Deprecated SetAudioPacketSize from RTPSender and removed calls to it.

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.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 242 matching lines @@
   // (XR) VOIP metric.
   int32_t SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric) override;
 
   // (XR) Receiver reference time report.
   void SetRtcpXrRrtrStatus(bool enable) override;
 
   bool RtcpXrRrtrStatus() const override;
 
   // Audio part.
 
-  // Set 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).
   int32_t SetAudioPacketSize(uint16_t packet_size_samples) override;
 
   // Send a TelephoneEvent tone using RFC 2833 (4733).
   int32_t SendTelephoneEventOutband(uint8_t key,
                                     uint16_t time_ms,
                                     uint8_t level) override;
 
   // Store the audio level in d_bov for header-extension-for-audio-level-
   // indication.
   int32_t SetAudioLevel(uint8_t level_d_bov) override;
@@ skipping 81 matching lines @@
   PacketLossStats receive_loss_stats_;
 
   // The processed RTT from RtcpRttStats.
   rtc::CriticalSection critical_section_rtt_;
   int64_t rtt_ms_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_