Renamed fields in rtp_rtcp_defines.h/RTCPReportBlock

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.
  */
 
@@ skipping 114 matching lines @@
   kRtxOff                 = 0x0,
   kRtxRetransmitted       = 0x1,  // Only send retransmissions over RTX.
   kRtxRedundantPayloads   = 0x2   // Preventively send redundant payloads
                                   // instead of padding.
 };
 
 const size_t kRtxHeaderSize = 2;
 
 struct RTCPReportBlock {
   RTCPReportBlock()
-      : remoteSSRC(0), sourceSSRC(0), fractionLost(0), cumulativeLost(0),
-        extendedHighSeqNum(0), jitter(0), lastSR(0),
-        delaySinceLastSR(0) {}
+      : sender_ssrc(0),
+        source_ssrc(0),
+        fraction_lost(0),
+        packets_lost(0),
+        extended_highest_sequence_number(0),
+        jitter(0),
+        last_sender_report_timestamp(0),
+        delay_since_last_sender_report(0) {}
 
-  RTCPReportBlock(uint32_t remote_ssrc,
+  RTCPReportBlock(uint32_t sender_ssrc,
                   uint32_t source_ssrc,
                   uint8_t fraction_lost,
-                  uint32_t cumulative_lost,
-                  uint32_t extended_high_sequence_number,
+                  uint32_t packets_lost,
+                  uint32_t extended_highest_sequence_number,
                   uint32_t jitter,
-                  uint32_t last_sender_report,
+                  uint32_t last_sender_report_timestamp,
                   uint32_t delay_since_last_sender_report)
-      : remoteSSRC(remote_ssrc),
-        sourceSSRC(source_ssrc),
-        fractionLost(fraction_lost),
-        cumulativeLost(cumulative_lost),
-        extendedHighSeqNum(extended_high_sequence_number),
+      : sender_ssrc(sender_ssrc),
+        source_ssrc(source_ssrc),
+        fraction_lost(fraction_lost),
+        packets_lost(packets_lost),
+        extended_highest_sequence_number(extended_highest_sequence_number),
         jitter(jitter),
-        lastSR(last_sender_report),
-        delaySinceLastSR(delay_since_last_sender_report) {}
+        last_sender_report_timestamp(last_sender_report_timestamp),
+        delay_since_last_sender_report(delay_since_last_sender_report) {}
 
   // Fields as described by RFC 3550 6.4.2.
-  uint32_t remoteSSRC;  // SSRC of sender of this report.
-  uint32_t sourceSSRC;  // SSRC of the RTP packet sender.
-  uint8_t fractionLost;
-  uint32_t cumulativeLost;  // 24 bits valid.
-  uint32_t extendedHighSeqNum;
+  union {
+    uint32_t sender_ssrc;  // SSRC of sender of this report.
+    RTC_DEPRECATED uint32_t remoteSSRC;
+  };
+  union {
+    uint32_t source_ssrc;  // SSRC of the RTP packet sender.
+    RTC_DEPRECATED uint32_t sourceSSRC;
+  };
+  union {
+    RTC_DEPRECATED uint8_t fractionLost;
+    uint8_t fraction_lost;
+  };
+  union {
+    uint32_t packets_lost;  // 24 bits valid.
+    RTC_DEPRECATED uint32_t cumulativeLost;
+  };
+  union {
+    uint32_t extended_highest_sequence_number;
+    RTC_DEPRECATED uint32_t extendedHighSeqNum;
+  };
   uint32_t jitter;
-  uint32_t lastSR;
-  uint32_t delaySinceLastSR;
+  union {
+    uint32_t last_sender_report_timestamp;
+    RTC_DEPRECATED uint32_t lastSR;
+  };
+  union {
+    uint32_t delay_since_last_sender_report;
+    RTC_DEPRECATED uint32_t delaySinceLastSR;
+  };
 };
 
 typedef std::list<RTCPReportBlock> ReportBlockList;
 
 struct RtpState {
   RtpState()
       : sequence_number(0),
         start_timestamp(0),
         timestamp(0),
         capture_time_ms(-1),
@@ skipping 274 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_