Mercurial > mplayer.hg
annotate libao2/pl_eq.c @ 8660:39476cbd1673
I have looked at the fullscreen code and realized that there are
generally two types of layer support for window managers:
- NetWM states (FULLSCREEN, STAYS_ON_TOP, ABOVE)
sawfish, metacity, kwin
- _WIN_LAYER
IceWM, WindowMaker
So we don't need any other window manager detection functions, we need
only to check for these two.
Code tested on lots of windowmanagers.
patch by Filip Kalinski <filon@pld.org.pl>
author | arpi |
---|---|
date | Mon, 30 Dec 2002 18:50:15 +0000 |
parents | 9fc45fe0d444 |
children | 46d21c0f36aa |
rev | line source |
---|---|
6430 | 1 /*============================================================================= |
2 // | |
3 // This software has been released under the terms of the GNU Public | |
4 // license. See http://www.gnu.org/copyleft/gpl.html for details. | |
5 // | |
6 // Copyright 2001 Anders Johansson ajh@atri.curtin.edu.au | |
7 // | |
8 //============================================================================= | |
9 */ | |
10 | |
11 /* Equalizer plugin, implementation of a 10 band time domain graphic | |
12 equalizer using IIR filters. The IIR filters are implemented using a | |
13 Direct Form II approach. But has been modified (b1 == 0 always) to | |
14 save computation. | |
15 */ | |
16 #define PLUGIN | |
17 | |
18 #include <stdio.h> | |
19 #include <stdlib.h> | |
20 #include <unistd.h> | |
21 #include <inttypes.h> | |
22 #include <math.h> | |
23 | |
24 #include "audio_out.h" | |
25 #include "audio_plugin.h" | |
26 #include "audio_plugin_internal.h" | |
27 #include "afmt.h" | |
28 #include "eq.h" | |
29 | |
30 static ao_info_t info = | |
31 { | |
32 "Equalizer audio plugin", | |
33 "eq", | |
34 "Anders", | |
35 "" | |
36 }; | |
37 | |
38 LIBAO_PLUGIN_EXTERN(eq) | |
39 | |
40 | |
41 #define CH 6 // Max number of channels | |
42 #define L 2 // Storage for filter taps | |
43 #define KM 10 // Max number of octaves | |
44 | |
45 #define Q 1.2247 /* Q value for band-pass filters 1.2247=(3/2)^(1/2) | |
7507
a2173bfdc133
Correction of spelling errors and removal of old code
anders
parents:
6839
diff
changeset
|
46 gives 4dB suppression @ Fc*2 and Fc/2 */ |
6430 | 47 |
48 // Center frequencies for band-pass filters | |
49 #define CF {31.25,62.5,125,250,500,1000,2000,4000,8000,16000} | |
50 | |
51 // local data | |
52 typedef struct pl_eq_s | |
53 { | |
54 int16_t a[KM][L]; // A weights | |
55 int16_t b[KM][L]; // B weights | |
56 int16_t wq[CH][KM][L]; // Circular buffer for W data | |
57 int16_t g[CH][KM]; // Gain factor for each channel and band | |
58 int16_t K; // Number of used eq bands | |
59 int channels; // Number of channels | |
60 } pl_eq_t; | |
61 | |
62 static pl_eq_t pl_eq; | |
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 return CONTROL_OK; | |
69 case AOCONTROL_PLUGIN_EQ_SET_GAIN:{ | |
70 float gain = ((equalizer_t*)arg)->gain; | |
71 int ch =((equalizer_t*)arg)->channel; | |
72 int band =((equalizer_t*)arg)->band; | |
73 if(ch > CH || ch < 0 || band > KM || band < 0) | |
74 return CONTROL_ERROR; | |
75 | |
76 pl_eq.g[ch][band]=(int16_t) 4096 * (pow(10.0,gain/20.0)-1.0); | |
77 return CONTROL_OK; | |
78 } | |
79 case AOCONTROL_PLUGIN_EQ_GET_GAIN:{ | |
80 int ch =((equalizer_t*)arg)->channel; | |
81 int band =((equalizer_t*)arg)->band; | |
82 if(ch > CH || ch < 0 || band > KM || band < 0) | |
83 return CONTROL_ERROR; | |
84 | |
85 ((equalizer_t*)arg)->gain = log10((float)pl_eq.g[ch][band]/4096.0+1) * 20.0; | |
86 return CONTROL_OK; | |
87 } | |
88 } | |
89 return CONTROL_UNKNOWN; | |
90 } | |
91 | |
6439 | 92 // return rounded 16bit int |
93 static inline int16_t lround16(double n){ | |
94 return (int16_t)((n)>=0.0?(n)+0.5:(n)-0.5); | |
95 } | |
96 | |
6430 | 97 // 2nd order Band-pass Filter design |
98 void bp2(int16_t* a, int16_t* b, float fc, float q){ | |
99 double th=2*3.141592654*fc; | |
100 double C=(1 - tan(th*q/2))/(1 + tan(th*q/2)); | |
101 | |
6439 | 102 a[0] = lround16( 16383.0 * (1 + C) * cos(th)); |
103 a[1] = lround16(-16383.0 * C); | |
6430 | 104 |
6439 | 105 b[0] = lround16(-16383.0 * (C - 1)/2); |
106 b[1] = lround16(-16383.0 * 1.0050); | |
6430 | 107 } |
108 | |
109 // empty buffers | |
110 static void reset(){ | |
111 int k,l,c; | |
112 for(c=0;c<pl_eq.channels;c++) | |
113 for(k=0;k<pl_eq.K;k++) | |
114 for(l=0;l<L*2;l++) | |
115 pl_eq.wq[c][k][l]=0; | |
116 } | |
117 | |
118 // open & setup audio device | |
119 // return: 1=success 0=fail | |
120 static int init(){ | |
8123
9fc45fe0d444
*HUGE* set of compiler warning fixes, unused variables removal
arpi
parents:
7507
diff
changeset
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121 int k = 0; |
6430 | 122 float F[KM] = CF; |
123 | |
124 // Check input format | |
125 if(ao_plugin_data.format != AFMT_S16_LE){ | |
126 fprintf(stderr,"[pl_eq] Input audio format not yet supported. \n"); | |
127 return 0; | |
128 } | |
129 | |
130 // Check number of channels | |
131 if(ao_plugin_data.channels>CH){ | |
132 fprintf(stderr,"[pl_eq] Too many channels, max is 6.\n"); | |
133 return 0; | |
134 } | |
135 pl_eq.channels=ao_plugin_data.channels; | |
136 | |
137 // Calculate number of active filters | |
138 pl_eq.K=KM; | |
139 while(F[pl_eq.K-1] > (float)ao_plugin_data.rate/2) | |
140 pl_eq.K--; | |
141 | |
142 // Generate filter taps | |
143 for(k=0;k<pl_eq.K;k++) | |
144 bp2(pl_eq.a[k],pl_eq.b[k],F[k]/((float)ao_plugin_data.rate),Q); | |
145 | |
146 // Reset buffers | |
147 reset(); | |
148 | |
149 // Tell ao_plugin how much this plugin adds to the overall time delay | |
150 ao_plugin_data.delay_fix-=2/((float)pl_eq.channels*(float)ao_plugin_data.rate); | |
151 // Print some cool remark of what the plugin does | |
152 printf("[pl_eq] Equalizer in use.\n"); | |
153 return 1; | |
154 } | |
155 | |
156 // close plugin | |
157 static void uninit(){ | |
158 } | |
159 | |
160 // processes 'ao_plugin_data.len' bytes of 'data' | |
161 // called for every block of data | |
162 static int play(){ | |
163 uint16_t ci = pl_eq.channels; // Index for channels | |
164 uint16_t nch = pl_eq.channels; // Number of channels | |
165 | |
166 while(ci--){ | |
167 int16_t* g = pl_eq.g[ci]; // Gain factor | |
168 int16_t* in = ((int16_t*)ao_plugin_data.data)+ci; | |
169 int16_t* out = ((int16_t*)ao_plugin_data.data)+ci; | |
170 int16_t* end = in+ao_plugin_data.len/2; // Block loop end | |
171 | |
172 while(in < end){ | |
173 register int16_t k = 0; // Frequency band index | |
174 register int32_t yt = 0; // Total output from filters | |
7507
a2173bfdc133
Correction of spelling errors and removal of old code
anders
parents:
6839
diff
changeset
|
175 register int16_t x = *in; // Current input sample |
6430 | 176 in+=nch; |
177 | |
178 // Run the filters | |
179 for(;k<pl_eq.K;k++){ | |
180 // Pointer to circular buffer wq | |
181 register int16_t* wq = pl_eq.wq[ci][k]; | |
6839 | 182 #if 0 |
6430 | 183 // Calculate output from AR part of current filter |
184 register int32_t xt = (x*pl_eq.b[k][0]) >> 4; | |
185 register int32_t w = xt + wq[0]*pl_eq.a[k][0] + wq[1]*pl_eq.a[k][1]; | |
186 // Calculate output form MA part of current filter | |
187 yt+=(((w + wq[1]*pl_eq.b[k][1]) >> 10)*g[k]) >> 12; | |
188 // Update circular buffer | |
189 wq[1] = wq[0]; wq[0] = w >> 14; | |
190 } | |
191 | |
192 // Calculate output | |
193 *out=(int16_t)(yt+x); | |
6839 | 194 #else |
195 // Calculate output from AR part of current filter | |
196 register int32_t xt = (x*pl_eq.b[k][0]) / 48; | |
197 register int32_t w = xt + wq[0]*pl_eq.a[k][0] + wq[1]*pl_eq.a[k][1]; | |
198 // Calculate output form MA part of current filter | |
199 yt+=(((w + wq[1]*pl_eq.b[k][1]) >> 10)*g[k]) >> 12; | |
200 // Update circular buffer | |
201 wq[1] = wq[0]; wq[0] = w / 24576; | |
202 } | |
203 | |
204 // Calculate output | |
205 *out=(int16_t)(yt * 0.25 + x * 0.5); | |
206 #endif | |
6430 | 207 out+=nch; |
208 } | |
209 } | |
210 return 1; | |
211 } | |
212 | |
213 | |
214 | |
215 | |
216 | |
217 | |
218 |