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/*
* Copyright (c) 2012 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.
*/
// Unit tests for BufferLevelFilter class.
#include "modules/audio_coding/neteq/buffer_level_filter.h"
#include <math.h> // Access to pow function.
#include "rtc_base/strings/string_builder.h"
#include "test/gtest.h"
namespace webrtc {
TEST(BufferLevelFilter, CreateAndDestroy) {
BufferLevelFilter* filter = new BufferLevelFilter();
EXPECT_EQ(0, filter->filtered_current_level());
delete filter;
}
TEST(BufferLevelFilter, ConvergenceTest) {
BufferLevelFilter filter;
for (int times = 10; times <= 50; times += 10) {
for (int value = 100; value <= 200; value += 10) {
filter.Reset();
filter.SetTargetBufferLevel(1); // Makes filter coefficient 0.98.
rtc::StringBuilder ss;
ss << "times = " << times << ", value = " << value;
SCOPED_TRACE(ss.str()); // Print out the parameter values on failure.
for (int i = 0; i < times; ++i) {
filter.Update(value, 0 /* time_stretched_samples */);
}
// Expect the filtered value to be (theoretically)
// (1 - 0.980 ^ |times|) * |value|.
double expected_value_double = (1 - pow(0.98, times)) * value;
int expected_value = static_cast<int>(expected_value_double);
// filtered_current_level() returns the value in Q8.
// The actual value may differ slightly from the expected value due to
// intermediate-stage rounding errors in the filter implementation.
// This is why we have to use EXPECT_NEAR with a tolerance of +/-1.
EXPECT_NEAR(expected_value, filter.filtered_current_level(), 1);
}
}
}
// Verify that target buffer level impacts on the filter convergence.
TEST(BufferLevelFilter, FilterFactor) {
BufferLevelFilter filter;
// Update 10 times with value 100.
const int kTimes = 10;
const int kValue = 100;
filter.SetTargetBufferLevel(3); // Makes filter coefficient 0.984.
for (int i = 0; i < kTimes; ++i) {
filter.Update(kValue, 0 /* time_stretched_samples */);
}
// Expect the filtered value to be
// (1 - 0.984 ^ |kTimes|) * |kValue|.
int expected_value = 15;
EXPECT_EQ(expected_value, filter.filtered_current_level());
filter.Reset();
filter.SetTargetBufferLevel(7); // Makes filter coefficient 0.988.
for (int i = 0; i < kTimes; ++i) {
filter.Update(kValue, 0 /* time_stretched_samples */);
}
// Expect the filtered value to be
// (1 - 0.988 ^ |kTimes|) * |kValue|.
expected_value = 11;
EXPECT_EQ(expected_value, filter.filtered_current_level());
filter.Reset();
filter.SetTargetBufferLevel(8); // Makes filter coefficient 0.992.
for (int i = 0; i < kTimes; ++i) {
filter.Update(kValue, 0 /* time_stretched_samples */);
}
// Expect the filtered value to be
// (1 - 0.992 ^ |kTimes|) * |kValue|.
expected_value = 8;
EXPECT_EQ(expected_value, filter.filtered_current_level());
}
TEST(BufferLevelFilter, TimeStretchedSamples) {
BufferLevelFilter filter;
filter.SetTargetBufferLevel(1); // Makes filter coefficient 0.98.
// Update 10 times with value 100.
const int kTimes = 10;
const int kValue = 100;
const int kTimeStretchedSamples = 3;
for (int i = 0; i < kTimes; ++i) {
filter.Update(kValue, 0);
}
// Expect the filtered value to be
// (1 - 0.98 ^ |kTimes|) * |kValue|.
const int kExpectedValue = 18;
EXPECT_EQ(kExpectedValue, filter.filtered_current_level());
// Update filter again, now with non-zero value for packet length.
// Set the current filtered value to be the input, in order to isolate the
// impact of |kTimeStretchedSamples|.
filter.Update(filter.filtered_current_level(), kTimeStretchedSamples);
EXPECT_EQ(kExpectedValue - kTimeStretchedSamples,
filter.filtered_current_level());
// Try negative value and verify that we come back to the previous result.
filter.Update(filter.filtered_current_level(), -kTimeStretchedSamples);
EXPECT_EQ(kExpectedValue, filter.filtered_current_level());
}
} // namespace webrtc