Refactor NACK bitrate allocation

webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.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.
  */
 
@@ skipping 379 matching lines @@
 };
 
 class TransportSequenceNumberAllocator {
  public:
   TransportSequenceNumberAllocator() {}
   virtual ~TransportSequenceNumberAllocator() {}
 
   virtual uint16_t AllocateSequenceNumber() = 0;
 };
 
+class NackRateLimiter {

[danilchap, 2016/06/28 14:26:18]

Probably Retransmission instead of Nack.
Since it limits retransmitting packets, not rtcp nack packets.

[sprang_webrtc, 2016/07/04 09:33:03]

Done.

+ public:
+  NackRateLimiter() {}
+  virtual ~NackRateLimiter() {}
+
+  // Try to use rate to send bytes. Returns true on success and if so updates
+  // current rate.

[danilchap, 2016/06/28 14:26:18]

bytes -> |bytes|.
May be rename it to |packet_size_bytes|

[sprang_webrtc, 2016/07/04 09:33:03]

Done.

+  virtual bool TryUseRate(size_t bytes) = 0;
+};
+
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_