feat: integra VAD ao controle de turnos da conversa
This commit is contained in:
32
src/voice_assistant/application/audio_history.py
Normal file
32
src/voice_assistant/application/audio_history.py
Normal file
@@ -0,0 +1,32 @@
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"""A bounded PCM history addressed by absolute input sample positions."""
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from voice_assistant.audio.models import CANONICAL_AUDIO_FORMAT
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class AudioHistory:
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def __init__(self, history_ms: int) -> None:
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if type(history_ms) is not int or history_ms <= 0:
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raise ValueError("history_ms must be a positive integer.")
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self._retained_samples = history_ms * CANONICAL_AUDIO_FORMAT.sample_rate // 1000
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self._audio = bytearray()
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self.start_sample = 0
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self.end_sample = 0
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def append(self, audio: bytes) -> None:
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self._audio.extend(audio)
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self.end_sample += len(audio) // CANONICAL_AUDIO_FORMAT.frame_size_bytes
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def read(self, start: int, end: int) -> bytes:
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if not self.start_sample <= start <= end <= self.end_sample:
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raise ValueError("Requested audio is outside the retained input history.")
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width = CANONICAL_AUDIO_FORMAT.frame_size_bytes
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return bytes(self._audio[(start - self.start_sample) * width:(end - self.start_sample) * width])
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def trim(self) -> None:
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start = max(self.start_sample, self.end_sample - self._retained_samples)
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del self._audio[:(start - self.start_sample) * CANONICAL_AUDIO_FORMAT.frame_size_bytes]
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self.start_sample = start
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def clear(self) -> None:
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self._audio.clear()
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self.start_sample = self.end_sample
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@@ -41,6 +41,18 @@ class SpeechChunk:
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audio_format: AudioFormat
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@dataclass(frozen=True, slots=True)
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class TurnOpened:
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session_id: str
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turn_id: int
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@dataclass(frozen=True, slots=True)
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class TurnInterrupted:
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session_id: str
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turn_id: int
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@dataclass(frozen=True, slots=True)
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class TurnCompleted:
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session_id: str
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@@ -58,5 +70,5 @@ class SessionFailure:
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type VoiceEvent = (
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TranscriptPartial | TranscriptFinal | AgentTextDelta | AgentReply
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| SpeechChunk | TurnCompleted | SessionFailure
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| SpeechChunk | TurnOpened | TurnInterrupted | TurnCompleted | SessionFailure
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)
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@@ -6,7 +6,12 @@ from collections.abc import Awaitable, Callable
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from contextlib import aclosing
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from voice_assistant.agent.adapter import StreamingAgentAdapter
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from voice_assistant.application.audio_history import AudioHistory
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from voice_assistant.vad.adapter import VADAdapter
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from voice_assistant.vad.models import SpeechStarted, SpeechEnded, VADEvent
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from voice_assistant.application.events import (
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TurnOpened,
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TurnInterrupted,
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AgentReply,
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AgentTextDelta,
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SessionFailure,
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@@ -18,6 +23,7 @@ from voice_assistant.application.events import (
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)
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from voice_assistant.audio.models import AudioFormat, CANONICAL_AUDIO_FORMAT
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from voice_assistant.stt.adapter import STTAdapter
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from voice_assistant.turn_manager import TurnState
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from voice_assistant.voice_pipeline import (
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PipelineAgentResponse,
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PipelineAudioChunk,
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@@ -64,9 +70,30 @@ class VoiceSession:
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emit: EventSink,
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stt_stop_timeout: float = 10.0,
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finalization_cancel_timeout: float = 5.0,
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vad: VADAdapter | None = None,
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pre_roll_ms: int = 200,
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history_ms: int = 2000,
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max_input_chunk_bytes: int = 65536,
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) -> None:
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if stt_stop_timeout <= 0 or finalization_cancel_timeout <= 0:
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raise ValueError("Session timeouts must be positive.")
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if type(pre_roll_ms) is not int or pre_roll_ms < 0:
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raise ValueError("pre_roll_ms must be a nonnegative integer.")
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if type(history_ms) is not int or history_ms <= 0 or pre_roll_ms > history_ms:
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raise ValueError("history_ms must be positive and cover pre_roll_ms.")
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if type(max_input_chunk_bytes) is not int or max_input_chunk_bytes <= 0:
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raise ValueError("max_input_chunk_bytes must be a positive integer.")
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self._vad = vad
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self._input_lock = asyncio.Lock()
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self._history = AudioHistory(history_ms)
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self._pre_roll_samples = pre_roll_ms * CANONICAL_AUDIO_FORMAT.sample_rate // 1000
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self._max_input_chunk_bytes = max_input_chunk_bytes
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self._input_failed = False
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self._input_finished = False
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self._closing = False
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self._capture_turn_id: int | None = None
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self._sent_until = 0
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self._last_voice_end = 0
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self._pipeline = pipeline
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self._create_stt = create_stt
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self._emit = emit
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@@ -116,7 +143,17 @@ class VoiceSession:
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"SESSION_ALREADY_CLOSED", "This session is already closed."
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)
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def _require_manual_input(self) -> None:
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if self._vad is not None:
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raise VoiceSessionError(
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"TURN_CONTROLLED_BY_VAD", "This session uses server-side turn detection.",
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)
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async def start_turn(self) -> int:
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self._require_manual_input()
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return await self._start_turn()
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async def _start_turn(self) -> int:
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self._require_open()
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if self.active_turn_id is not None:
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raise VoiceSessionError(
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@@ -168,6 +205,12 @@ class VoiceSession:
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async def send_audio(
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self, turn_id: int, sequence_number: int, audio: bytes,
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) -> None:
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self._require_manual_input()
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await self._send_audio(turn_id, sequence_number, audio)
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async def _send_audio(
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self, turn_id: int, sequence_number: int, audio: bytes,
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) -> None:
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self.validate_audio_input(turn_id)
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CANONICAL_AUDIO_FORMAT.validate_chunk(audio)
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@@ -187,6 +230,10 @@ class VoiceSession:
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) from exc
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def stop_turn(self, turn_id: int) -> None:
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self._require_manual_input()
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self._stop_turn(turn_id)
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def _stop_turn(self, turn_id: int) -> None:
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self._require_open()
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if self._stt is None:
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raise RuntimeError("STT session has not been started.")
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@@ -200,6 +247,12 @@ class VoiceSession:
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)
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async def interrupt(self, turn_id: int) -> None:
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async with self._input_lock:
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await self._interrupt(turn_id)
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if self._capture_turn_id == turn_id:
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self._capture_turn_id = None
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async def _interrupt(self, turn_id: int) -> None:
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self._require_open()
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if not self._pipeline.is_active_turn(turn_id):
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raise VoiceSessionError(
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@@ -221,6 +274,11 @@ class VoiceSession:
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) from exc
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async def end(self) -> None:
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self._closing = True
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async with self._input_lock:
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await self._end()
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async def _end(self) -> None:
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self._require_open()
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try:
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async with self._turn_transition_lock:
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@@ -306,6 +364,12 @@ class VoiceSession:
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or not self._pipeline.is_active_turn(turn_id)
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):
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return
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turn = self._pipeline.turn_manager.active_turn
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if self._vad is not None and turn is not None and turn.state == TurnState.LISTENING:
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# A VAD trigger may produce no recognizable transcript.
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self._pipeline.interrupt_if_active(turn_id)
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await self._emit(TurnInterrupted(self.session_id, turn_id))
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return
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self._pipeline.complete_turn(turn_id)
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self._completed_turns.add(turn_id)
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await self._emit(TurnCompleted(self.session_id, turn_id))
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@@ -389,6 +453,11 @@ class VoiceSession:
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self._stt_events_task = None
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async def aclose(self) -> None:
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self._closing = True
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async with self._input_lock:
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await self._close_resources()
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async def _close_resources(self) -> None:
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"""Attempt every cleanup operation, including after partial startup.
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The caller must shield this awaited cleanup from transport-scope
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@@ -425,3 +494,113 @@ class VoiceSession:
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"Failed to close %s provider. session_id=%s",
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name, self.session_id,
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)
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if self._vad is not None:
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try:
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await self._vad.aclose()
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except Exception:
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logger.exception("Failed to close VAD. session_id=%s", self.session_id)
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self._history.clear()
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self._capture_turn_id = None
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def _require_continuous_input(self) -> VADAdapter:
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self._require_open()
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if self._vad is None:
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raise VoiceSessionError("INVALID_INPUT_MODE", "Continuous input requires VAD.")
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if self._closing or self._input_finished:
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raise VoiceSessionError("INPUT_CLOSED", "Continuous audio input is closed.")
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if self._input_failed:
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raise VoiceSessionError("VOICE_INPUT_FAILED", "Create a new session after input failure.")
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return self._vad
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async def receive_audio(self, audio: bytes) -> None:
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"""Receive continuous PCM independently of client-managed turn IDs.
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The channel serializes and validates its own packet sequence. This
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operation validates PCM before advancing the application's timeline.
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"""
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CANONICAL_AUDIO_FORMAT.validate_chunk(audio)
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if len(audio) > self._max_input_chunk_bytes:
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raise ValueError("Continuous audio chunk exceeds its configured size limit.")
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async with self._input_lock:
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vad = self._require_continuous_input()
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self._history.append(audio)
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try:
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events = await vad.process(audio)
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if self._closing:
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raise VoiceSessionError("INPUT_CLOSED", "Session shutdown is in progress.")
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await self._handle_voice_activity(events)
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await self._forward_input_until(self._history.end_sample)
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except asyncio.CancelledError:
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self._input_failed = True
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raise
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except Exception as exc:
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self._input_failed = True
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if isinstance(exc, VoiceSessionError):
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raise
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raise VoiceSessionError(
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"VOICE_INPUT_FAILED", str(exc), turn_id=self.active_turn_id,
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) from exc
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finally:
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self._history.trim()
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async def finish_audio_input(self) -> None:
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"""Finish a normal input stream; session end/disconnect aborts instead."""
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async with self._input_lock:
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vad = self._require_continuous_input()
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try:
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events = await vad.flush()
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if self._closing:
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raise VoiceSessionError("INPUT_CLOSED", "Session shutdown is in progress.")
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await self._handle_voice_activity(events)
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self._input_finished = True
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except asyncio.CancelledError:
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self._input_failed = True
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raise
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except Exception as exc:
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self._input_failed = True
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if isinstance(exc, VoiceSessionError):
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raise
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raise VoiceSessionError(
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"VOICE_INPUT_FAILED", str(exc), turn_id=self.active_turn_id,
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) from exc
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finally:
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self._history.trim()
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async def _handle_voice_activity(self, events: tuple[VADEvent, ...]) -> None:
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for event in events:
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if isinstance(event, SpeechStarted):
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if self._capture_turn_id is not None:
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raise ValueError("VAD emitted a duplicate speech start.")
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if not self._history.start_sample <= event.sample_offset < self._history.end_sample:
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raise ValueError("Speech start is outside retained audio; increase history_ms.")
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previous = self.active_turn_id
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if previous is not None:
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await self._interrupt(previous)
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await self._emit(TurnInterrupted(self.session_id, previous))
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else:
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await self._cancel_finalization_task()
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turn_id = await self._start_turn()
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self._capture_turn_id = turn_id
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self._sent_until = max(
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self._history.start_sample, self._last_voice_end,
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event.sample_offset - self._pre_roll_samples,
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)
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await self._emit(TurnOpened(self.session_id, turn_id))
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elif isinstance(event, SpeechEnded):
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if not 0 <= event.sample_offset <= self._history.end_sample:
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raise ValueError("Speech end is outside input audio.")
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self._last_voice_end = event.sample_offset
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if self._capture_turn_id is not None:
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await self._forward_input_until(event.sample_offset)
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turn_id = self._capture_turn_id
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self._capture_turn_id = None
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self._stop_turn(turn_id)
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async def _forward_input_until(self, end_sample: int) -> None:
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turn_id = self._capture_turn_id
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if turn_id is None or end_sample <= self._sent_until:
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return
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audio = self._history.read(self._sent_until, end_sample)
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await self._send_audio(turn_id, self.expected_audio_sequence, audio)
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self._sent_until = end_sample
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@@ -13,7 +13,7 @@ from pydantic import BaseModel, ValidationError
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from voice_assistant.application.events import (
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AgentReply, AgentTextDelta, SessionFailure, SpeechChunk,
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TranscriptFinal, TranscriptPartial, TurnCompleted, VoiceEvent,
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TranscriptFinal, TranscriptPartial, TurnCompleted, TurnOpened, TurnInterrupted, VoiceEvent,
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)
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from voice_assistant.application.voice_session import VoiceSession, VoiceSessionError
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from voice_assistant.audio.models import CANONICAL_AUDIO_FORMAT
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@@ -26,9 +26,8 @@ from voice_assistant.audio_gateway.app.protocol import (
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from voice_assistant.config.loader import load_config, load_provider_catalog
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from voice_assistant.domain.session import SessionState
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from voice_assistant.providers import (
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create_agent_adapter, create_stt_adapter, create_tts_adapter,
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create_agent_adapter, create_stt_adapter, create_tts_adapter, create_vad_adapter,
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)
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from voice_assistant.tts.adapter import TTSAdapter
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from voice_assistant.turn_manager import TurnManager
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from voice_assistant.voice_pipeline import VoicePipeline
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@@ -81,6 +80,10 @@ async def send_voice_event(websocket: WebSocket, event: VoiceEvent) -> None:
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payload = AgentDelta(session_id=sid, turn_id=tid, text=text)
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case AgentReply(session_id=sid, turn_id=tid, text=text):
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payload = AgentResponse(session_id=sid, turn_id=tid, text=text)
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case TurnOpened(session_id=sid, turn_id=tid):
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payload = TurnStarted(session_id=sid, turn_id=tid)
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case TurnInterrupted(session_id=sid, turn_id=tid):
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payload = Interrupted(session_id=sid, turn_id=tid)
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case TurnCompleted(session_id=sid, turn_id=tid):
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payload = TurnEnded(session_id=sid, turn_id=tid)
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case SessionFailure():
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@@ -93,20 +96,34 @@ async def send_voice_event(websocket: WebSocket, event: VoiceEvent) -> None:
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await send_json(websocket, payload)
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async def create_voice_session(websocket: WebSocket, session_id: str | None) -> VoiceSession:
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"""Compose providers at the channel boundary, injecting application ports."""
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async def create_voice_session(
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websocket: WebSocket, session_id: str | None, *, server_vad: bool = False,
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) -> VoiceSession:
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"""Compose manual input or continuous input controlled by backend VAD."""
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state = SessionState(session_id=session_id) if session_id is not None else SessionState()
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agent = create_agent_adapter(config=app_config, catalog=provider_catalog)
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tts: TTSAdapter | None = None
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agent = None
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tts = None
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vad = None
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try:
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agent = create_agent_adapter(config=app_config, catalog=provider_catalog)
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tts = create_tts_adapter(config=app_config, catalog=provider_catalog)
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if server_vad:
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vad = create_vad_adapter(config=app_config, catalog=provider_catalog)
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pipeline = VoicePipeline(
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session=state, turn_manager=create_turn_manager(), agent=agent, tts=tts,
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)
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return VoiceSession(
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pipeline=pipeline,
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create_stt=partial(create_stt_adapter, config=app_config, catalog=provider_catalog),
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emit=partial(send_voice_event, websocket),
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stt_stop_timeout=STT_STOP_TIMEOUT_SECONDS,
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finalization_cancel_timeout=FINALIZATION_CANCEL_TIMEOUT_SECONDS,
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vad=vad,
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**app_config.voice_input,
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)
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except BaseException:
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# A partially composed application has not yet acquired ownership.
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with anyio.CancelScope(shield=True):
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for name, provider in (("Agent", agent), ("TTS", tts)):
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for name, provider in (("Agent", agent), ("TTS", tts), ("VAD", vad)):
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if provider is None:
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continue
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try:
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@@ -116,13 +133,6 @@ async def create_voice_session(websocket: WebSocket, session_id: str | None) ->
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"Failed to close %s after session composition failure.", name,
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)
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raise
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return VoiceSession(
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pipeline=pipeline,
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create_stt=partial(create_stt_adapter, config=app_config, catalog=provider_catalog),
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emit=partial(send_voice_event, websocket),
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stt_stop_timeout=STT_STOP_TIMEOUT_SECONDS,
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finalization_cancel_timeout=FINALIZATION_CANCEL_TIMEOUT_SECONDS,
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)
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@app.websocket("/ws/audio")
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@@ -72,3 +72,4 @@ class AppConfig(BaseModel):
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tts: ComponentConfig
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agent: ComponentConfig
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vad: ComponentConfig | None = None
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voice_input: dict[str, Any] = Field(default_factory=dict)
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Reference in New Issue
Block a user