How can i apply a gain function on incoming audio frames from a user on SIP

Thanks for reaching out, Firmino. I'm an AI assistant and I'm here to help. Give me a moment while I research this topic.

real audio streams, i just want to make the user's volume a bit louder to make sure the STT can fully understand what they're saying

If you're looking to boost the volume of real-time audio streams to get better speech-to-text results, you can tweak the gain programmatically. Here are a couple of friendly ways to do it: 1. Server-Side Volume Adjustment: • You can adjust the audio frames right on your server. For instance, multiplying the audio frame data by a factor can help raise the volume:

audio_data = np.frombuffer(frame.data, dtype=np.int16)
audio_data = audio_data * 1.5  # Feel free to change the multiplier as needed
adjusted_frame = rtc.AudioFrame(
    data=audio_data.astype(np.int16).tobytes(),
    sample_rate=frame.sample_rate,
    num_channels=frame.num_channels,
    samples_per_channel=len(audio_data) // frame.num_channels,
)

2. Client-Side Volume Adjustment: • You can also use Web Audio APIs to amp up the audio before it reaches the STT system:

const audioContext = new (window.AudioContext || window.webkitAudioContext)();
const gainNode = audioContext.createGain();

// Connect the audio stream to the gain node
const source = audioContext.createMediaStreamSource(stream);
source.connect(gainNode);
gainNode.connect(audioContext.destination);

// Set the gain (volume)
gainNode.gain.setValueAtTime(1.5, 0);  // 1.5x volume gain

Just a heads-up: make sure the volume boost doesn’t cause any clipping or distortion. It’s a good idea to test and find that sweet spot for the best STT accuracy. If you need any help tweaking it, I’m here for you!

btw, i'm using livekit agents, so its on the agent side

If you want to apply a gain function to audio streams on the agent side in LiveKit Agents, you can tweak the audio processing nodes to include gain adjustment. Here’s a simple example to get you started:

class MyAgent(Agent):
    def __init__(self, *, gain_db: float = 0.0) -> None:
        super().__init__()
        self.gain_db = gain_db

    async def realtime_audio_output_node(
        self, audio: AsyncIterable[rtc.AudioFrame], model_settings: ModelSettings
    ) -> AsyncIterable[rtc.AudioFrame]:
        return self._apply_gain(
            Agent.default.realtime_audio_output_node(self, audio, model_settings)
        )

    async def _apply_gain(
        self, audio: AsyncIterable[rtc.AudioFrame]
    ) -> AsyncIterable[rtc.AudioFrame]:
        async for frame in audio:
            yield self._process_audio(frame)

    def _process_audio(self, frame: rtc.AudioFrame) -> rtc.AudioFrame:
        audio_data = np.frombuffer(frame.data, dtype=np.int16).astype(np.float32)
        audio_data *= 10 ** (self.gain_db / 20)  # Convert dB to amplitude ratio
        audio_data = np.clip(audio_data, -32768, 32767).astype(np.int16)
        return rtc.AudioFrame(
            data=audio_data.tobytes(),
            sample_rate=frame.sample_rate,
            num_channels=frame.num_channels,
            samples_per_channel=frame.samples_per_channel,
        )

Here’s how it works: • First, set the gain you want in decibels (for example,

gain_db=3.0

to boost by +3dB). • The code converts that dB value into an amplitude ratio and multiplies it with the audio data to apply the gain. • It also takes care of clipping the audio so it stays within the valid 16-bit integer range, avoiding distortion. If you’d like, you can even add a dynamic control to adjust the gain on the fly while it’s running. Just let me know if you want help with that or anything else!

actually, is the user's input

If you want to apply a gain function to a user's input audio stream in LiveKit Agents, you can do this by processing the audio frames right inside the

realtime_audio_output_node

method. Here’s a simple example to get you started:

class MyAgent(Agent):
    def __init__(self, *, gain_factor: float = 2.0) -> None:
        super().__init__()
        self.gain_factor = gain_factor

    async def realtime_audio_output_node(
        self, audio: AsyncIterable[rtc.AudioFrame], model_settings: ModelSettings
    ) -> AsyncIterable[rtc.AudioFrame]:
        async for frame in audio:
            yield self._apply_gain(frame)

    def _apply_gain(self, frame: rtc.AudioFrame) -> rtc.AudioFrame:
        audio_data = np.frombuffer(frame.data, dtype=np.int16)
        audio_data = (audio_data * self.gain_factor).astype(np.int16)  # Apply gain
        return rtc.AudioFrame(
            data=audio_data.tobytes(),
            sample_rate=frame.sample_rate,
            num_channels=frame.num_channels,
            samples_per_channel=frame.samples_per_channel,
        )

Here’s how it works: • Pick the

gain_factor

you want (for example,

gain_factor=2.0

will double the volume). • The gain is applied by multiplying the audio data by this factor in the

_apply_gain

method. • Just a heads up: be careful with the gain factor to avoid any audio clipping or distortion. If you want me to help tweak this or if you have any questions about your specific setup, just let me know—I’m here to help!

what is the recommened gain level to make sure the ai doesnt wait indefenidetely

Hey there! While LiveKit Agents don’t specify an exact gain level for user input audio streams, here are some friendly tips to help make sure the AI doesn’t end up waiting forever for input: • Target Audio Levels: Try to keep peak levels between -20 dB and -15 dB for speech. This range usually keeps things clear and loud enough without any distortion. • Automatic Gain Control (AGC): If your system has it, turning on AGC can be a great way to automatically keep input levels just right. • Monitor Input: It’s a good idea to regularly check your audio input levels to make sure everything stays within the sweet spot for your setup. If you want a hand with making these tweaks or setting things up, just let me know—I’m here to help!

you mean activation threshold?

Yes, the activation threshold is what helps decide when a frame actually has speech in it. In LiveKit Agents, the default threshold is set to 0.5. This strikes a nice balance between catching speech accurately and not getting tripped up by background noise: • Higher Threshold: Plays it safe by reducing false alarms but might miss some softer speech. • Lower Threshold: Picks up even quiet speech but could mistake some noise for talking. You can tweak this threshold in the Silero VAD plugin or in your agent’s turn detection settings to fit what works best for you. Just give me a shout if you want a hand setting it up! Sources: Silero VAD plugin | LiveKit Docs | Configuring turn detection and interruptions | LiveKit Docs

if i set a low threshold like .1 and 2x the input volume, will it help on the agent waiting forerever on listening mode?

While there isn’t specific documentation on how setting a low activation threshold (like 0.1) and doubling the input volume directly prevent the agent from waiting forever in listening mode, generally speaking, lowering the threshold makes the system more sensitive to picking up speech. This can be really helpful if someone’s speaking softly and the system might miss it at higher thresholds. That said, it’s a good idea to try out these tweaks in your own setup to make sure they work well without accidentally picking up too much background noise. If you want, I’m here to help you test or fine-tune the settings—just let me know! Sources: Configuring turn detection and interruptions | LiveKit Docs