They are very confusing for people, and more often than not
also just not very accurate. Seeing 'last reviewed: 2005' in
your docs is not very confidence-inspiring. Let's just remove
those comments.
Add an accumulator that stops the signal emission as soon as a caps has
been retrieved. Otherwise the default handler would continue emitting
the signal and possibly overwrite the result with NULL again.
AUX elements are elements that can be inserted into the rtpbin
pipeline right before or after 1 or more session elements.
The AUX elements are essential for implementing functionality such
as error correction (FEC) and retransmission (RTX).
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711087
Keep track of elements that are added to multiple sessions and make sure
we only add them to the rtpbin once and that we clean them when no
session refers to them anymore.
* gst/rtpmanager/gstrtpbin.[ch]: four new action signals have been
added (request-rtp-encoder, request-rtp-decoder, request-rtcp-encoder
and request-rtcp-decoder). The user will be able to provide encoders
or decoders dynamically. The encoders must follow the srtpenc API and
the decoders the srtpdec API. Having separate signals for RTP and RTCP
allows the user to use different encoders/decoders or provide the same
one (e.g. that would be the case for srtpenc).
Also, rtpbin now allows application/x-srtp in its pads.
https://bugzilla.gnome.org/show_bug.cgi?id=719938
rtpbin can now send a custom in-band downstream event which informs
downstream that the bin has received an RTCP SR packet. This is useful
for applications which want to drop the initial unsynchronized received
RTP packets.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703560
Signed-off-by: Carlos Rafael Giani <dv@pseudoterminal.org>
Move the work of cleaning up the client streams in the free_stream
function. This allows us to properly clean up the client streams when we
remove an RTP stream as well.
Based on patch by Sujay <sdatar@cisco.com>
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=660156
When use-pipeline-clock is set, use the running-time of the
pipeline to calculate the NTP timestamps. This method would previously
only work when the base-time is set to 0 but with this change it can
also work with different offsets and we can also implement pause/resume
of the sender and receiver now.
Add private replacements for deprecated functions such as
g_mutex_new(), g_mutex_free(), g_cond_new() etc., mostly
to avoid the deprecation warnings. We'll change these
over to the new API once we depend on glib >= 2.32.
... to at least having it trigger a/v synchronization, possibly without
using provided values which are still not considered sane
(as previously dropped).
... at least if not syncing to NPT time:
* either sync using RTCP SR data (as currently)
* only perform the above once using initial RTCP SR packets
* discard RTCP and sync by equating provided stream's clock-base rtptime,
as provided by jitterbuffer (typically obtained from RTP-Info in RTSP).
