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
|
/*
* audio encoder psychoacoustic model
* Copyright (C) 2008 Konstantin Shishkov
*
* 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
*/
#include <string.h>
#include "avcodec.h"
#include "psymodel.h"
#include "iirfilter.h"
#include "libavutil/mem.h"
extern const FFPsyModel ff_aac_psy_model;
av_cold int ff_psy_init(FFPsyContext *ctx, AVCodecContext *avctx, int num_lens,
const uint8_t **bands, const int* num_bands,
int num_groups, const uint8_t *group_map)
{
int i, j, k = 0;
ctx->avctx = avctx;
ctx->ch = av_mallocz_array(sizeof(ctx->ch[0]), avctx->channels * 2);
ctx->group = av_mallocz_array(sizeof(ctx->group[0]), num_groups);
ctx->bands = av_malloc_array (sizeof(ctx->bands[0]), num_lens);
ctx->num_bands = av_malloc_array (sizeof(ctx->num_bands[0]), num_lens);
if (!ctx->ch || !ctx->group || !ctx->bands || !ctx->num_bands) {
ff_psy_end(ctx);
return AVERROR(ENOMEM);
}
memcpy(ctx->bands, bands, sizeof(ctx->bands[0]) * num_lens);
memcpy(ctx->num_bands, num_bands, sizeof(ctx->num_bands[0]) * num_lens);
/* assign channels to groups (with virtual channels for coupling) */
for (i = 0; i < num_groups; i++) {
/* NOTE: Add 1 to handle the AAC chan_config without modification.
* This has the side effect of allowing an array of 0s to map
* to one channel per group.
*/
ctx->group[i].num_ch = group_map[i] + 1;
for (j = 0; j < ctx->group[i].num_ch * 2; j++)
ctx->group[i].ch[j] = &ctx->ch[k++];
}
switch (ctx->avctx->codec_id) {
case AV_CODEC_ID_AAC:
ctx->model = &ff_aac_psy_model;
break;
}
if (ctx->model->init)
return ctx->model->init(ctx);
return 0;
}
FFPsyChannelGroup *ff_psy_find_group(FFPsyContext *ctx, int channel)
{
int i = 0, ch = 0;
while (ch <= channel)
ch += ctx->group[i++].num_ch;
return &ctx->group[i-1];
}
av_cold void ff_psy_end(FFPsyContext *ctx)
{
if (ctx->model && ctx->model->end)
ctx->model->end(ctx);
av_freep(&ctx->bands);
av_freep(&ctx->num_bands);
av_freep(&ctx->group);
av_freep(&ctx->ch);
}
typedef struct FFPsyPreprocessContext{
AVCodecContext *avctx;
float stereo_att;
struct FFIIRFilterCoeffs *fcoeffs;
struct FFIIRFilterState **fstate;
struct FFIIRFilterContext fiir;
}FFPsyPreprocessContext;
#define FILT_ORDER 4
av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *avctx)
{
FFPsyPreprocessContext *ctx;
int i;
float cutoff_coeff = 0;
ctx = av_mallocz(sizeof(FFPsyPreprocessContext));
if (!ctx)
return NULL;
ctx->avctx = avctx;
/* AAC has its own LP method */
if (avctx->codec_id != AV_CODEC_ID_AAC) {
if (avctx->cutoff > 0)
cutoff_coeff = 2.0 * avctx->cutoff / avctx->sample_rate;
if (cutoff_coeff && cutoff_coeff < 0.98)
ctx->fcoeffs = ff_iir_filter_init_coeffs(avctx, FF_FILTER_TYPE_BUTTERWORTH,
FF_FILTER_MODE_LOWPASS, FILT_ORDER,
cutoff_coeff, 0.0, 0.0);
if (ctx->fcoeffs) {
ctx->fstate = av_mallocz(sizeof(ctx->fstate[0]) * avctx->channels);
for (i = 0; i < avctx->channels; i++)
ctx->fstate[i] = ff_iir_filter_init_state(FILT_ORDER);
}
}
ff_iir_filter_init(&ctx->fiir);
return ctx;
}
void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx, float **audio, int channels)
{
int ch;
int frame_size = ctx->avctx->frame_size;
FFIIRFilterContext *iir = &ctx->fiir;
if (ctx->fstate) {
for (ch = 0; ch < channels; ch++)
iir->filter_flt(ctx->fcoeffs, ctx->fstate[ch], frame_size,
&audio[ch][frame_size], 1, &audio[ch][frame_size], 1);
}
}
av_cold void ff_psy_preprocess_end(struct FFPsyPreprocessContext *ctx)
{
int i;
ff_iir_filter_free_coeffsp(&ctx->fcoeffs);
if (ctx->fstate)
for (i = 0; i < ctx->avctx->channels; i++)
ff_iir_filter_free_statep(&ctx->fstate[i]);
av_freep(&ctx->fstate);
av_free(ctx);
}
|
