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>
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>
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>
Publish fragment-id in the messages that splitmuxsink and splitmuxsrc
send, so when they are received out of order (due to async finalization,
for example), they can still be identified / ordered correctly.
Fix a race in the splitmuxsink unit test where messages might be
received out of order
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7053>
Publish the playback offset for and duration into the
splitmuxsink-fragment-closed bus message as each fragment
finishes.
These can be passed to splitmuxsrc via the 'add-fragment'
signal to avoid splitmuxsrc measuring all files on startup
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7053>
Add a signal that allows adding fragments with a specific offset
and duration directly to splitmuxsrc's list. By providing the
fragment's offset on the playback timeline and duration directly,
splitmuxsrc doesn't need to measure the fragment making for faster
startup times.
Add a bus message that's published when fragments are measured,
reporting the offset and duration, so they can be cached by an
application and used on future invocations.
Add examples for handling the bus message and using the 'add-fragment'
signal.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7053>
Specify "layout" field in src template to make sure it's
set and gets fixated properly if the downstream element
supports both interleaved and non-interleaved caps.
Fixes
gst_pad_set_caps: assertion 'caps != NULL && gst_caps_is_fixed (caps)' failed
critical with e.g.
gst-launch-1.0 rtpdtmfsrc ! rtpdtmfdepay ! audioconvert ! fakesink
Not that the layout really matters in our case since we always
output mono anyway, but non-interleaved requires adding AudioMeta,
so this is the easiest fix.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7036>
This elements pass RTP packets along unchanged and appear as a RTP
payloader element.
This is useful, for example when using the gstreamer-rtsp-server
library, in the case where you are receiving RTP packets from a
different source and want to serve them over RTSP. Since the
gst-rtsp-server library expect the element marked as payX to be a RTP
payloader element and assumes certain properties are available.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5204>
The aim of this example is to show how to make use of the accept-certificate
signal from a GTK GUI, and prompt user in case of invalid certificate.
There are two subtleties to be aware of:
1. the signal is emitted from the GStreamer streaming thread, therefore the
caller can't modify the GUI straight away, instead they must do it from the
main thread (eg. by using g_idle_add())
2. in case of a redirection, then a TLS failure, the caller won't know
about the redirection. Actually, it's possible to be notified of the
redirection by watching "message:element" and inspecting http-headers,
but even in that case, the signal will be received *after* the signal
"accept-certificate" (even though the redirection happened *before*).
This second point is tricky. It's not uncommon to have servers that redirect
http requests to https. So errors of the type "HTTP -> HTTPS -> TLS error"
happen, and if the caller doesn't care about redirection, they might prompt
users with a message such as "TLS error for URL http://...", which wouldn't make
much sense.
This example shows how to handle that right, by connecting to the signal
"message:element", inspecting the http-headers, and in case of redirection,
updating the TLS error dialog to indicate that the request was redirected.
Here are a few examples of streams that exhibit TLS failure (at the time of
this commit, of course):
* https://radiolive.sanjavier.es:8443/stream: unknown-ca
* https://am981.ddns.net:9005/stream.ogg: unknown-ca
* http://stream.diazol.hu:7092/zene.mp3: redir then bad-identity
* https://streaming.fabrik.fm/izwi/echocast/audio/index.m3u8: unknown-ca
(this one is a HLS stream)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4925>
When doing a segment seek, the base offset in the new segment
would be increased by segment.position which is basically the
timestamp of the last packet. This does not include the duration
of the last packet though, so might be slightly shorter than the
actual duration of the clip or the requested segment.
Increase the base offset by the segment duration instead when
accumulating segments, which is more correct as it doesn't cut
off the last frame and makes the effective loop segment duration
consistent with the actual duration returned from a duration
query.
In case a segment stop was specified it's also possible that
some data was sent beyond the stop that's necessary for decoding
so the base offset increment should be based on that then and
not on the timestamp of the last buffer pushed out.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4604>
Allow a project to use gstreamer-full as a static library
and link to create a binary without dependencies.
Introduce the option 'gst-full-target-type' to
select the build type, dynamic(default) or static.
In gstreamer-full/static build configuration gstreamer (gst.c)
needs the symbol gst_init_static_plugins which is defined
in gstreamer-full.
All the tests and examples are linking with gstreamer but the
symbol gst_init_static_plugins is only defined in the gstreamer-full
library. gstreamer-full can not be built first as it needs to know what plugins
will be built.
One option would be to build all the examples and tests after
gstreamer-full as the tools.
Disable tools build in subprojects too as it will be built at the end of
build process.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4128>
In recent versions of Chrome (M106) a change on their jitter buffer means that
they are very susceptible to PictureID discontinuities.
Then avoid at all cost resetting the PictureID. Moreover, according to
the RFCs for VP8 and VP9 payloads; the PictureID can start off at any
random value. So there is no logical problem of incrementing it here
rather than resetting it, as long as it is a different PictureID.
WebRTC's recent corruption issue:
https://bugs.chromium.org/p/webrtc/issues/detail?id=15101
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4530>
By keeping async to TRUE, a deadlock is avoided where the appsink is
filled with data after a flushing seek but before its PAUSED->PLAYING
state change finishes. If that happens, the appsink is stuck, because
its internal condition variable waits for the appsink to have more room
for data. The basesink's preroll lock is held during this, and it also
tries to acquire that lock during the state change -> deadlock.
By keeping async to TRUE, this flood of data does not happen.
Also, setting the max-buffers property to 1 is unnecessary - the test
runner will anyway detect excess memory usage if it happens.
Other property adjustments turned out to just be redundant.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4200>
