Fixing the issue when using the TavusTransport with some TTS services.
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@@ -11,6 +11,8 @@ from pydantic import BaseModel
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from pipecat.audio.utils import create_default_resampler
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from pipecat.frames.frames import (
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BotStartedSpeakingFrame,
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BotStoppedSpeakingFrame,
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CancelFrame,
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EndFrame,
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Frame,
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@@ -20,8 +22,6 @@ from pipecat.frames.frames import (
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StartInterruptionFrame,
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TransportMessageFrame,
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TransportMessageUrgentFrame,
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TTSStartedFrame,
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TTSStoppedFrame,
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)
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from pipecat.processors.frame_processor import FrameDirection, FrameProcessor, FrameProcessorSetup
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from pipecat.transports.base_input import BaseInputTransport
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@@ -365,7 +365,8 @@ class TavusOutputTransport(BaseOutputTransport):
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self._client = client
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self._params = params
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self._samples_sent = 0
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self._start_time = time.time()
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self._start_time = None
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self._current_idx_str: Optional[str] = None
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async def setup(self, setup: FrameProcessorSetup):
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await super().setup(setup)
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@@ -377,8 +378,6 @@ class TavusOutputTransport(BaseOutputTransport):
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async def start(self, frame: StartFrame):
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await super().start(frame)
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self._samples_sent = 0
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self._start_time = time.time()
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await self._client.start(frame)
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await self.set_transport_ready(frame)
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@@ -394,25 +393,41 @@ class TavusOutputTransport(BaseOutputTransport):
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logger.info(f"TavusOutputTransport sending message {frame}")
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await self._client.send_message(frame)
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async def push_frame(self, frame: Frame, direction: FrameDirection = FrameDirection.DOWNSTREAM):
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# The BotStartedSpeakingFrame and BotStoppedSpeakingFrame are created inside BaseOutputTransport
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# so TavusOutputTransport never receives these frames.
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# This is a workaround, so we can more reliably be aware when the bot has started or stopped speaking
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if direction == FrameDirection.DOWNSTREAM:
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if isinstance(frame, BotStartedSpeakingFrame):
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if self._current_idx_str is not None:
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logger.warning("TavusOutputTransport self._current_idx_str is already defined!")
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self._current_idx_str = str(frame.id)
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self._start_time = time.time()
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self._samples_sent = 0
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elif isinstance(frame, BotStoppedSpeakingFrame):
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await self._client.encode_audio_and_send(b"\x00\x00", True, self._current_idx_str)
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self._current_idx_str = None
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await super().push_frame(frame, direction)
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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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if isinstance(frame, StartInterruptionFrame):
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await self._handle_interruptions()
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elif isinstance(frame, TTSStartedFrame):
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self._current_idx_str = str(frame.id)
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elif isinstance(frame, TTSStoppedFrame):
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logger.debug(f"TAVUS: {self}: stopped speaking")
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await self._client.encode_audio_and_send(b"\x00\x00", True, self._current_idx_str)
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async def _handle_interruptions(self):
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await self._client.send_interrupt_message()
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async def write_raw_audio_frames(self, frames: bytes, destination: Optional[str] = None):
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# Compute wait time for synchronization
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wait = self._start_time + (self._samples_sent / self._sample_rate) - time.time()
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wait = self._start_time + (self._samples_sent / self.sample_rate) - time.time()
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if wait > 0:
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logger.trace(f"TavusOutputTransport write_raw_audio_frames wait: {wait}")
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await asyncio.sleep(wait)
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if self._current_idx_str is None:
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logger.warning("TavusOutputTransport self._current_idx_str not defined yet!")
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return
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await self._client.encode_audio_and_send(frames, False, self._current_idx_str)
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# Update timestamp based on number of samples sent
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