update: 音频流式优化

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

@@ -1,6 +1,4 @@
 import json
-import asyncio
-import time
 from core.providers.tts.dto.dto import SentenceType
 from core.utils import textUtils
 
@@ -8,20 +6,18 @@ 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
-        pre_buffer = True
         await send_tts_message(conn, "start", None)
 
     if sentenceType == SentenceType.FIRST:
         await send_tts_message(conn, "sentence_start", text)
 
-    await sendAudio(conn, audios, pre_buffer)
+    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}")
 
     # 发送结束消息（如果是最后一个文本）
     if conn.llm_finish_task and sentenceType == SentenceType.LAST:
@@ -32,46 +28,12 @@ async def sendAudioMessage(conn, sentenceType, audios, text):
 
 
 # 播放音频
-async def sendAudio(conn, audios, pre_buffer=True):
+async def sendAudio(conn, audios):
     if audios is None:
         return
     # 如果audios不是opus数组，则不需要进行遍历，可以直接发送;这里需要进行流控管理，防止发送过快引发客户端溢出
-    if isinstance(audios ,bytes):
+    if isinstance(audios, bytes):
         await conn.websocket.send(audios)
-    else:
-        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
-        # 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:
-                break
-
-            # 重置没有声音的状态
-            conn.last_activity_time = time.time() * 1000
-
-            # 计算预期发送时间
-            expected_time = start_time + (play_position / 1000)
-            current_time = time.perf_counter()
-            delay = expected_time - current_time
-            if delay > 0:
-                await asyncio.sleep(delay)
-
-            await conn.websocket.send(opus_packet)
-
-            play_position += frame_duration
 
 
 async def send_tts_message(conn, state, text=None):
@@ -100,13 +62,12 @@ 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: