Gstreamer test udpsink udpsrc to play MP3 files

Send: Send

GST-launch filesrc location =/root/Media/test.pdf! Mad! Audioconvert! Audio/X-raw-int, channels = 1, depth = 16, width = 16, rate = 44100! Rtpl16pay! Udpsink host = 127.0.0.1 Port = 5000

Receive: 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

You can add one music file by yourself. Please try it !!!

C language programming files will be added slowly in the future. Please give me some advice !!! For details, refer !!!

C language programming has finally been implemented, and I am not very familiar with the principle. If there is anything unreasonable, you are welcome to correct and make progress together:

Sending part:

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"intmain (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;}

Receiving part:

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 voidprint_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 voidon_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 voidpad_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;}

I hope you can support it. I suggest you continue to transfer files and videos, and transfer the camera data. Come on !!!!!