skvlad e0d4637bea Allow applications to control audio send bitrate through RtpParameters.
This change builds on top of the refactoring in https://codereview.webrtc.org/1841083008/, and enables WebRTC client applications to control the max send bitrate for every audio stream through RtpParameters.

The AudioSendStream now stores the last codec spec, and whenever a global or per-stream bitrate limit changes, the effective limit (smaller of the two) is recomputed and the codec is reconfigured with that bitrate.

TBR=pthatcher
BUG=

Review URL: https://codereview.webrtc.org/1847353004

Cr-Commit-Position: refs/heads/master@{#12290}
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