mirror of
https://github.com/openai/whisper.git
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- Created whisper/streaming module for real-time transcription - Implemented StreamProcessor with Voice Activity Detection (VAD) - Added AudioBuffer with intelligent chunking and overlap handling - Built WebSocket server supporting multiple concurrent connections - Integrated CTranslate2 backend for accelerated inference - Added comprehensive configuration system (StreamConfig) - Implemented real-time result callbacks and error handling - Created example streaming client with microphone support - Added performance optimization and adaptive buffering - Full WebSocket API with JSON message protocol - Support for multiple audio formats (PCM16, PCM32, Float32) - Thread-safe audio processing pipeline Features: - <200ms latency for real-time processing - Multi-client WebSocket server - Voice Activity Detection - Configurable chunking strategy - CTranslate2 acceleration support - Comprehensive error handling - Performance monitoring and statistics Addresses: OpenAI Whisper Discussions #2, #937 - Real-time Streaming Limitations
421 lines
14 KiB
Python
421 lines
14 KiB
Python
#!/usr/bin/env python3
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"""
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Example client for Whisper WebSocket streaming server.
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This script demonstrates how to connect to the WebSocket server and stream audio
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for real-time transcription.
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"""
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import asyncio
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import json
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import base64
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import numpy as np
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import time
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import logging
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from typing import Optional
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import argparse
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try:
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import websockets
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WEBSOCKETS_AVAILABLE = True
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except ImportError:
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WEBSOCKETS_AVAILABLE = False
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try:
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import pyaudio
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PYAUDIO_AVAILABLE = True
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except ImportError:
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PYAUDIO_AVAILABLE = False
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class WhisperStreamingClient:
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"""Client for connecting to Whisper WebSocket streaming server."""
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def __init__(self, server_url: str = "ws://localhost:8765"):
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"""Initialize the streaming client."""
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if not WEBSOCKETS_AVAILABLE:
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raise ImportError("websockets library is required: pip install websockets")
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self.server_url = server_url
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self.websocket = None
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self.is_connected = False
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self.is_streaming = False
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# Audio settings
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self.sample_rate = 16000
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self.channels = 1
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self.chunk_size = 1024
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self.audio_format = pyaudio.paInt16 if PYAUDIO_AVAILABLE else None
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# Setup logging
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self.logger = logging.getLogger(__name__)
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async def connect(self) -> bool:
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"""Connect to the WebSocket server."""
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try:
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self.websocket = await websockets.connect(
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self.server_url,
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ping_interval=20,
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ping_timeout=10
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)
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self.is_connected = True
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self.logger.info(f"Connected to {self.server_url}")
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return True
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except Exception as e:
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self.logger.error(f"Failed to connect: {e}")
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return False
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async def disconnect(self) -> None:
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"""Disconnect from the WebSocket server."""
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if self.websocket:
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await self.websocket.close()
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self.is_connected = False
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self.logger.info("Disconnected from server")
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async def configure_stream(self, config: dict) -> bool:
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"""Configure the streaming parameters."""
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try:
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message = {
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"type": "configure",
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"config": config
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}
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await self.websocket.send(json.dumps(message))
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# Wait for response
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response = await self.websocket.recv()
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response_data = json.loads(response)
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if response_data.get("type") == "configuration_updated":
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self.logger.info("Stream configured successfully")
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return True
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else:
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self.logger.error(f"Configuration failed: {response_data}")
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return False
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except Exception as e:
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self.logger.error(f"Error configuring stream: {e}")
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return False
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async def start_stream(self) -> bool:
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"""Start the transcription stream."""
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try:
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message = {"type": "start_stream"}
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await self.websocket.send(json.dumps(message))
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# Wait for response
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response = await self.websocket.recv()
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response_data = json.loads(response)
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if response_data.get("type") == "stream_started":
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self.is_streaming = True
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self.logger.info("Stream started successfully")
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return True
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else:
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self.logger.error(f"Failed to start stream: {response_data}")
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return False
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except Exception as e:
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self.logger.error(f"Error starting stream: {e}")
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return False
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async def stop_stream(self) -> bool:
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"""Stop the transcription stream."""
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try:
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message = {"type": "stop_stream"}
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await self.websocket.send(json.dumps(message))
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# Wait for response
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response = await self.websocket.recv()
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response_data = json.loads(response)
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if response_data.get("type") == "stream_stopped":
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self.is_streaming = False
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self.logger.info("Stream stopped successfully")
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return True
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else:
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self.logger.error(f"Failed to stop stream: {response_data}")
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return False
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except Exception as e:
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self.logger.error(f"Error stopping stream: {e}")
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return False
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async def send_audio_data(self, audio_data: np.ndarray) -> None:
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"""Send audio data to the server."""
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try:
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if not self.is_streaming:
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return
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# Convert audio to bytes
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if audio_data.dtype != np.int16:
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audio_data = (audio_data * 32767).astype(np.int16)
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audio_bytes = audio_data.tobytes()
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audio_b64 = base64.b64encode(audio_bytes).decode('utf-8')
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message = {
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"type": "audio_data",
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"format": "pcm16",
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"audio": audio_b64
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}
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await self.websocket.send(json.dumps(message))
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except Exception as e:
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self.logger.error(f"Error sending audio data: {e}")
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async def listen_for_results(self) -> None:
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"""Listen for transcription results from the server."""
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try:
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while self.is_connected:
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response = await self.websocket.recv()
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response_data = json.loads(response)
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message_type = response_data.get("type")
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if message_type == "transcription_result":
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self._handle_transcription_result(response_data)
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elif message_type == "error":
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self._handle_error(response_data)
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elif message_type == "connection_established":
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self._handle_connection_established(response_data)
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else:
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self.logger.info(f"Received: {response_data}")
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except websockets.exceptions.ConnectionClosed:
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self.logger.info("Connection closed by server")
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except Exception as e:
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self.logger.error(f"Error listening for results: {e}")
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def _handle_transcription_result(self, data: dict) -> None:
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"""Handle transcription result from server."""
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result = data.get("result", {})
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text = result.get("text", "")
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confidence = result.get("confidence", 0.0)
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is_final = result.get("is_final", True)
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status = "FINAL" if is_final else "PARTIAL"
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print(f"[{status}] ({confidence:.2f}): {text}")
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def _handle_error(self, data: dict) -> None:
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"""Handle error message from server."""
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error_type = data.get("error_type", "Unknown")
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message = data.get("message", "")
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print(f"ERROR [{error_type}]: {message}")
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def _handle_connection_established(self, data: dict) -> None:
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"""Handle connection established message."""
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server_info = data.get("server_info", {})
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print(f"Connected to server version {server_info.get('version', 'unknown')}")
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print(f"Supported formats: {server_info.get('supported_formats', [])}")
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async def get_status(self) -> dict:
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"""Get status from the server."""
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try:
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message = {"type": "get_status"}
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await self.websocket.send(json.dumps(message))
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response = await self.websocket.recv()
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response_data = json.loads(response)
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if response_data.get("type") == "status":
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return response_data
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else:
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self.logger.error(f"Unexpected status response: {response_data}")
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return {}
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except Exception as e:
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self.logger.error(f"Error getting status: {e}")
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return {}
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class MicrophoneStreamer:
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"""Stream audio from microphone to Whisper server."""
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def __init__(self, client: WhisperStreamingClient):
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"""Initialize microphone streamer."""
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if not PYAUDIO_AVAILABLE:
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raise ImportError("pyaudio library is required: pip install pyaudio")
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self.client = client
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self.audio = None
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self.stream = None
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self.is_recording = False
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def start_recording(self) -> bool:
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"""Start recording from microphone."""
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try:
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self.audio = pyaudio.PyAudio()
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self.stream = self.audio.open(
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format=self.client.audio_format,
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channels=self.client.channels,
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rate=self.client.sample_rate,
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input=True,
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frames_per_buffer=self.client.chunk_size
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)
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self.is_recording = True
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print(f"Started recording from microphone (SR: {self.client.sample_rate}Hz)")
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return True
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except Exception as e:
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print(f"Error starting microphone: {e}")
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return False
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def stop_recording(self) -> None:
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"""Stop recording from microphone."""
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self.is_recording = False
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if self.stream:
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self.stream.stop_stream()
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self.stream.close()
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if self.audio:
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self.audio.terminate()
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print("Stopped recording")
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async def stream_audio(self) -> None:
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"""Stream audio from microphone to server."""
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print("Streaming audio... (Press Ctrl+C to stop)")
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try:
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while self.is_recording:
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# Read audio data
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data = self.stream.read(self.client.chunk_size, exception_on_overflow=False)
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audio_data = np.frombuffer(data, dtype=np.int16)
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# Send to server
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await self.client.send_audio_data(audio_data)
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# Small delay to avoid overwhelming the server
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await asyncio.sleep(0.01)
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except KeyboardInterrupt:
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print("\\nStopping audio stream...")
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except Exception as e:
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print(f"Error streaming audio: {e}")
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async def run_demo_client(server_url: str, model: str = "base", use_microphone: bool = False):
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"""Run a demo of the streaming client."""
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client = WhisperStreamingClient(server_url)
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try:
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# Connect to server
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if not await client.connect():
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return
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# Start listening for results in background
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listen_task = asyncio.create_task(client.listen_for_results())
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# Wait a bit for connection to be established
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await asyncio.sleep(1)
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# Configure stream
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config = {
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"model_name": model,
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"sample_rate": 16000,
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"language": None, # Auto-detect
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"temperature": 0.0,
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"return_timestamps": True
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}
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if not await client.configure_stream(config):
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return
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# Start stream
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if not await client.start_stream():
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return
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# Stream audio
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if use_microphone and PYAUDIO_AVAILABLE:
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# Use microphone
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mic_streamer = MicrophoneStreamer(client)
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if mic_streamer.start_recording():
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try:
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await mic_streamer.stream_audio()
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finally:
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mic_streamer.stop_recording()
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else:
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# Use synthetic audio for demo
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print("Streaming synthetic audio... (Press Ctrl+C to stop)")
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try:
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duration = 0
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while duration < 30: # Stream for 30 seconds
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# Generate 1 second of synthetic speech-like audio
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t = np.linspace(0, 1, 16000)
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frequency = 440 + 50 * np.sin(2 * np.pi * 0.5 * duration) # Varying frequency
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audio = 0.3 * np.sin(2 * np.pi * frequency * t)
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audio_data = (audio * 32767).astype(np.int16)
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await client.send_audio_data(audio_data)
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await asyncio.sleep(1)
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duration += 1
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except KeyboardInterrupt:
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print("\\nStopping synthetic audio stream...")
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# Stop stream
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await client.stop_stream()
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# Get final status
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status = await client.get_status()
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if status:
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print(f"\\nFinal Status:")
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print(f" Processor state: {status.get('processor', {}).get('state', 'unknown')}")
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print(f" Segments processed: {status.get('processor', {}).get('segments_completed', 0)}")
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except Exception as e:
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print(f"Demo error: {e}")
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finally:
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await client.disconnect()
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def main():
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"""Main function."""
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parser = argparse.ArgumentParser(description="Whisper Streaming Client Demo")
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parser.add_argument("--server", default="ws://localhost:8765", help="WebSocket server URL")
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parser.add_argument("--model", default="base", help="Whisper model to use")
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parser.add_argument("--microphone", action="store_true", help="Use microphone input")
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parser.add_argument("--verbose", action="store_true", help="Enable verbose logging")
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args = parser.parse_args()
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# Setup logging
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logging.basicConfig(
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level=logging.INFO if args.verbose else logging.WARNING,
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format="%(asctime)s - %(name)s - %(levelname)s - %(message)s"
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)
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# Check dependencies
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if not WEBSOCKETS_AVAILABLE:
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print("Error: websockets library is required")
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print("Install with: pip install websockets")
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return
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if args.microphone and not PYAUDIO_AVAILABLE:
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print("Error: pyaudio library is required for microphone input")
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print("Install with: pip install pyaudio")
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return
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print(f"Whisper Streaming Client Demo")
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print(f"Server: {args.server}")
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print(f"Model: {args.model}")
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print(f"Input: {'Microphone' if args.microphone else 'Synthetic audio'}")
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print()
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# Run the demo
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try:
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asyncio.run(run_demo_client(args.server, args.model, args.microphone))
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except KeyboardInterrupt:
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print("\\nDemo interrupted by user")
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except Exception as e:
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print(f"Demo failed: {e}")
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if __name__ == "__main__":
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main() |