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
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
|
/*
Copyright (C) 2013-2021 Carl Hetherington <cth@carlh.net>
This file is part of DCP-o-matic.
DCP-o-matic is free software; you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation; either version 2 of the License, or
(at your option) any later version.
DCP-o-matic is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with DCP-o-matic. If not, see <http://www.gnu.org/licenses/>.
*/
#include "resampler.h"
#include "audio_buffers.h"
#include "exceptions.h"
#include "compose.hpp"
#include "dcpomatic_assert.h"
#include <samplerate.h>
#include <iostream>
#include <cmath>
#include "i18n.h"
using std::cout;
using std::make_pair;
using std::make_shared;
using std::pair;
using std::runtime_error;
using std::shared_ptr;
/** @param in Input sampling rate (Hz)
* @param out Output sampling rate (Hz)
* @param channels Number of channels.
*/
Resampler::Resampler (int in, int out, int channels)
: _in_rate (in)
, _out_rate (out)
, _channels (channels)
{
int error;
_src = src_new (SRC_SINC_BEST_QUALITY, _channels, &error);
if (!_src) {
throw runtime_error (String::compose(N_("could not create sample-rate converter (%1)"), error));
}
}
Resampler::~Resampler ()
{
if (_src) {
src_delete (_src);
}
}
void
Resampler::set_fast ()
{
src_delete (_src);
_src = nullptr;
int error;
_src = src_new (SRC_LINEAR, _channels, &error);
if (!_src) {
throw runtime_error (String::compose(N_("could not create sample-rate converter (%1)"), error));
}
}
shared_ptr<const AudioBuffers>
Resampler::run (shared_ptr<const AudioBuffers> in)
{
DCPOMATIC_ASSERT(in->channels() == _channels);
int in_frames = in->frames ();
int in_offset = 0;
int out_offset = 0;
auto resampled = make_shared<AudioBuffers>(_channels, 0);
while (in_frames > 0) {
/* Compute the resampled frames count and add 32 for luck */
int const max_resampled_frames = ceil (static_cast<double>(in_frames) * _out_rate / _in_rate) + 32;
SRC_DATA data;
std::vector<float> in_buffer(in_frames * _channels);
std::vector<float> out_buffer(max_resampled_frames * _channels);
{
auto p = in->data ();
auto q = in_buffer.data();
for (int i = 0; i < in_frames; ++i) {
for (int j = 0; j < _channels; ++j) {
*q++ = p[j][in_offset + i];
}
}
}
data.data_in = in_buffer.data();
data.input_frames = in_frames;
data.data_out = out_buffer.data();
data.output_frames = max_resampled_frames;
data.end_of_input = 0;
data.src_ratio = double (_out_rate) / _in_rate;
int const r = src_process (_src, &data);
if (r) {
throw EncodeError (
String::compose (
N_("could not run sample-rate converter (%1) [processing %2 to %3, %4 channels]"),
src_strerror (r),
in_frames,
max_resampled_frames,
_channels
)
);
}
if (data.output_frames_gen == 0) {
break;
}
resampled->set_frames (out_offset + data.output_frames_gen);
{
auto p = data.data_out;
auto q = resampled->data ();
for (int i = 0; i < data.output_frames_gen; ++i) {
for (int j = 0; j < _channels; ++j) {
q[j][out_offset + i] = *p++;
}
}
}
in_frames -= data.input_frames_used;
in_offset += data.input_frames_used;
out_offset += data.output_frames_gen;
}
return resampled;
}
shared_ptr<const AudioBuffers>
Resampler::flush ()
{
auto out = make_shared<AudioBuffers>(_channels, 0);
int out_offset = 0;
int64_t const output_size = 65536;
float dummy[1];
std::vector<float> buffer(output_size);
SRC_DATA data;
data.data_in = dummy;
data.input_frames = 0;
data.data_out = buffer.data();
data.output_frames = output_size;
data.end_of_input = 1;
data.src_ratio = double (_out_rate) / _in_rate;
int const r = src_process (_src, &data);
if (r) {
throw EncodeError (String::compose(N_("could not run sample-rate converter (%1)"), src_strerror(r)));
}
out->set_frames (out_offset + data.output_frames_gen);
auto p = data.data_out;
auto q = out->data ();
for (int i = 0; i < data.output_frames_gen; ++i) {
for (int j = 0; j < _channels; ++j) {
q[j][out_offset + i] = *p++;
}
}
out_offset += data.output_frames_gen;
return out;
}
void
Resampler::reset ()
{
src_reset (_src);
}
|
