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This can reduce audio latency according to
https://developer.android.com/ndk/guides/audio/opensl/opensl-prog-notes#perform.
Previously we were using the hardcoded values of 48000 Hz and 256 frames
per buffer. The sample rate we use with this change is 48000 Hz on all
devices I'm aware of, but the buffer size does vary across devices.
Terminology note: The old code used the term "sample" to refer to what
Android refers to as a "frame". "Frame" is a clearer term to use for
this, so I've changed OpenSLESStream's terminology. One frame consists
of one sample per channel.
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Not sure if we're ever going to want to have more than one of these at
the same time, but these global variables are a code smell nonetheless.
I'm also deleting the existing member variables because they were
unused.
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SPDX standardizes how source code conveys its copyright and licensing
information. See https://spdx.github.io/spdx-spec/1-rationale/ . SPDX
tags are adopted in many large projects, including things like the Linux
kernel.
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Macros should be all upper-cased. This is also kind of a wart that's
been sticking out for quite a while now (we avoid prefixing
underscores).
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A failure on my part. I was updating the two buffers in the wrong order, so we were always a buffer behind in sending audio out to OpenSLES.
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Instead of creating the mixer externally and then passing it in, it can just be made within the class.
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I was statically setting the sample rate to 44.1Khz when we default to run at 48Khz.
This was causing audio to sound much too low.
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After calling enqueue the callback would fire before we had assigned g_mixer a value.
This would cause a fun crash to happen.
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and s/\bNULL\b/nullptr/g for *.cpp/h/mm files not compiled on my machine
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