Added it right before pushing the previous commit, it is
incorrect and deadlocks because this function gets called
from the join_bin thread, which already holds the lock,
that's the reason why request_aux_sender didn't take the
lock either.
"do-retransmission" was previously set when rtx-time != 0,
which made no sense as do-retransmission is used to enable
the sending of retransmission requests, where as rtx-time
is used by the peer to enable storing of buffers in order
to respond to retransmission requests.
rtsp-media now also provides a callback for the
request-aux-receiver signal.
https://bugzilla.gnome.org/show_bug.cgi?id=794822
...and replace all checks for RECORD in GstRTSPMedia which are really
for "sender-only". This way the code becomes more generic and introducing
support for onvif-backchannel later on will require no changes in
GstRTSPMedia.
In the multicast case (as in test-multicast, not test-multicast2), the
address could be allocated/reserved (and thus set) already without
allocating the actual socket. We need to allocate the socket here still
instead of just claiming that it was already allocated.
See https://bugzilla.gnome.org/show_bug.cgi?id=791743#c2
SDP are now provided *before* the pipeline is fully complete. In order
to know whether a media is seekable or not therefore requires asking
the invididual streams.
API: gst_rtsp_stream_seekable
https://bugzilla.gnome.org/show_bug.cgi?id=790674
The initial pipeline does not contain specific transport
elements. The receiver and the sender parts are added
after PLAY.
If the media is shared, the streams are dynamically
reconfigured after each PLAY.
https://bugzilla.gnome.org/show_bug.cgi?id=788340
If no sinks have been added yet, obtain the current and
the stop position of the stream from the send_src pad.
Change-Id: Iacd4ab4bdc69f6b49370d06012880ce48a7d595a
https://bugzilla.gnome.org/show_bug.cgi?id=788340
Commit 852cc09f54 assumed that
multiudpsink's last-sample always comes from the payloader. Which
is wrong if auxiliary streams are multiplexed in the same stream.
So check the buffer's ssrc against the caps'ssrc before to use its
seqnum. If not the same ssrc just use the payloader as done prior
the commit above or when there is no last-sample yet.
https://bugzilla.gnome.org/show_bug.cgi?id=784094
Calling function gst_rtsp_stream_get_server_port() results in
segmenation fault in the RTP/RTSP/TCP case.
Port that the server will use to receive RTCP makes only
sense in the UDP case, however the function should handle
the TCP case in a nicer way.
https://bugzilla.gnome.org/show_bug.cgi?id=776345
85c52e194b introduced a more correct
detection of the srtp rollover counter to add to the SDP.
Unfortunately, it was incomplete for live pipelines where the logic
blocks the source bin before creating the SDP and thus would never have
the necessary informaiton to create a correct SDP with srtp encryption.
Move the pad blocks to rtpbin's output pads instead so that the
necessary information can be created before we need the information for
the SDP.
https://bugzilla.gnome.org/show_bug.cgi?id=770239
With this RTSP server can use the sockets independent on the udpsrc
state.
When the udp src is finalized it will unref socket and when g_socket
is finalized the socket will be closed.
https://bugzilla.gnome.org/show_bug.cgi?id=765673
When using dynamic elements, gst_rtsp_stream_join_bin() is called from
"pad-added" signal. In that case priv->srcpad could already have its caps,
and they'll be sent to priv->send_src[0] pad. That means that when it
connects "notify::caps" signal, that pad could already have received its
caps and the signal won't be emitted anymore.
In that case priv->caps stay to NULL and when building the SDP that stream
gets ignored. Leading to missing video or audio when playing in client side.
https://bugzilla.gnome.org/show_bug.cgi?id=772478
Adding them later will cause deadlocks due to
1) pre-rolling and staying in PAUSED with the unicast/TCP sinks
2) adding the multicast sink
3) waiting for it to get data to preroll again
3) never happens because the queues after the tee are full.
This is basically reverting changes introduced in commit f62a9a7,
because it was introducing various regressions:
- It introduces a leak of udpsrc elements that got wrongly fixed by adding
an hash table in commit cba045e. We should have at most 4 udpsrc for unicast:
ipv4/ipv6, rtp/rtcp. They can be reused for all unicast clients.
- If a mcast client connects, it creates a new socket in SETUP to try to respect
the destination/port given by the client in the transport, and overrides the
socket already set on the udpsink element. That means that if we already had a
client connected, the source address on the udp packets it receives suddenly
changes.
- If a 2nd mcast client connects, the destination/port in its transport is
ignored but its transport wasn't updated.
What this patch does:
- Revert back to create udpsrc/udpsink for unicast clients on DESCRIBE.
- Always have a tee+queue when udp is enabled. This could be optimized
again in a later patch, but is more complicated. If no unicast clients
connects then those elements are useless, this could be also optimized
in a later patch.
- When mcast transport is added, it creates a new set of udpsrc/udpsink,
seperated from those for unicast clients. Since we already support only
one mcast address, we also create only one set of elements.
https://bugzilla.gnome.org/show_bug.cgi?id=766612
We add different crypto sessions in MIKEY, one for each sender
SSRC. Currently, all of them will have the same security policy, 0.
The rollover counters are obtained from the srtpenc element using the
"stats" property.
https://bugzilla.gnome.org/show_bug.cgi?id=730539
- Unicast udpsrcs are now managed in a hash table. This allows for proper cleanup in with shared streams and fixes a memory leak.
- Unicast udpsrcs are now properly cleaned up when shared connections exit. See the update_transport() function.
- Create unit test for shared media.
https://bugzilla.gnome.org/show_bug.cgi?id=764744
