Index: talk/app/webrtc/test/fakeaudiocapturemodule.h |
diff --git a/talk/app/webrtc/test/fakeaudiocapturemodule.h b/talk/app/webrtc/test/fakeaudiocapturemodule.h |
index 65a03c84045fb2c6a3e90be0104aefe3feb76c9f..4284b9ed51b89778e33aa0f2893be4a98fff9603 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. |
@@ -242,7 +242,7 @@ 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_; |
@@ -262,7 +262,7 @@ 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_; |