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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 <stdint.h>
#include "absl/types/optional.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
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