Remove chain of methods in RtpRtcp module to get current payload frequency for RTCP SRs

webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc

 /*
  *  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 32 matching lines @@
       inband_vad_active_(false),
       cngnb_payload_type_(-1),
       cngwb_payload_type_(-1),
       cngswb_payload_type_(-1),
       cngfb_payload_type_(-1),
       last_payload_type_(-1),
       audio_level_dbov_(0) {}
 
 RTPSenderAudio::~RTPSenderAudio() {}
 
-int RTPSenderAudio::AudioFrequency() const {
-  return kDtmfFrequencyHz;
-}
-
 // set audio packet size, used to determine when it's time to send a DTMF packet
 // in silence (CNG)
 int32_t RTPSenderAudio::SetAudioPacketSize(uint16_t packet_size_samples) {
   rtc::CritScope cs(&send_audio_critsect_);
   packet_size_samples_ = packet_size_samples;
   return 0;
 }
 
 int32_t RTPSenderAudio::RegisterAudioPayload(
     const char payloadName[RTP_PAYLOAD_NAME_SIZE],
@@ skipping 298 matching lines @@
         TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "Audio::SendTelephoneEvent",
         "timestamp", packet->Timestamp(), "seqnum", packet->SequenceNumber());
     result = rtp_sender_->SendToNetwork(std::move(packet), kAllowRetransmission,
                                         RtpPacketSender::kHighPriority);
     send_count--;
   } while (send_count > 0 && result);
 
   return result;
 }
 }  // namespace webrtc