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/*
* audio encoder psychoacoustic model
* Copyright (C) 2008 Konstantin Shishkov
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; 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);
}
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