PeerStreamer

/*

 * 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;

}

FFPsyWindowInfo ff_psy_suggest_window(FFPsyContext *ctx,

                                      const int16_t *audio, const int16_t *la,

                                      int channel, int prev_type)

{

    return ctx->model->window(ctx, audio, la, channel, prev_type);

}

void ff_psy_set_band_info(FFPsyContext *ctx, int channel,

                          const float *coeffs, const FFPsyWindowInfo *wi)

{

    ctx->model->analyze(ctx, channel, coeffs, wi);

}

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);

}