block_delay_buffer_unittest.cc (3591B)
1 /* 2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #include "modules/audio_processing/aec3/block_delay_buffer.h" 12 13 #include <cstddef> 14 #include <string> 15 #include <tuple> 16 17 #include "modules/audio_processing/aec3/aec3_common.h" 18 #include "modules/audio_processing/audio_buffer.h" 19 #include "rtc_base/strings/string_builder.h" 20 #include "test/gtest.h" 21 22 namespace webrtc { 23 24 namespace { 25 26 float SampleValue(size_t sample_index) { 27 return sample_index % 32768; 28 } 29 30 // Populates the frame with linearly increasing sample values for each band. 31 void PopulateInputFrame(size_t frame_length, 32 size_t num_bands, 33 size_t first_sample_index, 34 float* const* frame) { 35 for (size_t k = 0; k < num_bands; ++k) { 36 for (size_t i = 0; i < frame_length; ++i) { 37 frame[k][i] = SampleValue(first_sample_index + i); 38 } 39 } 40 } 41 42 std::string ProduceDebugText(int sample_rate_hz, size_t delay) { 43 char log_stream_buffer[8 * 1024]; 44 SimpleStringBuilder ss(log_stream_buffer); 45 ss << "Sample rate: " << sample_rate_hz; 46 ss << ", Delay: " << delay; 47 return ss.str(); 48 } 49 50 } // namespace 51 52 class BlockDelayBufferTest 53 : public ::testing::Test, 54 public ::testing::WithParamInterface<std::tuple<size_t, int, size_t>> {}; 55 56 INSTANTIATE_TEST_SUITE_P( 57 ParameterCombinations, 58 BlockDelayBufferTest, 59 ::testing::Combine(::testing::Values(0, 1, 27, 160, 4321, 7021), 60 ::testing::Values(16000, 32000, 48000), 61 ::testing::Values(1, 2, 4))); 62 63 // Verifies that the correct signal delay is achived. 64 TEST_P(BlockDelayBufferTest, CorrectDelayApplied) { 65 const size_t delay = std::get<0>(GetParam()); 66 const int rate = std::get<1>(GetParam()); 67 const size_t num_channels = std::get<2>(GetParam()); 68 69 SCOPED_TRACE(ProduceDebugText(rate, delay)); 70 size_t num_bands = NumBandsForRate(rate); 71 size_t subband_frame_length = 160; 72 73 BlockDelayBuffer delay_buffer(num_channels, num_bands, subband_frame_length, 74 delay); 75 76 static constexpr size_t kNumFramesToProcess = 20; 77 for (size_t frame_index = 0; frame_index < kNumFramesToProcess; 78 ++frame_index) { 79 AudioBuffer audio_buffer(rate, num_channels, rate, num_channels, rate, 80 num_channels); 81 if (rate > 16000) { 82 audio_buffer.SplitIntoFrequencyBands(); 83 } 84 size_t first_sample_index = frame_index * subband_frame_length; 85 for (size_t ch = 0; ch < num_channels; ++ch) { 86 PopulateInputFrame(subband_frame_length, num_bands, first_sample_index, 87 &audio_buffer.split_bands(ch)[0]); 88 } 89 delay_buffer.DelaySignal(&audio_buffer); 90 91 for (size_t ch = 0; ch < num_channels; ++ch) { 92 for (size_t band = 0; band < num_bands; ++band) { 93 size_t sample_index = first_sample_index; 94 for (size_t i = 0; i < subband_frame_length; ++i, ++sample_index) { 95 if (sample_index < delay) { 96 EXPECT_EQ(0.f, audio_buffer.split_bands(ch)[band][i]); 97 } else { 98 EXPECT_EQ(SampleValue(sample_index - delay), 99 audio_buffer.split_bands(ch)[band][i]); 100 } 101 } 102 } 103 } 104 } 105 } 106 107 } // namespace webrtc