ChanServ 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 7.1.1 is released
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<galad> intrac: it's full of out of specs mp4 out there, so even if there is a proper way to signal format changes by using multiple sample descriptions in mp4, often you find just random data put into tracks hoping the player is broken enough to ignore the mp4 container and to just read the bitsream
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<intrac> galad: yeah, I wish people wouldn't drift away from standards
<intrac> I have some variable framerate video, how can I force ffmpeg to generate fixed fps output and dup/drop frames as needed to achieve it?
<intrac> I tried -vsync 1 and -r 30000/1001 just before the output filename
<intrac> but mediainfo still reports the framerate as 'variable'
<intrac> I've also tried a -vf fps=30000/1001
<furq> is that with a new mediainfo
<furq> that was broken for a long time
<intrac> furq: it's mediainfo version 24.01
<intrac> can I check with ffprobe as an alternative?
<furq> yeah
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<furq> ffprobe -select_streams v:0 -show_entries frame=pts_time -of csv=p=0 foo.mkv
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<furq> i think that mediainfo is new enough but i'd trust ffprobe more
<furq> although it is obviously much more effort
<intrac> furq: that last command outputs the basic metadata, video/audio details (but not vfr/cfr) and then a lot of blank lines
<furq> it should output the timestamp for every frame
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<furq> also is this mpegts
<intrac> it seems to output just a lot of white lines
<intrac> someone suggested: ffmpeg -i input.mp4 -vf vfrdet -an -f null -
<furq> oh yeah that's a lot easier
<intrac> it outputs "Application provided invalid, non monotonically increasing dts to muxer in stream 0: 4 >= 4"
<intrac> also "[Parsed_vfrdet_0 @ 0x55c00af2ad80] VFR:0.734001 (1319/478) min: 12 max: 55 avg: 33"
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<intrac> so that seems to be saying VFR
<intrac> hrm
<furq> yeah that's not great
<intrac> my encode command is: ffmpeg -i input.mp4 -c:v libx264 -c:a aac -filter:v fps=30000/1001 -vsync cfr -t 60 -r 30000/1001 output.mp4
<furq> i take it you tried with -vf and -r individually already
<furq> using both probably doesn't do anything good
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<intrac> I tried with -r on its own, but not -vf fps
<furq> either should work
<intrac> neither works :(
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<intrac> will look at this again later
<intrac> thanks for the suggestions
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<intrac> furq: did you have any further ideas on how to get constant fps? :|
<intrac> (or anyone else)
<intrac> also, does this have any meaning with the output of fvrdet:
<intrac> Application provided invalid, non monotonically increasing dts to muxer in stream 0: 4 >= 4
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<intrac> it seems there are several different ways to generate dts/pts timestamps. maybe I need to set something there.
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<kepstin> also keep in mind that mkv doesn't _actually_ support CFR with any framerate that's not a power of 10
<kepstin> (although ffmpeg and other tools have heuristics to guess the actual framerate ignoring the timing jitter)
<intrac> kepstin: doh. that seems it might be the issue.
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<intrac> I just tried an encode to .mp4 and vfrdet reports CFR (VFR:0.000000)
<intrac> thanks!
<intrac> but now I'll see if this plays any better on my standalone media box
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<intrac> nope, still stutters :/
<intrac> ok, I think it relates to my earlier processing and resampling of the audio
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<intrac> it seems the stuttering issue is caused by the inclusion of "asetpts=1/0.91875*PTS"
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<intrac> but without that, the audio/video drift when asetrate=48000 is used - which is needed to correct the pitch error
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<kepstin> you need to resample the audio after running asetpts to make the audio continuous again
<kepstin> actually, using asetpts to change audio speed isn't generally a good idea
<kepstin> since that changes the timestamp on packets, causing there either to be overlaps or gaps
<johnjaye> i wanted to mix in 2 audio tracks into an existing video but at different time stamps
<johnjaye> is there a more practical way to do that than measuring the time, adding silence, concatting the 2 audios together then mixing with amix?
<johnjaye> i thought -itsoffset might work but it didn't
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<johnjaye> hmm i just added the silence
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<intrac_> It looks like the audio codec is a factor. If I use MP3 audio instead of AAC, the stuttering on my standalone player goes away
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