update:流式优化暂不稳定，先回滚到早期代码

main/xiaozhi-server/core/handle/sendAudioHandle.py

@@ -1,25 +1,25 @@
 import json
-import time
 import asyncio
-from core.utils import textUtils
+import time
 from core.providers.tts.dto.dto import SentenceType
+from core.utils import textUtils
 
 TAG = __name__
 
 
 async def sendAudioMessage(conn, sentenceType, audios, text):
+    # 发送句子开始消息
+    conn.logger.bind(tag=TAG).info(f"发送音频消息: {sentenceType}, {text}")
+
+    pre_buffer = False
     if conn.tts.tts_audio_first_sentence:
         conn.logger.bind(tag=TAG).info(f"发送第一段语音: {text}")
         conn.tts.tts_audio_first_sentence = False
-        await send_tts_message(conn, "start", None)
+        pre_buffer = True
 
-    if sentenceType == SentenceType.FIRST:
-        await send_tts_message(conn, "sentence_start", text)
+    await send_tts_message(conn, "sentence_start", text)
 
-    await sendAudio(conn, audios)
-    # 发送句子开始消息
-    if sentenceType is not SentenceType.MIDDLE:
-        conn.logger.bind(tag=TAG).info(f"发送音频消息: {sentenceType}, {text}")
+    await sendAudio(conn, audios, pre_buffer)
 
     # 发送结束消息（如果是最后一个文本）
     if conn.llm_finish_task and sentenceType == SentenceType.LAST:
@@ -30,59 +30,45 @@ async def sendAudioMessage(conn, sentenceType, audios, text):
 
 
 # 播放音频
-async def sendAudio(conn, audios, pre_buffer=False):
-    """
-    发送单个opus包，支持流控
-    Args:
-        conn: 连接对象
-        opus_packet: 单个opus数据包
-        pre_buffer: 快速发送音频
-    """
+async def sendAudio(conn, audios, pre_buffer=True):
     if audios is None or len(audios) == 0:
         return
+    # 流控参数优化
+    frame_duration = 60  # 帧时长（毫秒），匹配 Opus 编码
+    start_time = time.perf_counter()
+    play_position = 0
 
-    if isinstance(audios, bytes):
+    # 仅当第一句话时执行预缓冲
+    if pre_buffer:
+        pre_buffer_frames = min(3, len(audios))
+        for i in range(pre_buffer_frames):
+            await conn.websocket.send(audios[i])
+        remaining_audios = audios[pre_buffer_frames:]
+    else:
+        remaining_audios = audios
+
+    # 播放剩余音频帧
+    for opus_packet in remaining_audios:
         if conn.client_abort:
-            return
-
-        # 短音频直接发送（例如：提示音）
-        if pre_buffer:
-            await conn.websocket.send(audios)
-            return
+            break
 
+        # 重置没有声音的状态
         conn.last_activity_time = time.time() * 1000
 
-        # 获取或初始化流控状态
-        if not hasattr(conn, "audio_flow_control"):
-            conn.audio_flow_control = {
-                "last_send_time": 0,
-                "packet_count": 0,
-                "start_time": time.perf_counter(),
-            }
-
-        frame_duration=60
-        flow_control = conn.audio_flow_control
-        current_time = time.perf_counter()
         # 计算预期发送时间
-        expected_time = flow_control["start_time"] + (
-            flow_control["packet_count"] * frame_duration / 1000
-        )
+        expected_time = start_time + (play_position / 1000)
+        current_time = time.perf_counter()
         delay = expected_time - current_time
         if delay > 0:
             await asyncio.sleep(delay)
 
-        # 发送数据包
-        await conn.websocket.send(audios)
+        await conn.websocket.send(opus_packet)
 
-        # 更新流控状态
-        flow_control["packet_count"] += 1
-        flow_control["last_send_time"] = time.perf_counter()
+        play_position += frame_duration
 
 
 async def send_tts_message(conn, state, text=None):
     """发送 TTS 状态消息"""
-    if text is None and state == "sentence_start":
-        return
     message = {"type": "tts", "state": state, "session_id": conn.session_id}
     if text is not None:
         message["text"] = textUtils.check_emoji(text)
@@ -95,12 +81,8 @@ async def send_tts_message(conn, state, text=None):
             stop_tts_notify_voice = conn.config.get(
                 "stop_tts_notify_voice", "config/assets/tts_notify.mp3"
             )
-            conn.tts.audio_to_opus_data_stream(
-                stop_tts_notify_voice,
-                callback=lambda audio_data: asyncio.run_coroutine_threadsafe(
-                    sendAudio(conn, audio_data, True), conn.loop
-                ),
-            )
+            audios, _ = conn.tts.audio_to_opus_data(stop_tts_notify_voice)
+            await sendAudio(conn, audios)
         # 清除服务端讲话状态
         conn.clearSpeakStatus()
 
@@ -109,17 +91,18 @@ async def send_tts_message(conn, state, text=None):
 
 
 async def send_stt_message(conn, text):
-    """发送 STT 状态消息"""
     end_prompt_str = conn.config.get("end_prompt", {}).get("prompt")
     if end_prompt_str and end_prompt_str == text:
         await send_tts_message(conn, "start")
         return
 
+    """发送 STT 状态消息"""
+    
     # 解析JSON格式，提取实际的用户说话内容
     display_text = text
     try:
         # 尝试解析JSON格式
-        if text.strip().startswith("{") and text.strip().endswith("}"):
+        if text.strip().startswith('{') and text.strip().endswith('}'):
             parsed_data = json.loads(text)
             if isinstance(parsed_data, dict) and "content" in parsed_data:
                 # 如果是包含说话人信息的JSON格式，只显示content部分