The functionality of all the tests was kept exactly the same. Some tests
were renamed:
test_push_forward_seq -> test_rtxsend_rtxreceive
test_drop_one_sender -> test_rtxsend_rtxreceive_with_packet_loss
test_drop_multiple_sender -> test_multi_rtxsend_rtxreceive_with_packet_loss
test_rtxreceive_data_reconstruction was testing that retransmitted
buffer produced by rtxsend was correctly transformed to the original
buffer by rtxreceive. Now we are checking for this in all the tests
where both rtxsend & rtxreceive are involved. That's why the test was
removed.
Commit #1018aa made rtprtxsend handle buffer lists, breaking
the test which probes for buffers, but not buffer lists.
Use a utility function to run the probe callback on each buffer
in the list in turn and remove any buffers that are dropped.
Fix the raciness by iterating on a condition instead of using the gmainloop.
Don't use the EOS as the target, otherwise the retransmission of the last
packets are lost. Also count the retranmissions requests that are dropped.
Check the condition before blocking on the GCond
https://bugzilla.gnome.org/show_bug.cgi?id=728501
Now with rtprtxsend pushing rtx buffers from a different thread,
this is necessary to ensure that the result of the test is deterministic.
This code makes use of GstCheck's global GMutex and GCond that are
being used inside GstCheck's sink pad chain() function in order
to synchronize with it.
Now with rtprtxsend pushing rtx buffers from a different thread,
this is necessary to ensure that the result of the test is deterministic.
This code makes use of GstCheck's global GMutex and GCond that are
being used inside GstCheck's sink pad chain() function in order
to synchronize with it.
Now with rtprtxsend pushing rtx buffers from a different thread,
this is necessary to ensure that the result of the test is deterministic.
This code makes use of GstCheck's global GMutex and GCond that are
being used inside GstCheck's sink pad chain() function in order
to synchronize with it.
This unit test verifies that the rtxsend element correctly maintains
a buffer of already transmitted rtp packets and that it can
re-transmit all of them correctly on demand. It also verifies
that the limit of this buffer (max-size-packets property) is respected.
Several senders / one receiver
Similar than test_drop_one_sender but with multiple senders
mixed through the funnel element.
It drops some packets and checks that they are retransmited
correctly.
Test for one sender / one receiver
Build the pipeline
videotestsrc ! rtpvrawpay ! rtprtxsend ! rtprtxreceive ! fakesink
and drop some buffers between rtprtxsend and rtprtxreceive
Then it checks that every dropped packet has been re-sent.
It also checks that not too much requests has been sent.
