The diff between compared timestamps might be outside the gint range
resulting in wrong sorting results. This patch corrects that by
comparing the timestamps and then returning -1, 0 or 1 depending on the
result.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7726>
The wraparound handling code assumes that the PCR gets updated regularly for
being able to detect wraparounds. With ignore-pcr=true that was not the case and
it stayed initialized at 1h forever.
To avoid this problem, update the fake PCR whenever the PTS advanced by more
than 5s, and also detect wraparounds in these fake PCRs.
Problem can be reproduced with
$ gst-launch-1.0 videotestsrc pattern=black ! video/x-raw,framerate=1/5 ! \
x264enc speed-preset=ultrafast tune=zerolatency ! mpegtsmux ! \
tsdemux ignore-pcr=true ! fakesink
which restarts timestamps at 0 after around 26h30m.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7588>
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>
When force-live is TRUE, aggregator will correctly change its state with
NO_PREROLL, but unless something upstream is live did not previously set
live to TRUE on the latency query.
Fix this by or'ing force_live into the result.
Also improve debug
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7718>
Clients that already gotten a signal for synced clock, may rely on
getting the same when marked as corrupted to take appropriate action. So
send clock signal indicating no sync at identified corrupted state.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7664>
This allows the stream to drive the buffers submitted to the display server.
If the application does not receive frame events for a period of time due to
minimization or tty switch for example, instead of waiting to process and
then catching up when frame events resume, the stream will resume instantly.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7691>
There is no requirement for a base DRM format to be supported by libgstvideo
in order to be uploaded to. Don't limite to DRM fourcc that have a libgstvideo
format mapping. This notably enabled AFBC support, which uses an opaque based
format that does not have a linear definition. This also adds R8/RG88 and
simimlar other formats that are not yet mapped in libgstvideo.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7689>
When the stream resolution change it is needed to negotiate
a new pools and to update the caps.
Resolution change could occurs on a new sequence or a new
picture so move resolution change detection code in a common
function.
For memory allocation reasons, only allows resolution change
on non keyframe if the driver support remove buffer feature.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7684>
We must drain the pending output picture so that subclass can renegotiate
the caps. Not doing so while still renegotiating would mean that the
subclass would have to do an allocation query before pushing the caps.
Pushing the caps now without this would also not work since these caps
won't match the pending buffers format.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7684>
Add helpers function to call VIDIOC_REMOVE_BUFS ioctl.
If the driver support this feature buffers are removed from the queue when:
- the pool when is detached from the decoded.
- the pool is released.
- allocation failed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7684>
Use VIDIOC_CREATE_BUFS ioctl to create buffers instead of VIDIOC_REQBUFS
because it allows to create buffers also while streaming.
To prepare the introduction of VIDIOC_REMOVE_BUFFERS create
the buffers one per one instead of a range of them. This way
it can, in the futur, fill the holes.
gst_v4l2_decoder_request_buffers() is stil used to remove all
the buffers of the queue.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7684>
When a datachannel within a session is removed after proper close,
reference to the error_ignore_bin elements of the datachannel
appsrc/appsink were left in webrtcbin.
This caused the bin-objects to be left and not freed until the whole
webrtc session was terminated. Among other things that includes a thread
from the appsrc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7675>
