Why does Mixbus require more CPU usage than my other DAW?

Because Mixbus is emulating the operation of an “analog console”, it requires significantly more CPU resources than a typical DAW. Mixbus’s EQs, compressors, and other features are always active. The benefit of this operation is that you can always confidently enable the channelstrip features without worry of overloading your system. Once the mixer is running, it does not increase the load when you operate the controls.

Although Mixbus requires a high CPU usage by default, it is actually quite efficient compared to other DAWs, once you’ve added plugins equivalent to the high-quality EQ, compression, bussing, tape saturation, limiting, metering, and the other built-in features of Mixbus.

How do I increase the number of tracks or plugins that I can use?

  • Increase the “buffer size” in the Audio Setup dialog
  • On Windows and Linux, you must take steps to optimize your system for audio ( see our Installation section )
  • Use high quality audio devices and drivers
  • And of course using a faster computer, with more CPU cores, is desirable.

General comments about computer audio performance:

The CPU meter on your computer averages the cpu usage over a very long period (perhaps one second).
In digital audio, the timing is much more sensitive. When the soundcard passes us a buffer, then we have to wake up, process the audio buffer, and return it to the soundcard before the soundcard needs to play it out. If we don’t wake up in time, or we don’t get finished in time, then you hear a “click” caused by the dropout (we call these xruns, short for over-run or under-run).

In the case of a 1024 buffer size, this has to be done within (1024/44100) 25ms, or about 1/40th of a second.

This is complicated by the fact that it’s the OS’s job to “wake us up” and tell us that the soundcard has some data for us. Some OS’s and drivers are better at this than others. But let’s say it takes 5ms before we are even alerted that some data is available. We now have 20ms left to process the audio. If it takes us 10ms to process the audio, then we are finished within 50% of the allotted time and that’s what we display in the meter. This is a very accurate indication of your computer’s ability to process audio.

This also explains why the selected buffersize is so critical to the DSP load. A 256-sample buffersize is about 6ms, so if your computer sometimes waits 5ms after receiving the soundcard interrupt before it “wakes us up”, we only have 1ms remaining to do our work.

For more information on factors impacting the performance of your computer please see this video:
CPU Performance vs. Real-Time Performance in Digital Audio Workstations


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