Mercurial > SDL_sound_CoreAudio
changeset 360:c984aa6990f7
Fixes and enhancements from Frank Ranostaj.
| author | Ryan C. Gordon <icculus@icculus.org> |
|---|---|
| date | Wed, 12 Jun 2002 08:35:23 +0000 |
| parents | 2b7969823778 |
| children | 473e6217fdfa |
| files | alt_audio_convert.c alt_audio_convert.h |
| diffstat | 2 files changed, 647 insertions(+), 523 deletions(-) [+] |
line wrap: on
line diff
--- a/alt_audio_convert.c Tue Jun 11 23:33:27 2002 +0000 +++ b/alt_audio_convert.c Wed Jun 12 08:35:23 2002 +0000 @@ -1,49 +1,50 @@ /* - Extended Audio Converter for SDL (Simple DirectMedia Layer) - Copyright (C) 2002 Frank Ranostaj - Institute of Applied Physik - Johann Wolfgang Goethe-Universität - Frankfurt am Main, Germany + Extended Audio Converter for SDL (Simple DirectMedia Layer) + Copyright (C) 2002 Frank Ranostaj + Institute of Applied Physik + Johann Wolfgang Goethe-Universität + Frankfurt am Main, Germany - This library is free software; you can redistribute it and/or - modify it under the terms of the GNU Library General Public - License as published by the Free Software Foundation; either - version 2 of the License, or (at your option) any later version. + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Library General Public + License as published by the Free Software Foundation; either + version 2 of the License, or (at your option) any later version. - This library is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the implied warranty of - MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - Library General Public License for more details. + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Library General Public License for more details. - You should have received a copy of the GNU Library General Public - License along with this library; if not, write to the Free - Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + You should have received a copy of the GNU Library General Public + License along with this library; if not, write to the Free + Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - Frank Ranostaj - ranostaj@stud.uni-frankfurt.de + Frank Ranostaj + ranostaj@stud.uni-frankfurt.de (This code blatantly abducted for SDL_sound. Thanks, Frank! --ryan.) + */ - #include "alt_audio_convert.h" - #include <stdlib.h> #include <math.h> -/*provisorisch*/ -#define AUDIO_8 (4) -#define AUDIO_16WRONG (8) -#define AUDIO_FORMAT (AUDIO_8 | AUDIO_16WRONG) -#define AUDIO_SIGN (1) +/* some macros for "parsing" format */ + +#define IS_8BIT(x) ((x).format & 0x0008) +#define IS_16BIT(x) ((x).format & 0x0010) +#define IS_FLOAT(x) ((x).format & 0x0020) /* !!! FIXME: is this ok? */ +#define IS_SIGNED(x) ((x).format & 0x8000) +#define IS_SYSENDIAN(x) ((AUDIO_U16SYS ^ (x).format) & 0x1000) /*-------------------------------------------------------------------------*/ -/* this filter (Kaiser-window beta=6.8) gives a decent -80dB attentuation */ +/* this filter (Kaiser-window beta=6.8) gives a decent -80dB attentuation */ static const int filter[_fsize/2] = { - 0, 20798, 0, -6764, 0, 3863, 0, -2560, 0, 1800, 0, -1295, 0, 936, 0, --671, - 0, 474, 0, -326, 0, 217, 0, -138, 0, 83, 0, -46, 0, 23, 0, -9 -}; + 0, 20798, 0, -6764, 0, 3863, 0, -2560, + 0, 1800, 0, -1295, 0, 936, 0, -671, + 0, 474, 0, -326, 0, 217, 0, -138, + 0, 83, 0, -46, 0, 23, 0, -9 }; /* Mono (1 channel ) */ #define Suffix(x) x##1 @@ -55,633 +56,745 @@ #include "filter_templates.h" #undef Suffix -/* !!! FIXME: Lose all the "short" vars for "Sint16", etc. */ +/*-------------------------------------------------------------------------*/ +static int ConvertAudio( Sound_AudioCVT *Data, + Uint8* buffer, int length, int mode ) +{ + AdapterC Temp; + int i; + + /* Make sure there's a converter */ + if( Data == NULL ) { + SDL_SetError("No converter given"); + return(-1); + } -/*-------------------------------------------------------------------------*/ + /* Make sure there's data to convert */ + if( buffer == NULL ) { + SDL_SetError("No buffer allocated for conversion"); + return(-1); + } + + /* Set up the conversion and go! */ + Temp.buffer = buffer; + Temp.mode = mode; + Temp.filter = &Data->filter; + + for( i = 0; Data->adapter[i] != NULL; i++ ) + length = (*Data->adapter[i])( Temp, length); + + return length; +} + int Sound_ConvertAudio( Sound_AudioCVT *Data ) { - AdapterC Temp; - int i; - - /* !!! FIXME: Try the looping stuff under certain circumstances? --ryan. */ - int mode = 0; + /* !!! FIXME: Try the looping stuff under certain circumstances? --ryan. */ + return ConvertAudio( Data, Data->buf, Data->len, 0 ); +} - /* Make sure there's a converter */ - if( Data == NULL ) { - SDL_SetError("No converter given"); - return(-1); - } - /* Make sure there's data to convert */ - if( Data->buf == NULL ) { - SDL_SetError("No buffer allocated for conversion"); - return(-1); - } +/*-------------------------------------------------------------------------*/ +static int expand8BitTo16BitSys( AdapterC Data, int length ) +{ + int i; + Uint8* inp = Data.buffer; + Uint16* buffer = (Uint16*)Data.buffer; + for( i = length; i--; ) + buffer[i] = inp[i]<<8; + return 2*length; +} - /* Set up the conversion and go! */ - Temp.buffer = (short*)Data->buf; - Temp.mode = mode; - Temp.filter = &Data->filter; - - Data->len_cvt = Data->len; - for( i = 0; Data->adapter[i] != NULL; i++ ) - Data->len_cvt = (*Data->adapter[i])( Temp, Data->len_cvt); - - return(0); +static int expand8BitTo16BitWrong( AdapterC Data, int length ) +{ + int i; + Uint8* inp = Data.buffer; + Uint16* buffer = (Uint16*)Data.buffer; + for( i = length; i--; ) + buffer[i] = inp[i]; + return 2*length; } /*-------------------------------------------------------------------------*/ -static int expand8BitTo16Bit( AdapterC Data, int length ) +static int expand16BitToFloat( AdapterC Data, int length ) { - int i; - char* inp = (char*)Data.buffer-1; - short* buffer = Data.buffer-1; - for( i = length; i; i-- ) - buffer[i] = inp[i]<<8; - return length*2; + int i; + Sint16* inp = (Sint16*)Data.buffer; + float* buffer = (float*)Data.buffer; + for( i = length>>1; i--; ) + buffer[i] = inp[i]*(1./32767); + return 2*length; } /*-------------------------------------------------------------------------*/ static int swapBytes( AdapterC Data, int length ) { - int i; - unsigned short a,b; - short* buffer = Data.buffer; - for( i = 0; i < length; i++ ) - { - a = b = buffer[i]; - a <<= 8; - b >>= 8; - buffer[i] = a | b; - } - return length; + /* + * !!! FIXME !!! + * + * + * Use the faster SDL-Macros to swap + * - Frank + */ + + int i; + Uint16 a,b; + Uint16* buffer = (Uint16*) Data.buffer; + for( i = length>>1; i --; ) + { + a = b = buffer[i]; + buffer[i] = ( a << 8 ) | ( b >> 8 ); + } + return length; } /*-------------------------------------------------------------------------*/ -static int cut16BitTo8Bit( AdapterC Data, int length ) +static int cutFloatTo16Bit( AdapterC Data, int length ) { - int i; - short* inp = Data.buffer-1; - char* buffer = (char*)Data.buffer-1; - for( i = 0; i < length; i++ ) - buffer[i] = inp[i]>>8; - return length/2; + int i; + float* inp = (float*) Data.buffer; + Sint16* buffer = (Sint16*) Data.buffer; + length>>=2; + for( i = 0; i < length; i++ ) + { + if( inp[i] > 1. ) + buffer[i] = 32767; + else if( inp[i] < -1. ) + buffer[i] = -32768; + else + buffer[i] = 32767 * inp[i]; + } + return 2*length; +} + +/*-------------------------------------------------------------------------*/ +static int cut16BitSysTo8Bit( AdapterC Data, int length ) +{ + int i; + Uint16* inp = (Uint16*) Data.buffer; + Uint8* buffer = Data.buffer; + length >>= 1; + for( i = 0; i < length; i++ ) + buffer[i] = inp[i]>>8; + return length; +} + +static int cut16BitWrongTo8Bit( AdapterC Data, int length ) +{ + int i; + Uint16* inp = (Uint16*) Data.buffer; + Uint8* buffer = Data.buffer; + length >>= 1; + for( i = 0; i < length; i++ ) + buffer[i] = inp[i] & 0xff; + return length; } /*-------------------------------------------------------------------------*/ -static int changeSigned( AdapterC Data, int length ) +static int changeSigned( AdapterC Data, int length, int XOR ) { - int i; - short* buffer = Data.buffer; - for( i = 0; i < length; i++ ) - buffer[i] ^= 0x8000; - return length; + int i; + Uint32* buffer = (Uint32*) Data.buffer; + for( i = length>>2; i--; ) + buffer[i] ^= XOR; + for( i = 4*(length>>2); i < length; i++) + ((Uint8*)buffer)[i] ^= ((Uint8*)&XOR)[i&3]; + return length; +} + +static int changeSigned16BitSys( AdapterC Data, int length ) +{ + return changeSigned( Data, length, 0x80008000 ); +} + +static int changeSigned16BitWrong( AdapterC Data, int length ) +{ + return changeSigned( Data, length, 0x00800080 ); +} + +static int changeSigned8Bit( AdapterC Data, int length ) +{ + return changeSigned( Data, length, 0x80808080 ); } /*-------------------------------------------------------------------------*/ -static int convertStereoToMono( AdapterC Data, int length ) +static int convertStereoToMonoS16Bit( AdapterC Data, int length ) { - int i; - short* buffer = Data.buffer; + int i; + Sint16* buffer = (Sint16*) Data.buffer; + Sint16* src = (Sint16*) Data.buffer; + length >>= 2; + for( i = 0; i < length; i++ ) + buffer[i] = ((int) *(src++) + *(src++) ) >> 1; + return 2*length; +} - /* - * !!! FIXME: Can we avoid the division in this loop and just keep - * !!! FIXME: a second index variable? --ryan. - */ - for( i = 0; i < length; i+=2 ) - buffer[i/2] = ((int)buffer[i] + buffer[i+1] ) >> 1; - return length/2; +static int convertStereoToMonoU16Bit( AdapterC Data, int length ) +{ + int i; + Uint16* buffer = (Uint16*) Data.buffer; + Uint16* src = (Uint16*) Data.buffer; + length >>= 2; + for( i = 0; i < length; i++ ) + buffer[i] = ((int) *(src++) + *(src++) ) >> 1; + return 2*length; +} + +static int convertStereoToMonoS8Bit( AdapterC Data, int length ) +{ + int i; + Sint8* buffer = (Sint8*) Data.buffer; + Sint8* src = (Sint8*) Data.buffer; + length >>= 1; + for( i = 0; i < length; i++ ) + buffer[i] = ((int) *(src++) + *(src++) ) >> 1; + return length; +} + +static int convertStereoToMonoU8Bit( AdapterC Data, int length ) +{ + int i; + Uint8* buffer = (Uint8*) Data.buffer; + Uint8* src = (Uint8*) Data.buffer; + length >>= 1; + for( i = 0; i < length; i++ ) + buffer[i] = ((int) *(src++) + *(src++) ) >> 1; + return length; } /*-------------------------------------------------------------------------*/ -static int convertMonoToStereo( AdapterC Data, int length ) +static int convertMonoToStereo16Bit( AdapterC Data, int length ) { - int i; - short* buffer = Data.buffer-1; - length *= 2; + int i; + Uint16* buffer = (Uint16*) Data.buffer; + Uint16* dst = (Uint16*)Data.buffer + length; + for( i = length>>1; i--; ) + *(--dst) = *(--dst) = buffer[i]; + return 2*length; +} - /* - * !!! FIXME: Can we avoid the division in this loop and just keep - * !!! FIXME: a second index variable? --ryan. - */ - for( i = length; i; i-=2 ) - buffer[i] = buffer [i-1] = buffer[i/2]; - return length; +static int convertMonoToStereo8Bit( AdapterC Data, int length ) +{ + int i; + Uint8* buffer = Data.buffer; + Uint8* dst = Data.buffer + 2*length; + for( i = length; i--; ) + *(--dst) = *(--dst) = buffer[i]; + return 2*length; } /*-------------------------------------------------------------------------*/ static int minus5dB( AdapterC Data, int length ) { - int i; - short* buffer = Data.buffer; - for(i = length; i >= 0; i--) - buffer[i]= 38084 * buffer[i] >> 16; - return length; + int i; + Sint16* buffer = (Sint16*) Data.buffer; + for(i = length>>1; i--; ) + buffer[i]= 38084 * buffer[i] >> 16; + return length; } /*-------------------------------------------------------------------------*/ static int doubleRateStereo( AdapterC Data, int length ) { - _doubleRate2( Data.buffer, Data.mode, length/2 ); - return 2*_doubleRate2( Data.buffer+1, Data.mode, length/2 ); + length >>= 2; + _doubleRate2( (Sint16*)Data.buffer, Data.mode, length ); + return 4*_doubleRate2( (Sint16*)Data.buffer+1, Data.mode, length ); } static int doubleRateMono( AdapterC Data, int length ) { - return _doubleRate1( Data.buffer, Data.mode, length ); + return 2*_doubleRate1( (Sint16*)Data.buffer, Data.mode, length>>1 ); } /*-------------------------------------------------------------------------*/ static int halfRateStereo( AdapterC Data, int length ) { - _halfRate2( Data.buffer, Data.mode, length/2 ); - return 2*_halfRate2( Data.buffer+1, Data.mode, length/2 ); + length >>= 2; + _halfRate2( (Sint16*)Data.buffer, Data.mode, length ); + return 4*_halfRate2( (Sint16*)Data.buffer+1, Data.mode, length ); } static int halfRateMono( AdapterC Data, int length ) { - return _halfRate2( Data.buffer, Data.mode, length ); + return 2*_halfRate2( (Sint16*)Data.buffer, Data.mode, length>>1 ); } /*-------------------------------------------------------------------------*/ -static int varRateStereo( AdapterC Data, int length ) +static int varRateUpStereo( AdapterC Data, int length ) { - _varRate2( Data.buffer, Data.mode, Data.filter, length/2 ); - return 2*_varRate2( Data.buffer+1, Data.mode, Data.filter, length/2 ); + length >>= 2; + _varRateUp2( (Sint16*)Data.buffer, Data.mode, Data.filter, length ); + return 4 * _varRateUp2( (Sint16*)Data.buffer+1, + Data.mode, Data.filter, length ); } -static int varRateMono( AdapterC Data, int length ) +static int varRateUpMono( AdapterC Data, int length ) +{ + return 2 * _varRateUp1( (Sint16*)Data.buffer, + Data.mode, Data.filter, length>>1 ); +} + +static int varRateDownStereo( AdapterC Data, int length ) { - return _varRate1( Data.buffer, Data.mode, Data.filter, length ); + length >>= 2; + _varRateDown2( (Sint16*)Data.buffer, Data.mode, Data.filter, length ); + return 2 * _varRateDown2( (Sint16*)Data.buffer+1, + Data.mode, Data.filter, length ); +} + +static int varRateDownMono( AdapterC Data, int length ) +{ + return _varRateDown1( (Sint16*)Data.buffer, + Data.mode, Data.filter, length>>1 ); } /*-------------------------------------------------------------------------*/ typedef struct{ - short denominator; - short numerator; + Sint16 denominator; + Sint16 numerator; } Fraction; -/*-------------------------------------------------------------------------*/ static Fraction findFraction( float Value ) { /* gives a maximal error of 3% and typical less than 0.2% */ - const char frac[96]={ - 1, 2, -1, /* /1 */ - 1, 3, -1, /* /2 */ - 2, 4, 5, -1, /* /3 */ - 3, 5, 7, -1, /* /4 */ - 3, 4, 6, 7, 8, 9, -1, /* /5 */ - 5, 7, 11, -1, /* /6 */ - 4, 5, 6, 8, 9, 10, 11, 12, 13, -1, /* /7 */ - 5, 7, 9, 11, 13, 15, -1, /* /8 */ - 5, 7, 8, 10, 11, 13, 14, 16, -1, /* /9 */ - 7, 9, 11, 13, -1, /* /10 */ - 6, 7, 8, 9, 10, 12, 13, 14, 15, 16, -1, /* /11 */ - 7, 11, 13, -1, /* /12 */ - 7, 8, 9, 10, 11, 12, 14, 15, 16, -1, /* /13 */ - 9, 11, 13, 15, -1, /* /14 */ - 8, 11, 13, 14, 16, -1, /* /15 */ - 9, 11, 13, 15 }; /* /16 */ + const Uint8 frac[96]={ + 1, 2, -1, /* /1 */ + 1, 3, -1, /* /2 */ + 2, 4, 5, -1, /* /3 */ + 3, 5, 7, -1, /* /4 */ + 3, 4, 6, 7, 8, 9, -1, /* /5 */ + 5, 7, 11, -1, /* /6 */ + 4, 5, 6, 8, 9, 10, 11, 12, 13, -1, /* /7 */ + 5, 7, 9, 11, 13, 15, -1, /* /8 */ + 5, 7, 8, 10, 11, 13, 14, 16, -1, /* /9 */ + 7, 9, 11, 13, -1, /* /10 */ + 6, 7, 8, 9, 10, 12, 13, 14, 15, 16, -1, /* /11 */ + 7, 11, 13, -1, /* /12 */ + 7, 8, 9, 10, 11, 12, 14, 15, 16, -1, /* /13 */ + 9, 11, 13, 15, -1, /* /14 */ + 8, 11, 13, 14, 16, -1, /* /15 */ + 9, 11, 13, 15 }; /* /16 */ - Fraction Result = {0,0}; - int n,num,den=2; + Fraction Result = {0,0}; + int n,num,den=1; - float RelErr, BestErr = 0; - if( Value < 31/64. || Value > 64/31. ) return Result; + float RelErr, BestErr = 0; + if( Value < 31/64. || Value > 64/31. ) return Result; - for( n = 0; n < sizeof(frac); num=frac[++n] ) - { - if( num < 0 ) den++; - RelErr = Value * num / den; - RelErr = ( RelErr < (1/RelErr) ? RelErr : 1/RelErr ); - if( RelErr > BestErr ) - { - BestErr = RelErr; - Result.denominator = den; - Result.numerator = num; - } - } - return Result; + for( n = 0; n < sizeof(frac); num=frac[++n] ) + { + if( num < 0 ) den++; + RelErr = Value * num / den; + RelErr = min( RelErr, 1/RelErr ); + if( RelErr > BestErr ) + { + BestErr = RelErr; + Result.denominator = den; + Result.numerator = num; + } + } + return Result; } - +/*-------------------------------------------------------------------------*/ static float sinc( float x ) { - if( x > -1e-24 && x < 1e-24 ) return 1.; - else return sin(x)/x; + if( x > -1e-24 && x < 1e-24 ) return 1.; + else return sin(x)/x; } -static void calculateVarFilter( short* dst, float Ratio, float phase, - float scale ) +static void calculateVarFilter( Sint16* dst, float Ratio, float phase, float scale ) { - const unsigned short KaiserWindow7[]= { - 22930, 16292, 14648, 14288, 14470, 14945, 15608, 16404, - 17304, 18289, 19347, 20467, 21644, 22872, 24145, 25460, - 26812, 28198, 29612, 31052, 32513, 33991, 35482, 36983, - 38487, 39993, 41494, 42986, 44466, 45928, 47368, 48782, - 50165, 51513, 52821, 54086, 55302, 56466, 57575, 58624, - 59610, 60529, 61379, 62156, 62858, 63483, 64027, 64490, - 64870, 65165, 65375, 65498, 65535, 65484, 65347, 65124, - 64815, 64422, 63946, 63389, 62753, 62039, 61251, 60391 }; - int i; - float w; - const float fg = -.018 + .5 / Ratio; - const float omega = 2 * M_PI * fg; - phase -= 63; - for( i = 0; i < 64; i++) - { - w = scale * ( KaiserWindow7[i] * ( i + 1 )); - dst[i] = w * sinc( omega * (i+phase) ); - dst[127-i] = w * sinc( omega * (127-i+phase) ); - } + const Uint16 KaiserWindow7[]= { + 22930, 16292, 14648, 14288, 14470, 14945, 15608, 16404, + 17304, 18289, 19347, 20467, 21644, 22872, 24145, 25460, + 26812, 28198, 29612, 31052, 32513, 33991, 35482, 36983, + 38487, 39993, 41494, 42986, 44466, 45928, 47368, 48782, + 50165, 51513, 52821, 54086, 55302, 56466, 57575, 58624, + 59610, 60529, 61379, 62156, 62858, 63483, 64027, 64490, + 64870, 65165, 65375, 65498, 65535, 65484, 65347, 65124, + 64815, 64422, 63946, 63389, 62753, 62039, 61251, 60391 }; + int i; + float w; + const float fg = -.018 + .5 * Ratio; + const float omega = 2 * M_PI * fg; + phase -= 63; + for( i = 0; i < 64; i++) + { + w = scale * ( KaiserWindow7[i] * ( i + 1 )); + dst[i] = w * sinc( omega * (i+phase) ); + dst[127-i] = w * sinc( omega * (127-i+phase) ); + } } typedef struct{ - float scale; - int incr; + float scale; + int incr; } VarFilterMode; -static const VarFilterMode Up = { 0.0211952, 0 }; -static const VarFilterMode Down = { 0.0364733, 2 }; - +const VarFilterMode Up = { 0.0211952, -1 }; +const VarFilterMode Down = { 0.0364733, 1 }; static void setupVarFilter( VarFilter* filter, - float Ratio, VarFilterMode Direction ) + float Ratio, VarFilterMode Direction ) { - int i,n,d; - Fraction IRatio; - float phase; - IRatio = findFraction( Ratio ); + int i,n,d; + Fraction IRatio; + float phase; + IRatio = findFraction( Ratio ); + Ratio = min( Ratio, 1/Ratio ); + + n = IRatio.numerator; + d = IRatio.denominator; + filter->pos_mod = d; - if ( (1/Ratio) < Ratio ) - Ratio = 1/Ratio; + for( i = 0; i < d; i++ ) + { + if( phase >= n ) + { + phase -= d; + filter->incr[i] = Direction.incr; + } + else + filter->incr[i] = 1+Direction.incr; - n = IRatio.numerator; - d = IRatio.denominator; - filter->pos_mod = n; + calculateVarFilter( filter->c[i], Ratio, phase/(float)n, + Direction.scale ); + phase += d; + } +} + +/*-------------------------------------------------------------------------*/ +static void createRateConverter( Sound_AudioCVT *Data, int* fip, + int SrcRate, int DestRate, int Channel ) +{ + int filter_index = *fip; + + int VarPos = 0; + int Mono = 2 - Channel; + float Ratio = DestRate; + *fip = -1; + - for( i = 0; i < d; i++ ) - { - if( phase >= n ) - { - phase -= d; - filter->incr[i] = Direction.incr; - } - else - filter->incr[i] = 1; + if( SrcRate < 1 || SrcRate > 1<<18 || + DestRate < 1 || DestRate > 1<<18 ) return; + Ratio /= SrcRate; + + if( Ratio > 1.) + VarPos = filter_index++; + else + Data->adapter[filter_index++] = minus5dB; + + while( Ratio > 64./31.) + { + Data->adapter[filter_index++] = + Mono ? doubleRateMono : doubleRateStereo; + Ratio /= 2.; + Data->len_mult *= 2; + Data->add *= 2; + Data->add += _fsize; + } - calculateVarFilter( filter->c[i], Ratio, phase/(float)n, - Direction.scale ); - phase += d; - } + while( Ratio < 31./64. ) + { + Data->adapter[filter_index++] = + Mono ? halfRateMono : halfRateStereo; + Ratio *= 2; + } + + if( Ratio > 1. ) + { + setupVarFilter( &Data->filter, Ratio, Up ); + Data->adapter[VarPos] = + Mono ? varRateUpMono : varRateUpStereo; + Data->len_mult *= 2; + Data->add *= 2; + Data->add += _fsize; + } + else + { + setupVarFilter( &Data->filter, Ratio, Down ); + Data->adapter[filter_index++] = + Mono ? varRateDownMono : varRateDownStereo; + } + *fip = filter_index; } -static int createRateConverter( Sound_AudioCVT *Data, int filter_index, - int SrcRate, int DestRate, int Channel ) +/*-------------------------------------------------------------------------*/ +static void createFormatConverter16Bit(Sound_AudioCVT *Data, int* fip, + SDL_AudioSpec src, SDL_AudioSpec dst ) { - int VarPos = 0; - int Mono = 2 - Channel; - float Ratio = DestRate; - if( SrcRate < 1 || SrcRate > 1<<18 || - DestRate < 1 || DestRate > 1<<18 ) return -1; - if( SrcRate == DestRate ) return filter_index; - Ratio /= SrcRate; + int filter_index = *fip; + + if( src.channels == 2 && dst.channels == 1 ) + { + Data->add /= 2; + Data->len_mult /= 2; - if( Ratio > 1.0) - VarPos = filter_index++; - else - { - fprintf (stderr, "Filter: minus5dB\n"); - Data->adapter[filter_index++] = minus5dB; - } + if( !IS_SYSENDIAN(src) ) + Data->adapter[filter_index++] = swapBytes; + + if( IS_SIGNED(src) ) + Data->adapter[filter_index++] = convertStereoToMonoS16Bit; + else + Data->adapter[filter_index++] = convertStereoToMonoU16Bit; - while( Ratio > 64.0/31.0) - { - fprintf (stderr, "Filter: %s\n", - Mono ? "doubleRateMono" : "doubleRateStereo"); - Data->adapter[filter_index++] = - Mono ? doubleRateMono : doubleRateStereo; - Ratio /= 2; - Data->len_mult *= 2; - Data->add *= 2; - Data->add += _fsize; - } - - while( Ratio < 31.0/64.0 ) - { - fprintf (stderr, "Filter: %s\n", - Mono ? "halfRateMono" : "halfRateStereo"); - Data->adapter[filter_index++] = - Mono ? halfRateMono : halfRateStereo; - Ratio *= 2; - } + if( !IS_SYSENDIAN(dst) ) + Data->adapter[filter_index++] = swapBytes; + } + else if( IS_SYSENDIAN(src) != IS_SYSENDIAN(dst) ) + Data->adapter[filter_index++] = swapBytes; - if( Ratio > 1.0 ) - { - fprintf (stderr, "Filter: %s\n", - Mono ? "varRateMono" : "varRateStereo"); - setupVarFilter( &Data->filter, Ratio, Up ); - Data->adapter[VarPos] = - Mono ? varRateMono : varRateStereo; - Data->len_mult *= 2; - Data->add *= 2; - Data->add += _fsize; - } - else - { - fprintf (stderr, "Filter: %s\n", - Mono ? "varRateMono" : "varRateStereo"); - setupVarFilter( &Data->filter, Ratio, Down ); - Data->adapter[filter_index++] = - Mono ? varRateMono : varRateStereo; - } - return filter_index; + if( IS_SIGNED(src) != IS_SIGNED(dst) ) + { + if( IS_SYSENDIAN(dst) ) + Data->adapter[filter_index++] = changeSigned16BitSys; + else + Data->adapter[filter_index++] = changeSigned16BitSys; + } + + if( src.channels == 1 && dst.channels == 2 ) + { + Data->add *= 2; + Data->len_mult *= 2; + Data->adapter[filter_index++] = convertMonoToStereo16Bit; + } + + *fip = filter_index; } -static int BuildAudioCVT(Sound_AudioCVT *Data, - Uint16 src_format, Uint8 src_channels, int src_rate, - Uint16 dst_format, Uint8 dst_channels, int dst_rate) +/*-------------------------------------------------------------------------*/ +static void createFormatConverter8Bit(Sound_AudioCVT *Data, int *fip, + SDL_AudioSpec src, SDL_AudioSpec dst ) { - int filter_index = 0; + int filter_index = *fip; + if( IS_16BIT(src) ) + { + Data->add /= 2; + Data->len_mult /= 2; - if( Data == NULL ) return -1; - Data->len_mult = 1.0; - Data->add = 0; + if( IS_SYSENDIAN(src) ) + Data->adapter[filter_index++] = cut16BitSysTo8Bit; + else + Data->adapter[filter_index++] = cut16BitWrongTo8Bit; + } - /* Check channels */ - if( src_channels < 1 || src_channels > 2 || - dst_channels < 1 || dst_channels > 2 ) goto error_exit; + if( src.channels == 2 && dst.channels == 1 ) + { + Data->add /= 2; + Data->len_mult /= 2; + + if( IS_SIGNED(src) ) + Data->adapter[filter_index++] = convertStereoToMonoS8Bit; + else + Data->adapter[filter_index++] = convertStereoToMonoU8Bit; + } - /* First filter: Size/Endian conversion */ - switch( src_format & AUDIO_FORMAT) - { - case AUDIO_8: - fprintf (stderr, "Filter: expand8BitTo16Bit\n"); - Data->adapter[filter_index++] = expand8BitTo16Bit; - Data->len_mult *= 2; - break; - case AUDIO_16WRONG: - fprintf (stderr, "Filter: swapBytes\n"); - Data->adapter[filter_index++] = swapBytes; - break; - } + if( IS_SIGNED(src) != IS_SIGNED(dst) ) + Data->adapter[filter_index++] = changeSigned8Bit; + + if( src.channels == 1 && dst.channels == 2 ) + { + Data->add *= 2; + Data->len_mult *= 2; + Data->adapter[filter_index++] = convertMonoToStereo8Bit; + } - /* Second adapter: Sign conversion -- unsigned/signed */ - if( src_format & AUDIO_SIGN ) - { - fprintf (stderr, "Filter: changeSigned\n"); - Data->adapter[filter_index++] = changeSigned; - } + if( !IS_8BIT(dst) ) + { + Data->add *= 2; + Data->len_mult *= 2; + if( IS_SYSENDIAN(dst) ) + Data->adapter[filter_index++] = expand8BitTo16BitSys; + else + Data->adapter[filter_index++] = expand8BitTo16BitWrong; + } - /* Third adapter: Stereo->Mono conversion */ - if( src_channels == 2 && dst_channels == 1 ) - { - fprintf (stderr, "convertStereoToMono\n"); - Data->adapter[filter_index++] = convertStereoToMono; - } + *fip = filter_index; +} + +/*-------------------------------------------------------------------------*/ +static void createFormatConverter(Sound_AudioCVT *Data, int *fip, + SDL_AudioSpec src, SDL_AudioSpec dst ) +{ + int filter_index = *fip; - /* Do rate conversion */ - if( src_channels == 2 && dst_channels == 2 ) - filter_index = createRateConverter( Data, filter_index, - src_rate, dst_rate, 2 ); - else - filter_index = createRateConverter( Data, filter_index, - src_rate, dst_rate, 1 ); + if( IS_FLOAT(src) ) + { + Data->adapter[filter_index++] = cutFloatTo16Bit; + Data->len_mult /= 2; + Data->add /= 2; + } + + if( IS_8BIT(src) || IS_8BIT(dst) ) + createFormatConverter8Bit( Data, &filter_index, src, dst); + else + createFormatConverter16Bit( Data, &filter_index, src, dst); - if( filter_index < 0 ) goto error_exit; /* propagate error */ + if( IS_FLOAT(dst) ) + { + Data->adapter[filter_index++] = expand16BitToFloat; + Data->len_mult *= 2; + Data->add *= 2; + } + + *fip = filter_index; +} + - /* adapter: Mono->Stereo conversion */ - if( src_channels == 1 && dst_channels == 2 ){ - fprintf (stderr, "Filter: convertMonoToStereo\n"); - Data->adapter[filter_index++] = convertMonoToStereo; - Data->add *= 2; - Data->len_mult *= 2; - } +/*-------------------------------------------------------------------------*/ +DECLSPEC int BuildAudioCVT(Sound_AudioCVT *Data, + SDL_AudioSpec src, SDL_AudioSpec dst ) +{ + SDL_AudioSpec intrm; + int filter_index = 0; + + if( Data == NULL ) return -1; + Data->len_mult = 1.; + Data->add = 0; - /* adapter: final Sign conversion -- unsigned/signed */ - if( dst_format & AUDIO_SIGN ) - { - fprintf (stderr, "Filter: changeSigned\n"); - Data->adapter[filter_index++] = changeSigned; - } + /* Check channels */ + if( src.channels < 1 || src.channels > 2 || + dst.channels < 1 || dst.channels > 2 ) goto error_exit; + + /* If no frequency conversion is needed, go straight to dst format */ + if( src.freq == dst.freq ) + { + createFormatConverter( Data, &filter_index, src, dst ); + goto sucess_exit; + } + + /* Convert to signed 16Bit System-Endian */ + intrm.format = AUDIO_S16SYS; + intrm.channels = min( src.channels, dst.channels ); + createFormatConverter( Data, &filter_index, src, intrm ); - /* final adapter: Size/Endian conversion */ - switch( dst_format & AUDIO_FORMAT) - { - case AUDIO_8: - fprintf (stderr, "Filter: cut16BitTo8Bit\n"); - Data->adapter[filter_index++] = cut16BitTo8Bit; - break; - case AUDIO_16WRONG: - fprintf (stderr, "Filter: swapBytes\n"); - Data->adapter[filter_index++] = swapBytes; - break; - } - /* Set up the filter information */ - Data->adapter[filter_index] = NULL; - Data->needed = (filter_index > 0); - return 0; + /* Do rate conversion */ + if( src.channels == 2 && dst.channels == 2 ) + createRateConverter( Data, &filter_index, src.freq, dst.freq, 2 ); + else + createRateConverter( Data, &filter_index, src.freq, dst.freq, 1 ); + /* propagate error */ + if( filter_index < 0 ) goto error_exit; + + /* Convert to final format */ + createFormatConverter( Data, &filter_index, intrm, dst ); + + /* Set up the filter information */ +sucess_exit: +/* !!! FIXME: Is it okay to assign NULL to a function pointer? + Borland says no. -frank */ + Data->adapter[filter_index] = NULL; + return 0; error_exit: - Data->adapter[0] = NULL; - return -1; +/* !!! FIXME: Is it okay to assign NULL to a function pointer? + Borland says no. -frank */ + Data->adapter[0] = NULL; + return -1; } -/* - * Frank's audio converter has its own ideas about how to represent audio - * format, so at least for a transition period we use this to glue his code - * to our's. - * - * + The expand8BitTo16Bit filter will only convert to system byte order. - * + The cut16BitTo8Bit filter will only convert from system byte order. - * + The changeSigned filter only works on 16-bit samples, system byte order. - */ - +/*-------------------------------------------------------------------------*/ static char *fmt_to_str(Uint16 fmt) { switch (fmt) { - case AUDIO_U8: return " U8"; break; - case AUDIO_S8: return " S8"; break; - case AUDIO_U16MSB: return "U16MSB"; break; - case AUDIO_S16MSB: return "S16MSB"; break; - case AUDIO_U16LSB: return "U16LSB"; break; - case AUDIO_S16LSB: return "S16LSB"; break; + case AUDIO_U8: return " U8"; + case AUDIO_S8: return " S8"; + case AUDIO_U16MSB: return "U16MSB"; + case AUDIO_S16MSB: return "S16MSB"; + case AUDIO_U16LSB: return "U16LSB"; + case AUDIO_S16LSB: return "S16LSB"; } return "??????"; } -#define IS_8BIT(x) ((x) & 0x0008) -#define IS_16BIT(x) ((x) & 0x0010) -#define ENDIAN(x) ((x) & 0x1000) -#define SIGNED(x) ((x) & 0x8000) +#define AdapterDesc(x) { x, #x } + +static void show_AudioCVT( Sound_AudioCVT *Data ) +{ + int i,j; + const struct{ int (*adapter) ( AdapterC, int); Sint8 *name; } + AdapterDescription[] = { + AdapterDesc(expand8BitTo16BitSys), + AdapterDesc(expand8BitTo16BitWrong), + AdapterDesc(expand16BitToFloat), + AdapterDesc(swapBytes), + AdapterDesc(cut16BitSysTo8Bit), + AdapterDesc(cut16BitWrongTo8Bit), + AdapterDesc(cutFloatTo16Bit), + AdapterDesc(changeSigned16BitSys), + AdapterDesc(changeSigned16BitWrong), + AdapterDesc(changeSigned8Bit), + AdapterDesc(convertStereoToMonoS16Bit), + AdapterDesc(convertStereoToMonoU16Bit), + AdapterDesc(convertStereoToMonoS8Bit), + AdapterDesc(convertStereoToMonoU8Bit), + AdapterDesc(convertMonoToStereo16Bit), + AdapterDesc(convertMonoToStereo8Bit), + AdapterDesc(minus5dB), + AdapterDesc(doubleRateStereo), + AdapterDesc(doubleRateMono), + AdapterDesc(halfRateStereo), + AdapterDesc(halfRateMono), + AdapterDesc(varRateUpStereo), + AdapterDesc(varRateUpMono), + AdapterDesc(varRateDownStereo), + AdapterDesc(varRateDownMono), + { NULL, "----------NULL-----------" } + }; + const int AdapterDescMax = sizeof(AdapterDescription) + / sizeof(*AdapterDescription); + + fprintf( stderr, "\nAdapter List: \n" ); + for( i = 0; i < 32; i++ ) + { + for( j = 0; j < AdapterDescMax; j++ ) + { + if( Data->adapter[i] == AdapterDescription[j].adapter ) + { + fprintf( stderr, " %s\n", AdapterDescription[j].name ); + if( Data->adapter[i] == NULL ) return; + goto cont; + } + } + fprintf( stderr, " Error: unknown adapter\n" ); + cont: + } + fprintf( stderr, " Error: NULL adapter missing\n" ); +} + int Sound_BuildAudioCVT(Sound_AudioCVT *Data, - Uint16 src_in_format, Uint8 src_channels, int src_rate, - Uint16 dst_in_format, Uint8 dst_channels, int dst_rate) + Uint16 src_format, Uint8 src_channels, int src_rate, + Uint16 dst_format, Uint8 dst_channels, int dst_rate) { - Uint16 src_format = 0; - Uint16 dst_format = 0; + SDL_AudioSpec src, dst; + int ret; fprintf (stderr, + "Sound_BuildAudioCVT() :\n" + "-----------------------\n" "format: %s -> %s\n" "channels: %6d -> %6d\n" "rate: %6d -> %6d\n", - fmt_to_str (src_in_format), fmt_to_str (dst_in_format), + fmt_to_str (src_format), fmt_to_str (dst_format), src_channels, dst_channels, src_rate, dst_rate); - if ( IS_8BIT(src_in_format) && IS_16BIT(dst_in_format) ) - { - src_format |= AUDIO_8; - - /* - * Signedness and byte-order changes must wait until the data - * has been converted to 16-bit samples. - */ - if ( SIGNED(src_in_format) != SIGNED(dst_in_format) ) - { - dst_format |= AUDIO_SIGN; - } /* if */ - - if ( ENDIAN(dst_in_format) != ENDIAN(AUDIO_U16SYS) ) - { - dst_format |= AUDIO_16WRONG; - } /* if */ - } /* if */ - else if ( IS_16BIT(src_in_format) && IS_8BIT(dst_in_format) ) - { - dst_format |= AUDIO_8; + src.format = src_format; + src.channels = src_channels; + src.freq = src_rate; - /* - * Byte-order and signedness changes must be made before the data - * has been converted to 8-bit samples. - */ - if ( ENDIAN(src_in_format) != ENDIAN(AUDIO_U16SYS) ) - { - src_format |= AUDIO_16WRONG; - } /* if */ - - if ( SIGNED(src_in_format) != SIGNED(dst_in_format) ) - { - src_format |= AUDIO_SIGN; - } /* if */ - } /* else if */ - else if ( IS_16BIT(src_in_format) && IS_16BIT(dst_in_format) ) - { - if ( ENDIAN(src_in_format) != ENDIAN(dst_in_format) ) - { - if ( ENDIAN(src_in_format) == ENDIAN(AUDIO_U16SYS) ) - { - dst_format |= AUDIO_16WRONG; - - /* - * The data is already is system byte order, so any - * signedness change has to be made before changing byte - * order. - */ - if ( SIGNED(src_in_format) != SIGNED(dst_in_format) ) - { - src_format |= AUDIO_SIGN; - } /* if */ - } /* if */ - else - { - src_format |= AUDIO_16WRONG; + dst.format = dst_format; + dst.channels = dst_channels; + dst.freq = dst_rate; - /* - * The data is not in system byte order, so any signedness - * change has to be made after changing byte order. - */ - if ( SIGNED(src_in_format) != SIGNED(dst_in_format) ) - { - dst_format |= AUDIO_SIGN; - } /* if */ - } /* else */ - } /* if */ - else if ( ENDIAN(src_in_format) != SIGNED(AUDIO_U16SYS) ) - { - if ( SIGNED(src_in_format) != SIGNED(dst_in_format) ) - { - /* - * !!! FIXME !!! - * - * The changeSigned filter only works on system byte - * order. In this case, both source and destination is - * in opposite byte order, but the sign has to changed - * so we need to convert to system byte order, change - * sign, and then convert back to the original byte - * order again. This is not an optimal solution. - */ - src_format |= ( AUDIO_16WRONG | AUDIO_SIGN ); - dst_format |= AUDIO_16WRONG; - } /* if */ + ret = BuildAudioCVT( Data, src, dst ); - if ( src_rate != dst_rate ) - { - /* - * !!! FIXME !!! - * - * The audio conversion filter probably only works if the - * data is in system byte order. So we need to convert to - * system byte order, and then back to original byte - * order. This is not an optimal solution. - */ - src_format |= AUDIO_16WRONG; - dst_format |= AUDIO_16WRONG; - } - } /* else if */ - else if ( SIGNED(src_in_format) != SIGNED(dst_in_format) ) - { - src_format |= AUDIO_SIGN; - } /* else if */ - } /* else if */ - else if ( IS_8BIT(src_in_format) && IS_8BIT(dst_in_format) ) - { - /* - * !!! FIXME !!! - * - * The changeSigned filter only works on 16-bit samples, so if - * the signedness differs we have to convert from 8 to 16 bits, - * change the sign and then convert back to 8 bits again. This - * is not an optimal solution. - */ - if ( SIGNED(src_in_format) != SIGNED(dst_in_format) ) - { - src_format |= ( AUDIO_8 | AUDIO_SIGN ); - dst_format |= AUDIO_8; - } /* if */ + show_AudioCVT( Data ); + fprintf (stderr, "\n" + "return value: %d \n", ret ); - /* - * !!! FIXME !!! - * - * The convertMonoToStereo and convertStereoToMono filters only - * work with 16-bit samples. So if those are to be applied, we - * need to convert to 16-bit samples, and then back again. - */ - if ( src_channels != dst_channels ) - { - src_format |= AUDIO_8; - dst_format |= AUDIO_8; - } /* if */ - - /* - * !!! FIXME !!! - * - * The rate conversion filters almost certainly only work with - * 16-bit samples. Yadda, yadda, yadda. - */ - if ( src_rate != dst_rate ) - { - src_format |= AUDIO_8; - dst_format |= AUDIO_8; - } /* if */ - } /* else if */ - - return BuildAudioCVT(Data, src_format, src_channels, src_rate, - dst_format, dst_channels, dst_rate); + return ret; } - -/*-------------------------------------------------------------------------*/ -
--- a/alt_audio_convert.h Tue Jun 11 23:33:27 2002 +0000 +++ b/alt_audio_convert.h Wed Jun 12 08:35:23 2002 +0000 @@ -35,31 +35,42 @@ typedef struct{ - short c[16][2*_fsize]; + Sint16 c[16][2*_fsize]; char incr[16]; int pos_mod; } VarFilter; typedef struct{ - short* buffer; + Uint8* buffer; int mode; VarFilter *filter; } AdapterC; +/* +typedef struct{ + VarFilter filter; + double mult; // buffer must be len*buf_mult big + int add; + int (*adapter[32]) ( AdapterC Data, int length ); +} SDL_AudioC; + */ + typedef struct{ int needed; VarFilter filter; double len_mult; /* buffer must be len*len_mult big*/ Uint8* buf; int len; - int len_cvt; /* Length of converted audio buffer */ - int add; + int len_cvt; /* Length of converted audio buffer */ + int add; int (*adapter[32]) ( AdapterC Data, int length ); } Sound_AudioCVT; +#define SDL_AI_Loop 0x01 + extern DECLSPEC int Sound_ConvertAudio( Sound_AudioCVT *Data ); -extern DECLSPEC int Sound_BuildAudioConverter( Sound_AudioCVT *Data, +extern DECLSPEC int Sound_BuildAudioCVT( Sound_AudioCVT *Data, Uint16 src_format, Uint8 src_channels, int src_rate, Uint16 dst_format, Uint8 dst_channels, int dst_rate );