0.1.15 | BingBong live voice (client groundwork)

Lets a live microphone be streamed through BingBong from a relay rather
than over PEAK's Photon servers. This is the in-game client only: it
connects over a WebSocket, decodes the audio and plays it. The relay and
the web app that captures the microphone are not built yet, so the
feature does nothing on its own.

Playback uses OnAudioFilterRead rather than a streaming AudioClip. A clip
created with stream:true fills Unity's decode buffer ahead of playback,
and that read-ahead cannot be satisfied from a live stream because the
audio has not arrived yet -- the buffer drained to empty on a cycle no
matter how deep it was made, which was audible as constant clicking. The
filter callback runs in lockstep with playback, so demand can never
outrun real time.

Includes tools/stub_relay.py, a development stub speaking the same wire
format, so the client can be tested end to end without the real backend,
a microphone or a browser.

Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
This commit is contained in:
2026-09-05 16:18:04 +02:00
parent 50a7845345
commit 0d2350553a
8 changed files with 1137 additions and 1 deletions

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"""Development stub relay for BingBong Live Voice.
Speaks the exact wire format the real relay (an endpoint on the AudioBank FastAPI service)
will use, so the mod can be built and tested before that backend exists. Deliberately minimal:
no auth, no session directory, no persistence. It is a test fixture, not a reference
implementation — build the real thing from the plan, not from this.
Requires `websockets` (pip install websockets).
python tools/stub_relay.py serve # relay on ws://localhost:8787/voice/ws
python tools/stub_relay.py tone # stream a 1 kHz sine into channel "test"
python tools/stub_relay.py tone --sweep # sweep 200-2000 Hz (resampling bugs are obvious)
python tools/stub_relay.py listen # dump received frames, to verify fan-out
Point the mod at it with, in BepInEx config:
[BingBong Voice]
RelayURL = ws://localhost:8787/voice/ws?token=listen:test
Leaving RelayURL empty instead derives the URL from BugleSoundAPIURL, which is where the
relay lives in production.
Connect as: ws://localhost:8787/voice/ws?token=listen:test
ws://localhost:8787/voice/ws?token=speak:test
Wire format — 12-byte little-endian header, then PCM16LE mono payload:
0 u8 magic 0xB1
1 u8 version 1
2 u8 codec 0=PCM16LE 1=mu-law
3 u8 rate code 0=16k 1=8k 2=24k 3=48k
4 u32 seq
8 u32 sampleTimestamp (samples since stream start)
Binary frame = audio. Text frame = JSON control.
"""
import argparse
import asyncio
import json
import math
import struct
import sys
from collections import defaultdict
from urllib.parse import urlparse, parse_qs
from websockets.asyncio.server import serve
from websockets.asyncio.client import connect
HOST = "localhost"
PORT = 8787
PATH = "/voice/ws"
MAGIC = 0xB1
VERSION = 1
CODEC_PCM16 = 0
RATE_CODES = {16000: 0, 8000: 1, 24000: 2, 48000: 3}
FRAME_MS = 20
HEADER = struct.Struct("<BBBBII")
def pack_header(seq: int, ts: int, rate: int = 16000, codec: int = CODEC_PCM16) -> bytes:
return HEADER.pack(MAGIC, VERSION, codec, RATE_CODES[rate], seq & 0xFFFFFFFF, ts & 0xFFFFFFFF)
# ---------------------------------------------------------------- relay
channels: dict[str, dict] = defaultdict(lambda: {"speaker": None, "listeners": set()})
def parse_token(raw_path: str):
"""token=<role>:<channel>. Returns (role, channel) or (None, None)."""
q = parse_qs(urlparse(raw_path).query)
token = (q.get("token") or [""])[0]
if ":" not in token:
return None, None
role, _, channel = token.partition(":")
if role not in ("listen", "speak") or not channel:
return None, None
return role, channel
async def handler(ws):
role, channel = parse_token(ws.request.path)
if role is None:
await ws.close(1008, "bad token")
return
ch = channels[channel]
if role == "speak":
old = ch["speaker"]
if old is not None:
try:
await old.send(json.dumps({"type": "preempted"}))
await old.close(1000, "preempted")
except Exception:
pass
ch["speaker"] = ws
print(f"[relay] speaker joined '{channel}' ({len(ch['listeners'])} listening)")
await broadcast_control(channel, {"type": "speaker_joined"})
try:
async for msg in ws:
if isinstance(msg, bytes):
await fanout(channel, msg)
except Exception:
pass
finally:
if ch["speaker"] is ws:
ch["speaker"] = None
print(f"[relay] speaker left '{channel}'")
await broadcast_control(channel, {"type": "speaker_left"})
return
ch["listeners"].add(ws)
print(f"[relay] listener joined '{channel}' -> {len(ch['listeners'])} listening")
try:
await ws.wait_closed()
finally:
ch["listeners"].discard(ws)
print(f"[relay] listener left '{channel}' -> {len(ch['listeners'])} listening")
async def fanout(channel: str, payload: bytes):
listeners = list(channels[channel]["listeners"])
if not listeners:
return
# Real relay uses a bounded per-listener queue and drops oldest; the stub just fires and
# forgets, which is fine locally and keeps this file short.
await asyncio.gather(*(safe_send(w, payload) for w in listeners), return_exceptions=True)
async def broadcast_control(channel: str, obj: dict):
msg = json.dumps(obj)
await asyncio.gather(
*(safe_send(w, msg) for w in list(channels[channel]["listeners"])),
return_exceptions=True,
)
async def safe_send(ws, payload):
try:
await ws.send(payload)
except Exception:
pass
async def cmd_serve(args):
async with serve(handler, HOST, args.port, max_size=None):
print(f"[relay] listening on ws://{HOST}:{args.port}{PATH}")
print(f"[relay] listen: ws://{HOST}:{args.port}{PATH}?token=listen:test")
print(f"[relay] speak : ws://{HOST}:{args.port}{PATH}?token=speak:test")
await asyncio.Future()
# ---------------------------------------------------------------- tone generator
try:
import numpy as _np
except ImportError:
_np = None
def generate_frame(phase, ts, n, rate, args):
"""Returns (new_phase, pcm16_bytes) for one frame, phase-continuous across frames."""
if _np is not None:
idx = _np.arange(n, dtype=_np.float64)
if args.sweep:
# 200-2000 Hz over 4 s: any resampling error shows up as an obvious warble
t = (ts + idx) / rate
freq = 200.0 + 1800.0 * (0.5 - 0.5 * _np.cos(2 * _np.pi * t / 4.0))
else:
freq = _np.full(n, float(args.freq))
ph = phase + _np.cumsum(2 * _np.pi * freq / rate)
samples = (_np.sin(ph) * 0.35 * 32767).astype("<i2")
return float(ph[-1] % (2 * _np.pi)), samples.tobytes()
# Fallback without numpy. Fine for a steady tone; sweep may not hold real time.
out = []
for i in range(n):
freq = args.freq
if args.sweep:
t = (ts + i) / rate
freq = 200.0 + 1800.0 * (0.5 - 0.5 * math.cos(2 * math.pi * t / 4.0))
phase = (phase + 2 * math.pi * freq / rate) % (2 * math.pi)
out.append(int(math.sin(phase) * 0.35 * 32767))
return phase, struct.pack(f"<{n}h", *out)
async def cmd_tone(args):
url = f"ws://{HOST}:{args.port}{PATH}?token=speak:{args.channel}"
rate = args.rate
n = int(rate * FRAME_MS / 1000)
print(f"[tone] {url} {rate} Hz, {n} samples/frame ({FRAME_MS} ms)")
loop = asyncio.get_running_loop()
async with connect(url, max_size=None) as ws:
seq = 0
ts = 0
phase = 0.0
# Deadline-based pacing. A plain `sleep(FRAME_MS)` per iteration sleeps for *at least*
# that long and then adds the generate+send time on top, so the sender silently drifts
# behind real time and starves the receiver's jitter buffer. Track an absolute deadline
# and sleep only for the remainder instead.
next_deadline = loop.time()
late_frames = 0
while True:
# Vectorised: generating 320 samples a frame in a Python loop costs more than the
# 20 ms frame budget once sweep mode adds a second transcendental per sample, and the
# sender then silently falls behind real time and starves the receiver.
phase, payload = generate_frame(phase, ts, n, rate, args)
await ws.send(pack_header(seq, ts, rate) + payload)
seq += 1
ts += n
next_deadline += FRAME_MS / 1000
delay = next_deadline - loop.time()
if delay > 0:
await asyncio.sleep(delay)
else:
# Generation itself overran the frame budget; we can't catch up by sleeping.
late_frames += 1
next_deadline = loop.time()
if seq % 50 == 0:
print(f"\r[tone] sent {seq} frames ({ts / rate:6.1f}s), {late_frames} late", end="", flush=True)
# ---------------------------------------------------------------- listener (verification)
async def cmd_listen(args):
url = f"ws://{HOST}:{args.port}{PATH}?token=listen:{args.channel}"
print(f"[listen] {url}")
async with connect(url, max_size=None) as ws:
count = 0
last_seq = None
gaps = 0
async for msg in ws:
if isinstance(msg, str):
print(f"\n[listen] control: {msg}")
continue
magic, ver, codec, rc, seq, ts = HEADER.unpack_from(msg, 0)
if magic != MAGIC or ver != VERSION:
print(f"\n[listen] BAD HEADER magic={magic:#x} ver={ver}")
continue
if last_seq is not None and seq != last_seq + 1:
gaps += 1
print(f"\n[listen] SEQ GAP {last_seq} -> {seq}")
last_seq = seq
count += 1
if count % 50 == 0:
print(
f"\r[listen] {count} frames, {len(msg) - 12} B payload, codec={codec} "
f"rate={rc} ts={ts} gaps={gaps}",
end="",
flush=True,
)
def main():
p = argparse.ArgumentParser(description=__doc__, formatter_class=argparse.RawDescriptionHelpFormatter)
p.add_argument("--port", type=int, default=PORT)
sub = p.add_subparsers(dest="cmd", required=True)
sub.add_parser("serve")
t = sub.add_parser("tone")
t.add_argument("--channel", default="test")
t.add_argument("--freq", type=float, default=1000.0)
t.add_argument("--rate", type=int, default=16000, choices=sorted(RATE_CODES))
t.add_argument("--sweep", action="store_true")
l = sub.add_parser("listen")
l.add_argument("--channel", default="test")
args = p.parse_args()
fn = {"serve": cmd_serve, "tone": cmd_tone, "listen": cmd_listen}[args.cmd]
try:
asyncio.run(fn(args))
except KeyboardInterrupt:
print("\nbye")
if __name__ == "__main__":
sys.exit(main())