Deprecated SetAudioPacketSize from RTPSender and removed calls to it.

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 339e4f0d5654333f6225da5d240b59c211bab225..ca7ba3502983032d6448ae527dad94c133899683 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -777,11 +777,9 @@ int32_t ModuleRtpRtcpImpl::SendTelephoneEventOutband(
   return rtp_sender_.SendTelephoneEvent(key, time_ms, level);
 }
 
-// Set audio packet size, used to determine when it's time to send a DTMF
-// packet in silence (CNG).
 int32_t ModuleRtpRtcpImpl::SetAudioPacketSize(
     const uint16_t packet_size_samples) {
-  return rtp_sender_.SetAudioPacketSize(packet_size_samples);
+  return audio_ ? 0 : -1;
 }
 
 int32_t ModuleRtpRtcpImpl::SetAudioLevel(