Reland: Deliver reordered packets in arrival time order after parsing feedback message.

webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc

 /*
  *  Copyright (c) 2015 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/remote_bitrate_estimator/transport_feedback_adapter.h"
 
+#include <algorithm>
 #include <limits>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 #include "webrtc/modules/utility/include/process_thread.h"
 
 namespace webrtc {
 
 const int64_t kNoTimestamp = -1;
 const int64_t kSendTimeHistoryWindowMs = 10000;
 const int64_t kBaseTimestampScaleFactor =
     rtcp::TransportFeedback::kDeltaScaleFactor * (1 << 8);
 const int64_t kBaseTimestampRangeSizeUs = kBaseTimestampScaleFactor * (1 << 24);
 
+class PacketInfoComparator {
+ public:
+  inline bool operator()(const PacketInfo& lhs, const PacketInfo& rhs) {
+    if (lhs.arrival_time_ms != rhs.arrival_time_ms)
+      return lhs.arrival_time_ms < rhs.arrival_time_ms;
+    if (lhs.send_time_ms != rhs.send_time_ms)
+      return lhs.send_time_ms < rhs.send_time_ms;
+    return lhs.sequence_number < rhs.sequence_number;
+  }
+};
+
 TransportFeedbackAdapter::TransportFeedbackAdapter(
     BitrateController* bitrate_controller,
     Clock* clock)
     : send_time_history_(clock, kSendTimeHistoryWindowMs),
       bitrate_controller_(bitrate_controller),
       clock_(clock),
       current_offset_ms_(kNoTimestamp),
       last_timestamp_us_(kNoTimestamp) {}
 
 TransportFeedbackAdapter::~TransportFeedbackAdapter() {
@@ skipping 58 matching lines @@
         int64_t timestamp_ms = current_offset_ms_ + (offset_us / 1000);
         PacketInfo info(timestamp_ms, sequence_number);
         if (send_time_history_.GetInfo(&info, true) && info.send_time_ms >= 0) {
           packet_feedback_vector.push_back(info);
         } else {
           ++failed_lookups;
         }
       }
       ++sequence_number;
     }
+    std::sort(packet_feedback_vector.begin(), packet_feedback_vector.end(),
+              PacketInfoComparator());
     RTC_DCHECK(delta_it == delta_vec.end());
     if (failed_lookups > 0) {
       LOG(LS_WARNING) << "Failed to lookup send time for " << failed_lookups
                       << " packet" << (failed_lookups > 1 ? "s" : "")
                       << ". Send time history too small?";
     }
   }
 
   RTC_DCHECK(bitrate_estimator_.get() != nullptr);
   bitrate_estimator_->IncomingPacketFeedbackVector(packet_feedback_vector);
 }
 
 void TransportFeedbackAdapter::OnReceiveBitrateChanged(
     const std::vector<uint32_t>& ssrcs,
     uint32_t bitrate) {
   bitrate_controller_->UpdateDelayBasedEstimate(bitrate);
 }
 
 void TransportFeedbackAdapter::OnRttUpdate(int64_t avg_rtt_ms,
                                            int64_t max_rtt_ms) {
   RTC_DCHECK(bitrate_estimator_.get() != nullptr);
   bitrate_estimator_->OnRttUpdate(avg_rtt_ms, max_rtt_ms);
 }
 
 }  // namespace webrtc