Let this be mentioned in our code base history that I wrote this patch
with tears in my eyes. This is the saddest thing I've ever had to do to
Gecko. :(
--HG--
extra : rebase_source : 798e5b1dc8af51a399189dd05bc59ac077670820
Backed out changeset e811d4258d45 (bug 849713)
Backed out changeset 26aa58e87d5d (bug 849713)
Backed out changeset 9a6552161eff (bug 849713)
Backed out changeset 3551877d9b92 (bug 849713)
Landing on a CLOSED TREE
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.
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.
This is the bare minimum that one needs in order to get those interfaces
implemented. The work to make the simplest of Web Audio test cases
actually pass will be done in bug 792649.
This is the bare minimum that one needs in order to get those interfaces
implemented. The work to make the simplest of Web Audio test cases
actually pass will be done in bug 792649.
