forked from PaulStoffregen/Audio
-
Notifications
You must be signed in to change notification settings - Fork 1
/
async_input_spdif3.h
88 lines (81 loc) · 3.57 KB
/
async_input_spdif3.h
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
/* 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_h_
#define async_input_spdif3_h_
#include "Resampler.h"
#include "Quantizer.h"
#include "Arduino.h"
#include "AudioStream.h"
#include "DMAChannel.h"
#include <arm_math.h>
//#define DEBUG_SPDIF_IN //activates debug output
class AsyncAudioInputSPDIF3 : public AudioStream
{
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(bool dither=false, bool noiseshaping=false,float attenuation=100, int32_t minHalfFilterLength=20, int32_t maxHalfFilterLength=80);
~AsyncAudioInputSPDIF3();
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(int16_t* data_left, int16_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;
Quantizer* 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;
#ifdef DEBUG_SPDIF_IN
static volatile bool bufferOverflow;
#endif
};
#endif