mirror of
https://github.com/xinnan-tech/xiaozhi-esp32-server.git
synced 2026-07-27 01:23:55 +08:00
兼容udp和websocket协议传输音频
This commit is contained in:
+2
-2
@@ -181,9 +181,9 @@ public class DeviceServiceImpl extends BaseServiceImpl<DeviceDao, DeviceEntity>
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response.setWebsocket(websocket);
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response.setWebsocket(websocket);
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// 添加MQTT UDP配置
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// 添加MQTT UDP配置
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// 从系统参数获取MQTT Gateway地址,如果未配置不使用默认值
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// 从系统参数获取MQTT Gateway地址,仅在配置有效时使用
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String mqttUdpConfig = sysParamsService.getValue(Constant.SERVER_MQTT_GATEWAY, false);
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String mqttUdpConfig = sysParamsService.getValue(Constant.SERVER_MQTT_GATEWAY, false);
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if(!StringUtils.isBlank(mqttUdpConfig) && deviceById != null) {
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if(mqttUdpConfig != null && !mqttUdpConfig.equals("null") && !mqttUdpConfig.isEmpty() && deviceById != null) {
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try {
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try {
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DeviceReportRespDTO.MQTT mqtt = buildMqttConfig(macAddress, clientId, deviceById);
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DeviceReportRespDTO.MQTT mqtt = buildMqttConfig(macAddress, clientId, deviceById);
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if (mqtt != null) {
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if (mqtt != null) {
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@@ -25,6 +25,8 @@ server:
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# 所以如果你使用docker部署时,将vision_explain设置成局域网地址
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# 所以如果你使用docker部署时,将vision_explain设置成局域网地址
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# 如果你使用公网部署时,将vision_explain设置成公网地址
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# 如果你使用公网部署时,将vision_explain设置成公网地址
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vision_explain: http://你的ip或者域名:端口号/mcp/vision/explain
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vision_explain: http://你的ip或者域名:端口号/mcp/vision/explain
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# mqtt网关地址,当这个值为null时,mqtt网关桥接功能不开启,使用websocket双向通信,不使用mqtt和udp协议
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mqtt_gateway: null
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# OTA返回信息时区偏移量
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# OTA返回信息时区偏移量
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timezone_offset: +8
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timezone_offset: +8
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# 认证配置
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# 认证配置
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@@ -282,8 +282,17 @@ class ConnectionHandler:
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if self.asr is None:
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if self.asr is None:
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return
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return
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# 检查是否需要处理头部(只有当mqtt_gateway有实际值时才处理头部)
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mqtt_gateway = self.config.get("server", {}).get("mqtt_gateway")
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# 当mqtt_gateway为None, "null", "", 或实际的null值时,不处理头部
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need_header_processing = mqtt_gateway and mqtt_gateway not in [None, "null", ""] and str(mqtt_gateway).strip() != ""
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if len(message) >= 16:
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# 调试日志:首次连接时记录配置
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if not hasattr(self, '_logged_mqtt_config'):
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self.logger.bind(tag=TAG).info(f"MQTT Gateway配置: '{mqtt_gateway}', 头部处理: {need_header_processing}")
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self._logged_mqtt_config = True
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if need_header_processing and len(message) >= 16:
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try:
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try:
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timestamp = int.from_bytes(message[8:12], 'big')
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timestamp = int.from_bytes(message[8:12], 'big')
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audio_length = int.from_bytes(message[12:16], 'big')
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audio_length = int.from_bytes(message[12:16], 'big')
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@@ -304,6 +313,7 @@ class ConnectionHandler:
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except Exception as e:
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except Exception as e:
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self.logger.bind(tag=TAG).error(f"解析WebSocket音频包失败: {e}")
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self.logger.bind(tag=TAG).error(f"解析WebSocket音频包失败: {e}")
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# 不需要头部处理或没有头部时,直接处理原始消息
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self.asr_audio_queue.put(message)
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self.asr_audio_queue.put(message)
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def _process_websocket_audio(self, audio_data, timestamp):
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def _process_websocket_audio(self, audio_data, timestamp):
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@@ -43,6 +43,11 @@ async def sendAudio(conn, audios, frame_duration=60):
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if audios is None or len(audios) == 0:
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if audios is None or len(audios) == 0:
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return
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return
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# 检查是否需要添加头部(只有当mqtt_gateway有实际值时才添加头部)
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mqtt_gateway = conn.config.get("server", {}).get("mqtt_gateway")
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# 当mqtt_gateway为None, "null", "", 或实际的null值时,不添加头部
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need_header = mqtt_gateway and mqtt_gateway not in [None, "null", ""] and str(mqtt_gateway).strip() != ""
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if isinstance(audios, bytes):
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if isinstance(audios, bytes):
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if conn.client_abort:
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if conn.client_abort:
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return
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return
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@@ -68,18 +73,22 @@ async def sendAudio(conn, audios, frame_duration=60):
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if delay > 0:
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if delay > 0:
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await asyncio.sleep(delay)
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await asyncio.sleep(delay)
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# 为opus数据包添加16字节头部
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if need_header:
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timestamp = int((flow_control["start_time"] + flow_control["packet_count"] * frame_duration / 1000) * 1000) % (2**32)
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# 为opus数据包添加16字节头部
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header = bytearray(16)
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timestamp = int((flow_control["start_time"] + flow_control["packet_count"] * frame_duration / 1000) * 1000) % (2**32)
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header[0] = 1 # type
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header = bytearray(16)
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header[2:4] = len(audios).to_bytes(2, 'big') # payload length
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header[0] = 1 # type
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header[4:8] = flow_control["sequence"].to_bytes(4, 'big') # connection id/sequence
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header[2:4] = len(audios).to_bytes(2, 'big') # payload length
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header[8:12] = timestamp.to_bytes(4, 'big') # 时间戳
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header[4:8] = flow_control["sequence"].to_bytes(4, 'big') # connection id/sequence
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header[12:16] = len(audios).to_bytes(4, 'big') # opus长度
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header[8:12] = timestamp.to_bytes(4, 'big') # 时间戳
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header[12:16] = len(audios).to_bytes(4, 'big') # opus长度
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# 发送包含头部的完整数据包
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# 发送包含头部的完整数据包
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complete_packet = bytes(header) + audios
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complete_packet = bytes(header) + audios
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await conn.websocket.send(complete_packet)
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await conn.websocket.send(complete_packet)
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else:
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# 直接发送opus数据包,不添加头部
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await conn.websocket.send(audios)
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# 更新流控状态
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# 更新流控状态
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flow_control["packet_count"] += 1
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flow_control["packet_count"] += 1
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@@ -90,20 +99,29 @@ async def sendAudio(conn, audios, frame_duration=60):
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start_time = time.perf_counter()
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start_time = time.perf_counter()
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play_position = 0
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play_position = 0
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# 检查是否需要添加头部(只有当mqtt_gateway有实际值时才添加头部)
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mqtt_gateway = conn.config.get("server", {}).get("mqtt_gateway")
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# 当mqtt_gateway为None, "null", "", 或实际的null值时,不添加头部
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need_header = mqtt_gateway and mqtt_gateway not in [None, "null", ""] and str(mqtt_gateway).strip() != ""
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# 执行预缓冲
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# 执行预缓冲
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pre_buffer_frames = min(3, len(audios))
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pre_buffer_frames = min(3, len(audios))
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for i in range(pre_buffer_frames):
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for i in range(pre_buffer_frames):
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# 为预缓冲包添加头部
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if need_header:
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timestamp = int((start_time + i * frame_duration / 1000) * 1000) % (2**32)
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# 为预缓冲包添加头部
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header = bytearray(16)
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timestamp = int((start_time + i * frame_duration / 1000) * 1000) % (2**32)
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header[0] = 1 # type
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header = bytearray(16)
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header[2:4] = len(audios[i]).to_bytes(2, 'big') # payload length
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header[0] = 1 # type
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header[4:8] = i.to_bytes(4, 'big') # sequence
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header[2:4] = len(audios[i]).to_bytes(2, 'big') # payload length
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header[8:12] = timestamp.to_bytes(4, 'big') # 时间戳
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header[4:8] = i.to_bytes(4, 'big') # sequence
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header[12:16] = len(audios[i]).to_bytes(4, 'big') # opus长度
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header[8:12] = timestamp.to_bytes(4, 'big') # 时间戳
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header[12:16] = len(audios[i]).to_bytes(4, 'big') # opus长度
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complete_packet = bytes(header) + audios[i]
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complete_packet = bytes(header) + audios[i]
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await conn.websocket.send(complete_packet)
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await conn.websocket.send(complete_packet)
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else:
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# 直接发送预缓冲包,不添加头部
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await conn.websocket.send(audios[i])
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remaining_audios = audios[pre_buffer_frames:]
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remaining_audios = audios[pre_buffer_frames:]
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# 播放剩余音频帧
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# 播放剩余音频帧
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@@ -121,19 +139,23 @@ async def sendAudio(conn, audios, frame_duration=60):
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if delay > 0:
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if delay > 0:
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await asyncio.sleep(delay)
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await asyncio.sleep(delay)
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# 为opus数据包添加16字节头部 (timestamp at offset 8, length at offset 12)
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if need_header:
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timestamp = int((start_time + play_position / 1000) * 1000) % (2**32) # 使用播放位置计算时间戳
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# 为opus数据包添加16字节头部 (timestamp at offset 8, length at offset 12)
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sequence = pre_buffer_frames + i # 确保序列号连续
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timestamp = int((start_time + play_position / 1000) * 1000) % (2**32) # 使用播放位置计算时间戳
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header = bytearray(16)
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sequence = pre_buffer_frames + i # 确保序列号连续
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header[0] = 1 # type
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header = bytearray(16)
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header[2:4] = len(opus_packet).to_bytes(2, 'big') # payload length
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header[0] = 1 # type
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header[4:8] = sequence.to_bytes(4, 'big') # sequence
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header[2:4] = len(opus_packet).to_bytes(2, 'big') # payload length
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header[8:12] = timestamp.to_bytes(4, 'big') # 时间戳在第8-11字节
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header[4:8] = sequence.to_bytes(4, 'big') # sequence
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header[12:16] = len(opus_packet).to_bytes(4, 'big') # opus长度在第12-15字节
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header[8:12] = timestamp.to_bytes(4, 'big') # 时间戳在第8-11字节
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header[12:16] = len(opus_packet).to_bytes(4, 'big') # opus长度在第12-15字节
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# 发送包含头部的完整数据包
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# 发送包含头部的完整数据包
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complete_packet = bytes(header) + opus_packet
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complete_packet = bytes(header) + opus_packet
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await conn.websocket.send(complete_packet)
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await conn.websocket.send(complete_packet)
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else:
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# 直接发送opus数据包,不添加头部
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await conn.websocket.send(opus_packet)
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play_position += frame_duration
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play_position += frame_duration
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