A javascript library to encode the output of Web Audio API nodes in Ogg Opus or WAV format using WebAssembly. Audio encoded and decoded using libopus v1.2.1. Audio resampling is performed by speexDSP 1.2RC3.
Encoded and muxed audio will be returned as typedArray in dataAvailable event.
For legacy asm.js, please use version 1.2.0
The Recorder object is available in the global namespace and supports CommonJS and AMD imports.
var rec = new Recorder([config]);Creates a recorder instance.
- config - An optional configuration object (see config section below)
- bufferLength - (optional) The length of the buffer that the internal JavaScriptNode uses to capture the audio. Can be tweaked if experiencing performance issues. Defaults to
4096. - encoderApplication - (optional) Supported values are:
2048- Voice,2049- Full Band Audio,2051- Restricted Low Delay. Defaults to2049. - encoderBitRate - (optional) Target bitrate in bits/sec. The encoder selects an application-specific default when this is not specified.
- encoderComplexity - (optional) Value between 0 and 10 which determines latency and processing for encoding.
0is fastest with lowest complexity.10is slowest with highest complexity. The encoder selects a default when this is not specified. - encoderFrameSize - (optional) Specifies the frame size in ms used for encoding. Defaults to
20. - encoderPath - (optional) Path to
encoderWorker.min.jsorwaveWorker.min.jsworker script. Defaults toencoderWorker.min.js - encoderSampleRate - (optional) Specifies the sample rate to encode at. Defaults to
48000. Supported values are8000,12000,16000,24000or48000. - leaveStreamOpen - (optional) Keep the stream around when trying to
stoprecording, so you can re-startwithout re-initStream. Defaults tofalse. - maxBuffersPerPage - (optional) Maximum number of buffers to use before generating an Ogg page. This can be used to lower the streaming latency. The lower the value the more overhead the ogg stream will incur. Defaults to
40. - mediaTrackConstraints - (optional) Object to specify media track constraints. Defaults to
true. - monitorGain - (optional) Sets the gain of the monitoring output. Gain is an a-weighted value between
0and1. Defaults to0 - numberOfChannels - (optional) The number of channels to record.
1= mono,2= stereo. Defaults to1. Maximum2channels are supported. - originalSampleRateOverride - (optional) Override the ogg opus 'input sample rate' field. Google Speech API requires this field to be
16000. - resampleQuality - (optional) Value between 0 and 10 which determines latency and processing for resampling.
0is fastest with lowest quality.10is slowest with highest quality. Defaults to3. - streamPages - (optional)
dataAvailableevent will fire after each encoded page. Defaults tofalse.
- bufferLength - (optional) The length of the buffer that the internal JavaScriptNode uses to capture the audio. Can be tweaked if experiencing performance issues. Defaults to
4096. - encoderPath - (optional) Path to
encoderWorker.min.jsorwaveWorker.min.jsworker script. Defaults toencoderWorker.min.js - leaveStreamOpen - (optional) Keep the stream around when trying to
stoprecording, so you can re-startwithout re-initStream. Defaults tofalse. - mediaTrackConstraints - (optional) Object to specify media track constraints. Defaults to
true. - monitorGain - (optional) Sets the gain of the monitoring output. Gain is an a-weighted value between
0and1. Defaults to0 - numberOfChannels - (optional) The number of channels to record.
1= mono,2= stereo. Defaults to1. Maximum2channels are supported. - wavBitDepth - (optional) Desired bit depth of the WAV file. Defaults to
16. Supported values are8,16,24and32bits per sample.
rec.addEventListener( type, listener[, useCapture] )addEventListener will add an event listener to the event target. Available events are streamError, streamReady, dataAvailable, start, pause, resume and stop.
rec.initStream()initStream will request the user for permission to access the the audio stream and raise streamReady or streamError.
Returns a Promise which resolves the audio stream when it is ready.
rec.pause()pause will keep the stream and monitoring alive, but will not be recording the buffers. Will raise the pause event. Subsequent calls to resume will add to the current recording.
rec.removeEventListener( type, listener[, useCapture] )removeEventListener will remove an event listener from the event target.
rec.resume()resume will resume the recording if paused. Will raise the resume event.
rec.setMonitorGain( gain )setMonitorGain will set the volume on what will be passed to the monitor. Monitor level does not affect the recording volume. Gain is an a-weighted value between 0 and 1.
rec.start()start will initalize the worker and begin capturing audio if the audio stream is ready. Will raise the start event when started.
rec.stop()stop will cease capturing audio and disable the monitoring and mic input stream. Will request the recorded data and then terminate the worker once the final data has been published. Will raise the stop event when stopped.
rec.clearStream()clearStream will stop and delete the stream got from initStream, you will only ever call this manually if you have config.leaveStreamOpen set to true.
Recorder.isRecordingSupported()Returns a truthy value indicating if the browser supports recording.
Supported:
- Chrome v58
- Firefox v53
- Microsoft Edge v41
- Opera v44
- macOS Safari v11
- iOS Safari v11
Unsupported:
- IE 11 and below
- iOS 11 Chrome
- Firefox does not support sample rates above 48000Hz: https://bugzilla.mozilla.org/show_bug.cgi?id=1124981
- macOS Safari v11 does not sample rates above 44100Hz
- macOS Safari v11 native opus playback not yet supported
- iOS Safari v11 native opus playback not yet supported
- Microsoft Edge native opus playback not yet supported
Prebuilt sources are included in the dist folder. However below are instructions if you want to build them yourself. Opus and speex are compiled without SIMD optimizations. Performace is significantly worse with SIMD optimizations enabled.
Mac: Install autotools using MacPorts
port install automake autoconf libtool pkgconfigWindows: Install autotools using MSYS2
pacman -S make autoconf automake libtool pkgconfigInstall npm dependencies:
npm installcheckout, compile and create the dist from sources:
npm run makeRunning the unit tests:
npm testClean the dist folder and git submodules:
make cleanThe required files to record audio to ogg/opus are dist/recorder.min.js and dist/encoderWorker.min.js. Optionally dist/decoderWorker.min.js will help decode ogg/opus files and dist/waveWorker.min.js is a helper to transform floating point PCM data into wave/pcm. The source files src/encoderWorker.js and src/decoderWorker.js do not work without building process; it will produce an error ReferenceError: _malloc is not defined. You need to either use compiled file in dist/ folder or build by yourself.