udp videostreaming与python的gstreamer

Question

所有这些都在同一台计算机上完成。我想从Gstreamer使用GST使用python接收视频流。首先，我将介绍GStreamer的管道

gst-launch-1.0 v4l2src ! video/x-raw,width=640,height=480 ! timeoverlay ! tee name="local" ! queue ! autovideosink local. ! queue ! jpegenc! rtpjpegpay ! udpsink host=127.0.0.1 port= 5000 

那么我想从GStreamer的视频流与蟒蛇输出脚本的OpenCV

import sys 
from gi.repository import Gst 
import cv2 
import numpy 

Gst.init(None) 


image_arr = None 

def gst_to_opencv(sample): 
    buf = sample.get_buffer() 
    caps = sample.get_caps() 
    print caps.get_structure(0).get_value('format') 
    print caps.get_structure(0).get_value('height') 
    print caps.get_structure(0).get_value('width') 

    print buf.get_size() 

    arr = numpy.ndarray(
     (caps.get_structure(0).get_value('height'), 
     caps.get_structure(0).get_value('width'), 
     3), 
     buffer=buf.extract_dup(0, buf.get_size()), 
     dtype=numpy.uint8) 
    return arr 

def new_buffer(sink, data): 
    global image_arr 
    sample = sink.emit("pull-sample") 
    buf = sample.get_buffer() 
    print "Timestamp: ", buf.pts 
    arr = gst_to_opencv(sample) 
    image_arr = arr 
    return Gst.FlowReturn.OK 

# Create the elements 
source = Gst.ElementFactory.make("udpsrc", None) 
depay = Gst.ElementFactory.make("rtpjpegdepay", None) 
decoder = Gst.ElementFactory.make("jpegdec", None) 
sink = Gst.ElementFactory.make("appsink", None) 



# Create the empty pipeline111 
pipeline = Gst.Pipeline.new("test-pipeline") 

if not pipeline: 
    print("Not all elements could be created.") 
    exit(-1) 


sink.set_property("emit-signals", True) 
#sink.set_property("max-buffers", 2) 
# # sink.set_property("drop", True) 
# # sink.set_property("sync", False) 

#caps = Gst.caps_from_string("application/x-rtp, encoding name=JPEG, payload=26, width=640, height=480; video/x-bayer,format=(string){rggb,bggr,grbg,gbrg}") 
caps = Gst.caps_from_string(" application/x-rtp, encoding-name=JPEG, payload=26, format=(string)[]{rggb,bggr,grbg,gbrg}") 
sink.set_property("caps", caps) 


sink.connect("new-sample", new_buffer, sink) 

# Build the pipeline 
pipeline.add(source) 
pipeline.add(depay) 
pipeline.add(decoder) 
pipeline.add(sink) 
if not Gst.Element.link(source, depay): 
    print("Elements could not be linked.") 
    exit(-1) 
if not Gst.Element.link(depay, decoder): 
    print("Elements could not be linked.") 
    exit(-1) 
if not Gst.Element.link(decoder, sink): 
    print("Elements could not be linked.") 
    exit(-1) 

# Modify the source's properties 
source.set_property("port", 5000) 

# Start playing 
ret = pipeline.set_state(Gst.State.PLAYING) 
if ret == Gst.StateChangeReturn.FAILURE: 
    print("Unable to set the pipeline to the playing state.") 
    exit(-1) 

# Wait until error or EOS 
bus = pipeline.get_bus() 


# Parse message 
while True: 
    message = bus.timed_pop_filtered(10000, Gst.MessageType.ANY) 
    # print "image_arr: ", image_arr 
    if image_arr is not None: 
     cv2.imshow("appsink image arr", image_arr) 
     cv2.waitKey(1) 
    if message: 
     if message.type == Gst.MessageType.ERROR: 
      err, debug = message.parse_error() 
      print("Error received from element %s: %s" % (
       message.src.get_name(), err)) 
      print("Debugging information: %s" % debug) 
      break 
     elif message.type == Gst.MessageType.EOS: 
      print("End-Of-Stream reached.") 
      break 
     elif message.type == Gst.MessageType.STATE_CHANGED: 
      if isinstance(message.src, Gst.Pipeline): 
       old_state, new_state, pending_state = message.parse_state_changed() 
       print("Pipeline state changed from %s to %s." % 
         (old_state.value_nick, new_state.value_nick)) 
     else: 
      print("Unexpected message received.") 

# Free resources 
pipeline.set_state(Gst.State.NULL) 

我看到这个错误

Pipeline state changed from null to ready. 
Unexpected message received. 
Pipeline state changed from ready to paused. 
Unexpected message received. 
Unexpected message received. 
Unexpected message received. 
Error received from element rtpjpegdepay0: No RTP format was negotiated. 
Debugging information: gstrtpbasedepayload.c(373): gst_rtp_base_depayload_chain(): /GstPipeline:test-pipeline/GstRtpJPEGDepay:rtpjpegdepay0: 
Input buffers need to have RTP caps set on them. This is usually achieved by setting the 'caps' property of the upstream source element (often udpsrc or appsrc), or by putting a capsfilter element before the depayloader and setting the 'caps' property on that. Also see http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/gst/rtp/README 

如何解决它？

接收器的GStreamer管道

gst-launch-1.0 udpsrc port=5000 ! application/x-rtp, encoding-name=JPEG,payload=26 ! rtpjpegdepay ! jpegdec ! autovideosink 

Answer 1

你有你如何发送数据，基于这些上限会有所不同谈判上udpsrc帽状

gst-launch-1.0 udpsrc port=9090 caps=application/x-rtp,media=video,encoding-name=MP2T 

。您必须协商RTP格式，因为您在udpsink之前使用了rtp付费元素。