opus-recorder

Recorderjs

A javascript library to encode the output of Web Audio API nodes in Ogg Opus format. Audio encoded and decoded using libopus v1.1.4. Audio resampling is performed by speexDSP 1.2RC3. Encoded and muxed audio will be returned as typedArray in dataAvailable event.

Usage

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Constructor

The Recorder object is available in the global namespace and supports importing from module exports and AMD.

var rec = new Recorder([config]);

Creates a recorder instance.

• config - An optional configuration object (see config section below)

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Config

• bitRate (optional) Specifies the target bitrate in bits/sec. The encoder selects an application-specific default when this is not specified.
• 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) Specifies the encoder application. Supported values are 2048 - Voice, 2049 - Full Band Audio, 2051 - Restricted Low Delay. Defaults to 2049.
• encoderComplexity - (optional) Value between 0 and 10 which determines latency and processing for resampling. 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 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) Specifies the 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.
• 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.
• 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.

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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.

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Static Methods

Recorder.isRecordingSupported()

Will return a truthy value indicating if the browser supports recording.

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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.

Install EMScripten

Install autoconf, automake, libtool and pckconfig. On Mac you can do this using MacPorts sudo port install automake autoconf libtool pkgconfig

Make the dependencies using command make!

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Running the unit tests

make test

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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.