appsrc audio pipe line.

[Haiming Wang]

Hi chmario,
   Below is the pipeline I used which works.  I think maybe you need to add   " audioconvert ".  

mAudioSendPipeline = GstAppSrcUdpSinkPipeline.createInstance("appsrc block=true is-live=true name=app_src  ! audioconvert ! audioresample ! mulawenc ! rtppcmupay ! udpsink name=udp_sink clients=192.168.1.106:50000","audio/x-raw-int,rate=48000,width=16,channels=1,endianness=1234,depth=16,signed=(boolean)true", 1920, "127.0.0.1", 5000);
    	mAudioSendPipeline.startPipeline();
> Date: Tue, 15 Jul 2014 18:05:14 -0700
> From: chmario at hotmail.com
> To: gstreamer-devel at lists.freedesktop.org
> Subject: appsrc audio pipe line.
> 
> I am trying to write an application using appsrc plugin. With the simplest
> pipeline the data flows ok, (appsrc -> autoaudiosink), I am able to get the
> "need-data" signal and push the data buffer, but if  I add elements  between
> appsrc and autoaudiosink for example  (appsrc -> queue -> autoaudiosink) I
> got a bus error form appsrc "Internal data flow error".
> The properties I am setting in appsrc are:
> 	GstCaps *caps = gst_caps_new_full(gst_structure_new ("audio/x-raw-int",
> 							"channels", G_TYPE_INT, 1,
> 							"rate", G_TYPE_INT, 44100,	
> 							"width", G_TYPE_INT, 16,	
> 							"depth", G_TYPE_INT, 16,
> 							"signed", G_TYPE_BOOLEAN, 1,		 				 					 				 
> 							"endianness", G_TYPE_INT, 1234,					 			 
> 							NULL),
> 	      						NULL);
> 	g_object_set (G_OBJECT (streamSrc),  "stream-type",
> GST_APP_STREAM_TYPE_STREAM,
>                                                  "format", GST_FORMAT_TIME,
> NULL);
> 	g_object_set (G_OBJECT (streamSrc), "is-live", true, NULL);
> 
> I would apreciate if some one can help me to figure  out why I am unable to
> add more elements between appsrc and autoaudiosink.
> Eventually I need to change autoaudiosink for some audio streaming element. 
> 
> Regards
> Mario
> 
> 
> 
> -----
> MChC
> --
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