Merge pull request #1129 from pipecat-ai/instant_voice_demo

Pipecat improvements for the instant voice demo
This commit is contained in:
Filipi da Silva Fuchter
2025-02-12 11:53:40 -03:00
committed by GitHub
2 changed files with 40 additions and 6 deletions

View File

@@ -6,6 +6,7 @@
import asyncio
import io
import json
import time
import wave
from typing import Awaitable, Callable, Optional
@@ -21,6 +22,9 @@ from pipecat.frames.frames import (
OutputAudioRawFrame,
StartFrame,
StartInterruptionFrame,
TextFrame,
TransportMessageFrame,
TransportMessageUrgentFrame,
)
from pipecat.processors.frame_processor import FrameDirection
from pipecat.serializers.base_serializer import FrameSerializer
@@ -46,6 +50,7 @@ class WebsocketServerCallbacks(BaseModel):
on_client_connected: Callable[[websockets.WebSocketServerProtocol], Awaitable[None]]
on_client_disconnected: Callable[[websockets.WebSocketServerProtocol], Awaitable[None]]
on_session_timeout: Callable[[websockets.WebSocketServerProtocol], Awaitable[None]]
on_websocket_ready: Callable[[], Awaitable[None]]
class WebsocketServerInputTransport(BaseInputTransport):
@@ -96,6 +101,7 @@ class WebsocketServerInputTransport(BaseInputTransport):
async def _server_task_handler(self):
logger.info(f"Starting websocket server on {self._host}:{self._port}")
async with websockets.serve(self._client_handler, self._host, self._port) as server:
await self._callbacks.on_websocket_ready()
await self._stop_server_event.wait()
async def _client_handler(self, websocket: websockets.WebSocketServerProtocol, path):
@@ -186,6 +192,12 @@ class WebsocketServerOutputTransport(BaseOutputTransport):
await self._write_frame(frame)
self._next_send_time = 0
async def send_message(self, frame: TransportMessageFrame | TransportMessageUrgentFrame):
message_frame = TextFrame(
text=json.dumps(frame.message),
)
await self._write_frame(message_frame)
async def write_raw_audio_frames(self, frames: bytes):
if not self._websocket:
# Simulate audio playback with a sleep.
@@ -254,6 +266,7 @@ class WebsocketServerTransport(BaseTransport):
on_client_connected=self._on_client_connected,
on_client_disconnected=self._on_client_disconnected,
on_session_timeout=self._on_session_timeout,
on_websocket_ready=self._on_websocket_ready,
)
self._input: Optional[WebsocketServerInputTransport] = None
self._output: Optional[WebsocketServerOutputTransport] = None
@@ -264,6 +277,7 @@ class WebsocketServerTransport(BaseTransport):
self._register_event_handler("on_client_connected")
self._register_event_handler("on_client_disconnected")
self._register_event_handler("on_session_timeout")
self._register_event_handler("on_websocket_ready")
def input(self) -> WebsocketServerInputTransport:
if not self._input:
@@ -293,3 +307,6 @@ class WebsocketServerTransport(BaseTransport):
async def _on_session_timeout(self, websocket):
await self._call_event_handler("on_session_timeout", websocket)
async def _on_websocket_ready(self):
await self._call_event_handler("on_websocket_ready")

View File

@@ -5,6 +5,7 @@
#
import asyncio
import base64
import time
import warnings
from concurrent.futures import ThreadPoolExecutor
@@ -13,9 +14,6 @@ from typing import Any, Awaitable, Callable, Mapping, Optional
import aiohttp
from daily import (
CallClient,
Daily,
EventHandler,
VirtualCameraDevice,
VirtualMicrophoneDevice,
VirtualSpeakerDevice,
@@ -291,6 +289,7 @@ class DailyTransportClient(EventHandler):
self._transcription_ids = []
self._transcription_status = None
self._joining = False
self._joined = False
self._joined_event = asyncio.Event()
self._leave_counter = 0
@@ -423,13 +422,14 @@ class DailyTransportClient(EventHandler):
)
async def join(self):
# Transport already joined, ignore.
if self._joined:
# Transport already joined or joining, ignore.
if self._joined or self._joining:
# Increment leave counter if we already joined.
self._leave_counter += 1
return
logger.info(f"Joining {self._room_url}")
self._joining = True
# For performance reasons, never subscribe to video streams (unless a
# video renderer is registered).
@@ -444,6 +444,7 @@ class DailyTransportClient(EventHandler):
if not error:
self._joined = True
self._joining = False
# Increment leave counter if we successfully joined.
self._leave_counter += 1
@@ -462,6 +463,7 @@ class DailyTransportClient(EventHandler):
except asyncio.TimeoutError:
error_msg = f"Time out joining {self._room_url}"
logger.error(error_msg)
self._joining = False
await self._callbacks.on_error(error_msg)
async def _start_transcription(self):
@@ -1198,7 +1200,22 @@ class DailyTransport(BaseTransport):
async def _on_app_message(self, message: Any, sender: str):
if self._input:
await self._input.push_app_message(message, sender)
if message["type"] in {"raw-audio", "raw-audio-batch"}:
data = message["data"]
audio_list = data.get(
"base64AudioBatch", [data.get("base64Audio")]
) # Ensure a list
for base64_audio in filter(None, audio_list): # Filter out None values
pcm_bytes = base64.b64decode(base64_audio)
frame = InputAudioRawFrame(
audio=pcm_bytes,
sample_rate=data["sampleRate"],
num_channels=data["numChannels"],
)
await self._input.push_audio_frame(frame)
else:
await self._input.push_app_message(message, sender)
await self._call_event_handler("on_app_message", message, sender)
async def _on_call_state_updated(self, state: str):