Remove unnecessary interface TelephoneEventHandler

webrtc/modules/rtp_rtcp/test/testAPI/test_api_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 144 matching lines @@
   uint16_t test_sequence_number;
   uint32_t test_CSRC[webrtc::kRtpCsrcSize];
   SimulatedClock fake_clock;
   RateLimiter retransmission_rate_limiter_;
 };
 
 TEST_F(RtpRtcpAudioTest, Basic) {
   module1->SetSSRC(test_ssrc);
   module1->SetStartTimestamp(test_timestamp);
 
-  // Test detection at the end of a DTMF tone.
-  // EXPECT_EQ(0, module2->SetTelephoneEventForwardToDecoder(true));
-
   EXPECT_EQ(0, module1->SetSendingStatus(true));
 
   // Start basic RTP test.
 
   // Send an empty RTP packet.
   // Should fail since we have not registered the payload type.
   EXPECT_FALSE(module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96, 0, -1,
                                          nullptr, 0, nullptr, nullptr,
                                          nullptr));
 
@@ skipping 96 matching lines @@
     EXPECT_TRUE(module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
                                           timeStamp, -1, test, 4, nullptr,
                                           nullptr, nullptr));
     fake_clock.AdvanceTimeMilliseconds(20);
     module1->Process();
   }
 }
 
 }  // namespace
 }  // namespace webrtc