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OpenAudio_ArduinoLibrary/async_input_spdif3_F32.h

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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
*/
#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