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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<johnjaye> i don't mean for this to sound nitpicky, but why does .ass/.ssa format require a decimal point in lieu of a hundreths digit?
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<johnjaye> i.e. instead of 1:22:333:444 you can type 1:22:333.444 and it parses fine?
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<johnjaye> I donwloaded a specification document but the exact timecode format is only described as the former. but the libass option in ffmpeg apparently allows either
<johnjaye> I only noticed because ffmpeg spits out a bad spec error if you write 1:22:33 in a .ass file
<johnjaye> hmm. the libass github page doesn't have any specs. but it does have a rough guide that describes this format
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<welder> BtbN: I am still debugging the pulse cut issue. The culprit seems to be --enable-libx264 in Debian (it's not enabled by default), that's so odd! Do you have any clue why it may be so?
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<BtbN> libx264 got absolutely nothing to do with pulse
<BtbN> no idea how you would come to that conclusion
<welder> well I ran configure with the extra Debian flags - the issue appeared with adding this one
<welder> I know - that's unrealted to pulse...
<BtbN> That's virtually impossible to be the cause
<BtbN> those two don't even get linked into the same library
<BtbN> sounds to me like whatever test you are using just switched from some random other encoder to libx264, and that triggers the delay to show up
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<welder> In layman terms, what exactly triggers the delay?
<welder> omg, -v:c libx265 introduces 1 second shorter audio, libx264 2 and a half, and mpeg4 does not cut anything!!!!!!!!
<welder> Why?!
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<welder> the video codec expects several frames before it starts the encoding process and that blocks the second input from being started?
<welder> But it does not explain why the audio is cut from the END, and not beginning (it's perfectly in sync)?
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<Dan55> I'm trying to segment a wav file to 5 parts
<Dan55> 4 parts
<Dan55> if I use absolute timestamps it works but if I use relative timestamps, the third part duration is wrong
<Dan55> ffmpeg -i test.wav -filter_complex asegment=timestamps="1000ms|2000ms|4000ms|[1][2][3][4]" -map "[1]" test1.wav -map "[2]" test2.wav -map "[3]" test3.wav -map "[4]" test4.wav
<Dan55> ffmpeg -i test.wav -filter_complex asegment=timestamps="1000ms|+1000ms|+1000ms|[1][2][3][4]" -map "[1]" test1.wav -map "[2]" test2.wav -map "[3]" test3.wav -map "[4]" test4.wav
<Dan55> is this a but or do I use  "+<timestamp>ms" wrong?
<Dan55> *is this a bug
<Dan55> docs say: "In all cases, prefixing an each segment with ’+’ will make it relative to the previous segment. "
<BtbN> welder: I think it's something along the lines of the avformat being opened, and then the encoder pipeline needs a few seconds to spin up
<BtbN> so by that time, the audio source already has built a buffer of data
<BtbN> And in the end, it seems like the audio source cuts off immediately, while the video one finishes
<furq> Dan55: shouldn't the last segment be +2000ms
<Dan55> furq: yes. the time in the example is wrong but it still doesn't work for me
<furq> how far out is it
<Dan55> furq: on a specific file I checked the first two segments where correct. The third gave me 212.300000ms instead of 272.140000ms
<Dan55> the 4th gave me empty file
<Dan55> using absolute timestamps gave me exact results
<furq> huh
<furq> i get an error with that
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<furq> asegment=timestamps=1|2|3 works as expected
<furq> so that seems busted
<Dan55> I'm testing on the first wav file from here: https://www2.cs.uic.edu/~i101/SoundFiles/
<Dan55> ffmpeg -i test.wav -filter_complex asegment=timestamps="1000ms|2000ms|3000ms[1][2][3][4]" -map "[1]" test1.wav -map "[2]" test2.wav -map "[3]" test3.wav -map "[4]" test4.wav
<furq> well it happens with a synthetic input so it's definitely nothing to do with the input file
<Dan55> ffmpeg -i test.wav -filter_complex asegment=timestamps="+1000ms|+1000ms|+1000ms[1][2][3][4]" -map "[1]" test1.wav -map "[2]" test2.wav -map "[3]" test3.wav -map "[4]" test4.wav
<Dan55> first command is ok. with the second command the third file has no duration
<furq> maybe worth a bug report
<Dan55> ok
<Dan55> another question
<Dan55> I have an opus file with gaps (silence)
<Dan55> I'm trying to convert it to wav with
<Dan55> ffmpeg -i test.opus -af aresample=async=1 test.wav
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<Dan55> sometimes the duration of the wav file is different than the opus file. Shouldn't the duration be exactly the same?
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<Dan55> furq: can you please give me the synthetic input command so I can submit a but report?
<furq> although bizarrely
<furq> but then the third output is only one second long
<furq> so maybe an off by one error somewhere
<Dan55> thanks
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<furq> that's with a build from today as well
<Dan55> furq: reported. thanks
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<johnjaye> maybe i'm being forgetful again.
<johnjaye> but why is -map "[0:a]" not allowed at the end of the ffmpeg command?
<johnjaye> i guess it has to be "defined" in the filtergraph?
<furq> [] refers to an output label
<furq> labels can't contain :
<johnjaye> i was told to use -map "0:a" instead
<furq> right
<furq> you don't need the quotes unless you have a very unusual shell
<johnjaye> ok. but for input labels [0:a] is fine
<johnjaye> i guess the syntax is still a bit subtle for me
<furq> input labels are just stream specifiers in brackets
<furq> it would maybe be clearer if link/output labels couldn't be valid stream specifiers
<johnjaye> i see
<furq> but 0, 1 etc are valid which isn't great
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<KillerWasp> hello
<KillerWasp> This time I come not to ask for help, but to say that I have already managed to save an mp4 with video and audio, and everything works perfectly.
<KillerWasp> I give thanks for the help received here.
<KillerWasp> I only have one question left regarding RTMP transmissions, is the data sent raw data that is stored in the AVFrame buffer?
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<welder> Can someone please reproduce this? https://0x0.st/s/OVUz2ZBcex5HE6EICSrp-g/XhiA.txt Is it the expected behavior?
<KillerWasp> mmmmm, ok, seem not all are fine... Something is wrong where i use it for twitch...
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<BtbN> rtmp is not uncompressed video/audio, if you mean that
<BtbN> it's typically h264+aac
<KillerWasp> BtbN: but what have in the data of RTMP? Because until now their protocol is a damn mystery, and I don't know what goes there.
<BtbN> What do you mean?
<BtbN> It just streams h264 video and aac audio to a remote server
<KillerWasp> BtbN: That its documentation does not explain anything about what data to transmit.
<furq> flv
<KillerWasp> lol, but what?
<BtbN> Whatever you send it?
<BtbN> I really don't understand what you mean
<KillerWasp> the content of AVFrame? i guessing.
<BtbN> h264 and aac data
<BtbN> again, it's not uncompressed
<KillerWasp> BtbN: you're saying that av_interleaved_write_frame compress the raw data from AVFrame to a compressed data using the codec of h264 and aac?
<BtbN> av_interleaved_write_frame, despite it names, takes packets.
<BtbN> *its name
<KillerWasp> takes packets and send to mp4 or stream of rtmp. But what happen with rtmp? what are inside of datas of video/audio in a rtmp?
<BtbN> If you want to know how flv and rtmp are muxed, you'll need to read the spec
<BtbN> but normally you don't need to care about that from a user level
<KillerWasp> BtbN: i already read, and don't say nothing, that's i'm ask here.
<BtbN> Well, I don't get what you're asking then
<BtbN> It sends whatever data you give it
<KillerWasp> but what data? i don't believe that i can put in the data any garbage for RTMP.
<BtbN> Well, a codec it supports.
<BtbN> technically, rtmp only supports flv, and then flv a couple codecs
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<KillerWasp> well, nothing is useful. So far I have not found any documentation that says, for the data section you have to store the raw data of 1000 audio samples in 2-channel planar style float, and for the videos also raw data of a size WxH with 32 bits per pixel RGBX, transparency is not supported. This does explain to you exactly what is in the data.
<KillerWasp> It doesn't even explain if the raw data is involved or if there is some type of compression involved.
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<another|> Is your question "What codecs does flv support?" ?
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<welder> When recording video and audio, is there some flag that can ensure that both streams are of equal length before exiting? So that when pressing "q" ffmpeg will wait for more audio if video is bit longer and vice versa?
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<fairway> everybody, this channel is not acknowledging huge bug i found in ffmpeg
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<Marth64> ?
<furq> they left
<furq> the bug is only huge enough to be worth 71 seconds
<furq> 21:28:33 ( fairway) i found a huge bug with ffmpeg
<furq> 21:29:04 ( fairway) when i do ffprobe.exe video.wmv with v4.2.1 it showed 29.97 fps but with latest version it's showingg 59.94
<furq> oh no!
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<KillerWasp> another|: When I said about flv? I'm talking about RTMP.
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<another|> KillerWasp: rtmp uses flv as container