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 f501d27a723c62f745b3120d76e7e338734e1da0..e8334e5b60b2406940308acbc79d8144b59b743b 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
@@ -33,9 +33,10 @@
 
 namespace webrtc {
 
+class RtcEventLog;
+class RtpPacketToSend;
 class RTPSenderAudio;
 class RTPSenderVideo;
-class RtcEventLog;
 
 class RTPSenderInterface {
  public:
@@ -255,12 +256,16 @@ 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.
 
@@ -333,9 +338,7 @@ class RTPSender : public RTPSenderInterface {
 
   void UpdateNACKBitRate(uint32_t bytes, int64_t now);
 
-  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);
 
@@ -343,15 +346,10 @@ class RTPSender : public RTPSenderInterface {
   // return a larger value that their argument.
   size_t TrySendRedundantPayloads(size_t bytes);
 
-  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);
@@ -367,28 +365,12 @@ class RTPSender : public RTPSenderInterface {
                                    const RTPHeader& rtp_header,
                                    size_t* position) const;
 
-  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(uint16_t sequence_number,
-                                     uint8_t* rtp_packet,
-                                     size_t rtp_packet_length,
-                                     const RTPHeader& rtp_header) 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;
 
   class BitrateAggregator {
    public:
@@ -459,7 +441,7 @@ class RTPSender : public RTPSenderInterface {
   size_t nack_byte_count_[NACK_BYTECOUNT_SIZE];
   Bitrate nack_bitrate_;
 
-  RTPPacketHistory packet_history_;
+  RtpPacketHistory packet_history_;
 
   // Statistics
   rtc::CriticalSection statistics_crit_;