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| 1 /* |
| 2 * Copyright (c) 2017 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 #include <memory> |
| 12 #include <string> |
| 13 |
| 14 #include "webrtc/base/array_view.h" |
| 15 #include "webrtc/modules/audio_processing/audio_buffer.h" |
| 16 #include "webrtc/modules/audio_processing/agc2/gain_controller2.h" |
| 17 #include "webrtc/modules/audio_processing/agc2/digital_gain_applier.h" |
| 18 #include "webrtc/test/gtest.h" |
| 19 |
| 20 namespace webrtc { |
| 21 namespace test { |
| 22 |
| 23 namespace { |
| 24 |
| 25 constexpr size_t kNumFrames = 480u; |
| 26 constexpr size_t kStereo = 2u; |
| 27 |
| 28 void SetAudioBufferSamples(float value, AudioBuffer* ab) { |
| 29 for (size_t k = 0; k < ab->num_channels(); ++k) { |
| 30 auto channel = rtc::ArrayView<float>(ab->channels_f()[k], ab->num_frames()); |
| 31 for (auto& sample : channel) { sample = value; } |
| 32 } |
| 33 } |
| 34 |
| 35 template<typename Functor> |
| 36 bool CheckAudioBufferSamples(Functor validator, AudioBuffer* ab) { |
| 37 for (size_t k = 0; k < ab->num_channels(); ++k) { |
| 38 auto channel = rtc::ArrayView<float>(ab->channels_f()[k], ab->num_frames()); |
| 39 for (auto& sample : channel) { if (!validator(sample)) { return false; } } |
| 40 } |
| 41 return true; |
| 42 } |
| 43 |
| 44 bool TestDigitalGainApplier(float sample_value, float gain, float expected) { |
| 45 AudioBuffer ab(kNumFrames, kStereo, kNumFrames, kStereo, kNumFrames); |
| 46 SetAudioBufferSamples(sample_value, &ab); |
| 47 |
| 48 DigitalGainApplier gain_applier; |
| 49 for (size_t k = 0; k < ab.num_channels(); ++k) { |
| 50 auto channel_view = rtc::ArrayView<float>( |
| 51 ab.channels_f()[k], ab.num_frames()); |
| 52 gain_applier.Process(gain, channel_view); |
| 53 } |
| 54 |
| 55 auto check_expectation = [expected](float sample) { |
| 56 return sample == expected; }; |
| 57 return CheckAudioBufferSamples(check_expectation, &ab); |
| 58 } |
| 59 |
| 60 } // namespace |
| 61 |
| 62 TEST(GainController2, Instance) { |
| 63 std::unique_ptr<GainController2> gain_controller2; |
| 64 gain_controller2.reset(new GainController2( |
| 65 AudioProcessing::kSampleRate48kHz)); |
| 66 } |
| 67 |
| 68 TEST(GainController2, ToString) { |
| 69 AudioProcessing::Config config; |
| 70 |
| 71 config.gain_controller2.enabled = false; |
| 72 EXPECT_EQ("{enabled: false}", |
| 73 GainController2::ToString(config.gain_controller2)); |
| 74 |
| 75 config.gain_controller2.enabled = true; |
| 76 EXPECT_EQ("{enabled: true}", |
| 77 GainController2::ToString(config.gain_controller2)); |
| 78 } |
| 79 |
| 80 TEST(GainController2, DigitalGainApplierProcess) { |
| 81 EXPECT_TRUE(TestDigitalGainApplier(1000.0f, 0.5, 500.0f)); |
| 82 } |
| 83 |
| 84 TEST(GainController2, DigitalGainApplierCheckClipping) { |
| 85 EXPECT_TRUE(TestDigitalGainApplier(30000.0f, 1.5, 32767.0f)); |
| 86 EXPECT_TRUE(TestDigitalGainApplier(-30000.0f, 1.5, -32767.0f)); |
| 87 } |
| 88 |
| 89 TEST(GainController2, Usage) { |
| 90 std::unique_ptr<GainController2> gain_controller2; |
| 91 gain_controller2.reset(new GainController2( |
| 92 AudioProcessing::kSampleRate48kHz)); |
| 93 AudioBuffer ab(kNumFrames, kStereo, kNumFrames, kStereo, kNumFrames); |
| 94 SetAudioBufferSamples(1000.0f, &ab); |
| 95 gain_controller2->Process(&ab); |
| 96 } |
| 97 |
| 98 } // namespace test |
| 99 } // namespace webrtc |
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