Refactor NACK bitrate allocation

webrtc/voice_engine/channel.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 131 matching lines @@
                     int64_t capture_time_ms,
                     size_t bytes,
                     bool retransmission) override {
     rtc::CritScope lock(&crit_);
     if (rtp_packet_sender_) {
       rtp_packet_sender_->InsertPacket(priority, ssrc, sequence_number,
                                        capture_time_ms, bytes, retransmission);
     }
   }
 
+  bool AllocateRetransmissionBitrate(size_t bytes) override {
+    rtc::CritScope lock(&crit_);
+    if (rtp_packet_sender_)
+      return rtp_packet_sender_->AllocateRetransmissionBitrate(bytes);
+    return false;
+  }
+
+  int CurrentRetransmissionBitrate() override {
+    rtc::CritScope lock(&crit_);
+    if (rtp_packet_sender_)
+      return rtp_packet_sender_->CurrentRetransmissionBitrate();
+    return 0;
+  }
+
  private:
   rtc::ThreadChecker thread_checker_;
   rtc::CriticalSection crit_;
   RtpPacketSender* rtp_packet_sender_ GUARDED_BY(&crit_);
 };
 
 // Extend the default RTCP statistics struct with max_jitter, defined as the
 // maximum jitter value seen in an RTCP report block.
 struct ChannelStatistics : public RtcpStatistics {
   ChannelStatistics() : rtcp(), max_jitter(0) {}
@@ skipping 3320 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc