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1 | /* |
2 | * audio encoder psychoacoustic model | |
3 | * Copyright (C) 2008 Konstantin Shishkov | |
4 | * | |
5 | * This file is part of FFmpeg. | |
6 | * | |
7 | * FFmpeg is free software; you can redistribute it and/or | |
8 | * modify it under the terms of the GNU Lesser General Public | |
9 | * License as published by the Free Software Foundation; either | |
10 | * version 2.1 of the License, or (at your option) any later version. | |
11 | * | |
12 | * FFmpeg is distributed in the hope that it will be useful, | |
13 | * but WITHOUT ANY WARRANTY; without even the implied warranty of | |
14 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
15 | * Lesser General Public License for more details. | |
16 | * | |
17 | * You should have received a copy of the GNU Lesser General Public | |
18 | * License along with FFmpeg; if not, write to the Free Software | |
19 | * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA | |
20 | */ | |
21 | ||
22 | #include <string.h> | |
23 | ||
24 | #include "avcodec.h" | |
25 | #include "psymodel.h" | |
26 | #include "iirfilter.h" | |
27 | #include "libavutil/mem.h" | |
28 | ||
29 | extern const FFPsyModel ff_aac_psy_model; | |
30 | ||
31 | av_cold int ff_psy_init(FFPsyContext *ctx, AVCodecContext *avctx, int num_lens, | |
32 | const uint8_t **bands, const int* num_bands, | |
33 | int num_groups, const uint8_t *group_map) | |
34 | { | |
35 | int i, j, k = 0; | |
36 | ||
37 | ctx->avctx = avctx; | |
38 | ctx->ch = av_mallocz_array(sizeof(ctx->ch[0]), avctx->channels * 2); | |
39 | ctx->group = av_mallocz_array(sizeof(ctx->group[0]), num_groups); | |
40 | ctx->bands = av_malloc_array (sizeof(ctx->bands[0]), num_lens); | |
41 | ctx->num_bands = av_malloc_array (sizeof(ctx->num_bands[0]), num_lens); | |
42 | memcpy(ctx->bands, bands, sizeof(ctx->bands[0]) * num_lens); | |
43 | memcpy(ctx->num_bands, num_bands, sizeof(ctx->num_bands[0]) * num_lens); | |
44 | ||
45 | /* assign channels to groups (with virtual channels for coupling) */ | |
46 | for (i = 0; i < num_groups; i++) { | |
47 | /* NOTE: Add 1 to handle the AAC chan_config without modification. | |
48 | * This has the side effect of allowing an array of 0s to map | |
49 | * to one channel per group. | |
50 | */ | |
51 | ctx->group[i].num_ch = group_map[i] + 1; | |
52 | for (j = 0; j < ctx->group[i].num_ch * 2; j++) | |
53 | ctx->group[i].ch[j] = &ctx->ch[k++]; | |
54 | } | |
55 | ||
56 | switch (ctx->avctx->codec_id) { | |
57 | case AV_CODEC_ID_AAC: | |
58 | ctx->model = &ff_aac_psy_model; | |
59 | break; | |
60 | } | |
61 | if (ctx->model->init) | |
62 | return ctx->model->init(ctx); | |
63 | return 0; | |
64 | } | |
65 | ||
66 | FFPsyChannelGroup *ff_psy_find_group(FFPsyContext *ctx, int channel) | |
67 | { | |
68 | int i = 0, ch = 0; | |
69 | ||
70 | while (ch <= channel) | |
71 | ch += ctx->group[i++].num_ch; | |
72 | ||
73 | return &ctx->group[i-1]; | |
74 | } | |
75 | ||
76 | av_cold void ff_psy_end(FFPsyContext *ctx) | |
77 | { | |
78 | if (ctx->model && ctx->model->end) | |
79 | ctx->model->end(ctx); | |
80 | av_freep(&ctx->bands); | |
81 | av_freep(&ctx->num_bands); | |
82 | av_freep(&ctx->group); | |
83 | av_freep(&ctx->ch); | |
84 | } | |
85 | ||
86 | typedef struct FFPsyPreprocessContext{ | |
87 | AVCodecContext *avctx; | |
88 | float stereo_att; | |
89 | struct FFIIRFilterCoeffs *fcoeffs; | |
90 | struct FFIIRFilterState **fstate; | |
91 | struct FFIIRFilterContext fiir; | |
92 | }FFPsyPreprocessContext; | |
93 | ||
94 | #define FILT_ORDER 4 | |
95 | ||
96 | av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *avctx) | |
97 | { | |
98 | FFPsyPreprocessContext *ctx; | |
99 | int i; | |
100 | float cutoff_coeff = 0; | |
101 | ctx = av_mallocz(sizeof(FFPsyPreprocessContext)); | |
102 | ctx->avctx = avctx; | |
103 | ||
104 | if (avctx->cutoff > 0) | |
105 | cutoff_coeff = 2.0 * avctx->cutoff / avctx->sample_rate; | |
106 | ||
107 | if (!cutoff_coeff && avctx->codec_id == AV_CODEC_ID_AAC) | |
108 | cutoff_coeff = 2.0 * AAC_CUTOFF(avctx) / avctx->sample_rate; | |
109 | ||
110 | if (cutoff_coeff && cutoff_coeff < 0.98) | |
111 | ctx->fcoeffs = ff_iir_filter_init_coeffs(avctx, FF_FILTER_TYPE_BUTTERWORTH, | |
112 | FF_FILTER_MODE_LOWPASS, FILT_ORDER, | |
113 | cutoff_coeff, 0.0, 0.0); | |
114 | if (ctx->fcoeffs) { | |
115 | ctx->fstate = av_mallocz_array(sizeof(ctx->fstate[0]), avctx->channels); | |
116 | for (i = 0; i < avctx->channels; i++) | |
117 | ctx->fstate[i] = ff_iir_filter_init_state(FILT_ORDER); | |
118 | } | |
119 | ||
120 | ff_iir_filter_init(&ctx->fiir); | |
121 | ||
122 | return ctx; | |
123 | } | |
124 | ||
125 | void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx, float **audio, int channels) | |
126 | { | |
127 | int ch; | |
128 | int frame_size = ctx->avctx->frame_size; | |
129 | FFIIRFilterContext *iir = &ctx->fiir; | |
130 | ||
131 | if (ctx->fstate) { | |
132 | for (ch = 0; ch < channels; ch++) | |
133 | iir->filter_flt(ctx->fcoeffs, ctx->fstate[ch], frame_size, | |
134 | &audio[ch][frame_size], 1, &audio[ch][frame_size], 1); | |
135 | } | |
136 | } | |
137 | ||
138 | av_cold void ff_psy_preprocess_end(struct FFPsyPreprocessContext *ctx) | |
139 | { | |
140 | int i; | |
f6fa7814 | 141 | ff_iir_filter_free_coeffsp(&ctx->fcoeffs); |
2ba45a60 DM |
142 | if (ctx->fstate) |
143 | for (i = 0; i < ctx->avctx->channels; i++) | |
f6fa7814 | 144 | ff_iir_filter_free_statep(&ctx->fstate[i]); |
2ba45a60 DM |
145 | av_freep(&ctx->fstate); |
146 | av_free(ctx); | |
147 | } |