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In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
202 lines
7.8 KiB
C++
202 lines
7.8 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_processing/aec3/echo_remover.h"
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#include <algorithm>
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#include <memory>
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#include <numeric>
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#include <sstream>
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#include <string>
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#include "modules/audio_processing/aec3/aec3_common.h"
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#include "modules/audio_processing/aec3/render_buffer.h"
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#include "modules/audio_processing/aec3/render_delay_buffer.h"
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#include "modules/audio_processing/logging/apm_data_dumper.h"
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#include "modules/audio_processing/test/echo_canceller_test_tools.h"
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#include "rtc_base/random.h"
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#include "test/gtest.h"
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namespace webrtc {
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namespace {
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std::string ProduceDebugText(int sample_rate_hz) {
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std::ostringstream ss;
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ss << "Sample rate: " << sample_rate_hz;
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return ss.str();
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}
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std::string ProduceDebugText(int sample_rate_hz, int delay) {
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std::ostringstream ss(ProduceDebugText(sample_rate_hz));
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ss << ", Delay: " << delay;
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return ss.str();
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}
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} // namespace
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// Verifies the basic API call sequence
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TEST(EchoRemover, BasicApiCalls) {
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for (auto rate : {8000, 16000, 32000, 48000}) {
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SCOPED_TRACE(ProduceDebugText(rate));
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std::unique_ptr<EchoRemover> remover(
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EchoRemover::Create(AudioProcessing::Config::EchoCanceller3(), rate));
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std::unique_ptr<RenderDelayBuffer> render_buffer(
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RenderDelayBuffer::Create(NumBandsForRate(rate)));
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std::vector<std::vector<float>> render(NumBandsForRate(rate),
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std::vector<float>(kBlockSize, 0.f));
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std::vector<std::vector<float>> capture(
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NumBandsForRate(rate), std::vector<float>(kBlockSize, 0.f));
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for (size_t k = 0; k < 100; ++k) {
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EchoPathVariability echo_path_variability(k % 3 == 0 ? true : false,
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k % 5 == 0 ? true : false);
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rtc::Optional<size_t> echo_path_delay_samples =
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(k % 6 == 0 ? rtc::Optional<size_t>(k * 10)
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: rtc::Optional<size_t>());
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render_buffer->Insert(render);
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render_buffer->UpdateBuffers();
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remover->ProcessCapture(echo_path_delay_samples, echo_path_variability,
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k % 2 == 0 ? true : false,
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render_buffer->GetRenderBuffer(), &capture);
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}
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}
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}
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#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
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// Verifies the check for the samplerate.
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// TODO(peah): Re-enable the test once the issue with memory leaks during DEATH
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// tests on test bots has been fixed.
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TEST(EchoRemover, DISABLED_WrongSampleRate) {
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EXPECT_DEATH(std::unique_ptr<EchoRemover>(EchoRemover::Create(
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AudioProcessing::Config::EchoCanceller3(), 8001)),
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"");
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}
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// Verifies the check for the capture block size.
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TEST(EchoRemover, WrongCaptureBlockSize) {
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for (auto rate : {8000, 16000, 32000, 48000}) {
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SCOPED_TRACE(ProduceDebugText(rate));
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std::unique_ptr<EchoRemover> remover(
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EchoRemover::Create(AudioProcessing::Config::EchoCanceller3(), rate));
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std::unique_ptr<RenderDelayBuffer> render_buffer(
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RenderDelayBuffer::Create(NumBandsForRate(rate)));
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std::vector<std::vector<float>> capture(
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NumBandsForRate(rate), std::vector<float>(kBlockSize - 1, 0.f));
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EchoPathVariability echo_path_variability(false, false);
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rtc::Optional<size_t> echo_path_delay_samples;
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EXPECT_DEATH(remover->ProcessCapture(
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echo_path_delay_samples, echo_path_variability, false,
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render_buffer->GetRenderBuffer(), &capture),
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"");
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}
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}
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// Verifies the check for the number of capture bands.
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// TODO(peah): Re-enable the test once the issue with memory leaks during DEATH
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// tests on test bots has been fixed.c
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TEST(EchoRemover, DISABLED_WrongCaptureNumBands) {
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for (auto rate : {16000, 32000, 48000}) {
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SCOPED_TRACE(ProduceDebugText(rate));
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std::unique_ptr<EchoRemover> remover(
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EchoRemover::Create(AudioProcessing::Config::EchoCanceller3(), rate));
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std::unique_ptr<RenderDelayBuffer> render_buffer(
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RenderDelayBuffer::Create(NumBandsForRate(rate)));
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std::vector<std::vector<float>> capture(
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NumBandsForRate(rate == 48000 ? 16000 : rate + 16000),
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std::vector<float>(kBlockSize, 0.f));
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EchoPathVariability echo_path_variability(false, false);
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rtc::Optional<size_t> echo_path_delay_samples;
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EXPECT_DEATH(remover->ProcessCapture(
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echo_path_delay_samples, echo_path_variability, false,
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render_buffer->GetRenderBuffer(), &capture),
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"");
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}
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}
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// Verifies the check for non-null capture block.
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TEST(EchoRemover, NullCapture) {
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std::unique_ptr<EchoRemover> remover(
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EchoRemover::Create(AudioProcessing::Config::EchoCanceller3(), 8000));
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std::unique_ptr<RenderDelayBuffer> render_buffer(
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RenderDelayBuffer::Create(3));
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EchoPathVariability echo_path_variability(false, false);
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rtc::Optional<size_t> echo_path_delay_samples;
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EXPECT_DEATH(
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remover->ProcessCapture(echo_path_delay_samples, echo_path_variability,
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false, render_buffer->GetRenderBuffer(), nullptr),
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"");
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}
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#endif
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// Performs a sanity check that the echo_remover is able to properly
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// remove echoes.
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TEST(EchoRemover, BasicEchoRemoval) {
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constexpr int kNumBlocksToProcess = 500;
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Random random_generator(42U);
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for (auto rate : {8000, 16000, 32000, 48000}) {
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std::vector<std::vector<float>> x(NumBandsForRate(rate),
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std::vector<float>(kBlockSize, 0.f));
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std::vector<std::vector<float>> y(NumBandsForRate(rate),
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std::vector<float>(kBlockSize, 0.f));
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EchoPathVariability echo_path_variability(false, false);
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for (size_t delay_samples : {0, 64, 150, 200, 301}) {
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SCOPED_TRACE(ProduceDebugText(rate, delay_samples));
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std::unique_ptr<EchoRemover> remover(
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EchoRemover::Create(AudioProcessing::Config::EchoCanceller3(), rate));
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std::unique_ptr<RenderDelayBuffer> render_buffer(
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RenderDelayBuffer::Create(NumBandsForRate(rate)));
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std::vector<std::unique_ptr<DelayBuffer<float>>> delay_buffers(x.size());
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for (size_t j = 0; j < x.size(); ++j) {
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delay_buffers[j].reset(new DelayBuffer<float>(delay_samples));
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}
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float input_energy = 0.f;
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float output_energy = 0.f;
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for (int k = 0; k < kNumBlocksToProcess; ++k) {
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const bool silence = k < 100 || (k % 100 >= 10);
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for (size_t j = 0; j < x.size(); ++j) {
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if (silence) {
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std::fill(x[j].begin(), x[j].end(), 0.f);
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} else {
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RandomizeSampleVector(&random_generator, x[j]);
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}
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delay_buffers[j]->Delay(x[j], y[j]);
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}
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if (k > kNumBlocksToProcess / 2) {
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for (size_t j = 0; j < x.size(); ++j) {
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input_energy = std::inner_product(y[j].begin(), y[j].end(),
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y[j].begin(), input_energy);
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}
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}
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render_buffer->Insert(x);
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render_buffer->UpdateBuffers();
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remover->ProcessCapture(rtc::Optional<size_t>(delay_samples),
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echo_path_variability, false,
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render_buffer->GetRenderBuffer(), &y);
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if (k > kNumBlocksToProcess / 2) {
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for (size_t j = 0; j < x.size(); ++j) {
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output_energy = std::inner_product(y[j].begin(), y[j].end(),
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y[j].begin(), output_energy);
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}
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}
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}
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EXPECT_GT(input_energy, 10.f * output_energy);
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}
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}
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}
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} // namespace webrtc
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