Remove now unused code in RtpHeaderExtensionMap

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 193 matching lines @@
   // Called on update of RTP statistics.
   void RegisterRtpStatisticsCallback(StreamDataCountersCallback* callback);
   StreamDataCountersCallback* GetRtpStatisticsCallback() const;
 
   uint32_t BitrateSent() const;
 
   void SetRtpState(const RtpState& rtp_state);
   RtpState GetRtpState() const;
   void SetRtxRtpState(const RtpState& rtp_state);
   RtpState GetRtxRtpState() const;
-  bool ActivateCVORtpHeaderExtension();
 
  protected:
   int32_t CheckPayloadType(int8_t payload_type, RtpVideoCodecTypes* video_type);
 
  private:
   // Maps capture time in milliseconds to send-side delay in milliseconds.
   // Send-side delay is the difference between transmission time and capture
   // time.
   typedef std::map<int64_t, int> SendDelayMap;
 
@@ skipping 18 matching lines @@
       const RtpPacketToSend& packet);
 
   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_);
-
   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;
-
   void UpdateRtpStats(const RtpPacketToSend& packet,
                       bool is_rtx,
                       bool is_retransmit);
   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_;
   TransportSequenceNumberAllocator* const transport_sequence_number_allocator_;
   TransportFeedbackObserver* const transport_feedback_observer_;
   int64_t last_capture_time_ms_sent_;
   rtc::CriticalSection send_critsect_;
 
   Transport *transport_;
   bool sending_media_ GUARDED_BY(send_critsect_);
 
   size_t max_payload_length_;
 
   int8_t payload_type_ GUARDED_BY(send_critsect_);
   std::map<int8_t, RtpUtility::Payload*> payload_type_map_;
 
   RtpHeaderExtensionMap rtp_header_extension_map_ GUARDED_BY(send_critsect_);
-  bool video_rotation_active_;
 
   // 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_;
 
   // 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_);
@@ skipping 26 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_