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398689f581
This CL adds functionality for applying an optional fixed delay in AEC3 to the capture signal Bug: webrtc:9647 Change-Id: Id3b3f896bcf203e6611298dc804c3c80da9f1883 Reviewed-on: https://webrtc-review.googlesource.com/95142 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24399}
91 lines
3.1 KiB
C++
91 lines
3.1 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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// Verifies that the correct signal delay is achived.
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TEST(BlockDelayBuffer, CorrectDelayApplied) {
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for (size_t delay : {0, 1, 27, 160, 4321, 7021}) {
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for (auto rate : {8000, 16000, 32000, 48000}) {
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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 fullband_frame_length = rate / 100;
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size_t subband_frame_length = rate == 8000 ? 80 : 160;
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BlockDelayBuffer delay_buffer(num_bands, subband_frame_length, 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(fullband_frame_length, 1,
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fullband_frame_length, 1,
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fullband_frame_length);
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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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PopulateInputFrame(subband_frame_length, num_bands, first_sample_index,
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&audio_buffer.split_bands_f(0)[0]);
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delay_buffer.DelaySignal(&audio_buffer);
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for (size_t k = 0; k < num_bands; ++k) {
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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_f(0)[k][i]);
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} else {
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EXPECT_EQ(SampleValue(sample_index - delay),
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audio_buffer.split_bands_f(0)[k][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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}
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} // namespace webrtc
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