When a new time segment is received upstream is going to restart
with a new atom. Make the neededbytes and todrop variables
reflect that to avoid waiting too much or dropping the
initial bytes that contain the header.
The adapter might have data remaining from the previous segment,
push it all before clearing the adapter and starting a new segment.
It can accumulate data if it had pushed and got not-linked, returning
immediately without processing all the data. Before starting a new
segment this data should be handled.
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
This reverts commit 05bd708fc5.
The reverted patch is wrong and introduces a regression because there
may still be time to receive some of the packets included in the gap
if they are reordered.
Avoids accumulating all samples from a fragmented stream that could
lead to a 'index-too-big' error once it goes over 50MB of data. It
could reach that before 2h of playback so it doesn't take that long.
As upstream elements are providing data in time format they should
be the ones that have more information about the full media index
and should be able to seek if possible.
upstream_newsegment isn't really clear on what it means, it is set
to TRUE when the upstream element sends a segment in TIME format, so
rename it to be more clear about it.
It is important to know this because it means that upstream has
a notion of time and qtdemux is likely being driven by an upstream
element that is reading from a higher level abstraction than a file,
such as a DASH, MSS or DLNA element.
In fragmented streaming, multiple moov/moof will be parsed and their
previously stored samples array might leak when new values are parsed.
The parse_trak and callees won't free the previously stored values
before parsing the new ones.
In step-by-step, this is what happens:
1) initial moov is parsed, traks as well, streams are created. The
trak doesn't contain samples because they are in the moof's trun
boxes. n_samples is set to 0 while parsing the trak and the samples
array is still NULL.
2) moofs are parsed, and their trun boxes will increase n_samples and
create/extend the samples array
3) At some point a new moov might be sent (bitrate switching, for example)
and parsing the trak will overwrite n_samples with the values from
this trak. If the n_samples is set to 0 qtdemux will assume that
the samples array is NULL and will leak it when a new one is
created for the subsequent moofs.
This patch makes qtdemux properly free previous sample data before
creating new ones and adds an assert to catch future occurrences of
this issue when the code changes.
This reverts commit d46631c5c7.
pad only handle EOS events but not EOS flow, and will push the buffer again
resulting in an assertion error. So we should not handle the buffer
and return EOS flow.
goom_core.c: In function 'goom_update':
goom_core.c:685:5: error: 'param2' may be used uninitialized in this function [-Werror=maybe-uninitialized]
goom_lines_switch_to (goomInfo->gmline2, mode, param2, amplitude, couleur);
^
goom_core.c:684:5: error: 'param1' may be used uninitialized in this function [-Werror=maybe-uninitialized]
goom_lines_switch_to (goomInfo->gmline1, mode, param1, amplitude, couleur);
^
https://bugzilla.gnome.org/show_bug.cgi?id=752053
endpos variable does not correctly understand in the
4.6.3 GCC version. So compile error appears when we do
compile rtph261pay using jhbuild.
This patch is fixed the compile error in 4.6.3 GCC version.
https://bugzilla.gnome.org/show_bug.cgi?id=751985
Draft 16 of "RTP Payload Format for VP8" states in section 4.2 that:
R: Bit reserved for future use. MUST be set to zero and MUST be
ignored by the receiver.
https://bugzilla.gnome.org/show_bug.cgi?id=751929
gstrtph261pay.c: In function 'gst_rtp_h261_pay_class_init':
gstrtph261pay.c:1003:17: error: variable 'gobject_class' set but not used [-Werror=unused-but-set-variable]
GObjectClass *gobject_class;
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
We always pushed one buffer into the adapter, then handled exactly that one
buffer and flushed it from the adapter. Now also don't memcpy() the actual
payload but just attach the input buffer's data to the output buffer.
This code still needs some serious refactoring/rewriting.
