update:优化代码

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

@@ -1,12 +1,11 @@
-from core.handle.sendAudioHandle import send_stt_message
+import time
+import json
+import asyncio
+from core.utils.util import audio_to_data
+from core.handle.abortHandle import handleAbortMessage
 from core.handle.intentHandler import handle_user_intent
 from core.utils.output_counter import check_device_output_limit
-from core.handle.abortHandle import handleAbortMessage
-import time
-import asyncio
-import json
-from core.handle.sendAudioHandle import SentenceType
-from core.utils.util import audio_to_data
+from core.handle.sendAudioHandle import send_stt_message, SentenceType
 
 TAG = __name__
 
@@ -22,7 +21,6 @@ async def handleAudioMessage(conn, audio):
         if not hasattr(conn, "vad_resume_task") or conn.vad_resume_task.done():
             conn.vad_resume_task = asyncio.create_task(resume_vad_detection(conn))
         return
-
     if have_voice:
         if conn.client_is_speaking:
             await handleAbortMessage(conn)
@@ -42,22 +40,22 @@ async def startToChat(conn, text):
     # 检查输入是否是JSON格式（包含说话人信息）
     speaker_name = None
     actual_text = text
-    
+
     try:
         # 尝试解析JSON格式的输入
-        if text.strip().startswith('{') and text.strip().endswith('}'):
+        if text.strip().startswith("{") and text.strip().endswith("}"):
             data = json.loads(text)
-            if 'speaker' in data and 'content' in data:
-                speaker_name = data['speaker']
-                actual_text = data['content']
+            if "speaker" in data and "content" in data:
+                speaker_name = data["speaker"]
+                actual_text = data["content"]
                 conn.logger.bind(tag=TAG).info(f"解析到说话人信息: {speaker_name}")
-                
+
                 # 直接使用JSON格式的文本，不解析
                 actual_text = text
     except (json.JSONDecodeError, KeyError):
         # 如果解析失败，继续使用原始文本
         pass
-    
+
     # 保存说话人信息到连接对象
     if speaker_name:
         conn.current_speaker = speaker_name
@@ -118,10 +116,12 @@ async def no_voice_close_connect(conn, have_voice):
 
 
 async def max_out_size(conn):
+    # 播放超出最大输出字数的提示
+    conn.client_abort = False
     text = "不好意思，我现在有点事情要忙，明天这个时候我们再聊，约好了哦！明天不见不散，拜拜！"
     await send_stt_message(conn, text)
     file_path = "config/assets/max_output_size.wav"
-    opus_packets, _ = audio_to_data(file_path)
+    opus_packets = audio_to_data(file_path)
     conn.tts.tts_audio_queue.put((SentenceType.LAST, opus_packets, text))
     conn.close_after_chat = True
 
@@ -140,7 +140,7 @@ async def check_bind_device(conn):
 
         # 播放提示音
         music_path = "config/assets/bind_code.wav"
-        opus_packets, _ = audio_to_data(music_path)
+        opus_packets = audio_to_data(music_path)
         conn.tts.tts_audio_queue.put((SentenceType.FIRST, opus_packets, text))
 
         # 逐个播放数字
@@ -148,15 +148,17 @@ async def check_bind_device(conn):
             try:
                 digit = conn.bind_code[i]
                 num_path = f"config/assets/bind_code/{digit}.wav"
-                num_packets, _ = audio_to_data(num_path)
+                num_packets = audio_to_data(num_path)
                 conn.tts.tts_audio_queue.put((SentenceType.MIDDLE, num_packets, None))
             except Exception as e:
                 conn.logger.bind(tag=TAG).error(f"播放数字音频失败: {e}")
                 continue
         conn.tts.tts_audio_queue.put((SentenceType.LAST, [], None))
     else:
+        # 播放未绑定提示
+        conn.client_abort = False
         text = f"没有找到该设备的版本信息，请正确配置 OTA地址，然后重新编译固件。"
         await send_stt_message(conn, text)
         music_path = "config/assets/bind_not_found.wav"
-        opus_packets, _ = audio_to_data(music_path)
+        opus_packets = audio_to_data(music_path)
         conn.tts.tts_audio_queue.put((SentenceType.LAST, opus_packets, text))

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

@@ -30,6 +30,28 @@ async def sendAudioMessage(conn, sentenceType, audios, text):
             await conn.close()
 
 
+async def _send_to_mqtt_gateway(conn, opus_packet, timestamp, sequence):
+    """
+    发送带16字节头部的opus数据包给mqtt_gateway
+    Args:
+        conn: 连接对象
+        opus_packet: opus数据包
+        timestamp: 时间戳
+        sequence: 序列号
+    """
+    # 为opus数据包添加16字节头部
+    header = bytearray(16)
+    header[0] = 1  # type
+    header[2:4] = len(opus_packet).to_bytes(2, "big")  # payload length
+    header[4:8] = sequence.to_bytes(4, "big")  # sequence
+    header[8:12] = timestamp.to_bytes(4, "big")  # 时间戳
+    header[12:16] = len(opus_packet).to_bytes(4, "big")  # opus长度
+
+    # 发送包含头部的完整数据包
+    complete_packet = bytes(header) + opus_packet
+    await conn.websocket.send(complete_packet)
+
+
 # 播放音频
 async def sendAudio(conn, audios, frame_duration=60):
     """
@@ -42,9 +64,6 @@ async def sendAudio(conn, audios, frame_duration=60):
     """
     if audios is None or len(audios) == 0:
         return
-    # 检查是否需要处理头部（只有当websocket URL以"?from=mqtt"为结尾时才处理头部）
-    request_path = conn.websocket.request.path
-    need_header = request_path.endswith("?from=mqtt")
 
     if isinstance(audios, bytes):
         if conn.client_abort:
@@ -71,19 +90,19 @@ async def sendAudio(conn, audios, frame_duration=60):
         if delay > 0:
             await asyncio.sleep(delay)
 
-        if need_header:
-            # 为opus数据包添加16字节头部
-            timestamp = int((flow_control["start_time"] + flow_control["packet_count"] * frame_duration / 1000) * 1000) % (2**32)
-            header = bytearray(16)
-            header[0] = 1  # type
-            header[2:4] = len(audios).to_bytes(2, 'big')  # payload length
-            header[4:8] = flow_control["sequence"].to_bytes(4, 'big')  # connection id/sequence
-            header[8:12] = timestamp.to_bytes(4, 'big')  # 时间戳
-            header[12:16] = len(audios).to_bytes(4, 'big')  # opus长度
-            
-            # 发送包含头部的完整数据包
-            complete_packet = bytes(header) + audios
-            await conn.websocket.send(complete_packet)
+        if conn.conn_from_mqtt_gateway:
+            # 计算时间戳
+            timestamp = int(
+                (
+                    flow_control["start_time"]
+                    + flow_control["packet_count"] * frame_duration / 1000
+                )
+                * 1000
+            ) % (2**32)
+            # 调用通用函数发送带头部的数据包
+            await _send_to_mqtt_gateway(
+                conn, audios, timestamp, flow_control["sequence"]
+            )
         else:
             # 直接发送opus数据包，不添加头部
             await conn.websocket.send(audios)
@@ -97,30 +116,21 @@ async def sendAudio(conn, audios, frame_duration=60):
         start_time = time.perf_counter()
         play_position = 0
 
-        # 检查是否需要添加头部（只有当websocket URL以"?from=mqtt"为结尾时才添加头部）
-        request_path = conn.websocket.request.path
-        need_header = request_path.endswith("?from=mqtt")
-
         # 执行预缓冲
         pre_buffer_frames = min(3, len(audios))
         for i in range(pre_buffer_frames):
-            if need_header:
-                # 为预缓冲包添加头部
-                timestamp = int((start_time + i * frame_duration / 1000) * 1000) % (2**32)
-                header = bytearray(16)
-                header[0] = 1  # type
-                header[2:4] = len(audios[i]).to_bytes(2, 'big')  # payload length
-                header[4:8] = i.to_bytes(4, 'big')  # sequence
-                header[8:12] = timestamp.to_bytes(4, 'big')  # 时间戳
-                header[12:16] = len(audios[i]).to_bytes(4, 'big')  # opus长度
-                
-                complete_packet = bytes(header) + audios[i]
-                await conn.websocket.send(complete_packet)
+            if conn.conn_from_mqtt_gateway:
+                # 计算时间戳
+                timestamp = int((start_time + i * frame_duration / 1000) * 1000) % (
+                    2**32
+                )
+                # 调用通用函数发送带头部的数据包
+                await _send_to_mqtt_gateway(conn, audios[i], timestamp, i)
             else:
                 # 直接发送预缓冲包，不添加头部
                 await conn.websocket.send(audios[i])
         remaining_audios = audios[pre_buffer_frames:]
-        
+
         # 播放剩余音频帧
         for i, opus_packet in enumerate(remaining_audios):
             if conn.client_abort:
@@ -135,25 +145,19 @@ async def sendAudio(conn, audios, frame_duration=60):
             delay = expected_time - current_time
             if delay > 0:
                 await asyncio.sleep(delay)
-            
-            if need_header:
-                # 为opus数据包添加16字节头部 (timestamp at offset 8, length at offset 12)
-                timestamp = int((start_time + play_position / 1000) * 1000) % (2**32)  # 使用播放位置计算时间戳
+
+            if conn.conn_from_mqtt_gateway:
+                # 计算时间戳和序列号
+                timestamp = int((start_time + play_position / 1000) * 1000) % (
+                    2**32
+                )  # 使用播放位置计算时间戳
                 sequence = pre_buffer_frames + i  # 确保序列号连续
-                header = bytearray(16)
-                header[0] = 1  # type
-                header[2:4] = len(opus_packet).to_bytes(2, 'big')  # payload length
-                header[4:8] = sequence.to_bytes(4, 'big')  # sequence
-                header[8:12] = timestamp.to_bytes(4, 'big')  # 时间戳在第8-11字节
-                header[12:16] = len(opus_packet).to_bytes(4, 'big')  # opus长度在第12-15字节
-                
-                # 发送包含头部的完整数据包
-                complete_packet = bytes(header) + opus_packet
-                await conn.websocket.send(complete_packet)
+                # 调用通用函数发送带头部的数据包
+                await _send_to_mqtt_gateway(conn, opus_packet, timestamp, sequence)
             else:
                 # 直接发送opus数据包，不添加头部
                 await conn.websocket.send(opus_packet)
-            
+
             play_position += frame_duration