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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
11 // Unit tests for Normal class. | 11 // Unit tests for Normal class. |
12 | 12 |
13 #include "webrtc/modules/audio_coding/neteq/normal.h" | 13 #include "webrtc/modules/audio_coding/neteq/normal.h" |
14 | 14 |
15 #include <vector> | 15 #include <vector> |
16 | 16 |
17 #include "testing/gtest/include/gtest/gtest.h" | 17 #include "testing/gtest/include/gtest/gtest.h" |
18 #include "webrtc/base/scoped_ptr.h" | 18 #include "webrtc/base/scoped_ptr.h" |
19 #include "webrtc/common_audio/signal_processing/include/signal_processing_librar
y.h" | 19 #include "webrtc/common_audio/signal_processing/include/signal_processing_librar
y.h" |
20 #include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h" | 20 #include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h" |
21 #include "webrtc/modules/audio_coding/neteq/background_noise.h" | 21 #include "webrtc/modules/audio_coding/neteq/background_noise.h" |
22 #include "webrtc/modules/audio_coding/neteq/expand.h" | 22 #include "webrtc/modules/audio_coding/neteq/expand.h" |
23 #include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h" | 23 #include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h" |
24 #include "webrtc/modules/audio_coding/neteq/mock/mock_expand.h" | 24 #include "webrtc/modules/audio_coding/neteq/mock/mock_expand.h" |
25 #include "webrtc/modules/audio_coding/neteq/random_vector.h" | 25 #include "webrtc/modules/audio_coding/neteq/random_vector.h" |
| 26 #include "webrtc/modules/audio_coding/neteq/statistics_calculator.h" |
26 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h" | 27 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h" |
27 | 28 |
28 using ::testing::_; | 29 using ::testing::_; |
29 | 30 |
30 namespace webrtc { | 31 namespace webrtc { |
31 | 32 |
32 TEST(Normal, CreateAndDestroy) { | 33 TEST(Normal, CreateAndDestroy) { |
33 MockDecoderDatabase db; | 34 MockDecoderDatabase db; |
34 int fs = 8000; | 35 int fs = 8000; |
35 size_t channels = 1; | 36 size_t channels = 1; |
36 BackgroundNoise bgn(channels); | 37 BackgroundNoise bgn(channels); |
37 SyncBuffer sync_buffer(1, 1000); | 38 SyncBuffer sync_buffer(1, 1000); |
38 RandomVector random_vector; | 39 RandomVector random_vector; |
39 Expand expand(&bgn, &sync_buffer, &random_vector, fs, channels); | 40 StatisticsCalculator statistics; |
| 41 Expand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs, channels); |
40 Normal normal(fs, &db, bgn, &expand); | 42 Normal normal(fs, &db, bgn, &expand); |
41 EXPECT_CALL(db, Die()); // Called when |db| goes out of scope. | 43 EXPECT_CALL(db, Die()); // Called when |db| goes out of scope. |
42 } | 44 } |
43 | 45 |
44 TEST(Normal, AvoidDivideByZero) { | 46 TEST(Normal, AvoidDivideByZero) { |
45 WebRtcSpl_Init(); | 47 WebRtcSpl_Init(); |
46 MockDecoderDatabase db; | 48 MockDecoderDatabase db; |
47 int fs = 8000; | 49 int fs = 8000; |
48 size_t channels = 1; | 50 size_t channels = 1; |
49 BackgroundNoise bgn(channels); | 51 BackgroundNoise bgn(channels); |
50 SyncBuffer sync_buffer(1, 1000); | 52 SyncBuffer sync_buffer(1, 1000); |
51 RandomVector random_vector; | 53 RandomVector random_vector; |
52 MockExpand expand(&bgn, &sync_buffer, &random_vector, fs, channels); | 54 StatisticsCalculator statistics; |
| 55 MockExpand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs, |
| 56 channels); |
53 Normal normal(fs, &db, bgn, &expand); | 57 Normal normal(fs, &db, bgn, &expand); |
54 | 58 |
55 int16_t input[1000] = {0}; | 59 int16_t input[1000] = {0}; |
56 rtc::scoped_ptr<int16_t[]> mute_factor_array(new int16_t[channels]); | 60 rtc::scoped_ptr<int16_t[]> mute_factor_array(new int16_t[channels]); |
57 for (size_t i = 0; i < channels; ++i) { | 61 for (size_t i = 0; i < channels; ++i) { |
58 mute_factor_array[i] = 16384; | 62 mute_factor_array[i] = 16384; |
59 } | 63 } |
60 AudioMultiVector output(channels); | 64 AudioMultiVector output(channels); |
61 | 65 |
62 // Zero input length. | 66 // Zero input length. |
(...skipping 23 matching lines...) Expand all Loading... |
86 } | 90 } |
87 | 91 |
88 TEST(Normal, InputLengthAndChannelsDoNotMatch) { | 92 TEST(Normal, InputLengthAndChannelsDoNotMatch) { |
89 WebRtcSpl_Init(); | 93 WebRtcSpl_Init(); |
90 MockDecoderDatabase db; | 94 MockDecoderDatabase db; |
91 int fs = 8000; | 95 int fs = 8000; |
92 size_t channels = 2; | 96 size_t channels = 2; |
93 BackgroundNoise bgn(channels); | 97 BackgroundNoise bgn(channels); |
94 SyncBuffer sync_buffer(channels, 1000); | 98 SyncBuffer sync_buffer(channels, 1000); |
95 RandomVector random_vector; | 99 RandomVector random_vector; |
96 MockExpand expand(&bgn, &sync_buffer, &random_vector, fs, channels); | 100 StatisticsCalculator statistics; |
| 101 MockExpand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs, |
| 102 channels); |
97 Normal normal(fs, &db, bgn, &expand); | 103 Normal normal(fs, &db, bgn, &expand); |
98 | 104 |
99 int16_t input[1000] = {0}; | 105 int16_t input[1000] = {0}; |
100 rtc::scoped_ptr<int16_t[]> mute_factor_array(new int16_t[channels]); | 106 rtc::scoped_ptr<int16_t[]> mute_factor_array(new int16_t[channels]); |
101 for (size_t i = 0; i < channels; ++i) { | 107 for (size_t i = 0; i < channels; ++i) { |
102 mute_factor_array[i] = 16384; | 108 mute_factor_array[i] = 16384; |
103 } | 109 } |
104 AudioMultiVector output(channels); | 110 AudioMultiVector output(channels); |
105 | 111 |
106 // Let the number of samples be one sample less than 80 samples per channel. | 112 // Let the number of samples be one sample less than 80 samples per channel. |
107 size_t input_len = 80 * channels - 1; | 113 size_t input_len = 80 * channels - 1; |
108 EXPECT_EQ( | 114 EXPECT_EQ( |
109 0, | 115 0, |
110 normal.Process( | 116 normal.Process( |
111 input, input_len, kModeExpand, mute_factor_array.get(), &output)); | 117 input, input_len, kModeExpand, mute_factor_array.get(), &output)); |
112 EXPECT_EQ(0u, output.Size()); | 118 EXPECT_EQ(0u, output.Size()); |
113 | 119 |
114 EXPECT_CALL(db, Die()); // Called when |db| goes out of scope. | 120 EXPECT_CALL(db, Die()); // Called when |db| goes out of scope. |
115 EXPECT_CALL(expand, Die()); // Called when |expand| goes out of scope. | 121 EXPECT_CALL(expand, Die()); // Called when |expand| goes out of scope. |
116 } | 122 } |
117 | 123 |
118 // TODO(hlundin): Write more tests. | 124 // TODO(hlundin): Write more tests. |
119 | 125 |
120 } // namespace webrtc | 126 } // namespace webrtc |
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