Index: webrtc/modules/audio_device/dummy/file_audio_device.h |
diff --git a/webrtc/modules/audio_device/dummy/file_audio_device.h b/webrtc/modules/audio_device/dummy/file_audio_device.h |
index a69b47ecc6fd62ed46eb846e0217c92c338f248c..ae4737cb9aaf5e23618a331df05b7d1788fc6458 100644 |
--- a/webrtc/modules/audio_device/dummy/file_audio_device.h |
+++ b/webrtc/modules/audio_device/dummy/file_audio_device.h |
@@ -8,17 +8,17 @@ |
* be found in the AUTHORS file in the root of the source tree. |
*/ |
-#ifndef WEBRTC_MODULES_AUDIO_DEVICE_DUMMY_FILE_AUDIO_DEVICE_H_ |
-#define WEBRTC_MODULES_AUDIO_DEVICE_DUMMY_FILE_AUDIO_DEVICE_H_ |
+#ifndef WEBRTC_AUDIO_DEVICE_FILE_AUDIO_DEVICE_H |
+#define WEBRTC_AUDIO_DEVICE_FILE_AUDIO_DEVICE_H |
#include <stdio.h> |
#include <memory> |
#include <string> |
-#include "webrtc/base/file.h" |
#include "webrtc/modules/audio_device/audio_device_generic.h" |
#include "webrtc/system_wrappers/include/critical_section_wrapper.h" |
+#include "webrtc/system_wrappers/include/file_wrapper.h" |
#include "webrtc/system_wrappers/include/clock.h" |
namespace rtc { |
@@ -41,6 +41,7 @@ |
FileAudioDevice(const int32_t id, |
const char* inputFilename, |
const char* outputFilename); |
+ virtual ~FileAudioDevice(); |
// Retrieve the currently utilized audio layer |
int32_t ActiveAudioLayer( |
@@ -190,8 +191,8 @@ |
uint64_t _lastCallPlayoutMillis; |
uint64_t _lastCallRecordMillis; |
- rtc::File _outputFile; |
- rtc::File _inputFile; |
+ FileWrapper& _outputFile; |
+ FileWrapper& _inputFile; |
std::string _outputFilename; |
std::string _inputFilename; |
@@ -200,4 +201,4 @@ |
} // namespace webrtc |
-#endif // WEBRTC_MODULES_AUDIO_DEVICE_DUMMY_FILE_AUDIO_DEVICE_H_ |
+#endif // WEBRTC_AUDIO_DEVICE_FILE_AUDIO_DEVICE_H |