annotate celp_filters.c @ 10139:de9335d4b6b1 libavcodec

do not decode more than one audio frame in a decode packet call

/*
 * various filters for ACELP-based codecs
 *
 * Copyright (c) 2008 Vladimir Voroshilov
 *
 * 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 <inttypes.h>

#include "avcodec.h"
#include "celp_filters.h"

void ff_celp_convolve_circ(int16_t* fc_out,
                           const int16_t* fc_in,
                           const int16_t* filter,
                           int len)
{
    int i, k;

    memset(fc_out, 0, len * sizeof(int16_t));

    /* Since there are few pulses over an entire subframe (i.e. almost
       all fc_in[i] are zero) it is faster to loop over fc_in first. */
    for (i = 0; i < len; i++) {
        if (fc_in[i]) {
            for (k = 0; k < i; k++)
                fc_out[k] += (fc_in[i] * filter[len + k - i]) >> 15;

            for (k = i; k < len; k++)
                fc_out[k] += (fc_in[i] * filter[      k - i]) >> 15;
        }
    }
}

void ff_celp_circ_addf(float *out, const float *in,
                       const float *lagged, int lag, float fac, int n)
{
    int k;
    for (k = 0; k < lag; k++)
        out[k] = in[k] + fac * lagged[n + k - lag];
    for (; k < n; k++)
        out[k] = in[k] + fac * lagged[    k - lag];
}

int ff_celp_lp_synthesis_filter(int16_t *out,
                                const int16_t* filter_coeffs,
                                const int16_t* in,
                                int buffer_length,
                                int filter_length,
                                int stop_on_overflow,
                                int rounder)
{
    int i,n;

    // Avoids a +1 in the inner loop.
    filter_length++;

    for (n = 0; n < buffer_length; n++) {
        int sum = rounder;
        for (i = 1; i < filter_length; i++)
            sum -= filter_coeffs[i-1] * out[n-i];

        sum = (sum >> 12) + in[n];

        if (sum + 0x8000 > 0xFFFFU) {
            if (stop_on_overflow)
                return 1;
            sum = (sum >> 31) ^ 32767;
        }
        out[n] = sum;
    }

    return 0;
}

void ff_celp_lp_synthesis_filterf(float *out,
                                  const float* filter_coeffs,
                                  const float* in,
                                  int buffer_length,
                                  int filter_length)
{
    int i,n;

    // Avoids a +1 in the inner loop.
    filter_length++;

    for (n = 0; n < buffer_length; n++) {
        out[n] = in[n];
        for (i = 1; i < filter_length; i++)
            out[n] -= filter_coeffs[i-1] * out[n-i];
    }
}

void ff_celp_lp_zero_synthesis_filterf(float *out,
                                       const float* filter_coeffs,
                                       const float* in,
                                       int buffer_length,
                                       int filter_length)
{
    int i,n;

    // Avoids a +1 in the inner loop.
    filter_length++;

    for (n = 0; n < buffer_length; n++) {
        out[n] = in[n];
        for (i = 1; i < filter_length; i++)
            out[n] += filter_coeffs[i-1] * in[n-i];
    }
}