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Tweak a bunch of settings

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This commit is contained in:
Travis Ralston 2021-03-19 17:08:01 -06:00
parent 0f09eb3214
commit 207ba11da1
1 changed files with 7 additions and 2 deletions
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9
src/voice/VoiceRecorder.ts
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@ -26,10 +26,15 @@ export class VoiceRecorder {
mediaTrackConstraints: <MediaTrackConstraints>{
deviceId: CallMediaHandler.getAudioInput(),
},
encoderSampleRate: 16000, // we could go down to 12khz, but we lose quality
encoderSampleRate: 48000, // we could go down to 12khz, but we lose quality. 48khz is a webrtc default
encoderApplication: 2048, // voice (default is "audio")
streamPages: true, // so we can have a live EQ for the user
encoderFrameSize: 10, // we want updates fairly regularly for the UI
encoderFrameSize: 20, // ms, we want updates fairly regularly for the UI
numberOfChannels: 1, // stereo isn't important for us
//sourceNode: instanceof MediaStreamAudioSourceNode, // TODO: @@ Travis: Use this for EQ stuff.
encoderBitRate: 64000, // 64kbps is average for webrtc
encoderComplexity: 3, // 0-10, 0 is fast and low complexity
resampleQuality: 3, // 0-10, 10 is slow and high quality
});
private buffer = new Uint8Array(0);
private mxc: string;