annotate resample.c @ 5247:32b69770ae23 libavcodec

simplify ff_copy_bits() when compiled with config-small
author aurel
date Sat, 07 Jul 2007 23:03:14 +0000
parents 5c0c96d437f2
children 45d083bbbbe7
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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 Fabrice Bellard.
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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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21
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22 /**
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23 * @file resample.c
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24 * Sample rate convertion for both audio and video.
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25 */
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26
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27 #include "avcodec.h"
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28
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29 struct AVResampleContext;
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30
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31 struct ReSampleContext {
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32 struct AVResampleContext *resample_context;
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33 short *temp[2];
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34 int temp_len;
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35 float ratio;
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36 /* channel convert */
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37 int input_channels, output_channels, filter_channels;
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38 };
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39
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40 /* n1: number of samples */
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41 static void stereo_to_mono(short *output, short *input, int n1)
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42 {
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43 short *p, *q;
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44 int n = n1;
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45
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46 p = input;
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47 q = output;
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48 while (n >= 4) {
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49 q[0] = (p[0] + p[1]) >> 1;
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50 q[1] = (p[2] + p[3]) >> 1;
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51 q[2] = (p[4] + p[5]) >> 1;
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52 q[3] = (p[6] + p[7]) >> 1;
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53 q += 4;
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54 p += 8;
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55 n -= 4;
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56 }
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57 while (n > 0) {
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58 q[0] = (p[0] + p[1]) >> 1;
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59 q++;
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60 p += 2;
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61 n--;
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62 }
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63 }
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64
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65 /* n1: number of samples */
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66 static void mono_to_stereo(short *output, short *input, int n1)
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67 {
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68 short *p, *q;
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69 int n = n1;
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70 int v;
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71
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72 p = input;
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73 q = output;
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74 while (n >= 4) {
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75 v = p[0]; q[0] = v; q[1] = v;
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76 v = p[1]; q[2] = v; q[3] = v;
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77 v = p[2]; q[4] = v; q[5] = v;
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78 v = p[3]; q[6] = v; q[7] = v;
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79 q += 8;
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80 p += 4;
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81 n -= 4;
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82 }
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83 while (n > 0) {
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84 v = p[0]; q[0] = v; q[1] = v;
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85 q += 2;
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86 p += 1;
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87 n--;
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88 }
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89 }
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90
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91 /* XXX: should use more abstract 'N' channels system */
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92 static void stereo_split(short *output1, short *output2, short *input, int n)
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93 {
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94 int i;
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95
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96 for(i=0;i<n;i++) {
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97 *output1++ = *input++;
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98 *output2++ = *input++;
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99 }
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100 }
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101
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102 static void stereo_mux(short *output, short *input1, short *input2, int n)
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103 {
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104 int i;
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105
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106 for(i=0;i<n;i++) {
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107 *output++ = *input1++;
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108 *output++ = *input2++;
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109 }
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110 }
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111
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112 static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
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113 {
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114 int i;
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115 short l,r;
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116
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117 for(i=0;i<n;i++) {
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118 l=*input1++;
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119 r=*input2++;
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120 *output++ = l; /* left */
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121 *output++ = (l/2)+(r/2); /* center */
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122 *output++ = r; /* right */
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123 *output++ = 0; /* left surround */
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124 *output++ = 0; /* right surroud */
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125 *output++ = 0; /* low freq */
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126 }
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127 }
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128
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129 ReSampleContext *audio_resample_init(int output_channels, int input_channels,
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130 int output_rate, int input_rate)
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131 {
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132 ReSampleContext *s;
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133
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134 if ( input_channels > 2)
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135 {
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136 av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported.");
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137 return NULL;
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138 }
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139
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140 s = av_mallocz(sizeof(ReSampleContext));
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141 if (!s)
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142 {
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143 av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.");
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144 return NULL;
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145 }
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146
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147 s->ratio = (float)output_rate / (float)input_rate;
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148
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149 s->input_channels = input_channels;
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150 s->output_channels = output_channels;
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151
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152 s->filter_channels = s->input_channels;
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153 if (s->output_channels < s->filter_channels)
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154 s->filter_channels = s->output_channels;
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155
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156 /*
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157 * ac3 output is the only case where filter_channels could be greater than 2.
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158 * input channels can't be greater than 2, so resample the 2 channels and then
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159 * expand to 6 channels after the resampling.
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160 */
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161 if(s->filter_channels>2)
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162 s->filter_channels = 2;
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163
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164 #define TAPS 16
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165 s->resample_context= av_resample_init(output_rate, input_rate, TAPS, 10, 0, 0.8);
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166
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167 return s;
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168 }
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169
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170 /* resample audio. 'nb_samples' is the number of input samples */
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171 /* XXX: optimize it ! */
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172 int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
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173 {
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174 int i, nb_samples1;
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175 short *bufin[2];
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176 short *bufout[2];
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177 short *buftmp2[2], *buftmp3[2];
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178 int lenout;
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179
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180 if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) {
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181 /* nothing to do */
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182 memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
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183 return nb_samples;
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184 }
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185
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186 /* XXX: move those malloc to resample init code */
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187 for(i=0; i<s->filter_channels; i++){
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188 bufin[i]= (short*) av_malloc( (nb_samples + s->temp_len) * sizeof(short) );
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189 memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
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190 buftmp2[i] = bufin[i] + s->temp_len;
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191 }
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192
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193 /* make some zoom to avoid round pb */
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194 lenout= (int)(nb_samples * s->ratio) + 16;
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195 bufout[0]= (short*) av_malloc( lenout * sizeof(short) );
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196 bufout[1]= (short*) av_malloc( lenout * sizeof(short) );
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197
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198 if (s->input_channels == 2 &&
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199 s->output_channels == 1) {
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200 buftmp3[0] = output;
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201 stereo_to_mono(buftmp2[0], input, nb_samples);
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202 } else if (s->output_channels >= 2 && s->input_channels == 1) {
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203 buftmp3[0] = bufout[0];
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204 memcpy(buftmp2[0], input, nb_samples*sizeof(short));
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205 } else if (s->output_channels >= 2) {
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206 buftmp3[0] = bufout[0];
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207 buftmp3[1] = bufout[1];
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208 stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
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209 } else {
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210 buftmp3[0] = output;
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211 memcpy(buftmp2[0], input, nb_samples*sizeof(short));
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212 }
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213
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214 nb_samples += s->temp_len;
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215
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216 /* resample each channel */
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217 nb_samples1 = 0; /* avoid warning */
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218 for(i=0;i<s->filter_channels;i++) {
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219 int consumed;
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220 int is_last= i+1 == s->filter_channels;
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221
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222 nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i], &consumed, nb_samples, lenout, is_last);
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223 s->temp_len= nb_samples - consumed;
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224 s->temp[i]= av_realloc(s->temp[i], s->temp_len*sizeof(short));
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225 memcpy(s->temp[i], bufin[i] + consumed, s->temp_len*sizeof(short));
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226 }
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227
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228 if (s->output_channels == 2 && s->input_channels == 1) {
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229 mono_to_stereo(output, buftmp3[0], nb_samples1);
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230 } else if (s->output_channels == 2) {
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231 stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
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michaelni
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232 } else if (s->output_channels == 6) {
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233 ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
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234 }
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235
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236 for(i=0; i<s->filter_channels; i++)
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237 av_free(bufin[i]);
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238
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239 av_free(bufout[0]);
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240 av_free(bufout[1]);
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241 return nb_samples1;
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242 }
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243
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244 void audio_resample_close(ReSampleContext *s)
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245 {
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246 av_resample_close(s->resample_context);
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247 av_freep(&s->temp[0]);
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248 av_freep(&s->temp[1]);
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249 av_free(s);
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250 }