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431 lines
15 KiB
431 lines
15 KiB
/* 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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