| Index: talk/app/webrtc/test/fakeaudiocapturemodule.h
|
| diff --git a/talk/app/webrtc/test/fakeaudiocapturemodule.h b/talk/app/webrtc/test/fakeaudiocapturemodule.h
|
| index 9f36ed80af45e87ba029d6da170bbc6dc018ae41..be2966e7d971882e20bc6117e53cecb0eff0647f 100644
|
| --- a/talk/app/webrtc/test/fakeaudiocapturemodule.h
|
| +++ b/talk/app/webrtc/test/fakeaudiocapturemodule.h
|
| @@ -53,7 +53,7 @@ class FakeAudioCaptureModule
|
| : public webrtc::AudioDeviceModule,
|
| public rtc::MessageHandler {
|
| public:
|
| - typedef uint16 Sample;
|
| + typedef uint16_t Sample;
|
|
|
| // The value for the following constants have been derived by running VoE
|
| // using a real ADM. The constants correspond to 10ms of mono audio at 44kHz.
|
| @@ -238,7 +238,7 @@ class FakeAudioCaptureModule
|
|
|
| // The time in milliseconds when Process() was last called or 0 if no call
|
| // has been made.
|
| - uint32 last_process_time_ms_;
|
| + uint32_t last_process_time_ms_;
|
|
|
| // Callback for playout and recording.
|
| webrtc::AudioTransport* audio_callback_;
|
| @@ -258,7 +258,7 @@ class FakeAudioCaptureModule
|
| // wall clock time the next frame should be generated and received. started_
|
| // ensures that next_frame_time_ can be initialized properly on first call.
|
| bool started_;
|
| - uint32 next_frame_time_;
|
| + uint32_t next_frame_time_;
|
|
|
| rtc::scoped_ptr<rtc::Thread> process_thread_;
|
|
|
|
|