Recent kernels allow REQBUFS(0) on a queue that still has buffers in
use (mmapped or exported via dmabuf), orphaning all buffers on the queue.
If this is supported, the v4l2videodec element does not have to send a
drain request downstream.
Now that the v4l2allocator allows orphaning the V4L2 buffer queue, add
support for orphaning in the v4l2bufferpool. gst_v4l2_buffer_pool_orphan
can be used as a replacement for gst_v4l2_buffer_pool_stop, without
having to wait for buffers to be returned to the pool.
Recent kernels allow REQBUFS(0) on a queue that still has buffers in
use (mmapped or exported via dmabuf), orphaning all buffers on the queue.
Orphaning the allocator causes it to release all buffers with
REQBUFS(0), even if they are still in use. An orphaned allocator can
only be stopped. It can not be restarted or create new buffers.
Update to the latest installed headers (output of make headers_install)
from the media tree, keeping the slight modifications to the includes.
This includes new HEVC controls, the AdobeRGB -> opRGB rename, a new
capabilities field for v4l2_requestbuffers and v4l2_create_buffers, new
32-bit YUV formats, and request_fd changes.
We used to split the NACK if a smaller seqnum of a range of seqnum was
submited. This test also make sure that the three operations (append,
prepend, update) works properly.
In order to do that, we now split the nacks registration from the actual
FB nack packet construction. We then try and add as many FB Nacks as
possible into the active packets and leave the remaining seqnums in the
RTPSource. In order to avoid sending outdated NACK later on, we save the
seqnum calculated deadline and cleanup the outdated seqnums before the
next RTCP send.
Fixes#583
Calling rtp_session_send_rtcp before marking the source as requiring a
pli/fir/nack meant the rtcp_thread could be scheduled and start running
before the source was updated. This meant the request would not be sent
early but instead was transmitted with the next regular RTCP packet.
Add test for nack generation.
If the current time is equal to the early rtcp time deadline, there is
no need to schedule a timer. This ensure that immediate feedback is
really immediate and simplify implementing unit tests with the test
clock, which stops perfectly on the timeout time.
This fix has been extracted from Pexip feature patch called
"rtpsession: Allow instant transmission of RTCP packets"
The caps features were lost when sorting caps structures in
gst_v4l2src_fixate(). This was breaking alternate as
GST_CAPS_FEATURE_FORMAT_INTERLACED was removed from the caps.
When used in combination with a rtponviftimestamp element
downstream, forwarding this flag ensures it gets correctly
serialized in the ONVIF header extension.
A missing colon after G_DEFINE_TYPE declaration was confusing gst-indent
and causing problem in the pre-commit hook.
Add the missing colon and fix the following function declaration to
follow the normal GStreamer style.
One comments in gst_rtp_session_chain_send_rtp_common() is referring to
groups in a buffer list, however this concept of "group" comes from
GStreamer 0.10 and does not exist anymore in GStreamer 1.0, so update the
comment to refer to buffers instead.
Add a test to verify that stats about sent and received packets are
correct even when using buffer lists.
NOTE: the newly introduced get_session_source_stats() selects the
desired source (sender or receiver) by filtering them by type (using the
get_sender parameter) rather than by ssrc because this simplifies the
code and it's good enough for testing purposes as there is usually one
source per type in the test setup.
Filtering by ssrc would have required handling asynchronous signals like
"on-new-sender-ssrc", with the relative locking, just to retrieve the
actual ssrc of the sender.
The update_receiver_stats() function is called also when sending packets
in rtp_source_send_rtp(), and sending packets may happen using a buffer
list rather than individual buffers.
So update the stats using the actual number of packets sent.
NOTE: this is fine for the receive path too (rtp_process_send_rtp)
because the receive path does not support buffer lists and
pinfo->packets would always be equal to 1 in this case.
The tests create a buffer list and then use the chain_list callback to
verify that the correct packets have been pushed.
Move the creation and validation code next to each other so that the
reader can more easily understand what is going on.
While at it add some comments to introduce the two related functions.
Make it possible to differentiate between the position in the list and
the packet index in the global structures in check_packet, in some
future case the list may change, in case some element removes a buffer
from the list, and the two indices may not coincide.
Port the rtpbin_buffer_list test to GStreamer 1.0 and re-enable it.
Some other changes include:
- the check on the caps has been moved from the buffer level to the
pad level;
- remove underscore prefix from static functions names, this is not
idiomatic in C and rarely used in the other tests;
- the unused header_buffer variable has been removed;
- check_group() has been renamed to check_packet() because in
GStreamer 1.0 there is no concept of "group" anymore, the comments
have also been updated to reflect this.
Tests might take a bit longer, esp. when run under valgrind
and/or they're running on the CI with other things going on,
so let's just bump the timeout to something higher and let
the test runner time us out if needed.
On other OSes, it's not possible to have qmake or the qt5 pkg-config
files and not have moc, and `moc` will not be in `PATH`, so this only
causes problems.
The FLAC specification states that the data is processed in blocks, regardless of the number of channels. Thus, The latency can be calculated using the blocksize and rate. For example a 1 second block sampled at 44.1KHz has a blocksize of 44100
This is needed for the case you don't know in advance all the sessions
you will be using, but would like to place all the related AUX element
in the same GstBin. As per current implementation, each time an sender
AUX bin is requested and returned, RTPBin will walk the src pads and
create sessions for these pads.
In the current implementation, if a src pad already have a sessions, it
returns an error and stops. As a side effect, if an AUX bin is reused in
a following AUX bin request, it can only work if the pads are created on
the last request.
This change simply relax the restriction in order to keep walking, and
just ensure that all newly created pads have a sessions.
The V4L2 elements already set the delta unit buffer flag when dequeueing
the buffer, but gst_video_encoder_finish_frame overwrites it from the
passed codec frame's sync point flag. Set the flag correctly.
And check if argument is supported instead of just passing it blindly,
and make meson code slightly cleaner, centralising the argument setting
in one place.
Need to respect return of gst_video_guess_framerate() to ensure
non-zero denominator.
This patch is to fix below error with an abnormal (but has valid frame) file.
(gst-play-1.0:17940): GStreamer-CRITICAL **: passed '0' as denominator for `GstFraction'
