'commit'

dsLightRag/Test/TTS/T3_TTS_Pipeline.py

@@ -1,12 +1,137 @@
 import asyncio
+import json
 import os
-import sys
+import re
 import uuid
+import websockets
+from queue import Queue
 
-# 添加路径以导入其他模块
-sys.path.append(os.path.dirname(os.path.abspath(__file__)))
-from T1_LLM import stream_and_split_text
-from T2_StreamingVolanoTTS import StreamingVolcanoTTS
+from Config import Config
+from Util.LlmUtil import get_llm_response
+from Util.TTS_Protocols import full_client_request, receive_message, MsgType, EventType
+
+
+def stream_and_split_text(prompt):
+    """
+    流式获取LLM输出并按句子分割
+    @param prompt: 提示文本
+    @return: 生成器，每次产生一个完整句子
+    """
+    buffer = ""
+    
+    # 使用LlmUtil中的get_llm_response函数获取流式响应
+    for content in get_llm_response(prompt, stream=True):
+        buffer += content
+        
+        # 使用正则表达式检测句子结束
+        sentences = re.split(r'([。！？.!?])', buffer)
+        if len(sentences) > 1:
+            # 提取完整句子
+            for i in range(0, len(sentences)-1, 2):
+                if i+1 < len(sentences):
+                    sentence = sentences[i] + sentences[i+1]
+                    yield sentence
+                    
+            # 保留不完整的部分
+            buffer = sentences[-1]
+    
+    # 处理最后剩余的部分
+    if buffer:
+        yield buffer
+
+
+class StreamingVolcanoTTS:
+    def __init__(self, voice_type='zh_female_wanwanxiaohe_moon_bigtts', encoding='wav', max_concurrency=2):
+        self.voice_type = voice_type
+        self.encoding = encoding
+        self.app_key = Config.HS_APP_ID
+        self.access_token = Config.HS_ACCESS_TOKEN
+        self.endpoint = "wss://openspeech.bytedance.com/api/v3/tts/unidirectional/stream"
+        self.audio_queue = Queue()
+        self.max_concurrency = max_concurrency  # 最大并发数
+        self.semaphore = asyncio.Semaphore(max_concurrency)  # 并发控制信号量
+        
+    @staticmethod
+    def get_resource_id(voice: str) -> str:
+        if voice.startswith("S_"):
+            return "volc.megatts.default"
+        return "volc.service_type.10029"
+        
+    async def synthesize_stream(self, text_stream, audio_callback):
+        """
+        流式合成语音
+        
+        Args:
+            text_stream: 文本流生成器
+            audio_callback: 音频数据回调函数，接收音频片段
+        """
+        # 为每个文本片段创建一个WebSocket连接，但限制并发数
+        tasks = []
+        for text in text_stream:
+            if text.strip():  # 忽略空文本
+                task = asyncio.create_task(self._synthesize_single_with_semaphore(text, audio_callback))
+                tasks.append(task)
+                
+        # 等待所有任务完成
+        await asyncio.gather(*tasks)
+        
+    async def _synthesize_single_with_semaphore(self, text, audio_callback):
+        """使用信号量控制并发数的单个文本合成"""
+        async with self.semaphore:  # 获取信号量，限制并发数
+            await self._synthesize_single(text, audio_callback)
+            
+    async def _synthesize_single(self, text, audio_callback):
+        """合成单个文本片段"""
+        headers = {
+            "X-Api-App-Key": self.app_key,
+            "X-Api-Access-Key": self.access_token,
+            "X-Api-Resource-Id": self.get_resource_id(self.voice_type),
+            "X-Api-Connect-Id": str(uuid.uuid4()),
+        }
+
+        websocket = await websockets.connect(
+            self.endpoint, additional_headers=headers, max_size=10 * 1024 * 1024
+        )
+
+        try:
+            request = {
+                "user": {
+                    "uid": str(uuid.uuid4()),
+                },
+                "req_params": {
+                    "speaker": self.voice_type,
+                    "audio_params": {
+                        "format": self.encoding,
+                        "sample_rate": 24000,
+                        "enable_timestamp": True,
+                    },
+                    "text": text,
+                    "additions": json.dumps({"disable_markdown_filter": False}),
+                },
+            }
+
+            # 发送请求
+            await full_client_request(websocket, json.dumps(request).encode())
+
+            # 接收音频数据
+            audio_data = bytearray()
+            while True:
+                msg = await receive_message(websocket)
+
+                if msg.type == MsgType.FullServerResponse:
+                    if msg.event == EventType.SessionFinished:
+                        break
+                elif msg.type == MsgType.AudioOnlyServer:
+                    audio_data.extend(msg.payload)
+                else:
+                    raise RuntimeError(f"TTS conversion failed: {msg}")
+
+            # 通过回调函数返回音频数据
+            if audio_data:
+                audio_callback(audio_data)
+
+        finally:
+            await websocket.close()
 
 
 async def streaming_tts_pipeline(prompt, audio_callback):