1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3 *
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
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 // Unit tests for Expand class.
12
13 #include "modules/audio_coding/neteq/expand.h"
14
15 #include "common_audio/signal_processing/include/signal_processing_library.h"
16 #include "modules/audio_coding/neteq/background_noise.h"
17 #include "modules/audio_coding/neteq/random_vector.h"
18 #include "modules/audio_coding/neteq/statistics_calculator.h"
19 #include "modules/audio_coding/neteq/sync_buffer.h"
20 #include "modules/audio_coding/neteq/tools/resample_input_audio_file.h"
21 #include "rtc_base/numerics/safe_conversions.h"
22 #include "test/gtest.h"
23 #include "test/testsupport/file_utils.h"
24
25 namespace webrtc {
26
TEST(Expand,CreateAndDestroy)27 TEST(Expand, CreateAndDestroy) {
28 int fs = 8000;
29 size_t channels = 1;
30 BackgroundNoise bgn(channels);
31 SyncBuffer sync_buffer(1, 1000);
32 RandomVector random_vector;
33 StatisticsCalculator statistics;
34 Expand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs, channels);
35 }
36
TEST(Expand,CreateUsingFactory)37 TEST(Expand, CreateUsingFactory) {
38 int fs = 8000;
39 size_t channels = 1;
40 BackgroundNoise bgn(channels);
41 SyncBuffer sync_buffer(1, 1000);
42 RandomVector random_vector;
43 StatisticsCalculator statistics;
44 ExpandFactory expand_factory;
45 Expand* expand = expand_factory.Create(&bgn, &sync_buffer, &random_vector,
46 &statistics, fs, channels);
47 EXPECT_TRUE(expand != NULL);
48 delete expand;
49 }
50
51 namespace {
52 class FakeStatisticsCalculator : public StatisticsCalculator {
53 public:
LogDelayedPacketOutageEvent(int num_samples,int fs_hz)54 void LogDelayedPacketOutageEvent(int num_samples, int fs_hz) override {
55 last_outage_duration_samples_ = num_samples;
56 }
57
last_outage_duration_samples() const58 int last_outage_duration_samples() const {
59 return last_outage_duration_samples_;
60 }
61
62 private:
63 int last_outage_duration_samples_ = 0;
64 };
65
66 // This is the same size that is given to the SyncBuffer object in NetEq.
67 const size_t kNetEqSyncBufferLengthMs = 720;
68 } // namespace
69
70 class ExpandTest : public ::testing::Test {
71 protected:
ExpandTest()72 ExpandTest()
73 : input_file_(test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
74 32000),
75 test_sample_rate_hz_(32000),
76 num_channels_(1),
77 background_noise_(num_channels_),
78 sync_buffer_(num_channels_,
79 kNetEqSyncBufferLengthMs * test_sample_rate_hz_ / 1000),
80 expand_(&background_noise_,
81 &sync_buffer_,
82 &random_vector_,
83 &statistics_,
84 test_sample_rate_hz_,
85 num_channels_) {
86 input_file_.set_output_rate_hz(test_sample_rate_hz_);
87 }
88
SetUp()89 void SetUp() override {
90 // Fast-forward the input file until there is speech (about 1.1 second into
91 // the file).
92 const int speech_start_samples =
93 static_cast<int>(test_sample_rate_hz_ * 1.1f);
94 ASSERT_TRUE(input_file_.Seek(speech_start_samples));
95
96 // Pre-load the sync buffer with speech data.
97 std::unique_ptr<int16_t[]> temp(new int16_t[sync_buffer_.Size()]);
98 ASSERT_TRUE(input_file_.Read(sync_buffer_.Size(), temp.get()));
99 sync_buffer_.Channel(0).OverwriteAt(temp.get(), sync_buffer_.Size(), 0);
100 ASSERT_EQ(1u, num_channels_) << "Fix: Must populate all channels.";
101 }
102
103 test::ResampleInputAudioFile input_file_;
104 int test_sample_rate_hz_;
105 size_t num_channels_;
106 BackgroundNoise background_noise_;
107 SyncBuffer sync_buffer_;
108 RandomVector random_vector_;
109 FakeStatisticsCalculator statistics_;
110 Expand expand_;
111 };
112
113 // This test calls the expand object to produce concealment data a few times,
114 // and then ends by calling SetParametersForNormalAfterExpand. This simulates
115 // the situation where the packet next up for decoding was just delayed, not
116 // lost.
TEST_F(ExpandTest,DelayedPacketOutage)117 TEST_F(ExpandTest, DelayedPacketOutage) {
118 AudioMultiVector output(num_channels_);
119 size_t sum_output_len_samples = 0;
120 for (int i = 0; i < 10; ++i) {
121 EXPECT_EQ(0, expand_.Process(&output));
122 EXPECT_GT(output.Size(), 0u);
123 sum_output_len_samples += output.Size();
124 EXPECT_EQ(0, statistics_.last_outage_duration_samples());
125 }
126 expand_.SetParametersForNormalAfterExpand();
127 // Convert `sum_output_len_samples` to milliseconds.
128 EXPECT_EQ(rtc::checked_cast<int>(sum_output_len_samples),
129 statistics_.last_outage_duration_samples());
130 }
131
132 // This test is similar to DelayedPacketOutage, but ends by calling
133 // SetParametersForMergeAfterExpand. This simulates the situation where the
134 // packet next up for decoding was actually lost (or at least a later packet
135 // arrived before it).
TEST_F(ExpandTest,LostPacketOutage)136 TEST_F(ExpandTest, LostPacketOutage) {
137 AudioMultiVector output(num_channels_);
138 for (int i = 0; i < 10; ++i) {
139 EXPECT_EQ(0, expand_.Process(&output));
140 EXPECT_GT(output.Size(), 0u);
141 EXPECT_EQ(0, statistics_.last_outage_duration_samples());
142 }
143 expand_.SetParametersForMergeAfterExpand();
144 EXPECT_EQ(0, statistics_.last_outage_duration_samples());
145 }
146
147 // This test is similar to the DelayedPacketOutage test above, but with the
148 // difference that Expand::Reset() is called after 5 calls to Expand::Process().
149 // This should reset the statistics, and will in the end lead to an outage of
150 // 5 periods instead of 10.
TEST_F(ExpandTest,CheckOutageStatsAfterReset)151 TEST_F(ExpandTest, CheckOutageStatsAfterReset) {
152 AudioMultiVector output(num_channels_);
153 size_t sum_output_len_samples = 0;
154 for (int i = 0; i < 10; ++i) {
155 EXPECT_EQ(0, expand_.Process(&output));
156 EXPECT_GT(output.Size(), 0u);
157 sum_output_len_samples += output.Size();
158 if (i == 5) {
159 expand_.Reset();
160 sum_output_len_samples = 0;
161 }
162 EXPECT_EQ(0, statistics_.last_outage_duration_samples());
163 }
164 expand_.SetParametersForNormalAfterExpand();
165 // Convert `sum_output_len_samples` to milliseconds.
166 EXPECT_EQ(rtc::checked_cast<int>(sum_output_len_samples),
167 statistics_.last_outage_duration_samples());
168 }
169
170 namespace {
171 // Runs expand until Muted() returns true. Times out after 1000 calls.
ExpandUntilMuted(size_t num_channels,Expand * expand)172 void ExpandUntilMuted(size_t num_channels, Expand* expand) {
173 EXPECT_FALSE(expand->Muted()) << "Instance is muted from the start";
174 AudioMultiVector output(num_channels);
175 int num_calls = 0;
176 while (!expand->Muted()) {
177 ASSERT_LT(num_calls++, 1000) << "Test timed out";
178 EXPECT_EQ(0, expand->Process(&output));
179 }
180 }
181 } // namespace
182
183 // Verifies that Muted() returns true after a long expand period. Also verifies
184 // that Muted() is reset to false after calling Reset(),
185 // SetParametersForMergeAfterExpand() and SetParametersForNormalAfterExpand().
TEST_F(ExpandTest,Muted)186 TEST_F(ExpandTest, Muted) {
187 ExpandUntilMuted(num_channels_, &expand_);
188 expand_.Reset();
189 EXPECT_FALSE(expand_.Muted()); // Should be back to unmuted.
190
191 ExpandUntilMuted(num_channels_, &expand_);
192 expand_.SetParametersForMergeAfterExpand();
193 EXPECT_FALSE(expand_.Muted()); // Should be back to unmuted.
194
195 expand_.Reset(); // Must reset in order to start a new expand period.
196 ExpandUntilMuted(num_channels_, &expand_);
197 expand_.SetParametersForNormalAfterExpand();
198 EXPECT_FALSE(expand_.Muted()); // Should be back to unmuted.
199 }
200
201 // TODO(hlundin): Write more tests.
202
203 } // namespace webrtc
204