2 * audio encoder psychoacoustic model
3 * Copyright (C) 2008 Konstantin Shishkov
5 * This file is part of FFmpeg.
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.
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.
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
26 #include "iirfilter.h"
27 #include "libavutil/mem.h"
29 extern const FFPsyModel ff_aac_psy_model;
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)
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);
43 if (!ctx->ch || !ctx->group || !ctx->bands || !ctx->num_bands) {
45 return AVERROR(ENOMEM);
48 memcpy(ctx->bands, bands, sizeof(ctx->bands[0]) * num_lens);
49 memcpy(ctx->num_bands, num_bands, sizeof(ctx->num_bands[0]) * num_lens);
51 /* assign channels to groups (with virtual channels for coupling) */
52 for (i = 0; i < num_groups; i++) {
53 /* NOTE: Add 1 to handle the AAC chan_config without modification.
54 * This has the side effect of allowing an array of 0s to map
55 * to one channel per group.
57 ctx->group[i].num_ch = group_map[i] + 1;
58 for (j = 0; j < ctx->group[i].num_ch * 2; j++)
59 ctx->group[i].ch[j] = &ctx->ch[k++];
62 switch (ctx->avctx->codec_id) {
64 ctx->model = &ff_aac_psy_model;
68 return ctx->model->init(ctx);
72 FFPsyChannelGroup *ff_psy_find_group(FFPsyContext *ctx, int channel)
77 ch += ctx->group[i++].num_ch;
79 return &ctx->group[i-1];
82 av_cold void ff_psy_end(FFPsyContext *ctx)
84 if (ctx->model && ctx->model->end)
86 av_freep(&ctx->bands);
87 av_freep(&ctx->num_bands);
88 av_freep(&ctx->group);
92 typedef struct FFPsyPreprocessContext{
93 AVCodecContext *avctx;
95 struct FFIIRFilterCoeffs *fcoeffs;
96 struct FFIIRFilterState **fstate;
97 struct FFIIRFilterContext fiir;
98 }FFPsyPreprocessContext;
102 av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *avctx)
104 FFPsyPreprocessContext *ctx;
106 float cutoff_coeff = 0;
107 ctx = av_mallocz(sizeof(FFPsyPreprocessContext));
112 /* AAC has its own LP method */
113 if (avctx->codec_id != AV_CODEC_ID_AAC) {
114 if (avctx->cutoff > 0)
115 cutoff_coeff = 2.0 * avctx->cutoff / avctx->sample_rate;
117 if (cutoff_coeff && cutoff_coeff < 0.98)
118 ctx->fcoeffs = ff_iir_filter_init_coeffs(avctx, FF_FILTER_TYPE_BUTTERWORTH,
119 FF_FILTER_MODE_LOWPASS, FILT_ORDER,
120 cutoff_coeff, 0.0, 0.0);
122 ctx->fstate = av_mallocz(sizeof(ctx->fstate[0]) * avctx->channels);
123 for (i = 0; i < avctx->channels; i++)
124 ctx->fstate[i] = ff_iir_filter_init_state(FILT_ORDER);
128 ff_iir_filter_init(&ctx->fiir);
133 void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx, float **audio, int channels)
136 int frame_size = ctx->avctx->frame_size;
137 FFIIRFilterContext *iir = &ctx->fiir;
140 for (ch = 0; ch < channels; ch++)
141 iir->filter_flt(ctx->fcoeffs, ctx->fstate[ch], frame_size,
142 &audio[ch][frame_size], 1, &audio[ch][frame_size], 1);
146 av_cold void ff_psy_preprocess_end(struct FFPsyPreprocessContext *ctx)
149 ff_iir_filter_free_coeffsp(&ctx->fcoeffs);
151 for (i = 0; i < ctx->avctx->channels; i++)
152 ff_iir_filter_free_statep(&ctx->fstate[i]);
153 av_freep(&ctx->fstate);