Files
pipecat/examples/video-avatar/video-avatar-tavus-video-service.py
2026-05-20 19:03:01 -03:00

184 lines
6.5 KiB
Python

#
# Copyright (c) 2024-2026, Daily
#
# SPDX-License-Identifier: BSD 2-Clause License
#
import datetime
import io
import os
import wave
import aiofiles
import aiohttp
from dotenv import load_dotenv
from loguru import logger
from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.frames.frames import LLMRunFrame
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineParams, PipelineTask
from pipecat.processors.aggregators.llm_context import LLMContext
from pipecat.processors.aggregators.llm_response_universal import (
LLMContextAggregatorPair,
LLMUserAggregatorParams,
)
from pipecat.processors.audio.audio_buffer_processor import AudioBufferProcessor
from pipecat.runner.types import RunnerArguments
from pipecat.runner.utils import create_transport
from pipecat.services.cartesia.tts import CartesiaTTSService
from pipecat.services.deepgram.stt import DeepgramSTTService
from pipecat.services.google.llm import GoogleLLMService
from pipecat.services.tavus.video import TavusVideoService
from pipecat.transports.base_transport import BaseTransport, TransportParams
from pipecat.transports.daily.transport import DailyParams
load_dotenv(override=True)
async def save_audio_file(audio: bytes, filename: str, sample_rate: int, num_channels: int):
"""Save audio data to a WAV file."""
if len(audio) > 0:
with io.BytesIO() as buffer:
with wave.open(buffer, "wb") as wf:
wf.setsampwidth(2)
wf.setnchannels(num_channels)
wf.setframerate(sample_rate)
wf.writeframes(audio)
async with aiofiles.open(filename, "wb") as file:
await file.write(buffer.getvalue())
logger.info(f"Audio saved to {filename}")
# We use lambdas to defer transport parameter creation until the transport
# type is selected at runtime.
transport_params = {
"daily": lambda: DailyParams(
audio_in_enabled=True,
audio_out_enabled=True,
video_out_enabled=True,
video_out_is_live=True,
video_out_width=1280,
video_out_height=720,
audio_out_auto_silence=False,
),
"webrtc": lambda: TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
video_out_enabled=True,
video_out_is_live=True,
video_out_width=1280,
video_out_height=720,
audio_out_auto_silence=False,
),
}
async def run_bot(transport: BaseTransport, runner_args: RunnerArguments):
logger.info(f"Starting bot")
async with aiohttp.ClientSession() as session:
stt = DeepgramSTTService(api_key=os.environ["DEEPGRAM_API_KEY"], audio_passthrough=True)
tts = CartesiaTTSService(
api_key=os.environ["CARTESIA_API_KEY"],
settings=CartesiaTTSService.Settings(
voice="a167e0f3-df7e-4d52-a9c3-f949145efdab",
),
)
llm = GoogleLLMService(
api_key=os.environ["GOOGLE_API_KEY"],
settings=GoogleLLMService.Settings(
system_instruction="You are a helpful assistant in a voice conversation. Your responses will be spoken aloud, so avoid emojis, bullet points, or other formatting that can't be spoken. Respond to what the user said in a creative, helpful, and brief way.",
),
)
tavus = TavusVideoService(
api_key=os.environ["TAVUS_API_KEY"],
replica_id=os.environ["TAVUS_REPLICA_ID"],
session=session,
)
context = LLMContext()
user_aggregator, assistant_aggregator = LLMContextAggregatorPair(
context,
user_params=LLMUserAggregatorParams(vad_analyzer=SileroVADAnalyzer()),
)
audiobuffer = AudioBufferProcessor()
pipeline = Pipeline(
[
transport.input(), # Transport user input
stt, # STT
user_aggregator, # User responses
llm, # LLM
tts, # TTS
tavus, # Tavus output layer
transport.output(), # Transport bot output
audiobuffer, # Audio recording
assistant_aggregator, # Assistant spoken responses
]
)
task = PipelineTask(
pipeline,
params=PipelineParams(
audio_in_sample_rate=16000,
audio_out_sample_rate=24000,
enable_metrics=True,
enable_usage_metrics=True,
),
idle_timeout_secs=runner_args.pipeline_idle_timeout_secs,
)
@transport.event_handler("on_client_connected")
async def on_client_connected(transport, client):
logger.info(f"Client connected")
await audiobuffer.start_recording()
# Kick off the conversation.
context.add_message(
{
"role": "user",
"content": "Start by greeting the user and ask how you can help.",
}
)
await task.queue_frames([LLMRunFrame()])
@transport.event_handler("on_client_disconnected")
async def on_client_disconnected(transport, client):
logger.info(f"Client disconnected")
await task.cancel()
@audiobuffer.event_handler("on_audio_data")
async def on_audio_data(buffer, audio, sample_rate, num_channels):
timestamp = datetime.datetime.now().strftime("%Y%m%d_%H%M%S")
filename = f"recordings/merged_{timestamp}.wav"
os.makedirs("recordings", exist_ok=True)
await save_audio_file(audio, filename, sample_rate, num_channels)
@audiobuffer.event_handler("on_track_audio_data")
async def on_track_audio_data(buffer, user_audio, bot_audio, sample_rate, num_channels):
timestamp = datetime.datetime.now().strftime("%Y%m%d_%H%M%S")
os.makedirs("recordings", exist_ok=True)
await save_audio_file(user_audio, f"recordings/user_{timestamp}.wav", sample_rate, 1)
await save_audio_file(bot_audio, f"recordings/bot_{timestamp}.wav", sample_rate, 1)
runner = PipelineRunner(handle_sigint=runner_args.handle_sigint)
await runner.run(task)
async def bot(runner_args: RunnerArguments):
"""Main bot entry point compatible with Pipecat Cloud."""
transport = await create_transport(runner_args, transport_params)
await run_bot(transport, runner_args)
if __name__ == "__main__":
from pipecat.runner.run import main
main()