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The time of the first RTCP packet is semi-random, so
sometimes it was produced before enough packets from
the second SSRC were received. First drop queued RTCP
packets, then advance the clock enough to ensure
that at least one new RTCP packet is produced.
https://bugzilla.gnome.org/show_bug.cgi?id=750731
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Change the test to verify the read-index for multiple messages per buffer.
See https://bugzilla.gnome.org/show_bug.cgi?id=754144
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The accept-caps query just does a shallow check at the current
element while at this test we want it to also look at downstream.
So use caps query there.
https://bugzilla.gnome.org/show_bug.cgi?id=753623
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Makes it easier to differentiate from other modules states unit test
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1) Tests that using dynamic PT instead of the default ones work
2) If we ever decide to change the codec here we don't need to
worry about change the PT for the default one of the new codec
in the test
https://bugzilla.gnome.org/show_bug.cgi?id=746445
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The RTP PT for alaw is 8.
Less than 50 packets are received in the length of this test so it
would never drop a buffer or would drop only the last buffer and
it would fail sometimes when the received wouldn't receive the
retransmission packet in time.
https://bugzilla.gnome.org/show_bug.cgi?id=746445
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Some of the subtitle chunks will have embedded
NUL-terminators (last three), some don't (first three),
some will have markup, some won't, some will be valid
UTF-8 (all but last), some won't (last stanza).
https://bugzilla.gnome.org/show_bug.cgi?id=752421
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Do this by not putting the ssrc from the src pads in the caps used to
probe other sinkpads, and then intersecting with it later.
https://bugzilla.gnome.org/show_bug.cgi?id=752491
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Replace static constants with macros to make gcc happy
CC elements/elements_rtpjitterbuffer-rtpjitterbuffer.o
elements/rtpjitterbuffer.c:387:1: error: initializer element is not constant
static const GstClockTime PCMU_BUF_DURATION = PCMU_BUF_MS * GST_MSECOND;
^
elements/rtpjitterbuffer.c:388:1: error: initializer element is not constant
static const guint PCMU_BUF_SIZE = 64000 * PCMU_BUF_MS / 1000;
^
elements/rtpjitterbuffer.c:390:5: error: initializer element is not constant
PCMU_BUF_CLOCK_RATE * PCMU_BUF_MS / 1000;
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Fix indent on this file and break some comment lines into two to make
it fit 80 chars per line
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The amount of time that is completely expired and not worth waiting for,
is the duration of the packets in the gap (gap * duration) - the
latency (size) of the jitterbuffer (priv->latency_ns). This is the duration
that we make a "multi-lost" packet for.
The "late" concept made some sense in 0.10 as it reflected that a buffer
coming in had not been waited for at all, but had a timestamp that was
outside the jitterbuffer to wait for. With the rewrite of the waiting
(timeout) mechanism in 1.0, this no longer makes any sense, and the
variable no longer reflects anything meaningful (num > 0 is useless,
the duration is what matters)
Fixed up the tests that had been slightly modified in 1.0 to allow faulty
behavior to sneak in, and port some of them to use GstHarness.
https://bugzilla.gnome.org/show_bug.cgi?id=738363
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Plus new unit test.
https://bugzilla.gnome.org/show_bug.cgi?id=752112
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Also make it so that the mtu is always set if specified, not
only in case of the rather weird bufferlist test code path.
This allows us to easily make the payloader fragment a payload
across multiple output packets by setting a small MTU on it.
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Implementation according to RFC 4587.
Payloader create fragments on MB boundaries in order to match MTU size
the best it can. Some decoders/depayloaders in the wild are very strict
about receiving a continuous bit-stream (e.g. no no-op bits between
frames), so the payloader will shift the compressed bit-stream of a
frame to align with the last significant bit of the previous frame.
Depayloader does not try to be fancy in case of packet loss. It simply
drops all packets for a frame if there is a loss, keeping it simple.
https://bugzilla.gnome.org/show_bug.cgi?id=751886
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over all timers
If we have a clock, update "now" now with the very latest running time we have.
If timers are unscheduled below we otherwise wouldn't update now (it's only updated
when timers expire), and also for the very first loop iteration now would otherwise
always be 0.
Also the time is used for the timeout functions, e.g. to calculate any times
for the next timeouts and we would otherwise pass too old times there.
https://bugzilla.gnome.org/show_bug.cgi?id=751636
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Let's have an example with modern codecs.
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In presence of gaps, qtdemux will emit multiple segments. The
second segment start should match the CTTS.
https://bugzilla.gnome.org/show_bug.cgi?id=751361
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https://bugzilla.gnome.org/show_bug.cgi?id=751242
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https://bugzilla.gnome.org/show_bug.cgi?id=740575
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The mp4 muxer now writes a place-holder mdat as a free
atom followed by a 0-byte mdat that covers the rest of the
file, making it possible to rewrite it as 64-bit, or leave
it as-is if nothing else is written afterward
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caps or property
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Makes source code smaller, and ensures we go through common initialization
path (like the one that sets up XML unit test output ...)
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https://bugzilla.gnome.org/show_bug.cgi?id=746543
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See also https://bugzilla.gnome.org/show_bug.cgi?id=738237
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Mostly gst-launch -> gst-launch-1.0
Use autovideosink/autoaudiosink more often.
Sprinkle some converters here and there.
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The calculations were a bit off everywhere, even before the changes done
recently to the delay for RTX of expected future packets. It only worked by
accident, but now the calculations are all correct again. Hopefully.
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Make sure the test environment is set up.
https://bugzilla.gnome.org//show_bug.cgi?id=747624
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Adding an option in the equalizer demo to make the number of bands selectable.
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Switch the equalizer-nbands demo to use uridecodebin, so users can listen to
something more pleasant than white noise. If anybody misses the white noise
a uri handler to audiotestsrc can be used.
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Rename variable name to make it more readable, add comments for the three
scales created per block, and set the window title.
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A buffer overread.
https://bugzilla.gnome.org/show_bug.cgi?id=747554
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Both input streams in this test have a segment.start = 10s, so
output should start from 0 anyway.
Another test has both starting at non-0 segments, but the running
time of both streams should still start from 0
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Adds a new simple test that verifies that data is properly muxed
and preserved. PTS, DTS, duration and caps are verified.
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Added test suite for alpha element with test cases
1. alpha
2. chroma keying
https://bugzilla.gnome.org/show_bug.cgi?id=747595
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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.
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Needed by the new automake test runner
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Sometimes we can get segment-done before we got async-done. If we waited
for async-done only, the segment-done would be dropped and we would wait
forever for it a few lines below.
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Also verify that the multifilesink file messages are being correctly
posted to the bus
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Use a GstBus and wait for EOS to finish the tests instead of
relying on sleeping
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When not in fast-start or fragmented mode, we need to be able
to rewrite the size of the mdat atom, or else the output just
won't be playable - the mdat placeholder with size == 0 will
cover the rest of the file, including any moov atom we write out.
https://bugzilla.gnome.org/show_bug.cgi?id=708808
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726416
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See https://bugzilla.gnome.org/show_bug.cgi?id=745539
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Just avoid using the deprecated function entirely,
it's easy enough. Defining the macro is not enough.
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gdk_pixbuf_new_from_inline() has been deprecated in favour
of GResource.
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The test had a function to print the error, but was not parsing it.
This was causing warning about dbg_info being used uninitialized. If
the test was testing any errors, this would have crashed.
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