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92ea95e34a
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}
261 lines
10 KiB
C++
261 lines
10 KiB
C++
/*
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* Copyright (c) 2016 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/block_processor.h"
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#include <memory>
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#include <sstream>
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#include <string>
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#include <vector>
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#include "modules/audio_processing/aec3/aec3_common.h"
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#include "modules/audio_processing/aec3/mock/mock_echo_remover.h"
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#include "modules/audio_processing/aec3/mock/mock_render_delay_buffer.h"
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#include "modules/audio_processing/aec3/mock/mock_render_delay_controller.h"
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#include "modules/audio_processing/test/echo_canceller_test_tools.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/random.h"
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#include "test/gmock.h"
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#include "test/gtest.h"
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namespace webrtc {
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namespace {
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using testing::AtLeast;
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using testing::Return;
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using testing::StrictMock;
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using testing::_;
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// Verifies that the basic BlockProcessor functionality works and that the API
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// methods are callable.
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void RunBasicSetupAndApiCallTest(int sample_rate_hz) {
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std::unique_ptr<BlockProcessor> block_processor(BlockProcessor::Create(
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AudioProcessing::Config::EchoCanceller3(), sample_rate_hz));
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std::vector<std::vector<float>> block(NumBandsForRate(sample_rate_hz),
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std::vector<float>(kBlockSize, 0.f));
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block_processor->BufferRender(block);
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block_processor->ProcessCapture(false, false, &block);
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block_processor->UpdateEchoLeakageStatus(false);
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}
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#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
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void RunRenderBlockSizeVerificationTest(int sample_rate_hz) {
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std::unique_ptr<BlockProcessor> block_processor(BlockProcessor::Create(
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AudioProcessing::Config::EchoCanceller3(), sample_rate_hz));
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std::vector<std::vector<float>> block(
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NumBandsForRate(sample_rate_hz), std::vector<float>(kBlockSize - 1, 0.f));
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EXPECT_DEATH(block_processor->BufferRender(block), "");
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}
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void RunCaptureBlockSizeVerificationTest(int sample_rate_hz) {
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std::unique_ptr<BlockProcessor> block_processor(BlockProcessor::Create(
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AudioProcessing::Config::EchoCanceller3(), sample_rate_hz));
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std::vector<std::vector<float>> block(
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NumBandsForRate(sample_rate_hz), std::vector<float>(kBlockSize - 1, 0.f));
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EXPECT_DEATH(block_processor->ProcessCapture(false, false, &block), "");
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}
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void RunRenderNumBandsVerificationTest(int sample_rate_hz) {
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const size_t wrong_num_bands = NumBandsForRate(sample_rate_hz) < 3
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? NumBandsForRate(sample_rate_hz) + 1
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: 1;
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std::unique_ptr<BlockProcessor> block_processor(BlockProcessor::Create(
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AudioProcessing::Config::EchoCanceller3(), sample_rate_hz));
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std::vector<std::vector<float>> block(wrong_num_bands,
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std::vector<float>(kBlockSize, 0.f));
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EXPECT_DEATH(block_processor->BufferRender(block), "");
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}
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void RunCaptureNumBandsVerificationTest(int sample_rate_hz) {
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const size_t wrong_num_bands = NumBandsForRate(sample_rate_hz) < 3
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? NumBandsForRate(sample_rate_hz) + 1
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: 1;
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std::unique_ptr<BlockProcessor> block_processor(BlockProcessor::Create(
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AudioProcessing::Config::EchoCanceller3(), sample_rate_hz));
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std::vector<std::vector<float>> block(wrong_num_bands,
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std::vector<float>(kBlockSize, 0.f));
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EXPECT_DEATH(block_processor->ProcessCapture(false, false, &block), "");
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}
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#endif
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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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} // namespace
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// Verifies that the delay controller functionality is properly integrated with
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// the render delay buffer inside block processor.
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// TODO(peah): Activate the unittest once the required code has been landed.
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TEST(BlockProcessor, DISABLED_DelayControllerIntegration) {
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constexpr size_t kNumBlocks = 310;
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constexpr size_t kDelayInSamples = 640;
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constexpr size_t kDelayHeadroom = 1;
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constexpr size_t kDelayInBlocks =
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kDelayInSamples / kBlockSize - kDelayHeadroom;
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Random random_generator(42U);
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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<testing::StrictMock<webrtc::test::MockRenderDelayBuffer>>
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render_delay_buffer_mock(
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new StrictMock<webrtc::test::MockRenderDelayBuffer>(rate));
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EXPECT_CALL(*render_delay_buffer_mock, Insert(_))
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.Times(kNumBlocks)
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.WillRepeatedly(Return(true));
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EXPECT_CALL(*render_delay_buffer_mock, IsBlockAvailable())
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.Times(kNumBlocks)
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.WillRepeatedly(Return(true));
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EXPECT_CALL(*render_delay_buffer_mock, SetDelay(kDelayInBlocks))
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.Times(AtLeast(1));
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EXPECT_CALL(*render_delay_buffer_mock, MaxDelay()).WillOnce(Return(30));
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EXPECT_CALL(*render_delay_buffer_mock, Delay())
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.Times(kNumBlocks + 1)
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.WillRepeatedly(Return(0));
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std::unique_ptr<BlockProcessor> block_processor(
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BlockProcessor::Create(AudioProcessing::Config::EchoCanceller3(), rate,
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std::move(render_delay_buffer_mock)));
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std::vector<std::vector<float>> render_block(
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NumBandsForRate(rate), std::vector<float>(kBlockSize, 0.f));
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std::vector<std::vector<float>> capture_block(
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NumBandsForRate(rate), std::vector<float>(kBlockSize, 0.f));
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DelayBuffer<float> signal_delay_buffer(kDelayInSamples);
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for (size_t k = 0; k < kNumBlocks; ++k) {
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RandomizeSampleVector(&random_generator, render_block[0]);
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signal_delay_buffer.Delay(render_block[0], capture_block[0]);
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block_processor->BufferRender(render_block);
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block_processor->ProcessCapture(false, false, &capture_block);
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}
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}
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}
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// Verifies that BlockProcessor submodules are called in a proper manner.
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TEST(BlockProcessor, DISABLED_SubmoduleIntegration) {
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constexpr size_t kNumBlocks = 310;
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Random random_generator(42U);
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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<testing::StrictMock<webrtc::test::MockRenderDelayBuffer>>
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render_delay_buffer_mock(
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new StrictMock<webrtc::test::MockRenderDelayBuffer>(rate));
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std::unique_ptr<
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testing::StrictMock<webrtc::test::MockRenderDelayController>>
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render_delay_controller_mock(
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new StrictMock<webrtc::test::MockRenderDelayController>());
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std::unique_ptr<testing::StrictMock<webrtc::test::MockEchoRemover>>
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echo_remover_mock(new StrictMock<webrtc::test::MockEchoRemover>());
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EXPECT_CALL(*render_delay_buffer_mock, Insert(_))
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.Times(kNumBlocks - 1)
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.WillRepeatedly(Return(true));
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EXPECT_CALL(*render_delay_buffer_mock, IsBlockAvailable())
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.Times(kNumBlocks)
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.WillRepeatedly(Return(true));
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EXPECT_CALL(*render_delay_buffer_mock, UpdateBuffers()).Times(kNumBlocks);
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EXPECT_CALL(*render_delay_buffer_mock, SetDelay(9)).Times(AtLeast(1));
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EXPECT_CALL(*render_delay_buffer_mock, Delay())
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.Times(kNumBlocks)
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.WillRepeatedly(Return(0));
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EXPECT_CALL(*render_delay_controller_mock, GetDelay(_, _))
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.Times(kNumBlocks)
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.WillRepeatedly(Return(9));
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EXPECT_CALL(*render_delay_controller_mock, AlignmentHeadroomSamples())
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.Times(kNumBlocks);
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EXPECT_CALL(*echo_remover_mock, ProcessCapture(_, _, _, _, _))
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.Times(kNumBlocks);
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EXPECT_CALL(*echo_remover_mock, UpdateEchoLeakageStatus(_))
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.Times(kNumBlocks);
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std::unique_ptr<BlockProcessor> block_processor(BlockProcessor::Create(
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AudioProcessing::Config::EchoCanceller3(), rate,
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std::move(render_delay_buffer_mock),
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std::move(render_delay_controller_mock), std::move(echo_remover_mock)));
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std::vector<std::vector<float>> render_block(
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NumBandsForRate(rate), std::vector<float>(kBlockSize, 0.f));
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std::vector<std::vector<float>> capture_block(
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NumBandsForRate(rate), std::vector<float>(kBlockSize, 0.f));
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DelayBuffer<float> signal_delay_buffer(640);
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for (size_t k = 0; k < kNumBlocks; ++k) {
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RandomizeSampleVector(&random_generator, render_block[0]);
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signal_delay_buffer.Delay(render_block[0], capture_block[0]);
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block_processor->BufferRender(render_block);
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block_processor->ProcessCapture(false, false, &capture_block);
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block_processor->UpdateEchoLeakageStatus(false);
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}
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}
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}
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TEST(BlockProcessor, BasicSetupAndApiCalls) {
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for (auto rate : {8000, 16000, 32000, 48000}) {
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SCOPED_TRACE(ProduceDebugText(rate));
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RunBasicSetupAndApiCallTest(rate);
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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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TEST(BlockProcessor, VerifyRenderBlockSizeCheck) {
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for (auto rate : {8000, 16000, 32000, 48000}) {
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SCOPED_TRACE(ProduceDebugText(rate));
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RunRenderBlockSizeVerificationTest(rate);
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}
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}
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TEST(BlockProcessor, VerifyCaptureBlockSizeCheck) {
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for (auto rate : {8000, 16000, 32000, 48000}) {
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SCOPED_TRACE(ProduceDebugText(rate));
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RunCaptureBlockSizeVerificationTest(rate);
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}
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}
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TEST(BlockProcessor, VerifyRenderNumBandsCheck) {
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for (auto rate : {8000, 16000, 32000, 48000}) {
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SCOPED_TRACE(ProduceDebugText(rate));
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RunRenderNumBandsVerificationTest(rate);
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}
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}
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// TODO(peah): Verify the check for correct number of bands in the capture
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// signal.
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TEST(BlockProcessor, VerifyCaptureNumBandsCheck) {
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for (auto rate : {8000, 16000, 32000, 48000}) {
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SCOPED_TRACE(ProduceDebugText(rate));
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RunCaptureNumBandsVerificationTest(rate);
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}
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}
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// Verifiers that the verification for null ProcessCapture input works.
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TEST(BlockProcessor, NullProcessCaptureParameter) {
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EXPECT_DEATH(std::unique_ptr<BlockProcessor>(
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BlockProcessor::Create(
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AudioProcessing::Config::EchoCanceller3(), 8000))
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->ProcessCapture(false, false, nullptr),
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"");
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}
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// Verifies the check for correct sample rate.
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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(BlockProcessor, DISABLED_WrongSampleRate) {
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EXPECT_DEATH(std::unique_ptr<BlockProcessor>(BlockProcessor::Create(
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AudioProcessing::Config::EchoCanceller3(), 8001)),
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"");
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}
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#endif
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} // namespace webrtc
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