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251 Commits

Author SHA1 Message Date
Michael Niedermayer
fcbd117df3 Changelog: update
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-11-03 00:51:11 +01:00
Michael Niedermayer
b8aa7b9a6d avutil/integer: Fix integer overflow in av_mul_i()
Found-by: fate
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 3cc3cb663b)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-11-03 00:50:15 +01:00
Michael Niedermayer
d2ce6472a7 avcodec/msrle: Check that the input is large enough to contain a end of picture code
Fixes: Timeout
Fixes: 10625/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_MSRLE_fuzzer-5659651283091456

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 203ccb8746)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-11-03 00:50:15 +01:00
Marton Balint
ebc01c8f6d avformat/ftp: return AVERROR_EOF for EOF
Without this FTP just hangs on eof...

Signed-off-by: Marton Balint <cus@passwd.hu>
(cherry picked from commit 6a034adf75)
2018-11-01 23:12:19 +01:00
Marton Balint
00e7e4b188 avcodec/libx264: remove FF_CODEC_CAP_INIT_THREADSAFE flag
Libx264 uses strtok which is not thread safe. Strtok is used in
x264_param_default_preset in param_apply_tune in x264/common/base.c.
Therefore the flag must be removed.

x264 fixed the issue, once the fix is pushed to stable, an #if can be added
to re-enable the flag based on X264_BUILD number.

Fixes ticket #7446.

Signed-off-by: Marton Balint <cus@passwd.hu>
(cherry picked from commit b02490a497)
2018-11-01 23:12:03 +01:00
Michael Niedermayer
938bc919ea Prepare 4.0.3 release
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-23 01:26:15 +02:00
Michael Niedermayer
afebdc3ed7 avcodec/jpeg2000dec: Fix off by 1 error in JPEG2000_PGOD_CPRL handling
Fixes: assertion failure
Fixes: 10785/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_JPEG2000_fuzzer-5672160496975872

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 305e523105)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-23 01:00:10 +02:00
Michael Niedermayer
6e14fc4aa1 avcodec/mpeg4videodec: Fix typo in sprite delta check
Fixes: Integer overflow
Fixes: 10890/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_MPEG4_fuzzer-5636062181851136

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 b737317a88)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-23 01:00:10 +02:00
Michael Niedermayer
1061a2e2ef avcodec/h264_cavlc: Check mb_skip_run
Fixes: 10300/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_H264_fuzzer-6292205497483264
Fixes: signed integer overflow: -2147483648 - 1 cannot be represented in type 'int'

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 f72b9904fe)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-23 01:00:10 +02:00
Michael Niedermayer
65b0caf47d avcodec/ra144: Fix integer overflow in add_wav()
Fixes: signed integer overflow: -2144033225 + -5208934 cannot be represented in type 'int'
Fixes: 10633/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_RA_144_fuzzer-5679133791617024

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 c6282141cb)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-23 01:00:10 +02:00
Michael Niedermayer
bccddd7fcc avformat/utils: Never store negative values in last_IP_duration
Fixes: integer overflow compute_pkt_fields()
Fixes: compute_pkt_usan

Reported-by: Thomas Guilbert <tguilbert@chromium.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 079d1a7175)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-23 01:00:10 +02:00
Michael Niedermayer
b2e9d3da81 avformat/utils: Fix integer overflow in discontinuity check
Fixes: signed integer overflow: 7738135736989908991 - -7954308516317364223 cannot be represented in type 'long'
Fixes: find_stream_info_usan

Reported-by: Thomas Guilbert <tguilbert@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 4e19cfcfa3)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-23 01:00:10 +02:00
Aman Gupta
380bc8585c Revert "avcodec/cbs_h264: silence errors about end_of_seq nalus"
This reverts commit aec3daa8b4.

This is an incomplete fix, and did not quality for a backport.
2018-10-18 09:41:13 -07:00
Aman Gupta
8791a1e7de avcodec/cbs: ensure user_data is padded for GBC parsing
Fixes crash noticed in the cbs_userdata patchset.

====ERROR: AddressSanitizer: heap-buffer-overflow on address 0x609000026c89 at pc 0x00010725d37b bp 0x7ffeea04e750 sp 0x7ffeea04e748
READ of size 4 at 0x609000026c89 thread T0
    #0 0x10725d37a in ff_cbs_read_unsigned get_bits.h:274
    #1 0x1072d2767 in ff_cbs_read_a53_user_data cbs_misc_syntax_template.c:119
    #2 0x1078251a7 in h264_metadata_filter h264_metadata_bsf.c:595
    #3 0x105c1321d in output_packet ffmpeg.c:853

0x609000026c89 is located 1 bytes to the right of 8-byte region [0x609000026c80,0x609000026c88)
allocated by thread T0 here:
    #0 0x10aef08d7 in wrap_posix_memalign (libclang_rt.asan_osx_dynamic.dylib:x86_64h+0x578d7)
    #1 0x10aca95e6 in av_malloc mem.c:87
    #2 0x10ac545fe in av_buffer_allocz buffer.c:72
    #3 0x107263b27 in cbs_h264_read_nal_unit cbs_h264_syntax_template.c:722
    #4 0x10725b688 in cbs_read_fragment_content cbs.c:155

Signed-off-by: Aman Gupta <aman@tmm1.net>
(cherry picked from commit 41ed2c3849)
2018-10-16 11:54:52 -07:00
Aman Gupta
70d0d83d4d avcodec/cbs: fix crash in sei_pic_timestamp
Signed-off-by: Aman Gupta <aman@tmm1.net>
(cherry picked from commit b6c3a02740)
2018-10-16 11:54:42 -07:00
Aman Gupta
aec3daa8b4 avcodec/cbs_h264: silence errors about end_of_seq nalus
[ffmpeg] AVBSFContext: Decomposition unimplemented for unit 4 (type 10).

Signed-off-by: Aman Gupta <aman@tmm1.net>
(cherry picked from commit 64c50c0e97)
2018-10-16 11:54:39 -07:00
Timo Rothenpieler
7b23dd0f41 avcodec/cuviddec: properly take deinterlacing and display delay into account for buffer_full check
Signed-off-by: Timo Rothenpieler <timo@rothenpieler.org>
2018-10-14 18:14:18 +02:00
James Almer
967604fecf avcodec/h2645_parse: skip NALUs with no content after stripping all the trailing zeros
The GetBitContext is effectively empty in them.

Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit 9a09f4c54a)
2018-10-11 22:31:17 -07:00
Avi Halachmi (:avih)
3fb09dba40 configure: <fflib>_deps: validate, reduce sensitivity
- Allow to add deps in any order rather than "in linking order".
- Expand deps chains as required rather than just once.
- Validate that there are no cycles.
- Validate that [after expansion] deps are limited to other fflibs.
- Remove expectation for a specific output order of unique().

Previously when adding items to <fflib>_deps, developers were
required to add them in linking order. This can be awkward and
bug-prone, especially when a list is not empty, e.g. when adding
conditional deps.

It also implicitly expected unique() to keep the last instance of
recurring items such that these lists maintain their linking order
after removing duplicate items.

This patch mainly allows to add deps in any order by keeping just
one master list in linking order, and then reordering all the
<fflib>_deps lists to align with the master list order.

This master list is LIBRARY_LIST itself, where otherwise its order
doesn't matter.

The patch also removes a limit where these deps lists were expanded
only once. This could have resulted in incomplete expanded lists,
or forcing devs to add already-deducable deps to avoid this issue.

Note: it is possible to deduce the master list order automatically
from the deps lists, but in this case it's probably not worth the
added complexity, even if minor. Maintaining one list should be OK.

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-08 21:01:24 -03:00
Avi Halachmi (:avih)
a634da282b configure: speed up check_deps()
x4 - x25 faster.

check_deps() recursively enables/disables components, and its loop is
iterated nearly 6000 times. It's particularly slow in bash - currently
consuming more than 50% of configure runtime, and about 20% with other
shells.

This commit applies few local optimizations, most effective first:
- Use $1 $2 ... instead of pushvar/popvar, and same at enable_deep*
- Abort early in one notable case - empty deps, to avoid costly no-op.
- Smaller changes which do add up:
  - Handle ${cfg}_checking locally instead of via enable[d]/disable
  - ${cfg}_checking: test done before inprogress - x2 faster in 50%+
  - one eval instead of several at the empty-deps early abort path.

- The "actual work" part is unmodified - just its surroundings.

Biggest speedups (relative and absolute) are observed with bash.

Tested-by: Michael Niedermayer <michael@niedermayer.cc>
Tested-by: Helmut K. C. Tessarek <tessarek@evermeet.cx>
Tested-by: Dave Yeo <daveryeo@telus.net>
Tested-by: Reino Wijnsma <rwijnsma@xs4all.nl>
Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit 45499e557c)
2018-10-08 20:56:08 -03:00
Avi Halachmi (:avih)
df6e929e89 configure: speed up print_enabled_components()
x4 - x10 faster.

Inside print_enabled components, the filter_list case invokes sed
about 350 times to parse the same source file and extract different
info for each arg. This is never instant, and on systems where fork is
slow (notably MSYS2/Cygwin on windows) it takes many seconds.

Change it to use sed once on the source file and set env vars with the
parse results, then use these results inside the loop.

Additionally, the cases of indev_list and outdev_list are very
infrequent, but nevertheless they're faster, and arguably cleaner, with
shell parameter substitutions than with command substitutions.

Tested-by: Michael Niedermayer <michael@niedermayer.cc>
Tested-by: Helmut K. C. Tessarek <tessarek@evermeet.cx>
Tested-by: Dave Yeo <daveryeo@telus.net>
Tested-by: Reino Wijnsma <rwijnsma@xs4all.nl>
Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit 923586a58f)
2018-10-08 20:56:02 -03:00
Avi Halachmi (:avih)
42355d12db configure: speed up flatten_extralibs_wrapper()
x50 - x200 faster.

Currently configure spends 50-70% of its runtime inside a single
function: flatten_extralibs[_wrapper] - which does string processing.

During its run, nearly 20K command substitutions (subshells) are used,
including its callees unique() and resolve(), which is the reason
for its lengthy run.

This commit avoids all subshells during its execution, speeding it up
by about two orders of magnitude, and reducing the overall configure
runtime by 50-70% .

resolve() is rewritten to avoid subshells, and in unique() and
flatten_extralibs() we "inline" the filter[_out] functionality.

Note that logically, "unique" functionality has more than one possible
output (depending on which of the recurring items is kept). As it
turns out, other parts expect the last recurring item to be kept
(which was the original behavior of uniqie()). This patch preservs
its output order.

Tested-by: Michael Niedermayer <michael@niedermayer.cc>
Tested-by: Helmut K. C. Tessarek <tessarek@evermeet.cx>
Tested-by: Dave Yeo <daveryeo@telus.net>
Tested-by: Reino Wijnsma <rwijnsma@xs4all.nl>
Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit 58b81ac621)
2018-10-08 20:55:53 -03:00
Michael Niedermayer
eb495b20e5 avformat/utils: Fix potential integer overflow in extract_extradata()
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 0a41a8bf29)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:01 +02:00
Michael Niedermayer
7f521fae2b avcodec/unary: Improve get_unary() docs
Found-by: kierank
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit ad89e203bf)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
531ebb7506 avcodec/gdv: Replace divisions by shifts in rescale()
Divisions tend to be slower than shifts unless the compiler optimizes them out.
And some of these are in inner loops.

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit b90d8cc746)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
d7973cf03d avcodec/ac3dec: Fix shift signedness in mask creation
Fixes: 9924/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_EAC3_fuzzer-5473421772193792
Fixes: left shift of 1 by 63 places cannot be represented in type 'long 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 148a21611d)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
45fb50b4bc avcodec/eac3dec: Check that channel_map does not contain more than EAC3_MAX_CHANNELS
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit fe315feab5)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
f645fd64c3 doc/examples/vaapi_transcode: Fix the typo
Reviewed-by: "mypopy@gmail.com" <mypopy@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit c02ff77681)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
1a05e6ced3 avcodec/dvdsubdec: Sanity check len in decode_rle()
Fixes: Timeout
Fixes: 9778/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_DVDSUB_fuzzer-5186007132536832

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 e7b023e1db)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
5a4234de5e avcodec/mpeg4videodec: Fix undefined shift in get_amv()
Fixes: runtime error: shift exponent -1 is negative
Fixes: 9938/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_MPEG4_fuzzer-5653783529914368

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 c88afa44c4)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
47f608a7e1 avcodec/zmbv: Check that the decompressed data size is correct
This checks the value exactly for intra frames and checks it against a
minimum for inter frames as they can be variable.

Fixes: Timeout
Fixes: 10182/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_ZMBV_fuzzer-6245951174344704

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit e33b28cc79)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
5af88171e7 avcodec/zmbv: Update decomp_len in raw frames
decomp_len is used in raw frames, so it should not be left at the value from
whatever was decoded previously (which may be any other frame)

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 3d201b83cd)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
6d2b2ee3a5 avcodec/shorten: Fix bitstream end check in read_header()
Fixes: Timeout
Fixes: 9961/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SHORTEN_fuzzer-5687856176562176

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 28b80c2d52)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
9afcf994f6 avcodec/dvdsubdec: Avoid branch in decode_run_8bit()
Speed improvment 35.5 sec -> 34.7sec

Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 71bf033050)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
7da37aa980 avcodec/h264_refs: Document last if() in ff_h264_execute_ref_pic_marking()
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 697984b9db)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
318b13a5ad avcodec/ra144: Fix undefined integer overflow in add_wav()
Fixes: signed integer overflow: -26884 * 91439 cannot be represented in type 'int'
Fixes: 9687/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_RA_144_fuzzer-4995588121690112

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 93a203662f)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
17a5df4ab8 avcodec/indeo4: Check dimensions in decode_pic_hdr()
Fixes: Timeout
Fixes: 9654/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_INDEO4_fuzzer-6289863463665664

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 7592e88bfe)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Dale Curtis
968ffb93af avformat/mov: Error on too large stsd entry counts.
Entries are always at least 8 bytes per the parsing code, so if we
see an impossible entry count avoid massive allocations. This is
similar to an existing check in mov_read_stsc().

Since ff_mov_read_stsd_entries() does eof checks, an alternative
approach could be to clamp the entry count to atom.size / 8.

Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 320b631a99)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Bunk
84ab680624 examples: Fix use of AV_CODEC_FLAG_GLOBAL_HEADER
AV_CODEC_FLAG_GLOBAL_HEADER should be set before calling avcodec_open2() to have any effect.

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit a82e4fb8c6)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
8327559fe8 avcodec/hq_hqa: Check remaining input bits in hqa_decode_mb()
Fixes: Timeout
Fixes: 9634/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_HQ_HQA_fuzzer-6267852259590144

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 c9222b972d)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
229025799f avcodec/vb: Check for end of bytestream before reading blocktype
Fixes: Timeout
Fixes: 9601/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_VB_fuzzer-4550228702134272

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 1cbac9ce20)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
a04ff0c054 avcodec/snowdec: Fix integer overflow with motion vector residual
Fixes: signed integer overflow: -19818 + -2147483648 cannot be represented in type 'int'
Fixes: 9545/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SNOW_fuzzer-4928769537081344

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 acba153a14)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
ce828a247d avcodec/mpeg4videodec: Fix slice end detection in mpeg4_decode_studio_mb()
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 168d8d56bf)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
6e60a38322 avformat/nsvdec: Do not parse multiple NSVf
The specification states "NSV files may contain a single file header. "
Fixes: out of array access
Fixes: nsv-asan-002f473f726a0dcbd3bd53e422c4fc40b3cf3421

Found-by: Paul Ch <paulcher@icloud.com>
Tested-by: Paul Ch <paulcher@icloud.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 78d4b6bd43)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
f2b51fd54c avformat/dashdec: Fix strlen(rep_id_val) with it being NULL
Fixes: dash-crash-da39a3ee5e6b4b0d3255bfef95601890afd80709.xml

Found-by: Paul Ch <paulcher@icloud.com>
Reviewed-by: Steven Liu <lq@chinaffmpeg.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 46753bfdd0)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
0f36c5f5c7 avformat/mlvdec: read_string() received unsigned size, make the argument unsigned
Fixes: infinite loop
Fixes: mlv-timeout-e3b8cab9835edecad6823baa057e029671329d04

Found-by: Paul Ch <paulcher@icloud.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 1e71cb2c8e)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
3f095c5b56 avformat/rmdec: Fix EOF check in the stream loop in ivr_read_header()
Fixes: long running loop
Fixes: ivr-timeout-42468cb797f52f025fb329394702f5d4d64322d6

Found-by: Paul Ch <paulcher@icloud.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit c2eec1762d)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
b95c209a61 avcodec/scpr: Check for min > max in decompress_p()
Fixes: Timeout
Fixes: 9342/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SCPR_fuzzer-4795990841229312

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 3378194ce8)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
cd42c19f53 avcodec/shorten: Fix signed 32bit overflow in shift in shorten_decode_frame()
Fixes: runtime error: left shift of 1 by 31 places cannot be represented in type 'int'
Fixes: 9480/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SHORTEN_fuzzer-6647324284551168 -rss_limit_mb=2000

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 9b604e96a5)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
32353f8bcb avcodec/shorten: Fix integer overflow in residual/LPC combination
Fixes: signed integer overflow: -540538872 + -2012739576 cannot be represented in type 'int'
Fixes: 9255/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SHORTEN_fuzzer-5758630052757504

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 db7e9082e1)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
2c3ea34082 avcodec/shorten: Check verbatim length
Fixes: Timeout
Fixes: 9252/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SHORTEN_fuzzer-5780720709533696

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 7007dabec0)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
c0a6febf32 avcodec/mpegaudio_parser: Initialize poutbuf*
Possibly fixes: null pointer dereference
Possibly fixes: 9352/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_MP3ADUFLOAT_fuzzer-5146068961460224
Fixes: Heap-use-after-free
Fixes: 9453/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_MP3ADUFLOAT_fuzzer-5137954375729152

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 0f4c3b0b8e)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
cc0817af0d avcodec/aacpsdsp_template: Fix integer overflow in ps_stereo_interpolate_c()
Fixes: signed integer overflow: -1813244069 + -1407981383 cannot be represented in type 'int'
Fixes: 8823/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_AAC_FIXED_fuzzer-5643295618236416

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 47db5763e2)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
386975d7a4 avformat/flvenc: Check audio packet size
Fixes: Assertion failure
Fixes: assert_flvenc.c:941_1.swf

Found-by: #CHEN HONGXU# <HCHEN017@e.ntu.edu.sg>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 6b67d7f059)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Nikolas Bowe
8591d16ce5 lavc/svq3: Fix regression decoding some files.
Fixes some SVQ3 encoded files which fail to decode correctly after 6d6faa2a2d.
These files exhibit lots of artifacts and logs show "Media key encryption is not implemented".
However they decode without artifacts before 6d6faa2a2d.
The attatched patch allows these files to successfully decode, but also reject media key files.

Tested on the files in #6094 and http://samples.mplayerhq.hu/V-codecs/SVQ3/Vertical400kbit.sorenson3.mov

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 5aeb3b0080)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
07255282d0 avcodec/mlp_parser: Check if synccode is within buffer
Fixes: undefined shift
Fixes: 9216/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_TRUEHD_fuzzer-6281404575907840

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 51ac3f43b8)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
f9235773d6 avcodec/qtrle: Check remaining bytestream in qtrle_decode_XYbpp()
Fixes: Timeout
Fixes: 9213/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_QTRLE_fuzzer-5649753332252672

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 7dd836a3f9)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
4f51a21c30 avcodec/diracdec: Check bytes count in else branch in decode_lowdelay() too
Fixes: signed integer overflow: 8 * 340018243 cannot be represented in type 'int'
Fixes: 9441/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_DIRAC_fuzzer-5194665207791616

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 bed125b710)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
6cf72a56e7 avcodec/diracdec: Check slice numbers for overflows in relation to picture dimensions
Fixes: signed integer overflow: 88 * 33685506 cannot be represented in type 'int'
Fixes: 9433/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_DIRAC_fuzzer-5725943535501312

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 f457c0ad7f)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
88093d2c1f avcodec/diracdec: Change frame_number to 64bit as its a 32bit from the bitstream and we also have a -1 special case
Fixes: signed integer overflow: 2147483647 + 1 cannot be represented in type 'int'
Fixes: 9291/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_DIRAC_fuzzer-6324345860259840

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 462d1be6de)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
8147da2bad avcodec/dirac_dwt_template: Fix several integer overflows in horizontal_compose_daub97i()
Fixes: signed integer overflow: 2147483647 + 1 cannot be represented in type 'int'
Fixes: 8926/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_DIRAC_fuzzer-6047609228623872

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 69cac9e130)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
f291acafbb avcodec/diracdec: Prevent integer overflow in intermediate in global_mv()
Fixes: signed integer overflow: -393471 * 5460 cannot be represented in type 'int'
Fixes: 8890/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_DIRAC_fuzzer-6299775379963904

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 5129040646)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
cec6df48ba swresample/swresample: Fix input channel count in resample_first computation
Found-by: Marcin Gorzel <gorzel@google.com>
Reviewed-by: Marcin Gorzel <gorzel@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit bce4da85e8)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Michael Niedermayer
1a4a6d94cc avutil/pixfmt: Document chroma plane size for odd resolutions
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit be0b77e6e8)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-10-07 21:34:00 +02:00
Nicolas George
2be51cbeea lavf/libsmbclient: return AVERROR_EOF for EOF.
Fix trac ticket #7387.
2018-09-02 18:42:47 +02:00
Thilo Borgmann
49a90d5d31 lavc/videotoolboxenc: Fix compilation on osx 10.10.5 Yosemite
Signed-off-by: Aman Gupta <aman@tmm1.net>
(cherry picked from commit 72d9b8f4c5)
2018-08-14 10:02:05 -07:00
Aman Gupta
fab3418cb9 avcodec/mediacodecdec: fix SEGV on modern nvidia decoders
This code came originally from gstreamer, where it was added in [1]
as a work-around for the Tegra 3. (The alignment was changed in [2]
as a response to [3], from 32-bit to 16-bit).

gstreamer only used this workaround in the case where the decoder
didn't return a slice-height property, but when the code was copied
into avcodec the conditional got lost. This commit restores the guard
and prefers the slice-height from the decoder when it is available.

This fixes segfaults decoding 1920x1080 h264 and mpeg2 videos on the
NVidia SHIELD after upgrading to Android Oreo.

[1] a870e6a5c3
[2] 21ff3ae0b0
[3] https://bugzilla.gnome.org/show_bug.cgi?id=748867

Signed-off-by: Aman Gupta <aman@tmm1.net>
(cherry picked from commit 476fd6ba3a)
2018-08-03 11:07:31 -07:00
James Almer
9cc5337247 avcodec/bitstream_filters: check the input argument of av_bsf_get_by_name() for NULL
Fixes crashes like "ffmpeg -h bsf" caused by passing NULL to strcmp()

Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit 3258cc6507)
2018-07-28 22:43:09 -03:00
Timo Rothenpieler
d6d7853b4b avformat/librtmp: fix returning EOF from Read/Write
Ticket #7052
2018-07-28 01:11:30 +02:00
Thomas Guillem
db923b3fbd avcodec/videotoolboxenc: fix undefined behavior with rc_max_rate=0
On macOS, a zero rc_max_rate cause an error from
VTSessionSetProperty(kVTCompressionPropertyKey_DataRateLimits).

on iOS (depending on device/version), a zero rc_max_rate cause invalid
arguments from the vtenc_output_callback after few frames and then a crash
within the VideoToolbox library.

Signed-off-by: Aman Gupta <aman@tmm1.net>
(cherry picked from commit 93e157f40f)
2018-07-19 09:26:30 -07:00
Michael Niedermayer
0a155c57bd Update for 4.0.2
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-18 14:04:51 +02:00
Michael Niedermayer
3ef38c414e avcodec/dvdsub_parser: Allocate input padding
Fixes: out of array read
Fixes: 9350/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_DVDSUB_fuzzer-5746777750765568

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 cd86b5cfe2)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
40ed40902a avcodec/dvdsub_parser: Init output buf/size
No testcase

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 9e6c843776)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
0561cde128 avcodec/dirac_dwt_template: Fix signedness regression in interleave()
Found-by: <jdarnley>
Tested-by: James Darnley <james.darnley@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 181435a4de)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
670b565ba2 avformat/mov: Simplify last element computation in mov_estimate_video_delay()
Reviewed-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Reviewed-by: Sasi Inguva <isasi@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit b0644f7f72)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
6b65f46673 avformat/mov: Break out of inner loop early in mov_estimate_video_delay()
0.266 <- 0.299 sec (this is time ffmpeg so containing alot other things)

Sample for benchmark was: ffmpeg -f rawvideo -pix_fmt yuv420p -s 32x32 -i /dev/zero -t 24:00:00.00 out.mp4

Reviewed-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Reviewed-by: Sasi Inguva <isasi@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit aba13dc13e)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
052edeec55 avformat/mov: Eliminate variable buf_size from mov_estimate_video_delay()
Reviewed-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Reviewed-by: Sasi Inguva <isasi@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 3ce4034308)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
48479937c3 avformat/mov: remove modulo operations from mov_estimate_video_delay()
0.324 <-0.491 sec

Reviewed-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Reviewed-by: Sasi Inguva <isasi@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit c995e01b1e)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
fd53179f4a avformat/movenc: Write version 2 of audio atom if channels is not known
The version 1 needs the channel count and would divide by 0
Fixes: division by 0
Fixes: fpe_movenc.c_1108_1.ogg
Fixes: fpe_movenc.c_1108_2.ogg
Fixes: fpe_movenc.c_1108_3.wav

Found-by: #CHEN HONGXU# <HCHEN017@e.ntu.edu.sg>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit fa19fbcf71)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Rahul Chaudhry
5db47b3983 swresample/arm: rename labels to fix xcode build error
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit e84212b78e)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
0981dfee7d avformat/movenc: Check input sample count
Fixes: division by 0
Fixes: fpe_movenc.c_199_1.wav
Fixes: fpe_movenc.c_199_2.wav
Fixes: fpe_movenc.c_199_3.wav
Fixes: fpe_movenc.c_199_4.wav
Fixes: fpe_movenc.c_199_5.wav
Fixes: fpe_movenc.c_199_6.wav
Fixes: fpe_movenc.c_199_7.wav

Found-by: #CHEN HONGXU# <HCHEN017@e.ntu.edu.sg>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 3a2d21bc5f)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
d8c4b2ae57 avcodec/mjpegdec: Check for odd progressive RGB
Fixes: out of array access
Fixes: 9225/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_JPEGLS_fuzzer-5684770334834688

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 ee1e3ca5eb)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
fc92ca5b8e avcodec/vp8_parser: Do not leave data/size uninitialized
This is identical to what the VP9 parser does

Fixes: 9215/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_LIBVPX_VP8_fuzzer-5768227253649408
Fixes: out of memory access

This may also fix oss fuzz issue 9212

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 284dde24da)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
6d992a51c7 avformat/mms: Add missing chunksize check
Fixes: out of array read
Fixes: mms-crash-01b6c5d85f9d9f40f4e879896103e9f5b222816a

Found-by: Paul Ch <paulcher@icloud.com>
1st hunk by Paul Ch <paulcher@icloud.com>
Tested-by: Paul Ch <paulcher@icloud.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit cced03dd66)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
6f4b82cc3a avformat/pva: Check for EOF before retrying in read_part_of_packet()
Fixes: Infinite loop
Fixes: pva-4b1835dbc2027bf3c567005dcc78e85199240d06

Found-by: Paul Ch <paulcher@icloud.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 9807d3976b)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
37f505cc85 avformat/rmdec: Do not pass mime type in rm_read_multi() to ff_rm_read_mdpr_codecdata()
Fixes: use after free()
Fixes: rmdec-crash-ffe85b4cab1597d1cfea6955705e53f1f5c8a362

Found-by: Paul Ch <paulcher@icloud.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit a7e032a277)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
a21703ca5d avformat/asfdec_o: Check size_bmp more fully
Fixes: integer overflow and out of array access
Fixes: asfo-crash-46080c4341572a7137a162331af77f6ded45cbd7

Found-by: Paul Ch <paulcher@icloud.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 2b46ebdbff)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
a28ab09e2a avformat/mxfdec: Fix av_log context
Fixes: out of array access
Fixes: mxf-crash-1c2e59bf07a34675bfb3ada5e1ec22fa9f38f923

Found-by: Paul Ch <paulcher@icloud.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit bab0716c7f)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
4439d6aa69 avcodec/mpeg4videodec: Check for bitstream end in read_quant_matrix_ext()
Fixes: out of array read
Fixes: asff-crash-0e53d0dc491dfdd507530b66562812fbd4c36678

Found-by: Paul Ch <paulcher@icloud.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 5aba5b89d0)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
3bf80c7b22 avcodec/indeo4: Check for end of bitstream in decode_mb_info()
Fixes: Timeout
Fixes: 8776/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_INDEO4_fuzzer-5361788798369792

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 267ba2aa96)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
1361e4abb8 avcodec/ac3dec: Check channel_map index
Fixes: out of array read
Fixes: 8924/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_EAC3_fuzzer-5851861780267008

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 00f98d23b1)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
5fd1dce39a avcodec/mpeg4videodec: Remove use of FF_PROFILE_MPEG4_SIMPLE_STUDIO as indicator of studio profile
The profile field is changed by code inside and outside the decoder,
its not a reliable indicator of the internal codec state.
Maintaining it consistency with studio_profile is messy.
Its easier to just avoid it and use only studio_profile

Fixes: assertion failure
Fixes: ffmpeg_crash_9.avi

Found-by: Thuan Pham, Marcel Böhme, Andrew Santosa and Alexandru Razvan Caciulescu with AFLSmart
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit bd27a9364c)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
de0a1d01ba avcodec/shorten: Fix undefined addition in shorten_decode_frame()
Fixes: signed integer overflow: 1139785606 + 1454196085 cannot be represented in type 'int'
Fixes: 8937/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SHORTEN_fuzzer-6202943597445120

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 3b10bb8772)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
c4b23793d4 avcodec/shorten: Fix undefined integer overflow
Fixes: signed integer overflow: 8454144 * 256 cannot be represented in type 'int'
Fixes: 8788/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SHORTEN_fuzzer-5728205041303552

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 70832333bb)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
e21e5c95c1 avcodec/jpeg2000dec: Fixes invalid shifts in jpeg2000_decode_packets_po_iteration()
Fixes: shift exponent 47 is too large for 32-bit type 'int'
Fixes: 9163/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_JPEG2000_fuzzer-5661750182543360

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 652d7c6348)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
2b13c136c4 avcodec/jpeg2000dec: Check that there are enough bytes for all tiles
Fixes: OOM
Fixes: 8781/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_JPEG2000_fuzzer-5810709081358336

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 0898a3d990)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
d3536ce839 avformat/movenc: Use mov->fc consistently for av_log()
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 872ea3dfe5)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
679d749eab avcodec/mpeg4videodec: Check read profile before setting it
Fixes: null pointer dereference
Fixes: ffmpeg_crash_7.avi

Found-by: Thuan Pham, Marcel Böhme, Andrew Santosa and Alexandru Razvan Caciulescu with AFLSmart
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 2aa9047486)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
7610538224 avformat/movenc: Do not pass AVCodecParameters in avpriv_request_sample
Fixes: out of array read
Fixes: ffmpeg_crash_8.avi

Found-by: Thuan Pham, Marcel Böhme, Andrew Santosa and Alexandru Razvan Caciulescu with AFLSmart
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 95556e27e2)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
0003ace83b avcodec/ac3_parser: Check init_get_bits8() for failure
Fixes: null pointer dereference
Fixes: ffmpeg_crash_6.avi

Found-by: Thuan Pham, Marcel Böhme, Andrew Santosa and Alexandru Razvan Caciulescu with AFLSmart
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 00e8181bd9)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
20c5fb9721 avformat/movenc: Check that frame_types other than EAC3_FRAME_TYPE_INDEPENDENT have a supported substream id
Fixes: out of array access
Fixes: ffmpeg_bof_1.avi

Found-by: Thuan Pham, Marcel Böhme, Andrew Santosa and Alexandru Razvan Caciulescu with AFLSmart
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit ed22dc2221)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
841e1399e6 avcodec/dpx: Check elements in 12bps planar path
Fixes: null pointer dereference
Fixes: 8946/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_DPX_fuzzer-5078915222601728

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Carl Eugen Hoyos <ceffmpeg@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 75a2db5524)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
9f76f0fab8 avcodec/escape124: Fix spelling errors in comment
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit f59c4e4391)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
bf3e331b76 avcodec/ra144: Fix integer overflow in ff_eval_refl()
Fixes: signed integer overflow: -4096 * -524288 cannot be represented in type 'int'
Fixes: 8650/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_RA_144_fuzzer-5734816036159488

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 b31189881a)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
21732c1adc avcodec/cscd: Check output buffer size for lzo.
Fixes: Timeout
Fixes: 8665/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_CSCD_fuzzer-5768442610188288

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
(cherry picked from commit 78167b498f)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
d00548f2c1 avcodec/escape124: Check buf_size against num_superblocks
Fixes: Timeout
Fixes: 8722/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_ESCAPE124_fuzzer-4843268402577408

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 6677c98626)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
10e5302db4 avcodec/h264_parser: Reduce needed history for parsing mb index
This fixes a bug/regression with very small packets
Fixes: output_file

Regression since: 0782fb6bcb

Reported-by: Thierry Foucu <tfoucu@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit d25c945247)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
84280dc7cf avcodec/magicyuv: Check bits left in flags&1 branch
Fixes: Timeout
Fixes: 8690/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_MAGICYUV_fuzzer-6542020913922048

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 7719b8ccc7)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Michael Niedermayer
2c404cc11a avcodec/mjpegdec: Check for end of bitstream in ljpeg_decode_rgb_scan()
Fixes: Timeout
Fixes: 8648/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_MJPEG_fuzzer-5108395525799936

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 540e8c2d64)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-07-16 19:02:12 +02:00
Marton Balint
acb7907319 ffmpeg: fix -stream_loop with multiple inputs
The input thread needs to be properly cleaned up and re-initalized before we
can start reading again in threaded mode. (Threaded input reading is used when
there is mode than one input file).

Fixes ticket #6121 and #7043.

Signed-off-by: Marton Balint <cus@passwd.hu>
(cherry picked from commit da36bcbeb7)
2018-07-08 12:52:49 +02:00
Marton Balint
2c138c2d8c ffmpeg: factorize input thread creation and destruction
Signed-off-by: Marton Balint <cus@passwd.hu>
(cherry picked from commit b181cd359b)
2018-07-08 12:52:40 +02:00
Aman Gupta
b5106c5aa2 avformat/mpegts: parse large PMTs with multiple tables
In 9152c1e495, the mpegts parser was taught how to parse
PMT sections which contained multiple tables. That commit
fixed parsing of PMT packets from some cable providers,
which included a special SCTE table (0xc0) before the
standard program map table (0x2).

Sometimes, however, the combined 0xc0 and 0x2 tables are
larger than a single TS packet (188 bytes). The mpegts parser
already attempts to parse sections which span multiple packets,
but still assumed that the split section only contained one
table.

This patch fixes parsing of such a sample[1].

Before:

    Input #0, mpegts, from 'combined-pmt-tids-split.ts':
      Duration: 00:00:01.26, start: 39188.931756, bitrate: 597 kb/s
      Program 1
      No Program
        Stream #0:0[0xeff]: Audio: ac3, 48000 Hz, mono, fltp, 64 kb/s
        Stream #0:1[0xefd]: Audio: mp3, 0 channels, fltp
        Stream #0:2[0xefe]: Unknown: none

After:

    Input #0, mpegts, from 'combined-pmt-tids-split.ts':
      Duration: 00:00:01.27, start: 39188.931756, bitrate: 589 kb/s
      Program 1
        Stream #0:0[0xefd]: Video: h264 ([27][0][0][0] / 0x001B), none, 59.94 fps, 59.94 tbr, 90k tbn, 180k tbc
        Stream #0:1[0xefe](eng): Audio: ac3 ([129][0][0][0] / 0x0081), 48000 Hz, stereo, fltp, 384 kb/s
        Stream #0:2[0xeff](spa): Audio: ac3 ([129][0][0][0] / 0x0081), 48000 Hz, mono, fltp, 64 kb/s
        Stream #0:3[0xf00]: Data: scte_35
        Stream #0:4[0xf01]: Unknown: none (ETV1 / 0x31565445)
        Stream #0:5[0xf02]: Unknown: none (ETV1 / 0x31565445)
        Stream #0:6[0xf03]: Unknown: none ([192][0][0][0] / 0x00C0)

With the patch, the PMT is parsed correctly so the streams are
created in the correct order, are associated with "Program 1",
and their codecs are set correctly.

[1] https://s3.amazonaws.com/tmm1/combined-pmt-tids-split.ts

Signed-off-by: Aman Gupta <aman@tmm1.net>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit cd86c5dbcc)
2018-06-18 12:05:41 -07:00
Aman Gupta
789bac72ed Revert "avcodec/mediacodecdec: wait on first frame after input buffers are full"
@xyz reported a regression on his Sony Xperia Z3 Tablet Compact where
playback would intermittently fail to start, essentially deadlocking in
the decoder. Bisecting narrowed down the issue to this commit, which was
meant as an optimization but is not necessary.

This reverts commit a75bb5496a.

Signed-off-by: Aman Gupta <aman@tmm1.net>
(cherry picked from commit 37c2cb6a68)
2018-06-18 12:01:12 -07:00
Thomas Guillem
33fcbb4372 avcodec/videotoolboxenc: fix invalid session on iOS
Cf. comment. Restart the VT session when the APP goes from foreground to
background and vice versa.

Signed-off-by: Aman Gupta <aman@tmm1.net>
(cherry picked from commit 513e6a30fb)
2018-06-18 11:49:38 -07:00
Thomas Guillem
a56eb4d56c avcodec/videotoolboxenc: split initialization
Split vtenc_init() into vtenc_init() (VTEncContext initialization) and
vtenc_configure_encoder() (creates the vt session).

This commit will allow to restart the vt session while encoding.

Signed-off-by: Aman Gupta <aman@tmm1.net>
(cherry picked from commit 9e11d27c25)
2018-06-18 11:49:38 -07:00
Thomas Guillem
70799fae35 avcodec/videotoolboxenc: fix mutex/cond leak in error path
The leak could happen when the vtenc_create_encoder() function failed.

Signed-off-by: Aman Gupta <aman@tmm1.net>
(cherry picked from commit ce2330bdf8)
2018-06-18 11:49:38 -07:00
Michael Niedermayer
e049f7c24f Update for 4.0.1
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-16 00:18:40 +02:00
Michael Niedermayer
83a737aa70 avcodec/aacdec_fixed: Fix undefined integer overflow in apply_independent_coupling_fixed()
Fixes: signed integer overflow: 1195517 * 2048 cannot be represented in type 'int'
Fixes: 8636/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_AAC_FIXED_fuzzer-4695836326887424

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 8bd514d934)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
2deeb2eaef avcodec/dirac_dwt_template: Fix undefined behavior in interleave()
Fixes: signed integer overflow: 2147483647 + 1 cannot be represented in type 'int'
Fixes: 8697/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_DIRAC_fuzzer-5197148130902016

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 575d8ca026)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
9eaf908897 avutil/common: Fix undefined behavior in av_clip_uintp2_c()
Fixes: negation of -2147483648 cannot be represented in type 'int'; cast to an unsigned type to negate this value to itself
Fixes: 8521/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_DIRAC_fuzzer-5639024952737792

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 aa41d322be)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
1ca157b026 fftools/ffmpeg: Fallback to duration if sample rate is unavailable
Regression since: af1761f7
Fixes: Division by 0
Fixes: ffmpeg_crash_1

Found-by: Thuan Pham, Marcel Böhme, Andrew Santosa and Alexandru Razvan Caciulescu with AFLSmart
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 16d8b13b3b)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
f36128518b avformat/mov: Only set pkt->duration to non negative values
Reviewed-by: Sasi Inguva <isasi@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 8176799f31)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
897524954b avcodec/mpeg4videodec: Clear bits_per_raw_sample if it has originated from a previous instance
Fixes: assertion failure
Fixes: ffmpeg_crash_5.avi

Found-by: Thuan Pham <thuanpv@comp.nus.edu.sg>, Marcel Böhme, Andrew Santosa and Alexandru Razvan Caciulescu with AFLSmart
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 2fc108f60f)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Timo Teräs
f0e4bc61e3 avformat/movenc: fix recognization of cover image streams
For chapter images, the mov demux produces streams with disposition set
to attached_pic+timed_thumbnails. This patch fixes to properly recognize
streams that should be encoded as cover image (ones with only and only
attached_pic disposition set).

Signed-off-by: Timo Teräs <timo.teras@iki.fi>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 2223811b01)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Timo Teräs
36628bd215 avformat/movenc: properly handle cover image codecs
Find codec tag for attached images using appropriate list of
supported image formats.

This fixes writing the cover image to m4v/m4a and other container
formats that do not allow these codecs as a track.

Signed-off-by: Timo Teräs <timo.teras@iki.fi>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 12205d2c89)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
e42ab0115e avcodec/h264_slice: Fix overflow in recovery_frame computation
Fixes: signed integer overflow: 15 + 2147483646 cannot be represented in type 'int'
Fixes: 8381/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_H264_fuzzer-6225533137321984

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 8c20ea8ee0)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
10f68641ae avcodec/h264_ps: Move MAX_LOG2_MAX_FRAME_NUM to header so it can be used in h264_sei
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit b796c5ae92)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
f0a10f6376 avcodec/h264_mc_template: Only prefetch motion if the list is used.
Fixes: index 59 out of bounds for type 'H264Ref [48]'
Fixes: 8232/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_H264_fuzzer-5703295145345024

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 8b55591757)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
0b4d76d891 avcodec/xwddec: Use ff_set_dimensions()
Fixes: OOM
Fixes: 8178/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_XWD_fuzzer-4844793342459904

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit c2852e4e00)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
4fa2078217 avcodec/wavpack: Fix overflow in adding tail
Fixes: signed integer overflow: 2146907204 + 26846088 cannot be represented in type 'int'
Fixes: 8105/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_WAVPACK_fuzzer-6233036682166272

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 d13379fb79)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
081874a050 avcodec/shorten: Fix multiple integer overflows
Fixes: signed integer overflow: 3 * 1006632960 cannot be represented in type 'int'
Fixes: 8278/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SHORTEN_fuzzer-5692857166856192

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 f2abd36b38)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
028a0c9148 avcodec/shorten: Fix undefined shift in fix_bitshift()
Fixes: left shift of negative value -9
Fixes: 8571/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SHORTEN_fuzzer-5715966875926528

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 606c714823)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
9a53e8572a avcodec/shorten: Fix a negative left shift in shorten_decode_frame()
Fixes: left shift of negative value -9057
Fixes: 8527/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SHORTEN_fuzzer-5666853924896768

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 a711efe922)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
3aa3b05d64 avcodec/shorten: Sanity check nmeans
Fixes: OOM
Fixes: 8195/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SHORTEN_fuzzer-5179785826271232

The reference software appears to use longs for 32bits and it uses int for nmeans
hinting that the intended maximum size was not 32bit.

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 d91a0b503d)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
7a5ddf731b avcodec/shorten: Check non COMM chunk len before skip in decode_aiff_header()
Fixes: signed integer overflow: 2147483647 + 1 cannot be represented in type 'int'
Fixes: 8024/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SHORTEN_fuzzer-5109204648984576

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 424a81df10)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
4b12afccb2 avcodec/mjpegdec: Fix integer overflow in ljpeg_decode_rgb_scan()
Fixes: signed integer overflow: 32768 + 2147450880 cannot be represented in type 'int'
Fixes: 7885/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_THP_fuzzer-5298834394578944

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 936f4a2c2e)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
b55c824ee7 avcodec/truemotion2: Fix overflow in tm2_apply_deltas()
Fixes: signed integer overflow: 1077952576 + 1077952576 cannot be represented in type 'int'
Fixes: 7712/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_TRUEMOTION2_fuzzer-5056281753681920

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 79c6047c36)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
a90497c183 avcodec/opus_silk: Change silk_lsf2lpc() slightly toward silk/NLSF2A.c
Fixes: runtime error: signed integer overflow: -1440457022 - 785819492 cannot be represented in type 'int'
Fixes: 7700/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_OPUS_fuzzer-6595838684954624

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 e7dda51150)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
65b2b0d98a avcodec/amrwbdec: Fix division by 0 in find_hb_gain()
This restructures the code slightly toward D_UTIL_dec_synthesis()

Fixes: 7420/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_AMRWB_fuzzer-6577305112543232

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 dce80a4b47)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
0097cc0ea3 avcodec/h263dec: Reinitialize idct context if it has not been setup for the active profile
The profile after reading headers can be different from when the context was initialized

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 44a2415a6d)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
07767c704b avcodec/idctdsp: Clear idct/idct_add for studio profile
This does not leave them "as before" which may be a value from a previous profile

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 8c50d0cccf)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
2e7830e5ff avformat/mov: replace a value error by clipping into valid range in mov_read_stsc()
Fixes: #7165

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit fe84f70819)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
4f644b2632 avformat/bintext: Reduce detection for random .bin files as it more likely is not a multimedia related file
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 919e37377a)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
1d01a3b34c avformat/mov: Break out early if chunk_count is 0 in mov_build_index()
Without this some operations might overflow (undefined behavior)
even though the index adding loop would never execute

No testcase known

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 56e76bd057)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
2742cb10c7 avcodec/fic: Avoid some magic numbers related to cursors
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit c6a11714c4)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
8229afc3a9 avcodec/mpeg4video: Detect reference studio streams as studio streams
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit ba97d75ac6)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
76f8c8cd05 avcodec/mpeg4videodec: Do not corrupt bits_per_raw_sample
Reviewed-by: Kieran Kunhya <kierank@obe.tv>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 9e5d0860c0)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
786834a693 avcodec/mpeg4videode: Eliminate out of loop VOP startcode reading for studio profile
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 9f73ae31e0)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
f2c253f083 avcodec/g2meet: ask for sample with overflowing RGB
Suggested-by: Tomas Härdin <tjoppen@acc.umu.se>

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit ab834b8f36)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
e26be20a27 avcodec/idctdsp: Transmit studio_profile to init instead of using AVCodecContext profile
These 2 fields are not always the same, it is simpler to always use the same field
for detecting studio profile

Fixes: null pointer dereference
Fixes: ffmpeg_crash_3.avi

Found-by: Thuan Pham <thuanpv@comp.nus.edu.sg>, Marcel Böhme, Andrew Santosa and Alexandru RazvanCaciulescu with AFLSmart
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit b3332a182f)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
7b7c582c15 avcodec/ac3dec: Check that the number of channels with dependant streams is valid
Fixes: left shift of 1 by 63 places cannot be represented in type 'long long'
Fixes: out of array access
Fixes: 7284/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_AC3_fuzzer-5767914968842240

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 e3275f937d)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
06b84f7271 avcodec/ac3dec: Fix null pointer dereference in ac3_decode_frame()
Fixes: index 8 out of bounds for type 'uint8_t *[8]'
Fixes: 7273/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_EAC3_fuzzer-6296497667702784

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 e3f656f2de)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
f974cc9830 avcodec/aacdec_fixed: use 64bit to avoid overflow in rounding in apply_dependent_coupling_fixed()
Fixes: signed integer overflow: -2141499320 + -14469590 cannot be represented in type 'int'
Fixes: 7351/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_AAC_FIXED_fuzzer-6351214791884800

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 90475db97e)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
af0ba288e7 oavcodec/aacpsdsp_template: Use unsigned for hs0X to prevent undefined behavior
Fixes: signed integer overflow: 1073741842 + 1784008138 cannot be represented in type 'int'
Fixes: 6792/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_AAC_FIXED_fuzzer-5677589835284480

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 62cb6fadf3)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
c98d84e229 avcodec/g723_1dec: Clip bits2 in both directions
Fixes: shift exponent 33 is too large for 32-bit type 'int'
Fixes: 6743/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_G723_1_fuzzer-5823772687859712

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 53f241218d)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
b3d740263c avcodec/mpeg4videoenc: Use 64 bit for times in mpeg4_encode_gop_header()
Fixes truncation
Fixes Assertion n <= 31 && value < (1U << n) failed at libavcodec/put_bits.h:169
Fixes: ffmpeg_crash_2.avi

Found-by: Thuan Pham <thuanpv@comp.nus.edu.sg>, Marcel Böhme, Andrew Santosa and Alexandru RazvanCaciulescu with AFLSmart

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit e1182fac1a)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
6edf0ecab0 avcodec/mlpdec: Only change noise_type if the related fields are valid
Fixes: inconsistency
Fixes:runtime error: index 8 out of bounds for type 'int32_t [8]'
Fixes: 6686/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_TRUEHD_fuzzer-5191383498358784

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 63c4a4b0d6)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
b2aaf5de42 indeo4: Decode all or nothing of a band header.
This avoids inconsistent value combinations.
Alternatively it would be possible to add more checks and careful use of
temporary variables, but my try of this quickly seemed to become
a rather large change.
The disadvantage of this, is that the struct is copied back and forth.

Fixes: index 6 out of bounds for type 'const uint16_t [5][16]'
Fixes: 6557/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_INDEO4_fuzzer-4787296550256640

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 10c8521265)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
217367b5eb avcodec/ac3dec: Use frame_size if superframe_size is 0
Fixes: Infinite loop
Fixes: 7669/clusterfuzz-testcase-ffmpeg_AV_CODEC_ID_AC3_FIXED_fuzzer-4689042185650176
Fixes: 7670/clusterfuzz-testcase-ffmpeg_AV_CODEC_ID_AC3_fuzzer-4706306762997760
Fixes: 7672/clusterfuzz-testcase-ffmpeg_AV_CODEC_ID_EAC3_fuzzer-4702108499574784

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 f77eee67e2)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
c97f9ed53f avformat/mov: Only fail for STCO/STSC contradictions if both exist
Fixes regression with playback of GF9720Repeal20the20Eighth20with20Helen20Linehan.m4a
See: crbug 822666

Found-by: "Mattias Wadman <mattias.wadman@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 2c2d689c56)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
05ac7fdeeb avcodec/dirac_dwt: Fix integer overflow in COMPOSE_DD97iH0 / COMPOSE_DD137iL0
Fixes: negation of -2147483648 cannot be represented in type 'int32_t' (aka 'int');
Fixes: 6500/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_DIRAC_fuzzer-4523620274536448

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 cb944fc7f1)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
c071618ba6 avcodec/fic: Check available input space for cursor
Fixes: out of array read
Fixes: 6546/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_FIC_fuzzer-6317064647081984

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 cb2f7ea96b)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
b367c23da1 avcodec/mpeg4videodec: Check bps (VOL header) before VOP for studio profile
Fixes: runtime error: shift exponent -1 is negative
Fixes: 7486/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_MPEG4_fuzzer-4977380939530240

Fixes: runtime error: index 36 out of bounds for type 'const uint8_t [32]'
Fixes: 7566/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_MPEG4_fuzzer-6536620682510336

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 b3a18511cc)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
58a03420be avcodec/g2meet: Check RGB upper limit
Fixes: runtime error: left shift of 1876744317 by 16 places cannot be represented in type 'int'
Fixes: 6799/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_G2M_fuzzer-5115274731716608

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 4dd2c8b9ea)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
cbe442048f avcodec/jpeg2000dec: Fix undefined shift in the jpeg2000_decode_packets_po_iteration() CPRL case
Fixes: shift exponent 47 is too large for 32-bit type 'int'
Fixes: 7955/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_JPEG2000_fuzzer-6016721977606144

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 652ba72ed3)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
87e9f5e118 avcodec/jpeg2000dec: Skip init for component in CPRL if nothing is to be done
Fixes: assertion failure
Fixes: 7949/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_JPEG2000_fuzzer-4819602782552064

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 a96c131eb5)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
2cffce26a7 avcodec/g2meet: Change order of operations to avoid undefined behavior
Fixes: signed integer overflow: 65280 * 196032 cannot be represented in type 'int'
Fixes: 7279/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_G2M_fuzzer-5977332473921536

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 0a47451458)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
c26e101654 avcodec/flac_parser: Fix infinite loop
Fixes: crbug/827204

Reported-by: Frank Liberato <liberato@google.com>
Reviewed-by: Frank Liberato <liberato@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 15a2e35e9e)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
60e408f252 avcodec/mpeg4videodec: Split decode_studio_vol_header() out of decode_studiovisualobject()
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 177133a0f4)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
e61dcd2c86 avcodec/mpeg4videodec: Move decode_studiovisualobject() parsing in the branch for visual object parsing
Fixes: runtime error: shift exponent -1 is negative
Fixes: 7510/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_MPEG4_fuzzer-5024523356209152

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 e03bf251d8)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
dd3914c5b5 avcodec/mpeg4video_parser: Avoid litteral 0x1B6, use named constant instead
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit c0aa89eeee)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
90c4c076c7 avcodec/mpeg4video_parser: Fix incorrect spliting of MPEG-4 studio frames
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit a47bd1cd1c)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
d976855c00 avformat/m4vdec: Use the same constant names as libavcodec
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 0f176bb8e0)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
7bc5d49c60 avformat/m4vdec: Fix detection of raw MPEG-4 ES Studio
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 34dbdcfc20)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
36c4995428 avcodec/wavpack: Fix integer overflow in DEC_MED() / INC_MED()
Fixes: runtime error: signed integer overflow: 2147483637 + 128 cannot be represented in type 'int'
Fixes: 6701/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_WAVPACK_fuzzer-5358324934508544

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 6e95d80e6f)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
57bb78d980 avcodec/wavpack: Fix integer overflow in wv_unpack_stereo()
Fixes: runtime error: signed integer overflow: 2147483531 + 16384 cannot be represented in type 'int'
Fixes: 6615/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_WAVPACK_fuzzer-5165715515506688

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 da038c07f0)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
b2cb42f1c3 avcodec/error_resilience: Fix integer overflow in filter181()
Fixes: runtime error: signed integer overflow: 197710 * 10923 cannot be represented in type 'int'
Fixes: 7010/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_MPEG4_fuzzer-5667127596941312

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 1c97035e3b)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
fe7f2a77c7 avcodec/h263dec: Check slice_ret in mspeg4 slice loop
Fixes infinite loop
Fixes: 6858/clusterfuzz-testcase-ffmpeg_AV_CODEC_ID_MSMPEG4V3_fuzzer-4681563766784000
Fixes: 6890/clusterfuzz-testcase-ffmpeg_AV_CODEC_ID_WMV1_fuzzer-4756103142309888

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 de841fbea7)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
9dfe36616f avcodec/elsdec: Fix memleaks
Fixes: 6798/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_G2M_fuzzer-5135899701542912

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 0bd0401336)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
4ace1597a2 avcodec/vc1_block: simplify ac_val computation
also fixes: runtime error: index 1456 out of bounds for type 'int16_t [16]'

Found-by: durandal_1707
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit d06b01fc2d)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Michael Niedermayer
6d2c5bb5d2 avcodec/ffv1enc: Check that the crc + version combination is supported
The crc flag is only stored since version 3 thus before this crcs do not
work. We increase the version as needed same as we do with pix_fmts

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit d9706f79c1)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-06-15 22:31:13 +02:00
Carl Eugen Hoyos
a6d85a97d0 configure: The eac3_core bitstream filter needs the ac3 parser.
Fixes linking with "--disable-everything --enable-bsf=eac3_core".
(cherry picked from commit 9461e7d3a5)
2018-06-13 12:04:12 +02:00
John Cox
72a34d2332 configure: fix arm inline asm checks
Commit 8c893aa3cd removed quotes that were required to detect
inline asm in clang:

check_insn armv5te qadd r0, r0, r0
.../test.c:1:34: error: expected string literal in 'asm'
void foo(void){ __asm__ volatile(qadd r0, r0, r0); }

The correct code is:

void foo(void){ __asm__ volatile("qadd r0, r0, r0"); }

Commit message written by Frank Liberato <liberato@chromium.org>

Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit ad94f1c8ab)
2018-06-06 10:49:31 -03:00
Jan Ekström
c8b57d4333 lavf/libssh: translate a read of 0 to EOF
Yet another case of forgotten 0 =! EOF translation.

While the documentation for this specific synchronous read
function does not mention it, the documentation for
`sftp_async_read` documents it, as well as looking at the
implementation of this function leads one to find
`if (handle->eof) { return 0; }`.

Reported by stnutt on IRC.

(cherry picked from commit 26892c7615)
2018-05-28 20:34:53 +03:00
Aman Gupta
da399903c7 ffprobe: fix SEGV when new streams are added
Signed-off-by: Aman Gupta <aman@tmm1.net>
(cherry picked from commit 12ceaf0fba)
2018-05-18 19:16:35 -07:00
Aman Gupta
8336a66270 avformat/mpegts: fix incorrect indentation
Signed-off-by: Aman Gupta <aman@tmm1.net>
(cherry picked from commit 64bf915cd8)
2018-05-18 19:16:35 -07:00
Aman Gupta
d1845e7f1a avformat/mpegts: initialize section_buf to fix valgrind test failure
http://fate.ffmpeg.org/report.cgi?slot=x86_64-archlinux-gcc-valgrind&time=20180513001958

Signed-off-by: Aman Gupta <aman@tmm1.net>
(cherry picked from commit 42a03e7700)
2018-05-18 19:16:35 -07:00
Aman Gupta
852f78443a avformat/mpegts: reindent after last change
Signed-off-by: Aman Gupta <aman@tmm1.net>
(cherry picked from commit 7db022e67b)
2018-05-18 19:16:35 -07:00
Aman Gupta
c343eabfb7 avformat/mpegts: parse sections with multiple tables
Fixes PMT parsing in some mpegts streams which contain
multiple tables within the PMT pid. Previously, the parser
assumed only one table was present in each packet, and discarded
the rest of the section data after attempting to parse the first
table.

A similar issue was documented in the BeyondTV software[1], which
helped me diagnose the same bug in the ffmpeg mpegts demuxer. I also
tried DVBInspector, libdvbpsi's dvbinfo, and tstools' tsinfo to
help debug. The former two properly read PMTs with multiple tables,
whereas the last has the same bug as ffmpeg.

I've created a minimal sample[2] which contains the combined PMT.
Here's what ffmpeg probe shows before and after this patch:

Before:

    Input #0, mpegts, from 'combined-pmt-tids.ts':
      Duration: 00:00:01.08, start: 4932.966167, bitrate: 741 kb/s
      Program 1
      No Program
        Stream #0:0[0xf9d]: Audio: ac3, 48000 Hz, mono, fltp, 96 kb/s
        Stream #0:1[0xf9b]: Audio: mp3, 0 channels, fltp
        Stream #0:2[0xf9c]: Unknown: none

After:

    Input #0, mpegts, from 'combined-pmt-tids.ts':
      Duration: 00:00:01.11, start: 4932.966167, bitrate: 718 kb/s
      Program 1
        Stream #0:0[0xf9b]: Video: mpeg2video ([2][0][0][0] / 0x0002), none(tv, top first), 29.97 fps, 29.97 tbr, 90k tbn, 90k tbc
        Stream #0:1[0xf9c](eng): Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), fltp, 384 kb/s
        Stream #0:2[0xf9d](spa): Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, mono, fltp, 96 kb/s

With the patch, the PMT is parsed correctly so the streams are
created in the correct order, are associated with "Program 1",
and their codecs are set correctly.

[1] http://forums.snapstream.com/vb/showpost.php?p=343816&postcount=201
[2] https://s3.amazonaws.com/tmm1/combined-pmt-tids.ts

Signed-off-by: Aman Gupta <aman@tmm1.net>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 9152c1e495)
2018-05-18 19:16:35 -07:00
Aman Gupta
7ad163c258 avformat/mpegts: clean up whitespace
Signed-off-by: Aman Gupta <aman@tmm1.net>
(cherry picked from commit 07d9c31055)
2018-05-18 19:16:35 -07:00
Aman Gupta
ef28571efe avformat/mpegts: use MAX_SECTION_SIZE instead of hardcoded value
Signed-off-by: Aman Gupta <aman@tmm1.net>
(cherry picked from commit 1a14e39145)
2018-05-18 19:16:35 -07:00
Aman Gupta
97aea63340 avformat/mpegts: skip non-PMT tids earlier
This mimics the logic flow in all the other callbacks
(pat_cb, sdt_cb, m4sl_cb), and avoids calling skip_identical()
for non PMT_TID packets.

Since skip_identical modifies internal state like
MpegTSSectionFilter.last_ver, this change prevents unnecessary
reprocessing on some streams which contain multiple tables in
the PMT pid. This can be observed with streams from certain US
cable providers, which include both tid=0x2 and another unspecified
tid=0xc0.

Signed-off-by: Aman Gupta <aman@tmm1.net>
(cherry picked from commit 2c500f5097)
2018-05-18 19:16:35 -07:00
Aman Gupta
bb6a34f237 avcodec/mediacodecdec: add workaround for buggy amlogic mpeg2 decoder
I tested the previous mediacodec changes on seven different Android
TV devices, with both mpeg2 and h264 content. All except one worked
as expected. The exception was the MiBox3 running Android 6.0.1,
where playback would freeze on a frame every few seconds. I tested
two other AMLogic devices with newer Android versions that did not
show the same problem. H264 decoding on the MiBox3 was also not affected,
so this workaround applies only to OMX.amlogic.mpeg2.decoder.awesome
on Android API22.

There is a rumor that Xiaomi is planning to release Android Oreo for
the MiBox3, so I will revisit in a few months to confirm whether this
is specific to os/driver version or the chipset used in that device.

Signed-off-by: Aman Gupta <aman@tmm1.net>
Signed-off-by: Matthieu Bouron <matthieu.bouron@gmail.com>
(cherry picked from commit 9b563f6584)
2018-05-18 19:16:34 -07:00
Aman Gupta
db5631e408 avcodec/mediacodecdec: wait on first frame after input buffers are full
The output_buffer_count==0 special case is no longer required, and
can cause spurious EAGAIN to surface to the user when input buffers
are filled up. Since the caller now knows if the decoder is accepting
new input (via current_input_buffer>=0), let the wait parameter
control whether we block or not.

Signed-off-by: Aman Gupta <aman@tmm1.net>
Signed-off-by: Matthieu Bouron <matthieu.bouron@gmail.com>
(cherry picked from commit a75bb5496a)
2018-05-18 19:16:34 -07:00
Aman Gupta
df2c811b7c avcodec/mediacodecdec: restructure mediacodec_receive_frame
The new logic follows a recommendation by @rcombs to use
dequeueInputBuffer with a timeout of 0 as a way to detect
whether the codec wants more data. The dequeued buffer index is
kept in MediaCodecDecContext until it can be used next.

A similar technique is also used by the Google's official media
player Exoplayer: see MediaCodecRenderer.feedInputBuffer().

Signed-off-by: Aman Gupta <aman@tmm1.net>
Signed-off-by: Matthieu Bouron <matthieu.bouron@gmail.com>
(cherry picked from commit f6681feda6)
2018-05-18 19:16:34 -07:00
Aman Gupta
6f55a36be9 avcodec/mediacodec_wrapper: add helper to fetch SDK_INT
Signed-off-by: Matthieu Bouron <matthieu.bouron@gmail.com>
Signed-off-by: Aman Gupta <aman@tmm1.net>
(cherry picked from commit fe0a6bcbda)
2018-05-18 19:16:34 -07:00
Aman Gupta
33042d632d avcodec/mediacodecdec: refactor pts handling
Also fixes a bug where EOS buffer was sent with incorrect
pts when not using surface generation.

Signed-off-by: Matthieu Bouron <matthieu.bouron@gmail.com>
Signed-off-by: Aman Gupta <aman@tmm1.net>
(cherry picked from commit d8e92a89ed)
2018-05-18 19:16:34 -07:00
Aman Gupta
3054e53ddc avcodec/mediacodecdec: use AV_TIME_BASE_Q
Signed-off-by: Matthieu Bouron <matthieu.bouron@gmail.com>
Signed-off-by: Jan Ekström <jeebjp@gmail.com>
Signed-off-by: Aman Gupta <aman@tmm1.net>
(cherry picked from commit 7a4639b1eb)
2018-05-18 19:16:34 -07:00
Aman Gupta
84bf631018 avcodec/mediacodecdec: clarify delay_flush specific code
As of 2a0eb8685, ff_mediacodec_dec_is_flushing() only returns
true in delay_flush mode. Make this more obvious by adding
delay_flush to the if statement.

Signed-off-by: Matthieu Bouron <matthieu.bouron@gmail.com>
Signed-off-by: Aman Gupta <aman@tmm1.net>
(cherry picked from commit 6a7a84b2d1)
2018-05-18 19:16:34 -07:00
Aman Gupta
2884575d97 avcodec/videotoolbox: fix decoding of some HEVC videos
In a normal hwaccel, the AVHWFramesContext sets AVFrame.hw_frames_ctx
when it initializes a new AVFrame in av_hwframe_get_buffer().

But the VT hwaccel doesn't know what hw_frames_ctx to assign when
the AVFrame is first created, because it depends on the format of
the pixbuf that the decoder eventually decides to return. Thus
newly created AVFrames always have a NULL hw_frames_ctx, and the
hwaccel would only assign the ctx once a frame was done decoding.
This worked fine with the H264 decoder, but with the HEVC decoder
the frame's data may be moved to another empty AVFrame. Since the
empty AVFrame never had hw_frames_ctx set, a frame with a NULL
ctx could be returned to the API user.

This patch works around the issue by moving the derived
hw_frames_ctx from the AVFrame to a new VTHWFrame which now holds
both the CVPixelBufferRef and the AVBuffer. The hw_frames_ctx
is only copied to the AVFrame right before it is about to be
returned to the user in videotoolbox_postproc_frame() (since
in the case of VT, the hw_frames_ctx is only there for the API
user anyway).

Fixes playback on macOS and iOS of some hevc videos like
https://s3.amazonaws.com/tmm1/videotoolbox/germany-hevc-zdf.ts

Signed-off-by: Aman Gupta <aman@tmm1.net>
(cherry picked from commit 8f146b526f)
2018-05-18 19:16:34 -07:00
Aman Gupta
76716518a8 avcodec/hevc: remove videotoolbox hack
No longer required since 63d875772d. The equivalent hack
for h264 was removed in that commit, but this one was missed.

Signed-off-by: Aman Gupta <aman@tmm1.net>
(cherry picked from commit a19bac8fc8)
2018-05-18 19:16:34 -07:00
Aman Gupta
e8caf67f56 avcodec/videotoolbox: split h264/hevc callbacks
Previously the shared callbacks were trying to interpret
avctx->priv_data as H264Context*

Signed-off-by: Aman Gupta <aman@tmm1.net>
(cherry picked from commit 07d175d0b0)
2018-05-18 19:07:06 -07:00
Aman Gupta
e40922c16c avcodec/videotoolbox: cleanups
No functional changes.

Signed-off-by: Aman Gupta <aman@tmm1.net>
(cherry picked from commit dd77cca1c4)
2018-05-18 19:07:06 -07:00
Aman Gupta
ef9478d264 avcodec/videotoolbox: fix kVTCouldNotFindVideoDecoderErr trying to decode HEVC on iOS
Older iOS devices don't have a hardware HEVC decoder, but the
software decoder offered by VideoToolbox is well-optimized and
performs much better than the ffmpeg decoder.

Signed-off-by: Aman Gupta <aman@tmm1.net>
(cherry picked from commit bcff983dc3)
2018-05-18 19:07:06 -07:00
Aman Gupta
13d83899df avcodec/videotoolbox: improve logging of decoder errors
Signed-off-by: Aman Gupta <aman@tmm1.net>
(cherry picked from commit 84e03db9a3)
2018-05-18 19:07:06 -07:00
Marton Balint
61fed89ad4 avcodec/xwddec: fix palette alpha
Signed-off-by: Marton Balint <cus@passwd.hu>
(cherry picked from commit 50d6b7bd83)
2018-05-16 22:50:18 +02:00
Marton Balint
7931e01540 avformat/webm_chunk: always use a static buffer for get_chunk_filename
My conversation from AVFormatContext->filename to AVFormatContext->url was
wrong in this case because get_chunk_filename uses filename as an output
buffer, and not as an input buffer.

Fixes ticket #7188.

Signed-off-by: Marton Balint <cus@passwd.hu>
(cherry picked from commit 2dbe936bf7)
2018-05-15 23:09:51 +02:00
Niklas Haas
93cee87b13 configure: fix configure check for lilv-0
This should be included as `<lilv/lilv.h>`, same as is done in af_lv2.c.
Forcing the extra lilv-0 breaks platforms where the include dir is
`/usr/include/lilv/lilv.h` rather than
`/usr/include/lilv-0/lilv/lilv.h`.

The new include path works for both, because the `pkg-config --cflags`
includes `-I/usr/include/lilv-0`.

(cherry picked from commit 32234e03a7)
2018-05-13 19:18:48 +03:00
Philip Langdale
2a44f706aa avcodec/nvdec_hevc: fix scaling lists
The main issue here was the use of [i] instead of [i * 3] for the 32x32
matrix. As part of fixing this, I changed the code to match that used
in vdpau_hevc, which I spent a lot of time verifying.

I also changed to calculating NumPocTotalCurr using the existing helper,
which is what vdpau does.

Signed-off-by: Timo Rothenpieler <timo@rothenpieler.org>
(cherry picked from commit 1261003700)
2018-05-10 18:44:02 +02:00
Timo Rothenpieler
61b673b1f1 avcodec/hevcdec: make ff_hevc_frame_nb_refs take a const pointer
(cherry picked from commit 46c1ee1917)
2018-05-10 18:44:00 +02:00
Jan Ekström
8fde71acd9 lavf/bluray: translate a read of 0 to EOF
Yet another case of forgotten 0 =! EOF translation. The libbluray
documentation specifically mentions that a read of 0 is EOF.

Reported by Fyr on IRC.

(cherry picked from commit b995ec078f)
2018-05-10 17:59:55 +03:00
Rodger Combs
b32f865969 lavf/dashenc: don't call flush_init_segment before avformat_write_header
Fixes crash when muxing MKV-in-DASH
2018-05-04 13:38:18 +05:30
Marton Balint
d89eea3455 avdevice/decklink_dec: unref packets on avpacket_queue_put error
Signed-off-by: Marton Balint <cus@passwd.hu>
(cherry picked from commit 649087fa83)
2018-05-01 21:22:33 +02:00
Marton Balint
0a22e31fbb avcodec/hnm4video: fix palette alpha
Signed-off-by: Marton Balint <cus@passwd.hu>
(cherry picked from commit 4c501bafc0)
2018-05-01 21:22:19 +02:00
Marton Balint
70a01aa490 avcodec/anm: fix palette alpha
Signed-off-by: Marton Balint <cus@passwd.hu>
(cherry picked from commit e894d958fc)
2018-05-01 21:22:03 +02:00
Marton Balint
da6c519f6e avformat/qtpalette: parse color table according to the QuickTime file format specs
The specs says that the the first color component in the color array is
not alpha, but simply 0.

Fixes 0 alpha of fate-suite/cvid/catfight-cvid-pal8-partial.mov

Signed-off-by: Marton Balint <cus@passwd.hu>
(cherry picked from commit c60a824ee8)
2018-05-01 21:21:49 +02:00
Matt Oliver
29328d96b9 ffplay: Fix realloc_texture when input texture is NULL.
SDL_QueryTexture and SDL_DestroyTexture require that the input texture
pointer be non-null. Debug builds of SDL will correctly check for this
and break program execution. This patch fixes this by checking the
status of the texture pointer.

Signed-off-by: Matt Oliver <protogonoi@gmail.com>
(cherry picked from commit 6be690685a)
2018-05-01 21:21:02 +02:00
Mark Thompson
44cb647477 hwcontext_vaapi: Fix compilation with libva versions < 1.4.0
The BufferHandle API was added in libva 1.4.0 / VAAPI 0.36.0.

(cherry picked from commit 92a0a6bea9)
2018-04-27 23:42:07 +01:00
Ruiling Song
a768c0a3e1 lavf/qsv: clone the frame which may be managed by framework
For filters based on framesync, the input frame was managed
by framesync, so we should not directly keep and destroy it,
instead we make a clone of it here, or else double-free will occur.
But for other filters not based on framesync, we still need to
free the input frame inside filter_frame.

Signed-off-by: Ruiling Song <ruiling.song@intel.com>
(cherry picked from commit d865783b6c)
2018-04-23 22:56:18 +01:00
Ruiling Song
58569162c2 lavf: make overlay_qsv work based on framesync
The existing version which was cherry-picked from Libav does not work
with FFmpeg framework, because ff_request_frame() was totally
different between Libav (recursive) and FFmpeg (non-recursive).
The existing overlay_qsv implementation depends on the recursive version
of ff_request_frame to trigger immediate call to request_frame() on input pad.
But this has been removed in FFmpeg since "lavfi: make request_frame() non-recursive."
Now that we have handy framesync support in FFmpeg, so I make it work
based on framesync. Some other fixing which is also needed to make
overlay_qsv work are put in a separate patch.

Signed-off-by: Ruiling Song <ruiling.song@intel.com>
(cherry picked from commit f3341a0452)
2018-04-23 22:56:06 +01:00
Gyan Doshi
6b2fee19a7 avformat/segafilm - revert keyframe detection
Keyframe detection was inverted in cfe1a9d311 in order to fix keyframe
flags set for the sample attached to trac #7091. However, that sample is
errantly muxed.

As noted at
https://web.archive.org/web/20020803104640/http://www.pcisys.net:80/~melanson/codecs/film-format.txt,
the original keyframe detection logic is correct, and this patch
restores it.

Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit 9f9f56e679)
2018-04-22 16:34:39 -03:00
Aman Gupta
8cd79c2e73 avformat/utils: refactor upstream_stream_timings
Signed-off-by: Aman Gupta <aman@tmm1.net>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 7b8daa771c)
2018-04-20 12:25:32 -07:00
Aman Gupta
0502602d37 avformat/utils: ignore outlier durations on subtitle/data streams as well
Similar to 4c9c4fe8b2, but for durations. This fixes #7151, where
the report duration and bitrate on a mpegts stream is wildly off
due to the dvb_teletext stream's timings.

Signed-off-by: Aman Gupta <aman@tmm1.net>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit fd6e89586c)
2018-04-20 12:25:26 -07:00
Michael Niedermayer
ace829cb45 Changelog: replace <next> by 4.0
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-04-20 01:55:05 +02:00
Stephan Holljes
b9b3ef4f5a lavf/http.c: Free allocated client URLContext in case of error.
Signed-off-by: Stephan Holljes <klaxa1337@googlemail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 7b6b8c9265)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-04-20 01:53:41 +02:00
Michael Niedermayer
b2b7cb0f60 avdevice/android_camera: Fix AVClass.version
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 13b77af2f0)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-04-20 01:53:41 +02:00
Michael Niedermayer
5cc6370a15 avcodec: Fix AVClass .version
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit c0bce367e4)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-04-20 01:53:41 +02:00
Michael Niedermayer
8b019be79b avcodec/sheervideodata: Fix libavutil include
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 3dfe3436ac)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-04-20 01:53:41 +02:00
Michael Niedermayer
e36830c695 avcodec/sbc: Fix non static function prefix
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 9f1b99e7d0)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-04-20 01:53:41 +02:00
Michael Niedermayer
bc2ceeb3ac avcodec/opusenc_psy: Fix warning: ISO C90 forbids mixed declarations and code
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit f8b17fe332)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-04-20 01:53:41 +02:00
Michael Niedermayer
66bdf8f145 avcodec/dsicinvideo: Fail if there is only a small fraction of the data available that comprises a full frame
Fixes: Timeout
Fixes: 6306/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_DSICINVIDEO_fuzzer-5079253549842432

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 5549488bbf)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-04-20 01:53:41 +02:00
Michael Niedermayer
bfe61bbd00 avcodec/dsicinvideo: Propagate errors from cin_decode_rle()
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 942217b153)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-04-20 01:53:41 +02:00
Michael Niedermayer
5888679ae3 avcodec/dfa: Check dimension against maximum
The headers from where the dimensions are read in actual files
are limited to 16bit per component.

Fixes: Timeout
Fixes: 6305/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_DFA_fuzzer-4824270749302784

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 9d5a4fcfbb)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-04-20 01:53:41 +02:00
Michael Niedermayer
ecb375684d avcodec/cinepak: Skip empty frames
Speeds up decoding from 3 to 0.1 seconds for 6302/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_CINEPAK_fuzzer-5626371985375232
Fixes: Timeout

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 9033920bec)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-04-20 01:53:41 +02:00
Michael Niedermayer
df56bc18ef avcodec/cinepak: move some checks prior to frame allocation
Speeds up decoding from 8 to 3 seconds for 6302/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_CINEPAK_fuzzer-5626371985375232
Fixes: Timeout

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 2324ef1ff3)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-04-20 01:53:41 +02:00
Rahul Chaudhry
ef99025603 swresample/arm: remove unintentional relocation.
Branch to global symbol results in reference to PLT, and when compiling
for THUMB-2 - in a R_ARM_THM_JUMP19 relocation. Some linkers don't
support this relocation (ld.gold), while others can end up truncating
the relocation to fit (ld.bfd).

Convert this branch through PLT into a direct branch that the assembler
can resolve locally.

See https://github.com/android-ndk/ndk/issues/337 for background.

The current workaround is to disable neon during gstreamer build,
which is not optimal and can be reverted after this patch:
41556c4157

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit b22db4f465)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-04-20 01:53:41 +02:00
Michael Niedermayer
860293a9a2 doc/APIchanges: Fix typos in hashes
Thanks-to: Moritz Barsnick <barsnick@gmx.net> for finding the correct ones

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit ec8a5262b0)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-04-20 01:53:41 +02:00
Matthieu Bouron
9b71114247 avcodec/mediacodecdec_common: make stride and slice-height non-mandatory fields
Fixes decoding on the Samsung Chromebook Pro which do not set the codec
output format stride and slice-height fields.

(cherry picked from commit 67d0911f27)
2018-04-19 14:25:23 +02:00
Hendrik Leppkes
0b6de235b9 avformat/tls_schannel: fix handling of EOF after avio changes
(cherry picked from commit 5c6365af45)
2018-04-19 13:27:49 +02:00
Hendrik Leppkes
a73b464118 configure: fix clang-cl check in the MSVC section
Without properly grouping the checks, the second test would execute for
MSVC cl.exe, which results in configure getting stuck since cl.exe -? is
an interactive paginated help screen, waiting for input.
2018-04-19 09:58:48 +02:00
James Almer
d9e9e97e5f avdevice/iec61883: free the private context at the end
Fixes part of ticket #7146.

Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit 5079e96bcc)
2018-04-18 23:04:04 -03:00
James Almer
d52676da38 avdevice/iec61883: return reference counted packets
Fixes part of ticket #7146, dealing with leaks of packet data since
commit 87c8812270.

Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit b8629654c6)
2018-04-18 23:04:01 -03:00
Timo Teräs
ca85c3cd7d avformat/movenc: support writing iTunes cover image
Fixes https://trac.ffmpeg.org/ticket/2798

This makes movenc handle AV_DISPOSITION_ATTACHED_PIC and write
the associated pictures in iTunes cover atom. This corresponds
to how 'mov' demuxer parses and exposes the cover images when
reading.

Most of the existing track handling loops properly ignore
these 'virtual streams' as MOVTrack->entry is never incremented
for them. However, additional tests are added as needed to ignore
them.

Tested to produce valid output with:
  ffmpeg -i movie.mp4 -i thumb.jpg -disposition:v:1 attached_pic \
         -map 0 -map 1 -c copy movie-with-cover.mp4

The cover image is also copied correctly with:
  ffmpeg -i movie-with-cover.mp4 -map 0 -c copy out.mp4

AtomicParseley says that the attached_pic stream is properly
not visible in the main tracks of the file.

Signed-off-by: Timo Teräs <timo.teras@iki.fi>
(cherry picked from commit 9af71b326f)
2018-04-17 20:13:37 +01:00
Timo Teräs
de253343c1 ffprobe: report unavailable SAR correctly in stream info
av_guess_sample_aspect_ratio() will return undefined or missing
value as {0,1}. This fixes show_stream() to check numerator to
display 'N/A' when appropriate. show_frame() does this already
correctly.

Signed-off-by: Timo Teräs <timo.teras@iki.fi>
(cherry picked from commit c663dce031)
2018-04-17 20:13:25 +01:00
Timo Rothenpieler
9c787a21ce configure: add nvcc to CMDLINE_SET
This was somehow forgotten and nobody noticed until now.
2018-04-16 22:36:43 +02:00
James Almer
7e11a86175 avformat/flacenc: add flac_init()
Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit 6838359448)
2018-04-16 14:12:48 -03:00
Rodger Combs
9ef90ff0a2 avformat/flacenc: support writing attached pictures
Usage of packet queueing API and some cleaning done by the committer.

Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit 00d8598eba)
2018-04-16 14:12:37 -03:00
James Almer
6c95a26c1a avformat/movenc: forbid muxing AV1 streams until the spec is finished
This prevents creating potentially broken files, as both the AV1 and
the AV1 in ISOMBFF specs are unfinished.

Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit 62bdbb5ce0)
2018-04-16 10:50:38 -03:00
James Almer
b6ec181240 configure: extend the check for bcrypt
Some old mingw-w64 builds seem to provide an incomplete implementation
of the API. Add an extra check to make sure it's disabled for those.

Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit c04609eefc)
2018-04-16 10:50:30 -03:00
Michael Niedermayer
b42e135614 Update bunch of versions to 4.0
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-04-16 12:42:29 +02:00
Michael Niedermayer
0564e8ee49 add release notes based on release 3.4
Name suggestion was from Kieran

(cherry picked from commit b1ec41a64f)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2018-04-16 12:41:03 +02:00
2785 changed files with 59458 additions and 213361 deletions

1
.gitignore vendored
View File

@@ -36,4 +36,3 @@
/lcov/
/src
/mapfile
/tools/python/__pycache__/

View File

@@ -1,21 +0,0 @@
<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>
<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.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>
<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>

View File

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

View File

@@ -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.

1215
Changelog

File diff suppressed because it is too large Load Diff

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

View File

@@ -21,11 +21,10 @@ Specifically, the GPL parts of FFmpeg are:
- `compat/solaris/make_sunver.pl`
- `doc/t2h.pm`
- `doc/texi2pod.pl`
- `libswresample/tests/swresample.c`
- `libswresample/swresample-test.c`
- `tests/checkasm/*`
- `tests/tiny_ssim.c`
- the following filters in libavfilter:
- `signature_lookup.c`
- `vf_blackframe.c`
- `vf_boxblur.c`
- `vf_colormatrix.c`
@@ -35,13 +34,13 @@ Specifically, the GPL parts of FFmpeg are:
- `vf_eq.c`
- `vf_find_rect.c`
- `vf_fspp.c`
- `vf_geq.c`
- `vf_histeq.c`
- `vf_hqdn3d.c`
- `vf_interlace.c`
- `vf_kerndeint.c`
- `vf_lensfun.c` (GPL version 3 or later)
- `vf_mcdeint.c`
- `vf_mpdecimate.c`
- `vf_nnedi.c`
- `vf_owdenoise.c`
- `vf_perspective.c`
- `vf_phase.c`
@@ -50,14 +49,12 @@ Specifically, the GPL parts of FFmpeg are:
- `vf_pullup.c`
- `vf_repeatfields.c`
- `vf_sab.c`
- `vf_signature.c`
- `vf_smartblur.c`
- `vf_spp.c`
- `vf_stereo3d.c`
- `vf_super2xsai.c`
- `vf_tinterlace.c`
- `vf_uspp.c`
- `vf_vaguedenoiser.c`
- `vsrc_mptestsrc.c`
Should you, for whatever reason, prefer to use version 3 of the (L)GPL, then
@@ -83,47 +80,41 @@ affect the licensing of binaries resulting from the combination.
### Compatible libraries
The following libraries are under GPL version 2:
- avisynth
The following libraries are under GPL:
- frei0r
- libcdio
- libdavs2
- librubberband
- libvidstab
- libx264
- libx265
- libxavs
- libxavs2
- libxvid
When combining them with FFmpeg, FFmpeg needs to be licensed as GPL as well by
passing `--enable-gpl` to configure.
The following libraries are under LGPL version 3:
- gmp
- libaribb24
- liblensfun
When combining them with FFmpeg, use the configure option `--enable-version3` to
upgrade FFmpeg to the LGPL v3.
The VMAF, mbedTLS, RK MPI, OpenCORE and VisualOn libraries are under the Apache License
2.0. That license is incompatible with the LGPL v2.1 and the GPL v2, but not with
The OpenCORE and VisualOn libraries are under the Apache License 2.0. That
license is incompatible with the LGPL v2.1 and the GPL v2, but not with
version 3 of those licenses. So to combine these libraries with FFmpeg, the
license version needs to be upgraded by passing `--enable-version3` to configure.
The smbclient library is under the GPL v3, to combine it with FFmpeg,
the options `--enable-gpl` and `--enable-version3` have to be passed to
configure to upgrade FFmpeg to the GPL v3.
### Incompatible libraries
There are certain libraries you can combine with FFmpeg whose licenses are not
compatible with the GPL and/or the LGPL. If you wish to enable these
libraries, even in circumstances that their license may be incompatible, pass
`--enable-nonfree` to configure. This will cause the resulting binary to be
`--enable-nonfree` to configure. But note that if you enable any of these
libraries the resulting binary will be under a complex license mix that is
more restrictive than the LGPL and that may result in additional obligations.
It is possible that these restrictions cause the resulting binary to be
unredistributable.
The Fraunhofer FDK AAC and OpenSSL libraries are under licenses which are
incompatible with the GPLv2 and v3. To the best of our knowledge, they are
compatible with the LGPL.
The NVENC library, while its header file is licensed under the compatible MIT
license, requires a proprietary binary blob at run time, and is deemed to be
incompatible with the GPL. We are not certain if it is compatible with the
LGPL, but we require `--enable-nonfree` even with LGPL configurations in case
it is not.

View File

@@ -39,7 +39,7 @@ QuickTime faststart:
Miscellaneous Areas
===================
documentation Stefano Sabatini, Mike Melanson, Timothy Gu, Gyan Doshi
documentation Stefano Sabatini, Mike Melanson, Timothy Gu, Lou Logan, Gyan Doshi
project server Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser, Nikolay Aleksandrov
presets Robert Swain
metadata subsystem Aurelien Jacobs
@@ -52,9 +52,9 @@ Communication
website Deby Barbara Lepage
fate.ffmpeg.org Timothy Gu
Trac bug tracker Alexander Strasser, Michael Niedermayer, Carl Eugen Hoyos
Patchwork Andriy Gelman
mailing lists Baptiste Coudurier
Trac bug tracker Alexander Strasser, Michael Niedermayer, Carl Eugen Hoyos, Lou Logan
mailing lists Baptiste Coudurier, Lou Logan
Google+ Paul B Mahol, Michael Niedermayer, Alexander Strasser
Twitter Lou Logan, Reynaldo H. Verdejo Pinochet
Launchpad Timothy Gu
ffmpeg-security Andreas Cadhalpun, Carl Eugen Hoyos, Clément Bœsch, Michael Niedermayer, Reimar Doeffinger, Rodger Combs, wm4
@@ -78,7 +78,6 @@ Other:
float_dsp Loren Merritt
hash Reimar Doeffinger
hwcontext_cuda* Timo Rothenpieler
hwcontext_vulkan* Lynne
intfloat* Michael Niedermayer
integer.c, integer.h Michael Niedermayer
lzo Reimar Doeffinger
@@ -89,7 +88,6 @@ Other:
rational.c, rational.h Michael Niedermayer
rc4 Reimar Doeffinger
ripemd.c, ripemd.h James Almer
tx* Lynne
libavcodec
@@ -123,6 +121,7 @@ Generic Parts:
motion* Michael Niedermayer
rate control:
ratecontrol.c Michael Niedermayer
libxvid_rc.c Michael Niedermayer
simple IDCT:
simple_idct.c, simple_idct.h Michael Niedermayer
postprocessing:
@@ -145,7 +144,6 @@ Codecs:
asv* Michael Niedermayer
atrac3plus* Maxim Poliakovski
audiotoolbox* Rodger Combs
avs2* Huiwen Ren
bgmc.c, bgmc.h Thilo Borgmann
binkaudio.c Peter Ross
cavs* Stefan Gehrer
@@ -170,6 +168,7 @@ Codecs:
eacmv*, eaidct*, eat* Peter Ross
evrc* Paul B Mahol
exif.c, exif.h Thilo Borgmann
exr.c Martin Vignali
ffv1* Michael Niedermayer
ffwavesynth.c Nicolas George
fifo.c Jan Sebechlebsky
@@ -191,17 +190,14 @@ Codecs:
libcelt_dec.c Nicolas George
libcodec2.c Tomas Härdin
libdirac* David Conrad
libdavs2.c Huiwen Ren
libgsm.c Michel Bardiaux
libkvazaar.c Arttu Ylä-Outinen
libopenh264enc.c Martin Storsjo, Linjie Fu
libopenjpeg.c Jaikrishnan Menon
libopenjpegenc.c Michael Bradshaw
libtheoraenc.c David Conrad
libvorbis.c David Conrad
libvpx* James Zern
libxavs.c Stefan Gehrer
libxavs2.c Huiwen Ren
libzvbi-teletextdec.c Marton Balint
lzo.h, lzo.c Reimar Doeffinger
mdec.c Michael Niedermayer
@@ -217,7 +213,6 @@ Codecs:
msvideo1.c Mike Melanson
nuv.c Reimar Doeffinger
nvdec*, nvenc* Timo Rothenpieler
omx.c Martin Storsjo, Aman Gupta
opus* Rostislav Pehlivanov
paf.* Paul B Mahol
pcx.c Ivo van Poorten
@@ -225,7 +220,7 @@ Codecs:
ptx.c Ivo van Poorten
qcelp* Reynaldo H. Verdejo Pinochet
qdm2.c, qdm2data.h Roberto Togni
qsv* Mark Thompson, Zhong Li
qsv* Mark Thompson
qtrle.c Mike Melanson
ra144.c, ra144.h, ra288.c, ra288.h Roberto Togni
resample2.c Michael Niedermayer
@@ -338,7 +333,6 @@ Filters:
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
@@ -366,15 +360,12 @@ Filters:
vf_ssim.c Paul B Mahol
vf_stereo3d.c Paul B Mahol
vf_telecine.c Paul B Mahol
vf_tonemap_opencl.c Ruiling Song
vf_yadif.c Michael Niedermayer
vf_zoompan.c Paul B Mahol
Sources:
vsrc_mandelbrot.c Michael Niedermayer
dnn Yejun Guo
libavformat
===========
@@ -422,6 +413,7 @@ Muxers/Demuxers:
flvenc.c Michael Niedermayer, Steven Liu
gxf.c Reimar Doeffinger
gxfenc.c Baptiste Coudurier
hls.c Anssi Hannula
hlsenc.c Christian Suloway, Steven Liu
idcin.c Mike Melanson
idroqdec.c Mike Melanson
@@ -436,9 +428,9 @@ Muxers/Demuxers:
lmlm4.c Ivo van Poorten
lvfdec.c Paul B Mahol
lxfdec.c Tomas Härdin
matroska.c Aurelien Jacobs, Andreas Rheinhardt
matroskadec.c Aurelien Jacobs, Andreas Rheinhardt
matroskaenc.c David Conrad, Andreas Rheinhardt
matroska.c Aurelien Jacobs
matroskadec.c Aurelien Jacobs
matroskaenc.c David Conrad
matroska subtitles (matroskaenc.c) John Peebles
metadata* Aurelien Jacobs
mgsts.c Paul B Mahol
@@ -453,7 +445,7 @@ Muxers/Demuxers:
mpegtsenc.c Baptiste Coudurier
msnwc_tcp.c Ramiro Polla
mtv.c Reynaldo H. Verdejo Pinochet
mxf* Baptiste Coudurier, Tomas Härdin
mxf* Baptiste Coudurier
nistspheredec.c Paul B Mahol
nsvdec.c Francois Revol
nut* Michael Niedermayer
@@ -461,6 +453,7 @@ Muxers/Demuxers:
oggdec.c, oggdec.h David Conrad
oggenc.c Baptiste Coudurier
oggparse*.c David Conrad
oggparsedaala* Rostislav Pehlivanov
oma.c Maxim Poliakovski
paf.c Paul B Mahol
psxstr.c Mike Melanson
@@ -508,7 +501,6 @@ Protocols:
ftp.c Lukasz Marek
http.c Ronald S. Bultje
libssh.c Lukasz Marek
libzmq.c Andriy Gelman
mms*.c Ronald S. Bultje
udp.c Luca Abeni
icecast.c Marvin Scholz
@@ -532,10 +524,9 @@ Operating systems / CPU architectures
=====================================
Alpha Falk Hueffner
MIPS Manojkumar Bhosale, Shiyou Yin
MIPS Manojkumar Bhosale
Mac OS X / PowerPC Romain Dolbeau, Guillaume Poirier
Amiga / PowerPC Colin Ward
Linux / PowerPC Lauri Kasanen
Windows MinGW Alex Beregszaszi, Ramiro Polla
Windows Cygwin Victor Paesa
Windows MSVC Matthew Oliver, Hendrik Leppkes
@@ -564,7 +555,6 @@ Joakim Plate
Jun Zhao
Kieran Kunhya
Kirill Gavrilov
Limin Wang
Martin Storsjö
Panagiotis Issaris
Pedro Arthur
@@ -585,11 +575,8 @@ Releases
If you want to maintain an older release, please contact us
GnuPG Fingerprints and IRC nicknames of maintainers and contributors
====================================================================
IRC nicknames are in parentheses. These apply
to the IRC channels listed on the website.
GnuPG Fingerprints of maintainers and contributors
==================================================
Alexander Strasser 1C96 78B7 83CB 8AA7 9AF5 D1EB A7D8 A57B A876 E58F
Anssi Hannula 1A92 FF42 2DD9 8D2E 8AF7 65A9 4278 C520 513D F3CB
@@ -607,16 +594,13 @@ 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
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
Lou Logan 7D68 DC73 CBEF EABB 671A B6CF 621C 2E28 82F8 DC3A
Michael Niedermayer 9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
DD1E C9E8 DE08 5C62 9B3E 1846 B18E 8928 B394 8D64
Nicolas George 24CE 01CE 9ACC 5CEB 74D8 8D9D B063 D997 36E5 4C93
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
Ramiro Polla 7859 C65B 751B 1179 792E DAE8 8E95 8B2F 9B6C 5700
Reimar Doeffinger C61D 16E5 9E2C D10C 8958 38A4 0899 A2B9 06D4 D9C7
Reinhard Tartler 9300 5DC2 7E87 6C37 ED7B CA9A 9808 3544 9453 48A4
Reynaldo H. Verdejo Pinochet 6E27 CD34 170C C78E 4D4F 5F40 C18E 077F 3114 452A
@@ -625,8 +609,7 @@ Sascha Sommer 38A0 F88B 868E 9D3A 97D4 D6A0 E823 706F 1E07 0D3C
Stefano Sabatini 0D0B AD6B 5330 BBAD D3D6 6A0C 719C 2839 FC43 2D5F
Steinar H. Gunderson C2E9 004F F028 C18E 4EAD DB83 7F61 7561 7797 8F76
Stephan Hilb 4F38 0B3A 5F39 B99B F505 E562 8D5C 5554 4E17 8863
Thilo Borgmann (thilo) CE1D B7F4 4D20 FC3A DD9F FE5A 257C 5B8F 1D20 B92F
Tiancheng "Timothy" Gu 9456 AFC0 814A 8139 E994 8351 7FE6 B095 B582 B0D4
Tim Nicholson 38CF DB09 3ED0 F607 8B67 6CED 0C0B FC44 8B0B FC83
Tomas Härdin (thardin) A79D 4E3D F38F 763F 91F5 8B33 A01E 8AE0 41BB 2551
Tomas Härdin A79D 4E3D F38F 763F 91F5 8B33 A01E 8AE0 41BB 2551
Wei Gao 4269 7741 857A 0E60 9EC5 08D2 4744 4EFA 62C1 87B9

View File

@@ -50,12 +50,6 @@ $(TOOLS): %$(EXESUF): %.o
target_dec_%_fuzzer$(EXESUF): target_dec_%_fuzzer.o $(FF_DEP_LIBS)
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
tools/target_bsf_%_fuzzer$(EXESUF): tools/target_bsf_%_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) $(LIBFUZZER_PATH)
tools/sofa2wavs$(EXESUF): ELIBS = $(FF_EXTRALIBS)
tools/uncoded_frame$(EXESUF): $(FF_DEP_LIBS)
tools/uncoded_frame$(EXESUF): ELIBS = $(FF_EXTRALIBS)
@@ -64,7 +58,6 @@ tools/target_dec_%_fuzzer$(EXESUF): $(FF_DEP_LIBS)
CONFIGURABLE_COMPONENTS = \
$(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c)) \
$(SRC_PATH)/libavcodec/bitstream_filters.c \
$(SRC_PATH)/libavcodec/parsers.c \
$(SRC_PATH)/libavformat/protocols.c \
config.h: ffbuild/.config
@@ -141,7 +134,7 @@ uninstall-data:
clean::
$(RM) $(CLEANSUFFIXES)
$(RM) $(addprefix compat/,$(CLEANSUFFIXES)) $(addprefix compat/*/,$(CLEANSUFFIXES)) $(addprefix compat/*/*/,$(CLEANSUFFIXES))
$(RM) $(addprefix compat/,$(CLEANSUFFIXES)) $(addprefix compat/*/,$(CLEANSUFFIXES))
$(RM) -r coverage-html
$(RM) -rf coverage.info coverage.info.in lcov
@@ -151,7 +144,6 @@ distclean:: clean
version.h libavutil/ffversion.h libavcodec/codec_names.h \
libavcodec/bsf_list.c libavformat/protocol_list.c \
libavcodec/codec_list.c libavcodec/parser_list.c \
libavfilter/filter_list.c libavdevice/indev_list.c libavdevice/outdev_list.c \
libavformat/muxer_list.c libavformat/demuxer_list.c
ifeq ($(SRC_LINK),src)
$(RM) src
@@ -166,7 +158,7 @@ check: all alltools examples testprogs fate
include $(SRC_PATH)/tests/Makefile
$(sort $(OUTDIRS)):
$(sort $(OBJDIRS)):
$(Q)mkdir -p $@
# Dummy rule to stop make trying to rebuild removed or renamed headers

View File

@@ -1 +1 @@
4.3.6
4.0.3

View File

@@ -1,15 +1,15 @@
┌───────────────────────────────────
│ RELEASE NOTES for FFmpeg 4.3 "4:3" │
└───────────────────────────────────
┌───────────────────────────────────┐
│ RELEASE NOTES for FFmpeg 4.0 "Wu" │
└───────────────────────────────────┘
The FFmpeg Project proudly presents FFmpeg 4.3 "4:3", about 10
months after the release of FFmpeg 4.2.
The FFmpeg Project proudly presents FFmpeg 4.0 "Wu", about 6
months after the release of FFmpeg 3.4.
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.

1064
compat/avisynth/avisynth_c.h Normal file

File diff suppressed because it is too large Load Diff

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

View File

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

View File

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

View File

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

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

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

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@@ -1,131 +0,0 @@
/*
* Minimum CUDA compatibility definitions header
*
* Copyright (c) 2019 Rodger Combs
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef COMPAT_CUDA_CUDA_RUNTIME_H
#define COMPAT_CUDA_CUDA_RUNTIME_H
// Common macros
#define __global__ __attribute__((global))
#define __device__ __attribute__((device))
#define __device_builtin__ __attribute__((device_builtin))
#define __align__(N) __attribute__((aligned(N)))
#define __inline__ __inline__ __attribute__((always_inline))
#define max(a, b) ((a) > (b) ? (a) : (b))
#define min(a, b) ((a) < (b) ? (a) : (b))
#define abs(x) ((x) < 0 ? -(x) : (x))
#define atomicAdd(a, b) (__atomic_fetch_add(a, b, __ATOMIC_SEQ_CST))
// Basic typedefs
typedef __device_builtin__ unsigned long long cudaTextureObject_t;
typedef struct __device_builtin__ __align__(2) uchar2
{
unsigned char x, y;
} uchar2;
typedef struct __device_builtin__ __align__(4) ushort2
{
unsigned short x, y;
} ushort2;
typedef struct __device_builtin__ uint3
{
unsigned int x, y, z;
} 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;
} uchar4;
typedef struct __device_builtin__ __align__(8) ushort4
{
unsigned char x, y, z, w;
} ushort4;
typedef struct __device_builtin__ __align__(16) int4
{
int x, y, z, w;
} int4;
// Accessors for special registers
#define GETCOMP(reg, comp) \
asm("mov.u32 %0, %%" #reg "." #comp ";" : "=r"(tmp)); \
ret.comp = tmp;
#define GET(name, reg) static inline __device__ uint3 name() {\
uint3 ret; \
unsigned tmp; \
GETCOMP(reg, x) \
GETCOMP(reg, y) \
GETCOMP(reg, z) \
return ret; \
}
GET(getBlockIdx, ctaid)
GET(getBlockDim, ntid)
GET(getThreadIdx, tid)
// Instead of externs for these registers, we turn access to them into calls into trivial ASM
#define blockIdx (getBlockIdx())
#define blockDim (getBlockDim())
#define threadIdx (getThreadIdx())
// Basic initializers (simple macros rather than inline functions)
#define make_uchar2(a, b) ((uchar2){.x = a, .y = b})
#define make_ushort2(a, b) ((ushort2){.x = a, .y = b})
#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})
// 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(uchar2, make_uchar2(a & 0xFF, b & 0xFF))
TEX2D(ushort2, make_ushort2(a & 0xFFFF, b & 0xFFFF))
TEX2D(uchar4, make_uchar4(a & 0xFF, b & 0xFF, c & 0xFF, d & 0xFF))
TEX2D(ushort4, make_ushort4(a & 0xFFFF, b & 0xFFFF, c & 0xFFFF, d & 0xFFFF))
// Template calling tex instruction and converting the output to the selected type
template <class T>
static inline __device__ T tex2D(cudaTextureObject_t texObject, float x, float y)
{
T ret;
unsigned ret1, ret2, ret3, ret4;
asm("tex.2d.v4.u32.f32 {%0, %1, %2, %3}, [%4, {%5, %6}];" :
"=r"(ret1), "=r"(ret2), "=r"(ret3), "=r"(ret4) :
"l"(texObject), "f"(x), "f"(y));
conv(&ret, ret1, ret2, ret3, ret4);
return ret;
}
#endif /* COMPAT_CUDA_CUDA_RUNTIME_H */

View File

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

View File

@@ -27,8 +27,10 @@ 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"
while read LINE
do
printf "\n\t\"%s\\\n\"" "$(printf "%s" "$LINE" | sed -e 's/\r//g' -e 's/["\\]/\\&/g')" >> "$OUT"
done < "$IN"
printf ";\n" >> "$OUT"
exit 0

View File

@@ -1,47 +0,0 @@
/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <math.h>
#define FUN(name, type, op) \
type name(type x, type y) \
{ \
if (fpclassify(x) == FP_NAN) return y; \
if (fpclassify(y) == FP_NAN) return x; \
return x op y ? x : y; \
}
FUN(fmin, double, <)
FUN(fmax, double, >)
FUN(fminf, float, <)
FUN(fmaxf, float, >)
long double fmodl(long double x, long double y)
{
return fmod(x, y);
}
long double scalbnl(long double x, int exp)
{
return scalbn(x, exp);
}
long double copysignl(long double x, long double y)
{
return copysign(x, y);
}

View File

@@ -1,25 +0,0 @@
/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
double fmin(double, double);
double fmax(double, double);
float fminf(float, float);
float fmaxf(float, float);
long double fmodl(long double, long double);
long double scalbnl(long double, int);
long double copysignl(long double, long double);

View File

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

View File

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

View File

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

View File

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

924
configure vendored

File diff suppressed because it is too large Load Diff

View File

@@ -15,186 +15,6 @@ libavutil: 2017-10-21
API changes, most recent first:
2020-06-05 - ec39c2276a - lavu 56.50.100 - buffer.h
Passing NULL as alloc argument to av_buffer_pool_init2() is now allowed.
2020-05-27 - ba6cada92e - lavc 58.88.100 - avcodec.h codec.h
Move AVCodec-related public API to new header codec.h.
2020-05-23 - 064b875e89 - lavu 56.49.100 - video_enc_params.h
Add AV_VIDEO_ENC_PARAMS_H264.
2020-05-23 - 2e08b39444 - lavu 56.48.100 - hwcontext.h
Add av_hwdevice_ctx_create_derived_opts.
2020-05-23 - 6b65c4ec54 - lavu 56.47.100 - rational.h
Add av_gcd_q().
2020-05-22 - af9e622776 - lavu 56.46.101 - opt.h
Add AV_OPT_FLAG_CHILD_CONSTS.
2020-05-22 - 9d443c3e68 - lavc 58.87.100 - avcodec.h codec_par.h
Move AVBitstreamFilter-related public API to new header bsf.h.
Move AVCodecParameters-related public API to new header codec_par.h.
2020-05-21 - 13b1bbff0b - lavc 58.86.101 - avcodec.h
Deprecated AV_CODEC_CAP_INTRA_ONLY and AV_CODEC_CAP_LOSSLESS.
2020-05-17 - 84af196c65 - lavu 56.46.100 - common.h
Add av_sat_add64() and av_sat_sub64()
2020-05-12 - 991d417692 - lavu 56.45.100 - video_enc_params.h
lavc 58.84.100 - avcodec.h
Add a new API for exporting video encoding information.
Replaces the deprecated API for exporting QP tables from decoders.
Add AV_CODEC_EXPORT_DATA_VIDEO_ENC_PARAMS to request this information from
decoders.
2020-05-10 - dccd07f66d - lavu 56.44.100 - hwcontext_vulkan.h
Add enabled_inst_extensions, num_enabled_inst_extensions, enabled_dev_extensions
and num_enabled_dev_extensions fields to AVVulkanDeviceContext
2020-04-22 - 0e1db79e37 - lavc 58.81.100 - packet.h
- lavu 56.43.100 - dovi_meta.h
Add AV_PKT_DATA_DOVI_CONF and AVDOVIDecoderConfigurationRecord.
2020-04-15 - 22b25b3ea5 - lavc 58.79.100 - avcodec.h
Add formal support for calling avcodec_flush_buffers() on encoders.
Encoders that set the cap AV_CODEC_CAP_ENCODER_FLUSH will be flushed.
For all other encoders, the call is now a no-op rather than undefined
behaviour.
2020-04-10 - 672946c7fe - lavc 58.78.100 - avcodec.h codec_desc.h codec_id.h packet.h
Move AVCodecDesc-related public API to new header codec_desc.h.
Move AVCodecID enum to new header codec_id.h.
Move AVPacket-related public API to new header packet.h.
2020-03-29 - 4cb0dda555 - lavf 58.42.100 - avformat.h
av_read_frame() now guarantees to handle uninitialized input packets
and to return refcounted packets on success.
2020-03-27 - c52ec0367d - lavc 58.77.100 - avcodec.h
av_packet_ref() now guarantees to return the destination packet
in a blank state on error.
2020-03-10 - 05d27f342b - lavc 58.75.100 - avcodec.h
Add AV_PKT_DATA_ICC_PROFILE.
2020-02-21 - d005a7cdfd - lavc 58.73.101 - avcodec.h
Add AV_CODEC_EXPORT_DATA_PRFT.
2020-02-21 - c666689491 - lavc 58.73.100 - avcodec.h
Add AVCodecContext.export_side_data and AV_CODEC_EXPORT_DATA_MVS.
2020-02-13 - e8f054b095 - lavu 56.41.100 - tx.h
Add AV_TX_INT32_FFT and AV_TX_INT32_MDCT
2020-02-12 - 3182114f88 - lavu 56.40.100 - log.h
Add av_log_once().
2020-02-04 - a88449ffb2 - lavu 56.39.100 - hwcontext.h
Add AV_PIX_FMT_VULKAN
Add AV_HWDEVICE_TYPE_VULKAN and implementation.
2020-01-30 - 27529eeb27 - lavf 58.37.100 - avio.h
Add avio_protocol_get_class().
2020-01-15 - 717b2074ec - lavc 58.66.100 - avcodec.h
Add AV_PKT_DATA_PRFT and AVProducerReferenceTime.
2019-12-27 - 45259a0ee4 - lavu 56.38.100 - eval.h
Add av_expr_count_func().
2019-12-26 - 16685114d5 - lavu 56.37.100 - buffer.h
Add av_buffer_pool_buffer_get_opaque().
2019-11-17 - 1c23abc88f - lavu 56.36.100 - eval API
Add av_expr_count_vars().
2019-10-14 - f3746d31f9 - lavu 56.35.101 - opt.h
Add AV_OPT_FLAG_RUNTIME_PARAM.
2019-09-25 - f8406ab4b9 - lavc 58.59.100 - avcodec.h
Add max_samples
2019-09-04 - 2a9d461abc - lavu 56.35.100 - hwcontext_videotoolbox.h
Add av_map_videotoolbox_format_from_pixfmt2() for full range pixfmt
2019-09-01 - 8821d1f56e - lavu 56.34.100 - pixfmt.h
Add EBU Tech. 3213-E AVColorPrimaries value
2019-08-17 - 95fa73a2b4 - lavf 58.31.101 - avio.h
4K limit removed from avio_printf.
2019-08-17 - a82f8f2f10 - lavf 58.31.100 - avio.h
Add avio_print_string_array and avio_print.
2019-07-27 - 42e2319ba9 - lavu 56.33.100 - tx.h
Add AV_TX_DOUBLE_FFT and AV_TX_DOUBLE_MDCT
-------- 8< --------- FFmpeg 4.2 was cut here -------- 8< ---------
2019-06-21 - a30e44098a - lavu 56.30.100 - frame.h
Add FF_DECODE_ERROR_DECODE_SLICES
2019-06-14 - edfced8c04 - lavu 56.29.100 - frame.h
Add FF_DECODE_ERROR_CONCEALMENT_ACTIVE
2019-05-15 - b79b29ddb1 - lavu 56.28.100 - tx.h
Add av_tx_init(), av_tx_uninit() and related definitions.
2019-04-20 - 3153a6502a - lavc 58.52.100 - avcodec.h
Add AV_CODEC_FLAG_DROPCHANGED to allow avcodec_receive_frame to drop
frames whose parameters differ from first decoded frame in stream.
2019-04-12 - abfeba9724 - lavf 58.27.102
Rename hls,applehttp demuxer to hls
2019-01-27 - 5bcefceec8 - lavc 58.46.100 - avcodec.h
Add discard_damaged_percentage
2019-01-08 - 1ef4828276 - lavu 56.26.100 - frame.h
Add AV_FRAME_DATA_REGIONS_OF_INTEREST
2018-12-21 - 2744d6b364 - lavu 56.25.100 - hdr_dynamic_metadata.h
Add AV_FRAME_DATA_DYNAMIC_HDR_PLUS enum value, av_dynamic_hdr_plus_alloc(),
av_dynamic_hdr_plus_create_side_data() functions, and related structs.
-------- 8< --------- FFmpeg 4.1 was cut here -------- 8< ---------
2018-10-27 - 718044dc19 - lavu 56.21.100 - pixdesc.h
Add av_read_image_line2(), av_write_image_line2()
2018-10-24 - f9d4126f28 - lavu 56.20.100 - frame.h
Add AV_FRAME_DATA_S12M_TIMECODE
2018-10-11 - f6d48b618a - lavc 58.33.100 - mediacodec.h
Add av_mediacodec_render_buffer_at_time().
2018-09-09 - 35498c124a - lavc 58.29.100 - avcodec.h
Add AV_PKT_DATA_AFD
2018-08-16 - b33f5299a5 - lavc 58.23.100 - avcodec.h
Add av_bsf_flush().
2018-05-18 - 2b2f2f65f3 - lavf 58.15.100 - avformat.h
Add pmt_version field to AVProgram
2018-05-17 - 5dfeb7f081 - lavf 58.14.100 - avformat.h
Add AV_DISPOSITION_STILL_IMAGE
2018-05-10 - c855683427 - lavu 56.18.101 - hwcontext_cuda.h
Add AVCUDADeviceContext.stream.
2018-04-30 - 56b081da57 - lavu 56.18.100 - pixdesc.h
Add AV_PIX_FMT_FLAG_ALPHA to AV_PIX_FMT_PAL8.
2018-04-26 - 5be0410cb3 - lavu 56.17.100 - opt.h
Add AV_OPT_FLAG_DEPRECATED.
2018-04-26 - 71fa82bed6 - lavu 56.16.100 - threadmessage.h
Add av_thread_message_queue_nb_elems().
-------- 8< --------- FFmpeg 4.0 was cut here -------- 8< ---------
2018-04-03 - d6fc031caf - lavu 56.13.100 - pixdesc.h

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 = 4.3.6
PROJECT_NUMBER = 4.0.3
# 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

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

@@ -37,61 +37,6 @@ raw ADTS AAC or an MPEG-TS container to MP4A-LATM, to an FLV file, or
to MOV/MP4 files and related formats such as 3GP or M4A. Please note
that it is auto-inserted for MP4A-LATM and MOV/MP4 and related formats.
@section av1_metadata
Modify metadata embedded in an AV1 stream.
@table @option
@item td
Insert or remove temporal delimiter OBUs in all temporal units of the
stream.
@table @samp
@item insert
Insert a TD at the beginning of every TU which does not already have one.
@item remove
Remove the TD from the beginning of every TU which has one.
@end table
@item color_primaries
@item transfer_characteristics
@item matrix_coefficients
Set the color description fields in the stream (see AV1 section 6.4.2).
@item color_range
Set the color range in the stream (see AV1 section 6.4.2; note that
this cannot be set for streams using BT.709 primaries, sRGB transfer
characteristic and identity (RGB) matrix coefficients).
@table @samp
@item tv
Limited range.
@item pc
Full range.
@end table
@item chroma_sample_position
Set the chroma sample location in the stream (see AV1 section 6.4.2).
This can only be set for 4:2:0 streams.
@table @samp
@item vertical
Left position (matching the default in MPEG-2 and H.264).
@item colocated
Top-left position.
@end table
@item tick_rate
Set the tick rate (@emph{num_units_in_display_tick / time_scale}) in
the timing info in the sequence header.
@item num_ticks_per_picture
Set the number of ticks in each picture, to indicate that the stream
has a fixed framerate. Ignored if @option{tick_rate} is not also set.
@item delete_padding
Deletes Padding OBUs.
@end table
@section chomp
Remove zero padding at the end of a packet.
@@ -103,9 +48,7 @@ DTS-HD.
@section dump_extra
Add extradata to the beginning of the filtered packets except when
said packets already exactly begin with the extradata that is intended
to be added.
Add extradata to the beginning of the filtered packets.
@table @option
@item freq
@@ -122,7 +65,7 @@ add extradata to all packets
@end table
@end table
If not specified it is assumed @samp{k}.
If not specified it is assumed @samp{e}.
For example the following @command{ffmpeg} command forces a global
header (thus disabling individual packet headers) in the H.264 packets
@@ -224,10 +167,6 @@ Insert or remove AUD NAL units in all access units of the stream.
@item sample_aspect_ratio
Set the sample aspect ratio of the stream in the VUI parameters.
@item overscan_appropriate_flag
Set whether the stream is suitable for display using overscan
or not (see H.264 section E.2.1).
@item video_format
@item video_full_range_flag
Set the video format in the stream (see H.264 section E.2.1 and
@@ -276,15 +215,6 @@ insert the string ``hello'' associated with the given UUID.
@item delete_filler
Deletes both filler NAL units and filler SEI messages.
@item level
Set the level in the SPS. Refer to H.264 section A.3 and tables A-1
to A-5.
The argument must be the name of a level (for example, @samp{4.2}), a
level_idc value (for example, @samp{42}), or the special name @samp{auto}
indicating that the filter should attempt to guess the level from the
input stream properties.
@end table
@section h264_mp4toannexb
@@ -367,15 +297,6 @@ will replace the current ones if the stream is already cropped.
These fields are set in pixels. Note that some sizes may not be
representable if the chroma is subsampled (H.265 section 7.4.3.2.1).
@item level
Set the level in the VPS and SPS. See H.265 section A.4 and tables
A.6 and A.7.
The argument must be the name of a level (for example, @samp{5.1}), a
@emph{general_level_idc} value (for example, @samp{153} for level 5.1),
or the special name @samp{auto} indicating that the filter should
attempt to guess the level from the input stream properties.
@end table
@section hevc_mp4toannexb
@@ -548,111 +469,6 @@ ffmpeg -i INPUT -c copy -bsf noise[=1] output.mkv
@section null
This bitstream filter passes the packets through unchanged.
@section pcm_rechunk
Repacketize PCM audio to a fixed number of samples per packet or a fixed packet
rate per second. This is similar to the @ref{asetnsamples,,asetnsamples audio
filter,ffmpeg-filters} but works on audio packets instead of audio frames.
@table @option
@item nb_out_samples, n
Set the number of samples per each output audio packet. The number is intended
as the number of samples @emph{per each channel}. Default value is 1024.
@item pad, p
If set to 1, the filter will pad the last audio packet with silence, so that it
will contain the same number of samples (or roughly the same number of samples,
see @option{frame_rate}) as the previous ones. Default value is 1.
@item frame_rate, r
This option makes the filter output a fixed number of packets per second instead
of a fixed number of samples per packet. If the audio sample rate is not
divisible by the frame rate then the number of samples will not be constant but
will vary slightly so that each packet will start as close to the frame
boundary as possible. Using this option has precedence over @option{nb_out_samples}.
@end table
You can generate the well known 1602-1601-1602-1601-1602 pattern of 48kHz audio
for NTSC frame rate using the @option{frame_rate} option.
@example
ffmpeg -f lavfi -i sine=r=48000:d=1 -c pcm_s16le -bsf pcm_rechunk=r=30000/1001 -f framecrc -
@end example
@section prores_metadata
Modify color property metadata embedded in prores stream.
@table @option
@item color_primaries
Set the color primaries.
Available values are:
@table @samp
@item auto
Keep the same color primaries property (default).
@item unknown
@item bt709
@item bt470bg
BT601 625
@item smpte170m
BT601 525
@item bt2020
@item smpte431
DCI P3
@item smpte432
P3 D65
@end table
@item transfer_characteristics
Set the color transfer.
Available values are:
@table @samp
@item auto
Keep the same transfer characteristics property (default).
@item unknown
@item bt709
BT 601, BT 709, BT 2020
@item smpte2084
SMPTE ST 2084
@item arib-std-b67
ARIB STD-B67
@end table
@item matrix_coefficients
Set the matrix coefficient.
Available values are:
@table @samp
@item auto
Keep the same colorspace property (default).
@item unknown
@item bt709
@item smpte170m
BT 601
@item bt2020nc
@end table
@end table
Set Rec709 colorspace for each frame of the file
@example
ffmpeg -i INPUT -c copy -bsf:v prores_metadata=color_primaries=bt709:color_trc=bt709:colorspace=bt709 output.mov
@end example
Set Hybrid Log-Gamma parameters for each frame of the file
@example
ffmpeg -i INPUT -c copy -bsf:v prores_metadata=color_primaries=bt2020:color_trc=arib-std-b67:colorspace=bt2020nc output.mov
@end example
@section remove_extra
Remove extradata from packets.
@@ -689,40 +505,7 @@ Log trace output containing all syntax elements in the coded stream
headers (everything above the level of individual coded blocks).
This can be useful for debugging low-level stream issues.
Supports AV1, H.264, H.265, (M)JPEG, MPEG-2 and VP9, but depending
on the build only a subset of these may be available.
@section truehd_core
Extract the core from a TrueHD stream, dropping ATMOS data.
@section vp9_metadata
Modify metadata embedded in a VP9 stream.
@table @option
@item color_space
Set the color space value in the frame header. Note that any frame
set to RGB will be implicitly set to PC range and that RGB is
incompatible with profiles 0 and 2.
@table @samp
@item unknown
@item bt601
@item bt709
@item smpte170
@item smpte240
@item bt2020
@item rgb
@end table
@item color_range
Set the color range value in the frame header. Note that any value
imposed by the color space will take precedence over this value.
@table @samp
@item tv
@item pc
@end table
@end table
Supports H.264, H.265 and MPEG-2.
@section vp9_superframe

View File

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

View File

@@ -55,10 +55,6 @@ Do not draw edges.
@item psnr
Set error[?] variables during encoding.
@item truncated
Input bitstream might be randomly truncated.
@item drop_changed
Don't output frames whose parameters differ from first decoded frame in stream.
Error AVERROR_INPUT_CHANGED is returned when a frame is dropped.
@item ildct
Use interlaced DCT.
@@ -80,8 +76,6 @@ Deprecated, use mpegvideo private options instead.
Apply interlaced motion estimation.
@item cgop
Use closed gop.
@item output_corrupt
Output even potentially corrupted frames.
@end table
@item me_method @var{integer} (@emph{encoding,video})
@@ -646,24 +640,6 @@ noise preserving sum of squared differences
@item dia_size @var{integer} (@emph{encoding,video})
Set diamond type & size for motion estimation.
@table @samp
@item (1024, INT_MAX)
full motion estimation(slowest)
@item (768, 1024]
umh motion estimation
@item (512, 768]
hex motion estimation
@item (256, 512]
l2s diamond motion estimation
@item [2,256]
var diamond motion estimation
@item (-1, 2)
small diamond motion estimation
@item -1
funny diamond motion estimation
@item (INT_MIN, -1)
sab diamond motion estimation
@end table
@item last_pred @var{integer} (@emph{encoding,video})
Set amount of motion predictors from the previous frame.
@@ -781,12 +757,14 @@ Set noise reduction.
Set number of bits which should be loaded into the rc buffer before
decoding starts.
@item flags2 @var{flags} (@emph{decoding/encoding,audio,video,subtitles})
@item flags2 @var{flags} (@emph{decoding/encoding,audio,video})
Possible values:
@table @samp
@item fast
Allow non spec compliant speedup tricks.
@item sgop
Deprecated, use mpegvideo private options instead.
@item noout
Skip bitstream encoding.
@item ignorecrop
@@ -797,25 +775,11 @@ Place global headers at every keyframe instead of in extradata.
Frame data might be split into multiple chunks.
@item showall
Show all frames before the first keyframe.
@item skiprd
Deprecated, use mpegvideo private options instead.
@item export_mvs
Export motion vectors into frame side-data (see @code{AV_FRAME_DATA_MOTION_VECTORS})
for codecs that support it. See also @file{doc/examples/export_mvs.c}.
@item skip_manual
Do not skip samples and export skip information as frame side data.
@item ass_ro_flush_noop
Do not reset ASS ReadOrder field on flush.
@end table
@item export_side_data @var{flags} (@emph{decoding/encoding,audio,video,subtitles})
Possible values:
@table @samp
@item mvs
Export motion vectors into frame side-data (see @code{AV_FRAME_DATA_MOTION_VECTORS})
for codecs that support it. See also @file{doc/examples/export_mvs.c}.
@item prft
Export encoder Producer Reference Time into packet side-data (see @code{AV_PKT_DATA_PRFT})
for codecs that support it.
@end table
@item error @var{integer} (@emph{encoding,video})
@@ -855,8 +819,49 @@ Set number of macroblock rows at the bottom which are skipped.
@item profile @var{integer} (@emph{encoding,audio,video})
Set encoder codec profile. Default value is @samp{unknown}. Encoder specific
profiles are documented in the relevant encoder documentation.
Possible values:
@table @samp
@item unknown
@item aac_main
@item aac_low
@item aac_ssr
@item aac_ltp
@item aac_he
@item aac_he_v2
@item aac_ld
@item aac_eld
@item mpeg2_aac_low
@item mpeg2_aac_he
@item mpeg4_sp
@item mpeg4_core
@item mpeg4_main
@item mpeg4_asp
@item dts
@item dts_es
@item dts_96_24
@item dts_hd_hra
@item dts_hd_ma
@end table
@item level @var{integer} (@emph{encoding,audio,video})
@@ -957,9 +962,6 @@ Discard all bidirectional frames.
@item nokey
Discard all frames excepts keyframes.
@item nointra
Discard all frames except I frames.
@item all
Discard all frames.
@end table
@@ -984,6 +986,10 @@ Set chroma qp offset from luma.
@item trellis @var{integer} (@emph{encoding,audio,video})
Set rate-distortion optimal quantization.
@item sc_factor @var{integer} (@emph{encoding,video})
Set value multiplied by qscale for each frame and added to
scene_change_score.
@item mv0_threshold @var{integer} (@emph{encoding,video})
@item b_sensitivity @var{integer} (@emph{encoding,video})
Adjust sensitivity of b_frame_strategy 1.
@@ -1230,7 +1236,7 @@ instead of alpha. Default is 0.
@item dump_separator @var{string} (@emph{input})
Separator used to separate the fields printed on the command line about the
Stream parameters.
For example, to separate the fields with newlines and indentation:
For example to separate the fields with newlines and indention:
@example
ffprobe -dump_separator "
" -i ~/videos/matrixbench_mpeg2.mpg

View File

@@ -47,45 +47,6 @@ top-field-first is assumed
@end table
@section libdav1d
dav1d AV1 decoder.
libdav1d allows libavcodec to decode the AOMedia Video 1 (AV1) codec.
Requires the presence of the libdav1d headers and library during configuration.
You need to explicitly configure the build with @code{--enable-libdav1d}.
@subsection Options
The following options are supported by the libdav1d wrapper.
@table @option
@item framethreads
Set amount of frame threads to use during decoding. The default value is 0 (autodetect).
@item tilethreads
Set amount of tile threads to use during decoding. 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.
@item oppoint
Select an operating point of a scalable AV1 bitstream (0 - 31). Defaults to the
internal default of the library.
@item alllayers
Output all spatial layers of a scalable AV1 bitstream. The default value is false.
@end table
@section libdavs2
AVS2-P2/IEEE1857.4 video decoder wrapper.
This decoder allows libavcodec to decode AVS2 streams with davs2 library.
@c man end VIDEO DECODERS
@chapter Audio Decoders
@@ -227,31 +188,6 @@ without this library.
@chapter Subtitles Decoders
@c man begin SUBTILES DECODERS
@section libaribb24
ARIB STD-B24 caption decoder.
Implements profiles A and C of the ARIB STD-B24 standard.
@subsection libaribb24 Decoder Options
@table @option
@item -aribb24-base-path @var{path}
Sets the base path for the libaribb24 library. This is utilized for reading of
configuration files (for custom unicode conversions), and for dumping of
non-text symbols as images under that location.
Unset by default.
@item -aribb24-skip-ruby-text @var{boolean}
Tells the decoder wrapper to skip text blocks that contain half-height ruby
text.
Enabled by default.
@end table
@section dvbsub
@subsection Options
@@ -287,7 +223,7 @@ palette is stored in the IFO file, and therefore not available when reading
from dumped VOB files.
The format for this option is a string containing 16 24-bits hexadecimal
numbers (without 0x prefix) separated by commas, for example @code{0d00ee,
numbers (without 0x prefix) separated by comas, for example @code{0d00ee,
ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1,
7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b}.
@@ -312,30 +248,18 @@ configuration. You need to explicitly configure the build with
@table @option
@item txt_page
List of teletext page numbers to decode. Pages that do not match the specified
list are dropped. You may use the special @code{*} string to match all pages,
or @code{subtitle} to match all subtitle pages.
List of teletext page numbers to decode. You may use the special * string to
match all pages. Pages that do not match the specified list are dropped.
Default value is *.
@item txt_default_region
Set default character set used for decoding, a value between 0 and 87 (see
ETS 300 706, Section 15, Table 32). Default value is -1, which does not
override the libzvbi default. This option is needed for some legacy level 1.0
transmissions which cannot signal the proper charset.
@item txt_chop_top
Discards the top teletext line. Default value is 1.
@item txt_format
Specifies the format of the decoded subtitles.
@table @option
@item bitmap
The default format, you should use this for teletext pages, because certain
graphics and colors cannot be expressed in simple text or even ASS.
@item text
Simple text based output without formatting.
@item ass
Formatted ASS output, subtitle pages and teletext pages are returned in
different styles, subtitle pages are stripped down to text, but an effort is
made to keep the text alignment and the formatting.
@end table
Specifies the format of the decoded subtitles. The teletext decoder is capable
of decoding the teletext pages to bitmaps or to simple text, you should use
"bitmap" for teletext pages, because certain graphics and colors cannot be
expressed in simple text. You might use "text" for teletext based subtitles if
your application can handle simple text based subtitles. Default value is
bitmap.
@item txt_left
X offset of generated bitmaps, default is 0.
@item txt_top
@@ -348,8 +272,7 @@ present between the subtitle lines because of double-sized teletext characters.
Default value is 1.
@item txt_duration
Sets the display duration of the decoded teletext pages or subtitles in
milliseconds. Default value is -1 which means infinity or until the next
subtitle event comes.
milliseconds. Default value is 30000 which is 30 seconds.
@item txt_transparent
Force transparent background of the generated teletext bitmaps. Default value
is 0 which means an opaque background.

View File

@@ -25,6 +25,17 @@ Audible Format 2, 3, and 4 demuxer.
This demuxer is used to demux Audible Format 2, 3, and 4 (.aa) files.
@section applehttp
Apple HTTP Live Streaming demuxer.
This demuxer presents all AVStreams from all variant streams.
The id field is set to the bitrate variant index number. By setting
the discard flags on AVStreams (by pressing 'a' or 'v' in ffplay),
the caller can decide which variant streams to actually receive.
The total bitrate of the variant that the stream belongs to is
available in a metadata key named "variant_bitrate".
@section apng
Animated Portable Network Graphics demuxer.
@@ -258,12 +269,6 @@ ffmpeg -f live_flv -i rtmp://<any.server>/anything/key ....
@table @option
@item -flv_metadata @var{bool}
Allocate the streams according to the onMetaData array content.
@item -flv_ignore_prevtag @var{bool}
Ignore the size of previous tag value.
@item -flv_full_metadata @var{bool}
Output all context of the onMetadata.
@end table
@section gif
@@ -309,15 +314,6 @@ infinitely.
HLS demuxer
Apple HTTP Live Streaming demuxer.
This demuxer presents all AVStreams from all variant streams.
The id field is set to the bitrate variant index number. By setting
the discard flags on AVStreams (by pressing 'a' or 'v' in ffplay),
the caller can decide which variant streams to actually receive.
The total bitrate of the variant that the stream belongs to is
available in a metadata key named "variant_bitrate".
It accepts the following options:
@table @option
@@ -331,10 +327,6 @@ segment index to start live streams at (negative values are from the end).
Maximum number of times a insufficient list is attempted to be reloaded.
Default value is 1000.
@item m3u8_hold_counters
The maximum number of times to load m3u8 when it refreshes without new segments.
Default value is 1000.
@item http_persistent
Use persistent HTTP connections. Applicable only for HTTP streams.
Enabled by default.
@@ -342,10 +334,6 @@ Enabled by default.
@item http_multiple
Use multiple HTTP connections for downloading HTTP segments.
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.
@end table
@section image2
@@ -456,17 +444,6 @@ nanosecond precision.
@item video_size
Set the video size of the images to read. If not specified the video
size is guessed from the first image file in the sequence.
@item export_path_metadata
If set to 1, will add two extra fields to the metadata found in input, making them
also available for other filters (see @var{drawtext} filter for examples). Default
value is 0. The extra fields are described below:
@table @option
@item lavf.image2dec.source_path
Corresponds to the full path to the input file being read.
@item lavf.image2dec.source_basename
Corresponds to the name of the file being read.
@end table
@end table
@subsection Examples
@@ -498,84 +475,14 @@ ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv
The Game Music Emu library is a collection of video game music file emulators.
See @url{https://bitbucket.org/mpyne/game-music-emu/overview} for more information.
See @url{http://code.google.com/p/game-music-emu/} for more information.
It accepts the following options:
Some files have multiple tracks. The demuxer will pick the first track by
default. The @option{track_index} option can be used to select a different
track. Track indexes start at 0. The demuxer exports the number of tracks as
@var{tracks} meta data entry.
@table @option
@item track_index
Set the index of which track to demux. The demuxer can only export one track.
Track indexes start at 0. Default is to pick the first track. Number of tracks
is exported as @var{tracks} metadata entry.
@item sample_rate
Set the sampling rate of the exported track. Range is 1000 to 999999. Default is 44100.
@item max_size @emph{(bytes)}
The demuxer buffers the entire file into memory. Adjust this value to set the maximum buffer size,
which in turn, acts as a ceiling for the size of files that can be read.
Default is 50 MiB.
@end table
@section libmodplug
ModPlug based module demuxer
See @url{https://github.com/Konstanty/libmodplug}
It will export one 2-channel 16-bit 44.1 kHz audio stream.
Optionally, a @code{pal8} 16-color video stream can be exported with or without printed metadata.
It accepts the following options:
@table @option
@item noise_reduction
Apply a simple low-pass filter. Can be 1 (on) or 0 (off). Default is 0.
@item reverb_depth
Set amount of reverb. Range 0-100. Default is 0.
@item reverb_delay
Set delay in ms, clamped to 40-250 ms. Default is 0.
@item bass_amount
Apply bass expansion a.k.a. XBass or megabass. Range is 0 (quiet) to 100 (loud). Default is 0.
@item bass_range
Set cutoff i.e. upper-bound for bass frequencies. Range is 10-100 Hz. Default is 0.
@item surround_depth
Apply a Dolby Pro-Logic surround effect. Range is 0 (quiet) to 100 (heavy). Default is 0.
@item surround_delay
Set surround delay in ms, clamped to 5-40 ms. Default is 0.
@item max_size
The demuxer buffers the entire file into memory. Adjust this value to set the maximum buffer size,
which in turn, acts as a ceiling for the size of files that can be read. Range is 0 to 100 MiB.
0 removes buffer size limit (not recommended). Default is 5 MiB.
@item video_stream_expr
String which is evaluated using the eval API to assign colors to the generated video stream.
Variables which can be used are @code{x}, @code{y}, @code{w}, @code{h}, @code{t}, @code{speed},
@code{tempo}, @code{order}, @code{pattern} and @code{row}.
@item video_stream
Generate video stream. Can be 1 (on) or 0 (off). Default is 0.
@item video_stream_w
Set video frame width in 'chars' where one char indicates 8 pixels. Range is 20-512. Default is 30.
@item video_stream_h
Set video frame height in 'chars' where one char indicates 8 pixels. Range is 20-512. Default is 30.
@item video_stream_ptxt
Print metadata on video stream. Includes @code{speed}, @code{tempo}, @code{order}, @code{pattern},
@code{row} and @code{ts} (time in ms). Can be 1 (on) or 0 (off). Default is 1.
@end table
For very large files, the @option{max_size} option may have to be adjusted.
@section libopenmpt
@@ -604,13 +511,9 @@ Set the sample rate for libopenmpt to output.
Range is from 1000 to INT_MAX. The value default is 48000.
@end table
@section mov/mp4/3gp
@section mov/mp4/3gp/QuickTime
Demuxer for Quicktime File Format & ISO/IEC Base Media File Format (ISO/IEC 14496-12 or MPEG-4 Part 12, ISO/IEC 15444-12 or JPEG 2000 Part 12).
Registered extensions: mov, mp4, m4a, 3gp, 3g2, mj2, psp, m4b, ism, ismv, isma, f4v
@subsection Options
QuickTime / MP4 demuxer.
This demuxer accepts the following options:
@table @option
@@ -621,73 +524,10 @@ Enabling this can theoretically leak information in some use cases.
@item use_absolute_path
Allows loading of external tracks via absolute paths, disabled by default.
Enabling this poses a security risk. It should only be enabled if the source
is known to be non-malicious.
is known to be non malicious.
@item seek_streams_individually
When seeking, identify the closest point in each stream individually and demux packets in
that stream from identified point. This can lead to a different sequence of packets compared
to demuxing linearly from the beginning. Default is true.
@item ignore_editlist
Ignore any edit list atoms. The demuxer, by default, modifies the stream index to reflect the
timeline described by the edit list. Default is false.
@item advanced_editlist
Modify the stream index to reflect the timeline described by the edit list. @code{ignore_editlist}
must be set to false for this option to be effective.
If both @code{ignore_editlist} and this option are set to false, then only the
start of the stream index is modified to reflect initial dwell time or starting timestamp
described by the edit list. Default is true.
@item ignore_chapters
Don't parse chapters. This includes GoPro 'HiLight' tags/moments. Note that chapters are
only parsed when input is seekable. Default is false.
@item use_mfra_for
For seekable fragmented input, set fragment's starting timestamp from media fragment random access box, if present.
Following options are available:
@table @samp
@item auto
Auto-detect whether to set mfra timestamps as PTS or DTS @emph{(default)}
@item dts
Set mfra timestamps as DTS
@item pts
Set mfra timestamps as PTS
@item 0
Don't use mfra box to set timestamps
@end table
@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.
@item export_xmp
Export entire contents of @var{XMP_} box and @var{uuid} box as a string with key @code{xmp}. Note that
if @code{export_all} is set and this option isn't, the contents of @var{XMP_} box are still exported
but with key @code{XMP_}. Default is false.
@item activation_bytes
4-byte key required to decrypt Audible AAX and AAX+ files. See Audible AAX subsection below.
@item audible_fixed_key
Fixed key used for handling Audible AAX/AAX+ files. It has been pre-set so should not be necessary to
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).
@end table
@subsection Audible AAX
Audible AAX files are encrypted M4B files, and they can be decrypted by specifying a 4 byte activation secret.
@example
ffmpeg -activation_bytes 1CEB00DA -i test.aax -vn -c:a copy output.mp4
@end example
@section mpegts
MPEG-2 transport stream demuxer.
@@ -698,9 +538,6 @@ This demuxer accepts the following options:
Set size limit for looking up a new synchronization. Default value is
65536.
@item skip_unknown_pmt
Skip PMTs for programs not defined in the PAT. Default value is 0.
@item fix_teletext_pts
Override teletext packet PTS and DTS values with the timestamps calculated
from the PCR of the first program which the teletext stream is part of and is
@@ -715,10 +552,6 @@ Show the detected raw packet size, cannot be set by the user.
Scan and combine all PMTs. The value is an integer with value from -1
to 1 (-1 means automatic setting, 1 means enabled, 0 means
disabled). Default value is -1.
@item merge_pmt_versions
Re-use existing streams when a PMT's version is updated and elementary
streams move to different PIDs. Default value is 0.
@end table
@section mpjpeg
@@ -816,20 +649,4 @@ Example: convert the captions to a format most players understand:
ffmpeg -i http://www.ted.com/talks/subtitles/id/1/lang/en talk1-en.srt
@end example
@section vapoursynth
Vapoursynth wrapper.
Due to security concerns, Vapoursynth scripts will not
be autodetected so the input format has to be forced. For ff* CLI tools,
add @code{-f vapoursynth} before the input @code{-i yourscript.vpy}.
This demuxer accepts the following option:
@table @option
@item max_script_size
The demuxer buffers the entire script into memory. Adjust this value to set the maximum buffer size,
which in turn, acts as a ceiling for the size of scripts that can be read.
Default is 1 MiB.
@end table
@c man end DEMUXERS

View File

@@ -128,12 +128,6 @@ designated struct initializers (@samp{struct s x = @{ .i = 17 @};});
@item
compound literals (@samp{x = (struct s) @{ 17, 23 @};}).
@item
for loops with variable definition (@samp{for (int i = 0; i < 8; i++)});
@item
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
@@ -625,7 +619,7 @@ If the patch fixes a bug, did you provide a verbose analysis of the bug?
If the patch fixes a bug, did you provide enough information, including
a sample, so the bug can be reproduced and the fix can be verified?
Note please do not attach samples >100k to mails but rather provide a
URL, you can upload to @url{https://streams.videolan.org/upload/}.
URL, you can upload to ftp://upload.ffmpeg.org.
@item
Did you provide a verbose summary about what the patch does change?

View File

@@ -30,7 +30,11 @@ follows.
Advanced Audio Coding (AAC) encoder.
This encoder is the default AAC encoder, natively implemented into FFmpeg.
This encoder is the default AAC encoder, natively implemented into FFmpeg. Its
quality is on par or better than libfdk_aac at the default bitrate of 128kbps.
This encoder also implements more options, profiles and samplerates than
other encoders (with only the AAC-HE profile pending to be implemented) so this
encoder has become the default and is the recommended choice.
@subsection Options
@@ -647,7 +651,10 @@ configuration. You need to explicitly configure the build with
so if you allow the use of GPL, you should configure with
@code{--enable-gpl --enable-nonfree --enable-libfdk-aac}.
This encoder has support for the AAC-HE profiles.
This encoder is considered to produce output on par or worse at 128kbps to the
@ref{aacenc,,the native FFmpeg AAC encoder} but can often produce better
sounding audio at identical or lower bitrates and has support for the
AAC-HE profiles.
VBR encoding, enabled through the @option{vbr} or @option{flags
+qscale} options, is experimental and only works with some
@@ -726,14 +733,6 @@ if set to 0.
Default value is 0.
@item eld_v2
Enable ELDv2 (LD-MPS extension for ELD stereo signals) for ELDv2 if set to 1,
disabled if set to 0.
Note that option is available when fdk-aac version (AACENCODER_LIB_VL0.AACENCODER_LIB_VL1.AACENCODER_LIB_VL2) > (4.0.0).
Default value is 0.
@item signaling
Set SBR/PS signaling style.
@@ -1371,236 +1370,6 @@ makes it possible to store non-rgb pix_fmts.
@end table
@section librav1e
rav1e AV1 encoder wrapper.
Requires the presence of the rav1e headers and library during configuration.
You need to explicitly configure the build with @code{--enable-librav1e}.
@subsection Options
@table @option
@item qmax
Sets the maximum quantizer to use when using bitrate mode.
@item qmin
Sets the minimum quantizer to use when using bitrate mode.
@item qp
Uses quantizer mode to encode at the given quantizer (0-255).
@item speed
Selects the speed preset (0-10) to encode with.
@item tiles
Selects how many tiles to encode with.
@item tile-rows
Selects how many rows of tiles to encode with.
@item tile-columns
Selects how many columns of tiles to encode with.
@item rav1e-params
Set rav1e options using a list of @var{key}=@var{value} pairs separated
by ":". See @command{rav1e --help} for a list of options.
For example to specify librav1e encoding options with @option{-rav1e-params}:
@example
ffmpeg -i input -c:v librav1e -b:v 500K -rav1e-params speed=5:low_latency=true output.mp4
@end example
@end table
@section libaom-av1
libaom AV1 encoder wrapper.
Requires the presence of the libaom headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libaom}.
@subsection Options
The wrapper supports the following standard libavcodec options:
@table @option
@item b
Set bitrate target in bits/second. By default this will use
variable-bitrate mode. If @option{maxrate} and @option{minrate} are
also set to the same value then it will use constant-bitrate mode,
otherwise if @option{crf} is set as well then it will use
constrained-quality mode.
@item g keyint_min
Set key frame placement. The GOP size sets the maximum distance between
key frames; if zero the output stream will be intra-only. The minimum
distance is ignored unless it is the same as the GOP size, in which case
key frames will always appear at a fixed interval. Not set by default,
so without this option the library has completely free choice about
where to place key frames.
@item qmin qmax
Set minimum/maximum quantisation values. Valid range is from 0 to 63
(warning: this does not match the quantiser values actually used by AV1
- divide by four to map real quantiser values to this range). Defaults
to min/max (no constraint).
@item minrate maxrate bufsize rc_init_occupancy
Set rate control buffering parameters. Not used if not set - defaults
to unconstrained variable bitrate.
@item threads
Set the number of threads to use while encoding. This may require the
@option{tiles} or @option{row-mt} options to also be set to actually
use the specified number of threads fully. Defaults to the number of
hardware threads supported by the host machine.
@item profile
Set the encoding profile. Defaults to using the profile which matches
the bit depth and chroma subsampling of the input.
@end table
The wrapper also has some specific options:
@table @option
@item cpu-used
Set the quality/encoding speed tradeoff. Valid range is from 0 to 8,
higher numbers indicating greater speed and lower quality. The default
value is 1, which will be slow and high quality.
@item auto-alt-ref
Enable use of alternate reference frames. Defaults to the internal
default of the library.
@item arnr-max-frames (@emph{frames})
Set altref noise reduction max frame count. Default is -1.
@item arnr-strength (@emph{strength})
Set altref noise reduction filter strength. Range is -1 to 6. Default is -1.
@item aq-mode (@emph{aq-mode})
Set adaptive quantization mode. Possible values:
@table @samp
@item none (@emph{0})
Disabled.
@item variance (@emph{1})
Variance-based.
@item complexity (@emph{2})
Complexity-based.
@item cyclic (@emph{3})
Cyclic refresh.
@end table
@item tune (@emph{tune})
Set the distortion metric the encoder is tuned with. Default is @code{psnr}.
@table @samp
@item psnr (@emph{0})
@item ssim (@emph{1})
@end table
@item lag-in-frames
Set the maximum number of frames which the encoder may keep in flight
at any one time for lookahead purposes. Defaults to the internal
default of the library.
@item error-resilience
Enable error resilience features:
@table @option
@item default
Improve resilience against losses of whole frames.
@end table
Not enabled by default.
@item crf
Set the quality/size tradeoff for constant-quality (no bitrate target)
and constrained-quality (with maximum bitrate target) modes. Valid
range is 0 to 63, higher numbers indicating lower quality and smaller
output size. Only used if set; by default only the bitrate target is
used.
@item static-thresh
Set a change threshold on blocks below which they will be skipped by
the encoder. Defined in arbitrary units as a nonnegative integer,
defaulting to zero (no blocks are skipped).
@item drop-threshold
Set a threshold for dropping frames when close to rate control bounds.
Defined as a percentage of the target buffer - when the rate control
buffer falls below this percentage, frames will be dropped until it
has refilled above the threshold. Defaults to zero (no frames are
dropped).
@item denoise-noise-level (@emph{level})
Amount of noise to be removed for grain synthesis. Grain synthesis is disabled if
this option is not set or set to 0.
@item denoise-block-size (@emph{pixels})
Block size used for denoising for grain synthesis. If not set, AV1 codec
uses the default value of 32.
@item undershoot-pct (@emph{pct})
Set datarate undershoot (min) percentage of the target bitrate. Range is -1 to 100.
Default is -1.
@item overshoot-pct (@emph{pct})
Set datarate overshoot (max) percentage of the target bitrate. Range is -1 to 1000.
Default is -1.
@item minsection-pct (@emph{pct})
Minimum percentage variation of the GOP bitrate from the target bitrate. If minsection-pct
is not set, the libaomenc wrapper computes it as follows: @code{(minrate * 100 / bitrate)}.
Range is -1 to 100. Default is -1 (unset).
@item maxsection-pct (@emph{pct})
Maximum percentage variation of the GOP bitrate from the target bitrate. If maxsection-pct
is not set, the libaomenc wrapper computes it as follows: @code{(maxrate * 100 / bitrate)}.
Range is -1 to 5000. Default is -1 (unset).
@item frame-parallel (@emph{boolean})
Enable frame parallel decodability features. Default is true.
@item tiles
Set the number of tiles to encode the input video with, as columns x
rows. Larger numbers allow greater parallelism in both encoding and
decoding, but may decrease coding efficiency. Defaults to the minimum
number of tiles required by the size of the input video (this is 1x1
(that is, a single tile) for sizes up to and including 4K).
@item tile-columns tile-rows
Set the number of tiles as log2 of the number of tile rows and columns.
Provided for compatibility with libvpx/VP9.
@item row-mt (Requires libaom >= 1.0.0-759-g90a15f4f2)
Enable row based multi-threading. Disabled by default.
@item enable-cdef (@emph{boolean})
Enable Constrained Directional Enhancement Filter. The libaom-av1
encoder enables CDEF by default.
@item enable-restoration (@emph{boolean})
Enable Loop Restoration Filter. Default is true for libaom-av1.
@item enable-global-motion (@emph{boolean})
Enable the use of global motion for block prediction. Default is true.
@item enable-intrabc (@emph{boolean})
Enable block copy mode for intra block prediction. This mode is
useful for screen content. Default is true.
@end table
@section libkvazaar
Kvazaar H.265/HEVC encoder.
@@ -1872,8 +1641,7 @@ means unlimited.
@table @option
@item auto-alt-ref
Enable use of alternate reference frames (2-pass only).
Values greater than 1 enable multi-layer alternate reference frames (VP9 only).
@item arnr-maxframes
@item arnr-max-frames
Set altref noise reduction max frame count.
@item arnr-type
Set altref noise reduction filter type: backward, forward, centered.
@@ -1886,61 +1654,6 @@ Set number of frames to look ahead for frametype and ratecontrol.
@item error-resilient
Enable error resiliency features.
@item sharpness @var{integer}
Increase sharpness at the expense of lower PSNR.
The valid range is [0, 7].
@item ts-parameters
Sets the temporal scalability configuration using a :-separated list of
key=value pairs. For example, to specify temporal scalability parameters
with @code{ffmpeg}:
@example
ffmpeg -i INPUT -c:v libvpx -ts-parameters ts_number_layers=3:\
ts_target_bitrate=250,500,1000:ts_rate_decimator=4,2,1:\
ts_periodicity=4:ts_layer_id=0,2,1,2:ts_layering_mode=3 OUTPUT
@end example
Below is a brief explanation of each of the parameters, please
refer to @code{struct vpx_codec_enc_cfg} in @code{vpx/vpx_encoder.h} for more
details.
@table @option
@item ts_number_layers
Number of temporal coding layers.
@item ts_target_bitrate
Target bitrate for each temporal layer (in kbps).
(bitrate should be inclusive of the lower temporal layer).
@item ts_rate_decimator
Frame rate decimation factor for each temporal layer.
@item ts_periodicity
Length of the sequence defining frame temporal layer membership.
@item ts_layer_id
Template defining the membership of frames to temporal layers.
@item ts_layering_mode
(optional) Selecting the temporal structure from a set of pre-defined temporal layering modes.
Currently supports the following options.
@table @option
@item 0
No temporal layering flags are provided internally,
relies on flags being passed in using @code{metadata} field in @code{AVFrame}
with following keys.
@table @option
@item vp8-flags
Sets the flags passed into the encoder to indicate the referencing scheme for
the current frame.
Refer to function @code{vpx_codec_encode} in @code{vpx/vpx_encoder.h} for more
details.
@item temporal_id
Explicitly sets the temporal id of the current frame to encode.
@end table
@item 2
Two temporal layers. 0-1...
@item 3
Three temporal layers. 0-2-1-2...; with single reference frame.
@item 4
Same as option "3", except there is a dependency between
the two temporal layer 2 frames within the temporal period.
@end table
@end table
@item VP9-specific options
@table @option
@item lossless
@@ -1979,8 +1692,6 @@ Corpus VBR mode is a variant of standard VBR where the complexity distribution
midpoint is passed in rather than calculated for a specific clip or chunk.
The valid range is [0, 10000]. 0 (default) uses standard VBR.
@item enable-tpl @var{boolean}
Enable temporal dependency model.
@end table
@end table
@@ -2442,20 +2153,6 @@ during configuration. You need to explicitly configure the build with
@subsection Options
@table @option
@item b
Sets target video bitrate.
@item bf
@item g
Set the GOP size.
@item keyint_min
Minimum GOP size.
@item refs
Number of reference frames each P-frame can use. The range is from @var{1-16}.
@item preset
Set the x265 preset.
@@ -2468,28 +2165,6 @@ Set profile restrictions.
@item crf
Set the quality for constant quality mode.
@item qp
Set constant quantization rate control method parameter.
@item qmin
Minimum quantizer scale.
@item qmax
Maximum quantizer scale.
@item qdiff
Maximum difference between quantizer scales.
@item qblur
Quantizer curve blur
@item qcomp
Quantizer curve compression factor
@item i_qfactor
@item b_qfactor
@item forced-idr
Normally, when forcing a I-frame type, the encoder can select any type
of I-frame. This option forces it to choose an IDR-frame.
@@ -2505,63 +2180,6 @@ ffmpeg -i input -c:v libx265 -x265-params crf=26:psy-rd=1 output.mp4
@end example
@end table
@section libxavs2
xavs2 AVS2-P2/IEEE1857.4 encoder wrapper.
This encoder requires the presence of the libxavs2 headers and library
during configuration. You need to explicitly configure the build with
@option{--enable-libxavs2}.
The following standard libavcodec options are used:
@itemize
@item
@option{b} / @option{bit_rate}
@item
@option{g} / @option{gop_size}
@item
@option{bf} / @option{max_b_frames}
@end itemize
The encoder also has its own specific options:
@subsection Options
@table @option
@item lcu_row_threads
Set the number of parallel threads for rows from 1 to 8 (default 5).
@item initial_qp
Set the xavs2 quantization parameter from 1 to 63 (default 34). This is
used to set the initial qp for the first frame.
@item qp
Set the xavs2 quantization parameter from 1 to 63 (default 34). This is
used to set the qp value under constant-QP mode.
@item max_qp
Set the max qp for rate control from 1 to 63 (default 55).
@item min_qp
Set the min qp for rate control from 1 to 63 (default 20).
@item speed_level
Set the Speed level from 0 to 9 (default 0). Higher is better but slower.
@item log_level
Set the log level from -1 to 3 (default 0). -1: none, 0: error,
1: warning, 2: info, 3: debug.
@item xavs2-params
Set xavs2 options using a list of @var{key}=@var{value} couples separated
by ":".
For example to specify libxavs2 encoding options with @option{-xavs2-params}:
@example
ffmpeg -i input -c:v libxavs2 -xavs2-params RdoqLevel=0 output.avs2
@end example
@end table
@section libxvid
Xvid MPEG-4 Part 2 encoder wrapper.
@@ -2725,14 +2343,6 @@ fastest.
@end table
@section MediaFoundation
This provides wrappers to encoders (both audio and video) in the
MediaFoundation framework. It can access both SW and HW encoders.
Video encoders can take input in either of nv12 or yuv420p form
(some encoders support both, some support only either - in practice,
nv12 is the safer choice, especially among HW encoders).
@section mpeg2
MPEG-2 video encoder.
@@ -2740,20 +2350,6 @@ MPEG-2 video encoder.
@subsection Options
@table @option
@item profile @var{integer}
Select the mpeg2 profile to encode:
@table @samp
@item 422
@item main
@item ss
Spatially Scalable
@item snr
SNR Scalable
@item high
@item simple
@end table
@item seq_disp_ext @var{integer}
Specifies if the encoder should write a sequence_display_extension to the
output.
@@ -2774,9 +2370,6 @@ Specifies the video_format written into the sequence display extension
indicating the source of the video pictures. The default is @samp{unspecified},
can be @samp{component}, @samp{pal}, @samp{ntsc}, @samp{secam} or @samp{mac}.
For maximum compatibility, use @samp{component}.
@item a53cc @var{boolean}
Import closed captions (which must be ATSC compatible format) into output.
Default is 1 (on).
@end table
@section png
@@ -2862,7 +2455,7 @@ recommended value) and do not set a size constraint.
@section QSV encoders
The family of Intel QuickSync Video encoders (MPEG-2, H.264, HEVC, JPEG/MJPEG and VP9)
The family of Intel QuickSync Video encoders (MPEG-2, H.264 and HEVC)
The ratecontrol method is selected as follows:
@@ -2972,9 +2565,6 @@ The following standard libavcodec options are used:
@option{bf} / @option{max_b_frames}
@item
@option{profile}
If not set, this will be determined automatically from the format of the input
frames and the profiles supported by the driver.
@item
@option{level}
@item
@@ -2995,8 +2585,7 @@ Speed / quality tradeoff: higher values are faster / worse quality.
Size / quality tradeoff: higher values are smaller / worse quality.
@item
@option{qmin}
@item
@option{qmax}
(only: @option{qmax} is not supported)
@item
@option{i_qfactor} / @option{i_quant_factor}
@item
@@ -3005,55 +2594,8 @@ Size / quality tradeoff: higher values are smaller / worse quality.
@option{b_qfactor} / @option{b_quant_factor}
@item
@option{b_qoffset} / @option{b_quant_offset}
@item
@option{slices}
@end itemize
All encoders support the following options:
@table @option
@item low_power
Some drivers/platforms offer a second encoder for some codecs intended to use
less power than the default encoder; setting this option will attempt to use
that encoder. Note that it may support a reduced feature set, so some other
options may not be available in this mode.
@item idr_interval
Set the number of normal intra frames between full-refresh (IDR) frames in
open-GOP mode. The intra frames are still IRAPs, but will not include global
headers and may have non-decodable leading pictures.
@item b_depth
Set the B-frame reference depth. When set to one (the default), all B-frames
will refer only to P- or I-frames. When set to greater values multiple layers
of B-frames will be present, frames in each layer only referring to frames in
higher layers.
@item rc_mode
Set the rate control mode to use. A given driver may only support a subset of
modes.
Possible modes:
@table @option
@item auto
Choose the mode automatically based on driver support and the other options.
This is the default.
@item CQP
Constant-quality.
@item CBR
Constant-bitrate.
@item VBR
Variable-bitrate.
@item ICQ
Intelligent constant-quality.
@item QVBR
Quality-defined variable-bitrate.
@item AVBR
Average variable bitrate.
@end table
@end table
Each encoder also has its own specific options:
@table @option
@item h264_vaapi
@@ -3061,6 +2603,8 @@ Each encoder also has its own specific options:
@option{level} sets the value of @emph{level_idc}.
@table @option
@item low_power
Use low-power encoding mode.
@item coder
Set entropy encoder (default is @emph{cabac}). Possible values:
@@ -3073,70 +2617,21 @@ Use CABAC.
@item cavlc
Use CAVLC.
@end table
@item aud
Include access unit delimiters in the stream (not included by default).
@item sei
Set SEI message types to include.
Some combination of the following values:
@table @samp
@item identifier
Include a @emph{user_data_unregistered} message containing information about
the encoder.
@item timing
Include picture timing parameters (@emph{buffering_period} and
@emph{pic_timing} messages).
@item recovery_point
Include recovery points where appropriate (@emph{recovery_point} messages).
@end table
@end table
@item hevc_vaapi
@option{profile} and @option{level} set the values of
@emph{general_profile_idc} and @emph{general_level_idc} respectively.
@table @option
@item aud
Include access unit delimiters in the stream (not included by default).
@item tier
Set @emph{general_tier_flag}. This may affect the level chosen for the stream
if it is not explicitly specified.
@item sei
Set SEI message types to include.
Some combination of the following values:
@table @samp
@item hdr
Include HDR metadata if the input frames have it
(@emph{mastering_display_colour_volume} and @emph{content_light_level}
messages).
@end table
@end table
@item mjpeg_vaapi
Only baseline DCT encoding is supported. The encoder always uses the standard
quantisation and huffman tables - @option{global_quality} scales the standard
quantisation table (range 1-100).
For YUV, 4:2:0, 4:2:2 and 4:4:4 subsampling modes are supported. RGB is also
supported, and will create an RGB JPEG.
@table @option
@item jfif
Include JFIF header in each frame (not included by default).
@item huffman
Include standard huffman tables (on by default). Turning this off will save
a few hundred bytes in each output frame, but may lose compatibility with some
JPEG decoders which don't fully handle MJPEG.
@end table
Always encodes using the standard quantisation and huffman tables -
@option{global_quality} scales the standard quantisation table (range 1-100).
@item mpeg2_vaapi
@option{profile} and @option{level} set the value of @emph{profile_and_level_indication}.
No rate control is supported.
@item vp8_vaapi
B-frames are not supported.
@@ -3245,14 +2740,6 @@ and they can also be used in Matroska files.
@subsection Options
@table @option
@item palette
Specify the global palette used by the bitmaps.
The format for this option is a string containing 16 24-bits hexadecimal
numbers (without 0x prefix) separated by commas, for example @code{0d00ee,
ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1,
7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b}.
@item even_rows_fix
When set to 1, enable a work-around that makes the number of pixel rows
even in all subtitles. This fixes a problem with some players that

View File

@@ -1,4 +1,4 @@
/avio_list_dir
/avio_dir_cmd
/avio_reading
/decode_audio
/decode_video
@@ -20,5 +20,3 @@
/scaling_video
/transcode_aac
/transcoding
/vaapi_encode
/vaapi_transcode

View File

@@ -1,4 +1,4 @@
EXAMPLES-$(CONFIG_AVIO_LIST_DIR_EXAMPLE) += avio_list_dir
EXAMPLES-$(CONFIG_AVIO_DIR_CMD_EXAMPLE) += avio_dir_cmd
EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE) += avio_reading
EXAMPLES-$(CONFIG_DECODE_AUDIO_EXAMPLE) += decode_audio
EXAMPLES-$(CONFIG_DECODE_VIDEO_EXAMPLE) += decode_video
@@ -37,7 +37,7 @@ $(EXAMPLES_G): %$(PROGSSUF)_g$(EXESUF): %.o
examples: $(EXAMPLES)
$(EXAMPLES:%$(PROGSSUF)$(EXESUF)=%.o): | doc/examples
OUTDIRS += doc/examples
OBJDIRS += doc/examples
DOXY_INPUT += $(EXAMPLES:%$(PROGSSUF)$(EXESUF)=%.c)

View File

@@ -11,7 +11,7 @@ CFLAGS += -Wall -g
CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
EXAMPLES= avio_list_dir \
EXAMPLES= avio_dir_cmd \
avio_reading \
decode_audio \
decode_video \

View File

@@ -102,15 +102,38 @@ static int list_op(const char *input_dir)
return ret;
}
static int del_op(const char *url)
{
int ret = avpriv_io_delete(url);
if (ret < 0)
av_log(NULL, AV_LOG_ERROR, "Cannot delete '%s': %s.\n", url, av_err2str(ret));
return ret;
}
static int move_op(const char *src, const char *dst)
{
int ret = avpriv_io_move(src, dst);
if (ret < 0)
av_log(NULL, AV_LOG_ERROR, "Cannot move '%s' into '%s': %s.\n", src, dst, av_err2str(ret));
return ret;
}
static void usage(const char *program_name)
{
fprintf(stderr, "usage: %s input_dir\n"
"API example program to show how to list files in directory "
"accessed through AVIOContext.\n", program_name);
fprintf(stderr, "usage: %s OPERATION entry1 [entry2]\n"
"API example program to show how to manipulate resources "
"accessed through AVIOContext.\n"
"OPERATIONS:\n"
"list list content of the directory\n"
"move rename content in directory\n"
"del delete content in directory\n",
program_name);
}
int main(int argc, char *argv[])
{
const char *op = NULL;
int ret;
av_log_set_level(AV_LOG_DEBUG);
@@ -122,7 +145,32 @@ int main(int argc, char *argv[])
avformat_network_init();
ret = list_op(argv[1]);
op = argv[1];
if (strcmp(op, "list") == 0) {
if (argc < 3) {
av_log(NULL, AV_LOG_INFO, "Missing argument for list operation.\n");
ret = AVERROR(EINVAL);
} else {
ret = list_op(argv[2]);
}
} else if (strcmp(op, "del") == 0) {
if (argc < 3) {
av_log(NULL, AV_LOG_INFO, "Missing argument for del operation.\n");
ret = AVERROR(EINVAL);
} else {
ret = del_op(argv[2]);
}
} else if (strcmp(op, "move") == 0) {
if (argc < 4) {
av_log(NULL, AV_LOG_INFO, "Missing argument for move operation.\n");
ret = AVERROR(EINVAL);
} else {
ret = move_op(argv[2], argv[3]);
}
} else {
av_log(NULL, AV_LOG_INFO, "Invalid operation %s\n", op);
ret = AVERROR(EINVAL);
}
avformat_network_deinit();

View File

@@ -117,12 +117,11 @@ int main(int argc, char *argv[])
end:
avformat_close_input(&fmt_ctx);
/* note: the internal buffer could have changed, and be != avio_ctx_buffer */
if (avio_ctx)
if (avio_ctx) {
av_freep(&avio_ctx->buffer);
avio_context_free(&avio_ctx);
av_freep(&avio_ctx);
}
av_file_unmap(buffer, buffer_size);
if (ret < 0) {

View File

@@ -39,35 +39,6 @@
#define AUDIO_INBUF_SIZE 20480
#define AUDIO_REFILL_THRESH 4096
static int get_format_from_sample_fmt(const char **fmt,
enum AVSampleFormat sample_fmt)
{
int i;
struct sample_fmt_entry {
enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
} sample_fmt_entries[] = {
{ AV_SAMPLE_FMT_U8, "u8", "u8" },
{ AV_SAMPLE_FMT_S16, "s16be", "s16le" },
{ AV_SAMPLE_FMT_S32, "s32be", "s32le" },
{ AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
{ AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
};
*fmt = NULL;
for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
struct sample_fmt_entry *entry = &sample_fmt_entries[i];
if (sample_fmt == entry->sample_fmt) {
*fmt = AV_NE(entry->fmt_be, entry->fmt_le);
return 0;
}
}
fprintf(stderr,
"sample format %s is not supported as output format\n",
av_get_sample_fmt_name(sample_fmt));
return -1;
}
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame,
FILE *outfile)
{
@@ -115,9 +86,6 @@ int main(int argc, char **argv)
size_t data_size;
AVPacket *pkt;
AVFrame *decoded_frame = NULL;
enum AVSampleFormat sfmt;
int n_channels = 0;
const char *fmt;
if (argc <= 2) {
fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
@@ -204,26 +172,6 @@ int main(int argc, char **argv)
pkt->size = 0;
decode(c, pkt, decoded_frame, outfile);
/* print output pcm infomations, because there have no metadata of pcm */
sfmt = c->sample_fmt;
if (av_sample_fmt_is_planar(sfmt)) {
const char *packed = av_get_sample_fmt_name(sfmt);
printf("Warning: the sample format the decoder produced is planar "
"(%s). This example will output the first channel only.\n",
packed ? packed : "?");
sfmt = av_get_packed_sample_fmt(sfmt);
}
n_channels = c->channels;
if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
goto end;
printf("Play the output audio file with the command:\n"
"ffplay -f %s -ac %d -ar %d %s\n",
fmt, n_channels, c->sample_rate,
outfilename);
end:
fclose(outfile);
fclose(f);

View File

@@ -95,8 +95,7 @@ int main(int argc, char **argv)
AVPacket *pkt;
if (argc <= 2) {
fprintf(stderr, "Usage: %s <input file> <output file>\n"
"And check your input file is encoded by mpeg1video please.\n", argv[0]);
fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
exit(0);
}
filename = argv[1];

View File

@@ -55,93 +55,95 @@ static AVPacket pkt;
static int video_frame_count = 0;
static int audio_frame_count = 0;
static int output_video_frame(AVFrame *frame)
{
if (frame->width != width || frame->height != height ||
frame->format != pix_fmt) {
/* To handle this change, one could call av_image_alloc again and
* decode the following frames into another rawvideo file. */
fprintf(stderr, "Error: Width, height and pixel format have to be "
"constant in a rawvideo file, but the width, height or "
"pixel format of the input video changed:\n"
"old: width = %d, height = %d, format = %s\n"
"new: width = %d, height = %d, format = %s\n",
width, height, av_get_pix_fmt_name(pix_fmt),
frame->width, frame->height,
av_get_pix_fmt_name(frame->format));
return -1;
}
/* Enable or disable frame reference counting. You are not supposed to support
* both paths in your application but pick the one most appropriate to your
* needs. Look for the use of refcount in this example to see what are the
* differences of API usage between them. */
static int refcount = 0;
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_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);
return 0;
}
static int output_audio_frame(AVFrame *frame)
{
size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format);
printf("audio_frame n:%d nb_samples:%d pts:%s\n",
audio_frame_count++, frame->nb_samples,
av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));
/* Write the raw audio data samples of the first plane. This works
* fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However,
* most audio decoders output planar audio, which uses a separate
* plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P).
* In other words, this code will write only the first audio channel
* in these cases.
* You should use libswresample or libavfilter to convert the frame
* to packed data. */
fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file);
return 0;
}
static int decode_packet(AVCodecContext *dec, const AVPacket *pkt)
static int decode_packet(int *got_frame, int cached)
{
int ret = 0;
int decoded = pkt.size;
// submit the packet to the decoder
ret = avcodec_send_packet(dec, pkt);
if (ret < 0) {
fprintf(stderr, "Error submitting a packet for decoding (%s)\n", av_err2str(ret));
return ret;
}
*got_frame = 0;
// get all the available frames from the decoder
while (ret >= 0) {
ret = avcodec_receive_frame(dec, frame);
if (pkt.stream_index == video_stream_idx) {
/* decode video frame */
ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
if (ret < 0) {
// those two return values are special and mean there is no output
// frame available, but there were no errors during decoding
if (ret == AVERROR_EOF || ret == AVERROR(EAGAIN))
return 0;
fprintf(stderr, "Error during decoding (%s)\n", av_err2str(ret));
fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret));
return ret;
}
// write the frame data to output file
if (dec->codec->type == AVMEDIA_TYPE_VIDEO)
ret = output_video_frame(frame);
else
ret = output_audio_frame(frame);
if (*got_frame) {
av_frame_unref(frame);
if (ret < 0)
if (frame->width != width || frame->height != height ||
frame->format != pix_fmt) {
/* To handle this change, one could call av_image_alloc again and
* decode the following frames into another rawvideo file. */
fprintf(stderr, "Error: Width, height and pixel format have to be "
"constant in a rawvideo file, but the width, height or "
"pixel format of the input video changed:\n"
"old: width = %d, height = %d, format = %s\n"
"new: width = %d, height = %d, format = %s\n",
width, height, av_get_pix_fmt_name(pix_fmt),
frame->width, frame->height,
av_get_pix_fmt_name(frame->format));
return -1;
}
printf("video_frame%s n:%d coded_n:%d\n",
cached ? "(cached)" : "",
video_frame_count++, frame->coded_picture_number);
/* copy decoded frame to destination buffer:
* this is required since rawvideo expects non aligned data */
av_image_copy(video_dst_data, video_dst_linesize,
(const uint8_t **)(frame->data), frame->linesize,
pix_fmt, width, height);
/* write to rawvideo file */
fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file);
}
} else if (pkt.stream_index == audio_stream_idx) {
/* decode audio frame */
ret = avcodec_decode_audio4(audio_dec_ctx, frame, got_frame, &pkt);
if (ret < 0) {
fprintf(stderr, "Error decoding audio frame (%s)\n", av_err2str(ret));
return ret;
}
/* Some audio decoders decode only part of the packet, and have to be
* called again with the remainder of the packet data.
* Sample: fate-suite/lossless-audio/luckynight-partial.shn
* Also, some decoders might over-read the packet. */
decoded = FFMIN(ret, pkt.size);
if (*got_frame) {
size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format);
printf("audio_frame%s n:%d nb_samples:%d pts:%s\n",
cached ? "(cached)" : "",
audio_frame_count++, frame->nb_samples,
av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));
/* Write the raw audio data samples of the first plane. This works
* fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However,
* most audio decoders output planar audio, which uses a separate
* plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P).
* In other words, this code will write only the first audio channel
* in these cases.
* You should use libswresample or libavfilter to convert the frame
* to packed data. */
fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file);
}
}
return 0;
/* If we use frame reference counting, we own the data and need
* to de-reference it when we don't use it anymore */
if (*got_frame && refcount)
av_frame_unref(frame);
return decoded;
}
static int open_codec_context(int *stream_idx,
@@ -184,7 +186,8 @@ static int open_codec_context(int *stream_idx,
return ret;
}
/* Init the decoders */
/* Init the decoders, with or without reference counting */
av_dict_set(&opts, "refcounted_frames", refcount ? "1" : "0", 0);
if ((ret = avcodec_open2(*dec_ctx, dec, &opts)) < 0) {
fprintf(stderr, "Failed to open %s codec\n",
av_get_media_type_string(type));
@@ -227,17 +230,24 @@ static int get_format_from_sample_fmt(const char **fmt,
int main (int argc, char **argv)
{
int ret = 0;
int ret = 0, got_frame;
if (argc != 4) {
fprintf(stderr, "usage: %s input_file video_output_file audio_output_file\n"
if (argc != 4 && argc != 5) {
fprintf(stderr, "usage: %s [-refcount] input_file video_output_file audio_output_file\n"
"API example program to show how to read frames from an input file.\n"
"This program reads frames from a file, decodes them, and writes decoded\n"
"video frames to a rawvideo file named video_output_file, and decoded\n"
"audio frames to a rawaudio file named audio_output_file.\n",
argv[0]);
"audio frames to a rawaudio file named audio_output_file.\n\n"
"If the -refcount option is specified, the program use the\n"
"reference counting frame system which allows keeping a copy of\n"
"the data for longer than one decode call.\n"
"\n", argv[0]);
exit(1);
}
if (argc == 5 && !strcmp(argv[1], "-refcount")) {
refcount = 1;
argv++;
}
src_filename = argv[1];
video_dst_filename = argv[2];
audio_dst_filename = argv[3];
@@ -315,22 +325,23 @@ int main (int argc, char **argv)
/* read frames from the file */
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 (ret < 0)
break;
AVPacket orig_pkt = pkt;
do {
ret = decode_packet(&got_frame, 0);
if (ret < 0)
break;
pkt.data += ret;
pkt.size -= ret;
} while (pkt.size > 0);
av_packet_unref(&orig_pkt);
}
/* flush the decoders */
if (video_dec_ctx)
decode_packet(video_dec_ctx, NULL);
if (audio_dec_ctx)
decode_packet(audio_dec_ctx, NULL);
/* flush cached frames */
pkt.data = NULL;
pkt.size = 0;
do {
decode_packet(&got_frame, 1);
} while (got_frame);
printf("Demuxing succeeded.\n");

View File

@@ -145,7 +145,7 @@ int main(int argc, char **argv)
frame->width = c->width;
frame->height = c->height;
ret = av_frame_get_buffer(frame, 0);
ret = av_frame_get_buffer(frame, 32);
if (ret < 0) {
fprintf(stderr, "Could not allocate the video frame data\n");
exit(1);
@@ -186,8 +186,7 @@ int main(int argc, char **argv)
encode(c, NULL, pkt, f);
/* 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);
fwrite(endcode, 1, sizeof(endcode), f);
fclose(f);
avcodec_free_context(&c);

View File

@@ -74,6 +74,7 @@ static int open_input_file(const char *filename)
if (!dec_ctx)
return AVERROR(ENOMEM);
avcodec_parameters_to_context(dec_ctx, fmt_ctx->streams[audio_stream_index]->codecpar);
av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
/* init the audio decoder */
if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {

View File

@@ -29,8 +29,6 @@
#define _XOPEN_SOURCE 600 /* for usleep */
#include <unistd.h>
#include <stdio.h>
#include <stdlib.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
@@ -79,6 +77,7 @@ static int open_input_file(const char *filename)
if (!dec_ctx)
return AVERROR(ENOMEM);
avcodec_parameters_to_context(dec_ctx, fmt_ctx->streams[video_stream_index]->codecpar);
av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
/* init the video decoder */
if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
@@ -211,20 +210,17 @@ int main(int argc, char **argv)
{
int ret;
AVPacket packet;
AVFrame *frame;
AVFrame *filt_frame;
AVFrame *frame = av_frame_alloc();
AVFrame *filt_frame = av_frame_alloc();
if (argc != 2) {
fprintf(stderr, "Usage: %s file\n", argv[0]);
exit(1);
}
frame = av_frame_alloc();
filt_frame = av_frame_alloc();
if (!frame || !filt_frame) {
perror("Could not allocate frame");
exit(1);
}
if (argc != 2) {
fprintf(stderr, "Usage: %s file\n", argv[0]);
exit(1);
}
if ((ret = open_input_file(argv[1])) < 0)
goto end;
@@ -252,25 +248,27 @@ int main(int argc, char **argv)
goto end;
}
frame->pts = frame->best_effort_timestamp;
if (ret >= 0) {
frame->pts = frame->best_effort_timestamp;
/* push the decoded frame into the filtergraph */
if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
break;
}
/* pull filtered frames from the filtergraph */
while (1) {
ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
/* push the decoded frame into the filtergraph */
if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
break;
if (ret < 0)
goto end;
display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base);
av_frame_unref(filt_frame);
}
/* pull filtered frames from the filtergraph */
while (1) {
ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
if (ret < 0)
goto end;
display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base);
av_frame_unref(filt_frame);
}
av_frame_unref(frame);
}
av_frame_unref(frame);
}
}
av_packet_unref(&packet);

View File

@@ -4,23 +4,21 @@
*
* HW Acceleration API (video decoding) decode sample
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
* This file is part of FFmpeg.
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* 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.
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
@@ -213,6 +211,7 @@ int main(int argc, char *argv[])
return -1;
decoder_ctx->get_format = get_hw_format;
av_opt_set_int(decoder_ctx, "refcounted_frames", 1, 0);
if (hw_decoder_init(decoder_ctx, type) < 0)
return -1;

View File

@@ -47,11 +47,6 @@ int main (int argc, char **argv)
if ((ret = avformat_open_input(&fmt_ctx, argv[1], NULL, NULL)))
return ret;
if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
while ((tag = av_dict_get(fmt_ctx->metadata, "", tag, AV_DICT_IGNORE_SUFFIX)))
printf("%s=%s\n", tag->key, tag->value);

View File

@@ -78,45 +78,15 @@ static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt)
pkt->stream_index);
}
static int write_frame(AVFormatContext *fmt_ctx, AVCodecContext *c,
AVStream *st, AVFrame *frame)
static int write_frame(AVFormatContext *fmt_ctx, const AVRational *time_base, AVStream *st, AVPacket *pkt)
{
int ret;
/* rescale output packet timestamp values from codec to stream timebase */
av_packet_rescale_ts(pkt, *time_base, st->time_base);
pkt->stream_index = st->index;
// send the frame to the encoder
ret = avcodec_send_frame(c, frame);
if (ret < 0) {
fprintf(stderr, "Error sending a frame to the encoder: %s\n",
av_err2str(ret));
exit(1);
}
while (ret >= 0) {
AVPacket pkt = { 0 };
ret = avcodec_receive_packet(c, &pkt);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
else if (ret < 0) {
fprintf(stderr, "Error encoding a frame: %s\n", av_err2str(ret));
exit(1);
}
/* 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;
/* Write the compressed frame to the media file. */
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);
}
}
return ret == AVERROR_EOF ? 1 : 0;
/* Write the compressed frame to the media file. */
log_packet(fmt_ctx, pkt);
return av_interleaved_write_frame(fmt_ctx, pkt);
}
/* Add an output stream. */
@@ -315,7 +285,7 @@ static AVFrame *get_audio_frame(OutputStream *ost)
/* check if we want to generate more frames */
if (av_compare_ts(ost->next_pts, ost->enc->time_base,
STREAM_DURATION, (AVRational){ 1, 1 }) > 0)
STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
return NULL;
for (j = 0; j <frame->nb_samples; j++) {
@@ -339,10 +309,13 @@ static AVFrame *get_audio_frame(OutputStream *ost)
static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
{
AVCodecContext *c;
AVPacket pkt = { 0 }; // data and size must be 0;
AVFrame *frame;
int ret;
int got_packet;
int dst_nb_samples;
av_init_packet(&pkt);
c = ost->enc;
frame = get_audio_frame(ost);
@@ -376,7 +349,22 @@ static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
ost->samples_count += dst_nb_samples;
}
return write_frame(oc, c, ost->st, frame);
ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
if (ret < 0) {
fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
exit(1);
}
if (got_packet) {
ret = write_frame(oc, &c->time_base, ost->st, &pkt);
if (ret < 0) {
fprintf(stderr, "Error while writing audio frame: %s\n",
av_err2str(ret));
exit(1);
}
}
return (frame || got_packet) ? 0 : 1;
}
/**************************************************************/
@@ -396,7 +384,7 @@ static AVFrame *alloc_picture(enum AVPixelFormat pix_fmt, int width, int height)
picture->height = height;
/* allocate the buffers for the frame data */
ret = av_frame_get_buffer(picture, 0);
ret = av_frame_get_buffer(picture, 32);
if (ret < 0) {
fprintf(stderr, "Could not allocate frame data.\n");
exit(1);
@@ -476,7 +464,7 @@ static AVFrame *get_video_frame(OutputStream *ost)
/* check if we want to generate more frames */
if (av_compare_ts(ost->next_pts, c->time_base,
STREAM_DURATION, (AVRational){ 1, 1 }) > 0)
STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
return NULL;
/* when we pass a frame to the encoder, it may keep a reference to it
@@ -518,8 +506,37 @@ 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));
int ret;
AVCodecContext *c;
AVFrame *frame;
int got_packet = 0;
AVPacket pkt = { 0 };
c = ost->enc;
frame = get_video_frame(ost);
av_init_packet(&pkt);
/* encode the image */
ret = avcodec_encode_video2(c, &pkt, frame, &got_packet);
if (ret < 0) {
fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret));
exit(1);
}
if (got_packet) {
ret = write_frame(oc, &c->time_base, ost->st, &pkt);
} else {
ret = 0;
}
if (ret < 0) {
fprintf(stderr, "Error while writing video frame: %s\n", av_err2str(ret));
exit(1);
}
return (frame || got_packet) ? 0 : 1;
}
static void close_stream(AVFormatContext *oc, OutputStream *ost)

View File

@@ -1,23 +1,21 @@
/*
* Video Acceleration API (video encoding) encode sample
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
* This file is part of FFmpeg.
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* 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.
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
@@ -172,7 +170,7 @@ int main(int argc, char *argv[])
sw_frame->width = width;
sw_frame->height = height;
sw_frame->format = AV_PIX_FMT_NV12;
if ((err = av_frame_get_buffer(sw_frame, 0)) < 0)
if ((err = av_frame_get_buffer(sw_frame, 32)) < 0)
goto close;
if ((err = fread((uint8_t*)(sw_frame->data[0]), size, 1, fin)) <= 0)
break;

View File

@@ -1,23 +1,21 @@
/*
* Video Acceleration API (video transcoding) transcode sample
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
* This file is part of FFmpeg.
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* 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.
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**

View File

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

View File

@@ -149,18 +149,12 @@ the synchronisation of the samples directory.
@chapter Uploading new samples to the fate suite
If you need a sample uploaded send a mail to samples-request.
This is for developers who have an account on the fate suite server.
If you upload new samples, please make sure they are as small as possible,
space on each client, network bandwidth and so on benefit from smaller test cases.
Also keep in mind older checkouts use existing sample files, that means in
practice generally do not replace, remove or overwrite files as it likely would
break older checkouts or releases.
Also all needed samples for a commit should be uploaded, ideally 24
hours, before the push.
If you need an account for frequently uploading samples or you wish to help
others by doing that send a mail to ffmpeg-devel.
@example
#First update your local samples copy:
@@ -228,11 +222,6 @@ Set to @samp{1} to generate the missing or mismatched references.
Specify which hardware acceleration to use while running regression tests,
by default @samp{none} is used.
@item KEEP
Set to @samp{1} to keep temp files generated by fate test(s) when test is successful.
Default is @samp{0}, which removes these files. Files are always kept when a test
fails.
@end table
@section Examples

View File

@@ -216,208 +216,16 @@ filters is obviously also impossible, since filters work on uncompressed data.
@chapter Stream selection
@c man begin STREAM SELECTION
@command{ffmpeg} provides the @code{-map} option for manual control of stream selection in each
output file. Users can skip @code{-map} and let ffmpeg perform automatic stream selection as
described below. The @code{-vn / -an / -sn / -dn} options can be used to skip inclusion of
video, audio, subtitle and data streams respectively, whether manually mapped or automatically
selected, except for those streams which are outputs of complex filtergraphs.
By default, @command{ffmpeg} includes only one stream of each type (video, audio, subtitle)
present in the input files and adds them to each output file. It picks the
"best" of each based upon the following criteria: for video, it is the stream
with the highest resolution, for audio, it is the stream with the most channels, for
subtitles, it is the first subtitle stream. In the case where several streams of
the same type rate equally, the stream with the lowest index is chosen.
@section Description
The sub-sections that follow describe the various rules that are involved in stream selection.
The examples that follow next show how these rules are applied in practice.
While every effort is made to accurately reflect the behavior of the program, FFmpeg is under
continuous development and the code may have changed since the time of this writing.
@subsection Automatic stream selection
In the absence of any map options for a particular output file, ffmpeg inspects the output
format to check which type of streams can be included in it, viz. video, audio and/or
subtitles. For each acceptable stream type, ffmpeg will pick one stream, when available,
from among all the inputs.
It will select that stream based upon the following criteria:
@itemize
@item
for video, it is the stream with the highest resolution,
@item
for audio, it is the stream with the most channels,
@item
for subtitles, it is the first subtitle stream found but there's a caveat.
The output format's default subtitle encoder can be either text-based or image-based,
and only a subtitle stream of the same type will be chosen.
@end itemize
In the case where several streams of the same type rate equally, the stream with the lowest
index is chosen.
Data or attachment streams are not automatically selected and can only be included
using @code{-map}.
@subsection Manual stream selection
When @code{-map} is used, only user-mapped streams are included in that output file,
with one possible exception for filtergraph outputs described below.
@subsection Complex filtergraphs
If there are any complex filtergraph output streams with unlabeled pads, they will be added
to the first output file. This will lead to a fatal error if the stream type is not supported
by the output format. In the absence of the map option, the inclusion of these streams leads
to the automatic stream selection of their types being skipped. If map options are present,
these filtergraph streams are included in addition to the mapped streams.
Complex filtergraph output streams with labeled pads must be mapped once and exactly once.
@subsection Stream handling
Stream handling is independent of stream selection, with an exception for subtitles described
below. Stream handling is set via the @code{-codec} option addressed to streams within a
specific @emph{output} file. In particular, codec options are applied by ffmpeg after the
stream selection process and thus do not influence the latter. If no @code{-codec} option is
specified for a stream type, ffmpeg will select the default encoder registered by the output
file muxer.
An exception exists for subtitles. If a subtitle encoder is specified for an output file, the
first subtitle stream found of any type, text or image, will be included. ffmpeg does not validate
if the specified encoder can convert the selected stream or if the converted stream is acceptable
within the output format. This applies generally as well: when the user sets an encoder manually,
the stream selection process cannot check if the encoded stream can be muxed into the output file.
If it cannot, ffmpeg will abort and @emph{all} output files will fail to be processed.
@section Examples
The following examples illustrate the behavior, quirks and limitations of ffmpeg's stream
selection methods.
They assume the following three input files.
@verbatim
input file 'A.avi'
stream 0: video 640x360
stream 1: audio 2 channels
input file 'B.mp4'
stream 0: video 1920x1080
stream 1: audio 2 channels
stream 2: subtitles (text)
stream 3: audio 5.1 channels
stream 4: subtitles (text)
input file 'C.mkv'
stream 0: video 1280x720
stream 1: audio 2 channels
stream 2: subtitles (image)
@end verbatim
@subsubheading Example: automatic stream selection
@example
ffmpeg -i A.avi -i B.mp4 out1.mkv out2.wav -map 1:a -c:a copy out3.mov
@end example
There are three output files specified, and for the first two, no @code{-map} options
are set, so ffmpeg will select streams for these two files automatically.
@file{out1.mkv} is a Matroska container file and accepts video, audio and subtitle streams,
so ffmpeg will try to select one of each type.@*
For video, it will select @code{stream 0} from @file{B.mp4}, which has the highest
resolution among all the input video streams.@*
For audio, it will select @code{stream 3} from @file{B.mp4}, since it has the greatest
number of channels.@*
For subtitles, it will select @code{stream 2} from @file{B.mp4}, which is the first subtitle
stream from among @file{A.avi} and @file{B.mp4}.
@file{out2.wav} accepts only audio streams, so only @code{stream 3} from @file{B.mp4} is
selected.
For @file{out3.mov}, since a @code{-map} option is set, no automatic stream selection will
occur. The @code{-map 1:a} option will select all audio streams from the second input
@file{B.mp4}. No other streams will be included in this output file.
For the first two outputs, all included streams will be transcoded. The encoders chosen will
be the default ones registered by each output format, which may not match the codec of the
selected input streams.
For the third output, codec option for audio streams has been set
to @code{copy}, so no decoding-filtering-encoding operations will occur, or @emph{can} occur.
Packets of selected streams shall be conveyed from the input file and muxed within the output
file.
@subsubheading Example: automatic subtitles selection
@example
ffmpeg -i C.mkv out1.mkv -c:s dvdsub -an out2.mkv
@end example
Although @file{out1.mkv} is a Matroska container file which accepts subtitle streams, only a
video and audio stream shall be selected. The subtitle stream of @file{C.mkv} is image-based
and the default subtitle encoder of the Matroska muxer is text-based, so a transcode operation
for the subtitles is expected to fail and hence the stream isn't selected. However, in
@file{out2.mkv}, a subtitle encoder is specified in the command and so, the subtitle stream is
selected, in addition to the video stream. The presence of @code{-an} disables audio stream
selection for @file{out2.mkv}.
@subsubheading Example: unlabeled filtergraph outputs
@example
ffmpeg -i A.avi -i C.mkv -i B.mp4 -filter_complex "overlay" out1.mp4 out2.srt
@end example
A filtergraph is setup here using the @code{-filter_complex} option and consists of a single
video filter. The @code{overlay} filter requires exactly two video inputs, but none are
specified, so the first two available video streams are used, those of @file{A.avi} and
@file{C.mkv}. The output pad of the filter has no label and so is sent to the first output file
@file{out1.mp4}. Due to this, automatic selection of the video stream is skipped, which would
have selected the stream in @file{B.mp4}. The audio stream with most channels viz. @code{stream 3}
in @file{B.mp4}, is chosen automatically. No subtitle stream is chosen however, since the MP4
format has no default subtitle encoder registered, and the user hasn't specified a subtitle encoder.
The 2nd output file, @file{out2.srt}, only accepts text-based subtitle streams. So, even though
the first subtitle stream available belongs to @file{C.mkv}, it is image-based and hence skipped.
The selected stream, @code{stream 2} in @file{B.mp4}, is the first text-based subtitle stream.
@subsubheading Example: labeled filtergraph outputs
@example
ffmpeg -i A.avi -i B.mp4 -i C.mkv -filter_complex "[1:v]hue=s=0[outv];overlay;aresample" \
-map '[outv]' -an out1.mp4 \
out2.mkv \
-map '[outv]' -map 1:a:0 out3.mkv
@end example
The above command will fail, as the output pad labelled @code{[outv]} has been mapped twice.
None of the output files shall be processed.
@example
ffmpeg -i A.avi -i B.mp4 -i C.mkv -filter_complex "[1:v]hue=s=0[outv];overlay;aresample" \
-an out1.mp4 \
out2.mkv \
-map 1:a:0 out3.mkv
@end example
This command above will also fail as the hue filter output has a label, @code{[outv]},
and hasn't been mapped anywhere.
The command should be modified as follows,
@example
ffmpeg -i A.avi -i B.mp4 -i C.mkv -filter_complex "[1:v]hue=s=0,split=2[outv1][outv2];overlay;aresample" \
-map '[outv1]' -an out1.mp4 \
out2.mkv \
-map '[outv2]' -map 1:a:0 out3.mkv
@end example
The video stream from @file{B.mp4} is sent to the hue filter, whose output is cloned once using
the split filter, and both outputs labelled. Then a copy each is mapped to the first and third
output files.
The overlay filter, requiring two video inputs, uses the first two unused video streams. Those
are the streams from @file{A.avi} and @file{C.mkv}. The overlay output isn't labelled, so it is
sent to the first output file @file{out1.mp4}, regardless of the presence of the @code{-map} option.
The aresample filter is sent the first unused audio stream, that of @file{A.avi}. Since this filter
output is also unlabelled, it too is mapped to the first output file. The presence of @code{-an}
only suppresses automatic or manual stream selection of audio streams, not outputs sent from
filtergraphs. Both these mapped streams shall be ordered before the mapped stream in @file{out1.mp4}.
The video, audio and subtitle streams mapped to @code{out2.mkv} are entirely determined by
automatic stream selection.
@file{out3.mkv} consists of the cloned video output from the hue filter and the first audio
stream from @file{B.mp4}.
@*
You can disable some of those defaults by using the @code{-vn/-an/-sn/-dn} options. For
full manual control, use the @code{-map} option, which disables the defaults just
described.
@c man end STREAM SELECTION
@@ -508,7 +316,7 @@ input until the timestamps reach @var{position}.
@var{position} must be a time duration specification,
see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
@item -sseof @var{position} (@emph{input})
@item -sseof @var{position} (@emph{input/output})
Like the @code{-ss} option but relative to the "end of file". That is negative
values are earlier in the file, 0 is at EOF.
@@ -523,9 +331,6 @@ The offset is added to the timestamps of the input files. Specifying
a positive offset means that the corresponding streams are delayed by
the time duration specified in @var{offset}.
@item -itsscale @var{scale} (@emph{input,per-stream})
Rescale input timestamps. @var{scale} should be a floating point number.
@item -timestamp @var{date} (@emph{output})
Set the recording timestamp in the container.
@@ -570,31 +375,22 @@ The following dispositions are recognized:
@item hearing_impaired
@item visual_impaired
@item clean_effects
@item attached_pic
@item captions
@item descriptions
@item dependent
@item metadata
@end table
For example, to make the second audio stream the default stream:
@example
ffmpeg -i in.mkv -c copy -disposition:a:1 default out.mkv
ffmpeg -i in.mkv -disposition:a:1 default out.mkv
@end example
To make the second subtitle stream the default stream and remove the default
disposition from the first subtitle stream:
@example
ffmpeg -i in.mkv -c copy -disposition:s:0 0 -disposition:s:1 default out.mkv
ffmpeg -i INPUT -disposition:s:0 0 -disposition:s:1 default OUTPUT
@end example
To add an embedded cover/thumbnail:
@example
ffmpeg -i in.mp4 -i IMAGE -map 0 -map 1 -c copy -c:v:1 png -disposition:v:1 attached_pic out.mp4
@end example
Not all muxers support embedded thumbnails, and those who do, only support a few formats, like JPEG or PNG.
@item -program [title=@var{title}:][program_num=@var{program_num}:]st=@var{stream}[:st=@var{stream}...] (@emph{output})
Creates a program with the specified @var{title}, @var{program_num} and adds the specified
@@ -617,13 +413,8 @@ they do not conflict with the standard, as in:
ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg
@end example
@item -dn (@emph{input/output})
As an input option, blocks all data streams of a file from being filtered or
being automatically selected or mapped for any output. See @code{-discard}
option to disable streams individually.
As an output option, disables data recording i.e. automatic selection or
mapping of any data stream. For full manual control see the @code{-map}
@item -dn (@emph{output})
Disable data recording. For full manual control see the @code{-map}
option.
@item -dframes @var{number} (@emph{output})
@@ -783,13 +574,8 @@ If used together with @option{-vcodec copy}, it will affect the aspect ratio
stored at container level, but not the aspect ratio stored in encoded
frames, if it exists.
@item -vn (@emph{input/output})
As an input option, blocks all video streams of a file from being filtered or
being automatically selected or mapped for any output. See @code{-discard}
option to disable streams individually.
As an output option, disables video recording i.e. automatic selection or
mapping of any video stream. For full manual control see the @code{-map}
@item -vn (@emph{output})
Disable video recording. For full manual control see the @code{-map}
option.
@item -vcodec @var{codec} (@emph{output})
@@ -837,6 +623,8 @@ as the input (or graph output) and automatic conversions are disabled.
@item -sws_flags @var{flags} (@emph{input/output})
Set SwScaler flags.
@item -vdt @var{n}
Discard threshold.
@item -rc_override[:@var{stream_specifier}] @var{override} (@emph{output,per-stream})
Rate control override for specific intervals, formatted as "int,int,int"
@@ -879,19 +667,12 @@ Deprecated see -bsf
@item -force_key_frames[:@var{stream_specifier}] @var{time}[,@var{time}...] (@emph{output,per-stream})
@item -force_key_frames[:@var{stream_specifier}] expr:@var{expr} (@emph{output,per-stream})
@item -force_key_frames[:@var{stream_specifier}] source (@emph{output,per-stream})
Force key frames at the specified timestamps, more precisely at the first
frames after each specified time.
@var{force_key_frames} can take arguments of the following form:
@table @option
@item @var{time}[,@var{time}...]
If the argument consists of timestamps, ffmpeg will round the specified times to the nearest
output timestamp as per the encoder time base and force a keyframe at the first frame having
timestamp equal or greater than the computed timestamp. Note that if the encoder time base is too
coarse, then the keyframes may be forced on frames with timestamps lower than the specified time.
The default encoder time base is the inverse of the output framerate but may be set otherwise
via @code{-enc_time_base}.
If the argument is prefixed with @code{expr:}, the string @var{expr}
is interpreted like an expression and is evaluated for each frame. A
key frame is forced in case the evaluation is non-zero.
If one of the times is "@code{chapters}[@var{delta}]", it is expanded into
the time of the beginning of all chapters in the file, shifted by
@@ -905,11 +686,6 @@ before the beginning of every chapter:
-force_key_frames 0:05:00,chapters-0.1
@end example
@item expr:@var{expr}
If the argument is prefixed with @code{expr:}, the string @var{expr}
is interpreted like an expression and is evaluated for each frame. A
key frame is forced in case the evaluation is non-zero.
The expression in @var{expr} can contain the following constants:
@table @option
@item n
@@ -937,12 +713,6 @@ starting from second 13:
-force_key_frames expr:if(isnan(prev_forced_t),gte(t,13),gte(t,prev_forced_t+5))
@end example
@item source
If the argument is @code{source}, ffmpeg will force a key frame if
the current frame being encoded is marked as a key frame in its source.
@end table
Note that forcing too many keyframes is very harmful for the lookahead
algorithms of certain encoders: using fixed-GOP options or similar
would be more efficient.
@@ -1029,35 +799,6 @@ Choose the GPU device on the second platform supporting the @emph{cl_khr_fp16}
extension.
@end table
@item vulkan
If @var{device} is an integer, it selects the device by its index in a
system-dependent list of devices. If @var{device} is any other string, it
selects the first device with a name containing that string as a substring.
The following options are recognized:
@table @option
@item debug
If set to 1, enables the validation layer, if installed.
@item linear_images
If set to 1, images allocated by the hwcontext will be linear and locally mappable.
@item instance_extensions
A plus separated list of additional instance extensions to enable.
@item device_extensions
A plus separated list of additional device extensions to enable.
@end table
Examples:
@table @emph
@item -init_hw_device vulkan:1
Choose the second device on the system.
@item -init_hw_device vulkan:RADV
Choose the first device with a name containing the string @emph{RADV}.
@item -init_hw_device vulkan:0,instance_extensions=VK_KHR_wayland_surface+VK_KHR_xcb_surface
Choose the first device and enable the Wayland and XCB instance extensions.
@end table
@end table
@item -init_hw_device @var{type}[=@var{name}]@@@var{source}
@@ -1149,13 +890,8 @@ Set the number of audio channels. For output streams it is set by
default to the number of input audio channels. For input streams
this option only makes sense for audio grabbing devices and raw demuxers
and is mapped to the corresponding demuxer options.
@item -an (@emph{input/output})
As an input option, blocks all audio streams of a file from being filtered or
being automatically selected or mapped for any output. See @code{-discard}
option to disable streams individually.
As an output option, disables audio recording i.e. automatic selection or
mapping of any audio stream. For full manual control see the @code{-map}
@item -an (@emph{output})
Disable audio recording. For full manual control see the @code{-map}
option.
@item -acodec @var{codec} (@emph{input/output})
Set the audio codec. This is an alias for @code{-codec:a}.
@@ -1190,13 +926,8 @@ stereo but not 6 channels as 5.1. The default is to always try to guess. Use
@table @option
@item -scodec @var{codec} (@emph{input/output})
Set the subtitle codec. This is an alias for @code{-codec:s}.
@item -sn (@emph{input/output})
As an input option, blocks all subtitle streams of a file from being filtered or
being automatically selected or mapped for any output. See @code{-discard}
option to disable streams individually.
As an output option, disables subtitle recording i.e. automatic selection or
mapping of any subtitle stream. For full manual control see the @code{-map}
@item -sn (@emph{output})
Disable subtitle recording. For full manual control see the @code{-map}
option.
@item -sbsf @var{bitstream_filter}
Deprecated, see -bsf
@@ -1423,14 +1154,14 @@ disable any chapter copying.
@item -benchmark (@emph{global})
Show benchmarking information at the end of an encode.
Shows real, system and user time used and maximum memory consumption.
Shows CPU time used and maximum memory consumption.
Maximum memory consumption is not supported on all systems,
it will usually display as 0 if not supported.
@item -benchmark_all (@emph{global})
Show benchmarking information during the encode.
Shows real, system and user time used in various steps (audio/video encode/decode).
Shows CPU time used in various steps (audio/video encode/decode).
@item -timelimit @var{duration} (@emph{global})
Exit after ffmpeg has been running for @var{duration} seconds in CPU user time.
Exit after ffmpeg has been running for @var{duration} seconds.
@item -dump (@emph{global})
Dump each input packet to stderr.
@item -hex (@emph{global})
@@ -1443,6 +1174,10 @@ loss).
By default @command{ffmpeg} attempts to read the input(s) as fast as possible.
This option will slow down the reading of the input(s) to the native frame rate
of the input(s). It is useful for real-time output (e.g. live streaming).
@item -loop_output @var{number_of_times}
Repeatedly loop output for formats that support looping such as animated GIF
(0 will loop the output infinitely).
This option is deprecated, use -loop.
@item -vsync @var{parameter}
Video sync method.
For compatibility reasons old values can be specified as numbers.
@@ -1562,13 +1297,9 @@ Enable bitexact mode for (de)muxer and (de/en)coder
Finish encoding when the shortest input stream ends.
@item -dts_delta_threshold
Timestamp discontinuity delta threshold.
@item -dts_error_threshold @var{seconds}
Timestamp error delta threshold. This threshold use to discard crazy/damaged
timestamps and the default is 30 hours which is arbitrarily picked and quite
conservative.
@item -muxdelay @var{seconds} (@emph{output})
@item -muxdelay @var{seconds} (@emph{input})
Set the maximum demux-decode delay.
@item -muxpreload @var{seconds} (@emph{output})
@item -muxpreload @var{seconds} (@emph{input})
Set the initial demux-decode delay.
@item -streamid @var{output-stream-index}:@var{new-value} (@emph{output})
Assign a new stream-id value to an output stream. This option should be
@@ -1690,10 +1421,8 @@ This allows dumping sdp information when at least one output isn't an
rtp stream. (Requires at least one of the output formats to be rtp).
@item -discard (@emph{input})
Allows discarding specific streams or frames from streams.
Any input stream can be fully discarded, using value @code{all} whereas
selective discarding of frames from a stream occurs at the demuxer
and is not supported by all demuxers.
Allows discarding specific streams or frames of streams at the demuxer.
Not all demuxers support this.
@table @option
@item none
@@ -1721,8 +1450,6 @@ Stop and abort on various conditions. The following flags are available:
@table @option
@item empty_output
No packets were passed to the muxer, the output is empty.
@item empty_output_stream
No packets were passed to the muxer in some of the output streams.
@end table
@item -xerror (@emph{global})

View File

@@ -60,14 +60,10 @@ Play @var{duration} seconds of audio/video.
see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
@item -bytes
Seek by bytes.
@item -seek_interval
Set custom interval, in seconds, for seeking using left/right keys. Default is 10 seconds.
@item -nodisp
Disable graphical display.
@item -noborder
Borderless window.
@item -alwaysontop
Window always on top. Available on: X11 with SDL >= 2.0.5, Windows SDL >= 2.0.6.
@item -volume
Set the startup volume. 0 means silence, 100 means no volume reduction or
amplification. Negative values are treated as 0, values above 100 are treated
@@ -76,10 +72,6 @@ as 100.
Force format.
@item -window_title @var{title}
Set window title (default is the input filename).
@item -left @var{title}
Set the x position for the left of the window (default is a centered window).
@item -top @var{title}
Set the y position for the top of the window (default is a centered window).
@item -loop @var{number}
Loops movie playback <number> times. 0 means forever.
@item -showmode @var{mode}
@@ -133,9 +125,8 @@ 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
the audio/video synchronisation drift. It is shown by default, unless the
log level is lower than @code{info}. Its display can be forced by manually
specifying this option. To disable it, you need to specify @code{-nostats}.
the audio/video synchronisation drift. It is on by default, to
explicitly disable it you need to specify @code{-nostats}.
@item -fast
Non-spec-compliant optimizations.
@@ -198,12 +189,6 @@ input as soon as possible. Enabled by default for realtime streams, where data
may be dropped if not read in time. Use this option to enable infinite buffers
for all inputs, use @option{-noinfbuf} to disable it.
@item -filter_threads @var{nb_threads}
Defines how many threads are used to process a filter pipeline. Each pipeline
will produce a thread pool with this many threads available for parallel
processing. The default is 0 which means that the thread count will be
determined by the number of available CPUs.
@end table
@section While playing

View File

@@ -425,7 +425,7 @@ The @code{csv} writer is equivalent to @code{compact}, but supports
different defaults.
Each section is printed on a single line.
If no option is specified, the output has the form:
If no option is specifid, the output has the form:
@example
section|key1=val1| ... |keyN=valN
@end example
@@ -584,14 +584,14 @@ value is 0.
This is required for generating an XML file which can be validated
through an XSD file.
@item xsd_strict, x
@item xsd_compliant, x
If set to 1 perform more checks for ensuring that the output is XSD
compliant. Default value is 0.
This option automatically sets @option{fully_qualified} to 1.
@end table
For more information about the XML format, see
@url{https://www.w3.org/XML/}.
@url{http://www.w3.org/XML/}.
@c man end WRITERS
@chapter Timecode

View File

@@ -147,25 +147,11 @@
</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" />
@@ -226,7 +212,6 @@
<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"/>

View File

@@ -34,24 +34,27 @@ Possible forms of stream specifiers are:
@table @option
@item @var{stream_index}
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 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}]
thread count for the second stream to 4.
@item @var{stream_type}[:@var{stream_index}]
@var{stream_type} is one of following: 'v' or 'V' for video, 'a' for audio, 's'
for subtitle, 'd' for data, and 't' for attachments. 'v' matches all video
streams, 'V' only matches video streams which are not attached pictures, video
thumbnails or cover arts. If @var{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 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
are part of the program and match the @var{additional_stream_specifier}.
thumbnails or cover arts. If @var{stream_index} is given, then it matches
stream number @var{stream_index} of this type. Otherwise, it matches all
streams of this type.
@item p:@var{program_id}[:@var{stream_index}] or p:@var{program_id}[:@var{stream_type}[:@var{stream_index}]] or
p:@var{program_id}:m:@var{key}[:@var{value}]
In first version, if @var{stream_index} is given, then it matches the stream with number @var{stream_index}
in the program with the id @var{program_id}. Otherwise, it matches all streams in the
program. In the second version, @var{stream_type} is one of following: 'v' for video, 'a' for audio, 's'
for subtitle, 'd' for data. If @var{stream_index} is also given, then it matches
stream number @var{stream_index} of this type in the program with the id @var{program_id}.
Otherwise, if only @var{stream_type} is given, it matches all
streams of this type in the program with the id @var{program_id}.
In the third version matches streams in the program with the id @var{program_id} with the metadata
tag @var{key} having the specified value. If
@var{value} is not given, matches streams that contain the given tag with any
value.
@item #@var{stream_id} or i:@var{stream_id}
Match the stream by stream id (e.g. PID in MPEG-TS container).
@@ -109,10 +112,6 @@ Print detailed information about the muxer named @var{muxer_name}. Use the
@item filter=@var{filter_name}
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 name @var{bitstream_filter_name}.
Use the @option{-bsfs} option to get a list of all bitstream filters.
@end table
@item -version
@@ -236,15 +235,17 @@ ffmpeg [...] -loglevel +repeat
By default the program logs to stderr. If coloring is supported by the
terminal, colors are used to mark errors and warnings. Log coloring
can be disabled setting the environment variable
@env{AV_LOG_FORCE_NOCOLOR}, or can be forced setting
@env{AV_LOG_FORCE_NOCOLOR} or @env{NO_COLOR}, or can be forced setting
the environment variable @env{AV_LOG_FORCE_COLOR}.
The use of the environment variable @env{NO_COLOR} is deprecated and
will be dropped in a future FFmpeg version.
@item -report
Dump full command line and log output to a file named
Dump full command line and console output to a file named
@code{@var{program}-@var{YYYYMMDD}-@var{HHMMSS}.log} in the current
directory.
This file can be useful for bug reports.
It also implies @code{-loglevel debug}.
It also implies @code{-loglevel verbose}.
Setting the environment variable @env{FFREPORT} to any value has the
same effect. If the value is a ':'-separated key=value sequence, these
@@ -370,15 +371,7 @@ ffmpeg -i input.flac -id3v2_version 3 out.mp3
@end example
All codec AVOptions are per-stream, and thus a stream specifier
should be attached to them:
@example
ffmpeg -i multichannel.mxf -map 0:v:0 -map 0:a:0 -map 0:a:0 -c:a:0 ac3 -b:a:0 640k -ac:a:1 2 -c:a:1 aac -b:2 128k out.mp4
@end example
In the above example, a multichannel audio stream is mapped twice for output.
The first instance is encoded with codec ac3 and bitrate 640k.
The second instance is downmixed to 2 channels and encoded with codec aac. A bitrate of 128k is specified for it using
absolute index of the output stream.
should be attached to them.
Note: the @option{-nooption} syntax cannot be used for boolean
AVOptions, use @option{-option 0}/@option{-option 1}.

File diff suppressed because it is too large Load Diff

View File

@@ -27,50 +27,40 @@ stream information. A higher value will enable detecting more
information in case it is dispersed into the stream, but will increase
latency. Must be an integer not lesser than 32. It is 5000000 by default.
@item max_probe_packets @var{integer} (@emph{input})
Set the maximum number of buffered packets when probing a codec.
Default is 2500 packets.
@item packetsize @var{integer} (@emph{output})
Set packet size.
@item fflags @var{flags}
Set format flags. Some are implemented for a limited number of formats.
@item fflags @var{flags} (@emph{input/output})
Set format flags.
Possible values for input files:
Possible values:
@table @samp
@item discardcorrupt
Discard corrupted packets.
@item ignidx
Ignore index.
@item fastseek
Enable fast, but inaccurate seeks for some formats.
@item genpts
Generate missing PTS if DTS is present.
@item igndts
Ignore DTS if PTS is set. Inert when nofillin is set.
@item ignidx
Ignore index.
@item keepside (@emph{deprecated},@emph{inert})
@item nobuffer
Reduce the latency introduced by buffering during initial input streams analysis.
Generate PTS.
@item nofillin
Do not fill in missing values in packet fields that can be exactly calculated.
Do not fill in missing values that can be exactly calculated.
@item noparse
Disable AVParsers, this needs @code{+nofillin} too.
@item igndts
Ignore DTS.
@item discardcorrupt
Discard corrupted frames.
@item sortdts
Try to interleave output packets by DTS. At present, available only for AVIs with an index.
@end table
Possible values for output files:
@table @samp
@item autobsf
Automatically apply bitstream filters as required by the output format. Enabled by default.
Try to interleave output packets by DTS.
@item keepside
Do not merge side data.
@item latm
Enable RTP MP4A-LATM payload.
@item nobuffer
Reduce the latency introduced by optional buffering
@item bitexact
Only write platform-, build- and time-independent data.
This ensures that file and data checksums are reproducible and match between
platforms. Its primary use is for regression testing.
@item flush_packets
Write out packets immediately.
@item latm (@emph{deprecated},@emph{inert})
@item shortest
Stop muxing at the end of the shortest stream.
It may be needed to increase max_interleave_delta to avoid flushing the longer
@@ -143,7 +133,7 @@ Consider things that a sane encoder should not do as an error.
@item max_interleave_delta @var{integer} (@emph{output})
Set maximum buffering duration for interleaving. The duration is
expressed in microseconds, and defaults to 10000000 (10 seconds).
expressed in microseconds, and defaults to 1000000 (1 second).
To ensure all the streams are interleaved correctly, libavformat will
wait until it has at least one packet for each stream before actually
@@ -215,7 +205,7 @@ is @code{0} (meaning that no offset is applied).
@item dump_separator @var{string} (@emph{input})
Separator used to separate the fields printed on the command line about the
Stream parameters.
For example, to separate the fields with newlines and indentation:
For example to separate the fields with newlines and indention:
@example
ffprobe -dump_separator "
" -i ~/videos/matrixbench_mpeg2.mpg
@@ -224,32 +214,6 @@ ffprobe -dump_separator "
@item max_streams @var{integer} (@emph{input})
Specifies the maximum number of streams. This can be used to reject files that
would require too many resources due to a large number of streams.
@item skip_estimate_duration_from_pts @var{bool} (@emph{input})
Skip estimation of input duration when calculated using PTS.
At present, applicable for MPEG-PS and MPEG-TS.
@item strict, f_strict @var{integer} (@emph{input/output})
Specify how strictly to follow the standards. @code{f_strict} is deprecated and
should be used only via the @command{ffmpeg} tool.
Possible values:
@table @samp
@item very
strictly conform to an older more strict version of the spec or reference software
@item strict
strictly conform to all the things in the spec no matter what consequences
@item normal
@item unofficial
allow unofficial extensions
@item experimental
allow non standardized experimental things, experimental
(unfinished/work in progress/not well tested) decoders and encoders.
Note: experimental decoders can pose a security risk, do not use this for
decoding untrusted input.
@end table
@end table
@c man end FORMAT OPTIONS
@@ -260,10 +224,30 @@ decoding untrusted input.
Format stream specifiers allow selection of one or more streams that
match specific properties.
Possible forms of stream specifiers are:
@table @option
@item @var{stream_index}
Matches the stream with this index.
@item @var{stream_type}[:@var{stream_index}]
@var{stream_type} is one of following: 'v' for video, 'a' for audio,
's' for subtitle, 'd' for data, and 't' for attachments. If
@var{stream_index} is given, then it matches the stream number
@var{stream_index} of this type. Otherwise, it matches all streams of
this type.
@item p:@var{program_id}[:@var{stream_index}]
If @var{stream_index} is given, then it matches the stream with number
@var{stream_index} in the program with the id
@var{program_id}. Otherwise, it matches all streams in the program.
@item #@var{stream_id}
Matches the stream by a format-specific ID.
@end table
The exact semantics of stream specifiers is defined by the
@code{avformat_match_stream_specifier()} function declared in the
@file{libavformat/avformat.h} header and documented in the
@ref{Stream specifiers,,Stream specifiers section in the ffmpeg(1) manual,ffmpeg}.
@file{libavformat/avformat.h} header.
@ifclear config-writeonly
@include demuxers.texi

View File

@@ -17,177 +17,21 @@ for more formats. None of them are used by default, their use has to be
explicitly requested by passing the appropriate flags to
@command{./configure}.
@section Alliance for Open Media (AOM)
@section Alliance for Open Media libaom
FFmpeg can make use of the AOM library for AV1 decoding and encoding.
FFmpeg can make use of the libaom library for AV1 decoding.
Go to @url{http://aomedia.org/} and follow the instructions for
installing the library. Then pass @code{--enable-libaom} to configure to
enable it.
@section AMD AMF/VCE
@section OpenJPEG
FFmpeg can use the AMD Advanced Media Framework library
for accelerated H.264 and HEVC(only windows) encoding on hardware with Video Coding Engine (VCE).
FFmpeg can use the OpenJPEG libraries for encoding/decoding J2K videos. Go to
@url{http://www.openjpeg.org/} to get the libraries and follow the installation
instructions. To enable using OpenJPEG in FFmpeg, pass @code{--enable-libopenjpeg} to
@file{./configure}.
To enable support you must obtain the AMF framework header files(version 1.4.9+) from
@url{https://github.com/GPUOpen-LibrariesAndSDKs/AMF.git}.
Create an @code{AMF/} directory in the system include path.
Copy the contents of @code{AMF/amf/public/include/} into that directory.
Then configure FFmpeg with @code{--enable-amf}.
Initialization of amf encoder occurs in this order:
1) trying to initialize through dx11(only windows)
2) trying to initialize through dx9(only windows)
3) trying to initialize through vulkan
To use h.264(AMD VCE) encoder on linux amdgru-pro version 19.20+ and amf-amdgpu-pro
package(amdgru-pro contains, but does not install automatically) are required.
This driver can be installed using amdgpu-pro-install script in official amd driver archive.
@section AviSynth
FFmpeg can read AviSynth scripts as input. To enable support, pass
@code{--enable-avisynth} to configure after installing the headers
provided by @url{https://github.com/AviSynth/AviSynthPlus, AviSynth+}.
AviSynth+ can be configured to install only the headers by either
passing @code{-DHEADERS_ONLY:bool=on} to the normal CMake-based build
system, or by using the supplied @code{GNUmakefile}.
For Windows, supported AviSynth variants are
@url{http://avisynth.nl, AviSynth 2.6 RC1 or higher} for 32-bit builds and
@url{http://avisynth.nl/index.php/AviSynth+, AviSynth+ r1718 or higher} for 32-bit and 64-bit builds.
For Linux, macOS, and BSD, the only supported AviSynth variant is
@url{https://github.com/AviSynth/AviSynthPlus, AviSynth+}, starting with version 3.5.
@float NOTE
In 2016, AviSynth+ added support for building with GCC. However, due to
the eccentricities of Windows' calling conventions, 32-bit GCC builds
of AviSynth+ are not compatible with typical 32-bit builds of FFmpeg.
By default, FFmpeg assumes compatibility with 32-bit MSVC builds of
AviSynth+ since that is the most widely-used and entrenched build
configuration. Users can override this and enable support for 32-bit
GCC builds of AviSynth+ by passing @code{-DAVSC_WIN32_GCC32} to
@code{--extra-cflags} when configuring FFmpeg.
64-bit builds of FFmpeg are not affected, and can use either MSVC or
GCC builds of AviSynth+ without any special flags.
@end float
@float NOTE
AviSynth(+) is loaded dynamically. Distributors can build FFmpeg
with @code{--enable-avisynth}, and the binaries will work regardless
of the end user having AviSynth installed. If/when an end user
would like to use AviSynth scripts, then they can install AviSynth(+)
and FFmpeg will be able to find and use it to open scripts.
@end float
@section Chromaprint
FFmpeg can make use of the Chromaprint library for generating audio fingerprints.
Pass @code{--enable-chromaprint} to configure to
enable it. See @url{https://acoustid.org/chromaprint}.
@section codec2
FFmpeg can make use of the codec2 library for codec2 decoding and encoding.
There is currently no native decoder, so libcodec2 must be used for decoding.
Go to @url{http://freedv.org/}, download "Codec 2 source archive".
Build and install using CMake. Debian users can install the libcodec2-dev package instead.
Once libcodec2 is installed you can pass @code{--enable-libcodec2} to configure to enable it.
The easiest way to use codec2 is with .c2 files, since they contain the mode information required for decoding.
To encode such a file, use a .c2 file extension and give the libcodec2 encoder the -mode option:
@code{ffmpeg -i input.wav -mode 700C output.c2}.
Playback is as simple as @code{ffplay output.c2}.
For a list of supported modes, run @code{ffmpeg -h encoder=libcodec2}.
Raw codec2 files are also supported.
To make sense of them the mode in use needs to be specified as a format option:
@code{ffmpeg -f codec2raw -mode 1300 -i input.raw output.wav}.
@section dav1d
FFmpeg can make use of the dav1d library for AV1 video decoding.
Go to @url{https://code.videolan.org/videolan/dav1d} and follow the instructions for
installing the library. Then pass @code{--enable-libdav1d} to configure to enable it.
@section davs2
FFmpeg can make use of the davs2 library for AVS2-P2/IEEE1857.4 video decoding.
Go to @url{https://github.com/pkuvcl/davs2} and follow the instructions for
installing the library. Then pass @code{--enable-libdavs2} to configure to
enable it.
@float NOTE
libdavs2 is under the GNU Public License Version 2 or later
(see @url{http://www.gnu.org/licenses/old-licenses/gpl-2.0.html} for
details), you must upgrade FFmpeg's license to GPL in order to use it.
@end float
@section Game Music Emu
FFmpeg can make use of the Game Music Emu library to read audio from supported video game
music file formats. Pass @code{--enable-libgme} to configure to
enable it. See @url{https://bitbucket.org/mpyne/game-music-emu/overview}.
@section Intel QuickSync Video
FFmpeg can use Intel QuickSync Video (QSV) for accelerated decoding and encoding
of multiple codecs. To use QSV, FFmpeg must be linked against the @code{libmfx}
dispatcher, which loads the actual decoding libraries.
The dispatcher is open source and can be downloaded from
@url{https://github.com/lu-zero/mfx_dispatch.git}. FFmpeg needs to be configured
with the @code{--enable-libmfx} option and @code{pkg-config} needs to be able to
locate the dispatcher's @code{.pc} files.
@section Kvazaar
FFmpeg can make use of the Kvazaar library for HEVC encoding.
Go to @url{https://github.com/ultravideo/kvazaar} and follow the
instructions for installing the library. Then pass
@code{--enable-libkvazaar} to configure to enable it.
@section LAME
FFmpeg can make use of the LAME library for MP3 encoding.
Go to @url{http://lame.sourceforge.net/} and follow the
instructions for installing the library.
Then pass @code{--enable-libmp3lame} to configure to enable it.
@section libilbc
iLBC is a narrowband speech codec that has been made freely available
by Google as part of the WebRTC project. libilbc is a packaging friendly
copy of the iLBC codec. FFmpeg can make use of the libilbc library for
iLBC decoding and encoding.
Go to @url{https://github.com/TimothyGu/libilbc} and follow the instructions for
installing the library. Then pass @code{--enable-libilbc} to configure to
enable it.
@section libvpx
FFmpeg can make use of the libvpx library for VP8/VP9 decoding and encoding.
Go to @url{http://www.webmproject.org/} and follow the instructions for
installing the library. Then pass @code{--enable-libvpx} to configure to
enable it.
@section ModPlug
FFmpeg can make use of this library, originating in Modplug-XMMS, to read from MOD-like music files.
See @url{https://github.com/Konstanty/libmodplug}. Pass @code{--enable-libmodplug} to configure to
enable it.
@section OpenCORE, VisualOn, and Fraunhofer libraries
@@ -202,10 +46,9 @@ upgrade FFmpeg's license to LGPL version 3 (or if you have enabled
GPL components, GPL version 3) by passing @code{--enable-version3} to configure in
order to use it.
The license of the Fraunhofer AAC library is incompatible with the GPL.
Therefore, for GPL builds, you have to pass @code{--enable-nonfree} to
configure in order to use it. To the best of our knowledge, it is
compatible with the LGPL.
The Fraunhofer AAC library is licensed under a license incompatible to the GPL
and is not known to be compatible to the LGPL. Therefore, you have to pass
@code{--enable-nonfree} to configure to use it.
@end float
@subsection OpenCORE AMR
@@ -228,15 +71,73 @@ Then pass @code{--enable-libvo-amrwbenc} to configure to enable it.
@subsection Fraunhofer AAC library
FFmpeg can make use of the Fraunhofer AAC library for AAC decoding & encoding.
FFmpeg can make use of the Fraunhofer AAC library for AAC encoding.
Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the
instructions for installing the library.
Then pass @code{--enable-libfdk-aac} to configure to enable it.
@section LAME
FFmpeg can make use of the LAME library for MP3 encoding.
Go to @url{http://lame.sourceforge.net/} and follow the
instructions for installing the library.
Then pass @code{--enable-libmp3lame} to configure to enable it.
@section TwoLAME
FFmpeg can make use of the TwoLAME library for MP2 encoding.
Go to @url{http://www.twolame.org/} and follow the
instructions for installing the library.
Then pass @code{--enable-libtwolame} to configure to enable it.
@section libcodec2 / codec2 general
FFmpeg can make use of libcodec2 for codec2 encoding and decoding.
There is currently no native decoder, so libcodec2 must be used for decoding.
Go to @url{http://freedv.org/}, download "Codec 2 source archive".
Build and install using CMake. Debian users can install the libcodec2-dev package instead.
Once libcodec2 is installed you can pass @code{--enable-libcodec2} to configure to enable it.
The easiest way to use codec2 is with .c2 files, since they contain the mode information required for decoding.
To encode such a file, use a .c2 file extension and give the libcodec2 encoder the -mode option:
@code{ffmpeg -i input.wav -mode 700C output.c2}.
Playback is as simple as @code{ffplay output.c2}.
For a list of supported modes, run @code{ffmpeg -h encoder=libcodec2}.
Raw codec2 files are also supported.
To make sense of them the mode in use needs to be specified as a format option:
@code{ffmpeg -f codec2raw -mode 1300 -i input.raw output.wav}.
@section libvpx
FFmpeg can make use of the libvpx library for VP8/VP9 encoding.
Go to @url{http://www.webmproject.org/} and follow the instructions for
installing the library. Then pass @code{--enable-libvpx} to configure to
enable it.
@section libwavpack
FFmpeg can make use of the libwavpack library for WavPack encoding.
Go to @url{http://www.wavpack.com/} and follow the instructions for
installing the library. Then pass @code{--enable-libwavpack} to configure to
enable it.
@section libxavs
FFmpeg can make use of the libxavs library for Xavs encoding.
Go to @url{http://xavs.sf.net/} and follow the instructions for
installing the library. Then pass @code{--enable-libxavs} to configure to
enable it.
@section OpenH264
FFmpeg can make use of the OpenH264 library for H.264 decoding and encoding.
FFmpeg can make use of the OpenH264 library for H.264 encoding and decoding.
Go to @url{http://www.openh264.org/} and follow the instructions for
installing the library. Then pass @code{--enable-libopenh264} to configure to
@@ -249,47 +150,6 @@ constrained baseline profile and CABAC.) Using it is mostly useful for
testing and for taking advantage of Cisco's patent portfolio license
(@url{http://www.openh264.org/BINARY_LICENSE.txt}).
@section OpenJPEG
FFmpeg can use the OpenJPEG libraries for decoding/encoding J2K videos. Go to
@url{http://www.openjpeg.org/} to get the libraries and follow the installation
instructions. To enable using OpenJPEG in FFmpeg, pass @code{--enable-libopenjpeg} to
@file{./configure}.
@section rav1e
FFmpeg can make use of rav1e (Rust AV1 Encoder) via its C bindings to encode videos.
Go to @url{https://github.com/xiph/rav1e/} and follow the instructions to build
the C library. To enable using rav1e in FFmpeg, pass @code{--enable-librav1e}
to @file{./configure}.
@section TwoLAME
FFmpeg can make use of the TwoLAME library for MP2 encoding.
Go to @url{http://www.twolame.org/} and follow the
instructions for installing the library.
Then pass @code{--enable-libtwolame} to configure to enable it.
@section VapourSynth
FFmpeg can read VapourSynth scripts as input. To enable support, pass
@code{--enable-vapoursynth} to configure. Vapoursynth is detected via
@code{pkg-config}. Versions 42 or greater supported.
See @url{http://www.vapoursynth.com/}.
Due to security concerns, Vapoursynth scripts will not
be autodetected so the input format has to be forced. For ff* CLI tools,
add @code{-f vapoursynth} before the input @code{-i yourscript.vpy}.
@section WavPack
FFmpeg can make use of the libwavpack library for WavPack encoding.
Go to @url{http://www.wavpack.com/} and follow the instructions for
installing the library. Then pass @code{--enable-libwavpack} to configure to
enable it.
@section x264
FFmpeg can make use of the x264 library for H.264 encoding.
@@ -318,37 +178,92 @@ x265 is under the GNU Public License Version 2 or later
details), you must upgrade FFmpeg's license to GPL in order to use it.
@end float
@section xavs
@section kvazaar
FFmpeg can make use of the xavs library for AVS encoding.
FFmpeg can make use of the kvazaar library for HEVC encoding.
Go to @url{http://xavs.sf.net/} and follow the instructions for
installing the library. Then pass @code{--enable-libxavs} to configure to
Go to @url{https://github.com/ultravideo/kvazaar} and follow the
instructions for installing the library. Then pass
@code{--enable-libkvazaar} to configure to enable it.
@section libilbc
iLBC is a narrowband speech codec that has been made freely available
by Google as part of the WebRTC project. libilbc is a packaging friendly
copy of the iLBC codec. FFmpeg can make use of the libilbc library for
iLBC encoding and decoding.
Go to @url{https://github.com/TimothyGu/libilbc} and follow the instructions for
installing the library. Then pass @code{--enable-libilbc} to configure to
enable it.
@section xavs2
@section libzvbi
FFmpeg can make use of the xavs2 library for AVS2-P2/IEEE1857.4 video encoding.
Go to @url{https://github.com/pkuvcl/xavs2} and follow the instructions for
installing the library. Then pass @code{--enable-libxavs2} to configure to
enable it.
@float NOTE
libxavs2 is under the GNU Public License Version 2 or later
(see @url{http://www.gnu.org/licenses/old-licenses/gpl-2.0.html} for
details), you must upgrade FFmpeg's license to GPL in order to use it.
@end float
@section ZVBI
ZVBI is a VBI decoding library which can be used by FFmpeg to decode DVB
libzvbi is a VBI decoding library which can be used by FFmpeg to decode DVB
teletext pages and DVB teletext subtitles.
Go to @url{http://sourceforge.net/projects/zapping/} and follow the instructions for
installing the library. Then pass @code{--enable-libzvbi} to configure to
enable it.
@section AviSynth
FFmpeg can read AviSynth scripts as input. To enable support, pass
@code{--enable-avisynth} to configure. The correct headers are
included in compat/avisynth/, which allows the user to enable support
without needing to search for these headers themselves.
For Windows, supported AviSynth variants are
@url{http://avisynth.nl, AviSynth 2.6 RC1 or higher} for 32-bit builds and
@url{http://avs-plus.net, AviSynth+ r1718 or higher} for 32-bit and 64-bit builds.
For Linux and OS X, the supported AviSynth variant is
@url{https://github.com/avxsynth/avxsynth, AvxSynth}.
@float NOTE
There is currently a regression in AviSynth+'s @code{capi.h} header as of
October 2016, which interferes with the ability for builds of FFmpeg to use
MSVC-built binaries of AviSynth. Until this is resolved, you can make sure
a known good version is installed by checking out a version from before
the regression occurred:
@code{git clone -b MT git://github.com/AviSynth/AviSynthPlus.git @*
cd AviSynthPlus @*
git checkout -b oldheader b4f292b4dbfad149697fb65c6a037bb3810813f9 @*
make install PREFIX=/install/prefix}
@end float
@float NOTE
AviSynth and AvxSynth are loaded dynamically. Distributors can build FFmpeg
with @code{--enable-avisynth}, and the binaries will work regardless of the
end user having AviSynth or AvxSynth installed - they'll only need to be
installed to use AviSynth scripts (obviously).
@end float
@section Intel QuickSync Video
FFmpeg can use Intel QuickSync Video (QSV) for accelerated encoding and decoding
of multiple codecs. To use QSV, FFmpeg must be linked against the @code{libmfx}
dispatcher, which loads the actual decoding libraries.
The dispatcher is open source and can be downloaded from
@url{https://github.com/lu-zero/mfx_dispatch.git}. FFmpeg needs to be configured
with the @code{--enable-libmfx} option and @code{pkg-config} needs to be able to
locate the dispatcher's @code{.pc} files.
@section AMD VCE
FFmpeg can use the AMD Advanced Media Framework library for accelerated H.264
and HEVC encoding on VCE enabled hardware under Windows.
To enable support you must obtain the AMF framework header files from
@url{https://github.com/GPUOpen-LibrariesAndSDKs/AMF.git}.
Create an @code{AMF/} directory in the system include path.
Copy the contents of @code{AMF/amf/public/include/} into that directory.
Then configure FFmpeg with @code{--enable-amf}.
@chapter Supported File Formats, Codecs or Features
You can use the @code{-formats} and @code{-codecs} options to have an exhaustive list.
@@ -419,9 +334,6 @@ library:
@tab Contains header with version and mode info, simplifying playback.
@item CRI ADX @tab X @tab X
@tab Audio-only format used in console video games.
@item CRI AIX @tab @tab X
@item CRI HCA @tab @tab X
@tab Audio-only format used in console video games.
@item Discworld II BMV @tab @tab X
@item Interplay C93 @tab @tab X
@tab Used in the game Cyberia from Interplay.
@@ -491,8 +403,6 @@ library:
@item IEC61937 encapsulation @tab X @tab X
@item IFF @tab @tab X
@tab Interchange File Format
@item IFV @tab @tab X
@tab A format used by some old CCTV DVRs.
@item iLBC @tab X @tab X
@item Interplay MVE @tab @tab X
@tab Format used in various Interplay computer games.
@@ -577,6 +487,7 @@ library:
@item raw aptX @tab X @tab X
@item raw aptX HD @tab X @tab X
@item raw Chinese AVS video @tab X @tab X
@item raw CRI ADX @tab X @tab X
@item raw Dirac @tab X @tab X
@item raw DNxHD @tab X @tab X
@item raw DTS @tab X @tab X
@@ -605,7 +516,6 @@ library:
@item raw VC-1 @tab X @tab X
@item raw PCM A-law @tab X @tab X
@item raw PCM mu-law @tab X @tab X
@item raw PCM Archimedes VIDC @tab X @tab X
@item raw PCM signed 8 bit @tab X @tab X
@item raw PCM signed 16 bit big-endian @tab X @tab X
@item raw PCM signed 16 bit little-endian @tab X @tab X
@@ -649,7 +559,6 @@ library:
@item SAP @tab X @tab X
@item SBG @tab @tab X
@item SDP @tab @tab X
@item SER @tab @tab X
@item Sega FILM/CPK @tab X @tab X
@tab Used in many Sega Saturn console games.
@item Silicon Graphics Movie @tab @tab X
@@ -710,7 +619,7 @@ library:
@item Psygnosis YOP @tab @tab X
@end multitable
@code{X} means that the feature in that column (encoding / decoding) is supported.
@code{X} means that encoding (resp. decoding) is supported.
@section Image Formats
@@ -780,7 +689,7 @@ following image formats are supported:
@tab X Window Dump image format
@end multitable
@code{X} means that the feature in that column (encoding / decoding) is supported.
@code{X} means that encoding (resp. decoding) is supported.
@code{E} means that support is provided through an external library.
@@ -818,14 +727,12 @@ following image formats are supported:
@item Autodesk Animator Flic video @tab @tab X
@item Autodesk RLE @tab @tab X
@tab fourcc: AASC
@item AV1 @tab E @tab E
@tab Supported through external libraries libaom, libdav1d and librav1e
@item AV1 @tab @tab E
@tab Supported through external library libaom
@item Avid 1:1 10-bit RGB Packer @tab X @tab X
@tab fourcc: AVrp
@item AVS (Audio Video Standard) video @tab @tab X
@tab Video encoding used by the Creature Shock game.
@item AVS2-P2/IEEE1857.4 @tab E @tab E
@tab Supported through external libraries libxavs2 and libdavs2
@item AYUV @tab X @tab X
@tab Microsoft uncompressed packed 4:4:4:4
@item Beam Software VB @tab @tab X
@@ -851,8 +758,6 @@ following image formats are supported:
@tab Codec used in Delphine Software International games.
@item Discworld II BMV Video @tab @tab X
@item Canopus Lossless Codec @tab @tab X
@item CDToons @tab @tab X
@tab Codec used in various Broderbund games.
@item Cinepak @tab @tab X
@item Cirrus Logic AccuPak @tab X @tab X
@tab fourcc: CLJR
@@ -897,7 +802,6 @@ following image formats are supported:
@tab fourcc: G2M2, G2M3
@item Go2Webinar @tab @tab X
@tab fourcc: G2M4
@item Gremlin Digital Video @tab @tab X
@item H.261 @tab X @tab X
@item H.263 / H.263-1996 @tab X @tab X
@item H.263+ / H.263-1998 / H.263 version 2 @tab X @tab X
@@ -918,7 +822,6 @@ following image formats are supported:
@tab IFF interleaved bitmap
@item IFF ByteRun1 @tab @tab X
@tab IFF run length encoded bitmap
@item Infinity IMM4 @tab @tab X
@item Intel H.263 @tab @tab X
@item Intel Indeo 2 @tab @tab X
@item Intel Indeo 3 @tab @tab X
@@ -972,8 +875,6 @@ following image formats are supported:
@tab Video encoding used in NuppelVideo files.
@item On2 VP3 @tab @tab X
@tab still experimental
@item On2 VP4 @tab @tab X
@tab fourcc: VP40
@item On2 VP5 @tab @tab X
@tab fourcc: VP50
@item On2 VP6 @tab @tab X
@@ -1068,7 +969,7 @@ following image formats are supported:
@tab Encoder works only in PAL8.
@end multitable
@code{X} means that the feature in that column (encoding / decoding) is supported.
@code{X} means that encoding (resp. decoding) is supported.
@code{E} means that support is provided through an external library.
@@ -1083,10 +984,8 @@ following image formats are supported:
@item AAC+ @tab E @tab IX
@tab encoding supported through external library libfdk-aac
@item AC-3 @tab IX @tab IX
@item ACELP.KELVIN @tab @tab X
@item ADPCM 4X Movie @tab @tab X
@item APDCM Yamaha AICA @tab @tab X
@item ADPCM Argonaut Games @tab @tab X
@item ADPCM CDROM XA @tab @tab X
@item ADPCM Creative Technology @tab @tab X
@tab 16 -> 4, 8 -> 4, 8 -> 3, 8 -> 2
@@ -1102,14 +1001,10 @@ following image formats are supported:
@item ADPCM G.726 @tab X @tab X
@item ADPCM IMA AMV @tab @tab X
@tab Used in AMV files
@item ADPCM IMA Cunning Developments @tab @tab X
@item ADPCM IMA Electronic Arts EACS @tab @tab X
@item ADPCM IMA Electronic Arts SEAD @tab @tab X
@item ADPCM IMA Funcom @tab @tab X
@item ADPCM IMA High Voltage Software ALP @tab @tab X
@item ADPCM IMA QuickTime @tab X @tab X
@item ADPCM IMA Simon & Schuster Interactive @tab X @tab X
@item ADPCM IMA Ubisoft APM @tab @tab X
@item ADPCM IMA Loki SDL MJPEG @tab @tab X
@item ADPCM IMA WAV @tab X @tab X
@item ADPCM IMA Westwood @tab @tab X
@@ -1139,7 +1034,6 @@ following image formats are supported:
@item ADPCM Westwood Studios IMA @tab @tab X
@tab Used in Westwood Studios games like Command and Conquer.
@item ADPCM Yamaha @tab X @tab X
@item ADPCM Zork @tab @tab X
@item AMR-NB @tab E @tab X
@tab encoding supported through external library libopencore-amrnb
@item AMR-WB @tab E @tab X
@@ -1154,14 +1048,12 @@ following image formats are supported:
@item ATRAC1 @tab @tab X
@item ATRAC3 @tab @tab X
@item ATRAC3+ @tab @tab X
@item ATRAC9 @tab @tab X
@item Bink Audio @tab @tab X
@tab Used in Bink and Smacker files in many games.
@item CELT @tab @tab E
@tab decoding supported through external library libcelt
@item codec2 @tab E @tab E
@tab en/decoding supported through external library libcodec2
@item CRI HCA @tab @tab X
@item Delphine Software International CIN audio @tab @tab X
@tab Codec used in Delphine Software International games.
@item Digital Speech Standard - Standard Play mode (DSS SP) @tab @tab X
@@ -1171,7 +1063,6 @@ following image formats are supported:
@item DCA (DTS Coherent Acoustics) @tab X @tab X
@tab supported extensions: XCh, XXCH, X96, XBR, XLL, LBR (partially)
@item Dolby E @tab @tab X
@item DPCM Gremlin @tab @tab X
@item DPCM id RoQ @tab X @tab X
@tab Used in Quake III, Jedi Knight 2 and other computer games.
@item DPCM Interplay @tab @tab X
@@ -1183,11 +1074,10 @@ following image formats are supported:
@item DPCM Sol @tab @tab X
@item DPCM Xan @tab @tab X
@tab Used in Origin's Wing Commander IV AVI files.
@item DPCM Xilam DERF @tab @tab X
@item DSD (Direct Stream Digital), least significant bit first @tab @tab X
@item DSD (Direct Stream Digital), most significant bit first @tab @tab X
@item DSD (Direct Stream Digital), least significant bit first, planar @tab @tab X
@item DSD (Direct Stream Digital), most significant bit first, planar @tab @tab X
@item DSD (Direct Stream Digitial), least significant bit first @tab @tab X
@item DSD (Direct Stream Digitial), most significant bit first @tab @tab X
@item DSD (Direct Stream Digitial), least significant bit first, planar @tab @tab X
@item DSD (Direct Stream Digitial), most significant bit first, planar @tab @tab X
@item DSP Group TrueSpeech @tab @tab X
@item DST (Direct Stream Transfer) @tab @tab X
@item DV audio @tab @tab X
@@ -1224,7 +1114,6 @@ following image formats are supported:
@tab encoding supported through external library libopus
@item PCM A-law @tab X @tab X
@item PCM mu-law @tab X @tab X
@item PCM Archimedes VIDC @tab X @tab X
@item PCM signed 8-bit planar @tab X @tab X
@item PCM signed 16-bit big-endian planar @tab X @tab X
@item PCM signed 16-bit little-endian planar @tab X @tab X
@@ -1250,6 +1139,7 @@ following image formats are supported:
@item PCM unsigned 24-bit little-endian @tab X @tab X
@item PCM unsigned 32-bit big-endian @tab X @tab X
@item PCM unsigned 32-bit little-endian @tab X @tab X
@item PCM Zork @tab @tab X
@item QCELP / PureVoice @tab @tab X
@item QDesign Music Codec 1 @tab @tab X
@item QDesign Music Codec 2 @tab @tab X
@@ -1296,7 +1186,7 @@ following image formats are supported:
@item Xbox Media Audio 2 @tab @tab X
@end multitable
@code{X} means that the feature in that column (encoding / decoding) is supported.
@code{X} means that encoding (resp. decoding) is supported.
@code{E} means that support is provided through an external library.
@@ -1340,7 +1230,6 @@ performance on systems without hardware floating point support).
@multitable @columnfractions .4 .1
@item Name @tab Support
@item AMQP @tab E
@item file @tab X
@item FTP @tab X
@item Gopher @tab X
@@ -1365,7 +1254,6 @@ performance on systems without hardware floating point support).
@item TCP @tab X
@item TLS @tab X
@item UDP @tab X
@item ZMQ @tab E
@end multitable
@code{X} means that the protocol is supported.

View File

@@ -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>}.
@@ -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.
@@ -400,19 +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

@@ -178,9 +178,6 @@ Capture the mouse pointer. Default is 0.
@item -capture_mouse_clicks
Capture the screen mouse clicks. Default is 0.
@item -capture_raw_data
Capture the raw device data. Default is 0.
Using this option may result in receiving the underlying data delivered to the AVFoundation framework. E.g. for muxed devices that sends raw DV data to the framework (like tape-based camcorders), setting this option to false results in extracted video frames captured in the designated pixel format only. Setting this option to true results in receiving the raw DV stream untouched.
@end table
@subsection Examples
@@ -211,13 +208,6 @@ Record video from the system default video device using the pixel format bgr0 an
$ ffmpeg -f avfoundation -pixel_format bgr0 -i "default:none" out.avi
@end example
@item
Record raw DV data from a suitable input device and write the output into out.dv:
@example
$ ffmpeg -f avfoundation -capture_raw_data true -i "zr100:none" out.dv
@end example
@end itemize
@section bktr
@@ -277,8 +267,7 @@ audio track.
@item list_devices
If set to @option{true}, print a list of devices and exit.
Defaults to @option{false}. This option is deprecated, please use the
@code{-sources} option of ffmpeg to list the available input devices.
Defaults to @option{false}.
@item list_formats
If set to @option{true}, print a list of supported formats and exit.
@@ -292,6 +281,11 @@ as @option{pal} (3 letters).
Default behavior is autodetection of the input video format, if the hardware
supports it.
@item bm_v210
This is a deprecated option, you can use @option{raw_format} instead.
If set to @samp{1}, video is captured in 10 bit v210 instead
of uyvy422. Not all Blackmagic devices support this option.
@item raw_format
Set the pixel format of the captured video.
Available values are:
@@ -332,12 +326,6 @@ Defaults to @samp{2}.
Sets the decklink device duplex mode. Must be @samp{unset}, @samp{half} or @samp{full}.
Defaults to @samp{unset}.
@item timecode_format
Timecode type to include in the frame and video stream metadata. Must be
@samp{none}, @samp{rp188vitc}, @samp{rp188vitc2}, @samp{rp188ltc},
@samp{rp188any}, @samp{vitc}, @samp{vitc2}, or @samp{serial}. Defaults to
@samp{none} (not included).
@item video_input
Sets the video input source. Must be @samp{unset}, @samp{sdi}, @samp{hdmi},
@samp{optical_sdi}, @samp{component}, @samp{composite} or @samp{s_video}.
@@ -376,28 +364,6 @@ If set to @option{true}, timestamps are forwarded as they are without removing
the initial offset.
Defaults to @option{false}.
@item timestamp_align
Capture start time alignment in seconds. If set to nonzero, input frames are
dropped till the system timestamp aligns with configured value.
Alignment difference of up to one frame duration is tolerated.
This is useful for maintaining input synchronization across N different
hardware devices deployed for 'N-way' redundancy. The system time of different
hardware devices should be synchronized with protocols such as NTP or PTP,
before using this option.
Note that this method is not foolproof. In some border cases input
synchronization may not happen due to thread scheduling jitters in the OS.
Either sync could go wrong by 1 frame or in a rarer case
@option{timestamp_align} seconds.
Defaults to @samp{0}.
@item wait_for_tc (@emph{bool})
Drop frames till a frame with timecode is received. Sometimes serial timecode
isn't received with the first input frame. If that happens, the stored stream
timecode will be inaccurate. If this option is set to @option{true}, input frames
are dropped till a frame with timecode is received.
Option @var{timecode_format} must be specified.
Defaults to @option{false}.
@end table
@subsection Examples
@@ -407,7 +373,7 @@ Defaults to @option{false}.
@item
List input devices:
@example
ffmpeg -sources decklink
ffmpeg -f decklink -list_devices 1 -i dummy
@end example
@item
@@ -425,7 +391,7 @@ ffmpeg -format_code Hi50 -f decklink -i 'Intensity Pro' -c:a copy -c:v copy outp
@item
Capture video clip at 1080i50 10 bit:
@example
ffmpeg -raw_format yuv422p10 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder' -c:a copy -c:v copy output.avi
ffmpeg -bm_v210 1 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder' -c:a copy -c:v copy output.avi
@end example
@item
@@ -436,6 +402,116 @@ ffmpeg -channels 16 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder'
@end itemize
@section kmsgrab
KMS video input device.
Captures the KMS scanout framebuffer associated with a specified CRTC or plane as a
DRM object that can be passed to other hardware functions.
Requires either DRM master or CAP_SYS_ADMIN to run.
If you don't understand what all of that means, you probably don't want this. Look at
@option{x11grab} instead.
@subsection Options
@table @option
@item device
DRM device to capture on. Defaults to @option{/dev/dri/card0}.
@item format
Pixel format of the framebuffer. Defaults to @option{bgr0}.
@item format_modifier
Format modifier to signal on output frames. This is necessary to import correctly into
some APIs, but can't be autodetected. See the libdrm documentation for possible values.
@item crtc_id
KMS CRTC ID to define the capture source. The first active plane on the given CRTC
will be used.
@item plane_id
KMS plane ID to define the capture source. Defaults to the first active plane found if
neither @option{crtc_id} nor @option{plane_id} are specified.
@item framerate
Framerate to capture at. This is not synchronised to any page flipping or framebuffer
changes - it just defines the interval at which the framebuffer is sampled. Sampling
faster than the framebuffer update rate will generate independent frames with the same
content. Defaults to @code{30}.
@end table
@subsection Examples
@itemize
@item
Capture from the first active plane, download the result to normal frames and encode.
This will only work if the framebuffer is both linear and mappable - if not, the result
may be scrambled or fail to download.
@example
ffmpeg -f kmsgrab -i - -vf 'hwdownload,format=bgr0' output.mp4
@end example
@item
Capture from CRTC ID 42 at 60fps, map the result to VAAPI, convert to NV12 and encode as H.264.
@example
ffmpeg -crtc_id 42 -framerate 60 -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,scale_vaapi=w=1920:h=1080:format=nv12' -c:v h264_vaapi output.mp4
@end example
@end itemize
@section libndi_newtek
The libndi_newtek input device provides capture capabilities for using NDI (Network
Device Interface, standard created by NewTek).
Input filename is a NDI source name that could be found by sending -find_sources 1
to command line - it has no specific syntax but human-readable formatted.
To enable this input device, you need the NDI SDK and you
need to configure with the appropriate @code{--extra-cflags}
and @code{--extra-ldflags}.
@subsection Options
@table @option
@item find_sources
If set to @option{true}, print a list of found/available NDI sources and exit.
Defaults to @option{false}.
@item wait_sources
Override time to wait until the number of online sources have changed.
Defaults to @option{0.5}.
@item allow_video_fields
When this flag is @option{false}, all video that you receive will be progressive.
Defaults to @option{true}.
@end table
@subsection Examples
@itemize
@item
List input devices:
@example
ffmpeg -f libndi_newtek -find_sources 1 -i dummy
@end example
@item
Restream to NDI:
@example
ffmpeg -f libndi_newtek -i "DEV-5.INTERNAL.M1STEREO.TV (NDI_SOURCE_NAME_1)" -f libndi_newtek -y NDI_SOURCE_NAME_2
@end example
@end itemize
@section dshow
Windows DirectShow input device.
@@ -800,7 +876,7 @@ ffplay -f iec61883 -i auto
Grab and record the input of a FireWire DV/HDV device,
using a packet buffer of 100000 packets if the source is HDV.
@example
ffmpeg -f iec61883 -i auto -dvbuffer 100000 out.mpg
ffmpeg -f iec61883 -i auto -hdvbuffer 100000 out.mpg
@end example
@end itemize
@@ -863,76 +939,6 @@ Set the number of channels. Default is 2.
@end table
@section kmsgrab
KMS video input device.
Captures the KMS scanout framebuffer associated with a specified CRTC or plane as a
DRM object that can be passed to other hardware functions.
Requires either DRM master or CAP_SYS_ADMIN to run.
If you don't understand what all of that means, you probably don't want this. Look at
@option{x11grab} instead.
@subsection Options
@table @option
@item device
DRM device to capture on. Defaults to @option{/dev/dri/card0}.
@item format
Pixel format of the framebuffer. Defaults to @option{bgr0}.
@item format_modifier
Format modifier to signal on output frames. This is necessary to import correctly into
some APIs, but can't be autodetected. See the libdrm documentation for possible values.
@item crtc_id
KMS CRTC ID to define the capture source. The first active plane on the given CRTC
will be used.
@item plane_id
KMS plane ID to define the capture source. Defaults to the first active plane found if
neither @option{crtc_id} nor @option{plane_id} are specified.
@item framerate
Framerate to capture at. This is not synchronised to any page flipping or framebuffer
changes - it just defines the interval at which the framebuffer is sampled. Sampling
faster than the framebuffer update rate will generate independent frames with the same
content. Defaults to @code{30}.
@end table
@subsection Examples
@itemize
@item
Capture from the first active plane, download the result to normal frames and encode.
This will only work if the framebuffer is both linear and mappable - if not, the result
may be scrambled or fail to download.
@example
ffmpeg -f kmsgrab -i - -vf 'hwdownload,format=bgr0' output.mp4
@end example
@item
Capture from CRTC ID 42 at 60fps, map the result to VAAPI, convert to NV12 and encode as H.264.
@example
ffmpeg -crtc_id 42 -framerate 60 -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,scale_vaapi=w=1920:h=1080:format=nv12' -c:v h264_vaapi output.mp4
@end example
@item
To capture only part of a plane the output can be cropped - this can be used to capture
a single window, as long as it has a known absolute position and size. For example, to
capture and encode the middle quarter of a 1920x1080 plane:
@example
ffmpeg -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,crop=960:540:480:270,scale_vaapi=960:540:nv12' -c:v h264_vaapi output.mp4
@end example
@end itemize
@section lavfi
Libavfilter input virtual device.
@@ -1071,21 +1077,6 @@ IIDC1394 input device, based on libdc1394 and libraw1394.
Requires the configure option @code{--enable-libdc1394}.
@subsection Options
@table @option
@item framerate
Set the frame rate. Default is @code{ntsc}, corresponding to a frame
rate of @code{30000/1001}.
@item pixel_format
Select the pixel format. Default is @code{uyvy422}.
@item video_size
Set the video size given as a string such as @code{640x480} or @code{hd720}.
Default is @code{qvga}.
@end table
@section openal
The OpenAL input device provides audio capture on all systems with a
@@ -1204,6 +1195,7 @@ Set the number of channels. Default is 2.
@end table
@section pulse
PulseAudio input device.
@@ -1527,7 +1519,7 @@ ffmpeg -f x11grab -follow_mouse centered -show_region 1 -framerate 25 -video_siz
@end example
@item video_size
Set the video frame size. Default is the full desktop.
Set the video frame size. Default value is @code{vga}.
@item grab_x
@item grab_y

View File

@@ -95,16 +95,17 @@ Stuff that didn't reach the codebase:
- 0cef06df0 checkasm: add HEVC MC tests
- e7078e842 hevcdsp: add x86 SIMD for MC
- 7993ec19a hevc: Add hevc_get_pixel_4/8/12/16/24/32/48/64
- new bitstream reader (see http://ffmpeg.org/pipermail/ffmpeg-devel/2017-April/209609.html)
- use av_cpu_max_align() instead of hardcoding alignment requirements (see https://ffmpeg.org/pipermail/ffmpeg-devel/2017-September/215834.html)
- f44ec22e0 lavc: use av_cpu_max_align() instead of hardcoding alignment requirements
- 4de220d2e frame: allow align=0 (meaning automatic) for av_frame_get_buffer()
- Support recovery from an already present HLS playlist (see 16cb06bb30)
- Remove all output devices (see 8e7e042d41, 8d3db95f20, 6ce13070bd, d46cd24986 and https://ffmpeg.org/pipermail/ffmpeg-devel/2017-September/216904.html)
- avcodec/libaomenc: export the Sequence Header OBU as extradata (See a024c3ce9a)
Collateral damage that needs work locally:
------------------------------------------
- Merge proresdec2.c and proresdec_lgpl.c
- Merge proresenc_anatoliy.c and proresenc_kostya.c
- Fix MIPS AC3 downmix

View File

@@ -47,8 +47,7 @@ We cannot provide help for scripts and/or third-party tools.
@anchor{How do I ask a question or send a message to a mailing list?}
@section How do I ask a question or send a message to a mailing list?
First you must @ref{How do I subscribe?, subscribe}. Then all you have to do is
send an email:
All you have to do is send an email:
@itemize
@item
@@ -58,14 +57,49 @@ ffmpeg-user mailing list.
@item
Email @email{libav-user@@ffmpeg.org} to send a message to the
libav-user mailing list.
@item
Email @email{ffmpeg-devel@@ffmpeg.org} to send a message to the
ffmpeg-devel mailing list.
@end itemize
If you are not subscribed to the mailing list then your question must be
manually approved. Approval may take several days, but the wait is
usually less. If you want the message to be sent with no delay then you
must subscribe first. See @ref{How do I subscribe?}
Please do not send a message, subscribe, and re-send the message: this
results in duplicates, causes more work for the admins, and may lower
your chance at getting an answer. However, you may do so if you first
@ref{How do I delete my message in the moderation queue?, delete your original message from the moderation queue}.
@chapter Subscribing / Unsubscribing
@section What does subscribing do?
Subscribing allows two things:
@itemize
@item
Your messages will show up in the mailing list without waiting in the
moderation queue and needing to be manually approved by a mailing list
admin.
@item
You will receive all messages to the mailing list including replies to
your messages. Non-subscribed users do not receive any messages.
@end itemize
@section Do I need to subscribe?
No. You can still send a message to the mailing list without
subscribing. See @ref{How do I ask a question or send a message to a mailing list?}
However, your message will need to be manually approved by a mailing
list admin, and you will not receive any mailing list messages or
replies.
You can ask to be CCd in your message, but replying users will
sometimes forget to do so.
You may also view and reply to messages via the @ref{Where are the archives?, archives}.
@anchor{How do I subscribe?}
@section How do I subscribe?
@@ -90,9 +124,6 @@ The process is the same for the other mailing lists.
Please avoid asking a mailing list admin to unsubscribe you unless you
are absolutely unable to do so by yourself. See @ref{Who do I contact if I have a problem with the mailing list?}
Note that it is possible to temporarily halt message delivery (vacation mode).
See @ref{How do I disable mail delivery without unsubscribing?}
@chapter Moderation Queue
@anchor{Why is my message awaiting moderator approval?}
@section Why is my message awaiting moderator approval?
@@ -103,6 +134,8 @@ must be manually approved by a mailing list admin:
These are:
@itemize
@item
Messages from users who are @strong{not} subscribed.
@item
Messages that exceed the @ref{What is the message size limit?, message size limit}.
@@ -115,13 +148,13 @@ 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 Freenode for someone to approve your message.
The queue is usually checked once or twice a day, but on occasion
several days may pass before someone checks the queue.
@anchor{How do I delete my message in the moderation queue?}
@section How do I delete my message in the moderation queue?
You should have received an email with the subject @emph{Your message to <mailing list name> awaits moderator approval}.
You should have received an email with the subject @emph{Your message to ffmpeg-user awaits moderator approval}.
A link is in the message that will allow you to delete your message
unless a mailing list admin already approved or rejected it.
@@ -142,9 +175,6 @@ Click the email link at the top of the message just under the subject
title. The link will provide the proper headers to keep the message
within the thread.
Note that you must be subscribed to send a message to the ffmpeg-user or
libav-user mailing lists.
@section How do I search the archives?
Perform a site search using your favorite search engine. Example:
@@ -157,12 +187,11 @@ Perform a site search using your favorite search engine. Example:
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.
Some users prefer the third-party Nabble interface which presents the
mailing lists in a typical forum layout.
There are also numerous third-party help sites such as
@url{https://superuser.com/tags/ffmpeg, Super User} and
@url{https://www.reddit.com/r/ffmpeg/, r/ffmpeg on reddit}.
There are also numerous third-party help sites such as Super User and
r/ffmpeg on reddit.
@anchor{What is top-posting?}
@section What is top-posting?
@@ -174,15 +203,16 @@ Instead, use trimmed interleaved/inline replies (@url{https://lists.ffmpeg.org/p
@anchor{What is the message size limit?}
@section What is the message size limit?
The message size limit is 1000 kilobytes. Please provide links to larger files
instead of attaching them.
The message size limit is 500 kilobytes for the user lists and 1000
kilobytes for ffmpeg-devel. Please provide links to larger files instead
of attaching them.
@section Where can I upload sample files?
Anywhere that is not too annoying for us to use.
Google Drive and Dropbox are acceptable if you need a file host, and
@url{https://0x0.st/, 0x0.st} is good for files under 256 MiB.
0x0.st is good for files under 256 MiB.
Small, short samples are preferred if possible.
@@ -229,54 +259,6 @@ or headers.
You can then filter the mailing list messages to their own folder.
@anchor{How do I disable mail delivery without unsubscribing?}
@section How do I disable mail delivery without unsubscribing?
Sometimes you may want to temporarily stop receiving all mailing list
messages. This "vacation mode" is simple to do:
@enumerate
@item
Go to the @url{https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-user/, ffmpeg-user mailing list info page}
@item
Enter your email address in the box at very bottom of the page and click the
@emph{Unsubscribe or edit options} box.
@item
Enter your password and click the @emph{Log in} button.
@item
Look for the @emph{Mail delivery} option. Here you can disable/enable mail
delivery. If you check @emph{Set globally} it will apply your choice to all
other FFmpeg mailing lists you are subscribed to.
@end enumerate
Alternatively, from your subscribed address, send a message to @email{ffmpeg-user-request@@ffmpeg.org}
with the subject @emph{set delivery off}. To re-enable mail delivery send a
message to @email{ffmpeg-user-request@@ffmpeg.org} with the subject
@emph{set delivery on}.
@anchor{Why is the mailing list munging my address?}
@section Why is the mailing list munging my address?
This is due to subscribers that use an email service with a DMARC reject policy
which adds difficulties to mailing list operators.
The mailing list must re-write (munge) the @emph{From:} header for such users;
otherwise their email service will reject and bounce the message resulting in
automatic unsubscribing from the mailing list.
When sending a message these users will see @emph{via <mailing list name>}
added to their name and the @emph{From:} address munged to the address of
the particular mailing list.
If you want to avoid this then please use a different email service.
Note that ffmpeg-devel does not apply any munging as it causes issues with
patch authorship. As a result users with an email service with a DMARC reject
policy may be automatically unsubscribed due to rejected and bounced messages.
@chapter Rules and Etiquette
@section What are the rules and the proper etiquette?
@@ -375,15 +357,6 @@ form a multi-part message is recommended by email standards.
Check your spam folder.
@end itemize
@anchor{Why do I keep getting unsubscribed from ffmpeg-devel?}
@section Why do I keep getting unsubscribed from ffmpeg-devel?
Users with an email service that has a DMARC reject or quarantine policy may be
automatically unsubscribed from the ffmpeg-devel mailing list due to the mailing
list messages being continuously rejected and bounced back.
Consider using a different email service.
@anchor{Who do I contact if I have a problem with the mailing list?}
@section Who do I contact if I have a problem with the mailing list?

View File

@@ -33,7 +33,7 @@ At the beginning of a chapter section there may be an optional timebase to be
used for start/end values. It must be in form
@samp{TIMEBASE=@var{num}/@var{den}}, where @var{num} and @var{den} are
integers. If the timebase is missing then start/end times are assumed to
be in nanoseconds.
be in milliseconds.
Next a chapter section must contain chapter start and end times in form
@samp{START=@var{num}}, @samp{END=@var{num}}, where @var{num} is a positive

View File

@@ -51,14 +51,16 @@ the decode process starts. Call ff_thread_finish_setup() afterwards. If
some code can't be moved, have update_thread_context() run it in the next
thread.
If the codec allocates writable tables in its init(), add an init_thread_copy()
which re-allocates them for other threads.
Add AV_CODEC_CAP_FRAME_THREADS to the codec capabilities. There will be very little
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
AVCodec.caps_internal and use ff_thread_get_buffer() to allocate frames. The
ff_thread_report/await_progress(), set AVCodecInternal.allocate_progress. The
frames must then be freed with ff_thread_release_buffer().
Otherwise decode directly into the user-supplied frames.
Otherwise leave it at zero and decode directly into the user-supplied frames.
Call ff_thread_report_progress() after some part of the current picture has decoded.
A good place to put this is where draw_horiz_band() is called - add this if it isn't

View File

@@ -94,25 +94,21 @@ compatibility with software that only supports a single audio stream in AVI
@anchor{chromaprint}
@section chromaprint
Chromaprint fingerprinter.
Chromaprint fingerprinter
This muxer feeds audio data to the Chromaprint library,
which generates a fingerprint for the provided audio data. See @url{https://acoustid.org/chromaprint}
It takes a single signed native-endian 16-bit raw audio stream of at most 2 channels.
This muxer feeds audio data to the Chromaprint library, which generates
a fingerprint for the provided audio data. It takes a single signed
native-endian 16-bit raw audio stream.
@subsection Options
@table @option
@item silence_threshold
Threshold for detecting silence. Range is from -1 to 32767, where -1 disables
silence detection. Silence detection can only be used with version 3 of the
algorithm.
Silence detection must be disabled for use with the AcoustID service. Default is -1.
Threshold for detecting silence, ranges from 0 to 32767. -1 for default
(required for use with the AcoustID service).
@item algorithm
Version of algorithm to fingerprint with. Range is 0 to 4.
Version 3 enables silence detection. Default is 1.
Algorithm index to fingerprint with.
@item fp_format
Format to output the fingerprint as. Accepts the following options:
@@ -124,7 +120,7 @@ Binary raw fingerprint
Binary compressed fingerprint
@item base64
Base64 compressed fingerprint @emph{(default)}
Base64 compressed fingerprint
@end table
@@ -218,152 +214,58 @@ It creates a MPD manifest file and segment files for each stream.
The segment filename might contain pre-defined identifiers used with SegmentTemplate
as defined in section 5.3.9.4.4 of the standard. Available identifiers are "$RepresentationID$",
"$Number$", "$Bandwidth$" and "$Time$".
In addition to the standard identifiers, an ffmpeg-specific "$ext$" identifier is also supported.
When specified ffmpeg will replace $ext$ in the file name with muxing format's extensions such as mp4, webm etc.,
@example
ffmpeg -re -i <input> -map 0 -map 0 -c:a libfdk_aac -c:v libx264 \
-b:v:0 800k -b:v:1 300k -s:v:1 320x170 -profile:v:1 baseline \
-profile:v:0 main -bf 1 -keyint_min 120 -g 120 -sc_threshold 0 \
-b_strategy 0 -ar:a:1 22050 -use_timeline 1 -use_template 1 \
-window_size 5 -adaptation_sets "id=0,streams=v id=1,streams=a" \
ffmpeg -re -i <input> -map 0 -map 0 -c:a libfdk_aac -c:v libx264
-b:v:0 800k -b:v:1 300k -s:v:1 320x170 -profile:v:1 baseline
-profile:v:0 main -bf 1 -keyint_min 120 -g 120 -sc_threshold 0
-b_strategy 0 -ar:a:1 22050 -use_timeline 1 -use_template 1
-window_size 5 -adaptation_sets "id=0,streams=v id=1,streams=a"
-f dash /path/to/out.mpd
@end example
@table @option
@item min_seg_duration @var{microseconds}
This is a deprecated option to set the segment length in microseconds, use @var{seg_duration} instead.
@item seg_duration @var{duration}
Set the segment length in seconds (fractional value can be set). The value is
treated as average segment duration when @var{use_template} is enabled and
@item frag_duration @var{duration}
Set the length in seconds of fragments within segments (fractional value can be set).
@item frag_type @var{type}
Set the type of interval for fragmentation.
@item window_size @var{size}
@item -min_seg_duration @var{microseconds}
Set the segment length in microseconds.
@item -window_size @var{size}
Set the maximum number of segments kept in the manifest.
@item extra_window_size @var{size}
@item -extra_window_size @var{size}
Set the maximum number of segments kept outside of the manifest before removing from disk.
@item remove_at_exit @var{remove}
@item -remove_at_exit @var{remove}
Enable (1) or disable (0) removal of all segments when finished.
@item use_template @var{template}
@item -use_template @var{template}
Enable (1) or disable (0) use of SegmentTemplate instead of SegmentList.
@item use_timeline @var{timeline}
@item -use_timeline @var{timeline}
Enable (1) or disable (0) use of SegmentTimeline in SegmentTemplate.
@item single_file @var{single_file}
@item -single_file @var{single_file}
Enable (1) or disable (0) storing all segments in one file, accessed using byte ranges.
@item single_file_name @var{file_name}
DASH-templated name to be used for baseURL. Implies @var{single_file} set to "1". In the template, "$ext$" is replaced with the file name extension specific for the segment format.
@item init_seg_name @var{init_name}
DASH-templated name to used for the initialization segment. Default is "init-stream$RepresentationID$.$ext$". "$ext$" is replaced with the file name extension specific for the segment format.
@item media_seg_name @var{segment_name}
DASH-templated name to used for the media segments. Default is "chunk-stream$RepresentationID$-$Number%05d$.$ext$". "$ext$" is replaced with the file name extension specific for the segment format.
@item utc_timing_url @var{utc_url}
@item -single_file_name @var{file_name}
DASH-templated name to be used for baseURL. Implies @var{single_file} set to "1".
@item -init_seg_name @var{init_name}
DASH-templated name to used for the initialization segment. Default is "init-stream$RepresentationID$.m4s"
@item -media_seg_name @var{segment_name}
DASH-templated name to used for the media segments. Default is "chunk-stream$RepresentationID$-$Number%05d$.m4s"
@item -utc_timing_url @var{utc_url}
URL of the page that will return the UTC timestamp in ISO format. Example: "https://time.akamai.com/?iso"
@item method @var{method}
Use the given HTTP method to create output files. Generally set to PUT or POST.
@item http_user_agent @var{user_agent}
@item -http_user_agent @var{user_agent}
Override User-Agent field in HTTP header. Applicable only for HTTP output.
@item http_persistent @var{http_persistent}
@item -http_persistent @var{http_persistent}
Use persistent HTTP connections. Applicable only for HTTP output.
@item hls_playlist @var{hls_playlist}
@item -hls_playlist @var{hls_playlist}
Generate HLS playlist files as well. The master playlist is generated with the filename master.m3u8.
One media playlist file is generated for each stream with filenames media_0.m3u8, media_1.m3u8, etc.
@item streaming @var{streaming}
@item -streaming @var{streaming}
Enable (1) or disable (0) chunk streaming mode of output. In chunk streaming
mode, each frame will be a moof fragment which forms a chunk.
@item adaptation_sets @var{adaptation_sets}
@item -adaptation_sets @var{adaptation_sets}
Assign streams to AdaptationSets. Syntax is "id=x,streams=a,b,c id=y,streams=d,e" with x and y being the IDs
of the adaptation sets and a,b,c,d and e are the indices of the mapped streams.
To map all video (or audio) streams to an AdaptationSet, "v" (or "a") can be used as stream identifier instead of IDs.
When no assignment is defined, this defaults to an AdaptationSet for each stream.
Optional syntax is "id=x,seg_duration=x,frag_duration=x,frag_type=type,descriptor=descriptor_string,streams=a,b,c id=y,seg_duration=y,frag_type=type,streams=d,e" and so on,
descriptor is useful to the scheme defined by ISO/IEC 23009-1:2014/Amd.2:2015.
For example, -adaptation_sets "id=0,descriptor=<SupplementalProperty schemeIdUri=\"urn:mpeg:dash:srd:2014\" value=\"0,0,0,1,1,2,2\"/>,streams=v".
Please note that descriptor string should be a self-closing xml tag.
seg_duration, frag_duration and frag_type override the global option values for each adaptation set.
For example, -adaptation_sets "id=0,seg_duration=2,frag_duration=1,frag_type=duration,streams=v id=1,seg_duration=2,frag_type=none,streams=a"
type_id marks an adaptation set as containing streams meant to be used for Trick Mode for the referenced adaptation set.
For example, -adaptation_sets "id=0,seg_duration=2,frag_type=none,streams=0 id=1,seg_duration=10,frag_type=none,trick_id=0,streams=1"
@item timeout @var{timeout}
@item -timeout @var{timeout}
Set timeout for socket I/O operations. Applicable only for HTTP output.
@item index_correction @var{index_correction}
Enable (1) or Disable (0) segment index correction logic. Applicable only when
@var{use_template} is enabled and @var{use_timeline} is disabled.
When enabled, the logic monitors the flow of segment indexes. If a streams's
segment index value is not at the expected real time position, then the logic
corrects that index value.
Typically this logic is needed in live streaming use cases. The network bandwidth
fluctuations are common during long run streaming. Each fluctuation can cause
the segment indexes fall behind the expected real time position.
@item format_options @var{options_list}
Set container format (mp4/webm) options using a @code{:} separated list of
key=value parameters. Values containing @code{:} special characters must be
escaped.
@item global_sidx @var{global_sidx}
Write global SIDX atom. Applicable only for single file, mp4 output, non-streaming mode.
@item dash_segment_type @var{dash_segment_type}
Possible values:
@table @option
@item auto
If this flag is set, the dash segment files format will be selected based on the stream codec. This is the default mode.
@item mp4
If this flag is set, the dash segment files will be in in ISOBMFF format.
@item webm
If this flag is set, the dash segment files will be in in WebM format.
@end table
@item ignore_io_errors @var{ignore_io_errors}
Ignore IO errors during open and write. Useful for long-duration runs with network output.
@item lhls @var{lhls}
Enable Low-latency HLS(LHLS). Adds #EXT-X-PREFETCH tag with current segment's URI.
Apple doesn't have an official spec for LHLS. Meanwhile hls.js player folks are
trying to standardize a open LHLS spec. The draft spec is available in https://github.com/video-dev/hlsjs-rfcs/blob/lhls-spec/proposals/0001-lhls.md
This option will also try to comply with the above open spec, till Apple's spec officially supports it.
Applicable only when @var{streaming} and @var{hls_playlist} options are enabled.
This is an experimental feature.
@item ldash @var{ldash}
Enable Low-latency Dash by constraining the presence and values of some elements.
@item master_m3u8_publish_rate @var{master_m3u8_publish_rate}
Publish master playlist repeatedly every after specified number of segment intervals.
@item write_prft @var{write_prft}
Write Producer Reference Time elements on supported streams. This also enables writing
prft boxes in the underlying muxer. Applicable only when the @var{utc_url} option is enabled.
It's set to auto by default, in which case the muxer will attempt to enable it only in modes
that require it.
@item mpd_profile @var{mpd_profile}
Set one or more manifest profiles.
@item http_opts @var{http_opts}
A :-separated list of key=value options to pass to the underlying HTTP
protocol. Applicable only for HTTP output.
@item target_latency @var{target_latency}
Set an intended target latency in seconds (fractional value can be set) for serving. Applicable only when @var{streaming} and @var{write_prft} options are enabled.
This is an informative fields clients can use to measure the latency of the service.
@item min_playback_rate @var{min_playback_rate}
Set the minimum playback rate indicated as appropriate for the purposes of automatically
adjusting playback latency and buffer occupancy during normal playback by clients.
@item max_playback_rate @var{max_playback_rate}
Set the maximum playback rate indicated as appropriate for the purposes of automatically
adjusting playback latency and buffer occupancy during normal playback by clients.
@end table
@anchor{framecrc}
@@ -648,9 +550,6 @@ Set the starting sequence numbers according to @var{start_number} option value.
@item epoch
The start number will be the seconds since epoch (1970-01-01 00:00:00)
@item epoch_us
The start number will be the microseconds since epoch (1970-01-01 00:00:00)
@item datetime
The start number will be based on the current date/time as YYYYmmddHHMMSS. e.g. 20161231235759.
@@ -687,7 +586,7 @@ This example will produce the playlist, @file{out.m3u8}, and segment files:
but only the file name part without any path info will be contained in the m3u8 segment list.
Should a relative path be specified, the path of the created segment
files will be relative to the current working directory.
When strftime_mkdir is set, the whole expanded value of @var{filename} will be written into the m3u8 segment list.
When use_localtime_mkdir is set, the whole expanded value of @var{filename} will be written into the m3u8 segment list.
When @code{var_stream_map} is set with two or more variant streams, the
@var{filename} pattern must contain the string "%v", this string specifies
@@ -702,8 +601,7 @@ This example will produce the playlists segment file sets:
@file{file_1_000.ts}, @file{file_1_001.ts}, @file{file_1_002.ts}, etc.
The string "%v" may be present in the filename or in the last directory name
containing the file, but only in one of them. (Additionally, %v may appear multiple times in the last
sub-directory or filename.) If the string %v is present in the directory name, then
containing the file. If the string is present in the directory name, then
sub-directories are created after expanding the directory name pattern. This
enables creation of segments corresponding to different variant streams in
subdirectories.
@@ -717,40 +615,34 @@ This example will produce the playlists segment file sets:
@file{vs1/file_000.ts}, @file{vs1/file_001.ts}, @file{vs1/file_002.ts}, etc.
@item use_localtime
Same as strftime option, will be deprecated.
@item strftime
Use strftime() on @var{filename} to expand the segment filename with localtime.
The segment number is also available in this mode, but to use it, you need to specify second_level_segment_index
hls_flag and %%d will be the specifier.
@example
ffmpeg -i in.nut -strftime 1 -hls_segment_filename 'file-%Y%m%d-%s.ts' out.m3u8
ffmpeg -i in.nut -use_localtime 1 -hls_segment_filename 'file-%Y%m%d-%s.ts' out.m3u8
@end example
This example will produce the playlist, @file{out.m3u8}, and segment files:
@file{file-20160215-1455569023.ts}, @file{file-20160215-1455569024.ts}, etc.
Note: On some systems/environments, the @code{%s} specifier is not available. See
@code{strftime()} documentation.
@example
ffmpeg -i in.nut -strftime 1 -hls_flags second_level_segment_index -hls_segment_filename 'file-%Y%m%d-%%04d.ts' out.m3u8
ffmpeg -i in.nut -use_localtime 1 -hls_flags second_level_segment_index -hls_segment_filename 'file-%Y%m%d-%%04d.ts' out.m3u8
@end example
This example will produce the playlist, @file{out.m3u8}, and segment files:
@file{file-20160215-0001.ts}, @file{file-20160215-0002.ts}, etc.
@item use_localtime_mkdir
Same as strftime_mkdir option, will be deprecated .
@item strftime_mkdir
Used together with -strftime_mkdir, it will create all subdirectories which
Used together with -use_localtime, it will create all subdirectories which
is expanded in @var{filename}.
@example
ffmpeg -i in.nut -strftime 1 -strftime_mkdir 1 -hls_segment_filename '%Y%m%d/file-%Y%m%d-%s.ts' out.m3u8
ffmpeg -i in.nut -use_localtime 1 -use_localtime_mkdir 1 -hls_segment_filename '%Y%m%d/file-%Y%m%d-%s.ts' out.m3u8
@end example
This example will create a directory 201560215 (if it does not exist), and then
produce the playlist, @file{out.m3u8}, and segment files:
@file{20160215/file-20160215-1455569023.ts}, @file{20160215/file-20160215-1455569024.ts}, etc.
@example
ffmpeg -i in.nut -strftime 1 -strftime_mkdir 1 -hls_segment_filename '%Y/%m/%d/file-%Y%m%d-%s.ts' out.m3u8
ffmpeg -i in.nut -use_localtime 1 -use_localtime_mkdir 1 -hls_segment_filename '%Y/%m/%d/file-%Y%m%d-%s.ts' out.m3u8
@end example
This example will create a directory hierarchy 2016/02/15 (if any of them do not exist), and then
produce the playlist, @file{out.m3u8}, and segment files:
@@ -835,20 +727,17 @@ Possible values:
@table @samp
@item mpegts
Output segment files in MPEG-2 Transport Stream format. This is
compatible with all HLS versions.
If this flag is set, the hls segment files will format to mpegts.
the mpegts files is used in all hls versions.
@item fmp4
Output segment files in fragmented MP4 format, similar to MPEG-DASH.
fmp4 files may be used in HLS version 7 and above.
If this flag is set, the hls segment files will format to fragment mp4 looks like dash.
the fmp4 files is used in hls after version 7.
@end table
@item hls_fmp4_init_filename @var{filename}
Set filename to the fragment files header file, default filename is @file{init.mp4}.
@item hls_fmp4_init_resend
Resend init file after m3u8 file refresh every time, default is @var{0}.
set filename to the fragment files header file, default filename is @file{init.mp4}.
When @code{var_stream_map} is set with two or more variant streams, the
@var{filename} pattern must contain the string "%v", this string specifies
@@ -902,10 +791,6 @@ including the file containing the AES encryption key.
Add the @code{#EXT-X-INDEPENDENT-SEGMENTS} to playlists that has video segments
and when all the segments of that playlist are guaranteed to start with a Key frame.
@item iframes_only
Add the @code{#EXT-X-I-FRAMES-ONLY} to playlists that has video segments
and can play only I-frames in the @code{#EXT-X-BYTERANGE} mode.
@item split_by_time
Allow segments to start on frames other than keyframes. This improves
behavior on some players when the time between keyframes is inconsistent,
@@ -917,24 +802,24 @@ Generate @code{EXT-X-PROGRAM-DATE-TIME} tags.
@item second_level_segment_index
Makes it possible to use segment indexes as %%d in hls_segment_filename expression
besides date/time values when strftime is on.
besides date/time values when use_localtime is on.
To get fixed width numbers with trailing zeroes, %%0xd format is available where x is the required width.
@item second_level_segment_size
Makes it possible to use segment sizes (counted in bytes) as %%s in hls_segment_filename
expression besides date/time values when strftime is on.
expression besides date/time values when use_localtime is on.
To get fixed width numbers with trailing zeroes, %%0xs format is available where x is the required width.
@item second_level_segment_duration
Makes it possible to use segment duration (calculated in microseconds) as %%t in hls_segment_filename
expression besides date/time values when strftime is on.
expression besides date/time values when use_localtime is on.
To get fixed width numbers with trailing zeroes, %%0xt format is available where x is the required width.
@example
ffmpeg -i sample.mpeg \
-f hls -hls_time 3 -hls_list_size 5 \
-hls_flags second_level_segment_index+second_level_segment_size+second_level_segment_duration \
-strftime 1 -strftime_mkdir 1 -hls_segment_filename "segment_%Y%m%d%H%M%S_%%04d_%%08s_%%013t.ts" stream.m3u8
-use_localtime 1 -use_localtime_mkdir 1 -hls_segment_filename "segment_%Y%m%d%H%M%S_%%04d_%%08s_%%013t.ts" stream.m3u8
@end example
This will produce segments like this:
@file{segment_20170102194334_0003_00122200_0000003000000.ts}, @file{segment_20170102194334_0004_00120072_0000003000000.ts} etc.
@@ -942,11 +827,7 @@ This will produce segments like this:
@item temp_file
Write segment data to filename.tmp and rename to filename only once the segment is complete. A webserver
serving up segments can be configured to reject requests to *.tmp to prevent access to in-progress segments
before they have been added to the m3u8 playlist. This flag also affects how m3u8 playlist files are created.
If this flag is set, all playlist files will written into temporary file and renamed after they are complete, similarly as segments are handled.
But playlists with @code{file} protocol and with type (@code{hls_playlist_type}) other than @code{vod}
are always written into temporary file regardless of this flag. Master playlist files (@code{master_pl_name}), if any, with @code{file} protocol,
are always written into temporary file regardless of this flag if @code{master_pl_publish_rate} value is other than zero.
before they have been added to the m3u8 playlist.
@end table
@@ -997,21 +878,7 @@ This example creates two hls variant streams. The first variant stream will
contain video stream of bitrate 1000k and audio stream of bitrate 64k and the
second variant stream will contain video stream of bitrate 256k and audio
stream of bitrate 32k. Here, two media playlist with file names out_0.m3u8 and
out_1.m3u8 will be created. If you want something meaningful text instead of indexes
in result names, you may specify names for each or some of the variants
as in the following example.
@example
ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \
-map 0:v -map 0:a -map 0:v -map 0:a -f hls -var_stream_map "v:0,a:0,name:my_hd v:1,a:1,name:my_sd" \
http://example.com/live/out_%v.m3u8
@end example
This example creates two hls variant streams as in the previous one.
But here, the two media playlist with file names out_my_hd.m3u8 and
out_my_sd.m3u8 will be created.
out_1.m3u8 will be created.
@example
ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k \
-map 0:v -map 0:a -map 0:v -f hls -var_stream_map "v:0 a:0 v:1" \
@@ -1045,52 +912,6 @@ and they are mapped to the two video only variant streams with audio group names
By default, a single hls variant containing all the encoded streams is created.
@example
ffmpeg -re -i in.ts -b:a:0 32k -b:a:1 64k -b:v:0 1000k \
-map 0:a -map 0:a -map 0:v -f hls \
-var_stream_map "a:0,agroup:aud_low,default:yes a:1,agroup:aud_low v:0,agroup:aud_low" \
-master_pl_name master.m3u8 \
http://example.com/live/out_%v.m3u8
@end example
This example creates two audio only and one video only variant streams. In
addition to the #EXT-X-STREAM-INF tag for each variant stream in the master
playlist, #EXT-X-MEDIA tag is also added for the two audio only variant streams
and they are mapped to the one video only variant streams with audio group name
'aud_low', and the audio group have default stat is NO or YES.
By default, a single hls variant containing all the encoded streams is created.
@example
ffmpeg -re -i in.ts -b:a:0 32k -b:a:1 64k -b:v:0 1000k \
-map 0:a -map 0:a -map 0:v -f hls \
-var_stream_map "a:0,agroup:aud_low,default:yes,language:ENG a:1,agroup:aud_low,language:CHN v:0,agroup:aud_low" \
-master_pl_name master.m3u8 \
http://example.com/live/out_%v.m3u8
@end example
This example creates two audio only and one video only variant streams. In
addition to the #EXT-X-STREAM-INF tag for each variant stream in the master
playlist, #EXT-X-MEDIA tag is also added for the two audio only variant streams
and they are mapped to the one video only variant streams with audio group name
'aud_low', and the audio group have default stat is NO or YES, and one audio
have and language is named ENG, the other audio language is named CHN.
By default, a single hls variant containing all the encoded streams is created.
@example
ffmpeg -y -i input_with_subtitle.mkv \
-b:v:0 5250k -c:v h264 -pix_fmt yuv420p -profile:v main -level 4.1 \
-b:a:0 256k \
-c:s webvtt -c:a mp2 -ar 48000 -ac 2 -map 0:v -map 0:a:0 -map 0:s:0 \
-f hls -var_stream_map "v:0,a:0,s:0,sgroup:subtitle" \
-master_pl_name master.m3u8 -t 300 -hls_time 10 -hls_init_time 4 -hls_list_size \
10 -master_pl_publish_rate 10 -hls_flags \
delete_segments+discont_start+split_by_time ./tmp/video.m3u8
@end example
This example adds @code{#EXT-X-MEDIA} tag with @code{TYPE=SUBTITLES} in
the master playlist with webvtt subtitle group name 'subtitle'. Please make sure
the input file has one text subtitle stream at least.
@item cc_stream_map
Map string which specifies different closed captions groups and their
attributes. The closed captions stream groups are separated by space.
@@ -1111,7 +932,7 @@ ffmpeg -re -i in.ts -b:v 1000k -b:a 64k -a53cc 1 -f hls \
http://example.com/live/out.m3u8
@end example
This example adds @code{#EXT-X-MEDIA} tag with @code{TYPE=CLOSED-CAPTIONS} in
the master playlist with group name 'cc', language 'en' (english) and
the master playlist with group name 'cc', langauge 'en' (english) and
INSTREAM-ID 'CC1'. Also, it adds @code{CLOSED-CAPTIONS} attribute with group
name 'cc' for the output variant stream.
@example
@@ -1154,12 +975,6 @@ Use persistent HTTP connections. Applicable only for HTTP output.
@item timeout
Set timeout for socket I/O operations. Applicable only for HTTP output.
@item -ignore_io_errors
Ignore IO errors during open, write and delete. Useful for long-duration runs with network output.
@item headers
Set custom HTTP headers, can override built in default headers. Applicable only for HTTP output.
@end table
@anchor{ico}
@@ -1225,37 +1040,6 @@ The pattern "img%%-%d.jpg" will specify a sequence of filenames of the
form @file{img%-1.jpg}, @file{img%-2.jpg}, ..., @file{img%-10.jpg},
etc.
The image muxer supports the .Y.U.V image file format. This format is
special in that that each image frame consists of three files, for
each of the YUV420P components. To read or write this image file format,
specify the name of the '.Y' file. The muxer will automatically open the
'.U' and '.V' files as required.
@subsection Options
@table @option
@item frame_pts
If set to 1, expand the filename with pts from pkt->pts.
Default value is 0.
@item start_number
Start the sequence from the specified number. Default value is 1.
@item update
If set to 1, the filename will always be interpreted as just a
filename, not a pattern, and the corresponding file will be continuously
overwritten with new images. Default value is 0.
@item strftime
If set to 1, expand the filename with date and time information from
@code{strftime()}. Default value is 0.
@item protocol_opts @var{options_list}
Set protocol options as a :-separated list of key=value parameters. Values
containing the @code{:} special character must be escaped.
@end table
@subsection Examples
The following example shows how to use @command{ffmpeg} for creating a
@@ -1296,11 +1080,31 @@ You can set the file name with current frame's PTS:
ffmpeg -f v4l2 -r 1 -i /dev/video0 -copyts -f image2 -frame_pts true %d.jpg"
@end example
A more complex example is to publish contents of your desktop directly to a
WebDAV server every second:
@example
ffmpeg -f x11grab -framerate 1 -i :0.0 -q:v 6 -update 1 -protocol_opts method=PUT http://example.com/desktop.jpg
@end example
@subsection Options
@table @option
@item frame_pts
If set to 1, expand the filename with pts from pkt->pts.
Default value is 0.
@item start_number
Start the sequence from the specified number. Default value is 1.
@item update
If set to 1, the filename will always be interpreted as just a
filename, not a pattern, and the corresponding file will be continuously
overwritten with new images. Default value is 0.
@item strftime
If set to 1, expand the filename with date and time information from
@code{strftime()}. Default value is 0.
@end table
The image muxer supports the .Y.U.V image file format. This format is
special in that that each image frame consists of three files, for
each of the YUV420P components. To read or write this image file format,
specify the name of the '.Y' file. The muxer will automatically open the
'.U' and '.V' files as required.
@section matroska
@@ -1314,8 +1118,7 @@ The recognized metadata settings in this muxer are:
@table @option
@item title
Set title name provided to a single track. This gets mapped to
the FileDescription element for a stream written as attachment.
Set title name provided to a single track.
@item language
Specify the language of the track in the Matroska languages form.
@@ -1382,31 +1185,11 @@ index at the beginning of the file.
If this option is set to a non-zero value, the muxer will reserve a given amount
of space in the file header and then try to write the cues there when the muxing
finishes. If the reserved space does not suffice, no Cues will be written, the
file will be finalized and writing the trailer will return an error.
A safe size for most use cases should be about 50kB per hour of video.
finishes. If the available space does not suffice, muxing will fail. A safe size
for most use cases should be about 50kB per hour of video.
Note that cues are only written if the output is seekable and this option will
have no effect if it is not.
@item default_mode
This option controls how the FlagDefault of the output tracks will be set.
It influences which tracks players should play by default. The default mode
is @samp{infer}.
@table @samp
@item infer
In this mode, for each type of track (audio, video or subtitle), if there is
a track with disposition default of this type, then the first such track
(i.e. the one with the lowest index) will be marked as default; if no such
track exists, the first track of this type will be marked as default instead
(if existing). This ensures that the default flag is set in a sensible way even
if the input originated from containers that lack the concept of default tracks.
@item infer_no_subs
This mode is the same as infer except that if no subtitle track with
disposition default exists, no subtitle track will be marked as default.
@item passthrough
In this mode the FlagDefault is set if and only if the AV_DISPOSITION_DEFAULT
flag is set in the disposition of the corresponding stream.
@end table
@end table
@anchor{md5}
@@ -1499,10 +1282,6 @@ more efficient), but with this option set, the muxer writes one moof/mdat
pair for each track, making it easier to separate tracks.
This option is implicitly set when writing ismv (Smooth Streaming) files.
@item -movflags skip_sidx
Skip writing of sidx atom. When bitrate overhead due to sidx atom is high,
this option could be used for cases where sidx atom is not mandatory.
When global_sidx flag is enabled, this option will be ignored.
@item -movflags faststart
Run a second pass moving the index (moov atom) to the beginning of the file.
This operation can take a while, and will not work in various situations such
@@ -1534,18 +1313,6 @@ be negative. This enables the initial sample to have DTS/CTS of zero, and
reduces the need for edit lists for some cases such as video tracks with
B-frames. Additionally, eases conformance with the DASH-IF interoperability
guidelines.
This option is implicitly set when writing ismv (Smooth Streaming) files.
@item -write_prft
Write producer time reference box (PRFT) with a specified time source for the
NTP field in the PRFT box. Set value as @samp{wallclock} to specify timesource
as wallclock time and @samp{pts} to specify timesource as input packets' PTS
values.
Setting value to @samp{pts} is applicable only for a live encoding use case,
where PTS values are set as as wallclock time at the source. For example, an
encoding use case with decklink capture source where @option{video_pts} and
@option{audio_pts} are set to @samp{abs_wallclock}.
@end table
@subsection Example
@@ -1556,6 +1323,13 @@ point on IIS with this muxer. Example:
ffmpeg -re @var{<normal input/transcoding options>} -movflags isml+frag_keyframe -f ismv http://server/publishingpoint.isml/Streams(Encoder1)
@end example
@subsection Audible AAX
Audible AAX files are encrypted M4B files, and they can be decrypted by specifying a 4 byte activation secret.
@example
ffmpeg -activation_bytes 1CEB00DA -i test.aax -vn -c:a copy output.mp4
@end example
@section mp3
The MP3 muxer writes a raw MP3 stream with the following optional features:
@@ -1644,7 +1418,7 @@ Set the program @samp{service_type}. Default is @code{digital_tv}.
Accepts the following options:
@table @samp
@item hex_value
Any hexadecimal value between @code{0x01} and @code{0xff} as defined in
Any hexdecimal value between @code{0x01} to @code{0xff} as defined in
ETSI 300 468.
@item digital_tv
Digital TV service.
@@ -1663,14 +1437,11 @@ Advanced Codec Digital HDTV service.
@end table
@item mpegts_pmt_start_pid @var{integer}
Set the first PID for PMTs. Default is @code{0x1000}, minimum is @code{0x0020},
maximum is @code{0x1ffa}. This option has no effect in m2ts mode where the PMT
PID is fixed @code{0x0100}.
Set the first PID for PMT. Default is @code{0x1000}. Max is @code{0x1f00}.
@item mpegts_start_pid @var{integer}
Set the first PID for elementary streams. Default is @code{0x0100}, minimum is
@code{0x0020}, maximum is @code{0x1ffa}. This option has no effect in m2ts mode
where the elementary stream PIDs are fixed.
Set the first PID for data packets. Default is @code{0x0100}. Max is
@code{0x0f00}.
@item mpegts_m2ts_mode @var{boolean}
Enable m2ts mode if set to @code{1}. Default value is @code{-1} which
@@ -1697,6 +1468,10 @@ Conform to System B (DVB) instead of System A (ATSC).
Mark the initial packet of each stream as discontinuity.
@end table
@item resend_headers @var{integer}
Reemit PAT/PMT before writing the next packet. This option is deprecated:
use @option{mpegts_flags} instead.
@item mpegts_copyts @var{boolean}
Preserve original timestamps, if value is set to @code{1}. Default value
is @code{-1}, which results in shifting timestamps so that they start from 0.
@@ -1705,16 +1480,14 @@ is @code{-1}, which results in shifting timestamps so that they start from 0.
Omit the PES packet length for video packets. Default is @code{1} (true).
@item pcr_period @var{integer}
Override the default PCR retransmission time in milliseconds. Default is
@code{-1} which means that the PCR interval will be determined automatically:
20 ms is used for CBR streams, the highest multiple of the frame duration which
is less than 100 ms is used for VBR streams.
Override the default PCR retransmission time in milliseconds. Ignored if
variable muxrate is selected. Default is @code{20}.
@item pat_period @var{duration}
Maximum time in seconds between PAT/PMT tables. Default is @code{0.1}.
@item pat_period @var{double}
Maximum time in seconds between PAT/PMT tables.
@item sdt_period @var{duration}
Maximum time in seconds between SDT tables. Default is @code{0.5}.
@item sdt_period @var{double}
Maximum time in seconds between SDT tables.
@item tables_version @var{integer}
Set PAT, PMT and SDT version (default @code{0}, valid values are from 0 to 31, inclusively).
@@ -1748,7 +1521,7 @@ ffmpeg -i file.mpg -c copy \
out.ts
@end example
@section mxf, mxf_d10, mxf_opatom
@section mxf, mxf_d10
MXF muxer.
@@ -1760,7 +1533,7 @@ The muxer options are:
@item store_user_comments @var{bool}
Set if user comments should be stored if available or never.
IRT D-10 does not allow user comments. The default is thus to write them for
mxf and mxf_opatom but not for mxf_d10
mxf but not for mxf_d10
@end table
@section null
@@ -2150,53 +1923,6 @@ Specify whether to remove all fragments when finished. Default 0 (do not remove)
@end table
@anchor{streamhash}
@section streamhash
Per stream hash testing format.
This muxer computes and prints a cryptographic hash of all the input frames,
on a per-stream basis. This can be used for equality checks without having
to do a complete binary comparison.
By default audio frames are converted to signed 16-bit raw audio and
video frames to raw video before computing the hash, but the output
of explicit conversions to other codecs can also be used. Timestamps
are ignored. It uses the SHA-256 cryptographic hash function by default,
but supports several other algorithms.
The output of the muxer consists of one line per stream of the form:
@var{streamindex},@var{streamtype},@var{algo}=@var{hash}, where
@var{streamindex} is the index of the mapped stream, @var{streamtype} is a
single character indicating the type of stream, @var{algo} is a short string
representing the hash function used, and @var{hash} is a hexadecimal number
representing the computed hash.
@table @option
@item hash @var{algorithm}
Use the cryptographic hash function specified by the string @var{algorithm}.
Supported values include @code{MD5}, @code{murmur3}, @code{RIPEMD128},
@code{RIPEMD160}, @code{RIPEMD256}, @code{RIPEMD320}, @code{SHA160},
@code{SHA224}, @code{SHA256} (default), @code{SHA512/224}, @code{SHA512/256},
@code{SHA384}, @code{SHA512}, @code{CRC32} and @code{adler32}.
@end table
@subsection Examples
To compute the SHA-256 hash of the input converted to raw audio and
video, and store it in the file @file{out.sha256}:
@example
ffmpeg -i INPUT -f streamhash out.sha256
@end example
To print an MD5 hash to stdout use the command:
@example
ffmpeg -i INPUT -f streamhash -hash md5 -
@end example
See also the @ref{hash} and @ref{framehash} muxers.
@anchor{fifo}
@section fifo
@@ -2295,35 +2021,20 @@ ffmpeg -re -i ... -c:v libx264 -c:a aac -f fifo -fifo_format flv -map 0:v -map 0
@anchor{tee}
@section tee
The tee muxer can be used to write the same data to several outputs, such as files or streams.
It can be used, for example, to stream a video over a network and save it to disk at the same time.
The tee muxer can be used to write the same data to several files or any
other kind of muxer. It can be used, for example, to both stream a video to
the network and save it to disk at the same time.
It is different from specifying several outputs to the @command{ffmpeg}
command-line tool. With the tee muxer, the audio and video data will be encoded only once.
With conventional multiple outputs, multiple encoding operations in parallel are initiated,
which can be a very expensive process. The tee muxer is not useful when using the libavformat API
directly because it is then possible to feed the same packets to several muxers directly.
Since the tee muxer does not represent any particular output format, ffmpeg cannot auto-select
output streams. So all streams intended for output must be specified using @code{-map}. See
the examples below.
Some encoders may need different options depending on the output format;
the auto-detection of this can not work with the tee muxer, so they need to be explicitly specified.
The main example is the @option{global_header} flag.
The slave outputs are specified in the file name given to the muxer,
separated by '|'. If any of the slave name contains the '|' separator,
leading or trailing spaces or any special character, those must be
escaped (see @ref{quoting_and_escaping,,the "Quoting and escaping"
section in the ffmpeg-utils(1) manual,ffmpeg-utils}).
@subsection Options
command-line tool because the audio and video data will be encoded only once
with the tee muxer; encoding can be a very expensive process. It is not
useful when using the libavformat API directly because it is then possible
to feed the same packets to several muxers directly.
@table @option
@item use_fifo @var{bool}
If set to 1, slave outputs will be processed in separate threads using the @ref{fifo}
If set to 1, slave outputs will be processed in separate thread using @ref{fifo}
muxer. This allows to compensate for different speed/latency/reliability of
outputs and setup transparent recovery. By default this feature is turned off.
@@ -2332,6 +2043,12 @@ Options to pass to fifo pseudo-muxer instances. See @ref{fifo}.
@end table
The slave outputs are specified in the file name given to the muxer,
separated by '|'. If any of the slave name contains the '|' separator,
leading or trailing spaces or any special character, it must be
escaped (see @ref{quoting_and_escaping,,the "Quoting and escaping"
section in the ffmpeg-utils(1) manual,ffmpeg-utils}).
Muxer options can be specified for each slave by prepending them as a list of
@var{key}=@var{value} pairs separated by ':', between square brackets. If
the options values contain a special character or the ':' separator, they
@@ -2340,27 +2057,13 @@ must be escaped; note that this is a second level escaping.
The following special options are also recognized:
@table @option
@item f
Specify the format name. Required if it cannot be guessed from the
output URL.
Specify the format name. Useful if it cannot be guessed from the
output name suffix.
@item bsfs[/@var{spec}]
Specify a list of bitstream filters to apply to the specified
output.
It is possible to specify to which streams a given bitstream filter
applies, by appending a stream specifier to the option separated by
@code{/}. @var{spec} must be a stream specifier (see @ref{Format
stream specifiers}).
If the stream specifier is not specified, the bitstream filters will be
applied to all streams in the output. This will cause that output operation
to fail if the output contains streams to which the bitstream filter cannot
be applied e.g. @code{h264_mp4toannexb} being applied to an output containing an audio stream.
Options for a bitstream filter must be specified in the form of @code{opt=value}.
Several bitstream filters can be specified, separated by ",".
@item use_fifo @var{bool}
This allows to override tee muxer use_fifo option for individual slave muxer.
@@ -2368,13 +2071,19 @@ This allows to override tee muxer use_fifo option for individual slave muxer.
This allows to override tee muxer fifo_options for individual slave muxer.
See @ref{fifo}.
It is possible to specify to which streams a given bitstream filter
applies, by appending a stream specifier to the option separated by
@code{/}. @var{spec} must be a stream specifier (see @ref{Format
stream specifiers}). If the stream specifier is not specified, the
bitstream filters will be applied to all streams in the output.
Several bitstream filters can be specified, separated by ",".
@item select
Select the streams that should be mapped to the slave output,
specified by a stream specifier. If not specified, this defaults to
all the mapped streams. This will cause that output operation to fail
if the output format does not accept all mapped streams.
You may use multiple stream specifiers separated by commas (@code{,}) e.g.: @code{a:0,v}
all the input streams. You may use multiple stream specifiers
separated by commas (@code{,}) e.g.: @code{a:0,v}
@item onfail
Specify behaviour on output failure. This can be set to either @code{abort} (which is
@@ -2388,7 +2097,7 @@ will continue without being affected.
@itemize
@item
Encode something and both archive it in a WebM file and stream it
as MPEG-TS over UDP:
as MPEG-TS over UDP (the streams need to be explicitly mapped):
@example
ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a
"archive-20121107.mkv|[f=mpegts]udp://10.0.1.255:1234/"
@@ -2411,19 +2120,23 @@ option is applied to @file{out.aac} in order to make it contain only
audio packets.
@example
ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac
-f tee "[bsfs/v=dump_extra=freq=keyframe]out.ts|[movflags=+faststart]out.mp4|[select=a]out.aac"
-f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=a]out.aac"
@end example
@item
As above, but select only stream @code{a:1} for the audio output. Note
As below, but select only stream @code{a:1} for the audio output. Note
that a second level escaping must be performed, as ":" is a special
character used to separate options.
@example
ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac
-f tee "[bsfs/v=dump_extra=freq=keyframe]out.ts|[movflags=+faststart]out.mp4|[select=\'a:1\']out.aac"
-f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=\'a:1\']out.aac"
@end example
@end itemize
Note: some codecs may need different options depending on the output format;
the auto-detection of this can not work with the tee muxer. The main example
is the @option{global_header} flag.
@section webm_dash_manifest
WebM DASH Manifest muxer.

View File

@@ -140,8 +140,7 @@ device with @command{-list_formats 1}. Audio sample rate is always 48 kHz.
@item list_devices
If set to @option{true}, print a list of devices and exit.
Defaults to @option{false}. This option is deprecated, please use the
@code{-sinks} option of ffmpeg to list the available output devices.
Defaults to @option{false}.
@item list_formats
If set to @option{true}, print a list of supported formats and exit.
@@ -151,14 +150,6 @@ Defaults to @option{false}.
Amount of time to preroll video in seconds.
Defaults to @option{0.5}.
@item duplex_mode
Sets the decklink device duplex mode. Must be @samp{unset}, @samp{half} or @samp{full}.
Defaults to @samp{unset}.
@item timing_offset
Sets the genlock timing pixel offset on the used output.
Defaults to @samp{unset}.
@end table
@subsection Examples
@@ -168,7 +159,7 @@ Defaults to @samp{unset}.
@item
List output devices:
@example
ffmpeg -sinks decklink
ffmpeg -i test.avi -f decklink -list_devices 1 dummy
@end example
@item
@@ -191,6 +182,51 @@ ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 -s 720x486 -r 24000/1001 'DeckLi
@end itemize
@section libndi_newtek
The libndi_newtek output device provides playback capabilities for using NDI (Network
Device Interface, standard created by NewTek).
Output filename is a NDI name.
To enable this output device, you need the NDI SDK and you
need to configure with the appropriate @code{--extra-cflags}
and @code{--extra-ldflags}.
NDI uses uyvy422 pixel format natively, but also supports bgra, bgr0, rgba and
rgb0.
@subsection Options
@table @option
@item reference_level
The audio reference level in dB. This specifies how many dB above the
reference level (+4dBU) is the full range of 16 bit audio.
Defaults to @option{0}.
@item clock_video
These specify whether video "clock" themselves.
Defaults to @option{false}.
@item clock_audio
These specify whether audio "clock" themselves.
Defaults to @option{false}.
@end table
@subsection Examples
@itemize
@item
Play video clip:
@example
ffmpeg -i "udp://@@239.1.1.1:10480?fifo_size=1000000&overrun_nonfatal=1" -vf "scale=720:576,fps=fps=25,setdar=dar=16/9,format=pix_fmts=uyvy422" -f libndi_newtek NEW_NDI1
@end example
@end itemize
@section fbdev
Linux framebuffer output device.
@@ -329,8 +365,6 @@ ffmpeg -i INPUT -f pulse "stream name"
SDL (Simple DirectMedia Layer) output device.
"sdl2" can be used as alias for "sdl".
This output device allows one to show a video stream in an SDL
window. Only one SDL window is allowed per application, so you can
have only one instance of this output device in an application.
@@ -359,18 +393,9 @@ 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_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
@@ -395,10 +420,6 @@ ffmpeg -i INPUT -c:v rawvideo -pix_fmt yuv420p -window_size qcif -f sdl "SDL out
sndio audio output device.
@section v4l2
Video4Linux2 output device.
@section xv
XV (XVideo) output device.

View File

@@ -148,11 +148,16 @@ To target 32 bits replace @code{x86_64} with @code{i686} in the command above.
@section Microsoft Visual C++ or Intel C++ Compiler for Windows
FFmpeg can be built with MSVC 2013 or later.
FFmpeg can be built with MSVC 2012 or earlier using a C99-to-C89 conversion utility
and wrapper, or with MSVC 2013 and ICL natively.
You will need the following prerequisites:
@itemize
@item @uref{https://github.com/libav/c99-to-c89/, C99-to-C89 Converter & Wrapper}
(if using MSVC 2012 or earlier)
@item @uref{http://code.google.com/p/msinttypes/, msinttypes}
(if using MSVC 2012 or earlier)
@item @uref{http://msys2.github.io/, MSYS2}
@item @uref{http://www.nasm.us/, NASM}
(Also available via MSYS2's package manager.)
@@ -161,13 +166,16 @@ You will need the following prerequisites:
To set up a proper environment in MSYS2, you need to run @code{msys_shell.bat} from
the Visual Studio or Intel Compiler command prompt.
Place @code{yasm.exe} somewhere in your @code{PATH}.
Place @code{yasm.exe} somewhere in your @code{PATH}. If using MSVC 2012 or
earlier, place @code{c99wrap.exe} and @code{c99conv.exe} somewhere in your
@code{PATH} as well.
Next, make sure any other headers and libs you want to use, such as zlib, are
located in a spot that the compiler can see. Do so by modifying the @code{LIB}
and @code{INCLUDE} environment variables to include the @strong{Windows-style}
paths to these directories. Alternatively, you can try to use the
@code{--extra-cflags}/@code{--extra-ldflags} configure options.
@code{--extra-cflags}/@code{--extra-ldflags} configure options. If using MSVC
2012 or earlier, place @code{inttypes.h} somewhere the compiler can see too.
Finally, run:
@@ -209,6 +217,8 @@ can see.
@item FFmpeg has been tested with the following on i686 and x86_64:
@itemize
@item Visual Studio 2010 Pro and Express
@item Visual Studio 2012 Pro and Express
@item Visual Studio 2013 Pro and Express
@item Intel Composer XE 2013
@item Intel Composer XE 2013 SP1

View File

@@ -51,66 +51,6 @@ in microseconds.
A description of the currently available protocols follows.
@section amqp
Advanced Message Queueing Protocol (AMQP) version 0-9-1 is a broker based
publish-subscribe communication protocol.
FFmpeg must be compiled with --enable-librabbitmq to support AMQP. A separate
AMQP broker must also be run. An example open-source AMQP broker is RabbitMQ.
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]
@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".
Muliple subscribers may stream from the broker using the command:
@example
ffplay amqp://[[user]:[password]@@]hostname[:port]
@end example
In RabbitMQ all data published to the broker flows through a specific exchange,
and each subscribing client has an assigned queue/buffer. When a packet arrives
at an exchange, it may be copied to a client's queue depending on the exchange
and routing_key fields.
The following options are supported:
@table @option
@item exchange
Sets the exchange to use on the broker. RabbitMQ has several predefined
exchanges: "amq.direct" is the default exchange, where the publisher and
subscriber must have a matching routing_key; "amq.fanout" is the same as a
broadcast operation (i.e. the data is forwarded to all queues on the fanout
exchange independent of the routing_key); and "amq.topic" is similar to
"amq.direct", but allows for more complex pattern matching (refer to the RabbitMQ
documentation).
@item routing_key
Sets the routing key. The default value is "amqp". The routing key is used on
the "amq.direct" and "amq.topic" exchanges to decide whether packets are written
to the queue of a subscriber.
@item pkt_size
Maximum size of each packet sent/received to the broker. Default is 131072.
Minimum is 4096 and max is any large value (representable by an int). When
receiving packets, this sets an internal buffer size in FFmpeg. It should be
equal to or greater than the size of the published packets to the broker. Otherwise
the received message may be truncated causing decoding errors.
@item connection_timeout
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.
@end table
@section async
Asynchronous data filling wrapper for input stream.
@@ -253,20 +193,6 @@ 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 for files on slow medium.
@item follow
If set to 1, the protocol will retry reading at the end of the file, allowing
reading files that still are being written. In order for this to terminate,
you either need to use the rw_timeout option, or use the interrupt callback
(for API users).
@item seekable
Controls if seekability is advertised on the file. 0 means non-seekable, -1
means auto (seekable for normal files, non-seekable for named pipes).
Many demuxers handle seekable and non-seekable resources differently,
overriding this might speed up opening certain files at the cost of losing some
features (e.g. accurate seeking).
@end table
@section ftp
@@ -288,14 +214,6 @@ Set timeout in microseconds of socket I/O operations used by the underlying low
operation. By default it is set to -1, which means that the timeout is
not specified.
@item ftp-user
Set a user to be used for authenticating to the FTP server. This is overridden by the
user in the FTP URL.
@item ftp-password
Set a password to be used for authenticating to the FTP server. This is overridden by
the password in the FTP URL, or by @option{ftp-anonymous-password} if no user is set.
@item ftp-anonymous-password
Password used when login as anonymous user. Typically an e-mail address
should be used.
@@ -311,6 +229,17 @@ it, unless special care is taken (tests, customized server configuration
etc.). Different FTP servers behave in different way during seek
operation. ff* tools may produce incomplete content due to server limitations.
This protocol accepts the following options:
@table @option
@item follow
If set to 1, the protocol will retry reading at the end of the file, allowing
reading files that still are being written. In order for this to terminate,
you either need to use the rw_timeout option, or use the interrupt callback
(for API users).
@end table
@section gopher
Gopher protocol.
@@ -461,11 +390,6 @@ ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://@var{server}:@var{p
wget --post-file=somefile.ogg http://@var{server}:@var{port}
@end example
@item send_expect_100
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.
@end table
@subsection HTTP Cookies
@@ -1255,7 +1179,7 @@ options.
This protocol accepts the following options.
@table @option
@item connect_timeout=@var{milliseconds}
@item connect_timeout
Connection timeout; SRT cannot connect for RTT > 1500 msec
(2 handshake exchanges) with the default connect timeout of
3 seconds. This option applies to the caller and rendezvous
@@ -1286,18 +1210,7 @@ IP Type of Service. Applies to sender only. Default value is 0xB8.
@item ipttl=@var{ttl}
IP Time To Live. Applies to sender only. Default value is 64.
@item latency=@var{microseconds}
Timestamp-based Packet Delivery Delay.
Used to absorb bursts of missed packet retransmissions.
This flag sets both @option{rcvlatency} and @option{peerlatency}
to the same value. Note that prior to version 1.3.0
this is the only flag to set the latency, however
this is effectively equivalent to setting @option{peerlatency},
when side is sender and @option{rcvlatency}
when side is receiver, and the bidirectional stream
sending is not supported.
@item listen_timeout=@var{microseconds}
@item listen_timeout
Set socket listen timeout.
@item maxbw=@var{bytes/seconds}
@@ -1342,45 +1255,6 @@ only if @option{pbkeylen} is non-zero. It is used on
the receiver only if the received data is encrypted.
The configured passphrase cannot be recovered (write-only).
@item enforced_encryption=@var{1|0}
If true, both connection parties must have the same password
set (including empty, that is, with no encryption). If the
password doesn't match or only one side is unencrypted,
the connection is rejected. Default is true.
@item kmrefreshrate=@var{packets}
The number of packets to be transmitted after which the
encryption key is switched to a new key. Default is -1.
-1 means auto (0x1000000 in srt library). The range for
this option is integers in the 0 - @code{INT_MAX}.
@item kmpreannounce=@var{packets}
The interval between when a new encryption key is sent and
when switchover occurs. This value also applies to the
subsequent interval between when switchover occurs and
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 payload_size=@var{bytes}
Sets the maximum declared size of a packet transferred
during the single call to the sending function in Live
mode. Use 0 if this value isn't used (which is default in
file mode).
Default is -1 (automatic), which typically means MPEG-TS;
if you are going to use SRT
to send any different kind of payload, such as, for example,
wrapping a live stream in very small frames, then you can
use a bigger maximum frame size, though not greater than
1456 bytes.
@item pkt_size=@var{bytes}
Alias for @samp{payload_size}.
@item peerlatency=@var{microseconds}
The latency value (as described in @option{rcvlatency}) that is
set by the sender side as a minimum value for the receiver.
@item pbkeylen=@var{bytes}
Sender encryption key length, in bytes.
Only can be set to 0, 16, 24 and 32.
@@ -1389,28 +1263,18 @@ Not required on receiver (set to 0),
key size obtained from sender in HaiCrypt handshake.
Default value is 0.
@item rcvlatency=@var{microseconds}
The time that should elapse since the moment when the
packet was sent and the moment when it's delivered to
the receiver application in the receiving function.
This time should be a buffer time large enough to cover
the time spent for sending, unexpectedly extended RTT
time, and the time needed to retransmit the lost UDP
packet. The effective latency value will be the maximum
of this options' value and the value of @option{peerlatency}
set by the peer side. Before version 1.3.0 this option
is only available as @option{latency}.
@item recv_buffer_size=@var{bytes}
Set UDP receive buffer size, expressed in bytes.
Set receive buffer size, expressed in bytes.
@item send_buffer_size=@var{bytes}
Set UDP send buffer size, expressed in bytes.
Set send buffer size, expressed in bytes.
@item timeout=@var{microseconds}
Set raise error timeouts for read, write and connect operations. Note that the
SRT library has internal timeouts which can be controlled separately, the
value set here is only a cap on those.
@item rw_timeout
Set raise error timeout for read/write optations.
This option is only relevant in read mode:
if no data arrived in more than this time
interval, raise error.
@item tlpktdrop=@var{1|0}
Too-late Packet Drop. When enabled on receiver, it skips
@@ -1423,92 +1287,9 @@ have no chance of being delivered in time. It was
automatically enabled in the sender if the receiver
supports it.
@item sndbuf=@var{bytes}
Set send buffer size, expressed in bytes.
@item rcvbuf=@var{bytes}
Set receive buffer size, expressed in bytes.
Receive buffer must not be greater than @option{ffs}.
@item lossmaxttl=@var{packets}
The value up to which the Reorder Tolerance may grow. When
Reorder Tolerance is > 0, then packet loss report is delayed
until that number of packets come in. Reorder Tolerance
increases every time a "belated" packet has come, but it
wasn't due to retransmission (that is, when UDP packets tend
to come out of order), with the difference between the latest
sequence and this packet's sequence, and not more than the
value of this option. By default it's 0, which means that this
mechanism is turned off, and the loss report is always sent
immediately upon experiencing a "gap" in sequences.
@item minversion
The minimum SRT version that is required from the peer. A connection
to a peer that does not satisfy the minimum version requirement
will be rejected.
The version format in hex is 0xXXYYZZ for x.y.z in human readable
form.
@item streamid=@var{string}
A string limited to 512 characters that can be set on the socket prior
to connecting. This stream ID will be able to be retrieved by the
listener side from the socket that is returned from srt_accept and
was connected by a socket with that set stream ID. SRT does not enforce
any special interpretation of the contents of this string.
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 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
type must be exactly the same on both connecting parties, otherwise
the connection is rejected.
@item messageapi=@var{1|0}
When set, this socket uses the Message API, otherwise it uses Buffer
API. Note that in live mode (see @option{transtype}) theres only
message API available. In File mode you can chose to use one of two modes:
Stream API (default, when this option is false). In this mode you may
send as many data as you wish with one sending instruction, or even use
dedicated functions that read directly from a file. The internal facility
will take care of any speed and congestion control. When receiving, you
can also receive as many data as desired, the data not extracted will be
waiting for the next call. There is no boundary between data portions in
the Stream mode.
Message API. In this mode your single sending instruction passes exactly
one piece of data that has boundaries (a message). Contrary to Live mode,
this message may span across multiple UDP packets and the only size
limitation is that it shall fit as a whole in the sending buffer. The
receiver shall use as large buffer as necessary to receive the message,
otherwise the message will not be given up. When the message is not
complete (not all packets received or there was a packet loss) it will
not be given up.
@item transtype=@var{live|file}
Sets the transmission type for the socket, in particular, setting this
option sets multiple other parameters to their default values as required
for a particular transmission type.
live: Set options as for live transmission. In this mode, you should
send by one sending instruction only so many data that fit in one UDP packet,
and limited to the value defined first in @option{payload_size} (1316 is
default in this mode). There is no speed control in this mode, only the
bandwidth control, if configured, in order to not exceed the bandwidth with
the overhead transmission (retransmitted and control packets).
file: Set options as for non-live transmission. See @option{messageapi}
for further explanations
@item linger=@var{seconds}
The number of seconds that the socket waits for unsent data when closing.
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 tsbpddelay
Timestamp-based Packet Delivery Delay.
Used to absorb burst of missed packet retransmission.
@end table
@@ -1616,9 +1397,6 @@ Set send buffer size, expressed bytes.
@item tcp_nodelay=@var{1|0}
Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
@item tcp_mss=@var{bytes}
Set maximum segment size for outgoing TCP packets, expressed in bytes.
@end table
The following example shows how to setup a listening TCP connection
@@ -1711,7 +1489,7 @@ The list of supported options follows.
@item buffer_size=@var{size}
Set the UDP maximum socket buffer size in bytes. This is used to set either
the receive or send buffer size, depending on what the socket is used for.
Default is 32 KB for output, 384 KB for input. See also @var{fifo_size}.
Default is 64KB. See also @var{fifo_size}.
@item bitrate=@var{bitrate}
If set to nonzero, the output will have the specified constant bitrate if the
@@ -1725,8 +1503,9 @@ packet bursts.
Override the local UDP port to bind with.
@item localaddr=@var{addr}
Local IP address of a network interface used for sending packets or joining
multicast groups.
Choose the local IP address. This is useful e.g. if sending multicast
and the host has multiple interfaces, where the user can choose
which interface to send on by specifying the IP address of that interface.
@item pkt_size=@var{size}
Set the size in bytes of UDP packets.
@@ -1749,12 +1528,12 @@ For receiving, this gives the benefit of only receiving packets from
the specified peer address/port.
@item sources=@var{address}[,@var{address}]
Only receive packets sent from the specified addresses. In case of multicast,
also subscribe to multicast traffic coming from these addresses only.
Only receive packets sent to the multicast group from one of the
specified sender IP addresses.
@item block=@var{address}[,@var{address}]
Ignore packets sent from the specified addresses. In case of multicast, also
exclude the source addresses in the multicast subscription.
Ignore packets sent to the multicast group from the specified
sender IP addresses.
@item fifo_size=@var{units}
Set the UDP receiving circular buffer size, expressed as a number of
@@ -1820,51 +1599,4 @@ Timeout in ms.
Create the Unix socket in listening mode.
@end table
@section zmq
ZeroMQ asynchronous messaging using the libzmq library.
This library supports unicast streaming to multiple clients without relying on
an external server.
The required syntax for streaming or connecting to a stream is:
@example
zmq:tcp://ip-address:port
@end example
Example:
Create a localhost stream on port 5555:
@example
ffmpeg -re -i input -f mpegts zmq:tcp://127.0.0.1:5555
@end example
Multiple clients may connect to the stream using:
@example
ffplay zmq:tcp://127.0.0.1:5555
@end example
Streaming to multiple clients is implemented using a ZeroMQ Pub-Sub pattern.
The server side binds to a port and publishes data. Clients connect to the
server (via IP address/port) and subscribe to the stream. The order in which
the server and client start generally does not matter.
ffmpeg must be compiled with the --enable-libzmq option to support
this protocol.
Options can be set on the @command{ffmpeg}/@command{ffplay} command
line. The following options are supported:
@table @option
@item pkt_size
Forces the maximum packet size for sending/receiving data. The default value is
131,072 bytes. On the server side, this sets the maximum size of sent packets
via ZeroMQ. On the clients, it sets an internal buffer size for receiving
packets. Note that pkt_size on the clients should be equal to or greater than
pkt_size on the server. Otherwise the received message may be truncated causing
decoding errors.
@end table
@c man end PROTOCOLS

View File

@@ -5,8 +5,7 @@
The video scaler supports the following named options.
Options may be set by specifying -@var{option} @var{value} in the
FFmpeg tools, with a few API-only exceptions noted below.
For programmatic use, they can be set explicitly in the
FFmpeg tools. For programmatic use, they can be set explicitly in the
@code{SwsContext} options or through the @file{libavutil/opt.h} API.
@table @option
@@ -48,8 +47,7 @@ Select Gaussian rescaling algorithm.
Select sinc rescaling algorithm.
@item lanczos
Select Lanczos rescaling algorithm. The default width (alpha) is 3 and can be
changed by setting @code{param0}.
Select Lanczos rescaling algorithm.
@item spline
Select natural bicubic spline rescaling algorithm.
@@ -70,31 +68,29 @@ Select full chroma input.
Enable bitexact output.
@end table
@item srcw @var{(API only)}
@item srcw
Set source width.
@item srch @var{(API only)}
@item srch
Set source height.
@item dstw @var{(API only)}
@item dstw
Set destination width.
@item dsth @var{(API only)}
@item dsth
Set destination height.
@item src_format @var{(API only)}
@item src_format
Set source pixel format (must be expressed as an integer).
@item dst_format @var{(API only)}
@item dst_format
Set destination pixel format (must be expressed as an integer).
@item src_range @var{(boolean)}
If value is set to @code{1}, indicates source is full range. Default value is
@code{0}, which indicates source is limited range.
@item src_range
Select source range.
@item dst_range @var{(boolean)}
If value is set to @code{1}, enable full range for destination. Default value
is @code{0}, which enables limited range.
@item dst_range
Select destination range.
@anchor{sws_params}
@item param0, param1

View File

@@ -172,7 +172,7 @@ spatial_decomposition_count
FIXME
colorspace_type
0 unspecified YCbCr
0 unspecified YcbCr
1 Gray
2 Gray + Alpha
3 GBR
@@ -235,7 +235,7 @@ spatial_decomposition_type
stored as delta from last, last is reset to 0 if always_reset || keyframe
qlog
quality (logarithmic quantizer scale)
quality (logarthmic quantizer scale)
stored as delta from last, last is reset to 0 if always_reset || keyframe
mv_scale
@@ -251,11 +251,11 @@ block_max_depth
stored as delta from last, last is reset to 0 if always_reset || keyframe
quant_table
quantization table
quantiztation table
Highlevel bitstream structure:
==============================
=============================
--------------------------------------------
| Header |
--------------------------------------------
@@ -303,7 +303,7 @@ Decoding process:
| Intra DC | |
| | LL0 subband prediction
------------ |
\ Dequantization
\ Dequantizaton
------------------- \ |
| Reference frames | \ IDWT
| ------- ------- | Motion \ |
@@ -390,8 +390,8 @@ motion vector prediction
(mvx_diff, mvy_diff)*mv_scale
Intra DC Prediction:
====================
Intra DC Predicton:
======================
the luma and chroma values of the left block are used as predictors
the used luma and chroma is the sum of the predictor and y_diff, cb_diff, cr_diff
@@ -407,7 +407,7 @@ Motion Compensation:
Halfpel interpolation:
----------------------
Halfpel interpolation is done by convolution with the halfpel filter stored
halfpel interpolation is done by convolution with the halfpel filter stored
in the header:
horizontal halfpel samples are found by
@@ -463,8 +463,8 @@ to the closest available fullpel sample
Smaller pel interpolation:
--------------------------
if diag_mc is set then points which lie on a line between 2 vertically,
horizontally or diagonally adjacent halfpel points shall be interpolated
linearly with rounding to nearest and halfway values rounded up.
horiziontally or diagonally adjacent halfpel points shall be interpolated
linearls with rounding to nearest and halfway values rounded up.
points which lie on 2 diagonals at the same time should only use the one
diagonal not containing the fullpel point
@@ -519,8 +519,8 @@ width,height here are the width and height of the LL0 subband not of the final
video
Dequantization:
===============
Dequantizaton:
==============
FIXME
Wavelet Transform:

View File

@@ -126,15 +126,6 @@ The following examples are all valid time duration:
@item 55
55 seconds
@item 0.2
0.2 seconds
@item 200ms
200 milliseconds, that's 0.2s
@item 200000us
200000 microseconds, that's 0.2s
@item 12:03:45
12 hours, 03 minutes and 45 seconds
@@ -713,8 +704,6 @@ FL+FR+FC+LFE+BL+BR+FLC+FRC
FL+FR+FC+LFE+FLC+FRC+SL+SR
@item octagonal
FL+FR+FC+BL+BR+BC+SL+SR
@item hexadecagonal
FL+FR+FC+BL+BR+BC+SL+SR+WL+WR+TBL+TBR+TBC+TFC+TFL+TFR
@item downmix
DL+DR
@end table
@@ -931,9 +920,6 @@ corresponding input value will be returned.
@item round(expr)
Round the value of expression @var{expr} to the nearest integer. For example, "round(1.5)" is "2.0".
@item sgn(x)
Compute sign of @var{x}.
@item sin(x)
Compute sine of @var{x}.

View File

@@ -389,7 +389,7 @@ distributor with something like this:
td.in = in;
td.out = out;
ctx->internal->execute(ctx, filter_slice, &td, NULL, FFMIN(outlink->h, ff_filter_get_nb_threads(ctx)));
ctx->internal->execute(ctx, filter_slice, &td, NULL, FFMIN(outlink->h, ctx->graph->nb_threads));
// ...
@@ -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.

View File

@@ -38,6 +38,7 @@ OBJCCFLAGS = $(CPPFLAGS) $(CFLAGS) $(OBJCFLAGS)
ASFLAGS := $(CPPFLAGS) $(ASFLAGS)
CXXFLAGS := $(CPPFLAGS) $(CFLAGS) $(CXXFLAGS)
X86ASMFLAGS += $(IFLAGS:%=%/) -I$(<D)/ -Pconfig.asm
NVCCFLAGS += -ptx
HOSTCCFLAGS = $(IFLAGS) $(HOSTCPPFLAGS) $(HOSTCFLAGS)
LDFLAGS := $(ALLFFLIBS:%=$(LD_PATH)lib%) $(LDFLAGS)
@@ -90,7 +91,7 @@ COMPILE_NVCC = $(call COMPILE,NVCC)
%.h.c:
$(Q)echo '#include "$*.h"' >$@
%.ptx: %.cu $(SRC_PATH)/compat/cuda/cuda_runtime.h
%.ptx: %.cu
$(COMPILE_NVCC)
%.ptx.c: %.ptx
@@ -160,9 +161,9 @@ $(SLIBOBJS): | $(sort $(dir $(SLIBOBJS)))
$(TESTOBJS): | $(sort $(dir $(TESTOBJS)))
$(TOOLOBJS): | tools
OUTDIRS := $(OUTDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(SLIBOBJS) $(TESTOBJS))
OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(SLIBOBJS) $(TESTOBJS))
CLEANSUFFIXES = *.d *.gcda *.gcno *.h.c *.ho *.map *.o *.pc *.ptx *.ptx.c *.ver *.version *$(DEFAULT_X86ASMD).asm *~ *.ilk *.pdb
CLEANSUFFIXES = *.d *.gcda *.gcno *.h.c *.ho *.map *.o *.pc *.ptx *.ptx.c *.ver *.version *$(DEFAULT_X86ASMD).asm *~
LIBSUFFIXES = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a
define RULES

View File

@@ -10,6 +10,7 @@ ALLAVPROGS = $(AVBASENAMES:%=%$(PROGSSUF)$(EXESUF))
ALLAVPROGS_G = $(AVBASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
OBJS-ffmpeg += fftools/ffmpeg_opt.o fftools/ffmpeg_filter.o fftools/ffmpeg_hw.o
OBJS-ffmpeg-$(CONFIG_CUVID) += fftools/ffmpeg_cuvid.o
OBJS-ffmpeg-$(CONFIG_LIBMFX) += fftools/ffmpeg_qsv.o
ifndef CONFIG_VIDEOTOOLBOX
OBJS-ffmpeg-$(CONFIG_VDA) += fftools/ffmpeg_videotoolbox.o
@@ -31,7 +32,7 @@ $(foreach P,$(AVPROGS-yes),$(eval $(call DOFFTOOL,$(P))))
all: $(AVPROGS)
fftools/ffprobe.o fftools/cmdutils.o: libavutil/ffversion.h | fftools
OUTDIRS += fftools
OBJDIRS += fftools
ifdef AVPROGS
install: install-progs install-data

View File

@@ -55,6 +55,9 @@
#include "libavutil/ffversion.h"
#include "libavutil/version.h"
#include "cmdutils.h"
#if CONFIG_NETWORK
#include "libavformat/network.h"
#endif
#if HAVE_SYS_RESOURCE_H
#include <sys/time.h>
#include <sys/resource.h>
@@ -116,7 +119,7 @@ static void log_callback_report(void *ptr, int level, const char *fmt, va_list v
void init_dynload(void)
{
#if HAVE_SETDLLDIRECTORY && defined(_WIN32)
#ifdef _WIN32
/* Calling SetDllDirectory with the empty string (but not NULL) removes the
* current working directory from the DLL search path as a security pre-caution. */
SetDllDirectory("");
@@ -179,7 +182,7 @@ void show_help_options(const OptionDef *options, const char *msg, int req_flags,
first = 1;
for (po = options; po->name; po++) {
char buf[128];
char buf[64];
if (((po->flags & req_flags) != req_flags) ||
(alt_flags && !(po->flags & alt_flags)) ||
@@ -845,8 +848,8 @@ do { \
}
if (octx->cur_group.nb_opts || codec_opts || format_opts || resample_opts)
av_log(NULL, AV_LOG_WARNING, "Trailing option(s) found in the "
"command: may be ignored.\n");
av_log(NULL, AV_LOG_WARNING, "Trailing options were found on the "
"commandline.\n");
av_log(NULL, AV_LOG_DEBUG, "Finished splitting the commandline.\n");
@@ -977,7 +980,6 @@ static int init_report(const char *env)
char *filename_template = NULL;
char *key, *val;
int ret, count = 0;
int prog_loglevel, envlevel = 0;
time_t now;
struct tm *tm;
AVBPrint filename;
@@ -1009,7 +1011,6 @@ static int init_report(const char *env)
av_log(NULL, AV_LOG_FATAL, "Invalid report file level\n");
exit_program(1);
}
envlevel = 1;
} else {
av_log(NULL, AV_LOG_ERROR, "Unknown key '%s' in FFREPORT\n", key);
}
@@ -1017,7 +1018,7 @@ static int init_report(const char *env)
av_free(key);
}
av_bprint_init(&filename, 0, AV_BPRINT_SIZE_AUTOMATIC);
av_bprint_init(&filename, 0, 1);
expand_filename_template(&filename,
av_x_if_null(filename_template, "%p-%t.log"), tm);
av_free(filename_template);
@@ -1026,10 +1027,6 @@ static int init_report(const char *env)
return AVERROR(ENOMEM);
}
prog_loglevel = av_log_get_level();
if (!envlevel)
report_file_level = FFMAX(report_file_level, prog_loglevel);
report_file = fopen(filename.str, "w");
if (!report_file) {
int ret = AVERROR(errno);
@@ -1040,17 +1037,16 @@ static int init_report(const char *env)
av_log_set_callback(log_callback_report);
av_log(NULL, AV_LOG_INFO,
"%s started on %04d-%02d-%02d at %02d:%02d:%02d\n"
"Report written to \"%s\"\n"
"Log level: %d\n",
"Report written to \"%s\"\n",
program_name,
tm->tm_year + 1900, tm->tm_mon + 1, tm->tm_mday,
tm->tm_hour, tm->tm_min, tm->tm_sec,
filename.str, report_file_level);
filename.str);
av_bprint_finalize(&filename, NULL);
return 0;
}
int opt_report(void *optctx, const char *opt, const char *arg)
int opt_report(const char *opt)
{
return init_report(NULL);
}
@@ -1418,12 +1414,6 @@ static void print_codec(const AVCodec *c)
AV_CODEC_CAP_SLICE_THREADS |
AV_CODEC_CAP_AUTO_THREADS))
printf("threads ");
if (c->capabilities & AV_CODEC_CAP_AVOID_PROBING)
printf("avoidprobe ");
if (c->capabilities & AV_CODEC_CAP_HARDWARE)
printf("hardware ");
if (c->capabilities & AV_CODEC_CAP_HYBRID)
printf("hybrid ");
if (!c->capabilities)
printf("none");
printf("\n");
@@ -1444,17 +1434,6 @@ static void print_codec(const AVCodec *c)
printf("\n");
}
if (avcodec_get_hw_config(c, 0)) {
printf(" Supported hardware devices: ");
for (int i = 0;; i++) {
const AVCodecHWConfig *config = avcodec_get_hw_config(c, i);
if (!config)
break;
printf("%s ", av_hwdevice_get_type_name(config->device_type));
}
printf("\n");
}
if (c->supported_framerates) {
const AVRational *fps = c->supported_framerates;
@@ -1493,14 +1472,13 @@ static char get_media_type_char(enum AVMediaType type)
}
}
static const AVCodec *next_codec_for_id(enum AVCodecID id, void **iter,
static const AVCodec *next_codec_for_id(enum AVCodecID id, const AVCodec *prev,
int encoder)
{
const AVCodec *c;
while ((c = av_codec_iterate(iter))) {
if (c->id == id &&
(encoder ? av_codec_is_encoder(c) : av_codec_is_decoder(c)))
return c;
while ((prev = av_codec_next(prev))) {
if (prev->id == id &&
(encoder ? av_codec_is_encoder(prev) : av_codec_is_decoder(prev)))
return prev;
}
return NULL;
}
@@ -1537,12 +1515,11 @@ static unsigned get_codecs_sorted(const AVCodecDescriptor ***rcodecs)
static void print_codecs_for_id(enum AVCodecID id, int encoder)
{
void *iter = NULL;
const AVCodec *codec;
const AVCodec *codec = NULL;
printf(" (%s: ", encoder ? "encoders" : "decoders");
while ((codec = next_codec_for_id(id, &iter, encoder)))
while ((codec = next_codec_for_id(id, codec, encoder)))
printf("%s ", codec->name);
printf(")");
@@ -1565,8 +1542,7 @@ int show_codecs(void *optctx, const char *opt, const char *arg)
" -------\n");
for (i = 0; i < nb_codecs; i++) {
const AVCodecDescriptor *desc = codecs[i];
const AVCodec *codec;
void *iter = NULL;
const AVCodec *codec = NULL;
if (strstr(desc->name, "_deprecated"))
continue;
@@ -1584,14 +1560,14 @@ int show_codecs(void *optctx, const char *opt, const char *arg)
/* print decoders/encoders when there's more than one or their
* names are different from codec name */
while ((codec = next_codec_for_id(desc->id, &iter, 0))) {
while ((codec = next_codec_for_id(desc->id, codec, 0))) {
if (strcmp(codec->name, desc->name)) {
print_codecs_for_id(desc->id, 0);
break;
}
}
iter = NULL;
while ((codec = next_codec_for_id(desc->id, &iter, 1))) {
codec = NULL;
while ((codec = next_codec_for_id(desc->id, codec, 1))) {
if (strcmp(codec->name, desc->name)) {
print_codecs_for_id(desc->id, 1);
break;
@@ -1622,10 +1598,9 @@ static void print_codecs(int encoder)
encoder ? "Encoders" : "Decoders");
for (i = 0; i < nb_codecs; i++) {
const AVCodecDescriptor *desc = codecs[i];
const AVCodec *codec;
void *iter = NULL;
const AVCodec *codec = NULL;
while ((codec = next_codec_for_id(desc->id, &iter, encoder))) {
while ((codec = next_codec_for_id(desc->id, codec, encoder))) {
printf(" %c", get_media_type_char(desc->type));
printf((codec->capabilities & AV_CODEC_CAP_FRAME_THREADS) ? "F" : ".");
printf((codec->capabilities & AV_CODEC_CAP_SLICE_THREADS) ? "S" : ".");
@@ -1830,10 +1805,9 @@ static void show_help_codec(const char *name, int encoder)
if (codec)
print_codec(codec);
else if ((desc = avcodec_descriptor_get_by_name(name))) {
void *iter = NULL;
int printed = 0;
while ((codec = next_codec_for_id(desc->id, &iter, encoder))) {
while ((codec = next_codec_for_id(desc->id, codec, encoder))) {
printed = 1;
print_codec(codec);
}
@@ -1868,24 +1842,6 @@ static void show_help_demuxer(const char *name)
show_help_children(fmt->priv_class, AV_OPT_FLAG_DECODING_PARAM);
}
static void show_help_protocol(const char *name)
{
const AVClass *proto_class;
if (!name) {
av_log(NULL, AV_LOG_ERROR, "No protocol name specified.\n");
return;
}
proto_class = avio_protocol_get_class(name);
if (!proto_class) {
av_log(NULL, AV_LOG_ERROR, "Unknown protocol '%s'.\n", name);
return;
}
show_help_children(proto_class, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_ENCODING_PARAM);
}
static void show_help_muxer(const char *name)
{
const AVCodecDescriptor *desc;
@@ -1979,10 +1935,7 @@ static void show_help_bsf(const char *name)
{
const AVBitStreamFilter *bsf = av_bsf_get_by_name(name);
if (!name) {
av_log(NULL, AV_LOG_ERROR, "No bitstream filter name specified.\n");
return;
} else if (!bsf) {
if (!bsf) {
av_log(NULL, AV_LOG_ERROR, "Unknown bit stream filter '%s'.\n", name);
return;
}
@@ -2016,8 +1969,6 @@ int show_help(void *optctx, const char *opt, const char *arg)
show_help_demuxer(par);
} else if (!strcmp(topic, "muxer")) {
show_help_muxer(par);
} else if (!strcmp(topic, "protocol")) {
show_help_protocol(par);
#if CONFIG_AVFILTER
} else if (!strcmp(topic, "filter")) {
show_help_filter(par);
@@ -2057,7 +2008,7 @@ FILE *get_preset_file(char *filename, size_t filename_size,
av_strlcpy(filename, preset_name, filename_size);
f = fopen(filename, "r");
} else {
#if HAVE_GETMODULEHANDLE && defined(_WIN32)
#ifdef _WIN32
char datadir[MAX_PATH], *ls;
base[2] = NULL;
@@ -2210,7 +2161,7 @@ double get_rotation(AVStream *st)
if (fabs(theta - 90*round(theta/90)) > 2)
av_log(NULL, AV_LOG_WARNING, "Odd rotation angle.\n"
"If you want to help, upload a sample "
"of this file to https://streams.videolan.org/upload/ "
"of this file to ftp://upload.ffmpeg.org/incoming/ "
"and contact the ffmpeg-devel mailing list. (ffmpeg-devel@ffmpeg.org)");
return theta;

View File

@@ -99,7 +99,7 @@ int opt_default(void *optctx, const char *opt, const char *arg);
*/
int opt_loglevel(void *optctx, const char *opt, const char *arg);
int opt_report(void *optctx, const char *opt, const char *arg);
int opt_report(const char *opt);
int opt_max_alloc(void *optctx, const char *opt, const char *arg);
@@ -236,7 +236,7 @@ void show_help_options(const OptionDef *options, const char *msg, int req_flags,
{ "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" }, \
{ "report", 0, { (void*)opt_report }, "generate a report" }, \
{ "max_alloc", HAS_ARG, { .func_arg = opt_max_alloc }, "set maximum size of a single allocated block", "bytes" }, \
{ "cpuflags", HAS_ARG | OPT_EXPERT, { .func_arg = opt_cpuflags }, "force specific cpu flags", "flags" }, \
{ "hide_banner", OPT_BOOL | OPT_EXPERT, {&hide_banner}, "do not show program banner", "hide_banner" }, \

File diff suppressed because it is too large Load Diff

View File

@@ -61,6 +61,7 @@ enum HWAccelID {
HWACCEL_GENERIC,
HWACCEL_VIDEOTOOLBOX,
HWACCEL_QSV,
HWACCEL_CUVID,
};
typedef struct HWAccel {
@@ -71,7 +72,7 @@ typedef struct HWAccel {
} HWAccel;
typedef struct HWDevice {
const char *name;
char *name;
enum AVHWDeviceType type;
AVBufferRef *device_ref;
} HWDevice;
@@ -348,7 +349,6 @@ typedef struct InputStream {
AVFifoBuffer *sub_queue; ///< queue of AVSubtitle* before filter init
AVFrame *frame;
int w, h;
unsigned int initialize; ///< marks if sub2video_update should force an initialization
} sub2video;
int dr1;
@@ -430,8 +430,7 @@ enum forced_keyframes_const {
FKF_NB
};
#define ABORT_ON_FLAG_EMPTY_OUTPUT (1 << 0)
#define ABORT_ON_FLAG_EMPTY_OUTPUT_STREAM (1 << 1)
#define ABORT_ON_FLAG_EMPTY_OUTPUT (1 << 0)
extern const char *const forced_keyframes_const_names[];
@@ -460,7 +459,8 @@ typedef struct OutputStream {
AVRational mux_timebase;
AVRational enc_timebase;
AVBSFContext *bsf_ctx;
int nb_bitstream_filters;
AVBSFContext **bsf_ctx;
AVCodecContext *enc_ctx;
AVCodecParameters *ref_par; /* associated input codec parameters with encoders options applied */
@@ -484,7 +484,6 @@ typedef struct OutputStream {
AVRational frame_aspect_ratio;
/* forced key frames */
int64_t forced_kf_ref_pts;
int64_t *forced_kf_pts;
int forced_kf_count;
int forced_kf_index;
@@ -615,6 +614,7 @@ extern const AVIOInterruptCB int_cb;
extern const OptionDef options[];
extern const HWAccel hwaccels[];
extern AVBufferRef *hw_device_ctx;
#if CONFIG_QSV
extern char *qsv_device;
#endif
@@ -645,7 +645,7 @@ int filtergraph_is_simple(FilterGraph *fg);
int init_simple_filtergraph(InputStream *ist, OutputStream *ost);
int init_complex_filtergraph(FilterGraph *fg);
void sub2video_update(InputStream *ist, int64_t heartbeat_pts, AVSubtitle *sub);
void sub2video_update(InputStream *ist, AVSubtitle *sub);
int ifilter_parameters_from_frame(InputFilter *ifilter, const AVFrame *frame);
@@ -653,6 +653,7 @@ int ffmpeg_parse_options(int argc, char **argv);
int videotoolbox_init(AVCodecContext *s);
int qsv_init(AVCodecContext *s);
int cuvid_init(AVCodecContext *s);
HWDevice *hw_device_get_by_name(const char *name);
int hw_device_init_from_string(const char *arg, HWDevice **dev);
@@ -660,7 +661,6 @@ void hw_device_free_all(void);
int hw_device_setup_for_decode(InputStream *ist);
int hw_device_setup_for_encode(OutputStream *ost);
int hw_device_setup_for_filter(FilterGraph *fg);
int hwaccel_decode_init(AVCodecContext *avctx);

73
fftools/ffmpeg_cuvid.c Normal file
View File

@@ -0,0 +1,73 @@
/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/hwcontext.h"
#include "libavutil/pixdesc.h"
#include "ffmpeg.h"
static void cuvid_uninit(AVCodecContext *avctx)
{
InputStream *ist = avctx->opaque;
av_buffer_unref(&ist->hw_frames_ctx);
}
int cuvid_init(AVCodecContext *avctx)
{
InputStream *ist = avctx->opaque;
AVHWFramesContext *frames_ctx;
int ret;
av_log(avctx, AV_LOG_VERBOSE, "Initializing cuvid hwaccel\n");
if (!hw_device_ctx) {
ret = av_hwdevice_ctx_create(&hw_device_ctx, AV_HWDEVICE_TYPE_CUDA,
ist->hwaccel_device, NULL, 0);
if (ret < 0) {
av_log(avctx, AV_LOG_ERROR, "Error creating a CUDA device\n");
return ret;
}
}
av_buffer_unref(&ist->hw_frames_ctx);
ist->hw_frames_ctx = av_hwframe_ctx_alloc(hw_device_ctx);
if (!ist->hw_frames_ctx) {
av_log(avctx, AV_LOG_ERROR, "Error creating a CUDA frames context\n");
return AVERROR(ENOMEM);
}
frames_ctx = (AVHWFramesContext*)ist->hw_frames_ctx->data;
frames_ctx->format = AV_PIX_FMT_CUDA;
frames_ctx->sw_format = avctx->sw_pix_fmt;
frames_ctx->width = avctx->width;
frames_ctx->height = avctx->height;
av_log(avctx, AV_LOG_DEBUG, "Initializing CUDA frames context: sw_format = %s, width = %d, height = %d\n",
av_get_pix_fmt_name(frames_ctx->sw_format), frames_ctx->width, frames_ctx->height);
ret = av_hwframe_ctx_init(ist->hw_frames_ctx);
if (ret < 0) {
av_log(avctx, AV_LOG_ERROR, "Error initializing a CUDA frame pool\n");
return ret;
}
ist->hwaccel_uninit = cuvid_uninit;
return 0;
}

View File

@@ -65,7 +65,6 @@ enum AVPixelFormat choose_pixel_fmt(AVStream *st, AVCodecContext *enc_ctx, AVCod
if (codec && codec->pix_fmts) {
const enum AVPixelFormat *p = codec->pix_fmts;
const AVPixFmtDescriptor *desc = av_pix_fmt_desc_get(target);
//FIXME: This should check for AV_PIX_FMT_FLAG_ALPHA after PAL8 pixel format without alpha is implemented
int has_alpha = desc ? desc->nb_components % 2 == 0 : 0;
enum AVPixelFormat best= AV_PIX_FMT_NONE;
@@ -99,8 +98,7 @@ void choose_sample_fmt(AVStream *st, AVCodec *codec)
break;
}
if (*p == -1) {
const AVCodecDescriptor *desc = avcodec_descriptor_get(codec->id);
if(desc && (desc->props & AV_CODEC_PROP_LOSSLESS) && av_get_sample_fmt_name(st->codecpar->format) > av_get_sample_fmt_name(codec->sample_fmts[0]))
if((codec->capabilities & AV_CODEC_CAP_LOSSLESS) && av_get_sample_fmt_name(st->codecpar->format) > av_get_sample_fmt_name(codec->sample_fmts[0]))
av_log(NULL, AV_LOG_ERROR, "Conversion will not be lossless.\n");
if(av_get_sample_fmt_name(st->codecpar->format))
av_log(NULL, AV_LOG_WARNING,
@@ -294,17 +292,10 @@ static void init_input_filter(FilterGraph *fg, AVFilterInOut *in)
exit_program(1);
}
ist = input_streams[input_files[file_idx]->ist_index + st->index];
if (ist->user_set_discard == AVDISCARD_ALL) {
av_log(NULL, AV_LOG_FATAL, "Stream specifier '%s' in filtergraph description %s "
"matches a disabled input stream.\n", p, fg->graph_desc);
exit_program(1);
}
} else {
/* find the first unused stream of corresponding type */
for (i = 0; i < nb_input_streams; i++) {
ist = input_streams[i];
if (ist->user_set_discard == AVDISCARD_ALL)
continue;
if (ist->dec_ctx->codec_type == type && ist->discard)
break;
}
@@ -740,13 +731,6 @@ static int sub2video_prepare(InputStream *ist, InputFilter *ifilter)
if (!ist->sub2video.frame)
return AVERROR(ENOMEM);
ist->sub2video.last_pts = INT64_MIN;
ist->sub2video.end_pts = INT64_MIN;
/* sub2video structure has been (re-)initialized.
Mark it as such so that the system will be
initialized with the first received heartbeat. */
ist->sub2video.initialize = 1;
return 0;
}
@@ -790,12 +774,13 @@ static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter,
sar = ifilter->sample_aspect_ratio;
if(!sar.den)
sar = (AVRational){0,1};
av_bprint_init(&args, 0, AV_BPRINT_SIZE_AUTOMATIC);
av_bprint_init(&args, 0, 1);
av_bprintf(&args,
"video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:"
"pixel_aspect=%d/%d",
"pixel_aspect=%d/%d:sws_param=flags=%d",
ifilter->width, ifilter->height, ifilter->format,
tb.num, tb.den, sar.num, sar.den);
tb.num, tb.den, sar.num, sar.den,
SWS_BILINEAR + ((ist->dec_ctx->flags&AV_CODEC_FLAG_BITEXACT) ? SWS_BITEXACT:0));
if (fr.num && fr.den)
av_bprintf(&args, ":frame_rate=%d/%d", fr.num, fr.den);
snprintf(name, sizeof(name), "graph %d input from stream %d:%d", fg->index,
@@ -1062,9 +1047,17 @@ int configure_filtergraph(FilterGraph *fg)
if ((ret = avfilter_graph_parse2(fg->graph, graph_desc, &inputs, &outputs)) < 0)
goto fail;
ret = hw_device_setup_for_filter(fg);
if (ret < 0)
goto fail;
if (filter_hw_device || hw_device_ctx) {
AVBufferRef *device = filter_hw_device ? filter_hw_device->device_ref
: hw_device_ctx;
for (i = 0; i < fg->graph->nb_filters; i++) {
fg->graph->filters[i]->hw_device_ctx = av_buffer_ref(device);
if (!fg->graph->filters[i]->hw_device_ctx) {
ret = AVERROR(ENOMEM);
goto fail;
}
}
}
if (simple && (!inputs || inputs->next || !outputs || outputs->next)) {
const char *num_inputs;
@@ -1167,7 +1160,7 @@ int configure_filtergraph(FilterGraph *fg)
while (av_fifo_size(ist->sub2video.sub_queue)) {
AVSubtitle tmp;
av_fifo_generic_read(ist->sub2video.sub_queue, &tmp, sizeof(tmp), NULL);
sub2video_update(ist, INT64_MIN, &tmp);
sub2video_update(ist, &tmp);
avsubtitle_free(&tmp);
}
}

View File

@@ -19,8 +19,6 @@
#include <string.h>
#include "libavutil/avstring.h"
#include "libavutil/pixdesc.h"
#include "libavfilter/buffersink.h"
#include "ffmpeg.h"
@@ -101,7 +99,7 @@ int hw_device_init_from_string(const char *arg, HWDevice **dev_out)
// -> av_hwdevice_ctx_create_derived()
AVDictionary *options = NULL;
const char *type_name = NULL, *name = NULL, *device = NULL;
char *type_name = NULL, *name = NULL, *device = NULL;
enum AVHWDeviceType type;
HWDevice *dev, *src;
AVBufferRef *device_ref = NULL;
@@ -157,12 +155,10 @@ int hw_device_init_from_string(const char *arg, HWDevice **dev_out)
++p;
q = strchr(p, ',');
if (q) {
if (q - p > 0) {
device = av_strndup(p, q - p);
if (!device) {
err = AVERROR(ENOMEM);
goto fail;
}
device = av_strndup(p, q - p);
if (!device) {
err = AVERROR(ENOMEM);
goto fail;
}
err = av_dict_parse_string(&options, q + 1, "=", ",", 0);
if (err < 0) {
@@ -172,8 +168,7 @@ int hw_device_init_from_string(const char *arg, HWDevice **dev_out)
}
err = av_hwdevice_ctx_create(&device_ref, type,
q ? device : p[0] ? p : NULL,
options, 0);
device ? device : p, options, 0);
if (err < 0)
goto fail;
@@ -418,57 +413,18 @@ int hw_device_setup_for_decode(InputStream *ist)
int hw_device_setup_for_encode(OutputStream *ost)
{
const AVCodecHWConfig *config;
HWDevice *dev = NULL;
AVBufferRef *frames_ref = NULL;
int i;
if (ost->filter) {
frames_ref = av_buffersink_get_hw_frames_ctx(ost->filter->filter);
if (frames_ref &&
((AVHWFramesContext*)frames_ref->data)->format ==
ost->enc_ctx->pix_fmt) {
// Matching format, will try to use hw_frames_ctx.
} else {
frames_ref = NULL;
}
}
for (i = 0;; i++) {
config = avcodec_get_hw_config(ost->enc, i);
if (!config)
break;
if (frames_ref &&
config->methods & AV_CODEC_HW_CONFIG_METHOD_HW_FRAMES_CTX &&
(config->pix_fmt == AV_PIX_FMT_NONE ||
config->pix_fmt == ost->enc_ctx->pix_fmt)) {
av_log(ost->enc_ctx, AV_LOG_VERBOSE, "Using input "
"frames context (format %s) with %s encoder.\n",
av_get_pix_fmt_name(ost->enc_ctx->pix_fmt),
ost->enc->name);
ost->enc_ctx->hw_frames_ctx = av_buffer_ref(frames_ref);
if (!ost->enc_ctx->hw_frames_ctx)
return AVERROR(ENOMEM);
return 0;
}
if (!dev &&
config->methods & AV_CODEC_HW_CONFIG_METHOD_HW_DEVICE_CTX)
dev = hw_device_get_by_type(config->device_type);
}
HWDevice *dev;
dev = hw_device_match_by_codec(ost->enc);
if (dev) {
av_log(ost->enc_ctx, AV_LOG_VERBOSE, "Using device %s "
"(type %s) with %s encoder.\n", dev->name,
av_hwdevice_get_type_name(dev->type), ost->enc->name);
ost->enc_ctx->hw_device_ctx = av_buffer_ref(dev->device_ref);
if (!ost->enc_ctx->hw_device_ctx)
return AVERROR(ENOMEM);
return 0;
} else {
// No device required, or no device available.
return 0;
}
return 0;
}
static int hwaccel_retrieve_data(AVCodecContext *avctx, AVFrame *input)
@@ -521,31 +477,3 @@ int hwaccel_decode_init(AVCodecContext *avctx)
return 0;
}
int hw_device_setup_for_filter(FilterGraph *fg)
{
HWDevice *dev;
int i;
// If the user has supplied exactly one hardware device then just
// give it straight to every filter for convenience. If more than
// one device is available then the user needs to pick one explcitly
// with the filter_hw_device option.
if (filter_hw_device)
dev = filter_hw_device;
else if (nb_hw_devices == 1)
dev = hw_devices[0];
else
dev = NULL;
if (dev) {
for (i = 0; i < fg->graph->nb_filters; i++) {
fg->graph->filters[i]->hw_device_ctx =
av_buffer_ref(dev->device_ref);
if (!fg->graph->filters[i]->hw_device_ctx)
return AVERROR(ENOMEM);
}
}
return 0;
}

View File

@@ -1,4 +1,3 @@
/*
* ffmpeg option parsing
*
@@ -44,80 +43,16 @@
#define DEFAULT_PASS_LOGFILENAME_PREFIX "ffmpeg2pass"
#define SPECIFIER_OPT_FMT_str "%s"
#define SPECIFIER_OPT_FMT_i "%i"
#define SPECIFIER_OPT_FMT_i64 "%"PRId64
#define SPECIFIER_OPT_FMT_ui64 "%"PRIu64
#define SPECIFIER_OPT_FMT_f "%f"
#define SPECIFIER_OPT_FMT_dbl "%lf"
static const char *opt_name_codec_names[] = {"c", "codec", "acodec", "vcodec", "scodec", "dcodec", NULL};
static const char *opt_name_audio_channels[] = {"ac", NULL};
static const char *opt_name_audio_sample_rate[] = {"ar", NULL};
static const char *opt_name_frame_rates[] = {"r", NULL};
static const char *opt_name_frame_sizes[] = {"s", NULL};
static const char *opt_name_frame_pix_fmts[] = {"pix_fmt", NULL};
static const char *opt_name_ts_scale[] = {"itsscale", NULL};
static const char *opt_name_hwaccels[] = {"hwaccel", NULL};
static const char *opt_name_hwaccel_devices[] = {"hwaccel_device", NULL};
static const char *opt_name_hwaccel_output_formats[] = {"hwaccel_output_format", NULL};
static const char *opt_name_autorotate[] = {"autorotate", NULL};
static const char *opt_name_max_frames[] = {"frames", "aframes", "vframes", "dframes", NULL};
static const char *opt_name_bitstream_filters[] = {"bsf", "absf", "vbsf", NULL};
static const char *opt_name_codec_tags[] = {"tag", "atag", "vtag", "stag", NULL};
static const char *opt_name_sample_fmts[] = {"sample_fmt", NULL};
static const char *opt_name_qscale[] = {"q", "qscale", NULL};
static const char *opt_name_forced_key_frames[] = {"forced_key_frames", NULL};
static const char *opt_name_force_fps[] = {"force_fps", NULL};
static const char *opt_name_frame_aspect_ratios[] = {"aspect", NULL};
static const char *opt_name_rc_overrides[] = {"rc_override", NULL};
static const char *opt_name_intra_matrices[] = {"intra_matrix", NULL};
static const char *opt_name_inter_matrices[] = {"inter_matrix", NULL};
static const char *opt_name_chroma_intra_matrices[] = {"chroma_intra_matrix", NULL};
static const char *opt_name_top_field_first[] = {"top", NULL};
static const char *opt_name_presets[] = {"pre", "apre", "vpre", "spre", NULL};
static const char *opt_name_copy_initial_nonkeyframes[] = {"copyinkfr", NULL};
static const char *opt_name_copy_prior_start[] = {"copypriorss", NULL};
static const char *opt_name_filters[] = {"filter", "af", "vf", NULL};
static const char *opt_name_filter_scripts[] = {"filter_script", NULL};
static const char *opt_name_reinit_filters[] = {"reinit_filter", NULL};
static const char *opt_name_fix_sub_duration[] = {"fix_sub_duration", NULL};
static const char *opt_name_canvas_sizes[] = {"canvas_size", NULL};
static const char *opt_name_pass[] = {"pass", NULL};
static const char *opt_name_passlogfiles[] = {"passlogfile", NULL};
static const char *opt_name_max_muxing_queue_size[] = {"max_muxing_queue_size", NULL};
static const char *opt_name_guess_layout_max[] = {"guess_layout_max", NULL};
static const char *opt_name_apad[] = {"apad", NULL};
static const char *opt_name_discard[] = {"discard", NULL};
static const char *opt_name_disposition[] = {"disposition", NULL};
static const char *opt_name_time_bases[] = {"time_base", NULL};
static const char *opt_name_enc_time_bases[] = {"enc_time_base", NULL};
#define WARN_MULTIPLE_OPT_USAGE(name, type, so, st)\
{\
char namestr[128] = "";\
const char *spec = so->specifier && so->specifier[0] ? so->specifier : "";\
for (i = 0; opt_name_##name[i]; i++)\
av_strlcatf(namestr, sizeof(namestr), "-%s%s", opt_name_##name[i], opt_name_##name[i+1] ? (opt_name_##name[i+2] ? ", " : " or ") : "");\
av_log(NULL, AV_LOG_WARNING, "Multiple %s options specified for stream %d, only the last option '-%s%s%s "SPECIFIER_OPT_FMT_##type"' will be used.\n",\
namestr, st->index, opt_name_##name[0], spec[0] ? ":" : "", spec, so->u.type);\
}
#define MATCH_PER_STREAM_OPT(name, type, outvar, fmtctx, st)\
{\
int i, ret, matches = 0;\
SpecifierOpt *so;\
int i, ret;\
for (i = 0; i < o->nb_ ## name; i++) {\
char *spec = o->name[i].specifier;\
if ((ret = check_stream_specifier(fmtctx, st, spec)) > 0) {\
if ((ret = check_stream_specifier(fmtctx, st, spec)) > 0)\
outvar = o->name[i].u.type;\
so = &o->name[i];\
matches++;\
} else if (ret < 0)\
else if (ret < 0)\
exit_program(1);\
}\
if (matches > 1)\
WARN_MULTIPLE_OPT_USAGE(name, type, so, st);\
}
#define MATCH_PER_TYPE_OPT(name, type, outvar, fmtctx, mediatype)\
@@ -136,9 +71,13 @@ const HWAccel hwaccels[] = {
#endif
#if CONFIG_LIBMFX
{ "qsv", qsv_init, HWACCEL_QSV, AV_PIX_FMT_QSV },
#endif
#if CONFIG_CUVID
{ "cuvid", cuvid_init, HWACCEL_CUVID, AV_PIX_FMT_CUDA },
#endif
{ 0 },
};
AVBufferRef *hw_device_ctx;
HWDevice *filter_hw_device;
char *vstats_filename;
@@ -232,11 +171,14 @@ static void init_options(OptionsContext *o)
static int show_hwaccels(void *optctx, const char *opt, const char *arg)
{
enum AVHWDeviceType type = AV_HWDEVICE_TYPE_NONE;
int i;
printf("Hardware acceleration methods:\n");
while ((type = av_hwdevice_iterate_types(type)) !=
AV_HWDEVICE_TYPE_NONE)
printf("%s\n", av_hwdevice_get_type_name(type));
for (i = 0; hwaccels[i].name; i++)
printf("%s\n", hwaccels[i].name);
printf("\n");
return 0;
}
@@ -262,9 +204,8 @@ static AVDictionary *strip_specifiers(AVDictionary *dict)
static int opt_abort_on(void *optctx, const char *opt, const char *arg)
{
static const AVOption opts[] = {
{ "abort_on" , NULL, 0, AV_OPT_TYPE_FLAGS, { .i64 = 0 }, INT64_MIN, INT64_MAX, .unit = "flags" },
{ "empty_output" , NULL, 0, AV_OPT_TYPE_CONST, { .i64 = ABORT_ON_FLAG_EMPTY_OUTPUT }, .unit = "flags" },
{ "empty_output_stream", NULL, 0, AV_OPT_TYPE_CONST, { .i64 = ABORT_ON_FLAG_EMPTY_OUTPUT_STREAM }, .unit = "flags" },
{ "abort_on" , NULL, 0, AV_OPT_TYPE_FLAGS, { .i64 = 0 }, INT64_MIN, INT64_MAX, .unit = "flags" },
{ "empty_output" , NULL, 0, AV_OPT_TYPE_CONST, { .i64 = ABORT_ON_FLAG_EMPTY_OUTPUT }, .unit = "flags" },
{ NULL },
};
static const AVClass class = {
@@ -327,7 +268,7 @@ static int opt_map(void *optctx, const char *opt, const char *arg)
{
OptionsContext *o = optctx;
StreamMap *m = NULL;
int i, negative = 0, file_idx, disabled = 0;
int i, negative = 0, file_idx;
int sync_file_idx = -1, sync_stream_idx = 0;
char *p, *sync;
char *map;
@@ -362,11 +303,6 @@ static int opt_map(void *optctx, const char *opt, const char *arg)
"match any streams.\n", arg);
exit_program(1);
}
if (input_streams[input_files[sync_file_idx]->ist_index + sync_stream_idx]->user_set_discard == AVDISCARD_ALL) {
av_log(NULL, AV_LOG_FATAL, "Sync stream specification in map %s matches a disabled input "
"stream.\n", arg);
exit_program(1);
}
}
@@ -403,10 +339,6 @@ static int opt_map(void *optctx, const char *opt, const char *arg)
if (check_stream_specifier(input_files[file_idx]->ctx, input_files[file_idx]->ctx->streams[i],
*p == ':' ? p + 1 : p) <= 0)
continue;
if (input_streams[input_files[file_idx]->ist_index + i]->user_set_discard == AVDISCARD_ALL) {
disabled = 1;
continue;
}
GROW_ARRAY(o->stream_maps, o->nb_stream_maps);
m = &o->stream_maps[o->nb_stream_maps - 1];
@@ -426,10 +358,6 @@ static int opt_map(void *optctx, const char *opt, const char *arg)
if (!m) {
if (allow_unused) {
av_log(NULL, AV_LOG_VERBOSE, "Stream map '%s' matches no streams; ignoring.\n", arg);
} else if (disabled) {
av_log(NULL, AV_LOG_FATAL, "Stream map '%s' matches disabled streams.\n"
"To ignore this, add a trailing '?' to the map.\n", arg);
exit_program(1);
} else {
av_log(NULL, AV_LOG_FATAL, "Stream map '%s' matches no streams.\n"
"To ignore this, add a trailing '?' to the map.\n", arg);
@@ -509,8 +437,7 @@ static int opt_map_channel(void *optctx, const char *opt, const char *arg)
/* allow trailing ? to map_channel */
if (allow_unused = strchr(mapchan, '?'))
*allow_unused = 0;
if (m->channel_idx < 0 || m->channel_idx >= st->codecpar->channels ||
input_streams[input_files[m->file_idx]->ist_index + m->stream_idx]->user_set_discard == AVDISCARD_ALL) {
if (m->channel_idx < 0 || m->channel_idx >= st->codecpar->channels) {
if (allow_unused) {
av_log(NULL, AV_LOG_VERBOSE, "mapchan: invalid audio channel #%d.%d.%d\n",
m->file_idx, m->stream_idx, m->channel_idx);
@@ -536,15 +463,21 @@ static int opt_sdp_file(void *optctx, const char *opt, const char *arg)
#if CONFIG_VAAPI
static int opt_vaapi_device(void *optctx, const char *opt, const char *arg)
{
HWDevice *dev;
const char *prefix = "vaapi:";
char *tmp;
int err;
tmp = av_asprintf("%s%s", prefix, arg);
if (!tmp)
return AVERROR(ENOMEM);
err = hw_device_init_from_string(tmp, NULL);
err = hw_device_init_from_string(tmp, &dev);
av_free(tmp);
return err;
if (err < 0)
return err;
hw_device_ctx = av_buffer_ref(dev->device_ref);
if (!hw_device_ctx)
return AVERROR(ENOMEM);
return 0;
}
#endif
@@ -813,13 +746,6 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic)
MATCH_PER_STREAM_OPT(discard, str, discard_str, ic, st);
ist->user_set_discard = AVDISCARD_NONE;
if ((o->video_disable && ist->st->codecpar->codec_type == AVMEDIA_TYPE_VIDEO) ||
(o->audio_disable && ist->st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) ||
(o->subtitle_disable && ist->st->codecpar->codec_type == AVMEDIA_TYPE_SUBTITLE) ||
(o->data_disable && ist->st->codecpar->codec_type == AVMEDIA_TYPE_DATA))
ist->user_set_discard = AVDISCARD_ALL;
if (discard_str && av_opt_eval_int(&cc, discard_opt, discard_str, &ist->user_set_discard) < 0) {
av_log(NULL, AV_LOG_ERROR, "Error parsing discard %s.\n",
discard_str);
@@ -872,28 +798,9 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic)
MATCH_PER_STREAM_OPT(top_field_first, i, ist->top_field_first, ic, st);
MATCH_PER_STREAM_OPT(hwaccels, str, hwaccel, ic, st);
MATCH_PER_STREAM_OPT(hwaccel_output_formats, str,
hwaccel_output_format, ic, st);
if (!hwaccel_output_format && hwaccel && !strcmp(hwaccel, "cuvid")) {
av_log(NULL, AV_LOG_WARNING,
"WARNING: defaulting hwaccel_output_format to cuda for compatibility "
"with old commandlines. This behaviour is DEPRECATED and will be removed "
"in the future. Please explicitly set \"-hwaccel_output_format cuda\".\n");
ist->hwaccel_output_format = AV_PIX_FMT_CUDA;
} else if (hwaccel_output_format) {
ist->hwaccel_output_format = av_get_pix_fmt(hwaccel_output_format);
if (ist->hwaccel_output_format == AV_PIX_FMT_NONE) {
av_log(NULL, AV_LOG_FATAL, "Unrecognised hwaccel output "
"format: %s", hwaccel_output_format);
}
} else {
ist->hwaccel_output_format = AV_PIX_FMT_NONE;
}
if (hwaccel) {
// The NVDEC hwaccels use a CUDA device, so remap the name here.
if (!strcmp(hwaccel, "nvdec") || !strcmp(hwaccel, "cuvid"))
if (!strcmp(hwaccel, "nvdec"))
hwaccel = "cuda";
if (!strcmp(hwaccel, "none"))
@@ -927,6 +834,8 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic)
AV_HWDEVICE_TYPE_NONE)
av_log(NULL, AV_LOG_FATAL, "%s ",
av_hwdevice_get_type_name(type));
for (i = 0; hwaccels[i].name; i++)
av_log(NULL, AV_LOG_FATAL, "%s ", hwaccels[i].name);
av_log(NULL, AV_LOG_FATAL, "\n");
exit_program(1);
}
@@ -940,6 +849,18 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic)
exit_program(1);
}
MATCH_PER_STREAM_OPT(hwaccel_output_formats, str,
hwaccel_output_format, ic, st);
if (hwaccel_output_format) {
ist->hwaccel_output_format = av_get_pix_fmt(hwaccel_output_format);
if (ist->hwaccel_output_format == AV_PIX_FMT_NONE) {
av_log(NULL, AV_LOG_FATAL, "Unrecognised hwaccel output "
"format: %s", hwaccel_output_format);
}
} else {
ist->hwaccel_output_format = AV_PIX_FMT_NONE;
}
ist->hwaccel_pix_fmt = AV_PIX_FMT_NONE;
break;
@@ -979,17 +900,16 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic)
static void assert_file_overwrite(const char *filename)
{
const char *proto_name = avio_find_protocol_name(filename);
if (file_overwrite && no_file_overwrite) {
fprintf(stderr, "Error, both -y and -n supplied. Exiting.\n");
exit_program(1);
}
if (!file_overwrite) {
const char *proto_name = avio_find_protocol_name(filename);
if (proto_name && !strcmp(proto_name, "file") && avio_check(filename, 0) == 0) {
if (stdin_interaction && !no_file_overwrite) {
fprintf(stderr,"File '%s' already exists. Overwrite? [y/N] ", filename);
fprintf(stderr,"File '%s' already exists. Overwrite ? [y/N] ", filename);
fflush(stderr);
term_exit();
signal(SIGINT, SIG_DFL);
@@ -1005,19 +925,6 @@ static void assert_file_overwrite(const char *filename)
}
}
}
if (proto_name && !strcmp(proto_name, "file")) {
for (int i = 0; i < nb_input_files; i++) {
InputFile *file = input_files[i];
if (file->ctx->iformat->flags & AVFMT_NOFILE)
continue;
if (!strcmp(filename, file->ctx->url)) {
av_log(NULL, AV_LOG_FATAL, "Output %s same as Input #%d - exiting\n", filename, i);
av_log(NULL, AV_LOG_WARNING, "FFmpeg cannot edit existing files in-place.\n");
exit_program(1);
}
}
}
}
static void dump_attachment(AVStream *st, const char *filename)
@@ -1196,22 +1103,9 @@ static int open_input_file(OptionsContext *o, const char *filename)
}
}
if (o->start_time != AV_NOPTS_VALUE && o->start_time_eof != AV_NOPTS_VALUE) {
av_log(NULL, AV_LOG_WARNING, "Cannot use -ss and -sseof both, using -ss for %s\n", filename);
o->start_time_eof = AV_NOPTS_VALUE;
}
if (o->start_time_eof != AV_NOPTS_VALUE) {
if (o->start_time_eof >= 0) {
av_log(NULL, AV_LOG_ERROR, "-sseof value must be negative; aborting\n");
exit_program(1);
}
if (ic->duration > 0) {
if (ic->duration>0) {
o->start_time = o->start_time_eof + ic->duration;
if (o->start_time < 0) {
av_log(NULL, AV_LOG_WARNING, "-sseof value seeks to before start of file %s; ignored\n", filename);
o->start_time = AV_NOPTS_VALUE;
}
} else
av_log(NULL, AV_LOG_WARNING, "Cannot use -sseof, duration of %s not known\n", filename);
}
@@ -1228,10 +1122,8 @@ static int open_input_file(OptionsContext *o, const char *filename)
int dts_heuristic = 0;
for (i=0; i<ic->nb_streams; i++) {
const AVCodecParameters *par = ic->streams[i]->codecpar;
if (par->video_delay) {
if (par->video_delay)
dts_heuristic = 1;
break;
}
}
if (dts_heuristic) {
seek_timestamp -= 3*AV_TIME_BASE / 23;
@@ -1432,7 +1324,6 @@ static OutputStream *new_output_stream(OptionsContext *o, AVFormatContext *oc, e
ost->file_index = nb_output_files - 1;
ost->index = idx;
ost->st = st;
ost->forced_kf_ref_pts = AV_NOPTS_VALUE;
st->codecpar->codec_type = type;
ret = choose_encoder(o, oc, ost);
@@ -1529,12 +1420,54 @@ static OutputStream *new_output_stream(OptionsContext *o, AVFormatContext *oc, e
MATCH_PER_STREAM_OPT(copy_prior_start, i, ost->copy_prior_start, oc ,st);
MATCH_PER_STREAM_OPT(bitstream_filters, str, bsfs, oc, st);
if (bsfs && *bsfs) {
ret = av_bsf_list_parse_str(bsfs, &ost->bsf_ctx);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error parsing bitstream filter sequence '%s': %s\n", bsfs, av_err2str(ret));
while (bsfs && *bsfs) {
const AVBitStreamFilter *filter;
char *bsf, *bsf_options_str, *bsf_name;
bsf = av_get_token(&bsfs, ",");
if (!bsf)
exit_program(1);
bsf_name = av_strtok(bsf, "=", &bsf_options_str);
if (!bsf_name)
exit_program(1);
filter = av_bsf_get_by_name(bsf_name);
if (!filter) {
av_log(NULL, AV_LOG_FATAL, "Unknown bitstream filter %s\n", bsf_name);
exit_program(1);
}
ost->bsf_ctx = av_realloc_array(ost->bsf_ctx,
ost->nb_bitstream_filters + 1,
sizeof(*ost->bsf_ctx));
if (!ost->bsf_ctx)
exit_program(1);
ret = av_bsf_alloc(filter, &ost->bsf_ctx[ost->nb_bitstream_filters]);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error allocating a bitstream filter context\n");
exit_program(1);
}
ost->nb_bitstream_filters++;
if (bsf_options_str && filter->priv_class) {
const AVOption *opt = av_opt_next(ost->bsf_ctx[ost->nb_bitstream_filters-1]->priv_data, NULL);
const char * shorthand[2] = {NULL};
if (opt)
shorthand[0] = opt->name;
ret = av_opt_set_from_string(ost->bsf_ctx[ost->nb_bitstream_filters-1]->priv_data, bsf_options_str, shorthand, "=", ":");
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error parsing options for bitstream filter %s\n", bsf_name);
exit_program(1);
}
}
av_freep(&bsf);
if (*bsfs)
bsfs++;
}
MATCH_PER_STREAM_OPT(codec_tags, str, codec_tag, oc, st);
@@ -2210,10 +2143,7 @@ static int open_output_file(OptionsContext *o, const char *filename)
for (i = 0; i < nb_input_streams; i++) {
int new_area;
ist = input_streams[i];
new_area = ist->st->codecpar->width * ist->st->codecpar->height + 100000000*!!ist->st->codec_info_nb_frames
+ 5000000*!!(ist->st->disposition & AV_DISPOSITION_DEFAULT);
if (ist->user_set_discard == AVDISCARD_ALL)
continue;
new_area = ist->st->codecpar->width * ist->st->codecpar->height + 100000000*!!ist->st->codec_info_nb_frames;
if((qcr!=MKTAG('A', 'P', 'I', 'C')) && (ist->st->disposition & AV_DISPOSITION_ATTACHED_PIC))
new_area = 1;
if (ist->st->codecpar->codec_type == AVMEDIA_TYPE_VIDEO &&
@@ -2234,10 +2164,7 @@ static int open_output_file(OptionsContext *o, const char *filename)
for (i = 0; i < nb_input_streams; i++) {
int score;
ist = input_streams[i];
score = ist->st->codecpar->channels + 100000000*!!ist->st->codec_info_nb_frames
+ 5000000*!!(ist->st->disposition & AV_DISPOSITION_DEFAULT);
if (ist->user_set_discard == AVDISCARD_ALL)
continue;
score = ist->st->codecpar->channels + 100000000*!!ist->st->codec_info_nb_frames;
if (ist->st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO &&
score > best_score) {
best_score = score;
@@ -2259,8 +2186,6 @@ static int open_output_file(OptionsContext *o, const char *filename)
AVCodec const *output_codec =
avcodec_find_encoder(oc->oformat->subtitle_codec);
int input_props = 0, output_props = 0;
if (input_streams[i]->user_set_discard == AVDISCARD_ALL)
continue;
if (output_codec)
output_descriptor = avcodec_descriptor_get(output_codec->id);
if (input_descriptor)
@@ -2282,8 +2207,6 @@ static int open_output_file(OptionsContext *o, const char *filename)
if (!o->data_disable ) {
enum AVCodecID codec_id = av_guess_codec(oc->oformat, NULL, filename, NULL, AVMEDIA_TYPE_DATA);
for (i = 0; codec_id != AV_CODEC_ID_NONE && i < nb_input_streams; i++) {
if (input_streams[i]->user_set_discard == AVDISCARD_ALL)
continue;
if (input_streams[i]->st->codecpar->codec_type == AVMEDIA_TYPE_DATA
&& input_streams[i]->st->codecpar->codec_id == codec_id )
new_data_stream(o, oc, i);
@@ -2322,11 +2245,6 @@ loop_end:
int src_idx = input_files[map->file_index]->ist_index + map->stream_index;
ist = input_streams[input_files[map->file_index]->ist_index + map->stream_index];
if (ist->user_set_discard == AVDISCARD_ALL) {
av_log(NULL, AV_LOG_FATAL, "Stream #%d:%d is disabled and cannot be mapped.\n",
map->file_index, map->stream_index);
exit_program(1);
}
if(o->subtitle_disable && ist->st->codecpar->codec_type == AVMEDIA_TYPE_SUBTITLE)
continue;
if(o-> audio_disable && ist->st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO)
@@ -2384,14 +2302,12 @@ loop_end:
o->attachments[i]);
exit_program(1);
}
if (len > INT_MAX - AV_INPUT_BUFFER_PADDING_SIZE ||
!(attachment = av_malloc(len + AV_INPUT_BUFFER_PADDING_SIZE))) {
av_log(NULL, AV_LOG_FATAL, "Attachment %s too large.\n",
if (!(attachment = av_malloc(len))) {
av_log(NULL, AV_LOG_FATAL, "Attachment %s too large to fit into memory.\n",
o->attachments[i]);
exit_program(1);
}
avio_read(pb, attachment, len);
memset(attachment + len, 0, AV_INPUT_BUFFER_PADDING_SIZE);
ost = new_attachment_stream(o, oc, -1);
ost->stream_copy = 0;
@@ -2783,14 +2699,13 @@ static int opt_target(void *optctx, const char *opt, const char *arg)
} else {
/* Try to determine PAL/NTSC by peeking in the input files */
if (nb_input_files) {
int i, j;
int i, j, fr;
for (j = 0; j < nb_input_files; j++) {
for (i = 0; i < input_files[j]->nb_streams; i++) {
AVStream *st = input_files[j]->ctx->streams[i];
int64_t fr;
if (st->codecpar->codec_type != AVMEDIA_TYPE_VIDEO)
continue;
fr = st->time_base.den * 1000LL / st->time_base.num;
fr = st->time_base.den * 1000 / st->time_base.num;
if (fr == 25000) {
norm = PAL;
break;
@@ -3020,11 +2935,8 @@ static int opt_preset(void *optctx, const char *opt, const char *arg)
static int opt_old2new(void *optctx, const char *opt, const char *arg)
{
OptionsContext *o = optctx;
int ret;
char *s = av_asprintf("%s:%c", opt + 1, *opt);
if (!s)
return AVERROR(ENOMEM);
ret = parse_option(o, s, arg, options);
int ret = parse_option(o, s, arg, options);
av_free(s);
return ret;
}
@@ -3055,8 +2967,6 @@ static int opt_qscale(void *optctx, const char *opt, const char *arg)
return parse_option(o, "q:v", arg, options);
}
s = av_asprintf("q%s", opt + 6);
if (!s)
return AVERROR(ENOMEM);
ret = parse_option(o, s, arg, options);
av_free(s);
return ret;
@@ -3101,11 +3011,8 @@ static int opt_vsync(void *optctx, const char *opt, const char *arg)
static int opt_timecode(void *optctx, const char *opt, const char *arg)
{
OptionsContext *o = optctx;
int ret;
char *tcr = av_asprintf("timecode=%s", arg);
if (!tcr)
return AVERROR(ENOMEM);
ret = parse_option(o, "metadata:g", tcr, options);
int ret = parse_option(o, "metadata:g", tcr, options);
if (ret >= 0)
ret = av_dict_set(&o->g->codec_opts, "gop_timecode", arg, 0);
av_free(tcr);
@@ -3207,7 +3114,7 @@ void show_help_default(const char *opt, const char *arg)
" -h -- print basic options\n"
" -h long -- print more options\n"
" -h full -- print all options (including all format and codec specific options, very long)\n"
" -h type=name -- print all options for the named decoder/encoder/demuxer/muxer/filter/bsf/protocol\n"
" -h type=name -- print all options for the named decoder/encoder/demuxer/muxer/filter/bsf\n"
" See man %s for detailed description of the options.\n"
"\n", program_name);
@@ -3215,7 +3122,7 @@ void show_help_default(const char *opt, const char *arg)
OPT_EXIT, 0, 0);
show_help_options(options, "Global options (affect whole program "
"instead of just one file):",
"instead of just one file:",
0, per_file | OPT_EXIT | OPT_EXPERT, 0);
if (show_advanced)
show_help_options(options, "Advanced global options:", OPT_EXPERT,
@@ -3250,9 +3157,7 @@ void show_help_default(const char *opt, const char *arg)
#if CONFIG_SWSCALE
show_help_children(sws_get_class(), flags);
#endif
#if CONFIG_SWRESAMPLE
show_help_children(swr_get_class(), AV_OPT_FLAG_AUDIO_PARAM);
#endif
show_help_children(avfilter_get_class(), AV_OPT_FLAG_VIDEO_PARAM | AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM);
show_help_children(av_bsf_get_class(), AV_OPT_FLAG_VIDEO_PARAM | AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_BSF_PARAM);
}
@@ -3291,7 +3196,6 @@ static int open_files(OptionGroupList *l, const char *inout,
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error parsing options for %s file "
"%s.\n", inout, g->arg);
uninit_options(&o);
return ret;
}
@@ -3433,7 +3337,7 @@ const OptionDef options[] = {
OPT_INPUT | OPT_OUTPUT, { .off = OFFSET(start_time) },
"set the start time offset", "time_off" },
{ "sseof", HAS_ARG | OPT_TIME | OPT_OFFSET |
OPT_INPUT, { .off = OFFSET(start_time_eof) },
OPT_INPUT | OPT_OUTPUT, { .off = OFFSET(start_time_eof) },
"set the start time offset relative to EOF", "time_off" },
{ "seek_timestamp", HAS_ARG | OPT_INT | OPT_OFFSET |
OPT_INPUT, { .off = OFFSET(seek_timestamp) },
@@ -3465,7 +3369,7 @@ const OptionDef options[] = {
{ "stdin", OPT_BOOL | OPT_EXPERT, { &stdin_interaction },
"enable or disable interaction on standard input" },
{ "timelimit", HAS_ARG | OPT_EXPERT, { .func_arg = opt_timelimit },
"set max runtime in seconds in CPU user time", "limit" },
"set max runtime in seconds", "limit" },
{ "dump", OPT_BOOL | OPT_EXPERT, { &do_pkt_dump },
"dump each input packet" },
{ "hex", OPT_BOOL | OPT_EXPERT, { &do_hex_dump },

View File

@@ -28,7 +28,6 @@
#include "ffmpeg.h"
static AVBufferRef *hw_device_ctx;
char *qsv_device = NULL;
static int qsv_get_buffer(AVCodecContext *s, AVFrame *frame, int flags)
@@ -94,7 +93,7 @@ int qsv_init(AVCodecContext *s)
frames_ctx->height = FFALIGN(s->coded_height, 32);
frames_ctx->format = AV_PIX_FMT_QSV;
frames_ctx->sw_format = s->sw_pix_fmt;
frames_ctx->initial_pool_size = 64 + s->extra_hw_frames;
frames_ctx->initial_pool_size = 64;
frames_hwctx->frame_type = MFX_MEMTYPE_VIDEO_MEMORY_DECODER_TARGET;
ret = av_hwframe_ctx_init(ist->hw_frames_ctx);

View File

@@ -51,12 +51,7 @@ static int videotoolbox_retrieve_data(AVCodecContext *s, AVFrame *frame)
case kCVPixelFormatType_422YpCbCr8: vt->tmp_frame->format = AV_PIX_FMT_UYVY422; break;
case kCVPixelFormatType_32BGRA: vt->tmp_frame->format = AV_PIX_FMT_BGRA; break;
#ifdef kCFCoreFoundationVersionNumber10_7
case kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange:
case kCVPixelFormatType_420YpCbCr8BiPlanarFullRange: vt->tmp_frame->format = AV_PIX_FMT_NV12; break;
#endif
#if HAVE_KCVPIXELFORMATTYPE_420YPCBCR10BIPLANARVIDEORANGE
case kCVPixelFormatType_420YpCbCr10BiPlanarVideoRange:
case kCVPixelFormatType_420YpCbCr10BiPlanarFullRange: vt->tmp_frame->format = AV_PIX_FMT_P010; break;
case kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange: vt->tmp_frame->format = AV_PIX_FMT_NV12; break;
#endif
default:
av_log(NULL, AV_LOG_ERROR,
@@ -67,7 +62,7 @@ static int videotoolbox_retrieve_data(AVCodecContext *s, AVFrame *frame)
vt->tmp_frame->width = frame->width;
vt->tmp_frame->height = frame->height;
ret = av_frame_get_buffer(vt->tmp_frame, 0);
ret = av_frame_get_buffer(vt->tmp_frame, 32);
if (ret < 0)
return ret;

View File

@@ -40,7 +40,6 @@
#include "libavutil/samplefmt.h"
#include "libavutil/avassert.h"
#include "libavutil/time.h"
#include "libavutil/bprint.h"
#include "libavformat/avformat.h"
#include "libavdevice/avdevice.h"
#include "libswscale/swscale.h"
@@ -315,19 +314,15 @@ static int default_width = 640;
static int default_height = 480;
static int screen_width = 0;
static int screen_height = 0;
static int screen_left = SDL_WINDOWPOS_CENTERED;
static int screen_top = SDL_WINDOWPOS_CENTERED;
static int audio_disable;
static int video_disable;
static int subtitle_disable;
static const char* wanted_stream_spec[AVMEDIA_TYPE_NB] = {0};
static int seek_by_bytes = -1;
static float seek_interval = 10;
static int display_disable;
static int borderless;
static int alwaysontop;
static int startup_volume = 100;
static int show_status = -1;
static int show_status = 1;
static int av_sync_type = AV_SYNC_AUDIO_MASTER;
static int64_t start_time = AV_NOPTS_VALUE;
static int64_t duration = AV_NOPTS_VALUE;
@@ -355,7 +350,6 @@ static char *afilters = NULL;
#endif
static int autorotate = 1;
static int find_stream_info = 1;
static int filter_nbthreads = 0;
/* current context */
static int is_full_screen;
@@ -645,10 +639,7 @@ static int decoder_decode_frame(Decoder *d, AVFrame *frame, AVSubtitle *sub) {
if (packet_queue_get(d->queue, &pkt, 1, &d->pkt_serial) < 0)
return -1;
}
if (d->queue->serial == d->pkt_serial)
break;
av_packet_unref(&pkt);
} while (1);
} while (d->queue->serial != d->pkt_serial);
if (pkt.data == flush_pkt.data) {
avcodec_flush_buffers(d->avctx);
@@ -867,27 +858,31 @@ static void calculate_display_rect(SDL_Rect *rect,
int scr_xleft, int scr_ytop, int scr_width, int scr_height,
int pic_width, int pic_height, AVRational pic_sar)
{
AVRational aspect_ratio = pic_sar;
int64_t width, height, x, y;
float aspect_ratio;
int width, height, x, y;
if (av_cmp_q(aspect_ratio, av_make_q(0, 1)) <= 0)
aspect_ratio = av_make_q(1, 1);
if (pic_sar.num == 0)
aspect_ratio = 0;
else
aspect_ratio = av_q2d(pic_sar);
aspect_ratio = av_mul_q(aspect_ratio, av_make_q(pic_width, pic_height));
if (aspect_ratio <= 0.0)
aspect_ratio = 1.0;
aspect_ratio *= (float)pic_width / (float)pic_height;
/* XXX: we suppose the screen has a 1.0 pixel ratio */
height = scr_height;
width = av_rescale(height, aspect_ratio.num, aspect_ratio.den) & ~1;
width = lrint(height * aspect_ratio) & ~1;
if (width > scr_width) {
width = scr_width;
height = av_rescale(width, aspect_ratio.den, aspect_ratio.num) & ~1;
height = lrint(width / aspect_ratio) & ~1;
}
x = (scr_width - width) / 2;
y = (scr_height - height) / 2;
rect->x = scr_xleft + x;
rect->y = scr_ytop + y;
rect->w = FFMAX((int)width, 1);
rect->h = FFMAX((int)height, 1);
rect->w = FFMAX(width, 1);
rect->h = FFMAX(height, 1);
}
static void get_sdl_pix_fmt_and_blendmode(int format, Uint32 *sdl_pix_fmt, SDL_BlendMode *sdl_blendmode)
@@ -959,22 +954,6 @@ static int upload_texture(SDL_Texture **tex, AVFrame *frame, struct SwsContext *
return ret;
}
static void set_sdl_yuv_conversion_mode(AVFrame *frame)
{
#if SDL_VERSION_ATLEAST(2,0,8)
SDL_YUV_CONVERSION_MODE mode = SDL_YUV_CONVERSION_AUTOMATIC;
if (frame && (frame->format == AV_PIX_FMT_YUV420P || frame->format == AV_PIX_FMT_YUYV422 || frame->format == AV_PIX_FMT_UYVY422)) {
if (frame->color_range == AVCOL_RANGE_JPEG)
mode = SDL_YUV_CONVERSION_JPEG;
else if (frame->colorspace == AVCOL_SPC_BT709)
mode = SDL_YUV_CONVERSION_BT709;
else if (frame->colorspace == AVCOL_SPC_BT470BG || frame->colorspace == AVCOL_SPC_SMPTE170M || frame->colorspace == AVCOL_SPC_SMPTE240M)
mode = SDL_YUV_CONVERSION_BT601;
}
SDL_SetYUVConversionMode(mode);
#endif
}
static void video_image_display(VideoState *is)
{
Frame *vp;
@@ -1036,9 +1015,7 @@ static void video_image_display(VideoState *is)
vp->flip_v = vp->frame->linesize[0] < 0;
}
set_sdl_yuv_conversion_mode(vp->frame);
SDL_RenderCopyEx(renderer, is->vid_texture, NULL, &rect, 0, NULL, vp->flip_v ? SDL_FLIP_VERTICAL : 0);
set_sdl_yuv_conversion_mode(NULL);
if (sp) {
#if USE_ONEPASS_SUBTITLE_RENDER
SDL_RenderCopy(renderer, is->sub_texture, NULL, &rect);
@@ -1151,8 +1128,6 @@ static void video_audio_display(VideoState *s)
if (realloc_texture(&s->vis_texture, SDL_PIXELFORMAT_ARGB8888, s->width, s->height, SDL_BLENDMODE_NONE, 1) < 0)
return;
if (s->xpos >= s->width)
s->xpos = 0;
nb_display_channels= FFMIN(nb_display_channels, 2);
if (rdft_bits != s->rdft_bits) {
av_rdft_end(s->rdft);
@@ -1202,6 +1177,8 @@ static void video_audio_display(VideoState *s)
}
if (!s->paused)
s->xpos++;
if (s->xpos >= s->width)
s->xpos= s->xleft;
}
}
@@ -1328,11 +1305,7 @@ static void sigterm_handler(int sig)
static void set_default_window_size(int width, int height, AVRational sar)
{
SDL_Rect rect;
int max_width = screen_width ? screen_width : INT_MAX;
int max_height = screen_height ? screen_height : INT_MAX;
if (max_width == INT_MAX && max_height == INT_MAX)
max_height = height;
calculate_display_rect(&rect, 0, 0, max_width, max_height, width, height, sar);
calculate_display_rect(&rect, 0, 0, INT_MAX, height, width, height, sar);
default_width = rect.w;
default_height = rect.h;
}
@@ -1341,15 +1314,20 @@ static int video_open(VideoState *is)
{
int w,h;
w = screen_width ? screen_width : default_width;
h = screen_height ? screen_height : default_height;
if (screen_width) {
w = screen_width;
h = screen_height;
} else {
w = default_width;
h = default_height;
}
if (!window_title)
window_title = input_filename;
SDL_SetWindowTitle(window, window_title);
SDL_SetWindowSize(window, w, h);
SDL_SetWindowPosition(window, screen_left, screen_top);
SDL_SetWindowPosition(window, SDL_WINDOWPOS_CENTERED, SDL_WINDOWPOS_CENTERED);
if (is_full_screen)
SDL_SetWindowFullscreen(window, SDL_WINDOW_FULLSCREEN_DESKTOP);
SDL_ShowWindow(window);
@@ -1693,7 +1671,6 @@ display:
}
is->force_refresh = 0;
if (show_status) {
AVBPrint buf;
static int64_t last_time;
int64_t cur_time;
int aqsize, vqsize, sqsize;
@@ -1717,28 +1694,18 @@ display:
av_diff = get_master_clock(is) - get_clock(&is->vidclk);
else if (is->audio_st)
av_diff = get_master_clock(is) - get_clock(&is->audclk);
av_bprint_init(&buf, 0, AV_BPRINT_SIZE_AUTOMATIC);
av_bprintf(&buf,
"%7.2f %s:%7.3f fd=%4d aq=%5dKB vq=%5dKB sq=%5dB f=%"PRId64"/%"PRId64" \r",
get_master_clock(is),
(is->audio_st && is->video_st) ? "A-V" : (is->video_st ? "M-V" : (is->audio_st ? "M-A" : " ")),
av_diff,
is->frame_drops_early + is->frame_drops_late,
aqsize / 1024,
vqsize / 1024,
sqsize,
is->video_st ? is->viddec.avctx->pts_correction_num_faulty_dts : 0,
is->video_st ? is->viddec.avctx->pts_correction_num_faulty_pts : 0);
if (show_status == 1 && AV_LOG_INFO > av_log_get_level())
fprintf(stderr, "%s", buf.str);
else
av_log(NULL, AV_LOG_INFO, "%s", buf.str);
fflush(stderr);
av_bprint_finalize(&buf, NULL);
av_log(NULL, AV_LOG_INFO,
"%7.2f %s:%7.3f fd=%4d aq=%5dKB vq=%5dKB sq=%5dB f=%"PRId64"/%"PRId64" \r",
get_master_clock(is),
(is->audio_st && is->video_st) ? "A-V" : (is->video_st ? "M-V" : (is->audio_st ? "M-A" : " ")),
av_diff,
is->frame_drops_early + is->frame_drops_late,
aqsize / 1024,
vqsize / 1024,
sqsize,
is->video_st ? is->viddec.avctx->pts_correction_num_faulty_dts : 0,
is->video_st ? is->viddec.avctx->pts_correction_num_faulty_pts : 0);
fflush(stdout);
last_time = cur_time;
}
}
@@ -1971,7 +1938,6 @@ static int configure_audio_filters(VideoState *is, const char *afilters, int for
avfilter_graph_free(&is->agraph);
if (!(is->agraph = avfilter_graph_alloc()))
return AVERROR(ENOMEM);
is->agraph->nb_threads = filter_nbthreads;
while ((e = av_dict_get(swr_opts, "", e, AV_DICT_IGNORE_SUFFIX)))
av_strlcatf(aresample_swr_opts, sizeof(aresample_swr_opts), "%s=%s:", e->key, e->value);
@@ -2121,10 +2087,10 @@ static int audio_thread(void *arg)
return ret;
}
static int decoder_start(Decoder *d, int (*fn)(void *), const char *thread_name, void* arg)
static int decoder_start(Decoder *d, int (*fn)(void *), void *arg)
{
packet_queue_start(d->queue);
d->decoder_tid = SDL_CreateThread(fn, thread_name, arg);
d->decoder_tid = SDL_CreateThread(fn, "decoder", arg);
if (!d->decoder_tid) {
av_log(NULL, AV_LOG_ERROR, "SDL_CreateThread(): %s\n", SDL_GetError());
return AVERROR(ENOMEM);
@@ -2143,17 +2109,26 @@ static int video_thread(void *arg)
AVRational frame_rate = av_guess_frame_rate(is->ic, is->video_st, NULL);
#if CONFIG_AVFILTER
AVFilterGraph *graph = NULL;
AVFilterGraph *graph = avfilter_graph_alloc();
AVFilterContext *filt_out = NULL, *filt_in = NULL;
int last_w = 0;
int last_h = 0;
enum AVPixelFormat last_format = -2;
int last_serial = -1;
int last_vfilter_idx = 0;
if (!graph) {
av_frame_free(&frame);
return AVERROR(ENOMEM);
}
#endif
if (!frame)
if (!frame) {
#if CONFIG_AVFILTER
avfilter_graph_free(&graph);
#endif
return AVERROR(ENOMEM);
}
for (;;) {
ret = get_video_frame(is, frame);
@@ -2176,11 +2151,6 @@ static int video_thread(void *arg)
(const char *)av_x_if_null(av_get_pix_fmt_name(frame->format), "none"), is->viddec.pkt_serial);
avfilter_graph_free(&graph);
graph = avfilter_graph_alloc();
if (!graph) {
ret = AVERROR(ENOMEM);
goto the_end;
}
graph->nb_threads = filter_nbthreads;
if ((ret = configure_video_filters(graph, is, vfilters_list ? vfilters_list[is->vfilter_idx] : NULL, frame)) < 0) {
SDL_Event event;
event.type = FF_QUIT_EVENT;
@@ -2223,8 +2193,6 @@ static int video_thread(void *arg)
ret = queue_picture(is, frame, pts, duration, frame->pkt_pos, is->viddec.pkt_serial);
av_frame_unref(frame);
#if CONFIG_AVFILTER
if (is->videoq.serial != is->viddec.pkt_serial)
break;
}
#endif
@@ -2610,7 +2578,7 @@ static int stream_component_open(VideoState *is, int stream_index)
if (forced_codec_name) av_log(NULL, AV_LOG_WARNING,
"No codec could be found with name '%s'\n", forced_codec_name);
else av_log(NULL, AV_LOG_WARNING,
"No decoder could be found for codec %s\n", avcodec_get_name(avctx->codec_id));
"No codec could be found with id %d\n", avctx->codec_id);
ret = AVERROR(EINVAL);
goto fail;
}
@@ -2690,7 +2658,7 @@ static int stream_component_open(VideoState *is, int stream_index)
is->auddec.start_pts = is->audio_st->start_time;
is->auddec.start_pts_tb = is->audio_st->time_base;
}
if ((ret = decoder_start(&is->auddec, audio_thread, "audio_decoder", is)) < 0)
if ((ret = decoder_start(&is->auddec, audio_thread, is)) < 0)
goto out;
SDL_PauseAudioDevice(audio_dev, 0);
break;
@@ -2699,7 +2667,7 @@ static int stream_component_open(VideoState *is, int stream_index)
is->video_st = ic->streams[stream_index];
decoder_init(&is->viddec, avctx, &is->videoq, is->continue_read_thread);
if ((ret = decoder_start(&is->viddec, video_thread, "video_decoder", is)) < 0)
if ((ret = decoder_start(&is->viddec, video_thread, is)) < 0)
goto out;
is->queue_attachments_req = 1;
break;
@@ -2708,7 +2676,7 @@ static int stream_component_open(VideoState *is, int stream_index)
is->subtitle_st = ic->streams[stream_index];
decoder_init(&is->subdec, avctx, &is->subtitleq, is->continue_read_thread);
if ((ret = decoder_start(&is->subdec, subtitle_thread, "subtitle_decoder", is)) < 0)
if ((ret = decoder_start(&is->subdec, subtitle_thread, is)) < 0)
goto out;
break;
default:
@@ -2775,6 +2743,9 @@ static int read_thread(void *arg)
}
memset(st_index, -1, sizeof(st_index));
is->last_video_stream = is->video_stream = -1;
is->last_audio_stream = is->audio_stream = -1;
is->last_subtitle_stream = is->subtitle_stream = -1;
is->eof = 0;
ic = avformat_alloc_context();
@@ -2986,7 +2957,7 @@ static int read_thread(void *arg)
}
if (is->queue_attachments_req) {
if (is->video_st && is->video_st->disposition & AV_DISPOSITION_ATTACHED_PIC) {
AVPacket copy;
AVPacket copy = { 0 };
if ((ret = av_packet_ref(&copy, &is->video_st->attached_pic)) < 0)
goto fail;
packet_queue_put(&is->videoq, &copy);
@@ -3080,9 +3051,6 @@ static VideoState *stream_open(const char *filename, AVInputFormat *iformat)
is = av_mallocz(sizeof(VideoState));
if (!is)
return NULL;
is->last_video_stream = is->video_stream = -1;
is->last_audio_stream = is->audio_stream = -1;
is->last_subtitle_stream = is->subtitle_stream = -1;
is->filename = av_strdup(filename);
if (!is->filename)
goto fail;
@@ -3283,14 +3251,15 @@ static void event_loop(VideoState *cur_stream)
refresh_loop_wait_event(cur_stream, &event);
switch (event.type) {
case SDL_KEYDOWN:
if (exit_on_keydown || event.key.keysym.sym == SDLK_ESCAPE || event.key.keysym.sym == SDLK_q) {
if (exit_on_keydown) {
do_exit(cur_stream);
break;
}
// If we don't yet have a window, skip all key events, because read_thread might still be initializing...
if (!cur_stream->width)
continue;
switch (event.key.keysym.sym) {
case SDLK_ESCAPE:
case SDLK_q:
do_exit(cur_stream);
break;
case SDLK_f:
toggle_full_screen(cur_stream);
cur_stream->force_refresh = 1;
@@ -3355,10 +3324,10 @@ static void event_loop(VideoState *cur_stream)
seek_chapter(cur_stream, -1);
break;
case SDLK_LEFT:
incr = seek_interval ? -seek_interval : -10.0;
incr = -10.0;
goto do_seek;
case SDLK_RIGHT:
incr = seek_interval ? seek_interval : 10.0;
incr = 10.0;
goto do_seek;
case SDLK_UP:
incr = 60.0;
@@ -3451,7 +3420,7 @@ static void event_loop(VideoState *cur_stream)
break;
case SDL_WINDOWEVENT:
switch (event.window.event) {
case SDL_WINDOWEVENT_SIZE_CHANGED:
case SDL_WINDOWEVENT_RESIZED:
screen_width = cur_stream->width = event.window.data1;
screen_height = cur_stream->height = event.window.data2;
if (cur_stream->vis_texture) {
@@ -3594,10 +3563,8 @@ static const OptionDef options[] = {
{ "ss", HAS_ARG, { .func_arg = opt_seek }, "seek to a given position in seconds", "pos" },
{ "t", HAS_ARG, { .func_arg = opt_duration }, "play \"duration\" seconds of audio/video", "duration" },
{ "bytes", OPT_INT | HAS_ARG, { &seek_by_bytes }, "seek by bytes 0=off 1=on -1=auto", "val" },
{ "seek_interval", OPT_FLOAT | HAS_ARG, { &seek_interval }, "set seek interval for left/right keys, in seconds", "seconds" },
{ "nodisp", OPT_BOOL, { &display_disable }, "disable graphical display" },
{ "noborder", OPT_BOOL, { &borderless }, "borderless window" },
{ "alwaysontop", OPT_BOOL, { &alwaysontop }, "window always on top" },
{ "volume", OPT_INT | HAS_ARG, { &startup_volume}, "set startup volume 0=min 100=max", "volume" },
{ "f", HAS_ARG, { .func_arg = opt_format }, "force format", "fmt" },
{ "pix_fmt", HAS_ARG | OPT_EXPERT | OPT_VIDEO, { .func_arg = opt_frame_pix_fmt }, "set pixel format", "format" },
@@ -3614,8 +3581,6 @@ static const OptionDef options[] = {
{ "framedrop", OPT_BOOL | OPT_EXPERT, { &framedrop }, "drop frames when cpu is too slow", "" },
{ "infbuf", OPT_BOOL | OPT_EXPERT, { &infinite_buffer }, "don't limit the input buffer size (useful with realtime streams)", "" },
{ "window_title", OPT_STRING | HAS_ARG, { &window_title }, "set window title", "window title" },
{ "left", OPT_INT | HAS_ARG | OPT_EXPERT, { &screen_left }, "set the x position for the left of the window", "x pos" },
{ "top", OPT_INT | HAS_ARG | OPT_EXPERT, { &screen_top }, "set the y position for the top of the window", "y pos" },
#if CONFIG_AVFILTER
{ "vf", OPT_EXPERT | HAS_ARG, { .func_arg = opt_add_vfilter }, "set video filters", "filter_graph" },
{ "af", OPT_STRING | HAS_ARG, { &afilters }, "set audio filters", "filter_graph" },
@@ -3631,7 +3596,6 @@ static const OptionDef options[] = {
{ "autorotate", OPT_BOOL, { &autorotate }, "automatically rotate video", "" },
{ "find_stream_info", OPT_BOOL | OPT_INPUT | OPT_EXPERT, { &find_stream_info },
"read and decode the streams to fill missing information with heuristics" },
{ "filter_threads", HAS_ARG | OPT_INT | OPT_EXPERT, { &filter_nbthreads }, "number of filter threads per graph" },
{ NULL, },
};
@@ -3669,7 +3633,7 @@ void show_help_default(const char *opt, const char *arg)
"c cycle program\n"
"w cycle video filters or show modes\n"
"s activate frame-step mode\n"
"left/right seek backward/forward 10 seconds or to custom interval if -seek_interval is set\n"
"left/right seek backward/forward 10 seconds\n"
"down/up seek backward/forward 1 minute\n"
"page down/page up seek backward/forward 10 minutes\n"
"right mouse click seek to percentage in file corresponding to fraction of width\n"
@@ -3739,12 +3703,6 @@ int main(int argc, char **argv)
if (!display_disable) {
int flags = SDL_WINDOW_HIDDEN;
if (alwaysontop)
#if SDL_VERSION_ATLEAST(2,0,5)
flags |= SDL_WINDOW_ALWAYS_ON_TOP;
#else
av_log(NULL, AV_LOG_WARNING, "Your SDL version doesn't support SDL_WINDOW_ALWAYS_ON_TOP. Feature will be inactive.\n");
#endif
if (borderless)
flags |= SDL_WINDOW_BORDERLESS;
else

View File

@@ -36,7 +36,6 @@
#include "libavutil/display.h"
#include "libavutil/hash.h"
#include "libavutil/mastering_display_metadata.h"
#include "libavutil/dovi_meta.h"
#include "libavutil/opt.h"
#include "libavutil/pixdesc.h"
#include "libavutil/spherical.h"
@@ -166,8 +165,6 @@ typedef enum {
SECTION_ID_FRAME_TAGS,
SECTION_ID_FRAME_SIDE_DATA_LIST,
SECTION_ID_FRAME_SIDE_DATA,
SECTION_ID_FRAME_SIDE_DATA_TIMECODE_LIST,
SECTION_ID_FRAME_SIDE_DATA_TIMECODE,
SECTION_ID_FRAME_LOG,
SECTION_ID_FRAME_LOGS,
SECTION_ID_LIBRARY_VERSION,
@@ -212,9 +209,7 @@ static struct section sections[] = {
[SECTION_ID_FRAME] = { SECTION_ID_FRAME, "frame", 0, { SECTION_ID_FRAME_TAGS, SECTION_ID_FRAME_SIDE_DATA_LIST, SECTION_ID_FRAME_LOGS, -1 } },
[SECTION_ID_FRAME_TAGS] = { SECTION_ID_FRAME_TAGS, "tags", SECTION_FLAG_HAS_VARIABLE_FIELDS, { -1 }, .element_name = "tag", .unique_name = "frame_tags" },
[SECTION_ID_FRAME_SIDE_DATA_LIST] ={ SECTION_ID_FRAME_SIDE_DATA_LIST, "side_data_list", SECTION_FLAG_IS_ARRAY, { SECTION_ID_FRAME_SIDE_DATA, -1 }, .element_name = "side_data", .unique_name = "frame_side_data_list" },
[SECTION_ID_FRAME_SIDE_DATA] = { SECTION_ID_FRAME_SIDE_DATA, "side_data", 0, { SECTION_ID_FRAME_SIDE_DATA_TIMECODE_LIST, -1 } },
[SECTION_ID_FRAME_SIDE_DATA_TIMECODE_LIST] = { SECTION_ID_FRAME_SIDE_DATA_TIMECODE_LIST, "timecodes", SECTION_FLAG_IS_ARRAY, { SECTION_ID_FRAME_SIDE_DATA_TIMECODE, -1 } },
[SECTION_ID_FRAME_SIDE_DATA_TIMECODE] = { SECTION_ID_FRAME_SIDE_DATA_TIMECODE, "timecode", 0, { -1 } },
[SECTION_ID_FRAME_SIDE_DATA] = { SECTION_ID_FRAME_SIDE_DATA, "side_data", 0, { -1 } },
[SECTION_ID_FRAME_LOGS] = { SECTION_ID_FRAME_LOGS, "logs", SECTION_FLAG_IS_ARRAY, { SECTION_ID_FRAME_LOG, -1 } },
[SECTION_ID_FRAME_LOG] = { SECTION_ID_FRAME_LOG, "log", 0, { -1 }, },
[SECTION_ID_LIBRARY_VERSIONS] = { SECTION_ID_LIBRARY_VERSIONS, "library_versions", SECTION_FLAG_IS_ARRAY, { SECTION_ID_LIBRARY_VERSION, -1 } },
@@ -255,7 +250,6 @@ static const OptionDef *options;
/* FFprobe context */
static const char *input_filename;
static const char *print_input_filename;
static AVInputFormat *iformat = NULL;
static struct AVHashContext *hash;
@@ -1085,12 +1079,12 @@ typedef struct CompactContext {
#define OFFSET(x) offsetof(CompactContext, x)
static const AVOption compact_options[]= {
{"item_sep", "set item separator", OFFSET(item_sep_str), AV_OPT_TYPE_STRING, {.str="|"}, 0, 0 },
{"s", "set item separator", OFFSET(item_sep_str), AV_OPT_TYPE_STRING, {.str="|"}, 0, 0 },
{"item_sep", "set item separator", OFFSET(item_sep_str), AV_OPT_TYPE_STRING, {.str="|"}, CHAR_MIN, CHAR_MAX },
{"s", "set item separator", OFFSET(item_sep_str), AV_OPT_TYPE_STRING, {.str="|"}, CHAR_MIN, CHAR_MAX },
{"nokey", "force no key printing", OFFSET(nokey), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1 },
{"nk", "force no key printing", OFFSET(nokey), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1 },
{"escape", "set escape mode", OFFSET(escape_mode_str), AV_OPT_TYPE_STRING, {.str="c"}, 0, 0 },
{"e", "set escape mode", OFFSET(escape_mode_str), AV_OPT_TYPE_STRING, {.str="c"}, 0, 0 },
{"escape", "set escape mode", OFFSET(escape_mode_str), AV_OPT_TYPE_STRING, {.str="c"}, CHAR_MIN, CHAR_MAX },
{"e", "set escape mode", OFFSET(escape_mode_str), AV_OPT_TYPE_STRING, {.str="c"}, CHAR_MIN, CHAR_MAX },
{"print_section", "print section name", OFFSET(print_section), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1 },
{"p", "print section name", OFFSET(print_section), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1 },
{NULL},
@@ -1201,12 +1195,12 @@ static const Writer compact_writer = {
#define OFFSET(x) offsetof(CompactContext, x)
static const AVOption csv_options[] = {
{"item_sep", "set item separator", OFFSET(item_sep_str), AV_OPT_TYPE_STRING, {.str=","}, 0, 0 },
{"s", "set item separator", OFFSET(item_sep_str), AV_OPT_TYPE_STRING, {.str=","}, 0, 0 },
{"item_sep", "set item separator", OFFSET(item_sep_str), AV_OPT_TYPE_STRING, {.str=","}, CHAR_MIN, CHAR_MAX },
{"s", "set item separator", OFFSET(item_sep_str), AV_OPT_TYPE_STRING, {.str=","}, CHAR_MIN, CHAR_MAX },
{"nokey", "force no key printing", OFFSET(nokey), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1 },
{"nk", "force no key printing", OFFSET(nokey), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1 },
{"escape", "set escape mode", OFFSET(escape_mode_str), AV_OPT_TYPE_STRING, {.str="csv"}, 0, 0 },
{"e", "set escape mode", OFFSET(escape_mode_str), AV_OPT_TYPE_STRING, {.str="csv"}, 0, 0 },
{"escape", "set escape mode", OFFSET(escape_mode_str), AV_OPT_TYPE_STRING, {.str="csv"}, CHAR_MIN, CHAR_MAX },
{"e", "set escape mode", OFFSET(escape_mode_str), AV_OPT_TYPE_STRING, {.str="csv"}, CHAR_MIN, CHAR_MAX },
{"print_section", "print section name", OFFSET(print_section), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1 },
{"p", "print section name", OFFSET(print_section), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1 },
{NULL},
@@ -1239,8 +1233,8 @@ typedef struct FlatContext {
#define OFFSET(x) offsetof(FlatContext, x)
static const AVOption flat_options[]= {
{"sep_char", "set separator", OFFSET(sep_str), AV_OPT_TYPE_STRING, {.str="."}, 0, 0 },
{"s", "set separator", OFFSET(sep_str), AV_OPT_TYPE_STRING, {.str="."}, 0, 0 },
{"sep_char", "set separator", OFFSET(sep_str), AV_OPT_TYPE_STRING, {.str="."}, CHAR_MIN, CHAR_MAX },
{"s", "set separator", OFFSET(sep_str), AV_OPT_TYPE_STRING, {.str="."}, CHAR_MIN, CHAR_MAX },
{"hierarchical", "specify if the section specification should be hierarchical", OFFSET(hierarchical), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1 },
{"h", "specify if the section specification should be hierarchical", OFFSET(hierarchical), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1 },
{NULL},
@@ -1537,7 +1531,7 @@ static void json_print_section_header(WriterContext *wctx)
if (parent_section && parent_section->id == SECTION_ID_PACKETS_AND_FRAMES) {
if (!json->compact)
JSON_INDENT();
printf("\"type\": \"%s\"", section->name);
printf("\"type\": \"%s\"%s", section->name, json->item_sep);
}
}
av_bprint_finalize(&buf, NULL);
@@ -1581,10 +1575,8 @@ static inline void json_print_item_str(WriterContext *wctx,
static void json_print_str(WriterContext *wctx, const char *key, const char *value)
{
JSONContext *json = wctx->priv;
const struct section *parent_section = wctx->level ?
wctx->section[wctx->level-1] : NULL;
if (wctx->nb_item[wctx->level] || (parent_section && parent_section->id == SECTION_ID_PACKETS_AND_FRAMES))
if (wctx->nb_item[wctx->level])
printf("%s", json->item_sep);
if (!json->compact)
JSON_INDENT();
@@ -1594,11 +1586,9 @@ static void json_print_str(WriterContext *wctx, const char *key, const char *val
static void json_print_int(WriterContext *wctx, const char *key, long long int value)
{
JSONContext *json = wctx->priv;
const struct section *parent_section = wctx->level ?
wctx->section[wctx->level-1] : NULL;
AVBPrint buf;
if (wctx->nb_item[wctx->level] || (parent_section && parent_section->id == SECTION_ID_PACKETS_AND_FRAMES))
if (wctx->nb_item[wctx->level])
printf("%s", json->item_sep);
if (!json->compact)
JSON_INDENT();
@@ -1929,16 +1919,6 @@ static void print_pkt_side_data(WriterContext *w,
AVContentLightMetadata *metadata = (AVContentLightMetadata *)sd->data;
print_int("max_content", metadata->MaxCLL);
print_int("max_average", metadata->MaxFALL);
} else if (sd->type == AV_PKT_DATA_DOVI_CONF) {
AVDOVIDecoderConfigurationRecord *dovi = (AVDOVIDecoderConfigurationRecord *)sd->data;
print_int("dv_version_major", dovi->dv_version_major);
print_int("dv_version_minor", dovi->dv_version_minor);
print_int("dv_profile", dovi->dv_profile);
print_int("dv_level", dovi->dv_level);
print_int("rpu_present_flag", dovi->rpu_present_flag);
print_int("el_present_flag", dovi->el_present_flag);
print_int("bl_present_flag", dovi->bl_present_flag);
print_int("dv_bl_signal_compatibility_id", dovi->dv_bl_signal_compatibility_id);
}
writer_print_section_footer(w);
}
@@ -2219,18 +2199,6 @@ static void show_frame(WriterContext *w, AVFrame *frame, AVStream *stream,
char tcbuf[AV_TIMECODE_STR_SIZE];
av_timecode_make_mpeg_tc_string(tcbuf, *(int64_t *)(sd->data));
print_str("timecode", tcbuf);
} else if (sd->type == AV_FRAME_DATA_S12M_TIMECODE && sd->size == 16) {
uint32_t *tc = (uint32_t*)sd->data;
int m = FFMIN(tc[0],3);
writer_print_section_header(w, SECTION_ID_FRAME_SIDE_DATA_TIMECODE_LIST);
for (int j = 1; j <= m ; j++) {
char tcbuf[AV_TIMECODE_STR_SIZE];
av_timecode_make_smpte_tc_string(tcbuf, tc[j], 0);
writer_print_section_header(w, SECTION_ID_FRAME_SIDE_DATA_TIMECODE);
print_str("value", tcbuf);
writer_print_section_footer(w);
}
writer_print_section_footer(w);
} else if (sd->type == AV_FRAME_DATA_MASTERING_DISPLAY_METADATA) {
AVMasteringDisplayMetadata *metadata = (AVMasteringDisplayMetadata *)sd->data;
@@ -2445,7 +2413,9 @@ static int read_interval_packets(WriterContext *w, InputFile *ifile,
}
av_packet_unref(&pkt);
}
av_packet_unref(&pkt);
av_init_packet(&pkt);
pkt.data = NULL;
pkt.size = 0;
//Flush remaining frames that are cached in the decoder
for (i = 0; i < fmt_ctx->nb_streams; i++) {
pkt.stream_index = i;
@@ -2547,7 +2517,6 @@ static int show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_id
if (dec_ctx) {
print_int("coded_width", dec_ctx->coded_width);
print_int("coded_height", dec_ctx->coded_height);
print_int("closed_captions", !!(dec_ctx->properties & FF_CODEC_PROPERTY_CLOSED_CAPTIONS));
}
#endif
print_int("has_b_frames", par->video_delay);
@@ -2677,20 +2646,20 @@ static int show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_id
} while (0)
if (do_show_stream_disposition) {
writer_print_section_header(w, in_program ? SECTION_ID_PROGRAM_STREAM_DISPOSITION : SECTION_ID_STREAM_DISPOSITION);
PRINT_DISPOSITION(DEFAULT, "default");
PRINT_DISPOSITION(DUB, "dub");
PRINT_DISPOSITION(ORIGINAL, "original");
PRINT_DISPOSITION(COMMENT, "comment");
PRINT_DISPOSITION(LYRICS, "lyrics");
PRINT_DISPOSITION(KARAOKE, "karaoke");
PRINT_DISPOSITION(FORCED, "forced");
PRINT_DISPOSITION(HEARING_IMPAIRED, "hearing_impaired");
PRINT_DISPOSITION(VISUAL_IMPAIRED, "visual_impaired");
PRINT_DISPOSITION(CLEAN_EFFECTS, "clean_effects");
PRINT_DISPOSITION(ATTACHED_PIC, "attached_pic");
PRINT_DISPOSITION(TIMED_THUMBNAILS, "timed_thumbnails");
writer_print_section_footer(w);
writer_print_section_header(w, in_program ? SECTION_ID_PROGRAM_STREAM_DISPOSITION : SECTION_ID_STREAM_DISPOSITION);
PRINT_DISPOSITION(DEFAULT, "default");
PRINT_DISPOSITION(DUB, "dub");
PRINT_DISPOSITION(ORIGINAL, "original");
PRINT_DISPOSITION(COMMENT, "comment");
PRINT_DISPOSITION(LYRICS, "lyrics");
PRINT_DISPOSITION(KARAOKE, "karaoke");
PRINT_DISPOSITION(FORCED, "forced");
PRINT_DISPOSITION(HEARING_IMPAIRED, "hearing_impaired");
PRINT_DISPOSITION(VISUAL_IMPAIRED, "visual_impaired");
PRINT_DISPOSITION(CLEAN_EFFECTS, "clean_effects");
PRINT_DISPOSITION(ATTACHED_PIC, "attached_pic");
PRINT_DISPOSITION(TIMED_THUMBNAILS, "timed_thumbnails");
writer_print_section_footer(w);
}
if (do_show_stream_tags)
@@ -2849,8 +2818,7 @@ static void show_error(WriterContext *w, int err)
writer_print_section_footer(w);
}
static int open_input_file(InputFile *ifile, const char *filename,
const char *print_filename)
static int open_input_file(InputFile *ifile, const char *filename)
{
int err, i;
AVFormatContext *fmt_ctx = NULL;
@@ -2872,10 +2840,6 @@ static int open_input_file(InputFile *ifile, const char *filename,
print_error(filename, err);
return err;
}
if (print_filename) {
av_freep(&fmt_ctx->url);
fmt_ctx->url = av_strdup(print_filename);
}
ifile->fmt_ctx = fmt_ctx;
if (scan_all_pmts_set)
av_dict_set(&format_opts, "scan_all_pmts", NULL, AV_DICT_MATCH_CASE);
@@ -2952,7 +2916,6 @@ static int open_input_file(InputFile *ifile, const char *filename,
ist->dec_ctx->pkt_timebase = stream->time_base;
ist->dec_ctx->framerate = stream->avg_frame_rate;
#if FF_API_LAVF_AVCTX
ist->dec_ctx->properties = stream->codec->properties;
ist->dec_ctx->coded_width = stream->codec->coded_width;
ist->dec_ctx->coded_height = stream->codec->coded_height;
#endif
@@ -2990,8 +2953,7 @@ static void close_input_file(InputFile *ifile)
avformat_close_input(&ifile->fmt_ctx);
}
static int probe_file(WriterContext *wctx, const char *filename,
const char *print_filename)
static int probe_file(WriterContext *wctx, const char *filename)
{
InputFile ifile = { 0 };
int ret, i;
@@ -3000,7 +2962,7 @@ static int probe_file(WriterContext *wctx, const char *filename,
do_read_frames = do_show_frames || do_count_frames;
do_read_packets = do_show_packets || do_count_packets;
ret = open_input_file(&ifile, filename, print_filename);
ret = open_input_file(&ifile, filename);
if (ret < 0)
goto end;
@@ -3306,12 +3268,6 @@ static int opt_input_file_i(void *optctx, const char *opt, const char *arg)
return 0;
}
static int opt_print_filename(void *optctx, const char *opt, const char *arg)
{
print_input_filename = arg;
return 0;
}
void show_help_default(const char *opt, const char *arg)
{
av_log_set_callback(log_callback_help);
@@ -3501,7 +3457,7 @@ static int opt_sections(void *optctx, const char *opt, const char *arg)
return 0;
}
static int opt_show_versions(void *optctx, const char *opt, const char *arg)
static int opt_show_versions(const char *opt, const char *arg)
{
mark_section_show_entries(SECTION_ID_PROGRAM_VERSION, 1, NULL);
mark_section_show_entries(SECTION_ID_LIBRARY_VERSION, 1, NULL);
@@ -3509,7 +3465,7 @@ static int opt_show_versions(void *optctx, const char *opt, const char *arg)
}
#define DEFINE_OPT_SHOW_SECTION(section, target_section_id) \
static int opt_show_##section(void *optctx, const char *opt, const char *arg) \
static int opt_show_##section(const char *opt, const char *arg) \
{ \
mark_section_show_entries(SECTION_ID_##target_section_id, 1, NULL); \
return 0; \
@@ -3537,40 +3493,39 @@ static const OptionDef real_options[] = {
"use sexagesimal format HOURS:MM:SS.MICROSECONDS for time units" },
{ "pretty", 0, {.func_arg = opt_pretty},
"prettify the format of displayed values, make it more human readable" },
{ "print_format", OPT_STRING | HAS_ARG, { &print_format },
{ "print_format", OPT_STRING | HAS_ARG, {(void*)&print_format},
"set the output printing format (available formats are: default, compact, csv, flat, ini, json, xml)", "format" },
{ "of", OPT_STRING | HAS_ARG, { &print_format }, "alias for -print_format", "format" },
{ "select_streams", OPT_STRING | HAS_ARG, { &stream_specifier }, "select the specified streams", "stream_specifier" },
{ "of", OPT_STRING | HAS_ARG, {(void*)&print_format}, "alias for -print_format", "format" },
{ "select_streams", OPT_STRING | HAS_ARG, {(void*)&stream_specifier}, "select the specified streams", "stream_specifier" },
{ "sections", OPT_EXIT, {.func_arg = opt_sections}, "print sections structure and section information, and exit" },
{ "show_data", OPT_BOOL, { &do_show_data }, "show packets data" },
{ "show_data_hash", OPT_STRING | HAS_ARG, { &show_data_hash }, "show packets data hash" },
{ "show_error", 0, { .func_arg = &opt_show_error }, "show probing error" },
{ "show_format", 0, { .func_arg = &opt_show_format }, "show format/container info" },
{ "show_frames", 0, { .func_arg = &opt_show_frames }, "show frames info" },
{ "show_data", OPT_BOOL, {(void*)&do_show_data}, "show packets data" },
{ "show_data_hash", OPT_STRING | HAS_ARG, {(void*)&show_data_hash}, "show packets data hash" },
{ "show_error", 0, {(void*)&opt_show_error}, "show probing error" },
{ "show_format", 0, {(void*)&opt_show_format}, "show format/container info" },
{ "show_frames", 0, {(void*)&opt_show_frames}, "show frames info" },
{ "show_format_entry", HAS_ARG, {.func_arg = opt_show_format_entry},
"show a particular entry from the format/container info", "entry" },
{ "show_entries", HAS_ARG, {.func_arg = opt_show_entries},
"show a set of specified entries", "entry_list" },
#if HAVE_THREADS
{ "show_log", OPT_INT|HAS_ARG, { &do_show_log }, "show log" },
{ "show_log", OPT_INT|HAS_ARG, {(void*)&do_show_log}, "show log" },
#endif
{ "show_packets", 0, { .func_arg = &opt_show_packets }, "show packets info" },
{ "show_programs", 0, { .func_arg = &opt_show_programs }, "show programs info" },
{ "show_streams", 0, { .func_arg = &opt_show_streams }, "show streams info" },
{ "show_chapters", 0, { .func_arg = &opt_show_chapters }, "show chapters info" },
{ "count_frames", OPT_BOOL, { &do_count_frames }, "count the number of frames per stream" },
{ "count_packets", OPT_BOOL, { &do_count_packets }, "count the number of packets per stream" },
{ "show_program_version", 0, { .func_arg = &opt_show_program_version }, "show ffprobe version" },
{ "show_library_versions", 0, { .func_arg = &opt_show_library_versions }, "show library versions" },
{ "show_versions", 0, { .func_arg = &opt_show_versions }, "show program and library versions" },
{ "show_pixel_formats", 0, { .func_arg = &opt_show_pixel_formats }, "show pixel format descriptions" },
{ "show_private_data", OPT_BOOL, { &show_private_data }, "show private data" },
{ "private", OPT_BOOL, { &show_private_data }, "same as show_private_data" },
{ "show_packets", 0, {(void*)&opt_show_packets}, "show packets info" },
{ "show_programs", 0, {(void*)&opt_show_programs}, "show programs info" },
{ "show_streams", 0, {(void*)&opt_show_streams}, "show streams info" },
{ "show_chapters", 0, {(void*)&opt_show_chapters}, "show chapters info" },
{ "count_frames", OPT_BOOL, {(void*)&do_count_frames}, "count the number of frames per stream" },
{ "count_packets", OPT_BOOL, {(void*)&do_count_packets}, "count the number of packets per stream" },
{ "show_program_version", 0, {(void*)&opt_show_program_version}, "show ffprobe version" },
{ "show_library_versions", 0, {(void*)&opt_show_library_versions}, "show library versions" },
{ "show_versions", 0, {(void*)&opt_show_versions}, "show program and library versions" },
{ "show_pixel_formats", 0, {(void*)&opt_show_pixel_formats}, "show pixel format descriptions" },
{ "show_private_data", OPT_BOOL, {(void*)&show_private_data}, "show private data" },
{ "private", OPT_BOOL, {(void*)&show_private_data}, "same as show_private_data" },
{ "bitexact", OPT_BOOL, {&do_bitexact}, "force bitexact output" },
{ "read_intervals", HAS_ARG, {.func_arg = opt_read_intervals}, "set read intervals", "read_intervals" },
{ "default", HAS_ARG | OPT_AUDIO | OPT_VIDEO | OPT_EXPERT, {.func_arg = opt_default}, "generic catch all option", "" },
{ "i", HAS_ARG, {.func_arg = opt_input_file_i}, "read specified file", "input_file"},
{ "print_filename", HAS_ARG, {.func_arg = opt_print_filename}, "override the printed input filename", "print_file"},
{ "find_stream_info", OPT_BOOL | OPT_INPUT | OPT_EXPERT, { &find_stream_info },
"read and decode the streams to fill missing information with heuristics" },
{ NULL, },
@@ -3719,7 +3674,7 @@ int main(int argc, char **argv)
av_log(NULL, AV_LOG_ERROR, "Use -h to get full help or, even better, run 'man %s'.\n", program_name);
ret = AVERROR(EINVAL);
} else if (input_filename) {
ret = probe_file(wctx, input_filename, print_input_filename);
ret = probe_file(wctx, input_filename);
if (ret < 0 && do_show_error)
show_error(wctx, ret);
}

View File

@@ -131,8 +131,8 @@ static int zero12v_decode_frame(AVCodecContext *avctx, void *data,
u = x/2 + (uint16_t *)(pic->data[1] + line * pic->linesize[1]);
v = x/2 + (uint16_t *)(pic->data[2] + line * pic->linesize[2]);
memcpy(y, y_temp, sizeof(*y) * (width - x));
memcpy(u, u_temp, sizeof(*u) * ((width - x + 1) / 2));
memcpy(v, v_temp, sizeof(*v) * ((width - x + 1) / 2));
memcpy(u, u_temp, sizeof(*u) * (width - x + 1) / 2);
memcpy(v, v_temp, sizeof(*v) * (width - x + 1) / 2);
}
line_end += stride;

View File

@@ -145,7 +145,7 @@ typedef struct FourXContext {
int mv[256];
VLC pre_vlc;
int last_dc;
DECLARE_ALIGNED(32, int16_t, block)[6][64];
DECLARE_ALIGNED(16, int16_t, block)[6][64];
void *bitstream_buffer;
unsigned int bitstream_buffer_size;
int version;
@@ -158,7 +158,7 @@ typedef struct FourXContext {
#define FIX_1_847759065 121095
#define FIX_2_613125930 171254
#define MULTIPLY(var, const) ((int)((var) * (unsigned)(const)) >> 16)
#define MULTIPLY(var, const) (((var) * (const)) >> 16)
static void idct(int16_t block[64])
{
@@ -351,8 +351,6 @@ static int decode_p_block(FourXContext *f, uint16_t *dst, const uint16_t *src,
index = size2index[log2h][log2w];
av_assert0(index >= 0);
if (get_bits_left(&f->gb) < 1)
return AVERROR_INVALIDDATA;
h = 1 << log2h;
code = get_vlc2(&f->gb, block_type_vlc[1 - (f->version > 1)][index].table,
BLOCK_TYPE_VLC_BITS, 1);
@@ -498,9 +496,9 @@ static int decode_i_block(FourXContext *f, int16_t *block)
{
int code, i, j, level, val;
if (get_bits_left(&f->pre_gb) < 2) {
av_log(f->avctx, AV_LOG_ERROR, "%d bits left before decode_i_block()\n", get_bits_left(&f->pre_gb));
return AVERROR_INVALIDDATA;
if (get_bits_left(&f->gb) < 2){
av_log(f->avctx, AV_LOG_ERROR, "%d bits left before decode_i_block()\n", get_bits_left(&f->gb));
return -1;
}
/* DC coef */
@@ -525,10 +523,6 @@ static int decode_i_block(FourXContext *f, int16_t *block)
break;
if (code == 0xf0) {
i += 16;
if (i >= 64) {
av_log(f->avctx, AV_LOG_ERROR, "run %d overflow\n", i);
return 0;
}
} else {
if (code & 0xf) {
level = get_xbits(&f->gb, code & 0xf);
@@ -703,7 +697,6 @@ static const uint8_t *read_huffman_tables(FourXContext *f,
len_tab[j] = len;
}
ff_free_vlc(&f->pre_vlc);
if (init_vlc(&f->pre_vlc, ACDC_VLC_BITS, 257, len_tab, 1, 1,
bits_tab, 4, 4, 0))
return NULL;
@@ -739,7 +732,7 @@ static int decode_i2_frame(FourXContext *f, const uint8_t *buf, int length)
for (x = 0; x < width; x += 16) {
unsigned int color[4] = { 0 }, bits;
if (buf_end - buf < 8)
return AVERROR_INVALIDDATA;
return -1;
// warning following is purely guessed ...
color[0] = bytestream2_get_le16u(&g3);
color[1] = bytestream2_get_le16u(&g3);

View File

@@ -70,9 +70,6 @@ static int decode_frame(AVCodecContext *avctx, void *data,
unsigned char *planemap = c->planemap;
int ret;
if (buf_size < planes * height *2)
return AVERROR_INVALIDDATA;
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;

View File

@@ -164,7 +164,8 @@ static av_cold int eightsvx_decode_init(AVCodecContext *avctx)
case AV_CODEC_ID_8SVX_FIB: esc->table = fibonacci; break;
case AV_CODEC_ID_8SVX_EXP: esc->table = exponential; break;
default:
av_assert1(0);
av_log(avctx, AV_LOG_ERROR, "Invalid codec id %d.\n", avctx->codec->id);
return AVERROR_INVALIDDATA;
}
avctx->sample_fmt = AV_SAMPLE_FMT_U8P;

View File

@@ -6,18 +6,12 @@ HEADERS = ac3_parser.h \
avcodec.h \
avdct.h \
avfft.h \
bsf.h \
codec.h \
codec_desc.h \
codec_id.h \
codec_par.h \
d3d11va.h \
dirac.h \
dv_profile.h \
dxva2.h \
jni.h \
mediacodec.h \
packet.h \
qsv.h \
vaapi.h \
vdpau.h \
@@ -50,7 +44,6 @@ OBJS = ac3_parser.o \
options.o \
mjpegenc_huffman.o \
parser.o \
parsers.o \
profiles.o \
qsv_api.o \
raw.o \
@@ -69,12 +62,9 @@ OBJS-$(CONFIG_BLOCKDSP) += blockdsp.o
OBJS-$(CONFIG_BSWAPDSP) += bswapdsp.o
OBJS-$(CONFIG_CABAC) += cabac.o
OBJS-$(CONFIG_CBS) += cbs.o
OBJS-$(CONFIG_CBS_AV1) += cbs_av1.o
OBJS-$(CONFIG_CBS_H264) += cbs_h2645.o h2645_parse.o
OBJS-$(CONFIG_CBS_H265) += cbs_h2645.o h2645_parse.o
OBJS-$(CONFIG_CBS_JPEG) += cbs_jpeg.o
OBJS-$(CONFIG_CBS_MPEG2) += cbs_mpeg2.o
OBJS-$(CONFIG_CBS_VP9) += cbs_vp9.o
OBJS-$(CONFIG_CRYSTALHD) += crystalhd.o
OBJS-$(CONFIG_DCT) += dct.o dct32_fixed.o dct32_float.o
OBJS-$(CONFIG_ERROR_RESILIENCE) += error_resilience.o
@@ -173,16 +163,12 @@ OBJS-$(CONFIG_AAC_ENCODER) += aacenc.o aaccoder.o aacenctab.o \
aacenc_ltp.o \
aacenc_pred.o \
psymodel.o mpeg4audio.o kbdwin.o cbrt_data.o
OBJS-$(CONFIG_AAC_MF_ENCODER) += mfenc.o mf_utils.o
OBJS-$(CONFIG_AASC_DECODER) += aasc.o msrledec.o
OBJS-$(CONFIG_AC3_DECODER) += ac3dec_float.o ac3dec_data.o ac3.o kbdwin.o ac3tab.o
OBJS-$(CONFIG_AC3_FIXED_DECODER) += ac3dec_fixed.o ac3dec_data.o ac3.o kbdwin.o ac3tab.o
OBJS-$(CONFIG_AC3_ENCODER) += ac3enc_float.o ac3enc.o ac3tab.o \
ac3.o kbdwin.o
OBJS-$(CONFIG_AC3_FIXED_ENCODER) += ac3enc_fixed.o ac3enc.o ac3tab.o ac3.o
OBJS-$(CONFIG_AC3_MF_ENCODER) += mfenc.o mf_utils.o
OBJS-$(CONFIG_ACELP_KELVIN_DECODER) += g729dec.o lsp.o celp_math.o celp_filters.o acelp_filters.o acelp_pitch_delay.o acelp_vectors.o g729postfilter.o
OBJS-$(CONFIG_AGM_DECODER) += agm.o
OBJS-$(CONFIG_AIC_DECODER) += aic.o
OBJS-$(CONFIG_ALAC_DECODER) += alac.o alac_data.o alacdsp.o
OBJS-$(CONFIG_ALAC_ENCODER) += alacenc.o alac_data.o
@@ -202,13 +188,12 @@ OBJS-$(CONFIG_AMV_ENCODER) += mjpegenc.o mjpegenc_common.o \
OBJS-$(CONFIG_ANM_DECODER) += anm.o
OBJS-$(CONFIG_ANSI_DECODER) += ansi.o cga_data.o
OBJS-$(CONFIG_APE_DECODER) += apedec.o
OBJS-$(CONFIG_APTX_DECODER) += aptxdec.o aptx.o
OBJS-$(CONFIG_APTX_ENCODER) += aptxenc.o aptx.o
OBJS-$(CONFIG_APTX_HD_DECODER) += aptxdec.o aptx.o
OBJS-$(CONFIG_APTX_HD_ENCODER) += aptxenc.o aptx.o
OBJS-$(CONFIG_APTX_DECODER) += aptx.o
OBJS-$(CONFIG_APTX_ENCODER) += aptx.o
OBJS-$(CONFIG_APTX_HD_DECODER) += aptx.o
OBJS-$(CONFIG_APTX_HD_ENCODER) += aptx.o
OBJS-$(CONFIG_APNG_DECODER) += png.o pngdec.o pngdsp.o
OBJS-$(CONFIG_APNG_ENCODER) += png.o pngenc.o
OBJS-$(CONFIG_ARBC_DECODER) += arbc.o
OBJS-$(CONFIG_SSA_DECODER) += assdec.o ass.o
OBJS-$(CONFIG_SSA_ENCODER) += assenc.o ass.o
OBJS-$(CONFIG_ASS_DECODER) += assdec.o ass.o
@@ -224,7 +209,6 @@ OBJS-$(CONFIG_ATRAC3P_DECODER) += atrac3plusdec.o atrac3plus.o \
atrac3plusdsp.o atrac.o
OBJS-$(CONFIG_ATRAC3PAL_DECODER) += atrac3plusdec.o atrac3plus.o \
atrac3plusdsp.o atrac.o
OBJS-$(CONFIG_ATRAC9_DECODER) += atrac9dec.o
OBJS-$(CONFIG_AURA_DECODER) += cyuv.o
OBJS-$(CONFIG_AURA2_DECODER) += aura.o
OBJS-$(CONFIG_AVRN_DECODER) += avrndec.o mjpegdec.o
@@ -250,9 +234,8 @@ OBJS-$(CONFIG_BRENDER_PIX_DECODER) += brenderpix.o
OBJS-$(CONFIG_C93_DECODER) += c93.o
OBJS-$(CONFIG_CAVS_DECODER) += cavs.o cavsdec.o cavsdsp.o \
cavsdata.o
OBJS-$(CONFIG_CCAPTION_DECODER) += ccaption_dec.o ass.o
OBJS-$(CONFIG_CCAPTION_DECODER) += ccaption_dec.o
OBJS-$(CONFIG_CDGRAPHICS_DECODER) += cdgraphics.o
OBJS-$(CONFIG_CDTOONS_DECODER) += cdtoons.o
OBJS-$(CONFIG_CDXL_DECODER) += cdxl.o
OBJS-$(CONFIG_CFHD_DECODER) += cfhd.o cfhddata.o
OBJS-$(CONFIG_CINEPAK_DECODER) += cinepak.o
@@ -273,7 +256,6 @@ OBJS-$(CONFIG_DCA_DECODER) += dcadec.o dca.o dcadata.o dcahuff.o \
OBJS-$(CONFIG_DCA_ENCODER) += dcaenc.o dca.o dcadata.o dcahuff.o \
dcaadpcm.o
OBJS-$(CONFIG_DDS_DECODER) += dds.o
OBJS-$(CONFIG_DERF_DPCM_DECODER) += dpcm.o
OBJS-$(CONFIG_DIRAC_DECODER) += diracdec.o dirac.o diracdsp.o diractab.o \
dirac_arith.o dirac_dwt.o dirac_vlc.o
OBJS-$(CONFIG_DFA_DECODER) += dfa.o
@@ -292,8 +274,8 @@ OBJS-$(CONFIG_DSS_SP_DECODER) += dss_sp.o
OBJS-$(CONFIG_DST_DECODER) += dstdec.o dsd.o
OBJS-$(CONFIG_DVBSUB_DECODER) += dvbsubdec.o
OBJS-$(CONFIG_DVBSUB_ENCODER) += dvbsub.o
OBJS-$(CONFIG_DVDSUB_DECODER) += dvdsubdec.o dvdsub.o
OBJS-$(CONFIG_DVDSUB_ENCODER) += dvdsubenc.o dvdsub.o
OBJS-$(CONFIG_DVDSUB_DECODER) += dvdsubdec.o
OBJS-$(CONFIG_DVDSUB_ENCODER) += dvdsubenc.o
OBJS-$(CONFIG_DVAUDIO_DECODER) += dvaudiodec.o
OBJS-$(CONFIG_DVVIDEO_DECODER) += dvdec.o dv.o dvdata.o
OBJS-$(CONFIG_DVVIDEO_ENCODER) += dvenc.o dv.o dvdata.o
@@ -361,7 +343,6 @@ OBJS-$(CONFIG_H264_DECODER) += h264dec.o h264_cabac.o h264_cavlc.o \
OBJS-$(CONFIG_H264_AMF_ENCODER) += amfenc_h264.o
OBJS-$(CONFIG_H264_CUVID_DECODER) += cuviddec.o
OBJS-$(CONFIG_H264_MEDIACODEC_DECODER) += mediacodecdec.o
OBJS-$(CONFIG_H264_MF_ENCODER) += mfenc.o mf_utils.o
OBJS-$(CONFIG_H264_MMAL_DECODER) += mmaldec.o
OBJS-$(CONFIG_H264_NVENC_ENCODER) += nvenc_h264.o
OBJS-$(CONFIG_NVENC_ENCODER) += nvenc_h264.o
@@ -370,28 +351,25 @@ OBJS-$(CONFIG_H264_OMX_ENCODER) += omx.o
OBJS-$(CONFIG_H264_QSV_DECODER) += qsvdec_h2645.o
OBJS-$(CONFIG_H264_QSV_ENCODER) += qsvenc_h264.o
OBJS-$(CONFIG_H264_RKMPP_DECODER) += rkmppdec.o
OBJS-$(CONFIG_H264_VAAPI_ENCODER) += vaapi_encode_h264.o h264_levels.o
OBJS-$(CONFIG_H264_VAAPI_ENCODER) += vaapi_encode_h264.o
OBJS-$(CONFIG_H264_VIDEOTOOLBOX_ENCODER) += videotoolboxenc.o
OBJS-$(CONFIG_H264_V4L2M2M_DECODER) += v4l2_m2m_dec.o
OBJS-$(CONFIG_H264_V4L2M2M_ENCODER) += v4l2_m2m_enc.o
OBJS-$(CONFIG_HAP_DECODER) += hapdec.o hap.o
OBJS-$(CONFIG_HAP_ENCODER) += hapenc.o hap.o
OBJS-$(CONFIG_HCA_DECODER) += hcadec.o
OBJS-$(CONFIG_HCOM_DECODER) += hcom.o
OBJS-$(CONFIG_HEVC_DECODER) += hevcdec.o hevc_mvs.o \
hevc_cabac.o hevc_refs.o hevcpred.o \
hevcdsp.o hevc_filter.o hevc_data.o
OBJS-$(CONFIG_HEVC_AMF_ENCODER) += amfenc_hevc.o
OBJS-$(CONFIG_HEVC_CUVID_DECODER) += cuviddec.o
OBJS-$(CONFIG_HEVC_MEDIACODEC_DECODER) += mediacodecdec.o
OBJS-$(CONFIG_HEVC_MF_ENCODER) += mfenc.o mf_utils.o
OBJS-$(CONFIG_HEVC_NVENC_ENCODER) += nvenc_hevc.o
OBJS-$(CONFIG_NVENC_HEVC_ENCODER) += nvenc_hevc.o
OBJS-$(CONFIG_HEVC_QSV_DECODER) += qsvdec_h2645.o
OBJS-$(CONFIG_HEVC_QSV_ENCODER) += qsvenc_hevc.o hevc_ps_enc.o \
hevc_data.o
OBJS-$(CONFIG_HEVC_RKMPP_DECODER) += rkmppdec.o
OBJS-$(CONFIG_HEVC_VAAPI_ENCODER) += vaapi_encode_h265.o h265_profile_level.o
OBJS-$(CONFIG_HEVC_VAAPI_ENCODER) += vaapi_encode_h265.o
OBJS-$(CONFIG_HEVC_V4L2M2M_DECODER) += v4l2_m2m_dec.o
OBJS-$(CONFIG_HEVC_V4L2M2M_ENCODER) += v4l2_m2m_enc.o
OBJS-$(CONFIG_HNM4_VIDEO_DECODER) += hnm4video.o
@@ -400,14 +378,10 @@ OBJS-$(CONFIG_HQ_HQA_DECODER) += hq_hqa.o hq_hqadata.o hq_hqadsp.o \
OBJS-$(CONFIG_HQX_DECODER) += hqx.o hqxvlc.o hqxdsp.o canopus.o
OBJS-$(CONFIG_HUFFYUV_DECODER) += huffyuv.o huffyuvdec.o
OBJS-$(CONFIG_HUFFYUV_ENCODER) += huffyuv.o huffyuvenc.o
OBJS-$(CONFIG_HYMT_DECODER) += huffyuv.o huffyuvdec.o
OBJS-$(CONFIG_IDCIN_DECODER) += idcinvideo.o
OBJS-$(CONFIG_IDF_DECODER) += bintext.o cga_data.o
OBJS-$(CONFIG_IFF_ILBM_DECODER) += iff.o
OBJS-$(CONFIG_ILBC_DECODER) += ilbcdec.o
OBJS-$(CONFIG_IMC_DECODER) += imc.o
OBJS-$(CONFIG_IMM4_DECODER) += imm4.o
OBJS-$(CONFIG_IMM5_DECODER) += imm5.o
OBJS-$(CONFIG_INDEO2_DECODER) += indeo2.o
OBJS-$(CONFIG_INDEO3_DECODER) += indeo3.o
OBJS-$(CONFIG_INDEO4_DECODER) += indeo4.o ivi.o
@@ -428,7 +402,6 @@ OBJS-$(CONFIG_KMVC_DECODER) += kmvc.o
OBJS-$(CONFIG_LAGARITH_DECODER) += lagarith.o lagarithrac.o
OBJS-$(CONFIG_LJPEG_ENCODER) += ljpegenc.o mjpegenc_common.o
OBJS-$(CONFIG_LOCO_DECODER) += loco.o
OBJS-$(CONFIG_LSCR_DECODER) += png.o pngdec.o pngdsp.o
OBJS-$(CONFIG_M101_DECODER) += m101.o
OBJS-$(CONFIG_MACE3_DECODER) += mace.o
OBJS-$(CONFIG_MACE6_DECODER) += mace.o
@@ -440,7 +413,6 @@ OBJS-$(CONFIG_METASOUND_DECODER) += metasound.o metasound_data.o \
OBJS-$(CONFIG_MICRODVD_DECODER) += microdvddec.o ass.o
OBJS-$(CONFIG_MIMIC_DECODER) += mimic.o
OBJS-$(CONFIG_MJPEG_DECODER) += mjpegdec.o
OBJS-$(CONFIG_MJPEG_QSV_DECODER) += qsvdec_other.o
OBJS-$(CONFIG_MJPEG_ENCODER) += mjpegenc.o mjpegenc_common.o \
mjpegenc_huffman.o
OBJS-$(CONFIG_MJPEGB_DECODER) += mjpegbdec.o
@@ -462,7 +434,6 @@ OBJS-$(CONFIG_MP2FIXED_ENCODER) += mpegaudioenc_fixed.o mpegaudio.o \
mpegaudiodata.o mpegaudiodsp_data.o
OBJS-$(CONFIG_MP2FLOAT_DECODER) += mpegaudiodec_float.o
OBJS-$(CONFIG_MP3_DECODER) += mpegaudiodec_fixed.o
OBJS-$(CONFIG_MP3_MF_ENCODER) += mfenc.o mf_utils.o
OBJS-$(CONFIG_MP3ADU_DECODER) += mpegaudiodec_fixed.o
OBJS-$(CONFIG_MP3ADUFLOAT_DECODER) += mpegaudiodec_float.o
OBJS-$(CONFIG_MP3FLOAT_DECODER) += mpegaudiodec_float.o
@@ -505,23 +476,17 @@ OBJS-$(CONFIG_MSVIDEO1_DECODER) += msvideo1.o
OBJS-$(CONFIG_MSVIDEO1_ENCODER) += msvideo1enc.o elbg.o
OBJS-$(CONFIG_MSZH_DECODER) += lcldec.o
OBJS-$(CONFIG_MTS2_DECODER) += mss4.o
OBJS-$(CONFIG_MV30_DECODER) += mv30.o
OBJS-$(CONFIG_MVC1_DECODER) += mvcdec.o
OBJS-$(CONFIG_MVC2_DECODER) += mvcdec.o
OBJS-$(CONFIG_MVDV_DECODER) += midivid.o
OBJS-$(CONFIG_MVHA_DECODER) += mvha.o
OBJS-$(CONFIG_MWSC_DECODER) += mwsc.o
OBJS-$(CONFIG_MXPEG_DECODER) += mxpegdec.o
OBJS-$(CONFIG_NELLYMOSER_DECODER) += nellymoserdec.o nellymoser.o
OBJS-$(CONFIG_NELLYMOSER_ENCODER) += nellymoserenc.o nellymoser.o
OBJS-$(CONFIG_NOTCHLC_DECODER) += notchlc.o
OBJS-$(CONFIG_NUV_DECODER) += nuv.o rtjpeg.o
OBJS-$(CONFIG_ON2AVC_DECODER) += on2avc.o on2avcdata.o
OBJS-$(CONFIG_OPUS_DECODER) += opusdec.o opus.o opus_celt.o opus_rc.o \
opus_pvq.o opus_silk.o opustab.o vorbis_data.o \
opusdsp.o
opus_pvq.o opus_silk.o opustab.o vorbis_data.o
OBJS-$(CONFIG_OPUS_ENCODER) += opusenc.o opus.o opus_rc.o opustab.o opus_pvq.o \
opusenc_psy.o vorbis_data.o
opusenc_psy.o
OBJS-$(CONFIG_PAF_AUDIO_DECODER) += pafaudio.o
OBJS-$(CONFIG_PAF_VIDEO_DECODER) += pafvideo.o
OBJS-$(CONFIG_PAM_DECODER) += pnmdec.o pnm.o
@@ -530,7 +495,6 @@ OBJS-$(CONFIG_PBM_DECODER) += pnmdec.o pnm.o
OBJS-$(CONFIG_PBM_ENCODER) += pnmenc.o
OBJS-$(CONFIG_PCX_DECODER) += pcx.o
OBJS-$(CONFIG_PCX_ENCODER) += pcxenc.o
OBJS-$(CONFIG_PFM_DECODER) += pnmdec.o pnm.o
OBJS-$(CONFIG_PGM_DECODER) += pnmdec.o pnm.o
OBJS-$(CONFIG_PGM_ENCODER) += pnmenc.o
OBJS-$(CONFIG_PGMYUV_DECODER) += pnmdec.o pnm.o
@@ -544,10 +508,10 @@ OBJS-$(CONFIG_PNG_ENCODER) += png.o pngenc.o
OBJS-$(CONFIG_PPM_DECODER) += pnmdec.o pnm.o
OBJS-$(CONFIG_PPM_ENCODER) += pnmenc.o
OBJS-$(CONFIG_PRORES_DECODER) += proresdec2.o proresdsp.o proresdata.o
OBJS-$(CONFIG_PRORES_ENCODER) += proresenc_anatoliy.o proresdata.o
OBJS-$(CONFIG_PRORES_AW_ENCODER) += proresenc_anatoliy.o proresdata.o
OBJS-$(CONFIG_PRORES_LGPL_DECODER) += proresdec_lgpl.o proresdsp.o proresdata.o
OBJS-$(CONFIG_PRORES_ENCODER) += proresenc_anatoliy.o
OBJS-$(CONFIG_PRORES_AW_ENCODER) += proresenc_anatoliy.o
OBJS-$(CONFIG_PRORES_KS_ENCODER) += proresenc_kostya.o proresdata.o
OBJS-$(CONFIG_PROSUMER_DECODER) += prosumer.o
OBJS-$(CONFIG_PSD_DECODER) += psd.o
OBJS-$(CONFIG_PTX_DECODER) += ptx.o
OBJS-$(CONFIG_QCELP_DECODER) += qcelpdec.o \
@@ -567,7 +531,6 @@ OBJS-$(CONFIG_RA_144_DECODER) += ra144dec.o ra144.o celp_filters.o
OBJS-$(CONFIG_RA_144_ENCODER) += ra144enc.o ra144.o celp_filters.o
OBJS-$(CONFIG_RA_288_DECODER) += ra288.o celp_filters.o
OBJS-$(CONFIG_RALF_DECODER) += ralf.o
OBJS-$(CONFIG_RASC_DECODER) += rasc.o
OBJS-$(CONFIG_RAWVIDEO_DECODER) += rawdec.o
OBJS-$(CONFIG_RAWVIDEO_ENCODER) += rawenc.o
OBJS-$(CONFIG_REALTEXT_DECODER) += realtextdec.o ass.o
@@ -600,19 +563,18 @@ OBJS-$(CONFIG_SIPR_DECODER) += sipr.o acelp_pitch_delay.o \
celp_math.o acelp_vectors.o \
acelp_filters.o celp_filters.o \
sipr16k.o
OBJS-$(CONFIG_SIREN_DECODER) += siren.o
OBJS-$(CONFIG_SMACKAUD_DECODER) += smacker.o
OBJS-$(CONFIG_SMACKER_DECODER) += smacker.o
OBJS-$(CONFIG_SMC_DECODER) += smc.o
OBJS-$(CONFIG_SMVJPEG_DECODER) += smvjpegdec.o
OBJS-$(CONFIG_SNOW_DECODER) += snowdec.o snow.o snow_dwt.o
OBJS-$(CONFIG_SNOW_ENCODER) += snowenc.o snow.o snow_dwt.o \
h263.o h263data.o ituh263enc.o
h263.o ituh263enc.o
OBJS-$(CONFIG_SOL_DPCM_DECODER) += dpcm.o
OBJS-$(CONFIG_SONIC_DECODER) += sonic.o
OBJS-$(CONFIG_SONIC_ENCODER) += sonic.o
OBJS-$(CONFIG_SONIC_LS_ENCODER) += sonic.o
OBJS-$(CONFIG_SPEEDHQ_DECODER) += speedhq.o mpeg12.o mpeg12data.o simple_idct.o
OBJS-$(CONFIG_SPEEDHQ_DECODER) += speedhq.o simple_idct.o
OBJS-$(CONFIG_SP5X_DECODER) += sp5xdec.o
OBJS-$(CONFIG_SRGC_DECODER) += mscc.o
OBJS-$(CONFIG_SRT_DECODER) += srtdec.o ass.o htmlsubtitles.o
@@ -627,10 +589,10 @@ OBJS-$(CONFIG_SUNRAST_ENCODER) += sunrastenc.o
OBJS-$(CONFIG_LIBRSVG_DECODER) += librsvgdec.o
OBJS-$(CONFIG_SBC_DECODER) += sbcdec.o sbcdec_data.o sbc.o
OBJS-$(CONFIG_SBC_ENCODER) += sbcenc.o sbc.o sbcdsp.o sbcdsp_data.o
OBJS-$(CONFIG_SVQ1_DECODER) += svq1dec.o svq1.o h263data.o
OBJS-$(CONFIG_SVQ1_DECODER) += svq1dec.o svq1.o svq13.o h263data.o
OBJS-$(CONFIG_SVQ1_ENCODER) += svq1enc.o svq1.o h263data.o \
h263.o ituh263enc.o
OBJS-$(CONFIG_SVQ3_DECODER) += svq3.o mpegutils.o h264data.o
OBJS-$(CONFIG_SVQ3_DECODER) += svq3.o svq13.o mpegutils.o h264data.o
OBJS-$(CONFIG_TEXT_DECODER) += textdec.o ass.o
OBJS-$(CONFIG_TEXT_ENCODER) += srtenc.o ass_split.o
OBJS-$(CONFIG_TAK_DECODER) += takdec.o tak.o takdsp.o
@@ -639,7 +601,7 @@ OBJS-$(CONFIG_TARGA_ENCODER) += targaenc.o rle.o
OBJS-$(CONFIG_TARGA_Y216_DECODER) += targa_y216dec.o
OBJS-$(CONFIG_TDSC_DECODER) += tdsc.o
OBJS-$(CONFIG_TIERTEXSEQVIDEO_DECODER) += tiertexseqv.o
OBJS-$(CONFIG_TIFF_DECODER) += tiff.o lzw.o faxcompr.o tiff_data.o tiff_common.o mjpegdec.o
OBJS-$(CONFIG_TIFF_DECODER) += tiff.o lzw.o faxcompr.o tiff_data.o tiff_common.o
OBJS-$(CONFIG_TIFF_ENCODER) += tiffenc.o rle.o lzwenc.o tiff_data.o
OBJS-$(CONFIG_TMV_DECODER) += tmv.o cga_data.o
OBJS-$(CONFIG_TRUEHD_DECODER) += mlpdec.o mlpdsp.o
@@ -705,13 +667,11 @@ OBJS-$(CONFIG_VP9_CUVID_DECODER) += cuviddec.o
OBJS-$(CONFIG_VP9_MEDIACODEC_DECODER) += mediacodecdec.o
OBJS-$(CONFIG_VP9_RKMPP_DECODER) += rkmppdec.o
OBJS-$(CONFIG_VP9_VAAPI_ENCODER) += vaapi_encode_vp9.o
OBJS-$(CONFIG_VP9_QSV_ENCODER) += qsvenc_vp9.o
OBJS-$(CONFIG_VPLAYER_DECODER) += textdec.o ass.o
OBJS-$(CONFIG_VP9_V4L2M2M_DECODER) += v4l2_m2m_dec.o
OBJS-$(CONFIG_VQA_DECODER) += vqavideo.o
OBJS-$(CONFIG_WAVPACK_DECODER) += wavpack.o dsd.o
OBJS-$(CONFIG_WAVPACK_DECODER) += wavpack.o
OBJS-$(CONFIG_WAVPACK_ENCODER) += wavpackenc.o
OBJS-$(CONFIG_WCMV_DECODER) += wcmv.o
OBJS-$(CONFIG_WEBP_DECODER) += webp.o
OBJS-$(CONFIG_WEBVTT_DECODER) += webvttdec.o ass.o
OBJS-$(CONFIG_WEBVTT_ENCODER) += webvttenc.o ass_split.o
@@ -725,7 +685,7 @@ OBJS-$(CONFIG_WMAVOICE_DECODER) += wmavoice.o \
celp_filters.o \
acelp_vectors.o acelp_filters.o
OBJS-$(CONFIG_WMV1_DECODER) += msmpeg4dec.o msmpeg4.o msmpeg4data.o
OBJS-$(CONFIG_WMV1_ENCODER) += msmpeg4enc.o msmpeg4.o msmpeg4data.o
OBJS-$(CONFIG_WMV1_ENCODER) += msmpeg4enc.o
OBJS-$(CONFIG_WMV2_DECODER) += wmv2dec.o wmv2.o wmv2data.o \
msmpeg4dec.o msmpeg4.o msmpeg4data.o
OBJS-$(CONFIG_WMV2_ENCODER) += wmv2enc.o wmv2.o wmv2data.o \
@@ -767,7 +727,6 @@ OBJS-$(CONFIG_PCM_ALAW_DECODER) += pcm.o
OBJS-$(CONFIG_PCM_ALAW_ENCODER) += pcm.o
OBJS-$(CONFIG_PCM_BLURAY_DECODER) += pcm-bluray.o
OBJS-$(CONFIG_PCM_DVD_DECODER) += pcm-dvd.o
OBJS-$(CONFIG_PCM_DVD_ENCODER) += pcm-dvdenc.o
OBJS-$(CONFIG_PCM_F16LE_DECODER) += pcm.o
OBJS-$(CONFIG_PCM_F24LE_DECODER) += pcm.o
OBJS-$(CONFIG_PCM_F32BE_DECODER) += pcm.o
@@ -825,16 +784,13 @@ OBJS-$(CONFIG_PCM_U32BE_DECODER) += pcm.o
OBJS-$(CONFIG_PCM_U32BE_ENCODER) += pcm.o
OBJS-$(CONFIG_PCM_U32LE_DECODER) += pcm.o
OBJS-$(CONFIG_PCM_U32LE_ENCODER) += pcm.o
OBJS-$(CONFIG_PCM_VIDC_DECODER) += pcm.o
OBJS-$(CONFIG_PCM_VIDC_ENCODER) += pcm.o
OBJS-$(CONFIG_PCM_ZORK_DECODER) += pcm.o
OBJS-$(CONFIG_ADPCM_4XM_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_ADX_DECODER) += adxdec.o adx.o
OBJS-$(CONFIG_ADPCM_ADX_ENCODER) += adxenc.o adx.o
OBJS-$(CONFIG_ADPCM_AFC_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_AGM_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_AICA_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_ARGO_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_CT_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_DTK_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_EA_DECODER) += adpcm.o adpcm_data.o
@@ -850,23 +806,17 @@ OBJS-$(CONFIG_ADPCM_G726_ENCODER) += g726.o
OBJS-$(CONFIG_ADPCM_G726LE_DECODER) += g726.o
OBJS-$(CONFIG_ADPCM_G726LE_ENCODER) += g726.o
OBJS-$(CONFIG_ADPCM_IMA_AMV_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_ALP_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_APC_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_APM_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_CUNNING_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_DAT4_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_DK3_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_DK4_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_EA_EACS_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_EA_SEAD_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_ISS_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_MTF_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_OKI_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_QT_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_QT_ENCODER) += adpcmenc.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_RAD_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_SSI_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_SSI_ENCODER) += adpcmenc.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_SMJPEG_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_WAV_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_WAV_ENCODER) += adpcmenc.o adpcm_data.o
@@ -886,7 +836,6 @@ OBJS-$(CONFIG_ADPCM_VIMA_DECODER) += vima.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_XA_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_YAMAHA_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_YAMAHA_ENCODER) += adpcmenc.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_ZORK_DECODER) += adpcm.o adpcm_data.o
# hardware accelerators
OBJS-$(CONFIG_D3D11VA) += dxva2.o
@@ -909,7 +858,7 @@ OBJS-$(CONFIG_HEVC_D3D11VA_HWACCEL) += dxva2_hevc.o
OBJS-$(CONFIG_HEVC_DXVA2_HWACCEL) += dxva2_hevc.o
OBJS-$(CONFIG_HEVC_NVDEC_HWACCEL) += nvdec_hevc.o
OBJS-$(CONFIG_HEVC_QSV_HWACCEL) += qsvdec_h2645.o
OBJS-$(CONFIG_HEVC_VAAPI_HWACCEL) += vaapi_hevc.o h265_profile_level.o
OBJS-$(CONFIG_HEVC_VAAPI_HWACCEL) += vaapi_hevc.o
OBJS-$(CONFIG_HEVC_VDPAU_HWACCEL) += vdpau_hevc.o
OBJS-$(CONFIG_MJPEG_NVDEC_HWACCEL) += nvdec_mjpeg.o
OBJS-$(CONFIG_MJPEG_VAAPI_HWACCEL) += vaapi_mjpeg.o
@@ -941,7 +890,6 @@ OBJS-$(CONFIG_VP9_D3D11VA_HWACCEL) += dxva2_vp9.o
OBJS-$(CONFIG_VP9_DXVA2_HWACCEL) += dxva2_vp9.o
OBJS-$(CONFIG_VP9_NVDEC_HWACCEL) += nvdec_vp9.o
OBJS-$(CONFIG_VP9_VAAPI_HWACCEL) += vaapi_vp9.o
OBJS-$(CONFIG_VP9_VDPAU_HWACCEL) += vdpau_vp9.o
OBJS-$(CONFIG_VP8_QSV_HWACCEL) += qsvdec_other.o
# libavformat dependencies
@@ -993,12 +941,9 @@ OBJS-$(CONFIG_PCM_ALAW_AT_ENCODER) += audiotoolboxenc.o
OBJS-$(CONFIG_PCM_MULAW_AT_ENCODER) += audiotoolboxenc.o
OBJS-$(CONFIG_LIBAOM_AV1_DECODER) += libaomdec.o
OBJS-$(CONFIG_LIBAOM_AV1_ENCODER) += libaomenc.o
OBJS-$(CONFIG_LIBARIBB24_DECODER) += libaribb24.o ass.o
OBJS-$(CONFIG_LIBCELT_DECODER) += libcelt_dec.o
OBJS-$(CONFIG_LIBCODEC2_DECODER) += libcodec2.o codec2utils.o
OBJS-$(CONFIG_LIBCODEC2_ENCODER) += libcodec2.o codec2utils.o
OBJS-$(CONFIG_LIBDAV1D_DECODER) += libdav1d.o
OBJS-$(CONFIG_LIBDAVS2_DECODER) += libdavs2.o
OBJS-$(CONFIG_LIBFDK_AAC_DECODER) += libfdk-aacdec.o
OBJS-$(CONFIG_LIBFDK_AAC_ENCODER) += libfdk-aacenc.o
OBJS-$(CONFIG_LIBGSM_DECODER) += libgsmdec.o
@@ -1020,7 +965,6 @@ OBJS-$(CONFIG_LIBOPUS_DECODER) += libopusdec.o libopus.o \
vorbis_data.o
OBJS-$(CONFIG_LIBOPUS_ENCODER) += libopusenc.o libopus.o \
vorbis_data.o
OBJS-$(CONFIG_LIBRAV1E_ENCODER) += librav1e.o
OBJS-$(CONFIG_LIBSHINE_ENCODER) += libshine.o
OBJS-$(CONFIG_LIBSPEEX_DECODER) += libspeexdec.o
OBJS-$(CONFIG_LIBSPEEX_ENCODER) += libspeexenc.o
@@ -1041,8 +985,7 @@ OBJS-$(CONFIG_LIBX262_ENCODER) += libx264.o
OBJS-$(CONFIG_LIBX264_ENCODER) += libx264.o
OBJS-$(CONFIG_LIBX265_ENCODER) += libx265.o
OBJS-$(CONFIG_LIBXAVS_ENCODER) += libxavs.o
OBJS-$(CONFIG_LIBXAVS2_ENCODER) += libxavs2.o
OBJS-$(CONFIG_LIBXVID_ENCODER) += libxvid.o
OBJS-$(CONFIG_LIBXVID_ENCODER) += libxvid.o libxvid_rc.o
OBJS-$(CONFIG_LIBZVBI_TELETEXT_DECODER) += libzvbi-teletextdec.o ass.o
# parsers
@@ -1051,8 +994,6 @@ OBJS-$(CONFIG_AAC_PARSER) += aac_parser.o aac_ac3_parser.o \
mpeg4audio.o
OBJS-$(CONFIG_AC3_PARSER) += ac3tab.o aac_ac3_parser.o
OBJS-$(CONFIG_ADX_PARSER) += adx_parser.o adx.o
OBJS-$(CONFIG_AV1_PARSER) += av1_parser.o av1_parse.o
OBJS-$(CONFIG_AVS2_PARSER) += avs2_parser.o
OBJS-$(CONFIG_BMP_PARSER) += bmp_parser.o
OBJS-$(CONFIG_CAVSVIDEO_PARSER) += cavs_parser.o
OBJS-$(CONFIG_COOK_PARSER) += cook_parser.o
@@ -1066,20 +1007,18 @@ OBJS-$(CONFIG_DVD_NAV_PARSER) += dvd_nav_parser.o
OBJS-$(CONFIG_DVDSUB_PARSER) += dvdsub_parser.o
OBJS-$(CONFIG_FLAC_PARSER) += flac_parser.o flacdata.o flac.o \
vorbis_data.o
OBJS-$(CONFIG_G723_1_PARSER) += g723_1_parser.o
OBJS-$(CONFIG_G729_PARSER) += g729_parser.o
OBJS-$(CONFIG_GIF_PARSER) += gif_parser.o
OBJS-$(CONFIG_GSM_PARSER) += gsm_parser.o
OBJS-$(CONFIG_H261_PARSER) += h261_parser.o
OBJS-$(CONFIG_H263_PARSER) += h263_parser.o
OBJS-$(CONFIG_H264_PARSER) += h264_parser.o h264_sei.o h264data.o
OBJS-$(CONFIG_HEVC_PARSER) += hevc_parser.o hevc_data.o
OBJS-$(CONFIG_JPEG2000_PARSER) += jpeg2000_parser.o
OBJS-$(CONFIG_MJPEG_PARSER) += mjpeg_parser.o
OBJS-$(CONFIG_MLP_PARSER) += mlp_parse.o mlp_parser.o mlp.o
OBJS-$(CONFIG_MLP_PARSER) += mlp_parser.o mlp.o
OBJS-$(CONFIG_MPEG4VIDEO_PARSER) += mpeg4video_parser.o h263.o \
mpeg4videodec.o mpeg4video.o \
ituh263dec.o h263dec.o h263data.o
OBJS-$(CONFIG_PNG_PARSER) += png_parser.o
OBJS-$(CONFIG_MPEGAUDIO_PARSER) += mpegaudio_parser.o
OBJS-$(CONFIG_MPEGVIDEO_PARSER) += mpegvideo_parser.o \
mpeg12.o mpeg12data.o
@@ -1097,26 +1036,22 @@ OBJS-$(CONFIG_VC1_PARSER) += vc1_parser.o vc1.o vc1data.o \
OBJS-$(CONFIG_VP3_PARSER) += vp3_parser.o
OBJS-$(CONFIG_VP8_PARSER) += vp8_parser.o
OBJS-$(CONFIG_VP9_PARSER) += vp9_parser.o
OBJS-$(CONFIG_WEBP_PARSER) += webp_parser.o
OBJS-$(CONFIG_XMA_PARSER) += xma_parser.o
# bitstream filters
OBJS-$(CONFIG_AAC_ADTSTOASC_BSF) += aac_adtstoasc_bsf.o mpeg4audio.o
OBJS-$(CONFIG_AV1_METADATA_BSF) += av1_metadata_bsf.o
OBJS-$(CONFIG_AV1_FRAME_MERGE_BSF) += av1_frame_merge_bsf.o
OBJS-$(CONFIG_AV1_FRAME_SPLIT_BSF) += av1_frame_split_bsf.o
OBJS-$(CONFIG_CHOMP_BSF) += chomp_bsf.o
OBJS-$(CONFIG_DUMP_EXTRADATA_BSF) += dump_extradata_bsf.o
OBJS-$(CONFIG_DCA_CORE_BSF) += dca_core_bsf.o
OBJS-$(CONFIG_EAC3_CORE_BSF) += eac3_core_bsf.o
OBJS-$(CONFIG_EXTRACT_EXTRADATA_BSF) += extract_extradata_bsf.o \
av1_parse.o h2645_parse.o
h2645_parse.o
OBJS-$(CONFIG_FILTER_UNITS_BSF) += filter_units_bsf.o
OBJS-$(CONFIG_H264_METADATA_BSF) += h264_metadata_bsf.o h264_levels.o
OBJS-$(CONFIG_H264_METADATA_BSF) += h264_metadata_bsf.o
OBJS-$(CONFIG_H264_MP4TOANNEXB_BSF) += h264_mp4toannexb_bsf.o
OBJS-$(CONFIG_H264_REDUNDANT_PPS_BSF) += h264_redundant_pps_bsf.o
OBJS-$(CONFIG_HAPQA_EXTRACT_BSF) += hapqa_extract_bsf.o hap.o
OBJS-$(CONFIG_HEVC_METADATA_BSF) += h265_metadata_bsf.o h265_profile_level.o
OBJS-$(CONFIG_HEVC_METADATA_BSF) += h265_metadata_bsf.o
OBJS-$(CONFIG_HEVC_MP4TOANNEXB_BSF) += hevc_mp4toannexb_bsf.o
OBJS-$(CONFIG_IMX_DUMP_HEADER_BSF) += imx_dump_header_bsf.o
OBJS-$(CONFIG_MJPEG2JPEG_BSF) += mjpeg2jpeg_bsf.o
@@ -1128,14 +1063,9 @@ OBJS-$(CONFIG_MP3_HEADER_DECOMPRESS_BSF) += mp3_header_decompress_bsf.o \
OBJS-$(CONFIG_MPEG2_METADATA_BSF) += mpeg2_metadata_bsf.o
OBJS-$(CONFIG_NOISE_BSF) += noise_bsf.o
OBJS-$(CONFIG_NULL_BSF) += null_bsf.o
OBJS-$(CONFIG_OPUS_METADATA_BSF) += opus_metadata_bsf.o
OBJS-$(CONFIG_PCM_RECHUNK_BSF) += pcm_rechunk_bsf.o
OBJS-$(CONFIG_PRORES_METADATA_BSF) += prores_metadata_bsf.o
OBJS-$(CONFIG_REMOVE_EXTRADATA_BSF) += remove_extradata_bsf.o
OBJS-$(CONFIG_TEXT2MOVSUB_BSF) += movsub_bsf.o
OBJS-$(CONFIG_TRACE_HEADERS_BSF) += trace_headers_bsf.o
OBJS-$(CONFIG_TRUEHD_CORE_BSF) += truehd_core_bsf.o mlp_parse.o mlp.o
OBJS-$(CONFIG_VP9_METADATA_BSF) += vp9_metadata_bsf.o
OBJS-$(CONFIG_VP9_RAW_REORDER_BSF) += vp9_raw_reorder_bsf.o
OBJS-$(CONFIG_VP9_SUPERFRAME_BSF) += vp9_superframe_bsf.o
OBJS-$(CONFIG_VP9_SUPERFRAME_SPLIT_BSF) += vp9_superframe_split_bsf.o
@@ -1167,7 +1097,6 @@ SKIPHEADERS-$(CONFIG_JNI) += ffjni.h
SKIPHEADERS-$(CONFIG_LIBVPX) += libvpx.h
SKIPHEADERS-$(CONFIG_LIBWEBP_ENCODER) += libwebpenc_common.h
SKIPHEADERS-$(CONFIG_MEDIACODEC) += mediacodecdec_common.h mediacodec_surface.h mediacodec_wrapper.h mediacodec_sw_buffer.h
SKIPHEADERS-$(CONFIG_MEDIAFOUNDATION) += mf_utils.h
SKIPHEADERS-$(CONFIG_NVDEC) += nvdec.h
SKIPHEADERS-$(CONFIG_NVENC) += nvenc.h
SKIPHEADERS-$(CONFIG_QSV) += qsv.h qsv_internal.h
@@ -1198,8 +1127,6 @@ TESTPROGS-$(CONFIG_IDCTDSP) += dct
TESTPROGS-$(CONFIG_IIRFILTER) += iirfilter
TESTPROGS-$(HAVE_MMX) += motion
TESTPROGS-$(CONFIG_MPEGVIDEO) += mpeg12framerate
TESTPROGS-$(CONFIG_H264_METADATA_BSF) += h264_levels
TESTPROGS-$(CONFIG_HEVC_METADATA_BSF) += h265_levels
TESTPROGS-$(CONFIG_RANGECODER) += rangecoder
TESTPROGS-$(CONFIG_SNOW_ENCODER) += snowenc

View File

@@ -60,11 +60,11 @@ typedef struct A64Context {
} A64Context;
/* gray gradient */
static const uint8_t mc_colors[5]={0x0,0xb,0xc,0xf,0x1};
static const int mc_colors[5]={0x0,0xb,0xc,0xf,0x1};
/* other possible gradients - to be tested */
//static const uint8_t mc_colors[5]={0x0,0x8,0xa,0xf,0x7};
//static const uint8_t mc_colors[5]={0x0,0x9,0x8,0xa,0x3};
//static const int mc_colors[5]={0x0,0x8,0xa,0xf,0x7};
//static const int mc_colors[5]={0x0,0x9,0x8,0xa,0x3};
static void to_meta_with_crop(AVCodecContext *avctx,
const AVFrame *p, int *dest)
@@ -407,7 +407,6 @@ AVCodec ff_a64multi_encoder = {
.close = a64multi_close_encoder,
.pix_fmts = (const enum AVPixelFormat[]) {AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE},
.capabilities = AV_CODEC_CAP_DELAY,
.caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
};
#endif
#if CONFIG_A64MULTI5_ENCODER
@@ -422,6 +421,5 @@ AVCodec ff_a64multi5_encoder = {
.close = a64multi_close_encoder,
.pix_fmts = (const enum AVPixelFormat[]) {AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE},
.capabilities = AV_CODEC_CAP_DELAY,
.caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
};
#endif

View File

@@ -356,7 +356,7 @@ struct AACContext {
OutputConfiguration oc[2];
int warned_num_aac_frames;
int warned_960_sbr;
unsigned warned_71_wide;
int warned_gain_control;
/* aacdec functions pointers */
@@ -368,7 +368,7 @@ struct AACContext {
INTFLOAT *in, IndividualChannelStream *ics);
void (*update_ltp)(AACContext *ac, SingleChannelElement *sce);
void (*vector_pow43)(int *coefs, int len);
void (*subband_scale)(int *dst, int *src, int scale, int offset, int len, void *log_context);
void (*subband_scale)(int *dst, int *src, int scale, int offset, int len);
};

View File

@@ -21,8 +21,8 @@
#include "adts_header.h"
#include "adts_parser.h"
#include "avcodec.h"
#include "bsf.h"
#include "bsf_internal.h"
#include "put_bits.h"
#include "get_bits.h"
#include "mpeg4audio.h"
@@ -134,8 +134,8 @@ static int aac_adtstoasc_init(AVBSFContext *ctx)
/* Validate the extradata if the stream is already MPEG-4 AudioSpecificConfig */
if (ctx->par_in->extradata) {
MPEG4AudioConfig mp4ac;
int ret = avpriv_mpeg4audio_get_config2(&mp4ac, ctx->par_in->extradata,
ctx->par_in->extradata_size, 1, ctx);
int ret = avpriv_mpeg4audio_get_config(&mp4ac, ctx->par_in->extradata,
ctx->par_in->extradata_size * 8, 1);
if (ret < 0) {
av_log(ctx, AV_LOG_ERROR, "Error parsing AudioSpecificConfig extradata!\n");
return ret;

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