Remove unnecessary interface TelephoneEventHandler

webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.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 182 matching lines @@
       if (last_received_timestamp_ != rtp_header.timestamp) {
         last_received_timestamp_ = rtp_header.timestamp;
         last_received_frame_time_ms_ = clock_->TimeInMilliseconds();
       }
       last_received_sequence_number_ = rtp_header.sequenceNumber;
     }
   }
   return true;
 }
 
-TelephoneEventHandler* RtpReceiverImpl::GetTelephoneEventHandler() {
-  return rtp_media_receiver_->GetTelephoneEventHandler();
-}
-
 bool RtpReceiverImpl::Timestamp(uint32_t* timestamp) const {
   rtc::CritScope lock(&critical_section_rtp_receiver_);
   if (!HaveReceivedFrame())
     return false;
   *timestamp = last_received_timestamp_;
   return true;
 }
 
 bool RtpReceiverImpl::LastReceivedTimeMs(int64_t* receive_time_ms) const {
   rtc::CritScope lock(&critical_section_rtp_receiver_);
@@ skipping 239 matching lines @@
     // implementations might have CSRC 0 as a valid value.
     if (num_csrcs_diff > 0) {
       cb_rtp_feedback_->OnIncomingCSRCChanged(0, true);
     } else if (num_csrcs_diff < 0) {
       cb_rtp_feedback_->OnIncomingCSRCChanged(0, false);
     }
   }
 }
 
 }  // namespace webrtc