FFmpeg/libavcodec/acelp_filters.c

/*
 * 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 "acelp_filters.h"

const int16_t ff_acelp_interp_filter[61] =
{ /* (0.15) */
  29443, 28346, 25207, 20449, 14701,  8693,
   3143, -1352, -4402, -5865, -5850, -4673,
  -2783,  -672,  1211,  2536,  3130,  2991,
   2259,  1170,     0, -1001, -1652, -1868,
  -1666, -1147,  -464,   218,   756,  1060,
   1099,   904,   550,   135,  -245,  -514,
   -634,  -602,  -451,  -231,     0,   191,
    308,   340,   296,   198,    78,   -36,
   -120,  -163,  -165,  -132,   -79,   -19,
     34,    73,    91,    89,    70,    38,
      0,
};

void ff_acelp_interpolate(
        int16_t* out,
        const int16_t* in,
        const int16_t* filter_coeffs,
        int precision,
        int frac_pos,
        int filter_length,
        int length)
{
    int n, i;

    assert(pitch_delay_frac >= 0 && pitch_delay_frac < precision);

    for(n=0; n<length; n++)
    {
        int idx = 0;
        int v = 0x4000;

        for(i=0; i<filter_length;)
        {

            /* The reference G.729 and AMR fixed point code performs clipping after
               each of the two following accumulations.
               Since clipping affects only the synthetic OVERFLOW test without
               causing an int type overflow, it was moved outside the loop. */

            /*  R(x):=ac_v[-k+x]
                v += R(n-i)*ff_acelp_interp_filter(t+6i)
                v += R(n+i+1)*ff_acelp_interp_filter(6-t+6i) */

            v += in[n + i] * filter_coeffs[idx + frac_pos];
            idx += precision;
            i++;
            v += in[n - i] * filter_coeffs[idx - frac_pos];
        }
        if(av_clip_int16(v>>15) != (v>>15))
            av_log(NULL, AV_LOG_WARNING, "overflow that would need cliping in ff_acelp_interpolate()\n");
        out[n] = v >> 15;
    }
}

void ff_acelp_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;
        }
    }
}

int ff_acelp_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;

    // These two lines are to avoid a -1 subtraction in the main loop
    filter_length++;
    filter_coeffs--;

    for(n=0; n<buffer_length; n++)
    {
        int sum = rounder;
        for(i=1; i<filter_length; i++)
            sum -= filter_coeffs[i] * 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_acelp_high_pass_filter(
        int16_t* out,
        int hpf_f[2],
        const int16_t* in,
        int length)
{
    int i;
    int tmp;

    for(i=0; i<length; i++)
    {
        tmp =  (hpf_f[0]* 15836LL)>>13;
        tmp += (hpf_f[1]* -7667LL)>>13;
        tmp += 7699 * (in[i] - 2*in[i-1] + in[i-2]);

        /* With "+0x800" rounding, clipping is needed
           for ALGTHM and SPEECH tests. */
        out[i] = av_clip_int16((tmp + 0x800) >> 12);

        hpf_f[1] = hpf_f[0];
        hpf_f[0] = tmp;
    }
}