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
|
/*
* Copyright (c) 2016 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_framer.h"
#include <algorithm>
#include "modules/audio_processing/aec3/aec3_common.h"
#include "rtc_base/checks.h"
namespace webrtc {
BlockFramer::BlockFramer(size_t num_bands, size_t num_channels)
: num_bands_(num_bands),
num_channels_(num_channels),
buffer_(num_bands_,
std::vector<std::vector<float>>(
num_channels,
std::vector<float>(kBlockSize, 0.f))) {
RTC_DCHECK_LT(0, num_bands);
RTC_DCHECK_LT(0, num_channels);
}
BlockFramer::~BlockFramer() = default;
// All the constants are chosen so that the buffer is either empty or has enough
// samples for InsertBlockAndExtractSubFrame to produce a frame. In order to
// achieve this, the InsertBlockAndExtractSubFrame and InsertBlock methods need
// to be called in the correct order.
void BlockFramer::InsertBlock(const Block& block) {
RTC_DCHECK_EQ(num_bands_, block.NumBands());
RTC_DCHECK_EQ(num_channels_, block.NumChannels());
for (size_t band = 0; band < num_bands_; ++band) {
for (size_t channel = 0; channel < num_channels_; ++channel) {
RTC_DCHECK_EQ(0, buffer_[band][channel].size());
buffer_[band][channel].insert(buffer_[band][channel].begin(),
block.begin(band, channel),
block.end(band, channel));
}
}
}
void BlockFramer::InsertBlockAndExtractSubFrame(
const Block& block,
std::vector<std::vector<rtc::ArrayView<float>>>* sub_frame) {
RTC_DCHECK(sub_frame);
RTC_DCHECK_EQ(num_bands_, block.NumBands());
RTC_DCHECK_EQ(num_channels_, block.NumChannels());
RTC_DCHECK_EQ(num_bands_, sub_frame->size());
for (size_t band = 0; band < num_bands_; ++band) {
RTC_DCHECK_EQ(num_channels_, (*sub_frame)[0].size());
for (size_t channel = 0; channel < num_channels_; ++channel) {
RTC_DCHECK_LE(kSubFrameLength,
buffer_[band][channel].size() + kBlockSize);
RTC_DCHECK_GE(kBlockSize, buffer_[band][channel].size());
RTC_DCHECK_EQ(kSubFrameLength, (*sub_frame)[band][channel].size());
const int samples_to_frame =
kSubFrameLength - buffer_[band][channel].size();
std::copy(buffer_[band][channel].begin(), buffer_[band][channel].end(),
(*sub_frame)[band][channel].begin());
std::copy(
block.begin(band, channel),
block.begin(band, channel) + samples_to_frame,
(*sub_frame)[band][channel].begin() + buffer_[band][channel].size());
buffer_[band][channel].clear();
buffer_[band][channel].insert(
buffer_[band][channel].begin(),
block.begin(band, channel) + samples_to_frame,
block.end(band, channel));
}
}
}
} // namespace webrtc
|
