/* * 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 #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 // Verifies that the correct signal delay is achived. TEST(BlockDelayBuffer, CorrectDelayApplied) { for (size_t delay : {0, 1, 27, 160, 4321, 7021}) { for (auto rate : {8000, 16000, 32000, 48000}) { SCOPED_TRACE(ProduceDebugText(rate, delay)); size_t num_bands = NumBandsForRate(rate); size_t fullband_frame_length = rate / 100; size_t subband_frame_length = rate == 8000 ? 80 : 160; BlockDelayBuffer delay_buffer(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(fullband_frame_length, 1, fullband_frame_length, 1, fullband_frame_length); if (rate > 16000) { audio_buffer.SplitIntoFrequencyBands(); } size_t first_sample_index = frame_index * subband_frame_length; PopulateInputFrame(subband_frame_length, num_bands, first_sample_index, &audio_buffer.split_bands_f(0)[0]); delay_buffer.DelaySignal(&audio_buffer); for (size_t k = 0; k < num_bands; ++k) { 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_f(0)[k][i]); } else { EXPECT_EQ(SampleValue(sample_index - delay), audio_buffer.split_bands_f(0)[k][i]); } } } } } } } } // namespace webrtc