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:
@@ -4,14 +4,10 @@
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# SPDX-License-Identifier: BSD 2-Clause License
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#
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import asyncio
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import os
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import sys
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import aiohttp
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from dotenv import load_dotenv
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from loguru import logger
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from runner import configure
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from pipecat.frames.frames import EndFrame, LLMMessagesFrame
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from pipecat.pipeline.pipeline import Pipeline
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@@ -19,46 +15,51 @@ from pipecat.pipeline.runner import PipelineRunner
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from pipecat.pipeline.task import PipelineTask
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from pipecat.services.cartesia.tts import CartesiaTTSService
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from pipecat.services.openai.llm import OpenAILLMService
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from pipecat.transports.services.daily import DailyParams, DailyTransport
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from pipecat.transports.base_transport import TransportParams
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from pipecat.transports.network.small_webrtc import SmallWebRTCTransport
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from pipecat.transports.network.webrtc_connection import SmallWebRTCConnection
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load_dotenv(override=True)
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logger.remove(0)
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logger.add(sys.stderr, level="DEBUG")
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async def run_bot(webrtc_connection: SmallWebRTCConnection):
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logger.info(f"Starting bot")
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async def main():
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async with aiohttp.ClientSession() as session:
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(room_url, _) = await configure(session)
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# Create a transport using the WebRTC connection
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transport = SmallWebRTCTransport(
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webrtc_connection=webrtc_connection,
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params=TransportParams(
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audio_out_enabled=True,
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),
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)
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transport = DailyTransport(
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room_url, None, "Say One Thing From an LLM", DailyParams(audio_out_enabled=True)
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)
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tts = CartesiaTTSService(
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api_key=os.getenv("CARTESIA_API_KEY"),
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voice_id="71a7ad14-091c-4e8e-a314-022ece01c121", # British Reading Lady
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)
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tts = CartesiaTTSService(
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api_key=os.getenv("CARTESIA_API_KEY"),
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voice_id="71a7ad14-091c-4e8e-a314-022ece01c121", # British Reading Lady
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)
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llm = OpenAILLMService(api_key=os.getenv("OPENAI_API_KEY"), model="gpt-4o")
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llm = OpenAILLMService(api_key=os.getenv("OPENAI_API_KEY"), model="gpt-4o")
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messages = [
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{
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"role": "system",
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"content": "You are an LLM in a WebRTC session, and this is a 'hello world' demo. Say hello to the world.",
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}
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]
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messages = [
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{
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"role": "system",
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"content": "You are an LLM in a WebRTC session, and this is a 'hello world' demo. Say hello to the world.",
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}
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]
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task = PipelineTask(Pipeline([llm, tts, transport.output()]))
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runner = PipelineRunner()
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# Register an event handler so we can play the audio when the client joins
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@transport.event_handler("on_client_connected")
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async def on_client_connected(transport, client):
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await task.queue_frames([LLMMessagesFrame(messages), EndFrame()])
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task = PipelineTask(Pipeline([llm, tts, transport.output()]))
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runner = PipelineRunner(handle_sigint=False)
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@transport.event_handler("on_first_participant_joined")
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async def on_first_participant_joined(transport, participant):
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await task.queue_frames([LLMMessagesFrame(messages), EndFrame()])
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await runner.run(task)
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await runner.run(task)
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if __name__ == "__main__":
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asyncio.run(main())
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from run import main
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main()
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