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Opus & Wave Recorder

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

Usage

Constructor

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)

Config options for OGG OPUS encoder

  • 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 to 2049.
  • 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. 0 is fastest with lowest complexity. 10 is 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.js or waveWorker.min.js worker script. Defaults to encoderWorker.min.js
  • encoderSampleRate - (optional) Specifies the sample rate to encode at. Defaults to 48000. Supported values are 8000, 12000, 16000, 24000 or 48000.
  • leaveStreamOpen - (optional) Keep the stream around when trying to stop recording, so you can re-start without re-initStream. Defaults to false.
  • 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 0 and 1. Defaults to 0
  • numberOfChannels - (optional) The number of channels to record. 1 = mono, 2 = stereo. Defaults to 1. Maximum 2 channels 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. 0 is fastest with lowest quality. 10 is slowest with highest quality. Defaults to 3.
  • streamPages - (optional) dataAvailable event will fire after each encoded page. Defaults to false.

Config Options for WAV recorder

  • 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.js or waveWorker.min.js worker script. Defaults to encoderWorker.min.js
  • leaveStreamOpen - (optional) Keep the stream around when trying to stop recording, so you can re-start without re-initStream. Defaults to false.
  • 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 0 and 1. Defaults to 0
  • numberOfChannels - (optional) The number of channels to record. 1 = mono, 2 = stereo. Defaults to 1. Maximum 2 channels are supported.
  • wavBitDepth - (optional) Desired bit depth of the WAV file. Defaults to 16. Supported values are 8, 16, 24 and 32 bits per sample.

Instance Methods

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.

Static Methods

Recorder.isRecordingSupported()

Returns a truthy value indicating if the browser supports recording.

Browser Support

Supported:

  • Chrome v58
  • Firefox v53
  • Microsoft Edge v41
  • Opera v44
  • macOS Safari v11
  • iOS Safari v11

Unsupported:

  • IE 11 and below
  • iOS 11 Chrome

Known Issues

  • 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

Building from sources

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 pkgconfig

Windows: Install autotools using MSYS2

pacman -S make autoconf automake libtool pkgconfig

Inatall Node.js

Install EMScripten

Install npm dependencies:

npm install

checkout, compile and create the dist from sources:

npm run make

Running the unit tests:

npm test

Clean the dist folder and git submodules:

make clean

Required Files

The 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.