Repo created

TMessagesProj/jni/voip/webrtc/pc/jitter_buffer_delay_unittest.cc

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+/*
+ *  Copyright 2019 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.
+ */
+
+#include "pc/jitter_buffer_delay.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+class JitterBufferDelayTest : public ::testing::Test {
+ public:
+  JitterBufferDelayTest() {}
+
+ protected:
+  JitterBufferDelay delay_;
+};
+
+TEST_F(JitterBufferDelayTest, Set) {
+  // Delay in seconds.
+  delay_.Set(3.0);
+  EXPECT_EQ(delay_.GetMs(), 3000);
+}
+
+TEST_F(JitterBufferDelayTest, DefaultValue) {
+  EXPECT_EQ(delay_.GetMs(), 0);  // Default value is 0ms.
+}
+
+TEST_F(JitterBufferDelayTest, Clamping) {
+  // In current Jitter Buffer implementation (Audio or Video) maximum supported
+  // value is 10000 milliseconds.
+  delay_.Set(10.5);
+  EXPECT_EQ(delay_.GetMs(), 10000);
+
+  // Test int overflow.
+  delay_.Set(21474836470.0);
+  EXPECT_EQ(delay_.GetMs(), 10000);
+
+  delay_.Set(-21474836470.0);
+  EXPECT_EQ(delay_.GetMs(), 0);
+
+  // Boundary value in seconds to milliseconds conversion.
+  delay_.Set(0.0009);
+  EXPECT_EQ(delay_.GetMs(), 0);
+
+  delay_.Set(-2.0);
+  EXPECT_EQ(delay_.GetMs(), 0);
+}
+
+}  // namespace webrtc