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twiliohandle interruption (#422)

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* add interuption handler in twilio serializer

* fix autopep8

* revert ruff autoformatting

* address pr comments

* change interruption frame to user started frame in serializer

* remove overrrident handle interrupt

* remove unused import

* change userstarted to interuption frame
This commit is contained in:
Aashraya
2024-09-02 23:36:38 +05:30
committed by GitHub
parent aba5f89174
commit 51cd7fd285
2 changed files with 34 additions and 18 deletions
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21
src/pipecat/transports/network/fastapi_websocket.py
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@@ -12,8 +12,8 @@ import wave
from typing import Awaitable, Callable
from pydantic.main import BaseModel
from pipecat.frames.frames import AudioRawFrame, CancelFrame, EndFrame, StartFrame
from pipecat.processors.frame_processor import FrameProcessor
from pipecat.frames.frames import AudioRawFrame, CancelFrame, EndFrame, Frame, StartFrame, StartInterruptionFrame
from pipecat.processors.frame_processor import FrameDirection, FrameProcessor
from pipecat.serializers.base_serializer import FrameSerializer
from pipecat.transports.base_input import BaseInputTransport
from pipecat.transports.base_output import BaseOutputTransport
@@ -93,11 +93,18 @@ class FastAPIWebsocketOutputTransport(BaseOutputTransport):
self._params = params
self._websocket_audio_buffer = bytes()
async def process_frame(self, frame: Frame, direction: FrameDirection):
await super().process_frame(frame, direction)
if isinstance(frame, StartInterruptionFrame):
await self._write_frame(frame)
async def write_raw_audio_frames(self, frames: bytes):
self._websocket_audio_buffer += frames
while len(self._websocket_audio_buffer) >= self._params.audio_frame_size:
frame = AudioRawFrame(
audio=self._websocket_audio_buffer[:self._params.audio_frame_size],
audio=self._websocket_audio_buffer[:
self._params.audio_frame_size],
sample_rate=self._params.audio_out_sample_rate,
num_channels=self._params.audio_out_channels
)
@@ -121,7 +128,13 @@ class FastAPIWebsocketOutputTransport(BaseOutputTransport):
if payload and self._websocket.client_state == WebSocketState.CONNECTED:
await self._websocket.send_text(payload)
self._websocket_audio_buffer = self._websocket_audio_buffer[self._params.audio_frame_size:]
self._websocket_audio_buffer = self._websocket_audio_buffer[
self._params.audio_frame_size:]
async def _write_frame(self, frame: Frame):
payload = self._params.serializer.serialize(frame)
if payload and self._websocket.client_state == WebSocketState.CONNECTED:
await self._websocket.send_text(payload)
class FastAPIWebsocketTransport(BaseTransport):