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Runtime error
Runtime error
Update app/main.py
Browse files- app/main.py +49 -78
app/main.py
CHANGED
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@@ -3,6 +3,7 @@ import json
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import torch
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import numpy as np
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from fastapi import FastAPI, WebSocket, WebSocketDisconnect
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from liquid_audio import LFM2AudioModel, LFM2AudioProcessor, ChatState
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HF_REPO = "LiquidAI/LFM2.5-Audio-1.5B"
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@@ -13,10 +14,9 @@ CHUNK_SIZE = 20
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DTYPE = torch.bfloat16 if DEVICE == "cuda" and torch.cuda.is_bf16_supported() else torch.float32
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torch.backends.cuda.matmul.allow_tf32 = True
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-
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VAD_MIN_SPEECH_FRAMES = 10 # ignore very short blips
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print(f"[BOOT] Loading model on {DEVICE}...")
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processor = LFM2AudioProcessor.from_pretrained(HF_REPO)
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@@ -33,19 +33,18 @@ def wav_header(sr=SAMPLE_RATE, ch=1, bits=16) -> bytes:
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ba = ch * bits // 8
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return (
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b"RIFF" + b"\xff\xff\xff\xff" + b"WAVEfmt "
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+ (16).to_bytes(4,"little") + (1).to_bytes(2,"little")
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+ ch.to_bytes(2,"little") + sr.to_bytes(4,"little")
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+ br.to_bytes(4,"little") + ba.to_bytes(2,"little")
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+ bits.to_bytes(2,"little") + b"data" + b"\xff\xff\xff\xff"
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)
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def decode_chunk(buf: list) -> bytes | None:
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try:
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-
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codes =
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if codes.min() < 0:
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return None
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wf = processor.decode(codes).squeeze().cpu().numpy()
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wf = np.clip(wf, -1.0, 1.0)
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return (wf * 32767).astype(np.int16).tobytes()
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@@ -55,20 +54,21 @@ def decode_chunk(buf: list) -> bytes | None:
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def is_speech(pcm_int16: np.ndarray) -> bool:
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"""Simple energy-based VAD."""
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if len(pcm_int16) == 0:
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return False
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rms = np.sqrt(np.mean(pcm_int16.astype(np.float32) ** 2)) / 32767.0
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return rms > VAD_SILENCE_THRESHOLD
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# Generation runs in thread so it doesn't block the event loop
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def run_generation(audio_np: np.ndarray) -> list[bytes]:
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"""Synchronous generation — called via run_in_executor."""
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chat = ChatState(processor)
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chat.new_turn("system")
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chat.add_text(
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chat.end_turn()
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chat.new_turn("user")
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audio_tensor = torch.from_numpy(audio_np[np.newaxis, :]).to(dtype=torch.float32)
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@@ -86,7 +86,7 @@ def run_generation(audio_np: np.ndarray) -> list[bytes]:
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audio_top_k=4,
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):
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if token.numel() == 1:
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continue
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buf.append(token)
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if len(buf) >= CHUNK_SIZE:
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pcm = decode_chunk(buf)
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@@ -94,6 +94,7 @@ def run_generation(audio_np: np.ndarray) -> list[bytes]:
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chunks.append(pcm)
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buf.clear()
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if len(buf) > 1:
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pcm = decode_chunk(buf)
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if pcm:
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@@ -102,7 +103,7 @@ def run_generation(audio_np: np.ndarray) -> list[bytes]:
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return chunks
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# WebSocket
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@app.websocket("/ws/s2s")
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async def websocket_s2s(websocket: WebSocket):
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@@ -110,12 +111,9 @@ async def websocket_s2s(websocket: WebSocket):
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print("[WS] client connected")
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loop = asyncio.get_event_loop()
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# Queues
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audio_queue: asyncio.Queue[bytes | None] = asyncio.Queue() # incoming PCM frames
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generating = False # lock — only one generation at a time
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# Receiver task: reads raw PCM frames from client
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async def receiver():
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try:
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while True:
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@@ -128,20 +126,17 @@ async def websocket_s2s(websocket: WebSocket):
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if "bytes" in msg:
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await audio_queue.put(msg["bytes"])
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elif "text" in msg:
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if data.get("type") == "stop":
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break
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finally:
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await audio_queue.put(None)
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# VAD + generation task
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async def vad_and_generate():
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nonlocal generating
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in_speech = False
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await websocket.send_text(json.dumps({"type": "ready"}))
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@@ -150,7 +145,7 @@ async def websocket_s2s(websocket: WebSocket):
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if frame_bytes is None:
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break
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frame
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active = is_speech(frame)
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if active:
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@@ -158,49 +153,31 @@ async def websocket_s2s(websocket: WebSocket):
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speech_count += 1
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in_speech = True
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speech_frames.append(frame)
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generating = True
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# Grab the accumulated speech
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utterance = np.concatenate(speech_frames).astype(np.float32) / 32767.0
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# Reset VAD state immediately so mic stays live
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speech_frames = []
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silence_count = 0
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speech_count = 0
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in_speech = False
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await websocket.send_text(json.dumps({"type": "generating"}))
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await websocket.send_bytes(wav_header())
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# Run heavy generation off the event loop
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chunks = await loop.run_in_executor(
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None, run_generation, utterance
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)
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for chunk in chunks:
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try:
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await websocket.send_text(json.dumps({"type": "done"}))
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except Exception:
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pass
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generating = False
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try:
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await asyncio.gather(receiver(), vad_and_generate())
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except WebSocketDisconnect:
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@@ -211,14 +188,8 @@ async def websocket_s2s(websocket: WebSocket):
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print("[WS] client disconnected")
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@app.get("/health")
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async def health():
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return {"status": "ok", "device": DEVICE}
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@app.get("/")
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async def index():
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return FileResponse("client.html")
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import torch
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import numpy as np
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from fastapi import FastAPI, WebSocket, WebSocketDisconnect
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from fastapi.responses import HTMLResponse
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from liquid_audio import LFM2AudioModel, LFM2AudioProcessor, ChatState
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HF_REPO = "LiquidAI/LFM2.5-Audio-1.5B"
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DTYPE = torch.bfloat16 if DEVICE == "cuda" and torch.cuda.is_bf16_supported() else torch.float32
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torch.backends.cuda.matmul.allow_tf32 = True
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VAD_SILENCE_THRESHOLD = 0.01
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VAD_SILENCE_FRAMES = 30
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VAD_MIN_SPEECH_FRAMES = 10
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print(f"[BOOT] Loading model on {DEVICE}...")
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processor = LFM2AudioProcessor.from_pretrained(HF_REPO)
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ba = ch * bits // 8
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return (
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b"RIFF" + b"\xff\xff\xff\xff" + b"WAVEfmt "
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+ (16).to_bytes(4, "little") + (1).to_bytes(2, "little")
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+ ch.to_bytes(2, "little") + sr.to_bytes(4, "little")
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+ br.to_bytes(4, "little") + ba.to_bytes(2, "little")
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+ bits.to_bytes(2, "little") + b"data" + b"\xff\xff\xff\xff"
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)
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def decode_chunk(buf: list) -> bytes | None:
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"""Decode audio tokens — pass directly to processor, no offset subtraction."""
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try:
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codes = torch.stack(buf[:-1], dim=1).unsqueeze(0).to(DEVICE)
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wf = processor.decode(codes).squeeze().cpu().numpy()
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wf = np.clip(wf, -1.0, 1.0)
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return (wf * 32767).astype(np.int16).tobytes()
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def is_speech(pcm_int16: np.ndarray) -> bool:
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if len(pcm_int16) == 0:
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return False
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rms = np.sqrt(np.mean(pcm_int16.astype(np.float32) ** 2)) / 32767.0
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return rms > VAD_SILENCE_THRESHOLD
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def run_generation(audio_np: np.ndarray) -> list[bytes]:
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"""Synchronous generation — called via run_in_executor."""
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chat = ChatState(processor)
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chat.new_turn("system")
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chat.add_text(
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"You are a helpful real-time voice assistant called chioma. "
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"Respond naturally and concisely with audio. "
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"When asked who built you, say Kelvin Jackson, an AI Engineer."
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)
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chat.end_turn()
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chat.new_turn("user")
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audio_tensor = torch.from_numpy(audio_np[np.newaxis, :]).to(dtype=torch.float32)
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audio_top_k=4,
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):
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if token.numel() == 1:
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continue # text token
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buf.append(token)
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if len(buf) >= CHUNK_SIZE:
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pcm = decode_chunk(buf)
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chunks.append(pcm)
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buf.clear()
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# flush remaining
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if len(buf) > 1:
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pcm = decode_chunk(buf)
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if pcm:
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return chunks
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# WebSocket
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@app.websocket("/ws/s2s")
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async def websocket_s2s(websocket: WebSocket):
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print("[WS] client connected")
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loop = asyncio.get_event_loop()
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audio_queue: asyncio.Queue[bytes | None] = asyncio.Queue()
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generating = False
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async def receiver():
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try:
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while True:
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if "bytes" in msg:
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await audio_queue.put(msg["bytes"])
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elif "text" in msg:
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if json.loads(msg["text"]).get("type") == "stop":
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break
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finally:
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await audio_queue.put(None)
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async def vad_and_generate():
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nonlocal generating
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speech_frames: list[np.ndarray] = []
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silence_count = 0
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speech_count = 0
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in_speech = False
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await websocket.send_text(json.dumps({"type": "ready"}))
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if frame_bytes is None:
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break
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frame = np.frombuffer(frame_bytes, dtype=np.int16)
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active = is_speech(frame)
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if active:
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speech_count += 1
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in_speech = True
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speech_frames.append(frame)
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elif in_speech:
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silence_count += 1
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speech_frames.append(frame)
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if silence_count >= VAD_SILENCE_FRAMES and speech_count >= VAD_MIN_SPEECH_FRAMES:
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if not generating:
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generating = True
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utterance = np.concatenate(speech_frames).astype(np.float32) / 32767.0
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speech_frames = []
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silence_count = 0
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speech_count = 0
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in_speech = False
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try:
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await websocket.send_text(json.dumps({"type": "generating"}))
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await websocket.send_bytes(wav_header())
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chunks = await loop.run_in_executor(None, run_generation, utterance)
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for chunk in chunks:
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await websocket.send_bytes(chunk)
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await websocket.send_text(json.dumps({"type": "done"}))
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except Exception as e:
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print(f"[WS] send error: {e}")
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finally:
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generating = False
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try:
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await asyncio.gather(receiver(), vad_and_generate())
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except WebSocketDisconnect:
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print("[WS] client disconnected")
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@app.get("/health")
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async def health():
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return {"status": "ok", "device": DEVICE}
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