Rename webrtc::PacketInfo to webrtc::PacketFeedback

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 29 matching lines @@
       bitrate_bps_(bitrate_bps),
       next_rtp_time_(0),
       sequence_number_(0) {
   RTC_CHECK_GT(fps_, 0);
 }
 
 // Generates a new frame for this stream. If called too soon after the
 // previous frame, no frame will be generated. The frame is split into
 // packets.
 int64_t RtpStream::GenerateFrame(int64_t time_now_us,
-                                 std::vector<PacketInfo>* packets) {
+                                 std::vector<PacketFeedback>* packets) {
   if (time_now_us < next_rtp_time_) {
     return next_rtp_time_;
   }
   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_++);
+    PacketFeedback packet(-1, sequence_number_++);
     packet.send_time_ms = (time_now_us + kSendSideOffsetUs) / 1000;
     packet.payload_size = payload_size;
     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 {
@@ skipping 45 matching lines @@
         total_bitrate_before;
     stream->set_bitrate_bps(bitrate_after - total_bitrate_after);
     total_bitrate_after += stream->bitrate_bps();
   }
   ASSERT_EQ(bitrate_before, total_bitrate_before);
   EXPECT_EQ(total_bitrate_after, bitrate_bps);
 }
 
 // TODO(holmer): Break out the channel simulation part from this class to make
 // it possible to simulate different types of channels.
-int64_t StreamGenerator::GenerateFrame(std::vector<PacketInfo>* packets,
+int64_t StreamGenerator::GenerateFrame(std::vector<PacketFeedback>* packets,
                                        int64_t time_now_us) {
   RTC_CHECK(packets != NULL);
   RTC_CHECK(packets->empty());
   RTC_CHECK_GT(capacity_, 0);
   auto it =
       std::min_element(streams_.begin(), streams_.end(), RtpStream::Compare);
   (*it)->GenerateFrame(time_now_us, packets);
   int i = 0;
-  for (PacketInfo& packet : *packets) {
+  for (PacketFeedback& packet : *packets) {
     int capacity_bpus = capacity_ / 1000;
     int64_t required_network_time_us =
         (8 * 1000 * packet.payload_size + capacity_bpus / 2) / capacity_bpus;
     prev_arrival_time_us_ =
         std::max(time_now_us + required_network_time_us,
                  prev_arrival_time_us_ + required_network_time_us);
     packet.arrival_time_ms = prev_arrival_time_us_ / 1000;
     ++i;
   }
   it = std::min_element(streams_.begin(), streams_.end(), RtpStream::Compare);
@@ skipping 24 matching lines @@
   IncomingFeedback(arrival_time_ms, send_time_ms, sequence_number, payload_size,
                    PacedPacketInfo());
 }
 
 void DelayBasedBweTest::IncomingFeedback(int64_t arrival_time_ms,
                                          int64_t send_time_ms,
                                          uint16_t sequence_number,
                                          size_t payload_size,
                                          const PacedPacketInfo& pacing_info) {
   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, pacing_info);
-  std::vector<PacketInfo> packets;
+  PacketFeedback packet(arrival_time_ms + arrival_time_offset_ms_, send_time_ms,
+                        sequence_number, payload_size, pacing_info);
+  std::vector<PacketFeedback> packets;
   packets.push_back(packet);
   DelayBasedBwe::Result result =
       bitrate_estimator_->IncomingPacketFeedbackVector(packets);
   const uint32_t kDummySsrc = 0;
   if (result.updated) {
     bitrate_observer_.OnReceiveBitrateChanged({kDummySsrc},
                                               result.target_bitrate_bps);
   }
 }
 
 // Generates a frame of packets belonging to a stream at a given bitrate and
 // with a given ssrc. The stream is pushed through a very simple simulated
 // network, and is then given to the receive-side bandwidth estimator.
 // Returns true if an over-use was seen, false otherwise.
 // The StreamGenerator::updated() should be used to check for any changes in
 // target bitrate after the call to this function.
 bool DelayBasedBweTest::GenerateAndProcessFrame(uint32_t ssrc,
                                                 uint32_t bitrate_bps) {
   stream_generator_->SetBitrateBps(bitrate_bps);
-  std::vector<PacketInfo> packets;
+  std::vector<PacketFeedback> packets;
   int64_t next_time_us =
       stream_generator_->GenerateFrame(&packets, clock_.TimeInMicroseconds());
   if (packets.empty())
     return false;
 
   bool overuse = false;
   bitrate_observer_.Reset();
   clock_.AdvanceTimeMicroseconds(1000 * packets.back().arrival_time_ms -
                                  clock_.TimeInMicroseconds());
   for (auto& packet : packets) {
@@ skipping 283 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