Mercurial > libavcodec.hg
comparison psymodel.c @ 9935:d09283aeeef8 libavcodec
Merge the AAC encoder from SoC svn. It is still considered experimental.
author | alexc |
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date | Wed, 08 Jul 2009 20:01:31 +0000 |
parents | |
children | 7f42ae22c351 |
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9934:ff96ee73b08b | 9935:d09283aeeef8 |
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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 | |
26 extern const FFPsyModel ff_aac_psy_model; | |
27 | |
28 av_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); | |
38 switch(ctx->avctx->codec_id){ | |
39 case CODEC_ID_AAC: | |
40 ctx->model = &ff_aac_psy_model; | |
41 break; | |
42 } | |
43 if(ctx->model->init) | |
44 return ctx->model->init(ctx); | |
45 return 0; | |
46 } | |
47 | |
48 FFPsyWindowInfo 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 | |
55 void 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 | |
61 av_cold void ff_psy_end(FFPsyContext *ctx) | |
62 { | |
63 if(ctx->model->end) | |
64 ctx->model->end(ctx); | |
65 av_freep(&ctx->bands); | |
66 av_freep(&ctx->num_bands); | |
67 av_freep(&ctx->psy_bands); | |
68 } | |
69 | |
70 typedef 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 | |
79 av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *avctx) | |
80 { | |
81 FFPsyPreprocessContext *ctx; | |
82 int i; | |
83 float cutoff_coeff; | |
84 ctx = av_mallocz(sizeof(FFPsyPreprocessContext)); | |
85 ctx->avctx = avctx; | |
86 | |
87 if(avctx->flags & CODEC_FLAG_QSCALE) | |
88 cutoff_coeff = 1.0f / av_clip(1 + avctx->global_quality / FF_QUALITY_SCALE, 1, 8); | |
89 else | |
90 cutoff_coeff = avctx->bit_rate / (4.0f * avctx->sample_rate * avctx->channels); | |
91 | |
92 ctx->fcoeffs = ff_iir_filter_init_coeffs(FF_FILTER_TYPE_BUTTERWORTH, FF_FILTER_MODE_LOWPASS, | |
93 FILT_ORDER, cutoff_coeff, 0.0, 0.0); | |
94 if(ctx->fcoeffs){ | |
95 ctx->fstate = av_mallocz(sizeof(ctx->fstate[0]) * avctx->channels); | |
96 for(i = 0; i < avctx->channels; i++) | |
97 ctx->fstate[i] = ff_iir_filter_init_state(FILT_ORDER); | |
98 } | |
99 return ctx; | |
100 } | |
101 | |
102 void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx, | |
103 const int16_t *audio, int16_t *dest, | |
104 int tag, int channels) | |
105 { | |
106 int ch, i; | |
107 if(ctx->fstate){ | |
108 for(ch = 0; ch < channels; ch++){ | |
109 ff_iir_filter(ctx->fcoeffs, ctx->fstate[tag+ch], ctx->avctx->frame_size, | |
110 audio + ch, ctx->avctx->channels, | |
111 dest + ch, ctx->avctx->channels); | |
112 } | |
113 }else{ | |
114 for(ch = 0; ch < channels; ch++){ | |
115 for(i = 0; i < ctx->avctx->frame_size; i++) | |
116 dest[i*ctx->avctx->channels + ch] = audio[i*ctx->avctx->channels + ch]; | |
117 } | |
118 } | |
119 } | |
120 | |
121 av_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 |