tts-dgx-spark-faster-qwen3-tts/config/run_voicedesign_server.py

"""
OpenAI-compatible TTS server for Qwen3-TTS-12Hz-1.7B-VoiceDesign.

Voices are defined in voicedesign_voices.json as:
  { "voice_id": { "instruct": "...", "language": "..." } }

No ref_audio needed — the instruct text fully describes the voice.
"""
import json
import logging
import queue
import threading
import asyncio
import argparse
import numpy as np
import sys

import uvicorn
from fastapi import FastAPI, HTTPException
from fastapi.responses import Response, StreamingResponse, JSONResponse
from pydantic import BaseModel

sys.path.append("/app")
from faster_qwen3_tts.model import FasterQwen3TTS

logger = logging.getLogger(__name__)
logging.basicConfig(level=logging.INFO)

app = FastAPI()
tts_model: FasterQwen3TTS = None
voices: dict = {}
default_voice: str = None
SAMPLE_RATE = 24000
_model_lock = threading.Lock()


# ---------------------------------------------------------------------------
# Request schema (OpenAI TTS compatible)
# ---------------------------------------------------------------------------

class SpeechRequest(BaseModel):
    model: str = "tts-1"
    input: str
    voice: str = "vd_british_male"
    response_format: str = "wav"
    speed: float = 1.0


# ---------------------------------------------------------------------------
# Audio helpers
# ---------------------------------------------------------------------------

def _to_pcm16(audio: np.ndarray) -> bytes:
    return (audio * 32767).clip(-32768, 32767).astype(np.int16).tobytes()


def _wav_header(sample_rate: int) -> bytes:
    import struct
    return struct.pack(
        "<4sI4s4sIHHIIHH4sI",
        b"RIFF", 0xFFFFFFFF, b"WAVE",
        b"fmt ", 16, 1, 1,
        sample_rate, sample_rate * 2, 2, 16,
        b"data", 0xFFFFFFFF,
    )


def _to_mp3_bytes(audio: np.ndarray, sr: int) -> bytes:
    from pydub import AudioSegment
    import io
    pcm = _to_pcm16(audio)
    seg = AudioSegment(pcm, frame_rate=sr, sample_width=2, channels=1)
    buf = io.BytesIO()
    seg.export(buf, format="mp3")
    return buf.getvalue()


def resolve_voice(name: str) -> dict:
    cfg = voices.get(name)
    if cfg:
        return cfg
    if default_voice and default_voice in voices:
        logger.warning("Voice %r not found, falling back to %r", name, default_voice)
        return voices[default_voice]
    raise HTTPException(status_code=404, detail=f"Voice {name!r} not found")


# ---------------------------------------------------------------------------
# Generation helpers
# ---------------------------------------------------------------------------

async def _stream_chunks(voice_cfg: dict, text: str):
    q: queue.Queue = queue.Queue()
    _DONE = object()

    def producer():
        try:
            with _model_lock:
                for chunk, _sr, _timing in tts_model.generate_voice_design_streaming(
                    text=text,
                    instruct=voice_cfg["instruct"],
                    language=voice_cfg.get("language", "English"),
                ):
                    q.put(chunk)
        except Exception as exc:
            q.put(exc)
        finally:
            q.put(_DONE)

    threading.Thread(target=producer, daemon=True).start()
    loop = asyncio.get_event_loop()
    while True:
        item = await loop.run_in_executor(None, q.get)
        if item is _DONE:
            break
        if isinstance(item, Exception):
            raise item
        yield _to_pcm16(item)


# ---------------------------------------------------------------------------
# Endpoints
# ---------------------------------------------------------------------------

@app.get("/health")
async def health():
    return {"status": "ok", "model_loaded": tts_model is not None}


@app.post("/v1/audio/speech")
async def create_speech(req: SpeechRequest):
    if tts_model is None:
        raise HTTPException(status_code=503, detail="Model not loaded")
    if not req.input.strip():
        raise HTTPException(status_code=400, detail="'input' text is empty")

    voice_cfg = resolve_voice(req.voice)
    fmt = req.response_format.lower()

    _CONTENT_TYPES = {"wav": "audio/wav", "pcm": "audio/pcm", "mp3": "audio/mpeg"}
    if fmt not in _CONTENT_TYPES:
        raise HTTPException(status_code=400, detail=f"Unsupported format: {fmt!r}")

    if fmt == "mp3":
        loop = asyncio.get_event_loop()
        def _gen():
            with _model_lock:
                return tts_model.generate_voice_design(
                    text=req.input,
                    instruct=voice_cfg["instruct"],
                    language=voice_cfg.get("language", "English"),
                )
        audio_arrays, sr = await loop.run_in_executor(None, _gen)
        audio = audio_arrays[0] if audio_arrays else np.zeros(1, dtype=np.float32)
        return Response(content=_to_mp3_bytes(audio, sr), media_type="audio/mpeg")

    async def audio_stream():
        if fmt == "wav":
            yield _wav_header(SAMPLE_RATE)
        async for raw in _stream_chunks(voice_cfg, req.input):
            yield raw

    return StreamingResponse(audio_stream(), media_type=_CONTENT_TYPES[fmt])


_voice_list = None
_models_response = None


def _build_voice_list():
    global _voice_list, _models_response
    _voice_list = [{"id": v, "object": "model", "created": 1686935002, "owned_by": "qwen"} for v in voices]
    _models_response = {"object": "list", "data": _voice_list}


@app.get("/v1/models")
async def list_models():
    return _models_response

@app.get("/v1/audio/voices")
async def list_audio_voices():
    return _models_response

@app.get("/v1/audio/models")
async def list_audio_models():
    return _models_response

@app.get("/speakers")
async def get_speakers():
    return list(voices.keys())

@app.options("/{path:path}")
async def options_handler(path: str):
    return JSONResponse(content={"status": "ok"})


# ---------------------------------------------------------------------------
# Entry point
# ---------------------------------------------------------------------------

def main():
    global tts_model, voices, default_voice, SAMPLE_RATE

    parser = argparse.ArgumentParser()
    parser.add_argument("--model", default="/models/Qwen3-TTS-VoiceDesign")
    parser.add_argument("--voices", default="/config/voicedesign_voices.json")
    parser.add_argument("--port", type=int, default=8000)
    parser.add_argument("--host", default="0.0.0.0")
    parser.add_argument("--device", default="cuda")
    parser.add_argument("--max-seq-len", type=int, default=2048)
    args = parser.parse_args()

    with open(args.voices) as f:
        voices = json.load(f)
    default_voice = next(iter(voices), None)
    _build_voice_list()

    import torch
    logger.info("Loading VoiceDesign model %s …", args.model)
    tts_model = FasterQwen3TTS.from_pretrained(
        args.model,
        device=args.device,
        dtype=torch.bfloat16,
        attn_implementation="sdpa",
        max_seq_len=args.max_seq_len,
    )
    SAMPLE_RATE = tts_model.sample_rate
    logger.info("Model ready. Sample rate: %d Hz", SAMPLE_RATE)

    uvicorn.run(app, host=args.host, port=args.port, log_level="info")


if __name__ == "__main__":
    main()