diff options
author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
---|---|---|
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/block_delay_buffer_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/block_delay_buffer_unittest.cc')
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/aec3/block_delay_buffer_unittest.cc | 105 |
1 files changed, 105 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/block_delay_buffer_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/block_delay_buffer_unittest.cc new file mode 100644 index 0000000000..011ab49651 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/aec3/block_delay_buffer_unittest.cc @@ -0,0 +1,105 @@ +/* + * 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/block_delay_buffer.h" + +#include <string> + +#include "modules/audio_processing/aec3/aec3_common.h" +#include "modules/audio_processing/audio_buffer.h" +#include "rtc_base/strings/string_builder.h" +#include "test/gtest.h" + +namespace webrtc { + +namespace { + +float SampleValue(size_t sample_index) { + return sample_index % 32768; +} + +// Populates the frame with linearly increasing sample values for each band. +void PopulateInputFrame(size_t frame_length, + size_t num_bands, + size_t first_sample_index, + float* const* frame) { + for (size_t k = 0; k < num_bands; ++k) { + for (size_t i = 0; i < frame_length; ++i) { + frame[k][i] = SampleValue(first_sample_index + i); + } + } +} + +std::string ProduceDebugText(int sample_rate_hz, size_t delay) { + char log_stream_buffer[8 * 1024]; + rtc::SimpleStringBuilder ss(log_stream_buffer); + ss << "Sample rate: " << sample_rate_hz; + ss << ", Delay: " << delay; + return ss.str(); +} + +} // namespace + +class BlockDelayBufferTest + : public ::testing::Test, + public ::testing::WithParamInterface<std::tuple<size_t, int, size_t>> {}; + +INSTANTIATE_TEST_SUITE_P( + ParameterCombinations, + BlockDelayBufferTest, + ::testing::Combine(::testing::Values(0, 1, 27, 160, 4321, 7021), + ::testing::Values(16000, 32000, 48000), + ::testing::Values(1, 2, 4))); + +// Verifies that the correct signal delay is achived. +TEST_P(BlockDelayBufferTest, CorrectDelayApplied) { + const size_t delay = std::get<0>(GetParam()); + const int rate = std::get<1>(GetParam()); + const size_t num_channels = std::get<2>(GetParam()); + + SCOPED_TRACE(ProduceDebugText(rate, delay)); + size_t num_bands = NumBandsForRate(rate); + size_t subband_frame_length = 160; + + BlockDelayBuffer delay_buffer(num_channels, num_bands, subband_frame_length, + delay); + + static constexpr size_t kNumFramesToProcess = 20; + for (size_t frame_index = 0; frame_index < kNumFramesToProcess; + ++frame_index) { + AudioBuffer audio_buffer(rate, num_channels, rate, num_channels, rate, + num_channels); + if (rate > 16000) { + audio_buffer.SplitIntoFrequencyBands(); + } + size_t first_sample_index = frame_index * subband_frame_length; + for (size_t ch = 0; ch < num_channels; ++ch) { + PopulateInputFrame(subband_frame_length, num_bands, first_sample_index, + &audio_buffer.split_bands(ch)[0]); + } + delay_buffer.DelaySignal(&audio_buffer); + + for (size_t ch = 0; ch < num_channels; ++ch) { + for (size_t band = 0; band < num_bands; ++band) { + size_t sample_index = first_sample_index; + for (size_t i = 0; i < subband_frame_length; ++i, ++sample_index) { + if (sample_index < delay) { + EXPECT_EQ(0.f, audio_buffer.split_bands(ch)[band][i]); + } else { + EXPECT_EQ(SampleValue(sample_index - delay), + audio_buffer.split_bands(ch)[band][i]); + } + } + } + } + } +} + +} // namespace webrtc |