使用python-gstreamer播放RTSP
我用gstreamer来播放来自IP摄像头（比如Axis）的RTSP流。
我使用这样的命令行：
gst-launch-0.10 rtspsrc location=rtsp://192.168.0.127/axis-media/media.amp latency=0 ! decodebin ! autovideosink
这个方法运行得很好。
我想用pygtk做一个图形界面来控制它，所以我使用了gstreamer的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)
[...]
但是它不工作，程序退出时显示了这个信息：
gst.element_link_many(source, decoder,sink)
gst.LinkError: failed to link source with decoder
我还尝试改进我的命令行，只使用h264格式：
gst-launch-0.10 -v rtspsrc location=rtsp://192.168.0.127/axis-media/media.amp ! rtph264depay ! ffdec_h264 ! xvimagesink
然后在我的Python代码中这样实现：
[...]
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)
[...]
但我还是遇到了同样的错误 :(
gst.LinkError: failed to link source with depay
我的源头（rtspsrc）之间似乎有问题，因为它在使用decodebin和filesrc时可以正常工作（当然，rtph264depay是不行的）。
我不明白为什么它在命令行中可以工作，但在这里却不行。
有没有gstreamer的专家能帮我一下？
提前谢谢你们。
祝好，
2 个回答
2
这个回答解释了为什么你会遇到一个 gst.LinkError 的问题：关于Python中Gstreamer的gst.LinkError问题
通过 gst.parse_launch，你可以给元素命名，然后再把它们取出来设置属性：
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'
6
我有一个用"C"语言写的代码实现，觉得把它转换成"Python"应该挺简单的。
//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;
}
构建文件
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)
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();
}
});
}
}