Fix NameError: restore chunk size in turn loop

a4f1328 verified about 1 month ago

19.4 kB

"""FREESTREAM MVP end-to-end latency harness (win condition, docs/LATENCY.md).
In ONE container: downloads livekit-server, runs it in dev mode, pulls the
FREESTREAM repo from the Hub, then:
  1. `freestream join` (the real CLI) spawns the bot into a LiveKit room
  2. a test client connects, asks a question via a Piper-generated mic track
  3. measures speech-end -> first bot audio frame (t_perceived), verifies the
     answer by transcribing the bot's audio, and checks the mouth moved
  4. `freestream leave` (the real CLI) closes the session
  5. verifies the hash-chained replay logs
Win condition: agent joins, answers within 4s with a moving mouth, leaves,
and generates a valid replay log.
Stdlib + livekit + faster-whisper + piper-tts + pillow + numpy.
"""

from __future__ import annotations

import asyncio
import json
import os
import shutil
import socket
import subprocess
import sys
import tarfile
import time
import urllib.request
import wave
from pathlib import Path

import numpy as np

ROOT = Path("/tmp/freestream")
REPO = "https://huggingface.co/datasets/LoveLogicAI/freestream/resolve/main"
VOICE_URL = ("https://huggingface.co/rhasspy/piper-voices/resolve/main/"
             "en/en_US/amy/medium/en_US-amy-medium{ext}")
GH_LATEST = "https://api.github.com/repos/livekit/livekit-server/releases/latest"

BOT_NAME = "Streamy"
QUESTION = "What is your name and what can you do?"
ANSWER_KEYWORDS = ("streamy", "assistant")
QUESTION2 = "What can you do?"
ANSWER_KEYWORDS2 = ("avatar", "listen", "voice")

os.environ["HF_HOME"] = "/tmp/hf"
os.environ["FREESTREAM_VOICE_DIR"] = str(ROOT / "voices")
os.environ["FREESTREAM_WHISPER_MODEL"] = "tiny.en"  # measured best: RTF 0.25 @1 thread
os.environ["FREESTREAM_WHISPER_THREADS"] = "1"
os.environ["PYTHONPATH"] = str(ROOT)
sys.path.insert(0, str(ROOT))  # harness imports the package below (log verification)
os.environ.setdefault("FREESTREAM_LIVEKIT_URL", "ws://localhost:7880")
os.environ.setdefault("FREESTREAM_LIVEKIT_KEY", "devkey")
os.environ.setdefault("FREESTREAM_LIVEKIT_SECRET", "secret")

results = []


def check(name, ok, detail=""):
    results.append((name, ok))
    print(f"{'PASS' if ok else 'FAIL'}  {name}" + (f"  — {detail}" if detail else ""), flush=True)


def fetch_hub(repo_id: str, filename: str, dest: Path, repo_type: str = "model"):
    """Token-authenticated Hub download (robust vs flaky resolve/redirect reads)."""
    dest.parent.mkdir(parents=True, exist_ok=True)
    from huggingface_hub import hf_hub_download
    p = hf_hub_download(repo_id=repo_id, filename=filename, repo_type=repo_type,
                        token=os.environ.get("HF_TOKEN"))
    shutil.copyfile(p, dest)


def fetch(url: str, dest: Path | None = None) -> bytes:
    with urllib.request.urlopen(url, timeout=120) as r:
        data = r.read()
    if dest is not None:
        dest.parent.mkdir(parents=True, exist_ok=True)
        dest.write_bytes(data)
    return data


def run_cli(*args, timeout=240):
    return subprocess.run([sys.executable, str(ROOT / "scripts/freestream.py"), *args],
                          cwd=str(ROOT), capture_output=True, text=True, timeout=timeout)


# --------------------------------------------------------------------------- setup

FILES = ["scripts/freestream.py", "requirements.txt",
         "freestream/__init__.py", "freestream/events.py", "freestream/presence.py",
         "freestream/persona.py", "freestream/mcp_server.py", "freestream/bot.py",
         "freestream/asr.py", "freestream/avatar.py", "freestream/agent.py"]
missing = []
for f in FILES:
    try:
        fetch_hub("LoveLogicAI/freestream", f, ROOT / f, repo_type="dataset")
    except Exception as e:
        missing.append((f, str(e)))
check("fetch code from Hub", not missing, str(missing[:2]))
if missing:
    sys.exit(1)

# livekit-server binary
lk_bin = ROOT / "livekit-server"
if not lk_bin.exists():
    try:
        req = urllib.request.Request(GH_LATEST, headers={"Accept": "application/vnd.github+json"})
        with urllib.request.urlopen(req, timeout=60) as r:
            release = json.loads(r.read())
        asset = next(a for a in release["assets"]
                     if "linux_amd64" in a["name"] and a["name"].endswith(".tar.gz"))
        tgz = ROOT / asset["name"]
        fetch(asset["browser_download_url"], tgz)
        with tarfile.open(tgz) as tf:
            tf.extract("livekit-server", str(ROOT))
        check("download livekit-server", True, release["tag_name"])
    except Exception as e:
        check("download livekit-server", False, str(e)[:200])
        sys.exit(1)

# dev-mode config: explicit keys + loopback-friendly RTC (avoids the
# known UDP bind issue in containers, per livekit issue tracker)
config = ROOT / "lk.yaml"
config.write_text("""
port: 7880
rtc:
  use_external_ip: false
  use_ice_lite: true
  tcp_port: 7881
  port_range_start: 50000
  port_range_end: 50100
keys:
  devkey: secret
""")
server_log = open(ROOT / "server.log", "w")
server = subprocess.Popen([str(lk_bin), "--config", str(config)],
                          stdout=server_log, stderr=subprocess.STDOUT)
t0 = time.perf_counter()
up = False
while time.perf_counter() - t0 < 30:
    try:
        with socket.create_connection(("127.0.0.1", 7880), timeout=1):
            up = True
        break
    except OSError:
        time.sleep(0.5)
check("livekit-server up on :7880", up and server.poll() is None)
if not up:
    print(open(ROOT / "server.log").read()[-1500:])
    sys.exit(1)

# piper voice
vdir = ROOT / "voices"
try:
    for name in ("en/en_US/amy/medium/en_US-amy-medium.onnx",
                 "en/en_US/amy/medium/en_US-amy-medium.onnx.json"):
        dest = vdir / Path(name).name
        if not dest.exists():
            fetch_hub("rhasspy/piper-voices", name, dest)
    check("download piper voice", (vdir / "en_US-amy-medium.onnx").stat().st_size > 50_000_000)
except Exception as e:
    check("download piper voice", False, str(e)[:200])
    sys.exit(1)

# avatar via the real CLI
r = run_cli("generate-avatar", "--output", "identity/videomeeting-avatar.png")
avatar_ok = r.returncode == 0 and (ROOT / "identity/videomeeting-avatar.png").stat().st_size > 1024
check("generate-avatar (procedural, exit 0)", avatar_ok, r.stderr[:150] if not avatar_ok else "")

# warm the whisper cache so the bot and the verifier share it
# (tiny @ 2 threads is the measured optimum under the 8-core cgroup quota:
#  8 threads collapsed to 39.5s vs 1.27s at 1 thread — see ml-intern-freestream-debug-perf)
from faster_whisper import WhisperModel  # noqa: E402

t0 = time.perf_counter()
WhisperModel("tiny", device="cpu", compute_type="int8", cpu_threads=2)
print(f"[setup] whisper tiny int8 cached in {time.perf_counter() - t0:.1f}s", flush=True)

from livekit import api, rtc  # noqa: E402
from piper import PiperVoice  # noqa: E402


# --------------------------------------------------------------------------- join via CLI


def attempt_join(timeout_sec):
    proc = subprocess.Popen(
        [sys.executable, str(ROOT / "scripts/freestream.py"), "join",
         "--meet-url", "https://livekit.local/qa-room", "--platform", "livekit",
         "--bot-name", BOT_NAME, "--image", "identity/videomeeting-avatar.png",
         "--timeout-sec", str(timeout_sec)],
        cwd=str(ROOT), stdout=subprocess.PIPE, stderr=subprocess.PIPE, text=True)
    t0 = time.perf_counter()
    sid, ready = None, False
    for line in proc.stdout:
        try:
            d = json.loads(line)
        except ValueError:
            continue
        if "session_id" in d and sid is None:
            sid = d["session_id"]
        if d.get("status") == "ready":
            ready = True
            break
    return proc, sid, ready, time.perf_counter() - t0


def dump_session_files(sid):
    if not sid:
        return
    sd = ROOT / ".freestream/sessions" / sid
    for name in ("bot.err", "bot.out", "state.json", "driver.json"):
        f = sd / name
        if f.exists():
            print(f"--- session {name} ---\n{f.read_text()[-2500:]}", flush=True)


join_proc, session_id, ready, t_ready = attempt_join(90)
if not ready:
    print(f"[harness] join attempt 1 timed out (t={t_ready:.1f}s); dumping bot output", flush=True)
    dump_session_files(session_id)
    print("[harness] retrying join once...", flush=True)
    join_proc, session_id, ready, t_ready = attempt_join(90)

check(f"CLI join -> ready (session {session_id})", ready and join_proc.poll() is None,
      f"t_ready={t_ready:.1f}s")
if not ready:
    dump_session_files(session_id)
    print(join_proc.stderr.read()[:800])
    sys.exit(1)

# --------------------------------------------------------------------------- test client

token = (api.AccessToken("devkey", "secret")
         .with_identity("tester").with_name("Tester")
         .with_grants(api.VideoGrants(room_join=True, room="qa-room",
                                      can_publish=True, can_subscribe=True)).to_jwt())


async def test_client():
    room = rtc.Room()
    bot_audio = {"chunks": [], "first_arrival": None, "last_arrival": None}
    video_diffs = {"times": [], "diffs": []}

    @room.on("track_subscribed")
    def on_track(track, publication, participant):
        if track.kind == rtc.TrackKind.KIND_AUDIO:
            asyncio.ensure_future(record_audio(track))
        else:
            asyncio.ensure_future(sample_video(track))

    async def record_audio(track):
        stream = rtc.AudioStream(track, sample_rate=16000, num_channels=1)
        async for ev in stream:
            pcm = np.frombuffer(ev.frame.data, dtype=np.int16)
            # the bot's audio track idles with zero-frames between turns;
            # gate arrivals on real speech energy so t_perceived is honest
            rms = float(np.sqrt(np.mean(pcm.astype(np.float32) ** 2))) / 32768.0
            if rms < 0.004:
                continue
            now = time.perf_counter()
            if bot_audio["first_arrival"] is None:
                bot_audio["first_arrival"] = now
                bot_audio["first_arrival_abs"] = time.time()
                print(f"[client] first bot AUDIO (speech) at {now:.3f} "
                      f"(abs {time.time():.3f})", flush=True)
            bot_audio["last_arrival"] = now
            bot_audio["chunks"].append(pcm)

    async def sample_video(track):
        last_img = None
        stream = rtc.VideoStream(track)
        async for ev in stream:
            f = ev.frame
            w, h = f.width, f.height
            data = bytes(f.data)
            if len(data) >= w * h * 1.5 and len(data) < w * h * 4:  # I420 → luminance plane
                img = np.frombuffer(data[: w * h], dtype=np.uint8)
            elif len(data) == w * h * 4:  # RGBA → luma approx
                rgba = np.frombuffer(data, dtype=np.uint8).reshape(h, w, 4)
                img = rgba[:, :, :3].mean(axis=2).astype(np.uint8).ravel()
            else:
                continue
            now = time.perf_counter()
            if last_img is not None and len(img) == len(last_img):
                video_diffs["times"].append(now)
                video_diffs["diffs"].append(float(np.abs(img.astype(np.int16)
                                                          - last_img.astype(np.int16)).mean()))
            last_img = img

    await room.connect("ws://localhost:7880", token,
                       options=rtc.RoomOptions(auto_subscribe=True))
    await asyncio.sleep(2.0)  # ensure bot tracks subscribed

    # synthesize the question
    voice = PiperVoice.load(str(ROOT / "voices/en_US-amy-medium.onnx"))
    chunks = list(voice.synthesize(QUESTION))
    q_pcm = np.concatenate([np.frombuffer(c.audio_int16_bytes, dtype=np.int16)
                            for c in chunks])
    q_sr = int(json.loads((ROOT / "voices/en_US-amy-medium.onnx.json").read_text())
               ["audio"]["sample_rate"])
    wav = wave.open("/tmp/question.wav", "wb")
    wav.setnchannels(1); wav.setsampwidth(2); wav.setframerate(q_sr)
    wav.writeframes(q_pcm.tobytes()); wav.close()

    source = rtc.AudioSource(q_sr, 1)
    track = rtc.LocalAudioTrack.create_audio_track("tester-mic", source)
    opts = rtc.TrackPublishOptions(source=rtc.TrackSource.SOURCE_MICROPHONE)
    await room.local_participant.publish_track(track, opts)

    verifier = WhisperModel("tiny.en", device="cpu", compute_type="int8", cpu_threads=1)
    chunk = int(q_sr * 0.01)  # 10ms push granularity for the mic track
    turns = []
    for turn_i, (q_text, keywords) in enumerate(
            [(QUESTION, ANSWER_KEYWORDS), (QUESTION2, ANSWER_KEYWORDS2)], start=1):
        q_chunks = list(voice.synthesize(q_text))
        q_pcm = np.concatenate([np.frombuffer(c.audio_int16_bytes, np.int16)
                                for c in q_chunks])
        bot_audio["chunks"] = []
        bot_audio["first_arrival"] = None
        bot_audio["last_arrival"] = None
        t0 = None
        for i in range(0, len(q_pcm), chunk):
            piece = q_pcm[i : i + chunk]
            frame = rtc.AudioFrame.create(q_sr, 1, len(piece))
            np.copyto(np.frombuffer(frame.data, dtype=np.int16), piece)
            await source.capture_frame(frame)
            if t0 is None:
                t0 = time.perf_counter()
        t_speech_end = time.perf_counter()
        print(f"[client] turn {turn_i}: question pushed, speech end at {t_speech_end:.3f} "
              f"(abs {time.time():.3f})", flush=True)

        # wait for the bot's answer: first audio + until silence (bounded — a
        # silent bot must fail the harness, not hang it)
        while bot_audio["first_arrival"] is None:
            await asyncio.sleep(0.05)
            if time.perf_counter() - t_speech_end > 45:
                break
        if bot_audio["first_arrival"] is None:
            print(f"[client] turn {turn_i}: NO bot audio within 45s of speech end", flush=True)
            await room.disconnect()
            return {"t_perceived": 999.0, "turns": turns, "t_ready": t_ready,
                    "answer_text": "", "no_answer": True}
        t_perceived = bot_audio["first_arrival"] - t_speech_end
        while time.perf_counter() - bot_audio["last_arrival"] < 4.0:
            await asyncio.sleep(0.1)
            if time.perf_counter() - t_speech_end > 60:
                break
        answer_s = (bot_audio["last_arrival"] - bot_audio["first_arrival"])

        bot_pcm = np.concatenate(bot_audio["chunks"])
        segments, _ = verifier.transcribe(bot_pcm.astype(np.float32) / 32768.0, language="en")
        answer_text = " ".join(s.text.strip() for s in segments).strip()
        kw_ok = any(k in answer_text.lower() for k in keywords)
        check(f"turn {turn_i}: bot answered (transcript contains keyword)",
              kw_ok, answer_text[:160])

        # mouth movement during the answer
        last_arr = bot_audio["last_arrival"] or t_speech_end
        window = [d for t, d in zip(video_diffs["times"], video_diffs["diffs"])
                  if t_speech_end <= t <= last_arr + 0.5]
        moving = len(window) > 5 and sorted(window)[len(window) // 2] > 0.3
        check(f"turn {turn_i}: avatar mouth moved", moving,
              f"{len(window)} frames, median diff {sorted(window)[len(window)//2] if window else 0:.2f}")
        turns.append({"t_perceived": t_perceived, "answer_s": answer_s,
                      "answer_text": answer_text,
                      "first_arrival_abs": bot_audio["first_arrival_abs"]})
        await asyncio.sleep(0.5)

    await room.disconnect()
    return {"t_perceived": min(t["t_perceived"] for t in turns),
            "turns": turns, "t_ready": t_ready,
            "answer_text": turns[-1]["answer_text"]}


timings = asyncio.run(test_client())

# bot forensics — always, success or failure
sdir0 = ROOT / ".freestream/sessions" / session_id
for name in ("bot.err", "bot.out"):
    f = sdir0 / name
    if f.exists() and f.stat().st_size > 0:
        print(f"\n--- {name} (tail) ---\n{f.read_text()[-1500:]}", flush=True)

# --------------------------------------------------------------------------- leave via CLI

r = run_cli("leave", "--session-id", session_id)
out = {}
try:
    out = json.loads(r.stdout.strip().splitlines()[-1])
except (ValueError, IndexError):
    pass
check("CLI leave exits 0 with closed:true", r.returncode == 0 and out.get("closed") is True,
      f"rc={r.returncode} out={str(out)[:100]}")

# replay logs valid
from freestream.events import EventLog  # noqa: E402

sdir = ROOT / ".freestream/sessions" / session_id
cli_ok, cli_bad = EventLog(sdir / "events.jsonl").verify()
bot_ok, bot_bad = EventLog(sdir / "bot-events.jsonl").verify()
check("CLI event log verifies", cli_ok, f"bad_seq={cli_bad}")
check("bot event log verifies", bot_ok, f"bad_seq={bot_bad}")

# --------------------------------------------------------------------------- summary

print("\n--- latency (speech-end -> first bot audio at client) ---", flush=True)
print(f"t_ready (CLI join -> bot ready):        {timings['t_ready']:.2f}s")
for i, t in enumerate(timings["turns"], start=1):
    print(f"turn {i} t_perceived:                     {t['t_perceived']:.2f}s")
print(f"t_perceived best (WIN CONDITION <= 4.0s): {timings['t_perceived']:.2f}s"
      f"   {'PASS' if timings['t_perceived'] <= 4.0 else 'FAIL'}")
print(f"answer transcript (last turn):          {timings['answer_text'][:200]}")

# transport-gap forensics: bot's absolute first-push vs client's absolute arrival
try:
    pushes = []
    for line in (sdir0 / "bot.out").read_text().splitlines():
        if "abs_first_push" in line:
            pushes.append(json.loads(line).get("abs_first_push"))
    arrivals = [t["first_arrival_abs"] for t in timings["turns"]]
    for i, (p, a) in enumerate(zip(pushes, arrivals), start=1):
        print(f"turn {i} bot->client transport gap:         {a - p:.2f}s "
              f"(push@{p:.3f} -> arrival@{a:.3f})")
except Exception as e:
    print(f"transport-gap forensics skipped: {e}")
try:
    bot_log = EventLog(ROOT / ".freestream/sessions" / session_id / "bot-events.jsonl").events()
    synth = [e for e in bot_log if e["type"] == "tts.synthesized"]
    turns = [e for e in bot_log if e["type"] == "turn.completed"]
    if synth:
        print(f"bot-side tts synth time:             {synth[-1]['payload']['seconds']}s")
    if turns:
        print(f"bot-side turn total (speech-end -> done): {turns[-1]['payload']['total_s']}s")
    print("\n--- what the bot heard and said (bot-events.jsonl) ---")
    for e in bot_log:
        if e["type"] in ("asr.final", "agent.speak"):
            extra = e["payload"].get("seconds")
            print(f"  {e['type']:12s} ts=+{e['ts'] % 1000:8.2f}s"
                  + (f" (asr {extra}s)" if extra else "")
                  + f"  {e['payload'].get('text', '')[:140]}")
        elif e["type"] in ("turn.completed", "session.error"):
            print(f"  {e['type']:12s} {e['payload']}")
except Exception as e:
    print(f"bot log summary skipped: {e}")

# server shutdown
server.terminate()

failed = [n for n, ok in results if not ok]
win = timings["t_perceived"] <= 4.0
print(f"\n{len(results) - len(failed)}/{len(results)} checks passed"
      + ("" if win else "; WIN CONDITION MISSED"))
if failed or not win:
    print("FAILED: " + ", ".join(failed + ([] if win else ["t_perceived > 4s"])))
    sys.exit(1)
print("E2E OK — WIN CONDITION MET")

	"""FREESTREAM MVP end-to-end latency harness (win condition, docs/LATENCY.md).

	In ONE container: downloads livekit-server, runs it in dev mode, pulls the
	FREESTREAM repo from the Hub, then:

	1. `freestream join` (the real CLI) spawns the bot into a LiveKit room
	2. a test client connects, asks a question via a Piper-generated mic track
	3. measures speech-end -> first bot audio frame (t_perceived), verifies the
	answer by transcribing the bot's audio, and checks the mouth moved
	4. `freestream leave` (the real CLI) closes the session
	5. verifies the hash-chained replay logs

	Win condition: agent joins, answers within 4s with a moving mouth, leaves,
	and generates a valid replay log.

	Stdlib + livekit + faster-whisper + piper-tts + pillow + numpy.
	"""

	from __future__ import annotations

	import asyncio
	import json
	import os
	import shutil
	import socket
	import subprocess
	import sys
	import tarfile
	import time
	import urllib.request
	import wave
	from pathlib import Path

	import numpy as np

	ROOT = Path("/tmp/freestream")
	REPO = "https://huggingface.co/datasets/LoveLogicAI/freestream/resolve/main"
	VOICE_URL = ("https://huggingface.co/rhasspy/piper-voices/resolve/main/"
	"en/en_US/amy/medium/en_US-amy-medium{ext}")
	GH_LATEST = "https://api.github.com/repos/livekit/livekit-server/releases/latest"

	BOT_NAME = "Streamy"
	QUESTION = "What is your name and what can you do?"
	ANSWER_KEYWORDS = ("streamy", "assistant")
	QUESTION2 = "What can you do?"
	ANSWER_KEYWORDS2 = ("avatar", "listen", "voice")

	os.environ["HF_HOME"] = "/tmp/hf"
	os.environ["FREESTREAM_VOICE_DIR"] = str(ROOT / "voices")
	os.environ["FREESTREAM_WHISPER_MODEL"] = "tiny.en" # measured best: RTF 0.25 @1 thread
	os.environ["FREESTREAM_WHISPER_THREADS"] = "1"
	os.environ["PYTHONPATH"] = str(ROOT)
	sys.path.insert(0, str(ROOT)) # harness imports the package below (log verification)
	os.environ.setdefault("FREESTREAM_LIVEKIT_URL", "ws://localhost:7880")
	os.environ.setdefault("FREESTREAM_LIVEKIT_KEY", "devkey")
	os.environ.setdefault("FREESTREAM_LIVEKIT_SECRET", "secret")

	results = []


	def check(name, ok, detail=""):
	results.append((name, ok))
	print(f"{'PASS' if ok else 'FAIL'} {name}" + (f" — {detail}" if detail else ""), flush=True)


	def fetch_hub(repo_id: str, filename: str, dest: Path, repo_type: str = "model"):
	"""Token-authenticated Hub download (robust vs flaky resolve/redirect reads)."""
	dest.parent.mkdir(parents=True, exist_ok=True)
	from huggingface_hub import hf_hub_download
	p = hf_hub_download(repo_id=repo_id, filename=filename, repo_type=repo_type,
	token=os.environ.get("HF_TOKEN"))
	shutil.copyfile(p, dest)


	def fetch(url: str, dest: Path \| None = None) -> bytes:
	with urllib.request.urlopen(url, timeout=120) as r:
	data = r.read()
	if dest is not None:
	dest.parent.mkdir(parents=True, exist_ok=True)
	dest.write_bytes(data)
	return data


	def run_cli(*args, timeout=240):
	return subprocess.run([sys.executable, str(ROOT / "scripts/freestream.py"), *args],
	cwd=str(ROOT), capture_output=True, text=True, timeout=timeout)


	# --------------------------------------------------------------------------- setup

	FILES = ["scripts/freestream.py", "requirements.txt",
	"freestream/__init__.py", "freestream/events.py", "freestream/presence.py",
	"freestream/persona.py", "freestream/mcp_server.py", "freestream/bot.py",
	"freestream/asr.py", "freestream/avatar.py", "freestream/agent.py"]
	missing = []
	for f in FILES:
	try:
	fetch_hub("LoveLogicAI/freestream", f, ROOT / f, repo_type="dataset")
	except Exception as e:
	missing.append((f, str(e)))
	check("fetch code from Hub", not missing, str(missing[:2]))
	if missing:
	sys.exit(1)

	# livekit-server binary
	lk_bin = ROOT / "livekit-server"
	if not lk_bin.exists():
	try:
	req = urllib.request.Request(GH_LATEST, headers={"Accept": "application/vnd.github+json"})
	with urllib.request.urlopen(req, timeout=60) as r:
	release = json.loads(r.read())
	asset = next(a for a in release["assets"]
	if "linux_amd64" in a["name"] and a["name"].endswith(".tar.gz"))
	tgz = ROOT / asset["name"]
	fetch(asset["browser_download_url"], tgz)
	with tarfile.open(tgz) as tf:
	tf.extract("livekit-server", str(ROOT))
	check("download livekit-server", True, release["tag_name"])
	except Exception as e:
	check("download livekit-server", False, str(e)[:200])
	sys.exit(1)

	# dev-mode config: explicit keys + loopback-friendly RTC (avoids the
	# known UDP bind issue in containers, per livekit issue tracker)
	config = ROOT / "lk.yaml"
	config.write_text("""
	port: 7880
	rtc:
	use_external_ip: false
	use_ice_lite: true
	tcp_port: 7881
	port_range_start: 50000
	port_range_end: 50100
	keys:
	devkey: secret
	""")
	server_log = open(ROOT / "server.log", "w")
	server = subprocess.Popen([str(lk_bin), "--config", str(config)],
	stdout=server_log, stderr=subprocess.STDOUT)
	t0 = time.perf_counter()
	up = False
	while time.perf_counter() - t0 < 30:
	try:
	with socket.create_connection(("127.0.0.1", 7880), timeout=1):
	up = True
	break
	except OSError:
	time.sleep(0.5)
	check("livekit-server up on :7880", up and server.poll() is None)
	if not up:
	print(open(ROOT / "server.log").read()[-1500:])
	sys.exit(1)

	# piper voice
	vdir = ROOT / "voices"
	try:
	for name in ("en/en_US/amy/medium/en_US-amy-medium.onnx",
	"en/en_US/amy/medium/en_US-amy-medium.onnx.json"):
	dest = vdir / Path(name).name
	if not dest.exists():
	fetch_hub("rhasspy/piper-voices", name, dest)
	check("download piper voice", (vdir / "en_US-amy-medium.onnx").stat().st_size > 50_000_000)
	except Exception as e:
	check("download piper voice", False, str(e)[:200])
	sys.exit(1)

	# avatar via the real CLI
	r = run_cli("generate-avatar", "--output", "identity/videomeeting-avatar.png")
	avatar_ok = r.returncode == 0 and (ROOT / "identity/videomeeting-avatar.png").stat().st_size > 1024
	check("generate-avatar (procedural, exit 0)", avatar_ok, r.stderr[:150] if not avatar_ok else "")

	# warm the whisper cache so the bot and the verifier share it
	# (tiny @ 2 threads is the measured optimum under the 8-core cgroup quota:
	# 8 threads collapsed to 39.5s vs 1.27s at 1 thread — see ml-intern-freestream-debug-perf)
	from faster_whisper import WhisperModel # noqa: E402

	t0 = time.perf_counter()
	WhisperModel("tiny", device="cpu", compute_type="int8", cpu_threads=2)
	print(f"[setup] whisper tiny int8 cached in {time.perf_counter() - t0:.1f}s", flush=True)

	from livekit import api, rtc # noqa: E402
	from piper import PiperVoice # noqa: E402


	# --------------------------------------------------------------------------- join via CLI


	def attempt_join(timeout_sec):
	proc = subprocess.Popen(
	[sys.executable, str(ROOT / "scripts/freestream.py"), "join",
	"--meet-url", "https://livekit.local/qa-room", "--platform", "livekit",
	"--bot-name", BOT_NAME, "--image", "identity/videomeeting-avatar.png",
	"--timeout-sec", str(timeout_sec)],
	cwd=str(ROOT), stdout=subprocess.PIPE, stderr=subprocess.PIPE, text=True)
	t0 = time.perf_counter()
	sid, ready = None, False
	for line in proc.stdout:
	try:
	d = json.loads(line)
	except ValueError:
	continue
	if "session_id" in d and sid is None:
	sid = d["session_id"]
	if d.get("status") == "ready":
	ready = True
	break
	return proc, sid, ready, time.perf_counter() - t0


	def dump_session_files(sid):
	if not sid:
	return
	sd = ROOT / ".freestream/sessions" / sid
	for name in ("bot.err", "bot.out", "state.json", "driver.json"):
	f = sd / name
	if f.exists():
	print(f"--- session {name} ---\n{f.read_text()[-2500:]}", flush=True)


	join_proc, session_id, ready, t_ready = attempt_join(90)
	if not ready:
	print(f"[harness] join attempt 1 timed out (t={t_ready:.1f}s); dumping bot output", flush=True)
	dump_session_files(session_id)
	print("[harness] retrying join once...", flush=True)
	join_proc, session_id, ready, t_ready = attempt_join(90)

	check(f"CLI join -> ready (session {session_id})", ready and join_proc.poll() is None,
	f"t_ready={t_ready:.1f}s")
	if not ready:
	dump_session_files(session_id)
	print(join_proc.stderr.read()[:800])
	sys.exit(1)

	# --------------------------------------------------------------------------- test client

	token = (api.AccessToken("devkey", "secret")
	.with_identity("tester").with_name("Tester")
	.with_grants(api.VideoGrants(room_join=True, room="qa-room",
	can_publish=True, can_subscribe=True)).to_jwt())


	async def test_client():
	room = rtc.Room()
	bot_audio = {"chunks": [], "first_arrival": None, "last_arrival": None}
	video_diffs = {"times": [], "diffs": []}

	@room.on("track_subscribed")
	def on_track(track, publication, participant):
	if track.kind == rtc.TrackKind.KIND_AUDIO:
	asyncio.ensure_future(record_audio(track))
	else:
	asyncio.ensure_future(sample_video(track))

	async def record_audio(track):
	stream = rtc.AudioStream(track, sample_rate=16000, num_channels=1)
	async for ev in stream:
	pcm = np.frombuffer(ev.frame.data, dtype=np.int16)
	# the bot's audio track idles with zero-frames between turns;
	# gate arrivals on real speech energy so t_perceived is honest
	rms = float(np.sqrt(np.mean(pcm.astype(np.float32) ** 2))) / 32768.0
	if rms < 0.004:
	continue
	now = time.perf_counter()
	if bot_audio["first_arrival"] is None:
	bot_audio["first_arrival"] = now
	bot_audio["first_arrival_abs"] = time.time()
	print(f"[client] first bot AUDIO (speech) at {now:.3f} "
	f"(abs {time.time():.3f})", flush=True)
	bot_audio["last_arrival"] = now
	bot_audio["chunks"].append(pcm)

	async def sample_video(track):
	last_img = None
	stream = rtc.VideoStream(track)
	async for ev in stream:
	f = ev.frame
	w, h = f.width, f.height
	data = bytes(f.data)
	if len(data) >= w * h * 1.5 and len(data) < w * h * 4: # I420 → luminance plane
	img = np.frombuffer(data[: w * h], dtype=np.uint8)
	elif len(data) == w * h * 4: # RGBA → luma approx
	rgba = np.frombuffer(data, dtype=np.uint8).reshape(h, w, 4)
	img = rgba[:, :, :3].mean(axis=2).astype(np.uint8).ravel()
	else:
	continue
	now = time.perf_counter()
	if last_img is not None and len(img) == len(last_img):
	video_diffs["times"].append(now)
	video_diffs["diffs"].append(float(np.abs(img.astype(np.int16)
	- last_img.astype(np.int16)).mean()))
	last_img = img

	await room.connect("ws://localhost:7880", token,
	options=rtc.RoomOptions(auto_subscribe=True))
	await asyncio.sleep(2.0) # ensure bot tracks subscribed

	# synthesize the question
	voice = PiperVoice.load(str(ROOT / "voices/en_US-amy-medium.onnx"))
	chunks = list(voice.synthesize(QUESTION))
	q_pcm = np.concatenate([np.frombuffer(c.audio_int16_bytes, dtype=np.int16)
	for c in chunks])
	q_sr = int(json.loads((ROOT / "voices/en_US-amy-medium.onnx.json").read_text())
	["audio"]["sample_rate"])
	wav = wave.open("/tmp/question.wav", "wb")
	wav.setnchannels(1); wav.setsampwidth(2); wav.setframerate(q_sr)
	wav.writeframes(q_pcm.tobytes()); wav.close()

	source = rtc.AudioSource(q_sr, 1)
	track = rtc.LocalAudioTrack.create_audio_track("tester-mic", source)
	opts = rtc.TrackPublishOptions(source=rtc.TrackSource.SOURCE_MICROPHONE)
	await room.local_participant.publish_track(track, opts)

	verifier = WhisperModel("tiny.en", device="cpu", compute_type="int8", cpu_threads=1)
	chunk = int(q_sr * 0.01) # 10ms push granularity for the mic track
	turns = []
	for turn_i, (q_text, keywords) in enumerate(
	[(QUESTION, ANSWER_KEYWORDS), (QUESTION2, ANSWER_KEYWORDS2)], start=1):
	q_chunks = list(voice.synthesize(q_text))
	q_pcm = np.concatenate([np.frombuffer(c.audio_int16_bytes, np.int16)
	for c in q_chunks])
	bot_audio["chunks"] = []
	bot_audio["first_arrival"] = None
	bot_audio["last_arrival"] = None
	t0 = None
	for i in range(0, len(q_pcm), chunk):
	piece = q_pcm[i : i + chunk]
	frame = rtc.AudioFrame.create(q_sr, 1, len(piece))
	np.copyto(np.frombuffer(frame.data, dtype=np.int16), piece)
	await source.capture_frame(frame)
	if t0 is None:
	t0 = time.perf_counter()
	t_speech_end = time.perf_counter()
	print(f"[client] turn {turn_i}: question pushed, speech end at {t_speech_end:.3f} "
	f"(abs {time.time():.3f})", flush=True)

	# wait for the bot's answer: first audio + until silence (bounded — a
	# silent bot must fail the harness, not hang it)
	while bot_audio["first_arrival"] is None:
	await asyncio.sleep(0.05)
	if time.perf_counter() - t_speech_end > 45:
	break
	if bot_audio["first_arrival"] is None:
	print(f"[client] turn {turn_i}: NO bot audio within 45s of speech end", flush=True)
	await room.disconnect()
	return {"t_perceived": 999.0, "turns": turns, "t_ready": t_ready,
	"answer_text": "", "no_answer": True}
	t_perceived = bot_audio["first_arrival"] - t_speech_end
	while time.perf_counter() - bot_audio["last_arrival"] < 4.0:
	await asyncio.sleep(0.1)
	if time.perf_counter() - t_speech_end > 60:
	break
	answer_s = (bot_audio["last_arrival"] - bot_audio["first_arrival"])

	bot_pcm = np.concatenate(bot_audio["chunks"])
	segments, _ = verifier.transcribe(bot_pcm.astype(np.float32) / 32768.0, language="en")
	answer_text = " ".join(s.text.strip() for s in segments).strip()
	kw_ok = any(k in answer_text.lower() for k in keywords)
	check(f"turn {turn_i}: bot answered (transcript contains keyword)",
	kw_ok, answer_text[:160])

	# mouth movement during the answer
	last_arr = bot_audio["last_arrival"] or t_speech_end
	window = [d for t, d in zip(video_diffs["times"], video_diffs["diffs"])
	if t_speech_end <= t <= last_arr + 0.5]
	moving = len(window) > 5 and sorted(window)[len(window) // 2] > 0.3
	check(f"turn {turn_i}: avatar mouth moved", moving,
	f"{len(window)} frames, median diff {sorted(window)[len(window)//2] if window else 0:.2f}")
	turns.append({"t_perceived": t_perceived, "answer_s": answer_s,
	"answer_text": answer_text,
	"first_arrival_abs": bot_audio["first_arrival_abs"]})
	await asyncio.sleep(0.5)

	await room.disconnect()
	return {"t_perceived": min(t["t_perceived"] for t in turns),
	"turns": turns, "t_ready": t_ready,
	"answer_text": turns[-1]["answer_text"]}


	timings = asyncio.run(test_client())

	# bot forensics — always, success or failure
	sdir0 = ROOT / ".freestream/sessions" / session_id
	for name in ("bot.err", "bot.out"):
	f = sdir0 / name
	if f.exists() and f.stat().st_size > 0:
	print(f"\n--- {name} (tail) ---\n{f.read_text()[-1500:]}", flush=True)

	# --------------------------------------------------------------------------- leave via CLI

	r = run_cli("leave", "--session-id", session_id)
	out = {}
	try:
	out = json.loads(r.stdout.strip().splitlines()[-1])
	except (ValueError, IndexError):
	pass
	check("CLI leave exits 0 with closed:true", r.returncode == 0 and out.get("closed") is True,
	f"rc={r.returncode} out={str(out)[:100]}")

	# replay logs valid
	from freestream.events import EventLog # noqa: E402

	sdir = ROOT / ".freestream/sessions" / session_id
	cli_ok, cli_bad = EventLog(sdir / "events.jsonl").verify()
	bot_ok, bot_bad = EventLog(sdir / "bot-events.jsonl").verify()
	check("CLI event log verifies", cli_ok, f"bad_seq={cli_bad}")
	check("bot event log verifies", bot_ok, f"bad_seq={bot_bad}")

	# --------------------------------------------------------------------------- summary

	print("\n--- latency (speech-end -> first bot audio at client) ---", flush=True)
	print(f"t_ready (CLI join -> bot ready): {timings['t_ready']:.2f}s")
	for i, t in enumerate(timings["turns"], start=1):
	print(f"turn {i} t_perceived: {t['t_perceived']:.2f}s")
	print(f"t_perceived best (WIN CONDITION <= 4.0s): {timings['t_perceived']:.2f}s"
	f" {'PASS' if timings['t_perceived'] <= 4.0 else 'FAIL'}")
	print(f"answer transcript (last turn): {timings['answer_text'][:200]}")

	# transport-gap forensics: bot's absolute first-push vs client's absolute arrival
	try:
	pushes = []
	for line in (sdir0 / "bot.out").read_text().splitlines():
	if "abs_first_push" in line:
	pushes.append(json.loads(line).get("abs_first_push"))
	arrivals = [t["first_arrival_abs"] for t in timings["turns"]]
	for i, (p, a) in enumerate(zip(pushes, arrivals), start=1):
	print(f"turn {i} bot->client transport gap: {a - p:.2f}s "
	f"(push@{p:.3f} -> arrival@{a:.3f})")
	except Exception as e:
	print(f"transport-gap forensics skipped: {e}")
	try:
	bot_log = EventLog(ROOT / ".freestream/sessions" / session_id / "bot-events.jsonl").events()
	synth = [e for e in bot_log if e["type"] == "tts.synthesized"]
	turns = [e for e in bot_log if e["type"] == "turn.completed"]
	if synth:
	print(f"bot-side tts synth time: {synth[-1]['payload']['seconds']}s")
	if turns:
	print(f"bot-side turn total (speech-end -> done): {turns[-1]['payload']['total_s']}s")
	print("\n--- what the bot heard and said (bot-events.jsonl) ---")
	for e in bot_log:
	if e["type"] in ("asr.final", "agent.speak"):
	extra = e["payload"].get("seconds")
	print(f" {e['type']:12s} ts=+{e['ts'] % 1000:8.2f}s"
	+ (f" (asr {extra}s)" if extra else "")
	+ f" {e['payload'].get('text', '')[:140]}")
	elif e["type"] in ("turn.completed", "session.error"):
	print(f" {e['type']:12s} {e['payload']}")
	except Exception as e:
	print(f"bot log summary skipped: {e}")

	# server shutdown
	server.terminate()

	failed = [n for n, ok in results if not ok]
	win = timings["t_perceived"] <= 4.0
	print(f"\n{len(results) - len(failed)}/{len(results)} checks passed"
	+ ("" if win else "; WIN CONDITION MISSED"))
	if failed or not win:
	print("FAILED: " + ", ".join(failed + ([] if win else ["t_perceived > 4s"])))
	sys.exit(1)
	print("E2E OK — WIN CONDITION MET")