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.
Always wait with starting the RTCP thread until either a RTP or RTCP
packet is sent or received. Special handling is needed to make sure the
RTCP thread is started when requesting an early RTCP packet.
We want to wait with starting the RTCP thread until it's needed in order
to not send RTCP packets for an inactive source.
https://bugzilla.gnome.org/show_bug.cgi?id=795139
This mode is useful for muxers that can take a long time to finalize a
file. Instead of blocking the whole upstream pipeline while the muxer is
doing its stuff, we can unlink it and spawn a new muxer+sink combination
to continue running normally.
This requires us to receive the muxer and sink (if needed) as factories,
optionally accompanied by their respective properties structures. Also
added the muxer-added and sink-added signals, in case custom code has to
be called for them.
https://bugzilla.gnome.org/show_bug.cgi?id=783754
If obtain_internal_source() returns a source that is not internal it
means there exists a non-internal source with the same ssrc. Such an
ssrc collision should be handled by sending a GstRTPCollision event
upstream and choose a new ssrc, but for now we simply drop the packet.
Trying to process the packet further will cause it to be pushed
usptream (!) since the source is not internal (see source_push_rtp()).
https://bugzilla.gnome.org/show_bug.cgi?id=795139
