Add time line for acked bitrate.

webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.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 118 matching lines @@
 };
 
 class MockSendPacketObserver : public SendPacketObserver {
  public:
   MOCK_METHOD3(OnSendPacket, void(uint16_t, int64_t, uint32_t));
 };
 
 class MockTransportFeedbackObserver : public TransportFeedbackObserver {
  public:
   MOCK_METHOD3(AddPacket, void(uint16_t, size_t, int));
-  MOCK_METHOD1(OnTransportFeedback, void(const rtcp::TransportFeedback&));
+  MOCK_METHOD1(OnTransportFeedback,
+               std::vector<PacketInfo>(const rtcp::TransportFeedback&));
 };
 
 class RtpSenderTest : public ::testing::Test {
  protected:
   RtpSenderTest()
       : fake_clock_(kStartTime),
         mock_rtc_event_log_(),
         mock_paced_sender_(),
         retransmission_rate_limiter_(&fake_clock_, 1000),
         rtp_sender_(),
@@ skipping 1633 matching lines @@
       reinterpret_cast<uint8_t*>(transport_.sent_packets_[0]->data()),
       transport_.sent_packets_[0]->size(), true, &map, kSeqNum, hdr.rotation);
 
   // Verify that this packet does have CVO byte.
   VerifyCVOPacket(
       reinterpret_cast<uint8_t*>(transport_.sent_packets_[1]->data()),
       transport_.sent_packets_[1]->size(), true, &map, kSeqNum + 1,
       hdr.rotation);
 }
 }  // namespace webrtc