teensy-audio/Audio.cpp

#include "Audio.h"
#include "arm_math.h"




static arm_cfft_radix4_instance_q15 fft_inst;

void AudioAnalyzeFFT256::init(void)
{
	// TODO: replace this with static const version
	arm_cfft_radix4_init_q15(&fft_inst, 256, 0, 1);

	//for (int i=0; i<2048; i++) {
		//buffer[i] = i * 3;
	//}
	//__disable_irq();
	//ARM_DEMCR |= ARM_DEMCR_TRCENA;
	//ARM_DWT_CTRL |= ARM_DWT_CTRL_CYCCNTENA;
	//uint32_t n = ARM_DWT_CYCCNT;
	//arm_cfft_radix2_q15(&fft_inst, buffer);
	//n = ARM_DWT_CYCCNT - n;
	//__enable_irq();
	//cycles = n;
	//arm_cmplx_mag_q15(buffer, buffer, 512);

	// each audio block is 278525 cycles @ 96 MHz
	// 256 point fft2 takes 65408 cycles
	// 256 point fft4 takes 49108 cycles
	// 128 point cmag takes 10999 cycles
	// 1024 point fft2 takes 125948 cycles
	// 1024 point fft4 takes 125840 cycles
	// 512 point cmag takes 43764 cycles

	//state = 0;
	//outputflag = false;
}

static void copy_to_fft_buffer(void *destination, const void *source)
{
	const int16_t *src = (const int16_t *)source;
	int16_t *dst = (int16_t *)destination;

	// TODO: optimize this
	for (int i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
		*dst++ = *src++;  // real
		*dst++ = 0;       // imaginary
	}
}

// computes limit((val >> rshift), 2**bits)
static inline int32_t signed_saturate_rshift(int32_t val, int bits, int rshift) __attribute__((always_inline));
static inline int32_t signed_saturate_rshift(int32_t val, int bits, int rshift)
{
	int32_t out;
	asm volatile("ssat %0, %1, %2, asr %3" : "=r" (out) : "I" (bits), "r" (val), "I" (rshift));
	return out;
}

static void apply_window_to_fft_buffer(void *buffer, const void *window)
{
	int16_t *buf = (int16_t *)buffer;
	const int16_t *win = (int16_t *)window;;

	for (int i=0; i < 256; i++) {
		int32_t val = *buf * *win++;
		//*buf = signed_saturate_rshift(val, 16, 15);
		*buf = val >> 15;
		buf += 2;
	}

}

void AudioAnalyzeFFT256::update(void)
{
	audio_block_t *block;

	block = receiveReadOnly();
	if (!block) return;
	if (!prevblock) {
		prevblock = block;
		return;
	}
	copy_to_fft_buffer(buffer, prevblock->data);
	copy_to_fft_buffer(buffer+256, block->data);
	//window = AudioWindowBlackmanNuttall256;
	//window = NULL;
	if (window) apply_window_to_fft_buffer(buffer, window);
	arm_cfft_radix4_q15(&fft_inst, buffer);
	// TODO: is this averaging correct?  G. Heinzel's paper says we're
	// supposed to average the magnitude squared, then do the square
	// root at the end after dividing by naverage.
	arm_cmplx_mag_q15(buffer, buffer, 128);
	if (count == 0) {
		for (int i=0; i < 128; i++) {
			output[i] = buffer[i];
		}
	} else {
		for (int i=0; i < 128; i++) {
			output[i] += buffer[i];
		}
	}
	if (++count == naverage) {
		count = 0;
		for (int i=0; i < 128; i++) {
			output[i] /= naverage;
		}
		outputflag = true;
	}

	release(prevblock);
	prevblock = block;

#if 0
	block = receiveReadOnly();
	if (state == 0) {
		//Serial.print("0");
		if (block == NULL) return;
		copy_to_fft_buffer(buffer, block->data);
		state = 1;
	} else if (state == 1) {
		//Serial.print("1");
		if (block == NULL) return;
		copy_to_fft_buffer(buffer+256, block->data);
		arm_cfft_radix4_q15(&fft_inst, buffer);
		state = 2;
	} else {
		//Serial.print("2");
		arm_cmplx_mag_q15(buffer, output, 128);
		outputflag = true;
		if (block == NULL) return;
		copy_to_fft_buffer(buffer, block->data);
		state = 1;
	}
	release(block);
#endif
}



#ifdef ORIGINAL_AUDIOSYNTHWAVEFORM
/******************************************************************/
// PAH - add ramp-up and ramp-down to the onset of the wave
// the length is specified in samples
void AudioSynthWaveform::set_ramp_length(uint16_t r_length)
{
	if(r_length < 0) {
		ramp_length = 0;
		return;
	}
	// Don't set the ramp length longer than about 4 milliseconds
	if(r_length > 44*4) {
		ramp_length = 44*4;
		return;
	}
	ramp_length = r_length;
}

void AudioSynthWaveform::update(void)
{
	audio_block_t *block;
	uint32_t i, ph, inc, index, scale;
	int32_t val1, val2, val3;

	//Serial.println("AudioSynthWaveform::update");
	if (((magnitude > 0) || ramp_down) && (block = allocate()) != NULL) {
		ph = phase;
		inc = phase_increment;
		for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
			index = ph >> 24;
			val1 = wavetable[index];
			val2 = wavetable[index+1];
			scale = (ph >> 8) & 0xFFFF;
			val2 *= scale;
			val1 *= 0xFFFF - scale;
			val3 = (val1 + val2) >> 16;


// The value of ramp_up is always initialized to RAMP_LENGTH and then is
// decremented each time through here until it reaches zero.
// The value of ramp_up is used to generate a Q15 fraction which varies
// from [0 - 1), and multiplies this by the current sample
			if(ramp_up) {
				// ramp up to the new magnitude
				// ramp_mag is the Q15 representation of the fraction
				// Since ramp_up can't be zero, this cannot generate +1
				ramp_mag = ((ramp_length-ramp_up)<<15)/ramp_length;
				ramp_up--;
				block->data[i] = (val3 * ((ramp_mag * magnitude)>>15)) >> 15;

			} else if(ramp_down) {
				// ramp down to zero from the last magnitude
// The value of ramp_down is always initialized to RAMP_LENGTH and then is
// decremented each time through here until it reaches zero.
// The value of ramp_down is used to generate a Q15 fraction which varies
// from (1 - 0], and multiplies this by the current sample
				// avoid RAMP_LENGTH/RAMP_LENGTH because Q15 format
				// cannot represent +1
				ramp_mag = ((ramp_down - 1)<<15)/ramp_length;
				ramp_down--;
				block->data[i] = (val3 * ((ramp_mag * last_magnitude)>>15)) >> 15;
			} else {			
				block->data[i] = (val3 * magnitude) >> 15;
			}

			 //Serial.print(block->data[i]);
			 //Serial.print(", ");
			 //if ((i % 12) == 11) Serial.println();
			ph += inc;
		}
		 //Serial.println();
		phase = ph;
		transmit(block);
		release(block);
	} else {
		// is this numerical overflow ok?
		phase += phase_increment * AUDIO_BLOCK_SAMPLES;
	}
}
#else
/******************************************************************/
// PAH - add ramp-up and ramp-down to the onset of the wave
// the length is specified in samples
void AudioSynthWaveform::set_ramp_length(uint16_t r_length)
{
  if(r_length < 0) {
    ramp_length = 0;
    return;
  }
  // Don't set the ramp length longer than about 4 milliseconds
  if(r_length > 44*4) {
    ramp_length = 44*4;
    return;
  }
  ramp_length = r_length;
}

boolean AudioSynthWaveform::begin(float t_amp,int t_hi,short type)
{
  tone_type = type;
//  tone_amp = t_amp;
  amplitude(t_amp);
  tone_freq = t_hi;
  if(t_hi < 1)return false;
  if(t_hi >= AUDIO_SAMPLE_RATE_EXACT/2)return false;
  tone_phase = 0;
  tone_incr = (0x100000000LL*t_hi)/AUDIO_SAMPLE_RATE_EXACT;
  if(0) {
    Serial.print("AudioSynthWaveform.begin(tone_amp = ");
    Serial.print(t_amp);
    Serial.print(", tone_hi = ");
    Serial.print(t_hi);
    Serial.print(", tone_incr = ");
    Serial.print(tone_incr,HEX);
    //  Serial.print(", tone_hi = ");
    //  Serial.print(t_hi);
    Serial.println(")");
  }
  return(true);
}


void AudioSynthWaveform::update(void)
{
  audio_block_t *block;
  short *bp;
  // temporary for ramp in sine
  uint32_t ramp_mag;
  // temporaries for TRIANGLE
  uint32_t mag;
  short tmp_amp;

  
  if(tone_freq == 0)return;
  //          L E F T  C H A N N E L  O N L Y
  block = allocate();
  if(block) {
    bp = block->data;
    switch(tone_type) {
    case TONE_TYPE_SINE:
      for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
        // The value of ramp_up is always initialized to RAMP_LENGTH and then is
        // decremented each time through here until it reaches zero.
        // The value of ramp_up is used to generate a Q15 fraction which varies
        // from [0 - 1), and multiplies this by the current sample
        if(ramp_up) {
          // ramp up to the new magnitude
          // ramp_mag is the Q15 representation of the fraction
          // Since ramp_up can't be zero, this cannot generate +1
          ramp_mag = ((ramp_length-ramp_up)<<15)/ramp_length;
          ramp_up--;
          // adjust tone_phase to Q15 format and then adjust the result
          // of the multiplication
          *bp = (short)((arm_sin_q15(tone_phase>>17) * tone_amp) >> 15);
          *bp++ = (*bp * ramp_mag)>>15;
        } 
        else if(ramp_down) {
          // ramp down to zero from the last magnitude
          // The value of ramp_down is always initialized to RAMP_LENGTH and then is
          // decremented each time through here until it reaches zero.
          // The value of ramp_down is used to generate a Q15 fraction which varies
          // from (1 - 0], and multiplies this by the current sample
          // avoid RAMP_LENGTH/RAMP_LENGTH because Q15 format
          // cannot represent +1
          ramp_mag = ((ramp_down - 1)<<15)/ramp_length;
          ramp_down--;
          // adjust tone_phase to Q15 format and then adjust the result
          // of the multiplication
          *bp = (short)((arm_sin_q15(tone_phase>>17) * last_tone_amp) >> 15);
          *bp++ = (*bp * ramp_mag)>>15;
        } else {
          // adjust tone_phase to Q15 format and then adjust the result
          // of the multiplication
          *bp++ = (short)((arm_sin_q15(tone_phase>>17) * tone_amp) >> 15);
        } 
        
        // phase and incr are both unsigned 32-bit fractions
        tone_phase += tone_incr;
      }
      break;
      
    case TONE_TYPE_SQUARE:
      for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
        if(tone_phase & 0x80000000)*bp++ = -tone_amp;
        else *bp++ = tone_amp;
        // phase and incr are both unsigned 32-bit fractions
        tone_phase += tone_incr;
      }
      break;
      
    case TONE_TYPE_SAWTOOTH:
      for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
        *bp = ((short)(tone_phase>>16)*tone_amp) >> 15;
        bp++;
        // phase and incr are both unsigned 32-bit fractions
        tone_phase += tone_incr;
      }
      break;

    case TONE_TYPE_TRIANGLE:
      for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
        if(tone_phase & 0x80000000) {
          // negative half-cycle
          tmp_amp = -tone_amp;
        } 
        else {
          // positive half-cycle
          tmp_amp = tone_amp;
        }
        mag = tone_phase << 2;
        // Determine which quadrant
        if(tone_phase & 0x40000000) {
          // negate the magnitude
          mag = ~mag + 1;
        }
        *bp++ = ((short)(mag>>17)*tmp_amp) >> 15;
        tone_phase += tone_incr;
      }
      break;
    }
    // send the samples to the left channel
    transmit(block,0);
    release(block);
  }
}


#endif








#if 0
void AudioSineWaveMod::frequency(float f)
{
	if (f > AUDIO_SAMPLE_RATE_EXACT / 2 || f < 0.0) return;
	phase_increment = (f / AUDIO_SAMPLE_RATE_EXACT) * 4294967296.0f;
}

void AudioSineWaveMod::update(void)
{
	audio_block_t *block, *modinput;
	uint32_t i, ph, inc, index, scale;
	int32_t val1, val2;

	//Serial.println("AudioSineWave::update");
	modinput = receiveReadOnly();
	ph = phase;
	inc = phase_increment;
	block = allocate();
	if (!block) {
		// unable to allocate memory, so we'll send nothing
		if (modinput) {
			// but if we got modulation data, update the phase
			for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
				ph += inc + modinput->data[i] * modulation_factor;
			}
			release(modinput);
		} else {
			ph += phase_increment * AUDIO_BLOCK_SAMPLES;
		}
		phase = ph;
		return;
	}
	if (modinput) {
		for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
			index = ph >> 24;
			val1 = sine_table[index];
			val2 = sine_table[index+1];
			scale = (ph >> 8) & 0xFFFF;
			val2 *= scale;
			val1 *= 0xFFFF - scale;
			block->data[i] = (val1 + val2) >> 16;
			 //Serial.print(block->data[i]);
			 //Serial.print(", ");
			 //if ((i % 12) == 11) Serial.println();
			ph += inc + modinput->data[i] * modulation_factor;
		}
		release(modinput);
	} else {
		ph = phase;
		inc = phase_increment;
		for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
			index = ph >> 24;
			val1 = sine_table[index];
			val2 = sine_table[index+1];
			scale = (ph >> 8) & 0xFFFF;
			val2 *= scale;
			val1 *= 0xFFFF - scale;
			block->data[i] = (val1 + val2) >> 16;
			 //Serial.print(block->data[i]);
			 //Serial.print(", ");
			 //if ((i % 12) == 11) Serial.println();
			ph += inc;
		}
	}
	phase = ph;
	transmit(block);
	release(block);
}
#endif






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


void AudioPrint::update(void)
{
	audio_block_t *block;
	uint32_t i;

	Serial.println("AudioPrint::update");
	Serial.println(name);
	block = receiveReadOnly();
	if (block) {
		for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
			Serial.print(block->data[i]);
			Serial.print(", ");
			if ((i % 12) == 11) Serial.println();
		}
		Serial.println();
		release(block);
	}
}



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


#define STATE_DIRECT_8BIT_MONO		0  // playing mono at native sample rate
#define STATE_DIRECT_8BIT_STEREO	1  // playing stereo at native sample rate
#define STATE_DIRECT_16BIT_MONO		2  // playing mono at native sample rate
#define STATE_DIRECT_16BIT_STEREO	3  // playing stereo at native sample rate
#define STATE_CONVERT_8BIT_MONO		4  // playing mono, converting sample rate
#define STATE_CONVERT_8BIT_STEREO	5  // playing stereo, converting sample rate
#define STATE_CONVERT_16BIT_MONO	6  // playing mono, converting sample rate
#define STATE_CONVERT_16BIT_STEREO	7  // playing stereo, converting sample rate
#define STATE_PARSE1			8  // looking for 20 byte ID header
#define STATE_PARSE2			9  // looking for 16 byte format header
#define STATE_PARSE3			10 // looking for 8 byte data header
#define STATE_PARSE4			11 // ignoring unknown chunk
#define STATE_STOP			12

void AudioPlaySDcardWAV::begin(void)
{
	state = STATE_STOP;
	state_play = STATE_STOP;
	data_length = 0;
	if (block_left) {
		release(block_left);
		block_left = NULL;
	}
	if (block_right) {
		release(block_right);
		block_right = NULL;
	}
}


bool AudioPlaySDcardWAV::play(const char *filename)
{
	stop();
	wavfile = SD.open(filename);
	if (!wavfile) return false;
	buffer_remaining = 0;
	state_play = STATE_STOP;
	data_length = 0;
	state = STATE_PARSE1;
	return true;
}

void AudioPlaySDcardWAV::stop(void)
{
	__disable_irq();
	if (state != STATE_STOP) {
		audio_block_t *b1 = block_left;
		block_left = NULL;
		audio_block_t *b2 = block_right;
		block_right = NULL;
		state = STATE_STOP;
		__enable_irq();
		if (b1) release(b1);
		if (b2) release(b2);
		wavfile.close();
	} else {
		__enable_irq();
	}
}

bool AudioPlaySDcardWAV::start(void)
{
	__disable_irq();
	if (state == STATE_STOP) {
		if (state_play == STATE_STOP) {
			__enable_irq();
			return false;
		}
		state = state_play;
	}
	__enable_irq();
	return true;
}


void AudioPlaySDcardWAV::update(void)
{
	// only update if we're playing
	if (state == STATE_STOP) return;

	// allocate the audio blocks to transmit
	block_left = allocate();
	if (block_left == NULL) return;
	if (state < 8 && (state & 1) == 1) {
		// if we're playing stereo, allocate another
		// block for the right channel output
		block_right = allocate();
		if (block_right == NULL) {
			release(block_left);
			return;
		}
	} else {
		// if we're playing mono or just parsing
		// the WAV file header, no right-side block
		block_right = NULL;
	}
	block_offset = 0;

	//Serial.println("update");

	// is there buffered data?
	if (buffer_remaining > 0) {
		// we have buffered data
		if (consume()) return; // it was enough to transmit audio
	}

	// we only get to this point when buffer[512] is empty
	if (state != STATE_STOP && wavfile.available()) {
		// we can read more data from the file...
		buffer_remaining = wavfile.read(buffer, 512);
		if (consume()) {
			// good, it resulted in audio transmit
			return;
		} else {
			// not good, no audio was transmitted
			buffer_remaining = 0;
			if (block_left) {
				release(block_left);
				block_left = NULL;
			}
			if (block_right) {
				release(block_right);
				block_right = NULL;
			}
			// if we're still playing, well, there's going to
			// be a gap in output, but we can't keep burning
			// time trying to read more data.  Hopefully things
			// will go better next time?
			if (state != STATE_STOP) return;
		}
	}
	// end of file reached or other reason to stop
	wavfile.close();
	if (block_left) {
		release(block_left);
		block_left = NULL;
	}
	if (block_right) {
		release(block_right);
		block_right = NULL;
	}
	state_play = STATE_STOP;
	state = STATE_STOP;
}


// https://ccrma.stanford.edu/courses/422/projects/WaveFormat/

// Consume already buffered data.  Returns true if audio transmitted.
bool AudioPlaySDcardWAV::consume(void)
{
	uint32_t len, size;
	uint8_t lsb, msb;
	const uint8_t *p;

	size = buffer_remaining;
	p = buffer + 512 - size;
start:
	if (size == 0) return false;
	//Serial.print("AudioPlaySDcardWAV write, size = ");
	//Serial.print(size);
	//Serial.print(", data_length = ");
	//Serial.print(data_length);
	//Serial.print(", state = ");
	//Serial.println(state);
	switch (state) {
	  // parse wav file header, is this really a .wav file?
	  case STATE_PARSE1:
		len = 20 - data_length;
		if (size < len) len = size;
		memcpy((uint8_t *)header + data_length, p, len);
		data_length += len;
		if (data_length < 20) return false;
		// parse the header...
		if (header[0] == 0x46464952 && header[2] == 0x45564157
		  && header[3] == 0x20746D66 && header[4] == 16) {
			//Serial.println("header ok");
			state = STATE_PARSE2;
			p += len;
			size -= len;
			data_length = 0;
			goto start;
		}
		//Serial.println("unknown WAV header");
		break;

	  // check & extract key audio parameters
	  case STATE_PARSE2:
		len = 16 - data_length;
		if (size < len) len = size;
		memcpy((uint8_t *)header + data_length, p, len);
		data_length += len;
		if (data_length < 16) return false;
		if (parse_format()) {
			//Serial.println("audio format ok");
			p += len;
			size -= len;
			data_length = 0;
			state = STATE_PARSE3;
			goto start;
		}
		//Serial.println("unknown audio format");
		break;

	  // find the data chunk
	  case STATE_PARSE3:
		len = 8 - data_length;
		if (size < len) len = size;
		memcpy((uint8_t *)header + data_length, p, len);
		data_length += len;
		if (data_length < 8) return false;
		//Serial.print("chunk id = ");
		//Serial.print(header[0], HEX);
		//Serial.print(", length = ");
		//Serial.println(header[1]);
		p += len;
		size -= len;
		data_length = header[1];
		if (header[0] == 0x61746164) {
			//Serial.println("found data chunk");
			// TODO: verify offset in file is an even number
			// as required by WAV format.  abort if odd.  Code
			// below will depend upon this and fail if not even.
			leftover_bytes = 0;
			state = state_play;
			if (state & 1) {
				// if we're going to start stereo
				// better allocate another output block
				block_right = allocate();
				if (!block_right) return false;
			}
		} else {
			state = STATE_PARSE4;
		}
		goto start;

	  // ignore any extra unknown chunks (title & artist info)
	  case STATE_PARSE4:
		if (size < data_length) {
			data_length -= size;
			return false;
		}
		p += data_length;
		size -= data_length;
		data_length = 0;
		state = STATE_PARSE3;
		goto start;

	  // playing mono at native sample rate
	  case STATE_DIRECT_8BIT_MONO:
		return false;

	  // playing stereo at native sample rate
	  case STATE_DIRECT_8BIT_STEREO:
		return false;

	  // playing mono at native sample rate
	  case STATE_DIRECT_16BIT_MONO:
		if (size > data_length) size = data_length;
		data_length -= size;
		while (1) {
			lsb = *p++;
			msb = *p++;
			size -= 2;
			block_left->data[block_offset++] = (msb << 8) | lsb;
			if (block_offset >= AUDIO_BLOCK_SAMPLES) {
				transmit(block_left, 0);
				transmit(block_left, 1);
				 //Serial1.print('%');
				 //delayMicroseconds(90);
				release(block_left);
				block_left = NULL;
				data_length += size;
				buffer_remaining = size;
				if (block_right) release(block_right);
				return true;
			}
			if (size == 0) {
				if (data_length == 0) break;
				return false;
			}
		}
		// end of file reached
		if (block_offset > 0) {
			// TODO: fill remainder of last block with zero and transmit
		}
		state = STATE_STOP;
		return false;

	  // playing stereo at native sample rate
	  case STATE_DIRECT_16BIT_STEREO:
		if (size > data_length) size = data_length;
		data_length -= size;
		if (leftover_bytes) {
			block_left->data[block_offset] = header[0];
			goto right16;
		}
		while (1) {
			lsb = *p++;
			msb = *p++;
			size -= 2;
			if (size == 0) {
				if (data_length == 0) break;
				header[0] = (msb << 8) | lsb;
				leftover_bytes = 2;
				return false;
			}
			block_left->data[block_offset] = (msb << 8) | lsb;
			right16:
			lsb = *p++;
			msb = *p++;
			size -= 2;
			block_right->data[block_offset++] = (msb << 8) | lsb;
			if (block_offset >= AUDIO_BLOCK_SAMPLES) {
				transmit(block_left, 0);
				release(block_left);
				block_left = NULL;
				transmit(block_right, 1);
				release(block_right);
				block_right = NULL;
				data_length += size;
				buffer_remaining = size;
				return true;
			}
			if (size == 0) {
				if (data_length == 0) break;
				leftover_bytes = 0;
				return false;
			}
		}
		// end of file reached
		if (block_offset > 0) {
			// TODO: fill remainder of last block with zero and transmit
		}
		state = STATE_STOP;
		return false;

	  // playing mono, converting sample rate
	  case STATE_CONVERT_8BIT_MONO :
		return false;

	  // playing stereo, converting sample rate
	  case STATE_CONVERT_8BIT_STEREO:
		return false;

	  // playing mono, converting sample rate
	  case STATE_CONVERT_16BIT_MONO:
		return false;

	  // playing stereo, converting sample rate
	  case STATE_CONVERT_16BIT_STEREO:
		return false;

	  // ignore any extra data after playing
	  // or anything following any error
	  case STATE_STOP:
		return false;

	  // this is not supposed to happen!
	  //default:
		//Serial.println("AudioPlaySDcardWAV, unknown state");
	}
	state_play = STATE_STOP;
	state = STATE_STOP;
	return false;
}


/*
00000000  52494646 66EA6903 57415645 666D7420  RIFFf.i.WAVEfmt 
00000010  10000000 01000200 44AC0000 10B10200  ........D.......
00000020  04001000 4C495354 3A000000 494E464F  ....LIST:...INFO
00000030  494E414D 14000000 49205761 6E742054  INAM....I Want T
00000040  6F20436F 6D65204F 76657200 49415254  o Come Over.IART
00000050  12000000 4D656C69 73736120 45746865  ....Melissa Ethe
00000060  72696467 65006461 746100EA 69030100  ridge.data..i...
00000070  FEFF0300 FCFF0400 FDFF0200 0000FEFF  ................
00000080  0300FDFF 0200FFFF 00000100 FEFF0300  ................
00000090  FDFF0300 FDFF0200 FFFF0100 0000FFFF  ................
*/





// SD library on Teensy3 at 96 MHz
//  256 byte chunks, speed is 443272 bytes/sec
//  512 byte chunks, speed is 468023 bytes/sec





bool AudioPlaySDcardWAV::parse_format(void)
{
	uint8_t num = 0;
	uint16_t format;
	uint16_t channels;
	uint32_t rate;
	uint16_t bits;

	format = header[0];
	//Serial.print("  format = ");
	//Serial.println(format);
	if (format != 1) return false;

	channels = header[0] >> 16;
	//Serial.print("  channels = ");
	//Serial.println(channels);
	if (channels == 1) {
	} else if (channels == 2) {
		num = 1;
	} else {
		return false;
	}

	bits = header[3] >> 16;
	//Serial.print("  bits = ");
	//Serial.println(bits);
	if (bits == 8) {
	} else if (bits == 16) {
		num |= 2;
	} else {
		return false;
	}

	rate = header[1];
	//Serial.print("  rate = ");
	//Serial.println(rate);
	if (rate == AUDIO_SAMPLE_RATE) {
	} else if (rate >= 8000 && rate <= 48000) {
		num |= 4;
	} else {
		return false;
	}
	// we're not checking the byte rate and block align fields
	// if they're not the expected values, all we could do is
	// return false.  Do any real wav files have unexpected
	// values in these other fields?
	state_play = num;
	return true;
}



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


void AudioPlaySDcardRAW::begin(void)
{
	playing = false;
	if (block) {
		release(block);
		block = NULL;
	}
}


bool AudioPlaySDcardRAW::play(const char *filename)
{
	stop();
	rawfile = SD.open(filename);
	if (!rawfile) {
		Serial.println("unable to open file");
		return false;
	}
	Serial.println("able to open file");
	playing = true;
	return true;
}

void AudioPlaySDcardRAW::stop(void)
{
	__disable_irq();
	if (playing) {
		playing = false;
		__enable_irq();
		rawfile.close();
	} else {
		__enable_irq();
	}
}


void AudioPlaySDcardRAW::update(void)
{
	unsigned int i, n;

	// only update if we're playing
	if (!playing) return;

	// allocate the audio blocks to transmit
	block = allocate();
	if (block == NULL) return;

	if (rawfile.available()) {
		// we can read more data from the file...
		n = rawfile.read(block->data, AUDIO_BLOCK_SAMPLES*2);
		for (i=n/2; i < AUDIO_BLOCK_SAMPLES; i++) {
			block->data[i] = 0;
		}
		transmit(block);
		release(block);
	} else {
		rawfile.close();
		playing = false;
	}
}


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



void AudioPlayMemory::play(const unsigned int *data)
{
	uint32_t format;

	playing = 0;
	prior = 0;
	format = *data++;
	next = data;
	length = format & 0xFFFFFF;
	playing = format >> 24;
}

void AudioPlayMemory::stop(void)
{
	playing = 0;
}

extern "C" {
extern const int16_t ulaw_decode_table[256];
};

void AudioPlayMemory::update(void)
{
	audio_block_t *block;
	const unsigned int *in;
	int16_t *out;
	uint32_t tmp32, consumed;
	int16_t s0, s1, s2, s3, s4;
	int i;

	if (!playing) return;
	block = allocate();
	if (block == NULL) return;

	//Serial.write('.');

	out = block->data;
	in = next;
	s0 = prior;

	switch (playing) {
	  case 0x01: // u-law encoded, 44100 Hz
		for (i=0; i < AUDIO_BLOCK_SAMPLES; i += 4) {
			tmp32 = *in++;
			*out++ = ulaw_decode_table[(tmp32 >> 0) & 255];
			*out++ = ulaw_decode_table[(tmp32 >> 8) & 255];
			*out++ = ulaw_decode_table[(tmp32 >> 16) & 255];
			*out++ = ulaw_decode_table[(tmp32 >> 24) & 255];
		}
		consumed = 128;
		break;

	  case 0x81: // 16 bit PCM, 44100 Hz
		for (i=0; i < AUDIO_BLOCK_SAMPLES; i += 2) {
			tmp32 = *in++;
			*out++ = (int16_t)(tmp32 & 65535);
			*out++ = (int16_t)(tmp32 >> 16);
		}
		consumed = 128;
		break;

	  case 0x02: // u-law encoded, 22050 Hz 
		for (i=0; i < AUDIO_BLOCK_SAMPLES; i += 8) {
			tmp32 = *in++;
			s1 = ulaw_decode_table[(tmp32 >> 0) & 255];
			s2 = ulaw_decode_table[(tmp32 >> 8) & 255];
			s3 = ulaw_decode_table[(tmp32 >> 16) & 255];
			s4 = ulaw_decode_table[(tmp32 >> 24) & 255];
			*out++ = (s0 + s1) >> 1;
			*out++ = s1;
			*out++ = (s1 + s2) >> 1;
			*out++ = s2;
			*out++ = (s2 + s3) >> 1;
			*out++ = s3;
			*out++ = (s3 + s4) >> 1;
			*out++ = s4;
			s0 = s4;
		}
		consumed = 64;
		break;

	  case 0x82: // 16 bits PCM, 22050 Hz
		for (i=0; i < AUDIO_BLOCK_SAMPLES; i += 4) {
			tmp32 = *in++;
			s1 = (int16_t)(tmp32 & 65535);
			s2 = (int16_t)(tmp32 >> 16);
			*out++ = (s0 + s1) >> 1;
			*out++ = s1;
			*out++ = (s1 + s2) >> 1;
			*out++ = s2;
			s0 = s2;
		}
		consumed = 64;
		break;

	  case 0x03: // u-law encoded, 11025 Hz
		for (i=0; i < AUDIO_BLOCK_SAMPLES; i += 16) {
			tmp32 = *in++;
			s1 = ulaw_decode_table[(tmp32 >> 0) & 255];
			s2 = ulaw_decode_table[(tmp32 >> 8) & 255];
			s3 = ulaw_decode_table[(tmp32 >> 16) & 255];
			s4 = ulaw_decode_table[(tmp32 >> 24) & 255];
			*out++ = (s0 * 3 + s1) >> 2;
			*out++ = (s0 + s1)     >> 1;
			*out++ = (s0 + s1 * 3) >> 2;
			*out++ = s1;
			*out++ = (s1 * 3 + s2) >> 2;
			*out++ = (s1 + s2)     >> 1;
			*out++ = (s1 + s2 * 3) >> 2;
			*out++ = s2;
			*out++ = (s2 * 3 + s3) >> 2;
			*out++ = (s2 + s3)     >> 1;
			*out++ = (s2 + s3 * 3) >> 2;
			*out++ = s3;
			*out++ = (s3 * 3 + s4) >> 2;
			*out++ = (s3 + s4)     >> 1;
			*out++ = (s3 + s4 * 3) >> 2;
			*out++ = s4;
			s0 = s4;
		}
		consumed = 32;
		break;

	  case 0x83: // 16 bit PCM, 11025 Hz
		for (i=0; i < AUDIO_BLOCK_SAMPLES; i += 8) {
			tmp32 = *in++;
			s1 = (int16_t)(tmp32 & 65535);
			s2 = (int16_t)(tmp32 >> 16);
			*out++ = (s0 * 3 + s1) >> 2;
			*out++ = (s0 + s1)     >> 1;
			*out++ = (s0 + s1 * 3) >> 2;
			*out++ = s1;
			*out++ = (s1 * 3 + s2) >> 2;
			*out++ = (s1 + s2)     >> 1;
			*out++ = (s1 + s2 * 3) >> 2;
			*out++ = s2;
			s0 = s2;
		}
		consumed = 32;
		break;

	  default:
		release(block);
		playing = 0;
		return;
	}
	prior = s0;
	next = in;
	if (length > consumed) {
		length -= consumed;
	} else {
		playing = 0;
	}
	transmit(block);
	release(block);
}










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




// computes ((a[31:0] * b[15:0]) >> 16)
static inline int32_t signed_multiply_32x16b(int32_t a, uint32_t b) __attribute__((always_inline));
static inline int32_t signed_multiply_32x16b(int32_t a, uint32_t b)
{
	int32_t out;
	asm volatile("smulwb %0, %1, %2" : "=r" (out) : "r" (a), "r" (b));
	return out;
}

// computes ((a[31:0] * b[31:16]) >> 16)
static inline int32_t signed_multiply_32x16t(int32_t a, uint32_t b) __attribute__((always_inline));
static inline int32_t signed_multiply_32x16t(int32_t a, uint32_t b)
{
	int32_t out;
	asm volatile("smulwt %0, %1, %2" : "=r" (out) : "r" (a), "r" (b));
	return out;
}

// computes (((int64_t)a[31:0] * (int64_t)b[31:0]) >> 32)
static inline int32_t multiply_32x32_rshift32(int32_t a, int32_t b) __attribute__((always_inline));
static inline int32_t multiply_32x32_rshift32(int32_t a, int32_t b)
{
	int32_t out;
	asm volatile("smmul %0, %1, %2" : "=r" (out) : "r" (a), "r" (b));
	return out;
}

// computes (((int64_t)a[31:0] * (int64_t)b[31:0] + 0x8000000) >> 32)
static inline int32_t multiply_32x32_rshift32_rounded(int32_t a, int32_t b) __attribute__((always_inline));
static inline int32_t multiply_32x32_rshift32_rounded(int32_t a, int32_t b)
{
	int32_t out;
	asm volatile("smmulr %0, %1, %2" : "=r" (out) : "r" (a), "r" (b));
	return out;
}

// computes sum + (((int64_t)a[31:0] * (int64_t)b[31:0] + 0x8000000) >> 32)
static inline int32_t multiply_accumulate_32x32_rshift32_rounded(int32_t sum, int32_t a, int32_t b) __attribute__((always_inline));
static inline int32_t multiply_accumulate_32x32_rshift32_rounded(int32_t sum, int32_t a, int32_t b)
{
	int32_t out;
	asm volatile("smmlar %0, %2, %3, %1" : "=r" (out) : "r" (sum), "r" (a), "r" (b));
	return out;
}

// computes sum - (((int64_t)a[31:0] * (int64_t)b[31:0] + 0x8000000) >> 32)
static inline int32_t multiply_subtract_32x32_rshift32_rounded(int32_t sum, int32_t a, int32_t b) __attribute__((always_inline));
static inline int32_t multiply_subtract_32x32_rshift32_rounded(int32_t sum, int32_t a, int32_t b)
{
	int32_t out;
	asm volatile("smmlsr %0, %2, %3, %1" : "=r" (out) : "r" (sum), "r" (a), "r" (b));
	return out;
}

// computes ((a[15:0] << 16) | b[15:0])
static inline uint32_t pack_16x16(int32_t a, int32_t b) __attribute__((always_inline));
static inline uint32_t pack_16x16(int32_t a, int32_t b)
{
	int32_t out;
	asm volatile("pkhbt %0, %1, %2, lsl #16" : "=r" (out) : "r" (b), "r" (a));
	return out;
}

// computes (((a[31:16] + b[31:16]) << 16) | (a[15:0 + b[15:0]))
static inline uint32_t signed_add_16_and_16(uint32_t a, uint32_t b) __attribute__((always_inline));
static inline uint32_t signed_add_16_and_16(uint32_t a, uint32_t b)
{
	int32_t out;
	asm volatile("qadd16 %0, %1, %2" : "=r" (out) : "r" (a), "r" (b));
	return out;
}

// computes (sum + ((a[31:0] * b[15:0]) >> 16))
static inline int32_t signed_multiply_accumulate_32x16b(int32_t sum, int32_t a, uint32_t b)
{
	int32_t out;
	asm volatile("smlawb %0, %2, %3, %1" : "=r" (out) : "r" (sum), "r" (a), "r" (b));
	return out;
}

// computes (sum + ((a[31:0] * b[31:16]) >> 16))
static inline int32_t signed_multiply_accumulate_32x16t(int32_t sum, int32_t a, uint32_t b)
{
	int32_t out;
	asm volatile("smlawt %0, %2, %3, %1" : "=r" (out) : "r" (sum), "r" (a), "r" (b));
	return out;
}

// computes logical and, forces compiler to allocate register and use single cycle instruction
static inline uint32_t logical_and(uint32_t a, uint32_t b)
{
	asm volatile("and %0, %1" : "+r" (a) : "r" (b));
	return a;
}




void applyGain(int16_t *data, int32_t mult)
{
	uint32_t *p = (uint32_t *)data;
	const uint32_t *end = (uint32_t *)(data + AUDIO_BLOCK_SAMPLES);

	do {
		uint32_t tmp32 = *p; // read 2 samples from *data
		int32_t val1 = signed_multiply_32x16b(mult, tmp32);
		int32_t val2 = signed_multiply_32x16t(mult, tmp32);
		val1 = signed_saturate_rshift(val1, 16, 0);
		val2 = signed_saturate_rshift(val2, 16, 0);
		*p++ = pack_16x16(val2, val1);
	} while (p < end);
}

// page 133

void applyGainThenAdd(int16_t *data, const int16_t *in, int32_t mult)
{
	uint32_t *dst = (uint32_t *)data;
	const uint32_t *src = (uint32_t *)in;
	const uint32_t *end = (uint32_t *)(data + AUDIO_BLOCK_SAMPLES);

	if (mult == 65536) {
		do {
			uint32_t tmp32 = *dst;
			*dst++ = signed_add_16_and_16(tmp32, *src++);
			tmp32 = *dst;
			*dst++ = signed_add_16_and_16(tmp32, *src++);
		} while (dst < end);
	} else {
		do {
			uint32_t tmp32 = *src++; // read 2 samples from *data
			int32_t val1 = signed_multiply_32x16b(mult, tmp32);
			int32_t val2 = signed_multiply_32x16t(mult, tmp32);
			val1 = signed_saturate_rshift(val1, 16, 0);
			val2 = signed_saturate_rshift(val2, 16, 0);
			tmp32 = pack_16x16(val2, val1);
			uint32_t tmp32b = *dst;
			*dst++ = signed_add_16_and_16(tmp32, tmp32b);
		} while (dst < end);
	}
}


void AudioMixer4::update(void)
{
	audio_block_t *in, *out=NULL;
	unsigned int channel;

	for (channel=0; channel < 4; channel++) {
		if (!out) {
			out = receiveWritable(channel);
			if (out) {
				int32_t mult = multiplier[channel];
				if (mult != 65536) applyGain(out->data, mult);
			}
		} else {
			in = receiveReadOnly(channel);
			if (in) {
				applyGainThenAdd(out->data, in->data, multiplier[channel]);
				release(in);
			}
		}
	}
	if (out) {
		transmit(out);
		release(out);
	}
}


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





void AudioFilterBiquad::update(void)
{
	audio_block_t *block;
	int32_t a0, a1, a2, b1, b2, sum;
	uint32_t in2, out2, aprev, bprev, flag;
	uint32_t *data, *end;
	int32_t *state;

	block = receiveWritable();
	if (!block) return;
	data = (uint32_t *)(block->data);
	end = data + AUDIO_BLOCK_SAMPLES/2;
	state = (int32_t *)definition;
	do {
		a0 = *state++;
		a1 = *state++;
		a2 = *state++;
		b1 = *state++;
		b2 = *state++;
		aprev = *state++;
		bprev = *state++;
		sum = *state & 0x3FFF;
		do {
			in2 = *data;
			sum = signed_multiply_accumulate_32x16b(sum, a0, in2);
			sum = signed_multiply_accumulate_32x16t(sum, a1, aprev);
			sum = signed_multiply_accumulate_32x16b(sum, a2, aprev);
			sum = signed_multiply_accumulate_32x16t(sum, b1, bprev);
			sum = signed_multiply_accumulate_32x16b(sum, b2, bprev);
			out2 = (uint32_t)sum >> 14;
			sum &= 0x3FFF;
			sum = signed_multiply_accumulate_32x16t(sum, a0, in2);
			sum = signed_multiply_accumulate_32x16b(sum, a1, in2);
			sum = signed_multiply_accumulate_32x16t(sum, a2, aprev);
			sum = signed_multiply_accumulate_32x16b(sum, b1, out2);
			sum = signed_multiply_accumulate_32x16t(sum, b2, bprev);
			aprev = in2;
			bprev = pack_16x16(sum >> 14, out2);
			sum &= 0x3FFF;
			aprev = in2;
			*data++ = bprev;
		} while (data < end);
		flag = *state & 0x80000000;
		*state++ = sum | flag;
		*(state-2) = bprev;
		*(state-3) = aprev;
	} while (flag);
	transmit(block);
	release(block);
}

void AudioFilterBiquad::updateCoefs(int *source, bool doReset)
{
	int32_t *dest=(int32_t *)definition;
	int32_t *src=(int32_t *)source;
	__disable_irq();
	for(uint8_t index=0;index<5;index++)
	{
		*dest++=*src++;
	}
	if(doReset)
	{
		*dest++=0;
		*dest++=0;
		*dest++=0;
	}
	__enable_irq();
}

void AudioFilterBiquad::updateCoefs(int *source)
{
	updateCoefs(source,false);
}

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


extern "C" {
extern const int16_t fader_table[256];
};


void AudioEffectFade::update(void)
{
	audio_block_t *block;
	uint32_t i, pos, inc, index, scale;
	int32_t val1, val2, val, sample;
	uint8_t dir;

	pos = position;
	if (pos == 0) {
		// output is silent
		block = receiveReadOnly();
		if (block) release(block);
		return;
	} else if (pos == 0xFFFFFFFF) {
		// output is 100%
		block = receiveReadOnly();
		if (!block) return;
		transmit(block);
		release(block);
		return;
	}
	block = receiveWritable();
	if (!block) return;
	inc = rate;
	dir = direction;
	for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
		index = pos >> 24;
		val1 = fader_table[index];
		val2 = fader_table[index+1];
		scale = (pos >> 8) & 0xFFFF;
		val2 *= scale;
		val1 *= 0x10000 - scale;
		val = (val1 + val2) >> 16;
		sample = block->data[i];
		sample = (sample * val) >> 15;
		block->data[i] = sample;
		if (dir > 0) {
			// output is increasing
			if (inc < 0xFFFFFFFF - pos) pos += inc;
			else pos = 0xFFFFFFFF;
		} else {
			// output is decreasing
			if (inc < pos) pos -= inc;
			else pos = 0;
		}
	}
	position = pos;
	transmit(block);
	release(block);
}

void AudioEffectFade::fadeBegin(uint32_t newrate, uint8_t dir)
{
	__disable_irq();
	uint32_t pos = position;
	if (pos == 0) position = 1;
	else if (pos == 0xFFFFFFFF) position = 0xFFFFFFFE;
	rate = newrate;
	direction = dir;
	__enable_irq();
}



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


static inline int32_t multiply_32x32_rshift30(int32_t a, int32_t b) __attribute__((always_inline));
static inline int32_t multiply_32x32_rshift30(int32_t a, int32_t b)
{
	return ((int64_t)a * (int64_t)b) >> 30;
}

//#define TONE_DETECT_FAST

void AudioAnalyzeToneDetect::update(void)
{
	audio_block_t *block;
	int32_t q0, q1, q2, coef;
	const int16_t *p, *end;
	uint16_t n;

	block = receiveReadOnly();
	if (!block) return;
	if (!enabled) {
		release(block);
		return;
	}
	p = block->data;
	end = p + AUDIO_BLOCK_SAMPLES;
	n = count;
	coef = coefficient;
	q1 = s1;
	q2 = s2;
	do {
		// the Goertzel algorithm is kinda magical ;-)
#ifdef TONE_DETECT_FAST
		q0 = (*p++) + (multiply_32x32_rshift32_rounded(coef, q1) << 2) - q2;
#else
		q0 = (*p++) + multiply_32x32_rshift30(coef, q1) - q2;
		// TODO: is this only 1 cycle slower?  if so, always use it
#endif
		q2 = q1;
		q1 = q0;
		if (--n == 0) {
			out1 = q1;
			out2 = q2;
			q1 = 0;  // TODO: does clearing these help or hinder?
			q2 = 0;
			new_output = true;
			n = length;
		}
	} while (p < end);
	count = n;
	s1 = q1;
	s2 = q2;
	release(block);
}

void AudioAnalyzeToneDetect::set_params(int32_t coef, uint16_t cycles, uint16_t len)
{
	__disable_irq();
	coefficient = coef;
	ncycles = cycles;
	length = len;
	count = len;
	s1 = 0;
	s2 = 0;
	enabled = true;
	__enable_irq();
	//Serial.printf("Tone: coef=%d, ncycles=%d, length=%d\n", coefficient, ncycles, length);
}

float AudioAnalyzeToneDetect::read(void)
{
	int32_t coef, q1, q2, power;
	uint16_t len;

	__disable_irq();
	coef = coefficient;
	q1 = out1;
	q2 = out2;
	len = length;
	__enable_irq();
#ifdef TONE_DETECT_FAST
	power = multiply_32x32_rshift32_rounded(q2, q2);
	power = multiply_accumulate_32x32_rshift32_rounded(power, q1, q1);
	power = multiply_subtract_32x32_rshift32_rounded(power,
		multiply_32x32_rshift30(q1, q2), coef);
	power <<= 4;
#else
	int64_t power64;
	power64 = (int64_t)q2 * (int64_t)q2;
	power64 += (int64_t)q1 * (int64_t)q1;
	power64 -= (((int64_t)q1 * (int64_t)q2) >> 30) * (int64_t)coef;
	power = power64 >> 28;
#endif
	return sqrtf((float)power) / (float)len;
}


AudioAnalyzeToneDetect::operator bool()
{
	int32_t coef, q1, q2, power, trigger;
	uint16_t len;

	__disable_irq();
	coef = coefficient;
	q1 = out1;
	q2 = out2;
	len = length;
	__enable_irq();
#ifdef TONE_DETECT_FAST
	power = multiply_32x32_rshift32_rounded(q2, q2);
	power = multiply_accumulate_32x32_rshift32_rounded(power, q1, q1);
	power = multiply_subtract_32x32_rshift32_rounded(power,
		multiply_32x32_rshift30(q1, q2), coef);
	power <<= 4;
#else
	int64_t power64;
	power64 = (int64_t)q2 * (int64_t)q2;
	power64 += (int64_t)q1 * (int64_t)q1;
	power64 -= (((int64_t)q1 * (int64_t)q2) >> 30) * (int64_t)coef;
	power = power64 >> 28;
#endif
	trigger = (uint32_t)len * thresh;
	trigger = multiply_32x32_rshift32(trigger, trigger);

	Serial.printf("bool: power=%d, trig=%d\n", power, trigger);
	return (power >= trigger);
}




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

#include "Wire.h"

#define WM8731_I2C_ADDR 0x1A
//#define WM8731_I2C_ADDR 0x1B

#define WM8731_REG_LLINEIN	0
#define WM8731_REG_RLINEIN	1
#define WM8731_REG_LHEADOUT	2
#define WM8731_REG_RHEADOUT	3
#define WM8731_REG_ANALOG	4
#define WM8731_REG_DIGITAL	5
#define WM8731_REG_POWERDOWN	6
#define WM8731_REG_INTERFACE	7
#define WM8731_REG_SAMPLING	8
#define WM8731_REG_ACTIVE	9
#define WM8731_REG_RESET	15

bool AudioControlWM8731::enable(void)
{
	Wire.begin();
	delay(5);
	//write(WM8731_REG_RESET, 0);

	write(WM8731_REG_INTERFACE, 0x02); // I2S, 16 bit, MCLK slave
	write(WM8731_REG_SAMPLING, 0x20);  // 256*Fs, 44.1 kHz, MCLK/1

	// In order to prevent pops, the DAC should first be soft-muted (DACMU),
	// the output should then be de-selected from the line and headphone output
	// (DACSEL), then the DAC powered down (DACPD).

	write(WM8731_REG_DIGITAL, 0x08);   // DAC soft mute
	write(WM8731_REG_ANALOG, 0x00);    // disable all

	write(WM8731_REG_POWERDOWN, 0x00); // codec powerdown

	write(WM8731_REG_LHEADOUT, 0x80);      // volume off
	write(WM8731_REG_RHEADOUT, 0x80);

	delay(100); // how long to power up?

	write(WM8731_REG_ACTIVE, 1);
	delay(5);
	write(WM8731_REG_DIGITAL, 0x00);   // DAC unmuted
	write(WM8731_REG_ANALOG, 0x10);    // DAC selected

	return true;
}


bool AudioControlWM8731::write(unsigned int reg, unsigned int val)
{
	Wire.beginTransmission(WM8731_I2C_ADDR);
	Wire.write((reg << 1) | ((val >> 8) & 1));
	Wire.write(val & 0xFF);
	Wire.endTransmission();
	return true;
}

bool AudioControlWM8731::volumeInteger(unsigned int n)
{
	if (n > 127) n = 127;
	 //Serial.print("volumeInteger, n = ");
	 //Serial.println(n);
	write(WM8731_REG_LHEADOUT, n | 0x180);
	write(WM8731_REG_RHEADOUT, n | 0x80);
	return true;
}



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


bool AudioControlWM8731master::enable(void)
{
	Wire.begin();
	delay(5);
	//write(WM8731_REG_RESET, 0);

	write(WM8731_REG_INTERFACE, 0x42); // I2S, 16 bit, MCLK master
	write(WM8731_REG_SAMPLING, 0x20);  // 256*Fs, 44.1 kHz, MCLK/1

	// In order to prevent pops, the DAC should first be soft-muted (DACMU),
	// the output should then be de-selected from the line and headphone output
	// (DACSEL), then the DAC powered down (DACPD).

	write(WM8731_REG_DIGITAL, 0x08);   // DAC soft mute
	write(WM8731_REG_ANALOG, 0x00);    // disable all
	
	write(WM8731_REG_POWERDOWN, 0x00); // codec powerdown

	write(WM8731_REG_LHEADOUT, 0x80);      // volume off
	write(WM8731_REG_RHEADOUT, 0x80);

	delay(100); // how long to power up?

	write(WM8731_REG_ACTIVE, 1);
	delay(5);
	write(WM8731_REG_DIGITAL, 0x00);   // DAC unmuted
	write(WM8731_REG_ANALOG, 0x10);    // DAC selected

	return true;
}

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

#define CHIP_ID				0x0000
// 15:8 PARTID		0xA0 - 8 bit identifier for SGTL5000
// 7:0  REVID		0x00 - revision number for SGTL5000.

#define CHIP_DIG_POWER			0x0002
// 6	ADC_POWERUP	1=Enable, 0=disable the ADC block, both digital & analog, 
// 5	DAC_POWERUP	1=Enable, 0=disable the DAC block, both analog and digital
// 4	DAP_POWERUP	1=Enable, 0=disable the DAP block
// 1	I2S_OUT_POWERUP	1=Enable, 0=disable the I2S data output
// 0	I2S_IN_POWERUP	1=Enable, 0=disable the I2S data input

#define CHIP_CLK_CTRL			0x0004
// 5:4	RATE_MODE	Sets the sample rate mode. MCLK_FREQ is still specified
//			relative to the rate in SYS_FS
//				0x0 = SYS_FS specifies the rate
//				0x1 = Rate is 1/2 of the SYS_FS rate
//				0x2 = Rate is 1/4 of the SYS_FS rate
//				0x3 = Rate is 1/6 of the SYS_FS rate
// 3:2	SYS_FS		Sets the internal system sample rate (default=2)
//				0x0 = 32 kHz
//				0x1 = 44.1 kHz
//				0x2 = 48 kHz
//				0x3 = 96 kHz
// 1:0	MCLK_FREQ	Identifies incoming SYS_MCLK frequency and if the PLL should be used
//				0x0 = 256*Fs
//				0x1 = 384*Fs
//				0x2 = 512*Fs
//				0x3 = Use PLL
//				The 0x3 (Use PLL) setting must be used if the SYS_MCLK is not
//				a standard multiple of Fs (256, 384, or 512). This setting can
//				also be used if SYS_MCLK is a standard multiple of Fs.
//				Before this field is set to 0x3 (Use PLL), the PLL must be
//				powered up by setting CHIP_ANA_POWER->PLL_POWERUP and
//				CHIP_ANA_POWER->VCOAMP_POWERUP.  Also, the PLL dividers must
//				be calculated based on the external MCLK rate and
//				CHIP_PLL_CTRL register must be set (see CHIP_PLL_CTRL register
//				description details on how to calculate the divisors).

#define CHIP_I2S_CTRL			0x0006
// 8	SCLKFREQ	Sets frequency of I2S_SCLK when in master mode (MS=1). When in slave
//			mode (MS=0), this field must be set appropriately to match SCLK input
//			rate.
//				0x0 = 64Fs
//				0x1 = 32Fs - Not supported for RJ mode (I2S_MODE = 1)
// 7	MS		Configures master or slave of I2S_LRCLK and I2S_SCLK.
//				0x0 = Slave: I2S_LRCLK an I2S_SCLK are inputs
//				0x1 = Master: I2S_LRCLK and I2S_SCLK are outputs
//				NOTE: If the PLL is used (CHIP_CLK_CTRL->MCLK_FREQ==0x3),
//				the SGTL5000 must be a master of the I2S port (MS==1)
// 6	SCLK_INV	Sets the edge that data (input and output) is clocked in on for I2S_SCLK
//				0x0 = data is valid on rising edge of I2S_SCLK
//				0x1 = data is valid on falling edge of I2S_SCLK
// 5:4	DLEN		I2S data length (default=1)
//				0x0 = 32 bits (only valid when SCLKFREQ=0),
//					not valid for Right Justified Mode
//				0x1 = 24 bits (only valid when SCLKFREQ=0)
//				0x2 = 20 bits
//				0x3 = 16 bits
// 3:2	I2S_MODE	Sets the mode for the I2S port
//				0x0 = I2S mode or Left Justified (Use LRALIGN to select)
//				0x1 = Right Justified Mode
//				0x2 = PCM Format A/B
//				0x3 = RESERVED
// 1	LRALIGN		I2S_LRCLK Alignment to data word. Not used for Right Justified mode
//				0x0 = Data word starts 1 I2S_SCLK delay after I2S_LRCLK
//					transition (I2S format, PCM format A)
//				0x1 = Data word starts after I2S_LRCLK transition (left
//					justified format, PCM format B)
// 0	LRPOL		I2S_LRCLK Polarity when data is presented.
//				0x0 = I2S_LRCLK = 0 - Left, 1 - Right
//				1x0 = I2S_LRCLK = 0 - Right, 1 - Left
//				The left subframe should be presented first regardless of
//				the setting of LRPOL.

#define CHIP_SSS_CTRL			0x000A
// 14	DAP_MIX_LRSWAP	DAP Mixer Input Swap
//				0x0 = Normal Operation
//				0x1 = Left and Right channels for the DAP MIXER Input are swapped.
// 13	DAP_LRSWAP	DAP Mixer Input Swap
//				0x0 = Normal Operation
//				0x1 = Left and Right channels for the DAP Input are swapped
// 12	DAC_LRSWAP	DAC Input Swap
//				0x0 = Normal Operation
//				0x1 = Left and Right channels for the DAC are swapped
// 10	I2S_LRSWAP	I2S_DOUT Swap
//				0x0 = Normal Operation
//				0x1 = Left and Right channels for the I2S_DOUT are swapped
// 9:8	DAP_MIX_SELECT	Select data source for DAP mixer
//				0x0 = ADC
//				0x1 = I2S_IN
//				0x2 = Reserved
//				0x3 = Reserved
// 7:6	DAP_SELECT	Select data source for DAP
//				0x0 = ADC
//				0x1 = I2S_IN
//				0x2 = Reserved
//				0x3 = Reserved
// 5:4	DAC_SELECT	Select data source for DAC (default=1)
//				0x0 = ADC
//				0x1 = I2S_IN
//				0x2 = Reserved
//				0x3 = DAP
// 1:0	I2S_SELECT	Select data source for I2S_DOUT
//				0x0 = ADC
//				0x1 = I2S_IN
//				0x2 = Reserved
//				0x3 = DAP

#define CHIP_ADCDAC_CTRL		0x000E
// 13	VOL_BUSY_DAC_RIGHT Volume Busy DAC Right
//				0x0 = Ready
//				0x1 = Busy - This indicates the channel has not reached its
//					programmed volume/mute level
// 12	VOL_BUSY_DAC_LEFT  Volume Busy DAC Left
//				0x0 = Ready
//				0x1 = Busy - This indicates the channel has not reached its
//					programmed volume/mute level
// 9	VOL_RAMP_EN	Volume Ramp Enable (default=1)
//				0x0 = Disables volume ramp. New volume settings take immediate
//					effect without a ramp
//				0x1 = Enables volume ramp
//				This field affects DAC_VOL. The volume ramp effects both
//				volume settings and mute When set to 1 a soft mute is enabled.
// 8	VOL_EXPO_RAMP	Exponential Volume Ramp Enable
//				0x0 = Linear ramp over top 4 volume octaves
//				0x1 = Exponential ramp over full volume range
//				This bit only takes effect if VOL_RAMP_EN is 1.
// 3	DAC_MUTE_RIGHT	DAC Right Mute (default=1)
//				0x0 = Unmute
//				0x1 = Muted
//				If VOL_RAMP_EN = 1, this is a soft mute.
// 2	DAC_MUTE_LEFT	DAC Left Mute (default=1)
//				0x0 = Unmute
//				0x1 = Muted
//				If VOL_RAMP_EN = 1, this is a soft mute.
// 1	ADC_HPF_FREEZE	ADC High Pass Filter Freeze
//				0x0 = Normal operation
//				0x1 = Freeze the ADC high-pass filter offset register.  The
//				offset continues to be subtracted from the ADC data stream.
// 0	ADC_HPF_BYPASS	ADC High Pass Filter Bypass
//				0x0 = Normal operation
//				0x1 = Bypassed and offset not updated

#define CHIP_DAC_VOL			0x0010
// 15:8	DAC_VOL_RIGHT	DAC Right Channel Volume.  Set the Right channel DAC volume
//			with 0.5017 dB steps from 0 to -90 dB
//				0x3B and less = Reserved
//				0x3C = 0 dB
//				0x3D = -0.5 dB
//				0xF0 = -90 dB
//				0xFC and greater = Muted
//				If VOL_RAMP_EN = 1, there is an automatic ramp to the
//				new volume setting.
// 7:0	DAC_VOL_LEFT	DAC Left Channel Volume.  Set the Left channel DAC volume
//			with 0.5017 dB steps from 0 to -90 dB
//				0x3B and less = Reserved
//				0x3C = 0 dB
//				0x3D = -0.5 dB
//				0xF0 = -90 dB
//				0xFC and greater = Muted
//				If VOL_RAMP_EN = 1, there is an automatic ramp to the
//				new volume setting.

#define CHIP_PAD_STRENGTH		0x0014
// 9:8	I2S_LRCLK	I2S LRCLK Pad Drive Strength (default=1)
//				Sets drive strength for output pads per the table below.
//				 VDDIO    1.8 V     2.5 V     3.3 V
//				0x0 = Disable
//				0x1 =     1.66 mA   2.87 mA   4.02 mA
//				0x2 =     3.33 mA   5.74 mA   8.03 mA
//				0x3 =     4.99 mA   8.61 mA   12.05 mA
// 7:6	I2S_SCLK	I2S SCLK Pad Drive Strength (default=1)
// 5:4	I2S_DOUT	I2S DOUT Pad Drive Strength (default=1)
// 3:2	CTRL_DATA	I2C DATA Pad Drive Strength (default=3)
// 1:0	CTRL_CLK	I2C CLK Pad Drive Strength (default=3)
//				(all use same table as I2S_LRCLK)

#define CHIP_ANA_ADC_CTRL		0x0020
// 8	ADC_VOL_M6DB	ADC Volume Range Reduction
//				This bit shifts both right and left analog ADC volume
//				range down by 6.0 dB.
//				0x0 = No change in ADC range
//				0x1 = ADC range reduced by 6.0 dB
// 7:4	ADC_VOL_RIGHT	ADC Right Channel Volume
//				Right channel analog ADC volume control in 1.5 dB steps.
//				0x0 = 0 dB
//				0x1 = +1.5 dB
//				...
//				0xF = +22.5 dB
//				This range is -6.0 dB to +16.5 dB if ADC_VOL_M6DB is set to 1.
// 3:0	ADC_VOL_LEFT	ADC Left Channel Volume
//				(same scale as ADC_VOL_RIGHT)

#define CHIP_ANA_HP_CTRL		0x0022
// 14:8	HP_VOL_RIGHT	Headphone Right Channel Volume  (default 0x18)
//				Right channel headphone volume control with 0.5 dB steps.
//				0x00 = +12 dB
//				0x01 = +11.5 dB
//				0x18 = 0 dB
//				...
//				0x7F = -51.5 dB
// 6:0	HP_VOL_LEFT	Headphone Left Channel Volume  (default 0x18)
//				(same scale as HP_VOL_RIGHT)

#define CHIP_ANA_CTRL			0x0024
// 8	MUTE_LO		LINEOUT Mute, 0 = Unmute, 1 = Mute  (default 1)
// 6	SELECT_HP	Select the headphone input, 0 = DAC, 1 = LINEIN
// 5	EN_ZCD_HP	Enable the headphone zero cross detector (ZCD)
//				0x0 = HP ZCD disabled
//				0x1 = HP ZCD enabled
// 4	MUTE_HP		Mute the headphone outputs, 0 = Unmute, 1 = Mute (default)
// 2	SELECT_ADC	Select the ADC input, 0 = Microphone, 1 = LINEIN
// 1	EN_ZCD_ADC	Enable the ADC analog zero cross detector (ZCD)
//				0x0 = ADC ZCD disabled
//				0x1 = ADC ZCD enabled
// 0	MUTE_ADC	Mute the ADC analog volume, 0 = Unmute, 1 = Mute (default)

#define CHIP_LINREG_CTRL		0x0026
// 6	VDDC_MAN_ASSN	Determines chargepump source when VDDC_ASSN_OVRD is set.
//				0x0 = VDDA
//				0x1 = VDDIO
// 5	VDDC_ASSN_OVRD	Charge pump Source Assignment Override
//				0x0 = Charge pump source is automatically assigned based
//					on higher of VDDA and VDDIO
//				0x1 = the source of charge pump is manually assigned by
//					VDDC_MAN_ASSN If VDDIO and VDDA are both the same
//					and greater than 3.1 V, VDDC_ASSN_OVRD and
//					VDDC_MAN_ASSN should be used to manually assign
//					VDDIO as the source for charge pump.
// 3:0	D_PROGRAMMING	Sets the VDDD linear regulator output voltage in 50 mV steps.
//			Must clear the LINREG_SIMPLE_POWERUP and STARTUP_POWERUP bits
//			in the 0x0030 (CHIP_ANA_POWER) register after power-up, for
//			this setting to produce the proper VDDD voltage.
//				0x0 = 1.60
//				0xF = 0.85

#define CHIP_REF_CTRL			0x0028 // bandgap reference bias voltage and currents
// 8:4	VAG_VAL		Analog Ground Voltage Control
//				These bits control the analog ground voltage in 25 mV steps.
//				This should usually be set to VDDA/2 or lower for best
//				performance (maximum output swing at minimum THD). This VAG
//				reference is also used for the DAC and ADC voltage reference.
//				So changing this voltage scales the output swing of the DAC
//				and the output signal of the ADC.
//				0x00 = 0.800 V
//				0x1F = 1.575 V
// 3:1	BIAS_CTRL	Bias control
//				These bits adjust the bias currents for all of the analog
//				blocks. By lowering the bias current a lower quiescent power
//				is achieved. It should be noted that this mode can affect
//				performance by 3-4 dB.
//				0x0 = Nominal
//				0x1-0x3=+12.5%
//				0x4=-12.5%
//				0x5=-25%
//				0x6=-37.5%
//				0x7=-50%
// 0	SMALL_POP	VAG Ramp Control
//				Setting this bit slows down the VAG ramp from ~200 to ~400 ms
//				to reduce the startup pop, but increases the turn on/off time.
//				0x0 = Normal VAG ramp
//				0x1 = Slow down VAG ramp

#define CHIP_MIC_CTRL			0x002A // microphone gain & internal microphone bias
// 9:8	BIAS_RESISTOR	MIC Bias Output Impedance Adjustment
//				Controls an adjustable output impedance for the microphone bias.
//				If this is set to zero the micbias block is powered off and
//				the output is highZ.
//				0x0 = Powered off
//				0x1 = 2.0 kohm
//				0x2 = 4.0 kohm
//				0x3 = 8.0 kohm
// 6:4	BIAS_VOLT	MIC Bias Voltage Adjustment
//				Controls an adjustable bias voltage for the microphone bias
//				amp in 250 mV steps. This bias voltage setting should be no
//				more than VDDA-200 mV for adequate power supply rejection.
//				0x0 = 1.25 V
//				...
//				0x7 = 3.00 V
// 1:0	GAIN		MIC Amplifier Gain
//				Sets the microphone amplifier gain. At 0 dB setting the THD
//				can be slightly higher than other paths- typically around
//				~65 dB. At other gain settings the THD are better.
//				0x0 = 0 dB
//				0x1 = +20 dB
//				0x2 = +30 dB
//				0x3 = +40 dB

#define CHIP_LINE_OUT_CTRL		0x002C
// 11:8	OUT_CURRENT	Controls the output bias current for the LINEOUT amplifiers.  The
//			nominal recommended setting for a 10 kohm load with 1.0 nF load cap
//			is 0x3. There are only 5 valid settings.
//				0x0=0.18 mA
//				0x1=0.27 mA
//				0x3=0.36 mA
//				0x7=0.45 mA
//				0xF=0.54 mA
// 5:0	LO_VAGCNTRL	LINEOUT Amplifier Analog Ground Voltage
//				Controls the analog ground voltage for the LINEOUT amplifiers
//				in 25 mV steps. This should usually be set to VDDIO/2.
//				0x00 = 0.800 V
//				...
//				0x1F = 1.575 V
//				...
//				0x23 = 1.675 V
//				0x24-0x3F are invalid

#define CHIP_LINE_OUT_VOL		0x002E
// 12:8	LO_VOL_RIGHT	LINEOUT Right Channel Volume (default=4)
//				Controls the right channel LINEOUT volume in 0.5 dB steps.
//				Higher codes have more attenuation.
// 4:0	LO_VOL_LEFT	LINEOUT Left Channel Output Level (default=4)
//				Used to normalize the output level of the left line output
//				to full scale based on the values used to set
//				LINE_OUT_CTRL->LO_VAGCNTRL and CHIP_REF_CTRL->VAG_VAL.
//				In general this field should be set to:
//				40*log((VAG_VAL)/(LO_VAGCNTRL)) + 15
//				Suggested values based on typical VDDIO and VDDA voltages.
//					VDDA  VAG_VAL VDDIO  LO_VAGCNTRL LO_VOL_*
//					1.8 V    0.9   3.3 V     1.55      0x06
//					1.8 V    0.9   1.8 V      0.9      0x0F
//					3.3 V   1.55   1.8 V      0.9      0x19
//					3.3 V   1.55   3.3 V     1.55      0x0F
//				After setting to the nominal voltage, this field can be used
//				to adjust the output level in +/-0.5 dB increments by using
//				values higher or lower than the nominal setting.

#define CHIP_ANA_POWER			0x0030 // power down controls for the analog blocks.
		// The only other power-down controls are BIAS_RESISTOR in the MIC_CTRL register
		//  and the EN_ZCD control bits in ANA_CTRL.
// 14	DAC_MONO	While DAC_POWERUP is set, this allows the DAC to be put into left only
//				mono operation for power savings. 0=mono, 1=stereo (default)
// 13	LINREG_SIMPLE_POWERUP  Power up the simple (low power) digital supply regulator.
//				After reset, this bit can be cleared IF VDDD is driven
//				externally OR the primary digital linreg is enabled with
//				LINREG_D_POWERUP
// 12	STARTUP_POWERUP	Power up the circuitry needed during the power up ramp and reset.
//				After reset this bit can be cleared if VDDD is coming from
//				an external source.
// 11	VDDC_CHRGPMP_POWERUP Power up the VDDC charge pump block. If neither VDDA or VDDIO
//				is 3.0 V or larger this bit should be cleared before analog
//				blocks are powered up.
// 10	PLL_POWERUP	PLL Power Up, 0 = Power down, 1 = Power up
//				When cleared, the PLL is turned off. This must be set before
//				CHIP_CLK_CTRL->MCLK_FREQ is programmed to 0x3. The
//				CHIP_PLL_CTRL register must be configured correctly before
//				setting this bit.
// 9	LINREG_D_POWERUP Power up the primary VDDD linear regulator, 0 = Power down, 1 = Power up
// 8	VCOAMP_POWERUP	Power up the PLL VCO amplifier, 0 = Power down, 1 = Power up
// 7	VAG_POWERUP	Power up the VAG reference buffer.
//				Setting this bit starts the power up ramp for the headphone
//				and LINEOUT. The headphone (and/or LINEOUT) powerup should
//				be set BEFORE clearing this bit. When this bit is cleared
//				the power-down ramp is started. The headphone (and/or LINEOUT)
//				powerup should stay set until the VAG is fully ramped down
//				(200 to 400 ms after clearing this bit).
//				0x0 = Power down, 0x1 = Power up
// 6	ADC_MONO	While ADC_POWERUP is set, this allows the ADC to be put into left only
//				mono operation for power savings. This mode is useful when
//				only using the microphone input.
//				0x0 = Mono (left only), 0x1 = Stereo
// 5	REFTOP_POWERUP	Power up the reference bias currents
//				0x0 = Power down, 0x1 = Power up
//				This bit can be cleared when the part is a sleep state
//				to minimize analog power.
// 4	HEADPHONE_POWERUP Power up the headphone amplifiers
//				0x0 = Power down, 0x1 = Power up
// 3	DAC_POWERUP	Power up the DACs
//				0x0 = Power down, 0x1 = Power up
// 2	CAPLESS_HEADPHONE_POWERUP Power up the capless headphone mode
//				0x0 = Power down, 0x1 = Power up
// 1	ADC_POWERUP	Power up the ADCs
//				0x0 = Power down, 0x1 = Power up
// 0	LINEOUT_POWERUP	Power up the LINEOUT amplifiers
//				0x0 = Power down, 0x1 = Power up

#define CHIP_PLL_CTRL			0x0032
// 15:11 INT_DIVISOR
// 10:0 FRAC_DIVISOR

#define CHIP_CLK_TOP_CTRL		0x0034
// 11	ENABLE_INT_OSC	Setting this bit enables an internal oscillator to be used for the
//				zero cross detectors, the short detect recovery, and the
//				charge pump. This allows the I2S clock to be shut off while
//				still operating an analog signal path. This bit can be kept
//				on when the I2S clock is enabled, but the I2S clock is more
//				accurate so it is preferred to clear this bit when I2S is present.
// 3	INPUT_FREQ_DIV2	SYS_MCLK divider before PLL input
//				0x0 = pass through
//				0x1 = SYS_MCLK is divided by 2 before entering PLL
//				This must be set when the input clock is above 17 Mhz. This
//				has no effect when the PLL is powered down.

#define CHIP_ANA_STATUS			0x0036
// 9	LRSHORT_STS	This bit is high whenever a short is detected on the left or right
//				channel headphone drivers.
// 8	CSHORT_STS	This bit is high whenever a short is detected on the capless headphone
//				common/center channel driver.
// 4	PLL_IS_LOCKED	This bit goes high after the PLL is locked.

#define CHIP_ANA_TEST1			0x0038 //  intended only for debug.
#define CHIP_ANA_TEST2			0x003A //  intended only for debug.

#define CHIP_SHORT_CTRL			0x003C
// 14:12 LVLADJR	Right channel headphone	short detector in 25 mA steps.
//				0x3=25 mA
//				0x2=50 mA
//				0x1=75 mA
//				0x0=100 mA
//				0x4=125 mA
//				0x5=150 mA
//				0x6=175 mA
//				0x7=200 mA
//				This trip point can vary by ~30% over process so leave plenty
//				of guard band to avoid false trips.  This short detect trip
//				point is also effected by the bias current adjustments made
//				by CHIP_REF_CTRL->BIAS_CTRL and by CHIP_ANA_TEST1->HP_IALL_ADJ.
// 10:8	LVLADJL		Left channel headphone short detector in 25 mA steps.
//				(same scale as LVLADJR)
// 6:4	LVLADJC		Capless headphone center channel short detector in 50 mA steps.
//				0x3=50 mA
//				0x2=100 mA
//				0x1=150 mA
//				0x0=200 mA
//				0x4=250 mA
//				0x5=300 mA
//				0x6=350 mA
//				0x7=400 mA
// 3:2	MODE_LR		Behavior of left/right short detection
//				0x0 = Disable short detector, reset short detect latch,
//					software view non-latched short signal
//				0x1 = Enable short detector and reset the latch at timeout
//					(every ~50 ms)
//				0x2 = This mode is not used/invalid
//				0x3 = Enable short detector with only manual reset (have
//					to return to 0x0 to reset the latch)
// 1:0	MODE_CM		Behavior of capless headphone central short detection
//				(same settings as MODE_LR)

#define DAP_CONTROL			0x0100
#define DAP_PEQ				0x0102
#define DAP_BASS_ENHANCE		0x0104
#define DAP_BASS_ENHANCE_CTRL		0x0106
#define DAP_AUDIO_EQ			0x0108
#define DAP_SGTL_SURROUND		0x010A
#define DAP_FILTER_COEF_ACCESS		0x010C
#define DAP_COEF_WR_B0_MSB		0x010E
#define DAP_COEF_WR_B0_LSB		0x0110
#define DAP_AUDIO_EQ_BASS_BAND0		0x0116 // 115 Hz
#define DAP_AUDIO_EQ_BAND1		0x0118 // 330 Hz
#define DAP_AUDIO_EQ_BAND2		0x011A // 990 Hz
#define DAP_AUDIO_EQ_BAND3		0x011C // 3000 Hz
#define DAP_AUDIO_EQ_TREBLE_BAND4	0x011E // 9900 Hz
#define DAP_MAIN_CHAN			0x0120
#define DAP_MIX_CHAN			0x0122
#define DAP_AVC_CTRL			0x0124
#define DAP_AVC_THRESHOLD		0x0126
#define DAP_AVC_ATTACK			0x0128
#define DAP_AVC_DECAY			0x012A
#define DAP_COEF_WR_B1_MSB		0x012C
#define DAP_COEF_WR_B1_LSB		0x012E
#define DAP_COEF_WR_B2_MSB		0x0130
#define DAP_COEF_WR_B2_LSB		0x0132
#define DAP_COEF_WR_A1_MSB		0x0134
#define DAP_COEF_WR_A1_LSB		0x0136
#define DAP_COEF_WR_A2_MSB		0x0138
#define DAP_COEF_WR_A2_LSB		0x013A

#define SGTL5000_I2C_ADDR  0x0A  // CTRL_ADR0_CS pin low (normal configuration)
//#define SGTL5000_I2C_ADDR  0x2A // CTRL_ADR0_CS  pin high



bool AudioControlSGTL5000::enable(void)
{
	//unsigned int n;

	muted = true;
	Wire.begin();
	delay(5);
	//Serial.print("chip ID = ");
	//delay(5);
	//n = read(CHIP_ID);
	//Serial.println(n, HEX);

	write(CHIP_ANA_POWER, 0x4060);  // VDDD is externally driven with 1.8V
	write(CHIP_LINREG_CTRL, 0x006C);  // VDDA & VDDIO both over 3.1V
	write(CHIP_REF_CTRL, 0x01F1); // VAG=1.575 slow ramp, normal bias current
	write(CHIP_LINE_OUT_CTRL, 0x0322); // LO_VAGCNTRL=1.65V, OUT_CURRENT=0.36mA
	write(CHIP_SHORT_CTRL, 0x4446);  // allow up to 125mA
	write(CHIP_ANA_CTRL, 0x0137);  // enable zero cross detectors
	write(CHIP_ANA_POWER, 0x40FF); // power up: lineout, hp, adc, dac
	write(CHIP_DIG_POWER, 0x0073); // power up all digital stuff
	delay(400);
	// 40*log((1.575)/(1.65)) + 15 = 13.1391993746043 but it seems wrong, 5 is better...
	write(CHIP_LINE_OUT_VOL, 0x0505); // TODO: correct value for 3.3V
	write(CHIP_CLK_CTRL, 0x0004);  // 44.1 kHz, 256*Fs
	write(CHIP_I2S_CTRL, 0x0130); // SCLK=32*Fs, 16bit, I2S format
	// default signal routing is ok?
	write(CHIP_SSS_CTRL, 0x0010); // ADC->I2S, I2S->DAC
	write(CHIP_ADCDAC_CTRL, 0x0000); // disable dac mute
	write(CHIP_DAC_VOL, 0x3C3C); // digital gain, 0dB
	write(CHIP_ANA_HP_CTRL, 0x7F7F); // set volume (lowest level)
	write(CHIP_ANA_CTRL, 0x0136);  // enable zero cross detectors
	//mute = false;
	return true;
}

unsigned int AudioControlSGTL5000::read(unsigned int reg)
{
	unsigned int val;

	Wire.beginTransmission(SGTL5000_I2C_ADDR);
	Wire.write(reg >> 8);
	Wire.write(reg);
	if (Wire.endTransmission(false) != 0) return 0;
	if (Wire.requestFrom(SGTL5000_I2C_ADDR, 2) < 2) return 0;
	val = Wire.read() << 8;
	val |= Wire.read();
	return val;
}

bool AudioControlSGTL5000::write(unsigned int reg, unsigned int val)
{
	if (reg == CHIP_ANA_CTRL) ana_ctrl = val;
	Wire.beginTransmission(SGTL5000_I2C_ADDR);
	Wire.write(reg >> 8);
	Wire.write(reg);
	Wire.write(val >> 8);
	Wire.write(val);
	if (Wire.endTransmission() == 0) return true;
	return false;
}

unsigned int AudioControlSGTL5000::modify(unsigned int reg, unsigned int val, unsigned int iMask)
{
	unsigned int val1 = (read(reg)&(~iMask))|val;
	if(!write(reg,val1)) return 0;
	return val1;
}

bool AudioControlSGTL5000::volumeInteger(unsigned int n)
{
	if (n == 0) {
		muted = true;
		write(CHIP_ANA_HP_CTRL, 0x7F7F);
		return muteHeadphone();
	} else if (n > 0x80) {
		n = 0;
	} else {
		n = 0x80 - n;
	}
	if (muted) {
		muted = false;
		unmuteHeadphone();
	}
	n = n | (n << 8);
	return write(CHIP_ANA_HP_CTRL, n);  // set volume
}

bool AudioControlSGTL5000::volume(float left, float right)
{
	unsigned short m=((0x7F-calcVol(right,0x7F))<<8)|(0x7F-calcVol(left,0x7F));
	return write(CHIP_ANA_HP_CTRL, m);
}


// CHIP_LINE_OUT_VOL
unsigned short AudioControlSGTL5000::lo_lvl(uint8_t n)
{
	n&=31;
	return modify(CHIP_LINE_OUT_VOL,(n<<8)|n,(31<<8)|31);
}

unsigned short AudioControlSGTL5000::lo_lvl(uint8_t left, uint8_t right)
{
	left&=31;
	right&=31;
	return modify(CHIP_LINE_OUT_VOL,(right<<8)|left,(31<<8)|31);
}

unsigned short AudioControlSGTL5000::dac_vol(float n) // set both directly
{
	if(read(CHIP_ADCDAC_CTRL)&(3<<2)!=((n>0 ? 0:3)<<2)) modify(CHIP_ADCDAC_CTRL,(n>0 ? 0:3)<<2,3<<2);
	unsigned char m=calcVol(n,0xC0);
	return modify(CHIP_DAC_VOL,((0xFC-m)<<8)|(0xFC-m),65535);
}
unsigned short AudioControlSGTL5000::dac_vol(float left, float right)
{
	unsigned short adcdac=((right>0 ? 0:2)|(left>0 ? 0:1))<<2;
	if(read(CHIP_ADCDAC_CTRL)&(3<<2)!=adcdac)  modify(CHIP_ADCDAC_CTRL,adcdac,1<<2);
	unsigned short m=(0xFC-calcVol(right,0xC0))<<8|(0xFC-calcVol(left,0xC0));
	return modify(CHIP_DAC_VOL,m,65535);
}
// DAP_CONTROL
unsigned short AudioControlSGTL5000::dap_mix_enable(uint8_t n)
{
	return modify(DAP_CONTROL,(n&1)<<4,1<<4);
}
unsigned short AudioControlSGTL5000::dap_enable(uint8_t n)
{
	if(n) n=1;
	unsigned char DAC=1+(2*n); // I2S_IN if n==0 else DAP
	modify(DAP_CONTROL,n,1);
	return modify(CHIP_SSS_CTRL,(0<<6)|(DAC<<4),(3<<6)|(3<<4));
}

unsigned short AudioControlSGTL5000::dap_enable(void)
{
	return dap_enable(1);
}

// DAP_PEQ
unsigned short AudioControlSGTL5000::dap_peqs(uint8_t n) // valid to n&7, 0 thru 7 filters enabled.
{
	return modify(DAP_PEQ,(n&7),7);
}

// DAP_AUDIO_EQ
unsigned short AudioControlSGTL5000::dap_audio_eq(uint8_t n) // 0=NONE, 1=PEQ (7 IIR Biquad filters), 2=TONE (tone), 3=GEQ (5 band EQ)
{
	return modify(DAP_AUDIO_EQ,n&3,3);
}


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

void AudioFilterFIR::begin(short *cp,int n_coeffs)
{
  // pointer to coefficients
  coeff_p = cp;
  // Initialize FIR instances for the left and right channels
  if(coeff_p && (coeff_p != FIR_PASSTHRU)) {
    arm_fir_init_q15(&l_fir_inst, n_coeffs, coeff_p, &l_StateQ15[0], AUDIO_BLOCK_SAMPLES);
    arm_fir_init_q15(&r_fir_inst, n_coeffs, coeff_p, &r_StateQ15[0], AUDIO_BLOCK_SAMPLES);
  }
}

// This has the same effect as begin(NULL,0);
void AudioFilterFIR::stop(void)
{
  coeff_p = NULL;
}


void AudioFilterFIR::update(void)
{
  audio_block_t *block,*b_new;
  
  // If there's no coefficient table, give up.  
  if(coeff_p == NULL)return;

  // do passthru
  if(coeff_p == FIR_PASSTHRU) {
    // Just passthrough
    block = receiveWritable(0);
    if(block) {
      transmit(block,0);
      release(block);
    }
    block = receiveWritable(1);
    if(block) {
      transmit(block,1);
      release(block);
    }
    return;
  }
  // Left Channel
  block = receiveWritable(0);
  // get a block for the FIR output
  b_new = allocate();
  if(block && b_new) {
    arm_fir_q15(&l_fir_inst, (q15_t *)block->data, (q15_t *)b_new->data, AUDIO_BLOCK_SAMPLES);
    // send the FIR output to the left channel
    transmit(b_new,0);
  }
  if(block)release(block);
  if(b_new)release(b_new);

  // Right Channel
  block = receiveWritable(1);
  b_new = allocate();
  if(block && b_new) {
    arm_fir_q15(&r_fir_inst, (q15_t *)block->data, (q15_t *)b_new->data, AUDIO_BLOCK_SAMPLES);
    transmit(b_new,1);
  }
  if(block)release(block);
  if(b_new)release(b_new);
}


/******************************************************************/
//                A u d i o E f f e c t F l a n g e
// Written by Pete (El Supremo) Jan 2014
// 140207 - fix calculation of delay_rate_incr which is expressed as
//			a fraction of 2*PI
// 140207 - cosmetic fix to begin()

// circular addressing indices for left and right channels
short AudioEffectFlange::l_circ_idx;
short AudioEffectFlange::r_circ_idx;

short * AudioEffectFlange::l_delayline = NULL;
short * AudioEffectFlange::r_delayline = NULL;

// User-supplied offset for the delayed sample
// but start with passthru
int AudioEffectFlange::delay_offset_idx = DELAY_PASSTHRU;
int AudioEffectFlange::delay_length;

int AudioEffectFlange::delay_depth;
int AudioEffectFlange::delay_rate_incr;
unsigned int AudioEffectFlange::l_delay_rate_index;
unsigned int AudioEffectFlange::r_delay_rate_index;
// fails if the user provides unreasonable values but will
// coerce them and go ahead anyway. e.g. if the delay offset
// is >= CHORUS_DELAY_LENGTH, the code will force it to
// CHORUS_DELAY_LENGTH-1 and return false.
// delay_rate is the rate (in Hz) of the sine wave modulation
// delay_depth is the maximum variation around delay_offset
// i.e. the total offset is delay_offset + delay_depth * sin(delay_rate)
boolean AudioEffectFlange::begin(short *delayline,int d_length,int delay_offset,int d_depth,float delay_rate)
{
  boolean all_ok = true;

if(0) {
  Serial.print("AudioEffectFlange.begin(offset = ");
  Serial.print(delay_offset);
  Serial.print(", depth = ");
  Serial.print(d_depth);
  Serial.print(", rate = ");
  Serial.print(delay_rate,3);
  Serial.println(")");
  Serial.print("    FLANGE_DELAY_LENGTH = ");
  Serial.println(d_length);
}
  delay_length = d_length/2;
  l_delayline = delayline;
  r_delayline = delayline + delay_length;
  
  delay_depth = d_depth;
  // initial index
  l_delay_rate_index = 0;
  r_delay_rate_index = 0;
  l_circ_idx = 0;
  r_circ_idx = 0;
  delay_rate_incr = delay_rate/44100.*2147483648.; 
//Serial.println(delay_rate_incr,HEX);

  delay_offset_idx = delay_offset;
  // Allow the passthru code to go through
  if(delay_offset_idx < -1) {
    delay_offset_idx = 0;
    all_ok = false;
  }
  if(delay_offset_idx >= delay_length) {
    delay_offset_idx = delay_length - 1;
    all_ok = false;
  }  
  return(all_ok);
}


boolean AudioEffectFlange::modify(int delay_offset,int d_depth,float delay_rate)
{
  boolean all_ok = true;
  
  delay_depth = d_depth;

  delay_rate_incr = delay_rate/44100.*2147483648.;
  
  delay_offset_idx = delay_offset;
  // Allow the passthru code to go through
  if(delay_offset_idx < -1) {
    delay_offset_idx = 0;
    all_ok = false;
  }
  if(delay_offset_idx >= delay_length) {
    delay_offset_idx = delay_length - 1;
    all_ok = false;
  }
  l_delay_rate_index = 0;
  r_delay_rate_index = 0;
  l_circ_idx = 0;
  r_circ_idx = 0;
  return(all_ok);
}

void AudioEffectFlange::update(void)
{
  audio_block_t *block;
  int idx;
  short *bp;
  short frac;
  int idx1;

  if(l_delayline == NULL)return;
  if(r_delayline == NULL)return; 

  // do passthru
  if(delay_offset_idx == DELAY_PASSTHRU) {
    // Just passthrough
    block = receiveWritable(0);
    if(block) {
      bp = block->data;
      for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
        l_circ_idx++;
        if(l_circ_idx >= delay_length) {
          l_circ_idx = 0;
        }
        l_delayline[l_circ_idx] = *bp++;
      }
      transmit(block,0);
      release(block);
    }
    block = receiveWritable(1);
    if(block) {
      bp = block->data;
      for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
        r_circ_idx++;
        if(r_circ_idx >= delay_length) {
          r_circ_idx = 0;
        }
        r_delayline[r_circ_idx] = *bp++;
      }
      transmit(block,1);
      release(block);
    }
    return;
  }

  //          L E F T  C H A N N E L

  block = receiveWritable(0);
  if(block) {
    bp = block->data;
    for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
      l_circ_idx++;
      if(l_circ_idx >= delay_length) {
        l_circ_idx = 0;
      }
      l_delayline[l_circ_idx] = *bp;
      idx = arm_sin_q15( (q15_t)((l_delay_rate_index >> 16) & 0x7fff));
      idx = (idx * delay_depth) >> 15;
//Serial.println(idx);
      idx = l_circ_idx - (delay_offset_idx + idx);
      if(idx < 0) {
        idx += delay_length;
      }
      if(idx >= delay_length) {
        idx -= delay_length;
      }

      if(frac < 0)
        idx1 = idx - 1;
      else
        idx1 = idx + 1;
      if(idx1 < 0) {
        idx1 += delay_length;
      }
      if(idx1 >= delay_length) {
        idx1 -= delay_length;
      }
      frac = (l_delay_rate_index >> 1) &0x7fff;
      frac = (( (int)(l_delayline[idx1] - l_delayline[idx])*frac) >> 15);

      *bp++ = (l_delayline[l_circ_idx]
                + l_delayline[idx] + frac               
              )/2;

      l_delay_rate_index += delay_rate_incr;
      if(l_delay_rate_index & 0x80000000) {
        l_delay_rate_index &= 0x7fffffff;
      }
    }
    // send the effect output to the left channel
    transmit(block,0);
    release(block);
  }

  //          R I G H T  C H A N N E L

  block = receiveWritable(1);
  if(block) {
    bp = block->data;
    for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
      r_circ_idx++;
      if(r_circ_idx >= delay_length) {
        r_circ_idx = 0;
      }
      r_delayline[r_circ_idx] = *bp;
      idx = arm_sin_q15( (q15_t)((r_delay_rate_index >> 16)&0x7fff));
       idx = (idx * delay_depth) >> 15;

      idx = r_circ_idx - (delay_offset_idx + idx);
      if(idx < 0) {
        idx += delay_length;
      }
      if(idx >= delay_length) {
        idx -= delay_length;
      }

      if(frac < 0)
        idx1 = idx - 1;
      else
        idx1 = idx + 1;
      if(idx1 < 0) {
        idx1 += delay_length;
      }
      if(idx1 >= delay_length) {
        idx1 -= delay_length;
      }
      frac = (r_delay_rate_index >> 1) &0x7fff;
      frac = (( (int)(r_delayline[idx1] - r_delayline[idx])*frac) >> 15);

      *bp++ = (r_delayline[r_circ_idx]
                + r_delayline[idx] + frac
               )/2;

      r_delay_rate_index += delay_rate_incr;
      if(r_delay_rate_index & 0x80000000) {
        r_delay_rate_index &= 0x7fffffff;
      }

    }
    // send the effect output to the right channel
    transmit(block,1);
    release(block);
  }
}



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

//                A u d i o E f f e c t C h o r u s
// Written by Pete (El Supremo) Jan 2014

// circular addressing indices for left and right channels
short AudioEffectChorus::l_circ_idx;
short AudioEffectChorus::r_circ_idx;

short * AudioEffectChorus::l_delayline = NULL;
short * AudioEffectChorus::r_delayline = NULL;
int AudioEffectChorus::delay_length;
// An initial value of zero indicates passthru
int AudioEffectChorus::num_chorus = 0;


// All three must be valid.
boolean AudioEffectChorus::begin(short *delayline,int d_length,int n_chorus)
{
Serial.print("AudioEffectChorus.begin(Chorus delay line length = ");
Serial.print(d_length);
Serial.print(", n_chorus = ");
Serial.print(n_chorus);
Serial.println(")");

l_delayline = NULL;
r_delayline = NULL;
delay_length = 0;
l_circ_idx = 0;
r_circ_idx = 0;

  if(delayline == NULL) {
    return(false);
  }
  if(d_length < 10) {
    return(false);
  }
  if(n_chorus < 1) {
    return(false);
  }
  
  l_delayline = delayline;
  r_delayline = delayline + d_length/2;
  delay_length = d_length/2;
  num_chorus = n_chorus;
 
  return(true);
}

// This has the same effect as begin(NULL,0);
void AudioEffectChorus::stop(void)
{

}

void AudioEffectChorus::modify(int n_chorus)
{
  num_chorus = n_chorus;
}

int iabs(int x)
{
  if(x < 0)return(-x);
  return(x);
}
//static int d_count = 0;

int last_idx = 0;
void AudioEffectChorus::update(void)
{
  audio_block_t *block;
  short *bp;
  int sum;
  int c_idx;

  if(l_delayline == NULL)return;
  if(r_delayline == NULL)return;  
  
  // do passthru
  // It stores the unmodified data in the delay line so that
  // it isn't as likely to click
  if(num_chorus < 1) {
    // Just passthrough
    block = receiveWritable(0);
    if(block) {
      bp = block->data;
      for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
        l_circ_idx++;
        if(l_circ_idx >= delay_length) {
          l_circ_idx = 0;
        }
        l_delayline[l_circ_idx] = *bp++;
      }
      transmit(block,0);
      release(block);
    }
    block = receiveWritable(1);
    if(block) {
      bp = block->data;
      for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
        r_circ_idx++;
        if(r_circ_idx >= delay_length) {
          r_circ_idx = 0;
        }
        r_delayline[r_circ_idx] = *bp++;
      }
      transmit(block,1);
      release(block);
    }
    return;
  }

  //          L E F T  C H A N N E L

  block = receiveWritable(0);
  if(block) {
    bp = block->data;
    for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
      l_circ_idx++;
      if(l_circ_idx >= delay_length) {
        l_circ_idx = 0;
      }
      l_delayline[l_circ_idx] = *bp;
      sum = 0;
      c_idx = l_circ_idx;
      for(int k = 0; k < num_chorus; k++) {
        sum += l_delayline[c_idx];
        if(num_chorus > 1)c_idx -= delay_length/(num_chorus - 1) - 1;
        if(c_idx < 0) {
          c_idx += delay_length;
        }
      }
      *bp++ = sum/num_chorus;
    }

    // send the effect output to the left channel
    transmit(block,0);
    release(block);
  }

  //          R I G H T  C H A N N E L

  block = receiveWritable(1);
  if(block) {
    bp = block->data;
    for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
      r_circ_idx++;
      if(r_circ_idx >= delay_length) {
        r_circ_idx = 0;
      }
      r_delayline[r_circ_idx] = *bp;
      sum = 0;
      c_idx = r_circ_idx;
      for(int k = 0; k < num_chorus; k++) {
        sum += r_delayline[c_idx];
        if(num_chorus > 1)c_idx -= delay_length/(num_chorus - 1) - 1;
        if(c_idx < 0) {
          c_idx += delay_length;
        }
      }
      *bp++ = sum/num_chorus;
    }

    // send the effect output to the left channel
    transmit(block,1);
    release(block);
  }
}


// DAP_AUDIO_EQ_BASS_BAND0 & DAP_AUDIO_EQ_BAND1 & DAP_AUDIO_EQ_BAND2 etc etc
unsigned short AudioControlSGTL5000::dap_audio_eq_band(uint8_t bandNum, float n) // by signed percentage -100/+100; dap_audio_eq(3);
{ // 0x00==-12dB, 0x2F==0dB, 0x5F==12dB
	n=((n/100)*48)+0.499;
	if(n<-47) n=-47;
	if(n>48) n=48;
	n+=47;
	return modify(DAP_AUDIO_EQ_BASS_BAND0+(bandNum*2),(unsigned int)n,127);
}
void AudioControlSGTL5000::dap_audio_eq_geq(float bass, float mid_bass, float midrange, float mid_treble, float treble)
{
	dap_audio_eq_band(0,bass);
	dap_audio_eq_band(1,mid_bass);
	dap_audio_eq_band(2,midrange);
	dap_audio_eq_band(3,mid_treble);
	dap_audio_eq_band(4,treble);
}
void AudioControlSGTL5000::dap_audio_eq_tone(float bass, float treble) // dap_audio_eq(2);
{
	dap_audio_eq_band(0,bass);
	dap_audio_eq_band(4,treble);
}

// SGTL5000 PEQ Coefficient loader
void AudioControlSGTL5000::load_peq(uint8_t filterNum, int *filterParameters)
{
	// 1111 11111111 11111111

	write(DAP_COEF_WR_B0_MSB,(*filterParameters>>4)&65535);
	write(DAP_COEF_WR_B0_LSB,(*filterParameters++)&15);
	write(DAP_COEF_WR_B1_MSB,(*filterParameters>>4)&65535);
	write(DAP_COEF_WR_B1_LSB,(*filterParameters++)&15);
	write(DAP_COEF_WR_B2_MSB,(*filterParameters>>4)&65535);
	write(DAP_COEF_WR_B2_LSB,(*filterParameters++)&15);
	write(DAP_COEF_WR_A1_MSB,(*filterParameters>>4)&65535);
	write(DAP_COEF_WR_A1_LSB,(*filterParameters++)&15);
	write(DAP_COEF_WR_A2_MSB,(*filterParameters>>4)&65535);
	write(DAP_COEF_WR_A2_LSB,(*filterParameters++)&15);
	write(DAP_FILTER_COEF_ACCESS,(uint16_t)0x100|filterNum);
	delay(10); // seems necessary, didn't work for 1ms.
	modify(DAP_FILTER_COEF_ACCESS,(uint16_t)filterNum,15); 
}

unsigned char AudioControlSGTL5000::calcVol(float n, unsigned char range)
{
	n=(n*(((float)range)/100))+0.499;
	if ((unsigned char)n>range) n=range;
	return (unsigned char)n;
}

// if(SGTL5000_PEQ) quantization_unit=524288; if(AudioFilterBiquad) quantization_unit=2147483648;
void calcBiquad(uint8_t filtertype, float fC, float dB_Gain, float Q, uint32_t quantization_unit, uint32_t fS, int *coef)
{

// I used resources like http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt
// to make this routine, I tested most of the filter types and they worked. Such filters have limits and
// before calling this routine with varying values the end user should check that those values are limited
// to valid results.

  float A;
  if(filtertype<FILTER_PARAEQ) A=pow(10,dB_Gain/20); else A=pow(10,dB_Gain/40);
  float W0 = 2*3.14159265358979323846*fC/fS; 
  float cosw=cos(W0);
  float sinw=sin(W0);
  //float alpha = sinw*sinh((log(2)/2)*BW*W0/sinw);
  //float beta = sqrt(2*A);
  float alpha = sinw / (2 * Q); 
  float beta = sqrt(A)/Q;
  float b0,b1,b2,a0,a1,a2;

  switch(filtertype) {
  case FILTER_LOPASS:
    b0 = (1.0F - cosw) * 0.5F; // =(1-COS($H$2))/2
    b1 = 1.0F - cosw;
    b2 = (1.0F - cosw) * 0.5F;
    a0 = 1.0F + alpha;
    a1 = 2.0F * cosw;
    a2 = alpha - 1.0F;
  break;
  case FILTER_HIPASS:
    b0 = (1.0F + cosw) * 0.5F;
    b1 = -(cosw + 1.0F);
    b2 = (1.0F + cosw) * 0.5F;
    a0 = 1.0F + alpha;
    a1 = 2.0F * cosw;
    a2 = alpha - 1.0F;
  break;
  case FILTER_BANDPASS:
    b0 = alpha;
    b1 = 0.0F;
    b2 = -alpha;
    a0 = 1.0F + alpha;
    a1 = 2.0F * cosw;
    a2 = alpha - 1.0F;
   break;
  case FILTER_NOTCH:
    b0=1;
    b1=-2*cosw;
    b2=1;
    a0=1+alpha;
    a1=2*cosw;
    a2=-(1-alpha);
  break;
  case FILTER_PARAEQ:
    b0 = 1 + (alpha*A);
    b1 =-2 * cosw;
    b2 = 1 - (alpha*A);
    a0 = 1 + (alpha/A);
    a1 = 2 * cosw;
    a2 =-(1-(alpha/A));
  break;
  case FILTER_LOSHELF:
    b0 = A * ((A+1.0F) - ((A-1.0F)*cosw) + (beta*sinw));
    b1 = 2.0F * A * ((A-1.0F) - ((A+1.0F)*cosw));
    b2 = A * ((A+1.0F) - ((A-1.0F)*cosw) - (beta*sinw));
    a0 = (A+1.0F) + ((A-1.0F)*cosw) + (beta*sinw);
    a1 = 2.0F * ((A-1.0F) + ((A+1.0F)*cosw));
    a2 = -((A+1.0F) + ((A-1.0F)*cosw) - (beta*sinw));
  break;
  case FILTER_HISHELF:
    b0 = A * ((A+1.0F) + ((A-1.0F)*cosw) + (beta*sinw));
    b1 = -2.0F * A * ((A-1.0F) + ((A+1.0F)*cosw));
    b2 = A * ((A+1.0F) + ((A-1.0F)*cosw) - (beta*sinw));
    a0 = (A+1.0F) - ((A-1.0F)*cosw) + (beta*sinw);
    a1 = -2.0F * ((A-1.0F) - ((A+1.0F)*cosw));
    a2 = -((A+1.0F) - ((A-1.0F)*cosw) - (beta*sinw));
  }

  a0=(a0*2)/(float)quantization_unit; // once here instead of five times there...
  b0/=a0;
  *coef++=(int)(b0+0.499);
  b1/=a0;
  *coef++=(int)(b1+0.499);
  b2/=a0;
  *coef++=(int)(b2+0.499);
  a1/=a0;
  *coef++=(int)(a1+0.499);
  a2/=a0;
  *coef++=(int)(a2+0.499);
}

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

//                A u d i o T o n e S w e e p
// Written by Pete (El Supremo) Feb 2014


boolean AudioToneSweep::begin(short t_amp,int t_lo,int t_hi,float t_time)
{
  double tone_tmp;
  
if(0) {
  Serial.print("AudioToneSweep.begin(tone_amp = ");
  Serial.print(t_amp);
  Serial.print(", tone_lo = ");
  Serial.print(t_lo);
  Serial.print(", tone_hi = ");
  Serial.print(t_hi);
  Serial.print(", tone_time = ");
  Serial.print(t_time,1);
  Serial.println(")");
}
  tone_amp = 0;
  if(t_amp < 0)return false;
  if(t_lo < 1)return false;
  if(t_hi < 1)return false;
  if(t_hi >= 44100/2)return false;
  if(t_lo >= 44100/2)return false;
  if(t_time < 0)return false;
  tone_lo = t_lo;
  tone_hi = t_hi;
  tone_phase = 0;

  tone_amp = t_amp;
  // Limit the output amplitude to prevent aliasing
  // until I can figure out why this "overtops"
  // above 29000.
  if(tone_amp > 29000)tone_amp = 29000;
  tone_tmp = tone_hi - tone_lo;
  tone_sign = 1;
  tone_freq = tone_lo*0x100000000LL;
  if(tone_tmp < 0) {
    tone_sign = -1;
    tone_tmp = -tone_tmp;
  }
  tone_tmp = tone_tmp/t_time/44100.;
  tone_incr = (tone_tmp * 0x100000000LL);
  sweep_busy = 1;
  return(true);
}



unsigned char AudioToneSweep::busy(void)
{
  return(sweep_busy);
}

int b_count = 0;
void AudioToneSweep::update(void)
{
  audio_block_t *block;
  short *bp;
  int i;
  
  if(!sweep_busy)return;

  //          L E F T  C H A N N E L  O N L Y
  block = allocate();
  if(block) {
    bp = block->data;
    // Generate the sweep
    for(i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
      *bp++ = (short)(( (short)(arm_sin_q31((uint32_t)((tone_phase >> 15)&0x7fffffff))>>16) *tone_amp) >> 16);
      uint64_t tone_tmp = (0x400000000000LL * (int)((tone_freq >> 32)&0x7fffffff))/44100;

      tone_phase +=  tone_tmp;
      if(tone_phase & 0x800000000000LL)tone_phase &= 0x7fffffffffffLL;

      if(tone_sign > 0) {
        if((tone_freq >> 32) > tone_hi) {
          sweep_busy = 0;
          break;
        }
        tone_freq += tone_incr;
      } else {
        if((tone_freq >> 32) < tone_hi) {
          sweep_busy = 0;

          break;
        }
        tone_freq -= tone_incr;        
      }
    }
    while(i < AUDIO_BLOCK_SAMPLES) {
      *bp++ = 0;
      i++;
    }
    b_count++;
    // send the samples to the left channel
    transmit(block,0);
    release(block);
  }
}