Revert of Remove unnecessary interface TelephoneEventHandler

webrtc/voice_engine/channel.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 791 matching lines @@
       rtp_header_parser_(RtpHeaderParser::Create()),
       rtp_payload_registry_(
           new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(true))),
       rtp_receive_statistics_(
           ReceiveStatistics::Create(Clock::GetRealTimeClock())),
       rtp_receiver_(
           RtpReceiver::CreateAudioReceiver(Clock::GetRealTimeClock(),
                                            this,
                                            this,
                                            rtp_payload_registry_.get())),
+      telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()),
       _outputAudioLevel(),
       _externalTransport(false),
       // Avoid conflict with other channels by adding 1024 - 1026,
       // won't use as much as 1024 channels.
       _inputFilePlayerId(VoEModuleId(instanceId, channelId) + 1024),
       _outputFilePlayerId(VoEModuleId(instanceId, channelId) + 1025),
       _outputFileRecorderId(VoEModuleId(instanceId, channelId) + 1026),
       _outputFileRecording(false),
       _outputExternalMedia(false),
       _inputExternalMediaCallbackPtr(NULL),
@@ skipping 149 matching lines @@
     return -1;
   }
 
   // --- RTP/RTCP module initialization
 
   // Ensure that RTCP is enabled by default for the created channel.
   // Note that, the module will keep generating RTCP until it is explicitly
   // disabled by the user.
   // After StopListen (when no sockets exists), RTCP packets will no longer
   // be transmitted since the Transport object will then be invalid.
+  telephone_event_handler_->SetTelephoneEventForwardToDecoder(true);
   // RTCP is enabled by default.
   _rtpRtcpModule->SetRTCPStatus(RtcpMode::kCompound);
   // --- Register all permanent callbacks
   const bool fail = (audio_coding_->RegisterTransportCallback(this) == -1) ||
                     (audio_coding_->RegisterVADCallback(this) == -1);
 
   if (fail) {
     _engineStatisticsPtr->SetLastError(
         VE_CANNOT_INIT_CHANNEL, kTraceError,
         "Channel::Init() callbacks not registered");
@@ skipping 2212 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc