Removes unused APIs from the ADM (part II)

webrtc/modules/audio_device/audio_device_buffer.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 262 matching lines @@
 
 void AudioDeviceBuffer::SetVQEData(int play_delay_ms,
                                    int rec_delay_ms,
                                    int clock_drift) {
   RTC_DCHECK_RUN_ON(&recording_thread_checker_);
   play_delay_ms_ = play_delay_ms;
   rec_delay_ms_ = rec_delay_ms;
   clock_drift_ = clock_drift;
 }
 
-int32_t AudioDeviceBuffer::StartInputFileRecording(
-    const char fileName[kAdmMaxFileNameSize]) {
-  LOG(LS_WARNING) << "Not implemented";
-  return 0;
-}
-
-int32_t AudioDeviceBuffer::StopInputFileRecording() {
-  LOG(LS_WARNING) << "Not implemented";
-  return 0;
-}
-
-int32_t AudioDeviceBuffer::StartOutputFileRecording(
-    const char fileName[kAdmMaxFileNameSize]) {
-  LOG(LS_WARNING) << "Not implemented";
-  return 0;
-}
-
-int32_t AudioDeviceBuffer::StopOutputFileRecording() {
-  LOG(LS_WARNING) << "Not implemented";
-  return 0;
-}
-
 int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audio_buffer,
                                              size_t samples_per_channel) {
   RTC_DCHECK_RUN_ON(&recording_thread_checker_);
   // Copy the complete input buffer to the local buffer.
   const size_t old_size = rec_buffer_.size();
   rec_buffer_.SetData(static_cast<const int16_t*>(audio_buffer),
                       rec_channels_ * samples_per_channel);
   // Keep track of the size of the recording buffer. Only updated when the
   // size changes, which is a rare event.
   if (old_size != rec_buffer_.size()) {
@@ skipping 212 matching lines @@
   RTC_DCHECK_RUN_ON(&playout_thread_checker_);
   rtc::CritScope cs(&lock_);
   ++stats_.play_callbacks;
   stats_.play_samples += samples_per_channel;
   if (max_abs > stats_.max_play_level) {
     stats_.max_play_level = max_abs;
   }
 }
 
 }  // namespace webrtc