Merge remote-tracking branch 'origin/main' into py_test

2026-07-29 12:23:55 +08:00 · 2025-09-05 14:56:08 +08:00
parent c82cf670de d04ec9d510
commit 257e365299
51 changed files with 1641 additions and 507 deletions
@@ -1,14 +1,14 @@
 import os
+import io
 import wave
 import uuid
+import json
+import time
 import queue
 import asyncio
 import traceback
 import threading
 import opuslib_next
-import json
-import io
-import time
 import concurrent.futures
 from abc import ABC, abstractmethod
 from config.logger import setup_logging
@@ -87,11 +87,9 @@ class ASRProviderBase(ABC):
            
            # 预先准备WAV数据
            wav_data = None
-            # 使用连接的声纹识别提供者
            if conn.voiceprint_provider and combined_pcm_data:
                wav_data = self._pcm_to_wav(combined_pcm_data)
            
-            
            # 定义ASR任务
            def run_asr():
                start_time = time.monotonic()
@@ -149,7 +147,7 @@ class ASRProviderBase(ABC):
            
            
            # 处理结果
-            raw_text, file_path = results.get("asr", ("", None))
+            raw_text, _ = results.get("asr", ("", None))
            speaker_name = results.get("voiceprint", None)
            
            # 记录识别结果
@@ -1,8 +1,10 @@
 from config.logger import setup_logging
 from http import HTTPStatus
+import dashscope
 from dashscope import Application
 from core.providers.llm.base import LLMProviderBase
 from core.utils.util import check_model_key
+import time

 TAG = __name__
 logger = setup_logging()
@@ -15,6 +17,7 @@ class LLMProvider(LLMProviderBase):
        self.base_url = config.get("base_url")
        self.is_No_prompt = config.get("is_no_prompt")
        self.memory_id = config.get("ali_memory_id")
+        self.streaming_chunk_size = config.get("streaming_chunk_size", 3)  # 每次流式返回的字符数
        check_model_key("AliBLLLM", self.api_key)

    def response(self, session_id, dialogue):
@@ -32,6 +35,8 @@ class LLMProvider(LLMProviderBase):
                "app_id": self.app_id,
                "session_id": session_id,
                "messages": dialogue,
+                # 开启SDK原生流式
+                "stream": True,
            }
            if self.memory_id != False:
                # 百练memory需要prompt参数
@@ -42,25 +47,63 @@ class LLMProvider(LLMProviderBase):
                    f"【阿里百练API服务】处理后的prompt: {prompt}"
                )

+            # 可选地设置自定义API基地址（若配置为兼容模式URL则忽略）
+            if self.base_url and ("/api/" in self.base_url):
+                dashscope.base_http_api_url = self.base_url
+
            responses = Application.call(**call_params)
-            if responses.status_code != HTTPStatus.OK:
-                logger.bind(tag=TAG).error(
-                    f"code={responses.status_code}, "
-                    f"message={responses.message}, "
-                    f"请参考文档：https://help.aliyun.com/zh/model-studio/developer-reference/error-code"
-                )
-                yield "【阿里百练API服务响应异常】"
-            else:
-                logger.bind(tag=TAG).debug(
-                    f"【阿里百练API服务】构造参数: {call_params}"
-                )
-                yield responses.output.text
+
+            # 流式处理（SDK在stream=True时返回可迭代对象；否则返回单次响应对象）
+            logger.bind(tag=TAG).debug(
+                f"【阿里百练API服务】构造参数: {dict(call_params, api_key='***')}"
+            )
+
+            last_text = ""
+            try:
+                for resp in responses:
+                    if resp.status_code != HTTPStatus.OK:
+                        logger.bind(tag=TAG).error(
+                            f"code={resp.status_code}, message={resp.message}, 请参考文档：https://help.aliyun.com/zh/model-studio/developer-reference/error-code"
+                        )
+                        continue
+                    current_text = getattr(getattr(resp, "output", None), "text", None)
+                    if current_text is None:
+                        continue
+                    # SDK流式为增量覆盖，计算差量输出
+                    if len(current_text) >= len(last_text):
+                        delta = current_text[len(last_text):]
+                    else:
+                        # 避免偶发回退
+                        delta = current_text
+                    if delta:
+                        yield delta
+                    last_text = current_text
+            except TypeError:
+                # 非流式回落（一次性返回）
+                if responses.status_code != HTTPStatus.OK:
+                    logger.bind(tag=TAG).error(
+                        f"code={responses.status_code}, message={responses.message}, 请参考文档：https://help.aliyun.com/zh/model-studio/developer-reference/error-code"
+                    )
+                    yield "【阿里百练API服务响应异常】"
+                else:
+                    full_text = getattr(getattr(responses, "output", None), "text", "")
+                    logger.bind(tag=TAG).info(
+                        f"【阿里百练API服务】完整响应长度: {len(full_text)}"
+                    )
+                    for i in range(0, len(full_text), self.streaming_chunk_size):
+                        chunk = full_text[i:i + self.streaming_chunk_size]
+                        if chunk:
+                            yield chunk

        except Exception as e:
            logger.bind(tag=TAG).error(f"【阿里百练API服务】响应异常: {e}")
            yield "【LLM服务响应异常】"

    def response_with_functions(self, session_id, dialogue, functions=None):
-        logger.bind(tag=TAG).error(
-            f"阿里百练暂未实现完整的工具调用（function call），建议使用其他意图识别"
+        # 阿里百练当前未支持原生的 function call。为保持兼容，这里回退到普通文本流式输出。
+        # 上层会按 (content, tool_calls) 的形式消费，这里始终返回 (token, None)
+        logger.bind(tag=TAG).warning(
+            "阿里百练未实现原生 function call，已回退为纯文本流式输出"
        )
+        for token in self.response(session_id, dialogue):
+            yield token, None
@@ -18,8 +18,8 @@ class ServerMCPExecutor(ToolExecutor):
        """初始化MCP管理器"""
        if not self._initialized:
            self.mcp_manager = ServerMCPManager(self.conn)
-            await self.mcp_manager.initialize_servers()
            self._initialized = True
+            await self.mcp_manager.initialize_servers()

    async def execute(
        self, conn, tool_name: str, arguments: Dict[str, Any]
@@ -68,6 +68,9 @@ class ServerMCPManager:

        # 输出当前支持的服务端MCP工具列表
        if hasattr(self.conn, "func_handler") and self.conn.func_handler:
+            # 刷新工具缓存以确保服务端MCP工具被正确加载
+            if hasattr(self.conn.func_handler, "tool_manager"):
+                self.conn.func_handler.tool_manager.refresh_tools()
            self.conn.func_handler.current_support_functions()

    def get_all_tools(self) -> List[Dict[str, Any]]:
@@ -478,3 +478,142 @@ class TTSProvider(TTSProviderBase):
        finally:
            self._monitor_task = None

+    def to_tts(self, text: str) -> list:
+        """非流式TTS处理，用于测试及保存音频文件的场景"""
+        try:
+            # 创建新的事件循环
+            loop = asyncio.new_event_loop()
+            asyncio.set_event_loop(loop)
+
+            # 生成会话ID
+            session_id = uuid.uuid4().hex
+            # 存储音频数据
+            audio_data = []
+
+            async def _generate_audio():
+                # 刷新Token（如果需要）
+                if self._is_token_expired():
+                    self._refresh_token()
+
+                # 建立WebSocket连接
+                ws = await websockets.connect(
+                    self.ws_url,
+                    additional_headers={"X-NLS-Token": self.token},
+                    ping_interval=30,
+                    ping_timeout=10,
+                    close_timeout=10,
+                )
+                try:
+                    # 发送StartSynthesis请求
+                    start_message_id = str(uuid.uuid4().hex)
+                    start_request = {
+                        "header": {
+                            "message_id": start_message_id,
+                            "task_id": session_id,
+                            "namespace": "FlowingSpeechSynthesizer",
+                            "name": "StartSynthesis",
+                            "appkey": self.appkey,
+                        },
+                        "payload": {
+                            "voice": self.voice,
+                            "format": self.format,
+                            "sample_rate": self.sample_rate,
+                            "volume": self.volume,
+                            "speech_rate": self.speech_rate,
+                            "pitch_rate": self.pitch_rate,
+                            "enable_subtitle": True,
+                        },
+                    }
+                    await ws.send(json.dumps(start_request))
+
+                    # 等待SynthesisStarted响应
+                    synthesis_started = False
+                    while not synthesis_started:
+                        msg = await ws.recv()
+                        if isinstance(msg, str):
+                            data = json.loads(msg)
+                            header = data.get("header", {})
+                            if header.get("name") == "SynthesisStarted":
+                                synthesis_started = True
+                                logger.bind(tag=TAG).debug("TTS合成已启动")
+                            elif header.get("name") == "TaskFailed":
+                                error_info = data.get("payload", {}).get(
+                                    "error_info", {}
+                                )
+                                error_code = error_info.get("error_code")
+                                error_message = error_info.get(
+                                    "error_message", "未知错误"
+                                )
+                                raise Exception(
+                                    f"启动合成失败: {error_code} - {error_message}"
+                                )
+
+                    # 发送文本合成请求
+                    filtered_text = MarkdownCleaner.clean_markdown(text)
+                    run_message_id = str(uuid.uuid4().hex)
+                    run_request = {
+                        "header": {
+                            "message_id": run_message_id,
+                            "task_id": session_id,
+                            "namespace": "FlowingSpeechSynthesizer",
+                            "name": "RunSynthesis",
+                            "appkey": self.appkey,
+                        },
+                        "payload": {"text": filtered_text},
+                    }
+                    await ws.send(json.dumps(run_request))
+
+                    # 发送停止合成请求
+                    stop_message_id = str(uuid.uuid4().hex)
+                    stop_request = {
+                        "header": {
+                            "message_id": stop_message_id,
+                            "task_id": session_id,
+                            "namespace": "FlowingSpeechSynthesizer",
+                            "name": "StopSynthesis",
+                            "appkey": self.appkey,
+                        }
+                    }
+                    await ws.send(json.dumps(stop_request))
+
+                    # 接收音频数据
+                    synthesis_completed = False
+                    while not synthesis_completed:
+                        msg = await ws.recv()
+                        if isinstance(msg, (bytes, bytearray)):
+                            self.opus_encoder.encode_pcm_to_opus_stream(
+                                msg,
+                                end_of_stream=False,
+                                callback=lambda opus: audio_data.append(opus)
+                            )
+                        elif isinstance(msg, str):
+                            data = json.loads(msg)
+                            header = data.get("header", {})
+                            event_name = header.get("name")
+                            if event_name == "SynthesisCompleted":
+                                synthesis_completed = True
+                                logger.bind(tag=TAG).debug("TTS合成完成")
+                            elif event_name == "TaskFailed":
+                                error_info = data.get("payload", {}).get(
+                                    "error_info", {}
+                                )
+                                error_code = error_info.get("error_code")
+                                error_message = error_info.get(
+                                    "error_message", "未知错误"
+                                )
+                                raise Exception(
+                                    f"合成失败: {error_code} - {error_message}"
+                                )
+                finally:
+                    try:
+                        await ws.close()
+                    except:
+                        pass
+
+            loop.run_until_complete(_generate_audio())
+            loop.close()
+
+            return audio_data
+        except Exception as e:
+            logger.bind(tag=TAG).error(f"生成音频数据失败: {str(e)}")
+            return []
@@ -1,21 +1,22 @@
 import os
 import re
-import queue
+import time
 import uuid
+import queue
 import asyncio
 import threading
-from typing import Callable, Any
+import traceback
 from core.utils import p3
-import time
 from datetime import datetime
 from core.utils import textUtils
+from typing import Callable, Any
 from abc import ABC, abstractmethod
 from config.logger import setup_logging
-from core.utils.util import audio_bytes_to_data_stream, audio_to_data_stream
 from core.utils.tts import MarkdownCleaner
 from core.utils.output_counter import add_device_output
 from core.handle.reportHandle import enqueue_tts_report
 from core.handle.sendAudioHandle import sendAudioMessage
+from core.utils.util import audio_bytes_to_data_stream, audio_to_data_stream
 from core.providers.tts.dto.dto import (
    TTSMessageDTO,
    SentenceType,
@@ -23,8 +24,6 @@ from core.providers.tts.dto.dto import (
    InterfaceType,
 )

-import traceback
-
 TAG = __name__
 logger = setup_logging()

@@ -144,6 +143,68 @@ class TTSProviderBase(ABC):
            except Exception as e:
                logger.bind(tag=TAG).error(f"Failed to generate TTS file: {e}")
                return None
+    
+    def to_tts(self, text):
+        text = MarkdownCleaner.clean_markdown(text)
+        max_repeat_time = 5
+        if self.delete_audio_file:
+            # 需要删除文件的直接转为音频数据
+            while max_repeat_time > 0:
+                try:
+                    audio_bytes = asyncio.run(self.text_to_speak(text, None))
+                    if audio_bytes:
+                        audio_datas = []
+                        audio_bytes_to_data_stream(
+                            audio_bytes,
+                            file_type=self.audio_file_type,
+                            is_opus=True,
+                            callback=lambda data: audio_datas.append(data)
+                        )
+                        return audio_datas
+                    else:
+                        max_repeat_time -= 1
+                except Exception as e:
+                    logger.bind(tag=TAG).warning(
+                        f"语音生成失败{5 - max_repeat_time + 1}次: {text}，错误: {e}"
+                    )
+                    max_repeat_time -= 1
+            if max_repeat_time > 0:
+                logger.bind(tag=TAG).info(
+                    f"语音生成成功: {text}，重试{5 - max_repeat_time}次"
+                )
+            else:
+                logger.bind(tag=TAG).error(
+                    f"语音生成失败: {text}，请检查网络或服务是否正常"
+                )
+            return None
+        else:
+            tmp_file = self.generate_filename()
+            try:
+                while not os.path.exists(tmp_file) and max_repeat_time > 0:
+                    try:
+                        asyncio.run(self.text_to_speak(text, tmp_file))
+                    except Exception as e:
+                        logger.bind(tag=TAG).warning(
+                            f"语音生成失败{5 - max_repeat_time + 1}次: {text}，错误: {e}"
+                        )
+                        # 未执行成功，删除文件
+                        if os.path.exists(tmp_file):
+                            os.remove(tmp_file)
+                        max_repeat_time -= 1
+
+                if max_repeat_time > 0:
+                    logger.bind(tag=TAG).info(
+                        f"语音生成成功: {text}:{tmp_file}，重试{5 - max_repeat_time}次"
+                    )
+                else:
+                    logger.bind(tag=TAG).error(
+                        f"语音生成失败: {text}，请检查网络或服务是否正常"
+                    )
+
+                return tmp_file
+            except Exception as e:
+                logger.bind(tag=TAG).error(f"Failed to generate TTS file: {e}")
+                return None

    @abstractmethod
    async def text_to_speak(self, text, output_file):
@@ -284,8 +345,8 @@ class TTSProviderBase(ABC):
                    enqueue_audio = []
                    enqueue_text = text

-                # 计算音频数据的帧数
-                if isinstance(audio_datas, bytes):
+                # 收集上报音频数据
+                if isinstance(audio_datas, bytes) and enqueue_audio is not None:
                    enqueue_audio.append(audio_datas)

                # 发送音频
@@ -143,8 +143,8 @@ class TTSProvider(TTSProviderBase):
        data = {
            "text": text,
            "references": [
-                ServeReferenceAudio(audio=audio if audio else b"", text=text)
-                for text, audio in zip(ref_texts, byte_audios)
+                ServeReferenceAudio(audio=audio if audio else b"", text=ref_text)
+                for ref_text, audio in zip(ref_texts, byte_audios)
            ],
            "reference_id": self.reference_id,
            "normalize": self.normalize,
@@ -628,3 +628,104 @@ class TTSProvider(TTSProviderBase):

    def wav_to_opus_data_audio_raw_stream(self, raw_data_var, is_end=False, callback: Callable[[Any], Any]=None):
        return self.opus_encoder.encode_pcm_to_opus_stream(raw_data_var, is_end, callback=callback)
+
+    def to_tts(self, text: str) -> list:
+        """非流式生成音频数据，用于生成音频及测试场景
+        Args:
+            text: 要转换的文本
+        Returns:
+            list: 音频数据列表
+        """
+        try:
+            # 创建事件循环
+            loop = asyncio.new_event_loop()
+            asyncio.set_event_loop(loop)
+
+            # 生成会话ID
+            session_id = uuid.uuid4().__str__().replace("-", "")
+
+            # 存储音频数据
+            audio_data = []
+
+            async def _generate_audio():
+                # 创建新的WebSocket连接
+                ws_header = {
+                    "X-Api-App-Key": self.appId,
+                    "X-Api-Access-Key": self.access_token,
+                    "X-Api-Resource-Id": self.resource_id,
+                    "X-Api-Connect-Id": uuid.uuid4(),
+                }
+                ws = await websockets.connect(
+                    self.ws_url, additional_headers=ws_header, max_size=1000000000
+                )
+
+                try:
+                    # 启动会话
+                    header = Header(
+                        message_type=FULL_CLIENT_REQUEST,
+                        message_type_specific_flags=MsgTypeFlagWithEvent,
+                        serial_method=JSON,
+                    ).as_bytes()
+                    optional = Optional(
+                        event=EVENT_StartSession, sessionId=session_id
+                    ).as_bytes()
+                    payload = self.get_payload_bytes(
+                        event=EVENT_StartSession, speaker=self.voice
+                    )
+                    await self.send_event(ws, header, optional, payload)
+
+                    # 发送文本
+                    header = Header(
+                        message_type=FULL_CLIENT_REQUEST,
+                        message_type_specific_flags=MsgTypeFlagWithEvent,
+                        serial_method=JSON,
+                    ).as_bytes()
+                    optional = Optional(
+                        event=EVENT_TaskRequest, sessionId=session_id
+                    ).as_bytes()
+                    payload = self.get_payload_bytes(
+                        event=EVENT_TaskRequest, text=text, speaker=self.voice
+                    )
+                    await self.send_event(ws, header, optional, payload)
+
+                    # 发送结束会话请求
+                    header = Header(
+                        message_type=FULL_CLIENT_REQUEST,
+                        message_type_specific_flags=MsgTypeFlagWithEvent,
+                        serial_method=JSON,
+                    ).as_bytes()
+                    optional = Optional(
+                        event=EVENT_FinishSession, sessionId=session_id
+                    ).as_bytes()
+                    payload = str.encode("{}")
+                    await self.send_event(ws, header, optional, payload)
+
+                    # 接收音频数据
+                    while True:
+                        msg = await ws.recv()
+                        res = self.parser_response(msg)
+
+                        if (
+                            res.optional.event == EVENT_TTSResponse
+                            and res.header.message_type == AUDIO_ONLY_RESPONSE
+                        ):
+                            self.wav_to_opus_data_audio_raw_stream(res.payload, callback=lambda opus_frame: audio_data.append(opus_frame))
+                        elif res.optional.event == EVENT_SessionFinished:
+                            break
+
+                finally:
+                    # 清理资源
+                    try:
+                        await ws.close()
+                    except:
+                        pass
+
+            # 运行异步任务
+            loop.run_until_complete(_generate_audio())
+            loop.close()
+
+            return audio_data
+
+        except Exception as e:
+            logger.bind(tag=TAG).error(f"生成音频数据失败: {str(e)}")
+            return []
@@ -1,8 +1,10 @@
 import os
+import time
 import queue
-import asyncio
-import traceback
 import aiohttp
+import asyncio
+import requests
+import traceback
 from config.logger import setup_logging
 from core.utils.tts import MarkdownCleaner
 from core.providers.tts.base import TTSProviderBase
@@ -177,3 +179,57 @@ class TTSProvider(TTSProviderBase):
        await super().close()
        if hasattr(self, "opus_encoder"):
            self.opus_encoder.close()
+
+    def to_tts(self, text: str) -> list:
+        """非流式TTS处理，用于测试及保存音频文件的场景
+        Args:
+            text: 要转换的文本
+        Returns:
+            list: 返回opus编码后的音频数据列表
+        """
+        start_time = time.time()
+        text = MarkdownCleaner.clean_markdown(text)
+
+        payload = {"text": text, "character": self.character}
+
+        try:
+            with requests.post(self.api_url, json=payload, timeout=5) as response:
+                if response.status_code != 200:
+                    logger.bind(tag=TAG).error(
+                        f"TTS请求失败: {response.status_code}, {response.text}"
+                    )
+                    return []
+
+                logger.info(f"TTS请求成功: {text}, 耗时: {time.time() - start_time}秒")
+
+                # 使用opus编码器处理PCM数据
+                opus_datas = []
+                pcm_data = response.content
+
+                # 计算每帧的字节数
+                frame_bytes = int(
+                    self.opus_encoder.sample_rate
+                    * self.opus_encoder.channels
+                    * self.opus_encoder.frame_size_ms
+                    / 1000
+                    * 2
+                )
+
+                # 分帧处理PCM数据
+                for i in range(0, len(pcm_data), frame_bytes):
+                    frame = pcm_data[i : i + frame_bytes]
+                    if len(frame) < frame_bytes:
+                        # 最后一帧可能不足，用0填充
+                        frame = frame + b"\x00" * (frame_bytes - len(frame))
+
+                    self.opus_encoder.encode_pcm_to_opus_stream(
+                        frame,
+                        end_of_stream=(i + frame_bytes >= len(pcm_data)),
+                        callback=lambda opus: opus_datas.append(opus)
+                    )
+
+                return opus_datas
+
+        except Exception as e:
+            logger.bind(tag=TAG).error(f"TTS请求异常: {e}")
+            return []
@@ -1,8 +1,10 @@
 import os
+import time
 import queue
-import asyncio
-import traceback
 import aiohttp
+import asyncio
+import requests
+import traceback
 from config.logger import setup_logging
 from core.utils.tts import MarkdownCleaner
 from core.providers.tts.base import TTSProviderBase
@@ -109,10 +111,6 @@ class TTSProvider(TTSProviderBase):
        finally:
            return None

-    ###################################################################################
-    # linkerai单流式TTS重写父类的方法--结束
-    ###################################################################################
-
    async def text_to_speak(self, text, is_last):
        """流式处理TTS音频，每句只推送一次音频列表"""
        await self._tts_request(text, is_last)
@@ -199,3 +197,71 @@ class TTSProvider(TTSProviderBase):
        except Exception as e:
            logger.bind(tag=TAG).error(f"TTS请求异常: {e}")
            self.tts_audio_queue.put((SentenceType.LAST, [], None))
+
+    def to_tts(self, text: str) -> list:
+        """非流式TTS处理，用于测试及保存音频文件的场景
+        Args:
+            text: 要转换的文本
+        Returns:
+            list: 返回opus编码后的音频数据列表
+        """
+        start_time = time.time()
+        text = MarkdownCleaner.clean_markdown(text)
+
+        params = {
+            "tts_text": text,
+            "spk_id": self.voice,
+            "frame_duration": 60,
+            "stream": False,
+            "target_sr": 16000,
+            "audio_format": self.audio_format,
+            "instruct_text": "请生成一段自然流畅的语音",
+        }
+        headers = {
+            "Authorization": f"Bearer {self.access_token}",
+            "Content-Type": "application/json",
+        }
+
+        try:
+            with requests.get(
+                self.api_url, params=params, headers=headers, timeout=5
+            ) as response:
+                if response.status_code != 200:
+                    logger.bind(tag=TAG).error(
+                        f"TTS请求失败: {response.status_code}, {response.text}"
+                    )
+                    return []
+
+                logger.info(f"TTS请求成功: {text}, 耗时: {time.time() - start_time}秒")
+
+                # 使用opus编码器处理PCM数据
+                opus_datas = []
+                pcm_data = response.content
+
+                # 计算每帧的字节数
+                frame_bytes = int(
+                    self.opus_encoder.sample_rate
+                    * self.opus_encoder.channels
+                    * self.opus_encoder.frame_size_ms
+                    / 1000
+                    * 2
+                )
+
+                # 分帧处理PCM数据
+                for i in range(0, len(pcm_data), frame_bytes):
+                    frame = pcm_data[i : i + frame_bytes]
+                    if len(frame) < frame_bytes:
+                        # 最后一帧可能不足，用0填充
+                        frame = frame + b"\x00" * (frame_bytes - len(frame))
+
+                    self.opus_encoder.encode_pcm_to_opus_stream(
+                        frame,
+                        end_of_stream=(i + frame_bytes >= len(pcm_data)),
+                        callback=lambda opus: opus_datas.append(opus)
+                    )
+
+                return opus_datas
+
+        except Exception as e:
+            logger.bind(tag=TAG).error(f"TTS请求异常: {e}")
+            return []
@@ -1,13 +1,15 @@
-import asyncio
-import json
-import base64
-import aiohttp
-import numpy as np
 import io
 import wave
+import json
+import base64
+import asyncio
 import websockets
-from core.providers.tts.base import TTSProviderBase
+import numpy as np
+from datetime import datetime
 from config.logger import setup_logging
+from core.providers.tts.base import TTSProviderBase
+
+

 TAG = __name__
 logger = setup_logging()
@@ -18,11 +20,12 @@ class TTSProvider(TTSProviderBase):
        super().__init__(config, delete_audio_file)
        self.url = config.get("url", "ws://192.168.1.10:8092/paddlespeech/tts/streaming")
        self.protocol = config.get("protocol", "websocket")
+        
        if config.get("private_voice"):
            self.spk_id = int(config.get("private_voice"))
        else:
-            self.spk_id = int(config.get("spk_id", "0"))  
-        
+            self.spk_id = int(config.get("spk_id", "0"))
+            
        sample_rate = config.get("sample_rate", 24000)
        self.sample_rate = float(sample_rate) if sample_rate else 24000
        
@@ -32,7 +35,21 @@ class TTSProvider(TTSProviderBase):
        volume = config.get("volume", 1.0)
        self.volume = float(volume) if volume else 1.0
        
-        self.save_path = config.get("save_path", "./streaming_tts.wav")
+        self.delete_audio_file = config.get("delete_audio", True)
+        if not self.delete_audio_file:
+            timestamp = datetime.now().strftime("%Y%m%d_%H%M%S")
+            save_path = config.get("save_path")
+            if save_path:
+                if not save_path.endswith('.wav'):
+                    save_path = f"{save_path}_{timestamp}.wav"
+                else:
+                    other_path = save_path[:-4]
+                    save_path = f"{other_path}_{timestamp}.wav"
+                self.save_path = save_path
+            else:
+                self.save_path = f"./streaming_tts_{timestamp}.wav"
+        else:
+            self.save_path = None

    async def pcm_to_wav(self, pcm_data: bytes, sample_rate: int = 24000, num_channels: int = 1,
                         bits_per_sample: int = 16) -> bytes:
@@ -58,43 +75,9 @@ class TTSProvider(TTSProviderBase):
    async def text_to_speak(self, text, output_file):
        if self.protocol == "websocket":
            return await self.text_streaming(text, output_file)
-        elif self.protocol == "http":
-            return await self.text(text, output_file)
        else:
            raise ValueError("Unsupported protocol. Please use 'websocket' or 'http'.")

-    async def text(self, text, output_file):
-        request_json = {
-            "text": text,
-            "spk_id": self.spk_id,
-            "speed": self.speed,
-            "volume": self.volume,
-            "sample_rate": self.sample_rate,
-            "save_path": self.save_path
-        }
-
-        try:
-            async with aiohttp.ClientSession() as session:
-                async with session.post(self.url, json=request_json) as resp:
-                    if resp.status == 200:
-                        resp_json = await resp.json()
-                        if resp_json.get("success"):
-                            data = resp_json["result"]
-                            audio_bytes = base64.b64decode(data["audio"])
-                            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"Error: {resp_json.get('message', 'Unknown error')} while processing text: {text}")
-                    else:
-                        raise Exception(
-                            f"HTTP Error: {resp.status} - {await resp.text()} while processing text: {text}")
-        except Exception as e:
-            raise Exception(f"Error during TTS HTTP request: {e} while processing text: {text}")
-
    async def text_streaming(self, text, output_file):
        try:
            # 使用 websockets 异步连接到 WebSocket 服务器
@@ -151,6 +134,12 @@ class TTSProvider(TTSProviderBase):
                # 接收结束响应避免服务抛出异常
                await ws.recv()

+                # 根据配置决定是否保存文件
+                if not self.delete_audio_file and self.save_path:
+                    with open(self.save_path, "wb") as f:
+                        f.write(wav_data)
+                    logger.bind(tag=TAG).info(f"音频文件已保存到: {self.save_path}")
+                
                # 返回或保存音频数据
                if output_file:
                    with open(output_file, "wb") as file_to_save:
@@ -159,4 +148,4 @@ class TTSProvider(TTSProviderBase):
                    return wav_data

        except Exception as e:
-            raise Exception(f"Error during TTS WebSocket request: {e} while processing text: {text}")
+            raise Exception(f"Error during TTS WebSocket request: {e} while processing text: {text}")
@@ -33,8 +33,8 @@ class VADProvider(VADProviderBase):
            int(min_silence_duration_ms) if min_silence_duration_ms else 1000
        )

-        # 至少要多少帧才算有语音,增加灵敏度
-        self.frame_window_threshold = 1
+        # 至少要多少帧才算有语音
+        self.frame_window_threshold = 3

    def is_vad(self, conn, opus_packet):
        try: