blob: f9aee62ba9aab410af1be2d019587fdc2cf5639a [file] [log] [blame]
/*
* Copyright (c) 2018 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/agc2/vad_with_level.h"
#include "rtc_base/gunit.h"
namespace webrtc {
namespace test {
TEST(AutomaticGainController2VadWithLevelEstimator,
PeakLevelGreaterThanRmsLevel) {
constexpr size_t kSampleRateHz = 8000;
// 10 ms input frame, constant except for one peak value.
// Handcrafted so that the average is lower than the peak value.
std::array<float, kSampleRateHz / 100> frame;
frame.fill(1000.f);
frame[10] = 2000.f;
float* const channel0 = frame.data();
AudioFrameView<float> frame_view(&channel0, 1, frame.size());
// Compute audio frame levels (the VAD result is ignored).
VadWithLevel vad_with_level;
auto levels_and_vad_prob = vad_with_level.AnalyzeFrame(frame_view);
// Compare peak and RMS levels.
EXPECT_LT(levels_and_vad_prob.speech_rms_dbfs,
levels_and_vad_prob.speech_peak_dbfs);
}
} // namespace test
} // namespace webrtc