Files
ai-podcast/backend/services/caller_service.py
tcpsyn d4e25ceb88 Stream TTS audio to caller in real-time chunks
TTS audio was sent as a single huge WebSocket frame that overflowed the
browser's 3s ring buffer. Now streams in 60ms chunks at real-time rate.
Also increased browser ring buffer from 3s to 10s as safety net.

Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
2026-02-05 16:56:22 -07:00

182 lines
6.5 KiB
Python

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