Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
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Mirko Bonadei
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/*
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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 "webrtc/modules/audio_processing/aec3/render_delay_controller.h"
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#include <algorithm>
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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 "webrtc/modules/audio_processing/aec3/aec3_common.h"
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#include "webrtc/modules/audio_processing/aec3/block_processor.h"
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#include "webrtc/modules/audio_processing/aec3/decimator_by_4.h"
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#include "webrtc/modules/audio_processing/aec3/render_delay_buffer.h"
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#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
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#include "webrtc/modules/audio_processing/test/echo_canceller_test_tools.h"
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#include "webrtc/rtc_base/random.h"
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#include "webrtc/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, size_t delay) {
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std::ostringstream ss;
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ss << ProduceDebugText(sample_rate_hz) << ", Delay: " << delay;
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return ss.str();
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}
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} // namespace
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// Verifies the output of GetDelay when there are no AnalyzeRender calls.
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TEST(RenderDelayController, NoRenderSignal) {
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std::vector<float> block(kBlockSize, 0.f);
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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<RenderDelayBuffer> delay_buffer(
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RenderDelayBuffer::Create(NumBandsForRate(rate)));
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std::unique_ptr<RenderDelayController> delay_controller(
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RenderDelayController::Create(AudioProcessing::Config::EchoCanceller3(),
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rate));
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for (size_t k = 0; k < 100; ++k) {
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EXPECT_EQ(kMinEchoPathDelayBlocks,
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delay_controller->GetDelay(
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delay_buffer->GetDownsampledRenderBuffer(), block));
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}
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}
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}
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// Verifies the basic API call sequence.
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TEST(RenderDelayController, BasicApiCalls) {
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std::vector<float> capture_block(kBlockSize, 0.f);
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size_t delay_blocks = 0;
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for (auto rate : {8000, 16000, 32000, 48000}) {
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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::unique_ptr<RenderDelayBuffer> render_delay_buffer(
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RenderDelayBuffer::Create(NumBandsForRate(rate)));
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std::unique_ptr<RenderDelayController> delay_controller(
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RenderDelayController::Create(AudioProcessing::Config::EchoCanceller3(),
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rate));
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for (size_t k = 0; k < 10; ++k) {
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render_delay_buffer->Insert(render_block);
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render_delay_buffer->UpdateBuffers();
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delay_blocks = delay_controller->GetDelay(
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render_delay_buffer->GetDownsampledRenderBuffer(), capture_block);
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}
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EXPECT_FALSE(delay_controller->AlignmentHeadroomSamples());
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EXPECT_EQ(kMinEchoPathDelayBlocks, delay_blocks);
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}
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}
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// Verifies that the RenderDelayController is able to align the signals for
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// simple timeshifts between the signals.
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TEST(RenderDelayController, Alignment) {
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Random random_generator(42U);
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std::vector<float> capture_block(kBlockSize, 0.f);
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size_t delay_blocks = 0;
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for (auto rate : {8000, 16000, 32000, 48000}) {
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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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for (size_t delay_samples : {15, 50, 150, 200, 800, 4000}) {
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SCOPED_TRACE(ProduceDebugText(rate, delay_samples));
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std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
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RenderDelayBuffer::Create(NumBandsForRate(rate)));
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std::unique_ptr<RenderDelayController> delay_controller(
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RenderDelayController::Create(
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AudioProcessing::Config::EchoCanceller3(), rate));
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DelayBuffer<float> signal_delay_buffer(delay_samples);
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for (size_t k = 0; k < (400 + delay_samples / kBlockSize); ++k) {
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RandomizeSampleVector(&random_generator, render_block[0]);
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signal_delay_buffer.Delay(render_block[0], capture_block);
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render_delay_buffer->Insert(render_block);
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render_delay_buffer->UpdateBuffers();
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delay_blocks = delay_controller->GetDelay(
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render_delay_buffer->GetDownsampledRenderBuffer(), capture_block);
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}
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constexpr int kDelayHeadroomBlocks = 1;
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size_t expected_delay_blocks =
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std::max(0, static_cast<int>(delay_samples / kBlockSize) -
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kDelayHeadroomBlocks);
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EXPECT_EQ(expected_delay_blocks, delay_blocks);
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const rtc::Optional<size_t> headroom_samples =
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delay_controller->AlignmentHeadroomSamples();
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ASSERT_TRUE(headroom_samples);
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EXPECT_NEAR(delay_samples - delay_blocks * kBlockSize, *headroom_samples,
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4);
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}
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}
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}
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// Verifies that the RenderDelayController is able to align the signals for
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// simple timeshifts between the signals when there is jitter in the API calls.
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TEST(RenderDelayController, AlignmentWithJitter) {
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Random random_generator(42U);
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std::vector<float> capture_block(kBlockSize, 0.f);
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for (auto rate : {8000, 16000, 32000, 48000}) {
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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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for (size_t delay_samples : {15, 50, 300, 800}) {
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size_t delay_blocks = 0;
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SCOPED_TRACE(ProduceDebugText(rate, delay_samples));
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std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
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RenderDelayBuffer::Create(NumBandsForRate(rate)));
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std::unique_ptr<RenderDelayController> delay_controller(
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RenderDelayController::Create(
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AudioProcessing::Config::EchoCanceller3(), rate));
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DelayBuffer<float> signal_delay_buffer(delay_samples);
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for (size_t j = 0;
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j <
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(1000 + delay_samples / kBlockSize) / kMaxApiCallsJitterBlocks + 1;
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++j) {
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std::vector<std::vector<float>> capture_block_buffer;
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for (size_t k = 0; k < (kMaxApiCallsJitterBlocks - 1); ++k) {
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RandomizeSampleVector(&random_generator, render_block[0]);
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signal_delay_buffer.Delay(render_block[0], capture_block);
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capture_block_buffer.push_back(capture_block);
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render_delay_buffer->Insert(render_block);
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}
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for (size_t k = 0; k < (kMaxApiCallsJitterBlocks - 1); ++k) {
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render_delay_buffer->UpdateBuffers();
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delay_blocks = delay_controller->GetDelay(
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render_delay_buffer->GetDownsampledRenderBuffer(),
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capture_block_buffer[k]);
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}
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}
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constexpr int kDelayHeadroomBlocks = 1;
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size_t expected_delay_blocks =
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std::max(0, static_cast<int>(delay_samples / kBlockSize) -
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kDelayHeadroomBlocks);
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if (expected_delay_blocks < 2) {
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expected_delay_blocks = 0;
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}
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EXPECT_EQ(expected_delay_blocks, delay_blocks);
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const rtc::Optional<size_t> headroom_samples =
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delay_controller->AlignmentHeadroomSamples();
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ASSERT_TRUE(headroom_samples);
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EXPECT_NEAR(delay_samples - delay_blocks * kBlockSize, *headroom_samples,
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4);
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}
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}
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}
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// Verifies the initial value for the AlignmentHeadroomSamples.
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TEST(RenderDelayController, InitialHeadroom) {
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std::vector<float> render_block(kBlockSize, 0.f);
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std::vector<float> capture_block(kBlockSize, 0.f);
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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<RenderDelayBuffer> render_delay_buffer(
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RenderDelayBuffer::Create(NumBandsForRate(rate)));
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std::unique_ptr<RenderDelayController> delay_controller(
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RenderDelayController::Create(AudioProcessing::Config::EchoCanceller3(),
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rate));
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EXPECT_FALSE(delay_controller->AlignmentHeadroomSamples());
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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 capture signal block size.
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TEST(RenderDelayController, WrongCaptureSize) {
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std::vector<float> block(kBlockSize - 1, 0.f);
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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<RenderDelayBuffer> render_delay_buffer(
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RenderDelayBuffer::Create(NumBandsForRate(rate)));
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EXPECT_DEATH(
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std::unique_ptr<RenderDelayController>(
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RenderDelayController::Create(
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AudioProcessing::Config::EchoCanceller3(), rate))
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->GetDelay(render_delay_buffer->GetDownsampledRenderBuffer(),
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block),
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"");
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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(RenderDelayController, DISABLED_WrongSampleRate) {
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for (auto rate : {-1, 0, 8001, 16001}) {
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SCOPED_TRACE(ProduceDebugText(rate));
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std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
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RenderDelayBuffer::Create(NumBandsForRate(rate)));
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EXPECT_DEATH(
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std::unique_ptr<RenderDelayController>(RenderDelayController::Create(
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AudioProcessing::Config::EchoCanceller3(), rate)),
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"");
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}
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}
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#endif
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} // namespace webrtc
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