[This CL is a rebase of an original CL by solenberg@:
https://codereview.webrtc.org/2948763002/ which in turn was a
rebase of an original CL by peah@:
https://chromium-review.googlesource.com/c/527032/]
Allow an external audio processing module to be used in WebRTC
This CL adds support for optionally using an externally created audio
processing module in a peerconnection. The ownership is shared
between the peerconnection and the external creator of the module.
As part of this the internal ownership of the audio processing module
is moved from VoiceEngine to WebRtcVoiceEngine.
BUG=webrtc:7775
Review-Url: https://codereview.webrtc.org/2961723004
Cr-Commit-Position: refs/heads/master@{#18837}
Changes:
1. Added MockTransport.
2. Added VoiceEngineFixture class to be used in VoE unit tests. This class is in no way final as it is - it will grow as we add more unit tests and our understanding of the Voice Engine grows.
3. Modified VoEBase unit test to use VoiceEngineFixture.
4. Added VoENetwork unit test.
R=henrika@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49759004
Cr-Commit-Position: refs/heads/master@{#9118}
