view psymodel.c @ 12197:fbf4d5b1b664 libavcodec

Remove FF_MM_SSE2/3 flags for CPUs where this is generally not faster than regular MMX code. Examples of this are the Core1 CPU. Instead, set a new flag, FF_MM_SSE2/3SLOW, which can be checked for particular SSE2/3 functions that have been checked specifically on such CPUs and are actually faster than their MMX counterparts. In addition, use this flag to enable particular VP8 and LPC SSE2 functions that are faster than their MMX counterparts. Based on a patch by Loren Merritt <lorenm AT u washington edu>.

/*
 * 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 = 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(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);
}