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/* Audio Library for Teensy 3.X
* Copyright (c) 2014, Paul Stoffregen, paul@pjrc.com
*
* Development of this audio library was funded by PJRC.COM, LLC by sales of
* Teensy and Audio Adaptor boards. Please support PJRC's efforts to develop
* open source software by purchasing Teensy or other PJRC products.
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice, development funding notice, and this permission
* notice shall be included in all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
#include <Arduino.h>
#include "input_adcs.h"
#include "utility/pdb.h"
#include "utility/dspinst.h"
#if defined(__MK20DX256__) || defined(__MK64FX512__) || defined(__MK66FX1M0__)
#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];
audio_block_t * AudioInputAnalogStereo::block_left = NULL;
audio_block_t * AudioInputAnalogStereo::block_right = NULL;
uint16_t AudioInputAnalogStereo::offset_left = 0;
uint16_t AudioInputAnalogStereo::offset_right = 0;
int32_t AudioInputAnalogStereo::hpf_y1[2] = { 0, 0 };
int32_t AudioInputAnalogStereo::hpf_x1[2] = { 0, 0 };
bool AudioInputAnalogStereo::update_responsibility = false;
DMAChannel AudioInputAnalogStereo::dma0(false);
DMAChannel AudioInputAnalogStereo::dma1(false);
static int analogReadADC1(uint8_t pin);
void AudioInputAnalogStereo::init(uint8_t pin0, uint8_t pin1)
{
uint32_t tmp;
//pinMode(32, OUTPUT);
//pinMode(33, OUTPUT);
// 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
#if F_BUS == 96000000 || F_BUS == 48000000 || F_BUS == 24000000
analogReadAveraging(8);
ADC1_SC3 = ADC_SC3_AVGE + ADC_SC3_AVGS(1);
#else
analogReadAveraging(4);
ADC1_SC3 = ADC_SC3_AVGE + ADC_SC3_AVGS(0);
#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(pin0);
tmp = ( ((int32_t) tmp) << 14);
hpf_x1[0] = tmp; // With constant DC level x1 would be x0
hpf_y1[0] = 0; // Output will settle here when stable
tmp = (uint16_t) analogReadADC1(pin1);
tmp = ( ((int32_t) tmp) << 14);
hpf_x1[1] = tmp; // With constant DC level x1 would be x0
hpf_y1[1] = 0; // Output will settle here when stable
// set the programmable delay block to trigger the ADC at 44.1 kHz
//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;
PDB0_CH1C1 = 0x0101;
//}
// enable the ADC for hardware trigger and DMA
ADC0_SC2 |= ADC_SC2_ADTRG | ADC_SC2_DMAEN;
ADC1_SC2 |= ADC_SC2_ADTRG | ADC_SC2_DMAEN;
// set up a DMA channel to store the ADC data
dma0.begin(true);
dma1.begin(true);
// ADC0_RA = 0x4003B010
// ADC1_RA = 0x400BB010
dma0.TCD->SADDR = &ADC0_RA;
dma0.TCD->SOFF = 0;
dma0.TCD->ATTR = DMA_TCD_ATTR_SSIZE(1) | DMA_TCD_ATTR_DSIZE(1);
dma0.TCD->NBYTES_MLNO = 2;
dma0.TCD->SLAST = 0;
dma0.TCD->DADDR = left_buffer;
dma0.TCD->DOFF = 2;
dma0.TCD->CITER_ELINKNO = sizeof(left_buffer) / 2;
dma0.TCD->DLASTSGA = -sizeof(left_buffer);
dma0.TCD->BITER_ELINKNO = sizeof(left_buffer) / 2;
dma0.TCD->CSR = DMA_TCD_CSR_INTHALF | DMA_TCD_CSR_INTMAJOR;
dma1.TCD->SADDR = &ADC1_RA;
dma1.TCD->SOFF = 0;
dma1.TCD->ATTR = DMA_TCD_ATTR_SSIZE(1) | DMA_TCD_ATTR_DSIZE(1);
dma1.TCD->NBYTES_MLNO = 2;
dma1.TCD->SLAST = 0;
dma1.TCD->DADDR = right_buffer;
dma1.TCD->DOFF = 2;
dma1.TCD->CITER_ELINKNO = sizeof(right_buffer) / 2;
dma1.TCD->DLASTSGA = -sizeof(right_buffer);
dma1.TCD->BITER_ELINKNO = sizeof(right_buffer) / 2;
dma1.TCD->CSR = DMA_TCD_CSR_INTHALF | DMA_TCD_CSR_INTMAJOR;
dma0.triggerAtHardwareEvent(DMAMUX_SOURCE_ADC0);
//dma1.triggerAtHardwareEvent(DMAMUX_SOURCE_ADC1);
dma1.triggerAtTransfersOf(dma0);
dma1.triggerAtCompletionOf(dma0);
update_responsibility = update_setup();
dma0.enable();
dma1.enable();
dma0.attachInterrupt(isr0);
dma1.attachInterrupt(isr1);
}
void AudioInputAnalogStereo::isr0(void)
{
uint32_t daddr, offset;
const uint16_t *src, *end;
uint16_t *dest;
daddr = (uint32_t)(dma0.TCD->DADDR);
dma0.clearInterrupt();
//digitalWriteFast(32, HIGH);
if (daddr < (uint32_t)left_buffer + sizeof(left_buffer) / 2) {
// DMA is receiving to the first half of the buffer
// need to remove data from the second half
src = (uint16_t *)&left_buffer[AUDIO_BLOCK_SAMPLES/2];
end = (uint16_t *)&left_buffer[AUDIO_BLOCK_SAMPLES];
} else {
// DMA is receiving to the second half of the buffer
// need to remove data from the first half
src = (uint16_t *)&left_buffer[0];
end = (uint16_t *)&left_buffer[AUDIO_BLOCK_SAMPLES/2];
//if (update_responsibility) AudioStream::update_all();
}
if (block_left != NULL) {
offset = offset_left;
if (offset > AUDIO_BLOCK_SAMPLES/2) offset = AUDIO_BLOCK_SAMPLES/2;
offset_left = offset + AUDIO_BLOCK_SAMPLES/2;
dest = (uint16_t *)&(block_left->data[offset]);
do {
*dest++ = *src++;
} while (src < end);
}
//digitalWriteFast(32, LOW);
}
void AudioInputAnalogStereo::isr1(void)
{
uint32_t daddr, offset;
const uint16_t *src, *end;
uint16_t *dest;
daddr = (uint32_t)(dma1.TCD->DADDR);
dma1.clearInterrupt();
//digitalWriteFast(33, HIGH);
if (daddr < (uint32_t)right_buffer + sizeof(right_buffer) / 2) {
// DMA is receiving to the first half of the buffer
// need to remove data from the second half
src = (uint16_t *)&right_buffer[AUDIO_BLOCK_SAMPLES/2];
end = (uint16_t *)&right_buffer[AUDIO_BLOCK_SAMPLES];
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 *)&right_buffer[0];
end = (uint16_t *)&right_buffer[AUDIO_BLOCK_SAMPLES/2];
}
if (block_right != NULL) {
offset = offset_right;
if (offset > AUDIO_BLOCK_SAMPLES/2) offset = AUDIO_BLOCK_SAMPLES/2;
offset_right = offset + AUDIO_BLOCK_SAMPLES/2;
dest = (uint16_t *)&(block_right->data[offset]);
do {
*dest++ = *src++;
} while (src < end);
}
//digitalWriteFast(33, LOW);
}
void AudioInputAnalogStereo::update(void)
{
audio_block_t *new_left=NULL, *out_left=NULL;
audio_block_t *new_right=NULL, *out_right=NULL;
int32_t tmp;
int16_t s, *p, *end;
//Serial.println("update");
// allocate new block (ok if both NULL)
new_left = allocate();
if (new_left == NULL) {
new_right = NULL;
} else {
new_right = allocate();
if (new_right == NULL) {
release(new_left);
new_left = NULL;
}
}
__disable_irq();
if (offset_left < AUDIO_BLOCK_SAMPLES || offset_right < AUDIO_BLOCK_SAMPLES) {
// the DMA hasn't filled up both blocks
if (block_left == NULL) {
block_left = new_left;
offset_left = 0;
new_left = NULL;
}
if (block_right == NULL) {
block_right = new_right;
offset_right = 0;
new_right = NULL;
}
__enable_irq();
if (new_left) release(new_left);
if (new_right) release(new_right);
return;
}
// the DMA filled blocks, so grab them and get the
// new blocks to the DMA, as quickly as possible
out_left = block_left;
out_right = block_right;
block_left = new_left;
block_right = new_right;
offset_left = 0;
offset_right = 0;
__enable_irq();
//
// DC Offset Removal Filter
// 1-pole digital high-pass filter implementation
// y = a*(x[n] - x[n-1] + y[n-1])
// The coefficient "a" is as follows:
// a = UNITY*e^(-2*pi*fc/fs)
// fc = 2 @ fs = 44100
//
// DC removal, LEFT
p = out_left->data;
end = p + AUDIO_BLOCK_SAMPLES;
do {
tmp = (uint16_t)(*p);
tmp = ( ((int32_t) tmp) << 14);
int32_t acc = hpf_y1[0] - hpf_x1[0];
acc += tmp;
hpf_y1[0] = FRACMUL_SHL(acc, COEF_HPF_DCBLOCK, 1);
hpf_x1[0] = tmp;
s = signed_saturate_rshift(hpf_y1[0], 16, 14);
*p++ = s;
} while (p < end);
// DC removal, RIGHT
p = out_right->data;
end = p + AUDIO_BLOCK_SAMPLES;
do {
tmp = (uint16_t)(*p);
tmp = ( ((int32_t) tmp) << 14);
int32_t acc = hpf_y1[1] - hpf_x1[1];
acc += tmp;
hpf_y1[1]= FRACMUL_SHL(acc, COEF_HPF_DCBLOCK, 1);
hpf_x1[1] = tmp;
s = signed_saturate_rshift(hpf_y1[1], 16, 14);
*p++ = s;
} while (p < end);
// then transmit the AC data
transmit(out_left, 0);
release(out_left);
transmit(out_right, 1);
release(out_right);
}
#if defined(__MK20DX256__)
static const uint8_t pin2sc1a[] = {
5, 14, 8+128, 9+128, 13, 12, 6, 7, 15, 4, 0, 19, 3, 19+128, // 0-13 -> A0-A13
5, 14, 8+128, 9+128, 13, 12, 6, 7, 15, 4, // 14-23 are A0-A9
255, 255, // 24-25 are digital only
5+192, 5+128, 4+128, 6+128, 7+128, 4+192, // 26-31 are A15-A20
255, 255, // 32-33 are digital only
0, 19, 3, 19+128, // 34-37 are A10-A13
26, // 38 is temp sensor,
18+128, // 39 is vref
23 // 40 is A14
};
#elif defined(__MK64FX512__) || defined(__MK66FX1M0__)
static const uint8_t pin2sc1a[] = {
5, 14, 8+128, 9+128, 13, 12, 6, 7, 15, 4, 3, 19+128, 14+128, 15+128, // 0-13 -> A0-A13
5, 14, 8+128, 9+128, 13, 12, 6, 7, 15, 4, // 14-23 are A0-A9
255, 255, 255, 255, 255, 255, 255, // 24-30 are digital only
14+128, 15+128, 17, 18, 4+128, 5+128, 6+128, 7+128, 17+128, // 31-39 are A12-A20
255, 255, 255, 255, 255, 255, 255, 255, 255, // 40-48 are digital only
10+128, 11+128, // 49-50 are A23-A24
255, 255, 255, 255, 255, 255, 255, // 51-57 are digital only
255, 255, 255, 255, 255, 255, // 58-63 (sd card pins) are digital only
3, 19+128, // 64-65 are A10-A11
23, 23+128,// 66-67 are A21-A22 (DAC pins)
1, 1+128, // 68-69 are A25-A26 (unused USB host port on Teensy 3.5)
26, // 70 is Temperature Sensor
18+128 // 71 is Vref
};
#endif
static int analogReadADC1(uint8_t pin)
{
ADC1_SC1A = 9;
while (1) {
if ((ADC1_SC1A & ADC_SC1_COCO)) {
return ADC1_RA;
}
}
if (pin >= sizeof(pin2sc1a)) return 0;
uint8_t channel = pin2sc1a[pin];
if ((channel & 0x80) == 0) return 0;
if (channel == 255) return 0;
if (channel & 0x40) {
ADC1_CFG2 &= ~ADC_CFG2_MUXSEL;
} else {
ADC1_CFG2 |= ADC_CFG2_MUXSEL;
}
ADC1_SC1A = channel & 0x3F;
while (1) {
if ((ADC1_SC1A & ADC_SC1_COCO)) {
return ADC1_RA;
}
}
}
#else
void AudioInputAnalogStereo::init(uint8_t pin0, uint8_t pin1)
{
}
void AudioInputAnalogStereo::update(void)
{
}
#endif
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