ACM: Removing runtime APIs related to playout mode

webrtc/modules/audio_coding/main/acm2/acm_receiver.cc

 /*
  *  Copyright (c) 2013 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 195 matching lines @@
 
 int AcmReceiver::LeastRequiredDelayMs() const {
   return neteq_->LeastRequiredDelayMs();
 }
 
 int AcmReceiver::current_sample_rate_hz() const {
   CriticalSectionScoped lock(crit_sect_.get());
   return current_sample_rate_hz_;
 }
 
-// TODO(turajs): use one set of enumerators, e.g. the one defined in
-// common_types.h
-// TODO(henrik.lundin): This method is not used any longer. The call hierarchy
-// stops in voe::Channel::SetNetEQPlayoutMode(). Remove it.
-void AcmReceiver::SetPlayoutMode(AudioPlayoutMode mode) {
-  enum NetEqPlayoutMode playout_mode = kPlayoutOn;
-  switch (mode) {
-    case voice:
-      playout_mode = kPlayoutOn;
-      break;
-    case fax:  // No change to background noise mode.
-      playout_mode = kPlayoutFax;
-      break;
-    case streaming:
-      playout_mode = kPlayoutStreaming;
-      break;
-    case off:
-      playout_mode = kPlayoutOff;
-      break;
-  }
-  neteq_->SetPlayoutMode(playout_mode);
-}
-
-AudioPlayoutMode AcmReceiver::PlayoutMode() const {
-  AudioPlayoutMode acm_mode = voice;
-  NetEqPlayoutMode mode = neteq_->PlayoutMode();
-  switch (mode) {
-    case kPlayoutOn:
-      acm_mode = voice;
-      break;
-    case kPlayoutOff:
-      acm_mode = off;
-      break;
-    case kPlayoutFax:
-      acm_mode = fax;
-      break;
-    case kPlayoutStreaming:
-      acm_mode = streaming;
-      break;
-    default:
-      assert(false);
-  }
-  return acm_mode;
-}
-
 int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header,
                               const uint8_t* incoming_payload,
                               size_t length_payload) {
   uint32_t receive_timestamp = 0;
   InitialDelayManager::PacketType packet_type =
       InitialDelayManager::kUndefinedPacket;
   bool new_codec = false;
   const RTPHeader* header = &rtp_header.header;  // Just a shorthand.
 
   {
@@ skipping 551 matching lines @@
 
 void AcmReceiver::GetDecodingCallStatistics(
     AudioDecodingCallStats* stats) const {
   CriticalSectionScoped lock(crit_sect_.get());
   *stats = call_stats_.GetDecodingStatistics();
 }
 
 }  // namespace acm2
 
 }  // namespace webrtc