Question

所有这些都在同一台计算机上完成。我希望使用 Gst 使用 python 从 Gstreamer 接收视频流。首先我将介绍gstreamer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

然后我想从gstreamer获取一个视频流，用python中的脚本输出到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)

我看到了这个错误

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？

接收器gstreamer管道

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

Answer 1

您必须在udpsrc上协商上限，如

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

您如何发送数据，因为这些上限会有所不同。您必须协商RTP格式，因为您在udpsink之前使用了rtp pay元素。

udp使用python在gstreamer中播放视频

1 个答案: