BWE allocation strategy

webrtc/call/call.h

 /*
  *  Copyright (c) 2013 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.
  */
 #ifndef WEBRTC_CALL_CALL_H_
 #define WEBRTC_CALL_CALL_H_
 
 #include <algorithm>
 #include <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/api/rtcerror.h"
 #include "webrtc/call/audio_receive_stream.h"
 #include "webrtc/call/audio_send_stream.h"
 #include "webrtc/call/audio_state.h"
 #include "webrtc/call/flexfec_receive_stream.h"
 #include "webrtc/call/rtp_transport_controller_send_interface.h"
 #include "webrtc/common_types.h"
+#include "webrtc/rtc_base/bitrateallocationstrategy.h"
 #include "webrtc/rtc_base/networkroute.h"
 #include "webrtc/rtc_base/platform_file.h"
 #include "webrtc/rtc_base/socket.h"
 #include "webrtc/video_receive_stream.h"
 #include "webrtc/video_send_stream.h"
 
 namespace webrtc {
 
 class AudioProcessing;
 class RtcEventLog;
@@ skipping 141 matching lines @@
   virtual void SetBitrateConfig(
       const Config::BitrateConfig& bitrate_config) = 0;
 
   // The greater min and smaller max set by this and SetBitrateConfig will be
   // used. The latest non-negative start value form either call will be used.
   // Specifying a start bitrate will reset the current bitrate estimate.
   // Assumes 0 <= min <= start <= max holds for set parameters.
   virtual void SetBitrateConfigMask(
       const Config::BitrateConfigMask& bitrate_mask) = 0;
 
+  virtual void SetBitrateAllocationStrategy(
+      rtc::scoped_refptr<rtc::BitrateAllocationStrategy>
+          bitrate_allocation_strategy) = 0;
+
   // TODO(skvlad): When the unbundled case with multiple streams for the same
   // media type going over different networks is supported, track the state
   // for each stream separately. Right now it's global per media type.
   virtual void SignalChannelNetworkState(MediaType media,
                                          NetworkState state) = 0;
 
   virtual void OnTransportOverheadChanged(
       MediaType media,
       int transport_overhead_per_packet) = 0;
 
   virtual void OnNetworkRouteChanged(
       const std::string& transport_name,
       const rtc::NetworkRoute& network_route) = 0;
 
   virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
 
   virtual ~Call() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_CALL_H_