Index: webrtc/voice_engine/transmit_mixer.h |
diff --git a/webrtc/voice_engine/transmit_mixer.h b/webrtc/voice_engine/transmit_mixer.h |
index 483af0518ab5d5e7c2092e41557bae1f27d95f22..f697d3adedb75496af563be6417a4ddfe6353629 100644 |
--- a/webrtc/voice_engine/transmit_mixer.h |
+++ b/webrtc/voice_engine/transmit_mixer.h |
@@ -11,6 +11,8 @@ |
#ifndef WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H |
#define WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H |
+#include <memory> |
+ |
#include "webrtc/base/criticalsection.h" |
#include "webrtc/common_audio/resampler/include/push_resampler.h" |
#include "webrtc/common_types.h" |
@@ -196,9 +198,9 @@ private: |
MonitorModule _monitorModule; |
AudioFrame _audioFrame; |
PushResampler<int16_t> resampler_; // ADM sample rate -> mixing rate |
- FilePlayer* _filePlayerPtr; |
- FileRecorder* _fileRecorderPtr; |
- FileRecorder* _fileCallRecorderPtr; |
+ std::unique_ptr<FilePlayer> file_player_; |
+ std::unique_ptr<FileRecorder> file_recorder_; |
+ std::unique_ptr<FileRecorder> file_call_recorder_; |
int _filePlayerId; |
int _fileRecorderId; |
int _fileCallRecorderId; |