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N
ffmpeg-0.6.3
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${ noResults }
Anton Khirnov
d85c6aba0c
Stop using InputStream.dts for generating missing timestamps for decoded frames, because it contains pre-decoding timestamps and there may be arbitrary amount of delay between input packets and output frames (e.g. dependent on the thread count when frame threading is used). It is also in AV_TIME_BASE (i.e. microseconds), which may introduce unnecessary rounding issues. New code maintains a timebase that is the inverse of the LCM of all the samplerates seen so far, and thus can accurately represent every audio sample. This timebase is used to generate missing timestamps after decoding. Changes the result of the following FATE tests * pcm_dvd-16-5.1-96000 * lavf-smjpeg * adpcm-ima-smjpeg In all of these the timestamps now better correspond to actual frame durations. |
2 years ago | |
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.. | ||
acodec | fftools/ffmpeg: propagate frame durations to packets when encoding | 2 years ago |
fate | fftools/ffmpeg: rework audio-decode timestamp handling | 2 years ago |
lavf | fftools/ffmpeg: rework audio-decode timestamp handling | 2 years ago |
lavf-fate | avformat/matroskaenc: support writing Dynamic HDR10+ packet side data | 2 years ago |
pixfmt | fate/pixfmt: test xyz12le | 3 years ago |
seek | lavf/dv: use a more granular timebase for audio | 2 years ago |
vsynth | tests: Add more j2k tests to cover a wider range of cases | 2 years ago |