first tests
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"""Check StreamingSession: streamed ASR must match offline ASR; two-turn chat from the cache; latency after release."""
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import sys
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import time
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from pathlib import Path
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import librosa
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import numpy as np
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import soundfile as sf
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import torch
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sys.path.insert(0, str(Path(__file__).resolve().parent.parent))
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import server # noqa: E402
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from session import StreamingSession # noqa: E402
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engine = server.Engine()
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tok = engine.tokenizer
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wav, rate = sf.read(Path(__file__).resolve().parent.parent / "smoke_out/tts.wav", dtype="float32")
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CHUNK = 2048 # what the browser worklet sends
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def system_block(text):
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return engine.processor.apply_chat_template([{"role": "system", "content": text}], tokenize=False)
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def stream_in(s, audio, rate):
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for i in range(0, len(audio), CHUNK):
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s.feed_audio(audio[i:i + CHUNK], rate)
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def decode_text(out, suffix_len):
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seq = out.sequences[0, suffix_len:].tolist()
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return tok.decode([t for t in seq if t != engine.model.config.audio_out_token_id], skip_special_tokens=True)
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# offline reference
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_, inputs = engine.build_inputs([{"role": "system", "content": "Perform ASR."}, {"role": "user", "content": [{"type": "audio"}]}],
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[librosa.resample(wav, orig_sr=rate, target_sr=16000)], None)
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with torch.inference_mode():
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ref = engine.model.generate(**inputs, max_new_tokens=64, generation_mode="text", return_dict_in_generate=True)
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print("offline ASR :", tok.decode(ref.sequences[0, inputs["input_ids"].shape[1]:], skip_special_tokens=True))
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s = StreamingSession(engine)
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s.begin_turn(("asr", ""), system_block("Perform ASR."), keep_history=False)
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stream_in(s, wav, rate)
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t = time.perf_counter()
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s.feed_audio(np.zeros(0, np.float32), rate, final=True)
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out = s.generate("<|im_end|>\n<|im_start|>assistant\n", max_new_tokens=64, generation_mode="text")
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print(f"streamed ASR: {decode_text(out, 5)} ({(time.perf_counter() - t) * 1000:.0f} ms after release, {s.audio_tokens} audio tokens)")
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class FirstFrame:
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def __init__(self): self.t = None
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def put(self, v): self.t = self.t or time.perf_counter()
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def end(self): pass
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sysb = system_block("Respond with interleaved text and audio.")
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conv = [{"role": "system", "content": "Respond with interleaved text and audio."}, {"role": "user", "content": [{"type": "audio"}]}]
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w16 = librosa.resample(wav, orig_sr=rate, target_sr=16000)
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for label in ("warmup", "offline"):
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ff = None
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class FF:
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t = None
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def put(self, v): self.t = self.t or time.perf_counter()
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def end(self): pass
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ff = FF()
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t = time.perf_counter()
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_, inputs = engine.build_inputs(conv, [w16], "<|mixed_start|>")
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with torch.inference_mode():
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engine.model.generate(**inputs, max_new_tokens=512, audio_streamer=ff, return_dict_in_generate=True)
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print(f"{label} chat: first_frame={(ff.t - t) * 1000:.0f}ms after release (prep + full prefill)")
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s = StreamingSession(engine)
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for turn in range(3):
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info = s.begin_turn(("chat", ""), sysb, keep_history=True)
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stream_in(s, wav, rate)
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t = time.perf_counter()
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s.feed_audio(np.zeros(0, np.float32), rate, final=True)
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t_flush = time.perf_counter()
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ff = FirstFrame()
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out = s.generate("<|im_end|>\n<|im_start|>assistant\n<|mixed_start|>", max_new_tokens=512, audio_streamer=ff)
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print(f"chat turn {turn}: {info} flush={(t_flush - t) * 1000:.0f}ms first_frame={(ff.t - t) * 1000:.0f}ms "
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f"total={(time.perf_counter() - t):.1f}s ctx={s.length} close={s.needs_close}\n {decode_text(out, 6)!r}")
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# snapshot / restore round trip
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snap = s.snapshot()
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n = s.length
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s.begin_turn(("chat", ""), sysb, keep_history=True)
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s.restore(snap)
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print("restore ok:", s.length == n)
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@@ -0,0 +1,52 @@
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"""Headless check of the server's generation path: TTS -> ASR -> voice chat, with streaming."""
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import asyncio
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import sys
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import threading
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import time
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from pathlib import Path
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import librosa
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import soundfile as sf
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import torch
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sys.path.insert(0, str(Path(__file__).resolve().parent.parent))
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import server # noqa: E402
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OUT = Path(__file__).resolve().parent.parent / "smoke_out"
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OUT.mkdir(exist_ok=True)
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def run(engine, conv, clips, prefill, mode):
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loop = asyncio.new_event_loop()
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queue = asyncio.Queue()
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stats, cancel = {}, threading.Event()
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emit = server.Emitter(loop, queue, time.perf_counter())
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emit.send = lambda item: None
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_, inputs = engine.build_inputs(conv, clips, prefill)
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text = server.TextStream(engine, emit, cancel, stats)
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audio = server.AudioStream(engine, emit, cancel, stats) if mode != "asr" else None
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t = time.perf_counter()
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with torch.inference_mode():
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engine.model.generate(**inputs, max_new_tokens=512, streamer=text, audio_streamer=audio,
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return_dict_in_generate=True, **({"generation_mode": "text"} if mode == "asr" else {}))
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gen = time.perf_counter() - t
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pcm = torch.cat([torch.from_numpy(c) for c in audio.collected]).numpy() if audio and audio.collected else None
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dur = pcm.size / server.OUT_RATE if pcm is not None else 0
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print(f"[{mode}] prompt={inputs['input_ids'].shape[1]} gen={gen:.2f}s audio={dur:.2f}s stats={stats}\n text={text.text!r}")
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return text.text, pcm
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engine = server.Engine()
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print(engine.info())
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_, pcm = run(engine, [{"role": "system", "content": "Perform TTS in English using a female voice."},
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{"role": "user", "content": "What is the tallest mountain in Europe?"}], [], "<|audio_start|>", "tts")
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sf.write(OUT / "tts.wav", pcm, server.OUT_RATE)
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q16 = librosa.resample(pcm, orig_sr=server.OUT_RATE, target_sr=server.IN_RATE)
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run(engine, [{"role": "system", "content": "Perform ASR."}, {"role": "user", "content": [{"type": "audio"}]}], [q16], None, "asr")
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reply, pcm = run(engine, [{"role": "system", "content": "Respond with interleaved text and audio."},
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{"role": "user", "content": [{"type": "audio"}]}], [q16], "<|mixed_start|>", "chat")
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sf.write(OUT / "chat.wav", pcm, server.OUT_RATE)
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# sanity-check the chat reply's speech by transcribing it
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r16 = librosa.resample(pcm, orig_sr=server.OUT_RATE, target_sr=server.IN_RATE)
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run(engine, [{"role": "system", "content": "Perform ASR."}, {"role": "user", "content": [{"type": "audio"}]}], [r16], None, "asr")
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"""Exercise "Voice + tools" over the WebSocket: typed requests, then a spoken one (made with the server's TTS)."""
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import asyncio
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import json
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import numpy as np
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import websockets
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URL = "ws://127.0.0.1:8000/ws"
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PROMPTS = [
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"What's the weather like in Paris right now?",
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"What is 17 times 23, divided by 4?",
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"Please make the screen green.",
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"What time is it in Tokyo?",
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"Tell me a fun fact about octopuses.", # should answer without a tool
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]
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async def turn(ws, send):
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await send()
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pcm, events = [], []
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while True:
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m = await ws.recv()
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if isinstance(m, bytes):
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pcm.append(np.frombuffer(m, dtype=np.float32))
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continue
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m = json.loads(m)
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if m["type"] in ("tool_call", "tool_result", "ui"):
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events.append({k: v for k, v in m.items() if k not in ("t_ms",)})
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if m["type"] in ("done", "error"):
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return m, events, np.concatenate(pcm) if pcm else np.zeros(0, np.float32)
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async def main():
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async with websockets.connect(URL, max_size=None) as ws:
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await ws.recv()
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await ws.send(json.dumps({"type": "settings", "mode": "tools", "keep_history": False}))
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for p in PROMPTS:
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done, events, pcm = await turn(ws, lambda: ws.send(json.dumps({"type": "text", "text": p})))
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print(f"\n> {p}")
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for e in events:
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print(" ", e)
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s = done.get("stats", {})
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print(f" reply: {done.get('text')!r}\n rounds={s.get('rounds')} audio={s.get('audio_out_s')}s "
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f"first_text={s.get('first_text_ms')}ms gen={s.get('gen_ms')}ms")
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# spoken request: synthesize it with TTS, then send it as mic audio
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q = "What's the weather in Stockholm?"
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_, _, speech = await turn(ws, lambda: ws.send(json.dumps({"type": "tts", "text": q})))
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async def speak():
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await ws.send(json.dumps({"type": "start", "rate": 24000}))
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for i in range(0, len(speech), 2048):
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await ws.send(speech[i:i + 2048].tobytes())
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await ws.send(json.dumps({"type": "stop"}))
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done, events, _ = await turn(ws, speak)
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print(f"\n> (spoken) {q}")
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for e in events:
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print(" ", e)
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print(f" reply: {done.get('text')!r} rounds={done['stats'].get('rounds')}")
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asyncio.run(main())
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"""Drive the server's WebSocket like the browser does: stream a wav in real time in 2048-sample chunks, 3 turns."""
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import asyncio
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import json
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import sys
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import numpy as np
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import soundfile as sf
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import websockets
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async def main(path, url="ws://127.0.0.1:8000/ws"):
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audio, rate = sf.read(path, dtype="float32")
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async with websockets.connect(url, max_size=None) as ws:
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print("<", json.loads(await ws.recv())["type"])
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for turn in range(3): # later turns exercise the cached history
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await ws.send(json.dumps({"type": "start", "rate": rate}))
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for i in range(0, len(audio), 2048):
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await ws.send(audio[i:i + 2048].tobytes())
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await asyncio.sleep(2048 / rate) # real-time pacing, like a mic
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await ws.send(json.dumps({"type": "stop"}))
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pcm_bytes = 0
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while True:
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msg = await ws.recv()
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if isinstance(msg, bytes):
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pcm_bytes += len(msg)
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continue
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m = json.loads(msg)
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if m["type"] in ("token", "audio_chunk", "log"):
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continue
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if m["type"] == "done":
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keys = ("first_text_ms", "first_frame_ms", "first_audio_ms", "flush_ms", "prefill_calls",
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"prefill_busy_ms", "context_tokens", "context_tokens_after", "rtf", "gen_ms", "codec_decode_ms", "audio_out_s")
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print("< done", m["text"][:80], {k: m["stats"].get(k) for k in keys})
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elif m["type"] != "prompt":
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print("<", m["type"], {k: v for k, v in m.items() if k != "type"})
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if m["type"] in ("done", "error"):
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break
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print(f" received {pcm_bytes / 4 / 24000:.2f}s of PCM")
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asyncio.run(main(sys.argv[1]))
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