Fixes the following error when building in osx.
error: implicit conversion from enumeration type
'GstJPEG2000Colorspace' to different enumeration type
'GstJPEG2000Sampling'
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
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
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...
The gst_dash_demux_get_live_seek_range () function returns a stop value
that is beyond the available range. The functions
gst_mpd_client_check_time_position() and
gst_mpd_client_get_next_segment_availability_end_time() in
gstmpdparser.c include the segment duration when checking if a segment
is available. The gst_dash_demux_get_live_seek_range() function
in gstdashdemux.c ignores the segment duration.
According to the DASH specification, if maxSegmentDuration is not present,
then the maximum Segment duration is the maximum duration of any Segment
documented in the MPD.
https://bugzilla.gnome.org/show_bug.cgi?id=753751
There's no need for the jump to an extra thread in most cases, especially
when relying solely on a shader to render. We can use the provided
render_to_target() functions to simplify filter writing.
Facilities are given to create fbo's and attach GL memory (renderbuffers
or textures). It also keeps track of the renderable size for use with
effective use with glViewport().
Don't clear decryption state immediately after
initialising it in the start_fragment. Don't clear
the state of all streams when we want to only clear
the current stream.
https://bugzilla.gnome.org//show_bug.cgi?id=768757
Add demuxer instance-wide decryption key cache. The current and
last key url are per-stream, so make a shared cache. Move the
decryption handling into the stream object, and use the shared
cache for the keys.
Prepare hlsdemux for more than one single stream. Currently hlsdemux
assumes there'll only ever be one stream and most of the stream-specific
state is actually in the hlsdemux structure. Add a stream subclass
instead and move some stream-specific members there instead.
In this mode, we let WebRTC Audio Processing figure-out the delay. This
is useful when the latency reported by the stack cannot be trusted. Note
that in this mode, the leaking of echo during packet lost is much worst.
It is recommanded to use PLC (e.g. spanplc, or opus built-in plc).
In this mode, we don't do any synchronization. Instead, we simply process all
the available reverse stream data as it comes.
