Here we make the non-realtime graphs to go to sleep until they're shut down
from the main thread. This allows us to use the common forced shutdown code
path in MediaStreamGraphImpl::RunThread. We also need to delete the graph
object when the last message is dispatched to it.
In addition, we need to make sure that the AudioNodes also get released when
they're no longer needed. To do this, we need for force the SelfReference of
AudioBufferSourceNodes to be released when the context is shut down, and also
trigger the destruction of the graph there.
We use a custom AudioNodeEngine for the destination nodes belonging to
OfflineAudioContexts, and there we record the rendered buffer. Once the buffer
is full, we resample it if the sampling rate of the OfflineAudioContext is
different than the ideal sampling rate, and then we hand it off to the main
thread in order for the complete event to be dispatched.
We offload most of the logic for OfflineAudioContext to the destination node,
since that's where the sample recording needs to happen, so doing this will
make the code simpler.
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
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.
