I am building a web app which allows users to listen to a loop of instrumental music and then record vocals on top. This is all working using Recorder.js however there are a few problems:
- There is latency with recording, so this needs to be set by the user before pressing record.
- The exported loop is not always the same length as the sample rate might not match the time needed exactly
However since then I went back to the drawing board and asked: What's best for the user?. This gave me a new set of requirements:
- Backing loop plays continuously in the background
- Recording starts and stops whenever the user chooses
- Recording then plays back in sync with loop (the dead time between loops is automatically filled with blank audio)
- User can slide an offset slider to adjust for small timing issues with latency
- User can select which portion of the recording to save (same length as original backing loop)
Here's a diagram of how that would look:
Logic I have so far:
// backing loop
a.startTime = 5
a.duration = 10
a.loop = true
// recording
b.startTime = 22.5
b.duration = 15
b.loop = false
// fill blank space + loop
fill = a.duration - (b.duration % a.duration) // 5
c = b.buffers + (fill * blankBuffers)
c.startTime = (context.currentTime - a.startTime) % a.duration
c.duration = 20
c.loop = true
// user corrects timing offset
c.startTime = ((context.currentTime - a.startTime) % a.duration) - offset
// user choose favourite loop
? this is where I start to lose the plot!
Here is an example of chopping the buffers sent from Recorder.js:
// shorten the length of buffers
start = context.sampleRate * 2; // start at 2 seconds
end = context.sampleRate * 3; // end at 3 seconds
buffers.push(buffers.subarray(start, end));
And more example code from the previous versions i've been working on: https://github.com/mattdiamond/Recorderjs/issues/105
Any help in working out how to slice the buffers for the exported loop or improving this logic would be greatly appreciated!
UPDATE
Using this example I was able to find out how to insert blank space into the recording:
http://mdn.github.io/audio-buffer/
I've now managed to almost replicate the functionality I need, however the white noise seems off. Is there a miscalculation somewhere?