If not waiting for free thread pool before clean transport caches, there
can be a crash if a thread is executing in transport list loop in
function send_tcp_message.
Also add a check if priv->send_pool in on_message_sent to avoid that a
new thread is pushed during wait of free thread pool. This is possible
since when waiting for free thread pool mutex have to be unlocked.
This adds new functions for passing buffer lists through the different
layers without breaking API/ABI, and enables the appsink to actually
provide buffer lists.
This should already reduce CPU usage and potentially context switches a
bit by passing a whole buffer list from the appsink instead of
individual buffers. As a next step it would be necessary to
a) Add support for a vector of data for the GstRTSPMessage body
b) Add support for sending multiple messages at once to the
GstRTSPWatch and let it be handled internally
c) Adding API to GOutputStream that works like writev()
Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
In plug_src we changed the element state before adding it to
the owner container. This prevented the pipeline from intercepting
a GST_STREAM_STATUS_TYPE_CREATE message from the pad in order
to assign a custom task pool.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/53
The sequence number in the rtpinfo is supposed to be the first RTP
sequence number. The "seqnum" property on a payloader is supposed to be
the number from the last processed RTP packet. The sequence number for
payloaders that inherit gstrtpbasepayload will not be correct in case of
buffer lists. In order to fix the seqnum property on the payloaders
gst-rtsp-server must get the sequence number for rtpinfo elsewhere and
"seqnum-offset" from the "stats" property contains the value of the
very first RTP packet in a stream. The server will, however, try to look
at the last simple in the sink element and only use properties on the
payloader in case there no sink elements yet, and by looking at the last
sample of the sink gives the server full control of which RTP packet it
looks at. If the payloader does not have the "stats" property, "seqnum"
is still used since "seqnum-offset" is only present in as part of
"stats" and this is still an issue not solved with this patch.
Needed for gst-plugins-base!17
When the underlying layers are running on_message_sent, this sometimes
causes the underlying layer to send more data, which will cause the
underlying layer to run callback on_message_sent again. This can go on
and on.
To break this chain, we introduce an idle source that takes care of
sending data if there are more to send when running callback
https://bugzilla.gnome.org/show_bug.cgi?id=797289
By default the multicast sockets are bound to INADDR_ANY,
as it's not allowed to bind sockets to multicast addresses
in Windows. This default behaviour can be changed by setting
bind-mcast-address property on the media-factory object.
https://bugzilla.gnome.org/show_bug.cgi?id=797059
Export rtsp-server library API in headers when we're building the
library itself, otherwise import the API from the headers.
This fixes linker warnings on Windows when building with MSVC.
Fix up some missing config.h includes when building the lib which
is needed to get the export api define from config.h
https://bugzilla.gnome.org/show_bug.cgi?id=797185
When media is shared, the same media stream can be sent
to multiple multicast groups. Currently, there is no API
to retrieve multicast addresses from the stream.
When calling gst_rtsp_stream_get_multicast_address() function,
only the first multicast address is returned.
With this patch, each multicast destination requested in SETUP
will be stored in an internal list (call to
gst_rtsp_stream_add_multicast_client_address()).
The list of multicast groups requested by the clients can be
retrieved by calling gst_rtsp_stream_get_multicast_client_addresses().
There still exist some problems with the current implementation
in the multicast case:
1) The receiving part is currently only configured with
regard to the first multicast client (see
https://bugzilla.gnome.org/show_bug.cgi?id=796917).
2) Secondly, of security reasons, some constraints should be
put on the requested multicast destinations (see
https://bugzilla.gnome.org/show_bug.cgi?id=796916).
Change-Id: I6b060746e472a0734cc2fd828ffe4ea2956733ea
https://bugzilla.gnome.org/show_bug.cgi?id=793441
The maximum ttl value provided so far by the multicast clients
will be chosen and reported in the response to the current
client request.
Change-Id: I5408646e3b5a0a224d907ae215bdea60c4f1905f
https://bugzilla.gnome.org/show_bug.cgi?id=793441
If "transport.client-settings" parameter is set to true, the client is
allowed to specify destination, ports and ttl.
There is no need for pre-configured address pool.
Change-Id: I6ae578fb5164d78e8ec1e2ee82dc4eaacd0912d1
https://bugzilla.gnome.org/show_bug.cgi?id=793441
If "transport.client-settings" parameter is set to true, the client is
allowed to specify destination, ports and ttl.
There is no need for pre-configured address pool.
https://bugzilla.gnome.org/show_bug.cgi?id=793441
multiudpsink does not support setting the socket* properties
after it has started, which meant that rtsp-server could no
longer serve on both IPV4 and IPV6 sockets since the patches
from https://bugzilla.gnome.org/show_bug.cgi?id=757488 were
merged.
When first connecting an IPV6 client then an IPV4 client,
multiudpsink fell back to using the IPV6 socket.
When first connecting an IPV4 client, then an IPV6 client,
multiudpsink errored out, released the IPV4 socket, then
crashed when trying to send a message on NULL nevertheless,
that is however a separate issue.
This could probably be fixed by handling the setting of
sockets in multiudpsink after it has started, that will
however be a much more significant effort.
For now, this commit simply partially reverts the behaviour
of rtsp-stream: it will continue to only create the udpsinks
when needed, as was the case since the patches were merged,
it will however when creating them, always allocate both
sockets and set them on the sink before it starts, as was
the case prior to the patches.
Transport configuration will only error out if the allocation
of UDP sockets fails for the actual client's family, this
also downgrades the GST_ERRORs in alloc_ports_one_family
to GST_WARNINGs, as failing to allocate is no longer
necessarily fatal.
https://bugzilla.gnome.org/show_bug.cgi?id=796875
Before, the watch backlog size in GstRTSPClient was changed
dynamically between unlimited and a fixed size, trying to avoid both
unlimited memory usage and deadlocks while waiting for place in the
queue. (Some of the deadlocks were described in a long comment in
handle_request().)
In the previous commit, we changed to a fixed backlog size of 100.
This is possible, because we now handle RTP/RTCP data messages differently
from RTSP request/response messages.
The data messages are messages tunneled over TCP. We allow at most one
queued data message per stream in GstRTSPClient at a time, and
successfully sent data messages are acked by sending a "message-sent"
callback from the GstStreamTransport. Until that ack comes, the
GstRTSPStream does not call pull_sample() on its appsink, and
therefore the streaming thread in the pipeline will not be blocked
inside GstRTSPClient, waiting for a place in the queue.
pull_sample() is called when we have both an ack and a "new-sample"
signal from the appsink. Then, we know there is a buffer to write.
RTSP request/response messages are not acked in the same way as data
messages. The rest of the 100 places in the queue are used for
them. If the queue becomes full of request/response messages, we
return an error and close the connection to the client.
Change-Id: I275310bc90a219ceb2473c098261acc78be84c97
Fix race when setting up source elements.
Since we set the source element(s) to PLAYING state before hooking
them up to the downstream funnel, it's possible for the source element
to receive packets before we actually get to linking it to the funnel,
in which case buffers would be pushed out on an unlinked pad, causing
it to error out and stop receiving more data.
We fix this by blocking the source's srcpad until we have linked it.
https://bugzilla.gnome.org/show_bug.cgi?id=796160
Transport specific sink elements are added to the pipeline
in PLAY request and sockets are already created in SETUP so
it's actually wrong to require the presence of sinks in
_get_*_socket() functions.
https://bugzilla.gnome.org/show_bug.cgi?id=793441
If a multicast client requests different transport settings
than the existing one make sure that this new transport
configuruation is propagated to the multicast udp sink.
https://bugzilla.gnome.org/show_bug.cgi?id=793441
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
