This was never reset when going from PAUSED->READY and resulted
in encoders being not reusable after EOS. They just rejected any
buffer because they received EOS in their previous life.
The flag wasn't used anywhere except for rejecting buffers after
EOS, and this is now handled by GstPad directly.
vorbis_reorder_map is defined for eight channels max. If we have more
than eight channels, it's the application which shall define the order.
Since we set audio position to none, we just interleave all the channels
without any particular reordering.
https://bugzilla.gnome.org/show_bug.cgi?id=737742
The "iradio-mode" property used to have a default FALSE value in HTTP
source elements but now it should default to TRUE or just do not exist
as a property so it is not really needed to set it any more in
uridecodebin.
Apart from that this code could've never worked as uridecodebin looks for a
string-typed iradio-mode property, but it's a boolean in all sources.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725383
The spec mentions a version of the MPEG-2 frame with a base frame and
extension frame. I don't have IEC 13818-3 to figure out what that is,
and don't see any references in search results, so it's a FIXME for now.
https://bugzilla.gnome.org/show_bug.cgi?id=736797
It is better to use storel to splat the variable into the destination.
ORC doesn't know when a variable is last written to so it can't yet optimize
away the copy operation.
Support lanczos scaling method for NV12 and NV21 formats.
Scale the 'Y' plane and scale 'NV' plane.
Implementation for submethods - int16, int32, float and double
https://bugzilla.gnome.org/show_bug.cgi?id=737400
Move the conversion code used in videoconvert to the video library
and expose a simple but generic API to do arbitrary conversion. It can
currently do colorspace conversion but the plan is to add videoscale to
it as well.
See https://bugzilla.gnome.org/show_bug.cgi?id=732415
When playing chained data the audio ringbuffer is released and
then acquired again. This makes it reset the segbase/segdone
variables, but the next sample will be scheduled to play in
the next position (right after the sample from the previous media)
and, as the segdone is at 0, the audiosink will wait the duration
of this previous media before it can write and play the new data.
What happens is this:
pointer at 0, write to 698-1564, diff 698, segtotal 20, segsize 1764, base 0
it will have to wait the length of 698 samples before being able to write.
In a regular sample playback it looks like:
pointer at 677, write to 696-1052, diff 19, segtotal 20, segsize 1764, base 0
In this case it will write to the next available position and it
doesn't need to wait or fill with silence.
This solution is borrowed from pulsesink that resets the clock to
start again from 0, which makes it reset the time_offset to the time
of the last played sample. This is used to correct the place of
writing in the ringbuffer to the new start (0 again)
https://bugzilla.gnome.org/show_bug.cgi?id=737055
Move the assert to the error handling block at the end of the function so the
the logging is still triggered. Reword the logging slightly and add another
comment to hint what went wrong.
Fixes#737138
The allocation query failure doesn't mean that the negotiation
has failed as the element can allocate buffers itself.
Instead, only fail if the pads are flushing and the allocation
query failed.
https://bugzilla.gnome.org/show_bug.cgi?id=735844
audioresample and videoscale is something the application will have to do if
required, but we can at least help here by adding the
audioconvert/videoconvert elements.
https://bugzilla.gnome.org/show_bug.cgi?id=735748
