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author | Justin Ruggles <justin.ruggles@gmail.com> | 2012-01-11 10:22:47 -0500 |
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committer | Justin Ruggles <justin.ruggles@gmail.com> | 2012-03-03 17:03:27 -0500 |
commit | ea289186f04fa8b9f2cc3bc1aba1cb239ab56f47 (patch) | |
tree | cb64dac5deff6614f0645954c065268fffeee7bb /libavformat/asfdec.c | |
parent | 01be6fa926dc3de593756ffd1e09f9523be5fd00 (diff) | |
download | ffmpeg-ea289186f04fa8b9f2cc3bc1aba1cb239ab56f47.tar.gz |
ea: fix audio pts
The time base is 1 / sample_rate, not 90000.
Several more codecs encode the sample count in the first 4 bytes of the
chunk, so we set the durations accordingly. Also, we can set start_time and
packet duration instead of keeping track of the sample count in the demuxer.
Diffstat (limited to 'libavformat/asfdec.c')
0 files changed, 0 insertions, 0 deletions