- Mute host mic forwarding while TTS is streaming to prevent interleaving both audio sources into the same playback buffer - Replace nearest-neighbor downsampling with box-filter averaging on both server (host mic) and browser (caller mic) for anti-aliased resampling Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
186 lines
6.6 KiB
Python
186 lines
6.6 KiB
Python
"""Browser caller queue and audio stream service"""
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import asyncio
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import time
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import threading
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import numpy as np
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from typing import Optional
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class CallerService:
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"""Manages browser caller queue, channel allocation, and WebSocket streams"""
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FIRST_REAL_CHANNEL = 3
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def __init__(self):
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self._queue: list[dict] = []
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self.active_calls: dict[str, dict] = {}
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self._allocated_channels: set[int] = set()
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self._caller_counter: int = 0
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self._lock = threading.Lock()
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self._websockets: dict[str, any] = {} # caller_id -> WebSocket
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self.streaming_tts: bool = False # True while TTS audio is being streamed
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def add_to_queue(self, caller_id: str, name: str):
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with self._lock:
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self._queue.append({
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"caller_id": caller_id,
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"name": name,
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"queued_at": time.time(),
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})
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print(f"[Caller] {name} added to queue (ID: {caller_id})")
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def remove_from_queue(self, caller_id: str):
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with self._lock:
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self._queue = [c for c in self._queue if c["caller_id"] != caller_id]
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print(f"[Caller] {caller_id} removed from queue")
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def get_queue(self) -> list[dict]:
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now = time.time()
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with self._lock:
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return [
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{
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"caller_id": c["caller_id"],
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"name": c["name"],
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"wait_time": int(now - c["queued_at"]),
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}
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for c in self._queue
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]
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def allocate_channel(self) -> int:
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with self._lock:
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ch = self.FIRST_REAL_CHANNEL
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while ch in self._allocated_channels:
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ch += 1
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self._allocated_channels.add(ch)
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return ch
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def release_channel(self, channel: int):
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with self._lock:
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self._allocated_channels.discard(channel)
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def take_call(self, caller_id: str) -> dict:
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caller = None
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with self._lock:
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for c in self._queue:
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if c["caller_id"] == caller_id:
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caller = c
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break
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if caller:
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self._queue = [c for c in self._queue if c["caller_id"] != caller_id]
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if not caller:
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raise ValueError(f"Caller {caller_id} not in queue")
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channel = self.allocate_channel()
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self._caller_counter += 1
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name = caller["name"]
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call_info = {
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"caller_id": caller_id,
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"name": name,
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"channel": channel,
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"started_at": time.time(),
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}
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self.active_calls[caller_id] = call_info
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print(f"[Caller] {name} taken on air — channel {channel}")
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return call_info
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def hangup(self, caller_id: str):
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call_info = self.active_calls.pop(caller_id, None)
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if call_info:
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self.release_channel(call_info["channel"])
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print(f"[Caller] {call_info['name']} hung up — channel {call_info['channel']} released")
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self._websockets.pop(caller_id, None)
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def reset(self):
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with self._lock:
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for call_info in self.active_calls.values():
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self._allocated_channels.discard(call_info["channel"])
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self._queue.clear()
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self.active_calls.clear()
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self._allocated_channels.clear()
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self._caller_counter = 0
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self._websockets.clear()
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print("[Caller] Service reset")
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def register_websocket(self, caller_id: str, websocket):
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"""Register a WebSocket for a caller"""
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self._websockets[caller_id] = websocket
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def unregister_websocket(self, caller_id: str):
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"""Unregister a WebSocket"""
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self._websockets.pop(caller_id, None)
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async def send_audio_to_caller(self, caller_id: str, pcm_data: bytes, sample_rate: int):
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"""Send small audio chunk to real caller via WebSocket binary frame.
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For short chunks (host mic, ≤960 samples), sends immediately.
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For large chunks (TTS), use stream_audio_to_caller instead.
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"""
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ws = self._websockets.get(caller_id)
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if not ws:
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return
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try:
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if sample_rate != 16000:
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audio = np.frombuffer(pcm_data, dtype=np.int16).astype(np.float32) / 32768.0
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ratio = 16000 / sample_rate
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out_len = int(len(audio) * ratio)
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indices = (np.arange(out_len) / ratio).astype(int)
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indices = np.clip(indices, 0, len(audio) - 1)
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audio = audio[indices]
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pcm_data = (audio * 32767).astype(np.int16).tobytes()
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await ws.send_bytes(pcm_data)
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except Exception as e:
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print(f"[Caller] Failed to send audio: {e}")
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async def stream_audio_to_caller(self, caller_id: str, pcm_data: bytes, sample_rate: int):
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"""Stream large audio (TTS) to caller in real-time chunks to avoid buffer overflow."""
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ws = self._websockets.get(caller_id)
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if not ws:
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return
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self.streaming_tts = True
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try:
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audio = np.frombuffer(pcm_data, dtype=np.int16).astype(np.float32) / 32768.0
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if sample_rate != 16000:
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ratio = 16000 / sample_rate
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out_len = int(len(audio) * ratio)
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indices = (np.arange(out_len) / ratio).astype(int)
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indices = np.clip(indices, 0, len(audio) - 1)
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audio = audio[indices]
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# Send in 60ms chunks at real-time rate
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chunk_samples = 960
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for i in range(0, len(audio), chunk_samples):
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if caller_id not in self._websockets:
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break
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chunk = audio[i:i + chunk_samples]
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pcm_chunk = (chunk * 32767).astype(np.int16).tobytes()
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await ws.send_bytes(pcm_chunk)
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await asyncio.sleep(0.055) # ~60ms, slightly under to stay ahead
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except Exception as e:
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print(f"[Caller] Failed to stream audio: {e}")
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finally:
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self.streaming_tts = False
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async def notify_caller(self, caller_id: str, message: dict):
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"""Send JSON control message to caller"""
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ws = self._websockets.get(caller_id)
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if ws:
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import json
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await ws.send_text(json.dumps(message))
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async def disconnect_caller(self, caller_id: str):
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"""Disconnect a caller's WebSocket"""
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ws = self._websockets.get(caller_id)
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if ws:
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try:
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import json
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await ws.send_text(json.dumps({"status": "disconnected"}))
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await ws.close()
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except Exception:
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pass
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self._websockets.pop(caller_id, None)
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