| /* | 
 |  *  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_impl.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) { | 
 |   rtc::CriticalSection crit_capture; | 
 |   LevelEstimatorImpl level_estimator(&crit_capture); | 
 |   level_estimator.Initialize(); | 
 |   level_estimator.Enable(true); | 
 |  | 
 |   int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100); | 
 |   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_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 |