Canceling the accept/select happens when the source is shut down. This is
not an error and the GST_FLOW_ERROR causes problems when only part of the
pipeline is shut down.
https://bugzilla.gnome.org/show_bug.cgi?id=731567
We are scaling from a unit in microseconds to a unit in ((1 << 32) per seconds).
We therefore scale the microseconds values by:
value of a second in the target unit (1 << 32)
--------------------------------------------------------------
value of a second in the origin format (1 000 000 microsecond)
When playing RTSP streams there will be one decodebin per stream. If some of
them fail because of a missing plugin we should not fail completely but play
the supported streams at least.
https://bugzilla.gnome.org/show_bug.cgi?id=730868
A simple '&' is not sufficiant. With mmapping_flags == PROT_READ and
prot == PROT_READ|PROT_WRITE the check produces the wrong result.
Change the check to make sure that prot is a subset of mmapping_flags.
https://bugzilla.gnome.org/show_bug.cgi?id=730559
When a pipeline using alsasink and push mode upstream fails
to preroll, the following state will be the case:
- A loop upstream will be PAUSED, pushing a first buffer
- alsasink will be READY, pending PAUSED, because async
On error, the pipeline will switch to NULL. alsasink is in
READY, so goes to NULL immediately. It zeroes its cached
caps. Meanwhile, the upstream loop can cause a caps query,
conccurent with the state change. This will use those cached
caps. If the zeroing happens between the NULL test and the
dereferencing, GStreamer will critical down in the GstValue
code.
Since it appears that such a gap between states (PAUSED
and pushing upstream, and NULL downstream) is expected, we
need to protect the read/write access to the cached caps.
This fixes the critical.
See https://bugzilla.gnome.org/show_bug.cgi?id=731121
This lets oggdemux determine they are not delta units, and removes
spurious per packet warnings about being unable to determine the
packet's keyframeness.
Aggregate buffering messages to only post the lower value
to avoid setting pipeline to playing while any multiqueue
is still buffering.
There are 3 scenarios where the entries should be removed from
the list:
1) When decodebin is set to READY
2) When an element posts a 100% buffering (already implemented)
3) When a multiqueue is removed from decodebin.
For item 3 we don't need to handle it because this should only
happen when either 1 is hapenning or when it is playing a
chained file, for which number 2 should have happened for the
previous stream to finish
https://bugzilla.gnome.org/show_bug.cgi?id=726423
With lots of shared memory instances (e.g. created by a RTP payloader) the
overhead of duplicating the file descriptor and creating extra mappings is
significant. To avoid this, the parent memory maps the whole region and the
shared copies just reuse the same mapping.
https://bugzilla.gnome.org/show_bug.cgi?id=730441
Add a read source on write socket when lost tunnel.
To be able to detect when clint closes get channel.
This is already done in gst_rtsp_source_dispatch_write but
only when the queue is empty.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730368
Otherwise we might end up inside the callback without having stored
the probe id... then try to remove that probe (not!) from the callback
and wait forever for the pad to unblock.
By re-using the uri argument for storing local data, we could end up in
a situation where we would free uri ... which would actually be the
string passed in argument.
Instead explicitely use a local variable. Fixes double-free issues.
CID #1212176
