<div dir="ltr"><div><div><div><div><div>Hi, </div>I am writing pipeline in a C code using gstreamer-1.0 which send the audio and receive audio on same port. But I am unable to do that. It is able send the audio but receiving process not working. </div>Please help about this. What is going to wrong in this ? Any help/pointer appericiated ... </div>My code is here : #include <stdio.h> #include <gst/gst.h> #include <gio/gio.h> #include <stdlib.h> #include <sys/socket.h> #include <netinet/in.h> /* Structure to contain all our information, so we can pass it to callbacks */ typedef struct _CustomData { GstElement *pipeline; GstElement *source; GstElement *convert; GstElement *audio_resample; GstElement *audio_encoder; GstElement *audio_rtp; GstElement *audio_sink; GstElement *colorspace; GstElement *video_encoder; GstElement *video_rtp; GstElement *video_sink; GstElement *recvsource; GstElement *recvdepay; GstElement *recvsink; } CustomData; /* Handler for the pad-added signal */ static void pad_added_handler (GstElement *src, GstPad *pad, CustomData *data); int main(int argc, char *argv[]) { CustomData data; GstBus *bus; GstMessage *msg; GstStateChangeReturn ret; gboolean terminate = FALSE; /* Initialize GStreamer */ gst_init (&argc, &argv); /* Create the elements */ data.source = gst_element_factory_make ("uridecodebin", "source"); data.convert = gst_element_factory_make ("audioconvert", "convert"); data.audio_resample = gst_element_factory_make ("audioresample", "resample"); data.audio_encoder = gst_element_factory_make ("mulawenc", "aencoder"); data.audio_rtp = gst_element_factory_make ("rtppcmupay", "artppay"); data.audio_sink = gst_element_factory_make ("udpsink", "audio_sink"); data.colorspace = gst_element_factory_make ("autovideoconvert", "colorspace"); data.video_encoder = gst_element_factory_make ("avenc_h263p", "vencoder"); data.video_rtp = gst_element_factory_make ("rtph263ppay", "video_rtp"); data.video_sink = gst_element_factory_make ("udpsink", "video_sink"); data.recvsource = gst_element_factory_make ("udpsrc", "recvsrc");; data.recvdepay = gst_element_factory_make ("rtppcmudepay", "artdepay");; data.recvsink = gst_element_factory_make ("filesink", "recvsink");; /* Create the empty pipeline */ data.pipeline = gst_pipeline_new ("test-pipeline"); if (!data.pipeline || !data.source || !data.convert || !data.audio_sink || !data.colorspace || !data.video_sink) { g_printerr ("Not all elements could be created.\n"); return -1; } /* Build the pipeline. Note that we are NOT linking the source at this * point. We will do it later. */ gst_bin_add_many (GST_BIN (data.pipeline), data.source, data.convert ,data.audio_resample,data.audio_encoder,data.audio_rtp, data.audio_sink, data.colorspace,data.video_encoder,data.video_rtp, data.video_sink, NULL); if (!(gst_element_link_many (data.convert, data.audio_resample,data.audio_encoder,data.audio_rtp,data.audio_sink,NULL) )) { g_printerr ("audio Elements could not be linked.\n"); gst_object_unref (data.pipeline); return -1; } if (!( gst_element_link_many (data.colorspace,data.video_encoder,data.video_rtp, data.video_sink,NULL))) { g_printerr ("video Elements could not be linked.\n"); gst_object_unref (data.pipeline); return -1; } if (!( gst_element_link_many (data.recvsource,data.recvdepay,data.recvsink,NULL))) { g_printerr ("video Elements could not be linked.\n"); gst_object_unref (data.pipeline); return -1; } struct sockaddr_in artp_addr; memset(&artp_addr, 0, sizeof(struct sockaddr_in)); int artp_sockfd = socket (AF_INET, SOCK_DGRAM, 0); char on =1; setsockopt(artp_sockfd, NULL, SO_REUSEADDR, (const char *) &on, sizeof(on)); perror("setsockopt"); if (artp_sockfd > 0) { int res; artp_addr.sin_family = AF_INET; artp_addr.sin_port = htons(7878); artp_addr.sin_addr.s_addr = inet_addr("192.168.0.227");; res = bind(artp_sockfd, (struct sockaddr*)&artp_addr,sizeof(artp_addr)); if (res == 0) { printf("Succesfully bound to audio local RTP port : 7878 \t sockfd : %d.\n",artp_sockfd); } else { printf("Unable to bind to local audio RTP port 7878."); } } /* Set the URI to play */ g_object_set (data.source, "uri", "file:///home/amar/KRSNA.mpg", NULL); GstCaps *caps; caps =gst_caps_from_string("application/x-rtp,media=(string)audio,encoding-name=PCMU,payload=0,clock-rate=8000"); g_object_set (data.audio_sink, "port", 3333 , NULL); g_object_set (data.audio_sink, "host", "127.0.0.1" , NULL); GSocket * s = g_socket_new_from_fd(artp_sockfd, NULL); g_object_set (data.audio_sink, "socket", s , NULL); g_object_set (data.video_sink, "port", 9078 , NULL); g_object_set (data.video_sink, "host", "127.0.0.1" , NULL); g_object_set (data.recvsource, "caps", caps, NULL); g_object_set (data.recvsource, "socket", s , NULL); g_object_set (data.recvsink, "location", "new.wav", NULL); GstPad *srcpad, *sinkpad; GstPadLinkReturn lres; g_signal_connect (data.source, "pad-added", G_CALLBACK (pad_added_handler), &data); /* Start playing */ ret = gst_element_set_state (data.pipeline, GST_STATE_PLAYING); if (ret == GST_STATE_CHANGE_FAILURE) { g_printerr ("Unable to set the pipeline to the playing state.\n"); gst_object_unref (data.pipeline); return -1; } /* Listen to the bus */ bus = gst_element_get_bus (data.pipeline); do { msg = gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE, GST_MESSAGE_STATE_CHANGED | GST_MESSAGE_ERROR | GST_MESSAGE_EOS); /* Parse message */ if (msg != NULL) { GError *err; gchar *debug_info; switch (GST_MESSAGE_TYPE (msg)) { case GST_MESSAGE_ERROR: gst_message_parse_error (msg, &err, &debug_info); g_printerr ("Error received from element %s: %s\n", GST_OBJECT_NAME (msg->src), err->message); g_printerr ("Debugging information: %s\n", debug_info ? debug_info : "none"); g_clear_error (&err); g_free (debug_info); terminate = TRUE; break; case GST_MESSAGE_EOS: g_print ("End-Of-Stream reached.\n"); terminate = TRUE; break; case GST_MESSAGE_STATE_CHANGED: /* We are only interested in state-changed messages from the pipeline */ if (GST_MESSAGE_SRC (msg) == GST_OBJECT (data.pipeline)) { GstState old_state, new_state, pending_state; gst_message_parse_state_changed (msg, &old_state, &new_state, &pending_state); g_print ("Pipeline state changed from %s to %s:\n", gst_element_state_get_name (old_state), gst_element_state_get_name (new_state)); } break; default: /* We should not reach here */ g_printerr ("Unexpected message received.\n"); break; } gst_message_unref (msg); } } while (!terminate); /* Free resources */ gst_object_unref (bus); gst_element_set_state (data.pipeline, GST_STATE_NULL); gst_object_unref (data.pipeline); return 0; } /* This function will be called by the pad-added signal */ static void pad_added_handler (GstElement *src, GstPad *new_pad, CustomData *data) { GstPad *sink_pad_audio = gst_element_get_static_pad (data->convert, "sink"); GstPad *sink_pad_video = gst_element_get_static_pad (data->colorspace, "sink"); GstPadLinkReturn ret; GstCaps *new_pad_caps = NULL; GstStructure *new_pad_struct = NULL; const gchar *new_pad_type = NULL; g_print ("Received new pad '%s' from '%s':\n", GST_PAD_NAME (new_pad), GST_ELEMENT_NAME (src)); /* Check the new pad's type */ new_pad_caps = gst_pad_query_caps (new_pad,NULL); new_pad_struct = gst_caps_get_structure (new_pad_caps, 0); new_pad_type = gst_structure_get_name (new_pad_struct); if (!g_str_has_prefix (new_pad_type, "audio/x-raw")) { g_print (" It has type '%s' which is raw video. Connecting.\n", new_pad_type); /* Attempt the link */ ret = gst_pad_link (new_pad, sink_pad_video); if (GST_PAD_LINK_FAILED (ret)) { g_print (" Type is '%s' but link failed.\n", new_pad_type); } else { g_print (" Link succeeded (type '%s').\n", new_pad_type); } goto exit; } /* Attempt the link */ ret = gst_pad_link (new_pad, sink_pad_audio); if (GST_PAD_LINK_FAILED (ret)) { g_print (" Type is '%s' but link failed.\n", new_pad_type); } else { g_print (" Link succeeded (type '%s').\n", new_pad_type); } exit: /* Unreference the new pad's caps, if we got them */ if (new_pad_caps != NULL) gst_caps_unref (new_pad_caps); /* Unreference the sink pad */ gst_object_unref (sink_pad_audio); gst_object_unref (sink_pad_video); } </div>Thanks, </div>Amar </div>