annotate libao2/pl_surround.c @ 5041:8a708971d372

Fixed swapping of UV planes in single buffered mode
author nick
date Mon, 11 Mar 2002 07:46:04 +0000
parents d358dc143a9e
children 8336b1cf8d70
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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 <unistd.h>
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42
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43 #include "audio_out.h"
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44 #include "audio_plugin.h"
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45 #include "audio_plugin_internal.h"
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46 #include "afmt.h"
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47
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48 #include "remez.h"
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49 #include "firfilter.c"
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50
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51 static ao_info_t info =
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52 {
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53 "Surround decoder plugin",
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54 "surround",
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55 "Steve Davies <steve@daviesfam.org>",
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56 ""
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57 };
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58
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59 LIBAO_PLUGIN_EXTERN(surround)
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60
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61 // local data
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62 typedef struct pl_surround_s
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63 {
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64 int passthrough; // Just be a "NO-OP"
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65 int msecs; // Rear channel delay in milliseconds
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66 int16_t* databuf; // Output audio buffer
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67 int16_t* Ls_delaybuf; // circular buffer to be used for delaying Ls audio
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68 int16_t* Rs_delaybuf; // circular buffer to be used for delaying Rs audio
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69 int delaybuf_len; // delaybuf buffer length in samples
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70 int delaybuf_pos; // offset in buffer where we are reading/writing
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71 double* filter_coefs_surround; // FIR filter coefficients for surround sound 7kHz lowpass
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72 int rate; // input data rate
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73 int format; // input format
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74 int input_channels; // input channels
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75
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76 } pl_surround_t;
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77
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78 static pl_surround_t pl_surround={0,20,NULL,NULL,NULL,0,0,NULL,0,0,0};
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79
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80 // to set/get/query special features/parameters
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81 static int control(int cmd,int arg){
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82 switch(cmd){
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83 case AOCONTROL_PLUGIN_SET_LEN:
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84 if (pl_surround.passthrough) return CONTROL_OK;
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85 //fprintf(stderr, "pl_surround: AOCONTROL_PLUGIN_SET_LEN with arg=%d\n", arg);
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86 //fprintf(stderr, "pl_surround: ao_plugin_data.len=%d\n", ao_plugin_data.len);
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87 // Allocate an output buffer
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88 if (pl_surround.databuf != NULL) {
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89 free(pl_surround.databuf); pl_surround.databuf = NULL;
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90 }
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91 // Allocate output buffer
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92 pl_surround.databuf = calloc(ao_plugin_data.len, 1);
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93 // Return back smaller len so we don't get overflowed...
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94 ao_plugin_data.len /= 2;
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95 return CONTROL_OK;
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96 }
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97 return -1;
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98 }
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99
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100 // open & setup audio device
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101 // return: 1=success 0=fail
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102 static int init(){
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103
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104 fprintf(stderr, "pl_surround: init input rate=%d, channels=%d\n", ao_plugin_data.rate, ao_plugin_data.channels);
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105 if (ao_plugin_data.channels != 2) {
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106 fprintf(stderr, "pl_surround: source audio must have 2 channels, using passthrough mode\n");
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107 pl_surround.passthrough = 1;
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108 return 1;
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109 }
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110 if (ao_plugin_data.format != AFMT_S16_LE) {
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111 fprintf(stderr, "pl_surround: I'm dumb and can only handle AFMT_S16_LE audio format, using passthrough mode\n");
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112 pl_surround.passthrough = 1;
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113 return 1;
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114 }
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115
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116 pl_surround.passthrough = 0;
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117
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118 /* Store info on input format to expect */
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119 pl_surround.rate=ao_plugin_data.rate;
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120 pl_surround.format=ao_plugin_data.format;
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121 pl_surround.input_channels=ao_plugin_data.channels;
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122
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123 // Input 2 channels, output will be 4 - tell ao_plugin
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124 ao_plugin_data.channels = 4;
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125 ao_plugin_data.sz_mult /= 2;
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126
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127 // Figure out buffer space (in int16_ts) needed for the 15msec delay
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128 // Extra 31 samples allow for lowpass filter delay (taps-1)
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129 pl_surround.delaybuf_len = (pl_surround.rate * pl_surround.msecs / 1000) + 31;
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130 // Allocate delay buffers
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131 pl_surround.Ls_delaybuf=(void*)calloc(pl_surround.delaybuf_len,sizeof(int16_t));
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132 pl_surround.Rs_delaybuf=(void*)calloc(pl_surround.delaybuf_len,sizeof(int16_t));
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133 fprintf(stderr, "pl_surround: %dmsec surround delay, rate %d - buffers are %d bytes each\n",
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134 pl_surround.msecs,pl_surround.rate, pl_surround.delaybuf_len*sizeof(int16_t));
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135 pl_surround.delaybuf_pos = 0;
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136 // Surround filer coefficients
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137 pl_surround.filter_coefs_surround = calc_coefficients_7kHz_lowpass(pl_surround.rate);
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138 //dump_filter_coefficients(pl_surround.filter_coefs_surround);
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139 //testfilter(pl_surround.filter_coefs_surround, 32, pl_surround.rate);
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140 return 1;
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141 }
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142
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143 // close plugin
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144 static void uninit(){
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145 // fprintf(stderr, "pl_surround: uninit called!\n");
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146 if (pl_surround.passthrough) return;
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147 if(pl_surround.Ls_delaybuf)
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148 free(pl_surround.Ls_delaybuf);
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149 if(pl_surround.Rs_delaybuf)
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150 free(pl_surround.Rs_delaybuf);
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151 if(pl_surround.databuf)
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152 free(pl_surround.databuf);
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153 pl_surround.delaybuf_len=0;
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154 }
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155
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156 // empty buffers
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157 static void reset()
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158 {
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159 if (pl_surround.passthrough) return;
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160 //fprintf(stderr, "pl_surround: reset called\n");
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161 pl_surround.delaybuf_pos = 0;
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162 memset(pl_surround.Ls_delaybuf, 0, sizeof(int16_t)*pl_surround.delaybuf_len);
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163 memset(pl_surround.Rs_delaybuf, 0, sizeof(int16_t)*pl_surround.delaybuf_len);
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164 }
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165
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166 // The beginnings of an active matrix...
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167 static double steering_matrix[][12] = {
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168 // LL RL LR RR LS RS LLs RLs LRs RRs LC RC
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169 {.707, .0, .0, .707, .5, -.5, .5878, -.3928, .3928, -.5878, .5, .5},
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170 };
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171
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172 // Experimental moving average dominances
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173 static int amp_L = 0, amp_R = 0, amp_C = 0, amp_S = 0;
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174
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175 // processes 'ao_plugin_data.len' bytes of 'data'
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176 // called for every block of data
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177 static int play(){
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178 int16_t *in, *out;
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179 int i, samples;
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180 double *matrix = steering_matrix[0]; // later we'll index based on detected dominance
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181
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182 if (pl_surround.passthrough) return 1;
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183
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184 // fprintf(stderr, "pl_surround: play %d bytes, %d samples\n", ao_plugin_data.len, samples);
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185
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186 samples = ao_plugin_data.len / sizeof(int16_t) / pl_surround.input_channels;
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187 out = pl_surround.databuf; in = (int16_t *)ao_plugin_data.data;
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188
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189 // Testing - place a 1kHz tone on Lt and Rt in anti-phase: should decode in S
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190 //sinewave(in, samples, pl_surround.input_channels, 1000, 0.0, pl_surround.rate);
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191 //sinewave(&in[1], samples, pl_surround.input_channels, 1000, PI, pl_surround.rate);
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192
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193 for (i=0; i<samples; i++) {
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194
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195 // Dominance:
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196 //abs(in[0]) abs(in[1]);
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197 //abs(in[0]+in[1]) abs(in[0]-in[1]);
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198 //10 * log( abs(in[0]) / (abs(in[1])|1) );
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199 //10 * log( abs(in[0]+in[1]) / (abs(in[0]-in[1])|1) );
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200
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201 // About volume balancing...
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202 // Surround encoding does the following:
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203 // Lt=L+.707*C+.707*S, Rt=R+.707*C-.707*S
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204 // So S should be extracted as:
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205 // (Lt-Rt)
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206 // But we are splitting the S to two output channels, so we
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207 // must take 3dB off as we split it:
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208 // Ls=Rs=.707*(Lt-Rt)
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209 // Trouble is, Lt could be +32767, Rt -32768, so possibility that S will
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210 // overflow. So to avoid that, we cut L/R by 3dB (*.707), and S by 6dB (/2).
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211 // this keeps the overall balance, but guarantees no overflow.
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212
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213 // output front left and right
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214 out[0] = matrix[0]*in[0] + matrix[1]*in[1];
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215 out[1] = matrix[2]*in[0] + matrix[3]*in[1];
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216 // output Ls and Rs - from 20msec ago, lowpass filtered @ 7kHz
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217 out[2] = firfilter(pl_surround.Ls_delaybuf, pl_surround.delaybuf_pos,
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218 pl_surround.delaybuf_len, 32, pl_surround.filter_coefs_surround);
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219 #ifdef SPLITREAR
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220 out[3] = firfilter(pl_surround.Rs_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 #else
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223 out[3] = -out[2];
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224 #endif
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225 // calculate and save surround for 20msecs time
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226 #ifdef SPLITREAR
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227 pl_surround.Ls_delaybuf[pl_surround.delaybuf_pos] =
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228 matrix[6]*in[0] + matrix[7]*in[1];
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229 pl_surround.Rs_delaybuf[pl_surround.delaybuf_pos++] =
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230 matrix[8]*in[0] + matrix[9]*in[1];
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231 #else
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232 pl_surround.Ls_delaybuf[pl_surround.delaybuf_pos++] =
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233 matrix[4]*in[0] + matrix[5]*in[1];
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234 #endif
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235 pl_surround.delaybuf_pos %= pl_surround.delaybuf_len;
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236
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237 // next samples...
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238 in = &in[pl_surround.input_channels]; out = &out[4];
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239 }
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240
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241 // Show some state
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242 //printf("\npl_surround: delaybuf_pos=%d, samples=%d\r\033[A", pl_surround.delaybuf_pos, samples);
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243
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244 // Set output block/len
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245 ao_plugin_data.data=pl_surround.databuf;
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246 ao_plugin_data.len=samples*sizeof(int16_t)*4;
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247 return 1;
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248 }