1. The spec indicates that the notification should occur near the end of
'setting the description' processing
2. The current location with the drop of the lock could cause the 'check
if negotiation is needed' logic to execute and become confused about
the state of the webrtcbin's current local descriptions.
In the bad case, the following assertions could be hit:
g_assert (trans->mline < gst_sdp_message_medias_len (webrtc->current_local_description->sdp));
g_assert (trans->mline < gst_sdp_message_medias_len (webrtc->current_remote_description->sdp));
Moving the signalling state change later in the set description task
means that checking for a renegotiation will early abort as the
signalling state is not STABLE before the session description and
transceivers have been updated.
The start of each segment is relative to the Period start, minus
the presentation time offset.
As specified in section 5.3.9.6 of the MPEG DASH specification:
The value of the @t attribute minus the value of the
@presentationTimeOffset specifies the MPD start time of
the first Segment in the series.
dashdemux was not taking account of presentationTimeOffset and in
some methods was not taking into account the Period start time.
This commit modifies the segment->start value to always be
relative to the MPD start time (zero for VOD,
availabilityStartTime for live streams). This makes all uses of
the segment list consistent.
Fixes#841
If both data channels become ready simultaneously, then the two integer
read-add-update cycles can execute concurrently and only ever increment
once instead of the required twice. Use an atomic add instead.
It is very possible for badly behaving signalling or peers to send
us ICE candidates before we receive an SDP. While we had consideration
for that on the first set SDP, subsequent SDP's could result in
misconfigured ICE transports. Expand the previous code to also take
into account reconfigurations.
Limitations:
- No transport changes at all (ICE, DTLS)
- Codec changes are untested and probably don't work
- Stream removal doesn't remove transports (i.e. non-bundled transports
will stay around until webrtcbin is shutdown)
- Unified Plan SDP only. No Plan-B support.
Currently, if one was to set -Dhls-crypto to either libgcrypt or openssl
instead of auto, the following lines would fail because hls_crypto_dep is not
yet set:
if not hls_crypto_dep.found() and ['auto', 'libgcrypt'].contains(hls_crypto)
if not hls_crypto_dep.found() and ['auto', 'openssl'].contains(hls_crypto)
Instead, change "if not hls_crypto_dep.found()" to "if not have_hls_crypto"
which fixes the error.
Use a timeout to limit that amount of time we wait after the compositor
for the initial configure event. Compositor are support to emit a
configure event before any wl_buffer can be attached. The problem is
that Weston strongly enforce this, while gnome-shell simply does not
emit such an event.
When buffer is used by compositor, we don't need attach it and hold one
more reference. Just check used_by_compositor, just return if it is true.
Assert error log is not need, this is normal behavior.
... when it has not yet been connected to.
Also, a condition variable is not a semaphore, so a lock/wait/unlock
sequence is inherently racy without any state checking. So switch to
a different lock and check the intended state.
Some GIR annotations were incorrect or even missing. The former isn't
good for bindings, while the latter is especially annoying for signal
handlers, as that means your arguments will get the wrong names in the
rendered documentation.
GST_VIDEO_BUFFER_FLAG_INTERLACED and GST_VIDEO_BUFFER_FLAG_TFF
flags are needed when processing SCTE 20 closed captions for an interlaced
stream, when we need to convert back to analog, in which case we need to match
the caption to the top or bottom field
... if (x265 version >= 1.9) requirement is satisfied.
The SEI messages were supported since x265 version 1.8
but there was API change from version 1.9
(contentLightLevelInfo was renamed to maxCLL and maxFALL)
This change makes it possible to create more than just 5 webrtc
data channels. The maximum number of data channels is exactly
DEFAULT_NUMBER_OF_SCTP_STREAMS / 2, therefore the limit is now
512.
Use proper API to flush libass events when we do
a flushing seek, and also do it in FLUSH_STOP
rather than FLUSH_START, so we can be sure
streaming has stopped.
Fixes seeking back in time.
Something seems to have changed in libass that
renders the old manual way of flushing events
ineffective and libass then seems to ignore
timestamps that are older than the ones last
seen then if we do it the old way.
Fixes#916
With latest XDG shell, we need to fait for the surface to have been
configured before we can attach a buffer to it. This is being enforce by
Weston with an error.
Fixes#933
Fixes the following error.
gstccconverter.c:677:7: error: variable 'len' is used uninitialized whenever 'if' condition is false
[-Werror,-Wsometimes-uninitialized]
if (flags & 0x40) {
^~~~~~~~~~~~
gstccconverter.c:698:10: note: uninitialized use occurs here
return len;
^~~
gstccconverter.c:677:3: note: remove the 'if' if its condition is always true
if (flags & 0x40) {
^~~~~~~~~~~~~~~~~~
gstccconverter.c:572:12: note: initialize the variable 'len' to silence this warning
guint len;
^
= 0
Prior to this, cccombiner stopped consuming video buffers when
data wasn't arriving on its caption pad. In a live situation,
when aggregator is timing out we should still output whatever
video buffers are present, even if no caption buffers can be
aggregated with them.
Since the addition of BUNDLE support, the pads and the transceivers
share a single transport stream. When getting stats from the stream,
filter by the ssrc of the current pad to avoid merging the stats for
different pads.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/889
It is necessary to implement this vmethod, as when the src pad
is marked as reconfigure, the base class will reset to src caps,
and the default update_src_caps simply queries the caps allowed
downstream without taking into account the caps set by
gst_aggregator_set_src_caps.
