Propagate packet pacing information to SenTimeHistory

webrtc/modules/congestion_controller/delay_based_bwe_unittest_helper.cc

 /*
  *  Copyright (c) 2016 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/congestion_controller/delay_based_bwe_unittest_helper.h"
@@ skipping 42 matching lines @@
   }
   RTC_CHECK(packets != NULL);
   size_t bits_per_frame = (bitrate_bps_ + fps_ / 2) / fps_;
   size_t n_packets =
       std::max<size_t>((bits_per_frame + 4 * kMtu) / (8 * kMtu), 1u);
   size_t payload_size = (bits_per_frame + 4 * n_packets) / (8 * n_packets);
   for (size_t i = 0; i < n_packets; ++i) {
     PacketInfo packet(-1, sequence_number_++);
     packet.send_time_ms = (time_now_us + kSendSideOffsetUs) / 1000;
     packet.payload_size = payload_size;
-    packet.probe_cluster_id = PacketInfo::kNotAProbe;
+    packet.probe_cluster_id = PacedPacketInfo::kNotAProbe;
     packets->push_back(packet);
   }
   next_rtp_time_ = time_now_us + (1000000 + fps_ / 2) / fps_;
   return next_rtp_time_;
 }
 
 // The send-side time when the next frame can be generated.
 int64_t RtpStream::next_rtp_time() const {
   return next_rtp_time_;
 }
@@ skipping 88 matching lines @@
   stream_generator_->AddStream(new test::RtpStream(30, 3e5));
 }
 
 const uint32_t DelayBasedBweTest::kDefaultSsrc = 0;
 
 void DelayBasedBweTest::IncomingFeedback(int64_t arrival_time_ms,
                                          int64_t send_time_ms,
                                          uint16_t sequence_number,
                                          size_t payload_size) {
   IncomingFeedback(arrival_time_ms, send_time_ms, sequence_number, payload_size,
-                   PacketInfo::kNotAProbe);
+                   PacedPacketInfo::kNotAProbe);
 }
 
 void DelayBasedBweTest::IncomingFeedback(int64_t arrival_time_ms,
                                          int64_t send_time_ms,
                                          uint16_t sequence_number,
                                          size_t payload_size,
                                          int probe_cluster_id) {
   RTC_CHECK_GE(arrival_time_ms + arrival_time_offset_ms_, 0);
   PacketInfo packet(arrival_time_ms + arrival_time_offset_ms_, send_time_ms,
                     sequence_number, payload_size, probe_cluster_id);
@@ skipping 90 matching lines @@
   EXPECT_EQ(0u, ssrcs.size());
   clock_.AdvanceTimeMilliseconds(1000);
   EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps));
   EXPECT_FALSE(bitrate_observer_.updated());
   bitrate_observer_.Reset();
   clock_.AdvanceTimeMilliseconds(1000);
   // Inserting packets for 5 seconds to get a valid estimate.
   for (int i = 0; i < 5 * kFramerate + 1 + kNumInitialPackets; ++i) {
     // NOTE!!! If the following line is moved under the if case then this test
     //         wont work on windows realease bots.
-    int cluster_id = i < kInitialProbingPackets ? 0 : PacketInfo::kNotAProbe;
+    int cluster_id =
+        i < kInitialProbingPackets ? 0 : PacedPacketInfo::kNotAProbe;
 
     if (i == kNumInitialPackets) {
       EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps));
       EXPECT_EQ(0u, ssrcs.size());
       EXPECT_FALSE(bitrate_observer_.updated());
       bitrate_observer_.Reset();
     }
     IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms,
                      sequence_number++, kMtu, cluster_id);
     clock_.AdvanceTimeMilliseconds(1000 / kFramerate);
@@ skipping 11 matching lines @@
 void DelayBasedBweTest::RateIncreaseReorderingTestHelper(
     uint32_t expected_bitrate_bps) {
   const int kFramerate = 50;  // 50 fps to avoid rounding errors.
   const int kFrameIntervalMs = 1000 / kFramerate;
   int64_t send_time_ms = 0;
   uint16_t sequence_number = 0;
   // Inserting packets for five seconds to get a valid estimate.
   for (int i = 0; i < 5 * kFramerate + 1 + kNumInitialPackets; ++i) {
     // NOTE!!! If the following line is moved under the if case then this test
     //         wont work on windows realease bots.
-    int cluster_id = i < kInitialProbingPackets ? 0 : PacketInfo::kNotAProbe;
+    int cluster_id =
+        i < kInitialProbingPackets ? 0 : PacedPacketInfo::kNotAProbe;
 
     // TODO(sprang): Remove this hack once the single stream estimator is gone,
     // as it doesn't do anything in Process().
     if (i == kNumInitialPackets) {
       // Process after we have enough frames to get a valid input rate estimate.
 
       EXPECT_FALSE(bitrate_observer_.updated());  // No valid estimate.
     }
     IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms,
                      sequence_number++, kMtu, cluster_id);
@@ skipping 167 matching lines @@
     IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms,
                      sequence_number++, 1000);
     clock_.AdvanceTimeMilliseconds(2 * kFrameIntervalMs);
     send_time_ms += kFrameIntervalMs;
   }
   uint32_t bitrate_after = 0;
   bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_after);
   EXPECT_LT(bitrate_after, bitrate_before);
 }
 }  // namespace webrtc