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By default, web-audio-api doesn't play back the sound it generates. In fact, an AudioContext has no default output, and you need to give it a writable node stream to which it can write raw PCM audio. After creating an AudioContext, set its output stream like this : audioContext.outStream = writableStream.
Example: Playing back sound with node-speaker
This is probably the simplest way to play back audio. Install node-speaker with npm install speaker, then do something like this :
import{AudioContext}from'web-audio-api'importSpeakerfrom'speaker'constcontext=newAudioContextcontext.outStream=newSpeaker({channels: context.format.numberOfChannels,bitDepth: context.format.bitDepth,sampleRate: context.sampleRate})// Create some audio nodes here to make some noise ...
Example : playing back sound with aplay
Linux users can play back sound from web-audio-api by piping its output to aplay. For this, simply send the generated sound straight to stdout like this :
import{AudioContext}from'web-audio-api'constcontext=newAudioContext()context.outStream=process.stdout// Create some audio nodes here to make some noise ...
Then start your script, piping it to aplay like so :
node myScript.js | aplay -f cd
Example : creating an audio stream with icecast2
icecast is a open-source streaming server. It works great, and is very easy to setup. icecast accepts connections from different source clients which provide the sound to encode and stream. ices is a client for icecast which accepts raw PCM audio from its standard input, and you can send sound from web-audio-api to ices (which will send it to icecast) by simply doing :
Gibber is a great audiovisual live coding environment for the browser made by Charlie Roberts. For audio, it uses Web Audio API, so you can run it on web-audio-api. First install gibber with npm :
npm install gibber.audio.lib
Then to you can run the following test to see that everything works:
npm test gibber.audio.lib
Overall view of implementation
Each time you create an AudioNode (like for instance an AudioBufferSourceNode or a GainNode), it inherits from DspObject which is in charge of two things:
register schedule events with _schedule
compute the appropriate digital signal processing with _tick
Each time you connect an AudioNode using source.connect(destination, output, input) it connects the relevant AudioOutput instances of source node to the relevant AudioInput instance of the destination node.
To instantiate all of these AudioNode, you needed an overall AudioContext instance. This latter has a destination property (where the sound will flow out), instance of AudioDestinationNode, which inherits from AudioNode. The AudioContext instance keeps track of connections to the destination. When that happens, it triggers the audio loop, calling _tick infinitely on the destination, which will itself call _tick on its input ... and so forth go up on the whole audio graph.
Running the debugger
Right now everything runs in one process, so if you set a break point in your code, there's going to be a lot of buffer underflows, and you won't be able to debug anything.
One trick is to kill the AudioContext right before the break point, like this:
context[Symbol.dispose]()debugger
that way the audio loop is stopped, and you can inspect your objects in peace.