Add backend api and engine
This commit is contained in:
Xin Wang
601
engine/examples/mic_client.py
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601
engine/examples/mic_client.py
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@@ -0,0 +1,601 @@
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#!/usr/bin/env python3
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"""
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Microphone client for testing duplex voice conversation.
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This client captures audio from the microphone, sends it to the server,
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and plays back the AI's voice response through the speakers.
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It also displays the LLM's text responses in the console.
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Usage:
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python examples/mic_client.py --url ws://localhost:8000/ws
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python examples/mic_client.py --url ws://localhost:8000/ws --chat "Hello!"
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python examples/mic_client.py --url ws://localhost:8000/ws --verbose
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Requirements:
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pip install sounddevice soundfile websockets numpy
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"""
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import argparse
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import asyncio
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import json
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import sys
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import time
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import threading
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import queue
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from pathlib import Path
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try:
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import numpy as np
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except ImportError:
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print("Please install numpy: pip install numpy")
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sys.exit(1)
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try:
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import sounddevice as sd
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except ImportError:
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print("Please install sounddevice: pip install sounddevice")
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sys.exit(1)
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try:
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import websockets
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except ImportError:
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print("Please install websockets: pip install websockets")
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sys.exit(1)
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class MicrophoneClient:
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"""
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Full-duplex microphone client for voice conversation.
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Features:
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- Real-time microphone capture
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- Real-time speaker playback
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- WebSocket communication
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- Text chat support
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"""
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def __init__(
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self,
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url: str,
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sample_rate: int = 16000,
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chunk_duration_ms: int = 20,
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input_device: int = None,
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output_device: int = None
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):
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"""
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Initialize microphone client.
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Args:
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url: WebSocket server URL
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sample_rate: Audio sample rate (Hz)
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chunk_duration_ms: Audio chunk duration (ms)
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input_device: Input device ID (None for default)
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output_device: Output device ID (None for default)
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"""
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self.url = url
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self.sample_rate = sample_rate
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self.chunk_duration_ms = chunk_duration_ms
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self.chunk_samples = int(sample_rate * chunk_duration_ms / 1000)
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self.input_device = input_device
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self.output_device = output_device
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# WebSocket connection
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self.ws = None
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self.running = False
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# Audio buffers
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self.audio_input_queue = queue.Queue()
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self.audio_output_buffer = b"" # Continuous buffer for smooth playback
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self.audio_output_lock = threading.Lock()
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# Statistics
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self.bytes_sent = 0
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self.bytes_received = 0
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# State
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self.is_recording = True
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self.is_playing = True
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# TTFB tracking (Time to First Byte)
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self.request_start_time = None
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self.first_audio_received = False
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# Interrupt handling - discard audio until next trackStart
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self._discard_audio = False
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self._audio_sequence = 0 # Track audio sequence to detect stale chunks
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# Verbose mode for streaming LLM responses
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self.verbose = False
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async def connect(self) -> None:
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"""Connect to WebSocket server."""
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print(f"Connecting to {self.url}...")
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self.ws = await websockets.connect(self.url)
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self.running = True
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print("Connected!")
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# Send invite command
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await self.send_command({
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"command": "invite",
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"option": {
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"codec": "pcm",
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"sampleRate": self.sample_rate
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}
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})
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async def send_command(self, cmd: dict) -> None:
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"""Send JSON command to server."""
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if self.ws:
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await self.ws.send(json.dumps(cmd))
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print(f"→ Command: {cmd.get('command', 'unknown')}")
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async def send_chat(self, text: str) -> None:
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"""Send chat message (text input)."""
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# Reset TTFB tracking for new request
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self.request_start_time = time.time()
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self.first_audio_received = False
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await self.send_command({
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"command": "chat",
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"text": text
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})
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print(f"→ Chat: {text}")
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async def send_interrupt(self) -> None:
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"""Send interrupt command."""
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await self.send_command({
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"command": "interrupt"
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})
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async def send_hangup(self, reason: str = "User quit") -> None:
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"""Send hangup command."""
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await self.send_command({
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"command": "hangup",
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"reason": reason
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})
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def _audio_input_callback(self, indata, frames, time, status):
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"""Callback for audio input (microphone)."""
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if status:
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print(f"Input status: {status}")
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if self.is_recording and self.running:
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# Convert to 16-bit PCM
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audio_data = (indata[:, 0] * 32767).astype(np.int16).tobytes()
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self.audio_input_queue.put(audio_data)
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def _add_audio_to_buffer(self, audio_data: bytes):
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"""Add audio data to playback buffer."""
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with self.audio_output_lock:
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self.audio_output_buffer += audio_data
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def _playback_thread_func(self):
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"""Thread function for continuous audio playback."""
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import time
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# Chunk size: 50ms of audio
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chunk_samples = int(self.sample_rate * 0.05)
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chunk_bytes = chunk_samples * 2
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print(f"Audio playback thread started (device: {self.output_device or 'default'})")
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try:
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# Create output stream with callback
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with sd.OutputStream(
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samplerate=self.sample_rate,
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channels=1,
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dtype='int16',
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blocksize=chunk_samples,
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device=self.output_device,
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latency='low'
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) as stream:
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while self.running:
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# Get audio from buffer
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with self.audio_output_lock:
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if len(self.audio_output_buffer) >= chunk_bytes:
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audio_data = self.audio_output_buffer[:chunk_bytes]
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self.audio_output_buffer = self.audio_output_buffer[chunk_bytes:]
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else:
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# Not enough audio - output silence
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audio_data = b'\x00' * chunk_bytes
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# Convert to numpy array and write to stream
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samples = np.frombuffer(audio_data, dtype=np.int16).reshape(-1, 1)
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stream.write(samples)
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except Exception as e:
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print(f"Playback thread error: {e}")
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import traceback
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traceback.print_exc()
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async def _playback_task(self):
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"""Start playback thread and monitor it."""
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# Run playback in a dedicated thread for reliable timing
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playback_thread = threading.Thread(target=self._playback_thread_func, daemon=True)
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playback_thread.start()
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# Wait for client to stop
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while self.running and playback_thread.is_alive():
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await asyncio.sleep(0.1)
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print("Audio playback stopped")
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async def audio_sender(self) -> None:
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"""Send audio from microphone to server."""
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while self.running:
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try:
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# Get audio from queue with timeout
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try:
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audio_data = await asyncio.get_event_loop().run_in_executor(
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None, lambda: self.audio_input_queue.get(timeout=0.1)
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)
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except queue.Empty:
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continue
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# Send to server
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if self.ws and self.is_recording:
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await self.ws.send(audio_data)
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self.bytes_sent += len(audio_data)
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except asyncio.CancelledError:
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break
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except Exception as e:
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print(f"Audio sender error: {e}")
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break
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async def receiver(self) -> None:
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"""Receive messages from server."""
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try:
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while self.running:
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try:
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message = await asyncio.wait_for(self.ws.recv(), timeout=0.1)
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if isinstance(message, bytes):
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# Audio data received
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self.bytes_received += len(message)
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# Check if we should discard this audio (after interrupt)
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if self._discard_audio:
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duration_ms = len(message) / (self.sample_rate * 2) * 1000
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print(f"← Audio: {duration_ms:.0f}ms (DISCARDED - waiting for new track)")
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continue
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if self.is_playing:
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self._add_audio_to_buffer(message)
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# Calculate and display TTFB for first audio packet
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if not self.first_audio_received and self.request_start_time:
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client_ttfb_ms = (time.time() - self.request_start_time) * 1000
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self.first_audio_received = True
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print(f"← [TTFB] Client first audio latency: {client_ttfb_ms:.0f}ms")
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# Show progress (less verbose)
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with self.audio_output_lock:
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buffer_ms = len(self.audio_output_buffer) / (self.sample_rate * 2) * 1000
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duration_ms = len(message) / (self.sample_rate * 2) * 1000
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print(f"← Audio: {duration_ms:.0f}ms (buffer: {buffer_ms:.0f}ms)")
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else:
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# JSON event
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event = json.loads(message)
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await self._handle_event(event)
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except asyncio.TimeoutError:
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continue
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except websockets.ConnectionClosed:
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print("Connection closed")
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self.running = False
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break
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except asyncio.CancelledError:
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pass
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except Exception as e:
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print(f"Receiver error: {e}")
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self.running = False
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async def _handle_event(self, event: dict) -> None:
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"""Handle incoming event."""
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event_type = event.get("event", "unknown")
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if event_type == "answer":
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print("← Session ready!")
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elif event_type == "speaking":
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print("← User speech detected")
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elif event_type == "silence":
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print("← User silence detected")
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elif event_type == "transcript":
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# Display user speech transcription
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text = event.get("text", "")
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is_final = event.get("isFinal", False)
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if is_final:
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# Clear the interim line and print final
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print(" " * 80, end="\r") # Clear previous interim text
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print(f"→ You: {text}")
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else:
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# Interim result - show with indicator (overwrite same line)
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display_text = text[:60] + "..." if len(text) > 60 else text
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print(f" [listening] {display_text}".ljust(80), end="\r")
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elif event_type == "ttfb":
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# Server-side TTFB event
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latency_ms = event.get("latencyMs", 0)
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print(f"← [TTFB] Server reported latency: {latency_ms}ms")
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elif event_type == "llmResponse":
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# LLM text response
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text = event.get("text", "")
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is_final = event.get("isFinal", False)
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if is_final:
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# Print final LLM response
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print(f"← AI: {text}")
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elif self.verbose:
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# Show streaming chunks only in verbose mode
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display_text = text[:60] + "..." if len(text) > 60 else text
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print(f" [streaming] {display_text}")
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elif event_type == "trackStart":
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print("← Bot started speaking")
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# IMPORTANT: Accept audio again after trackStart
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self._discard_audio = False
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self._audio_sequence += 1
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# Reset TTFB tracking for voice responses (when no chat was sent)
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if self.request_start_time is None:
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self.request_start_time = time.time()
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self.first_audio_received = False
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# Clear any old audio in buffer
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with self.audio_output_lock:
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self.audio_output_buffer = b""
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elif event_type == "trackEnd":
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print("← Bot finished speaking")
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# Reset TTFB tracking after response completes
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self.request_start_time = None
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self.first_audio_received = False
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elif event_type == "interrupt":
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print("← Bot interrupted!")
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# IMPORTANT: Discard all audio until next trackStart
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self._discard_audio = True
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# Clear audio buffer immediately
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with self.audio_output_lock:
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buffer_ms = len(self.audio_output_buffer) / (self.sample_rate * 2) * 1000
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self.audio_output_buffer = b""
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print(f" (cleared {buffer_ms:.0f}ms, discarding audio until new track)")
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elif event_type == "error":
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print(f"← Error: {event.get('error')}")
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elif event_type == "hangup":
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print(f"← Hangup: {event.get('reason')}")
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self.running = False
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else:
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print(f"← Event: {event_type}")
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async def interactive_mode(self) -> None:
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"""Run interactive mode for text chat."""
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print("\n" + "=" * 50)
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print("Voice Conversation Client")
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print("=" * 50)
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print("Speak into your microphone to talk to the AI.")
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print("Or type messages to send text.")
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print("")
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print("Commands:")
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print(" /quit - End conversation")
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print(" /mute - Mute microphone")
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print(" /unmute - Unmute microphone")
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print(" /interrupt - Interrupt AI speech")
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print(" /stats - Show statistics")
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print("=" * 50 + "\n")
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while self.running:
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try:
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user_input = await asyncio.get_event_loop().run_in_executor(
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None, input, ""
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)
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if not user_input:
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continue
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# Handle commands
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if user_input.startswith("/"):
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cmd = user_input.lower().strip()
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if cmd == "/quit":
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await self.send_hangup("User quit")
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break
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elif cmd == "/mute":
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self.is_recording = False
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print("Microphone muted")
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elif cmd == "/unmute":
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self.is_recording = True
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print("Microphone unmuted")
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elif cmd == "/interrupt":
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await self.send_interrupt()
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elif cmd == "/stats":
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print(f"Sent: {self.bytes_sent / 1024:.1f} KB")
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print(f"Received: {self.bytes_received / 1024:.1f} KB")
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else:
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print(f"Unknown command: {cmd}")
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else:
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# Send as chat message
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await self.send_chat(user_input)
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except EOFError:
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break
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except Exception as e:
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print(f"Input error: {e}")
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async def run(self, chat_message: str = None, interactive: bool = True) -> None:
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"""
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Run the client.
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Args:
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chat_message: Optional single chat message to send
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interactive: Whether to run in interactive mode
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"""
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try:
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await self.connect()
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# Wait for answer
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await asyncio.sleep(0.5)
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# Start audio input stream
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print("Starting audio streams...")
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input_stream = sd.InputStream(
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samplerate=self.sample_rate,
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channels=1,
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dtype=np.float32,
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blocksize=self.chunk_samples,
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device=self.input_device,
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callback=self._audio_input_callback
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)
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input_stream.start()
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print("Audio streams started")
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# Start background tasks
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sender_task = asyncio.create_task(self.audio_sender())
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receiver_task = asyncio.create_task(self.receiver())
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playback_task = asyncio.create_task(self._playback_task())
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if chat_message:
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# Send single message and wait
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await self.send_chat(chat_message)
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await asyncio.sleep(15)
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elif interactive:
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# Run interactive mode
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await self.interactive_mode()
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else:
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# Just wait
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while self.running:
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await asyncio.sleep(0.1)
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# Cleanup
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self.running = False
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sender_task.cancel()
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receiver_task.cancel()
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playback_task.cancel()
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try:
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await sender_task
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except asyncio.CancelledError:
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pass
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try:
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await receiver_task
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except asyncio.CancelledError:
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pass
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try:
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await playback_task
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except asyncio.CancelledError:
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pass
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input_stream.stop()
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except ConnectionRefusedError:
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print(f"Error: Could not connect to {self.url}")
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print("Make sure the server is running.")
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except Exception as e:
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print(f"Error: {e}")
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finally:
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await self.close()
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async def close(self) -> None:
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"""Close the connection."""
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self.running = False
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if self.ws:
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await self.ws.close()
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print(f"\nSession ended")
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print(f" Total sent: {self.bytes_sent / 1024:.1f} KB")
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print(f" Total received: {self.bytes_received / 1024:.1f} KB")
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def list_devices():
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"""List available audio devices."""
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print("\nAvailable audio devices:")
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||||
print("-" * 60)
|
||||
devices = sd.query_devices()
|
||||
for i, device in enumerate(devices):
|
||||
direction = []
|
||||
if device['max_input_channels'] > 0:
|
||||
direction.append("IN")
|
||||
if device['max_output_channels'] > 0:
|
||||
direction.append("OUT")
|
||||
direction_str = "/".join(direction) if direction else "N/A"
|
||||
|
||||
default = ""
|
||||
if i == sd.default.device[0]:
|
||||
default += " [DEFAULT INPUT]"
|
||||
if i == sd.default.device[1]:
|
||||
default += " [DEFAULT OUTPUT]"
|
||||
|
||||
print(f" {i:2d}: {device['name'][:40]:40s} ({direction_str}){default}")
|
||||
print("-" * 60)
|
||||
|
||||
|
||||
async def main():
|
||||
parser = argparse.ArgumentParser(
|
||||
description="Microphone client for duplex voice conversation"
|
||||
)
|
||||
parser.add_argument(
|
||||
"--url",
|
||||
default="ws://localhost:8000/ws",
|
||||
help="WebSocket server URL"
|
||||
)
|
||||
parser.add_argument(
|
||||
"--chat",
|
||||
help="Send a single chat message instead of using microphone"
|
||||
)
|
||||
parser.add_argument(
|
||||
"--sample-rate",
|
||||
type=int,
|
||||
default=16000,
|
||||
help="Audio sample rate (default: 16000)"
|
||||
)
|
||||
parser.add_argument(
|
||||
"--input-device",
|
||||
type=int,
|
||||
help="Input device ID"
|
||||
)
|
||||
parser.add_argument(
|
||||
"--output-device",
|
||||
type=int,
|
||||
help="Output device ID"
|
||||
)
|
||||
parser.add_argument(
|
||||
"--list-devices",
|
||||
action="store_true",
|
||||
help="List available audio devices and exit"
|
||||
)
|
||||
parser.add_argument(
|
||||
"--no-interactive",
|
||||
action="store_true",
|
||||
help="Disable interactive mode"
|
||||
)
|
||||
parser.add_argument(
|
||||
"--verbose", "-v",
|
||||
action="store_true",
|
||||
help="Show streaming LLM response chunks"
|
||||
)
|
||||
|
||||
args = parser.parse_args()
|
||||
|
||||
if args.list_devices:
|
||||
list_devices()
|
||||
return
|
||||
|
||||
client = MicrophoneClient(
|
||||
url=args.url,
|
||||
sample_rate=args.sample_rate,
|
||||
input_device=args.input_device,
|
||||
output_device=args.output_device
|
||||
)
|
||||
client.verbose = args.verbose
|
||||
|
||||
await client.run(
|
||||
chat_message=args.chat,
|
||||
interactive=not args.no_interactive
|
||||
)
|
||||
|
||||
|
||||
if __name__ == "__main__":
|
||||
try:
|
||||
asyncio.run(main())
|
||||
except KeyboardInterrupt:
|
||||
print("\nInterrupted by user")
|
||||
Reference in New Issue
Block a user