Add TransportFeedback adapter, adapting remote feedback to bwe estiamtor

webrtc/modules/rtp_rtcp/interface/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_INTERFACE_RTP_RTCP_DEFINES_H_
 #define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_
 
 #include <stddef.h>
 #include <list>
 
 #include "webrtc/modules/interface/module_common_types.h"
 #include "webrtc/system_wrappers/interface/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
 
 namespace webrtc {
+namespace rtcp {
+class TransportFeedback;
+}
 
 const int kVideoPayloadTypeFrequency = 90000;
 
 // Minimum RTP header size in bytes.
 const uint8_t kRtpHeaderSize = 12;
 
 struct AudioPayload
 {
     uint32_t    frequency;
     uint8_t     channels;
@@ skipping 249 matching lines @@
   virtual void OnReceivedEstimatedBitrate(uint32_t bitrate) = 0;
 
   virtual void OnReceivedRtcpReceiverReport(
       const ReportBlockList& report_blocks,
       int64_t rtt,
       int64_t now_ms) = 0;
 
   virtual ~RtcpBandwidthObserver() {}
 };
 
-class SendTimeObserver {
+struct PacketInfo {
+  PacketInfo(int64_t arrival_time_ms,
+             int64_t send_time_ms,
+             uint16_t sequence_number,
+             size_t payload_size,
+             bool was_paced)
+      : arrival_time_ms(arrival_time_ms),
+        send_time_ms(send_time_ms),
+        sequence_number(sequence_number),
+        payload_size(payload_size),
+        was_paced(was_paced) {}
+  // Time corresponding to when the packet was received. Timestamped with the
+  // receiver's clock.
+  int64_t arrival_time_ms;
+  // Time corresponding to when the packet was sent, timestamped with the
+  // sender's clock.
+  int64_t send_time_ms;
+  // Packet identifier, incremented with 1 for every packet generated by the
+  // sender.
+  uint16_t sequence_number;
+  // Size of the packet excluding RTP headers.
+  size_t payload_size;
+  // True if the packet was paced out by the pacer.
+  bool was_paced;
+};
+
+class TransportFeedbackObserver {
  public:
-  SendTimeObserver() {}
-  virtual ~SendTimeObserver() {}
+  TransportFeedbackObserver() {}
+  virtual ~TransportFeedbackObserver() {}
 
-  // Transport-wide sequence number and timestamp (system time in milliseconds),
-  // of when the packet was put on the wire.
-  virtual void OnPacketSent(uint16_t transport_sequence_number,
-                            int64_t send_time) = 0;
+  // Note: Transport-wide sequence number as sequence number. Arrival time
+  // must be set to 0.
+  virtual void OnPacketSent(const PacketInfo& info) = 0;
+
+  virtual void 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 54 matching lines @@
   uint64_t single_packet_loss_count;
   // The number of events in which more than one adjacent packet was lost.
   uint64_t multiple_packet_loss_event_count;
   // The number of packets lost in events where more than one adjacent packet
   // was lost.
   uint64_t multiple_packet_loss_packet_count;
 };
 
 }  // namespace webrtc
 #endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_