comparison ac3dec.c @ 7549:60b5c1e5a7ee libavcodec

use imdct_half in ac3

7548:3d1b177a1b8c

        }
    }
}

/**
 * Perform the 256-point IMDCT
 */
static void do_imdct_256(AC3DecodeContext *s, int chindex)
{
    int i, k;
    DECLARE_ALIGNED_16(float, x[128]);
    FFTComplex z[2][64];
    float *o_ptr = s->tmp_output;

    for(i=0; i<2; i++) {
        /* de-interleave coefficients */
        for(k=0; k<128; k++) {
            x[k] = s->transform_coeffs[chindex][2*k+i];
        }

        /* run standard IMDCT */
        ff_imdct_calc(&s->imdct_256, o_ptr, x);

        /* reverse the post-rotation & reordering from standard IMDCT */
        for(k=0; k<32; k++) {
            z[i][32+k].re = -o_ptr[128+2*k];
            z[i][32+k].im = -o_ptr[2*k];
            z[i][31-k].re =  o_ptr[2*k+1];
            z[i][31-k].im =  o_ptr[128+2*k+1];
        }
    }

    /* apply AC-3 post-rotation & reordering */
    for(k=0; k<64; k++) {
        o_ptr[    2*k  ] = -z[0][   k].im;
        o_ptr[    2*k+1] =  z[0][63-k].re;
        o_ptr[128+2*k  ] = -z[0][   k].re;
        o_ptr[128+2*k+1] =  z[0][63-k].im;
        o_ptr[256+2*k  ] = -z[1][   k].re;
        o_ptr[256+2*k+1] =  z[1][63-k].im;
        o_ptr[384+2*k  ] =  z[1][   k].im;
        o_ptr[384+2*k+1] = -z[1][63-k].re;
    }
}

/**
 * Inverse MDCT Transform.
 * Convert frequency domain coefficients to time-domain audio samples.
 * reference: Section 7.9.4 Transformation Equations
 */
static inline void do_imdct(AC3DecodeContext *s, int channels)
{
    int ch;

    for (ch=1; ch<=channels; ch++) {
        if (s->block_switch[ch]) {
            do_imdct_256(s, ch);
        } else {
            ff_imdct_calc(&s->imdct_512, s->tmp_output, s->transform_coeffs[ch]);
        }
        /* For the first half of the block, apply the window, add the delay
           from the previous block, and send to output */
        s->dsp.vector_fmul_add_add(s->output[ch-1], s->tmp_output,
                                     s->window, s->delay[ch-1], 0, 256, 1);
        /* For the second half of the block, apply the window and store the
           samples to delay, to be combined with the next block */
        s->dsp.vector_fmul_reverse(s->delay[ch-1], s->tmp_output+256,
                                   s->window, 256);
    }
}

/**
 * Downmix the output to mono or stereo.

/**
 * Upmix delay samples from stereo to original channel layout.
 */
static void ac3_upmix_delay(AC3DecodeContext *s)
{
    int channel_data_size = sizeof(s->delay[0]);
    switch(s->channel_mode) {
        case AC3_CHMODE_DUALMONO:
        case AC3_CHMODE_STEREO:
            /* upmix mono to stereo */
            memcpy(s->delay[1], s->delay[0], channel_data_size);

            ac3_downmix(s, s->transform_coeffs, 1);
        }

        if(!s->downmixed) {
            s->downmixed = 1;
            ac3_downmix(s, s->delay, 0);
        }

        do_imdct(s, s->out_channels);
    }

7549:60b5c1e5a7ee

        }
    }
}

/**
 * Inverse MDCT Transform.
 * Convert frequency domain coefficients to time-domain audio samples.
 * reference: Section 7.9.4 Transformation Equations
 */
static inline void do_imdct(AC3DecodeContext *s, int channels)
{
    int ch;

    for (ch=1; ch<=channels; ch++) {
        if (s->block_switch[ch]) {
            int i;
            float *x = s->tmp_output+128;
            for(i=0; i<128; i++)
                x[i] = s->transform_coeffs[ch][2*i];
            ff_imdct_half(&s->imdct_256, s->tmp_output, x);
            s->dsp.vector_fmul_window(s->output[ch-1], s->delay[ch-1], s->tmp_output, s->window, 0, 128);
            for(i=0; i<128; i++)
                x[i] = s->transform_coeffs[ch][2*i+1];
            ff_imdct_half(&s->imdct_256, s->delay[ch-1], x);
        } else {
            ff_imdct_half(&s->imdct_512, s->tmp_output, s->transform_coeffs[ch]);
            s->dsp.vector_fmul_window(s->output[ch-1], s->delay[ch-1], s->tmp_output, s->window, 0, 128);
            memcpy(s->delay[ch-1], s->tmp_output+128, 128*sizeof(float));
        }
    }
}

/**
 * Downmix the output to mono or stereo.

/**
 * Upmix delay samples from stereo to original channel layout.
 */
static void ac3_upmix_delay(AC3DecodeContext *s)
{
    int channel_data_size = 128*sizeof(float);
    switch(s->channel_mode) {
        case AC3_CHMODE_DUALMONO:
        case AC3_CHMODE_STEREO:
            /* upmix mono to stereo */
            memcpy(s->delay[1], s->delay[0], channel_data_size);

            ac3_downmix(s, s->transform_coeffs, 1);
        }

        if(!s->downmixed) {
            s->downmixed = 1;
            // FIXME delay[] is half the size of the other downmixes
            ac3_downmix(s, s->delay, 0);
        }

        do_imdct(s, s->out_channels);
    }