Make input ADC and output DAC play nicely together

examples/HardwareTesting/PassThroughAnalog/PassThroughAnalog.ino

@@ -1,41 +0,0 @@
-#include <Audio.h>
-#include <Wire.h>
-#include <SPI.h>
-#include <SD.h>
-
-
-// Create the Audio components.  These should be created in the
-// order data flows, inputs/sources -> processing -> outputs
-//
-AudioInputAnalog analogPinInput(A2); // analog A2 (pin 16)
-AudioOutputI2S   audioOutput;        // audio shield: headphones & line-out
-AudioOutputPWM   pwmOutput;          // audio output with PWM on pins 3 & 4
-
-// Create Audio connections between the components
-//
-AudioConnection c1(analogPinInput, 0, audioOutput, 0);
-AudioConnection c2(analogPinInput, 0, audioOutput, 1);
-AudioConnection c3(analogPinInput, 0, pwmOutput, 0);
-
-// 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 and set the output volume.
-  audioShield.enable();
-  audioShield.volume(0.6);
-}
-
-void loop() {
-  // Do nothing here.  The Audio flows automatically
-
-  // When AudioInputAnalog is running, analogRead() must NOT be used.
-}
-
-

examples/HardwareTesting/PassThroughMono/PassThroughMono.ino

@@ -0,0 +1,41 @@
+#include <Audio.h>
+#include <Wire.h>
+#include <SPI.h>
+#include <SD.h>
+#include "utility/pdb.h"
+
+// GUItool: begin automatically generated code
+AudioInputAnalog         adc1;           //xy=161,80
+AudioOutputAnalog        dac1;           //xy=329,47
+AudioOutputPWM           pwm1;           //xy=331,125
+AudioConnection          patchCord1(adc1, dac1);
+AudioConnection          patchCord2(adc1, pwm1);
+// GUItool: end automatically generated code
+
+void setup() {
+  // Audio connections require memory to work.  For more
+  // detailed information, see the MemoryAndCpuUsage example
+  AudioMemory(12);
+}
+
+void loop() {
+  // Do nothing here.  The Audio flows automatically
+
+  // When AudioInputAnalog is running, analogRead() must NOT be used.
+  delay(200);
+  Serial.print("PDB: SC=");
+  Serial.print(PDB0_SC, HEX);
+  Serial.print(", CONFIG=");
+  Serial.print(PDB_CONFIG, HEX);
+  Serial.print(", MOD=");
+  Serial.print(PDB0_MOD);
+  Serial.print(", IDLY=");
+  Serial.print(PDB0_IDLY);
+  Serial.print(", CH0C1=");
+  Serial.print(PDB0_CH0C1);
+  Serial.print(", DMA: ");
+  Serial.print(dma_channel_allocated_mask, HEX);
+  Serial.println();
+}
+
+

examples/HardwareTesting/PassThroughPlusTone/PassThroughPlusTone.ino

@@ -1,95 +0,0 @@
-#include <Audio.h>
-#include <Bounce.h>
-#include <Wire.h>
-#include <SPI.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
-//
-//AudioInputAnalog analogPinInput(16); // analog A2 (pin 16)
-AudioInputI2S       audioInput;         // audio shield: mic or line-in
-AudioSynthWaveform  toneLow(AudioWaveformSine);
-AudioSynthWaveform  toneHigh(AudioWaveformTriangle);
-AudioMixer4         mixerLeft;
-AudioMixer4         mixerRight;
-AudioOutputI2S      audioOutput;        // audio shield: headphones & line-out
-AudioOutputPWM      pwmOutput;          // audio output with PWM on pins 3 & 4
-
-// Create Audio connections between the components
-//
-AudioConnection c1(audioInput, 0, mixerLeft, 0);
-AudioConnection c2(audioInput, 1, mixerRight, 0);
-AudioConnection c3(toneHigh, 0, mixerLeft, 1);
-AudioConnection c4(toneHigh, 0, mixerRight, 1);
-AudioConnection c5(toneLow, 0, mixerLeft, 2);
-AudioConnection c6(toneLow, 0, mixerRight, 2);
-AudioConnection c7(mixerLeft, 0, audioOutput, 0);
-AudioConnection c8(mixerRight, 0, audioOutput, 1);
-AudioConnection c9(mixerLeft, 0, pwmOutput, 0);
-
-// Create an object to control the audio shield.
-// 
-AudioControlSGTL5000 audioShield;
-
-
-// Bounce objects to read two pushbuttons (pins 0 and 1)
-//
-Bounce button0 = Bounce(0, 12);
-Bounce button1 = Bounce(1, 12);  // 12 ms debounce time
-
-
-void setup() {
-  pinMode(0, INPUT_PULLUP);
-  pinMode(1, INPUT_PULLUP);
-
-  // Audio connections require memory to work.  For more
-  // detailed information, see the MemoryAndCpuUsage example
-  AudioMemory(12);
-
-  // Enable the audio shield and set the output volume.
-  audioShield.enable();
-  audioShield.inputSelect(myInput);
-  audioShield.volume(0.60);
-}
-
-elapsedMillis volmsec=0;
-
-void loop() {
-  // Check each button
-  button0.update();
-  button1.update();
-
-  // fallingEdge = high (not pressed - voltage from pullup resistor)
-  //               to low (pressed - button connects pin to ground)
-  if (button0.fallingEdge()) {
-    toneLow.frequency(256);
-    toneLow.amplitude(0.4);
-  }
-  if (button1.fallingEdge()) {
-    toneHigh.frequency(980);
-    toneHigh.amplitude(0.25);
-  } 
-
-  // risingEdge = low (pressed - button connects pin to ground)
-  //              to high (not pressed - voltage from pullup resistor)
-  if (button0.risingEdge()) {
-    toneLow.amplitude(0);
-  }
-  if (button1.risingEdge()) {
-    toneHigh.amplitude(0);
-  }
-
-  // every 50 ms, adjust the volume
-  if (volmsec > 50) {
-    float vol = analogRead(15);
-    vol = vol / 10.24;
-    audioShield.volume(vol);
-    volmsec = 0;
-  }
-}
-
-

gui/list.html

@@ -112,7 +112,7 @@
 
 
 <script type="text/javascript">
-	RED.nodes.registerType('AudioInputADC',{
+	RED.nodes.registerType('AudioInputAnalog',{
 	shortName: "adc",
 		inputs:0,
 		outputs:1,
@@ -121,7 +121,7 @@
 		icon: "arrow-in.png"
 	});
 </script>
-<script type="text/x-red" data-help-name="AudioInputADC">
+<script type="text/x-red" data-help-name="AudioInputAnalog">
 	<h3>Summary</h3>
 	<p>Receive audio using the built-in analog to digital converter.</p>
 	<h3>Audio Connections</h3>
@@ -129,18 +129,12 @@
 		<tr class=top><th>Port</th><th>Purpose</th></tr>
 		<tr class=odd><td align=center>Out 0</td><td>Audio Channel</td></tr>
 	</table>
-	<h3>Parameters</h3>
-	<table class=doc align=center cellpadding=3>
-		<tr class=top><th>Name</th><th>Type</th><th>Function</th></tr>
-		<tr class=odd><td align=center>Pin</td><td>Integer</td><td>Analog Pin To Use</td></tr>
-	</table>
-	<p>The pin number should be specified as "A0" to "A20"</p>
-	<p align=center><img src="adcpins2.jpg"></p>
-	<p align=center><img src="adcpins1.jpg"></p>
 	<h3>Functions</h3>
 	<p>This object has no functions to call from the Arduino sketch.  It
 		simply streams data from the ADC to its output port.</p>
 	<h3>Hardware</h3>
+	<p>Pin A2 is used for audio input.</p>
+	<p align=center><img src="adcpins2.jpg"></p>
 	<p>Signal range is 0 to 1.2V</p>
 	<p>Need for DC bias, approx 0.6V</p>
 	<p>TODO: suggested circuity for signal input</p>
@@ -151,7 +145,7 @@
 	<p>TODO: actual noise measurements with different input circuitry
 		(it's not nearly as quiet as the audio shield)</p>
 </script>
-<script type="text/x-red" data-template-name="AudioInputADC">
+<script type="text/x-red" data-template-name="AudioInputAnalog">
 	<div class="form-row">
 		<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
 		<input type="text" id="node-input-name" placeholder="Name">

input_adc.cpp

@@ -33,34 +33,17 @@ uint16_t AudioInputAnalog::block_offset = 0;
 bool AudioInputAnalog::update_responsibility = false;
 DMAChannel AudioInputAnalog::dma;
 
-// #define PDB_CONFIG (PDB_SC_TRGSEL(15) | PDB_SC_PDBEN | PDB_SC_CONT)
-// #define PDB_PERIOD 1087 // 48e6 / 44100
 
-void AudioInputAnalog::begin(unsigned int pin)
+AudioInputAnalog::AudioInputAnalog() : AudioStream(0, NULL)
 {
+	unsigned int pin = A2;
 	uint32_t i, sum=0;
 
-	// pin specified in user sketches should be A0 to A13
-	// numbers can be used, but the recommended usage is
-	// with the named constants A0 to A13
-	//   constants A0-A9 are actually 14 to 23
-	//   constants A10-A13 are actually 34 to 37
-	if (pin > 23 && !(pin >= 34 && pin <= 37)) return;
-
-	dma.begin(true); // Allocate the DMA channel first
-
-	//pinMode(2, OUTPUT);
-	//pinMode(3, OUTPUT);
-	//digitalWriteFast(3, HIGH);
-	//delayMicroseconds(500);
-	//digitalWriteFast(3, LOW);
-
 	// Configure the ADC and run at least one software-triggered
 	// conversion.  This completes the self calibration stuff and
 	// leaves the ADC in a state that's mostly ready to use
 	analogReadRes(16);
 	analogReference(INTERNAL); // range 0 to 1.2 volts
-	//analogReference(DEFAULT); // range 0 to 3.3 volts
 	analogReadAveraging(8);
 	// Actually, do many normal reads, to start with a nice DC level
 	for (i=0; i < 1024; i++) {
@@ -68,22 +51,25 @@ void AudioInputAnalog::begin(unsigned int pin)
 	}
 	dc_average = sum >> 10;
 
-	// testing only, enable adc interrupt
-	//ADC0_SC1A |= ADC_SC1_AIEN;
-	//while ((ADC0_SC1A & ADC_SC1_COCO) == 0) ; // wait
-	//NVIC_ENABLE_IRQ(IRQ_ADC0);
-
 	// set the programmable delay block to trigger the ADC at 44.1 kHz
-	SIM_SCGC6 |= SIM_SCGC6_PDB;
-	PDB0_MOD = PDB_PERIOD;
-	PDB0_SC = PDB_CONFIG | PDB_SC_LDOK;
-	PDB0_SC = PDB_CONFIG | PDB_SC_SWTRIG;
-	PDB0_CH0C1 = 0x0101;
+	if (!(SIM_SCGC6 & SIM_SCGC6_PDB)
+	  || (PDB0_SC & PDB_CONFIG) != PDB_CONFIG
+	  || PDB0_MOD != PDB_PERIOD
+	  || PDB0_IDLY != 1
+	  || PDB0_CH0C1 != 0x0101) {
+		SIM_SCGC6 |= SIM_SCGC6_PDB;
+		PDB0_IDLY = 1;
+		PDB0_MOD = PDB_PERIOD;
+		PDB0_SC = PDB_CONFIG | PDB_SC_LDOK;
+		PDB0_SC = PDB_CONFIG | PDB_SC_SWTRIG;
+		PDB0_CH0C1 = 0x0101;
+	}
 
 	// enable the ADC for hardware trigger and DMA
 	ADC0_SC2 |= ADC_SC2_ADTRG | ADC_SC2_DMAEN;
 
 	// set up a DMA channel to store the ADC data
+	dma.begin(true);
 	dma.TCD->SADDR = &ADC0_RA;
 	dma.TCD->SOFF = 0;
 	dma.TCD->ATTR = DMA_TCD_ATTR_SSIZE(1) | DMA_TCD_ATTR_DSIZE(1);
@@ -101,6 +87,7 @@ void AudioInputAnalog::begin(unsigned int pin)
 	dma.attachInterrupt(isr);
 }
 
+
 void AudioInputAnalog::isr(void)
 {
 	uint32_t daddr, offset;
@@ -108,7 +95,7 @@ void AudioInputAnalog::isr(void)
 	uint16_t *dest_left;
 	audio_block_t *left;
 
-	//digitalWriteFast(3, HIGH);
+	digitalWriteFast(2, HIGH);
 	daddr = (uint32_t)(dma.TCD->DADDR);
 	dma.clearInterrupt();
 
@@ -117,26 +104,26 @@ void AudioInputAnalog::isr(void)
 		// need to remove data from the second half
 		src = (uint16_t *)&analog_rx_buffer[AUDIO_BLOCK_SAMPLES/2];
 		end = (uint16_t *)&analog_rx_buffer[AUDIO_BLOCK_SAMPLES];
-		if (AudioInputAnalog::update_responsibility) AudioStream::update_all();
+		if (update_responsibility) AudioStream::update_all();
 	} else {
 		// DMA is receiving to the second half of the buffer
 		// need to remove data from the first half
 		src = (uint16_t *)&analog_rx_buffer[0];
 		end = (uint16_t *)&analog_rx_buffer[AUDIO_BLOCK_SAMPLES/2];
 	}
-	left = AudioInputAnalog::block_left;
+	left = block_left;
 	if (left != NULL) {
-		offset = AudioInputAnalog::block_offset;
+		offset = block_offset;
 		if (offset > AUDIO_BLOCK_SAMPLES/2) offset = AUDIO_BLOCK_SAMPLES/2;
 		//if (offset <= AUDIO_BLOCK_SAMPLES/2) {
 			dest_left = (uint16_t *)&(left->data[offset]);
-			AudioInputAnalog::block_offset = offset + AUDIO_BLOCK_SAMPLES/2;
+			block_offset = offset + AUDIO_BLOCK_SAMPLES/2;
 			do {
 				*dest_left++ = *src++;
 			} while (src < end);
 		//}
 	}
-	//digitalWriteFast(3, LOW);
+	digitalWriteFast(2, LOW);
 }
 
 
@@ -157,6 +144,8 @@ void AudioInputAnalog::update(void)
 	unsigned int dc, offset;
 	int16_t s, *p, *end;
 
+	//Serial.println("update");
+
 	// allocate new block (ok if NULL)
 	new_left = allocate();
 

input_adc.h

@@ -33,9 +33,9 @@
 class AudioInputAnalog : public AudioStream
 {
 public:
-        AudioInputAnalog(unsigned int pin) : AudioStream(0, NULL) { begin(pin); }
+        AudioInputAnalog();
         virtual void update(void);
-        void begin(unsigned int pin);
+        friend void dma_ch9_isr(void);
 private:
         static audio_block_t *block_left;
         static uint16_t block_offset;

output_dac.cpp

@@ -48,11 +48,18 @@ void AudioOutputAnalog::begin(void)
 	}
 
 	// set the programmable delay block to trigger DMA requests
-	SIM_SCGC6 |= SIM_SCGC6_PDB;
-	PDB0_IDLY = 1;
-	PDB0_MOD = PDB_PERIOD;
-	PDB0_SC = PDB_CONFIG | PDB_SC_LDOK;
-	PDB0_SC = PDB_CONFIG | PDB_SC_SWTRIG | PDB_SC_PDBIE | PDB_SC_DMAEN;
+	if (!(SIM_SCGC6 & SIM_SCGC6_PDB)
+	  || (PDB0_SC & PDB_CONFIG) != PDB_CONFIG
+	  || PDB0_MOD != PDB_PERIOD
+	  || PDB0_IDLY != 1
+	  || PDB0_CH0C1 != 0x0101) {
+		SIM_SCGC6 |= SIM_SCGC6_PDB;
+		PDB0_IDLY = 1;
+		PDB0_MOD = PDB_PERIOD;
+		PDB0_SC = PDB_CONFIG | PDB_SC_LDOK;
+		PDB0_SC = PDB_CONFIG | PDB_SC_SWTRIG;
+		PDB0_CH0C1 = 0x0101;
+	}
 
 	dma.TCD->SADDR = dac_buffer;
 	dma.TCD->SOFF = 2;

output_pwm.h

@@ -35,7 +35,6 @@ class AudioOutputPWM : public AudioStream
 public:
 	AudioOutputPWM(void) : AudioStream(1, inputQueueArray) { begin(); }
 	virtual void update(void);
-	void begin(void);
 private:
 	static audio_block_t *block_1st;
 	static audio_block_t *block_2nd;
@@ -45,6 +44,7 @@ private:
 	audio_block_t *inputQueueArray[1];
 	static DMAChannel dma;
 	static void isr(void);
+	void begin(void);
 };
 
 #endif

utility/pdb.h

@@ -33,7 +33,7 @@
 // to set their sample rate.  They must all configure the same
 // period to avoid chaos.
 
-#define PDB_CONFIG (PDB_SC_TRGSEL(15) | PDB_SC_PDBEN | PDB_SC_CONT)
+#define PDB_CONFIG (PDB_SC_TRGSEL(15) | PDB_SC_PDBEN | PDB_SC_CONT | PDB_SC_PDBIE | PDB_SC_DMAEN)
 
 #if F_BUS == 60000000
   #define PDB_PERIOD (1360-1)