Use cutoff frequency to adjust bandwidth in the generic psymodel preprocess.
[libav.git] / libavcodec / psymodel.c
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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 "avcodec.h"
23#include "psymodel.h"
24#include "iirfilter.h"
25
26extern const FFPsyModel ff_aac_psy_model;
27
28av_cold int ff_psy_init(FFPsyContext *ctx, AVCodecContext *avctx,
29 int num_lens,
30 const uint8_t **bands, const int* num_bands)
31{
32 ctx->avctx = avctx;
33 ctx->psy_bands = av_mallocz(sizeof(FFPsyBand) * PSY_MAX_BANDS * avctx->channels);
34 ctx->bands = av_malloc (sizeof(ctx->bands[0]) * num_lens);
35 ctx->num_bands = av_malloc (sizeof(ctx->num_bands[0]) * num_lens);
36 memcpy(ctx->bands, bands, sizeof(ctx->bands[0]) * num_lens);
37 memcpy(ctx->num_bands, num_bands, sizeof(ctx->num_bands[0]) * num_lens);
fd257dc4 38 switch (ctx->avctx->codec_id) {
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39 case CODEC_ID_AAC:
40 ctx->model = &ff_aac_psy_model;
41 break;
42 }
fd257dc4 43 if (ctx->model->init)
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44 return ctx->model->init(ctx);
45 return 0;
46}
47
48FFPsyWindowInfo ff_psy_suggest_window(FFPsyContext *ctx,
49 const int16_t *audio, const int16_t *la,
50 int channel, int prev_type)
51{
52 return ctx->model->window(ctx, audio, la, channel, prev_type);
53}
54
55void ff_psy_set_band_info(FFPsyContext *ctx, int channel,
56 const float *coeffs, FFPsyWindowInfo *wi)
57{
58 ctx->model->analyze(ctx, channel, coeffs, wi);
59}
60
61av_cold void ff_psy_end(FFPsyContext *ctx)
62{
fd257dc4 63 if (ctx->model->end)
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64 ctx->model->end(ctx);
65 av_freep(&ctx->bands);
66 av_freep(&ctx->num_bands);
67 av_freep(&ctx->psy_bands);
68}
69
70typedef struct FFPsyPreprocessContext{
71 AVCodecContext *avctx;
72 float stereo_att;
73 struct FFIIRFilterCoeffs *fcoeffs;
74 struct FFIIRFilterState **fstate;
75}FFPsyPreprocessContext;
76
77#define FILT_ORDER 4
78
79av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *avctx)
80{
81 FFPsyPreprocessContext *ctx;
82 int i;
83 float cutoff_coeff;
99d61d34 84 ctx = av_mallocz(sizeof(FFPsyPreprocessContext));
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85 ctx->avctx = avctx;
86
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87 if (avctx->cutoff > 0)
88 cutoff_coeff = 2.0 * avctx->cutoff / avctx->sample_rate;
89 else if (avctx->flags & CODEC_FLAG_QSCALE)
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90 cutoff_coeff = 1.0f / av_clip(1 + avctx->global_quality / FF_QUALITY_SCALE, 1, 8);
91 else
92 cutoff_coeff = avctx->bit_rate / (4.0f * avctx->sample_rate * avctx->channels);
93
94 ctx->fcoeffs = ff_iir_filter_init_coeffs(FF_FILTER_TYPE_BUTTERWORTH, FF_FILTER_MODE_LOWPASS,
99d61d34 95 FILT_ORDER, cutoff_coeff, 0.0, 0.0);
fd257dc4 96 if (ctx->fcoeffs) {
78e65cd7 97 ctx->fstate = av_mallocz(sizeof(ctx->fstate[0]) * avctx->channels);
fd257dc4 98 for (i = 0; i < avctx->channels; i++)
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99 ctx->fstate[i] = ff_iir_filter_init_state(FILT_ORDER);
100 }
101 return ctx;
102}
103
104void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx,
105 const int16_t *audio, int16_t *dest,
106 int tag, int channels)
107{
108 int ch, i;
fd257dc4 109 if (ctx->fstate) {
c8f47d8b 110 for (ch = 0; ch < channels; ch++)
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111 ff_iir_filter(ctx->fcoeffs, ctx->fstate[tag+ch], ctx->avctx->frame_size,
112 audio + ch, ctx->avctx->channels,
113 dest + ch, ctx->avctx->channels);
fd257dc4 114 } else {
c8f47d8b 115 for (ch = 0; ch < channels; ch++)
fd257dc4 116 for (i = 0; i < ctx->avctx->frame_size; i++)
78e65cd7 117 dest[i*ctx->avctx->channels + ch] = audio[i*ctx->avctx->channels + ch];
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118 }
119}
120
121av_cold void ff_psy_preprocess_end(struct FFPsyPreprocessContext *ctx)
122{
123 int i;
124 ff_iir_filter_free_coeffs(ctx->fcoeffs);
125 if (ctx->fstate)
126 for (i = 0; i < ctx->avctx->channels; i++)
127 ff_iir_filter_free_state(ctx->fstate[i]);
128 av_freep(&ctx->fstate);
129}
130