Deprecated SetAudioPacketSize from RTPSender and removed calls to it.

webrtc/modules/rtp_rtcp/include/rtp_rtcp.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 410 matching lines @@
   virtual void RegisterRtcpStatisticsCallback(
       RtcpStatisticsCallback* callback) = 0;
   virtual RtcpStatisticsCallback* GetRtcpStatisticsCallback() = 0;
   // BWE feedback packets.
   virtual bool SendFeedbackPacket(const rtcp::TransportFeedback& packet) = 0;
 
   // **************************************************************************
   // Audio
   // **************************************************************************
 
-  // Sets audio packet size, used to determine when it's time to send a DTMF
-  // packet in silence (CNG).
+  // This function is deprecated. It was previously used to determine when it
+  // was time to send a DTMF packet in silence (CNG).
   // Returns -1 on failure else 0.
-  virtual int32_t SetAudioPacketSize(uint16_t packet_size_samples) = 0;
+  RTC_DEPRECATED virtual int32_t SetAudioPacketSize(
+      uint16_t packet_size_samples) = 0;
 
   // Sends a TelephoneEvent tone using RFC 2833 (4733).
   // Returns -1 on failure else 0.
   virtual int32_t SendTelephoneEventOutband(uint8_t key,
                                             uint16_t time_ms,
                                             uint8_t level) = 0;
 
   // Store the audio level in dBov for header-extension-for-audio-level-
   // indication.
   // This API shall be called before transmision of an RTP packet to ensure
@@ skipping 32 matching lines @@
   virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0;
 
   // Sends a request for a keyframe.
   // Returns -1 on failure else 0.
   virtual int32_t RequestKeyFrame() = 0;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_