gstreamer

问题 I am working on an application that uses GStreamer to send a Motion JPEG video stream through a tcpclientsink element. The application works fine, except if I disrupt the network by switching the connection from wired to wireless or wireless to wired. When that happens, it looks like the tcpclientsink element waits 15 minutes before responding to messages. That becomes a problem if I try to shut down the application during this time. Here is what I've observed: Start a Motion JPEG media

问题 I am trying learning how to use dynamic pads in gstreamer. So I tried to add pad-added signal so I can get a message once an element is created. However, I didn't get any message. Here is the code: #include <gst/gst.h> static void cb_new_pad (GstElement *element, GstPad *pad, gpointer data) { gchar *name; name = gst_pad_get_name (pad); g_print ("A new pad %s was created\n", name); g_free (name); /* here, you would setup a new pad link for the newly created pad */ } int main (int argc, char

问题 I'm following this: http://docs.gstreamer.com/display/GstSDK/Installing+for+Android+development I'm using the Windows x86 ADT bundle, the Windows x86 r9d NDK, and the Gstreamer SDK linked to in the guide. I was able to get the first tutorial to add native support, build, and run. Then I tried the 3rd tutorial, and Eclipse becomes unresponsive. Same thing happened with the 2nd tutorial. I left it to do it over the weekend in case it just took a long time. When I change anything for another

问题 I am trying to capture the microphone audio and save it as a file. But its not working, i can only play the file while assign. How can i enable the microphone and buffer it or save or dump as raw .odd/vorbis ? #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 ("End of stream\n"); g_main_loop_quit (loop); break; case GST_MESSAGE

问题 I'm trying to play a local mpeg2 TS file with gstreamer with this: gst-launch filesrc location=open_season.mpg ! mpeg2dec ! xvimagesink The first frame appears as big blocks of color and then stops. Any thoughts about what I'm doing wrong here? Does a -TS file need to be handled differently than this? Here's the log: $ gst-launch filesrc location=open_season.mpg ! mpeg2dec ! xvimagesink Setting pipeline to PAUSED ... Pipeline is PREROLLING .... WARNING: from element /GstPipeline:pipeline0

问题 I am trying to build a program or robot that can generate audio and video on the fly and stream it to a rtmp server such a Twitch or Youtube. I have found that I can pull in streams of video and audio using Gstreamer. I also have found that Gstreamer has a library in Python which is good because my robot is already written in Python. The problem is, when testing with the testvideosrc input, I can make the command work just fine with both audio and video on the command line, but it exits

问题 Closed . This question needs to be more focused. It is not currently accepting answers. Want to improve this question? Update the question so it focuses on one problem only by editing this post. Closed 4 months ago . I wish to implement that command line in code. But "src." part => I didn't understand how to do. At gstreamer website there is a tee - queue sample but rtspsrc cant be link with tee element. And wherenever I remove decodebin it crashes.How can I code that ? gst-launch-1.0 rtspsrc

问题 I can compile the first gstreamer tutorial found here. However, when try to run it, I get a warning like this: (basic-tutorial-1:27661): GStreamer-WARNING **: Failed to load plugin '/usr/lib/gstreamer-0.10/libgstffmpeg.so': /usr/lib/x86_64-linux-gnu/libavcodec.so.53: undefined symbol: lame_set_VBR_quality Cannot connect to server socket err = No such file or directory Cannot connect to server socket jack server is not running or cannot be started How can I solve the lame_set_VBR_quality

问题 Looking for explanation how to using named elements in respect with muxing two inputs in one module. For instance muxing audio and video in one mpegtsmux modle gst-launch filesrc location=surround.mp4 ! decodebin name=dmux ! queue ! audioconvert ! lamemp3enc dmux. ! queue ! x264enc ! mpegtsmux name=mux ! queue ! filesink location=out.ts Above pipeline gives plugins interconnection like below So it shows audio doesn't connect to mpegtsmus. How to modify command line to have audio and video

问题 Here is what I have come up with so far for seeking while playing: case GDK_Up: { gint64 pos_ns, dur_ns, seek_ns; GstFormat format; format = GST_FORMAT_TIME; gst_element_query_duration(pipeline,&format,&dur_ns); gst_element_query_position(pipeline,&format,&pos_ns); g_print ("Time: %" GST_TIME_FORMAT " / %" GST_TIME_FORMAT "\n", GST_TIME_ARGS (pos_ns), GST_TIME_ARGS (dur_ns)); seek_ns = pos_ns + 60*GST_SECOND; if (!gst_element_seek (pipeline, 1.0, GST_FORMAT_TIME, GST_SEEK_FLAG_FLUSH | GST