GPT-SoVITS 接口测试台
+选择 3-10 秒参考音频或视频(视频会自动提取音频),在波形图上拖拽选取裁剪区域,填写后端接口地址和生成文本,直接调用中间层 /api/tts。
diff --git a/README.md b/README.md
index 2511c73c..819242b6 100644
--- a/README.md
+++ b/README.md
@@ -27,6 +27,44 @@ A Powerful Few-shot Voice Conversion and Text-to-Speech WebUI.
---
+## 简化接口 / 测试前端
+
+本工作区新增了一个用于 MVP 调用的中间层接口和测试前端:
+
+| 页面 | 地址 | 说明 |
+|------|------|------|
+| 测试前端 | http://127.0.0.1:9881/test/ | 带波形裁剪、视频转音频的测试 UI |
+| Swagger UI | http://127.0.0.1:9881/docs | 交互式 API 文档 |
+| ReDoc | http://127.0.0.1:9881/redoc | 可读式 API 文档 |
+
+文档:
+
+- 快速启动:[docs/simple_api_quickstart.md](./docs/simple_api_quickstart.md)
+- 完整教程:[docs/simple_api.md](./docs/simple_api.md)
+
+核心功能:
+
+- `/api/tts` — 上传参考音频/视频 + 文字,直接返回生成的音频
+- 前端支持视频上传,自动提取音频
+- 前端波形裁剪工具,可选择 3-10 秒片段
+- 5 种情绪预设(neutral/happy/calm/sad/angry)
+- 7 个契约测试,无需 GPU 即可运行
+
+启动命令:
+
+```
+cd D:\tts\GPT-SoVITS
+.\go-simple-api.ps1
+```
+
+运行测试:
+
+```
+python -m unittest tests.test_simple_api_contract -v
+```
+
+---
+
## Features:
1. **Zero-shot TTS:** Input a 5-second vocal sample and experience instant text-to-speech conversion.
@@ -478,3 +516,4 @@ Thankful to @Naozumi520 for providing the Cantonese training set and for the gui
+
diff --git a/docs/simple_api.md b/docs/simple_api.md
new file mode 100644
index 00000000..f9e5cc09
--- /dev/null
+++ b/docs/simple_api.md
@@ -0,0 +1,402 @@
+# GPT-SoVITS 简化接口文档
+
+本项目新增 `simple_api.py` 作为中间层,封装 GPT-SoVITS 推理引擎,提供更简洁的调用方式。
+
+## 快速开始
+
+```bash
+# 安装依赖
+python -m pip install -r requirements.txt
+
+# 启动
+python simple_api.py -c simple_api.yaml
+
+# 访问
+Swagger UI: http://127.0.0.1:9881/docs
+ReDoc: http://127.0.0.1:9881/redoc
+测试前端: http://127.0.0.1:9881/test/
+```
+
+## 接口总览
+
+| 方法 | 路径 | 说明 | 标签 |
+|------|------|------|------|
+| GET | `/health` | 健康检查(含 GPU 信息) | System |
+| GET | `/voices` | 列出 voice profiles | System |
+| **POST** | **`/api/tts`** | **核心 TTS 接口(MVP)** | **MVP** |
+| GET | `/speak` | voice profile TTS (GET) | Profile |
+| POST | `/speak` | voice profile TTS (POST) | Profile |
+| POST | `/v1/tts` | OpenAI 兼容格式 TTS | Profile |
+| POST | `/speak/base64` | 返回 Base64 音频 | Profile |
+| POST | `/admin/reload-config` | 热加载配置 | Admin |
+| POST | `/admin/weights` | 切换模型权重 | Admin |
+
+---
+
+## 1. POST /api/tts — 核心 TTS 接口
+
+**推荐使用此接口**。上传参考音频和文字,直接返回生成的音频。
+
+### 请求格式
+
+```
+Content-Type: multipart/form-data
+```
+
+### 字段说明
+
+| 字段 | 类型 | 必填 | 默认值 | 说明 |
+|------|------|------|--------|------|
+| `text` | string | **是** | — | 需要生成的文字 |
+| `ref_audio` | file | **是** | — | 主参考音频,3-10 秒(支持 wav/flac/ogg/mp3/m4a/aac) |
+| `aux_ref_audio` | file[] | 否 | — | 辅助参考音频,可上传多个 |
+| `prompt_text` | string | 否 | `""` | 主参考音频对应文字(v2 可留空;v3/v4 必填) |
+| `text_lang` | string | 否 | `zh` | 生成文字语言:zh/en/ja/ko/yue/auto |
+| `prompt_lang` | string | 否 | `zh` | 参考音频语言:zh/en/ja/ko/yue/auto |
+| `format` | string | 否 | `wav` | 返回格式:wav/ogg/aac/raw |
+| `emotion` | string | 否 | `neutral` | 情绪预设:neutral/happy/calm/sad/angry |
+| `speed` | float | 否 | — | 语速(0.5-2.0),覆盖情绪预设中的语速 |
+| `seed` | int | 否 | `-1` | 随机种子,-1 为随机 |
+
+### 情绪预设参数映射
+
+| 情绪 | temperature | top_p | top_k | speed_factor | repetition_penalty |
+|------|-------------|-------|-------|--------------|-------------------|
+| neutral | — | — | — | — | — |
+| happy | 1.1 | 0.95 | — | — | — |
+| calm | 0.8 | 0.85 | — | 0.92 | — |
+| sad | 0.75 | 0.85 | — | 0.9 | — |
+| angry | 1.2 | — | 20 | — | 1.25 |
+
+> 显式传入 `speed` 会覆盖情绪预设中的 `speed_factor`。
+
+### curl 示例
+
+**基础调用:**
+
+```powershell
+curl.exe -X POST http://127.0.0.1:9881/api/tts `
+ -F "text=你好,欢迎使用这个声音。" `
+ -F "ref_audio=@D:\audio\ref.wav" `
+ --output output.wav
+```
+
+**带辅助参考音频和情绪:**
+
+```powershell
+curl.exe -X POST http://127.0.0.1:9881/api/tts `
+ -F "text=你好,欢迎使用这个声音。" `
+ -F "ref_audio=@D:\audio\ref.wav" `
+ -F "aux_ref_audio=@D:\audio\aux1.wav" `
+ -F "emotion=happy" `
+ -F "speed=1.1" `
+ --output output.wav
+```
+
+**Linux/macOS:**
+
+```bash
+curl -X POST http://127.0.0.1:9881/api/tts \
+ -F "text=你好,欢迎使用这个声音。" \
+ -F "ref_audio=@/path/to/ref.wav" \
+ -F "emotion=calm" \
+ --output output.wav
+```
+
+### 返回
+
+- 成功:音频二进制流(Content-Type: `audio/wav` 等)
+- 失败:JSON 错误信息
+
+```json
+{"message": "tts failed", "exception": "..."}
+```
+
+### 常见错误
+
+| HTTP 状态码 | 原因 |
+|------------|------|
+| 400 | text 为空 / ref_audio 缺失 / 音频时长不在 3-10 秒 / 不支持的 format / v3/v4 时 prompt_text 为空 |
+| 404 | voice profile 不存在(仅 /speak 接口) |
+| 503 | TTS pipeline 未就绪(模型未加载) |
+
+---
+
+## 2. GET /health — 健康检查
+
+```bash
+curl http://127.0.0.1:9881/health
+```
+
+返回示例:
+
+```json
+{
+ "status": "ok",
+ "tts_config": "GPT_SoVITS/configs/tts_infer.yaml",
+ "version": "v2",
+ "languages": ["auto", "en", "zh"],
+ "pid": 12345,
+ "memory_mb": 2048.5,
+ "gpu": {
+ "name": "NVIDIA GeForce RTX 3080",
+ "memory_used_mb": 4096.2,
+ "memory_total_mb": 10240.0
+ }
+}
+```
+
+---
+
+## 3. GET /voices — 列出 voice profiles
+
+```bash
+curl http://127.0.0.1:9881/voices
+```
+
+返回示例:
+
+```json
+{
+ "default_voice": "default",
+ "voices": [
+ {
+ "name": "default",
+ "description": "Replace this profile with your reference voice.",
+ "text_lang": "zh",
+ "prompt_lang": "zh",
+ "ref_audio_path": "reference.wav",
+ "ready": true
+ }
+ ]
+}
+```
+
+---
+
+## 4. POST /speak — voice profile TTS
+
+基于 `simple_api.yaml` 中配置的 voice profile 调用 TTS。
+
+### 请求体(JSON)
+
+```json
+{
+ "text": "hello world",
+ "voice": "default",
+ "text_lang": "zh",
+ "format": "wav",
+ "speed": 1.0
+}
+```
+
+| 字段 | 类型 | 必填 | 说明 |
+|------|------|------|------|
+| `text` | string | **是** | 需要生成的文字 |
+| `voice` | string | 否 | voice profile 名称,不传则使用 default |
+| `text_lang` | string | 否 | 生成文字语言 |
+| `format` | string | 否 | 返回格式 |
+| `stream` | bool | 否 | 是否流式返回 |
+| `speed` | float | 否 | 语速 |
+
+### curl 示例
+
+```bash
+curl -X POST http://127.0.0.1:9881/speak \
+ -H "Content-Type: application/json" \
+ -d '{"text":"你好世界","voice":"default"}' \
+ --output output.wav
+```
+
+---
+
+## 5. GET /speak — voice profile TTS (GET)
+
+与 POST /speak 相同,但通过 URL 参数传递。
+
+```
+GET /speak?text=hello&voice=default&format=wav
+```
+
+---
+
+## 6. POST /speak/base64 — 返回 Base64 音频
+
+返回 Base64 编码的音频,适合 Web 前端直接使用。
+
+```bash
+curl -X POST http://127.0.0.1:9881/speak/base64 \
+ -H "Content-Type: application/json" \
+ -d '{"text":"hello","voice":"default"}'
+```
+
+返回:
+
+```json
+{
+ "media_type": "audio/wav",
+ "audio_base64": "UklGRi..."
+}
+```
+
+---
+
+## 7. POST /v1/tts — OpenAI 兼容格式
+
+请求格式与 POST /speak 相同,路径兼容 OpenAI TTS API 风格。
+
+---
+
+## 8. POST /admin/reload-config — 热加载配置
+
+重新加载 `simple_api.yaml`,无需重启服务。
+
+```bash
+curl -X POST http://127.0.0.1:9881/admin/reload-config
+```
+
+返回:`{"message": "success", "default_voice": "default"}`
+
+---
+
+## 9. POST /admin/weights — 切换模型权重
+
+运行时切换 GPT-SoVITS 模型权重文件。
+
+```bash
+curl -X POST http://127.0.0.1:9881/admin/weights \
+ -H "Content-Type: application/json" \
+ -d '{"gpt_weights_path":"path/to/gpt.pt","sovits_weights_path":"path/to/sovits.pt"}'
+```
+
+---
+
+## 配置文件
+
+`simple_api.yaml`:
+
+```yaml
+server:
+ host: 127.0.0.1
+ port: 9881
+ tts_config: GPT_SoVITS/configs/tts_infer.yaml
+
+cors_allow_origins:
+ - "*"
+
+upload:
+ dir: runtime/uploads
+ min_ref_seconds: 3
+ max_ref_seconds: 10
+ max_upload_mb: 80
+
+defaults:
+ text_lang: zh
+ prompt_lang: zh
+ media_type: wav
+ text_split_method: cut5
+ batch_size: 1
+ speed_factor: 1.0
+ seed: -1
+
+emotion_presets:
+ neutral: {}
+ happy:
+ temperature: 1.1
+ top_p: 0.95
+ calm:
+ temperature: 0.8
+ top_p: 0.85
+ speed_factor: 0.92
+ sad:
+ temperature: 0.75
+ top_p: 0.85
+ speed_factor: 0.9
+ angry:
+ temperature: 1.2
+ top_k: 20
+ repetition_penalty: 1.25
+
+voices:
+ default:
+ description: Replace this profile with your reference voice.
+ ref_audio_path: reference.wav
+ prompt_text: Replace this with the exact text spoken in reference.wav.
+ prompt_lang: zh
+ text_lang: zh
+```
+
+### 配置说明
+
+| 配置项 | 说明 |
+|--------|------|
+| `server.host` | 监听地址 |
+| `server.port` | 监听端口 |
+| `server.tts_config` | GPT-SoVITS 推理配置文件路径 |
+| `upload.dir` | 临时上传目录 |
+| `upload.min_ref_seconds` | 主参考音频最短秒数 |
+| `upload.max_ref_seconds` | 主参考音频最长秒数 |
+| `upload.max_upload_mb` | 单个上传文件最大体积 (MB) |
+| `defaults.*` | 所有接口的默认参数 |
+| `emotion_presets.*` | 情绪预设参数映射 |
+| `voices.*` | 固定音色 profile |
+
+---
+
+## 添加自定义音色
+
+编辑 `simple_api.yaml`,在 `voices` 下添加:
+
+```yaml
+voices:
+ narrator:
+ description: "男声旁白"
+ ref_audio_path: voices/narrator.wav
+ prompt_text: "旁白参考音频的逐字稿"
+ prompt_lang: zh
+ text_lang: zh
+```
+
+然后热加载:
+
+```bash
+curl -X POST http://127.0.0.1:9881/admin/reload-config
+```
+
+---
+
+## 测试
+
+### 契约测试(无需 GPU)
+
+```bash
+python -m unittest tests.test_simple_api_contract -v
+```
+
+覆盖:
+
+- `/api/tts` 路由注册
+- 上传接口参数构造
+- 主参考音频 3-10 秒校验
+- v2 空 prompt_text 允许 / v3/v4 空 prompt_text 拒绝
+- 临时上传目录清理
+- 情绪预设应用与 speed 覆盖
+
+### 前端测试
+
+1. 启动后端
+2. 访问 `http://127.0.0.1:9881/test/`
+3. 上传音频或视频(视频会自动提取音频)
+4. 使用波形裁剪工具选择 3-10 秒片段
+5. 填写文字,选择情绪和语速
+6. 点击生成
+
+---
+
+## 启动脚本
+
+| 脚本 | 平台 | 说明 |
+|------|------|------|
+| `go-simple-api.ps1` | Windows PowerShell | 自动检测 runtime\python.exe |
+| `go-simple-api.bat` | Windows CMD | 同上 |
+| `open-test-frontend.ps1` | Windows PowerShell | 直接打开测试前端 HTML |
diff --git a/docs/simple_api_quickstart.md b/docs/simple_api_quickstart.md
new file mode 100644
index 00000000..b3865fa9
--- /dev/null
+++ b/docs/simple_api_quickstart.md
@@ -0,0 +1,85 @@
+# 简化 TTS 接口快速启动
+
+完整教程见:`docs/simple_api.md`
+
+## 一句话流程
+
+启动后端,打开测试页,上传 3-10 秒参考音频或视频(视频自动提取音频),填写生成文字,点击生成。
+
+## 1. 启动后端
+
+```powershell
+cd D:\tts\GPT-SoVITS
+python -m pip install -r requirements.txt
+.\go-simple-api.ps1
+```
+
+也可以运行:
+
+```bat
+go-simple-api.bat
+```
+
+默认后端地址:
+
+```text
+http://127.0.0.1:9881
+```
+
+## 2. 打开页面
+
+| 页面 | 地址 | 说明 |
+|------|------|------|
+| 测试前端 | http://127.0.0.1:9881/test/ | 带波形裁剪的测试 UI |
+| Swagger UI | http://127.0.0.1:9881/docs | 交互式 API 文档 |
+| ReDoc | http://127.0.0.1:9881/redoc | 可读式 API 文档 |
+
+## 3. 调用接口
+
+接口:
+
+```http
+POST /api/tts
+Content-Type: multipart/form-data
+```
+
+最小字段:
+
+```text
+text 要生成的文字
+ref_audio 3-10 秒主参考音频(或视频,前端自动提取)
+```
+
+常用可选字段:
+
+```text
+aux_ref_audio 辅助参考音频,可多个
+prompt_text 参考音频文本,可留空
+text_lang 默认 zh
+prompt_lang 默认 zh
+emotion neutral / happy / calm / sad / angry
+speed 语速,默认 1
+seed 默认 -1
+```
+
+## 4. PowerShell 示例
+
+```powershell
+curl.exe -X POST http://127.0.0.1:9881/api/tts `
+ -F "text=你好,欢迎使用这个声音。" `
+ -F "ref_audio=@D:\audio\ref.wav" `
+ -F "prompt_text=" `
+ -F "text_lang=zh" `
+ -F "prompt_lang=zh" `
+ --output output.wav
+```
+
+## 5. 注意事项
+
+- 主参考音频必须是 3-10 秒。
+- 前端支持上传视频文件,会自动提取音频。
+- 前端提供波形裁剪工具,可直接选择 3-10 秒片段。
+- `prompt_text` 在当前 v2 配置下可以为空。
+- 如果切换到 v3/v4,空 `prompt_text` 会返回 400。
+- 生成文字固定按标点符号切句。
+- 更详细的配置、profile、base64 接口见 `docs/simple_api.md`。
diff --git a/go-simple-api.bat b/go-simple-api.bat
new file mode 100644
index 00000000..2c2c7d62
--- /dev/null
+++ b/go-simple-api.bat
@@ -0,0 +1,12 @@
+@echo off
+set "SCRIPT_DIR=%~dp0"
+set "SCRIPT_DIR=%SCRIPT_DIR:~0,-1%"
+cd /d "%SCRIPT_DIR%"
+
+if exist "%SCRIPT_DIR%\runtime\python.exe" (
+ "%SCRIPT_DIR%\runtime\python.exe" "%SCRIPT_DIR%\simple_api.py" -c "%SCRIPT_DIR%\simple_api.yaml" %*
+) else (
+ python "%SCRIPT_DIR%\simple_api.py" -c "%SCRIPT_DIR%\simple_api.yaml" %*
+)
+
+pause
diff --git a/go-simple-api.ps1 b/go-simple-api.ps1
new file mode 100644
index 00000000..4d1336c5
--- /dev/null
+++ b/go-simple-api.ps1
@@ -0,0 +1,13 @@
+$ErrorActionPreference = "Stop"
+chcp 65001
+Set-Location $PSScriptRoot
+
+$runtimePython = Join-Path $PSScriptRoot "runtime\python.exe"
+$apiScript = Join-Path $PSScriptRoot "simple_api.py"
+$apiConfig = Join-Path $PSScriptRoot "simple_api.yaml"
+
+if (Test-Path $runtimePython) {
+ & $runtimePython $apiScript -c $apiConfig @args
+} else {
+ python $apiScript -c $apiConfig @args
+}
diff --git a/open-test-frontend.ps1 b/open-test-frontend.ps1
new file mode 100644
index 00000000..7e26d5a6
--- /dev/null
+++ b/open-test-frontend.ps1
@@ -0,0 +1,3 @@
+$ErrorActionPreference = "Stop"
+Set-Location $PSScriptRoot
+Start-Process (Join-Path $PSScriptRoot "test_frontend\index.html")
diff --git a/requirements.txt b/requirements.txt
index 00344838..9c468e65 100644
--- a/requirements.txt
+++ b/requirements.txt
@@ -3,6 +3,7 @@ numpy<2.0
scipy
tensorboard
librosa==0.10.2
+soundfile
numba
pytorch-lightning>=2.4
gradio<5
@@ -22,6 +23,7 @@ transformers>=4.43,<=4.50
peft<0.18.0
chardet
PyYAML
+python-multipart
psutil
jieba_fast
jieba
diff --git a/simple_api.py b/simple_api.py
new file mode 100644
index 00000000..eb8d635b
--- /dev/null
+++ b/simple_api.py
@@ -0,0 +1,787 @@
+"""
+Small profile-based API layer for GPT-SoVITS.
+
+Run:
+ python simple_api.py -c simple_api.yaml
+
+Then call:
+ POST http://127.0.0.1:9881/speak
+ {"text": "hello", "voice": "default"}
+"""
+
+from __future__ import annotations
+
+import argparse
+import base64
+import os
+import shutil
+import subprocess
+import sys
+import threading
+import traceback
+import uuid
+import wave
+from copy import deepcopy
+from io import BytesIO
+from pathlib import Path
+from typing import Any, Dict, Generator, List, Optional, Tuple, Union
+
+import numpy as np
+import soundfile as sf
+import yaml
+from fastapi import FastAPI, File, Form, HTTPException, Response, UploadFile
+from fastapi.middleware.cors import CORSMiddleware
+from fastapi.responses import JSONResponse, StreamingResponse
+from fastapi.staticfiles import StaticFiles
+from pydantic import BaseModel, Field
+
+PROJECT_ROOT = Path(__file__).resolve().parent
+os.chdir(PROJECT_ROOT)
+sys.path.append(str(PROJECT_ROOT))
+sys.path.append(str(PROJECT_ROOT / "GPT_SoVITS"))
+
+from GPT_SoVITS.TTS_infer_pack.TTS import TTS, TTS_Config # noqa: E402
+from GPT_SoVITS.TTS_infer_pack.text_segmentation_method import ( # noqa: E402
+ get_method_names as get_cut_method_names,
+)
+
+
+DEFAULT_TTS_PARAMS: Dict[str, Any] = {
+ "text": "",
+ "text_lang": "zh",
+ "ref_audio_path": "",
+ "aux_ref_audio_paths": [],
+ "prompt_text": "",
+ "prompt_lang": "zh",
+ "top_k": 15,
+ "top_p": 1.0,
+ "temperature": 1.0,
+ "text_split_method": "cut5",
+ "batch_size": 1,
+ "batch_threshold": 0.75,
+ "split_bucket": True,
+ "speed_factor": 1.0,
+ "fragment_interval": 0.3,
+ "seed": -1,
+ "media_type": "wav",
+ "streaming_mode": False,
+ "return_fragment": False,
+ "fixed_length_chunk": False,
+ "parallel_infer": True,
+ "repetition_penalty": 1.35,
+ "sample_steps": 32,
+ "super_sampling": False,
+ "overlap_length": 2,
+ "min_chunk_length": 16,
+}
+
+SUPPORTED_MEDIA_TYPES = {"wav", "raw", "ogg", "aac"}
+SUPPORTED_UPLOAD_EXTENSIONS = {".wav", ".flac", ".ogg", ".mp3", ".m4a", ".aac"}
+VOICE_META_KEYS = {"description"}
+
+
+class SpeakRequest(BaseModel):
+ text: str
+ voice: Optional[str] = None
+ text_lang: Optional[str] = None
+ ref_audio_path: Optional[str] = None
+ aux_ref_audio_paths: Optional[List[str]] = None
+ prompt_text: Optional[str] = None
+ prompt_lang: Optional[str] = None
+ format: Optional[str] = Field(default=None, description="wav, raw, ogg, or aac")
+ stream: Optional[bool] = Field(default=None, description="true maps to streaming mode 2")
+ streaming_mode: Optional[Union[bool, int]] = Field(default=None, description="0, 1, 2, 3, true, or false")
+ speed: Optional[float] = Field(default=None, description="Alias of speed_factor")
+ speed_factor: Optional[float] = None
+ top_k: Optional[int] = None
+ top_p: Optional[float] = None
+ temperature: Optional[float] = None
+ text_split_method: Optional[str] = None
+ batch_size: Optional[int] = None
+ batch_threshold: Optional[float] = None
+ split_bucket: Optional[bool] = None
+ fragment_interval: Optional[float] = None
+ seed: Optional[int] = None
+ parallel_infer: Optional[bool] = None
+ repetition_penalty: Optional[float] = None
+ sample_steps: Optional[int] = None
+ super_sampling: Optional[bool] = None
+ overlap_length: Optional[int] = None
+ min_chunk_length: Optional[int] = None
+
+
+class WeightsRequest(BaseModel):
+ gpt_weights_path: Optional[str] = None
+ sovits_weights_path: Optional[str] = None
+
+
+def parse_args() -> argparse.Namespace:
+ parser = argparse.ArgumentParser(description="GPT-SoVITS simple API")
+ parser.add_argument("-c", "--config", default="simple_api.yaml", help="simple API config path")
+ parser.add_argument("--tts-config", default=None, help="GPT-SoVITS tts_infer.yaml path")
+ parser.add_argument("-a", "--bind-addr", default=None, help="bind address")
+ parser.add_argument("-p", "--port", type=int, default=None, help="bind port")
+ return parser.parse_args()
+
+
+def load_yaml_config(config_path: Union[str, Path]) -> Dict[str, Any]:
+ path = Path(config_path)
+ if not path.is_absolute():
+ path = PROJECT_ROOT / path
+ if not path.exists():
+ raise FileNotFoundError(f"simple API config not found: {path}")
+ with path.open("r", encoding="utf-8") as f:
+ data = yaml.safe_load(f) or {}
+ if not isinstance(data, dict):
+ raise ValueError("simple API config must be a YAML object")
+ return data
+
+
+args = parse_args()
+simple_config = load_yaml_config(args.config)
+server_config = simple_config.get("server", {}) or {}
+host = args.bind_addr or server_config.get("host", "127.0.0.1")
+port = args.port or int(server_config.get("port", 9881))
+tts_config_path = args.tts_config or server_config.get("tts_config", "GPT_SoVITS/configs/tts_infer.yaml")
+
+cut_method_names = get_cut_method_names()
+tts_config: Optional[TTS_Config] = None
+tts_pipeline: Optional[TTS] = None
+infer_lock = threading.Lock()
+
+APP = FastAPI(
+ title="GPT-SoVITS Simple API",
+ description=(
+ "简化接口层,封装 GPT-SoVITS 推理引擎。\n\n"
+ "## 核心流程\n"
+ "1. 上传 3-10 秒参考音频(或视频,前端自动提取音频)\n"
+ "2. 填写需要生成的文字\n"
+ "3. 调用 `/api/tts` 获取生成的音频\n\n"
+ "## 其他接口\n"
+ "- `/speak` — 基于 voice profile 的调用方式\n"
+ "- `/v1/tts` — OpenAI 兼容格式\n"
+ "- `/admin/*` — 管理接口(热加载配置、切换模型)"
+ ),
+ version="1.1.0",
+ docs_url="/docs",
+ redoc_url="/redoc",
+ openapi_tags=[
+ {"name": "MVP", "description": "核心 TTS 接口,上传参考音频直接生成"},
+ {"name": "Profile", "description": "基于 voice profile 的调用方式"},
+ {"name": "Admin", "description": "管理接口:热加载配置、切换模型权重"},
+ {"name": "System", "description": "健康检查与信息查询"},
+ ],
+)
+APP.add_middleware(
+ CORSMiddleware,
+ allow_origins=simple_config.get("cors_allow_origins", ["*"]),
+ allow_credentials=False,
+ allow_methods=["*"],
+ allow_headers=["*"],
+)
+test_frontend_dir = PROJECT_ROOT / "test_frontend"
+if test_frontend_dir.exists():
+ APP.mount("/test", StaticFiles(directory=str(test_frontend_dir), html=True), name="test_frontend")
+
+
+def pack_ogg(io_buffer: BytesIO, data: np.ndarray, rate: int) -> BytesIO:
+ def handle_pack_ogg() -> None:
+ with sf.SoundFile(io_buffer, mode="w", samplerate=rate, channels=1, format="ogg") as audio_file:
+ audio_file.write(data)
+
+ try:
+ threading.stack_size(4096 * 4096)
+ pack_thread = threading.Thread(target=handle_pack_ogg)
+ pack_thread.start()
+ pack_thread.join()
+ except (RuntimeError, ValueError):
+ handle_pack_ogg()
+ return io_buffer
+
+
+def pack_raw(io_buffer: BytesIO, data: np.ndarray, rate: int) -> BytesIO:
+ del rate
+ io_buffer.write(data.tobytes())
+ return io_buffer
+
+
+def pack_wav(io_buffer: BytesIO, data: np.ndarray, rate: int) -> BytesIO:
+ del io_buffer
+ wav_buffer = BytesIO()
+ sf.write(wav_buffer, data, rate, format="wav")
+ return wav_buffer
+
+
+def pack_aac(io_buffer: BytesIO, data: np.ndarray, rate: int) -> BytesIO:
+ process = subprocess.Popen(
+ [
+ "ffmpeg",
+ "-f",
+ "s16le",
+ "-ar",
+ str(rate),
+ "-ac",
+ "1",
+ "-i",
+ "pipe:0",
+ "-c:a",
+ "aac",
+ "-b:a",
+ "192k",
+ "-vn",
+ "-f",
+ "adts",
+ "pipe:1",
+ ],
+ stdin=subprocess.PIPE,
+ stdout=subprocess.PIPE,
+ stderr=subprocess.PIPE,
+ )
+ out, _ = process.communicate(input=data.tobytes())
+ io_buffer.write(out)
+ return io_buffer
+
+
+def pack_audio(io_buffer: BytesIO, data: np.ndarray, rate: int, media_type: str) -> BytesIO:
+ if media_type == "ogg":
+ io_buffer = pack_ogg(io_buffer, data, rate)
+ elif media_type == "aac":
+ io_buffer = pack_aac(io_buffer, data, rate)
+ elif media_type == "wav":
+ io_buffer = pack_wav(io_buffer, data, rate)
+ else:
+ io_buffer = pack_raw(io_buffer, data, rate)
+ io_buffer.seek(0)
+ return io_buffer
+
+
+def wave_header_chunk(frame_input: bytes = b"", channels: int = 1, sample_width: int = 2, sample_rate: int = 32000) -> bytes:
+ wav_buf = BytesIO()
+ with wave.open(wav_buf, "wb") as vfout:
+ vfout.setnchannels(channels)
+ vfout.setsampwidth(sample_width)
+ vfout.setframerate(sample_rate)
+ vfout.writeframes(frame_input)
+ wav_buf.seek(0)
+ return wav_buf.read()
+
+
+def request_to_dict(request: BaseModel) -> Dict[str, Any]:
+ if hasattr(request, "model_dump"):
+ return request.model_dump(exclude_none=True)
+ return request.dict(exclude_none=True)
+
+
+def get_default_voice_name() -> Optional[str]:
+ default_voice = simple_config.get("default_voice")
+ if default_voice:
+ return str(default_voice)
+ voices = simple_config.get("voices", {}) or {}
+ if "default" in voices:
+ return "default"
+ return next(iter(voices.keys()), None)
+
+
+def resolve_project_path(path_value: Optional[str]) -> Optional[str]:
+ if path_value in [None, ""]:
+ return None
+ path = Path(str(path_value))
+ if not path.is_absolute():
+ path = PROJECT_ROOT / path
+ return str(path)
+
+
+def normalize_streaming(streaming_mode: Optional[Union[bool, int]], stream: Optional[bool]) -> Tuple[bool, bool, bool, bool]:
+ if streaming_mode is None:
+ streaming_mode = 2 if stream else 0
+ elif isinstance(streaming_mode, bool):
+ streaming_mode = 2 if streaming_mode else 0
+
+ if streaming_mode == 0:
+ return False, False, False, False
+ if streaming_mode == 1:
+ return False, True, False, True
+ if streaming_mode == 2:
+ return True, False, False, True
+ if streaming_mode == 3:
+ return True, False, True, True
+ raise HTTPException(status_code=400, detail="streaming_mode must be 0, 1, 2, 3, true, or false")
+
+
+def get_upload_config() -> Dict[str, Any]:
+ return simple_config.get("upload", {}) or {}
+
+
+def get_upload_root() -> Path:
+ upload_dir = str(get_upload_config().get("dir", "runtime/uploads"))
+ root = Path(upload_dir)
+ if not root.is_absolute():
+ root = PROJECT_ROOT / root
+ root.mkdir(parents=True, exist_ok=True)
+ return root
+
+
+def get_request_upload_dir() -> Path:
+ request_dir = get_upload_root() / uuid.uuid4().hex
+ request_dir.mkdir(parents=True, exist_ok=True)
+ return request_dir
+
+
+def get_upload_limits() -> Tuple[float, float, int]:
+ upload_config = get_upload_config()
+ min_seconds = float(upload_config.get("min_ref_seconds", 3.0))
+ max_seconds = float(upload_config.get("max_ref_seconds", 10.0))
+ max_upload_mb = int(upload_config.get("max_upload_mb", 80))
+ return min_seconds, max_seconds, max_upload_mb
+
+
+def get_upload_suffix(upload: UploadFile) -> str:
+ suffix = Path(upload.filename or "").suffix.lower()
+ if suffix:
+ return suffix
+ content_type = (upload.content_type or "").lower()
+ if content_type in {"audio/wav", "audio/x-wav", "audio/wave"}:
+ return ".wav"
+ if content_type == "audio/mpeg":
+ return ".mp3"
+ if content_type == "audio/ogg":
+ return ".ogg"
+ if content_type in {"audio/aac", "audio/aacp"}:
+ return ".aac"
+ return ".wav"
+
+
+def validate_audio_upload(upload: UploadFile) -> None:
+ suffix = get_upload_suffix(upload)
+ content_type = (upload.content_type or "").lower()
+ is_audio_type = content_type.startswith("audio/") or content_type in {"application/octet-stream", ""}
+ if suffix not in SUPPORTED_UPLOAD_EXTENSIONS or not is_audio_type:
+ raise HTTPException(status_code=400, detail=f"unsupported audio upload: {upload.filename or content_type}")
+
+
+async def save_upload_file(upload: UploadFile, target_dir: Path, name_prefix: str) -> str:
+ validate_audio_upload(upload)
+ _, _, max_upload_mb = get_upload_limits()
+ max_bytes = max_upload_mb * 1024 * 1024
+ suffix = get_upload_suffix(upload)
+ target_path = target_dir / f"{name_prefix}{suffix}"
+ total_size = 0
+
+ try:
+ with target_path.open("wb") as f:
+ while True:
+ chunk = await upload.read(1024 * 1024)
+ if not chunk:
+ break
+ total_size += len(chunk)
+ if total_size > max_bytes:
+ raise HTTPException(status_code=400, detail=f"audio upload exceeds {max_upload_mb} MB")
+ f.write(chunk)
+ if total_size == 0:
+ raise HTTPException(status_code=400, detail=f"audio upload is empty: {upload.filename or name_prefix}")
+ finally:
+ await upload.close()
+
+ return str(target_path)
+
+
+def get_audio_duration_with_ffprobe(audio_path: str) -> float:
+ result = subprocess.run(
+ [
+ "ffprobe",
+ "-v",
+ "error",
+ "-show_entries",
+ "format=duration",
+ "-of",
+ "default=noprint_wrappers=1:nokey=1",
+ audio_path,
+ ],
+ capture_output=True,
+ text=True,
+ timeout=15,
+ )
+ if result.returncode != 0:
+ raise RuntimeError(result.stderr.strip() or "ffprobe failed")
+ return float(result.stdout.strip())
+
+
+def get_audio_duration_seconds(audio_path: str) -> float:
+ try:
+ info = sf.info(audio_path)
+ if info.samplerate <= 0:
+ raise ValueError("invalid sample rate")
+ return float(info.frames) / float(info.samplerate)
+ except Exception as sf_exc:
+ try:
+ return get_audio_duration_with_ffprobe(audio_path)
+ except Exception as ffprobe_exc:
+ raise HTTPException(
+ status_code=400,
+ detail=f"unable to read audio duration: {audio_path}. soundfile={sf_exc}; ffprobe={ffprobe_exc}",
+ ) from ffprobe_exc
+
+
+def validate_main_ref_duration(audio_path: str) -> None:
+ min_seconds, max_seconds, _ = get_upload_limits()
+ duration = get_audio_duration_seconds(audio_path)
+ if duration < min_seconds or duration > max_seconds:
+ raise HTTPException(
+ status_code=400,
+ detail=f"ref_audio duration must be between {min_seconds:g}s and {max_seconds:g}s, got {duration:.2f}s",
+ )
+
+
+def apply_emotion_preset(payload: Dict[str, Any], emotion: Optional[str]) -> None:
+ if emotion in [None, ""]:
+ return
+ emotion_name = str(emotion).strip().lower()
+ emotion_presets = simple_config.get("emotion_presets", {}) or {}
+ if emotion_name not in emotion_presets:
+ supported = ", ".join(sorted(emotion_presets.keys())) or "none"
+ raise HTTPException(status_code=400, detail=f"emotion is not configured: {emotion_name}. supported: {supported}")
+ payload.update(emotion_presets[emotion_name] or {})
+
+
+def prompt_text_required_for_current_model() -> bool:
+ if tts_config is None:
+ return False
+ use_vocoder = getattr(tts_config, "use_vocoder", None)
+ if use_vocoder is not None:
+ return bool(use_vocoder)
+ return getattr(tts_config, "version", None) in {"v3", "v4"}
+
+
+def voice_public_info(name: str, profile: Dict[str, Any]) -> Dict[str, Any]:
+ ref_audio_path = resolve_project_path(profile.get("ref_audio_path"))
+ return {
+ "name": name,
+ "description": profile.get("description", ""),
+ "text_lang": profile.get("text_lang"),
+ "prompt_lang": profile.get("prompt_lang"),
+ "ref_audio_path": profile.get("ref_audio_path"),
+ "ready": bool(ref_audio_path and Path(ref_audio_path).exists() and profile.get("prompt_lang")),
+ }
+
+
+def build_tts_request(payload: Dict[str, Any]) -> Tuple[Dict[str, Any], str, bool]:
+ voices = simple_config.get("voices", {}) or {}
+ explicit_ref_audio = bool(payload.get("ref_audio_path"))
+ voice_name = payload.pop("voice", None)
+ if voice_name is None and not explicit_ref_audio:
+ voice_name = get_default_voice_name()
+ voice_profile: Dict[str, Any] = {}
+
+ if voice_name:
+ if voice_name not in voices:
+ raise HTTPException(status_code=404, detail=f"voice not found: {voice_name}")
+ voice_profile = deepcopy(voices[voice_name] or {})
+
+ tts_req = deepcopy(DEFAULT_TTS_PARAMS)
+ tts_req.update(simple_config.get("defaults", {}) or {})
+ tts_req.update({k: v for k, v in voice_profile.items() if k not in VOICE_META_KEYS})
+
+ stream = payload.pop("stream", None)
+ media_type = payload.pop("format", None)
+ speed = payload.pop("speed", None)
+ if media_type is not None:
+ tts_req["media_type"] = media_type
+
+ for key, value in payload.items():
+ if key in DEFAULT_TTS_PARAMS:
+ tts_req[key] = value
+ if speed is not None:
+ tts_req["speed_factor"] = speed
+
+ ref_audio_path = resolve_project_path(tts_req.get("ref_audio_path"))
+ if ref_audio_path:
+ tts_req["ref_audio_path"] = ref_audio_path
+
+ aux_paths = tts_req.get("aux_ref_audio_paths") or []
+ tts_req["aux_ref_audio_paths"] = [resolve_project_path(item) for item in aux_paths if item]
+
+ media_type = str(tts_req.get("media_type", "wav")).lower()
+ tts_req["media_type"] = media_type
+ if media_type not in SUPPORTED_MEDIA_TYPES:
+ raise HTTPException(status_code=400, detail=f"format is not supported: {media_type}")
+
+ text = str(tts_req.get("text") or "").strip()
+ if not text:
+ raise HTTPException(status_code=400, detail="text is required")
+ tts_req["text"] = text
+
+ if not tts_req.get("ref_audio_path"):
+ raise HTTPException(status_code=400, detail="ref_audio_path is required in voice profile or request")
+ if not Path(tts_req["ref_audio_path"]).exists():
+ raise HTTPException(status_code=400, detail=f"ref_audio_path does not exist: {tts_req['ref_audio_path']}")
+
+ for aux_path in tts_req["aux_ref_audio_paths"]:
+ if aux_path and not Path(aux_path).exists():
+ raise HTTPException(status_code=400, detail=f"aux_ref_audio_path does not exist: {aux_path}")
+
+ if tts_config is None:
+ raise HTTPException(status_code=503, detail="TTS pipeline is not ready")
+
+ text_lang = str(tts_req.get("text_lang") or "").lower()
+ prompt_lang = str(tts_req.get("prompt_lang") or "").lower()
+ tts_req["text_lang"] = text_lang
+ tts_req["prompt_lang"] = prompt_lang
+
+ if text_lang not in tts_config.languages:
+ raise HTTPException(status_code=400, detail=f"text_lang is not supported: {text_lang}")
+ if prompt_lang not in tts_config.languages:
+ raise HTTPException(status_code=400, detail=f"prompt_lang is not supported: {prompt_lang}")
+ if not str(tts_req.get("prompt_text") or "").strip() and prompt_text_required_for_current_model():
+ version = getattr(tts_config, "version", "current")
+ raise HTTPException(status_code=400, detail=f"prompt_text is required when using GPT-SoVITS {version}")
+
+ split_method = str(tts_req.get("text_split_method") or "cut5")
+ if split_method not in cut_method_names:
+ raise HTTPException(status_code=400, detail=f"text_split_method is not supported: {split_method}")
+
+ streaming_mode = tts_req.pop("streaming_mode", None)
+ streaming_enabled, return_fragment, fixed_length_chunk, response_stream = normalize_streaming(streaming_mode, stream)
+ tts_req["streaming_mode"] = streaming_enabled
+ tts_req["return_fragment"] = return_fragment
+ tts_req["fixed_length_chunk"] = fixed_length_chunk
+
+ return tts_req, media_type, response_stream
+
+
+def synthesize_once(payload: Dict[str, Any]) -> Tuple[bytes, str]:
+ tts_req, media_type, response_stream = build_tts_request(payload)
+ if response_stream:
+ raise HTTPException(status_code=400, detail="base64 output does not support streaming")
+
+ if tts_pipeline is None:
+ raise HTTPException(status_code=503, detail="TTS pipeline is not ready")
+
+ try:
+ with infer_lock:
+ sr, audio_data = next(tts_pipeline.run(tts_req))
+ audio_bytes = pack_audio(BytesIO(), audio_data, sr, media_type).getvalue()
+ return audio_bytes, media_type
+ except Exception as exc:
+ raise HTTPException(status_code=400, detail={"message": "tts failed", "exception": str(exc)}) from exc
+
+
+def synthesize_response(payload: Dict[str, Any]) -> Response:
+ tts_req, media_type, response_stream = build_tts_request(payload)
+ if tts_pipeline is None:
+ raise HTTPException(status_code=503, detail="TTS pipeline is not ready")
+
+ if response_stream:
+
+ def streaming_generator() -> Generator[bytes, None, None]:
+ first_chunk = True
+ chunk_media_type = media_type
+ with infer_lock:
+ for sr, chunk in tts_pipeline.run(tts_req):
+ if first_chunk and chunk_media_type == "wav":
+ yield wave_header_chunk(sample_rate=sr)
+ chunk_media_type = "raw"
+ first_chunk = False
+ yield pack_audio(BytesIO(), chunk, sr, chunk_media_type).getvalue()
+
+ return StreamingResponse(streaming_generator(), media_type=f"audio/{media_type}")
+
+ try:
+ with infer_lock:
+ sr, audio_data = next(tts_pipeline.run(tts_req))
+ audio_bytes = pack_audio(BytesIO(), audio_data, sr, media_type).getvalue()
+ return Response(audio_bytes, media_type=f"audio/{media_type}")
+ except Exception as exc:
+ return JSONResponse(status_code=400, content={"message": "tts failed", "exception": str(exc)})
+
+
+@APP.on_event("startup")
+def startup() -> None:
+ global tts_config, tts_pipeline
+ tts_config = TTS_Config(tts_config_path)
+ print(tts_config)
+ tts_pipeline = TTS(tts_config)
+
+
+@APP.get("/", tags=["System"])
+def index() -> Dict[str, Any]:
+ return {
+ "name": "GPT-SoVITS Simple API",
+ "version": "1.1.0",
+ "docs": "/docs",
+ "endpoints": {
+ "system": ["/health", "/voices"],
+ "mvp": ["/api/tts"],
+ "profile": ["/speak", "/speak/base64", "/v1/tts"],
+ "admin": ["/admin/reload-config", "/admin/weights"],
+ },
+ }
+
+
+@APP.get("/health", tags=["System"], summary="健康检查")
+def health() -> Dict[str, Any]:
+ import os
+
+ result: Dict[str, Any] = {
+ "status": "ok" if tts_pipeline is not None else "starting",
+ "tts_config": tts_config_path,
+ "version": getattr(tts_config, "version", None),
+ "languages": getattr(tts_config, "languages", []),
+ "pid": os.getpid(),
+ }
+ try:
+ import psutil
+ result["memory_mb"] = round(psutil.Process(os.getpid()).memory_info().rss / 1024 / 1024, 1)
+ except Exception:
+ pass
+ try:
+ import torch
+ if torch.cuda.is_available():
+ result["gpu"] = {
+ "name": torch.cuda.get_device_name(0),
+ "memory_used_mb": round(torch.cuda.memory_allocated(0) / 1024 / 1024, 1),
+ "memory_total_mb": round(torch.cuda.get_device_properties(0).total_mem / 1024 / 1024, 1),
+ }
+ except Exception:
+ pass
+ return result
+
+
+@APP.get("/voices", tags=["System"], summary="列出可用 voice profiles")
+def list_voices() -> Dict[str, Any]:
+ voices = simple_config.get("voices", {}) or {}
+ return {
+ "default_voice": get_default_voice_name(),
+ "voices": [voice_public_info(name, profile or {}) for name, profile in voices.items()],
+ }
+
+
+@APP.post(
+ "/api/tts",
+ tags=["MVP"],
+ summary="核心 TTS 接口",
+ description=(
+ "上传参考音频和需要生成的文字,返回生成的音频。\n\n"
+ "**主参考音频要求**:3-10 秒,支持 wav/flac/ogg/mp3/m4a/aac 格式。\n\n"
+ "**文字切句**:固定使用 `cut5`(按标点符号切句)。\n\n"
+ "**情绪预设**:neutral / happy / calm / sad / angry,本质是映射到采样和语速参数。"
+ ),
+)
+async def mvp_tts(
+ text: str = Form(...),
+ ref_audio: UploadFile = File(...),
+ aux_ref_audio: Optional[List[UploadFile]] = File(default=None),
+ prompt_text: str = Form(default=""),
+ text_lang: str = Form(default="zh"),
+ prompt_lang: str = Form(default="zh"),
+ format: str = Form(default="wav"),
+ emotion: Optional[str] = Form(default=None),
+ speed: Optional[float] = Form(default=None),
+ seed: int = Form(default=-1),
+) -> Response:
+ request_dir = get_request_upload_dir()
+ try:
+ ref_audio_path = await save_upload_file(ref_audio, request_dir, "ref")
+ validate_main_ref_duration(ref_audio_path)
+
+ aux_ref_audio_paths = []
+ for index, upload in enumerate(aux_ref_audio or []):
+ aux_path = await save_upload_file(upload, request_dir, f"aux_{index}")
+ get_audio_duration_seconds(aux_path)
+ aux_ref_audio_paths.append(aux_path)
+
+ payload: Dict[str, Any] = {
+ "text": text,
+ "text_lang": text_lang,
+ "ref_audio_path": ref_audio_path,
+ "aux_ref_audio_paths": aux_ref_audio_paths,
+ "prompt_text": prompt_text or "",
+ "prompt_lang": prompt_lang,
+ "format": format,
+ "text_split_method": "cut5",
+ "streaming_mode": 0,
+ "seed": seed,
+ }
+ apply_emotion_preset(payload, emotion)
+ payload["text_split_method"] = "cut5"
+ payload["streaming_mode"] = 0
+ if speed is not None:
+ payload["speed"] = speed
+
+ return synthesize_response(payload)
+ finally:
+ shutil.rmtree(request_dir, ignore_errors=True)
+
+
+@APP.get("/speak", tags=["Profile"], summary="GET 方式调用 voice profile TTS")
+def speak_get(
+ text: str,
+ voice: Optional[str] = None,
+ text_lang: Optional[str] = None,
+ format: Optional[str] = None,
+ stream: Optional[bool] = None,
+ speed: Optional[float] = None,
+) -> Response:
+ payload = {
+ "text": text,
+ "voice": voice,
+ "text_lang": text_lang,
+ "format": format,
+ "stream": stream,
+ "speed": speed,
+ }
+ return synthesize_response({k: v for k, v in payload.items() if v is not None})
+
+
+@APP.post("/speak", tags=["Profile"], summary="POST 方式调用 voice profile TTS")
+def speak_post(request: SpeakRequest) -> Response:
+ return synthesize_response(request_to_dict(request))
+
+
+@APP.post("/v1/tts", tags=["Profile"], summary="OpenAI 兼容格式 TTS")
+def openai_style_tts(request: SpeakRequest) -> Response:
+ return synthesize_response(request_to_dict(request))
+
+
+@APP.post("/speak/base64", tags=["Profile"], summary="返回 Base64 编码的音频")
+def speak_base64(request: SpeakRequest) -> Dict[str, Any]:
+ audio_bytes, media_type = synthesize_once(request_to_dict(request))
+ return {
+ "media_type": f"audio/{media_type}",
+ "audio_base64": base64.b64encode(audio_bytes).decode("ascii"),
+ }
+
+
+@APP.post("/admin/reload-config", tags=["Admin"], summary="热加载 simple_api.yaml 配置")
+def reload_config() -> Dict[str, Any]:
+ global simple_config
+ simple_config = load_yaml_config(args.config)
+ return {"message": "success", "default_voice": get_default_voice_name()}
+
+
+@APP.post("/admin/weights", tags=["Admin"], summary="运行时切换 GPT-SoVITS 模型权重")
+def set_weights(request: WeightsRequest) -> Dict[str, Any]:
+ if tts_pipeline is None:
+ raise HTTPException(status_code=503, detail="TTS pipeline is not ready")
+ if not request.gpt_weights_path and not request.sovits_weights_path:
+ raise HTTPException(status_code=400, detail="gpt_weights_path or sovits_weights_path is required")
+
+ try:
+ with infer_lock:
+ if request.gpt_weights_path:
+ tts_pipeline.init_t2s_weights(request.gpt_weights_path)
+ if request.sovits_weights_path:
+ tts_pipeline.init_vits_weights(request.sovits_weights_path)
+ except Exception as exc:
+ raise HTTPException(status_code=400, detail={"message": "change weights failed", "exception": str(exc)}) from exc
+
+ return {"message": "success"}
+
+
+if __name__ == "__main__":
+ import uvicorn
+
+ try:
+ uvicorn.run(app=APP, host=None if host == "None" else host, port=port, workers=1)
+ except Exception:
+ traceback.print_exc()
+ sys.exit(1)
diff --git a/simple_api.yaml b/simple_api.yaml
new file mode 100644
index 00000000..4c02394f
--- /dev/null
+++ b/simple_api.yaml
@@ -0,0 +1,59 @@
+server:
+ host: 127.0.0.1
+ port: 9881
+ tts_config: GPT_SoVITS/configs/tts_infer.yaml
+
+cors_allow_origins:
+ - "*"
+
+upload:
+ dir: runtime/uploads
+ min_ref_seconds: 3
+ max_ref_seconds: 10
+ max_upload_mb: 80
+
+default_voice: default
+
+defaults:
+ text_lang: zh
+ prompt_lang: zh
+ media_type: wav
+ text_split_method: cut5
+ batch_size: 1
+ batch_threshold: 0.75
+ split_bucket: true
+ speed_factor: 1.0
+ fragment_interval: 0.3
+ seed: -1
+ parallel_infer: true
+ repetition_penalty: 1.35
+ sample_steps: 32
+ super_sampling: false
+ overlap_length: 2
+ min_chunk_length: 16
+
+emotion_presets:
+ neutral: {}
+ happy:
+ temperature: 1.1
+ top_p: 0.95
+ calm:
+ temperature: 0.8
+ top_p: 0.85
+ speed_factor: 0.92
+ sad:
+ temperature: 0.75
+ top_p: 0.85
+ speed_factor: 0.9
+ angry:
+ temperature: 1.2
+ top_k: 20
+ repetition_penalty: 1.25
+
+voices:
+ default:
+ description: Replace this profile with your reference voice.
+ ref_audio_path: reference.wav
+ prompt_text: Replace this with the exact text spoken in reference.wav.
+ prompt_lang: zh
+ text_lang: zh
diff --git a/test_frontend/index.html b/test_frontend/index.html
new file mode 100644
index 00000000..ed441135
--- /dev/null
+++ b/test_frontend/index.html
@@ -0,0 +1,1006 @@
+
+
+
选择 3-10 秒参考音频或视频(视频会自动提取音频),在波形图上拖拽选取裁剪区域,填写后端接口地址和生成文本,直接调用中间层 /api/tts。