This fixes a regression from commit "srtp: Support libsrtp2"
e9aa117200 where an internal
set of ssrc(s) was added because the libsrtp v2 keeps its
internal streams as private. But the change prevented that
ssrc(s) that not in the caps from being added to the stats.
This patch ensures that all ssrc(s) are inserted to this set
instead of only inserting those from the caps.
During element shutdown, the srtp encryption session
object can be cleaned up. In that case, return GST_FLOW_FLUSHING
from the chain function. Also properly return GST_FLOW_ERROR
upstream during actual errors.
https://bugzilla.gnome.org/show_bug.cgi?id=790508
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
We remove get-rollover-counter signal in favor of the "stats"
property. The "stats" property is a GstStructure with caps
application/x-srtp-encoder-stats that contains an array of
structures with caps application/x-srtp-stream.
Each stream structure contains "ssrc" and "roc" fields.
https://bugzilla.gnome.org/show_bug.cgi?id=733265
We add a new signal, get-rollover-counter, to the SRTP encoder. Given a
ssrc the signal will return the currently internal SRTP rollover counter
for the given stream.
For the SRTP decoder we have a new SRTP caps parameter "roc" that needs
to be set when a new SRTP stream is created for a given SSRC.
https://bugzilla.gnome.org/show_bug.cgi?id=726861
* 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
