RtpPacketHistory rewritten to use RtpPacket class.

webrtc/modules/rtp_rtcp/source/rtp_sender.h

Index: webrtc/modules/rtp_rtcp/source/rtp_sender.h
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
index f39309a5d1243eb013e8fb88e5d3e2513a4ec688..edf27b10bba078cad5a032738bb656da965228e8 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
@@ -35,9 +35,10 @@
 namespace webrtc {
 
 class RateLimiter;
+class RtcEventLog;
+class RtpPacketToSend;
 class RTPSenderAudio;
 class RTPSenderVideo;
-class RtcEventLog;
 
 class RTPSenderInterface {
  public:
@@ -262,12 +263,16 @@ class RTPSender : public RTPSenderInterface {
   uint32_t Timestamp() const override;
   uint32_t SSRC() const override;
 
+  // Deprecated. Create RtpPacketToSend instead and use next function.
   int32_t SendToNetwork(uint8_t* data_buffer,
                         size_t payload_length,
                         size_t rtp_header_length,
                         int64_t capture_time_ms,
                         StorageType storage,
                         RtpPacketSender::Priority priority) override;
+  bool SendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
+                     StorageType storage,
+                     RtpPacketSender::Priority priority);
 
   // Audio.
 
@@ -345,9 +350,7 @@ class RTPSender : public RTPSenderInterface {
                          const std::vector<uint32_t>& csrcs) const
       EXCLUSIVE_LOCKS_REQUIRED(send_critsect_);
 
-  bool PrepareAndSendPacket(uint8_t* buffer,
-                            size_t length,
-                            int64_t capture_time_ms,
+  bool PrepareAndSendPacket(std::unique_ptr<RtpPacketToSend> packet,
                             bool send_over_rtx,
                             bool is_retransmit,
                             int probe_cluster_id);
@@ -356,15 +359,10 @@ class RTPSender : public RTPSenderInterface {
   // return a larger value that their argument.
   size_t TrySendRedundantPayloads(size_t bytes, int probe_cluster_id);
 
-  void BuildPaddingPacket(uint8_t* packet,
-                          size_t header_length,
-                          size_t padding_length);
-
-  void BuildRtxPacket(uint8_t* buffer, size_t* length,
-                      uint8_t* buffer_rtx);
+  std::unique_ptr<RtpPacketToSend> BuildRtxPacket(
+      const RtpPacketToSend& packet);
 
-  bool SendPacketToNetwork(const uint8_t* packet,
-                           size_t size,
+  bool SendPacketToNetwork(const RtpPacketToSend& packet,
                            const PacketOptions& options);
 
   void UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms);
@@ -381,19 +379,8 @@ class RTPSender : public RTPSenderInterface {
                                    size_t* position) const
       EXCLUSIVE_LOCKS_REQUIRED(send_critsect_);
 
-  void UpdateTransmissionTimeOffset(uint8_t* rtp_packet,
-                                    size_t rtp_packet_length,
-                                    const RTPHeader& rtp_header,
-                                    int64_t time_diff_ms) const;
-  void UpdateAbsoluteSendTime(uint8_t* rtp_packet,
-                              size_t rtp_packet_length,
-                              const RTPHeader& rtp_header,
-                              int64_t now_ms) const;
-
-  bool UpdateTransportSequenceNumber(uint8_t* rtp_packet,
-                                     size_t rtp_packet_length,
-                                     const RTPHeader& rtp_header,
-                                     int* sequence_number) const;
+  bool UpdateTransportSequenceNumber(RtpPacketToSend* packet,
+                                     int* packet_id) const;
 
   void UpdatePlayoutDelayLimits(uint8_t* rtp_packet,
                                 size_t rtp_packet_length,
@@ -401,14 +388,10 @@ class RTPSender : public RTPSenderInterface {
                                 uint16_t min_playout_delay,
                                 uint16_t max_playout_delay) const;
 
-  bool AllocateTransportSequenceNumber(int* packet_id) const;
-
-  void UpdateRtpStats(const uint8_t* buffer,
-                      size_t packet_length,
-                      const RTPHeader& header,
+  void UpdateRtpStats(const RtpPacketToSend& packet,
                       bool is_rtx,
                       bool is_retransmit);
-  bool IsFecPacket(const uint8_t* buffer, const RTPHeader& header) const;
+  bool IsFecPacket(const RtpPacketToSend& packet) const;
 
   Clock* const clock_;
   const int64_t clock_delta_ms_;
@@ -445,7 +428,7 @@ class RTPSender : public RTPSenderInterface {
   PlayoutDelayOracle playout_delay_oracle_;
   bool playout_delay_active_ GUARDED_BY(send_critsect_);
 
-  RTPPacketHistory packet_history_;
+  RtpPacketHistory packet_history_;
 
   // Statistics
   rtc::CriticalSection statistics_crit_;