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Thread: Help! Need fix for glitch every 96 samples.

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    Help! Need fix for glitch every 96 samples.

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    Help, please!

    I need help fixing these audio files. They were recorded at 48khz, 24bit through a Focusrite liquid 56 interface. The ASIO buffer was set at 96.

    This has never been an issue before and I was monitoring direct (not through the DAW) so there was no apparent distortion. There are 4, 20-25 minute podcasts. 2 microphones in the same room.

    As you will hear from this example, there is a crackle/distortion throughout, which gets more apparent as the signal gets louder.

    When you inspect the file, there is a single sample out of place every 96 samples. If I select the area (adjacent samples) and use a repair tool in Audacity, or SoundForge etc. I've had good success in "fixing" it. (much better than any decrackle I've tried)

    I can't find a tool that will do just this task. It seems that all "declick/decrackle" plugins "listen" for the crackle and then try to remove it. But if I could tell them that it happens every 96 samples then I'm sure it would work better. If you can find a tool that does this I would greatly appreciate.

    I've considered scripting. I don't have access to soundforge .. just a demo, but I understand you can do some scripting with it. Apparently if you have some programming skills you can do scripting in Audacity. (requires a recompile of the software and then knowledge of python) I created a keyboard macro that would select the bits, apply the repair tool, then move ahead 96 samples, then repeat. It works for a while but the software gets bogged down with too many commands after a while. And it is very slow.

    Re-recording is not an option. The whole project would be scrapped if it can't be repaired. If you can help, please download the attached file and then contatct me. Thanks in advance!

    https://www.dropbox.com/s/bymb0lh7kv...Audio%2007.wav

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    Sorry I can't help with the problem at hand, but as far as keeping the problem from happening again: I would suspect the last thing updated. If you recently installed a driver, try Roll Back Driver (in device manager). If you had an upgrade to your DAW, try reinstalling an older version. If you just upgraded your ASIO, roll it back, etc. See what keeps the noise from happening.
    You may have had something else running in the background that ran your computer into processor starve. Buffer that low is tight on some processors, and if you had anything extra running it would cause a rhythmic glitch like you describe.
    Sorry about your project, dude.
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    Also, since you're monitoring directly via your interface, you could get away with a larger ASIO buffer. The added latency wouldn't affect your direct monitoring, and it would ease the strain on your CPU. That might alleviate the problem in the future.

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    Thanks for the response. It isn't a problem with drivers or anything. At least I can't reproduce the effect, although it has happened maybe once before.. I just caught it in time. The nature of the podcast session was that we didn't listen back on anything until it was too late. It may be that I had another project at 44.1 open before and when I switched to 48 for the podcast session, the ASIO driver somehow got F#$%d. I have a core i7 4770 which has no problem with the 96 sample buffer most of the time. I can generally record 16+ simultaneous channels with some effects/plugins and a total latency of under 8ms without more than 20% cpu. This was just an unfortunate time to not just record a minute and check that everything was ok. We had done so many of these in the past that it just seemed to be going as usual.

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