Sound becomes distorted after ~15 seconds of use.

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putyo$onme
Posts: 9
Joined: Tue Aug 01, 2017 8:10 pm

Sound becomes distorted after ~15 seconds of use.

Post by putyo$onme »

I was able to record a couple tracks for one project. However, suddenly I have a problem where the sound becomes distorted and choppy during both playback and recording. If I restart the program, this problem is fixed, but if I adjust any settings on my tracks, the problem returns. It doesn't matter what I'm adjusting, or even if I am simply pressing buttons on my midi controller, the sound of everything from playback to input, is very choppy and distorted. What am I doing wrong? Thanks in advance :)
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Ian Craig
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Re: Sound becomes distorted after ~15 seconds of use.

Post by Ian Craig »

Hi, switch from using the ASIO subsystem to CoreAudio (WaveRT) and increase the latency of your sound device (the number of milliseconds of lagtime), or switch to ASIO if using Core Audio and increase latency settings. Read here
http://www.acoustica.com/mixcraft/v7/he ... ation.html
Mixcraft 9 Pro Studio (build 470) recording output using MRecorder
AMD Ryzen 8 Core 3.0 GHz (40 GB Ram) & Intel i9 11th Gen 3.5 GHz (64GB Ram),
Windows 10 Professional
Focusrite Scarlett 2i2 & 8i6
putyo$onme
Posts: 9
Joined: Tue Aug 01, 2017 8:10 pm

Re: Sound becomes distorted after ~15 seconds of use.

Post by putyo$onme »

I have found that the lowest I can get input latency under WaveRT is 3ms, which is unfortunately too noticeable to allow for playback + input synchronization. With ASIO it sounds much closer to instantaneous. Is this an unfortunate trade off or are there some options I could tweak to lower my latency further on WaveRT?
JonInc
Posts: 536
Joined: Sat Dec 05, 2009 2:30 pm
Location: East of Santa Monica

Re: Sound becomes distorted after ~15 seconds of use.

Post by JonInc »

Would help if we had some info on your setup....

Mixcraft version/Build #
Win 7/8/10? All the latest updates?
Your sound card/audio interface (Latest drivers?)

With ASIO, does increasing the buffer size get rid of the audio problems (Apart from the latency lag)?
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