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260c788d77
This CL adds the missing support for multi-channel in the code that provides an optional and configurable delay to be added to the microphone signal. The CL also makes the creation of the delay object conditional on the need for that support (this is important since this adds a significant heap memory footprint) Bug: webrtc:11314,chromium:1045910 Change-Id: I92d577e31af830945fe9d5ca2032000aad4266be Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167525 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30392}
105 lines
3.5 KiB
C++
105 lines
3.5 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/block_delay_buffer.h"
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#include <string>
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#include "modules/audio_processing/aec3/aec3_common.h"
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#include "modules/audio_processing/audio_buffer.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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float SampleValue(size_t sample_index) {
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return sample_index % 32768;
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}
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// Populates the frame with linearly increasing sample values for each band.
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void PopulateInputFrame(size_t frame_length,
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size_t num_bands,
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size_t first_sample_index,
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float* const* frame) {
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for (size_t k = 0; k < num_bands; ++k) {
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for (size_t i = 0; i < frame_length; ++i) {
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frame[k][i] = SampleValue(first_sample_index + i);
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}
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}
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}
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std::string ProduceDebugText(int sample_rate_hz, size_t delay) {
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char log_stream_buffer[8 * 1024];
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rtc::SimpleStringBuilder ss(log_stream_buffer);
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ss << "Sample rate: " << sample_rate_hz;
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ss << ", Delay: " << delay;
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return ss.str();
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}
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} // namespace
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class BlockDelayBufferTest
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: public ::testing::Test,
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public ::testing::WithParamInterface<std::tuple<size_t, int, size_t>> {};
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INSTANTIATE_TEST_SUITE_P(
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ParameterCombinations,
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BlockDelayBufferTest,
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::testing::Combine(::testing::Values(0, 1, 27, 160, 4321, 7021),
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::testing::Values(16000, 32000, 48000),
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::testing::Values(1, 2, 4)));
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// Verifies that the correct signal delay is achived.
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TEST_P(BlockDelayBufferTest, CorrectDelayApplied) {
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const size_t delay = std::get<0>(GetParam());
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const int rate = std::get<1>(GetParam());
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const size_t num_channels = std::get<2>(GetParam());
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SCOPED_TRACE(ProduceDebugText(rate, delay));
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size_t num_bands = NumBandsForRate(rate);
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size_t subband_frame_length = 160;
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BlockDelayBuffer delay_buffer(num_channels, num_bands, subband_frame_length,
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delay);
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static constexpr size_t kNumFramesToProcess = 20;
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for (size_t frame_index = 0; frame_index < kNumFramesToProcess;
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++frame_index) {
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AudioBuffer audio_buffer(rate, num_channels, rate, num_channels, rate,
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num_channels);
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if (rate > 16000) {
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audio_buffer.SplitIntoFrequencyBands();
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}
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size_t first_sample_index = frame_index * subband_frame_length;
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for (size_t ch = 0; ch < num_channels; ++ch) {
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PopulateInputFrame(subband_frame_length, num_bands, first_sample_index,
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&audio_buffer.split_bands(ch)[0]);
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}
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delay_buffer.DelaySignal(&audio_buffer);
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for (size_t ch = 0; ch < num_channels; ++ch) {
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for (size_t band = 0; band < num_bands; ++band) {
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size_t sample_index = first_sample_index;
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for (size_t i = 0; i < subband_frame_length; ++i, ++sample_index) {
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if (sample_index < delay) {
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EXPECT_EQ(0.f, audio_buffer.split_bands(ch)[band][i]);
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} else {
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EXPECT_EQ(SampleValue(sample_index - delay),
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audio_buffer.split_bands(ch)[band][i]);
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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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} // namespace webrtc
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