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video/rtc/rtc.py

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import queue
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import asyncio
import aiohttp
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import threading
import time
from aiortc import RTCPeerConnection, RTCSessionDescription, RTCConfiguration
from aiortc.mediastreams import MediaStreamTrack
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# 创建一个长度为1的队列用于生产者和消费者之间的通信
frame_queue = queue.Queue(maxsize=1)
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class VideoTrack(MediaStreamTrack):
"""自定义视频轨道类继承自MediaStreamTrack"""
kind = "video"
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def __init__(self, max_frames=100):
super().__init__()
self.frames = queue.Queue(maxsize=max_frames)
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async def recv(self):
return await super().recv()
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def webrtc_producer(webrtc_url):
"""
生产者方法从WEBRTC读取视频帧并放入队列
仅当队列空时才放入新帧否则丢弃
"""
loop = asyncio.new_event_loop()
asyncio.set_event_loop(loop)
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# 创建RTCPeerConnection对象不使用ICE服务器
pc = RTCPeerConnection(RTCConfiguration(iceServers=[]))
video_track = VideoTrack()
pc.addTrack(video_track)
@pc.on("track")
async def on_track(track):
if track.kind == "video":
print("接收到视频轨道,开始接收视频帧")
while True:
# 从轨道接收视频帧
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frame = await track.recv()
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# 转换为BGR24格式的NumPy数组
frame_bgr24 = frame.to_ndarray(format='bgr24')
# 检查队列是否为空,为空则加入,否则丢弃
if frame_queue.empty():
try:
frame_queue.put_nowait(frame_bgr24)
print("帧已放入队列")
except queue.Full:
print("队列已满,丢弃帧")
else:
print("队列非空,丢弃帧")
async def main():
# 创建并发送SDP Offer
offer = await pc.createOffer()
print("已创建本地SDP Offer")
await pc.setLocalDescription(offer)
# 发送Offer到服务器并接收Answer
async with aiohttp.ClientSession() as session:
print(f"开始向服务器 {webrtc_url} 发送SDP Offer")
async with session.post(
webrtc_url,
data=offer.sdp.encode(),
headers={
"Content-Type": "application/sdp",
"Content-Length": str(len(offer.sdp))
},
ssl=False
) as response:
print("已接收到服务器的响应")
answer_sdp = await response.text()
await pc.setRemoteDescription(RTCSessionDescription(sdp=answer_sdp, type='answer'))
# 保持连接
try:
while True:
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await asyncio.sleep(0.1)
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except KeyboardInterrupt:
pass
finally:
print("关闭RTCPeerConnection")
await pc.close()
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try:
loop.run_until_complete(main())
finally:
loop.close()
def frame_consumer(ip):
"""
消费者方法从队列中读取帧并处理
每次处理后休眠200ms模拟延迟
"""
print("消费者启动,开始等待帧...")
try:
while True:
# 阻塞等待队列中的帧
frame = frame_queue.get()
print(f"消费帧,大小: {frame.shape}")
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has_violation, violations, confidences = OCRViolationDetector.detect(frame)
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# 输出检测结果
if has_violation:
detector.logger.info(f"在图片中检测到 {len(violations)} 个违禁词:")
for word, conf in zip(violations, confidences):
detector.logger.info(f"- {word} (置信度: {conf:.4f})")
else:
detector.logger.info("图片中未检测到违禁词")
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# 标记任务完成
frame_queue.task_done()
except KeyboardInterrupt:
print("消费者退出")
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def start_webrtc_stream(ip, webrtc_url):
"""
启动WebRTC视频流处理的主方法
参数: webrtc_url - WebRTC服务器地址
"""
print(f"开始连接到WebRTC服务器: {webrtc_url}")
# 启动生产者线程
producer_thread = threading.Thread(
target=webrtc_producer,
args=(webrtc_url,),
daemon=True,
name="webrtc-producer"
)
# 启动消费者线程
consumer_thread = threading.Thread(
target=frame_consumer(ip),
daemon=True,
name="frame-consumer"
)
producer_thread.start()
consumer_thread.start()
print("生产者和消费者线程已启动")
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try:
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# 保持主线程运行
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while True:
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time.sleep(1)
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except KeyboardInterrupt:
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print("程序正在退出...")
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if __name__ == "__main__":
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# 示例用法
# 实际使用时替换为真实的WebRTC服务器地址
webrtc_server_url = "http://192.168.110.65:1985/rtc/v1/whep/?app=live&stream=677a4845aa48cb8526c811ad56fc5e60"
start_webrtc_stream(webrtc_server_url)