X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Facelp_filters.c;h=93bec6589a48ab6aa0a492ac7507b28f587115b1;hb=b7254288d222013e20539c530b1ec5d324ed5352;hp=78937050c353b9b9816451c7181144c6481dcc4b;hpb=cd523888f304d297bb7dec5d358d0ee92576cc44;p=ffmpeg

diff --git a/libavcodec/acelp_filters.c b/libavcodec/acelp_filters.c
index 78937050c35..93bec6589a4 100644
--- a/libavcodec/acelp_filters.c
+++ b/libavcodec/acelp_filters.c
@@ -3,32 +3,30 @@
  *
  * 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 <inttypes.h>
 
+#include "libavutil/common.h"
 #include "avcodec.h"
 #include "acelp_filters.h"
-#define FRAC_BITS 13
-#include "mathops.h"
 
-const int16_t ff_acelp_interp_filter[61] =
-{ /* (0.15) */
+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,
@@ -42,26 +40,19 @@ const int16_t ff_acelp_interp_filter[61] =
       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)
+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);
+    assert(frac_pos >= 0 && frac_pos < precision);
 
-    for(n=0; n<length; n++)
-    {
+    for (n = 0; n < length; n++) {
         int idx = 0;
         int v = 0x4000;
 
-        for(i=0; i<filter_length;)
-        {
+        for (i = 0; i < filter_length;) {
 
             /* The reference G.729 and AMR fixed point code performs clipping after
                each of the two following accumulations.
@@ -77,88 +68,78 @@ void ff_acelp_interpolate(
             i++;
             v += in[n - i] * filter_coeffs[idx - frac_pos];
         }
-        out[n] = av_clip_int16(v >> 15);
+        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)
+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, k;
-
-    memset(fc_out, 0, len * sizeof(int16_t));
+    int n, i;
 
-    /* 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 (n = 0; n < length; n++) {
+        int idx = 0;
+        float v = 0;
 
-            for(k=i; k<len; k++)
-                fc_out[k] += (fc_in[i] * filter[k - i]) >> 15;
+        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] = v;
     }
 }
 
-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--;
+void ff_acelp_high_pass_filter(int16_t* out, int hpf_f[2],
+                               const int16_t* in, int length)
+{
+    int i;
+    int tmp;
 
-    for(n=0; n<buffer_length; n++)
-    {
-        int sum = rounder;
-        for(i=1; i<filter_length; i++)
-            sum -= filter_coeffs[i] * out[n-i];
+    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]);
 
-        sum = (sum >> 12) + in[n];
+        /* With "+0x800" rounding, clipping is needed
+           for ALGTHM and SPEECH tests. */
+        out[i] = av_clip_int16((tmp + 0x800) >> 12);
 
-        if(sum + 0x8000 > 0xFFFFU)
-        {
-            if(stop_on_overflow)
-                return 1;
-            sum = (sum >> 31) ^ 32767;
-        }
-        out[n] = sum;
+        hpf_f[1] = hpf_f[0];
+        hpf_f[0] = tmp;
     }
-
-    return 0;
 }
 
-void ff_acelp_high_pass_filter(
-        int16_t* out,
-        int hpf_f[2],
-        const int16_t* in,
-        int length)
+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;
-    int tmp;
+    float 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) */
+    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];
 
-        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;
+        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;
+}