The basic idea is to write out the signal that came in with the same number of
channels as it had when it came in. Things get a bit more complicated when
one output block may be derived from more than one input block, each having
different numbers of channels. When this happens, the input blocks with fewer
channels are upmixed, so as not to lose (or distort) any signal in the block
with more channels.
HRTFPanner no longer uses exponential decay (with time constant 20ms) for
delay changes, but a smoother linear transition during cross-fade time (~45ms).
--HG--
rename : content/media/webaudio/DelayProcessor.cpp => content/media/webaudio/DelayBuffer.cpp
rename : content/media/webaudio/DelayProcessor.h => content/media/webaudio/DelayBuffer.h
extra : rebase_source : 18453d631779cd7d0672b5325e110b107ab4237d
The azimuth calculation in the Web Audio spec becomes undefined in this
situation as it requires normalizing a zero vector. The panning effect should
not depend on azimuth when the source is directly above the listener (because
position does not depend on azimuth), but the specified "equalpower" panning
model does depend on azimuth even at this elevation. Setting azimuth to zero
produces the same result as if the normalized projection were replaced with a
zero vector.
--HG--
extra : transplant_source : f%A4h%CB7%7Bp%87%AE%09%9F%2Cu%D7%CD%9D%5E%A8%EC%0D
Normalizing the AudioListener front orientation vectors before taking their
cross product avoids the possibility of overflow.
The panning effect and the azimuth and elevation calculation in the Web Audio
spec becomes undefined with linearly dependent listener vectors, so keep
existing state in these situations.
PannerNode orientation is normalized for consistency, but zero is permitted
for this vector because the sound cone algorithm in the Web Audio specifies
behavior for this case.
--HG--
extra : transplant_source : %DA%C7e%E7%90%14%AF%EA%08%94x%C1%A2g%F1%9A%EE%16%EB%29
Here is what this patch does:
* Got rid of the JSBindingFinalized stuff
* Made all nodes wrappercached
* Started to hold a self reference while the AudioBufferSourceNode is playing back
* Converted the input references to weak references
* Got rid of all of the SetProduceOwnOutput and UpdateOutputEnded logic
The nodes are now collected by the cycle collector which calls into
DisconnectFromGraph which drops the references to other nodes and destroys the
media stream. Note that most of the cycles that are now inherent in the
ownership model are between nodes and their AudioParams (that is, the cycles
not created by content.)
Here's what this patch does:
-- Makes AudioNodes mostly not use nsWrapperCache. AudioDestinationNode
still does.
-- Rename MaxNumberOfInputs/Outputs to NumberOfInputs/Outputs, and have them
default to 1 in AudioNode.
-- Allow any number of nodes to be connected to any given input/output port.
