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Create stt_google.py
Browse files- stt_google.py +143 -0
stt_google.py
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| 1 |
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"""
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| 2 |
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Google Cloud Speech-to-Text Implementation
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"""
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import os
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import asyncio
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from typing import AsyncIterator, Optional, List
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from google.cloud import speech_v1p1beta1 as speech
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from google.api_core import exceptions
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from utils import log
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from stt_interface import STTInterface, STTConfig, TranscriptionResult
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class GoogleCloudSTT(STTInterface):
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"""Google Cloud Speech-to-Text implementation"""
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def __init__(self, credentials_path: str):
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if credentials_path and os.path.exists(credentials_path):
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os.environ['GOOGLE_APPLICATION_CREDENTIALS'] = credentials_path
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log(f"✅ Google credentials set from: {credentials_path}")
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else:
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log("⚠️ Google credentials path not found, using default credentials")
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self.client = speech.SpeechAsyncClient()
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self.streaming_config = None
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self.is_streaming = False
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self.audio_queue = asyncio.Queue()
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async def start_streaming(self, config: STTConfig) -> None:
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"""Initialize streaming session"""
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try:
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recognition_config = speech.RecognitionConfig(
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encoding=self._get_encoding(config.encoding),
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sample_rate_hertz=config.sample_rate,
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language_code=config.language,
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enable_automatic_punctuation=config.enable_punctuation,
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enable_word_time_offsets=config.enable_word_timestamps,
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model=config.model,
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use_enhanced=config.use_enhanced,
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metadata=speech.RecognitionMetadata(
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interaction_type=speech.RecognitionMetadata.InteractionType.VOICE_SEARCH,
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recording_device_type=speech.RecognitionMetadata.RecordingDeviceType.PC,
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audio_topic="general"
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)
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)
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self.streaming_config = speech.StreamingRecognitionConfig(
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config=recognition_config,
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interim_results=config.interim_results,
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single_utterance=config.single_utterance
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)
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self.is_streaming = True
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log("✅ Google STT streaming session started")
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except Exception as e:
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log(f"❌ Failed to start Google STT streaming: {e}")
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raise
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async def stream_audio(self, audio_chunk: bytes) -> AsyncIterator[TranscriptionResult]:
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"""Stream audio chunk and get transcription results"""
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if not self.is_streaming:
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log("⚠️ STT streaming not started")
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return
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try:
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# Add audio chunk to queue
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await self.audio_queue.put(audio_chunk)
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# Process audio stream
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async def audio_generator():
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while self.is_streaming:
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chunk = await self.audio_queue.get()
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yield speech.StreamingRecognizeRequest(audio_content=chunk)
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# Get responses
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responses = await self.client.streaming_recognize(
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self.streaming_config,
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audio_generator()
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)
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async for response in responses:
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for result in response.results:
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if result.alternatives:
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yield TranscriptionResult(
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text=result.alternatives[0].transcript,
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is_final=result.is_final,
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confidence=result.alternatives[0].confidence,
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timestamp=asyncio.get_event_loop().time()
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)
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except exceptions.OutOfRange:
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log("⚠️ Google STT: Exceeded maximum audio duration")
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self.is_streaming = False
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except Exception as e:
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log(f"❌ Google STT streaming error: {e}")
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raise
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async def stop_streaming(self) -> Optional[TranscriptionResult]:
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"""Stop streaming and get final result"""
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self.is_streaming = False
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log("🛑 Google STT streaming stopped")
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# Process any remaining audio in queue
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if not self.audio_queue.empty():
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# TODO: Process remaining audio
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pass
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return None
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def supports_realtime(self) -> bool:
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"""Google Cloud Speech supports real-time streaming"""
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return True
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def get_supported_languages(self) -> List[str]:
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"""Get list of supported language codes"""
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return [
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"tr-TR", # Turkish
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"en-US", # English (US)
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"en-GB", # English (UK)
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"de-DE", # German
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"fr-FR", # French
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"es-ES", # Spanish
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"it-IT", # Italian
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"pt-BR", # Portuguese (Brazil)
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"ru-RU", # Russian
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"ja-JP", # Japanese
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"ko-KR", # Korean
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"zh-CN", # Chinese (Simplified)
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]
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def _get_encoding(self, encoding: str):
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| 132 |
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"""Convert encoding string to Google Cloud Speech encoding"""
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| 133 |
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encoding_map = {
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| 134 |
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"LINEAR16": speech.RecognitionConfig.AudioEncoding.LINEAR16,
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"FLAC": speech.RecognitionConfig.AudioEncoding.FLAC,
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"MULAW": speech.RecognitionConfig.AudioEncoding.MULAW,
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"AMR": speech.RecognitionConfig.AudioEncoding.AMR,
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| 138 |
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"AMR_WB": speech.RecognitionConfig.AudioEncoding.AMR_WB,
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"OGG_OPUS": speech.RecognitionConfig.AudioEncoding.OGG_OPUS,
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"SPEEX_WITH_HEADER_BYTE": speech.RecognitionConfig.AudioEncoding.SPEEX_WITH_HEADER_BYTE,
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| 141 |
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"WEBM_OPUS": speech.RecognitionConfig.AudioEncoding.WEBM_OPUS,
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}
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return encoding_map.get(encoding, speech.RecognitionConfig.AudioEncoding.WEBM_OPUS)
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