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input_adc.cpp
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input_adc.cpp
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/* Audio Library for Teensy 3.X
* Copyright (c) 2014, Paul Stoffregen, [email protected]
*
* 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 "input_adc.h"
#include "utility/pdb.h"
#include "utility/dspinst.h"
DMAMEM static uint16_t analog_rx_buffer[AUDIO_BLOCK_SAMPLES];
audio_block_t * AudioInputAnalog::block_left = NULL;
uint16_t AudioInputAnalog::block_offset = 0;
uint16_t AudioInputAnalog::dc_average = 0;
bool AudioInputAnalog::update_responsibility = false;
DMAChannel AudioInputAnalog::dma(false);
void AudioInputAnalog::init(uint8_t pin)
{
uint32_t i, sum=0;
// 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
analogReadAveraging(8);
// Actually, do many normal reads, to start with a nice DC level
for (i=0; i < 1024; i++) {
sum += analogRead(pin);
}
dc_average = sum >> 10;
// 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;
}
// 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);
dma.TCD->NBYTES_MLNO = 2;
dma.TCD->SLAST = 0;
dma.TCD->DADDR = analog_rx_buffer;
dma.TCD->DOFF = 2;
dma.TCD->CITER_ELINKNO = sizeof(analog_rx_buffer) / 2;
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;
dma.triggerAtHardwareEvent(DMAMUX_SOURCE_ADC0);
update_responsibility = update_setup();
dma.enable();
dma.attachInterrupt(isr);
}
void AudioInputAnalog::isr(void)
{
uint32_t daddr, offset;
const uint16_t *src, *end;
uint16_t *dest_left;
audio_block_t *left;
daddr = (uint32_t)(dma.TCD->DADDR);
dma.clearInterrupt();
if (daddr < (uint32_t)analog_rx_buffer + sizeof(analog_rx_buffer) / 2) {
// DMA is receiving to the first half of the buffer
// 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 (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 = block_left;
if (left != NULL) {
offset = block_offset;
if (offset > AUDIO_BLOCK_SAMPLES/2) offset = AUDIO_BLOCK_SAMPLES/2;
dest_left = (uint16_t *)&(left->data[offset]);
block_offset = offset + AUDIO_BLOCK_SAMPLES/2;
do {
*dest_left++ = *src++;
} while (src < end);
}
}
void AudioInputAnalog::update(void)
{
audio_block_t *new_left=NULL, *out_left=NULL;
unsigned int dc, offset;
int32_t tmp;
int16_t s, *p, *end;
//Serial.println("update");
// allocate new block (ok if NULL)
new_left = allocate();
__disable_irq();
offset = block_offset;
if (offset < AUDIO_BLOCK_SAMPLES) {
// the DMA didn't fill a block
if (new_left != NULL) {
// but we allocated a block
if (block_left == NULL) {
// the DMA doesn't have any blocks to fill, so
// give it the one we just allocated
block_left = new_left;
block_offset = 0;
__enable_irq();
//Serial.println("fail1");
} else {
// the DMA already has blocks, doesn't need this
__enable_irq();
release(new_left);
//Serial.print("fail2, offset=");
//Serial.println(offset);
}
} else {
// The DMA didn't fill a block, and we could not allocate
// memory... the system is likely starving for memory!
// Sadly, there's nothing we can do.
__enable_irq();
//Serial.println("fail3");
}
return;
}
// the DMA filled a block, so grab it and get the
// new block to the DMA, as quickly as possible
out_left = block_left;
block_left = new_left;
block_offset = 0;
__enable_irq();
// find and subtract DC offset....
// TODO: this may not be correct, needs testing with more types of signals
dc = dc_average;
p = out_left->data;
end = p + AUDIO_BLOCK_SAMPLES;
do {
tmp = (uint16_t)(*p) - (int32_t)dc;
s = signed_saturate_rshift(tmp, 16, 0);
*p++ = s;
dc += s / 12000; // slow response, remove DC component
} while (p < end);
dc_average = dc;
// then transmit the AC data
transmit(out_left);
release(out_left);
}