Testing the Web Audio API

Testing the Web Audio API. The current setup is not suitable for live coding, so there is a break on each code change that ideally should happen after each bar loop completion.

I can tell you that this is not suitable for audio applications, but can be useful for now, for audio fx for web apps and games. Of course you’ll find web applications done on the browser, but there’s no way they’d compete with a proper real-time audio application such as Ableton, Cubase, etc - I shouldn’t even have had to point this out…

Maybe in 10 years with WebAssembly and better soundcards for the general population computer systems (such as mobile, laptops and tablets) this might be reconsiderate. Latency is a major problem. Audio recording might have > 100ms latency, which requires the user to either determinate the latency by himself if the application has a settings option for that, or a computation by the app to determinate how long is the latency - this is not trivial, as it’d require the user to go through a process where it’d record some audio and through a system of loopback have the computation for the recording time and playback difference. This can also be done through generating a signal automatically, but would require the user to place the mic recorder or earplugs onto the computer mic recorder (which is an horrible user experience).

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