Mercurial > mplayer.hg
annotate libao2/pl_surround.c @ 6713:cc917a581b6e
add simple playlist support
author | pontscho |
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date | Fri, 12 Jul 2002 00:53:26 +0000 |
parents | 2eec40929570 |
children | 9fc45fe0d444 |
rev | line source |
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3313 | 1 /* |
2 This is an ao2 plugin to do simple decoding of matrixed surround | |
3 sound. This will provide a (basic) surround-sound effect from | |
4 audio encoded for Dolby Surround, Pro Logic etc. | |
5 | |
6 * This program is free software; you can redistribute it and/or modify | |
7 * it under the terms of the GNU General Public License as published by | |
8 * the Free Software Foundation; either version 2 of the License, or | |
9 * (at your option) any later version. | |
10 * | |
11 * This program is distributed in the hope that it will be useful, | |
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of | |
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the | |
14 * GNU General Public License for more details. | |
15 * | |
16 * You should have received a copy of the GNU General Public License | |
17 * along with this program; if not, write to the Free Software | |
18 * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. | |
19 | |
20 Original author: Steve Davies <steve@daviesfam.org> | |
21 */ | |
22 | |
23 /* The principle: Make rear channels by extracting anti-phase data | |
3681 | 24 from the front channels, delay by 20msec and feed to rear in anti-phase |
3313 | 25 */ |
26 | |
27 | |
3681 | 28 // SPLITREAR: Define to decode two distinct rear channels - |
29 // this doesn't work so well in practice because | |
30 // separation in a passive matrix is not high. | |
31 // C (dialogue) to Ls and Rs 14dB or so - | |
32 // so dialogue leaks to the rear. | |
33 // Still - give it a try and send feedback. | |
34 // comment this define for old behaviour of a single | |
35 // surround sent to rear in anti-phase | |
36 #define SPLITREAR | |
37 | |
38 | |
3313 | 39 #include <stdio.h> |
40 #include <stdlib.h> | |
6237 | 41 #include <string.h> |
3317 | 42 #include <unistd.h> |
3313 | 43 |
44 #include "audio_out.h" | |
45 #include "audio_plugin.h" | |
46 #include "audio_plugin_internal.h" | |
47 #include "afmt.h" | |
48 | |
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49 #include "remez.h" |
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50 #include "firfilter.c" |
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51 |
3313 | 52 static ao_info_t info = |
53 { | |
54 "Surround decoder plugin", | |
55 "surround", | |
56 "Steve Davies <steve@daviesfam.org>", | |
57 "" | |
58 }; | |
59 | |
60 LIBAO_PLUGIN_EXTERN(surround) | |
61 | |
62 // local data | |
63 typedef struct pl_surround_s | |
64 { | |
65 int passthrough; // Just be a "NO-OP" | |
66 int msecs; // Rear channel delay in milliseconds | |
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 |
3313 | 73 int rate; // input data rate |
74 int format; // input format | |
75 int input_channels; // input channels | |
76 | |
77 } pl_surround_t; | |
78 | |
3681 | 79 static pl_surround_t pl_surround={0,20,NULL,NULL,NULL,0,0,NULL,0,0,0}; |
3313 | 80 |
81 // to set/get/query special features/parameters | |
82 static int control(int cmd,int arg){ | |
83 switch(cmd){ | |
84 case AOCONTROL_PLUGIN_SET_LEN: | |
85 if (pl_surround.passthrough) return CONTROL_OK; | |
86 //fprintf(stderr, "pl_surround: AOCONTROL_PLUGIN_SET_LEN with arg=%d\n", arg); | |
87 //fprintf(stderr, "pl_surround: ao_plugin_data.len=%d\n", ao_plugin_data.len); | |
88 // Allocate an output buffer | |
89 if (pl_surround.databuf != NULL) { | |
90 free(pl_surround.databuf); pl_surround.databuf = NULL; | |
91 } | |
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92 // Allocate output buffer |
3313 | 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... |
3313 | 95 ao_plugin_data.len /= 2; |
96 return CONTROL_OK; | |
97 } | |
98 return -1; | |
99 } | |
100 | |
101 // open & setup audio device | |
102 // return: 1=success 0=fail | |
103 static int init(){ | |
104 | |
105 fprintf(stderr, "pl_surround: init input rate=%d, channels=%d\n", ao_plugin_data.rate, ao_plugin_data.channels); | |
106 if (ao_plugin_data.channels != 2) { | |
107 fprintf(stderr, "pl_surround: source audio must have 2 channels, using passthrough mode\n"); | |
108 pl_surround.passthrough = 1; | |
109 return 1; | |
110 } | |
111 if (ao_plugin_data.format != AFMT_S16_LE) { | |
112 fprintf(stderr, "pl_surround: I'm dumb and can only handle AFMT_S16_LE audio format, using passthrough mode\n"); | |
113 pl_surround.passthrough = 1; | |
114 return 1; | |
115 } | |
116 | |
117 pl_surround.passthrough = 0; | |
118 | |
119 /* Store info on input format to expect */ | |
120 pl_surround.rate=ao_plugin_data.rate; | |
121 pl_surround.format=ao_plugin_data.format; | |
122 pl_surround.input_channels=ao_plugin_data.channels; | |
123 | |
124 // Input 2 channels, output will be 4 - tell ao_plugin | |
125 ao_plugin_data.channels = 4; | |
126 ao_plugin_data.sz_mult /= 2; | |
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); |
3313 | 141 return 1; |
142 } | |
143 | |
144 // close plugin | |
145 static void uninit(){ | |
146 // fprintf(stderr, "pl_surround: uninit called!\n"); | |
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); |
5501 | 152 if(pl_surround.databuf) { |
3313 | 153 free(pl_surround.databuf); |
5501 | 154 pl_surround.databuf = NULL; |
155 } | |
3313 | 156 pl_surround.delaybuf_len=0; |
157 } | |
158 | |
159 // empty buffers | |
160 static void reset() | |
161 { | |
162 if (pl_surround.passthrough) return; | |
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); |
3313 | 167 } |
168 | |
3681 | 169 // The beginnings of an active matrix... |
170 static double steering_matrix[][12] = { | |
171 // LL RL LR RR LS RS LLs RLs LRs RRs LC RC | |
172 {.707, .0, .0, .707, .5, -.5, .5878, -.3928, .3928, -.5878, .5, .5}, | |
173 }; | |
174 | |
175 // Experimental moving average dominances | |
176 static int amp_L = 0, amp_R = 0, amp_C = 0, amp_S = 0; | |
3313 | 177 |
178 // processes 'ao_plugin_data.len' bytes of 'data' | |
179 // called for every block of data | |
180 static int play(){ | |
181 int16_t *in, *out; | |
182 int i, samples; | |
3681 | 183 double *matrix = steering_matrix[0]; // later we'll index based on detected dominance |
3313 | 184 |
185 if (pl_surround.passthrough) return 1; | |
186 | |
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187 // fprintf(stderr, "pl_surround: play %d bytes, %d samples\n", ao_plugin_data.len, samples); |
3313 | 188 |
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; |
3313 | 191 |
3681 | 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 |
3313 | 196 for (i=0; i<samples; i++) { |
3373 | 197 |
3681 | 198 // Dominance: |
199 //abs(in[0]) abs(in[1]); | |
200 //abs(in[0]+in[1]) abs(in[0]-in[1]); | |
201 //10 * log( abs(in[0]) / (abs(in[1])|1) ); | |
202 //10 * log( abs(in[0]+in[1]) / (abs(in[0]-in[1])|1) ); | |
203 | |
3410 | 204 // About volume balancing... |
3373 | 205 // Surround encoding does the following: |
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) |
3410 | 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 |
3681 | 213 // overflow. So to avoid that, we cut L/R by 3dB (*.707), and S by 6dB (/2). |
214 // this keeps the overall balance, but guarantees no overflow. | |
3373 | 215 |
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216 // output front left and right |
3681 | 217 out[0] = matrix[0]*in[0] + matrix[1]*in[1]; |
218 out[1] = matrix[2]*in[0] + matrix[3]*in[1]; | |
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); |
3681 | 222 #ifdef SPLITREAR |
223 out[3] = firfilter(pl_surround.Rs_delaybuf, pl_surround.delaybuf_pos, | |
224 pl_surround.delaybuf_len, 32, pl_surround.filter_coefs_surround); | |
225 #else | |
226 out[3] = -out[2]; | |
227 #endif | |
228 // calculate and save surround for 20msecs time | |
229 #ifdef SPLITREAR | |
230 pl_surround.Ls_delaybuf[pl_surround.delaybuf_pos] = | |
231 matrix[6]*in[0] + matrix[7]*in[1]; | |
232 pl_surround.Rs_delaybuf[pl_surround.delaybuf_pos++] = | |
233 matrix[8]*in[0] + matrix[9]*in[1]; | |
234 #else | |
3718 | 235 pl_surround.Ls_delaybuf[pl_surround.delaybuf_pos++] = |
3681 | 236 matrix[4]*in[0] + matrix[5]*in[1]; |
237 #endif | |
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238 pl_surround.delaybuf_pos %= pl_surround.delaybuf_len; |
3681 | 239 |
3313 | 240 // next samples... |
241 in = &in[pl_surround.input_channels]; out = &out[4]; | |
242 } | |
3681 | 243 |
244 // Show some state | |
245 //printf("\npl_surround: delaybuf_pos=%d, samples=%d\r\033[A", pl_surround.delaybuf_pos, samples); | |
3313 | 246 |
247 // Set output block/len | |
248 ao_plugin_data.data=pl_surround.databuf; | |
249 ao_plugin_data.len=samples*sizeof(int16_t)*4; | |
250 return 1; | |
251 } |