Deprecated SetAudioPacketSize from RTPSender and removed calls to it.

webrtc/modules/rtp_rtcp/include/rtp_rtcp.h

Index: webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
index bb9e1f2959878b77886e45652e2e0c6eafd60b72..98442ad6bc7f06496fb0db8f095d528e4103f459 100644
--- a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
@@ -428,10 +428,11 @@ class RtpRtcp : public Module {
   // Audio
   // **************************************************************************
 
-  // Sets 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).
   // Returns -1 on failure else 0.
-  virtual int32_t SetAudioPacketSize(uint16_t packet_size_samples) = 0;
+  RTC_DEPRECATED virtual int32_t SetAudioPacketSize(
+      uint16_t packet_size_samples) = 0;
 
   // Sends a TelephoneEvent tone using RFC 2833 (4733).
   // Returns -1 on failure else 0.