Merge remote-tracking branch 'upstream/master'

control_sgtl5000.cpp

@@ -531,7 +531,7 @@ bool AudioControlSGTL5000::enable(void)
 	delay(400);
 	write(CHIP_LINE_OUT_VOL, 0x1D1D); // default approx 1.3 volts peak-to-peak
 	write(CHIP_CLK_CTRL, 0x0004);  // 44.1 kHz, 256*Fs
-	write(CHIP_I2S_CTRL, 0x0130); // SCLK=32*Fs, 16bit, I2S format
+	write(CHIP_I2S_CTRL, 0x0030); // SCLK=64*Fs, 16bit, I2S format
 	// default signal routing is ok?
 	write(CHIP_SSS_CTRL, 0x0010); // ADC->I2S, I2S->DAC
 	write(CHIP_ADCDAC_CTRL, 0x0000); // disable dac mute

examples/HardwareTesting/PassThroughADCtoI2S/PassThroughADCtoI2S.ino

@@ -0,0 +1,39 @@
+/*
+ * A simple hardware test which receives audio on the A2 analog pin
+ * and sends it to the audio shield (I2S digital audio)
+ *
+ * This example code is in the public domain.
+ */
+
+#include <Audio.h>
+#include <Wire.h>
+#include <SPI.h>
+#include <SD.h>
+#include <SerialFlash.h>
+
+// GUItool: begin automatically generated code
+AudioInputAnalog         adc1;           //xy=197,73
+AudioOutputI2S           i2s1;           //xy=375,85
+AudioConnection          patchCord1(adc1, 0, i2s1, 0);
+AudioConnection          patchCord2(adc1, 0, i2s1, 1);
+AudioControlSGTL5000     sgtl5000_1;     //xy=314,158
+// GUItool: end automatically generated code
+
+
+void setup() {
+  // Audio connections require memory to work.  For more
+  // detailed information, see the MemoryAndCpuUsage example
+  AudioMemory(12);
+
+  // Enable the audio shield
+  sgtl5000_1.enable();
+  sgtl5000_1.volume(0.5);
+}
+
+void loop() {
+  // Do nothing here.  The Audio flows automatically
+
+  // When AudioInputAnalog is running, analogRead() must NOT be used.
+}
+
+

gui/index.html

@@ -3236,7 +3236,7 @@ double s_freq = .0625;</p>
 		at altered speed.  The grainLength is specified in milliseconds, up to
 		one third of the memory from begin();
 		</p>
-	<p class=func><span class=keyword>end</span>();</p>
+	<p class=func><span class=keyword>stop</span>();</p>
 	<p class=desc>Stop granual processing.  The input signal is passed to the
 		output without any changes.
 		</p>

input_adc.cpp

@@ -24,7 +24,7 @@
  * THE SOFTWARE.
  */
 
-#if !defined(__IMXRT1052__) && !defined(__IMXRT1062__)
+#if defined(KINETISK)
  
 #include <Arduino.h>
 #include "input_adc.h"
@@ -33,7 +33,7 @@
 
 #define COEF_HPF_DCBLOCK    (1048300<<10)  // DC Removal filter coefficient in S1.30
 
-DMAMEM static uint16_t analog_rx_buffer[AUDIO_BLOCK_SAMPLES];
+DMAMEM __attribute__((aligned(32))) static uint16_t analog_rx_buffer[AUDIO_BLOCK_SAMPLES];
 audio_block_t * AudioInputAnalog::block_left = NULL;
 uint16_t AudioInputAnalog::block_offset = 0;
 int32_t AudioInputAnalog::hpf_y1 = 0;
@@ -44,7 +44,7 @@ DMAChannel AudioInputAnalog::dma(false);
 
 void AudioInputAnalog::init(uint8_t pin)
 {
-    int32_t tmp;
+	int32_t tmp;
 
 	// Configure the ADC and run at least one software-triggered
 	// conversion.  This completes the self calibration stuff and
@@ -57,15 +57,14 @@ void AudioInputAnalog::init(uint8_t pin)
 	analogReadAveraging(4);
 #endif
 	// Note for review:
-    // Probably not useful to spin cycles here stabilizing
-    // since DC blocking is similar to te external analog filters
-    tmp = (uint16_t) analogRead(pin);
-    tmp = ( ((int32_t) tmp) << 14);
-    hpf_x1 = tmp;   // With constant DC level x1 would be x0
-    hpf_y1 = 0;     // Output will settle here when stable
+	// Probably not useful to spin cycles here stabilizing
+	// since DC blocking is similar to te external analog filters
+	tmp = (uint16_t) analogRead(pin);
+	tmp = ( ((int32_t) tmp) << 14);
+	hpf_x1 = tmp;   // With constant DC level x1 would be x0
+	hpf_y1 = 0;     // Output will settle here when stable
 
 	// set the programmable delay block to trigger the ADC at 44.1 kHz
-#if defined(KINETISK)
 	if (!(SIM_SCGC6 & SIM_SCGC6_PDB)
 	  || (PDB0_SC & PDB_CONFIG) != PDB_CONFIG
 	  || PDB0_MOD != PDB_PERIOD
@@ -78,13 +77,11 @@ void AudioInputAnalog::init(uint8_t pin)
 		PDB0_SC = PDB_CONFIG | PDB_SC_SWTRIG;
 		PDB0_CH0C1 = 0x0101;
 	}
-#endif
 	// 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);
-#if defined(KINETISK)
 	dma.TCD->SADDR = &ADC0_RA;
 	dma.TCD->SOFF = 0;
 	dma.TCD->ATTR = DMA_TCD_ATTR_SSIZE(1) | DMA_TCD_ATTR_DSIZE(1);
@@ -96,7 +93,6 @@ void AudioInputAnalog::init(uint8_t pin)
 	dma.TCD->DLASTSGA = -sizeof(analog_rx_buffer);
 	dma.TCD->BITER_ELINKNO = sizeof(analog_rx_buffer) / 2;
 	dma.TCD->CSR = DMA_TCD_CSR_INTHALF | DMA_TCD_CSR_INTMAJOR;
-#endif
 	dma.triggerAtHardwareEvent(DMAMUX_SOURCE_ADC0);
 	update_responsibility = update_setup();
 	dma.enable();
@@ -111,9 +107,7 @@ void AudioInputAnalog::isr(void)
 	uint16_t *dest_left;
 	audio_block_t *left;
 
-#if defined(KINETISK)
 	daddr = (uint32_t)(dma.TCD->DADDR);
-#endif
 	dma.clearInterrupt();
 
 	if (daddr < (uint32_t)analog_rx_buffer + sizeof(analog_rx_buffer) / 2) {
@@ -213,4 +207,221 @@ void AudioInputAnalog::update(void)
 	transmit(out_left);
 	release(out_left);
 }
-#endif
+#endif
+
+
+
+#if defined(__IMXRT1062__)
+
+#include <Arduino.h>
+#include "input_adc.h"
+
+extern "C" void xbar_connect(unsigned int input, unsigned int output);
+
+#define FILTERLEN 15
+
+DMAChannel AudioInputAnalog::dma(false);
+// TODO: how much extra space is needed to avoid wrap-around timing?  200 seems a safe guess
+static __attribute__((aligned(32))) uint16_t adc_buffer[AUDIO_BLOCK_SAMPLES*4+200];
+static int16_t capture_buffer[AUDIO_BLOCK_SAMPLES*4+FILTERLEN];
+// TODO: these big buffers should be in DMAMEM, rather than consuming precious DTCM
+
+PROGMEM static const uint8_t adc2_pin_to_channel[] = {
+	7,      // 0/A0  AD_B1_02
+	8,      // 1/A1  AD_B1_03
+	12,     // 2/A2  AD_B1_07
+	11,     // 3/A3  AD_B1_06
+	6,      // 4/A4  AD_B1_01
+	5,      // 5/A5  AD_B1_00
+	15,     // 6/A6  AD_B1_10
+	0,      // 7/A7  AD_B1_11
+	13,     // 8/A8  AD_B1_08
+	14,     // 9/A9  AD_B1_09
+	255,	// 10/A10 AD_B0_12 - only on ADC1, 1 - can't use for audio
+	255,	// 11/A11 AD_B0_13 - only on ADC1, 2 - can't use for audio
+	3,      // 12/A12 AD_B1_14
+	4,      // 13/A13 AD_B1_15
+	7,      // 14/A0  AD_B1_02
+	8,      // 15/A1  AD_B1_03
+	12,     // 16/A2  AD_B1_07
+	11,     // 17/A3  AD_B1_06
+	6,      // 18/A4  AD_B1_01
+	5,      // 19/A5  AD_B1_00
+	15,     // 20/A6  AD_B1_10
+	0,      // 21/A7  AD_B1_11
+	13,     // 22/A8  AD_B1_08
+	14,     // 23/A9  AD_B1_09
+	255,    // 24/A10 AD_B0_12 - only on ADC1, 1 - can't use for audio
+	255,    // 25/A11 AD_B0_13 - only on ADC1, 2 - can't use for audio
+	3,      // 26/A12 AD_B1_14 - only on ADC2, do not use analogRead()
+	4,      // 27/A13 AD_B1_15 - only on ADC2, do not use analogRead()
+#ifdef ARDUINO_TEENSY41
+	255,    // 28
+	255,    // 29
+	255,    // 30
+	255,    // 31
+	255,    // 32
+	255,    // 33
+	255,    // 34
+	255,    // 35
+	255,    // 36
+	255,    // 37
+	1,      // 38/A14 AD_B1_12 - only on ADC2, do not use analogRead()
+	2,      // 39/A15 AD_B1_13 - only on ADC2, do not use analogRead()
+	9,      // 40/A16 AD_B1_04
+	10,     // 41/A17 AD_B1_05
+#endif
+};
+
+static const int16_t filter[FILTERLEN] = {
+	1449,
+	3676,
+	6137,
+	9966,
+	13387,
+	16896,
+	18951,
+	19957,
+	18951,
+	16896,
+	13387,
+	9966,
+	6137,
+	3676,
+	1449
+};
+
+
+void AudioInputAnalog::init(uint8_t pin)
+{
+	if (pin >= sizeof(adc2_pin_to_channel)) return;
+	const uint8_t adc_channel = adc2_pin_to_channel[pin];
+	if (adc_channel == 255) return;
+
+	// configure a timer to trigger ADC
+	// TODO: sample rate should be slightly lower than 4X AUDIO_SAMPLE_RATE_EXACT
+	//       linear interpolation is supposed to resample it to exactly 4X
+	//       the sample rate, so we avoid artifacts boundaries between captures
+	const int comp1 = ((float)F_BUS_ACTUAL) / (AUDIO_SAMPLE_RATE_EXACT * 4.0f) / 2.0f + 0.5f;
+	TMR4_ENBL &= ~(1<<3);
+	TMR4_SCTRL3 = TMR_SCTRL_OEN | TMR_SCTRL_FORCE;
+	TMR4_CSCTRL3 = TMR_CSCTRL_CL1(1) | TMR_CSCTRL_TCF1EN;
+	TMR4_CNTR3 = 0;
+	TMR4_LOAD3 = 0;
+	TMR4_COMP13 = comp1;
+	TMR4_CMPLD13 = comp1;
+	TMR4_CTRL3 = TMR_CTRL_CM(1) | TMR_CTRL_PCS(8) | TMR_CTRL_LENGTH | TMR_CTRL_OUTMODE(3);
+	TMR4_DMA3 = TMR_DMA_CMPLD1DE;
+	TMR4_CNTR3 = 0;
+	TMR4_ENBL |= (1<<3);
+
+	// connect the timer output the ADC_ETC input
+	const int trigger = 4; // 0-3 for ADC1, 4-7 for ADC2
+	CCM_CCGR2 |= CCM_CCGR2_XBAR1(CCM_CCGR_ON);
+	xbar_connect(XBARA1_IN_QTIMER4_TIMER3, XBARA1_OUT_ADC_ETC_TRIG00 + trigger);
+
+	// turn on ADC_ETC and configure to receive trigger
+	if (ADC_ETC_CTRL & (ADC_ETC_CTRL_SOFTRST | ADC_ETC_CTRL_TSC_BYPASS)) {
+		ADC_ETC_CTRL = 0; // clears SOFTRST only
+		ADC_ETC_CTRL = 0; // clears TSC_BYPASS
+	}
+	ADC_ETC_CTRL |= ADC_ETC_CTRL_TRIG_ENABLE(1 << trigger) | ADC_ETC_CTRL_DMA_MODE_SEL;
+	ADC_ETC_DMA_CTRL |= ADC_ETC_DMA_CTRL_TRIQ_ENABLE(trigger);
+
+	// configure ADC_ETC trigger4 to make one ADC2 measurement on pin A2
+	const int len = 1;
+	IMXRT_ADC_ETC.TRIG[trigger].CTRL = ADC_ETC_TRIG_CTRL_TRIG_CHAIN(len - 1) |
+		ADC_ETC_TRIG_CTRL_TRIG_PRIORITY(7);
+	IMXRT_ADC_ETC.TRIG[trigger].CHAIN_1_0 = ADC_ETC_TRIG_CHAIN_HWTS0(1) |
+		ADC_ETC_TRIG_CHAIN_CSEL0(adc2_pin_to_channel[pin]) | ADC_ETC_TRIG_CHAIN_B2B0;
+
+	// set up ADC2 for 12 bit mode, hardware trigger
+	Serial.printf("ADC2_CFG = %08X\n", ADC2_CFG);
+	ADC2_CFG |= ADC_CFG_ADTRG;
+	ADC2_CFG = ADC_CFG_MODE(2) | ADC_CFG_ADSTS(3) | ADC_CFG_ADLSMP | ADC_CFG_ADTRG |
+		ADC_CFG_ADICLK(1) | ADC_CFG_ADIV(0) /*| ADC_CFG_ADHSC*/;
+	ADC2_GC &= ~ADC_GC_AVGE; // single sample, no averaging
+	ADC2_HC0 = ADC_HC_ADCH(16); // 16 = controlled by ADC_ETC
+
+	// use a DMA channel to capture ADC_ETC output
+	dma.begin();
+	dma.TCD->SADDR = &(IMXRT_ADC_ETC.TRIG[4].RESULT_1_0);
+	dma.TCD->SOFF = 0;
+	dma.TCD->ATTR = DMA_TCD_ATTR_SSIZE(1) | DMA_TCD_ATTR_DSIZE(1);
+	dma.TCD->NBYTES_MLNO = 2;
+	dma.TCD->SLAST = 0;
+	dma.TCD->DADDR = adc_buffer;
+	dma.TCD->DOFF = 2;
+	dma.TCD->CITER_ELINKNO = sizeof(adc_buffer) / 2;
+	dma.TCD->DLASTSGA = -sizeof(adc_buffer);
+	dma.TCD->BITER_ELINKNO = sizeof(adc_buffer) / 2;
+	dma.TCD->CSR = 0;
+	dma.triggerAtHardwareEvent(DMAMUX_SOURCE_ADC_ETC);
+	dma.enable();
+
+	// TODO: configure I2S1 to interrupt every 128 audio samples
+}
+
+static int16_t fir(const int16_t *data, const int16_t *impulse, int len)
+{
+	int64_t sum=0;
+
+	while (len > 0) {
+		sum += *data++ * *impulse++; // TODO: optimize with DSP inst and filter symmetry
+		len --;
+	}
+	sum = sum >> 15; // TODO: adjust filter coefficients for proper gain, 12 to 16 bits
+	if (sum > 32767) return 32767;
+	if (sum < -32768) return -32768;
+	return sum;
+}
+
+void AudioInputAnalog::update(void)
+{
+	audio_block_t *output=NULL;
+	output = allocate();
+	if (output == NULL) return;
+
+	uint16_t *p = (uint16_t *)dma.TCD->DADDR;
+	//int offset = p - adc_buffer;
+	//if (--offset < 0) offset = sizeof(adc_buffer) / 2 - 1;
+	//Serial.printf("offset = %4d, val = %4d\n", offset + 1, adc_buffer[offset]);
+
+	// copy adc buffer to capture buffer
+	//  FIXME: this should be done from the I2S interrupt, for precise capture timing
+	const unsigned int capture_len = sizeof(capture_buffer) / 2;
+	for (unsigned int i=0; i < capture_len; i++) {
+		// TODO: linear interpolate to exactly 4X sample rate
+		if (--p < adc_buffer) p = adc_buffer + (sizeof(adc_buffer) / 2 - 1);
+
+		// remove DC offset
+		// TODO: very slow low pass filter for DC offset
+		int dc_offset = 550; // FIXME: quick kludge for testing!!
+
+		int n = (int)*p - dc_offset;
+		if (n > 4095) n = 4095;
+		if (n < -4095) n = -4095;
+
+		capture_buffer[i] = n;
+	}
+	//printbuf(capture_buffer, 8);
+
+	// low pass filter and subsample (this part belongs here)
+	int16_t *dest = output->data + AUDIO_BLOCK_SAMPLES - 1;
+	for (int i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
+#if 1
+		// proper low-pass filter sounds pretty good
+		*dest-- = fir(capture_buffer + i * 4, filter, sizeof(filter)/2);
+#else
+		// just averge 4 samples together, lower quality but much faster
+		*dest-- = capture_buffer[i * 4] + capture_buffer[i * 4 + 1]
+			+ capture_buffer[i * 4 + 2] + capture_buffer[i * 4 + 3];
+#endif
+	}
+	transmit(output);
+	release(output);
+}
+
+
+
+#endif

input_adc.h

@@ -37,7 +37,6 @@ public:
         AudioInputAnalog() : AudioStream(0, NULL) { init(A2); }
         AudioInputAnalog(uint8_t pin) : AudioStream(0, NULL) { init(pin); }
         virtual void update(void);
-        friend void dma_ch9_isr(void);
 private:
         static audio_block_t *block_left;
         static uint16_t block_offset;

input_adcs.cpp

@@ -33,8 +33,8 @@
 
 #define COEF_HPF_DCBLOCK    (1048300<<10)  // DC Removal filter coefficient in S1.30
 
-DMAMEM static uint16_t left_buffer[AUDIO_BLOCK_SAMPLES];
-DMAMEM static uint16_t right_buffer[AUDIO_BLOCK_SAMPLES];
+DMAMEM __attribute__((aligned(32))) static uint16_t left_buffer[AUDIO_BLOCK_SAMPLES];
+DMAMEM __attribute__((aligned(32))) static uint16_t right_buffer[AUDIO_BLOCK_SAMPLES];
 audio_block_t * AudioInputAnalogStereo::block_left = NULL;
 audio_block_t * AudioInputAnalogStereo::block_right = NULL;
 uint16_t AudioInputAnalogStereo::offset_left = 0;

input_i2s.cpp

@@ -30,7 +30,7 @@
 #include "input_i2s.h"
 #include "output_i2s.h"
 
-static uint32_t i2s_rx_buffer[AUDIO_BLOCK_SAMPLES];
+DMAMEM __attribute__((aligned(32))) static uint32_t i2s_rx_buffer[AUDIO_BLOCK_SAMPLES];
 audio_block_t * AudioInputI2S::block_left = NULL;
 audio_block_t * AudioInputI2S::block_right = NULL;
 uint16_t AudioInputI2S::block_offset = 0;
@@ -122,9 +122,7 @@ void AudioInputI2S::isr(void)
 			dest_left = &(left->data[offset]);
 			dest_right = &(right->data[offset]);
 			AudioInputI2S::block_offset = offset + AUDIO_BLOCK_SAMPLES/2;
-			#if IMXRT_CACHE_ENABLED >=1
-			arm_dcache_delete(src, sizeof(i2s_rx_buffer) / 2);
-			#endif
+			arm_dcache_delete((void*)src, sizeof(i2s_rx_buffer) / 2);
 			do {
 				*dest_left++ = *src++;
 				*dest_right++ = *src++;

input_i2s2.cpp

@@ -30,7 +30,7 @@
 #include "input_i2s2.h"
 #include "output_i2s2.h"
 
-static uint32_t i2s2_rx_buffer[AUDIO_BLOCK_SAMPLES];
+DMAMEM __attribute__((aligned(32))) static uint32_t i2s2_rx_buffer[AUDIO_BLOCK_SAMPLES];
 audio_block_t * AudioInputI2S2::block_left = NULL;
 audio_block_t * AudioInputI2S2::block_right = NULL;
 uint16_t AudioInputI2S2::block_offset = 0;
@@ -102,9 +102,9 @@ void AudioInputI2S2::isr(void)
 			dest_left = &(left->data[offset]);
 			dest_right = &(right->data[offset]);
 			AudioInputI2S2::block_offset = offset + AUDIO_BLOCK_SAMPLES/2;
-			#if IMXRT_CACHE_ENABLED >=1
-			arm_dcache_delete(src, sizeof(i2s_rx_buffer) / 2);
-			#endif
+
+			arm_dcache_delete((void*)src, sizeof(i2s2_rx_buffer) / 2);
+
 			do {
 				*dest_left++ = *src++;
 				*dest_right++ = *src++;

input_i2s_hex.cpp

@@ -112,7 +112,7 @@ void AudioInputI2SHex::isr(void)
 	if (block_ch1) {
 		offset = block_offset;
 		if (offset <= AUDIO_BLOCK_SAMPLES/2) {
-			arm_dcache_delete(src, sizeof(i2s_rx_buffer) / 2);
+			arm_dcache_delete((void*)src, sizeof(i2s_rx_buffer) / 2);
 			block_offset = offset + AUDIO_BLOCK_SAMPLES/2;
 			dest1 = &(block_ch1->data[offset]);
 			dest2 = &(block_ch2->data[offset]);

input_i2s_quad.cpp

@@ -148,7 +148,7 @@ void AudioInputI2SQuad::isr(void)
 	if (block_ch1) {
 		offset = block_offset;
 		if (offset <= AUDIO_BLOCK_SAMPLES/2) {
-			arm_dcache_delete(src, sizeof(i2s_rx_buffer) / 2);
+			arm_dcache_delete((void*)src, sizeof(i2s_rx_buffer) / 2);
 			block_offset = offset + AUDIO_BLOCK_SAMPLES/2;
 			dest1 = &(block_ch1->data[offset]);
 			dest2 = &(block_ch2->data[offset]);

input_pdm.cpp

@@ -52,7 +52,7 @@
 // but its performance should be *much* better than the rapid passband rolloff
 // of Cascaded Integrator Comb (CIC) or moving average filters.
 
-DMAMEM static uint32_t pdm_buffer[AUDIO_BLOCK_SAMPLES*4];
+DMAMEM __attribute__((aligned(32))) static uint32_t pdm_buffer[AUDIO_BLOCK_SAMPLES*4];
 static uint32_t leftover[14];
 audio_block_t * AudioInputPDM::block_left = NULL;
 bool AudioInputPDM::update_responsibility = false;

output_adat.cpp

@@ -56,7 +56,7 @@ uint16_t  AudioOutputADAT::ch8_offset = 0;
 bool AudioOutputADAT::update_responsibility = false;
 //uint32_t  AudioOutputADAT::vucp = VUCP_VALID;
 
-DMAMEM static uint32_t ADAT_tx_buffer[AUDIO_BLOCK_SAMPLES * 8]; //4 KB, AUDIO_BLOCK_SAMPLES is usually 128
+DMAMEM __attribute__((aligned(32))) static uint32_t ADAT_tx_buffer[AUDIO_BLOCK_SAMPLES * 8]; //4 KB, AUDIO_BLOCK_SAMPLES is usually 128
 
 DMAChannel AudioOutputADAT::dma(false);
 

output_dac.cpp

@@ -30,7 +30,7 @@
 
 #if defined(__MK20DX256__) || defined(__MK64FX512__) || defined(__MK66FX1M0__)
 
-DMAMEM static uint16_t dac_buffer[AUDIO_BLOCK_SAMPLES*2];
+DMAMEM __attribute__((aligned(32))) static uint16_t dac_buffer[AUDIO_BLOCK_SAMPLES*2];
 audio_block_t * AudioOutputAnalog::block_left_1st = NULL;
 audio_block_t * AudioOutputAnalog::block_left_2nd = NULL;
 bool AudioOutputAnalog::update_responsibility = false;

output_dacs.cpp

@@ -30,7 +30,7 @@
 
 #if defined(__MK64FX512__) || defined(__MK66FX1M0__)
 
-DMAMEM static uint32_t dac_buffer[AUDIO_BLOCK_SAMPLES*2];
+DMAMEM __attribute__((aligned(32))) static uint32_t dac_buffer[AUDIO_BLOCK_SAMPLES*2];
 audio_block_t * AudioOutputAnalogStereo::block_left_1st = NULL;
 audio_block_t * AudioOutputAnalogStereo::block_left_2nd = NULL;
 audio_block_t * AudioOutputAnalogStereo::block_right_1st = NULL;

output_i2s.cpp

@@ -475,6 +475,7 @@ void AudioOutputI2Sslave::begin(void)
 	dma.TCD->DLASTSGA = 0;
 	dma.TCD->BITER_ELINKNO = sizeof(i2s_tx_buffer) / 2;
 	dma.TCD->DADDR = (void *)((uint32_t)&I2S1_TDR0 + 2);
+	dma.TCD->CSR = DMA_TCD_CSR_INTHALF | DMA_TCD_CSR_INTMAJOR;
 	dma.triggerAtHardwareEvent(DMAMUX_SOURCE_SAI1_TX);
 	dma.enable();
 

output_pt8211.cpp

@@ -40,9 +40,9 @@ uint16_t  AudioOutputPT8211::block_left_offset = 0;
 uint16_t  AudioOutputPT8211::block_right_offset = 0;
 bool AudioOutputPT8211::update_responsibility = false;
 #if defined(AUDIO_PT8211_OVERSAMPLING)
-	static uint32_t i2s_tx_buffer[AUDIO_BLOCK_SAMPLES*4];
+DMAMEM __attribute__((aligned(32))) static uint32_t i2s_tx_buffer[AUDIO_BLOCK_SAMPLES*4];
 #else
-	static uint32_t i2s_tx_buffer[AUDIO_BLOCK_SAMPLES];
+DMAMEM __attribute__((aligned(32))) static uint32_t i2s_tx_buffer[AUDIO_BLOCK_SAMPLES];
 #endif
 DMAChannel AudioOutputPT8211::dma(false);
 

output_pt8211_2.cpp

@@ -40,9 +40,9 @@ uint16_t  AudioOutputPT8211_2::block_left_offset = 0;
 uint16_t  AudioOutputPT8211_2::block_right_offset = 0;
 bool AudioOutputPT8211_2::update_responsibility = false;
 #if defined(AUDIO_PT8211_OVERSAMPLING)
-	static uint32_t i2s_tx_buffer[AUDIO_BLOCK_SAMPLES*4];
+DMAMEM __attribute__((aligned(32))) static uint32_t i2s_tx_buffer[AUDIO_BLOCK_SAMPLES*4];
 #else
-	static uint32_t i2s_tx_buffer[AUDIO_BLOCK_SAMPLES];
+DMAMEM __attribute__((aligned(32))) static uint32_t i2s_tx_buffer[AUDIO_BLOCK_SAMPLES];
 #endif
 DMAChannel AudioOutputPT8211_2::dma(false);
 

output_pwm.cpp

@@ -213,7 +213,7 @@ extern uint8_t analog_write_res;
 extern const struct _pwm_pin_info_struct pwm_pin_info[];
 audio_block_t * AudioOutputPWM::block = NULL;
 DMAMEM __attribute__((aligned(32))) static uint16_t pwm_tx_buffer[2][AUDIO_BLOCK_SAMPLES * 2];
-DMAChannel AudioOutputPWM::dma[2](false);
+DMAChannel AudioOutputPWM::dma[2];
 _audio_info_flexpwm AudioOutputPWM::apins[2];
 
 FLASHMEM