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/*
* 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
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