js/defaults.js
/**
* @file defaults.js
* @since 2.0.0
*/
// plugin defaults
const pluginDefaultOptions = {
// Single snapshot image.
image: false,
// Include audio in the recorded clip.
audio: false,
// Include video in the recorded clip.
video: false,
// Animated GIF.
animation: false,
// Screen capture.
screen: false,
// Maximum length of the recorded clip.
maxLength: 10,
// Maximum file size of the recorded clip. Works only when the timeSlice
// option is also enabled.
maxFileSize: 0,
// Boolean indicating if milliseconds should be included,
// e.g. "00:00:000" vs "00:00".
displayMilliseconds: false,
// Use a custom time format function, for example: (seconds, guide) => `test:${seconds}:${guide}`
formatTime: undefined,
// Width of the recorded video frames.
frameWidth: 320,
// Height of the recorded video frames.
frameHeight: 240,
// Enables console logging for debugging purposes.
debug: false,
// Enable Picture-in-Picture support.
pip: false,
// Turn off the camera/mic (and light) when audio and/or video recording
// stops, and turns them on again when you resume recording.
autoMuteDevice: false,
// The video bitrate in kbps (only used in webm-wasm plugin).
videoBitRate: 1200,
// Video recording library to use. Legal values are 'recordrtc' (default)
// and 'webm-wasm'.
videoEngine: 'recordrtc',
// The video frame rate in frames per second (only used in webm-wasm plugin).
videoFrameRate: 30,
// The mime type for the video recorder.
// Use 'video/mp4' (Firefox) or 'video/webm;codecs=H264' (Chrome 52 and
// newer) for MP4.
videoMimeType: 'video/webm',
// Video recorder type to use. This allows you to specify an alternative
// recorder class, e.g. WhammyRecorder from recordrtc. Defaults to 'auto'
// which let's recordrtc specify the best available recorder type.
videoRecorderType: 'auto',
// URL for the video worker. Use an empty string '' to disable (default).
videoWorkerURL: '',
// URL for the video worker WebAssembly file. Use an empty string '' to
// disable (default).
videoWebAssemblyURL: '',
// Audio recording library to use. Legal values are 'recordrtc' (default),
// 'libvorbis.js', 'opus-recorder', 'opus-media-recorder', 'lamejs', 'vmsg'
// and 'recorder.js'.
audioEngine: 'recordrtc',
// Audio recorder type to use. This allows you to specify an alternative
// recorder class, e.g. StereoAudioRecorder. Defaults to 'auto' which let's
// recordrtc specify the best available recorder type. Currently this
// setting is only used with the 'recordrtc' audioEngine.
audioRecorderType: 'auto',
// The mime type for the audio recorder. Defaults to 'auto' which will pick
// the best option available in the browser (e.g. either 'audio/wav',
// 'audio/ogg' or 'audio/webm').
audioMimeType: 'auto',
// The size of the audio buffer (in sample-frames) which needs to
// be processed each time onprocessaudio is called.
// From the spec: This value controls how frequently the audioprocess event is
// dispatched and how many sample-frames need to be processed each call.
// Lower values for buffer size will result in a lower (better) latency.
// Higher values will be necessary to avoid audio breakup and glitches.
// Legal values are 256, 512, 1024, 2048, 4096, 8192 or 16384.
audioBufferSize: 4096,
// The audio sample rate (in sample-frames per second) at which the
// AudioContext handles audio. It is assumed that all AudioNodes
// in the context run at this rate. In making this assumption,
// sample-rate converters or "varispeed" processors are not supported
// in real-time processing.
// The sampleRate parameter describes the sample-rate of the
// linear PCM audio data in the buffer in sample-frames per second.
// An implementation must support sample-rates in at least
// the range 22050 to 96000.
audioSampleRate: 44100,
// The audio bitrate in kbps (only used in lamejs plugin).
audioBitRate: 128,
// Allows you to record single-channel audio, which can reduce the
// file size.
audioChannels: 2,
// URL for the audio worker. Use an empty string '' to disable (default).
audioWorkerURL: '',
// URL for the audio worker WebAssembly file. Use an empty string '' to
// disable (default).
audioWebAssemblyURL: '',
// Enables the audioBufferUpdate event that provides realtime AudioBuffer
// instances from the input audio device.
audioBufferUpdate: false,
// Frame rate in frames per second.
animationFrameRate: 200,
// Sets quality of color quantization (conversion of images to the
// maximum 256 colors allowed by the GIF specification).
// Lower values (minimum = 1) produce better colors,
// but slow processing significantly. 10 is the default,
// and produces good color mapping at reasonable speeds.
// Values greater than 20 do not yield significant improvements
// in speed.
animationQuality: 10,
// A string indicating the output type: dataURL (base64 string) or blob.
// The default output is dataURL.
imageOutputType: 'dataURL',
// A string indicating the image format used in image-only mode. The default
// format type is image/png.
imageOutputFormat: 'image/png',
// A number between 0 and 1 indicating the image quality to use for image
// formats that use lossy compression such as image/jpeg and image/webp.
// If this argument is anything else, the default value for image quality
// is used. The default value is 0.92.
imageOutputQuality: 0.92,
// Accepts numbers in milliseconds; use this to force intervals-based blobs.
timeSlice: 0,
// Media converter library to use. Legal values are 'ts-ebml', 'ffmpeg.wasm'
// and 'ffmpeg.js'. Use an empty string '' to disable (default).
convertEngine: '',
// URL for the converter worker.
convertWorkerURL: '',
// List of string options to pass to the convert engine.
convertOptions: [],
// If the converter should automatically start after recording (default),
// or manually by calling `convert()`
convertAuto: true,
// Enable keyboard hotkeys.
hotKeys: false,
// Use this object to specify additional settings for the library used by the
// plugin (only used in opus-recorder, ffmpeg.js, ffmpeg.wasm and vmsg plugins).
pluginLibraryOptions: {}
};
export default pluginDefaultOptions;
