mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-20 09:07:52 +01:00
1b4059e84f
This CL changes the AEC3 behavior to be more transparent when there is uncertainty about the amount of echo in the microphone signal. Bug: webrtc:8398, chromium:774868 Change-Id: I88e681f8decd892f44397b753df371a1c4b90af0 Reviewed-on: https://webrtc-review.googlesource.com/10801 Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20319}
95 lines
3.4 KiB
C++
95 lines
3.4 KiB
C++
/*
|
|
* Copyright (c) 2017 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/aec3/residual_echo_estimator.h"
|
|
|
|
#include "modules/audio_processing/aec3/aec3_fft.h"
|
|
#include "modules/audio_processing/aec3/aec_state.h"
|
|
#include "modules/audio_processing/include/audio_processing.h"
|
|
#include "modules/audio_processing/test/echo_canceller_test_tools.h"
|
|
#include "rtc_base/random.h"
|
|
#include "test/gtest.h"
|
|
|
|
namespace webrtc {
|
|
|
|
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
|
|
|
|
// Verifies that the check for non-null output residual echo power works.
|
|
TEST(ResidualEchoEstimator, NullResidualEchoPowerOutput) {
|
|
AecState aec_state(AudioProcessing::Config::EchoCanceller3{});
|
|
RenderBuffer render_buffer(Aec3Optimization::kNone, 3, 10,
|
|
std::vector<size_t>(1, 10));
|
|
std::vector<std::array<float, kFftLengthBy2Plus1>> H2;
|
|
std::array<float, kFftLengthBy2Plus1> S2_linear;
|
|
std::array<float, kFftLengthBy2Plus1> Y2;
|
|
EXPECT_DEATH(ResidualEchoEstimator(AudioProcessing::Config::EchoCanceller3{})
|
|
.Estimate(aec_state, render_buffer, S2_linear, Y2, nullptr),
|
|
"");
|
|
}
|
|
|
|
#endif
|
|
|
|
TEST(ResidualEchoEstimator, BasicTest) {
|
|
ResidualEchoEstimator estimator(AudioProcessing::Config::EchoCanceller3{});
|
|
AudioProcessing::Config::EchoCanceller3 config;
|
|
config.param.ep_strength.default_len = 0.f;
|
|
AecState aec_state(config);
|
|
RenderBuffer render_buffer(Aec3Optimization::kNone, 3, 10,
|
|
std::vector<size_t>(1, 10));
|
|
std::array<float, kFftLengthBy2Plus1> E2_main;
|
|
std::array<float, kFftLengthBy2Plus1> E2_shadow;
|
|
std::array<float, kFftLengthBy2Plus1> S2_linear;
|
|
std::array<float, kFftLengthBy2Plus1> S2_fallback;
|
|
std::array<float, kFftLengthBy2Plus1> Y2;
|
|
std::array<float, kFftLengthBy2Plus1> R2;
|
|
EchoPathVariability echo_path_variability(false, false);
|
|
std::vector<std::vector<float>> x(3, std::vector<float>(kBlockSize, 0.f));
|
|
std::vector<std::array<float, kFftLengthBy2Plus1>> H2(10);
|
|
Random random_generator(42U);
|
|
FftData X;
|
|
std::array<float, kBlockSize> x_old;
|
|
std::array<float, kBlockSize> s;
|
|
Aec3Fft fft;
|
|
|
|
for (auto& H2_k : H2) {
|
|
H2_k.fill(0.01f);
|
|
}
|
|
H2[2].fill(10.f);
|
|
H2[2][0] = 0.1f;
|
|
|
|
std::array<float, kAdaptiveFilterTimeDomainLength> h;
|
|
h.fill(0.f);
|
|
|
|
s.fill(100.f);
|
|
|
|
constexpr float kLevel = 10.f;
|
|
E2_shadow.fill(kLevel);
|
|
E2_main.fill(kLevel);
|
|
S2_linear.fill(kLevel);
|
|
S2_fallback.fill(kLevel);
|
|
Y2.fill(kLevel);
|
|
|
|
for (int k = 0; k < 2000; ++k) {
|
|
RandomizeSampleVector(&random_generator, x[0]);
|
|
std::for_each(x[0].begin(), x[0].end(), [](float& a) { a /= 30.f; });
|
|
fft.PaddedFft(x[0], x_old, &X);
|
|
render_buffer.Insert(x);
|
|
|
|
aec_state.HandleEchoPathChange(echo_path_variability);
|
|
aec_state.Update(H2, h, true, rtc::Optional<size_t>(2), render_buffer,
|
|
E2_main, Y2, x[0], s, false);
|
|
|
|
estimator.Estimate(aec_state, render_buffer, S2_linear, Y2, &R2);
|
|
}
|
|
std::for_each(R2.begin(), R2.end(),
|
|
[&](float a) { EXPECT_NEAR(kLevel, a, 0.1f); });
|
|
}
|
|
|
|
} // namespace webrtc
|