Some filter additions, some SGTL5000 additionsdds
@@ -2204,7 +2204,28 @@ void AudioFilterBiquad::update(void) | |||
release(block); | |||
} | |||
void AudioFilterBiquad::updateCoefs(int *source, bool doReset) | |||
{ | |||
int32_t *dest=(int32_t *)definition; | |||
int32_t *src=(int32_t *)source; | |||
__disable_irq(); | |||
for(uint8_t index=0;index<5;index++) | |||
{ | |||
*dest++=*src++; | |||
} | |||
if(doReset) | |||
{ | |||
*dest++=0; | |||
*dest++=0; | |||
*dest++=0; | |||
} | |||
__enable_irq(); | |||
} | |||
void AudioFilterBiquad::updateCoefs(int *source) | |||
{ | |||
updateCoefs(source,false); | |||
} | |||
/******************************************************************/ | |||
@@ -2962,7 +2983,7 @@ bool AudioControlWM8731master::enable(void) | |||
#define DAP_BASS_ENHANCE_CTRL 0x0106 | |||
#define DAP_AUDIO_EQ 0x0108 | |||
#define DAP_SGTL_SURROUND 0x010A | |||
#define DAP_FILTER_COEF_ACCES 0x010C | |||
#define DAP_FILTER_COEF_ACCESS 0x010C | |||
#define DAP_COEF_WR_B0_MSB 0x010E | |||
#define DAP_COEF_WR_B0_LSB 0x0110 | |||
#define DAP_AUDIO_EQ_BASS_BAND0 0x0116 // 115 Hz | |||
@@ -3005,12 +3026,13 @@ bool AudioControlSGTL5000::enable(void) | |||
write(CHIP_ANA_POWER, 0x4060); // VDDD is externally driven with 1.8V | |||
write(CHIP_LINREG_CTRL, 0x006C); // VDDA & VDDIO both over 3.1V | |||
write(CHIP_REF_CTRL, 0x01F1); // VAG=1.575 slow ramp, normal bias current | |||
write(CHIP_LINE_OUT_CTRL, 0x0322); | |||
write(CHIP_LINE_OUT_CTRL, 0x0322); // LO_VAGCNTRL=1.65V, OUT_CURRENT=0.36mA | |||
write(CHIP_SHORT_CTRL, 0x4446); // allow up to 125mA | |||
write(CHIP_ANA_CTRL, 0x0137); // enable zero cross detectors | |||
write(CHIP_ANA_POWER, 0x40FF); // power up: lineout, hp, adc, dac | |||
write(CHIP_DIG_POWER, 0x0073); // power up all digital stuff | |||
delay(400); | |||
// 40*log((1.575)/(1.65)) + 15 = 13.1391993746043 but it seems wrong, 5 is better... | |||
write(CHIP_LINE_OUT_VOL, 0x0505); // TODO: correct value for 3.3V | |||
write(CHIP_CLK_CTRL, 0x0004); // 44.1 kHz, 256*Fs | |||
write(CHIP_I2S_CTRL, 0x0130); // SCLK=32*Fs, 16bit, I2S format | |||
@@ -3050,6 +3072,13 @@ bool AudioControlSGTL5000::write(unsigned int reg, unsigned int val) | |||
return false; | |||
} | |||
unsigned int AudioControlSGTL5000::modify(unsigned int reg, unsigned int val, unsigned int iMask) | |||
{ | |||
unsigned int val1 = (read(reg)&(~iMask))|val; | |||
if(!write(reg,val1)) return 0; | |||
return val1; | |||
} | |||
bool AudioControlSGTL5000::volumeInteger(unsigned int n) | |||
{ | |||
if (n == 0) { | |||
@@ -3069,6 +3098,207 @@ bool AudioControlSGTL5000::volumeInteger(unsigned int n) | |||
return write(CHIP_ANA_HP_CTRL, n); // set volume | |||
} | |||
bool AudioControlSGTL5000::volume(float left, float right) | |||
{ | |||
unsigned short m=(0x7F-calcVol(right,0x7F))<<8|0x7F-calcVol(left,0x7F); | |||
return write(CHIP_ANA_HP_CTRL, m); | |||
} | |||
// CHIP_LINE_OUT_VOL | |||
unsigned short AudioControlSGTL5000::lo_lvl(uint8_t n) | |||
{ | |||
n&=31; | |||
return modify(CHIP_LINE_OUT_VOL,(n<<8)|n,(31<<8)|31); | |||
} | |||
unsigned short AudioControlSGTL5000::lo_lvl(uint8_t left, uint8_t right) | |||
{ | |||
left&=31; | |||
right&=31; | |||
return modify(CHIP_LINE_OUT_VOL,(right<<8)|left,(31<<8)|31); | |||
} | |||
unsigned short AudioControlSGTL5000::dac_vol(float n) // set both directly | |||
{ | |||
if(read(CHIP_ADCDAC_CTRL)&(3<<2)!=((n>0 ? 0:3)<<2)) modify(CHIP_ADCDAC_CTRL,(n>0 ? 0:3)<<2,3<<2); | |||
unsigned char m=calcVol(n,0xC0); | |||
return modify(CHIP_DAC_VOL,((0xFC-m)<<8)|(0xFC-m),65535); | |||
} | |||
unsigned short AudioControlSGTL5000::dac_vol(float left, float right) | |||
{ | |||
unsigned short adcdac=((right>0 ? 0:2)|(left>0 ? 0:1))<<2; | |||
if(read(CHIP_ADCDAC_CTRL)&(3<<2)!=adcdac) modify(CHIP_ADCDAC_CTRL,adcdac,1<<2); | |||
unsigned short m=(0xFC-calcVol(right,0xC0))<<8|(0xFC-calcVol(left,0xC0)); | |||
return modify(CHIP_DAC_VOL,m,65535); | |||
} | |||
// DAP_CONTROL | |||
unsigned short AudioControlSGTL5000::dap_mix_enable(uint8_t n) | |||
{ | |||
return modify(DAP_CONTROL,(n&1)<<4,1<<4); | |||
} | |||
unsigned short AudioControlSGTL5000::dap_enable(uint8_t n) | |||
{ | |||
if(n) n=1; | |||
unsigned char DAC=1+(2*n); // I2S_IN if n==0 else DAP | |||
modify(DAP_CONTROL,n,1); | |||
return modify(CHIP_SSS_CTRL,(0<<6)|(DAC<<4),(3<<6)|(3<<4)); | |||
} | |||
unsigned short AudioControlSGTL5000::dap_enable(void) | |||
{ | |||
return dap_enable(1); | |||
} | |||
// DAP_PEQ | |||
unsigned short AudioControlSGTL5000::dap_peqs(uint8_t n) // valid to n&7, 0 thru 7 filters enabled. | |||
{ | |||
return modify(DAP_PEQ,(n&7),7); | |||
} | |||
// DAP_AUDIO_EQ | |||
unsigned short AudioControlSGTL5000::dap_audio_eq(uint8_t n) // 0=NONE, 1=PEQ (7 IIR Biquad filters), 2=TONE (tone), 3=GEQ (5 band EQ) | |||
{ | |||
return modify(DAP_AUDIO_EQ,n&3,3); | |||
} | |||
// DAP_AUDIO_EQ_BASS_BAND0 & DAP_AUDIO_EQ_BAND1 & DAP_AUDIO_EQ_BAND2 etc etc | |||
unsigned short AudioControlSGTL5000::dap_audio_eq_band(uint8_t bandNum, float n) // by signed percentage -100/+100; dap_audio_eq(3); | |||
{ // 0x00==-12dB, 0x2F==0dB, 0x5F==12dB | |||
n=((n/100)*48)+0.499; | |||
if(n<-47) n=-47; | |||
if(n>48) n=48; | |||
n+=47; | |||
return modify(DAP_AUDIO_EQ_BASS_BAND0+(bandNum*2),(unsigned int)n,127); | |||
} | |||
void AudioControlSGTL5000::dap_audio_eq_geq(float bass, float mid_bass, float midrange, float mid_treble, float treble) | |||
{ | |||
dap_audio_eq_band(0,bass); | |||
dap_audio_eq_band(1,mid_bass); | |||
dap_audio_eq_band(2,midrange); | |||
dap_audio_eq_band(3,mid_treble); | |||
dap_audio_eq_band(4,treble); | |||
} | |||
void AudioControlSGTL5000::dap_audio_eq_tone(float bass, float treble) // dap_audio_eq(2); | |||
{ | |||
dap_audio_eq_band(0,bass); | |||
dap_audio_eq_band(4,treble); | |||
} | |||
// SGTL5000 PEQ Coefficient loader | |||
void AudioControlSGTL5000::load_peq(uint8_t filterNum, int *filterParameters) | |||
{ | |||
// 1111 11111111 11111111 | |||
write(DAP_COEF_WR_B0_MSB,(*filterParameters>>4)&65535); | |||
write(DAP_COEF_WR_B0_LSB,(*filterParameters++)&15); | |||
write(DAP_COEF_WR_B1_MSB,(*filterParameters>>4)&65535); | |||
write(DAP_COEF_WR_B1_LSB,(*filterParameters++)&15); | |||
write(DAP_COEF_WR_B2_MSB,(*filterParameters>>4)&65535); | |||
write(DAP_COEF_WR_B2_LSB,(*filterParameters++)&15); | |||
write(DAP_COEF_WR_A1_MSB,(*filterParameters>>4)&65535); | |||
write(DAP_COEF_WR_A1_LSB,(*filterParameters++)&15); | |||
write(DAP_COEF_WR_A2_MSB,(*filterParameters>>4)&65535); | |||
write(DAP_COEF_WR_A2_LSB,(*filterParameters++)&15); | |||
write(DAP_FILTER_COEF_ACCESS,(uint16_t)0x100|filterNum); | |||
delay(10); // seems necessary, didn't work for 1ms. | |||
modify(DAP_FILTER_COEF_ACCESS,(uint16_t)filterNum,15); | |||
} | |||
unsigned char AudioControlSGTL5000::calcVol(float n, unsigned char range) | |||
{ | |||
n=(n*(((float)range)/100))+0.499; | |||
if ((unsigned char)n>range) n=range; | |||
return (unsigned char)n; | |||
} | |||
// if(SGTL5000_PEQ) quantization_unit=524288; if(AudioFilterBiquad) quantization_unit=2147483648; | |||
void calcBiquad(uint8_t filtertype, float fC, float dB_Gain, float Q, uint32_t quantization_unit, uint32_t fS, int *coef) | |||
{ | |||
// I used resources like http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt | |||
// to make this routine, I tested most of the filter types and they worked. Such filters have limits and | |||
// before calling this routine with varying values the end user should check that those values are limited | |||
// to valid results. | |||
float A; | |||
if(filtertype<FILTER_PARAEQ) A=pow(10,dB_Gain/20); else A=pow(10,dB_Gain/40); | |||
float W0 = 2*3.14159265358979323846*fC/fS; | |||
float cosw=cos(W0); | |||
float sinw=sin(W0); | |||
//float alpha = sinw*sinh((log(2)/2)*BW*W0/sinw); | |||
//float beta = sqrt(2*A); | |||
float alpha = sinw / (2 * Q); | |||
float beta = sqrt(A)/Q; | |||
float b0,b1,b2,a0,a1,a2; | |||
switch(filtertype) { | |||
case FILTER_LOPASS: | |||
b0 = (1.0F - cosw) * 0.5F; // =(1-COS($H$2))/2 | |||
b1 = 1.0F - cosw; | |||
b2 = (1.0F - cosw) * 0.5F; | |||
a0 = 1.0F + alpha; | |||
a1 = 2.0F * cosw; | |||
a2 = alpha - 1.0F; | |||
break; | |||
case FILTER_HIPASS: | |||
b0 = (1.0F + cosw) * 0.5F; | |||
b1 = -(cosw + 1.0F); | |||
b2 = (1.0F + cosw) * 0.5F; | |||
a0 = 1.0F + alpha; | |||
a1 = 2.0F * cosw; | |||
a2 = alpha - 1.0F; | |||
break; | |||
case FILTER_BANDPASS: | |||
b0 = alpha; | |||
b1 = 0.0F; | |||
b2 = -alpha; | |||
a0 = 1.0F + alpha; | |||
a1 = 2.0F * cosw; | |||
a2 = alpha - 1.0F; | |||
break; | |||
case FILTER_NOTCH: | |||
b0=1; | |||
b1=-2*cosw; | |||
b2=1; | |||
a0=1+alpha; | |||
a1=2*cosw; | |||
a2=-(1-alpha); | |||
break; | |||
case FILTER_PARAEQ: | |||
b0 = 1 + (alpha*A); | |||
b1 =-2 * cosw; | |||
b2 = 1 - (alpha*A); | |||
a0 = 1 + (alpha/A); | |||
a1 = 2 * cosw; | |||
a2 =-(1-(alpha/A)); | |||
break; | |||
case FILTER_LOSHELF: | |||
b0 = A * ((A+1.0F) - ((A-1.0F)*cosw) + (beta*sinw)); | |||
b1 = 2.0F * A * ((A-1.0F) - ((A+1.0F)*cosw)); | |||
b2 = A * ((A+1.0F) - ((A-1.0F)*cosw) - (beta*sinw)); | |||
a0 = (A+1.0F) + ((A-1.0F)*cosw) + (beta*sinw); | |||
a1 = 2.0F * ((A-1.0F) + ((A+1.0F)*cosw)); | |||
a2 = -((A+1.0F) + ((A-1.0F)*cosw) - (beta*sinw)); | |||
break; | |||
case FILTER_HISHELF: | |||
b0 = A * ((A+1.0F) + ((A-1.0F)*cosw) + (beta*sinw)); | |||
b1 = -2.0F * A * ((A-1.0F) + ((A+1.0F)*cosw)); | |||
b2 = A * ((A+1.0F) + ((A-1.0F)*cosw) - (beta*sinw)); | |||
a0 = (A+1.0F) - ((A-1.0F)*cosw) + (beta*sinw); | |||
a1 = -2.0F * ((A-1.0F) - ((A+1.0F)*cosw)); | |||
a2 = -((A+1.0F) - ((A-1.0F)*cosw) - (beta*sinw)); | |||
} | |||
a0=(a0*2)/(float)quantization_unit; // once here instead of five times there... | |||
b0/=a0; | |||
*coef++=(int)(b0+0.499); | |||
b1/=a0; | |||
*coef++=(int)(b1+0.499); | |||
b2/=a0; | |||
*coef++=(int)(b2+0.499); | |||
a1/=a0; | |||
*coef++=(int)(a1+0.499); | |||
a2/=a0; | |||
*coef++=(int)(a2+0.499); | |||
} |
@@ -14,7 +14,6 @@ | |||
#define AudioNoInterrupts() (NVIC_DISABLE_IRQ(IRQ_SOFTWARE)) | |||
#define AudioInterrupts() (NVIC_ENABLE_IRQ(IRQ_SOFTWARE)) | |||
// waveforms.c | |||
extern "C" { | |||
extern const int16_t AudioWaveformSine[257]; | |||
@@ -377,6 +376,9 @@ public: | |||
AudioFilterBiquad(int *parameters) | |||
: AudioStream(1, inputQueueArray), definition(parameters) { } | |||
virtual void update(void); | |||
void updateCoefs(int *source, bool doReset); | |||
void updateCoefs(int *source); | |||
private: | |||
int *definition; | |||
audio_block_t *inputQueueArray[1]; | |||
@@ -527,29 +529,43 @@ public: | |||
} | |||
//bool inputLinein(void) { return write(0x0024, ana_ctrl | (1<<2)); } | |||
//bool inputMic(void) { return write(0x002A, 0x0172) && write(0x0024, ana_ctrl & ~(1<<2)); } | |||
bool volume(float left, float right); | |||
unsigned short micGain(unsigned int n) { return modify(0x002A, n&3, 3); } | |||
unsigned short lo_lvl(uint8_t n); | |||
unsigned short lo_lvl(uint8_t left, uint8_t right); | |||
unsigned short dac_vol(float n); | |||
unsigned short dac_vol(float left, float right); | |||
unsigned short dap_mix_enable(uint8_t n); | |||
unsigned short dap_enable(uint8_t n); | |||
unsigned short dap_enable(void); | |||
unsigned short dap_peqs(uint8_t n); | |||
unsigned short dap_audio_eq(uint8_t n); | |||
unsigned short dap_audio_eq_band(uint8_t bandNum, float n); | |||
void dap_audio_eq_geq(float bass, float mid_bass, float midrange, float mid_treble, float treble); | |||
void dap_audio_eq_tone(float bass, float treble); | |||
void load_peq(uint8_t filterNum, int *filterParameters); | |||
protected: | |||
bool muted; | |||
bool volumeInteger(unsigned int n); // range: 0x00 to 0x80 | |||
uint16_t ana_ctrl; | |||
unsigned char calcVol(float n, unsigned char range); | |||
unsigned int read(unsigned int reg); | |||
bool write(unsigned int reg, unsigned int val); | |||
unsigned int modify(unsigned int reg, unsigned int val, unsigned int iMask); | |||
}; | |||
//For Filter Type: 0 = LPF, 1 = HPF, 2 = BPF, 3 = NOTCH, 4 = PeakingEQ, 5 = LowShelf, 6 = HighShelf | |||
#define FILTER_LOPASS 0 | |||
#define FILTER_HIPASS 1 | |||
#define FILTER_BANDPASS 2 | |||
#define FILTER_NOTCH 3 | |||
#define FILTER_PARAEQ 4 | |||
#define FILTER_LOSHELF 5 | |||
#define FILTER_HISHELF 6 | |||
void calcBiquad(uint8_t filtertype, float fC, float dB_Gain, float Q, uint32_t quantization_unit, uint32_t fS, int *coef); |
@@ -0,0 +1,78 @@ | |||
// Tone example using AudioFilterBiquad object and calcBiquad filter calculator routine. | |||
#include <Audio.h> | |||
#include <Wire.h> | |||
#include <SD.h> | |||
const int myInput = AUDIO_INPUT_LINEIN; | |||
// const int myInput = AUDIO_INPUT_MIC; | |||
int BassFilter_L[]={0,0,0,0,0,0,0,0}; | |||
int BassFilter_R[]={0,0,0,0,0,0,0,0}; | |||
int TrebFilter_L[]={0,0,0,0,0,0,0,0}; | |||
int TrebFilter_R[]={0,0,0,0,0,0,0,0}; | |||
int updateFilter[5]; | |||
AudioInputI2S audioInput; // audio shield: mic or line-in | |||
AudioFilterBiquad filterBass_L(BassFilter_L); | |||
AudioFilterBiquad filterBass_R(BassFilter_R); | |||
AudioFilterBiquad filterTreb_L(TrebFilter_L); | |||
AudioFilterBiquad filterTreb_R(TrebFilter_R); | |||
AudioOutputI2S audioOutput; // audio shield: headphones & line-out | |||
// Create Audio connections between the components | |||
// | |||
AudioConnection c1(audioInput,0,filterBass_L,0); | |||
AudioConnection c2(audioInput,1,filterBass_R,0); | |||
AudioConnection c3(filterBass_L,0,filterTreb_L,0); | |||
AudioConnection c4(filterBass_R,0,filterTreb_R,0); | |||
AudioConnection c5(filterTreb_L,0,audioOutput,0); | |||
AudioConnection c6(filterTreb_R,0,audioOutput,1); | |||
// Create an object to control the audio shield. | |||
// | |||
AudioControlSGTL5000 audioShield; | |||
void setup() { | |||
// Audio connections require memory to work. For more | |||
// detailed information, see the MemoryAndCpuUsage example | |||
AudioMemory(12); | |||
// Enable the audio shield, select the input and set the output volume. | |||
audioShield.enable(); | |||
audioShield.inputSelect(myInput); | |||
audioShield.volume(75); | |||
audioShield.unmuteLineout(); | |||
calcBiquad(FILTER_PARAEQ,110,0,0.2,2147483648,44100,updateFilter); | |||
filterBass_L.updateCoefs(updateFilter); | |||
filterBass_R.updateCoefs(updateFilter); | |||
calcBiquad(FILTER_PARAEQ,4400,0,0.167,2147483648,44100,updateFilter); | |||
filterTreb_L.updateCoefs(updateFilter); | |||
filterTreb_R.updateCoefs(updateFilter); | |||
} | |||
elapsedMillis chgMsec=0; | |||
float tone1=0; | |||
void loop() { | |||
// every 10 ms, check for adjustment the tone & vol | |||
if (chgMsec > 10) { // more regular updates for actual changes seems better. | |||
float tone2=analogRead(15); | |||
tone2=floor(((tone2-512)/512)*70)/10; | |||
if(tone2!=tone1) | |||
{ | |||
// calcBiquad(FilterType,FrequencyC,dBgain,Q,QuantizationUnit,SampleRate,int*); | |||
calcBiquad(FILTER_PARAEQ,110,-tone2,0.2,2147483648,44100,updateFilter); | |||
filterBass_L.updateCoefs(updateFilter); | |||
filterBass_R.updateCoefs(updateFilter); | |||
calcBiquad(FILTER_PARAEQ,4400,tone2,0.167,2147483648,44100,updateFilter); | |||
filterTreb_L.updateCoefs(updateFilter); | |||
filterTreb_R.updateCoefs(updateFilter); | |||
tone1=tone2; | |||
} | |||
chgMsec = 0; | |||
} | |||
} | |||
@@ -0,0 +1,64 @@ | |||
// Tone example using SGTL5000 DAP PEQ filters and calcBiquad filter calculator routine. | |||
#include <Audio.h> | |||
#include <Wire.h> | |||
#include <SD.h> | |||
const int myInput = AUDIO_INPUT_LINEIN; | |||
// const int myInput = AUDIO_INPUT_MIC; | |||
int updateFilter[5]; | |||
AudioInputI2S audioInput; // audio shield: mic or line-in | |||
AudioOutputI2S audioOutput; // audio shield: headphones & line-out | |||
// Create Audio connections between the components | |||
// | |||
AudioConnection c1(audioInput, 0, audioOutput, 0); // left passing through | |||
AudioConnection c2(audioInput, 1, audioOutput, 1); // right passing through | |||
// Create an object to control the audio shield. | |||
// | |||
AudioControlSGTL5000 audioShield; | |||
void setup() { | |||
// Audio connections require memory to work. For more | |||
// detailed information, see the MemoryAndCpuUsage example | |||
AudioMemory(4); | |||
// Enable the audio shield, select the input and set the output volume. | |||
audioShield.enable(); | |||
audioShield.inputSelect(myInput); | |||
audioShield.volume(75); | |||
audioShield.unmuteLineout(); | |||
audioShield.dap_enable(); // enable the DAP block in SGTL5000 | |||
audioShield.dap_audio_eq(1); // using PEQ Biquad filters | |||
audioShield.dap_peqs(2); // enable filter 0 & filter 1 | |||
calcBiquad(FILTER_PARAEQ,110,0,0.2,524288,44100,updateFilter); | |||
audioShield.load_peq(0,updateFilter); | |||
calcBiquad(FILTER_PARAEQ,4400,0,0.167,524288,44100,updateFilter); | |||
audioShield.load_peq(1,updateFilter); | |||
} | |||
elapsedMillis chgMsec=0; | |||
float tone1=0; | |||
void loop() { | |||
// every 10 ms, check for adjustment the tone & vol | |||
if (chgMsec > 10) { // more regular updates for actual changes seems better. | |||
float tone2=analogRead(15); | |||
tone2=floor(((tone2-512)/512)*70)/10; | |||
if(tone2!=tone1) | |||
{ | |||
// calcBiquad(FilterType,FrequencyC,dBgain,Q,QuantizationUnit,SampleRate,int*); | |||
calcBiquad(FILTER_PARAEQ,110,-tone2,0.2,524288,44100,updateFilter); | |||
audioShield.load_peq(0,updateFilter); | |||
calcBiquad(FILTER_PARAEQ,4400,tone2,0.167,524288,44100,updateFilter); | |||
audioShield.load_peq(1,updateFilter); | |||
tone1=tone2; | |||
} | |||
chgMsec = 0; | |||
} | |||
} | |||
@@ -0,0 +1,62 @@ | |||
// DAC balance example: Will influence both HP & LO outputs. | |||
#include <Audio.h> | |||
#include <Wire.h> | |||
#include <SD.h> | |||
const int myInput = AUDIO_INPUT_LINEIN; | |||
// const int myInput = AUDIO_INPUT_MIC; | |||
// Create the Audio components. These should be created in the | |||
// order data flows, inputs/sources -> processing -> outputs | |||
// | |||
AudioInputI2S audioInput; // audio shield: mic or line-in | |||
AudioOutputI2S audioOutput; // audio shield: headphones & line-out | |||
// Create Audio connections between the components | |||
// | |||
AudioConnection c1(audioInput, 0, audioOutput, 0); // left passing through | |||
AudioConnection c2(audioInput, 1, audioOutput, 1); // right passing through | |||
// Create an object to control the audio shield. | |||
// | |||
AudioControlSGTL5000 audioShield; | |||
void setup() { | |||
// Audio connections require memory to work. For more | |||
// detailed information, see the MemoryAndCpuUsage example | |||
AudioMemory(4); | |||
// Enable the audio shield and set the output volume. | |||
audioShield.enable(); | |||
audioShield.inputSelect(myInput); | |||
audioShield.volume(75); | |||
audioShield.unmuteLineout(); | |||
} | |||
elapsedMillis chgMsec=0; | |||
float lastBal=1024; | |||
void loop() { | |||
// every 10 ms, check for adjustment the balance & vol | |||
if (chgMsec > 10) { // more regular updates for actual changes seems better. | |||
float bal1=analogRead(15); | |||
bal1=((bal1-512)/512)*100; | |||
bal1=(int)bal1; | |||
if(lastBal!=bal1) | |||
{ | |||
if(bal1<0) | |||
{ // leaning toward left... | |||
audioShield.dac_vol(100,100+bal1); | |||
} else if(bal1>0) { // to the right | |||
audioShield.dac_vol(100-bal1,100); | |||
} else { // middle | |||
audioShield.dac_vol(100); | |||
} | |||
lastBal=bal1; | |||
} | |||
chgMsec = 0; | |||
} | |||
} | |||
@@ -0,0 +1,62 @@ | |||
// HP balance example: Will influence only HP output. | |||
#include <Audio.h> | |||
#include <Wire.h> | |||
#include <SD.h> | |||
const int myInput = AUDIO_INPUT_LINEIN; | |||
// const int myInput = AUDIO_INPUT_MIC; | |||
// Create the Audio components. These should be created in the | |||
// order data flows, inputs/sources -> processing -> outputs | |||
// | |||
AudioInputI2S audioInput; // audio shield: mic or line-in | |||
AudioOutputI2S audioOutput; // audio shield: headphones & line-out | |||
// Create Audio connections between the components | |||
// Just connecting in to out | |||
AudioConnection c1(audioInput, 0, audioOutput, 0); // left connection | |||
AudioConnection c2(audioInput, 1, audioOutput, 1); // right connection | |||
// Create an object to control the audio shield. | |||
// | |||
AudioControlSGTL5000 audioShield; | |||
void setup() { | |||
// Audio connections require memory to work. For more | |||
// detailed information, see the MemoryAndCpuUsage example | |||
AudioMemory(4); | |||
// Enable the audio shield and set the output volume. | |||
audioShield.enable(); | |||
audioShield.inputSelect(myInput); | |||
audioShield.volume(60); | |||
audioShield.unmuteLineout(); | |||
} | |||
elapsedMillis chgMsec=0; | |||
float lastBal=1024; | |||
float vol1=75; | |||
void loop() { | |||
// every 10 ms, check for adjustment the balance & vol | |||
if (chgMsec > 10) { // more regular updates for actual changes seems better. | |||
float bal1=analogRead(15); | |||
bal1=((bal1-512)/512)*100; | |||
bal1=(int)bal1; | |||
if(lastBal!=bal1) | |||
{ | |||
if(bal1<0) | |||
{ | |||
audioShield.volume(vol1,(vol1/100)*(100+bal1)); | |||
} else { | |||
audioShield.volume((vol1/100)*(100-bal1),vol1); | |||
} | |||
lastBal=bal1; | |||
} | |||
chgMsec = 0; | |||
} | |||
} | |||