[Matthias247]

> The method the author proposes of creating buffers one after the other is a bizarre solution.

I used the same solution when I tried to perform realtime audio streaming from a daemon on an embedded device to a browser (which is probably even a more realistic use-case for a browser audio API than generatic sine waves). I basically stumbled over the same issues than the author: A deprecated ScriptProcessorNode and high-level APIs which don't help me (like the oscillator one).

In the end I opted for a very similar solution as the author: Whenenver I got enough samples through websocket (I encoded them simply as raw 16bit samples there) I created a BufferSource, copied all samples in there (with conversion to floating point), and enqueued the buffer for playback at the position where the last buffer finished.

I really didn't expect that to work well due to all the overhead of creating and copying buffers and due to the uncertainity whether the browser will switch between 2 buffers without missing samples. But surprisingly it worked and did the job. I included a buffering of 200ms, which means I only started playback 200ms to be able to receive more data in the background and have a little bit more time to append further buffers. I experimented a little bit with that number but can't remember how deep the lower limit was before getting dropouts regurarly. It definitely wasn't usable for low-latency playback.