读取帧优化

This commit is contained in:
ZZX9599
2025-09-03 10:35:14 +08:00
parent be5383d752
commit 1816b5c5dd

View File

@ -1,64 +1,71 @@
import queue
import asyncio
from datetime import datetime
import aiohttp
import cv2
import numpy as np
import threading
import time
from aiortc import RTCPeerConnection, RTCSessionDescription, RTCConfiguration
from ocr.ocr_violation_detector import OCRViolationDetector
from ws.ws import send_message_to_client
from aiortc.mediastreams import MediaStreamTrack
# 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)
async def rtc_frame_receiver(url, frame_queue, stop_event):
class VideoTrack(MediaStreamTrack):
"""自定义视频轨道类继承自MediaStreamTrack"""
kind = "video"
def __init__(self, max_frames=100):
super().__init__()
self.frames = queue.Queue(maxsize=max_frames)
async def recv(self):
return await super().recv()
def webrtc_producer(webrtc_url):
"""
接收RTC帧并往队列放入cv2格式的帧数据
当队列已满时直接丢弃新帧,不阻塞等待
当stop_event被设置时停止接收
生产者方法从WEBRTC读取视频帧并放入队列
当队列空时才放入新帧,否则丢弃
"""
loop = asyncio.new_event_loop()
asyncio.set_event_loop(loop)
# 创建RTCPeerConnection对象不使用ICE服务器
pc = RTCPeerConnection(RTCConfiguration(iceServers=[]))
# 累计帧计数器和丢弃帧计数器
total_frames = 0
dropped_frames = 0
video_track = VideoTrack()
pc.addTrack(video_track)
@pc.on("track")
async def on_track(track):
nonlocal total_frames, dropped_frames
if track.kind == "video":
print("接收到视频轨道开始接收视频帧")
while not stop_event.is_set(): # 检查是否需要停止
# 接收当前帧并累计计数
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
frame = await track.recv()
total_frames += 1
# 转换为BGR24格式的NumPy数组
frame_bgr24 = frame.to_ndarray(format='bgr24')
# 转换为cv2兼容的BGR格式numpy数组
frame_cv2 = frame.to_ndarray(format='bgr24')
# 验证是否为cv2兼容格式
if isinstance(frame_cv2, np.ndarray) and frame_cv2.ndim == 3 and frame_cv2.shape[2] == 3:
# 检查队列是否已满
if frame_queue.full():
# 队列已满,丢弃当前帧
dropped_frames += 1
print(f"{total_frames}帧:队列已满,丢弃该帧(累计丢弃: {dropped_frames}")
# 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:
# 队列未满,放入当前帧
await frame_queue.put(frame_cv2)
print(f"{total_frames}帧:已放入队列")
else:
print("帧格式转换失败不是有效的cv2格式")
print("队列非空,丢弃帧")
# 创建并设置本地offer
async def main():
# 创建并发送SDP Offer
offer = await pc.createOffer()
print("已创建本地 SDP Offer")
print("已创建本地SDP Offer")
await pc.setLocalDescription(offer)
# 发送offer到服务器
# 发送Offer到服务器并接收Answer
async with aiohttp.ClientSession() as session:
print("开始向服务器发送 SDP Offer")
try:
print(f"开始向服务器 {webrtc_url} 发送SDP Offer")
async with session.post(
url,
webrtc_url,
data=offer.sdp.encode(),
headers={
"Content-Type": "application/sdp",
@ -66,107 +73,83 @@ async def rtc_frame_receiver(url, frame_queue, stop_event):
},
ssl=False
) as response:
if response.status == 200:
print("已接收到服务器的响应、开始处理 SDP Answer")
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'))
else:
print(f"服务器响应错误: {response.status}")
stop_event.set()
except Exception as e:
print(f"发送SDP Offer失败: {str(e)}")
stop_event.set()
# 保持连接
try:
# 保持连接,直到收到停止信号
while not stop_event.is_set():
await asyncio.sleep(1)
while True:
await asyncio.sleep(0.1)
except KeyboardInterrupt:
print("用户中断")
pass
finally:
print(f"开始关闭 RTCPeerConnection,共接收{total_frames}帧,丢弃{dropped_frames}")
print("关闭RTCPeerConnection")
await pc.close()
print("已关闭 RTCPeerConnection")
async def frame_consumer(ip, frame_queue, stop_event):
"""
从队列中阻塞读取cv2帧并处理队列为空时阻塞等待
检测到违规内容后设置stop_event以终止所有任务
"""
# 创建OCR检测器实例
ocr_detector = OCRViolationDetector(
forbidden_words_path=r"D:\Git\bin\video\ocr\forbidden_words.txt",
ocr_confidence_threshold=0.5, )
while not stop_event.is_set(): # 检查是否需要停止
try:
# 阻塞等待队列中的帧
current_frame = await frame_queue.get()
# 进行OCR检测
has_violation, words, confidences = ocr_detector.detect(current_frame)
print(f"检测结果: {'有违规内容' if has_violation else '无违规内容'}")
print(f"检测到的词: {words}")
print(f"置信度: {confidences}")
# 输出所有检测到的违禁词
if has_violation:
print(f"测试结果:图片中共检测到 {len(words)} 个违禁词:")
response_data = {
"status": "stop",
"timestamp": datetime.now().isoformat(),
"violations": [{"word": w, "confidence": c} for w, c in zip(words, confidences)]
}
await send_message_to_client(ip, response_data)
for word, conf in zip(words, confidences):
print(f"- {word}(置信度:{conf:.4f}")
# 检测到违规,设置停止事件
print("检测到违规内容准备关闭AI检测")
stop_event.set()
# 标记任务完成,允许生产者放入新的帧
frame_queue.task_done()
except Exception as e:
print(f"处理帧时发生错误: {str(e)}")
frame_queue.task_done()
def process_webrtc_stream(ip, webrtc_url):
"""
处理WEBRTC流并持续打印OCR检测结果检测到违规后关闭
队列大小为1满时直接丢弃新帧
Args:
ip: IP地址
webrtc_url: WEBRTC服务器地址
"""
# 创建队列大小为1和停止事件
frame_queue = asyncio.Queue(maxsize=1) # 只存储一帧
stop_event = asyncio.Event() # 用于控制任务停止的事件
# 定义事件循环中的主任务
async def main_task():
# 创建任务
receiver_task = asyncio.create_task(rtc_frame_receiver(webrtc_url, frame_queue, stop_event))
consumer_task = asyncio.create_task(frame_consumer(ip, frame_queue, stop_event))
# 等待任务完成
await asyncio.gather(receiver_task, consumer_task)
# 确保队列处理完毕
await frame_queue.join()
try:
# 运行事件循环
asyncio.run(main_task())
except KeyboardInterrupt:
print("用户中断处理流程")
stop_event.set()
loop.run_until_complete(main())
finally:
# 确保关闭所有cv2窗口
cv2.destroyAllWindows()
print("AI检测已关闭")
loop.close()
def frame_consumer():
"""
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}")
# 模拟处理延迟
time.sleep(0.2) # 200ms
# 标记任务完成
frame_queue.task_done()
except KeyboardInterrupt:
print("消费者退出")
def start_webrtc_stream(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,
daemon=True,
name="frame-consumer"
)
producer_thread.start()
consumer_thread.start()
print("生产者和消费者线程已启动")
try:
# 保持主线程运行
while True:
time.sleep(1)
except KeyboardInterrupt:
print("程序正在退出...")
if __name__ == "__main__":
# 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)