BtbN changed the topic of #ffmpeg to: Welcome to the FFmpeg USER support channel | Development channel: #ffmpeg-devel | Bug reports: https://ffmpeg.org/bugreports.html | Wiki: https://trac.ffmpeg.org/ | This channel is publically logged | FFmpeg 7.0 is released
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<aaabbb>
is there anything like "silenceremove" but for live streaming? with silenceremove the very instant the audio falls under the threshold, the audio stops. i want it to keep passing audio for a configurable number of seconds
<aaabbb>
i thought silenceremove=stop_periods=-1:stop_duration=2 but that only affects the very end of the audio stream it appears
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<winshixvxv>
h
<winshixvxv>
join #irssi
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<SuRGeoNix>
Hi Guys, a question about streams that the codec parameters changing... can you identify this change during demuxing or only during decoding? And is it possible also to have a change in the codec id or only for e.g. width/height/pixelformat?
<SuRGeoNix>
One more question, do the keyframes make sense only for video or you could have non-keyframes also for audio/data/subs streams?
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<aaabbb>
SuRGeoNix: keyframes are a video thing
<aaabbb>
when ffprobe is parsing audio packets it'll call each frame a keyframe iirc
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<blinux>
This may be more of a distro issue, but I think my ffmpeg package doesn't have libx264 installed, even though my OpenSUSE distro have libx264 installed. ffmpeg is using libopenh264, which is not good quality.
<blinux>
Or, I mean not compiled for use of libx264
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<Cracki>
find out who maintains that distro's package of ffmpeg?
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<blinux>
perhaps, I'm also trying handbrake, though it gave me a gstreamer error trying to preview with x264 10-bit. The banding problem I'm experiencing may be lack of 10-bit. If I know how to enable 10-bit on mpeg-4, then that might be helpful.
<microchip_>
blinux: add a bit of noise to the encode. Will reduce banding a lot if you can't do 10 bit
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<whatsupdoc>
Hi I am trying to use ffmpeg to merge multiple videos and it won't work
<whatsupdoc>
The video works fine, but the audio is scrambled. It's like it's fast-forwarded and high pitched and then it disappears
<whatsupdoc>
I am using this command: ffmpeg -f concat -safe 0 -i input.txt -c copy output.mp4
<whatsupdoc>
[aost#0:1/copy @ 0x142610220] Non-monotonic DTS; previous: 4095, current: 3203; changing to 4096. This may result in incorrect timestamps in the output file.
<whatsupdoc>
I also tried this command: ffmpeg -f concat -safe 0 -i input.txt -c:v copy -c:a aac -ac 2 -ar 48000 output.mp4
<whatsupdoc>
But this gives me a lot of these errors:
<whatsupdoc>
[aac @ 0x142e10050] Number of scalefactor bands in group (51) exceeds limit (49).
<whatsupdoc>
[aist#0:1/aac @ 0x142e05340] [dec:aac @ 0x142e0faa0] Error submitting packet to decoder: Invalid data found when processing input
<whatsupdoc>
And the result is the same
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<CounterPillow>
stream copying when concatenating will not work under most circumstances
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