Updating foundation examples to use SmallWebRTCTransport and pipecat-ai-small-webrtc-prebuilt (#1534)

Co-authored-by: Filipi Fuchter <filipi@daily.co>
This commit is contained in:
Mark Backman
2025-04-11 19:44:16 -04:00
committed by GitHub
parent 8186219879
commit f6accbd510
120 changed files with 7989 additions and 7179 deletions

View File

@@ -4,21 +4,18 @@
# SPDX-License-Identifier: BSD 2-Clause License
#
import asyncio
import os
import sys
import wave
import aiohttp
from dotenv import load_dotenv
from loguru import logger
from runner import configure
from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.frames.frames import (
Frame,
LLMFullResponseEndFrame,
OutputAudioRawFrame,
TTSSpeakFrame,
)
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
@@ -30,14 +27,14 @@ from pipecat.processors.aggregators.openai_llm_context import (
from pipecat.processors.frame_processor import FrameDirection, FrameProcessor
from pipecat.processors.logger import FrameLogger
from pipecat.services.cartesia.tts import CartesiaTTSService
from pipecat.services.deepgram.stt import DeepgramSTTService
from pipecat.services.openai.llm import OpenAILLMService
from pipecat.transports.services.daily import DailyParams, DailyTransport
from pipecat.transports.base_transport import TransportParams
from pipecat.transports.network.small_webrtc import SmallWebRTCTransport
from pipecat.transports.network.webrtc_connection import SmallWebRTCConnection
load_dotenv(override=True)
logger.remove(0)
logger.add(sys.stderr, level="DEBUG")
sounds = {}
sound_files = ["ding1.wav", "ding2.wav"]
@@ -80,70 +77,83 @@ class InboundSoundEffectWrapper(FrameProcessor):
await self.push_frame(frame, direction)
async def main():
async with aiohttp.ClientSession() as session:
(room_url, token) = await configure(session)
async def run_bot(webrtc_connection: SmallWebRTCConnection):
logger.info(f"Starting bot")
transport = DailyTransport(
room_url,
token,
"Respond bot",
DailyParams(
audio_out_enabled=True,
transcription_enabled=True,
vad_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
)
transport = SmallWebRTCTransport(
webrtc_connection=webrtc_connection,
params=TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
vad_audio_passthrough=True,
),
)
llm = OpenAILLMService(api_key=os.getenv("OPENAI_API_KEY"), model="gpt-4o")
stt = DeepgramSTTService(api_key=os.getenv("DEEPGRAM_API_KEY"))
tts = CartesiaTTSService(
api_key=os.getenv("CARTESIA_API_KEY"),
voice_id="71a7ad14-091c-4e8e-a314-022ece01c121", # British Reading Lady
)
llm = OpenAILLMService(api_key=os.getenv("OPENAI_API_KEY"), model="gpt-4o")
messages = [
{
"role": "system",
"content": "You are a helpful LLM in a WebRTC call. Your goal is to demonstrate your capabilities in a succinct way. Your output will be converted to audio. Respond to what the user said in a creative and helpful way.",
},
tts = CartesiaTTSService(
api_key=os.getenv("CARTESIA_API_KEY"),
voice_id="71a7ad14-091c-4e8e-a314-022ece01c121", # British Reading Lady
)
messages = [
{
"role": "system",
"content": "You are a helpful LLM in a WebRTC call. Your goal is to demonstrate your capabilities in a succinct way. Your output will be converted to audio. Respond to what the user said in a creative and helpful way.",
},
]
context = OpenAILLMContext(messages)
context_aggregator = llm.create_context_aggregator(context)
out_sound = OutboundSoundEffectWrapper()
in_sound = InboundSoundEffectWrapper()
fl = FrameLogger("LLM Out")
fl2 = FrameLogger("Transcription In")
pipeline = Pipeline(
[
transport.input(),
stt,
context_aggregator.user(),
in_sound,
fl2,
llm,
fl,
tts,
out_sound,
transport.output(),
context_aggregator.assistant(),
]
)
context = OpenAILLMContext(messages)
context_aggregator = llm.create_context_aggregator(context)
out_sound = OutboundSoundEffectWrapper()
in_sound = InboundSoundEffectWrapper()
fl = FrameLogger("LLM Out")
fl2 = FrameLogger("Transcription In")
task = PipelineTask(pipeline)
pipeline = Pipeline(
[
transport.input(),
context_aggregator.user(),
in_sound,
fl2,
llm,
fl,
tts,
out_sound,
transport.output(),
context_aggregator.assistant(),
]
)
@transport.event_handler("on_client_connected")
async def on_client_connected(transport, client):
logger.info(f"Client connected")
# Kick off the conversation.
await task.queue_frame(TTSSpeakFrame("Hi, I'm listening!"))
await transport.send_audio(sounds["ding1.wav"])
@transport.event_handler("on_first_participant_joined")
async def on_first_participant_joined(transport, participant):
await transport.capture_participant_transcription(participant["id"])
await tts.say("Hi, I'm listening!")
await transport.send_audio(sounds["ding1.wav"])
@transport.event_handler("on_client_disconnected")
async def on_client_disconnected(transport, client):
logger.info(f"Client disconnected")
runner = PipelineRunner()
@transport.event_handler("on_client_closed")
async def on_client_closed(transport, client):
logger.info(f"Client closed connection")
await task.cancel()
task = PipelineTask(pipeline)
runner = PipelineRunner(handle_sigint=False)
await runner.run(task)
await runner.run(task)
if __name__ == "__main__":
asyncio.run(main())
from run import main
main()