Remove unnecessary interface TelephoneEventHandler

webrtc/modules/rtp_rtcp/source/rtp_receiver_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.
  */
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h"
 
 #include <assert.h>  // assert
 #include <math.h>   // pow()
 #include <string.h>  // memcpy()
 
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
 
 namespace webrtc {
 RTPReceiverStrategy* RTPReceiverStrategy::CreateAudioStrategy(
     RtpData* data_callback) {
   return new RTPReceiverAudio(data_callback);
 }
 
 RTPReceiverAudio::RTPReceiverAudio(RtpData* data_callback)
     : RTPReceiverStrategy(data_callback),
-      TelephoneEventHandler(),
       last_received_frequency_(8000),
-      telephone_event_forward_to_decoder_(false),
       telephone_event_payload_type_(-1),
       cng_nb_payload_type_(-1),
       cng_wb_payload_type_(-1),
       cng_swb_payload_type_(-1),
       cng_fb_payload_type_(-1),
       cng_payload_type_(-1),
       g722_payload_type_(-1),
       last_received_g722_(false),
       num_energy_(0),
       current_remote_energy_() {
   last_payload_.Audio.channels = 1;
   memset(current_remote_energy_, 0, sizeof(current_remote_energy_));
 }
 
-// Outband TelephoneEvent(DTMF) detection
-void RTPReceiverAudio::SetTelephoneEventForwardToDecoder(
-    bool forward_to_decoder) {
-  rtc::CritScope lock(&crit_sect_);
-  telephone_event_forward_to_decoder_ = forward_to_decoder;
-}
-
-// Is forwarding of outband telephone events turned on/off?
-bool RTPReceiverAudio::TelephoneEventForwardToDecoder() const {
-  rtc::CritScope lock(&crit_sect_);
-  return telephone_event_forward_to_decoder_;
-}
-
 bool RTPReceiverAudio::TelephoneEventPayloadType(
     int8_t payload_type) const {
   rtc::CritScope lock(&crit_sect_);
   return telephone_event_payload_type_ == payload_type;
 }
 
 bool RTPReceiverAudio::CNGPayloadType(int8_t payload_type,
                                       uint32_t* frequency,
                                       bool* cng_payload_type_has_changed) {
   rtc::CritScope lock(&crit_sect_);
@@ skipping 281 matching lines @@
                        &also_ignored)) {
       rtp_header->type.Audio.isCNG = true;
       rtp_header->frameType = kAudioFrameCN;
     } else {
       rtp_header->frameType = kAudioFrameSpeech;
       rtp_header->type.Audio.isCNG = false;
     }
 
     // check if it's a DTMF event, hence something we can playout
     if (telephone_event_packet) {
-      if (!telephone_event_forward_to_decoder_) {
-        // don't forward event to decoder
-        return 0;
-      }
       std::set<uint8_t>::iterator first =
           telephone_event_reported_.begin();
       if (first != telephone_event_reported_.end() && *first > 15) {
         // don't forward non DTMF events
         return 0;
       }
     }
   }
   // TODO(holmer): Break this out to have RED parsing handled generically.
   if (is_red && !(payload_data[0] & 0x80)) {
     // we recive only one frame packed in a RED packet remove the RED wrapper
     rtp_header->header.payloadType = payload_data[0];
 
     // only one frame in the RED strip the one byte to help NetEq
     return data_callback_->OnReceivedPayloadData(
         payload_data + 1, payload_length - 1, rtp_header);
   }
 
   rtp_header->type.Audio.channel = audio_specific.channels;
   return data_callback_->OnReceivedPayloadData(
       payload_data, payload_length, rtp_header);
 }
 }  // namespace webrtc