使用python-gstreamer播放RTSP

Question

我使用gstreamer来播放来自IP摄像机(如Axis)的RTSP流。我使用如下命令行：

gst-launch-0.10 rtspsrc location=rtsp://192.168.0.127/axis-media/media.amp latency=0 ! decodebin ! autovideosink

它工作得很好。

我想用pygtk中的gui来控制它，所以我使用了gstreamer python绑定。我写了这段代码：

[...]
self.player = gst.Pipeline("player")
source = gst.element_factory_make("rtspsrc", "source")
source.set_property("location", "rtsp://192.168.0.127/axis-media/media.amp")
decoder = gst.element_factory_make("decodebin", "decoder")
sink = gst.element_factory_make("autovideosink", "sink")

self.player.add(source, decoder, sink)
gst.element_link_many(source, decoder, sink)

bus = self.player.get_bus()
bus.add_signal_watch()
bus.enable_sync_message_emission()
bus.connect("message", self.on_message)
bus.connect("sync-message::element", self.on_sync_message)
[...]

但它不工作并退出，并显示以下消息：

gst.element_link_many(source, decoder,sink)
gst.LinkError: failed to link source with decoder

我也试着用这个来改进我的命令行界面，因为我只使用h264：

gst-launch-0.10 -v rtspsrc location=rtsp://192.168.0.127/axis-media/media.amp ! rtph264depay !  ffdec_h264 ! xvimagesink

并在我的python代码中像这样实现它：

[...]
self.player = gst.Pipeline("player")
source = gst.element_factory_make("rtspsrc", "source")
depay = gst.element_factory_make("rtph264depay", "depay")
decoder = gst.element_factory_make("ffdec_h264", "decoder")
sink = gst.element_factory_make("xvimagesink", "output")

self.player.add(source, depay, decoder, sink)
gst.element_link_many(source, depay, decoder, sink)
[...]

但我得到了相同的错误:(

gst.LinkError: failed to link source with depay

我的源码(rtspsrc)有问题，因为它使用filesrc进行解码(当然不能使用rtph264depay)

我不明白为什么它不能工作，因为它可以在cli中工作。有没有gstreamer的专家可以帮我？

提前谢谢。

致以敬意，

Answer 1

我有一个你正在寻找的代码的"C“实现。我认为转换成"Python“应该相当简单。

 //Display RTSP streaming of video
 //(c) 2011 enthusiasticgeek
 // This code is distributed in the hope that it will be useful,
 // but WITHOUT ANY WARRANTY; without even the implied warranty of
 // MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  

#include <string.h>
#include <math.h>
#include <gst/gst.h>
#include <glib.h>

static gboolean bus_call (GstBus *bus,GstMessage *msg, gpointer data){
  GMainLoop *loop = (GMainLoop *) data;

  switch (GST_MESSAGE_TYPE (msg)) {

    case GST_MESSAGE_EOS:
      g_print ("Stream Ends\n");
      g_main_loop_quit (loop);
      break;

    case GST_MESSAGE_ERROR: {
      gchar  *debug;
      GError *error;

      gst_message_parse_error (msg, &error, &debug);
      g_free (debug);

      g_printerr ("Error: %s\n", error->message);
      g_error_free (error);

      g_main_loop_quit (loop);
      break;
    }
    default:
      break;
  }

  return TRUE;
}

static void on_pad_added (GstElement *element, GstPad *pad, gpointer data){

  GstPad *sinkpad;
  GstElement *decoder = (GstElement *) data;

  /* We can now link this pad with the rtsp-decoder sink pad */
  g_print ("Dynamic pad created, linking source/demuxer\n");

  sinkpad = gst_element_get_static_pad (decoder, "sink");

  gst_pad_link (pad, sinkpad);

  gst_object_unref (sinkpad);
}

int main (int argc, char *argv[])
{
  GMainLoop *loop;
  GstBus *bus;
  GstElement *source;
  GstElement *decoder;
  GstElement *sink;
  GstElement *pipeline;
  GstElement *demux;
  GstElement *colorspace;

  /* Initializing GStreamer */
  gst_init (&argc, &argv);
  loop = g_main_loop_new (NULL, FALSE);

 //gst-launch-0.10 rtspsrc location=rtsp://<ip> ! decodebin ! ffmpegcolorspace ! autovideosink
 //gst-launch -v rtspsrc location="rtsp://<ip> ! rtpmp4vdepay ! mpeg4videoparse ! ffdec_mpeg4 ! ffmpegcolorspace! autovideosink
 //gst-launch -v rtspsrc location="rtsp://<ip> ! rtpmp4vdepay ! ffdec_mpeg4 ! ffmpegcolorspace! autovideosink
  /* Create Pipe's Elements */
  pipeline = gst_pipeline_new ("video player");
  g_assert (pipeline);
  source   = gst_element_factory_make ("rtspsrc", "Source");
  g_assert (source);
  demux = gst_element_factory_make ("rtpmp4vdepay", "Depay");
  g_assert (demux);
  decoder = gst_element_factory_make ("ffdec_mpeg4", "Decoder");
  g_assert (decoder);
  colorspace     = gst_element_factory_make ("ffmpegcolorspace",  "Colorspace");
  g_assert(colorspace);
  sink     = gst_element_factory_make ("autovideosink", "Output");
  g_assert (sink);

  /*Make sure: Every elements was created ok*/
  if (!pipeline || !source || !demux || !decoder || !colorspace || !sink) {
    g_printerr ("One of the elements wasn't create... Exiting\n");
    return -1;
  }

  g_printf(" \nPipeline is Part(A) ->(dynamic/runtime link)  Part(B)[ Part(B-1) -> Part(B-2) -> Part(B-3) ]\n\n");
  g_printf(" [source](dynamic)->(dynamic)[demux]->[decoder]->[colorspace]->[videosink] \n\n");

  /* Set video Source */
  g_object_set (G_OBJECT (source), "location", argv[1], NULL);
  //g_object_set (G_OBJECT (source), "do-rtcp", TRUE, NULL);
  g_object_set (G_OBJECT (source), "latency", 0, NULL);

  /* Putting a Message handler */
  bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
  gst_bus_add_watch (bus, bus_call, loop);
  gst_object_unref (bus);

  /* Add Elements to the Bin */
  gst_bin_add_many (GST_BIN (pipeline), source, demux, decoder, colorspace, sink, NULL);

  /* Link confirmation */
  if (!gst_element_link_many (demux, decoder, colorspace, sink, NULL)){
     g_warning ("Linking part (B) Fail...");
  }

  g_printf("\nNote that the source will be linked to the demuxer(depayload) dynamically.\n\
     The reason is that rtspsrc may contain various elements (for example\n\
     audio and video). The source pad(s) will be created at run time,\n\
     by the rtspsrc when it detects the amount and nature of elements.\n\
     Therefore we connect a callback function which will be executed\n\
     when the \"pad-added\" is emitted.\n");

  /* Dynamic Pad Creation */
  if(! g_signal_connect (source, "pad-added", G_CALLBACK (on_pad_added),demux))
  {
    g_warning ("Linking part (A) with part (B) Fail...");
  }
  /* Run the pipeline */
  g_print ("Playing: %s\n", argv[1]);
  gst_element_set_state (pipeline, GST_STATE_PLAYING);
  g_main_loop_run (loop);

  /* Ending Playback */
  g_print ("End of the Streaming... ending the playback\n");
  gst_element_set_state (pipeline, GST_STATE_NULL);

  /* Eliminating Pipeline */
  g_print ("Eliminating Pipeline\n");
  gst_object_unref (GST_OBJECT (pipeline));

  return 0;
}

Makefile

test = test12
ext = c
CC = gcc
CPP = g++
gstreamer:
    $(CC) -g $(test).$(ext) -o $(test) `pkg-config gstreamer-0.10 --libs --cflags` `pkg-config gtk+-2.0 --libs --cflags`
clean:
    rm -rf $(test)

更新

等效的Java代码

 // Display RTSP streaming of video
 // (c) 2011 enthusiasticgeek
 // This code is distributed in the hope that it will be useful,
 // but WITHOUT ANY WARRANTY; without even the implied warranty of
 // MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE
 // Leave Credits intact

package video2; //replace this with your package
import java.awt.BorderLayout;
import java.awt.Dimension;

import javax.swing.JFrame;
import javax.swing.SwingUtilities;

//import org.gstreamer.Caps;
import org.gstreamer.Element;
import org.gstreamer.ElementFactory;
import org.gstreamer.Gst;
import org.gstreamer.Pad;
import org.gstreamer.PadDirection;
import org.gstreamer.Pipeline;
import org.gstreamer.swing.VideoComponent;

/**
 * A Simple videotest example.
 */
public class Main {
    public Main() {
    }
    private static Pipeline pipe;
    public static void main(String[] args) {
    // Quartz is abysmally slow at scaling video for some reason, so turn it off.
    System.setProperty("apple.awt.graphics.UseQuartz", "false");

    args = Gst.init("SwingVideoTest", args);

    pipe = new Pipeline("pipeline");
    /*
    final Element videosrc = ElementFactory.make("videotestsrc", "source");
    final Element videofilter = ElementFactory.make("capsfilter", "flt");
    videofilter.setCaps(Caps.fromString("video/x-raw-yuv, width=720, height=576"
            + ", bpp=32, depth=32, framerate=25/1"));
    */

     pipe.getBus().connect(new Bus.ERROR() {
        public void errorMessage(GstObject source, int code, String message) {
            System.out.println("Error occurred: " + message);
            Gst.quit();
        }
    });
    pipe.getBus().connect(new Bus.STATE_CHANGED() {
        public void stateChanged(GstObject source, State old, State current, State pending) {
            if (source == pipe) {
                System.out.println("Pipeline state changed from " + old + " to " + current);
            }
        }
    });
    pipe.getBus().connect(new Bus.EOS() {
        public void endOfStream(GstObject source) {
            System.out.println("Finished playing file");
            Gst.quit();
        }
    });        

     pipe.getBus().connect(new Bus.TAG() {
        public void tagsFound(GstObject source, TagList tagList) {
            for (String tag : tagList.getTagNames()) {
                System.out.println("Found tag " + tag + " = "
                        + tagList.getValue(tag, 0));
            }
        }
    });

    final Element source = ElementFactory.make("rtspsrc", "Source");
    final Element demux = ElementFactory.make("rtpmp4vdepay", "Depay");
    final Element decoder=ElementFactory.make("ffdec_mpeg4", "Decoder");
    final Element colorspace = ElementFactory.make("ffmpegcolorspace",  "Colorspace");
    //final Element sink = ElementFactory.make ("autovideosink", "Output");

    SwingUtilities.invokeLater(new Runnable() {

        public void run() {
            // Create the video component and link it in
            VideoComponent videoComponent = new VideoComponent();
            Element videosink = videoComponent.getElement();

           source.connect(new Element.PAD_ADDED() {
           public void padAdded(Element element, Pad pad) {
            pad.link(demux.getStaticPad("sink"));
           }
            });

           Pad p = new Pad(null, PadDirection.SRC);
           source.addPad(p);

            source.set("location","rtsp://<user>:<pass>@<ip>/mpeg4/1/media.amp");  //replace this with your source

            pipe.addMany(source, demux, decoder, colorspace, videosink);
            Element.linkMany(demux, decoder, colorspace, videosink);

            // Now create a JFrame to display the video output
            JFrame frame = new JFrame("Swing Video Test");
            frame.setDefaultCloseOperation(JFrame.EXIT_ON_CLOSE);
            frame.add(videoComponent, BorderLayout.CENTER);
            videoComponent.setPreferredSize(new Dimension(720, 576));
            frame.pack();
            frame.setVisible(true);

            // Start the pipeline processing
            pipe.play();
        }
    });
    }
}

Answer 2

这个答案解释了为什么你会得到一个gst.LinkError：Gstreamer of python's gst.LinkError problem

使用gst.parse_launch，您可以命名元素，然后检索它们以设置属性：

pipeline = gst.parse_launch('rtspsrc name=source latency=0 ! decodebin ! autovideosink')
source = pipeline.get_by_name('source')
source.props.location = 'rtsp://192.168.0.127/axis-media/media.amp'