ref: 9e6e95166ac606b05631c0ebacd81e93038f073e
dir: /src/ft2_mix_macros.h/
#pragma once
#include "ft2_header.h"
/* ----------------------------------------------------------------------- */
/* GENERAL MIXER MACROS */
/* ----------------------------------------------------------------------- */
#define GET_VOL \
CDA_LVol = v->SLVol2; \
CDA_RVol = v->SRVol2; \
#define SET_VOL_BACK \
v->SLVol2 = CDA_LVol; \
v->SRVol2 = CDA_RVol; \
#define GET_MIXER_VARS \
const uint32_t SFrq = v->SFrq; \
const uint32_t SFrqRev = v->SFrqRev; \
audioMixL = audio.mixBufferL; \
audioMixR = audio.mixBufferR; \
const bool mixInMono = (CDA_LVol == CDA_RVol); \
realPos = v->SPos; \
pos = v->SPosDec; \
#define GET_MIXER_VARS_RAMP \
const uint32_t SFrq = v->SFrq; \
const uint32_t SFrqRev = v->SFrqRev; \
audioMixL = audio.mixBufferL; \
audioMixR = audio.mixBufferR; \
CDA_LVolIP = v->SLVolIP; \
CDA_RVolIP = v->SRVolIP; \
const bool mixInMono = (v->SLVol2 == v->SRVol2) && (CDA_LVolIP == CDA_RVolIP); \
realPos = v->SPos; \
pos = v->SPosDec; \
#define SET_BASE8 \
CDA_LinearAdr = v->SBase8; \
smpPtr = CDA_LinearAdr + realPos; \
#define SET_BASE16 \
CDA_LinearAdr = v->SBase16; \
smpPtr = CDA_LinearAdr + realPos; \
#define SET_BASE8_BIDI \
CDA_LinearAdr = v->SBase8; \
CDA_LinAdrRev = v->SRevBase8; \
#define SET_BASE16_BIDI \
CDA_LinearAdr = v->SBase16; \
CDA_LinAdrRev = v->SRevBase16; \
#define INC_POS \
pos += SFrq; \
smpPtr += pos >> 16; \
pos &= 0xFFFF; \
#define INC_POS_BIDI \
pos += CDA_IPValL; \
smpPtr += pos >> 16; \
smpPtr += CDA_IPValH; \
pos &= 0xFFFF; \
#define SET_BACK_MIXER_POS \
v->SPosDec = pos; \
v->SPos = realPos; \
/* ----------------------------------------------------------------------- */
/* SAMPLE RENDERING MACROS */
/* ----------------------------------------------------------------------- */
#define VOLUME_RAMPING \
CDA_LVol += CDA_LVolIP; \
CDA_RVol += CDA_RVolIP; \
// all the 64-bit MULs here convert to fast logic on most 32-bit CPUs
#define RENDER_8BIT_SMP \
assert(smpPtr >= CDA_LinearAdr && smpPtr < CDA_LinearAdr+v->SLen); \
sample = *smpPtr << 20; \
*audioMixL++ += ((int64_t)sample * CDA_LVol) >> 32; \
*audioMixR++ += ((int64_t)sample * CDA_RVol) >> 32; \
#define RENDER_8BIT_SMP_MONO \
assert(smpPtr >= CDA_LinearAdr && smpPtr < CDA_LinearAdr+v->SLen); \
sample = *smpPtr << 20; \
sample = ((int64_t)sample * CDA_LVol) >> 32; \
*audioMixL++ += sample; \
*audioMixR++ += sample; \
#define RENDER_16BIT_SMP \
assert(smpPtr >= CDA_LinearAdr && smpPtr < CDA_LinearAdr+v->SLen); \
sample = *smpPtr << 12; \
*audioMixL++ += ((int64_t)sample * CDA_LVol) >> 32; \
*audioMixR++ += ((int64_t)sample * CDA_RVol) >> 32; \
#define RENDER_16BIT_SMP_MONO \
assert(smpPtr >= CDA_LinearAdr && smpPtr < CDA_LinearAdr+v->SLen); \
sample = *smpPtr << 12; \
sample = ((int64_t)sample * CDA_LVol) >> 32; \
*audioMixL++ += sample; \
*audioMixR++ += sample; \
#ifndef LERPMIX
// 4-tap cubic spline interpolation (default - slower than linear, but better quality)
// in: int32_t s0,s1,s2,s3 = -128..127 | f = 0..65535 (frac) | out: 16-bit s0 (will exceed 16-bits because of overshoot)
#define INTERPOLATE8(s0, s1, s2, s3, f) \
{ \
const int16_t *t = cubicSplineTable + ((f >> CUBIC_FSHIFT) & CUBIC_FMASK); \
s0 = ((s0 * t[0]) + (s1 * t[1]) + (s2 * t[2]) + (s3 * t[3])) >> (CUBIC_QUANTSHIFT-8); \
} \
// in: int32_t s0,s1,s2,s3 = -32768..32767 | f = 0..65535 (frac) | out: 16-bit s0 (will exceed 16-bits because of overshoot)
#define INTERPOLATE16(s0, s1, s2, s3, f) \
{ \
const int16_t *t = cubicSplineTable + ((f >> CUBIC_FSHIFT) & CUBIC_FMASK); \
s0 = ((s0 * t[0]) + (s1 * t[1]) + (s2 * t[2]) + (s3 * t[3])) >> CUBIC_QUANTSHIFT; \
} \
/* 8bitbubsy: It may look like we are potentially going out of bounds by looking up sample point
** -1, 1 and 2, but the sample data is actually padded on both the left (negative) and right side,
** where correct samples are stored according to loop mode (or no loop).
**
** The only issue is that the -1 look-up gets wrong information if loopStart>0 on looped-samples,
** and the sample-position is at loopStart. The spline will get ever so slighty wrong because of this,
** but it's barely audible anyway. Doing it elsewise would require a refactoring of how the audio mixer
** works!
*/
#define RENDER_8BIT_SMP_INTRP \
assert(smpPtr >= CDA_LinearAdr && smpPtr < CDA_LinearAdr+v->SLen); \
sample = smpPtr[-1]; \
sample2 = smpPtr[0]; \
sample3 = smpPtr[1]; \
sample4 = smpPtr[2]; \
INTERPOLATE8(sample, sample2, sample3, sample4, pos) \
sample <<= 12; \
*audioMixL++ += ((int64_t)sample * CDA_LVol) >> 32; \
*audioMixR++ += ((int64_t)sample * CDA_RVol) >> 32; \
#define RENDER_8BIT_SMP_MONO_INTRP \
assert(smpPtr >= CDA_LinearAdr && smpPtr < CDA_LinearAdr+v->SLen); \
sample = smpPtr[-1]; \
sample2 = smpPtr[0]; \
sample3 = smpPtr[1]; \
sample4 = smpPtr[2]; \
INTERPOLATE8(sample, sample2, sample3, sample4, pos) \
sample <<= 12; \
sample = ((int64_t)sample * CDA_LVol) >> 32; \
*audioMixL++ += sample; \
*audioMixR++ += sample; \
#define RENDER_16BIT_SMP_INTRP \
assert(smpPtr >= CDA_LinearAdr && smpPtr < CDA_LinearAdr+v->SLen); \
sample = smpPtr[-1]; \
sample2 = smpPtr[0]; \
sample3 = smpPtr[1]; \
sample4 = smpPtr[2]; \
INTERPOLATE16(sample, sample2, sample3, sample4, pos) \
sample <<= 12; \
*audioMixL++ += ((int64_t)sample * CDA_LVol) >> 32; \
*audioMixR++ += ((int64_t)sample * CDA_RVol) >> 32; \
#define RENDER_16BIT_SMP_MONO_INTRP \
assert(smpPtr >= CDA_LinearAdr && smpPtr < CDA_LinearAdr+v->SLen); \
sample = smpPtr[-1]; \
sample2 = smpPtr[0]; \
sample3 = smpPtr[1]; \
sample4 = smpPtr[2]; \
INTERPOLATE16(sample, sample2, sample3, sample4, pos) \
sample <<= 12; \
sample = ((int64_t)sample * CDA_LVol) >> 32; \
*audioMixL++ += sample; \
*audioMixR++ += sample; \
#else
// 2-tap linear interpolation (like FT2 - faster, but bad quality)
// in: int32_t s1,s2 = -128..127 | f = 0..65535 (frac) | out: s1 = -32768..32767
#define INTERPOLATE8(s1, s2, f) \
s2 -= s1; \
s2 = (s2 * (int32_t)f) >> (16 - 8); \
s1 <<= 8; \
s1 += s2; \
// in: int32_t s1,s2 = -32768..32767 | f = 0..65535 (frac) | out: s1 = -32768..32767
#define INTERPOLATE16(s1, s2, f) \
s2 = (s2 - s1) >> 1; \
s2 = (s2 * (int32_t)f) >> (16 - 1); \
s1 += s2; \
/* 8bitbubsy: It may look like we are potentially going out of bounds by looking up sample point
** 1, but the sample data is actually padded on the right side, where a correct sample is stored
** according to loop mode (or no loop). This is how original FT2 (MS-DOS) does it as well.
*/
#define RENDER_8BIT_SMP_INTRP \
assert(smpPtr >= CDA_LinearAdr && smpPtr < CDA_LinearAdr+v->SLen); \
sample = smpPtr[0]; \
sample2 = smpPtr[1]; \
INTERPOLATE8(sample, sample2, pos) \
sample <<= 12; \
*audioMixL++ += ((int64_t)sample * CDA_LVol) >> 32; \
*audioMixR++ += ((int64_t)sample * CDA_RVol) >> 32; \
#define RENDER_8BIT_SMP_MONO_INTRP \
assert(smpPtr >= CDA_LinearAdr && smpPtr < CDA_LinearAdr+v->SLen); \
sample = smpPtr[0]; \
sample2 = smpPtr[1]; \
INTERPOLATE8(sample, sample2, pos) \
sample <<= 12; \
sample = ((int64_t)sample * CDA_LVol) >> 32; \
*audioMixL++ += sample; \
*audioMixR++ += sample; \
#define RENDER_16BIT_SMP_INTRP \
assert(smpPtr >= CDA_LinearAdr && smpPtr < CDA_LinearAdr+v->SLen); \
sample = smpPtr[0]; \
sample2 = smpPtr[1]; \
INTERPOLATE16(sample, sample2, pos) \
sample <<= 12; \
*audioMixL++ += ((int64_t)sample * CDA_LVol) >> 32; \
*audioMixR++ += ((int64_t)sample * CDA_RVol) >> 32; \
#define RENDER_16BIT_SMP_MONO_INTRP \
assert(smpPtr >= CDA_LinearAdr && smpPtr < CDA_LinearAdr+v->SLen); \
sample = smpPtr[0]; \
sample2 = smpPtr[1]; \
INTERPOLATE16(sample, sample2, pos) \
sample <<= 12; \
sample = ((int64_t)sample * CDA_LVol) >> 32; \
*audioMixL++ += sample; \
*audioMixR++ += sample; \
#endif
/* ----------------------------------------------------------------------- */
/* SAMPLES-TO-MIX LIMITING MACROS */
/* ----------------------------------------------------------------------- */
#define LIMIT_MIX_NUM \
i = (v->SLen - 1) - realPos; \
if (i > 65535) \
i = 65535; \
\
i = (i << 16) | (pos ^ 0xFFFF); \
samplesToMix = ((int64_t)i * SFrqRev) >> 32; \
samplesToMix++; \
\
if (samplesToMix > CDA_BytesLeft) \
samplesToMix = CDA_BytesLeft; \
#define LIMIT_MIX_NUM_RAMP \
if (v->SVolIPLen == 0) \
{ \
CDA_LVolIP = 0; \
CDA_RVolIP = 0; \
\
if (v->isFadeOutVoice) \
{ \
v->mixRoutine = NULL; /* fade out voice is done, shut it down */ \
return; \
} \
} \
else \
{ \
if (samplesToMix > v->SVolIPLen) \
samplesToMix = v->SVolIPLen; \
\
v->SVolIPLen -= samplesToMix; \
} \
#define HANDLE_SAMPLE_END \
realPos = (int32_t)(smpPtr - CDA_LinearAdr); \
if (realPos >= v->SLen) \
{ \
v->mixRoutine = NULL; \
return; \
} \
#define WRAP_LOOP \
realPos = (int32_t)(smpPtr - CDA_LinearAdr); \
while (realPos >= v->SLen) \
realPos -= v->SRepL; \
smpPtr = CDA_LinearAdr + realPos; \
#define WRAP_BIDI_LOOP \
while (realPos >= v->SLen) \
{ \
realPos -= v->SRepL; \
v->backwards ^= 1; \
} \
#define START_BIDI \
if (v->backwards) \
{ \
delta = 0 - SFrq; \
assert(realPos >= v->SRepS && realPos < v->SLen); \
realPos = ~realPos; \
smpPtr = CDA_LinAdrRev + realPos; \
pos ^= 0xFFFF; \
} \
else \
{ \
delta = SFrq; \
assert(realPos >= 0 && realPos < v->SLen); \
smpPtr = CDA_LinearAdr + realPos; \
} \
\
const int32_t CDA_IPValH = (int32_t)delta >> 16; \
const int32_t CDA_IPValL = delta & 0xFFFF; \
#define END_BIDI \
if (v->backwards) \
{ \
pos ^= 0xFFFF; \
realPos = ~(int32_t)(smpPtr - CDA_LinAdrRev); \
} \
else \
{ \
realPos = (int32_t)(smpPtr - CDA_LinearAdr); \
} \
\
/* ----------------------------------------------------------------------- */
/* VOLUME=0 OPTIMIZATION MACROS */
/* ----------------------------------------------------------------------- */
#define VOL0_OPTIMIZATION_NO_LOOP \
assert(numSamples <= 65536); \
\
pos = v->SPosDec + ((v->SFrq & 0xFFFF) * numSamples); \
realPos = v->SPos + ((v->SFrq >> 16) * numSamples) + (pos >> 16); \
\
if (realPos >= v->SLen) \
{ \
v->mixRoutine = NULL; /* shut down voice */ \
return; \
} \
\
pos &= 0xFFFF; \
SET_BACK_MIXER_POS
#define VOL0_OPTIMIZATION_LOOP \
assert(numSamples <= 65536); \
\
pos = v->SPosDec + ((v->SFrq & 0xFFFF) * numSamples); \
realPos = v->SPos + ((v->SFrq >> 16) * numSamples) + (pos >> 16); \
\
while (realPos >= v->SLen) \
realPos -= v->SRepL; \
\
pos &= 0xFFFF; \
SET_BACK_MIXER_POS
#define VOL0_OPTIMIZATION_BIDI_LOOP \
assert(numSamples <= 65536); \
\
pos = v->SPosDec + ((v->SFrq & 0xFFFF) * numSamples); \
realPos = v->SPos + ((v->SFrq >> 16) * numSamples) + (pos >> 16); \
\
while (realPos >= v->SLen) \
{ \
realPos -= v->SRepL; \
v->backwards ^= 1; \
} \
\
pos &= 0xFFFF; \
SET_BACK_MIXER_POS