Add PeerConnectionInterface::UpdateCallBitrate with call tests.

webrtc/api/peerconnectioninterface.h

Index: webrtc/api/peerconnectioninterface.h
diff --git a/webrtc/api/peerconnectioninterface.h b/webrtc/api/peerconnectioninterface.h
index a3a07b6313cc55eab982b538304f51d8568b62eb..a70f9d09f2bb90d9582b1cf155ed317c341de2b1 100644
--- a/webrtc/api/peerconnectioninterface.h
+++ b/webrtc/api/peerconnectioninterface.h
@@ -731,6 +731,21 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
   // destroyed, RegisterUMAOberver(nullptr) should be called.
   virtual void RegisterUMAObserver(UMAObserver* observer) = 0;
 
+  // 0 <= min <= current <= max should hold for set parameters.
+  struct BitrateParameters {
+    rtc::Optional<int> min_bitrate_bps;
+    rtc::Optional<int> current_bitrate_bps;
+    rtc::Optional<int> max_bitrate_bps;
+  };
+
+  // SetBitrate limits the bandwidth allocated for all RTP streams sent by
+  // this PeerConnection. Other limitations might affect these limits and
+  // are respected (for example "b=AS" in SDP).
+  //
+  // Changing |current_bitrate_bps| to a new value will reset the current
+  // bitrate estimate to the provided value.
+  virtual RTCError SetBitrate(const BitrateParameters& bitrate) = 0;
+
   // Returns the current SignalingState.
   virtual SignalingState signaling_state() = 0;
   virtual IceConnectionState ice_connection_state() = 0;