forked from len0rd/rockbox
duke3d: further optimize audio mixing
Rather than holding intermediate results as fixed-point, this converts them directly to normal integers (in the range of the PCM sample) while mixing, instead of waiting till the end to perform a separate shifting step. Also, this precalculates some constants in the reverb code. Change-Id: Ie04e444d145bc28ce67eef9ae0ead6d328acf28a
This commit is contained in:
parent
69ff35ac49
commit
cf07bb328d
5 changed files with 65 additions and 52 deletions
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@ -1368,9 +1368,13 @@ int VBE_setPalette(uint8_t *palettebuffer)
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memcpy(lastPalette, palettebuffer, 768);
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for (i = 0; i < 256; i++){
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sdlp->b = (Uint8) ((((float) *p++) / 63.0) * 255.0);
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sdlp->g = (Uint8) ((((float) *p++) / 63.0) * 255.0);
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sdlp->r = (Uint8) ((((float) *p++) / 63.0) * 255.0);
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/* doesn't map perfectly */
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sdlp->b = (Uint8) (*p << 2) | (*p >> 4);
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p++;
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sdlp->g = (Uint8) (*p << 2) | (*p >> 4);
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p++;
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sdlp->r = (Uint8) (*p << 2) | (*p >> 4);
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p++;
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sdlp->unused = *p++; /* This byte is unused in BUILD, too. */
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sdlp++;
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}
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@ -122,7 +122,7 @@ static int MV_FooMemory;
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static int MV_BufferDescriptor;
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static int MV_BufferEmpty[ NumberOfBuffers ];
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char *MV_MixBuffer[ NumberOfBuffers + 1 ];
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/* fixed-point */
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/* raw samples in range [-2^15, 2^15-1], interleaved stereo */
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long *MV_FooBuffer = NULL;
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static VoiceNode *MV_Voices = NULL;
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@ -465,8 +465,8 @@ void MV_ServiceVoc
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}
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{
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ClearBuffer_DW( MV_FooBuffer, 0, sizeof(long) / 4 * MV_BufferSize / MV_SampleSize * MV_Channels);
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MV_BufferEmpty[ MV_MixPage ] = TRUE;
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ClearBuffer_DW( MV_FooBuffer, 0, (sizeof(long) * MV_BufferSize / MV_SampleSize * MV_Channels) / 4);
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MV_BufferEmpty[ MV_MixPage ] = TRUE;
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}
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// Play any waiting voices
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@ -65,7 +65,7 @@ int MV_cubic(int position)
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}
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*/
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static int MV_cubic16(const short *src, int position, int rate)
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static inline int MV_cubic16(const short *src, int position, int rate)
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{
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int temp, hpos = position >> 16;
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@ -88,7 +88,7 @@ static int MV_cubic16(const short *src, int position, int rate)
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return do_cubic ? MV_cubic(position) : gval(3);
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}
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static int MV_cubic8to16(const unsigned char *src, int position, int rate)
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static inline int MV_cubic8to16(const unsigned char *src, int position, int rate)
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{
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int temp, hpos = position >> 16;
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@ -394,9 +394,11 @@ void MV_MixFPMono8( uint32_t position,
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for (i = 0; i < length; i++) {
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int s = MV_cubic8to16(src, position, rate) << FRACBITS;
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long out;
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int out;
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/* output is long in range [0, 2^16) */
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out = (s * MV_LeftScale) >> (FRACBITS * 2);
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out = (s * MV_LeftScale) >> FRACBITS;
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*dest += out;
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position += rate;
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@ -423,8 +425,8 @@ void MV_MixFPStereo8( uint32_t position,
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int s = MV_cubic8to16(src, position, rate) << FRACBITS;
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long left, right;
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left = (s * MV_LeftScale) >> FRACBITS;
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right = (s * MV_RightScale) >> FRACBITS;
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left = (s * MV_LeftScale) >> (FRACBITS * 2);
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right = (s * MV_RightScale) >> (FRACBITS * 2);
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dest[0] += left;
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dest[1] += right;
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@ -452,9 +454,9 @@ void MV_MixFPMono16( uint32_t position,
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for (i = 0; i < length; i++) {
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int s = MV_cubic16(src, position, rate) << FRACBITS;
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long out;
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int out;
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out = (s * MV_LeftScale) >> FRACBITS;
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out = (s * MV_LeftScale) >> (FRACBITS * 2);
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*dest += out;
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position += rate;
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@ -480,10 +482,10 @@ void MV_MixFPStereo16( uint32_t position,
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for (i = 0; i < length; i++) {
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int s = MV_cubic16(src, position, rate) << FRACBITS;
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long left, right;
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int left, right;
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left = (s * MV_LeftScale) >> FRACBITS;
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right = (s * MV_RightScale) >> FRACBITS;
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left = (s * MV_LeftScale) >> (FRACBITS * 2);
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right = (s * MV_RightScale) >> (FRACBITS * 2);
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dest[0] += left;
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dest[1] += right;
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@ -18,24 +18,40 @@ static int FB_SRC_A, FB_SRC_B, IIR_DEST_A0, IIR_DEST_A1, ACC_SRC_A0, ACC_SRC_A1,
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ACC_SRC_C1, ACC_SRC_D0, ACC_SRC_D1, IIR_SRC_B1, IIR_SRC_B0, MIX_DEST_A0,
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MIX_DEST_A1, MIX_DEST_B0, MIX_DEST_B1;
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static long IIR_ALPHA, ACC_COEF_A, ACC_COEF_B, ACC_COEF_C, ACC_COEF_D, IIR_COEF, FB_ALPHA, FB_X,
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IN_COEF_L, IN_COEF_R;
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//static long IIR_ALPHA, ACC_COEF_A, ACC_COEF_B, ACC_COEF_C, ACC_COEF_D, IIR_COEF, FB_ALPHA, FB_X,
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// IN_COEF_L, IN_COEF_R;
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static long iRVBLeft, iRVBRight;
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static int cnv_offset(int src)
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#define fp_scale ((double)(1<<FRACBITS))
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//static const double fp_scale = (double) (1 << FRACBITS);
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static const long IIR_ALPHA = 0.8701171875 * fp_scale,
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ACC_COEF_A = 0.622314453125 * fp_scale,
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ACC_COEF_B = -0.5244140625 * fp_scale,
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ACC_COEF_C = 0.53955078125 * fp_scale,
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ACC_COEF_D = -0.50830078125 * fp_scale,
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IIR_COEF = -0.69921875 * fp_scale,
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FB_ALPHA = 0.67578125 * fp_scale,
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FB_X = 0.646484375 * fp_scale,
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IN_COEF_L = -2. * fp_scale,
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IN_COEF_R = -2. * fp_scale;
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static long iRVBLeft, iRVBRight;
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static inline int cnv_offset(int src)
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{
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int64_t temp = ((int64_t)src * (int64_t)MV_MixRate) / 22050;
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return (int)temp;
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/* no need for 64-bit ints here */
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/* src can be no greater than 2^16-1, which allows sample rates up
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* to 65KHz */
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int temp = (src * MV_MixRate) / 22050;
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return temp;
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}
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// static char err[256];
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// extern __stdcall OutputDebugStringA(char *);
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static const double fp_scale = (double) (1 << FRACBITS);
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static void check_buffer()
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static inline void check_buffer()
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{
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int new_delay = cnv_offset(MV_ReverbDelay);
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@ -64,16 +80,6 @@ static void check_buffer()
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MIX_DEST_A1 = cnv_offset(0x238);
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MIX_DEST_B0 = cnv_offset(0x154);
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MIX_DEST_B1 = cnv_offset(0xAA);
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IIR_ALPHA = 0.8701171875 * fp_scale;
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ACC_COEF_A = 0.622314453125 * fp_scale;
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ACC_COEF_B = -0.5244140625 * fp_scale;
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ACC_COEF_C = 0.53955078125 * fp_scale;
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ACC_COEF_D = -0.50830078125 * fp_scale;
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IIR_COEF = -0.69921875 * fp_scale;
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FB_ALPHA = 0.67578125 * fp_scale;
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FB_X = 0.646484375 * fp_scale;
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IN_COEF_L = -2. * fp_scale;
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IN_COEF_R = -2. * fp_scale;
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if (reverbBuffer) reverbBuffer = (long*) realloc(reverbBuffer, new_delay * sizeof(long));
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else reverbBuffer = (long*) malloc(new_delay * sizeof(long));
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memset(reverbBuffer, 0, new_delay * sizeof(long));
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@ -83,7 +89,7 @@ static void check_buffer()
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}
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long g_buffer(int iOff, long *ptr) // get_buffer content helper: takes care about wraps
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static inline long g_buffer(int iOff, long *ptr) // get_buffer content helper: takes care about wraps
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{
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int correctDelay = delay;
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if(!correctDelay)
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@ -104,7 +110,7 @@ long g_buffer(int iOff, long *ptr) // get_buffer conten
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return (long)*(ptr+iOff);
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}
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void s_buffer(int iOff,long iVal, long *ptr) // set_buffer content helper: takes care about wraps and clipping
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static inline void s_buffer(int iOff,long iVal, long *ptr) // set_buffer content helper: takes care about wraps and clipping
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{
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int correctDelay = delay;
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if(!correctDelay)
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@ -125,7 +131,7 @@ void s_buffer(int iOff,long iVal, long *ptr) // set_buffer conten
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*(ptr+iOff)=iVal;
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}
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void s_buffer1(int iOff,long iVal, long *ptr) // set_buffer (+1 sample) content helper: takes care about wraps and clipping
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static inline void s_buffer1(int iOff,long iVal, long *ptr) // set_buffer (+1 sample) content helper: takes care about wraps and clipping
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{
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int correctDelay = delay;
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if(!correctDelay)
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@ -146,7 +152,7 @@ void s_buffer1(int iOff,long iVal, long *ptr) // set_buffer (+1 s
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*(ptr+iOff)=iVal;
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}
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long MixREVERBLeft(long INPUT_SAMPLE_L, long INPUT_SAMPLE_R, long *ptr)
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static inline long MixREVERBLeft(long INPUT_SAMPLE_L, long INPUT_SAMPLE_R, long *ptr)
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{
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long ACC0,ACC1,FB_A0,FB_A1,FB_B0,FB_B1;
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@ -186,8 +192,8 @@ long MixREVERBLeft(long INPUT_SAMPLE_L, long INPUT_SAMPLE_R, long *ptr)
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s_buffer(MIX_DEST_B0, fp_mul(FB_ALPHA , ACC0, FRACBITS) - fp_mul(FB_A0, (FB_ALPHA - one), FRACBITS) - fp_mul(FB_B0, FB_X, FRACBITS), ptr);
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s_buffer(MIX_DEST_B1, fp_mul(FB_ALPHA , ACC1, FRACBITS) - fp_mul(FB_A1, (FB_ALPHA - one), FRACBITS) - fp_mul(FB_B1, FB_X, FRACBITS), ptr);
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iRVBLeft = fp_div((g_buffer(MIX_DEST_A0, ptr)+g_buffer(MIX_DEST_B0, ptr)), 3 << FRACBITS, FRACBITS);
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iRVBRight = fp_div((g_buffer(MIX_DEST_A1, ptr)+g_buffer(MIX_DEST_B1, ptr)), 3 << FRACBITS, FRACBITS);
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iRVBLeft = (g_buffer(MIX_DEST_A0, ptr)+g_buffer(MIX_DEST_B0, ptr)) / 3;
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iRVBRight = (g_buffer(MIX_DEST_A1, ptr)+g_buffer(MIX_DEST_B1, ptr)) / 3;
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CurrAddr++;
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if(CurrAddr>delay-1) CurrAddr=0;
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@ -195,7 +201,7 @@ long MixREVERBLeft(long INPUT_SAMPLE_L, long INPUT_SAMPLE_R, long *ptr)
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return (long)iRVBLeft;
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}
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long MixREVERBRight(void)
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static inline long MixREVERBRight(void)
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{
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return (long)iRVBRight;
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}
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@ -227,12 +233,12 @@ void MV_FPReverb(int volume)
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{
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for (i = 0; i < count; i++)
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{
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long temp = MV_FooBuffer[i];
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long temp = MV_FooBuffer[i] << FRACBITS;
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/* evaluation order matters */
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long total = MixREVERBLeft(temp, temp, reverbBuffer);
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total += MixREVERBRight();
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total /= 2;
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MV_FooBuffer[i] += (scale * total) >> FRACBITS;
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MV_FooBuffer[i] += (scale * total) >> (FRACBITS * 2);
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}
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}
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else
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count /= 2;
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for (i = 0; i < count; i++)
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{
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long left = MV_FooBuffer[i*2];
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long right = MV_FooBuffer[i*2+1];
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long left = MV_FooBuffer[i*2] << FRACBITS;
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long right = MV_FooBuffer[i*2+1] << FRACBITS;
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left += (scale * MixREVERBLeft(left, right, reverbBuffer)) >> FRACBITS;
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right += (scale * MixREVERBRight()) >> FRACBITS;
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MV_FooBuffer[i*2] = left;
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MV_FooBuffer[i*2+1] = right;
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MV_FooBuffer[i*2] = left >> FRACBITS;
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MV_FooBuffer[i*2+1] = right >> FRACBITS;
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}
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}
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@ -275,7 +281,7 @@ void MV_16BitDownmix(char *dest, int count)
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for (i = 0; i < count; i++)
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{
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int out = MV_FooBuffer[i] >> FRACBITS;
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int out = MV_FooBuffer[i];
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if (out < -32768) pdest[i] = -32768;
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else if (out > 32767) pdest[i] = 32767;
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else pdest[i] = out;
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for (i = 0; i < count; i++)
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{
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int out = MV_FooBuffer[i] >> FRACBITS;
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int out = MV_FooBuffer[i] >> 8;
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if (out < -128) dest[i] = 0;
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else if (out > 127) dest[i] = 255;
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else dest[i] = out + 0x80;
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@ -728,6 +728,7 @@ void CONFIG_ReadSetup( void )
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NumBits = 16;
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SCRIPT_GetNumber( scripthandle, "Sound Setup", "MixRate",&MixRate);
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MixRate = RB_SAMPR;
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printf("MixRate = %d Hz", MixRate);
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SCRIPT_GetNumber( scripthandle, "Sound Setup", "MidiPort",&MidiPort);
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SCRIPT_GetNumber( scripthandle, "Sound Setup", "BlasterAddress",&dummy);
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BlasterConfig.Address = dummy;
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