Tweak a bunch of settings
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1 changed files with 7 additions and 2 deletions
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@ -26,10 +26,15 @@ export class VoiceRecorder {
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mediaTrackConstraints: <MediaTrackConstraints>{
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mediaTrackConstraints: <MediaTrackConstraints>{
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deviceId: CallMediaHandler.getAudioInput(),
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deviceId: CallMediaHandler.getAudioInput(),
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},
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},
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encoderSampleRate: 16000, // we could go down to 12khz, but we lose quality
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encoderSampleRate: 48000, // we could go down to 12khz, but we lose quality. 48khz is a webrtc default
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encoderApplication: 2048, // voice (default is "audio")
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encoderApplication: 2048, // voice (default is "audio")
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streamPages: true, // so we can have a live EQ for the user
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streamPages: true, // so we can have a live EQ for the user
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encoderFrameSize: 10, // we want updates fairly regularly for the UI
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encoderFrameSize: 20, // ms, we want updates fairly regularly for the UI
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numberOfChannels: 1, // stereo isn't important for us
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//sourceNode: instanceof MediaStreamAudioSourceNode, // TODO: @@ Travis: Use this for EQ stuff.
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encoderBitRate: 64000, // 64kbps is average for webrtc
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encoderComplexity: 3, // 0-10, 0 is fast and low complexity
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resampleQuality: 3, // 0-10, 10 is slow and high quality
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});
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});
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private buffer = new Uint8Array(0);
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private buffer = new Uint8Array(0);
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private mxc: string;
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private mxc: string;
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