blob: 94b84bbdc3555bb1b144de9f5e8ae224fffba70b [file] [log] [blame]
/*
* 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