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various filters for ACELP-based codecs

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Originally committed as revision 13110 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Vladimir Voroshilov 2008-05-11 03:42:53 +00:00
parent f863bee841
commit a52000f291
2 changed files with 238 additions and 0 deletions
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125
libavcodec/acelp_filters.c Normal file
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/*
* 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"
#define FRAC_BITS 13
#include "mathops.h"
void ff_acelp_convolve_circ(
int16_t* fc_out,
const int16_t* fc_in,
const int16_t* filter,
int subframe_size)
{
int i, k;
memset(fc_out, 0, subframe_size * sizeof(int16_t));
/* Since there are few pulses over entire subframe (i.e. almost all
fc_in[i] are zero, in case of G.729D the buffer contains two non-zero
samples before the call to ff_acelp_enhance_harmonics, and (due to
pitch_delay bounded to [20; 143]) a maximum four non-zero samples
for a total of 40 after the call to it), it is faster to swap two loops
and process non-zero samples only. This will reduce the number of
multiplications from 40*40 to 4*40 for G.729D */
for(i=0; i<subframe_size; i++)
{
if(fc_in[i])
{
for(k=0; k<i; k++)
fc_out[k] += (fc_in[i] * filter[subframe_size + k - i]) >> 15;
for(k=i; k<subframe_size; 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 i,n;
for(n=0; n<buffer_length; n++)
{
int sum = 0x800;
for(i=1; i<filter_length; i++)
sum -= filter_coeffs[i] * out[n-i];
sum = (sum >> 12) + in[n];
/* Check for overflow */
if(sum + 0x8000 > 0xFFFFU)
{
if(stop_on_overflow)
return 1;
sum = (sum >> 31) ^ 32767;
}
out[n] = sum;
}
return 0;
}
void ff_acelp_weighted_filter(
int16_t *out,
const int16_t* in,
const int16_t *weight_pow,
int filter_length)
{
int n;
for(n=0; n<filter_length; n++)
out[n] = (in[n] * weight_pow[n] + 0x4000) >> 15; /* (3.12) = (0.15) * (3.12) with rounding */
}
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 = MULL(hpf_f[0], 15836); /* (14.13) = (13.13) * (1.13) */
tmp += MULL(hpf_f[1], -7667); /* (13.13) = (13.13) * (0.13) */
tmp += 7699 * (in[i] - 2*in[i-1] + in[i-2]); /* (14.13) = (0.13) * (14.0) */
/* Multiplication by 2 with rounding can cause short type
overflow, thus clipping is required. */
out[i] = av_clip_int16((tmp + 0x800) >> 12); /* (15.0) = 2 * (13.13) = (14.13) */
hpf_f[1] = hpf_f[0];
hpf_f[0] = tmp;
}
}

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libavcodec/acelp_filters.h Normal file
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/*
* 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
*/
#ifndef FFMPEG_ACELP_FILTERS_H
#define FFMPEG_ACELP_FILTERS_H
/**
* \brief Circularly convolve fixed vector with a phase dispersion impulse
* response filter (D.6.2 of G.729 and 6.1.5 of AMR).
* \param fc_out vector with filter applied
* \param fc_in source vector
* \param filter phase filter coefficients
*
* fc_out[n] = sum(i,0,len-1){ fc_in[i] * filter[(len + n - i)%len] }
*
* \note fc_in and fc_out should not overlap!
*/
void ff_acelp_convolve_circ(
int16_t* fc_out,
const int16_t* fc_in,
const int16_t* filter,
int subframe_size);
/**
* \brief LP synthesis filter
* \param out [out] pointer to output buffer
* \param filter_coeffs filter coefficients (-0x8000 <= (3.12) < 0x8000)
* \param in input signal
* \param buffer_length amount of data to process
* \param filter_length filter length (11 for 10th order LP filter)
* \param stop_on_overflow 1 - return immediately if overflow occurs
* 0 - ignore overflows
*
* \return 1 if overflow occurred, 0 - otherwise
*
* \note Output buffer must contain 10 samples of past
* speech data before pointer.
*
* Routine applies 1/A(z) filter to given speech data.
*/
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);
/**
* \brief Calculates coefficients of weighted A(z/weight) filter.
* \param out [out] weighted A(z/weight) result
* filter (-0x8000 <= (3.12) < 0x8000)
* \param in source filter (-0x8000 <= (3.12) < 0x8000)
* \param weight_pow array containing weight^i (-0x8000 <= (0.15) < 0x8000)
* \param filter_length filter length (11 for 10th order LP filter)
*
* out[i]=weight_pow[i]*in[i] , i=0..9
*/
void ff_acelp_weighted_filter(
int16_t *out,
const int16_t* in,
const int16_t *weight_pow,
int filter_length);
/**
* \brief high-pass filtering and upscaling (4.2.5 of G.729)
* \param out [out] output buffer for filtered speech data
* \param hpf_f [in/out] past filtered data from previous (2 items long)
* frames (-0x20000000 <= (14.13) < 0x20000000)
* \param in speech data to process
* \param length input data size
*
* out[i] = 0.93980581 * in[i] - 1.8795834 * in[i-1] + 0.93980581 * in[i-2] +
* 1.9330735 * out[i-1] - 0.93589199 * out[i-2]
*
* The filter has a cut-off frequency of 100Hz
*
* \note Two items before the top of the out buffer must contain two items from the
* tail of the previous subframe.
*
* \remark It is safe to pass the same array in in and out parameters
*
* \remark AMR uses mostly the same filter (cut-off frequency 60Hz, same formula,
* but constants differs in 5th sign after comma). Fortunately in
* fixed-point all coefficients are the same as in G.729. Thus this
* routine can be used for the fixed-point AMR decoder, too.
*/
void ff_acelp_high_pass_filter(
int16_t* out,
int hpf_f[2],
const int16_t* in,
int length);
#endif // FFMPEG_ACELP_FILTERS_H