For libsrtp 1, add defines that translate the new namespaced identifiers
to the old unnamespaced ones. Also move the code for setting and getting
a stream's ROC into two compat functions that match libsrtp2's API.
It seems that libsrtp2 properly supports changing the ROC without having
to touch the sequence numbers afterwards, given that srtp_set_stream_roc
sets a pending_roc field, so the entire roc_changed dance should not be
needed anymore. The compat functions for libsrtp 1 just contain our
preexisting hacks, however, so it's still needed there.
libsrtp2 has no means of discovering the streams in the session, so to
create the stats structure we need to iterate over our own set of SSRCs.
For this we also need to re-add the previously removed ssrcs_set to the
encoder.
https://bugzilla.gnome.org/show_bug.cgi?id=776901
rtcp_buffer_get_ssrc is called even with RTP buffers. this means we
might end up with an exception and not find any valid RTCP packet type
and thus hit GST_RTCP_TYPE_INVALID. we now take care of this.
https://bugzilla.gnome.org/show_bug.cgi?id=727512
* ext/srtp/gstsrtp.[ch]: added GST_SRTP_CIPHER_AES_256_ICM to
GstSrtpCipherType and new function cipher_key_size.
* ext/srtp/gstsrtpenc.c: maximum key size is now 46 characters (14 for
the salt plus the key). If different ciphers are chosen for RTP and
RTCP the maximum needed key size is expected.
* ext/srtp/gstsrtpdec.c: minor documentation updates.
https://bugzilla.gnome.org/show_bug.cgi?id=720434
