Merge pull request #14 from IDC-Dragon/master
high resolution for I2S input/output, added S/PDIF input/output with configurable sample rate and 24 bitpull/16/head
Bob Larkin
3 years ago
committed by
GitHub
commit
e1a2fcce5b
@ -0,0 +1,431 @@ |
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/* Audio Library for Teensy 3.X
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* Copyright (c) 2019, 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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/*
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by Alexander Walch |
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*/ |
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#if defined(__IMXRT1062__) |
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#include "async_input_spdif3_F32.h" |
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#include "output_spdif3_F32.h" |
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#include "biquad.h" |
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#include <utility/imxrt_hw.h> |
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//Parameters
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namespace { |
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#define SPDIF_RX_BUFFER_LENGTH AUDIO_BLOCK_SAMPLES |
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const int32_t bufferLength=8*AUDIO_BLOCK_SAMPLES; |
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const uint16_t noSamplerPerIsr=SPDIF_RX_BUFFER_LENGTH/4; |
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const float toFloatAudio= (float)(1./pow(2., 23.)); |
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} |
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// dummy class, no quantization
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class Scaler_F32 { |
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public: |
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Scaler_F32() { |
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_factor = 1.0; |
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}; |
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void configure() { |
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}; |
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void quantize(float* input, float32_t* output, uint16_t length) { |
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memcpy(output, input, length * sizeof(float)); |
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/*
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for (uint16_t i =0; i< length; i++){ |
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*output++ = *input++ * _factor; |
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} |
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*/ |
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}; |
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private: |
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float _factor; |
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}; |
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#ifdef DEBUG_SPDIF_IN |
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volatile bool AsyncAudioInputSPDIF3_F32::bufferOverflow=false; |
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#endif |
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volatile uint32_t AsyncAudioInputSPDIF3_F32::microsLast; |
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DMAMEM __attribute__((aligned(32))) |
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static int32_t spdif_rx_buffer[SPDIF_RX_BUFFER_LENGTH]; |
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static float bufferR[bufferLength]; |
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static float bufferL[bufferLength]; |
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volatile int32_t AsyncAudioInputSPDIF3_F32::buffer_offset = 0; // read by resample/ written in spdif input isr -> copied at the beginning of 'resmaple' protected by __disable_irq() in resample
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int32_t AsyncAudioInputSPDIF3_F32::resample_offset = 0; // read/written by resample/ read in spdif input isr -> no protection needed?
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float AsyncAudioInputSPDIF3_F32::sample_rate_Hz = AUDIO_SAMPLE_RATE_EXACT; |
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DMAChannel AsyncAudioInputSPDIF3_F32::dma(false); |
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AsyncAudioInputSPDIF3_F32::~AsyncAudioInputSPDIF3_F32(){ |
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delete [] _bufferLPFilter.pCoeffs; |
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delete [] _bufferLPFilter.pState; |
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delete quantizer[0]; |
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delete quantizer[1]; |
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} |
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FLASHMEM |
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AsyncAudioInputSPDIF3_F32::AsyncAudioInputSPDIF3_F32(const AudioSettings_F32 &settings, float attenuation, int32_t minHalfFilterLength, int32_t maxHalfFilterLength): |
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AudioStream_F32(0, NULL), |
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_resampler(attenuation, minHalfFilterLength, maxHalfFilterLength) |
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{ |
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sample_rate_Hz = settings.sample_rate_Hz; |
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quantizer[0]=new Scaler_F32(); |
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quantizer[0]->configure(); |
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quantizer[1]=new Scaler_F32(); |
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quantizer[1]->configure(); |
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begin(); |
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} |
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FLASHMEM |
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void AsyncAudioInputSPDIF3_F32::begin() |
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{ |
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AudioOutputSPDIF3_F32::config_spdif3(sample_rate_Hz); |
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dma.begin(true); // Allocate the DMA channel first
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const uint32_t noByteMinorLoop=2*4; |
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dma.TCD->SOFF = 4; |
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dma.TCD->ATTR = DMA_TCD_ATTR_SSIZE(2) | DMA_TCD_ATTR_DSIZE(2); |
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dma.TCD->NBYTES_MLNO = DMA_TCD_NBYTES_MLOFFYES_NBYTES(noByteMinorLoop) | DMA_TCD_NBYTES_SMLOE |
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DMA_TCD_NBYTES_MLOFFYES_MLOFF(-8); |
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dma.TCD->SLAST = -8; |
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dma.TCD->DOFF = 4; |
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dma.TCD->CITER_ELINKNO = sizeof(spdif_rx_buffer) / noByteMinorLoop; |
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dma.TCD->DLASTSGA = -sizeof(spdif_rx_buffer); |
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dma.TCD->BITER_ELINKNO = sizeof(spdif_rx_buffer) / noByteMinorLoop; |
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dma.TCD->CSR = DMA_TCD_CSR_INTHALF | DMA_TCD_CSR_INTMAJOR;
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dma.TCD->SADDR = (void *)((uint32_t)&SPDIF_SRL); |
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dma.TCD->DADDR = spdif_rx_buffer; |
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dma.triggerAtHardwareEvent(DMAMUX_SOURCE_SPDIF_RX); |
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//SPDIF_SCR |=SPDIF_SCR_DMA_RX_EN; //DMA Receive Request Enable
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dma.enable(); |
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dma.attachInterrupt(isr); |
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#ifdef DEBUG_SPDIF_IN |
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while (!Serial); |
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#endif |
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_bufferLPFilter.pCoeffs=new float[5]; |
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_bufferLPFilter.numStages=1; |
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_bufferLPFilter.pState=new float[2]; |
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getCoefficients(_bufferLPFilter.pCoeffs, BiquadType::LOW_PASS, 0., 5., sample_rate_Hz/AUDIO_BLOCK_SAMPLES, 0.5); |
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SPDIF_SCR &=(~SPDIF_SCR_RXFIFO_OFF_ON); //receive fifo is turned on again
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SPDIF_SRCD = 0; |
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SPDIF_SCR |= SPDIF_SCR_DMA_RX_EN; |
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CORE_PIN15_CONFIG = 3; |
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IOMUXC_SPDIF_IN_SELECT_INPUT = 0; // GPIO_AD_B1_03_ALT3
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} |
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bool AsyncAudioInputSPDIF3_F32::isLocked() { |
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return (SPDIF_SRPC & SPDIF_SRPC_LOCK) == SPDIF_SRPC_LOCK; |
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} |
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void AsyncAudioInputSPDIF3_F32::resample(float32_t* data_left, float32_t* data_right, int32_t& block_offset){ |
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block_offset=0; |
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if(!_resampler.initialized() || !isLocked()){ |
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return; |
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} |
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int32_t bOffset=buffer_offset; |
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int32_t resOffset=resample_offset; |
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uint16_t inputBufferStop = bOffset >= resOffset ? bOffset-resOffset : bufferLength-resOffset; |
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if (inputBufferStop==0){ |
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return; |
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} |
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uint16_t processedLength; |
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uint16_t outputCount=0; |
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uint16_t outputLength=AUDIO_BLOCK_SAMPLES; |
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float resampledBufferL[AUDIO_BLOCK_SAMPLES]; |
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float resampledBufferR[AUDIO_BLOCK_SAMPLES]; |
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_resampler.resample(&bufferL[resOffset],&bufferR[resOffset], inputBufferStop, processedLength, resampledBufferL, resampledBufferR, outputLength, outputCount); |
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resOffset=(resOffset+processedLength)%bufferLength; |
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block_offset=outputCount;
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if (bOffset > resOffset && block_offset< AUDIO_BLOCK_SAMPLES){ |
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inputBufferStop= bOffset-resOffset; |
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outputLength=AUDIO_BLOCK_SAMPLES-block_offset; |
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_resampler.resample(&bufferL[resOffset],&bufferR[resOffset], inputBufferStop, processedLength, resampledBufferL+block_offset, resampledBufferR+block_offset, outputLength, outputCount); |
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resOffset=(resOffset+processedLength)%bufferLength; |
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block_offset+=outputCount; |
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} |
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quantizer[0]->quantize(resampledBufferL, data_left, block_offset); // TODO: degenerated to a copy, perhaps directly resample into here?
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quantizer[1]->quantize(resampledBufferR, data_right, block_offset); |
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__disable_irq(); |
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resample_offset=resOffset; |
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__enable_irq();
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} |
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void AsyncAudioInputSPDIF3_F32::isr(void) |
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{ |
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dma.clearInterrupt(); |
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microsLast=micros(); |
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const int32_t *src, *end; |
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uint32_t daddr = (uint32_t)(dma.TCD->DADDR); |
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if (daddr < (uint32_t)spdif_rx_buffer + sizeof(spdif_rx_buffer) / 2) { |
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// DMA is receiving to the first half of the buffer
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// need to remove data from the second half
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src = (int32_t *)&spdif_rx_buffer[SPDIF_RX_BUFFER_LENGTH/2]; |
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end = (int32_t *)&spdif_rx_buffer[SPDIF_RX_BUFFER_LENGTH]; |
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//if (AsyncAudioInputSPDIF3_F32::update_responsibility) AudioStream::update_all();
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} else { |
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// DMA is receiving to the second half of the buffer
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// need to remove data from the first half
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src = (int32_t *)&spdif_rx_buffer[0]; |
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end = (int32_t *)&spdif_rx_buffer[SPDIF_RX_BUFFER_LENGTH/2]; |
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} |
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if (buffer_offset >=resample_offset || |
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(buffer_offset + SPDIF_RX_BUFFER_LENGTH/4) < resample_offset) { |
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#if IMXRT_CACHE_ENABLED >=1 |
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arm_dcache_delete((void*)src, sizeof(spdif_rx_buffer) / 2); |
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#endif |
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float *destR = &(bufferR[buffer_offset]); |
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float *destL = &(bufferL[buffer_offset]); |
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do {
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int32_t n=(*src) & 0x800000 ? (*src)|0xFF800000 : (*src) & 0xFFFFFF; |
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*destL++ = (float)(n)*toFloatAudio; |
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++src; |
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n=(*src) & 0x800000 ? (*src)|0xFF800000 : (*src) & 0xFFFFFF; |
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*destR++ = (float)(n)*toFloatAudio; |
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++src; |
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} while (src < end); |
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buffer_offset=(buffer_offset+SPDIF_RX_BUFFER_LENGTH/4)%bufferLength; |
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} |
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#ifdef DEBUG_SPDIF_IN |
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else { |
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bufferOverflow=true; |
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} |
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#endif |
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} |
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double AsyncAudioInputSPDIF3_F32::getNewValidInputFrequ(){ |
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//page 2129: FrequMeas[23:0]=FreqMeas_CLK / BUS_CLK * 2^10 * GAIN
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if (isLocked()){ |
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const double f=(double)F_BUS_ACTUAL/(1048576.*(double)AudioOutputSPDIF3_F32::dpll_Gain()*128.);// bit clock = 128 * sampling frequency
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const double freqMeas=(double)(SPDIF_SRFM & 0xFFFFFF)*f; |
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if (_lastValidInputFrequ != freqMeas){//frequency not stable yet;
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_lastValidInputFrequ=freqMeas; |
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return -1.;
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} |
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return _lastValidInputFrequ; |
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} |
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return -1.; |
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} |
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double AsyncAudioInputSPDIF3_F32::getBufferedTime() const{ |
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__disable_irq(); |
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double n=_bufferedTime; |
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__enable_irq(); |
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return n; |
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} |
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void AsyncAudioInputSPDIF3_F32::configure(){ |
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if(!isLocked()){ |
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_resampler.reset(); |
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return; |
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} |
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#ifdef DEBUG_SPDIF_IN |
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const bool bOverf=bufferOverflow; |
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bufferOverflow=false; |
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if (bOverf){ |
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Serial.print("buffer overflow, buffer offset: "); |
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Serial.print(buffer_offset); |
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Serial.print(", resample_offset: "); |
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Serial.println(resample_offset); |
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if (!_resampler.initialized()){ |
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Serial.println("_resampler not initialized. "); |
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} |
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} |
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#endif |
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const double inputF=getNewValidInputFrequ(); //returns: -1 ... invalid frequency
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if (inputF > 0.){ |
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//we got a valid sample frequency
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const double frequDiff=inputF/_inputFrequency-1.; |
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if (abs(frequDiff) > 0.01 || !_resampler.initialized()){ |
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//the new sample frequency differs from the last one -> configure the _resampler again
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_inputFrequency=inputF;
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_targetLatencyS=max(0.001,(noSamplerPerIsr*3./2./_inputFrequency)); |
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_maxLatency=max(2.*_blockDuration, 2*noSamplerPerIsr/_inputFrequency); |
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const int32_t targetLatency=round(_targetLatencyS*inputF); |
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__disable_irq(); |
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resample_offset = targetLatency <= buffer_offset ? buffer_offset - targetLatency : bufferLength -(targetLatency-buffer_offset); |
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__enable_irq(); |
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_resampler.configure(inputF, sample_rate_Hz); |
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#ifdef DEBUG_SPDIF_IN |
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Serial.print("_maxLatency: "); |
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Serial.println(_maxLatency); |
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Serial.print("targetLatency: "); |
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Serial.println(targetLatency); |
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Serial.print("relative frequ diff: "); |
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Serial.println(frequDiff, 8); |
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Serial.print("configure _resampler with frequency "); |
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Serial.println(inputF,8);
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#endif |
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} |
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} |
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} |
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void AsyncAudioInputSPDIF3_F32::monitorResampleBuffer(){ |
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if(!_resampler.initialized()){ |
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return; |
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} |
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__disable_irq(); |
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const double dmaOffset=(micros()-microsLast)*1e-6; //[seconds]
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double bTime = resample_offset <= buffer_offset ? (buffer_offset-resample_offset-_resampler.getXPos())/_lastValidInputFrequ+dmaOffset : (bufferLength-resample_offset +buffer_offset-_resampler.getXPos())/_lastValidInputFrequ+dmaOffset; //[seconds]
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double diff = bTime- (_blockDuration+ _targetLatencyS); //seconds
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biquad_cascade_df2T<double, arm_biquad_cascade_df2T_instance_f32, float>(&_bufferLPFilter, &diff, &diff, 1); |
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bool settled=_resampler.addToSampleDiff(diff); |
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if (bTime > _maxLatency || bTime-dmaOffset<= _blockDuration || settled) {
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double distance=(_blockDuration+_targetLatencyS-dmaOffset)*_lastValidInputFrequ+_resampler.getXPos(); |
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diff=0.; |
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if (distance > bufferLength-noSamplerPerIsr){ |
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diff=bufferLength-noSamplerPerIsr-distance; |
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distance=bufferLength-noSamplerPerIsr; |
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} |
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if (distance < 0.){ |
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distance=0.; |
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diff=- (_blockDuration+ _targetLatencyS); |
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} |
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double resample_offsetF=buffer_offset-distance; |
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resample_offset=(int32_t)floor(resample_offsetF); |
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_resampler.addToPos(resample_offsetF-resample_offset); |
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while (resample_offset<0){ |
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resample_offset+=bufferLength; |
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}
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#ifdef DEBUG_SPDIF_IN |
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double bTimeFixed = resample_offset <= buffer_offset ? (buffer_offset-resample_offset-_resampler.getXPos())/_lastValidInputFrequ+dmaOffset : (bufferLength-resample_offset +buffer_offset-_resampler.getXPos())/_lastValidInputFrequ+dmaOffset; //[seconds]
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#endif |
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__enable_irq(); |
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#ifdef DEBUG_SPDIF_IN |
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Serial.print("settled: "); |
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Serial.println(settled); |
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Serial.print("bTime: "); |
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Serial.println(bTime*1e6,3); |
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Serial.print("_maxLatency: "); |
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Serial.println(_maxLatency*1e6,3); |
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Serial.print("bTime-dmaOffset: "); |
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Serial.println((bTime-dmaOffset)*1e6,3); |
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Serial.print(", _blockDuration: "); |
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Serial.println(_blockDuration*1e6,3); |
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Serial.print("bTimeFixed: "); |
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Serial.println(bTimeFixed*1e6,3); |
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#endif |
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preload(&_bufferLPFilter, (float)diff); |
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_resampler.fixStep();
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} |
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else {
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__enable_irq(); |
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} |
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_bufferedTime=_targetLatencyS+diff; |
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} |
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void AsyncAudioInputSPDIF3_F32::update(void) |
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{ |
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configure(); |
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monitorResampleBuffer(); //important first call 'monitorResampleBuffer' then 'resample'
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audio_block_f32_t *block_left = allocate_f32(); |
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audio_block_f32_t *block_right = nullptr; |
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if (block_left!= nullptr) { |
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block_right = allocate_f32(); |
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if (block_right == nullptr) { |
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release(block_left); |
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block_left = nullptr; |
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} |
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} |
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if (block_left && block_right) { |
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int32_t block_offset; |
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resample(block_left->data, block_right->data,block_offset); |
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if(block_offset < AUDIO_BLOCK_SAMPLES){ |
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memset(block_left->data+block_offset, 0, (AUDIO_BLOCK_SAMPLES-block_offset)*sizeof(float32_t));
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memset(block_right->data+block_offset, 0, (AUDIO_BLOCK_SAMPLES-block_offset)*sizeof(float32_t));
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#ifdef DEBUG_SPDIF_IN |
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Serial.print("filled only "); |
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Serial.print(block_offset); |
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Serial.println(" samples."); |
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#endif |
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} |
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transmit(block_left, 0); |
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release(block_left); |
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block_left=nullptr; |
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transmit(block_right, 1); |
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release(block_right); |
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block_right=nullptr;
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} |
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#ifdef DEBUG_SPDIF_IN |
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else {
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Serial.println("Not enough blocks available. Too few audio memory?"); |
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} |
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#endif |
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} |
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double AsyncAudioInputSPDIF3_F32::getInputFrequency() const{ |
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__disable_irq(); |
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double f=_lastValidInputFrequ; |
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__enable_irq(); |
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return isLocked() ? f : 0.; |
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} |
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double AsyncAudioInputSPDIF3_F32::getTargetLantency() const { |
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__disable_irq(); |
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double l=_targetLatencyS; |
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__enable_irq(); |
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return l ; |
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} |
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double AsyncAudioInputSPDIF3_F32::getAttenuation() const{ |
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return _resampler.getAttenuation(); |
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} |
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int32_t AsyncAudioInputSPDIF3_F32::getHalfFilterLength() const{ |
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return _resampler.getHalfFilterLength(); |
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} |
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#endif // __IMXRT1062__
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#if defined(__MK66FX1M0__) || defined(__MK64FX512__) || defined(__MK20DX256__) || defined(__MKL26Z64__) |
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// empty code to allow compile (but no sound input) on other Teensy models
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#include "async_input_spdif3.h" |
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AsyncAudioInputSPDIF3_F32::AsyncAudioInputSPDIF3_F32(bool dither, bool noiseshaping,float attenuation, int32_t minHalfFilterLength, int32_t maxHalfFilterLength): |
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AudioStream(0, NULL), _resampler(attenuation, minHalfFilterLength, maxHalfFilterLength) |
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{ } |
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void AsyncAudioInputSPDIF3_F32::begin() { } |
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void AsyncAudioInputSPDIF3_F32::update(void) { } |
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double AsyncAudioInputSPDIF3_F32::getBufferedTime() const { return 0; } |
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double AsyncAudioInputSPDIF3_F32::getInputFrequency() const { return 0; } |
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bool AsyncAudioInputSPDIF3_F32::isLocked() { return false; } |
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double AsyncAudioInputSPDIF3_F32::getTargetLantency() const { return 0; } |
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double AsyncAudioInputSPDIF3_F32::getAttenuation() const { return 0; } |
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int32_t AsyncAudioInputSPDIF3_F32::getHalfFilterLength() const { return 0; } |
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AsyncAudioInputSPDIF3_F32::~AsyncAudioInputSPDIF3_F32() { } |
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#endif |
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@ -0,0 +1,91 @@ |
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/* Audio Library for Teensy 3.X
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* Copyright (c) 2019, 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 |
||||
* 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. |
||||
*/ |
||||
/*
|
||||
by Alexander Walch |
||||
*/ |
||||
#ifndef async_input_spdif3_f32_h_ |
||||
#define async_input_spdif3_f32_h_ |
||||
#include "Resampler.h" |
||||
#include "Quantizer.h" |
||||
#include "Arduino.h" |
||||
#include "AudioStream_F32.h" |
||||
#include "DMAChannel.h" |
||||
#include <arm_math.h> |
||||
|
||||
//#define DEBUG_SPDIF_IN //activates debug output
|
||||
|
||||
class Scaler_F32; // internal
|
||||
|
||||
class AsyncAudioInputSPDIF3_F32 : public AudioStream_F32 |
||||
{ |
||||
public: |
||||
///@param attenuation target attenuation [dB] of the anti-aliasing filter. Only used if AUDIO_SAMPLE_RATE_EXACT < input sample rate (input fs). The attenuation can't be reached if the needed filter length exceeds 2*MAX_FILTER_SAMPLES+1
|
||||
///@param minHalfFilterLength If AUDIO_SAMPLE_RATE_EXACT >= input fs), the filter length of the resampling filter is 2*minHalfFilterLength+1. If AUDIO_SAMPLE_RATE_EXACT < input fs the filter is maybe longer to reach the desired attenuation
|
||||
///@param maxHalfFilterLength Can be used to restrict the maximum filter length at the cost of a lower attenuation
|
||||
AsyncAudioInputSPDIF3_F32(const AudioSettings_F32 &settings, float attenuation=100, int32_t minHalfFilterLength=20, int32_t maxHalfFilterLength=80); |
||||
~AsyncAudioInputSPDIF3_F32(); |
||||
void begin(); |
||||
virtual void update(void); |
||||
double getBufferedTime() const; |
||||
double getInputFrequency() const; |
||||
static bool isLocked(); |
||||
double getTargetLantency() const; |
||||
double getAttenuation() const; |
||||
int32_t getHalfFilterLength() const; |
||||
protected:
|
||||
static DMAChannel dma; |
||||
static void isr(void); |
||||
private: |
||||
void resample(float32_t* data_left, float32_t* data_right, int32_t& block_offset); |
||||
void monitorResampleBuffer(); |
||||
void configure(); |
||||
double getNewValidInputFrequ(); |
||||
void config_spdifIn(); |
||||
|
||||
//accessed in isr ====
|
||||
static volatile int32_t buffer_offset; |
||||
static int32_t resample_offset; |
||||
static volatile uint32_t microsLast; |
||||
//====================
|
||||
|
||||
Resampler _resampler; |
||||
Scaler_F32* quantizer[2]; |
||||
arm_biquad_cascade_df2T_instance_f32 _bufferLPFilter; |
||||
|
||||
volatile double _bufferedTime; |
||||
volatile double _lastValidInputFrequ; |
||||
double _inputFrequency=0.; |
||||
double _targetLatencyS; //target latency [seconds]
|
||||
const double _blockDuration=AUDIO_BLOCK_SAMPLES/AUDIO_SAMPLE_RATE_EXACT; //[seconds]
|
||||
double _maxLatency=2.*_blockDuration; |
||||
static float sample_rate_Hz; // configured output sample rate
|
||||
|
||||
#ifdef DEBUG_SPDIF_IN |
||||
static volatile bool bufferOverflow; |
||||
#endif |
||||
}; |
||||
|
||||
#endif |
||||
@ -0,0 +1,260 @@ |
||||
/* Audio Library for Teensy 3.X
|
||||
* Copyright (c) 2019, 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. |
||||
*/ |
||||
/*
|
||||
by Frank Bösing |
||||
*/ |
||||
|
||||
#if defined(__IMXRT1052__) || defined(__IMXRT1062__) |
||||
#include <Arduino.h> |
||||
#include "input_spdif3_F32.h" |
||||
#include "output_spdif3_F32.h" |
||||
#include "utility/imxrt_hw.h" |
||||
|
||||
// sign extend and scale
|
||||
static inline float32_t i24_to_f32(int32_t n) { |
||||
const float32_t scale = 1.0 / (1LL << 31); |
||||
int32_t leftaligned = (uint32_t)n << 8; // to avoid manual sign extension
|
||||
return scale * leftaligned; |
||||
} |
||||
|
||||
DMAMEM __attribute__((aligned(32))) |
||||
static uint32_t spdif_rx_buffer[AUDIO_BLOCK_SAMPLES * 4]; |
||||
audio_block_f32_t * AudioInputSPDIF3_F32::block_left = NULL; |
||||
audio_block_f32_t * AudioInputSPDIF3_F32::block_right = NULL; |
||||
uint16_t AudioInputSPDIF3_F32::block_offset = 0; |
||||
bool AudioInputSPDIF3_F32::update_responsibility = false; |
||||
DMAChannel AudioInputSPDIF3_F32::dma(false); |
||||
|
||||
FLASHMEM |
||||
void AudioInputSPDIF3_F32::begin(void) |
||||
{ |
||||
dma.begin(true); // Allocate the DMA channel first
|
||||
|
||||
AudioOutputSPDIF3_F32::config_spdif3(sample_rate_Hz); |
||||
|
||||
const int nbytes_mlno = 2 * 4; // 8 Bytes per minor loop
|
||||
dma.TCD->SADDR = &SPDIF_SRL; |
||||
dma.TCD->SOFF = 4; |
||||
dma.TCD->ATTR = DMA_TCD_ATTR_SSIZE(2) | DMA_TCD_ATTR_DSIZE(2); |
||||
dma.TCD->NBYTES_MLNO = DMA_TCD_NBYTES_MLOFFYES_NBYTES(nbytes_mlno) | DMA_TCD_NBYTES_SMLOE | |
||||
DMA_TCD_NBYTES_MLOFFYES_MLOFF(-8); |
||||
dma.TCD->SLAST = -8; |
||||
dma.TCD->DADDR = spdif_rx_buffer; |
||||
dma.TCD->DOFF = 4; |
||||
dma.TCD->DLASTSGA = -sizeof(spdif_rx_buffer); |
||||
dma.TCD->CITER_ELINKNO = sizeof(spdif_rx_buffer) / nbytes_mlno; |
||||
dma.TCD->BITER_ELINKNO = sizeof(spdif_rx_buffer) / nbytes_mlno; |
||||
dma.TCD->CSR = DMA_TCD_CSR_INTHALF | DMA_TCD_CSR_INTMAJOR; |
||||
|
||||
dma.triggerAtHardwareEvent(DMAMUX_SOURCE_SPDIF_RX); |
||||
update_responsibility = update_setup(); |
||||
dma.attachInterrupt(isr); |
||||
dma.enable(); |
||||
|
||||
SPDIF_SRCD = 0; |
||||
SPDIF_SCR |= SPDIF_SCR_DMA_RX_EN; |
||||
CORE_PIN15_CONFIG = 3; |
||||
IOMUXC_SPDIF_IN_SELECT_INPUT = 0; // GPIO_AD_B1_03_ALT3
|
||||
//pinMode(13, OUTPUT);
|
||||
} |
||||
|
||||
void AudioInputSPDIF3_F32::isr(void) |
||||
{ |
||||
uint32_t daddr, offset; |
||||
const int32_t *src, *end; |
||||
float32_t *dest_left, *dest_right; |
||||
audio_block_f32_t *left, *right; |
||||
|
||||
dma.clearInterrupt(); |
||||
//digitalWriteFast(13, !digitalReadFast(13));
|
||||
if (AudioInputSPDIF3_F32::update_responsibility) AudioStream::update_all(); |
||||
|
||||
daddr = (uint32_t)(dma.TCD->DADDR); |
||||
|
||||
if (daddr < (uint32_t)spdif_rx_buffer + sizeof(spdif_rx_buffer) / 2) { |
||||
// DMA is receiving to the first half of the buffer
|
||||
// need to remove data from the second half
|
||||
src = (int32_t *)&spdif_rx_buffer[AUDIO_BLOCK_SAMPLES * 2]; |
||||
end = (int32_t *)&spdif_rx_buffer[AUDIO_BLOCK_SAMPLES * 4]; |
||||
} else { |
||||
// DMA is receiving to the second half of the buffer
|
||||
// need to remove data from the first half
|
||||
src = (int32_t *)&spdif_rx_buffer[0]; |
||||
end = (int32_t *)&spdif_rx_buffer[AUDIO_BLOCK_SAMPLES*2]; |
||||
} |
||||
|
||||
left = AudioInputSPDIF3_F32::block_left; |
||||
right = AudioInputSPDIF3_F32::block_right; |
||||
|
||||
if (left != NULL && right != NULL) { |
||||
offset = AudioInputSPDIF3_F32::block_offset; |
||||
if (offset <= AUDIO_BLOCK_SAMPLES*2) { |
||||
dest_left = &(left->data[offset]); |
||||
dest_right = &(right->data[offset]); |
||||
AudioInputSPDIF3_F32::block_offset = offset + AUDIO_BLOCK_SAMPLES*2; |
||||
|
||||
do { |
||||
#if IMXRT_CACHE_ENABLED >=1 |
||||
SCB_CACHE_DCIMVAC = (uintptr_t)src; |
||||
asm("dsb":::"memory"); |
||||
#endif |
||||
|
||||
*dest_left++ = i24_to_f32(*src++); |
||||
*dest_right++ = i24_to_f32(*src++); |
||||
|
||||
*dest_left++ = i24_to_f32(*src++); |
||||
*dest_right++ = i24_to_f32(*src++); |
||||
|
||||
*dest_left++ = i24_to_f32(*src++); |
||||
*dest_right++ = i24_to_f32(*src++); |
||||
|
||||
*dest_left++ = i24_to_f32(*src++); |
||||
*dest_right++ = i24_to_f32(*src++); |
||||
|
||||
} while (src < end); |
||||
} |
||||
} |
||||
else if (left != NULL) { |
||||
offset = AudioInputSPDIF3_F32::block_offset; |
||||
if (offset <= AUDIO_BLOCK_SAMPLES*2) { |
||||
dest_left = &(left->data[offset]); |
||||
AudioInputSPDIF3_F32::block_offset = offset + AUDIO_BLOCK_SAMPLES*2; |
||||
|
||||
do { |
||||
#if IMXRT_CACHE_ENABLED >=1 |
||||
SCB_CACHE_DCIMVAC = (uintptr_t)src; |
||||
asm("dsb":::"memory"); |
||||
#endif |
||||
|
||||
*dest_left++ = i24_to_f32(*src++); |
||||
src++; |
||||
|
||||
*dest_left++ = i24_to_f32(*src++); |
||||
src++; |
||||
|
||||
*dest_left++ = i24_to_f32(*src++); |
||||
src++; |
||||
|
||||
*dest_left++ = i24_to_f32(*src++); |
||||
src++; |
||||
|
||||
} while (src < end); |
||||
}
|
||||
} |
||||
else if (right != NULL) { |
||||
offset = AudioInputSPDIF3_F32::block_offset; |
||||
if (offset <= AUDIO_BLOCK_SAMPLES*2) { |
||||
dest_right = &(right->data[offset]); |
||||
AudioInputSPDIF3_F32::block_offset = offset + AUDIO_BLOCK_SAMPLES*2; |
||||
|
||||
do { |
||||
#if IMXRT_CACHE_ENABLED >=1 |
||||
SCB_CACHE_DCIMVAC = (uintptr_t)src; |
||||
asm("dsb":::"memory"); |
||||
#endif |
||||
|
||||
src++; |
||||
*dest_right++ = i24_to_f32(*src++); |
||||
|
||||
src++; |
||||
*dest_right++ = i24_to_f32(*src++); |
||||
|
||||
src++; |
||||
*dest_right++ = i24_to_f32(*src++); |
||||
|
||||
src++; |
||||
*dest_right++ = i24_to_f32(*src++); |
||||
|
||||
} while (src < end); |
||||
}
|
||||
} |
||||
|
||||
} |
||||
|
||||
|
||||
void AudioInputSPDIF3_F32::update(void) |
||||
{ |
||||
audio_block_f32_t *new_left=NULL, *new_right=NULL, *out_left=NULL, *out_right=NULL; |
||||
|
||||
// allocate 2 new blocks, but if one fails, allocate neither
|
||||
new_left = allocate_f32(); |
||||
if (new_left != NULL) { |
||||
new_right = allocate_f32(); |
||||
if (new_right == NULL) { |
||||
release(new_left); |
||||
new_left = NULL; |
||||
} |
||||
} |
||||
__disable_irq(); |
||||
if (block_offset >= AUDIO_BLOCK_SAMPLES) { |
||||
// the DMA filled 2 blocks, so grab them and get the
|
||||
// 2 new blocks to the DMA, as quickly as possible
|
||||
out_left = block_left; |
||||
block_left = new_left; |
||||
out_right = block_right; |
||||
block_right = new_right; |
||||
block_offset = 0; |
||||
__enable_irq(); |
||||
// then transmit the DMA's former blocks
|
||||
transmit(out_left, 0); |
||||
release(out_left); |
||||
transmit(out_right, 1); |
||||
release(out_right); |
||||
//Serial.print(".");
|
||||
} else if (new_left != NULL) { |
||||
// the DMA didn't fill blocks, but we allocated blocks
|
||||
if (block_left == NULL) { |
||||
// the DMA doesn't have any blocks to fill, so
|
||||
// give it the ones we just allocated
|
||||
block_left = new_left; |
||||
block_right = new_right; |
||||
block_offset = 0; |
||||
__enable_irq(); |
||||
} else { |
||||
// the DMA already has blocks, doesn't need these
|
||||
__enable_irq(); |
||||
release(new_left); |
||||
release(new_right); |
||||
} |
||||
} else { |
||||
// The DMA didn't fill blocks, and we could not allocate
|
||||
// memory... the system is likely starving for memory!
|
||||
// Sadly, there's nothing we can do.
|
||||
__enable_irq(); |
||||
} |
||||
} |
||||
|
||||
bool AudioInputSPDIF3_F32::pllLocked(void) |
||||
{ |
||||
return (SPDIF_SRPC & SPDIF_SRPC_LOCK) == SPDIF_SRPC_LOCK ? true:false; |
||||
} |
||||
|
||||
unsigned int AudioInputSPDIF3_F32::sampleRate(void) { |
||||
if (!pllLocked()) return 0; |
||||
return (float)((uint64_t)F_BUS_ACTUAL * SPDIF_SRFM) / (0x8000000ULL * AudioOutputSPDIF3_F32::dpll_Gain()) + 0.5F; |
||||
} |
||||
|
||||
#endif |
||||
@ -0,0 +1,60 @@ |
||||
/* Audio Library for Teensy 3.X
|
||||
* Copyright (c) 2019, 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. |
||||
*/ |
||||
// Frank Bösing
|
||||
|
||||
#if defined(__IMXRT1052__) || defined(__IMXRT1062__) |
||||
#ifndef _input_spdif3_f32_h_ |
||||
#define _input_spdif3_f32_h_ |
||||
|
||||
#include "Arduino.h" |
||||
#include "AudioStream_F32.h" |
||||
#include "DMAChannel.h" |
||||
|
||||
class AudioInputSPDIF3_F32 : public AudioStream_F32 |
||||
{ |
||||
public: |
||||
AudioInputSPDIF3_F32(const AudioSettings_F32 &settings) : AudioStream_F32(0, NULL) {
|
||||
sample_rate_Hz = settings.sample_rate_Hz; |
||||
begin();
|
||||
} |
||||
virtual void update(void); |
||||
void begin(void); |
||||
static bool pllLocked(void); |
||||
static unsigned int sampleRate(void); |
||||
protected: |
||||
//AudioInputSPDIF3_F32(int dummy): AudioStream_F32(0, NULL) {} // to be used only inside AudioInputSPDIF3slave !!
|
||||
static bool update_responsibility; |
||||
static DMAChannel dma; |
||||
static void isr(void); |
||||
private: |
||||
static audio_block_f32_t *block_left; |
||||
static audio_block_f32_t *block_right; |
||||
static uint16_t block_offset; |
||||
static float sample_rate_Hz; |
||||
}; |
||||
|
||||
#endif |
||||
#endif |
||||
@ -0,0 +1,308 @@ |
||||
/* Hardware-SPDIF for Teensy 4
|
||||
* Copyright (c) 2019, Frank Bösing, f.boesing@gmx.de |
||||
* |
||||
* 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. |
||||
*/ |
||||
/*
|
||||
http://www.hardwarebook.info/S/PDIF
|
||||
https://www.mikrocontroller.net/articles/S/PDIF
|
||||
https://en.wikipedia.org/wiki/S/PDIF
|
||||
*/ |
||||
#include <Arduino.h> |
||||
#include "output_spdif3_f32.h" |
||||
|
||||
#if defined(__IMXRT1062__) |
||||
|
||||
#include "utility/imxrt_hw.h" |
||||
#include "memcpy_audio.h" |
||||
#include <math.h> |
||||
|
||||
// TODO: convert within update() instead of isr(), into buffer
|
||||
static inline int32_t f32_to_i24(float32_t f) { |
||||
const float32_t fullscale = (1LL << 23) - 1; |
||||
if (f > 1.0) return fullscale; |
||||
if (f < -1.0) return -fullscale; |
||||
return (int32_t)(f * fullscale); |
||||
} |
||||
|
||||
audio_block_f32_t * AudioOutputSPDIF3_F32::block_left_1st = nullptr; |
||||
audio_block_f32_t * AudioOutputSPDIF3_F32::block_right_1st = nullptr; |
||||
audio_block_f32_t * AudioOutputSPDIF3_F32::block_left_2nd = nullptr; |
||||
audio_block_f32_t * AudioOutputSPDIF3_F32::block_right_2nd = nullptr; |
||||
bool AudioOutputSPDIF3_F32::update_responsibility = false; |
||||
float AudioOutputSPDIF3_F32::sample_rate_Hz = AUDIO_SAMPLE_RATE_EXACT; |
||||
DMAChannel AudioOutputSPDIF3_F32::dma(false); |
||||
|
||||
DMAMEM __attribute__((aligned(32))) |
||||
static int32_t SPDIF_tx_buffer[AUDIO_BLOCK_SAMPLES * 4]; |
||||
DMAMEM __attribute__((aligned(32))) |
||||
audio_block_f32_t AudioOutputSPDIF3_F32::block_silent; |
||||
|
||||
#define SPDIF_DPLL_GAIN24 0 |
||||
#define SPDIF_DPLL_GAIN16 1 |
||||
#define SPDIF_DPLL_GAIN12 2 |
||||
#define SPDIF_DPLL_GAIN8 3 |
||||
#define SPDIF_DPLL_GAIN6 4 |
||||
#define SPDIF_DPLL_GAIN4 5 |
||||
#define SPDIF_DPLL_GAIN3 6 |
||||
#define SPDIF_DPLL_GAIN1 7 |
||||
|
||||
#define SPDIF_DPLL_GAIN SPDIF_DPLL_GAIN8 //Actual Gain
|
||||
static const uint8_t spdif_gain[8] = {24, 16, 12, 8, 6, 4, 3, 1}; |
||||
|
||||
FLASHMEM |
||||
void AudioOutputSPDIF3_F32::begin(void) |
||||
{ |
||||
|
||||
dma.begin(true); // Allocate the DMA channel first
|
||||
|
||||
block_left_1st = nullptr; |
||||
block_right_1st = nullptr; |
||||
memset(&block_silent, 0, sizeof(block_silent)); |
||||
|
||||
config_spdif3(sample_rate_Hz); |
||||
const int nbytes_mlno = 2 * 4; // 8 Bytes per minor loop
|
||||
|
||||
dma.TCD->SADDR = SPDIF_tx_buffer; |
||||
dma.TCD->SOFF = 4; |
||||
dma.TCD->ATTR = DMA_TCD_ATTR_SSIZE(2) | DMA_TCD_ATTR_DSIZE(2); |
||||
dma.TCD->NBYTES_MLNO = DMA_TCD_NBYTES_MLOFFYES_NBYTES(nbytes_mlno) | DMA_TCD_NBYTES_DMLOE | |
||||
DMA_TCD_NBYTES_MLOFFYES_MLOFF(-8); |
||||
dma.TCD->SLAST = -sizeof(SPDIF_tx_buffer); |
||||
dma.TCD->DADDR = &SPDIF_STL; |
||||
dma.TCD->DOFF = 4; |
||||
dma.TCD->DLASTSGA = -8; |
||||
//dma.TCD->ATTR_DST = ((31 - __builtin_clz(8)) << 3);
|
||||
dma.TCD->CITER_ELINKNO = sizeof(SPDIF_tx_buffer) / nbytes_mlno; |
||||
dma.TCD->BITER_ELINKNO = sizeof(SPDIF_tx_buffer) / nbytes_mlno; |
||||
dma.TCD->CSR = DMA_TCD_CSR_INTHALF | DMA_TCD_CSR_INTMAJOR; |
||||
|
||||
dma.triggerAtHardwareEvent(DMAMUX_SOURCE_SPDIF_TX); |
||||
|
||||
update_responsibility = update_setup(); |
||||
dma.enable(); |
||||
dma.attachInterrupt(isr); |
||||
|
||||
CORE_PIN14_CONFIG = 3; //3:SPDIF_OUT
|
||||
SPDIF_SCR |= SPDIF_SCR_DMA_TX_EN; |
||||
SPDIF_STC |= SPDIF_STC_TX_ALL_CLK_EN; |
||||
// pinMode(13, OUTPUT);
|
||||
} |
||||
|
||||
void AudioOutputSPDIF3_F32::isr(void) |
||||
{ |
||||
|
||||
const float32_t *src_left, *src_right; |
||||
const int32_t *end; |
||||
int32_t *dest; |
||||
audio_block_f32_t *block_left, *block_right; |
||||
uint32_t saddr; |
||||
|
||||
saddr = (uint32_t)(dma.TCD->SADDR); |
||||
dma.clearInterrupt(); |
||||
if (saddr < (uint32_t)SPDIF_tx_buffer + sizeof(SPDIF_tx_buffer) / 2) { |
||||
// DMA is transmitting the first half of the buffer
|
||||
// so we must fill the second half
|
||||
dest = SPDIF_tx_buffer + AUDIO_BLOCK_SAMPLES*2; |
||||
end = SPDIF_tx_buffer + AUDIO_BLOCK_SAMPLES*4; |
||||
} else { |
||||
// DMA is transmitting the second half of the buffer
|
||||
// so we must fill the first half
|
||||
dest = SPDIF_tx_buffer; |
||||
end = SPDIF_tx_buffer + AUDIO_BLOCK_SAMPLES*2; |
||||
} |
||||
block_left = block_left_1st; |
||||
if (!block_left) block_left = &block_silent; |
||||
block_right = block_right_1st; |
||||
if (!block_right) block_right = &block_silent; |
||||
|
||||
src_left = (const float32_t *)(block_left->data); |
||||
src_right = (const float32_t *)(block_right->data); |
||||
|
||||
do { |
||||
#if IMXRT_CACHE_ENABLED >= 2 |
||||
SCB_CACHE_DCCIMVAC = (uintptr_t) dest; |
||||
asm volatile("dsb"); |
||||
#endif |
||||
|
||||
*dest++ = f32_to_i24(*src_left++); |
||||
*dest++ = f32_to_i24(*src_right++); |
||||
|
||||
*dest++ = f32_to_i24(*src_left++); |
||||
*dest++ = f32_to_i24(*src_right++); |
||||
|
||||
*dest++ = f32_to_i24(*src_left++); |
||||
*dest++ = f32_to_i24(*src_right++); |
||||
|
||||
*dest++ = f32_to_i24(*src_left++); |
||||
*dest++ = f32_to_i24(*src_right++); |
||||
|
||||
} while (dest < end); |
||||
|
||||
if (block_left != &block_silent) { |
||||
release(block_left); |
||||
block_left_1st = block_left_2nd; |
||||
block_left_2nd = nullptr; |
||||
} |
||||
if (block_right != &block_silent) { |
||||
release(block_right); |
||||
block_right_1st = block_right_2nd; |
||||
block_right_2nd = nullptr; |
||||
} |
||||
|
||||
if (update_responsibility) update_all(); |
||||
//digitalWriteFast(13,!digitalReadFast(13));
|
||||
} |
||||
|
||||
void AudioOutputSPDIF3_F32::update(void) |
||||
{ |
||||
audio_block_f32_t *block_left, *block_right; |
||||
|
||||
block_left = receiveReadOnly_f32(0); // input 0
|
||||
block_right = receiveReadOnly_f32(1); // input 1
|
||||
__disable_irq(); |
||||
if (block_left) { |
||||
if (block_left_1st == nullptr) { |
||||
block_left_1st = block_left; |
||||
block_left = nullptr; |
||||
} else if (block_left_2nd == nullptr) { |
||||
block_left_2nd = block_left; |
||||
block_left = nullptr; |
||||
} else { |
||||
audio_block_f32_t *tmp = block_left_1st; |
||||
block_left_1st = block_left_2nd; |
||||
block_left_2nd = block_left; |
||||
block_left = tmp; |
||||
} |
||||
} |
||||
if (block_right) { |
||||
if (block_right_1st == nullptr) { |
||||
block_right_1st = block_right; |
||||
block_right = nullptr; |
||||
} else if (block_right_2nd == nullptr) { |
||||
block_right_2nd = block_right; |
||||
block_right = nullptr; |
||||
} else { |
||||
audio_block_f32_t *tmp = block_right_1st; |
||||
block_right_1st = block_right_2nd; |
||||
block_right_2nd = block_right; |
||||
block_right = tmp; |
||||
} |
||||
} |
||||
__enable_irq(); |
||||
if (block_left) { |
||||
release(block_left); |
||||
} |
||||
if (block_right) { |
||||
release(block_right); |
||||
} |
||||
|
||||
} |
||||
|
||||
void AudioOutputSPDIF3_F32::mute_PCM(const bool mute) |
||||
{ |
||||
if (mute) |
||||
SPDIF_SCR |= SPDIF_SCR_VALCTRL; |
||||
else |
||||
SPDIF_SCR &= ~SPDIF_SCR_VALCTRL; |
||||
} |
||||
|
||||
uint32_t AudioOutputSPDIF3_F32::dpll_Gain(void) |
||||
{ |
||||
return spdif_gain[SPDIF_DPLL_GAIN]; |
||||
} |
||||
|
||||
FLASHMEM |
||||
void AudioOutputSPDIF3_F32::config_spdif3(float fs_Hz) |
||||
{ |
||||
delay(1); //WHY IS THIS NEEDED?
|
||||
|
||||
uint32_t fs = fs_Hz; |
||||
// PLL between 27*24 = 648MHz und 54*24=1296MHz
|
||||
// n1, n2 choosen for compatibility with I2S (same PLL frequency) :
|
||||
int n1 = 4; //SAI prescaler 4 => (n1*n2) = multiple of 4
|
||||
int n2 = 1 + (24000000 * 27) / (fs * 256 * n1); |
||||
double C = ((double)fs * 256 * n1 * n2) / 24000000; |
||||
int c0 = C; |
||||
int c2 = 10000; |
||||
int c1 = C * c2 - (c0 * c2); |
||||
set_audioClock(c0, c1, c2); |
||||
|
||||
//use new pred/podf values
|
||||
n1 = 7; //0: divide by 1 (do not use with high input frequencies), 1:/2, 2: /3, 7:/8
|
||||
n2 = 0; //0: divide by 1, 7: divide by 8
|
||||
|
||||
CCM_CCGR5 &= ~CCM_CCGR5_SPDIF(CCM_CCGR_ON); //Clock gate off
|
||||
|
||||
CCM_CDCDR = (CCM_CDCDR & ~(CCM_CDCDR_SPDIF0_CLK_SEL_MASK | CCM_CDCDR_SPDIF0_CLK_PRED_MASK | CCM_CDCDR_SPDIF0_CLK_PODF_MASK)) |
||||
| CCM_CDCDR_SPDIF0_CLK_SEL(0) // 0 PLL4, 1 PLL3 PFD2, 2 PLL5, 3 pll3_sw_clk
|
||||
| CCM_CDCDR_SPDIF0_CLK_PRED(n1) |
||||
| CCM_CDCDR_SPDIF0_CLK_PODF(n2); |
||||
|
||||
CCM_CCGR5 |= CCM_CCGR5_SPDIF(CCM_CCGR_ON); //Clock gate on
|
||||
|
||||
if (!(SPDIF_SCR & (SPDIF_SCR_DMA_RX_EN | SPDIF_SCR_DMA_TX_EN))) { |
||||
//Serial.print("Reset SPDIF3");
|
||||
SPDIF_SCR = SPDIF_SCR_SOFT_RESET; //Reset SPDIF
|
||||
while (SPDIF_SCR & SPDIF_SCR_SOFT_RESET) {;} //Wait for Reset (takes 8 cycles)
|
||||
} else return; |
||||
|
||||
SPDIF_SCR = |
||||
SPDIF_SCR_RXFIFOFULL_SEL(0) | // Full interrupt if at least 1 sample in Rx left and right FIFOs
|
||||
SPDIF_SCR_RXAUTOSYNC | |
||||
SPDIF_SCR_TXAUTOSYNC | |
||||
SPDIF_SCR_TXFIFOEMPTY_SEL(2) | // Empty interrupt if at most 8 samples in Tx left and right FIFOs
|
||||
SPDIF_SCR_TXFIFO_CTRL(1) | // 0:Send zeros 1: normal operation
|
||||
SPDIF_SCR_VALCTRL | // Outgoing Validity always clear
|
||||
SPDIF_SCR_TXSEL(5) | // 0:off and output 0, 1:Feed-though SPDIFIN, 5:Tx Normal operation
|
||||
SPDIF_SCR_USRC_SEL(3); |
||||
|
||||
SPDIF_SRPC = |
||||
SPDIF_SRPC_CLKSRC_SEL(1) | //if (DPLL Locked) SPDIF_RxClk else tx_clk (SPDIF0_CLK_ROOT)
|
||||
SPDIF_SRPC_GAINSEL(SPDIF_DPLL_GAIN); |
||||
|
||||
uint32_t pllclock = (c0 + (float)c1 / c2) * 24000000ULL; //677376000 Hz
|
||||
uint32_t clock = pllclock / (1 + n1) / (1 + n2); |
||||
uint32_t clkdiv = clock / (fs * 64); // 1 .. 128
|
||||
uint32_t mod = clock % (fs * 64); |
||||
if (mod > ((fs * 64) / 2)) clkdiv += 1; //nearest divider
|
||||
|
||||
#if 0 |
||||
Serial.printf("PLL: %d\n", pllclock); |
||||
Serial.printf("clock: %d\n", clock); |
||||
Serial.printf("clkdiv: %d\n", clkdiv); |
||||
#endif |
||||
|
||||
SPDIF_STC = |
||||
SPDIF_STC_TXCLK_SOURCE(1) | //tx_clk input (from SPDIF0_CLK_ROOT)
|
||||
SPDIF_STC_TXCLK_DF(clkdiv - 1); |
||||
} |
||||
|
||||
#endif // __IMXRT1062__
|
||||
|
||||
|
||||
#if defined(__MK66FX1M0__) || defined(__MK64FX512__) || defined(__MK20DX256__) || defined(__MKL26Z64__) |
||||
// empty code to allow compile (but no sound output) on other Teensy models
|
||||
|
||||
void AudioOutputSPDIF3_F32::update(void) { } |
||||
void AudioOutputSPDIF3_F32::begin(void) { } |
||||
void AudioOutputSPDIF3_F32::mute_PCM(const bool mute) { } |
||||
bool AudioOutputSPDIF3_F32::pll_locked(void) { return false; } |
||||
|
||||
#endif |
||||
@ -0,0 +1,62 @@ |
||||
/* Hardware-SPDIF for Teensy 4
|
||||
* Copyright (c) 2019, Frank Bösing, f.boesing@gmx.de |
||||
* |
||||
* 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. |
||||
*/ |
||||
|
||||
#ifndef output_SPDIF3_f32_h_ |
||||
#define output_SPDIF3_f32_h_ |
||||
|
||||
#include <Arduino.h> |
||||
#include "AudioStream_F32.h" |
||||
//include "AudioStream.h"
|
||||
#include <DMAChannel.h> |
||||
|
||||
class AudioOutputSPDIF3_F32 : public AudioStream_F32 |
||||
{ |
||||
public: |
||||
AudioOutputSPDIF3_F32(const AudioSettings_F32 &settings) : AudioStream_F32(2, inputQueueArray) { |
||||
sample_rate_Hz = settings.sample_rate_Hz; |
||||
begin(); |
||||
} |
||||
virtual void update(void); |
||||
void begin(void); |
||||
friend class AudioInputSPDIF3_F32; |
||||
friend class AsyncAudioInputSPDIF3_F32; |
||||
static void mute_PCM(const bool mute); |
||||
static bool pll_locked(void); |
||||
protected: |
||||
//AudioOutputSPDIF3_F32(int dummy): AudioStream(2, inputQueueArray) {}
|
||||
static void config_spdif3(float fs_Hz); |
||||
static audio_block_f32_t *block_left_1st; |
||||
static audio_block_f32_t *block_right_1st; |
||||
static bool update_responsibility; |
||||
static DMAChannel dma; |
||||
static void isr(void);
|
||||
private: |
||||
static uint32_t dpll_Gain() __attribute__ ((const)); |
||||
static audio_block_f32_t *block_left_2nd; |
||||
static audio_block_f32_t *block_right_2nd; |
||||
static audio_block_f32_t block_silent; |
||||
audio_block_f32_t *inputQueueArray[2]; |
||||
static float sample_rate_Hz; |
||||
}; |
||||
|
||||
|
||||
#endif |
||||
Loading…
Reference in new issue