On 12/12/25 08:34, 5112 Ian via ffmpeg-user wrote:
With these parameters, the transcoding latency and playback are normal.
However, when RTMP becomes unstable—such as during network instability,
jitter, or when the stream reconnects—the transcoding sometimes breaks and
playback fails. The log shows:
[aost#0:1/aac @ 0x6036eb9941c0] [warning] Non-monotonic DTS; previous:
88537152, current: 1209813; changing to 88537152. This may result in
incorrect timestamps in the output file.
With network instability, you need some syncing/buffering. I think you
should try aresample:
https://ffmpeg.org/ffmpeg-filters.html#aresample-1
"This filter is also able to stretch/squeeze the audio data to make it
match the timestamps or to inject silence / cut out audio to make it
match the timestamps, do a combination of both or do neither."
This should cope with audio pts discontinuities, typically inserting
silence when audio packets are lost.
Nicolas
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