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OpenAudio_ArduinoLibrary/async_input_spdif3_F32.cpp

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/* 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(__IMXRT1062__)
#include "async_input_spdif3_F32.h"
#include "output_spdif3_F32.h"
#include "biquad.h"
#include <utility/imxrt_hw.h>
//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;
const float toFloatAudio= (float)(1./pow(2., 23.));
}
// dummy class, no quantization
class Scaler_F32 {
public:
Scaler_F32() {
_factor = 1.0;
};
void configure() {
};
void quantize(float* input, float32_t* output, uint16_t length) {
memcpy(output, input, length * sizeof(float));
/*
for (uint16_t i =0; i< length; i++){
*output++ = *input++ * _factor;
}
*/
};
private:
float _factor;
};
#ifdef DEBUG_SPDIF_IN
volatile bool AsyncAudioInputSPDIF3_F32::bufferOverflow=false;
#endif
volatile uint32_t AsyncAudioInputSPDIF3_F32::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_F32::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_F32::resample_offset = 0; // read/written by resample/ read in spdif input isr -> no protection needed?
float AsyncAudioInputSPDIF3_F32::sample_rate_Hz = AUDIO_SAMPLE_RATE_EXACT;
DMAChannel AsyncAudioInputSPDIF3_F32::dma(false);
AsyncAudioInputSPDIF3_F32::~AsyncAudioInputSPDIF3_F32(){
delete [] _bufferLPFilter.pCoeffs;
delete [] _bufferLPFilter.pState;
delete quantizer[0];
delete quantizer[1];
}
FLASHMEM
AsyncAudioInputSPDIF3_F32::AsyncAudioInputSPDIF3_F32(const AudioSettings_F32 &settings, float attenuation, int32_t minHalfFilterLength, int32_t maxHalfFilterLength):
AudioStream_F32(0, NULL),
_resampler(attenuation, minHalfFilterLength, maxHalfFilterLength)
{
sample_rate_Hz = settings.sample_rate_Hz;
quantizer[0]=new Scaler_F32();
quantizer[0]->configure();
quantizer[1]=new Scaler_F32();
quantizer[1]->configure();
begin();
}
FLASHMEM
void AsyncAudioInputSPDIF3_F32::begin()
{
AudioOutputSPDIF3_F32::config_spdif3(sample_rate_Hz);
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);
#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., sample_rate_Hz/AUDIO_BLOCK_SAMPLES, 0.5);
SPDIF_SCR &=(~SPDIF_SCR_RXFIFO_OFF_ON); //receive fifo is turned on again
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
}
bool AsyncAudioInputSPDIF3_F32::isLocked() {
return (SPDIF_SRPC & SPDIF_SRPC_LOCK) == SPDIF_SRPC_LOCK;
}
void AsyncAudioInputSPDIF3_F32::resample(float32_t* data_left, float32_t* data_right, int32_t& block_offset){
block_offset=0;
if(!_resampler.initialized() || !isLocked()){
return;
}
int32_t bOffset=buffer_offset;
int32_t resOffset=resample_offset;
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); // TODO: degenerated to a copy, perhaps directly resample into here?
quantizer[1]->quantize(resampledBufferR, data_right, block_offset);
__disable_irq();
resample_offset=resOffset;
__enable_irq();
}
void AsyncAudioInputSPDIF3_F32::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_F32::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]);
do {
int32_t n=(*src) & 0x800000 ? (*src)|0xFF800000 : (*src) & 0xFFFFFF;
*destL++ = (float)(n)*toFloatAudio;
++src;
n=(*src) & 0x800000 ? (*src)|0xFF800000 : (*src) & 0xFFFFFF;
*destR++ = (float)(n)*toFloatAudio;
++src;
} while (src < end);
buffer_offset=(buffer_offset+SPDIF_RX_BUFFER_LENGTH/4)%bufferLength;
}
#ifdef DEBUG_SPDIF_IN
else {
bufferOverflow=true;
}
#endif
}
double AsyncAudioInputSPDIF3_F32::getNewValidInputFrequ(){
//page 2129: FrequMeas[23:0]=FreqMeas_CLK / BUS_CLK * 2^10 * GAIN
if (isLocked()){
const double f=(double)F_BUS_ACTUAL/(1048576.*(double)AudioOutputSPDIF3_F32::dpll_Gain()*128.);// bit clock = 128 * sampling frequency
const double freqMeas=(double)(SPDIF_SRFM & 0xFFFFFF)*f;
if (_lastValidInputFrequ != freqMeas){//frequency not stable yet;
_lastValidInputFrequ=freqMeas;
return -1.;
}
return _lastValidInputFrequ;
}
return -1.;
}
double AsyncAudioInputSPDIF3_F32::getBufferedTime() const{
__disable_irq();
double n=_bufferedTime;
__enable_irq();
return n;
}
void AsyncAudioInputSPDIF3_F32::configure(){
if(!isLocked()){
_resampler.reset();
return;
}
#ifdef DEBUG_SPDIF_IN
const bool bOverf=bufferOverflow;
bufferOverflow=false;
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
const double inputF=getNewValidInputFrequ(); //returns: -1 ... invalid frequency
if (inputF > 0.){
//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;
_targetLatencyS=max(0.001,(noSamplerPerIsr*3./2./_inputFrequency));
_maxLatency=max(2.*_blockDuration, 2*noSamplerPerIsr/_inputFrequency);
const int32_t targetLatency=round(_targetLatencyS*inputF);
__disable_irq();
resample_offset = targetLatency <= buffer_offset ? buffer_offset - targetLatency : bufferLength -(targetLatency-buffer_offset);
__enable_irq();
_resampler.configure(inputF, sample_rate_Hz);
#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_F32::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<double, arm_biquad_cascade_df2T_instance_f32, float>(&_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;
}
#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.println(bTime*1e6,3);
Serial.print("_maxLatency: ");
Serial.println(_maxLatency*1e6,3);
Serial.print("bTime-dmaOffset: ");
Serial.println((bTime-dmaOffset)*1e6,3);
Serial.print(", _blockDuration: ");
Serial.println(_blockDuration*1e6,3);
Serial.print("bTimeFixed: ");
Serial.println(bTimeFixed*1e6,3);
#endif
preload(&_bufferLPFilter, (float)diff);
_resampler.fixStep();
}
else {
__enable_irq();
}
_bufferedTime=_targetLatencyS+diff;
}
void AsyncAudioInputSPDIF3_F32::update(void)
{
configure();
monitorResampleBuffer(); //important first call 'monitorResampleBuffer' then 'resample'
audio_block_f32_t *block_left = allocate_f32();
audio_block_f32_t *block_right = nullptr;
if (block_left!= nullptr) {
block_right = allocate_f32();
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(float32_t));
memset(block_right->data+block_offset, 0, (AUDIO_BLOCK_SAMPLES-block_offset)*sizeof(float32_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_F32::getInputFrequency() const{
__disable_irq();
double f=_lastValidInputFrequ;
__enable_irq();
return isLocked() ? f : 0.;
}
double AsyncAudioInputSPDIF3_F32::getTargetLantency() const {
__disable_irq();
double l=_targetLatencyS;
__enable_irq();
return l ;
}
double AsyncAudioInputSPDIF3_F32::getAttenuation() const{
return _resampler.getAttenuation();
}
int32_t AsyncAudioInputSPDIF3_F32::getHalfFilterLength() const{
return _resampler.getHalfFilterLength();
}
#endif // __IMXRT1062__
#if defined(__MK66FX1M0__) || defined(__MK64FX512__) || defined(__MK20DX256__) || defined(__MKL26Z64__)
// empty code to allow compile (but no sound input) on other Teensy models
#include "async_input_spdif3.h"
AsyncAudioInputSPDIF3_F32::AsyncAudioInputSPDIF3_F32(bool dither, bool noiseshaping,float attenuation, int32_t minHalfFilterLength, int32_t maxHalfFilterLength):
AudioStream(0, NULL), _resampler(attenuation, minHalfFilterLength, maxHalfFilterLength)
{ }
void AsyncAudioInputSPDIF3_F32::begin() { }
void AsyncAudioInputSPDIF3_F32::update(void) { }
double AsyncAudioInputSPDIF3_F32::getBufferedTime() const { return 0; }
double AsyncAudioInputSPDIF3_F32::getInputFrequency() const { return 0; }
bool AsyncAudioInputSPDIF3_F32::isLocked() { return false; }
double AsyncAudioInputSPDIF3_F32::getTargetLantency() const { return 0; }
double AsyncAudioInputSPDIF3_F32::getAttenuation() const { return 0; }
int32_t AsyncAudioInputSPDIF3_F32::getHalfFilterLength() const { return 0; }
AsyncAudioInputSPDIF3_F32::~AsyncAudioInputSPDIF3_F32() { }
#endif