Switch AudioRendererAlgorithm to buffer based on time.

Switch AudioRendererAlgorithm to buffer based on time.

It doesn't make sense to buffer based on a frame count when the
amount of time that data represent varies with sample rate; it's
the opposite of what we want.  I.e., a 48kHz playback will exhaust
this buffer faster than a 44.1kHz playback.

Instead, switch to time based initial capacity using the sample
rate provided during initialization.  This provides a greater
resilience against underflow when video decoding is overloading
the media thread.

At 48kHz we were previously buffering ~171ms and at 44.1kHz we were
buffering ~186ms of audio data.  I've chosen a nice round 200ms for
simplicity and since it works well on an Android One device.

BUG=519108
TEST=media_unittests still pass.

Review URL: https://codereview.chromium.org/1278733004

Cr-Commit-Position: refs/heads/master@{#343279}