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1 /* | |
2 * Copyright (c) 2016 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 #include <vector> | |
11 | |
12 #include "testing/gtest/include/gtest/gtest.h" | |
13 #include "webrtc/base/array_view.h" | |
14 #include "webrtc/modules/audio_processing/audio_buffer.h" | |
15 #include "webrtc/modules/audio_processing/intelligibility/intelligibility_enhanc er.h" | |
16 #include "webrtc/modules/audio_processing/noise_suppression_impl.h" | |
17 #include "webrtc/modules/audio_processing/test/audio_buffer_tools.h" | |
18 #include "webrtc/modules/audio_processing/test/bitexactness_tools.h" | |
19 | |
20 namespace webrtc { | |
21 namespace { | |
22 | |
23 const size_t kNumFramesToProcess = 1000; | |
24 | |
25 int IntelligibilityEnhancerSampleRate(int sample_rate_hz) { | |
26 return (sample_rate_hz > AudioProcessing::kSampleRate16kHz | |
27 ? AudioProcessing::kSampleRate16kHz | |
28 : sample_rate_hz); | |
29 } | |
30 | |
31 // Process one frame of data and produce the output. | |
32 void ProcessOneFrame(int sample_rate_hz, | |
33 AudioBuffer* render_audio_buffer, | |
34 AudioBuffer* capture_audio_buffer, | |
35 NoiseSuppressionImpl* noise_suppressor, | |
36 IntelligibilityEnhancer* intelligibility_enhancer) { | |
37 if (sample_rate_hz > AudioProcessing::kSampleRate16kHz) { | |
38 render_audio_buffer->SplitIntoFrequencyBands(); | |
39 capture_audio_buffer->SplitIntoFrequencyBands(); | |
40 } | |
41 | |
42 intelligibility_enhancer->ProcessRenderAudio( | |
43 render_audio_buffer->split_channels_f(kBand0To8kHz), | |
44 IntelligibilityEnhancerSampleRate(sample_rate_hz), | |
45 render_audio_buffer->num_channels()); | |
46 | |
47 noise_suppressor->AnalyzeCaptureAudio(capture_audio_buffer); | |
48 noise_suppressor->ProcessCaptureAudio(capture_audio_buffer); | |
49 | |
50 intelligibility_enhancer->SetCaptureNoiseEstimate( | |
51 noise_suppressor->NoiseEstimate()); | |
52 | |
53 if (sample_rate_hz > AudioProcessing::kSampleRate16kHz) { | |
54 render_audio_buffer->MergeFrequencyBands(); | |
55 } | |
56 } | |
57 | |
58 // Processes a specified amount of frames, verifies the results and reports | |
59 // any errors. | |
60 void RunBitexactnessTest(int sample_rate_hz, | |
61 size_t num_channels, | |
62 rtc::ArrayView<const float> output_reference) { | |
63 const StreamConfig render_config(sample_rate_hz, num_channels, false); | |
64 AudioBuffer render_buffer( | |
65 render_config.num_frames(), render_config.num_channels(), | |
66 render_config.num_frames(), render_config.num_channels(), | |
67 render_config.num_frames()); | |
68 test::InputAudioFile render_file( | |
69 test::GetApmRenderTestVectorFileName(sample_rate_hz)); | |
70 std::vector<float> render_input(render_buffer.num_frames() * | |
71 render_buffer.num_channels()); | |
72 | |
73 const StreamConfig capture_config(sample_rate_hz, num_channels, false); | |
74 AudioBuffer capture_buffer( | |
75 capture_config.num_frames(), capture_config.num_channels(), | |
76 capture_config.num_frames(), capture_config.num_channels(), | |
77 capture_config.num_frames()); | |
78 test::InputAudioFile capture_file( | |
79 test::GetApmCaptureTestVectorFileName(sample_rate_hz)); | |
80 std::vector<float> capture_input(render_buffer.num_frames() * | |
81 capture_buffer.num_channels()); | |
82 | |
83 rtc::CriticalSection crit_capture; | |
84 NoiseSuppressionImpl noise_suppressor(&crit_capture); | |
85 noise_suppressor.Initialize(capture_config.num_channels(), sample_rate_hz); | |
86 noise_suppressor.Enable(true); | |
87 | |
88 IntelligibilityEnhancer intelligibility_enhancer( | |
89 IntelligibilityEnhancerSampleRate(sample_rate_hz), | |
90 render_config.num_channels(), NoiseSuppressionImpl::num_noise_bins()); | |
91 | |
92 for (size_t frame_no = 0u; frame_no < kNumFramesToProcess; ++frame_no) { | |
93 ReadFloatSamplesFromStereoFile(render_buffer.num_frames(), | |
94 render_buffer.num_channels(), &render_file, | |
95 render_input); | |
96 ReadFloatSamplesFromStereoFile(render_buffer.num_frames(), | |
aluebs-webrtc
2016/03/22 11:53:29
capture_buffer
peah-webrtc
2016/03/23 22:13:53
Good catch!!!
Done.
| |
97 capture_buffer.num_channels(), &capture_file, | |
98 capture_input); | |
99 | |
100 test::CopyVectorToAudioBuffer(render_config, render_input, &render_buffer); | |
101 test::CopyVectorToAudioBuffer(capture_config, capture_input, | |
102 &capture_buffer); | |
103 | |
104 ProcessOneFrame(sample_rate_hz, &render_buffer, &capture_buffer, | |
105 &noise_suppressor, &intelligibility_enhancer); | |
106 } | |
107 | |
108 // Extract and verify the test results. | |
109 std::vector<float> render_output; | |
110 test::ExtractVectorFromAudioBuffer(render_config, &render_buffer, | |
111 &render_output); | |
112 | |
113 const float kTolerance = 1.f / static_cast<float>(1 << 15); | |
114 | |
115 // Compare the output with the reference. Only the first values of the output | |
116 // from last frame processed are compared in order not having to specify all | |
117 // preceeding frames as testvectors. As the algorithm being tested has a | |
118 // memory, testing only the last frame implicitly also tests the preceeding | |
119 // frames. | |
120 EXPECT_TRUE(test::BitExactFrame(render_buffer.num_frames(), | |
121 render_config.num_channels(), | |
122 output_reference, render_output, kTolerance)); | |
123 } | |
124 | |
125 } // namespace | |
126 | |
127 TEST(IntelligibilityEnhancerBitExactnessTest, Mono8kHz) { | |
128 const float kOutputReference[] = {-0.001892f, -0.003296f, -0.001953f}; | |
129 | |
130 RunBitexactnessTest(AudioProcessing::kSampleRate8kHz, 1, kOutputReference); | |
131 } | |
132 | |
133 TEST(IntelligibilityEnhancerBitExactnessTest, Mono16kHz) { | |
134 const float kOutputReference[] = {-0.000977f, -0.003296f, -0.002441f}; | |
135 | |
136 RunBitexactnessTest(AudioProcessing::kSampleRate16kHz, 1, kOutputReference); | |
137 } | |
138 | |
139 TEST(IntelligibilityEnhancerBitExactnessTest, Mono32kHz) { | |
140 const float kOutputReference[] = {0.003021f, -0.011780f, -0.008209f}; | |
141 | |
142 RunBitexactnessTest(AudioProcessing::kSampleRate32kHz, 1, kOutputReference); | |
143 } | |
144 | |
145 TEST(IntelligibilityEnhancerBitExactnessTest, Mono48kHz) { | |
146 const float kOutputReference[] = {-0.027696f, -0.026253f, -0.018001f}; | |
147 | |
148 RunBitexactnessTest(AudioProcessing::kSampleRate48kHz, 1, kOutputReference); | |
149 } | |
150 | |
151 TEST(IntelligibilityEnhancerBitExactnessTest, Stereo8kHz) { | |
152 const float kOutputReference[] = {0.021454f, 0.035919f, 0.026428f, | |
153 -0.000641f, 0.000366f, 0.000641f}; | |
154 | |
155 RunBitexactnessTest(AudioProcessing::kSampleRate8kHz, 2, kOutputReference); | |
156 } | |
157 | |
158 TEST(IntelligibilityEnhancerBitExactnessTest, Stereo16kHz) { | |
159 const float kOutputReference[] = {0.021362f, 0.035736f, 0.023895f, | |
160 -0.001404f, -0.001465f, 0.000549f}; | |
161 | |
162 RunBitexactnessTest(AudioProcessing::kSampleRate16kHz, 2, kOutputReference); | |
163 } | |
164 | |
165 TEST(IntelligibilityEnhancerBitExactnessTest, Stereo32kHz) { | |
166 const float kOutputReference[] = {0.030641f, 0.027406f, 0.028321f, | |
167 -0.001343f, -0.004578f, 0.000977f}; | |
168 | |
169 RunBitexactnessTest(AudioProcessing::kSampleRate32kHz, 2, kOutputReference); | |
170 } | |
171 | |
172 TEST(IntelligibilityEnhancerBitExactnessTest, Stereo48kHz) { | |
173 const float kOutputReference[] = {-0.009276f, -0.001601f, -0.008255f, | |
174 -0.012975f, -0.015940f, -0.017820f}; | |
175 | |
176 RunBitexactnessTest(AudioProcessing::kSampleRate48kHz, 2, kOutputReference); | |
177 } | |
178 | |
179 } // namespace webrtc | |
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