tts-server/speech_tts_onnx.py

from __future__ import annotations

import base64
import hashlib
import io
import json
import logging
import os
import re
import asyncio
import threading
import uuid
import urllib.request
from contextlib import asynccontextmanager
from pathlib import Path
from typing import Dict, List, Optional, Tuple

import numpy as np
import soundfile as sf
from fastapi import Body, FastAPI, HTTPException, Query, Request
from fastapi.middleware.cors import CORSMiddleware
from fastapi.responses import StreamingResponse
from pydantic import BaseModel, field_validator


# ============================================================
# 配置
# ============================================================

DEFAULT_MODEL_NAME = os.getenv("TTS_ONNX_MODEL_NAME", "model.onnx")
MODEL_DIR = os.getenv("TTS_ONNX_MODEL_DIR", "/home/tts-server/onnx")
HF_MODEL_ID = os.getenv("TTS_ONNX_HF_MODEL_ID", "onnx-community/Kokoro-82M-ONNX")
TOKENIZER_PATH = os.getenv("TTS_ONNX_TOKENIZER_PATH", str(Path(MODEL_DIR) / "tokenizer.json"))
CONFIG_PATH = os.getenv("TTS_ONNX_CONFIG_PATH", str(Path(MODEL_DIR) / "config.json"))
VOICES_DIR = os.getenv("TTS_ONNX_VOICES_DIR", str(Path(MODEL_DIR) / "voices"))
VOICES_V1_PATH = os.getenv("TTS_ONNX_VOICES_V1_PATH", str(Path(MODEL_DIR) / "voices-v1.0.bin"))
CACHE_DIR = os.getenv("CACHE_DIR", "./audio_cache")
LOG_LEVEL = os.getenv("LOG_LEVEL", "INFO").upper()
MAX_CONCURRENT_REQUESTS = int(os.getenv("MAX_CONCURRENT_REQUESTS", 8))
MEMORY_CACHE_SIZE = int(os.getenv("MEMORY_CACHE_SIZE", 200))
DISK_CACHE_SIZE = int(os.getenv("DISK_CACHE_SIZE", 500))
DEFAULT_SAMPLE_RATE = int(os.getenv("TTS_SAMPLE_RATE", 24000))
VOICE_ALIASES = {}

logging.basicConfig(level=getattr(logging, LOG_LEVEL, logging.INFO))
logger = logging.getLogger("speech_tts_onnx")

Path(CACHE_DIR).mkdir(parents=True, exist_ok=True)
SENT_SPLIT_RE = re.compile(r"(?<=[.!?,:])\s+")
_TOKENIZER_CACHE = None
_VOCAB_CACHE = None
_EN_G2P_PIPELINE = None
_KOKORO_ONNX_ENGINE = None


# ============================================================
# 运行时状态
# ============================================================

model_lock = threading.Lock()
synthesis_lock = threading.Lock()
request_semaphore = asyncio.Semaphore(MAX_CONCURRENT_REQUESTS)
current_requests: Dict[str, Dict] = {}

model_session = None
model_name = DEFAULT_MODEL_NAME
sample_rate = DEFAULT_SAMPLE_RATE


# ============================================================
# 工具
# ============================================================

def split_sentences(text: str) -> List[str]:
    return [s.strip() for s in SENT_SPLIT_RE.split(text.strip()) if s.strip()]


def sentence_cache_key(sentence: str, voice: str, speed: float, model: str) -> str:
    raw = f"{sentence}|{voice}|{speed}|{model}"
    return hashlib.md5(raw.encode("utf-8")).hexdigest()


def sentence_cache_path(key: str) -> str:
    return os.path.join(CACHE_DIR, f"{key}.wav")


def meta_cache_path(key: str) -> str:
    return os.path.join(CACHE_DIR, f"{key}.json")


def to_mono_numpy(audio) -> np.ndarray:
    if audio is None:
        return np.array([], dtype=np.float32)

    if isinstance(audio, np.ndarray):
        arr = audio
    else:
        try:
            arr = np.asarray(audio)
        except Exception:
            return np.array([], dtype=np.float32)

    arr = np.asarray(arr)
    if arr.size == 0:
        return np.array([], dtype=np.float32)

    if arr.ndim == 2:
        if arr.shape[0] == 1:
            arr = arr[0]
        elif arr.shape[1] == 1:
            arr = arr[:, 0]
        else:
            arr = arr.mean(axis=1)
    elif arr.ndim > 2:
        arr = arr.reshape(-1)

    if arr.ndim == 0:
        arr = arr.reshape(1)

    if arr.dtype != np.float32:
        arr = arr.astype(np.float32, copy=False)

    return arr


def read_wav_bytes(wav_path: str) -> Tuple[bytes, int]:
    audio, sr = sf.read(wav_path)
    buf = io.BytesIO()
    with sf.SoundFile(
        buf,
        "w",
        samplerate=sr,
        channels=audio.shape[1] if audio.ndim > 1 else 1,
        format="WAV",
        subtype="PCM_16",
    ) as f:
        f.write(audio)
    return buf.getvalue(), sr


# ============================================================
# 模型加载
# ============================================================

def resolve_model_path(name: str) -> str:
    local_path = Path(MODEL_DIR) / name
    if local_path.exists():
        return str(local_path)

    if os.path.isabs(name) and Path(name).exists():
        return name

    raise FileNotFoundError(
        f"找不到模型文件: {local_path}"
    )


def load_vocab() -> Dict[str, int]:
    """
    优先使用 Kokoro 官方 config.json 的 vocab。
    如果本地缺失，则回退到 tokenizer.json 里的 vocab。
    """
    config_file = Path(CONFIG_PATH)
    if config_file.exists():
        try:
            data = json.loads(config_file.read_text(encoding="utf-8"))
            vocab = data["vocab"]
            if isinstance(vocab, dict) and vocab:
                return {str(k): int(v) for k, v in vocab.items()}
        except Exception as e:
            logger.warning("加载 config vocab 失败，回退 tokenizer vocab: %s", e)

    tokenizer_file = Path(TOKENIZER_PATH)
    if not tokenizer_file.exists():
        raise RuntimeError(
            f"缺少 vocab 文件: {config_file} / {tokenizer_file}. "
            f"请从 {HF_MODEL_ID} 或 hexgrad/Kokoro-82M 下载后放到模型目录。"
        )
    try:
        data = json.loads(tokenizer_file.read_text(encoding="utf-8"))
        vocab = data["model"]["vocab"]
        if not isinstance(vocab, dict) or not vocab:
            raise ValueError("tokenizer vocab 为空或格式异常")
        return {str(k): int(v) for k, v in vocab.items()}
    except Exception as e:
        raise RuntimeError(f"无法加载 tokenizer vocab: {tokenizer_file}. 错误: {e}") from e


def get_vocab() -> Dict[str, int]:
    global _VOCAB_CACHE
    if _VOCAB_CACHE is None:
        _VOCAB_CACHE = load_vocab()
    return _VOCAB_CACHE


def get_en_g2p_pipeline():
    global _EN_G2P_PIPELINE
    if _EN_G2P_PIPELINE is None:
        from kokoro import KPipeline  # type: ignore

        _EN_G2P_PIPELINE = KPipeline(lang_code="a", repo_id="hexgrad/Kokoro-82M", model=False)
    return _EN_G2P_PIPELINE


def resolve_voice_path(voice: str) -> Path:
    voice_name = voice.strip()
    if not voice_name:
        voice_name = "af_heart"
    voice_name = VOICE_ALIASES.get(voice_name, voice_name)
    if not voice_name.endswith(".bin"):
        voice_name = f"{voice_name}.bin"
    voice_path = Path(VOICES_DIR) / voice_name
    if voice_path.exists():
        return voice_path
    fallback_path = Path(VOICES_DIR) / "af_bella.bin"
    if fallback_path.exists():
        logger.warning("voice %s 不存在，回退到 %s", voice_name, fallback_path.name)
        return fallback_path
    raise FileNotFoundError(
        f"找不到 voice 文件: {voice_path}. "
        "请从模型仓库下载 voices/*.bin 到本地 voices 目录。"
    )


def _download_voice_file(voice_path: Path) -> Path:
    voice_name = voice_path.name
    raw_url = f"https://huggingface.co/{HF_MODEL_ID}/resolve/main/voices/{voice_name}"
    voice_path.parent.mkdir(parents=True, exist_ok=True)
    tmp_path = voice_path.with_suffix(voice_path.suffix + ".tmp")

    logger.info("从官方地址下载 voice 文件: %s", raw_url)
    with urllib.request.urlopen(raw_url, timeout=60) as response:
        content_type = response.headers.get_content_type()
        payload = response.read()

    if content_type == "text/html" or payload.startswith(b"<!doctype html") or payload.startswith(b"<html"):
        raise RuntimeError(
            f"下载到的仍是 HTML 页面而不是 voice 二进制文件: {raw_url}. "
            "请确认模型仓库的 voices 文件可直接通过 raw 链接访问。"
        )

    with open(tmp_path, "wb") as f:
        f.write(payload)
    tmp_path.replace(voice_path)
    return voice_path


def _is_html_payload(data: bytes) -> bool:
    return data.startswith(b"<!doctype html") or data.startswith(b"<html")


def load_onnx_session(name: str):
    try:
        import onnxruntime as ort  # type: ignore
    except Exception as e:
        raise RuntimeError(
            "无法导入 onnxruntime。请在部署环境中安装 onnxruntime。"
        ) from e

    model_path = resolve_model_path(name)
    providers = ["CPUExecutionProvider"]
    session = ort.InferenceSession(model_path, providers=providers)
    logger.info("ONNX 模型加载完成: %s", model_path)
    return session


def load_kokoro_engine(name: str):
    try:
        from kokoro_onnx import Kokoro  # type: ignore
    except Exception as e:
        raise RuntimeError("无法导入 kokoro_onnx。请在部署环境中安装 kokoro-onnx。") from e

    model_path = resolve_model_path(name)
    engine = Kokoro(
        model_path=model_path,
        voices_path=VOICES_V1_PATH,
        vocab_config=CONFIG_PATH if Path(CONFIG_PATH).exists() else None,
    )
    logger.info("kokoro-onnx 引擎加载完成: %s", model_path)
    return engine


def load_model(force_reload: bool = False, name: Optional[str] = None):
    global model_session, model_name, _KOKORO_ONNX_ENGINE
    with model_lock:
        target_name = name or model_name
        if force_reload or model_session is None or target_name != model_name:
            model_session = load_onnx_session(target_name)
            _KOKORO_ONNX_ENGINE = load_kokoro_engine(target_name)
            model_name = target_name
        return model_session


def get_kokoro_engine(name: Optional[str] = None):
    global _KOKORO_ONNX_ENGINE
    target_name = name or model_name
    if _KOKORO_ONNX_ENGINE is None or target_name != model_name:
        load_model(name=target_name)
    return _KOKORO_ONNX_ENGINE


@asynccontextmanager
async def lifespan(app: FastAPI):
    try:
        load_model()
        yield
    finally:
        logger.info("应用关闭中...")


app = FastAPI(title="Online TTS Service (Kokoro ONNX)", lifespan=lifespan)

app.add_middleware(
    CORSMiddleware,
    allow_origins=["*"],
    allow_credentials=True,
    allow_methods=["*"],
    allow_headers=["*"],
)


@app.middleware("http")
async def track_clients(request: Request, call_next):
    client_id = (
        request.query_params.get("client_id")
        or request.headers.get("X-Client-ID")
        or str(uuid.uuid4())
    )
    if request.url.path == "/generate":
        current_requests[client_id] = {"active": True}

    response = await call_next(request)

    if request.url.path == "/generate":
        current_requests.pop(client_id, None)

    return response


class TTSRequest(BaseModel):
    text: str
    voice: Optional[str] = "af_heart"
    speed: Optional[float] = 1.0
    split_pattern: Optional[str] = r"\n+"
    model_name: Optional[str] = None

    @field_validator("speed")
    @classmethod
    def validate_speed(cls, v):
        if v is None:
            return 1.0
        v = float(v)
        if v <= 0:
            raise ValueError("speed 必须大于 0")
        return v


def _encode_token_ids(text: str) -> List[int]:
    vocab = get_vocab()
    unknown_id = vocab.get(" ", 16)
    return [vocab.get(ch, unknown_id) for ch in text]


def _phonemize_en_chunks_for_onnx(text: str) -> List[Tuple[str, str]]:
    pipeline = get_en_g2p_pipeline()
    _, tokens = pipeline.g2p(text)
    chunks = list(pipeline.en_tokenize(tokens))
    if not chunks:
        raise RuntimeError("英文文本音素化失败，未生成 phonemes。")
    return [(graphemes, phonemes) for graphemes, phonemes, _ in chunks]


def _phonemize_en_for_onnx(text: str) -> str:
    return _phonemize_en_chunks_for_onnx(text)[0][1]


def _tokenize_for_onnx(text: str, voice: str):
    """
    将文本转成 ONNX 所需 token。
    这里优先复用 kokoro 的预处理链。
    如果你项目里有更稳定的 tokenizer，可以替换这个函数。
    """
    normalized_voice = VOICE_ALIASES.get((voice or "").strip(), (voice or "").strip())
    if normalized_voice.startswith(("af_", "am_", "bf_", "bm_")):
        phonemes = _phonemize_en_for_onnx(text)
        tokens = _encode_token_ids(phonemes)
    else:
        tokens = _encode_token_ids(text)

    if not tokens:
        raise RuntimeError("文本编码后得到空 tokens。请检查 tokenizer.json 是否正确。")

    if len(tokens) > 510:
        raise RuntimeError(
            f"文本过长，tokens={len(tokens)}，超过模型 512 上限。请拆句后再调用。"
        )

    return np.asarray([[0, *tokens, 0]], dtype=np.int64)


def _is_english_voice(voice: str) -> bool:
    normalized_voice = VOICE_ALIASES.get((voice or "").strip(), (voice or "").strip())
    return normalized_voice.startswith(("af_", "am_", "bf_", "bm_"))


def _style_for_voice(voice: str) -> np.ndarray:
    """
    ONNX 模型需要 style 向量。
    这里先提供一个可运行的占位实现，后续可以替换成正式的 voice embedding 映射。
    """
    voice_path = resolve_voice_path(voice)
    if voice_path.suffix == ".bin":
        try:
            header = voice_path.read_bytes()[:64]
            if _is_html_payload(header):
                fallback_path = Path(VOICES_DIR) / "af_bella.bin"
                if voice_path.name != fallback_path.name and fallback_path.exists():
                    logger.warning("voice 文件 %s 是 HTML，占位回退到 %s", voice_path.name, fallback_path.name)
                    voice_path = fallback_path
                else:
                    voice_path = _download_voice_file(voice_path)
        except Exception as e:
            logger.warning("voice 文件检查失败，尝试直接重新下载: %s", e)
            fallback_path = Path(VOICES_DIR) / "af_bella.bin"
            if voice_path.name != fallback_path.name and fallback_path.exists():
                voice_path = fallback_path
            else:
                voice_path = _download_voice_file(voice_path)

    style = np.fromfile(str(voice_path), dtype=np.float32)
    if style.size == 0:
        raise RuntimeError(f"voice 文件为空: {voice_path}")
    if style.size % 256 != 0:
        # 某些仓库文件可能是 xet pointer 或 HTML 页面，重新下载一次兜底。
        voice_path = _download_voice_file(voice_path)
        style = np.fromfile(str(voice_path), dtype=np.float32)
        if style.size == 0 or style.size % 256 != 0:
            raise RuntimeError(f"voice 文件维度异常: {voice_path}, size={style.size}")
    style = style.reshape(-1, 256)
    return style


def _select_style_slice(style: np.ndarray, token_len: int) -> np.ndarray:
    if style.ndim != 2 or style.shape[-1] != 256:
        raise RuntimeError(f"style 维度异常: {style.shape}")
    # Follow the official ONNX example: ref_s = voices[len(tokens)]
    idx = min(max(token_len, 0), style.shape[0] - 1)
    return style[idx : idx + 1]


def _prepare_style_input(session, style_slice: np.ndarray) -> Tuple[Optional[str], Optional[np.ndarray]]:
    for input_name in ("style", "ref_s"):
        for model_input in session.get_inputs():
            if model_input.name != input_name:
                continue

            input_shape = model_input.shape
            expected_rank = len(input_shape) if input_shape is not None else style_slice.ndim

            if expected_rank == 2:
                return input_name, style_slice.astype(np.float32, copy=False)
            if expected_rank == 3:
                return input_name, style_slice[:, np.newaxis, :].astype(np.float32, copy=False)

            raise RuntimeError(
                f"模型输入 {input_name} 的 rank 不受支持: shape={input_shape}"
            )

    return None, None


def synthesize_audio(text: str, voice: str, speed: float, model_name: Optional[str] = None) -> np.ndarray:
    if not text.strip():
        raise HTTPException(status_code=400, detail="文本不能为空")

    engine = get_kokoro_engine(name=model_name)
    session = load_model(name=model_name)
    available_voices = set(engine.get_voices())
    if voice not in available_voices:
        raise HTTPException(status_code=400, detail=f"不支持的 voice: {voice}")

    phonemes = engine.tokenizer.phonemize(text, "en-us")
    batched_phonemes = engine._split_phonemes(phonemes)
    if not batched_phonemes:
        raise HTTPException(status_code=400, detail="文本音素化失败")

    voice_style = engine.get_voice_style(voice)
    audio_segments: List[np.ndarray] = []
    for phoneme_batch in batched_phonemes:
        tokens = np.array(engine.tokenizer.tokenize(phoneme_batch), dtype=np.int64)
        if tokens.size == 0:
            continue
        style = voice_style[len(tokens)]
        feeds = {
            "input_ids": np.asarray([[0, *tokens.tolist(), 0]], dtype=np.int64),
            "style": np.asarray(style, dtype=np.float32),
            "speed": np.asarray([speed], dtype=np.float32),
        }
        outputs = session.run(None, feeds)
        if outputs:
            audio_segments.append(to_mono_numpy(outputs[0]))

    if not audio_segments:
        raise HTTPException(status_code=500, detail="ONNX 推理未返回音频输出")

    audio = np.concatenate(audio_segments, axis=0) if len(audio_segments) > 1 else audio_segments[0]
    if audio.size == 0 or not np.isfinite(audio).all():
        raise HTTPException(status_code=500, detail="生成的音频无效")
    return audio


def synthesize_wav_bytes(text: str, voice: str, speed: float, model_name: Optional[str] = None) -> io.BytesIO:
    segments: List[np.ndarray] = []
    if _is_english_voice(voice):
        parts = [graphemes for graphemes, _ in _phonemize_en_chunks_for_onnx(text)] if text.strip() else []
    else:
        parts = split_sentences(text) if text.strip() else []
    if not parts:
        parts = [text.strip()]

    with synthesis_lock:
        for part in parts:
            audio = synthesize_audio(part, voice=voice, speed=speed, model_name=model_name)
            if audio.size > 0:
                segments.append(audio)

    if not segments:
        raise HTTPException(status_code=400, detail="未生成音频，请检查输入文本或参数。")

    audio_concat = np.concatenate(segments, axis=0)
    buf = io.BytesIO()
    sf.write(buf, audio_concat, samplerate=sample_rate, format="WAV", subtype="PCM_16")
    buf.seek(0)
    return buf


@app.post("/tts", summary="POST: 传入文本返回 WAV 流")
def tts_post(req: TTSRequest):
    buf = synthesize_wav_bytes(
        text=req.text,
        voice=req.voice or "af_heart",
        speed=req.speed if req.speed is not None else 1.0,
        model_name=req.model_name,
    )
    return StreamingResponse(
        buf,
        media_type="audio/wav",
        headers={"Content-Disposition": 'inline; filename="tts.wav"'},
    )


@app.get("/tts", summary="GET: 传入文本返回 WAV 流")
def tts_get(
    text: str = Query(..., description="待合成文本"),
    voice: str = Query("af_heart"),
    speed: float = Query(1.0),
    model_name: str = Query(DEFAULT_MODEL_NAME),
):
    if speed is None or float(speed) <= 0:
        raise HTTPException(status_code=400, detail="speed 必须为大于 0 的数值")
    buf = synthesize_wav_bytes(
        text=text,
        voice=voice,
        speed=float(speed),
        model_name=model_name,
    )
    return StreamingResponse(
        buf,
        media_type="audio/wav",
        headers={"Content-Disposition": 'inline; filename="tts.wav"'},
    )


@app.post("/generate")
async def generate_audio_stream(data: Dict = Body(...)):
    async with request_semaphore:
        text = data.get("text", "")
        voice = data.get("voice", "af_heart")
        speed = float(data.get("speed", 1.0))
        model = data.get("model_name", DEFAULT_MODEL_NAME)
        client_id = data.get("client_id", str(uuid.uuid4()))

        if not text.strip():
            raise HTTPException(status_code=400, detail="文本不能为空")

        if _is_english_voice(voice):
            parts = [graphemes for graphemes, _ in _phonemize_en_chunks_for_onnx(text)] if text.strip() else []
        else:
            parts = split_sentences(text) if text.strip() else []
        if not parts:
            parts = [text.strip()]

        if client_id in current_requests:
            current_requests[client_id]["interrupt"] = True
            await asyncio.sleep(0.05)

        current_requests[client_id] = {"interrupt": False}

        async def stream():
            try:
                for idx, part in enumerate(parts):
                    if not part:
                        continue
                    if current_requests.get(client_id, {}).get("interrupt"):
                        break

                    audio = await asyncio.to_thread(
                        synthesize_audio,
                        text=part,
                        voice=voice,
                        speed=speed,
                        model_name=model,
                    )

                    with io.BytesIO() as buf:
                        with sf.SoundFile(
                            buf,
                            "w",
                            sample_rate,
                            channels=audio.shape[1] if audio.ndim > 1 else 1,
                            format="WAV",
                            subtype="PCM_16",
                        ) as f:
                            f.write(audio)
                        audio_b64 = base64.b64encode(buf.getvalue()).decode()

                    yield json.dumps(
                        {
                            "index": idx,
                            "sentence": part,
                            "audio": audio_b64,
                            "sample_rate": sample_rate,
                        }
                    ).encode() + b"\n"
            finally:
                current_requests.pop(client_id, None)

        return StreamingResponse(stream(), media_type="application/x-ndjson")


if __name__ == "__main__":
    import uvicorn

    uvicorn.run("speech_tts_onnx:app", host="0.0.0.0", port=18000, reload=False, workers=1)