Mercurial > libavcodec.hg
annotate resample2.c @ 12197:fbf4d5b1b664 libavcodec
Remove FF_MM_SSE2/3 flags for CPUs where this is generally not faster than
regular MMX code. Examples of this are the Core1 CPU. Instead, set a new flag,
FF_MM_SSE2/3SLOW, which can be checked for particular SSE2/3 functions that
have been checked specifically on such CPUs and are actually faster than
their MMX counterparts.
In addition, use this flag to enable particular VP8 and LPC SSE2 functions
that are faster than their MMX counterparts.
Based on a patch by Loren Merritt <lorenm AT u washington edu>.
author | rbultje |
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date | Mon, 19 Jul 2010 22:38:23 +0000 |
parents | ee69b90df099 |
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1 /* |
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2 * audio resampling |
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3 * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at> |
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4 * |
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5 * This file is part of FFmpeg. |
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6 * |
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7 * FFmpeg is free software; you can redistribute it and/or |
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8 * modify it under the terms of the GNU Lesser General Public |
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9 * License as published by the Free Software Foundation; either |
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10 * version 2.1 of the License, or (at your option) any later version. |
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11 * |
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12 * FFmpeg is distributed in the hope that it will be useful, |
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13 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
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14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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15 * Lesser General Public License for more details. |
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16 * |
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17 * You should have received a copy of the GNU Lesser General Public |
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18 * License along with FFmpeg; if not, write to the Free Software |
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19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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20 */ |
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22 /** |
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23 * @file |
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24 * audio resampling |
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25 * @author Michael Niedermayer <michaelni@gmx.at> |
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26 */ |
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27 |
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28 #include "avcodec.h" |
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29 #include "dsputil.h" |
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30 |
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31 #ifndef CONFIG_RESAMPLE_HP |
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32 #define FILTER_SHIFT 15 |
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33 |
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34 #define FELEM int16_t |
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35 #define FELEM2 int32_t |
4702 | 36 #define FELEML int64_t |
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37 #define FELEM_MAX INT16_MAX |
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38 #define FELEM_MIN INT16_MIN |
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39 #define WINDOW_TYPE 9 |
4702 | 40 #elif !defined(CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE) |
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41 #define FILTER_SHIFT 30 |
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42 |
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43 #define FELEM int32_t |
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44 #define FELEM2 int64_t |
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46 #define FELEM_MAX INT32_MAX |
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47 #define FELEM_MIN INT32_MIN |
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48 #define WINDOW_TYPE 12 |
4702 | 49 #else |
50 #define FILTER_SHIFT 0 | |
51 | |
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52 #define FELEM double |
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53 #define FELEM2 double |
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54 #define FELEML double |
4702 | 55 #define WINDOW_TYPE 24 |
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56 #endif |
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57 |
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58 |
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59 typedef struct AVResampleContext{ |
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60 const AVClass *av_class; |
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61 FELEM *filter_bank; |
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62 int filter_length; |
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63 int ideal_dst_incr; |
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64 int dst_incr; |
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65 int index; |
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66 int frac; |
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67 int src_incr; |
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68 int compensation_distance; |
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69 int phase_shift; |
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70 int phase_mask; |
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71 int linear; |
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72 }AVResampleContext; |
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73 |
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74 /** |
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75 * 0th order modified bessel function of the first kind. |
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76 */ |
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77 static double bessel(double x){ |
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78 double v=1; |
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79 double lastv=0; |
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80 double t=1; |
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81 int i; |
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83 x= x*x/4; |
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84 for(i=1; v != lastv; i++){ |
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85 lastv=v; |
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86 t *= x/(i*i); |
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87 v += t; |
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88 } |
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89 return v; |
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90 } |
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91 |
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92 /** |
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93 * builds a polyphase filterbank. |
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94 * @param factor resampling factor |
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95 * @param scale wanted sum of coefficients for each filter |
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96 * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2..16->kaiser windowed sinc beta=2..16 |
12121 | 97 * @return 0 on success, negative on error |
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98 */ |
12121 | 99 static int build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){ |
4822 | 100 int ph, i; |
12121 | 101 double x, y, w; |
102 double *tab = av_malloc(tap_count * sizeof(*tab)); | |
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103 const int center= (tap_count-1)/2; |
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104 |
12121 | 105 if (!tab) |
106 return AVERROR(ENOMEM); | |
107 | |
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108 /* if upsampling, only need to interpolate, no filter */ |
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109 if (factor > 1.0) |
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110 factor = 1.0; |
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111 |
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112 for(ph=0;ph<phase_count;ph++) { |
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113 double norm = 0; |
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114 for(i=0;i<tap_count;i++) { |
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115 x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor; |
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116 if (x == 0) y = 1.0; |
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117 else y = sin(x) / x; |
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118 switch(type){ |
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119 case 0:{ |
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120 const float d= -0.5; //first order derivative = -0.5 |
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121 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor); |
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122 if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x); |
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123 else y= d*(-4 + 8*x - 5*x*x + x*x*x); |
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124 break;} |
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125 case 1: |
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126 w = 2.0*x / (factor*tap_count) + M_PI; |
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127 y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w); |
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128 break; |
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129 default: |
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130 w = 2.0*x / (factor*tap_count*M_PI); |
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131 y *= bessel(type*sqrt(FFMAX(1-w*w, 0))); |
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132 break; |
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133 } |
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134 |
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135 tab[i] = y; |
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136 norm += y; |
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137 } |
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138 |
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139 /* normalize so that an uniform color remains the same */ |
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140 for(i=0;i<tap_count;i++) { |
4702 | 141 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE |
142 filter[ph * tap_count + i] = tab[i] / norm; | |
143 #else | |
144 filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX); | |
145 #endif | |
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146 } |
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147 } |
4695 | 148 #if 0 |
149 { | |
150 #define LEN 1024 | |
151 int j,k; | |
152 double sine[LEN + tap_count]; | |
153 double filtered[LEN]; | |
154 double maxff=-2, minff=2, maxsf=-2, minsf=2; | |
155 for(i=0; i<LEN; i++){ | |
156 double ss=0, sf=0, ff=0; | |
157 for(j=0; j<LEN+tap_count; j++) | |
158 sine[j]= cos(i*j*M_PI/LEN); | |
159 for(j=0; j<LEN; j++){ | |
160 double sum=0; | |
161 ph=0; | |
162 for(k=0; k<tap_count; k++) | |
163 sum += filter[ph * tap_count + k] * sine[k+j]; | |
164 filtered[j]= sum / (1<<FILTER_SHIFT); | |
165 ss+= sine[j + center] * sine[j + center]; | |
166 ff+= filtered[j] * filtered[j]; | |
167 sf+= sine[j + center] * filtered[j]; | |
168 } | |
169 ss= sqrt(2*ss/LEN); | |
170 ff= sqrt(2*ff/LEN); | |
171 sf= 2*sf/LEN; | |
172 maxff= FFMAX(maxff, ff); | |
173 minff= FFMIN(minff, ff); | |
174 maxsf= FFMAX(maxsf, sf); | |
175 minsf= FFMIN(minsf, sf); | |
176 if(i%11==0){ | |
4702 | 177 av_log(NULL, AV_LOG_ERROR, "i:%4d ss:%f ff:%13.6e-%13.6e sf:%13.6e-%13.6e\n", i, ss, maxff, minff, maxsf, minsf); |
4695 | 178 minff=minsf= 2; |
179 maxff=maxsf= -2; | |
180 } | |
181 } | |
182 } | |
183 #endif | |
12121 | 184 |
185 av_free(tab); | |
186 return 0; | |
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187 } |
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188 |
2308 | 189 AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){ |
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190 AVResampleContext *c= av_mallocz(sizeof(AVResampleContext)); |
2308 | 191 double factor= FFMIN(out_rate * cutoff / in_rate, 1.0); |
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192 int phase_count= 1<<phase_shift; |
2967 | 193 |
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194 if (!c) |
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195 return NULL; |
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196 |
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197 c->phase_shift= phase_shift; |
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198 c->phase_mask= phase_count-1; |
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199 c->linear= linear; |
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200 |
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201 c->filter_length= FFMAX((int)ceil(filter_size/factor), 1); |
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202 c->filter_bank= av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM)); |
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203 if (!c->filter_bank) |
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204 goto error; |
12121 | 205 if (build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE)) |
206 goto error; | |
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207 memcpy(&c->filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM)); |
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208 c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1]; |
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209 |
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210 c->src_incr= out_rate; |
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211 c->ideal_dst_incr= c->dst_incr= in_rate * phase_count; |
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212 c->index= -phase_count*((c->filter_length-1)/2); |
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213 |
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214 return c; |
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215 error: |
12121 | 216 av_free(c->filter_bank); |
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217 av_free(c); |
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218 return NULL; |
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219 } |
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220 |
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221 void av_resample_close(AVResampleContext *c){ |
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222 av_freep(&c->filter_bank); |
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223 av_freep(&c); |
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224 } |
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225 |
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226 void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){ |
2083 | 227 // sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr; |
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228 c->compensation_distance= compensation_distance; |
2083 | 229 c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance; |
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230 } |
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231 |
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232 int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){ |
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233 int dst_index, i; |
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234 int index= c->index; |
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235 int frac= c->frac; |
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236 int dst_incr_frac= c->dst_incr % c->src_incr; |
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237 int dst_incr= c->dst_incr / c->src_incr; |
2277 | 238 int compensation_distance= c->compensation_distance; |
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239 |
2403 | 240 if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){ |
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241 int64_t index2= ((int64_t)index)<<32; |
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242 int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr; |
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243 dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr); |
2967 | 244 |
2403 | 245 for(dst_index=0; dst_index < dst_size; dst_index++){ |
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246 dst[dst_index] = src[index2>>32]; |
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247 index2 += incr; |
2403 | 248 } |
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249 frac += dst_index * dst_incr_frac; |
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250 index += dst_index * dst_incr; |
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251 index += frac / c->src_incr; |
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252 frac %= c->src_incr; |
2403 | 253 }else{ |
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254 for(dst_index=0; dst_index < dst_size; dst_index++){ |
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255 FELEM *filter= c->filter_bank + c->filter_length*(index & c->phase_mask); |
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256 int sample_index= index >> c->phase_shift; |
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257 FELEM2 val=0; |
2967 | 258 |
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259 if(sample_index < 0){ |
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260 for(i=0; i<c->filter_length; i++) |
4001 | 261 val += src[FFABS(sample_index + i) % src_size] * filter[i]; |
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262 }else if(sample_index + c->filter_length > src_size){ |
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263 break; |
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264 }else if(c->linear){ |
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265 FELEM2 v2=0; |
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266 for(i=0; i<c->filter_length; i++){ |
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267 val += src[sample_index + i] * (FELEM2)filter[i]; |
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268 v2 += src[sample_index + i] * (FELEM2)filter[i + c->filter_length]; |
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269 } |
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270 val+=(v2-val)*(FELEML)frac / c->src_incr; |
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271 }else{ |
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272 for(i=0; i<c->filter_length; i++){ |
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273 val += src[sample_index + i] * (FELEM2)filter[i]; |
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274 } |
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275 } |
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276 |
4702 | 277 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE |
5523 | 278 dst[dst_index] = av_clip_int16(lrintf(val)); |
4702 | 279 #else |
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280 val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT; |
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281 dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val; |
4702 | 282 #endif |
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283 |
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284 frac += dst_incr_frac; |
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285 index += dst_incr; |
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286 if(frac >= c->src_incr){ |
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287 frac -= c->src_incr; |
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288 index++; |
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289 } |
2277 | 290 |
291 if(dst_index + 1 == compensation_distance){ | |
292 compensation_distance= 0; | |
293 dst_incr_frac= c->ideal_dst_incr % c->src_incr; | |
294 dst_incr= c->ideal_dst_incr / c->src_incr; | |
295 } | |
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296 } |
2403 | 297 } |
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298 *consumed= FFMAX(index, 0) >> c->phase_shift; |
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299 if(index>=0) index &= c->phase_mask; |
2109 | 300 |
2277 | 301 if(compensation_distance){ |
302 compensation_distance -= dst_index; | |
303 assert(compensation_distance > 0); | |
304 } | |
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305 if(update_ctx){ |
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306 c->frac= frac; |
2109 | 307 c->index= index; |
2277 | 308 c->dst_incr= dst_incr_frac + c->src_incr*dst_incr; |
309 c->compensation_distance= compensation_distance; | |
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310 } |
2967 | 311 #if 0 |
2083 | 312 if(update_ctx && !c->compensation_distance){ |
313 #undef rand | |
314 av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2); | |
315 av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance); | |
316 } | |
317 #endif | |
2967 | 318 |
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319 return dst_index; |
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320 } |