/* 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 Alexander Walch */ #if defined(__IMXRT1052__) || defined(__IMXRT1062__) #include "async_input_spdif3.h" #include "biquad.h" #include //Parameters namespace { #define SPDIF_RX_BUFFER_LENGTH AUDIO_BLOCK_SAMPLES const int32_t bufferLength=8*AUDIO_BLOCK_SAMPLES; const uint16_t noSamplerPerIsr=SPDIF_RX_BUFFER_LENGTH/4; } volatile bool AsyncAudioInputSPDIF3::resetResampler=true; #ifdef DEBUG_SPDIF_IN volatile bool AsyncAudioInputSPDIF3::bufferOverflow=false; #endif volatile uint32_t AsyncAudioInputSPDIF3::microsLast; DMAMEM __attribute__((aligned(32))) static int32_t spdif_rx_buffer[SPDIF_RX_BUFFER_LENGTH]; static float bufferR[bufferLength]; static float bufferL[bufferLength]; volatile int32_t AsyncAudioInputSPDIF3::buffer_offset = 0; // read by resample/ written in spdif input isr -> copied at the beginning of 'resmaple' protected by __disable_irq() in resample int32_t AsyncAudioInputSPDIF3::resample_offset = 0; // read/written by resample/ read in spdif input isr -> no protection needed? volatile bool AsyncAudioInputSPDIF3::lockChanged=false; volatile bool AsyncAudioInputSPDIF3::locked=false; DMAChannel AsyncAudioInputSPDIF3::dma(false); AsyncAudioInputSPDIF3::~AsyncAudioInputSPDIF3(){ delete [] _bufferLPFilter.pCoeffs; delete [] _bufferLPFilter.pState; delete quantizer[0]; delete quantizer[1]; } PROGMEM AsyncAudioInputSPDIF3::AsyncAudioInputSPDIF3(bool dither, bool noiseshaping,float attenuation, int32_t minHalfFilterLength) : AudioStream(0, NULL) { _attenuation=attenuation; _minHalfFilterLength=minHalfFilterLength; const float factor = powf(2, 15)-1.f; // to 16 bit audio quantizer[0]=new Quantizer(AUDIO_SAMPLE_RATE_EXACT); quantizer[0]->configure(noiseshaping, dither, factor); quantizer[1]=new Quantizer(AUDIO_SAMPLE_RATE_EXACT); quantizer[1]->configure(noiseshaping, dither, factor); begin(); } PROGMEM void AsyncAudioInputSPDIF3::begin() { dma.begin(true); // Allocate the DMA channel first const uint32_t noByteMinorLoop=2*4; 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(noByteMinorLoop) | DMA_TCD_NBYTES_SMLOE | DMA_TCD_NBYTES_MLOFFYES_MLOFF(-8); dma.TCD->SLAST = -8; dma.TCD->DOFF = 4; dma.TCD->CITER_ELINKNO = sizeof(spdif_rx_buffer) / noByteMinorLoop; dma.TCD->DLASTSGA = -sizeof(spdif_rx_buffer); dma.TCD->BITER_ELINKNO = sizeof(spdif_rx_buffer) / noByteMinorLoop; dma.TCD->CSR = DMA_TCD_CSR_INTHALF | DMA_TCD_CSR_INTMAJOR; dma.TCD->SADDR = (void *)((uint32_t)&SPDIF_SRL); dma.TCD->DADDR = spdif_rx_buffer; dma.triggerAtHardwareEvent(DMAMUX_SOURCE_SPDIF_RX); SPDIF_SCR |=SPDIF_SCR_DMA_RX_EN; //DMA Receive Request Enable dma.enable(); dma.attachInterrupt(isr); config_spdifIn(); #ifdef DEBUG_SPDIF_IN while (!Serial); #endif _bufferLPFilter.pCoeffs=new float[5]; _bufferLPFilter.numStages=1; _bufferLPFilter.pState=new float[2]; getCoefficients(_bufferLPFilter.pCoeffs, BiquadType::LOW_PASS, 0., 5., AUDIO_SAMPLE_RATE_EXACT/AUDIO_BLOCK_SAMPLES, 0.5); } bool AsyncAudioInputSPDIF3::isLocked() const { __disable_irq(); bool l=locked; __enable_irq(); return l; } void AsyncAudioInputSPDIF3::spdif_interrupt(){ if(SPDIF_SIS & SPDIF_SIS_LOCK){ if (!locked){ locked=true; lockChanged=true; } } else if(SPDIF_SIS & SPDIF_SIS_LOCKLOSS){ if (locked){ locked=false; lockChanged=true; resetResampler=true; } } SPDIF_SIC |= SPDIF_SIC_LOCKLOSS;//clear SPDIF_SIC_LOCKLOSS interrupt SPDIF_SIC |= SPDIF_SIC_LOCK; //clear SPDIF_SIC_LOCK interrupt } void AsyncAudioInputSPDIF3::resample(int16_t* data_left, int16_t* data_right, int32_t& block_offset){ block_offset=0; if(!_resampler.initialized()){ return; } __disable_irq(); if(!locked){ __enable_irq(); return; } int32_t bOffset=buffer_offset; int32_t resOffset=resample_offset; __enable_irq(); uint16_t inputBufferStop = bOffset >= resOffset ? bOffset-resOffset : bufferLength-resOffset; if (inputBufferStop==0){ return; } uint16_t processedLength; uint16_t outputCount=0; uint16_t outputLength=AUDIO_BLOCK_SAMPLES; float resampledBufferL[AUDIO_BLOCK_SAMPLES]; float resampledBufferR[AUDIO_BLOCK_SAMPLES]; _resampler.resample(&bufferL[resOffset],&bufferR[resOffset], inputBufferStop, processedLength, resampledBufferL, resampledBufferR, outputLength, outputCount); resOffset=(resOffset+processedLength)%bufferLength; block_offset=outputCount; if (bOffset > resOffset && block_offset< AUDIO_BLOCK_SAMPLES){ inputBufferStop= bOffset-resOffset; outputLength=AUDIO_BLOCK_SAMPLES-block_offset; _resampler.resample(&bufferL[resOffset],&bufferR[resOffset], inputBufferStop, processedLength, resampledBufferL+block_offset, resampledBufferR+block_offset, outputLength, outputCount); resOffset=(resOffset+processedLength)%bufferLength; block_offset+=outputCount; } quantizer[0]->quantize(resampledBufferL, data_left, block_offset); quantizer[1]->quantize(resampledBufferR, data_right, block_offset); __disable_irq(); resample_offset=resOffset; __enable_irq(); } void AsyncAudioInputSPDIF3::isr(void) { dma.clearInterrupt(); microsLast=micros(); const int32_t *src, *end; uint32_t 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[SPDIF_RX_BUFFER_LENGTH/2]; end = (int32_t *)&spdif_rx_buffer[SPDIF_RX_BUFFER_LENGTH]; //if (AsyncAudioInputSPDIF3::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 = (int32_t *)&spdif_rx_buffer[0]; end = (int32_t *)&spdif_rx_buffer[SPDIF_RX_BUFFER_LENGTH/2]; } if (buffer_offset >=resample_offset || (buffer_offset + SPDIF_RX_BUFFER_LENGTH/4) < resample_offset) { #if IMXRT_CACHE_ENABLED >=1 arm_dcache_delete((void*)src, sizeof(spdif_rx_buffer) / 2); #endif float *destR = &(bufferR[buffer_offset]); float *destL = &(bufferL[buffer_offset]); const float factor= pow(2., 23.)+1; do { int32_t n=(*src) & 0x800000 ? (*src)|0xFF800000 : (*src) & 0xFFFFFF; *destL++ = (float)(n)/factor; ++src; n=(*src) & 0x800000 ? (*src)|0xFF800000 : (*src) & 0xFFFFFF; *destR++ = (float)(n)/factor; ++src; } while (src < end); buffer_offset=(buffer_offset+SPDIF_RX_BUFFER_LENGTH/4)%bufferLength; } #ifdef DEBUG_SPDIF_IN else { bufferOverflow=true; } #endif } double AsyncAudioInputSPDIF3::getNewValidInputFrequ(){ //page 2129: FrequMeas[23:0]=FreqMeas_CLK / BUS_CLK * 2^10 * GAIN if (SPDIF_SRPC & SPDIF_SRPC_LOCK){ const double f=(float)F_BUS_ACTUAL/(1024.*1024.*24.*128.);// bit clock = 128 * sampling frequency const double freqMeas=(SPDIF_SRFM & 0xFFFFFF)*f; if (_lastValidInputFrequ != freqMeas){//frequency not stable yet; _lastValidInputFrequ=freqMeas; return -1.; } return _lastValidInputFrequ; } return -1.; } double AsyncAudioInputSPDIF3::getBufferedTime() const{ __disable_irq(); double n=_bufferedTime; __enable_irq(); return n; } void AsyncAudioInputSPDIF3::configure(){ __disable_irq(); if(resetResampler){ _resampler.reset(); resetResampler=false; } if(!locked){ __enable_irq(); #ifdef DEBUG_SPDIF_IN Serial.println("lock lost"); #endif return; } #ifdef DEBUG_SPDIF_IN const bool bOverf=bufferOverflow; bufferOverflow=false; #endif const bool lc=lockChanged; __enable_irq(); #ifdef DEBUG_SPDIF_IN if (bOverf){ Serial.print("buffer overflow, buffer offset: "); Serial.print(buffer_offset); Serial.print(", resample_offset: "); Serial.println(resample_offset); if (!_resampler.initialized()){ Serial.println("_resampler not initialized. "); } } #endif if (lc || !_resampler.initialized()){ const double inputF=getNewValidInputFrequ(); //returns: -1 ... invalid frequency if (inputF > 0.){ __disable_irq(); lockChanged=false; //only reset lockChanged if a valid frequency was received (inputFrequ > 0.) __enable_irq(); //we got a valid sample frequency const double frequDiff=inputF/_inputFrequency-1.; if (abs(frequDiff) > 0.01 || !_resampler.initialized()){ //the new sample frequency differs from the last one -> configure the _resampler again _inputFrequency=inputF; const int32_t targetLatency=round(_targetLatencyS*inputF); _targetLatencyS=max(0.001,(noSamplerPerIsr*3./2./_inputFrequency)); __disable_irq(); resample_offset = targetLatency <= buffer_offset ? buffer_offset - targetLatency : bufferLength -(targetLatency-buffer_offset); __enable_irq(); _resampler.configure(inputF, AUDIO_SAMPLE_RATE_EXACT, _attenuation, _minHalfFilterLength); #ifdef DEBUG_SPDIF_IN Serial.print("_maxLatency: "); Serial.println(_maxLatency); Serial.print("targetLatency: "); Serial.println(targetLatency); Serial.print("relative frequ diff: "); Serial.println(frequDiff, 8); Serial.print("configure _resampler with frequency "); Serial.println(inputF,8); #endif } } } } void AsyncAudioInputSPDIF3::monitorResampleBuffer(){ if(!_resampler.initialized()){ return; } __disable_irq(); const double dmaOffset=(micros()-microsLast)*1e-6; //[seconds] double bTime = resample_offset <= buffer_offset ? (buffer_offset-resample_offset-_resampler.getXPos())/_lastValidInputFrequ+dmaOffset : (bufferLength-resample_offset +buffer_offset-_resampler.getXPos())/_lastValidInputFrequ+dmaOffset; //[seconds] double diff = bTime- (_blockDuration+ _targetLatencyS); //seconds biquad_cascade_df2T(&_bufferLPFilter, &diff, &diff, 1); bool settled=_resampler.addToSampleDiff(diff); if (bTime > _maxLatency || bTime-dmaOffset<= _blockDuration || settled) { double distance=(_blockDuration+_targetLatencyS-dmaOffset)*_lastValidInputFrequ+_resampler.getXPos(); diff=0.; if (distance > bufferLength-noSamplerPerIsr){ diff=bufferLength-noSamplerPerIsr-distance; distance=bufferLength-noSamplerPerIsr; } if (distance < 0.){ distance=0.; diff=- (_blockDuration+ _targetLatencyS); } double resample_offsetF=buffer_offset-distance; resample_offset=(int32_t)floor(resample_offsetF); _resampler.addToPos(resample_offsetF-resample_offset); while (resample_offset<0){ resample_offset+=bufferLength; } //int32_t b_offset=buffer_offset; #ifdef DEBUG_SPDIF_IN double bTimeFixed = resample_offset <= buffer_offset ? (buffer_offset-resample_offset-_resampler.getXPos())/_lastValidInputFrequ+dmaOffset : (bufferLength-resample_offset +buffer_offset-_resampler.getXPos())/_lastValidInputFrequ+dmaOffset; //[seconds] #endif __enable_irq(); #ifdef DEBUG_SPDIF_IN // Serial.print("settled: "); // Serial.println(settled); Serial.print("bTime: "); Serial.print(bTime*1e6,3); Serial.print("_maxLatency: "); Serial.println(_maxLatency*1e6,3); Serial.print("bTime-dmaOffset: "); Serial.print((bTime-dmaOffset)*1e6,3); Serial.print(", _blockDuration: "); Serial.print(_blockDuration*1e6,3); Serial.print("bTimeFixed: "); Serial.print(bTimeFixed*1e6,3); #endif preload(&_bufferLPFilter, diff); _resampler.fixStep(); } else { __enable_irq(); } _bufferedTime=_targetLatencyS+diff; } void AsyncAudioInputSPDIF3::update(void) { configure(); monitorResampleBuffer(); //important first call 'monitorResampleBuffer' then 'resample' audio_block_t *block_left =allocate(); audio_block_t *block_right =nullptr; if (block_left!= nullptr) { block_right = allocate(); if (block_right == nullptr) { release(block_left); block_left = nullptr; } } if (block_left && block_right) { int32_t block_offset; resample(block_left->data, block_right->data,block_offset); if(block_offset < AUDIO_BLOCK_SAMPLES){ memset(block_left->data+block_offset, 0, (AUDIO_BLOCK_SAMPLES-block_offset)*sizeof(int16_t)); memset(block_right->data+block_offset, 0, (AUDIO_BLOCK_SAMPLES-block_offset)*sizeof(int16_t)); #ifdef DEBUG_SPDIF_IN Serial.print("filled only "); Serial.print(block_offset); Serial.println(" samples."); #endif } transmit(block_left, 0); release(block_left); block_left=nullptr; transmit(block_right, 1); release(block_right); block_right=nullptr; } #ifdef DEBUG_SPDIF_IN else { Serial.println("Not enough blocks available. Too few audio memory?"); } #endif } double AsyncAudioInputSPDIF3::getInputFrequency() const{ __disable_irq(); double f=_lastValidInputFrequ; __enable_irq(); return f; } double AsyncAudioInputSPDIF3::getTargetLantency() const { __disable_irq(); double l=_targetLatencyS; __enable_irq(); return l ; } void AsyncAudioInputSPDIF3::config_spdifIn(){ //CCM Clock Gating Register 5, imxrt1060_rev1.pdf page 1145 CCM_CCGR5 |=CCM_CCGR5_SPDIF(CCM_CCGR_ON); //turn spdif clock on - necessary for receiver! SPDIF_SCR |=SPDIF_SCR_RXFIFO_OFF_ON; //turn receive fifo off 1->off, 0->on SPDIF_SCR&=~(SPDIF_SCR_RXFIFO_CTR); //reset rx fifo control: normal opertation SPDIF_SCR&=~(SPDIF_SCR_RXFIFOFULL_SEL(3)); //reset rx full select SPDIF_SCR|=SPDIF_SCR_RXFIFOFULL_SEL(2); //full interrupt if at least 8 sample in Rx left and right FIFOs SPDIF_SCR|=SPDIF_SCR_RXAUTOSYNC; //Rx FIFO auto sync on SPDIF_SCR&=(~SPDIF_SCR_USRC_SEL(3)); //No embedded U channel CORE_PIN15_CONFIG = 3; //pin 15 set to alt3 -> spdif input /// from eval board sample code // IOMUXC_SetPinConfig( // IOMUXC_GPIO_AD_B1_03_SPDIF_IN, /* GPIO_AD_B1_03 PAD functional properties : */ // 0x10B0u); /* Slew Rate Field: Slow Slew Rate // Drive Strength Field: R0/6 // Speed Field: medium(100MHz) // Open Drain Enable Field: Open Drain Disabled // Pull / Keep Enable Field: Pull/Keeper Enabled // Pull / Keep Select Field: Keeper // Pull Up / Down Config. Field: 100K Ohm Pull Down // Hyst. Enable Field: Hysteresis Disabled */ CORE_PIN15_PADCONFIG=0x10B0; SPDIF_SCR &=(~SPDIF_SCR_RXFIFO_OFF_ON); //receive fifo is turned on again SPDIF_SRPC &= ~SPDIF_SRPC_CLKSRC_SEL(15); //reset clock selection page 2136 //SPDIF_SRPC |=SPDIF_SRPC_CLKSRC_SEL(6); //if (DPLL Locked) SPDIF_RxClk else tx_clk (SPDIF0_CLK_ROOT) //page 2129: FrequMeas[23:0]=FreqMeas_CLK / BUS_CLK * 2^10 * GAIN SPDIF_SRPC &=~SPDIF_SRPC_GAINSEL(7); //reset gain select 0 -> gain = 24*2^10 //SPDIF_SRPC |= SPDIF_SRPC_GAINSEL(3); //gain select: 8*2^10 //============================================== //interrupts SPDIF_SIE |= SPDIF_SIE_LOCK; //enable spdif receiver lock interrupt SPDIF_SIE |=SPDIF_SIE_LOCKLOSS; lockChanged=true; attachInterruptVector(IRQ_SPDIF, spdif_interrupt); NVIC_SET_PRIORITY(IRQ_SPDIF, 208); // 255 = lowest priority, 208 = priority of update NVIC_ENABLE_IRQ(IRQ_SPDIF); SPDIF_SIC |= SPDIF_SIC_LOCK; //clear SPDIF_SIC_LOCK interrupt SPDIF_SIC |= SPDIF_SIC_LOCKLOSS;//clear SPDIF_SIC_LOCKLOSS interrupt locked=(SPDIF_SRPC & SPDIF_SRPC_LOCK); } #endif