使用python-gstreamer播放RTSP

Question

我使用gstreamer從IP攝像機（如Axis）播放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

它工作正常。

我想用pygtk中的gui控制它，所以我使用gstreamer python綁定。 我寫了這段代碼：

[...]
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)
[...]

但它不起作用並退出此消息：

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

我也嘗試用這個來改進我的CLI，因為我只使用h264：

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

並在我的python代碼中實現它：

[...]
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)
[...]

但我得到了同樣的錯誤:(

gst.LinkError: failed to link source with depay

我的源（rtspsrc）之間有一些問題，因為它與decodebin一起使用filesrc（當然不能用rtph264depay工作）

我不明白為什么它不起作用，因為它在cli中工作。 gstreamer的任何專家誰可以幫助我？

提前致謝。

問候，

Answer 1

我有一個你正在尋找的代碼的“C”實現。 我認為轉換為“Python”應該相當簡單

 //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文件

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)

UPDATE

等效Java代碼

 // 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

這個答案解釋了為什么你得到一個gst.LinkError ： python的gst.LinkError問題的Gstreamer

使用gst.parse_launch ，您可以命名元素，然后檢索它們以設置屬性：

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'