| /* |
| * 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 |