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Diffstat (limited to 'ffmpeg/libavcodec/psymodel.c')
| -rw-r--r-- | ffmpeg/libavcodec/psymodel.c | 147 |
1 files changed, 147 insertions, 0 deletions
diff --git a/ffmpeg/libavcodec/psymodel.c b/ffmpeg/libavcodec/psymodel.c new file mode 100644 index 0000000..ea11636 --- /dev/null +++ b/ffmpeg/libavcodec/psymodel.c @@ -0,0 +1,147 @@ +/* + * audio encoder psychoacoustic model + * Copyright (C) 2008 Konstantin Shishkov + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include <string.h> + +#include "avcodec.h" +#include "psymodel.h" +#include "iirfilter.h" +#include "libavutil/mem.h" + +extern const FFPsyModel ff_aac_psy_model; + +av_cold int ff_psy_init(FFPsyContext *ctx, AVCodecContext *avctx, int num_lens, + const uint8_t **bands, const int* num_bands, + int num_groups, const uint8_t *group_map) +{ + int i, j, k = 0; + + ctx->avctx = avctx; + ctx->ch = av_mallocz(sizeof(ctx->ch[0]) * avctx->channels * 2); + ctx->group = av_mallocz(sizeof(ctx->group[0]) * num_groups); + ctx->bands = av_malloc (sizeof(ctx->bands[0]) * num_lens); + ctx->num_bands = av_malloc (sizeof(ctx->num_bands[0]) * num_lens); + memcpy(ctx->bands, bands, sizeof(ctx->bands[0]) * num_lens); + memcpy(ctx->num_bands, num_bands, sizeof(ctx->num_bands[0]) * num_lens); + + /* assign channels to groups (with virtual channels for coupling) */ + for (i = 0; i < num_groups; i++) { + /* NOTE: Add 1 to handle the AAC chan_config without modification. + * This has the side effect of allowing an array of 0s to map + * to one channel per group. + */ + ctx->group[i].num_ch = group_map[i] + 1; + for (j = 0; j < ctx->group[i].num_ch * 2; j++) + ctx->group[i].ch[j] = &ctx->ch[k++]; + } + + switch (ctx->avctx->codec_id) { + case AV_CODEC_ID_AAC: + ctx->model = &ff_aac_psy_model; + break; + } + if (ctx->model->init) + return ctx->model->init(ctx); + return 0; +} + +FFPsyChannelGroup *ff_psy_find_group(FFPsyContext *ctx, int channel) +{ + int i = 0, ch = 0; + + while (ch <= channel) + ch += ctx->group[i++].num_ch; + + return &ctx->group[i-1]; +} + +av_cold void ff_psy_end(FFPsyContext *ctx) +{ + if (ctx->model->end) + ctx->model->end(ctx); + av_freep(&ctx->bands); + av_freep(&ctx->num_bands); + av_freep(&ctx->group); + av_freep(&ctx->ch); +} + +typedef struct FFPsyPreprocessContext{ + AVCodecContext *avctx; + float stereo_att; + struct FFIIRFilterCoeffs *fcoeffs; + struct FFIIRFilterState **fstate; + struct FFIIRFilterContext fiir; +}FFPsyPreprocessContext; + +#define FILT_ORDER 4 + +av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *avctx) +{ + FFPsyPreprocessContext *ctx; + int i; + float cutoff_coeff = 0; + ctx = av_mallocz(sizeof(FFPsyPreprocessContext)); + ctx->avctx = avctx; + + if (avctx->cutoff > 0) + cutoff_coeff = 2.0 * avctx->cutoff / avctx->sample_rate; + + if (!cutoff_coeff && avctx->codec_id == AV_CODEC_ID_AAC) + cutoff_coeff = 2.0 * AAC_CUTOFF(avctx) / avctx->sample_rate; + + if (cutoff_coeff && cutoff_coeff < 0.98) + ctx->fcoeffs = ff_iir_filter_init_coeffs(avctx, FF_FILTER_TYPE_BUTTERWORTH, + FF_FILTER_MODE_LOWPASS, FILT_ORDER, + cutoff_coeff, 0.0, 0.0); + if (ctx->fcoeffs) { + ctx->fstate = av_mallocz(sizeof(ctx->fstate[0]) * avctx->channels); + for (i = 0; i < avctx->channels; i++) + ctx->fstate[i] = ff_iir_filter_init_state(FILT_ORDER); + } + + ff_iir_filter_init(&ctx->fiir); + + return ctx; +} + +void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx, float **audio, int channels) +{ + int ch; + int frame_size = ctx->avctx->frame_size; + FFIIRFilterContext *iir = &ctx->fiir; + + if (ctx->fstate) { + for (ch = 0; ch < channels; ch++) + iir->filter_flt(ctx->fcoeffs, ctx->fstate[ch], frame_size, + &audio[ch][frame_size], 1, &audio[ch][frame_size], 1); + } +} + +av_cold void ff_psy_preprocess_end(struct FFPsyPreprocessContext *ctx) +{ + int i; + ff_iir_filter_free_coeffs(ctx->fcoeffs); + if (ctx->fstate) + for (i = 0; i < ctx->avctx->channels; i++) + ff_iir_filter_free_state(ctx->fstate[i]); + av_freep(&ctx->fstate); + av_free(ctx); +} |