Cleaned out candidateSet member from TMMBRHelp class

webrtc/modules/rtp_rtcp/source/rtcp_receiver.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.
  */
 
 #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
 
 #include <assert.h>
 #include <string.h>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
 #include "webrtc/modules/rtp_rtcp/source/time_util.h"
 #include "webrtc/modules/rtp_rtcp/source/tmmbr_help.h"
 #include "webrtc/system_wrappers/include/ntp_time.h"
 
 namespace webrtc {
 using RTCPHelp::RTCPPacketInformation;
 using RTCPHelp::RTCPReceiveInformation;
 using RTCPHelp::RTCPReportBlockInformation;
 using RTCPUtility::kBtVoipMetric;
@@ skipping 1195 matching lines @@
     RTCPHelp::RTCPPacketInformation* rtcp_packet_information) {
   rtcp::RtcpPacket* packet = rtcp_parser->ReleaseRtcpPacket();
   RTC_DCHECK(packet != nullptr);
   rtcp_packet_information->rtcpPacketTypeFlags |= kRtcpTransportFeedback;
   rtcp_packet_information->transport_feedback_.reset(
       static_cast<rtcp::TransportFeedback*>(packet));
 
   rtcp_parser->Iterate();
 }
 int32_t RTCPReceiver::UpdateTMMBR() {
-  TMMBRHelp tmmbr_help;
-  uint32_t bitrate = 0;
-  uint32_t accNumCandidates = 0;
-
   int32_t size = TMMBRReceived(0, 0, NULL);
+  TMMBRSet candidates;
   if (size > 0) {
-    TMMBRSet* candidateSet = tmmbr_help.VerifyAndAllocateCandidateSet(size);
+    candidates.reserve(size);
     // Get candidate set from receiver.
-    accNumCandidates = TMMBRReceived(size, accNumCandidates, candidateSet);
+    TMMBRReceived(size, 0, &candidates);
   }
   // Find bounding set
-  std::vector<rtcp::TmmbItem> bounding = tmmbr_help.FindTMMBRBoundingSet();
+  std::vector<rtcp::TmmbItem> bounding =
+      TMMBRHelp::FindBoundingSet(std::move(candidates));
   // Set bounding set
   // Inform remote clients about the new bandwidth
   // inform the remote client
   _rtpRtcp.SetTMMBN(&bounding);
 
   // might trigger a TMMBN
   if (bounding.empty()) {
     // owner of max bitrate request has timed out
     // empty bounding set has been sent
     return 0;
   }
-  // Get net bitrate from bounding set depending on sent packet rate
-  if (tmmbr_help.CalcMinBitRate(&bitrate)) {
-    // we have a new bandwidth estimate on this channel
-    if (_cbRtcpBandwidthObserver) {
-        _cbRtcpBandwidthObserver->OnReceivedEstimatedBitrate(bitrate * 1000);
-    }
+  // We have a new bandwidth estimate on this channel.
+  if (_cbRtcpBandwidthObserver) {
+    constexpr uint64_t kMinVideoBwManagementBitrateBps =
+        MIN_VIDEO_BW_MANAGEMENT_BITRATE * 1000;
+    uint64_t bitrate_bps = std::max(TMMBRHelp::CalcMinBitrateBps(bounding),
+                                    kMinVideoBwManagementBitrateBps);

[philipel, 2016/08/11 11:57:12]

Remove kMinVideoBwManagementBitrateBps and just use
MIN_VIDEO_BW_MANAGEMENT_BITRATE * 1000 here instead.

[danilchap, 2016/08/11 13:08:36]

On the 2nd thought I think it is better to remove forcing low-limit completely:
it is up to to BandwidthObserver to decide what to do with oversmall bandwidth
(current implementation actually has different, smaller value for minimum
bitrate)
Stefan, do you think it is better that way or should rtcp receiver still report
30000 bps when it calculate less with Tmmbr?

+    _cbRtcpBandwidthObserver->OnReceivedEstimatedBitrate(bitrate_bps);
   }
   return 0;
 }
 
 void RTCPReceiver::RegisterRtcpStatisticsCallback(
     RtcpStatisticsCallback* callback) {
   rtc::CritScope cs(&_criticalSectionFeedbacks);
   stats_callback_ = callback;
 }
 
@@ skipping 159 matching lines @@
         return -1;
       }
       num += receiveInfo->TmmbrSet.lengthOfSet();
       receiveInfoIt++;
     }
   }
   return num;
 }
 
 }  // namespace webrtc