mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-15 06:40:43 +01:00
ee84d39fce
Multichannel signals are downmixed to mono before decimation and delay estimation. This is useful when not all channels play audio content. The feature can be toggled in the AEC3 configuration. Bug: webrtc:10913 Change-Id: I7d40edf7732bb51fec69e7f3ca063d821c5069c4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151762 Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29126}
137 lines
4.7 KiB
C++
137 lines
4.7 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/decimator.h"
|
|
|
|
#include <math.h>
|
|
|
|
#include <algorithm>
|
|
#include <array>
|
|
#include <cmath>
|
|
#include <cstring>
|
|
#include <numeric>
|
|
#include <string>
|
|
#include <vector>
|
|
|
|
#include "modules/audio_processing/aec3/aec3_common.h"
|
|
#include "rtc_base/strings/string_builder.h"
|
|
#include "test/gtest.h"
|
|
|
|
namespace webrtc {
|
|
|
|
namespace {
|
|
|
|
std::string ProduceDebugText(int sample_rate_hz) {
|
|
rtc::StringBuilder ss;
|
|
ss << "Sample rate: " << sample_rate_hz;
|
|
return ss.Release();
|
|
}
|
|
|
|
constexpr size_t kDownSamplingFactors[] = {2, 4, 8};
|
|
constexpr float kPi = 3.141592f;
|
|
constexpr size_t kNumStartupBlocks = 50;
|
|
constexpr size_t kNumBlocks = 1000;
|
|
|
|
void ProduceDecimatedSinusoidalOutputPower(int sample_rate_hz,
|
|
size_t down_sampling_factor,
|
|
float sinusoidal_frequency_hz,
|
|
float* input_power,
|
|
float* output_power) {
|
|
float input[kBlockSize * kNumBlocks];
|
|
const size_t sub_block_size = kBlockSize / down_sampling_factor;
|
|
|
|
// Produce a sinusoid of the specified frequency.
|
|
for (size_t k = 0; k < kBlockSize * kNumBlocks; ++k) {
|
|
input[k] = 32767.f * std::sin(2.f * kPi * sinusoidal_frequency_hz * k /
|
|
sample_rate_hz);
|
|
}
|
|
|
|
Decimator decimator(down_sampling_factor);
|
|
std::vector<float> output(sub_block_size * kNumBlocks);
|
|
|
|
for (size_t k = 0; k < kNumBlocks; ++k) {
|
|
std::vector<float> sub_block(sub_block_size);
|
|
std::vector<std::vector<float>> input_multichannel(
|
|
1, std::vector<float>(kBlockSize));
|
|
memcpy(input_multichannel[0].data(), &input[k * kBlockSize],
|
|
kBlockSize * sizeof(float));
|
|
decimator.Decimate(input_multichannel, true, sub_block);
|
|
|
|
std::copy(sub_block.begin(), sub_block.end(),
|
|
output.begin() + k * sub_block_size);
|
|
}
|
|
|
|
ASSERT_GT(kNumBlocks, kNumStartupBlocks);
|
|
rtc::ArrayView<const float> input_to_evaluate(
|
|
&input[kNumStartupBlocks * kBlockSize],
|
|
(kNumBlocks - kNumStartupBlocks) * kBlockSize);
|
|
rtc::ArrayView<const float> output_to_evaluate(
|
|
&output[kNumStartupBlocks * sub_block_size],
|
|
(kNumBlocks - kNumStartupBlocks) * sub_block_size);
|
|
*input_power =
|
|
std::inner_product(input_to_evaluate.begin(), input_to_evaluate.end(),
|
|
input_to_evaluate.begin(), 0.f) /
|
|
input_to_evaluate.size();
|
|
*output_power =
|
|
std::inner_product(output_to_evaluate.begin(), output_to_evaluate.end(),
|
|
output_to_evaluate.begin(), 0.f) /
|
|
output_to_evaluate.size();
|
|
}
|
|
|
|
} // namespace
|
|
|
|
// Verifies that there is little aliasing from upper frequencies in the
|
|
// downsampling.
|
|
TEST(Decimator, NoLeakageFromUpperFrequencies) {
|
|
float input_power;
|
|
float output_power;
|
|
for (auto rate : {16000, 32000, 48000}) {
|
|
for (auto down_sampling_factor : kDownSamplingFactors) {
|
|
ProduceDebugText(rate);
|
|
ProduceDecimatedSinusoidalOutputPower(rate, down_sampling_factor,
|
|
3.f / 8.f * rate, &input_power,
|
|
&output_power);
|
|
EXPECT_GT(0.0001f * input_power, output_power);
|
|
}
|
|
}
|
|
}
|
|
|
|
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
|
|
// Verifies the check for the input size.
|
|
TEST(Decimator, WrongInputSize) {
|
|
Decimator decimator(4);
|
|
std::vector<std::vector<float>> x(1, std::vector<float>(kBlockSize - 1, 0.f));
|
|
std::array<float, kBlockSize / 4> x_downsampled;
|
|
EXPECT_DEATH(decimator.Decimate(x, true, x_downsampled), "");
|
|
}
|
|
|
|
// Verifies the check for non-null output parameter.
|
|
TEST(Decimator, NullOutput) {
|
|
Decimator decimator(4);
|
|
std::vector<std::vector<float>> x(1, std::vector<float>(kBlockSize, 0.f));
|
|
EXPECT_DEATH(decimator.Decimate(x, true, nullptr), "");
|
|
}
|
|
|
|
// Verifies the check for the output size.
|
|
TEST(Decimator, WrongOutputSize) {
|
|
Decimator decimator(4);
|
|
std::vector<std::vector<float>> x(1, std::vector<float>(kBlockSize, 0.f));
|
|
std::array<float, kBlockSize / 4 - 1> x_downsampled;
|
|
EXPECT_DEATH(decimator.Decimate(x, true, x_downsampled), "");
|
|
}
|
|
|
|
// Verifies the check for the correct downsampling factor.
|
|
TEST(Decimator, CorrectDownSamplingFactor) {
|
|
EXPECT_DEATH(Decimator(3), "");
|
|
}
|
|
|
|
#endif
|
|
|
|
} // namespace webrtc
|