add background_noise service and example

This commit is contained in:
daniil
2024-10-02 12:51:17 +03:00
committed by Aleix Conchillo Flaqué
parent f83381860c
commit cff62650ee
2 changed files with 220 additions and 0 deletions

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#
# Copyright (c) 2024, Daily
#
# SPDX-License-Identifier: BSD 2-Clause License
#
import os
import sys
from pipecat.frames.frames import EndFrame, LLMMessagesFrame
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineParams, PipelineTask
from pipecat.processors.aggregators.llm_response import (
LLMAssistantResponseAggregator,
LLMUserResponseAggregator,
)
from pipecat.services.background_noise import BackgroundNoiseEffect
from pipecat.services.cartesia import CartesiaTTSService
from pipecat.services.openai import OpenAILLMService
from pipecat.services.deepgram import DeepgramSTTService
from pipecat.transports.network.fastapi_websocket import (
FastAPIWebsocketTransport,
FastAPIWebsocketParams,
)
from pipecat.vad.silero import SileroVADAnalyzer
from pipecat.serializers.twilio import TwilioFrameSerializer
from loguru import logger
from dotenv import load_dotenv
load_dotenv(override=True)
logger.remove(0)
logger.add(sys.stderr, level="DEBUG")
async def run_bot(websocket_client, stream_sid):
transport = FastAPIWebsocketTransport(
websocket=websocket_client,
params=FastAPIWebsocketParams(
audio_out_enabled=True,
add_wav_header=False,
vad_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
vad_audio_passthrough=True,
serializer=TwilioFrameSerializer(stream_sid),
),
)
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="79a125e8-cd45-4c13-8a67-188112f4dd22", # British Lady
)
messages = [
{
"role": "system",
"content": "You are a helpful LLM in an audio call. Your goal is to demonstrate your capabilities in a succinct way. Your output will be converted to audio so don't include special characters in your answers. Respond to what the user said in a creative and helpful way.",
},
]
tma_in = LLMUserResponseAggregator(messages)
tma_out = LLMAssistantResponseAggregator(messages)
background_noise = BackgroundNoiseEffect(websocket_client, stream_sid, "your_path_to_audio_in_format_pcm16000")
pipeline = Pipeline(
[
transport.input(), # Websocket input from client
stt, # Speech-To-Text
tma_in, # User responses
llm, # LLM
tts, # Text-To-Speech
background_noise,
transport.output(), # Websocket output to client
tma_out, # LLM responses
]
)
task = PipelineTask(pipeline, params=PipelineParams(allow_interruptions=True))
@transport.event_handler("on_client_connected")
async def on_client_connected(transport, client):
# Kick off the conversation.
messages.append({"role": "system", "content": "Please introduce yourself to the user."})
await task.queue_frames([LLMMessagesFrame(messages)])
@transport.event_handler("on_client_disconnected")
async def on_client_disconnected(transport, client):
await task.queue_frames([EndFrame()])
runner = PipelineRunner(handle_sigint=False)
await runner.run(task)

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import asyncio
import json
import time
from asyncio import sleep
from io import BytesIO
import loguru
from pipecat.processors.frame_processor import FrameProcessor, FrameDirection
from pydub import AudioSegment
from pipecat.frames.frames import AudioRawFrame, OutputAudioRawFrame, Frame, BotStartedSpeakingFrame, \
BotStoppedSpeakingFrame, EndFrame
class BackgroundNoiseEffect(FrameProcessor):
def __init__(self, websocket_client, stream_sid, music_path):
super().__init__(sync=False)
self._speaking = True
self._audio_task = self.get_event_loop().create_task(self._audio_task_handler())
self._audio_queue = asyncio.Queue()
self._stop = False
self.stream_sid = stream_sid
self.websocket_client = websocket_client
self.music_path = music_path
self.get_music_part_gen = self._get_music_part()
self.emptied = False
async def process_frame(self, frame: Frame, direction: FrameDirection):
await super().process_frame(frame, direction)
if isinstance(frame, BotStartedSpeakingFrame):
self._speaking = True
if isinstance(frame, BotStoppedSpeakingFrame):
self._speaking = False
self.emptied = False
if isinstance(frame, AudioRawFrame) and self._speaking:
if not self.emptied:
self.emptied = True
buffer_clear_message = {"event": "clear", "streamSid": self.stream_sid}
await self.websocket_client.send_text(json.dumps(buffer_clear_message))
frame.audio = self._combine_with_music(frame)
if isinstance(frame, EndFrame):
self._stop = True
await self.push_frame(frame, direction)
def _combine_with_music(self, frame: AudioRawFrame):
"""
Combines small raw audio segments from the frame with chunks of a music file.
"""
small_audio_bytes = frame.audio
music_audio = AudioSegment.from_wav(self.music_path)
music_audio = music_audio - 15
music_position = 0
small_audio = AudioSegment(
data=small_audio_bytes,
sample_width=2,
frame_rate=16000,
channels=1
)
small_audio_length = len(small_audio)
music_chunk = music_audio[music_position:music_position + small_audio_length]
if len(music_chunk) < small_audio_length:
music_position = 0
music_chunk += music_audio[:small_audio_length - len(music_chunk)]
combined_audio = music_chunk.overlay(small_audio)
music_position += small_audio_length
output_buffer = BytesIO()
try:
combined_audio.export(output_buffer, format="raw")
return output_buffer.getvalue()
finally:
output_buffer.close()
def _get_music_part(self):
"""
Generator that yields chunks of background music audio.
"""
music_audio = AudioSegment.from_wav(self.music_path)
music_audio = music_audio - 15
music_position = 0
small_audio_length = 6400
while True:
if music_position + small_audio_length > len(music_audio):
music_chunk = music_audio[music_position:] + music_audio[
:(music_position + small_audio_length) % len(music_audio)]
music_position = (music_position + small_audio_length) % len(music_audio)
else:
music_chunk = music_audio[music_position:music_position + small_audio_length]
music_position += small_audio_length
output_buffer = BytesIO()
try:
music_chunk.export(output_buffer, format="raw")
frame = OutputAudioRawFrame(audio=output_buffer.getvalue(), sample_rate=16000, num_channels=1)
yield frame
finally:
output_buffer.close()
async def _audio_task_handler(self):
while True:
await sleep(0.005)
if self._stop:
break
if not self._speaking:
frame = next(self.get_music_part_gen)
await self.push_frame(frame, FrameDirection.DOWNSTREAM)