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rtp: also support shrinking the extension data

Matthew Waters requested to merge ystreet/gstreamer:gst-plugins-base-rtp-extension-shrink into main

Copied from https://gitlab.freedesktop.org//gstreamer/gst-plugins-base/-/merge_requests/1259

Currently the extension data length specified in the RTP header would say it was shorter then the data serialised to a packet. When combining the resulting buffer, the underlying memory would still contain the extra (now 0-filled) padding data.

This would mean that parsing the resulting RTP packet would potentially start with a number of 0-filled bytes which many RTP formats are not expecting.

Such usage is found by e.g. RTP header extension when allocating the maximum buffer (which may be larger than the written size) and shrinking to the required size the data once all the rtp header extension data has been written.

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