webrtc/modules/audio_processing/agc2/gain_curve_applier_unittest.cc
Alex Loiko a05ee82c4c Fixed Digital mode of AGC2 implementation finished.
This CL adds the GainCurveApplier (GCA). It owns a
FixedDigitalLevelEstimator (LE) and an InterpolatedGainCurve
(IGC). The GCA uses the LE to compute the input signal level, looks up
a gain from IGC and applies it on the signal.

The other IGC and LE submodules were added in previous CLs [1] and
[2].

This CL also turns on AGC2 in the APM fuzzer.

[1] https://webrtc-review.googlesource.com/c/src/+/51920
[2] https://webrtc-review.googlesource.com/c/src/+/52381

Bug: webrtc:7949
Change-Id: Idb10cc3ca9d6d2e4ac5824cc3391ed8aa680f6cd
Reviewed-on: https://webrtc-review.googlesource.com/54361
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22103}
2018-02-20 15:59:25 +00:00

59 lines
2.2 KiB
C++

/*
* 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/gain_curve_applier.h"
#include "common_audio/include/audio_util.h"
#include "modules/audio_processing/agc2/agc2_common.h"
#include "modules/audio_processing/agc2/agc2_testing_common.h"
#include "modules/audio_processing/agc2/vector_float_frame.h"
#include "rtc_base/gunit.h"
namespace webrtc {
TEST(GainCurveApplier, GainCurveApplierShouldConstructAndRun) {
const int sample_rate_hz = 48000;
ApmDataDumper apm_data_dumper(0);
GainCurveApplier gain_curve_applier(sample_rate_hz, &apm_data_dumper);
VectorFloatFrame vectors_with_float_frame(1, sample_rate_hz / 100,
kMaxAbsFloatS16Value);
gain_curve_applier.Process(vectors_with_float_frame.float_frame_view());
}
TEST(GainCurveApplier, OutputVolumeAboveThreshold) {
const int sample_rate_hz = 48000;
const float input_level =
(kMaxAbsFloatS16Value + DbfsToFloatS16(test::kLimiterMaxInputLevelDbFs)) /
2.f;
ApmDataDumper apm_data_dumper(0);
GainCurveApplier gain_curve_applier(sample_rate_hz, &apm_data_dumper);
// Give the level estimator time to adapt.
for (int i = 0; i < 5; ++i) {
VectorFloatFrame vectors_with_float_frame(1, sample_rate_hz / 100,
input_level);
gain_curve_applier.Process(vectors_with_float_frame.float_frame_view());
}
VectorFloatFrame vectors_with_float_frame(1, sample_rate_hz / 100,
input_level);
gain_curve_applier.Process(vectors_with_float_frame.float_frame_view());
rtc::ArrayView<const float> channel =
vectors_with_float_frame.float_frame_view().channel(0);
for (const auto& sample : channel) {
EXPECT_LT(0.9f * kMaxAbsFloatS16Value, sample);
}
}
} // namespace webrtc