Deprecated SetAudioPacketSize from RTPSender and removed calls to it.

webrtc/modules/rtp_rtcp/source/rtp_sender.h

Index: webrtc/modules/rtp_rtcp/source/rtp_sender.h
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
index 66db7502e98c8e58f237a880cc0c26eea5687f44..080165a0bdf350c87c1bf1179a81c55117a03d07 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
@@ -179,9 +179,9 @@ class RTPSender {
   // 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.