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https://github.com/mollyim/webrtc.git
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d3ead1a942
This change adds a Block class to reduce the need for std::vector<std::vector<std::vector<float>>>. This make the code easier to read and less error prone. It also enables future changes to the underlying data structure of a block. For instance, the data of all bands and channels could be stored in a single vector. The change has been verified to be bit-exact. Bug: webrtc:14089 Change-Id: Ied9a78124c0bbafe0e912017aef91f7c311de2ae Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262252 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36968}
334 lines
15 KiB
C++
334 lines
15 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/render_delay_controller.h"
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#include <algorithm>
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#include <memory>
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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/block_processor.h"
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#include "modules/audio_processing/aec3/decimator.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 "rtc_base/strings/string_builder.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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rtc::StringBuilder ss;
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ss << "Sample rate: " << sample_rate_hz;
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return ss.Release();
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}
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std::string ProduceDebugText(int sample_rate_hz,
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size_t delay,
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size_t num_render_channels,
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size_t num_capture_channels) {
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rtc::StringBuilder ss;
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ss << ProduceDebugText(sample_rate_hz) << ", Delay: " << delay
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<< ", Num render channels: " << num_render_channels
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<< ", Num capture channels: " << num_capture_channels;
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return ss.Release();
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}
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constexpr size_t kDownSamplingFactors[] = {2, 4, 8};
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} // namespace
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// Verifies the output of GetDelay when there are no AnalyzeRender calls.
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// TODO(bugs.webrtc.org/11161): Re-enable tests.
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TEST(RenderDelayController, DISABLED_NoRenderSignal) {
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for (size_t num_render_channels : {1, 2, 8}) {
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Block block(/*num_bands=1*/ 1, /*num_channels=*/1);
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EchoCanceller3Config config;
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for (size_t num_matched_filters = 4; num_matched_filters <= 10;
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num_matched_filters++) {
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for (auto down_sampling_factor : kDownSamplingFactors) {
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config.delay.down_sampling_factor = down_sampling_factor;
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config.delay.num_filters = num_matched_filters;
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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<RenderDelayBuffer> delay_buffer(
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RenderDelayBuffer::Create(config, rate, num_render_channels));
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std::unique_ptr<RenderDelayController> delay_controller(
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RenderDelayController::Create(config, rate,
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/*num_capture_channels*/ 1));
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for (size_t k = 0; k < 100; ++k) {
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auto delay = delay_controller->GetDelay(
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delay_buffer->GetDownsampledRenderBuffer(),
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delay_buffer->Delay(), block);
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EXPECT_FALSE(delay->delay);
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}
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}
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}
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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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// TODO(bugs.webrtc.org/11161): Re-enable tests.
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TEST(RenderDelayController, DISABLED_BasicApiCalls) {
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for (size_t num_capture_channels : {1, 2, 4}) {
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for (size_t num_render_channels : {1, 2, 8}) {
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Block capture_block(/*num_bands=*/1, num_capture_channels);
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absl::optional<DelayEstimate> delay_blocks;
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for (size_t num_matched_filters = 4; num_matched_filters <= 10;
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num_matched_filters++) {
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for (auto down_sampling_factor : kDownSamplingFactors) {
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EchoCanceller3Config config;
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config.delay.down_sampling_factor = down_sampling_factor;
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config.delay.num_filters = num_matched_filters;
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config.delay.capture_alignment_mixing.downmix = false;
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config.delay.capture_alignment_mixing.adaptive_selection = false;
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for (auto rate : {16000, 32000, 48000}) {
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Block render_block(NumBandsForRate(rate), num_render_channels);
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std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
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RenderDelayBuffer::Create(config, rate, num_render_channels));
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std::unique_ptr<RenderDelayController> delay_controller(
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RenderDelayController::Create(EchoCanceller3Config(), rate,
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num_capture_channels));
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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->PrepareCaptureProcessing();
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delay_blocks = delay_controller->GetDelay(
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render_delay_buffer->GetDownsampledRenderBuffer(),
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render_delay_buffer->Delay(), capture_block);
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}
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EXPECT_TRUE(delay_blocks);
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EXPECT_FALSE(delay_blocks->delay);
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}
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}
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}
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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.
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// TODO(bugs.webrtc.org/11161): Re-enable tests.
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TEST(RenderDelayController, DISABLED_Alignment) {
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Random random_generator(42U);
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for (size_t num_capture_channels : {1, 2, 4}) {
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Block capture_block(/*num_bands=*/1, num_capture_channels);
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for (size_t num_matched_filters = 4; num_matched_filters <= 10;
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num_matched_filters++) {
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for (auto down_sampling_factor : kDownSamplingFactors) {
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EchoCanceller3Config config;
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config.delay.down_sampling_factor = down_sampling_factor;
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config.delay.num_filters = num_matched_filters;
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config.delay.capture_alignment_mixing.downmix = false;
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config.delay.capture_alignment_mixing.adaptive_selection = false;
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for (size_t num_render_channels : {1, 2, 8}) {
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for (auto rate : {16000, 32000, 48000}) {
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Block render_block(NumBandsForRate(rate), num_render_channels);
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for (size_t delay_samples : {15, 50, 150, 200, 800, 4000}) {
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absl::optional<DelayEstimate> delay_blocks;
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SCOPED_TRACE(ProduceDebugText(rate, delay_samples,
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num_render_channels,
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num_capture_channels));
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std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
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RenderDelayBuffer::Create(config, rate, num_render_channels));
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std::unique_ptr<RenderDelayController> delay_controller(
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RenderDelayController::Create(config, rate,
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num_capture_channels));
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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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for (int band = 0; band < render_block.NumBands(); ++band) {
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for (int channel = 0; channel < render_block.NumChannels();
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++channel) {
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RandomizeSampleVector(&random_generator,
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render_block.View(band, channel));
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}
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}
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signal_delay_buffer.Delay(
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render_block.View(/*band=*/0, /*channel=*/0),
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capture_block.View(/*band=*/0, /*channel=*/0));
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render_delay_buffer->Insert(render_block);
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render_delay_buffer->PrepareCaptureProcessing();
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delay_blocks = delay_controller->GetDelay(
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render_delay_buffer->GetDownsampledRenderBuffer(),
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render_delay_buffer->Delay(), capture_block);
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}
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ASSERT_TRUE(!!delay_blocks);
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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->delay);
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}
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}
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}
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}
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}
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}
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}
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// Verifies that the RenderDelayController is able to properly handle noncausal
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// delays.
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// TODO(bugs.webrtc.org/11161): Re-enable tests.
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TEST(RenderDelayController, DISABLED_NonCausalAlignment) {
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Random random_generator(42U);
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for (size_t num_capture_channels : {1, 2, 4}) {
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for (size_t num_render_channels : {1, 2, 8}) {
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for (size_t num_matched_filters = 4; num_matched_filters <= 10;
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num_matched_filters++) {
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for (auto down_sampling_factor : kDownSamplingFactors) {
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EchoCanceller3Config config;
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config.delay.down_sampling_factor = down_sampling_factor;
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config.delay.num_filters = num_matched_filters;
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config.delay.capture_alignment_mixing.downmix = false;
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config.delay.capture_alignment_mixing.adaptive_selection = false;
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for (auto rate : {16000, 32000, 48000}) {
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Block render_block(NumBandsForRate(rate), num_render_channels);
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Block capture_block(NumBandsForRate(rate), num_capture_channels);
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for (int delay_samples : {-15, -50, -150, -200}) {
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absl::optional<DelayEstimate> delay_blocks;
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SCOPED_TRACE(ProduceDebugText(rate, -delay_samples,
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num_render_channels,
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num_capture_channels));
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std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
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RenderDelayBuffer::Create(config, rate, num_render_channels));
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std::unique_ptr<RenderDelayController> delay_controller(
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RenderDelayController::Create(EchoCanceller3Config(), rate,
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num_capture_channels));
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DelayBuffer<float> signal_delay_buffer(-delay_samples);
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for (int k = 0;
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k < (400 - delay_samples / static_cast<int>(kBlockSize));
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++k) {
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RandomizeSampleVector(
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&random_generator,
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capture_block.View(/*band=*/0, /*channel=*/0));
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signal_delay_buffer.Delay(
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capture_block.View(/*band=*/0, /*channel=*/0),
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render_block.View(/*band=*/0, /*channel=*/0));
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render_delay_buffer->Insert(render_block);
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render_delay_buffer->PrepareCaptureProcessing();
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delay_blocks = delay_controller->GetDelay(
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render_delay_buffer->GetDownsampledRenderBuffer(),
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render_delay_buffer->Delay(), capture_block);
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}
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ASSERT_FALSE(delay_blocks);
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}
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}
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}
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}
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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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// TODO(bugs.webrtc.org/11161): Re-enable tests.
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TEST(RenderDelayController, DISABLED_AlignmentWithJitter) {
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Random random_generator(42U);
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for (size_t num_capture_channels : {1, 2, 4}) {
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for (size_t num_render_channels : {1, 2, 8}) {
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Block capture_block(
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/*num_bands=*/1, num_capture_channels);
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for (size_t num_matched_filters = 4; num_matched_filters <= 10;
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num_matched_filters++) {
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for (auto down_sampling_factor : kDownSamplingFactors) {
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EchoCanceller3Config config;
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config.delay.down_sampling_factor = down_sampling_factor;
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config.delay.num_filters = num_matched_filters;
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config.delay.capture_alignment_mixing.downmix = false;
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config.delay.capture_alignment_mixing.adaptive_selection = false;
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for (auto rate : {16000, 32000, 48000}) {
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Block render_block(NumBandsForRate(rate), num_render_channels);
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for (size_t delay_samples : {15, 50, 300, 800}) {
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absl::optional<DelayEstimate> delay_blocks;
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SCOPED_TRACE(ProduceDebugText(rate, delay_samples,
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num_render_channels,
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num_capture_channels));
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std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
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RenderDelayBuffer::Create(config, rate, num_render_channels));
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std::unique_ptr<RenderDelayController> delay_controller(
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RenderDelayController::Create(config, rate,
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num_capture_channels));
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DelayBuffer<float> signal_delay_buffer(delay_samples);
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constexpr size_t kMaxTestJitterBlocks = 26;
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for (size_t j = 0; j < (1000 + delay_samples / kBlockSize) /
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kMaxTestJitterBlocks +
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1;
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++j) {
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std::vector<Block> capture_block_buffer;
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for (size_t k = 0; k < (kMaxTestJitterBlocks - 1); ++k) {
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RandomizeSampleVector(
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&random_generator,
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render_block.View(/*band=*/0, /*channel=*/0));
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signal_delay_buffer.Delay(
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render_block.View(/*band=*/0, /*channel=*/0),
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capture_block.View(/*band=*/0, /*channel=*/0));
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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 < (kMaxTestJitterBlocks - 1); ++k) {
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render_delay_buffer->PrepareCaptureProcessing();
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delay_blocks = delay_controller->GetDelay(
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render_delay_buffer->GetDownsampledRenderBuffer(),
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render_delay_buffer->Delay(), 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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ASSERT_TRUE(delay_blocks);
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EXPECT_EQ(expected_delay_blocks, delay_blocks->delay);
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}
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}
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}
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}
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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 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(RenderDelayControllerDeathTest, 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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EchoCanceller3Config config;
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std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
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RenderDelayBuffer::Create(config, rate, 1));
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EXPECT_DEATH(
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std::unique_ptr<RenderDelayController>(
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RenderDelayController::Create(EchoCanceller3Config(), rate, 1)),
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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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