RtpPacketHistory rewritten to use RtpPacket class.

webrtc/modules/rtp_rtcp/source/rtp_sender.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 17 matching lines @@
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_history.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/modules/rtp_rtcp/source/ssrc_database.h"
 #include "webrtc/transport.h"
 
 namespace webrtc {
 
 class RateLimiter;
+class RtcEventLog;
+class RtpPacketToSend;
 class RTPSenderAudio;
 class RTPSenderVideo;
-class RtcEventLog;
 
 class RTPSenderInterface {
  public:
   RTPSenderInterface() {}
   virtual ~RTPSenderInterface() {}
 
   virtual uint32_t SSRC() const = 0;
   virtual uint32_t Timestamp() const = 0;
 
   virtual int32_t BuildRTPheader(uint8_t* data_buffer,
@@ skipping 204 matching lines @@
                          const bool inc_sequence_number = true) override;
 
   size_t RtpHeaderLength() const override;
   uint16_t AllocateSequenceNumber(uint16_t packets_to_send) override;
   size_t MaxPayloadLength() const override;
 
   // Current timestamp.
   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.
 
   // Send a DTMF tone using RFC 2833 (4733).
   int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level);
 
   // Set audio packet size, used to determine when it's time to send a DTMF
   // packet in silence (CNG).
   int32_t SetAudioPacketSize(uint16_t packet_size_samples);
 
@@ skipping 57 matching lines @@
 
   size_t CreateRtpHeader(uint8_t* header,
                          int8_t payload_type,
                          uint32_t ssrc,
                          bool marker_bit,
                          uint32_t timestamp,
                          uint16_t sequence_number,
                          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);
 
   // Return the number of bytes sent.  Note that both of these functions may
   // 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);
+  std::unique_ptr<RtpPacketToSend> BuildRtxPacket(
+      const RtpPacketToSend& packet);
 
-  void BuildRtxPacket(uint8_t* buffer, size_t* length,
-                      uint8_t* buffer_rtx);
-
-  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);
   void UpdateOnSendPacket(int packet_id,
                           int64_t capture_time_ms,
                           uint32_t ssrc);
 
   // Find the byte position of the RTP extension as indicated by |type| in
   // |rtp_packet|. Return false if such extension doesn't exist.
   bool FindHeaderExtensionPosition(RTPExtensionType type,
                                    const uint8_t* rtp_packet,
                                    size_t rtp_packet_length,
                                    const RTPHeader& rtp_header,
                                    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,
                                 const RTPHeader& rtp_header,
                                 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_;
   Random random_ GUARDED_BY(send_critsect_);
 
   const bool audio_configured_;
   const std::unique_ptr<RTPSenderAudio> audio_;
   const std::unique_ptr<RTPSenderVideo> video_;
 
   RtpPacketSender* const paced_sender_;
@@ skipping 16 matching lines @@
   VideoRotation rotation_;
   bool video_rotation_active_;
   uint16_t transport_sequence_number_;
 
   // Tracks the current request for playout delay limits from application
   // and decides whether the current RTP frame should include the playout
   // delay extension on header.
   PlayoutDelayOracle playout_delay_oracle_;
   bool playout_delay_active_ GUARDED_BY(send_critsect_);
 
-  RTPPacketHistory packet_history_;
+  RtpPacketHistory packet_history_;
 
   // Statistics
   rtc::CriticalSection statistics_crit_;
   SendDelayMap send_delays_ GUARDED_BY(statistics_crit_);
   FrameCounts frame_counts_ GUARDED_BY(statistics_crit_);
   StreamDataCounters rtp_stats_ GUARDED_BY(statistics_crit_);
   StreamDataCounters rtx_rtp_stats_ GUARDED_BY(statistics_crit_);
   StreamDataCountersCallback* rtp_stats_callback_ GUARDED_BY(statistics_crit_);
   RateStatistics total_bitrate_sent_ GUARDED_BY(statistics_crit_);
   RateStatistics nack_bitrate_sent_ GUARDED_BY(statistics_crit_);
@@ skipping 25 matching lines @@
   std::map<int8_t, int8_t> rtx_payload_type_map_ GUARDED_BY(send_critsect_);
 
   RateLimiter* const retransmission_rate_limiter_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_