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/* Audio Library for Teensy 3.X |
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* Copyright (c) 2016, Paul Stoffregen, paul@pjrc.com |
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* |
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* Development of this audio library was funded by PJRC.COM, LLC by sales of |
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* Teensy and Audio Adaptor boards. Please support PJRC's efforts to develop |
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* open source software by purchasing Teensy or other PJRC products. |
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* |
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* Permission is hereby granted, free of charge, to any person obtaining a copy |
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* of this software and associated documentation files (the "Software"), to deal |
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* in the Software without restriction, including without limitation the rights |
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* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell |
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* copies of the Software, and to permit persons to whom the Software is |
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* furnished to do so, subject to the following conditions: |
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* |
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* The above copyright notice, development funding notice, and this permission |
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* notice shall be included in all copies or substantial portions of the Software. |
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* |
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR |
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, |
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE |
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER |
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, |
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN |
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* THE SOFTWARE. |
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*/ |
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#include "output_dacs.h" |
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#include "utility/pdb.h" |
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#if defined(__MK64FX512__) || defined(__MK66FX1M0__) |
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DMAMEM static uint32_t dac_buffer[AUDIO_BLOCK_SAMPLES*2]; |
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audio_block_t * AudioOutputAnalogStereo::block_left_1st = NULL; |
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audio_block_t * AudioOutputAnalogStereo::block_left_2nd = NULL; |
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audio_block_t * AudioOutputAnalogStereo::block_right_1st = NULL; |
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audio_block_t * AudioOutputAnalogStereo::block_right_2nd = NULL; |
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audio_block_t AudioOutputAnalogStereo::block_silent; |
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bool AudioOutputAnalogStereo::update_responsibility = false; |
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DMAChannel AudioOutputAnalogStereo::dma(false); |
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void AudioOutputAnalogStereo::begin(void) |
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{ |
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dma.begin(true); // Allocate the DMA channel first |
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SIM_SCGC2 |= SIM_SCGC2_DAC0 | SIM_SCGC2_DAC1; |
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DAC0_C0 = DAC_C0_DACEN; // 1.2V VDDA is DACREF_2 |
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DAC1_C0 = DAC_C0_DACEN; |
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memset(&block_silent, 0, sizeof(block_silent)); |
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// slowly ramp up to DC voltage, approx 1/4 second |
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for (int16_t i=0; i<2048; i+=8) { |
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*(int16_t *)&(DAC0_DAT0L) = i; |
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*(int16_t *)&(DAC1_DAT0L) = i; |
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delay(1); |
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} |
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// set the programmable delay block to trigger DMA requests |
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if (!(SIM_SCGC6 & SIM_SCGC6_PDB) |
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|| (PDB0_SC & PDB_CONFIG) != PDB_CONFIG |
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|| PDB0_MOD != PDB_PERIOD |
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|| PDB0_IDLY != 1 |
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|| PDB0_CH0C1 != 0x0101) { |
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SIM_SCGC6 |= SIM_SCGC6_PDB; |
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PDB0_IDLY = 1; |
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PDB0_MOD = PDB_PERIOD; |
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PDB0_SC = PDB_CONFIG | PDB_SC_LDOK; |
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PDB0_SC = PDB_CONFIG | PDB_SC_SWTRIG; |
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PDB0_CH0C1 = 0x0101; |
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} |
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dma.TCD->SADDR = dac_buffer; |
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dma.TCD->SOFF = 2; |
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dma.TCD->ATTR = DMA_TCD_ATTR_SSIZE(DMA_TCD_ATTR_SIZE_16BIT) | |
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DMA_TCD_ATTR_DSIZE(DMA_TCD_ATTR_SIZE_16BIT); |
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dma.TCD->NBYTES_MLNO = DMA_TCD_NBYTES_MLOFFYES_NBYTES(4) | DMA_TCD_NBYTES_DMLOE | |
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DMA_TCD_NBYTES_MLOFFYES_MLOFF((&DAC0_DAT0L - &DAC1_DAT0L) * 2); |
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dma.TCD->SLAST = -sizeof(dac_buffer); |
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dma.TCD->DADDR = &DAC0_DAT0L; |
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dma.TCD->DOFF = &DAC1_DAT0L - &DAC0_DAT0L; |
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dma.TCD->CITER_ELINKNO = sizeof(dac_buffer) / 4; |
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dma.TCD->DLASTSGA = (&DAC0_DAT0L - &DAC1_DAT0L) * 2; |
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dma.TCD->BITER_ELINKNO = sizeof(dac_buffer) / 4; |
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dma.TCD->CSR = DMA_TCD_CSR_INTHALF | DMA_TCD_CSR_INTMAJOR; |
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dma.triggerAtHardwareEvent(DMAMUX_SOURCE_PDB); |
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update_responsibility = update_setup(); |
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dma.enable(); |
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dma.attachInterrupt(isr); |
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} |
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void AudioOutputAnalogStereo::analogReference(int ref) |
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{ |
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// TODO: this should ramp gradually to the new DC level |
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if (ref == INTERNAL) { |
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DAC0_C0 &= ~DAC_C0_DACRFS; // 1.2V |
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} else { |
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DAC0_C0 |= DAC_C0_DACRFS; // 3.3V |
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} |
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} |
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void AudioOutputAnalogStereo::update(void) |
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{ |
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audio_block_t *block_left, *block_right; |
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block_left = receiveReadOnly(0); // input 0 |
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block_right = receiveReadOnly(1); // input 1 |
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__disable_irq(); |
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if (block_left) { |
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if (block_left_1st == NULL) { |
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block_left_1st = block_left; |
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block_left = NULL; |
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} else if (block_left_2nd == NULL) { |
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block_left_2nd = block_left; |
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block_left = NULL; |
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} else { |
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audio_block_t *tmp = block_left_1st; |
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block_left_1st = block_left_2nd; |
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block_left_2nd = block_left; |
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block_left = tmp; |
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} |
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} |
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if (block_right) { |
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if (block_right_1st == NULL) { |
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block_right_1st = block_right; |
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block_right = NULL; |
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} else if (block_right_2nd == NULL) { |
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block_right_2nd = block_right; |
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block_right = NULL; |
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} else { |
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audio_block_t *tmp = block_right_1st; |
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block_right_1st = block_right_2nd; |
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block_right_2nd = block_right; |
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block_right = tmp; |
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} |
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} |
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__enable_irq(); |
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if (block_left) release(block_left); |
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if (block_right) release(block_right); |
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} |
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// TODO: the DAC has much higher bandwidth than the datasheet says |
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// can we output a 2X oversampled output, for easier filtering? |
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void AudioOutputAnalogStereo::isr(void) |
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{ |
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const uint32_t *src_left, *src_right, *end; |
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uint32_t *dest; |
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audio_block_t *block_left, *block_right; |
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uint32_t saddr; |
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saddr = (uint32_t)(dma.TCD->SADDR); |
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dma.clearInterrupt(); |
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if (saddr < (uint32_t)dac_buffer + sizeof(dac_buffer) / 2) { |
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// DMA is transmitting the first half of the buffer |
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// so we must fill the second half |
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dest = dac_buffer + AUDIO_BLOCK_SAMPLES; |
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end = dac_buffer + AUDIO_BLOCK_SAMPLES*2; |
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} else { |
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// DMA is transmitting the second half of the buffer |
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// so we must fill the first half |
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dest = dac_buffer; |
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end = dac_buffer + AUDIO_BLOCK_SAMPLES; |
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} |
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block_left = block_left_1st; |
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if (!block_left) block_left = &block_silent; |
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block_right = block_right_1st; |
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if (!block_right) block_right = &block_silent; |
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src_left = (const uint32_t *)(block_left->data); |
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src_right = (const uint32_t *)(block_right->data); |
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do { |
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// TODO: can this be optimized? |
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uint32_t left = *src_left++; |
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uint32_t right = *src_right++; |
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uint32_t out1 = ((left & 0xFFFF) + 32767) >> 4; |
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out1 |= (((right & 0xFFFF) + 32767) >> 4) << 16; |
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uint32_t out2 = ((left >> 16) + 32767) >> 4; |
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out2 |= (((right >> 16) + 32767) >> 4) << 16; |
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*dest++ = out1; |
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*dest++ = out2; |
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} while (dest < end); |
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if (block_left != &block_silent) { |
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release(block_left); |
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block_left_1st = block_left_2nd; |
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block_left_2nd = NULL; |
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} |
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if (block_right != &block_silent) { |
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release(block_right); |
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block_left_1st = block_left_2nd; |
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block_left_2nd = NULL; |
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} |
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if (update_responsibility) update_all(); |
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} |
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#else // not __MK64FX512__ or __MK66FX1M0__ |
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void AudioOutputAnalogStereo::begin(void) |
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{ |
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} |
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void AudioOutputAnalogStereo::update(void) |
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{ |
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audio_block_t *block; |
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block = receiveReadOnly(0); // input 0 |
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if (block) release(block); |
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block = receiveReadOnly(1); // input 1 |
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if (block) release(block); |
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} |
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#endif |
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PaulStoffregen
8 年之前