mirror of https://git.ffmpeg.org/ffmpeg.git
9c2c0c37f8
- Remove the 1024 cap on the number of samples, for high sample rate audio it was suboptimal, calculate the low neighbour power of two for the number of samples (audio blocks) instead. - Make the function work correctly also for non-pcm codecs by using the stream bitrate to estimate the target packet size. A previous version of this patch used av_get_audio_frame_duration2() the estimate the desired packet size, but for some codecs that returns the duration of a single audio frame regardless of frame_bytes. - Fallback to 4096/block_align*block_align if bitrate is not available. Signed-off-by: Marton Balint <cus@passwd.hu> |
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| acodec | ||
| fate | ||
| lavf | ||
| lavf-fate | ||
| pixfmt | ||
| seek | ||
| vsynth | ||