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# | ||
# Copyright (c) 2024, Daily | ||
# | ||
# SPDX-License-Identifier: BSD 2-Clause License | ||
# | ||
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import asyncio | ||
import aiohttp | ||
import os | ||
import sys | ||
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from pipecat.frames.frames import EndFrame, TTSSpeakFrame | ||
from pipecat.pipeline.pipeline import Pipeline | ||
from pipecat.pipeline.task import PipelineTask | ||
from pipecat.pipeline.runner import PipelineRunner | ||
from pipecat.services.riva import FastpitchTTSService | ||
from pipecat.transports.services.daily import DailyParams, DailyTransport | ||
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from runner import configure | ||
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from loguru import logger | ||
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from dotenv import load_dotenv | ||
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load_dotenv(override=True) | ||
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logger.remove(0) | ||
logger.add(sys.stderr, level="DEBUG") | ||
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async def main(): | ||
async with aiohttp.ClientSession() as session: | ||
(room_url, _) = await configure(session) | ||
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transport = DailyTransport( | ||
room_url, None, "Say One Thing", DailyParams(audio_out_enabled=True) | ||
) | ||
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tts = FastpitchTTSService(api_key=os.getenv("NVIDIA_API_KEY")) | ||
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runner = PipelineRunner() | ||
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task = PipelineTask(Pipeline([tts, transport.output()])) | ||
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# Register an event handler so we can play the audio when the | ||
# participant joins. | ||
@transport.event_handler("on_first_participant_joined") | ||
async def on_first_participant_joined(transport, participant): | ||
participant_name = participant.get("info", {}).get("userName", "") | ||
await task.queue_frames([TTSSpeakFrame(f"Aloha, {participant_name}!"), EndFrame()]) | ||
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await runner.run(task) | ||
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if __name__ == "__main__": | ||
asyncio.run(main()) |
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# | ||
# Copyright (c) 2024, Daily | ||
# | ||
# SPDX-License-Identifier: BSD 2-Clause License | ||
# | ||
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import asyncio | ||
import os | ||
import sys | ||
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import aiohttp | ||
from dotenv import load_dotenv | ||
from loguru import logger | ||
from runner import configure | ||
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from pipecat.audio.vad.silero import SileroVADAnalyzer | ||
from pipecat.frames.frames import LLMMessagesFrame | ||
from pipecat.pipeline.pipeline import Pipeline | ||
from pipecat.pipeline.runner import PipelineRunner | ||
from pipecat.pipeline.task import PipelineParams, PipelineTask | ||
from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext | ||
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from pipecat.services.deepgram import DeepgramSTTService | ||
from pipecat.services.riva import FastpitchTTSService, ParakeetSTTService | ||
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from pipecat.services.openai import OpenAILLMService | ||
from pipecat.transports.services.daily import DailyParams, DailyTransport | ||
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load_dotenv(override=True) | ||
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logger.remove(0) | ||
logger.add(sys.stderr, level="DEBUG") | ||
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async def main(): | ||
async with aiohttp.ClientSession() as session: | ||
(room_url, _) = await configure(session) | ||
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transport = DailyTransport( | ||
room_url, | ||
None, | ||
"Respond bot", | ||
DailyParams( | ||
audio_out_enabled=True, | ||
vad_enabled=True, | ||
vad_analyzer=SileroVADAnalyzer(), | ||
vad_audio_passthrough=True, | ||
), | ||
) | ||
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# stt = DeepgramSTTService(api_key=os.getenv("DEEPGRAM_API_KEY")) | ||
stt = ParakeetSTTService(api_key=os.getenv("NVIDIA_API_KEY")) | ||
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tts = FastpitchTTSService(api_key=os.getenv("NVIDIA_API_KEY")) | ||
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llm = OpenAILLMService(api_key=os.getenv("OPENAI_API_KEY"), model="gpt-4o") | ||
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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 so don't include special characters in your answers. Respond to what the user said in a creative and helpful way.", | ||
}, | ||
] | ||
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context = OpenAILLMContext(messages) | ||
context_aggregator = llm.create_context_aggregator(context) | ||
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pipeline = Pipeline( | ||
[ | ||
transport.input(), # Transport user input | ||
stt, # STT | ||
context_aggregator.user(), # User responses | ||
llm, # LLM | ||
tts, # TTS | ||
transport.output(), # Transport bot output | ||
context_aggregator.assistant(), # Assistant spoken responses | ||
] | ||
) | ||
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task = PipelineTask(pipeline, PipelineParams(allow_interruptions=True)) | ||
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@transport.event_handler("on_first_participant_joined") | ||
async def on_first_participant_joined(transport, participant): | ||
# Kick off the conversation. | ||
messages.append({"role": "system", "content": "Please introduce yourself to the user."}) | ||
await task.queue_frames([LLMMessagesFrame(messages)]) | ||
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runner = PipelineRunner() | ||
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await runner.run(task) | ||
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if __name__ == "__main__": | ||
asyncio.run(main()) |
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# | ||
# Copyright (c) 2024, Daily | ||
# | ||
# SPDX-License-Identifier: BSD 2-Clause License | ||
# | ||
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import asyncio | ||
from typing import AsyncGenerator, List, Optional, Union, Iterator | ||
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from loguru import logger | ||
from pydantic.main import BaseModel | ||
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from pipecat.frames.frames import ( | ||
CancelFrame, | ||
EndFrame, | ||
ErrorFrame, | ||
Frame, | ||
InterimTranscriptionFrame, | ||
StartFrame, | ||
TranscriptionFrame, | ||
TTSAudioRawFrame, | ||
TTSStartedFrame, | ||
TTSStoppedFrame, | ||
) | ||
from pipecat.services.ai_services import STTService, TTSService | ||
from pipecat.transcriptions.language import Language | ||
from pipecat.utils.time import time_now_iso8601 | ||
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try: | ||
import riva.client | ||
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except ModuleNotFoundError as e: | ||
logger.error(f"Exception: {e}") | ||
logger.error( | ||
"In order to use nvidia riva TTS or STT, you need to `pip install pipecat-ai[riva]`. Also, set `NVIDIA_API_KEY` environment variable." | ||
) | ||
raise Exception(f"Missing module: {e}") | ||
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class FastpitchTTSService(TTSService): | ||
class InputParams(BaseModel): | ||
language: Optional[str] = "en-US" | ||
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def __init__( | ||
self, | ||
*, | ||
api_key: str, | ||
server: str = "grpc.nvcf.nvidia.com:443", | ||
voice_id: str = "English-US.Female-1", | ||
sample_rate_hz: int = 24000, | ||
# nvidia riva calls this 'function-id' | ||
model: str = "0149dedb-2be8-4195-b9a0-e57e0e14f972", | ||
params: InputParams = InputParams(), | ||
**kwargs, | ||
): | ||
super().__init__(sample_rate=sample_rate_hz, **kwargs) | ||
self._api_key = api_key | ||
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self.set_model_name("fastpitch-hifigan-tts") | ||
self.set_voice(voice_id) | ||
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self.voice_id = voice_id | ||
self.sample_rate_hz = sample_rate_hz | ||
self.language_code = params.language | ||
self.nchannels = 1 | ||
self.sampwidth = 2 | ||
self.quality = None | ||
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metadata = [ | ||
["function-id", model], | ||
["authorization", f"Bearer {api_key}"], | ||
] | ||
auth = riva.client.Auth(None, True, server, metadata) | ||
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self.service = riva.client.SpeechSynthesisService(auth) | ||
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async def stop(self, frame: EndFrame): | ||
await super().stop(frame) | ||
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async def cancel(self, frame: CancelFrame): | ||
await super().cancel(frame) | ||
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async def run_tts(self, text: str) -> AsyncGenerator[Frame, None]: | ||
logger.debug(f"Generating TTS: [{text}]") | ||
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await self.start_ttfb_metrics() | ||
yield TTSStartedFrame() | ||
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try: | ||
custom_dictionary_input = {} | ||
responses = self.service.synthesize_online( | ||
text, | ||
self.voice_id, | ||
self.language_code, | ||
sample_rate_hz=self.sample_rate_hz, | ||
audio_prompt_file=None, | ||
quality=20 if self.quality is None else self.quality, | ||
custom_dictionary=custom_dictionary_input, | ||
) | ||
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for resp in responses: | ||
await self.stop_ttfb_metrics() | ||
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frame = TTSAudioRawFrame( | ||
audio=resp.audio, | ||
sample_rate=self.sample_rate_hz, | ||
num_channels=self.nchannels, | ||
) | ||
yield frame | ||
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except Exception as e: | ||
logger.error(f"{self} exception: {e}") | ||
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await self.start_tts_usage_metrics(text) | ||
yield TTSStoppedFrame() | ||
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class ParakeetSTTService(STTService): | ||
def __init__( | ||
self, | ||
*, | ||
api_key: str, | ||
server: str = "grpc.nvcf.nvidia.com:443", | ||
# nvidia calls this 'function-id' | ||
model: str = "1598d209-5e27-4d3c-8079-4751568b1081", | ||
**kwargs, | ||
): | ||
super().__init__(**kwargs) | ||
self._api_key = api_key | ||
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self.set_model_name("parakeet-ctc-1.1b-asr") | ||
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input_device = 0 | ||
list_devices = False | ||
profanity_filter = False | ||
automatic_punctuation = False | ||
no_verbatim_transcripts = False | ||
language_code = "en-US" | ||
model_name = "" | ||
boosted_lm_words = None | ||
boosted_lm_score = 4.0 | ||
speaker_diarization = False | ||
diarization_max_speakers = 3 | ||
start_history = -1 | ||
start_threshold = -1.0 | ||
stop_history = -1 | ||
stop_threshold = -1.0 | ||
stop_history_eou = -1 | ||
stop_threshold_eou = -1.0 | ||
custom_configuration = "" | ||
server = "grpc.nvcf.nvidia.com:443" | ||
ssl_cert = None | ||
use_ssl = True | ||
sample_rate_hz: int = 16000 | ||
file_streaming_chunk = 1600 | ||
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metadata = [ | ||
["function-id", model], | ||
["authorization", f"Bearer {api_key}"], | ||
] | ||
auth = riva.client.Auth(None, True, server, metadata) | ||
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self.asr_service = riva.client.ASRService(auth) | ||
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config = riva.client.StreamingRecognitionConfig( | ||
config=riva.client.RecognitionConfig( | ||
encoding=riva.client.AudioEncoding.LINEAR_PCM, | ||
language_code=language_code, | ||
model=model_name, | ||
max_alternatives=1, | ||
profanity_filter=profanity_filter, | ||
enable_automatic_punctuation=automatic_punctuation, | ||
verbatim_transcripts=not no_verbatim_transcripts, | ||
sample_rate_hertz=sample_rate_hz, | ||
audio_channel_count=1, | ||
), | ||
interim_results=True, | ||
) | ||
self.config = config | ||
riva.client.add_word_boosting_to_config(config, boosted_lm_words, boosted_lm_score) | ||
riva.client.add_endpoint_parameters_to_config( | ||
config, | ||
start_history, | ||
start_threshold, | ||
stop_history, | ||
stop_history_eou, | ||
stop_threshold, | ||
stop_threshold_eou, | ||
) | ||
riva.client.add_custom_configuration_to_config(config, custom_configuration) | ||
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def can_generate_metrics(self) -> bool: | ||
return False | ||
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async def start(self, frame: StartFrame): | ||
await super().start(frame) | ||
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async def stop(self, frame: EndFrame): | ||
await super().stop(frame) | ||
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async def cancel(self, frame: CancelFrame): | ||
await super().cancel(frame) | ||
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async def run_stt(self, audio: bytes) -> AsyncGenerator[Frame, None]: | ||
responses = self.asr_service.streaming_response_generator( | ||
audio_chunks=audio, | ||
streaming_config=self.config, | ||
) | ||
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for response in responses: | ||
if not response.results: | ||
continue | ||
partial_transcript = "" | ||
for result in response.results: | ||
if result: | ||
if not result.alternatives: | ||
continue | ||
transcript = result.alternatives[0].transcript | ||
if transcript: | ||
language = None | ||
if len(transcript) > 0: | ||
await self.stop_ttfb_metrics() | ||
if result.is_final: | ||
await self.stop_processing_metrics() | ||
yield TranscriptionFrame( | ||
transcript, "", time_now_iso8601(), language | ||
) | ||
else: | ||
yield InterimTranscriptionFrame( | ||
transcript, "", time_now_iso8601(), language | ||
) | ||
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yield None | ||
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async def _on_speech_started(self, *args, **kwargs): | ||
await self.start_ttfb_metrics() | ||
await self.start_processing_metrics() |