rtpbin can still emit signals when it is being disposed, and while
rtpbin is inside ristsrc/ristsink it can still live longer.
So we either have disconnect all signals at some point, or let GObject
take care of that automatically.
Related to !1412
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1413>
In file included from ../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/gstbin.h:27,
from ../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/gst.h:35,
from ../gst/rtp/gstrtpsink.h:23,
from ../gst/rtp/gstrtpsink.c:49:
In function ‘gst_rtp_sink_start’,
inlined from ‘gst_rtp_sink_change_state’ at ../gst/rtp/gstrtpsink.c:509:11:
../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/gstelement.h:422:18: error: ‘%s’ directive argument is null [-Werror=format-overflow=]
422 | gchar *__txt = _gst_element_error_printf text; \
../gst/rtp/gstrtpsink.c:476:3: note: in expansion of macro ‘GST_ELEMENT_ERROR’
476 | GST_ELEMENT_ERROR (self, RESOURCE, NOT_FOUND,
| ^~~~~~~~~~~~~~~~~
../gst/rtp/gstrtpsink.c: In function ‘gst_rtp_sink_change_state’:
../gst/rtp/gstrtpsink.c:477:37: note: format string is defined here
477 | ("Could not resolve hostname '%s'", remote_addr),
| ^~
In file included from ../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/gstbin.h:27,
from ../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/gst.h:35,
from ../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/rtp/gstrtpdefs.h:27,
from ../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/rtp/rtp.h:25,
from ../gst/rist/gstristsink.c:72:
In function ‘gst_rist_sink_setup_rtcp_socket’,
inlined from ‘gst_rist_sink_start’ at ../gst/rist/gstristsink.c:658:10,
inlined from ‘gst_rist_sink_change_state’ at ../gst/rist/gstristsink.c:801:13:
../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/gstelement.h:422:18: error: ‘%s’ directive argument is null [-Werror=format-overflow=]
422 | gchar *__txt = _gst_element_error_printf text; \
../gst/rist/gstristsink.c:595:3: note: in expansion of macro ‘GST_ELEMENT_ERROR’
595 | GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND,
| ^~~~~~~~~~~~~~~~~
../gst/rist/gstristsink.c: In function ‘gst_rist_sink_change_state’:
../gst/rist/gstristsink.c:596:37: note: format string is defined here
596 | ("Could not resolve hostname '%s'", remote_addr),
| ^~
This patchs add support for configuring the bonding method used. There is
two method specified
- redundant: All the RTP packets are replicated
- combined: RTP packet are evenly distributed over each links
Additionally, an application can set the "dispatcher" property in order
to implement custom dispatching method. Whenever the "dispatcher"
property is set, "bonding-method" property will be ignored.
As we can now have multiple sessions, stats need to be implemented per
session. This follow RTPSession model with sources. The stats are now:
dropped: 0
received: 0
recovered: 0
permanently-lost: 0
duplicates: 0
retransmission-requests-sent: 0
rtx-roundtrip-time: 0
session-stats:
session-id=0
rtp-from=""
rtcp-from=""
dropped=0
received=0
session-id=1
rtp-from=""
rtcp-from=""
dropped=0
received=0
. . .
session-stats is a GValueArray as there is no better alternatives.
As we can now have multiple sessions, stats need to be implemented per
session. This follow RTPSession model with sources. The stats are now:
sent-original-packets: 0
sent-retransmitted-packets: 0
session-stats:
session-id=0
sent-original-packets=0
sent-retransmitted-packets=0
round-trip-time=0
session-id=1
sent-original-packets=0
sent-retransmitted-packets=0
round-trip-time=0
. . .
session-stats is a GValueArray as there is no better alternatives.
RIST TR-06-1 is a specification for video streaming made by the VSF
group. It is using a subset of RTP specification to which some
modification has been made to improve RTX behaviour and avoid any need
for signaling. The plugin implement ristrtxsend / ristrtxreceive element
which are the RIST specific equivalent of rtprtxsend/rtprtxreceive and
ristsink / ristsrc which implement rist transmitter and receiver. The
RIST protocol is meant to be used in unidirectional way. Typically, MPEG
TS over RTP is used.
Currently we support unicast and multicast streaming according to the
specification. This patch does not include any bonding support yet. The
ristsrc element introduce rist:// URI handling in parallel to it's
property configuration interface.
