| /* |
| * 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 "avcodec.h" |
| #include "psymodel.h" |
| #include "iirfilter.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) |
| { |
| ctx->avctx = avctx; |
| ctx->psy_bands = av_mallocz(sizeof(FFPsyBand) * PSY_MAX_BANDS * avctx->channels); |
| 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); |
| switch (ctx->avctx->codec_id) { |
| case CODEC_ID_AAC: |
| ctx->model = &ff_aac_psy_model; |
| break; |
| } |
| if (ctx->model->init) |
| return ctx->model->init(ctx); |
| return 0; |
| } |
| |
| 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->psy_bands); |
| } |
| |
| typedef struct FFPsyPreprocessContext{ |
| AVCodecContext *avctx; |
| float stereo_att; |
| struct FFIIRFilterCoeffs *fcoeffs; |
| struct FFIIRFilterState **fstate; |
| }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) |
| 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); |
| } |
| return ctx; |
| } |
| |
| void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx, |
| const int16_t *audio, int16_t *dest, |
| int tag, int channels) |
| { |
| int ch, i; |
| if (ctx->fstate) { |
| for (ch = 0; ch < channels; ch++) |
| ff_iir_filter(ctx->fcoeffs, ctx->fstate[tag+ch], ctx->avctx->frame_size, |
| audio + ch, ctx->avctx->channels, |
| dest + ch, ctx->avctx->channels); |
| } else { |
| for (ch = 0; ch < channels; ch++) |
| for (i = 0; i < ctx->avctx->frame_size; i++) |
| dest[i*ctx->avctx->channels + ch] = audio[i*ctx->avctx->channels + ch]; |
| } |
| } |
| |
| 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); |
| } |
| |