Enable setting the maximum bitrate limit in RtpSender.

This change allows the application to limit the bitrate of the outgoing
audio and video streams at runtime. The API roughly follows the WebRTC
API draft, defining the RTCRtpParameters structure witn exactly one
encoding (simulcast streams are not exposed in the API for now).
(https://www.w3.org/TR/webrtc/#idl-def-RTCRtpParameters)

BUG=

Review URL: https://codereview.webrtc.org/1788583004

Cr-Commit-Position: refs/heads/master@{#12025}

webrtc/api/webrtcsession.h

@@ -245,6 +245,10 @@ class WebRtcSession : public AudioProviderInterface,
   void SetRawAudioSink(uint32_t ssrc,
                        rtc::scoped_ptr<AudioSinkInterface> sink) override;
 
+  RtpParameters GetAudioRtpParameters(uint32_t ssrc) const override;
+  bool SetAudioRtpParameters(uint32_t ssrc,
+                             const RtpParameters& parameters) override;
+
   // Implements VideoMediaProviderInterface.
   bool SetCaptureDevice(uint32_t ssrc, cricket::VideoCapturer* camera) override;
   void SetVideoPlayout(
@@ -255,6 +259,10 @@ class WebRtcSession : public AudioProviderInterface,
                     bool enable,
                     const cricket::VideoOptions* options) override;
 
+  RtpParameters GetVideoRtpParameters(uint32_t ssrc) const override;
+  bool SetVideoRtpParameters(uint32_t ssrc,
+                             const RtpParameters& parameters) override;
+
   // Implements DtmfProviderInterface.
   virtual bool CanInsertDtmf(const std::string& track_id);
   virtual bool InsertDtmf(const std::string& track_id,