This is a mega-patch that was too hard to disentangle. Here's what it does:
-- Create infrastructure around AudioNode::UpdateOutputEnded to detect
when a node can no longer produce any output. When that becomes true,
disconnect it from the AudioNode graph.
-- Have AudioNode implement JSBindingFinalized to use as input in
UpdateOutputEnded.
-- Give every AudioNode a MediaStream, and give every connection
a MediaInputPort.
-- Actually play the audio that reaches the AudioContext's destination node.
-- Force AudioContext to use the audio sample rate defined by MediaStreamGraph.
-- Fix AudioBufferSourceNode's start and stop methods to possibly throw and
take default 'when' parameters.
-- Create an AudioNodeStream for AudioBufferSourceNode and give it a
AudioBufferSourceNodeEngine that does what's needed. Set parameters for
this engine in the start() and stop() methods.
-- Create AudioBuffer::GetThreadSharedChannelsForRate, which is responsible
for stealing the contents of any JS array buffers, and bundling them up
into a thread-shared read-only buffer object which can be used as
part of an AudioChunk. This method will also be responsible for
resampling and caching as necessary.
--HG--
rename : content/media/MediaStreamGraph.cpp => content/media/MediaStreamGraphImpl.h
extra : rebase_source : 9fa0ec0efa304acd6513e427103d6339c78efa53
Modifies MediaStreamGraph to always advance its time by a multiple of
WEBAUDIO_BLOCK_SIZE.
--HG--
extra : rebase_source : 99524b09edd4ac0e1bc6607f2ba14925bc2f11c2
There is no need for these to be independent objects in general and we
don't need to addref/release them. We can just require the caller to
remove them before they die.
We can also save some refcount churn by having
DispatchToMainThreadAfterStreamStateUpdate take already_AddRefed.
--HG--
extra : rebase_source : 751114a1befd73b405dff3ee986496efb6f76201
We need this in order to update the MediaStreamGraph thread when an
AudioParam changes. This enables each AudioParam to be registered with
a callback from its owner node, so that the owner node can have custom
processing code for each AudioParam's mutation.
This is a mega-patch that was too hard to disentangle. Here's what it does:
-- Create infrastructure around AudioNode::UpdateOutputEnded to detect
when a node can no longer produce any output. When that becomes true,
disconnect it from the AudioNode graph.
-- Have AudioNode implement JSBindingFinalized to use as input in
UpdateOutputEnded.
-- Give every AudioNode a MediaStream, and give every connection
a MediaInputPort.
-- Actually play the audio that reaches the AudioContext's destination node.
-- Force AudioContext to use the audio sample rate defined by MediaStreamGraph.
-- Fix AudioBufferSourceNode's start and stop methods to possibly throw and
take default 'when' parameters.
-- Create an AudioNodeStream for AudioBufferSourceNode and give it a
AudioBufferSourceNodeEngine that does what's needed. Set parameters for
this engine in the start() and stop() methods.
-- Create AudioBuffer::GetThreadSharedChannelsForRate, which is responsible
for stealing the contents of any JS array buffers, and bundling them up
into a thread-shared read-only buffer object which can be used as
part of an AudioChunk. This method will also be responsible for
resampling and caching as necessary.
There is no need for these to be independent objects in general and we
don't need to addref/release them. We can just require the caller to
remove them before they die.
We can also save some refcount churn by having
DispatchToMainThreadAfterStreamStateUpdate take already_AddRefed.
