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dd57723337
This is a mega-patch that was too hard to disentangle. Here's what it does: -- Create infrastructure around AudioNode::UpdateOutputEnded to detect when a node can no longer produce any output. When that becomes true, disconnect it from the AudioNode graph. -- Have AudioNode implement JSBindingFinalized to use as input in UpdateOutputEnded. -- Give every AudioNode a MediaStream, and give every connection a MediaInputPort. -- Actually play the audio that reaches the AudioContext's destination node. -- Force AudioContext to use the audio sample rate defined by MediaStreamGraph. -- Fix AudioBufferSourceNode's start and stop methods to possibly throw and take default 'when' parameters. -- Create an AudioNodeStream for AudioBufferSourceNode and give it a AudioBufferSourceNodeEngine that does what's needed. Set parameters for this engine in the start() and stop() methods. -- Create AudioBuffer::GetThreadSharedChannelsForRate, which is responsible for stealing the contents of any JS array buffers, and bundling them up into a thread-shared read-only buffer object which can be used as part of an AudioChunk. This method will also be responsible for resampling and caching as necessary.