Previously, we weren't too consistent about the definition of frame and
sample when it relates to raw audio data. This brings all the usages in
the context of raw data in line (hopefully), with samples referring to
a single PCM value, and frames referring to the multiple samples that
make up an instant's audio across all channels.
This will allow reuse of the allocations, instead of reallocating a
FixedArray for every block that changes size. Generally, it will be
possible to reuse AudioBlock memory throughout most of the pipeline
at least while in a single process.
For web audio, I reckon an occasional misjudged channel layout is
better than more frequent exceptions.
Signed PCM is normalized with unsigned max divided by 2, not
signed max. If you divide by the signed max (32767), you get headroom
that can exceed the threshold below -1.0. It's not audible, this mostly
matters for tests that assume correct normalization. But it turns out
there's no shortage of "golden ears" jackholes out there who swear they
can hear the difference.
Audio blocks now contain a sample specification with the sample rate
and channel map for the audio data they contain. This will facilitate
conversion from one sample specification to another in order to allow
playback on devices with more or less speakers than the audio data
contains.
This commit implements the functionality to play back audio through
PlaybackManager.
To decode the audio data, AudioDataProviders are created for each track
in the provided media data. These providers will fill their audio block
queue, then sit idle until their corresponding tracks are enabled.
In order to output the audio, one AudioMixingSink is created which
manages a PlaybackStream which requests audio blocks from multiple
AudioDataProviders and mixes them into one buffer with sample-perfect
precision.
