annotate resample.c @ 36:23723a0ebd24 libavcodec

permuted coefs in normal IDCT to avoid having different cases there
author glantau
date Tue, 07 Aug 2001 22:45:35 +0000
parents 986e461dc072
children 5aa6292a1660
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1 /*
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2 * Sample rate convertion for both audio and video
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3 * Copyright (c) 2000 Gerard Lantau.
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4 *
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5 * This program is free software; you can redistribute it and/or modify
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6 * it under the terms of the GNU General Public License as published by
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7 * the Free Software Foundation; either version 2 of the License, or
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8 * (at your option) any later version.
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9 *
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10 * This program is distributed in the hope that it will be useful,
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11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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13 * GNU General Public License for more details.
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14 *
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15 * You should have received a copy of the GNU General Public License
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16 * along with this program; if not, write to the Free Software
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17 * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
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18 */
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19 #include <stdlib.h>
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20 #include <stdio.h>
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21 #include <string.h>
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22 #include <math.h>
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23 #include "avcodec.h"
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24
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25 #define NDEBUG
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26 #include <assert.h>
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27
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28 typedef struct {
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29 /* fractional resampling */
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30 UINT32 incr; /* fractional increment */
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31 UINT32 frac;
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32 int last_sample;
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33 /* integer down sample */
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34 int iratio; /* integer divison ratio */
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35 int icount, isum;
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36 int inv;
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37 } ReSampleChannelContext;
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38
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39 struct ReSampleContext {
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40 ReSampleChannelContext channel_ctx[2];
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41 float ratio;
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42 /* channel convert */
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43 int input_channels, output_channels, filter_channels;
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44 };
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45
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46
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47 #define FRAC_BITS 16
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48 #define FRAC (1 << FRAC_BITS)
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49
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50 static void init_mono_resample(ReSampleChannelContext *s, float ratio)
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51 {
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52 ratio = 1.0 / ratio;
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53 s->iratio = (int)floor(ratio);
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54 if (s->iratio == 0)
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55 s->iratio = 1;
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56 s->incr = (int)((ratio / s->iratio) * FRAC);
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57 s->frac = 0;
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58 s->last_sample = 0;
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59 s->icount = s->iratio;
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60 s->isum = 0;
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61 s->inv = (FRAC / s->iratio);
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62 }
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63
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64 /* fractional audio resampling */
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65 static int fractional_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
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66 {
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67 unsigned int frac, incr;
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68 int l0, l1;
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69 short *q, *p, *pend;
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70
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71 l0 = s->last_sample;
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72 incr = s->incr;
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73 frac = s->frac;
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74
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75 p = input;
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76 pend = input + nb_samples;
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77 q = output;
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78
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79 l1 = *p++;
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80 for(;;) {
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81 /* interpolate */
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82 *q++ = (l0 * (FRAC - frac) + l1 * frac) >> FRAC_BITS;
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83 frac = frac + s->incr;
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84 while (frac >= FRAC) {
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85 if (p >= pend)
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86 goto the_end;
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87 frac -= FRAC;
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88 l0 = l1;
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89 l1 = *p++;
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90 }
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91 }
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92 the_end:
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93 s->last_sample = l1;
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94 s->frac = frac;
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95 return q - output;
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96 }
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97
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98 static int integer_downsample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
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99 {
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100 short *q, *p, *pend;
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101 int c, sum;
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102
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103 p = input;
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104 pend = input + nb_samples;
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105 q = output;
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106
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107 c = s->icount;
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108 sum = s->isum;
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109
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110 for(;;) {
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111 sum += *p++;
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112 if (--c == 0) {
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113 *q++ = (sum * s->inv) >> FRAC_BITS;
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114 c = s->iratio;
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115 sum = 0;
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116 }
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117 if (p >= pend)
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118 break;
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119 }
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120 s->isum = sum;
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121 s->icount = c;
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122 return q - output;
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123 }
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124
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125 /* n1: number of samples */
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126 static void stereo_to_mono(short *output, short *input, int n1)
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127 {
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128 short *p, *q;
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129 int n = n1;
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130
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131 p = input;
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132 q = output;
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133 while (n >= 4) {
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134 q[0] = (p[0] + p[1]) >> 1;
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135 q[1] = (p[2] + p[3]) >> 1;
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136 q[2] = (p[4] + p[5]) >> 1;
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137 q[3] = (p[6] + p[7]) >> 1;
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138 q += 4;
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139 p += 8;
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140 n -= 4;
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141 }
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142 while (n > 0) {
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143 q[0] = (p[0] + p[1]) >> 1;
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144 q++;
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145 p += 2;
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146 n--;
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147 }
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148 }
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149
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150 /* n1: number of samples */
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151 static void mono_to_stereo(short *output, short *input, int n1)
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152 {
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153 short *p, *q;
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154 int n = n1;
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155 int v;
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156
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157 p = input;
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158 q = output;
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159 while (n >= 4) {
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160 v = p[0]; q[0] = v; q[1] = v;
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161 v = p[1]; q[2] = v; q[3] = v;
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162 v = p[2]; q[4] = v; q[5] = v;
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163 v = p[3]; q[6] = v; q[7] = v;
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164 q += 8;
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165 p += 4;
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166 n -= 4;
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167 }
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168 while (n > 0) {
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169 v = p[0]; q[0] = v; q[1] = v;
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170 q += 2;
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171 p += 1;
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172 n--;
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173 }
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174 }
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175
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176 /* XXX: should use more abstract 'N' channels system */
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177 static void stereo_split(short *output1, short *output2, short *input, int n)
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178 {
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179 int i;
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180
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181 for(i=0;i<n;i++) {
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182 *output1++ = *input++;
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183 *output2++ = *input++;
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184 }
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185 }
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186
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187 static void stereo_mux(short *output, short *input1, short *input2, int n)
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188 {
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189 int i;
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190
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191 for(i=0;i<n;i++) {
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192 *output++ = *input1++;
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193 *output++ = *input2++;
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194 }
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195 }
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196
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197 static int mono_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
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198 {
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199 short buf1[nb_samples];
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200 short *buftmp;
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201
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202 /* first downsample by an integer factor with averaging filter */
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203 if (s->iratio > 1) {
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204 buftmp = buf1;
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205 nb_samples = integer_downsample(s, buftmp, input, nb_samples);
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206 } else {
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207 buftmp = input;
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208 }
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209
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210 /* then do a fractional resampling with linear interpolation */
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211 if (s->incr != FRAC) {
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212 nb_samples = fractional_resample(s, output, buftmp, nb_samples);
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213 } else {
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214 memcpy(output, buftmp, nb_samples * sizeof(short));
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215 }
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216 return nb_samples;
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217 }
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218
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219 ReSampleContext *audio_resample_init(int output_channels, int input_channels,
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220 int output_rate, int input_rate)
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221 {
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222 ReSampleContext *s;
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223 int i;
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224
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225 if (output_channels > 2 || input_channels > 2)
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226 return NULL;
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227
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228 s = av_mallocz(sizeof(ReSampleContext));
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229 if (!s)
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230 return NULL;
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231
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232 s->ratio = (float)output_rate / (float)input_rate;
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233
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234 s->input_channels = input_channels;
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235 s->output_channels = output_channels;
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236
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237 s->filter_channels = s->input_channels;
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238 if (s->output_channels < s->filter_channels)
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239 s->filter_channels = s->output_channels;
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240
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241 for(i=0;i<s->filter_channels;i++) {
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242 init_mono_resample(&s->channel_ctx[i], s->ratio);
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243 }
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244 return s;
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245 }
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246
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247 /* resample audio. 'nb_samples' is the number of input samples */
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248 /* XXX: optimize it ! */
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249 /* XXX: do it with polyphase filters, since the quality here is
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250 HORRIBLE. Return the number of samples available in output */
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251 int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
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252 {
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253 int i, nb_samples1;
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254 short bufin[2][nb_samples];
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255 short bufout[2][(int)(nb_samples * s->ratio) + 16]; /* make some zoom to avoid round pb */
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256 short *buftmp2[2], *buftmp3[2];
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257
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258 if (s->input_channels == s->output_channels && s->ratio == 1.0) {
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259 /* nothing to do */
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260 memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
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261 return nb_samples;
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262 }
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263
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264 if (s->input_channels == 2 &&
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265 s->output_channels == 1) {
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266 buftmp2[0] = bufin[0];
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267 buftmp3[0] = output;
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268 stereo_to_mono(buftmp2[0], input, nb_samples);
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269 } else if (s->output_channels == 2 && s->input_channels == 1) {
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270 buftmp2[0] = input;
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271 buftmp3[0] = bufout[0];
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272 } else if (s->output_channels == 2) {
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273 buftmp2[0] = bufin[0];
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274 buftmp2[1] = bufin[1];
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275 buftmp3[0] = bufout[0];
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276 buftmp3[1] = bufout[1];
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277 stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
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278 } else {
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279 buftmp2[0] = input;
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280 buftmp3[0] = output;
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281 }
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282
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283 /* resample each channel */
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284 nb_samples1 = 0; /* avoid warning */
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285 for(i=0;i<s->filter_channels;i++) {
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286 nb_samples1 = mono_resample(&s->channel_ctx[i], buftmp3[i], buftmp2[i], nb_samples);
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287 }
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288
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289 if (s->output_channels == 2 && s->input_channels == 1) {
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290 mono_to_stereo(output, buftmp3[0], nb_samples1);
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291 } else if (s->output_channels == 2) {
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292 stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
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293 }
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294
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295 return nb_samples1;
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296 }
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297
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298 void audio_resample_close(ReSampleContext *s)
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299 {
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300 free(s);
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301 }