We leave the nsIDOMEventTarget* versions fallible for now, but this makes the
common case a lot simpler. Note that this means that pushing a null JSContext,
a bug, is no longer handled at runtime. But I think we should just assert
against it, since there are already callers that don't check the return value.
The goal here is to get rid of this crap entirely, and make nsCxPusher always
push. But that's a scary change, so we do it in chunks. This patch, in particular,
should have zero behavioral change. This means preserving some very wrong behavior.
For instance, currently SafeAutoJSContext never pushes a damn thing, because the
safe JSContext doesn't have an associated nsIScriptContext. We preserve this
behavior, and in fact convert various similarly-buggy consumers to
SafeAutoJSContext, so that we can hoist the behavioral change into a subsequent
patch.
All the code is just moving except the Optional<nsAString>::operator=
that takes a FakeDependentString. That had to be changed so it has no
dependency on the actual definition of FakeDependentString.
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
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.
