How to use g_signal_connect to connect gstrtpbin with depayloader(in both case Audio+Video together)

[arpi d]

Thanks its working.I have used the recv_rtp_src_m (where m is the session
identifier) and used the same function  GST_PAD_NAME (new_pad) to get this
name. passed the depayloaders in an array in g_signal_connect.We can do
other way also checking the caps from pad and from there we can search
payload type or media type .

On Mon, Aug 1, 2011 at 12:53 AM, Andy_Milestone <
andy87.milestone at googlemail.com> wrote:

> Well, I had the same problem, because I am working on softphone with video-
> and audio support.
>
> I am not sure If there is an "intelligent" solution for this problem, but I
> checked the payload-type, which is given in this line:
>
> >    g_print ("new payload on pad: %s\n", GST_PAD_NAME (new_pad));
> If its payloadtype 0 then I connect the rtpbin with the audio-depay and if
> its 96 I connect it to the video-depay. In my case the rtpbin is defined as
> a pointer in a global structure, therefore I can decide in the callback
> which depayloader should be used.
>
> Do want to realize undirectional or bidirectional streaming?
>
>
> Andreas
>
>
> Am 01.08.2011 06:15, schrieb arpi d:
>
>> I am writing client code which receives audio+video.
>>
>> When I am trying to connect rtpbin and depayloader how to know which
>> session is this?
>>
>> Actually,I am writing the code in C and am trying to use g_signal_connect.
>> So that I can use
>>  g_signal_connect (rtpbin, "pad-added", G_CALLBACK (pad_added_cb),
>> videodepay);
>>
>>  g_signal_connect (rtpbin, "pad-added", G_CALLBACK (pad_added_cb),
>> audiodepay);
>>
>> and the pad_added_cb handler is:
>>
>> |/* will be called when rtpbin has validated a payload that we can
>> depayload */
>> |static void
>>
>> |pad_added_cb (GstElement * rtpbin, GstPad * new_pad, GstElement * depay)
>> {
>>   GstPad *sinkpad;
>>   GstPadLinkReturn lres;
>>
>>   g_print ("new payload on pad: %s\n", GST_PAD_NAME (new_pad));
>>
>>   sinkpad = gst_element_get_static_pad (depay,"sink");
>>   g_assert (sinkpad);
>>
>>   lres = gst_pad_link (new_pad, sinkpad);
>>   g_assert (lres == GST_PAD_LINK_OK);
>>   gst_object_unref (sinkpad);
>> }|
>>
>>
>> my client pipeline is:
>>
>> VIDEO_DEC="rtph264depay ! ffdec_h264"
>> AUDIO_DEC="rtpmp4gdepay ! faad"
>>
>> VIDEO_SINK="ffmpegcolorspace ! autovideosink"
>> AUDIO_SINK="autoaudiosink"
>> DEST=127.0.0.1
>>
>> LATENCY=100
>>
>> gst-launch -v gstrtpbin name=rtpbin                   \
>>     udpsrc caps=$VIDEO_CAPS port=5000 ! rtpbin.recv_rtp_sink_0         \
>>       rtpbin. ! $VIDEO_DEC ! $VIDEO_SINK                               \
>>     udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0                         \
>>  rtpbin.send_rtcp_src_0 ! udpsink port=5005 host=$DEST sync=false
>> async=false
>> udpsrc caps=$AUDIO_CAPS port=5002 ! rtpbin.recv_rtp_sink_1              \
>> rtpbin. ! $AUDIO_DEC ! $AUDIO_SINK                                \
>> udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1                              \
>> rtpbin.send_rtcp_src_1 ! udpsink port=5007 host=$DEST sync=false
>> async=false
>> Same thing as above pipeline in C language I am writing.
>> Please reply.
>> If you know other way please do let  me know.
>> Thaks in advance.
>>
>>
>>
>> ______________________________**_________________
>> gstreamer-devel mailing list
>> gstreamer-devel at lists.**freedesktop.org<gstreamer-devel at lists.freedesktop.org>
>> http://lists.freedesktop.org/**mailman/listinfo/gstreamer-**devel<http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel>
>>
>
> ______________________________**_________________
> gstreamer-devel mailing list
> gstreamer-devel at lists.**freedesktop.org<gstreamer-devel at lists.freedesktop.org>
> http://lists.freedesktop.org/**mailman/listinfo/gstreamer-**devel<http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freedesktop.org/archives/gstreamer-devel/attachments/20110802/581e18c8/attachment.html>