Move RTCPHelp::RTCPReceiveInformation inside RTCPReceiver

webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.cc

Index: webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.cc b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.cc
index a80866c3f8613e4a3d6c2a89eb120140e7b980bd..ecfeb4fc837ad1cb193110ec0c9013eff72a98cb 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.cc
@@ -32,37 +32,5 @@ RTCPReportBlockInformation::RTCPReportBlockInformation()
 
 RTCPReportBlockInformation::~RTCPReportBlockInformation() {}
 
-RTCPReceiveInformation::RTCPReceiveInformation() = default;
-RTCPReceiveInformation::~RTCPReceiveInformation() = default;
-
-void RTCPReceiveInformation::InsertTmmbrItem(uint32_t sender_ssrc,
-                                             const rtcp::TmmbItem& tmmbr_item,
-                                             int64_t current_time_ms) {
-  TimedTmmbrItem* entry = &tmmbr_[sender_ssrc];
-  entry->tmmbr_item = rtcp::TmmbItem(sender_ssrc, tmmbr_item.bitrate_bps(),
-                                     tmmbr_item.packet_overhead());
-  entry->last_updated_ms = current_time_ms;
-}
-
-void RTCPReceiveInformation::GetTmmbrSet(
-    int64_t current_time_ms,
-    std::vector<rtcp::TmmbItem>* candidates) {
-  // Use audio define since we don't know what interval the remote peer use.
-  int64_t timeouted_ms = current_time_ms - 5 * RTCP_INTERVAL_AUDIO_MS;
-  for (auto it = tmmbr_.begin(); it != tmmbr_.end();) {
-    if (it->second.last_updated_ms < timeouted_ms) {
-      // Erase timeout entries.
-      it = tmmbr_.erase(it);
-    } else {
-      candidates->push_back(it->second.tmmbr_item);
-      ++it;
-    }
-  }
-}
-
-void RTCPReceiveInformation::ClearTmmbr() {
-  tmmbr_.clear();
-}
-
 }  // namespace RTCPHelp
 }  // namespace webrtc