| /* |
| * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include <vector> |
| |
| #include "testing/gtest/include/gtest/gtest.h" |
| #include "webrtc/base/array_view.h" |
| #include "webrtc/modules/audio_processing/audio_buffer.h" |
| #include "webrtc/modules/audio_processing/include/audio_processing.h" |
| #include "webrtc/modules/audio_processing/level_controller/level_controller.h" |
| #include "webrtc/modules/audio_processing/test/audio_buffer_tools.h" |
| #include "webrtc/modules/audio_processing/test/bitexactness_tools.h" |
| |
| namespace webrtc { |
| namespace { |
| |
| const int kNumFramesToProcess = 1000; |
| |
| // Processes a specified amount of frames, verifies the results and reports |
| // any errors. |
| void RunBitexactnessTest(int sample_rate_hz, |
| size_t num_channels, |
| rtc::ArrayView<const float> output_reference) { |
| LevelController level_controller; |
| level_controller.Initialize(sample_rate_hz); |
| |
| int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100); |
| const StreamConfig capture_config(sample_rate_hz, num_channels, false); |
| AudioBuffer capture_buffer( |
| capture_config.num_frames(), capture_config.num_channels(), |
| capture_config.num_frames(), capture_config.num_channels(), |
| capture_config.num_frames()); |
| test::InputAudioFile capture_file( |
| test::GetApmCaptureTestVectorFileName(sample_rate_hz)); |
| std::vector<float> capture_input(samples_per_channel * num_channels); |
| for (size_t frame_no = 0; frame_no < kNumFramesToProcess; ++frame_no) { |
| ReadFloatSamplesFromStereoFile(samples_per_channel, num_channels, |
| &capture_file, capture_input); |
| |
| test::CopyVectorToAudioBuffer(capture_config, capture_input, |
| &capture_buffer); |
| |
| level_controller.Process(&capture_buffer); |
| } |
| |
| // Extract test results. |
| std::vector<float> capture_output; |
| test::ExtractVectorFromAudioBuffer(capture_config, &capture_buffer, |
| &capture_output); |
| |
| // Compare the output with the reference. Only the first values of the output |
| // from last frame processed are compared in order not having to specify all |
| // preceding frames as testvectors. As the algorithm being tested has a |
| // memory, testing only the last frame implicitly also tests the preceeding |
| // frames. |
| const float kVectorElementErrorBound = 1.0f / 32768.0f; |
| EXPECT_TRUE(test::VerifyDeinterleavedArray( |
| capture_config.num_frames(), capture_config.num_channels(), |
| output_reference, capture_output, kVectorElementErrorBound)); |
| } |
| |
| } // namespace |
| |
| TEST(LevelControlBitExactnessTest, DISABLED_Mono8kHz) { |
| const float kOutputReference[] = {-0.023242f, -0.020266f, -0.015097f}; |
| RunBitexactnessTest(AudioProcessing::kSampleRate8kHz, 1, kOutputReference); |
| } |
| |
| TEST(LevelControlBitExactnessTest, DISABLED_Mono16kHz) { |
| const float kOutputReference[] = {-0.019461f, -0.018761f, -0.018481f}; |
| RunBitexactnessTest(AudioProcessing::kSampleRate16kHz, 1, kOutputReference); |
| } |
| |
| TEST(LevelControlBitExactnessTest, DISABLED_Mono32kHz) { |
| const float kOutputReference[] = {-0.016872f, -0.019118f, -0.018722f}; |
| RunBitexactnessTest(AudioProcessing::kSampleRate32kHz, 1, kOutputReference); |
| } |
| |
| // TODO(peah): Investigate why this particular testcase differ between Android |
| // and the rest of the platforms. |
| TEST(LevelControlBitExactnessTest, DISABLED_Mono48kHz) { |
| #if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \ |
| defined(WEBRTC_ANDROID)) |
| const float kOutputReference[] = {-0.016771f, -0.017831f, -0.020482f}; |
| #else |
| const float kOutputReference[] = {-0.015949f, -0.016957f, -0.019478f}; |
| #endif |
| RunBitexactnessTest(AudioProcessing::kSampleRate48kHz, 1, kOutputReference); |
| } |
| |
| TEST(LevelControlBitExactnessTest, DISABLED_Stereo8kHz) { |
| const float kOutputReference[] = {-0.019304f, -0.011600f, -0.016690f, |
| -0.071335f, -0.031849f, -0.065694f}; |
| RunBitexactnessTest(AudioProcessing::kSampleRate8kHz, 2, kOutputReference); |
| } |
| |
| TEST(LevelControlBitExactnessTest, DISABLED_Stereo16kHz) { |
| const float kOutputReference[] = {-0.016302f, -0.007559f, -0.015668f, |
| -0.068346f, -0.031476f, -0.066065f}; |
| RunBitexactnessTest(AudioProcessing::kSampleRate16kHz, 2, kOutputReference); |
| } |
| |
| TEST(LevelControlBitExactnessTest, DISABLED_Stereo32kHz) { |
| const float kOutputReference[] = {-0.013944f, -0.008337f, -0.015972f, |
| -0.063563f, -0.031233f, -0.066784f}; |
| RunBitexactnessTest(AudioProcessing::kSampleRate32kHz, 2, kOutputReference); |
| } |
| |
| TEST(LevelControlBitExactnessTest, DISABLED_Stereo48kHz) { |
| const float kOutputReference[] = {-0.013652f, -0.008125f, -0.014593f, |
| -0.062963f, -0.030270f, -0.064727f}; |
| RunBitexactnessTest(AudioProcessing::kSampleRate48kHz, 2, kOutputReference); |
| } |
| |
| } // namespace webrtc |