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104 Commits

Author SHA1 Message Date
Michael Niedermayer
6b6b9e593d Changelog: update
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-07-11 00:26:17 +02:00
Michael Niedermayer
5086d22697 avcodec/tiff: Check input space in dng_decode_jpeg()
Fixes: out of array read
Fixes: 24034/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_TIFF_fuzzer-5111884337119232

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 79e8d17024)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-07-11 00:25:33 +02:00
Michael Niedermayer
3c4679c430 avcodec/mjpeg_parser: Adjust size rejection threshold
Fixes: 86987846-429c8d80-c197-11ea-916b-bb4738e09687.jpg
Fixes: Regression since ec3d8a0e69

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit dde6077297)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-07-11 00:25:33 +02:00
Michael Niedermayer
832652a9d1 avcodec/cbs_jpeg: Fix uninitialized end index in cbs_jpeg_split_fragment()
Fixes: Out of array read
Fixes: 24043/clusterfuzz-testcase-minimized-ffmpeg_BSF_TRACE_HEADERS_fuzzer-5084566275751936.fuzz

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 4a10bc8f6f)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-07-11 00:25:33 +02:00
Andreas Rheinhardt
9ee65bf88d avformat/sdp: Fix potential write beyond end of buffer
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit 5d91b7718e)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
2020-07-10 20:52:00 +02:00
Andreas Rheinhardt
be84216c53 avformat/mm: Check for existence of audio stream
No audio stream is created unconditionally and if none has been created,
no packet with stream_index 1 may be returned. This fixes an assert in
ff_read_packet() in libavformat/utils reported in ticket #8782.

Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit ec59dc73f0)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
2020-07-10 20:52:00 +02:00
Michael Niedermayer
401b59e4c3 Update for 4.3.1
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-07-09 22:17:30 +02:00
Zhao Zhili
d4ced9ebb7 avformat/mov: Fix unaligned read of uint32_t and endian-dependance in mov_read_default
Reviewed-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 806a4d5187)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-07-09 12:35:39 +02:00
Michael Niedermayer
b021eba8b6 avcodec/apedec: Fix undefined integer overflow with 24bit
Fixes: signed integer overflow: 8683744 * 256 cannot be represented in type 'int'
Fixes: 23527/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_APE_fuzzer-5679885932822528

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 9f7b252cdf)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-07-09 12:35:39 +02:00
Michael Niedermayer
093c2dd644 avcodec/loco: Fix integer overflow with large values from loco_get_rice()
Fixes: signed integer overflow: 155 + 2147483647 cannot be represented in type 'int'
Fixes: 23421/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_LOCO_fuzzer-5652849097965568

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 3ddc5e1f3c)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-07-09 12:35:39 +02:00
Michael Niedermayer
99eb08f390 avformat/smjpegdec: Check the existence of referred streams
Fixes: Assertion failure
Fixes: 23758/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-5160954605338624.fuzz

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 321ea59dac)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-07-09 12:35:39 +02:00
Michael Niedermayer
b228e0c5f6 avcodec/tiff: Check frame parameters before blit for DNG
Fixes: out of array access
Fixes: 23888/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_TIFF_fuzzer-6021365974171648.fuzz

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 4091f4f780)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-07-09 12:35:39 +02:00
Michael Niedermayer
11a10e30a9 avcodec/mjpegdec: Limit bayer to single plane outputting format
This reduces the number of paths reachable with DNG and should
improve security

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 865a34970e)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-07-09 12:35:39 +02:00
Michael Niedermayer
f98f29de5e avcodec/pnmdec: Fix misaligned reads
Found-by: "Steinar H. Gunderson" <steinar+ffmpeg@gunderson.no>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit ea28ce9bc1)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-07-09 12:35:39 +02:00
Michael Niedermayer
531ddbacb5 avcodec/mv30: Fix integer overflows in idct2_1d()
Fixes: signed integer overflow: 6500736 * 473 cannot be represented in type 'int'
Fixes: 23259/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_MV30_fuzzer-5179394271477760

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 3b8d5bcc31)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-07-09 12:35:39 +02:00
Michael Niedermayer
d25345bb00 avcodec/hcadec: Check total_band_count against imdct_in size
Fixes: index 128 out of bounds for type 'float [128]'
Fixes: 23465/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_HCA_fuzzer-5089866596745216

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 2d96c94531)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-07-09 12:35:39 +02:00
Michael Niedermayer
1ff86cb452 avcodec/scpr3: Fix out of array access with dectab
Fixes: 23721/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SCPR_fuzzer-5914074721550336

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit c8de8dfba6)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-07-09 12:35:39 +02:00
Michael Niedermayer
f1ebea7c91 avcodec/tiff: Do not overrun the array ends in dng_blit()
Fixes: out of array access
Fixes: 23589/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_TIFF_fuzzer-5110559589793792.fuzz

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit f35caea77f)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-07-09 12:35:39 +02:00
Michael Niedermayer
c86a9d5b82 avcodec/dstdec: Replace AC overread check by sample rate check
Real files do skip coding 0 bits at the end, thus this kind of check
does not work reliable.

Fixes: Ticket 8770
Fixes: dst-256fs44-6ch-refdstencoder.dff

The samplerate is specified in ISO/IEC 14496-3:2005(E) as one of 3 fixed
values, this also can be used to limit the duration and avoid the timeout

This reverts commit f6df99dba1.

(cherry picked from commit 1679f23beb)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-07-09 12:35:39 +02:00
Reimar Döffinger
1f32d8ea23 dnn_backend_native: Add overflow check for length calculation.
We should not silently allocate an incorrect sized buffer.
Fixes trac issue #8718.

Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Reviewed-by: Guo, Yejun <yejun.guo@intel.com>
2020-07-06 20:25:50 +08:00
Andreas Rheinhardt
7cbb6ee2ee avcodec/h264_metadata_bsf: Fix invalid av_freep
This bug was introduced in 3c8a2a1180.

Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit 04e06beb0a)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
2020-07-04 22:33:21 +02:00
James Almer
acefb59ac5 avcodec/cbs_h265: set default VUI parameters when vui_parameters_present_flag is false
Based on cbs_h264 code.

Should fix ticket #8752.

Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit d1c55fc460)
2020-07-02 22:26:39 -03:00
Manoj Bonda
797574400d avcodec/av1_parser: initialize avctx->pix_fmt
Initialize avctx->pix_fmt in av1_parser.c
AV1 Chroma format is invalid when quering using below code if no AV1 decoder
is available:

iVideoStream = av_find_best_stream(fmtc, AVMEDIA_TYPE_VIDEO, -1, -1, NULL, 0);
eChromaFormat = (AVPixelFormat)fmtc->streams[iVideoStream]->codecpar->format;

Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit 23d06f606e)
2020-07-02 22:26:39 -03:00
James Almer
b303fe926e avcodec/av1_parser: add missing parsing for RGB pixel format signaling
Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit af6cddae1f)
2020-07-02 22:26:39 -03:00
James Almer
8f5f453998 avcodec/av1_parser: set context values outside the OBU parsing loop
Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit 634a44db5a)
2020-07-02 22:26:39 -03:00
Michael Niedermayer
836f6fb567 avutil/avsscanf: Add () to avoid integer overflow in scanexp()
Fixes: signed integer overflow: 2147483610 + 52 cannot be represented in type 'int'
Fixes: 23260/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_PBM_fuzzer-5187871274434560

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 42b28565aa)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-07-02 10:20:36 +02:00
Michael Niedermayer
3571d9d654 avformat/utils: reorder duration computation to avoid overflow
Fixes: signed integer overflow: 8 * 9223372036854774783 cannot be represented in type 'long'
Fixes: 23381/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-4818340509122560

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 10cc82c35b)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-07-02 10:20:36 +02:00
Michael Niedermayer
f27a510211 avcodec/pngdec: Check for fctl after idat
Fixes: out of array access
Fixes: 23554/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_APNG_fuzzer-4796622520451072.fuzz

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 65b1ba680f)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-07-02 10:20:36 +02:00
Michael Niedermayer
a3fdeb0c3a avformat/hls: Pass a copy of the URL for probing
The segments / url can be modified by the io read when reloading

This may be an alternative or additional fix for Ticket8673
as a further alternative the reload stuff could be disabled during
probing

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit b5e39880fb)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-07-02 10:20:36 +02:00
Michael Niedermayer
199d6a049a avutil/common: Fix integer overflow in av_ceil_log2_c()
Fixes: left shift of 1913647649 by 1 places cannot be represented in type 'int'
Fixes: 23572/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_WMALOSSLESS_fuzzer-5082619795734528

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit e409262837)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-07-02 10:20:36 +02:00
Michael Niedermayer
f4affa071a avcodec/wmalosslessdec: fix overflow with pred in revert_cdlms
Fixes: signed integer overflow: 2048 + 2147483646 cannot be represented in type 'int'
Fixes: 23538/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_WMALOSSLESS_fuzzer-5227567073460224

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 21598d711d)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-07-02 10:20:36 +02:00
Michael Niedermayer
c05d51c067 avformat/mvdec: Fix integer overflow with billions of channels
Fixes: signed integer overflow: 1394614304 * 2 cannot be represented in type 'int'
Fixes: 23491/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-5697377020411904

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit b6fbbe08c3)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-07-02 10:20:36 +02:00
Michael Niedermayer
3ce81bf960 avformat/microdvddec: skip malformed lines without frame number.
Fixes: signed integer overflow: 1 - -9223372036854775808 cannot be represented in type 'long'
Fixes: 23490/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-5133490093031424

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit a8fb7612a9)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-07-02 10:20:36 +02:00
Guo Yejun
dd273d359e dnn_backend_native: check operand index
it fixed the issue in https://trac.ffmpeg.org/ticket/8716
(cherry-pick from 0b3bd001ac)
Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
2020-07-02 09:03:24 +08:00
Guo Yejun
5530748bfd dnn_backend_native.c: refine code for fail case
(cherry-pick from fc932195ab)
Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
2020-07-02 09:01:41 +08:00
Zhao Zhili
143e2d0d66 avformat/mov: fix memleaks
Fix two cases of memleaks:
1. The leak of dv_demux
2. The leak of dv_fctx upon dv_demux allocate failure

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit f3dc38a186)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
2020-07-01 19:02:29 +02:00
Andreas Rheinhardt
7c1ad9d151 libavformat/mov: Fix memleaks when demuxing DV audio
The code for demuxing DV audio predates the introduction of refcounted
packets and when the latter was added, changes to the former were
forgotten. This meant that when avpriv_dv_produce_packet initialized the
packet containing the AVBufferRef, the AVBufferRef as well as the
underlying AVBuffer leaked; the actual packet data didn't leak: They
were directly freed, but not via their AVBuffer's free function.

https://samples.ffmpeg.org/ffmpeg-bugs/trac/ticket4671/dir1.tar.bz2
contains samples for this (enable_drefs needs to be enabled for them).

Moreover, errors in avpriv_dv_produce_packet were ignored; this has been
changed, too.

Furthermore, in the hypothetical scenario that the track has a palette,
this would leak, too, so reorder the code so that the palette code
appears after the DV audio code.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit 61f5c6ab06)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
2020-07-01 19:01:56 +02:00
Andreas Rheinhardt
b3d8e13a88 avcodec/cbs_av1: Fix writing uvlc numbers >= INT_MAX
Fixes: assertion failure
Fixes: left shift of 1 by 31 places cannot be represented in type 'int'
Fixes: 23264/clusterfuzz-testcase-minimized-ffmpeg_BSF_AV1_METADATA_fuzzer-6308429248593920

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit 6f06c17a55)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
2020-07-01 19:01:31 +02:00
Andreas Rheinhardt
3cf212f6c8 avformat/avc, mxfenc: Avoid allocation of H264 SPS structure, fix memleak
Up until now, ff_avc_decode_sps would parse a SPS and return some
properties from it in a freshly allocated structure. Yet said structure
is very small and completely internal to libavformat, so there is no
reason to use the heap for it. This commit therefore changes the
function to return an int and to modify a caller-provided structure.
This will also allow ff_avc_decode_sps to return better error codes in
the future.

It also fixes a memleak in mxfenc: If a packet contained multiple SPS,
only the SPS structure belonging to the last SPS would be freed, the
other ones would leak when the pointer is overwritten to point to the
new SPS structure. Of course, without allocations there are no leaks.
This is Coverity issue #1445194.

Furthermore, the SPS structure has been renamed from
H264SequenceParameterSet to H264SPS in order to avoid overlong lines.

Reviewed-by: Tomas Härdin <tjoppen@acc.umu.se>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit a0b6df0a39)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
2020-07-01 19:00:19 +02:00
Andreas Rheinhardt
284fffa92f avcodec/bitstream: Don't check for undefined behaviour after it happened
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit 5e196dac22)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
2020-07-01 18:59:57 +02:00
Andreas Rheinhardt
d8407afe02 avformat/aviobuf: Also return truncated buffer in avio_get_dyn_buf()
Two kinds of errors can happen when working with dynamic buffers:
(Re)allocation errors or truncation errors (one has to truncate the
buffer to a size of INT_MAX because avio_close_dyn_buf() and
avio_get_dyn_buf() both return an int). Right now, avio_get_dyn_buf()
returns an empty buffer in either case. But given that
avio_get_dyn_buf() does not destroy the dynamic buffer, one can return
the buffer in case of truncation and let the user check the error flags
and decide for himself instead of hardcoding a single way to proceed
in case of truncation.

(This actually restores the behaviour from before commit
163bb9ac0af495a5cb95441bdb5c02170440d28c.)

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit c33e56c7a6)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
2020-07-01 18:59:27 +02:00
Andreas Rheinhardt
b6546add07 avformat/aviobuf: Don't check for overflow after it happened
If adding two ints overflows, it doesn't matter whether the result will
be stored in an unsigned or not; and checking afterwards does not make it
retroactively defined.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit 28a078eded)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
2020-07-01 18:58:10 +02:00
Michael Niedermayer
8e12af29d1 avcodec/tiff: Check stride for dng
Fixes: assertion failure
Fixes: 23422/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_TIFF_fuzzer-5746026064642048

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 276dfa9d91)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-06-15 20:41:15 +02:00
Andreas Rheinhardt
716b5c6ec9 avformat/mov: Fix reel_name size check
Only read str_size bytes from offset 30 of extradata if the extradata is
indeed at least 30 + str_size bytes long.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit ff3fad6b0e)
2020-06-15 17:30:33 +02:00
Andreas Rheinhardt
9d921e38f4 avformat/mov: Fix memleak upon encountering repeating tags
mov_read_custom tries to read three strings belonging to three different
tags. When an already encountered tag is encountered again, a new buffer
for the string to be read is allocated and stored in the pointer
destined for this particular tag. But in this scenario, said pointer
already holds the address of the string read earlier, leading to a leak.

This commit therefore aborts the reading process upon encountering
an already encountered tag.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit dfef1d5e3c)
2020-06-15 17:30:33 +02:00
Andreas Rheinhardt
c49dfee90b avformat/matroskaenc: Don't use NULL for %s format string
The argument pertaining to a printf %s conversion specifier must not
be NULL, even if the precision (i.e. the number of characters to write)
is zero. If it is NULL, it is undefined behaviour.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit 6de6ce7bc8)
2020-06-15 17:30:33 +02:00
Andreas Rheinhardt
3f3cfddb37 avformat/webvttdec: Fix memleak upon read header failure
The already parsed subtitles (contained in an FFDemuxSubtitlesQueue)
would leak if an error happened upon reading a subsequent subtitle.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit c784fe8b86)
2020-06-15 17:30:33 +02:00
Andreas Rheinhardt
b7897f0319 avformat/vplayerdec: Fix memleak upon read header failure
The already parsed subtitles (contained in an FFDemuxSubtitlesQueue)
would leak if an error happened upon reading a subsequent subtitle.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit 67434afa7f)
2020-06-15 17:30:33 +02:00
Andreas Rheinhardt
6eac7d79f4 avformat/tedcaptionsdec: Fix memleak upon read header failure
The already parsed subtitles (contained in an FFDemuxSubtitlesQueue)
would leak if allocating the AVStream for the subtitles fails.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit 337783b118)
2020-06-15 17:30:33 +02:00
Andreas Rheinhardt
04e1d16f65 avformat/subviewerdec: Fix memleak upon read header failure
The already parsed subtitles (contained in an FFDemuxSubtitlesQueue)
would leak if an error happened upon reading a subsequent subtitle.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit a708f65273)
2020-06-15 17:30:32 +02:00
Andreas Rheinhardt
49b60a9a52 avformat/subviewer1dec: Fix memleak upon read header failure
The already parsed subtitles (contained in an FFDemuxSubtitlesQueue)
would leak if an error happened upon reading a subsequent subtitle.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit 9751d75152)
2020-06-15 17:30:32 +02:00
Andreas Rheinhardt
3201350dc7 avformat/stldec: Fix memleak upon read header failure
The already parsed subtitles (contained in an FFDemuxSubtitlesQueue)
would leak if an error happened upon reading a subsequent subtitle.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit e13874b9ea)
2020-06-15 17:30:32 +02:00
Andreas Rheinhardt
157bbc779c avformat/srtdec: Fix memleak upon read header failure
The already parsed subtitles (contained in an FFDemuxSubtitlesQueue)
would leak if an error happened upon reading a subsequent subtitle.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit c70409957c)
2020-06-15 17:30:32 +02:00
Andreas Rheinhardt
bf29cf8eb6 avformat/sccdec: Fix memleak upon read header failure
The already parsed subtitles (contained in an FFDemuxSubtitlesQueue)
would leak if an error happened upon reading a subsequent subtitle.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit f3c63e67bb)
2020-06-15 17:30:28 +02:00
Andreas Rheinhardt
6e64260a19 avformat/samidec: Fix memleak upon read header failure
The already parsed subtitles (contained in an FFDemuxSubtitlesQueue)
would leak if an error happened upon reading a subsequent subtitle
or when creating extradata.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit f161f8e4ad)
2020-06-15 17:25:47 +02:00
Andreas Rheinhardt
7754a2ea12 avformat/pjsdec: Fix memleak upon read header failure
The already parsed subtitles (contained in an FFDemuxSubtitlesQueue)
would leak if an error happened upon reading a subsequent subtitle.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit 9df560e898)
2020-06-15 17:25:47 +02:00
Andreas Rheinhardt
d84b9ab4ab avformat/mpsubdec: Fix memleak upon read header failure
The already parsed subtitles (contained in an FFDemuxSubtitlesQueue)
would leak if an error happened upon creating an AVStream.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit a5ed8aeea4)
2020-06-15 17:25:47 +02:00
Andreas Rheinhardt
f172490742 avformat/mpl2dec: Fix memleak upon read header failure
The already parsed subtitles (contained in an FFDemuxSubtitlesQueue)
would leak if an error happened upon reading a subsequent subtitle.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit 331799747e)
2020-06-15 17:25:47 +02:00
Andreas Rheinhardt
330a757d41 avformat/microdvddec: Fix memleak upon read header failure
The already parsed subtitles (contained in an FFDemuxSubtitlesQueue)
would leak if an error happened upon reading a subsequent subtitle
or when allocating extradata.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit b12014a5b8)
2020-06-15 17:25:46 +02:00
Andreas Rheinhardt
ea27fe480e avformat/lrcdec: Fix memleak upon read header failure
The already parsed subtitles (contained in an FFDemuxSubtitlesQueue)
would leak if an error happened upon reading a subsequent subtitle.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit d38694cea9)
2020-06-15 17:25:46 +02:00
Andreas Rheinhardt
db2002aee7 avformat/jacosubdec: Fix memleak upon read header failure
The already parsed subtitles (contained in an FFDemuxSubtitlesQueue)
would leak if an error happened upon reading a subsequent subtitle.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit c13a752733)
2020-06-15 17:25:46 +02:00
Andreas Rheinhardt
788a7c027b avformat/assdec: Fix memleak upon read header failure
The already parsed subtitles (contained in an FFDemuxSubtitlesQueue)
would leak if an error happened upon reading a subsequent subtitle
or if creating the extradata failed.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit 5ab39c2d8c)
2020-06-15 17:25:46 +02:00
Andreas Rheinhardt
7c0a9ff9c0 avformat/aqtitledec: Fix memleak upon read header failure
The already parsed subtitles (contained in an FFDemuxSubtitlesQueue)
would leak if an error happened upon reading a subsequent subtitle.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit a86a5d06d8)
2020-06-15 17:25:46 +02:00
Andreas Rheinhardt
30d66abc80 avformat/mov: Fix memleaks upon read_header failure
By default, a demuxer's read_close function is not called automatically
if an error happens when reading the header; instead it is up to the
demuxer to clean up after itself in this case. The mov demuxer did this
by calling its read_close function when it encountered some errors when
reading the header. Yet for other errors (mostly adding side-data to
streams) this has been forgotten, so that all the internal structures
of the demuxer leak.

This commit fixes this by making sure mov_read_close is called when
necessary.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit ac378c535b)
2020-06-15 17:25:46 +02:00
Andreas Rheinhardt
5171e0ee18 avformat/omadec: Fix memleaks upon read_header failure
Fixes possible leaks of id3v2 metadata as well as an AVDES struct in
case the content is encrypted and an error happens lateron.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit 3d3ba43bc6)
2020-06-15 17:25:46 +02:00
Andreas Rheinhardt
245d0f1889 avformat/matroskadec: Fix memleaks in WebM DASH manifest demuxer
In certain error scenarios, the underlying Matroska demuxer was not
properly closed, causing leaks.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit 0841063ce6)
2020-06-15 17:25:46 +02:00
Andreas Rheinhardt
0260352d92 avformat/matroskadec: Use right number of tracks
When demuxing a Matroska/WebM file, streams are added for tracks and for
attachments, so that the array containing the former can be NULL even
when the corresponding AVFormatContext has streams. So check for there
to be tracks in the MatroskaDemuxContext instead of just streams in the
AVFormatContext before dereferencing the pointer to the tracks.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit 1ef30571a0)
2020-06-15 17:25:46 +02:00
Andreas Rheinhardt
a2ab8babef avformat/matroskadec: Fix handling gigantic durations
matroska_parse_block currently asserts that the duration is not equal to
AV_NOPTS_VALUE, but there is nothing that actually guarantees this. It
is easy to create (spec-compliant) files which run into this assert;
so replace it and instead cap the duration to INT64_MAX, as the duration
field of an AVPacket is an int64_t.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit 3714d452b8)
2020-06-15 17:25:46 +02:00
Andreas Rheinhardt
751f285152 avformat/matroskadec: Move AVBufferRef instead of copying, fix memleak
EBML binary elements are already made reference-counted when read;
so when populating the AVStream.attached_pic, one does not need to
allocate a new buffer for the data; instead the current code just
creates a new reference to the underlying AVBuffer. But this can be
improved even further: Just move the already existing reference.

This also fixes a memleak that happens upon error because
matroska_read_close has not been called in this scenario.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit cbe336c9e8)
2020-06-15 17:25:46 +02:00
Andreas Rheinhardt
2c738c7521 avformat/hlsenc: Always treat numbers as decimal
c801ab43c3 caused a regression: The stream
number is now parsed with strtoll without a fixed basis; as a
consequence, the "010" in a variant stream mapping like "a:010" is now
treated as an octal number (i.e. as eight, not ten). This was not
intended and may break some scripts, so this commit restores the old
behaviour.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit 19a876fd69)
2020-06-15 05:35:07 +02:00
Andreas Rheinhardt
82d70d8038 avcodec/hevc_mp4toannexb_bsf: Check NAL size against available input
The hevc_mp4toannexb bsf does not explicitly check whether a NAL unit
is so big that it extends beyond the end of the input packet; it does so
only implicitly by using the checked version of the bytestream2 API.
But this has downsides compared to real checks: It can lead to huge
allocations (up to 2GiB) even when the input packet is just a few bytes.
And furthermore it leads to uninitialized data being output.
So add a check to error out early if it happens.

Also check directly whether there is enough data for the length field.

Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit ea1b71e82f)
2020-06-15 04:18:16 +02:00
Michael Niedermayer
cc948a1c8c RELEASE_NOTES: Based on the version from 4.1
Name suggested by Kieran O Leary and Reto Kromer

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-06-14 22:02:34 +02:00
Michael Niedermayer
5c1e458b34 avformat/mxfdec: free duplicated utf16 strings
Fixes: memleak
Fixes: 23415/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-5124814510751744

Suggested-by: Marton Balint <cus@passwd.hu>
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 0aa2768cb2)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-06-14 22:02:34 +02:00
Michael Niedermayer
8bdc64d45f avformat/4xm: Check that a video stream was created before returning packets for it
Fixes: assertion failure
Fixes: 23434/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-5227750851084288.fuzz

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit c517c3f474)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-06-14 22:02:34 +02:00
Michael Niedermayer
a3e0c9f8f0 avcodec/ffwavesynth: Avoid undefined operation on ts overflow
Alternatively these conditions could be treated as errors
Fixes: 23147/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_FFWAVESYNTH_fuzzer-5639254549200896
Fixes: signed integer overflow: 9223372036854775807 + 1 cannot be represented in type 'int64_t' (aka 'long')

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 584d334afd)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-06-14 22:02:34 +02:00
Michael Niedermayer
95b9ac040e avcodec/mv30: check mode_size vs. input space
Fixes: Timeout (longer than my patience vs 1sec)
Fixes: 22984/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_MV30_fuzzer-5630021988515840

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 75e2ac4f07)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-06-14 22:02:34 +02:00
Michael Niedermayer
f823932349 avcodec/mpeg4videodec: Fix 2 integer overflows in get_amv()
Fixes: signed integer overflow: -144876608 * 16 cannot be represented in type 'int'
Fixes: 22782/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_MPEG4_fuzzer-6039584977977344

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit e361785ee0)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-06-14 22:02:34 +02:00
Michael Niedermayer
fa0a71ac41 avcodec/jpeg2000dec: Fix/check for multiple integer overflows
Fixes: shift exponent 35 is too large for 32-bit type 'int'
Fixes: 22857/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_JPEG2000_fuzzer-5202709358837760

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit c579ceffbe)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-06-14 22:02:34 +02:00
Michael Niedermayer
e149b24c63 avcodec/lossless_audiodsp: Fix undefined overflows in scalarproduct_and_madd_int16_c()
Fixes: signed integer overflow: 2142077091 + 6881070 cannot be represented in type 'int'
Fixes: 22737/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_WMALOSSLESS_fuzzer-5958388889681920

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit c0dfe134be)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-06-14 22:02:34 +02:00
Michael Niedermayer
2ce670fc48 avcodec/sonic: Fix several integer overflows
Fixes: signed integer overflow: 2129689466 + 2129689466 cannot be represented in type 'int'
Fixes: 20715/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SONIC_fuzzer-5155263109922816

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 75d520e337)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-06-14 22:02:34 +02:00
Michael Niedermayer
6011484167 avformat/oggdec: Disable mid stream codec changes
Fixes: 22082/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-5688619118624768
Fixes: crash from V-codecs/Theora/theora_testsuite_broken/multi2.ogg

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Suggested-by: Lynne on IRC
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 70277f1232)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-06-14 22:02:34 +02:00
Michael Niedermayer
c372189443 avcodec/mpeg4videodec: avoid invalid values and reinitialize in format changes for studio profile
Fixes: out of array access
Fixes: 23327/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_MPEG4_fuzzer-5134822992510976

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit e53235f06c)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-06-14 22:02:34 +02:00
Michael Niedermayer
335ddf2fe9 avcodec/pixlet: Fix log(0) check
Fixes: passing zero to clz(), which is not a valid argument
Fixes: 23337/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_PIXLET_fuzzer-5179131989065728

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit bd0f81526d)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-06-14 22:02:34 +02:00
Michael Niedermayer
6514919306 avformat/ape: Cleanup after ape_read_header() failure
Fixes: memleaks
Fixes: 23306/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-5635436931448832

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 9b5fc789fb)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-06-14 22:02:34 +02:00
Michael Niedermayer
0e51c7b64a avcodec/iff: Fix off by x error
Fixes: out of array access
Fixes: 23245/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_IFF_ILBM_fuzzer-5723121327013888.fuzz

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 51225dee0a)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-06-14 22:02:34 +02:00
Michael Niedermayer
28460ece95 avcodec/wmalosslessdec: Check block_align maximum
Fixes: Assertion failure
Fixes: 22737/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_WMALOSSLESS_fuzzer-5958388889681920

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 314d10f7a6)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-06-14 22:02:34 +02:00
Michael Niedermayer
63d14168a5 avcodec/loco: Fix signed integer overflow in loco_get_rice()
Fixes: signed integer overflow: 2147483647 + 1 cannot be represented in type 'int'
Fixes: 22975/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_LOCO_fuzzer-5658160970072064

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit aa88cdfd90)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-06-14 22:02:34 +02:00
Michael Niedermayer
838e17ffec avformat/thp: Check fps
Fixes: division by zero
Fixes: 23162/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-4856420817436672

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 0e15b01b4e)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-06-14 22:02:34 +02:00
Michael Niedermayer
d078f39a51 avformat/mpl2dec: Fix integer overflow with duration
Fixes: signed integer overflow: 9223372036854775807 - -1 cannot be represented in type 'long'
Fixes: 23167/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-6425051741290496

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 9a42a67c5c)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-06-14 22:02:34 +02:00
Michael Niedermayer
e468d9248c avcodec/cbs: Allocate more CodedBitstreamUnit at once in cbs_insert_unit()
Fixes: Timeout (85sec -> 0.5sec)
Fixes: 20791/clusterfuzz-testcase-minimized-ffmpeg_BSF_AV1_FRAME_SPLIT_fuzzer-5659537719951360
Fixes: 21214/clusterfuzz-testcase-minimized-ffmpeg_BSF_MPEG2_METADATA_fuzzer-5165560875974656
Fixes: 21247/clusterfuzz-testcase-minimized-ffmpeg_BSF_H264_METADATA_fuzzer-5715175257931776

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 49ba60fed0)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-06-14 22:02:34 +02:00
Michael Niedermayer
e625d40b93 avcodec/mpeg12dec: remove outdated comments
Found-by: Kieran
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 48de8f5816)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-06-14 22:02:34 +02:00
Michael Niedermayer
bb788dec83 avcodec/snowdec: Avoid integer overflow with huge qlog
Fixes: integer overflow
Fixes: 22285/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SNOW_fuzzer-5682428762128384

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 38fbf33c72)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-06-14 22:02:34 +02:00
Michael Niedermayer
611fc7244a avcodec/movtextdec: Fix shift overflows in mov_text_init()
Fixes: left shift of 243 by 24 places cannot be represented in type 'int'
Fixes: 22716/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_MOVTEXT_fuzzer-5704263425851392

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit d7a2311a2c)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-06-14 22:02:34 +02:00
Dale Curtis
8dee726b1a avformat/mov: Check if DTS is AV_NOPTS_VALUE in mov_find_next_sample().
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit bf446711bc)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-06-14 22:02:34 +02:00
James Almer
dba8e32e44 avcodec/cbs_av1: abort when written inferred values don't match
If this happens, it's a sign of parsing issues earlier in the process, or
misuse by the calling module.

Prevents writing invalid bitstreams.

Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit 318a1a383d)
2020-06-14 16:45:05 -03:00
James Almer
e6ab99f324 avcodec/cbs_h2645: abort when written inferred values don't match
If this happens, it's a sign of parsing issues earlier in the process, or
misuse by the calling module.

Prevents writing invalid bitstreams.

Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit ef13fafe22)
2020-06-14 16:44:57 -03:00
Marton Balint
cdf88b5a0c avcodec/libzvbi-teletextdec: fix txt_default_region limits
Max region ID is 87. Also the region affects not only the G0 charset but G2 and
the national subset as well.

Signed-off-by: Marton Balint <cus@passwd.hu>
(cherry picked from commit 16d29c1be8)
2020-06-14 21:10:41 +02:00
David Holroyd
3a390eadd2 lavf/prompeg: prompeg_write() must report data all was written
Previously, prompeg_write() would only report to caller that bytes we
written when a FEC packet was actually created.  Not all RTP packets are
expected to generate a FEC packet however, so this behavior was causing
avio to retry writing the RTP packet, eventually forcing the FEC state
machine to send a FEC packet erroneously (and so breaking out of the
retry loop).

This was resulting in incorrect FEC data being generated, and far too
many FEC packets to be sent (~100% FEC overhead).

fix #7863

Signed-off-by: David Holroyd <david.holroyd@m2amedia.tv>
(cherry picked from commit ffc1208266)
2020-06-14 21:09:05 +02:00
Steven Liu
e929799065 avformat/hls: check segment duration value of EXTINF
fix ticket: 8673
set the default EXTINF duration to 1ms if duration is smaller than 1ms

Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
(cherry picked from commit 9dfb19baeb)
2020-06-14 21:04:45 +02:00
Steven Liu
0c37321362 avformat/hls: check output string is usable of ff_make_absolute_url
fix ticket: 8688
should goto failed workflow if cannot get usable string by ff_make_absolute_url

Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
(cherry picked from commit ea1940c6e2)
2020-06-14 21:04:30 +02:00
Steven Liu
cfec756a6d avformat/url: check return value of strchr
fix ticket: 8687
workflow should return if there have no value of strchr

Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
(cherry picked from commit 029ff31af6)
2020-06-14 21:04:07 +02:00
Anton Khirnov
569a9d3d70 pthread_frame: change the way delay is set
It is a constant known at codec init, so set it in
ff_frame_thread_init(). Also, only set it for video, since the meaning
of this field is not well-defined for audio with frame threading.

Fixes availability of delay in callbacks invoked from the per-thread
contexts after 1f4cf92cfb.

(cherry picked from commit 6943ab688d)
2020-06-11 10:08:58 -03:00
James Almer
52dc21a68d avcodec/snow: ensure current_picture is writable before modifying its data
current_picture was not writable here because a reference existed in
at least avctx->coded_frame, and potentially elsewhere if the caller
created new ones from it.

Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit 1ee3c984b9)
2020-06-09 18:21:59 -03:00
Michael Niedermayer
c1ebaffba9 Update for version 4.3
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-06-08 22:51:03 +02:00
5360 changed files with 277174 additions and 581263 deletions

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

6
.gitignore vendored
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@@ -19,12 +19,8 @@
*.swp
*.ver
*.version
*.metal.air
*.metallib
*.metallib.c
*.ptx
*.ptx.c
*.ptx.gz
*_g
\#*
.\#*
@@ -35,8 +31,8 @@
/ffprobe
/config.asm
/config.h
/config_components.h
/coverage.info
/avversion.h
/lcov/
/src
/mapfile

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@@ -1,26 +1,21 @@
<james.darnley@gmail.com> <jdarnley@obe.tv>
<jeebjp@gmail.com> <jan.ekstrom@aminocom.com>
<sw@jkqxz.net> <mrt@jkqxz.net>
<u@pkh.me> <cboesch@gopro.com>
<quinkblack@foxmail.com> <wantlamy@gmail.com>
<quinkblack@foxmail.com> <zhilizhao@tencent.com>
<zhilizhao@tencent.com> <quinkblack@foxmail.com>
<zhilizhao@tencent.com> <wantlamy@gmail.com>
<modmaker@google.com> <modmaker-at-google.com@ffmpeg.org>
<stebbins@jetheaddev.com> <jstebbins@jetheaddev.com>
<barryjzhao@tencent.com> <mypopydev@gmail.com>
<barryjzhao@tencent.com> <jun.zhao@intel.com>
<josh@itanimul.li> <joshdk@obe.tv>
<michael@niedermayer.cc> <michaelni@gmx.at>
<linjie.justin.fu@gmail.com> <linjie.fu@intel.com>
<linjie.justin.fu@gmail.com> <fulinjie@zju.edu.cn>
<linjie.fu@intel.com> <fulinjie@zju.edu.cn>
<ceffmpeg@gmail.com> <cehoyos@ag.or.at>
<ceffmpeg@gmail.com> <cehoyos@rainbow.studorg.tuwien.ac.at>
<ffmpeg@gyani.pro> <gyandoshi@gmail.com>
<atomnuker@gmail.com> <rpehlivanov@obe.tv>
<lizhong1008@gmail.com> <zhong.li@intel.com>
<lizhong1008@gmail.com> <zhongli_dev@126.com>
<andreas.rheinhardt@outlook.com> <andreas.rheinhardt@gmail.com>
<andreas.rheinhardt@outlook.com> <andreas.rheinhardt@googlemail.com>
<zhong.li@intel.com> <zhongli_dev@126.com>
<andreas.rheinhardt@gmail.com> <andreas.rheinhardt@googlemail.com>
rcombs <rcombs@rcombs.me> <rodger.combs@gmail.com>
<thilo.borgmann@mail.de> <thilo.borgmann@googlemail.com>
<lq@chinaffmpeg.org> <liuqi05@kuaishou.com>
<ruiling.song83@gmail.com> <ruiling.song@intel.com>
Cosmin Stejerean <cosmin@cosmin.at> Cosmin Stejerean via ffmpeg-devel <ffmpeg-devel@ffmpeg.org>

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@@ -1,6 +1,6 @@
See the Git history of the project (https://git.ffmpeg.org/ffmpeg) to
See the Git history of the project (git://source.ffmpeg.org/ffmpeg) to
get the names of people who have contributed to FFmpeg.
To check the log, you can type the command "git log" in the FFmpeg
source directory, or browse the online repository at
https://git.ffmpeg.org/ffmpeg
http://source.ffmpeg.org.

656
Changelog
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@@ -1,621 +1,47 @@
Entries are sorted chronologically from oldest to youngest within each release,
releases are sorted from youngest to oldest.
version 7.0.2:
avcodec/snow: Fix off by 1 error in run_buffer
avcodec/utils: apply the same alignment to YUV410 as we do to YUV420 for snow
avformat/iamf_parse: Check for 0 samples
swscale: [loongarch] Fix checkasm-sw_yuv2rgb failure.
avcodec/aacps_tablegen_template: don't redefine CONFIG_HARDCODED_TABLES
avutil/hwcontext_vaapi: use the correct type for VASurfaceAttribExternalBuffers.buffers
avcodec/pcm-bluray/dvd: Use correct pointer types on BE
avcodec/pngenc: fix sBIT writing for indexed-color PNGs
avcodec/pngdec: use 8-bit sBIT cap for indexed PNGs per spec
avformat/mov: check that child boxes of trak are only present inside it
avformat/mov: check that sample and chunk count is 1 for HEIF
avcodec/videotoolboxenc: Fix bitrate doesn't work as expected
avdevice/dshow: Don't skip audio devices if no video device is present
avcodec/hdrenc: Allocate more space
avcodec/cfhdenc: Height of 16 is not supported
avcodec/cfhdenc: Allocate more space
avcodec/osq: fix integer overflow when applying factor
avcodec/osq: avoid using too large numbers for shifts and integers in update_residue_parameter()
avcodec/vaapi_encode: Check hwctx
avcodec/proresdec: Consider negative bits left
avcodec/alsdec: Clear shift_value
avcodec/hevc/hevcdec: Do not allow slices to depend on failed slices
avformat/mov: add an EOF check in IPRP
avfilter/vf_xfade: Check ff_inlink_consume_frame() for failure
avutil/slicethread: Check pthread_*_init() for failure
avutil/frame: Check log2_crop_align
avutil/buffer: Check ff_mutex_init() for failure
avformat/xmv: Check this_packet_size
avformat/webpenc: Check filesize in trailer
avformat/ty: rec_size seems to only need 32bit
avformat/tty: Check avio_size()
avformat/siff: Basic pkt_size check
avformat/sauce: Check avio_size() for failure
avformat/sapdec: Check ffurl_get_file_handle() for error
avformat/nsvdec: Check asize for PCM
avformat/mp3dec: Check header_filesize
avformat/mp3dec; Check for avio_size() failure
avformat/mov: Use 64bit for str_size
avformat/mm: Check length
avformat/hnm: Check *chunk_size
avformat/hlsenc: Check ret
avformat/bintext: Check avio_size() return
avformat/asfdec_o: Check size of index object
avfilter/vf_scale: Check ff_scale_adjust_dimensions() for failure
avfilter/scale_eval: Use 64bit, check values in ff_scale_adjust_dimensions()
avfilter/vf_lut3d: Check av_scanf()
avfilter/vf_elbg: Use unsigned for shifting into the top bit
avfilter/vf_premultiply: Use AV_PIX_MAX_PLANES
avfilter/vf_deshake_opencl: Ensure that the first iteration initializes the best variables
avformat/iamf_parse: Check for negative sample sizes
swscale/output: Fix integer overflows in yuv2rgba64_X_c_template
avformat/mxfdec: Reorder elements of expression in bisect loop
avutil/timecode: Use a 64bit framenum internally
avcodec/pnmdec: Use 64bit for input size check
avformat/mov: Check extradata in mov_read_iacb()
avcodec/mpeg12enc: Use av_rescale() in vbv_buffer_size computation
avcodec/utvideoenc: Use unsigned shift to build flags
avcodec/j2kenc: Merge dwt_norm into lambda
avcodec/vc2enc: Fix overflows with storing large values
avcodec/mpegvideo_enc: Do not duplicate pictures on shifting
avdevice/dshow_capture: Fix error handling in ff_dshow_##prefix##_Create()
avcodec/tiff: Check value on positive signed targets
avfilter/vf_convolution_opencl: Assert that the filter name is one of the filters
avfilter/vf_bm3d: Dont round MSE2SSE to an integer
avdevice/dshow: Remove NULL check on pin
avdevice/dshow: check ff_dshow_pin_ConnectionMediaType() for failure
avdevice/dshow: Check device_filter_unique_name before use
avdevice/dshow: Cleanup also on av_log case
avdevice/dshow_filter: Use wcscpy_s()
avcodec/flac_parser: Assert that we do not overrun the link_penalty array
avcodec/osq: avoid signed overflow in downsample path
avcodec/pixlet: Simplify pfx computation
avcodec/motion_est: Fix score squaring overflow
avcodec/mlpenc: Use 64 for ml, mr
avcodec/loco: Check loco_get_rice() for failure
avcodec/loco: check get_ur_golomb_jpegls() for failure
avcodec/leaddec: Check init_get_bits8() for failure
avcodec/imm4: check cbphi for error
avcodec/iff: Use signed count
avcodec/golomb: Assert that k is in the supported range for get_ur/sr_golomb()
avcodec/golomb: Document return for get_ur_golomb_jpegls() and get_sr_golomb_flac()
avcodec/dxv: Fix type in get_opcodes()
avcodec/cri: Check length
avcodec/xsubdec: Check parse_timecode()
avutil/imgutils: av_image_check_size2() ensure width and height fit in 32bit
avfilter/vf_tiltandshift: Free dst on error
doc/examples/mux: remove nop
avcodec/proresenc_kostya: use unsigned alpha for rotation
avformat/rtpenc_rfc4175: Use 64bit in computation if copy_offset
avformat/rtmpproto: Use AV_DICT_MATCH_CASE instead of litteral number
avformat/rtmppkt: Simplify and deobfuscate amf_tag_skip() slightly
avformat/rmdec: use 64bit for audio_framesize checks
avutil/wchar_filename: Correct sizeof
avutil/hwcontext_d3d11va: correct sizeof IDirect3DSurface9
avutil/hwcontext_d3d11va: Free AVD3D11FrameDescriptor on error
avutil/hwcontext_d3d11va: correct sizeof AVD3D11FrameDescriptor
avcodec/vvc/refs: Use unsigned mask
doc/examples/vaapi_encode: Try to check fwrite() for failure
avformat/usmdec: Initialize value
avformat/tls_schannel: Initialize ret
avformat/subfile: Assert that whence is a known case
avformat/subfile: Merge if into switch()
avformat/rtsp: Check that lower transport is handled in one of the if()
avformat/rtsp: initialize reply1
avformat/rtsp: use < 0 for error check
avformat/rtpenc_vc2hq: Check sizes
avfilter/af_aderivative: Free out on error
swscale/swscale: Use ptrdiff_t for linesize computations
avfilter/af_amerge: Cleanup on av_channel_layout_copy() failure
avfilter/af_afir: Assert format
avfilter/af_afftdn: Assert format
avfilter/af_pan: check nb_output_channels before use
cbs_av1: Reject thirty-two zero bits in uvlc code
avfilter/af_mcompand: compute half frequency in double
avfilter/af_channelsplit: Assert that av_channel_layout_channel_from_index() succeeds
avfilter/af_aresample: Cleanup on av_channel_layout_copy() failure
tools/coverity: Phase 1 study of anti-halicogenic for coverity av_rescale()
avfilter/vf_avgblur: Check plane instead of AVFrame
avfilter/drawutils: Fix depthb computation
avfilter/avf_showcwt: Check av_parse_video_rate() for failure
avformat/rdt: Check pkt_len
avformat/mpeg: Check len in mpegps_probe()
avformat/mxfenc: resurrects the error print
avdevice/dshow: Check ICaptureGraphBuilder2_SetFiltergraph() for failure
avcodec/mfenc: check IMFSample_ConvertToContiguousBuffer() for failure
avcodec/vc1_loopfilter: Factor duplicate code in vc1_b_h_intfi_loop_filter()
avcodec/vvc/ctu: Remove dead ret check
avcodec/vvc/dec: Remove constant eos_at_start
avformat/img2dec: assert no pipe on ts_from_file
avcodec/cbs_jpeg: Try to move the read entity to one side in a test
fftools/ffplay: Check vulkan_params
fftools/ffmpeg_enc: Initialize Decoder
fftools/ffmpeg_enc: Initialize fd
fftools/ffmpeg_enc: simplify opaque_ref check
avformat/mov: Check edit list for overflow
fftools/ffmpeg: Check read() for failure
avcodec/vvc/dec: Check ff_init_cabac_decoder() for failure
MAINTAINERS: Add Timo Rothenpieler to server admins
swscale/output: Avoid undefined overflow in yuv2rgb_write_full()
swscale/output: alpha can become negative after scaling, use multiply
avcodec/targaenc: Allocate space for the palette
avcodec/r210enc: Use av_rescale for bitrate
avcodec/jfdctint_template: Fewer integer anomalies
avcodec/snowenc: MV limits due to mv_penalty table size
tools/target_dec_fuzzer: Adjust threshold for MV30
tools/target_dec_fuzzer: Adjust threshold for jpeg2000
avformat/mxfdec: Check container_ul->desc before use
avcodec/libvpxenc: Cleanup on error
MAINTAINERS: Update the entries for the release maintainer for FFmpeg
doc/developer: Provide information about git send-email and gmail
avfilter/vf_rotate: Check ff_draw_init2() return value
avformat/mov: Use int64_t in intermediate for corrected_dts
avformat/mov: Use 64bit in intermediate for current_dts
avformat/matroskadec: Assert that num_levels is non negative
avformat/libzmq: Check av_strstart()
avformat/img2dec: Little JFIF / Exif cleanup
avformat/img2dec: Move DQT after unrelated if()
avformat/imfdec: Simplify get_next_track_with_minimum_timestamp()
avdevice/xcbgrab: Check sscanf() return
fftools/cmdutils: Add protective () to FLAGS
avformat/sdp: Check before appending ","
avcodec/libx264: Check init_get_bits8() return code
avcodec/ilbcdec: Remove dead code
avcodec/vp8: Check cond init
avcodec/vp8: Check mutex init
avcodec/proresenc_anatoliy: Assert that AV_PROFILE_UNKNOWN is replaced
avcodec/pcm-dvdenc: 64bit pkt-size
avcodec/notchlc: Check init_get_bits8() for failure
avcodec/tests/dct: Use 64bit in intermediate for error computation
avcodec/scpr3: Check add_dec() for failure
avcodec/rv34: assert that size is not 0 in rv34_gen_vlc_ext()
avcodec/wavpackenc: Use unsigned for potential 31bit shift
avcodec/vvc/mvs: Initialize mvf
avcodec/tests/jpeg2000dwt: Use 64bit in comparission
avcodec/tests/jpeg2000dwt: Use 64bit in err2 computation
avformat/fwse: Remove always false expression
avcodec/sga: Make it clear that the return is intentionally not checked
avformat/asfdec_f: Use 64bit for preroll computation
avformat/argo_asf: Use 64bit in offset intermediate
avformat/ape: Use 64bit for final frame size
avformat/ac4dec: Check remaining space in ac4_probe()
avdevice/pulse_audio_enc: Use av_rescale() to avoid integer overflow
avcodec/vlc: Cleanup on multi table alloc failure in ff_vlc_init_multi_from_lengths()
avcodec/tiff: Assert init_get_bits8() success in unpack_gray()
avcodec/tiff: Assert init_get_bits8() success in horizontal_fill()
tools/decode_simple: Check avcodec_send_packet() for errors on flushing
swscale/yuv2rgb: Use 64bit for brightness computation
swscale/x86/swscale: use a clearer name for INPUT_PLANER_RGB_A_FUNC_CASE
avutil/tests/opt: Check av_set_options_string() for failure
avutil/tests/dict: Check av_dict_set() before get for failure
avdevice/dshow: fix badly indented line
avformat/demux: resurrect dead stores
avcodec/tests/bitstream_template: Assert bits_init8() return
tools/enc_recon_frame_test: Assert that av_image_get_linesize() succeeds
avformat/iamf_writer: disallow Opus extradata with mapping family other than 0
avformat/iamf_parse: sanitize audio_roll_distance values
avformat/iamf: byteswap values in OpusHeader
avformat/iamf: rename Codec Config seek_preroll to audio_roll_distance
avformat/iamf_writer: fix coded audio_roll_distance values
avformat/iamf_writer: fix PCM endian-ness flag
avformat/movenc: fix channel count and samplerate fields for IAMF tracks
avformat/iamf_parse: keep substream count consistent
avformat/iamf_parse: add missing padding to AAC extradata
avformat/iamf_parse: 0 layers are not allowed
avformat/iamf_parse: consider nb_substreams when accessing substreams array
avformat/iamf_parse: Remove dead case
avcodec/png: more informative error message for invalid sBIT size
avcodec/pngdec: avoid erroring with sBIT on indexed-color images
avfilter/vf_tiltandshift: fix buffer offset for yuv422p input
avutil/timestamp: avoid possible FPE when 0 is passed to av_ts_make_time_string2()
avformat/mov: add more checks for infe atom size
avformat/mov: check for EOF inside the infe list parsing loop
avformat/mov: check extent_offset calculation for overflow
avformat/mov: check that iloc offset values fit on an int64_t
avcodec/pngenc: fix mDCv typo
avcodec/pngdec: fix mDCv typo
avcodec/nvenc: fix segfault in intra-only mode
avdevice/avfoundation: add external video devices
aarch64: Add OpenBSD runtime detection of dotprod and i8mm using sysctl
fftools/ffplay_renderer: use correct NULL value for Vulkan type
qsv: Initialize impl_value
avutil/hwcontext_qsv: fix GCC 14.1 warnings
avcodec/mediacodecenc: workaround the alignment requirement for H.265
avcodec/mediacodecenc: workaround the alignment requirement only for H.264
lavc/lpc: fix off-by-one in R-V V compute_autocorr
lavc/vp9: reset segmentation fields when segmentation isn't enabled
configure: enable ffnvcodec, nvenc, nvdec for FreeBSD
lavc/sbrdsp: fix potential overflow in noise table
version 7.0.1:
lavc/flacdsp: do not assume maximum R-V VL
avformat/flacdec: Reorder allocations to avoid leak on error
avcodec/adts_parser: Don't presume buffer to be padded
avformat/movenc: Check av_malloc()
avcodec/vp8: Return error on error
avformat/mov: store sample_sizes as unsigned ints
avformat/vvc: fix parsing sps_subpic_id
avformat/vvc: initialize some ptl flags
avcodec/mscc & mwsc: Check loop counts before use
avcodec/mpegvideo_enc: Fix potential overflow in RD
avcodec/mpeg4videodec: assert impossible wrap points
avcodec/mpeg12dec: Use 64bit in bit computation
avcodec/vqcdec: Check init_get_bits8() for failure
avcodec/vvc/dec: Check init_get_bits8() for failure
avcodec/vble: Check av_image_get_buffer_size() for failure
avcodec/vp3: Replace check by assert
avcodec/vp8: Forward return of ff_vpx_init_range_decoder()
avcodec/jpeg2000dec: remove ST=3 case
avcodec/qsvdec: Check av_image_get_buffer_size() for failure
avcodec/exr: Fix preview overflow
avcodec/decode: decode_simple_internal() only implements audio and video
avcodec/fmvc: remove dead assignment
avcodec/h2645_sei: Remove dead checks
avcodec/h264_slice: Remove dead sps check
avcodec/lpc: copy levenson coeffs only when they have been computed
avutil/tests/base64: Check with too short output array
libavutil/base64: Try not to write over the array end
avcodec/cbs_av1: Avoid shift overflow
fftools/ffplay: Check return of swr_alloc_set_opts2()
tools/opt_common: Check for malloc failure
doc/examples/demux_decode: Simplify loop
avformat/concatdec: Check file
avcodec/mpegvideo_enc: Fix 1 line and one column images
avcodec/amrwbdec: assert mode to be valid in decode_fixed_vector()
avcodec/wavarc: fix integer overflow in decode_5elp() block type 2
swscale/output: Fix integer overflow in yuv2rgba64_full_1_c_template()
swscale/output: Fix integer overflow in yuv2rgba64_1_c_template
avcodec/av1dec: Change bit_depth to int
avcodec/av1dec: bit_depth cannot be another values than 8,10,12
avcodec/avs3_parser: assert the return value of init_get_bits()
avcodec/avs2_parser: Assert init_get_bits8() success with const size 15
avfilter/avfiltergraph: return value of ff_request_frame() is unused
avformat/mxfdec: Check body_offset
avformat/kvag: Check sample_rate
avcodec/atrac9dec: Check init_get_bits8() for failure
avcodec/ac3_parser: Check init_get_bits8() for failure
avcodec/pngdec: Check last AVFrame before deref
avcodec/hevcdec: Check ref frame
doc/examples/qsv_transcode: Initialize pointer before free
doc/examples/qsv_transcode: Simplify str_to_dict() loop
doc/examples/vaapi_transcode: Simplify loop
doc/examples/qsv_transcode: Simplify loop
avcodec/cbs_h2645: Check NAL space
avfilter/vf_thumbnail_cuda: Set ret before checking it
avfilter/signature_lookup: Dont copy uninitialized stuff around
avfilter/signature_lookup: Fix 2 differences to the refernce SW
avcodec/x86/vp3dsp_init: Set correct function pointer, fix crash
avformat/mp3dec: change bogus error message if read_header encounters EOF
avformat/mp3dec: simplify inner frame size check in mp3_read_header
avformat/mp3dec: only call ffio_ensure_seekback once
avcodec/cbs_h266: read vps_ptl_max_tid before using it
avcodec/cbs_h266: fix sh_collocated_from_l0_flag and sh_collocated_ref_idx infer
avformat/vvc: fix parsing some early VPS bitstream values
avformat/vvc: fix writing general_constraint_info bytes
avutil/ppc/cpu: Also use the machdep.altivec sysctl on NetBSD
lavd/v4l2: Use proper field type for second parameter of ioctl() with BSD's
vulkan_av1: Fix force_integer_mv value
vaapi_av1: Fix force_integer_mv value
av1dec: Add force_integer_mv derived field for decoder use
avutil/iamf: fix offsets for mix_gain options
avformat/iamfdec: check nb_streams in header read
avformat/mov: free the infe allocated item data on failure
avformat/iamf_writer: reject duplicated stream ids in a stream group
avformat/mov: don't read key_size bytes twice in the keys atom
avformat/mov: take into account the first eight bytes in the keys atom
avformat/mov: fix the check for the heif item parsing loop
avutil/iamf: fix mix_gain_class name
av1dec: Fix RefFrameSignBias calculation
avcodec/codec_par: always clear extradata_size in avcodec_parameters_to_context()
avcodec/mediacodecenc: Fix return empty packet when bsf is used
avcodec/hevcdec: Fix precedence, bogus film grain warning
avcodec/hevcdec: fix segfault on invalid film grain metadata
lavc/vvc: Skip enhancement layer NAL units
avformat/mov: ignore old infe box versions
vulkan_av1: add workaround for NVIDIA drivers tested on broken CTS
lavc/vulkan_av1: Use av1dec reference order hint information
lavc/av1: Record reference ordering information for each frame
doc/encoders: add missing libxvid option
doc/encoders: remove non-existent flag
fate/ffmpeg: Avoid dependency on samples
avcodec/wavpack: Remove always-false check
avcodec/wavpack: Fix leak and segfault on reallocation error
avcodec/lossless_videoencdsp: Don't presume alignment in diff_bytes
avcodec/ppc/h264dsp: Fix left shifts of negative numbers
version 7.0:
- DXV DXT1 encoder
- LEAD MCMP decoder
- EVC decoding using external library libxevd
- EVC encoding using external library libxeve
- QOA decoder and demuxer
- aap filter
- demuxing, decoding, filtering, encoding, and muxing in the
ffmpeg CLI now all run in parallel
- enable gdigrab device to grab a window using the hwnd=HANDLER syntax
- IAMF raw demuxer and muxer
- D3D12VA hardware accelerated H264, HEVC, VP9, AV1, MPEG-2 and VC1 decoding
- tiltandshift filter
- qrencode filter and qrencodesrc source
- quirc filter
- lavu/eval: introduce randomi() function in expressions
- VVC decoder (experimental)
- fsync filter
- Raw Captions with Time (RCWT) closed caption muxer
- ffmpeg CLI -bsf option may now be used for input as well as output
- ffmpeg CLI options may now be used as -/opt <path>, which is equivalent
to -opt <contents of file <path>>
- showinfo bitstream filter
- a C11-compliant compiler is now required; note that this requirement
will be bumped to C17 in the near future, so consider updating your
build environment if it lacks C17 support
- Change the default bitrate control method from VBR to CQP for QSV encoders.
- removed deprecated ffmpeg CLI options -psnr and -map_channel
- DVD-Video demuxer, powered by libdvdnav and libdvdread
- ffprobe -show_stream_groups option
- ffprobe (with -export_side_data film_grain) now prints film grain metadata
- AEA muxer
- ffmpeg CLI loopback decoders
- Support PacketTypeMetadata of PacketType in enhanced flv format
- ffplay with hwaccel decoding support (depends on vulkan renderer via libplacebo)
- dnn filter libtorch backend
- Android content URIs protocol
- AOMedia Film Grain Synthesis 1 (AFGS1)
- RISC-V optimizations for AAC, FLAC, JPEG-2000, LPC, RV4.0, SVQ, VC1, VP8, and more
- Loongarch optimizations for HEVC decoding
- Important AArch64 optimizations for HEVC
- IAMF support inside MP4/ISOBMFF
- Support for HEIF/AVIF still images and tiled still images
- Dolby Vision profile 10 support in AV1
- Support for Ambient Viewing Environment metadata in MP4/ISOBMFF
- HDR10 metadata passthrough when encoding with libx264, libx265, and libsvtav1
version 6.1:
- libaribcaption decoder
- Playdate video decoder and demuxer
- Extend VAAPI support for libva-win32 on Windows
- afireqsrc audio source filter
- arls filter
- ffmpeg CLI new option: -readrate_initial_burst
- zoneplate video source filter
- command support in the setpts and asetpts filters
- Vulkan decode hwaccel, supporting H264, HEVC and AV1
- color_vulkan filter
- bwdif_vulkan filter
- nlmeans_vulkan filter
- RivaTuner video decoder
- xfade_vulkan filter
- vMix video decoder
- Essential Video Coding parser, muxer and demuxer
- Essential Video Coding frame merge bsf
- bwdif_cuda filter
- Microsoft RLE video encoder
- Raw AC-4 muxer and demuxer
- Raw VVC bitstream parser, muxer and demuxer
- Bitstream filter for editing metadata in VVC streams
- Bitstream filter for converting VVC from MP4 to Annex B
- scale_vt filter for videotoolbox
- transpose_vt filter for videotoolbox
- support for the P_SKIP hinting to speed up libx264 encoding
- Support HEVC,VP9,AV1 codec in enhanced flv format
- apsnr and asisdr audio filters
- OSQ demuxer and decoder
- Support HEVC,VP9,AV1 codec fourcclist in enhanced rtmp protocol
- CRI USM demuxer
- ffmpeg CLI '-top' option deprecated in favor of the setfield filter
- VAAPI AV1 encoder
- ffprobe XML output schema changed to account for multiple
variable-fields elements within the same parent element
- ffprobe -output_format option added as an alias of -of
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 4.3.1:
avcodec/tiff: Check input space in dng_decode_jpeg()
avcodec/mjpeg_parser: Adjust size rejection threshold
avcodec/cbs_jpeg: Fix uninitialized end index in cbs_jpeg_split_fragment()
avformat/sdp: Fix potential write beyond end of buffer
avformat/mm: Check for existence of audio stream
avformat/mov: Fix unaligned read of uint32_t and endian-dependance in mov_read_default
avcodec/apedec: Fix undefined integer overflow with 24bit
avcodec/loco: Fix integer overflow with large values from loco_get_rice()
avformat/smjpegdec: Check the existence of referred streams
avcodec/tiff: Check frame parameters before blit for DNG
avcodec/mjpegdec: Limit bayer to single plane outputting format
avcodec/pnmdec: Fix misaligned reads
avcodec/mv30: Fix integer overflows in idct2_1d()
avcodec/hcadec: Check total_band_count against imdct_in size
avcodec/scpr3: Fix out of array access with dectab
avcodec/tiff: Do not overrun the array ends in dng_blit()
avcodec/dstdec: Replace AC overread check by sample rate check
dnn_backend_native: Add overflow check for length calculation.
avcodec/h264_metadata_bsf: Fix invalid av_freep
avcodec/cbs_h265: set default VUI parameters when vui_parameters_present_flag is false
avcodec/av1_parser: initialize avctx->pix_fmt
avcodec/av1_parser: add missing parsing for RGB pixel format signaling
avcodec/av1_parser: set context values outside the OBU parsing loop
avutil/avsscanf: Add () to avoid integer overflow in scanexp()
avformat/utils: reorder duration computation to avoid overflow
avcodec/pngdec: Check for fctl after idat
avformat/hls: Pass a copy of the URL for probing
avutil/common: Fix integer overflow in av_ceil_log2_c()
avcodec/wmalosslessdec: fix overflow with pred in revert_cdlms
avformat/mvdec: Fix integer overflow with billions of channels
avformat/microdvddec: skip malformed lines without frame number.
dnn_backend_native: check operand index
dnn_backend_native.c: refine code for fail case
avformat/mov: fix memleaks
libavformat/mov: Fix memleaks when demuxing DV audio
avcodec/cbs_av1: Fix writing uvlc numbers >= INT_MAX
avformat/avc, mxfenc: Avoid allocation of H264 SPS structure, fix memleak
avcodec/bitstream: Don't check for undefined behaviour after it happened
avformat/aviobuf: Also return truncated buffer in avio_get_dyn_buf()
avformat/aviobuf: Don't check for overflow after it happened
version 4.3:
- v360 filter

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
@@ -34,8 +40,7 @@ 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, Timo Rothenpieler
project server emergencies Árpád Gereöffy, Reimar Doeffinger, Alexander Strasser, Nikolay Aleksandrov, Timo Rothenpieler
project server Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser, Nikolay Aleksandrov
presets Robert Swain
metadata subsystem Aurelien Jacobs
release management Michael Niedermayer
@@ -50,9 +55,9 @@ 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
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
@@ -110,6 +115,8 @@ 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:
@@ -131,29 +138,28 @@ 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
audiotoolbox* Rodger Combs
avs2* Huiwen Ren
bgmc.c, bgmc.h Thilo Borgmann
binkaudio.c Peter Ross
cavs* Stefan Gehrer
cdxl.c Paul B Mahol
celp_filters.* Vitor Sessak
cinepak.c Roberto Togni
cinepakenc.c Rl / Aetey G.T. AB
ccaption_dec.c Anshul Maheshwari, Aman Gupta
cljr Alex Beregszaszi
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
@@ -162,6 +168,7 @@ Codecs:
dv.c Roman Shaposhnik
dvbsubdec.c Anshul Maheshwari
eacmv*, eaidct*, eat* Peter Ross
evrc* Paul B Mahol
exif.c, exif.h Thilo Borgmann
ffv1* Michael Niedermayer
ffwavesynth.c Nicolas George
@@ -179,14 +186,12 @@ Codecs:
interplayvideo.c Mike Melanson
jni*, ffjni* Matthieu Bouron
jpeg2000* Nicolas Bertrand
jpegxl* Leo Izen
jvdec.c Peter Ross
lcl*.c Roberto Togni, Reimar Doeffinger
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
@@ -209,18 +214,18 @@ Codecs:
mqc* Nicolas Bertrand
msmpeg4.c, msmpeg4data.h Michael Niedermayer
msrle.c Mike Melanson
msrleenc.c Tomas Härdin
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
pgssubdec.c Reimar Doeffinger
ptx.c Ivo van Poorten
qcelp* Reynaldo H. Verdejo Pinochet
qdm2.c, qdm2data.h Roberto Togni
qsv* Mark Thompson, Zhong Li, 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
@@ -228,20 +233,25 @@ Codecs:
rpza.c Roberto Togni
rtjpeg.c, rtjpeg.h Reimar Doeffinger
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
srt* Aurelien Jacobs
sunrast.c Ivo van Poorten
svq3.c Michael Niedermayer
tak* Paul B Mahol
truemotion1* Mike Melanson
tta.c Alex Beregszaszi, Jaikrishnan Menon
ttaenc.c Paul B Mahol
txd.c Ivo van Poorten
v4l2_* Jorge Ramirez-Ortiz
vc2* Rostislav Pehlivanov
vcr1.c Michael Niedermayer
videotoolboxenc.c Rick Kern, Aman Gupta
vima.c Paul B Mahol
vorbisdec.c Denes Balatoni, David Conrad
vorbisenc.c Oded Shimon
vp3* Mike Melanson
@@ -250,21 +260,24 @@ Codecs:
vp8 David Conrad, Ronald Bultje
vp9 Ronald Bultje
vqavideo.c Mike Melanson
vvc Nuo Mi
wmaprodec.c Sascha Sommer
wmavoice.c Ronald S. Bultje
wmv2.c Michael Niedermayer
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, Zhao Zhili
vaapi* Haihao Xiang
vaapi_encode* Mark Thompson, Haihao Xiang
mediacodec* Matthieu Bouron, Aman Gupta
vaapi* Gwenole Beauchesne
vaapi_encode* Mark Thompson
vdpau* Philip Langdale, Carl Eugen Hoyos
videotoolbox* Rick Kern, Aman Gupta, Zhao Zhili
videotoolbox* Rick Kern, Aman Gupta
libavdevice
@@ -299,34 +312,63 @@ Generic parts:
motion_estimation.c Davinder Singh
Filters:
f_drawgraph.c Paul B Mahol
af_adelay.c Paul B Mahol
af_aecho.c Paul B Mahol
af_afade.c Paul B Mahol
af_amerge.c Nicolas George
af_aphaser.c Paul B Mahol
af_aresample.c Michael Niedermayer
af_astats.c Paul B Mahol
af_atempo.c Pavel Koshevoy
af_biquads.c Paul B Mahol
af_chorus.c Paul B Mahol
af_compand.c Paul B Mahol
af_firequalizer.c Muhammad Faiz
af_hdcd.c Burt P.
af_ladspa.c Paul B Mahol
af_loudnorm.c Kyle Swanson
af_pan.c Nicolas George
af_sidechaincompress.c Paul B Mahol
af_silenceremove.c Paul B Mahol
avf_aphasemeter.c Paul B Mahol
avf_avectorscope.c Paul B Mahol
avf_showcqt.c Muhammad Faiz
vf_blend.c Paul B Mahol
vf_bwdif Thomas Mundt (CC <thomas.mundt@hr.de>)
vf_chromakey.c Timo Rothenpieler
vf_colorchannelmixer.c Paul B Mahol
vf_colorconstancy.c Mina Sami (CC <minas.gorgy@gmail.com>)
vf_colorbalance.c Paul B Mahol
vf_colorkey.c Timo Rothenpieler
vf_colorlevels.c Paul B Mahol
vf_coreimage.m Thilo Borgmann
vf_deband.c Paul B Mahol
vf_dejudder.c Nicholas Robbins
vf_delogo.c Jean Delvare (CC <jdelvare@suse.com>)
vf_drawbox.c/drawgrid Andrey Utkin
vf_fsync.c Thilo Borgmann
vf_extractplanes.c Paul B Mahol
vf_histogram.c Paul B Mahol
vf_hqx.c Clément Bœsch
vf_idet.c Pascal Massimino
vf_il.c Paul B Mahol
vf_(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
vf_neighbor.c Paul B Mahol
vf_psnr.c Paul B Mahol
vf_random.c Paul B Mahol
vf_readvitc.c Tobias Rapp (CC t.rapp at noa-archive dot com)
vf_scale.c Michael Niedermayer
vf_separatefields.c Paul B Mahol
vf_ssim.c Paul B Mahol
vf_stereo3d.c Paul B Mahol
vf_telecine.c Paul B Mahol
vf_tonemap_opencl.c Ruiling Song
vf_yadif.c Michael Niedermayer
vf_zoompan.c Paul B Mahol
Sources:
vsrc_mandelbrot.c Michael Niedermayer
@@ -348,34 +390,32 @@ Muxers/Demuxers:
4xm.c Mike Melanson
aadec.c Vesselin Bontchev (vesselin.bontchev at yandex dot com)
adtsenc.c Robert Swain
afc.c Paul B Mahol
aiffdec.c Baptiste Coudurier, Matthieu Bouron
aiffenc.c Baptiste Coudurier, Matthieu Bouron
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
avi* Michael Niedermayer
avisynth.c Stephen Hutchinson
avr.c Paul B Mahol
bink.c Peter Ross
boadec.c Michael Niedermayer
brstm.c Paul B Mahol
caf* Peter Ross
cdxl.c Paul B Mahol
codec2.c Tomas Härdin
crc.c Michael Niedermayer
dashdec.c Steven Liu
dashenc.c Karthick Jeyapal
daud.c Reimar Doeffinger
dfpwmdec.c Jack Bruienne
dss.c Oleksij Rempel
dtsdec.c foo86
dtshddec.c Paul B Mahol
dv.c Roman Shaposhnik
dvdvideodec.c Marth64
electronicarts.c Peter Ross
evc* Samsung (Dawid Kozinski)
epafdec.c Paul B Mahol
ffm* Baptiste Coudurier
flic.c Mike Melanson
flvdec.c Michael Niedermayer
@@ -386,22 +426,22 @@ 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* 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 subtitles (matroskaenc.c) John Peebles
metadata* Aurelien Jacobs
mgsts.c Paul B Mahol
microdvd* Aurelien Jacobs
mm.c Peter Ross
mov.c Baptiste Coudurier
@@ -414,6 +454,7 @@ Muxers/Demuxers:
msnwc_tcp.c Ramiro Polla
mtv.c Reynaldo H. Verdejo Pinochet
mxf* Baptiste Coudurier, Tomas Härdin
nistspheredec.c Paul B Mahol
nsvdec.c Francois Revol
nut* Michael Niedermayer
nuv.c Reimar Doeffinger
@@ -421,12 +462,12 @@ Muxers/Demuxers:
oggenc.c Baptiste Coudurier
oggparse*.c David Conrad
oma.c Maxim Poliakovski
pp_bnk.c Zane van Iperen
paf.c Paul B Mahol
psxstr.c Mike Melanson
pva.c Ivo van Poorten
pvfdec.c Paul B Mahol
r3d.c Baptiste Coudurier
raw.c Michael Niedermayer
rcwtenc.c Marth64
rdt.c Ronald S. Bultje
rl2.c Sascha Sommer
rmdec.c, rmenc.c Ronald S. Bultje
@@ -445,9 +486,11 @@ Muxers/Demuxers:
sdp.c Martin Storsjo
segafilm.c Mike Melanson
segment.c Stefano Sabatini
smjpeg* Paul B Mahol
spdif* Anssi Hannula
srtdec.c Aurelien Jacobs
swf.c Baptiste Coudurier
takdec.c Paul B Mahol
tta.c Alex Beregszaszi
txd.c Ivo van Poorten
voc.c Aurelien Jacobs
@@ -457,13 +500,13 @@ Muxers/Demuxers:
webvtt* Matthew J Heaney
westwood.c Mike Melanson
wtv.c Peter Ross
wvenc.c Paul B Mahol
Protocols:
async.c Zhang Rui
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
@@ -490,14 +533,13 @@ 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
RISC-V Rémi Denis-Courmont
Windows MinGW Alex Beregszaszi, Ramiro Polla
Windows Cygwin Victor Paesa
Windows MSVC Hendrik Leppkes
Windows MSVC Matthew Oliver, Hendrik Leppkes
Windows ICL Matthew Oliver
ADI/Blackfin DSP Marc Hoffman
Sparc Roman Shaposhnik
OS/2 KO Myung-Hun
@@ -535,12 +577,10 @@ wm4
Releases
========
7.0 Michael Niedermayer
6.1 Michael Niedermayer
5.1 Michael Niedermayer
4.4 Michael Niedermayer
3.4 Michael Niedermayer
2.8 Michael Niedermayer
2.7 Michael Niedermayer
2.6 Michael Niedermayer
2.5 Michael Niedermayer
If you want to maintain an older release, please contact us
@@ -563,23 +603,18 @@ 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 (Traneptora) B6FD 3CFC 7ACF 83FC 9137 6945 5A71 C331 FD2F A19A
Leo Izen (Traneptora) 1D83 0A0B CE46 709E 203B 26FC 764E 48EA 4822 1833
Loren Merritt ABD9 08F4 C920 3F65 D8BE 35D7 1540 DAA7 060F 56DE
Lou Logan (llogan) 7D68 DC73 CBEF EABB 671A B6CF 621C 2E28 82F8 DC3A
Lynne FE50 139C 6805 72CA FD52 1F8D A2FE A5F0 3F03 4464
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
@@ -594,4 +629,3 @@ 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

View File

@@ -13,19 +13,17 @@ vpath %.v $(SRC_PATH)
vpath %.texi $(SRC_PATH)
vpath %.cu $(SRC_PATH)
vpath %.ptx $(SRC_PATH)
vpath %.metal $(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
ALLFFLIBS = avcodec avdevice avfilter avformat avutil postproc swscale swresample
# $(FFLIBS-yes) needs to be in linking order
FFLIBS-$(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
@@ -47,7 +45,7 @@ FF_DEP_LIBS := $(DEP_LIBS)
FF_STATIC_DEP_LIBS := $(STATIC_DEP_LIBS)
$(TOOLS): %$(EXESUF): %.o
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $(filter-out $(FF_DEP_LIBS), $^) $(EXTRALIBS-$(*F)) $(EXTRALIBS) $(ELIBS)
$(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) $(LIBFUZZER_PATH)
@@ -55,42 +53,24 @@ target_dec_%_fuzzer$(EXESUF): target_dec_%_fuzzer.o $(FF_DEP_LIBS)
tools/target_bsf_%_fuzzer$(EXESUF): tools/target_bsf_%_fuzzer.o $(FF_DEP_LIBS)
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
target_dem_%_fuzzer$(EXESUF): target_dem_%_fuzzer.o $(FF_DEP_LIBS)
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
tools/target_dem_fuzzer$(EXESUF): tools/target_dem_fuzzer.o $(FF_DEP_LIBS)
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
tools/target_io_dem_fuzzer$(EXESUF): tools/target_io_dem_fuzzer.o $(FF_DEP_LIBS)
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
tools/target_sws_fuzzer$(EXESUF): tools/target_sws_fuzzer.o $(FF_DEP_LIBS)
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
tools/enum_options$(EXESUF): ELIBS = $(FF_EXTRALIBS)
tools/enum_options$(EXESUF): $(FF_DEP_LIBS)
tools/enc_recon_frame_test$(EXESUF): $(FF_DEP_LIBS)
tools/enc_recon_frame_test$(EXESUF): ELIBS = $(FF_EXTRALIBS)
tools/scale_slice_test$(EXESUF): $(FF_DEP_LIBS)
tools/scale_slice_test$(EXESUF): ELIBS = $(FF_EXTRALIBS)
tools/sofa2wavs$(EXESUF): ELIBS = $(FF_EXTRALIBS)
tools/uncoded_frame$(EXESUF): $(FF_DEP_LIBS)
tools/uncoded_frame$(EXESUF): ELIBS = $(FF_EXTRALIBS)
tools/target_dec_%_fuzzer$(EXESUF): $(FF_DEP_LIBS)
tools/target_dem_%_fuzzer$(EXESUF): $(FF_DEP_LIBS)
CONFIGURABLE_COMPONENTS = \
$(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c)) \
$(SRC_PATH)/libavcodec/bitstream_filters.c \
$(SRC_PATH)/libavcodec/hwaccels.h \
$(SRC_PATH)/libavcodec/parsers.c \
$(SRC_PATH)/libavformat/protocols.c \
config_components.h: ffbuild/.config
config.h: ffbuild/.config
ffbuild/.config: $(CONFIGURABLE_COMPONENTS)
@-tput bold 2>/dev/null
@-printf '\nWARNING: $(?) newer than config_components.h, rerun configure\n\n'
@-printf '\nWARNING: $(?) newer than config.h, rerun configure\n\n'
@-tput sgr0 2>/dev/null
SUBDIR_VARS := CLEANFILES FFLIBS HOSTPROGS TESTPROGS TOOLS \
@@ -98,8 +78,7 @@ SUBDIR_VARS := CLEANFILES FFLIBS HOSTPROGS TESTPROGS TOOLS \
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 LSX-OBJS LASX-OBJS RV-OBJS RVV-OBJS \
OBJS SLIBOBJS SHLIBOBJS STLIBOBJS HOSTOBJS TESTOBJS
MMI-OBJS OBJS SLIBOBJS HOSTOBJS TESTOBJS
define RESET
$(1) :=
@@ -121,13 +100,12 @@ include $(SRC_PATH)/fftools/Makefile
include $(SRC_PATH)/doc/Makefile
include $(SRC_PATH)/doc/examples/Makefile
$(ALLFFLIBS:%=lib%/version.o): libavutil/ffversion.h
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
$(RM) $@
$(CP) $< $@
$(STRIP) $@
endif
@@ -136,18 +114,13 @@ endif
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
VERSION_SH = $(SRC_PATH)/ffbuild/version.sh
ifeq ($(VERSION_TRACKING),yes)
GIT_LOG = $(SRC_PATH)/.git/logs/HEAD
endif
.version: $(wildcard $(GIT_LOG)) $(VERSION_SH) ffbuild/config.mak
.version: M=@
ifneq ($(VERSION_TRACKING),yes)
libavutil/ffversion.h .version: REVISION=unknown
endif
libavutil/ffversion.h .version:
$(M)revision=$(REVISION) $(VERSION_SH) $(SRC_PATH) libavutil/ffversion.h $(EXTRA_VERSION)
$(M)$(VERSION_SH) $(SRC_PATH) libavutil/ffversion.h $(EXTRA_VERSION)
$(Q)touch .version
# force version.sh to run whenever version might have changed
@@ -173,7 +146,7 @@ clean::
$(RM) -rf coverage.info coverage.info.in lcov
distclean:: clean
$(RM) .version config.asm config.h config_components.h mapfile \
$(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 \

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.

View File

@@ -1 +1 @@
7.0.2
4.3.1

View File

@@ -1,15 +1,15 @@
─────────────────────────────────────────┐
│ RELEASE NOTES for FFmpeg 7.0 "Dijkstra" │
─────────────────────────────────────────┘
┌────────────────────────────────────┐
│ RELEASE NOTES for FFmpeg 4.3 "4:3" │
└────────────────────────────────────┘
The FFmpeg Project proudly presents FFmpeg 7.0 "Dijkstra", about 6
months after the release of FFmpeg 6.1.
The FFmpeg Project proudly presents FFmpeg 4.3 "4:3", about 10
months after the release of FFmpeg 4.2.
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.libera.chat) or ask
feel free to join us on the #ffmpeg IRC channel (on irc.freenode.net) or ask
on the mailing-lists.

View File

@@ -19,6 +19,7 @@
#ifndef COMPAT_ATOMICS_WIN32_STDATOMIC_H
#define COMPAT_ATOMICS_WIN32_STDATOMIC_H
#define WIN32_LEAN_AND_MEAN
#include <stddef.h>
#include <stdint.h>
#include <windows.h>
@@ -95,7 +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)

View File

@@ -1,7 +1,7 @@
/*
* Minimum CUDA compatibility definitions header
*
* Copyright (c) 2019 rcombs
* Copyright (c) 2019 Rodger Combs
*
* This file is part of FFmpeg.
*
@@ -49,16 +49,6 @@ typedef struct __device_builtin__ __align__(4) ushort2
unsigned short x, y;
} ushort2;
typedef struct __device_builtin__ __align__(8) float2
{
float x, y;
} float2;
typedef struct __device_builtin__ __align__(8) int2
{
int x, y;
} int2;
typedef struct __device_builtin__ uint3
{
unsigned int x, y, z;
@@ -66,6 +56,11 @@ typedef struct __device_builtin__ uint3
typedef struct uint3 dim3;
typedef struct __device_builtin__ __align__(8) int2
{
int x, y;
} int2;
typedef struct __device_builtin__ __align__(4) uchar4
{
unsigned char x, y, z, w;
@@ -73,7 +68,7 @@ typedef struct __device_builtin__ __align__(4) uchar4
typedef struct __device_builtin__ __align__(8) ushort4
{
unsigned short x, y, z, w;
unsigned char x, y, z, w;
} ushort4;
typedef struct __device_builtin__ __align__(16) int4
@@ -81,11 +76,6 @@ typedef struct __device_builtin__ __align__(16) int4
int x, y, z, w;
} int4;
typedef struct __device_builtin__ __align__(16) float4
{
float x, y, z, w;
} float4;
// Accessors for special registers
#define GETCOMP(reg, comp) \
asm("mov.u32 %0, %%" #reg "." #comp ";" : "=r"(tmp)); \
@@ -110,31 +100,24 @@ GET(getThreadIdx, tid)
#define threadIdx (getThreadIdx())
// Basic initializers (simple macros rather than inline functions)
#define make_int2(a, b) ((int2){.x = a, .y = b})
#define make_uchar2(a, b) ((uchar2){.x = a, .y = b})
#define make_ushort2(a, b) ((ushort2){.x = a, .y = b})
#define make_float2(a, b) ((float2){.x = a, .y = b})
#define make_int4(a, b, c, d) ((int4){.x = a, .y = b, .z = c, .w = d})
#define make_uchar4(a, b, c, d) ((uchar4){.x = a, .y = b, .z = c, .w = d})
#define make_ushort4(a, b, c, d) ((ushort4){.x = a, .y = b, .z = c, .w = d})
#define make_float4(a, b, c, d) ((float4){.x = a, .y = b, .z = c, .w = d})
// Conversions from the tex instruction's 4-register output to various types
#define TEX2D(type, ret) static inline __device__ void conv(type* out, unsigned a, unsigned b, unsigned c, unsigned d) {*out = (ret);}
TEX2D(unsigned char, a & 0xFF)
TEX2D(unsigned short, a & 0xFFFF)
TEX2D(float, a)
TEX2D(uchar2, make_uchar2(a & 0xFF, b & 0xFF))
TEX2D(ushort2, make_ushort2(a & 0xFFFF, b & 0xFFFF))
TEX2D(float2, make_float2(a, b))
TEX2D(uchar4, make_uchar4(a & 0xFF, b & 0xFF, c & 0xFF, d & 0xFF))
TEX2D(ushort4, make_ushort4(a & 0xFFFF, b & 0xFFFF, c & 0xFFFF, d & 0xFFFF))
TEX2D(float4, make_float4(a, b, c, d))
// Template calling tex instruction and converting the output to the selected type
template<typename T>
inline __device__ T tex2D(cudaTextureObject_t texObject, float x, float y)
template <class T>
static inline __device__ T tex2D(cudaTextureObject_t texObject, float x, float y)
{
T ret;
unsigned ret1, ret2, ret3, ret4;
@@ -145,48 +128,4 @@ inline __device__ T tex2D(cudaTextureObject_t texObject, float x, float y)
return ret;
}
template<>
inline __device__ float4 tex2D<float4>(cudaTextureObject_t texObject, float x, float y)
{
float4 ret;
asm("tex.2d.v4.f32.f32 {%0, %1, %2, %3}, [%4, {%5, %6}];" :
"=r"(ret.x), "=r"(ret.y), "=r"(ret.z), "=r"(ret.w) :
"l"(texObject), "f"(x), "f"(y));
return ret;
}
template<>
inline __device__ float tex2D<float>(cudaTextureObject_t texObject, float x, float y)
{
return tex2D<float4>(texObject, x, y).x;
}
template<>
inline __device__ float2 tex2D<float2>(cudaTextureObject_t texObject, float x, float y)
{
float4 ret = tex2D<float4>(texObject, x, y);
return make_float2(ret.x, ret.y);
}
// Math helper functions
static inline __device__ float floorf(float a) { return __builtin_floorf(a); }
static inline __device__ float floor(float a) { return __builtin_floorf(a); }
static inline __device__ double floor(double a) { return __builtin_floor(a); }
static inline __device__ float ceilf(float a) { return __builtin_ceilf(a); }
static inline __device__ float ceil(float a) { return __builtin_ceilf(a); }
static inline __device__ double ceil(double a) { return __builtin_ceil(a); }
static inline __device__ float truncf(float a) { return __builtin_truncf(a); }
static inline __device__ float trunc(float a) { return __builtin_truncf(a); }
static inline __device__ double trunc(double a) { return __builtin_trunc(a); }
static inline __device__ float fabsf(float a) { return __builtin_fabsf(a); }
static inline __device__ float fabs(float a) { return __builtin_fabsf(a); }
static inline __device__ double fabs(double a) { return __builtin_fabs(a); }
static inline __device__ float sqrtf(float a) { return __builtin_sqrtf(a); }
static inline __device__ float __saturatef(float a) { return __nvvm_saturate_f(a); }
static inline __device__ float __sinf(float a) { return __nvvm_sin_approx_f(a); }
static inline __device__ float __cosf(float a) { return __nvvm_cos_approx_f(a); }
static inline __device__ float __expf(float a) { return __nvvm_ex2_approx_f(a * (float)__builtin_log2(__builtin_exp(1))); }
static inline __device__ float __powf(float a, float b) { return __nvvm_ex2_approx_f(__nvvm_lg2_approx_f(a) * b); }
#endif /* COMPAT_CUDA_CUDA_RUNTIME_H */

34
compat/cuda/ptx2c.sh Executable file
View File

@@ -0,0 +1,34 @@
#!/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" > "$OUT"
echo >> "$OUT"
sed -e "$(printf 's/\r//g')" -e 's/["\\]/\\&/g' -e "$(printf 's/^/\t"/')" -e 's/$/\\n"/' < "$IN" >> "$OUT"
echo ";" >> "$OUT"
exit 0

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);

View File

@@ -20,40 +20,11 @@
#define COMPAT_W32DLFCN_H
#ifdef _WIN32
#include <stdint.h>
#include <windows.h>
#include "config.h"
#include "libavutil/macros.h"
#if (_WIN32_WINNT < 0x0602) || HAVE_WINRT
#include "libavutil/wchar_filename.h"
static inline wchar_t *get_module_filename(HMODULE module)
{
wchar_t *path = NULL, *new_path;
DWORD path_size = 0, path_len;
do {
path_size = path_size ? FFMIN(2 * path_size, INT16_MAX + 1) : MAX_PATH;
new_path = av_realloc_array(path, path_size, sizeof *path);
if (!new_path) {
av_free(path);
return NULL;
}
path = new_path;
// Returns path_size in case of insufficient buffer.
// Whether the error is set or not and whether the output
// is null-terminated or not depends on the version of Windows.
path_len = GetModuleFileNameW(module, path, path_size);
} while (path_len && path_size <= INT16_MAX && path_size <= path_len);
if (!path_len) {
av_free(path);
return NULL;
}
return path;
}
#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
@@ -63,53 +34,29 @@ static inline wchar_t *get_module_filename(HMODULE module)
*/
static inline HMODULE win32_dlopen(const char *name)
{
wchar_t *name_w;
HMODULE module = NULL;
if (utf8towchar(name, &name_w))
name_w = NULL;
#if _WIN32_WINNT < 0x0602
// On Win7 and earlier we check if KB2533623 is available
// Need to check if KB2533623 is available
if (!GetProcAddress(GetModuleHandleW(L"kernel32.dll"), "SetDefaultDllDirectories")) {
wchar_t *path = NULL, *new_path;
DWORD pathlen, pathsize, namelen;
if (!name_w)
HMODULE module = NULL;
wchar_t *path = NULL, *name_w = NULL;
DWORD pathlen;
if (utf8towchar(name, &name_w))
goto exit;
namelen = wcslen(name_w);
path = (wchar_t *)av_mallocz_array(MAX_PATH, sizeof(wchar_t));
// Try local directory first
path = get_module_filename(NULL);
if (!path)
pathlen = GetModuleFileNameW(NULL, path, MAX_PATH);
pathlen = wcsrchr(path, '\\') - path;
if (pathlen == 0 || pathlen + wcslen(name_w) + 2 > MAX_PATH)
goto exit;
new_path = wcsrchr(path, '\\');
if (!new_path)
goto exit;
pathlen = new_path - path;
pathsize = pathlen + namelen + 2;
new_path = av_realloc_array(path, pathsize, sizeof *path);
if (!new_path)
goto exit;
path = new_path;
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, pathsize);
if (!pathlen)
pathlen = GetSystemDirectoryW(path, MAX_PATH);
if (pathlen == 0 || pathlen + wcslen(name_w) + 2 > MAX_PATH)
goto exit;
// Buffer is not enough in two cases:
// 1. system directory + \ + module name
// 2. system directory even without the module name.
if (pathlen + namelen + 2 > pathsize) {
pathsize = pathlen + namelen + 2;
new_path = av_realloc_array(path, pathsize, sizeof *path);
if (!new_path)
goto exit;
path = new_path;
// Query again to handle the case #2.
pathlen = GetSystemDirectoryW(path, pathsize);
if (!pathlen)
goto exit;
}
path[pathlen] = L'\\';
path[pathlen] = '\\';
wcscpy(path + pathlen + 1, name_w);
module = LoadLibraryExW(path, NULL, LOAD_WITH_ALTERED_SEARCH_PATH);
}
@@ -126,19 +73,16 @@ exit:
# define LOAD_LIBRARY_SEARCH_SYSTEM32 0x00000800
#endif
#if HAVE_WINRT
if (!name_w)
wchar_t *name_w = NULL;
int ret;
if (utf8towchar(name, &name_w))
return NULL;
module = LoadPackagedLibrary(name_w, 0);
#else
#define LOAD_FLAGS (LOAD_LIBRARY_SEARCH_APPLICATION_DIR | LOAD_LIBRARY_SEARCH_SYSTEM32)
/* filename may be be in CP_ACP */
if (!name_w)
return LoadLibraryExA(name, NULL, LOAD_FLAGS);
module = LoadLibraryExW(name_w, NULL, LOAD_FLAGS);
#undef LOAD_FLAGS
#endif
ret = LoadPackagedLibrary(name_w, 0);
av_free(name_w);
return module;
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

View File

@@ -35,6 +35,7 @@
* As most functions here are used without checking return values,
* only implement return values as necessary. */
#define WIN32_LEAN_AND_MEAN
#include <windows.h>
#include <process.h>
#include <time.h>
@@ -65,14 +66,7 @@ typedef CONDITION_VARIABLE pthread_cond_t;
#define PTHREAD_CANCEL_ENABLE 1
#define PTHREAD_CANCEL_DISABLE 0
#if HAVE_WINRT
#define THREADFUNC_RETTYPE DWORD
#else
#define THREADFUNC_RETTYPE unsigned
#endif
static av_unused THREADFUNC_RETTYPE
__stdcall attribute_align_arg win32thread_worker(void *arg)
static av_unused unsigned __stdcall attribute_align_arg win32thread_worker(void *arg)
{
pthread_t *h = (pthread_t*)arg;
h->ret = h->func(h->arg);

View File

@@ -1,32 +0,0 @@
#!/bin/sh
if [ "$1" = "--version" ]; then
rc.exe -?
exit $?
fi
if [ $# -lt 2 ]; then
echo "Usage: mswindres [-I/include/path ...] [-DSOME_DEFINE ...] [-o output.o] input.rc [output.o]" >&2
exit 0
fi
EXTRA_OPTS="-nologo"
while [ $# -gt 2 ]; do
case $1 in
-D*) EXTRA_OPTS="$EXTRA_OPTS -d$(echo $1 | sed -e "s/^..//" -e "s/ /\\\\ /g")" ;;
-I*) EXTRA_OPTS="$EXTRA_OPTS -i$(echo $1 | sed -e "s/^..//" -e "s/ /\\\\ /g")" ;;
-o) OPT_OUT="$2"; shift ;;
esac
shift
done
IN="$1"
if [ -z "$OPT_OUT" ]; then
OUT="$2"
else
OUT="$OPT_OUT"
fi
eval set -- $EXTRA_OPTS
rc.exe "$@" -fo "$OUT" "$IN"

1844
configure vendored

File diff suppressed because it is too large Load Diff

View File

@@ -1,849 +1,20 @@
The last version increases of all libraries were on 2024-03-07
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 7.0 was cut here -------- 8< ---------
2024-03-25 - 5df901ffa56 - lavu 59.7.100 - timestamp.h
Add av_ts_make_time_string2() for better timestamp precision, the new
function accepts AVRational as time base instead of *AVRational, and is not
an inline function like its predecessor.
2024-03-23 - a9023377b22 - lavu 59.6.100 - film_grain_params.h
Add av_film_grain_params_select().
2024-03-23 - 35d2960dcd0 - lavu 59.5.100 - film_grain_params.h
Add AVFilmGrainParams.color_range, color_primaries, color_trc, color_space,
width, height, subsampling_x, subsampling_y, bit_depth_luma and
bit_depth_chroma. Deprecate the corresponding fields from
AVFilmGrainH274Params.
2024-03-23 - f17e18d2922 - lavc 61.3.100 - jni.h
Add av_jni_set_android_app_ctx() and av_jni_get_android_app_ctx().
2024-03-22 - 26398da8f30 - lavu 59.4.100 - frame.h
Constified the first-level pointee of av_frame_side_data_get()
and renamed it to av_frame_side_data_get_c(). From now on,
av_frame_side_data_get() is a wrapper around av_frame_side_data_get_c()
that accepts AVFrameSideData * const *sd.
2024-03-20 - 0d36844ddf9 - lavc 61.2.100 - avcodec.h
Add AVCodecContext.[nb_]decoded_side_data.
2024-03-20 - d9ade14c5c5 - lavu 59.3.100 - frame.h
Add av_frame_side_data_free(), av_frame_side_data_new(),
av_frame_side_data_clone(), av_frame_side_data_get() as well
as AV_FRAME_SIDE_DATA_FLAG_UNIQUE.
2024-03-16 - ed6207274e6 - lavu 59.2.100 - channel_layout.h
Add AV_CHANNEL_LAYOUT_RETYPE_FLAG_CANONICAL.
2024-03-08 - 68a8eca7523 - lavc 61.1.100 - avcodec.h
Add AVCodecContext.[nb_]side_data_prefer_packet.
2024-03-08 - efe44787781 - lavu 59.1.100 - opt.h
Add AV_OPT_TYPE_FLAG_ARRAY and AVOptionArrayDef.
2024-03-08 - c9f5cea9cca - lavc 61.0.100 - vdpau.h
Deprecate av_vdpau_alloc_context(), av_alloc_vdpaucontext(),
av_vdpau_hwaccel_get_render2() and av_vdpau_hwaccel_set_render2().
The former are superseded by av_vdpau_bind_context(), the latter
are unneeded as the relevant field is public and can be accessed directly.
2024-03-06 - 49707b05900 - lavf 60.25.100 - avformat.h
Deprecate av_fmt_ctx_get_duration_estimation_method().
The relevant field is public and needs no getter to access.
2024-03-05 - ab15c04dee5 - lavf 60.24.100 - avformat.h
Add avformat_stream_group_name().
2024-02-28 - b295aafb082 - swr 4.14.100 - swresample.h
swr_convert() now accepts arrays of const pointers (to input and output).
2024-02-28 - 58e3ef7f546 - lavu 58.40.100 - timestamp.h
av_ts_make_time_string() now accepts a pointer to const AVRational.
2024-02-28 - dfb9d8a5a2f - lavf 60.23.100 - avio.h
avio_print_string_array() now accepts an array of const pointers.
2024-02-26 - 41e349c24a7 - lavf 60.22.101 - avformat.h
AV_DISPOSITION_DEPENDENT may now also be used for video streams
intended to be merged with other video streams for presentation.
2024-02-26 - 25a10677d12 - lavf 60.22.100 - avformat.h
Add AVStreamGroupTileGrid
Add AV_STREAM_GROUP_PARAMS_TILE_GRID
Add AVStreamGroup.params.tile_grid
2024-02-21 - 1d66a122df9 - lavc 60.40.100 - avcodec.h
Deprecate AV_INPUT_BUFFER_MIN_SIZE without replacement.
2024-02-16 - eea9bd88a5f - lavu 58.39.100 - pixfmt.h
Add AV_VIDEO_MAX_PLANES
2024-02-13 - ec2036454bc - lavf 60.21.100 - avformat.h
Add AVStreamGroup.disposition.
2024-02-12 - 66386bf2a2a - lavu 58.38.100 - channel_layout.h
Add av_channel_layout_retype().
2024-02-12 - 4569b861322 - lavu 58.37.100 - channel_layout.h
Add av_channel_layout_custom_init().
2024-02-04 - 45697e6a512 - lavc 60.39.100 - packet.h
Add AV_PKT_DATA_AMBIENT_VIEWING_ENVIRONMENT.
2023-11-xx - xxxxxxxxxx - lavfi 9.16.100 - buffersink.h buffersrc.h
Add av_buffersink_get_colorspace and av_buffersink_get_color_range.
Add AVBufferSrcParameters.color_space and AVBufferSrcParameters.color_range.
2023-11-xx - xxxxxxxxxx - lavfi 9.15.100 - avfilter.h
Add AVFilterLink.colorspace, AVFilterLink.color_range
2023-12-21 - 142f727b9ca - lavu 58.36.100 - pixfmt.h hwcontext.h hwcontext_d3d12va.h
Add AV_HWDEVICE_TYPE_D3D12VA and AV_PIX_FMT_D3D12.
Add AVD3D12VADeviceContext, AVD3D12VASyncContext, AVD3D12VAFrame and
AVD3D12VAFramesContext.
2023-12-18 - 74279227dd2 - lavc 60.36.100 - packet.h
Add AV_PKT_DATA_IAMF_MIX_GAIN_PARAM, AV_PKT_DATA_IAMF_DEMIXING_INFO_PARAM
and AV_PKT_DATA_IAMF_RECON_GAIN_INFO_PARAM.
2023-12-18 - 556b596d1d9 - lavc 60.19.100 - avformat.h
Add AVStreamGroup struct.
Add AVFormatContext.stream_groups and AVFormatContext.nb_stream_groups
Add avformat_stream_group_create(), avformat_stream_group_add_stream(),
and av_stream_group_get_class().
Add enum AVStreamGroupParamsType with values AV_STREAM_GROUP_PARAMS_NONE,
AV_STREAM_GROUP_PARAMS_IAMF_AUDIO_ELEMENT and
AV_STREAM_GROUP_PARAMS_IAMF_MIX_PRESENTATION.
2023-12-18 - d2af93bbefc - lavu 58.35.100 - iamf.h
Add a new API to support Immersive Audio Model and Formats.
2023-12-13 - 5475f665f60 - lavu 58.33.100 - imgutils.h
Add av_image_fill_color().
2023-11-08 - b82957a66a7 - lavu 58.32.100 - channel_layout.h
Add AV_CH_LAYOUT_7POINT2POINT3 and AV_CHANNEL_LAYOUT_7POINT2POINT3.
Add AV_CH_LAYOUT_9POINT1POINT4_BACK and AV_CHANNEL_LAYOUT_9POINT1POINT4_BACK.
2023-10-31 - 57c16323f26 - lavu 58.31.100 - pixdesc.h
Add AV_PIX_FMT_FLAG_XYZ.
-------- 8< --------- FFmpeg 6.1 was cut here -------- 8< ---------
2023-10-27 - 52a97642604 - lavu 58.28.100 - channel_layout.h
Add AV_CH_LAYOUT_3POINT1POINT2 and AV_CHANNEL_LAYOUT_3POINT1POINT2.
Add AV_CH_LAYOUT_5POINT1POINT2_BACK and AV_CHANNEL_LAYOUT_5POINT1POINT2_BACK.
Add AV_CH_LAYOUT_5POINT1POINT4_BACK and AV_CHANNEL_LAYOUT_5POINT1POINT4_BACK.
Add AV_CH_LAYOUT_7POINT1POINT2 and AV_CHANNEL_LAYOUT_7POINT1POINT2.
Add AV_CH_LAYOUT_7POINT1POINT4_BACK and AV_CHANNEL_LAYOUT_7POINT1POINT4_BACK.
2023-10-06 - 804be7f9e3c - lavc 60.30.101 - avcodec.h
AVCodecContext.coded_side_data may now be used during decoding, to be set
by user before calling avcodec_open2() for initialization.
2023-10-06 - 5432d2aacad - lavc 60.15.100 - avformat.h
Deprecate AVFormatContext.{nb_,}side_data, av_stream_add_side_data(),
av_stream_new_side_data(), and av_stream_get_side_data(). Side data fields
from AVFormatContext.codecpar should be used from now on.
2023-10-06 - 21d7cc6fa9a - lavc 60.30.100 - codec_par.h
Added {nb_,}coded_side_data to AVCodecParameters.
The AVCodecParameters helpers will copy it to and from its AVCodecContext
namesake.
2023-10-06 - 74279227dd2 - lavc 60.29.100 - packet.h
Added av_packet_side_data_new(), av_packet_side_data_add(),
av_packet_side_data_get(), av_packet_side_data_remove, and
av_packet_side_data_free().
2023-10-03 - ea14e8bc302 - lavc 60.28.100 - codec_par.h defs.h
Move the definition of enum AVFieldOrder from codec_par.h to defs.h.
2023-10-03 - dd48e49d547 - lavf 60.14.100 - avformat.h
Deprecate AVFMT_ALLOW_FLUSH without replacement. Users can always
flush any muxer by sending a NULL packet.
2023-09-28 - 8e1ef7c38f6 - lavu 58.27.100 - pixfmt.h
Add AV_PIX_FMT_GBRAP14BE, AV_PIX_FMT_GBRAP14LE pixel formats.
2023-09-28 - 05f8b2ca0f7 - lavu 58.26.100 - hwcontext_cuda.h
Add AV_CUDA_USE_CURRENT_CONTEXT.
2023-09-19 - ba9cd06c763 - lavu 58.25.100 - avutil.h
Make AV_TIME_BASE_Q compatible with C++.
2023-09-18 - 85e075587dc - lavf 60 - avformat.h
Deprecate AVFMT_FLAG_SHORTEST without replacement.
2023-09-07 - 423b6a7e493 - lavu 58.24.100 - imgutils.h
Add av_image_copy2(), a wrapper around the av_image_copy()
to overcome limitations of automatic conversions.
2023-09-07 - 5094d1f429e - lavu 58.23.100 - fifo.h
Constify the AVFifo pointees in av_fifo_peek() and av_fifo_peek_to_cb().
2023-09-07 - fa4bf5793a0 - lavu 58.22.100 - audio_fifo.h
Constify some pointees in av_audio_fifo_write(), av_audio_fifo_read(),
av_audio_fifo_peek() and av_audio_fifo_peek_at().
2023-09-07 - 9bf31f60960 - lavu 58.21.100 - samplefmt.h
Constify some pointees in av_samples_copy() and av_samples_set_silence().
2023-09-07 - 41285890e03 - lavu 58.20.100 - imgutils.h
Constify some pointees in av_image_copy(), av_image_copy_uc_from() and
av_image_fill_black().
2023-09-07 - 2a68d945cd7 - lavf 60.12.100 - avio.h
Constify the buffer pointees in the write_packet and write_data_type
callbacks of AVIOContext on the next major bump.
2023-09-07 - 8238bc0b5e3 - lavc 60.26.100 - defs.h
Add AV_PROFILE_* and AV_LEVEL_* replacements in defs.h for the
defines from avcodec.h. The latter are deprecated.
2023-09-06 - b6627a57f41 - lavc 60.25.101 - avcodec.h
AVCodecContext.rc_buffer_size may now be set by decoders.
2023-09-02 - 25ecc94d58f - lavu 58.19.100 - executor.h
Add AVExecutor API
2023-09-01 - 139e54911c8 - lavc 60.25.100 - avfft.h
The entire header will be deprecated and removed in two major bumps.
For a replacement to av_dct, av_rdft, av_fft and av_mdct, use
the new API from libavutil/tx.h.
2023-09-01 - 11e22730e1e - lavu 58.18.100 - tx.h
Add AV_TX_REAL_TO_REAL and AV_TX_REAL_TO_IMAGINARY
2023-08-18 - ff094f5ebbd - lavu 58.17.100 - channel_layout.h
All AV_CHANNEL_LAYOUT_* macros are now compatible with C++ 17 and older.
2023-08-08 - 5012b4ab4ca - lavu 58.15.100 - video_hint.h
Add AVVideoHint API.
2023-08-08 - 5012b4ab4ca - lavc 60 - avcodec.h
Deprecate AV_CODEC_FLAG_DROPCHANGED without replacement.
2023-07-05 - d694c25b44c - lavu 58.14.100 - random_seed.h
Add av_random_bytes()
2023-05-29 - 637afea88ed - lavc 60.16.100 - avcodec.h codec_id.h
Add AV_CODEC_ID_EVC, FF_PROFILE_EVC_BASELINE, and FF_PROFILE_EVC_MAIN.
2023-05-29 - 75918016ab1 - lavu 58.12.100 - mathematics.h
Add av_bessel_i0()
2023-05-29 - f3795e18574 - lavc 60.15.100 - avcodec.h
Add AVHWAccel.update_thread_context, AVHWAccel.free_frame_priv,
AVHWAccel.flush.
2023-05-29 - db1d0227812 - lavu 58.11.100 - hwcontext_vulkan.h
Add AVVulkanDeviceContext.lock_queue, AVVulkanDeviceContext.unlock_queue,
AVVulkanFramesContext.format, AVVulkanFramesContext.lock_frame,
AVVulkanFramesContext.unlock_frame, AVVkFrame.queue_family.
Deprecate AV_VK_FRAME_FLAG_CONTIGUOUS_MEMORY (use multiplane images instead).
2023-05-29 - bef86ba86cc - lavu 58.10.100 - pixfmt.h
Add AV_PIX_FMT_P212BE, AV_PIX_FMT_P212LE, AV_PIX_FMT_P412BE,
AV_PIX_FMT_P412LE.
2023-05-18 - 01d444c077e - lavu 58.8.100 - frame.h
Add av_frame_replace().
2023-05-18 - 63767b79a57 - lavu 58 - frame.h
Deprecate AVFrame.palette_has_changed without replacement.
2023-05-15 - 7d1d61cc5f5 - lavc 60 - avcodec.h
Depreate AVCodecContext.ticks_per_frame in favor of
AVCodecContext.framerate (encoding) and
AV_CODEC_PROP_FIELDS (decoding).
2023-05-15 - 70433abf7fb - lavc 60.12.100 - codec_desc.h
Add AV_CODEC_PROP_FIELDS.
2023-05-15 - 8b20d0dcb5c - lavc 60 - codec.h
Depreate AV_CODEC_CAP_SUBFRAMES without replacement.
2023-05-07 - c2ae8e30b7f - lavc 60.11.100 - codec_par.h
Add AVCodecParameters.framerate.
2023-05-04 - 0fc9c1f6828 - lavu 58.7.100 - frame.h
Deprecate AVFrame.interlaced_frame, AVFrame.top_field_first, and
AVFrame.key_frame.
Add AV_FRAME_FLAG_INTERLACED, AV_FRAME_FLAG_TOP_FIELD_FIRST, and
AV_FRAME_FLAG_KEY flags as replacement.
2023-04-10 - 4eaaa38d3df - lavu 58.6.100 - frame.h
av_frame_get_plane_buffer() now accepts const AVFrame*.
2023-04-04 - 61b27b15fc9 - lavu 58.6.100 - hdr_dynamic_metadata.h
Add AV_HDR_PLUS_MAX_PAYLOAD_SIZE.
av_dynamic_hdr_plus_create_side_data() now accepts a user provided
buffer.
2023-03-24 - 632c3499319 - lavfi 9.5.100 - avfilter.h
Add AVFILTER_FLAG_HWDEVICE.
2023-03-21 - 0a3ce5f7384 - lavu 58.5.100 - hdr_dynamic_metadata.h
Add av_dynamic_hdr_plus_from_t35() and av_dynamic_hdr_plus_to_t35()
functions to convert between raw T.35 payloads containing dynamic
HDR10+ metadata and their parsed representations as AVDynamicHDRPlus.
2023-03-17 - 3be46ee7672 - lavu 58.4.100 - hdr_dynamic_vivid_metadata.h
Add two group of three spline params.
Deprecate previous define which only supports one group of params.
2023-03-02 - 373ef1c4fae - lavc 60.6.100 - avcodec.h
Add FF_PROFILE_EAC3_DDP_ATMOS, FF_PROFILE_TRUEHD_ATMOS,
FF_PROFILE_DTS_HD_MA_X and FF_PROFILE_DTS_HD_MA_X_IMAX.
2023-02-25 - f4593775436 - lavc 60.5.100 - avcodec.h
Add FF_PROFILE_HEVC_SCC.
-------- 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.
@@ -2111,7 +1282,7 @@ 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

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 = 7.0.2
PROJECT_NUMBER = 4.3.1
# 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

View File

@@ -19,7 +19,6 @@ 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/community.html \
doc/developer.html \
doc/faq.html \
doc/fate.html \
@@ -28,9 +27,6 @@ HTMLPAGES = $(AVPROGS-yes:%=doc/%.html) $(AVPROGS-yes:%=doc/%-all.html) $(COMP
doc/mailing-list-faq.html \
doc/nut.html \
doc/platform.html \
$(SRC_PATH)/doc/bootstrap.min.css \
$(SRC_PATH)/doc/style.min.css \
$(SRC_PATH)/doc/default.css \
TXTPAGES = doc/fate.txt \
@@ -106,7 +102,7 @@ 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)
$(M)$(SRC_PATH)/doc/doxy-wrapper.sh $$PWD/doc/doxy $(SRC_PATH) doc/Doxyfile $(DOXYGEN) $(DOXY_INPUT);
$(M)OUT_DIR=$$PWD/doc/doxy; cd $(SRC_PATH); ./doc/doxy-wrapper.sh $$OUT_DIR $< $(DOXYGEN) $(DOXY_INPUT);
install-doc: install-html install-man

View File

@@ -3,9 +3,9 @@
The FFmpeg developers.
For details about the authorship, see the Git history of the project
(https://git.ffmpeg.org/ffmpeg), e.g. by typing the command
(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
@command{git log} in the FFmpeg source directory, or browsing the
online repository at @url{https://git.ffmpeg.org/ffmpeg}.
online repository at @url{http://source.ffmpeg.org}.
Maintainers for the specific components are listed in the file
@file{MAINTAINERS} in the source code tree.

View File

@@ -81,7 +81,7 @@ Top-left position.
@end table
@item tick_rate
Set the tick rate (@emph{time_scale / num_units_in_display_tick}) in
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
@@ -132,36 +132,6 @@ the header stored in extradata to the key packets:
ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts
@end example
@section dv_error_marker
Blocks in DV which are marked as damaged are replaced by blocks of the specified color.
@table @option
@item color
The color to replace damaged blocks by
@item sta
A 16 bit mask which specifies which of the 16 possible error status values are
to be replaced by colored blocks. 0xFFFE is the default which replaces all non 0
error status values.
@table @samp
@item ok
No error, no concealment
@item err
Error, No concealment
@item res
Reserved
@item notok
Error or concealment
@item notres
Not reserved
@item Aa, Ba, Ca, Ab, Bb, Cb, A, B, C, a, b, erri, erru
The specific error status code
@end table
see page 44-46 or section 5.5 of
@url{http://web.archive.org/web/20060927044735/http://www.smpte.org/smpte_store/standards/pdf/s314m.pdf}
@end table
@section eac3_core
Extract the core from a E-AC-3 stream, dropping extra channels.
@@ -199,13 +169,6 @@ Identical to @option{pass_types}, except the units in the given set
removed and all others passed through.
@end table
The types used by pass_types and remove_types correspond to NAL unit types
(nal_unit_type) in H.264, HEVC and H.266 (see Table 7-1 in the H.264
and HEVC specifications or Table 5 in the H.266 specification), to
marker values for JPEG (without 0xFF prefix) and to start codes without
start code prefix (i.e. the byte following the 0x000001) for MPEG-2.
For VP8 and VP9, every unit has type zero.
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.
@@ -220,21 +183,6 @@ To remove all AUDs, SEI and filler from an H.265 stream:
ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=remove_types=35|38-40' OUTPUT
@end example
To remove all user data from a MPEG-2 stream, including Closed Captions:
@example
ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=remove_types=178' OUTPUT
@end example
To remove all SEI from a H264 stream, including Closed Captions:
@example
ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=remove_types=6' OUTPUT
@end example
To remove all prefix and suffix SEI from a HEVC stream, including Closed Captions and dynamic HDR:
@example
ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=remove_types=39|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.
@@ -269,16 +217,12 @@ Modify metadata embedded in an H.264 stream.
Insert or remove AUD NAL units in all access units of the stream.
@table @samp
@item pass
@item insert
@item remove
@end table
Default is pass.
@item sample_aspect_ratio
Set the sample aspect ratio of the stream in the VUI parameters.
See H.264 table E-1.
@item overscan_appropriate_flag
Set whether the stream is suitable for display using overscan
@@ -300,7 +244,7 @@ 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 (time_scale / num_units_in_tick) in the VUI
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).
@@ -309,11 +253,6 @@ 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 zero_new_constraint_set_flags
Zero constraint_set4_flag and constraint_set5_flag in the SPS. These
bits were reserved in a previous version of the H.264 spec, and thus
some hardware decoders require these to be zero. The result of zeroing
this is still a valid bitstream.
@item crop_left
@item crop_right
@@ -337,37 +276,6 @@ insert the string ``hello'' associated with the given UUID.
@item delete_filler
Deletes both filler NAL units and filler SEI messages.
@item display_orientation
Insert, extract or remove Display orientation SEI messages.
See H.264 section D.1.27 and D.2.27 for syntax and semantics.
@table @samp
@item pass
@item insert
@item remove
@item extract
@end table
Default is pass.
Insert mode works in conjunction with @code{rotate} and @code{flip} options.
Any pre-existing Display orientation messages will be removed in insert or remove mode.
Extract mode attaches the display matrix to the packet as side data.
@item rotate
Set rotation in display orientation SEI (anticlockwise angle in degrees).
Range is -360 to +360. Default is NaN.
@item flip
Set flip in display orientation SEI.
@table @samp
@item horizontal
@item vertical
@end table
Default is unset.
@item level
Set the level in the SPS. Refer to H.264 section A.3 and tables A-1
to A-5.
@@ -404,6 +312,9 @@ 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.
@@ -436,8 +347,8 @@ 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 (time_scale /
num_units_in_tick). Combined with @option{num_ticks_poc_diff_one}, this can
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.
@@ -550,6 +461,10 @@ 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.
@@ -614,67 +529,20 @@ container. Can be used for fuzzing or testing error resilience/concealment.
Parameters:
@table @option
@item amount
Accepts an expression whose evaluation per-packet determines how often bytes in that
packet will be modified. A value below 0 will result in a variable frequency.
Default is 0 which results in no modification. However, if neither amount nor drop is specified,
amount will be set to @var{-1}. See below for accepted variables.
@item drop
Accepts an expression evaluated per-packet whose value determines whether that packet is dropped.
Evaluation to a positive value results in the packet being dropped. Evaluation to a negative
value results in a variable chance of it being dropped, roughly inverse in proportion to the magnitude
of the value. Default is 0 which results in no drops. See below for accepted variables.
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
Accepts a non-negative integer, which assigns a variable chance of it being dropped, roughly inverse
in proportion to the value. Default is 0 which results in no drops. This option is kept for backwards
compatibility and is equivalent to setting drop to a negative value with the same magnitude
i.e. @code{dropamount=4} is the same as @code{drop=-4}. Ignored if drop is also specified.
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
Both @code{amount} and @code{drop} accept expressions containing the following variables:
@table @samp
@item n
The index of the packet, starting from zero.
@item tb
The timebase for packet timestamps.
@item pts
Packet presentation timestamp.
@item dts
Packet decoding timestamp.
@item nopts
Constant representing AV_NOPTS_VALUE.
@item startpts
First non-AV_NOPTS_VALUE PTS seen in the stream.
@item startdts
First non-AV_NOPTS_VALUE DTS seen in the stream.
@item duration
@itemx d
Packet duration, in timebase units.
@item pos
Packet position in input; may be -1 when unknown or not set.
@item size
Packet size, in bytes.
@item key
Whether packet is marked as a keyframe.
@item state
A pseudo random integer, primarily derived from the content of packet payload.
@end table
@subsection Examples
Apply modification to every byte but don't drop any packets.
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
Drop every video packet not marked as a keyframe after timestamp 30s but do not
modify any of the remaining packets.
@example
ffmpeg -i INPUT -c copy -bsf:v noise=drop='gt(t\,30)*not(key)' output.mkv
@end example
Drop one second of audio every 10 seconds and add some random noise to the rest.
@example
ffmpeg -i INPUT -c copy -bsf:a noise=amount=-1:drop='between(mod(t\,10)\,9\,10)' output.mkv
ffmpeg -i INPUT -c copy -bsf noise[=1] output.mkv
@end example
@section null
@@ -710,14 +578,6 @@ for NTSC frame rate using the @option{frame_rate} option.
ffmpeg -f lavfi -i sine=r=48000:d=1 -c pcm_s16le -bsf pcm_rechunk=r=30000/1001 -f framecrc -
@end example
@section pgs_frame_merge
Merge a sequence of PGS Subtitle segments ending with an "end of display set"
segment into a single packet.
This is required by some containers that support PGS subtitles
(muxer @code{matroska}).
@section prores_metadata
Modify color property metadata embedded in prores stream.
@@ -815,100 +675,6 @@ Remove extradata from all frames.
@end table
@end table
@section setts
Set PTS and DTS in packets.
It accepts the following parameters:
@table @option
@item ts
@item pts
@item dts
Set expressions for PTS, DTS or both.
@item duration
Set expression for duration.
@item time_base
Set output time base.
@end table
The expressions are evaluated through the eval API and can contain the following
constants:
@table @option
@item N
The count of the input packet. Starting from 0.
@item TS
The demux timestamp in input in case of @code{ts} or @code{dts} option or presentation
timestamp in case of @code{pts} option.
@item POS
The original position in the file of the packet, or undefined if undefined
for the current packet
@item DTS
The demux timestamp in input.
@item PTS
The presentation timestamp in input.
@item DURATION
The duration in input.
@item STARTDTS
The DTS of the first packet.
@item STARTPTS
The PTS of the first packet.
@item PREV_INDTS
The previous input DTS.
@item PREV_INPTS
The previous input PTS.
@item PREV_INDURATION
The previous input duration.
@item PREV_OUTDTS
The previous output DTS.
@item PREV_OUTPTS
The previous output PTS.
@item PREV_OUTDURATION
The previous output duration.
@item NEXT_DTS
The next input DTS.
@item NEXT_PTS
The next input PTS.
@item NEXT_DURATION
The next input duration.
@item TB
The timebase of stream packet belongs.
@item TB_OUT
The output timebase.
@item SR
The sample rate of stream packet belongs.
@item NOPTS
The AV_NOPTS_VALUE constant.
@end table
For example, to set PTS equal to DTS (not recommended if B-frames are involved):
@example
ffmpeg -i INPUT -c:a copy -bsf:a setts=pts=DTS out.mkv
@end example
@section showinfo
Log basic packet information. Mainly useful for testing, debugging,
and development.
@anchor{text2movsub}
@section text2movsub

File diff suppressed because one or more lines are too long

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@@ -3,7 +3,7 @@
@c man begin CODEC OPTIONS
libavcodec provides some generic global options, which can be set on
all the encoders and decoders. In addition, each codec may support
all the encoders and decoders. In addition each codec may support
so-called private options, which are specific for a given codec.
Sometimes, a global option may only affect a specific kind of codec,
@@ -50,6 +50,8 @@ 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
@@ -70,6 +72,10 @@ 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
@@ -78,6 +84,40 @@ Use closed gop.
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 +184,24 @@ 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 +210,8 @@ 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 +220,8 @@ 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 +242,14 @@ 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 +303,29 @@ 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 +336,18 @@ 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 +412,11 @@ 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 +439,19 @@ 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.
@@ -554,6 +668,9 @@ sab diamond motion estimation
@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 +714,40 @@ 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})
@@ -617,6 +763,20 @@ 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.
@@ -644,8 +804,6 @@ for codecs that support it. See also @file{doc/examples/export_mvs.c}.
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})
@@ -658,14 +816,13 @@ 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 error @var{integer} (@emph{encoding,video})
@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
implementation supports multi-threading.
@@ -678,6 +835,12 @@ 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.
@@ -697,24 +860,76 @@ profiles are documented in the relevant encoder documentation.
@item level @var{integer} (@emph{encoding,audio,video})
Set the encoder level. This level depends on the specific codec, and
might correspond to the profile level. It is set by default to
@samp{unknown}.
Possible values:
@table @samp
@item unknown
@end table
@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})
@@ -754,24 +969,48 @@ 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})
See @ref{channel layout syntax,,the Channel Layout section in the ffmpeg-utils(1) manual,ffmpeg-utils}
for the required syntax.
Possible values:
@table @samp
@end table
@item request_channel_layout @var{integer} (@emph{decoding,audio})
Possible values:
@table @samp
@end table
@item rc_max_vbv_use @var{float} (@emph{encoding,video})
@item rc_min_vbv_use @var{float} (@emph{encoding,video})
@item ticks_per_frame @var{integer} (@emph{decoding/encoding,audio,video})
@item color_primaries @var{integer} (@emph{decoding/encoding,video})
Possible values:
@@ -871,12 +1110,6 @@ 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})
@@ -886,11 +1119,9 @@ Possible values:
@table @samp
@item tv
@item mpeg
@item limited
MPEG (219*2^(n-8))
@item pc
@item jpeg
@item full
JPEG (2^n-1)
@end table

View File

@@ -1,177 +0,0 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle Community
@titlepage
@center @titlefont{Community}
@end titlepage
@top
@contents
@anchor{Organisation}
@chapter Organisation
The FFmpeg project is organized through a community working on global consensus.
Decisions are taken by the ensemble of active members, through voting and are aided by two committees.
@anchor{General Assembly}
@chapter General Assembly
The ensemble of active members is called the General Assembly (GA).
The General Assembly is sovereign and legitimate for all its decisions regarding the FFmpeg project.
The General Assembly is made up of active contributors.
Contributors are considered "active contributors" if they have authored more than 20 patches in the last 36 months in the main FFmpeg repository, or if they have been voted in by the GA.
The list of active contributors is updated twice each year, on 1st January and 1st July, 0:00 UTC.
Additional members are added to the General Assembly through a vote after proposal by a member of the General Assembly. They are part of the GA for two years, after which they need a confirmation by the GA.
A script to generate the current members of the general assembly (minus members voted in) can be found in `tools/general_assembly.pl`.
@anchor{Voting}
@chapter Voting
Voting is done using a ranked voting system, currently running on https://vote.ffmpeg.org/ .
Majority vote means more than 50% of the expressed ballots.
@anchor{Technical Committee}
@chapter Technical Committee
The Technical Committee (TC) is here to arbitrate and make decisions when technical conflicts occur in the project. They will consider the merits of all the positions, judge them and make a decision.
The TC resolves technical conflicts but is not a technical steering committee.
Decisions by the TC are binding for all the contributors.
Decisions made by the TC can be re-opened after 1 year or by a majority vote of the General Assembly, requested by one of the member of the GA.
The TC is elected by the General Assembly for a duration of 1 year, and is composed of 5 members. Members can be re-elected if they wish. A majority vote in the General Assembly can trigger a new election of the TC.
The members of the TC can be elected from outside of the GA. Candidates for election can either be suggested or self-nominated.
The conflict resolution process is detailed in the resolution process document.
The TC can be contacted at <tc@@ffmpeg>.
@anchor{Resolution Process}
@section Resolution Process
The Technical Committee (TC) is here to arbitrate and make decisions when technical conflicts occur in the project.
The TC main role is to resolve technical conflicts. It is therefore not a technical steering committee, but it is understood that some decisions might impact the future of the project.
@subsection Seizing
The TC can take possession of any technical matter that it sees fit.
To involve the TC in a matter, email tc@ or CC them on an ongoing discussion.
As members of TC are developers, they also can email tc@ to raise an issue.
@subsection Announcement
The TC, once seized, must announce itself on the main mailing list, with a [TC] tag.
The TC has 2 modes of operation: a RFC one and an internal one.
If the TC thinks it needs the input from the larger community, the TC can call for a RFC. Else, it can decide by itself.
If the disagreement involves a member of the TC, that member should recuse themselves from the decision.
The decision to use a RFC process or an internal discussion is a discretionary decision of the TC.
The TC can also reject a seizure for a few reasons such as: the matter was not discussed enough previously; it lacks expertise to reach a beneficial decision on the matter; or the matter is too trivial.
@subsection RFC call
In the RFC mode, one person from the TC posts on the mailing list the technical question and will request input from the community.
The mail will have the following specification:
a precise title
a specific tag [TC RFC]
a top-level email
contain a precise question that does not exceed 100 words and that is answerable by developers
may have an extra description, or a link to a previous discussion, if deemed necessary,
contain a precise end date for the answers.
The answers from the community must be on the main mailing list and must have the following specification:
keep the tag and the title unchanged
limited to 400 words
a first-level, answering directly to the main email
answering to the question.
Further replies to answers are permitted, as long as they conform to the community standards of politeness, they are limited to 100 words, and are not nested more than once. (max-depth=2)
After the end-date, mails on the thread will be ignored.
Violations of those rules will be escalated through the Community Committee.
After all the emails are in, the TC has 96 hours to give its final decision. Exceptionally, the TC can request an extra delay, that will be notified on the mailing list.
@subsection Within TC
In the internal case, the TC has 96 hours to give its final decision. Exceptionally, the TC can request an extra delay.
@subsection Decisions
The decisions from the TC will be sent on the mailing list, with the [TC] tag.
Internally, the TC should take decisions with a majority, or using ranked-choice voting.
The decision from the TC should be published with a summary of the reasons that lead to this decision.
The decisions from the TC are final, until the matters are reopened after no less than one year.
@anchor{Community Committee}
@chapter Community Committee
The Community Committee (CC) is here to arbitrage and make decisions when inter-personal conflicts occur in the project. It will decide quickly and take actions, for the sake of the project.
The CC can remove privileges of offending members, including removal of commit access and temporary ban from the community.
Decisions made by the CC can be re-opened after 1 year or by a majority vote of the General Assembly. Indefinite bans from the community must be confirmed by the General Assembly, in a majority vote.
The CC is elected by the General Assembly for a duration of 1 year, and is composed of 5 members. Members can be re-elected if they wish. A majority vote in the General Assembly can trigger a new election of the CC.
The members of the CC can be elected from outside of the GA. Candidates for election can either be suggested or self-nominated.
The CC is governed by and responsible for enforcing the Code of Conduct.
The CC can be contacted at <cc@@ffmpeg>.
@anchor{Code of Conduct}
@chapter Code of Conduct
Be friendly and respectful towards others and third parties.
Treat others the way you yourself want to be treated.
Be considerate. Not everyone shares the same viewpoint and priorities as you do.
Different opinions and interpretations help the project.
Looking at issues from a different perspective assists development.
Do not assume malice for things that can be attributed to incompetence. Even if
it is malice, it's rarely good to start with that as initial assumption.
Stay friendly even if someone acts contrarily. Everyone has a bad day
once in a while.
If you yourself have a bad day or are angry then try to take a break and reply
once you are calm and without anger if you have to.
Try to help other team members and cooperate if you can.
The goal of software development is to create technical excellence, not for any
individual to be better and "win" against the others. Large software projects
are only possible and successful through teamwork.
If someone struggles do not put them down. Give them a helping hand
instead and point them in the right direction.
Finally, keep in mind the immortal words of Bill and Ted,
"Be excellent to each other."
@bye

View File

@@ -25,19 +25,6 @@ enabled decoders.
A description of some of the currently available video decoders
follows.
@section av1
AOMedia Video 1 (AV1) decoder.
@subsection Options
@table @option
@item operating_point
Select an operating point of a scalable AV1 bitstream (0 - 31). Default is 0.
@end table
@section rawvideo
Raw video decoder.
@@ -76,23 +63,13 @@ The following options are supported by the libdav1d wrapper.
@item framethreads
Set amount of frame threads to use during decoding. The default value is 0 (autodetect).
This option is deprecated for libdav1d >= 1.0 and will be removed in the future. Use the
option @code{max_frame_delay} and the global option @code{threads} instead.
@item tilethreads
Set amount of tile threads to use during decoding. The default value is 0 (autodetect).
This option is deprecated for libdav1d >= 1.0 and will be removed in the future. Use the
global option @code{threads} instead.
@item max_frame_delay
Set max amount of frames the decoder may buffer internally. The default value is 0
(autodetect).
@item filmgrain
Apply film grain to the decoded video if present in the bitstream. Defaults to the
internal default of the library.
This option is deprecated and will be removed in the future. See the global option
@code{export_side_data} to export Film Grain parameters instead of applying it.
@item oppoint
Select an operating point of a scalable AV1 bitstream (0 - 31). Defaults to the
@@ -111,108 +88,6 @@ This decoder allows libavcodec to decode AVS2 streams with davs2 library.
@c man end VIDEO DECODERS
@section libuavs3d
AVS3-P2/IEEE1857.10 video decoder.
libuavs3d allows libavcodec to decode AVS3 streams.
Requires the presence of the libuavs3d headers and library during configuration.
You need to explicitly configure the build with @code{--enable-libuavs3d}.
@subsection Options
The following option is supported by the libuavs3d wrapper.
@table @option
@item frame_threads
Set amount of frame threads to use during decoding. The default value is 0 (autodetect).
@end table
@section libxevd
eXtra-fast Essential Video Decoder (XEVD) MPEG-5 EVC decoder wrapper.
This decoder requires the presence of the libxevd headers and library
during configuration. You need to explicitly configure the build with
@option{--enable-libxevd}.
The xevd project website is at @url{https://github.com/mpeg5/xevd}.
@subsection Options
The following options are supported by the libxevd wrapper.
The xevd-equivalent options or values are listed in parentheses for easy migration.
To get a more accurate and extensive documentation of the libxevd options,
invoke the command @code{xevd_app --help} or consult the libxevd documentation.
@table @option
@item threads (@emph{threads})
Force to use a specific number of threads
@end table
@section QSV Decoders
The family of Intel QuickSync Video decoders (VC1, MPEG-2, H.264, HEVC,
JPEG/MJPEG, VP8, VP9, AV1).
@subsection Common Options
The following options are supported by all qsv decoders.
@table @option
@item @var{async_depth}
Internal parallelization depth, the higher the value the higher the latency.
@item @var{gpu_copy}
A GPU-accelerated copy between video and system memory
@table @samp
@item default
@item on
@item off
@end table
@end table
@subsection HEVC Options
Extra options for hevc_qsv.
@table @option
@item @var{load_plugin}
A user plugin to load in an internal session
@table @samp
@item none
@item hevc_sw
@item hevc_hw
@end table
@item @var{load_plugins}
A :-separate list of hexadecimal plugin UIDs to load in an internal session
@end table
@section v210
Uncompressed 4:2:2 10-bit decoder.
@subsection Options
@table @option
@item custom_stride
Set the line size of the v210 data in bytes. The default value is 0
(autodetect). You can use the special -1 value for a strideless v210 as seen in
BOXX files.
@end table
@c man end VIDEO DECODERS
@chapter Audio Decoders
@c man begin AUDIO DECODERS
@@ -232,7 +107,7 @@ the undocumented RealAudio 3 (a.k.a. dnet).
@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. The default value is 1.
from the AC-3 stream. This factor is applied exponentially.
There are 3 notable scale factor ranges:
@table @option
@item drc_scale == 0
@@ -377,169 +252,6 @@ Enabled by default.
@end table
@section libaribcaption
Yet another ARIB STD-B24 caption decoder using external @dfn{libaribcaption}
library.
Implements profiles A and C of the Japanse ARIB STD-B24 standard,
Brazilian ABNT NBR 15606-1, and Philippines version of ISDB-T.
Requires the presence of the libaribcaption headers and library
(@url{https://github.com/xqq/libaribcaption}) during configuration.
You need to explicitly configure the build with @code{--enable-libaribcaption}.
If both @dfn{libaribb24} and @dfn{libaribcaption} are enabled, @dfn{libaribcaption}
decoder precedes.
@subsection libaribcaption Decoder Options
@table @option
@item -sub_type @var{subtitle_type}
Specifies the format of the decoded subtitles.
@table @samp
@item bitmap
Graphical image.
@item ass
ASS formatted text.
@item text
Simple text based output without formatting.
@end table
The default is @dfn{ass} as same as @dfn{libaribb24} decoder.
Some present players (e.g., @dfn{mpv}) expect ASS format for ARIB caption.
@item -caption_encoding @var{encoding_scheme}
Specifies the encoding scheme of input subtitle text.
@table @samp
@item auto
Automatically detect text encoding (default).
@item jis
8bit-char JIS encoding defined in ARIB STD B24.
This encoding used in Japan for ISDB captions.
@item utf8
UTF-8 encoding defined in ARIB STD B24.
This encoding is used in Philippines for ISDB-T captions.
@item latin
Latin character encoding defined in ABNT NBR 15606-1.
This encoding is used in South America for SBTVD / ISDB-Tb captions.
@end table
@item -font @var{font_name[,font_name2,...]}
Specify comma-separated list of font family names to be used for @dfn{bitmap}
or @dfn{ass} type subtitle rendering.
Only first font name is used for @dfn{ass} type subtitle.
If not specified, use internaly defined default font family.
@item -ass_single_rect @var{boolean}
ARIB STD-B24 specifies that some captions may be displayed at different
positions at a time (multi-rectangle subtitle).
Since some players (e.g., old @dfn{mpv}) can't handle multiple ASS rectangles
in a single AVSubtitle, or multiple ASS rectangles of indeterminate duration
with the same start timestamp, this option can change the behavior so that
all the texts are displayed in a single ASS rectangle.
The default is @var{false}.
If your player cannot handle AVSubtitles with multiple ASS rectangles properly,
set this option to @var{true} or define @env{ASS_SINGLE_RECT=1} to change
default behavior at compilation.
@item -force_outline_text @var{boolean}
Specify whether always render outline text for all characters regardless of
the indication by charactor style.
The default is @var{false}.
@item -outline_width @var{number} (0.0 - 3.0)
Specify width for outline text, in dots (relative).
The default is @var{1.5}.
@item -ignore_background @var{boolean}
Specify whether to ignore background color rendering.
The default is @var{false}.
@item -ignore_ruby @var{boolean}
Specify whether to ignore rendering for ruby-like (furigana) characters.
The default is @var{false}.
@item -replace_drcs @var{boolean}
Specify whether to render replaced DRCS characters as Unicode characters.
The default is @var{true}.
@item -replace_msz_ascii @var{boolean}
Specify whether to replace MSZ (Middle Size; half width) fullwidth
alphanumerics with halfwidth alphanumerics.
The default is @var{true}.
@item -replace_msz_japanese @var{boolean}
Specify whether to replace some MSZ (Middle Size; half width) fullwidth
japanese special characters with halfwidth ones.
The default is @var{true}.
@item -replace_msz_glyph @var{boolean}
Specify whether to replace MSZ (Middle Size; half width) characters
with halfwidth glyphs if the fonts supports it.
This option works under FreeType or DirectWrite renderer
with Adobe-Japan1 compliant fonts.
e.g., IBM Plex Sans JP, Morisawa BIZ UDGothic, Morisawa BIZ UDMincho,
Yu Gothic, Yu Mincho, and Meiryo.
The default is @var{true}.
@item -canvas_size @var{image_size}
Specify the resolution of the canvas to render subtitles to; usually, this
should be frame size of input video.
This only applies when @code{-subtitle_type} is set to @var{bitmap}.
The libaribcaption decoder assumes input frame size for bitmap rendering as below:
@enumerate
@item
PROFILE_A : 1440 x 1080 with SAR (PAR) 4:3
@item
PROFILE_C : 320 x 180 with SAR (PAR) 1:1
@end enumerate
If actual frame size of input video does not match above assumption,
the rendered captions may be distorted.
To make the captions undistorted, add @code{-canvas_size} option to specify
actual input video size.
Note that the @code{-canvas_size} option is not required for video with
different size but same aspect ratio.
In such cases, the caption will be stretched or shrunk to actual video size
if @code{-canvas_size} option is not specified.
If @code{-canvas_size} option is specified with different size,
the caption will be stretched or shrunk as specified size with calculated SAR.
@end table
@subsection libaribcaption decoder usage examples
Display MPEG-TS file with ARIB subtitle by @code{ffplay} tool:
@example
ffplay -sub_type bitmap MPEG.TS
@end example
Display MPEG-TS file with input frame size 1920x1080 by @code{ffplay} tool:
@example
ffplay -sub_type bitmap -canvas_size 1920x1080 MPEG.TS
@end example
Embed ARIB subtitle in transcoded video:
@example
ffmpeg -sub_type bitmap -i src.m2t -filter_complex "[0:v][0:s]overlay" -vcodec h264 dest.mp4
@end example
@section dvbsub
@subsection Options
@@ -547,8 +259,6 @@ ffmpeg -sub_type bitmap -i src.m2t -filter_complex "[0:v][0:s]overlay" -vcodec h
@table @option
@item compute_clut
@table @option
@item -2
Compute clut once if no matching CLUT is in the stream.
@item -1
Compute clut if no matching CLUT is in the stream.
@item 0

View File

@@ -25,13 +25,6 @@ Audible Format 2, 3, and 4 demuxer.
This demuxer is used to demux Audible Format 2, 3, and 4 (.aa) files.
@section aac
Raw Audio Data Transport Stream AAC demuxer.
This demuxer is used to demux an ADTS input containing a single AAC stream
alongwith any ID3v1/2 or APE tags in it.
@section apng
Animated Portable Network Graphics demuxer.
@@ -44,15 +37,12 @@ between the last fcTL and IEND chunks.
@table @option
@item -ignore_loop @var{bool}
Ignore the loop variable in the file if set. Default is enabled.
Ignore the loop variable in the file if set.
@item -max_fps @var{int}
Maximum framerate in frames per second. Default of 0 imposes no limit.
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). Default is 15.
(0 meaning as fast as possible).
@end table
@section asf
@@ -103,7 +93,8 @@ 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.
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
@@ -157,16 +148,6 @@ directive) will be reduced based on their specified Out point.
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.
This directive is deprecated, use @code{file_packet_meta} instead.
@item @code{file_packet_meta @var{key} @var{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{option @var{key} @var{value}}
Option to access, open and probe the file.
Can be present multiple times.
@item @code{stream}
Introduce a stream in the virtual file.
@@ -184,20 +165,6 @@ subfiles will be used.
This is especially useful for MPEG-PS (VOB) files, where the order of the
streams is not reliable.
@item @code{stream_meta @var{key} @var{value}}
Metadata for the stream.
Can be present multiple times.
@item @code{stream_codec @var{value}}
Codec for the stream.
@item @code{stream_extradata @var{hex_string}}
Extradata for the string, encoded in hexadecimal.
@item @code{chapter @var{id} @var{start} @var{end}}
Add a chapter. @var{id} is an unique identifier, possibly small and
consecutive.
@end table
@subsection Options
@@ -207,8 +174,7 @@ This demuxer accepts the following option:
@table @option
@item safe
If set to 1, reject unsafe file paths and directives.
A file path is considered safe if it
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
@@ -218,6 +184,9 @@ 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.
@@ -274,217 +243,11 @@ 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.
@subsection Options
This demuxer accepts the following option:
@table @option
@item cenc_decryption_key
16-byte key, in hex, to decrypt files encrypted using ISO Common Encryption (CENC/AES-128 CTR; ISO/IEC 23001-7).
@end table
@section dvdvideo
DVD-Video demuxer, powered by libdvdnav and libdvdread.
Can directly ingest DVD titles, specifically sequential PGCs, into
a conversion pipeline. Menu assets, such as background video or audio,
can also be demuxed given the menu's coordinates (at best effort).
Seeking is not supported at this time.
Block devices (DVD drives), ISO files, and directory structures are accepted.
Activate with @code{-f dvdvideo} in front of one of these inputs.
This demuxer does NOT have decryption code of any kind. You are on your own
working with encrypted DVDs, and should not expect support on the matter.
Underlying playback is handled by libdvdnav, and structure parsing by libdvdread.
FFmpeg must be built with GPL library support available as well as the
configure switches @code{--enable-libdvdnav} and @code{--enable-libdvdread}.
You will need to provide either the desired "title number" or exact PGC/PG coordinates.
Many open-source DVD players and tools can aid in providing this information.
If not specified, the demuxer will default to title 1 which works for many discs.
However, due to the flexibility of the format, it is recommended to check manually.
There are many discs that are authored strangely or with invalid headers.
If the input is a real DVD drive, please note that there are some drives which may
silently fail on reading bad sectors from the disc, returning random bits instead
which is effectively corrupt data. This is especially prominent on aging or rotting discs.
A second pass and integrity checks would be needed to detect the corruption.
This is not an FFmpeg issue.
@subsection Background
DVD-Video is not a directly accessible, linear container format in the
traditional sense. Instead, it allows for complex and programmatic playback of
carefully muxed MPEG-PS streams that are stored in headerless VOB files.
To the end-user, these streams are known simply as "titles", but the actual
logical playback sequence is defined by one or more "PGCs", or Program Group Chains,
within the title. The PGC is in turn comprised of multiple "PGs", or Programs",
which are the actual video segments (and for a typical video feature, sequentially
ordered). The PGC structure, along with stream layout and metadata, are stored in
IFO files that need to be parsed. PGCs can be thought of as playlists in easier terms.
An actual DVD player relies on user GUI interaction via menus and an internal VM
to drive the direction of demuxing. Generally, the user would either navigate (via menus)
or automatically be redirected to the PGC of their choice. During this process and
the subsequent playback, the DVD player's internal VM also maintains a state and
executes instructions that can create jumps to different sectors during playback.
This is why libdvdnav is involved, as a linear read of the MPEG-PS blobs on the
disc (VOBs) is not enough to produce the right sequence in many cases.
There are many other DVD structures (a long subject) that will not be discussed here.
NAV packets, in particular, are handled by this demuxer to build accurate timing
but not emitted as a stream. For a good high-level understanding, refer to:
@url{https://code.videolan.org/videolan/libdvdnav/-/blob/master/doc/dvd_structures}
@subsection Options
This demuxer accepts the following options:
@table @option
@item title @var{int}
The title number to play. Must be set if @option{pgc} and @option{pg} are not set.
Not applicable to menus.
Default is 0 (auto), which currently only selects the first available title (title 1)
and notifies the user about the implications.
@item chapter_start @var{int}
The chapter, or PTT (part-of-title), number to start at. Not applicable to menus.
Default is 1.
@item chapter_end @var{int}
The chapter, or PTT (part-of-title), number to end at. Not applicable to menus.
Default is 0, which is a special value to signal end at the last possible chapter.
@item angle @var{int}
The video angle number, referring to what is essentially an additional
video stream that is composed from alternate frames interleaved in the VOBs.
Not applicable to menus.
Default is 1.
@item region @var{int}
The region code to use for playback. Some discs may use this to default playback
at a particular angle in different regions. This option will not affect the region code
of a real DVD drive, if used as an input. Not applicable to menus.
Default is 0, "world".
@item menu @var{bool}
Demux menu assets instead of navigating a title. Requires exact coordinates
of the menu (@option{menu_lu}, @option{menu_vts}, @option{pgc}, @option{pg}).
Default is false.
@item menu_lu @var{int}
The menu language to demux. In DVD, menus are grouped by language.
Default is 0, the first language unit.
@item menu_vts @var{int}
The VTS where the menu lives, or 0 if it is a VMG menu (root-level).
Default is 0, VMG menu.
@item pgc @var{int}
The entry PGC to start playback, in conjunction with @option{pg}.
Alternative to setting @option{title}.
Chapter markers are not supported at this time.
Must be explicitly set for menus.
Default is 0, automatically resolve from value of @option{title}.
@item pg @var{int}
The entry PG to start playback, in conjunction with @option{pgc}.
Alternative to setting @option{title}.
Chapter markers are not supported at this time.
Default is 0, automatically resolve from value of @option{title}, or
start from the beginning (PG 1) of the menu.
@item preindex @var{bool}
Enable this to have accurate chapter (PTT) markers and duration measurement,
which requires a slow second pass read in order to index the chapter marker
timestamps from NAV packets. This is non-ideal extra work for real optical drives.
It is recommended and faster to use this option with a backup of the DVD structure
stored on a hard drive. Not compatible with @option{pgc} and @option{pg}.
Not applicable to menus.
Default is 0, false.
@item trim @var{bool}
Skip padding cells (i.e. cells shorter than 1 second) from the beginning.
There exist many discs with filler segments at the beginning of the PGC,
often with junk data intended for controlling a real DVD player's
buffering speed and with no other material data value.
Not applicable to menus.
Default is 1, true.
@end table
@subsection Examples
@itemize
@item
Open title 3 from a given DVD structure:
@example
ffmpeg -f dvdvideo -title 3 -i <path to DVD> ...
@end example
@item
Open chapters 3-6 from title 1 from a given DVD structure:
@example
ffmpeg -f dvdvideo -chapter_start 3 -chapter_end 6 -title 1 -i <path to DVD> ...
@end example
@item
Open only chapter 5 from title 1 from a given DVD structure:
@example
ffmpeg -f dvdvideo -chapter_start 5 -chapter_end 5 -title 1 -i <path to DVD> ...
@end example
@item
Demux menu with language 1 from VTS 1, PGC 1, starting at PG 1:
@example
ffmpeg -f dvdvideo -menu 1 -menu_lu 1 -menu_vts 1 -pgc 1 -pg 1 -i <path to DVD> ...
@end example
@end itemize
@section ea
Electronic Arts Multimedia format demuxer.
This format is used by various Electronic Arts games.
@subsection Options
@table @option
@item merge_alpha @var{bool}
Normally the VP6 alpha channel (if exists) is returned as a secondary video
stream, by setting this option you can make the demuxer return a single video
stream which contains the alpha channel in addition to the ordinary video.
@end table
@section imf
Interoperable Master Format demuxer.
This demuxer presents audio and video streams found in an IMF Composition, as
specified in @url{https://doi.org/10.5594/SMPTE.ST2067-2.2020, SMPTE ST 2067-2}.
@example
ffmpeg [-assetmaps <path of ASSETMAP1>,<path of ASSETMAP2>,...] -i <path of CPL> ...
@end example
If @code{-assetmaps} is not specified, the demuxer looks for a file called
@file{ASSETMAP.xml} in the same directory as the CPL.
@section flv, live_flv, kux
@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.
KUX is a flv variant used on the Youku platform.
@example
ffmpeg -f flv -i myfile.flv ...
@@ -561,9 +324,6 @@ It accepts the following options:
@item live_start_index
segment index to start live streams at (negative values are from the end).
@item prefer_x_start
prefer to use #EXT-X-START if it's in playlist instead of live_start_index.
@item allowed_extensions
',' separated list of file extensions that hls is allowed to access.
@@ -586,13 +346,6 @@ Enabled by default for HTTP/1.1 servers.
@item http_seekable
Use HTTP partial requests for downloading HTTP segments.
0 = disable, 1 = enable, -1 = auto, Default is auto.
@item seg_format_options
Set options for the demuxer of media segments using a list of key=value pairs separated by @code{:}.
@item seg_max_retry
Maximum number of times to reload a segment on error, useful when segment skip on network error is not desired.
Default value is 0.
@end table
@section image2
@@ -908,12 +661,6 @@ Set mfra timestamps as PTS
Don't use mfra box to set timestamps
@end table
@item use_tfdt
For fragmented input, set fragment's starting timestamp to @code{baseMediaDecodeTime} from the @code{tfdt} box.
Default is enabled, which will prefer to use the @code{tfdt} box to set DTS. Disable to use the @code{earliest_presentation_time} from the @code{sidx} box.
In either case, the timestamp from the @code{mfra} box will be used if it's available and @code{use_mfra_for} is
set to pts or dts.
@item export_all
Export unrecognized boxes within the @var{udta} box as metadata entries. The first four
characters of the box type are set as the key. Default is false.
@@ -932,22 +679,6 @@ specify.
@item decryption_key
16-byte key, in hex, to decrypt files encrypted using ISO Common Encryption (CENC/AES-128 CTR; ISO/IEC 23001-7).
@item max_stts_delta
Very high sample deltas written in a trak's stts box may occasionally be intended but usually they are written in
error or used to store a negative value for dts correction when treated as signed 32-bit integers. This option lets
the user set an upper limit, beyond which the delta is clamped to 1. Values greater than the limit if negative when
cast to int32 are used to adjust onward dts.
Unit is the track time scale. Range is 0 to UINT_MAX. Default is @code{UINT_MAX - 48000*10} which allows up to
a 10 second dts correction for 48 kHz audio streams while accommodating 99.9% of @code{uint32} range.
@item interleaved_read
Interleave packets from multiple tracks at demuxer level. For badly interleaved files, this prevents playback issues
caused by large gaps between packets in different tracks, as MOV/MP4 do not have packet placement requirements.
However, this can cause excessive seeking on very badly interleaved files, due to seeking between tracks, so disabling
it may prevent I/O issues, at the expense of playback.
@end table
@subsection Audible AAX
@@ -988,10 +719,6 @@ 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.
@item max_packet_size
Set maximum size, in bytes, of packet emitted by the demuxer. Payloads above this size
are split across multiple packets. Range is 1 to INT_MAX/2. Default is 204800 bytes.
@end table
@section mpjpeg
@@ -1105,27 +832,4 @@ which in turn, acts as a ceiling for the size of scripts that can be read.
Default is 1 MiB.
@end table
@section w64
Sony Wave64 Audio demuxer.
This demuxer accepts the following options:
@table @option
@item max_size
See the same option for the @ref{wav} demuxer.
@end table
@anchor{wav}
@section wav
RIFF Wave Audio demuxer.
This demuxer accepts the following options:
@table @option
@item max_size
Specify the maximum packet size in bytes for the demuxed packets. By default
this is set to 0, which means that a sensible value is chosen based on the
input format.
@end table
@c man end DEMUXERS

View File

@@ -10,109 +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.
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}.
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:
@chapter Contributing
There are 2 ways by which code gets into FFmpeg:
@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.
@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
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}.
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 Language
FFmpeg is mainly programmed in the ISO C11 language, except for the public
headers which must stay C99 compatible.
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
@subsection SIMD/DSP
@anchor{SIMD/DSP}
As modern compilers are unable to generate efficient SIMD or other
performance-critical DSP code from plain C, handwritten assembly is used.
Usually such code is isolated in a separate function. Then the standard approach
is writing multiple versions of this function a plain C one that works
everywhere and may also be useful for debugging, and potentially multiple
architecture-specific optimized implementations. Initialization code then
chooses the best available version at runtime and loads it into a function
pointer; the function in question is then always called through this pointer.
The specific syntax used for writing assembly is:
@itemize @bullet
@item
NASM on x86;
@item
GAS on ARM and RISC-V.
@end itemize
A unit testing framework for assembly called @code{checkasm} lives under
@file{tests/checkasm}. All new assembly should come with @code{checkasm} tests;
adding tests for existing assembly that lacks them is also strongly encouraged.
@subsection Other languages
Other languages than C may be used in special cases:
@itemize @bullet
@item
Compiler intrinsics or inline assembly when the code in question cannot be
written in the standard way described in the @ref{SIMD/DSP} section. This
typically applies to code that needs to be inlined.
@item
Objective-C where required for interacting with macOS-specific interfaces.
@end itemize
@section Code formatting conventions
There are the following guidelines regarding the indentation in files:
@@ -135,39 +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
@@ -209,52 +110,92 @@ int myfunc(int my_parameter)
...
@end example
@anchor{Naming conventions}
@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
Variadic macros (@samp{#define ARRAY(nb, ...) (int[nb + 1])@{ nb, __VA_ARGS__ @}});
@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.
@@ -268,50 +209,50 @@ symbols. If in doubt, just avoid names starting with @code{_} altogether.
@section Miscellaneous conventions
@itemize @bullet
@item
fprintf and printf are forbidden in libavformat and libavcodec,
please use av_log() instead.
@item
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 Code behaviour
@subheading Correctness
The code must be valid. It must not crash, abort, access invalid pointers, leak
memory, cause data races or signed integer overflow, or otherwise cause
undefined behaviour. Error codes should be checked and, when applicable,
forwarded to the caller.
@subheading Thread- and library-safety
Our libraries may be called by multiple independent callers in the same process.
These calls may happen from any number of threads and the different call sites
may not be aware of each other - e.g. a user program may be calling our
libraries directly, and use one or more libraries that also call our libraries.
The code must behave correctly under such conditions.
@subheading Robustness
The code must treat as untrusted any bytestream received from a caller or read
from a file, network, etc. It must not misbehave when arbitrary data is sent to
it - typically it should print an error message and return
@code{AVERROR_INVALIDDATA} on encountering invalid input data.
@subheading Memory allocation
The code must use the @code{av_malloc()} family of functions from
@file{libavutil/mem.h} to perform all memory allocation, except in special cases
(e.g. when interacting with an external library that requires a specific
allocator to be used).
All allocations should be checked and @code{AVERROR(ENOMEM)} returned on
failure. A common mistake is that error paths leak memory - make sure that does
not happen.
@subheading stdio
Our libraries must not access the stdio streams stdin/stdout/stderr directly
(e.g. via @code{printf()} family of functions), as that is not library-safe. For
logging, use @code{av_log()}.
@section Patches/Committing
@subheading Licenses for patches must be compatible with FFmpeg.
Contributions should be licensed under the
@@ -334,24 +275,13 @@ missing samples or an implementation with a small subset of features.
Always check the mailing list for any reviewers with issues and test
FATE before you push.
@subheading Commit messages
Commit messages are highly important tools for informing other developers on
what a given change does and why. Every commit must always have a properly
filled out commit message with the following format:
@example
area changed: short 1 line description
details describing what and why and giving references.
@end example
If the commit addresses a known bug on our bug tracker or other external issue
(e.g. CVE), the commit message should include the relevant bug ID(s) or other
external identifiers. Note that this should be done in addition to a proper
explanation and not instead of it. Comments such as "fixed!" or "Changed it."
are not acceptable.
When applying patches that have been discussed at length on the mailing list,
reference the thread in the commit message.
@subheading Keep the main commit message short with an extended description below.
The commit message should have a short first line in the form of
a @samp{topic: short description} as a header, separated by a newline
from the body consisting of an explanation of why the change is necessary.
If the commit fixes a known bug on the bug tracker, the commit message
should include its bug ID. Referring to the issue on the bug tracker does
not exempt you from writing an excerpt of the bug in the commit message.
@subheading Testing must be adequate but not excessive.
If it works for you, others, and passes FATE then it should be OK to commit
@@ -370,6 +300,15 @@ later on.
Also if you have doubts about splitting or not splitting, do not hesitate to
ask/discuss it on the developer mailing list.
@subheading Ask before you change the build system (configure, etc).
Do not commit changes to the build system (Makefiles, configure script)
which change behavior, defaults etc, without asking first. The same
applies to compiler warning fixes, trivial looking fixes and to code
maintained by other developers. We usually have a reason for doing things
the way we do. Send your changes as patches to the ffmpeg-devel mailing
list, and if the code maintainers say OK, you may commit. This does not
apply to files you wrote and/or maintain.
@subheading Cosmetic changes should be kept in separate patches.
We refuse source indentation and other cosmetic changes if they are mixed
with functional changes, such commits will be rejected and removed. Every
@@ -384,15 +323,27 @@ NOTE: If you had to put if()@{ .. @} over a large (> 5 lines) chunk of code,
then either do NOT change the indentation of the inner part within (do not
move it to the right)! or do so in a separate commit
@subheading Commit messages should always be filled out properly.
Always fill out the commit log message. Describe in a few lines what you
changed and why. You can refer to mailing list postings if you fix a
particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.
Recommended format:
@example
area changed: Short 1 line description
details describing what and why and giving references.
@end example
@subheading Credit the author of the patch.
Make sure the author of the commit is set correctly. (see git commit --author)
If you apply a patch, send an
answer to ffmpeg-devel (or wherever you got the patch from) saying that
you applied the patch.
@subheading Credit any researchers
If a commit/patch fixes an issues found by some researcher, always credit the
researcher in the commit message for finding/reporting the issue.
@subheading Complex patches should refer to discussion surrounding them.
When applying patches that have been discussed (at length) on the mailing
list, reference the thread in the log message.
@subheading Always wait long enough before pushing changes
Do NOT commit to code actively maintained by others without permission.
@@ -402,6 +353,22 @@ time-frame (12h for build failures and security fixes, 3 days small changes,
Also note, the maintainer can simply ask for more time to review!
@section Code
@subheading API/ABI changes should be discussed before they are made.
Do not change behavior of the programs (renaming options etc) or public
API or ABI without first discussing it on the ffmpeg-devel mailing list.
Do not remove widely used functionality or features (redundant code can be removed).
@subheading Remember to check if you need to bump versions for libav*.
Depending on the change, you may need to change the version integer.
Incrementing the first component means no backward compatibility to
previous versions (e.g. removal of a function from the public API).
Incrementing the second component means backward compatible change
(e.g. addition of a function to the public API or extension of an
existing data structure).
Incrementing the third component means a noteworthy binary compatible
change (e.g. encoder bug fix that matters for the decoder). The third
component always starts at 100 to distinguish FFmpeg from Libav.
@subheading Warnings for correct code may be disabled if there is no other option.
Compiler warnings indicate potential bugs or code with bad style. If a type of
warning always points to correct and clean code, that warning should
@@ -411,150 +378,10 @@ If it is a bug, the bug has to be fixed. If it is not, the code should
be changed to not generate a warning unless that causes a slowdown
or obfuscates the code.
@section Library public interfaces
Every library in FFmpeg provides a set of public APIs in its installed headers,
which are those listed in the variable @code{HEADERS} in that library's
@file{Makefile}. All identifiers defined in those headers (except for those
explicitly documented otherwise), and corresponding symbols exported from
compiled shared or static libraries are considered public interfaces and must
comply with the API and ABI compatibility rules described in this section.
Public APIs must be backward compatible within a given major version. I.e. any
valid user code that compiles and works with a given library version must still
compile and work with any later version, as long as the major version number is
unchanged. "Valid user code" here means code that is calling our APIs in a
documented and/or intended manner and is not relying on any undefined behavior.
Incrementing the major version may break backward compatibility, but only to the
extent described in @ref{Major version bumps}.
We also guarantee backward ABI compatibility for shared and static libraries.
I.e. it should be possible to replace a shared or static build of our library
with a build of any later version (re-linking the user binary in the static
case) without breaking any valid user binaries, as long as the major version
number remains unchanged.
@subsection Adding new interfaces
Any new public identifiers in installed headers are considered new API - this
includes new functions, structs, macros, enum values, typedefs, new fields in
existing structs, new installed headers, etc. Consider the following
guidelines when adding new APIs.
@subsubheading Motivation
While new APIs can be added relatively easily, changing or removing them is much
harder due to abovementioned compatibility requirements. You should then
consider carefully whether the functionality you are adding really needs to be
exposed to our callers as new public API.
Your new API should have at least one well-established use case outside of the
library that cannot be easily achieved with existing APIs. Every library in
FFmpeg also has a defined scope - your new API must fit within it.
@subsubheading Replacing existing APIs
If your new API is replacing an existing one, it should be strictly superior to
it, so that the advantages of using the new API outweight the cost to the
callers of changing their code. After adding the new API you should then
deprecate the old one and schedule it for removal, as described in
@ref{Removing interfaces}.
If you deem an existing API deficient and want to fix it, the preferred approach
in most cases is to add a differently-named replacement and deprecate the
existing API rather than modify it. It is important to make the changes visible
to our callers (e.g. through compile- or run-time deprecation warnings) and make
it clear how to transition to the new API (e.g. in the Doxygen documentation or
on the wiki).
@subsubheading API design
The FFmpeg libraries are used by a variety of callers to perform a wide range of
multimedia-related processing tasks. You should therefore - within reason - try
to design your new API for the broadest feasible set of use cases and avoid
unnecessarily limiting it to a specific type of callers (e.g. just media
playback or just transcoding).
@subsubheading Consistency
Check whether similar APIs already exist in FFmpeg. If they do, try to model
your new addition on them to achieve better overall consistency.
The naming of your new identifiers should follow the @ref{Naming conventions}
and be aligned with other similar APIs, if applicable.
@subsubheading Extensibility
You should also consider how your API might be extended in the future in a
backward-compatible way. If you are adding a new struct @code{AVFoo}, the
standard approach is requiring the caller to always allocate it through a
constructor function, typically named @code{av_foo_alloc()}. This way new fields
may be added to the end of the struct without breaking ABI compatibility.
Typically you will also want a destructor - @code{av_foo_free(AVFoo**)} that
frees the indirectly supplied object (and its contents, if applicable) and
writes @code{NULL} to the supplied pointer, thus eliminating the potential
dangling pointer in the caller's memory.
If you are adding new functions, consider whether it might be desirable to tweak
their behavior in the future - you may want to add a flags argument, even though
it would be unused initially.
@subsubheading Documentation
All new APIs must be documented as Doxygen-formatted comments above the
identifiers you add to the public headers. You should also briefly mention the
change in @file{doc/APIchanges}.
@subsubheading Bump the version
Backward-incompatible API or ABI changes require incrementing (bumping) the
major version number, as described in @ref{Major version bumps}. Major
bumps are significant events that happen on a schedule - so if your change
strictly requires one you should add it under @code{#if} preprocesor guards that
disable it until the next major bump happens.
New APIs that can be added without breaking API or ABI compatibility require
bumping the minor version number.
Incrementing the third (micro) version component means a noteworthy binary
compatible change (e.g. encoder bug fix that matters for the decoder). The third
component always starts at 100 to distinguish FFmpeg from Libav.
@anchor{Removing interfaces}
@subsection Removing interfaces
Due to abovementioned compatibility guarantees, removing APIs is an involved
process that should only be undertaken with good reason. Typically a deficient,
restrictive, or otherwise inadequate API is replaced by a superior one, though
it does at times happen that we remove an API without any replacement (e.g. when
the feature it provides is deemed not worth the maintenance effort, out of scope
of the project, fundamentally flawed, etc.).
The removal has two steps - first the API is deprecated and scheduled for
removal, but remains present and functional. The second step is actually
removing the API - this is described in @ref{Major version bumps}.
To deprecate an API you should signal to our users that they should stop using
it. E.g. if you intend to remove struct members or functions, you should mark
them with @code{attribute_deprecated}. When this cannot be done, it may be
possible to detect the use of the deprecated API at runtime and print a warning
(though take care not to print it too often). You should also document the
deprecation (and the replacement, if applicable) in the relevant Doxygen
documentation block.
Finally, you should define a deprecation guard along the lines of
@code{#define FF_API_<FOO> (LIBAVBAR_VERSION_MAJOR < XX)} (where XX is the major
version in which the API will be removed) in @file{libavbar/version_major.h}
(@file{version.h} in case of @code{libavutil}). Then wrap all uses of the
deprecated API in @code{#if FF_API_<FOO> .... #endif}, so that the code will
automatically get disabled once the major version reaches XX. You can also use
@code{FF_DISABLE_DEPRECATION_WARNINGS} and @code{FF_ENABLE_DEPRECATION_WARNINGS}
to suppress compiler deprecation warnings inside these guards. You should test
that the code compiles and works with the guard macro evaluating to both true
and false.
@anchor{Major version bumps}
@subsection Major version bumps
A major version bump signifies an API and/or ABI compatibility break. To reduce
the negative effects on our callers, who are required to adapt their code,
backward-incompatible changes during a major bump should be limited to:
@itemize @bullet
@item
Removing previously deprecated APIs.
@item
Performing ABI- but not API-breaking changes, like reordering struct contents.
@end itemize
@subheading Check untrusted input properly.
Never write to unallocated memory, never write over the end of arrays,
always check values read from some untrusted source before using them
as array index or other risky things.
@section Documentation/Other
@subheading Subscribe to the ffmpeg-devel mailing list.
@@ -598,6 +425,35 @@ finding a new maintainer and also don't forget to update the @file{MAINTAINERS}
We think our rules are not too hard. If you have comments, contact us.
@chapter Code of conduct
Be friendly and respectful towards others and third parties.
Treat others the way you yourself want to be treated.
Be considerate. Not everyone shares the same viewpoint and priorities as you do.
Different opinions and interpretations help the project.
Looking at issues from a different perspective assists development.
Do not assume malice for things that can be attributed to incompetence. Even if
it is malice, it's rarely good to start with that as initial assumption.
Stay friendly even if someone acts contrarily. Everyone has a bad day
once in a while.
If you yourself have a bad day or are angry then try to take a break and reply
once you are calm and without anger if you have to.
Try to help other team members and cooperate if you can.
The goal of software development is to create technical excellence, not for any
individual to be better and "win" against the others. Large software projects
are only possible and successful through teamwork.
If someone struggles do not put them down. Give them a helping hand
instead and point them in the right direction.
Finally, keep in mind the immortal words of Bill and Ted,
"Be excellent to each other."
@anchor{Submitting patches}
@chapter Submitting patches
@@ -638,27 +494,6 @@ 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 How to setup git send-email?
Please see @url{https://git-send-email.io/}.
For gmail additionally see @url{https://shallowsky.com/blog/tech/email/gmail-app-passwds.html}.
@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
@@ -819,14 +654,16 @@ Lines with similar content should be aligned vertically when doing so
improves readability.
@item
Consider adding a regression test for your code. All new modules
should be covered by tests. That includes demuxers, muxers, decoders, encoders
filters, bitstream filters, parsers. If its not possible to do that, add
an explanation why to your patchset, its ok to not test if theres a reason.
Consider adding a regression test for your code.
@item
If you added YASM code please check that things still work with --disable-yasm.
@item
Make sure you check the return values of function and return appropriate
error codes. Especially memory allocation functions like @code{av_malloc()}
are notoriously left unchecked, which is a serious problem.
@item
Test your code with valgrind and or Address Sanitizer to ensure it's free
of leaks, out of array accesses, etc.
@@ -876,8 +713,6 @@ accordingly].
@section Adding files to the fate-suite dataset
If you need a sample uploaded send a mail to samples-request.
When there is no muxer or encoder available to generate test media for a
specific test then the media has to be included in the fate-suite.
First please make sure that the sample file is as small as possible to test the

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"`

File diff suppressed because it is too large Load Diff

View File

@@ -22,4 +22,3 @@
/transcoding
/vaapi_encode
/vaapi_transcode
/qsv_transcode

View File

@@ -1,27 +1,26 @@
EXAMPLES-$(CONFIG_AVIO_HTTP_SERVE_FILES) += avio_http_serve_files
EXAMPLES-$(CONFIG_AVIO_LIST_DIR_EXAMPLE) += avio_list_dir
EXAMPLES-$(CONFIG_AVIO_READ_CALLBACK_EXAMPLE) += avio_read_callback
EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE) += avio_reading
EXAMPLES-$(CONFIG_DECODE_AUDIO_EXAMPLE) += decode_audio
EXAMPLES-$(CONFIG_DECODE_FILTER_AUDIO_EXAMPLE) += decode_filter_audio
EXAMPLES-$(CONFIG_DECODE_FILTER_VIDEO_EXAMPLE) += decode_filter_video
EXAMPLES-$(CONFIG_DECODE_VIDEO_EXAMPLE) += decode_video
EXAMPLES-$(CONFIG_DEMUX_DECODE_EXAMPLE) += demux_decode
EXAMPLES-$(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-$(CONFIG_MUX_EXAMPLE) += mux
EXAMPLES-$(CONFIG_QSV_DECODE_EXAMPLE) += qsv_decode
EXAMPLES-$(CONFIG_REMUX_EXAMPLE) += remux
EXAMPLES-$(CONFIG_RESAMPLE_AUDIO_EXAMPLE) += resample_audio
EXAMPLES-$(CONFIG_SCALE_VIDEO_EXAMPLE) += scale_video
EXAMPLES-$(CONFIG_SHOW_METADATA_EXAMPLE) += show_metadata
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_TRANSCODE_EXAMPLE) += transcode
EXAMPLES-$(CONFIG_TRANSCODING_EXAMPLE) += transcoding
EXAMPLES-$(CONFIG_VAAPI_ENCODE_EXAMPLE) += vaapi_encode
EXAMPLES-$(CONFIG_VAAPI_TRANSCODE_EXAMPLE) += vaapi_transcode
EXAMPLES-$(CONFIG_QSV_TRANSCODE_EXAMPLE) += qsv_transcode
EXAMPLES := $(EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)$(EXESUF))
EXAMPLES_G := $(EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)_g$(EXESUF))

View File

@@ -11,40 +11,33 @@ CFLAGS += -Wall -g
CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
# missing the following targets, since they need special options in the FFmpeg build:
# qsv_decode
# qsv_transcode
# vaapi_encode
# vaapi_transcode
EXAMPLES=\
avio_http_serve_files \
avio_list_dir \
avio_read_callback \
EXAMPLES= avio_list_dir \
avio_reading \
decode_audio \
decode_filter_audio \
decode_filter_video \
decode_video \
demux_decode \
demuxing_decoding \
encode_audio \
encode_video \
extract_mvs \
filtering_video \
filtering_audio \
http_multiclient \
hw_decode \
mux \
remux \
resample_audio \
scale_video \
show_metadata \
metadata \
muxing \
remuxing \
resampling_audio \
scaling_video \
transcode_aac \
transcode
transcoding \
OBJS=$(addsuffix .o,$(EXAMPLES))
# the following examples make explicit use of the math library
avcodec: LDLIBS += -lm
encode_audio: LDLIBS += -lm
mux: LDLIBS += -lm
resample_audio: LDLIBS += -lm
muxing: LDLIBS += -lm
resampling_audio: LDLIBS += -lm
.phony: all clean-test clean

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.

View File

@@ -20,13 +20,6 @@
* THE SOFTWARE.
*/
/**
* @file libavformat AVIOContext list directory API usage example
* @example avio_list_dir.c
*
* Show how to list directories through the libavformat AVIOContext API.
*/
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavformat/avio.h>

View File

@@ -21,11 +21,12 @@
*/
/**
* @file libavformat AVIOContext read callback API usage example
* @example avio_read_callback.c
* @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>

View File

@@ -21,11 +21,10 @@
*/
/**
* @file libavcodec audio decoding API usage example
* @example decode_audio.c
* @file
* audio decoding with libavcodec API example
*
* Decode data from an MP2 input file and generate a raw audio file to
* be played with ffplay.
* @example decode_audio.c
*/
#include <stdio.h>
@@ -98,7 +97,7 @@ static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame,
exit(1);
}
for (i = 0; i < frame->nb_samples; i++)
for (ch = 0; ch < dec_ctx->ch_layout.nb_channels; ch++)
for (ch = 0; ch < dec_ctx->channels; ch++)
fwrite(frame->data[ch] + data_size*i, 1, data_size, outfile);
}
}
@@ -216,7 +215,7 @@ int main(int argc, char **argv)
sfmt = av_get_packed_sample_fmt(sfmt);
}
n_channels = c->ch_layout.nb_channels;
n_channels = c->channels;
if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
goto end;

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);
}
@@ -93,7 +92,6 @@ int main(int argc, char **argv)
uint8_t *data;
size_t data_size;
int ret;
int eof;
AVPacket *pkt;
if (argc <= 2) {
@@ -152,16 +150,15 @@ 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 +170,8 @@ 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);

View File

@@ -21,18 +21,17 @@
*/
/**
* @file libavformat and libavcodec demuxing and decoding API usage example
* @example demux_decode.c
* @file
* Demuxing and decoding example.
*
* Show how to use the libavformat and libavcodec API to demux and decode audio
* and video data. Write the output as raw audio and input files to be played by
* ffplay.
* 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 <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
static AVFormatContext *fmt_ctx = NULL;
@@ -52,7 +51,7 @@ static int video_dst_bufsize;
static int video_stream_idx = -1, audio_stream_idx = -1;
static AVFrame *frame = NULL;
static AVPacket *pkt = NULL;
static AVPacket pkt;
static int video_frame_count = 0;
static int audio_frame_count = 0;
@@ -73,14 +72,14 @@ static int output_video_frame(AVFrame *frame)
return -1;
}
printf("video_frame n:%d\n",
video_frame_count++);
printf("video_frame n:%d coded_n:%d\n",
video_frame_count++, frame->coded_picture_number);
/* copy decoded frame to destination buffer:
* this is required since rawvideo expects non aligned data */
av_image_copy2(video_dst_data, video_dst_linesize,
frame->data, frame->linesize,
pix_fmt, width, height);
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);
@@ -138,9 +137,11 @@ static int decode_packet(AVCodecContext *dec, const AVPacket *pkt)
ret = output_audio_frame(frame);
av_frame_unref(frame);
if (ret < 0)
return ret;
}
return ret;
return 0;
}
static int open_codec_context(int *stream_idx,
@@ -148,7 +149,8 @@ static int open_codec_context(int *stream_idx,
{
int ret, stream_index;
AVStream *st;
const AVCodec *dec = NULL;
AVCodec *dec = NULL;
AVDictionary *opts = NULL;
ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
if (ret < 0) {
@@ -183,7 +185,7 @@ static int open_codec_context(int *stream_idx,
}
/* Init the decoders */
if ((ret = avcodec_open2(*dec_ctx, dec, NULL)) < 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;
@@ -301,12 +303,10 @@ int main (int argc, char **argv)
goto end;
}
pkt = av_packet_alloc();
if (!pkt) {
fprintf(stderr, "Could not allocate packet\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);
@@ -314,14 +314,14 @@ int main (int argc, char **argv)
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) {
while (av_read_frame(fmt_ctx, &pkt) >= 0) {
// check if the packet belongs to a stream we are interested in, otherwise
// skip it
if (pkt->stream_index == video_stream_idx)
ret = decode_packet(video_dec_ctx, pkt);
else if (pkt->stream_index == audio_stream_idx)
ret = decode_packet(audio_dec_ctx, pkt);
av_packet_unref(pkt);
if (pkt.stream_index == video_stream_idx)
ret = decode_packet(video_dec_ctx, &pkt);
else if (pkt.stream_index == audio_stream_idx)
ret = decode_packet(audio_dec_ctx, &pkt);
av_packet_unref(&pkt);
if (ret < 0)
break;
}
@@ -343,7 +343,7 @@ int main (int argc, char **argv)
if (audio_stream) {
enum AVSampleFormat sfmt = audio_dec_ctx->sample_fmt;
int n_channels = audio_dec_ctx->ch_layout.nb_channels;
int n_channels = audio_dec_ctx->channels;
const char *fmt;
if (av_sample_fmt_is_planar(sfmt)) {
@@ -372,7 +372,6 @@ end:
fclose(video_dst_file);
if (audio_dst_file)
fclose(audio_dst_file);
av_packet_free(&pkt);
av_frame_free(&frame);
av_free(video_dst_data[0]);

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;
}

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>
@@ -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++) {
@@ -198,12 +185,7 @@ 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.
*/
/* add sequence end code to have a real MPEG file */
if (codec->id == AV_CODEC_ID_MPEG1VIDEO || codec->id == AV_CODEC_ID_MPEG2VIDEO)
fwrite(endcode, 1, sizeof(endcode), f);
fclose(f);

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;
}

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;

View File

@@ -23,11 +23,9 @@
*/
/**
* @file audio decoding and filtering usage example
* @example decode_filter_audio.c
*
* Demux, decode and filter audio input file, generate a raw audio
* file to be played with ffplay.
* @file
* API example for audio decoding and filtering
* @example filtering_audio.c
*/
#include <unistd.h>
@@ -36,7 +34,6 @@
#include <libavformat/avformat.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/channel_layout.h>
#include <libavutil/opt.h>
static const char *filter_descr = "aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono";
@@ -51,8 +48,8 @@ static int audio_stream_index = -1;
static int open_input_file(const char *filename)
{
const AVCodec *dec;
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");
@@ -96,6 +93,7 @@ static int init_filters(const char *filters_descr)
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;
@@ -107,13 +105,12 @@ static int init_filters(const char *filters_descr)
}
/* buffer audio source: the decoded frames from the decoder will be inserted here. */
if (dec_ctx->ch_layout.order == AV_CHANNEL_ORDER_UNSPEC)
av_channel_layout_default(&dec_ctx->ch_layout, dec_ctx->ch_layout.nb_channels);
ret = snprintf(args, sizeof(args),
"time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=",
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));
av_channel_layout_describe(&dec_ctx->ch_layout, args + ret, sizeof(args) - ret);
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) {
@@ -136,7 +133,7 @@ static int init_filters(const char *filters_descr)
goto end;
}
ret = av_opt_set(buffersink_ctx, "ch_layouts", "mono",
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");
@@ -187,7 +184,7 @@ static int init_filters(const char *filters_descr)
/* Print summary of the sink buffer
* Note: args buffer is reused to store channel layout string */
outlink = buffersink_ctx->inputs[0];
av_channel_layout_describe(&outlink->ch_layout, args, sizeof(args));
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), "?"),
@@ -202,7 +199,7 @@ end:
static void print_frame(const AVFrame *frame)
{
const int n = frame->nb_samples * frame->ch_layout.nb_channels;
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;
@@ -217,12 +214,12 @@ static void print_frame(const AVFrame *frame)
int main(int argc, char **argv)
{
int ret;
AVPacket *packet = av_packet_alloc();
AVPacket packet;
AVFrame *frame = av_frame_alloc();
AVFrame *filt_frame = av_frame_alloc();
if (!packet || !frame || !filt_frame) {
fprintf(stderr, "Could not allocate frame or packet\n");
if (!frame || !filt_frame) {
perror("Could not allocate frame");
exit(1);
}
if (argc != 2) {
@@ -237,11 +234,11 @@ int main(int argc, char **argv)
/* read all packets */
while (1) {
if ((ret = av_read_frame(fmt_ctx, packet)) < 0)
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 (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;
@@ -277,13 +274,12 @@ int main(int argc, char **argv)
}
}
}
av_packet_unref(packet);
av_packet_unref(&packet);
}
end:
avfilter_graph_free(&filter_graph);
avcodec_free_context(&dec_ctx);
avformat_close_input(&fmt_ctx);
av_packet_free(&packet);
av_frame_free(&frame);
av_frame_free(&filt_frame);

View File

@@ -24,7 +24,7 @@
/**
* @file
* API example for decoding and filtering
* @example decode_filter_video.c
* @example filtering_video.c
*/
#define _XOPEN_SOURCE 600 /* for usleep */
@@ -53,8 +53,8 @@ static int64_t last_pts = AV_NOPTS_VALUE;
static int open_input_file(const char *filename)
{
const AVCodec *dec;
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");
@@ -210,7 +210,7 @@ static void display_frame(const AVFrame *frame, AVRational time_base)
int main(int argc, char **argv)
{
int ret;
AVPacket *packet;
AVPacket packet;
AVFrame *frame;
AVFrame *filt_frame;
@@ -221,9 +221,8 @@ int main(int argc, char **argv)
frame = av_frame_alloc();
filt_frame = av_frame_alloc();
packet = av_packet_alloc();
if (!frame || !filt_frame || !packet) {
fprintf(stderr, "Could not allocate frame or packet\n");
if (!frame || !filt_frame) {
perror("Could not allocate frame");
exit(1);
}
@@ -234,11 +233,11 @@ int main(int argc, char **argv)
/* read all packets */
while (1) {
if ((ret = av_read_frame(fmt_ctx, packet)) < 0)
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 (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;
@@ -274,7 +273,7 @@ int main(int argc, char **argv)
av_frame_unref(frame);
}
}
av_packet_unref(packet);
av_packet_unref(&packet);
}
end:
avfilter_graph_free(&filter_graph);
@@ -282,7 +281,6 @@ end:
avformat_close_input(&fmt_ctx);
av_frame_free(&frame);
av_frame_free(&filt_frame);
av_packet_free(&packet);
if (ret < 0 && ret != AVERROR_EOF) {
fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));

View File

@@ -21,11 +21,12 @@
*/
/**
* @file libavformat multi-client network API usage example
* @example avio_http_serve_files.c
* @file
* libavformat multi-client network API usage example.
*
* Serve a file without decoding or demuxing it over the HTTP protocol. Multiple
* clients can connect and will receive the same file.
* @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>

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);

View File

@@ -21,10 +21,9 @@
*/
/**
* @file libavformat metadata extraction API usage example
* @example show_metadata.c
*
* Show metadata from an input file.
* @file
* Shows how the metadata API can be used in application programs.
* @example metadata.c
*/
#include <stdio.h>
@@ -35,7 +34,7 @@
int main (int argc, char **argv)
{
AVFormatContext *fmt_ctx = NULL;
const AVDictionaryEntry *tag = NULL;
AVDictionaryEntry *tag = NULL;
int ret;
if (argc != 2) {
@@ -53,7 +52,7 @@ int main (int argc, char **argv)
return ret;
}
while ((tag = av_dict_iterate(fmt_ctx->metadata, tag)))
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);

View File

@@ -21,11 +21,12 @@
*/
/**
* @file libavformat muxing API usage example
* @example mux.c
* @file
* libavformat API example.
*
* Generate a synthetic audio and video signal and mux them to a media file in
* any supported libavformat format. The default codecs are used.
* Output a media file in any supported libavformat format. The default
* codecs are used.
* @example muxing.c
*/
#include <stdlib.h>
@@ -38,7 +39,6 @@
#include <libavutil/opt.h>
#include <libavutil/mathematics.h>
#include <libavutil/timestamp.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libswscale/swscale.h>
#include <libswresample/swresample.h>
@@ -61,8 +61,6 @@ typedef struct OutputStream {
AVFrame *frame;
AVFrame *tmp_frame;
AVPacket *tmp_pkt;
float t, tincr, tincr2;
struct SwsContext *sws_ctx;
@@ -81,7 +79,7 @@ static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt)
}
static int write_frame(AVFormatContext *fmt_ctx, AVCodecContext *c,
AVStream *st, AVFrame *frame, AVPacket *pkt)
AVStream *st, AVFrame *frame)
{
int ret;
@@ -94,7 +92,9 @@ static int write_frame(AVFormatContext *fmt_ctx, AVCodecContext *c,
}
while (ret >= 0) {
ret = avcodec_receive_packet(c, pkt);
AVPacket pkt = { 0 };
ret = avcodec_receive_packet(c, &pkt);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
else if (ret < 0) {
@@ -103,15 +103,13 @@ static int write_frame(AVFormatContext *fmt_ctx, AVCodecContext *c,
}
/* rescale output packet timestamp values from codec to stream timebase */
av_packet_rescale_ts(pkt, c->time_base, st->time_base);
pkt->stream_index = st->index;
av_packet_rescale_ts(&pkt, c->time_base, st->time_base);
pkt.stream_index = st->index;
/* Write the compressed frame to the media file. */
log_packet(fmt_ctx, pkt);
ret = av_interleaved_write_frame(fmt_ctx, pkt);
/* pkt is now blank (av_interleaved_write_frame() takes ownership of
* its contents and resets pkt), so that no unreferencing is necessary.
* This would be different if one used av_write_frame(). */
log_packet(fmt_ctx, &pkt);
ret = av_interleaved_write_frame(fmt_ctx, &pkt);
av_packet_unref(&pkt);
if (ret < 0) {
fprintf(stderr, "Error while writing output packet: %s\n", av_err2str(ret));
exit(1);
@@ -123,7 +121,7 @@ static int write_frame(AVFormatContext *fmt_ctx, AVCodecContext *c,
/* Add an output stream. */
static void add_stream(OutputStream *ost, AVFormatContext *oc,
const AVCodec **codec,
AVCodec **codec,
enum AVCodecID codec_id)
{
AVCodecContext *c;
@@ -137,12 +135,6 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
exit(1);
}
ost->tmp_pkt = av_packet_alloc();
if (!ost->tmp_pkt) {
fprintf(stderr, "Could not allocate AVPacket\n");
exit(1);
}
ost->st = avformat_new_stream(oc, NULL);
if (!ost->st) {
fprintf(stderr, "Could not allocate stream\n");
@@ -169,7 +161,16 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
c->sample_rate = 44100;
}
}
av_channel_layout_copy(&c->ch_layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO);
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;
@@ -199,7 +200,7 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
* the motion of the chroma plane does not match the luma plane. */
c->mb_decision = 2;
}
break;
break;
default:
break;
@@ -214,22 +215,25 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
/* audio output */
static AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt,
const AVChannelLayout *channel_layout,
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;
av_channel_layout_copy(&frame->ch_layout, channel_layout);
frame->channel_layout = channel_layout;
frame->sample_rate = sample_rate;
frame->nb_samples = nb_samples;
if (nb_samples) {
if (av_frame_get_buffer(frame, 0) < 0) {
ret = av_frame_get_buffer(frame, 0);
if (ret < 0) {
fprintf(stderr, "Error allocating an audio buffer\n");
exit(1);
}
@@ -238,8 +242,7 @@ static AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt,
return frame;
}
static void open_audio(AVFormatContext *oc, const AVCodec *codec,
OutputStream *ost, AVDictionary *opt_arg)
static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
{
AVCodecContext *c;
int nb_samples;
@@ -268,9 +271,9 @@ static void open_audio(AVFormatContext *oc, const AVCodec *codec,
else
nb_samples = c->frame_size;
ost->frame = alloc_audio_frame(c->sample_fmt, &c->ch_layout,
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->ch_layout,
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 */
@@ -281,25 +284,25 @@ static void open_audio(AVFormatContext *oc, const AVCodec *codec,
}
/* create resampler context */
ost->swr_ctx = swr_alloc();
if (!ost->swr_ctx) {
fprintf(stderr, "Could not allocate resampler context\n");
exit(1);
}
ost->swr_ctx = swr_alloc();
if (!ost->swr_ctx) {
fprintf(stderr, "Could not allocate resampler context\n");
exit(1);
}
/* set options */
av_opt_set_chlayout (ost->swr_ctx, "in_chlayout", &c->ch_layout, 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_chlayout (ost->swr_ctx, "out_chlayout", &c->ch_layout, 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);
/* 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);
}
/* 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
@@ -317,7 +320,7 @@ static AVFrame *get_audio_frame(OutputStream *ost)
for (j = 0; j <frame->nb_samples; j++) {
v = (int)(sin(ost->t) * 10000);
for (i = 0; i < ost->enc->ch_layout.nb_channels; i++)
for (i = 0; i < ost->enc->channels; i++)
*q++ = v;
ost->t += ost->tincr;
ost->tincr += ost->tincr2;
@@ -346,9 +349,10 @@ static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
if (frame) {
/* convert samples from native format to destination codec format, using the resampler */
/* compute destination number of samples */
dst_nb_samples = swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples;
av_assert0(dst_nb_samples == frame->nb_samples);
/* 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;
@@ -372,37 +376,36 @@ static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
ost->samples_count += dst_nb_samples;
}
return write_frame(oc, c, ost->st, frame, ost->tmp_pkt);
return write_frame(oc, c, ost->st, frame);
}
/**************************************************************/
/* video output */
static AVFrame *alloc_frame(enum AVPixelFormat pix_fmt, int width, int height)
static AVFrame *alloc_picture(enum AVPixelFormat pix_fmt, int width, int height)
{
AVFrame *frame;
AVFrame *picture;
int ret;
frame = av_frame_alloc();
if (!frame)
picture = av_frame_alloc();
if (!picture)
return NULL;
frame->format = pix_fmt;
frame->width = width;
frame->height = height;
picture->format = pix_fmt;
picture->width = width;
picture->height = height;
/* allocate the buffers for the frame data */
ret = av_frame_get_buffer(frame, 0);
ret = av_frame_get_buffer(picture, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate frame data.\n");
exit(1);
}
return frame;
return picture;
}
static void open_video(AVFormatContext *oc, const AVCodec *codec,
OutputStream *ost, AVDictionary *opt_arg)
static void open_video(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
{
int ret;
AVCodecContext *c = ost->enc;
@@ -419,7 +422,7 @@ static void open_video(AVFormatContext *oc, const AVCodec *codec,
}
/* allocate and init a re-usable frame */
ost->frame = alloc_frame(c->pix_fmt, c->width, c->height);
ost->frame = alloc_picture(c->pix_fmt, c->width, c->height);
if (!ost->frame) {
fprintf(stderr, "Could not allocate video frame\n");
exit(1);
@@ -430,9 +433,9 @@ static void open_video(AVFormatContext *oc, const AVCodec *codec,
* output format. */
ost->tmp_frame = NULL;
if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
ost->tmp_frame = alloc_frame(AV_PIX_FMT_YUV420P, c->width, c->height);
ost->tmp_frame = alloc_picture(AV_PIX_FMT_YUV420P, c->width, c->height);
if (!ost->tmp_frame) {
fprintf(stderr, "Could not allocate temporary video frame\n");
fprintf(stderr, "Could not allocate temporary picture\n");
exit(1);
}
}
@@ -515,7 +518,8 @@ static AVFrame *get_video_frame(OutputStream *ost)
*/
static int write_video_frame(AVFormatContext *oc, OutputStream *ost)
{
return write_frame(oc, ost->enc, ost->st, get_video_frame(ost), ost->tmp_pkt);
return write_frame(oc, ost->enc, ost->st, get_video_frame(ost));
}
static void close_stream(AVFormatContext *oc, OutputStream *ost)
@@ -523,7 +527,6 @@ static void close_stream(AVFormatContext *oc, OutputStream *ost)
avcodec_free_context(&ost->enc);
av_frame_free(&ost->frame);
av_frame_free(&ost->tmp_frame);
av_packet_free(&ost->tmp_pkt);
sws_freeContext(ost->sws_ctx);
swr_free(&ost->swr_ctx);
}
@@ -534,10 +537,10 @@ static void close_stream(AVFormatContext *oc, OutputStream *ost)
int main(int argc, char **argv)
{
OutputStream video_st = { 0 }, audio_st = { 0 };
const AVOutputFormat *fmt;
const char *filename;
AVOutputFormat *fmt;
AVFormatContext *oc;
const AVCodec *audio_codec, *video_codec;
AVCodec *audio_codec, *video_codec;
int ret;
int have_video = 0, have_audio = 0;
int encode_video = 0, encode_audio = 0;
@@ -624,6 +627,10 @@ int main(int argc, char **argv)
}
}
/* 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. */

View File

@@ -1,435 +0,0 @@
/*
* 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 video transcoding API usage example
* @example qsv_transcode.c
*
* Perform QSV-accelerated transcoding and show to dynamically change
* encoder's options.
*
* Usage: qsv_transcode input_stream codec output_stream initial option
* { frame_number new_option }
* e.g: - qsv_transcode input.mp4 h264_qsv output_h264.mp4 "g 60"
* - qsv_transcode input.mp4 hevc_qsv output_hevc.mp4 "g 60 async_depth 1"
* 100 "g 120"
* (initialize codec with gop_size 60 and change it to 120 after 100
* frames)
*/
#include <stdio.h>
#include <errno.h>
#include <libavutil/hwcontext.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavutil/opt.h>
static AVFormatContext *ifmt_ctx = NULL, *ofmt_ctx = NULL;
static AVBufferRef *hw_device_ctx = NULL;
static AVCodecContext *decoder_ctx = NULL, *encoder_ctx = NULL;
static int video_stream = -1;
typedef struct DynamicSetting {
int frame_number;
char* optstr;
} DynamicSetting;
static DynamicSetting *dynamic_setting;
static int setting_number;
static int current_setting_number;
static int str_to_dict(char* optstr, AVDictionary **opt)
{
char *key, *value;
if (strlen(optstr) == 0)
return 0;
key = strtok(optstr, " ");
if (key == NULL)
return AVERROR(EINVAL);
value = strtok(NULL, " ");
if (value == NULL)
return AVERROR(EINVAL);
av_dict_set(opt, key, value, 0);
do {
key = strtok(NULL, " ");
if (key == NULL)
return 0;
value = strtok(NULL, " ");
if (value == NULL)
return AVERROR(EINVAL);
av_dict_set(opt, key, value, 0);
} while(1);
}
static int dynamic_set_parameter(AVCodecContext *avctx)
{
AVDictionary *opts = NULL;
int ret = 0;
static int frame_number = 0;
frame_number++;
if (current_setting_number < setting_number &&
frame_number == dynamic_setting[current_setting_number].frame_number) {
AVDictionaryEntry *e = NULL;
ret = str_to_dict(dynamic_setting[current_setting_number++].optstr, &opts);
if (ret < 0) {
fprintf(stderr, "The dynamic parameter is wrong\n");
goto fail;
}
/* Set common option. The dictionary will be freed and replaced
* by a new one containing all options not found in common option list.
* Then this new dictionary is used to set private option. */
if ((ret = av_opt_set_dict(avctx, &opts)) < 0)
goto fail;
/* Set codec specific option */
if ((ret = av_opt_set_dict(avctx->priv_data, &opts)) < 0)
goto fail;
/* There is no "framerate" option in commom option list. Use "-r" to set
* framerate, which is compatible with ffmpeg commandline. The video is
* assumed to be average frame rate, so set time_base to 1/framerate. */
e = av_dict_get(opts, "r", NULL, 0);
if (e) {
avctx->framerate = av_d2q(atof(e->value), INT_MAX);
encoder_ctx->time_base = av_inv_q(encoder_ctx->framerate);
}
}
fail:
av_dict_free(&opts);
return ret;
}
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) {
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 open_input_file(char *filename)
{
int ret;
const AVCodec *decoder = NULL;
AVStream *video = NULL;
if ((ret = avformat_open_input(&ifmt_ctx, filename, NULL, NULL)) < 0) {
fprintf(stderr, "Cannot open input file '%s', Error code: %s\n",
filename, av_err2str(ret));
return ret;
}
if ((ret = avformat_find_stream_info(ifmt_ctx, NULL)) < 0) {
fprintf(stderr, "Cannot find input stream information. Error code: %s\n",
av_err2str(ret));
return ret;
}
ret = av_find_best_stream(ifmt_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, NULL, 0);
if (ret < 0) {
fprintf(stderr, "Cannot find a video stream in the input file. "
"Error code: %s\n", av_err2str(ret));
return ret;
}
video_stream = ret;
video = ifmt_ctx->streams[video_stream];
switch(video->codecpar->codec_id) {
case AV_CODEC_ID_H264:
decoder = avcodec_find_decoder_by_name("h264_qsv");
break;
case AV_CODEC_ID_HEVC:
decoder = avcodec_find_decoder_by_name("hevc_qsv");
break;
case AV_CODEC_ID_VP9:
decoder = avcodec_find_decoder_by_name("vp9_qsv");
break;
case AV_CODEC_ID_VP8:
decoder = avcodec_find_decoder_by_name("vp8_qsv");
break;
case AV_CODEC_ID_AV1:
decoder = avcodec_find_decoder_by_name("av1_qsv");
break;
case AV_CODEC_ID_MPEG2VIDEO:
decoder = avcodec_find_decoder_by_name("mpeg2_qsv");
break;
case AV_CODEC_ID_MJPEG:
decoder = avcodec_find_decoder_by_name("mjpeg_qsv");
break;
default:
fprintf(stderr, "Codec is not supportted by qsv\n");
return AVERROR(EINVAL);
}
if (!(decoder_ctx = avcodec_alloc_context3(decoder)))
return AVERROR(ENOMEM);
if ((ret = avcodec_parameters_to_context(decoder_ctx, video->codecpar)) < 0) {
fprintf(stderr, "avcodec_parameters_to_context error. Error code: %s\n",
av_err2str(ret));
return ret;
}
decoder_ctx->framerate = av_guess_frame_rate(ifmt_ctx, video, NULL);
decoder_ctx->hw_device_ctx = av_buffer_ref(hw_device_ctx);
if (!decoder_ctx->hw_device_ctx) {
fprintf(stderr, "A hardware device reference create failed.\n");
return AVERROR(ENOMEM);
}
decoder_ctx->get_format = get_format;
decoder_ctx->pkt_timebase = video->time_base;
if ((ret = avcodec_open2(decoder_ctx, decoder, NULL)) < 0)
fprintf(stderr, "Failed to open codec for decoding. Error code: %s\n",
av_err2str(ret));
return ret;
}
static int encode_write(AVPacket *enc_pkt, AVFrame *frame)
{
int ret = 0;
av_packet_unref(enc_pkt);
if((ret = dynamic_set_parameter(encoder_ctx)) < 0) {
fprintf(stderr, "Failed to set dynamic parameter. Error code: %s\n",
av_err2str(ret));
goto end;
}
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) {
if (ret = avcodec_receive_packet(encoder_ctx, enc_pkt))
break;
enc_pkt->stream_index = 0;
av_packet_rescale_ts(enc_pkt, encoder_ctx->time_base,
ofmt_ctx->streams[0]->time_base);
if ((ret = av_interleaved_write_frame(ofmt_ctx, enc_pkt)) < 0) {
fprintf(stderr, "Error during writing data to output file. "
"Error code: %s\n", av_err2str(ret));
return ret;
}
}
end:
if (ret == AVERROR_EOF)
return 0;
ret = ((ret == AVERROR(EAGAIN)) ? 0:-1);
return ret;
}
static int dec_enc(AVPacket *pkt, const AVCodec *enc_codec, char *optstr)
{
AVFrame *frame;
int ret = 0;
ret = avcodec_send_packet(decoder_ctx, pkt);
if (ret < 0) {
fprintf(stderr, "Error during decoding. Error code: %s\n", av_err2str(ret));
return ret;
}
while (ret >= 0) {
if (!(frame = av_frame_alloc()))
return AVERROR(ENOMEM);
ret = avcodec_receive_frame(decoder_ctx, frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
av_frame_free(&frame);
return 0;
} else if (ret < 0) {
fprintf(stderr, "Error while decoding. Error code: %s\n", av_err2str(ret));
goto fail;
}
if (!encoder_ctx->hw_frames_ctx) {
AVDictionaryEntry *e = NULL;
AVDictionary *opts = NULL;
AVStream *ost;
/* we need to ref hw_frames_ctx of decoder to initialize encoder's codec.
Only after we get a decoded frame, can we obtain its hw_frames_ctx */
encoder_ctx->hw_frames_ctx = av_buffer_ref(decoder_ctx->hw_frames_ctx);
if (!encoder_ctx->hw_frames_ctx) {
ret = AVERROR(ENOMEM);
goto fail;
}
/* set AVCodecContext Parameters for encoder, here we keep them stay
* the same as decoder.
*/
encoder_ctx->time_base = av_inv_q(decoder_ctx->framerate);
encoder_ctx->pix_fmt = AV_PIX_FMT_QSV;
encoder_ctx->width = decoder_ctx->width;
encoder_ctx->height = decoder_ctx->height;
if ((ret = str_to_dict(optstr, &opts)) < 0) {
fprintf(stderr, "Failed to set encoding parameter.\n");
goto fail;
}
/* There is no "framerate" option in commom option list. Use "-r" to
* set framerate, which is compatible with ffmpeg commandline. The
* video is assumed to be average frame rate, so set time_base to
* 1/framerate. */
e = av_dict_get(opts, "r", NULL, 0);
if (e) {
encoder_ctx->framerate = av_d2q(atof(e->value), INT_MAX);
encoder_ctx->time_base = av_inv_q(encoder_ctx->framerate);
}
if ((ret = avcodec_open2(encoder_ctx, enc_codec, &opts)) < 0) {
fprintf(stderr, "Failed to open encode codec. Error code: %s\n",
av_err2str(ret));
av_dict_free(&opts);
goto fail;
}
av_dict_free(&opts);
if (!(ost = avformat_new_stream(ofmt_ctx, enc_codec))) {
fprintf(stderr, "Failed to allocate stream for output format.\n");
ret = AVERROR(ENOMEM);
goto fail;
}
ost->time_base = encoder_ctx->time_base;
ret = avcodec_parameters_from_context(ost->codecpar, encoder_ctx);
if (ret < 0) {
fprintf(stderr, "Failed to copy the stream parameters. "
"Error code: %s\n", av_err2str(ret));
goto fail;
}
/* write the stream header */
if ((ret = avformat_write_header(ofmt_ctx, NULL)) < 0) {
fprintf(stderr, "Error while writing stream header. "
"Error code: %s\n", av_err2str(ret));
goto fail;
}
}
frame->pts = av_rescale_q(frame->pts, decoder_ctx->pkt_timebase,
encoder_ctx->time_base);
if ((ret = encode_write(pkt, frame)) < 0)
fprintf(stderr, "Error during encoding and writing.\n");
fail:
av_frame_free(&frame);
}
return ret;
}
int main(int argc, char **argv)
{
const AVCodec *enc_codec;
int ret = 0;
AVPacket *dec_pkt = NULL;
if (argc < 5 || (argc - 5) % 2) {
av_log(NULL, AV_LOG_ERROR, "Usage: %s <input file> <encoder> <output file>"
" <\"encoding option set 0\"> [<frame_number> <\"encoding options set 1\">]...\n", argv[0]);
return 1;
}
setting_number = (argc - 5) / 2;
dynamic_setting = av_malloc(setting_number * sizeof(*dynamic_setting));
current_setting_number = 0;
for (int i = 0; i < setting_number; i++) {
dynamic_setting[i].frame_number = atoi(argv[i*2 + 5]);
dynamic_setting[i].optstr = argv[i*2 + 6];
}
ret = av_hwdevice_ctx_create(&hw_device_ctx, AV_HWDEVICE_TYPE_QSV, NULL, NULL, 0);
if (ret < 0) {
fprintf(stderr, "Failed to create a QSV device. Error code: %s\n", av_err2str(ret));
goto end;
}
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;
if (!(enc_codec = avcodec_find_encoder_by_name(argv[2]))) {
fprintf(stderr, "Could not find encoder '%s'\n", argv[2]);
ret = -1;
goto end;
}
if ((ret = (avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, argv[3]))) < 0) {
fprintf(stderr, "Failed to deduce output format from file extension. Error code: "
"%s\n", av_err2str(ret));
goto end;
}
if (!(encoder_ctx = avcodec_alloc_context3(enc_codec))) {
ret = AVERROR(ENOMEM);
goto end;
}
ret = avio_open(&ofmt_ctx->pb, argv[3], AVIO_FLAG_WRITE);
if (ret < 0) {
fprintf(stderr, "Cannot open output file. "
"Error code: %s\n", av_err2str(ret));
goto end;
}
/* read all packets and only transcoding video */
while (ret >= 0) {
if ((ret = av_read_frame(ifmt_ctx, dec_pkt)) < 0)
break;
if (video_stream == dec_pkt->stream_index)
ret = dec_enc(dec_pkt, enc_codec, argv[4]);
av_packet_unref(dec_pkt);
}
/* flush decoder */
av_packet_unref(dec_pkt);
if ((ret = dec_enc(dec_pkt, enc_codec, argv[4])) < 0) {
fprintf(stderr, "Failed to flush decoder %s\n", av_err2str(ret));
goto end;
}
/* flush encoder */
if ((ret = encode_write(dec_pkt, NULL)) < 0) {
fprintf(stderr, "Failed to flush encoder %s\n", av_err2str(ret));
goto end;
}
/* write the trailer for output stream */
if ((ret = av_write_trailer(ofmt_ctx)) < 0)
fprintf(stderr, "Failed to write trailer %s\n", av_err2str(ret));
end:
avformat_close_input(&ifmt_ctx);
avformat_close_input(&ofmt_ctx);
avcodec_free_context(&decoder_ctx);
avcodec_free_context(&encoder_ctx);
av_buffer_unref(&hw_device_ctx);
av_packet_free(&dec_pkt);
av_freep(&dynamic_setting);
return ret;
}

View File

@@ -21,13 +21,16 @@
*/
/**
* @file Intel QSV-accelerated H.264 decoding API usage example
* @example qsv_decode.c
* @file
* Intel QSV-accelerated H.264 decoding example.
*
* Perform QSV-accelerated H.264 decoding with output frames in the
* GPU video surfaces, write the decoded frames to an output file.
* @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"
@@ -41,10 +44,38 @@
#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;
}
@@ -56,7 +87,7 @@ static int get_format(AVCodecContext *avctx, const enum AVPixelFormat *pix_fmts)
return AV_PIX_FMT_NONE;
}
static int decode_packet(AVCodecContext *decoder_ctx,
static int decode_packet(DecodeContext *decode, AVCodecContext *decoder_ctx,
AVFrame *frame, AVFrame *sw_frame,
AVPacket *pkt, AVIOContext *output_ctx)
{
@@ -110,15 +141,15 @@ int main(int argc, char **argv)
AVCodecContext *decoder_ctx = NULL;
const AVCodec *decoder;
AVPacket *pkt = NULL;
AVPacket pkt = { 0 };
AVFrame *frame = NULL, *sw_frame = NULL;
DecodeContext decode = { NULL };
AVIOContext *output_ctx = NULL;
int ret, i;
AVBufferRef *device_ref = NULL;
if (argc < 3) {
fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
return 1;
@@ -146,7 +177,7 @@ int main(int argc, char **argv)
}
/* open the hardware device */
ret = av_hwdevice_ctx_create(&device_ref, AV_HWDEVICE_TYPE_QSV,
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");
@@ -178,8 +209,7 @@ int main(int argc, char **argv)
decoder_ctx->extradata_size = video_st->codecpar->extradata_size;
}
decoder_ctx->hw_device_ctx = av_buffer_ref(device_ref);
decoder_ctx->opaque = &decode;
decoder_ctx->get_format = get_format;
ret = avcodec_open2(decoder_ctx, NULL, NULL);
@@ -197,26 +227,27 @@ int main(int argc, char **argv)
frame = av_frame_alloc();
sw_frame = av_frame_alloc();
pkt = av_packet_alloc();
if (!frame || !sw_frame || !pkt) {
if (!frame || !sw_frame) {
ret = AVERROR(ENOMEM);
goto finish;
}
/* actual decoding */
while (ret >= 0) {
ret = av_read_frame(input_ctx, pkt);
ret = av_read_frame(input_ctx, &pkt);
if (ret < 0)
break;
if (pkt->stream_index == video_st->index)
ret = decode_packet(decoder_ctx, frame, sw_frame, pkt, output_ctx);
if (pkt.stream_index == video_st->index)
ret = decode_packet(&decode, decoder_ctx, frame, sw_frame, &pkt, output_ctx);
av_packet_unref(pkt);
av_packet_unref(&pkt);
}
/* flush the decoder */
ret = decode_packet(decoder_ctx, frame, sw_frame, NULL, output_ctx);
pkt.data = NULL;
pkt.size = 0;
ret = decode_packet(&decode, decoder_ctx, frame, sw_frame, &pkt, output_ctx);
finish:
if (ret < 0) {
@@ -229,11 +260,10 @@ finish:
av_frame_free(&frame);
av_frame_free(&sw_frame);
av_packet_free(&pkt);
avcodec_free_context(&decoder_ctx);
av_buffer_unref(&device_ref);
av_buffer_unref(&decode.hw_device_ref);
avio_close(output_ctx);

View File

@@ -21,11 +21,11 @@
*/
/**
* @file libavformat/libavcodec demuxing and muxing API usage example
* @example remux.c
* @file
* libavformat/libavcodec demuxing and muxing API example.
*
* Remux streams from one container format to another. Data is copied from the
* input to the output without transcoding.
* Remux streams from one container format to another.
* @example remuxing.c
*/
#include <libavutil/timestamp.h>
@@ -45,9 +45,9 @@ static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt, cons
int main(int argc, char **argv)
{
const AVOutputFormat *ofmt = NULL;
AVOutputFormat *ofmt = NULL;
AVFormatContext *ifmt_ctx = NULL, *ofmt_ctx = NULL;
AVPacket *pkt = NULL;
AVPacket pkt;
const char *in_filename, *out_filename;
int ret, i;
int stream_index = 0;
@@ -65,12 +65,6 @@ int main(int argc, char **argv)
in_filename = argv[1];
out_filename = argv[2];
pkt = av_packet_alloc();
if (!pkt) {
fprintf(stderr, "Could not allocate AVPacket\n");
return 1;
}
if ((ret = avformat_open_input(&ifmt_ctx, in_filename, 0, 0)) < 0) {
fprintf(stderr, "Could not open input file '%s'", in_filename);
goto end;
@@ -91,7 +85,7 @@ int main(int argc, char **argv)
}
stream_mapping_size = ifmt_ctx->nb_streams;
stream_mapping = av_calloc(stream_mapping_size, sizeof(*stream_mapping));
stream_mapping = av_mallocz_array(stream_mapping_size, sizeof(*stream_mapping));
if (!stream_mapping) {
ret = AVERROR(ENOMEM);
goto end;
@@ -146,39 +140,38 @@ int main(int argc, char **argv)
while (1) {
AVStream *in_stream, *out_stream;
ret = av_read_frame(ifmt_ctx, pkt);
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);
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");
pkt.stream_index = stream_mapping[pkt.stream_index];
out_stream = ofmt_ctx->streams[pkt.stream_index];
log_packet(ifmt_ctx, &pkt, "in");
/* copy packet */
av_packet_rescale_ts(pkt, in_stream->time_base, out_stream->time_base);
pkt->pos = -1;
log_packet(ofmt_ctx, pkt, "out");
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);
/* pkt is now blank (av_interleaved_write_frame() takes ownership of
* its contents and resets pkt), so that no unreferencing is necessary.
* This would be different if one used av_write_frame(). */
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:
av_packet_free(&pkt);
avformat_close_input(&ifmt_ctx);

View File

@@ -21,12 +21,8 @@
*/
/**
* @file audio resampling API usage example
* @example resample_audio.c
*
* Generate a synthetic audio signal, and Use libswresample API to perform audio
* resampling. The output is written to a raw audio file to be played with
* ffplay.
* @example resampling_audio.c
* libswresample API use example.
*/
#include <libavutil/opt.h>
@@ -84,7 +80,7 @@ static void fill_samples(double *dst, int nb_samples, int nb_channels, int sampl
int main(int argc, char **argv)
{
AVChannelLayout src_ch_layout = AV_CHANNEL_LAYOUT_STEREO, dst_ch_layout = AV_CHANNEL_LAYOUT_SURROUND;
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;
@@ -96,7 +92,6 @@ int main(int argc, char **argv)
int dst_bufsize;
const char *fmt;
struct SwrContext *swr_ctx;
char buf[64];
double t;
int ret;
@@ -125,11 +120,11 @@ int main(int argc, char **argv)
}
/* set options */
av_opt_set_chlayout(swr_ctx, "in_chlayout", &src_ch_layout, 0);
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_chlayout(swr_ctx, "out_chlayout", &dst_ch_layout, 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);
@@ -141,7 +136,7 @@ int main(int argc, char **argv)
/* allocate source and destination samples buffers */
src_nb_channels = src_ch_layout.nb_channels;
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) {
@@ -156,7 +151,7 @@ int main(int argc, char **argv)
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 = dst_ch_layout.nb_channels;
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) {
@@ -199,10 +194,9 @@ int main(int argc, char **argv)
if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0)
goto end;
av_channel_layout_describe(&dst_ch_layout, buf, sizeof(buf));
fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"
"ffplay -f %s -channel_layout %s -channels %d -ar %d %s\n",
fmt, buf, dst_nb_channels, dst_rate, dst_filename);
"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);

View File

@@ -21,10 +21,9 @@
*/
/**
* @file libswscale API usage example
* @example scale_video.c
*
* Generate a synthetic video signal and use libswscale to perform rescaling.
* @file
* libswscale API use example.
* @example scaling_video.c
*/
#include <libavutil/imgutils.h>

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;
}
@@ -447,17 +441,26 @@ static int init_converted_samples(uint8_t ***converted_input_samples,
int error;
/* Allocate as many pointers as there are audio channels.
* Each pointer will point to the audio samples of the corresponding
* Each pointer will later point to the audio samples of the corresponding
* channels (although it may be NULL for interleaved formats).
* Allocate memory for the samples of all channels in one consecutive
*/
if (!(*converted_input_samples = calloc(output_codec_context->channels,
sizeof(**converted_input_samples)))) {
fprintf(stderr, "Could not allocate converted input sample pointers\n");
return AVERROR(ENOMEM);
}
/* Allocate memory for the samples of all channels in one consecutive
* block for convenience. */
if ((error = av_samples_alloc_array_and_samples(converted_input_samples, NULL,
output_codec_context->ch_layout.nb_channels,
if ((error = av_samples_alloc(*converted_input_samples, NULL,
output_codec_context->channels,
frame_size,
output_codec_context->sample_fmt, 0)) < 0) {
fprintf(stderr,
"Could not allocate converted input samples (error '%s')\n",
av_err2str(error));
av_freep(&(*converted_input_samples)[0]);
free(*converted_input_samples);
return error;
}
return 0;
@@ -550,7 +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. */
@@ -589,9 +592,10 @@ static int read_decode_convert_and_store(AVAudioFifo *fifo,
ret = 0;
cleanup:
if (converted_input_samples)
if (converted_input_samples) {
av_freep(&converted_input_samples[0]);
av_freep(&converted_input_samples);
free(converted_input_samples);
}
av_frame_free(&input_frame);
return ret;
@@ -623,7 +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;
@@ -657,12 +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) {
@@ -670,20 +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)) {
@@ -704,14 +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;
}
@@ -850,6 +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;

View File

@@ -23,18 +23,15 @@
*/
/**
* @file demuxing, decoding, filtering, encoding and muxing API usage example
* @example transcode.c
*
* Convert input to output file, applying some hard-coded filter-graph on both
* audio and video streams.
* @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/channel_layout.h>
#include <libavutil/opt.h>
#include <libavutil/pixdesc.h>
@@ -44,17 +41,12 @@ typedef struct FilteringContext {
AVFilterContext *buffersink_ctx;
AVFilterContext *buffersrc_ctx;
AVFilterGraph *filter_graph;
AVPacket *enc_pkt;
AVFrame *filtered_frame;
} FilteringContext;
static FilteringContext *filter_ctx;
typedef struct StreamContext {
AVCodecContext *dec_ctx;
AVCodecContext *enc_ctx;
AVFrame *dec_frame;
} StreamContext;
static StreamContext *stream_ctx;
@@ -74,13 +66,13 @@ static int open_input_file(const char *filename)
return ret;
}
stream_ctx = av_calloc(ifmt_ctx->nb_streams, sizeof(*stream_ctx));
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];
const AVCodec *dec = avcodec_find_decoder(stream->codecpar->codec_id);
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);
@@ -97,11 +89,6 @@ static int open_input_file(const char *filename)
"for stream #%u\n", i);
return ret;
}
/* Inform the decoder about the timebase for the packet timestamps.
* This is highly recommended, but not mandatory. */
codec_ctx->pkt_timebase = stream->time_base;
/* Reencode video & audio and remux subtitles etc. */
if (codec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
|| codec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
@@ -115,10 +102,6 @@ static int open_input_file(const char *filename)
}
}
stream_ctx[i].dec_ctx = codec_ctx;
stream_ctx[i].dec_frame = av_frame_alloc();
if (!stream_ctx[i].dec_frame)
return AVERROR(ENOMEM);
}
av_dump_format(ifmt_ctx, 0, filename, 0);
@@ -130,7 +113,7 @@ static int open_output_file(const char *filename)
AVStream *out_stream;
AVStream *in_stream;
AVCodecContext *dec_ctx, *enc_ctx;
const AVCodec *encoder;
AVCodec *encoder;
int ret;
unsigned int i;
@@ -182,9 +165,8 @@ static int open_output_file(const char *filename)
enc_ctx->time_base = av_inv_q(dec_ctx->framerate);
} else {
enc_ctx->sample_rate = dec_ctx->sample_rate;
ret = av_channel_layout_copy(&enc_ctx->ch_layout, &dec_ctx->ch_layout);
if (ret < 0)
return ret;
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};
@@ -196,7 +178,7 @@ static int open_output_file(const char *filename)
/* 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 %s encoder for stream #%u\n", encoder->name, i);
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);
@@ -271,7 +253,7 @@ static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
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->pkt_timebase.num, dec_ctx->pkt_timebase.den,
dec_ctx->time_base.num, dec_ctx->time_base.den,
dec_ctx->sample_aspect_ratio.num,
dec_ctx->sample_aspect_ratio.den);
@@ -297,7 +279,6 @@ static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
goto end;
}
} else if (dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
char buf[64];
buffersrc = avfilter_get_by_name("abuffer");
buffersink = avfilter_get_by_name("abuffersink");
if (!buffersrc || !buffersink) {
@@ -306,14 +287,14 @@ static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
goto end;
}
if (dec_ctx->ch_layout.order == AV_CHANNEL_ORDER_UNSPEC)
av_channel_layout_default(&dec_ctx->ch_layout, dec_ctx->ch_layout.nb_channels);
av_channel_layout_describe(&dec_ctx->ch_layout, buf, sizeof(buf));
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=%s",
dec_ctx->pkt_timebase.num, dec_ctx->pkt_timebase.den, dec_ctx->sample_rate,
"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),
buf);
dec_ctx->channel_layout);
ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
args, NULL, filter_graph);
if (ret < 0) {
@@ -336,9 +317,9 @@ static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
goto end;
}
av_channel_layout_describe(&enc_ctx->ch_layout, buf, sizeof(buf));
ret = av_opt_set(buffersink_ctx, "ch_layouts",
buf, AV_OPT_SEARCH_CHILDREN);
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;
@@ -417,67 +398,54 @@ static int init_filters(void)
stream_ctx[i].enc_ctx, filter_spec);
if (ret)
return ret;
filter_ctx[i].enc_pkt = av_packet_alloc();
if (!filter_ctx[i].enc_pkt)
return AVERROR(ENOMEM);
filter_ctx[i].filtered_frame = av_frame_alloc();
if (!filter_ctx[i].filtered_frame)
return AVERROR(ENOMEM);
}
return 0;
}
static int encode_write_frame(unsigned int stream_index, int flush)
{
StreamContext *stream = &stream_ctx[stream_index];
FilteringContext *filter = &filter_ctx[stream_index];
AVFrame *filt_frame = flush ? NULL : filter->filtered_frame;
AVPacket *enc_pkt = filter->enc_pkt;
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 */
av_packet_unref(enc_pkt);
if (filt_frame && filt_frame->pts != AV_NOPTS_VALUE)
filt_frame->pts = av_rescale_q(filt_frame->pts, filt_frame->time_base,
stream->enc_ctx->time_base);
ret = avcodec_send_frame(stream->enc_ctx, filt_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;
while (ret >= 0) {
ret = avcodec_receive_packet(stream->enc_ctx, enc_pkt);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
return 0;
/* prepare packet for muxing */
enc_pkt->stream_index = stream_index;
av_packet_rescale_ts(enc_pkt,
stream->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);
}
/* 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)
{
FilteringContext *filter = &filter_ctx[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->buffersrc_ctx,
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");
@@ -486,9 +454,14 @@ static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index)
/* 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->buffersink_ctx,
filter->filtered_frame);
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
@@ -496,13 +469,12 @@ static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index)
*/
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
ret = 0;
av_frame_free(&filt_frame);
break;
}
filter->filtered_frame->time_base = av_buffersink_get_time_base(filter->buffersink_ctx);;
filter->filtered_frame->pict_type = AV_PICTURE_TYPE_NONE;
ret = encode_write_frame(stream_index, 0);
av_frame_unref(filter->filtered_frame);
filt_frame->pict_type = AV_PICTURE_TYPE_NONE;
ret = encode_write_frame(filt_frame, stream_index, NULL);
if (ret < 0)
break;
}
@@ -512,20 +484,34 @@ static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index)
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;
av_log(NULL, AV_LOG_INFO, "Flushing stream #%u encoder\n", stream_index);
return encode_write_frame(stream_index, 1);
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 = NULL;
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]);
@@ -538,85 +524,63 @@ int main(int argc, char **argv)
goto end;
if ((ret = init_filters()) < 0)
goto end;
if (!(packet = av_packet_alloc()))
goto end;
/* read all packets */
while (1) {
if ((ret = av_read_frame(ifmt_ctx, packet)) < 0)
if ((ret = av_read_frame(ifmt_ctx, &packet)) < 0)
break;
stream_index = packet->stream_index;
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) {
StreamContext *stream = &stream_ctx[stream_index];
av_log(NULL, AV_LOG_DEBUG, "Going to reencode&filter the frame\n");
ret = avcodec_send_packet(stream->dec_ctx, packet);
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;
}
while (ret >= 0) {
ret = avcodec_receive_frame(stream->dec_ctx, stream->dec_frame);
if (ret == AVERROR_EOF || ret == AVERROR(EAGAIN))
break;
else if (ret < 0)
goto end;
stream->dec_frame->pts = stream->dec_frame->best_effort_timestamp;
ret = filter_encode_write_frame(stream->dec_frame, stream_index);
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,
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);
ret = av_interleaved_write_frame(ofmt_ctx, &packet);
if (ret < 0)
goto end;
}
av_packet_unref(packet);
av_packet_unref(&packet);
}
/* flush decoders, filters and encoders */
/* flush filters and encoders */
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
StreamContext *stream;
/* flush filter */
if (!filter_ctx[i].filter_graph)
continue;
stream = &stream_ctx[i];
av_log(NULL, AV_LOG_INFO, "Flushing stream %u decoder\n", i);
/* flush decoder */
ret = avcodec_send_packet(stream->dec_ctx, NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Flushing decoding failed\n");
goto end;
}
while (ret >= 0) {
ret = avcodec_receive_frame(stream->dec_ctx, stream->dec_frame);
if (ret == AVERROR_EOF)
break;
else if (ret < 0)
goto end;
stream->dec_frame->pts = stream->dec_frame->best_effort_timestamp;
ret = filter_encode_write_frame(stream->dec_frame, i);
if (ret < 0)
goto end;
}
/* flush filter */
ret = filter_encode_write_frame(NULL, i);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Flushing filter failed\n");
@@ -633,18 +597,14 @@ int main(int argc, char **argv)
av_write_trailer(ofmt_ctx);
end:
av_packet_free(&packet);
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) {
if (filter_ctx && filter_ctx[i].filter_graph)
avfilter_graph_free(&filter_ctx[i].filter_graph);
av_packet_free(&filter_ctx[i].enc_pkt);
av_frame_free(&filter_ctx[i].filtered_frame);
}
av_frame_free(&stream_ctx[i].dec_frame);
}
av_free(filter_ctx);
av_free(stream_ctx);

View File

@@ -1,4 +1,6 @@
/*
* 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
@@ -19,12 +21,13 @@
*/
/**
* @file Intel VAAPI-accelerated encoding API usage example
* @example vaapi_encode.c
* @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
*
* Perform VAAPI-accelerated encoding. Read input from an NV12 raw
* file, and write the H.264 encoded data to an output raw file.
* Usage: vaapi_encode 1920 1080 input.yuv output.h264
*/
#include <stdio.h>
@@ -71,31 +74,27 @@ static int set_hwframe_ctx(AVCodecContext *ctx, AVBufferRef *hw_device_ctx)
static int encode_write(AVCodecContext *avctx, AVFrame *frame, FILE *fout)
{
int ret = 0;
AVPacket *enc_pkt;
AVPacket enc_pkt;
if (!(enc_pkt = av_packet_alloc()))
return AVERROR(ENOMEM);
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);
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);
if (ret != enc_pkt->size) {
ret = AVERROR(errno);
break;
}
enc_pkt.stream_index = 0;
ret = fwrite(enc_pkt.data, enc_pkt.size, 1, fout);
av_packet_unref(&enc_pkt);
}
end:
av_packet_free(&enc_pkt);
ret = ((ret == AVERROR(EAGAIN)) ? 0 : -1);
return ret;
}
@@ -106,7 +105,7 @@ int main(int argc, char *argv[])
FILE *fin = NULL, *fout = NULL;
AVFrame *sw_frame = NULL, *hw_frame = NULL;
AVCodecContext *avctx = NULL;
const AVCodec *codec = NULL;
AVCodec *codec = NULL;
const char *enc_name = "h264_vaapi";
if (argc < 5) {

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;
@@ -210,20 +216,22 @@ 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:
av_frame_free(&frame);
if (ret < 0)
return ret;
}
return ret;
return 0;
}
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"
@@ -238,12 +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;
@@ -273,21 +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);
@@ -298,6 +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;
}

View File

@@ -450,7 +450,7 @@ work with streams that were detected during the initial scan; streams that
are detected later are ignored.
The size of the initial scan is controlled by two options: @code{probesize}
(default ~5@tie{}Mo) and @code{analyzeduration} (default 5,000,000@tie{}µs = 5@tie{}s). For
(default ~5 Mo) and @code{analyzeduration} (default 5,000,000 µs = 5 s). For
the subtitle stream to be detected, both values must be large enough.
@section Why was the @command{ffmpeg} @option{-sameq} option removed? What to use instead?
@@ -467,7 +467,7 @@ point acceptable for your tastes. The most common options to do that are
@option{-qscale} and @option{-qmax}, but you should peruse the documentation
of the encoder you chose.
@section I have a stretched video, why does scaling not fix it?
@section I have a stretched video, why does scaling does not fix it?
A lot of video codecs and formats can store the @emph{aspect ratio} of the
video: this is the ratio between the width and the height of either the full

View File

@@ -79,29 +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
Beware that some assertions are disabled by default, so mind setting
@option{--assert-level=<level>} at configuration time, e.g. when seeking
the highest possible test coverage:
@example
./configure --assert-level=2
@end example
Note that raising the assert level could have a performance impact.
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.
@@ -231,14 +208,6 @@ meaning only while running the regression tests.
Specify how many threads to use while running regression tests, it is
quite useful to detect thread-related regressions.
This variable may be set to the string "random", optionally followed by a
number, like "random99", This will cause each test to use a random number of
threads. If a number is specified, it is used as a maximum number of threads,
otherwise 16 is the maximum.
In case a test fails, the thread count used for it will be written into the
errfile.
@item THREAD_TYPE
Specify which threading strategy test, either @samp{slice} or @samp{frame},
by default @samp{slice+frame}

View File

@@ -31,25 +31,3 @@ makeopts= # extra options passed to 'make'
# defaulting to makeopts above if this is not set
#tar= # command to create a tar archive from its arguments on stdout,
# defaults to 'tar c'
#fate_targets= # targets to make when running fate; defaults to "fate",
# can be set to run a subset of tests, e.g. "fate-checkasm".
#fate_environments= # a list of names of configurations to run tests for;
# each round is run with variables from ${${name}_env} set.
# One example of using fate_environments:
# target_exec="qemu-aarch64-static"
# fate_targets="fate-checkasm fate-cpu"
# fate_environments="sve128 sve256"
# sve128_env="QEMU_CPU=max,sve128=on"
# sve256_env="QEMU_CPU=max,sve256=on"
# The variables set by fate_environments can also be used explicitly
# by target_exec, e.g. like this:
# target_exec="qemu-aarch64-static -cpu \$(MY_CPU)"
# fate_targets="fate-checkasm fate-cpu"
# fate_environments="sve128 sve256"
# sve128_env="MY_CPU=max,sve128=on"
# sve256_env="MY_CPU=max,sve256=on"

File diff suppressed because it is too large Load Diff

View File

@@ -34,6 +34,10 @@ various FFmpeg APIs.
Force displayed width.
@item -y @var{height}
Force displayed height.
@item -s @var{size}
Set frame size (WxH or abbreviation), needed for videos which do
not contain a header with the frame size like raw YUV. This option
has been deprecated in favor of private options, try -video_size.
@item -fs
Start in fullscreen mode.
@item -an
@@ -122,6 +126,10 @@ Read @var{input_url}.
@section Advanced options
@table @option
@item -pix_fmt @var{format}
Set pixel format.
This option has been deprecated in favor of private options, try -pixel_format.
@item -stats
Print several playback statistics, in particular show the stream
duration, the codec parameters, the current position in the stream and
@@ -196,18 +204,6 @@ will produce a thread pool with this many threads available for parallel
processing. The default is 0 which means that the thread count will be
determined by the number of available CPUs.
@item -enable_vulkan
Use vulkan renderer rather than SDL builtin renderer. Depends on libplacebo.
@item -vulkan_params
Vulkan configuration using a list of @var{key}=@var{value} pairs separated by
":".
@item -hwaccel
Use HW accelerated decoding. Enable this option will enable vulkan renderer
automatically.
@end table
@section While playing
@@ -226,6 +222,8 @@ Pause.
Toggle mute.
@item 9, 0
Decrease and increase volume respectively.
@item /, *
Decrease and increase volume respectively.
@@ -297,7 +295,6 @@ Toggle full screen.
@ifset config-avfilter
@include filters.texi
@end ifset
@include general_contents.texi
@end ifset
@chapter See Also

View File

@@ -12,7 +12,7 @@
@chapter Synopsis
ffprobe [@var{options}] @file{input_url}
ffprobe [@var{options}] [@file{input_url}]
@chapter Description
@c man begin DESCRIPTION
@@ -28,9 +28,6 @@ If a url is specified in input, ffprobe will try to open and
probe the url content. If the url cannot be opened or recognized as
a multimedia file, a positive exit code is returned.
If no output is specified as output with @option{o} ffprobe will write
to stdout.
ffprobe may be employed both as a standalone application or in
combination with a textual filter, which may perform more
sophisticated processing, e.g. statistical processing or plotting.
@@ -41,15 +38,15 @@ ffprobe will show it.
ffprobe output is designed to be easily parsable by a textual filter,
and consists of one or more sections of a form defined by the selected
writer, which is specified by the @option{output_format} option.
writer, which is specified by the @option{print_format} option.
Sections may contain other nested sections, and are identified by a
name (which may be shared by other sections), and an unique
name. See the output of @option{sections}.
Metadata tags stored in the container or in the streams are recognized
and printed in the corresponding "FORMAT", "STREAM", "STREAM_GROUP_STREAM"
or "PROGRAM_STREAM" section.
and printed in the corresponding "FORMAT", "STREAM" or "PROGRAM_STREAM"
section.
@c man end
@@ -83,7 +80,7 @@ Use sexagesimal format HH:MM:SS.MICROSECONDS for time values.
Prettify the format of the displayed values, it corresponds to the
options "-unit -prefix -byte_binary_prefix -sexagesimal".
@item -output_format, -of, -print_format @var{writer_name}[=@var{writer_options}]
@item -of, -print_format @var{writer_name}[=@var{writer_options}]
Set the output printing format.
@var{writer_name} specifies the name of the writer, and
@@ -91,7 +88,7 @@ Set the output printing format.
For example for printing the output in JSON format, specify:
@example
-output_format json
-print_format json
@end example
For more details on the available output printing formats, see the
@@ -232,13 +229,6 @@ multimedia stream.
Each media stream information is printed within a dedicated section
with name "PROGRAM_STREAM".
@item -show_stream_groups
Show information about stream groups and their streams contained in the
input multimedia stream.
Each media stream information is printed within a dedicated section
with name "STREAM_GROUP_STREAM".
@item -show_chapters
Show information about chapters stored in the format.
@@ -345,12 +335,6 @@ Show information about all pixel formats supported by FFmpeg.
Pixel format information for each format is printed within a section
with name "PIXEL_FORMAT".
@item -show_optional_fields @var{value}
Some writers viz. JSON and XML, omit the printing of fields with invalid or non-applicable values,
while other writers always print them. This option enables one to control this behaviour.
Valid values are @code{always}/@code{1}, @code{never}/@code{0} and @code{auto}/@code{-1}.
Default is @var{auto}.
@item -bitexact
Force bitexact output, useful to produce output which is not dependent
on the specific build.
@@ -358,10 +342,6 @@ on the specific build.
@item -i @var{input_url}
Read @var{input_url}.
@item -o @var{output_url}
Write output to @var{output_url}. If not specified, the output is sent
to stdout.
@end table
@c man end
@@ -422,9 +402,8 @@ keyN=valN
[/SECTION]
@end example
Metadata tags are printed as a line in the corresponding FORMAT, STREAM,
STREAM_GROUP_STREAM or PROGRAM_STREAM section, and are prefixed by the
string "TAG:".
Metadata tags are printed as a line in the corresponding FORMAT, STREAM or
PROGRAM_STREAM section, and are prefixed by the string "TAG:".
A description of the accepted options follows.
@@ -663,7 +642,6 @@ DV, GXF and AVI timecodes are available in format metadata
@ifset config-avfilter
@include filters.texi
@end ifset
@include general_contents.texi
@end ifset
@chapter See Also

View File

@@ -1,527 +1,394 @@
<?xml version="1.0" encoding="UTF-8"?>
<xsd:schema xmlns:xsd="http://www.w3.org/2001/XMLSchema"
targetNamespace="http://www.ffmpeg.org/schema/ffprobe"
xmlns:ffprobe="http://www.ffmpeg.org/schema/ffprobe">
<xsd:element name="ffprobe" type="ffprobe:ffprobeType"/>
<xsd:complexType name="ffprobeType">
<xsd:sequence>
<xsd:element name="program_version" type="ffprobe:programVersionType" minOccurs="0" maxOccurs="1" />
<xsd:element name="library_versions" type="ffprobe:libraryVersionsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="pixel_formats" type="ffprobe:pixelFormatsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="packets" type="ffprobe:packetsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="frames" type="ffprobe:framesType" minOccurs="0" maxOccurs="1" />
<xsd:element name="packets_and_frames" type="ffprobe:packetsAndFramesType" minOccurs="0" maxOccurs="1" />
<xsd:element name="programs" type="ffprobe:programsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="stream_groups" type="ffprobe:StreamGroupsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="streams" type="ffprobe:streamsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="chapters" type="ffprobe:chaptersType" minOccurs="0" maxOccurs="1" />
<xsd:element name="format" type="ffprobe:formatType" minOccurs="0" maxOccurs="1" />
<xsd:element name="error" type="ffprobe:errorType" minOccurs="0" maxOccurs="1" />
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="packetsType">
<xsd:sequence>
<xsd:element name="packet" type="ffprobe:packetType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="framesType">
<xsd:choice minOccurs="0" maxOccurs="unbounded">
<xsd:element name="frame" type="ffprobe:frameType"/>
<xsd:element name="subtitle" type="ffprobe:subtitleType"/>
</xsd:choice>
</xsd:complexType>
<xsd:complexType name="packetsAndFramesType">
<xsd:choice minOccurs="0" maxOccurs="unbounded">
<xsd:element name="packet" type="ffprobe:packetType"/>
<xsd:element name="frame" type="ffprobe:frameType"/>
<xsd:element name="subtitle" type="ffprobe:subtitleType"/>
</xsd:choice>
</xsd:complexType>
<xsd:complexType name="tagsType">
<xsd:sequence>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="packetType">
<xsd:sequence>
<xsd:element name="tags" type="ffprobe:tagsType" minOccurs="0" maxOccurs="1"/>
<xsd:element name="side_data_list" type="ffprobe:packetSideDataListType" minOccurs="0" maxOccurs="1" />
</xsd:sequence>
<xsd:attribute name="codec_type" type="xsd:string" use="required" />
<xsd:attribute name="stream_index" type="xsd:int" use="required" />
<xsd:attribute name="pts" type="xsd:long" />
<xsd:attribute name="pts_time" type="xsd:float" />
<xsd:attribute name="dts" type="xsd:long" />
<xsd:attribute name="dts_time" type="xsd:float" />
<xsd:attribute name="duration" type="xsd:long" />
<xsd:attribute name="duration_time" type="xsd:float" />
<xsd:attribute name="size" type="xsd:long" use="required" />
<xsd:attribute name="pos" type="xsd:long" />
<xsd:attribute name="flags" type="xsd:string" use="required" />
<xsd:attribute name="data" type="xsd:string" />
<xsd:attribute name="data_hash" type="xsd:string" />
</xsd:complexType>
<xsd:complexType name="packetSideDataListType">
<xsd:sequence>
<xsd:element name="side_data" type="ffprobe:packetSideDataType" minOccurs="1" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="packetSideDataType">
<xsd:sequence>
<xsd:element name="side_datum" type="ffprobe:packetSideDatumType" minOccurs="1" maxOccurs="unbounded"/>
</xsd:sequence>
<xsd:attribute name="type" type="xsd:string"/>
</xsd:complexType>
<xsd:complexType name="packetSideDatumType">
<xsd:attribute name="key" type="xsd:string"/>
<xsd:attribute name="value" type="xsd:string"/>
</xsd:complexType>
<xsd:complexType name="frameType">
<xsd:sequence>
<xsd:element name="tags" type="ffprobe:tagsType" minOccurs="0" maxOccurs="1"/>
<xsd:element name="logs" type="ffprobe:logsType" minOccurs="0" maxOccurs="1"/>
<xsd:element name="side_data_list" type="ffprobe:frameSideDataListType" minOccurs="0" maxOccurs="1" />
</xsd:sequence>
<xsd:attribute name="media_type" type="xsd:string" use="required"/>
<xsd:attribute name="stream_index" type="xsd:int" />
<xsd:attribute name="key_frame" type="xsd:int" use="required"/>
<xsd:attribute name="pts" type="xsd:long" />
<xsd:attribute name="pts_time" type="xsd:float"/>
<xsd:attribute name="pkt_dts" type="xsd:long" />
<xsd:attribute name="pkt_dts_time" type="xsd:float"/>
<xsd:attribute name="best_effort_timestamp" type="xsd:long" />
<xsd:attribute name="best_effort_timestamp_time" type="xsd:float" />
<xsd:attribute name="duration" type="xsd:long" />
<xsd:attribute name="duration_time" type="xsd:float"/>
<xsd:attribute name="pkt_pos" type="xsd:long" />
<xsd:attribute name="pkt_size" type="xsd:int" />
<!-- audio attributes -->
<xsd:attribute name="sample_fmt" type="xsd:string"/>
<xsd:attribute name="nb_samples" type="xsd:long" />
<xsd:attribute name="channels" type="xsd:int" />
<xsd:attribute name="channel_layout" type="xsd:string"/>
<!-- video attributes -->
<xsd:attribute name="width" type="xsd:long" />
<xsd:attribute name="height" type="xsd:long" />
<xsd:attribute name="crop_top" type="xsd:long" />
<xsd:attribute name="crop_bottom" type="xsd:long" />
<xsd:attribute name="crop_left" type="xsd:long" />
<xsd:attribute name="crop_right" type="xsd:long" />
<xsd:attribute name="pix_fmt" type="xsd:string"/>
<xsd:attribute name="sample_aspect_ratio" type="xsd:string"/>
<xsd:attribute name="pict_type" type="xsd:string"/>
<xsd:attribute name="interlaced_frame" type="xsd:int" />
<xsd:attribute name="top_field_first" type="xsd:int" />
<xsd:attribute name="repeat_pict" type="xsd:int" />
<xsd:attribute name="color_range" type="xsd:string"/>
<xsd:attribute name="color_space" type="xsd:string"/>
<xsd:attribute name="color_primaries" type="xsd:string"/>
<xsd:attribute name="color_transfer" type="xsd:string"/>
<xsd:attribute name="chroma_location" type="xsd:string"/>
</xsd:complexType>
<xsd:complexType name="logsType">
<xsd:sequence>
<xsd:element name="log" type="ffprobe:logType" minOccurs="1" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="logType">
<xsd:attribute name="context" type="xsd:string"/>
<xsd:attribute name="level" type="xsd:int" />
<xsd:attribute name="category" type="xsd:int" />
<xsd:attribute name="parent_context" type="xsd:string"/>
<xsd:attribute name="parent_category" type="xsd:int" />
<xsd:attribute name="message" type="xsd:string"/>
</xsd:complexType>
<xsd:complexType name="frameSideDataListType">
<xsd:sequence>
<xsd:element name="side_data" type="ffprobe:frameSideDataType" minOccurs="1" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="frameSideDataType">
<xsd:sequence>
<xsd:element name="timecodes" type="ffprobe:frameSideDataTimecodeList" minOccurs="0" maxOccurs="1"/>
<xsd:element name="components" type="ffprobe:frameSideDataComponentList" minOccurs="0" maxOccurs="1"/>
<xsd:element name="side_datum" type="ffprobe:frameSideDatumType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
<xsd:attribute name="side_data_type" type="xsd:string"/>
<xsd:attribute name="side_data_size" type="xsd:int" />
<xsd:attribute name="timecode" type="xsd:string"/>
</xsd:complexType>
<xsd:complexType name="frameSideDatumType">
<xsd:attribute name="key" type="xsd:string"/>
<xsd:attribute name="value" type="xsd:string"/>
</xsd:complexType>
<xsd:complexType name="frameSideDataTimecodeList">
<xsd:sequence>
<xsd:element name="timecode" type="ffprobe:frameSideDataTimecodeType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="frameSideDataTimecodeType">
<xsd:attribute name="value" type="xsd:string"/>
</xsd:complexType>
<xsd:complexType name="frameSideDataComponentList">
<xsd:sequence>
<xsd:element name="component" type="ffprobe:frameSideDataComponentType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="frameSideDataComponentType">
<xsd:sequence>
<xsd:element name="pieces" type="ffprobe:frameSideDataPieceList" minOccurs="0" maxOccurs="1"/>
<xsd:element name="side_datum" type="ffprobe:frameSideDatumType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="frameSideDataPieceList">
<xsd:sequence>
<xsd:element name="piece" type="ffprobe:frameSideDataPieceType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="frameSideDataPieceType">
<xsd:sequence>
<xsd:element name="side_datum" type="ffprobe:frameSideDatumType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="subtitleType">
<xsd:attribute name="media_type" type="xsd:string" fixed="subtitle" use="required"/>
<xsd:attribute name="pts" type="xsd:long" />
<xsd:attribute name="pts_time" type="xsd:float"/>
<xsd:attribute name="format" type="xsd:int" />
<xsd:attribute name="start_display_time" type="xsd:int" />
<xsd:attribute name="end_display_time" type="xsd:int" />
<xsd:attribute name="num_rects" type="xsd:int" />
</xsd:complexType>
<xsd:complexType name="streamsType">
<xsd:sequence>
<xsd:element name="stream" type="ffprobe:streamType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="programsType">
<xsd:sequence>
<xsd:element name="program" type="ffprobe:programType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="StreamGroupsType">
<xsd:sequence>
<xsd:element name="stream_group" type="ffprobe:streamGroupType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="streamDispositionType">
<xsd:attribute name="default" type="xsd:int" use="required" />
<xsd:attribute name="dub" type="xsd:int" use="required" />
<xsd:attribute name="original" type="xsd:int" use="required" />
<xsd:attribute name="comment" type="xsd:int" use="required" />
<xsd:attribute name="lyrics" type="xsd:int" use="required" />
<xsd:attribute name="karaoke" type="xsd:int" use="required" />
<xsd:attribute name="forced" type="xsd:int" use="required" />
<xsd:attribute name="hearing_impaired" type="xsd:int" use="required" />
<xsd:attribute name="visual_impaired" type="xsd:int" use="required" />
<xsd:attribute name="clean_effects" type="xsd:int" use="required" />
<xsd:attribute name="attached_pic" type="xsd:int" use="required" />
<xsd:attribute name="timed_thumbnails" type="xsd:int" use="required" />
<xsd:attribute name="non_diegetic" type="xsd:int" use="required" />
<xsd:attribute name="captions" type="xsd:int" use="required" />
<xsd:attribute name="descriptions" type="xsd:int" use="required" />
<xsd:attribute name="metadata" type="xsd:int" use="required" />
<xsd:attribute name="dependent" type="xsd:int" use="required" />
<xsd:attribute name="still_image" type="xsd:int" use="required" />
</xsd:complexType>
<xsd:complexType name="streamType">
<xsd:sequence>
<xsd:element name="disposition" type="ffprobe:streamDispositionType" minOccurs="0" maxOccurs="1"/>
<xsd:element name="tags" type="ffprobe:tagsType" minOccurs="0" maxOccurs="1"/>
<xsd:element name="side_data_list" type="ffprobe:packetSideDataListType" minOccurs="0" maxOccurs="1" />
</xsd:sequence>
<xsd:attribute name="index" type="xsd:int" use="required"/>
<xsd:attribute name="codec_name" type="xsd:string" />
<xsd:attribute name="codec_long_name" type="xsd:string" />
<xsd:attribute name="profile" type="xsd:string" />
<xsd:attribute name="codec_type" type="xsd:string" />
<xsd:attribute name="codec_tag" type="xsd:string" use="required"/>
<xsd:attribute name="codec_tag_string" type="xsd:string" use="required"/>
<xsd:attribute name="extradata" type="xsd:string" />
<xsd:attribute name="extradata_size" type="xsd:int" />
<xsd:attribute name="extradata_hash" type="xsd:string" />
<!-- video attributes -->
<xsd:attribute name="width" type="xsd:int"/>
<xsd:attribute name="height" type="xsd:int"/>
<xsd:attribute name="coded_width" type="xsd:int"/>
<xsd:attribute name="coded_height" type="xsd:int"/>
<xsd:attribute name="closed_captions" type="xsd:boolean"/>
<xsd:attribute name="film_grain" type="xsd:boolean"/>
<xsd:attribute name="has_b_frames" type="xsd:int"/>
<xsd:attribute name="sample_aspect_ratio" type="xsd:string"/>
<xsd:attribute name="display_aspect_ratio" type="xsd:string"/>
<xsd:attribute name="pix_fmt" type="xsd:string"/>
<xsd:attribute name="level" type="xsd:int"/>
<xsd:attribute name="color_range" type="xsd:string"/>
<xsd:attribute name="color_space" type="xsd:string"/>
<xsd:attribute name="color_transfer" type="xsd:string"/>
<xsd:attribute name="color_primaries" type="xsd:string"/>
<xsd:attribute name="chroma_location" type="xsd:string"/>
<xsd:attribute name="field_order" type="xsd:string"/>
<xsd:attribute name="refs" type="xsd:int"/>
<!-- audio attributes -->
<xsd:attribute name="sample_fmt" type="xsd:string"/>
<xsd:attribute name="sample_rate" type="xsd:int"/>
<xsd:attribute name="channels" type="xsd:int"/>
<xsd:attribute name="channel_layout" type="xsd:string"/>
<xsd:attribute name="bits_per_sample" type="xsd:int"/>
<xsd:attribute name="initial_padding" type="xsd:int"/>
<xsd:attribute name="id" type="xsd:string"/>
<xsd:attribute name="r_frame_rate" type="xsd:string" use="required"/>
<xsd:attribute name="avg_frame_rate" type="xsd:string" use="required"/>
<xsd:attribute name="time_base" type="xsd:string" use="required"/>
<xsd:attribute name="start_pts" type="xsd:long"/>
<xsd:attribute name="start_time" type="xsd:float"/>
<xsd:attribute name="duration_ts" type="xsd:long"/>
<xsd:attribute name="duration" type="xsd:float"/>
<xsd:attribute name="bit_rate" type="xsd:int"/>
<xsd:attribute name="max_bit_rate" type="xsd:int"/>
<xsd:attribute name="bits_per_raw_sample" type="xsd:int"/>
<xsd:attribute name="nb_frames" type="xsd:int"/>
<xsd:attribute name="nb_read_frames" type="xsd:int"/>
<xsd:attribute name="nb_read_packets" type="xsd:int"/>
</xsd:complexType>
<xsd:complexType name="programType">
<xsd:sequence>
<xsd:element name="tags" type="ffprobe:tagsType" minOccurs="0" maxOccurs="1"/>
<xsd:element name="streams" type="ffprobe:streamsType" minOccurs="0" maxOccurs="1"/>
</xsd:sequence>
<xsd:attribute name="program_id" type="xsd:int" use="required"/>
<xsd:attribute name="program_num" type="xsd:int" use="required"/>
<xsd:attribute name="nb_streams" type="xsd:int" use="required"/>
<xsd:attribute name="pmt_pid" type="xsd:int" use="required"/>
<xsd:attribute name="pcr_pid" type="xsd:int" use="required"/>
</xsd:complexType>
<xsd:complexType name="streamGroupType">
<xsd:sequence>
<xsd:element name="disposition" type="ffprobe:streamDispositionType" minOccurs="0" maxOccurs="1"/>
<xsd:element name="tags" type="ffprobe:tagsType" minOccurs="0" maxOccurs="1"/>
<xsd:element name="streams" type="ffprobe:streamsType" minOccurs="0" maxOccurs="1"/>
<xsd:element name="components" type="ffprobe:streamGroupComponentList" minOccurs="0" maxOccurs="1"/>
</xsd:sequence>
<xsd:attribute name="index" type="xsd:int" use="required"/>
<xsd:attribute name="nb_streams" type="xsd:int" use="required"/>
<xsd:attribute name="type" type="xsd:string" use="required"/>
</xsd:complexType>
<xsd:complexType name="streamGroupComponentList">
<xsd:sequence>
<xsd:element name="component" type="ffprobe:streamGroupComponentType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="streamGroupComponentType">
<xsd:sequence>
<xsd:element name="subcomponents" type="ffprobe:streamGroupSubComponentList" minOccurs="0" maxOccurs="1"/>
<xsd:element name="component_entry" type="ffprobe:streamGroupEntryType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="streamGroupSubComponentList">
<xsd:sequence>
<xsd:element name="subcomponent" type="ffprobe:streamGroupSubComponentType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="streamGroupSubComponentType">
<xsd:sequence>
<xsd:element name="pieces" type="ffprobe:streamGroupPieceList" minOccurs="0" maxOccurs="1"/>
<xsd:element name="subcomponent_entry" type="ffprobe:streamGroupEntryType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="streamGroupPieceList">
<xsd:sequence>
<xsd:element name="piece" type="ffprobe:streamGroupPieceType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="streamGroupPieceType">
<xsd:sequence>
<xsd:element name="subpieces" type="ffprobe:streamGroupSubPieceList" minOccurs="0" maxOccurs="1"/>
<xsd:element name="piece_entry" type="ffprobe:streamGroupEntryType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="streamGroupSubPieceList">
<xsd:sequence>
<xsd:element name="subpiece" type="ffprobe:streamGroupSubPieceType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="streamGroupSubPieceType">
<xsd:sequence>
<xsd:element name="blocks" type="ffprobe:streamGroupBlockList" minOccurs="0" maxOccurs="1"/>
<xsd:element name="subpiece_entry" type="ffprobe:streamGroupEntryType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="streamGroupBlockList">
<xsd:sequence>
<xsd:element name="block" type="ffprobe:streamGroupBlockType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="streamGroupBlockType">
<xsd:sequence>
<xsd:element name="block_entry" type="ffprobe:streamGroupEntryType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="streamGroupEntryType">
<xsd:attribute name="key" type="xsd:string"/>
<xsd:attribute name="value" type="xsd:string"/>
</xsd:complexType>
<xsd:complexType name="formatType">
<xsd:sequence>
<xsd:element name="tags" type="ffprobe:tagsType" minOccurs="0" maxOccurs="1"/>
</xsd:sequence>
<xsd:attribute name="filename" type="xsd:string" use="required"/>
<xsd:attribute name="nb_streams" type="xsd:int" use="required"/>
<xsd:attribute name="nb_programs" type="xsd:int" use="required"/>
<xsd:attribute name="nb_stream_groups" type="xsd:int" use="required"/>
<xsd:attribute name="format_name" type="xsd:string" use="required"/>
<xsd:attribute name="format_long_name" type="xsd:string"/>
<xsd:attribute name="start_time" type="xsd:float"/>
<xsd:attribute name="duration" type="xsd:float"/>
<xsd:attribute name="size" type="xsd:long"/>
<xsd:attribute name="bit_rate" type="xsd:long"/>
<xsd:attribute name="probe_score" type="xsd:int"/>
</xsd:complexType>
<xsd:complexType name="tagType">
<xsd:attribute name="key" type="xsd:string" use="required"/>
<xsd:attribute name="value" type="xsd:string" use="required"/>
</xsd:complexType>
<xsd:complexType name="errorType">
<xsd:attribute name="code" type="xsd:int" use="required"/>
<xsd:attribute name="string" type="xsd:string" use="required"/>
</xsd:complexType>
<xsd:complexType name="programVersionType">
<xsd:attribute name="version" type="xsd:string" use="required"/>
<xsd:attribute name="copyright" type="xsd:string" use="required"/>
<xsd:attribute name="build_date" type="xsd:string"/>
<xsd:attribute name="build_time" type="xsd:string"/>
<xsd:attribute name="compiler_ident" type="xsd:string" use="required"/>
<xsd:attribute name="configuration" type="xsd:string" use="required"/>
</xsd:complexType>
<xsd:complexType name="chaptersType">
<xsd:sequence>
<xsd:element name="chapter" type="ffprobe:chapterType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="chapterType">
<xsd:sequence>
<xsd:element name="tags" type="ffprobe:tagsType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
<xsd:attribute name="id" type="xsd:int" use="required"/>
<xsd:attribute name="time_base" type="xsd:string" use="required"/>
<xsd:attribute name="start" type="xsd:int" use="required"/>
<xsd:attribute name="start_time" type="xsd:float"/>
<xsd:attribute name="end" type="xsd:int" use="required"/>
<xsd:attribute name="end_time" type="xsd:float" use="required"/>
</xsd:complexType>
<xsd:complexType name="libraryVersionType">
<xsd:attribute name="name" type="xsd:string" use="required"/>
<xsd:attribute name="major" type="xsd:int" use="required"/>
<xsd:attribute name="minor" type="xsd:int" use="required"/>
<xsd:attribute name="micro" type="xsd:int" use="required"/>
<xsd:attribute name="version" type="xsd:int" use="required"/>
<xsd:attribute name="ident" type="xsd:string" use="required"/>
</xsd:complexType>
<xsd:complexType name="libraryVersionsType">
<xsd:sequence>
<xsd:element name="library_version" type="ffprobe:libraryVersionType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="pixelFormatFlagsType">
<xsd:attribute name="big_endian" type="xsd:int" use="required"/>
<xsd:attribute name="palette" type="xsd:int" use="required"/>
<xsd:attribute name="bitstream" type="xsd:int" use="required"/>
<xsd:attribute name="hwaccel" type="xsd:int" use="required"/>
<xsd:attribute name="planar" type="xsd:int" use="required"/>
<xsd:attribute name="rgb" type="xsd:int" use="required"/>
<xsd:attribute name="alpha" type="xsd:int" use="required"/>
</xsd:complexType>
<xsd:complexType name="pixelFormatComponentType">
<xsd:attribute name="index" type="xsd:int" use="required"/>
<xsd:attribute name="bit_depth" type="xsd:int" use="required"/>
</xsd:complexType>
<xsd:complexType name="pixelFormatComponentsType">
<xsd:sequence>
<xsd:element name="component" type="ffprobe:pixelFormatComponentType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="pixelFormatType">
<xsd:sequence>
<xsd:element name="flags" type="ffprobe:pixelFormatFlagsType" minOccurs="0" maxOccurs="1"/>
<xsd:element name="components" type="ffprobe:pixelFormatComponentsType" minOccurs="0" maxOccurs="1"/>
</xsd:sequence>
<xsd:attribute name="name" type="xsd:string" use="required"/>
<xsd:attribute name="nb_components" type="xsd:int" use="required"/>
<xsd:attribute name="log2_chroma_w" type="xsd:int"/>
<xsd:attribute name="log2_chroma_h" type="xsd:int"/>
<xsd:attribute name="bits_per_pixel" type="xsd:int"/>
</xsd:complexType>
<xsd:complexType name="pixelFormatsType">
<xsd:sequence>
<xsd:element name="pixel_format" type="ffprobe:pixelFormatType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
targetNamespace="http://www.ffmpeg.org/schema/ffprobe"
xmlns:ffprobe="http://www.ffmpeg.org/schema/ffprobe">
<xsd:element name="ffprobe" type="ffprobe:ffprobeType"/>
<xsd:complexType name="ffprobeType">
<xsd:sequence>
<xsd:element name="program_version" type="ffprobe:programVersionType" minOccurs="0" maxOccurs="1" />
<xsd:element name="library_versions" type="ffprobe:libraryVersionsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="pixel_formats" type="ffprobe:pixelFormatsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="packets" type="ffprobe:packetsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="frames" type="ffprobe:framesType" minOccurs="0" maxOccurs="1" />
<xsd:element name="packets_and_frames" type="ffprobe:packetsAndFramesType" minOccurs="0" maxOccurs="1" />
<xsd:element name="programs" type="ffprobe:programsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="streams" type="ffprobe:streamsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="chapters" type="ffprobe:chaptersType" minOccurs="0" maxOccurs="1" />
<xsd:element name="format" type="ffprobe:formatType" minOccurs="0" maxOccurs="1" />
<xsd:element name="error" type="ffprobe:errorType" minOccurs="0" maxOccurs="1" />
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="packetsType">
<xsd:sequence>
<xsd:element name="packet" type="ffprobe:packetType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="framesType">
<xsd:sequence>
<xsd:choice minOccurs="0" maxOccurs="unbounded">
<xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="subtitle" type="ffprobe:subtitleType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:choice>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="packetsAndFramesType">
<xsd:sequence>
<xsd:choice minOccurs="0" maxOccurs="unbounded">
<xsd:element name="packet" type="ffprobe:packetType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="subtitle" type="ffprobe:subtitleType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:choice>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="packetType">
<xsd:sequence>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="side_data_list" type="ffprobe:packetSideDataListType" minOccurs="0" maxOccurs="1" />
</xsd:sequence>
<xsd:attribute name="codec_type" type="xsd:string" use="required" />
<xsd:attribute name="stream_index" type="xsd:int" use="required" />
<xsd:attribute name="pts" type="xsd:long" />
<xsd:attribute name="pts_time" type="xsd:float" />
<xsd:attribute name="dts" type="xsd:long" />
<xsd:attribute name="dts_time" type="xsd:float" />
<xsd:attribute name="duration" type="xsd:long" />
<xsd:attribute name="duration_time" type="xsd:float" />
<xsd:attribute name="convergence_duration" type="xsd:long" />
<xsd:attribute name="convergence_duration_time" type="xsd:float" />
<xsd:attribute name="size" type="xsd:long" use="required" />
<xsd:attribute name="pos" type="xsd:long" />
<xsd:attribute name="flags" type="xsd:string" use="required" />
<xsd:attribute name="data" type="xsd:string" />
<xsd:attribute name="data_hash" type="xsd:string" />
</xsd:complexType>
<xsd:complexType name="packetSideDataListType">
<xsd:sequence>
<xsd:element name="side_data" type="ffprobe:packetSideDataType" minOccurs="1" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="packetSideDataType">
<xsd:attribute name="side_data_type" type="xsd:string"/>
<xsd:attribute name="side_data_size" type="xsd:int" />
</xsd:complexType>
<xsd:complexType name="frameType">
<xsd:sequence>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="logs" type="ffprobe:logsType" minOccurs="0" maxOccurs="1"/>
<xsd:element name="side_data_list" type="ffprobe:frameSideDataListType" minOccurs="0" maxOccurs="1" />
</xsd:sequence>
<xsd:attribute name="media_type" type="xsd:string" use="required"/>
<xsd:attribute name="stream_index" type="xsd:int" />
<xsd:attribute name="key_frame" type="xsd:int" use="required"/>
<xsd:attribute name="pts" type="xsd:long" />
<xsd:attribute name="pts_time" type="xsd:float"/>
<xsd:attribute name="pkt_pts" type="xsd:long" />
<xsd:attribute name="pkt_pts_time" type="xsd:float"/>
<xsd:attribute name="pkt_dts" type="xsd:long" />
<xsd:attribute name="pkt_dts_time" type="xsd:float"/>
<xsd:attribute name="best_effort_timestamp" type="xsd:long" />
<xsd:attribute name="best_effort_timestamp_time" type="xsd:float" />
<xsd:attribute name="pkt_duration" type="xsd:long" />
<xsd:attribute name="pkt_duration_time" type="xsd:float"/>
<xsd:attribute name="pkt_pos" type="xsd:long" />
<xsd:attribute name="pkt_size" type="xsd:int" />
<!-- audio attributes -->
<xsd:attribute name="sample_fmt" type="xsd:string"/>
<xsd:attribute name="nb_samples" type="xsd:long" />
<xsd:attribute name="channels" type="xsd:int" />
<xsd:attribute name="channel_layout" type="xsd:string"/>
<!-- video attributes -->
<xsd:attribute name="width" type="xsd:long" />
<xsd:attribute name="height" type="xsd:long" />
<xsd:attribute name="pix_fmt" type="xsd:string"/>
<xsd:attribute name="sample_aspect_ratio" type="xsd:string"/>
<xsd:attribute name="pict_type" type="xsd:string"/>
<xsd:attribute name="coded_picture_number" type="xsd:long" />
<xsd:attribute name="display_picture_number" type="xsd:long" />
<xsd:attribute name="interlaced_frame" type="xsd:int" />
<xsd:attribute name="top_field_first" type="xsd:int" />
<xsd:attribute name="repeat_pict" type="xsd:int" />
<xsd:attribute name="color_range" type="xsd:string"/>
<xsd:attribute name="color_space" type="xsd:string"/>
<xsd:attribute name="color_primaries" type="xsd:string"/>
<xsd:attribute name="color_transfer" type="xsd:string"/>
<xsd:attribute name="chroma_location" type="xsd:string"/>
</xsd:complexType>
<xsd:complexType name="logsType">
<xsd:sequence>
<xsd:element name="log" type="ffprobe:logType" minOccurs="1" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="logType">
<xsd:attribute name="context" type="xsd:string"/>
<xsd:attribute name="level" type="xsd:int" />
<xsd:attribute name="category" type="xsd:int" />
<xsd:attribute name="parent_context" type="xsd:string"/>
<xsd:attribute name="parent_category" type="xsd:int" />
<xsd:attribute name="message" type="xsd:string"/>
</xsd:complexType>
<xsd:complexType name="frameSideDataListType">
<xsd:sequence>
<xsd:element name="side_data" type="ffprobe:frameSideDataType" minOccurs="1" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="frameSideDataType">
<xsd:sequence>
<xsd:element name="timecodes" type="ffprobe:frameSideDataTimecodeList" minOccurs="0" maxOccurs="1"/>
</xsd:sequence>
<xsd:attribute name="side_data_type" type="xsd:string"/>
<xsd:attribute name="side_data_size" type="xsd:int" />
<xsd:attribute name="timecode" type="xsd:string"/>
</xsd:complexType>
<xsd:complexType name="frameSideDataTimecodeList">
<xsd:sequence>
<xsd:element name="timecode" type="ffprobe:frameSideDataTimecodeType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="frameSideDataTimecodeType">
<xsd:attribute name="value" type="xsd:string"/>
</xsd:complexType>
<xsd:complexType name="subtitleType">
<xsd:attribute name="media_type" type="xsd:string" fixed="subtitle" use="required"/>
<xsd:attribute name="pts" type="xsd:long" />
<xsd:attribute name="pts_time" type="xsd:float"/>
<xsd:attribute name="format" type="xsd:int" />
<xsd:attribute name="start_display_time" type="xsd:int" />
<xsd:attribute name="end_display_time" type="xsd:int" />
<xsd:attribute name="num_rects" type="xsd:int" />
</xsd:complexType>
<xsd:complexType name="streamsType">
<xsd:sequence>
<xsd:element name="stream" type="ffprobe:streamType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="programsType">
<xsd:sequence>
<xsd:element name="program" type="ffprobe:programType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="streamDispositionType">
<xsd:attribute name="default" type="xsd:int" use="required" />
<xsd:attribute name="dub" type="xsd:int" use="required" />
<xsd:attribute name="original" type="xsd:int" use="required" />
<xsd:attribute name="comment" type="xsd:int" use="required" />
<xsd:attribute name="lyrics" type="xsd:int" use="required" />
<xsd:attribute name="karaoke" type="xsd:int" use="required" />
<xsd:attribute name="forced" type="xsd:int" use="required" />
<xsd:attribute name="hearing_impaired" type="xsd:int" use="required" />
<xsd:attribute name="visual_impaired" type="xsd:int" use="required" />
<xsd:attribute name="clean_effects" type="xsd:int" use="required" />
<xsd:attribute name="attached_pic" type="xsd:int" use="required" />
<xsd:attribute name="timed_thumbnails" type="xsd:int" use="required" />
</xsd:complexType>
<xsd:complexType name="streamType">
<xsd:sequence>
<xsd:element name="disposition" type="ffprobe:streamDispositionType" minOccurs="0" maxOccurs="1"/>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="side_data_list" type="ffprobe:packetSideDataListType" minOccurs="0" maxOccurs="1" />
</xsd:sequence>
<xsd:attribute name="index" type="xsd:int" use="required"/>
<xsd:attribute name="codec_name" type="xsd:string" />
<xsd:attribute name="codec_long_name" type="xsd:string" />
<xsd:attribute name="profile" type="xsd:string" />
<xsd:attribute name="codec_type" type="xsd:string" />
<xsd:attribute name="codec_time_base" type="xsd:string" use="required"/>
<xsd:attribute name="codec_tag" type="xsd:string" use="required"/>
<xsd:attribute name="codec_tag_string" type="xsd:string" use="required"/>
<xsd:attribute name="extradata" type="xsd:string" />
<xsd:attribute name="extradata_hash" type="xsd:string" />
<!-- video attributes -->
<xsd:attribute name="width" type="xsd:int"/>
<xsd:attribute name="height" type="xsd:int"/>
<xsd:attribute name="coded_width" type="xsd:int"/>
<xsd:attribute name="coded_height" type="xsd:int"/>
<xsd:attribute name="closed_captions" type="xsd:boolean"/>
<xsd:attribute name="has_b_frames" type="xsd:int"/>
<xsd:attribute name="sample_aspect_ratio" type="xsd:string"/>
<xsd:attribute name="display_aspect_ratio" type="xsd:string"/>
<xsd:attribute name="pix_fmt" type="xsd:string"/>
<xsd:attribute name="level" type="xsd:int"/>
<xsd:attribute name="color_range" type="xsd:string"/>
<xsd:attribute name="color_space" type="xsd:string"/>
<xsd:attribute name="color_transfer" type="xsd:string"/>
<xsd:attribute name="color_primaries" type="xsd:string"/>
<xsd:attribute name="chroma_location" type="xsd:string"/>
<xsd:attribute name="field_order" type="xsd:string"/>
<xsd:attribute name="timecode" type="xsd:string"/>
<xsd:attribute name="refs" type="xsd:int"/>
<!-- audio attributes -->
<xsd:attribute name="sample_fmt" type="xsd:string"/>
<xsd:attribute name="sample_rate" type="xsd:int"/>
<xsd:attribute name="channels" type="xsd:int"/>
<xsd:attribute name="channel_layout" type="xsd:string"/>
<xsd:attribute name="bits_per_sample" type="xsd:int"/>
<xsd:attribute name="id" type="xsd:string"/>
<xsd:attribute name="r_frame_rate" type="xsd:string" use="required"/>
<xsd:attribute name="avg_frame_rate" type="xsd:string" use="required"/>
<xsd:attribute name="time_base" type="xsd:string" use="required"/>
<xsd:attribute name="start_pts" type="xsd:long"/>
<xsd:attribute name="start_time" type="xsd:float"/>
<xsd:attribute name="duration_ts" type="xsd:long"/>
<xsd:attribute name="duration" type="xsd:float"/>
<xsd:attribute name="bit_rate" type="xsd:int"/>
<xsd:attribute name="max_bit_rate" type="xsd:int"/>
<xsd:attribute name="bits_per_raw_sample" type="xsd:int"/>
<xsd:attribute name="nb_frames" type="xsd:int"/>
<xsd:attribute name="nb_read_frames" type="xsd:int"/>
<xsd:attribute name="nb_read_packets" type="xsd:int"/>
</xsd:complexType>
<xsd:complexType name="programType">
<xsd:sequence>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="streams" type="ffprobe:streamsType" minOccurs="0" maxOccurs="1"/>
</xsd:sequence>
<xsd:attribute name="program_id" type="xsd:int" use="required"/>
<xsd:attribute name="program_num" type="xsd:int" use="required"/>
<xsd:attribute name="nb_streams" type="xsd:int" use="required"/>
<xsd:attribute name="start_time" type="xsd:float"/>
<xsd:attribute name="start_pts" type="xsd:long"/>
<xsd:attribute name="end_time" type="xsd:float"/>
<xsd:attribute name="end_pts" type="xsd:long"/>
<xsd:attribute name="pmt_pid" type="xsd:int" use="required"/>
<xsd:attribute name="pcr_pid" type="xsd:int" use="required"/>
</xsd:complexType>
<xsd:complexType name="formatType">
<xsd:sequence>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
<xsd:attribute name="filename" type="xsd:string" use="required"/>
<xsd:attribute name="nb_streams" type="xsd:int" use="required"/>
<xsd:attribute name="nb_programs" type="xsd:int" use="required"/>
<xsd:attribute name="format_name" type="xsd:string" use="required"/>
<xsd:attribute name="format_long_name" type="xsd:string"/>
<xsd:attribute name="start_time" type="xsd:float"/>
<xsd:attribute name="duration" type="xsd:float"/>
<xsd:attribute name="size" type="xsd:long"/>
<xsd:attribute name="bit_rate" type="xsd:long"/>
<xsd:attribute name="probe_score" type="xsd:int"/>
</xsd:complexType>
<xsd:complexType name="tagType">
<xsd:attribute name="key" type="xsd:string" use="required"/>
<xsd:attribute name="value" type="xsd:string" use="required"/>
</xsd:complexType>
<xsd:complexType name="errorType">
<xsd:attribute name="code" type="xsd:int" use="required"/>
<xsd:attribute name="string" type="xsd:string" use="required"/>
</xsd:complexType>
<xsd:complexType name="programVersionType">
<xsd:attribute name="version" type="xsd:string" use="required"/>
<xsd:attribute name="copyright" type="xsd:string" use="required"/>
<xsd:attribute name="build_date" type="xsd:string"/>
<xsd:attribute name="build_time" type="xsd:string"/>
<xsd:attribute name="compiler_ident" type="xsd:string" use="required"/>
<xsd:attribute name="configuration" type="xsd:string" use="required"/>
</xsd:complexType>
<xsd:complexType name="chaptersType">
<xsd:sequence>
<xsd:element name="chapter" type="ffprobe:chapterType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="chapterType">
<xsd:sequence>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
<xsd:attribute name="id" type="xsd:int" use="required"/>
<xsd:attribute name="time_base" type="xsd:string" use="required"/>
<xsd:attribute name="start" type="xsd:int" use="required"/>
<xsd:attribute name="start_time" type="xsd:float"/>
<xsd:attribute name="end" type="xsd:int" use="required"/>
<xsd:attribute name="end_time" type="xsd:float" use="required"/>
</xsd:complexType>
<xsd:complexType name="libraryVersionType">
<xsd:attribute name="name" type="xsd:string" use="required"/>
<xsd:attribute name="major" type="xsd:int" use="required"/>
<xsd:attribute name="minor" type="xsd:int" use="required"/>
<xsd:attribute name="micro" type="xsd:int" use="required"/>
<xsd:attribute name="version" type="xsd:int" use="required"/>
<xsd:attribute name="ident" type="xsd:string" use="required"/>
</xsd:complexType>
<xsd:complexType name="libraryVersionsType">
<xsd:sequence>
<xsd:element name="library_version" type="ffprobe:libraryVersionType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="pixelFormatFlagsType">
<xsd:attribute name="big_endian" type="xsd:int" use="required"/>
<xsd:attribute name="palette" type="xsd:int" use="required"/>
<xsd:attribute name="bitstream" type="xsd:int" use="required"/>
<xsd:attribute name="hwaccel" type="xsd:int" use="required"/>
<xsd:attribute name="planar" type="xsd:int" use="required"/>
<xsd:attribute name="rgb" type="xsd:int" use="required"/>
<xsd:attribute name="pseudopal" type="xsd:int" use="required"/>
<xsd:attribute name="alpha" type="xsd:int" use="required"/>
</xsd:complexType>
<xsd:complexType name="pixelFormatComponentType">
<xsd:attribute name="index" type="xsd:int" use="required"/>
<xsd:attribute name="bit_depth" type="xsd:int" use="required"/>
</xsd:complexType>
<xsd:complexType name="pixelFormatComponentsType">
<xsd:sequence>
<xsd:element name="component" type="ffprobe:pixelFormatComponentType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="pixelFormatType">
<xsd:sequence>
<xsd:element name="flags" type="ffprobe:pixelFormatFlagsType" minOccurs="0" maxOccurs="1"/>
<xsd:element name="components" type="ffprobe:pixelFormatComponentsType" minOccurs="0" maxOccurs="1"/>
</xsd:sequence>
<xsd:attribute name="name" type="xsd:string" use="required"/>
<xsd:attribute name="nb_components" type="xsd:int" use="required"/>
<xsd:attribute name="log2_chroma_w" type="xsd:int"/>
<xsd:attribute name="log2_chroma_h" type="xsd:int"/>
<xsd:attribute name="bits_per_pixel" type="xsd:int"/>
</xsd:complexType>
<xsd:complexType name="pixelFormatsType">
<xsd:sequence>
<xsd:element name="pixel_format" type="ffprobe:pixelFormatType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
</xsd:schema>

View File

@@ -13,15 +13,6 @@ corresponding value to true. They can be set to false by prefixing
the option name with "no". For example using "-nofoo"
will set the boolean option with name "foo" to false.
Options that take arguments support a special syntax where the argument given on
the command line is interpreted as a path to the file from which the actual
argument value is loaded. To use this feature, add a forward slash '/'
immediately before the option name (after the leading dash). E.g.
@example
ffmpeg -i INPUT -/filter:v filter.script OUTPUT
@end example
will load a filtergraph description from the file named @file{filter.script}.
@anchor{Stream specifiers}
@section Stream specifiers
Some options are applied per-stream, e.g. bitrate or codec. Stream specifiers
@@ -46,9 +37,9 @@ Matches the stream with this index. E.g. @code{-threads:1 4} would set the
thread count for the second stream to 4. If @var{stream_index} is used as an
additional stream specifier (see below), then it selects stream number
@var{stream_index} from the matching streams. Stream numbering is based on the
order of the streams as detected by libavformat except when a stream group
specifier or program ID is also specified. In this case it is based on the
ordering of the streams in the group or program.
order of the streams as detected by libavformat except when a program ID is
also specified. In this case it is based on the ordering of the streams in the
program.
@item @var{stream_type}[:@var{additional_stream_specifier}]
@var{stream_type} is one of following: 'v' or 'V' for video, 'a' for audio, 's'
for subtitle, 'd' for data, and 't' for attachments. 'v' matches all video
@@ -57,17 +48,6 @@ thumbnails or cover arts. If @var{additional_stream_specifier} is used, then
it matches streams which both have this type and match the
@var{additional_stream_specifier}. Otherwise, it matches all streams of the
specified type.
@item g:@var{group_specifier}[:@var{additional_stream_specifier}]
Matches streams which are in the group with the specifier @var{group_specifier}.
if @var{additional_stream_specifier} is used, then it matches streams which both
are part of the group and match the @var{additional_stream_specifier}.
@var{group_specifier} may be one of the following:
@table @option
@item @var{group_index}
Match the stream with this group index.
@item #@var{group_id} or i:@var{group_id}
Match the stream with this group id.
@end table
@item p:@var{program_id}[:@var{additional_stream_specifier}]
Matches streams which are in the program with the id @var{program_id}. If
@var{additional_stream_specifier} is used, then it matches streams which both
@@ -127,24 +107,17 @@ Print detailed information about the muxer named @var{muxer_name}. Use the
@option{-formats} option to get a list of all muxers and demuxers.
@item filter=@var{filter_name}
Print detailed information about the filter named @var{filter_name}. Use the
Print detailed information about the filter name @var{filter_name}. Use the
@option{-filters} option to get a list of all filters.
@item bsf=@var{bitstream_filter_name}
Print detailed information about the bitstream filter named @var{bitstream_filter_name}.
Print detailed information about the bitstream filter name @var{bitstream_filter_name}.
Use the @option{-bsfs} option to get a list of all bitstream filters.
@item protocol=@var{protocol_name}
Print detailed information about the protocol named @var{protocol_name}.
Use the @option{-protocols} option to get a list of all protocols.
@end table
@item -version
Show version.
@item -buildconf
Show the build configuration, one option per line.
@item -formats
Show available formats (including devices).
@@ -187,9 +160,6 @@ Show available sample formats.
@item -layouts
Show channel names and standard channel layouts.
@item -dispositions
Show stream dispositions.
@item -colors
Show recognized color names.
@@ -375,19 +345,6 @@ Possible flags for this option are:
@item k8
@end table
@end table
@item -cpucount @var{count} (@emph{global})
Override detection of CPU count. This option is intended
for testing. Do not use it unless you know what you're doing.
@example
ffmpeg -cpucount 2
@end example
@item -max_alloc @var{bytes}
Set the maximum size limit for allocating a block on the heap by ffmpeg's
family of malloc functions. Exercise @strong{extreme caution} when using
this option. Don't use if you do not understand the full consequence of doing so.
Default is INT_MAX.
@end table
@section AVOptions

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@@ -46,10 +46,10 @@ Enable fast, but inaccurate seeks for some formats.
@item genpts
Generate missing PTS if DTS is present.
@item igndts
Ignore DTS if PTS is also set. In case the PTS is set, the DTS value
is set to NOPTS. This is ignored when the @code{nofillin} flag is set.
Ignore DTS if PTS is set. Inert when nofillin is set.
@item ignidx
Ignore index.
@item keepside (@emph{deprecated},@emph{inert})
@item nobuffer
Reduce the latency introduced by buffering during initial input streams analysis.
@item nofillin
@@ -70,6 +70,7 @@ This ensures that file and data checksums are reproducible and match between
platforms. Its primary use is for regression testing.
@item flush_packets
Write out packets immediately.
@item latm (@emph{deprecated},@emph{inert})
@item shortest
Stop muxing at the end of the shortest stream.
It may be needed to increase max_interleave_delta to avoid flushing the longer

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@@ -53,7 +53,7 @@ Most distribution and operating system provide a package for it.
@section Cloning the source tree
@example
git clone https://git.ffmpeg.org/ffmpeg.git <target>
git clone git://source.ffmpeg.org/ffmpeg <target>
@end example
This will put the FFmpeg sources into the directory @var{<target>}.
@@ -66,7 +66,7 @@ This will put the FFmpeg sources into the directory @var{<target>} and let
you push back your changes to the remote repository.
@example
git clone git@@ffmpeg.org:ffmpeg-web <target>
git clone gil@@ffmpeg.org:ffmpeg-web <target>
@end example
This will put the source of the FFmpeg website into the directory
@@ -187,18 +187,11 @@ to make sure you don't have untracked files or deletions.
git add [-i|-p|-A] <filenames/dirnames>
@end example
Make sure you have told Git your name, email address and GPG key
Make sure you have told Git your name and email address
@example
git config --global user.name "My Name"
git config --global user.email my@@email.invalid
git config --global user.signingkey ABCDEF0123245
@end example
Enable signing all commits or use -S
@example
git config --global commit.gpgsign true
@end example
Use @option{--global} to set the global configuration for all your Git checkouts.
@@ -224,46 +217,16 @@ git config --global core.editor
or set by one of the following environment variables:
@var{GIT_EDITOR}, @var{VISUAL} or @var{EDITOR}.
@section Writing a commit message
Log messages should be concise but descriptive. Explain why you made a change,
what you did will be obvious from the changes themselves most of the time.
Saying just "bug fix" or "10l" is bad. Remember that people of varying skill
levels look at and educate themselves while reading through your code. Don't
include filenames in log messages, Git provides that information.
Log messages should be concise but descriptive.
The first line must contain the context, a colon and a very short
summary of what the commit does. Details can be added, if necessary,
separated by an empty line. These details should not exceed 60-72 characters
per line, except when containing code.
Example of a good commit message:
@example
avcodec/cbs: add a helper to read extradata within packet side data
Using ff_cbs_read() on the raw buffer will not parse it as extradata,
resulting in parsing errors for example when handling ISOBMFF avcC.
This helper works around that.
@end example
@example
ptr might be NULL
@end example
If the summary on the first line is not enough, in the body of the message,
explain why you made a change, what you did will be obvious from the changes
themselves most of the time. Saying just "bug fix" or "10l" is bad. Remember
that people of varying skill levels look at and educate themselves while
reading through your code. Don't include filenames in log messages except in
the context, Git provides that information.
If the commit fixes a registered issue, state it in a separate line of the
body: @code{Fix Trac ticket #42.}
The first line will be used to name
Possibly make the commit message have a terse, descriptive first line, an
empty line and then a full description. The first line will be used to name
the patch by @command{git format-patch}.
Common mistakes for the first line, as seen in @command{git log --oneline}
include: missing context at the beginning; description of what the code did
before the patch; line too long or wrapped to the second line.
@section Preparing a patchset
@example
@@ -430,19 +393,6 @@ git checkout -b svn_23456 $SHA1
where @var{$SHA1} is the commit hash from the @command{git log} output.
@chapter gpg key generation
If you have no gpg key yet, we recommend that you create a ed25519 based key as it
is small, fast and secure. Especially it results in small signatures in git.
@example
gpg --default-new-key-algo "ed25519/cert,sign+cv25519/encr" --quick-generate-key "human@@server.com"
@end example
When generating a key, make sure the email specified matches the email used in git as some sites like
github consider mismatches a reason to declare such commits unverified. After generating a key you
can add it to the MAINTAINER file and upload it to a keyserver.
@chapter Pre-push checklist
Once you have a set of commits that you feel are ready for pushing,

View File

@@ -222,8 +222,7 @@ $ ffmpeg -f avfoundation -capture_raw_data true -i "zr100:none" out.dv
@section bktr
BSD video input device. Deprecated and will be removed - please contact
the developers if you are interested in maintaining it.
BSD video input device.
@subsection Options
@@ -297,31 +296,16 @@ supports it.
Set the pixel format of the captured video.
Available values are:
@table @samp
@item auto
This is the default which means 8-bit YUV 422 or 8-bit ARGB if format
autodetection is used, 8-bit YUV 422 otherwise.
@item uyvy422
8-bit YUV 422.
@item yuv422p10
10-bit YUV 422.
@item argb
8-bit RGB.
@item bgra
8-bit RGB.
@item rgb10
10-bit RGB.
@end table
@item teletext_lines
@@ -345,33 +329,14 @@ Defines number of audio channels to capture. Must be @samp{2}, @samp{8} or @samp
Defaults to @samp{2}.
@item duplex_mode
Sets the decklink device duplex/profile mode. Must be @samp{unset}, @samp{half}, @samp{full},
@samp{one_sub_device_full}, @samp{one_sub_device_half}, @samp{two_sub_device_full},
@samp{four_sub_device_half}
Sets the decklink device duplex mode. Must be @samp{unset}, @samp{half} or @samp{full}.
Defaults to @samp{unset}.
Note: DeckLink SDK 11.0 have replaced the duplex property by a profile property.
For the DeckLink Duo 2 and DeckLink Quad 2, a profile is shared between any 2
sub-devices that utilize the same connectors. For the DeckLink 8K Pro, a profile
is shared between all 4 sub-devices. So DeckLink 8K Pro support four profiles.
Valid profile modes for DeckLink 8K Pro(with DeckLink SDK >= 11.0):
@samp{one_sub_device_full}, @samp{one_sub_device_half}, @samp{two_sub_device_full},
@samp{four_sub_device_half}
Valid profile modes for DeckLink Quad 2 and DeckLink Duo 2:
@samp{half}, @samp{full}
@item timecode_format
Timecode type to include in the frame and video stream metadata. Must be
@samp{none}, @samp{rp188vitc}, @samp{rp188vitc2}, @samp{rp188ltc},
@samp{rp188hfr}, @samp{rp188any}, @samp{vitc}, @samp{vitc2}, or @samp{serial}.
Defaults to @samp{none} (not included).
In order to properly support 50/60 fps timecodes, the ordering of the queried
timecode types for @samp{rp188any} is HFR, VITC1, VITC2 and LTC for >30 fps
content. Note that this is slightly different to the ordering used by the
DeckLink API, which is HFR, VITC1, LTC, VITC2.
@samp{rp188any}, @samp{vitc}, @samp{vitc2}, or @samp{serial}. Defaults to
@samp{none} (not included).
@item video_input
Sets the video input source. Must be @samp{unset}, @samp{sdi}, @samp{hdmi},
@@ -433,12 +398,6 @@ are dropped till a frame with timecode is received.
Option @var{timecode_format} must be specified.
Defaults to @option{false}.
@item enable_klv(@emph{bool})
If set to @option{true}, extracts KLV data from VANC and outputs KLV packets.
KLV VANC packets are joined based on MID and PSC fields and aggregated into
one KLV packet.
Defaults to @option{false}.
@end table
@subsection Examples
@@ -626,12 +585,6 @@ Save the currently used video capture filter device and its
parameters (if the filter supports it) to a file.
If a file with the same name exists it will be overwritten.
@item use_video_device_timestamps
If set to @option{false}, the timestamp for video frames will be
derived from the wallclock instead of the timestamp provided by
the capture device. This allows working around devices that
provide unreliable timestamps.
@end table
@subsection Examples
@@ -723,7 +676,7 @@ Win32 GDI-based screen capture device.
This device allows you to capture a region of the display on Windows.
Amongst options for the imput filenames are such elements as:
There are two options for the input filename:
@example
desktop
@end example
@@ -731,13 +684,9 @@ or
@example
title=@var{window_title}
@end example
or
@example
hwnd=@var{window_hwnd}
@end example
The first option will capture the entire desktop, or a fixed region of the
desktop. The second and third options will instead capture the contents of a single
desktop. The second option will instead capture the contents of a single
window, regardless of its position on the screen.
For example, to grab the entire desktop using @command{ffmpeg}:
@@ -934,15 +883,11 @@ If you don't understand what all of that means, you probably don't want this. L
DRM device to capture on. Defaults to @option{/dev/dri/card0}.
@item format
Pixel format of the framebuffer. This can be autodetected if you are running Linux 5.7
or later, but needs to be provided for earlier versions. Defaults to @option{bgr0},
which is the most common format used by the Linux console and Xorg X server.
Pixel format of the framebuffer. Defaults to @option{bgr0}.
@item format_modifier
Format modifier to signal on output frames. This is necessary to import correctly into
some APIs. It can be autodetected if you are running Linux 5.7 or later, but will need
to be provided explicitly when needed in earlier versions. See the libdrm documentation
for possible values.
some APIs, but can't be autodetected. See the libdrm documentation for possible values.
@item crtc_id
KMS CRTC ID to define the capture source. The first active plane on the given CRTC
@@ -996,8 +941,9 @@ This input device reads data from the open output pads of a libavfilter
filtergraph.
For each filtergraph open output, the input device will create a
corresponding stream which is mapped to the generated output.
The filtergraph is specified through the option @option{graph}.
corresponding stream which is mapped to the generated output. Currently
only video data is supported. The filtergraph is specified through the
option @option{graph}.
@subsection Options
@@ -1293,11 +1239,11 @@ Specify the samplerate in Hz, by default 48kHz is used.
Specify the channels in use, by default 2 (stereo) is set.
@item frame_size
This option does nothing and is deprecated.
Specify the number of bytes per frame, by default it is set to 1024.
@item fragment_size
Specify the size in bytes of the minimal buffering fragment in PulseAudio, it
will affect the audio latency. By default it is set to 50 ms amount of data.
Specify the minimal buffering fragment in PulseAudio, it will affect the
audio latency. By default it is unset.
@item wallclock
Set the initial PTS using the current time. Default is 1.
@@ -1532,14 +1478,6 @@ ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0+10,20 out.mpg
@subsection Options
@table @option
@item select_region
Specify whether to select the grabbing area graphically using the pointer.
A value of @code{1} prompts the user to select the grabbing area graphically
by clicking and dragging. A single click with no dragging will select the
whole screen. A region with zero width or height will also select the whole
screen. This option overwrites the @var{video_size}, @var{grab_x}, and
@var{grab_y} options. Default value is @code{0}.
@item draw_mouse
Specify whether to draw the mouse pointer. A value of @code{0} specifies
not to draw the pointer. Default value is @code{1}.
@@ -1588,21 +1526,8 @@ With @var{follow_mouse}:
ffmpeg -f x11grab -follow_mouse centered -show_region 1 -framerate 25 -video_size cif -i :0.0 out.mpg
@end example
@item window_id
Grab this window, instead of the whole screen. Default value is 0, which maps to
the whole screen (root window).
The id of a window can be found using the @command{xwininfo} program, possibly with options -tree and
-root.
If the window is later enlarged, the new area is not recorded. Video ends when
the window is closed, unmapped (i.e., iconified) or shrunk beyond the video
size (which defaults to the initial window size).
This option disables options @option{follow_mouse} and @option{select_region}.
@item video_size
Set the video frame size. Default is the full desktop or window.
Set the video frame size. Default is the full desktop.
@item grab_x
@item grab_y

View File

@@ -1,94 +0,0 @@
FFmpeg Infrastructure:
======================
Servers:
~~~~~~~~
Main Server:
------------
Our Main server is hosted at telepoint.bg
for more details see: https://www.ffmpeg.org/#thanks_sponsor_0001
Nothing runs on our main server directly, instead several VMs run on it.
ffmpeg.org VM:
--------------
Web, mail, and public facing git, also website git
fftrac VM:
----------
trac.ffmpeg.org Issue tracking
ffaux VM:
---------
patchwork.ffmpeg.org Patch tracking
vote.ffmpeg.org Condorcet voting
fate:
-----
fate.ffmpeg.org FFmpeg automated testing environment
coverage:
---------
coverage.ffmpeg.org Fate code coverage
The main and fate server as well as VMs currently run ubuntu
Cronjobs:
~~~~~~~~~
Part of the docs is in the main ffmpeg repository as texi files, this part is build by a cronjob. So is the
doxygen stuff as well as the FFmpeg git snapshot.
These 3 scripts are under the ffcron user
Git:
~~~~
Public facing git is provided by our infra, (https://git.ffmpeg.org/gitweb)
main developer ffmpeg git repository for historic reasons is provided by (git@source.ffmpeg.org:ffmpeg)
Other developer git repositories are provided via git@git.ffmpeg.org:<NAME_OF_REPOSITORY>
git mirrors are available on https://github.com/FFmpeg
(there are some exceptions where primary repositories are on github or elsewhere instead of the mirrors)
Github mirrors are redundantly synced by multiple people
You need a new git repository related to FFmpeg ? contact root at ffmpeg.org
Fate:
~~~~~
fatesamples are provided via rsync. Every FFmpeg developer who has a shell account in ffmpeg.org
should be in the samples group and be able to upload samples.
See https://www.ffmpeg.org/fate.html#Uploading-new-samples-to-the-fate-suite
Accounts:
~~~~~~~~~
You need an account for some FFmpeg work? Send mail to root at ffmpeg.org
VMs:
~~~~
You need a VM, docker container for FFmpeg? contact root at ffmpeg.org
(for docker, CC Andriy)
IRC:
~~~~
irc channels are at https://libera.chat/
irc channel archives are at https://libera.irclog.whitequark.org

View File

@@ -116,7 +116,7 @@ or is abusive towards others).
@section How long does it take for my message in the moderation queue to be approved?
The queue is not checked on a regular basis. You can ask on the
@t{#ffmpeg-devel} IRC channel on Libera Chat for someone to approve your message.
@t{#ffmpeg-devel} IRC channel on Freenode for someone to approve your message.
@anchor{How do I delete my message in the moderation queue?}
@section How do I delete my message in the moderation queue?
@@ -155,7 +155,7 @@ Perform a site search using your favorite search engine. Example:
@section Is there an alternative to the mailing list?
You can ask for help in the official @t{#ffmpeg} IRC channel on Libera Chat.
You can ask for help in the official @t{#ffmpeg} IRC channel on Freenode.
Some users prefer the third-party @url{http://www.ffmpeg-archive.org/, Nabble}
interface which presents the mailing lists in a typical forum layout.
@@ -344,7 +344,7 @@ recommended.
Avoid sending the same message to multiple mailing lists.
@item
Please follow our @url{https://ffmpeg.org/community.html#Code-of-conduct, Code of Conduct}.
Please follow our @url{https://ffmpeg.org/developer.html#Code-of-conduct, Code of Conduct}.
@end itemize
@chapter Help

View File

@@ -1,4 +1,3 @@
@anchor{metadata}
@chapter Metadata
@c man begin METADATA

View File

@@ -48,6 +48,11 @@ Files that have MIPS copyright notice in them:
float_dsp_mips.c
libm_mips.h
softfloat_tables.h
* libavcodec/
fft_fixed_32.c
fft_init_table.c
fft_table.h
mdct_fixed_32.c
* libavcodec/mips/
aacdec_fixed.c
aacsbr_fixed.c
@@ -65,6 +70,9 @@ Files that have MIPS copyright notice in them:
compute_antialias_float.h
lsp_mips.h
dsputil_mips.c
fft_mips.c
fft_table.h
fft_init_table.c
fmtconvert_mips.c
iirfilter_mips.c
mpegaudiodsp_mips_fixed.c

View File

@@ -20,7 +20,8 @@ Slice threading -
Frame threading -
* Restrictions with slice threading also apply.
* Custom get_buffer2() and get_format() callbacks must be thread-safe.
* For best performance, the client should set thread_safe_callbacks if it
provides a thread-safe get_buffer() callback.
* There is one frame of delay added for every thread beyond the first one.
Clients must be able to handle this; the pkt_dts and pkt_pts fields in
AVFrame will work as usual.
@@ -55,7 +56,8 @@ speed gain at this point but it should work.
If there are inter-frame dependencies, so the codec calls
ff_thread_report/await_progress(), set FF_CODEC_CAP_ALLOCATE_PROGRESS in
FFCodec.caps_internal and use ff_thread_get_buffer() to allocate frames.
AVCodec.caps_internal and use ff_thread_get_buffer() to allocate frames. The
frames must then be freed with ff_thread_release_buffer().
Otherwise decode directly into the user-supplied frames.
Call ff_thread_report_progress() after some part of the current picture has decoded.

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@@ -267,11 +267,6 @@ CELL/SPU:
http://www-01.ibm.com/chips/techlib/techlib.nsf/techdocs/30B3520C93F437AB87257060006FFE5E/$file/Language_Extensions_for_CBEA_2.4.pdf
http://www-01.ibm.com/chips/techlib/techlib.nsf/techdocs/9F820A5FFA3ECE8C8725716A0062585F/$file/CBE_Handbook_v1.1_24APR2007_pub.pdf
RISC-V-specific:
----------------
The RISC-V Instruction Set Manual, Volume 1, Unprivileged ISA:
https://riscv.org/technical/specifications/
GCC asm links:
--------------
official doc but quite ugly

View File

@@ -38,52 +38,6 @@ ffmpeg -i INPUT -f alsa hw:1,7
@end example
@end itemize
@section AudioToolbox
AudioToolbox output device.
Allows native output to CoreAudio devices on OSX.
The output filename can be empty (or @code{-}) to refer to the default system output device or a number that refers to the device index as shown using: @code{-list_devices true}.
Alternatively, the audio input device can be chosen by index using the
@option{
-audio_device_index <INDEX>
}
, overriding any device name or index given in the input filename.
All available devices can be enumerated by using @option{-list_devices true}, listing
all device names, UIDs and corresponding indices.
@subsection Options
AudioToolbox supports the following options:
@table @option
@item -audio_device_index <INDEX>
Specify the audio device by its index. Overrides anything given in the output filename.
@end table
@subsection Examples
@itemize
@item
Print the list of supported devices and output a sine wave to the default device:
@example
$ ffmpeg -f lavfi -i sine=r=44100 -f audiotoolbox -list_devices true -
@end example
@item
Output a sine wave to the device with the index 2, overriding any output filename:
@example
$ ffmpeg -f lavfi -i sine=r=44100 -f audiotoolbox -audio_device_index 2 -
@end example
@end itemize
@section caca
CACA output device.
@@ -198,48 +152,13 @@ Amount of time to preroll video in seconds.
Defaults to @option{0.5}.
@item duplex_mode
Sets the decklink device duplex/profile mode. Must be @samp{unset}, @samp{half}, @samp{full},
@samp{one_sub_device_full}, @samp{one_sub_device_half}, @samp{two_sub_device_full},
@samp{four_sub_device_half}
Sets the decklink device duplex mode. Must be @samp{unset}, @samp{half} or @samp{full}.
Defaults to @samp{unset}.
Note: DeckLink SDK 11.0 have replaced the duplex property by a profile property.
For the DeckLink Duo 2 and DeckLink Quad 2, a profile is shared between any 2
sub-devices that utilize the same connectors. For the DeckLink 8K Pro, a profile
is shared between all 4 sub-devices. So DeckLink 8K Pro support four profiles.
Valid profile modes for DeckLink 8K Pro(with DeckLink SDK >= 11.0):
@samp{one_sub_device_full}, @samp{one_sub_device_half}, @samp{two_sub_device_full},
@samp{four_sub_device_half}
Valid profile modes for DeckLink Quad 2 and DeckLink Duo 2:
@samp{half}, @samp{full}
@item timing_offset
Sets the genlock timing pixel offset on the used output.
Defaults to @samp{unset}.
@item link
Sets the SDI video link configuration on the used output. Must be
@samp{unset}, @samp{single} link SDI, @samp{dual} link SDI or @samp{quad} link
SDI.
Defaults to @samp{unset}.
@item sqd
Enable Square Division Quad Split mode for Quad-link SDI output.
Must be @samp{unset}, @samp{true} or @samp{false}.
Defaults to @option{unset}.
@item level_a
Enable SMPTE Level A mode on the used output.
Must be @samp{unset}, @samp{true} or @samp{false}.
Defaults to @option{unset}.
@item vanc_queue_size
Sets maximum output buffer size in bytes for VANC data. If the buffering reaches this value,
outgoing VANC data will be dropped.
Defaults to @samp{1048576}.
@end table
@subsection Examples
@@ -302,7 +221,7 @@ ffmpeg -re -i INPUT -c:v rawvideo -pix_fmt bgra -f fbdev /dev/fb0
See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1).
@section opengl
OpenGL output device. Deprecated and will be removed.
OpenGL output device.
To enable this output device you need to configure FFmpeg with @code{--enable-opengl}.
@@ -408,13 +327,7 @@ ffmpeg -i INPUT -f pulse "stream name"
@section sdl
SDL (Simple DirectMedia Layer) output device. Deprecated and will be removed.
For monitoring purposes in FFmpeg, pipes and a video player such as ffplay can be used:
@example
ffmpeg -i INPUT -f nut -c:v rawvideo - | ffplay -
@end example
SDL (Simple DirectMedia Layer) output device.
"sdl2" can be used as alias for "sdl".
@@ -432,18 +345,13 @@ For more information about SDL, check:
@table @option
@item window_borderless
Set SDL window border off.
Default value is 0 (enable window border).
@item window_title
Set the SDL window title, if not specified default to the filename
specified for the output device.
@item window_enable_quit
Enable quit action (using window button or keyboard key)
when non-zero value is provided.
Default value is 1 (enable quit action).
@item window_fullscreen
Set fullscreen mode when non-zero value is provided.
Default value is zero.
@item icon_title
Set the name of the iconified SDL window, if not specified it is set
to the same value of @var{window_title}.
@item window_size
Set the SDL window size, can be a string of the form
@@ -451,13 +359,18 @@ Set the SDL window size, can be a string of the form
If not specified it defaults to the size of the input video,
downscaled according to the aspect ratio.
@item window_title
Set the SDL window title, if not specified default to the filename
specified for the output device.
@item window_x
@item window_y
Set the position of the window on the screen.
@item window_fullscreen
Set fullscreen mode when non-zero value is provided.
Default value is zero.
@item window_enable_quit
Enable quit action (using window button or keyboard key)
when non-zero value is provided.
Default value is 1 (enable quit action)
@end table
@subsection Interactive commands

View File

@@ -92,6 +92,9 @@ For information about compiling FFmpeg on OS/2 see
@chapter Windows
To get help and instructions for building FFmpeg under Windows, check out
the FFmpeg Windows Help Forum at @url{http://ffmpeg.zeranoe.com/forum/}.
@section Native Windows compilation using MinGW or MinGW-w64
FFmpeg can be built to run natively on Windows using the MinGW-w64

View File

@@ -63,17 +63,16 @@ After starting the broker, an FFmpeg client may stream data to the broker using
the command:
@example
ffmpeg -re -i input -f mpegts amqp://[[user]:[password]@@]hostname[:port][/vhost]
ffmpeg -re -i input -f mpegts amqp://[[user]:[password]@@]hostname[:port]
@end example
Where hostname and port (default is 5672) is the address of the broker. The
client may also set a user/password for authentication. The default for both
fields is "guest". Name of virtual host on broker can be set with vhost. The
default value is "/".
fields is "guest".
Muliple subscribers may stream from the broker using the command:
@example
ffplay amqp://[[user]:[password]@@]hostname[:port][/vhost]
ffplay amqp://[[user]:[password]@@]hostname[:port]
@end example
In RabbitMQ all data published to the broker flows through a specific exchange,
@@ -110,21 +109,6 @@ the received message may be truncated causing decoding errors.
The timeout in seconds during the initial connection to the broker. The
default value is rw_timeout, or 5 seconds if rw_timeout is not set.
@item delivery_mode @var{mode}
Sets the delivery mode of each message sent to broker.
The following values are accepted:
@table @samp
@item persistent
Delivery mode set to "persistent" (2). This is the default value.
Messages may be written to the broker's disk depending on its setup.
@item non-persistent
Delivery mode set to "non-persistent" (1).
Messages will stay in broker's memory unless the broker is under memory
pressure.
@end table
@end table
@section async
@@ -175,16 +159,6 @@ Caching wrapper for input stream.
Cache the input stream to temporary file. It brings seeking capability to live streams.
The accepted options are:
@table @option
@item read_ahead_limit
Amount in bytes that may be read ahead when seeking isn't supported. Range is -1 to INT_MAX.
-1 for unlimited. Default is 65536.
@end table
URL Syntax is
@example
cache:@var{URL}
@end example
@@ -215,38 +189,6 @@ ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
Note that you may need to escape the character "|" which is special for
many shells.
@section concatf
Physical concatenation protocol using a line break delimited list of
resources.
Read and seek from many resources in sequence as if they were
a unique resource.
A URL accepted by this protocol has the syntax:
@example
concatf:@var{URL}
@end example
where @var{URL} is the url containing a line break delimited list of
resources to be concatenated, each one possibly specifying a distinct
protocol. Special characters must be escaped with backslash or single
quotes. See @ref{quoting_and_escaping,,the "Quoting and escaping"
section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
For example to read a sequence of files @file{split1.mpeg},
@file{split2.mpeg}, @file{split3.mpeg} listed in separate lines within
a file @file{split.txt} with @command{ffplay} use the command:
@example
ffplay concatf:split.txt
@end example
Where @file{split.txt} contains the lines:
@example
split1.mpeg
split2.mpeg
split3.mpeg
@end example
@section crypto
AES-encrypted stream reading protocol.
@@ -275,33 +217,6 @@ For example, to convert a GIF file given inline with @command{ffmpeg}:
ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png
@end example
@section fd
File descriptor access protocol.
The accepted syntax is:
@example
fd: -fd @var{file_descriptor}
@end example
If @option{fd} is not specified, by default the stdout file descriptor will be
used for writing, stdin for reading. Unlike the pipe protocol, fd protocol has
seek support if it corresponding to a regular file. fd protocol doesn't support
pass file descriptor via URL for security.
This protocol accepts the following options:
@table @option
@item blocksize
Set I/O operation maximum block size, in bytes. Default value is
@code{INT_MAX}, which results in not limiting the requested block size.
Setting this value reasonably low improves user termination request reaction
time, which is valuable if data transmission is slow.
@item fd
Set file descriptor.
@end table
@section file
File access protocol.
@@ -400,12 +315,6 @@ operation. ff* tools may produce incomplete content due to server limitations.
Gopher protocol.
@section gophers
Gophers protocol.
The Gopher protocol with TLS encapsulation.
@section hls
Read Apple HTTP Live Streaming compliant segmented stream as
@@ -465,6 +374,14 @@ Set the Referer header. Include 'Referer: URL' header in HTTP request.
Override the User-Agent header. If not specified the protocol will use a
string describing the libavformat build. ("Lavf/<version>")
@item user-agent
This is a deprecated option, you can use user_agent instead it.
@item timeout
Set timeout in microseconds of socket I/O operations used by the underlying low level
operation. By default it is set to -1, which means that the timeout is
not specified.
@item reconnect_at_eof
If set then eof is treated like an error and causes reconnection, this is useful
for live / endless streams.
@@ -472,13 +389,6 @@ for live / endless streams.
@item reconnect_streamed
If set then even streamed/non seekable streams will be reconnected on errors.
@item reconnect_on_network_error
Reconnect automatically in case of TCP/TLS errors during connect.
@item reconnect_on_http_error
A comma separated list of HTTP status codes to reconnect on. The list can
include specific status codes (e.g. '503') or the strings '4xx' / '5xx'.
@item reconnect_delay_max
Sets the maximum delay in seconds after which to give up reconnecting
@@ -556,28 +466,6 @@ Send an Expect: 100-continue header for POST. If set to 1 it will send, if set
to 0 it won't, if set to -1 it will try to send if it is applicable. Default
value is -1.
@item auth_type
Set HTTP authentication type. No option for Digest, since this method requires
getting nonce parameters from the server first and can't be used straight away like
Basic.
@table @option
@item none
Choose the HTTP authentication type automatically. This is the default.
@item basic
Choose the HTTP basic authentication.
Basic authentication sends a Base64-encoded string that contains a user name and password
for the client. Base64 is not a form of encryption and should be considered the same as
sending the user name and password in clear text (Base64 is a reversible encoding).
If a resource needs to be protected, strongly consider using an authentication scheme
other than basic authentication. HTTPS/TLS should be used with basic authentication.
Without these additional security enhancements, basic authentication should not be used
to protect sensitive or valuable information.
@end table
@end table
@subsection HTTP Cookies
@@ -632,44 +520,12 @@ audio/mpeg.
This enables support for Icecast versions < 2.4.0, that do not support the
HTTP PUT method but the SOURCE method.
@item tls
Establish a TLS (HTTPS) connection to Icecast.
@end table
@example
icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint}
@end example
@section ipfs
InterPlanetary File System (IPFS) protocol support. One can access files stored
on the IPFS network through so-called gateways. These are http(s) endpoints.
This protocol wraps the IPFS native protocols (ipfs:// and ipns://) to be sent
to such a gateway. Users can (and should) host their own node which means this
protocol will use one's local gateway to access files on the IPFS network.
This protocol accepts the following options:
@table @option
@item gateway
Defines the gateway to use. When not set, the protocol will first try
locating the local gateway by looking at @code{$IPFS_GATEWAY}, @code{$IPFS_PATH}
and @code{$HOME/.ipfs/}, in that order.
@end table
One can use this protocol in 2 ways. Using IPFS:
@example
ffplay ipfs://<hash>
@end example
Or the IPNS protocol (IPNS is mutable IPFS):
@example
ffplay ipns://<hash>
@end example
@section mmst
MMS (Microsoft Media Server) protocol over TCP.
@@ -714,7 +570,7 @@ The accepted syntax is:
pipe:[@var{number}]
@end example
If @option{fd} isn't specified, @var{number} is the number corresponding to the file descriptor of the
@var{number} is the number corresponding to the file descriptor of the
pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
is not specified, by default the stdout file descriptor will be used
for writing, stdin for reading.
@@ -741,8 +597,6 @@ Set I/O operation maximum block size, in bytes. Default value is
@code{INT_MAX}, which results in not limiting the requested block size.
Setting this value reasonably low improves user termination request reaction
time, which is valuable if data transmission is slow.
@item fd
Set file descriptor.
@end table
Note that some formats (typically MOV), require the output protocol to
@@ -783,50 +637,6 @@ Example usage:
-f rtp_mpegts -fec prompeg=l=8:d=4 rtp://@var{hostname}:@var{port}
@end example
@section rist
Reliable Internet Streaming Transport protocol
The accepted options are:
@table @option
@item rist_profile
Supported values:
@table @samp
@item simple
@item main
This one is default.
@item advanced
@end table
@item buffer_size
Set internal RIST buffer size in milliseconds for retransmission of data.
Default value is 0 which means the librist default (1 sec). Maximum value is 30
seconds.
@item fifo_size
Size of the librist receiver output fifo in number of packets. This must be a
power of 2.
Defaults to 8192 (vs the librist default of 1024).
@item overrun_nonfatal=@var{1|0}
Survive in case of librist fifo buffer overrun. Default value is 0.
@item pkt_size
Set maximum packet size for sending data. 1316 by default.
@item log_level
Set loglevel for RIST logging messages. You only need to set this if you
explicitly want to enable debug level messages or packet loss simulation,
otherwise the regular loglevel is respected.
@item secret
Set override of encryption secret, by default is unset.
@item encryption
Set encryption type, by default is disabled.
Acceptable values are 128 and 256.
@end table
@section rtmp
Real-Time Messaging Protocol.
@@ -896,13 +706,6 @@ be named, by prefixing the type with 'N' and specifying the name before
the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
times to construct arbitrary AMF sequences.
@item rtmp_enhanced_codecs
Specify the list of codecs the client advertises to support in an
enhanced RTMP stream. This option should be set to a comma separated
list of fourcc values, like @code{hvc1,av01,vp09} for multiple codecs
or @code{hvc1} for only one codec. The specified list will be presented
in the "fourCcLive" property of the Connect Command Message.
@item rtmp_flashver
Version of the Flash plugin used to run the SWF player. The default
is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
@@ -948,11 +751,6 @@ URL to player swf file, compute hash/size automatically.
@item rtmp_tcurl
URL of the target stream. Defaults to proto://host[:port]/app.
@item tcp_nodelay=@var{1|0}
Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
@emph{Remark: Writing to the socket is currently not optimized to minimize system calls and reduces the efficiency / effect of TCP_NODELAY.}
@end table
For example to read with @command{ffplay} a multimedia resource named
@@ -1140,9 +938,6 @@ Set the local RTCP port to @var{n}.
@item pkt_size=@var{n}
Set max packet size (in bytes) to @var{n}.
@item buffer_size=@var{size}
Set the maximum UDP socket buffer size in bytes.
@item connect=0|1
Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
to 0).
@@ -1160,13 +955,6 @@ set to 1) or to a default remote address (if set to 0).
@item localport=@var{n}
Set the local RTP port to @var{n}.
@item localaddr=@var{addr}
Local IP address of a network interface used for sending packets or joining
multicast groups.
@item timeout=@var{n}
Set timeout (in microseconds) of socket I/O operations to @var{n}.
This is a deprecated option. Instead, @option{localrtpport} should be
used.
@@ -1211,59 +999,6 @@ Options can be set on the @command{ffmpeg}/@command{ffplay} command
line, or set in code via @code{AVOption}s or in
@code{avformat_open_input}.
@subsection Muxer
The following options are supported.
@table @option
@item rtsp_transport
Set RTSP transport protocols.
It accepts the following values:
@table @samp
@item udp
Use UDP as lower transport protocol.
@item tcp
Use TCP (interleaving within the RTSP control channel) as lower
transport protocol.
@end table
Default value is @samp{0}.
@item rtsp_flags
Set RTSP flags.
The following values are accepted:
@table @samp
@item latm
Use MP4A-LATM packetization instead of MPEG4-GENERIC for AAC.
@item rfc2190
Use RFC 2190 packetization instead of RFC 4629 for H.263.
@item skip_rtcp
Don't send RTCP sender reports.
@item h264_mode0
Use mode 0 for H.264 in RTP.
@item send_bye
Send RTCP BYE packets when finishing.
@end table
Default value is @samp{0}.
@item min_port
Set minimum local UDP port. Default value is 5000.
@item max_port
Set maximum local UDP port. Default value is 65000.
@item buffer_size
Set the maximum socket buffer size in bytes.
@item pkt_size
Set max send packet size (in bytes). Default value is 1472.
@end table
@subsection Demuxer
The following options are supported.
@table @option
@@ -1289,10 +1024,6 @@ Use UDP multicast as lower transport protocol.
@item http
Use HTTP tunneling as lower transport protocol, which is useful for
passing proxies.
@item https
Use HTTPs tunneling as lower transport protocol, which is useful for
passing proxies and widely used for security consideration.
@end table
Multiple lower transport protocols may be specified, in that case they are
@@ -1310,9 +1041,6 @@ Accept packets only from negotiated peer address and port.
Act as a server, listening for an incoming connection.
@item prefer_tcp
Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
@item satip_raw
Export raw MPEG-TS stream instead of demuxing. The flag will simply write out
the raw stream, with the original PAT/PMT/PIDs intact.
@end table
Default value is @samp{none}.
@@ -1325,7 +1053,6 @@ The following flags are accepted:
@item video
@item audio
@item data
@item subtitle
@end table
By default it accepts all media types.
@@ -1336,23 +1063,21 @@ Set minimum local UDP port. Default value is 5000.
@item max_port
Set maximum local UDP port. Default value is 65000.
@item listen_timeout
Set maximum timeout (in seconds) to establish an initial connection. Setting
@option{listen_timeout} > 0 sets @option{rtsp_flags} to @samp{listen}. Default is -1
which means an infinite timeout when @samp{listen} mode is set.
@item timeout
Set maximum timeout (in seconds) to wait for incoming connections.
A value of -1 means infinite (default). This option implies the
@option{rtsp_flags} set to @samp{listen}.
@item reorder_queue_size
Set number of packets to buffer for handling of reordered packets.
@item timeout
@item stimeout
Set socket TCP I/O timeout in microseconds.
@item user_agent
@item user-agent
Override User-Agent header. If not specified, it defaults to the
libavformat identifier string.
@item buffer_size
Set the maximum socket buffer size in bytes.
@end table
When receiving data over UDP, the demuxer tries to reorder received packets
@@ -1637,12 +1362,6 @@ when the old encryption key is decommissioned. Default is -1.
-1 means auto (0x1000 in srt library). The range for
this option is integers in the 0 - @code{INT_MAX}.
@item snddropdelay=@var{microseconds}
The sender's extra delay before dropping packets. This delay is
added to the default drop delay time interval value.
Special value -1: Do not drop packets on the sender at all.
@item payload_size=@var{bytes}
Sets the maximum declared size of a packet transferred
during the single call to the sending function in Live
@@ -1742,9 +1461,6 @@ This option doesnt make sense in Rendezvous connection; the result
might be that simply one side will override the value from the other
side and its the matter of luck which one would win
@item srt_streamid=@var{string}
Alias for @samp{streamid} to avoid conflict with ffmpeg command line option.
@item smoother=@var{live|file}
The type of Smoother used for the transmission for that socket, which
is responsible for the transmission and congestion control. The Smoother
@@ -1794,11 +1510,6 @@ Default is -1. -1 means auto (off with 0 seconds in live mode, on with 180
seconds in file mode). The range for this option is integers in the
0 - @code{INT_MAX}.
@item tsbpd=@var{1|0}
When true, use Timestamp-based Packet Delivery mode. The default behavior
depends on the transmission type: enabled in live mode, disabled in file
mode.
@end table
For more information see: @url{https://github.com/Haivision/srt}.
@@ -1885,15 +1596,8 @@ tcp://@var{hostname}:@var{port}[?@var{options}]
The list of supported options follows.
@table @option
@item listen=@var{2|1|0}
Listen for an incoming connection. 0 disables listen, 1 enables listen in
single client mode, 2 enables listen in multi-client mode. Default value is 0.
@item local_addr=@var{addr}
Local IP address of a network interface used for tcp socket connect.
@item local_port=@var{port}
Local port used for tcp socket connect.
@item listen=@var{1|0}
Listen for an incoming connection. Default value is 0.
@item timeout=@var{microseconds}
Set raise error timeout, expressed in microseconds.
@@ -1913,8 +1617,6 @@ Set send buffer size, expressed bytes.
@item tcp_nodelay=@var{1|0}
Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
@emph{Remark: Writing to the socket is currently not optimized to minimize system calls and reduces the efficiency / effect of TCP_NODELAY.}
@item tcp_mss=@var{bytes}
Set maximum segment size for outgoing TCP packets, expressed in bytes.
@end table
@@ -1971,10 +1673,6 @@ A file containing the private key for the certificate.
If enabled, listen for connections on the provided port, and assume
the server role in the handshake instead of the client role.
@item http_proxy
The HTTP proxy to tunnel through, e.g. @code{http://example.com:1234}.
The proxy must support the CONNECT method.
@end table
Example command lines:
@@ -2168,4 +1866,5 @@ decoding errors.
@end table
@c man end PROTOCOLS

View File

@@ -11,8 +11,18 @@ programmatic use.
@table @option
@item uchl, used_chlayout
Set used input channel layout. Default is unset. This option is
@item ich, in_channel_count
Set the number of input channels. Default value is 0. Setting this
value is not mandatory if the corresponding channel layout
@option{in_channel_layout} is set.
@item och, out_channel_count
Set the number of output channels. Default value is 0. Setting this
value is not mandatory if the corresponding channel layout
@option{out_channel_layout} is set.
@item uch, used_channel_count
Set the number of used input channels. Default value is 0. This option is
only used for special remapping.
@item isr, in_sample_rate
@@ -31,8 +41,8 @@ Specify the output sample format. It is set by default to @code{none}.
Set the internal sample format. Default value is @code{none}.
This will automatically be chosen when it is not explicitly set.
@item ichl, in_chlayout
@item ochl, out_chlayout
@item icl, in_channel_layout
@item ocl, out_channel_layout
Set the input/output channel layout.
See @ref{channel layout syntax,,the Channel Layout section in the ffmpeg-utils(1) manual,ffmpeg-utils}

View File

@@ -20,45 +20,8 @@
# License along with FFmpeg; if not, write to the Free Software
# Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
# Texinfo 7.0 changed the syntax of various functions.
# Provide a shim for older versions.
sub ff_set_from_init_file($$) {
my $key = shift;
my $value = shift;
if (exists &{'texinfo_set_from_init_file'}) {
texinfo_set_from_init_file($key, $value);
} else {
set_from_init_file($key, $value);
}
}
sub ff_get_conf($) {
my $key = shift;
if (exists &{'texinfo_get_conf'}) {
texinfo_get_conf($key);
} else {
get_conf($key);
}
}
sub get_formatting_function($$) {
my $obj = shift;
my $func = shift;
my $sub = $obj->can('formatting_function');
if ($sub) {
return $obj->formatting_function($func);
} else {
return $obj->{$func};
}
}
# determine texinfo version
my $program_version_num = version->declare(ff_get_conf('PACKAGE_VERSION'))->numify;
my $program_version_6_8 = $program_version_num >= 6.008000;
# no navigation elements
ff_set_from_init_file('HEADERS', 0);
set_from_init_file('HEADERS', 0);
sub ffmpeg_heading_command($$$$$)
{
@@ -92,7 +55,7 @@ sub ffmpeg_heading_command($$$$$)
$element = $command->{'parent'};
}
if ($element) {
$result .= &{get_formatting_function($self, 'format_element_header')}($self, $cmdname,
$result .= &{$self->{'format_element_header'}}($self, $cmdname,
$command, $element);
}
@@ -149,11 +112,7 @@ sub ffmpeg_heading_command($$$$$)
$cmdname
= $Texinfo::Common::level_to_structuring_command{$cmdname}->[$heading_level];
}
# format_heading_text expects an array of headings for texinfo >= 7.0
if ($program_version_num >= 7.000000) {
$heading = [$heading];
}
$result .= &{get_formatting_function($self,'format_heading_text')}(
$result .= &{$self->{'format_heading_text'}}(
$self, $cmdname, $heading,
$heading_level +
$self->get_conf('CHAPTER_HEADER_LEVEL') - 1, $command);
@@ -168,18 +127,14 @@ foreach my $command (keys(%Texinfo::Common::sectioning_commands), 'node') {
}
# print the TOC where @contents is used
if ($program_version_6_8) {
ff_set_from_init_file('CONTENTS_OUTPUT_LOCATION', 'inline');
} else {
ff_set_from_init_file('INLINE_CONTENTS', 1);
}
set_from_init_file('INLINE_CONTENTS', 1);
# make chapters <h2>
ff_set_from_init_file('CHAPTER_HEADER_LEVEL', 2);
set_from_init_file('CHAPTER_HEADER_LEVEL', 2);
# Do not add <hr>
ff_set_from_init_file('DEFAULT_RULE', '');
ff_set_from_init_file('BIG_RULE', '');
set_from_init_file('DEFAULT_RULE', '');
set_from_init_file('BIG_RULE', '');
# Customized file beginning
sub ffmpeg_begin_file($$$)
@@ -196,18 +151,7 @@ sub ffmpeg_begin_file($$$)
my ($title, $description, $encoding, $date, $css_lines,
$doctype, $bodytext, $copying_comment, $after_body_open,
$extra_head, $program_and_version, $program_homepage,
$program, $generator);
if ($program_version_num >= 7.000000) {
($title, $description, $encoding, $date, $css_lines,
$doctype, $bodytext, $copying_comment, $after_body_open,
$extra_head, $program_and_version, $program_homepage,
$program, $generator) = $self->_file_header_information($command);
} else {
($title, $description, $encoding, $date, $css_lines,
$doctype, $bodytext, $copying_comment, $after_body_open,
$extra_head, $program_and_version, $program_homepage,
$program, $generator) = $self->_file_header_informations($command);
}
$program, $generator) = $self->_file_header_informations($command);
my $links = $self->_get_links ($filename, $element);
@@ -240,11 +184,7 @@ EOT
return $head1 . $head_title . $head2 . $head_title . $head3;
}
if ($program_version_6_8) {
texinfo_register_formatting_function('format_begin_file', \&ffmpeg_begin_file);
} else {
texinfo_register_formatting_function('begin_file', \&ffmpeg_begin_file);
}
texinfo_register_formatting_function('begin_file', \&ffmpeg_begin_file);
sub ffmpeg_program_string($)
{
@@ -261,17 +201,13 @@ sub ffmpeg_program_string($)
$self->gdt('This document was generated automatically.'));
}
}
if ($program_version_6_8) {
texinfo_register_formatting_function('format_program_string', \&ffmpeg_program_string);
} else {
texinfo_register_formatting_function('program_string', \&ffmpeg_program_string);
}
texinfo_register_formatting_function('program_string', \&ffmpeg_program_string);
# Customized file ending
sub ffmpeg_end_file($)
{
my $self = shift;
my $program_string = &{get_formatting_function($self,'format_program_string')}($self);
my $program_string = &{$self->{'format_program_string'}}($self);
my $program_text = <<EOT;
<p style="font-size: small;">
$program_string
@@ -284,15 +220,11 @@ EOT
EOT
return $program_text . $footer;
}
if ($program_version_6_8) {
texinfo_register_formatting_function('format_end_file', \&ffmpeg_end_file);
} else {
texinfo_register_formatting_function('end_file', \&ffmpeg_end_file);
}
texinfo_register_formatting_function('end_file', \&ffmpeg_end_file);
# Dummy title command
# Ignore title. Title is handled through ffmpeg_begin_file().
ff_set_from_init_file('USE_TITLEPAGE_FOR_TITLE', 1);
set_from_init_file('USE_TITLEPAGE_FOR_TITLE', 1);
sub ffmpeg_title($$$$)
{
return '';
@@ -310,14 +242,8 @@ sub ffmpeg_float($$$$$)
my $args = shift;
my $content = shift;
my ($caption, $prepended);
if ($program_version_num >= 7.000000) {
($caption, $prepended) = Texinfo::Convert::Converter::float_name_caption($self,
$command);
} else {
($caption, $prepended) = Texinfo::Common::float_name_caption($self,
$command);
}
my ($caption, $prepended) = Texinfo::Common::float_name_caption($self,
$command);
my $caption_text = '';
my $prepended_text;
my $prepended_save = '';
@@ -389,13 +315,8 @@ sub ffmpeg_float($$$$$)
$caption->{'args'}->[0], 'float caption');
}
if ($prepended_text.$caption_text ne '') {
if ($program_version_num >= 7.000000) {
$prepended_text = $self->html_attribute_class('div',['float-caption']). '>'
. $prepended_text;
} else {
$prepended_text = $self->_attribute_class('div','float-caption'). '>'
. $prepended_text;
}
$prepended_text = $self->_attribute_class('div','float-caption'). '>'
. $prepended_text;
$caption_text .= '</div>';
}
my $html_class = '';
@@ -408,13 +329,8 @@ sub ffmpeg_float($$$$$)
$prepended_text = '';
$caption_text = '';
}
if ($program_version_num >= 7.000000) {
return $self->html_attribute_class('div', [$html_class]). '>' . "\n" .
$prepended_text . $caption_text . $content . '</div>';
} else {
return $self->_attribute_class('div', $html_class). '>' . "\n" .
$prepended_text . $caption_text . $content . '</div>';
}
return $self->_attribute_class('div', $html_class). '>' . "\n" .
$prepended_text . $caption_text . $content . '</div>';
}
texinfo_register_command_formatting('float',

View File

@@ -172,9 +172,6 @@ INF: while(<$inf>) {
} elsif ($ended =~ /^(?:itemize|enumerate|(?:multi|[fv])?table)$/) {
$_ = "\n=back\n";
$ic = pop @icstack;
} elsif ($ended =~ /^float$/) {
$_ = "\n=back\n";
$ic = pop @icstack;
} else {
die "unknown command \@end $ended at line $.\n";
}
@@ -300,12 +297,6 @@ INF: while(<$inf>) {
$_ = ""; # need a paragraph break
};
/^\@(float)\s+\w+/ and do {
push @endwstack, $endw;
$endw = $1;
$_ = "\n=over 4\n";
};
/^\@item\s+(.*\S)\s*$/ and $endw eq "multitable" and do {
my $columns = $1;
$columns =~ s/\@tab/ : /;

File diff suppressed because it is too large Load Diff

View File

@@ -110,13 +110,11 @@ maximum of 2 digits. The @var{m} at the end expresses decimal value for
@emph{or}
@example
[-]@var{S}+[.@var{m}...][s|ms|us]
[-]@var{S}+[.@var{m}...]
@end example
@var{S} expresses the number of seconds, with the optional decimal part
@var{m}. The optional literal suffixes @samp{s}, @samp{ms} or @samp{us}
indicate to interpret the value as seconds, milliseconds or microseconds,
respectively.
@var{m}.
In both expressions, the optional @samp{-} indicates negative duration.
@@ -695,8 +693,6 @@ FL+FR+FC+LFE+SL+SR
FL+FR+FC+BC+SL+SR
@item 6.0(front)
FL+FR+FLC+FRC+SL+SR
@item 3.1.2
FL+FR+FC+LFE+TFL+TFR
@item hexagonal
FL+FR+FC+BL+BR+BC
@item 6.1
@@ -715,52 +711,28 @@ FL+FR+FC+LFE+BL+BR+SL+SR
FL+FR+FC+LFE+BL+BR+FLC+FRC
@item 7.1(wide-side)
FL+FR+FC+LFE+FLC+FRC+SL+SR
@item 5.1.2
FL+FR+FC+LFE+BL+BR+TFL+TFR
@item octagonal
FL+FR+FC+BL+BR+BC+SL+SR
@item cube
FL+FR+BL+BR+TFL+TFR+TBL+TBR
@item 5.1.4
FL+FR+FC+LFE+BL+BR+TFL+TFR+TBL+TBR
@item 7.1.2
FL+FR+FC+LFE+BL+BR+SL+SR+TFL+TFR
@item 7.1.4
FL+FR+FC+LFE+BL+BR+SL+SR+TFL+TFR+TBL+TBR
@item 7.2.3
FL+FR+FC+LFE+BL+BR+SL+SR+TFL+TFR+TBC+LFE2
@item 9.1.4
FL+FR+FC+LFE+BL+BR+FLC+FRC+SL+SR+TFL+TFR+TBL+TBR
@item hexadecagonal
FL+FR+FC+BL+BR+BC+SL+SR+WL+WR+TBL+TBR+TBC+TFC+TFL+TFR
@item downmix
DL+DR
@item 22.2
FL+FR+FC+LFE+BL+BR+FLC+FRC+BC+SL+SR+TC+TFL+TFC+TFR+TBL+TBC+TBR+LFE2+TSL+TSR+BFC+BFL+BFR
@end table
A custom channel layout can be specified as a sequence of terms, separated by '+'.
Each term can be:
A custom channel layout can be specified as a sequence of terms, separated by
'+' or '|'. Each term can be:
@itemize
@item
the name of a single channel (e.g. @samp{FL}, @samp{FR}, @samp{FC}, @samp{LFE}, etc.),
each optionally containing a custom name after a '@@', (e.g. @samp{FL@@Left},
@samp{FR@@Right}, @samp{FC@@Center}, @samp{LFE@@Low_Frequency}, etc.)
@end itemize
A standard channel layout can be specified by the following:
@itemize
@item
the name of a single channel (e.g. @samp{FL}, @samp{FR}, @samp{FC}, @samp{LFE}, etc.)
@item
the name of a standard channel layout (e.g. @samp{mono},
@samp{stereo}, @samp{4.0}, @samp{quad}, @samp{5.0}, etc.)
@item
the name of a single channel (e.g. @samp{FL}, @samp{FR}, @samp{FC}, @samp{LFE}, etc.)
@item
a number of channels, in decimal, followed by 'c', yielding the default channel
layout for that number of channels (see the function
@code{av_channel_layout_default}). Note that not all channel counts have a
@code{av_get_default_channel_layout}). Note that not all channel counts have a
default layout.
@item
@@ -777,7 +749,7 @@ Before libavutil version 53 the trailing character "c" to specify a number of
channels was optional, but now it is required, while a channel layout mask can
also be specified as a decimal number (if and only if not followed by "c" or "C").
See also the function @code{av_channel_layout_from_string} defined in
See also the function @code{av_get_channel_layout} defined in
@file{libavutil/channel_layout.h}.
@c man end SYNTAX
@@ -815,7 +787,7 @@ Compute arcsine of @var{x}.
@item atan(x)
Compute arctangent of @var{x}.
@item atan2(y, x)
@item atan2(x, y)
Compute principal value of the arc tangent of @var{y}/@var{x}.
@item between(x, min, max)
@@ -939,15 +911,9 @@ Returns the value of the expression printed.
Prints t with loglevel l
@item random(idx)
Return a pseudo random value between 0.0 and 1.0. @var{idx} is the
index of the internal variable which will be used to save the
seed/state.
@item randomi(idx, min, max)
Return a pseudo random value in the interval between @var{min} and
@var{max}. @var{idx} is the index of the internal variable which will
be used to save the seed/state.
@item random(x)
Return a pseudo random value between 0.0 and 1.0. @var{x} is the index of the
internal variable which will be used to save the seed/state.
@item root(expr, max)
Find an input value for which the function represented by @var{expr}
@@ -1095,13 +1061,13 @@ indication of the corresponding powers of 10 and of 2.
@item T
10^12 / 2^40
@item P
10^15 / 2^50
10^15 / 2^40
@item E
10^18 / 2^60
10^18 / 2^50
@item Z
10^21 / 2^70
10^21 / 2^60
@item Y
10^24 / 2^80
10^24 / 2^70
@end table
@c man end EXPRESSION EVALUATION

View File

@@ -418,4 +418,4 @@ done:
When all of this is done, you can submit your patch to the ffmpeg-devel
mailing-list for review. If you need any help, feel free to come on our IRC
channel, #ffmpeg-devel on irc.libera.chat.
channel, #ffmpeg-devel on irc.freenode.net.

2
ffbuild/.gitignore vendored
View File

@@ -1,6 +1,4 @@
/.config
/bin2c
/bin2c.exe
/config.fate
/config.log
/config.mak

View File

@@ -8,15 +8,10 @@ OBJS-$(HAVE_MIPSFPU) += $(MIPSFPU-OBJS) $(MIPSFPU-OBJS-yes)
OBJS-$(HAVE_MIPSDSP) += $(MIPSDSP-OBJS) $(MIPSDSP-OBJS-yes)
OBJS-$(HAVE_MIPSDSPR2) += $(MIPSDSPR2-OBJS) $(MIPSDSPR2-OBJS-yes)
OBJS-$(HAVE_MSA) += $(MSA-OBJS) $(MSA-OBJS-yes)
OBJS-$(HAVE_MMI) += $(MMI-OBJS) $(MMI-OBJS-yes)
OBJS-$(HAVE_LSX) += $(LSX-OBJS) $(LSX-OBJS-yes)
OBJS-$(HAVE_LASX) += $(LASX-OBJS) $(LASX-OBJS-yes)
OBJS-$(HAVE_MMI) += $(MMI-OBJS) $(MMI-OBJS-yes)
OBJS-$(HAVE_ALTIVEC) += $(ALTIVEC-OBJS) $(ALTIVEC-OBJS-yes)
OBJS-$(HAVE_VSX) += $(VSX-OBJS) $(VSX-OBJS-yes)
OBJS-$(HAVE_RV) += $(RV-OBJS) $(RV-OBJS-yes)
OBJS-$(HAVE_RVV) += $(RVV-OBJS) $(RVV-OBJS-yes)
OBJS-$(HAVE_MMX) += $(MMX-OBJS) $(MMX-OBJS-yes)
OBJS-$(HAVE_X86ASM) += $(X86ASM-OBJS) $(X86ASM-OBJS-yes)

View File

@@ -1,76 +0,0 @@
/*
* This file is part of FFmpeg.
*
* 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 <string.h>
#include <stdio.h>
int main(int argc, char **argv)
{
const char *name;
FILE *input, *output;
unsigned int length = 0;
unsigned char data;
if (argc < 3 || argc > 4)
return 1;
input = fopen(argv[1], "rb");
if (!input)
return -1;
output = fopen(argv[2], "wb");
if (!output)
return -1;
if (argc == 4) {
name = argv[3];
} else {
size_t arglen = strlen(argv[1]);
name = argv[1];
for (int i = 0; i < arglen; i++) {
if (argv[1][i] == '.')
argv[1][i] = '_';
else if (argv[1][i] == '/')
name = &argv[1][i+1];
}
}
fprintf(output, "const unsigned char ff_%s_data[] = { ", name);
while (fread(&data, 1, 1, input) > 0) {
fprintf(output, "0x%02x, ", data);
length++;
}
fprintf(output, "0x00 };\n");
fprintf(output, "const unsigned int ff_%s_len = %u;\n", name, length);
fclose(output);
if (ferror(input) || !feof(input))
return -1;
fclose(input);
return 0;
}

View File

@@ -12,13 +12,10 @@ endif
ifndef SUBDIR
BIN2CEXE = ffbuild/bin2c$(HOSTEXESUF)
BIN2C = $(BIN2CEXE)
ifndef V
Q = @
ECHO = printf "$(1)\t%s\n" $(2)
BRIEF = CC CXX OBJCC HOSTCC HOSTLD AS X86ASM AR LD STRIP CP WINDRES NVCC BIN2C
BRIEF = CC CXX OBJCC HOSTCC HOSTLD AS X86ASM AR LD STRIP CP WINDRES NVCC
SILENT = DEPCC DEPHOSTCC DEPAS DEPX86ASM RANLIB RM
MSG = $@
@@ -29,8 +26,7 @@ $(foreach VAR,$(SILENT),$(eval override $(VAR) = @$($(VAR))))
$(eval INSTALL = @$(call ECHO,INSTALL,$$(^:$(SRC_DIR)/%=%)); $(INSTALL))
endif
# Prepend to a recursively expanded variable without making it simply expanded.
PREPEND = $(eval $(1) = $(patsubst %,$$(%), $(2)) $(value $(1)))
ALLFFLIBS = avcodec avdevice avfilter avformat avresample avutil postproc swscale swresample
# NASM requires -I path terminated with /
IFLAGS := -I. -I$(SRC_LINK)/
@@ -40,9 +36,7 @@ CCFLAGS = $(CPPFLAGS) $(CFLAGS)
OBJCFLAGS += $(EOBJCFLAGS)
OBJCCFLAGS = $(CPPFLAGS) $(CFLAGS) $(OBJCFLAGS)
ASFLAGS := $(CPPFLAGS) $(ASFLAGS)
# Use PREPEND here so that later (target-dependent) additions to CPPFLAGS
# end up in CXXFLAGS.
$(call PREPEND,CXXFLAGS, CPPFLAGS CFLAGS)
CXXFLAGS := $(CPPFLAGS) $(CFLAGS) $(CXXFLAGS)
X86ASMFLAGS += $(IFLAGS:%=%/) -I$(<D)/ -Pconfig.asm
HOSTCCFLAGS = $(IFLAGS) $(HOSTCPPFLAGS) $(HOSTCFLAGS)
@@ -50,7 +44,7 @@ LDFLAGS := $(ALLFFLIBS:%=$(LD_PATH)lib%) $(LDFLAGS)
define COMPILE
$(call $(1)DEP,$(1))
$($(1)) $($(1)FLAGS) $($(2)) $($(1)_DEPFLAGS) $($(1)_C) $($(1)_O) $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<)
$($(1)) $($(1)FLAGS) $($(1)_DEPFLAGS) $($(1)_C) $($(1)_O) $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<)
endef
COMPILE_C = $(call COMPILE,CC)
@@ -60,22 +54,6 @@ COMPILE_M = $(call COMPILE,OBJCC)
COMPILE_X86ASM = $(call COMPILE,X86ASM)
COMPILE_HOSTC = $(call COMPILE,HOSTCC)
COMPILE_NVCC = $(call COMPILE,NVCC)
COMPILE_MMI = $(call COMPILE,CC,MMIFLAGS)
COMPILE_MSA = $(call COMPILE,CC,MSAFLAGS)
COMPILE_LSX = $(call COMPILE,CC,LSXFLAGS)
COMPILE_LASX = $(call COMPILE,CC,LASXFLAGS)
%_mmi.o: %_mmi.c
$(COMPILE_MMI)
%_msa.o: %_msa.c
$(COMPILE_MSA)
%_lsx.o: %_lsx.c
$(COMPILE_LSX)
%_lasx.o: %_lasx.c
$(COMPILE_LASX)
%.o: %.c
$(COMPILE_C)
@@ -104,7 +82,7 @@ COMPILE_LASX = $(call COMPILE,CC,LASXFLAGS)
-$(if $(ASMSTRIPFLAGS), $(STRIP) $(ASMSTRIPFLAGS) $@)
%.o: %.rc
$(WINDRES) $(IFLAGS) $(foreach ARG,$(CC_DEPFLAGS),--preprocessor-arg "$(ARG)") -o $@ $<
$(WINDRES) $(IFLAGS) --preprocessor "$(DEPWINDRES) -E -xc-header -DRC_INVOKED $(CC_DEPFLAGS)" -o $@ $<
%.i: %.c
$(CC) $(CCFLAGS) $(CC_E) $<
@@ -112,40 +90,16 @@ COMPILE_LASX = $(call COMPILE,CC,LASXFLAGS)
%.h.c:
$(Q)echo '#include "$*.h"' >$@
$(BIN2CEXE): ffbuild/bin2c_host.o
$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $^ $(HOSTEXTRALIBS)
%.metal.air: %.metal
$(METALCC) $< -o $@
%.metallib: %.metal.air
$(METALLIB) --split-module-without-linking $< -o $@
%.metallib.c: %.metallib $(BIN2CEXE)
$(BIN2C) $< $@ $(subst .,_,$(basename $(notdir $@)))
%.ptx: %.cu $(SRC_PATH)/compat/cuda/cuda_runtime.h
$(COMPILE_NVCC)
ifdef CONFIG_PTX_COMPRESSION
%.ptx.gz: TAG = GZIP
%.ptx.gz: %.ptx
$(M)gzip -nc9 $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<) >$@
%.ptx.c: %.ptx.gz $(BIN2CEXE)
$(BIN2C) $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<) $@ $(subst .,_,$(basename $(notdir $@)))
else
%.ptx.c: %.ptx $(BIN2CEXE)
$(BIN2C) $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<) $@ $(subst .,_,$(basename $(notdir $@)))
endif
clean::
$(RM) $(BIN2CEXE) $(CLEANSUFFIXES:%=ffbuild/%)
%.ptx.c: %.ptx
$(Q)sh $(SRC_PATH)/compat/cuda/ptx2c.sh $@ $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<)
%.c %.h %.pc %.ver %.version: TAG = GEN
# Dummy rule to stop make trying to rebuild removed or renamed headers
%.h %_template.c:
%.h:
@:
# Disable suffix rules. Most of the builtin rules are suffix rules,
@@ -160,8 +114,6 @@ include $(SRC_PATH)/ffbuild/arch.mak
OBJS += $(OBJS-yes)
SLIBOBJS += $(SLIBOBJS-yes)
SHLIBOBJS += $(SHLIBOBJS-yes)
STLIBOBJS += $(STLIBOBJS-yes)
FFLIBS := $($(NAME)_FFLIBS) $(FFLIBS-yes) $(FFLIBS)
TESTPROGS += $(TESTPROGS-yes)
@@ -170,8 +122,6 @@ FFEXTRALIBS := $(LDLIBS:%=$(LD_LIB)) $(foreach lib,EXTRALIBS-$(NAME) $(FFLIBS:%=
OBJS := $(sort $(OBJS:%=$(SUBDIR)%))
SLIBOBJS := $(sort $(SLIBOBJS:%=$(SUBDIR)%))
SHLIBOBJS := $(sort $(SHLIBOBJS:%=$(SUBDIR)%))
STLIBOBJS := $(sort $(STLIBOBJS:%=$(SUBDIR)%))
TESTOBJS := $(TESTOBJS:%=$(SUBDIR)tests/%) $(TESTPROGS:%=$(SUBDIR)tests/%.o)
TESTPROGS := $(TESTPROGS:%=$(SUBDIR)tests/%$(EXESUF))
HOSTOBJS := $(HOSTPROGS:%=$(SUBDIR)%.o)
@@ -193,7 +143,7 @@ HOBJS = $(filter-out $(SKIPHEADERS:.h=.h.o),$(ALLHEADERS:.h=.h.o))
PTXOBJS = $(filter %.ptx.o,$(OBJS))
$(HOBJS): CCFLAGS += $(CFLAGS_HEADERS)
checkheaders: $(HOBJS)
.SECONDARY: $(HOBJS:.o=.c) $(PTXOBJS:.o=.c) $(PTXOBJS:.o=.gz) $(PTXOBJS:.o=)
.SECONDARY: $(HOBJS:.o=.c) $(PTXOBJS:.o=.c) $(PTXOBJS:.o=)
alltools: $(TOOLS)
@@ -207,14 +157,12 @@ $(OBJS): | $(sort $(dir $(OBJS)))
$(HOBJS): | $(sort $(dir $(HOBJS)))
$(HOSTOBJS): | $(sort $(dir $(HOSTOBJS)))
$(SLIBOBJS): | $(sort $(dir $(SLIBOBJS)))
$(SHLIBOBJS): | $(sort $(dir $(SHLIBOBJS)))
$(STLIBOBJS): | $(sort $(dir $(STLIBOBJS)))
$(TESTOBJS): | $(sort $(dir $(TESTOBJS)))
$(TOOLOBJS): | tools
OUTDIRS := $(OUTDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(SLIBOBJS) $(SHLIBOBJS) $(STLIBOBJS) $(TESTOBJS))
OUTDIRS := $(OUTDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(SLIBOBJS) $(TESTOBJS))
CLEANSUFFIXES = *.d *.gcda *.gcno *.h.c *.ho *.map *.o *.pc *.ptx *.ptx.gz *.ptx.c *.ver *.version *$(DEFAULT_X86ASMD).asm *~ *.ilk *.pdb
CLEANSUFFIXES = *.d *.gcda *.gcno *.h.c *.ho *.map *.o *.pc *.ptx *.ptx.c *.ver *.version *$(DEFAULT_X86ASMD).asm *~ *.ilk *.pdb
LIBSUFFIXES = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a
define RULES
@@ -224,4 +172,4 @@ endef
$(eval $(RULES))
-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d) $(SHLIBOBJS:.o=.d) $(STLIBOBJS:.o=.d) $(SLIBOBJS:.o=.d)) $(OBJS:.o=$(DEFAULT_X86ASMD).d)
-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d) $(SLIBOBJS:.o=.d)) $(OBJS:.o=$(DEFAULT_X86ASMD).d)

View File

@@ -14,26 +14,10 @@ INSTHEADERS := $(INSTHEADERS) $(HEADERS:%=$(SUBDIR)%)
all-$(CONFIG_STATIC): $(SUBDIR)$(LIBNAME) $(SUBDIR)lib$(FULLNAME).pc
all-$(CONFIG_SHARED): $(SUBDIR)$(SLIBNAME) $(SUBDIR)lib$(FULLNAME).pc
LIBOBJS := $(OBJS) $(SHLIBOBJS) $(STLIBOBJS) $(SUBDIR)%.h.o $(TESTOBJS)
LIBOBJS := $(OBJS) $(SUBDIR)%.h.o $(TESTOBJS)
$(LIBOBJS) $(LIBOBJS:.o=.s) $(LIBOBJS:.o=.i): CPPFLAGS += -DHAVE_AV_CONFIG_H
ifdef CONFIG_SHARED
# In case both shared libs and static libs are enabled, it can happen
# that a user might want to link e.g. libavformat statically, but
# libavcodec and the other libs dynamically. In this case
# libavformat won't be able to access libavcodec's internal symbols,
# so that they have to be duplicated into the archive just like
# for purely shared builds.
# Test programs are always statically linked against their library
# to be able to access their library's internals, even with shared builds.
# Yet linking against dependend libraries still uses dynamic linking.
# This means that we are in the scenario described above.
# In case only static libs are used, the linker will only use
# one of these copies; this depends on the duplicated object files
# containing exactly the same symbols.
OBJS += $(SHLIBOBJS)
endif
$(SUBDIR)$(LIBNAME): $(OBJS) $(STLIBOBJS)
$(SUBDIR)$(LIBNAME): $(OBJS)
$(RM) $@
$(AR) $(ARFLAGS) $(AR_O) $^
$(RANLIB) $@
@@ -52,8 +36,8 @@ $(LIBOBJS): CPPFLAGS += -DBUILDING_$(NAME)
$(TESTPROGS) $(TOOLS): %$(EXESUF): %.o
$$(LD) $(LDFLAGS) $(LDEXEFLAGS) $$(LD_O) $$(filter %.o,$$^) $$(THISLIB) $(FFEXTRALIBS) $$(EXTRALIBS-$$(*F)) $$(ELIBS)
$(SUBDIR)lib$(NAME).version: $(SUBDIR)version.h $(SUBDIR)version_major.h | $(SUBDIR)
$$(M) $$(SRC_PATH)/ffbuild/libversion.sh $(NAME) $$^ > $$@
$(SUBDIR)lib$(NAME).version: $(SUBDIR)version.h | $(SUBDIR)
$$(M) $$(SRC_PATH)/ffbuild/libversion.sh $(NAME) $$< > $$@
$(SUBDIR)lib$(FULLNAME).pc: $(SUBDIR)version.h ffbuild/config.sh | $(SUBDIR)
$$(M) $$(SRC_PATH)/ffbuild/pkgconfig_generate.sh $(NAME) "$(DESC)"
@@ -64,7 +48,7 @@ $(SUBDIR)lib$(NAME).ver: $(SUBDIR)lib$(NAME).v $(OBJS)
$(SUBDIR)$(SLIBNAME): $(SUBDIR)$(SLIBNAME_WITH_MAJOR)
$(Q)cd ./$(SUBDIR) && $(LN_S) $(SLIBNAME_WITH_MAJOR) $(SLIBNAME)
$(SUBDIR)$(SLIBNAME_WITH_MAJOR): $(OBJS) $(SHLIBOBJS) $(SLIBOBJS) $(SUBDIR)lib$(NAME).ver
$(SUBDIR)$(SLIBNAME_WITH_MAJOR): $(OBJS) $(SLIBOBJS) $(SUBDIR)lib$(NAME).ver
$(SLIB_CREATE_DEF_CMD)
$$(LD) $(SHFLAGS) $(LDFLAGS) $(LDSOFLAGS) $$(LD_O) $$(filter %.o,$$^) $(FFEXTRALIBS)
$(SLIB_EXTRA_CMD)

View File

@@ -5,12 +5,8 @@ toupper(){
name=lib$1
ucname=$(toupper ${name})
file=$2
file2=$3
eval $(awk "/#define ${ucname}_VERSION_M/ { print \$2 \"=\" \$3 }" "$file")
if [ -f "$file2" ]; then
eval $(awk "/#define ${ucname}_VERSION_M/ { print \$2 \"=\" \$3 }" "$file2")
fi
eval ${ucname}_VERSION=\$${ucname}_VERSION_MAJOR.\$${ucname}_VERSION_MINOR.\$${ucname}_VERSION_MICRO
eval echo "${name}_VERSION=\$${ucname}_VERSION"
eval echo "${name}_VERSION_MAJOR=\$${ucname}_VERSION_MAJOR"

View File

@@ -9,27 +9,15 @@ AVBASENAMES = ffmpeg ffplay ffprobe
ALLAVPROGS = $(AVBASENAMES:%=%$(PROGSSUF)$(EXESUF))
ALLAVPROGS_G = $(AVBASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
OBJS-ffmpeg += \
fftools/ffmpeg_dec.o \
fftools/ffmpeg_demux.o \
fftools/ffmpeg_enc.o \
fftools/ffmpeg_filter.o \
fftools/ffmpeg_hw.o \
fftools/ffmpeg_mux.o \
fftools/ffmpeg_mux_init.o \
fftools/ffmpeg_opt.o \
fftools/ffmpeg_sched.o \
fftools/objpool.o \
fftools/sync_queue.o \
fftools/thread_queue.o \
OBJS-ffplay += fftools/ffplay_renderer.o
OBJS-ffmpeg += fftools/ffmpeg_opt.o fftools/ffmpeg_filter.o fftools/ffmpeg_hw.o
OBJS-ffmpeg-$(CONFIG_LIBMFX) += fftools/ffmpeg_qsv.o
ifndef CONFIG_VIDEOTOOLBOX
OBJS-ffmpeg-$(CONFIG_VDA) += fftools/ffmpeg_videotoolbox.o
endif
OBJS-ffmpeg-$(CONFIG_VIDEOTOOLBOX) += fftools/ffmpeg_videotoolbox.o
define DOFFTOOL
OBJS-$(1) += fftools/cmdutils.o fftools/opt_common.o fftools/$(1).o $(OBJS-$(1)-yes)
ifdef HAVE_GNU_WINDRES
OBJS-$(1) += fftools/fftoolsres.o
endif
OBJS-$(1) += fftools/cmdutils.o fftools/$(1).o $(OBJS-$(1)-yes)
$(1)$(PROGSSUF)_g$(EXESUF): $$(OBJS-$(1))
$$(OBJS-$(1)): | fftools
$$(OBJS-$(1)): CFLAGS += $(CFLAGS-$(1))

File diff suppressed because it is too large Load Diff

View File

@@ -44,16 +44,33 @@ extern const char program_name[];
*/
extern const int program_birth_year;
extern AVCodecContext *avcodec_opts[AVMEDIA_TYPE_NB];
extern AVFormatContext *avformat_opts;
extern AVDictionary *sws_dict;
extern AVDictionary *swr_opts;
extern AVDictionary *format_opts, *codec_opts;
extern AVDictionary *format_opts, *codec_opts, *resample_opts;
extern int hide_banner;
/**
* Register a program-specific cleanup routine.
*/
void register_exit(void (*cb)(int ret));
/**
* Wraps exit with a program-specific cleanup routine.
*/
void exit_program(int ret) av_noreturn;
/**
* Initialize dynamic library loading
*/
void init_dynload(void);
/**
* Initialize the cmdutils option system, in particular
* allocate the *_opts contexts.
*/
void init_opts(void);
/**
* Uninitialize the cmdutils option system, in particular
* free the *_opts contexts and their contents.
@@ -66,30 +83,37 @@ void uninit_opts(void);
*/
void log_callback_help(void* ptr, int level, const char* fmt, va_list vl);
/**
* Override the cpuflags.
*/
int opt_cpuflags(void *optctx, const char *opt, const char *arg);
/**
* Fallback for options that are not explicitly handled, these will be
* parsed through AVOptions.
*/
int opt_default(void *optctx, const char *opt, const char *arg);
/**
* Set the libav* libraries log level.
*/
int opt_loglevel(void *optctx, const char *opt, const char *arg);
int opt_report(void *optctx, const char *opt, const char *arg);
int opt_max_alloc(void *optctx, const char *opt, const char *arg);
int opt_codec_debug(void *optctx, const char *opt, const char *arg);
/**
* Limit the execution time.
*/
int opt_timelimit(void *optctx, const char *opt, const char *arg);
enum OptionType {
OPT_TYPE_FUNC,
OPT_TYPE_BOOL,
OPT_TYPE_STRING,
OPT_TYPE_INT,
OPT_TYPE_INT64,
OPT_TYPE_FLOAT,
OPT_TYPE_DOUBLE,
OPT_TYPE_TIME,
};
/**
* Parse a string and return its corresponding value as a double.
* Exit from the application if the string cannot be correctly
* parsed or the corresponding value is invalid.
*
* @param context the context of the value to be set (e.g. the
* corresponding command line option name)
@@ -99,8 +123,25 @@ enum OptionType {
* @param min the minimum valid accepted value
* @param max the maximum valid accepted value
*/
int parse_number(const char *context, const char *numstr, enum OptionType type,
double min, double max, double *dst);
double parse_number_or_die(const char *context, const char *numstr, int type,
double min, double max);
/**
* Parse a string specifying a time and return its corresponding
* value as a number of microseconds. Exit from the application if
* the string cannot be correctly parsed.
*
* @param context the context of the value to be set (e.g. the
* corresponding command line option name)
* @param timestr the string to be parsed
* @param is_duration a flag which tells how to interpret timestr, if
* not zero timestr is interpreted as a duration, otherwise as a
* date
*
* @see av_parse_time()
*/
int64_t parse_time_or_die(const char *context, const char *timestr,
int is_duration);
typedef struct SpecifierOpt {
char *specifier; /**< stream/chapter/program/... specifier */
@@ -114,70 +155,31 @@ typedef struct SpecifierOpt {
} u;
} SpecifierOpt;
typedef struct SpecifierOptList {
SpecifierOpt *opt;
int nb_opt;
/* Canonical option definition that was parsed into this list. */
const struct OptionDef *opt_canon;
enum OptionType type;
} SpecifierOptList;
typedef struct OptionDef {
const char *name;
enum OptionType type;
int flags;
/* The OPT_TYPE_FUNC option takes an argument.
* Must not be used with other option types, as for those it holds:
* - OPT_TYPE_BOOL do not take an argument
* - all other types do
*/
#define OPT_FUNC_ARG (1 << 0)
/* Program will immediately exit after processing this option */
#define OPT_EXIT (1 << 1)
/* Option is intended for advanced users. Only affects
* help output.
*/
#define OPT_EXPERT (1 << 2)
#define OPT_VIDEO (1 << 3)
#define OPT_AUDIO (1 << 4)
#define OPT_SUBTITLE (1 << 5)
#define OPT_DATA (1 << 6)
/* The option is per-file (currently ffmpeg-only). At least one of OPT_INPUT,
* OPT_OUTPUT, OPT_DECODER must be set when this flag is in use.
*/
#define OPT_PERFILE (1 << 7)
/* Option is specified as an offset in a passed optctx.
* Always use as OPT_OFFSET in option definitions. */
#define OPT_FLAG_OFFSET (1 << 8)
#define OPT_OFFSET (OPT_FLAG_OFFSET | OPT_PERFILE)
/* Option is to be stored in a SpecifierOptList.
Always use as OPT_SPEC in option definitions. */
#define OPT_FLAG_SPEC (1 << 9)
#define OPT_SPEC (OPT_FLAG_SPEC | OPT_OFFSET)
/* Option applies per-stream (implies OPT_SPEC). */
#define OPT_FLAG_PERSTREAM (1 << 10)
#define OPT_PERSTREAM (OPT_FLAG_PERSTREAM | OPT_SPEC)
/* ffmpeg-only - specifies whether an OPT_PERFILE option applies to input,
* output, or both. */
#define OPT_INPUT (1 << 11)
#define OPT_OUTPUT (1 << 12)
/* This option is a "canonical" form, to which one or more alternatives
* exist. These alternatives are listed in u1.names_alt. */
#define OPT_HAS_ALT (1 << 13)
/* This option is an alternative form of some other option, whose
* name is stored in u1.name_canon */
#define OPT_HAS_CANON (1 << 14)
/* ffmpeg-only - OPT_PERFILE may apply to standalone decoders */
#define OPT_DECODER (1 << 15)
#define HAS_ARG 0x0001
#define OPT_BOOL 0x0002
#define OPT_EXPERT 0x0004
#define OPT_STRING 0x0008
#define OPT_VIDEO 0x0010
#define OPT_AUDIO 0x0020
#define OPT_INT 0x0080
#define OPT_FLOAT 0x0100
#define OPT_SUBTITLE 0x0200
#define OPT_INT64 0x0400
#define OPT_EXIT 0x0800
#define OPT_DATA 0x1000
#define OPT_PERFILE 0x2000 /* the option is per-file (currently ffmpeg-only).
implied by OPT_OFFSET or OPT_SPEC */
#define OPT_OFFSET 0x4000 /* option is specified as an offset in a passed optctx */
#define OPT_SPEC 0x8000 /* option is to be stored in an array of SpecifierOpt.
Implies OPT_OFFSET. Next element after the offset is
an int containing element count in the array. */
#define OPT_TIME 0x10000
#define OPT_DOUBLE 0x20000
#define OPT_INPUT 0x40000
#define OPT_OUTPUT 0x80000
union {
void *dst_ptr;
int (*func_arg)(void *, const char *, const char *);
@@ -185,15 +187,6 @@ typedef struct OptionDef {
} u;
const char *help;
const char *argname;
union {
/* Name of the canonical form of this option.
* Is valid when OPT_HAS_CANON is set. */
const char *name_canon;
/* A NULL-terminated list of alternate forms of this option.
* Is valid when OPT_HAS_ALT is set. */
const char * const *names_alt;
} u1;
} OptionDef;
/**
@@ -203,9 +196,51 @@ typedef struct OptionDef {
* @param msg title of this group. Only printed if at least one option matches.
* @param req_flags print only options which have all those flags set.
* @param rej_flags don't print options which have any of those flags set.
* @param alt_flags print only options that have at least one of those flags set
*/
void show_help_options(const OptionDef *options, const char *msg, int req_flags,
int rej_flags);
int rej_flags, int alt_flags);
#if CONFIG_AVDEVICE
#define CMDUTILS_COMMON_OPTIONS_AVDEVICE \
{ "sources" , OPT_EXIT | HAS_ARG, { .func_arg = show_sources }, \
"list sources of the input device", "device" }, \
{ "sinks" , OPT_EXIT | HAS_ARG, { .func_arg = show_sinks }, \
"list sinks of the output device", "device" }, \
#else
#define CMDUTILS_COMMON_OPTIONS_AVDEVICE
#endif
#define CMDUTILS_COMMON_OPTIONS \
{ "L", OPT_EXIT, { .func_arg = show_license }, "show license" }, \
{ "h", OPT_EXIT, { .func_arg = show_help }, "show help", "topic" }, \
{ "?", OPT_EXIT, { .func_arg = show_help }, "show help", "topic" }, \
{ "help", OPT_EXIT, { .func_arg = show_help }, "show help", "topic" }, \
{ "-help", OPT_EXIT, { .func_arg = show_help }, "show help", "topic" }, \
{ "version", OPT_EXIT, { .func_arg = show_version }, "show version" }, \
{ "buildconf", OPT_EXIT, { .func_arg = show_buildconf }, "show build configuration" }, \
{ "formats", OPT_EXIT, { .func_arg = show_formats }, "show available formats" }, \
{ "muxers", OPT_EXIT, { .func_arg = show_muxers }, "show available muxers" }, \
{ "demuxers", OPT_EXIT, { .func_arg = show_demuxers }, "show available demuxers" }, \
{ "devices", OPT_EXIT, { .func_arg = show_devices }, "show available devices" }, \
{ "codecs", OPT_EXIT, { .func_arg = show_codecs }, "show available codecs" }, \
{ "decoders", OPT_EXIT, { .func_arg = show_decoders }, "show available decoders" }, \
{ "encoders", OPT_EXIT, { .func_arg = show_encoders }, "show available encoders" }, \
{ "bsfs", OPT_EXIT, { .func_arg = show_bsfs }, "show available bit stream filters" }, \
{ "protocols", OPT_EXIT, { .func_arg = show_protocols }, "show available protocols" }, \
{ "filters", OPT_EXIT, { .func_arg = show_filters }, "show available filters" }, \
{ "pix_fmts", OPT_EXIT, { .func_arg = show_pix_fmts }, "show available pixel formats" }, \
{ "layouts", OPT_EXIT, { .func_arg = show_layouts }, "show standard channel layouts" }, \
{ "sample_fmts", OPT_EXIT, { .func_arg = show_sample_fmts }, "show available audio sample formats" }, \
{ "colors", OPT_EXIT, { .func_arg = show_colors }, "show available color names" }, \
{ "loglevel", HAS_ARG, { .func_arg = opt_loglevel }, "set logging level", "loglevel" }, \
{ "v", HAS_ARG, { .func_arg = opt_loglevel }, "set logging level", "loglevel" }, \
{ "report", 0, { .func_arg = opt_report }, "generate a report" }, \
{ "max_alloc", HAS_ARG, { .func_arg = opt_max_alloc }, "set maximum size of a single allocated block", "bytes" }, \
{ "cpuflags", HAS_ARG | OPT_EXPERT, { .func_arg = opt_cpuflags }, "force specific cpu flags", "flags" }, \
{ "hide_banner", OPT_BOOL | OPT_EXPERT, {&hide_banner}, "do not show program banner", "hide_banner" }, \
CMDUTILS_COMMON_OPTIONS_AVDEVICE \
/**
* Show help for all options with given flags in class and all its
@@ -219,6 +254,11 @@ void show_help_children(const AVClass *class, int flags);
*/
void show_help_default(const char *opt, const char *arg);
/**
* Generic -h handler common to all fftools.
*/
int show_help(void *optctx, const char *opt, const char *arg);
/**
* Parse the command line arguments.
*
@@ -231,8 +271,8 @@ void show_help_default(const char *opt, const char *arg);
* argument without a leading option name flag. NULL if such arguments do
* not have to be processed.
*/
int parse_options(void *optctx, int argc, char **argv, const OptionDef *options,
int (* parse_arg_function)(void *optctx, const char*));
void parse_options(void *optctx, int argc, char **argv, const OptionDef *options,
void (* parse_arg_function)(void *optctx, const char*));
/**
* Parse one given option.
@@ -277,6 +317,7 @@ typedef struct OptionGroup {
AVDictionary *codec_opts;
AVDictionary *format_opts;
AVDictionary *resample_opts;
AVDictionary *sws_dict;
AVDictionary *swr_opts;
} OptionGroup;
@@ -307,7 +348,7 @@ typedef struct OptionParseContext {
*
* @param optctx an app-specific options context. NULL for global options group
*/
int parse_optgroup(void *optctx, OptionGroup *g, const OptionDef *defs);
int parse_optgroup(void *optctx, OptionGroup *g);
/**
* Split the commandline into an intermediate form convenient for further
@@ -370,12 +411,10 @@ int check_stream_specifier(AVFormatContext *s, AVStream *st, const char *spec);
* @param st A stream from s for which the options should be filtered.
* @param codec The particular codec for which the options should be filtered.
* If null, the default one is looked up according to the codec id.
* @param dst a pointer to the created dictionary
* @return a non-negative number on success, a negative error code on failure
* @return a pointer to the created dictionary
*/
int filter_codec_opts(const AVDictionary *opts, enum AVCodecID codec_id,
AVFormatContext *s, AVStream *st, const AVCodec *codec,
AVDictionary **dst);
AVDictionary *filter_codec_opts(AVDictionary *opts, enum AVCodecID codec_id,
AVFormatContext *s, AVStream *st, AVCodec *codec);
/**
* Setup AVCodecContext options for avformat_find_stream_info().
@@ -384,10 +423,12 @@ int filter_codec_opts(const AVDictionary *opts, enum AVCodecID codec_id,
* contained in s.
* Each dictionary will contain the options from codec_opts which can
* be applied to the corresponding stream codec context.
*
* @return pointer to the created array of dictionaries, NULL if it
* cannot be created
*/
int setup_find_stream_info_opts(AVFormatContext *s,
AVDictionary *codec_opts,
AVDictionary ***dst);
AVDictionary **setup_find_stream_info_opts(AVFormatContext *s,
AVDictionary *codec_opts);
/**
* Print an error message to stderr, indicating filename and a human
@@ -398,10 +439,7 @@ int setup_find_stream_info_opts(AVFormatContext *s,
*
* @see av_strerror()
*/
static inline void print_error(const char *filename, int err)
{
av_log(NULL, AV_LOG_ERROR, "%s: %s\n", filename, av_err2str(err));
}
void print_error(const char *filename, int err);
/**
* Print the program banner to stderr. The banner contents depend on the
@@ -410,6 +448,136 @@ static inline void print_error(const char *filename, int err)
*/
void show_banner(int argc, char **argv, const OptionDef *options);
/**
* Print the version of the program to stdout. The version message
* depends on the current versions of the repository and of the libav*
* libraries.
* This option processing function does not utilize the arguments.
*/
int show_version(void *optctx, const char *opt, const char *arg);
/**
* Print the build configuration of the program to stdout. The contents
* depend on the definition of FFMPEG_CONFIGURATION.
* This option processing function does not utilize the arguments.
*/
int show_buildconf(void *optctx, const char *opt, const char *arg);
/**
* Print the license of the program to stdout. The license depends on
* the license of the libraries compiled into the program.
* This option processing function does not utilize the arguments.
*/
int show_license(void *optctx, const char *opt, const char *arg);
/**
* Print a listing containing all the formats supported by the
* program (including devices).
* This option processing function does not utilize the arguments.
*/
int show_formats(void *optctx, const char *opt, const char *arg);
/**
* Print a listing containing all the muxers supported by the
* program (including devices).
* This option processing function does not utilize the arguments.
*/
int show_muxers(void *optctx, const char *opt, const char *arg);
/**
* Print a listing containing all the demuxer supported by the
* program (including devices).
* This option processing function does not utilize the arguments.
*/
int show_demuxers(void *optctx, const char *opt, const char *arg);
/**
* Print a listing containing all the devices supported by the
* program.
* This option processing function does not utilize the arguments.
*/
int show_devices(void *optctx, const char *opt, const char *arg);
#if CONFIG_AVDEVICE
/**
* Print a listing containing autodetected sinks of the output device.
* Device name with options may be passed as an argument to limit results.
*/
int show_sinks(void *optctx, const char *opt, const char *arg);
/**
* Print a listing containing autodetected sources of the input device.
* Device name with options may be passed as an argument to limit results.
*/
int show_sources(void *optctx, const char *opt, const char *arg);
#endif
/**
* Print a listing containing all the codecs supported by the
* program.
* This option processing function does not utilize the arguments.
*/
int show_codecs(void *optctx, const char *opt, const char *arg);
/**
* Print a listing containing all the decoders supported by the
* program.
*/
int show_decoders(void *optctx, const char *opt, const char *arg);
/**
* Print a listing containing all the encoders supported by the
* program.
*/
int show_encoders(void *optctx, const char *opt, const char *arg);
/**
* Print a listing containing all the filters supported by the
* program.
* This option processing function does not utilize the arguments.
*/
int show_filters(void *optctx, const char *opt, const char *arg);
/**
* Print a listing containing all the bit stream filters supported by the
* program.
* This option processing function does not utilize the arguments.
*/
int show_bsfs(void *optctx, const char *opt, const char *arg);
/**
* Print a listing containing all the protocols supported by the
* program.
* This option processing function does not utilize the arguments.
*/
int show_protocols(void *optctx, const char *opt, const char *arg);
/**
* Print a listing containing all the pixel formats supported by the
* program.
* This option processing function does not utilize the arguments.
*/
int show_pix_fmts(void *optctx, const char *opt, const char *arg);
/**
* Print a listing containing all the standard channel layouts supported by
* the program.
* This option processing function does not utilize the arguments.
*/
int show_layouts(void *optctx, const char *opt, const char *arg);
/**
* Print a listing containing all the sample formats supported by the
* program.
*/
int show_sample_fmts(void *optctx, const char *opt, const char *arg);
/**
* Print a listing containing all the color names and values recognized
* by the program.
*/
int show_colors(void *optctx, const char *opt, const char *arg);
/**
* Return a positive value if a line read from standard input
* starts with [yY], otherwise return 0.
@@ -439,31 +607,20 @@ FILE *get_preset_file(char *filename, size_t filename_size,
/**
* Realloc array to hold new_size elements of elem_size.
* Calls exit() on failure.
*
* @param array pointer to the array to reallocate, will be updated
* with a new pointer on success
* @param array array to reallocate
* @param elem_size size in bytes of each element
* @param size new element count will be written here
* @param new_size number of elements to place in reallocated array
* @return a non-negative number on success, a negative error code on failure
* @return reallocated array
*/
int grow_array(void **array, int elem_size, int *size, int new_size);
void *grow_array(void *array, int elem_size, int *size, int new_size);
/**
* Atomically add a new element to an array of pointers, i.e. allocate
* a new entry, reallocate the array of pointers and make the new last
* member of this array point to the newly allocated buffer.
*
* @param array array of pointers to reallocate
* @param elem_size size of the new element to allocate
* @param nb_elems pointer to the number of elements of the array array;
* *nb_elems will be incremented by one by this function.
* @return pointer to the newly allocated entry or NULL on failure
*/
void *allocate_array_elem(void *array, size_t elem_size, int *nb_elems);
#define media_type_string av_get_media_type_string
#define GROW_ARRAY(array, nb_elems)\
grow_array((void**)&array, sizeof(*array), &nb_elems, nb_elems + 1)
array = grow_array(array, sizeof(*array), &nb_elems, nb_elems + 1)
#define GET_PIX_FMT_NAME(pix_fmt)\
const char *name = av_get_pix_fmt_name(pix_fmt);
@@ -478,9 +635,14 @@ void *allocate_array_elem(void *array, size_t elem_size, int *nb_elems);
char name[16];\
snprintf(name, sizeof(name), "%d", rate);
double get_rotation(const int32_t *displaymatrix);
#define GET_CH_LAYOUT_NAME(ch_layout)\
char name[16];\
snprintf(name, sizeof(name), "0x%"PRIx64, ch_layout);
/* read file contents into a string */
char *file_read(const char *filename);
#define GET_CH_LAYOUT_DESC(ch_layout)\
char name[128];\
av_get_channel_layout_string(name, sizeof(name), 0, ch_layout);
double get_rotation(AVStream *st);
#endif /* FFTOOLS_CMDUTILS_H */

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