Remove unnecessary interface TelephoneEventHandler

webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h

 /*
  *  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 26 matching lines @@
   // provides audio or video-specific data. The is_first_packet argument is true
   // if this packet is either the first packet ever or the first in its frame.
   virtual int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header,
                                  const PayloadUnion& specific_payload,
                                  bool is_red,
                                  const uint8_t* payload,
                                  size_t payload_length,
                                  int64_t timestamp_ms,
                                  bool is_first_packet) = 0;
 
-  virtual TelephoneEventHandler* GetTelephoneEventHandler() = 0;
-
   // Retrieves the last known applicable frequency.
   virtual int GetPayloadTypeFrequency() const = 0;
 
   // Computes the current dead-or-alive state.
   virtual RTPAliveType ProcessDeadOrAlive(
       uint16_t last_payload_length) const = 0;
 
   // Returns true if we should report CSRC changes for this payload type.
   // TODO(phoglund): should move out of here along with other payload stuff.
   virtual bool ShouldReportCsrcChanges(uint8_t payload_type) const = 0;
@@ skipping 35 matching lines @@
   // packet.
   explicit RTPReceiverStrategy(RtpData* data_callback);
 
   rtc::CriticalSection crit_sect_;
   PayloadUnion last_payload_;
   RtpData* data_callback_;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_