The level in GstAudioLevelMeta is represented as a signed 8bit value from 0 to
127 (with 127 meaning silence). When converting from double, make sure to clip
the value, this also prevent integer overflow in the conversion. This fixes an
issue where a lower then -127db is reported and random level with near silent
streams (due to integer overflow).
Fixes#4068
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8012>
Most importantly, this ensures that UDP streams still continue to run even if
they are not linked for a while. With decodebin3 the pads will all be unlinked
unless selected, and selecting a stream at a later time would otherwise switch
to a stream with a stopped udpsrc.
Apart from that this also ensures that actual errors from handling RTP packets
between udpsrc and the source pads are not silently ignored but considered
errors like they would be for TCP/interleaved.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7946>
A pipeline always has an async bus, which involves allocating an fd pair. As
splitmuxsrc only uses the bus' sync handler, this is not required and can easily
cause splitmuxsrc to exceed the fd limit for no good reason.
The other features of GstPipeline are also not needed here, e.g. clock selection.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7952>
But also don't wait for a buffer on both pads, which might take forever in case
of gaps in one of the streams.
The muxer can only advance the time if it has a timestamped buffer that can be
output, otherwise it will just busy-wait and use up a lot of CPU.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7871>
Some special videos with mlv fourcc can't be recognized by
qtdemux when the subtype of the video is vide instead of
m1v, and will cause negotiation error in subsequent plugin.
So make the handle in qtdemux_video_caps. It might be better
than nothing.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7855>
The `reuse` property end up setting the SO_REUSEADDR socket option for
the UDP socket. This setting have surprising effects.
On Linux systems the man page (`socket(7)`) states:
```
SO_REUSEADDR
Indicates that the rules used in validating addresses supplied
in a bind(2) call should allow reuse of local addresses. For
AF_INET sockets this means that a socket may bind, except when
there is an active listening socket bound to the address.
```
But since UDP does not listen this ends up meaning that when an
ephemeral port is allocated (setting the `port` to `0`) the kernel is
free to reuse any other UDP port that has `SO_REUSEADDR` set.
Tests checking the likelyhood of port conflict when using multiple
`udpsrc` shows port conflicts starting to occur after ~100-300 udpsrc
with port allocation enabled. See issue #3411 for more details.
Changing the default value of a property is not a small thing we risk
breaking application that rely on the current default value. But since
the effects of having `reuse` default `TRUE` on can also have damaging
and hard-to-debug consequences, it might be worth to consider.
Having `SO_REUSEADDR` enabled for multicast, might have some use cases
but for unicast, with dynamic port allocation, it does not make sense.
When not using an multicast address we will disable port reuse if the
`port` property is set to 0 (=allocate) and warn the user that we did
so.
Closes#3411
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7841>
Previously the wrapping of the 24-bit reference time was not handled
correctly when transforming it into GstClockTime. Given the unit of 64ms
the span that could be represented by 24 bits is 12 days and depending
on the start value we could get a wrapping problem anytime within this
time frame. This turned out to be particularly problematic for the GCC
algorithm in gst-plugins-rs which tried to evict old packages based on
the "oldest" timestamp, which due to wrapping problems could be in the
future. Thus, the container managing the packets could grow without
limits for a long time thereby creating both CPU and memory problems.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7527>
If a stream has an 'irregular' frame rate (e.g. metadata) RTCP SR
may be generated way too early, before the RTPSource has received
the first packet after Latency was configured in the pipeline.
We skip such RTPSources in the RTCP generation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7740>
Some servers (e.g. Axis cameras) expect the client to propose the encryption
key(s) to be used for SRTP / SRTCP. This is required to allow re-keying so
as to evade cryptanalysis. Note that the behaviour is not specified by the
RFCs. By setting the 'client-managed-mikey-mode' property to 'true', rtspsrc
acts as follows:
* For a secured profile (RTP/SAVP or RTP/SAVPF), any media in the SDP
returned by the server for which a MIKEY key management applies is
elligible for client managed mode. The MIKEY from the server is then
ignored.
* rtspsrc sends a SETUP with a MIKEY payload proposed by the user. The
payload is formed by calling the 'request-rtp-key' signal for each
elligible stream. During initialisation, 'request-rtcp-key' is also
called as usual. The keys returned by both signals should be the same
for a single stream, but the mechanism allows a different approach.
* The user can start re-keying of a stream by calling SET_PARAMETER.
The convenience signal 'set-mikey-parameter' can be used to build a
'KeyMgmt' parameter with a MIKEY payload.
* After the server accepts the new parameter, the user can call
'remove-key' and prepare for the new key(s) to be served by signals
'request-rtp-key' & 'request-rtcp-key'.
* The signals 'soft-limit' & 'hard-limit' are called when a key
reaches the limits of its utilisation.
This commit adds support for:
* client-managed MIKEY mode to srtpsrc.
* Master Key Index (MKI) parsing and encoding to GstMIKEYMessage.
* re-keying using the signals 'set-mikey-parameter' & 'remove-key' and
then by serving the new key via 'request-rtp-key' & 'request-rtcp-key'.
* 'soft-limit' & 'hard-limit' signals, similar to those provided by srtpdec.
See also:
* https://www.rfc-editor.org/rfc/rfc3830
* https://www.rfc-editor.org/rfc/rfc4567
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7587>
Move RB info from receiver reports into the internal source that the RR
are about, and deprecate (but retain) the old mapping where each
external source has only a single RB entry in the rtp statistics.
The old method is broken if a remote peer uses a single ssrc to send
receiver reports for more than one of our internal sources, other
as multiple RB in a single packet, or alternate RB in different reports.
In each case only the most recent entry was kept, overwriting data for
other internal sources.
In multicast scenarios each internal source may receive multiple
receiver reports from different peers. To support that, all received
RR's are now stored into a hash table indexed by the sender's SSRC,
and all RRs are placed into an array when generating statistics, so
that the information from all peers is retrievable.
The current deficient behaviour (adding RB info into non-internal RTPSources) is
deprecated but kept in order to be backward compatible, and retained
that way in the generated statistics structure.
Refs
[1] https://tools.ietf.org/html/rfc3550#section-6.4.1
Based on a patch by Fede Claramonte <fclaramonte@twilio.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7424>
A zero-sized box is not really a problem and can be skipped to look at any
possibly following ones.
BMD ATEM devices specifically write a zero-sized bmdc box in the sample
description, followed by the avcC box in case of h264. Previously the avcC box
would simply not be read at all and the file would be unplayable.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7564>
Timestamps are untouched by default, but the new mode can now be enabled to replace RTP timestamps
with ones generated from the buffer PTS. Making it an enum in case different modes are needed in the future.
That allows for a rtpjitterbuffer to do proper drift compensation, so that the stream coming out of gst-rtsp-server
is not drifting compared to the pipeline clock and also not compared to the RTCP NTP times.
Most of the code is borrowed from rtpbasepayload, as it's exactly its behaviour which I wanted to bring here.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7526>
It has to be included in the block duration but in GStreamer we're not
including it in the buffer duration, so it has to be added again here.
Not including it in the block duration can lead to fatal errors when playing
back with Firefox if there are more padding samples than actual samples, e.g.
> D/MediaDemuxer WebMDemuxer[7f6a0808b900] ::GetNextPacket: Padding frames larger
> than packet size, flagging the packet for error (padding: {13500000,1000000000},
> duration: {6000,1000000}, already processed: false)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7502>
By setting the earliest time to timestamp + 2 * diff there would be a difference
of 1 * diff between the current clock time and the earliest time the element
would let through in the future. If e.g. a frame is arriving 30s late at the
sink, then not just all frames up to that point would be dropped but also 30s of
frames after the current clock time.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7459>
splitmuxsink can't possibly know how much latency it will introduce as it always
keeps one GOP around before outputting something. This breaks the latency
configuration of the pipeline and we're better off just pretending that
everything downstream of the sinkpads is not live.
Especially muxers that are based on aggregator and time out on the latency
deadline can easily misbehave otherwise as the deadline will be exceeded usually.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7499>
If two (or more) rtpfunnel elements are cascaded, then only one will
realistically have information on the particular ssrc that is in use for a
particular input stream. As such, any key unit requests may never reach the
corresponding encoder.
This has been discovered by combining simulcast and BUNDLE with webrtcbin.
simulcast uses one rtpfunnel, and BUNDLE uses another rtpfunnel.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7405>
