view psymodel.c @ 10952:ea8f891d997d libavcodec

H264 DXVA2 implementation It allows VLD H264 decoding using DXVA2 (GPU assisted decoding API under VISTA and Windows 7). It is implemented by using AVHWAccel API. It has been tested successfully for some time in VLC using an nvidia card on Windows 7. To compile it, you need to have the system header dxva2api.h (either from microsoft or using http://downloads.videolan.org/pub/videolan/testing/contrib/dxva2api.h) The generated libavcodec.dll does not depend directly on any new lib as the necessary objects are given by the application using FFmpeg.

/*
 * 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);
}