Add PeerConnectionInterface::UpdateCallBitrate.

webrtc/api/peerconnectioninterface.h

 /*
  *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 711 matching lines @@
       const std::vector<cricket::Candidate>& candidates) {
     return false;
   }
 
   // Register a metric observer (used by chromium).
   //
   // There can only be one observer at a time. Before the observer is
   // destroyed, RegisterUMAOberver(nullptr) should be called.
   virtual void RegisterUMAObserver(UMAObserver* observer) = 0;
 
+  // 0 <= min <= current <= max should hold for set parameters.

[kwiberg-webrtc, 2017/05/24 08:27:55]

Are all 8 combinations of set/unset supported?

[Zach Stein, 2017/05/24 21:10:02]

Yes.

+  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

[Taylor Brandstetter, 2017/05/25 15:33:52]

This comment may need to be updated. It's not just changing to a new value that
will reset the current estimate, passing in *any* value will do this.

[Zach Stein, 2017/05/25 20:26:31]

Done.

+  // 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;
   virtual IceGatheringState ice_gathering_state() = 0;
 
   // Starts RtcEventLog using existing file. Takes ownership of |file| and
   // passes it on to Call, which will take the ownership. If the
   // operation fails the file will be closed. The logging will stop
   // automatically after 10 minutes have passed, or when the StopRtcEventLog
   // function is called.
@@ skipping 363 matching lines @@
     cricket::WebRtcVideoEncoderFactory* encoder_factory,
     cricket::WebRtcVideoDecoderFactory* decoder_factory) {
   return CreatePeerConnectionFactory(
       worker_and_network_thread, worker_and_network_thread, signaling_thread,
       default_adm, encoder_factory, decoder_factory);
 }
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_PEERCONNECTIONINTERFACE_H_