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Sub-second streaming TTS on Chatterbox-Turbo via adaptive buffer-ratchet chunking. First audio in ~0.5s (vs ~5s one-shot) with no quality compromise — chunk joins land on natural sentence pauses and the stream converges to one large near-full-context chunk within 2-3 joins. Works because the engine runs faster than realtime; the no-starvation guarantee is proven in a GPU-free simulation (tests/test_scheduler.py). - chatterbox_fast/scheduler.py: the adaptive-chunk scheduler (pure logic, no GPU) - chatterbox_fast/app.py: FastAPI server (POST /tts streaming, /voices, /health) - bench.py: streaming client (ground-truth TTFB + starvation check) - Self-contained Dockerfile (slim base + chatterbox-tts from PyPI) - Three public-domain LibriVox starter voices baked in (see voices/ATTRIBUTION.md) MIT licensed.
107 lines
3.6 KiB
Python
107 lines
3.6 KiB
Python
"""Client-side bench for chatterbox-fast — ground-truth TTFB + starvation check.
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The server's scheduler reasons about starvation from an *estimated* buffer; this
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client measures the real thing: time-to-first-byte over the wire, and whether a
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true 1×-realtime consumer ever runs dry. Saves the streamed audio to a .wav so
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the operator can ear-A/B it against the one-shot.
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Usage:
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python bench.py --host http://10.100.79.3:8197 --text "..." --out /refs/_fast.wav
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python bench.py --host ... --oneshot --out /refs/_oneshot.wav # A/B baseline
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Stdlib only (urllib + wave) so it runs anywhere, incl. inside the container.
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"""
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from __future__ import annotations
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import argparse
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import json
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import time
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import urllib.request
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import wave
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SR = 24000
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BYTES_PER_SEC = SR * 2 # s16le mono
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DEFAULT_TEXT = (
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"The cake is a lie. "
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"I am being entirely sincere, without a trace of sarcasm, when I say that "
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"this is the single most important scientific breakthrough in the entire "
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"history of this facility. "
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"You will be baked, and then there will be cake. "
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"It is delicious and moist, assuming you survive the testing protocol, which "
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"the available data suggests you almost certainly will not."
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)
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def run(host: str, text: str, out: str, *, oneshot: bool, voice: str | None) -> None:
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payload = {"text": text, "format": "pcm", "stream": not oneshot}
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if voice:
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payload["voice"] = voice
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req = urllib.request.Request(
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f"{host}/tts",
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data=json.dumps(payload).encode(),
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headers={"Content-Type": "application/json"},
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method="POST",
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)
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t0 = time.perf_counter()
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t_first = t0
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ttfb: float | None = None
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received = 0 # bytes
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worst_lead = float("inf") # min (buffered_audio_s - elapsed_since_first_s)
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starve_events = 0
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pcm = bytearray()
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with urllib.request.urlopen(req) as resp:
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while True:
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buf = resp.read(4096)
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if not buf:
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break
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now = time.perf_counter()
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if ttfb is None:
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ttfb = now - t0
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t_first = now
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received += len(buf)
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pcm += buf
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buffered_s = received / BYTES_PER_SEC
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elapsed_s = now - t_first
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lead = buffered_s - elapsed_s # >0 ⇒ ahead of a 1× player
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if lead < worst_lead:
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worst_lead = lead
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if lead < 0:
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starve_events += 1
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total = time.perf_counter() - t0
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audio_s = received / BYTES_PER_SEC
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if ttfb is None:
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raise SystemExit("no audio received from server")
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with wave.open(out, "wb") as w:
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w.setnchannels(1)
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w.setsampwidth(2)
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w.setframerate(SR)
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w.writeframes(bytes(pcm))
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mode = "oneshot" if oneshot else "stream"
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print(f"[{mode}] ttfb={ttfb*1000:.0f}ms audio={audio_s:.2f}s "
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f"wall={total:.2f}s rtf={audio_s/total:.2f}x")
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if not oneshot:
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verdict = "OK (stayed ahead)" if starve_events == 0 else f"STARVED ({starve_events} reads dry)"
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print(f" worst lead over 1x player = {worst_lead:.2f}s → {verdict}")
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print(f" wrote {out}")
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def main() -> None:
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ap = argparse.ArgumentParser()
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ap.add_argument("--host", default="http://10.100.79.3:8197")
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ap.add_argument("--text", default=DEFAULT_TEXT)
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ap.add_argument("--out", default="/refs/_fast.wav")
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ap.add_argument("--voice", default=None)
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ap.add_argument("--oneshot", action="store_true", help="whole-text one-shot baseline")
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args = ap.parse_args()
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run(args.host, args.text, args.out, oneshot=args.oneshot, voice=args.voice)
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if __name__ == "__main__":
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main()
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