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
|
/*
* Interface to libaacplus for aac+ (sbr+ps) encoding
* Copyright (c) 2010 tipok <piratfm@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
* Interface to libaacplus for aac+ (sbr+ps) encoding.
*/
#include "avcodec.h"
#include <aacplus.h>
typedef struct aacPlusAudioContext {
aacplusEncHandle aacplus_handle;
} aacPlusAudioContext;
static av_cold int aacPlus_encode_init(AVCodecContext *avctx)
{
aacPlusAudioContext *s = avctx->priv_data;
aacplusEncConfiguration *aacplus_cfg;
unsigned long samples_input, max_bytes_output;
/* number of channels */
if (avctx->channels < 1 || avctx->channels > 2) {
av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed\n", avctx->channels);
return -1;
}
s->aacplus_handle = aacplusEncOpen(avctx->sample_rate,
avctx->channels,
&samples_input, &max_bytes_output);
if(!s->aacplus_handle) {
av_log(avctx, AV_LOG_ERROR, "can't open encoder\n");
return -1;
}
/* check aacplus version */
aacplus_cfg = aacplusEncGetCurrentConfiguration(s->aacplus_handle);
/* put the options in the configuration struct */
if(avctx->profile != FF_PROFILE_AAC_LOW && avctx->profile != FF_PROFILE_UNKNOWN) {
av_log(avctx, AV_LOG_ERROR, "invalid AAC profile: %d, only LC supported\n", avctx->profile);
aacplusEncClose(s->aacplus_handle);
return -1;
}
aacplus_cfg->bitRate = avctx->bit_rate;
aacplus_cfg->bandWidth = avctx->cutoff;
aacplus_cfg->outputFormat = !(avctx->flags & CODEC_FLAG_GLOBAL_HEADER);
aacplus_cfg->inputFormat = AACPLUS_INPUT_16BIT;
if (!aacplusEncSetConfiguration(s->aacplus_handle, aacplus_cfg)) {
av_log(avctx, AV_LOG_ERROR, "libaacplus doesn't support this output format!\n");
return -1;
}
avctx->frame_size = samples_input / avctx->channels;
avctx->coded_frame= avcodec_alloc_frame();
avctx->coded_frame->key_frame= 1;
/* Set decoder specific info */
avctx->extradata_size = 0;
if (avctx->flags & CODEC_FLAG_GLOBAL_HEADER) {
unsigned char *buffer = NULL;
unsigned long decoder_specific_info_size;
if (aacplusEncGetDecoderSpecificInfo(s->aacplus_handle, &buffer,
&decoder_specific_info_size) == 1) {
avctx->extradata = av_malloc(decoder_specific_info_size + FF_INPUT_BUFFER_PADDING_SIZE);
avctx->extradata_size = decoder_specific_info_size;
memcpy(avctx->extradata, buffer, avctx->extradata_size);
}
#undef free
free(buffer);
#define free please_use_av_free
}
return 0;
}
static int aacPlus_encode_frame(AVCodecContext *avctx,
unsigned char *frame, int buf_size, void *data)
{
aacPlusAudioContext *s = avctx->priv_data;
int bytes_written;
bytes_written = aacplusEncEncode(s->aacplus_handle,
data,
avctx->frame_size * avctx->channels,
frame,
buf_size);
return bytes_written;
}
static av_cold int aacPlus_encode_close(AVCodecContext *avctx)
{
aacPlusAudioContext *s = avctx->priv_data;
av_freep(&avctx->coded_frame);
av_freep(&avctx->extradata);
aacplusEncClose(s->aacplus_handle);
return 0;
}
AVCodec ff_libaacplus_encoder = {
.name = "libaacplus",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_AAC,
.priv_data_size = sizeof(aacPlusAudioContext),
.init = aacPlus_encode_init,
.encode = aacPlus_encode_frame,
.close = aacPlus_encode_close,
.sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("libaacplus AAC+ (Advanced Audio Codec with SBR+PS)"),
};
|
