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 "webrtc/modules/audio_coding/neteq/expand.h"
14 
15 #include "testing/gtest/include/gtest/gtest.h"
16 #include "webrtc/base/safe_conversions.h"
17 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
18 #include "webrtc/modules/audio_coding/neteq/background_noise.h"
19 #include "webrtc/modules/audio_coding/neteq/random_vector.h"
20 #include "webrtc/modules/audio_coding/neteq/statistics_calculator.h"
21 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
22 #include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
23 #include "webrtc/test/testsupport/fileutils.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 outage_duration_ms)54   void LogDelayedPacketOutageEvent(int outage_duration_ms) override {
55     last_outage_duration_ms_ = outage_duration_ms;
56   }
57 
last_outage_duration_ms() const58   int last_outage_duration_ms() const { return last_outage_duration_ms_; }
59 
60  private:
61   int last_outage_duration_ms_ = 0;
62 };
63 
64 // This is the same size that is given to the SyncBuffer object in NetEq.
65 const size_t kNetEqSyncBufferLengthMs = 720;
66 }  // namespace
67 
68 class ExpandTest : public ::testing::Test {
69  protected:
ExpandTest()70   ExpandTest()
71       : input_file_(test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
72                     32000),
73         test_sample_rate_hz_(32000),
74         num_channels_(1),
75         background_noise_(num_channels_),
76         sync_buffer_(num_channels_,
77                      kNetEqSyncBufferLengthMs * test_sample_rate_hz_ / 1000),
78         expand_(&background_noise_,
79                 &sync_buffer_,
80                 &random_vector_,
81                 &statistics_,
82                 test_sample_rate_hz_,
83                 num_channels_) {
84     WebRtcSpl_Init();
85     input_file_.set_output_rate_hz(test_sample_rate_hz_);
86   }
87 
SetUp()88   void SetUp() override {
89     // Fast-forward the input file until there is speech (about 1.1 second into
90     // the file).
91     const size_t speech_start_samples =
92         static_cast<size_t>(test_sample_rate_hz_ * 1.1f);
93     ASSERT_TRUE(input_file_.Seek(speech_start_samples));
94 
95     // Pre-load the sync buffer with speech data.
96     ASSERT_TRUE(
97         input_file_.Read(sync_buffer_.Size(), &sync_buffer_.Channel(0)[0]));
98     ASSERT_EQ(1u, num_channels_) << "Fix: Must populate all channels.";
99   }
100 
101   test::ResampleInputAudioFile input_file_;
102   int test_sample_rate_hz_;
103   size_t num_channels_;
104   BackgroundNoise background_noise_;
105   SyncBuffer sync_buffer_;
106   RandomVector random_vector_;
107   FakeStatisticsCalculator statistics_;
108   Expand expand_;
109 };
110 
111 // This test calls the expand object to produce concealment data a few times,
112 // and then ends by calling SetParametersForNormalAfterExpand. This simulates
113 // the situation where the packet next up for decoding was just delayed, not
114 // lost.
TEST_F(ExpandTest,DelayedPacketOutage)115 TEST_F(ExpandTest, DelayedPacketOutage) {
116   AudioMultiVector output(num_channels_);
117   size_t sum_output_len_samples = 0;
118   for (int i = 0; i < 10; ++i) {
119     EXPECT_EQ(0, expand_.Process(&output));
120     EXPECT_GT(output.Size(), 0u);
121     sum_output_len_samples += output.Size();
122     EXPECT_EQ(0, statistics_.last_outage_duration_ms());
123   }
124   expand_.SetParametersForNormalAfterExpand();
125   // Convert |sum_output_len_samples| to milliseconds.
126   EXPECT_EQ(rtc::checked_cast<int>(sum_output_len_samples /
127                                    (test_sample_rate_hz_ / 1000)),
128             statistics_.last_outage_duration_ms());
129 }
130 
131 // This test is similar to DelayedPacketOutage, but ends by calling
132 // SetParametersForMergeAfterExpand. This simulates the situation where the
133 // packet next up for decoding was actually lost (or at least a later packet
134 // arrived before it).
TEST_F(ExpandTest,LostPacketOutage)135 TEST_F(ExpandTest, LostPacketOutage) {
136   AudioMultiVector output(num_channels_);
137   size_t sum_output_len_samples = 0;
138   for (int i = 0; i < 10; ++i) {
139     EXPECT_EQ(0, expand_.Process(&output));
140     EXPECT_GT(output.Size(), 0u);
141     sum_output_len_samples += output.Size();
142     EXPECT_EQ(0, statistics_.last_outage_duration_ms());
143   }
144   expand_.SetParametersForMergeAfterExpand();
145   EXPECT_EQ(0, statistics_.last_outage_duration_ms());
146 }
147 
148 // This test is similar to the DelayedPacketOutage test above, but with the
149 // difference that Expand::Reset() is called after 5 calls to Expand::Process().
150 // This should reset the statistics, and will in the end lead to an outage of
151 // 5 periods instead of 10.
TEST_F(ExpandTest,CheckOutageStatsAfterReset)152 TEST_F(ExpandTest, CheckOutageStatsAfterReset) {
153   AudioMultiVector output(num_channels_);
154   size_t sum_output_len_samples = 0;
155   for (int i = 0; i < 10; ++i) {
156     EXPECT_EQ(0, expand_.Process(&output));
157     EXPECT_GT(output.Size(), 0u);
158     sum_output_len_samples += output.Size();
159     if (i == 5) {
160       expand_.Reset();
161       sum_output_len_samples = 0;
162     }
163     EXPECT_EQ(0, statistics_.last_outage_duration_ms());
164   }
165   expand_.SetParametersForNormalAfterExpand();
166   // Convert |sum_output_len_samples| to milliseconds.
167   EXPECT_EQ(rtc::checked_cast<int>(sum_output_len_samples /
168                                    (test_sample_rate_hz_ / 1000)),
169             statistics_.last_outage_duration_ms());
170 }
171 
172 // TODO(hlundin): Write more tests.
173 
174 }  // namespace webrtc
175