view acelp_vectors.c @ 12089:5638941ec8ef libavcodec

PPC: convert Altivec FFT to pure assembler On PPC a leaf function has a 288-byte red zone below the stack pointer, sparing these functions the chore of setting up a full stack frame. When a function call is disguised within an inline asm block, the compiler might not adjust the stack pointer as required before a function call, resulting in the red zone being clobbered. Moving the entire function to pure asm avoids this problem and also results in somewhat better code.

/*
 * adaptive and fixed codebook vector operations 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_vectors.h"
#include "celp_math.h"

const uint8_t ff_fc_2pulses_9bits_track1[16] =
{
    1,  3,
    6,  8,
    11, 13,
    16, 18,
    21, 23,
    26, 28,
    31, 33,
    36, 38
};
const uint8_t ff_fc_2pulses_9bits_track1_gray[16] =
{
  1,  3,
  8,  6,
  18, 16,
  11, 13,
  38, 36,
  31, 33,
  21, 23,
  28, 26,
};

const uint8_t ff_fc_2pulses_9bits_track2_gray[32] =
{
  0,  2,
  5,  4,
  12, 10,
  7,  9,
  25, 24,
  20, 22,
  14, 15,
  19, 17,
  36, 31,
  21, 26,
  1,  6,
  16, 11,
  27, 29,
  32, 30,
  39, 37,
  34, 35,
};

const uint8_t ff_fc_4pulses_8bits_tracks_13[16] =
{
  0, 5, 10, 15, 20, 25, 30, 35, 40, 45, 50, 55, 60, 65, 70, 75,
};

const uint8_t ff_fc_4pulses_8bits_track_4[32] =
{
    3,  4,
    8,  9,
    13, 14,
    18, 19,
    23, 24,
    28, 29,
    33, 34,
    38, 39,
    43, 44,
    48, 49,
    53, 54,
    58, 59,
    63, 64,
    68, 69,
    73, 74,
    78, 79,
};

#if 0
static uint8_t gray_decode[32] =
{
    0,  1,  3,  2,  7,  6,  4,  5,
   15, 14, 12, 13,  8,  9, 11, 10,
   31, 30, 28, 29, 24, 25, 27, 26,
   16, 17, 19, 18, 23, 22, 20, 21
};
#endif

const float ff_pow_0_7[10] = {
    0.700000, 0.490000, 0.343000, 0.240100, 0.168070,
    0.117649, 0.082354, 0.057648, 0.040354, 0.028248
};

const float ff_pow_0_75[10] = {
    0.750000, 0.562500, 0.421875, 0.316406, 0.237305,
    0.177979, 0.133484, 0.100113, 0.075085, 0.056314
};

const float ff_pow_0_55[10] = {
    0.550000, 0.302500, 0.166375, 0.091506, 0.050328,
    0.027681, 0.015224, 0.008373, 0.004605, 0.002533
};

const float ff_b60_sinc[61] = {
 0.898529  ,  0.865051  ,  0.769257  ,  0.624054  ,  0.448639  ,  0.265289   ,
 0.0959167 , -0.0412598 , -0.134338  , -0.178986  , -0.178528  , -0.142609   ,
-0.0849304 , -0.0205078 ,  0.0369568 ,  0.0773926 ,  0.0955200 ,  0.0912781  ,
 0.0689392 ,  0.0357056 ,  0.        , -0.0305481 , -0.0504150 , -0.0570068  ,
-0.0508423 , -0.0350037 , -0.0141602 ,  0.00665283,  0.0230713 ,  0.0323486  ,
 0.0335388 ,  0.0275879 ,  0.0167847 ,  0.00411987, -0.00747681, -0.0156860  ,
-0.0193481 , -0.0183716 , -0.0137634 , -0.00704956,  0.        ,  0.00582886 ,
 0.00939941,  0.0103760 ,  0.00903320,  0.00604248,  0.00238037, -0.00109863 ,
-0.00366211, -0.00497437, -0.00503540, -0.00402832, -0.00241089, -0.000579834,
 0.00103760,  0.00222778,  0.00277710,  0.00271606,  0.00213623,  0.00115967 ,
 0.
};

void ff_acelp_fc_pulse_per_track(
        int16_t* fc_v,
        const uint8_t *tab1,
        const uint8_t *tab2,
        int pulse_indexes,
        int pulse_signs,
        int pulse_count,
        int bits)
{
    int mask = (1 << bits) - 1;
    int i;

    for(i=0; i<pulse_count; i++)
    {
        fc_v[i + tab1[pulse_indexes & mask]] +=
                (pulse_signs & 1) ? 8191 : -8192; // +/-1 in (2.13)

        pulse_indexes >>= bits;
        pulse_signs >>= 1;
    }

    fc_v[tab2[pulse_indexes]] += (pulse_signs & 1) ? 8191 : -8192;
}

void ff_decode_10_pulses_35bits(const int16_t *fixed_index,
                                AMRFixed *fixed_sparse,
                                const uint8_t *gray_decode,
                                int half_pulse_count, int bits)
{
    int i;
    int mask = (1 << bits) - 1;

    fixed_sparse->no_repeat_mask = 0;
    fixed_sparse->n = 2 * half_pulse_count;
    for (i = 0; i < half_pulse_count; i++) {
        const int pos1   = gray_decode[fixed_index[2*i+1] & mask] + i;
        const int pos2   = gray_decode[fixed_index[2*i  ] & mask] + i;
        const float sign = (fixed_index[2*i+1] & (1 << bits)) ? -1.0 : 1.0;
        fixed_sparse->x[2*i+1] = pos1;
        fixed_sparse->x[2*i  ] = pos2;
        fixed_sparse->y[2*i+1] = sign;
        fixed_sparse->y[2*i  ] = pos2 < pos1 ? -sign : sign;
    }
}

void ff_acelp_weighted_vector_sum(
        int16_t* out,
        const int16_t *in_a,
        const int16_t *in_b,
        int16_t weight_coeff_a,
        int16_t weight_coeff_b,
        int16_t rounder,
        int shift,
        int length)
{
    int i;

    // Clipping required here; breaks OVERFLOW test.
    for(i=0; i<length; i++)
        out[i] = av_clip_int16((
                 in_a[i] * weight_coeff_a +
                 in_b[i] * weight_coeff_b +
                 rounder) >> shift);
}

void ff_weighted_vector_sumf(float *out, const float *in_a, const float *in_b,
                             float weight_coeff_a, float weight_coeff_b, int length)
{
    int i;

    for(i=0; i<length; i++)
        out[i] = weight_coeff_a * in_a[i]
               + weight_coeff_b * in_b[i];
}

void ff_adaptive_gain_control(float *out, const float *in, float speech_energ,
                              int size, float alpha, float *gain_mem)
{
    int i;
    float postfilter_energ = ff_dot_productf(in, in, size);
    float gain_scale_factor = 1.0;
    float mem = *gain_mem;

    if (postfilter_energ)
        gain_scale_factor = sqrt(speech_energ / postfilter_energ);

    gain_scale_factor *= 1.0 - alpha;

    for (i = 0; i < size; i++) {
        mem = alpha * mem + gain_scale_factor;
        out[i] = in[i] * mem;
    }

    *gain_mem = mem;
}

void ff_scale_vector_to_given_sum_of_squares(float *out, const float *in,
                                             float sum_of_squares, const int n)
{
    int i;
    float scalefactor = ff_dot_productf(in, in, n);
    if (scalefactor)
        scalefactor = sqrt(sum_of_squares / scalefactor);
    for (i = 0; i < n; i++)
        out[i] = in[i] * scalefactor;
}

void ff_set_fixed_vector(float *out, const AMRFixed *in, float scale, int size)
{
    int i;

    for (i=0; i < in->n; i++) {
        int x   = in->x[i], repeats = !((in->no_repeat_mask >> i) & 1);
        float y = in->y[i] * scale;

        do {
            out[x] += y;
            y *= in->pitch_fac;
            x += in->pitch_lag;
        } while (x < size && repeats);
    }
}

void ff_clear_fixed_vector(float *out, const AMRFixed *in, int size)
{
    int i;

    for (i=0; i < in->n; i++) {
        int x  = in->x[i], repeats = !((in->no_repeat_mask >> i) & 1);

        do {
            out[x] = 0.0;
            x += in->pitch_lag;
        } while (x < size && repeats);
    }
}