Index: webrtc/modules/audio_device/audio_device_buffer.h |
diff --git a/webrtc/modules/audio_device/audio_device_buffer.h b/webrtc/modules/audio_device/audio_device_buffer.h |
index 2a5a84b22fa60ad8066e629538c67591e963054a..a9163ea0a611570792a3863c40220fe63ee68ec4 100644 |
--- a/webrtc/modules/audio_device/audio_device_buffer.h |
+++ b/webrtc/modules/audio_device/audio_device_buffer.h |
@@ -108,14 +108,6 @@ class AudioDeviceBuffer { |
virtual int32_t RequestPlayoutData(size_t samples_per_channel); |
virtual int32_t GetPlayoutData(void* audio_buffer); |
- // TODO(henrika): these methods should not be used and does not contain any |
- // valid implementation. Investigate the possibility to either remove them |
- // or add a proper implementation if needed. |
- int32_t StartInputFileRecording(const char fileName[kAdmMaxFileNameSize]); |
- int32_t StopInputFileRecording(); |
- int32_t StartOutputFileRecording(const char fileName[kAdmMaxFileNameSize]); |
- int32_t StopOutputFileRecording(); |
- |
int32_t SetTypingStatus(bool typing_status); |
// Called on iOS where the native audio layer can be interrupted by other |