Index: webrtc/modules/audio_processing/test/performance_timer.h |
diff --git a/webrtc/modules/audio_processing/test/performance_timer.h b/webrtc/modules/audio_processing/test/performance_timer.h |
new file mode 100644 |
index 0000000000000000000000000000000000000000..5921fae07011a16aa8d0a83be8ed92fbb78a4c76 |
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+++ b/webrtc/modules/audio_processing/test/performance_timer.h |
@@ -0,0 +1,42 @@ |
+/* |
+ * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. |
+ * |
+ * Use of this source code is governed by a BSD-style license |
+ * that can be found in the LICENSE file in the root of the source |
+ * tree. An additional intellectual property rights grant can be found |
+ * in the file PATENTS. All contributing project authors may |
+ * be found in the AUTHORS file in the root of the source tree. |
+ */ |
+ |
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_PERFORMANCE_TIMER_H_ |
+#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_PERFORMANCE_TIMER_H_ |
+ |
+#include <vector> |
+ |
+#include "webrtc/base/optional.h" |
+#include "webrtc/system_wrappers/include/clock.h" |
+ |
+namespace webrtc { |
+namespace test { |
+ |
+class PerformanceTimer { |
+ public: |
+ explicit PerformanceTimer(int num_frames_to_process); |
+ ~PerformanceTimer(); |
+ |
+ void StartTimer(); |
+ void StopTimer(); |
+ |
+ double GetDurationAverage() const; |
+ double GetDurationStandardDeviation() const; |
+ |
+ private: |
+ webrtc::Clock* clock_; |
+ rtc::Optional<int64_t> start_timestamp_us_; |
+ std::vector<int64_t> timestamps_us_; |
+}; |
+ |
+} // namespace test |
+} // namespace webrtc |
+ |
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_PERFORMANCE_TIMER_H_ |