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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/modules/audio_processing/aec3/echo_path_delay_estimator_unittest.cc | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/aec3/echo_path_delay_estimator_unittest.cc')
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/aec3/echo_path_delay_estimator_unittest.cc | 184 |
1 files changed, 184 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/echo_path_delay_estimator_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/echo_path_delay_estimator_unittest.cc new file mode 100644 index 0000000000..e2c101fb04 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/aec3/echo_path_delay_estimator_unittest.cc @@ -0,0 +1,184 @@ +/* + * 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/echo_path_delay_estimator.h" + +#include <algorithm> +#include <string> + +#include "api/audio/echo_canceller3_config.h" +#include "modules/audio_processing/aec3/aec3_common.h" +#include "modules/audio_processing/aec3/render_delay_buffer.h" +#include "modules/audio_processing/logging/apm_data_dumper.h" +#include "modules/audio_processing/test/echo_canceller_test_tools.h" +#include "rtc_base/random.h" +#include "rtc_base/strings/string_builder.h" +#include "test/gtest.h" + +namespace webrtc { +namespace { + +std::string ProduceDebugText(size_t delay, size_t down_sampling_factor) { + rtc::StringBuilder ss; + ss << "Delay: " << delay; + ss << ", Down sampling factor: " << down_sampling_factor; + return ss.Release(); +} + +} // namespace + +class EchoPathDelayEstimatorMultiChannel + : public ::testing::Test, + public ::testing::WithParamInterface<std::tuple<size_t, size_t>> {}; + +INSTANTIATE_TEST_SUITE_P(MultiChannel, + EchoPathDelayEstimatorMultiChannel, + ::testing::Combine(::testing::Values(1, 2, 3, 6, 8), + ::testing::Values(1, 2, 4))); + +// Verifies that the basic API calls work. +TEST_P(EchoPathDelayEstimatorMultiChannel, BasicApiCalls) { + const size_t num_render_channels = std::get<0>(GetParam()); + const size_t num_capture_channels = std::get<1>(GetParam()); + constexpr int kSampleRateHz = 48000; + constexpr size_t kNumBands = NumBandsForRate(kSampleRateHz); + ApmDataDumper data_dumper(0); + EchoCanceller3Config config; + std::unique_ptr<RenderDelayBuffer> render_delay_buffer( + RenderDelayBuffer::Create(config, kSampleRateHz, num_render_channels)); + EchoPathDelayEstimator estimator(&data_dumper, config, num_capture_channels); + Block render(kNumBands, num_render_channels); + Block capture(/*num_bands=*/1, num_capture_channels); + for (size_t k = 0; k < 100; ++k) { + render_delay_buffer->Insert(render); + estimator.EstimateDelay(render_delay_buffer->GetDownsampledRenderBuffer(), + capture); + } +} + +// Verifies that the delay estimator produces correct delay for artificially +// delayed signals. +TEST(EchoPathDelayEstimator, DelayEstimation) { + constexpr size_t kNumRenderChannels = 1; + constexpr size_t kNumCaptureChannels = 1; + constexpr int kSampleRateHz = 48000; + constexpr size_t kNumBands = NumBandsForRate(kSampleRateHz); + Random random_generator(42U); + Block render(kNumBands, kNumRenderChannels); + Block capture(/*num_bands=*/1, kNumCaptureChannels); + ApmDataDumper data_dumper(0); + constexpr size_t kDownSamplingFactors[] = {2, 4, 8}; + for (auto down_sampling_factor : kDownSamplingFactors) { + EchoCanceller3Config config; + config.delay.delay_headroom_samples = 0; + config.delay.down_sampling_factor = down_sampling_factor; + config.delay.num_filters = 10; + for (size_t delay_samples : {30, 64, 150, 200, 800, 4000}) { + SCOPED_TRACE(ProduceDebugText(delay_samples, down_sampling_factor)); + std::unique_ptr<RenderDelayBuffer> render_delay_buffer( + RenderDelayBuffer::Create(config, kSampleRateHz, kNumRenderChannels)); + DelayBuffer<float> signal_delay_buffer(delay_samples); + EchoPathDelayEstimator estimator(&data_dumper, config, + kNumCaptureChannels); + + absl::optional<DelayEstimate> estimated_delay_samples; + for (size_t k = 0; k < (500 + (delay_samples) / kBlockSize); ++k) { + RandomizeSampleVector(&random_generator, + render.View(/*band=*/0, /*channel=*/0)); + signal_delay_buffer.Delay(render.View(/*band=*/0, /*channel=*/0), + capture.View(/*band=*/0, /*channel=*/0)); + render_delay_buffer->Insert(render); + + if (k == 0) { + render_delay_buffer->Reset(); + } + + render_delay_buffer->PrepareCaptureProcessing(); + + auto estimate = estimator.EstimateDelay( + render_delay_buffer->GetDownsampledRenderBuffer(), capture); + + if (estimate) { + estimated_delay_samples = estimate; + } + } + + if (estimated_delay_samples) { + // Allow estimated delay to be off by a block as internally the delay is + // quantized with an error up to a block. + size_t delay_ds = delay_samples / down_sampling_factor; + size_t estimated_delay_ds = + estimated_delay_samples->delay / down_sampling_factor; + EXPECT_NEAR(delay_ds, estimated_delay_ds, + kBlockSize / down_sampling_factor); + } else { + ADD_FAILURE(); + } + } + } +} + +// Verifies that the delay estimator does not produce delay estimates for render +// signals of low level. +TEST(EchoPathDelayEstimator, NoDelayEstimatesForLowLevelRenderSignals) { + constexpr size_t kNumRenderChannels = 1; + constexpr size_t kNumCaptureChannels = 1; + constexpr int kSampleRateHz = 48000; + constexpr size_t kNumBands = NumBandsForRate(kSampleRateHz); + Random random_generator(42U); + EchoCanceller3Config config; + Block render(kNumBands, kNumRenderChannels); + Block capture(/*num_bands=*/1, kNumCaptureChannels); + ApmDataDumper data_dumper(0); + EchoPathDelayEstimator estimator(&data_dumper, config, kNumCaptureChannels); + std::unique_ptr<RenderDelayBuffer> render_delay_buffer( + RenderDelayBuffer::Create(EchoCanceller3Config(), kSampleRateHz, + kNumRenderChannels)); + for (size_t k = 0; k < 100; ++k) { + RandomizeSampleVector(&random_generator, + render.View(/*band=*/0, /*channel=*/0)); + for (auto& render_k : render.View(/*band=*/0, /*channel=*/0)) { + render_k *= 100.f / 32767.f; + } + std::copy(render.begin(/*band=*/0, /*channel=*/0), + render.end(/*band=*/0, /*channel=*/0), + capture.begin(/*band*/ 0, /*channel=*/0)); + render_delay_buffer->Insert(render); + render_delay_buffer->PrepareCaptureProcessing(); + EXPECT_FALSE(estimator.EstimateDelay( + render_delay_buffer->GetDownsampledRenderBuffer(), capture)); + } +} + +#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) + +// Verifies the check for the render blocksize. +// TODO(peah): Re-enable the test once the issue with memory leaks during DEATH +// tests on test bots has been fixed. +TEST(EchoPathDelayEstimatorDeathTest, DISABLED_WrongRenderBlockSize) { + ApmDataDumper data_dumper(0); + EchoCanceller3Config config; + EchoPathDelayEstimator estimator(&data_dumper, config, 1); + std::unique_ptr<RenderDelayBuffer> render_delay_buffer( + RenderDelayBuffer::Create(config, 48000, 1)); + Block capture(/*num_bands=*/1, /*num_channels=*/1); + EXPECT_DEATH(estimator.EstimateDelay( + render_delay_buffer->GetDownsampledRenderBuffer(), capture), + ""); +} + +// Verifies the check for non-null data dumper. +TEST(EchoPathDelayEstimatorDeathTest, NullDataDumper) { + EXPECT_DEATH(EchoPathDelayEstimator(nullptr, EchoCanceller3Config(), 1), ""); +} + +#endif + +} // namespace webrtc |