annotate libao2/pl_surround.c @ 10524:3eeb6cdddea1

spelling corrections
author diego
date Mon, 04 Aug 2003 22:36:22 +0000
parents 12b1790038b0
children 815f03b7cee5
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1 /*
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2 This is an ao2 plugin to do simple decoding of matrixed surround
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3 sound. This will provide a (basic) surround-sound effect from
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4 audio encoded for Dolby Surround, Pro Logic etc.
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5
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6 * This program is free software; you can redistribute it and/or modify
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7 * it under the terms of the GNU General Public License as published by
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8 * the Free Software Foundation; either version 2 of the License, or
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9 * (at your option) any later version.
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10 *
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11 * This program is distributed in the hope that it will be useful,
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12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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14 * GNU General Public License for more details.
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15 *
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16 * You should have received a copy of the GNU General Public License
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17 * along with this program; if not, write to the Free Software
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18 * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
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19
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20 Original author: Steve Davies <steve@daviesfam.org>
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21 */
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22
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23 /* The principle: Make rear channels by extracting anti-phase data
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24 from the front channels, delay by 20msec and feed to rear in anti-phase
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25 */
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26
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27
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28 // SPLITREAR: Define to decode two distinct rear channels -
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29 // this doesn't work so well in practice because
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30 // separation in a passive matrix is not high.
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31 // C (dialogue) to Ls and Rs 14dB or so -
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32 // so dialogue leaks to the rear.
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33 // Still - give it a try and send feedback.
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34 // comment this define for old behaviour of a single
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35 // surround sent to rear in anti-phase
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36 #define SPLITREAR
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37
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38
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39 #include <stdio.h>
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40 #include <stdlib.h>
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41 #include <string.h>
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42 #include <unistd.h>
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43
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44 #include "audio_out.h"
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45 #include "audio_plugin.h"
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46 #include "audio_plugin_internal.h"
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47 #include "afmt.h"
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48
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49 #include "remez.h"
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50 #include "firfilter.c"
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51
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52 static ao_info_t info =
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53 {
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54 "Surround decoder plugin",
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55 "surround",
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56 "Steve Davies <steve@daviesfam.org>",
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57 ""
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58 };
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59
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60 LIBAO_PLUGIN_EXTERN(surround)
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61
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62 // local data
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63 typedef struct pl_surround_s
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64 {
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65 int passthrough; // Just be a "NO-OP"
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66 int msecs; // Rear channel delay in milliseconds
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67 int16_t* databuf; // Output audio buffer
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68 int16_t* Ls_delaybuf; // circular buffer to be used for delaying Ls audio
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69 int16_t* Rs_delaybuf; // circular buffer to be used for delaying Rs audio
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70 int delaybuf_len; // delaybuf buffer length in samples
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71 int delaybuf_pos; // offset in buffer where we are reading/writing
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72 double* filter_coefs_surround; // FIR filter coefficients for surround sound 7kHz lowpass
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73 int rate; // input data rate
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74 int format; // input format
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75 int input_channels; // input channels
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76
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77 } pl_surround_t;
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78
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79 static pl_surround_t pl_surround={0,20,NULL,NULL,NULL,0,0,NULL,0,0,0};
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80
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81 // to set/get/query special features/parameters
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82 static int control(int cmd,void *arg){
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83 switch(cmd){
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84 case AOCONTROL_PLUGIN_SET_LEN:
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85 if (pl_surround.passthrough) return CONTROL_OK;
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86 //fprintf(stderr, "pl_surround: AOCONTROL_PLUGIN_SET_LEN with arg=%d\n", arg);
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87 //fprintf(stderr, "pl_surround: ao_plugin_data.len=%d\n", ao_plugin_data.len);
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88 // Allocate an output buffer
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89 if (pl_surround.databuf != NULL) {
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90 free(pl_surround.databuf); pl_surround.databuf = NULL;
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91 }
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92 // Allocate output buffer
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93 pl_surround.databuf = calloc(ao_plugin_data.len, 1);
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94 // Return back smaller len so we don't get overflowed...
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95 ao_plugin_data.len /= 2;
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96 return CONTROL_OK;
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97 }
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98 return -1;
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99 }
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100
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101 // open & setup audio device
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102 // return: 1=success 0=fail
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103 static int init(){
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104
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105 fprintf(stderr, "pl_surround: init input rate=%d, channels=%d\n", ao_plugin_data.rate, ao_plugin_data.channels);
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106 if (ao_plugin_data.channels != 2) {
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107 fprintf(stderr, "pl_surround: source audio must have 2 channels, using passthrough mode\n");
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108 pl_surround.passthrough = 1;
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109 return 1;
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110 }
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111 if (ao_plugin_data.format != AFMT_S16_NE) {
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112 fprintf(stderr, "pl_surround: I'm dumb and can only handle AFMT_S16_NE audio format, using passthrough mode\n");
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113 pl_surround.passthrough = 1;
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114 return 1;
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115 }
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116
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117 pl_surround.passthrough = 0;
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118
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119 /* Store info on input format to expect */
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120 pl_surround.rate=ao_plugin_data.rate;
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121 pl_surround.format=ao_plugin_data.format;
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122 pl_surround.input_channels=ao_plugin_data.channels;
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123
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124 // Input 2 channels, output will be 4 - tell ao_plugin
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125 ao_plugin_data.channels = 4;
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126 ao_plugin_data.sz_mult /= 2;
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127
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128 // Figure out buffer space (in int16_ts) needed for the 15msec delay
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129 // Extra 31 samples allow for lowpass filter delay (taps-1)
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130 pl_surround.delaybuf_len = (pl_surround.rate * pl_surround.msecs / 1000) + 31;
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131 // Allocate delay buffers
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132 pl_surround.Ls_delaybuf=(void*)calloc(pl_surround.delaybuf_len,sizeof(int16_t));
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133 pl_surround.Rs_delaybuf=(void*)calloc(pl_surround.delaybuf_len,sizeof(int16_t));
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134 fprintf(stderr, "pl_surround: %dmsec surround delay, rate %d - buffers are %d bytes each\n",
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135 pl_surround.msecs,pl_surround.rate, pl_surround.delaybuf_len*sizeof(int16_t));
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136 pl_surround.delaybuf_pos = 0;
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137 // Surround filer coefficients
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138 pl_surround.filter_coefs_surround = calc_coefficients_7kHz_lowpass(pl_surround.rate);
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139 //dump_filter_coefficients(pl_surround.filter_coefs_surround);
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140 //testfilter(pl_surround.filter_coefs_surround, 32, pl_surround.rate);
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141 return 1;
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142 }
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143
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144 // close plugin
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145 static void uninit(){
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146 // fprintf(stderr, "pl_surround: uninit called!\n");
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147 if (pl_surround.passthrough) return;
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148 if(pl_surround.Ls_delaybuf)
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149 free(pl_surround.Ls_delaybuf);
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150 if(pl_surround.Rs_delaybuf)
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151 free(pl_surround.Rs_delaybuf);
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152 if(pl_surround.databuf) {
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153 free(pl_surround.databuf);
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154 pl_surround.databuf = NULL;
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155 }
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156 pl_surround.delaybuf_len=0;
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157 }
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158
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159 // empty buffers
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160 static void reset()
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161 {
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162 if (pl_surround.passthrough) return;
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163 //fprintf(stderr, "pl_surround: reset called\n");
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164 pl_surround.delaybuf_pos = 0;
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165 memset(pl_surround.Ls_delaybuf, 0, sizeof(int16_t)*pl_surround.delaybuf_len);
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166 memset(pl_surround.Rs_delaybuf, 0, sizeof(int16_t)*pl_surround.delaybuf_len);
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167 }
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168
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169 // The beginnings of an active matrix...
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170 static double steering_matrix[][12] = {
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171 // LL RL LR RR LS RS LLs RLs LRs RRs LC RC
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172 {.707, .0, .0, .707, .5, -.5, .5878, -.3928, .3928, -.5878, .5, .5},
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173 };
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174
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175 // Experimental moving average dominances
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176 //static int amp_L = 0, amp_R = 0, amp_C = 0, amp_S = 0;
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177
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178 // processes 'ao_plugin_data.len' bytes of 'data'
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179 // called for every block of data
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180 static int play(){
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181 int16_t *in, *out;
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182 int i, samples;
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183 double *matrix = steering_matrix[0]; // later we'll index based on detected dominance
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184
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185 if (pl_surround.passthrough) return 1;
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186
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187 // fprintf(stderr, "pl_surround: play %d bytes, %d samples\n", ao_plugin_data.len, samples);
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188
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189 samples = ao_plugin_data.len / sizeof(int16_t) / pl_surround.input_channels;
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190 out = pl_surround.databuf; in = (int16_t *)ao_plugin_data.data;
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191
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192 // Testing - place a 1kHz tone on Lt and Rt in anti-phase: should decode in S
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193 //sinewave(in, samples, pl_surround.input_channels, 1000, 0.0, pl_surround.rate);
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194 //sinewave(&in[1], samples, pl_surround.input_channels, 1000, PI, pl_surround.rate);
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195
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196 for (i=0; i<samples; i++) {
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197
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198 // Dominance:
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199 //abs(in[0]) abs(in[1]);
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200 //abs(in[0]+in[1]) abs(in[0]-in[1]);
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201 //10 * log( abs(in[0]) / (abs(in[1])|1) );
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202 //10 * log( abs(in[0]+in[1]) / (abs(in[0]-in[1])|1) );
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203
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204 // About volume balancing...
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205 // Surround encoding does the following:
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206 // Lt=L+.707*C+.707*S, Rt=R+.707*C-.707*S
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207 // So S should be extracted as:
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208 // (Lt-Rt)
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209 // But we are splitting the S to two output channels, so we
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210 // must take 3dB off as we split it:
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211 // Ls=Rs=.707*(Lt-Rt)
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212 // Trouble is, Lt could be +32767, Rt -32768, so possibility that S will
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213 // overflow. So to avoid that, we cut L/R by 3dB (*.707), and S by 6dB (/2).
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214 // this keeps the overall balance, but guarantees no overflow.
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215
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216 // output front left and right
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217 out[0] = matrix[0]*in[0] + matrix[1]*in[1];
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218 out[1] = matrix[2]*in[0] + matrix[3]*in[1];
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219 // output Ls and Rs - from 20msec ago, lowpass filtered @ 7kHz
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220 out[2] = firfilter(pl_surround.Ls_delaybuf, pl_surround.delaybuf_pos,
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221 pl_surround.delaybuf_len, 32, pl_surround.filter_coefs_surround);
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222 #ifdef SPLITREAR
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223 out[3] = firfilter(pl_surround.Rs_delaybuf, pl_surround.delaybuf_pos,
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224 pl_surround.delaybuf_len, 32, pl_surround.filter_coefs_surround);
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225 #else
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226 out[3] = -out[2];
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227 #endif
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228 // calculate and save surround for 20msecs time
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229 #ifdef SPLITREAR
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230 pl_surround.Ls_delaybuf[pl_surround.delaybuf_pos] =
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231 matrix[6]*in[0] + matrix[7]*in[1];
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232 pl_surround.Rs_delaybuf[pl_surround.delaybuf_pos++] =
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233 matrix[8]*in[0] + matrix[9]*in[1];
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234 #else
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235 pl_surround.Ls_delaybuf[pl_surround.delaybuf_pos++] =
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236 matrix[4]*in[0] + matrix[5]*in[1];
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237 #endif
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238 pl_surround.delaybuf_pos %= pl_surround.delaybuf_len;
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239
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240 // next samples...
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241 in = &in[pl_surround.input_channels]; out = &out[4];
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242 }
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243
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244 // Show some state
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245 //printf("\npl_surround: delaybuf_pos=%d, samples=%d\r\033[A", pl_surround.delaybuf_pos, samples);
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246
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247 // Set output block/len
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248 ao_plugin_data.data=pl_surround.databuf;
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249 ao_plugin_data.len=samples*sizeof(int16_t)*4;
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250 return 1;
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251 }