Appsink is only apparently processing 25% of data

[Michael Rappaport]

I have a pipeline that is taking in an RTP MPEG-2 TS, and doing a few things:
1) Playing back video and audio
2) Using *filesink* to save the file to disk
3) Using *appsink* to save samples to disk when they are received.

Basically, the program is creating two files, one by stdio, and one by the
filesink. Because I am using a tee, I would expect the appsink file and the
filesink file to be exactly the same. However, the appsink file is not
nearly as big as the filesink file. I've been struggling with this one for
quite some time, and any assistance would be greatly appreciated.

Here is my code:

#include <gst/gst.h>
#include <time.h>
#include <Windows.h>
#include <gst/app/gstappsink.h>
#include <string.h>
#include <stdio.h>
#include <gst/base/gstbasesink.h>
#include <gst/gstelement.h>

FILE *myfile;

typedef struct _CustomData {
	GstElement *pipeline;
	GstElement *source;
	GstElement *appsink;
} CustomData;

GstFlowReturn new_buffer(GstElement *sink) {
	GstSample *sample;
	GstFlowReturn ret = GST_FLOW_OK;
	/* Retrieve the buffer */
	sample = gst_app_sink_pull_sample(GST_APP_SINK(sink));
	//retreive buffer data
	GstMapInfo map;
	gst_buffer_map(gst_sample_get_buffer(sample), &map, GST_MAP_READ);
	//printe buffer data
	fprintf(myfile, "%s", map.data);
	return ret;
}



int main(int argc, char *argv[]) {
	CustomData data;
	//GstElement *pipeline, *textoverlay_element, *myTSsink;
	GstElement *LBR_Decode;
	GstBus *bus;
	GstMessage *msg;
	GstStateChangeReturn ret;
	GMainLoop *main_loop;

	myfile = fopen("app.ts", "w+");

	/* Initialize GStreamer */
	gst_init(&argc, &argv);


	//appsink + file
	data.pipeline = gst_parse_launch("udpsrc uri = udp://239.195.8.27:1234
caps=\"application/x-rtp, media=(string)video, clock-rate=(int)90000,
encoding-name=(string)MP2T-ES, payload=(int)33\" name=myudpsrc ! tee name=t
! queue2 ! rtpjitterbuffer latency=1000 drop-on-latency=false !
application/x-rtp ! rtpmp2tdepay ! video/mpegts ! queue2 ! appsink
name=myTSsink t. ! queue2 ! rtpjitterbuffer latency=1000
drop-on-latency=false ! application/x-rtp ! rtpmp2tdepay ! video/mpegts !
queue2 ! filesink location=file.ts t. ! queue2 ! rtpjitterbuffer
latency=1000 drop-on-latency=false ! application/x-rtp ! rtpmp2tdepay !
video/mpegts ! queue2 ! tsdemux name=demux demux. ! queue2 ! faad !
audioconvert ! audioresample ! autoaudiosink demux. ! queue2 ! decodebin
name=dec ! queue2 ! videoconvert ! autovideosink", NULL);


	//appsink stuff
	data.appsink = gst_bin_get_by_name(GST_BIN(data.pipeline), "myTSsink");
	GstAppSinkCallbacks mycallbacks;
	mycallbacks.eos = 0;
	mycallbacks.new_sample = new_buffer;
	mycallbacks.new_preroll = 0;
	GstCaps *caps = gst_caps_new_simple("video/mpegts", "packetsize",
G_TYPE_INT, 188, "systemstream", G_TYPE_BOOLEAN, TRUE, NULL);
	gst_app_sink_set_caps(GST_APP_SINK(data.appsink), caps);
	gst_app_sink_set_callbacks(GST_APP_SINK(data.appsink), &mycallbacks,
&data.appsink, 0);
	gst_app_sink_set_emit_signals(GST_APP_SINK(data.appsink), TRUE);
	gst_app_sink_set_max_buffers(GST_APP_SINK(data.appsink), 0);
	gst_app_sink_set_drop(GST_APP_SINK(data.appsink), FALSE);



	/* 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;
	}

	/* Wait until error or EOS */
	bus = gst_element_get_bus(data.pipeline);
	msg = gst_bus_timed_pop_filtered(bus, GST_CLOCK_TIME_NONE,
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);
			break;
		case GST_MESSAGE_EOS:
			g_print("End-Of-Stream reached.\n");
			break;
		default:
			/* We should not reach here because we only asked for ERRORs and EOS */
			g_printerr("Unexpected message received.\n");
			break;
		}
		gst_message_unref(msg);
	}

	/* Free resources */
	gst_object_unref(bus);
	gst_element_set_state(data.pipeline, GST_STATE_NULL);
	gst_object_unref(data.pipeline);
	return 0;
}



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