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
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.
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
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
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
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.
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.
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
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.
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.
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
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.
Implement 2 new elements - splitmuxsink and splitmuxsrc.
splitmuxsink is a bin which wraps a muxer and takes 1 video stream,
plus audio/subtitle streams, and starts a new file
whenever necessary to avoid overrunning a threshold of either bytes
or time. New files are started at a keyframe, and corresponding audio
and subtitle streams are split at packet boundaries to match
video GOP timestamps.
splitmuxsrc is a corresponding source element which handles
the splitmux:// URL and plays back all component files,
reconstructing the original elementary streams as it goes.
Our ones were expired. The new ones were copied from libsoup's
tests files.
Also sets the property to use our own cert to validate the
server, otherwise the default system certs would be used
and it would fail.
They should always be built, while the speex elements are not.
Need to check for a smaller number of buffers then (7->4) because
speexenc will add 3 header buffers while alawenc will just output
as many buffers as it receives as input.
https://bugzilla.gnome.org/show_bug.cgi?id=742098
rtpmux behaves like a funnel in that it forwards whatever upstream is
sending buffers. So setting proxy caps doesn't make sense as the
upstream don't have to have compatible caps, thus resulting in an empty
caps set as a result of a caps query. Instead set fixed caps just
as funnel does.
https://bugzilla.gnome.org/show_bug.cgi?id=738722
Actually look for error messages on the bus and fail if there
is one before the EOS message. Disable pull mode test which is
pointless as long as matroskaparse only supports push mode
(pull mode support has not been ported over to 1.0).
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
As we now replace the local RTPSource on a conflict, it's no longer possible
to keep local conflicts in the RTPSource, so they instead need to be kept
in the RTPSession.
Also fix the rtpcollision test to generate multiple collisions instead of
one by change the address, as otherwise we detected that it was a single one.
From libsoup docs:
Prior to 2.44 SoupStatus was called SoupKnownStatusCode,
but the individual values have always had the names they
have now.
Fixes:
https://bugzilla.gnome.org/show_bug.cgi?id=727329
With years the amount of ifdef have grown up and we are not even sure if the
old code path compiles. Each time we need to update the v4l2 framework to add
the new feature, we break compilation on older kernel. With exception of two
controls in the video orientation control, this patch get rid of all ifdef by
including the latest version of videodev2.h inside GStreamer.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723446
