Mercurial > libavcodec.hg
annotate psymodel.c @ 10069:8ac9bc10b485 libavcodec
Always allocate a buffer of AVPALETTE_SIZE for palette in the subtitle
decoders instead of as small as possible.
This avoids completely unnecessary issues with e.g. libswscale.
author | reimar |
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date | Thu, 20 Aug 2009 18:54:50 +0000 |
parents | a79d7debe431 |
children | 9db3fbaba639 |
rev | line source |
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9935
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1 /* |
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2 * audio encoder psychoacoustic model |
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3 * Copyright (C) 2008 Konstantin Shishkov |
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4 * |
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5 * This file is part of FFmpeg. |
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6 * |
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7 * FFmpeg is free software; you can redistribute it and/or |
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8 * modify it under the terms of the GNU Lesser General Public |
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9 * License as published by the Free Software Foundation; either |
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10 * version 2.1 of the License, or (at your option) any later version. |
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11 * |
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12 * FFmpeg is distributed in the hope that it will be useful, |
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13 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
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14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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15 * Lesser General Public License for more details. |
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16 * |
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17 * You should have received a copy of the GNU Lesser General Public |
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18 * License along with FFmpeg; if not, write to the Free Software |
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19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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20 */ |
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21 |
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22 #include "avcodec.h" |
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23 #include "psymodel.h" |
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24 #include "iirfilter.h" |
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25 |
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26 extern const FFPsyModel ff_aac_psy_model; |
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27 |
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28 av_cold int ff_psy_init(FFPsyContext *ctx, AVCodecContext *avctx, |
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29 int num_lens, |
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30 const uint8_t **bands, const int* num_bands) |
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31 { |
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32 ctx->avctx = avctx; |
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33 ctx->psy_bands = av_mallocz(sizeof(FFPsyBand) * PSY_MAX_BANDS * avctx->channels); |
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34 ctx->bands = av_malloc (sizeof(ctx->bands[0]) * num_lens); |
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35 ctx->num_bands = av_malloc (sizeof(ctx->num_bands[0]) * num_lens); |
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36 memcpy(ctx->bands, bands, sizeof(ctx->bands[0]) * num_lens); |
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37 memcpy(ctx->num_bands, num_bands, sizeof(ctx->num_bands[0]) * num_lens); |
9936 | 38 switch (ctx->avctx->codec_id) { |
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39 case CODEC_ID_AAC: |
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40 ctx->model = &ff_aac_psy_model; |
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41 break; |
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42 } |
9936 | 43 if (ctx->model->init) |
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44 return ctx->model->init(ctx); |
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45 return 0; |
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46 } |
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47 |
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48 FFPsyWindowInfo ff_psy_suggest_window(FFPsyContext *ctx, |
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49 const int16_t *audio, const int16_t *la, |
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50 int channel, int prev_type) |
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51 { |
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52 return ctx->model->window(ctx, audio, la, channel, prev_type); |
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53 } |
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54 |
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55 void ff_psy_set_band_info(FFPsyContext *ctx, int channel, |
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56 const float *coeffs, FFPsyWindowInfo *wi) |
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57 { |
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58 ctx->model->analyze(ctx, channel, coeffs, wi); |
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59 } |
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60 |
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61 av_cold void ff_psy_end(FFPsyContext *ctx) |
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62 { |
9936 | 63 if (ctx->model->end) |
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64 ctx->model->end(ctx); |
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65 av_freep(&ctx->bands); |
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66 av_freep(&ctx->num_bands); |
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67 av_freep(&ctx->psy_bands); |
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68 } |
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69 |
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70 typedef struct FFPsyPreprocessContext{ |
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71 AVCodecContext *avctx; |
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72 float stereo_att; |
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73 struct FFIIRFilterCoeffs *fcoeffs; |
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74 struct FFIIRFilterState **fstate; |
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75 }FFPsyPreprocessContext; |
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76 |
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77 #define FILT_ORDER 4 |
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78 |
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79 av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *avctx) |
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80 { |
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81 FFPsyPreprocessContext *ctx; |
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82 int i; |
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83 float cutoff_coeff; |
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84 ctx = av_mallocz(sizeof(FFPsyPreprocessContext)); |
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85 ctx->avctx = avctx; |
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86 |
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87 if (avctx->cutoff > 0) |
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88 cutoff_coeff = 2.0 * avctx->cutoff / avctx->sample_rate; |
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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); |
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91 else |
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92 cutoff_coeff = avctx->bit_rate / (4.0f * avctx->sample_rate * avctx->channels); |
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93 |
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94 ctx->fcoeffs = ff_iir_filter_init_coeffs(FF_FILTER_TYPE_BUTTERWORTH, FF_FILTER_MODE_LOWPASS, |
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95 FILT_ORDER, cutoff_coeff, 0.0, 0.0); |
9936 | 96 if (ctx->fcoeffs) { |
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97 ctx->fstate = av_mallocz(sizeof(ctx->fstate[0]) * avctx->channels); |
9936 | 98 for (i = 0; i < avctx->channels; i++) |
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99 ctx->fstate[i] = ff_iir_filter_init_state(FILT_ORDER); |
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100 } |
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101 return ctx; |
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102 } |
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103 |
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104 void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx, |
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105 const int16_t *audio, int16_t *dest, |
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106 int tag, int channels) |
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107 { |
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108 int ch, i; |
9936 | 109 if (ctx->fstate) { |
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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, |
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112 audio + ch, ctx->avctx->channels, |
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113 dest + ch, ctx->avctx->channels); |
9936 | 114 } else { |
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115 for (ch = 0; ch < channels; ch++) |
9936 | 116 for (i = 0; i < ctx->avctx->frame_size; i++) |
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117 dest[i*ctx->avctx->channels + ch] = audio[i*ctx->avctx->channels + ch]; |
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118 } |
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119 } |
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120 |
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121 av_cold void ff_psy_preprocess_end(struct FFPsyPreprocessContext *ctx) |
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122 { |
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123 int i; |
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124 ff_iir_filter_free_coeffs(ctx->fcoeffs); |
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125 if (ctx->fstate) |
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126 for (i = 0; i < ctx->avctx->channels; i++) |
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127 ff_iir_filter_free_state(ctx->fstate[i]); |
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128 av_freep(&ctx->fstate); |
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129 } |
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130 |