Mercurial > mplayer.hg
comparison libao2/pl_surround.c @ 3313:76a3421bc421
Dolby Surround decoding audio plugin
author | steve |
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date | Tue, 04 Dec 2001 15:42:44 +0000 |
parents | |
children | 614b4525d275 |
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3312:636d07d2654f | 3313:76a3421bc421 |
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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 | |
24 from the front channels, delay by 15msec and feed to rear in anti-phase | |
25 www.dolby.com has the background | |
26 */ | |
27 | |
28 | |
29 #include <stdio.h> | |
30 #include <stdlib.h> | |
31 | |
32 #include "audio_out.h" | |
33 #include "audio_plugin.h" | |
34 #include "audio_plugin_internal.h" | |
35 #include "afmt.h" | |
36 | |
37 static ao_info_t info = | |
38 { | |
39 "Surround decoder plugin", | |
40 "surround", | |
41 "Steve Davies <steve@daviesfam.org>", | |
42 "" | |
43 }; | |
44 | |
45 LIBAO_PLUGIN_EXTERN(surround) | |
46 | |
47 // local data | |
48 typedef struct pl_surround_s | |
49 { | |
50 int passthrough; // Just be a "NO-OP" | |
51 int msecs; // Rear channel delay in milliseconds | |
52 int16_t* databuf; // Output audio buffer | |
53 int16_t* delaybuf; // circular buffer to be used for delaying audio signal | |
54 int delaybuf_len; // local buffer length in samples | |
55 int delaybuf_ptr; // offset in buffer where we are reading/writing | |
56 int rate; // input data rate | |
57 int format; // input format | |
58 int input_channels; // input channels | |
59 | |
60 } pl_surround_t; | |
61 | |
62 static pl_surround_t pl_surround={0,15,NULL,NULL,0,0,0,0,0}; | |
63 | |
64 // to set/get/query special features/parameters | |
65 static int control(int cmd,int arg){ | |
66 switch(cmd){ | |
67 case AOCONTROL_PLUGIN_SET_LEN: | |
68 if (pl_surround.passthrough) return CONTROL_OK; | |
69 //fprintf(stderr, "pl_surround: AOCONTROL_PLUGIN_SET_LEN with arg=%d\n", arg); | |
70 //fprintf(stderr, "pl_surround: ao_plugin_data.len=%d\n", ao_plugin_data.len); | |
71 // Allocate an output buffer | |
72 if (pl_surround.databuf != NULL) { | |
73 free(pl_surround.databuf); pl_surround.databuf = NULL; | |
74 } | |
75 pl_surround.databuf = calloc(ao_plugin_data.len, 1); | |
76 // Return back smaller len so we don't get overflowed... (??seems the right thing to do?) | |
77 ao_plugin_data.len /= 2; | |
78 return CONTROL_OK; | |
79 } | |
80 return -1; | |
81 } | |
82 | |
83 // open & setup audio device | |
84 // return: 1=success 0=fail | |
85 static int init(){ | |
86 | |
87 fprintf(stderr, "pl_surround: init input rate=%d, channels=%d\n", ao_plugin_data.rate, ao_plugin_data.channels); | |
88 if (ao_plugin_data.channels != 2) { | |
89 fprintf(stderr, "pl_surround: source audio must have 2 channels, using passthrough mode\n"); | |
90 pl_surround.passthrough = 1; | |
91 return 1; | |
92 } | |
93 if (ao_plugin_data.format != AFMT_S16_LE) { | |
94 fprintf(stderr, "pl_surround: I'm dumb and can only handle AFMT_S16_LE audio format, using passthrough mode\n"); | |
95 pl_surround.passthrough = 1; | |
96 return 1; | |
97 } | |
98 | |
99 pl_surround.passthrough = 0; | |
100 | |
101 /* Store info on input format to expect */ | |
102 pl_surround.rate=ao_plugin_data.rate; | |
103 pl_surround.format=ao_plugin_data.format; | |
104 pl_surround.input_channels=ao_plugin_data.channels; | |
105 | |
106 // Input 2 channels, output will be 4 - tell ao_plugin | |
107 ao_plugin_data.channels = 4; | |
108 ao_plugin_data.sz_mult /= 2; | |
109 | |
110 // Figure out buffer space needed for the 15msec delay | |
111 pl_surround.delaybuf_len = pl_surround.rate * pl_surround.msecs / 1000; | |
112 // Allocate delay buffer | |
113 pl_surround.delaybuf=(void*)calloc(pl_surround.delaybuf_len,sizeof(int16_t)); | |
114 fprintf(stderr, "pl_surround: %dmsec surround delay, rate %d - buffer is %d samples\n", | |
115 pl_surround.msecs,pl_surround.rate, pl_surround.delaybuf_len); | |
116 pl_surround.delaybuf_ptr = 0; | |
117 | |
118 return 1; | |
119 } | |
120 | |
121 // close plugin | |
122 static void uninit(){ | |
123 // fprintf(stderr, "pl_surround: uninit called!\n"); | |
124 if (pl_surround.passthrough) return; | |
125 if(pl_surround.delaybuf) | |
126 free(pl_surround.delaybuf); | |
127 if(pl_surround.databuf) | |
128 free(pl_surround.databuf); | |
129 pl_surround.delaybuf_len=0; | |
130 } | |
131 | |
132 // empty buffers | |
133 static void reset() | |
134 { | |
135 if (pl_surround.passthrough) return; | |
136 //fprintf(stderr, "pl_surround: reset called\n"); | |
137 pl_surround.delaybuf_ptr = 0; | |
138 memset(pl_surround.delaybuf, 0, sizeof(int16_t)*pl_surround.delaybuf_len); | |
139 } | |
140 | |
141 | |
142 // processes 'ao_plugin_data.len' bytes of 'data' | |
143 // called for every block of data | |
144 static int play(){ | |
145 int16_t *in, *out; | |
146 int i, samples; | |
147 int surround; | |
148 | |
149 if (pl_surround.passthrough) return 1; | |
150 | |
151 // fprintf(stderr, "pl_surround: play %d bytes, %d samples\n", ao_plugin_data.len, samples); | |
152 | |
153 samples = ao_plugin_data.len / sizeof(int16_t) / pl_surround.input_channels; | |
154 | |
155 out = pl_surround.databuf; in = (int16_t *)ao_plugin_data.data; | |
156 for (i=0; i<samples; i++) { | |
157 // front left and right | |
158 out[0] = in[0]; | |
159 out[1] = in[1]; | |
160 // surround - from 15msec ago | |
161 out[2] = pl_surround.delaybuf[pl_surround.delaybuf_ptr]; | |
162 out[3] = -out[2]; | |
163 // calculate and save surround for 15msecs time | |
164 pl_surround.delaybuf[pl_surround.delaybuf_ptr++] = (in[0]/2 - in[1]/2); | |
165 pl_surround.delaybuf_ptr %= pl_surround.delaybuf_len; | |
166 // next samples... | |
167 in = &in[pl_surround.input_channels]; out = &out[4]; | |
168 } | |
169 | |
170 // Set output block/len | |
171 ao_plugin_data.data=pl_surround.databuf; | |
172 ao_plugin_data.len=samples*sizeof(int16_t)*4; | |
173 return 1; | |
174 } | |
175 | |
176 | |
177 |