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/*

 * 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 

#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 && 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);

}