view psymodel.c @ 10300:4d1b9ca628fc libavcodec

Drop unused args from vector_fmul_add_add, simpify code, and rename The src3 and step arguments to vector_fmul_add_add() are always zero and one, respectively. This removes these arguments from the function, simplifies the code accordingly, and renames the function to better match the new operation.
author mru
date Sun, 27 Sep 2009 16:51:54 +0000
parents a79d7debe431
children 9db3fbaba639
line wrap: on
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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 "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, 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;
    ctx        = av_mallocz(sizeof(FFPsyPreprocessContext));
    ctx->avctx = avctx;

    if (avctx->cutoff > 0)
        cutoff_coeff = 2.0 * avctx->cutoff / avctx->sample_rate;
    else if (avctx->flags & CODEC_FLAG_QSCALE)
        cutoff_coeff = 1.0f / av_clip(1 + avctx->global_quality / FF_QUALITY_SCALE, 1, 8);
    else
        cutoff_coeff = avctx->bit_rate / (4.0f * avctx->sample_rate * avctx->channels);

    ctx->fcoeffs = ff_iir_filter_init_coeffs(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);
}