Ensures the test can run on systems without alsa (or any audio output for
that matter), and will avoid people running build slaves wondering what
the hell was beeping during the night :)
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 example demonstrates how to use rtpbin with retransmission (rtx)
elements set in the place of rtpbin's "aux" elements in order to
enable RTP retransmission according to the rules of RFC4588.
It shows how to use "set-aux-receive" and "set-aux-send"
properties of rtpbin to set rtprtxsend and rtprtxreceive
Build 2 pipelines, one for rtpbin as a sender and one for
rtobin as a receive. Then transmit an audio stream.
It also drops some packets to activate restransmission and
check they are actually retransmited.
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.
Keep track of elements that are added to multiple sessions and make sure
we only add them to the rtpbin once and that we clean them when no
session refers to them anymore.
This test checks that when we have multiple internal sender sources
in rtpsession, SRs contain RBs for every other sender source, and that
they are included roundrobin when they exceed ST_RTCP_MAX_RB_COUNT,
which is the max number of RBs that can fit in a SR.
The parser can accept input that is not completely specified. Use the
ACCEPT_INTERSECT flag on the sinkpad to tweak the acceptcaps function to
check for intersection only. This allows us to proxy downstream
constraints while still allowing non-subset caps as input.
We can then also remove the appended template caps workaround.
Make a unit-test to check the new feature.
This reverts commit 26040ee38c
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=705024
These new tests send a tag event before seding the buffer. Tested case are an
empty tag list, a tag list with orientation-180 set and an invalid orientation value.
https://bugzilla.gnome.org/show_bug.cgi?id=719497
Don't reset the expected output seqnum when clearing the pt map because this
could stall the jitterbuffer forever.
Add a unit test for this.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=709800
Silencing this warning:
elements/souphttpsrc.c:533:14: error: comparison between ‘SoupKnownStatusCode’ and ‘enum <anonymous>’ [-Werror=enum-compare]
if (status != SOUP_STATUS_OK && !send_error_doc)
With gcc 4.8.2 (debian)
Store both DTS and PTS on buffers.
Make a queue for srcpad events.
Activate pads after linking so that we don't get RECONFIGURE events.
Add test for retransmission.
When we have a large number of missing packets, generate one lost event for all
the packets that have no chance of being pushed out in time.
Fix and activate unit test for large gaps.
In 0.10 elements would post tag messages on the bus
directly, and rganalysis would only post a tag message
when it changed tags. In 1.0, only sinks post tag
messages when they receive the serialised tag event.
This means that we get an additional tag message on
the bus now where we didn't expect one before.
https://bugzilla.gnome.org/show_bug.cgi?id=695090
The "message" property has been replaced by "post-messages".
Pre-patch output:
(test_spectrum:23101): GLib-GObject-WARNING **: g_object_set_valist:
object class `GstSpectrum' has no property named `message'
New spectrum message, endtime 0:00:00.100000000
(test_spectrum:23101): GStreamer-CRITICAL **:
gst_value_list_get_value: assertion `GST_VALUE_HOLDS_LIST (value)' failed
[...]
Post-patch:
New spectrum message, endtime 0:00:00.100000000
band 0 (freq 400): magnitude -65.988777 dB phase 1.533397
band 1 (freq 1200): magnitude -65.545563 dB phase -0.780900
band 2 (freq 2000): magnitude -64.791946 dB phase -0.799611
band 3 (freq 2800): magnitude -64.556175 dB phase -0.063615
[...]
https://bugzilla.gnome.org/show_bug.cgi?id=704179
The previous implementation had the formatting of SDP attributes happen
in each RTP payloader, now instead the constituent values are propagated
as caps fields. This allows for applications to do SDP offer/answer
based on caps negotiation.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700748
