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avcodec/g728_template: do_hybrid_window() template

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intended for use by RealAudio 2.0 (28.8k) and G.728 decoders.
This commit is contained in:
Peter Ross
2024-12-22 08:46:32 +11:00
parent b1172b8cc6
commit 93368029e3
2 changed files with 68 additions and 47 deletions
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65
libavcodec/g728_template.c Normal file
View File

@ -0,0 +1,65 @@
/*
* G.728 / RealAudio 2.0 (28.8K) decoder
*
* 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
*/
static void convolve(float *tgt, const float *src, int len, int n)
{
for (; n >= 0; n--)
tgt[n] = ff_scalarproduct_float_c(src, src - n, len);
}
/**
* Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
*
* @param order filter order
* @param n input length
* @param non_rec number of non-recursive samples
* @param out filter output
* @param hist pointer to the input history of the filter
* @param out pointer to the non-recursive part of the output
* @param out2 pointer to the recursive part of the output
* @param window pointer to the windowing function table
*/
static void do_hybrid_window(void (*vector_fmul)(float *dst, const float *src0, const float *src1, int len),
int order, int n, int non_rec, float *out,
float *hist, float *out2, const float *window)
{
int i;
float buffer1[MAX_BACKWARD_FILTER_ORDER + 1];
float buffer2[MAX_BACKWARD_FILTER_ORDER + 1];
LOCAL_ALIGNED(32, float, work, [FFALIGN(MAX_BACKWARD_FILTER_ORDER +
MAX_BACKWARD_FILTER_LEN +
MAX_BACKWARD_FILTER_NONREC, 16)]);
av_assert2(order>=0);
vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 16));
convolve(buffer1, work + order , n , order);
convolve(buffer2, work + order + n, non_rec, order);
for (i=0; i <= order; i++) {
out2[i] = out2[i] * ATTEN + buffer1[i];
out [i] = out2[i] + buffer2[i];
}
/* Multiply by the white noise correcting factor (WNCF). */
*out *= 257.0 / 256.0;
}

50
libavcodec/ra288.c
View File

@ -37,6 +37,8 @@
#define MAX_BACKWARD_FILTER_ORDER 36 #define MAX_BACKWARD_FILTER_ORDER 36
#define MAX_BACKWARD_FILTER_LEN 40 #define MAX_BACKWARD_FILTER_LEN 40
#define MAX_BACKWARD_FILTER_NONREC 35 #define MAX_BACKWARD_FILTER_NONREC 35
#define ATTEN 0.5625
#include "g728_template.c"
#define RA288_BLOCK_SIZE 5 #define RA288_BLOCK_SIZE 5
#define RA288_BLOCKS_PER_FRAME 32 #define RA288_BLOCKS_PER_FRAME 32
@ -87,13 +89,6 @@ static av_cold int ra288_decode_init(AVCodecContext *avctx)
return 0; return 0;
} }
static void convolve(float *tgt, const float *src, int len, int n)
{
for (; n >= 0; n--)
tgt[n] = ff_scalarproduct_float_c(src, src - n, len);
}
static void decode(RA288Context *ractx, float gain, int cb_coef) static void decode(RA288Context *ractx, float gain, int cb_coef)
{ {
int i; int i;
@ -131,45 +126,6 @@ static void decode(RA288Context *ractx, float gain, int cb_coef)
ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36); ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36);
} }
/**
* Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
*
* @param order filter order
* @param n input length
* @param non_rec number of non-recursive samples
* @param out filter output
* @param hist pointer to the input history of the filter
* @param out pointer to the non-recursive part of the output
* @param out2 pointer to the recursive part of the output
* @param window pointer to the windowing function table
*/
static void do_hybrid_window(RA288Context *ractx,
int order, int n, int non_rec, float *out,
float *hist, float *out2, const float *window)
{
int i;
float buffer1[MAX_BACKWARD_FILTER_ORDER + 1];
float buffer2[MAX_BACKWARD_FILTER_ORDER + 1];
LOCAL_ALIGNED(32, float, work, [FFALIGN(MAX_BACKWARD_FILTER_ORDER +
MAX_BACKWARD_FILTER_LEN +
MAX_BACKWARD_FILTER_NONREC, 16)]);
av_assert2(order>=0);
ractx->vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 16));
convolve(buffer1, work + order , n , order);
convolve(buffer2, work + order + n, non_rec, order);
for (i=0; i <= order; i++) {
out2[i] = out2[i] * 0.5625 + buffer1[i];
out [i] = out2[i] + buffer2[i];
}
/* Multiply by the white noise correcting factor (WNCF). */
*out *= 257.0 / 256.0;
}
/** /**
* Backward synthesis filter, find the LPC coefficients from past speech data. * Backward synthesis filter, find the LPC coefficients from past speech data.
*/ */
@ -180,7 +136,7 @@ static void backward_filter(RA288Context *ractx,
{ {
float temp[MAX_BACKWARD_FILTER_ORDER+1]; float temp[MAX_BACKWARD_FILTER_ORDER+1];
do_hybrid_window(ractx, order, n, non_rec, temp, hist, rec, window); do_hybrid_window(ractx->vector_fmul, order, n, non_rec, temp, hist, rec, window);
if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1)) if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1))
ractx->vector_fmul(lpc, lpc, tab, FFALIGN(order, 16)); ractx->vector_fmul(lpc, lpc, tab, FFALIGN(order, 16));