Remove deprecated RTPSender::SendPadData

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 179 matching lines @@
   RtpVideoCodecTypes VideoCodecType() const;
 
   uint32_t MaxConfiguredBitrateVideo() const;
 
   // ULPFEC.
   void SetUlpfecConfig(int red_payload_type, int ulpfec_payload_type);
 
   bool SetFecParameters(const FecProtectionParams& delta_params,
                         const FecProtectionParams& key_params);
 
-  RTC_DEPRECATED
-  size_t SendPadData(size_t bytes,
-                     bool timestamp_provided,
-                     uint32_t timestamp,
-                     int64_t capture_time_ms);
-
   // 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;
 
   void SetTransportOverhead(int transport_overhead);
 
  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;
 
   size_t SendPadData(size_t bytes, int probe_cluster_id);
 
-  size_t DeprecatedSendPadData(size_t bytes,
-                               bool timestamp_provided,
-                               uint32_t timestamp,
-                               int64_t capture_time_ms,
-                               int probe_cluster_id);
-
   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);
 
   std::unique_ptr<RtpPacketToSend> BuildRtxPacket(
@@ skipping 94 matching lines @@
 
   RateLimiter* const retransmission_rate_limiter_;
   OverheadObserver* overhead_observer_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_