CPU Resources + DSP Usage
By default, Mixbus requires significantly more CPU resources than a typical DAW, because it is emulating the operation of an analog console.Nevertheless, a modern 2GHz CPU should be able to handle dozens of tracks and several plugins, with reasonable settings. Note that, because of the way Mixbus works, it does not increase the cpu to enable all the EQs and compressors; the DSP load for that processing is already pre-allocated when you run Mixbus. This means that if Mixbus is running smoothly, you won't accidentally overload your system if you enable all the channelstrip features while you are mixing.
The typical methods to minimize the CPU load are: increase the "buffer size" in the Audio Setup dialog; use a lower sample rate, reduce the number of tracks, or reduce the number of plugins in your session. And of course using a faster computer, with more CPU cores, is desirable.
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, Mixbus has to be alerted by your OS, 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 lack of audio (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. There's plenty of CPU power to get the work done if we are woken-up in time; but if the OS is slow to wake us up, then the desktop CPU usage can look very low, but you are still encountering glitches.
If you have time to watch this (very good, but very long) video, you can learn more details: https://www.youtube.com/watch?v=GUsLLEkswzE
If you Google-search for "computer audio performance", you will find lots of resources online. Here are some of the best links we've found. Note that this information is provided "at your own risk":
Mac performance optimizations:
Windows performance optimizations: