diff --git a/src/pipecat/services/assemblyai/stt.py b/src/pipecat/services/assemblyai/stt.py index 0618a29fe..445ab7b73 100644 --- a/src/pipecat/services/assemblyai/stt.py +++ b/src/pipecat/services/assemblyai/stt.py @@ -13,7 +13,7 @@ WebSocket API for streaming audio transcription. import asyncio import json from dataclasses import dataclass, field -from typing import Any, AsyncGenerator, Dict, Mapping, Optional +from typing import Any, AsyncGenerator, Dict, Optional from urllib.parse import urlencode from loguru import logger @@ -211,18 +211,6 @@ class AssemblyAISTTService(WebsocketSTTService): self._user_speaking = False self._vad_speaking = False - # Log final connection params after any modifications - logger.info(f"{self} Final connection params being sent to AssemblyAI:") - logger.info(f" min_end_of_turn_silence_when_confident: {self._settings.connection_params.min_end_of_turn_silence_when_confident}") - logger.info(f" max_turn_silence: {self._settings.connection_params.max_turn_silence}") - - # Warn if min_end_of_turn_silence_when_confident is not 100ms - if self._settings.connection_params.min_end_of_turn_silence_when_confident != 100: - logger.warning( - f"For best latency, set min_end_of_turn_silence_when_confident to 100ms. " - f"Current value: {self._settings.connection_params.min_end_of_turn_silence_when_confident}ms" - ) - def _configure_pipecat_turn_mode( self, connection_params: AssemblyAIConnectionParams, is_u3_pro: bool ) -> AssemblyAIConnectionParams: @@ -461,13 +449,7 @@ class AssemblyAISTTService(WebsocketSTTService): if params: query_string = urlencode(params) - full_url = f"{self._api_endpoint_base_url}?{query_string}" - logger.info(f"{self} WebSocket URL being sent to AssemblyAI:") - logger.info(f" {full_url}") - logger.info(f" Parsed params:") - for k, v in params.items(): - logger.info(f" {k}: {v}") - return full_url + return f"{self._api_endpoint_base_url}?{query_string}" return self._api_endpoint_base_url async def _connect(self): @@ -672,17 +654,6 @@ class AssemblyAISTTService(WebsocketSTTService): - end_of_turn → TranscriptionFrame + UserStoppedSpeakingFrame - else → InterimTranscriptionFrame """ - # Log transcript details - logger.info(f"{self} ===== TRANSCRIPT RECEIVED =====") - logger.info(f" Text: \"{message.transcript}\"") - logger.info(f" end_of_turn: {message.end_of_turn}") - logger.info(f" turn_is_formatted: {message.turn_is_formatted}") - logger.info(f" turn_order: {message.turn_order}") - if message.end_of_turn_confidence is not None: - logger.info(f" end_of_turn_confidence: {message.end_of_turn_confidence}") - logger.info(f" speaker: {message.speaker}") - logger.info(f"===============================") - if not message.transcript: return @@ -709,11 +680,6 @@ class AssemblyAISTTService(WebsocketSTTService): speaker=message.speaker, text=message.transcript ) - logger.info(f"{self} 🤖 TEXT SENT TO LLM (with speaker format): \"{transcript_text}\"") - else: - logger.info(f"{self} 🤖 TEXT SENT TO LLM (speaker {message.speaker}): \"{transcript_text}\"") - else: - logger.info(f"{self} 🤖 TEXT SENT TO LLM: \"{transcript_text}\"") # Determine if this is a final turn from AssemblyAI is_final_turn = message.end_of_turn and (