Dale Curtis 460132c998 lavf/mp3dec: don't adjust start time; packets are not adjusted. ... 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>		2020-05-27 10:22:17 +02:00
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api	Stop hardcoding align=32 in av_frame_get_buffer() calls.	2020-05-22 14:38:57 +02:00
checkasm	checkasm/sw_scale: Fix stack-buffer-overflow	2020-05-20 23:18:50 +02:00
dnn	dnn-layer-mathbinary-test: add unit test for minimum	2020-05-08 15:22:44 +08:00
fate	fate: add tests for h264 and vp9 video enc parameters export	2020-05-25 11:59:45 +02:00
filtergraphs	…
ref	lavf/mp3dec: don't adjust start time; packets are not adjusted.	2020-05-27 10:22:17 +02:00
.gitignore	…
Makefile	fate: add tests for h264 and vp9 video enc parameters export	2020-05-25 11:59:45 +02:00
audiogen.c	…
audiomatch.c	…
base64.c	…
copycooker.sh	…
extended.ffconcat	…
fate-run.sh	fate: add tests for h264 and vp9 video enc parameters export	2020-05-25 11:59:45 +02:00
fate-valgrind.supp	…
fate.sh	…
md5.sh	…
refcmp-metadata.awk	…
reference.pnm	…
rotozoom.c	…
simple1.ffconcat	…
simple2.ffconcat	…
test.ffmeta	…
tiny_psnr.c	…
tiny_ssim.c	…
utils.c	…
videogen.c	…