Mercurial > SDL_sound_CoreAudio
annotate alt_audio_convert.c @ 347:778cee61e1be
Mono-to-stereo fixes, more fprintf debugging output.
| author | Ryan C. Gordon <icculus@icculus.org> |
|---|---|
| date | Thu, 23 May 2002 12:37:13 +0000 |
| parents | fbbb1f25b944 |
| children | c984aa6990f7 |
| rev | line source |
|---|---|
| 338 | 1 /* |
| 2 Extended Audio Converter for SDL (Simple DirectMedia Layer) | |
| 3 Copyright (C) 2002 Frank Ranostaj | |
| 4 Institute of Applied Physik | |
| 5 Johann Wolfgang Goethe-Universität | |
| 6 Frankfurt am Main, Germany | |
| 7 | |
| 8 This library is free software; you can redistribute it and/or | |
| 9 modify it under the terms of the GNU Library General Public | |
| 10 License as published by the Free Software Foundation; either | |
| 11 version 2 of the License, or (at your option) any later version. | |
| 12 | |
| 13 This library is distributed in the hope that it will be useful, | |
| 14 but WITHOUT ANY WARRANTY; without even the implied warranty of | |
| 15 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
| 16 Library General Public License for more details. | |
| 17 | |
| 18 You should have received a copy of the GNU Library General Public | |
| 19 License along with this library; if not, write to the Free | |
| 20 Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA | |
| 21 | |
| 22 Frank Ranostaj | |
| 23 ranostaj@stud.uni-frankfurt.de | |
| 24 | |
| 25 (This code blatantly abducted for SDL_sound. Thanks, Frank! --ryan.) | |
| 26 */ | |
| 27 | |
| 28 #include "alt_audio_convert.h" | |
| 29 | |
| 30 #include <stdlib.h> | |
| 31 #include <math.h> | |
| 32 | |
| 33 /*provisorisch*/ | |
| 34 #define AUDIO_8 (4) | |
| 35 #define AUDIO_16WRONG (8) | |
| 36 #define AUDIO_FORMAT (AUDIO_8 | AUDIO_16WRONG) | |
| 37 #define AUDIO_SIGN (1) | |
| 38 | |
| 39 | |
| 40 /*-------------------------------------------------------------------------*/ | |
| 41 /* this filter (Kaiser-window beta=6.8) gives a decent -80dB attentuation */ | |
| 42 static const int filter[_fsize/2] = { | |
| 43 0, 20798, 0, -6764, 0, 3863, 0, -2560, 0, 1800, 0, -1295, 0, 936, 0, | |
| 44 -671, | |
| 45 0, 474, 0, -326, 0, 217, 0, -138, 0, 83, 0, -46, 0, 23, 0, -9 | |
| 46 }; | |
| 47 | |
| 48 /* Mono (1 channel ) */ | |
| 49 #define Suffix(x) x##1 | |
| 50 #include "filter_templates.h" | |
| 51 #undef Suffix | |
| 52 | |
| 53 /* Stereo (2 channel ) */ | |
| 54 #define Suffix(x) x##2 | |
| 55 #include "filter_templates.h" | |
| 56 #undef Suffix | |
| 57 | |
| 58 /* !!! FIXME: Lose all the "short" vars for "Sint16", etc. */ | |
| 59 | |
| 60 /*-------------------------------------------------------------------------*/ | |
|
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61 int Sound_ConvertAudio( Sound_AudioCVT *Data ) |
| 338 | 62 { |
| 63 AdapterC Temp; | |
| 64 int i; | |
| 65 | |
| 66 /* !!! FIXME: Try the looping stuff under certain circumstances? --ryan. */ | |
| 67 int mode = 0; | |
| 68 | |
| 69 /* Make sure there's a converter */ | |
| 70 if( Data == NULL ) { | |
| 71 SDL_SetError("No converter given"); | |
| 72 return(-1); | |
| 73 } | |
| 74 | |
| 75 /* Make sure there's data to convert */ | |
| 76 if( Data->buf == NULL ) { | |
| 77 SDL_SetError("No buffer allocated for conversion"); | |
| 78 return(-1); | |
| 79 } | |
| 80 | |
| 81 /* Set up the conversion and go! */ | |
| 82 Temp.buffer = (short*)Data->buf; | |
| 83 Temp.mode = mode; | |
| 84 Temp.filter = &Data->filter; | |
| 85 | |
| 86 Data->len_cvt = Data->len; | |
| 87 for( i = 0; Data->adapter[i] != NULL; i++ ) | |
| 88 Data->len_cvt = (*Data->adapter[i])( Temp, Data->len_cvt); | |
| 89 | |
| 90 return(0); | |
| 91 } | |
| 92 | |
| 93 /*-------------------------------------------------------------------------*/ | |
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94 static int expand8BitTo16Bit( AdapterC Data, int length ) |
| 338 | 95 { |
| 96 int i; | |
| 97 char* inp = (char*)Data.buffer-1; | |
| 98 short* buffer = Data.buffer-1; | |
| 99 for( i = length; i; i-- ) | |
| 100 buffer[i] = inp[i]<<8; | |
| 101 return length*2; | |
| 102 } | |
| 103 | |
| 104 /*-------------------------------------------------------------------------*/ | |
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105 static int swapBytes( AdapterC Data, int length ) |
| 338 | 106 { |
| 107 int i; | |
| 108 unsigned short a,b; | |
| 109 short* buffer = Data.buffer; | |
| 110 for( i = 0; i < length; i++ ) | |
| 111 { | |
| 112 a = b = buffer[i]; | |
| 113 a <<= 8; | |
| 114 b >>= 8; | |
| 115 buffer[i] = a | b; | |
| 116 } | |
| 117 return length; | |
| 118 } | |
| 119 | |
| 120 /*-------------------------------------------------------------------------*/ | |
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121 static int cut16BitTo8Bit( AdapterC Data, int length ) |
| 338 | 122 { |
| 123 int i; | |
| 124 short* inp = Data.buffer-1; | |
| 125 char* buffer = (char*)Data.buffer-1; | |
| 126 for( i = 0; i < length; i++ ) | |
| 127 buffer[i] = inp[i]>>8; | |
| 128 return length/2; | |
| 129 } | |
| 130 | |
| 131 /*-------------------------------------------------------------------------*/ | |
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132 static int changeSigned( AdapterC Data, int length ) |
| 338 | 133 { |
| 134 int i; | |
| 135 short* buffer = Data.buffer; | |
| 136 for( i = 0; i < length; i++ ) | |
| 137 buffer[i] ^= 0x8000; | |
| 138 return length; | |
| 139 } | |
| 140 | |
| 141 /*-------------------------------------------------------------------------*/ | |
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142 static int convertStereoToMono( AdapterC Data, int length ) |
| 338 | 143 { |
| 144 int i; | |
| 145 short* buffer = Data.buffer; | |
| 146 | |
| 147 /* | |
| 148 * !!! FIXME: Can we avoid the division in this loop and just keep | |
| 149 * !!! FIXME: a second index variable? --ryan. | |
| 150 */ | |
| 151 for( i = 0; i < length; i+=2 ) | |
| 152 buffer[i/2] = ((int)buffer[i] + buffer[i+1] ) >> 1; | |
| 153 return length/2; | |
| 154 } | |
| 155 | |
| 156 /*-------------------------------------------------------------------------*/ | |
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157 static int convertMonoToStereo( AdapterC Data, int length ) |
| 338 | 158 { |
| 159 int i; | |
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160 short* buffer = Data.buffer-1; |
| 338 | 161 length *= 2; |
| 162 | |
| 163 /* | |
| 164 * !!! FIXME: Can we avoid the division in this loop and just keep | |
| 165 * !!! FIXME: a second index variable? --ryan. | |
| 166 */ | |
| 167 for( i = length; i; i-=2 ) | |
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168 buffer[i] = buffer [i-1] = buffer[i/2]; |
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169 return length; |
| 338 | 170 } |
| 171 | |
| 172 /*-------------------------------------------------------------------------*/ | |
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173 static int minus5dB( AdapterC Data, int length ) |
| 338 | 174 { |
| 175 int i; | |
| 176 short* buffer = Data.buffer; | |
| 177 for(i = length; i >= 0; i--) | |
| 178 buffer[i]= 38084 * buffer[i] >> 16; | |
| 179 return length; | |
| 180 } | |
| 181 | |
| 182 /*-------------------------------------------------------------------------*/ | |
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183 static int doubleRateStereo( AdapterC Data, int length ) |
| 338 | 184 { |
| 185 _doubleRate2( Data.buffer, Data.mode, length/2 ); | |
| 186 return 2*_doubleRate2( Data.buffer+1, Data.mode, length/2 ); | |
| 187 } | |
| 188 | |
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189 static int doubleRateMono( AdapterC Data, int length ) |
| 338 | 190 { |
| 191 return _doubleRate1( Data.buffer, Data.mode, length ); | |
| 192 } | |
| 193 | |
| 194 /*-------------------------------------------------------------------------*/ | |
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195 static int halfRateStereo( AdapterC Data, int length ) |
| 338 | 196 { |
| 197 _halfRate2( Data.buffer, Data.mode, length/2 ); | |
| 198 return 2*_halfRate2( Data.buffer+1, Data.mode, length/2 ); | |
| 199 } | |
| 200 | |
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201 static int halfRateMono( AdapterC Data, int length ) |
| 338 | 202 { |
| 203 return _halfRate2( Data.buffer, Data.mode, length ); | |
| 204 } | |
| 205 | |
| 206 /*-------------------------------------------------------------------------*/ | |
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207 static int varRateStereo( AdapterC Data, int length ) |
| 338 | 208 { |
| 209 _varRate2( Data.buffer, Data.mode, Data.filter, length/2 ); | |
| 210 return 2*_varRate2( Data.buffer+1, Data.mode, Data.filter, length/2 ); | |
| 211 } | |
| 212 | |
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213 static int varRateMono( AdapterC Data, int length ) |
| 338 | 214 { |
| 215 return _varRate1( Data.buffer, Data.mode, Data.filter, length ); | |
| 216 } | |
| 217 | |
| 218 /*-------------------------------------------------------------------------*/ | |
| 219 typedef struct{ | |
| 220 short denominator; | |
| 221 short numerator; | |
| 222 } Fraction; | |
| 223 | |
| 224 /*-------------------------------------------------------------------------*/ | |
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225 static Fraction findFraction( float Value ) |
| 338 | 226 { |
| 227 /* gives a maximal error of 3% and typical less than 0.2% */ | |
| 228 const char frac[96]={ | |
| 229 1, 2, -1, /* /1 */ | |
| 230 1, 3, -1, /* /2 */ | |
| 231 2, 4, 5, -1, /* /3 */ | |
| 232 3, 5, 7, -1, /* /4 */ | |
| 233 3, 4, 6, 7, 8, 9, -1, /* /5 */ | |
| 234 5, 7, 11, -1, /* /6 */ | |
| 235 4, 5, 6, 8, 9, 10, 11, 12, 13, -1, /* /7 */ | |
| 236 5, 7, 9, 11, 13, 15, -1, /* /8 */ | |
| 237 5, 7, 8, 10, 11, 13, 14, 16, -1, /* /9 */ | |
| 238 7, 9, 11, 13, -1, /* /10 */ | |
| 239 6, 7, 8, 9, 10, 12, 13, 14, 15, 16, -1, /* /11 */ | |
| 240 7, 11, 13, -1, /* /12 */ | |
| 241 7, 8, 9, 10, 11, 12, 14, 15, 16, -1, /* /13 */ | |
| 242 9, 11, 13, 15, -1, /* /14 */ | |
| 243 8, 11, 13, 14, 16, -1, /* /15 */ | |
| 244 9, 11, 13, 15 }; /* /16 */ | |
| 245 | |
| 246 | |
| 247 Fraction Result = {0,0}; | |
| 248 int n,num,den=2; | |
| 249 | |
| 250 float RelErr, BestErr = 0; | |
| 251 if( Value < 31/64. || Value > 64/31. ) return Result; | |
| 252 | |
| 253 for( n = 0; n < sizeof(frac); num=frac[++n] ) | |
| 254 { | |
| 255 if( num < 0 ) den++; | |
| 256 RelErr = Value * num / den; | |
| 257 RelErr = ( RelErr < (1/RelErr) ? RelErr : 1/RelErr ); | |
| 258 if( RelErr > BestErr ) | |
| 259 { | |
| 260 BestErr = RelErr; | |
| 261 Result.denominator = den; | |
| 262 Result.numerator = num; | |
| 263 } | |
| 264 } | |
| 265 return Result; | |
| 266 } | |
| 267 | |
| 268 | |
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269 static float sinc( float x ) |
| 338 | 270 { |
| 271 if( x > -1e-24 && x < 1e-24 ) return 1.; | |
| 272 else return sin(x)/x; | |
| 273 } | |
| 274 | |
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275 static void calculateVarFilter( short* dst, float Ratio, float phase, |
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276 float scale ) |
| 338 | 277 { |
| 278 const unsigned short KaiserWindow7[]= { | |
| 279 22930, 16292, 14648, 14288, 14470, 14945, 15608, 16404, | |
| 280 17304, 18289, 19347, 20467, 21644, 22872, 24145, 25460, | |
| 281 26812, 28198, 29612, 31052, 32513, 33991, 35482, 36983, | |
| 282 38487, 39993, 41494, 42986, 44466, 45928, 47368, 48782, | |
| 283 50165, 51513, 52821, 54086, 55302, 56466, 57575, 58624, | |
| 284 59610, 60529, 61379, 62156, 62858, 63483, 64027, 64490, | |
| 285 64870, 65165, 65375, 65498, 65535, 65484, 65347, 65124, | |
| 286 64815, 64422, 63946, 63389, 62753, 62039, 61251, 60391 }; | |
| 287 int i; | |
| 288 float w; | |
| 289 const float fg = -.018 + .5 / Ratio; | |
| 290 const float omega = 2 * M_PI * fg; | |
| 291 phase -= 63; | |
| 292 for( i = 0; i < 64; i++) | |
| 293 { | |
| 294 w = scale * ( KaiserWindow7[i] * ( i + 1 )); | |
| 295 dst[i] = w * sinc( omega * (i+phase) ); | |
| 296 dst[127-i] = w * sinc( omega * (127-i+phase) ); | |
| 297 } | |
| 298 } | |
| 299 | |
| 300 typedef struct{ | |
| 301 float scale; | |
| 302 int incr; | |
| 303 } VarFilterMode; | |
| 304 | |
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305 static const VarFilterMode Up = { 0.0211952, 0 }; |
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306 static const VarFilterMode Down = { 0.0364733, 2 }; |
| 338 | 307 |
| 308 | |
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309 static void setupVarFilter( VarFilter* filter, |
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310 float Ratio, VarFilterMode Direction ) |
| 338 | 311 { |
| 312 int i,n,d; | |
| 313 Fraction IRatio; | |
| 314 float phase; | |
| 315 IRatio = findFraction( Ratio ); | |
| 316 | |
| 317 if ( (1/Ratio) < Ratio ) | |
| 318 Ratio = 1/Ratio; | |
| 319 | |
| 320 n = IRatio.numerator; | |
| 321 d = IRatio.denominator; | |
| 322 filter->pos_mod = n; | |
| 323 | |
| 324 for( i = 0; i < d; i++ ) | |
| 325 { | |
| 326 if( phase >= n ) | |
| 327 { | |
| 328 phase -= d; | |
| 329 filter->incr[i] = Direction.incr; | |
| 330 } | |
| 331 else | |
| 332 filter->incr[i] = 1; | |
| 333 | |
| 334 calculateVarFilter( filter->c[i], Ratio, phase/(float)n, | |
| 335 Direction.scale ); | |
| 336 phase += d; | |
| 337 } | |
| 338 } | |
| 339 | |
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340 static int createRateConverter( Sound_AudioCVT *Data, int filter_index, |
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341 int SrcRate, int DestRate, int Channel ) |
| 338 | 342 { |
| 343 int VarPos = 0; | |
| 344 int Mono = 2 - Channel; | |
| 345 float Ratio = DestRate; | |
| 346 if( SrcRate < 1 || SrcRate > 1<<18 || | |
| 347 DestRate < 1 || DestRate > 1<<18 ) return -1; | |
| 348 if( SrcRate == DestRate ) return filter_index; | |
| 349 Ratio /= SrcRate; | |
| 350 | |
| 351 if( Ratio > 1.0) | |
| 352 VarPos = filter_index++; | |
| 353 else | |
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354 { |
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355 fprintf (stderr, "Filter: minus5dB\n"); |
| 338 | 356 Data->adapter[filter_index++] = minus5dB; |
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357 } |
| 338 | 358 |
| 359 while( Ratio > 64.0/31.0) | |
| 360 { | |
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361 fprintf (stderr, "Filter: %s\n", |
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362 Mono ? "doubleRateMono" : "doubleRateStereo"); |
| 338 | 363 Data->adapter[filter_index++] = |
| 364 Mono ? doubleRateMono : doubleRateStereo; | |
| 365 Ratio /= 2; | |
| 366 Data->len_mult *= 2; | |
| 367 Data->add *= 2; | |
| 368 Data->add += _fsize; | |
| 369 } | |
| 370 | |
| 371 while( Ratio < 31.0/64.0 ) | |
| 372 { | |
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373 fprintf (stderr, "Filter: %s\n", |
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374 Mono ? "halfRateMono" : "halfRateStereo"); |
| 338 | 375 Data->adapter[filter_index++] = |
| 376 Mono ? halfRateMono : halfRateStereo; | |
| 377 Ratio *= 2; | |
| 378 } | |
| 379 | |
| 380 if( Ratio > 1.0 ) | |
| 381 { | |
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382 fprintf (stderr, "Filter: %s\n", |
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383 Mono ? "varRateMono" : "varRateStereo"); |
| 338 | 384 setupVarFilter( &Data->filter, Ratio, Up ); |
| 385 Data->adapter[VarPos] = | |
| 386 Mono ? varRateMono : varRateStereo; | |
| 387 Data->len_mult *= 2; | |
| 388 Data->add *= 2; | |
| 389 Data->add += _fsize; | |
| 390 } | |
| 391 else | |
| 392 { | |
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393 fprintf (stderr, "Filter: %s\n", |
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394 Mono ? "varRateMono" : "varRateStereo"); |
| 338 | 395 setupVarFilter( &Data->filter, Ratio, Down ); |
| 396 Data->adapter[filter_index++] = | |
| 397 Mono ? varRateMono : varRateStereo; | |
| 398 } | |
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399 return filter_index; |
| 338 | 400 } |
| 401 | |
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402 static int BuildAudioCVT(Sound_AudioCVT *Data, |
| 338 | 403 Uint16 src_format, Uint8 src_channels, int src_rate, |
| 404 Uint16 dst_format, Uint8 dst_channels, int dst_rate) | |
| 405 { | |
| 406 int filter_index = 0; | |
| 407 | |
| 408 if( Data == NULL ) return -1; | |
| 409 Data->len_mult = 1.0; | |
| 410 Data->add = 0; | |
| 411 | |
| 412 /* Check channels */ | |
| 413 if( src_channels < 1 || src_channels > 2 || | |
| 414 dst_channels < 1 || dst_channels > 2 ) goto error_exit; | |
| 415 | |
| 416 /* First filter: Size/Endian conversion */ | |
| 417 switch( src_format & AUDIO_FORMAT) | |
| 418 { | |
| 419 case AUDIO_8: | |
|
342
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420 fprintf (stderr, "Filter: expand8BitTo16Bit\n"); |
| 338 | 421 Data->adapter[filter_index++] = expand8BitTo16Bit; |
| 422 Data->len_mult *= 2; | |
| 423 break; | |
| 424 case AUDIO_16WRONG: | |
|
342
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425 fprintf (stderr, "Filter: swapBytes\n"); |
| 338 | 426 Data->adapter[filter_index++] = swapBytes; |
|
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427 break; |
| 338 | 428 } |
| 429 | |
| 430 /* Second adapter: Sign conversion -- unsigned/signed */ | |
| 431 if( src_format & AUDIO_SIGN ) | |
|
342
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432 { |
|
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433 fprintf (stderr, "Filter: changeSigned\n"); |
| 338 | 434 Data->adapter[filter_index++] = changeSigned; |
|
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|
435 } |
| 338 | 436 |
| 437 /* Third adapter: Stereo->Mono conversion */ | |
| 438 if( src_channels == 2 && dst_channels == 1 ) | |
|
342
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|
439 { |
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440 fprintf (stderr, "convertStereoToMono\n"); |
| 338 | 441 Data->adapter[filter_index++] = convertStereoToMono; |
|
342
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442 } |
| 338 | 443 |
| 444 /* Do rate conversion */ | |
| 445 if( src_channels == 2 && dst_channels == 2 ) | |
| 446 filter_index = createRateConverter( Data, filter_index, | |
| 447 src_rate, dst_rate, 2 ); | |
| 448 else | |
| 449 filter_index = createRateConverter( Data, filter_index, | |
| 450 src_rate, dst_rate, 1 ); | |
| 451 | |
| 452 if( filter_index < 0 ) goto error_exit; /* propagate error */ | |
| 453 | |
| 454 /* adapter: Mono->Stereo conversion */ | |
| 455 if( src_channels == 1 && dst_channels == 2 ){ | |
|
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456 fprintf (stderr, "Filter: convertMonoToStereo\n"); |
| 338 | 457 Data->adapter[filter_index++] = convertMonoToStereo; |
| 458 Data->add *= 2; | |
| 459 Data->len_mult *= 2; | |
| 460 } | |
| 461 | |
| 462 /* adapter: final Sign conversion -- unsigned/signed */ | |
| 463 if( dst_format & AUDIO_SIGN ) | |
|
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464 { |
|
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465 fprintf (stderr, "Filter: changeSigned\n"); |
| 338 | 466 Data->adapter[filter_index++] = changeSigned; |
|
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467 } |
| 338 | 468 |
| 469 /* final adapter: Size/Endian conversion */ | |
| 470 switch( dst_format & AUDIO_FORMAT) | |
| 471 { | |
| 472 case AUDIO_8: | |
|
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473 fprintf (stderr, "Filter: cut16BitTo8Bit\n"); |
| 338 | 474 Data->adapter[filter_index++] = cut16BitTo8Bit; |
| 475 break; | |
| 476 case AUDIO_16WRONG: | |
|
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477 fprintf (stderr, "Filter: swapBytes\n"); |
| 338 | 478 Data->adapter[filter_index++] = swapBytes; |
|
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479 break; |
| 338 | 480 } |
| 481 /* Set up the filter information */ | |
| 482 Data->adapter[filter_index] = NULL; | |
| 483 Data->needed = (filter_index > 0); | |
| 484 return 0; | |
| 485 | |
| 486 error_exit: | |
| 487 Data->adapter[0] = NULL; | |
| 488 return -1; | |
| 489 } | |
|
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490 |
|
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491 /* |
|
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492 * Frank's audio converter has its own ideas about how to represent audio |
|
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493 * format, so at least for a transition period we use this to glue his code |
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494 * to our's. |
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495 * |
|
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496 * + The expand8BitTo16Bit filter will only convert to system byte order. |
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497 * + The cut16BitTo8Bit filter will only convert from system byte order. |
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498 * + The changeSigned filter only works on 16-bit samples, system byte order. |
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499 */ |
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500 |
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501 static char *fmt_to_str(Uint16 fmt) |
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502 { |
|
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503 switch (fmt) |
|
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|
504 { |
|
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505 case AUDIO_U8: return " U8"; break; |
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506 case AUDIO_S8: return " S8"; break; |
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|
507 case AUDIO_U16MSB: return "U16MSB"; break; |
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508 case AUDIO_S16MSB: return "S16MSB"; break; |
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509 case AUDIO_U16LSB: return "U16LSB"; break; |
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|
510 case AUDIO_S16LSB: return "S16LSB"; break; |
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|
511 } |
|
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512 return "??????"; |
|
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|
513 } |
|
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|
514 |
|
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|
515 #define IS_8BIT(x) ((x) & 0x0008) |
|
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|
516 #define IS_16BIT(x) ((x) & 0x0010) |
|
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|
517 #define ENDIAN(x) ((x) & 0x1000) |
|
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|
518 #define SIGNED(x) ((x) & 0x8000) |
|
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|
519 |
|
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|
520 int Sound_BuildAudioCVT(Sound_AudioCVT *Data, |
|
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|
521 Uint16 src_in_format, Uint8 src_channels, int src_rate, |
|
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|
522 Uint16 dst_in_format, Uint8 dst_channels, int dst_rate) |
|
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|
523 { |
|
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|
524 Uint16 src_format = 0; |
|
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|
525 Uint16 dst_format = 0; |
|
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|
526 |
|
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diff
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|
527 fprintf (stderr, |
|
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diff
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|
528 "format: %s -> %s\n" |
|
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diff
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|
529 "channels: %6d -> %6d\n" |
|
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diff
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|
530 "rate: %6d -> %6d\n", |
|
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diff
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|
531 fmt_to_str (src_in_format), fmt_to_str (dst_in_format), |
|
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|
532 src_channels, dst_channels, |
|
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|
533 src_rate, dst_rate); |
|
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diff
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|
534 |
|
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|
535 if ( IS_8BIT(src_in_format) && IS_16BIT(dst_in_format) ) |
|
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Cleanups by Torbj�rn.
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diff
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|
536 { |
|
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diff
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|
537 src_format |= AUDIO_8; |
|
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diff
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|
538 |
|
fbbb1f25b944
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|
539 /* |
|
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|
540 * Signedness and byte-order changes must wait until the data |
|
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|
541 * has been converted to 16-bit samples. |
|
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diff
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|
542 */ |
|
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diff
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|
543 if ( SIGNED(src_in_format) != SIGNED(dst_in_format) ) |
|
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Cleanups by Torbj�rn.
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|
544 { |
|
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diff
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|
545 dst_format |= AUDIO_SIGN; |
|
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diff
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|
546 } /* if */ |
|
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|
547 |
|
fbbb1f25b944
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diff
changeset
|
548 if ( ENDIAN(dst_in_format) != ENDIAN(AUDIO_U16SYS) ) |
|
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diff
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|
549 { |
|
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diff
changeset
|
550 dst_format |= AUDIO_16WRONG; |
|
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diff
changeset
|
551 } /* if */ |
|
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diff
changeset
|
552 } /* if */ |
|
fbbb1f25b944
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diff
changeset
|
553 else if ( IS_16BIT(src_in_format) && IS_8BIT(dst_in_format) ) |
|
fbbb1f25b944
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diff
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|
554 { |
|
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diff
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|
555 dst_format |= AUDIO_8; |
|
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diff
changeset
|
556 |
|
fbbb1f25b944
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diff
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|
557 /* |
|
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diff
changeset
|
558 * Byte-order and signedness changes must be made before the data |
|
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|
559 * has been converted to 8-bit samples. |
|
fbbb1f25b944
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diff
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|
560 */ |
|
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diff
changeset
|
561 if ( ENDIAN(src_in_format) != ENDIAN(AUDIO_U16SYS) ) |
|
fbbb1f25b944
Cleanups by Torbj�rn.
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diff
changeset
|
562 { |
|
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diff
changeset
|
563 src_format |= AUDIO_16WRONG; |
|
fbbb1f25b944
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diff
changeset
|
564 } /* if */ |
|
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diff
changeset
|
565 |
|
fbbb1f25b944
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diff
changeset
|
566 if ( SIGNED(src_in_format) != SIGNED(dst_in_format) ) |
|
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diff
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|
567 { |
|
fbbb1f25b944
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diff
changeset
|
568 src_format |= AUDIO_SIGN; |
|
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|
569 } /* if */ |
|
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|
570 } /* else if */ |
|
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changeset
|
571 else if ( IS_16BIT(src_in_format) && IS_16BIT(dst_in_format) ) |
|
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parents:
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diff
changeset
|
572 { |
|
fbbb1f25b944
Cleanups by Torbj�rn.
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parents:
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diff
changeset
|
573 if ( ENDIAN(src_in_format) != ENDIAN(dst_in_format) ) |
|
fbbb1f25b944
Cleanups by Torbj�rn.
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parents:
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diff
changeset
|
574 { |
|
fbbb1f25b944
Cleanups by Torbj�rn.
Ryan C. Gordon <icculus@icculus.org>
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338
diff
changeset
|
575 if ( ENDIAN(src_in_format) == ENDIAN(AUDIO_U16SYS) ) |
|
fbbb1f25b944
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parents:
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diff
changeset
|
576 { |
|
fbbb1f25b944
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diff
changeset
|
577 dst_format |= AUDIO_16WRONG; |
|
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diff
changeset
|
578 |
|
fbbb1f25b944
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Ryan C. Gordon <icculus@icculus.org>
parents:
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579 /* |
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580 * The data is already is system byte order, so any |
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581 * signedness change has to be made before changing byte |
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582 * order. |
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583 */ |
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584 if ( SIGNED(src_in_format) != SIGNED(dst_in_format) ) |
|
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585 { |
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586 src_format |= AUDIO_SIGN; |
|
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587 } /* if */ |
|
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588 } /* if */ |
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589 else |
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590 { |
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591 src_format |= AUDIO_16WRONG; |
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592 |
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593 /* |
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594 * The data is not in system byte order, so any signedness |
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595 * change has to be made after changing byte order. |
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596 */ |
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597 if ( SIGNED(src_in_format) != SIGNED(dst_in_format) ) |
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598 { |
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599 dst_format |= AUDIO_SIGN; |
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600 } /* if */ |
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601 } /* else */ |
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602 } /* if */ |
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603 else if ( ENDIAN(src_in_format) != SIGNED(AUDIO_U16SYS) ) |
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604 { |
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605 if ( SIGNED(src_in_format) != SIGNED(dst_in_format) ) |
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606 { |
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607 /* |
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608 * !!! FIXME !!! |
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609 * |
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610 * The changeSigned filter only works on system byte |
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611 * order. In this case, both source and destination is |
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612 * in opposite byte order, but the sign has to changed |
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613 * so we need to convert to system byte order, change |
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614 * sign, and then convert back to the original byte |
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615 * order again. This is not an optimal solution. |
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616 */ |
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617 src_format |= ( AUDIO_16WRONG | AUDIO_SIGN ); |
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618 dst_format |= AUDIO_16WRONG; |
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619 } /* if */ |
|
347
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620 |
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621 if ( src_rate != dst_rate ) |
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622 { |
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623 /* |
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624 * !!! FIXME !!! |
|
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625 * |
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626 * The audio conversion filter probably only works if the |
|
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627 * data is in system byte order. So we need to convert to |
|
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628 * system byte order, and then back to original byte |
|
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629 * order. This is not an optimal solution. |
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630 */ |
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631 src_format |= AUDIO_16WRONG; |
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632 dst_format |= AUDIO_16WRONG; |
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633 } |
|
342
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634 } /* else if */ |
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635 else if ( SIGNED(src_in_format) != SIGNED(dst_in_format) ) |
|
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|
636 { |
|
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|
637 src_format |= AUDIO_SIGN; |
|
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638 } /* else if */ |
|
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639 } /* else if */ |
|
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640 else if ( IS_8BIT(src_in_format) && IS_8BIT(dst_in_format) ) |
|
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|
641 { |
|
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|
642 /* |
|
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diff
changeset
|
643 * !!! FIXME !!! |
|
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|
644 * |
|
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changeset
|
645 * The changeSigned filter only works on 16-bit samples, so if |
|
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|
646 * the signedness differs we have to convert from 8 to 16 bits, |
|
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|
647 * change the sign and then convert back to 8 bits again. This |
|
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|
648 * is not an optimal solution. |
|
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parents:
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changeset
|
649 */ |
|
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changeset
|
650 if ( SIGNED(src_in_format) != SIGNED(dst_in_format) ) |
|
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|
651 { |
|
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changeset
|
652 src_format |= ( AUDIO_8 | AUDIO_SIGN ); |
|
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changeset
|
653 dst_format |= AUDIO_8; |
|
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parents:
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diff
changeset
|
654 } /* if */ |
|
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diff
changeset
|
655 |
|
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Cleanups by Torbj�rn.
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changeset
|
656 /* |
|
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Ryan C. Gordon <icculus@icculus.org>
parents:
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diff
changeset
|
657 * !!! FIXME !!! |
|
fbbb1f25b944
Cleanups by Torbj�rn.
Ryan C. Gordon <icculus@icculus.org>
parents:
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diff
changeset
|
658 * |
|
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Cleanups by Torbj�rn.
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parents:
338
diff
changeset
|
659 * The convertMonoToStereo and convertStereoToMono filters only |
|
fbbb1f25b944
Cleanups by Torbj�rn.
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diff
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|
660 * work with 16-bit samples. So if those are to be applied, we |
|
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Cleanups by Torbj�rn.
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parents:
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diff
changeset
|
661 * need to convert to 16-bit samples, and then back again. |
|
fbbb1f25b944
Cleanups by Torbj�rn.
Ryan C. Gordon <icculus@icculus.org>
parents:
338
diff
changeset
|
662 */ |
|
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Cleanups by Torbj�rn.
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parents:
338
diff
changeset
|
663 if ( src_channels != dst_channels ) |
|
fbbb1f25b944
Cleanups by Torbj�rn.
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parents:
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diff
changeset
|
664 { |
|
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parents:
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changeset
|
665 src_format |= AUDIO_8; |
|
fbbb1f25b944
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parents:
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diff
changeset
|
666 dst_format |= AUDIO_8; |
|
fbbb1f25b944
Cleanups by Torbj�rn.
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parents:
338
diff
changeset
|
667 } /* if */ |
|
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parents:
338
diff
changeset
|
668 |
|
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Cleanups by Torbj�rn.
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parents:
338
diff
changeset
|
669 /* |
|
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parents:
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diff
changeset
|
670 * !!! FIXME !!! |
|
fbbb1f25b944
Cleanups by Torbj�rn.
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parents:
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diff
changeset
|
671 * |
|
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Cleanups by Torbj�rn.
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parents:
338
diff
changeset
|
672 * The rate conversion filters almost certainly only work with |
|
fbbb1f25b944
Cleanups by Torbj�rn.
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parents:
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diff
changeset
|
673 * 16-bit samples. Yadda, yadda, yadda. |
|
fbbb1f25b944
Cleanups by Torbj�rn.
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parents:
338
diff
changeset
|
674 */ |
|
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parents:
338
diff
changeset
|
675 if ( src_rate != dst_rate ) |
|
fbbb1f25b944
Cleanups by Torbj�rn.
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parents:
338
diff
changeset
|
676 { |
|
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parents:
338
diff
changeset
|
677 src_format |= AUDIO_8; |
|
fbbb1f25b944
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parents:
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diff
changeset
|
678 dst_format |= AUDIO_8; |
|
fbbb1f25b944
Cleanups by Torbj�rn.
Ryan C. Gordon <icculus@icculus.org>
parents:
338
diff
changeset
|
679 } /* if */ |
|
fbbb1f25b944
Cleanups by Torbj�rn.
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parents:
338
diff
changeset
|
680 } /* else if */ |
|
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Cleanups by Torbj�rn.
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parents:
338
diff
changeset
|
681 |
|
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parents:
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diff
changeset
|
682 return BuildAudioCVT(Data, src_format, src_channels, src_rate, |
|
fbbb1f25b944
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parents:
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diff
changeset
|
683 dst_format, dst_channels, dst_rate); |
|
fbbb1f25b944
Cleanups by Torbj�rn.
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parents:
338
diff
changeset
|
684 } |
|
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diff
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|
685 |
| 338 | 686 /*-------------------------------------------------------------------------*/ |
| 687 |