/* * Copyright (c) 2013 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 "modules/audio_processing/transient/transient_detector.h" #include #include #include "modules/audio_processing/transient/common.h" #include "modules/audio_processing/transient/file_utils.h" #include "rtc_base/strings/string_builder.h" #include "rtc_base/system/file_wrapper.h" #include "test/gtest.h" #include "test/testsupport/file_utils.h" namespace webrtc { static const int kSampleRatesHz[] = {ts::kSampleRate8kHz, ts::kSampleRate16kHz, ts::kSampleRate32kHz, ts::kSampleRate48kHz}; static const size_t kNumberOfSampleRates = sizeof(kSampleRatesHz) / sizeof(*kSampleRatesHz); // This test is for the correctness of the transient detector. // Checks the results comparing them with the ones stored in the detect files in // the directory: resources/audio_processing/transient/ // The files contain all the results in double precision (Little endian). // The audio files used with different sample rates are stored in the same // directory. #if defined(WEBRTC_IOS) TEST(TransientDetectorTest, DISABLED_CorrectnessBasedOnFiles) { #else TEST(TransientDetectorTest, CorrectnessBasedOnFiles) { #endif for (size_t i = 0; i < kNumberOfSampleRates; ++i) { int sample_rate_hz = kSampleRatesHz[i]; // Prepare detect file. rtc::StringBuilder detect_file_name; detect_file_name << "audio_processing/transient/detect" << (sample_rate_hz / 1000) << "kHz"; FileWrapper detect_file = FileWrapper::OpenReadOnly( test::ResourcePath(detect_file_name.str(), "dat")); bool file_opened = detect_file.is_open(); ASSERT_TRUE(file_opened) << "File could not be opened.\n" << detect_file_name.str().c_str(); // Prepare audio file. rtc::StringBuilder audio_file_name; audio_file_name << "audio_processing/transient/audio" << (sample_rate_hz / 1000) << "kHz"; FileWrapper audio_file = FileWrapper::OpenReadOnly( test::ResourcePath(audio_file_name.str(), "pcm")); // Create detector. TransientDetector detector(sample_rate_hz); const size_t buffer_length = sample_rate_hz * ts::kChunkSizeMs / 1000; std::unique_ptr buffer(new float[buffer_length]); const float kTolerance = 0.02f; size_t frames_read = 0; while (ReadInt16FromFileToFloatBuffer(&audio_file, buffer_length, buffer.get()) == buffer_length) { ++frames_read; float detector_value = detector.Detect(buffer.get(), buffer_length, NULL, 0); double file_value; ASSERT_EQ(1u, ReadDoubleBufferFromFile(&detect_file, 1, &file_value)) << "Detect test file is malformed.\n"; // Compare results with data from the matlab test file. EXPECT_NEAR(file_value, detector_value, kTolerance) << "Frame: " << frames_read; } detect_file.Close(); audio_file.Close(); } } } // namespace webrtc