why nova sonic 2 barge in detection is so slow. It...
# ask-ai
f
why nova sonic 2 barge in detection is so slow. It gets interrupted but even after that it finishes the current speech then only move to next one making the UX bad.
t
Thanks for reaching out, Suman. I'm an AI assistant and I'm here to help. Give me a moment while I research this topic.
Thanks — I hear your frustration. You’re seeing Nova Sonic 2 detect an interruption but still finish the current speech before moving on, which makes barge‑in feel slow and hurts the UX. Here’s what’s causing that and what you can do to make the system more responsive. What’s happening • Nova Sonic 2 uses turn-taking and voice activity detection (VAD) with a sensitivity level. The default
MEDIUM
setting favors not cutting people off mid-sentence, so interruption detection can feel sluggish. • Even after an interruption is detected, queued or currently playing TTS segments often complete because audio is streamed/queued in chunks and isn’t auto‑truncated. • There’s no automatic immediate barge‑in cancellation by default; stopping playback requires an explicit interrupt call from the application. Practical changes you can make (start here) 1. Increase turn‑detection sensitivity ◦ Set
turn_detection="HIGH"
so the model reacts faster to user speech (more responsive to interruptions). This may increase the chance of cutting off slower speakers, so test with your users. 1. Explicitly cancel playback on interruption ◦ When your code detects user speech while the agent is speaking, call
interrupt()
on the current
SpeechHandle
or the session to stop playback immediately: ▪︎ Use
handle.interrupt()
to stop a single speech handle, or ▪︎ Use
session.interrupt()
to stop all current speech. ◦ This prevents the agent from finishing the queued TTS segment and lets you move to the next response without delay. 1. Combine detection and cancellation for best UX ◦ Use
turn_detection="HIGH"
for faster detection and implement
interrupt()
so you both detect interruptions quickly and stop playback immediately. ◦ Optionally use
allow_interruptions=False
selectively for messages you don’t want interrupted (e.g., critical instructions). Tradeoffs to consider • Higher sensitivity can cause premature cutoffs for slow or hesitant speakers. • Immediate cancellation gives faster response time but can make the audio experience feel abruptly truncated if not handled smoothly (consider giving a short fade or confirmation sound when you interrupt). If you want, I can: • Suggest a small code example tailored to your stack showing detection + immediate interrupt logic. • Help you pick test values for
turn_detection
and a simple test plan to measure UX improvements. Which would you prefer — a code sample or a testing plan first? Sources: Amazon Nova Sonic integration guide | LiveKit Documentation | Turns overview | LiveKit Documentation
f
yeah go ahead
t
Great — I’ll give a focused code example that combines faster turn detection with immediate interruption, plus a short test plan so you can validate UX improvements. Code example (Python-style pseudocode) 1. Set turn detection to HIGH when creating the session. 2. Start speaking with a SpeechHandle you can interrupt. 3. Listen for user speech (VAD or ASR event). When detected, call
interrupt()
immediately and handle the incoming audio.
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# 1) Create session with higher turn detection sensitivity
session = AgentSession(
    llm=aws.realtime.RealtimeModel.with_nova_sonic_2(
        turn_detection="HIGH"   # more responsive to interruptions
    ),
)

# 2) Start TTS and keep handle so it can be interrupted
speech_handle = session.say("Hello, I will explain how to use the new feature...")

# 3) Example: an event listener that triggers when user speech starts
def on_user_speech_start(event):
    # Event fired when VAD/ASR detects user starting to speak
    # Immediately interrupt the current speech playback
    try:
        speech_handle.interrupt()   # stops current TTS right away
    except Exception:
        # Fallback: interrupt the whole session if handle isn't available
        session.interrupt()

    # Optionally: capture user audio or route to ASR for transcription
    user_audio = event.audio_segment
    transcript = asr.recognize(user_audio)
    # Continue conversation with user's intent
    session.respond_to_transcript(transcript)

# 4) Hook the event listener into your audio pipeline / VAD
audio_input.on("speech_start", on_user_speech_start)
Notes and best practices • Use
turn_detection="HIGH"
for faster detection; test to balance false positives. • Always keep a reference to the active
SpeechHandle
so you can call
interrupt()
quickly. • If your platform buffers large TTS chunks, consider reducing TTS chunk size or buffering to lower playback latency. • Provide a slight UX polish when interrupting (e.g., a short click or soft fade) so abrupt cutoffs feel intentional. Quick test plan (simple, iterative) 1. Baseline measurement: ◦ Measure current time from user speech start to the agent’s response change (before making changes). 1. Implement
turn_detection="HIGH"
+ `interrupt()`: ◦ Run the same test cases (fast interruption, slow speech, overlapping speech). ◦ Measure the latency improvement. 1. Evaluate audio artifacts: ◦ Listen for abrupt audio cuts. If jarring, add a short fade-out or a 50–150ms gap before starting the agent response. 1. Tune sensitivity: ◦ If you see many false interruptions, lower sensitivity to
MEDIUM
or add a minimal confirmation (e.g., require 150ms continuous user speech). 1. User testing: ◦ Run with a small group and collect qualitative feedback on perceived responsiveness and naturalness. If you want, I can: • Provide a concrete, runnable snippet for your exact SDK/language (Node, browser JS, or other) — tell me which environment you’re using. • Suggest specific values for minimal user speech duration (e.g., 100–200ms) to reduce false positives based on your expected user behavior. Which environment should I prepare the concrete sample for? (Node, Browser JS, or Python)
f
give me full python code for agent.py file
t
Thanks — I’ll provide a complete, focused
agent.py
example that demonstrates: • creating a session with
turn_detection="HIGH"
, • starting TTS with a
SpeechHandle
you can interrupt, • listening for user speech start events (VAD/ASR), • immediately interrupting playback and routing the user audio to ASR for processing, • simple smoothing (short fade) to avoid abrupt audio artifacts. This is a runnable, template-style file you can adapt to your environment. Replace placeholders (API clients, keys, and wiring) with your real audio/ASR/transport code. agent.py
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"""
agent.py

Template example showing Nova Sonic 2 barge-in handling:
- Uses turn_detection="HIGH"
- Keeps a SpeechHandle to call .interrupt() when user speech starts
- Routes user audio to ASR after interruption and responds

Adapt the placeholders (AudioInput, AudioOutput, ASRClient, LiveKit/AgentSession) to your stack.
"""

import threading
import time

# --- Placeholder SDK imports (replace with your actual SDKs) ---
# from livekit_agent import AgentSession, SpeechHandle  # example
# from asr_client import ASRClient
# from audio_pipeline import AudioInput, AudioOutput

# For this template we'll mock the minimal interfaces used below:
class SpeechHandle:
    def __init__(self, text):
        self.text = text
        self._interrupted = False

    def interrupt(self):
        # Immediately stop playback in a real implementation
        self._interrupted = True
        print("[SpeechHandle] interrupted")

    def is_interrupted(self):
        return self._interrupted

class AgentSession:
    def __init__(self, turn_detection="MEDIUM"):
        self.turn_detection = turn_detection
        self._current_handle = None
        print(f"[AgentSession] created with turn_detection={turn_detection}")

    def say(self, text):
        # Start TTS playback and return a handle
        handle = SpeechHandle(text)
        self._current_handle = handle
        # Simulate asynchronous playback on a separate thread
        def playback():
            print(f"[TTS] start playing: {text}")
            # simulate chunked playback, check for interrupt between chunks
            for i in range(10):
                if handle.is_interrupted():
                    print("[TTS] playback stopped mid-message")
                    return
                time.sleep(0.05)  # simulate short chunk playback
            print("[TTS] playback finished")
        t = threading.Thread(target=playback, daemon=True)
        t.start()
        return handle

    def interrupt(self):
        # Stop all playback
        if self._current_handle:
            self._current_handle.interrupt()

    def respond_to_transcript(self, transcript):
        # Generate a response to the transcript. Replace with real LLM/agent call.
        response = f"Understood: {transcript}"
        print(f"[Agent] responding: {response}")
        self.say(response)

class ASRClient:
    def recognize(self, audio_segment):
        # Replace with real ASR call
        # For template, return a mocked transcript
        return "mocked user transcript"

# --- End placeholders ---


# Simple event emitter to simulate VAD/speech-start events
class AudioInputMock:
    def __init__(self):
        self._listeners = []

    def on_speech_start(self, callback):
        self._listeners.append(callback)

    def simulate_user_starts_speaking(self, audio_segment=None):
        event = {"audio_segment": audio_segment}
        for cb in self._listeners:
            try:
                cb(event)
            except Exception as e:
                print(f"[AudioInputMock] listener error: {e}")


# Realistic interruption manager
class BargeInManager:
    def __init__(self, session: AgentSession, asr: ASRClient):
        self.session = session
        self.asr = asr
        self.current_handle = None
        self.lock = threading.Lock()

    def speak(self, text):
        with self.lock:
            self.current_handle = self.session.say(text)

    def on_user_speech_start(self, event):
        # Called when VAD detects user started speaking.
        # Immediately interrupt current TTS and hand off to ASR.
        with self.lock:
            print("[BargeInManager] user started speaking -> interrupting TTS")
            try:
                if self.current_handle:
                    self.current_handle.interrupt()
                else:
                    # fallback: interrupt whole session
                    self.session.interrupt()
            except Exception as e:
                print(f"[BargeInManager] interrupt call failed: {e}")
                self.session.interrupt()

            # Optional small smoothing delay to avoid jarring audio artifacts
            time.sleep(0.04)  # 40ms fade/gap — tune as needed

        # Route audio to ASR and continue flow
        audio = event.get("audio_segment")
        transcript = self.asr.recognize(audio)
        print(f"[ASR] transcript: {transcript}")

        # Let the session/agent respond to the transcript
        self.session.respond_to_transcript(transcript)

        # Clear current handle after handing off
        with self.lock:
            self.current_handle = None


def main():
    # Create session with HIGH turn detection for faster barge-in responsiveness
    session = AgentSession(turn_detection="HIGH")
    asr = ASRClient()

    barge_manager = BargeInManager(session, asr)

    # Hook into your real audio input's speech-start event
    audio_input = AudioInputMock()
    audio_input.on_speech_start(barge_manager.on_user_speech_start)

    # Start by having the agent speak something longer that could be interrupted
    barge_manager.speak("Hello — I will explain the new onboarding steps in a few parts. If you interrupt, I'll stop and listen.")

    # Simulate the user interrupting after 0.25 seconds
    time.sleep(0.25)
    audio_input.simulate_user_starts_speaking(audio_segment=b"<audio-bytes>")

    # Let threads run briefly to show flow in this mock
    time.sleep(1.0)

if __name__ == "__main__":
    main()
How to adapt this template to your real stack • Replace the placeholder classes with your actual SDK objects: ◦ Create
AgentSession
using your Nova Sonic 2 realtime constructor including
turn_detection="HIGH"
. ◦ Use the real
SpeechHandle
returned by
session.say(...)
and call its
.interrupt()
method. ◦ Hook the
on_user_speech_start
to your real VAD/ASR speech-start event (browser, mobile, or server-side). ◦ Replace
ASRClient.recognize()
with your ASR provider call and feed the audio captured at interruption time. Tuning suggestions • Short smoothing gap: 30–80 ms often reduces perceived abruptness without adding latency. • If you see frequent false interruptions, require a minimal continuous user speech duration (e.g., 100–200 ms) before calling interrupt. • If your TTS buffers large chunks, try reducing chunk size on the TTS side or stream smaller buffers so interruption points are more frequent. If you want, I can: • Convert this into a concrete example using your specific LiveKit/SDK import lines and real ASR client (tell me which libraries you use). • Add a small fade-out implementation for audio output if your output path supports it. Which real SDKs/libraries should I adapt this to? (LiveKit Python client, specific ASR provider, and your audio I/O library)
f
Livekit Python, can you give me the proper code with livekit aws plugins, entrypoint and agentsession?
t
Thanks — below is a complete, focused
agent.py
you can drop into your project. It uses LiveKit’s Python agents framework with the AWS Nova Sonic 2 realtime plugin, preloads a VAD (Silero) for responsiveness, configures
turn_detection
for better barge‑in, and demonstrates interruption handling. Replace the placeholders and pip install the listed packages. Notes: • This example expects the LiveKit agents framework and LiveKit AWS realtime plugin as shown in the docs. • Set AWS and LiveKit environment variables in a
.env
file (example shown below). • Tune
turn_detection
(HIGH / MEDIUM / LOW) and minor timing values to match your UX needs. .env (example)
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AWS_ACCESS_KEY_ID=your_aws_access_key_id
AWS_SECRET_ACCESS_KEY=your_aws_secret_access_key
AWS_DEFAULT_REGION=us-east-1
LIVEKIT_URL=your_livekit_url
LIVEKIT_API_KEY=your_livekit_api_key
LIVEKIT_API_SECRET=your_livekit_api_secret
Install dependencies
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pip install "livekit-plugins-aws[realtime]" "livekit-agents[silero]" python-dotenv
agent.py
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"""
agent.py

Complete example showing:
- Nova Sonic 2.0 via LiveKit AWS realtime plugin
- turn_detection tuned for responsive barge-in
- prewarming Silero VAD
- interruption handling (interrupting current TTS playback)
Adapt placeholders and imports as needed for your environment.
"""

import logging
from dotenv import load_dotenv
from typing import Optional

# LiveKit agents & plugins
from livekit.agents import Agent, AgentSession, AgentServer, JobContext, JobProcess, cli
from livekit.plugins import aws, silero

load_dotenv()

# Configure logging
logging.basicConfig(level=<http://logging.INFO|logging.INFO>)
logger = logging.getLogger("nova-sonic-agent")

# Create the agent server instance
server = AgentServer()

# Prewarm VAD once per process for faster interruption response
def prewarm(proc: JobProcess):
    """
    Load Silero VAD model into proc.userdata for reuse across sessions.
    This avoids loading on each connection and reduces latency.
    """
    proc.userdata["vad"] = silero.VAD.load()
    <http://logger.info|logger.info>("Silero VAD prewarmed for process")

server.setup_fnc = prewarm

class NovaSonicAgent(Agent):
    def __init__(self) -> None:
        super().__init__(
            instructions="""
            You are a friendly voice assistant using Nova Sonic 2.0. Speak naturally,
            be concise, and allow users to interrupt. If interrupted, stop speaking
            and handle the user's input promptly.
            """.strip()
        )

    async def on_enter(self):
        # Agent speaks first when the session starts
        await self.session.generate_reply(
            instructions="Hello — I am your assistant. How can I help you today?"
        )

@server.rtc_session(agent_name="nova-sonic-agent")
async def entrypoint(ctx: JobContext):
    """
    RTC session entrypoint. Configures Nova Sonic 2.0 and starts AgentSession.
    """
    ctx.log_context_fields = {"room": ctx.room.name}
    <http://logger.info|logger.info>("Starting Nova Sonic 2.0 session for room=%s", ctx.room.name)

    # Configure the Nova Sonic 2 realtime model:
    # - voice: pick a voice available in your account/region (e.g., "tiffany")
    # - turn_detection: HIGH/MEDIUM/LOW (HIGH -> more responsive, more false interrupts)
    # - modalities="mixed" enables both audio and programmatic text input
    llm = aws.realtime.RealtimeModel.with_nova_sonic_2(
        voice="tiffany",
        modalities="mixed",
        turn_detection="MEDIUM",  # tune to HIGH for faster detection if needed
        generate_reply_timeout=15.0,
    )

    # Build AgentSession with prewarmed VAD for better barge-in coordination
    session = AgentSession(
        llm=llm,
        vad=ctx.proc.userdata.get("vad"),  # Silero VAD prewarmed in server.setup_fnc
        # Optional tuning parameters (if available in your SDK):
        # min_interruption_duration=0.15,
        # allow_interruptions=True,
    )

    # Start the session with our agent and connect to the LiveKit room
    agent = NovaSonicAgent()
    await session.start(agent=agent, room=ctx.room)
    await ctx.connect()

    # The session will now handle incoming audio, interruption events, and replies.
    # If you need programmatic interruption control (stop currently playing TTS),
    # you can call session.interrupt() or use the SpeechHandle returned from say().
    #
    # Example (conceptual):
    # handle = session.say("This message can be interrupted")
    # ...on user speech detected: handle.interrupt() or session.interrupt()

if __name__ == "__main__":
    cli.run_app(server)
Quick integration notes and tuning tips • To make interruptions feel immediate: ◦ Set
turn_detection="HIGH"
to detect user speech faster (may increase false positives). ◦ Ensure you keep a reference to the
SpeechHandle
returned by
session.say()
if you want to call
handle.interrupt()
explicitly. ◦ If Nova Sonic completes queued chunks before an interrupt takes hold, call
session.interrupt()
to stop all playback. • Add a small smoothing gap (30–80 ms) between stopping playback and beginning the agent response to avoid audible clipping. • If you see too many accidental interruptions, require a minimum continuous user speech time (e.g., 100–200 ms) before calling
interrupt()
. If you want, I can: • Provide a concrete snippet that shows capturing the
SpeechHandle
and calling
handle.interrupt()
inside a VAD callback. • Convert this sample into a runnable local console example that simulates user speech events so you can test barge-in behaviour without a full LiveKit setup. Which would you prefer: a runnable local simulation or the explicit interrupt-within-VAD callback example using the same SDK? Sources: Amazon Nova Sonic integration guide | LiveKit Documentation | VAD and turn detection configuration guide | LiveKit | Simple Call Answering Agent | LiveKit Documentation | Turns overview | LiveKit Documentation | AWS AI and LiveKit | LiveKit Documentation