Refactor NACK bitrate allocation

webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h

Index: webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
index 05320f72e1b5a5272bf7c35a2330ce7abe1d6f65..cbed14351089576eb289fce44c95867979bcfbf4 100644
--- a/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
+++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
@@ -397,5 +397,15 @@ class 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_