Is there an easy way to calculate user's input vol...
# ask-ai
f
Is there an easy way to calculate user's input volume when they speak?
t
Thanks for reaching out, Muthana. I'm an AI assistant and I'm here to help. Give me a moment while I research this topic.
Hey there! If you want to figure out a user's input volume when they speak, you can use the Web Audio API to analyze their audio input stream. Here’s a simple way to do it: 1. Create an AudioContext and AnalyserNode: Start by using the Web Audio API to set up an audio context and an analyser node to process the audio stream. 2. Connect the MediaStream: Link the user's audio track to the audio context using
createMediaStreamSource
. 3. Configure the AnalyserNode: Adjust settings like
fftSize
and
smoothingTimeConstant
to get the best results. 4. Calculate Volume: Regularly grab the frequency data from the analyser, calculate the root mean square (RMS) of the audio signal, and normalize it so you get a volume value between 0 and 1. Here’s a quick example of how you might do the calculation:
Copy code
function calculateVolume() {
    analyser.getByteFrequencyData(dataArray);
    let sum = 0;
    for (const amplitude of dataArray) {
        sum += Math.pow(amplitude / 255, 2);
    }
    const volume = Math.sqrt(sum / dataArray.length);
    return volume;
}
And if you’re working with LiveKit’s React components, there’s an even easier way! You can use the
useTrackVolume
hook like this:
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import { useTrackVolume } from "livekit/components-react";

function SpeakerVolumeIndicator({ track }) {
    const volume = useTrackVolume(track);

    return (
        <div>
            <p>Volume: {volume.toFixed(2)}</p>
        </div>
    );
}
If you want me to walk you through any part of this or need more examples, just let me know—I’m here to help!
f
In this call path (Telnyx SIP → LiveKit room → Python AgentSession), when we pass
RoomInputOptions(noise_cancellation=BVCTelephony())
, does BVC run upstream of the agent’s
stt_node
input
?
t
Yes, when you set up
RoomInputOptions(noise_cancellation=BVCTelephony())
, the BVCTelephony noise cancellation works right before the agent's
stt_node
input. This way, the audio gets cleaned up first, making sure it’s in great shape before it reaches the transcription system—helping everything run smoother and improving the STT results. Sources: Audio and video | LiveKit Docs | examples/voice_agents/multi_agent.py | Voice AI quickstart | LiveKit Docs | Enhanced noise cancellation | LiveKit Docs
f
Given we also use
AudioProcessingModule
(AGC/NS/HPF), what’s the cleanest hook to access frames before APM but after BVC (or raw if BVC happens later)? Is the
audio
iterable passed to
stt_node(self, audio, ...)
indeed pre-APM, and only our
apm.process_stream(...)
makes it post-APM?
t
I don't have the answer you're looking for. You could also try asking your question: • in one of the other Slack channels or • to https://deepwiki.com/livekit/livekit_composite which is trained on all LiveKit source code If you find the answer, please post it here to help others!
f
please show the canonical snippet to obtain a stream at the desired point.