ACM: Remove functions related to DTMF

webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.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 803 matching lines @@
   {
     CriticalSectionScoped lock(acm_crit_sect_.get());
     // Initialize receiver, if it is not initialized. Otherwise, initial delay
     // is reset upon initialization of the receiver.
     if (!receiver_initialized_)
       InitializeReceiverSafe();
   }
   return receiver_.SetInitialDelay(delay_ms);
 }
 
-int AudioCodingModuleImpl::SetDtmfPlayoutStatus(bool enable) {
-  return 0;
-}
-
-bool AudioCodingModuleImpl::DtmfPlayoutStatus() const {
-  return true;
-}
-
 int AudioCodingModuleImpl::EnableNack(size_t max_nack_list_size) {
   return receiver_.EnableNack(max_nack_list_size);
 }
 
 void AudioCodingModuleImpl::DisableNack() {
   receiver_.DisableNack();
 }
 
 std::vector<uint16_t> AudioCodingModuleImpl::GetNackList(
     int64_t round_trip_time_ms) const {
   return receiver_.GetNackList(round_trip_time_ms);
 }
 
 int AudioCodingModuleImpl::LeastRequiredDelayMs() const {
   return receiver_.LeastRequiredDelayMs();
 }
 
 void AudioCodingModuleImpl::GetDecodingCallStatistics(
       AudioDecodingCallStats* call_stats) const {
   receiver_.GetDecodingCallStatistics(call_stats);
 }
 
 }  // namespace acm2
 
 AudioCodingImpl::AudioCodingImpl(const Config& config) {
   AudioCodingModule::Config config_old = config.ToOldConfig();
   acm_old_.reset(new acm2::AudioCodingModuleImpl(config_old));
   acm_old_->RegisterTransportCallback(config.transport);
   acm_old_->RegisterVADCallback(config.vad_callback);
-  acm_old_->SetDtmfPlayoutStatus(config.play_dtmf);
   if (config.initial_playout_delay_ms > 0) {
     acm_old_->SetInitialPlayoutDelay(config.initial_playout_delay_ms);
   }
   playout_frequency_hz_ = config.playout_frequency_hz;
 }
 
 AudioCodingImpl::~AudioCodingImpl() = default;
 
 bool AudioCodingImpl::RegisterSendCodec(AudioEncoder* send_codec) {
   FATAL() << "Not implemented yet.";
@@ skipping 263 matching lines @@
       *sample_rate_hz = 8000;
       *channels = 1;
       break;
     default:
       FATAL() << "Codec type " << codec_type << " not supported.";
   }
   return true;
 }
 
 }  // namespace webrtc