/* 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_adcs.h" #include "utility/pdb.h" #include "utility/dspinst.h" #if defined(__MK20DX256__) || defined(__MK64FX512__) || defined(__MK66FX1M0__) 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::left_dc_average_hist[16]; int32_t AudioInputAnalogStereo::right_dc_average_hist[16]; int32_t AudioInputAnalogStereo::current_dc_average_index = 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 i, sum0=0, sum1=0; //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 // Actually, do many normal reads, to start with a nice DC level for (i=0; i < 1024; i++) { sum0 += analogRead(pin0); sum1 += analogReadADC1(pin1); } for (i = 0; i < 16; i++) { left_dc_average_hist[i] = sum0 >> 10; right_dc_average_hist[i] = sum1 >> 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; 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; uint32_t i, dc; 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(); // Find and subtract DC offset... We use an average of the // last 16 * AUDIO_BLOCK_SAMPLES samples. dc = 0; for (i = 0; i < 16; i++) { dc += left_dc_average_hist[i]; } dc /= 16 * AUDIO_BLOCK_SAMPLES; left_dc_average_hist[current_dc_average_index] = 0; p = out_left->data; end = p + AUDIO_BLOCK_SAMPLES; do { left_dc_average_hist[current_dc_average_index] += (uint16_t)(*p); tmp = (uint16_t)(*p) - (int32_t)dc; s = signed_saturate_rshift(tmp, 16, 0); *p++ = s; } while (p < end); dc = 0; for (i = 0; i < 16; i++) { dc += right_dc_average_hist[i]; } dc /= 16 * AUDIO_BLOCK_SAMPLES; right_dc_average_hist[current_dc_average_index] = 0; p = out_right->data; end = p + AUDIO_BLOCK_SAMPLES; do { right_dc_average_hist[current_dc_average_index] += (uint16_t)(*p); tmp = (uint16_t)(*p) - (int32_t)dc; s = signed_saturate_rshift(tmp, 16, 0); *p++ = s; } while (p < end); current_dc_average_index = (current_dc_average_index + 1) % 16; // 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