#include <gst/gst.h>
int main (int argc, char * argv[]) {

	if (argc != 3) {
		g_print ("Syntax error\n");
		return -1;
	}

	GstElement *pipeline, *src, *dst;
	/* Сюда будет читаться результат попытки запуска потока. */
	GstStateChangeReturn ret;
	/* bus - это шина конвейера. Через нее мы можем получать сообщения о событиях. */
	GstBus *bus;
	GstMessage *msg;

	/* Инициализация GStreamer */
	gst_init (NULL, NULL);

	/* Создаем элементы */
	pipeline = gst_element_factory_make ("pipeline", "pipe");
	src = gst_element_factory_make ("filesrc", "src");
	dst = gst_element_factory_make ("filesink", "dst");
	if ( !pipeline || !src || !dst ) {
		g_printerr ("Unable to create some elements\n");
		return -1;
	}

	/* Добавляем элементы в конвейер */
	gst_bin_add_many (GST_BIN(pipeline), src, dst, NULL);

	/* И связываем их */
	if ( gst_element_link (src, dst) != TRUE ) 	{
		g_printerr ("Elements can not be linked\n");
		gst_object_unref (pipeline);
		return -1;
	}

	/* Задаем элементам свойства */
	g_object_set (src, "location", argv[1], NULL);
	g_object_set (dst, "location", argv[2], NULL);

	/* Запускаем конвейер */
	ret = gst_element_set_state (pipeline, GST_STATE_PLAYING);
	if ( ret == GST_STATE_CHANGE_FAILURE ) {
		g_printerr ("Unable to set pipeline to the playing state\n");
		gst_object_unref (pipeline);
		return -1;
	}

	/* Мало просто установить режим PLAYING. Нужно ждать либо конца потока, либо 
	 * ошибок. Для начала подключаемся к шине конвейера (эти манипуляции будут 
	 * описаны в следующей статье) */
	bus = gst_element_get_bus (pipeline);

	/* И ожидаем события на шине. Когда событие произойдет, функция вернет 
	 * сообщение, которое мы будем парсить. */
	msg = gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE, GST_MESSAGE_ERROR | GST_MESSAGE_EOS);

	/* Парсим сообщение */
	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 ("We reach End-Of-Stream\n");
				break;

			default:
				g_printerr ("Unexpected message received\n");
				break;
		}
		gst_message_unref (msg);
	}

	/* Освобождаем ресурсы */
	gst_object_unref (bus);
	gst_element_set_state (pipeline, GST_STATE_NULL);
	gst_object_unref (pipeline);

	return 0;
}

pipeline = gst_pipeline_new ("my-pipeline");

bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
gst_bus_add_signal_watch (bus);
g_signal_connect (bus, "message", (GCallback) cb_message,
  pipeline);

src = gst_element_factory_make ("uridecodebin", "src");
if (src == NULL)
g_error ("Could not create 'uridecodebin' element");

g_object_set (src, "uri", argv[1], NULL);

csp = gst_element_factory_make ("videoconvert", "csp");
if (csp == NULL)
g_error ("Could not create 'videoconvert' element");

vs = gst_element_factory_make ("videoscale", "vs");
if (csp == NULL)
g_error ("Could not create 'videoscale' element");

sink = gst_element_factory_make ("autovideosink", "sink");
if (sink == NULL)
g_error ("Could not create 'autovideosink' element");

gst_bin_add_many (GST_BIN (pipeline), src, csp, vs, sink, NULL);

guint8 *data;
gsize size;

gst-launch-1.0 tcpserversrc port=9999 ! decodebin ! alsasink
gst-launch-1.0 filesrc location=/home/yury-timofeev/samples/1.mp3 ! tcpclientsink port=9999

gst-launch-1.0 tcpserversrc port=9999 ! decodebin ! audiomixer name=mix ! alsasink tcpserversrc port=9998 ! decodebin ! mix.
gst-launch-1.0 filesrc location=/home/yury-timofeev/samples/1.mp3 ! tcpclientsink port=9999
gst-launch-1.0 filesrc location=/home/yury-timofeev/samples/2.mp3 ! tcpclientsink port=9998

gst-launch-1.0 tcpserversrc port=9999 ! audio/x-raw,format=S8,channels=2,rate=48000 ! audioconvert ! alsasink

gst-launch-1.0 filesrc location=/home/yury-timofeev/samples/1.mp3 ! decodebin ! audioconvert ! audioresample ! audio/x-raw,format=S8,channels=2,rate=48000 ! tcpclientsink port=9999

GstElement *sink = gst_bin_get_by_name(GST_BIN(pipeline), "dest");
g_signal_emit_by_name(sink, "add", "192.168.1.25", 5004, NULL);
g_object_unref(sink);

#include <gobject/gsignal.h>
...
GstElement *pipeline = gst_pipeline_new("audio-sender");
GstElement *source = gst_element_factory_make("filesrc", "file-source");
GstElement *decodebin = gst_element_factory_make("decodebin", "decodebin");
GstElement *alsasink = gst_element_factory_make("alsasink", "alsasink");

GstBus* bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline));
guint bus_watch_id = gst_bus_add_watch(bus, bus_call, loop);
gst_object_unref(bus);

gst_bin_add_many(GST_BIN(pipeline), source, decodebin, alsasink, nullptr); // в бин добавляем все 3 элемента
gst_element_link(source, decodebin);  // связываем только первые 2 элемента: filesrc ! decodebin

g_signal_connect(decodebin, "pad-added", G_CALLBACK(on_pad_added), alsasink);	// на decodebin привязываемся к сигналу pad-added, который будет вызван, когда у decodebin появился pad, который sink (в начале работы такого пада у него нет)

static void on_pad_added (GstElement *element,
              GstPad     *pad,
              gpointer    data)
{
    GstPad *sinkpad;
    GstElement *alsasink = (GstElement *) data;

    sinkpad = gst_element_get_static_pad (alsasink, "sink");	// берем пад, равный синку (то есть входной) у элемента alsasink

    gst_pad_link (pad, sinkpad);	// связываем наш пад (т.е. decodebin) с синком алсасинка

    gst_object_unref (sinkpad);
}

gst-launch-1.0 tcpserversrc port=9999 ! audio/x-raw,format=S8,channels=2,rate=48000 ! audioconvert ! alsasink

gst-launch-1.0 tcpserversrc port=9999 ! capsfilter caps=audio/x-raw,format=S8,channels=2,rate=48000 ! audioconvert ! alsasink

g_print("Pipeline contains the following elements:\n");
GstIterator *it;
GValue elem = G_VALUE_INIT;
it = gst_bin_iterate_elements(GST_BIN(pipeline));
while (gst_iterator_next(it, &elem) == GST_ITERATOR_OK)
{
    g_print("%s\n",
        gst_element_get_name(g_value_get_object(&elem)));
    g_value_reset(&elem);
}

g_value_unset(&elem);
gst_iterator_free(it);

#define GET_BUFFER_LEVEL_FOR_QUANTITY(format,alt_max) \
    normalize_to_buffering_level (queue->cur_level.format,queue->max_level.format,(alt_max))

/* figure out the buffering level we are filled, we take the max of all formats. */
if (!QUEUE_IS_USING_RING_BUFFER (queue)) {
  buflevel = GET_BUFFER_LEVEL_FOR_QUANTITY (bytes, 0);
} else {
  guint64 rb_size = queue->ring_buffer_max_size;
  buflevel = GET_BUFFER_LEVEL_FOR_QUANTITY (bytes, rb_size);
}

buflevel2 = GET_BUFFER_LEVEL_FOR_QUANTITY (time, 0);
buflevel = MAX (buflevel, buflevel2);

buflevel2 = GET_BUFFER_LEVEL_FOR_QUANTITY (buffers, 0);
buflevel = MAX (buflevel, buflevel2);

/* also apply the rate estimate when we need to */
if (queue->use_rate_estimate) {
  buflevel2 = GET_BUFFER_LEVEL_FOR_QUANTITY (rate_time, 0);
  buflevel = MAX (buflevel, buflevel2);
}

teepad = gst_element_get_static_pad(tee, "src_<номер>");
sinkpad = gst_element_get_static_pad (queue, "sink");
gst_pad_unlink (teepad, sinkpad);
gst_object_unref (sinkpad);

gst_element_set_state (sink, GST_STATE_NULL);
gst_element_set_state (conv, GST_STATE_NULL);
gst_element_set_state (queue, GST_STATE_NULL);

gst_bin_remove (GST_BIN (pipeline), queue);
gst_bin_remove (GST_BIN (pipeline), conv);
gst_bin_remove (GST_BIN (pipeline), sink);

gst_object_unref (sink);
gst_object_unref (conv);
gst_object_unref (queue);

gst_element_release_request_pad (tee, teepad);
gst_object_unref (teepad);

return GST_PAD_PROBE_REMOVE;