Otherwise ssrc changes via rtpsession's (deprecated!) internal-ssrc property
are not possible anymore. rtpsession was now patched to only suggest an ssrc
if it makes sense to do so.
In 2.0 we should get rid of all the properties that are also negotiated via
caps, the code and behaviour is too confusing otherwise.
https://bugzilla.gnome.org/show_bug.cgi?id=749581
According to this section of the rfc.
https://tools.ietf.org/html/rfc5506#section-3.4.2
The validation should be updated to accept more types of RTCP
packages, with this mask change feedback packages will be also
accepted.
Change-Id: If5ead59e03c7c60bbe45a9b09f3ff680e7fa4868
Micro-optimisation: if the buffer consist of just one memory, we
know we have already mapped that memory to read the headers, so
no need to map it another time to get to the payload data, we
can just set up the payload data details right there and then
and avoid another map call in gst_rtp_buffer_get_payload().
Adds up when receiving RTP-payloaded raw video which can easily
be thousands of packets per frame.
Implement a chain_list function, which avoids lots of locking
compared to the default fallback implementation in GstPad.
We may also want to do some more sophisticated timestamp
tracking here at some point, but for now leave it up to the
jitterbuffer and/or subclasses (in case buffers in the
buffer list have no timestamp set on them, there may only
be a timestamp for the whole list on the first buffer).
This provides the exact same behaviour as the default
fallback implementation.
Summary:
So that the user can easily use the same encoding profile to render
with/without audio/video stream.
API:
gst_encoding_profile_is_disabled
gst_encoding_pofile_set_enabled
https://bugzilla.gnome.org/show_bug.cgi?id=749056
Instead of returning the first video meta found on a buffer, return the
one with the lowest id (which is usually the same thing, except on
multi-view buffers)
The original 0/1 framerate must still be allowed to be configured
on the upstream side of videorate, otherwise future caps renegotiation
is going to fail.
https://bugzilla.gnome.org/show_bug.cgi?id=750032
When a stream has a variable framerate, videorate calculates it and
forces it on the output caps. However, the code in _transform_caps()
currently also does that if the transform is going in the opposite
direction (GST_PAD_SRC), so during a renegotiation it tries to force
upstream to use the calculated framerate and it fails.
https://bugzilla.gnome.org/show_bug.cgi?id=750032
This part of pipeline is:
tee name=t ! visualizationbin ! streamsynchronizer name=s
t. ! s.
streamsynchronizer might block and it could starve the visualization
branch of the pipeline when it is enabled.
The visualization bin has queues internally but the other branch
that links the audiotee directly to the synchronizer is vulnerable
to block. Adding a queue between "t. ! s." fixes deadlocks.
https://bugzilla.gnome.org/show_bug.cgi?id=749676
Use g_utf16_to_utf8() instead of the more generic g_convert(), so
that we can extract text in UTF-16 format even on embedded systems
with crippled iconv support.
This code path is exercised by the id3demux test_unsync_v23
check in gst-plugins-good.
https://bugzilla.gnome.org/show_bug.cgi?id=741144
This reverts commit 76647f2710.
Avoiding pull mode activation is a feature regression, and
demuxers should always use pull mode where that is possible,
e.g. if there's an upstream queue2 with a ring buffer or
a download buffer.
This patch made reverse playback no longer possible over http.
If the goal is to minimise seeks, then that can still be done
by making the demuxer behave differently in pull mode if
the SEQUENTIAL flag is set. If there are bugs, like the demuxer
needlessly scanning the entire file on start-up in pull mode,
then those should be fixed instead.
https://bugzilla.gnome.org/show_bug.cgi?id=746010
