*
* Copyright (c) 2008 Vladimir Voroshilov
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav 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,
+ * Libav 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
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#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)
+
+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 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 n, i;
+
+ assert(frac_pos >= 0 && frac_pos < 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;
}
}
-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)
+void ff_acelp_interpolatef(float *out, const float *in,
+ const float *filter_coeffs, int precision,
+ int frac_pos, int filter_length, int length)
{
- 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;
+ int n, i;
+
+ for (n = 0; n < length; n++) {
+ int idx = 0;
+ float v = 0;
+
+ for (i = 0; i < filter_length;) {
+ v += in[n + i] * filter_coeffs[idx + frac_pos];
+ idx += precision;
+ i++;
+ v += in[n - i] * filter_coeffs[idx - frac_pos];
}
- out[n] = sum;
+ out[n] = v;
}
-
- 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)
+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. */
+ 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]);
- out[i] = av_clip_int16((tmp + 0x800) >> 12); /* (15.0) = 2 * (13.13) = (14.13) */
+ /* 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;
}
}
+
+void ff_acelp_apply_order_2_transfer_function(float *out, const float *in,
+ const float zero_coeffs[2],
+ const float pole_coeffs[2],
+ float gain, float mem[2], int n)
+{
+ int i;
+ float tmp;
+
+ for (i = 0; i < n; i++) {
+ tmp = gain * in[i] - pole_coeffs[0] * mem[0] - pole_coeffs[1] * mem[1];
+ out[i] = tmp + zero_coeffs[0] * mem[0] + zero_coeffs[1] * mem[1];
+
+ mem[1] = mem[0];
+ mem[0] = tmp;
+ }
+}
+
+void ff_tilt_compensation(float *mem, float tilt, float *samples, int size)
+{
+ float new_tilt_mem = samples[size - 1];
+ int i;
+
+ for (i = size - 1; i > 0; i--)
+ samples[i] -= tilt * samples[i - 1];
+
+ samples[0] -= tilt * *mem;
+ *mem = new_tilt_mem;
+}
+
projects / ffmpeg / blobdiff