github-mirror/Telegram-iOS
1
0
Fork 0
mirror of https://github.com/TelegramMessenger/Telegram-iOS.git synced 2026-07-06 03:33:41 +02:00
Code Issues Projects Releases Packages Wiki Activity Actions

[Temp] Roll back ffmpeg

Browse source
This commit is contained in:
Ali 2023-09-30 10:58:14 +04:00
parent cd2a4e0788
commit 552aede716
9753 changed files with 601514 additions and 887674 deletions
Download patch file Download diff file Expand all files Collapse all files

249
submodules/ffmpeg/BUILD vendored
View file

@ -1,148 +1,107 @@
avcodec_headers = [
"ac3_parser.h",
"adts_parser.h",
"avcodec.h",
"avdct.h",
"avfft.h",
"bsf.h",
"codec_desc.h",
"codec_id.h",
"codec_par.h",
"codec.h",
"d3d11va.h",
"defs.h",
"dirac.h",
"dv_profile.h",
"dxva2.h",
"jni.h",
"mediacodec.h",
"packet.h",
"qsv.h",
"vdpau.h",
"version_major.h",
"version.h",
"videotoolbox.h",
"vorbis_parser.h",
"xvmc.h",
]
avformat_headers = [
"avformat.h",
"avio.h",
"version_major.h",
"version.h",
]
avutil_headers = [
"adler32.h",
"aes_ctr.h",
"aes.h",
"ambient_viewing_environment.h",
"attributes.h",
"audio_fifo.h",
"avassert.h",
"avconfig.h",
"avstring.h",
"avutil.h",
"base64.h",
"blowfish.h",
"bprint.h",
"bswap.h",
"buffer.h",
"camellia.h",
"cast5.h",
"channel_layout.h",
"common.h",
"cpu.h",
"crc.h",
"csp.h",
"des.h",
"detection_bbox.h",
"dict.h",
"display.h",
"dovi_meta.h",
"downmix_info.h",
"encryption_info.h",
"error.h",
"eval.h",
"ffversion.h",
"fifo.h",
"file.h",
"film_grain_params.h",
"frame.h",
"hash.h",
"hdr_dynamic_metadata.h",
"hdr_dynamic_vivid_metadata.h",
"hmac.h",
"hwcontext_cuda.h",
"hwcontext_d3d11va.h",
"hwcontext_drm.h",
"hwcontext_dxva2.h",
"hwcontext_mediacodec.h",
"hwcontext_opencl.h",
"hwcontext_qsv.h",
"hwcontext_vaapi.h",
"hwcontext_vdpau.h",
"hwcontext_videotoolbox.h",
"hwcontext_vulkan.h",
"hwcontext.h",
"imgutils.h",
"intfloat.h",
"intreadwrite.h",
"lfg.h",
"log.h",
"lzo.h",
"macros.h",
"mastering_display_metadata.h",
"mathematics.h",
"md5.h",
"mem.h",
"motion_vector.h",
"murmur3.h",
"opt.h",
"parseutils.h",
"pixdesc.h",
"pixelutils.h",
"pixfmt.h",
"random_seed.h",
"rational.h",
"rc4.h",
"replaygain.h",
"ripemd.h",
"samplefmt.h",
"sha.h",
"sha512.h",
"spherical.h",
"stereo3d.h",
"tea.h",
"threadmessage.h",
"time.h",
"timecode.h",
"timestamp.h",
"tree.h",
"twofish.h",
"tx.h",
"uuid.h",
"version.h",
"video_enc_params.h",
"xtea.h",
]
swresample_headers = [
"swresample.h",
"version_major.h",
"version.h",
]
ffmpeg_header_paths = [
"libavcodec/{}".format(x) for x in avcodec_headers
] + [
"libavformat/{}".format(x) for x in avformat_headers
] + [
"libavutil/{}".format(x) for x in avutil_headers
] + [
"libswresample/{}".format(x) for x in swresample_headers
"libavutil/hwcontext.h",
"libavutil/time.h",
"libavutil/hwcontext_cuda.h",
"libavutil/intfloat.h",
"libavutil/error.h",
"libavutil/fifo.h",
"libavutil/blowfish.h",
"libavutil/hwcontext_mediacodec.h",
"libavutil/replaygain.h",
"libavutil/version.h",
"libavutil/murmur3.h",
"libavutil/stereo3d.h",
"libavutil/samplefmt.h",
"libavutil/pixdesc.h",
"libavutil/base64.h",
"libavutil/rational.h",
"libavutil/sha.h",
"libavutil/motion_vector.h",
"libavutil/avconfig.h",
"libavutil/lfg.h",
"libavutil/avutil.h",
"libavutil/xtea.h",
"libavutil/crc.h",
"libavutil/hwcontext_vdpau.h",
"libavutil/frame.h",
"libavutil/file.h",
"libavutil/md5.h",
"libavutil/cast5.h",
"libavutil/hwcontext_vaapi.h",
"libavutil/spherical.h",
"libavutil/ffversion.h",
"libavutil/audio_fifo.h",
"libavutil/tree.h",
"libavutil/threadmessage.h",
"libavutil/attributes.h",
"libavutil/adler32.h",
"libavutil/hwcontext_d3d11va.h",
"libavutil/timecode.h",
"libavutil/sha512.h",
"libavutil/hwcontext_dxva2.h",
"libavutil/display.h",
"libavutil/buffer.h",
"libavutil/camellia.h",
"libavutil/pixelutils.h",
"libavutil/hwcontext_drm.h",
"libavutil/common.h",
"libavutil/hmac.h",
"libavutil/eval.h",
"libavutil/dict.h",
"libavutil/random_seed.h",
"libavutil/opt.h",
"libavutil/mastering_display_metadata.h",
"libavutil/log.h",
"libavutil/aes.h",
"libavutil/macros.h",
"libavutil/bswap.h",
"libavutil/rc4.h",
"libavutil/tea.h",
"libavutil/cpu.h",
"libavutil/lzo.h",
"libavutil/des.h",
"libavutil/channel_layout.h",
"libavutil/encryption_info.h",
"libavutil/twofish.h",
"libavutil/imgutils.h",
"libavutil/hwcontext_videotoolbox.h",
"libavutil/mem.h",
"libavutil/parseutils.h",
"libavutil/ripemd.h",
"libavutil/bprint.h",
"libavutil/hwcontext_qsv.h",
"libavutil/pixfmt.h",
"libavutil/aes_ctr.h",
"libavutil/timestamp.h",
"libavutil/downmix_info.h",
"libavutil/avassert.h",
"libavutil/hash.h",
"libavutil/mathematics.h",
"libavutil/intreadwrite.h",
"libavutil/avstring.h",
"libavformat/version.h",
"libavformat/avio.h",
"libavformat/avformat.h",
"libavcodec/adts_parser.h",
"libavcodec/avcodec.h",
"libavcodec/version.h",
"libavcodec/vdpau.h",
"libavcodec/qsv.h",
"libavcodec/vaapi.h",
"libavcodec/videotoolbox.h",
"libavcodec/xvmc.h",
"libavcodec/mediacodec.h",
"libavcodec/d3d11va.h",
"libavcodec/avfft.h",
"libavcodec/jni.h",
"libavcodec/dirac.h",
"libavcodec/avdct.h",
"libavcodec/ac3_parser.h",
"libavcodec/vorbis_parser.h",
"libavcodec/dxva2.h",
"libavcodec/dv_profile.h",
"libswresample/version.h",
"libswresample/swresample.h",
]
ffmpeg_libs = [
@ -206,7 +165,7 @@ genrule(
],
cmd_bash =
"""
#set -x
set -x
ABS_RULEDIR="$$(pwd)/$(RULEDIR)"
echo "ABS_RULEDIR=$$ABS_RULEDIR"

10
submodules/ffmpeg/Sources/FFMpeg/build-ffmpeg-bazel.sh
View file

@ -1,8 +1,6 @@
#!/bin/sh
#set -x
RV=1
set -x
RAW_ARCHS="$2"
ARCHS=""
@ -21,7 +19,7 @@ done
BUILD_DIR=$3
SOURCE_DIR=$4
FF_VERSION="6.0"
FF_VERSION="4.1"
SOURCE="$SOURCE_DIR/ffmpeg-$FF_VERSION"
GAS_PREPROCESSOR_PATH="$SOURCE_DIR/gas-preprocessor.pl"
@ -49,14 +47,12 @@ CONFIGURE_FLAGS="--enable-cross-compile --disable-programs \
--enable-libopus \
--enable-libvpx \
--enable-audiotoolbox \
--enable-videotoolbox \
--enable-bsf=aac_adtstoasc \
--enable-decoder=h264,h264_videotoolbox,libvpx_vp9,hevc,libopus,mp3,aac,flac,alac_at,pcm_s16le,pcm_s24le,gsm_ms_at \
--enable-decoder=h264,libvpx_vp9,hevc,libopus,mp3,aac,flac,alac_at,pcm_s16le,pcm_s24le,gsm_ms_at \
--enable-demuxer=aac,mov,m4v,mp3,ogg,libopus,flac,wav,aiff,matroska \
--enable-parser=aac,h264,mp3,libopus \
--enable-protocol=file \
--enable-muxer=mp4 \
--enable-hwaccel=h264_videotoolbox,hevc_videotoolbox \
"

2
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/.gitattributes vendored Normal file
View file

@ -0,0 +1,2 @@
*.pnm -diff -text
tests/ref/fate/sub-scc eol=crlf

2
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/.travis.yml → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/.travis.yml
View file

@ -19,7 +19,7 @@ cache:
directories:
- ffmpeg-samples
before_install:
- if [ "$TRAVIS_OS_NAME" == "osx" ]; then brew update; fi
- if [ "$TRAVIS_OS_NAME" == "osx" ]; then brew update --all; fi
install:
- if [ "$TRAVIS_OS_NAME" == "osx" ]; then brew install nasm; fi
script:

0
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/CONTRIBUTING.md → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/CONTRIBUTING.md
View file

0
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/COPYING.GPLv2 → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/COPYING.GPLv2
View file

0
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/COPYING.GPLv3 → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/COPYING.GPLv3
View file

0
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/COPYING.LGPLv2.1 → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/COPYING.LGPLv2.1
View file

0
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/COPYING.LGPLv3 → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/COPYING.LGPLv3
View file

0
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/CREDITS → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/CREDITS
View file

317
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/Changelog → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/Changelog
View file

@ -1,323 +1,6 @@
Entries are sorted chronologically from oldest to youngest within each release,
releases are sorted from youngest to oldest.
version 6.0:
- Radiance HDR image support
- ddagrab (Desktop Duplication) video capture filter
- ffmpeg -shortest_buf_duration option
- ffmpeg now requires threading to be built
- ffmpeg now runs every muxer in a separate thread
- Add new mode to cropdetect filter to detect crop-area based on motion vectors and edges
- VAAPI decoding and encoding for 10/12bit 422, 10/12bit 444 HEVC and VP9
- WBMP (Wireless Application Protocol Bitmap) image format
- a3dscope filter
- bonk decoder and demuxer
- Micronas SC-4 audio decoder
- LAF demuxer
- APAC decoder and demuxer
- Media 100i decoders
- DTS to PTS reorder bsf
- ViewQuest VQC decoder
- backgroundkey filter
- nvenc AV1 encoding support
- MediaCodec decoder via NDKMediaCodec
- MediaCodec encoder
- oneVPL support for QSV
- QSV AV1 encoder
- QSV decoding and encoding for 10/12bit 422, 10/12bit 444 HEVC and VP9
- showcwt multimedia filter
- corr video filter
- adrc audio filter
- afdelaysrc audio filter
- WADY DPCM decoder and demuxer
- CBD2 DPCM decoder
- ssim360 video filter
- ffmpeg CLI new options: -stats_enc_pre[_fmt], -stats_enc_post[_fmt],
-stats_mux_pre[_fmt]
- hstack_vaapi, vstack_vaapi and xstack_vaapi filters
- XMD ADPCM decoder and demuxer
- media100 to mjpegb bsf
- ffmpeg CLI new option: -fix_sub_duration_heartbeat
- WavArc decoder and demuxer
- CrystalHD decoders deprecated
- SDNS demuxer
- RKA decoder and demuxer
- filtergraph syntax in ffmpeg CLI now supports passing file contents
as option values, by prefixing option name with '/'
- hstack_qsv, vstack_qsv and xstack_qsv filters
version 5.1:
- add ipfs/ipns gateway support
- dialogue enhance audio filter
- dropped obsolete XvMC hwaccel
- pcm-bluray encoder
- DFPWM audio encoder/decoder and raw muxer/demuxer
- SITI filter
- Vizrt Binary Image encoder/decoder
- avsynctest source filter
- feedback video filter
- pixelize video filter
- colormap video filter
- colorchart video source filter
- multiply video filter
- PGS subtitle frame merge bitstream filter
- blurdetect filter
- tiltshelf audio filter
- QOI image format support
- ffprobe -o option
- virtualbass audio filter
- VDPAU AV1 hwaccel
- PHM image format support
- remap_opencl filter
- added chromakey_cuda filter
- added bilateral_cuda filter
version 5.0:
- ADPCM IMA Westwood encoder
- Westwood AUD muxer
- ADPCM IMA Acorn Replay decoder
- Argonaut Games CVG demuxer
- Argonaut Games CVG muxer
- Concatf protocol
- afwtdn audio filter
- audio and video segment filters
- Apple Graphics (SMC) encoder
- hsvkey and hsvhold video filters
- adecorrelate audio filter
- atilt audio filter
- grayworld video filter
- AV1 Low overhead bitstream format muxer
- swscale slice threading
- MSN Siren decoder
- scharr video filter
- apsyclip audio filter
- morpho video filter
- amr parser
- (a)latency filters
- GEM Raster image decoder
- asdr audio filter
- speex decoder
- limitdiff video filter
- xcorrelate video filter
- varblur video filter
- huesaturation video filter
- colorspectrum source video filter
- RTP packetizer for uncompressed video (RFC 4175)
- bitpacked encoder
- VideoToolbox VP9 hwaccel
- VideoToolbox ProRes hwaccel
- support loongarch.
- aspectralstats audio filter
- adynamicsmooth audio filter
- libplacebo filter
- vflip_vulkan, hflip_vulkan and flip_vulkan filters
- adynamicequalizer audio filter
- yadif_videotoolbox filter
- VideoToolbox ProRes encoder
- anlmf audio filter
- IMF demuxer (experimental)
version 4.4:
- AudioToolbox output device
- MacCaption demuxer
- PGX decoder
- chromanr video filter
- VDPAU accelerated HEVC 10/12bit decoding
- ADPCM IMA Ubisoft APM encoder
- Rayman 2 APM muxer
- AV1 encoding support SVT-AV1
- Cineform HD encoder
- ADPCM Argonaut Games encoder
- Argonaut Games ASF muxer
- AV1 Low overhead bitstream format demuxer
- RPZA video encoder
- ADPCM IMA MOFLEX decoder
- MobiClip FastAudio decoder
- MobiClip video decoder
- MOFLEX demuxer
- MODS demuxer
- PhotoCD decoder
- MCA demuxer
- AV1 decoder (Hardware acceleration used only)
- SVS demuxer
- Argonaut Games BRP demuxer
- DAT demuxer
- aax demuxer
- IPU decoder, parser and demuxer
- Intel QSV-accelerated AV1 decoding
- Argonaut Games Video decoder
- libwavpack encoder removed
- ACE demuxer
- AVS3 demuxer
- AVS3 video decoder via libuavs3d
- Cintel RAW decoder
- VDPAU accelerated VP9 10/12bit decoding
- afreqshift and aphaseshift filters
- High Voltage Software ADPCM encoder
- LEGO Racers ALP (.tun & .pcm) muxer
- AV1 VAAPI decoder
- adenorm filter
- ADPCM IMA AMV encoder
- AMV muxer
- NVDEC AV1 hwaccel
- DXVA2/D3D11VA hardware accelerated AV1 decoding
- speechnorm filter
- SpeedHQ encoder
- asupercut filter
- asubcut filter
- Microsoft Paint (MSP) version 2 decoder
- Microsoft Paint (MSP) demuxer
- AV1 monochrome encoding support via libaom >= 2.0.1
- asuperpass and asuperstop filter
- shufflepixels filter
- tmidequalizer filter
- estdif filter
- epx filter
- Dolby E parser
- shear filter
- kirsch filter
- colortemperature filter
- colorcontrast filter
- PFM encoder
- colorcorrect filter
- binka demuxer
- XBM parser
- xbm_pipe demuxer
- colorize filter
- CRI parser
- aexciter audio filter
- exposure video filter
- monochrome video filter
- setts bitstream filter
- vif video filter
- OpenEXR image encoder
- Simbiosis IMX decoder
- Simbiosis IMX demuxer
- Digital Pictures SGA demuxer and decoders
- TTML subtitle encoder and muxer
- identity video filter
- msad video filter
- gophers protocol
- RIST protocol via librist
version 4.3:
- v360 filter
- Intel QSV-accelerated MJPEG decoding
- Intel QSV-accelerated VP9 decoding
- Support for TrueHD in mp4
- Support AMD AMF encoder on Linux (via Vulkan)
- IMM5 video decoder
- ZeroMQ protocol
- support Sipro ACELP.KELVIN decoding
- streamhash muxer
- sierpinski video source
- scroll video filter
- photosensitivity filter
- anlms filter
- arnndn filter
- bilateral filter
- maskedmin and maskedmax filters
- VDPAU VP9 hwaccel
- median filter
- QSV-accelerated VP9 encoding
- AV1 encoding support via librav1e
- AV1 frame merge bitstream filter
- AV1 Annex B demuxer
- axcorrelate filter
- mvdv decoder
- mvha decoder
- MPEG-H 3D Audio support in mp4
- thistogram filter
- freezeframes filter
- Argonaut Games ADPCM decoder
- Argonaut Games ASF demuxer
- xfade video filter
- xfade_opencl filter
- afirsrc audio filter source
- pad_opencl filter
- Simon & Schuster Interactive ADPCM decoder
- Real War KVAG demuxer
- CDToons video decoder
- siren audio decoder
- Rayman 2 ADPCM decoder
- Rayman 2 APM demuxer
- cas video filter
- High Voltage Software ADPCM decoder
- LEGO Racers ALP (.tun & .pcm) demuxer
- AMQP 0-9-1 protocol (RabbitMQ)
- Vulkan support
- avgblur_vulkan, overlay_vulkan, scale_vulkan and chromaber_vulkan filters
- ADPCM IMA MTF decoder
- FWSE demuxer
- DERF DPCM decoder
- DERF demuxer
- CRI HCA decoder
- CRI HCA demuxer
- overlay_cuda filter
- switch from AvxSynth to AviSynth+ on Linux
- mv30 decoder
- Expanded styling support for 3GPP Timed Text Subtitles (movtext)
- WebP parser
- tmedian filter
- maskedthreshold filter
- Support for muxing pcm and pgs in m2ts
- Cunning Developments ADPCM decoder
- asubboost filter
- Pro Pinball Series Soundbank demuxer
- pcm_rechunk bitstream filter
- scdet filter
- NotchLC decoder
- gradients source video filter
- MediaFoundation encoder wrapper
- untile filter
- Simon & Schuster Interactive ADPCM encoder
- PFM decoder
- dblur video filter
- Real War KVAG muxer
version 4.2:
- tpad filter
- AV1 decoding support through libdav1d
- dedot filter
- chromashift and rgbashift filters
- freezedetect filter
- truehd_core bitstream filter
- dhav demuxer
- PCM-DVD encoder
- GIF parser
- vividas demuxer
- hymt decoder
- anlmdn filter
- maskfun filter
- hcom demuxer and decoder
- ARBC decoder
- libaribb24 based ARIB STD-B24 caption support (profiles A and C)
- Support decoding of HEVC 4:4:4 content in nvdec and cuviddec
- removed libndi-newtek
- agm decoder
- KUX demuxer
- AV1 frame split bitstream filter
- lscr decoder
- lagfun filter
- asoftclip filter
- Support decoding of HEVC 4:4:4 content in vdpau
- colorhold filter
- xmedian filter
- asr filter
- showspatial multimedia filter
- VP4 video decoder
- IFV demuxer
- derain filter
- deesser filter
- mov muxer writes tracks with unspecified language instead of English by default
- add support for using clang to compile CUDA kernels
version 4.1:
- deblock filter
- tmix filter

2
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/INSTALL.md → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/INSTALL.md
View file

@ -1,4 +1,4 @@
## Installing FFmpeg
#Installing FFmpeg:
1. Type `./configure` to create the configuration. A list of configure
options is printed by running `configure --help`.

41
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/LICENSE.md → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/LICENSE.md
View file

@ -21,11 +21,10 @@ Specifically, the GPL parts of FFmpeg are:
- `compat/solaris/make_sunver.pl`
- `doc/t2h.pm`
- `doc/texi2pod.pl`
- `libswresample/tests/swresample.c`
- `libswresample/swresample-test.c`
- `tests/checkasm/*`
- `tests/tiny_ssim.c`
- the following filters in libavfilter:
- `signature_lookup.c`
- `vf_blackframe.c`
- `vf_boxblur.c`
- `vf_colormatrix.c`
@ -35,13 +34,13 @@ Specifically, the GPL parts of FFmpeg are:
- `vf_eq.c`
- `vf_find_rect.c`
- `vf_fspp.c`
- `vf_geq.c`
- `vf_histeq.c`
- `vf_hqdn3d.c`
- `vf_interlace.c`
- `vf_kerndeint.c`
- `vf_lensfun.c` (GPL version 3 or later)
- `vf_mcdeint.c`
- `vf_mpdecimate.c`
- `vf_nnedi.c`
- `vf_owdenoise.c`
- `vf_perspective.c`
- `vf_phase.c`
@ -50,14 +49,12 @@ Specifically, the GPL parts of FFmpeg are:
- `vf_pullup.c`
- `vf_repeatfields.c`
- `vf_sab.c`
- `vf_signature.c`
- `vf_smartblur.c`
- `vf_spp.c`
- `vf_stereo3d.c`
- `vf_super2xsai.c`
- `vf_tinterlace.c`
- `vf_uspp.c`
- `vf_vaguedenoiser.c`
- `vsrc_mptestsrc.c`
Should you, for whatever reason, prefer to use version 3 of the (L)GPL, then
@ -83,47 +80,41 @@ affect the licensing of binaries resulting from the combination.
### Compatible libraries
The following libraries are under GPL version 2:
- avisynth
The following libraries are under GPL:
- frei0r
- libcdio
- libdavs2
- librubberband
- libvidstab
- libx264
- libx265
- libxavs
- libxavs2
- libxvid
When combining them with FFmpeg, FFmpeg needs to be licensed as GPL as well by
passing `--enable-gpl` to configure.
The following libraries are under LGPL version 3:
- gmp
- libaribb24
- liblensfun
When combining them with FFmpeg, use the configure option `--enable-version3` to
upgrade FFmpeg to the LGPL v3.
The VMAF, mbedTLS, RK MPI, OpenCORE and VisualOn libraries are under the Apache License
2.0. That license is incompatible with the LGPL v2.1 and the GPL v2, but not with
The OpenCORE and VisualOn libraries are under the Apache License 2.0. That
license is incompatible with the LGPL v2.1 and the GPL v2, but not with
version 3 of those licenses. So to combine these libraries with FFmpeg, the
license version needs to be upgraded by passing `--enable-version3` to configure.
The smbclient library is under the GPL v3, to combine it with FFmpeg,
the options `--enable-gpl` and `--enable-version3` have to be passed to
configure to upgrade FFmpeg to the GPL v3.
### 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. This will cause the resulting binary to be
`--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.

88
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/MAINTAINERS → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/MAINTAINERS
View file

@ -11,11 +11,17 @@ A (CC <address>) after the name means that the maintainer prefers to be CC-ed on
patches and related discussions.
Project Leader
==============
final design decisions
Applications
============
ffmpeg:
ffmpeg.c Michael Niedermayer, Anton Khirnov
ffmpeg.c Michael Niedermayer
ffplay:
ffplay.c Marton Balint
@ -33,9 +39,8 @@ QuickTime faststart:
Miscellaneous Areas
===================
documentation Stefano Sabatini, Mike Melanson, Timothy Gu, Gyan Doshi
project server day to day operations Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser, Nikolay Aleksandrov
project server emergencies Árpád Gereöffy, Reimar Doeffinger, Alexander Strasser, Nikolay Aleksandrov
documentation Stefano Sabatini, Mike Melanson, Timothy Gu, Lou Logan, Gyan Doshi
project server Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser, Nikolay Aleksandrov
presets Robert Swain
metadata subsystem Aurelien Jacobs
release management Michael Niedermayer
@ -47,12 +52,12 @@ Communication
website Deby Barbara Lepage
fate.ffmpeg.org Timothy Gu
Trac bug tracker Alexander Strasser, Michael Niedermayer, Carl Eugen Hoyos
Patchwork Andriy Gelman
mailing lists Baptiste Coudurier
Twitter Reynaldo H. Verdejo Pinochet
Trac bug tracker Alexander Strasser, Michael Niedermayer, Carl Eugen Hoyos, Lou Logan
mailing lists 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, rcombs, wm4
ffmpeg-security Andreas Cadhalpun, Carl Eugen Hoyos, Clément Bœsch, Michael Niedermayer, Reimar Doeffinger, Rodger Combs, wm4
libavutil
@ -73,7 +78,6 @@ Other:
float_dsp Loren Merritt
hash Reimar Doeffinger
hwcontext_cuda* Timo Rothenpieler
hwcontext_vulkan* Lynne
intfloat* Michael Niedermayer
integer.c, integer.h Michael Niedermayer
lzo Reimar Doeffinger
@ -84,7 +88,6 @@ Other:
rational.c, rational.h Michael Niedermayer
rc4 Reimar Doeffinger
ripemd.c, ripemd.h James Almer
tx* Lynne
libavcodec
@ -110,6 +113,8 @@ 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
motion estimation:
@ -131,16 +136,13 @@ Codecs:
8bps.c Roberto Togni
8svx.c Jaikrishnan Menon
aacenc*, aaccoder.c Rostislav Pehlivanov
adpcm.c Zane van Iperen
alacenc.c Jaikrishnan Menon
alsdec.c Thilo Borgmann, Umair Khan
amfenc* Dmitrii Ovchinnikov
aptx.c Aurelien Jacobs
ass* Aurelien Jacobs
asv* Michael Niedermayer
atrac3plus* Maxim Poliakovski
audiotoolbox* rcombs
avs2* Huiwen Ren
audiotoolbox* Rodger Combs
bgmc.c, bgmc.h Thilo Borgmann
binkaudio.c Peter Ross
cavs* Stefan Gehrer
@ -151,10 +153,10 @@ Codecs:
ccaption_dec.c Anshul Maheshwari, Aman Gupta
cljr Alex Beregszaszi
cpia.c Stephan Hilb
crystalhd.c Philip Langdale
cscd.c Reimar Doeffinger
cuviddec.c Timo Rothenpieler
dca* foo86
dfpwm* Jack Bruienne
dirac* Rostislav Pehlivanov
dnxhd* Baptiste Coudurier
dolby_e* foo86
@ -165,6 +167,7 @@ Codecs:
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
@ -186,18 +189,14 @@ Codecs:
libcelt_dec.c Nicolas George
libcodec2.c Tomas Härdin
libdirac* David Conrad
libdavs2.c Huiwen Ren
libjxl*.c, libjxl.h Leo Izen
libgsm.c Michel Bardiaux
libkvazaar.c Arttu Ylä-Outinen
libopenh264enc.c Martin Storsjo, Linjie Fu
libopenjpeg.c Jaikrishnan Menon
libopenjpegenc.c Michael Bradshaw
libtheoraenc.c David Conrad
libvorbis.c David Conrad
libvpx* James Zern
libxavs.c Stefan Gehrer
libxavs2.c Huiwen Ren
libzvbi-teletextdec.c Marton Balint
lzo.h, lzo.c Reimar Doeffinger
mdec.c Michael Niedermayer
@ -213,7 +212,6 @@ Codecs:
msvideo1.c Mike Melanson
nuv.c Reimar Doeffinger
nvdec*, nvenc* Timo Rothenpieler
omx.c Martin Storsjo, Aman Gupta
opus* Rostislav Pehlivanov
paf.* Paul B Mahol
pcx.c Ivo van Poorten
@ -221,7 +219,7 @@ Codecs:
ptx.c Ivo van Poorten
qcelp* Reynaldo H. Verdejo Pinochet
qdm2.c, qdm2data.h Roberto Togni
qsv* Mark Thompson, Zhong Li, Haihao Xiang
qsv* Mark Thompson, Zhong Li
qtrle.c Mike Melanson
ra144.c, ra144.h, ra288.c, ra288.h Roberto Togni
resample2.c Michael Niedermayer
@ -231,6 +229,7 @@ Codecs:
rv10.c Michael Niedermayer
s3tc* Ivo van Poorten
smc.c Mike Melanson
smvjpegdec.c Ash Hughes
snow* Michael Niedermayer, Loren Merritt
sonic.c Alex Beregszaszi
speedhq.c Steinar H. Gunderson
@ -261,14 +260,16 @@ Codecs:
xan.c Mike Melanson
xbm* Paul B Mahol
xface Stefano Sabatini
xvmc.c Ivan Kalvachev
xwd* Paul B Mahol
Hardware acceleration:
crystalhd.c Philip Langdale
dxva2* Hendrik Leppkes, Laurent Aimar, Steve Lhomme
d3d11va* Steve Lhomme
mediacodec* Matthieu Bouron, Aman Gupta
vaapi* Haihao Xiang
vaapi_encode* Mark Thompson, Haihao Xiang
vaapi* Gwenole Beauchesne
vaapi_encode* Mark Thompson
vdpau* Philip Langdale, Carl Eugen Hoyos
videotoolbox* Rick Kern, Aman Gupta
@ -347,7 +348,6 @@ Filters:
vf_il.c Paul B Mahol
vf_(t)interlace Thomas Mundt (CC <thomas.mundt@hr.de>)
vf_lenscorrection.c Daniel Oberhoff
vf_libplacebo.c Niklas Haas
vf_mergeplanes.c Paul B Mahol
vf_mestimate.c Davinder Singh
vf_minterpolate.c Davinder Singh
@ -360,15 +360,12 @@ Filters:
vf_ssim.c Paul B Mahol
vf_stereo3d.c Paul B Mahol
vf_telecine.c Paul B Mahol
vf_tonemap_opencl.c Ruiling Song
vf_yadif.c Michael Niedermayer
vf_zoompan.c Paul B Mahol
Sources:
vsrc_mandelbrot.c Michael Niedermayer
dnn Yejun Guo
libavformat
===========
@ -387,13 +384,7 @@ Muxers/Demuxers:
afc.c Paul B Mahol
aiffdec.c Baptiste Coudurier, Matthieu Bouron
aiffenc.c Baptiste Coudurier, Matthieu Bouron
alp.c Zane van Iperen
amvenc.c Zane van Iperen
apm.c Zane van Iperen
apngdec.c Benoit Fouet
argo_asf.c Zane van Iperen
argo_brp.c Zane van Iperen
argo_cvg.c Zane van Iperen
ass* Aurelien Jacobs
astdec.c Paul B Mahol
astenc.c James Almer
@ -410,7 +401,6 @@ Muxers/Demuxers:
dashdec.c Steven Liu
dashenc.c Karthick Jeyapal
daud.c Reimar Doeffinger
dfpwmdec.c Jack Bruienne
dss.c Oleksij Rempel
dtsdec.c foo86
dtshddec.c Paul B Mahol
@ -427,22 +417,19 @@ Muxers/Demuxers:
idcin.c Mike Melanson
idroqdec.c Mike Melanson
iff.c Jaikrishnan Menon
imf* Pierre-Anthony Lemieux
img2*.c Michael Niedermayer
ipmovie.c Mike Melanson
ircam* Paul B Mahol
iss.c Stefan Gehrer
jpegxl_probe.* Leo Izen
jvdec.c Peter Ross
kvag.c Zane van Iperen
libmodplug.c Clément Bœsch
libopenmpt.c Josh de Kock
lmlm4.c Ivo van Poorten
lvfdec.c Paul B Mahol
lxfdec.c Tomas Härdin
matroska.c Aurelien Jacobs, Andreas Rheinhardt
matroskadec.c Aurelien Jacobs, Andreas Rheinhardt
matroskaenc.c David Conrad, Andreas Rheinhardt
matroska.c Aurelien Jacobs
matroskadec.c Aurelien Jacobs
matroskaenc.c David Conrad
matroska subtitles (matroskaenc.c) John Peebles
metadata* Aurelien Jacobs
mgsts.c Paul B Mahol
@ -457,7 +444,7 @@ Muxers/Demuxers:
mpegtsenc.c Baptiste Coudurier
msnwc_tcp.c Ramiro Polla
mtv.c Reynaldo H. Verdejo Pinochet
mxf* Baptiste Coudurier, Tomas Härdin
mxf* Baptiste Coudurier
nistspheredec.c Paul B Mahol
nsvdec.c Francois Revol
nut* Michael Niedermayer
@ -465,9 +452,9 @@ Muxers/Demuxers:
oggdec.c, oggdec.h David Conrad
oggenc.c Baptiste Coudurier
oggparse*.c David Conrad
oggparsedaala* Rostislav Pehlivanov
oma.c Maxim Poliakovski
paf.c Paul B Mahol
pp_bnk.c Zane van Iperen
psxstr.c Mike Melanson
pva.c Ivo van Poorten
pvfdec.c Paul B Mahol
@ -512,9 +499,7 @@ Protocols:
bluray.c Petri Hintukainen
ftp.c Lukasz Marek
http.c Ronald S. Bultje
libsrt.c Zhao Zhili
libssh.c Lukasz Marek
libzmq.c Andriy Gelman
mms*.c Ronald S. Bultje
udp.c Luca Abeni
icecast.c Marvin Scholz
@ -539,10 +524,8 @@ Operating systems / CPU architectures
Alpha Falk Hueffner
MIPS Manojkumar Bhosale, Shiyou Yin
LoongArch Shiyou Yin
Mac OS X / PowerPC Romain Dolbeau, Guillaume Poirier
Amiga / PowerPC Colin Ward
Linux / PowerPC Lauri Kasanen
Windows MinGW Alex Beregszaszi, Ramiro Polla
Windows Cygwin Victor Paesa
Windows MSVC Matthew Oliver, Hendrik Leppkes
@ -571,7 +554,6 @@ Joakim Plate
Jun Zhao
Kieran Kunhya
Kirill Gavrilov
Limin Wang
Martin Storsjö
Panagiotis Issaris
Pedro Arthur
@ -610,23 +592,17 @@ Daniel Verkamp 78A6 07ED 782C 653E C628 B8B9 F0EB 8DD8 2F0E 21C7
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
Haihao Xiang (haihao) 1F0C 31E8 B4FE F7A4 4DC1 DC99 E0F5 76D4 76FC 437F
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
Leo Izen (thebombzen) B6FD 3CFC 7ACF 83FC 9137 6945 5A71 C331 FD2F A19A
Loren Merritt ABD9 08F4 C920 3F65 D8BE 35D7 1540 DAA7 060F 56DE
Lynne FE50 139C 6805 72CA FD52 1F8D A2FE A5F0 3F03 4464
Lou Logan (llogan) 7D68 DC73 CBEF EABB 671A B6CF 621C 2E28 82F8 DC3A
Michael Niedermayer 9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
DD1E C9E8 DE08 5C62 9B3E 1846 B18E 8928 B394 8D64
Nicolas George 24CE 01CE 9ACC 5CEB 74D8 8D9D B063 D997 36E5 4C93
Niklas Haas (haasn) 1DDB 8076 B14D 5B48 32FC 99D9 EB52 DA9C 02BA 6FB4
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
Pierre-Anthony Lemieux (pal) F4B3 9492 E6F2 E4AF AEC8 46CB 698F A1F0 F8D4 EED4
Ramiro Polla 7859 C65B 751B 1179 792E DAE8 8E95 8B2F 9B6C 5700
Reimar Doeffinger C61D 16E5 9E2C D10C 8958 38A4 0899 A2B9 06D4 D9C7
Reinhard Tartler 9300 5DC2 7E87 6C37 ED7B CA9A 9808 3544 9453 48A4
Reynaldo H. Verdejo Pinochet 6E27 CD34 170C C78E 4D4F 5F40 C18E 077F 3114 452A
@ -635,9 +611,7 @@ 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
Thilo Borgmann (thilo) CE1D B7F4 4D20 FC3A DD9F FE5A 257C 5B8F 1D20 B92F
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
Wei Gao 4269 7741 857A 0E60 9EC5 08D2 4744 4EFA 62C1 87B9
Zane van Iperen (zane) 61AE D40F 368B 6F26 9DAE 3892 6861 6B2D 8AC4 DCC5

174
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/Makefile Normal file
View file

@ -0,0 +1,174 @@
MAIN_MAKEFILE=1
include ffbuild/config.mak
vpath %.c $(SRC_PATH)
vpath %.cpp $(SRC_PATH)
vpath %.h $(SRC_PATH)
vpath %.inc $(SRC_PATH)
vpath %.m $(SRC_PATH)
vpath %.S $(SRC_PATH)
vpath %.asm $(SRC_PATH)
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
HOSTPROGS := $(TESTTOOLS:%=tests/%) doc/print_options
# $(FFLIBS-yes) needs to be in linking order
FFLIBS-$(CONFIG_AVDEVICE) += avdevice
FFLIBS-$(CONFIG_AVFILTER) += avfilter
FFLIBS-$(CONFIG_AVFORMAT) += avformat
FFLIBS-$(CONFIG_AVCODEC) += avcodec
FFLIBS-$(CONFIG_AVRESAMPLE) += avresample
FFLIBS-$(CONFIG_POSTPROC) += postproc
FFLIBS-$(CONFIG_SWRESAMPLE) += swresample
FFLIBS-$(CONFIG_SWSCALE) += swscale
FFLIBS := avutil
DATA_FILES := $(wildcard $(SRC_PATH)/presets/*.ffpreset) $(SRC_PATH)/doc/ffprobe.xsd
SKIPHEADERS = compat/w32pthreads.h
# first so "all" becomes default target
all: all-yes
include $(SRC_PATH)/tools/Makefile
include $(SRC_PATH)/ffbuild/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) $(ELIBS)
target_dec_%_fuzzer$(EXESUF): target_dec_%_fuzzer.o $(FF_DEP_LIBS)
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
tools/sofa2wavs$(EXESUF): ELIBS = $(FF_EXTRALIBS)
tools/uncoded_frame$(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)
@-tput bold 2>/dev/null
@-printf '\nWARNING: $(?) newer than config.h, rerun configure\n\n'
@-tput sgr0 2>/dev/null
SUBDIR_VARS := CLEANFILES 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 \
MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSP-OBJS MSA-OBJS \
MMI-OBJS OBJS SLIBOBJS HOSTOBJS TESTOBJS
define RESET
$(1) :=
$(1)-yes :=
endef
define DOSUBDIR
$(foreach V,$(SUBDIR_VARS),$(eval $(call RESET,$(V))))
SUBDIR := $(1)/
include $(SRC_PATH)/$(1)/Makefile
-include $(SRC_PATH)/$(1)/$(ARCH)/Makefile
-include $(SRC_PATH)/$(1)/$(INTRINSICS)/Makefile
include $(SRC_PATH)/ffbuild/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
$(PROGS): %$(PROGSSUF)$(EXESUF): %$(PROGSSUF)_g$(EXESUF)
ifeq ($(STRIPTYPE),direct)
$(STRIP) -o $@ $<
else
$(CP) $< $@
$(STRIP) $@
endif
%$(PROGSSUF)_g$(EXESUF): $(FF_DEP_LIBS)
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
VERSION_SH = $(SRC_PATH)/ffbuild/version.sh
GIT_LOG = $(SRC_PATH)/.git/logs/HEAD
.version: $(wildcard $(GIT_LOG)) $(VERSION_SH) ffbuild/config.mak
.version: M=@
libavutil/ffversion.h .version:
$(M)$(VERSION_SH) $(SRC_PATH) libavutil/ffversion.h $(EXTRA_VERSION)
$(Q)touch .version
# force version.sh to run whenever version might have changed
-include .version
install: install-libs install-headers
install-libs: install-libs-yes
install-data: $(DATA_FILES)
$(Q)mkdir -p "$(DATADIR)"
$(INSTALL) -m 644 $(DATA_FILES) "$(DATADIR)"
uninstall: uninstall-data uninstall-headers uninstall-libs uninstall-pkgconfig
uninstall-data:
$(RM) -r "$(DATADIR)"
clean::
$(RM) $(CLEANSUFFIXES)
$(RM) $(addprefix compat/,$(CLEANSUFFIXES)) $(addprefix compat/*/,$(CLEANSUFFIXES))
$(RM) -r coverage-html
$(RM) -rf coverage.info coverage.info.in lcov
distclean:: clean
$(RM) .version avversion.h config.asm config.h mapfile \
ffbuild/.config ffbuild/config.* libavutil/avconfig.h \
version.h libavutil/ffversion.h libavcodec/codec_names.h \
libavcodec/bsf_list.c libavformat/protocol_list.c \
libavcodec/codec_list.c libavcodec/parser_list.c \
libavformat/muxer_list.c libavformat/demuxer_list.c
ifeq ($(SRC_LINK),src)
$(RM) src
endif
$(RM) -rf doc/examples/pc-uninstalled
config:
$(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION)
build: all alltools examples testprogs
check: all alltools examples testprogs fate
include $(SRC_PATH)/tests/Makefile
$(sort $(OBJDIRS)):
$(Q)mkdir -p $@
# Dummy rule to stop make trying to rebuild removed or renamed headers
%.h:
@:
# Disable suffix rules. Most of the builtin rules are suffix rules,
# so this saves some time on slow systems.
.SUFFIXES:
.PHONY: all all-yes alltools build check config testprogs
.PHONY: *clean install* uninstall*

2
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/README.md → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/README.md
View file

@ -9,7 +9,7 @@ such as audio, video, subtitles and related metadata.
* `libavcodec` provides implementation of a wider range of codecs.
* `libavformat` implements streaming protocols, container formats and basic I/O access.
* `libavutil` includes hashers, decompressors and miscellaneous utility functions.
* `libavfilter` provides means to alter decoded audio and video through a directed graph of connected filters.
* `libavfilter` provides a mean to alter decoded Audio and Video through chain of filters.
* `libavdevice` provides an abstraction to access capture and playback devices.
* `libswresample` implements audio mixing and resampling routines.
* `libswscale` implements color conversion and scaling routines.

1
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/RELEASE Normal file
View file

@ -0,0 +1 @@
4.1

15
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/RELEASE_NOTES Normal file
View file

@ -0,0 +1,15 @@
┌─────────────────────────────────────────────┐
│ RELEASE NOTES for FFmpeg 4.1 "al-Khwarizmi" │
└─────────────────────────────────────────────┘
The FFmpeg Project proudly presents FFmpeg 4.1 "al-Khwarizmi", about 6
months after the release of FFmpeg 4.0.
A complete Changelog is available at the root of the project, and the
complete Git history on https://git.ffmpeg.org/gitweb/ffmpeg.git
We hope you will like this release as much as we enjoyed working on it, and
as usual, if you have any questions about it, or any FFmpeg related topic,
feel free to join us on the #ffmpeg IRC channel (on irc.freenode.net) or ask
on the mailing-lists.

1
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/VERSION Normal file
View file

@ -0,0 +1 @@
4.1

0
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/compat/aix/math.h → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/compat/aix/math.h
View file

0
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/compat/atomics/dummy/stdatomic.h → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/compat/atomics/dummy/stdatomic.h
View file

0
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/compat/atomics/gcc/stdatomic.h → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/compat/atomics/gcc/stdatomic.h
View file

0
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/compat/atomics/pthread/stdatomic.c → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/compat/atomics/pthread/stdatomic.c
View file

0
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/compat/atomics/pthread/stdatomic.h → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/compat/atomics/pthread/stdatomic.h
View file

0
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/compat/atomics/suncc/stdatomic.h → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/compat/atomics/suncc/stdatomic.h
View file

2
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/compat/atomics/win32/stdatomic.h → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/compat/atomics/win32/stdatomic.h
View file

@ -96,7 +96,7 @@ do { \
atomic_load(object)
#define atomic_exchange(object, desired) \
InterlockedExchangePointer((PVOID volatile *)object, (PVOID)desired)
InterlockedExchangePointer(object, desired);
#define atomic_exchange_explicit(object, desired, order) \
atomic_exchange(object, desired)

1064
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/compat/avisynth/avisynth_c.h Normal file
View file

File diff suppressed because it is too large Load diff

62
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/compat/avisynth/avs/capi.h Normal file
View file

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

55
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/compat/avisynth/avs/config.h Normal file
View file

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

51
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/compat/avisynth/avs/types.h Normal file
View file

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

728
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/compat/avisynth/avxsynth_c.h Normal file
View file

@ -0,0 +1,728 @@
// Avisynth C Interface Version 0.20
// Copyright 2003 Kevin Atkinson
// This program is free software; you can redistribute it and/or modify
// it under the terms of the GNU General Public License as published by
// the Free Software Foundation; either version 2 of the License, or
// (at your option) any later version.
//
// This program is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
// GNU General Public License for more details.
//
// You should have received a copy of the GNU General Public License
// along with this program; if not, write to the Free Software
// Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston,
// MA 02110-1301 USA, or visit
// http://www.gnu.org/copyleft/gpl.html .
//
// As a special exception, I give you permission to link to the
// Avisynth C interface with independent modules that communicate with
// the Avisynth C interface solely through the interfaces defined in
// avisynth_c.h, regardless of the license terms of these independent
// modules, and to copy and distribute the resulting combined work
// under terms of your choice, provided that every copy of the
// combined work is accompanied by a complete copy of the source code
// of the Avisynth C interface and Avisynth itself (with the version
// used to produce the combined work), being distributed under the
// terms of the GNU General Public License plus this exception. An
// independent module is a module which is not derived from or based
// on Avisynth C Interface, such as 3rd-party filters, import and
// export plugins, or graphical user interfaces.
#ifndef __AVXSYNTH_C__
#define __AVXSYNTH_C__
#include "windowsPorts/windows2linux.h"
#include <stdarg.h>
#ifdef __cplusplus
# define EXTERN_C extern "C"
#else
# define EXTERN_C
#endif
#define AVSC_USE_STDCALL 1
#ifndef AVSC_USE_STDCALL
# define AVSC_CC __cdecl
#else
# define AVSC_CC __stdcall
#endif
#define AVSC_INLINE static __inline
#ifdef AVISYNTH_C_EXPORTS
# define AVSC_EXPORT EXTERN_C
# define AVSC_API(ret, name) EXTERN_C __declspec(dllexport) ret AVSC_CC name
#else
# define AVSC_EXPORT EXTERN_C __declspec(dllexport)
# ifndef AVSC_NO_DECLSPEC
# define AVSC_API(ret, name) EXTERN_C __declspec(dllimport) ret AVSC_CC name
# else
# define AVSC_API(ret, name) typedef ret (AVSC_CC *name##_func)
# endif
#endif
#ifdef __GNUC__
typedef long long int INT64;
#else
typedef __int64 INT64;
#endif
/////////////////////////////////////////////////////////////////////
//
// Constants
//
#ifndef __AVXSYNTH_H__
enum { AVISYNTH_INTERFACE_VERSION = 3 };
#endif
enum {AVS_SAMPLE_INT8 = 1<<0,
AVS_SAMPLE_INT16 = 1<<1,
AVS_SAMPLE_INT24 = 1<<2,
AVS_SAMPLE_INT32 = 1<<3,
AVS_SAMPLE_FLOAT = 1<<4};
enum {AVS_PLANAR_Y=1<<0,
AVS_PLANAR_U=1<<1,
AVS_PLANAR_V=1<<2,
AVS_PLANAR_ALIGNED=1<<3,
AVS_PLANAR_Y_ALIGNED=AVS_PLANAR_Y|AVS_PLANAR_ALIGNED,
AVS_PLANAR_U_ALIGNED=AVS_PLANAR_U|AVS_PLANAR_ALIGNED,
AVS_PLANAR_V_ALIGNED=AVS_PLANAR_V|AVS_PLANAR_ALIGNED};
// Colorspace properties.
enum {AVS_CS_BGR = 1<<28,
AVS_CS_YUV = 1<<29,
AVS_CS_INTERLEAVED = 1<<30,
AVS_CS_PLANAR = 1<<31};
// Specific colorformats
enum {
AVS_CS_UNKNOWN = 0,
AVS_CS_BGR24 = 1<<0 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
AVS_CS_BGR32 = 1<<1 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
AVS_CS_YUY2 = 1<<2 | AVS_CS_YUV | AVS_CS_INTERLEAVED,
AVS_CS_YV12 = 1<<3 | AVS_CS_YUV | AVS_CS_PLANAR, // y-v-u, planar
AVS_CS_I420 = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR, // y-u-v, planar
AVS_CS_IYUV = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR // same as above
};
enum {
AVS_IT_BFF = 1<<0,
AVS_IT_TFF = 1<<1,
AVS_IT_FIELDBASED = 1<<2};
enum {
AVS_FILTER_TYPE=1,
AVS_FILTER_INPUT_COLORSPACE=2,
AVS_FILTER_OUTPUT_TYPE=9,
AVS_FILTER_NAME=4,
AVS_FILTER_AUTHOR=5,
AVS_FILTER_VERSION=6,
AVS_FILTER_ARGS=7,
AVS_FILTER_ARGS_INFO=8,
AVS_FILTER_ARGS_DESCRIPTION=10,
AVS_FILTER_DESCRIPTION=11};
enum { //SUBTYPES
AVS_FILTER_TYPE_AUDIO=1,
AVS_FILTER_TYPE_VIDEO=2,
AVS_FILTER_OUTPUT_TYPE_SAME=3,
AVS_FILTER_OUTPUT_TYPE_DIFFERENT=4};
enum {
AVS_CACHE_NOTHING=0,
AVS_CACHE_RANGE=1,
AVS_CACHE_ALL=2,
AVS_CACHE_AUDIO=3,
AVS_CACHE_AUDIO_NONE=4,
AVS_CACHE_AUDIO_AUTO=5
};
#define AVS_FRAME_ALIGN 16
typedef struct AVS_Clip AVS_Clip;
typedef struct AVS_ScriptEnvironment AVS_ScriptEnvironment;
/////////////////////////////////////////////////////////////////////
//
// AVS_VideoInfo
//
// AVS_VideoInfo is layed out identicly to VideoInfo
typedef struct AVS_VideoInfo {
int width, height; // width=0 means no video
unsigned fps_numerator, fps_denominator;
int num_frames;
int pixel_type;
int audio_samples_per_second; // 0 means no audio
int sample_type;
INT64 num_audio_samples;
int nchannels;
// Imagetype properties
int image_type;
} AVS_VideoInfo;
// useful functions of the above
AVSC_INLINE int avs_has_video(const AVS_VideoInfo * p)
{ return (p->width!=0); }
AVSC_INLINE int avs_has_audio(const AVS_VideoInfo * p)
{ return (p->audio_samples_per_second!=0); }
AVSC_INLINE int avs_is_rgb(const AVS_VideoInfo * p)
{ return !!(p->pixel_type&AVS_CS_BGR); }
AVSC_INLINE int avs_is_rgb24(const AVS_VideoInfo * p)
{ return (p->pixel_type&AVS_CS_BGR24)==AVS_CS_BGR24; } // Clear out additional properties
AVSC_INLINE int avs_is_rgb32(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_BGR32) == AVS_CS_BGR32 ; }
AVSC_INLINE int avs_is_yuv(const AVS_VideoInfo * p)
{ return !!(p->pixel_type&AVS_CS_YUV ); }
AVSC_INLINE int avs_is_yuy2(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_YUY2) == AVS_CS_YUY2; }
AVSC_INLINE int avs_is_yv12(const AVS_VideoInfo * p)
{ return ((p->pixel_type & AVS_CS_YV12) == AVS_CS_YV12)||((p->pixel_type & AVS_CS_I420) == AVS_CS_I420); }
AVSC_INLINE int avs_is_color_space(const AVS_VideoInfo * p, int c_space)
{ return ((p->pixel_type & c_space) == c_space); }
AVSC_INLINE int avs_is_property(const AVS_VideoInfo * p, int property)
{ return ((p->pixel_type & property)==property ); }
AVSC_INLINE int avs_is_planar(const AVS_VideoInfo * p)
{ return !!(p->pixel_type & AVS_CS_PLANAR); }
AVSC_INLINE int avs_is_field_based(const AVS_VideoInfo * p)
{ return !!(p->image_type & AVS_IT_FIELDBASED); }
AVSC_INLINE int avs_is_parity_known(const AVS_VideoInfo * p)
{ return ((p->image_type & AVS_IT_FIELDBASED)&&(p->image_type & (AVS_IT_BFF | AVS_IT_TFF))); }
AVSC_INLINE int avs_is_bff(const AVS_VideoInfo * p)
{ return !!(p->image_type & AVS_IT_BFF); }
AVSC_INLINE int avs_is_tff(const AVS_VideoInfo * p)
{ return !!(p->image_type & AVS_IT_TFF); }
AVSC_INLINE int avs_bits_per_pixel(const AVS_VideoInfo * p)
{
switch (p->pixel_type) {
case AVS_CS_BGR24: return 24;
case AVS_CS_BGR32: return 32;
case AVS_CS_YUY2: return 16;
case AVS_CS_YV12:
case AVS_CS_I420: return 12;
default: return 0;
}
}
AVSC_INLINE int avs_bytes_from_pixels(const AVS_VideoInfo * p, int pixels)
{ return pixels * (avs_bits_per_pixel(p)>>3); } // Will work on planar images, but will return only luma planes
AVSC_INLINE int avs_row_size(const AVS_VideoInfo * p)
{ return avs_bytes_from_pixels(p,p->width); } // Also only returns first plane on planar images
AVSC_INLINE int avs_bmp_size(const AVS_VideoInfo * vi)
{ if (avs_is_planar(vi)) {int p = vi->height * ((avs_row_size(vi)+3) & ~3); p+=p>>1; return p; } return vi->height * ((avs_row_size(vi)+3) & ~3); }
AVSC_INLINE int avs_samples_per_second(const AVS_VideoInfo * p)
{ return p->audio_samples_per_second; }
AVSC_INLINE int avs_bytes_per_channel_sample(const AVS_VideoInfo * p)
{
switch (p->sample_type) {
case AVS_SAMPLE_INT8: return sizeof(signed char);
case AVS_SAMPLE_INT16: return sizeof(signed short);
case AVS_SAMPLE_INT24: return 3;
case AVS_SAMPLE_INT32: return sizeof(signed int);
case AVS_SAMPLE_FLOAT: return sizeof(float);
default: return 0;
}
}
AVSC_INLINE int avs_bytes_per_audio_sample(const AVS_VideoInfo * p)
{ return p->nchannels*avs_bytes_per_channel_sample(p);}
AVSC_INLINE INT64 avs_audio_samples_from_frames(const AVS_VideoInfo * p, INT64 frames)
{ return ((INT64)(frames) * p->audio_samples_per_second * p->fps_denominator / p->fps_numerator); }
AVSC_INLINE int avs_frames_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
{ return (int)(samples * (INT64)p->fps_numerator / (INT64)p->fps_denominator / (INT64)p->audio_samples_per_second); }
AVSC_INLINE INT64 avs_audio_samples_from_bytes(const AVS_VideoInfo * p, INT64 bytes)
{ return bytes / avs_bytes_per_audio_sample(p); }
AVSC_INLINE INT64 avs_bytes_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
{ return samples * avs_bytes_per_audio_sample(p); }
AVSC_INLINE int avs_audio_channels(const AVS_VideoInfo * p)
{ return p->nchannels; }
AVSC_INLINE int avs_sample_type(const AVS_VideoInfo * p)
{ return p->sample_type;}
// useful mutator
AVSC_INLINE void avs_set_property(AVS_VideoInfo * p, int property)
{ p->image_type|=property; }
AVSC_INLINE void avs_clear_property(AVS_VideoInfo * p, int property)
{ p->image_type&=~property; }
AVSC_INLINE void avs_set_field_based(AVS_VideoInfo * p, int isfieldbased)
{ if (isfieldbased) p->image_type|=AVS_IT_FIELDBASED; else p->image_type&=~AVS_IT_FIELDBASED; }
AVSC_INLINE void avs_set_fps(AVS_VideoInfo * p, unsigned numerator, unsigned denominator)
{
unsigned x=numerator, y=denominator;
while (y) { // find gcd
unsigned t = x%y; x = y; y = t;
}
p->fps_numerator = numerator/x;
p->fps_denominator = denominator/x;
}
AVSC_INLINE int avs_is_same_colorspace(AVS_VideoInfo * x, AVS_VideoInfo * y)
{
return (x->pixel_type == y->pixel_type)
|| (avs_is_yv12(x) && avs_is_yv12(y));
}
/////////////////////////////////////////////////////////////////////
//
// AVS_VideoFrame
//
// VideoFrameBuffer holds information about a memory block which is used
// for video data. For efficiency, instances of this class are not deleted
// when the refcount reaches zero; instead they're stored in a linked list
// to be reused. The instances are deleted when the corresponding AVS
// file is closed.
// AVS_VideoFrameBuffer is layed out identicly to VideoFrameBuffer
// DO NOT USE THIS STRUCTURE DIRECTLY
typedef struct AVS_VideoFrameBuffer {
unsigned char * data;
int data_size;
// sequence_number is incremented every time the buffer is changed, so
// that stale views can tell they're no longer valid.
long sequence_number;
long refcount;
} AVS_VideoFrameBuffer;
// VideoFrame holds a "window" into a VideoFrameBuffer.
// AVS_VideoFrame is layed out identicly to IVideoFrame
// DO NOT USE THIS STRUCTURE DIRECTLY
typedef struct AVS_VideoFrame {
int refcount;
AVS_VideoFrameBuffer * vfb;
int offset, pitch, row_size, height, offsetU, offsetV, pitchUV; // U&V offsets are from top of picture.
} AVS_VideoFrame;
// Access functions for AVS_VideoFrame
AVSC_INLINE int avs_get_pitch(const AVS_VideoFrame * p) {
return p->pitch;}
AVSC_INLINE int avs_get_pitch_p(const AVS_VideoFrame * p, int plane) {
switch (plane) {
case AVS_PLANAR_U: case AVS_PLANAR_V: return p->pitchUV;}
return p->pitch;}
AVSC_INLINE int avs_get_row_size(const AVS_VideoFrame * p) {
return p->row_size; }
AVSC_INLINE int avs_get_row_size_p(const AVS_VideoFrame * p, int plane) {
int r;
switch (plane) {
case AVS_PLANAR_U: case AVS_PLANAR_V:
if (p->pitchUV) return p->row_size>>1;
else return 0;
case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED:
if (p->pitchUV) {
r = ((p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)) )>>1; // Aligned rowsize
if (r < p->pitchUV)
return r;
return p->row_size>>1;
} else return 0;
case AVS_PLANAR_Y_ALIGNED:
r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
if (r <= p->pitch)
return r;
return p->row_size;
}
return p->row_size;
}
AVSC_INLINE int avs_get_height(const AVS_VideoFrame * p) {
return p->height;}
AVSC_INLINE int avs_get_height_p(const AVS_VideoFrame * p, int plane) {
switch (plane) {
case AVS_PLANAR_U: case AVS_PLANAR_V:
if (p->pitchUV) return p->height>>1;
return 0;
}
return p->height;}
AVSC_INLINE const unsigned char* avs_get_read_ptr(const AVS_VideoFrame * p) {
return p->vfb->data + p->offset;}
AVSC_INLINE const unsigned char* avs_get_read_ptr_p(const AVS_VideoFrame * p, int plane)
{
switch (plane) {
case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
default: return p->vfb->data + p->offset;}
}
AVSC_INLINE int avs_is_writable(const AVS_VideoFrame * p) {
return (p->refcount == 1 && p->vfb->refcount == 1);}
AVSC_INLINE unsigned char* avs_get_write_ptr(const AVS_VideoFrame * p)
{
if (avs_is_writable(p)) {
++p->vfb->sequence_number;
return p->vfb->data + p->offset;
} else
return 0;
}
AVSC_INLINE unsigned char* avs_get_write_ptr_p(const AVS_VideoFrame * p, int plane)
{
if (plane==AVS_PLANAR_Y && avs_is_writable(p)) {
++p->vfb->sequence_number;
return p->vfb->data + p->offset;
} else if (plane==AVS_PLANAR_Y) {
return 0;
} else {
switch (plane) {
case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
default: return p->vfb->data + p->offset;
}
}
}
#if defined __cplusplus
extern "C"
{
#endif // __cplusplus
AVSC_API(void, avs_release_video_frame)(AVS_VideoFrame *);
// makes a shallow copy of a video frame
AVSC_API(AVS_VideoFrame *, avs_copy_video_frame)(AVS_VideoFrame *);
#if defined __cplusplus
}
#endif // __cplusplus
#ifndef AVSC_NO_DECLSPEC
AVSC_INLINE void avs_release_frame(AVS_VideoFrame * f)
{avs_release_video_frame(f);}
AVSC_INLINE AVS_VideoFrame * avs_copy_frame(AVS_VideoFrame * f)
{return avs_copy_video_frame(f);}
#endif
/////////////////////////////////////////////////////////////////////
//
// AVS_Value
//
// Treat AVS_Value as a fat pointer. That is use avs_copy_value
// and avs_release_value appropiaty as you would if AVS_Value was
// a pointer.
// To maintain source code compatibility with future versions of the
// avisynth_c API don't use the AVS_Value directly. Use the helper
// functions below.
// AVS_Value is layed out identicly to AVSValue
typedef struct AVS_Value AVS_Value;
struct AVS_Value {
short type; // 'a'rray, 'c'lip, 'b'ool, 'i'nt, 'f'loat, 's'tring, 'v'oid, or 'l'ong
// for some function e'rror
short array_size;
union {
void * clip; // do not use directly, use avs_take_clip
char boolean;
int integer;
INT64 integer64; // match addition of __int64 to avxplugin.h
float floating_pt;
const char * string;
const AVS_Value * array;
} d;
};
// AVS_Value should be initilized with avs_void.
// Should also set to avs_void after the value is released
// with avs_copy_value. Consider it the equalvent of setting
// a pointer to NULL
static const AVS_Value avs_void = {'v'};
AVSC_API(void, avs_copy_value)(AVS_Value * dest, AVS_Value src);
AVSC_API(void, avs_release_value)(AVS_Value);
AVSC_INLINE int avs_defined(AVS_Value v) { return v.type != 'v'; }
AVSC_INLINE int avs_is_clip(AVS_Value v) { return v.type == 'c'; }
AVSC_INLINE int avs_is_bool(AVS_Value v) { return v.type == 'b'; }
AVSC_INLINE int avs_is_int(AVS_Value v) { return v.type == 'i'; }
AVSC_INLINE int avs_is_float(AVS_Value v) { return v.type == 'f' || v.type == 'i'; }
AVSC_INLINE int avs_is_string(AVS_Value v) { return v.type == 's'; }
AVSC_INLINE int avs_is_array(AVS_Value v) { return v.type == 'a'; }
AVSC_INLINE int avs_is_error(AVS_Value v) { return v.type == 'e'; }
#if defined __cplusplus
extern "C"
{
#endif // __cplusplus
AVSC_API(AVS_Clip *, avs_take_clip)(AVS_Value, AVS_ScriptEnvironment *);
AVSC_API(void, avs_set_to_clip)(AVS_Value *, AVS_Clip *);
#if defined __cplusplus
}
#endif // __cplusplus
AVSC_INLINE int avs_as_bool(AVS_Value v)
{ return v.d.boolean; }
AVSC_INLINE int avs_as_int(AVS_Value v)
{ return v.d.integer; }
AVSC_INLINE const char * avs_as_string(AVS_Value v)
{ return avs_is_error(v) || avs_is_string(v) ? v.d.string : 0; }
AVSC_INLINE double avs_as_float(AVS_Value v)
{ return avs_is_int(v) ? v.d.integer : v.d.floating_pt; }
AVSC_INLINE const char * avs_as_error(AVS_Value v)
{ return avs_is_error(v) ? v.d.string : 0; }
AVSC_INLINE const AVS_Value * avs_as_array(AVS_Value v)
{ return v.d.array; }
AVSC_INLINE int avs_array_size(AVS_Value v)
{ return avs_is_array(v) ? v.array_size : 1; }
AVSC_INLINE AVS_Value avs_array_elt(AVS_Value v, int index)
{ return avs_is_array(v) ? v.d.array[index] : v; }
// only use these functions on am AVS_Value that does not already have
// an active value. Remember, treat AVS_Value as a fat pointer.
AVSC_INLINE AVS_Value avs_new_value_bool(int v0)
{ AVS_Value v = {0}; v.type = 'b'; v.d.boolean = v0 == 0 ? 0 : 1; return v; }
AVSC_INLINE AVS_Value avs_new_value_int(int v0)
{ AVS_Value v = {0}; v.type = 'i'; v.d.integer = v0; return v; }
AVSC_INLINE AVS_Value avs_new_value_string(const char * v0)
{ AVS_Value v = {0}; v.type = 's'; v.d.string = v0; return v; }
AVSC_INLINE AVS_Value avs_new_value_float(float v0)
{ AVS_Value v = {0}; v.type = 'f'; v.d.floating_pt = v0; return v;}
AVSC_INLINE AVS_Value avs_new_value_error(const char * v0)
{ AVS_Value v = {0}; v.type = 'e'; v.d.string = v0; return v; }
#ifndef AVSC_NO_DECLSPEC
AVSC_INLINE AVS_Value avs_new_value_clip(AVS_Clip * v0)
{ AVS_Value v = {0}; avs_set_to_clip(&v, v0); return v; }
#endif
AVSC_INLINE AVS_Value avs_new_value_array(AVS_Value * v0, int size)
{ AVS_Value v = {0}; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; }
/////////////////////////////////////////////////////////////////////
//
// AVS_Clip
//
#if defined __cplusplus
extern "C"
{
#endif // __cplusplus
AVSC_API(void, avs_release_clip)(AVS_Clip *);
AVSC_API(AVS_Clip *, avs_copy_clip)(AVS_Clip *);
AVSC_API(const char *, avs_clip_get_error)(AVS_Clip *); // return 0 if no error
AVSC_API(const AVS_VideoInfo *, avs_get_video_info)(AVS_Clip *);
AVSC_API(int, avs_get_version)(AVS_Clip *);
AVSC_API(AVS_VideoFrame *, avs_get_frame)(AVS_Clip *, int n);
// The returned video frame must be released with avs_release_video_frame
AVSC_API(int, avs_get_parity)(AVS_Clip *, int n);
// return field parity if field_based, else parity of first field in frame
AVSC_API(int, avs_get_audio)(AVS_Clip *, void * buf,
INT64 start, INT64 count);
// start and count are in samples
AVSC_API(int, avs_set_cache_hints)(AVS_Clip *,
int cachehints, size_t frame_range);
#if defined __cplusplus
}
#endif // __cplusplus
// This is the callback type used by avs_add_function
typedef AVS_Value (AVSC_CC * AVS_ApplyFunc)
(AVS_ScriptEnvironment *, AVS_Value args, void * user_data);
typedef struct AVS_FilterInfo AVS_FilterInfo;
struct AVS_FilterInfo
{
// these members should not be modified outside of the AVS_ApplyFunc callback
AVS_Clip * child;
AVS_VideoInfo vi;
AVS_ScriptEnvironment * env;
AVS_VideoFrame * (AVSC_CC * get_frame)(AVS_FilterInfo *, int n);
int (AVSC_CC * get_parity)(AVS_FilterInfo *, int n);
int (AVSC_CC * get_audio)(AVS_FilterInfo *, void * buf,
INT64 start, INT64 count);
int (AVSC_CC * set_cache_hints)(AVS_FilterInfo *, int cachehints,
int frame_range);
void (AVSC_CC * free_filter)(AVS_FilterInfo *);
// Should be set when ever there is an error to report.
// It is cleared before any of the above methods are called
const char * error;
// this is to store whatever and may be modified at will
void * user_data;
};
// Create a new filter
// fi is set to point to the AVS_FilterInfo so that you can
// modify it once it is initilized.
// store_child should generally be set to true. If it is not
// set than ALL methods (the function pointers) must be defined
// If it is set than you do not need to worry about freeing the child
// clip.
#if defined __cplusplus
extern "C"
{
#endif // __cplusplus
AVSC_API(AVS_Clip *, avs_new_c_filter)(AVS_ScriptEnvironment * e,
AVS_FilterInfo * * fi,
AVS_Value child, int store_child);
#if defined __cplusplus
}
#endif // __cplusplus
/////////////////////////////////////////////////////////////////////
//
// AVS_ScriptEnvironment
//
// For GetCPUFlags. These are backwards-compatible with those in VirtualDub.
enum {
/* slowest CPU to support extension */
AVS_CPU_FORCE = 0x01, // N/A
AVS_CPU_FPU = 0x02, // 386/486DX
AVS_CPU_MMX = 0x04, // P55C, K6, PII
AVS_CPU_INTEGER_SSE = 0x08, // PIII, Athlon
AVS_CPU_SSE = 0x10, // PIII, Athlon XP/MP
AVS_CPU_SSE2 = 0x20, // PIV, Hammer
AVS_CPU_3DNOW = 0x40, // K6-2
AVS_CPU_3DNOW_EXT = 0x80, // Athlon
AVS_CPU_X86_64 = 0xA0, // Hammer (note: equiv. to 3DNow + SSE2,
// which only Hammer will have anyway)
};
#if defined __cplusplus
extern "C"
{
#endif // __cplusplus
AVSC_API(const char *, avs_get_error)(AVS_ScriptEnvironment *); // return 0 if no error
AVSC_API(long, avs_get_cpu_flags)(AVS_ScriptEnvironment *);
AVSC_API(int, avs_check_version)(AVS_ScriptEnvironment *, int version);
AVSC_API(char *, avs_save_string)(AVS_ScriptEnvironment *, const char* s, int length);
AVSC_API(char *, avs_sprintf)(AVS_ScriptEnvironment *, const char * fmt, ...);
AVSC_API(char *, avs_vsprintf)(AVS_ScriptEnvironment *, const char * fmt, va_list val);
// note: val is really a va_list; I hope everyone typedefs va_list to a pointer
AVSC_API(int, avs_add_function)(AVS_ScriptEnvironment *,
const char * name, const char * params,
AVS_ApplyFunc apply, void * user_data);
AVSC_API(int, avs_function_exists)(AVS_ScriptEnvironment *, const char * name);
AVSC_API(AVS_Value, avs_invoke)(AVS_ScriptEnvironment *, const char * name,
AVS_Value args, const char** arg_names);
// The returned value must be be released with avs_release_value
AVSC_API(AVS_Value, avs_get_var)(AVS_ScriptEnvironment *, const char* name);
// The returned value must be be released with avs_release_value
AVSC_API(int, avs_set_var)(AVS_ScriptEnvironment *, const char* name, AVS_Value val);
AVSC_API(int, avs_set_global_var)(AVS_ScriptEnvironment *, const char* name, const AVS_Value val);
//void avs_push_context(AVS_ScriptEnvironment *, int level=0);
//void avs_pop_context(AVS_ScriptEnvironment *);
AVSC_API(AVS_VideoFrame *, avs_new_video_frame_a)(AVS_ScriptEnvironment *,
const AVS_VideoInfo * vi, int align);
// align should be at least 16
#if defined __cplusplus
}
#endif // __cplusplus
#ifndef AVSC_NO_DECLSPEC
AVSC_INLINE
AVS_VideoFrame * avs_new_video_frame(AVS_ScriptEnvironment * env,
const AVS_VideoInfo * vi)
{return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
AVSC_INLINE
AVS_VideoFrame * avs_new_frame(AVS_ScriptEnvironment * env,
const AVS_VideoInfo * vi)
{return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
#endif
#if defined __cplusplus
extern "C"
{
#endif // __cplusplus
AVSC_API(int, avs_make_writable)(AVS_ScriptEnvironment *, AVS_VideoFrame * * pvf);
AVSC_API(void, avs_bit_blt)(AVS_ScriptEnvironment *, unsigned char* dstp, int dst_pitch, const unsigned char* srcp, int src_pitch, int row_size, int height);
typedef void (AVSC_CC *AVS_ShutdownFunc)(void* user_data, AVS_ScriptEnvironment * env);
AVSC_API(void, avs_at_exit)(AVS_ScriptEnvironment *, AVS_ShutdownFunc function, void * user_data);
AVSC_API(AVS_VideoFrame *, avs_subframe)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height);
// The returned video frame must be be released
AVSC_API(int, avs_set_memory_max)(AVS_ScriptEnvironment *, int mem);
AVSC_API(int, avs_set_working_dir)(AVS_ScriptEnvironment *, const char * newdir);
// avisynth.dll exports this; it's a way to use it as a library, without
// writing an AVS script or without going through AVIFile.
AVSC_API(AVS_ScriptEnvironment *, avs_create_script_environment)(int version);
#if defined __cplusplus
}
#endif // __cplusplus
// this symbol is the entry point for the plugin and must
// be defined
AVSC_EXPORT
const char * AVSC_CC avisynth_c_plugin_init(AVS_ScriptEnvironment* env);
#if defined __cplusplus
extern "C"
{
#endif // __cplusplus
AVSC_API(void, avs_delete_script_environment)(AVS_ScriptEnvironment *);
AVSC_API(AVS_VideoFrame *, avs_subframe_planar)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height, int rel_offsetU, int rel_offsetV, int new_pitchUV);
// The returned video frame must be be released
#if defined __cplusplus
}
#endif // __cplusplus
#endif //__AVXSYNTH_C__

85
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/compat/avisynth/windowsPorts/basicDataTypeConversions.h Normal file
View file

@ -0,0 +1,85 @@
#ifndef __DATA_TYPE_CONVERSIONS_H__
#define __DATA_TYPE_CONVERSIONS_H__
#include <stdint.h>
#include <wchar.h>
#ifdef __cplusplus
namespace avxsynth {
#endif // __cplusplus
typedef int64_t __int64;
typedef int32_t __int32;
#ifdef __cplusplus
typedef bool BOOL;
#else
typedef uint32_t BOOL;
#endif // __cplusplus
typedef void* HMODULE;
typedef void* LPVOID;
typedef void* PVOID;
typedef PVOID HANDLE;
typedef HANDLE HWND;
typedef HANDLE HINSTANCE;
typedef void* HDC;
typedef void* HBITMAP;
typedef void* HICON;
typedef void* HFONT;
typedef void* HGDIOBJ;
typedef void* HBRUSH;
typedef void* HMMIO;
typedef void* HACMSTREAM;
typedef void* HACMDRIVER;
typedef void* HIC;
typedef void* HACMOBJ;
typedef HACMSTREAM* LPHACMSTREAM;
typedef void* HACMDRIVERID;
typedef void* LPHACMDRIVER;
typedef unsigned char BYTE;
typedef BYTE* LPBYTE;
typedef char TCHAR;
typedef TCHAR* LPTSTR;
typedef const TCHAR* LPCTSTR;
typedef char* LPSTR;
typedef LPSTR LPOLESTR;
typedef const char* LPCSTR;
typedef LPCSTR LPCOLESTR;
typedef wchar_t WCHAR;
typedef unsigned short WORD;
typedef unsigned int UINT;
typedef UINT MMRESULT;
typedef uint32_t DWORD;
typedef DWORD COLORREF;
typedef DWORD FOURCC;
typedef DWORD HRESULT;
typedef DWORD* LPDWORD;
typedef DWORD* DWORD_PTR;
typedef int32_t LONG;
typedef int32_t* LONG_PTR;
typedef LONG_PTR LRESULT;
typedef uint32_t ULONG;
typedef uint32_t* ULONG_PTR;
//typedef __int64_t intptr_t;
typedef uint64_t _fsize_t;
//
// Structures
//
typedef struct _GUID {
DWORD Data1;
WORD Data2;
WORD Data3;
BYTE Data4[8];
} GUID;
typedef GUID REFIID;
typedef GUID CLSID;
typedef CLSID* LPCLSID;
typedef GUID IID;
#ifdef __cplusplus
}; // namespace avxsynth
#endif // __cplusplus
#endif // __DATA_TYPE_CONVERSIONS_H__

77
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/compat/avisynth/windowsPorts/windows2linux.h Normal file
View file

@ -0,0 +1,77 @@
#ifndef __WINDOWS2LINUX_H__
#define __WINDOWS2LINUX_H__
/*
* LINUX SPECIFIC DEFINITIONS
*/
//
// Data types conversions
//
#include <stdlib.h>
#include <string.h>
#include "basicDataTypeConversions.h"
#ifdef __cplusplus
namespace avxsynth {
#endif // __cplusplus
//
// purposefully define the following MSFT definitions
// to mean nothing (as they do not mean anything on Linux)
//
#define __stdcall
#define __cdecl
#define noreturn
#define __declspec(x)
#define STDAPI extern "C" HRESULT
#define STDMETHODIMP HRESULT __stdcall
#define STDMETHODIMP_(x) x __stdcall
#define STDMETHOD(x) virtual HRESULT x
#define STDMETHOD_(a, x) virtual a x
#ifndef TRUE
#define TRUE true
#endif
#ifndef FALSE
#define FALSE false
#endif
#define S_OK (0x00000000)
#define S_FALSE (0x00000001)
#define E_NOINTERFACE (0X80004002)
#define E_POINTER (0x80004003)
#define E_FAIL (0x80004005)
#define E_OUTOFMEMORY (0x8007000E)
#define INVALID_HANDLE_VALUE ((HANDLE)((LONG_PTR)-1))
#define FAILED(hr) ((hr) & 0x80000000)
#define SUCCEEDED(hr) (!FAILED(hr))
//
// Functions
//
#define MAKEDWORD(a,b,c,d) (((a) << 24) | ((b) << 16) | ((c) << 8) | (d))
#define MAKEWORD(a,b) (((a) << 8) | (b))
#define lstrlen strlen
#define lstrcpy strcpy
#define lstrcmpi strcasecmp
#define _stricmp strcasecmp
#define InterlockedIncrement(x) __sync_fetch_and_add((x), 1)
#define InterlockedDecrement(x) __sync_fetch_and_sub((x), 1)
// Windows uses (new, old) ordering but GCC has (old, new)
#define InterlockedCompareExchange(x,y,z) __sync_val_compare_and_swap(x,z,y)
#define UInt32x32To64(a, b) ( (uint64_t) ( ((uint64_t)((uint32_t)(a))) * ((uint32_t)(b)) ) )
#define Int64ShrlMod32(a, b) ( (uint64_t) ( (uint64_t)(a) >> (b) ) )
#define Int32x32To64(a, b) ((__int64)(((__int64)((long)(a))) * ((long)(b))))
#define MulDiv(nNumber, nNumerator, nDenominator) (int32_t) (((int64_t) (nNumber) * (int64_t) (nNumerator) + (int64_t) ((nDenominator)/2)) / (int64_t) (nDenominator))
#ifdef __cplusplus
}; // namespace avxsynth
#endif // __cplusplus
#endif // __WINDOWS2LINUX_H__

6
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/compat/cuda/dynlink_loader.h → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/compat/cuda/dynlink_loader.h
View file

@ -16,8 +16,8 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef COMPAT_CUDA_DYNLINK_LOADER_H
#define COMPAT_CUDA_DYNLINK_LOADER_H
#ifndef AV_COMPAT_CUDA_DYNLINK_LOADER_H
#define AV_COMPAT_CUDA_DYNLINK_LOADER_H
#include "libavutil/log.h"
#include "compat/w32dlfcn.h"
@ -30,4 +30,4 @@
#include <ffnvcodec/dynlink_loader.h>
#endif /* COMPAT_CUDA_DYNLINK_LOADER_H */
#endif

36
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/compat/cuda/ptx2c.sh Executable file
View file

@ -0,0 +1,36 @@
#!/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

0
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/compat/dispatch_semaphore/semaphore.h → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/compat/dispatch_semaphore/semaphore.h
View file

0
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/compat/float/float.h → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/compat/float/float.h
View file

0
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/compat/float/limits.h → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/compat/float/limits.h
View file

0
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/compat/getopt.c → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/compat/getopt.c
View file

2
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/compat/msvcrt/snprintf.c → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/compat/msvcrt/snprintf.c
View file

@ -59,7 +59,7 @@ int avpriv_vsnprintf(char *s, size_t n, const char *fmt,
* recommends to provide _snprintf/_vsnprintf() a buffer size that
* is one less than the actual buffer, and zero it before calling
* _snprintf/_vsnprintf() to workaround this problem.
* See https://web.archive.org/web/20151214111935/http://msdn.microsoft.com/en-us/library/1kt27hek(v=vs.80).aspx */
* See http://msdn.microsoft.com/en-us/library/1kt27hek(v=vs.80).aspx */
memset(s, 0, n);
va_copy(ap_copy, ap);
ret = _vsnprintf(s, n - 1, fmt, ap_copy);

0
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/compat/msvcrt/snprintf.h → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/compat/msvcrt/snprintf.h
View file

26
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/compat/os2threads.h → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/compat/os2threads.h
View file

@ -27,19 +27,15 @@
#define COMPAT_OS2THREADS_H
#define INCL_DOS
#define INCL_DOSERRORS
#include <os2.h>
#undef __STRICT_ANSI__ /* for _beginthread() */
#include <stdlib.h>
#include <time.h>
#include <sys/builtin.h>
#include <sys/fmutex.h>
#include "libavutil/attributes.h"
#include "libavutil/common.h"
#include "libavutil/time.h"
typedef struct {
TID tid;
@ -167,28 +163,6 @@ static av_always_inline int pthread_cond_broadcast(pthread_cond_t *cond)
return 0;
}
static av_always_inline int pthread_cond_timedwait(pthread_cond_t *cond,
pthread_mutex_t *mutex,
const struct timespec *abstime)
{
int64_t abs_milli = abstime->tv_sec * 1000LL + abstime->tv_nsec / 1000000;
ULONG t = av_clip64(abs_milli - av_gettime() / 1000, 0, ULONG_MAX);
__atomic_increment(&cond->wait_count);
pthread_mutex_unlock(mutex);
APIRET ret = DosWaitEventSem(cond->event_sem, t);
__atomic_decrement(&cond->wait_count);
DosPostEventSem(cond->ack_sem);
pthread_mutex_lock(mutex);
return (ret == ERROR_TIMEOUT) ? ETIMEDOUT : 0;
}
static av_always_inline int pthread_cond_wait(pthread_cond_t *cond,
pthread_mutex_t *mutex)
{

0
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/compat/solaris/make_sunver.pl → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/compat/solaris/make_sunver.pl
View file

0
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/compat/strtod.c → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/compat/strtod.c
View file

0
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/compat/va_copy.h → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/compat/va_copy.h
View file

94
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/compat/w32dlfcn.h Normal file
View file

@ -0,0 +1,94 @@
/*
* 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 */

26
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/compat/w32pthreads.h → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/compat/w32pthreads.h
View file

@ -38,13 +38,11 @@
#define WIN32_LEAN_AND_MEAN
#include <windows.h>
#include <process.h>
#include <time.h>
#include "libavutil/attributes.h"
#include "libavutil/common.h"
#include "libavutil/internal.h"
#include "libavutil/mem.h"
#include "libavutil/time.h"
typedef struct pthread_t {
void *handle;
@ -63,9 +61,6 @@ typedef CONDITION_VARIABLE pthread_cond_t;
#define InitializeCriticalSection(x) InitializeCriticalSectionEx(x, 0, 0)
#define WaitForSingleObject(a, b) WaitForSingleObjectEx(a, b, FALSE)
#define PTHREAD_CANCEL_ENABLE 1
#define PTHREAD_CANCEL_DISABLE 0
static av_unused unsigned __stdcall attribute_align_arg win32thread_worker(void *arg)
{
pthread_t *h = (pthread_t*)arg;
@ -161,31 +156,10 @@ static inline int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex
return 0;
}
static inline int pthread_cond_timedwait(pthread_cond_t *cond, pthread_mutex_t *mutex,
const struct timespec *abstime)
{
int64_t abs_milli = abstime->tv_sec * 1000LL + abstime->tv_nsec / 1000000;
DWORD t = av_clip64(abs_milli - av_gettime() / 1000, 0, UINT32_MAX);
if (!SleepConditionVariableSRW(cond, mutex, t, 0)) {
DWORD err = GetLastError();
if (err == ERROR_TIMEOUT)
return ETIMEDOUT;
else
return EINVAL;
}
return 0;
}
static inline int pthread_cond_signal(pthread_cond_t *cond)
{
WakeConditionVariable(cond);
return 0;
}
static inline int pthread_setcancelstate(int state, int *oldstate)
{
return 0;
}
#endif /* COMPAT_W32PTHREADS_H */

4
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/compat/windows/makedef → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/compat/windows/makedef
View file

@ -48,7 +48,7 @@ trap 'rm -f -- $libname' EXIT
if [ -n "$AR" ]; then
$AR rcs ${libname} $@ >/dev/null
else
lib.exe -out:${libname} $@ >/dev/null
lib -out:${libname} $@ >/dev/null
fi
if [ $? != 0 ]; then
echo "Could not create temporary library." >&2
@ -108,7 +108,7 @@ if [ -n "$NM" ]; then
cut -d' ' -f3 |
sed -e "s/^${prefix}//")
else
dump=$(dumpbin.exe -linkermember:1 ${libname} |
dump=$(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)

2
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/compat/windows/mslink → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/compat/windows/mslink
View file

@ -4,6 +4,6 @@ LINK_EXE_PATH=$(dirname "$(command -v cl)")/link
if [ -x "$LINK_EXE_PATH" ]; then
"$LINK_EXE_PATH" $@
else
link.exe $@
link $@
fi
exit $?

1738
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/configure → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/configure vendored
View file

File diff suppressed because it is too large Load diff

703
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/doc/APIchanges → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/APIchanges
View file

@ -1,693 +1,20 @@
The last version increases of all libraries were on 2023-02-09
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
API changes, most recent first:
-------- 8< --------- FFmpeg 6.0 was cut here -------- 8< ---------
2023-02-16 - 927042b409 - lavf 60.2.100 - avformat.h
Deprecate AVFormatContext io_close callback.
The superior io_close2 callback should be used instead.
2023-02-13 - 2296078397 - lavu 58.1.100 - frame.h
Deprecate AVFrame.coded_picture_number and display_picture_number.
Their usefulness is questionable and very few decoders set them.
2023-02-13 - 6b6f7db819 - lavc 60.2.100 - avcodec.h
Add AVCodecContext.frame_num as a 64bit version of frame_number.
Deprecate AVCodecContext.frame_number.
2023-02-12 - d1b9a3ddb4 - lavfi 9.1.100 - avfilter.h
Add filtergraph segment parsing API.
New structs:
- AVFilterGraphSegment
- AVFilterChain
- AVFilterParams
- AVFilterPadParams
New functions:
- avfilter_graph_segment_parse()
- avfilter_graph_segment_create_filters()
- avfilter_graph_segment_apply_opts()
- avfilter_graph_segment_init()
- avfilter_graph_segment_link()
- avfilter_graph_segment_apply()
2023-02-09 - 719a93f4e4 - lavu 58.0.100 - csp.h
Add av_csp_approximate_trc_gamma() and av_csp_trc_func_from_id().
Add av_csp_trc_function.
2023-02-09 - 868a31b42d - lavc 60.0.100 - avcodec.h
avcodec_decode_subtitle2() now accepts const AVPacket*.
2023-02-04 - d02340b9e3 - lavc 59.63.100
Allow AV_CODEC_FLAG_COPY_OPAQUE to be used with decoders.
2023-01-29 - a1a80f2e64 - lavc 59.59.100 - avcodec.h
Add AV_CODEC_FLAG_COPY_OPAQUE and AV_CODEC_FLAG_FRAME_DURATION.
2023-01-13 - 002d0ec740 - lavu 57.44.100 - ambient_viewing_environment.h frame.h
Adds a new structure for holding H.274 Ambient Viewing Environment metadata,
AVAmbientViewingEnvironment.
Adds a new AVFrameSideDataType entry AV_FRAME_DATA_AMBIENT_VIEWING_ENVIRONMENT
for it.
2022-12-10 - 7a8d78f7e3 - lavc 59.55.100 - avcodec.h
Add AV_HWACCEL_FLAG_UNSAFE_OUTPUT.
2022-11-24 - e97368eba5 - lavu 57.43.100 - tx.h
Add AV_TX_FLOAT_DCT, AV_TX_DOUBLE_DCT and AV_TX_INT32_DCT.
2022-11-06 - 9dad237928 - lavu 57.42.100 - dict.h
Add av_dict_iterate().
2022-11-03 - 6228ba141d - lavu 57.41.100 - channel_layout.h
Add AV_CH_LAYOUT_7POINT1_TOP_BACK and AV_CHANNEL_LAYOUT_7POINT1_TOP_BACK.
2022-10-30 - 83e918de71 - lavu 57.40.100 - channel_layout.h
Add AV_CH_LAYOUT_CUBE and AV_CHANNEL_LAYOUT_CUBE.
2022-10-11 - 479747645f - lavu 57.39.101 - pixfmt.h
Add AV_PIX_FMT_RGBF32 and AV_PIX_FMT_RGBAF32.
2022-10-05 - 37d5ddc317 - lavu 57.39.100 - cpu.h
Add AV_CPU_FLAG_RVB_BASIC.
2022-10-03 - d09776d486 - lavf 59.34.100 - avio.h
Make AVIODirContext an opaque type in a future major version bump.
2022-09-27 - 0c0a3deb18 - lavu 57.38.100 - cpu.h
Add CPU flags for RISC-V vector extensions:
AV_CPU_FLAG_RVV_I32, AV_CPU_FLAG_RVV_F32, AV_CPU_FLAG_RVV_I64,
AV_CPU_FLAG_RVV_F64
2022-09-26 - a02a0e8db4 - lavc 59.48.100 - avcodec.h
Deprecate avcodec_enum_to_chroma_pos() and avcodec_chroma_pos_to_enum().
Use av_chroma_location_enum_to_pos() or av_chroma_location_pos_to_enum()
instead.
2022-09-26 - xxxxxxxxxx - lavu 57.37.100 - pixdesc.h pixfmt.h
Add av_chroma_location_enum_to_pos() and av_chroma_location_pos_to_enum().
Add AV_PIX_FMT_RGBF32BE, AV_PIX_FMT_RGBF32LE, AV_PIX_FMT_RGBAF32BE,
AV_PIX_FMT_RGBAF32LE.
2022-09-26 - cf856d8957 - lavc 59.47.100 - avcodec.h defs.h
Move the AV_EF_* and FF_COMPLIANCE_* defines from avcodec.h to defs.h.
2022-09-03 - d75c4693fe - lavu 57.36.100 - pixfmt.h
Add AV_PIX_FMT_P012, AV_PIX_FMT_Y212, AV_PIX_FMT_XV30, AV_PIX_FMT_XV36
2022-09-03 - dea9744560 - lavu 57.35.100 - file.h
Deprecate av_tempfile() without replacement.
2022-08-03 - cc5a5c9860 - lavu 57.34.100 - pixfmt.h
Add AV_PIX_FMT_VUYX.
2022-08-22 - 14726571dd - lavf 59 - avformat.h
Deprecate av_stream_get_end_pts() without replacement.
2022-08-19 - 352799dca8 - lavc 59.42.102 - codec_id.h
Deprecate AV_CODEC_ID_AYUV and ayuv decoder/encoder. The rawvideo codec
and vuya pixel format combination will be used instead from now on.
2022-08-07 - e95b08a7dd - lavu 57.33.101 - pixfmt.h
Add AV_PIX_FMT_RGBAF16{BE,LE} pixel formats.
2022-08-12 - e0bbdbe0a6 - lavu 57.33.100 - hwcontext_qsv.h
Add loader field to AVQSVDeviceContext
2022-08-03 - 6ab8a9d375 - lavu 57.32.100 - pixfmt.h
Add AV_PIX_FMT_VUYA.
2022-08-02 - e3838b856f - lavc 59.41.100 - avcodec.h codec.h
Add AV_CODEC_FLAG_RECON_FRAME and AV_CODEC_CAP_ENCODER_RECON_FRAME.
avcodec_receive_frame() may now be used on encoders when
AV_CODEC_FLAG_RECON_FRAME is active.
2022-08-02 - eede1d2927 - lavu 57.31.100 - frame.h
av_frame_make_writable() may now be called on non-refcounted
frames and will make a refcounted copy out of them.
Previously an error was returned in such cases.
2022-07-30 - e1a0f2df3d - lavc 59.40.100 - avcodec.h
Add the AV_CODEC_FLAG2_ICC_PROFILES flag to AVCodecContext, to enable
automatic reading and writing of embedded ICC profiles in image files.
The "flags2" option now supports the corresponding flag "icc_profiles".
2022-07-19 - 4397f9a5a0 - lavu 57.30.100 - frame.h
Add AVFrame.duration, deprecate AVFrame.pkt_duration.
-------- 8< --------- FFmpeg 5.1 was cut here -------- 8< ---------
2022-06-12 - 7cae3d8b76 - lavf 59.25.100 - avio.h
Add avio_vprintf(), similar to avio_printf() but allow to use it
from within a function taking a variable argument list as input.
2022-06-12 - ff59ecc4de - lavu 57.27.100 - uuid.h
Add UUID handling functions.
Add av_uuid_parse(), av_uuid_urn_parse(), av_uuid_parse_range(),
av_uuid_parse_range(), av_uuid_equal(), av_uuid_copy(), and av_uuid_nil().
2022-06-01 - d42b410e05 - lavu 57.26.100 - csp.h
Add public API for colorspace structs.
Add av_csp_luma_coeffs_from_avcsp(), av_csp_primaries_desc_from_id(),
and av_csp_primaries_id_from_desc().
2022-05-23 - 4cdc14aa95 - lavu 57.25.100 - avutil.h
Deprecate av_fopen_utf8() without replacement.
2022-03-16 - f3a0e2ee2b - all libraries - version_major.h
Add lib<name>/version_major.h as new installed headers, which only
contain the major version number (and corresponding API deprecation
defines).
2022-03-15 - cdba98bb80 - swr 4.5.100 - swresample.h
Add swr_alloc_set_opts2() and swr_build_matrix2().
Deprecate swr_alloc_set_opts() and swr_build_matrix().
2022-03-15 - cdba98bb80 - lavfi 8.28.100 - avfilter.h buffersink.h buffersrc.h
Update AVFilterLink for the new channel layout API: add ch_layout,
deprecate channel_layout.
Update the buffersink filter sink for the new channel layout API:
add av_buffersink_get_ch_layout() and the ch_layouts option,
deprecate av_buffersink_get_channel_layout() and the channel_layouts option.
Update AVBufferSrcParameters for the new channel layout API:
add ch_layout, deprecate channel_layout.
2022-03-15 - cdba98bb80 - lavf 59.19.100 - avformat.h
Add AV_DISPOSITION_NON_DIEGETIC.
2022-03-15 - cdba98bb80 - lavc 59.24.100 - avcodec.h codec_par.h
Update AVCodecParameters for the new channel layout API: add ch_layout,
deprecate channels/channel_layout.
Update AVCodecContext for the new channel layout API: add ch_layout,
deprecate channels/channel_layout.
Update AVCodec for the new channel layout API: add ch_layouts,
deprecate channel_layouts.
2022-03-15 - cdba98bb80 - lavu 57.24.100 - channel_layout.h frame.h opt.h
Add new channel layout API based on the AVChannelLayout struct.
Add support for Ambisonic audio.
Deprecate previous channel layout API based on uint64 bitmasks.
Add AV_OPT_TYPE_CHLAYOUT option type, deprecate AV_OPT_TYPE_CHANNEL_LAYOUT.
Update AVFrame for the new channel layout API: add ch_layout, deprecate
channels/channel_layout.
2022-03-10 - f629ea2e18 - lavu 57.23.100 - cpu.h
Add AV_CPU_FLAG_AVX512ICL.
2022-02-07 - a10f1aec1f - lavu 57.21.100 - fifo.h
Deprecate AVFifoBuffer and the API around it, namely av_fifo_alloc(),
av_fifo_alloc_array(), av_fifo_free(), av_fifo_freep(), av_fifo_reset(),
av_fifo_size(), av_fifo_space(), av_fifo_generic_peek_at(),
av_fifo_generic_peek(), av_fifo_generic_read(), av_fifo_generic_write(),
av_fifo_realloc2(), av_fifo_grow(), av_fifo_drain() and av_fifo_peek2().
Users should switch to the AVFifo-API.
2022-02-07 - 7329b22c05 - lavu 57.20.100 - fifo.h
Add a new FIFO API, which allows setting a FIFO element size.
This API operates on these elements rather than on bytes.
Add av_fifo_alloc2(), av_fifo_elem_size(), av_fifo_can_read(),
av_fifo_can_write(), av_fifo_grow2(), av_fifo_drain2(), av_fifo_write(),
av_fifo_write_from_cb(), av_fifo_read(), av_fifo_read_to_cb(),
av_fifo_peek(), av_fifo_peek_to_cb(), av_fifo_drain2(), av_fifo_reset2(),
av_fifo_freep2(), av_fifo_auto_grow_limit().
2022-01-26 - af94ab7c7c0 - lavu 57.19.100 - tx.h
Add AV_TX_FLOAT_RDFT, AV_TX_DOUBLE_RDFT and AV_TX_INT32_RDFT.
-------- 8< --------- FFmpeg 5.0 was cut here -------- 8< ---------
2022-01-04 - 78dc21b123e - lavu 57.16.100 - frame.h
Add AV_FRAME_DATA_DOVI_METADATA.
2022-01-03 - 70f318e6b6c - lavf 59.13.100 - avformat.h
Add AVFMT_EXPERIMENTAL flag.
2021-12-22 - b7e1ec7bda9 - lavu 57.13.100 - hwcontext_videotoolbox.h
Add av_vt_pixbuf_set_attachments
2021-12-22 - 69bd95dcd8d - lavu 57.13.100 - hwcontext_videotoolbox.h
Add av_map_videotoolbox_chroma_loc_from_av
Add av_map_videotoolbox_color_matrix_from_av
Add av_map_videotoolbox_color_primaries_from_av
Add av_map_videotoolbox_color_trc_from_av
2021-12-21 - ffbab99f2c2 - lavu 57.12.100 - cpu.h
Add AV_CPU_FLAG_SLOW_GATHER.
2021-12-20 - 278068dc60d - lavu 57.11.101 - display.h
Modified the documentation of av_display_rotation_set()
to match its longstanding actual behaviour of treating
the angle as directed clockwise.
2021-12-12 - 64834bb86a1 - lavf 59.10.100 - avformat.h
Add AVFormatContext io_close2 which returns an int
2021-12-10 - f45cbb775e4 - lavu 57.11.100 - hwcontext_vulkan.h
Add AVVkFrame.offset and AVVulkanFramesContext.flags.
2021-12-04 - b9c928a486f - lavfi 8.19.100 - avfilter.h
Add AVFILTER_FLAG_METADATA_ONLY.
2021-12-03 - b236ef0a594 - lavu 57.10.100 - frame.h
Add AVFrame.time_base
2021-11-22 - b2cd1fb2ec6 - lavu 57.9.100 - pixfmt.h
Add AV_PIX_FMT_P210, AV_PIX_FMT_P410, AV_PIX_FMT_P216, and AV_PIX_FMT_P416.
2021-11-17 - 54e65aa38ab - lavf 57.9.100 - frame.h
Add AV_FRAME_DATA_DOVI_RPU_BUFFER.
2021-11-16 - ed75a08d36c - lavf 59.9.100 - avformat.h
Add av_stream_get_class(). Schedule adding AVStream.av_class at libavformat
major version 60.
Add av_disposition_to_string() and av_disposition_from_string().
Add "disposition" AVOption to AVStream's class.
2021-11-12 - 8478d60d5b5 - lavu 57.8.100 - hwcontext_vulkan.h
Added AVVkFrame.sem_value, AVVulkanDeviceContext.queue_family_encode_index,
nb_encode_queues, queue_family_decode_index, and nb_decode_queues.
2021-10-18 - 682bafdb125 - lavf 59.8.100 - avio.h
Introduce public bytes_{read,written} statistic fields to AVIOContext.
2021-10-13 - a5622ed16f8 - lavf 59.7.100 - avio.h
Deprecate AVIOContext.written. Originally added as a private entry in
commit 3f75e5116b900f1428aa13041fc7d6301bf1988a, its grouping with
the comment noting its private state was missed during merging of the field
from Libav (most likely due to an already existing field in between).
2021-09-21 - 0760d9153c3 - lavu 57.7.100 - pixfmt.h
Add AV_PIX_FMT_X2BGR10.
2021-09-20 - 8d5de914d31 - lavu 57.6.100 - mem.h
Deprecate av_mallocz_array() as it is identical to av_calloc().
2021-09-20 - 176b8d785bf - lavc 59.9.100 - avcodec.h
Deprecate AVCodecContext.sub_text_format and the corresponding
AVOptions. It is unused since the last major bump.
2021-09-20 - dd846bc4a91 - lavc 59.8.100 - avcodec.h codec.h
Deprecate AV_CODEC_FLAG_TRUNCATED and AV_CODEC_CAP_TRUNCATED,
as they are redundant with parsers.
2021-09-17 - ccfdef79b13 - lavu 57.5.101 - buffer.h
Constified the input parameters in av_buffer_replace(), av_buffer_ref(),
and av_buffer_pool_buffer_get_opaque().
2021-09-08 - 4f78711f9c2 - lavu 57.5.100 - hwcontext_d3d11va.h
Add AVD3D11VAFramesContext.texture_infos
2021-09-06 - 42cd64c1826 - lsws 6.1.100 - swscale.h
Add AVFrame-based scaling API:
- sws_scale_frame()
- sws_frame_start()
- sws_frame_end()
- sws_send_slice()
- sws_receive_slice()
- sws_receive_slice_alignment()
2021-09-02 - cbf111059d2 - lavc 59.7.100 - avcodec.h
Incremented the number of elements of AVCodecParser.codec_ids to seven.
2021-08-24 - 590a7e02f04 - lavc 59.6.100 - avcodec.h
Add FF_CODEC_PROPERTY_FILM_GRAIN
2021-08-20 - 7c5f998196d - lavfi 8.3.100 - avfilter.H
Add avfilter_filter_pad_count() as a replacement for avfilter_pad_count().
Deprecate avfilter_pad_count().
2021-08-17 - 8c53b145993 - lavu 57.4.101 - opt.h
av_opt_copy() now guarantees that allocated src and dst options
don't alias each other even on error.
2021-08-14 - d5de9965ef6 - lavu 57.4.100 - imgutils.h
Add av_image_copy_plane_uc_from()
2021-08-02 - a1a0fddfd05 - lavc 59.4.100 - packet.h
Add AVPacket.opaque, AVPacket.opaque_ref, AVPacket.time_base.
2021-07-23 - 2dd8acbe800 - lavu 57.3.100 - common.h macros.h
Move several macros (AV_NE, FFDIFFSIGN, FFMAX, FFMAX3, FFMIN, FFMIN3,
FFSWAP, FF_ARRAY_ELEMS, MKTAG, MKBETAG) from common.h to macros.h.
2021-07-22 - e3b5ff17c2e - lavu 57.2.100 - film_grain_params.h
Add AV_FILM_GRAIN_PARAMS_H274, AVFilmGrainH274Params
2021-07-19 - c1bf56a526f - lavu 57.1.100 - cpu.h
Add av_cpu_force_count()
2021-06-17 - aca923b3653 - lavc 59.2.100 - packet.h
Add AV_PKT_DATA_DYNAMIC_HDR10_PLUS
2021-06-09 - 2cccab96f6f - lavf 59.3.100 - avformat.h
Add pts_wrap_bits to AVStream
2021-06-10 - 7c9763070d9 - lavc 59.1.100 - avcodec.h codec.h
Move av_get_profile_name() from avcodec.h to codec.h.
2021-06-10 - bb3648e6766 - lavc 59.1.100 - avcodec.h codec_par.h
Move av_get_audio_frame_duration2() from avcodec.h to codec_par.h.
2021-06-10 - 881db34f6a0 - lavc 59.1.100 - avcodec.h codec_id.h
Move av_get_bits_per_sample(), av_get_exact_bits_per_sample(),
avcodec_profile_name(), and av_get_pcm_codec() from avcodec.h
to codec_id.h.
2021-06-10 - ff0a96046d8 - lavc 59.1.100 - avcodec.h defs.h
Add new installed header defs.h. The following definitions are moved
into it from avcodec.h:
- AVDiscard
- AVAudioServiceType
- AVPanScan
- AVCPBProperties and av_cpb_properties_alloc()
- AVProducerReferenceTime
- av_xiphlacing()
2021-04-27 - cb3ac722f4 - lavc 59.0.100 - avcodec.h
Constified AVCodecParserContext.parser.
2021-04-27 - 8b3e6ce5f4 - lavd 59.0.100 - avdevice.h
The av_*_device_next API functions now accept and return
pointers to const AVInputFormat resp. AVOutputFormat.
2021-04-27 - d7e0d428fa - lavd 59.0.100 - avdevice.h
avdevice_list_input_sources and avdevice_list_output_sinks now accept
pointers to const AVInputFormat resp. const AVOutputFormat.
2021-04-27 - 46dac8cf3d - lavf 59.0.100 - avformat.h
av_find_best_stream now uses a const AVCodec ** parameter
for the returned decoder.
2021-04-27 - 626535f6a1 - lavc 59.0.100 - codec.h
avcodec_find_encoder_by_name(), avcodec_find_encoder(),
avcodec_find_decoder_by_name() and avcodec_find_decoder()
now return a pointer to const AVCodec.
2021-04-27 - 14fa0a4efb - lavf 59.0.100 - avformat.h
Constified AVFormatContext.*_codec.
2021-04-27 - 56450a0ee4 - lavf 59.0.100 - avformat.h
Constified the pointers to AVInputFormats and AVOutputFormats
in AVFormatContext, avformat_alloc_output_context2(),
av_find_input_format(), av_probe_input_format(),
av_probe_input_format2(), av_probe_input_format3(),
av_probe_input_buffer2(), av_probe_input_buffer(),
avformat_open_input(), av_guess_format() and av_guess_codec().
Furthermore, constified the AVProbeData in av_probe_input_format(),
av_probe_input_format2() and av_probe_input_format3().
2021-04-19 - 18af1ea8d1 - lavu 56.74.100 - tx.h
Add AV_TX_FULL_IMDCT and AV_TX_UNALIGNED.
2021-04-17 - f1bf465aa0 - lavu 56.73.100 - frame.h detection_bbox.h
Add AV_FRAME_DATA_DETECTION_BBOXES
2021-04-06 - 557953a397 - lavf 58.78.100 - avformat.h
Add avformat_index_get_entries_count(), avformat_index_get_entry(),
and avformat_index_get_entry_from_timestamp().
2021-03-21 - a77beea6c8 - lavu 56.72.100 - frame.h
Deprecated av_get_colorspace_name().
Use av_color_space_name() instead.
-------- 8< --------- FFmpeg 4.4 was cut here -------- 8< ---------
2021-03-19 - e8c0bca6bd - lavu 56.69.100 - adler32.h
Added a typedef for the type of the Adler-32 checksums
used by av_adler32_update(). It will be changed to uint32_t
at the next major bump.
The type of the parameter for the length of the input buffer
will also be changed to size_t at the next major bump.
2021-03-19 - e318438f2f - lavf 58.75.100 - avformat.h
AVChapter.id will be changed from int to int64_t
on the next major version bump.
2021-03-17 - f7db77bd87 - lavc 58.133.100 - codec.h
Deprecated av_init_packet(). Once removed, sizeof(AVPacket) will
no longer be a part of the public ABI.
Deprecated AVPacketList.
2021-03-16 - 7d09579190 - lavc 58.132.100 - codec.h
Add AV_CODEC_CAP_OTHER_THREADS as a new name for
AV_CODEC_CAP_AUTO_THREADS. AV_CODEC_CAP_AUTO_THREADS
is now deprecated.
2021-03-12 - 6e7e3a3820 - lavc 58.131.100 - avcodec.h codec.h
Add a get_encode_buffer callback to AVCodecContext, similar to
get_buffer2 but for encoders.
Add avcodec_default_get_encode_buffer().
Add AV_GET_ENCODE_BUFFER_FLAG_REF.
Encoders may now be flagged as AV_CODEC_CAP_DR1 capable.
2021-03-10 - 42e68fe015 - lavf 58.72.100 - avformat.h
Change AVBufferRef related AVStream function and struct size
parameter and fields type to size_t at next major bump.
2021-03-10 - d79e0fe65c - lavc 58.130.100 - packet.h
Change AVBufferRef related AVPacket function and struct size
parameter and fields type to size_t at next major bump.
2021-03-10 - 14040a1d91 - lavu 56.68.100 - buffer.h frame.h
Change AVBufferRef and relevant AVFrame function and struct size
parameter and fields type to size_t at next major bump.
2021-03-04 - a0eec776b6 - lavc 58.128.101 - avcodec.h
Enable err_recognition to be set for encoders.
2021-03-03 - 2ff40b98ec - lavf 58.70.100 - avformat.h
Deprecate AVFMT_FLAG_PRIV_OPT. It will do nothing
as soon as av_demuxer_open() is removed.
2021-02-27 - dd9227e48f - lavc 58.126.100 - avcodec.h
Deprecated avcodec_get_frame_class().
2021-02-21 - 5ca40d6d94 - lavu 56.66.100 - tx.h
Add enum AVTXFlags and AVTXFlags.AV_TX_INPLACE
2021-02-14 - 4f49ca7bbc - lavd 58.12.100 - avdevice.h
Deprecated avdevice_capabilities_create() and
avdevice_capabilities_free().
2021-02-10 - 1bda9bb68a - lavu 56.65.100 - common.h
Add FFABS64U()
2021-01-26 - 5dd9567080 - lavu 56.64.100 - common.h
Add FFABSU()
2021-01-25 - 56709ca8aa - lavc 58.119.100 - avcodec.h
Deprecate AVCodecContext.debug_mv, FF_DEBUG_VIS_MV_P_FOR, FF_DEBUG_VIS_MV_B_FOR,
FF_DEBUG_VIS_MV_B_BACK
2021-01-11 - ebdd33086a - lavc 58.116.100 - avcodec.h
Add FF_PROFILE_VVC_MAIN_10 and FF_PROFILE_VVC_MAIN_10_444.
2020-01-01 - baecaa16c1 - lavu 56.63.100 - video_enc_params.h
Add AV_VIDEO_ENC_PARAMS_MPEG2
2020-12-03 - eca12f4d5a - lavu 56.62.100 - timecode.h
Add av_timecode_init_from_components.
2020-11-27 - a83098ab03 - lavc 58.114.100 - avcodec.h
Deprecate AVCodecContext.thread_safe_callbacks. Starting with
LIBAVCODEC_VERSION_MAJOR=60, user callbacks must always be
thread-safe when frame threading is used.
2020-11-25 - d243dd540a - lavc 58.113.100 - avcodec.h
Adds a new flag AV_CODEC_EXPORT_DATA_FILM_GRAIN for export_side_data.
2020-11-25 - 4f9ee87253 - lavu 56.61.100 - film_grain_params.h
Adds a new API for extracting codec film grain parameters as side data.
Adds a new AVFrameSideDataType entry AV_FRAME_DATA_FILM_GRAIN_PARAMS for it.
2020-10-28 - f95d9510ff - lavf 58.64.100 - avformat.h
Add AVSTREAM_EVENT_FLAG_NEW_PACKETS.
2020-09-28 - 68918d3b7f - lavu 56.60.100 - buffer.h
Add a av_buffer_replace() convenience function.
2020-09-13 - 837b6eb90e - lavu 56.59.100 - timecode.h
Add av_timecode_make_smpte_tc_string2.
2020-08-21 - 06f2651204 - lavu 56.58.100 - avstring.h
Deprecate av_d2str(). Use av_asprintf() instead.
2020-08-04 - 34de0abbe7 - lavu 56.58.100 - channel_layout.h
Add AV_CH_LAYOUT_22POINT2 together with its newly required pieces:
AV_CH_TOP_SIDE_LEFT, AV_CH_TOP_SIDE_RIGHT, AV_CH_BOTTOM_FRONT_CENTER,
AV_CH_BOTTOM_FRONT_LEFT, AV_CH_BOTTOM_FRONT_RIGHT.
2020-07-23 - 84655b7101 - lavu 56.57.100 - cpu.h
Add AV_CPU_FLAG_MMI and AV_CPU_FLAG_MSA.
2020-07-22 - 3a8e927176 - lavu 56.56.100 - imgutils.h
Add av_image_fill_plane_sizes().
2020-07-15 - 448a9aaa78 - lavc 58.96.100 - packet.h
Add AV_PKT_DATA_S12M_TIMECODE.
2020-06-12 - b09fb030c1 - lavu 56.55.100 - pixdesc.h
Add AV_PIX_FMT_X2RGB10.
2020-06-11 - bc8ab084fb - lavu 56.54.100 - frame.h
Add AV_FRAME_DATA_SEI_UNREGISTERED.
2020-06-10 - 1b4a98b029 - lavu 56.53.100 - log.h opt.h
Add av_opt_child_class_iterate() and AVClass.child_class_iterate().
Deprecate av_opt_child_class_next() and AVClass.child_class_next().
-------- 8< --------- FFmpeg 4.3 was cut here -------- 8< ---------
2020-06-05 - ec39c2276a - lavu 56.50.100 - buffer.h
Passing NULL as alloc argument to av_buffer_pool_init2() is now allowed.
2020-05-27 - ba6cada92e - lavc 58.88.100 - avcodec.h codec.h
Move AVCodec-related public API to new header codec.h.
2020-05-23 - 064b875e89 - lavu 56.49.100 - video_enc_params.h
Add AV_VIDEO_ENC_PARAMS_H264.
2020-05-23 - 2e08b39444 - lavu 56.48.100 - hwcontext.h
Add av_hwdevice_ctx_create_derived_opts.
2020-05-23 - 6b65c4ec54 - lavu 56.47.100 - rational.h
Add av_gcd_q().
2020-05-22 - af9e622776 - lavu 56.46.101 - opt.h
Add AV_OPT_FLAG_CHILD_CONSTS.
2020-05-22 - 9d443c3e68 - lavc 58.87.100 - avcodec.h codec_par.h
Move AVBitstreamFilter-related public API to new header bsf.h.
Move AVCodecParameters-related public API to new header codec_par.h.
2020-05-21 - 13b1bbff0b - lavc 58.86.101 - avcodec.h
Deprecated AV_CODEC_CAP_INTRA_ONLY and AV_CODEC_CAP_LOSSLESS.
2020-05-17 - 84af196c65 - lavu 56.46.100 - common.h
Add av_sat_add64() and av_sat_sub64()
2020-05-12 - 991d417692 - lavu 56.45.100 - video_enc_params.h
lavc 58.84.100 - avcodec.h
Add a new API for exporting video encoding information.
Replaces the deprecated API for exporting QP tables from decoders.
Add AV_CODEC_EXPORT_DATA_VIDEO_ENC_PARAMS to request this information from
decoders.
2020-05-10 - dccd07f66d - lavu 56.44.100 - hwcontext_vulkan.h
Add enabled_inst_extensions, num_enabled_inst_extensions, enabled_dev_extensions
and num_enabled_dev_extensions fields to AVVulkanDeviceContext
2020-04-22 - 0e1db79e37 - lavc 58.81.100 - packet.h
- lavu 56.43.100 - dovi_meta.h
Add AV_PKT_DATA_DOVI_CONF and AVDOVIDecoderConfigurationRecord.
2020-04-15 - 22b25b3ea5 - lavc 58.79.100 - avcodec.h
Add formal support for calling avcodec_flush_buffers() on encoders.
Encoders that set the cap AV_CODEC_CAP_ENCODER_FLUSH will be flushed.
For all other encoders, the call is now a no-op rather than undefined
behaviour.
2020-04-10 - 672946c7fe - lavc 58.78.100 - avcodec.h codec_desc.h codec_id.h packet.h
Move AVCodecDesc-related public API to new header codec_desc.h.
Move AVCodecID enum to new header codec_id.h.
Move AVPacket-related public API to new header packet.h.
2020-03-29 - 4cb0dda555 - lavf 58.42.100 - avformat.h
av_read_frame() now guarantees to handle uninitialized input packets
and to return refcounted packets on success.
2020-03-27 - c52ec0367d - lavc 58.77.100 - avcodec.h
av_packet_ref() now guarantees to return the destination packet
in a blank state on error.
2020-03-10 - 05d27f342b - lavc 58.75.100 - avcodec.h
Add AV_PKT_DATA_ICC_PROFILE.
2020-02-21 - d005a7cdfd - lavc 58.73.101 - avcodec.h
Add AV_CODEC_EXPORT_DATA_PRFT.
2020-02-21 - c666689491 - lavc 58.73.100 - avcodec.h
Add AVCodecContext.export_side_data and AV_CODEC_EXPORT_DATA_MVS.
2020-02-13 - e8f054b095 - lavu 56.41.100 - tx.h
Add AV_TX_INT32_FFT and AV_TX_INT32_MDCT
2020-02-12 - 3182114f88 - lavu 56.40.100 - log.h
Add av_log_once().
2020-02-04 - a88449ffb2 - lavu 56.39.100 - hwcontext.h
Add AV_PIX_FMT_VULKAN
Add AV_HWDEVICE_TYPE_VULKAN and implementation.
2020-01-30 - 27529eeb27 - lavf 58.37.100 - avio.h
Add avio_protocol_get_class().
2020-01-15 - 717b2074ec - lavc 58.66.100 - avcodec.h
Add AV_PKT_DATA_PRFT and AVProducerReferenceTime.
2019-12-27 - 45259a0ee4 - lavu 56.38.100 - eval.h
Add av_expr_count_func().
2019-12-26 - 16685114d5 - lavu 56.37.100 - buffer.h
Add av_buffer_pool_buffer_get_opaque().
2019-11-17 - 1c23abc88f - lavu 56.36.100 - eval API
Add av_expr_count_vars().
2019-10-14 - f3746d31f9 - lavu 56.35.101 - opt.h
Add AV_OPT_FLAG_RUNTIME_PARAM.
2019-09-25 - f8406ab4b9 - lavc 58.59.100 - avcodec.h
Add max_samples
2019-09-04 - 2a9d461abc - lavu 56.35.100 - hwcontext_videotoolbox.h
Add av_map_videotoolbox_format_from_pixfmt2() for full range pixfmt
2019-09-01 - 8821d1f56e - lavu 56.34.100 - pixfmt.h
Add EBU Tech. 3213-E AVColorPrimaries value
2019-08-17 - 95fa73a2b4 - lavf 58.31.101 - avio.h
4K limit removed from avio_printf.
2019-08-17 - a82f8f2f10 - lavf 58.31.100 - avio.h
Add avio_print_string_array and avio_print.
2019-07-27 - 42e2319ba9 - lavu 56.33.100 - tx.h
Add AV_TX_DOUBLE_FFT and AV_TX_DOUBLE_MDCT
-------- 8< --------- FFmpeg 4.2 was cut here -------- 8< ---------
2019-06-21 - a30e44098a - lavu 56.30.100 - frame.h
Add FF_DECODE_ERROR_DECODE_SLICES
2019-06-14 - edfced8c04 - lavu 56.29.100 - frame.h
Add FF_DECODE_ERROR_CONCEALMENT_ACTIVE
2019-05-15 - b79b29ddb1 - lavu 56.28.100 - tx.h
Add av_tx_init(), av_tx_uninit() and related definitions.
2019-04-20 - 3153a6502a - lavc 58.52.100 - avcodec.h
Add AV_CODEC_FLAG_DROPCHANGED to allow avcodec_receive_frame to drop
frames whose parameters differ from first decoded frame in stream.
2019-04-12 - abfeba9724 - lavf 58.27.102
Rename hls,applehttp demuxer to hls
2019-01-27 - 5bcefceec8 - lavc 58.46.100 - avcodec.h
Add discard_damaged_percentage
2019-01-08 - 1ef4828276 - lavu 56.26.100 - frame.h
Add AV_FRAME_DATA_REGIONS_OF_INTEREST
2018-12-21 - 2744d6b364 - lavu 56.25.100 - hdr_dynamic_metadata.h
Add AV_FRAME_DATA_DYNAMIC_HDR_PLUS enum value, av_dynamic_hdr_plus_alloc(),
av_dynamic_hdr_plus_create_side_data() functions, and related structs.
-------- 8< --------- FFmpeg 4.1 was cut here -------- 8< ---------
2018-10-27 - 718044dc19 - lavu 56.21.100 - pixdesc.h
@ -1810,7 +1137,7 @@ API changes, most recent first:
2014-04-15 - ef818d8 - lavf 55.37.101 - avformat.h
Add av_format_inject_global_side_data()
2014-04-12 - 4f698be8f - lavu 52.76.100 - log.h
2014-04-12 - 4f698be - lavu 52.76.100 - log.h
Add av_log_get_flags()
2014-04-11 - 6db42a2b - lavd 55.12.100 - avdevice.h

3
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/doc/Doxyfile → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/Doxyfile
View file

@ -38,7 +38,7 @@ PROJECT_NAME = FFmpeg
# could be handy for archiving the generated documentation or if some version
# control system is used.
PROJECT_NUMBER = 6.0
PROJECT_NUMBER = 4.1
# Using the PROJECT_BRIEF tag one can provide an optional one line description
# for a project that appears at the top of each page and should give viewer a
@ -1980,7 +1980,6 @@ PREDEFINED = __attribute__(x)= \
av_alloc_size(...)= \
AV_GCC_VERSION_AT_LEAST(x,y)=1 \
AV_GCC_VERSION_AT_MOST(x,y)=0 \
"FF_PAD_STRUCTURE(name,size,...)=typedef struct name { __VA_ARGS__ } name;" \
__GNUC__
# If the MACRO_EXPANSION and EXPAND_ONLY_PREDEF tags are set to YES then this

154
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/Makefile Normal file
View file

@ -0,0 +1,154 @@
LIBRARIES-$(CONFIG_AVUTIL) += libavutil
LIBRARIES-$(CONFIG_SWSCALE) += libswscale
LIBRARIES-$(CONFIG_SWRESAMPLE) += libswresample
LIBRARIES-$(CONFIG_AVCODEC) += libavcodec
LIBRARIES-$(CONFIG_AVFORMAT) += libavformat
LIBRARIES-$(CONFIG_AVDEVICE) += libavdevice
LIBRARIES-$(CONFIG_AVFILTER) += libavfilter
COMPONENTS-$(CONFIG_AVUTIL) += ffmpeg-utils
COMPONENTS-$(CONFIG_SWSCALE) += ffmpeg-scaler
COMPONENTS-$(CONFIG_SWRESAMPLE) += ffmpeg-resampler
COMPONENTS-$(CONFIG_AVCODEC) += ffmpeg-codecs ffmpeg-bitstream-filters
COMPONENTS-$(CONFIG_AVFORMAT) += ffmpeg-formats ffmpeg-protocols
COMPONENTS-$(CONFIG_AVDEVICE) += ffmpeg-devices
COMPONENTS-$(CONFIG_AVFILTER) += ffmpeg-filters
MANPAGES1 = $(AVPROGS-yes:%=doc/%.1) $(AVPROGS-yes:%=doc/%-all.1) $(COMPONENTS-yes:%=doc/%.1)
MANPAGES3 = $(LIBRARIES-yes:%=doc/%.3)
MANPAGES = $(MANPAGES1) $(MANPAGES3)
PODPAGES = $(AVPROGS-yes:%=doc/%.pod) $(AVPROGS-yes:%=doc/%-all.pod) $(COMPONENTS-yes:%=doc/%.pod) $(LIBRARIES-yes:%=doc/%.pod)
HTMLPAGES = $(AVPROGS-yes:%=doc/%.html) $(AVPROGS-yes:%=doc/%-all.html) $(COMPONENTS-yes:%=doc/%.html) $(LIBRARIES-yes:%=doc/%.html) \
doc/developer.html \
doc/faq.html \
doc/fate.html \
doc/general.html \
doc/git-howto.html \
doc/mailing-list-faq.html \
doc/nut.html \
doc/platform.html \
TXTPAGES = doc/fate.txt \
DOCS-$(CONFIG_HTMLPAGES) += $(HTMLPAGES)
DOCS-$(CONFIG_PODPAGES) += $(PODPAGES)
DOCS-$(CONFIG_MANPAGES) += $(MANPAGES)
DOCS-$(CONFIG_TXTPAGES) += $(TXTPAGES)
DOCS = $(DOCS-yes)
all-$(CONFIG_DOC): doc
doc: documentation
apidoc: doc/doxy/html
documentation: $(DOCS)
TEXIDEP = perl $(SRC_PATH)/doc/texidep.pl $(SRC_PATH) $< $@ >$(@:%=%.d)
doc/%.txt: TAG = TXT
doc/%.txt: doc/%.texi
$(Q)$(TEXIDEP)
$(M)makeinfo --force --no-headers -o $@ $< 2>/dev/null
GENTEXI = format codec
GENTEXI := $(GENTEXI:%=doc/avoptions_%.texi)
$(GENTEXI): TAG = GENTEXI
$(GENTEXI): doc/avoptions_%.texi: doc/print_options$(HOSTEXESUF)
$(M)doc/print_options $* > $@
doc/%.html: TAG = HTML
doc/%-all.html: TAG = HTML
ifdef HAVE_MAKEINFO_HTML
doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.pm $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)makeinfo --html -I doc --no-split -D config-not-all --init-file=$(SRC_PATH)/doc/t2h.pm --output $@ $<
doc/%-all.html: doc/%.texi $(SRC_PATH)/doc/t2h.pm $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)makeinfo --html -I doc --no-split -D config-all --init-file=$(SRC_PATH)/doc/t2h.pm --output $@ $<
else
doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)texi2html -I doc -monolithic --D=config-not-all --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
doc/%-all.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)texi2html -I doc -monolithic --D=config-all --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
endif
doc/%.pod: TAG = POD
doc/%.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)perl $(SRC_PATH)/doc/texi2pod.pl -Dconfig-not-all=yes -Idoc $< $@
doc/%-all.pod: TAG = POD
doc/%-all.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)perl $(SRC_PATH)/doc/texi2pod.pl -Dconfig-all=yes -Idoc $< $@
doc/%.1 doc/%.3: TAG = MAN
doc/%.1: doc/%.pod $(GENTEXI)
$(M)pod2man --section=1 --center=" " --release=" " --date=" " $< > $@
doc/%.3: doc/%.pod $(GENTEXI)
$(M)pod2man --section=3 --center=" " --release=" " --date=" " $< > $@
$(DOCS) doc/doxy/html: | doc/
DOXY_INPUT = $(INSTHEADERS)
DOXY_INPUT_DEPS = $(addprefix $(SRC_PATH)/, $(DOXY_INPUT)) ffbuild/config.mak
doc/doxy/html: TAG = DOXY
doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(SRC_PATH)/doc/doxy-wrapper.sh $(DOXY_INPUT_DEPS)
$(M)OUT_DIR=$$PWD/doc/doxy; cd $(SRC_PATH); ./doc/doxy-wrapper.sh $$OUT_DIR $< $(DOXYGEN) $(DOXY_INPUT);
install-doc: install-html install-man
install-html:
install-man:
ifdef CONFIG_HTMLPAGES
install-progs-$(CONFIG_DOC): install-html
install-html: $(HTMLPAGES)
$(Q)mkdir -p "$(DOCDIR)"
$(INSTALL) -m 644 $(HTMLPAGES) "$(DOCDIR)"
endif
ifdef CONFIG_MANPAGES
install-progs-$(CONFIG_DOC): install-man
install-man: $(MANPAGES)
$(Q)mkdir -p "$(MANDIR)/man1"
$(INSTALL) -m 644 $(MANPAGES1) "$(MANDIR)/man1"
$(Q)mkdir -p "$(MANDIR)/man3"
$(INSTALL) -m 644 $(MANPAGES3) "$(MANDIR)/man3"
endif
uninstall: uninstall-doc
uninstall-doc: uninstall-html uninstall-man
uninstall-html:
$(RM) -r "$(DOCDIR)"
uninstall-man:
$(RM) $(addprefix "$(MANDIR)/man1/",$(AVPROGS-yes:%=%.1) $(AVPROGS-yes:%=%-all.1) $(COMPONENTS-yes:%=%.1))
$(RM) $(addprefix "$(MANDIR)/man3/",$(LIBRARIES-yes:%=%.3))
clean:: docclean
distclean:: docclean
$(RM) doc/config.texi
docclean::
$(RM) $(CLEANSUFFIXES:%=doc/%)
$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 doc/*.3 doc/avoptions_*.texi
$(RM) -r doc/doxy/html
-include $(wildcard $(DOCS:%=%.d))
.PHONY: apidoc doc documentation

0
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/doc/authors.texi → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/authors.texi
View file

612
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/bitstream_filters.texi Normal file
View file

@ -0,0 +1,612 @@
@chapter Bitstream Filters
@c man begin BITSTREAM FILTERS
When you configure your FFmpeg build, all the supported bitstream
filters are enabled by default. You can list all available ones using
the configure option @code{--list-bsfs}.
You can disable all the bitstream filters using the configure option
@code{--disable-bsfs}, and selectively enable any bitstream filter using
the option @code{--enable-bsf=BSF}, or you can disable a particular
bitstream filter using the option @code{--disable-bsf=BSF}.
The option @code{-bsfs} of the ff* tools will display the list of
all the supported bitstream filters included in your build.
The ff* tools have a -bsf option applied per stream, taking a
comma-separated list of filters, whose parameters follow the filter
name after a '='.
@example
ffmpeg -i INPUT -c:v copy -bsf:v filter1[=opt1=str1:opt2=str2][,filter2] OUTPUT
@end example
Below is a description of the currently available bitstream filters,
with their parameters, if any.
@section aac_adtstoasc
Convert MPEG-2/4 AAC ADTS to an MPEG-4 Audio Specific Configuration
bitstream.
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
@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 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}.
For example the following @command{ffmpeg} command forces a global
header (thus disabling individual packet headers) in the H.264 packets
generated by the @code{libx264} encoder, but corrects them by adding
the header stored in extradata to the key packets:
@example
ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts
@end example
@section 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
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}).
For example to remux an MP4 file containing an H.264 stream to mpegts
format with @command{ffmpeg}, you can use the command:
@example
ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts
@end example
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
Pro decoder. This filter only applies to the mpeg2video codec, and is
likely not needed for Final Cut Pro 7 and newer with the appropriate
@option{-tag:v}.
For example, to remux 30 MB/sec NTSC IMX to MOV:
@example
ffmpeg -i input.mxf -c copy -bsf:v imxdump -tag:v mx3n output.mov
@end example
@section mjpeg2jpeg
Convert MJPEG/AVI1 packets to full JPEG/JFIF packets.
MJPEG is a video codec wherein each video frame is essentially a
JPEG image. The individual frames can be extracted without loss,
e.g. by
@example
ffmpeg -i ../some_mjpeg.avi -c:v copy frames_%d.jpg
@end example
Unfortunately, these chunks are incomplete JPEG images, because
they lack the DHT segment required for decoding. Quoting from
@url{http://www.digitalpreservation.gov/formats/fdd/fdd000063.shtml}:
Avery Lee, writing in the rec.video.desktop newsgroup in 2001,
commented that "MJPEG, or at least the MJPEG in AVIs having the
MJPG fourcc, is restricted JPEG with a fixed -- and *omitted* --
Huffman table. The JPEG must be YCbCr colorspace, it must be 4:2:2,
and it must use basic Huffman encoding, not arithmetic or
progressive. . . . You can indeed extract the MJPEG frames and
decode them with a regular JPEG decoder, but you have to prepend
the DHT segment to them, or else the decoder won't have any idea
how to decompress the data. The exact table necessary is given in
the OpenDML spec."
This bitstream filter patches the header of frames extracted from an MJPEG
stream (carrying the AVI1 header ID and lacking a DHT segment) to
produce fully qualified JPEG images.
@example
ffmpeg -i mjpeg-movie.avi -c:v copy -bsf:v mjpeg2jpeg frame_%d.jpg
exiftran -i -9 frame*.jpg
ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
@end example
@section mjpegadump
Add an MJPEG A header to the bitstream, to enable decoding by
Quicktime.
@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 mpeg4_unpack_bframes
Unpack DivX-style packed B-frames.
DivX-style packed B-frames are not valid MPEG-4 and were only a
workaround for the broken Video for Windows subsystem.
They use more space, can cause minor AV sync issues, require more
CPU power to decode (unless the player has some decoded picture queue
to compensate the 2,0,2,0 frame per packet style) and cause
trouble if copied into a standard container like mp4 or mpeg-ps/ts,
because MPEG-4 decoders may not be able to decode them, since they are
not valid MPEG-4.
For example to fix an AVI file containing an MPEG-4 stream with
DivX-style packed B-frames using @command{ffmpeg}, you can use the command:
@example
ffmpeg -i INPUT.avi -codec copy -bsf:v mpeg4_unpack_bframes OUTPUT.avi
@end example
@section noise
Damages the contents of packets or simply drops them 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.
@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

0
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/doc/bootstrap.min.css → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/bootstrap.min.css vendored
View file

10
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/doc/build_system.txt → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/build_system.txt
View file

@ -36,11 +36,11 @@ install
examples
Build all examples located in doc/examples.
checkheaders
Check headers dependencies.
libavformat/output-example
Build the libavformat basic example.
alltools
Build all tools in tools directory.
libswscale/swscale-test
Build the swscale self-test (useful also as an example).
config
Reconfigure the project with the current configuration.
@ -48,8 +48,6 @@ config
tools/target_dec_<decoder>_fuzzer
Build fuzzer to fuzz the specified decoder.
tools/target_bsf_<filter>_fuzzer
Build fuzzer to fuzz the specified bitstream filter.
Useful standard make commands:
make -t <target>

351
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/doc/codecs.texi → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/codecs.texi
View file

@ -50,13 +50,11 @@ Use internal 2pass ratecontrol in first pass mode.
Use internal 2pass ratecontrol in second pass mode.
@item gray
Only decode/encode grayscale.
@item emu_edge
Do not draw edges.
@item psnr
Set error[?] variables during encoding.
@item truncated
Input bitstream might be randomly truncated.
@item drop_changed
Don't output frames whose parameters differ from first decoded frame in stream.
Error AVERROR_INPUT_CHANGED is returned when a frame is dropped.
@item ildct
Use interlaced DCT.
@ -70,14 +68,50 @@ This ensures that file and data checksums are reproducible and match between
platforms. Its primary use is for regression testing.
@item aic
Apply H263 advanced intra coding / mpeg4 ac prediction.
@item cbp
Deprecated, use mpegvideo private options instead.
@item qprd
Deprecated, use mpegvideo private options instead.
@item ilme
Apply interlaced motion estimation.
@item cgop
Use closed gop.
@item output_corrupt
Output even potentially corrupted frames.
@end table
@item me_method @var{integer} (@emph{encoding,video})
Set motion estimation method.
Possible values:
@table @samp
@item zero
zero motion estimation (fastest)
@item full
full motion estimation (slowest)
@item epzs
EPZS motion estimation (default)
@item esa
esa motion estimation (alias for full)
@item tesa
tesa motion estimation
@item dia
dia motion estimation (alias for epzs)
@item log
log motion estimation
@item phods
phods motion estimation
@item x1
X1 motion estimation
@item hex
hex motion estimation
@item umh
umh motion estimation
@item iter
iter motion estimation
@end table
@item extradata_size @var{integer}
Set extradata size.
@item time_base @var{rational number}
Set codec time base.
@ -144,6 +178,24 @@ Default value is 0.
@item b_qfactor @var{float} (@emph{encoding,video})
Set qp factor between P and B frames.
@item rc_strategy @var{integer} (@emph{encoding,video})
Set ratecontrol method.
@item b_strategy @var{integer} (@emph{encoding,video})
Set strategy to choose between I/P/B-frames.
@item ps @var{integer} (@emph{encoding,video})
Set RTP payload size in bytes.
@item mv_bits @var{integer}
@item header_bits @var{integer}
@item i_tex_bits @var{integer}
@item p_tex_bits @var{integer}
@item i_count @var{integer}
@item p_count @var{integer}
@item skip_count @var{integer}
@item misc_bits @var{integer}
@item frame_bits @var{integer}
@item codec_tag @var{integer}
@item bug @var{flags} (@emph{decoding,video})
Workaround not auto detected encoder bugs.
@ -152,6 +204,8 @@ Possible values:
@table @samp
@item autodetect
@item old_msmpeg4
some old lavc generated msmpeg4v3 files (no autodetection)
@item xvid_ilace
Xvid interlacing bug (autodetected if fourcc==XVIX)
@item ump4
@ -160,6 +214,8 @@ Xvid interlacing bug (autodetected if fourcc==XVIX)
padding bug (autodetected)
@item amv
@item ac_vlc
illegal vlc bug (autodetected per fourcc)
@item qpel_chroma
@item std_qpel
@ -180,6 +236,14 @@ Workaround various bugs in microsoft broken decoders.
trancated frames
@end table
@item lelim @var{integer} (@emph{encoding,video})
Set single coefficient elimination threshold for luminance (negative
values also consider DC coefficient).
@item celim @var{integer} (@emph{encoding,video})
Set single coefficient elimination threshold for chrominance (negative
values also consider dc coefficient)
@item strict @var{integer} (@emph{decoding/encoding,audio,video})
Specify how strictly to follow the standards.
@ -233,8 +297,29 @@ consider things that a sane encoder should not do as an error
@item block_align @var{integer}
@item mpeg_quant @var{integer} (@emph{encoding,video})
Use MPEG quantizers instead of H.263.
@item qsquish @var{float} (@emph{encoding,video})
How to keep quantizer between qmin and qmax (0 = clip, 1 = use
differentiable function).
@item rc_qmod_amp @var{float} (@emph{encoding,video})
Set experimental quantizer modulation.
@item rc_qmod_freq @var{integer} (@emph{encoding,video})
Set experimental quantizer modulation.
@item rc_override_count @var{integer}
@item rc_eq @var{string} (@emph{encoding,video})
Set rate control equation. When computing the expression, besides the
standard functions defined in the section 'Expression Evaluation', the
following functions are available: bits2qp(bits), qp2bits(qp). Also
the following constants are available: iTex pTex tex mv fCode iCount
mcVar var isI isP isB avgQP qComp avgIITex avgPITex avgPPTex avgBPTex
avgTex.
@item maxrate @var{integer} (@emph{encoding,audio,video})
Set max bitrate tolerance (in bits/s). Requires bufsize to be set.
@ -245,12 +330,18 @@ encode. It is of little use elsewise.
@item bufsize @var{integer} (@emph{encoding,audio,video})
Set ratecontrol buffer size (in bits).
@item rc_buf_aggressivity @var{float} (@emph{encoding,video})
Currently useless.
@item i_qfactor @var{float} (@emph{encoding,video})
Set QP factor between P and I frames.
@item i_qoffset @var{float} (@emph{encoding,video})
Set QP offset between P and I frames.
@item rc_init_cplx @var{float} (@emph{encoding,video})
Set initial complexity for 1-pass encoding.
@item dct @var{integer} (@emph{encoding,video})
Set DCT algorithm.
@ -315,7 +406,11 @@ Automatically pick a IDCT compatible with the simple one
@item simpleneon
@item xvid
@item simplealpha
@item ipp
@item xvidmmx
@item faani
floating point AAN IDCT
@ -338,6 +433,19 @@ favor predicting from the previous frame instead of the current
@item bits_per_coded_sample @var{integer}
@item pred @var{integer} (@emph{encoding,video})
Set prediction method.
Possible values:
@table @samp
@item left
@item plane
@item median
@end table
@item aspect @var{rational number} (@emph{encoding,video})
Set sample aspect ratio.
@ -532,28 +640,13 @@ noise preserving sum of squared differences
@item dia_size @var{integer} (@emph{encoding,video})
Set diamond type & size for motion estimation.
@table @samp
@item (1024, INT_MAX)
full motion estimation(slowest)
@item (768, 1024]
umh motion estimation
@item (512, 768]
hex motion estimation
@item (256, 512]
l2s diamond motion estimation
@item [2,256]
var diamond motion estimation
@item (-1, 2)
small diamond motion estimation
@item -1
funny diamond motion estimation
@item (INT_MIN, -1)
sab diamond motion estimation
@end table
@item last_pred @var{integer} (@emph{encoding,video})
Set amount of motion predictors from the previous frame.
@item preme @var{integer} (@emph{encoding,video})
Set pre motion estimation.
@item precmp @var{integer} (@emph{encoding,video})
Set pre motion estimation compare function.
@ -597,11 +690,40 @@ Set diamond type & size for motion estimation pre-pass.
@item subq @var{integer} (@emph{encoding,video})
Set sub pel motion estimation quality.
@item dtg_active_format @var{integer}
@item me_range @var{integer} (@emph{encoding,video})
Set limit motion vectors range (1023 for DivX player).
@item ibias @var{integer} (@emph{encoding,video})
Set intra quant bias.
@item pbias @var{integer} (@emph{encoding,video})
Set inter quant bias.
@item color_table_id @var{integer}
@item global_quality @var{integer} (@emph{encoding,audio,video})
@item coder @var{integer} (@emph{encoding,video})
Possible values:
@table @samp
@item vlc
variable length coder / huffman coder
@item ac
arithmetic coder
@item raw
raw (no encoding)
@item rle
run-length coder
@item deflate
deflate-based coder
@end table
@item context @var{integer} (@emph{encoding,video})
Set context model.
@item slice_flags @var{integer}
@item mbd @var{integer} (@emph{encoding,video})
@ -617,16 +739,32 @@ use fewest bits
use best rate distortion
@end table
@item stream_codec_tag @var{integer}
@item sc_threshold @var{integer} (@emph{encoding,video})
Set scene change threshold.
@item lmin @var{integer} (@emph{encoding,video})
Set min lagrange factor (VBR).
@item lmax @var{integer} (@emph{encoding,video})
Set max lagrange factor (VBR).
@item nr @var{integer} (@emph{encoding,video})
Set noise reduction.
@item rc_init_occupancy @var{integer} (@emph{encoding,video})
Set number of bits which should be loaded into the rc buffer before
decoding starts.
@item flags2 @var{flags} (@emph{decoding/encoding,audio,video,subtitles})
@item flags2 @var{flags} (@emph{decoding/encoding,audio,video})
Possible values:
@table @samp
@item fast
Allow non spec compliant speedup tricks.
@item sgop
Deprecated, use mpegvideo private options instead.
@item noout
Skip bitstream encoding.
@item ignorecrop
@ -637,34 +775,17 @@ Place global headers at every keyframe instead of in extradata.
Frame data might be split into multiple chunks.
@item showall
Show all frames before the first keyframe.
@item skiprd
Deprecated, use mpegvideo private options instead.
@item export_mvs
Export motion vectors into frame side-data (see @code{AV_FRAME_DATA_MOTION_VECTORS})
for codecs that support it. See also @file{doc/examples/export_mvs.c}.
@item skip_manual
Do not skip samples and export skip information as frame side data.
@item ass_ro_flush_noop
Do not reset ASS ReadOrder field on flush.
@item icc_profiles
Generate/parse embedded ICC profiles from/to colorimetry tags.
@end table
@item export_side_data @var{flags} (@emph{decoding/encoding,audio,video,subtitles})
@item error @var{integer} (@emph{encoding,video})
Possible values:
@table @samp
@item mvs
Export motion vectors into frame side-data (see @code{AV_FRAME_DATA_MOTION_VECTORS})
for codecs that support it. See also @file{doc/examples/export_mvs.c}.
@item prft
Export encoder Producer Reference Time into packet side-data (see @code{AV_PKT_DATA_PRFT})
for codecs that support it.
@item venc_params
Export video encoding parameters through frame side data (see @code{AV_FRAME_DATA_VIDEO_ENC_PARAMS})
for codecs that support it. At present, those are H.264 and VP9.
@item film_grain
Export film grain parameters through frame side data (see @code{AV_FRAME_DATA_FILM_GRAIN_PARAMS}).
Supported at present by AV1 decoders.
@end table
@item qns @var{integer} (@emph{encoding,video})
Deprecated, use mpegvideo private options instead.
@item threads @var{integer} (@emph{decoding/encoding,video})
Set the number of threads to be used, in case the selected codec
@ -678,6 +799,12 @@ automatically select the number of threads to set
Default value is @samp{auto}.
@item me_threshold @var{integer} (@emph{encoding,video})
Set motion estimation threshold.
@item mb_threshold @var{integer} (@emph{encoding,video})
Set macroblock threshold.
@item dc @var{integer} (@emph{encoding,video})
Set intra_dc_precision.
@ -692,8 +819,49 @@ Set number of macroblock rows at the bottom which are skipped.
@item profile @var{integer} (@emph{encoding,audio,video})
Set encoder codec profile. Default value is @samp{unknown}. Encoder specific
profiles are documented in the relevant encoder documentation.
Possible values:
@table @samp
@item unknown
@item aac_main
@item aac_low
@item aac_ssr
@item aac_ltp
@item aac_he
@item aac_he_v2
@item aac_ld
@item aac_eld
@item mpeg2_aac_low
@item mpeg2_aac_he
@item mpeg4_sp
@item mpeg4_core
@item mpeg4_main
@item mpeg4_asp
@item dts
@item dts_es
@item dts_96_24
@item dts_hd_hra
@item dts_hd_ma
@end table
@item level @var{integer} (@emph{encoding,audio,video})
@ -706,12 +874,67 @@ Possible values:
@item lowres @var{integer} (@emph{decoding,audio,video})
Decode at 1= 1/2, 2=1/4, 3=1/8 resolutions.
@item skip_threshold @var{integer} (@emph{encoding,video})
Set frame skip threshold.
@item skip_factor @var{integer} (@emph{encoding,video})
Set frame skip factor.
@item skip_exp @var{integer} (@emph{encoding,video})
Set frame skip exponent.
Negative values behave identical to the corresponding positive ones, except
that the score is normalized.
Positive values exist primarily for compatibility reasons and are not so useful.
@item skipcmp @var{integer} (@emph{encoding,video})
Set frame skip compare function.
Possible values:
@table @samp
@item sad
sum of absolute differences, fast (default)
@item sse
sum of squared errors
@item satd
sum of absolute Hadamard transformed differences
@item dct
sum of absolute DCT transformed differences
@item psnr
sum of squared quantization errors (avoid, low quality)
@item bit
number of bits needed for the block
@item rd
rate distortion optimal, slow
@item zero
0
@item vsad
sum of absolute vertical differences
@item vsse
sum of squared vertical differences
@item nsse
noise preserving sum of squared differences
@item w53
5/3 wavelet, only used in snow
@item w97
9/7 wavelet, only used in snow
@item dctmax
@item chroma
@end table
@item border_mask @var{float} (@emph{encoding,video})
Increase the quantizer for macroblocks close to borders.
@item mblmin @var{integer} (@emph{encoding,video})
Set min macroblock lagrange factor (VBR).
@item mblmax @var{integer} (@emph{encoding,video})
Set max macroblock lagrange factor (VBR).
@item mepc @var{integer} (@emph{encoding,video})
Set motion estimation bitrate penalty compensation (1.0 = 256).
@item skip_loop_filter @var{integer} (@emph{decoding,video})
@item skip_idct @var{integer} (@emph{decoding,video})
@item skip_frame @var{integer} (@emph{decoding,video})
@ -739,9 +962,6 @@ Discard all bidirectional frames.
@item nokey
Discard all frames excepts keyframes.
@item nointra
Discard all frames except I frames.
@item all
Discard all frames.
@end table
@ -751,17 +971,34 @@ Default value is @samp{default}.
@item bidir_refine @var{integer} (@emph{encoding,video})
Refine the two motion vectors used in bidirectional macroblocks.
@item brd_scale @var{integer} (@emph{encoding,video})
Downscale frames for dynamic B-frame decision.
@item keyint_min @var{integer} (@emph{encoding,video})
Set minimum interval between IDR-frames.
@item refs @var{integer} (@emph{encoding,video})
Set reference frames to consider for motion compensation.
@item chromaoffset @var{integer} (@emph{encoding,video})
Set chroma qp offset from luma.
@item trellis @var{integer} (@emph{encoding,audio,video})
Set rate-distortion optimal quantization.
@item mv0_threshold @var{integer} (@emph{encoding,video})
@item b_sensitivity @var{integer} (@emph{encoding,video})
Adjust sensitivity of b_frame_strategy 1.
@item compression_level @var{integer} (@emph{encoding,audio,video})
@item min_prediction_order @var{integer} (@emph{encoding,audio})
@item max_prediction_order @var{integer} (@emph{encoding,audio})
@item timecode_frame_start @var{integer} (@emph{encoding,video})
Set GOP timecode frame start number, in non drop frame format.
@item request_channels @var{integer} (@emph{decoding,audio})
Set desired number of audio channels.
@item bits_per_raw_sample @var{integer}
@item channel_layout @var{integer} (@emph{decoding/encoding,audio})
@ -875,12 +1112,6 @@ BT.2020 NCL
BT.2020 CL
@item smpte2085
SMPTE 2085
@item chroma-derived-nc
Chroma-derived NCL
@item chroma-derived-c
Chroma-derived CL
@item ictcp
ICtCp
@end table
@item color_range @var{integer} (@emph{decoding/encoding,video})
@ -1001,7 +1232,7 @@ instead of alpha. Default is 0.
@item dump_separator @var{string} (@emph{input})
Separator used to separate the fields printed on the command line about the
Stream parameters.
For example, to separate the fields with newlines and indentation:
For example to separate the fields with newlines and indention:
@example
ffprobe -dump_separator "
" -i ~/videos/matrixbench_mpeg2.mpg

301
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/decoders.texi Normal file
View file

@ -0,0 +1,301 @@
@chapter Decoders
@c man begin DECODERS
Decoders are configured elements in FFmpeg which allow the decoding of
multimedia streams.
When you configure your FFmpeg build, all the supported native decoders
are enabled by default. Decoders requiring an external library must be enabled
manually via the corresponding @code{--enable-lib} option. You can list all
available decoders using the configure option @code{--list-decoders}.
You can disable all the decoders with the configure option
@code{--disable-decoders} and selectively enable / disable single decoders
with the options @code{--enable-decoder=@var{DECODER}} /
@code{--disable-decoder=@var{DECODER}}.
The option @code{-decoders} of the ff* tools will display the list of
enabled decoders.
@c man end DECODERS
@chapter Video Decoders
@c man begin VIDEO DECODERS
A description of some of the currently available video decoders
follows.
@section rawvideo
Raw video decoder.
This decoder decodes rawvideo streams.
@subsection Options
@table @option
@item top @var{top_field_first}
Specify the assumed field type of the input video.
@table @option
@item -1
the video is assumed to be progressive (default)
@item 0
bottom-field-first is assumed
@item 1
top-field-first is assumed
@end table
@end table
@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
@c man begin AUDIO DECODERS
A description of some of the currently available audio decoders
follows.
@section ac3
AC-3 audio decoder.
This decoder implements part of ATSC A/52:2010 and ETSI TS 102 366, as well as
the undocumented RealAudio 3 (a.k.a. dnet).
@subsection AC-3 Decoder Options
@table @option
@item -drc_scale @var{value}
Dynamic Range Scale Factor. The factor to apply to dynamic range values
from the AC-3 stream. This factor is applied exponentially.
There are 3 notable scale factor ranges:
@table @option
@item drc_scale == 0
DRC disabled. Produces full range audio.
@item 0 < drc_scale <= 1
DRC enabled. Applies a fraction of the stream DRC value.
Audio reproduction is between full range and full compression.
@item drc_scale > 1
DRC enabled. Applies drc_scale asymmetrically.
Loud sounds are fully compressed. Soft sounds are enhanced.
@end table
@end table
@section flac
FLAC audio decoder.
This decoder aims to implement the complete FLAC specification from Xiph.
@subsection FLAC Decoder options
@table @option
@item -use_buggy_lpc
The lavc FLAC encoder used to produce buggy streams with high lpc values
(like the default value). This option makes it possible to decode such streams
correctly by using lavc's old buggy lpc logic for decoding.
@end table
@section ffwavesynth
Internal wave synthesizer.
This decoder generates wave patterns according to predefined sequences. Its
use is purely internal and the format of the data it accepts is not publicly
documented.
@section libcelt
libcelt decoder wrapper.
libcelt allows libavcodec to decode the Xiph CELT ultra-low delay audio codec.
Requires the presence of the libcelt headers and library during configuration.
You need to explicitly configure the build with @code{--enable-libcelt}.
@section libgsm
libgsm decoder wrapper.
libgsm allows libavcodec to decode the GSM full rate audio codec. Requires
the presence of the libgsm headers and library during configuration. You need
to explicitly configure the build with @code{--enable-libgsm}.
This decoder supports both the ordinary GSM and the Microsoft variant.
@section libilbc
libilbc decoder wrapper.
libilbc allows libavcodec to decode the Internet Low Bitrate Codec (iLBC)
audio codec. Requires the presence of the libilbc headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libilbc}.
@subsection Options
The following option is supported by the libilbc wrapper.
@table @option
@item enhance
Enable the enhancement of the decoded audio when set to 1. The default
value is 0 (disabled).
@end table
@section libopencore-amrnb
libopencore-amrnb decoder wrapper.
libopencore-amrnb allows libavcodec to decode the Adaptive Multi-Rate
Narrowband audio codec. Using it requires the presence of the
libopencore-amrnb headers and library during configuration. You need to
explicitly configure the build with @code{--enable-libopencore-amrnb}.
An FFmpeg native decoder for AMR-NB exists, so users can decode AMR-NB
without this library.
@section libopencore-amrwb
libopencore-amrwb decoder wrapper.
libopencore-amrwb allows libavcodec to decode the Adaptive Multi-Rate
Wideband audio codec. Using it requires the presence of the
libopencore-amrwb headers and library during configuration. You need to
explicitly configure the build with @code{--enable-libopencore-amrwb}.
An FFmpeg native decoder for AMR-WB exists, so users can decode AMR-WB
without this library.
@section libopus
libopus decoder wrapper.
libopus allows libavcodec to decode the Opus Interactive Audio Codec.
Requires the presence of the libopus headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libopus}.
An FFmpeg native decoder for Opus exists, so users can decode Opus
without this library.
@c man end AUDIO DECODERS
@chapter Subtitles Decoders
@c man begin SUBTILES DECODERS
@section dvbsub
@subsection Options
@table @option
@item compute_clut
@table @option
@item -1
Compute clut if no matching CLUT is in the stream.
@item 0
Never compute CLUT
@item 1
Always compute CLUT and override the one provided in the stream.
@end table
@item dvb_substream
Selects the dvb substream, or all substreams if -1 which is default.
@end table
@section dvdsub
This codec decodes the bitmap subtitles used in DVDs; the same subtitles can
also be found in VobSub file pairs and in some Matroska files.
@subsection Options
@table @option
@item palette
Specify the global palette used by the bitmaps. When stored in VobSub, the
palette is normally specified in the index file; in Matroska, the palette is
stored in the codec extra-data in the same format as in VobSub. In DVDs, the
palette is stored in the IFO file, and therefore not available when reading
from dumped VOB files.
The format for this option is a string containing 16 24-bits hexadecimal
numbers (without 0x prefix) separated by comas, for example @code{0d00ee,
ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1,
7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b}.
@item ifo_palette
Specify the IFO file from which the global palette is obtained.
(experimental)
@item forced_subs_only
Only decode subtitle entries marked as forced. Some titles have forced
and non-forced subtitles in the same track. Setting this flag to @code{1}
will only keep the forced subtitles. Default value is @code{0}.
@end table
@section libzvbi-teletext
Libzvbi allows libavcodec to decode DVB teletext pages and DVB teletext
subtitles. Requires the presence of the libzvbi headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libzvbi}.
@subsection Options
@table @option
@item txt_page
List of teletext page numbers to decode. 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.
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
@item txt_left
X offset of generated bitmaps, default is 0.
@item txt_top
Y offset of generated bitmaps, default is 0.
@item txt_chop_spaces
Chops leading and trailing spaces and removes empty lines from the generated
text. This option is useful for teletext based subtitles where empty spaces may
be present at the start or at the end of the lines or empty lines may be
present between the subtitle lines because of double-sized teletext characters.
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.
@item txt_transparent
Force transparent background of the generated teletext bitmaps. Default value
is 0 which means an opaque background.
@item txt_opacity
Sets the opacity (0-255) of the teletext background. If
@option{txt_transparent} is not set, it only affects characters between a start
box and an end box, typically subtitles. Default value is 0 if
@option{txt_transparent} is set, 255 otherwise.
@end table
@c man end SUBTILES DECODERS

0
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/doc/default.css → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/default.css
View file

665
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/demuxers.texi Normal file
View file

@ -0,0 +1,665 @@
@chapter Demuxers
@c man begin DEMUXERS
Demuxers are configured elements in FFmpeg that can read the
multimedia streams from a particular type of file.
When you configure your FFmpeg build, all the supported demuxers
are enabled by default. You can list all available ones using the
configure option @code{--list-demuxers}.
You can disable all the demuxers using the configure option
@code{--disable-demuxers}, and selectively enable a single demuxer with
the option @code{--enable-demuxer=@var{DEMUXER}}, or disable it
with the option @code{--disable-demuxer=@var{DEMUXER}}.
The option @code{-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 description of some of the currently available demuxers follows.
@section aa
Audible Format 2, 3, and 4 demuxer.
This demuxer is used to demux Audible Format 2, 3, and 4 (.aa) files.
@section applehttp
Apple HTTP Live Streaming demuxer.
This demuxer presents all AVStreams from all variant streams.
The id field is set to the bitrate variant index number. By setting
the discard flags on AVStreams (by pressing 'a' or 'v' in ffplay),
the caller can decide which variant streams to actually receive.
The total bitrate of the variant that the stream belongs to is
available in a metadata key named "variant_bitrate".
@section apng
Animated Portable Network Graphics demuxer.
This demuxer is used to demux APNG files.
All headers, but the PNG signature, up to (but not including) the first
fcTL chunk are transmitted as extradata.
Frames are then split as being all the chunks between two fcTL ones, or
between the last fcTL and IEND chunks.
@table @option
@item -ignore_loop @var{bool}
Ignore the loop variable in the file if set.
@item -max_fps @var{int}
Maximum framerate in frames per second (0 for no limit).
@item -default_fps @var{int}
Default framerate in frames per second when none is specified in the file
(0 meaning as fast as possible).
@end table
@section asf
Advanced Systems Format demuxer.
This demuxer is used to demux ASF files and MMS network streams.
@table @option
@item -no_resync_search @var{bool}
Do not try to resynchronize by looking for a certain optional start code.
@end table
@anchor{concat}
@section concat
Virtual concatenation script demuxer.
This demuxer reads a list of files and other directives from a text file and
demuxes them one after the other, as if all their packets had been muxed
together.
The timestamps in the files are adjusted so that the first file starts at 0
and each next file starts where the previous one finishes. Note that it is
done globally and may cause gaps if all streams do not have exactly the same
length.
All files must have the same streams (same codecs, same time base, etc.).
The duration of each file is used to adjust the timestamps of the next file:
if the duration is incorrect (because it was computed using the bit-rate or
because the file is truncated, for example), it can cause artifacts. The
@code{duration} directive can be used to override the duration stored in
each file.
@subsection Syntax
The script is a text file in extended-ASCII, with one directive per line.
Empty lines, leading spaces and lines starting with '#' are ignored. The
following directive is recognized:
@table @option
@item @code{file @var{path}}
Path to a file to read; special characters and spaces must be escaped with
backslash or single quotes.
All subsequent 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 make FFmpeg recognize the format automatically, this directive must
appear exactly as is (no extra space or byte-order-mark) on the very first
line of the script.
@item @code{duration @var{dur}}
Duration of the file. This information can be specified from the file;
specifying it here may be more efficient or help if the information from the
file is not available or accurate.
If the duration is set for all files, then it is possible to seek in the
whole concatenated video.
@item @code{inpoint @var{timestamp}}
In point of the file. When the demuxer opens the file it instantly seeks to the
specified timestamp. Seeking is done so that all streams can be presented
successfully at In point.
This directive works best with intra frame codecs, because for non-intra frame
ones you will usually get extra packets before the actual In point and the
decoded content will most likely contain frames before In point too.
For each file, packets before the file In point will have timestamps less than
the calculated start timestamp of the file (negative in case of the first
file), and the duration of the files (if not specified by the @code{duration}
directive) will be reduced based on their specified In point.
Because of potential packets before the specified In point, packet timestamps
may overlap between two concatenated files.
@item @code{outpoint @var{timestamp}}
Out point of the file. When the demuxer reaches the specified decoding
timestamp in any of the streams, it handles it as an end of file condition and
skips the current and all the remaining packets from all streams.
Out point is exclusive, which means that the demuxer will not output packets
with a decoding timestamp greater or equal to Out point.
This directive works best with intra frame codecs and formats where all streams
are tightly interleaved. For non-intra frame codecs you will usually get
additional packets with presentation timestamp after Out point therefore the
decoded content will most likely contain frames after Out point too. If your
streams are not tightly interleaved you may not get all the packets from all
streams before Out point and you may only will be able to decode the earliest
stream until Out point.
The duration of the files (if not specified by the @code{duration}
directive) will be reduced based on their specified Out point.
@item @code{file_packet_metadata @var{key=value}}
Metadata of the packets of the file. The specified metadata will be set for
each file packet. You can specify this directive multiple times to add multiple
metadata entries.
@item @code{stream}
Introduce a stream in the virtual file.
All subsequent stream-related directives apply to the last introduced
stream.
Some streams properties must be set in order to allow identifying the
matching streams in the subfiles.
If no streams are defined in the script, the streams from the first file are
copied.
@item @code{exact_stream_id @var{id}}
Set the id of the stream.
If this directive is given, the string with the corresponding id in the
subfiles will be used.
This is especially useful for MPEG-PS (VOB) files, where the order of the
streams is not reliable.
@end table
@subsection Options
This demuxer accepts the following option:
@table @option
@item safe
If set to 1, reject unsafe file paths. A file path is considered safe if it
does not contain a protocol specification and is relative and all components
only contain characters from the portable character set (letters, digits,
period, underscore and hyphen) and have no period at the beginning of a
component.
If set to 0, any file name is accepted.
The default is 1.
-1 is equivalent to 1 if the format was automatically
probed and 0 otherwise.
@item auto_convert
If set to 1, try to perform automatic conversions on packet data to make the
streams concatenable.
The default is 1.
Currently, the only conversion is adding the h264_mp4toannexb bitstream
filter to H.264 streams in MP4 format. This is necessary in particular if
there are resolution changes.
@item segment_time_metadata
If set to 1, every packet will contain the @var{lavf.concat.start_time} and the
@var{lavf.concat.duration} packet metadata values which are the start_time and
the duration of the respective file segments in the concatenated output
expressed in microseconds. The duration metadata is only set if it is known
based on the concat file.
The default is 0.
@end table
@subsection Examples
@itemize
@item
Use absolute filenames and include some comments:
@example
# my first filename
file /mnt/share/file-1.wav
# my second filename including whitespace
file '/mnt/share/file 2.wav'
# my third filename including whitespace plus single quote
file '/mnt/share/file 3'\''.wav'
@end example
@item
Allow for input format auto-probing, use safe filenames and set the duration of
the first file:
@example
ffconcat version 1.0
file file-1.wav
duration 20.0
file subdir/file-2.wav
@end example
@end itemize
@section 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
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
@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 gif
Animated GIF demuxer.
It accepts the following options:
@table @option
@item min_delay
Set the minimum valid delay between frames in hundredths of seconds.
Range is 0 to 6000. Default value is 2.
@item max_gif_delay
Set the maximum valid delay between frames in hundredth of seconds.
Range is 0 to 65535. Default value is 65535 (nearly eleven minutes),
the maximum value allowed by the specification.
@item default_delay
Set the default delay between frames in hundredths of seconds.
Range is 0 to 6000. Default value is 10.
@item ignore_loop
GIF files can contain information to loop a certain number of times (or
infinitely). If @option{ignore_loop} is set to 1, then the loop setting
from the input will be ignored and looping will not occur. If set to 0,
then looping will occur and will cycle the number of times according to
the GIF. Default value is 1.
@end table
For example, with the overlay filter, place an infinitely looping GIF
over another video:
@example
ffmpeg -i input.mp4 -ignore_loop 0 -i input.gif -filter_complex overlay=shortest=1 out.mkv
@end example
Note that in the above example the shortest option for overlay filter is
used to end the output video at the length of the shortest input file,
which in this case is @file{input.mp4} as the GIF in this example loops
infinitely.
@section 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.
This demuxer reads from a list of image files specified by a pattern.
The syntax and meaning of the pattern is specified by the
option @var{pattern_type}.
The pattern may contain a suffix which is used to automatically
determine the format of the images contained in the files.
The size, the pixel format, and the format of each image must be the
same for all the files in the sequence.
This demuxer accepts the following options:
@table @option
@item framerate
Set the frame rate for the video stream. It defaults to 25.
@item loop
If set to 1, loop over the input. Default value is 0.
@item pattern_type
Select the pattern type used to interpret the provided filename.
@var{pattern_type} accepts one of the following values.
@table @option
@item none
Disable pattern matching, therefore the video will only contain the specified
image. You should use this option if you do not want to create sequences from
multiple images and your filenames may contain special pattern characters.
@item sequence
Select a sequence pattern type, used to specify a sequence of files
indexed by sequential numbers.
A sequence pattern may contain the string "%d" or "%0@var{N}d", which
specifies the position of the characters representing a sequential
number in each filename matched by the pattern. If the form
"%d0@var{N}d" is used, the string representing the number in each
filename is 0-padded and @var{N} is the total number of 0-padded
digits representing the number. The literal character '%' can be
specified in the pattern with the string "%%".
If the sequence pattern contains "%d" or "%0@var{N}d", the first filename of
the file list specified by the pattern must contain a number
inclusively contained between @var{start_number} and
@var{start_number}+@var{start_number_range}-1, and all the following
numbers must be sequential.
For example the pattern "img-%03d.bmp" will match a sequence of
filenames of the form @file{img-001.bmp}, @file{img-002.bmp}, ...,
@file{img-010.bmp}, etc.; the pattern "i%%m%%g-%d.jpg" will match a
sequence of filenames of the form @file{i%m%g-1.jpg},
@file{i%m%g-2.jpg}, ..., @file{i%m%g-10.jpg}, etc.
Note that the pattern must not necessarily contain "%d" or
"%0@var{N}d", for example to convert a single image file
@file{img.jpeg} you can employ the command:
@example
ffmpeg -i img.jpeg img.png
@end example
@item glob
Select a glob wildcard pattern type.
The pattern is interpreted like a @code{glob()} pattern. This is only
selectable if libavformat was compiled with globbing support.
@item glob_sequence @emph{(deprecated, will be removed)}
Select a mixed glob wildcard/sequence pattern.
If your version of libavformat was compiled with globbing support, and
the provided pattern contains at least one glob meta character among
@code{%*?[]@{@}} that is preceded by an unescaped "%", the pattern is
interpreted like a @code{glob()} pattern, otherwise it is interpreted
like a sequence pattern.
All glob special characters @code{%*?[]@{@}} must be prefixed
with "%". To escape a literal "%" you shall use "%%".
For example the pattern @code{foo-%*.jpeg} will match all the
filenames prefixed by "foo-" and terminating with ".jpeg", and
@code{foo-%?%?%?.jpeg} will match all the filenames prefixed with
"foo-", followed by a sequence of three characters, and terminating
with ".jpeg".
This pattern type is deprecated in favor of @var{glob} and
@var{sequence}.
@end table
Default value is @var{glob_sequence}.
@item pixel_format
Set the pixel format of the images to read. If not specified the pixel
format is guessed from the first image file in the sequence.
@item start_number
Set the index of the file matched by the image file pattern to start
to read from. Default value is 0.
@item start_number_range
Set the index interval range to check when looking for the first image
file in the sequence, starting from @var{start_number}. Default value
is 5.
@item ts_from_file
If set to 1, will set frame timestamp to modification time of image file. Note
that monotonity of timestamps is not provided: images go in the same order as
without this option. Default value is 0.
If set to 2, will set frame timestamp to the modification time of the image file in
nanosecond precision.
@item video_size
Set the video size of the images to read. If not specified the video
size is guessed from the first image file in the sequence.
@end table
@subsection Examples
@itemize
@item
Use @command{ffmpeg} for creating a video from the images in the file
sequence @file{img-001.jpeg}, @file{img-002.jpeg}, ..., assuming an
input frame rate of 10 frames per second:
@example
ffmpeg -framerate 10 -i 'img-%03d.jpeg' out.mkv
@end example
@item
As above, but start by reading from a file with index 100 in the sequence:
@example
ffmpeg -framerate 10 -start_number 100 -i 'img-%03d.jpeg' out.mkv
@end example
@item
Read images matching the "*.png" glob pattern , that is all the files
terminating with the ".png" suffix:
@example
ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv
@end example
@end itemize
@section libgme
The Game Music Emu library is a collection of video game music file emulators.
See @url{http://code.google.com/p/game-music-emu/} for more information.
Some files have multiple tracks. The demuxer will pick the first track by
default. The @option{track_index} option can be used to select a different
track. Track indexes start at 0. The demuxer exports the number of tracks as
@var{tracks} meta data entry.
For very large files, the @option{max_size} option may have to be adjusted.
@section libopenmpt
libopenmpt based module demuxer
See @url{https://lib.openmpt.org/libopenmpt/} for more information.
Some files have multiple subsongs (tracks) this can be set with the @option{subsong}
option.
It accepts the following options:
@table @option
@item subsong
Set the subsong index. This can be either 'all', 'auto', or the index of the
subsong. Subsong indexes start at 0. The default is 'auto'.
The default value is to let libopenmpt choose.
@item layout
Set the channel layout. Valid values are 1, 2, and 4 channel layouts.
The default value is STEREO.
@item sample_rate
Set the sample rate for libopenmpt to output.
Range is from 1000 to INT_MAX. The value default is 48000.
@end table
@section mov/mp4/3gp/QuickTime
QuickTime / MP4 demuxer.
This demuxer accepts the following options:
@table @option
@item enable_drefs
Enable loading of external tracks, disabled by default.
Enabling this can theoretically leak information in some use cases.
@item use_absolute_path
Allows loading of external tracks via absolute paths, disabled by default.
Enabling this poses a security risk. It should only be enabled if the source
is known to be non malicious.
@end table
@section mpegts
MPEG-2 transport stream demuxer.
This demuxer accepts the following options:
@table @option
@item resync_size
Set size limit for looking up a new synchronization. Default value is
65536.
@item 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
not discarded. Default value is 1, set this option to 0 if you want your
teletext packet PTS and DTS values untouched.
@item ts_packetsize
Output option carrying the raw packet size in bytes.
Show the detected raw packet size, cannot be set by the user.
@item scan_all_pmts
Scan and combine all PMTs. The value is an integer with value from -1
to 1 (-1 means automatic setting, 1 means enabled, 0 means
disabled). Default value is -1.
@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
MJPEG encapsulated in multi-part MIME demuxer.
This demuxer allows reading of MJPEG, where each frame is represented as a part of
multipart/x-mixed-replace stream.
@table @option
@item strict_mime_boundary
Default implementation applies a relaxed standard to multi-part MIME boundary detection,
to prevent regression with numerous existing endpoints not generating a proper MIME
MJPEG stream. Turning this option on by setting it to 1 will result in a stricter check
of the boundary value.
@end table
@section rawvideo
Raw video demuxer.
This demuxer allows one to read raw video data. Since there is no header
specifying the assumed video parameters, the user must specify them
in order to be able to decode the data correctly.
This demuxer accepts the following options:
@table @option
@item framerate
Set input video frame rate. Default value is 25.
@item pixel_format
Set the input video pixel format. Default value is @code{yuv420p}.
@item video_size
Set the input video size. This value must be specified explicitly.
@end table
For example to read a rawvideo file @file{input.raw} with
@command{ffplay}, assuming a pixel format of @code{rgb24}, a video
size of @code{320x240}, and a frame rate of 10 images per second, use
the command:
@example
ffplay -f rawvideo -pixel_format rgb24 -video_size 320x240 -framerate 10 input.raw
@end example
@section sbg
SBaGen script demuxer.
This demuxer reads the script language used by SBaGen
@url{http://uazu.net/sbagen/} to generate binaural beats sessions. A SBG
script looks like that:
@example
-SE
a: 300-2.5/3 440+4.5/0
b: 300-2.5/0 440+4.5/3
off: -
NOW == a
+0:07:00 == b
+0:14:00 == a
+0:21:00 == b
+0:30:00 off
@end example
A SBG script can mix absolute and relative timestamps. If the script uses
either only absolute timestamps (including the script start time) or only
relative ones, then its layout is fixed, and the conversion is
straightforward. On the other hand, if the script mixes both kind of
timestamps, then the @var{NOW} reference for relative timestamps will be
taken from the current time of day at the time the script is read, and the
script layout will be frozen according to that reference. That means that if
the script is directly played, the actual times will match the absolute
timestamps up to the sound controller's clock accuracy, but if the user
somehow pauses the playback or seeks, all times will be shifted accordingly.
@section tedcaptions
JSON captions used for @url{http://www.ted.com/, TED Talks}.
TED does not provide links to the captions, but they can be guessed from the
page. The file @file{tools/bookmarklets.html} from the FFmpeg source tree
contains a bookmarklet to expose them.
This demuxer accepts the following option:
@table @option
@item start_time
Set the start time of the TED talk, in milliseconds. The default is 15000
(15s). It is used to sync the captions with the downloadable videos, because
they include a 15s intro.
@end table
Example: convert the captions to a format most players understand:
@example
ffmpeg -i http://www.ted.com/talks/subtitles/id/1/lang/en talk1-en.srt
@end example
@c man end DEMUXERS

261
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/doc/developer.texi → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/developer.texi
View file

@ -10,79 +10,41 @@
@contents
@chapter Introduction
@chapter Notes for external developers
This text is concerned with the development @emph{of} FFmpeg itself. Information
on using the FFmpeg libraries in other programs can be found elsewhere, e.g. in:
@itemize @bullet
@item
the installed header files
@item
@url{http://ffmpeg.org/doxygen/trunk/index.html, the Doxygen documentation}
generated from the headers
@item
the examples under @file{doc/examples}
@end itemize
This document is mostly useful for internal FFmpeg developers.
External developers who need to use the API in their application should
refer to the API doxygen documentation in the public headers, and
check the examples in @file{doc/examples} and in the source code to
see how the public API is employed.
If you modify FFmpeg code for your own use case, you are highly encouraged to
@emph{submit your changes back to us}, using this document as a guide. There are
both pragmatic and ideological reasons to do so:
@itemize @bullet
@item
Maintaining external changes to keep up with upstream development is
time-consuming and error-prone. With your code in the main tree, it will be
maintained by FFmpeg developers.
@item
FFmpeg developers include leading experts in the field who can find bugs or
design flaws in your code.
@item
By supporting the project you find useful you ensure it continues to be
maintained and developed.
@end itemize
You can use the FFmpeg libraries in your commercial program, but you
are encouraged to @emph{publish any patch you make}. In this case the
best way to proceed is to send your patches to the ffmpeg-devel
mailing list following the guidelines illustrated in the remainder of
this document.
For more detailed legal information about the use of FFmpeg in
external programs read the @file{LICENSE} file in the source tree and
consult @url{https://ffmpeg.org/legal.html}.
@section Contributing code
@chapter Contributing
All proposed code changes should be submitted for review to
@url{mailto:ffmpeg-devel@@ffmpeg.org, the development mailing list}, as
described in more detail in the @ref{Submitting patches} chapter. The code
should comply with the @ref{Development Policy} and follow the @ref{Coding Rules}.
There are 2 ways by which code gets into FFmpeg:
@itemize @bullet
@item Submitting patches to the ffmpeg-devel mailing list.
See @ref{Submitting patches} for details.
@item Directly committing changes to the main tree.
@end itemize
Whichever way, changes should be reviewed by the maintainer of the code
before they are committed. And they should follow the @ref{Coding Rules}.
The developer making the commit and the author are responsible for their changes
and should try to fix issues their commit causes.
@anchor{Coding Rules}
@chapter Coding Rules
@section C language features
FFmpeg is programmed in the ISO C99 language, extended with:
@itemize @bullet
@item
Atomic operations from C11 @file{stdatomic.h}. They are emulated on
architectures/compilers that do not support them, so all FFmpeg-internal code
may use atomics without any extra checks. However, @file{stdatomic.h} must not
be included in public headers, so they stay C99-compatible.
@end itemize
Compiler-specific extensions may be used with good reason, but must not be
depended on, i.e. the code must still compile and work with compilers lacking
the extension.
The following C99 features must not be used anywhere in the codebase:
@itemize @bullet
@item
variable-length arrays;
@item
complex numbers;
@item
mixed statements and declarations.
@end itemize
@section Code formatting conventions
There are the following guidelines regarding the indentation in files:
@ -105,39 +67,8 @@ K&R coding style is used.
@end itemize
The presentation is one inspired by 'indent -i4 -kr -nut'.
@subsection Vim configuration
In order to configure Vim to follow FFmpeg formatting conventions, paste
the following snippet into your @file{.vimrc}:
@example
" indentation rules for FFmpeg: 4 spaces, no tabs
set expandtab
set shiftwidth=4
set softtabstop=4
set cindent
set cinoptions=(0
" Allow tabs in Makefiles.
autocmd FileType make,automake set noexpandtab shiftwidth=8 softtabstop=8
" Trailing whitespace and tabs are forbidden, so highlight them.
highlight ForbiddenWhitespace ctermbg=red guibg=red
match ForbiddenWhitespace /\s\+$\|\t/
" Do not highlight spaces at the end of line while typing on that line.
autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@@<!$/
@end example
@subsection Emacs configuration
For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
@lisp
(c-add-style "ffmpeg"
'("k&r"
(c-basic-offset . 4)
(indent-tabs-mode . nil)
(show-trailing-whitespace . t)
(c-offsets-alist
(statement-cont . (c-lineup-assignments +)))
)
)
(setq c-default-style "ffmpeg")
@end lisp
The main priority in FFmpeg is simplicity and small code size in order to
minimize the bug count.
@section Comments
Use the JavaDoc/Doxygen format (see examples below) so that code documentation
@ -179,51 +110,89 @@ int myfunc(int my_parameter)
...
@end example
@section Naming conventions
@section C language features
Names of functions, variables, and struct members must be lowercase, using
underscores (_) to separate words. For example, @samp{avfilter_get_video_buffer}
is an acceptable function name and @samp{AVFilterGetVideo} is not.
FFmpeg is programmed in the ISO C90 language with a few additional
features from ISO C99, namely:
Struct, union, enum, and typedeffed type names must use CamelCase. All structs
and unions should be typedeffed to the same name as the struct/union tag, e.g.
@code{typedef struct AVFoo @{ ... @} AVFoo;}. Enums are typically not
typedeffed.
Enumeration constants and macros must be UPPERCASE, except for macros
masquerading as functions, which should use the function naming convention.
All identifiers in the libraries should be namespaced as follows:
@itemize @bullet
@item
No namespacing for identifiers with file and lower scope (e.g. local variables,
static functions), and struct and union members,
the @samp{inline} keyword;
@item
The @code{ff_} prefix must be used for variables and functions visible outside
of file scope, but only used internally within a single library, e.g.
@samp{ff_w64_demuxer}. This prevents name collisions when FFmpeg is statically
linked.
@samp{//} comments;
@item
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
accept patches to remove their use unless they absolutely do not impair
clarity and performance.
All code must compile with recent versions of GCC and a number of other
currently supported compilers. To ensure compatibility, please do not use
additional C99 features or GCC extensions. Especially watch out for:
@itemize @bullet
@item
mixing statements and declarations;
@item
@samp{long long} (use @samp{int64_t} instead);
@item
@samp{__attribute__} not protected by @samp{#ifdef __GNUC__} or similar;
@item
GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}).
@end itemize
@section Naming conventions
All names should be composed with underscores (_), not CamelCase. For example,
@samp{avfilter_get_video_buffer} is an acceptable function name and
@samp{AVFilterGetVideo} is not. The exception from this are type names, like
for example structs and enums; they should always be in CamelCase.
There are the following conventions for naming variables and functions:
@itemize @bullet
@item
For local variables no prefix is required.
@item
For file-scope variables and functions declared as @code{static}, no prefix
is required.
@item
For variables and functions visible outside of file scope, but only used
internally by a library, an @code{ff_} prefix should be used,
e.g. @samp{ff_w64_demuxer}.
@item
For variables and functions visible outside of file scope, used internally
across multiple libraries, use @code{avpriv_} as prefix, for example,
@samp{avpriv_report_missing_feature}.
@item
All other internal identifiers, like private type or macro names, should be
namespaced only to avoid possible internal conflicts. E.g. @code{H264_NAL_SPS}
vs. @code{HEVC_NAL_SPS}.
@item
Each library has its own prefix for public symbols, in addition to the
commonly used @code{av_} (@code{avformat_} for libavformat,
@code{avcodec_} for libavcodec, @code{swr_} for libswresample, etc).
Check the existing code and choose names accordingly.
@item
Other public identifiers (struct, union, enum, macro, type names) must use their
library's public prefix (@code{AV}, @code{Sws}, or @code{Swr}).
Note that some symbols without these prefixes are also exported for
retro-compatibility reasons. These exceptions are declared in the
@code{lib<name>/lib<name>.v} files.
@end itemize
Furthermore, name space reserved for the system should not be invaded.
@ -246,7 +215,39 @@ 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
@anchor{Development Policy}
@section Editor configuration
In order to configure Vim to follow FFmpeg formatting conventions, paste
the following snippet into your @file{.vimrc}:
@example
" indentation rules for FFmpeg: 4 spaces, no tabs
set expandtab
set shiftwidth=4
set softtabstop=4
set cindent
set cinoptions=(0
" Allow tabs in Makefiles.
autocmd FileType make,automake set noexpandtab shiftwidth=8 softtabstop=8
" Trailing whitespace and tabs are forbidden, so highlight them.
highlight ForbiddenWhitespace ctermbg=red guibg=red
match ForbiddenWhitespace /\s\+$\|\t/
" Do not highlight spaces at the end of line while typing on that line.
autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@@<!$/
@end example
For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
@lisp
(c-add-style "ffmpeg"
'("k&r"
(c-basic-offset . 4)
(indent-tabs-mode . nil)
(show-trailing-whitespace . t)
(c-offsets-alist
(statement-cont . (c-lineup-assignments +)))
)
)
(setq c-default-style "ffmpeg")
@end lisp
@chapter Development Policy
@section Patches/Committing
@ -490,22 +491,6 @@ 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.
@subheading Sending patches from email clients
Using @code{git send-email} might not be desirable for everyone. The
following trick allows to send patches via email clients in a safe
way. It has been tested with Outlook and Thunderbird (with X-Unsent
extension) and might work with other applications.
Create your patch like this:
@verbatim
git format-patch -s -o "outputfolder" --add-header "X-Unsent: 1" --suffix .eml --to ffmpeg-devel@ffmpeg.org -1 1a2b3c4d
@end verbatim
Now you'll just need to open the eml file with the email application
and execute 'Send'.
@subheading Reviews
Your patch will be reviewed on the mailing list. You will likely be asked
to make some changes and are expected to send in an improved version that
incorporates the requests from the review. This process may go through
@ -637,7 +622,7 @@ If the patch fixes a bug, did you provide a verbose analysis of the bug?
If the patch fixes a bug, did you provide enough information, including
a sample, so the bug can be reproduced and the fix can be verified?
Note please do not attach samples >100k to mails but rather provide a
URL, you can upload to @url{https://streams.videolan.org/upload/}.
URL, you can upload to ftp://upload.ffmpeg.org.
@item
Did you provide a verbose summary about what the patch does change?

0
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/doc/devices.texi → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/devices.texi
View file

9
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/doc/doxy-wrapper.sh → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/doxy-wrapper.sh
View file

@ -1,13 +1,10 @@
#!/bin/sh
OUT_DIR="${1}"
SRC_DIR="${2}"
DOXYFILE="${3}"
DOXYGEN="${4}"
DOXYFILE="${2}"
DOXYGEN="${3}"
shift 4
cd ${SRC_DIR}
shift 3
if [ -e "VERSION" ]; then
VERSION=`cat "VERSION"`

2864
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/encoders.texi Normal file
View file

File diff suppressed because it is too large Load diff

0
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/doc/errno.txt → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/errno.txt
View file

64
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/examples/Makefile Normal file
View file

@ -0,0 +1,64 @@
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
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)
EXAMPLE_MAKEFILE := $(SRC_PATH)/doc/examples/Makefile
EXAMPLES_FILES := $(wildcard $(SRC_PATH)/doc/examples/*.c) $(SRC_PATH)/doc/examples/README $(EXAMPLE_MAKEFILE)
$(foreach P,$(EXAMPLES),$(eval OBJS-$(P:%$(PROGSSUF)$(EXESUF)=%) = $(P:%$(PROGSSUF)$(EXESUF)=%).o))
$(EXAMPLES_G): %$(PROGSSUF)_g$(EXESUF): %.o
examples: $(EXAMPLES)
$(EXAMPLES:%$(PROGSSUF)$(EXESUF)=%.o): | doc/examples
OBJDIRS += doc/examples
DOXY_INPUT += $(EXAMPLES:%$(PROGSSUF)$(EXESUF)=%.c)
install: install-examples
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

50
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/examples/Makefile.example Normal file
View file

@ -0,0 +1,50 @@
# 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)

8
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/doc/examples/README → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/examples/README
View file

@ -7,10 +7,8 @@ that you have them installed and working on your system.
Method 1: build the installed examples in a generic read/write user directory
Copy to a read/write user directory and run:
make -f Makefile.example
It will link to the libraries on your system, assuming the PKG_CONFIG_PATH is
Copy to a read/write user directory and just use "make", it will link
to the libraries on your system, assuming the PKG_CONFIG_PATH is
correctly configured.
Method 2: build the examples in-tree
@ -22,4 +20,4 @@ examples using "make examplesclean"
If you want to try the dedicated Makefile examples (to emulate the first
method), go into doc/examples and run a command such as
PKG_CONFIG_PATH=pc-uninstalled make -f Makefile.example
PKG_CONFIG_PATH=pc-uninstalled make.

178
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/examples/avio_dir_cmd.c Normal file
View file

@ -0,0 +1,178 @@
/*
* Copyright (c) 2014 Lukasz Marek
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavformat/avio.h>
static const char *type_string(int type)
{
switch (type) {
case AVIO_ENTRY_DIRECTORY:
return "<DIR>";
case AVIO_ENTRY_FILE:
return "<FILE>";
case AVIO_ENTRY_BLOCK_DEVICE:
return "<BLOCK DEVICE>";
case AVIO_ENTRY_CHARACTER_DEVICE:
return "<CHARACTER DEVICE>";
case AVIO_ENTRY_NAMED_PIPE:
return "<PIPE>";
case AVIO_ENTRY_SYMBOLIC_LINK:
return "<LINK>";
case AVIO_ENTRY_SOCKET:
return "<SOCKET>";
case AVIO_ENTRY_SERVER:
return "<SERVER>";
case AVIO_ENTRY_SHARE:
return "<SHARE>";
case AVIO_ENTRY_WORKGROUP:
return "<WORKGROUP>";
case AVIO_ENTRY_UNKNOWN:
default:
break;
}
return "<UNKNOWN>";
}
static int list_op(const char *input_dir)
{
AVIODirEntry *entry = NULL;
AVIODirContext *ctx = NULL;
int cnt, ret;
char filemode[4], uid_and_gid[20];
if ((ret = avio_open_dir(&ctx, input_dir, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open directory: %s.\n", av_err2str(ret));
goto fail;
}
cnt = 0;
for (;;) {
if ((ret = avio_read_dir(ctx, &entry)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot list directory: %s.\n", av_err2str(ret));
goto fail;
}
if (!entry)
break;
if (entry->filemode == -1) {
snprintf(filemode, 4, "???");
} else {
snprintf(filemode, 4, "%3"PRIo64, entry->filemode);
}
snprintf(uid_and_gid, 20, "%"PRId64"(%"PRId64")", entry->user_id, entry->group_id);
if (cnt == 0)
av_log(NULL, AV_LOG_INFO, "%-9s %12s %30s %10s %s %16s %16s %16s\n",
"TYPE", "SIZE", "NAME", "UID(GID)", "UGO", "MODIFIED",
"ACCESSED", "STATUS_CHANGED");
av_log(NULL, AV_LOG_INFO, "%-9s %12"PRId64" %30s %10s %s %16"PRId64" %16"PRId64" %16"PRId64"\n",
type_string(entry->type),
entry->size,
entry->name,
uid_and_gid,
filemode,
entry->modification_timestamp,
entry->access_timestamp,
entry->status_change_timestamp);
avio_free_directory_entry(&entry);
cnt++;
};
fail:
avio_close_dir(&ctx);
return ret;
}
static int del_op(const char *url)
{
int ret = avpriv_io_delete(url);
if (ret < 0)
av_log(NULL, AV_LOG_ERROR, "Cannot delete '%s': %s.\n", url, av_err2str(ret));
return ret;
}
static int move_op(const char *src, const char *dst)
{
int ret = avpriv_io_move(src, dst);
if (ret < 0)
av_log(NULL, AV_LOG_ERROR, "Cannot move '%s' into '%s': %s.\n", src, dst, av_err2str(ret));
return ret;
}
static void usage(const char *program_name)
{
fprintf(stderr, "usage: %s OPERATION entry1 [entry2]\n"
"API example program to show how to manipulate resources "
"accessed through AVIOContext.\n"
"OPERATIONS:\n"
"list list content of the directory\n"
"move rename content in directory\n"
"del delete content in directory\n",
program_name);
}
int main(int argc, char *argv[])
{
const char *op = NULL;
int ret;
av_log_set_level(AV_LOG_DEBUG);
if (argc < 2) {
usage(argv[0]);
return 1;
}
avformat_network_init();
op = argv[1];
if (strcmp(op, "list") == 0) {
if (argc < 3) {
av_log(NULL, AV_LOG_INFO, "Missing argument for list operation.\n");
ret = AVERROR(EINVAL);
} else {
ret = list_op(argv[2]);
}
} else if (strcmp(op, "del") == 0) {
if (argc < 3) {
av_log(NULL, AV_LOG_INFO, "Missing argument for del operation.\n");
ret = AVERROR(EINVAL);
} else {
ret = del_op(argv[2]);
}
} else if (strcmp(op, "move") == 0) {
if (argc < 4) {
av_log(NULL, AV_LOG_INFO, "Missing argument for move operation.\n");
ret = AVERROR(EINVAL);
} else {
ret = move_op(argv[2], argv[3]);
}
} else {
av_log(NULL, AV_LOG_INFO, "Invalid operation %s\n", op);
ret = AVERROR(EINVAL);
}
avformat_network_deinit();
return ret < 0 ? 1 : 0;
}

133
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/examples/avio_reading.c Normal file
View file

@ -0,0 +1,133 @@
/*
* Copyright (c) 2014 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* libavformat AVIOContext API example.
*
* Make libavformat demuxer access media content through a custom
* AVIOContext read callback.
* @example avio_reading.c
*/
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavformat/avio.h>
#include <libavutil/file.h>
struct buffer_data {
uint8_t *ptr;
size_t size; ///< size left in the buffer
};
static int read_packet(void *opaque, uint8_t *buf, int buf_size)
{
struct buffer_data *bd = (struct buffer_data *)opaque;
buf_size = FFMIN(buf_size, bd->size);
if (!buf_size)
return AVERROR_EOF;
printf("ptr:%p size:%zu\n", bd->ptr, bd->size);
/* copy internal buffer data to buf */
memcpy(buf, bd->ptr, buf_size);
bd->ptr += buf_size;
bd->size -= buf_size;
return buf_size;
}
int main(int argc, char *argv[])
{
AVFormatContext *fmt_ctx = NULL;
AVIOContext *avio_ctx = NULL;
uint8_t *buffer = NULL, *avio_ctx_buffer = NULL;
size_t buffer_size, avio_ctx_buffer_size = 4096;
char *input_filename = NULL;
int ret = 0;
struct buffer_data bd = { 0 };
if (argc != 2) {
fprintf(stderr, "usage: %s input_file\n"
"API example program to show how to read from a custom buffer "
"accessed through AVIOContext.\n", argv[0]);
return 1;
}
input_filename = argv[1];
/* slurp file content into buffer */
ret = av_file_map(input_filename, &buffer, &buffer_size, 0, NULL);
if (ret < 0)
goto end;
/* fill opaque structure used by the AVIOContext read callback */
bd.ptr = buffer;
bd.size = buffer_size;
if (!(fmt_ctx = avformat_alloc_context())) {
ret = AVERROR(ENOMEM);
goto end;
}
avio_ctx_buffer = av_malloc(avio_ctx_buffer_size);
if (!avio_ctx_buffer) {
ret = AVERROR(ENOMEM);
goto end;
}
avio_ctx = avio_alloc_context(avio_ctx_buffer, avio_ctx_buffer_size,
0, &bd, &read_packet, NULL, NULL);
if (!avio_ctx) {
ret = AVERROR(ENOMEM);
goto end;
}
fmt_ctx->pb = avio_ctx;
ret = avformat_open_input(&fmt_ctx, NULL, NULL, NULL);
if (ret < 0) {
fprintf(stderr, "Could not open input\n");
goto end;
}
ret = avformat_find_stream_info(fmt_ctx, NULL);
if (ret < 0) {
fprintf(stderr, "Could not find stream information\n");
goto end;
}
av_dump_format(fmt_ctx, 0, input_filename, 0);
end:
avformat_close_input(&fmt_ctx);
/* note: the internal buffer could have changed, and be != avio_ctx_buffer */
if (avio_ctx) {
av_freep(&avio_ctx->buffer);
av_freep(&avio_ctx);
}
av_file_unmap(buffer, buffer_size);
if (ret < 0) {
fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
return 1;
}
return 0;
}

184
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/examples/decode_audio.c Normal file
View file

@ -0,0 +1,184 @@
/*
* 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;
}

28
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/doc/examples/decode_video.c → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/examples/decode_video.c
View file

@ -21,11 +21,10 @@
*/
/**
* @file libavcodec video decoding API usage example
* @example decode_video.c *
* @file
* video decoding with libavcodec API example
*
* Read from an MPEG1 video file, decode frames, and generate PGM images as
* output.
* @example decode_video.c
*/
#include <stdio.h>
@ -42,7 +41,7 @@ static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
FILE *f;
int i;
f = fopen(filename,"wb");
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);
@ -70,12 +69,12 @@ static void decode(AVCodecContext *dec_ctx, AVFrame *frame, AVPacket *pkt,
exit(1);
}
printf("saving frame %3"PRId64"\n", dec_ctx->frame_num);
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-%"PRId64, filename, dec_ctx->frame_num);
snprintf(buf, sizeof(buf), "%s-%d", filename, dec_ctx->frame_number);
pgm_save(frame->data[0], frame->linesize[0],
frame->width, frame->height, buf);
}
@ -93,12 +92,10 @@ int main(int argc, char **argv)
uint8_t *data;
size_t data_size;
int ret;
int eof;
AVPacket *pkt;
if (argc <= 2) {
fprintf(stderr, "Usage: %s <input file> <output file>\n"
"And check your input file is encoded by mpeg1video please.\n", argv[0]);
fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
exit(0);
}
filename = argv[1];
@ -152,16 +149,15 @@ int main(int argc, char **argv)
exit(1);
}
do {
while (!feof(f)) {
/* read raw data from the input file */
data_size = fread(inbuf, 1, INBUF_SIZE, f);
if (ferror(f))
if (!data_size)
break;
eof = !data_size;
/* use the parser to split the data into frames */
data = inbuf;
while (data_size > 0 || eof) {
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) {
@ -173,10 +169,8 @@ int main(int argc, char **argv)
if (pkt->size)
decode(c, frame, pkt, outfilename);
else if (eof)
break;
}
} while (!eof);
}
/* flush the decoder */
decode(c, frame, NULL, outfilename);

390
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/examples/demuxing_decoding.c Normal file
View file

@ -0,0 +1,390 @@
/*
* Copyright (c) 2012 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* Demuxing and decoding example.
*
* Show how to use the libavformat and libavcodec API to demux and
* decode audio and video data.
* @example demuxing_decoding.c
*/
#include <libavutil/imgutils.h>
#include <libavutil/samplefmt.h>
#include <libavutil/timestamp.h>
#include <libavformat/avformat.h>
static AVFormatContext *fmt_ctx = NULL;
static AVCodecContext *video_dec_ctx = NULL, *audio_dec_ctx;
static int width, height;
static enum AVPixelFormat pix_fmt;
static AVStream *video_stream = NULL, *audio_stream = NULL;
static const char *src_filename = NULL;
static const char *video_dst_filename = NULL;
static const char *audio_dst_filename = NULL;
static FILE *video_dst_file = NULL;
static FILE *audio_dst_file = NULL;
static uint8_t *video_dst_data[4] = {NULL};
static int video_dst_linesize[4];
static int video_dst_bufsize;
static int video_stream_idx = -1, audio_stream_idx = -1;
static AVFrame *frame = NULL;
static AVPacket pkt;
static int video_frame_count = 0;
static int audio_frame_count = 0;
/* Enable or disable frame reference counting. You are not supposed to support
* both paths in your application but pick the one most appropriate to your
* needs. Look for the use of refcount in this example to see what are the
* differences of API usage between them. */
static int refcount = 0;
static int decode_packet(int *got_frame, int cached)
{
int ret = 0;
int decoded = pkt.size;
*got_frame = 0;
if (pkt.stream_index == video_stream_idx) {
/* decode video frame */
ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
if (ret < 0) {
fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret));
return ret;
}
if (*got_frame) {
if (frame->width != width || frame->height != height ||
frame->format != pix_fmt) {
/* To handle this change, one could call av_image_alloc again and
* decode the following frames into another rawvideo file. */
fprintf(stderr, "Error: Width, height and pixel format have to be "
"constant in a rawvideo file, but the width, height or "
"pixel format of the input video changed:\n"
"old: width = %d, height = %d, format = %s\n"
"new: width = %d, height = %d, format = %s\n",
width, height, av_get_pix_fmt_name(pix_fmt),
frame->width, frame->height,
av_get_pix_fmt_name(frame->format));
return -1;
}
printf("video_frame%s n:%d coded_n:%d\n",
cached ? "(cached)" : "",
video_frame_count++, frame->coded_picture_number);
/* copy decoded frame to destination buffer:
* this is required since rawvideo expects non aligned data */
av_image_copy(video_dst_data, video_dst_linesize,
(const uint8_t **)(frame->data), frame->linesize,
pix_fmt, width, height);
/* write to rawvideo file */
fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file);
}
} else if (pkt.stream_index == audio_stream_idx) {
/* decode audio frame */
ret = avcodec_decode_audio4(audio_dec_ctx, frame, got_frame, &pkt);
if (ret < 0) {
fprintf(stderr, "Error decoding audio frame (%s)\n", av_err2str(ret));
return ret;
}
/* Some audio decoders decode only part of the packet, and have to be
* called again with the remainder of the packet data.
* Sample: fate-suite/lossless-audio/luckynight-partial.shn
* Also, some decoders might over-read the packet. */
decoded = FFMIN(ret, pkt.size);
if (*got_frame) {
size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format);
printf("audio_frame%s n:%d nb_samples:%d pts:%s\n",
cached ? "(cached)" : "",
audio_frame_count++, frame->nb_samples,
av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));
/* Write the raw audio data samples of the first plane. This works
* fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However,
* most audio decoders output planar audio, which uses a separate
* plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P).
* In other words, this code will write only the first audio channel
* in these cases.
* You should use libswresample or libavfilter to convert the frame
* to packed data. */
fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file);
}
}
/* If we use frame reference counting, we own the data and need
* to de-reference it when we don't use it anymore */
if (*got_frame && refcount)
av_frame_unref(frame);
return decoded;
}
static int open_codec_context(int *stream_idx,
AVCodecContext **dec_ctx, AVFormatContext *fmt_ctx, enum AVMediaType type)
{
int ret, stream_index;
AVStream *st;
AVCodec *dec = NULL;
AVDictionary *opts = NULL;
ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
if (ret < 0) {
fprintf(stderr, "Could not find %s stream in input file '%s'\n",
av_get_media_type_string(type), src_filename);
return ret;
} else {
stream_index = ret;
st = fmt_ctx->streams[stream_index];
/* find decoder for the stream */
dec = avcodec_find_decoder(st->codecpar->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) {
fprintf(stderr, "Failed to open %s codec\n",
av_get_media_type_string(type));
return ret;
}
*stream_idx = stream_index;
}
return 0;
}
static int get_format_from_sample_fmt(const char **fmt,
enum AVSampleFormat sample_fmt)
{
int i;
struct sample_fmt_entry {
enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
} sample_fmt_entries[] = {
{ AV_SAMPLE_FMT_U8, "u8", "u8" },
{ AV_SAMPLE_FMT_S16, "s16be", "s16le" },
{ AV_SAMPLE_FMT_S32, "s32be", "s32le" },
{ AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
{ AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
};
*fmt = NULL;
for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
struct sample_fmt_entry *entry = &sample_fmt_entries[i];
if (sample_fmt == entry->sample_fmt) {
*fmt = AV_NE(entry->fmt_be, entry->fmt_le);
return 0;
}
}
fprintf(stderr,
"sample format %s is not supported as output format\n",
av_get_sample_fmt_name(sample_fmt));
return -1;
}
int main (int argc, char **argv)
{
int ret = 0, got_frame;
if (argc != 4 && argc != 5) {
fprintf(stderr, "usage: %s [-refcount] input_file video_output_file audio_output_file\n"
"API example program to show how to read frames from an input file.\n"
"This program reads frames from a file, decodes them, and writes decoded\n"
"video frames to a rawvideo file named video_output_file, and decoded\n"
"audio frames to a rawaudio file named audio_output_file.\n\n"
"If the -refcount option is specified, the program use the\n"
"reference counting frame system which allows keeping a copy of\n"
"the data for longer than one decode call.\n"
"\n", argv[0]);
exit(1);
}
if (argc == 5 && !strcmp(argv[1], "-refcount")) {
refcount = 1;
argv++;
}
src_filename = argv[1];
video_dst_filename = argv[2];
audio_dst_filename = argv[3];
/* open input file, and allocate format context */
if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
fprintf(stderr, "Could not open source file %s\n", src_filename);
exit(1);
}
/* retrieve stream information */
if (avformat_find_stream_info(fmt_ctx, NULL) < 0) {
fprintf(stderr, "Could not find stream information\n");
exit(1);
}
if (open_codec_context(&video_stream_idx, &video_dec_ctx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) {
video_stream = fmt_ctx->streams[video_stream_idx];
video_dst_file = fopen(video_dst_filename, "wb");
if (!video_dst_file) {
fprintf(stderr, "Could not open destination file %s\n", video_dst_filename);
ret = 1;
goto end;
}
/* allocate image where the decoded image will be put */
width = video_dec_ctx->width;
height = video_dec_ctx->height;
pix_fmt = video_dec_ctx->pix_fmt;
ret = av_image_alloc(video_dst_data, video_dst_linesize,
width, height, pix_fmt, 1);
if (ret < 0) {
fprintf(stderr, "Could not allocate raw video buffer\n");
goto end;
}
video_dst_bufsize = ret;
}
if (open_codec_context(&audio_stream_idx, &audio_dec_ctx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) {
audio_stream = fmt_ctx->streams[audio_stream_idx];
audio_dst_file = fopen(audio_dst_filename, "wb");
if (!audio_dst_file) {
fprintf(stderr, "Could not open destination file %s\n", audio_dst_filename);
ret = 1;
goto end;
}
}
/* dump input information to stderr */
av_dump_format(fmt_ctx, 0, src_filename, 0);
if (!audio_stream && !video_stream) {
fprintf(stderr, "Could not find audio or video stream in the input, aborting\n");
ret = 1;
goto end;
}
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate frame\n");
ret = AVERROR(ENOMEM);
goto end;
}
/* initialize packet, set data to NULL, let the demuxer fill it */
av_init_packet(&pkt);
pkt.data = NULL;
pkt.size = 0;
if (video_stream)
printf("Demuxing video from file '%s' into '%s'\n", src_filename, video_dst_filename);
if (audio_stream)
printf("Demuxing audio from file '%s' into '%s'\n", src_filename, audio_dst_filename);
/* read frames from the file */
while (av_read_frame(fmt_ctx, &pkt) >= 0) {
AVPacket orig_pkt = pkt;
do {
ret = decode_packet(&got_frame, 0);
if (ret < 0)
break;
pkt.data += ret;
pkt.size -= ret;
} while (pkt.size > 0);
av_packet_unref(&orig_pkt);
}
/* flush cached frames */
pkt.data = NULL;
pkt.size = 0;
do {
decode_packet(&got_frame, 1);
} while (got_frame);
printf("Demuxing succeeded.\n");
if (video_stream) {
printf("Play the output video file with the command:\n"
"ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
av_get_pix_fmt_name(pix_fmt), width, height,
video_dst_filename);
}
if (audio_stream) {
enum AVSampleFormat sfmt = audio_dec_ctx->sample_fmt;
int n_channels = audio_dec_ctx->channels;
const char *fmt;
if (av_sample_fmt_is_planar(sfmt)) {
const char *packed = av_get_sample_fmt_name(sfmt);
printf("Warning: the sample format the decoder produced is planar "
"(%s). This example will output the first channel only.\n",
packed ? packed : "?");
sfmt = av_get_packed_sample_fmt(sfmt);
n_channels = 1;
}
if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
goto end;
printf("Play the output audio file with the command:\n"
"ffplay -f %s -ac %d -ar %d %s\n",
fmt, n_channels, audio_dec_ctx->sample_rate,
audio_dst_filename);
}
end:
avcodec_free_context(&video_dec_ctx);
avcodec_free_context(&audio_dec_ctx);
avformat_close_input(&fmt_ctx);
if (video_dst_file)
fclose(video_dst_file);
if (audio_dst_file)
fclose(audio_dst_file);
av_frame_free(&frame);
av_free(video_dst_data[0]);
return ret < 0;
}

36
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/doc/examples/encode_audio.c → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/examples/encode_audio.c
View file

@ -21,10 +21,10 @@
*/
/**
* @file libavcodec encoding audio API usage examples
* @example encode_audio.c
* @file
* audio encoding with libavcodec API example.
*
* Generate a synthetic audio signal and encode it to an output MP2 file.
* @example encode_audio.c
*/
#include <stdint.h>
@ -70,25 +70,26 @@ static int select_sample_rate(const AVCodec *codec)
}
/* select layout with the highest channel count */
static int select_channel_layout(const AVCodec *codec, AVChannelLayout *dst)
static int select_channel_layout(const AVCodec *codec)
{
const AVChannelLayout *p, *best_ch_layout;
const uint64_t *p;
uint64_t best_ch_layout = 0;
int best_nb_channels = 0;
if (!codec->ch_layouts)
return av_channel_layout_copy(dst, &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO);
if (!codec->channel_layouts)
return AV_CH_LAYOUT_STEREO;
p = codec->ch_layouts;
while (p->nb_channels) {
int nb_channels = p->nb_channels;
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_ch_layout = *p;
best_nb_channels = nb_channels;
}
p++;
}
return av_channel_layout_copy(dst, best_ch_layout);
return best_ch_layout;
}
static void encode(AVCodecContext *ctx, AVFrame *frame, AVPacket *pkt,
@ -163,9 +164,8 @@ int main(int argc, char **argv)
/* select other audio parameters supported by the encoder */
c->sample_rate = select_sample_rate(codec);
ret = select_channel_layout(codec, &c->ch_layout);
if (ret < 0)
exit(1);
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) {
@ -195,9 +195,7 @@ int main(int argc, char **argv)
frame->nb_samples = c->frame_size;
frame->format = c->sample_fmt;
ret = av_channel_layout_copy(&frame->ch_layout, &c->ch_layout);
if (ret < 0)
exit(1);
frame->channel_layout = c->channel_layout;
/* allocate the data buffers */
ret = av_frame_get_buffer(frame, 0);
@ -220,7 +218,7 @@ int main(int argc, char **argv)
for (j = 0; j < c->frame_size; j++) {
samples[2*j] = (int)(sin(t) * 10000);
for (k = 1; k < c->ch_layout.nb_channels; k++)
for (k = 1; k < c->channels; k++)
samples[2*j + k] = samples[2*j];
t += tincr;
}

35
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/doc/examples/encode_video.c → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/examples/encode_video.c
View file

@ -21,10 +21,10 @@
*/
/**
* @file libavcodec encoding video API usage example
* @example encode_video.c
* @file
* video encoding with libavcodec API example
*
* Generate synthetic video data and encode it to an output file.
* @example encode_video.c
*/
#include <stdio.h>
@ -145,7 +145,7 @@ int main(int argc, char **argv)
frame->width = c->width;
frame->height = c->height;
ret = av_frame_get_buffer(frame, 0);
ret = av_frame_get_buffer(frame, 32);
if (ret < 0) {
fprintf(stderr, "Could not allocate the video frame data\n");
exit(1);
@ -155,25 +155,12 @@ int main(int argc, char **argv)
for (i = 0; i < 25; i++) {
fflush(stdout);
/* Make sure the frame data is writable.
On the first round, the frame is fresh from av_frame_get_buffer()
and therefore we know it is writable.
But on the next rounds, encode() will have called
avcodec_send_frame(), and the codec may have kept a reference to
the frame in its internal structures, that makes the frame
unwritable.
av_frame_make_writable() checks that and allocates a new buffer
for the frame only if necessary.
*/
/* make sure the frame data is writable */
ret = av_frame_make_writable(frame);
if (ret < 0)
exit(1);
/* Prepare a dummy image.
In real code, this is where you would have your own logic for
filling the frame. FFmpeg does not care what you put in the
frame.
*/
/* prepare a dummy image */
/* Y */
for (y = 0; y < c->height; y++) {
for (x = 0; x < c->width; x++) {
@ -198,14 +185,8 @@ int main(int argc, char **argv)
/* flush the encoder */
encode(c, NULL, pkt, f);
/* Add sequence end code to have a real MPEG file.
It makes only sense because this tiny examples writes packets
directly. This is called "elementary stream" and only works for some
codecs. To create a valid file, you usually need to write packets
into a proper file format or protocol; see mux.c.
*/
if (codec->id == AV_CODEC_ID_MPEG1VIDEO || codec->id == AV_CODEC_ID_MPEG2VIDEO)
fwrite(endcode, 1, sizeof(endcode), f);
/* add sequence end code to have a real MPEG file */
fwrite(endcode, 1, sizeof(endcode), f);
fclose(f);
avcodec_free_context(&c);

40
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/doc/examples/extract_mvs.c → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/examples/extract_mvs.c
View file

@ -21,16 +21,7 @@
* THE SOFTWARE.
*/
/**
* @file libavcodec motion vectors extraction API usage example
* @example extract_mvs.c
*
* Read from input file, decode video stream and print a motion vectors
* representation to stdout.
*/
#include <libavutil/motion_vector.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
static AVFormatContext *fmt_ctx = NULL;
@ -69,11 +60,10 @@ static int decode_packet(const AVPacket *pkt)
const AVMotionVector *mvs = (const AVMotionVector *)sd->data;
for (i = 0; i < sd->size / sizeof(*mvs); i++) {
const AVMotionVector *mv = &mvs[i];
printf("%d,%2d,%2d,%2d,%4d,%4d,%4d,%4d,0x%"PRIx64",%4d,%4d,%4d\n",
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,
mv->motion_x, mv->motion_y, mv->motion_scale);
mv->dst_x, mv->dst_y, mv->flags);
}
}
av_frame_unref(frame);
@ -88,7 +78,7 @@ static int open_codec_context(AVFormatContext *fmt_ctx, enum AVMediaType type)
int ret;
AVStream *st;
AVCodecContext *dec_ctx = NULL;
const AVCodec *dec = NULL;
AVCodec *dec = NULL;
AVDictionary *opts = NULL;
ret = av_find_best_stream(fmt_ctx, type, -1, -1, &dec, 0);
@ -114,9 +104,7 @@ static int open_codec_context(AVFormatContext *fmt_ctx, enum AVMediaType type)
/* Init the video decoder */
av_dict_set(&opts, "flags2", "+export_mvs", 0);
ret = avcodec_open2(dec_ctx, dec, &opts);
av_dict_free(&opts);
if (ret < 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;
@ -133,7 +121,7 @@ static int open_codec_context(AVFormatContext *fmt_ctx, enum AVMediaType type)
int main(int argc, char **argv)
{
int ret = 0;
AVPacket *pkt = NULL;
AVPacket pkt = { 0 };
if (argc != 2) {
fprintf(stderr, "Usage: %s <video>\n", argv[0]);
@ -168,20 +156,13 @@ int main(int argc, char **argv)
goto end;
}
pkt = av_packet_alloc();
if (!pkt) {
fprintf(stderr, "Could not allocate AVPacket\n");
ret = AVERROR(ENOMEM);
goto end;
}
printf("framenum,source,blockw,blockh,srcx,srcy,dstx,dsty,flags,motion_x,motion_y,motion_scale\n");
printf("framenum,source,blockw,blockh,srcx,srcy,dstx,dsty,flags\n");
/* 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);
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;
}
@ -193,6 +174,5 @@ end:
avcodec_free_context(&video_dec_ctx);
avformat_close_input(&fmt_ctx);
av_frame_free(&frame);
av_packet_free(&pkt);
return ret < 0;
}

23
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/doc/examples/filter_audio.c → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/examples/filter_audio.c
View file

@ -19,11 +19,13 @@
*/
/**
* @file libavfilter audio filtering API usage example
* @example filter_audio.c
* @file
* libavfilter API usage example.
*
* This example will generate a sine wave audio, pass it through a simple filter
* chain, and then compute the MD5 checksum of the output data.
* @example filter_audio.c
* This example will generate a sine wave audio,
* pass it through a simple filter chain, and then compute the MD5 checksum of
* the output data.
*
* The filter chain it uses is:
* (input) -> abuffer -> volume -> aformat -> abuffersink -> (output)
@ -53,7 +55,7 @@
#define INPUT_SAMPLERATE 48000
#define INPUT_FORMAT AV_SAMPLE_FMT_FLTP
#define INPUT_CHANNEL_LAYOUT (AVChannelLayout)AV_CHANNEL_LAYOUT_5POINT0
#define INPUT_CHANNEL_LAYOUT AV_CH_LAYOUT_5POINT0
#define VOLUME_VAL 0.90
@ -98,7 +100,7 @@ static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
}
/* Set the filter options through the AVOptions API. */
av_channel_layout_describe(&INPUT_CHANNEL_LAYOUT, ch_layout, sizeof(ch_layout));
av_get_channel_layout_string(ch_layout, sizeof(ch_layout), 0, INPUT_CHANNEL_LAYOUT);
av_opt_set (abuffer_ctx, "channel_layout", ch_layout, AV_OPT_SEARCH_CHILDREN);
av_opt_set (abuffer_ctx, "sample_fmt", av_get_sample_fmt_name(INPUT_FORMAT), AV_OPT_SEARCH_CHILDREN);
av_opt_set_q (abuffer_ctx, "time_base", (AVRational){ 1, INPUT_SAMPLERATE }, AV_OPT_SEARCH_CHILDREN);
@ -152,8 +154,9 @@ static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
/* A third way of passing the options is in a string of the form
* key1=value1:key2=value2.... */
snprintf(options_str, sizeof(options_str),
"sample_fmts=%s:sample_rates=%d:channel_layouts=stereo",
av_get_sample_fmt_name(AV_SAMPLE_FMT_S16), 44100);
"sample_fmts=%s:sample_rates=%d:channel_layouts=0x%"PRIx64,
av_get_sample_fmt_name(AV_SAMPLE_FMT_S16), 44100,
(uint64_t)AV_CH_LAYOUT_STEREO);
err = avfilter_init_str(aformat_ctx, options_str);
if (err < 0) {
av_log(NULL, AV_LOG_ERROR, "Could not initialize the aformat filter.\n");
@ -212,7 +215,7 @@ static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
static int process_output(struct AVMD5 *md5, AVFrame *frame)
{
int planar = av_sample_fmt_is_planar(frame->format);
int channels = frame->ch_layout.nb_channels;
int channels = av_get_channel_layout_nb_channels(frame->channel_layout);
int planes = planar ? channels : 1;
int bps = av_get_bytes_per_sample(frame->format);
int plane_size = bps * frame->nb_samples * (planar ? 1 : channels);
@ -245,7 +248,7 @@ static int get_input(AVFrame *frame, int frame_num)
/* Set up the frame properties and allocate the buffer for the data. */
frame->sample_rate = INPUT_SAMPLERATE;
frame->format = INPUT_FORMAT;
av_channel_layout_copy(&frame->ch_layout, &INPUT_CHANNEL_LAYOUT);
frame->channel_layout = INPUT_CHANNEL_LAYOUT;
frame->nb_samples = FRAME_SIZE;
frame->pts = frame_num * FRAME_SIZE;

292
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/examples/filtering_audio.c Normal file
View file

@ -0,0 +1,292 @@
/*
* Copyright (c) 2010 Nicolas George
* Copyright (c) 2011 Stefano Sabatini
* Copyright (c) 2012 Clément Bœsch
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* API example for audio decoding and filtering
* @example filtering_audio.c
*/
#include <unistd.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
static const char *filter_descr = "aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono";
static const char *player = "ffplay -f s16le -ar 8000 -ac 1 -";
static AVFormatContext *fmt_ctx;
static AVCodecContext *dec_ctx;
AVFilterContext *buffersink_ctx;
AVFilterContext *buffersrc_ctx;
AVFilterGraph *filter_graph;
static int audio_stream_index = -1;
static int open_input_file(const char *filename)
{
int ret;
AVCodec *dec;
if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
return ret;
}
if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
/* select the audio stream */
ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find an 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);
/* init the audio decoder */
if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n");
return ret;
}
return 0;
}
static int init_filters(const char *filters_descr)
{
char args[512];
int ret = 0;
const AVFilter *abuffersrc = avfilter_get_by_name("abuffer");
const AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
static const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 };
static const int out_sample_rates[] = { 8000, -1 };
const AVFilterLink *outlink;
AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;
filter_graph = avfilter_graph_alloc();
if (!outputs || !inputs || !filter_graph) {
ret = AVERROR(ENOMEM);
goto end;
}
/* buffer audio source: the decoded frames from the decoder will be inserted here. */
if (!dec_ctx->channel_layout)
dec_ctx->channel_layout = av_get_default_channel_layout(dec_ctx->channels);
snprintf(args, sizeof(args),
"time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
time_base.num, time_base.den, dec_ctx->sample_rate,
av_get_sample_fmt_name(dec_ctx->sample_fmt), dec_ctx->channel_layout);
ret = avfilter_graph_create_filter(&buffersrc_ctx, abuffersrc, "in",
args, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
goto end;
}
/* buffer audio sink: to terminate the filter chain. */
ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out",
NULL, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
goto end;
}
ret = av_opt_set_int_list(buffersink_ctx, "sample_fmts", out_sample_fmts, -1,
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
goto end;
}
ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts", out_channel_layouts, -1,
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
goto end;
}
ret = av_opt_set_int_list(buffersink_ctx, "sample_rates", out_sample_rates, -1,
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
goto end;
}
/*
* Set the endpoints for the filter graph. The filter_graph will
* be linked to the graph described by filters_descr.
*/
/*
* The buffer source output must be connected to the input pad of
* the first filter described by filters_descr; since the first
* filter input label is not specified, it is set to "in" by
* default.
*/
outputs->name = av_strdup("in");
outputs->filter_ctx = buffersrc_ctx;
outputs->pad_idx = 0;
outputs->next = NULL;
/*
* The buffer sink input must be connected to the output pad of
* the last filter described by filters_descr; since the last
* filter output label is not specified, it is set to "out" by
* default.
*/
inputs->name = av_strdup("out");
inputs->filter_ctx = buffersink_ctx;
inputs->pad_idx = 0;
inputs->next = NULL;
if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
&inputs, &outputs, NULL)) < 0)
goto end;
if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
goto end;
/* Print summary of the sink buffer
* Note: args buffer is reused to store channel layout string */
outlink = buffersink_ctx->inputs[0];
av_get_channel_layout_string(args, sizeof(args), -1, outlink->channel_layout);
av_log(NULL, AV_LOG_INFO, "Output: srate:%dHz fmt:%s chlayout:%s\n",
(int)outlink->sample_rate,
(char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
args);
end:
avfilter_inout_free(&inputs);
avfilter_inout_free(&outputs);
return ret;
}
static void print_frame(const AVFrame *frame)
{
const int n = frame->nb_samples * av_get_channel_layout_nb_channels(frame->channel_layout);
const uint16_t *p = (uint16_t*)frame->data[0];
const uint16_t *p_end = p + n;
while (p < p_end) {
fputc(*p & 0xff, stdout);
fputc(*p>>8 & 0xff, stdout);
p++;
}
fflush(stdout);
}
int main(int argc, char **argv)
{
int ret;
AVPacket packet;
AVFrame *frame = av_frame_alloc();
AVFrame *filt_frame = av_frame_alloc();
if (!frame || !filt_frame) {
perror("Could not allocate frame");
exit(1);
}
if (argc != 2) {
fprintf(stderr, "Usage: %s file | %s\n", argv[0], player);
exit(1);
}
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if ((ret = init_filters(filter_descr)) < 0)
goto end;
/* read all packets */
while (1) {
if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
break;
if (packet.stream_index == audio_stream_index) {
ret = avcodec_send_packet(dec_ctx, &packet);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while sending a packet to the decoder\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;
}
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");
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);
}
}
}
av_packet_unref(&packet);
}
end:
avfilter_graph_free(&filter_graph);
avcodec_free_context(&dec_ctx);
avformat_close_input(&fmt_ctx);
av_frame_free(&frame);
av_frame_free(&filt_frame);
if (ret < 0 && ret != AVERROR_EOF) {
fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
exit(1);
}
exit(0);
}

291
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/examples/filtering_video.c Normal file
View file

@ -0,0 +1,291 @@
/*
* Copyright (c) 2010 Nicolas George
* Copyright (c) 2011 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* API example for decoding and filtering
* @example filtering_video.c
*/
#define _XOPEN_SOURCE 600 /* for usleep */
#include <unistd.h>
#include <stdio.h>
#include <stdlib.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
const char *filter_descr = "scale=78:24,transpose=cclock";
/* other way:
scale=78:24 [scl]; [scl] transpose=cclock // assumes "[in]" and "[out]" to be input output pads respectively
*/
static AVFormatContext *fmt_ctx;
static AVCodecContext *dec_ctx;
AVFilterContext *buffersink_ctx;
AVFilterContext *buffersrc_ctx;
AVFilterGraph *filter_graph;
static int video_stream_index = -1;
static int64_t last_pts = AV_NOPTS_VALUE;
static int open_input_file(const char *filename)
{
int ret;
AVCodec *dec;
if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
return ret;
}
if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
/* select the video stream */
ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, &dec, 0);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find a video stream in the input file\n");
return ret;
}
video_stream_index = ret;
/* 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);
/* init the video decoder */
if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open video decoder\n");
return ret;
}
return 0;
}
static int init_filters(const char *filters_descr)
{
char args[512];
int ret = 0;
const AVFilter *buffersrc = avfilter_get_by_name("buffer");
const AVFilter *buffersink = avfilter_get_by_name("buffersink");
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
AVRational time_base = fmt_ctx->streams[video_stream_index]->time_base;
enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE };
filter_graph = avfilter_graph_alloc();
if (!outputs || !inputs || !filter_graph) {
ret = AVERROR(ENOMEM);
goto end;
}
/* buffer video source: the decoded frames from the decoder will be inserted here. */
snprintf(args, sizeof(args),
"video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
time_base.num, time_base.den,
dec_ctx->sample_aspect_ratio.num, dec_ctx->sample_aspect_ratio.den);
ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
args, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n");
goto end;
}
/* buffer video sink: to terminate the filter chain. */
ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
NULL, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
goto end;
}
ret = av_opt_set_int_list(buffersink_ctx, "pix_fmts", pix_fmts,
AV_PIX_FMT_NONE, AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n");
goto end;
}
/*
* Set the endpoints for the filter graph. The filter_graph will
* be linked to the graph described by filters_descr.
*/
/*
* The buffer source output must be connected to the input pad of
* the first filter described by filters_descr; since the first
* filter input label is not specified, it is set to "in" by
* default.
*/
outputs->name = av_strdup("in");
outputs->filter_ctx = buffersrc_ctx;
outputs->pad_idx = 0;
outputs->next = NULL;
/*
* The buffer sink input must be connected to the output pad of
* the last filter described by filters_descr; since the last
* filter output label is not specified, it is set to "out" by
* default.
*/
inputs->name = av_strdup("out");
inputs->filter_ctx = buffersink_ctx;
inputs->pad_idx = 0;
inputs->next = NULL;
if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
&inputs, &outputs, NULL)) < 0)
goto end;
if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
goto end;
end:
avfilter_inout_free(&inputs);
avfilter_inout_free(&outputs);
return ret;
}
static void display_frame(const AVFrame *frame, AVRational time_base)
{
int x, y;
uint8_t *p0, *p;
int64_t delay;
if (frame->pts != AV_NOPTS_VALUE) {
if (last_pts != AV_NOPTS_VALUE) {
/* sleep roughly the right amount of time;
* usleep is in microseconds, just like AV_TIME_BASE. */
delay = av_rescale_q(frame->pts - last_pts,
time_base, AV_TIME_BASE_Q);
if (delay > 0 && delay < 1000000)
usleep(delay);
}
last_pts = frame->pts;
}
/* Trivial ASCII grayscale display. */
p0 = frame->data[0];
puts("\033c");
for (y = 0; y < frame->height; y++) {
p = p0;
for (x = 0; x < frame->width; x++)
putchar(" .-+#"[*(p++) / 52]);
putchar('\n');
p0 += frame->linesize[0];
}
fflush(stdout);
}
int main(int argc, char **argv)
{
int ret;
AVPacket packet;
AVFrame *frame;
AVFrame *filt_frame;
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);
}
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if ((ret = init_filters(filter_descr)) < 0)
goto end;
/* read all packets */
while (1) {
if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
break;
if (packet.stream_index == video_stream_index) {
ret = avcodec_send_packet(dec_ctx, &packet);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while sending a packet to the decoder\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;
/* push the decoded frame into the filtergraph */
if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
break;
}
/* pull filtered frames from the filtergraph */
while (1) {
ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
if (ret < 0)
goto end;
display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base);
av_frame_unref(filt_frame);
}
av_frame_unref(frame);
}
}
av_packet_unref(&packet);
}
end:
avfilter_graph_free(&filter_graph);
avcodec_free_context(&dec_ctx);
avformat_close_input(&fmt_ctx);
av_frame_free(&frame);
av_frame_free(&filt_frame);
if (ret < 0 && ret != AVERROR_EOF) {
fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
exit(1);
}
exit(0);
}

156
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/examples/http_multiclient.c Normal file
View file

@ -0,0 +1,156 @@
/*
* Copyright (c) 2015 Stephan Holljes
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* libavformat multi-client network API usage example.
*
* @example http_multiclient.c
* This example will serve a file without decoding or demuxing it over http.
* Multiple clients can connect and will receive the same file.
*/
#include <libavformat/avformat.h>
#include <libavutil/opt.h>
#include <unistd.h>
static void process_client(AVIOContext *client, const char *in_uri)
{
AVIOContext *input = NULL;
uint8_t buf[1024];
int ret, n, reply_code;
uint8_t *resource = NULL;
while ((ret = avio_handshake(client)) > 0) {
av_opt_get(client, "resource", AV_OPT_SEARCH_CHILDREN, &resource);
// check for strlen(resource) is necessary, because av_opt_get()
// may return empty string.
if (resource && strlen(resource))
break;
av_freep(&resource);
}
if (ret < 0)
goto end;
av_log(client, AV_LOG_TRACE, "resource=%p\n", resource);
if (resource && resource[0] == '/' && !strcmp((resource + 1), in_uri)) {
reply_code = 200;
} else {
reply_code = AVERROR_HTTP_NOT_FOUND;
}
if ((ret = av_opt_set_int(client, "reply_code", reply_code, AV_OPT_SEARCH_CHILDREN)) < 0) {
av_log(client, AV_LOG_ERROR, "Failed to set reply_code: %s.\n", av_err2str(ret));
goto end;
}
av_log(client, AV_LOG_TRACE, "Set reply code to %d\n", reply_code);
while ((ret = avio_handshake(client)) > 0);
if (ret < 0)
goto end;
fprintf(stderr, "Handshake performed.\n");
if (reply_code != 200)
goto end;
fprintf(stderr, "Opening input file.\n");
if ((ret = avio_open2(&input, in_uri, AVIO_FLAG_READ, NULL, NULL)) < 0) {
av_log(input, AV_LOG_ERROR, "Failed to open input: %s: %s.\n", in_uri,
av_err2str(ret));
goto end;
}
for(;;) {
n = avio_read(input, buf, sizeof(buf));
if (n < 0) {
if (n == AVERROR_EOF)
break;
av_log(input, AV_LOG_ERROR, "Error reading from input: %s.\n",
av_err2str(n));
break;
}
avio_write(client, buf, n);
avio_flush(client);
}
end:
fprintf(stderr, "Flushing client\n");
avio_flush(client);
fprintf(stderr, "Closing client\n");
avio_close(client);
fprintf(stderr, "Closing input\n");
avio_close(input);
av_freep(&resource);
}
int main(int argc, char **argv)
{
AVDictionary *options = NULL;
AVIOContext *client = NULL, *server = NULL;
const char *in_uri, *out_uri;
int ret, pid;
av_log_set_level(AV_LOG_TRACE);
if (argc < 3) {
printf("usage: %s input http://hostname[:port]\n"
"API example program to serve http to multiple clients.\n"
"\n", argv[0]);
return 1;
}
in_uri = argv[1];
out_uri = argv[2];
avformat_network_init();
if ((ret = av_dict_set(&options, "listen", "2", 0)) < 0) {
fprintf(stderr, "Failed to set listen mode for server: %s\n", av_err2str(ret));
return ret;
}
if ((ret = avio_open2(&server, out_uri, AVIO_FLAG_WRITE, NULL, &options)) < 0) {
fprintf(stderr, "Failed to open server: %s\n", av_err2str(ret));
return ret;
}
fprintf(stderr, "Entering main loop.\n");
for(;;) {
if ((ret = avio_accept(server, &client)) < 0)
goto end;
fprintf(stderr, "Accepted client, forking process.\n");
// XXX: Since we don't reap our children and don't ignore signals
// this produces zombie processes.
pid = fork();
if (pid < 0) {
perror("Fork failed");
ret = AVERROR(errno);
goto end;
}
if (pid == 0) {
fprintf(stderr, "In child.\n");
process_client(client, in_uri);
avio_close(server);
exit(0);
}
if (pid > 0)
avio_close(client);
}
end:
avio_close(server);
if (ret < 0 && ret != AVERROR_EOF) {
fprintf(stderr, "Some errors occurred: %s\n", av_err2str(ret));
return 1;
}
return 0;
}

35
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/doc/examples/hw_decode.c → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/examples/hw_decode.c
View file

@ -24,11 +24,12 @@
*/
/**
* @file HW-accelerated decoding API usage.example
* @example hw_decode.c
* @file
* HW-Accelerated decoding example.
*
* Perform HW-accelerated decoding with output frames from HW video
* surfaces.
* @example hw_decode.c
* This example shows how to do HW-accelerated decoding with output
* frames from the HW video surfaces.
*/
#include <stdio.h>
@ -151,8 +152,8 @@ int main(int argc, char *argv[])
int video_stream, ret;
AVStream *video = NULL;
AVCodecContext *decoder_ctx = NULL;
const AVCodec *decoder = NULL;
AVPacket *packet = NULL;
AVCodec *decoder = NULL;
AVPacket packet;
enum AVHWDeviceType type;
int i;
@ -171,12 +172,6 @@ int main(int argc, char *argv[])
return -1;
}
packet = av_packet_alloc();
if (!packet) {
fprintf(stderr, "Failed to allocate AVPacket\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]);
@ -228,25 +223,27 @@ int main(int argc, char *argv[])
}
/* open the file to dump raw data */
output_file = fopen(argv[3], "w+b");
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)
if ((ret = av_read_frame(input_ctx, &packet)) < 0)
break;
if (video_stream == packet->stream_index)
ret = decode_write(decoder_ctx, packet);
if (video_stream == packet.stream_index)
ret = decode_write(decoder_ctx, &packet);
av_packet_unref(packet);
av_packet_unref(&packet);
}
/* flush the decoder */
ret = decode_write(decoder_ctx, NULL);
packet.data = NULL;
packet.size = 0;
ret = decode_write(decoder_ctx, &packet);
av_packet_unref(&packet);
if (output_file)
fclose(output_file);
av_packet_free(&packet);
avcodec_free_context(&decoder_ctx);
avformat_close_input(&input_ctx);
av_buffer_unref(&hw_device_ctx);

55
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/examples/metadata.c Normal file
View file

@ -0,0 +1,55 @@
/*
* Copyright (c) 2011 Reinhard Tartler
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* Shows how the metadata API can be used in application programs.
* @example metadata.c
*/
#include <stdio.h>
#include <libavformat/avformat.h>
#include <libavutil/dict.h>
int main (int argc, char **argv)
{
AVFormatContext *fmt_ctx = NULL;
AVDictionaryEntry *tag = NULL;
int ret;
if (argc != 2) {
printf("usage: %s <input_file>\n"
"example program to demonstrate the use of the libavformat metadata API.\n"
"\n", argv[0]);
return 1;
}
if ((ret = avformat_open_input(&fmt_ctx, argv[1], NULL, NULL)))
return ret;
while ((tag = av_dict_get(fmt_ctx->metadata, "", tag, AV_DICT_IGNORE_SUFFIX)))
printf("%s=%s\n", tag->key, tag->value);
avformat_close_input(&fmt_ctx);
return 0;
}

667
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/examples/muxing.c Normal file
View file

@ -0,0 +1,667 @@
/*
* Copyright (c) 2003 Fabrice Bellard
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* libavformat API example.
*
* Output a media file in any supported libavformat format. The default
* codecs are used.
* @example muxing.c
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <math.h>
#include <libavutil/avassert.h>
#include <libavutil/channel_layout.h>
#include <libavutil/opt.h>
#include <libavutil/mathematics.h>
#include <libavutil/timestamp.h>
#include <libavformat/avformat.h>
#include <libswscale/swscale.h>
#include <libswresample/swresample.h>
#define STREAM_DURATION 10.0
#define STREAM_FRAME_RATE 25 /* 25 images/s */
#define STREAM_PIX_FMT AV_PIX_FMT_YUV420P /* default pix_fmt */
#define SCALE_FLAGS SWS_BICUBIC
// a wrapper around a single output AVStream
typedef struct OutputStream {
AVStream *st;
AVCodecContext *enc;
/* pts of the next frame that will be generated */
int64_t next_pts;
int samples_count;
AVFrame *frame;
AVFrame *tmp_frame;
float t, tincr, tincr2;
struct SwsContext *sws_ctx;
struct SwrContext *swr_ctx;
} OutputStream;
static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt)
{
AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base;
printf("pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n",
av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base),
av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base),
av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base),
pkt->stream_index);
}
static int write_frame(AVFormatContext *fmt_ctx, const AVRational *time_base, AVStream *st, AVPacket *pkt)
{
/* rescale output packet timestamp values from codec to stream timebase */
av_packet_rescale_ts(pkt, *time_base, st->time_base);
pkt->stream_index = st->index;
/* Write the compressed frame to the media file. */
log_packet(fmt_ctx, pkt);
return av_interleaved_write_frame(fmt_ctx, pkt);
}
/* Add an output stream. */
static void add_stream(OutputStream *ost, AVFormatContext *oc,
AVCodec **codec,
enum AVCodecID codec_id)
{
AVCodecContext *c;
int i;
/* find the encoder */
*codec = avcodec_find_encoder(codec_id);
if (!(*codec)) {
fprintf(stderr, "Could not find encoder for '%s'\n",
avcodec_get_name(codec_id));
exit(1);
}
ost->st = avformat_new_stream(oc, NULL);
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;
switch ((*codec)->type) {
case AVMEDIA_TYPE_AUDIO:
c->sample_fmt = (*codec)->sample_fmts ?
(*codec)->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
c->bit_rate = 64000;
c->sample_rate = 44100;
if ((*codec)->supported_samplerates) {
c->sample_rate = (*codec)->supported_samplerates[0];
for (i = 0; (*codec)->supported_samplerates[i]; i++) {
if ((*codec)->supported_samplerates[i] == 44100)
c->sample_rate = 44100;
}
}
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
c->channel_layout = AV_CH_LAYOUT_STEREO;
if ((*codec)->channel_layouts) {
c->channel_layout = (*codec)->channel_layouts[0];
for (i = 0; (*codec)->channel_layouts[i]; i++) {
if ((*codec)->channel_layouts[i] == AV_CH_LAYOUT_STEREO)
c->channel_layout = AV_CH_LAYOUT_STEREO;
}
}
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
ost->st->time_base = (AVRational){ 1, c->sample_rate };
break;
case AVMEDIA_TYPE_VIDEO:
c->codec_id = codec_id;
c->bit_rate = 400000;
/* Resolution must be a multiple of two. */
c->width = 352;
c->height = 288;
/* timebase: This is the fundamental unit of time (in seconds) in terms
* of which frame timestamps are represented. For fixed-fps content,
* timebase should be 1/framerate and timestamp increments should be
* identical to 1. */
ost->st->time_base = (AVRational){ 1, STREAM_FRAME_RATE };
c->time_base = ost->st->time_base;
c->gop_size = 12; /* emit one intra frame every twelve frames at most */
c->pix_fmt = STREAM_PIX_FMT;
if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) {
/* just for testing, we also add B-frames */
c->max_b_frames = 2;
}
if (c->codec_id == AV_CODEC_ID_MPEG1VIDEO) {
/* Needed to avoid using macroblocks in which some coeffs overflow.
* This does not happen with normal video, it just happens here as
* the motion of the chroma plane does not match the luma plane. */
c->mb_decision = 2;
}
break;
default:
break;
}
/* Some formats want stream headers to be separate. */
if (oc->oformat->flags & AVFMT_GLOBALHEADER)
c->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
}
/**************************************************************/
/* audio output */
static AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt,
uint64_t channel_layout,
int sample_rate, int nb_samples)
{
AVFrame *frame = av_frame_alloc();
int ret;
if (!frame) {
fprintf(stderr, "Error allocating an audio frame\n");
exit(1);
}
frame->format = sample_fmt;
frame->channel_layout = channel_layout;
frame->sample_rate = sample_rate;
frame->nb_samples = nb_samples;
if (nb_samples) {
ret = av_frame_get_buffer(frame, 0);
if (ret < 0) {
fprintf(stderr, "Error allocating an audio buffer\n");
exit(1);
}
}
return frame;
}
static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
{
AVCodecContext *c;
int nb_samples;
int ret;
AVDictionary *opt = NULL;
c = ost->enc;
/* open it */
av_dict_copy(&opt, opt_arg, 0);
ret = avcodec_open2(c, codec, &opt);
av_dict_free(&opt);
if (ret < 0) {
fprintf(stderr, "Could not open audio codec: %s\n", av_err2str(ret));
exit(1);
}
/* init signal generator */
ost->t = 0;
ost->tincr = 2 * M_PI * 110.0 / c->sample_rate;
/* increment frequency by 110 Hz per second */
ost->tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
if (c->codec->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE)
nb_samples = 10000;
else
nb_samples = c->frame_size;
ost->frame = alloc_audio_frame(c->sample_fmt, c->channel_layout,
c->sample_rate, nb_samples);
ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, c->channel_layout,
c->sample_rate, nb_samples);
/* copy the stream parameters to the muxer */
ret = avcodec_parameters_from_context(ost->st->codecpar, c);
if (ret < 0) {
fprintf(stderr, "Could not copy the stream parameters\n");
exit(1);
}
/* create resampler context */
ost->swr_ctx = swr_alloc();
if (!ost->swr_ctx) {
fprintf(stderr, "Could not allocate resampler context\n");
exit(1);
}
/* set options */
av_opt_set_int (ost->swr_ctx, "in_channel_count", c->channels, 0);
av_opt_set_int (ost->swr_ctx, "in_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(ost->swr_ctx, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
av_opt_set_int (ost->swr_ctx, "out_channel_count", c->channels, 0);
av_opt_set_int (ost->swr_ctx, "out_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(ost->swr_ctx, "out_sample_fmt", c->sample_fmt, 0);
/* initialize the resampling context */
if ((ret = swr_init(ost->swr_ctx)) < 0) {
fprintf(stderr, "Failed to initialize the resampling context\n");
exit(1);
}
}
/* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
* 'nb_channels' channels. */
static AVFrame *get_audio_frame(OutputStream *ost)
{
AVFrame *frame = ost->tmp_frame;
int j, i, v;
int16_t *q = (int16_t*)frame->data[0];
/* check if we want to generate more frames */
if (av_compare_ts(ost->next_pts, ost->enc->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++)
*q++ = v;
ost->t += ost->tincr;
ost->tincr += ost->tincr2;
}
frame->pts = ost->next_pts;
ost->next_pts += frame->nb_samples;
return frame;
}
/*
* encode one audio frame and send it to the muxer
* return 1 when encoding is finished, 0 otherwise
*/
static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
{
AVCodecContext *c;
AVPacket pkt = { 0 }; // data and size must be 0;
AVFrame *frame;
int ret;
int got_packet;
int dst_nb_samples;
av_init_packet(&pkt);
c = ost->enc;
frame = get_audio_frame(ost);
if (frame) {
/* convert samples from native format to destination codec format, using the resampler */
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples,
c->sample_rate, c->sample_rate, AV_ROUND_UP);
av_assert0(dst_nb_samples == frame->nb_samples);
/* when we pass a frame to the encoder, it may keep a reference to it
* internally;
* make sure we do not overwrite it here
*/
ret = av_frame_make_writable(ost->frame);
if (ret < 0)
exit(1);
/* convert to destination format */
ret = swr_convert(ost->swr_ctx,
ost->frame->data, dst_nb_samples,
(const uint8_t **)frame->data, frame->nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
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;
}
ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
if (ret < 0) {
fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
exit(1);
}
if (got_packet) {
ret = write_frame(oc, &c->time_base, ost->st, &pkt);
if (ret < 0) {
fprintf(stderr, "Error while writing audio frame: %s\n",
av_err2str(ret));
exit(1);
}
}
return (frame || got_packet) ? 0 : 1;
}
/**************************************************************/
/* video output */
static AVFrame *alloc_picture(enum AVPixelFormat pix_fmt, int width, int height)
{
AVFrame *picture;
int ret;
picture = av_frame_alloc();
if (!picture)
return NULL;
picture->format = pix_fmt;
picture->width = width;
picture->height = height;
/* allocate the buffers for the frame data */
ret = av_frame_get_buffer(picture, 32);
if (ret < 0) {
fprintf(stderr, "Could not allocate frame data.\n");
exit(1);
}
return picture;
}
static void open_video(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
{
int ret;
AVCodecContext *c = ost->enc;
AVDictionary *opt = NULL;
av_dict_copy(&opt, opt_arg, 0);
/* open the codec */
ret = avcodec_open2(c, codec, &opt);
av_dict_free(&opt);
if (ret < 0) {
fprintf(stderr, "Could not open video codec: %s\n", av_err2str(ret));
exit(1);
}
/* allocate and init a re-usable frame */
ost->frame = alloc_picture(c->pix_fmt, c->width, c->height);
if (!ost->frame) {
fprintf(stderr, "Could not allocate video frame\n");
exit(1);
}
/* If the output format is not YUV420P, then a temporary YUV420P
* picture is needed too. It is then converted to the required
* output format. */
ost->tmp_frame = NULL;
if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
ost->tmp_frame = alloc_picture(AV_PIX_FMT_YUV420P, c->width, c->height);
if (!ost->tmp_frame) {
fprintf(stderr, "Could not allocate temporary picture\n");
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;
i = frame_index;
/* Y */
for (y = 0; y < height; y++)
for (x = 0; x < width; x++)
pict->data[0][y * pict->linesize[0] + x] = x + y + i * 3;
/* Cb and Cr */
for (y = 0; y < height / 2; y++) {
for (x = 0; x < width / 2; x++) {
pict->data[1][y * pict->linesize[1] + x] = 128 + y + i * 2;
pict->data[2][y * pict->linesize[2] + x] = 64 + x + i * 5;
}
}
}
static AVFrame *get_video_frame(OutputStream *ost)
{
AVCodecContext *c = ost->enc;
/* check if we want to generate more frames */
if (av_compare_ts(ost->next_pts, c->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 */
if (!ost->sws_ctx) {
ost->sws_ctx = sws_getContext(c->width, c->height,
AV_PIX_FMT_YUV420P,
c->width, c->height,
c->pix_fmt,
SCALE_FLAGS, NULL, NULL, NULL);
if (!ost->sws_ctx) {
fprintf(stderr,
"Could not initialize the conversion context\n");
exit(1);
}
}
fill_yuv_image(ost->tmp_frame, ost->next_pts, c->width, c->height);
sws_scale(ost->sws_ctx, (const uint8_t * const *) ost->tmp_frame->data,
ost->tmp_frame->linesize, 0, c->height, ost->frame->data,
ost->frame->linesize);
} else {
fill_yuv_image(ost->frame, ost->next_pts, c->width, c->height);
}
ost->frame->pts = ost->next_pts++;
return ost->frame;
}
/*
* encode one video frame and send it to the muxer
* return 1 when encoding is finished, 0 otherwise
*/
static int write_video_frame(AVFormatContext *oc, OutputStream *ost)
{
int ret;
AVCodecContext *c;
AVFrame *frame;
int got_packet = 0;
AVPacket pkt = { 0 };
c = ost->enc;
frame = get_video_frame(ost);
av_init_packet(&pkt);
/* encode the image */
ret = avcodec_encode_video2(c, &pkt, frame, &got_packet);
if (ret < 0) {
fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret));
exit(1);
}
if (got_packet) {
ret = write_frame(oc, &c->time_base, ost->st, &pkt);
} else {
ret = 0;
}
if (ret < 0) {
fprintf(stderr, "Error while writing video frame: %s\n", av_err2str(ret));
exit(1);
}
return (frame || got_packet) ? 0 : 1;
}
static void close_stream(AVFormatContext *oc, OutputStream *ost)
{
avcodec_free_context(&ost->enc);
av_frame_free(&ost->frame);
av_frame_free(&ost->tmp_frame);
sws_freeContext(ost->sws_ctx);
swr_free(&ost->swr_ctx);
}
/**************************************************************/
/* media file output */
int main(int argc, char **argv)
{
OutputStream video_st = { 0 }, audio_st = { 0 };
const char *filename;
AVOutputFormat *fmt;
AVFormatContext *oc;
AVCodec *audio_codec, *video_codec;
int ret;
int have_video = 0, have_audio = 0;
int encode_video = 0, encode_audio = 0;
AVDictionary *opt = NULL;
int i;
if (argc < 2) {
printf("usage: %s output_file\n"
"API example program to output a media file with libavformat.\n"
"This program generates a synthetic audio and video stream, encodes and\n"
"muxes them into a file named output_file.\n"
"The output format is automatically guessed according to the file extension.\n"
"Raw images can also be output by using '%%d' in the filename.\n"
"\n", argv[0]);
return 1;
}
filename = argv[1];
for (i = 2; i+1 < argc; i+=2) {
if (!strcmp(argv[i], "-flags") || !strcmp(argv[i], "-fflags"))
av_dict_set(&opt, argv[i]+1, argv[i+1], 0);
}
/* allocate the output media context */
avformat_alloc_output_context2(&oc, NULL, NULL, filename);
if (!oc) {
printf("Could not deduce output format from file extension: using MPEG.\n");
avformat_alloc_output_context2(&oc, NULL, "mpeg", filename);
}
if (!oc)
return 1;
fmt = oc->oformat;
/* Add the audio and video streams using the default format codecs
* and initialize the codecs. */
if (fmt->video_codec != AV_CODEC_ID_NONE) {
add_stream(&video_st, oc, &video_codec, fmt->video_codec);
have_video = 1;
encode_video = 1;
}
if (fmt->audio_codec != AV_CODEC_ID_NONE) {
add_stream(&audio_st, oc, &audio_codec, fmt->audio_codec);
have_audio = 1;
encode_audio = 1;
}
/* Now that all the parameters are set, we can open the audio and
* video codecs and allocate the necessary encode buffers. */
if (have_video)
open_video(oc, video_codec, &video_st, opt);
if (have_audio)
open_audio(oc, audio_codec, &audio_st, opt);
av_dump_format(oc, 0, filename, 1);
/* open the output file, if needed */
if (!(fmt->flags & AVFMT_NOFILE)) {
ret = avio_open(&oc->pb, filename, AVIO_FLAG_WRITE);
if (ret < 0) {
fprintf(stderr, "Could not open '%s': %s\n", filename,
av_err2str(ret));
return 1;
}
}
/* Write the stream header, if any. */
ret = avformat_write_header(oc, &opt);
if (ret < 0) {
fprintf(stderr, "Error occurred when opening output file: %s\n",
av_err2str(ret));
return 1;
}
while (encode_video || encode_audio) {
/* select the stream to encode */
if (encode_video &&
(!encode_audio || av_compare_ts(video_st.next_pts, video_st.enc->time_base,
audio_st.next_pts, audio_st.enc->time_base) <= 0)) {
encode_video = !write_video_frame(oc, &video_st);
} else {
encode_audio = !write_audio_frame(oc, &audio_st);
}
}
/* Write the trailer, if any. The trailer must be written before you
* close the CodecContexts open when you wrote the header; otherwise
* av_write_trailer() may try to use memory that was freed on
* av_codec_close(). */
av_write_trailer(oc);
/* Close each codec. */
if (have_video)
close_stream(oc, &video_st);
if (have_audio)
close_stream(oc, &audio_st);
if (!(fmt->flags & AVFMT_NOFILE))
/* Close the output file. */
avio_closep(&oc->pb);
/* free the stream */
avformat_free_context(oc);
return 0;
}

271
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/examples/qsvdec.c Normal file
View file

@ -0,0 +1,271 @@
/*
* Copyright (c) 2015 Anton Khirnov
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* Intel QSV-accelerated H.264 decoding example.
*
* @example qsvdec.c
* This example shows how to do QSV-accelerated H.264 decoding with output
* frames in the GPU video surfaces.
*/
#include "config.h"
#include <stdio.h>
#include "libavformat/avformat.h"
#include "libavformat/avio.h"
#include "libavcodec/avcodec.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;
} DecodeContext;
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;
/* 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;
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;
return AV_PIX_FMT_QSV;
}
pix_fmts++;
}
fprintf(stderr, "The QSV pixel format not offered in get_format()\n");
return AV_PIX_FMT_NONE;
}
static int decode_packet(DecodeContext *decode, AVCodecContext *decoder_ctx,
AVFrame *frame, AVFrame *sw_frame,
AVPacket *pkt, AVIOContext *output_ctx)
{
int ret = 0;
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) {
fprintf(stderr, "Error during decoding\n");
return 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;
}
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);
fail:
av_frame_unref(sw_frame);
av_frame_unref(frame);
if (ret < 0)
return ret;
}
return 0;
}
int main(int argc, char **argv)
{
AVFormatContext *input_ctx = NULL;
AVStream *video_st = NULL;
AVCodecContext *decoder_ctx = NULL;
const AVCodec *decoder;
AVPacket pkt = { 0 };
AVFrame *frame = NULL, *sw_frame = NULL;
DecodeContext decode = { NULL };
AVIOContext *output_ctx = NULL;
int ret, i;
if (argc < 3) {
fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
return 1;
}
/* open the input file */
ret = avformat_open_input(&input_ctx, argv[1], NULL, NULL);
if (ret < 0) {
fprintf(stderr, "Cannot open input file '%s': ", argv[1]);
goto finish;
}
/* find the first H.264 video stream */
for (i = 0; i < input_ctx->nb_streams; i++) {
AVStream *st = input_ctx->streams[i];
if (st->codecpar->codec_id == AV_CODEC_ID_H264 && !video_st)
video_st = st;
else
st->discard = AVDISCARD_ALL;
}
if (!video_st) {
fprintf(stderr, "No H.264 video stream in the input file\n");
goto finish;
}
/* 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");
goto finish;
}
/* initialize the decoder */
decoder = avcodec_find_decoder_by_name("h264_qsv");
if (!decoder) {
fprintf(stderr, "The QSV decoder is not present in libavcodec\n");
goto finish;
}
decoder_ctx = avcodec_alloc_context3(decoder);
if (!decoder_ctx) {
ret = AVERROR(ENOMEM);
goto finish;
}
decoder_ctx->codec_id = AV_CODEC_ID_H264;
if (video_st->codecpar->extradata_size) {
decoder_ctx->extradata = av_mallocz(video_st->codecpar->extradata_size +
AV_INPUT_BUFFER_PADDING_SIZE);
if (!decoder_ctx->extradata) {
ret = AVERROR(ENOMEM);
goto finish;
}
memcpy(decoder_ctx->extradata, video_st->codecpar->extradata,
video_st->codecpar->extradata_size);
decoder_ctx->extradata_size = video_st->codecpar->extradata_size;
}
decoder_ctx->opaque = &decode;
decoder_ctx->get_format = get_format;
ret = avcodec_open2(decoder_ctx, NULL, NULL);
if (ret < 0) {
fprintf(stderr, "Error opening the decoder: ");
goto finish;
}
/* open the output stream */
ret = avio_open(&output_ctx, argv[2], AVIO_FLAG_WRITE);
if (ret < 0) {
fprintf(stderr, "Error opening the output context: ");
goto finish;
}
frame = av_frame_alloc();
sw_frame = av_frame_alloc();
if (!frame || !sw_frame) {
ret = AVERROR(ENOMEM);
goto finish;
}
/* actual decoding */
while (ret >= 0) {
ret = av_read_frame(input_ctx, &pkt);
if (ret < 0)
break;
if (pkt.stream_index == video_st->index)
ret = decode_packet(&decode, decoder_ctx, frame, sw_frame, &pkt, output_ctx);
av_packet_unref(&pkt);
}
/* flush the decoder */
pkt.data = NULL;
pkt.size = 0;
ret = decode_packet(&decode, decoder_ctx, frame, sw_frame, &pkt, output_ctx);
finish:
if (ret < 0) {
char buf[1024];
av_strerror(ret, buf, sizeof(buf));
fprintf(stderr, "%s\n", buf);
}
avformat_close_input(&input_ctx);
av_frame_free(&frame);
av_frame_free(&sw_frame);
avcodec_free_context(&decoder_ctx);
av_buffer_unref(&decode.hw_device_ref);
avio_close(output_ctx);
return ret;
}

191
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/examples/remuxing.c Normal file
View file

@ -0,0 +1,191 @@
/*
* Copyright (c) 2013 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* libavformat/libavcodec demuxing and muxing API example.
*
* Remux streams from one container format to another.
* @example remuxing.c
*/
#include <libavutil/timestamp.h>
#include <libavformat/avformat.h>
static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt, const char *tag)
{
AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base;
printf("%s: pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n",
tag,
av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base),
av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base),
av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base),
pkt->stream_index);
}
int main(int argc, char **argv)
{
AVOutputFormat *ofmt = NULL;
AVFormatContext *ifmt_ctx = NULL, *ofmt_ctx = NULL;
AVPacket pkt;
const char *in_filename, *out_filename;
int ret, i;
int stream_index = 0;
int *stream_mapping = NULL;
int stream_mapping_size = 0;
if (argc < 3) {
printf("usage: %s input output\n"
"API example program to remux a media file with libavformat and libavcodec.\n"
"The output format is guessed according to the file extension.\n"
"\n", argv[0]);
return 1;
}
in_filename = argv[1];
out_filename = argv[2];
if ((ret = avformat_open_input(&ifmt_ctx, in_filename, 0, 0)) < 0) {
fprintf(stderr, "Could not open input file '%s'", in_filename);
goto end;
}
if ((ret = avformat_find_stream_info(ifmt_ctx, 0)) < 0) {
fprintf(stderr, "Failed to retrieve input stream information");
goto end;
}
av_dump_format(ifmt_ctx, 0, in_filename, 0);
avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, out_filename);
if (!ofmt_ctx) {
fprintf(stderr, "Could not create output context\n");
ret = AVERROR_UNKNOWN;
goto end;
}
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);
if (!out_stream) {
fprintf(stderr, "Failed allocating output stream\n");
ret = AVERROR_UNKNOWN;
goto end;
}
ret = avcodec_parameters_copy(out_stream->codecpar, in_codecpar);
if (ret < 0) {
fprintf(stderr, "Failed to copy codec parameters\n");
goto end;
}
out_stream->codecpar->codec_tag = 0;
}
av_dump_format(ofmt_ctx, 0, out_filename, 1);
if (!(ofmt->flags & AVFMT_NOFILE)) {
ret = avio_open(&ofmt_ctx->pb, out_filename, AVIO_FLAG_WRITE);
if (ret < 0) {
fprintf(stderr, "Could not open output file '%s'", out_filename);
goto end;
}
}
ret = avformat_write_header(ofmt_ctx, NULL);
if (ret < 0) {
fprintf(stderr, "Error occurred when opening output file\n");
goto end;
}
while (1) {
AVStream *in_stream, *out_stream;
ret = av_read_frame(ifmt_ctx, &pkt);
if (ret < 0)
break;
in_stream = ifmt_ctx->streams[pkt.stream_index];
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 */
pkt.pts = av_rescale_q_rnd(pkt.pts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
pkt.dts = av_rescale_q_rnd(pkt.dts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
pkt.duration = av_rescale_q(pkt.duration, in_stream->time_base, out_stream->time_base);
pkt.pos = -1;
log_packet(ofmt_ctx, &pkt, "out");
ret = av_interleaved_write_frame(ofmt_ctx, &pkt);
if (ret < 0) {
fprintf(stderr, "Error muxing packet\n");
break;
}
av_packet_unref(&pkt);
}
av_write_trailer(ofmt_ctx);
end:
avformat_close_input(&ifmt_ctx);
/* close output */
if (ofmt_ctx && !(ofmt->flags & AVFMT_NOFILE))
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;
}
return 0;
}

214
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/examples/resampling_audio.c Normal file
View file

@ -0,0 +1,214 @@
/*
* Copyright (c) 2012 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @example resampling_audio.c
* libswresample API use example.
*/
#include <libavutil/opt.h>
#include <libavutil/channel_layout.h>
#include <libavutil/samplefmt.h>
#include <libswresample/swresample.h>
static int get_format_from_sample_fmt(const char **fmt,
enum AVSampleFormat sample_fmt)
{
int i;
struct sample_fmt_entry {
enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
} sample_fmt_entries[] = {
{ AV_SAMPLE_FMT_U8, "u8", "u8" },
{ AV_SAMPLE_FMT_S16, "s16be", "s16le" },
{ AV_SAMPLE_FMT_S32, "s32be", "s32le" },
{ AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
{ AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
};
*fmt = NULL;
for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
struct sample_fmt_entry *entry = &sample_fmt_entries[i];
if (sample_fmt == entry->sample_fmt) {
*fmt = AV_NE(entry->fmt_be, entry->fmt_le);
return 0;
}
}
fprintf(stderr,
"Sample format %s not supported as output format\n",
av_get_sample_fmt_name(sample_fmt));
return AVERROR(EINVAL);
}
/**
* Fill dst buffer with nb_samples, generated starting from t.
*/
static void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t)
{
int i, j;
double tincr = 1.0 / sample_rate, *dstp = dst;
const double c = 2 * M_PI * 440.0;
/* generate sin tone with 440Hz frequency and duplicated channels */
for (i = 0; i < nb_samples; i++) {
*dstp = sin(c * *t);
for (j = 1; j < nb_channels; j++)
dstp[j] = dstp[0];
dstp += nb_channels;
*t += tincr;
}
}
int main(int argc, char **argv)
{
int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND;
int src_rate = 48000, dst_rate = 44100;
uint8_t **src_data = NULL, **dst_data = NULL;
int src_nb_channels = 0, dst_nb_channels = 0;
int src_linesize, dst_linesize;
int src_nb_samples = 1024, dst_nb_samples, max_dst_nb_samples;
enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_DBL, dst_sample_fmt = AV_SAMPLE_FMT_S16;
const char *dst_filename = NULL;
FILE *dst_file;
int dst_bufsize;
const char *fmt;
struct SwrContext *swr_ctx;
double t;
int ret;
if (argc != 2) {
fprintf(stderr, "Usage: %s output_file\n"
"API example program to show how to resample an audio stream with libswresample.\n"
"This program generates a series of audio frames, resamples them to a specified "
"output format and rate and saves them to an output file named output_file.\n",
argv[0]);
exit(1);
}
dst_filename = argv[1];
dst_file = fopen(dst_filename, "wb");
if (!dst_file) {
fprintf(stderr, "Could not open destination file %s\n", dst_filename);
exit(1);
}
/* create resampler context */
swr_ctx = swr_alloc();
if (!swr_ctx) {
fprintf(stderr, "Could not allocate resampler context\n");
ret = AVERROR(ENOMEM);
goto end;
}
/* set options */
av_opt_set_int(swr_ctx, "in_channel_layout", src_ch_layout, 0);
av_opt_set_int(swr_ctx, "in_sample_rate", src_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);
av_opt_set_int(swr_ctx, "out_channel_layout", dst_ch_layout, 0);
av_opt_set_int(swr_ctx, "out_sample_rate", dst_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);
/* initialize the resampling context */
if ((ret = swr_init(swr_ctx)) < 0) {
fprintf(stderr, "Failed to initialize the resampling context\n");
goto end;
}
/* allocate source and destination samples buffers */
src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_nb_channels,
src_nb_samples, src_sample_fmt, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate source samples\n");
goto end;
}
/* compute the number of converted samples: buffering is avoided
* ensuring that the output buffer will contain at least all the
* converted input samples */
max_dst_nb_samples = dst_nb_samples =
av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
/* buffer is going to be directly written to a rawaudio file, no alignment */
dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels,
dst_nb_samples, dst_sample_fmt, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate destination samples\n");
goto end;
}
t = 0;
do {
/* generate synthetic audio */
fill_samples((double *)src_data[0], src_nb_samples, src_nb_channels, src_rate, &t);
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) +
src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
if (dst_nb_samples > max_dst_nb_samples) {
av_freep(&dst_data[0]);
ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
dst_nb_samples, dst_sample_fmt, 1);
if (ret < 0)
break;
max_dst_nb_samples = dst_nb_samples;
}
/* convert to destination format */
ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)src_data, src_nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
goto end;
}
dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
ret, dst_sample_fmt, 1);
if (dst_bufsize < 0) {
fprintf(stderr, "Could not get sample buffer size\n");
goto end;
}
printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret);
fwrite(dst_data[0], 1, dst_bufsize, dst_file);
} while (t < 10);
if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0)
goto end;
fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"
"ffplay -f %s -channel_layout %"PRId64" -channels %d -ar %d %s\n",
fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);
end:
fclose(dst_file);
if (src_data)
av_freep(&src_data[0]);
av_freep(&src_data);
if (dst_data)
av_freep(&dst_data[0]);
av_freep(&dst_data);
swr_free(&swr_ctx);
return ret < 0;
}

140
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/examples/scaling_video.c Normal file
View file

@ -0,0 +1,140 @@
/*
* Copyright (c) 2012 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* libswscale API use example.
* @example scaling_video.c
*/
#include <libavutil/imgutils.h>
#include <libavutil/parseutils.h>
#include <libswscale/swscale.h>
static void fill_yuv_image(uint8_t *data[4], int linesize[4],
int width, int height, int frame_index)
{
int x, y;
/* Y */
for (y = 0; y < height; y++)
for (x = 0; x < width; x++)
data[0][y * linesize[0] + x] = x + y + frame_index * 3;
/* Cb and Cr */
for (y = 0; y < height / 2; y++) {
for (x = 0; x < width / 2; x++) {
data[1][y * linesize[1] + x] = 128 + y + frame_index * 2;
data[2][y * linesize[2] + x] = 64 + x + frame_index * 5;
}
}
}
int main(int argc, char **argv)
{
uint8_t *src_data[4], *dst_data[4];
int src_linesize[4], dst_linesize[4];
int src_w = 320, src_h = 240, dst_w, dst_h;
enum AVPixelFormat src_pix_fmt = AV_PIX_FMT_YUV420P, dst_pix_fmt = AV_PIX_FMT_RGB24;
const char *dst_size = NULL;
const char *dst_filename = NULL;
FILE *dst_file;
int dst_bufsize;
struct SwsContext *sws_ctx;
int i, ret;
if (argc != 3) {
fprintf(stderr, "Usage: %s output_file output_size\n"
"API example program to show how to scale an image with libswscale.\n"
"This program generates a series of pictures, rescales them to the given "
"output_size and saves them to an output file named output_file\n."
"\n", argv[0]);
exit(1);
}
dst_filename = argv[1];
dst_size = argv[2];
if (av_parse_video_size(&dst_w, &dst_h, dst_size) < 0) {
fprintf(stderr,
"Invalid size '%s', must be in the form WxH or a valid size abbreviation\n",
dst_size);
exit(1);
}
dst_file = fopen(dst_filename, "wb");
if (!dst_file) {
fprintf(stderr, "Could not open destination file %s\n", dst_filename);
exit(1);
}
/* create scaling context */
sws_ctx = sws_getContext(src_w, src_h, src_pix_fmt,
dst_w, dst_h, dst_pix_fmt,
SWS_BILINEAR, NULL, NULL, NULL);
if (!sws_ctx) {
fprintf(stderr,
"Impossible to create scale context for the conversion "
"fmt:%s s:%dx%d -> fmt:%s s:%dx%d\n",
av_get_pix_fmt_name(src_pix_fmt), src_w, src_h,
av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h);
ret = AVERROR(EINVAL);
goto end;
}
/* allocate source and destination image buffers */
if ((ret = av_image_alloc(src_data, src_linesize,
src_w, src_h, src_pix_fmt, 16)) < 0) {
fprintf(stderr, "Could not allocate source image\n");
goto end;
}
/* buffer is going to be written to rawvideo file, no alignment */
if ((ret = av_image_alloc(dst_data, dst_linesize,
dst_w, dst_h, dst_pix_fmt, 1)) < 0) {
fprintf(stderr, "Could not allocate destination image\n");
goto end;
}
dst_bufsize = ret;
for (i = 0; i < 100; i++) {
/* generate synthetic video */
fill_yuv_image(src_data, src_linesize, src_w, src_h, i);
/* convert to destination format */
sws_scale(sws_ctx, (const uint8_t * const*)src_data,
src_linesize, 0, src_h, dst_data, dst_linesize);
/* write scaled image to file */
fwrite(dst_data[0], 1, dst_bufsize, dst_file);
}
fprintf(stderr, "Scaling succeeded. Play the output file with the command:\n"
"ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h, dst_filename);
end:
fclose(dst_file);
av_freep(&src_data[0]);
av_freep(&dst_data[0]);
sws_freeContext(sws_ctx);
return ret < 0;
}

117
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/doc/examples/transcode_aac.c → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/examples/transcode_aac.c
View file

@ -1,5 +1,5 @@
/*
* Copyright (c) 2013-2022 Andreas Unterweger
* Copyright (c) 2013-2018 Andreas Unterweger
*
* This file is part of FFmpeg.
*
@ -19,11 +19,12 @@
*/
/**
* @file audio transcoding to MPEG/AAC API usage example
* @example transcode_aac.c
* @file
* Simple audio converter
*
* Convert an input audio file to AAC in an MP4 container. Formats other than
* MP4 are supported based on the output file extension.
* @example transcode_aac.c
* Convert an input audio file to AAC in an MP4 container using FFmpeg.
* Formats other than MP4 are supported based on the output file extension.
* @author Andreas Unterweger (dustsigns@gmail.com)
*/
@ -37,7 +38,6 @@
#include "libavutil/audio_fifo.h"
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/channel_layout.h"
#include "libavutil/frame.h"
#include "libavutil/opt.h"
@ -60,8 +60,7 @@ static int open_input_file(const char *filename,
AVCodecContext **input_codec_context)
{
AVCodecContext *avctx;
const AVCodec *input_codec;
const AVStream *stream;
AVCodec *input_codec;
int error;
/* Open the input file to read from it. */
@ -89,10 +88,8 @@ static int open_input_file(const char *filename,
return AVERROR_EXIT;
}
stream = (*input_format_context)->streams[0];
/* Find a decoder for the audio stream. */
if (!(input_codec = avcodec_find_decoder(stream->codecpar->codec_id))) {
if (!(input_codec = avcodec_find_decoder((*input_format_context)->streams[0]->codecpar->codec_id))) {
fprintf(stderr, "Could not find input codec\n");
avformat_close_input(input_format_context);
return AVERROR_EXIT;
@ -107,7 +104,7 @@ static int open_input_file(const char *filename,
}
/* Initialize the stream parameters with demuxer information. */
error = avcodec_parameters_to_context(avctx, stream->codecpar);
error = avcodec_parameters_to_context(avctx, (*input_format_context)->streams[0]->codecpar);
if (error < 0) {
avformat_close_input(input_format_context);
avcodec_free_context(&avctx);
@ -123,9 +120,6 @@ static int open_input_file(const char *filename,
return error;
}
/* Set the packet timebase for the decoder. */
avctx->pkt_timebase = stream->time_base;
/* Save the decoder context for easier access later. */
*input_codec_context = avctx;
@ -150,7 +144,7 @@ static int open_output_file(const char *filename,
AVCodecContext *avctx = NULL;
AVIOContext *output_io_context = NULL;
AVStream *stream = NULL;
const AVCodec *output_codec = NULL;
AVCodec *output_codec = NULL;
int error;
/* Open the output file to write to it. */
@ -205,11 +199,15 @@ static int open_output_file(const char *filename,
/* Set the basic encoder parameters.
* The input file's sample rate is used to avoid a sample rate conversion. */
av_channel_layout_default(&avctx->ch_layout, OUTPUT_CHANNELS);
avctx->channels = OUTPUT_CHANNELS;
avctx->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
avctx->sample_rate = input_codec_context->sample_rate;
avctx->sample_fmt = output_codec->sample_fmts[0];
avctx->bit_rate = OUTPUT_BIT_RATE;
/* Allow the use of the experimental AAC encoder. */
avctx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
/* Set the sample rate for the container. */
stream->time_base.den = input_codec_context->sample_rate;
stream->time_base.num = 1;
@ -247,16 +245,14 @@ cleanup:
/**
* Initialize one data packet for reading or writing.
* @param[out] packet Packet to be initialized
* @return Error code (0 if successful)
* @param packet Packet to be initialized
*/
static int init_packet(AVPacket **packet)
static void init_packet(AVPacket *packet)
{
if (!(*packet = av_packet_alloc())) {
fprintf(stderr, "Could not allocate packet\n");
return AVERROR(ENOMEM);
}
return 0;
av_init_packet(packet);
/* Set the packet data and size so that it is recognized as being empty. */
packet->data = NULL;
packet->size = 0;
}
/**
@ -291,18 +287,21 @@ static int init_resampler(AVCodecContext *input_codec_context,
/*
* Create a resampler context for the conversion.
* Set the conversion parameters.
* Default channel layouts based on the number of channels
* are assumed for simplicity (they are sometimes not detected
* properly by the demuxer and/or decoder).
*/
error = swr_alloc_set_opts2(resample_context,
&output_codec_context->ch_layout,
*resample_context = swr_alloc_set_opts(NULL,
av_get_default_channel_layout(output_codec_context->channels),
output_codec_context->sample_fmt,
output_codec_context->sample_rate,
&input_codec_context->ch_layout,
av_get_default_channel_layout(input_codec_context->channels),
input_codec_context->sample_fmt,
input_codec_context->sample_rate,
0, NULL);
if (error < 0) {
if (!*resample_context) {
fprintf(stderr, "Could not allocate resample context\n");
return error;
return AVERROR(ENOMEM);
}
/*
* Perform a sanity check so that the number of converted samples is
@ -330,7 +329,7 @@ static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
{
/* Create the FIFO buffer based on the specified output sample format. */
if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
output_codec_context->ch_layout.nb_channels, 1))) {
output_codec_context->channels, 1))) {
fprintf(stderr, "Could not allocate FIFO\n");
return AVERROR(ENOMEM);
}
@ -372,33 +371,28 @@ static int decode_audio_frame(AVFrame *frame,
int *data_present, int *finished)
{
/* Packet used for temporary storage. */
AVPacket *input_packet;
AVPacket input_packet;
int error;
init_packet(&input_packet);
error = init_packet(&input_packet);
if (error < 0)
return error;
*data_present = 0;
*finished = 0;
/* Read one audio frame from the input file into a temporary packet. */
if ((error = av_read_frame(input_format_context, input_packet)) < 0) {
if ((error = av_read_frame(input_format_context, &input_packet)) < 0) {
/* If we are at the end of the file, flush the decoder below. */
if (error == AVERROR_EOF)
*finished = 1;
else {
fprintf(stderr, "Could not read frame (error '%s')\n",
av_err2str(error));
goto cleanup;
return error;
}
}
/* Send the audio frame stored in the temporary packet to the decoder.
* The input audio stream decoder is used to do this. */
if ((error = avcodec_send_packet(input_codec_context, input_packet)) < 0) {
if ((error = avcodec_send_packet(input_codec_context, &input_packet)) < 0) {
fprintf(stderr, "Could not send packet for decoding (error '%s')\n",
av_err2str(error));
goto cleanup;
return error;
}
/* Receive one frame from the decoder. */
@ -424,7 +418,7 @@ static int decode_audio_frame(AVFrame *frame,
}
cleanup:
av_packet_free(&input_packet);
av_packet_unref(&input_packet);
return error;
}
@ -450,7 +444,7 @@ static int init_converted_samples(uint8_t ***converted_input_samples,
* Each pointer will later point to the audio samples of the corresponding
* channels (although it may be NULL for interleaved formats).
*/
if (!(*converted_input_samples = calloc(output_codec_context->ch_layout.nb_channels,
if (!(*converted_input_samples = calloc(output_codec_context->channels,
sizeof(**converted_input_samples)))) {
fprintf(stderr, "Could not allocate converted input sample pointers\n");
return AVERROR(ENOMEM);
@ -459,7 +453,7 @@ static int init_converted_samples(uint8_t ***converted_input_samples,
/* Allocate memory for the samples of all channels in one consecutive
* block for convenience. */
if ((error = av_samples_alloc(*converted_input_samples, NULL,
output_codec_context->ch_layout.nb_channels,
output_codec_context->channels,
frame_size,
output_codec_context->sample_fmt, 0)) < 0) {
fprintf(stderr,
@ -559,7 +553,7 @@ static int read_decode_convert_and_store(AVAudioFifo *fifo,
AVFrame *input_frame = NULL;
/* Temporary storage for the converted input samples. */
uint8_t **converted_input_samples = NULL;
int data_present;
int data_present = 0;
int ret = AVERROR_EXIT;
/* Initialize temporary storage for one input frame. */
@ -633,7 +627,7 @@ static int init_output_frame(AVFrame **frame,
* Default channel layouts based on the number of channels
* are assumed for simplicity. */
(*frame)->nb_samples = frame_size;
av_channel_layout_copy(&(*frame)->ch_layout, &output_codec_context->ch_layout);
(*frame)->channel_layout = output_codec_context->channel_layout;
(*frame)->format = output_codec_context->sample_fmt;
(*frame)->sample_rate = output_codec_context->sample_rate;
@ -667,12 +661,9 @@ static int encode_audio_frame(AVFrame *frame,
int *data_present)
{
/* Packet used for temporary storage. */
AVPacket *output_packet;
AVPacket output_packet;
int error;
error = init_packet(&output_packet);
if (error < 0)
return error;
init_packet(&output_packet);
/* Set a timestamp based on the sample rate for the container. */
if (frame) {
@ -680,20 +671,21 @@ static int encode_audio_frame(AVFrame *frame,
pts += frame->nb_samples;
}
*data_present = 0;
/* Send the audio frame stored in the temporary packet to the encoder.
* The output audio stream encoder is used to do this. */
error = avcodec_send_frame(output_codec_context, frame);
/* Check for errors, but proceed with fetching encoded samples if the
* encoder signals that it has nothing more to encode. */
if (error < 0 && error != AVERROR_EOF) {
fprintf(stderr, "Could not send packet for encoding (error '%s')\n",
av_err2str(error));
goto cleanup;
/* The encoder signals that it has nothing more to encode. */
if (error == AVERROR_EOF) {
error = 0;
goto cleanup;
} else if (error < 0) {
fprintf(stderr, "Could not send packet for encoding (error '%s')\n",
av_err2str(error));
return error;
}
/* Receive one encoded frame from the encoder. */
error = avcodec_receive_packet(output_codec_context, output_packet);
error = avcodec_receive_packet(output_codec_context, &output_packet);
/* If the encoder asks for more data to be able to provide an
* encoded frame, return indicating that no data is present. */
if (error == AVERROR(EAGAIN)) {
@ -714,14 +706,14 @@ static int encode_audio_frame(AVFrame *frame,
/* Write one audio frame from the temporary packet to the output file. */
if (*data_present &&
(error = av_write_frame(output_format_context, output_packet)) < 0) {
(error = av_write_frame(output_format_context, &output_packet)) < 0) {
fprintf(stderr, "Could not write frame (error '%s')\n",
av_err2str(error));
goto cleanup;
}
cleanup:
av_packet_free(&output_packet);
av_packet_unref(&output_packet);
return error;
}
@ -860,6 +852,7 @@ int main(int argc, char **argv)
int data_written;
/* Flush the encoder as it may have delayed frames. */
do {
data_written = 0;
if (encode_audio_frame(NULL, output_format_context,
output_codec_context, &data_written))
goto cleanup;

620
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/examples/transcoding.c Normal file
View file

@ -0,0 +1,620 @@
/*
* Copyright (c) 2010 Nicolas George
* Copyright (c) 2011 Stefano Sabatini
* Copyright (c) 2014 Andrey Utkin
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* API example for demuxing, decoding, filtering, encoding and muxing
* @example transcoding.c
*/
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
#include <libavutil/pixdesc.h>
static AVFormatContext *ifmt_ctx;
static AVFormatContext *ofmt_ctx;
typedef struct FilteringContext {
AVFilterContext *buffersink_ctx;
AVFilterContext *buffersrc_ctx;
AVFilterGraph *filter_graph;
} FilteringContext;
static FilteringContext *filter_ctx;
typedef struct StreamContext {
AVCodecContext *dec_ctx;
AVCodecContext *enc_ctx;
} StreamContext;
static StreamContext *stream_ctx;
static int open_input_file(const char *filename)
{
int ret;
unsigned int i;
ifmt_ctx = NULL;
if ((ret = avformat_open_input(&ifmt_ctx, filename, NULL, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
return ret;
}
if ((ret = avformat_find_stream_info(ifmt_ctx, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
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);
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;
}
/* 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);
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);
return 0;
}
static int open_output_file(const char *filename)
{
AVStream *out_stream;
AVStream *in_stream;
AVCodecContext *dec_ctx, *enc_ctx;
AVCodec *encoder;
int ret;
unsigned int i;
ofmt_ctx = NULL;
avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, filename);
if (!ofmt_ctx) {
av_log(NULL, AV_LOG_ERROR, "Could not create output context\n");
return AVERROR_UNKNOWN;
}
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
out_stream = avformat_new_stream(ofmt_ctx, NULL);
if (!out_stream) {
av_log(NULL, AV_LOG_ERROR, "Failed allocating output stream\n");
return AVERROR_UNKNOWN;
}
in_stream = ifmt_ctx->streams[i];
dec_ctx = stream_ctx[i].dec_ctx;
if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
|| dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
/* in this example, we choose transcoding to same codec */
encoder = avcodec_find_encoder(dec_ctx->codec_id);
if (!encoder) {
av_log(NULL, AV_LOG_FATAL, "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
* streams easily using filters */
if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) {
enc_ctx->height = dec_ctx->height;
enc_ctx->width = dec_ctx->width;
enc_ctx->sample_aspect_ratio = dec_ctx->sample_aspect_ratio;
/* take first format from list of supported formats */
if (encoder->pix_fmts)
enc_ctx->pix_fmt = encoder->pix_fmts[0];
else
enc_ctx->pix_fmt = dec_ctx->pix_fmt;
/* video time_base can be set to whatever is handy and supported by encoder */
enc_ctx->time_base = av_inv_q(dec_ctx->framerate);
} else {
enc_ctx->sample_rate = dec_ctx->sample_rate;
enc_ctx->channel_layout = dec_ctx->channel_layout;
enc_ctx->channels = av_get_channel_layout_nb_channels(enc_ctx->channel_layout);
/* take first format from list of supported formats */
enc_ctx->sample_fmt = encoder->sample_fmts[0];
enc_ctx->time_base = (AVRational){1, enc_ctx->sample_rate};
}
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);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Copying parameters for stream #%u failed\n", i);
return ret;
}
out_stream->time_base = in_stream->time_base;
}
}
av_dump_format(ofmt_ctx, 0, filename, 1);
if (!(ofmt_ctx->oformat->flags & AVFMT_NOFILE)) {
ret = avio_open(&ofmt_ctx->pb, filename, AVIO_FLAG_WRITE);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Could not open output file '%s'", filename);
return ret;
}
}
/* init muxer, write output file header */
ret = avformat_write_header(ofmt_ctx, NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error occurred when opening output file\n");
return ret;
}
return 0;
}
static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
AVCodecContext *enc_ctx, const char *filter_spec)
{
char args[512];
int ret = 0;
const AVFilter *buffersrc = NULL;
const AVFilter *buffersink = NULL;
AVFilterContext *buffersrc_ctx = NULL;
AVFilterContext *buffersink_ctx = NULL;
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
AVFilterGraph *filter_graph = avfilter_graph_alloc();
if (!outputs || !inputs || !filter_graph) {
ret = AVERROR(ENOMEM);
goto end;
}
if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) {
buffersrc = avfilter_get_by_name("buffer");
buffersink = avfilter_get_by_name("buffersink");
if (!buffersrc || !buffersink) {
av_log(NULL, AV_LOG_ERROR, "filtering source or sink element not found\n");
ret = AVERROR_UNKNOWN;
goto end;
}
snprintf(args, sizeof(args),
"video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
dec_ctx->time_base.num, dec_ctx->time_base.den,
dec_ctx->sample_aspect_ratio.num,
dec_ctx->sample_aspect_ratio.den);
ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
args, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n");
goto end;
}
ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
NULL, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
goto end;
}
ret = av_opt_set_bin(buffersink_ctx, "pix_fmts",
(uint8_t*)&enc_ctx->pix_fmt, sizeof(enc_ctx->pix_fmt),
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n");
goto end;
}
} else if (dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
buffersrc = avfilter_get_by_name("abuffer");
buffersink = avfilter_get_by_name("abuffersink");
if (!buffersrc || !buffersink) {
av_log(NULL, AV_LOG_ERROR, "filtering source or sink element not found\n");
ret = AVERROR_UNKNOWN;
goto end;
}
if (!dec_ctx->channel_layout)
dec_ctx->channel_layout =
av_get_default_channel_layout(dec_ctx->channels);
snprintf(args, sizeof(args),
"time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
dec_ctx->time_base.num, dec_ctx->time_base.den, dec_ctx->sample_rate,
av_get_sample_fmt_name(dec_ctx->sample_fmt),
dec_ctx->channel_layout);
ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
args, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
goto end;
}
ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
NULL, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
goto end;
}
ret = av_opt_set_bin(buffersink_ctx, "sample_fmts",
(uint8_t*)&enc_ctx->sample_fmt, sizeof(enc_ctx->sample_fmt),
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
goto end;
}
ret = av_opt_set_bin(buffersink_ctx, "channel_layouts",
(uint8_t*)&enc_ctx->channel_layout,
sizeof(enc_ctx->channel_layout), AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
goto end;
}
ret = av_opt_set_bin(buffersink_ctx, "sample_rates",
(uint8_t*)&enc_ctx->sample_rate, sizeof(enc_ctx->sample_rate),
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
goto end;
}
} else {
ret = AVERROR_UNKNOWN;
goto end;
}
/* Endpoints for the filter graph. */
outputs->name = av_strdup("in");
outputs->filter_ctx = buffersrc_ctx;
outputs->pad_idx = 0;
outputs->next = NULL;
inputs->name = av_strdup("out");
inputs->filter_ctx = buffersink_ctx;
inputs->pad_idx = 0;
inputs->next = NULL;
if (!outputs->name || !inputs->name) {
ret = AVERROR(ENOMEM);
goto end;
}
if ((ret = avfilter_graph_parse_ptr(filter_graph, filter_spec,
&inputs, &outputs, NULL)) < 0)
goto end;
if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
goto end;
/* Fill FilteringContext */
fctx->buffersrc_ctx = buffersrc_ctx;
fctx->buffersink_ctx = buffersink_ctx;
fctx->filter_graph = filter_graph;
end:
avfilter_inout_free(&inputs);
avfilter_inout_free(&outputs);
return ret;
}
static int init_filters(void)
{
const char *filter_spec;
unsigned int i;
int ret;
filter_ctx = av_malloc_array(ifmt_ctx->nb_streams, sizeof(*filter_ctx));
if (!filter_ctx)
return AVERROR(ENOMEM);
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
filter_ctx[i].buffersrc_ctx = NULL;
filter_ctx[i].buffersink_ctx = NULL;
filter_ctx[i].filter_graph = NULL;
if (!(ifmt_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO
|| ifmt_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_VIDEO))
continue;
if (ifmt_ctx->streams[i]->codecpar->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);
if (ret)
return ret;
}
return 0;
}
static int encode_write_frame(AVFrame *filt_frame, unsigned int stream_index, int *got_frame) {
int ret;
int got_frame_local;
AVPacket enc_pkt;
int (*enc_func)(AVCodecContext *, AVPacket *, const AVFrame *, int *) =
(ifmt_ctx->streams[stream_index]->codecpar->codec_type ==
AVMEDIA_TYPE_VIDEO) ? avcodec_encode_video2 : avcodec_encode_audio2;
if (!got_frame)
got_frame = &got_frame_local;
av_log(NULL, AV_LOG_INFO, "Encoding frame\n");
/* encode filtered frame */
enc_pkt.data = NULL;
enc_pkt.size = 0;
av_init_packet(&enc_pkt);
ret = enc_func(stream_ctx[stream_index].enc_ctx, &enc_pkt,
filt_frame, got_frame);
av_frame_free(&filt_frame);
if (ret < 0)
return ret;
if (!(*got_frame))
return 0;
/* prepare packet for muxing */
enc_pkt.stream_index = stream_index;
av_packet_rescale_ts(&enc_pkt,
stream_ctx[stream_index].enc_ctx->time_base,
ofmt_ctx->streams[stream_index]->time_base);
av_log(NULL, AV_LOG_DEBUG, "Muxing frame\n");
/* mux encoded frame */
ret = av_interleaved_write_frame(ofmt_ctx, &enc_pkt);
return ret;
}
static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index)
{
int ret;
AVFrame *filt_frame;
av_log(NULL, AV_LOG_INFO, "Pushing decoded frame to filters\n");
/* push the decoded frame into the filtergraph */
ret = av_buffersrc_add_frame_flags(filter_ctx[stream_index].buffersrc_ctx,
frame, 0);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
return ret;
}
/* pull filtered frames from the filtergraph */
while (1) {
filt_frame = av_frame_alloc();
if (!filt_frame) {
ret = AVERROR(ENOMEM);
break;
}
av_log(NULL, AV_LOG_INFO, "Pulling filtered frame from filters\n");
ret = av_buffersink_get_frame(filter_ctx[stream_index].buffersink_ctx,
filt_frame);
if (ret < 0) {
/* if no more frames for output - returns AVERROR(EAGAIN)
* if flushed and no more frames for output - returns AVERROR_EOF
* rewrite retcode to 0 to show it as normal procedure completion
*/
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
ret = 0;
av_frame_free(&filt_frame);
break;
}
filt_frame->pict_type = AV_PICTURE_TYPE_NONE;
ret = encode_write_frame(filt_frame, stream_index, NULL);
if (ret < 0)
break;
}
return ret;
}
static int flush_encoder(unsigned int stream_index)
{
int ret;
int got_frame;
if (!(stream_ctx[stream_index].enc_ctx->codec->capabilities &
AV_CODEC_CAP_DELAY))
return 0;
while (1) {
av_log(NULL, AV_LOG_INFO, "Flushing stream #%u encoder\n", stream_index);
ret = encode_write_frame(NULL, stream_index, &got_frame);
if (ret < 0)
break;
if (!got_frame)
return 0;
}
return ret;
}
int main(int argc, char **argv)
{
int ret;
AVPacket packet = { .data = NULL, .size = 0 };
AVFrame *frame = NULL;
enum AVMediaType type;
unsigned int stream_index;
unsigned int i;
int got_frame;
int (*dec_func)(AVCodecContext *, AVFrame *, int *, const AVPacket *);
if (argc != 3) {
av_log(NULL, AV_LOG_ERROR, "Usage: %s <input file> <output file>\n", argv[0]);
return 1;
}
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if ((ret = open_output_file(argv[2])) < 0)
goto end;
if ((ret = init_filters()) < 0)
goto end;
/* read all packets */
while (1) {
if ((ret = av_read_frame(ifmt_ctx, &packet)) < 0)
break;
stream_index = packet.stream_index;
type = ifmt_ctx->streams[packet.stream_index]->codecpar->codec_type;
av_log(NULL, AV_LOG_DEBUG, "Demuxer gave frame of stream_index %u\n",
stream_index);
if (filter_ctx[stream_index].filter_graph) {
av_log(NULL, AV_LOG_DEBUG, "Going to reencode&filter the frame\n");
frame = av_frame_alloc();
if (!frame) {
ret = AVERROR(ENOMEM);
break;
}
av_packet_rescale_ts(&packet,
ifmt_ctx->streams[stream_index]->time_base,
stream_ctx[stream_index].dec_ctx->time_base);
dec_func = (type == AVMEDIA_TYPE_VIDEO) ? avcodec_decode_video2 :
avcodec_decode_audio4;
ret = dec_func(stream_ctx[stream_index].dec_ctx, frame,
&got_frame, &packet);
if (ret < 0) {
av_frame_free(&frame);
av_log(NULL, AV_LOG_ERROR, "Decoding failed\n");
break;
}
if (got_frame) {
frame->pts = frame->best_effort_timestamp;
ret = filter_encode_write_frame(frame, stream_index);
av_frame_free(&frame);
if (ret < 0)
goto end;
} else {
av_frame_free(&frame);
}
} else {
/* remux this frame without reencoding */
av_packet_rescale_ts(&packet,
ifmt_ctx->streams[stream_index]->time_base,
ofmt_ctx->streams[stream_index]->time_base);
ret = av_interleaved_write_frame(ofmt_ctx, &packet);
if (ret < 0)
goto end;
}
av_packet_unref(&packet);
}
/* flush filters and encoders */
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
/* flush filter */
if (!filter_ctx[i].filter_graph)
continue;
ret = filter_encode_write_frame(NULL, i);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Flushing filter failed\n");
goto end;
}
/* flush encoder */
ret = flush_encoder(i);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Flushing encoder failed\n");
goto end;
}
}
av_write_trailer(ofmt_ctx);
end:
av_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);
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);
avformat_free_context(ofmt_ctx);
if (ret < 0)
av_log(NULL, AV_LOG_ERROR, "Error occurred: %s\n", av_err2str(ret));
return ret ? 1 : 0;
}

224
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/examples/vaapi_encode.c Normal file
View file

@ -0,0 +1,224 @@
/*
* 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;
}

57
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/doc/examples/vaapi_transcode.c → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/examples/vaapi_transcode.c
View file

@ -1,4 +1,6 @@
/*
* 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
@ -19,10 +21,11 @@
*/
/**
* @file Intel VAAPI-accelerated transcoding API usage example
* @example vaapi_transcode.c
* @file
* Intel VAAPI-accelerated transcoding example.
*
* Perform VAAPI-accelerated transcoding.
* @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
@ -59,7 +62,7 @@ static enum AVPixelFormat get_vaapi_format(AVCodecContext *ctx,
static int open_input_file(const char *filename)
{
int ret;
const AVCodec *decoder = NULL;
AVCodec *decoder = NULL;
AVStream *video = NULL;
if ((ret = avformat_open_input(&ifmt_ctx, filename, NULL, NULL)) < 0) {
@ -106,25 +109,28 @@ static int open_input_file(const char *filename)
return ret;
}
static int encode_write(AVPacket *enc_pkt, AVFrame *frame)
static int encode_write(AVFrame *frame)
{
int ret = 0;
AVPacket enc_pkt;
av_packet_unref(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);
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,
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);
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));
@ -139,7 +145,7 @@ end:
return ret;
}
static int dec_enc(AVPacket *pkt, const AVCodec *enc_codec)
static int dec_enc(AVPacket *pkt, AVCodec *enc_codec)
{
AVFrame *frame;
int ret = 0;
@ -210,7 +216,7 @@ static int dec_enc(AVPacket *pkt, const AVCodec *enc_codec)
initialized = 1;
}
if ((ret = encode_write(pkt, frame)) < 0)
if ((ret = encode_write(frame)) < 0)
fprintf(stderr, "Error during encoding and writing.\n");
fail:
@ -223,9 +229,9 @@ fail:
int main(int argc, char **argv)
{
const AVCodec *enc_codec;
int ret = 0;
AVPacket *dec_pkt;
AVPacket dec_pkt;
AVCodec *enc_codec;
if (argc != 4) {
fprintf(stderr, "Usage: %s <input file> <encode codec> <output file>\n"
@ -240,12 +246,6 @@ int main(int argc, char **argv)
return -1;
}
dec_pkt = av_packet_alloc();
if (!dec_pkt) {
fprintf(stderr, "Failed to allocate decode packet\n");
goto end;
}
if ((ret = open_input_file(argv[1])) < 0)
goto end;
@ -275,21 +275,23 @@ int main(int argc, char **argv)
/* read all packets and only transcoding video */
while (ret >= 0) {
if ((ret = av_read_frame(ifmt_ctx, dec_pkt)) < 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);
if (video_stream == dec_pkt.stream_index)
ret = dec_enc(&dec_pkt, enc_codec);
av_packet_unref(dec_pkt);
av_packet_unref(&dec_pkt);
}
/* flush decoder */
av_packet_unref(dec_pkt);
ret = dec_enc(dec_pkt, enc_codec);
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(dec_pkt, NULL);
ret = encode_write(NULL);
/* write the trailer for output stream */
av_write_trailer(ofmt_ctx);
@ -300,6 +302,5 @@ end:
avcodec_free_context(&decoder_ctx);
avcodec_free_context(&encoder_ctx);
av_buffer_unref(&hw_device_ctx);
av_packet_free(&dec_pkt);
return ret;
}

12
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/doc/faq.texi → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/faq.texi
View file

@ -76,7 +76,7 @@ the gcc developers. Note that we will not add workarounds for gcc bugs.
Also note that (some of) the gcc developers believe this is not a bug or
not a bug they should fix:
@url{https://gcc.gnu.org/bugzilla/show_bug.cgi?id=11203}.
@url{http://gcc.gnu.org/bugzilla/show_bug.cgi?id=11203}.
Then again, some of them do not know the difference between an undecidable
problem and an NP-hard problem...
@ -257,13 +257,13 @@ default.
@section Which are good parameters for encoding high quality MPEG-4?
'-mbd rd -flags +mv4+aic -trellis 2 -cmp 2 -subcmp 2 -g 300 -pass 1/2',
things to try: '-bf 2', '-mpv_flags qp_rd', '-mpv_flags mv0', '-mpv_flags skip_rd'.
things to try: '-bf 2', '-flags qprd', '-flags mv0', '-flags skiprd'.
@section Which are good parameters for encoding high quality MPEG-1/MPEG-2?
'-mbd rd -trellis 2 -cmp 2 -subcmp 2 -g 100 -pass 1/2'
but beware the '-g 100' might cause problems with some decoders.
Things to try: '-bf 2', '-mpv_flags qp_rd', '-mpv_flags mv0', '-mpv_flags skip_rd'.
Things to try: '-bf 2', '-flags qprd', '-flags mv0', '-flags skiprd.
@section Interlaced video looks very bad when encoded with ffmpeg, what is wrong?
@ -516,7 +516,7 @@ 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 macOS)
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.
@ -526,7 +526,7 @@ For example:
ffmpeg -nostdin -i INPUT OUTPUT
@end example
or (on Linux, macOS, and other UNIX-like shells):
or (on Linux, Mac OS, and other UNIX-like shells):
@example
ffmpeg -i INPUT OUTPUT </dev/null
@ -601,7 +601,7 @@ No. These tools are too bloated and they complicate the build.
FFmpeg is already organized in a highly modular manner and does not need to
be rewritten in a formal object language. Further, many of the developers
favor straight C; it works for them. For more arguments on this matter,
read @uref{https://web.archive.org/web/20111004021423/http://kernel.org/pub/linux/docs/lkml/#s15, "Programming Religion"}.
read @uref{http://www.tux.org/lkml/#s15, "Programming Religion"}.
@section Why are the ffmpeg programs devoid of debugging symbols?

19
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/doc/fate.texi → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/fate.texi
View file

@ -79,21 +79,6 @@ Do not put a '~' character in the samples path to indicate a home
directory. Because of shell nuances, this will cause FATE to fail.
@end float
To get the complete list of tests, run the command:
@example
make fate-list
@end example
You can specify a subset of tests to run by specifying the
corresponding elements from the list with the @code{fate-} prefix,
e.g. as in:
@example
make fate-ffprobe_compact fate-ffprobe_xml
@end example
This makes it easier to run a few tests in case of failure without
running the complete test suite.
To use a custom wrapper to run the test, pass @option{--target-exec} to
@command{configure} or set the @var{TARGET_EXEC} Make variable.
@ -164,8 +149,6 @@ the synchronisation of the samples directory.
@chapter Uploading new samples to the fate suite
If you need a sample uploaded send a mail to samples-request.
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.
@ -174,8 +157,6 @@ 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.
If you need an account for frequently uploading samples or you wish to help
others by doing that send a mail to ffmpeg-devel.
@example
#First update your local samples copy:

0
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/doc/fate_config.sh.template → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/fate_config.sh.template
View file

0
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/doc/ffmpeg-bitstream-filters.texi → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/ffmpeg-bitstream-filters.texi
View file

0
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/doc/ffmpeg-codecs.texi → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/ffmpeg-codecs.texi
View file

0
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/doc/ffmpeg-devices.texi → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/ffmpeg-devices.texi
View file

0
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/doc/ffmpeg-filters.texi → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/ffmpeg-filters.texi
View file

0
submodules/ffmpeg/Sources/FFMpeg/ffmpeg-6.0/doc/ffmpeg-formats.texi → submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/doc/ffmpeg-formats.texi
View file

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