1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
|
/*
* 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
// 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
|
