# HG changeset patch # User Ryan C. Gordon # Date 1022059674 0 # Node ID fbbb1f25b944605eaad58fa02fadf57544546241 # Parent 3466dde3a846cf2e072ad92b71ebd715ceda65de Cleanups by Torbj�rn. diff -r 3466dde3a846 -r fbbb1f25b944 alt_audio_convert.c --- a/alt_audio_convert.c Wed May 22 09:27:12 2002 +0000 +++ b/alt_audio_convert.c Wed May 22 09:27:54 2002 +0000 @@ -58,7 +58,7 @@ /* !!! FIXME: Lose all the "short" vars for "Sint16", etc. */ /*-------------------------------------------------------------------------*/ -int DECLSPEC Sound_ConvertAudio( Sound_AudioCVT *Data ) +int Sound_ConvertAudio( Sound_AudioCVT *Data ) { AdapterC Temp; int i; @@ -91,7 +91,7 @@ } /*-------------------------------------------------------------------------*/ -int expand8BitTo16Bit( AdapterC Data, int length ) +static int expand8BitTo16Bit( AdapterC Data, int length ) { int i; char* inp = (char*)Data.buffer-1; @@ -102,7 +102,7 @@ } /*-------------------------------------------------------------------------*/ -int swapBytes( AdapterC Data, int length ) +static int swapBytes( AdapterC Data, int length ) { int i; unsigned short a,b; @@ -118,7 +118,7 @@ } /*-------------------------------------------------------------------------*/ -int cut16BitTo8Bit( AdapterC Data, int length ) +static int cut16BitTo8Bit( AdapterC Data, int length ) { int i; short* inp = Data.buffer-1; @@ -129,7 +129,7 @@ } /*-------------------------------------------------------------------------*/ -int changeSigned( AdapterC Data, int length ) +static int changeSigned( AdapterC Data, int length ) { int i; short* buffer = Data.buffer; @@ -139,7 +139,7 @@ } /*-------------------------------------------------------------------------*/ -int convertStereoToMono( AdapterC Data, int length ) +static int convertStereoToMono( AdapterC Data, int length ) { int i; short* buffer = Data.buffer; @@ -154,7 +154,7 @@ } /*-------------------------------------------------------------------------*/ -int convertMonoToStereo( AdapterC Data, int length ) +static int convertMonoToStereo( AdapterC Data, int length ) { int i; short* buffer = Data.buffer-2; @@ -170,7 +170,7 @@ } /*-------------------------------------------------------------------------*/ -int minus5dB( AdapterC Data, int length ) +static int minus5dB( AdapterC Data, int length ) { int i; short* buffer = Data.buffer; @@ -180,37 +180,37 @@ } /*-------------------------------------------------------------------------*/ -int doubleRateStereo( AdapterC Data, int 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 ); } -int doubleRateMono( AdapterC Data, int length ) +static int doubleRateMono( AdapterC Data, int length ) { return _doubleRate1( Data.buffer, Data.mode, length ); } /*-------------------------------------------------------------------------*/ -int halfRateStereo( AdapterC Data, int length ) +static int halfRateStereo( AdapterC Data, int length ) { _halfRate2( Data.buffer, Data.mode, length/2 ); return 2*_halfRate2( Data.buffer+1, Data.mode, length/2 ); } -int halfRateMono( AdapterC Data, int length ) +static int halfRateMono( AdapterC Data, int length ) { return _halfRate2( Data.buffer, Data.mode, length ); } /*-------------------------------------------------------------------------*/ -int varRateStereo( AdapterC Data, int length ) +static int varRateStereo( 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 ); } -int varRateMono( AdapterC Data, int length ) +static int varRateMono( AdapterC Data, int length ) { return _varRate1( Data.buffer, Data.mode, Data.filter, length ); } @@ -222,7 +222,7 @@ } Fraction; /*-------------------------------------------------------------------------*/ -Fraction findFraction( float Value ) +static Fraction findFraction( float Value ) { /* gives a maximal error of 3% and typical less than 0.2% */ const char frac[96]={ @@ -266,13 +266,14 @@ } -float sinc( float x ) +static float sinc( float x ) { if( x > -1e-24 && x < 1e-24 ) return 1.; else return sin(x)/x; } -void calculateVarFilter( short* dst, float Ratio, float phase, float scale ) +static void calculateVarFilter( short* dst, float Ratio, float phase, + float scale ) { const unsigned short KaiserWindow7[]= { 22930, 16292, 14648, 14288, 14470, 14945, 15608, 16404, @@ -301,12 +302,12 @@ int incr; } VarFilterMode; -const VarFilterMode Up = { 0.0211952, 0 }; -const VarFilterMode Down = { 0.0364733, 2 }; +static const VarFilterMode Up = { 0.0211952, 0 }; +static const VarFilterMode Down = { 0.0364733, 2 }; -void setupVarFilter( VarFilter* filter, - float Ratio, VarFilterMode Direction ) +static void setupVarFilter( VarFilter* filter, + float Ratio, VarFilterMode Direction ) { int i,n,d; Fraction IRatio; @@ -336,8 +337,8 @@ } } -int createRateConverter( Sound_AudioCVT *Data, int filter_index, - int SrcRate, int DestRate, int Channel ) +static int createRateConverter( Sound_AudioCVT *Data, int filter_index, + int SrcRate, int DestRate, int Channel ) { int VarPos = 0; int Mono = 2 - Channel; @@ -387,7 +388,7 @@ return 0; } -int DECLSPEC Sound_BuildAudioCVT(Sound_AudioCVT *Data, +static int BuildAudioCVT(Sound_AudioCVT *Data, Uint16 src_format, Uint8 src_channels, int src_rate, Uint16 dst_format, Uint8 dst_channels, int dst_rate) { @@ -405,20 +406,29 @@ 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; } /* Second adapter: Sign conversion -- unsigned/signed */ if( src_format & AUDIO_SIGN ) + { + fprintf (stderr, "Filter: changeSigned\n"); Data->adapter[filter_index++] = changeSigned; + } /* Third adapter: Stereo->Mono conversion */ if( src_channels == 2 && dst_channels == 1 ) + { + fprintf (stderr, "convertStereoToMono\n"); Data->adapter[filter_index++] = convertStereoToMono; + } /* Do rate conversion */ if( src_channels == 2 && dst_channels == 2 ) @@ -432,6 +442,7 @@ /* 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; @@ -439,16 +450,22 @@ /* adapter: final Sign conversion -- unsigned/signed */ if( dst_format & AUDIO_SIGN ) + { + fprintf (stderr, "Filter: changeSigned\n"); Data->adapter[filter_index++] = changeSigned; + } /* 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; @@ -459,5 +476,187 @@ 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; + } + return "??????"; +} + +#define IS_8BIT(x) ((x) & 0x0008) +#define IS_16BIT(x) ((x) & 0x0010) +#define ENDIAN(x) ((x) & 0x1000) +#define SIGNED(x) ((x) & 0x8000) + +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 = 0; + Uint16 dst_format = 0; + + fprintf (stderr, + "format: %s -> %s\n" + "channels: %6d -> %6d\n" + "rate: %6d -> %6d\n", + fmt_to_str (src_in_format), fmt_to_str (dst_in_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; + + /* + * 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; + + /* + * 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 */ + } /* 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 */ + + /* + * !!! 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); +} + /*-------------------------------------------------------------------------*/ diff -r 3466dde3a846 -r fbbb1f25b944 audio_convert.c --- a/audio_convert.c Wed May 22 09:27:12 2002 +0000 +++ b/audio_convert.c Wed May 22 09:27:54 2002 +0000 @@ -45,7 +45,7 @@ * audio data. */ -void Sound_ConvertEndian(Sound_AudioCVT *cvt, Uint16 *format) +static void Sound_ConvertEndian(Sound_AudioCVT *cvt, Uint16 *format) { int i; Uint8 *data, tmp; @@ -71,7 +71,7 @@ * significant bit of each sample. */ -void Sound_ConvertSign(Sound_AudioCVT *cvt, Uint16 *format) +static void Sound_ConvertSign(Sound_AudioCVT *cvt, Uint16 *format) { int i; Uint8 *data; @@ -108,7 +108,7 @@ * of each 16-bit sample. */ -void Sound_Convert8(Sound_AudioCVT *cvt, Uint16 *format) +static void Sound_Convert8(Sound_AudioCVT *cvt, Uint16 *format) { int i; Uint8 *src, *dst; @@ -136,7 +136,7 @@ /* Convert 8-bit to 16-bit - LSB */ -void Sound_Convert16LSB(Sound_AudioCVT *cvt, Uint16 *format) +static void Sound_Convert16LSB(Sound_AudioCVT *cvt, Uint16 *format) { int i; Uint8 *src, *dst; @@ -161,7 +161,7 @@ /* Convert 8-bit to 16-bit - MSB */ -void Sound_Convert16MSB(Sound_AudioCVT *cvt, Uint16 *format) +static void Sound_Convert16MSB(Sound_AudioCVT *cvt, Uint16 *format) { int i; Uint8 *src, *dst; @@ -186,7 +186,7 @@ /* Duplicate a mono channel to both stereo channels */ -void Sound_ConvertStereo(Sound_AudioCVT *cvt, Uint16 *format) +static void Sound_ConvertStereo(Sound_AudioCVT *cvt, Uint16 *format) { int i; @@ -230,7 +230,7 @@ /* Effectively mix right and left channels into a single channel */ -void Sound_ConvertMono(Sound_AudioCVT *cvt, Uint16 *format) +static void Sound_ConvertMono(Sound_AudioCVT *cvt, Uint16 *format) { int i; Sint32 sample; @@ -386,7 +386,7 @@ /* Convert rate up by multiple of 2 */ -void Sound_RateMUL2(Sound_AudioCVT *cvt, Uint16 *format) +static void Sound_RateMUL2(Sound_AudioCVT *cvt, Uint16 *format) { int i; Uint8 *src, *dst; @@ -428,7 +428,7 @@ /* Convert rate down by multiple of 2 */ -void Sound_RateDIV2(Sound_AudioCVT *cvt, Uint16 *format) +static void Sound_RateDIV2(Sound_AudioCVT *cvt, Uint16 *format) { int i; Uint8 *src, *dst; @@ -467,7 +467,7 @@ /* Very slow rate conversion routine */ -void Sound_RateSLOW(Sound_AudioCVT *cvt, Uint16 *format) +static void Sound_RateSLOW(Sound_AudioCVT *cvt, Uint16 *format) { double ipos; int i, clen;