Fixes the following error when building in osx.
error: implicit conversion from enumeration type
'GstJPEG2000Colorspace' to different enumeration type
'GstJPEG2000Sampling'
After seeking in aiff files the information about the data end offset is
discarded, leading to audio artifacts with metadata chunks at the end of
a file.
This patch retains the end offset information after a seek event.
https://bugzilla.gnome.org//show_bug.cgi?id=769376
This reverts commit 947656cfd2.
This makes all dash seeking tests fail. Needs more testing to fully understand
what's going wrong. Revert ok'd by Sebastian
timecodewait receives a timecode as an argument (either as string or as
GstVideoTimeCode - one is gst-launch-friendly and the other is code-friendly),
and it will drop all audio and video buffers until that timecode has been
reached.
https://bugzilla.gnome.org/show_bug.cgi?id=766419
We don't need to call the latter at all as we're definitely in this period and
the segment is selected via the SIDX.
This is especially important when doing SNAP seeks, as otherwise we would
always start from the beginning of the period (usually 0) again.
After the check in line 1,111, media->uri can't be NULL. So the two checks
for GST_HLS_MEDIA_TYPE_CLOSED_CAPTIONS are the same, removing the redundant
one which goes to cc_unsupported.
CID 1364752
When draining a program, we might send a newsegment event on the pads
that are going to be removed (and then the pending data).
In order to do that, calculate_and_push_newsegment() needs to know
what list of streams it should take into account (instead of blindly
using the current one).
All callers to calculate_and_push_newsegment() and push_pending_data()
can now specify the program on which to act (or NULL for the default
one).
Create an output stream for each media when alternate renditions
are present. Update the manifests for all those streams, and
make sure that typefinding is still done for files smaller than 2KB
such as small WebVTT files.
When fetching a byte-region from a server resource,
adjust the downstream buffer offsets so that downstream
doesn't know. This is because id3demux insists on the
first offset being 0. Later we might strip ID3 headers
entirely and this will be unneeded.
Modify playlist updating to track information across updates
better, although still hackish.
When connection_speed == 0, choose the default variant
not the first one in the (now sorted) variant list, as that
will have the lowest bitrate.
Make M3U8 and GstM3U8MediaFile refcounted. The contents
of it and GstM3U8MediaFile are pretty much immutable
already, but if we make it refcounted we can just
return a ref to the media file from _get_next_fragment()
instead of copying over all fields one-by-one, and then
copying them all into the adaptive stream structure fields again.
Move state from client into m3u8 structure. This will
be useful later when we'll have multiple media playlists
being streamed at the same time, as will be the case with
alternative renditions.
This has the downside that we need to copy over some
state when we switch between variant streams.
The GstM3U8Client structure is gone, and main/current
lists are not directly in hlsdemux. hlsdemux had as
many CLIENT_LOCK/UNLOCK as the m3u8 code anyway...
When connect to qmlglsrc, x11 event loop will be replace by qt event loop
which will cause the window cannot receive event from xserver, such as resize
https://bugzilla.gnome.org/show_bug.cgi?id=768160
Makes infinitely more sense and implementation were expecting that behaviour
anyway and would enter a resize, draw, resize, draw, ... cycle instead of only
resizing once.
Add a test of the gst_mpd_client_get_maximum_segment_duration() function
to check that it first checks the MPD@maxSegmentDuration and then falls
back to checking all of the segment durations.
https://bugzilla.gnome.org/show_bug.cgi?id=753751
