Add time line for acked bitrate.

webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
 #define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
 
 #include <stddef.h>
 #include <list>
+#include <vector>
 
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/typedefs.h"
 
 #define RTCP_CNAME_SIZE 256    // RFC 3550 page 44, including null termination
 #define IP_PACKET_SIZE 1500    // we assume ethernet
 #define MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS 10
 #define TIMEOUT_SEI_MESSAGES_MS 30000   // in milliseconds
 
@@ skipping 276 matching lines @@
  public:
   TransportFeedbackObserver() {}
   virtual ~TransportFeedbackObserver() {}
 
   // Note: Transport-wide sequence number as sequence number. Arrival time
   // must be set to 0.
   virtual void AddPacket(uint16_t sequence_number,
                          size_t length,
                          int probe_cluster_id) = 0;
 
-  virtual void OnTransportFeedback(const rtcp::TransportFeedback& feedback) = 0;
+  virtual std::vector<PacketInfo> OnTransportFeedback(
+      const rtcp::TransportFeedback& feedback) = 0;
 };
 
 class RtcpRttStats {
  public:
   virtual void OnRttUpdate(int64_t rtt) = 0;
 
   virtual int64_t LastProcessedRtt() const = 0;
 
   virtual ~RtcpRttStats() {}
 };
@@ skipping 70 matching lines @@
 class TransportSequenceNumberAllocator {
  public:
   TransportSequenceNumberAllocator() {}
   virtual ~TransportSequenceNumberAllocator() {}
 
   virtual uint16_t AllocateSequenceNumber() = 0;
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_