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#include "AudioStream.h"
#include "arm_math.h"


// When changing multiple audio object settings that must update at
// the same time, these functions allow the audio library interrupt
// to be disabled. For example, you may wish to begin playing a note
// in response to reading an analog sensor. If you have "velocity"
// information, you might start the sample playing and also adjust
// the gain of a mixer channel. Use AudioNoInterrupts() first, then
// make both changes to the 2 separate objects. Then allow the audio
// library to update with AudioInterrupts(). Both changes will happen
// at the same time, because AudioNoInterrupts() prevents any updates
// while you make changes.
#define AudioNoInterrupts() (NVIC_DISABLE_IRQ(IRQ_SOFTWARE))
#define AudioInterrupts() (NVIC_ENABLE_IRQ(IRQ_SOFTWARE))


// waveforms.c
extern "C" {
extern const int16_t AudioWaveformSine[257];
extern const int16_t AudioWaveformTriangle[257];
extern const int16_t AudioWaveformSquare[257];
extern const int16_t AudioWaveformSawtooth[257];
}

// windows.c
extern "C" {
extern const int16_t AudioWindowHanning256[];
extern const int16_t AudioWindowBartlett256[];
extern const int16_t AudioWindowBlackman256[];
extern const int16_t AudioWindowFlattop256[];
extern const int16_t AudioWindowBlackmanHarris256[];
extern const int16_t AudioWindowNuttall256[];
extern const int16_t AudioWindowBlackmanNuttall256[];
extern const int16_t AudioWindowWelch256[];
extern const int16_t AudioWindowHamming256[];
extern const int16_t AudioWindowCosine256[];
extern const int16_t AudioWindowTukey256[];
}

class AudioAnalyzeFFT256 : public AudioStream
{
public:
AudioAnalyzeFFT256(uint8_t navg = 8, const int16_t *win = AudioWindowHanning256)
: AudioStream(1, inputQueueArray), outputflag(false),
prevblock(NULL), count(0), naverage(navg), window(win) { init(); }

bool available() {
if (outputflag == true) {
outputflag = false;
return true;
}
return false;
}
virtual void update(void);
//uint32_t cycles;
int32_t output[128] __attribute__ ((aligned (4)));
private:
void init(void);
const int16_t *window;
audio_block_t *prevblock;
int16_t buffer[512] __attribute__ ((aligned (4)));
uint8_t count;
uint8_t naverage;
bool outputflag;
audio_block_t *inputQueueArray[1];
};



class AudioSynthWaveform : public AudioStream
{
public:
AudioSynthWaveform(const int16_t *waveform)
: AudioStream(0, NULL), wavetable(waveform), magnitude(0), phase(0) { }
void frequency(float freq) {
if (freq > AUDIO_SAMPLE_RATE_EXACT / 2 || freq < 0.0) return;
phase_increment = (freq / AUDIO_SAMPLE_RATE_EXACT) * 4294967296.0f;
}
void amplitude(float n) { // 0 to 1.0
if (n < 0) n = 0;
else if (n > 1.0) n = 1.0;
magnitude = n * 32767.0;
}
virtual void update(void);
private:
const int16_t *wavetable;
uint16_t magnitude;
uint32_t phase;
uint32_t phase_increment;
};




#if 0
class AudioSineWaveMod : public AudioStream
{
public:
AudioSineWaveMod() : AudioStream(1, inputQueueArray) {}
void frequency(float freq);
//void amplitude(q15 n);
virtual void update(void);
private:
uint32_t phase;
uint32_t phase_increment;
uint32_t modulation_factor;
audio_block_t *inputQueueArray[1];
};
#endif





class AudioOutputPWM : public AudioStream
{
public:
AudioOutputPWM(void) : AudioStream(1, inputQueueArray) { begin(); }
virtual void update(void);
void begin(void);
friend void dma_ch3_isr(void);
private:
static audio_block_t *block_1st;
static audio_block_t *block_2nd;
static uint32_t block_offset;
static bool update_responsibility;
static uint8_t interrupt_count;
audio_block_t *inputQueueArray[1];
};





class AudioOutputAnalog : public AudioStream
{
public:
AudioOutputAnalog(void) : AudioStream(1, inputQueueArray) { begin(); }
virtual void update(void);
void begin(void);
void analogReference(int ref);
friend void dma_ch4_isr(void);
private:
static audio_block_t *block_left_1st;
static audio_block_t *block_left_2nd;
static bool update_responsibility;
audio_block_t *inputQueueArray[1];
};





class AudioPrint : public AudioStream
{
public:
AudioPrint(const char *str) : AudioStream(1, inputQueueArray), name(str) {}
virtual void update(void);
private:
const char *name;
audio_block_t *inputQueueArray[1];
};





















class AudioInputI2S : public AudioStream
{
public:
AudioInputI2S(void) : AudioStream(0, NULL) { begin(); }
virtual void update(void);
void begin(void);
friend void dma_ch1_isr(void);
protected:
AudioInputI2S(int dummy): AudioStream(0, NULL) {} // to be used only inside AudioInputI2Sslave !!
static bool update_responsibility;
private:
static audio_block_t *block_left;
static audio_block_t *block_right;
static uint16_t block_offset;
};


class AudioOutputI2S : public AudioStream
{
public:
AudioOutputI2S(void) : AudioStream(2, inputQueueArray) { begin(); }
virtual void update(void);
void begin(void);
friend void dma_ch0_isr(void);
friend class AudioInputI2S;
protected:
AudioOutputI2S(int dummy): AudioStream(2, inputQueueArray) {} // to be used only inside AudioOutputI2Sslave !!
static void config_i2s(void);
static audio_block_t *block_left_1st;
static audio_block_t *block_right_1st;
static bool update_responsibility;
private:
static audio_block_t *block_left_2nd;
static audio_block_t *block_right_2nd;
static uint16_t block_left_offset;
static uint16_t block_right_offset;
audio_block_t *inputQueueArray[2];
};


class AudioInputI2Sslave : public AudioInputI2S
{
public:
AudioInputI2Sslave(void) : AudioInputI2S(0) { begin(); }
void begin(void);
friend void dma_ch1_isr(void);
};


class AudioOutputI2Sslave : public AudioOutputI2S
{
public:
AudioOutputI2Sslave(void) : AudioOutputI2S(0) { begin(); } ;
void begin(void);
friend class AudioInputI2Sslave;
friend void dma_ch0_isr(void);
protected:
static void config_i2s(void);
};





class AudioInputAnalog : public AudioStream
{
public:
AudioInputAnalog(unsigned int pin) : AudioStream(0, NULL) { begin(pin); }
virtual void update(void);
void begin(unsigned int pin);
friend void dma_ch2_isr(void);
private:
static audio_block_t *block_left;
static uint16_t block_offset;
uint16_t dc_average;
static bool update_responsibility;
};




















#include "SD.h"

class AudioPlaySDcardWAV : public AudioStream
{
public:
AudioPlaySDcardWAV(void) : AudioStream(0, NULL) { begin(); }
void begin(void);

bool play(const char *filename);
void stop(void);
bool start(void);
virtual void update(void);
private:
File wavfile;
bool consume(void);
bool parse_format(void);
uint32_t header[5];
uint32_t data_length; // number of bytes remaining in data section
audio_block_t *block_left;
audio_block_t *block_right;
uint16_t block_offset;
uint8_t buffer[512];
uint16_t buffer_remaining;
uint8_t state;
uint8_t state_play;
uint8_t leftover_bytes;
};


class AudioPlaySDcardRAW : public AudioStream
{
public:
AudioPlaySDcardRAW(void) : AudioStream(0, NULL) { begin(); }
void begin(void);
bool play(const char *filename);
void stop(void);
virtual void update(void);
private:
File rawfile;
audio_block_t *block;
bool playing;
bool paused;
};



class AudioPlayMemory : public AudioStream
{
public:
AudioPlayMemory(void) : AudioStream(0, NULL), playing(0) { }
void play(const unsigned int *data);
void stop(void);
virtual void update(void);
private:
const unsigned int *next;
uint32_t length;
int16_t prior;
volatile uint8_t playing;
};










class AudioMixer4 : public AudioStream
{
public:
AudioMixer4(void) : AudioStream(4, inputQueueArray) {
for (int i=0; i<4; i++) multiplier[i] = 65536;
}
virtual void update(void);
void gain(unsigned int channel, float gain) {
if (channel >= 4) return;
if (gain > 32767.0f) gain = 32767.0f;
else if (gain < 0.0f) gain = 0.0f;
multiplier[channel] = gain * 65536.0f; // TODO: proper roundoff?
}
private:
int32_t multiplier[4];
audio_block_t *inputQueueArray[4];
};






class AudioFilterBiquad : public AudioStream
{
public:
AudioFilterBiquad(int *parameters)
: AudioStream(1, inputQueueArray), definition(parameters) { }
virtual void update(void);
private:
int *definition;
audio_block_t *inputQueueArray[1];
};



class AudioEffectFade : public AudioStream
{
public:
AudioEffectFade(void)
: AudioStream(1, inputQueueArray), position(0xFFFFFFFF) {}
void fadeIn(uint32_t milliseconds) {
uint32_t samples = (uint32_t)(milliseconds * 441u + 5u) / 10u;
//Serial.printf("fadeIn, %u samples\n", samples);
fadeBegin(0xFFFFFFFFu / samples, 1);
}
void fadeOut(uint32_t milliseconds) {
uint32_t samples = (uint32_t)(milliseconds * 441u + 5u) / 10u;
//Serial.printf("fadeOut, %u samples\n", samples);
fadeBegin(0xFFFFFFFFu / samples, 0);
}
virtual void update(void);
private:
void fadeBegin(uint32_t newrate, uint8_t dir);
uint32_t position; // 0 = off, 0xFFFFFFFF = on
uint32_t rate;
uint8_t direction; // 0 = fading out, 1 = fading in
audio_block_t *inputQueueArray[1];
};



class AudioAnalyzeToneDetect : public AudioStream
{
public:
AudioAnalyzeToneDetect(void)
: AudioStream(1, inputQueueArray), thresh(6554), enabled(false) { }
void frequency(float freq, uint16_t cycles=10) {
set_params((int32_t)(cos((double)freq
* (2.0 * 3.14159265358979323846 / AUDIO_SAMPLE_RATE_EXACT))
* (double)2147483647.999), cycles,
(float)AUDIO_SAMPLE_RATE_EXACT / freq * (float)cycles + 0.5f);
}
void set_params(int32_t coef, uint16_t cycles, uint16_t len);
bool available(void) {
__disable_irq();
bool flag = new_output;
if (flag) new_output = false;
__enable_irq();
return flag;
}
float read(void);
void threshold(float level) {
if (level < 0.01f) thresh = 655;
else if (level > 0.99f) thresh = 64881;
else thresh = level * 65536.0f + 0.5f;
}
operator bool(); // true if at or above threshold, false if below
virtual void update(void);
private:
int32_t coefficient; // Goertzel algorithm coefficient
int32_t s1, s2; // Goertzel algorithm state
int32_t out1, out2; // Goertzel algorithm state output
uint16_t length; // number of samples to analyze
uint16_t count; // how many left to analyze
uint16_t ncycles; // number of waveform cycles to seek
uint16_t thresh; // threshold, 655 to 64881 (1% to 99%)
bool enabled;
volatile bool new_output;
audio_block_t *inputQueueArray[1];
};







// TODO: more audio processing objects....
// sine wave with frequency modulation (phase)
// waveforms with bandwidth limited tables for synth
// envelope: attack-decay-sustain-release, maybe other more complex?
// MP3 decoding - it is possible with optimized code?
// other decompression, ADPCM, Vorbis, Speex, etc?




// A base class for all Codecs, DACs and ADCs, so at least the
// most basic functionality is consistent.

#define AUDIO_INPUT_LINEIN 0
#define AUDIO_INPUT_MIC 1

class AudioControl
{
public:
virtual bool enable(void) = 0;
virtual bool disable(void) = 0;
virtual bool volume(float volume) = 0; // volume 0.0 to 100.0
virtual bool inputLevel(float volume) = 0; // volume 0.0 to 100.0
virtual bool inputSelect(int n) = 0;
};



class AudioControlWM8731 : public AudioControl
{
public:
bool enable(void);
bool disable(void) { return false; }
bool volume(float n) { return volumeInteger(n * 0.8 + 47.499); }
bool inputLevel(float n) { return false; }
bool inputSelect(int n) { return false; }
protected:
bool write(unsigned int reg, unsigned int val);
bool volumeInteger(unsigned int n); // range: 0x2F to 0x7F
};

class AudioControlWM8731master : public AudioControlWM8731
{
public:
bool enable(void);
};


class AudioControlSGTL5000 : public AudioControl
{
public:
bool enable(void);
bool disable(void) { return false; }
bool volume(float n) { return volumeInteger(n * 1.29 + 0.499); }
bool inputLevel(float n) {return false;}
bool muteHeadphone(void) { return write(0x0024, ana_ctrl | (1<<4)); }
bool unmuteHeadphone(void) { return write(0x0024, ana_ctrl & ~(1<<4)); }
bool muteLineout(void) { return write(0x0024, ana_ctrl | (1<<8)); }
bool unmuteLineout(void) { return write(0x0024, ana_ctrl & ~(1<<8)); }
bool inputSelect(int n) {
if (n == AUDIO_INPUT_LINEIN) {
return write(0x0024, ana_ctrl | (1<<2));
} else if (n == AUDIO_INPUT_MIC) {
//return write(0x002A, 0x0172) && write(0x0024, ana_ctrl & ~(1<<2));
return write(0x002A, 0x0173) && write(0x0024, ana_ctrl & ~(1<<2)); // +40dB
} else {
return false;
}
}
//bool inputLinein(void) { return write(0x0024, ana_ctrl | (1<<2)); }
//bool inputMic(void) { return write(0x002A, 0x0172) && write(0x0024, ana_ctrl & ~(1<<2)); }
protected:
bool muted;
bool volumeInteger(unsigned int n); // range: 0x00 to 0x80
uint16_t ana_ctrl;



unsigned int read(unsigned int reg);
bool write(unsigned int reg, unsigned int val);
};



/******************************************************************/
/******************************************************************/

// Maximum number of coefficients in a FIR filter
// The audio breaks up with 128 coefficients so a
// maximum of 150 is more than sufficient
#define MAX_COEFFS 150

// Indicates that the code should just pass through the audio
// without any filtering (as opposed to doing nothing at all)
#define FIR_PASSTHRU ((short *) 1)

class AudioFilterFIR :
public AudioStream
{
public:
AudioFilterFIR(void):
AudioStream(2,inputQueueArray) {
}

void begin(short *coeff_p,int f_pin);
virtual void update(void);
void stop(void);
private:
audio_block_t *inputQueueArray[2];
static q15_t l_StateQ15[];
static q15_t r_StateQ15[];
static arm_fir_instance_q15 l_fir_inst;
static arm_fir_instance_q15 r_fir_inst;
static short *coeff_p;
};



/******************************************************************/
/******************************************************************/






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