how to play a background sound in loop while agent is speaking ? @millions-hair-24956

To play background sound in a loop while an agent is speaking, you can publish a separate audio track dedicated to the background sound. This allows you to control the playback independently from the agent's speech. According to LiveKit documentation, you can create a

LocalAudioTrack

for the background sound and manage its playback using the

publish_track

method. Here's a basic approach: 1. Create and Publish Track: Use

LocalAudioTrack.create_audio_track

to create a track for your background sound and publish it using

publish_track

. 2. Loop Playback: Implement a loop to continuously push audio frames to the track. Ensure the audio data is in the correct format (e.g.,

int16

). Would you like more detailed guidance or a code example? Sources: - LiveKit Slack Discussion - LiveKit documentation 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.

please provide code example @millions-hair-24956

Here's a Python code example to play a background sound in a loop while an agent is speaking using LiveKit:

import asyncio
import numpy as np
from livekit import rtc

async def play_background_sound(room, audio_file_path):
    # Load audio data
    with open(audio_file_path, 'rb') as audio_file:
        audio_data = audio_file.read()

    # Create an audio source and track
    sample_rate = 44100  # Example sample rate
    num_channels = 1  # Mono audio
    audio_source = rtc.AudioSource(sample_rate, num_channels)
    background_track = rtc.LocalAudioTrack.create_audio_track("background_audio", audio_source)

    # Publish the track
    await room.local_participant.publish_track(background_track)

    # Loop to continuously send audio frames
    chunk_size = 1024  # Define chunk size
    while True:
        for i in range(0, len(audio_data), chunk_size):
            chunk = audio_data[i:i + chunk_size]
            if len(chunk) < chunk_size:
                chunk = np.pad(chunk, (0, chunk_size - len(chunk)))
            audio_frame = rtc.AudioFrame(
                data=chunk.tobytes(),
                sample_rate=sample_rate,
                samples_per_channel=chunk_size,
                num_channels=num_channels
            )
            await audio_source.capture_frame(audio_frame)
            await asyncio.sleep(chunk_size / sample_rate)

# Example usage
async def main():
    # Connect to the room
    room = rtc.Room()
    await room.connect("your_livekit_url", "your_participant_token")

    # Play background sound
    await play_background_sound(room, "path_to_your_audio_file.wav")

if __name__ == '__main__':
    asyncio.run(main())

This code sets up a separate audio track for the background sound and continuously sends audio frames to it. Make sure your audio file is in the correct format and adjust the

sample_rate

and

chunk_size

as needed. Sources: - LiveKit Slack Discussion 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.

lets say that audio file is two minutes long, how to set chunk size ? @millions-hair-24956

To determine the chunk size for a two-minute audio file, you need to consider the sample rate and the number of channels. The chunk size is typically set to match the duration of audio you want to process in each iteration, often in milliseconds. For example, if your audio file is two minutes long, and you want to process it in 20ms chunks, you can calculate the chunk size as follows: 1. Calculate Samples per Chunk: - If the sample rate is 44100 Hz (samples per second), then for 20ms, the number of samples per chunk is

44100 * 0.02 = 882

samples. 2. Determine Chunk Size in Bytes: - For mono audio (1 channel) with 16-bit samples, each sample is 2 bytes. Therefore, the chunk size in bytes is

882 samples * 2 bytes/sample = 1764 bytes

. Here's how you can set the chunk size in the code:

sample_rate = 44100  # Sample rate in Hz
chunk_duration_ms = 20  # Chunk duration in milliseconds
samples_per_chunk = int(sample_rate * (chunk_duration_ms / 1000.0))
chunk_size = samples_per_chunk * 2  # 2 bytes per sample for 16-bit audio

# Use chunk_size in your loop
while True:
    for i in range(0, len(audio_data), chunk_size):
        chunk = audio_data[i:i + chunk_size]
        if len(chunk) < chunk_size:
            chunk = np.pad(chunk, (0, chunk_size - len(chunk)))
        # Process the chunk...

Would you like more details on this calculation? Sources: - LiveKit Slack Discussion 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.

auido channels : stereo sample rate : 44.1 khz bits per sample : 16 duration : 02:29 @millions-hair-24956

To handle a stereo audio file with a sample rate of 44.1 kHz, 16 bits per sample, and a duration of 2 minutes and 29 seconds, you can calculate the chunk size for processing in smaller segments. Here's how you can determine the chunk size: 1. Calculate Samples per Chunk: - For a 20ms chunk duration, the number of samples per chunk is

44100 * 0.02 = 882

samples. 2. Determine Chunk Size in Bytes: - For stereo audio (2 channels) with 16-bit samples, each sample is 2 bytes. Therefore, the chunk size in bytes is

882 samples * 2 bytes/sample * 2 channels = 3528 bytes

. Here's how you can set the chunk size in the code:

sample_rate = 44100  # Sample rate in Hz
num_channels = 2  # Stereo audio
chunk_duration_ms = 20  # Chunk duration in milliseconds
samples_per_chunk = int(sample_rate * (chunk_duration_ms / 1000.0))
chunk_size = samples_per_chunk * 2 * num_channels  # 2 bytes per sample for 16-bit audio

# Use chunk_size in your loop
while True:
    for i in range(0, len(audio_data), chunk_size):
        chunk = audio_data[i:i + chunk_size]
        if len(chunk) < chunk_size:
            chunk = np.pad(chunk, (0, chunk_size - len(chunk)))
        # Process the chunk...

This setup will allow you to process the audio in 20ms chunks, which is a common practice for real-time audio processing. Would you like more details or assistance with implementation? Sources: - LiveKit Slack Discussion 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.