[gst-devel] problem of sending out vediio through rtp protocol

[xuxin04072129]

hi all
   I come across a problem when sending out vedio through rtp protocol.Last time i ask the same question here and Wim Taymans told me that i had connected to the pad-added signal twice.I should thank him first .Later  I re-write the code , but i get error as flows 
       Error: internal data flow error.
I am new to gstreamer , can anyone help me to check my code and give me some suggestions . Thank you very much. the following is my code

/******************************************************
 #include <gst/gst.h>
#include <glib.h>
#include <unistd.h>
#include <stdlib.h>


GstElement *multiudpsink1, *multiudpsink2;

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_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;
}

// I have rewrite this function
static void on_pad_added(GstElement *element, GstPad *pad, gpointer data)
{
    GstCaps *caps;
    GstStructure *str;
    GstPad *t_pad;
    
    caps=gst_pad_get_caps(pad);
    str=gst_caps_get_structure(caps,0);
    
    //
         if (g_strrstr (gst_structure_get_name (str), "audio")) 
         {             
             t_pad = gst_element_get_static_pad (multiudpsink1, "sink");
             g_print("rtp catch audio\n");
         }
         else if(g_strrstr (gst_structure_get_name (str), "vedio"))
      {          
             t_pad = gst_element_get_static_pad (multiudpsink2, "sink");
             g_print("rtp catch vedio\n");
        }
        else
        {
            gst_caps_unref (caps);
            return;
        }
        
        if (GST_PAD_IS_LINKED (t_pad)) 
      {
          gst_caps_unref (caps);
            g_object_unref (t_pad);
            return;
      }
      //
      else
      {
          //
          gst_pad_link (pad, t_pad);
          g_print("Dynamic pad created, linking rtpbin/udp\n");
                    
          gst_caps_unref (caps);          
            g_object_unref (t_pad);
            return;
      }     
}

int main(int argc, char **argv)
{
    GMainLoop *loop;
    GstBus *bus;
    GstPad *pad;
    GstCaps *videocap, *audiocap;
    GstElement *pipeline, *gstrtpbin, *udpsrc1, *udpsrc2;
        
    
    gst_init(&argc, &argv);
    loop = g_main_loop_new(NULL, FALSE);
    
    pipeline = gst_pipeline_new("server");
    gstrtpbin = gst_element_factory_make("gstrtpbin", "gst_rtpbin");
    udpsrc1 = gst_element_factory_make("udpsrc", "udpsrc1");
    udpsrc2 = gst_element_factory_make("udpsrc", "udpsrc2");
    multiudpsink1 = gst_element_factory_make("multiudpsink", "multiudpsink1");
    multiudpsink2 = gst_element_factory_make("multiudpsink", "multiudpsink2");    
    
    
    bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline));
    gst_bus_add_watch(bus, bus_call, loop);
    gst_object_unref(bus);
    
    videocap = gst_caps_new_simple("application/x-rtp", 
        "media", G_TYPE_STRING, "video",
        "clock-rate", G_TYPE_INT, 90000,
        "encoding-name", G_TYPE_STRING, "H264", NULL);
        
    audiocap = gst_caps_new_simple("application/x-rtp",
        "media", G_TYPE_STRING, "audio",
        "clock-rate", G_TYPE_INT, 8000,
        "encoding-name", G_TYPE_STRING, "PCMA", NULL);
    
    g_object_set(G_OBJECT(udpsrc1), "caps", videocap, NULL);
    g_object_set(G_OBJECT(udpsrc2), "caps", audiocap, NULL);
    g_object_set(G_OBJECT(udpsrc1), "port", 5000, NULL);
    g_object_set(G_OBJECT(udpsrc2), "port", 5002, NULL);
    g_object_set(G_OBJECT(multiudpsink1), "clients","172.21.29.169:5000,172.21.29.168:5000", NULL);
    g_object_set(G_OBJECT(multiudpsink2), "clients","172.21.29.169:5002,172.21.29.168:5002", NULL);
    g_object_set(G_OBJECT(multiudpsink1), "sync",FALSE, NULL);
    g_object_set(G_OBJECT(multiudpsink2), "sync",FALSE, NULL);
    
    gst_caps_unref(videocap);
    gst_caps_unref(audiocap);
        
    gst_bin_add_many(GST_BIN(pipeline), udpsrc1, udpsrc2, gstrtpbin, multiudpsink1, multiudpsink2, NULL);
    
    pad = gst_element_get_request_pad(gstrtpbin, "recv_rtp_sink_0");
    gst_pad_link(gst_element_get_pad(udpsrc1, "src"), pad);
    
    pad = gst_element_get_request_pad(gstrtpbin, "recv_rtp_sink_1");
    gst_pad_link(gst_element_get_pad(udpsrc2, "src"), pad);
    
    g_signal_connect(gstrtpbin, "pad-added", G_CALLBACK(on_pad_added),NULL);    
    
    gst_element_set_state(pipeline, GST_STATE_PLAYING);
    
    g_print("Running...\n");
    g_main_loop_run(loop);
    
    /* Out of the main loop, clean up nicely */
    g_print("Returned, stopping playback\n");
    gst_element_set_state(pipeline, GST_STATE_NULL);
    
    g_print("Deleting pipeline\n");
    gst_object_unref(GST_OBJECT(pipeline));
    
    return 0;
}
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