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91 lines
3.7 KiB
91 lines
3.7 KiB
3 years ago
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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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#ifndef async_input_spdif3_f32_h_
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#define async_input_spdif3_f32_h_
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#include "Resampler.h"
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#include "Quantizer.h"
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#include "Arduino.h"
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#include "AudioStream_F32.h"
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#include "DMAChannel.h"
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#include <arm_math.h>
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//#define DEBUG_SPDIF_IN //activates debug output
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class Scaler_F32; // internal
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class AsyncAudioInputSPDIF3_F32 : public AudioStream_F32
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{
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public:
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///@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
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///@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
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///@param maxHalfFilterLength Can be used to restrict the maximum filter length at the cost of a lower attenuation
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AsyncAudioInputSPDIF3_F32(const AudioSettings_F32 &settings, float attenuation=100, int32_t minHalfFilterLength=20, int32_t maxHalfFilterLength=80);
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~AsyncAudioInputSPDIF3_F32();
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void begin();
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virtual void update(void);
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double getBufferedTime() const;
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double getInputFrequency() const;
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static bool isLocked();
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double getTargetLantency() const;
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double getAttenuation() const;
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int32_t getHalfFilterLength() const;
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protected:
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static DMAChannel dma;
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static void isr(void);
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private:
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void resample(float32_t* data_left, float32_t* data_right, int32_t& block_offset);
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void monitorResampleBuffer();
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void configure();
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double getNewValidInputFrequ();
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void config_spdifIn();
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//accessed in isr ====
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static volatile int32_t buffer_offset;
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static int32_t resample_offset;
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static volatile uint32_t microsLast;
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//====================
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Resampler _resampler;
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Scaler_F32* quantizer[2];
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arm_biquad_cascade_df2T_instance_f32 _bufferLPFilter;
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volatile double _bufferedTime;
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volatile double _lastValidInputFrequ;
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double _inputFrequency=0.;
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double _targetLatencyS; //target latency [seconds]
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const double _blockDuration=AUDIO_BLOCK_SAMPLES/AUDIO_SAMPLE_RATE_EXACT; //[seconds]
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double _maxLatency=2.*_blockDuration;
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static float sample_rate_Hz; // configured output sample rate
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#ifdef DEBUG_SPDIF_IN
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static volatile bool bufferOverflow;
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#endif
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};
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#endif
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