comparison vorbis.c @ 3571:3bfe36a9db73 libavcodec

vorbis cosmetics: mdct0,mdct1 => mdct[2]

3570:991ef6ade276

    av_freep(&vc->buf_tmp);

    av_freep(&vc->residues);
    av_freep(&vc->modes);

    ff_mdct_end(&vc->mdct0);
    ff_mdct_end(&vc->mdct1);

    for(i=0;i<vc->codebook_count;++i) {
        av_free(vc->codebooks[i].codevectors);
        free_vlc(&vc->codebooks[i].vlc);
    }

        av_free(vc->mappings[i].mux);
    }
    av_freep(&vc->mappings);

    if(vc->exp_bias){
        av_freep(&vc->swin);
        av_freep(&vc->lwin);
    }
}

// Parse setup header -------------------------------------------------

    int_fast32_t * map;
    int_fast32_t n; //TODO: could theoretically be smaller?

    for (blockflag=0;blockflag<2;++blockflag)
    {
    n=(blockflag ? vc->blocksize_1 : vc->blocksize_0) / 2;
    floors[floor_number].data.t0.map[blockflag]=
        av_malloc((n+1) * sizeof(int_fast32_t)); // n+sentinel

    map=floors[floor_number].data.t0.map[blockflag];
    vf=&floors[floor_number].data.t0;

    vc->bitrate_maximum=get_bits_long_le(gb, 32);
    vc->bitrate_nominal=get_bits_long_le(gb, 32);
    vc->bitrate_minimum=get_bits_long_le(gb, 32);
    bl0=get_bits(gb, 4);
    bl1=get_bits(gb, 4);
    vc->blocksize_0=(1<<bl0);
    vc->blocksize_1=(1<<bl1);
    if (bl0>13 || bl0<6 || bl1>13 || bl1<6 || bl1<bl0) {
        av_log(vc->avccontext, AV_LOG_ERROR, " Vorbis id header packet corrupt (illegal blocksize). \n");
        return 3;
    }
    // output format int16
    if (vc->blocksize_1/2 * vc->audio_channels * 2 >
                                             AVCODEC_MAX_AUDIO_FRAME_SIZE) {
        av_log(vc->avccontext, AV_LOG_ERROR, "Vorbis channel count makes "
               "output packets too large.\n");
        return 4;
    }
    vc->swin=vwin[bl0-6];
    vc->lwin=vwin[bl1-6];

    if(vc->exp_bias){
        int i;
        float *win;
        win = av_malloc(vc->blocksize_0/2 * sizeof(float));
        for(i=0; i<vc->blocksize_0/2; i++)
            win[i] = vc->swin[i] * (1<<15);
        vc->swin = win;
        win = av_malloc(vc->blocksize_1/2 * sizeof(float));
        for(i=0; i<vc->blocksize_1/2; i++)
            win[i] = vc->lwin[i] * (1<<15);
        vc->lwin = win;
    }

    if ((get_bits1(gb)) == 0) {
        av_log(vc->avccontext, AV_LOG_ERROR, " Vorbis id header packet corrupt (framing flag not set). \n");
        return 2;
    }

    vc->channel_residues=(float *)av_malloc((vc->blocksize_1/2)*vc->audio_channels * sizeof(float));
    vc->channel_floors=(float *)av_malloc((vc->blocksize_1/2)*vc->audio_channels * sizeof(float));
    vc->saved=(float *)av_malloc((vc->blocksize_1/2)*vc->audio_channels * sizeof(float));
    vc->ret=(float *)av_malloc((vc->blocksize_1/2)*vc->audio_channels * sizeof(float));
    vc->buf=(float *)av_malloc(vc->blocksize_1 * sizeof(float));
    vc->buf_tmp=(float *)av_malloc(vc->blocksize_1 * sizeof(float));
    vc->saved_start=0;

    ff_mdct_init(&vc->mdct0, bl0, 1);
    ff_mdct_init(&vc->mdct1, bl1, 1);

    AV_DEBUG(" vorbis version %d \n audio_channels %d \n audio_samplerate %d \n bitrate_max %d \n bitrate_nom %d \n bitrate_min %d \n blk_0 %d blk_1 %d \n ",
            vc->version, vc->audio_channels, vc->audio_samplerate, vc->bitrate_maximum, vc->bitrate_nominal, vc->bitrate_minimum, vc->blocksize_0, vc->blocksize_1);

/*
    BLK=vc->blocksize_0;
    for(i=0;i<BLK/2;++i) {
        vc->swin[i]=sin(0.5*3.14159265358*(sin(((float)i+0.5)/(float)BLK*3.14159265358))*(sin(((float)i+0.5)/(float)BLK*3.14159265358)));
    }
*/

    return 0;
}

    if (vc->modes[mode_number].blockflag) {
        previous_window=get_bits1(gb);
        next_window=get_bits1(gb);
    }

    blocksize=vc->modes[mode_number].blockflag ? vc->blocksize_1 : vc->blocksize_0;
    memset(ch_res_ptr, 0, sizeof(float)*vc->audio_channels*blocksize/2); //FIXME can this be removed ?
    memset(ch_floor_ptr, 0, sizeof(float)*vc->audio_channels*blocksize/2); //FIXME can this be removed ?

// Decode floor

// MDCT, overlap/add, save data for next overlapping  FPMATH

    for(j=0;j<vc->audio_channels;++j) {
        uint_fast8_t step=vc->audio_channels;
        uint_fast16_t k;
        float *saved=vc->saved+j*vc->blocksize_1/2;
        float *ret=vc->ret;
        const float *lwin=vc->lwin;
        const float *swin=vc->swin;
        float *buf=vc->buf;
        float *buf_tmp=vc->buf_tmp;

        ch_floor_ptr=vc->channel_floors+j*blocksize/2;

        saved_start=vc->saved_start;

        vc->mdct0.fft.imdct_calc(vc->modes[mode_number].blockflag ? &vc->mdct1 : &vc->mdct0, buf, ch_floor_ptr, buf_tmp);

        //FIXME process channels together, to allow faster simd vector_fmul_add_add?
        if (vc->modes[mode_number].blockflag) {
            // -- overlap/add
            if (previous_window) {
                vc->dsp.vector_fmul_add_add(ret+j, buf, lwin, saved, vc->add_bias, vc->blocksize_1/2, step);
                retlen=vc->blocksize_1/2;
            } else {
                int len = (vc->blocksize_1-vc->blocksize_0)/4;
                buf += len;
                vc->dsp.vector_fmul_add_add(ret+j, buf, swin, saved, vc->add_bias, vc->blocksize_0/2, step);
                k = vc->blocksize_0/2*step + j;
                buf += vc->blocksize_0/2;
                if(vc->exp_bias){
                    for(i=0; i<len; i++, k+=step)
                        ((uint32_t*)ret)[k] = ((uint32_t*)buf)[i] + vc->exp_bias; // ret[k]=buf[i]*(1<<bias)
                } else {
                    for(i=0; i<len; i++, k+=step)
                        ret[k] = buf[i] + fadd_bias;
                }
                buf=vc->buf;
                retlen=vc->blocksize_0/2+len;
            }
            // -- save
            if (next_window) {
                buf += vc->blocksize_1/2;
                vc->dsp.vector_fmul_reverse(saved, buf, lwin, vc->blocksize_1/2);
                saved_start=0;
            } else {
                saved_start=(vc->blocksize_1-vc->blocksize_0)/4;
                buf += vc->blocksize_1/2;
                for(i=0; i<saved_start; i++)
                    ((uint32_t*)saved)[i] = ((uint32_t*)buf)[i] + vc->exp_bias;
                vc->dsp.vector_fmul_reverse(saved+saved_start, buf+saved_start, swin, vc->blocksize_0/2);
            }
        } else {
            // --overlap/add
            if(vc->add_bias) {
                for(k=j, i=0;i<saved_start;++i, k+=step)
                    ret[k] = saved[i] + fadd_bias;
            } else {
                for(k=j, i=0;i<saved_start;++i, k+=step)
                    ret[k] = saved[i];
            }
            vc->dsp.vector_fmul_add_add(ret+k, buf, swin, saved+saved_start, vc->add_bias, vc->blocksize_0/2, step);
            retlen=saved_start+vc->blocksize_0/2;
            // -- save
            buf += vc->blocksize_0/2;
            vc->dsp.vector_fmul_reverse(saved, buf, swin, vc->blocksize_0/2);
            saved_start=0;
        }
    }
    vc->saved_start=saved_start;

3571:3bfe36a9db73

    av_freep(&vc->buf_tmp);

    av_freep(&vc->residues);
    av_freep(&vc->modes);

    ff_mdct_end(&vc->mdct[0]);
    ff_mdct_end(&vc->mdct[1]);

    for(i=0;i<vc->codebook_count;++i) {
        av_free(vc->codebooks[i].codevectors);
        free_vlc(&vc->codebooks[i].vlc);
    }

        av_free(vc->mappings[i].mux);
    }
    av_freep(&vc->mappings);

    if(vc->exp_bias){
        av_freep(&vc->win[0]);
        av_freep(&vc->win[1]);
    }
}

// Parse setup header -------------------------------------------------

    int_fast32_t * map;
    int_fast32_t n; //TODO: could theoretically be smaller?

    for (blockflag=0;blockflag<2;++blockflag)
    {
    n=(blockflag ? vc->blocksize[1] : vc->blocksize[0]) / 2;
    floors[floor_number].data.t0.map[blockflag]=
        av_malloc((n+1) * sizeof(int_fast32_t)); // n+sentinel

    map=floors[floor_number].data.t0.map[blockflag];
    vf=&floors[floor_number].data.t0;

    vc->bitrate_maximum=get_bits_long_le(gb, 32);
    vc->bitrate_nominal=get_bits_long_le(gb, 32);
    vc->bitrate_minimum=get_bits_long_le(gb, 32);
    bl0=get_bits(gb, 4);
    bl1=get_bits(gb, 4);
    vc->blocksize[0]=(1<<bl0);
    vc->blocksize[1]=(1<<bl1);
    if (bl0>13 || bl0<6 || bl1>13 || bl1<6 || bl1<bl0) {
        av_log(vc->avccontext, AV_LOG_ERROR, " Vorbis id header packet corrupt (illegal blocksize). \n");
        return 3;
    }
    // output format int16
    if (vc->blocksize[1]/2 * vc->audio_channels * 2 >
                                             AVCODEC_MAX_AUDIO_FRAME_SIZE) {
        av_log(vc->avccontext, AV_LOG_ERROR, "Vorbis channel count makes "
               "output packets too large.\n");
        return 4;
    }
    vc->win[0]=vwin[bl0-6];
    vc->win[1]=vwin[bl1-6];

    if(vc->exp_bias){
        int i, j;
        for(j=0; j<2; j++){
            float *win = av_malloc(vc->blocksize[j]/2 * sizeof(float));
            for(i=0; i<vc->blocksize[j]/2; i++)
                win[i] = vc->win[j][i] * (1<<15);
            vc->win[j] = win;
        }
    }

    if ((get_bits1(gb)) == 0) {
        av_log(vc->avccontext, AV_LOG_ERROR, " Vorbis id header packet corrupt (framing flag not set). \n");
        return 2;
    }

    vc->channel_residues=(float *)av_malloc((vc->blocksize[1]/2)*vc->audio_channels * sizeof(float));
    vc->channel_floors=(float *)av_malloc((vc->blocksize[1]/2)*vc->audio_channels * sizeof(float));
    vc->saved=(float *)av_malloc((vc->blocksize[1]/2)*vc->audio_channels * sizeof(float));
    vc->ret=(float *)av_malloc((vc->blocksize[1]/2)*vc->audio_channels * sizeof(float));
    vc->buf=(float *)av_malloc(vc->blocksize[1] * sizeof(float));
    vc->buf_tmp=(float *)av_malloc(vc->blocksize[1] * sizeof(float));
    vc->saved_start=0;

    ff_mdct_init(&vc->mdct[0], bl0, 1);
    ff_mdct_init(&vc->mdct[1], bl1, 1);

    AV_DEBUG(" vorbis version %d \n audio_channels %d \n audio_samplerate %d \n bitrate_max %d \n bitrate_nom %d \n bitrate_min %d \n blk_0 %d blk_1 %d \n ",
            vc->version, vc->audio_channels, vc->audio_samplerate, vc->bitrate_maximum, vc->bitrate_nominal, vc->bitrate_minimum, vc->blocksize[0], vc->blocksize[1]);

/*
    BLK=vc->blocksize[0];
    for(i=0;i<BLK/2;++i) {
        vc->win[0][i]=sin(0.5*3.14159265358*(sin(((float)i+0.5)/(float)BLK*3.14159265358))*(sin(((float)i+0.5)/(float)BLK*3.14159265358)));
    }
*/

    return 0;
}

    if (vc->modes[mode_number].blockflag) {
        previous_window=get_bits1(gb);
        next_window=get_bits1(gb);
    }

    blocksize=vc->blocksize[vc->modes[mode_number].blockflag];
    memset(ch_res_ptr, 0, sizeof(float)*vc->audio_channels*blocksize/2); //FIXME can this be removed ?
    memset(ch_floor_ptr, 0, sizeof(float)*vc->audio_channels*blocksize/2); //FIXME can this be removed ?

// Decode floor

// MDCT, overlap/add, save data for next overlapping  FPMATH

    for(j=0;j<vc->audio_channels;++j) {
        uint_fast8_t step=vc->audio_channels;
        uint_fast16_t k;
        float *saved=vc->saved+j*vc->blocksize[1]/2;
        float *ret=vc->ret;
        const float *lwin=vc->win[1];
        const float *swin=vc->win[0];
        float *buf=vc->buf;
        float *buf_tmp=vc->buf_tmp;

        ch_floor_ptr=vc->channel_floors+j*blocksize/2;

        saved_start=vc->saved_start;

        vc->mdct[0].fft.imdct_calc(&vc->mdct[vc->modes[mode_number].blockflag], buf, ch_floor_ptr, buf_tmp);

        //FIXME process channels together, to allow faster simd vector_fmul_add_add?
        if (vc->modes[mode_number].blockflag) {
            // -- overlap/add
            if (previous_window) {
                vc->dsp.vector_fmul_add_add(ret+j, buf, lwin, saved, vc->add_bias, vc->blocksize[1]/2, step);
                retlen=vc->blocksize[1]/2;
            } else {
                int len = (vc->blocksize[1]-vc->blocksize[0])/4;
                buf += len;
                vc->dsp.vector_fmul_add_add(ret+j, buf, swin, saved, vc->add_bias, vc->blocksize[0]/2, step);
                k = vc->blocksize[0]/2*step + j;
                buf += vc->blocksize[0]/2;
                if(vc->exp_bias){
                    for(i=0; i<len; i++, k+=step)
                        ((uint32_t*)ret)[k] = ((uint32_t*)buf)[i] + vc->exp_bias; // ret[k]=buf[i]*(1<<bias)
                } else {
                    for(i=0; i<len; i++, k+=step)
                        ret[k] = buf[i] + fadd_bias;
                }
                buf=vc->buf;
                retlen=vc->blocksize[0]/2+len;
            }
            // -- save
            if (next_window) {
                buf += vc->blocksize[1]/2;
                vc->dsp.vector_fmul_reverse(saved, buf, lwin, vc->blocksize[1]/2);
                saved_start=0;
            } else {
                saved_start=(vc->blocksize[1]-vc->blocksize[0])/4;
                buf += vc->blocksize[1]/2;
                for(i=0; i<saved_start; i++)
                    ((uint32_t*)saved)[i] = ((uint32_t*)buf)[i] + vc->exp_bias;
                vc->dsp.vector_fmul_reverse(saved+saved_start, buf+saved_start, swin, vc->blocksize[0]/2);
            }
        } else {
            // --overlap/add
            if(vc->add_bias) {
                for(k=j, i=0;i<saved_start;++i, k+=step)
                    ret[k] = saved[i] + fadd_bias;
            } else {
                for(k=j, i=0;i<saved_start;++i, k+=step)
                    ret[k] = saved[i];
            }
            vc->dsp.vector_fmul_add_add(ret+k, buf, swin, saved+saved_start, vc->add_bias, vc->blocksize[0]/2, step);
            retlen=saved_start+vc->blocksize[0]/2;
            // -- save
            buf += vc->blocksize[0]/2;
            vc->dsp.vector_fmul_reverse(saved, buf, swin, vc->blocksize[0]/2);
            saved_start=0;
        }
    }
    vc->saved_start=saved_start;