Deprecated SetAudioPacketSize from RTPSender and removed calls to it.

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 161 matching lines @@
 
   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);
+  // This function is deprecated. It was previously used to determine when it
+  // was time to send a DTMF packet in silence (CNG).
+  RTC_DEPRECATED int32_t SetAudioPacketSize(uint16_t packet_size_samples);
 
   // Store the audio level in d_bov for
   // header-extension-for-audio-level-indication.
   int32_t SetAudioLevel(uint8_t level_d_bov);
 
   RtpVideoCodecTypes VideoCodecType() const;
 
   uint32_t MaxConfiguredBitrateVideo() const;
 
   // ULPFEC.
@@ skipping 145 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_