| /* |
| * 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 "api/array_view.h" |
| #include "modules/audio_processing/audio_buffer.h" |
| #include "modules/audio_processing/level_estimator.h" |
| #include "modules/audio_processing/test/audio_buffer_tools.h" |
| #include "modules/audio_processing/test/bitexactness_tools.h" |
| #include "test/gtest.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, |
| int rms_reference) { |
| LevelEstimator level_estimator; |
| int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100); |
| StreamConfig capture_config(sample_rate_hz, num_channels, false); |
| AudioBuffer capture_buffer( |
| capture_config.sample_rate_hz(), capture_config.num_channels(), |
| capture_config.sample_rate_hz(), capture_config.num_channels(), |
| capture_config.sample_rate_hz(), capture_config.num_channels()); |
| |
| 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_estimator.ProcessStream(capture_buffer); |
| } |
| |
| // Extract test results. |
| int rms = level_estimator.RMS(); |
| |
| // Compare the output to the reference. |
| EXPECT_EQ(rms_reference, rms); |
| } |
| |
| } // namespace |
| |
| TEST(LevelEstimatorBitExactnessTest, Mono8kHz) { |
| const int kRmsReference = 31; |
| |
| RunBitexactnessTest(8000, 1, kRmsReference); |
| } |
| |
| TEST(LevelEstimatorBitExactnessTest, Mono16kHz) { |
| const int kRmsReference = 31; |
| |
| RunBitexactnessTest(16000, 1, kRmsReference); |
| } |
| |
| TEST(LevelEstimatorBitExactnessTest, Mono32kHz) { |
| const int kRmsReference = 31; |
| |
| RunBitexactnessTest(32000, 1, kRmsReference); |
| } |
| |
| TEST(LevelEstimatorBitExactnessTest, Mono48kHz) { |
| const int kRmsReference = 31; |
| |
| RunBitexactnessTest(48000, 1, kRmsReference); |
| } |
| |
| TEST(LevelEstimatorBitExactnessTest, Stereo16kHz) { |
| const int kRmsReference = 30; |
| |
| RunBitexactnessTest(16000, 2, kRmsReference); |
| } |
| |
| } // namespace webrtc |