Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.

This CL adds the following interfaces:
* RtpTransportController
* RtpTransport
* RtpSender
* RtpReceiver

They're implemented on top of the "BaseChannel" object, which is normally used
in a PeerConnection, and roughly corresponds to an SDP "m=" section. As a result
of this, there are several limitations:

* You can only have one of each type of sender and receiver (audio/video) on top
  of the same transport controller.
* The sender/receiver with the same media type must use the same RTP transport.
* You can't change the transport after creating the sender or receiver.
* Some of the parameters aren't supported.

Later, these "adapter" objects will be gradually replaced by real objects that don't
have these limitations, as "BaseChannel", "MediaChannel" and related code is
restructured. In this CL, we essentially have:

ORTC adapter objects -> BaseChannel -> Media engine
PeerConnection -> BaseChannel -> Media engine

And later we hope to have simply:

PeerConnection -> "Real" ORTC objects -> Media engine

See the linked bug for more context.

BUG=webrtc:7013
TBR=stefan@webrtc.org

Review-Url: https://codereview.webrtc.org/2675173003
Cr-Commit-Position: refs/heads/master@{#16842}

webrtc/api/peerconnectioninterface.h

@@ -68,7 +68,6 @@
 #define WEBRTC_API_PEERCONNECTIONINTERFACE_H_
 
 #include <memory>
-#include <ostream>
 #include <string>
 #include <utility>
 #include <vector>
@@ -901,7 +900,7 @@ class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
   virtual rtc::scoped_refptr<MediaStreamInterface>
       CreateLocalMediaStream(const std::string& label) = 0;
 
-  // Creates a AudioSourceInterface.
+  // Creates an AudioSourceInterface.
   // |options| decides audio processing settings.
   virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
       const cricket::AudioOptions& options) = 0;