# Source: https://docs.pipecat.ai/server/utilities/audio/audio-buffer-processor.md

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# AudioBufferProcessor

> Process and buffer audio frames from conversations with flexible event handling

## Overview

The `AudioBufferProcessor` captures and buffers audio frames from both input (user) and output (bot) sources during conversations. It provides synchronized audio streams with configurable sample rates, supports both mono and stereo output, and offers flexible event handlers for various audio processing workflows.

## Constructor

```python  theme={null}
AudioBufferProcessor(
    sample_rate=None,
    num_channels=1,
    buffer_size=0,
    enable_turn_audio=False,
    **kwargs
)
```

### Parameters

<ParamField path="sample_rate" type="Optional[int]" default="None">
  The desired output sample rate in Hz. If `None`, uses the transport's sample
  rate from the `StartFrame`.
</ParamField>

<ParamField path="num_channels" type="int" default="1">
  Number of output audio channels:

  * `1`: Mono output (user and bot audio are mixed together)
  * `2`: Stereo output (user audio on left channel, bot audio on right channel)
</ParamField>

<ParamField path="buffer_size" type="int" default="0">
  Buffer size in bytes that triggers audio data events:

  * `0`: Events only trigger when recording stops
  * `>0`: Events trigger whenever buffer reaches this size (useful for chunked processing)
</ParamField>

<ParamField path="enable_turn_audio" type="bool" default="False">
  Whether to enable per-turn audio event handlers (`on_user_turn_audio_data` and
  `on_bot_turn_audio_data`).
</ParamField>

## Properties

### sample\_rate

```python  theme={null}
@property
def sample_rate(self) -> int
```

The current sample rate of the audio processor in Hz.

### num\_channels

```python  theme={null}
@property
def num_channels(self) -> int
```

The number of channels in the audio output (1 for mono, 2 for stereo).

## Methods

### start\_recording()

```python  theme={null}
async def start_recording()
```

Start recording audio from both user and bot sources. Initializes recording state and resets audio buffers.

### stop\_recording()

```python  theme={null}
async def stop_recording()
```

Stop recording and trigger final audio data handlers with any remaining buffered audio.

### has\_audio()

```python  theme={null}
def has_audio() -> bool
```

Check if both user and bot audio buffers contain data.

**Returns:** `True` if both buffers contain audio data.

## Event Handlers

The processor supports multiple event handlers for different audio processing workflows. Register handlers using the `@processor.event_handler()` decorator.

### on\_audio\_data

Triggered when `buffer_size` is reached or recording stops, providing merged audio.

```python  theme={null}
@audiobuffer.event_handler("on_audio_data")
async def on_audio_data(buffer, audio: bytes, sample_rate: int, num_channels: int):
    # Handle merged audio data
    pass
```

**Parameters:**

* `buffer`: The AudioBufferProcessor instance
* `audio`: Merged audio data (format depends on `num_channels` setting)
* `sample_rate`: Sample rate in Hz
* `num_channels`: Number of channels (1 or 2)

### on\_track\_audio\_data

Triggered alongside `on_audio_data`, providing separate user and bot audio tracks.

```python  theme={null}
@audiobuffer.event_handler("on_track_audio_data")
async def on_track_audio_data(buffer, user_audio: bytes, bot_audio: bytes,
                             sample_rate: int, num_channels: int):
    # Handle separate audio tracks
    pass
```

**Parameters:**

* `buffer`: The AudioBufferProcessor instance
* `user_audio`: Raw user audio bytes (always mono)
* `bot_audio`: Raw bot audio bytes (always mono)
* `sample_rate`: Sample rate in Hz
* `num_channels`: Always 1 for individual tracks

### on\_user\_turn\_audio\_data

Triggered when a user speaking turn ends. Requires `enable_turn_audio=True`.

```python  theme={null}
@audiobuffer.event_handler("on_user_turn_audio_data")
async def on_user_turn_audio_data(buffer, audio: bytes, sample_rate: int, num_channels: int):
    # Handle user turn audio
    pass
```

**Parameters:**

* `buffer`: The AudioBufferProcessor instance
* `audio`: Audio data from the user's speaking turn
* `sample_rate`: Sample rate in Hz
* `num_channels`: Always 1 (mono)

### on\_bot\_turn\_audio\_data

Triggered when a bot speaking turn ends. Requires `enable_turn_audio=True`.

```python  theme={null}
@audiobuffer.event_handler("on_bot_turn_audio_data")
async def on_bot_turn_audio_data(buffer, audio: bytes, sample_rate: int, num_channels: int):
    # Handle bot turn audio
    pass
```

**Parameters:**

* `buffer`: The AudioBufferProcessor instance
* `audio`: Audio data from the bot's speaking turn
* `sample_rate`: Sample rate in Hz
* `num_channels`: Always 1 (mono)

## Audio Processing Features

* **Automatic resampling**: Converts incoming audio to the specified sample rate
* **Buffer synchronization**: Aligns user and bot audio streams temporally
* **Silence insertion**: Fills gaps in non-continuous audio streams to maintain timing
* **Turn tracking**: Monitors speaking turns when `enable_turn_audio=True`

## Integration Notes

### STT Audio Passthrough

If using an STT service in your pipeline, enable audio passthrough to make audio available to the AudioBufferProcessor:

```python  theme={null}
stt = DeepgramSTTService(
    api_key=os.getenv("DEEPGRAM_API_KEY"),
    audio_passthrough=True,
)
```

<Note>`audio_passthrough` is enabled by default.</Note>

### Pipeline Placement

Add the AudioBufferProcessor after `transport.output()` to capture both user and bot audio:

```python  theme={null}
pipeline = Pipeline([
    transport.input(),
    # ... other processors ...
    transport.output(),
    audiobuffer,  # Place after audio output
    # ... remaining processors ...
])
```