This functions more or less identically to using a media out value, but the
desired speed is preserved through block length changes, even if the block's
length is reduced to zero (i.e. no rounding errors).
frames
Previously, when the video renderer received a dirty cache signal, it would
proceed to extract all frames from the range and queue them. However, this could
be extremely slow for long ranges since it had to iterate through the entire
range and calculate the individual frames it contained. Now, we use the same
range combining system as audio and automatically calculate the next frame
within the range only when necessary. Essentially the same work, but split up
over time and done only when needed leading to no discernible UI pause when
invalidating cache.
Previously the audio output manager was a "hybrid device" that acted as a pull
device for audio devices and as a push device and device proxy for the rest of
the application. This approach turned out to be flawed, particularly in the
frequent opening and closing of the audio device (extremely slow). Since we
use both pulling (for constant playback) and pushing (audio scrubbing, short
bursts of sound), it needed a similar but different approach. This approach
will switch the output device from push mode (default) to pull mode (during
playback) only when necessary resulting in far less UI lag (basically
unnoticeable now) than the previous approach.
The renderer backend can now distinguish between jobs. Previously if two jobs
of the same frame were started (which is legal if the user made a change while
frames were still being rendered), an earlier job in some situations could
finish AFTER a later job, and the backend would have no way of distinguishing
between them. This meant a frame could be erroneously set to an old value
rather than the newest. This commit introduces job identification so that old
jobs are automatically discarded.
Since updating all input values per sample could be performance costly,
we now only update inputs that are either keyframing or connected to
another node
This mostly builds on the keyframing we already set up for video, but the
audio renderer will now appropriately updated keyframe inputs per sample in
accordance with keyframe values.