diff --git a/main/xiaozhi-server/config.yaml b/main/xiaozhi-server/config.yaml
index 57f35603..fb8189bb 100644
--- a/main/xiaozhi-server/config.yaml
+++ b/main/xiaozhi-server/config.yaml
@@ -710,6 +710,22 @@ TTS:
     #     voice_id: female-shaonv
     #     weight: 1
     # language_boost: auto
+  MinimaxTTSHTTPStream:
+  # Minimax流式语音合成服务
+    type: minimax_httpstream
+    output_dir: tmp/
+    group_id: 你的minimax平台groupID
+    api_key: 你的minimax平台接口密钥
+    model: "speech-01-turbo"
+    voice_id: "female-shaonv"
+
+  MinimaxTTSWebSocketStream:
+    type: minimax_webSocket
+    output_dir: tmp/
+    group_id: 你的minimax平台groupID
+    api_key: 你的minimax平台接口密钥
+    model: "speech-01-turbo"
+    voice_id: "female-shaonv"
   AliyunTTS:
     # 阿里云智能语音交互服务，需要先在阿里云平台开通服务，然后获取验证信息
     # 平台地址：https://nls-portal.console.aliyun.com/
diff --git a/main/xiaozhi-server/core/providers/tts/minimax_httpstream.py b/main/xiaozhi-server/core/providers/tts/minimax_httpstream.py
new file mode 100644
index 00000000..605b3efd
--- /dev/null
+++ b/main/xiaozhi-server/core/providers/tts/minimax_httpstream.py
@@ -0,0 +1,230 @@
+import os
+import uuid
+import json
+import requests
+from datetime import datetime
+from typing import Iterator, Optional, Union
+from core.providers.tts.base import TTSProviderBase
+from core.utils.util import parse_string_to_list
+
+
+class TTSProvider(TTSProviderBase):
+    def __init__(self, config, delete_audio_file):
+        super().__init__(config, delete_audio_file)
+        self.group_id = config.get("group_id")
+        self.api_key = config.get("api_key")
+        self.model = config.get("model")
+        if config.get("private_voice"):
+            self.voice = config.get("private_voice")
+        else:
+            self.voice = config.get("voice_id")
+
+        default_voice_setting = {
+            "voice_id": "female-shaonv",
+            "speed": 1,
+            "vol": 1,
+            "pitch": 0,
+            "emotion": "happy",
+        }
+        default_pronunciation_dict = {"tone": ["处理/(chu3)(li3)", "危险/dangerous"]}
+        defult_audio_setting = {
+            "sample_rate": 32000,
+            "bitrate": 128000,
+            "format": "mp3",
+            "channel": 1,
+        }
+        self.voice_setting = {
+            **default_voice_setting,
+            **config.get("voice_setting", {}),
+        }
+        self.pronunciation_dict = {
+            **default_pronunciation_dict,
+            **config.get("pronunciation_dict", {}),
+        }
+        self.audio_setting = {**defult_audio_setting, **config.get("audio_setting", {})}
+        self.timber_weights = parse_string_to_list(config.get("timber_weights"))
+
+        if self.voice:
+            self.voice_setting["voice_id"] = self.voice
+
+        self.host = "api.minimax.chat"
+        self.api_url = f"https://{self.host}/v1/t2a_v2?GroupId={self.group_id}"
+        self.header = {
+            "Content-Type": "application/json",
+            "Authorization": f"Bearer {self.api_key}",
+        }
+        self.audio_file_type = defult_audio_setting.get("format", "mp3")
+
+    def generate_filename(self, extension=".mp3"):
+        return os.path.join(
+            self.output_file,
+            f"tts-{__name__}{datetime.now().date()}@{uuid.uuid4().hex}{extension}",
+        )
+
+    async def text_to_speak(self, text, output_file):
+        """非流式语音合成（保留原有实现）"""
+        request_json = {
+            "model": self.model,
+            "text": text,
+            "stream": False,
+            "voice_setting": self.voice_setting,
+            "pronunciation_dict": self.pronunciation_dict,
+            "audio_setting": self.audio_setting,
+        }
+
+        if type(self.timber_weights) is list and len(self.timber_weights) > 0:
+            request_json["timber_weights"] = self.timber_weights
+            request_json["voice_setting"]["voice_id"] = ""
+
+        try:
+            resp = requests.post(
+                self.api_url, json.dumps(request_json), headers=self.header
+            )
+            if resp.json()["base_resp"]["status_code"] == 0:
+                data = resp.json()["data"]["audio"]
+                audio_bytes = bytes.fromhex(data)
+                if output_file:
+                    with open(output_file, "wb") as file_to_save:
+                        file_to_save.write(audio_bytes)
+                else:
+                    return audio_bytes
+            else:
+                raise Exception(
+                    f"{__name__} status_code: {resp.status_code} response: {resp.content}"
+                )
+        except Exception as e:
+            raise Exception(f"{__name__} error: {e}")
+
+    def text_to_speak_stream(
+        self, 
+        text: str, 
+        chunk_callback: Optional[callable] = None
+    ) -> Iterator[bytes]:
+        """
+        流式语音合成方法
+        :param text: 要合成的文本
+        :param chunk_callback: 可选的回调函数，用于处理每个音频块
+        :return: 生成器，每次产生一个音频数据块(bytes)
+        """
+        request_json = {
+            "model": self.model,
+            "text": text,
+            "stream": True,
+            "voice_setting": self.voice_setting,
+            "pronunciation_dict": self.pronunciation_dict,
+            "audio_setting": self.audio_setting,
+        }
+
+        if isinstance(self.timber_weights, list) and len(self.timber_weights) > 0:
+            request_json["timber_weights"] = self.timber_weights
+            request_json["voice_setting"]["voice_id"] = ""
+
+        try:
+            with requests.post(
+                self.api_url,
+                data=json.dumps(request_json),
+                headers=self.header,
+                stream=True
+            ) as response:
+                
+                # 检查HTTP状态码
+                if response.status_code != 200:
+                    raise Exception(
+                        f"HTTP error: {response.status_code}, response: {response.text}"
+                    )
+                
+                # 处理流式响应
+                for line in response.iter_lines():
+                    if line:  # 过滤空行
+                        # 检查是否为数据行 (SSE格式)
+                        if line.startswith(b'data:'):
+                            try:
+                                data = json.loads(line[5:].strip())  # 去掉"data:"前缀
+                                
+                                # 检查API状态码
+                                if data.get("base_resp", {}).get("status_code", -1) != 0:
+                                    raise Exception(
+                                        f"API error: {data.get('base_resp', {}).get('status_msg')}"
+                                    )
+                                
+                                # 跳过非音频数据块
+                                if "extra_info" in data:
+                                    continue
+                                    
+                                # 提取音频数据
+                                audio_hex = data.get("data", {}).get("audio")
+                                if audio_hex:
+                                    audio_chunk = bytes.fromhex(audio_hex)
+                                    if chunk_callback:
+                                        chunk_callback(audio_chunk)
+                                    yield audio_chunk
+                                    
+                            except json.JSONDecodeError:
+                                # 忽略JSON解析错误（可能是心跳包等）
+                                continue
+                            except Exception as e:
+                                raise e
+                                
+        except Exception as e:
+            raise Exception(f"{__name__} stream error: {e}")
+
+    def save_stream_to_file(
+        self, 
+        text: str, 
+        output_file: Optional[str] = None,
+        progress_callback: Optional[callable] = None
+    ) -> str:
+        """
+        流式合成并保存到文件
+        :param text: 要合成的文本
+        :param output_file: 输出文件路径，如果为None则自动生成
+        :param progress_callback: 可选的回调函数，接收已写入的字节数
+        :return: 保存的文件路径
+        """
+        if not output_file:
+            output_file = self.generate_filename(extension=f".{self.audio_file_type}")
+        
+        os.makedirs(os.path.dirname(output_file), exist_ok=True)
+        
+        total_bytes = 0
+        try:
+            with open(output_file, "wb") as audio_file:
+                for audio_chunk in self.text_to_speak_stream(text):
+                    audio_file.write(audio_chunk)
+                    audio_file.flush()
+                    total_bytes += len(audio_chunk)
+                    if progress_callback:
+                        progress_callback(total_bytes)
+            return output_file
+        except Exception as e:
+            # 清理可能创建的不完整文件
+            if os.path.exists(output_file):
+                os.remove(output_file)
+            raise e
+
+    def stream_to_audio_player(self, text: str, player_command: list = None):
+        """
+        流式合成并直接播放音频
+        :param text: 要合成的文本
+        :param player_command: 音频播放器命令，默认使用mpv
+        """
+        if player_command is None:
+            player_command = ["mpv", "--no-cache", "--no-terminal", "--", "fd://0"]
+        
+        try:
+            import subprocess
+            player_process = subprocess.Popen(
+                player_command,
+                stdin=subprocess.PIPE,
+                stdout=subprocess.DEVNULL,
+                stderr=subprocess.DEVNULL,
+            )
+            
+            for audio_chunk in self.text_to_speak_stream(text):
+                player_process.stdin.write(audio_chunk)
+                player_process.stdin.flush()
+                
+            player_process.stdin.close()
+            player_process.wait()
+        except Exception as e:
+            raise Exception(f"Audio player error: {e}")
\ No newline at end of file
diff --git a/main/xiaozhi-server/core/providers/tts/minimax_webSocket.py b/main/xiaozhi-server/core/providers/tts/minimax_webSocket.py
new file mode 100644
index 00000000..bda6b47f
--- /dev/null
+++ b/main/xiaozhi-server/core/providers/tts/minimax_webSocket.py
@@ -0,0 +1,180 @@
+import os
+import uuid
+import json
+import asyncio
+import websockets
+import ssl
+from datetime import datetime
+from core.providers.tts.base import TTSProviderBase
+from core.utils.util import parse_string_to_list
+
+
+class TTSProvider(TTSProviderBase):
+    def __init__(self, config, delete_audio_file):
+        super().__init__(config, delete_audio_file)
+        self.group_id = config.get("group_id")
+        self.api_key = config.get("api_key")
+        self.model = config.get("model")
+        
+        # 初始化语音设置
+        default_voice_setting = {
+            "voice_id": "female-shaonv",
+            "speed": 1,
+            "vol": 1,
+            "pitch": 0,
+            "emotion": "happy",
+        }
+        default_pronunciation_dict = {"tone": ["处理/(chu3)(li3)", "危险/dangerous"]}
+        default_audio_setting = {
+            "sample_rate": 32000,
+            "bitrate": 128000,
+            "format": "mp3",
+            "channel": 1,
+        }
+        
+        # 合并配置
+        self.voice_setting = {
+            **default_voice_setting,
+            **config.get("voice_setting", {}),
+        }
+        self.pronunciation_dict = {
+            **default_pronunciation_dict,
+            **config.get("pronunciation_dict", {}),
+        }
+        self.audio_setting = {
+            **default_audio_setting, 
+            **config.get("audio_setting", {})
+        }
+        self.timber_weights = parse_string_to_list(config.get("timber_weights"))
+        
+        # 设置语音ID
+        if config.get("private_voice"):
+            self.voice_setting["voice_id"] = config.get("private_voice")
+        elif config.get("voice_id"):
+            self.voice_setting["voice_id"] = config.get("voice_id")
+        
+        # WebSocket配置
+        self.ws_url = "wss://api.minimaxi.com/ws/v1/t2a_v2"
+        self.headers = {
+            "Authorization": f"Bearer {self.api_key}",
+            "GroupId": self.group_id
+        }
+        self.audio_file_type = self.audio_setting.get("format", "mp3")
+
+    def generate_filename(self, extension=".mp3"):
+        """生成唯一的音频文件名"""
+        return os.path.join(
+            self.output_file,
+            f"tts-{__name__}{datetime.now().date()}@{uuid.uuid4().hex}{extension}",
+        )
+
+    async def _establish_connection(self):
+        """建立WebSocket连接"""
+        ssl_context = ssl.create_default_context()
+        ssl_context.check_hostname = False
+        ssl_context.verify_mode = ssl.CERT_NONE
+
+        try:
+            ws = await websockets.connect(
+                self.ws_url,
+                additional_headers=self.headers,
+                ssl=ssl_context
+            )
+            connected = json.loads(await ws.recv())
+            if connected.get("event") == "connected_success":
+                print("连接成功")
+                return ws
+            return None
+        except Exception as e:
+            print(f"连接失败: {e}")
+            return None
+
+    async def _start_task(self, websocket):
+        """发送任务开始请求"""
+        start_msg = {
+            "event": "task_start",
+            "model": self.model,
+            "voice_setting": self.voice_setting,
+            "pronunciation_dict": self.pronunciation_dict,
+            "audio_setting": self.audio_setting
+        }
+        
+        if self.timber_weights and len(self.timber_weights) > 0:
+            start_msg["timber_weights"] = self.timber_weights
+            start_msg["voice_setting"]["voice_id"] = ""
+
+        await websocket.send(json.dumps(start_msg))
+        response = json.loads(await websocket.recv())
+        return response.get("event") == "task_started"
+
+    async def _continue_task(self, websocket, text):
+        """发送继续请求并收集音频数据"""
+        await websocket.send(json.dumps({
+            "event": "task_continue",
+            "text": text
+        }))
+
+        audio_chunks = []
+        while True:
+            response = json.loads(await websocket.recv())
+            if "data" in response and "audio" in response["data"]:
+                audio_chunks.append(response["data"]["audio"])
+            if response.get("is_final"):
+                break
+        return "".join(audio_chunks)
+
+    async def _close_connection(self, websocket):
+        """关闭连接"""
+        if websocket:
+            await websocket.send(json.dumps({"event": "task_finish"}))
+            await websocket.close()
+            print("连接已关闭")
+
+    async def text_to_speak(self, text, output_file=None):
+        """主方法：文本转语音"""
+        ws = await self._establish_connection()
+        if not ws:
+            raise Exception("无法建立WebSocket连接")
+
+        try:
+            if not await self._start_task(ws):
+                raise Exception("任务启动失败")
+
+            hex_audio = await self._continue_task(ws, text)
+            audio_bytes = bytes.fromhex(hex_audio)
+
+            # 保存到文件或返回二进制数据
+            if output_file:
+                with open(output_file, "wb") as f:
+                    f.write(audio_bytes)
+                print(f"音频已保存为{output_file}")
+                return output_file
+            else:
+                # 返回音频二进制数据（不播放）
+                return audio_bytes
+
+        finally:
+            await self._close_connection(ws)
+
+
+async def main():
+    """测试用主函数"""
+    # 示例配置
+    config = {
+        "group_id": "YOUR_GROUP_ID",  # 替换为实际的group_id
+        "api_key": "YOUR_API_KEY",    # 替换为实际的api_key
+        "model": "your-model",        # 替换为实际的模型名称
+        "voice_id": "male-qn-qingse",
+        "voice_setting": {
+            "speed": 1.2,
+            "emotion": "happy"
+        }
+    }
+    
+    tts = TTSProvider(config, delete_audio_file=True)
+    output_file = tts.generate_filename()
+    await tts.text_to_speak("这是一个测试文本，用于验证流式语音合成功能", output_file)
+
+
+if __name__ == "__main__":
+    asyncio.run(main())
\ No newline at end of file