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Initial support for ADC input on Teensy 4.0

dds
PaulStoffregen 4 yıl önce
ebeveyn
66d89d911c
işleme
5d6cdd51aa
2 değiştirilmiş dosya ile 226 ekleme ve 15 silme
Görünümü Böl Farklılık Durumunu Göster
  1. +226
    -14
      input_adc.cpp
  2. +0
    -1
      input_adc.h

+ 226
- 14
input_adc.cpp Dosyayı Görüntüle

@@ -24,7 +24,7 @@
* THE SOFTWARE.
*/

#if !defined(__IMXRT1052__) && !defined(__IMXRT1062__)
#if defined(KINETISK)
#include <Arduino.h>
#include "input_adc.h"
@@ -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) {
@@ -214,3 +208,221 @@ void AudioInputAnalog::update(void)
release(out_left);
}
#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;
TMR3_ENBL &= ~(1<<3);
TMR3_SCTRL3 = TMR_SCTRL_OEN | TMR_SCTRL_FORCE;
TMR3_CSCTRL3 = TMR_CSCTRL_CL1(1) | TMR_CSCTRL_TCF1EN;
TMR3_CNTR3 = 0;
TMR3_LOAD3 = 0;
TMR3_COMP13 = comp1;
TMR3_CMPLD13 = comp1;
TMR3_CTRL3 = TMR_CTRL_CM(1) | TMR_CTRL_PCS(8) | TMR_CTRL_LENGTH | TMR_CTRL_OUTMODE(3);
TMR3_DMA3 = TMR_DMA_CMPLD1DE;
CORE_PIN15_CONFIG = 1 ; // GPIO_AD_B1_03, ALT1 = QTIMER3_TIMER3, page 495
TMR3_CNTR3 = 0;
TMR3_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_QTIMER3_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

+ 0
- 1
input_adc.h Dosyayı Görüntüle

@@ -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;

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