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https://github.com/xinnan-tech/xiaozhi-esp32-server.git
synced 2026-07-29 02:43:55 +08:00
mqtt
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@@ -55,6 +55,7 @@ async def sendAudio(conn, audios, frame_duration=60):
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"last_send_time": 0,
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"packet_count": 0,
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"start_time": time.perf_counter(),
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"sequence": 0, # 添加序列号
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}
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flow_control = conn.audio_flow_control
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@@ -67,11 +68,22 @@ async def sendAudio(conn, audios, frame_duration=60):
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if delay > 0:
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await asyncio.sleep(delay)
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# 发送数据包
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await conn.websocket.send(audios)
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# 为opus数据包添加16字节头部
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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 = bytearray(16)
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header[0] = 1 # type
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header[2:4] = len(audios).to_bytes(2, 'big') # payload length
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header[4:8] = flow_control["sequence"].to_bytes(4, 'big') # connection id/sequence
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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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complete_packet = bytes(header) + audios
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await conn.websocket.send(complete_packet)
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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["sequence"] += 1
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flow_control["last_send_time"] = time.perf_counter()
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else:
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# 文件型音频走普通播放
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@@ -81,11 +93,21 @@ async def sendAudio(conn, audios, frame_duration=60):
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# 执行预缓冲
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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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await conn.websocket.send(audios[i])
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# 为预缓冲包添加头部
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timestamp = int((start_time + i * frame_duration / 1000) * 1000) % (2**32)
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header = bytearray(16)
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header[0] = 1 # type
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header[2:4] = len(audios[i]).to_bytes(2, 'big') # payload length
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header[4:8] = i.to_bytes(4, 'big') # sequence
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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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await conn.websocket.send(complete_packet)
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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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for opus_packet in remaining_audios:
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for i, opus_packet in enumerate(remaining_audios):
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if conn.client_abort:
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break
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@@ -98,9 +120,21 @@ async def sendAudio(conn, audios, frame_duration=60):
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delay = expected_time - current_time
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if delay > 0:
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await asyncio.sleep(delay)
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await conn.websocket.send(opus_packet)
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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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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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sequence = pre_buffer_frames + i # 确保序列号连续
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header = bytearray(16)
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header[0] = 1 # type
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header[2:4] = len(opus_packet).to_bytes(2, 'big') # payload length
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header[4:8] = sequence.to_bytes(4, 'big') # sequence
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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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complete_packet = bytes(header) + opus_packet
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await conn.websocket.send(complete_packet)
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play_position += frame_duration
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