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  1. Member
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    i got two sources

    1 is VC-1 movie with 24FPS BDinfo

    2 is AVC same movie but with 23.976 BDinfo

    now i wana add audio from 1st source to 2nd

    eac is tool that can change FPS mostly 25=>23.976 but how to from 24.000 to 23.976

    when i will use slowdown its working as 25=>23 even when source is 24.

    there is diffrence in audio after +- 45 minutes of movie and very pointless to use delay in Multiavc

    once i done it properl but audio from 90 minutes has been expanded to 180 yikes .
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  2. Member Cornucopia's Avatar
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    Forget FPS for a moment and check: are the lengths the same? And are you sure this is the IDENTICAL program (same edits, etc.)?

    If so (both ok), copy the stream from the 1st to 2nd. Done.
    If lengths same but not identical program, you cannot do this without re-editing, and maybe not even then.
    If program identical but not same length, demux stream 1, decode to LPCM wav, time compress/expand to match length, re-encode to appropriate codec, then mux into stream 2's file.

    Scott
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    audio from source was 24.000 FPS when movie was 23.976 after BDinfo didnt realized that after alltime fail synchro tests
    lenght was diffrent but it was matter like 1:47:58 1:48:08
    your last words arent help we are talkin about FPS change not compression and matching

    this helped me :
    eac3to.exe audioin audioout -24.000 -changeTo23.976 even in FAQ eac dont have it.

    anyway thanks for fast answer that should help for other users

    BD Rebuilder should contain also FPS change during reecoding.
    again thanks
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  4. Member Cornucopia's Avatar
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    No we are not. Audio does NOT EVER have a FPS. Only the video portion does. Audio has samplerate. In most cases, it is 44100Hz or 48000Hz.

    A change of 1001/1000 like you are suggesting would have given you 6484 seconds, not 6488. Overall, it would have been a LOT more accurate just to look at your target duration and TC/X to give you that duration.

    Scott
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    correct but we have to change speed of audio when we have movie 24fps and audio from 23 we need to change speed of audio slow down or speed up

    anyway as i said its easier to do that in Rebuilder if there someday will be such option

    or

    Multiavchd option slowdown/speedup audio.

    audio FPS is easier to recognize when some1 is asking " how to change audio fps" when we have 2 diffrent sources

    last time i was doin Curse of the Golden Flower too i got both Blu Ray and DVD so i extracted audio from DVD then eac 25 to 23.976
    synchro and finished . Many movies like that have dubbed audio and i hate that.

    Again thanks for answer and help
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  6. Member
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    Originally Posted by Cornucopia View Post
    No we are not. Audio does NOT EVER have a FPS. Only the video portion does. Audio has samplerate. In most cases, it is 44100Hz or 48000Hz.
    Scott
    Well, technically, audio does have frames per second. An AC3 frame lasts for 32ms, for DTS it's 10.67ms. A "sample" would be the equivalent of a pixel. But it's not really relevant to the discussion.
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  7. Member Cornucopia's Avatar
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    Not quite. Technically, AUDIO has a samplerate. Certain codecs store their compressed packets in "frames", and the fact that they get generated (and read) at a certain rate allows them to be calculated as "having a framerate", but this so-called framerate has NOTHING to do with the description (or quality) of the audio itself. It only has to do with the partitioning of grouped segments of audio for the decoder. And since the codecs' "framerate" is locked to the codec itself and it not adjustable, even mention of this kind of partitioning as "framerate" in the context of the OP's request to "change the audio framerate" is ludicrous.

    IOW, it is not the property of the audio to have a FPS, it is the property of a few particular data types' segmentation (which have no bearing on synchronization).

    The audio samplerate, which is equivalent in many ways to the visual pixel, is totally relevant to the discussion as one can just as easily change a header that signifies a samplerate in order to change the length or pitch of an (uncompressed, and a few compressed) audio file, which possibly will allow it to sync with the visual's length. This is analogous to the way one can change the header of an image's PAR in order to correct/coordinate the DAR.

    Similarly, just as one can do a resampling resize of pixels to a different resolution (up or down as necessary to match other images), one can do a resampling of the audio (up or down) can be done to mix/edit it smoothly with other audio tracks.

    You already know these things ndjamena, so I don't understand why you would throw out such an obvious red herring. The OP needs big-picture clarity & solutions, not tangential trivialities.

    Scott
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    Tangential trivialities, um, right. Oddly switching audio from 24fps to 24/1.001 makes perfect sense to me. He's saying the same thing about you that you're saying about me, just in different words.
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  9. Member Cornucopia's Avatar
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    Except there's nothing ODD about applying "pulldown" to samplerates, particularly in a production environment. It's done all the time, for the very reason of maintaining sync.

    Were you to ever apply your technicality & attempt to change the "fps" of any AC3 or DTS, you would render it unusable for everyone. That is quite a difference.

    Scott
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