ACM: Remove functions related to DTMF

webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.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 12 matching lines @@
 #include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
 #include "webrtc/modules/audio_coding/main/acm2/codec_manager.h"
 
 namespace webrtc {
 
 class CriticalSectionWrapper;
 class AudioCodingImpl;
 
 namespace acm2 {
 
-class ACMDTMFDetection;
-
 class AudioCodingModuleImpl final : public AudioCodingModule {
  public:
   friend webrtc::AudioCodingImpl;
 
   explicit AudioCodingModuleImpl(const AudioCodingModule::Config& config);
   ~AudioCodingModuleImpl() override;
 
   /////////////////////////////////////////
   //   Sender
   //
@@ skipping 104 matching lines @@
   // Maximum playout delay.
   int SetMaximumPlayoutDelay(int time_ms) override;
 
   // Smallest latency NetEq will maintain.
   int LeastRequiredDelayMs() const override;
 
   // Impose an initial delay on playout. ACM plays silence until |delay_ms|
   // audio is accumulated in NetEq buffer, then starts decoding payloads.
   int SetInitialPlayoutDelay(int delay_ms) override;
 
-  // TODO(turajs): DTMF playout is always activated in NetEq these APIs should
-  // be removed, as well as all VoE related APIs and methods.
-  //
-  // Configure Dtmf playout status i.e on/off playout the incoming outband Dtmf
-  // tone.
-  int SetDtmfPlayoutStatus(bool enable) override;
-
-  // Get Dtmf playout status.
-  bool DtmfPlayoutStatus() const override;
-
   // Set playout mode voice, fax.
   int SetPlayoutMode(AudioPlayoutMode mode) override;
 
   // Get playout mode voice, fax.
   AudioPlayoutMode PlayoutMode() const override;
 
   // Get playout timestamp.
   int PlayoutTimestamp(uint32_t* timestamp) override;
 
   // Get 10 milliseconds of raw audio data to play out, and
@@ skipping 190 matching lines @@
   int playout_frequency_hz_;
   // TODO(henrik.lundin): All members below this line are temporary and should
   // be removed after refactoring is completed.
   rtc::scoped_ptr<acm2::AudioCodingModuleImpl> acm_old_;
   CodecInst current_send_codec_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_