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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 Normal class.
#include "webrtc/modules/audio_coding/neteq/normal.h"
#include <vector>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
#include "webrtc/modules/audio_coding/neteq/background_noise.h"
#include "webrtc/modules/audio_coding/neteq/expand.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_expand.h"
#include "webrtc/modules/audio_coding/neteq/random_vector.h"
#include "webrtc/modules/audio_coding/neteq/statistics_calculator.h"
#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
using ::testing::_;
namespace webrtc {
TEST(Normal, CreateAndDestroy) {
MockDecoderDatabase db;
int fs = 8000;
size_t channels = 1;
BackgroundNoise bgn(channels);
SyncBuffer sync_buffer(1, 1000);
RandomVector random_vector;
StatisticsCalculator statistics;
Expand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs, channels);
Normal normal(fs, &db, bgn, &expand);
EXPECT_CALL(db, Die()); // Called when |db| goes out of scope.
}
TEST(Normal, AvoidDivideByZero) {
WebRtcSpl_Init();
MockDecoderDatabase db;
int fs = 8000;
size_t channels = 1;
BackgroundNoise bgn(channels);
SyncBuffer sync_buffer(1, 1000);
RandomVector random_vector;
StatisticsCalculator statistics;
MockExpand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs,
channels);
Normal normal(fs, &db, bgn, &expand);
int16_t input[1000] = {0};
rtc::scoped_ptr<int16_t[]> mute_factor_array(new int16_t[channels]);
for (size_t i = 0; i < channels; ++i) {
mute_factor_array[i] = 16384;
}
AudioMultiVector output(channels);
// Zero input length.
EXPECT_EQ(
0,
normal.Process(input, 0, kModeExpand, mute_factor_array.get(), &output));
EXPECT_EQ(0u, output.Size());
// Try to make energy_length >> scaling = 0;
EXPECT_CALL(expand, SetParametersForNormalAfterExpand());
EXPECT_CALL(expand, Process(_));
EXPECT_CALL(expand, Reset());
// If input_size_samples < 64, then energy_length in Normal::Process() will
// be equal to input_size_samples. Since the input is all zeros, decoded_max
// will be zero, and scaling will be >= 6. Thus, energy_length >> scaling = 0,
// and using this as a denominator would lead to problems.
int input_size_samples = 63;
EXPECT_EQ(input_size_samples,
normal.Process(input,
input_size_samples,
kModeExpand,
mute_factor_array.get(),
&output));
EXPECT_CALL(db, Die()); // Called when |db| goes out of scope.
EXPECT_CALL(expand, Die()); // Called when |expand| goes out of scope.
}
TEST(Normal, InputLengthAndChannelsDoNotMatch) {
WebRtcSpl_Init();
MockDecoderDatabase db;
int fs = 8000;
size_t channels = 2;
BackgroundNoise bgn(channels);
SyncBuffer sync_buffer(channels, 1000);
RandomVector random_vector;
StatisticsCalculator statistics;
MockExpand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs,
channels);
Normal normal(fs, &db, bgn, &expand);
int16_t input[1000] = {0};
rtc::scoped_ptr<int16_t[]> mute_factor_array(new int16_t[channels]);
for (size_t i = 0; i < channels; ++i) {
mute_factor_array[i] = 16384;
}
AudioMultiVector output(channels);
// Let the number of samples be one sample less than 80 samples per channel.
size_t input_len = 80 * channels - 1;
EXPECT_EQ(
0,
normal.Process(
input, input_len, kModeExpand, mute_factor_array.get(), &output));
EXPECT_EQ(0u, output.Size());
EXPECT_CALL(db, Die()); // Called when |db| goes out of scope.
EXPECT_CALL(expand, Die()); // Called when |expand| goes out of scope.
}
// TODO(hlundin): Write more tests.
} // namespace webrtc
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