teensy-audio/input_adc.cpp

/* 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 "input_adc.h"
#include "utility/pdb.h"
#include "utility/dspinst.h"

#define COEF_HPF_DCBLOCK    1048300  // DC Removal filter coefficient in S12.19

DMAMEM 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;
int32_t AudioInputAnalog::hpf_x1 = 0;

bool AudioInputAnalog::update_responsibility = false;
DMAChannel AudioInputAnalog::dma(false);

void AudioInputAnalog::init(uint8_t pin)
{
    int32_t tmp;

	// 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);
#else
	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) << 4);
    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
	  || 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;
	}
#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);
	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;
#endif
	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;

#if defined(KINETISK)
	daddr = (uint32_t)(dma.TCD->DADDR);
#endif
	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;
	uint32_t 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();

    //
	// 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)
    //  UNITY = 2^20
    //  fc = 2
    //  fs = 44100
    //
	p = out_left->data;
	end = p + AUDIO_BLOCK_SAMPLES;
	do {
		tmp = (uint16_t)(*p);
        tmp = ( ((int32_t) tmp) << 4);
        int32_t acc = tmp;
        acc += hpf_y1;
        acc -= hpf_x1;
        hpf_y1 = FRACMUL_SHL(acc, COEF_HPF_DCBLOCK, 11);
        hpf_x1 = tmp;
		s = signed_saturate_rshift(hpf_y1, 16, 4);
		*p++ = s;
	} while (p < end);

	// then transmit the AC data
	transmit(out_left);
	release(out_left);
}