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Audio/Video desync over time when audio is PCM 11.025kHz

stormy1777stormy1777
edited November 2010 in Support
Hi...

Using v2.6 r294 capturing 1024x768 (quad core no other activity on system) and noticed that audio @ 44kHz or "Use MCI Recording" produces huge audio portion of resulting file (e.g. 80% or so for audio, rest is video). Audio is minimal in my captures (no need to spend 80% on it), so figured to reduce it to PCM 11.025kHz (mono or stereo).

Capture and audio is great, however, over time, it gets out of sync to the video, the video comes first, then the audio!!

I do not want to re-encode the audio of the resulting AVI, nor do I want to spend huge portion of the AVI file on audio.

I tried all possible framerates 100/10/100, 40/25, any values I tried (all multipied to 1000) still produce a video that's out of sync to audio, even the 'auto adjust'.

The only workaround somewhat usable was to fix the framerate using virtualdub as described several times in this forum and blog:

http://camstudio.org/blog/audiovideo-desynch-workaround

In one case my framerate was "25" and VD suggested 24.831 and indeed the resulting AVI was in sync.

Is there any way to capture with reduced size of audio (11.025kHz) and keep it in sync? This sync issue doesn't occur with "use MCI recording" or @44kHz.

Tried the mp3 audio, but no matter what I always get PCM audio, which is good enough, only if it was in sync :)

Thanks much for any new ideas.

Comments

  • stormy1777stormy1777
    I tried other frame rate combinations, like, capture every 45 playback rate 22, that doesn't multiple to 1000, but it gives less skew. On a 1hr file virtual dub wanted to correct from "22" to "22.072", once that correction was made, the a/v are in sync.

    Through trial and error, found other combinations like that (e.g. 41/24, 43/23, 45/22, 47/21), but nothing gave perfect sync, always VD correction was needed.

    this only happens with 11.025kHz audio.
  • mltmlt
    It is probably the problem of how audio and video are muxed together after the capture. As far as I remember somewhere in r302 or so I added for my personal use an option Program.MergeAV so that CS can leave audio & video untouched for later merge in other software.

    I didn't try 11khz, but 22khz worked fine for me. MP3 _IS_ broken.

    Are you sure it is real desync? I mean if you seek it to some place towards the end & start playback from there, is it still out of sync? I remember some other problems while working with VirtualDub's frameserver and AviSynth, but those artifacts were purely for playback, and result was okay after ffmpeg re-encoding.
  • stormy1777stormy1777
    Yes, it is out of sync as checked using VLC, BSPlayer, and Virtual dub, both by playing, or seeking to end of file doing the abcdef check.

    I figured mp3 was broken, hence the attempt for smaller sampling on PCM, otherwise files are huge for no real reason.

    Have to use 16bits since 8bit produces clipping and hissing. Not using stereo since that produces double the size audio, also higher kHz produces larger file, so for me, 11kHz@16bit mono is the way to go (until mp3 is fixed). I produce hours of recordings, and that translates into huge files if that is not done, or extra work to reencode audio.

    as stated above, 44kHz works OK, also 'use MCI recording' which bypass anything u have above, also works well. I have not tried 22khz, but it's not relevant at this point..

    I also tried to interleave at different values, no change.. this can even be seen on a small file, like 5minutes.
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