There are no guarantees that all side data types have the same
representation on all platforms.
Tests that change output due to this:
id3v2-priv-remux, cover-art-mp3-id3v2-remux, gapless-mp3: SKIP_SAMPLES,
which is tested by fate-gapless-mp3-side-data
matroska-vp8-alpha-remux: MATROSKA_BLOCKADDITIONAL, which is tested by
remux itself (side data is written into output)
matroska-mastering-display-metadata: MASTERING_DISPLAY_METADATA and
CONTENT_LIGHT_LEVEL, which are tested by ffprobe invocation in the same
test
matroska-spherical-mono-remux: STEREO3D and SPHERICAL, which are tested
by ffprobe invocation in the same test
segment-mp4-to-ts: MPEGTS_STREAM_ID, which is tested by ts remuxing
tests
webm-webvtt-remux: WEBVTT_IDENTIFIER/SETTINGS, which is tested by the
ffprobe invocation in the same test
mxf-d10-user-comments: CPB_PROPERTIES, which is tested by mxf-probe-d10
mov-cover-image: SKIP_SAMPLES, which is tested for mov by
mov-aac-2048-priming
copy-trac3074: AUDIO_SERVICE_TYPE, which is tested by fate-hls-fmp4_ac3
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 460132c998.
7546ac2fee 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>
This should be more useful for users since numerical values for channel
layout can be confusing and unintuitive.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
Ideally this should be discarded by the demuxer but this is not
possible without fully parsing which would be then very similar
to this. The current ID3v1 discard code in the demuxer does not work
and will be removed in a subsequent commit
The discard code could be adjusted if needed to also discard tags at
other locations than the end or to limit this possibly to input
from the mp3 demuxer or even to move the discarding to the
decoder.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
"-usetoc 2" now invokes the generic seek and indexing mode. This mode
skips data until the seek target is reached, and this is exact. It also
makes gapless audio actually work if a seek past the start of the file
is involved.
Change the fate-gapless-mp3 test to use the new mode, and move the old
one to fate-gapless-mp3-toc (since the test forces use of the Xing TOC).
The new mode has a different result for the seek - this result is
actually correct.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Seeking to a negative time did not have the desired effect of seeking to
the next valid position (the file start). On the other hand, just
"-ss 0" will normally seek to a position higher than 0, because it adds
the start time of the file. (The start time is not 0 because the gapless
code skips a few samples from the start.)
Fix this by using the "-seek_timestamp 1" option, which makes "-ss 0" do
what you'd expect it would do.
Also put the -ss option at the right place, before -i. This actually
makes it seek, instead of something completely else. The ".out-3" test
is no different in the -usetoc 0/1 cases, because the seeking is
inaccurate (in both cases).
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
