changing rtp capabilities

[Saurabh Sabnis]

I mentioned *clock-rate* by mistake, I was actually talking about only the
random properties such as *seqnum-base* and *ssrc* which according to
RTP should
be random but I want to do something for which I require them to be a
particular value that I can assign.

I tried using *gst_caps_new_simple* and then *link_filtered* to assign
these properties some value, but it didn't take my values and generated
random values anyway.

Let me tell you what I want to do. I want to capture an RTP stream, (I know
the capabilities for this stream and hence I can capture it using udpsrc
and caps filter), vp8rtpdepay and vp8dec it, do some processing on it and
then send that stream out on the network after vp8enc and rtpvp8pay. The
catch is that I want to send it out with the same capabilities that it
originally had. (basically maintain the sequence numbers and ssrc value
that the original stream had). Is there any way of achieving this?



On Wed, Feb 20, 2013 at 1:45 PM, Saurabh Sabnis <sausab at gmail.com> wrote:

> I mentioned *clock-rate* by mistake, I was actually talking about only
> the random properties such as *seqnum-base* and *ssrc* which according to
> RTP should be random but I want to do something for which I require them to
> be a particular value that I can assign.
> I tried using gst_caps_new_simple and then link_filtered to
>
>
> On Wed, Feb 20, 2013 at 12:47 PM, Chuck Crisler <ccrisler at mutualink.net>wrote:
>
>> The generated capabilities aren't random, they are default, except for
>> the seqnum-base, which RTP says can (should?) be random. Different elements
>> require different capabilities, so these may not hold for your case.
>> However, you can always initialize the capabilities like this:
>>
>>             GstCaps *pSinkCaps = gst_caps_new_simple("application/x-rtp",
>> "media", G_TYPE_STRING, "video",
>>                      "encoding-name", G_TYPE_STRING, "H264", "payload",
>> G_TYPE_INT, 96,
>>                      "clock-rate", G_TYPE_INT, 90000, NULL);
>>
>>             // Link the pipeline together.
>>             if (gst_element_link_filtered(source_, queue_, pSinkCaps)) {
>>             ...
>>             }
>>             gst_caps_unref(pSinkCaps);
>>
>>
>> On Wed, Feb 20, 2013 at 10:05 AM, Saurabh Sabnis <sausab at gmail.com>wrote:
>>
>>> Hi,
>>>
>>> I am new to *gstreamer* and I am trying to send RTP stream over the
>>> network. The *'rtpvp8pay'* element automatically generates some random
>>> capabilities such as *seqnum-base, clock-rate* and *ssrc* that need to
>>> be used as cap filter for *'udpsrc'* element at the receiving side.
>>> Is it possible to manually set these capabilities at the sender side
>>> instead of letting 'rtpvp8pay' element to generate these randomly?
>>>
>>>
>>> Thanks and regards,
>>> --
>>> *Saurabh Sabnis*
>>> *Graduate Student*
>>> *College of Computing*
>>> *Georgia Institute of Technology*
>>> *Atlanta, GA 30332-0280*
>>> *ssabnis3 at gatech.edu*
>>>
>>> _______________________________________________
>>> gstreamer-devel mailing list
>>> gstreamer-devel at lists.freedesktop.org
>>> http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>>>
>>>
>>
>> _______________________________________________
>> gstreamer-devel mailing list
>> gstreamer-devel at lists.freedesktop.org
>> http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>>
>>
>
>
> --
> *Saurabh Sabnis*
> *Graduate Student*
> *College of Computing*
> *Georgia Institute of Technology*
> *Atlanta, GA 30332-0280*
> *ssabnis3 at gatech.edu*
>



-- 
*Saurabh Sabnis*
*Graduate Student*
*College of Computing*
*Georgia Institute of Technology*
*Atlanta, GA 30332-0280*
*ssabnis3 at gatech.edu*
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