And only set low-percent/high-percent if not using downloadbuffer, just
like in old uridecodebin. using the watermark based buffering causes
playback to hang never finish buffering with downloadbuffer.
With both audiorate and videorate, it seems more sensible to apply rate
adjustments after the first buffer appears. For example, with v4l2src,
there is often a small delay before the first video buffer turns up, and
this can cause a stuttery start because of videorate trying to ensure a
perfect stream.
Those multiqueue are the ones dealing with adaptive demuxers. They should
have a time limit set so that they don't end up buffering too much data.
They would previously be set with no limits at all, which would cause them
to grow indefinitely until downstream blocks.
And monitor no_more_pads.
With live sources such as rtsp, uridecodebin only creates its
child decodebins between PAUSED and PLAYING.
This means that the ASYNC_DONE it posts when getting NO_PREROLL
in its change_state method gets immediately propagated by the
GstBin parent class, as opposed to a situation where a
decodebin has been added to it already, and has posted ASYNC_START.
The proposed solution, instead of simply waiting for ASYNC_DONE,
and finishing prematurely in that case, waits for three conditions
to be true:
* the uridecodebin needs to have emitted no_more_pads
* its current state must be PAUSED if not live, PLAYING otherwise
* There must be no "pending subtitle pads", ie pads where we haven't
received tags yet.
All these conditions are checked in the message handler, as we
post custom messages on it when we get subtitle tags or no_more_pads.
https://bugzilla.gnome.org/show_bug.cgi?id=783257
When the input is TRICKMODE_KEY_UNITS, we expect to only receive keyframes
which we want to decode/push immediately. Therefore don't queue them.
If upstream didn't send just keyframes (which is the ideal situation), two
different things can happen:
1) Either the subclass checks the segment flags and properly configures
the decoder implementation to only decode/output keyframes,
2) Or the subclass really decodes and outputs everything, in which case
the reverse frames will end up arriving "late" downstream (and will
be dropped). If upstream did properly send GOP in reverse order, we
still end up just showing keyframes (but at the overhead of decoding
everything).
https://bugzilla.gnome.org/show_bug.cgi?id=777094
gst_video_rate_flush_prev() ensures that the pushed buffer is writable
by calling gst_buffer_make_writable() on videorate->prevbuf.
In drop-only mode we always push buffers directly when they are received
from GstBaseTransform (gst_video_rate_transform_ip()) and do not keep them
around. GstBaseTransform already ensures that those buffers are
writable so there is no need to do it twice.
This change saves us from copying buffers in drop-only mode as we no longer
calls gst_buffer_make_writable() with a buffer having a refcount of 2
(one ref owned by GstBaseTransform and one in videorate->prevbuf).
https://bugzilla.gnome.org/show_bug.cgi?id=780767
Always put multiview-caps onto the output caps, assuming
mono if we've got no other information. It's still easy for
downstream elements to override using a capssetter or event
probe if desired.
https://bugzilla.gnome.org/show_bug.cgi?id=776172
Child streams could have more accurate width/height or various other
information added. If they have the same name, they are likely to be the
same streams.
https://bugzilla.gnome.org/show_bug.cgi?id=782697
This is now needed as GstClock does not do that internally anymore,
because that broke bindings.
And mark the function correctly as (transfer full), which it already was
before.
https://bugzilla.gnome.org/show_bug.cgi?id=743062
