Gstreamer 测试udpsink udpsrc播放mp3文件

发送：send

     gst-launch filesrc location=/root/Media/test.mp3 ! mad ! audioconvert ! audio/x-raw-int,channels=1,depth=16,width=16, rate=44100 ! rtpL16pay  ! udpsink host=127.0.0.1 port=5000

接收：recv

   gst-launch-0.10 udpsrc port=5000 ! "application/x-rtp,media=(string)audio, clock-rate=(int)44100, width=16, height=16,\
 encoding-name=(string)L16, encoding-params=(string)1, channels=(int)1, channel-positions=(int)1, payload=(int)96" ! gstrtpjitterbuffer do-lost=true ! rtpL16depay ! audioconvert ! alsasink sync=false

音乐文件你自己添加一个就可以了，快去试试吧！！！

C语言编程文件会在后续慢慢添加，高手请指点吧！！！不知道可以参考一下！！！

C语言编程终于实现了，我自己也不是很懂原理，如果有不合理的地方欢迎大家指正共同进步：

发送部分：

server-sender.c

#include <string.h>
#include <math.h>

#include <gst/gst.h>
#define DEST_HOST "127.0.0.1"
#define AUDIO_CAPS "audio/x-raw-int,channels=1,depth=16,width=16, rate=44100"

int
main (int argc,char *argv[])
{
	GstElement *source,*maddecoder,*audioconv;
	GstElement *rtpbin,*rtpsink,*rtppay;
	GstElement *pipeline;
	GMainLoop *loop;
	GstCaps *caps;
	GstPad *srcpad,*sinkpad;


	gst_init(&argc,&argv);

	pipeline = gst_pipeline_new(NULL);
	g_assert(pipeline);

	source = gst_element_factory_make("filesrc","source");
	g_assert (pipeline);
	maddecoder=gst_element_factory_make("mad","maddecoder");
	g_assert (maddecoder);
	audioconv=gst_element_factory_make("audioconvert","audioconv");
	g_assert (audioconv);
/*
	caps=gst_caps_new_simple("audio/x-raw-int",
				"channels",G_TYPE_INT,1,
				"depth",G_TYPE_INT,16,
				"width",G_TYPE_INT,15,
				"rate",GST_TYPE_LIST,44100,NULL);
	
*/
	rtppay=gst_element_factory_make("rtpL16pay","rtppay");
	g_assert (rtppay);
	g_object_set(G_OBJECT(source),"location","/root/Media/test.mp3",NULL);

	gst_bin_add_many (GST_BIN (pipeline),source,maddecoder,audioconv,rtppay,NULL);

	caps=gst_caps_from_string(AUDIO_CAPS);
	

	if(!gst_element_link_many(source,maddecoder,audioconv,NULL)){
		g_error("Failed to link ");
	}
	if(!gst_element_link_filtered(audioconv,rtppay,caps))
        {
                 g_error("Failed to link caps");

        }
        gst_caps_unref(caps);
	rtpbin=gst_element_factory_make("gstrtpbin","rtpbin");
	g_assert(rtpbin);

	gst_bin_add(GST_BIN(pipeline),rtpbin);

	rtpsink=gst_element_factory_make ("udpsink","rtpsink");
	g_assert(rtpsink);
	g_object_set(rtpsink,"port",5000,"host","127.0.0.1",NULL);
	gst_bin_add_many(GST_BIN(pipeline),rtpsink,NULL);


	sinkpad = gst_element_get_request_pad (rtpbin, "send_rtp_sink_0");
  	srcpad = gst_element_get_static_pad (rtppay, "src");
  	if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK)
    		g_error ("Failed to link audio payloader to rtpbin");
  	gst_object_unref (srcpad);
	
	srcpad = gst_element_get_static_pad (rtpbin, "send_rtp_src_0");
  	sinkpad = gst_element_get_static_pad (rtpsink, "sink");
  	if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK)
    		g_error ("Failed to link rtpbin to rtpsink");
  	gst_object_unref (srcpad);
  	gst_object_unref (sinkpad);


	g_print("starting sender pipeline\n");
	gst_element_set_state(pipeline,GST_STATE_PLAYING);
	loop=g_main_loop_new(NULL,FALSE);
	g_main_loop_run(loop);
	g_print("stopping sender pipeline\n");
	gst_element_set_state(pipeline,GST_STATE_NULL);
	return 0;

}

接收部分：

client-recv.c

#include <string.h>
#include <math.h>

#include <gst/gst.h>

#define AUDIO_CAPS "application/x-rtp,media=(string)audio, clock-rate=(int)44100, width=16, height=16,encoding-name=(string)L16, encoding-params=(string)1, channels=(int)1, channel-positions=(int)1, payload=(int)96"
//"application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA"

static void
print_source_stats (GObject * source)
{
  GstStructure *stats;
  gchar *str;

  g_return_if_fail (source != NULL);

  /* get the source stats */
  g_object_get (source, "stats", &stats, NULL);

  /* simply dump the stats structure */
  str = gst_structure_to_string (stats);
  g_print ("source stats: %s\n", str);

  gst_structure_free (stats);
  g_free (str);
}
static void
on_ssrc_active_cb (GstElement * rtpbin, guint sessid, guint ssrc,
    GstElement * depay)
{
  GObject *session, *isrc, *osrc;

  g_print ("got RTCP from session %u, SSRC %u\n", sessid, ssrc);

  /* get the right session */
  g_signal_emit_by_name (rtpbin, "get-internal-session", sessid, &session);

  /* get the internal source (the SSRC allocated to us, the receiver */
  g_object_get (session, "internal-source", &isrc, NULL);
  print_source_stats (isrc);

  /* get the remote source that sent us RTCP */
  g_signal_emit_by_name (session, "get-source-by-ssrc", ssrc, &osrc);
  print_source_stats (osrc);
}
static void
pad_added_cb (GstElement * rtpbin, GstPad * new_pad, GstElement * depay)
{
  GstPad *sinkpad;
  GstPadLinkReturn lres;

  g_print ("new payload on pad: %s\n", GST_PAD_NAME (new_pad));

  sinkpad = gst_element_get_static_pad (depay, "sink");
  g_assert (sinkpad);

  lres = gst_pad_link (new_pad, sinkpad);
  g_assert (lres == GST_PAD_LINK_OK);
  gst_object_unref (sinkpad);
}






int 
main (int argc,char *argv[])
{
	GstElement *rtpbin,*rtpsrc,*buffer,*rtppay,*audioconver, *audiosink;
	GstElement *pipeline;
	GMainLoop *loop;
	GstCaps *caps;
	GstPadLinkReturn lres;
	GstPad *srcpad,*sinkpad;

	gst_init(&argc,&argv);
	
	pipeline=gst_pipeline_new(NULL);
	g_assert (pipeline);
	rtpsrc=gst_element_factory_make("udpsrc","rtpsrc");
	g_assert (rtpsrc);
	g_object_set (rtpsrc,"port",5000,NULL);
	caps=gst_caps_from_string(AUDIO_CAPS);
	g_object_set(rtpsrc,"caps",caps,NULL);
	gst_caps_unref(caps);
	
	gst_bin_add_many(GST_BIN (pipeline),rtpsrc,NULL);
	rtppay=gst_element_factory_make("rtpL16depay","rtppay");
	g_assert (rtppay);
	audioconver=gst_element_factory_make("audioconvert","audioconver");
	g_assert (audioconver);

	audiosink=gst_element_factory_make("alsasink","audiosink");
	g_assert (audiosink);


	gst_bin_add_many (GST_BIN(pipeline),rtppay,audioconver,audiosink,NULL);

	gboolean res=gst_element_link_many(rtppay,audioconver,audiosink,NULL);
	g_assert(res==TRUE);
	g_object_set (audiosink, "sync", FALSE, NULL);

	rtpbin=gst_element_factory_make("gstrtpbin","rtpbin");
	g_assert(rtpbin);
	gst_bin_add(GST_BIN(pipeline),rtpbin);
	srcpad = gst_element_get_static_pad (rtpsrc, "src");
  	sinkpad = gst_element_get_request_pad (rtpbin, "recv_rtp_sink_0");
  	lres = gst_pad_link (srcpad, sinkpad);
  	g_assert (lres == GST_PAD_LINK_OK);
  	gst_object_unref (srcpad);

	g_signal_connect (rtpbin, "pad-added", G_CALLBACK (pad_added_cb), rtppay);

  /* give some stats when we receive RTCP */
  //	g_signal_connect (rtpbin, "on-ssrc-active", G_CALLBACK (on_ssrc_active_cb),
 //     rtppay);


	g_print ("starting receiver pipeline\n");
        gst_element_set_state (pipeline, GST_STATE_PLAYING);

	loop = g_main_loop_new (NULL, FALSE);
  	g_main_loop_run (loop);

	g_print ("stopping receiver pipeline\n");
  	gst_element_set_state (pipeline, GST_STATE_NULL);

  	gst_object_unref (pipeline);

  	return 0;
}

希望大家支持，给建议，后面会继续些一下文件视频传输，摄像头camera数据传输，顶起来吧大家！！！！！