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  1. Member
    Join Date
    Nov 2005
    Location
    Hungary
    Search Comp PM
    Hello,

    I don't know where I make a mistake. I have used the following equipment and tools:

    - Sony DVR-240 video camera and Adobe Rremier Pro to capture
    - freeEnc, QuEnc, TMPGEnc Plus Trial to encode
    - VirtualDub to save the audio stream
    - GUI for dvdauthor to authore
    - VLC Media Player to check the result

    I would have recorded my VHS video to DVD. Therfore I made the followings:

    - I captured the film through the camera (as a video card) with
    the Adobe Premier. I got 20GB DV AVI files. It is probably Type2
    (Gspot displays this information: Type: OpenDML AVI, FOURCC:dvsd)
    Length: 1:31 hour
    - After that I tried to encode it different ways:

    - with QuEnc, freeEnc, TMPGEnc
    - I saved tha audio in wav format and encoded video and audio
    separetely.

    (My video system is PAL)

    But nothing helped me. Namely the audo end the video stream
    are not in same step increasingly at the end of the film (only sensiby the second part of the film). But

    - when I play the avi file it seems to correct (but the mpeg2 is wrong)
    - when I split the avi file to more parts and they are encoded particularly
    the mpeg2 parts are good.
    - when I save the wav file and it's encoded separately, it's length is
    same as video stream has. And inspite of this the final
    result is same: after The DVD authoring the voice was drifted increasingly

    I would be grateful for some help.

    Regards,
    D.A.
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  2. I have no experience with DV at all. But from MPEG it is known that frame drops during capturing can cause such an behavior. But it could also be a problem with the multiplexer (mplex is not the best). Just as a first check:
    - when I split the avi file to more parts and they are encoded particularly the mpeg2 parts are good.
    - when I save the wav file and it's encoded separately, it's length is
    same as video stream has.
    So at this stage, you have a fitting audio and video stream?
    Then you could try to multiplex the streams with another muxer (imago, bbmpeg or muxman). If the result is ok, you can use the 'premuxed' files in GfD.
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  3. Member rkr1958's Avatar
    Join Date
    Feb 2002
    Location
    Huntsville, AL, USA
    Search Comp PM
    If your audio and video start in sync but drift apart over time then you can usually fix that with Goldwave (30-day free trial) timewarp function. I have to do this all the time. Often times this drift is caused by a slight bit error rate in your audio capture. See ... https://www.videohelp.com/forum/viewtopic.php?p=1178036#1178036

    Goldwave will allow you to increase or decrease the length of the raw audio to match that of the video and, if necessary and desired, resample the audio to 48-KHz. Again, something I do all the time.
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  4. Member
    Join Date
    Nov 2005
    Location
    Hungary
    Search Comp PM
    Originally Posted by rkr1958
    If your audio and video start in sync but drift apart over time then you can usually fix that with Goldwave (30-day free trial) timewarp function. I have to do this all the time. Often times this drift is caused by a slight bit error rate in your audio capture. See ... https://www.videohelp.com/forum/viewtopic.php?p=1178036#1178036

    Goldwave will allow you to increase or decrease the length of the raw audio to match that of the video and, if necessary and desired, resample the audio to 48-KHz. Again, something I do all the time.
    Thank you for the advice, I will see this program. But I've been
    interested in why instead of how. Thanks for link, I started
    in this thread, but it seems to me this is a big jungle, and anybody who
    stumble across this problem only experiments and nobody knows any
    concrete thing.

    You wrote that tha audio sample rate sometimes is 47997 instead of
    48000. So that the lag time is about 0.5 sec in one hour. My
    experience is similar but there is a contradiction. Originally I captured the
    film with 32 kHz sample rate, and after that I saved the WAV file with
    VirtualDub, resampled to 48khz and encoded to mpeg1 audio stream.
    After multiplexing the annoying effect was remained, the audio was out of.

    Does anybody know why this effect exist?

    Thanks,
    d.a.
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