/* 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" DMAMEM static uint16_t analog_rx_buffer[AUDIO_BLOCK_SAMPLES]; audio_block_t * AudioInputAnalog::block_left = NULL; uint16_t AudioInputAnalog::block_offset = 0; bool AudioInputAnalog::update_responsibility = false; // #define PDB_CONFIG (PDB_SC_TRGSEL(15) | PDB_SC_PDBEN | PDB_SC_CONT) // #define PDB_PERIOD 1087 // 48e6 / 44100 void AudioInputAnalog::begin(unsigned int pin) { uint32_t i, sum=0; // pin must be 0 to 13 (for A0 to A13) // or 14 to 23 for digital pin numbers A0-A9 // or 34 to 37 corresponding to A10-A13 if (pin > 23 && !(pin >= 34 && pin <= 37)) return; //pinMode(2, OUTPUT); //pinMode(3, OUTPUT); //digitalWriteFast(3, HIGH); //delayMicroseconds(500); //digitalWriteFast(3, LOW); // 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 //analogReference(DEFAULT); // range 0 to 3.3 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; // testing only, enable adc interrupt //ADC0_SC1A |= ADC_SC1_AIEN; //while ((ADC0_SC1A & ADC_SC1_COCO) == 0) ; // wait //NVIC_ENABLE_IRQ(IRQ_ADC0); // set the programmable delay block to trigger the ADC at 44.1 kHz SIM_SCGC6 |= SIM_SCGC6_PDB; 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 SIM_SCGC7 |= SIM_SCGC7_DMA; SIM_SCGC6 |= SIM_SCGC6_DMAMUX; DMA_CR = 0; DMA_TCD2_SADDR = &ADC0_RA; DMA_TCD2_SOFF = 0; DMA_TCD2_ATTR = DMA_TCD_ATTR_SSIZE(1) | DMA_TCD_ATTR_DSIZE(1); DMA_TCD2_NBYTES_MLNO = 2; DMA_TCD2_SLAST = 0; DMA_TCD2_DADDR = analog_rx_buffer; DMA_TCD2_DOFF = 2; DMA_TCD2_CITER_ELINKNO = sizeof(analog_rx_buffer) / 2; DMA_TCD2_DLASTSGA = -sizeof(analog_rx_buffer); DMA_TCD2_BITER_ELINKNO = sizeof(analog_rx_buffer) / 2; DMA_TCD2_CSR = DMA_TCD_CSR_INTHALF | DMA_TCD_CSR_INTMAJOR; DMAMUX0_CHCFG2 = DMAMUX_DISABLE; DMAMUX0_CHCFG2 = DMAMUX_SOURCE_ADC0 | DMAMUX_ENABLE; update_responsibility = update_setup(); DMA_SERQ = 2; NVIC_ENABLE_IRQ(IRQ_DMA_CH2); } void dma_ch2_isr(void) { uint32_t daddr, offset; const uint16_t *src, *end; uint16_t *dest_left; audio_block_t *left; //digitalWriteFast(3, HIGH); daddr = (uint32_t)DMA_TCD2_DADDR; DMA_CINT = 2; 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 (AudioInputAnalog::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 = AudioInputAnalog::block_left; if (left != NULL) { offset = AudioInputAnalog::block_offset; if (offset > AUDIO_BLOCK_SAMPLES/2) offset = AUDIO_BLOCK_SAMPLES/2; //if (offset <= AUDIO_BLOCK_SAMPLES/2) { dest_left = (uint16_t *)&(left->data[offset]); AudioInputAnalog::block_offset = offset + AUDIO_BLOCK_SAMPLES/2; do { *dest_left++ = *src++; } while (src < end); //} } //digitalWriteFast(3, LOW); } #if 0 void adc0_isr(void) { uint32_t tmp = ADC0_RA; // read ADC result to clear interrupt digitalWriteFast(3, HIGH); delayMicroseconds(1); digitalWriteFast(3, LOW); } #endif void AudioInputAnalog::update(void) { audio_block_t *new_left=NULL, *out_left=NULL; unsigned int dc, offset; int16_t s, *p, *end; // 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 { s = (uint16_t)(*p) - dc; // TODO: should be saturating subtract *p++ = s; dc += s >> 13; // approx 5.38 Hz high pass filter } while (p < end); dc_average = dc; // then transmit the AC data transmit(out_left); release(out_left); }