Merge pull request #304 from pipecat-ai/khk/cartesia-continue
Cartesia streaming (WebSocket) and word-level timestamps support
This commit is contained in:
@@ -37,6 +37,7 @@ async def main(room_url: str, token):
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token,
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token,
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"Respond bot",
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"Respond bot",
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DailyParams(
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DailyParams(
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audio_out_sample_rate=44100,
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audio_out_enabled=True,
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audio_out_enabled=True,
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transcription_enabled=True,
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transcription_enabled=True,
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vad_enabled=True,
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vad_enabled=True,
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@@ -47,6 +48,7 @@ async def main(room_url: str, token):
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tts = CartesiaTTSService(
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tts = CartesiaTTSService(
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api_key=os.getenv("CARTESIA_API_KEY"),
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api_key=os.getenv("CARTESIA_API_KEY"),
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voice_id="a0e99841-438c-4a64-b679-ae501e7d6091", # Barbershop Man
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voice_id="a0e99841-438c-4a64-b679-ae501e7d6091", # Barbershop Man
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sample_rate=44100,
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)
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)
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llm = OpenAILLMService(
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llm = OpenAILLMService(
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@@ -68,11 +70,11 @@ async def main(room_url: str, token):
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tma_in, # User responses
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tma_in, # User responses
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llm, # LLM
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llm, # LLM
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tts, # TTS
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tts, # TTS
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tma_out, # Goes before the transport because cartesia has word-level timestamps!
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transport.output(), # Transport bot output
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transport.output(), # Transport bot output
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tma_out # Assistant spoken responses
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])
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])
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task = PipelineTask(pipeline, PipelineParams(allow_interruptions=True))
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task = PipelineTask(pipeline, PipelineParams(allow_interruptions=True, enable_metrics=True))
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@transport.event_handler("on_first_participant_joined")
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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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async def on_first_participant_joined(transport, participant):
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@@ -36,7 +36,7 @@ Website = "https://pipecat.ai"
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[project.optional-dependencies]
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[project.optional-dependencies]
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anthropic = [ "anthropic~=0.28.1" ]
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anthropic = [ "anthropic~=0.28.1" ]
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azure = [ "azure-cognitiveservices-speech~=1.38.0" ]
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azure = [ "azure-cognitiveservices-speech~=1.38.0" ]
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cartesia = [ "cartesia~=1.0.3" ]
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cartesia = [ "websockets~=12.0" ]
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daily = [ "daily-python~=0.10.1" ]
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daily = [ "daily-python~=0.10.1" ]
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deepgram = [ "deepgram-sdk~=3.2.7" ]
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deepgram = [ "deepgram-sdk~=3.2.7" ]
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examples = [ "python-dotenv~=1.0.0", "flask~=3.0.3", "flask_cors~=4.0.1" ]
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examples = [ "python-dotenv~=1.0.0", "flask~=3.0.3", "flask_cors~=4.0.1" ]
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@@ -136,9 +136,16 @@ class LLMService(AIService):
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class TTSService(AIService):
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class TTSService(AIService):
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def __init__(self, *, aggregate_sentences: bool = True, **kwargs):
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def __init__(
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self,
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*,
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aggregate_sentences: bool = True,
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# if True, subclass is responsible for pushing TextFrames and LLMFullResponseEndFrames
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push_text_frames: bool = True,
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**kwargs):
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super().__init__(**kwargs)
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super().__init__(**kwargs)
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self._aggregate_sentences: bool = aggregate_sentences
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self._aggregate_sentences: bool = aggregate_sentences
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self._push_text_frames: bool = push_text_frames
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self._current_sentence: str = ""
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self._current_sentence: str = ""
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# Converts the text to audio.
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# Converts the text to audio.
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@@ -149,6 +156,10 @@ class TTSService(AIService):
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async def say(self, text: str):
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async def say(self, text: str):
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await self.process_frame(TextFrame(text=text), FrameDirection.DOWNSTREAM)
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await self.process_frame(TextFrame(text=text), FrameDirection.DOWNSTREAM)
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async def _handle_interruption(self, frame: StartInterruptionFrame, direction: FrameDirection):
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self._current_sentence = ""
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await self.push_frame(frame, direction)
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async def _process_text_frame(self, frame: TextFrame):
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async def _process_text_frame(self, frame: TextFrame):
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text: str | None = None
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text: str | None = None
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if not self._aggregate_sentences:
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if not self._aggregate_sentences:
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@@ -172,9 +183,10 @@ class TTSService(AIService):
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await self.process_generator(self.run_tts(text))
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await self.process_generator(self.run_tts(text))
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await self.stop_processing_metrics()
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await self.stop_processing_metrics()
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await self.push_frame(TTSStoppedFrame())
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await self.push_frame(TTSStoppedFrame())
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# We send the original text after the audio. This way, if we are
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if self._push_text_frames:
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# interrupted, the text is not added to the assistant context.
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# We send the original text after the audio. This way, if we are
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await self.push_frame(TextFrame(text))
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# interrupted, the text is not added to the assistant context.
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await self.push_frame(TextFrame(text))
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async def process_frame(self, frame: Frame, direction: FrameDirection):
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async def process_frame(self, frame: Frame, direction: FrameDirection):
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await super().process_frame(frame, direction)
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await super().process_frame(frame, direction)
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@@ -182,12 +194,15 @@ class TTSService(AIService):
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if isinstance(frame, TextFrame):
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if isinstance(frame, TextFrame):
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await self._process_text_frame(frame)
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await self._process_text_frame(frame)
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elif isinstance(frame, StartInterruptionFrame):
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elif isinstance(frame, StartInterruptionFrame):
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self._current_sentence = ""
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await self._handle_interruption(frame, direction)
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await self.push_frame(frame, direction)
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elif isinstance(frame, LLMFullResponseEndFrame) or isinstance(frame, EndFrame):
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elif isinstance(frame, LLMFullResponseEndFrame) or isinstance(frame, EndFrame):
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self._current_sentence = ""
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self._current_sentence = ""
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await self._push_tts_frames(self._current_sentence)
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await self._push_tts_frames(self._current_sentence)
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await self.push_frame(frame)
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if isinstance(frame, LLMFullResponseEndFrame):
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if self._push_text_frames:
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await self.push_frame(frame, direction)
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else:
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await self.push_frame(frame, direction)
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else:
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else:
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await self.push_frame(frame, direction)
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await self.push_frame(frame, direction)
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@@ -4,15 +4,37 @@
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# SPDX-License-Identifier: BSD 2-Clause License
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# SPDX-License-Identifier: BSD 2-Clause License
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#
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#
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from cartesia import AsyncCartesia
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import json
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import uuid
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import base64
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import asyncio
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import time
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from typing import AsyncGenerator
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from typing import AsyncGenerator
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from pipecat.frames.frames import AudioRawFrame, CancelFrame, EndFrame, Frame, StartFrame
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from pipecat.processors.frame_processor import FrameDirection
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from pipecat.frames.frames import (
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Frame,
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AudioRawFrame,
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StartInterruptionFrame,
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StartFrame,
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EndFrame,
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TextFrame,
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LLMFullResponseEndFrame
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)
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from pipecat.services.ai_services import TTSService
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from pipecat.services.ai_services import TTSService
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from loguru import logger
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from loguru import logger
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# See .env.example for Cartesia configuration needed
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try:
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import websockets
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except ModuleNotFoundError as e:
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logger.error(f"Exception: {e}")
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logger.error(
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"In order to use Cartesia, you need to `pip install pipecat-ai[cartesia]`. Also, set `CARTESIA_API_KEY` environment variable.")
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raise Exception(f"Missing module: {e}")
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class CartesiaTTSService(TTSService):
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class CartesiaTTSService(TTSService):
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@@ -21,13 +43,30 @@ class CartesiaTTSService(TTSService):
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*,
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*,
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api_key: str,
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api_key: str,
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voice_id: str,
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voice_id: str,
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cartesia_version: str = "2024-06-10",
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url: str = "wss://api.cartesia.ai/tts/websocket",
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model_id: str = "sonic-english",
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model_id: str = "sonic-english",
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encoding: str = "pcm_s16le",
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encoding: str = "pcm_s16le",
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sample_rate: int = 16000,
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sample_rate: int = 16000,
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language: str = "en",
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**kwargs):
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**kwargs):
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super().__init__(**kwargs)
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super().__init__(**kwargs)
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# Aggregating sentences still gives cleaner-sounding results and fewer
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# artifacts than streaming one word at a time. On average, waiting for
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# a full sentence should only "cost" us 15ms or so with GPT-4o or a Llama 3
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# model, and it's worth it for the better audio quality.
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self._aggregate_sentences = True
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# we don't want to automatically push LLM response text frames, because the
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# context aggregators will add them to the LLM context even if we're
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# interrupted. cartesia gives us word-by-word timestamps. we can use those
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# to generate text frames ourselves aligned with the playout timing of the audio!
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self._push_text_frames = False
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self._api_key = api_key
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self._api_key = api_key
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self._cartesia_version = cartesia_version
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self._url = url
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self._voice_id = voice_id
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self._voice_id = voice_id
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self._model_id = model_id
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self._model_id = model_id
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self._output_format = {
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self._output_format = {
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@@ -35,42 +74,152 @@ class CartesiaTTSService(TTSService):
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"encoding": encoding,
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"encoding": encoding,
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"sample_rate": sample_rate,
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"sample_rate": sample_rate,
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}
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}
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self._client = None
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self._language = language
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self._websocket = None
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self._context_id = None
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self._context_id_start_timestamp = None
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self._timestamped_words_buffer = []
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self._receive_task = None
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self._context_appending_task = None
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self._waiting_for_ttfb = False
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def can_generate_metrics(self) -> bool:
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def can_generate_metrics(self) -> bool:
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return True
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return True
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async def start(self, frame: StartFrame):
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async def start(self, frame: StartFrame):
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try:
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await super().start(frame)
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self._client = AsyncCartesia(api_key=self._api_key)
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await self._connect()
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self._voice = self._client.voices.get(id=self._voice_id)
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except Exception as e:
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logger.exception(f"{self} initialization error: {e}")
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async def stop(self, frame: EndFrame):
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async def stop(self, frame: EndFrame):
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if self._client:
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await super().stop(frame)
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await self._client.close()
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await self._disconnect()
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async def cancel(self, frame: CancelFrame):
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async def _connect(self):
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if self._client:
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try:
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await self._client.close()
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self._websocket = await websockets.connect(
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f"{self._url}?api_key={self._api_key}&cartesia_version={self._cartesia_version}"
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)
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self._receive_task = self.get_event_loop().create_task(self._receive_task_handler())
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self._context_appending_task = self.get_event_loop().create_task(self._context_appending_task_handler())
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except Exception as e:
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logger.exception(f"{self} initialization error: {e}")
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self._websocket = None
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async def _disconnect(self):
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try:
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if self._context_appending_task:
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self._context_appending_task.cancel()
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self._context_appending_task = None
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if self._receive_task:
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self._receive_task.cancel()
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self._receive_task = None
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if self._websocket:
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ws = self._websocket
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self._websocket = None
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await ws.close()
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self._context_id = None
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self._context_id_start_timestamp = None
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self._timestamped_words_buffer = []
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self._waiting_for_ttfb = False
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await self.stop_all_metrics()
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except Exception as e:
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logger.exception(f"{self} error closing websocket: {e}")
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|
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async def _handle_interruption(self, frame: StartInterruptionFrame, direction: FrameDirection):
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|
await super()._handle_interruption(frame, direction)
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|
self._context_id = None
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self._context_id_start_timestamp = None
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self._timestamped_words_buffer = []
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await self.stop_all_metrics()
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await self.push_frame(LLMFullResponseEndFrame())
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|
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async def _receive_task_handler(self):
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|
try:
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async for message in self._websocket:
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|
msg = json.loads(message)
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|
# logger.debug(f"Received message: {msg['type']} {msg['context_id']}")
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|
if not msg or msg["context_id"] != self._context_id:
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|
continue
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|
if msg["type"] == "done":
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# unset _context_id but not the _context_id_start_timestamp because we are likely still
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# playing out audio and need the timestamp to set send context frames
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self._context_id = None
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self._timestamped_words_buffer.append(("LLMFullResponseEndFrame", 0))
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elif msg["type"] == "timestamps":
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|
# logger.debug(f"TIMESTAMPS: {msg}")
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self._timestamped_words_buffer.extend(
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|
list(zip(msg["word_timestamps"]["words"], msg["word_timestamps"]["end"]))
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)
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|
elif msg["type"] == "chunk":
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|
if not self._context_id_start_timestamp:
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|
self._context_id_start_timestamp = time.time()
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|
if self._waiting_for_ttfb:
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|
await self.stop_ttfb_metrics()
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|
self._waiting_for_ttfb = False
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|
frame = AudioRawFrame(
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|
audio=base64.b64decode(msg["data"]),
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|
sample_rate=self._output_format["sample_rate"],
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|
num_channels=1
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|
)
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|
await self.push_frame(frame)
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|
except Exception as e:
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|
logger.exception(f"{self} exception: {e}")
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|
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|
async def _context_appending_task_handler(self):
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|
try:
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|
while True:
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|
await asyncio.sleep(0.1)
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|
if not self._context_id_start_timestamp:
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|
continue
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|
elapsed_seconds = time.time() - self._context_id_start_timestamp
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|
# pop all words from self._timestamped_words_buffer that are older than the
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|
# elapsed time and print a message about them to the console
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|
while self._timestamped_words_buffer and self._timestamped_words_buffer[0][1] <= elapsed_seconds:
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|
word, timestamp = self._timestamped_words_buffer.pop(0)
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|
if word == "LLMFullResponseEndFrame" and timestamp == 0:
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|
await self.push_frame(LLMFullResponseEndFrame())
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|
continue
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|
# print(f"Word '{word}' with timestamp {timestamp:.2f}s has been spoken.")
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|
await self.push_frame(TextFrame(word))
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|
except Exception as e:
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|
logger.exception(f"{self} exception: {e}")
|
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|
|
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async def run_tts(self, text: str) -> AsyncGenerator[Frame, None]:
|
async def run_tts(self, text: str) -> AsyncGenerator[Frame, None]:
|
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logger.debug(f"Generating TTS: [{text}]")
|
logger.debug(f"Generating TTS: [{text}]")
|
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|
|
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try:
|
try:
|
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await self.start_ttfb_metrics()
|
if not self._websocket:
|
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|
await self._connect()
|
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|
|
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chunk_generator = await self._client.tts.sse(
|
if not self._waiting_for_ttfb:
|
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stream=True,
|
await self.start_ttfb_metrics()
|
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transcript=text,
|
self._waiting_for_ttfb = True
|
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voice_embedding=self._voice["embedding"],
|
|
||||||
model_id=self._model_id,
|
|
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output_format=self._output_format,
|
|
||||||
)
|
|
||||||
|
|
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async for chunk in chunk_generator:
|
if not self._context_id:
|
||||||
await self.stop_ttfb_metrics()
|
self._context_id = str(uuid.uuid4())
|
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yield AudioRawFrame(chunk["audio"], self._output_format["sample_rate"], 1)
|
|
||||||
|
msg = {
|
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|
"transcript": text + " ",
|
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|
"continue": True,
|
||||||
|
"context_id": self._context_id,
|
||||||
|
"model_id": self._model_id,
|
||||||
|
"voice": {
|
||||||
|
"mode": "id",
|
||||||
|
"id": self._voice_id
|
||||||
|
},
|
||||||
|
"output_format": self._output_format,
|
||||||
|
"language": self._language,
|
||||||
|
"add_timestamps": True,
|
||||||
|
}
|
||||||
|
# logger.debug(f"SENDING MESSAGE {json.dumps(msg)}")
|
||||||
|
try:
|
||||||
|
await self._websocket.send(json.dumps(msg))
|
||||||
|
except Exception as e:
|
||||||
|
logger.exception(f"{self} error sending message: {e}")
|
||||||
|
await self._disconnect()
|
||||||
|
await self._connect()
|
||||||
|
return
|
||||||
|
yield None
|
||||||
except Exception as e:
|
except Exception as e:
|
||||||
logger.exception(f"{self} exception: {e}")
|
logger.exception(f"{self} exception: {e}")
|
||||||
|
|||||||
Reference in New Issue
Block a user