Deprecated SetAudioPacketSize from RTPSender and removed calls to it.

webrtc/modules/rtp_rtcp/source/rtp_sender_audio.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 24 matching lines @@
                                uint32_t rate,
                                RtpUtility::Payload** payload);
 
   bool SendAudio(FrameType frame_type,
                  int8_t payload_type,
                  uint32_t capture_timestamp,
                  const uint8_t* payload_data,
                  size_t payload_size,
                  const RTPFragmentationHeader* fragmentation);
 
-  // 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);
-
   // Store the audio level in dBov for
   // header-extension-for-audio-level-indication.
   // Valid range is [0,100]. Actual value is negative.
   int32_t SetAudioLevel(uint8_t level_dbov);
 
   // Send a DTMF tone using RFC 2833 (4733)
   int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level);
 
  protected:
   bool SendTelephoneEventPacket(
       bool ended,
       uint32_t dtmf_timestamp,
       uint16_t duration,
       bool marker_bit);  // set on first packet in talk burst
 
   bool MarkerBit(FrameType frame_type, int8_t payload_type);
 
  private:
   Clock* const clock_ = nullptr;
   RTPSender* const rtp_sender_ = nullptr;
 
   rtc::CriticalSection send_audio_critsect_;
 
-  uint16_t packet_size_samples_ GUARDED_BY(send_audio_critsect_) = 160;
-
   // DTMF.
   bool dtmf_event_is_on_ = false;
   bool dtmf_event_first_packet_sent_ = false;
   int8_t dtmf_payload_type_ GUARDED_BY(send_audio_critsect_) = -1;
   uint32_t dtmf_payload_freq_ GUARDED_BY(send_audio_critsect_) = 8000;
   uint32_t dtmf_timestamp_ = 0;
   uint32_t dtmf_length_samples_ = 0;
   int64_t dtmf_time_last_sent_ = 0;
   uint32_t dtmf_timestamp_last_sent_ = 0;
   DtmfQueue::Event dtmf_current_event_;
@@ skipping 11 matching lines @@
   // (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/)
   uint8_t audio_level_dbov_ GUARDED_BY(send_audio_critsect_) = 0;
   OneTimeEvent first_packet_sent_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSenderAudio);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_