Rename webrtc::PacketInfo to webrtc::PacketFeedback

webrtc/modules/congestion_controller/delay_based_bwe_unittest_helper.h

 /*
  *  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.
  */
 
@@ skipping 36 matching lines @@
 
 class RtpStream {
  public:
   enum { kSendSideOffsetUs = 1000000 };
 
   RtpStream(int fps, int bitrate_bps);
 
   // 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 GenerateFrame(int64_t time_now_us, std::vector<PacketInfo>* packets);
+  int64_t GenerateFrame(int64_t time_now_us,
+                        std::vector<PacketFeedback>* packets);
 
   // The send-side time when the next frame can be generated.
   int64_t next_rtp_time() const;
 
   void set_bitrate_bps(int bitrate_bps);
 
   int bitrate_bps() const;
 
   static bool Compare(const std::unique_ptr<RtpStream>& lhs,
                       const std::unique_ptr<RtpStream>& rhs);
@@ skipping 19 matching lines @@
 
   // Divides |bitrate_bps| among all streams. The allocated bitrate per stream
   // is decided by the initial allocation ratios.
   void SetBitrateBps(int bitrate_bps);
 
   // Set the RTP timestamp offset for the stream identified by |ssrc|.
   void set_rtp_timestamp_offset(uint32_t ssrc, uint32_t offset);
 
   // TODO(holmer): Break out the channel simulation part from this class to make
   // it possible to simulate different types of channels.
-  int64_t GenerateFrame(std::vector<PacketInfo>* packets, int64_t time_now_us);
+  int64_t GenerateFrame(std::vector<PacketFeedback>* packets,
+                        int64_t time_now_us);
 
  private:
   // Capacity of the simulated channel in bits per second.
   int capacity_;
   // The time when the last packet arrived.
   int64_t prev_arrival_time_us_;
   // All streams being transmitted on this simulated channel.
   std::vector<std::unique_ptr<RtpStream>> streams_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(StreamGenerator);
@@ skipping 57 matching lines @@
   std::unique_ptr<DelayBasedBwe> bitrate_estimator_;
   std::unique_ptr<test::StreamGenerator> stream_generator_;
   int64_t arrival_time_offset_ms_;
   bool first_update_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(DelayBasedBweTest);
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_CONGESTION_CONTROLLER_DELAY_BASED_BWE_UNITTEST_HELPER_H_