I'm doing a project where I have a server offering up audio via WebSocket to the browser. The browser is then using WebSocket to capture the audio blocks and play them out via the Web Audio API.
I create an AudioContext which then has a predefined sample rate of X to play back this audio on the local hardware. Fair enough. However, the sample rate of the audio I am getting from the server is Y (which is not necessarily the same value as X, but I do know what it is via the WebSocket protocol I have defined).
After doing a lot of digging I can't seem to see a way in which the input audio would be resampled via Web Audio itself. I can't imagine this is an uncommon problem but since I'm not quite an expert JavaScript coder I have yet to discern the solution.
Just assigning the samples to getChannelData only really works when X==Y. Otherwise, the audio skips and is at the wrong pitch. Using decodeAudioData allows me access to do the resampling, it seems, but I don't want to have to write my own resampler.
Thoughts?
Confused, are you needing to use decodeAudioData?
If you just have the raw PCM data transferred, you just create an AudioBuffer in your local AudioContext at the served buffer's sampleRate (i.e. you set the third parameter of http://webaudio.github.io/web-audio-api/#widl-AudioContext-createBuffer-AudioBuffer-unsigned-long-numberOfChannels-unsigned-long-length-float-sampleRate based on the data you're getting), and call start() with an appropriate time (and a playbackRate of 1). You WILL get minor glitches at block boundaries, but only minor ones.
There's an active discussion on this, watch issue: https://github.com/WebAudio/web-audio-api/issues/118.