Deprecated SetAudioPacketSize from RTPSender and removed calls to it.

webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h

Index: webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
index ee4831c6348c5277efa7632cee974fb5a376f65a..cf79120bb9ae932fa7627a6fab4ee9305b916111 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
@@ -42,10 +42,6 @@ class RTPSenderAudio {
                  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.
@@ -69,8 +65,6 @@ class RTPSenderAudio {
 
   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;