| Commit message (Collapse) | Author | Age |
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Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
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Now we just use one ADTS raw frame to calculate the bit rate, it's
lead to a larger error when get the duration from bit rate, the
improvement cumulate Nth ADTS frames to get the average bit rate.
e,g used the command get the duration like:
ffprobe -show_entries format=duration -i fate-suite/aac/foo.aac
before this improvement dump the duration=2.173935
after this improvement dump the duration=1.979267
in fact, the real duration can be get by command like:
ffmpeg -i fate-suite/aac/foo.aac -f null /dev/null with time=00:00:01.97
Also update the fate-adtstoasc_ticket3715.
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
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Signed-off-by: Ting Fu <ting.fu@intel.com>
Signed-off-by: Guo Yejun <yejun.guo@intel.com>
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Signed-off-by: Ting Fu <ting.fu@intel.com>
Signed-off-by: Guo Yejun <yejun.guo@intel.com>
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Signed-off-by: Ting Fu <ting.fu@intel.com>
Signed-off-by: Guo Yejun <yejun.guo@intel.com>
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Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
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This is a requirement of the AV1-ISOBMFF spec. Section 2.1.
General Requirements & Brands states:
* It SHALL have the av01 brand among the compatible brands array of the FileTypeBox
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
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Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
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Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
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Signed-off-by: Fei Wang <fei.w.wang@intel.com>
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The format is packed RGB with each channel 10 bits available and
include 2 bits unused.
Signed-off-by: Fei Wang <fei.w.wang@intel.com>
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Signed-off-by: Ting Fu <ting.fu@intel.com>
Signed-off-by: Guo Yejun <yejun.guo@intel.com>
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Signed-off-by: Ting Fu <ting.fu@intel.com>
Signed-off-by: Guo Yejun <yejun.guo@intel.com>
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Signed-off-by: Ting Fu <ting.fu@intel.com>
Signed-off-by: Guo Yejun <yejun.guo@intel.com>
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Replace it with avcodec_free_context() or drop it completely as
appropriate.
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This causes regressions in end to end timestamps with mp3s and ffmpeg.
The revert is to avoid this regression in the 4.3 release
See: [FFmpeg-devel] [PATCH] Don't adjust start time for MP3 files; packets are not adjusted.
This reverts commit 460132c9980f8a1f501a1f69477bca49e1641233.
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Stop using the deprecated global option
Signed-off-by: James Almer <jamrial@gmail.com>
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Signed-off-by: Zane van Iperen <zane@zanevaniperen.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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Fix mixed declaration and code.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
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Signed-off-by: Ting Fu <ting.fu@intel.com>
Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
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7546ac2fee4 made it so that the start_time for mp3 files is
adjusted for skip_samples. However, this appears incorrect because
subsequent packet timestamps are not adjusted and skip_samples are
applied by deleting data from a packet without changing the timestamp.
E.g., we are told the start_time is ~25ms and we get a packet with a
timestamp of 0 that has had the skip_samples discarded from it. As such
rendering engines may incorrectly discard everything prior to the
25ms thinking that is where playback should officially start. Since the
samples were deleted without adjusting timestamps though, the true
start_time is still 0.
Other formats like MP4 with edit lists will adjust both the start
time and the timestamps of subsequent packets to avoid this issue.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
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Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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Use 0, which selects the alignment automatically.
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A buffer whose size is not a multiple of four has been initialized using
consecutive writes of 32bits. This results in a stack-buffer-overflow
reported by ASAN in the checkasm-sw_scale FATE-test.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
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single: Single-track
track{0,1}: Dual-track
trunc-t1: Truncated track 1
trunc-t2-track{0,1}: Fully-truncated track 2
trunc-t2a-track{0,1}: Partially-truncated track 2
trunc-h2: Truncated track 2 header
Signed-off-by: Zane van Iperen <zane@zanevaniperen.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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changes since v1
- default behavior, no longer hidden behind decoder parameter
- updated tests to reflect change
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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Also fixes colorhold filtering.
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The Matroska muxer writes the Chapters early when chapters were already
available when writing the header; in this case any tags pertaining to
these chapters get written, too.
Yet if no chapters had been supplied before writing the header, Chapters
can also be written when writing the trailer if any are supplied. Tags
belonging to these chapters were up until now completely ignored.
This commit changes this: Writing the tags belonging to chapters has
been moved to mkv_write_chapters(). If mkv_write_tags() has not been
called yet (i.e. when chapters are written when writing the header),
the AVIOContext for writing the ordinary Tags element is used, but not
output, as this is left to mkv_write_tags() in order to only write one
Tags element. Yet if mkv_write_tags() has already been called,
mkv_write_chapters() will output a Tags element of its own which only
contains the tags for chapters.
When chapters are available initially, the corresponding tags will now
be the first tags in the Tags element; but the ordering of tags in Tags
is irrelevant anyway.
This commit also makes chapter_id_offset local to mkv_write_chapters()
as it is used only there and not reused at all.
Potentially writing a second Tags element means that the maximum number
of SeekHead entries had to be incremented. All the changes to FATE
result from the ensuing increase in the amount of space reserved for the
SeekHead (21 bytes more).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
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We won't be able to seek back to write the actual duration anyway.
FATE-tests using the md5pipe command had to be updated due to this change.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
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This function doesn't call emms.
Signed-off-by: Martin Storsjö <martin@martin.st>
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Signed-off-by: Martin Storsjö <martin@martin.st>
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diff_pixels_unaligned
Signed-off-by: Martin Storsjö <martin@martin.st>
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Also fill x8-x17 with garbage before calling the function.
Figure out the number of stack parameters and make sure that the
value on the stack after those is untouched.
Signed-off-by: Martin Storsjö <martin@martin.st>
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Figure out the number of stack parameters and make sure that the
value on the stack after those is untouched.
Signed-off-by: Martin Storsjö <martin@martin.st>
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We should just use a normal bl here, and the linker will add the 'x'
bit if necessary.
This fixes calling the checkasm_fail_func on windows, where the
code is built in thumb mode (and the linker doesn't clear the 'x'
bit in the blx instruction).
Signed-off-by: Martin Storsjö <martin@martin.st>
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This makes it easier to share code with e.g. the dav1d implementation
of checkasm.
Signed-off-by: Martin Storsjö <martin@martin.st>
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This tests the hscale 8bpp to 14/18bpp functions with different filter
sizes.
Signed-off-by: Josh de Kock <josh@itanimul.li>
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This was ported from dav1d (c950e7101bdf5f7117bfca816984a21e550509f0).
Signed-off-by: Josh de Kock <josh@itanimul.li>
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have tested on linux x86_32/64, mingw32/64 arm & mips qemu
Tested-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
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Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
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Tested on x86-32/64, mingw32/64, arm & mips qemu
Tested-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
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Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
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Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
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The patch will make audio and subtitle packets be marked as AV_PKT_FLAG_KEY.
For audio, it'll caused the audio sample to be sync sample.
To verify ref/fate/movenc results:
1. Get the movenc test data
[lmwang@vpn ffmpeg]$ libavformat/tests/movenc -w && mkdir -p audio_old && mv *.mp4 audio_old_
After applied the patch:
[lmwang@vpn ffmpeg]$ make fate-movenc SAMPLES=../fate-suite
[lmwang@vpn ffmpeg]$ libavformat/tests/movenc -w && mkdir -p audio_key && mv *.mp4 audio_key
2. Get l-smash and build boxdumper
https://github.com/l-smash/l-smash.git
3. dump the box of crc change mp4 and diff -u
[lmwang@vpn ffmpeg]$ ../l-smash/cli/boxdumper --box audio_key/non-empty-moov-no-elst.mp4 > audio_key/non-empty-moov-no-elst.log
[lmwang@vpn ffmpeg]$ ../l-smash/cli/boxdumper --box audio_old/non-empty-moov-no-elst.mp4 > audio_old/non-empty-moov-no-elst.log
[lmwang@vpn ffmpeg]$ diff -u audio_key/non-empty-moov-no-elst.log audio_old/non-empty-moov-no-elst.log
- default_sample_flags = 0x02000000
- independent
- sync sample
+ default_sample_flags = 0x01010000
+ dependent
+ non-sync sample
4. have checked the change of crc are caused by default_sample_flags
non-empty-moov.mp4, non-empty-moov-elst.mp4,
non-empty-moov-no-elst.mp4, empty-moov.mp4, delay-moov-content.mp4,
empty-moov-second-frag.mp4, empty-moov-second-frag-discont.mp4,
delay-moov-second-frag-discont.mp4, delay-moov-elst-second-frag.mp4
etc
5 For subtitle, it'll effect for tests/ref/fate/binsub-movtextenc and
tests/ref/fate/sub2video, that's expecting result for the subtitle is
marked as keyframe. Below is the checking result of binsub-movtextenc:
[lmwang@vpn ffmpeg]$ ./ffmpeg -i ../fate-suite/sub/MovText_capability_tester.mp4 -map 0 -scodec mov_text -f mp4 -flags +bitexact -fflags +bitexact -movflags frag_keyframe+empty_moov audio_key/binsub-movtextenc.mp4
[lmwang@vpn ffmpeg]$ ./ffmpeg -i ../fate-suite/sub/MovText_capability_tester.mp4 -map 0 -scodec mov_text -f mp4 -flags +bitexact -fflags +bitexact -movflags frag_keyframe+empty_moov audio_old/binsub-movtextenc.mp4
[lmwang@vpn ffmpeg]$../l-smash/cli/boxdumper audio_key/binsub-movtextenc.mp4 > audio_key/binsub-movtextenc.log
[lmwang@vpn ffmpeg]$../l-smash/cli/boxdumper audio_old/binsub-movtextenc.mp4 > audio_old/binsub-movtextenc.log
[lmwang@vpn ffmpeg]$ diff -u audio_key/binsub-movtextenc.log audio_old/binsub-movtextenc.log
.... // the key difference is the flag for sync sample
- flags = 0x000701
+ flags = 0x000301
data-offset-present
sample-duration-present
sample-size-present
- sample-flags-present
sample_count = 6
- data_offset = 188
+ data_offset = 164
sample[0]
sample_duration = 1570000
sample_size = 21
- sample_flags = 0x02000000
- independent
- sync sample
- degradation_priority = 0
sample[1]
sample_duration = 510000
sample_size = 2
- sample_flags = 0x01010000
- dependent
- non-sync sample
- degradation_priority = 0
sample[2]
sample_duration = 1690000
sample_size = 9
- sample_flags = 0x02000000
- independent
- sync sample
- degradation_priority = 0
Suggested-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Suggested-by: Nicolas George <george@nsup.org>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
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double dot
fix ticket: 8625
and add testcase into url for double dot corner case
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
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and add fate-hlsenc for test all of the testcase
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
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