AW: AW: AW: Gstreamer output to a buffer then file On Jetson tegra Tx2

[aasim]

*a) *For audio i am not adding any PTS values only for video i have
incremented PTS values (appsrc filling encoded data ).

*b) *audio start at same time but video is captured + encoded by different
apis and then filled in appsrc and audio is captured and muxed by gstreamer 

*c) cmd*
sprintf(launch_string_,
			"appsrc name=mysource !
video/x-h264,width=%d,height=%d,stream-format=byte-stream !",
			STREAM_SIZE.width(), STREAM_SIZE.height());
	sprintf(launch_string_ + strlen(launch_string_),
	           " h264parse ! flvmux name=mux alsasrc device=plughw:2 !
audioresample ! audio/x-raw,rate=48000,channels=1 ! queue ! voaacenc
bitrate=32000 ! queue ! mux. mux.  ! queue ! "

*d)* audio* sampling rate 48000*

*e)* i am* unable to use mp4  muxer *as its giving problem(not going in
audio function so no output file write ) 
*
f) *After getting muxed output in local buffer i write in to file as .mp4
but muxer *i am using is flv *as mp4 muxer is not working.




For audio code just capturing from gstreamer and muxing with appsrcc . here
audio is playing properly but video is fast forward.
if i remove incremented PTS for video audio also plays in fast forward way.

Question 
How to sync video and audio when video is captured  separately 

*This is audio code *





/* called when the appsink notifies us that there is a new buffer ready for
 * processing */
static void  on_new_sample_from_sink (GstElement * elt)
{
	guint size;
	gpointer raw_buffer;
	GstBuffer *app_buffer, *buffer;
	GstElement *source;
	GstMapInfo map = {0};
	GstSample *sample;
	static GstClockTime timestamp;

	/* get the buffer from appsink */
	g_signal_emit_by_name (sink, "pull-sample", &sample,NULL);
	if(sample)
	{

		buffer = gst_sample_get_buffer (sample);
		gst_buffer_map (buffer, &map, GST_MAP_READ);
		raw_buffer = g_malloc0 (map.size);
		memcpy (raw_buffer, map.data, map.size);
		printf("\n output sample= %ld \n",map.size);
		m_outputFile1->write((char *) raw_buffer,map.size);

		gst_buffer_unmap (buffer,&map);
		gst_sample_unref(sample);
	}
} 


*main function*


static bool execute()
{
	GMainLoop *main_loop;
	GstPipeline *gst_pipeline = NULL;
	GError *err = NULL;
	GstElement *appsrc_;

	gst_init (0, NULL);
	main_loop = g_main_loop_new (NULL, FALSE);
	char launch_string_[1024];

	sprintf(launch_string_,
			"appsrc name=mysource !
video/x-h264,width=%d,height=%d,stream-format=byte-stream !",
			STREAM_SIZE.width(), STREAM_SIZE.height());
	sprintf(launch_string_ + strlen(launch_string_),
	           " h264parse ! flvmux name=mux alsasrc device=plughw:2 !
audioresample ! audio/x-raw,rate=48000,channels=1 ! queue ! voaacenc
bitrate=32000 ! queue ! mux. mux.  ! queue ! "
			"appsink name=sink ");
//	sprintf(launch_string_ + strlen(launch_string_),
//			" h264parse !  avmux_mp4 name=mux alsasrc device=plughw:2 !
audioresample ! audio/x-raw,rate=48000,channels=1 ! queue ! voaacenc
bitrate=32000 ! queue ! mux. mux.  ! queue ! "
//			"appsink name=sink ");
			// " h264parse ! qtmux ! filesink location=a.mp4 ");

	printf("\n cmd of gstremer = %s \n",launch_string_);

	gst_pipeline = (GstPipeline*)gst_parse_launch(launch_string_, &err);
	appsrc_ = gst_bin_get_by_name(GST_BIN(gst_pipeline), "mysource");

	/* get sink */
	sink = gst_bin_get_by_name (GST_BIN (gst_pipeline), "sink");


	g_object_set(G_OBJECT(sink), "emit-signals", TRUE, "sync", FALSE, NULL);
	g_signal_connect(sink, "new-sample", G_CALLBACK(on_new_sample_from_sink),
NULL);


	gst_app_src_set_stream_type(GST_APP_SRC(appsrc_),
GST_APP_STREAM_TYPE_STREAM);
	gst_element_set_state((GstElement*)gst_pipeline, GST_STATE_PLAYING);



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