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.
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.
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 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.
False positive for the three variables but some warnings like:
../tests/check/elements/matroskamux.c:875:10:
warning: 'chapters_offset' may be used uninitialized in this function [-Wmaybe-uninitialized]
*index = chapters_offset;
~~~~~~~^~~~~~~~~~~~~~~~~
The above is false positive as there is a gboolean to check if it was
initialized or not (found_chapters_declaration).
This reverts commit dcd3ce9751.
This functionality was implemented for gstopenwebrtc, but it
turned out this was not actually needed for webrtc bundling
support, as shown in webrtcbin. It also doesn't correspond
to any standards.
This is an API break, but nothing should actually depend on
this, at least not for its initial purpose.
Changes in rtpbin.c were reverted manually, to preserve some
refactoring that had occurred in the original commit.
Fixes#537
gstreamer!55 makes some changes to how the `error-after` counter works
which breaks this test. This change makes the test not rely on the
ability to alter `error-after` at runtime and explicitly stops and
starts the harness before pushing data.
An alternative would be to add another argument to
`harness_rtpulpfecdec` to set `error-after` on construction but that's
slightly more long-winded. so I went for this approach instead.
Fixes#532, even though that's already closed.
The initial mission statement for this test was:
* demonstrate usage of the request-aux-* signals in rtpbin
* test the rtx elements
We have examples that serve the first use case, and better
(harnessed) tests for the second use case.
This test is slow and racy, it served its purpose but can now
be removed.
Fixes#533
When the EOS event is received, run all timers immediately and avoid
pushing the EOS downstream before this has been run. This ensures that
the lost packet statistics are accurate.
The teardown of the pads checks the refcount, but there are timers
inside the jitterbuffer that can push things, so if we're not lucky,
things could be pushed while the pads are being shut down. Putting the
jitterbuffer to NULL first avoids this.
