[gst-devel] Problems with deinterleave plugin

[Alexey Chernov]

Hello,

I'm working on a sound editor based on GStreamer and I faced the problem with 
deinterleave plugin recently.

To load and decode file I use pipeline: filesrc, decodebin, audioconvert, 
audioresample, deinterleave and then several branches containing queue, 
audioconvert and appsink (for the experiment I changed it to fakesink).

Here's the code of main function:

GMainLoop* _loop;

GstElement* _pipeline;

void load_file(const char* filename)
{
	GstElement *source, *decodebin, *audio_convert, *audio_resample, *deint;
	GstBus* bus;



	if (_loop)
	{
		g_main_loop_quit(_loop);
		g_main_loop_unref(_loop);
	}
	_loop = g_main_loop_new (NULL, FALSE);

	/* Create gstreamer elements */
	if (_pipeline)
	{
		gst_element_set_state (_pipeline, GST_STATE_NULL);
		gst_object_unref (GST_OBJECT (_pipeline));
		_pipeline=0;
	}
	_pipeline      = gst_pipeline_new("decode_to_app");
	source        = gst_element_factory_make("filesrc",  "filesrc");
	decodebin     = gst_element_factory_make("decodebin",  "decode_bin");
	audio_convert = gst_element_factory_make("audioconvert","audio-convert");
	audio_resample = gst_element_factory_make("audioresample","audio-resample");
	deint		   = gst_element_factory_make("deinterleave", "deint");


	if (!_pipeline || !source || !decodebin || !audio_convert || !deint)
	{
		std::cerr<<"Elements could not be created. Exiting."<<std::endl;
	}

	/* Set up the pipeline */

	/* we set the properties to the source element to receive only rtp packets*/
	g_object_set(G_OBJECT (source), "location", filename, NULL);

	/* we add a message handler */
	bus = gst_pipeline_get_bus (GST_PIPELINE (_pipeline));
	
	gst_bus_add_watch (bus, bus_call, this);
	gst_object_unref (bus);

	/* we add all elements into the pipeline */
	gst_bin_add_many (GST_BIN (_pipeline), source, decodebin, audio_convert, 
audio_resample, deint, NULL);

	/* we link all the elements together */
	link_two_elements(source, decodebin);
	link_two_elements(audio_resample, audio_convert);
	link_two_elements(audio_convert,deint);
	
	g_signal_connect (decodebin, "new-decoded-pad", G_CALLBACK (cb_new_pad), 
audio_resample);
	g_signal_connect (deint, "pad-added", G_CALLBACK (il_new_pad), 0);

	/* Set the pipeline to "playing" state*/
	gst_element_set_state (_pipeline, GST_STATE_PLAYING);

	/* Iterate */
	g_print ("Running...\n");
	g_main_loop_run (_loop);

	/* Out of the main loop, clean up nicely */
	g_print ("Returned, stopping listening\n");
	gst_element_set_state (_pipeline, GST_STATE_NULL);
	
	g_print ("Deleting pipeline\n");
	gst_object_unref (GST_OBJECT (_pipeline));
}

Here's il_new_pad implementation:

int _channels=0;

void il_new_pad (GstElement *decodebin, GstPad *pad, gpointer data)
{
	GstElement* element=0;
	if (_pipeline)
	{
		GstElement *queue, *aconv, *ares, *appsink;

		char* num=itoa(_channels,num,10);

		char* name="queue";
		strcat(name,num);
		queue = gst_element_factory_make("queue",  name);

		char* name="aconv";
		strcat(name,num);
		aconv = gst_element_factory_make("audioconvert", name);

		char* name="sink";
		strcat(name,num);
		appsink = gst_element_factory_make("fakesink", name);		
		
		gst_bin_add_many (GST_BIN (_pipeline), queue, aconv, appsink, NULL);

		link_two_elements(queue, aconv);
		link_two_elements(aconv,appsink);
	
		g_object_set(G_OBJECT (appsink), "sync", FALSE, NULL);
		
		element=queue;

		++_channels;
	}

	GstCaps *caps;
	GstStructure *str;
	GstPad *audiopad;

	/* only link once */
	audiopad = gst_element_get_static_pad (element, "sink");
	if (GST_PAD_IS_LINKED (audiopad))
	{
		g_object_unref (audiopad);
	}

	/* check media type */
	caps = gst_pad_get_caps (pad);
	str = gst_caps_get_structure (caps, 0);
	if (!g_strrstr (gst_structure_get_name (str), "audio"))
	{
		std::cerr<<"won't connect!"<<std::endl;
		gst_caps_unref (caps);
		gst_object_unref (audiopad);
	}
	gst_caps_unref (caps);

	/* link'n'play */
	gst_pad_link (pad, audiopad);
}

Everything seem to start OK, il_new_pad procedure works two times (for stereo 
file), but then I've got the following messages in console:

0:00:01.703963841  5174      0x1d712a0 INFO             GST_EVENT 
gstpad.c:4675:gst_pad_send_event:<queue0:sink> Received event on flushing pad. 
Discarding

0:00:01.703978717  5174      0x1d712a0 INFO             GST_EVENT 
gstpad.c:4675:gst_pad_send_event:<queue1:sink> Received event on flushing pad. 
Discarding

0:00:01.703995479  5174      0x1d712a0 INFO             GST_EVENT 
gstpad.c:4675:gst_pad_send_event:<queue0:sink> Received event on flushing pad. 
Discarding

0:00:01.704007213  5174      0x1d712a0 INFO             GST_EVENT 
gstpad.c:4675:gst_pad_send_event:<queue1:sink> Received event on flushing pad. 
Discarding

0:00:01.704021111  5174      0x1d712a0 INFO             GST_EVENT 
gstpad.c:4675:gst_pad_send_event:<queue0:sink> Received event on flushing pad. 
Discarding

0:00:01.704032565  5174      0x1d712a0 INFO             GST_EVENT 
gstpad.c:4675:gst_pad_send_event:<queue1:sink> Received event on flushing pad. 
Discarding

0:00:01.704047371  5174      0x1d712a0 INFO             GST_EVENT 
gstpad.c:4675:gst_pad_send_event:<queue0:sink> Received event on flushing pad. 
Discarding

0:00:01.704058825  5174      0x1d712a0 INFO             GST_EVENT 
gstpad.c:4675:gst_pad_send_event:<queue1:sink> Received event on flushing pad. 
Discarding

0:00:01.704073143  5174      0x1d712a0 WARN          deinterleave 
deinterleave.c:810:gst_deinterleave_process: gst_pad_alloc_buffer() returned 
wrong-state

0:00:01.704371435  5174      0x1d712a0 WARN          deinterleave 
deinterleave.c:810:gst_deinterleave_process: gst_pad_alloc_buffer() returned 
wrong-state

0:00:01.704564057  5174      0x1d712a0 WARN          deinterleave 
deinterleave.c:810:gst_deinterleave_process: gst_pad_alloc_buffer() returned 
wrong-state

0:00:01.704730419  5174      0x1d712a0 WARN          deinterleave 
deinterleave.c:810:gst_deinterleave_process: gst_pad_alloc_buffer() returned 
wrong-state

0:00:01.704894197  5174      0x1d712a0 WARN          deinterleave 
deinterleave.c:810:gst_deinterleave_process: gst_pad_alloc_buffer() returned 
wrong-state

0:00:01.704937428  5174      0x1d712a0 INFO               basesrc 
gstbasesrc.c:2278:gst_base_src_loop:<filesrc> pausing after gst_pad_push() = 
wrong-state

What was the wrong in my setup? Could you please suggest how can I fix it to 
get the proper behavior (that new branch with appsink (fakesink) is added to 
pipeline when the new channel is recognized).

Thank you very much in advance!

Alexey Chernov