
{
/* Unlocked SWR locks use no resources */
DeleteCriticalSection(m);
return 0;
}
static inline int pthread_mutex_lock(pthread_mutex_t *m)
{
AcquireSRWLockExclusive(m);
EnterCriticalSection(m);
return 0;
}
static inline int pthread_mutex_unlock(pthread_mutex_t *m)
{
ReleaseSRWLockExclusive(m);
LeaveCriticalSection(m);
return 0;
}
#if _WIN32_WINNT >= 0x0600
typedef INIT_ONCE pthread_once_t;
#define PTHREAD_ONCE_INIT INIT_ONCE_STATIC_INIT
@@ -152,7 +167,7 @@ static inline int pthread_cond_broadcast(pthread_cond_t *cond)
static inline int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
{
SleepConditionVariableSRW(cond, mutex, INFINITE, 0);
SleepConditionVariableCS(cond, mutex, INFINITE);
return 0;
}
@@ -162,4 +177,242 @@ static inline int pthread_cond_signal(pthread_cond_t *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;
volatile int waiter_count;
HANDLE semaphore;
HANDLE waiters_done;
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)
{
win32_cond_t *win32_cond = NULL;
w32thread_once_fallback(&w32thread_init_state, w32thread_init);
if (cond_init) {
cond_init(cond);
return 0;
}
/* non native condition variables */
win32_cond = av_mallocz(sizeof(win32_cond_t));
if (!win32_cond)
return ENOMEM;
cond->Ptr = win32_cond;
win32_cond->semaphore = CreateSemaphore(NULL, 0, 0x7fffffff, NULL);
if (!win32_cond->semaphore)
return ENOMEM;
win32_cond->waiters_done = CreateEvent(NULL, TRUE, FALSE, NULL);
if (!win32_cond->waiters_done)
return ENOMEM;
pthread_mutex_init(&win32_cond->mtx_waiter_count, NULL);
pthread_mutex_init(&win32_cond->mtx_broadcast, NULL);
return 0;
}
static av_unused int pthread_cond_destroy(pthread_cond_t *cond)
{
win32_cond_t *win32_cond = cond->Ptr;
/* native condition variables do not destroy */
if (cond_init)
return 0;
/* non native condition variables */
CloseHandle(win32_cond->semaphore);
CloseHandle(win32_cond->waiters_done);
pthread_mutex_destroy(&win32_cond->mtx_waiter_count);
pthread_mutex_destroy(&win32_cond->mtx_broadcast);
av_freep(&win32_cond);
cond->Ptr = NULL;
return 0;
}
static av_unused int pthread_cond_broadcast(pthread_cond_t *cond)
{
win32_cond_t *win32_cond = cond->Ptr;
int have_waiter;
if (cond_broadcast) {
cond_broadcast(cond);
return 0;
}
/* non native condition variables */
pthread_mutex_lock(&win32_cond->mtx_broadcast);
pthread_mutex_lock(&win32_cond->mtx_waiter_count);
have_waiter = 0;
if (win32_cond->waiter_count) {
win32_cond->is_broadcast = 1;
have_waiter = 1;
}
if (have_waiter) {
ReleaseSemaphore(win32_cond->semaphore, win32_cond->waiter_count, NULL);
pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
WaitForSingleObject(win32_cond->waiters_done, INFINITE);
ResetEvent(win32_cond->waiters_done);
win32_cond->is_broadcast = 0;
} else
pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
pthread_mutex_unlock(&win32_cond->mtx_broadcast);
return 0;
}
static av_unused int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
{
win32_cond_t *win32_cond = cond->Ptr;
int last_waiter;
if (cond_wait) {
cond_wait(cond, mutex, INFINITE);
return 0;
}
/* non native condition variables */
pthread_mutex_lock(&win32_cond->mtx_broadcast);
pthread_mutex_lock(&win32_cond->mtx_waiter_count);
win32_cond->waiter_count++;
pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
pthread_mutex_unlock(&win32_cond->mtx_broadcast);
// unlock the external mutex
pthread_mutex_unlock(mutex);
WaitForSingleObject(win32_cond->semaphore, INFINITE);
pthread_mutex_lock(&win32_cond->mtx_waiter_count);
win32_cond->waiter_count--;
last_waiter = !win32_cond->waiter_count || !win32_cond->is_broadcast;
pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
if (last_waiter)
SetEvent(win32_cond->waiters_done);
// lock the external mutex
return pthread_mutex_lock(mutex);
}
static av_unused int pthread_cond_signal(pthread_cond_t *cond)
{
win32_cond_t *win32_cond = cond->Ptr;
int have_waiter;
if (cond_signal) {
cond_signal(cond);
return 0;
}
pthread_mutex_lock(&win32_cond->mtx_broadcast);
/* non-native condition variables */
pthread_mutex_lock(&win32_cond->mtx_waiter_count);
have_waiter = win32_cond->waiter_count;
pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
if (have_waiter) {
ReleaseSemaphore(win32_cond->semaphore, 1, NULL);
WaitForSingleObject(win32_cond->waiters_done, INFINITE);
ResetEvent(win32_cond->waiters_done);
}
pthread_mutex_unlock(&win32_cond->mtx_broadcast);
return 0;
}
#endif
static av_unused void w32thread_init(void)
{
#if _WIN32_WINNT < 0x0600
HANDLE kernel_dll = GetModuleHandle(TEXT("kernel32.dll"));
/* if one is available, then they should all be available */
cond_init =
(void*)GetProcAddress(kernel_dll, "InitializeConditionVariable");
cond_broadcast =
(void*)GetProcAddress(kernel_dll, "WakeAllConditionVariable");
cond_signal =
(void*)GetProcAddress(kernel_dll, "WakeConditionVariable");
cond_wait =
(void*)GetProcAddress(kernel_dll, "SleepConditionVariableCS");
initonce_begin =
(void*)GetProcAddress(kernel_dll, "InitOnceBeginInitialize");
initonce_complete =
(void*)GetProcAddress(kernel_dll, "InitOnceComplete");
#endif
}
#endif /* COMPAT_W32PTHREADS_H */

View File

@@ -45,11 +45,7 @@ libname=$(mktemp -u "library").lib
trap 'rm -f -- $libname' EXIT
if [ -n "$AR" ]; then
$AR rcs ${libname} $@ >/dev/null
else
lib -out:${libname} $@ >/dev/null
fi
lib -out:${libname} $@ >/dev/null
if [ $? != 0 ]; then
echo "Could not create temporary library." >&2
exit 1
@@ -58,7 +54,23 @@ fi
IFS='
'
prefix="$EXTERN_PREFIX"
# Determine if we're building for x86 or x86_64 and
# set the symbol prefix accordingly.
prefix=""
arch=$(dumpbin -headers ${libname} |
tr '\t' ' ' |
grep '^ \+.\+machine \+(.\+)' |
head -1 |
sed -e 's/^ \{1,\}.\{1,\} \{1,\}machine \{1,\}(\(...\)).*/\1/')
if [ "${arch}" = "x86" ]; then
prefix="_"
else
if [ "${arch}" != "ARM" ] && [ "${arch}" != "x64" ]; then
echo "Unknown machine type." >&2
exit 1
fi
fi
started=0
regex="none"
@@ -100,19 +112,7 @@ for line in $(cat ${vscript} | tr '\t' ' '); do
'
done
if [ -n "$NM" ]; then
# Use eval, since NM="nm -g"
dump=$(eval "$NM --defined-only -g ${libname}" |
grep -v : |
grep -v ^$ |
cut -d' ' -f3 |
sed -e "s/^${prefix}//")
else
dump=$(dumpbin -linkermember:1 ${libname} |
sed -e '/public symbols/,$!d' -e '/^ \{1,\}Summary/,$d' -e "s/ \{1,\}${prefix}/ /" -e 's/ \{1,\}/ /g' |
tail -n +2 |
cut -d' ' -f3)
fi
dump=$(dumpbin -linkermember:1 ${libname})
rm ${libname}
@@ -121,6 +121,9 @@ list=""
for exp in ${regex}; do
list="${list}"'
'$(echo "${dump}" |
sed -e '/public symbols/,$!d' -e '/^ \{1,\}Summary/,$d' -e "s/ \{1,\}${prefix}/ /" -e 's/ \{1,\}/ /g' |
tail -n +2 |
cut -d' ' -f3 |
grep "^${exp}" |
sed -e 's/^/ /')
done

4418
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

@@ -2,610 +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: 2017-10-21
libavdevice: 2017-10-21
libavfilter: 2017-10-21
libavformat: 2017-10-21
libavresample: 2017-10-21
libpostproc: 2017-10-21
libswresample: 2017-10-21
libswscale: 2017-10-21
libavutil: 2017-10-21
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
API changes, most recent first:
-------- 8< --------- FFmpeg 4.1 was cut here -------- 8< ---------
2018-10-27 - 718044dc19 - lavu 56.21.100 - pixdesc.h
Add av_read_image_line2(), av_write_image_line2()
2018-10-24 - f9d4126f28 - lavu 56.20.100 - frame.h
Add AV_FRAME_DATA_S12M_TIMECODE
2018-10-11 - f6d48b618a - lavc 58.33.100 - mediacodec.h
Add av_mediacodec_render_buffer_at_time().
2018-09-09 - 35498c124a - lavc 58.29.100 - avcodec.h
Add AV_PKT_DATA_AFD
2018-08-16 - b33f5299a5 - lavc 58.23.100 - avcodec.h
Add av_bsf_flush().
2018-05-18 - 2b2f2f65f3 - lavf 58.15.100 - avformat.h
Add pmt_version field to AVProgram
2018-05-17 - 5dfeb7f081 - lavf 58.14.100 - avformat.h
Add AV_DISPOSITION_STILL_IMAGE
2018-05-10 - c855683427 - lavu 56.18.101 - hwcontext_cuda.h
Add AVCUDADeviceContext.stream.
2018-04-30 - 56b081da57 - lavu 56.18.100 - pixdesc.h
Add AV_PIX_FMT_FLAG_ALPHA to AV_PIX_FMT_PAL8.
2018-04-26 - 5be0410cb3 - lavu 56.17.100 - opt.h
Add AV_OPT_FLAG_DEPRECATED.
2018-04-26 - 71fa82bed6 - lavu 56.16.100 - threadmessage.h
Add av_thread_message_queue_nb_elems().
-------- 8< --------- FFmpeg 4.0 was cut here -------- 8< ---------
2018-04-03 - d6fc031caf - lavu 56.13.100 - pixdesc.h
Deprecate AV_PIX_FMT_FLAG_PSEUDOPAL and make allocating a pseudo palette
optional for API users (see AV_PIX_FMT_FLAG_PSEUDOPAL doxygen for details).
2018-04-01 - 860086ee16 - lavc 58.17.100 - avcodec.h
Add av_packet_make_refcounted().
2018-04-01 - f1805d160d - lavfi 7.14.100 - avfilter.h
Deprecate use of avfilter_register(), avfilter_register_all(),
avfilter_next(). Add av_filter_iterate().
2018-03-25 - b7d0d912ef - lavc 58.16.100 - avcodec.h
Add FF_SUB_CHARENC_MODE_IGNORE.
2018-03-23 - db2a7c947e - lavu 56.12.100 - encryption_info.h
Add AVEncryptionInitInfo and AVEncryptionInfo structures to hold new side-data
for encryption info.
2018-03-21 - f14ca60001 - lavc 58.15.100 - avcodec.h
Add av_packet_make_writable().
2018-03-18 - 4b86ac27a0 - lavu 56.11.100 - frame.h
Add AV_FRAME_DATA_QP_TABLE_PROPERTIES and AV_FRAME_DATA_QP_TABLE_DATA.
2018-03-15 - e0e72539cf - lavu 56.10.100 - opt.h
Add AV_OPT_FLAG_BSF_PARAM
2018-03-07 - 950170bd3b - lavu 56.9.100 - crc.h
Add AV_CRC_8_EBU crc variant.
2018-03-07 - 2a0eb86857 - lavc 58.14.100 - mediacodec.h
Change the default behavior of avcodec_flush() on mediacodec
video decoders. To restore the previous behavior, use the new
delay_flush=1 option.
2018-03-01 - 6731f60598 - lavu 56.8.100 - frame.h
Add av_frame_new_side_data_from_buf().
2018-02-15 - 8a8d0b319a
Change av_ripemd_update(), av_murmur3_update() and av_hash_update() length
parameter type to size_t at next major bump.
2018-02-12 - bcab11a1a2 - lavfi 7.12.100 - avfilter.h
Add AVFilterContext.extra_hw_frames.
2018-02-12 - d23fff0d8a - lavc 58.11.100 - avcodec.h
Add AVCodecContext.extra_hw_frames.
2018-02-06 - 0694d87024 - lavf 58.9.100 - avformat.h
Deprecate use of av_register_input_format(), av_register_output_format(),
av_register_all(), av_iformat_next(), av_oformat_next().
Add av_demuxer_iterate(), and av_muxer_iterate().
2018-02-06 - 36c85d6e77 - lavc 58.10.100 - avcodec.h
Deprecate use of avcodec_register(), avcodec_register_all(),
av_codec_next(), av_register_codec_parser(), and av_parser_next().
Add av_codec_iterate() and av_parser_iterate().
2018-02-04 - ff46124b0d - lavf 58.8.100 - avformat.h
Deprecate the current names of the RTSP "timeout", "stimeout", "user-agent"
options. Introduce "listen_timeout" as replacement for the current "timeout"
option, and "user_agent" as replacement for "user-agent". Once the deprecation
is over, the old "timeout" option will be removed, and "stimeout" will be
renamed to "stimeout" (the "timeout" option will essentially change semantics).
2018-01-28 - ea3672b7d6 - lavf 58.7.100 - avformat.h
Deprecate AVFormatContext filename field which had limited length, use the
new dynamically allocated url field instead.
2018-01-28 - ea3672b7d6 - lavf 58.7.100 - avformat.h
Add url field to AVFormatContext and add ff_format_set_url helper function.
2018-01-27 - 6194d7e564 - lavf 58.6.100 - avformat.h
Add AVFMTCTX_UNSEEKABLE (for HLS demuxer).
2018-01-23 - 9f07cf7c00 - lavu 56.9.100 - aes_ctr.h
Add method to set the 16-byte IV.
2018-01-16 - 631c56a8e4 - lavf 58.5.100 - avformat.h
Explicitly make avformat_network_init() and avformat_network_deinit() optional.
If these are not called, network initialization and deinitialization is
automatic, and unlike in older versions, fully supported, unless libavformat
is linked to ancient GnuTLS and OpenSSL.
2018-01-16 - 6512ff72f9 - lavf 58.4.100 - avformat.h
Deprecate AVStream.recommended_encoder_configuration. It was useful only for
FFserver, which has been removed.
2018-01-05 - 798dcf2432 - lavfi 7.11.101 - avfilter.h
Deprecate avfilter_link_get_channels(). Use av_buffersink_get_channels().
2017-01-04 - c29038f304 - lavr 4.0.0 - avresample.h
Deprecate the entire library. Merged years ago to provide compatibility
with Libav, it remained unmaintained by the FFmpeg project and duplicated
functionality provided by libswresample.
In order to improve consistency and reduce attack surface, it has been deprecated.
Users of this library are asked to migrate to libswresample, which, as well as
providing more functionality, is faster and has higher accuracy.
2017-12-26 - a04c2c707d - lavc 58.9.100 - avcodec.h
Deprecate av_lockmgr_register(). You need to build FFmpeg with threading
support enabled to get basic thread-safety (which is the default build
configuration).
2017-12-24 - 8b81eabe57 - lavu 56.7.100 - cpu.h
AVX-512 flags added.
2017-12-16 - 8bf4e6d3ce - lavc 58.8.100 - avcodec.h
The MediaCodec decoders now support AVCodecContext.hw_device_ctx.
2017-12-16 - e4d9f05ca7 - lavu 56.6.100 - hwcontext.h hwcontext_mediacodec.h
Add AV_HWDEVICE_TYPE_MEDIACODEC and a new installed header with
MediaCodec-specific hwcontext definitions.
2017-12-14 - b945fed629 - lavc 58.7.100 - avcodec.h
Add AV_CODEC_CAP_HARDWARE, AV_CODEC_CAP_HYBRID, and AVCodec.wrapper_name,
and mark all AVCodecs accordingly.
2017-11-29 - d268094f88 - lavu 56.4.100 / 56.7.0 - stereo3d.h
Add view field to AVStereo3D structure and AVStereo3DView enum.
2017-11-26 - 3a71bcc213 - lavc 58.6.100 - avcodec.h
Add const to AVCodecContext.hwaccel.
2017-11-26 - 3536a3efb9 - lavc 58.5.100 - avcodec.h
Deprecate user visibility of the AVHWAccel structure and the functions
av_register_hwaccel() and av_hwaccel_next().
2017-11-26 - 24cc0a53e9 - lavc 58.4.100 - avcodec.h
Add AVCodecHWConfig and avcodec_get_hw_config().
2017-11-22 - 3650cb2dfa - lavu 56.3.100 - opencl.h
Remove experimental OpenCL API (av_opencl_*).
2017-11-22 - b25d8ef0a7 - lavu 56.2.100 - hwcontext.h hwcontext_opencl.h
Add AV_HWDEVICE_TYPE_OPENCL and a new installed header with
OpenCL-specific hwcontext definitions.
2017-11-22 - a050f56c09 - lavu 56.1.100 - pixfmt.h
Add AV_PIX_FMT_OPENCL.
2017-11-11 - 48e4eda11d - lavc 58.3.100 - avcodec.h
Add avcodec_get_hw_frames_parameters().
-------- 8< --------- FFmpeg 3.4 was cut here -------- 8< ---------
2017-09-28 - b6cf66ae1c - lavc 57.106.104 - avcodec.h
Add AV_PKT_DATA_A53_CC packet side data, to export closed captions
2017-09-27 - 7aa6b8a68f - lavu 55.77.101 / lavu 55.31.1 - frame.h
Allow passing the value of 0 (meaning "automatic") as the required alignment
to av_frame_get_buffer().
2017-09-27 - 522f877086 - lavu 55.77.100 / lavu 55.31.0 - cpu.h
Add av_cpu_max_align() for querying maximum required data alignment.
2017-09-26 - b1cf151c4d - lavc 57.106.102 - avcodec.h
Deprecate AVCodecContext.refcounted_frames. This was useful for deprecated
API only (avcodec_decode_video2/avcodec_decode_audio4). The new decode APIs
(avcodec_send_packet/avcodec_receive_frame) always work with reference
counted frames.
2017-09-21 - 6f15f1cdc8 - lavu 55.76.100 / 56.6.0 - pixdesc.h
Add av_color_range_from_name(), av_color_primaries_from_name(),
av_color_transfer_from_name(), av_color_space_from_name(), and
av_chroma_location_from_name().
2017-09-13 - 82342cead1 - lavc 57.106.100 - avcodec.h
Add AV_PKT_FLAG_TRUSTED.
2017-09-13 - 9cb23cd9fe - lavu 55.75.100 - hwcontext.h hwcontext_drm.h
Add AV_HWDEVICE_TYPE_DRM and implementation.
2017-09-08 - 5ba2aef6ec - lavfi 6.103.100 - buffersrc.h
Add av_buffersrc_close().
2017-09-04 - 6cadbb16e9 - lavc 57.105.100 - avcodec.h
Add AV_HWACCEL_CODEC_CAP_EXPERIMENTAL, replacing the deprecated
HWACCEL_CODEC_CAP_EXPERIMENTAL flag.
2017-09-01 - 5d76674756 - lavf 57.81.100 - avio.h
Add avio_read_partial().
2017-09-01 - xxxxxxx - lavf 57.80.100 / 57.11.0 - avio.h
Add avio_context_free(). From now on it must be used for freeing AVIOContext.
2017-08-08 - 1460408703 - lavu 55.74.100 - pixdesc.h
Add AV_PIX_FMT_FLAG_FLOAT pixel format flag.
2017-08-08 - 463b81de2b - lavu 55.72.100 - imgutils.h
Add av_image_fill_black().
2017-08-08 - caa12027ba - lavu 55.71.100 - frame.h
Add av_frame_apply_cropping().
2017-07-25 - 24de4fddca - lavu 55.69.100 - frame.h
Add AV_FRAME_DATA_ICC_PROFILE side data type.
2017-06-27 - 70143a3954 - lavc 57.100.100 - avcodec.h
DXVA2 and D3D11 hardware accelerated decoding now supports the new hwaccel API,
which can create the decoder context and allocate hardware frame automatically.
See AVCodecContext.hw_device_ctx and AVCodecContext.hw_frames_ctx. For D3D11,
the new AV_PIX_FMT_D3D11 pixfmt must be used with the new API.
2017-06-27 - 3303511f33 - lavu 56.67.100 - hwcontext.h
Add AV_HWDEVICE_TYPE_D3D11VA and AV_PIX_FMT_D3D11.
2017-06-24 - 09891c5391 - lavf 57.75.100 - avio.h
Add AVIO_DATA_MARKER_FLUSH_POINT to signal preferred flush points to aviobuf.
2017-06-14 - d59c6a3aeb - lavu 55.66.100 - hwcontext.h
av_hwframe_ctx_create_derived() now takes some AV_HWFRAME_MAP_* combination
as its flags argument (which was previously unused).
2017-06-14 - 49ae8a5e87 - lavc 57.99.100 - avcodec.h
Add AV_HWACCEL_FLAG_ALLOW_PROFILE_MISMATCH.
2017-06-14 - 0b1794a43e - lavu 55.65.100 - hwcontext.h
Add AV_HWDEVICE_TYPE_NONE, av_hwdevice_find_type_by_name(),
av_hwdevice_get_type_name() and av_hwdevice_iterate_types().
2017-06-14 - b22172f6f3 - lavu 55.64.100 - hwcontext.h
Add av_hwdevice_ctx_create_derived().
2017-05-15 - 532b23f079 - lavc 57.96.100 - avcodec.h
VideoToolbox hardware-accelerated decoding now supports the new hwaccel API,
which can create the decoder context and allocate hardware frames automatically.
See AVCodecContext.hw_device_ctx and AVCodecContext.hw_frames_ctx.
2017-05-15 - 532b23f079 - lavu 57.63.100 - hwcontext.h
Add AV_HWDEVICE_TYPE_VIDEOTOOLBOX and implementation.
2017-05-08 - f089e02fa2 - lavc 57.95.100 / 57.31.0 - avcodec.h
Add AVCodecContext.apply_cropping to control whether cropping
is handled by libavcodec or the caller.
2017-05-08 - a47bd5d77e - lavu 55.62.100 / 55.30.0 - frame.h
Add AVFrame.crop_left/right/top/bottom fields for attaching cropping
information to video frames.
2017-xx-xx - xxxxxxxxxx
Change av_sha_update(), av_sha512_update() and av_md5_sum()/av_md5_update() length
parameter type to size_t at next major bump.
2017-05-05 - c0f17a905f - lavc 57.94.100 - avcodec.h
The cuvid decoders now support AVCodecContext.hw_device_ctx, which removes
the requirement to set an incomplete AVCodecContext.hw_frames_ctx only to
set the Cuda device handle.
2017-04-11 - 8378466507 - lavu 55.61.100 - avstring.h
Add av_strireplace().
2016-04-06 - 157e57a181 - lavc 57.92.100 - avcodec.h
Add AV_PKT_DATA_CONTENT_LIGHT_LEVEL packet side data.
2016-04-06 - b378f5bd64 - lavu 55.60.100 - mastering_display_metadata.h
Add AV_FRAME_DATA_CONTENT_LIGHT_LEVEL value, av_content_light_metadata_alloc()
and av_content_light_metadata_create_side_data() API, and AVContentLightMetadata
type to export content light level video properties.
2017-03-31 - 9033e8723c - lavu 55.57.100 - spherical.h
Add av_spherical_projection_name().
Add av_spherical_from_name().
2017-03-30 - 4cda23f1f1 - lavu 55.53.100 / 55.27.0 - hwcontext.h
Add av_hwframe_map() and associated AV_HWFRAME_MAP_* flags.
Add av_hwframe_ctx_create_derived().
2017-03-29 - bfdcdd6d82 - lavu 55.52.100 - avutil.h
add av_fourcc_make_string() function and av_fourcc2str() macro to replace
av_get_codec_tag_string() from lavc.
2017-03-27 - ddef3d902f - lavf 57.68.100 - avformat.h
Deprecate that demuxers export the stream rotation angle in AVStream.metadata
(via an entry named "rotate"). Use av_stream_get_side_data() with
AV_PKT_DATA_DISPLAYMATRIX instead, and read the rotation angle with
av_display_rotation_get(). The same is done for muxing. Instead of adding a
"rotate" entry to AVStream.metadata, AV_PKT_DATA_DISPLAYMATRIX side data has
to be added to the AVStream.
2017-03-23 - 7e4ba776a2 - lavc 57.85.101 - avcodec.h
vdpau hardware accelerated decoding now supports the new hwaccel API, which
can create the decoder context and allocate hardware frame automatically.
See AVCodecContext.hw_device_ctx and AVCodecContext.hw_frames_ctx.
2017-03-23 - 156bd8278f - lavc 57.85.100 - avcodec.h
Add AVCodecContext.hwaccel_flags field. This will control some hwaccels at
a later point.
2017-03-21 - fc9f14c7de - lavf 57.67.100 / 57.08.0 - avio.h
Add AVIO_SEEKABLE_TIME flag.
2017-03-21 - d682ae70b4 - lavf 57.66.105, lavc 57.83.101 - avformat.h, avcodec.h
Deprecate AVFMT_FLAG_KEEP_SIDE_DATA. It will be ignored after the next major
bump, and libavformat will behave as if it were always set.
Deprecate av_packet_merge_side_data() and av_packet_split_side_data().
2016-03-20 - 8200b16a9c - lavu 55.50.100 / 55.21.0 - imgutils.h
Add av_image_copy_uc_from(), a version of av_image_copy() for copying
from GPU mapped memory.
2017-03-20 - 9c2436e - lavu 55.49.100 - pixdesc.h
Add AV_PIX_FMT_FLAG_BAYER pixel format flag.
2017-03-18 - 3796fb2692 - lavfi 6.77.100 - avfilter.h
Deprecate AVFilterGraph.resample_lavr_opts
It's never been used by avfilter nor passed to anything.
2017-02-10 - 1b7ffddb3a - lavu 55.48.100 / 55.33.0 - spherical.h
Add AV_SPHERICAL_EQUIRECTANGULAR_TILE, av_spherical_tile_bounds(),
and projection-specific properties (bound_left, bound_top, bound_right,
bound_bottom, padding) to AVSphericalMapping.
2017-03-02 - ade7c1a232 - lavc 57.81.104 - videotoolbox.h
AVVideotoolboxContext.cv_pix_fmt_type can now be set to 0 to output the
native decoder format. (The default value is not changed.)
2017-03-02 - 554bc4eea8 - lavu 55.47.101, lavc 57.81.102, lavf 57.66.103
Remove requirement to use AVOption or accessors to access certain fields
in AVFrame, AVCodecContext, and AVFormatContext that were previously
documented as "no direct access" allowed.
2017-02-13 - c1a5fca06f - lavc 57.80.100 - avcodec.h
Add AVCodecContext.hw_device_ctx.
2017-02-11 - e3af49b14b - lavu 55.47.100 - frame.h
Add AVFrame.opaque_ref.
2017-01-31 - 2eab48177d - lavu 55.46.100 / 55.20.0 - cpu.h
Add AV_CPU_FLAG_SSSE3SLOW.
2017-01-24 - c4618f842a - lavu 55.45.100 - channel_layout.h
Add av_get_extended_channel_layout()
2017-01-22 - 76c5a69e26 - lavu 55.44.100 - lfg.h
Add av_lfg_init_from_data().
2017-01-17 - 2a4a8653b6 - lavc 57.74.100 - vaapi.h
Deprecate struct vaapi_context and the vaapi.h installed header.
Callers should set AVCodecContext.hw_frames_ctx instead.
2017-01-12 - dbe9dbed31 - lavfi 6.69.100 - buffersink.h
Add av_buffersink_get_*() functions.
2017-01-06 - 9488032e10 - lavf 57.62.100 - avio.h
Add avio_get_dyn_buf()
2016-12-10 - f542b152aa - lavu 55.43.100 - imgutils.h
Add av_image_check_size2()
2016-12-07 - e7a6f8c972 - lavc 57.67.100 / 57.29.0 - avcodec.h
Add AV_PKT_DATA_SPHERICAL packet side data to export AVSphericalMapping
information from containers.
2016-12-07 - 8f58ecc344 - lavu 55.42.100 / 55.30.0 - spherical.h
Add AV_FRAME_DATA_SPHERICAL value, av_spherical_alloc() API and
AVSphericalMapping type to export and describe spherical video properties.
2016-11-18 - 2ab50647ff - lavf 57.58.100 - avformat.h
Add av_stream_add_side_data().
2016-11-13 - 775a8477b7 - lavu 55.39.100 - hwcontext_vaapi.h
Add AV_VAAPI_DRIVER_QUIRK_ATTRIB_MEMTYPE.
2016-11-13 - a8d51bb424 - lavu 55.38.100 - hwcontext_vaapi.h
Add driver quirks field to VAAPI-specific hwdevice and enum with
members AV_VAAPI_DRIVER_QUIRK_* to represent its values.
2016-11-10 - 638b216d4f - lavu 55.36.100 - pixfmt.h
Add AV_PIX_FMT_GRAY12(LE/BE).
-------- 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
@@ -924,7 +333,7 @@ API changes, most recent first:
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 - afbd4b7e09 - lavf 56.01.0 - avformat.h
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.
@@ -941,7 +350,7 @@ API changes, most recent first:
2014-08-08 - 5c3c671 - lavf 55.53.100 - avio.h
Add avio_feof() and deprecate url_feof().
2014-08-07 - bb789016d4 - lsws 2.1.3 - swscale.h
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
@@ -1675,7 +1084,7 @@ 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.

File diff suppressed because it is too large Load Diff

View File

@@ -24,7 +24,6 @@ HTMLPAGES = $(AVPROGS-yes:%=doc/%.html) $(AVPROGS-yes:%=doc/%-all.html) $(COMP
doc/fate.html \
doc/general.html \
doc/git-howto.html \
doc/mailing-list-faq.html \
doc/nut.html \
doc/platform.html \
@@ -37,6 +36,30 @@ DOCS-$(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_AVCODEC_EXAMPLE) += avcodec
DOC_EXAMPLES-$(CONFIG_DECODING_ENCODING_EXAMPLE) += decoding_encoding
DOC_EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding
DOC_EXAMPLES-$(CONFIG_EXTRACT_MVS_EXAMPLE) += extract_mvs
DOC_EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE) += filter_audio
DOC_EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE) += filtering_audio
DOC_EXAMPLES-$(CONFIG_FILTERING_VIDEO_EXAMPLE) += filtering_video
DOC_EXAMPLES-$(CONFIG_METADATA_EXAMPLE) += metadata
DOC_EXAMPLES-$(CONFIG_MUXING_EXAMPLE) += muxing
DOC_EXAMPLES-$(CONFIG_QSVDEC_EXAMPLE) += qsvdec
DOC_EXAMPLES-$(CONFIG_REMUXING_EXAMPLE) += remuxing
DOC_EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE) += resampling_audio
DOC_EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE) += scaling_video
DOC_EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE) += transcode_aac
DOC_EXAMPLES-$(CONFIG_TRANSCODING_EXAMPLE) += transcoding
ALL_DOC_EXAMPLES_LIST = $(DOC_EXAMPLES-) $(DOC_EXAMPLES-yes)
DOC_EXAMPLES := $(DOC_EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)$(EXESUF))
ALL_DOC_EXAMPLES := $(ALL_DOC_EXAMPLES_LIST:%=doc/examples/%$(PROGSSUF)$(EXESUF))
ALL_DOC_EXAMPLES_G := $(ALL_DOC_EXAMPLES_LIST:%=doc/examples/%$(PROGSSUF)_g$(EXESUF))
PROGS += $(DOC_EXAMPLES)
all-$(CONFIG_DOC): doc
doc: documentation
@@ -44,6 +67,8 @@ doc: documentation
apidoc: doc/doxy/html
documentation: $(DOCS)
examples: $(DOC_EXAMPLES)
TEXIDEP = perl $(SRC_PATH)/doc/texidep.pl $(SRC_PATH) $< $@ >$(@:%=%.d)
doc/%.txt: TAG = TXT
@@ -96,9 +121,11 @@ doc/%.3: doc/%.pod $(GENTEXI)
$(M)pod2man --section=3 --center=" " --release=" " --date=" " $< > $@
$(DOCS) doc/doxy/html: | doc/
$(DOC_EXAMPLES:%$(EXESUF)=%.o): | doc/examples
OBJDIRS += doc/examples
DOXY_INPUT = $(INSTHEADERS)
DOXY_INPUT_DEPS = $(addprefix $(SRC_PATH)/, $(DOXY_INPUT)) ffbuild/config.mak
DOXY_INPUT = $(INSTHEADERS) $(DOC_EXAMPLES:%$(EXESUF)=%.c) $(LIB_EXAMPLES:%$(EXESUF)=%.c)
DOXY_INPUT_DEPS = $(addprefix $(SRC_PATH)/, $(DOXY_INPUT))
doc/doxy/html: TAG = DOXY
doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(SRC_PATH)/doc/doxy-wrapper.sh $(DOXY_INPUT_DEPS)
@@ -144,7 +171,11 @@ clean:: docclean
distclean:: docclean
$(RM) doc/config.texi
docclean::
examplesclean:
$(RM) $(ALL_DOC_EXAMPLES) $(ALL_DOC_EXAMPLES_G)
$(RM) $(CLEANSUFFIXES:%=doc/examples/%)
docclean: examplesclean
$(RM) $(CLEANSUFFIXES:%=doc/%)
$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 doc/*.3 doc/avoptions_*.texi
$(RM) -r doc/doxy/html

View File

@@ -18,7 +18,7 @@ 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
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,
@@ -26,98 +26,38 @@ with their parameters, if any.
@section aac_adtstoasc
Convert MPEG-2/4 AAC ADTS to an MPEG-4 Audio Specific Configuration
bitstream.
Convert MPEG-2/4 AAC ADTS to MPEG-4 Audio Specific Configuration
bitstream filter.
This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4
ADTS header and removes the ADTS header.
This filter is required for example when copying an AAC stream from a
raw ADTS AAC or an MPEG-TS container to MP4A-LATM, to an FLV file, or
to MOV/MP4 files and related formats such as 3GP or M4A. Please note
that it is auto-inserted for MP4A-LATM and MOV/MP4 and related formats.
@section av1_metadata
Modify metadata embedded in an AV1 stream.
@table @option
@item td
Insert or remove temporal delimiter OBUs in all temporal units of the
stream.
@table @samp
@item insert
Insert a TD at the beginning of every TU which does not already have one.
@item remove
Remove the TD from the beginning of every TU which has one.
@end table
@item color_primaries
@item transfer_characteristics
@item matrix_coefficients
Set the color description fields in the stream (see AV1 section 6.4.2).
@item color_range
Set the color range in the stream (see AV1 section 6.4.2; note that
this cannot be set for streams using BT.709 primaries, sRGB transfer
characteristic and identity (RGB) matrix coefficients).
@table @samp
@item tv
Limited range.
@item pc
Full range.
@end table
@item chroma_sample_position
Set the chroma sample location in the stream (see AV1 section 6.4.2).
This can only be set for 4:2:0 streams.
@table @samp
@item vertical
Left position (matching the default in MPEG-2 and H.264).
@item colocated
Top-left position.
@end table
@item tick_rate
Set the tick rate (@emph{num_units_in_display_tick / time_scale}) in
the timing info in the sequence header.
@item num_ticks_per_picture
Set the number of ticks in each picture, to indicate that the stream
has a fixed framerate. Ignored if @option{tick_rate} is not also set.
@end table
This is required for example when copying an AAC stream from a raw
ADTS AAC container to a FLV or a MOV/MP4 file.
@section chomp
Remove zero padding at the end of a packet.
@section dca_core
Extract the core from a DCA/DTS stream, dropping extensions such as
DTS-HD.
@section dump_extra
Add extradata to the beginning of the filtered packets.
@table @option
@item freq
The additional argument specifies which packets should be filtered.
It accepts the values:
@table @samp
@item a
add extradata to all key packets, but only if @var{local_header} is
set in the @option{flags2} codec context field
@item k
@item keyframe
add extradata to all key packets
@item e
@item all
add extradata to all packets
@end table
@end table
If not specified it is assumed @samp{e}.
If not specified it is assumed @samp{k}.
For example the following @command{ffmpeg} command forces a global
header (thus disabling individual packet headers) in the H.264 packets
@@ -127,157 +67,6 @@ 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 eac3_core
Extract the core from a E-AC-3 stream, dropping extra channels.
@section extract_extradata
Extract the in-band extradata.
Certain codecs allow the long-term headers (e.g. MPEG-2 sequence headers,
or H.264/HEVC (VPS/)SPS/PPS) to be transmitted either "in-band" (i.e. as a part
of the bitstream containing the coded frames) or "out of band" (e.g. on the
container level). This latter form is called "extradata" in FFmpeg terminology.
This bitstream filter detects the in-band headers and makes them available as
extradata.
@table @option
@item remove
When this option is enabled, the long-term headers are removed from the
bitstream after extraction.
@end table
@section filter_units
Remove units with types in or not in a given set from the stream.
@table @option
@item pass_types
List of unit types or ranges of unit types to pass through while removing
all others. This is specified as a '|'-separated list of unit type values
or ranges of values with '-'.
@item remove_types
Identical to @option{pass_types}, except the units in the given set
removed and all others passed through.
@end table
Extradata is unchanged by this transformation, but note that if the stream
contains inline parameter sets then the output may be unusable if they are
removed.
For example, to remove all non-VCL NAL units from an H.264 stream:
@example
ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=pass_types=1-5' OUTPUT
@end example
To remove all AUDs, SEI and filler from an H.265 stream:
@example
ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=remove_types=35|38-40' OUTPUT
@end example
@section hapqa_extract
Extract Rgb or Alpha part of an HAPQA file, without recompression, in order to create an HAPQ or an HAPAlphaOnly file.
@table @option
@item texture
Specifies the texture to keep.
@table @option
@item color
@item alpha
@end table
@end table
Convert HAPQA to HAPQ
@example
ffmpeg -i hapqa_inputfile.mov -c copy -bsf:v hapqa_extract=texture=color -tag:v HapY -metadata:s:v:0 encoder="HAPQ" hapq_file.mov
@end example
Convert HAPQA to HAPAlphaOnly
@example
ffmpeg -i hapqa_inputfile.mov -c copy -bsf:v hapqa_extract=texture=alpha -tag:v HapA -metadata:s:v:0 encoder="HAPAlpha Only" hapalphaonly_file.mov
@end example
@section h264_metadata
Modify metadata embedded in an H.264 stream.
@table @option
@item aud
Insert or remove AUD NAL units in all access units of the stream.
@table @samp
@item insert
@item remove
@end table
@item sample_aspect_ratio
Set the sample aspect ratio of the stream in the VUI parameters.
@item video_format
@item video_full_range_flag
Set the video format in the stream (see H.264 section E.2.1 and
table E-2).
@item colour_primaries
@item transfer_characteristics
@item matrix_coefficients
Set the colour description in the stream (see H.264 section E.2.1
and tables E-3, E-4 and E-5).
@item chroma_sample_loc_type
Set the chroma sample location in the stream (see H.264 section
E.2.1 and figure E-1).
@item tick_rate
Set the tick rate (num_units_in_tick / time_scale) in the VUI
parameters. This is the smallest time unit representable in the
stream, and in many cases represents the field rate of the stream
(double the frame rate).
@item fixed_frame_rate_flag
Set whether the stream has fixed framerate - typically this indicates
that the framerate is exactly half the tick rate, but the exact
meaning is dependent on interlacing and the picture structure (see
H.264 section E.2.1 and table E-6).
@item crop_left
@item crop_right
@item crop_top
@item crop_bottom
Set the frame cropping offsets in the SPS. These values will replace
the current ones if the stream is already cropped.
These fields are set in pixels. Note that some sizes may not be
representable if the chroma is subsampled or the stream is interlaced
(see H.264 section 7.4.2.1.1).
@item sei_user_data
Insert a string as SEI unregistered user data. The argument must
be of the form @emph{UUID+string}, where the UUID is as hex digits
possibly separated by hyphens, and the string can be anything.
For example, @samp{086f3693-b7b3-4f2c-9653-21492feee5b8+hello} will
insert the string ``hello'' associated with the given UUID.
@item delete_filler
Deletes both filler NAL units and filler SEI messages.
@item level
Set the level in the SPS. Refer to H.264 section A.3 and tables A-1
to A-5.
The argument must be the name of a level (for example, @samp{4.2}), a
level_idc value (for example, @samp{42}), or the special name @samp{auto}
indicating that the filter should attempt to guess the level from the
input stream properties.
@end table
@section h264_mp4toannexb
Convert an H.264 bitstream from length prefixed mode to start code
@@ -285,7 +74,7 @@ prefixed mode (as defined in the Annex B of the ITU-T H.264
specification).
This is required by some streaming formats, typically the MPEG-2
transport stream format (muxer @code{mpegts}).
transport stream format ("mpegts").
For example to remux an MP4 file containing an H.264 stream to mpegts
format with @command{ffmpeg}, you can use the command:
@@ -294,92 +83,6 @@ format with @command{ffmpeg}, you can use the command:
ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts
@end example
Please note that this filter is auto-inserted for MPEG-TS (muxer
@code{mpegts}) and raw H.264 (muxer @code{h264}) output formats.
@section h264_redundant_pps
This applies a specific fixup to some Blu-ray streams which contain
redundant PPSs modifying irrelevant parameters of the stream which
confuse other transformations which require correct extradata.
A new single global PPS is created, and all of the redundant PPSs
within the stream are removed.
@section hevc_metadata
Modify metadata embedded in an HEVC stream.
@table @option
@item aud
Insert or remove AUD NAL units in all access units of the stream.
@table @samp
@item insert
@item remove
@end table
@item sample_aspect_ratio
Set the sample aspect ratio in the stream in the VUI parameters.
@item video_format
@item video_full_range_flag
Set the video format in the stream (see H.265 section E.3.1 and
table E.2).
@item colour_primaries
@item transfer_characteristics
@item matrix_coefficients
Set the colour description in the stream (see H.265 section E.3.1
and tables E.3, E.4 and E.5).
@item chroma_sample_loc_type
Set the chroma sample location in the stream (see H.265 section
E.3.1 and figure E.1).
@item tick_rate
Set the tick rate in the VPS and VUI parameters (num_units_in_tick /
time_scale). Combined with @option{num_ticks_poc_diff_one}, this can
set a constant framerate in the stream. Note that it is likely to be
overridden by container parameters when the stream is in a container.
@item num_ticks_poc_diff_one
Set poc_proportional_to_timing_flag in VPS and VUI and use this value
to set num_ticks_poc_diff_one_minus1 (see H.265 sections 7.4.3.1 and
E.3.1). Ignored if @option{tick_rate} is not also set.
@item crop_left
@item crop_right
@item crop_top
@item crop_bottom
Set the conformance window cropping offsets in the SPS. These values
will replace the current ones if the stream is already cropped.
These fields are set in pixels. Note that some sizes may not be
representable if the chroma is subsampled (H.265 section 7.4.3.2.1).
@end table
@section hevc_mp4toannexb
Convert an HEVC/H.265 bitstream from length prefixed mode to start code
prefixed mode (as defined in the Annex B of the ITU-T H.265
specification).
This is required by some streaming formats, typically the MPEG-2
transport stream format (muxer @code{mpegts}).
For example to remux an MP4 file containing an HEVC stream to mpegts
format with @command{ffmpeg}, you can use the command:
@example
ffmpeg -i INPUT.mp4 -codec copy -bsf:v hevc_mp4toannexb OUTPUT.ts
@end example
Please note that this filter is auto-inserted for MPEG-TS (muxer
@code{mpegts}) and raw HEVC/H.265 (muxer @code{h265} or
@code{hevc}) output formats.
@section imxdump
Modifies the bitstream to fit in MOV and to be usable by the Final Cut
@@ -430,58 +133,11 @@ exiftran -i -9 frame*.jpg
ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
@end example
@section mjpegadump
@section mjpega_dump_header
Add an MJPEG A header to the bitstream, to enable decoding by
Quicktime.
@section movsub
@anchor{mov2textsub}
@section mov2textsub
Extract a representable text file from MOV subtitles, stripping the
metadata header from each subtitle packet.
See also the @ref{text2movsub} filter.
@section mp3decomp
Decompress non-standard compressed MP3 audio headers.
@section mpeg2_metadata
Modify metadata embedded in an MPEG-2 stream.
@table @option
@item display_aspect_ratio
Set the display aspect ratio in the stream.
The following fixed values are supported:
@table @option
@item 4/3
@item 16/9
@item 221/100
@end table
Any other value will result in square pixels being signalled instead
(see H.262 section 6.3.3 and table 6-3).
@item frame_rate
Set the frame rate in the stream. This is constructed from a table
of known values combined with a small multiplier and divisor - if
the supplied value is not exactly representable, the nearest
representable value will be used instead (see H.262 section 6.3.3
and table 6-4).
@item video_format
Set the video format in the stream (see H.262 section 6.3.6 and
table 6-6).
@item colour_primaries
@item transfer_characteristics
@item matrix_coefficients
Set the colour description in the stream (see H.262 section 6.3.6
and tables 6-7, 6-8 and 6-9).
@end table
@section mp3_header_decompress
@section mpeg4_unpack_bframes
@@ -505,108 +161,20 @@ ffmpeg -i INPUT.avi -codec copy -bsf:v mpeg4_unpack_bframes OUTPUT.avi
@section noise
Damages the contents of packets or simply drops them without damaging the
container. Can be used for fuzzing or testing error resilience/concealment.
Damages the contents of packets without damaging the container. Can be
used for fuzzing or testing error resilience/concealment.
Parameters:
@table @option
@item amount
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.
@item dropamount
A numeral string, whose value is related to how often packets will be dropped.
Therefore, values below or equal to 0 are forbidden, and the lower the more
frequent packets will be dropped, with 1 meaning every packet is dropped.
@end table
The following example applies the modification to every byte but does not drop
any packets.
@example
ffmpeg -i INPUT -c copy -bsf noise[=1] output.mkv
@end example
@section null
This bitstream filter passes the packets through unchanged.
applies the modification to every byte.
@section remove_extra
Remove extradata from packets.
It accepts the following parameter:
@table @option
@item freq
Set which frame types to remove extradata from.
@table @samp
@item k
Remove extradata from non-keyframes only.
@item keyframe
Remove extradata from keyframes only.
@item e, all
Remove extradata from all frames.
@end table
@end table
@anchor{text2movsub}
@section text2movsub
Convert text subtitles to MOV subtitles (as used by the @code{mov_text}
codec) with metadata headers.
See also the @ref{mov2textsub} filter.
@section trace_headers
Log trace output containing all syntax elements in the coded stream
headers (everything above the level of individual coded blocks).
This can be useful for debugging low-level stream issues.
Supports H.264, H.265, MPEG-2 and VP9.
@section vp9_metadata
Modify metadata embedded in a VP9 stream.
@table @option
@item color_space
Set the color space value in the frame header.
@table @samp
@item unknown
@item bt601
@item bt709
@item smpte170
@item smpte240
@item bt2020
@item rgb
@end table
@item color_range
Set the color range value in the frame header. Note that this cannot
be set in RGB streams.
@table @samp
@item tv
@item pc
@end table
@end table
@section vp9_superframe
Merge VP9 invisible (alt-ref) frames back into VP9 superframes. This
fixes merging of split/segmented VP9 streams where the alt-ref frame
was split from its visible counterpart.
@section vp9_superframe_split
Split VP9 superframes into single frames.
@section vp9_raw_reorder
Given a VP9 stream with correct timestamps but possibly out of order,
insert additional show-existing-frame packets to correct the ordering.
@c man end BITSTREAM FILTERS

View File

@@ -39,15 +39,15 @@ 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).
config
Reconfigure the project with the current configuration.
tools/target_dec_<decoder>_fuzzer
Build fuzzer to fuzz the specified decoder.
Useful standard make commands:
make -t <target>

View File

@@ -44,6 +44,12 @@ Use 1/4 pel motion compensation.
Use loop filter.
@item qscale
Use fixed qscale.
@item gmc
Use gmc.
@item mv0
Always try a mb with mv=<0,0>.
@item input_preserved
@item pass1
Use internal 2pass ratecontrol in first pass mode.
@item pass2
@@ -56,6 +62,8 @@ Do not draw edges.
Set error[?] variables during encoding.
@item truncated
@item naq
Normalize adaptive quantization.
@item ildct
Use interlaced DCT.
@item low_delay
@@ -130,8 +138,7 @@ Set audio sampling rate (in Hz).
Set number of audio channels.
@item cutoff @var{integer} (@emph{encoding,audio})
Set cutoff bandwidth. (Supported only by selected encoders, see
their respective documentation sections.)
Set cutoff bandwidth.
@item frame_size @var{integer} (@emph{encoding,audio})
Set audio frame size.
@@ -250,7 +257,7 @@ 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
@@ -449,9 +456,6 @@ 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.
@@ -467,6 +471,8 @@ rate control
macroblock (MB) type
@item qp
per-block quantization parameter (QP)
@item mv
motion vector
@item dct_coeff
@item green_metadata
@@ -476,12 +482,18 @@ display complexity metadata for the upcoming frame, GoP or for a given duration.
@item startcode
@item pts
@item er
error recognition
@item mmco
memory management control operations (H.264)
@item bugs
@item vis_qp
visualize quantization parameter (QP), lower QP are tinted greener
@item vis_mb_type
visualize block types
@item buffers
picture buffer allocations
@item thread_ops
@@ -490,6 +502,21 @@ threading operations
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
forward predicted MVs of P-frames
@item bf
forward predicted MVs of B-frames
@item bb
backward predicted MVs of B-frames
@end table
@item cmp @var{integer} (@emph{encoding,video})
Set full pel me compare function.
@@ -726,6 +753,8 @@ Set context model.
@item slice_flags @var{integer}
@item xvmc_acceleration @var{integer}
@item mbd @var{integer} (@emph{encoding,video})
Set macroblock decision algorithm (high quality mode).
@@ -986,6 +1015,10 @@ Set chroma qp offset from luma.
@item trellis @var{integer} (@emph{encoding,audio,video})
Set rate-distortion optimal quantization.
@item sc_factor @var{integer} (@emph{encoding,video})
Set value multiplied by qscale for each frame and added to
scene_change_score.
@item mv0_threshold @var{integer} (@emph{encoding,video})
@item b_sensitivity @var{integer} (@emph{encoding,video})
Adjust sensitivity of b_frame_strategy 1.
@@ -1013,136 +1046,15 @@ 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
@item smpte428_1
SMPTE ST 428-1
@item smpte431
SMPTE 431-2
@item smpte432
SMPTE 432-1
@item jedec-p22
JEDEC P22
@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
@item log100
Log
@item log_sqrt
@item log316
Log square root
@item iec61966_2_4
@item iec61966-2-4
IEC 61966-2-4
@item bt1361
@item bt1361e
BT.1361
@item iec61966_2_1
@item iec61966-2-1
IEC 61966-2-1
@item bt2020_10
@item bt2020_10bit
BT.2020 - 10 bit
@item bt2020_12
@item bt2020_12bit
BT.2020 - 12 bit
@item smpte2084
SMPTE ST 2084
@item smpte428
@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 bt2020nc
@item bt2020_ncl
BT.2020 NCL
@item bt2020c
@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.
Possible values:
@table @samp
@item tv
@item mpeg
MPEG (219*2^(n-8))
@item pc
@item jpeg
JPEG (2^n-1)
@end table
@item chroma_sample_location @var{integer} (@emph{decoding/encoding,video})
Possible values:
@table @samp
@item left
@item center
@item topleft
@item top
@item bottomleft
@item bottom
@end table
@item log_level_offset @var{integer}
Set the log level offset.
@@ -1221,13 +1133,13 @@ Interlaced video, top coded first, bottom displayed first
Interlaced video, bottom coded first, top displayed first
@end table
@item skip_alpha @var{bool} (@emph{decoding,video})
@item skip_alpha @var{integer} (@emph{decoding,video})
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.
"," 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
@@ -1238,20 +1150,6 @@ ffprobe -dump_separator "
" -i ~/videos/matrixbench_mpeg2.mpg
@end example
@item max_pixels @var{integer} (@emph{decoding/encoding,video})
Maximum number of pixels per image. This value can be used to avoid out of
memory failures due to large images.
@item apply_cropping @var{bool} (@emph{decoding,video})
Enable cropping if cropping parameters are multiples of the required
alignment for the left and top parameters. If the alignment is not met the
cropping will be partially applied to maintain alignment.
Default is 1 (enabled).
Note: The required alignment depends on if @code{AV_CODEC_FLAG_UNALIGNED} is set and the
CPU. @code{AV_CODEC_FLAG_UNALIGNED} cannot be changed from the command line. Also hardware
decoders will not apply left/top Cropping.
@end table
@c man end CODEC OPTIONS

View File

@@ -25,6 +25,13 @@ 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.
@@ -47,12 +54,6 @@ top-field-first is assumed
@end table
@section libdavs2
AVS2-P2/IEEE1857.4 video decoder wrapper.
This decoder allows libavcodec to decode AVS2 streams with davs2 library.
@c man end VIDEO DECODERS
@chapter Audio Decoders
@@ -108,7 +109,7 @@ correctly by using lavc's old buggy lpc logic for decoding.
@section ffwavesynth
Internal wave synthesizer.
Internal wave synthetizer.
This decoder generates wave patterns according to predefined sequences. Its
use is purely internal and the format of the data it accepts is not publicly
@@ -254,25 +255,18 @@ configuration. You need to explicitly configure the build with
@table @option
@item txt_page
List of teletext page numbers to decode. Pages that do not match the specified
list are dropped. You may use the special @code{*} string to match all pages,
or @code{subtitle} to match all subtitle pages.
List of teletext page numbers to decode. You may use the special * string to
match all pages. Pages that do not match the specified list are dropped.
Default value is *.
@item txt_chop_top
Discards the top teletext line. Default value is 1.
@item txt_format
Specifies the format of the decoded subtitles.
@table @option
@item bitmap
The default format, you should use this for teletext pages, because certain
graphics and colors cannot be expressed in simple text or even ASS.
@item text
Simple text based output without formatting.
@item ass
Formatted ASS output, subtitle pages and teletext pages are returned in
different styles, subtitle pages are stripped down to text, but an effort is
made to keep the text alignment and the formatting.
@end table
Specifies the format of the decoded subtitles. The teletext decoder is capable
of decoding the teletext pages to bitmaps or to simple text, you should use
"bitmap" for teletext pages, because certain graphics and colors cannot be
expressed in simple text. You might use "text" for teletext based subtitles if
your application can handle simple text based subtitles. Default value is
bitmap.
@item txt_left
X offset of generated bitmaps, default is 0.
@item txt_top
@@ -281,12 +275,11 @@ Y offset of generated bitmaps, default is 0.
Chops leading and trailing spaces and removes empty lines from the generated
text. This option is useful for teletext based subtitles where empty spaces may
be present at the start or at the end of the lines or empty lines may be
present between the subtitle lines because of double-sized teletext characters.
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 -1 which means infinity or until the next
subtitle event comes.
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.

View File

@@ -13,9 +13,8 @@ You can disable all the demuxers using the configure option
the option @code{--enable-demuxer=@var{DEMUXER}}, or disable it
with the option @code{--disable-demuxer=@var{DEMUXER}}.
The option @code{-demuxers} of the ff* tools will display the list of
enabled demuxers. Use @code{-formats} to view a combined list of
enabled demuxers and muxers.
The option @code{-formats} of the ff* tools will display the list of
enabled demuxers.
The description of some of the currently available demuxers follows.
@@ -73,7 +72,7 @@ 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
@@ -105,10 +104,10 @@ 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}}
@@ -193,9 +192,7 @@ 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
@@ -244,39 +241,30 @@ file subdir/file-2.wav
@end example
@end itemize
@section dash
Dynamic Adaptive Streaming over HTTP demuxer.
This demuxer presents all AVStreams found in the manifest.
By setting the discard flags on AVStreams the caller can decide
which streams to actually receive.
Each stream mirrors the @code{id} and @code{bandwidth} properties from the
@code{<Representation>} as metadata keys named "id" and "variant_bitrate" respectively.
@section flv, live_flv
@section flv
Adobe Flash Video Format demuxer.
This demuxer is used to demux FLV files and RTMP network streams. In case of live network streams, if you force format, you may use live_flv option instead of flv to survive timestamp discontinuities.
@example
ffmpeg -f flv -i myfile.flv ...
ffmpeg -f live_flv -i rtmp://<any.server>/anything/key ....
@end example
This demuxer is used to demux FLV files and RTMP network streams.
@table @option
@item -flv_metadata @var{bool}
Allocate the streams according to the onMetaData array content.
@item -flv_ignore_prevtag @var{bool}
Ignore the size of previous tag value.
@item -flv_full_metadata @var{bool}
Output all context of the onMetadata.
@end table
@section libgme
The Game Music Emu library is a collection of video game music file emulators.
See @url{http://code.google.com/p/game-music-emu/} for more information.
Some files have multiple tracks. The demuxer will pick the first track by
default. The @option{track_index} option can be used to select a different
track. Track indexes start at 0. The demuxer exports the number of tracks as
@var{tracks} meta data entry.
For very large files, the @option{max_size} option may have to be adjusted.
@section gif
Animated GIF demuxer.
@@ -316,32 +304,6 @@ used to end the output video at the length of the shortest input file,
which in this case is @file{input.mp4} as the GIF in this example loops
infinitely.
@section hls
HLS demuxer
It accepts the following options:
@table @option
@item live_start_index
segment index to start live streams at (negative values are from the end).
@item allowed_extensions
',' separated list of file extensions that hls is allowed to access.
@item max_reload
Maximum number of times a insufficient list is attempted to be reloaded.
Default value is 1000.
@item http_persistent
Use persistent HTTP connections. Applicable only for HTTP streams.
Enabled by default.
@item http_multiple
Use multiple HTTP connections for downloading HTTP segments.
Enabled by default for HTTP/1.1 servers.
@end table
@section image2
Image file demuxer.
@@ -477,49 +439,9 @@ ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv
@end example
@end itemize
@section libgme
@section mov/mp4/3gp/Quicktme
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.
Quicktime / MP4 demuxer.
This demuxer accepts the following options:
@table @option
@@ -544,9 +466,6 @@ This demuxer accepts the following options:
Set size limit for looking up a new synchronization. Default value is
65536.
@item skip_unknown_pmt
Skip PMTs for programs not defined in the PAT. Default value is 0.
@item fix_teletext_pts
Override 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
@@ -561,10 +480,6 @@ Show the detected raw packet size, cannot be set by the user.
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.
@item merge_pmt_versions
Re-use existing streams when a PMT's version is updated and elementary
streams move to different PIDs. Default value is 0.
@end table
@section mpjpeg

View File

@@ -10,7 +10,9 @@
@contents
@chapter Notes for external developers
@chapter Developers Guide
@section Notes for external developers
This document is mostly useful for internal FFmpeg developers.
External developers who need to use the API in their application should
@@ -28,13 +30,15 @@ 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}.
@chapter Contributing
@section Contributing
There are 2 ways by which code gets into FFmpeg:
There are 3 ways by which code gets into FFmpeg.
@itemize @bullet
@item Submitting patches to the ffmpeg-devel mailing list.
@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.
@end itemize
Whichever way, changes should be reviewed by the maintainer of the code
@@ -43,9 +47,9 @@ The developer making the commit and the author are responsible for their changes
and should try to fix issues their commit causes.
@anchor{Coding Rules}
@chapter Coding Rules
@section Coding Rules
@section Code formatting conventions
@subsection Code formatting conventions
There are the following guidelines regarding the indentation in files:
@@ -70,7 +74,7 @@ The presentation is one inspired by 'indent -i4 -kr -nut'.
The main priority in FFmpeg is simplicity and small code size in order to
minimize the bug count.
@section Comments
@subsection Comments
Use the JavaDoc/Doxygen format (see examples below) so that code documentation
can be generated automatically. All nontrivial functions should have a comment
above them explaining what the function does, even if it is just one sentence.
@@ -110,7 +114,7 @@ int myfunc(int my_parameter)
...
@end example
@section C language features
@subsection C language features
FFmpeg is programmed in the ISO C90 language with a few additional
features from ISO C99, namely:
@@ -127,14 +131,6 @@ designated struct initializers (@samp{struct s x = @{ .i = 17 @};});
@item
compound literals (@samp{x = (struct s) @{ 17, 23 @};}).
@item
for loops with variable definition (@samp{for (int i = 0; i < 8; i++)});
@item
Implementation defined behavior for signed integers is assumed to match the
expected behavior for two's complement. Non representable values in integer
casts are binary truncated. Shift right of signed values uses sign extension.
@end itemize
These features are supported by all compilers we care about, so we will not
@@ -159,7 +155,7 @@ mixing statements and declarations;
GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}).
@end itemize
@section Naming conventions
@subsection Naming conventions
All names should be composed with underscores (_), not CamelCase. For example,
@samp{avfilter_get_video_buffer} is an acceptable function name and
@samp{AVFilterGetVideo} is not. The exception from this are type names, like
@@ -183,7 +179,7 @@ e.g. @samp{ff_w64_demuxer}.
@item
For variables and functions visible outside of file scope, used internally
across multiple libraries, use @code{avpriv_} as prefix, for example,
@samp{avpriv_report_missing_feature}.
@samp{avpriv_aac_parse_header}.
@item
Each library has its own prefix for public symbols, in addition to the
@@ -203,7 +199,7 @@ letter as they are reserved by the C standard. Names starting with @code{_}
are reserved at the file level and may not be used for externally visible
symbols. If in doubt, just avoid names starting with @code{_} altogether.
@section Miscellaneous conventions
@subsection Miscellaneous conventions
@itemize @bullet
@item
@@ -215,7 +211,7 @@ Casts should be used only when necessary. Unneeded parentheses
should also be avoided if they don't make the code easier to understand.
@end itemize
@section Editor configuration
@subsection Editor configuration
In order to configure Vim to follow FFmpeg formatting conventions, paste
the following snippet into your @file{.vimrc}:
@example
@@ -248,10 +244,10 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
(setq c-default-style "ffmpeg")
@end lisp
@chapter Development Policy
@section Development Policy
@section 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
@@ -264,15 +260,15 @@ 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.
@@ -280,24 +276,30 @@ 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
@@ -306,7 +308,7 @@ 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
@@ -320,7 +322,7 @@ 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.
@@ -332,31 +334,47 @@ 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!
@section 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
@@ -366,7 +384,7 @@ 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.
@@ -375,84 +393,18 @@ 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.
@section Documentation/Other
@subheading Subscribe to the ffmpeg-devel mailing list.
It is important to be subscribed to the
@uref{https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel, ffmpeg-devel}
mailing list. Almost any non-trivial patch is to be sent there for review.
Other developers may have comments about your contribution. We expect you see
those comments, and to improve it if requested. (N.B. Experienced committers
have other channels, and may sometimes skip review for trivial fixes.) Also,
discussion here about bug fixes and FFmpeg improvements by other developers may
be helpful information for you. Finally, by being a list subscriber, your
contribution will be posted immediately to the list, without the moderation
hold which messages from non-subscribers experience.
However, it is more important to the project that we receive your patch than
that you be subscribed to the ffmpeg-devel list. If you have a patch, and don't
want to subscribe and discuss the patch, then please do send it to the list
anyway.
@subheading Subscribe to the ffmpeg-cvslog mailing list.
Diffs of all commits are sent to the
@uref{https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-cvslog, ffmpeg-cvslog}
mailing list. Some developers read this list to review all code base changes
from all sources. Subscribing to this list is not mandatory.
@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.
@end enumerate
We think our rules are not too hard. If you have comments, contact us.
@chapter 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}
@chapter Submitting patches
@section Submitting patches
First, read the @ref{Coding Rules} above if you did not yet, in particular
the rules regarding patch submission.
@@ -485,11 +437,7 @@ Patches should be posted to the
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
@@ -501,7 +449,7 @@ Give us a few days to react. But if some time passes without reaction,
send a reminder by email. Your patch should eventually be dealt with.
@chapter New codecs or formats checklist
@section New codecs or formats checklist
@enumerate
@item
@@ -553,7 +501,7 @@ Did you make sure it compiles standalone, i.e. with
@end enumerate
@chapter Patch submission checklist
@section patch submission checklist
@enumerate
@item
@@ -563,9 +511,9 @@ Does @code{make fate} pass with the patch applied?
Was the patch generated with git format-patch or send-email?
@item
Did you sign-off your patch? (@code{git commit -s})
See @uref{https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/plain/Documentation/process/submitting-patches.rst, Sign your work} for the meaning
of @dfn{sign-off}.
Did you sign off your patch? (git commit -s)
See @url{http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob_plain;f=Documentation/SubmittingPatches} for the meaning
of sign off.
@item
Did you provide a clear git commit log message?
@@ -666,7 +614,7 @@ Test your code with valgrind and or Address Sanitizer to ensure it's free
of leaks, out of array accesses, etc.
@end enumerate
@chapter Patch review process
@section Patch review process
All patches posted to ffmpeg-devel will be reviewed, unless they contain a
clear note that the patch is not for the git master branch.
@@ -697,7 +645,7 @@ 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}
@chapter Regression tests
@section Regression tests
Before submitting a patch (or committing to the repository), you should at least
test that you did not break anything.
@@ -708,7 +656,7 @@ Running 'make fate' accomplishes this, please see @url{fate.html} for details.
this case, the reference results of the regression tests shall be modified
accordingly].
@section Adding files to the fate-suite dataset
@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.
@@ -719,7 +667,7 @@ 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
the ffmpeg-devel mailing list, a direct link to download the sample media.
@section Visualizing Test Coverage
@subsection Visualizing Test Coverage
The FFmpeg build system allows visualizing the test coverage in an easy
manner with the coverage tools @code{gcov}/@code{lcov}. This involves
@@ -746,7 +694,7 @@ You can use the command @code{make lcov-reset} to reset the coverage
measurements. You will need to rerun @code{make lcov} after running a
new test.
@section Using Valgrind
@subsection Using Valgrind
The configure script provides a shortcut for using valgrind to spot bugs
related to memory handling. Just add the option
@@ -760,7 +708,7 @@ In case you need finer control over how valgrind is invoked, use the
your configure line instead.
@anchor{Release process}
@chapter Release process
@section Release process
FFmpeg maintains a set of @strong{release branches}, which are the
recommended deliverable for system integrators and distributors (such as
@@ -792,7 +740,7 @@ adjustments to the symbol versioning file. Please discuss such changes
on the @strong{ffmpeg-devel} mailing list in time to allow forward planning.
@anchor{Criteria for Point Releases}
@section Criteria for Point Releases
@subsection Criteria for Point Releases
Changes that match the following criteria are valid candidates for
inclusion into a point release:
@@ -816,7 +764,7 @@ point releases of the same release branch.
The order for checking the rules is (1 OR 2 OR 3) AND 4.
@section Release Checklist
@subsection Release Checklist
The release process involves the following steps:

1
doc/doxy/.gitignore vendored
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@@ -1 +0,0 @@
/html/

View File

@@ -61,9 +61,9 @@ 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.
all quantizers and adjusting some individual quantizer a little.
Will tune itself based on whether aac_is/aac_ms/aac_pns are enabled.
This is the default choice for a coder.
@item anmr
Average noise to mask ratio (ANMR) trellis-based solution.
@@ -76,27 +76,27 @@ Not currently recommended.
@item fast
Constant quantizer method.
Uses a cheaper version of twoloop algorithm that doesn't try to do as many
clever adjustments. Worse with low bitrates (less than 64kbps), but is better
and much faster at higher bitrates.
This is the default choice for a coder
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.
@end table
@item aac_ms
Sets mid/side coding mode. The default value of "auto" will automatically use
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 beneficial.
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 imperceptible white noise during the decoding process. By default, it's
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
@@ -130,20 +130,20 @@ 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.
Equivalent to -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.
Long term prediction profile, is enabled by and will enable the 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.
Main-type prediction profile, is enabled by and will enable the aac_pred option.
Introduced in MPEG2.
@end table
If this option is unspecified it is set to @samp{aac_low}.
@end table
@end table
@section ac3 and ac3_fixed
@@ -487,10 +487,6 @@ is an optional AC-3 feature that increases quality by selectively encoding
the left/right channels as mid/side. This option is enabled by default, and it
is highly recommended that it be left as enabled except for testing purposes.
@item cutoff @var{frequency}
Set lowpass cutoff frequency. If unspecified, the encoder selects a default
determined by various other encoding parameters.
@end table
@subsection Floating-Point-Only AC-3 Encoding Options
@@ -548,8 +544,7 @@ The following options are supported by FFmpeg's flac encoder.
@table @option
@item compression_level
Sets the compression level, which chooses defaults for many other options
if they are not set explicitly. Valid values are from 0 to 12, 5 is the
default.
if they are not set explicitly.
@item frame_size
Sets the size of the frames in samples per channel.
@@ -598,7 +593,7 @@ Channel mode
@item auto
The mode is chosen automatically for each frame
@item indep
Channels are independently coded
Chanels are independently coded
@item left_side
@item right_side
@item mid_side
@@ -616,27 +611,111 @@ and slightly improves compression.
@end table
@anchor{opusenc}
@section opus
@anchor{libfaac}
@section libfaac
Opus encoder.
libfaac AAC (Advanced Audio Coding) encoder wrapper.
This is a native FFmpeg encoder for the Opus format. Currently its in development and
only implements the CELT part of the codec. Its quality is usually worse and at best
is equal to the libopus encoder.
This encoder is of much lower quality and is more unstable than any other AAC
encoders, so it's highly recommended to instead use other encoders, like
@ref{aacenc,,the native FFmpeg AAC encoder}.
This encoder also requires the presence of the libfaac headers and library
during configuration. You need to explicitly configure the build with
@code{--enable-libfaac --enable-nonfree}.
@subsection Options
@table @option
@item b
Set bit rate in bits/s. If unspecified it uses the number of channels and the layout
to make a good guess.
The following shared FFmpeg codec options are recognized.
@item opus_delay
Sets the maximum delay in milliseconds. Lower delays than 20ms will very quickly
decrease quality.
The following options are supported by the libfaac wrapper. The
@command{faac}-equivalent of the options are listed in parentheses.
@table @option
@item b (@emph{-b})
Set bit rate in bits/s for ABR (Average Bit Rate) mode. If the bit rate
is not explicitly specified, it is automatically set to a suitable
value depending on the selected profile. @command{faac} bitrate is
expressed in kilobits/s.
Note that libfaac does not support CBR (Constant Bit Rate) but only
ABR (Average Bit Rate).
If VBR mode is enabled this option is ignored.
@item ar (@emph{-R})
Set audio sampling rate (in Hz).
@item ac (@emph{-c})
Set the number of audio channels.
@item cutoff (@emph{-C})
Set cutoff frequency. If not specified (or explicitly set to 0) it
will use a value automatically computed by the library. Default value
is 0.
@item profile
Set audio profile.
The following profiles are recognized:
@table @samp
@item aac_main
Main AAC (Main)
@item aac_low
Low Complexity AAC (LC)
@item aac_ssr
Scalable Sample Rate (SSR)
@item aac_ltp
Long Term Prediction (LTP)
@end table
If not specified it is set to @samp{aac_low}.
@item flags +qscale
Set constant quality VBR (Variable Bit Rate) mode.
@item global_quality
Set quality in VBR mode as an integer number of lambda units.
Only relevant when VBR mode is enabled with @code{flags +qscale}. The
value is converted to QP units by dividing it by @code{FF_QP2LAMBDA},
and used to set the quality value used by libfaac. A reasonable range
for the option value in QP units is [10-500], the higher the value the
higher the quality.
@item q (@emph{-q})
Enable VBR mode when set to a non-negative value, and set constant
quality value as a double floating point value in QP units.
The value sets the quality value used by libfaac. A reasonable range
for the option value is [10-500], the higher the value the higher the
quality.
This option is valid only using the @command{ffmpeg} command-line
tool. For library interface users, use @option{global_quality}.
@end table
@subsection Examples
@itemize
@item
Use @command{ffmpeg} to convert an audio file to ABR 128 kbps AAC in an M4A (MP4)
container:
@example
ffmpeg -i input.wav -codec:a libfaac -b:a 128k -output.m4a
@end example
@item
Use @command{ffmpeg} to convert an audio file to VBR AAC, using the
LTP AAC profile:
@example
ffmpeg -i input.wav -c:a libfaac -profile:a aac_ltp -q:a 100 output.m4a
@end example
@end itemize
@anchor{libfdk-aac-enc}
@section libfdk_aac
@@ -842,10 +921,6 @@ Set algorithm quality. Valid arguments are integers in the 0-9 range,
with 0 meaning highest quality but slowest, and 9 meaning fastest
while producing the worst quality.
@item cutoff (@emph{--lowpass})
Set lowpass cutoff frequency. If unspecified, the encoder dynamically
adjusts the cutoff.
@item reservoir
Enable use of bit reservoir when set to 1. Default value is 1. LAME
has this enabled by default, but can be overridden by use
@@ -899,95 +974,6 @@ default value is 0 (disabled).
@end table
@section libopus
libopus Opus Interactive Audio Codec encoder wrapper.
Requires the presence of the libopus headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libopus}.
@subsection Option Mapping
Most libopus options are modelled 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.
@table @option
@item b (@emph{bitrate})
Set the bit rate in bits/s. FFmpeg's @option{b} option is
expressed in bits/s, while @command{opusenc}'s @option{bitrate} in
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
in parentheses:
@table @samp
@item off (@emph{hard-cbr})
Use constant bit rate encoding.
@item on (@emph{vbr})
Use variable bit rate encoding (the default).
@item constrained (@emph{cvbr})
Use constrained variable bit rate encoding.
@end table
@item compression_level (@emph{comp})
Set encoding algorithm complexity. Valid options are integers in
the 0-10 range. 0 gives the fastest encodes but lower quality, while 10
gives the highest quality but slowest encoding. The default is 10.
@item frame_duration (@emph{framesize})
Set maximum frame size, or duration of a frame in milliseconds. The
argument must be exactly the following: 2.5, 5, 10, 20, 40, 60. Smaller
frame sizes achieve lower latency but less quality at a given bitrate.
Sizes greater than 20ms are only interesting at fairly low bitrates.
The default is 20ms.
@item packet_loss (@emph{expect-loss})
Set expected packet loss percentage. The default is 0.
@item application (N.A.)
Set intended application type. Valid options are listed below:
@table @samp
@item voip
Favor improved speech intelligibility.
@item audio
Favor faithfulness to the input (the default).
@item lowdelay
Restrict to only the lowest delay modes.
@end table
@item cutoff (N.A.)
Set cutoff bandwidth in Hz. The argument must be exactly one of the
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.
@item apply_phase_inv (N.A.) (requires libopus >= 1.2)
If set to 0, disables the use of phase inversion for intensity stereo,
improving the quality of mono downmixes, but slightly reducing normal stereo
quality. The default is 1 (phase inversion enabled).
@end table
@anchor{libshine}
@section libshine
@@ -1127,6 +1113,79 @@ default value is 0 (disabled).
@end table
@section libopus
libopus Opus Interactive Audio Codec encoder wrapper.
Requires the presence of the libopus headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libopus}.
@subsection Option Mapping
Most libopus options are modelled 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.
@table @option
@item b (@emph{bitrate})
Set the bit rate in bits/s. FFmpeg's @option{b} option is
expressed in bits/s, while @command{opusenc}'s @option{bitrate} in
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
in parentheses:
@table @samp
@item off (@emph{hard-cbr})
Use constant bit rate encoding.
@item on (@emph{vbr})
Use variable bit rate encoding (the default).
@item constrained (@emph{cvbr})
Use constrained variable bit rate encoding.
@end table
@item compression_level (@emph{comp})
Set encoding algorithm complexity. Valid options are integers in
the 0-10 range. 0 gives the fastest encodes but lower quality, while 10
gives the highest quality but slowest encoding. The default is 10.
@item frame_duration (@emph{framesize})
Set maximum frame size, or duration of a frame in milliseconds. The
argument must be exactly the following: 2.5, 5, 10, 20, 40, 60. Smaller
frame sizes achieve lower latency but less quality at a given bitrate.
Sizes greater than 20ms are only interesting at fairly low bitrates.
The default is 20ms.
@item packet_loss (@emph{expect-loss})
Set expected packet loss percentage. The default is 0.
@item application (N.A.)
Set intended application type. Valid options are listed below:
@table @samp
@item voip
Favor improved speech intelligibility.
@item audio
Favor faithfulness to the input (the default).
@item lowdelay
Restrict to only the lowest delay modes.
@end table
@item cutoff (N.A.)
Set cutoff bandwidth in Hz. The argument must be exactly one of the
following: 4000, 6000, 8000, 12000, or 20000, corresponding to
narrowband, mediumband, wideband, super wideband, and fullband
respectively. The default is 0 (cutoff disabled).
@end table
@section libvorbis
libvorbis encoder wrapper.
@@ -1226,27 +1285,6 @@ Same as @samp{3}, but with extra processing enabled.
@end table
@end table
@anchor{mjpegenc}
@section mjpeg
Motion JPEG encoder.
@subsection Options
@table @option
@item huffman
Set the huffman encoding strategy. Possible values:
@table @samp
@item default
Use the default huffman tables. This is the default strategy.
@item optimal
Compute and use optimal huffman tables.
@end table
@end table
@anchor{wavpackenc}
@section wavpack
@@ -1316,81 +1354,6 @@ disabled
A description of some of the currently available video encoders
follows.
@section Hap
Vidvox Hap video encoder.
@subsection Options
@table @option
@item format @var{integer}
Specifies the Hap format to encode.
@table @option
@item hap
@item hap_alpha
@item hap_q
@end table
Default value is @option{hap}.
@item chunks @var{integer}
Specifies the number of chunks to split frames into, between 1 and 64. This
permits multithreaded decoding of large frames, potentially at the cost of
data-rate. The encoder may modify this value to divide frames evenly.
Default value is @var{1}.
@item compressor @var{integer}
Specifies the second-stage compressor to use. If set to @option{none},
@option{chunks} will be limited to 1, as chunked uncompressed frames offer no
benefit.
@table @option
@item none
@item snappy
@end table
Default value is @option{snappy}.
@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 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 libopenh264
Cisco libopenh264 H.264/MPEG-4 AVC encoder wrapper.
@@ -1427,7 +1390,7 @@ 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:
Set slice mode. Can assume one of the follwing possible values:
@table @samp
@item fixed
@@ -1457,6 +1420,30 @@ Set maximum NAL size in bytes.
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.
@@ -1670,7 +1657,7 @@ option to 2.
Enable frame parallel decodability features.
@item aq-mode
Set adaptive quantization mode (0: off (default), 1: variance 2: complexity, 3:
cyclic refresh, 4: equator360).
cyclic refresh).
@item colorspace @emph{color-space}
Set input color space. The VP9 bitstream supports signaling the following
colorspaces:
@@ -1683,15 +1670,6 @@ colorspaces:
@item @samp{smpte240m} @emph{smpte240}
@item @samp{bt2020_ncl} @emph{bt2020}
@end table
@item row-mt @var{boolean}
Enable row based multi-threading.
@item tune-content
Set content type: default (0), screen (1), film (2).
@item corpus-complexity
Corpus VBR mode is a variant of standard VBR where the complexity distribution
midpoint is passed in rather than calculated for a specific clip or chunk.
The valid range is [0, 10000]. 0 (default) uses standard VBR.
@end table
@end table
@@ -1699,6 +1677,7 @@ The valid range is [0, 10000]. 0 (default) uses standard VBR.
For more information about libvpx see:
@url{http://www.webmproject.org/}
@section libwebp
libwebp WebP Image encoder wrapper
@@ -1804,7 +1783,7 @@ the documentation of the undocumented generic options, see
@ref{codec-options,,the Codec Options chapter}.
To get a more accurate and extensive documentation of the libx264
options, invoke the command @command{x264 --fullhelp} or consult
options, invoke the command @command{x264 --full-help} or consult
the libx264 documentation.
@table @option
@@ -1865,10 +1844,6 @@ Exhaustive search.
Hadamard exhaustive search (slowest).
@end table
@item forced-idr
Normally, when forcing a I-frame type, the encoder can select any type
of I-frame. This option forces it to choose an IDR-frame.
@item subq (@emph{subme})
Sub-pixel motion estimation method.
@@ -1891,7 +1866,7 @@ 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
@@ -2117,12 +2092,12 @@ is kept undocumented for some reason.
For example to specify libx264 encoding options with @command{ffmpeg}:
@example
ffmpeg -i foo.mpg -c:v libx264 -x264opts keyint=123:min-keyint=20 -an out.mkv
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).
Only the mpeg2 and h264 decoders provide these. Default is 0 (off).
@item x264-params (N.A.)
Override the x264 configuration using a :-separated list of key=value
@@ -2159,16 +2134,6 @@ Set the x265 preset.
@item tune
Set the x265 tune parameter.
@item profile
Set profile restrictions.
@item crf
Set the quality for constant quality mode.
@item forced-idr
Normally, when forcing a I-frame type, the encoder can select any type
of I-frame. This option forces it to choose an IDR-frame.
@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.
@@ -2365,11 +2330,6 @@ Never write it.
@itemx always
Always write it.
@end table
@item video_format @var{integer}
Specifies the video_format written into the sequence display extension
indicating the source of the video pictures. The default is @samp{unspecified},
can be @samp{component}, @samp{pal}, @samp{ntsc}, @samp{secam} or @samp{mac}.
For maximum compatibility, use @samp{component}.
@end table
@section png
@@ -2403,7 +2363,6 @@ Select the ProRes profile to encode
@item standard
@item hq
@item 4444
@item 4444xq
@end table
@item quant_mat @var{integer}
@@ -2453,6 +2412,27 @@ 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)
@@ -2541,191 +2521,6 @@ encoder use CAVLC instead of CABAC.
@end itemize
@section snow
@subsection Options
@table @option
@item iterative_dia_size
dia size for the iterative motion estimation
@end table
@section VAAPI encoders
Wrappers for hardware encoders accessible via VAAPI.
These encoders only accept input in VAAPI hardware surfaces. If you have input
in software frames, use the @option{hwupload} filter to upload them to the GPU.
The following standard libavcodec options are used:
@itemize
@item
@option{g} / @option{gop_size}
@item
@option{bf} / @option{max_b_frames}
@item
@option{profile}
If not set, this will be determined automatically from the format of the input
frames and the profiles supported by the driver.
@item
@option{level}
@item
@option{b} / @option{bit_rate}
@item
@option{maxrate} / @option{rc_max_rate}
@item
@option{bufsize} / @option{rc_buffer_size}
@item
@option{rc_init_occupancy} / @option{rc_initial_buffer_occupancy}
@item
@option{compression_level}
Speed / quality tradeoff: higher values are faster / worse quality.
@item
@option{q} / @option{global_quality}
Size / quality tradeoff: higher values are smaller / worse quality.
@item
@option{qmin}
@item
@option{qmax}
@item
@option{i_qfactor} / @option{i_quant_factor}
@item
@option{i_qoffset} / @option{i_quant_offset}
@item
@option{b_qfactor} / @option{b_quant_factor}
@item
@option{b_qoffset} / @option{b_quant_offset}
@item
@option{slices}
@end itemize
All encoders support the following options:
@itemize
@item
@option{low_power}
Some drivers/platforms offer a second encoder for some codecs intended to use
less power than the default encoder; setting this option will attempt to use
that encoder. Note that it may support a reduced feature set, so some other
options may not be available in this mode.
@end itemize
Each encoder also has its own specific options:
@table @option
@item h264_vaapi
@option{profile} sets the value of @emph{profile_idc} and the @emph{constraint_set*_flag}s.
@option{level} sets the value of @emph{level_idc}.
@table @option
@item coder
Set entropy encoder (default is @emph{cabac}). Possible values:
@table @samp
@item ac
@item cabac
Use CABAC.
@item vlc
@item cavlc
Use CAVLC.
@end table
@item aud
Include access unit delimiters in the stream (not included by default).
@item sei
Set SEI message types to include.
Some combination of the following values:
@table @samp
@item identifier
Include a @emph{user_data_unregistered} message containing information about
the encoder.
@item timing
Include picture timing parameters (@emph{buffering_period} and
@emph{pic_timing} messages).
@item recovery_point
Include recovery points where appropriate (@emph{recovery_point} messages).
@end table
@end table
@item hevc_vaapi
@option{profile} and @option{level} set the values of
@emph{general_profile_idc} and @emph{general_level_idc} respectively.
@table @option
@item aud
Include access unit delimiters in the stream (not included by default).
@item tier
Set @emph{general_tier_flag}. This may affect the level chosen for the stream
if it is not explicitly specified.
@item sei
Set SEI message types to include.
Some combination of the following values:
@table @samp
@item hdr
Include HDR metadata if the input frames have it
(@emph{mastering_display_colour_volume} and @emph{content_light_level}
messages).
@end table
@end table
@item mjpeg_vaapi
Only baseline DCT encoding is supported. The encoder always uses the standard
quantisation and huffman tables - @option{global_quality} scales the standard
quantisation table (range 1-100).
For YUV, 4:2:0, 4:2:2 and 4:4:4 subsampling modes are supported. RGB is also
supported, and will create an RGB JPEG.
@table @option
@item jfif
Include JFIF header in each frame (not included by default).
@item huffman
Include standard huffman tables (on by default). Turning this off will save
a few hundred bytes in each output frame, but may lose compatibility with some
JPEG decoders which don't fully handle MJPEG.
@end table
@item mpeg2_vaapi
@option{profile} and @option{level} set the value of @emph{profile_and_level_indication}.
@item vp8_vaapi
B-frames are not supported.
@option{global_quality} sets the @emph{q_idx} used for non-key frames (range 0-127).
@table @option
@item loop_filter_level
@item loop_filter_sharpness
Manually set the loop filter parameters.
@end table
@item vp9_vaapi
@option{global_quality} sets the @emph{q_idx} used for P-frames (range 0-255).
@table @option
@item loop_filter_level
@item loop_filter_sharpness
Manually set the loop filter parameters.
@end table
B-frames are supported, but the output stream is always in encode order rather than display
order. If B-frames are enabled, it may be necessary to use the @option{vp9_raw_reorder}
bitstream filter to modify the output stream to display frames in the correct order.
Only normal frames are produced - the @option{vp9_superframe} bitstream filter may be
required to produce a stream usable with all decoders.
@end table
@section vc2
SMPTE VC-2 (previously BBC Dirac Pro). This codec was primarily aimed at
@@ -2791,52 +2586,6 @@ Reduces detail but attempts to preserve color at extremely low bitrates.
@end table
@section libxavs2
xavs2 AVS2-P2/IEEE1857.4 encoder wrapper.
This encoder requires the presence of the libxavs2 headers and library
during configuration. You need to explicitly configure the build with
@option{--enable-libxavs2}.
@subsection Options
@table @option
@item lcu_row_threads
Set the number of parallel threads for rows from 1 to 8 (default 5).
@item initial_qp
Set the xavs2 quantization parameter from 1 to 63 (default 34). This is
used to set the initial qp for the first frame.
@item qp
Set the xavs2 quantization parameter from 1 to 63 (default 34). This is
used to set the qp value under constant-QP mode.
@item max_qp
Set the max qp for rate control from 1 to 63 (default 55).
@item min_qp
Set the min qp for rate control from 1 to 63 (default 20).
@item speed_level
Set the Speed level from 0 to 9 (default 0). Higher is better but slower.
@item log_level
Set the log level from -1 to 3 (default 0). -1: none, 0: error,
1: warning, 2: info, 3: debug.
@item xavs2-params
Set xavs2 options using a list of @var{key}=@var{value} couples separated
by ":".
For example to specify libxavs2 encoding options with @option{-xavs2-params}:
@example
ffmpeg -i input -c:v libxavs2 -xavs2-params preset_level=5 output.avs2
@end example
@end table
@c man end VIDEO ENCODERS
@chapter Subtitles Encoders

View File

@@ -1,24 +0,0 @@
/avio_dir_cmd
/avio_reading
/decode_audio
/decode_video
/demuxing_decoding
/encode_audio
/encode_video
/extract_mvs
/filter_audio
/filtering_audio
/filtering_video
/http_multiclient
/hw_decode
/metadata
/muxing
/pc-uninstalled
/qsvdec
/remuxing
/resampling_audio
/scaling_video
/transcode_aac
/transcoding
/vaapi_encode
/vaapi_transcode

View File

@@ -1,64 +1,46 @@
EXAMPLES-$(CONFIG_AVIO_DIR_CMD_EXAMPLE) += avio_dir_cmd
EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE) += avio_reading
EXAMPLES-$(CONFIG_DECODE_AUDIO_EXAMPLE) += decode_audio
EXAMPLES-$(CONFIG_DECODE_VIDEO_EXAMPLE) += decode_video
EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding
EXAMPLES-$(CONFIG_ENCODE_AUDIO_EXAMPLE) += encode_audio
EXAMPLES-$(CONFIG_ENCODE_VIDEO_EXAMPLE) += encode_video
EXAMPLES-$(CONFIG_EXTRACT_MVS_EXAMPLE) += extract_mvs
EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE) += filter_audio
EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE) += filtering_audio
EXAMPLES-$(CONFIG_FILTERING_VIDEO_EXAMPLE) += filtering_video
EXAMPLES-$(CONFIG_HTTP_MULTICLIENT_EXAMPLE) += http_multiclient
EXAMPLES-$(CONFIG_HW_DECODE_EXAMPLE) += hw_decode
EXAMPLES-$(CONFIG_METADATA_EXAMPLE) += metadata
EXAMPLES-$(CONFIG_MUXING_EXAMPLE) += muxing
EXAMPLES-$(CONFIG_QSVDEC_EXAMPLE) += qsvdec
EXAMPLES-$(CONFIG_REMUXING_EXAMPLE) += remuxing
EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE) += resampling_audio
EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE) += scaling_video
EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE) += transcode_aac
EXAMPLES-$(CONFIG_TRANSCODING_EXAMPLE) += transcoding
EXAMPLES-$(CONFIG_VAAPI_ENCODE_EXAMPLE) += vaapi_encode
EXAMPLES-$(CONFIG_VAAPI_TRANSCODE_EXAMPLE) += vaapi_transcode
# use pkg-config for getting CFLAGS and LDLIBS
FFMPEG_LIBS= libavdevice \
libavformat \
libavfilter \
libavcodec \
libswresample \
libswscale \
libavutil \
EXAMPLES := $(EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)$(EXESUF))
EXAMPLES_G := $(EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)_g$(EXESUF))
ALL_EXAMPLES := $(EXAMPLES) $(EXAMPLES-:%=doc/examples/%$(PROGSSUF)$(EXESUF))
ALL_EXAMPLES_G := $(EXAMPLES_G) $(EXAMPLES-:%=doc/examples/%$(PROGSSUF)_g$(EXESUF))
PROGS += $(EXAMPLES)
CFLAGS += -Wall -g
CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
EXAMPLE_MAKEFILE := $(SRC_PATH)/doc/examples/Makefile
EXAMPLES_FILES := $(wildcard $(SRC_PATH)/doc/examples/*.c) $(SRC_PATH)/doc/examples/README $(EXAMPLE_MAKEFILE)
EXAMPLES= avio_dir_cmd \
avio_reading \
decoding_encoding \
demuxing_decoding \
extract_mvs \
filtering_video \
filtering_audio \
http_multiclient \
metadata \
muxing \
remuxing \
resampling_audio \
scaling_video \
transcode_aac \
transcoding \
$(foreach P,$(EXAMPLES),$(eval OBJS-$(P:%$(PROGSSUF)$(EXESUF)=%) = $(P:%$(PROGSSUF)$(EXESUF)=%).o))
$(EXAMPLES_G): %$(PROGSSUF)_g$(EXESUF): %.o
OBJS=$(addsuffix .o,$(EXAMPLES))
examples: $(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
$(EXAMPLES:%$(PROGSSUF)$(EXESUF)=%.o): | doc/examples
OBJDIRS += doc/examples
.phony: all clean-test clean
DOXY_INPUT += $(EXAMPLES:%$(PROGSSUF)$(EXESUF)=%.c)
all: $(OBJS) $(EXAMPLES)
install: install-examples
clean-test:
$(RM) test*.pgm test.h264 test.mp2 test.sw test.mpg
install-examples: $(EXAMPLES_FILES)
$(Q)mkdir -p "$(DATADIR)/examples"
$(INSTALL) -m 644 $(EXAMPLES_FILES) "$(DATADIR)/examples"
$(INSTALL) -m 644 $(EXAMPLE_MAKEFILE:%=%.example) "$(DATADIR)/examples/Makefile"
uninstall: uninstall-examples
uninstall-examples:
$(RM) -r "$(DATADIR)/examples"
examplesclean:
$(RM) $(ALL_EXAMPLES) $(ALL_EXAMPLES_G)
$(RM) $(CLEANSUFFIXES:%=doc/examples/%)
docclean:: examplesclean
-include $(wildcard $(EXAMPLES:%$(PROGSSUF)$(EXESUF)=%.d))
.PHONY: examples
clean: clean-test
$(RM) $(EXAMPLES) $(OBJS)

View File

@@ -1,50 +0,0 @@
# use pkg-config for getting CFLAGS and LDLIBS
FFMPEG_LIBS= libavdevice \
libavformat \
libavfilter \
libavcodec \
libswresample \
libswscale \
libavutil \
CFLAGS += -Wall -g
CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
EXAMPLES= avio_dir_cmd \
avio_reading \
decode_audio \
decode_video \
demuxing_decoding \
encode_audio \
encode_video \
extract_mvs \
filtering_video \
filtering_audio \
http_multiclient \
hw_decode \
metadata \
muxing \
remuxing \
resampling_audio \
scaling_video \
transcode_aac \
transcoding \
OBJS=$(addsuffix .o,$(EXAMPLES))
# the following examples make explicit use of the math library
avcodec: LDLIBS += -lm
encode_audio: LDLIBS += -lm
muxing: LDLIBS += -lm
resampling_audio: LDLIBS += -lm
.phony: all clean-test clean
all: $(OBJS) $(EXAMPLES)
clean-test:
$(RM) test*.pgm test.h264 test.mp2 test.sw test.mpg
clean: clean-test
$(RM) $(EXAMPLES) $(OBJS)

View File

@@ -143,6 +143,8 @@ int main(int argc, char *argv[])
return 1;
}
/* register codecs and formats and other lavf/lavc components*/
av_register_all();
avformat_network_init();
op = argv[1];

View File

@@ -44,8 +44,6 @@ static int read_packet(void *opaque, uint8_t *buf, int buf_size)
struct buffer_data *bd = (struct buffer_data *)opaque;
buf_size = FFMIN(buf_size, bd->size);
if (!buf_size)
return AVERROR_EOF;
printf("ptr:%p size:%zu\n", bd->ptr, bd->size);
/* copy internal buffer data to buf */
@@ -74,6 +72,9 @@ int main(int argc, char *argv[])
}
input_filename = argv[1];
/* register codecs and formats and other lavf/lavc components*/
av_register_all();
/* slurp file content into buffer */
ret = av_file_map(input_filename, &buffer, &buffer_size, 0, NULL);
if (ret < 0)

View File

@@ -1,184 +0,0 @@
/*
* Copyright (c) 2001 Fabrice Bellard
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* audio decoding with libavcodec API example
*
* @example decode_audio.c
*/
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <libavutil/frame.h>
#include <libavutil/mem.h>
#include <libavcodec/avcodec.h>
#define AUDIO_INBUF_SIZE 20480
#define AUDIO_REFILL_THRESH 4096
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame,
FILE *outfile)
{
int i, ch;
int ret, data_size;
/* send the packet with the compressed data to the decoder */
ret = avcodec_send_packet(dec_ctx, pkt);
if (ret < 0) {
fprintf(stderr, "Error submitting the packet to the decoder\n");
exit(1);
}
/* read all the output frames (in general there may be any number of them */
while (ret >= 0) {
ret = avcodec_receive_frame(dec_ctx, frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
return;
else if (ret < 0) {
fprintf(stderr, "Error during decoding\n");
exit(1);
}
data_size = av_get_bytes_per_sample(dec_ctx->sample_fmt);
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 < frame->nb_samples; i++)
for (ch = 0; ch < dec_ctx->channels; ch++)
fwrite(frame->data[ch] + data_size*i, 1, data_size, outfile);
}
}
int main(int argc, char **argv)
{
const char *outfilename, *filename;
const AVCodec *codec;
AVCodecContext *c= NULL;
AVCodecParserContext *parser = NULL;
int len, ret;
FILE *f, *outfile;
uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
uint8_t *data;
size_t data_size;
AVPacket *pkt;
AVFrame *decoded_frame = NULL;
if (argc <= 2) {
fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
exit(0);
}
filename = argv[1];
outfilename = argv[2];
pkt = av_packet_alloc();
/* find the MPEG audio decoder */
codec = avcodec_find_decoder(AV_CODEC_ID_MP2);
if (!codec) {
fprintf(stderr, "Codec not found\n");
exit(1);
}
parser = av_parser_init(codec->id);
if (!parser) {
fprintf(stderr, "Parser not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate audio codec context\n");
exit(1);
}
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
f = fopen(filename, "rb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
exit(1);
}
outfile = fopen(outfilename, "wb");
if (!outfile) {
av_free(c);
exit(1);
}
/* decode until eof */
data = inbuf;
data_size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
while (data_size > 0) {
if (!decoded_frame) {
if (!(decoded_frame = av_frame_alloc())) {
fprintf(stderr, "Could not allocate audio frame\n");
exit(1);
}
}
ret = av_parser_parse2(parser, c, &pkt->data, &pkt->size,
data, data_size,
AV_NOPTS_VALUE, AV_NOPTS_VALUE, 0);
if (ret < 0) {
fprintf(stderr, "Error while parsing\n");
exit(1);
}
data += ret;
data_size -= ret;
if (pkt->size)
decode(c, pkt, decoded_frame, outfile);
if (data_size < AUDIO_REFILL_THRESH) {
memmove(inbuf, data, data_size);
data = inbuf;
len = fread(data + data_size, 1,
AUDIO_INBUF_SIZE - data_size, f);
if (len > 0)
data_size += len;
}
}
/* flush the decoder */
pkt->data = NULL;
pkt->size = 0;
decode(c, pkt, decoded_frame, outfile);
fclose(outfile);
fclose(f);
avcodec_free_context(&c);
av_parser_close(parser);
av_frame_free(&decoded_frame);
av_packet_free(&pkt);
return 0;
}

View File

@@ -1,186 +0,0 @@
/*
* Copyright (c) 2001 Fabrice Bellard
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* video decoding with libavcodec API example
*
* @example decode_video.c
*/
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <libavcodec/avcodec.h>
#define INBUF_SIZE 4096
static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
char *filename)
{
FILE *f;
int i;
f = fopen(filename,"w");
fprintf(f, "P5\n%d %d\n%d\n", xsize, ysize, 255);
for (i = 0; i < ysize; i++)
fwrite(buf + i * wrap, 1, xsize, f);
fclose(f);
}
static void decode(AVCodecContext *dec_ctx, AVFrame *frame, AVPacket *pkt,
const char *filename)
{
char buf[1024];
int ret;
ret = avcodec_send_packet(dec_ctx, pkt);
if (ret < 0) {
fprintf(stderr, "Error sending a packet for decoding\n");
exit(1);
}
while (ret >= 0) {
ret = avcodec_receive_frame(dec_ctx, frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
return;
else if (ret < 0) {
fprintf(stderr, "Error during decoding\n");
exit(1);
}
printf("saving frame %3d\n", dec_ctx->frame_number);
fflush(stdout);
/* the picture is allocated by the decoder. no need to
free it */
snprintf(buf, sizeof(buf), "%s-%d", filename, dec_ctx->frame_number);
pgm_save(frame->data[0], frame->linesize[0],
frame->width, frame->height, buf);
}
}
int main(int argc, char **argv)
{
const char *filename, *outfilename;
const AVCodec *codec;
AVCodecParserContext *parser;
AVCodecContext *c= NULL;
FILE *f;
AVFrame *frame;
uint8_t inbuf[INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
uint8_t *data;
size_t data_size;
int ret;
AVPacket *pkt;
if (argc <= 2) {
fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
exit(0);
}
filename = argv[1];
outfilename = argv[2];
pkt = av_packet_alloc();
if (!pkt)
exit(1);
/* 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);
/* find the MPEG-1 video decoder */
codec = avcodec_find_decoder(AV_CODEC_ID_MPEG1VIDEO);
if (!codec) {
fprintf(stderr, "Codec not found\n");
exit(1);
}
parser = av_parser_init(codec->id);
if (!parser) {
fprintf(stderr, "parser not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate video codec context\n");
exit(1);
}
/* For some codecs, such as msmpeg4 and mpeg4, width and height
MUST be initialized there because this information is not
available in the bitstream. */
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
f = fopen(filename, "rb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
exit(1);
}
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate video frame\n");
exit(1);
}
while (!feof(f)) {
/* read raw data from the input file */
data_size = fread(inbuf, 1, INBUF_SIZE, f);
if (!data_size)
break;
/* use the parser to split the data into frames */
data = inbuf;
while (data_size > 0) {
ret = av_parser_parse2(parser, c, &pkt->data, &pkt->size,
data, data_size, AV_NOPTS_VALUE, AV_NOPTS_VALUE, 0);
if (ret < 0) {
fprintf(stderr, "Error while parsing\n");
exit(1);
}
data += ret;
data_size -= ret;
if (pkt->size)
decode(c, frame, pkt, outfilename);
}
}
/* flush the decoder */
decode(c, frame, NULL, outfilename);
fclose(f);
av_parser_close(parser);
avcodec_free_context(&c);
av_frame_free(&frame);
av_packet_free(&pkt);
return 0;
}

View File

@@ -0,0 +1,665 @@
/*
* Copyright (c) 2001 Fabrice Bellard
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* libavcodec API use example.
*
* @example decoding_encoding.c
* Note that libavcodec only handles codecs (mpeg, mpeg4, etc...),
* not file formats (avi, vob, mp4, mov, mkv, mxf, flv, mpegts, mpegps, etc...). See library 'libavformat' for the
* format handling
*/
#include <math.h>
#include <libavutil/opt.h>
#include <libavcodec/avcodec.h>
#include <libavutil/channel_layout.h>
#include <libavutil/common.h>
#include <libavutil/imgutils.h>
#include <libavutil/mathematics.h>
#include <libavutil/samplefmt.h>
#define INBUF_SIZE 4096
#define AUDIO_INBUF_SIZE 20480
#define AUDIO_REFILL_THRESH 4096
/* check that a given sample format is supported by the encoder */
static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt)
{
const enum AVSampleFormat *p = codec->sample_fmts;
while (*p != AV_SAMPLE_FMT_NONE) {
if (*p == sample_fmt)
return 1;
p++;
}
return 0;
}
/* just pick the highest supported samplerate */
static int select_sample_rate(AVCodec *codec)
{
const int *p;
int best_samplerate = 0;
if (!codec->supported_samplerates)
return 44100;
p = codec->supported_samplerates;
while (*p) {
best_samplerate = FFMAX(*p, best_samplerate);
p++;
}
return best_samplerate;
}
/* select layout with the highest channel count */
static int select_channel_layout(AVCodec *codec)
{
const uint64_t *p;
uint64_t best_ch_layout = 0;
int best_nb_channels = 0;
if (!codec->channel_layouts)
return AV_CH_LAYOUT_STEREO;
p = codec->channel_layouts;
while (*p) {
int nb_channels = av_get_channel_layout_nb_channels(*p);
if (nb_channels > best_nb_channels) {
best_ch_layout = *p;
best_nb_channels = nb_channels;
}
p++;
}
return best_ch_layout;
}
/*
* Audio encoding example
*/
static void audio_encode_example(const char *filename)
{
AVCodec *codec;
AVCodecContext *c= NULL;
AVFrame *frame;
AVPacket pkt;
int i, j, k, ret, got_output;
int buffer_size;
FILE *f;
uint16_t *samples;
float t, tincr;
printf("Encode audio file %s\n", filename);
/* find the MP2 encoder */
codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
if (!codec) {
fprintf(stderr, "Codec not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate audio codec context\n");
exit(1);
}
/* put sample parameters */
c->bit_rate = 64000;
/* check that the encoder supports s16 pcm input */
c->sample_fmt = AV_SAMPLE_FMT_S16;
if (!check_sample_fmt(codec, c->sample_fmt)) {
fprintf(stderr, "Encoder does not support sample format %s",
av_get_sample_fmt_name(c->sample_fmt));
exit(1);
}
/* select other audio parameters supported by the encoder */
c->sample_rate = select_sample_rate(codec);
c->channel_layout = select_channel_layout(codec);
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
f = fopen(filename, "wb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
exit(1);
}
/* frame containing input raw audio */
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate audio frame\n");
exit(1);
}
frame->nb_samples = c->frame_size;
frame->format = c->sample_fmt;
frame->channel_layout = c->channel_layout;
/* the codec gives us the frame size, in samples,
* we calculate the size of the samples buffer in bytes */
buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size,
c->sample_fmt, 0);
if (buffer_size < 0) {
fprintf(stderr, "Could not get sample buffer size\n");
exit(1);
}
samples = av_malloc(buffer_size);
if (!samples) {
fprintf(stderr, "Could not allocate %d bytes for samples buffer\n",
buffer_size);
exit(1);
}
/* setup the data pointers in the AVFrame */
ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
(const uint8_t*)samples, buffer_size, 0);
if (ret < 0) {
fprintf(stderr, "Could not setup audio frame\n");
exit(1);
}
/* encode a single tone sound */
t = 0;
tincr = 2 * M_PI * 440.0 / c->sample_rate;
for (i = 0; i < 200; i++) {
av_init_packet(&pkt);
pkt.data = NULL; // packet data will be allocated by the encoder
pkt.size = 0;
for (j = 0; j < c->frame_size; j++) {
samples[2*j] = (int)(sin(t) * 10000);
for (k = 1; k < c->channels; k++)
samples[2*j + k] = samples[2*j];
t += tincr;
}
/* encode the samples */
ret = avcodec_encode_audio2(c, &pkt, frame, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding audio frame\n");
exit(1);
}
if (got_output) {
fwrite(pkt.data, 1, pkt.size, f);
av_packet_unref(&pkt);
}
}
/* get the delayed frames */
for (got_output = 1; got_output; i++) {
ret = avcodec_encode_audio2(c, &pkt, NULL, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding frame\n");
exit(1);
}
if (got_output) {
fwrite(pkt.data, 1, pkt.size, f);
av_packet_unref(&pkt);
}
}
fclose(f);
av_freep(&samples);
av_frame_free(&frame);
avcodec_close(c);
av_free(c);
}
/*
* Audio decoding.
*/
static void audio_decode_example(const char *outfilename, const char *filename)
{
AVCodec *codec;
AVCodecContext *c= NULL;
int len;
FILE *f, *outfile;
uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
AVPacket avpkt;
AVFrame *decoded_frame = NULL;
av_init_packet(&avpkt);
printf("Decode audio file %s to %s\n", filename, outfilename);
/* find the mpeg audio decoder */
codec = avcodec_find_decoder(AV_CODEC_ID_MP2);
if (!codec) {
fprintf(stderr, "Codec not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate audio codec context\n");
exit(1);
}
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
f = fopen(filename, "rb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
exit(1);
}
outfile = fopen(outfilename, "wb");
if (!outfile) {
av_free(c);
exit(1);
}
/* decode until eof */
avpkt.data = inbuf;
avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
while (avpkt.size > 0) {
int i, ch;
int got_frame = 0;
if (!decoded_frame) {
if (!(decoded_frame = av_frame_alloc())) {
fprintf(stderr, "Could not allocate audio frame\n");
exit(1);
}
}
len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
if (len < 0) {
fprintf(stderr, "Error while decoding\n");
exit(1);
}
if (got_frame) {
/* if a frame has been decoded, output it */
int data_size = av_get_bytes_per_sample(c->sample_fmt);
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);
}
avpkt.size -= len;
avpkt.data += len;
avpkt.dts =
avpkt.pts = AV_NOPTS_VALUE;
if (avpkt.size < AUDIO_REFILL_THRESH) {
/* Refill the input buffer, to avoid trying to decode
* incomplete frames. Instead of this, one could also use
* a parser, or use a proper container format through
* libavformat. */
memmove(inbuf, avpkt.data, avpkt.size);
avpkt.data = inbuf;
len = fread(avpkt.data + avpkt.size, 1,
AUDIO_INBUF_SIZE - avpkt.size, f);
if (len > 0)
avpkt.size += len;
}
}
fclose(outfile);
fclose(f);
avcodec_close(c);
av_free(c);
av_frame_free(&decoded_frame);
}
/*
* Video encoding example
*/
static void video_encode_example(const char *filename, int codec_id)
{
AVCodec *codec;
AVCodecContext *c= NULL;
int i, ret, x, y, got_output;
FILE *f;
AVFrame *frame;
AVPacket pkt;
uint8_t endcode[] = { 0, 0, 1, 0xb7 };
printf("Encode video file %s\n", filename);
/* find the mpeg1 video encoder */
codec = avcodec_find_encoder(codec_id);
if (!codec) {
fprintf(stderr, "Codec not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate video codec context\n");
exit(1);
}
/* put sample parameters */
c->bit_rate = 400000;
/* resolution must be a multiple of two */
c->width = 352;
c->height = 288;
/* frames per second */
c->time_base = (AVRational){1,25};
/* emit one intra frame every ten frames
* check frame pict_type before passing frame
* to encoder, if frame->pict_type is AV_PICTURE_TYPE_I
* then gop_size is ignored and the output of encoder
* will always be I frame irrespective to gop_size
*/
c->gop_size = 10;
c->max_b_frames = 1;
c->pix_fmt = AV_PIX_FMT_YUV420P;
if (codec_id == AV_CODEC_ID_H264)
av_opt_set(c->priv_data, "preset", "slow", 0);
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
f = fopen(filename, "wb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
exit(1);
}
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate video frame\n");
exit(1);
}
frame->format = c->pix_fmt;
frame->width = c->width;
frame->height = c->height;
/* the image can be allocated by any means and av_image_alloc() is
* just the most convenient way if av_malloc() is to be used */
ret = av_image_alloc(frame->data, frame->linesize, c->width, c->height,
c->pix_fmt, 32);
if (ret < 0) {
fprintf(stderr, "Could not allocate raw picture buffer\n");
exit(1);
}
/* encode 1 second of video */
for (i = 0; i < 25; i++) {
av_init_packet(&pkt);
pkt.data = NULL; // packet data will be allocated by the encoder
pkt.size = 0;
fflush(stdout);
/* prepare a dummy image */
/* Y */
for (y = 0; y < c->height; y++) {
for (x = 0; x < c->width; x++) {
frame->data[0][y * frame->linesize[0] + x] = x + y + i * 3;
}
}
/* Cb and Cr */
for (y = 0; y < c->height/2; y++) {
for (x = 0; x < c->width/2; x++) {
frame->data[1][y * frame->linesize[1] + x] = 128 + y + i * 2;
frame->data[2][y * frame->linesize[2] + x] = 64 + x + i * 5;
}
}
frame->pts = i;
/* encode the image */
ret = avcodec_encode_video2(c, &pkt, frame, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding frame\n");
exit(1);
}
if (got_output) {
printf("Write frame %3d (size=%5d)\n", i, pkt.size);
fwrite(pkt.data, 1, pkt.size, f);
av_packet_unref(&pkt);
}
}
/* get the delayed frames */
for (got_output = 1; got_output; i++) {
fflush(stdout);
ret = avcodec_encode_video2(c, &pkt, NULL, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding frame\n");
exit(1);
}
if (got_output) {
printf("Write frame %3d (size=%5d)\n", i, pkt.size);
fwrite(pkt.data, 1, pkt.size, f);
av_packet_unref(&pkt);
}
}
/* add sequence end code to have a real mpeg file */
fwrite(endcode, 1, sizeof(endcode), f);
fclose(f);
avcodec_close(c);
av_free(c);
av_freep(&frame->data[0]);
av_frame_free(&frame);
printf("\n");
}
/*
* Video decoding example
*/
static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
char *filename)
{
FILE *f;
int i;
f = fopen(filename,"w");
fprintf(f, "P5\n%d %d\n%d\n", xsize, ysize, 255);
for (i = 0; i < ysize; i++)
fwrite(buf + i * wrap, 1, xsize, f);
fclose(f);
}
static int decode_write_frame(const char *outfilename, AVCodecContext *avctx,
AVFrame *frame, int *frame_count, AVPacket *pkt, int last)
{
int len, got_frame;
char buf[1024];
len = avcodec_decode_video2(avctx, frame, &got_frame, pkt);
if (len < 0) {
fprintf(stderr, "Error while decoding frame %d\n", *frame_count);
return len;
}
if (got_frame) {
printf("Saving %sframe %3d\n", last ? "last " : "", *frame_count);
fflush(stdout);
/* the picture is allocated by the decoder, no need to free it */
snprintf(buf, sizeof(buf), outfilename, *frame_count);
pgm_save(frame->data[0], frame->linesize[0],
frame->width, frame->height, buf);
(*frame_count)++;
}
if (pkt->data) {
pkt->size -= len;
pkt->data += len;
}
return 0;
}
static void video_decode_example(const char *outfilename, const char *filename)
{
AVCodec *codec;
AVCodecContext *c= NULL;
int frame_count;
FILE *f;
AVFrame *frame;
uint8_t inbuf[INBUF_SIZE + AV_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);
printf("Decode video file %s to %s\n", filename, outfilename);
/* find the mpeg1 video decoder */
codec = avcodec_find_decoder(AV_CODEC_ID_MPEG1VIDEO);
if (!codec) {
fprintf(stderr, "Codec not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate video codec context\n");
exit(1);
}
if (codec->capabilities & AV_CODEC_CAP_TRUNCATED)
c->flags |= AV_CODEC_FLAG_TRUNCATED; // we do not send complete frames
/* For some codecs, such as msmpeg4 and mpeg4, width and height
MUST be initialized there because this information is not
available in the bitstream. */
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
f = fopen(filename, "rb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
exit(1);
}
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate video frame\n");
exit(1);
}
frame_count = 0;
for (;;) {
avpkt.size = fread(inbuf, 1, INBUF_SIZE, f);
if (avpkt.size == 0)
break;
/* NOTE1: some codecs are stream based (mpegvideo, mpegaudio)
and this is the only method to use them because you cannot
know the compressed data size before analysing it.
BUT some other codecs (msmpeg4, mpeg4) are inherently frame
based, so you must call them with all the data for one
frame exactly. You must also initialize 'width' and
'height' before initializing them. */
/* NOTE2: some codecs allow the raw parameters (frame size,
sample rate) to be changed at any frame. We handle this, so
you should also take care of it */
/* here, we use a stream based decoder (mpeg1video), so we
feed decoder and see if it could decode a frame */
avpkt.data = inbuf;
while (avpkt.size > 0)
if (decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 0) < 0)
exit(1);
}
/* some codecs, such as MPEG, transmit the I and P frame with a
latency of one frame. You must do the following to have a
chance to get the last frame of the video */
avpkt.data = NULL;
avpkt.size = 0;
decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 1);
fclose(f);
avcodec_close(c);
av_free(c);
av_frame_free(&frame);
printf("\n");
}
int main(int argc, char **argv)
{
const char *output_type;
/* register all the codecs */
avcodec_register_all();
if (argc < 2) {
printf("usage: %s output_type\n"
"API example program to decode/encode a media stream with libavcodec.\n"
"This program generates a synthetic stream and encodes it to a file\n"
"named test.h264, test.mp2 or test.mpg depending on output_type.\n"
"The encoded stream is then decoded and written to a raw data output.\n"
"output_type must be chosen between 'h264', 'mp2', 'mpg'.\n",
argv[0]);
return 1;
}
output_type = argv[1];
if (!strcmp(output_type, "h264")) {
video_encode_example("test.h264", AV_CODEC_ID_H264);
} else if (!strcmp(output_type, "mp2")) {
audio_encode_example("test.mp2");
audio_decode_example("test.pcm", "test.mp2");
} else if (!strcmp(output_type, "mpg")) {
video_encode_example("test.mpg", AV_CODEC_ID_MPEG1VIDEO);
video_decode_example("test%02d.pgm", "test.mpg");
} else {
fprintf(stderr, "Invalid output type '%s', choose between 'h264', 'mp2', or 'mpg'\n",
output_type);
return 1;
}
return 0;
}

View File

@@ -93,9 +93,10 @@ static int decode_packet(int *got_frame, int cached)
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 */
@@ -147,10 +148,11 @@ static int decode_packet(int *got_frame, int cached)
}
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;
AVStream *st;
AVCodecContext *dec_ctx = NULL;
AVCodec *dec = NULL;
AVDictionary *opts = NULL;
@@ -164,31 +166,17 @@ static int open_codec_context(int *stream_idx,
st = fmt_ctx->streams[stream_index];
/* 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 ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) {
fprintf(stderr, "Failed to open %s codec\n",
av_get_media_type_string(type));
return ret;
@@ -252,6 +240,9 @@ int main (int argc, char **argv)
video_dst_filename = argv[2];
audio_dst_filename = argv[3];
/* register all formats and codecs */
av_register_all();
/* open input file, and allocate format context */
if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
fprintf(stderr, "Could not open source file %s\n", src_filename);
@@ -264,8 +255,9 @@ 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) {
@@ -287,8 +279,9 @@ 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);
@@ -376,8 +369,8 @@ 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);

View File

@@ -1,238 +0,0 @@
/*
* Copyright (c) 2001 Fabrice Bellard
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* audio encoding with libavcodec API example.
*
* @example encode_audio.c
*/
#include <stdint.h>
#include <stdio.h>
#include <stdlib.h>
#include <libavcodec/avcodec.h>
#include <libavutil/channel_layout.h>
#include <libavutil/common.h>
#include <libavutil/frame.h>
#include <libavutil/samplefmt.h>
/* check that a given sample format is supported by the encoder */
static int check_sample_fmt(const AVCodec *codec, enum AVSampleFormat sample_fmt)
{
const enum AVSampleFormat *p = codec->sample_fmts;
while (*p != AV_SAMPLE_FMT_NONE) {
if (*p == sample_fmt)
return 1;
p++;
}
return 0;
}
/* just pick the highest supported samplerate */
static int select_sample_rate(const AVCodec *codec)
{
const int *p;
int best_samplerate = 0;
if (!codec->supported_samplerates)
return 44100;
p = codec->supported_samplerates;
while (*p) {
if (!best_samplerate || abs(44100 - *p) < abs(44100 - best_samplerate))
best_samplerate = *p;
p++;
}
return best_samplerate;
}
/* select layout with the highest channel count */
static int select_channel_layout(const AVCodec *codec)
{
const uint64_t *p;
uint64_t best_ch_layout = 0;
int best_nb_channels = 0;
if (!codec->channel_layouts)
return AV_CH_LAYOUT_STEREO;
p = codec->channel_layouts;
while (*p) {
int nb_channels = av_get_channel_layout_nb_channels(*p);
if (nb_channels > best_nb_channels) {
best_ch_layout = *p;
best_nb_channels = nb_channels;
}
p++;
}
return best_ch_layout;
}
static void encode(AVCodecContext *ctx, AVFrame *frame, AVPacket *pkt,
FILE *output)
{
int ret;
/* send the frame for encoding */
ret = avcodec_send_frame(ctx, frame);
if (ret < 0) {
fprintf(stderr, "Error sending the frame to the encoder\n");
exit(1);
}
/* read all the available output packets (in general there may be any
* number of them */
while (ret >= 0) {
ret = avcodec_receive_packet(ctx, pkt);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
return;
else if (ret < 0) {
fprintf(stderr, "Error encoding audio frame\n");
exit(1);
}
fwrite(pkt->data, 1, pkt->size, output);
av_packet_unref(pkt);
}
}
int main(int argc, char **argv)
{
const char *filename;
const AVCodec *codec;
AVCodecContext *c= NULL;
AVFrame *frame;
AVPacket *pkt;
int i, j, k, ret;
FILE *f;
uint16_t *samples;
float t, tincr;
if (argc <= 1) {
fprintf(stderr, "Usage: %s <output file>\n", argv[0]);
return 0;
}
filename = argv[1];
/* find the MP2 encoder */
codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
if (!codec) {
fprintf(stderr, "Codec not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate audio codec context\n");
exit(1);
}
/* put sample parameters */
c->bit_rate = 64000;
/* check that the encoder supports s16 pcm input */
c->sample_fmt = AV_SAMPLE_FMT_S16;
if (!check_sample_fmt(codec, c->sample_fmt)) {
fprintf(stderr, "Encoder does not support sample format %s",
av_get_sample_fmt_name(c->sample_fmt));
exit(1);
}
/* select other audio parameters supported by the encoder */
c->sample_rate = select_sample_rate(codec);
c->channel_layout = select_channel_layout(codec);
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
f = fopen(filename, "wb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
exit(1);
}
/* packet for holding encoded output */
pkt = av_packet_alloc();
if (!pkt) {
fprintf(stderr, "could not allocate the packet\n");
exit(1);
}
/* frame containing input raw audio */
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate audio frame\n");
exit(1);
}
frame->nb_samples = c->frame_size;
frame->format = c->sample_fmt;
frame->channel_layout = c->channel_layout;
/* allocate the data buffers */
ret = av_frame_get_buffer(frame, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate audio data buffers\n");
exit(1);
}
/* encode a single tone sound */
t = 0;
tincr = 2 * M_PI * 440.0 / c->sample_rate;
for (i = 0; i < 200; i++) {
/* make sure the frame is writable -- makes a copy if the encoder
* kept a reference internally */
ret = av_frame_make_writable(frame);
if (ret < 0)
exit(1);
samples = (uint16_t*)frame->data[0];
for (j = 0; j < c->frame_size; j++) {
samples[2*j] = (int)(sin(t) * 10000);
for (k = 1; k < c->channels; k++)
samples[2*j + k] = samples[2*j];
t += tincr;
}
encode(c, frame, pkt, f);
}
/* flush the encoder */
encode(c, NULL, pkt, f);
fclose(f);
av_frame_free(&frame);
av_packet_free(&pkt);
avcodec_free_context(&c);
return 0;
}

View File

@@ -1,197 +0,0 @@
/*
* Copyright (c) 2001 Fabrice Bellard
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* video encoding with libavcodec API example
*
* @example encode_video.c
*/
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <libavcodec/avcodec.h>
#include <libavutil/opt.h>
#include <libavutil/imgutils.h>
static void encode(AVCodecContext *enc_ctx, AVFrame *frame, AVPacket *pkt,
FILE *outfile)
{
int ret;
/* send the frame to the encoder */
if (frame)
printf("Send frame %3"PRId64"\n", frame->pts);
ret = avcodec_send_frame(enc_ctx, frame);
if (ret < 0) {
fprintf(stderr, "Error sending a frame for encoding\n");
exit(1);
}
while (ret >= 0) {
ret = avcodec_receive_packet(enc_ctx, pkt);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
return;
else if (ret < 0) {
fprintf(stderr, "Error during encoding\n");
exit(1);
}
printf("Write packet %3"PRId64" (size=%5d)\n", pkt->pts, pkt->size);
fwrite(pkt->data, 1, pkt->size, outfile);
av_packet_unref(pkt);
}
}
int main(int argc, char **argv)
{
const char *filename, *codec_name;
const AVCodec *codec;
AVCodecContext *c= NULL;
int i, ret, x, y;
FILE *f;
AVFrame *frame;
AVPacket *pkt;
uint8_t endcode[] = { 0, 0, 1, 0xb7 };
if (argc <= 2) {
fprintf(stderr, "Usage: %s <output file> <codec name>\n", argv[0]);
exit(0);
}
filename = argv[1];
codec_name = argv[2];
/* find the mpeg1video encoder */
codec = avcodec_find_encoder_by_name(codec_name);
if (!codec) {
fprintf(stderr, "Codec '%s' not found\n", codec_name);
exit(1);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate video codec context\n");
exit(1);
}
pkt = av_packet_alloc();
if (!pkt)
exit(1);
/* put sample parameters */
c->bit_rate = 400000;
/* resolution must be a multiple of two */
c->width = 352;
c->height = 288;
/* frames per second */
c->time_base = (AVRational){1, 25};
c->framerate = (AVRational){25, 1};
/* emit one intra frame every ten frames
* check frame pict_type before passing frame
* to encoder, if frame->pict_type is AV_PICTURE_TYPE_I
* then gop_size is ignored and the output of encoder
* will always be I frame irrespective to gop_size
*/
c->gop_size = 10;
c->max_b_frames = 1;
c->pix_fmt = AV_PIX_FMT_YUV420P;
if (codec->id == AV_CODEC_ID_H264)
av_opt_set(c->priv_data, "preset", "slow", 0);
/* open it */
ret = avcodec_open2(c, codec, NULL);
if (ret < 0) {
fprintf(stderr, "Could not open codec: %s\n", av_err2str(ret));
exit(1);
}
f = fopen(filename, "wb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
exit(1);
}
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate video frame\n");
exit(1);
}
frame->format = c->pix_fmt;
frame->width = c->width;
frame->height = c->height;
ret = av_frame_get_buffer(frame, 32);
if (ret < 0) {
fprintf(stderr, "Could not allocate the video frame data\n");
exit(1);
}
/* encode 1 second of video */
for (i = 0; i < 25; i++) {
fflush(stdout);
/* make sure the frame data is writable */
ret = av_frame_make_writable(frame);
if (ret < 0)
exit(1);
/* prepare a dummy image */
/* Y */
for (y = 0; y < c->height; y++) {
for (x = 0; x < c->width; x++) {
frame->data[0][y * frame->linesize[0] + x] = x + y + i * 3;
}
}
/* Cb and Cr */
for (y = 0; y < c->height/2; y++) {
for (x = 0; x < c->width/2; x++) {
frame->data[1][y * frame->linesize[1] + x] = 128 + y + i * 2;
frame->data[2][y * frame->linesize[2] + x] = 64 + x + i * 5;
}
}
frame->pts = i;
/* encode the image */
encode(c, frame, pkt, f);
}
/* flush the encoder */
encode(c, NULL, pkt, f);
/* add sequence end code to have a real MPEG file */
fwrite(endcode, 1, sizeof(endcode), f);
fclose(f);
avcodec_free_context(&c);
av_frame_free(&frame);
av_packet_free(&pkt);
return 0;
}

View File

@@ -31,26 +31,23 @@ 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(const AVPacket *pkt)
static int decode_packet(int *got_frame, int cached)
{
int ret = avcodec_send_packet(video_dec_ctx, pkt);
if (ret < 0) {
fprintf(stderr, "Error while sending a packet to the decoder: %s\n", av_err2str(ret));
return ret;
}
int decoded = pkt.size;
while (ret >= 0) {
ret = avcodec_receive_frame(video_dec_ctx, frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
break;
} else if (ret < 0) {
fprintf(stderr, "Error while receiving a frame from the decoder: %s\n", av_err2str(ret));
*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 (ret >= 0) {
if (*got_frame) {
int i;
AVFrameSideData *sd;
@@ -61,19 +58,19 @@ static int decode_packet(const AVPacket *pkt)
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);
video_frame_count, mv->source,
mv->w, mv->h, mv->src_x, mv->src_y,
mv->dst_x, mv->dst_y, mv->flags);
}
}
av_frame_unref(frame);
}
}
return 0;
return decoded;
}
static int open_codec_context(AVFormatContext *fmt_ctx, enum AVMediaType type)
static int open_codec_context(int *stream_idx,
AVFormatContext *fmt_ctx, enum AVMediaType type)
{
int ret;
AVStream *st;
@@ -81,27 +78,24 @@ static int open_codec_context(AVFormatContext *fmt_ctx, enum AVMediaType type)
AVCodec *dec = NULL;
AVDictionary *opts = NULL;
ret = av_find_best_stream(fmt_ctx, type, -1, -1, &dec, 0);
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 {
int stream_idx = ret;
st = fmt_ctx->streams[stream_idx];
*stream_idx = ret;
st = fmt_ctx->streams[*stream_idx];
dec_ctx = avcodec_alloc_context3(dec);
if (!dec_ctx) {
fprintf(stderr, "Failed to allocate codec\n");
/* 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);
}
ret = avcodec_parameters_to_context(dec_ctx, st->codecpar);
if (ret < 0) {
fprintf(stderr, "Failed to copy codec parameters to codec context\n");
return ret;
}
/* Init the video decoder */
av_dict_set(&opts, "flags2", "+export_mvs", 0);
if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) {
@@ -109,10 +103,6 @@ static int open_codec_context(AVFormatContext *fmt_ctx, enum AVMediaType type)
av_get_media_type_string(type));
return ret;
}
video_stream_idx = stream_idx;
video_stream = fmt_ctx->streams[video_stream_idx];
video_dec_ctx = dec_ctx;
}
return 0;
@@ -120,8 +110,7 @@ static int open_codec_context(AVFormatContext *fmt_ctx, enum AVMediaType type)
int main(int argc, char **argv)
{
int ret = 0;
AVPacket pkt = { 0 };
int ret = 0, got_frame;
if (argc != 2) {
fprintf(stderr, "Usage: %s <video>\n", argv[0]);
@@ -129,6 +118,8 @@ int main(int argc, char **argv)
}
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);
@@ -139,7 +130,10 @@ int main(int argc, char **argv)
exit(1);
}
open_codec_context(fmt_ctx, AVMEDIA_TYPE_VIDEO);
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);
@@ -158,20 +152,33 @@ int main(int argc, char **argv)
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) {
if (pkt.stream_index == video_stream_idx)
ret = decode_packet(&pkt);
av_packet_unref(&pkt);
if (ret < 0)
break;
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 */
decode_packet(NULL);
pkt.data = NULL;
pkt.size = 0;
do {
decode_packet(&got_frame, 1);
} while (got_frame);
end:
avcodec_free_context(&video_dec_ctx);
avcodec_close(video_dec_ctx);
avformat_close_input(&fmt_ctx);
av_frame_free(&frame);
return ret < 0;

View File

@@ -64,13 +64,13 @@ static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
{
AVFilterGraph *filter_graph;
AVFilterContext *abuffer_ctx;
const AVFilter *abuffer;
AVFilter *abuffer;
AVFilterContext *volume_ctx;
const AVFilter *volume;
AVFilter *volume;
AVFilterContext *aformat_ctx;
const AVFilter *aformat;
AVFilter *aformat;
AVFilterContext *abuffersink_ctx;
const AVFilter *abuffersink;
AVFilter *abuffersink;
AVDictionary *options_dict = NULL;
uint8_t options_str[1024];
@@ -289,6 +289,8 @@ int main(int argc, char *argv[])
return 1;
}
avfilter_register_all();
/* Allocate the frame we will be using to store the data. */
frame = av_frame_alloc();
if (!frame) {

View File

@@ -32,6 +32,7 @@
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
@@ -64,16 +65,12 @@ 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;
/* create decoding context */
dec_ctx = avcodec_alloc_context3(dec);
if (!dec_ctx)
return AVERROR(ENOMEM);
avcodec_parameters_to_context(dec_ctx, fmt_ctx->streams[audio_stream_index]->codecpar);
dec_ctx = fmt_ctx->streams[audio_stream_index]->codec;
av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
/* init the audio decoder */
if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
@@ -88,8 +85,8 @@ static int init_filters(const char *filters_descr)
{
char args[512];
int ret = 0;
const AVFilter *abuffersrc = avfilter_get_by_name("abuffer");
const AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
AVFilter *abuffersrc = avfilter_get_by_name("abuffer");
AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
@@ -199,7 +196,7 @@ end:
static void print_frame(const AVFrame *frame)
{
const int n = frame->nb_samples * av_get_channel_layout_nb_channels(frame->channel_layout);
const int n = frame->nb_samples * av_get_channel_layout_nb_channels(av_frame_get_channel_layout(frame));
const uint16_t *p = (uint16_t*)frame->data[0];
const uint16_t *p_end = p + n;
@@ -214,9 +211,10 @@ static void print_frame(const AVFrame *frame)
int main(int argc, char **argv)
{
int ret;
AVPacket packet;
AVPacket packet0, packet;
AVFrame *frame = av_frame_alloc();
AVFrame *filt_frame = av_frame_alloc();
int got_frame;
if (!frame || !filt_frame) {
perror("Could not allocate frame");
@@ -227,58 +225,63 @@ int main(int argc, char **argv)
exit(1);
}
av_register_all();
avfilter_register_all();
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if ((ret = init_filters(filter_descr)) < 0)
goto end;
/* read all packets */
packet0.data = NULL;
packet.data = NULL;
while (1) {
if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
break;
if (!packet0.data) {
if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
break;
packet0 = packet;
}
if (packet.stream_index == audio_stream_index) {
ret = avcodec_send_packet(dec_ctx, &packet);
got_frame = 0;
ret = avcodec_decode_audio4(dec_ctx, frame, &got_frame, &packet);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while sending a packet to the decoder\n");
break;
av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n");
continue;
}
packet.size -= ret;
packet.data += ret;
while (ret >= 0) {
ret = avcodec_receive_frame(dec_ctx, frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
if (got_frame) {
/* push the audio data from decoded frame into the filtergraph */
if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, 0) < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
break;
} else if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while receiving a frame from the decoder\n");
goto end;
}
if (ret >= 0) {
/* push the audio data from decoded frame into the filtergraph */
if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
/* pull filtered audio from the filtergraph */
while (1) {
ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
}
/* pull filtered audio from the filtergraph */
while (1) {
ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
if (ret < 0)
goto end;
print_frame(filt_frame);
av_frame_unref(filt_frame);
}
av_frame_unref(frame);
if (ret < 0)
goto end;
print_frame(filt_frame);
av_frame_unref(filt_frame);
}
}
if (packet.size <= 0)
av_packet_unref(&packet0);
} else {
/* discard non-wanted packets */
av_packet_unref(&packet0);
}
av_packet_unref(&packet);
}
end:
avfilter_graph_free(&filter_graph);
avcodec_free_context(&dec_ctx);
avcodec_close(dec_ctx);
avformat_close_input(&fmt_ctx);
av_frame_free(&frame);
av_frame_free(&filt_frame);

View File

@@ -29,11 +29,10 @@
#define _XOPEN_SOURCE 600 /* for usleep */
#include <unistd.h>
#include <stdio.h>
#include <stdlib.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
@@ -73,12 +72,8 @@ static int open_input_file(const char *filename)
return ret;
}
video_stream_index = ret;
/* create decoding context */
dec_ctx = avcodec_alloc_context3(dec);
if (!dec_ctx)
return AVERROR(ENOMEM);
avcodec_parameters_to_context(dec_ctx, fmt_ctx->streams[video_stream_index]->codecpar);
dec_ctx = fmt_ctx->streams[video_stream_index]->codec;
av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
/* init the video decoder */
if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
@@ -93,8 +88,8 @@ static int init_filters(const char *filters_descr)
{
char args[512];
int ret = 0;
const AVFilter *buffersrc = avfilter_get_by_name("buffer");
const AVFilter *buffersink = avfilter_get_by_name("buffersink");
AVFilter *buffersrc = avfilter_get_by_name("buffer");
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;
@@ -211,20 +206,21 @@ int main(int argc, char **argv)
{
int ret;
AVPacket packet;
AVFrame *frame;
AVFrame *filt_frame;
AVFrame *frame = av_frame_alloc();
AVFrame *filt_frame = av_frame_alloc();
int got_frame;
if (!frame || !filt_frame) {
perror("Could not allocate frame");
exit(1);
}
if (argc != 2) {
fprintf(stderr, "Usage: %s file\n", argv[0]);
exit(1);
}
frame = av_frame_alloc();
filt_frame = av_frame_alloc();
if (!frame || !filt_frame) {
perror("Could not allocate frame");
exit(1);
}
av_register_all();
avfilter_register_all();
if ((ret = open_input_file(argv[1])) < 0)
goto end;
@@ -237,22 +233,15 @@ int main(int argc, char **argv)
break;
if (packet.stream_index == video_stream_index) {
ret = avcodec_send_packet(dec_ctx, &packet);
got_frame = 0;
ret = avcodec_decode_video2(dec_ctx, frame, &got_frame, &packet);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while sending a packet to the decoder\n");
av_log(NULL, AV_LOG_ERROR, "Error decoding video\n");
break;
}
while (ret >= 0) {
ret = avcodec_receive_frame(dec_ctx, frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
break;
} else if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while receiving a frame from the decoder\n");
goto end;
}
frame->pts = frame->best_effort_timestamp;
if (got_frame) {
frame->pts = av_frame_get_best_effort_timestamp(frame);
/* push the decoded frame into the filtergraph */
if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
@@ -277,7 +266,7 @@ int main(int argc, char **argv)
}
end:
avfilter_graph_free(&filter_graph);
avcodec_free_context(&dec_ctx);
avcodec_close(dec_ctx);
avformat_close_input(&fmt_ctx);
av_frame_free(&frame);
av_frame_free(&filt_frame);

View File

@@ -33,19 +33,18 @@
#include <libavutil/opt.h>
#include <unistd.h>
static void process_client(AVIOContext *client, const char *in_uri)
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;
char *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;
av_freep(&resource);
}
if (ret < 0)
goto end;
@@ -94,16 +93,15 @@ end:
avio_close(client);
fprintf(stderr, "Closing input\n");
avio_close(input);
av_freep(&resource);
}
int main(int argc, char **argv)
{
av_log_set_level(AV_LOG_TRACE);
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"
@@ -114,6 +112,7 @@ int main(int argc, char **argv)
in_uri = argv[1];
out_uri = argv[2];
av_register_all();
avformat_network_init();
if ((ret = av_dict_set(&options, "listen", "2", 0)) < 0) {

View File

@@ -1,252 +0,0 @@
/*
* Copyright (c) 2017 Jun Zhao
* Copyright (c) 2017 Kaixuan Liu
*
* HW Acceleration API (video decoding) decode sample
*
* 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
* HW-Accelerated decoding example.
*
* @example hw_decode.c
* This example shows how to do HW-accelerated decoding with output
* frames from the HW video surfaces.
*/
#include <stdio.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavutil/pixdesc.h>
#include <libavutil/hwcontext.h>
#include <libavutil/opt.h>
#include <libavutil/avassert.h>
#include <libavutil/imgutils.h>
static AVBufferRef *hw_device_ctx = NULL;
static enum AVPixelFormat hw_pix_fmt;
static FILE *output_file = NULL;
static int hw_decoder_init(AVCodecContext *ctx, const enum AVHWDeviceType type)
{
int err = 0;
if ((err = av_hwdevice_ctx_create(&hw_device_ctx, type,
NULL, NULL, 0)) < 0) {
fprintf(stderr, "Failed to create specified HW device.\n");
return err;
}
ctx->hw_device_ctx = av_buffer_ref(hw_device_ctx);
return err;
}
static enum AVPixelFormat get_hw_format(AVCodecContext *ctx,
const enum AVPixelFormat *pix_fmts)
{
const enum AVPixelFormat *p;
for (p = pix_fmts; *p != -1; p++) {
if (*p == hw_pix_fmt)
return *p;
}
fprintf(stderr, "Failed to get HW surface format.\n");
return AV_PIX_FMT_NONE;
}
static int decode_write(AVCodecContext *avctx, AVPacket *packet)
{
AVFrame *frame = NULL, *sw_frame = NULL;
AVFrame *tmp_frame = NULL;
uint8_t *buffer = NULL;
int size;
int ret = 0;
ret = avcodec_send_packet(avctx, packet);
if (ret < 0) {
fprintf(stderr, "Error during decoding\n");
return ret;
}
while (1) {
if (!(frame = av_frame_alloc()) || !(sw_frame = av_frame_alloc())) {
fprintf(stderr, "Can not alloc frame\n");
ret = AVERROR(ENOMEM);
goto fail;
}
ret = avcodec_receive_frame(avctx, frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
av_frame_free(&frame);
av_frame_free(&sw_frame);
return 0;
} else if (ret < 0) {
fprintf(stderr, "Error while decoding\n");
goto fail;
}
if (frame->format == hw_pix_fmt) {
/* retrieve data from GPU to CPU */
if ((ret = av_hwframe_transfer_data(sw_frame, frame, 0)) < 0) {
fprintf(stderr, "Error transferring the data to system memory\n");
goto fail;
}
tmp_frame = sw_frame;
} else
tmp_frame = frame;
size = av_image_get_buffer_size(tmp_frame->format, tmp_frame->width,
tmp_frame->height, 1);
buffer = av_malloc(size);
if (!buffer) {
fprintf(stderr, "Can not alloc buffer\n");
ret = AVERROR(ENOMEM);
goto fail;
}
ret = av_image_copy_to_buffer(buffer, size,
(const uint8_t * const *)tmp_frame->data,
(const int *)tmp_frame->linesize, tmp_frame->format,
tmp_frame->width, tmp_frame->height, 1);
if (ret < 0) {
fprintf(stderr, "Can not copy image to buffer\n");
goto fail;
}
if ((ret = fwrite(buffer, 1, size, output_file)) < 0) {
fprintf(stderr, "Failed to dump raw data.\n");
goto fail;
}
fail:
av_frame_free(&frame);
av_frame_free(&sw_frame);
av_freep(&buffer);
if (ret < 0)
return ret;
}
}
int main(int argc, char *argv[])
{
AVFormatContext *input_ctx = NULL;
int video_stream, ret;
AVStream *video = NULL;
AVCodecContext *decoder_ctx = NULL;
AVCodec *decoder = NULL;
AVPacket packet;
enum AVHWDeviceType type;
int i;
if (argc < 4) {
fprintf(stderr, "Usage: %s <device type> <input file> <output file>\n", argv[0]);
return -1;
}
type = av_hwdevice_find_type_by_name(argv[1]);
if (type == AV_HWDEVICE_TYPE_NONE) {
fprintf(stderr, "Device type %s is not supported.\n", argv[1]);
fprintf(stderr, "Available device types:");
while((type = av_hwdevice_iterate_types(type)) != AV_HWDEVICE_TYPE_NONE)
fprintf(stderr, " %s", av_hwdevice_get_type_name(type));
fprintf(stderr, "\n");
return -1;
}
/* open the input file */
if (avformat_open_input(&input_ctx, argv[2], NULL, NULL) != 0) {
fprintf(stderr, "Cannot open input file '%s'\n", argv[2]);
return -1;
}
if (avformat_find_stream_info(input_ctx, NULL) < 0) {
fprintf(stderr, "Cannot find input stream information.\n");
return -1;
}
/* find the video stream information */
ret = av_find_best_stream(input_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, &decoder, 0);
if (ret < 0) {
fprintf(stderr, "Cannot find a video stream in the input file\n");
return -1;
}
video_stream = ret;
for (i = 0;; i++) {
const AVCodecHWConfig *config = avcodec_get_hw_config(decoder, i);
if (!config) {
fprintf(stderr, "Decoder %s does not support device type %s.\n",
decoder->name, av_hwdevice_get_type_name(type));
return -1;
}
if (config->methods & AV_CODEC_HW_CONFIG_METHOD_HW_DEVICE_CTX &&
config->device_type == type) {
hw_pix_fmt = config->pix_fmt;
break;
}
}
if (!(decoder_ctx = avcodec_alloc_context3(decoder)))
return AVERROR(ENOMEM);
video = input_ctx->streams[video_stream];
if (avcodec_parameters_to_context(decoder_ctx, video->codecpar) < 0)
return -1;
decoder_ctx->get_format = get_hw_format;
if (hw_decoder_init(decoder_ctx, type) < 0)
return -1;
if ((ret = avcodec_open2(decoder_ctx, decoder, NULL)) < 0) {
fprintf(stderr, "Failed to open codec for stream #%u\n", video_stream);
return -1;
}
/* open the file to dump raw data */
output_file = fopen(argv[3], "w+");
/* actual decoding and dump the raw data */
while (ret >= 0) {
if ((ret = av_read_frame(input_ctx, &packet)) < 0)
break;
if (video_stream == packet.stream_index)
ret = decode_write(decoder_ctx, &packet);
av_packet_unref(&packet);
}
/* flush the decoder */
packet.data = NULL;
packet.size = 0;
ret = decode_write(decoder_ctx, &packet);
av_packet_unref(&packet);
if (output_file)
fclose(output_file);
avcodec_free_context(&decoder_ctx);
avformat_close_input(&input_ctx);
av_buffer_unref(&hw_device_ctx);
return 0;
}

View File

@@ -44,6 +44,7 @@ int main (int argc, char **argv)
return 1;
}
av_register_all();
if ((ret = avformat_open_input(&fmt_ctx, argv[1], NULL, NULL)))
return ret;

View File

@@ -52,7 +52,6 @@
// 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;
@@ -105,18 +104,13 @@ 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:
@@ -161,7 +155,7 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
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) {
@@ -219,7 +213,7 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, A
int ret;
AVDictionary *opt = NULL;
c = ost->enc;
c = ost->st->codec;
/* open it */
av_dict_copy(&opt, opt_arg, 0);
@@ -246,13 +240,6 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, A
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");
exit(1);
}
/* create resampler context */
ost->swr_ctx = swr_alloc();
if (!ost->swr_ctx) {
@@ -284,13 +271,13 @@ static AVFrame *get_audio_frame(OutputStream *ost)
int16_t *q = (int16_t*)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++) {
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;
@@ -316,7 +303,7 @@ 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);
@@ -335,15 +322,15 @@ static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
if (ret < 0)
exit(1);
/* convert to destination format */
ret = swr_convert(ost->swr_ctx,
ost->frame->data, dst_nb_samples,
(const uint8_t **)frame->data, frame->nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
exit(1);
}
frame = ost->frame;
/* convert to destination format */
ret = swr_convert(ost->swr_ctx,
ost->frame->data, dst_nb_samples,
(const uint8_t **)frame->data, frame->nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
exit(1);
}
frame = ost->frame;
frame->pts = av_rescale_q(ost->samples_count, (AVRational){1, c->sample_rate}, c->time_base);
ost->samples_count += dst_nb_samples;
@@ -396,7 +383,7 @@ static AVFrame *alloc_picture(enum AVPixelFormat pix_fmt, int width, int height)
static void open_video(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
{
int ret;
AVCodecContext *c = ost->enc;
AVCodecContext *c = ost->st->codec;
AVDictionary *opt = NULL;
av_dict_copy(&opt, opt_arg, 0);
@@ -427,20 +414,21 @@ 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. */
static void fill_yuv_image(AVFrame *pict, int frame_index,
int width, int height)
{
int x, y, i;
int x, y, i, ret;
/* when we pass a frame to the encoder, it may keep a reference to it
* internally;
* make sure we do not overwrite it here
*/
ret = av_frame_make_writable(pict);
if (ret < 0)
exit(1);
i = frame_index;
@@ -460,18 +448,13 @@ 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;
/* when we pass a frame to the encoder, it may keep a reference to it
* internally; make sure we do not overwrite it here */
if (av_frame_make_writable(ost->frame) < 0)
exit(1);
if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
/* as we only generate a YUV420P picture, we must convert it
* to the codec pixel format if needed */
@@ -488,9 +471,9 @@ static AVFrame *get_video_frame(OutputStream *ost)
}
}
fill_yuv_image(ost->tmp_frame, ost->next_pts, c->width, c->height);
sws_scale(ost->sws_ctx, (const uint8_t * const *) ost->tmp_frame->data,
ost->tmp_frame->linesize, 0, c->height, ost->frame->data,
ost->frame->linesize);
sws_scale(ost->sws_ctx,
(const uint8_t * const *)ost->tmp_frame->data, ost->tmp_frame->linesize,
0, c->height, ost->frame->data, ost->frame->linesize);
} else {
fill_yuv_image(ost->frame, ost->next_pts, c->width, c->height);
}
@@ -512,7 +495,7 @@ static int write_video_frame(AVFormatContext *oc, OutputStream *ost)
int got_packet = 0;
AVPacket pkt = { 0 };
c = ost->enc;
c = ost->st->codec;
frame = get_video_frame(ost);
@@ -541,7 +524,7 @@ 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);
@@ -562,7 +545,9 @@ int main(int argc, char **argv)
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) {
printf("usage: %s output_file\n"
@@ -576,9 +561,8 @@ 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);
if (argc > 3 && !strcmp(argv[2], "-flags")) {
av_dict_set(&opt, argv[2]+1, argv[3], 0);
}
/* allocate the output media context */
@@ -636,8 +620,8 @@ 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);

View File

@@ -26,55 +26,185 @@
*
* @example qsvdec.c
* This example shows how to do QSV-accelerated H.264 decoding with output
* frames in the GPU video surfaces.
* 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/buffer.h"
#include "libavutil/error.h"
#include "libavutil/hwcontext.h"
#include "libavutil/hwcontext_qsv.h"
#include "libavutil/mem.h"
typedef struct DecodeContext {
AVBufferRef *hw_device_ref;
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) {
DecodeContext *decode = avctx->opaque;
AVHWFramesContext *frames_ctx;
AVQSVFramesContext *frames_hwctx;
int ret;
if (!avctx->hwaccel_context) {
DecodeContext *decode = avctx->opaque;
AVQSVContext *qsv = av_qsv_alloc_context();
if (!qsv)
return AV_PIX_FMT_NONE;
/* create a pool of surfaces to be used by the decoder */
avctx->hw_frames_ctx = av_hwframe_ctx_alloc(decode->hw_device_ref);
if (!avctx->hw_frames_ctx)
return AV_PIX_FMT_NONE;
frames_ctx = (AVHWFramesContext*)avctx->hw_frames_ctx->data;
frames_hwctx = frames_ctx->hwctx;
qsv->session = decode->mfx_session;
qsv->iopattern = MFX_IOPATTERN_OUT_VIDEO_MEMORY;
frames_ctx->format = AV_PIX_FMT_QSV;
frames_ctx->sw_format = avctx->sw_pix_fmt;
frames_ctx->width = FFALIGN(avctx->coded_width, 32);
frames_ctx->height = FFALIGN(avctx->coded_height, 32);
frames_ctx->initial_pool_size = 32;
frames_hwctx->frame_type = MFX_MEMTYPE_VIDEO_MEMORY_DECODER_TARGET;
ret = av_hwframe_ctx_init(avctx->hw_frames_ctx);
if (ret < 0)
return AV_PIX_FMT_NONE;
avctx->hwaccel_context = qsv;
}
return AV_PIX_FMT_QSV;
}
@@ -88,47 +218,86 @@ static int get_format(AVCodecContext *avctx, const enum AVPixelFormat *pix_fmts)
}
static int decode_packet(DecodeContext *decode, AVCodecContext *decoder_ctx,
AVFrame *frame, AVFrame *sw_frame,
AVPacket *pkt, AVIOContext *output_ctx)
AVFrame *frame, AVPacket *pkt,
AVIOContext *output_ctx)
{
int ret = 0;
int got_frame = 1;
ret = avcodec_send_packet(decoder_ctx, pkt);
if (ret < 0) {
fprintf(stderr, "Error during decoding\n");
return ret;
}
while (ret >= 0) {
int i, j;
ret = avcodec_receive_frame(decoder_ctx, frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
else if (ret < 0) {
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. */
ret = av_hwframe_transfer_data(sw_frame, frame, 0);
if (ret < 0) {
fprintf(stderr, "Error transferring the data to system memory\n");
goto fail;
}
if (got_frame) {
mfxFrameSurface1 *surf = (mfxFrameSurface1*)frame->data[3];
VASurfaceID surface = *(VASurfaceID*)surf->Data.MemId;
for (i = 0; i < FF_ARRAY_ELEMS(sw_frame->data) && sw_frame->data[i]; i++)
for (j = 0; j < (sw_frame->height >> (i > 0)); j++)
avio_write(output_ctx, sw_frame->data[i] + j * sw_frame->linesize[i], sw_frame->width);
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:
av_frame_unref(sw_frame);
av_frame_unref(frame);
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;
if (ret < 0)
return ret;
}
}
return 0;
@@ -142,13 +311,30 @@ int main(int argc, char **argv)
const AVCodec *decoder;
AVPacket pkt = { 0 };
AVFrame *frame = NULL, *sw_frame = NULL;
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;
int ret, i, err;
av_register_all();
if (argc < 3) {
fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
@@ -166,7 +352,7 @@ int main(int argc, char **argv)
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)
if (st->codec->codec_id == AV_CODEC_ID_H264 && !video_st)
video_st = st;
else
st->discard = AVDISCARD_ALL;
@@ -176,13 +362,34 @@ int main(int argc, char **argv)
goto finish;
}
/* open the hardware device */
ret = av_hwdevice_ctx_create(&decode.hw_device_ref, AV_HWDEVICE_TYPE_QSV,
"auto", NULL, 0);
if (ret < 0) {
fprintf(stderr, "Cannot open the hardware device\n");
/* 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");
@@ -197,19 +404,21 @@ int main(int argc, char **argv)
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 +
if (video_st->codec->extradata_size) {
decoder_ctx->extradata = av_mallocz(video_st->codec->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;
memcpy(decoder_ctx->extradata, video_st->codec->extradata,
video_st->codec->extradata_size);
decoder_ctx->extradata_size = video_st->codec->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);
@@ -225,9 +434,8 @@ int main(int argc, char **argv)
goto finish;
}
frame = av_frame_alloc();
sw_frame = av_frame_alloc();
if (!frame || !sw_frame) {
frame = av_frame_alloc();
if (!frame) {
ret = AVERROR(ENOMEM);
goto finish;
}
@@ -239,7 +447,7 @@ int main(int argc, char **argv)
break;
if (pkt.stream_index == video_st->index)
ret = decode_packet(&decode, decoder_ctx, frame, sw_frame, &pkt, output_ctx);
ret = decode_packet(&decode, decoder_ctx, frame, &pkt, output_ctx);
av_packet_unref(&pkt);
}
@@ -247,7 +455,7 @@ int main(int argc, char **argv)
/* flush the decoder */
pkt.data = NULL;
pkt.size = 0;
ret = decode_packet(&decode, decoder_ctx, frame, sw_frame, &pkt, output_ctx);
ret = decode_packet(&decode, decoder_ctx, frame, &pkt, output_ctx);
finish:
if (ret < 0) {
@@ -259,11 +467,19 @@ finish:
avformat_close_input(&input_ctx);
av_frame_free(&frame);
av_frame_free(&sw_frame);
if (decoder_ctx)
av_freep(&decoder_ctx->hwaccel_context);
avcodec_free_context(&decoder_ctx);
av_buffer_unref(&decode.hw_device_ref);
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);

View File

@@ -50,9 +50,6 @@ int main(int argc, char **argv)
AVPacket pkt;
const char *in_filename, *out_filename;
int ret, i;
int stream_index = 0;
int *stream_mapping = NULL;
int stream_mapping_size = 0;
if (argc < 3) {
printf("usage: %s input output\n"
@@ -65,6 +62,8 @@ int main(int argc, char **argv)
in_filename = argv[1];
out_filename = argv[2];
av_register_all();
if ((ret = avformat_open_input(&ifmt_ctx, in_filename, 0, 0)) < 0) {
fprintf(stderr, "Could not open input file '%s'", in_filename);
goto end;
@@ -84,42 +83,25 @@ int main(int argc, char **argv)
goto end;
}
stream_mapping_size = ifmt_ctx->nb_streams;
stream_mapping = av_mallocz_array(stream_mapping_size, sizeof(*stream_mapping));
if (!stream_mapping) {
ret = AVERROR(ENOMEM);
goto end;
}
ofmt = ofmt_ctx->oformat;
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
AVStream *out_stream;
AVStream *in_stream = ifmt_ctx->streams[i];
AVCodecParameters *in_codecpar = in_stream->codecpar;
if (in_codecpar->codec_type != AVMEDIA_TYPE_AUDIO &&
in_codecpar->codec_type != AVMEDIA_TYPE_VIDEO &&
in_codecpar->codec_type != AVMEDIA_TYPE_SUBTITLE) {
stream_mapping[i] = -1;
continue;
}
stream_mapping[i] = stream_index++;
out_stream = avformat_new_stream(ofmt_ctx, NULL);
AVStream *out_stream = avformat_new_stream(ofmt_ctx, in_stream->codec->codec);
if (!out_stream) {
fprintf(stderr, "Failed allocating output stream\n");
ret = AVERROR_UNKNOWN;
goto end;
}
ret = avcodec_parameters_copy(out_stream->codecpar, in_codecpar);
ret = avcodec_copy_context(out_stream->codec, in_stream->codec);
if (ret < 0) {
fprintf(stderr, "Failed to copy codec parameters\n");
fprintf(stderr, "Failed to copy context from input to output stream codec context\n");
goto end;
}
out_stream->codecpar->codec_tag = 0;
out_stream->codec->codec_tag = 0;
if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
out_stream->codec->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
}
av_dump_format(ofmt_ctx, 0, out_filename, 1);
@@ -145,14 +127,8 @@ int main(int argc, char **argv)
break;
in_stream = ifmt_ctx->streams[pkt.stream_index];
if (pkt.stream_index >= stream_mapping_size ||
stream_mapping[pkt.stream_index] < 0) {
av_packet_unref(&pkt);
continue;
}
pkt.stream_index = stream_mapping[pkt.stream_index];
out_stream = ofmt_ctx->streams[pkt.stream_index];
log_packet(ifmt_ctx, &pkt, "in");
/* copy packet */
@@ -180,8 +156,6 @@ end:
avio_closep(&ofmt_ctx->pb);
avformat_free_context(ofmt_ctx);
av_freep(&stream_mapping);
if (ret < 0 && ret != AVERROR_EOF) {
fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
return 1;

File diff suppressed because it is too large Load Diff

View File

@@ -30,6 +30,7 @@
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
@@ -44,12 +45,6 @@ typedef struct FilteringContext {
} FilteringContext;
static FilteringContext *filter_ctx;
typedef struct StreamContext {
AVCodecContext *dec_ctx;
AVCodecContext *enc_ctx;
} StreamContext;
static StreamContext *stream_ctx;
static int open_input_file(const char *filename)
{
int ret;
@@ -66,42 +61,22 @@ static int open_input_file(const char *filename)
return ret;
}
stream_ctx = av_mallocz_array(ifmt_ctx->nb_streams, sizeof(*stream_ctx));
if (!stream_ctx)
return AVERROR(ENOMEM);
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
AVStream *stream = ifmt_ctx->streams[i];
AVCodec *dec = avcodec_find_decoder(stream->codecpar->codec_id);
AVStream *stream;
AVCodecContext *codec_ctx;
if (!dec) {
av_log(NULL, AV_LOG_ERROR, "Failed to find decoder for stream #%u\n", i);
return AVERROR_DECODER_NOT_FOUND;
}
codec_ctx = avcodec_alloc_context3(dec);
if (!codec_ctx) {
av_log(NULL, AV_LOG_ERROR, "Failed to allocate the decoder context for stream #%u\n", i);
return AVERROR(ENOMEM);
}
ret = avcodec_parameters_to_context(codec_ctx, stream->codecpar);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Failed to copy decoder parameters to input decoder context "
"for stream #%u\n", i);
return ret;
}
stream = ifmt_ctx->streams[i];
codec_ctx = stream->codec;
/* Reencode video & audio and remux subtitles etc. */
if (codec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
|| codec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
if (codec_ctx->codec_type == AVMEDIA_TYPE_VIDEO)
codec_ctx->framerate = av_guess_frame_rate(ifmt_ctx, stream, NULL);
/* Open decoder */
ret = avcodec_open2(codec_ctx, dec, NULL);
ret = avcodec_open2(codec_ctx,
avcodec_find_decoder(codec_ctx->codec_id), NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Failed to open decoder for stream #%u\n", i);
return ret;
}
}
stream_ctx[i].dec_ctx = codec_ctx;
}
av_dump_format(ifmt_ctx, 0, filename, 0);
@@ -133,7 +108,8 @@ static int open_output_file(const char *filename)
}
in_stream = ifmt_ctx->streams[i];
dec_ctx = stream_ctx[i].dec_ctx;
dec_ctx = in_stream->codec;
enc_ctx = out_stream->codec;
if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
|| dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
@@ -143,11 +119,6 @@ static int open_output_file(const char *filename)
av_log(NULL, AV_LOG_FATAL, "Necessary encoder not found\n");
return AVERROR_INVALIDDATA;
}
enc_ctx = avcodec_alloc_context3(encoder);
if (!enc_ctx) {
av_log(NULL, AV_LOG_FATAL, "Failed to allocate the encoder context\n");
return AVERROR(ENOMEM);
}
/* In this example, we transcode to same properties (picture size,
* sample rate etc.). These properties can be changed for output
@@ -157,12 +128,9 @@ 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 = av_inv_q(dec_ctx->framerate);
enc_ctx->time_base = dec_ctx->time_base;
} else {
enc_ctx->sample_rate = dec_ctx->sample_rate;
enc_ctx->channel_layout = dec_ctx->channel_layout;
@@ -172,36 +140,28 @@ static int open_output_file(const char *filename)
enc_ctx->time_base = (AVRational){1, enc_ctx->sample_rate};
}
if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
enc_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
/* Third parameter can be used to pass settings to encoder */
ret = avcodec_open2(enc_ctx, encoder, NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open video encoder for stream #%u\n", i);
return ret;
}
ret = avcodec_parameters_from_context(out_stream->codecpar, enc_ctx);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Failed to copy encoder parameters to output stream #%u\n", i);
return ret;
}
out_stream->time_base = enc_ctx->time_base;
stream_ctx[i].enc_ctx = enc_ctx;
} else if (dec_ctx->codec_type == AVMEDIA_TYPE_UNKNOWN) {
av_log(NULL, AV_LOG_FATAL, "Elementary stream #%d is of unknown type, cannot proceed\n", i);
return AVERROR_INVALIDDATA;
} else {
/* if this stream must be remuxed */
ret = avcodec_parameters_copy(out_stream->codecpar, in_stream->codecpar);
ret = avcodec_copy_context(ofmt_ctx->streams[i]->codec,
ifmt_ctx->streams[i]->codec);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Copying parameters for stream #%u failed\n", i);
av_log(NULL, AV_LOG_ERROR, "Copying stream context failed\n");
return ret;
}
out_stream->time_base = in_stream->time_base;
}
if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
enc_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
}
av_dump_format(ofmt_ctx, 0, filename, 1);
@@ -228,8 +188,8 @@ static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
{
char args[512];
int ret = 0;
const AVFilter *buffersrc = NULL;
const AVFilter *buffersink = NULL;
AVFilter *buffersrc = NULL;
AVFilter *buffersink = NULL;
AVFilterContext *buffersrc_ctx = NULL;
AVFilterContext *buffersink_ctx = NULL;
AVFilterInOut *outputs = avfilter_inout_alloc();
@@ -385,17 +345,17 @@ static int init_filters(void)
filter_ctx[i].buffersrc_ctx = NULL;
filter_ctx[i].buffersink_ctx = NULL;
filter_ctx[i].filter_graph = NULL;
if (!(ifmt_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO
|| ifmt_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_VIDEO))
if (!(ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO
|| ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO))
continue;
if (ifmt_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_VIDEO)
if (ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO)
filter_spec = "null"; /* passthrough (dummy) filter for video */
else
filter_spec = "anull"; /* passthrough (dummy) filter for audio */
ret = init_filter(&filter_ctx[i], stream_ctx[i].dec_ctx,
stream_ctx[i].enc_ctx, filter_spec);
ret = init_filter(&filter_ctx[i], ifmt_ctx->streams[i]->codec,
ofmt_ctx->streams[i]->codec, filter_spec);
if (ret)
return ret;
}
@@ -407,7 +367,7 @@ static int encode_write_frame(AVFrame *filt_frame, unsigned int stream_index, in
int got_frame_local;
AVPacket enc_pkt;
int (*enc_func)(AVCodecContext *, AVPacket *, const AVFrame *, int *) =
(ifmt_ctx->streams[stream_index]->codecpar->codec_type ==
(ifmt_ctx->streams[stream_index]->codec->codec_type ==
AVMEDIA_TYPE_VIDEO) ? avcodec_encode_video2 : avcodec_encode_audio2;
if (!got_frame)
@@ -418,7 +378,7 @@ static int encode_write_frame(AVFrame *filt_frame, unsigned int stream_index, in
enc_pkt.data = NULL;
enc_pkt.size = 0;
av_init_packet(&enc_pkt);
ret = enc_func(stream_ctx[stream_index].enc_ctx, &enc_pkt,
ret = enc_func(ofmt_ctx->streams[stream_index]->codec, &enc_pkt,
filt_frame, got_frame);
av_frame_free(&filt_frame);
if (ret < 0)
@@ -429,7 +389,7 @@ 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,
stream_ctx[stream_index].enc_ctx->time_base,
ofmt_ctx->streams[stream_index]->codec->time_base,
ofmt_ctx->streams[stream_index]->time_base);
av_log(NULL, AV_LOG_DEBUG, "Muxing frame\n");
@@ -487,7 +447,7 @@ static int flush_encoder(unsigned int stream_index)
int ret;
int got_frame;
if (!(stream_ctx[stream_index].enc_ctx->codec->capabilities &
if (!(ofmt_ctx->streams[stream_index]->codec->codec->capabilities &
AV_CODEC_CAP_DELAY))
return 0;
@@ -518,6 +478,9 @@ int main(int argc, char **argv)
return 1;
}
av_register_all();
avfilter_register_all();
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if ((ret = open_output_file(argv[2])) < 0)
@@ -530,7 +493,7 @@ int main(int argc, char **argv)
if ((ret = av_read_frame(ifmt_ctx, &packet)) < 0)
break;
stream_index = packet.stream_index;
type = ifmt_ctx->streams[packet.stream_index]->codecpar->codec_type;
type = ifmt_ctx->streams[packet.stream_index]->codec->codec_type;
av_log(NULL, AV_LOG_DEBUG, "Demuxer gave frame of stream_index %u\n",
stream_index);
@@ -543,10 +506,10 @@ int main(int argc, char **argv)
}
av_packet_rescale_ts(&packet,
ifmt_ctx->streams[stream_index]->time_base,
stream_ctx[stream_index].dec_ctx->time_base);
ifmt_ctx->streams[stream_index]->codec->time_base);
dec_func = (type == AVMEDIA_TYPE_VIDEO) ? avcodec_decode_video2 :
avcodec_decode_audio4;
ret = dec_func(stream_ctx[stream_index].dec_ctx, frame,
ret = dec_func(ifmt_ctx->streams[stream_index]->codec, frame,
&got_frame, &packet);
if (ret < 0) {
av_frame_free(&frame);
@@ -555,7 +518,7 @@ int main(int argc, char **argv)
}
if (got_frame) {
frame->pts = frame->best_effort_timestamp;
frame->pts = av_frame_get_best_effort_timestamp(frame);
ret = filter_encode_write_frame(frame, stream_index);
av_frame_free(&frame);
if (ret < 0)
@@ -600,14 +563,13 @@ end:
av_packet_unref(&packet);
av_frame_free(&frame);
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
avcodec_free_context(&stream_ctx[i].dec_ctx);
if (ofmt_ctx && ofmt_ctx->nb_streams > i && ofmt_ctx->streams[i] && stream_ctx[i].enc_ctx)
avcodec_free_context(&stream_ctx[i].enc_ctx);
avcodec_close(ifmt_ctx->streams[i]->codec);
if (ofmt_ctx && ofmt_ctx->nb_streams > i && ofmt_ctx->streams[i] && ofmt_ctx->streams[i]->codec)
avcodec_close(ofmt_ctx->streams[i]->codec);
if (filter_ctx && filter_ctx[i].filter_graph)
avfilter_graph_free(&filter_ctx[i].filter_graph);
}
av_free(filter_ctx);
av_free(stream_ctx);
avformat_close_input(&ifmt_ctx);
if (ofmt_ctx && !(ofmt_ctx->oformat->flags & AVFMT_NOFILE))
avio_closep(&ofmt_ctx->pb);

View File

@@ -1,224 +0,0 @@
/*
* Video Acceleration API (video encoding) encode sample
*
* 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 VAAPI-accelerated encoding example.
*
* @example vaapi_encode.c
* This example shows how to do VAAPI-accelerated encoding. now only support NV12
* raw file, usage like: vaapi_encode 1920 1080 input.yuv output.h264
*
*/
#include <stdio.h>
#include <string.h>
#include <errno.h>
#include <libavcodec/avcodec.h>
#include <libavutil/pixdesc.h>
#include <libavutil/hwcontext.h>
static int width, height;
static AVBufferRef *hw_device_ctx = NULL;
static int set_hwframe_ctx(AVCodecContext *ctx, AVBufferRef *hw_device_ctx)
{
AVBufferRef *hw_frames_ref;
AVHWFramesContext *frames_ctx = NULL;
int err = 0;
if (!(hw_frames_ref = av_hwframe_ctx_alloc(hw_device_ctx))) {
fprintf(stderr, "Failed to create VAAPI frame context.\n");
return -1;
}
frames_ctx = (AVHWFramesContext *)(hw_frames_ref->data);
frames_ctx->format = AV_PIX_FMT_VAAPI;
frames_ctx->sw_format = AV_PIX_FMT_NV12;
frames_ctx->width = width;
frames_ctx->height = height;
frames_ctx->initial_pool_size = 20;
if ((err = av_hwframe_ctx_init(hw_frames_ref)) < 0) {
fprintf(stderr, "Failed to initialize VAAPI frame context."
"Error code: %s\n",av_err2str(err));
av_buffer_unref(&hw_frames_ref);
return err;
}
ctx->hw_frames_ctx = av_buffer_ref(hw_frames_ref);
if (!ctx->hw_frames_ctx)
err = AVERROR(ENOMEM);
av_buffer_unref(&hw_frames_ref);
return err;
}
static int encode_write(AVCodecContext *avctx, AVFrame *frame, FILE *fout)
{
int ret = 0;
AVPacket enc_pkt;
av_init_packet(&enc_pkt);
enc_pkt.data = NULL;
enc_pkt.size = 0;
if ((ret = avcodec_send_frame(avctx, frame)) < 0) {
fprintf(stderr, "Error code: %s\n", av_err2str(ret));
goto end;
}
while (1) {
ret = avcodec_receive_packet(avctx, &enc_pkt);
if (ret)
break;
enc_pkt.stream_index = 0;
ret = fwrite(enc_pkt.data, enc_pkt.size, 1, fout);
av_packet_unref(&enc_pkt);
}
end:
ret = ((ret == AVERROR(EAGAIN)) ? 0 : -1);
return ret;
}
int main(int argc, char *argv[])
{
int size, err;
FILE *fin = NULL, *fout = NULL;
AVFrame *sw_frame = NULL, *hw_frame = NULL;
AVCodecContext *avctx = NULL;
AVCodec *codec = NULL;
const char *enc_name = "h264_vaapi";
if (argc < 5) {
fprintf(stderr, "Usage: %s <width> <height> <input file> <output file>\n", argv[0]);
return -1;
}
width = atoi(argv[1]);
height = atoi(argv[2]);
size = width * height;
if (!(fin = fopen(argv[3], "r"))) {
fprintf(stderr, "Fail to open input file : %s\n", strerror(errno));
return -1;
}
if (!(fout = fopen(argv[4], "w+b"))) {
fprintf(stderr, "Fail to open output file : %s\n", strerror(errno));
err = -1;
goto close;
}
err = av_hwdevice_ctx_create(&hw_device_ctx, AV_HWDEVICE_TYPE_VAAPI,
NULL, NULL, 0);
if (err < 0) {
fprintf(stderr, "Failed to create a VAAPI device. Error code: %s\n", av_err2str(err));
goto close;
}
if (!(codec = avcodec_find_encoder_by_name(enc_name))) {
fprintf(stderr, "Could not find encoder.\n");
err = -1;
goto close;
}
if (!(avctx = avcodec_alloc_context3(codec))) {
err = AVERROR(ENOMEM);
goto close;
}
avctx->width = width;
avctx->height = height;
avctx->time_base = (AVRational){1, 25};
avctx->framerate = (AVRational){25, 1};
avctx->sample_aspect_ratio = (AVRational){1, 1};
avctx->pix_fmt = AV_PIX_FMT_VAAPI;
/* set hw_frames_ctx for encoder's AVCodecContext */
if ((err = set_hwframe_ctx(avctx, hw_device_ctx)) < 0) {
fprintf(stderr, "Failed to set hwframe context.\n");
goto close;
}
if ((err = avcodec_open2(avctx, codec, NULL)) < 0) {
fprintf(stderr, "Cannot open video encoder codec. Error code: %s\n", av_err2str(err));
goto close;
}
while (1) {
if (!(sw_frame = av_frame_alloc())) {
err = AVERROR(ENOMEM);
goto close;
}
/* read data into software frame, and transfer them into hw frame */
sw_frame->width = width;
sw_frame->height = height;
sw_frame->format = AV_PIX_FMT_NV12;
if ((err = av_frame_get_buffer(sw_frame, 32)) < 0)
goto close;
if ((err = fread((uint8_t*)(sw_frame->data[0]), size, 1, fin)) <= 0)
break;
if ((err = fread((uint8_t*)(sw_frame->data[1]), size/2, 1, fin)) <= 0)
break;
if (!(hw_frame = av_frame_alloc())) {
err = AVERROR(ENOMEM);
goto close;
}
if ((err = av_hwframe_get_buffer(avctx->hw_frames_ctx, hw_frame, 0)) < 0) {
fprintf(stderr, "Error code: %s.\n", av_err2str(err));
goto close;
}
if (!hw_frame->hw_frames_ctx) {
err = AVERROR(ENOMEM);
goto close;
}
if ((err = av_hwframe_transfer_data(hw_frame, sw_frame, 0)) < 0) {
fprintf(stderr, "Error while transferring frame data to surface."
"Error code: %s.\n", av_err2str(err));
goto close;
}
if ((err = (encode_write(avctx, hw_frame, fout))) < 0) {
fprintf(stderr, "Failed to encode.\n");
goto close;
}
av_frame_free(&hw_frame);
av_frame_free(&sw_frame);
}
/* flush encoder */
err = encode_write(avctx, NULL, fout);
if (err == AVERROR_EOF)
err = 0;
close:
if (fin)
fclose(fin);
if (fout)
fclose(fout);
av_frame_free(&sw_frame);
av_frame_free(&hw_frame);
avcodec_free_context(&avctx);
av_buffer_unref(&hw_device_ctx);
return err;
}

View File

@@ -1,306 +0,0 @@
/*
* Video Acceleration API (video transcoding) transcode sample
*
* 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 VAAPI-accelerated transcoding example.
*
* @example vaapi_transcode.c
* This example shows how to do VAAPI-accelerated transcoding.
* Usage: vaapi_transcode input_stream codec output_stream
* e.g: - vaapi_transcode input.mp4 h264_vaapi output_h264.mp4
* - vaapi_transcode input.mp4 vp9_vaapi output_vp9.ivf
*/
#include <stdio.h>
#include <errno.h>
#include <libavutil/hwcontext.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
static AVFormatContext *ifmt_ctx = NULL, *ofmt_ctx = NULL;
static AVBufferRef *hw_device_ctx = NULL;
static AVCodecContext *decoder_ctx = NULL, *encoder_ctx = NULL;
static int video_stream = -1;
static AVStream *ost;
static int initialized = 0;
static enum AVPixelFormat get_vaapi_format(AVCodecContext *ctx,
const enum AVPixelFormat *pix_fmts)
{
const enum AVPixelFormat *p;
for (p = pix_fmts; *p != AV_PIX_FMT_NONE; p++) {
if (*p == AV_PIX_FMT_VAAPI)
return *p;
}
fprintf(stderr, "Unable to decode this file using VA-API.\n");
return AV_PIX_FMT_NONE;
}
static int open_input_file(const char *filename)
{
int ret;
AVCodec *decoder = NULL;
AVStream *video = NULL;
if ((ret = avformat_open_input(&ifmt_ctx, filename, NULL, NULL)) < 0) {
fprintf(stderr, "Cannot open input file '%s', Error code: %s\n",
filename, av_err2str(ret));
return ret;
}
if ((ret = avformat_find_stream_info(ifmt_ctx, NULL)) < 0) {
fprintf(stderr, "Cannot find input stream information. Error code: %s\n",
av_err2str(ret));
return ret;
}
ret = av_find_best_stream(ifmt_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, &decoder, 0);
if (ret < 0) {
fprintf(stderr, "Cannot find a video stream in the input file. "
"Error code: %s\n", av_err2str(ret));
return ret;
}
video_stream = ret;
if (!(decoder_ctx = avcodec_alloc_context3(decoder)))
return AVERROR(ENOMEM);
video = ifmt_ctx->streams[video_stream];
if ((ret = avcodec_parameters_to_context(decoder_ctx, video->codecpar)) < 0) {
fprintf(stderr, "avcodec_parameters_to_context error. Error code: %s\n",
av_err2str(ret));
return ret;
}
decoder_ctx->hw_device_ctx = av_buffer_ref(hw_device_ctx);
if (!decoder_ctx->hw_device_ctx) {
fprintf(stderr, "A hardware device reference create failed.\n");
return AVERROR(ENOMEM);
}
decoder_ctx->get_format = get_vaapi_format;
if ((ret = avcodec_open2(decoder_ctx, decoder, NULL)) < 0)
fprintf(stderr, "Failed to open codec for decoding. Error code: %s\n",
av_err2str(ret));
return ret;
}
static int encode_write(AVFrame *frame)
{
int ret = 0;
AVPacket enc_pkt;
av_init_packet(&enc_pkt);
enc_pkt.data = NULL;
enc_pkt.size = 0;
if ((ret = avcodec_send_frame(encoder_ctx, frame)) < 0) {
fprintf(stderr, "Error during encoding. Error code: %s\n", av_err2str(ret));
goto end;
}
while (1) {
ret = avcodec_receive_packet(encoder_ctx, &enc_pkt);
if (ret)
break;
enc_pkt.stream_index = 0;
av_packet_rescale_ts(&enc_pkt, ifmt_ctx->streams[video_stream]->time_base,
ofmt_ctx->streams[0]->time_base);
ret = av_interleaved_write_frame(ofmt_ctx, &enc_pkt);
if (ret < 0) {
fprintf(stderr, "Error during writing data to output file. "
"Error code: %s\n", av_err2str(ret));
return -1;
}
}
end:
if (ret == AVERROR_EOF)
return 0;
ret = ((ret == AVERROR(EAGAIN)) ? 0:-1);
return ret;
}
static int dec_enc(AVPacket *pkt, AVCodec *enc_codec)
{
AVFrame *frame;
int ret = 0;
ret = avcodec_send_packet(decoder_ctx, pkt);
if (ret < 0) {
fprintf(stderr, "Error during decoding. Error code: %s\n", av_err2str(ret));
return ret;
}
while (ret >= 0) {
if (!(frame = av_frame_alloc()))
return AVERROR(ENOMEM);
ret = avcodec_receive_frame(decoder_ctx, frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
av_frame_free(&frame);
return 0;
} else if (ret < 0) {
fprintf(stderr, "Error while decoding. Error code: %s\n", av_err2str(ret));
goto fail;
}
if (!initialized) {
/* we need to ref hw_frames_ctx of decoder to initialize encoder's codec.
Only after we get a decoded frame, can we obtain its hw_frames_ctx */
encoder_ctx->hw_frames_ctx = av_buffer_ref(decoder_ctx->hw_frames_ctx);
if (!encoder_ctx->hw_frames_ctx) {
ret = AVERROR(ENOMEM);
goto fail;
}
/* set AVCodecContext Parameters for encoder, here we keep them stay
* the same as decoder.
* xxx: now the sample can't handle resolution change case.
*/
encoder_ctx->time_base = av_inv_q(decoder_ctx->framerate);
encoder_ctx->pix_fmt = AV_PIX_FMT_VAAPI;
encoder_ctx->width = decoder_ctx->width;
encoder_ctx->height = decoder_ctx->height;
if ((ret = avcodec_open2(encoder_ctx, enc_codec, NULL)) < 0) {
fprintf(stderr, "Failed to open encode codec. Error code: %s\n",
av_err2str(ret));
goto fail;
}
if (!(ost = avformat_new_stream(ofmt_ctx, enc_codec))) {
fprintf(stderr, "Failed to allocate stream for output format.\n");
ret = AVERROR(ENOMEM);
goto fail;
}
ost->time_base = encoder_ctx->time_base;
ret = avcodec_parameters_from_context(ost->codecpar, encoder_ctx);
if (ret < 0) {
fprintf(stderr, "Failed to copy the stream parameters. "
"Error code: %s\n", av_err2str(ret));
goto fail;
}
/* write the stream header */
if ((ret = avformat_write_header(ofmt_ctx, NULL)) < 0) {
fprintf(stderr, "Error while writing stream header. "
"Error code: %s\n", av_err2str(ret));
goto fail;
}
initialized = 1;
}
if ((ret = encode_write(frame)) < 0)
fprintf(stderr, "Error during encoding and writing.\n");
fail:
av_frame_free(&frame);
if (ret < 0)
return ret;
}
return 0;
}
int main(int argc, char **argv)
{
int ret = 0;
AVPacket dec_pkt;
AVCodec *enc_codec;
if (argc != 4) {
fprintf(stderr, "Usage: %s <input file> <encode codec> <output file>\n"
"The output format is guessed according to the file extension.\n"
"\n", argv[0]);
return -1;
}
ret = av_hwdevice_ctx_create(&hw_device_ctx, AV_HWDEVICE_TYPE_VAAPI, NULL, NULL, 0);
if (ret < 0) {
fprintf(stderr, "Failed to create a VAAPI device. Error code: %s\n", av_err2str(ret));
return -1;
}
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if (!(enc_codec = avcodec_find_encoder_by_name(argv[2]))) {
fprintf(stderr, "Could not find encoder '%s'\n", argv[2]);
ret = -1;
goto end;
}
if ((ret = (avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, argv[3]))) < 0) {
fprintf(stderr, "Failed to deduce output format from file extension. Error code: "
"%s\n", av_err2str(ret));
goto end;
}
if (!(encoder_ctx = avcodec_alloc_context3(enc_codec))) {
ret = AVERROR(ENOMEM);
goto end;
}
ret = avio_open(&ofmt_ctx->pb, argv[3], AVIO_FLAG_WRITE);
if (ret < 0) {
fprintf(stderr, "Cannot open output file. "
"Error code: %s\n", av_err2str(ret));
goto end;
}
/* read all packets and only transcoding video */
while (ret >= 0) {
if ((ret = av_read_frame(ifmt_ctx, &dec_pkt)) < 0)
break;
if (video_stream == dec_pkt.stream_index)
ret = dec_enc(&dec_pkt, enc_codec);
av_packet_unref(&dec_pkt);
}
/* flush decoder */
dec_pkt.data = NULL;
dec_pkt.size = 0;
ret = dec_enc(&dec_pkt, enc_codec);
av_packet_unref(&dec_pkt);
/* flush encoder */
ret = encode_write(NULL);
/* write the trailer for output stream */
av_write_trailer(ofmt_ctx);
end:
avformat_close_input(&ifmt_ctx);
avformat_close_input(&ofmt_ctx);
avcodec_free_context(&decoder_ctx);
avcodec_free_context(&encoder_ctx);
av_buffer_unref(&hw_device_ctx);
return ret;
}

View File

@@ -311,18 +311,18 @@ 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{https://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{https://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{https://ffmpeg.org/ffmpeg-filters.html#overlay, @code{overlay}} video filter.
@end itemize
@@ -333,19 +333,19 @@ 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{https://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{https://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{https://ffmpeg.org/ffmpeg-protocols.html#concat,
@code{concat}} protocol designed specifically for that, with examples in the
documentation.
@@ -385,7 +385,7 @@ mkfifo intermediate2.mpg
ffmpeg -i input1.avi -qscale:v 1 -y intermediate1.mpg < /dev/null &
ffmpeg -i input2.avi -qscale:v 1 -y intermediate2.mpg < /dev/null &
cat intermediate1.mpg intermediate2.mpg |\
ffmpeg -f mpeg -i - -c:v mpeg4 -c:a libmp3lame output.avi
ffmpeg -f mpeg -i - -c:v mpeg4 -acodec libmp3lame output.avi
@end example
@subsection Concatenating using raw audio and video
@@ -407,13 +407,13 @@ mkfifo temp2.a
mkfifo temp2.v
mkfifo all.a
mkfifo all.v
ffmpeg -i input1.flv -vn -f u16le -c:a pcm_s16le -ac 2 -ar 44100 - > temp1.a < /dev/null &
ffmpeg -i input2.flv -vn -f u16le -c:a pcm_s16le -ac 2 -ar 44100 - > temp2.a < /dev/null &
ffmpeg -i input1.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 - > temp1.a < /dev/null &
ffmpeg -i input2.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 - > temp2.a < /dev/null &
ffmpeg -i input1.flv -an -f yuv4mpegpipe - > temp1.v < /dev/null &
@{ ffmpeg -i input2.flv -an -f yuv4mpegpipe - < /dev/null | tail -n +2 > temp2.v ; @} &
cat temp1.a temp2.a > all.a &
cat temp1.v temp2.v > all.v &
ffmpeg -f u16le -c:a pcm_s16le -ac 2 -ar 44100 -i all.a \
ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 44100 -i all.a \
-f yuv4mpegpipe -i all.v \
-y output.flv
rm temp[12].[av] all.[av]
@@ -485,7 +485,7 @@ 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}}.
@url{https://ffmpeg.org/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.
@@ -501,71 +501,6 @@ 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
@anchor{background task}
@section How do I run ffmpeg as a background task?
ffmpeg normally checks the console input, for entries like "q" to stop
and "?" to give help, while performing operations. ffmpeg does not have a way of
detecting when it is running as a background task.
When it checks the console input, that can cause the process running ffmpeg
in the background to suspend.
To prevent those input checks, allowing ffmpeg to run as a background task,
use the @url{ffmpeg.html#stdin-option, @code{-nostdin} option}
in the ffmpeg invocation. This is effective whether you run ffmpeg in a shell
or invoke ffmpeg in its own process via an operating system API.
As an alternative, when you are running ffmpeg in a shell, you can redirect
standard input to @code{/dev/null} (on Linux and Mac OS)
or @code{NUL} (on Windows). You can do this redirect either
on the ffmpeg invocation, or from a shell script which calls ffmpeg.
For example:
@example
ffmpeg -nostdin -i INPUT OUTPUT
@end example
or (on Linux, Mac OS, and other UNIX-like shells):
@example
ffmpeg -i INPUT OUTPUT </dev/null
@end example
or (on Windows):
@example
ffmpeg -i INPUT OUTPUT <NUL
@end example
@section How do I prevent ffmpeg from suspending with a message like @emph{suspended (tty output)}?
If you run ffmpeg in the background, you may find that its process suspends.
There may be a message like @emph{suspended (tty output)}. The question is how
to prevent the process from being suspended.
For example:
@example
% ffmpeg -i INPUT OUTPUT &> ~/tmp/log.txt &
[1] 93352
%
[1] + suspended (tty output) ffmpeg -i INPUT OUTPUT &>
@end example
The message "tty output" notwithstanding, the problem here is that
ffmpeg normally checks the console input when it runs. The operating system
detects this, and suspends the process until you can bring it to the
foreground and attend to it.
The solution is to use the right techniques to tell ffmpeg not to consult
console input. You can use the
@url{ffmpeg.html#stdin-option, @code{-nostdin} option},
or redirect standard input with @code{< /dev/null}.
See FAQ
@ref{background task, @emph{How do I run ffmpeg as a background task?}}
for details.
@chapter Development
@section Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat?

View File

@@ -147,28 +147,6 @@ process.
The only thing left is to automate the execution of the fate.sh script and
the synchronisation of the samples directory.
@chapter Uploading new samples to the fate suite
This is for developers who have an account on the fate suite server.
If you upload new samples, please make sure they are as small as possible,
space on each client, network bandwidth and so on benefit from smaller test cases.
Also keep in mind older checkouts use existing sample files, that means in
practice generally do not replace, remove or overwrite files as it likely would
break older checkouts or releases.
Also all needed samples for a commit should be uploaded, ideally 24
hours, before the push.
@example
#First update your local samples copy:
rsync -vauL --chmod=Dg+s,Duo+x,ug+rw,o+r,o-w,+X fate-suite.ffmpeg.org:/home/samples/fate-suite/ ~/fate-suite
#Then do a dry run checking what would be uploaded:
rsync -vanL --no-g --chmod=Dg+s,Duo+x,ug+rw,o+r,o-w,+X ~/fate-suite/ fate-suite.ffmpeg.org:/home/samples/fate-suite
#Upload the files:
rsync -vaL --no-g --chmod=Dg+s,Duo+x,ug+rw,o+r,o-w,+X ~/fate-suite/ fate-suite.ffmpeg.org:/home/samples/fate-suite
@end example
@chapter FATE makefile targets and variables
@@ -219,16 +197,6 @@ through @command{ssh}.
@item GEN
Set to @samp{1} to generate the missing or mismatched references.
@item HWACCEL
Specify which hardware acceleration to use while running regression tests,
by default @samp{none} is used.
@item KEEP
Set to @samp{1} to keep temp files generated by fate test(s) when test is successful.
Default is @samp{0}, which removes these files. Files are always kept when a test
fails.
@end table
@section Examples

View File

@@ -6,7 +6,6 @@ workdir= # directory in which to do all the work
#fate_recv="ssh -T fate@fate.ffmpeg.org" # command to submit report
comment= # optional description
build_only= # set to "yes" for a compile-only instance that skips tests
ignore_tests=
# the following are optional and map to configure options
arch=
@@ -27,7 +26,5 @@ extra_conf= # extra configure options not covered above
#make= # name of GNU make if not 'make'
makeopts= # extra options passed to 'make'
#makeopts_fate= # extra options passed to 'make' when running tests,
# defaulting to makeopts above if this is not set
#tar= # command to create a tar archive from its arguments on stdout,
# defaults to 'tar c'

View File

@@ -26,12 +26,12 @@ bitstream level modifications without performing decoding.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libavcodec.html,libavcodec}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), libavcodec(3)
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavcodec(3)
@end ifnothtml
@include authors.texi

View File

@@ -23,12 +23,12 @@ the libavcodec library.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libavcodec.html,libavcodec}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), libavcodec(3)
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavcodec(3)
@end ifnothtml
@include authors.texi

View File

@@ -23,12 +23,12 @@ libavdevice library.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libavdevice.html,libavdevice}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), libavdevice(3)
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavdevice(3)
@end ifnothtml
@include authors.texi

View File

@@ -23,12 +23,12 @@ libavfilter library.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libavfilter.html,libavfilter}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), libavfilter(3)
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavfilter(3)
@end ifnothtml
@include authors.texi

View File

@@ -23,12 +23,12 @@ provided by the libavformat library.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libavformat.html,libavformat}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), libavformat(3)
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavformat(3)
@end ifnothtml
@include authors.texi

View File

@@ -23,12 +23,12 @@ libavformat library.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libavformat.html,libavformat}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), libavformat(3)
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavformat(3)
@end ifnothtml
@include authors.texi

View File

@@ -25,12 +25,12 @@ and convert audio format and packing layout.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libswresample.html,libswresample}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), libswresample(3)
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libswresample(3)
@end ifnothtml
@include authors.texi

View File

@@ -24,12 +24,12 @@ image rescaling and pixel format conversion.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libswscale.html,libswscale}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), libswscale(3)
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libswscale(3)
@end ifnothtml
@include authors.texi

View File

@@ -23,12 +23,12 @@ by the libavutil library.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libavutil.html,libavutil}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), libavutil(3)
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavutil(3)
@end ifnothtml
@include authors.texi

View File

@@ -12,7 +12,7 @@
@chapter Synopsis
ffmpeg [@var{global_options}] @{[@var{input_file_options}] -i @file{input_url}@} ... @{[@var{output_file_options}] @file{output_url}@} ...
ffmpeg [@var{global_options}] @{[@var{input_file_options}] -i @file{input_file}@} ... @{[@var{output_file_options}] @file{output_file}@} ...
@chapter Description
@c man begin DESCRIPTION
@@ -24,10 +24,10 @@ rates and resize video on the fly with a high quality polyphase filter.
@command{ffmpeg} reads from an arbitrary number of input "files" (which can be regular
files, pipes, network streams, grabbing devices, etc.), specified by the
@code{-i} option, and writes to an arbitrary number of output "files", which are
specified by a plain output url. Anything found on the command line which
cannot be interpreted as an option is considered to be an output url.
specified by a plain output filename. Anything found on the command line which
cannot be interpreted as an option is considered to be an output filename.
Each input or output url can, in principle, contain any number of streams of
Each input or output file can, in principle, contain any number of streams of
different types (video/audio/subtitle/attachment/data). The allowed number and/or
types of streams may be limited by the container format. Selecting which
streams from which inputs will go into which output is either done automatically
@@ -216,208 +216,16 @@ filters is obviously also impossible, since filters work on uncompressed data.
@chapter Stream selection
@c man begin STREAM SELECTION
@command{ffmpeg} provides the @code{-map} option for manual control of stream selection in each
output file. Users can skip @code{-map} and let ffmpeg perform automatic stream selection as
described below. The @code{-vn / -an / -sn / -dn} options can be used to skip inclusion of
video, audio, subtitle and data streams respectively, whether manually mapped or automatically
selected, except for those streams which are outputs of complex filtergraphs.
By default, @command{ffmpeg} includes only one stream of each type (video, audio, subtitle)
present in the input files and adds them to each output file. It picks the
"best" of each based upon the following criteria: for video, it is the stream
with the highest resolution, for audio, it is the stream with the most channels, for
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.
@section Description
The sub-sections that follow describe the various rules that are involved in stream selection.
The examples that follow next show how these rules are applied in practice.
While every effort is made to accurately reflect the behavior of the program, FFmpeg is under
continuous development and the code may have changed since the time of this writing.
@subsection Automatic stream selection
In the absence of any map options for a particular output file, ffmpeg inspects the output
format to check which type of streams can be included in it, viz. video, audio and/or
subtitles. For each acceptable stream type, ffmpeg will pick one stream, when available,
from among all the inputs.
It will select that stream based upon the following criteria:
@itemize
@item
for video, it is the stream with the highest resolution,
@item
for audio, it is the stream with the most channels,
@item
for subtitles, it is the first subtitle stream found but there's a caveat.
The output format's default subtitle encoder can be either text-based or image-based,
and only a subtitle stream of the same type will be chosen.
@end itemize
In the case where several streams of the same type rate equally, the stream with the lowest
index is chosen.
Data or attachment streams are not automatically selected and can only be included
using @code{-map}.
@subsection Manual stream selection
When @code{-map} is used, only user-mapped streams are included in that output file,
with one possible exception for filtergraph outputs described below.
@subsection Complex filtergraphs
If there are any complex filtergraph output streams with unlabeled pads, they will be added
to the first output file. This will lead to a fatal error if the stream type is not supported
by the output format. In the absence of the map option, the inclusion of these streams leads
to the automatic stream selection of their types being skipped. If map options are present,
these filtergraph streams are included in addition to the mapped streams.
Complex filtergraph output streams with labeled pads must be mapped once and exactly once.
@subsection Stream handling
Stream handling is independent of stream selection, with an exception for subtitles described
below. Stream handling is set via the @code{-codec} option addressed to streams within a
specific @emph{output} file. In particular, codec options are applied by ffmpeg after the
stream selection process and thus do not influence the latter. If no @code{-codec} option is
specified for a stream type, ffmpeg will select the default encoder registered by the output
file muxer.
An exception exists for subtitles. If a subtitle encoder is specified for an output file, the
first subtitle stream found of any type, text or image, will be included. ffmpeg does not validate
if the specified encoder can convert the selected stream or if the converted stream is acceptable
within the output format. This applies generally as well: when the user sets an encoder manually,
the stream selection process cannot check if the encoded stream can be muxed into the output file.
If it cannot, ffmpeg will abort and @emph{all} output files will fail to be processed.
@section Examples
The following examples illustrate the behavior, quirks and limitations of ffmpeg's stream
selection methods.
They assume the following three input files.
@verbatim
input file 'A.avi'
stream 0: video 640x360
stream 1: audio 2 channels
input file 'B.mp4'
stream 0: video 1920x1080
stream 1: audio 2 channels
stream 2: subtitles (text)
stream 3: audio 5.1 channels
stream 4: subtitles (text)
input file 'C.mkv'
stream 0: video 1280x720
stream 1: audio 2 channels
stream 2: subtitles (image)
@end verbatim
@subsubheading Example: automatic stream selection
@example
ffmpeg -i A.avi -i B.mp4 out1.mkv out2.wav -map 1:a -c:a copy out3.mov
@end example
There are three output files specified, and for the first two, no @code{-map} options
are set, so ffmpeg will select streams for these two files automatically.
@file{out1.mkv} is a Matroska container file and accepts video, audio and subtitle streams,
so ffmpeg will try to select one of each type.@*
For video, it will select @code{stream 0} from @file{B.mp4}, which has the highest
resolution among all the input video streams.@*
For audio, it will select @code{stream 3} from @file{B.mp4}, since it has the greatest
number of channels.@*
For subtitles, it will select @code{stream 2} from @file{B.mp4}, which is the first subtitle
stream from among @file{A.avi} and @file{B.mp4}.
@file{out2.wav} accepts only audio streams, so only @code{stream 3} from @file{B.mp4} is
selected.
For @file{out3.mov}, since a @code{-map} option is set, no automatic stream selection will
occur. The @code{-map 1:a} option will select all audio streams from the second input
@file{B.mp4}. No other streams will be included in this output file.
For the first two outputs, all included streams will be transcoded. The encoders chosen will
be the default ones registered by each output format, which may not match the codec of the
selected input streams.
For the third output, codec option for audio streams has been set
to @code{copy}, so no decoding-filtering-encoding operations will occur, or @emph{can} occur.
Packets of selected streams shall be conveyed from the input file and muxed within the output
file.
@subsubheading Example: automatic subtitles selection
@example
ffmpeg -i C.mkv out1.mkv -c:s dvdsub -an out2.mkv
@end example
Although @file{out1.mkv} is a Matroska container file which accepts subtitle streams, only a
video and audio stream shall be selected. The subtitle stream of @file{C.mkv} is image-based
and the default subtitle encoder of the Matroska muxer is text-based, so a transcode operation
for the subtitles is expected to fail and hence the stream isn't selected. However, in
@file{out2.mkv}, a subtitle encoder is specified in the command and so, the subtitle stream is
selected, in addition to the video stream. The presence of @code{-an} disables audio stream
selection for @file{out2.mkv}.
@subsubheading Example: unlabeled filtergraph outputs
@example
ffmpeg -i A.avi -i C.mkv -i B.mp4 -filter_complex "overlay" out1.mp4 out2.srt
@end example
A filtergraph is setup here using the @code{-filter_complex} option and consists of a single
video filter. The @code{overlay} filter requires exactly two video inputs, but none are
specified, so the first two available video streams are used, those of @file{A.avi} and
@file{C.mkv}. The output pad of the filter has no label and so is sent to the first output file
@file{out1.mp4}. Due to this, automatic selection of the video stream is skipped, which would
have selected the stream in @file{B.mp4}. The audio stream with most channels viz. @code{stream 3}
in @file{B.mp4}, is chosen automatically. No subtitle stream is chosen however, since the MP4
format has no default subtitle encoder registered, and the user hasn't specified a subtitle encoder.
The 2nd output file, @file{out2.srt}, only accepts text-based subtitle streams. So, even though
the first subtitle stream available belongs to @file{C.mkv}, it is image-based and hence skipped.
The selected stream, @code{stream 2} in @file{B.mp4}, is the first text-based subtitle stream.
@subsubheading Example: labeled filtergraph outputs
@example
ffmpeg -i A.avi -i B.mp4 -i C.mkv -filter_complex "[1:v]hue=s=0[outv];overlay;aresample" \
-map '[outv]' -an out1.mp4 \
out2.mkv \
-map '[outv]' -map 1:a:0 out3.mkv
@end example
The above command will fail, as the output pad labelled @code{[outv]} has been mapped twice.
None of the output files shall be processed.
@example
ffmpeg -i A.avi -i B.mp4 -i C.mkv -filter_complex "[1:v]hue=s=0[outv];overlay;aresample" \
-an out1.mp4 \
out2.mkv \
-map 1:a:0 out3.mkv
@end example
This command above will also fail as the hue filter output has a label, @code{[outv]},
and hasn't been mapped anywhere.
The command should be modified as follows,
@example
ffmpeg -i A.avi -i B.mp4 -i C.mkv -filter_complex "[1:v]hue=s=0,split=2[outv1][outv2];overlay;aresample" \
-map '[outv1]' -an out1.mp4 \
out2.mkv \
-map '[outv2]' -map 1:a:0 out3.mkv
@end example
The video stream from @file{B.mp4} is sent to the hue filter, whose output is cloned once using
the split filter, and both outputs labelled. Then a copy each is mapped to the first and third
output files.
The overlay filter, requiring two video inputs, uses the first two unused video streams. Those
are the streams from @file{A.avi} and @file{C.mkv}. The overlay output isn't labelled, so it is
sent to the first output file @file{out1.mp4}, regardless of the presence of the @code{-map} option.
The aresample filter is sent the first unused audio stream, that of @file{A.avi}. Since this filter
output is also unlabelled, it too is mapped to the first output file. The presence of @code{-an}
only suppresses automatic or manual stream selection of audio streams, not outputs sent from
filtergraphs. Both these mapped streams shall be ordered before the mapped stream in @file{out1.mp4}.
The video, audio and subtitle streams mapped to @code{out2.mkv} are entirely determined by
automatic stream selection.
@file{out3.mkv} consists of the cloned video output from the hue filter and the first audio
stream from @file{B.mp4}.
@*
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.
@c man end STREAM SELECTION
@@ -435,8 +243,8 @@ Force input or output file format. The format is normally auto detected for inpu
files and guessed from the file extension for output files, so this option is not
needed in most cases.
@item -i @var{url} (@emph{input})
input file url
@item -i @var{filename} (@emph{input})
input file name
@item -y (@emph{global})
Overwrite output files without asking.
@@ -473,7 +281,7 @@ libx264, and the 138th audio, which will be encoded with libvorbis.
When used as an input option (before @code{-i}), limit the @var{duration} of
data read from the input file.
When used as an output option (before an output url), stop writing the
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,
@@ -481,8 +289,8 @@ see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1)
-to and -t are mutually exclusive and -t has priority.
@item -to @var{position} (@emph{input/output})
Stop writing the output or reading the input at @var{position}.
@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}.
@@ -502,13 +310,13 @@ 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
will be preserved.
When used as an output option (before an output url), decodes but discards
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})
@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.
@@ -549,49 +357,6 @@ To set the language of the first audio stream:
ffmpeg -i INPUT -metadata:s:a:0 language=eng OUTPUT
@end example
@item -disposition[:stream_specifier] @var{value} (@emph{output,per-stream})
Sets the disposition for a stream.
This option overrides the disposition copied from the input stream. It is also
possible to delete the disposition by setting it to 0.
The following dispositions are recognized:
@table @option
@item default
@item dub
@item original
@item comment
@item lyrics
@item karaoke
@item forced
@item hearing_impaired
@item visual_impaired
@item clean_effects
@item attached_pic
@item captions
@item descriptions
@item dependent
@item metadata
@end table
For example, to make the second audio stream the default stream:
@example
ffmpeg -i in.mkv -c copy -disposition:a:1 default out.mkv
@end example
To make the second subtitle stream the default stream and remove the default
disposition from the first subtitle stream:
@example
ffmpeg -i in.mkv -c copy -disposition:s:0 0 -disposition:s:1 default out.mkv
@end example
To add an embedded cover/thumbnail:
@example
ffmpeg -i in.mp4 -i IMAGE -map 0 -map 1 -c copy -c:v:1 png -disposition:v:1 attached_pic out.mp4
@end example
Not all muxers support embedded thumbnails, and those who do, only support a few formats, like JPEG or PNG.
@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
@@ -614,13 +379,8 @@ they do not conflict with the standard, as in:
ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg
@end example
@item -dn (@emph{output})
Disable data recording. For full manual control see the @code{-map}
option.
@item -dframes @var{number} (@emph{output})
Set the number of data frames to output. This is an obsolete alias for
@code{-frames:d}, which you should use instead.
Set the number of data frames to output. 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.
@@ -655,11 +415,6 @@ This option is similar to @option{-filter}, the only difference is that its
argument is the name of the file from which a filtergraph description is to be
read.
@item -filter_threads @var{nb_threads} (@emph{global})
Defines how many threads are used to process a filter pipeline. Each pipeline
will produce a thread pool with this many threads available for parallel processing.
The default is the number of available CPUs.
@item -pre[:@var{stream_specifier}] @var{preset_name} (@emph{output,per-stream})
Specify the preset for matching stream(s).
@@ -675,7 +430,6 @@ the encoding process. It is made of "@var{key}=@var{value}" lines. @var{key}
consists of only alphanumeric characters. The last key of a sequence of
progress information is always "progress".
@anchor{stdin option}
@item -stdin
Enable interaction on standard input. On by default unless standard input is
used as an input. To explicitly disable interaction you need to specify
@@ -736,8 +490,7 @@ Disable automatically rotating video based on file metadata.
@table @option
@item -vframes @var{number} (@emph{output})
Set the number of video frames to output. This is an obsolete alias for
@code{-frames:v}, which you should use instead.
Set the number of video frames to output. 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).
@@ -776,8 +529,7 @@ stored at container level, but not the aspect ratio stored in encoded
frames, if it exists.
@item -vn (@emph{output})
Disable video recording. For full manual control see the @code{-map}
option.
Disable video recording.
@item -vcodec @var{codec} (@emph{output})
Set the video codec. This is an alias for @code{-codec:v}.
@@ -824,6 +576,8 @@ as the input (or graph output) and automatic conversions are disabled.
@item -sws_flags @var{flags} (@emph{input/output})
Set SwScaler flags.
@item -vdt @var{n}
Discard threshold.
@item -rc_override[:@var{stream_specifier}] @var{override} (@emph{output,per-stream})
Rate control override for specific intervals, formatted as "int,int,int"
@@ -843,16 +597,6 @@ Calculate PSNR of compressed frames.
Dump video coding statistics to @file{vstats_HHMMSS.log}.
@item -vstats_file @var{file}
Dump video coding statistics to @var{file}.
@item -vstats_version @var{file}
Specifies which version of the vstats format to use. Default is 2.
version = 1 :
@code{frame= %5d q= %2.1f PSNR= %6.2f f_size= %6d s_size= %8.0fkB time= %0.3f br= %7.1fkbits/s avg_br= %7.1fkbits/s}
version > 1:
@code{out= %2d st= %2d frame= %5d q= %2.1f PSNR= %6.2f f_size= %6d s_size= %8.0fkB time= %0.3f br= %7.1fkbits/s avg_br= %7.1fkbits/s}
@item -top[:@var{stream_specifier}] @var{n} (@emph{output,per-stream})
top=1/bottom=0/auto=-1 field first
@item -dc @var{precision}
@@ -920,104 +664,6 @@ would be more efficient.
When doing stream copy, copy also non-key frames found at the
beginning.
@item -init_hw_device @var{type}[=@var{name}][:@var{device}[,@var{key=value}...]]
Initialise a new hardware device of type @var{type} called @var{name}, using the
given device parameters.
If no name is specified it will receive a default name of the form "@var{type}%d".
The meaning of @var{device} and the following arguments depends on the
device type:
@table @option
@item cuda
@var{device} is the number of the CUDA device.
@item dxva2
@var{device} is the number of the Direct3D 9 display adapter.
@item vaapi
@var{device} is either an X11 display name or a DRM render node.
If not specified, it will attempt to open the default X11 display (@emph{$DISPLAY})
and then the first DRM render node (@emph{/dev/dri/renderD128}).
@item vdpau
@var{device} is an X11 display name.
If not specified, it will attempt to open the default X11 display (@emph{$DISPLAY}).
@item qsv
@var{device} selects a value in @samp{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
If not specified, @samp{auto_any} is used.
(Note that it may be easier to achieve the desired result for QSV by creating the
platform-appropriate subdevice (@samp{dxva2} or @samp{vaapi}) and then deriving a
QSV device from that.)
@item opencl
@var{device} selects the platform and device as @emph{platform_index.device_index}.
The set of devices can also be filtered using the key-value pairs to find only
devices matching particular platform or device strings.
The strings usable as filters are:
@table @option
@item platform_profile
@item platform_version
@item platform_name
@item platform_vendor
@item platform_extensions
@item device_name
@item device_vendor
@item driver_version
@item device_version
@item device_profile
@item device_extensions
@item device_type
@end table
The indices and filters must together uniquely select a device.
Examples:
@table @emph
@item -init_hw_device opencl:0.1
Choose the second device on the first platform.
@item -init_hw_device opencl:,device_name=Foo9000
Choose the device with a name containing the string @emph{Foo9000}.
@item -init_hw_device opencl:1,device_type=gpu,device_extensions=cl_khr_fp16
Choose the GPU device on the second platform supporting the @emph{cl_khr_fp16}
extension.
@end table
@end table
@item -init_hw_device @var{type}[=@var{name}]@@@var{source}
Initialise a new hardware device of type @var{type} called @var{name},
deriving it from the existing device with the name @var{source}.
@item -init_hw_device list
List all hardware device types supported in this build of ffmpeg.
@item -filter_hw_device @var{name}
Pass the hardware device called @var{name} to all filters in any filter graph.
This can be used to set the device to upload to with the @code{hwupload} filter,
or the device to map to with the @code{hwmap} filter. Other filters may also
make use of this parameter when they require a hardware device. Note that this
is typically only required when the input is not already in hardware frames -
when it is, filters will derive the device they require from the context of the
frames they receive as input.
This is a global setting, so all filters will receive the same device.
@item -hwaccel[:@var{stream_specifier}] @var{hwaccel} (@emph{input,per-stream})
Use hardware acceleration to decode the matching stream(s). The allowed values
of @var{hwaccel} are:
@@ -1028,15 +674,15 @@ Do not use any hardware acceleration (the default).
@item auto
Automatically select the hardware acceleration method.
@item vda
Use Apple VDA hardware acceleration.
@item vdpau
Use VDPAU (Video Decode and Presentation API for Unix) hardware acceleration.
@item dxva2
Use DXVA2 (DirectX Video Acceleration) hardware acceleration.
@item vaapi
Use VAAPI (Video Acceleration API) hardware acceleration.
@item qsv
Use the Intel QuickSync Video acceleration for video transcoding.
@@ -1060,11 +706,33 @@ useful for testing.
@item -hwaccel_device[:@var{stream_specifier}] @var{hwaccel_device} (@emph{input,per-stream})
Select a device to use for hardware acceleration.
This option only makes sense when the @option{-hwaccel} option is also specified.
It can either refer to an existing device created with @option{-init_hw_device}
by name, or it can create a new device as if
@samp{-init_hw_device} @var{type}:@var{hwaccel_device}
were called immediately before.
This option only makes sense when the @option{-hwaccel} option is also
specified. Its exact meaning depends on the specific hardware acceleration
method chosen.
@table @option
@item vdpau
For VDPAU, this option specifies the X11 display/screen to use. If this option
is not specified, the value of the @var{DISPLAY} environment variable is used
@item dxva2
For DXVA2, this option should contain the number of the display adapter to use.
If this option is not specified, the default adapter is used.
@item qsv
For QSV, this option corresponds to the valus 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.
@@ -1075,8 +743,7 @@ List all hardware acceleration methods supported in this build of ffmpeg.
@table @option
@item -aframes @var{number} (@emph{output})
Set the number of audio frames to output. This is an obsolete alias for
@code{-frames:a}, which you should use instead.
Set the number of audio frames to output. 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
@@ -1090,8 +757,7 @@ default to the number of input audio channels. For input streams
this option only makes sense for audio grabbing devices and raw demuxers
and is mapped to the corresponding demuxer options.
@item -an (@emph{output})
Disable audio recording. For full manual control see the @code{-map}
option.
Disable audio recording.
@item -acodec @var{codec} (@emph{input/output})
Set the audio codec. This is an alias for @code{-codec:a}.
@item -sample_fmt[:@var{stream_specifier}] @var{sample_fmt} (@emph{output,per-stream})
@@ -1126,8 +792,7 @@ stereo but not 6 channels as 5.1. The default is to always try to guess. Use
@item -scodec @var{codec} (@emph{input/output})
Set the subtitle codec. This is an alias for @code{-codec:s}.
@item -sn (@emph{output})
Disable subtitle recording. For full manual control see the @code{-map}
option.
Disable subtitle recording.
@item -sbsf @var{bitstream_filter}
Deprecated, see -bsf
@end table
@@ -1157,7 +822,7 @@ Set the size of the canvas used to render subtitles.
@section Advanced options
@table @option
@item -map [-]@var{input_file_id}[:@var{stream_specifier}][?][,@var{sync_file_id}[:@var{stream_specifier}]] | @var{[linklabel]} (@emph{output})
@item -map [-]@var{input_file_id}[:@var{stream_specifier}][,@var{sync_file_id}[:@var{stream_specifier}]] | @var{[linklabel]} (@emph{output})
Designate one or more input streams as a source for the output file. Each input
stream is identified by the input file index @var{input_file_id} and
@@ -1173,11 +838,6 @@ the source for output stream 1, etc.
A @code{-} character before the stream identifier creates a "negative" mapping.
It disables matching streams from already created mappings.
A trailing @code{?} after the stream index will allow the map to be
optional: if the map matches no streams the map will be ignored instead
of failing. Note the map will still fail if an invalid input file index
is used; such as if the map refers to a non-existent input.
An alternative @var{[linklabel]} form will map outputs from complex filter
graphs (see the @option{-filter_complex} option) to the output file.
@var{linklabel} must correspond to a defined output link label in the graph.
@@ -1215,13 +875,6 @@ To map all the streams except the second audio, use negative mappings
ffmpeg -i INPUT -map 0 -map -0:a:1 OUTPUT
@end example
To map the video and audio streams from the first input, and using the
trailing @code{?}, ignore the audio mapping if no audio streams exist in
the first input:
@example
ffmpeg -i INPUT -map 0:v -map 0:a? OUTPUT
@end example
To pick the English audio stream:
@example
ffmpeg -i INPUT -map 0:m:language:eng OUTPUT
@@ -1237,7 +890,7 @@ such streams is attempted.
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}]
@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
be mapped on all the audio streams.
@@ -1246,10 +899,6 @@ Using "-1" instead of
@var{input_file_id}.@var{stream_specifier}.@var{channel_id} will map a muted
channel.
A trailing @code{?} will allow the map_channel to be
optional: if the map_channel matches no channel the map_channel will be ignored instead
of failing.
For example, assuming @var{INPUT} is a stereo audio file, you can switch the
two audio channels with the following command:
@example
@@ -1297,13 +946,6 @@ video stream), you can use the following command:
ffmpeg -i input.mkv -filter_complex "[0:1] [0:2] amerge" -c:a pcm_s16le -c:v copy output.mkv
@end example
To map the first two audio channels from the first input, and using the
trailing @code{?}, ignore the audio channel mapping if the first input is
mono instead of stereo:
@example
ffmpeg -i INPUT -map_channel 0.0.0 -map_channel 0.0.1? OUTPUT
@end example
@item -map_metadata[:@var{metadata_spec_out}] @var{infile}[:@var{metadata_spec_in}] (@emph{output,per-metadata})
Set metadata information of the next output file from @var{infile}. Note that
those are file indices (zero-based), not filenames.
@@ -1353,12 +995,12 @@ disable any chapter copying.
@item -benchmark (@emph{global})
Show benchmarking information at the end of an encode.
Shows real, system and user time used and maximum memory consumption.
Shows CPU time used and maximum memory consumption.
Maximum memory consumption is not supported on all systems,
it will usually display as 0 if not supported.
@item -benchmark_all (@emph{global})
Show benchmarking information during the encode.
Shows real, system and user time used in various steps (audio/video encode/decode).
Shows CPU time used in various steps (audio/video encode/decode).
@item -timelimit @var{duration} (@emph{global})
Exit after ffmpeg has been running for @var{duration} seconds.
@item -dump (@emph{global})
@@ -1366,13 +1008,17 @@ 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).
By default @command{ffmpeg} attempts to read the input(s) as fast as possible.
This option will slow down the reading of the input(s) to the native frame rate
of the input(s). It is useful for real-time output (e.g. live streaming).
@item -loop_input
Loop over the input stream. Currently it works only for image
streams. This option is used for automatic FFserver testing.
This option is deprecated, use -loop 1.
@item -loop_output @var{number_of_times}
Repeatedly loop output for formats that support looping such as animated GIF
(0 will loop the output infinitely).
@@ -1466,32 +1112,6 @@ Try to make the choice automatically, in order to generate a sane output.
Default value is -1.
@item -enc_time_base[:@var{stream_specifier}] @var{timebase} (@emph{output,per-stream})
Set the encoder timebase. @var{timebase} is a floating point number,
and can assume one of the following values:
@table @option
@item 0
Assign a default value according to the media type.
For video - use 1/framerate, for audio - use 1/samplerate.
@item -1
Use the input stream timebase when possible.
If an input stream is not available, the default timebase will be used.
@item >0
Use the provided number as the timebase.
This field can be provided as a ratio of two integers (e.g. 1:24, 1:48000)
or as a floating point number (e.g. 0.04166, 2.0833e-5)
@end table
Default value is 0.
@item -bitexact (@emph{input/output})
Enable bitexact mode for (de)muxer and (de/en)coder
@item -shortest (@emph{output})
Finish encoding when the shortest input stream ends.
@item -dts_delta_threshold
@@ -1509,7 +1129,7 @@ may be reassigned to a different value.
For example, to set the stream 0 PID to 33 and the stream 1 PID to 36 for
an output mpegts file:
@example
ffmpeg -i inurl -streamid 0:33 -streamid 1:36 out.ts
ffmpeg -i infile -streamid 0:33 -streamid 1:36 out.ts
@end example
@item -bsf[:@var{stream_specifier}] @var{bitstream_filters} (@emph{output,per-stream})
@@ -1581,11 +1201,6 @@ To generate 5 seconds of pure red video using lavfi @code{color} source:
ffmpeg -filter_complex 'color=c=red' -t 5 out.mkv
@end example
@item -filter_complex_threads @var{nb_threads} (@emph{global})
Defines how many threads are used to process a filter_complex graph.
Similar to filter_threads but used for @code{-filter_complex} graphs only.
The default is the number of available CPUs.
@item -lavfi @var{filtergraph} (@emph{global})
Define a complex filtergraph, i.e. one with arbitrary number of inputs and/or
outputs. Equivalent to @option{-filter_complex}.
@@ -1614,6 +1229,16 @@ 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
many aspects of the encoding from @command{ffmpeg}. Without this
option @command{ffmpeg} will transmit to @command{ffserver} what is
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
@@ -1654,15 +1279,6 @@ No packets were passed to the muxer, the output is empty.
@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
@@ -1868,7 +1484,7 @@ to enable LAME support by passing @code{--enable-libmp3lame} to configure.
The mapping is particularly useful for DVD transcoding
to get the desired audio language.
NOTE: To see the supported input formats, use @code{ffmpeg -demuxers}.
NOTE: To see the supported input formats, use @code{ffmpeg -formats}.
@item
You can extract images from a video, or create a video from many images:
@@ -1883,8 +1499,8 @@ output them in files named @file{foo-001.jpeg}, @file{foo-002.jpeg},
etc. Images will be rescaled to fit the new WxH values.
If you want to extract just a limited number of frames, you can use the
above command in combination with the @code{-frames:v} or @code{-t} option,
or in combination with -ss to start extracting from a certain point in time.
above command in combination with the -vframes or -t option, or in
combination with -ss to start extracting from a certain point in time.
For creating a video from many images:
@example
@@ -1968,7 +1584,7 @@ ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
@ifset config-not-all
@url{ffmpeg-all.html,ffmpeg-all},
@end ifset
@url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-utils.html,ffmpeg-utils},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@@ -1987,7 +1603,7 @@ ffmpeg(1),
@ifset config-not-all
ffmpeg-all(1),
@end ifset
ffplay(1), ffprobe(1),
ffplay(1), ffprobe(1), ffserver(1),
ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)

View File

@@ -12,7 +12,7 @@
@chapter Synopsis
ffplay [@var{options}] [@file{input_url}]
ffplay [@var{options}] [@file{input_file}]
@chapter Description
@c man begin DESCRIPTION
@@ -60,24 +60,12 @@ Play @var{duration} seconds of audio/video.
see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
@item -bytes
Seek by bytes.
@item -seek_interval
Set custom interval, in seconds, for seeking using left/right keys. Default is 10 seconds.
@item -nodisp
Disable graphical display.
@item -noborder
Borderless window.
@item -volume
Set the startup volume. 0 means silence, 100 means no volume reduction or
amplification. Negative values are treated as 0, values above 100 are treated
as 100.
@item -f @var{fmt}
Force format.
@item -window_title @var{title}
Set window title (default is the input filename).
@item -left @var{title}
Set the x position for the left of the window (default is a centered window).
@item -top @var{title}
Set the y position for the top of the window (default is a centered window).
@item -loop @var{number}
Loops movie playback <number> times. 0 means forever.
@item -showmode @var{mode}
@@ -118,8 +106,8 @@ the input audio.
Use the option "-filters" to show all the available filters (including
sources and sinks).
@item -i @var{input_url}
Read @var{input_url}.
@item -i @var{input_file}
Read @var{input_file}.
@end table
@section Advanced options
@@ -297,7 +285,7 @@ Toggle full screen.
@ifset config-not-all
@url{ffplay-all.html,ffmpeg-all},
@end ifset
@url{ffmpeg.html,ffmpeg}, @url{ffprobe.html,ffprobe},
@url{ffmpeg.html,ffmpeg}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-utils.html,ffmpeg-utils},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@@ -316,7 +304,7 @@ ffplay(1),
@ifset config-not-all
ffplay-all(1),
@end ifset
ffmpeg(1), ffprobe(1),
ffmpeg(1), ffprobe(1), ffserver(1),
ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)

View File

@@ -12,7 +12,7 @@
@chapter Synopsis
ffprobe [@var{options}] [@file{input_url}]
ffprobe [@var{options}] [@file{input_file}]
@chapter Description
@c man begin DESCRIPTION
@@ -24,8 +24,8 @@ For example it can be used to check the format of the container used
by a multimedia stream and the format and type of each media stream
contained in it.
If a url is specified in input, ffprobe will try to open and
probe the url content. If the url cannot be opened or recognized as
If a filename is specified in input, ffprobe will try to open and
probe the file content. If the file cannot be opened or recognized as
a multimedia file, a positive exit code is returned.
ffprobe may be employed both as a standalone application or in
@@ -208,13 +208,6 @@ multimedia stream.
The information for each single frame is printed within a dedicated
section with name "FRAME" or "SUBTITLE".
@item -show_log @var{loglevel}
Show logging information from the decoder about each frame according to
the value set in @var{loglevel}, (see @code{-loglevel}). This option requires @code{-show_frames}.
The information for each log message is printed within a dedicated
section with name "LOG".
@item -show_streams
Show information about each media stream contained in the input
multimedia stream.
@@ -252,7 +245,7 @@ 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.
@@ -339,8 +332,8 @@ with name "PIXEL_FORMAT".
Force bitexact output, useful to produce output which is not dependent
on the specific build.
@item -i @var{input_url}
Read @var{input_url}.
@item -i @var{input_file}
Read @var{input_file}.
@end table
@c man end
@@ -471,7 +464,7 @@ Perform no escaping.
@end table
@item print_section, p
Print the section name at the beginning of each line if the value is
Print the section name at the begin of each line if the value is
@code{1}, disable it with value set to @code{0}. Default value is
@code{1}.
@@ -584,7 +577,7 @@ value is 0.
This is required for generating an XML file which can be validated
through an XSD file.
@item xsd_strict, x
@item xsd_compliant, x
If set to 1 perform more checks for ensuring that the output is XSD
compliant. Default value is 0.
This option automatically sets @option{fully_qualified} to 1.
@@ -653,7 +646,7 @@ DV, GXF and AVI timecodes are available in format metadata
@ifset config-not-all
@url{ffprobe-all.html,ffprobe-all},
@end ifset
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffserver.html,ffserver},
@url{ffmpeg-utils.html,ffmpeg-utils},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@@ -672,7 +665,7 @@ ffprobe(1),
@ifset config-not-all
ffprobe-all(1),
@end ifset
ffmpeg(1), ffplay(1),
ffmpeg(1), ffplay(1), ffserver(1),
ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)

View File

@@ -83,7 +83,6 @@
<xsd:complexType name="frameType">
<xsd:sequence>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="logs" type="ffprobe:logsType" minOccurs="0" maxOccurs="1"/>
<xsd:element name="side_data_list" type="ffprobe:frameSideDataListType" minOccurs="0" maxOccurs="1" />
</xsd:sequence>
@@ -120,25 +119,6 @@
<xsd:attribute name="interlaced_frame" type="xsd:int" />
<xsd:attribute name="top_field_first" type="xsd:int" />
<xsd:attribute name="repeat_pict" type="xsd:int" />
<xsd:attribute name="color_range" type="xsd:string"/>
<xsd:attribute name="color_space" type="xsd:string"/>
<xsd:attribute name="color_primaries" type="xsd:string"/>
<xsd:attribute name="color_transfer" type="xsd:string"/>
<xsd:attribute name="chroma_location" type="xsd:string"/>
</xsd:complexType>
<xsd:complexType name="logsType">
<xsd:sequence>
<xsd:element name="log" type="ffprobe:logType" minOccurs="1" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="logType">
<xsd:attribute name="context" type="xsd:string"/>
<xsd:attribute name="level" type="xsd:int" />
<xsd:attribute name="category" type="xsd:int" />
<xsd:attribute name="parent_context" type="xsd:string"/>
<xsd:attribute name="parent_category" type="xsd:int" />
<xsd:attribute name="message" type="xsd:string"/>
</xsd:complexType>
<xsd:complexType name="frameSideDataListType">
@@ -149,7 +129,6 @@
<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">
@@ -186,7 +165,6 @@
<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">
@@ -222,7 +200,6 @@
<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"/>

372
doc/ffserver.conf Normal file
View File

@@ -0,0 +1,372 @@
# 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
# Address on which the server is bound. Only useful if you have
# several network interfaces.
HTTPBindAddress 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
# MaxClients maximum limit.
MaxHTTPConnections 2000
# Number of simultaneous requests that can be handled. Since FFServer
# is very fast, it is more likely that you will want to leave this high
# and use MaxBandwidth, below.
MaxClients 1000
# This the maximum amount of kbit/sec that you are prepared to
# consume when streaming to clients.
MaxBandwidth 1000
# Access log file (uses standard Apache log file format)
# '-' is the standard output.
CustomLog -
##################################################################
# Definition of the live feeds. Each live feed contains one video
# and/or audio sequence coming from an ffmpeg encoder or another
# ffserver. This sequence may be encoded simultaneously with several
# codecs at several resolutions.
<Feed feed1.ffm>
# You must use 'ffmpeg' to send a live feed to ffserver. In this
# example, you can type:
#
# ffmpeg http://localhost:8090/feed1.ffm
# ffserver can also do time shifting. It means that it can stream any
# previously recorded live stream. The request should contain:
# "http://xxxx?date=[YYYY-MM-DDT][[HH:]MM:]SS[.m...]".You must specify
# a path where the feed is stored on disk. You also specify the
# maximum size of the feed, where zero means unlimited. Default:
# File=/tmp/feed_name.ffm FileMaxSize=5M
File /tmp/feed1.ffm
FileMaxSize 200K
# You could specify
# ReadOnlyFile /saved/specialvideo.ffm
# This marks the file as readonly and it will not be deleted or updated.
# Specify launch in order to start ffmpeg automatically.
# First ffmpeg must be defined with an appropriate path if needed,
# after that options can follow, but avoid adding the http:// field
#Launch ffmpeg
# Only allow connections from localhost to the feed.
ACL allow 127.0.0.1
</Feed>
##################################################################
# Now you can define each stream which will be generated from the
# original audio and video stream. Each format has a filename (here
# 'test1.mpg'). FFServer will send this stream when answering a
# request containing this filename.
<Stream test1.mpg>
# coming from live feed 'feed1'
Feed feed1.ffm
# Format of the stream : you can choose among:
# mpeg : MPEG-1 multiplexed video and audio
# mpegvideo : only MPEG-1 video
# mp2 : MPEG-2 audio (use AudioCodec to select layer 2 and 3 codec)
# ogg : Ogg format (Vorbis audio codec)
# rm : RealNetworks-compatible stream. Multiplexed audio and video.
# 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)
Format mpeg
# Bitrate for the audio stream. Codecs usually support only a few
# different bitrates.
AudioBitRate 32
# Number of audio channels: 1 = mono, 2 = stereo
AudioChannels 1
# 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.
AudioSampleRate 44100
# Bitrate for the video stream
VideoBitRate 64
# Ratecontrol buffer size
VideoBufferSize 40
# Number of frames per second
VideoFrameRate 3
# Size of the video frame: WxH (default: 160x128)
# The following abbreviations are defined: sqcif, qcif, cif, 4cif, qqvga,
# qvga, vga, svga, xga, uxga, qxga, sxga, qsxga, hsxga, wvga, wxga, wsxga,
# wuxga, woxga, wqsxga, wquxga, whsxga, whuxga, cga, ega, hd480, hd720,
# hd1080
VideoSize 160x128
# Transmit only intra frames (useful for low bitrates, but kills frame rate).
#VideoIntraOnly
# If non-intra only, an intra frame is transmitted every VideoGopSize
# frames. Video synchronization can only begin at an intra frame.
VideoGopSize 12
# More MPEG-4 parameters
# VideoHighQuality
# Video4MotionVector
# Choose your codecs:
#AudioCodec mp2
#VideoCodec mpeg1video
# Suppress audio
#NoAudio
# Suppress video
#NoVideo
#VideoQMin 3
#VideoQMax 31
# Set this to the number of seconds backwards in time to start. Note that
# most players will buffer 5-10 seconds of video, and also you need to allow
# for a keyframe to appear in the data stream.
#Preroll 15
# ACL:
# You can allow ranges of addresses (or single addresses)
#ACL ALLOW <first address> <last address>
# You can deny ranges of addresses (or single addresses)
#ACL DENY <first address> <last address>
# You can repeat the ACL allow/deny as often as you like. It is on a per
# stream basis. The first match defines the action. If there are no matches,
# then the default is the inverse of the last ACL statement.
#
# Thus 'ACL allow localhost' only allows access from localhost.
# 'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and
# allow everybody else.
</Stream>
##################################################################
# Example streams
# Multipart JPEG
#<Stream test.mjpg>
#Feed feed1.ffm
#Format mpjpeg
#VideoFrameRate 2
#VideoIntraOnly
#NoAudio
#Strict -1
#</Stream>
# Single JPEG
#<Stream test.jpg>
#Feed feed1.ffm
#Format jpeg
#VideoFrameRate 2
#VideoIntraOnly
##VideoSize 352x240
#NoAudio
#Strict -1
#</Stream>
# Flash
#<Stream test.swf>
#Feed feed1.ffm
#Format swf
#VideoFrameRate 2
#VideoIntraOnly
#NoAudio
#</Stream>
# ASF compatible
<Stream test.asf>
Feed feed1.ffm
Format asf
VideoFrameRate 15
VideoSize 352x240
VideoBitRate 256
VideoBufferSize 40
VideoGopSize 30
AudioBitRate 64
StartSendOnKey
</Stream>
# MP3 audio
#<Stream test.mp3>
#Feed feed1.ffm
#Format mp2
#AudioCodec mp3
#AudioBitRate 64
#AudioChannels 1
#AudioSampleRate 44100
#NoVideo
#</Stream>
# Ogg Vorbis audio
#<Stream test.ogg>
#Feed feed1.ffm
#Metadata title "Stream title"
#AudioBitRate 64
#AudioChannels 2
#AudioSampleRate 44100
#NoVideo
#</Stream>
# Real with audio only at 32 kbits
#<Stream test.ra>
#Feed feed1.ffm
#Format rm
#AudioBitRate 32
#NoVideo
#NoAudio
#</Stream>
# Real with audio and video at 64 kbits
#<Stream test.rm>
#Feed feed1.ffm
#Format rm
#AudioBitRate 32
#VideoBitRate 128
#VideoFrameRate 25
#VideoGopSize 25
#NoAudio
#</Stream>
##################################################################
# A stream coming from a file: you only need to set the input
# filename and optionally a new format. Supported conversions:
# AVI -> ASF
#<Stream file.rm>
#File "/usr/local/httpd/htdocs/tlive.rm"
#NoAudio
#</Stream>
#<Stream file.asf>
#File "/usr/local/httpd/htdocs/test.asf"
#NoAudio
#Metadata author "Me"
#Metadata copyright "Super MegaCorp"
#Metadata title "Test stream from disk"
#Metadata comment "Test comment"
#</Stream>
##################################################################
# RTSP examples
#
# You can access this stream with the RTSP URL:
# rtsp://localhost:5454/test1-rtsp.mpg
#
# A non-standard RTSP redirector is also created. Its URL is:
# http://localhost:8090/test1-rtsp.rtsp
#<Stream test1-rtsp.mpg>
#Format rtp
#File "/usr/local/httpd/htdocs/test1.mpg"
#</Stream>
# Transcode an incoming live feed to another live feed,
# using libx264 and video presets
#<Stream live.h264>
#Format rtp
#Feed feed1.ffm
#VideoCodec libx264
#VideoFrameRate 24
#VideoBitRate 100
#VideoSize 480x272
#AVPresetVideo default
#AVPresetVideo baseline
#AVOptionVideo flags +global_header
#
#AudioCodec libfaac
#AudioBitRate 32
#AudioChannels 2
#AudioSampleRate 22050
#AVOptionAudio flags +global_header
#</Stream>
##################################################################
# SDP/multicast examples
#
# If you want to send your stream in multicast, you must set the
# multicast address with MulticastAddress. The port and the TTL can
# also be set.
#
# An SDP file is automatically generated by ffserver by adding the
# 'sdp' extension to the stream name (here
# http://localhost:8090/test1-sdp.sdp). You should usually give this
# file to your player to play the stream.
#
# The 'NoLoop' option can be used to avoid looping when the stream is
# terminated.
#<Stream test1-sdp.mpg>
#Format rtp
#File "/usr/local/httpd/htdocs/test1.mpg"
#MulticastAddress 224.124.0.1
#MulticastPort 5000
#MulticastTTL 16
#NoLoop
#</Stream>
##################################################################
# Special streams
# Server status
<Stream stat.html>
Format status
# Only allow local people to get the status
ACL allow localhost
ACL allow 192.168.0.0 192.168.255.255
#FaviconURL http://pond1.gladstonefamily.net:8080/favicon.ico
</Stream>
# Redirect index.html to the appropriate site
<Redirect index.html>
URL http://www.ffmpeg.org/
</Redirect>

923
doc/ffserver.texi Normal file
View File

@@ -0,0 +1,923 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle ffserver Documentation
@titlepage
@center @titlefont{ffserver Documentation}
@end titlepage
@top
@contents
@chapter Synopsis
ffserver [@var{options}]
@chapter Description
@c man begin DESCRIPTION
@command{ffserver} is a streaming server for both audio and video.
It supports several live feeds, streaming from files and time shifting
on live feeds. You can seek to positions in the past on each live
feed, provided you specify a big enough feed storage.
@command{ffserver} is configured through a configuration file, which
is read at startup. If not explicitly specified, it will read from
@file{/etc/ffserver.conf}.
@command{ffserver} receives prerecorded files or FFM streams from some
@command{ffmpeg} instance as input, then streams them over
RTP/RTSP/HTTP.
An @command{ffserver} instance will listen on some port as specified
in the configuration file. You can launch one or more instances of
@command{ffmpeg} and send one or more FFM streams to the port where
ffserver is expecting to receive them. Alternately, you can make
@command{ffserver} launch such @command{ffmpeg} instances at startup.
Input streams are called feeds, and each one is specified by a
@code{<Feed>} section in the configuration file.
For each feed you can have different output streams in various
formats, each one specified by a @code{<Stream>} section in the
configuration file.
@chapter Detailed description
@command{ffserver} works by forwarding streams encoded by
@command{ffmpeg}, or pre-recorded streams which are read from disk.
Precisely, @command{ffserver} acts as an HTTP server, accepting POST
requests from @command{ffmpeg} to acquire the stream to publish, and
serving RTSP clients or HTTP clients GET requests with the stream
media content.
A feed is an @ref{FFM} stream created by @command{ffmpeg}, and sent to
a port where @command{ffserver} is listening.
Each feed is identified by a unique name, corresponding to the name
of the resource published on @command{ffserver}, and is configured by
a dedicated @code{Feed} section in the configuration file.
The feed publish URL is given by:
@example
http://@var{ffserver_ip_address}:@var{http_port}/@var{feed_name}
@end example
where @var{ffserver_ip_address} is the IP address of the machine where
@command{ffserver} is installed, @var{http_port} is the port number of
the HTTP server (configured through the @option{HTTPPort} option), and
@var{feed_name} is the name of the corresponding feed defined in the
configuration file.
Each feed is associated to a file which is stored on disk. This stored
file is used to send pre-recorded data to a player as fast as
possible when new content is added in real-time to the stream.
A "live-stream" or "stream" is a resource published by
@command{ffserver}, and made accessible through the HTTP protocol to
clients.
A stream can be connected to a feed, or to a file. In the first case,
the published stream is forwarded from the corresponding feed
generated by a running instance of @command{ffmpeg}, in the second
case the stream is read from a pre-recorded file.
Each stream is identified by a unique name, corresponding to the name
of the resource served by @command{ffserver}, and is configured by
a dedicated @code{Stream} section in the configuration file.
The stream access HTTP URL is given by:
@example
http://@var{ffserver_ip_address}:@var{http_port}/@var{stream_name}[@var{options}]
@end example
The stream access RTSP URL is given by:
@example
http://@var{ffserver_ip_address}:@var{rtsp_port}/@var{stream_name}[@var{options}]
@end example
@var{stream_name} is the name of the corresponding stream defined in
the configuration file. @var{options} is a list of options specified
after the URL which affects how the stream is served by
@command{ffserver}. @var{http_port} and @var{rtsp_port} are the HTTP
and RTSP ports configured with the options @var{HTTPPort} and
@var{RTSPPort} respectively.
In case the stream is associated to a feed, the encoding parameters
must be configured in the stream configuration. They are sent to
@command{ffmpeg} when setting up the encoding. This allows
@command{ffserver} to define the encoding parameters used by
the @command{ffmpeg} encoders.
The @command{ffmpeg} @option{override_ffserver} commandline option
allows one to override the encoding parameters set by the server.
Multiple streams can be connected to the same feed.
For example, you can have a situation described by the following
graph:
@verbatim
_________ __________
| | | |
ffmpeg 1 -----| feed 1 |-----| stream 1 |
\ |_________|\ |__________|
\ \
\ \ __________
\ \ | |
\ \| stream 2 |
\ |__________|
\
\ _________ __________
\ | | | |
\| feed 2 |-----| stream 3 |
|_________| |__________|
_________ __________
| | | |
ffmpeg 2 -----| feed 3 |-----| stream 4 |
|_________| |__________|
_________ __________
| | | |
| file 1 |-----| stream 5 |
|_________| |__________|
@end verbatim
@anchor{FFM}
@section FFM, FFM2 formats
FFM and FFM2 are formats used by ffserver. They allow storing a wide variety of
video and audio streams and encoding options, and can store a moving time segment
of an infinite movie or a whole movie.
FFM is version specific, and there is limited compatibility of FFM files
generated by one version of ffmpeg/ffserver and another version of
ffmpeg/ffserver. It may work but it is not guaranteed to work.
FFM2 is extensible while maintaining compatibility and should work between
differing versions of tools. FFM2 is the default.
@section Status stream
@command{ffserver} supports an HTTP interface which exposes the
current status of the server.
Simply point your browser to the address of the special status stream
specified in the configuration file.
For example if you have:
@example
<Stream status.html>
Format status
# Only allow local people to get the status
ACL allow localhost
ACL allow 192.168.0.0 192.168.255.255
</Stream>
@end example
then the server will post a page with the status information when
the special stream @file{status.html} is requested.
@section How do I make it work?
As a simple test, just run the following two command lines where INPUTFILE
is some file which you can decode with ffmpeg:
@example
ffserver -f doc/ffserver.conf &
ffmpeg -i INPUTFILE http://localhost:8090/feed1.ffm
@end example
At this point you should be able to go to your Windows machine and fire up
Windows Media Player (WMP). Go to Open URL and enter
@example
http://<linuxbox>:8090/test.asf
@end example
You should (after a short delay) see video and hear audio.
WARNING: trying to stream test1.mpg doesn't work with WMP as it tries to
transfer the entire file before starting to play.
The same is true of AVI files.
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 else can it do?
You can replay video from .ffm files that was recorded earlier.
However, there are a number of caveats, including the fact that the
ffserver parameters must match the original parameters used to record the
file. If they do not, then ffserver deletes the file before recording into it.
(Now that I write this, it seems broken).
You can fiddle with many of the codec choices and encoding parameters, and
there are a bunch more parameters that you cannot control. Post a message
to the mailing list if there are some 'must have' parameters. Look in
ffserver.conf for a list of the currently available controls.
It will automatically generate the ASX or RAM files that are often used
in browsers. These files are actually redirections to the underlying ASF
or RM file. The reason for this is that the browser often fetches the
entire file before starting up the external viewer. The redirection files
are very small and can be transferred quickly. [The stream itself is
often 'infinite' and thus the browser tries to download it and never
finishes.]
@section Tips
* When you connect to a live stream, most players (WMP, RA, etc) want to
buffer a certain number of seconds of material so that they can display the
signal continuously. However, ffserver (by default) starts sending data
in realtime. This means that there is a pause of a few seconds while the
buffering is being done by the player. The good news is that this can be
cured by adding a '?buffer=5' to the end of the URL. This means that the
stream should start 5 seconds in the past -- and so the first 5 seconds
of the stream are sent as fast as the network will allow. It will then
slow down to real time. This noticeably improves the startup experience.
You can also add a 'Preroll 15' statement into the ffserver.conf that will
add the 15 second prebuffering on all requests that do not otherwise
specify a time. In addition, ffserver will skip frames until a key_frame
is found. This further reduces the startup delay by not transferring data
that will be discarded.
@section Why does the ?buffer / Preroll stop working after a time?
It turns out that (on my machine at least) the number of frames successfully
grabbed is marginally less than the number that ought to be grabbed. This
means that the timestamp in the encoded data stream gets behind realtime.
This means that if you say 'Preroll 10', then when the stream gets 10
or more seconds behind, there is no Preroll left.
Fixing this requires a change in the internals of how timestamps are
handled.
@section Does the @code{?date=} stuff work.
Yes (subject to the limitation outlined above). Also note that whenever you
start ffserver, it deletes the ffm file (if any parameters have changed),
thus wiping out what you had recorded before.
The format of the @code{?date=xxxxxx} is fairly flexible. You should use one
of the following formats (the 'T' is literal):
@example
* YYYY-MM-DDTHH:MM:SS (localtime)
* YYYY-MM-DDTHH:MM:SSZ (UTC)
@end example
You can omit the YYYY-MM-DD, and then it refers to the current day. However
note that @samp{?date=16:00:00} refers to 16:00 on the current day -- this
may be in the future and so is unlikely to be useful.
You use this by adding the ?date= to the end of the URL for the stream.
For example: @samp{http://localhost:8080/test.asf?date=2002-07-26T23:05:00}.
@c man end
@chapter Options
@c man begin OPTIONS
@include fftools-common-opts.texi
@section Main options
@table @option
@item -f @var{configfile}
Read configuration file @file{configfile}. If not specified it will
read by default from @file{/etc/ffserver.conf}.
@item -n
Enable no-launch mode. This option disables all the @code{Launch}
directives within the various @code{<Feed>} sections. Since
@command{ffserver} will not launch any @command{ffmpeg} instances, you
will have to launch them manually.
@item -d
Enable debug mode. This option increases log verbosity, and directs
log messages to stdout. When specified, the @option{CustomLog} option
is ignored.
@end table
@chapter Configuration file syntax
@command{ffserver} reads a configuration file containing global
options and settings for each stream and feed.
The configuration file consists of global options and dedicated
sections, which must be introduced by "<@var{SECTION_NAME}
@var{ARGS}>" on a separate line and must be terminated by a line in
the form "</@var{SECTION_NAME}>". @var{ARGS} is optional.
Currently the following sections are recognized: @samp{Feed},
@samp{Stream}, @samp{Redirect}.
A line starting with @code{#} is ignored and treated as a comment.
Name of options and sections are case-insensitive.
@section ACL syntax
An ACL (Access Control List) specifies the address which are allowed
to access a given stream, or to write a given feed.
It accepts the folling forms
@itemize
@item
Allow/deny access to @var{address}.
@example
ACL ALLOW <address>
ACL DENY <address>
@end example
@item
Allow/deny access to ranges of addresses from @var{first_address} to
@var{last_address}.
@example
ACL ALLOW <first_address> <last_address>
ACL DENY <first_address> <last_address>
@end example
@end itemize
You can repeat the ACL allow/deny as often as you like. It is on a per
stream basis. The first match defines the action. If there are no matches,
then the default is the inverse of the last ACL statement.
Thus 'ACL allow localhost' only allows access from localhost.
'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and
allow everybody else.
@section Global options
@table @option
@item HTTPPort @var{port_number}
@item Port @var{port_number}
@item RTSPPort @var{port_number}
@var{HTTPPort} sets the HTTP server listening TCP port number,
@var{RTSPPort} sets the RTSP server listening TCP port number.
@var{Port} is the equivalent of @var{HTTPPort} and is deprecated.
You must select a different port from your standard HTTP web server if
it is running on the same computer.
If not specified, no corresponding server will be created.
@item HTTPBindAddress @var{ip_address}
@item BindAddress @var{ip_address}
@item RTSPBindAddress @var{ip_address}
Set address on which the HTTP/RTSP server is bound. Only useful if you
have several network interfaces.
@var{BindAddress} is the equivalent of @var{HTTPBindAddress} and is
deprecated.
@item MaxHTTPConnections @var{n}
Set number of simultaneous HTTP connections that can be handled. It
has to be defined @emph{before} the @option{MaxClients} parameter,
since it defines the @option{MaxClients} maximum limit.
Default value is 2000.
@item MaxClients @var{n}
Set number of simultaneous requests that can be handled. Since
@command{ffserver} is very fast, it is more likely that you will want
to leave this high and use @option{MaxBandwidth}.
Default value is 5.
@item MaxBandwidth @var{kbps}
Set the maximum amount of kbit/sec that you are prepared to consume
when streaming to clients.
Default value is 1000.
@item CustomLog @var{filename}
Set access log file (uses standard Apache log file format). '-' is the
standard output.
If not specified @command{ffserver} will produce no log.
In case the commandline option @option{-d} is specified this option is
ignored, and the log is written to standard output.
@item NoDaemon
Set no-daemon mode. This option is currently ignored since now
@command{ffserver} will always work in no-daemon mode, and is
deprecated.
@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
A Feed section defines a feed provided to @command{ffserver}.
Each live feed contains one video and/or audio sequence coming from an
@command{ffmpeg} encoder or another @command{ffserver}. This sequence
may be encoded simultaneously with several codecs at several
resolutions.
A feed instance specification is introduced by a line in the form:
@example
<Feed FEED_FILENAME>
@end example
where @var{FEED_FILENAME} specifies the unique name of the FFM stream.
The following options are recognized within a Feed section.
@table @option
@item File @var{filename}
@item ReadOnlyFile @var{filename}
Set the path where the feed file is stored on disk.
If not specified, the @file{/tmp/FEED.ffm} is assumed, where
@var{FEED} is the feed name.
If @option{ReadOnlyFile} is used the file is marked as read-only and
it will not be deleted or updated.
@item Truncate
Truncate the feed file, rather than appending to it. By default
@command{ffserver} will append data to the file, until the maximum
file size value is reached (see @option{FileMaxSize} option).
@item FileMaxSize @var{size}
Set maximum size of the feed file in bytes. 0 means unlimited. The
postfixes @code{K} (2^10), @code{M} (2^20), and @code{G} (2^30) are
recognized.
Default value is 5M.
@item Launch @var{args}
Launch an @command{ffmpeg} command when creating @command{ffserver}.
@var{args} must be a sequence of arguments to be provided to an
@command{ffmpeg} instance. The first provided argument is ignored, and
it is replaced by a path with the same dirname of the @command{ffserver}
instance, followed by the remaining argument and terminated with a
path corresponding to the feed.
When the launched process exits, @command{ffserver} will launch
another program instance.
In case you need a more complex @command{ffmpeg} configuration,
e.g. if you need to generate multiple FFM feeds with a single
@command{ffmpeg} instance, you should launch @command{ffmpeg} by hand.
This option is ignored in case the commandline option @option{-n} is
specified.
@item ACL @var{spec}
Specify the list of IP address which are allowed or denied to write
the feed. Multiple ACL options can be specified.
@end table
@section Stream section
A Stream section defines a stream provided by @command{ffserver}, and
identified by a single name.
The stream is sent when answering a request containing the stream
name.
A stream section must be introduced by the line:
@example
<Stream STREAM_NAME>
@end example
where @var{STREAM_NAME} specifies the unique name of the stream.
The following options are recognized within a Stream section.
Encoding options are marked with the @emph{encoding} tag, and they are
used to set the encoding parameters, and are mapped to libavcodec
encoding options. Not all encoding options are supported, in
particular it is not possible to set encoder private options. In order
to override the encoding options specified by @command{ffserver}, you
can use the @command{ffmpeg} @option{override_ffserver} commandline
option.
Only one of the @option{Feed} and @option{File} options should be set.
@table @option
@item Feed @var{feed_name}
Set the input feed. @var{feed_name} must correspond to an existing
feed defined in a @code{Feed} section.
When this option is set, encoding options are used to setup the
encoding operated by the remote @command{ffmpeg} process.
@item File @var{filename}
Set the filename of the pre-recorded input file to stream.
When this option is set, encoding options are ignored and the input
file content is re-streamed as is.
@item Format @var{format_name}
Set the format of the output stream.
Must be the name of a format recognized by FFmpeg. If set to
@samp{status}, it is treated as a status stream.
@item InputFormat @var{format_name}
Set input format. If not specified, it is automatically guessed.
@item Preroll @var{n}
Set this to the number of seconds backwards in time to start. Note that
most players will buffer 5-10 seconds of video, and also you need to allow
for a keyframe to appear in the data stream.
Default value is 0.
@item StartSendOnKey
Do not send stream until it gets the first key frame. By default
@command{ffserver} will send data immediately.
@item MaxTime @var{n}
Set the number of seconds to run. This value set the maximum duration
of the stream a client will be able to receive.
A value of 0 means that no limit is set on the stream duration.
@item ACL @var{spec}
Set ACL for the stream.
@item DynamicACL @var{spec}
@item RTSPOption @var{option}
@item MulticastAddress @var{address}
@item MulticastPort @var{port}
@item MulticastTTL @var{integer}
@item NoLoop
@item FaviconURL @var{url}
Set favicon (favourite icon) for the server status page. It is ignored
for regular streams.
@item Author @var{value}
@item Comment @var{value}
@item Copyright @var{value}
@item Title @var{value}
Set metadata corresponding to the option. All these options are
deprecated in favor of @option{Metadata}.
@item Metadata @var{key} @var{value}
Set metadata value on the output stream.
@item 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.
@item AudioCodec @var{codec_name} (@emph{encoding,audio})
Set audio codec.
@item AudioBitRate @var{rate} (@emph{encoding,audio})
Set bitrate for the audio stream in kbits per second.
@item AudioChannels @var{n} (@emph{encoding,audio})
Set number of audio channels.
@item AudioSampleRate @var{n} (@emph{encoding,audio})
Set sampling frequency for audio. When using low bitrates, you should
lower this frequency to 22050 or 11025. The supported frequencies
depend on the selected audio codec.
@item AVOptionAudio [@var{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 AVPresetAudio @var{preset} (@emph{encoding,audio})
Set preset for audio stream.
@item VideoCodec @var{codec_name} (@emph{encoding,video})
Set video codec.
@item VideoBitRate @var{n} (@emph{encoding,video})
Set bitrate for the video stream in kbits per second.
@item VideoBitRateRange @var{range} (@emph{encoding,video})
Set video bitrate range.
A range must be specified in the form @var{minrate}-@var{maxrate}, and
specifies the @option{minrate} and @option{maxrate} encoding options
expressed in kbits per second.
@item VideoBitRateRangeTolerance @var{n} (@emph{encoding,video})
Set video bitrate tolerance in kbits per second.
@item PixelFormat @var{pixel_format} (@emph{encoding,video})
Set video pixel format.
@item Debug @var{integer} (@emph{encoding,video})
Set video @option{debug} encoding option.
@item Strict @var{integer} (@emph{encoding,video})
Set video @option{strict} encoding option.
@item VideoBufferSize @var{n} (@emph{encoding,video})
Set ratecontrol buffer size, expressed in KB.
@item VideoFrameRate @var{n} (@emph{encoding,video})
Set number of video frames per second.
@item VideoSize (@emph{encoding,video})
Set size of the video frame, must be an abbreviation or in the form
@var{W}x@var{H}. See @ref{video size syntax,,the Video size section
in the ffmpeg-utils(1) manual,ffmpeg-utils}.
Default value is @code{160x128}.
@item VideoIntraOnly (@emph{encoding,video})
Transmit only intra frames (useful for low bitrates, but kills frame rate).
@item VideoGopSize @var{n} (@emph{encoding,video})
If non-intra only, an intra frame is transmitted every VideoGopSize
frames. Video synchronization can only begin at an intra frame.
@item VideoTag @var{tag} (@emph{encoding,video})
Set video tag.
@item VideoHighQuality (@emph{encoding,video})
@item Video4MotionVector (@emph{encoding,video})
@item BitExact (@emph{encoding,video})
Set bitexact encoding flag.
@item IdctSimple (@emph{encoding,video})
Set simple IDCT algorithm.
@item Qscale @var{n} (@emph{encoding,video})
Enable constant quality encoding, and set video qscale (quantization
scale) value, expressed in @var{n} QP units.
@item VideoQMin @var{n} (@emph{encoding,video})
@item VideoQMax @var{n} (@emph{encoding,video})
Set video qmin/qmax.
@item VideoQDiff @var{integer} (@emph{encoding,video})
Set video @option{qdiff} encoding option.
@item LumiMask @var{float} (@emph{encoding,video})
@item DarkMask @var{float} (@emph{encoding,video})
Set @option{lumi_mask}/@option{dark_mask} encoding options.
@item AVOptionVideo [@var{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 AVPresetVideo @var{preset} (@emph{encoding,video})
Set preset for video stream.
@var{preset} must be the path of a preset file.
@end table
@subsection Server status stream
A server status stream is a special stream which is used to show
statistics about the @command{ffserver} operations.
It must be specified setting the option @option{Format} to
@samp{status}.
@section Redirect section
A redirect section specifies where to redirect the requested URL to
another page.
A redirect section must be introduced by the line:
@example
<Redirect NAME>
@end example
where @var{NAME} is the name of the page which should be redirected.
It only accepts the option @option{URL}, which specify the redirection
URL.
@chapter Stream examples
@itemize
@item
Multipart JPEG
@example
<Stream test.mjpg>
Feed feed1.ffm
Format mpjpeg
VideoFrameRate 2
VideoIntraOnly
NoAudio
Strict -1
</Stream>
@end example
@item
Single JPEG
@example
<Stream test.jpg>
Feed feed1.ffm
Format jpeg
VideoFrameRate 2
VideoIntraOnly
VideoSize 352x240
NoAudio
Strict -1
</Stream>
@end example
@item
Flash
@example
<Stream test.swf>
Feed feed1.ffm
Format swf
VideoFrameRate 2
VideoIntraOnly
NoAudio
</Stream>
@end example
@item
ASF compatible
@example
<Stream test.asf>
Feed feed1.ffm
Format asf
VideoFrameRate 15
VideoSize 352x240
VideoBitRate 256
VideoBufferSize 40
VideoGopSize 30
AudioBitRate 64
StartSendOnKey
</Stream>
@end example
@item
MP3 audio
@example
<Stream test.mp3>
Feed feed1.ffm
Format mp2
AudioCodec mp3
AudioBitRate 64
AudioChannels 1
AudioSampleRate 44100
NoVideo
</Stream>
@end example
@item
Ogg Vorbis audio
@example
<Stream test.ogg>
Feed feed1.ffm
Metadata title "Stream title"
AudioBitRate 64
AudioChannels 2
AudioSampleRate 44100
NoVideo
</Stream>
@end example
@item
Real with audio only at 32 kbits
@example
<Stream test.ra>
Feed feed1.ffm
Format rm
AudioBitRate 32
NoVideo
</Stream>
@end example
@item
Real with audio and video at 64 kbits
@example
<Stream test.rm>
Feed feed1.ffm
Format rm
AudioBitRate 32
VideoBitRate 128
VideoFrameRate 25
VideoGopSize 25
</Stream>
@end example
@item
For stream coming from a file: you only need to set the input filename
and optionally a new format.
@example
<Stream file.rm>
File "/usr/local/httpd/htdocs/tlive.rm"
NoAudio
</Stream>
@end example
@example
<Stream file.asf>
File "/usr/local/httpd/htdocs/test.asf"
NoAudio
Metadata author "Me"
Metadata copyright "Super MegaCorp"
Metadata title "Test stream from disk"
Metadata comment "Test comment"
</Stream>
@end example
@end itemize
@c man end
@include config.texi
@ifset config-all
@ifset config-avutil
@include utils.texi
@end ifset
@ifset config-avcodec
@include codecs.texi
@include bitstream_filters.texi
@end ifset
@ifset config-avformat
@include formats.texi
@include protocols.texi
@end ifset
@ifset config-avdevice
@include devices.texi
@end ifset
@ifset config-swresample
@include resampler.texi
@end ifset
@ifset config-swscale
@include scaler.texi
@end ifset
@ifset config-avfilter
@include filters.texi
@end ifset
@end ifset
@chapter See Also
@ifhtml
@ifset config-all
@url{ffserver.html,ffserver},
@end ifset
@ifset config-not-all
@url{ffserver-all.html,ffserver-all},
@end ifset
the @file{doc/ffserver.conf} example,
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg-utils.html,ffmpeg-utils},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@url{ffmpeg-codecs.html,ffmpeg-codecs},
@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters},
@url{ffmpeg-formats.html,ffmpeg-formats},
@url{ffmpeg-devices.html,ffmpeg-devices},
@url{ffmpeg-protocols.html,ffmpeg-protocols},
@url{ffmpeg-filters.html,ffmpeg-filters}
@end ifhtml
@ifnothtml
@ifset config-all
ffserver(1),
@end ifset
@ifset config-not-all
ffserver-all(1),
@end ifset
the @file{doc/ffserver.conf} example, ffmpeg(1), ffplay(1), ffprobe(1),
ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffserver
@settitle ffserver video server
@end ignore
@bye

View File

@@ -42,20 +42,10 @@ 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
stream number @var{stream_index} of this type. Otherwise, it matches all
streams of this type.
@item p:@var{program_id}[:@var{stream_index}] or p:@var{program_id}[:@var{stream_type}[:@var{stream_index}]] or
p:@var{program_id}:m:@var{key}[:@var{value}]
In first version, if @var{stream_index} is given, then it matches the stream with number @var{stream_index}
@item p:@var{program_id}[:@var{stream_index}]
If @var{stream_index} is given, then it matches the stream with number @var{stream_index}
in the program with the id @var{program_id}. Otherwise, it matches all streams in the
program. In the second version, @var{stream_type} is one of following: 'v' for video, 'a' for audio, 's'
for subtitle, 'd' for data. If @var{stream_index} is also given, then it matches
stream number @var{stream_index} of this type in the program with the id @var{program_id}.
Otherwise, if only @var{stream_type} is given, it matches all
streams of this type in the program with the id @var{program_id}.
In the third version matches streams in the program with the id @var{program_id} 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.
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}]
@@ -120,12 +110,6 @@ Show version.
@item -formats
Show available formats (including devices).
@item -demuxers
Show available demuxers.
@item -muxers
Show available muxers.
@item -devices
Show available devices.
@@ -163,7 +147,7 @@ Show channel names and standard channel layouts.
Show recognized color names.
@item -sources @var{device}[,@var{opt1}=@var{val1}[,@var{opt2}=@var{val2}]...]
Show autodetected sources of the input device.
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
@@ -178,34 +162,24 @@ The returned list cannot be assumed to be always complete.
ffmpeg -sinks pulse,server=192.168.0.4
@end example
@item -loglevel [@var{flags}+]@var{loglevel} | -v [@var{flags}+]@var{loglevel}
Set logging level and flags used by the library.
The optional @var{flags} prefix can consist of the following values:
@table @samp
@item repeat
Indicates that repeated log output should not be compressed to the first line
and the "Last message repeated n times" line will be omitted.
@item level
Indicates that log output should add a @code{[level]} prefix to each message
line. This can be used as an alternative to log coloring, e.g. when dumping the
log to file.
@end table
Flags can also be used alone by adding a '+'/'-' prefix to set/reset a single
flag without affecting other @var{flags} or changing @var{loglevel}. When
setting both @var{flags} and @var{loglevel}, a '+' separator is expected
between the last @var{flags} value and before @var{loglevel}.
@item -loglevel [repeat+]@var{loglevel} | -v [repeat+]@var{loglevel}
Set the logging level used by the library.
Adding "repeat+" indicates that repeated log output should not be compressed
to the first line and the "Last message repeated n times" line will be
omitted. "repeat" can also be used alone.
If "repeat" is used alone, and with no prior loglevel set, the default
loglevel will be used. If multiple loglevel parameters are given, using
'repeat' will not change the loglevel.
@var{loglevel} is a string or a number containing one of the following values:
@table @samp
@item quiet, -8
Show nothing at all; be silent.
@item panic, 0
Only show fatal errors which could lead the process to crash, such as
an assertion failure. This is not currently used for anything.
and assert failure. This is not currently used for anything.
@item fatal, 8
Only show fatal errors. These are errors after which the process absolutely
cannot continue.
cannot continue after.
@item error, 16
Show all errors, including ones which can be recovered from.
@item warning, 24
@@ -221,24 +195,13 @@ Show everything, including debugging information.
@item trace, 56
@end table
For example to enable repeated log output, add the @code{level} prefix, and set
@var{loglevel} to @code{verbose}:
@example
ffmpeg -loglevel repeat+level+verbose -i input output
@end example
Another example that enables repeated log output without affecting current
state of @code{level} prefix flag or @var{loglevel}:
@example
ffmpeg [...] -loglevel +repeat
@end example
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
@@ -346,6 +309,51 @@ Possible flags for this option are:
@item k8
@end table
@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
@item -opencl_options options (@emph{global})
Set OpenCL environment options. This option is only available when
FFmpeg has been compiled with @code{--enable-opencl}.
@var{options} must be a list of @var{key}=@var{value} option pairs
separated by ':'. See the ``OpenCL Options'' section in the
ffmpeg-utils manual for the list of supported options.
@end table
@section AVOptions

View File

@@ -5,7 +5,7 @@ This document explains guidelines that should be observed (or ignored with
good reason) when writing filters for libavfilter.
In this document, the word “frame” indicates either a video frame or a group
of audio samples, as stored in an AVFrame structure.
of audio samples, as stored in an AVFilterBuffer structure.
Format negotiation
@@ -35,31 +35,32 @@ Format negotiation
to set the formats supported on another.
Frame references ownership and permissions
==========================================
Buffer references ownership and permissions
===========================================
Principle
---------
Audio and video data are voluminous; the frame and frame reference
Audio and video data are voluminous; the buffer and buffer reference
mechanism is intended to avoid, as much as possible, expensive copies of
that data while still allowing the filters to produce correct results.
The data is stored in buffers represented by AVFrame structures.
Several references can point to the same frame buffer; the buffer is
automatically deallocated once all corresponding references have been
destroyed.
The data is stored in buffers represented by AVFilterBuffer structures.
They must not be accessed directly, but through references stored in
AVFilterBufferRef structures. Several references can point to the
same buffer; the buffer is automatically deallocated once all
corresponding references have been destroyed.
The characteristics of the data (resolution, sample rate, etc.) are
stored in the reference; different references for the same buffer can
show different characteristics. In particular, a video reference can
point to only a part of a video buffer.
A reference is usually obtained as input to the filter_frame method or
requested using the ff_get_video_buffer or ff_get_audio_buffer
functions. A new reference on an existing buffer can be created with
av_frame_ref(). A reference is destroyed using
the av_frame_free() function.
A reference is usually obtained as input to the start_frame or
filter_frame method or requested using the ff_get_video_buffer or
ff_get_audio_buffer functions. A new reference on an existing buffer can
be created with the avfilter_ref_buffer. A reference is destroyed using
the avfilter_unref_bufferp function.
Reference ownership
-------------------
@@ -72,13 +73,17 @@ Frame references ownership and permissions
Here are the (fairly obvious) rules for reference ownership:
* A reference received by the filter_frame method belongs to the
corresponding filter.
* A reference received by the filter_frame method (or its start_frame
deprecated version) belongs to the corresponding filter.
* A reference passed to ff_filter_frame is given away and must no longer
be used.
Special exception: for video references: the reference may be used
internally for automatic copying and must not be destroyed before
end_frame; it can be given away to ff_start_frame.
* A reference created with av_frame_ref() belongs to the code that
* A reference passed to ff_filter_frame (or the deprecated
ff_start_frame) is given away and must no longer be used.
* A reference created with avfilter_ref_buffer belongs to the code that
created it.
* A reference obtained with ff_get_video_buffer or ff_get_audio_buffer
@@ -90,32 +95,89 @@ Frame references ownership and permissions
Link reference fields
---------------------
The AVFilterLink structure has a few AVFrame fields.
partial_buf is used by libavfilter internally and must not be accessed
by filters.
fifo contains frames queued in the filter's input. They belong to the
framework until they are taken by the filter.
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
and must not be accessed by filters.
Reference permissions
---------------------
Since the same frame data can be shared by several frames, modifying may
have unintended consequences. A frame is considered writable if only one
reference to it exists. The code owning that reference it then allowed
to modify the data.
The AVFilterBufferRef structure has a perms field that describes what
the code that owns the reference is allowed to do to the buffer data.
Different references for the same buffer can have different permissions.
A filter can check if a frame is writable by using the
av_frame_is_writable() function.
For video filters that implement the deprecated
start_frame/draw_slice/end_frame API, the permissions only apply to the
parts of the buffer that have already been covered by the draw_slice
method.
A filter can ensure that a frame is writable at some point of the code
by using the ff_inlink_make_frame_writable() function. It will duplicate
the frame if needed.
The value is a binary OR of the following constants:
A filter can ensure that the frame passed to the filter_frame() callback
is writable by setting the needs_writable flag on the corresponding
input pad. It does not apply to the activate() callback.
* AV_PERM_READ: the owner can read the buffer data; this is essentially
always true and is there for self-documentation.
* AV_PERM_WRITE: the owner can modify the buffer data.
* AV_PERM_PRESERVE: the owner can rely on the fact that the buffer data
will not be modified by previous filters.
* AV_PERM_REUSE: the owner can output the buffer several times, without
modifying the data in between.
* AV_PERM_REUSE2: the owner can output the buffer several times and
modify the data in between (useless without the WRITE permissions).
* AV_PERM_ALIGN: the owner can access the data using fast operations
that require data alignment.
The READ, WRITE and PRESERVE permissions are about sharing the same
buffer between several filters to avoid expensive copies without them
doing conflicting changes on the data.
The REUSE and REUSE2 permissions are about special memory for direct
rendering. For example a buffer directly allocated in video memory must
not modified once it is displayed on screen, or it will cause tearing;
it will therefore not have the REUSE2 permission.
The ALIGN permission is about extracting part of the buffer, for
copy-less padding or cropping for example.
References received on input pads are guaranteed to have all the
permissions stated in the min_perms field and none of the permissions
stated in the rej_perms.
References obtained by ff_get_video_buffer and ff_get_audio_buffer are
guaranteed to have at least all the permissions requested as argument.
References created by avfilter_ref_buffer have the same permissions as
the original reference minus the ones explicitly masked; the mask is
usually ~0 to keep the same permissions.
Filters should remove permissions on reference they give to output
whenever necessary. It can be automatically done by setting the
rej_perms field on the output pad.
Here are a few guidelines corresponding to common situations:
* Filters that modify and forward their frame (like drawtext) need the
WRITE permission.
* Filters that read their input to produce a new frame on output (like
scale) need the READ permission on input and must request a buffer
with the WRITE permission.
* Filters that intend to keep a reference after the filtering process
is finished (after filter_frame returns) must have the PRESERVE
permission on it and remove the WRITE permission if they create a new
reference to give it away.
* Filters that intend to modify a reference they have kept after the end
of the filtering process need the REUSE2 permission and must remove
the PRESERVE permission if they create a new reference to give it
away.
Frame scheduling
@@ -127,100 +189,11 @@ Frame scheduling
Simple filters that output one frame for each input frame should not have
to worry about it.
There are two design for filters: one using the filter_frame() and
request_frame() callbacks and the other using the activate() callback.
The design using filter_frame() and request_frame() is legacy, but it is
suitable for filters that have a single input and process one frame at a
time. New filters with several inputs, that treat several frames at a time
or that require a special treatment at EOF should probably use the design
using activate().
activate
--------
This method is called when something must be done in a filter; the
definition of that "something" depends on the semantic of the filter.
The callback must examine the status of the filter's links and proceed
accordingly.
The status of output links is stored in the frame_wanted_out, status_in
and status_out fields and tested by the ff_outlink_frame_wanted()
function. If this function returns true, then the processing requires a
frame on this link and the filter is expected to make efforts in that
direction.
The status of input links is stored by the status_in, fifo and
status_out fields; they must not be accessed directly. The fifo field
contains the frames that are queued in the input for processing by the
filter. The status_in and status_out fields contains the queued status
(EOF or error) of the link; status_in is a status change that must be
taken into account after all frames in fifo have been processed;
status_out is the status that have been taken into account, it is final
when it is not 0.
The typical task of an activate callback is to first check the backward
status of output links, and if relevant forward it to the corresponding
input. Then, if relevant, for each input link: test the availability of
frames in fifo and process them; if no frame is available, test and
acknowledge a change of status using ff_inlink_acknowledge_status(); and
forward the result (frame or status change) to the corresponding input.
If nothing is possible, test the status of outputs and forward it to the
corresponding input(s). If still not possible, return FFERROR_NOT_READY.
If the filters stores internally one or a few frame for some input, it
can consider them to be part of the FIFO and delay acknowledging a
status change accordingly.
Example code:
ret = ff_outlink_get_status(outlink);
if (ret) {
ff_inlink_set_status(inlink, ret);
return 0;
}
if (priv->next_frame) {
/* use it */
return 0;
}
ret = ff_inlink_consume_frame(inlink, &frame);
if (ret < 0)
return ret;
if (ret) {
/* use it */
return 0;
}
ret = ff_inlink_acknowledge_status(inlink, &status, &pts);
if (ret) {
/* flush */
ff_outlink_set_status(outlink, status, pts);
return 0;
}
if (ff_outlink_frame_wanted(outlink)) {
ff_inlink_request_frame(inlink);
return 0;
}
return FFERROR_NOT_READY;
The exact code depends on how similar the /* use it */ blocks are and
how related they are to the /* flush */ block, and needs to apply these
operations to the correct inlink or outlink if there are several.
Macros are available to factor that when no extra processing is needed:
FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
FF_FILTER_FORWARD_STATUS_ALL(outlink, filter);
FF_FILTER_FORWARD_STATUS(inlink, outlink);
FF_FILTER_FORWARD_STATUS_ALL(inlink, filter);
FF_FILTER_FORWARD_WANTED(outlink, inlink);
filter_frame
------------
For filters that do not use the activate() callback, this method is
called when a frame is pushed to the filter's input. It can be called at
any time except in a reentrant way.
This method is called when a frame is pushed to the filter's input. It
can be called at any time except in a reentrant way.
If the input frame is enough to produce output, then the filter should
push the output frames on the output link immediately.
@@ -249,10 +222,9 @@ Frame scheduling
request_frame
-------------
For filters that do not use the activate() callback, this method is
called when a frame is wanted on an output.
This method is called when a frame is wanted on an output.
For a source, it should directly call filter_frame on the corresponding
For an input, it should directly call filter_frame on the corresponding
output.
For a filter, if there are queued frames already ready, one of these
@@ -282,7 +254,16 @@ Frame scheduling
}
return 0;
Note that, except for filters that can have queued frames and sources,
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.
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.
Legacy API
==========
Until libavfilter 3.23, the filter_frame method was split:
- for video filters, it was made of start_frame, draw_slice (that could be
called several times on distinct parts of the frame) and end_frame;
- for audio filters, it was called filter_samples.

File diff suppressed because it is too large Load Diff

View File

@@ -30,47 +30,37 @@ latency. Must be an integer not lesser than 32. It is 5000000 by default.
@item packetsize @var{integer} (@emph{output})
Set packet size.
@item fflags @var{flags}
Set format flags. Some are implemented for a limited number of formats.
@item fflags @var{flags} (@emph{input/output})
Set format flags.
Possible values for input files:
Possible values:
@table @samp
@item discardcorrupt
Discard corrupted packets.
@item ignidx
Ignore index.
@item fastseek
Enable fast, but inaccurate seeks for some formats.
@item genpts
Generate missing PTS if DTS is present.
@item igndts
Ignore DTS if PTS is set. Inert when nofillin is set.
@item ignidx
Ignore index.
@item keepside (@emph{deprecated},@emph{inert})
@item nobuffer
Reduce the latency introduced by buffering during initial input streams analysis.
Generate PTS.
@item nofillin
Do not fill in missing values in packet fields that can be exactly calculated.
Do not fill in missing values that can be exactly calculated.
@item noparse
Disable AVParsers, this needs @code{+nofillin} too.
@item igndts
Ignore DTS.
@item discardcorrupt
Discard corrupted frames.
@item sortdts
Try to interleave output packets by DTS. At present, available only for AVIs with an index.
@end table
Possible values for output files:
@table @samp
@item autobsf
Automatically apply bitstream filters as required by the output format. Enabled by default.
Try to interleave output packets by DTS.
@item keepside
Do not merge side data.
@item latm
Enable RTP MP4A-LATM payload.
@item nobuffer
Reduce the latency introduced by optional buffering
@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 flush_packets
Write out packets immediately.
@item latm (@emph{deprecated},@emph{inert})
@item shortest
Stop muxing at the end of the shortest stream.
It may be needed to increase max_interleave_delta to avoid flushing the longer
streams before EOF.
@end table
@item seek2any @var{integer} (@emph{input})
@@ -157,7 +147,7 @@ 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})
@@ -188,10 +178,9 @@ Default is 0.
Correct single timestamp overflows if set to 1. Default is 1.
@item flush_packets @var{integer} (@emph{output})
Flush the underlying I/O stream after each packet. Default is -1 (auto), which
means that the underlying protocol will decide, 1 enables it, and has the
effect of reducing the latency, 0 disables it and may increase IO throughput in
some cases.
Flush the underlying I/O stream after each packet. Default 1 enables it, and
has the effect of reducing the latency; 0 disables it and may slightly
increase performance in some cases.
@item output_ts_offset @var{offset} (@emph{output})
Set the output time offset.
@@ -206,7 +195,7 @@ 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.
"," 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
@@ -216,14 +205,6 @@ For example to separate the fields with newlines and indention:
ffprobe -dump_separator "
" -i ~/videos/matrixbench_mpeg2.mpg
@end example
@item max_streams @var{integer} (@emph{input})
Specifies the maximum number of streams. This can be used to reject files that
would require too many resources due to a large number of streams.
@item skip_estimate_duration_from_pts @var{bool} (@emph{input})
Skip estimation of input duration when calculated using PTS.
At present, applicable for MPEG-PS and MPEG-TS.
@end table
@c man end FORMAT OPTIONS

View File

@@ -17,42 +17,6 @@ for more formats. None of them are used by default, their use has to be
explicitly requested by passing the appropriate flags to
@command{./configure}.
@section libxavs2
FFmpeg can make use of the xavs2 library for AVS2-P2/IEEE1857.4 video encoding.
Go to @url{https://github.com/pkuvcl/xavs2} and follow the instructions for
installing the library. Then pass @code{--enable-libxavs2} to configure to
enable it.
@float NOTE
libxavs2 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 libdavs2
FFmpeg can make use of the davs2 library for AVS2-P2/IEEE1857.4 video decoding.
Go to @url{https://github.com/pkuvcl/davs2} and follow the instructions for
installing the library. Then pass @code{--enable-libdavs2} to configure to
enable it.
@float NOTE
libdavs2 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 Alliance for Open Media libaom
FFmpeg can make use of the libaom library for AV1 decoding.
Go to @url{http://aomedia.org/} and follow the instructions for
installing the library. Then pass @code{--enable-libaom} to configure to
enable it.
@section OpenJPEG
FFmpeg can use the OpenJPEG libraries for encoding/decoding J2K videos. Go to
@@ -74,10 +38,9 @@ upgrade FFmpeg's license to LGPL version 3 (or if you have enabled
GPL components, GPL version 3) by passing @code{--enable-version3} to configure in
order to use it.
The license of the Fraunhofer AAC library is incompatible with the GPL.
Therefore, for GPL builds, you have to pass @code{--enable-nonfree} to
configure in order to use it. To the best of our knowledge, it is
compatible with the LGPL.
The Fraunhofer AAC library is licensed under a license incompatible to the GPL
and is not known to be compatible to the LGPL. Therefore, you have to pass
@code{--enable-nonfree} to configure to use it.
@end float
@subsection OpenCORE AMR
@@ -100,7 +63,7 @@ Then pass @code{--enable-libvo-amrwbenc} to configure to enable it.
@subsection Fraunhofer AAC library
FFmpeg can make use of the Fraunhofer AAC library for AAC decoding & encoding.
FFmpeg can make use of the Fraunhofer AAC library for AAC encoding.
Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the
instructions for installing the library.
@@ -122,24 +85,6 @@ Go to @url{http://www.twolame.org/} and follow the
instructions for installing the library.
Then pass @code{--enable-libtwolame} to configure to enable it.
@section libcodec2 / codec2 general
FFmpeg can make use of libcodec2 for codec2 encoding and decoding.
There is currently no native decoder, so libcodec2 must be used for decoding.
Go to @url{http://freedv.org/}, download "Codec 2 source archive".
Build and install using CMake. Debian users can install the libcodec2-dev package instead.
Once libcodec2 is installed you can pass @code{--enable-libcodec2} to configure to enable it.
The easiest way to use codec2 is with .c2 files, since they contain the mode information required for decoding.
To encode such a file, use a .c2 file extension and give the libcodec2 encoder the -mode option:
@code{ffmpeg -i input.wav -mode 700C output.c2}.
Playback is as simple as @code{ffplay output.c2}.
For a list of supported modes, run @code{ffmpeg -h encoder=libcodec2}.
Raw codec2 files are also supported.
To make sense of them the mode in use needs to be specified as a format option:
@code{ffmpeg -f codec2raw -mode 1300 -i input.raw output.wav}.
@section libvpx
FFmpeg can make use of the libvpx library for VP8/VP9 encoding.
@@ -156,29 +101,14 @@ 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 libxavs
FFmpeg can make use of the libxavs library for Xavs encoding.
Go to @url{http://xavs.sf.net/} and follow the instructions for
installing the library. Then pass @code{--enable-libxavs} to configure to
enable it.
@section OpenH264
FFmpeg can make use of the OpenH264 library for H.264 encoding and decoding.
FFmpeg can make use of the OpenH264 library for H.264 encoding.
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.
@@ -249,19 +179,6 @@ For Windows, supported AviSynth variants are
For Linux and OS X, the supported AviSynth variant is
@url{https://github.com/avxsynth/avxsynth, AvxSynth}.
@float NOTE
There is currently a regression in AviSynth+'s @code{capi.h} header as of
October 2016, which interferes with the ability for builds of FFmpeg to use
MSVC-built binaries of AviSynth. Until this is resolved, you can make sure
a known good version is installed by checking out a version from before
the regression occurred:
@code{git clone -b MT git://github.com/AviSynth/AviSynthPlus.git @*
cd AviSynthPlus @*
git checkout -b oldheader b4f292b4dbfad149697fb65c6a037bb3810813f9 @*
make install PREFIX=/install/prefix}
@end float
@float NOTE
AviSynth and AvxSynth are loaded dynamically. Distributors can build FFmpeg
with @code{--enable-avisynth}, and the binaries will work regardless of the
@@ -280,18 +197,6 @@ The dispatcher is open source and can be downloaded from
with the @code{--enable-libmfx} option and @code{pkg-config} needs to be able to
locate the dispatcher's @code{.pc} files.
@section AMD VCE
FFmpeg can use the AMD Advanced Media Framework library for accelerated H.264
and HEVC encoding on VCE enabled hardware under Windows.
To enable support you must obtain the AMF framework header files from
@url{https://github.com/GPUOpen-LibrariesAndSDKs/AMF.git}.
Create an @code{AMF/} directory in the system include path.
Copy the contents of @code{AMF/amf/public/include/} into that directory.
Then configure FFmpeg with @code{--enable-amf}.
@chapter Supported File Formats, Codecs or Features
@@ -357,10 +262,6 @@ library:
@item BRSTM @tab @tab X
@tab Audio format used on the Nintendo Wii.
@item BWF @tab X @tab X
@item codec2 (raw) @tab X @tab X
@tab Must be given -mode format option to decode correctly.
@item codec2 (.c2 files) @tab X @tab X
@tab Contains header with version and mode info, simplifying playback.
@item CRI ADX @tab X @tab X
@tab Audio-only format used in console video games.
@item Discworld II BMV @tab @tab X
@@ -412,7 +313,6 @@ library:
@item FunCom ISS @tab @tab X
@tab Audio format used in various games from FunCom like The Longest Journey.
@item G.723.1 @tab X @tab X
@item G.726 @tab @tab X @tab Both left- and right-justified.
@item G.729 BIT @tab X @tab X
@item G.729 raw @tab @tab X
@item GENH @tab @tab X
@@ -435,8 +335,6 @@ library:
@item iLBC @tab X @tab X
@item Interplay MVE @tab @tab X
@tab Format used in various Interplay computer games.
@item Iterated Systems ClearVideo @tab @tab X
@tab I-frames only
@item IV8 @tab @tab X
@tab A format generated by IndigoVision 8000 video server.
@item IVF (On2) @tab X @tab X
@@ -482,7 +380,6 @@ library:
@tab Audio format used on the PS3.
@item Mirillis FIC video @tab @tab X
@tab No cursor rendering.
@item MIDI Sample Dump Standard @tab @tab X
@item MIME multipart JPEG @tab X @tab
@item MSN TCP webcam @tab @tab X
@tab Used by MSN Messenger webcam streams.
@@ -496,7 +393,6 @@ library:
@item NC camera feed @tab @tab X
@tab NC (AVIP NC4600) camera streams
@item NIST SPeech HEader REsources @tab @tab X
@item Computerized Speech Lab NSP @tab @tab X
@item NTT TwinVQ (VQF) @tab @tab X
@tab Nippon Telegraph and Telephone Corporation TwinVQ.
@item Nullsoft Streaming Video @tab @tab X
@@ -511,10 +407,6 @@ library:
@item QCP @tab @tab X
@item raw ADTS (AAC) @tab X @tab X
@item raw AC-3 @tab X @tab X
@item raw AMR-NB @tab @tab X
@item raw AMR-WB @tab @tab X
@item raw aptX @tab X @tab X
@item raw aptX HD @tab X @tab X
@item raw Chinese AVS video @tab X @tab X
@item raw CRI ADX @tab X @tab X
@item raw Dirac @tab X @tab X
@@ -538,14 +430,12 @@ library:
@item raw NULL @tab X @tab
@item raw video @tab X @tab X
@item raw id RoQ @tab X @tab
@item raw SBC @tab X @tab X
@item raw Shorten @tab @tab X
@item raw TAK @tab @tab X
@item raw TrueHD @tab X @tab X
@item raw VC-1 @tab X @tab X
@item raw PCM A-law @tab X @tab X
@item raw PCM mu-law @tab X @tab X
@item raw PCM Archimedes VIDC @tab X @tab X
@item raw PCM signed 8 bit @tab X @tab X
@item raw PCM signed 16 bit big-endian @tab X @tab X
@item raw PCM signed 16 bit little-endian @tab X @tab X
@@ -553,8 +443,6 @@ library:
@item raw PCM signed 24 bit little-endian @tab X @tab X
@item raw PCM signed 32 bit big-endian @tab X @tab X
@item raw PCM signed 32 bit little-endian @tab X @tab X
@item raw PCM signed 64 bit big-endian @tab X @tab X
@item raw PCM signed 64 bit little-endian @tab X @tab X
@item raw PCM unsigned 8 bit @tab X @tab X
@item raw PCM unsigned 16 bit big-endian @tab X @tab X
@item raw PCM unsigned 16 bit little-endian @tab X @tab X
@@ -562,8 +450,6 @@ library:
@item raw PCM unsigned 24 bit little-endian @tab X @tab X
@item raw PCM unsigned 32 bit big-endian @tab X @tab X
@item raw PCM unsigned 32 bit little-endian @tab X @tab X
@item raw PCM 16.8 floating point little-endian @tab @tab X
@item raw PCM 24.0 floating point little-endian @tab @tab X
@item raw PCM floating-point 32 bit big-endian @tab X @tab X
@item raw PCM floating-point 32 bit little-endian @tab X @tab X
@item raw PCM floating-point 64 bit big-endian @tab X @tab X
@@ -585,12 +471,10 @@ library:
@tab Output is performed by publishing stream to RTMP server
@item RTP @tab X @tab X
@item RTSP @tab X @tab X
@item Sample Dump eXchange @tab @tab X
@item SAP @tab X @tab X
@item SBG @tab @tab X
@item SDP @tab @tab X
@item SER @tab @tab X
@item Sega FILM/CPK @tab X @tab X
@item Sega FILM/CPK @tab @tab X
@tab Used in many Sega Saturn console games.
@item Silicon Graphics Movie @tab @tab X
@item Sierra SOL @tab @tab X
@@ -601,7 +485,6 @@ library:
@tab Multimedia format used by many games.
@item SMJPEG @tab X @tab X
@tab Used in certain Loki game ports.
@item SMPTE 337M encapsulation @tab @tab X
@item Smush @tab @tab X
@tab Multimedia format used in some LucasArts games.
@item Sony OpenMG (OMA) @tab X @tab X
@@ -610,7 +493,7 @@ 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 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
@@ -619,7 +502,7 @@ library:
@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 True Audio @tab @tab X
@item VAG @tab @tab X
@tab Audio format used in many Sony PS2 games.
@item VC-1 test bitstream @tab X @tab X
@@ -637,7 +520,6 @@ 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.
@@ -673,8 +555,6 @@ following image formats are supported:
@tab Digital Picture Exchange
@item EXR @tab @tab X
@tab OpenEXR
@item FITS @tab X @tab X
@tab Flexible Image Transport System
@item JPEG @tab X @tab X
@tab Progressive JPEG is not supported.
@item JPEG 2000 @tab X @tab X
@@ -696,8 +576,6 @@ following image formats are supported:
@item PNG @tab X @tab X
@item PPM @tab X @tab X
@tab Portable PixelMap image
@item PSD @tab @tab X
@tab Photoshop
@item PTX @tab @tab X
@tab V.Flash PTX format
@item SGI @tab X @tab X
@@ -714,8 +592,6 @@ following image formats are supported:
@tab X BitMap image format
@item XFace @tab X @tab X
@tab X-Face image format
@item XPM @tab @tab X
@tab X PixMap image format
@item XWD @tab X @tab X
@tab X Window Dump image format
@end multitable
@@ -741,7 +617,6 @@ following image formats are supported:
@item ANSI/ASCII art @tab @tab X
@item Apple Intermediate Codec @tab @tab X
@item Apple MJPEG-B @tab @tab X
@item Apple Pixlet @tab @tab X
@item Apple ProRes @tab X @tab X
@item Apple QuickDraw @tab @tab X
@tab fourcc: qdrw
@@ -758,8 +633,6 @@ following image formats are supported:
@item Autodesk Animator Flic video @tab @tab X
@item Autodesk RLE @tab @tab X
@tab fourcc: AASC
@item AV1 @tab @tab E
@tab Supported through external library libaom
@item Avid 1:1 10-bit RGB Packer @tab X @tab X
@tab fourcc: AVrp
@item AVS (Audio Video Standard) video @tab @tab X
@@ -770,7 +643,6 @@ 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
@@ -797,7 +669,7 @@ following image formats are supported:
@item DFA @tab @tab X
@tab Codec used in Chronomaster game.
@item Dirac @tab E @tab X
@tab supported though the native vc2 (Dirac Pro) encoder
@tab supported through external library libschroedinger
@item Deluxe Paint Animation @tab @tab X
@item DNxHD @tab X @tab X
@tab aka SMPTE VC3
@@ -805,8 +677,6 @@ 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
@@ -826,14 +696,12 @@ following image formats are supported:
@item Flash Screen Video v2 @tab X @tab X
@item Flash Video (FLV) @tab X @tab X
@tab Sorenson H.263 used in Flash
@item FM Screen Capture Codec @tab @tab X
@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 Gremlin Digital Video @tab @tab X
@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
@@ -854,7 +722,6 @@ following image formats are supported:
@tab IFF interleaved bitmap
@item IFF ByteRun1 @tab @tab X
@tab IFF run length encoded bitmap
@item Infinity IMM4 @tab @tab X
@item Intel H.263 @tab @tab X
@item Intel Indeo 2 @tab @tab X
@item Intel Indeo 3 @tab @tab X
@@ -876,8 +743,6 @@ 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 X @tab X
@item Mandsoft Screen Capture Codec @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
@@ -902,7 +767,6 @@ following image formats are supported:
@item MPEG-4 part 2 Microsoft variant version 1 @tab @tab X
@item MPEG-4 part 2 Microsoft variant version 2 @tab X @tab X
@item MPEG-4 part 2 Microsoft variant version 3 @tab X @tab X
@item Newtek SpeedHQ @tab @tab X
@item Nintendo Gamecube THP video @tab @tab X
@item NuppelVideo/RTjpeg @tab @tab X
@tab Video encoding used in NuppelVideo files.
@@ -943,9 +807,7 @@ 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 ScreenPressor @tab @tab X
@item Screenpresso @tab @tab X
@item Screen Recorder Gold Codec @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
@@ -1013,7 +875,7 @@ following image formats are supported:
@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
@tab encoding supported through internal encoder and external libraries libfaac and libfdk-aac
@item AAC+ @tab E @tab IX
@tab encoding supported through external library libfdk-aac
@item AC-3 @tab IX @tab IX
@@ -1074,20 +936,13 @@ following image formats are supported:
@item Amazing Studio PAF Audio @tab @tab X
@item Apple lossless audio @tab X @tab X
@tab QuickTime fourcc 'alac'
@item aptX @tab X @tab X
@tab Used in Bluetooth A2DP
@item aptX HD @tab X @tab X
@tab Used in Bluetooth A2DP
@item ATRAC1 @tab @tab X
@item ATRAC3 @tab @tab X
@item ATRAC3+ @tab @tab X
@item ATRAC9 @tab @tab X
@item Bink Audio @tab @tab X
@tab Used in Bink and Smacker files in many games.
@item CELT @tab @tab E
@tab decoding supported through external library libcelt
@item codec2 @tab E @tab E
@tab en/decoding supported through external library libcodec2
@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
@@ -1095,8 +950,7 @@ following image formats are supported:
@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 Dolby E @tab @tab X
@tab supported extensions: XCh, XLL (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
@@ -1113,7 +967,6 @@ 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
@@ -1131,7 +984,7 @@ following image formats are supported:
@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
@@ -1148,7 +1001,6 @@ following image formats are supported:
@tab encoding supported through external library libopus
@item PCM A-law @tab X @tab X
@item PCM mu-law @tab X @tab X
@item PCM Archimedes VIDC @tab X @tab X
@item PCM signed 8-bit planar @tab X @tab X
@item PCM signed 16-bit big-endian planar @tab X @tab X
@item PCM signed 16-bit little-endian planar @tab X @tab X
@@ -1176,7 +1028,6 @@ following image formats are supported:
@item PCM unsigned 32-bit little-endian @tab X @tab X
@item PCM Zork @tab @tab X
@item QCELP / PureVoice @tab @tab X
@item QDesign Music Codec 1 @tab @tab X
@item QDesign Music Codec 2 @tab @tab X
@tab There are still some distortions.
@item RealAudio 1.0 (14.4K) @tab X @tab X
@@ -1187,8 +1038,6 @@ following image formats are supported:
@tab Real low bitrate AC-3 codec
@item RealAudio Lossless @tab @tab X
@item RealAudio SIPR / ACELP.NET @tab @tab X
@item SBC (low-complexity subband codec) @tab X @tab X
@tab Used in Bluetooth A2DP
@item Shorten @tab @tab X
@item Sierra VMD audio @tab @tab X
@tab Used in Sierra VMD files.
@@ -1202,7 +1051,7 @@ 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
@@ -1275,7 +1124,6 @@ performance on systems without hardware floating point support).
@item MMSH @tab X
@item MMST @tab X
@item pipe @tab X
@item Pro-MPEG FEC @tab X
@item RTMP @tab X
@item RTMPE @tab X
@item RTMPS @tab X

View File

@@ -408,7 +408,7 @@ with @option{--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}
@@ -418,7 +418,7 @@ Next let the code pass through a full run of our test suite.
@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).

View File

@@ -63,51 +63,12 @@ Set the number of channels. Default is 2.
@end table
@section android_camera
Android camera input device.
This input devices uses the Android Camera2 NDK API which is
available on devices with API level 24+. The availability of
android_camera is autodetected during configuration.
This device allows capturing from all cameras on an Android device,
which are integrated into the Camera2 NDK API.
The available cameras are enumerated internally and can be selected
with the @var{camera_index} parameter. The input file string is
discarded.
Generally the back facing camera has index 0 while the front facing
camera has index 1.
@subsection Options
@table @option
@item video_size
Set the video size given as a string such as 640x480 or hd720.
Falls back to the first available configuration reported by
Android if requested video size is not available or by default.
@item framerate
Set the video framerate.
Falls back to the first available configuration reported by
Android if requested framerate is not available or by default (-1).
@item camera_index
Set the index of the camera to use. Default is 0.
@item input_queue_size
Set the maximum number of frames to buffer. Default is 5.
@end table
@section avfoundation
AVFoundation input device.
AVFoundation is the currently recommended framework by Apple for streamgrabbing on OSX >= 10.7 as well as on iOS.
The older QTKit framework has been marked deprecated since OSX version 10.7.
The input filename has to be given in the following syntax:
@example
@@ -254,9 +215,8 @@ 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 of the
input can be set with @option{raw_format}.
Framerate and video size must be determined for your device with
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.
@@ -267,124 +227,30 @@ audio track.
@item list_devices
If set to @option{true}, print a list of devices and exit.
Defaults to @option{false}. Alternatively you can use the @code{-sources}
option of ffmpeg to list the available input devices.
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 format_code <FourCC>
This sets the input video format to the format given by the FourCC. To see
the supported values of your device(s) use @option{list_formats}.
Note that there is a FourCC @option{'pal '} that can also be used
as @option{pal} (3 letters).
Default behavior is autodetection of the input video format, if the hardware
supports it.
@item bm_v210
This is a deprecated option, you can use @option{raw_format} instead.
If set to @samp{1}, video is captured in 10 bit v210 instead
of uyvy422. Not all Blackmagic devices support this option.
@item raw_format
Set the pixel format of the captured video.
Available values are:
@table @samp
@item uyvy422
@item yuv422p10
@item argb
@item bgra
@item rgb10
@end table
@item teletext_lines
If set to nonzero, an additional teletext stream will be captured from the
vertical ancillary data. Both SD PAL (576i) and HD (1080i or 1080p)
sources are supported. In case of HD sources, OP47 packets are decoded.
This option is a bitmask of the SD PAL VBI lines captured, 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.
For SD sources, ffmpeg needs to be compiled with @code{--enable-libzvbi}. For
HD sources, on older (pre-4K) DeckLink card models you have to capture in 10
bit mode.
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 timecode_format
Timecode type to include in the frame and video stream metadata. Must be
@samp{none}, @samp{rp188vitc}, @samp{rp188vitc2}, @samp{rp188ltc},
@samp{rp188any}, @samp{vitc}, @samp{vitc2}, or @samp{serial}. Defaults to
@samp{none} (not included).
@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}, @samp{wallclock} or @samp{abs_wallclock}.
Defaults to @samp{video}.
@item audio_pts
Sets the audio packet timestamp source. Must be @samp{video}, @samp{audio},
@samp{reference}, @samp{wallclock} or @samp{abs_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}.
@item queue_size
Sets maximum input buffer size in bytes. If the buffering reaches this value,
incoming frames will be dropped.
Defaults to @samp{1073741824}.
@item audio_depth
Sets the audio sample bit depth. Must be @samp{16} or @samp{32}.
Defaults to @samp{16}.
@item decklink_copyts
If set to @option{true}, timestamps are forwarded as they are without removing
the initial offset.
Defaults to @option{false}.
@item timestamp_align
Capture start time alignment in seconds. If set to nonzero, input frames are
dropped till the system timestamp aligns with configured value.
Alignment difference of up to one frame duration is tolerated.
This is useful for maintaining input synchronization across N different
hardware devices deployed for 'N-way' redundancy. The system time of different
hardware devices should be synchronized with protocols such as NTP or PTP,
before using this option.
Note that this method is not foolproof. In some border cases input
synchronization may not happen due to thread scheduling jitters in the OS.
Either sync could go wrong by 1 frame or in a rarer case
@option{timestamp_align} seconds.
Defaults to @samp{0}.
@end table
@subsection Examples
@@ -404,21 +270,21 @@ ffmpeg -f decklink -list_formats 1 -i 'Intensity Pro'
@end example
@item
Capture video clip at 1080i50:
Capture video clip at 1080i50 (format 11):
@example
ffmpeg -format_code Hi50 -f decklink -i 'Intensity Pro' -c:a copy -c:v copy output.avi
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 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder' -c:a copy -c:v copy output.avi
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 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder' -c:a copy -c:v copy output.avi
ffmpeg -channels 16 -f decklink -i 'UltraStudio Mini Recorder@@11' -acodec copy -vcodec copy output.avi
@end example
@end itemize
@@ -623,6 +489,31 @@ $ ffmpeg -f dshow -show_video_device_dialog true -crossbar_video_input_pin_numbe
@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.
@@ -765,7 +656,7 @@ 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.
@@ -850,68 +741,6 @@ Set the number of channels. Default is 2.
@end table
@section kmsgrab
KMS video input device.
Captures the KMS scanout framebuffer associated with a specified CRTC or plane as a
DRM object that can be passed to other hardware functions.
Requires either DRM master or CAP_SYS_ADMIN to run.
If you don't understand what all of that means, you probably don't want this. Look at
@option{x11grab} instead.
@subsection Options
@table @option
@item device
DRM device to capture on. Defaults to @option{/dev/dri/card0}.
@item format
Pixel format of the framebuffer. Defaults to @option{bgr0}.
@item format_modifier
Format modifier to signal on output frames. This is necessary to import correctly into
some APIs, but can't be autodetected. See the libdrm documentation for possible values.
@item crtc_id
KMS CRTC ID to define the capture source. The first active plane on the given CRTC
will be used.
@item plane_id
KMS plane ID to define the capture source. Defaults to the first active plane found if
neither @option{crtc_id} nor @option{plane_id} are specified.
@item framerate
Framerate to capture at. This is not synchronised to any page flipping or framebuffer
changes - it just defines the interval at which the framebuffer is sampled. Sampling
faster than the framebuffer update rate will generate independent frames with the same
content. Defaults to @code{30}.
@end table
@subsection Examples
@itemize
@item
Capture from the first active plane, download the result to normal frames and encode.
This will only work if the framebuffer is both linear and mappable - if not, the result
may be scrambled or fail to download.
@example
ffmpeg -f kmsgrab -i - -vf 'hwdownload,format=bgr0' output.mp4
@end example
@item
Capture from CRTC ID 42 at 60fps, map the result to VAAPI, convert to NV12 and encode as H.264.
@example
ffmpeg -crtc_id 42 -framerate 60 -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,scale_vaapi=w=1920:h=1080:format=nv12' -c:v h264_vaapi output.mp4
@end example
@end itemize
@section lavfi
Libavfilter input virtual device.
@@ -1050,71 +879,6 @@ IIDC1394 input device, based on libdc1394 and libraw1394.
Requires the configure option @code{--enable-libdc1394}.
@section libndi_newtek
The libndi_newtek input device provides capture capabilities for using NDI (Network
Device Interface, standard created by NewTek).
Input filename is a NDI source name that could be found by sending -find_sources 1
to command line - it has no specific syntax but human-readable formatted.
To enable this input device, you need the NDI SDK and you
need to configure with the appropriate @code{--extra-cflags}
and @code{--extra-ldflags}.
@subsection Options
@table @option
@item find_sources
If set to @option{true}, print a list of found/available NDI sources and exit.
Defaults to @option{false}.
@item wait_sources
Override time to wait until the number of online sources have changed.
Defaults to @option{0.5}.
@item allow_video_fields
When this flag is @option{false}, all video that you receive will be progressive.
Defaults to @option{true}.
@item extra_ips
If is set to list of comma separated ip addresses, scan for sources not only
using mDNS but also use unicast ip addresses specified by this list.
@end table
@subsection Examples
@itemize
@item
List input devices:
@example
ffmpeg -f libndi_newtek -find_sources 1 -i dummy
@end example
@item
List local and remote input devices:
@example
ffmpeg -f libndi_newtek -extra_ips "192.168.10.10" -find_sources 1 -i dummy
@end example
@item
Restream to NDI:
@example
ffmpeg -f libndi_newtek -i "DEV-5.INTERNAL.M1STEREO.TV (NDI_SOURCE_NAME_1)" -f libndi_newtek -y NDI_SOURCE_NAME_2
@end example
@item
Restream remote NDI to local NDI:
@example
ffmpeg -f libndi_newtek -extra_ips "192.168.10.10" -i "DEV-5.REMOTE.M1STEREO.TV (NDI_SOURCE_NAME_1)" -f libndi_newtek -y NDI_SOURCE_NAME_2
@end example
@end itemize
@section openal
The OpenAL input device provides audio capture on all systems with a
@@ -1233,6 +997,7 @@ Set the number of channels. Default is 2.
@end table
@section pulse
PulseAudio input device.
@@ -1285,6 +1050,49 @@ Record a stream from default device:
ffmpeg -f pulse -i default /tmp/pulse.wav
@end example
@section qtkit
QTKit input device.
The filename passed as input is parsed to contain either a device name or index.
The device index can also be given by using -video_device_index.
A given device index will override any given device name.
If the desired device consists of numbers only, use -video_device_index to identify it.
The default device will be chosen if an empty string or the device name "default" is given.
The available devices can be enumerated by using -list_devices.
@example
ffmpeg -f qtkit -i "0" out.mpg
@end example
@example
ffmpeg -f qtkit -video_device_index 0 -i "" out.mpg
@end example
@example
ffmpeg -f qtkit -i "default" out.mpg
@end example
@example
ffmpeg -f qtkit -list_devices true -i ""
@end example
@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.
@@ -1471,6 +1279,9 @@ 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:
@@ -1508,7 +1319,7 @@ 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
@@ -1558,6 +1369,11 @@ 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

View File

@@ -193,6 +193,9 @@ ffplay
ffprobe
issues in or related to ffprobe.c
ffserver
issues in or related to ffserver.c
postproc
issues in libpostproc/*

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