Mercurial > mplayer.hg
annotate mp3lib/decod386.c @ 21040:c7d3523c74ee
fix mp3lib compilation on OSX
author | nplourde |
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date | Sun, 19 Nov 2006 15:21:50 +0000 |
parents | f569aadfa921 |
children | 0db8fb3cc48d |
rev | line source |
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1 /* |
18783 | 2 * Modified for use with MPlayer, for details see the changelog at |
3 * http://svn.mplayerhq.hu/mplayer/trunk/ | |
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4 * $Id$ |
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5 */ |
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6 |
1 | 7 /* |
8 * Mpeg Layer-1,2,3 audio decoder | |
9 * ------------------------------ | |
10 * copyright (c) 1995,1996,1997 by Michael Hipp, All rights reserved. | |
11 * See also 'README' | |
12 * | |
13 * slighlty optimized for machines without autoincrement/decrement. | |
14 * The performance is highly compiler dependend. Maybe | |
15 * the decode.c version for 'normal' processor may be faster | |
16 * even for Intel processors. | |
17 */ | |
18 | |
19 | |
16989 | 20 #include "config.h" |
1 | 21 |
21040 | 22 /* Can't compile MMX assembly on Darwin */ |
23 #ifdef SYS_DARWIN | |
24 #undef HAVE_MMX | |
25 #endif /* SYS_DARWIN */ | |
26 | |
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27 #if 0 |
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28 /* old WRITE_SAMPLE */ |
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29 /* is portable */ |
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30 #define WRITE_SAMPLE(samples,sum,clip) { \ |
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31 if( (sum) > 32767.0) { *(samples) = 0x7fff; (clip)++; } \ |
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32 else if( (sum) < -32768.0) { *(samples) = -0x8000; (clip)++; }\ |
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33 else { *(samples) = sum; } \ |
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34 } |
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35 #else |
1 | 36 /* new WRITE_SAMPLE */ |
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37 |
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38 /* |
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39 * should be the same as the "old WRITE_SAMPLE" macro above, but uses |
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40 * some tricks to avoid double->int conversions and floating point compares. |
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41 * |
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42 * Here's how it works: |
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43 * ((((65536.0 * 65536.0 * 16)+(65536.0 * 0.5))* 65536.0)) is |
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44 * 0x0010000080000000LL in hex. It computes 0x0010000080000000LL + sum |
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45 * as a double IEEE fp value and extracts the low-order 32-bits from the |
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46 * IEEE fp representation stored in memory. The 2^56 bit in the constant |
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47 * is intended to force the bits of "sum" into the least significant bits |
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48 * of the double mantissa. After an integer substraction of 0x80000000 |
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49 * we have the original double value "sum" converted to an 32-bit int value. |
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50 * |
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51 * (Is that really faster than the clean and simple old version of the macro?) |
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52 */ |
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53 |
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54 /* |
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55 * On a SPARC cpu, we fetch the low-order 32-bit from the second 32-bit |
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56 * word of the double fp value stored in memory. On an x86 cpu, we fetch it |
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57 * from the first 32-bit word. |
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58 * I'm not sure if the WORDS_BIGENDIAN feature test covers all possible memory |
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59 * layouts of double floating point values an all cpu architectures. If |
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60 * it doesn't work for you, just enable the "old WRITE_SAMPLE" macro. |
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61 */ |
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62 #if WORDS_BIGENDIAN |
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63 #define MANTISSA_OFFSET 1 |
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64 #else |
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65 #define MANTISSA_OFFSET 0 |
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66 #endif |
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67 |
1 | 68 /* sizeof(int) == 4 */ |
69 #define WRITE_SAMPLE(samples,sum,clip) { \ | |
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70 union { double dtemp; int itemp[2]; } u; int v; \ |
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71 u.dtemp = ((((65536.0 * 65536.0 * 16)+(65536.0 * 0.5))* 65536.0)) + (sum);\ |
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72 v = u.itemp[MANTISSA_OFFSET] - 0x80000000; \ |
1 | 73 if( v > 32767) { *(samples) = 0x7fff; (clip)++; } \ |
74 else if( v < -32768) { *(samples) = -0x8000; (clip)++; } \ | |
75 else { *(samples) = v; } \ | |
76 } | |
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77 #endif |
1 | 78 |
79 | |
80 /* | |
81 #define WRITE_SAMPLE(samples,sum,clip) { \ | |
82 double dtemp; int v; \ | |
83 dtemp = ((((65536.0 * 65536.0 * 16)+(65536.0 * 0.5))* 65536.0)) + (sum);\ | |
84 v = ((*(int *)&dtemp) - 0x80000000); \ | |
85 if( v > 32767) { *(samples) = 0x7fff; (clip)++; } \ | |
86 else if( v < -32768) { *(samples) = -0x8000; (clip)++; } \ | |
87 else { *(samples) = v; } \ | |
88 } | |
89 */ | |
90 | |
13188 | 91 static int synth_1to1(real *bandPtr,int channel,unsigned char *out,int *pnt); |
92 | |
1 | 93 static int synth_1to1_mono(real *bandPtr,unsigned char *samples,int *pnt) |
94 { | |
95 short samples_tmp[64]; | |
96 short *tmp1 = samples_tmp; | |
97 int i,ret; | |
98 int pnt1 = 0; | |
99 | |
100 ret = synth_1to1(bandPtr,0,(unsigned char *) samples_tmp,&pnt1); | |
101 samples += *pnt; | |
102 | |
103 for(i=0;i<32;i++) { | |
104 *( (short *) samples) = *tmp1; | |
105 samples += 2; | |
106 tmp1 += 2; | |
107 } | |
108 *pnt += 64; | |
109 | |
110 return ret; | |
111 } | |
112 | |
113 | |
114 static int synth_1to1_mono2stereo(real *bandPtr,unsigned char *samples,int *pnt) | |
115 { | |
116 int i,ret; | |
117 | |
118 ret = synth_1to1(bandPtr,0,samples,pnt); | |
119 samples = samples + *pnt - 128; | |
120 | |
121 for(i=0;i<32;i++) { | |
122 ((short *)samples)[1] = ((short *)samples)[0]; | |
123 samples+=4; | |
124 } | |
125 | |
126 return ret; | |
127 } | |
128 | |
12134 | 129 static synth_func_t synth_func; |
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130 |
19133 | 131 #if defined(CAN_COMPILE_X86_ASM) && defined(HAVE_MMX) |
13188 | 132 int synth_1to1_MMX( real *bandPtr,int channel,short * samples) |
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133 { |
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134 static short buffs[2][2][0x110]; |
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135 static int bo = 1; |
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136 synth_1to1_MMX_s(bandPtr, channel, samples, (short *) buffs, &bo); |
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137 return 0; |
1258 | 138 } |
139 #endif | |
9002 | 140 |
141 #ifdef HAVE_ALTIVEC | |
142 #define dct64_base(a,b,c) if(gCpuCaps.hasAltiVec) dct64_altivec(a,b,c); else dct64(a,b,c) | |
143 #else /* HAVE_ALTIVEC */ | |
144 #define dct64_base(a,b,c) dct64(a,b,c) | |
145 #endif /* HAVE_ALTIVEC */ | |
146 | |
1 | 147 static int synth_1to1(real *bandPtr,int channel,unsigned char *out,int *pnt) |
148 { | |
149 static real buffs[2][2][0x110]; | |
150 static const int step = 2; | |
151 static int bo = 1; | |
152 short *samples = (short *) (out + *pnt); | |
153 real *b0,(*buf)[0x110]; | |
154 int clip = 0; | |
155 int bo1; | |
8543 | 156 |
157 *pnt += 128; | |
158 | |
4149 | 159 /* optimized for x86 */ |
4321 | 160 #if defined(CAN_COMPILE_X86_ASM) |
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161 if ( synth_func ) |
787 | 162 { |
8543 | 163 // printf("Calling %p, bandPtr=%p channel=%d samples=%p\n",synth_func,bandPtr,channel,samples); |
164 // FIXME: synth_func() may destroy EBP, don't rely on stack contents!!! | |
165 return (*synth_func)( bandPtr,channel,samples); | |
736 | 166 } |
1258 | 167 #endif |
1 | 168 if(!channel) { /* channel=0 */ |
169 bo--; | |
170 bo &= 0xf; | |
171 buf = buffs[0]; | |
172 } | |
173 else { | |
174 samples++; | |
175 buf = buffs[1]; | |
176 } | |
177 | |
178 if(bo & 0x1) { | |
179 b0 = buf[0]; | |
180 bo1 = bo; | |
9002 | 181 dct64_base(buf[1]+((bo+1)&0xf),buf[0]+bo,bandPtr); |
1 | 182 } |
183 else { | |
184 b0 = buf[1]; | |
185 bo1 = bo+1; | |
9002 | 186 dct64_base(buf[0]+bo,buf[1]+bo+1,bandPtr); |
1 | 187 } |
188 | |
189 { | |
190 register int j; | |
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191 real *window = mp3lib_decwin + 16 - bo1; |
1 | 192 |
193 for (j=16;j;j--,b0+=0x10,window+=0x20,samples+=step) | |
194 { | |
195 real sum; | |
196 sum = window[0x0] * b0[0x0]; | |
197 sum -= window[0x1] * b0[0x1]; | |
198 sum += window[0x2] * b0[0x2]; | |
199 sum -= window[0x3] * b0[0x3]; | |
200 sum += window[0x4] * b0[0x4]; | |
201 sum -= window[0x5] * b0[0x5]; | |
202 sum += window[0x6] * b0[0x6]; | |
203 sum -= window[0x7] * b0[0x7]; | |
204 sum += window[0x8] * b0[0x8]; | |
205 sum -= window[0x9] * b0[0x9]; | |
206 sum += window[0xA] * b0[0xA]; | |
207 sum -= window[0xB] * b0[0xB]; | |
208 sum += window[0xC] * b0[0xC]; | |
209 sum -= window[0xD] * b0[0xD]; | |
210 sum += window[0xE] * b0[0xE]; | |
211 sum -= window[0xF] * b0[0xF]; | |
212 | |
213 WRITE_SAMPLE(samples,sum,clip); | |
214 } | |
215 | |
216 { | |
217 real sum; | |
218 sum = window[0x0] * b0[0x0]; | |
219 sum += window[0x2] * b0[0x2]; | |
220 sum += window[0x4] * b0[0x4]; | |
221 sum += window[0x6] * b0[0x6]; | |
222 sum += window[0x8] * b0[0x8]; | |
223 sum += window[0xA] * b0[0xA]; | |
224 sum += window[0xC] * b0[0xC]; | |
225 sum += window[0xE] * b0[0xE]; | |
226 WRITE_SAMPLE(samples,sum,clip); | |
227 b0-=0x10,window-=0x20,samples+=step; | |
228 } | |
229 window += bo1<<1; | |
230 | |
231 for (j=15;j;j--,b0-=0x10,window-=0x20,samples+=step) | |
232 { | |
233 real sum; | |
234 sum = -window[-0x1] * b0[0x0]; | |
235 sum -= window[-0x2] * b0[0x1]; | |
236 sum -= window[-0x3] * b0[0x2]; | |
237 sum -= window[-0x4] * b0[0x3]; | |
238 sum -= window[-0x5] * b0[0x4]; | |
239 sum -= window[-0x6] * b0[0x5]; | |
240 sum -= window[-0x7] * b0[0x6]; | |
241 sum -= window[-0x8] * b0[0x7]; | |
242 sum -= window[-0x9] * b0[0x8]; | |
243 sum -= window[-0xA] * b0[0x9]; | |
244 sum -= window[-0xB] * b0[0xA]; | |
245 sum -= window[-0xC] * b0[0xB]; | |
246 sum -= window[-0xD] * b0[0xC]; | |
247 sum -= window[-0xE] * b0[0xD]; | |
248 sum -= window[-0xF] * b0[0xE]; | |
249 sum -= window[-0x0] * b0[0xF]; | |
250 | |
251 WRITE_SAMPLE(samples,sum,clip); | |
252 } | |
253 } | |
254 | |
255 return clip; | |
256 | |
257 } | |
258 | |
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259 #ifdef USE_FAKE_MONO |
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260 static int synth_1to1_l(real *bandPtr,int channel,unsigned char *out,int *pnt) |
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261 { |
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262 int i,ret; |
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263 |
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264 ret = synth_1to1(bandPtr,channel,out,pnt); |
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265 out = out + *pnt - 128; |
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266 |
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267 for(i=0;i<32;i++) { |
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268 ((short *)out)[1] = ((short *)out)[0]; |
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269 out+=4; |
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270 } |
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271 |
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272 return ret; |
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273 } |
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274 |
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275 static int synth_1to1_r(real *bandPtr,int channel,unsigned char *out,int *pnt) |
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276 { |
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277 int i,ret; |
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278 |
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279 ret = synth_1to1(bandPtr,channel,out,pnt); |
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280 out = out + *pnt - 128; |
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281 |
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282 for(i=0;i<32;i++) { |
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283 ((short *)out)[0] = ((short *)out)[1]; |
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284 out+=4; |
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285 } |
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286 |
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287 return ret; |
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288 } |
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289 #endif |