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The AudioData class represents raw audio data received in audio renderer callbacks. It provides access to audio frame metadata and the raw audio samples.

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Class Definition

class AudioData:
    @property
    def bits_per_sample(self) -> int: ...
    @property
    def sample_rate(self) -> int: ...
    @property
    def num_channels(self) -> int: ...
    @property
    def num_audio_frames(self) -> int: ...
    @property
    def audio_frames(self) -> bytes: ...

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Properties

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bits_per_sample

@property
def bits_per_sample(self) -> int
The number of bits per audio sample (typically 16). Returns: int - Bits per sample

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sample_rate

@property
def sample_rate(self) -> int
The audio sample rate in Hz (e.g., 16000, 44100, 48000). Returns: int - Sample rate in Hz

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num_channels

@property
def num_channels(self) -> int
The number of audio channels (1 for mono, 2 for stereo). Returns: int - Number of channels

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num_audio_frames

@property
def num_audio_frames(self) -> int
The number of audio frames (samples per channel) in this buffer. Returns: int - Number of audio frames

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audio_frames

@property
def audio_frames(self) -> bytes
The raw audio data as bytes. The format is interleaved PCM audio. Returns: bytes - Raw audio frame data

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Usage in Audio Renderers

AudioData objects are received in audio renderer callbacks that you set with set_audio_renderer().

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Example: Basic Audio Renderer

from daily import CallClient, AudioData
import wave

def audio_callback(participant_id: str, audio_data: AudioData, audio_source: str):
    # Access audio metadata
    print(f"Sample rate: {audio_data.sample_rate} Hz")
    print(f"Channels: {audio_data.num_channels}")
    print(f"Frames: {audio_data.num_audio_frames}")
    print(f"Bits per sample: {audio_data.bits_per_sample}")
    
    # Get raw audio frames
    raw_audio = audio_data.audio_frames
    print(f"Audio data size: {len(raw_audio)} bytes")

# Set up audio renderer
client = CallClient()
client.set_audio_renderer(
    participant_id="participant-id",
    callback=audio_callback,
    sample_rate=16000,
    callback_interval_ms=20
)

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Example: Saving Audio to WAV File

from daily import CallClient, AudioData
import wave

class AudioRecorder:
    def __init__(self, filename: str):
        self.filename = filename
        self.wav_file = None
        
    def audio_callback(self, participant_id: str, audio_data: AudioData, audio_source: str):
        # Initialize WAV file on first callback
        if self.wav_file is None:
            self.wav_file = wave.open(self.filename, 'wb')
            self.wav_file.setnchannels(audio_data.num_channels)
            self.wav_file.setsampwidth(audio_data.bits_per_sample // 8)
            self.wav_file.setframerate(audio_data.sample_rate)
        
        # Write audio frames to file
        self.wav_file.writeframes(audio_data.audio_frames)
    
    def close(self):
        if self.wav_file:
            self.wav_file.close()

# Usage
recorder = AudioRecorder("output.wav")
client = CallClient()
client.set_audio_renderer(
    participant_id="participant-id",
    callback=recorder.audio_callback,
    sample_rate=16000
)

# Later: stop recording
recorder.close()

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Example: Processing Audio Frames

from daily import CallClient, AudioData
import numpy as np

def process_audio(participant_id: str, audio_data: AudioData, audio_source: str):
    # Convert bytes to numpy array for processing
    # Assuming 16-bit PCM audio
    audio_array = np.frombuffer(
        audio_data.audio_frames,
        dtype=np.int16
    )
    
    # Reshape for multi-channel audio
    if audio_data.num_channels > 1:
        audio_array = audio_array.reshape(-1, audio_data.num_channels)
    
    # Process audio (e.g., compute volume)
    volume = np.abs(audio_array).mean()
    print(f"Average volume: {volume}")
    
    # Apply audio processing, filters, etc.
    # ...

client = CallClient()
client.set_audio_renderer(
    participant_id="participant-id",
    callback=process_audio,
    sample_rate=16000
)

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Notes

  • AudioData objects are read-only and provided by the SDK
  • The audio data format is interleaved PCM (Pulse Code Modulation)
  • For stereo audio, samples are interleaved: [L, R, L, R, …]
  • The total size of audio_frames in bytes is: num_audio_frames * num_channels * (bits_per_sample / 8)
  • Audio renderers are called at the interval specified by callback_interval_ms in set_audio_renderer()

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See Also