error after using the appropriate plugins.

[killerrats]

 -> queue1 -> rtph264depay -> h264parse -> vtee -> (FakesinkBin /
RecBin)
rtspsrc                                                                                                                                    
       -> queue2 -> rtpjitterbuffer -> rtpmp4gdepay -> aacparse -> atee ->
(FakesinkBin / RecBin)

_______
FakesinkBin
_______
queue5 -> fakesink
queue6 -> fakesink

_______
RecBin
_______
queue3->
             avimux -> appsink
queue4 ->

I have successfully loaded each plugin and other dependecies. The problem is
that I get an error. ---- video ---- and ---- audio ---- are in the
pad-added function for the rtspsrc. If i don't use the plugins and let
gstreamer load the plugins it works fine. anyone know why? method below is what I use to connect the audio and video. I use it for

void cb_new_rtspsrc_pad( GstElement* element, GstPad* pad, gpointer data)
{
       
        gchar *name;
        GstCaps * p_caps;
        GstElement* nextElement;
        gchar*str;
        PipelineClass* pipe = (PipelineClass*)data;
        name = gst_pad_get_name(pad);
        g_print("A new pad %s was created\n", name);
        p_caps = gst_pad_get_pad_template_caps (pad);
       
        str = gst_caps_to_string(p_caps);
        g_print("Caps: %s",str);

        if (strstr(name, "_97") != NULL)
        {
                pipe->AddToConsoleOutputList("------------------------ Video -------------------------------");
               
                        nextElement = gst_bin_get_by_name(GST_BIN(pipe->_videoRecClass.srcPipeline),"videodepay");
                        pipe->_videoRecClass.aVideoSrcPad = pad;

                        if(!gst_element_link_pads_filtered(element, name, nextElement, "sink",p_caps))
                {
                        pipe->AddToConsoleOutputList("Failed to link video element to src to sink");
                }
                gst_object_unref(nextElement);
               
        }
       
        else if (strstr(name, "_96") != NULL )
        {
                pipe->AddToConsoleOutputList("------------------------ Audio -------------------------------");

                nextElement = gst_bin_get_by_name(GST_BIN(pipe->_videoRecClass.srcPipeline),"audiodepay");
               
                if(nextElement != NULL)
                {
                if(!gst_element_link_pads_filtered(element, name, nextElement, "sink",p_caps))
                {
                        pipe->AddToConsoleOutputList("Failed to link audio element to src to sink");
                }
                }
                gst_object_unref(nextElement);

        }
        else
        {
                pipe->AddToConsoleOutputList("---------------------- CANNOT DETECT MEDIA ----------------------");
        }
        g_free(name);
        g_free(str);
        gst_caps_unref(p_caps);
}

(VideoConvert.exe:8424): GLib-GObject-WARNING **: gsignal.c:3406: signal
name 'reset-sync' is invalid for instance '047E2020' of type 'GstRTPDec'
A new pad recv_rtp_src_1_2158651129_97 was created
Caps: application/x-rtp; application/x-rdt
------------------------ Video -------------------------------

(VideoConvert.exe:8424): GStreamer-CRITICAL **: gst_segment_to_running_time:
assertion 'segment->format == format' failed
A new pad recv_rtp_src_0_2158651128_96 was created
Caps: application/x-rtp; application/x-rdt
------------------------ Audio -------------------------------
0:00:11.599951631  8424   00881CB0 WARN              capsfilter
gstcapsfilter.c:469:gst_capsfilter_prepare_buf:<capsfilter1> error: Filter
caps do not
 completely specify the output format
0:00:11.601438633  8424   00881CB0 WARN              capsfilter
gstcapsfilter.c:469:gst_capsfilter_prepare_buf:<capsfilter1> error: Output
caps are un
fixed: application/x-rtp, media=(string){ video, audio, application },
clock-rate=(int)[ 1, 2147483647 ], encoding-name=(string)MPEG4-GENERIC,
mode=(s
tring){ generic, CELP-cbr, CELP-vbr, AAC-lbr, AAC-hbr }
0:00:11.603494697 ERROR: from element
/GstPipeline:pipeline/GstCapsFilter:capsfilter1: Filter caps do not
completely specify the output format
 8424Additional debug info:
gstcapsfilter.c(469): gst_capsfilter_prepare_buf ():
/GstPipeline:pipeline/GstCapsFilter:capsfilter1:
Output caps are unfixed: application/x-rtp, media=(string){ video, audio,
application }, clock-rate=(int)[ 1, 2147483647 ], encoding-name=(string)MPEG
4-GENERIC, mode=(string){ generic, CELP-cbr, CELP-vbr, AAC-lbr, AAC-hbr }
   00881CB0 WARN           basetransform
gstbasetransform.c:2207:default_generate_output:<capsfilter1> could not get
buffer from pool: error
0:00:11.607876701  8424   00881CB0 WARN                 rtspsrc
gstrtspsrc.c:5483:gst_rtspsrc_loop:<source> error: Internal data flow error.
0:00:11.609185569  8424   00881CB0 WARN                 rtspsrc
gstrtspsrc.c:5483:gst_rtspsrc_loop:<source> error: streaming task paused,
reason error



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[Gst-Geek]

Output caps are in complete. Your blcok diagram is not clear. Can you  attach
png file ??



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[killerrats]

<http://gstreamer-devel.966125.n4.nabble.com/file/t377034/rtsp_rec_pipeline_diagram.jpg>

there is a correction. I don't have a rtpjitterbuffer element anymore.



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