Adjust some settings

src/components/views/voice_messages/LiveRecordingWaveform.tsx

@@ -49,12 +49,12 @@ export default class LiveRecordingWaveform extends React.PureComponent<IProps, I
         const bars = arrayFastResample(Array.from(update.waveform), DOWNSAMPLE_TARGET);
         this.setState({
             // The incoming data is between zero and one, but typically even screaming into a
-            // microphone won't send you over 0.6, so we "cap" the graph at about 0.35 for a
+            // microphone won't send you over 0.6, so we "cap" the graph at about 0.50 for a
             // point where the average user can still see feedback and be perceived as peaking
             // when talking "loudly".
             //
             // We multiply by 100 because the Waveform component wants values in 0-100 (percentages)
-            heights: bars.map(b => percentageOf(b, 0, 0.35) * 100),
+            heights: bars.map(b => percentageOf(b, 0, 0.50) * 100),
         });
     };
 

src/voice/VoiceRecorder.ts

@@ -73,10 +73,9 @@ export class VoiceRecorder {
         // The size of the audio buffer largely decides how quickly we push timing/waveform data
         // out of this class. Smaller buffers mean we update more frequently as we can't hold as
         // many bytes. Larger buffers mean slower updates. For scale, 1024 gives us about 30Hz of
-        // updates and 2048 gives us about 20Hz. We use 2048 because it updates frequently enough
-        // to feel realtime (~20fps, which is what humans perceive as "realtime"). Must be a power
-        // of 2.
-        this.recorderProcessor = this.recorderContext.createScriptProcessor(2048, CHANNELS, CHANNELS);
+        // updates and 2048 gives us about 20Hz. We use 1024 to get as close to perceived realtime
+        // as possible. Must be a power of 2.
+        this.recorderProcessor = this.recorderContext.createScriptProcessor(1024, CHANNELS, CHANNELS);
 
         // Connect our inputs and outputs
         this.recorderSource.connect(this.recorderFFT);