Tweaked version of BBR for WebRTC.

webrtc/modules/bitrate_controller/include/bitrate_controller.h

 /*
  *  Copyright (c) 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.
  *
  *  Usage: this class will register multiple RtcpBitrateObserver's one at each
  *  RTCP module. It will aggregate the results and run one bandwidth estimation
  *  and push the result to the encoders via BitrateObserver(s).
  */
 
 #ifndef WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_
 #define WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_
 
+#include <map>
+
 #include "webrtc/modules/congestion_controller/delay_based_bwe.h"
 #include "webrtc/modules/include/module.h"
 #include "webrtc/modules/pacing/paced_sender.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
 class RtcEventLog;
 
 // Deprecated
 // TODO(perkj): Remove BitrateObserver when no implementations use it.
 class BitrateObserver {
   // Observer class for bitrate changes announced due to change in bandwidth
   // estimate or due to bitrate allocation changes. Fraction loss and rtt is
   // also part of this callback to allow the obsevrer to optimize its settings
   // for different types of network environments. The bitrate does not include
   // packet headers and is measured in bits per second.
  public:
   virtual void OnNetworkChanged(uint32_t bitrate_bps,
                                 uint8_t fraction_loss,  // 0 - 255.
                                 int64_t rtt_ms) = 0;
-
+  // TODO(gnish): Merge these two into one function.
+  virtual void OnNetworkChanged(uint32_t bitrate_for_encoder_bps,
+                                uint32_t bitrate_for_pacer_bps,
+                                bool in_probe_rtt,
+                                int64_t target_set_time,
+                                uint64_t congestion_window) {}
+  virtual void OnBytesAcked(size_t bytes) {}
+  virtual size_t pacer_queue_size_in_bytes() { return 0; }
   virtual ~BitrateObserver() {}
 };
 
-class BitrateController : public Module,
-                          public RtcpBandwidthObserver {
+class BitrateController : public Module, public RtcpBandwidthObserver {
   // This class collects feedback from all streams sent to a peer (via
   // RTCPBandwidthObservers). It does one  aggregated send side bandwidth
   // estimation and divide the available bitrate between all its registered
   // BitrateObservers.
  public:
   static const int kDefaultStartBitratebps = 300000;
 
   // Deprecated:
   // TODO(perkj): BitrateObserver has been deprecated and is not used in WebRTC.
   // Remove this method once other other projects does not use it.
@@ skipping 28 matching lines @@
   virtual bool AvailableBandwidth(uint32_t* bandwidth) const = 0;
 
   virtual void SetReservedBitrate(uint32_t reserved_bitrate_bps) = 0;
 
   virtual bool GetNetworkParameters(uint32_t* bitrate,
                                     uint8_t* fraction_loss,
                                     int64_t* rtt) = 0;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_