Re: [w3ctag/design-reviews] WebRTC playoutDelayHint (#441)

Using double seconds is consistent with WebRTC's getStats() API which returns [jitterBufferDelay](https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-jitterbufferdelay). Setting this API affects the jitterBufferDelay, so it could be confusing if you set one to 1000 and expect the other to be 1.0. This is also consistent with the rest of getStats(), where time is always in seconds.

I don't know the reason why getStats() uses seconds though, but it's been the standard unit of measurement in WebRTC's stats. A wild guess would be that its an [SI unit](https://en.wikipedia.org/wiki/International_System_of_Units).

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