Removes unused APIs from the ADM (part II)

webrtc/modules/audio_device/audio_device_buffer.cc

Index: webrtc/modules/audio_device/audio_device_buffer.cc
diff --git a/webrtc/modules/audio_device/audio_device_buffer.cc b/webrtc/modules/audio_device/audio_device_buffer.cc
index 905c69681fa90f632e3335701757a6ebed7276f0..79a0ebc3b69cbecb0a837b64e9f4495c0e7deaef 100644
--- a/webrtc/modules/audio_device/audio_device_buffer.cc
+++ b/webrtc/modules/audio_device/audio_device_buffer.cc
@@ -280,28 +280,6 @@ void AudioDeviceBuffer::SetVQEData(int play_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_);