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Fix compiler issues

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This commit is contained in:
Ken Cooke 2015-11-18 10:06:40 -08:00
parent c61dad108c
commit 8ae3fa61c5
1 changed files with 30 additions and 30 deletions
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60
libraries/audio/src/AudioReverb.cpp
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@ -17,8 +17,8 @@
#include <intrin.h> #include <intrin.h>
inline static int MULHI(int a, int b) { inline static int MULHI(int a, int b) {
long long c = __emul(a, b); long long c = __emul(a, b);
return ((int*)&c)[1]; return ((int*)&c)[1];
} }
#else #else
@ -33,10 +33,10 @@ inline static int MULHI(int a, int b) {
#define MIN(a,b) (((a) < (b)) ? (a) : (b)) #define MIN(a,b) (((a) < (b)) ? (a) : (b))
#endif #endif
static const float M_PHI = 0.6180339887f; // maximum allpass diffusion static const float PHI = 0.6180339887f; // maximum allpass diffusion
static const double M_PI = 3.14159265358979323846; static const double PI = 3.14159265358979323846;
static const double M_SQRT2 = 1.41421356237309504880; static const double SQRT2 = 1.41421356237309504880;
static const double FIXQ31 = 2147483648.0; static const double FIXQ31 = 2147483648.0;
static const double FIXQ32 = 4294967296.0; static const double FIXQ32 = 4294967296.0;
@ -208,8 +208,8 @@ static void BQPeakBelowPi(double coef[5], double w0, double dbgain, double Q) {
G = MAX(G, 1.001); G = MAX(G, 1.001);
// compute the Nyquist gain // compute the Nyquist gain
wpi = w0 + 2.8 * (1.0 - w0/M_PI); // minimax-like error wpi = w0 + 2.8 * (1.0 - w0/PI); // minimax-like error
wpi = MIN(wpi, M_PI); wpi = MIN(wpi, PI);
G1 = analogPeak(w0, G, Q, wpi); G1 = analogPeak(w0, G, Q, wpi);
G1sq = G1 * G1; G1sq = G1 * G1;
@ -218,7 +218,7 @@ static void BQPeakBelowPi(double coef[5], double w0, double dbgain, double Q) {
// compute the analog half-gain frequency // compute the analog half-gain frequency
temp = G + 2.0 * Qsq - sqrt(Gsq + 4.0 * Qsq * G); temp = G + 2.0 * Qsq - sqrt(Gsq + 4.0 * Qsq * G);
wh = sqrt(temp) * w0 / (Q * M_SQRT2); wh = sqrt(temp) * w0 / (Q * SQRT2);
// prewarp freqs of w0 and wh // prewarp freqs of w0 and wh
w0 = tan(0.5 * w0); w0 = tan(0.5 * w0);
@ -305,8 +305,8 @@ static void BQPeakAbovePi(double coef[5], double w0, double dbgain, double Q) {
G = MAX(G, 1.001); G = MAX(G, 1.001);
// compute the Nyquist gain // compute the Nyquist gain
wpi = M_PI; wpi = PI;
if (w0 < M_PI) { if (w0 < PI) {
G1 = G; // use the peak gain G1 = G; // use the peak gain
} else { } else {
G1 = analogPeak(w0, G, Q, wpi); // use Nyquist gain of analog response G1 = analogPeak(w0, G, Q, wpi); // use Nyquist gain of analog response
@ -317,7 +317,7 @@ static void BQPeakAbovePi(double coef[5], double w0, double dbgain, double Q) {
// compute the analog half-gain frequency // compute the analog half-gain frequency
temp = G + 2.0 * Qsq - sqrt(Gsq + 4.0 * Qsq * G); temp = G + 2.0 * Qsq - sqrt(Gsq + 4.0 * Qsq * G);
wh = sqrt(temp) * w0 / (Q * M_SQRT2); wh = sqrt(temp) * w0 / (Q * SQRT2);
// approximate wn and wd // approximate wn and wd
// use half-gain frequency as mapping // use half-gain frequency as mapping
@ -425,8 +425,8 @@ static void BQShelf(double coef[5], double w0, double dbgain, double resonance,
G = MAX(G, 1.001); G = MAX(G, 1.001);
// compute the Nyquist gain // compute the Nyquist gain
wpi = w0 + 2.8 * (1.0 - w0/M_PI); // minimax-like error wpi = w0 + 2.8 * (1.0 - w0/PI); // minimax-like error
wpi = MIN(wpi, M_PI); wpi = MIN(wpi, PI);
G1 = analogShelf(w0, G, resonance, isHigh, wpi); G1 = analogShelf(w0, G, resonance, isHigh, wpi);
// approximate wn and wd // approximate wn and wd
@ -513,10 +513,10 @@ static void BQFilter(double coef[5], double w0, int isHigh) {
if (isHigh) { if (isHigh) {
w0 = MIN(w0, M_PI); // disallow w0 > pi for highpass w0 = MIN(w0, PI); // disallow w0 > pi for highpass
// compute the Nyquist gain // compute the Nyquist gain
wpi = M_PI; wpi = PI;
G1 = analogFilter(w0, isHigh, wpi); G1 = analogFilter(w0, isHigh, wpi);
// approximate wn and wd // approximate wn and wd
@ -528,13 +528,13 @@ static void BQFilter(double coef[5], double w0, int isHigh) {
// compute B and A // compute B and A
B = 0.0; B = 0.0;
A = M_SQRT2 * Wdsq / atan(wd); // Qd = sqrt(0.5) * atan(wd)/wd; A = SQRT2 * Wdsq / atan(wd); // Qd = sqrt(0.5) * atan(wd)/wd;
} else { } else {
// compute the Nyquist gain // compute the Nyquist gain
wpi = w0 + 2.8 * (1.0 - w0/M_PI); // minimax-like error wpi = w0 + 2.8 * (1.0 - w0/PI); // minimax-like error
wpi = MIN(wpi, M_PI); wpi = MIN(wpi, PI);
G1 = analogFilter(w0, isHigh, wpi); G1 = analogFilter(w0, isHigh, wpi);
// approximate wn and wd // approximate wn and wd
@ -653,7 +653,7 @@ public:
// lowpass filter, -3dB @ freq // lowpass filter, -3dB @ freq
double coef[5]; double coef[5];
BQFilter(coef, M_PI * freq / (0.5 * sampleRate), 0); BQFilter(coef, PI * freq / (0.5 * sampleRate), 0);
_b0 = (float)coef[0]; _b0 = (float)coef[0];
_b1 = (float)coef[1]; _b1 = (float)coef[1];
_b2 = (float)coef[2]; _b2 = (float)coef[2];
@ -661,7 +661,7 @@ public:
_a2 = (float)coef[4]; _a2 = (float)coef[4];
// DC-blocking filter, -3dB @ 10Hz // DC-blocking filter, -3dB @ 10Hz
_alpha = (float)(1.0 - exp(-M_PI * 10.0 / (0.5 * sampleRate))); _alpha = (float)(1.0 - exp(-PI * 10.0 / (0.5 * sampleRate)));
} }
void process(float input0, float input1, float& output0, float& output1) { void process(float input0, float input1, float& output0, float& output1) {
@ -809,9 +809,9 @@ public:
// amplitude slightly less than 1.0 // amplitude slightly less than 1.0
_y0 = (int32_t)(0.000 * FIXQ31); _y0 = (int32_t)(0.000 * FIXQ31);
_y1 = (int32_t)(0.999 * cos(M_PI * freq / sampleRate) * FIXQ31); _y1 = (int32_t)(0.999 * cos(PI * freq / sampleRate) * FIXQ31);
_k = (int32_t)(2.0 * sin(M_PI * freq / sampleRate) * FIXQ32); _k = (int32_t)(2.0 * sin(PI * freq / sampleRate) * FIXQ32);
} }
void setGain(int32_t gain) { void setGain(int32_t gain) {
@ -895,7 +895,7 @@ public:
output = _output; output = _output;
// add modulation to delay // add modulation to delay
uint32_t offset = _delay + (mod >> MOD_FRACBITS); int32_t offset = _delay + (mod >> MOD_FRACBITS);
float frac = (mod & MOD_FRACMASK) * QMOD_TO_FLOAT; float frac = (mod & MOD_FRACMASK) * QMOD_TO_FLOAT;
// 3rd-order Lagrange interpolation // 3rd-order Lagrange interpolation
@ -985,8 +985,8 @@ public:
freq1 = MIN(MAX(freq1, 1.0f), 24000.0f); freq1 = MIN(MAX(freq1, 1.0f), 24000.0f);
double coefLo[3], coefHi[3]; double coefLo[3], coefHi[3];
PZShelf(coefLo, M_PI * freq0 / (0.5 * sampleRate), dBgain0, 0); // low shelf PZShelf(coefLo, PI * freq0 / (0.5 * sampleRate), dBgain0, 0); // low shelf
PZShelf(coefHi, M_PI * freq1 / (0.5 * sampleRate), dBgain1, 1); // high shelf PZShelf(coefHi, PI * freq1 / (0.5 * sampleRate), dBgain1, 1); // high shelf
// convolve into a single biquad // convolve into a single biquad
_b0 = (float)(coefLo[0] * coefHi[0]); _b0 = (float)(coefLo[0] * coefHi[0]);
@ -1374,7 +1374,7 @@ static const float diffusionCoefTable[][2] = {
60.0000f, 0.2344f, 60.0000f, 0.2344f,
73.3333f, 0.3125f, 73.3333f, 0.3125f,
93.3333f, 0.5000f, 93.3333f, 0.5000f,
100.0000f, M_PHI, 100.0000f, PHI,
}; };
static const float roomSizeTable[][2] = { static const float roomSizeTable[][2] = {
@ -1552,13 +1552,13 @@ void ReverbImpl::setParameters(ReverbParameters *p) {
_ap8.setCoef(lateDiffusionCoef); _ap8.setCoef(lateDiffusionCoef);
_ap9.setCoef(lateDiffusionCoef); _ap9.setCoef(lateDiffusionCoef);
_ap10.setCoef(M_PHI); _ap10.setCoef(PHI);
_ap11.setCoef(M_PHI); _ap11.setCoef(PHI);
_ap12.setCoef(lateDiffusionCoef); _ap12.setCoef(lateDiffusionCoef);
_ap13.setCoef(lateDiffusionCoef); _ap13.setCoef(lateDiffusionCoef);
_ap14.setCoef(M_PHI); _ap14.setCoef(PHI);
_ap15.setCoef(M_PHI); _ap15.setCoef(PHI);
_ap16.setCoef(lateDiffusionCoef); _ap16.setCoef(lateDiffusionCoef);
_ap17.setCoef(lateDiffusionCoef); _ap17.setCoef(lateDiffusionCoef);