GstAppSrc Error Stopped Playing. Not Negotiated (-4)

[namakilam7]

Following the code I am using to build a pipeline to stream raw pcm samples
on rtp stream.

static gboolean
		push_buffer2(AppData * app)
	{
		gpointer raw_buffer;
		GstBuffer *app_buffer;
		GstMemory *mem;
		GstFlowReturn ret;

		app->num_frame++;
		
		if (!app->isCallConnected && connectedCalls != 0) {
			return TRUE;
		}
		char *data = app->dataQueue->pop();

		if (data == NULL) {
			data = (char*)malloc(sizeof(char) * 960);
			memset(data, 0, 960);
		}

		app_buffer = gst_buffer_new();

		//mem = gst_allocator_alloc(NULL, BUFFER_SIZE, NULL);
		mem = gst_memory_new_wrapped(GST_MEMORY_FLAG_READONLY, data, 960, 0, 960,
NULL, NULL);
		gst_buffer_append_memory(app_buffer, mem);
		gst_buffer_set_size(app_buffer, BUFFER_SIZE);

		/* Setting the correct timestamp for the buffer is very important,
otherwise the
		* resulting file won't be created correctly */
		GST_BUFFER_TIMESTAMP(app_buffer) = (GstClockTime)((app->num_frame * 0.06)
* 1e9);

		/* push new buffer */
		g_signal_emit_by_name(app->source, "push-buffer", app_buffer, &ret);
		gst_buffer_unref(app_buffer);

		if (ret != GST_FLOW_OK) {
			/* some error, stop sending data */
			return FALSE;
		}

		//return TRUE;

		return TRUE;
	}

	/* This signal callback is called when appsrc needs data, we add an idle
handler
	* to the mainloop to start pushing data into the appsrc */
	static void
		start_feed(GstElement * pipeline, guint size, AppData * app)
	{
		if (app->source_id == 0) {
			//g_print("start feeding at frame %i\n", app->num_frame);
			app->source_id = g_idle_add((GSourceFunc)push_buffer2, app);
		}
	}

	/* This callback is called when appsrc has enough data and we can stop
sending.
	* We remove the idle handler from the mainloop */
	static void
		stop_feed(GstElement * pipeline, AppData * app)
	{
		if (app->source_id != 0) {
			//g_print("stop feeding at frame %i\n", app->num_frame);
			g_source_remove(app->source_id);
			app->source_id = 0;
		}
	}

static void runGstPipeline() { 
GstBus *bus = NULL;
		//GstElement *appsrc = NULL;
		//GstElement *appsrc2 = NULL;
		GstElement *pipeline, *mixer, *ac1, *mulawenc, *rtppcmupay, *udpsink,
*appsrc, *ac2, *ar, *wavenc, *wavparse, *ac3;
		
		GstCaps *caps = gst_caps_new_simple("audio/x-raw",
			"format", G_TYPE_STRING, "S16LE",
			"channels", G_TYPE_INT, 1,
			"rate", G_TYPE_INT, 8000, NULL);
		GstPad *adder_sinkpad1;

		gst_init(NULL, NULL);

		pipeline = gst_pipeline_new("audio-send-pipeline");
		mixer = gst_element_factory_make("adder", "audio-mixer");
		
		ac1 = gst_element_factory_make("audioconvert", "conv1");
		mulawenc = gst_element_factory_make("mulawenc", "mulaw-enc");
		rtppcmupay = gst_element_factory_make("rtppcmupay", "pay");
		udpsink = gst_element_factory_make("udpsink", "sink");
		
		g_object_set(udpsink,
			"host", "127.0.0.1",
			"port", 5000,
			NULL);

		appsrc = gst_element_factory_make("appsrc", "source1");
		gst_app_src_set_caps(GST_APP_SRC(appsrc), caps);
		
		ac2 = gst_element_factory_make("audioconvert", "conv2");
		ar = gst_element_factory_make("audioresample", "resam1");
	
		wavenc = gst_element_factory_make("wavenc", "enc1");
		wavparse = gst_element_factory_make("wavparse", "parse1");

		ac3 = gst_element_factory_make("audioconvert", "conv3");


		if (!pipeline || !mixer || !ac1 || !mulawenc || !rtppcmupay || !udpsink ||
!appsrc || !ac2 || !ar || !wavenc || !wavparse || !ac3) {
			g_printerr("One or more elements could not be created. Exiting\n");
			return;
		}
		
		bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline));
		gst_bin_add_many(GST_BIN(pipeline),
			mixer ,ac1 ,mulawenc ,rtppcmupay ,udpsink ,appsrc ,ac2 ,ar ,wavenc
,wavparse ,ac3, NULL);

		int n = 0;
		if ((n = gst_element_link(mixer, ac1)) == 0) {
			g_print("link error: %d\n", n);
			g_print("cannot link mixer with caps-filter-1\n");
		}

		if ((n = gst_element_link(ac1, mulawenc)) == 0) {
			g_print("link error: %d\n", n);
			g_print("cannot link audio-convert-1 with mulawenc\n");
		}

		if ((n = gst_element_link(mulawenc, udpsink)) == 0) {
			g_print("link error: %d\n", n);
			g_print("cannot link mulawenc with udpsink\n");
		}

		if ((n = gst_element_link(appsrc, ac2)) == 0) {
			g_print("link error: %d\n", n);
			g_print("cannot link appsrc with caps-filter-2\n");
		}

		if ((n = gst_element_link(ac2, ar)) == 0) {
			g_print("link error: %d\n", n);
			g_print("cannot link audio-convert-2 with audio-resample\n");
		}

		if ((n = gst_element_link(ar, wavenc)) == 0) {
			g_print("link error: %d\n", n);
			g_print("cannot link audio-resample with wav-enc\n");
		}

		if ((n = gst_element_link(wavenc, wavparse)) == 0) {
			g_print("link error: %d\n", n);
			g_print("cannot link wav-enc with wav-parse\n");
		}

		if ((n = gst_element_link(wavparse, ac3)) == 0) {
			g_print("link error: %d\n", n);
			g_print("cannot link wav-parse with ac3\n");
		}

		adder_sinkpad1 = gst_element_get_request_pad(mixer, "sink_%u");
		g_print("PadName : %s", gst_pad_get_name(adder_sinkpad1));
		
		GstPad *srcPad = gst_element_get_static_pad(ac3, "src");
		if ((n = gst_pad_link(srcPad, adder_sinkpad1)) != 0) {
			g_print("pad error: %d\n", n);
			g_print("cannot link audio-convert-3 with adder");
		}


		sendAppData = g_new0(AppData, 1);
		sendAppData->pipeline = pipeline;
		sendAppData->loop = g_main_loop_new(NULL, FALSE);

		/* setting up pipeline, we push media data into this pipeline that will
		* then be recorded to a file, encoded with a codec*/
		if (sendAppData->pipeline == NULL) {
			g_print("Bad pipeline\n");
			return;
		}

		sendAppData->source = appsrc;
		sendAppData->dataQueue = &dataQueue;

		/* setting maximum of bytes queued */
		gst_app_src_set_max_bytes((GstAppSrc *)appsrc, QUEUED_FRAMES *
BUFFER_SIZE);

		g_signal_connect(sendAppData->source, "need-data", G_CALLBACK(start_feed),
sendAppData);
		g_signal_connect(sendAppData->source, "enough-data",
G_CALLBACK(stop_feed), sendAppData);

		/* add watch for messages */
		//bus = gst_element_get_bus(sendAppData->pipeline);
		gst_bus_add_watch(bus, (GstBusFunc)on_pipeline_message, sendAppData);
		gst_object_unref(bus);

		/* configure the appsrc, we will push data into the appsrc from the
		* mainloop */
		gst_element_set_state(sendAppData->pipeline, GST_STATE_PLAYING);

		sendAppData->dataQueue = new ThreadSafeQueue<char*>();

		available.push(sendAppData);

		/* this mainloop is stopped when we receive an error or EOS */
		g_print("Creating movie...\n");
		g_main_loop_run(sendAppData->loop);
		g_print("Done.\n");

		gst_element_set_state(sendAppData->pipeline, GST_STATE_NULL);

		/* Cleaning up */
		gst_object_unref(sendAppData->source);
		gst_object_unref(sendAppData->pipeline);
		g_main_loop_unref(sendAppData->loop);
		g_free(sendAppData);
		/* $$$$$$$$$$$$$$$$$$$$$$$$ STOP PIPELINE AND RETURN
$$$$$$$$$$$$$$$$$$$$$$$$ */
}



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