Refactor composing report blocks for rtcp Sender/Receiver reports

webrtc/modules/rtp_rtcp/source/rtcp_sender.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 437 matching lines @@
   uint32_t rtp_timestamp =
       timestamp_offset_ + last_rtp_timestamp_ +
       (clock_->TimeInMilliseconds() - last_frame_capture_time_ms_) * rtp_rate;
 
   rtcp::SenderReport* report = new rtcp::SenderReport();
   report->SetSenderSsrc(ssrc_);
   report->SetNtp(ctx.now_);
   report->SetRtpTimestamp(rtp_timestamp);
   report->SetPacketCount(ctx.feedback_state_.packets_sent);
   report->SetOctetCount(ctx.feedback_state_.media_bytes_sent);
-
-  for (auto it : report_blocks_)
-    report->AddReportBlock(it.second);
-
-  report_blocks_.clear();
+  report->SetReportBlocks(CreateReportBlocks(ctx.feedback_state_));
 
   return std::unique_ptr<rtcp::RtcpPacket>(report);
 }
 
 std::unique_ptr<rtcp::RtcpPacket> RTCPSender::BuildSDES(
     const RtcpContext& ctx) {
   size_t length_cname = cname_.length();
   RTC_CHECK_LT(length_cname, RTCP_CNAME_SIZE);
 
   rtcp::Sdes* sdes = new rtcp::Sdes();
   sdes->AddCName(ssrc_, cname_);
 
   for (const auto& it : csrc_cnames_)
     RTC_CHECK(sdes->AddCName(it.first, it.second));
 
   return std::unique_ptr<rtcp::RtcpPacket>(sdes);
 }
 
 std::unique_ptr<rtcp::RtcpPacket> RTCPSender::BuildRR(const RtcpContext& ctx) {
   rtcp::ReceiverReport* report = new rtcp::ReceiverReport();
   report->SetSenderSsrc(ssrc_);
-  for (auto it : report_blocks_)
-    report->AddReportBlock(it.second);
+  report->SetReportBlocks(CreateReportBlocks(ctx.feedback_state_));
 
-  report_blocks_.clear();
   return std::unique_ptr<rtcp::RtcpPacket>(report);
 }
 
 std::unique_ptr<rtcp::RtcpPacket> RTCPSender::BuildPLI(const RtcpContext& ctx) {
   rtcp::Pli* pli = new rtcp::Pli();
   pli->SetSenderSsrc(ssrc_);
   pli->SetMediaSsrc(remote_ssrc_);
 
   TRACE_EVENT_INSTANT0(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                        "RTCPSender::PLI");
@@ skipping 325 matching lines @@
       }
       if (minIntervalMs > RTCP_INTERVAL_VIDEO_MS)
         minIntervalMs = RTCP_INTERVAL_VIDEO_MS;
     }
     // The interval between RTCP packets is varied randomly over the
     // range [1/2,3/2] times the calculated interval.
     uint32_t timeToNext =
         random_.Rand(minIntervalMs * 1 / 2, minIntervalMs * 3 / 2);
     next_time_to_send_rtcp_ = clock_->TimeInMilliseconds() + timeToNext;
 
-    if (receive_statistics_) {
-      StatisticianMap statisticians =
-          receive_statistics_->GetActiveStatisticians();
-      RTC_DCHECK(report_blocks_.empty());
-      for (auto& it : statisticians) {
-        AddReportBlock(feedback_state, it.first, it.second);
-      }
-    }
+    // RtcpSender expected to be used for sending either just sender reports
+    // or just receiver reports.
+    RTC_DCHECK(!(IsFlagPresent(kRtcpSr) && IsFlagPresent(kRtcpRr)));
   }
 }
 
-bool RTCPSender::AddReportBlock(const FeedbackState& feedback_state,
-                                uint32_t ssrc,
-                                StreamStatistician* statistician) {
-  // Do we have receive statistics to send?
-  RtcpStatistics stats;
-  if (!statistician->GetStatistics(&stats, true))
-    return false;
+std::vector<rtcp::ReportBlock> RTCPSender::CreateReportBlocks(
+    const FeedbackState& feedback_state) {
+  std::vector<rtcp::ReportBlock> result;
+  if (!receive_statistics_)
+    return result;
 
-  if (report_blocks_.size() >= RTCP_MAX_REPORT_BLOCKS) {
-    LOG(LS_WARNING) << "Too many report blocks.";
-    return false;
+  StatisticianMap statisticians = receive_statistics_->GetActiveStatisticians();
+  result.reserve(statisticians.size());
+  for (auto& statistician : statisticians) {
+    // Do we have receive statistics to send?
+    RtcpStatistics stats;
+    if (!statistician.second->GetStatistics(&stats, true))
+      continue;
+    // TODO(danilchap): Support sending more than |RTCP_MAX_REPORT_BLOCKS| per
+    // compound rtcp packet when single rtcp module is used for multiple media
+    // streams.
+    if (result.size() >= RTCP_MAX_REPORT_BLOCKS) {
+      LOG(LS_WARNING) << "Too many report blocks.";
+      continue;
+    }
+    result.emplace_back();
+    rtcp::ReportBlock& block = result.back();

[eladalon, 2017/07/25 12:19:20]

It would probably be overkill to make sure the emplace_back didn't fail due to
OOM, right?

[danilchap, 2017/07/25 15:20:08]

yep, specially that emplace_back has no way to report an error (other than with
an exception that shouldn't be catched anyway)

+    block.SetMediaSsrc(statistician.first);
+    block.SetFractionLost(stats.fraction_lost);
+    if (!block.SetCumulativeLost(stats.cumulative_lost)) {
+      LOG(LS_WARNING) << "Cumulative lost is oversized.";
+      result.pop_back();
+      continue;
+    }
+    block.SetExtHighestSeqNum(stats.extended_max_sequence_number);
+    block.SetJitter(stats.jitter);
   }
-  RTC_DCHECK(report_blocks_.find(ssrc) == report_blocks_.end());
-  rtcp::ReportBlock* block = &report_blocks_[ssrc];
-  block->SetMediaSsrc(ssrc);
-  block->SetFractionLost(stats.fraction_lost);
-  if (!block->SetCumulativeLost(stats.cumulative_lost)) {
-    report_blocks_.erase(ssrc);
-    LOG(LS_WARNING) << "Cumulative lost is oversized.";
-    return false;
+
+  if (!result.empty() && ((feedback_state.last_rr_ntp_secs != 0) ||
+                          (feedback_state.last_rr_ntp_frac != 0))) {
+    // Get our NTP as late as possible to avoid a race.
+    uint32_t now = CompactNtp(clock_->CurrentNtpTime());
+
+    uint32_t receive_time = feedback_state.last_rr_ntp_secs & 0x0000FFFF;
+    receive_time <<= 16;
+    receive_time += (feedback_state.last_rr_ntp_frac & 0xffff0000) >> 16;
+
+    uint32_t delay_since_last_sr = now - receive_time;
+    // TODO(danilchap): Instead of setting same value on all report blocks,
+    // set only when media_ssrc match sender ssrc of the sender report
+    // remote times were taken from.
+    for (auto& report_block : result) {
+      report_block.SetLastSr(feedback_state.remote_sr);
+      report_block.SetDelayLastSr(delay_since_last_sr);
+    }
   }
-  block->SetExtHighestSeqNum(stats.extended_max_sequence_number);
-  block->SetJitter(stats.jitter);
-  block->SetLastSr(feedback_state.remote_sr);
-
-  // TODO(sprang): Do we really need separate time stamps for each report?
-  // Get our NTP as late as possible to avoid a race.
-  NtpTime ntp = clock_->CurrentNtpTime();
-
-  // Delay since last received report.
-  if ((feedback_state.last_rr_ntp_secs != 0) ||
-      (feedback_state.last_rr_ntp_frac != 0)) {
-    // Get the 16 lowest bits of seconds and the 16 highest bits of fractions.
-    uint32_t now = CompactNtp(ntp);
-
-    uint32_t receiveTime = feedback_state.last_rr_ntp_secs & 0x0000FFFF;
-    receiveTime <<= 16;
-    receiveTime += (feedback_state.last_rr_ntp_frac & 0xffff0000) >> 16;
-
-    block->SetDelayLastSr(now - receiveTime);
-  }
-  return true;
+  return result;
 }
 
 void RTCPSender::SetCsrcs(const std::vector<uint32_t>& csrcs) {
   RTC_DCHECK_LE(csrcs.size(), kRtpCsrcSize);
   rtc::CritScope lock(&critical_section_rtcp_sender_);
   csrcs_ = csrcs;
 }
 
 int32_t RTCPSender::SetApplicationSpecificData(uint8_t subType,
                                                uint32_t name,
@@ skipping 112 matching lines @@
     max_packet_size = max_packet_size_;
   }
 
   RTC_DCHECK_LE(max_packet_size, IP_PACKET_SIZE);
   uint8_t buffer[IP_PACKET_SIZE];
   return packet.BuildExternalBuffer(buffer, max_packet_size, &sender) &&
          !sender.send_failure_;
 }
 
 }  // namespace webrtc