Video and Audio streaming

[clust3r]

Hi all, I'm a n00b in Gstreamer devel..
I wrote a very simple code that creates different pipelines in order to
trasmit audio and video from source video and audio.
now I need to merge and syncronize audio and video ,  in 1 pipeline, but I
have no idea, how to do this.... please help..!! :-)

<code>
#include <gst/gst.h>
#include <stdbool.h>
#include <stdio.h>
#include <string.h>
#include <stdlib.h>
 
static GMainLoop *loop;
 
static gboolean bus_call(GstBus *bus, GstMessage *msg, void *user_data)
{
	switch (GST_MESSAGE_TYPE(msg)) {
	case GST_MESSAGE_EOS: {
		g_message("End-of-stream");
		g_main_loop_quit(loop);
		break;
	}
	case GST_MESSAGE_ERROR: {
		GError *err;
		gst_message_parse_error(msg, &err, NULL);
		g_error("%s", err->message);
		g_error_free(err);
 
		g_main_loop_quit(loop);
		break;
	}
	default:
		break;
	}
 
	return true;
}
 
static void play_uri(const char *uri)
{
	GstElement *pipeline;
	GstBus *bus;
 
	loop = g_main_loop_new(NULL, FALSE);
	pipeline = gst_element_factory_make("playbin", "player");
 
	if (uri)
		g_object_set(G_OBJECT(pipeline), "uri", uri, NULL);
 
	bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline));
	gst_bus_add_watch(bus, bus_call, NULL);
	gst_object_unref(bus);
 
	gst_element_set_state(GST_ELEMENT(pipeline), GST_STATE_PLAYING);
 
	g_main_loop_run(loop);
 
	gst_element_set_state(GST_ELEMENT(pipeline), GST_STATE_NULL);
	gst_object_unref(GST_OBJECT(pipeline));
}

static void stream (void)
{
	GstElement *video_pipeline, *video_source, *video_encoder, *video_payload,
*video_sink;
	GstElement *audio_pipeline_tx, *audio_source, *audio_encoder,
*audio_payload, *audio_tx;
	GstElement *audio_pipeline_rx, *audio_rx, *audio_depayload, *audio_decoder,
*audio_sink;
	GstBus *video_bus;
	GstBus *audio_tx_bus, *audio_rx_bus;
	GstCaps *rx_caps;
 
	loop = g_main_loop_new(NULL, FALSE);
	
	/*
	 * VIDEO PIPELINE
	 */
	video_pipeline = gst_pipeline_new("test-video-stream");
	g_assert (video_pipeline);
	
	video_source = gst_element_factory_make("mfw_v4lsrc", "video_source");
	video_encoder = gst_element_factory_make("mfw_vpuencoder",
"video_encoder");
	video_payload = gst_element_factory_make("rtph264pay","video_payload");
	video_sink = gst_element_factory_make("udpsink","video_sink");
	
	g_object_set(G_OBJECT(video_source), "capture-width", 352, NULL);
	g_object_set(G_OBJECT(video_source), "capture-height", 288, NULL);

	g_object_set(G_OBJECT(video_encoder), "codec-type", 2, NULL);	
	g_object_set(G_OBJECT(video_encoder), "width", 352, NULL);
	g_object_set(G_OBJECT(video_encoder), "height", 288, NULL);
	g_object_set(G_OBJECT(video_encoder), "loopback", FALSE, NULL);

	g_object_set(G_OBJECT(video_sink), "host", "192.168.3.140", NULL);
	g_object_set(G_OBJECT(video_sink), "port", 5500, NULL);
	
	
	video_bus = gst_pipeline_get_bus(GST_PIPELINE(video_pipeline));
	gst_bus_add_watch(video_bus, bus_call, NULL);
	gst_object_unref(video_bus);
 
	gst_bin_add_many(GST_BIN(video_pipeline), video_source, video_encoder,
			video_payload, video_sink, NULL);

	gst_element_link_many(video_source, video_encoder, video_payload,
video_sink, NULL);
	
	/*
	 * AUDIO PIPELINE
	 */
	audio_pipeline_tx = gst_pipeline_new("audio_tx");
	g_assert(audio_pipeline_tx);

	audio_source = gst_element_factory_make("alsasrc", "audio_source");
	audio_encoder = gst_element_factory_make("mulawenc", "audio_encoder");
	audio_payload = gst_element_factory_make("rtppcmupay","audio_payload");
	audio_tx = gst_element_factory_make("udpsink","audio_tx");
	
	g_object_set(G_OBJECT(audio_tx), "host", "192.168.3.140", NULL);
	g_object_set(G_OBJECT(audio_tx), "port", 5600, NULL);
	
	audio_tx_bus = gst_pipeline_get_bus(GST_PIPELINE(audio_pipeline_tx));
	gst_bus_add_watch(audio_tx_bus, bus_call, NULL);
	gst_object_unref(audio_tx_bus);
 
	gst_bin_add_many(GST_BIN(audio_pipeline_tx), audio_source, audio_encoder,
			audio_payload, audio_tx, NULL);

	gst_element_link_many(audio_source, audio_encoder, audio_payload, audio_tx,
NULL);
	
	// __________________________________________ //

	audio_pipeline_rx = gst_pipeline_new("audio_rx");
	
	audio_rx = gst_element_factory_make("udpsrc", "audio_receive");
	audio_depayload = gst_element_factory_make("rtppcmudepay",
"audio_depayload");
	audio_decoder = gst_element_factory_make("mulawdec", "audio_decoder");
	audio_sink = gst_element_factory_make("alsasink", "audio_sink");
	
	rx_caps = gst_caps_new_simple("application/x-rtp",
				"media", G_TYPE_STRING, "audio",
				"clock-rate", G_TYPE_INT, 8000,
				"encoding-name", G_TYPE_STRING, "PCMU",
				NULL);

	g_object_set(G_OBJECT(audio_rx), "port", 5600, NULL);
	g_object_set (G_OBJECT (audio_rx), "caps", rx_caps, NULL);
	gst_caps_unref (rx_caps);
	
	audio_rx_bus = gst_pipeline_get_bus(GST_PIPELINE(audio_pipeline_rx));
	gst_bus_add_watch(audio_rx_bus, bus_call, NULL);
	gst_object_unref(audio_rx_bus);
 
	gst_bin_add_many(GST_BIN(audio_pipeline_rx), audio_rx, audio_depayload,
			audio_decoder, audio_sink, NULL);

	gst_element_link_many(audio_rx, audio_depayload, audio_decoder, audio_sink,
NULL);

	/*
	 * TEST
	 */
	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);
	
	///



	/*
	 * PLAY
	 */
	gst_element_set_state(GST_ELEMENT(video_pipeline), GST_STATE_PLAYING);
	gst_element_set_state(GST_ELEMENT(audio_pipeline_tx), GST_STATE_PLAYING);
	gst_element_set_state(GST_ELEMENT(audio_pipeline_rx), GST_STATE_PLAYING);
 
	g_main_loop_run(loop);
 
	gst_element_set_state(GST_ELEMENT(video_pipeline), GST_STATE_NULL);
	gst_object_unref(GST_OBJECT(video_pipeline));
	gst_element_set_state(GST_ELEMENT(audio_pipeline_tx), GST_STATE_NULL);
	gst_object_unref(GST_OBJECT(audio_pipeline_tx));
	gst_element_set_state(GST_ELEMENT(audio_pipeline_rx), GST_STATE_NULL);
	gst_object_unref(GST_OBJECT(audio_pipeline_rx));

}
 
int main(int argc, char *argv[])
{
	gst_init(&argc, &argv);
	stream();
	//play_uri(argv[1]);
	return 0;
}
</code>


--
View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Video-and-Audio-streaming-tp4410471p4410471.html
Sent from the GStreamer-devel mailing list archive at Nabble.com.