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
|
/*
* Copyright (c) 2015 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/test/audio_buffer_tools.h"
#include <cstring>
#include <vector>
#include "api/array_view.h"
#include "api/audio/audio_processing.h"
#include "modules/audio_processing/audio_buffer.h"
#include "rtc_base/checks.h"
namespace webrtc {
namespace test {
void SetupFrame(const StreamConfig& stream_config,
std::vector<float*>* frame,
std::vector<float>* frame_samples) {
frame_samples->resize(stream_config.num_channels() *
stream_config.num_frames());
frame->resize(stream_config.num_channels());
for (size_t ch = 0; ch < stream_config.num_channels(); ++ch) {
(*frame)[ch] = &(*frame_samples)[ch * stream_config.num_frames()];
}
}
void CopyVectorToAudioBuffer(const StreamConfig& stream_config,
ArrayView<const float> source,
AudioBuffer* destination) {
std::vector<float*> input;
std::vector<float> input_samples;
SetupFrame(stream_config, &input, &input_samples);
RTC_CHECK_EQ(input_samples.size(), source.size());
memcpy(input_samples.data(), source.data(),
source.size() * sizeof(source[0]));
destination->CopyFrom(&input[0], stream_config);
}
void ExtractVectorFromAudioBuffer(const StreamConfig& stream_config,
AudioBuffer* source,
std::vector<float>* destination) {
std::vector<float*> output;
SetupFrame(stream_config, &output, destination);
source->CopyTo(stream_config, &output[0]);
}
void FillBuffer(float value, AudioBuffer& audio_buffer) {
for (size_t ch = 0; ch < audio_buffer.num_channels(); ++ch) {
FillBufferChannel(value, ch, audio_buffer);
}
}
void FillBufferChannel(float value, int channel, AudioBuffer& audio_buffer) {
RTC_CHECK_LT(channel, audio_buffer.num_channels());
for (size_t i = 0; i < audio_buffer.num_frames(); ++i) {
audio_buffer.channels()[channel][i] = value;
}
}
} // namespace test
} // namespace webrtc
|
