problem of sending video through rtp protocol

[xuxin04072129]

hi all
   
    I use rtp protocol to send a movie. On the "sender" when  I set the value of "async" property of udpsink true, the "receiver" can't receive any packets. Instead,when i set "async" false ,the receiver can receiver can receive the rtp packets and play the movie ,but the audio and the vedio are asynchronous, and the video is faster than normal.
   
    Any suggestion? What could be the problem?
   
 Regards

 xuxin
 
 the following is my code
 receiver:\
     gst-launch -v gstrtpbin name=rtpbin latency=200 \
    udpsrc caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)   H264" port=5000 ! rtpbin.recv_rtp_sink_0 \
rtpbin. ! rtph264depay ! decodebin ! xvimagesink \
udpsrc caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA" port=5002 ! rtpbin.recv_rtp_sink_1 \
rtpbin. ! rtppcmadepay ! decodebin ! audioconvert ! audioresample ! alsasink


sender :
#include <gst/gst.h>
#include <glib.h>
#include <unistd.h>
#include <stdlib.h>

#define SOURCE  "/home/xuxin/desktop/g_p/test/a.avi"
#define DEST_HOST  "172.21.29.169"
#define VIDEO_PORT 5000
#define AUDIO_PORT 5002

// global declaration
GstElement *pipeline,*udpsink0,*udpsink1;
GstElement *audio_bin,*video_bin;

//
gboolean res;//element link
GstPadLinkReturn lres;//pad link

//bus
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;
}

static void on_pad_added_rtpbin(GstElement *element, GstPad *pad, gpointer data)
{
    GstCaps *caps=NULL;
    GstStructure *str=NULL;
    GstPad *t_pad=NULL;
   
    const gchar* media_type=NULL;
    gint clock_rate;
    gboolean judgment;
    const gchar* encoding_name=NULL;
    const gchar* name=NULL;
    g_print("**************enter into rtpbin callback ****************\n");
    g_assert(pad!=NULL);
    caps=gst_pad_get_caps(pad);
    g_assert(caps!=NULL);
    str=gst_caps_get_structure(caps,0);
    g_assert(str!=NULL);
   
    media_type=gst_structure_get_string(str,"media");
    g_assert(media_type!=NULL);
    g_print("the media type is %s\n",media_type);
    //**************test*****************************
    name=gst_structure_get_name(str);
    g_print("the caps is %s\n",name);
    judgment=gst_structure_get_int(str,"clock-rate",&clock_rate);
    g_print("clock_rate is :%d\n",clock_rate);
    encoding_name=gst_structure_get_string(str,"encoding-name");
    g_assert(encoding_name!=NULL);
    g_print("the encoding_name is %s\n",encoding_name);
    //***********************************************
   
   
   
    //media type audio or video
         if (g_strrstr (media_type, "video"))
         {            
             g_print("rtp catch audio\n");
            t_pad = gst_element_get_static_pad (udpsink0, "sink");
            g_assert(t_pad!=NULL);      
            g_print("get the pad of udpsink0\n");     
            lres=gst_pad_link (pad, t_pad);
          g_assert (lres == GST_PAD_LINK_OK);
          g_print("Dynamic pad created, linking rtpbin/udp1\n");
         }
         else if(g_strrstr (media_type, "audio"))
      {         
          g_print("rtp catch video\n");
            t_pad = gst_element_get_static_pad (udpsink1, "sink");
            g_assert(t_pad!=NULL);  
            g_print("get the pad of udpsink1\n");
            lres=gst_pad_link (pad, t_pad);
          g_assert (lres == GST_PAD_LINK_OK);
          g_print("Dynamic pad created, linking rtpbin/udp0\n");     
             
        }
        else
        {
            gst_caps_unref (caps);
            return;
        }
   
}

static void on_pad_added_decodebin(GstElement *decodebin,GstPad *pad,gboolean last,gpointer data)
{
    GstCaps *caps=NULL;
      GstStructure *str=NULL;
      GstPad *t_pad=NULL;
   
    const gchar* media_type=NULL;
    g_print("**************** enter into the decodebin signal callback **********\n");
   
    g_assert(pad!=NULL);
   
    caps = gst_pad_get_caps (pad);
    g_assert(caps!=NULL);
      str = gst_caps_get_structure (caps, 0);
      g_assert(str!=NULL);
     
      media_type=gst_structure_get_name (str);
      g_assert(media_type!=NULL);
    g_print("the media type is %s\n",media_type);
      //check media type audio or video
         if (g_strrstr (media_type, "audio"))
         {            
             g_print("catch audio in decodebin\n");
            t_pad = gst_element_get_static_pad (audio_bin, "sink");
            g_assert(t_pad!=NULL);
            lres=gst_pad_link (pad, t_pad);
          g_assert (lres == GST_PAD_LINK_OK);
          g_print("Dynamic pad created, linking decoderbin/queue2\n");
           
             
         }
         else if(g_strrstr (media_type, "video"))
      {         
          g_print("catch video in decodebin\n");
            t_pad = gst_element_get_static_pad (video_bin, "sink");
            g_assert(t_pad!=NULL);
            lres=gst_pad_link (pad, t_pad);
          g_assert (lres == GST_PAD_LINK_OK);
          g_print("Dynamic pad created, linking decoderbin/queue1\n");
           
             
        }
        else
        {
            g_print("have not get suitable type\n");
            gst_caps_unref (caps);
            return;
        }
   
}

int main(int argc, char **argv)
{
    GMainLoop *loop;
    GstBus *bus;
   
    //pad of rtpbin
    GstPad *sink_pad;   
    GstPad *bin_pad;   
   
    GstPad *videopad_src,*videopad_sink;
    GstPad *audiopad_src,*audiopad_sink;
    //declare elements
    GstElement  *filesrc,*decodebin,*gstrtpbin;
    //element for video
    GstElement *queue1,*x264enc,*rtph264pay;
    //element for audio
    GstElement *queue2,*audioresample,*audioconvert,*alawenc,*rtppcmapay;
   
    gst_init(&argc, &argv);
    loop = g_main_loop_new(NULL, FALSE);
   
    //generate elements
    pipeline = gst_pipeline_new("server");
    filesrc=gst_element_factory_make("filesrc", "filesrc");
    decodebin=gst_element_factory_make("decodebin", "decodebin");
    gstrtpbin = gst_element_factory_make("gstrtpbin", "gstrtpbin");
    //for video
    queue1=gst_element_factory_make("queue", "queue1");
    x264enc=gst_element_factory_make("x264enc", "x264enc");
    rtph264pay=gst_element_factory_make("rtph264pay", "rtph264pay");
    udpsink0=gst_element_factory_make("udpsink", "udpsink0");
    //for audio
    queue2 = gst_element_factory_make("queue", "queue2");
    audioresample = gst_element_factory_make("audioresample", "audioresample");
    audioconvert = gst_element_factory_make("audioconvert", "audioconvert");
    alawenc = gst_element_factory_make("alawenc", "alawenc");   
    rtppcmapay = gst_element_factory_make("rtppcmapay", "rtppcmapay");
    udpsink1=gst_element_factory_make("udpsink", "udpsink1");
   
    bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline));
    gst_bus_add_watch(bus, bus_call, loop);
    gst_object_unref(bus);
   
    //set properties of elements
    g_object_set(G_OBJECT(filesrc), "location",SOURCE, NULL);   
    g_object_set(G_OBJECT(udpsink0), "host",DEST_HOST,"port",VIDEO_PORT, "sync",FALSE,
            "async", FALSE,NULL);
    g_object_set(G_OBJECT(udpsink1), "host",DEST_HOST,"port",AUDIO_PORT, "sync",FALSE,
            "async", FALSE,NULL);
    //g_object_set(G_OBJECT(udpsink0),  NULL);
    //g_object_set(G_OBJECT(udpsink1), "host",DEST_HOST, NULL);
    //g_object_set(G_OBJECT(udpsink0), "sync",FALSE, NULL);
    //g_object_set(G_OBJECT(udpsink1), "sync",FALSE, NULL);
   
    //create video_bin
    video_bin = gst_bin_new ("videobin");
    gst_bin_add_many (GST_BIN (video_bin),queue1,x264enc,rtph264pay,NULL);
    res=gst_element_link_many(queue1,x264enc,rtph264pay,NULL);
    g_assert (res == TRUE);
    //add pad to video_bin
    videopad_sink = gst_element_get_static_pad (queue1, "sink");
    videopad_src= gst_element_get_static_pad (rtph264pay, "src");   
    gst_element_add_pad (video_bin,gst_ghost_pad_new ("sink", videopad_sink));
    gst_element_add_pad (video_bin,gst_ghost_pad_new ("src",videopad_src));
      gst_object_unref (videopad_sink);
      gst_object_unref (videopad_src);
     
      //create audio_bin
      audio_bin = gst_bin_new ("audiobin");
    gst_bin_add_many (GST_BIN (audio_bin),queue2,audioresample,audioconvert,alawenc,rtppcmapay,NULL);
    res=gst_element_link_many(queue2,audioresample,audioconvert,alawenc,rtppcmapay,NULL);
    g_assert (res == TRUE);
    //add pad to audio_bin
    audiopad_sink = gst_element_get_static_pad (queue2, "sink");
    audiopad_src= gst_element_get_static_pad (rtppcmapay, "src");   
    res=gst_element_add_pad (audio_bin,gst_ghost_pad_new ("sink", audiopad_sink));
    g_assert (res == TRUE);
    res=gst_element_add_pad (audio_bin,gst_ghost_pad_new ("src", audiopad_src));
    g_assert (res == TRUE);
      gst_object_unref (audiopad_sink);
      gst_object_unref (audiopad_src);
   
    //add elements into pipeline
    gst_bin_add_many(GST_BIN(pipeline),
            filesrc,decodebin,audio_bin,video_bin,gstrtpbin,udpsink0,udpsink1,NULL);
   
    //static link filesrc and decoderbin
    res=gst_element_link(filesrc,decodebin);   
    g_assert (res == TRUE);   
   
    //get request pad from gstrtpbin and connect with video_bin
    bin_pad=gst_element_get_pad(video_bin, "src");
    g_assert(bin_pad!=NULL);   
    sink_pad = gst_element_get_request_pad(gstrtpbin, "recv_rtp_sink_0");
    g_assert(sink_pad!=NULL);   
    lres=gst_pad_link(bin_pad, sink_pad);
    g_assert (lres == GST_PAD_LINK_OK);
   
   
    //get request pad from gstrtpbin and connect with audio_bin
    bin_pad=gst_element_get_pad(audio_bin, "src");
    g_assert(bin_pad!=NULL);
    sink_pad = gst_element_get_request_pad(gstrtpbin, "recv_rtp_sink_1");
    g_assert(sink_pad!=NULL);
    lres=gst_pad_link(bin_pad, sink_pad);
    g_assert (lres == GST_PAD_LINK_OK);

    //signal link
    g_signal_connect(decodebin, "new-decoded-pad", G_CALLBACK(on_pad_added_decodebin),NULL);   
   
    g_signal_connect(gstrtpbin, "pad-added", G_CALLBACK(on_pad_added_rtpbin),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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