Refactor NACK bitrate allocation

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

 /*
  *  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.
  */
 
@@ skipping 45 matching lines @@
       transport_feedback_callback(nullptr),
       rtt_stats(nullptr),
       rtcp_packet_type_counter_observer(nullptr),
       remote_bitrate_estimator(nullptr),
       paced_sender(nullptr),
       transport_sequence_number_allocator(nullptr),
       send_bitrate_observer(nullptr),
       send_frame_count_observer(nullptr),
       send_side_delay_observer(nullptr),
       event_log(nullptr),
-      send_packet_observer(nullptr) {}
+      send_packet_observer(nullptr),
+      retransmission_rate_limiter(nullptr) {}
 
 RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) {
   if (configuration.clock) {
     return new ModuleRtpRtcpImpl(configuration);
   } else {
     // No clock implementation provided, use default clock.
     RtpRtcp::Configuration configuration_copy;
     memcpy(&configuration_copy, &configuration,
            sizeof(RtpRtcp::Configuration));
     configuration_copy.clock = Clock::GetRealTimeClock();
     return new ModuleRtpRtcpImpl(configuration_copy);
   }
 }
 
 ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
     : rtp_sender_(configuration.audio,
                   configuration.clock,
                   configuration.outgoing_transport,
                   configuration.paced_sender,
                   configuration.transport_sequence_number_allocator,
                   configuration.transport_feedback_callback,
                   configuration.send_bitrate_observer,
                   configuration.send_frame_count_observer,
                   configuration.send_side_delay_observer,
                   configuration.event_log,
-                  configuration.send_packet_observer),
+                  configuration.send_packet_observer,
+                  configuration.retransmission_rate_limiter),
       rtcp_sender_(configuration.audio,
                    configuration.clock,
                    configuration.receive_statistics,
                    configuration.rtcp_packet_type_counter_observer,
                    configuration.event_log,
                    configuration.outgoing_transport),
       rtcp_receiver_(configuration.clock,
                      configuration.receiver_only,
                      configuration.rtcp_packet_type_counter_observer,
                      configuration.bandwidth_callback,
@@ skipping 710 matching lines @@
 int32_t ModuleRtpRtcpImpl::SetSendREDPayloadType(
     const int8_t payload_type) {
   return rtp_sender_.SetRED(payload_type);
 }
 
 // Get payload type for Redundant Audio Data RFC 2198.
 int32_t ModuleRtpRtcpImpl::SendREDPayloadType(int8_t* payload_type) const {
   return rtp_sender_.RED(payload_type);
 }
 
-void ModuleRtpRtcpImpl::SetTargetSendBitrate(uint32_t bitrate_bps) {
-  rtp_sender_.SetTargetBitrate(bitrate_bps);
-}
-
 int32_t ModuleRtpRtcpImpl::SetKeyFrameRequestMethod(
     const KeyFrameRequestMethod method) {
   key_frame_req_method_ = method;
   return 0;
 }
 
 int32_t ModuleRtpRtcpImpl::RequestKeyFrame() {
   switch (key_frame_req_method_) {
     case kKeyFrameReqPliRtcp:
       return SendRTCP(kRtcpPli);
@@ skipping 159 matching lines @@
 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
     StreamDataCountersCallback* callback) {
   rtp_sender_.RegisterRtpStatisticsCallback(callback);
 }
 
 StreamDataCountersCallback*
     ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
   return rtp_sender_.GetRtpStatisticsCallback();
 }
 }  // namespace webrtc