Reland of actor NACK bitrate allocation

webrtc/modules/rtp_rtcp/include/rtp_rtcp.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.
  */
 
 #ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
 #define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
 
 #include <set>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/include/module.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/video_coding/include/video_coding_defines.h"
 
 namespace webrtc {
 // Forward declarations.
+class RateLimiter;
 class ReceiveStatistics;
 class RemoteBitrateEstimator;
+class RtcEventLog;
 class RtpReceiver;
 class Transport;
-class RtcEventLog;
 
 RTPExtensionType StringToRtpExtensionType(const std::string& extension);
 
 namespace rtcp {
 class TransportFeedback;
 }
 
 class RtpRtcp : public Module {
  public:
   struct Configuration {
@@ skipping 31 matching lines @@
     RtcpRttStats* rtt_stats;
     RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer;
     RemoteBitrateEstimator* remote_bitrate_estimator;
     RtpPacketSender* paced_sender;
     TransportSequenceNumberAllocator* transport_sequence_number_allocator;
     BitrateStatisticsObserver* send_bitrate_observer;
     FrameCountObserver* send_frame_count_observer;
     SendSideDelayObserver* send_side_delay_observer;
     RtcEventLog* event_log;
     SendPacketObserver* send_packet_observer;
+    RateLimiter* retransmission_rate_limiter;
     RTC_DISALLOW_COPY_AND_ASSIGN(Configuration);
   };
 
   /*
    *   Create a RTP/RTCP module object using the system clock.
    *
    *   configuration  - Configuration of the RTP/RTCP module.
    */
   static RtpRtcp* CreateRtpRtcp(const RtpRtcp::Configuration& configuration);
 
@@ skipping 517 matching lines @@
      */
      virtual int32_t SetAudioLevel(uint8_t level_dBov) = 0;
 
     /**************************************************************************
     *
     *   Video
     *
     ***************************************************************************/
 
     /*
-    *   Set the target send bitrate
-    */
-    virtual void SetTargetSendBitrate(uint32_t bitrate_bps) = 0;
-
-    /*
     *   Turn on/off generic FEC
     */
     virtual void SetGenericFECStatus(bool enable,
                                      uint8_t payload_type_red,
                                      uint8_t payload_type_fec) = 0;
 
     /*
     *   Get generic FEC setting
     */
     virtual void GenericFECStatus(bool* enable,
@@ skipping 13 matching lines @@
 
     /*
     *   send a request for a keyframe
     *
     *   return -1 on failure else 0
     */
     virtual int32_t RequestKeyFrame() = 0;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_