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
|
/*
* Audio FIFO
* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Audio FIFO
*/
#include "avutil.h"
#include "audio_fifo.h"
#include "common.h"
#include "fifo.h"
#include "mem.h"
#include "samplefmt.h"
struct AVAudioFifo {
AVFifoBuffer **buf; /**< single buffer for interleaved, per-channel buffers for planar */
int nb_buffers; /**< number of buffers */
int nb_samples; /**< number of samples currently in the FIFO */
int allocated_samples; /**< current allocated size, in samples */
int channels; /**< number of channels */
enum AVSampleFormat sample_fmt; /**< sample format */
int sample_size; /**< size, in bytes, of one sample in a buffer */
};
void av_audio_fifo_free(AVAudioFifo *af)
{
if (af) {
if (af->buf) {
int i;
for (i = 0; i < af->nb_buffers; i++) {
if (af->buf[i])
av_fifo_free(af->buf[i]);
}
av_freep(&af->buf);
}
av_free(af);
}
}
AVAudioFifo *av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels,
int nb_samples)
{
AVAudioFifo *af;
int buf_size, i;
/* get channel buffer size (also validates parameters) */
if (av_samples_get_buffer_size(&buf_size, channels, nb_samples, sample_fmt, 1) < 0)
return NULL;
af = av_mallocz(sizeof(*af));
if (!af)
return NULL;
af->channels = channels;
af->sample_fmt = sample_fmt;
af->sample_size = buf_size / nb_samples;
af->nb_buffers = av_sample_fmt_is_planar(sample_fmt) ? channels : 1;
af->buf = av_mallocz_array(af->nb_buffers, sizeof(*af->buf));
if (!af->buf)
goto error;
for (i = 0; i < af->nb_buffers; i++) {
af->buf[i] = av_fifo_alloc(buf_size);
if (!af->buf[i])
goto error;
}
af->allocated_samples = nb_samples;
return af;
error:
av_audio_fifo_free(af);
return NULL;
}
int av_audio_fifo_realloc(AVAudioFifo *af, int nb_samples)
{
int i, ret, buf_size;
if ((ret = av_samples_get_buffer_size(&buf_size, af->channels, nb_samples,
af->sample_fmt, 1)) < 0)
return ret;
for (i = 0; i < af->nb_buffers; i++) {
if ((ret = av_fifo_realloc2(af->buf[i], buf_size)) < 0)
return ret;
}
af->allocated_samples = nb_samples;
return 0;
}
int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
{
int i, ret, size;
/* automatically reallocate buffers if needed */
if (av_audio_fifo_space(af) < nb_samples) {
int current_size = av_audio_fifo_size(af);
/* check for integer overflow in new size calculation */
if (INT_MAX / 2 - current_size < nb_samples)
return AVERROR(EINVAL);
/* reallocate buffers */
if ((ret = av_audio_fifo_realloc(af, 2 * (current_size + nb_samples))) < 0)
return ret;
}
size = nb_samples * af->sample_size;
for (i = 0; i < af->nb_buffers; i++) {
ret = av_fifo_generic_write(af->buf[i], data[i], size, NULL);
if (ret != size)
return AVERROR_BUG;
}
af->nb_samples += nb_samples;
return nb_samples;
}
int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples)
{
int i, ret, size;
if (nb_samples < 0)
return AVERROR(EINVAL);
nb_samples = FFMIN(nb_samples, af->nb_samples);
if (!nb_samples)
return 0;
size = nb_samples * af->sample_size;
for (i = 0; i < af->nb_buffers; i++) {
if ((ret = av_fifo_generic_read(af->buf[i], data[i], size, NULL)) < 0)
return AVERROR_BUG;
}
af->nb_samples -= nb_samples;
return nb_samples;
}
int av_audio_fifo_drain(AVAudioFifo *af, int nb_samples)
{
int i, size;
if (nb_samples < 0)
return AVERROR(EINVAL);
nb_samples = FFMIN(nb_samples, af->nb_samples);
if (nb_samples) {
size = nb_samples * af->sample_size;
for (i = 0; i < af->nb_buffers; i++)
av_fifo_drain(af->buf[i], size);
af->nb_samples -= nb_samples;
}
return 0;
}
void av_audio_fifo_reset(AVAudioFifo *af)
{
int i;
for (i = 0; i < af->nb_buffers; i++)
av_fifo_reset(af->buf[i]);
af->nb_samples = 0;
}
int av_audio_fifo_size(AVAudioFifo *af)
{
return af->nb_samples;
}
int av_audio_fifo_space(AVAudioFifo *af)
{
return af->allocated_samples - af->nb_samples;
}
|
