text: /* Audio Library for Teensy 3.X
   * Copyright (c) 2014, Paul Stoffregen, paul@pjrc.com
   *
   * Development of this audio library was funded by PJRC.COM, LLC by sales of
   * Teensy and Audio Adaptor boards.  Please support PJRC's efforts to develop
   * open source software by purchasing Teensy or other PJRC products.
   *
   * Permission is hereby granted, free of charge, to any person obtaining a copy
   * of this software and associated documentation files (the "Software"), to deal
   * in the Software without restriction, including without limitation the rights
   * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
   * copies of the Software, and to permit persons to whom the Software is
   * furnished to do so, subject to the following conditions:
   *
   * The above copyright notice, development funding notice, and this permission
   * notice shall be included in all copies or substantial portions of the Software.
   *
   * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
   * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
   * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
   * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
   * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
   * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
   * THE SOFTWARE.
   */
   
   /* 
   *  Extended by Chip Audette, OpenAudio, May 2019
   *  Converted to F32 and to variable audio block length
   *	The F32 conversion is under the MIT License.  Use at your own risk.
   */
   
  #include "output_i2s_f32.h"
  //#include "input_i2s_f32.h"
  //include "memcpy_audio.h"
  //#include "memcpy_interleave.h"
  #include <arm_math.h>
  
  ////////////
  //
  // Changing the sample rate based on changing the I2S bus freuqency
  //
  //Here's the function to change the sample rate of the system (via changing the clocking of the I2S bus)
  //https://forum.pjrc.com/threads/38753-Discussion-about-a-simple-way-to-change-the-sample-rate?p=121365&viewfull=1#post121365
  //
  //And, a post on how to compute the frac and div portions?  I haven't checked the code presented in this post:
  //https://forum.pjrc.com/threads/38753-Discussion-about-a-simple-way-to-change-the-sample-rate?p=188812&viewfull=1#post188812
  //
  //Finally, here is my own Matlab code for computing the mult and div values...
  /*
  	%choose the sample rates that you are hoping to hit
  	targ_fs_Hz =  [2000, 8000, 11025, 16000, 22050, 24000, 32000, 44100, floor(44117.64706) , ...
  		48000, 88200, floor(44117.64706 * 2), (37000/256*662), 96000, 176400, floor(44117.64706 * 4), 192000];
  
  	F_PLL = 180e6;  %choose the clock rate used for this calculation
  	PLL_div = 256;
  
  	all_n=[];all_d=[];
  	for Itarg=1:length(targ_fs_Hz)
  		if (0)
  			[best_d,best_n]=rat((F_PLL/PLL_div)/targ_fs_Hz(Itarg));
  		else
  			best_n = 1; best_d = 1; best_err = 1e10;
  			for n=1:255
  				d = [1:4095];
  				act_fs_Hz = F_PLL / PLL_div * n ./ d;
  				[err,I] = min(abs(act_fs_Hz - targ_fs_Hz(Itarg)));
  				if err < best_err
  					best_n = n; best_d = d(I);
  					best_err = err;
  				end
  			end
  		end
  		all_n(Itarg) = best_n;
  		all_d(Itarg) = best_d;
  		disp(['fs = ' num2str(targ_fs_Hz(Itarg)) ', n = ' num2str(best_n) ', d = ' num2str(best_d) ', true = ' num2str(F_PLL/PLL_div * best_n / best_d)])
  	end
  */
  float AudioOutputI2S_F32::setI2SFreq(const float freq_Hz) {
  	int freq = (int)(freq_Hz+0.5);
    typedef struct {
      uint8_t mult;
      uint16_t div;
    } __attribute__((__packed__)) tmclk;
  
    const int numfreqs = 17;
    const int samplefreqs[numfreqs] = { 2000, 8000, 11025, 16000, 22050, 24000, 32000, 44100, (int)44117.64706 , 48000, 88200, (int)(44117.64706 * 2), (int)(95679.69+0.5), 96000, 176400, (int)(44117.64706 * 4), 192000};
  
  #if (F_PLL==16000000)
    const tmclk clkArr[numfreqs] = {{4, 125}, {16, 125}, {148, 839}, {32, 125}, {145, 411}, {48, 125}, {64, 125}, {151, 214}, {12, 17}, {96, 125}, {151, 107}, {24, 17}, {124,81}, {192, 125}, {127, 45}, {48, 17}, {255, 83} };
  #elif (F_PLL==72000000)
    const tmclk clkArr[numfreqs] = {{832, 1125}, {32, 1125}, {49, 1250}, {64, 1125}, {49, 625}, {32, 375}, {128, 1125}, {98, 625}, {8, 51}, {64, 375}, {196, 625}, {16, 51}, {248,729}, {128, 375}, {249, 397}, {32, 51}, {185, 271} };
  #elif (F_PLL==96000000)
    const tmclk clkArr[numfreqs] = {{2, 375},{8, 375}, {73, 2483}, {16, 375}, {147, 2500}, {8, 125},  {32, 375}, {147, 1250}, {2, 17}, {16, 125}, {147, 625}, {4, 17}, {62,243},{32, 125}, {151, 321}, {8, 17}, {64, 125} };
  #elif (F_PLL==120000000)
    const tmclk clkArr[numfreqs] = {{8, 1875},{32, 1875}, {89, 3784}, {64, 1875}, {147, 3125}, {32, 625},  {128, 1875}, {205, 2179}, {8, 85}, {64, 625}, {89, 473}, {16, 85}, {119,583}, {128, 625}, {178, 473}, {32, 85}, {145, 354} };
  #elif (F_PLL==144000000)
    const tmclk clkArr[numfreqs] = {{4, 1125},{16, 1125}, {49, 2500}, {32, 1125}, {49, 1250}, {16, 375},  {64, 1125}, {49, 625}, {4, 51}, {32, 375}, {98, 625}, {8, 51}, {157,923}, {64, 375}, {196, 625}, {16, 51}, {128, 375} };
  #elif (F_PLL==180000000)
    const tmclk clkArr[numfreqs] = {{9, 3164}, {46, 4043}, {49, 3125}, {73, 3208}, {98, 3125}, {64, 1875},  {183, 4021}, {196, 3125}, {16, 255}, {128, 1875}, {107, 853}, {32, 255}, {238,1749}, {219, 1604}, {214, 853}, {64, 255}, {219, 802} };
  #elif (F_PLL==192000000)
    const tmclk clkArr[numfreqs] = {{1, 375}, {4, 375}, {37, 2517}, {8, 375}, {73, 2483}, {4, 125}, {16, 375}, {147, 2500}, {1, 17}, {8, 125}, {147, 1250}, {2, 17}, {31,243}, {16, 125}, {147, 625}, {4, 17}, {32, 125} };
  #elif (F_PLL==216000000)
    const tmclk clkArr[numfreqs] = {{8, 3375}, {32, 3375}, {49, 3750}, {64, 3375}, {49, 1875}, {32, 1125},  {128, 3375}, {98, 1875}, {8, 153}, {64, 1125}, {196, 1875}, {16, 153}, {248,2187}, {128, 1125}, {226, 1081}, {32, 153}, {147, 646} };
  #elif (F_PLL==240000000)
    const tmclk clkArr[numfreqs] = {{4, 1875}, {16, 1875}, {29, 2466}, {32, 1875}, {89, 3784}, {16, 625}, {64, 1875}, {147, 3125}, {4, 85}, {32, 625}, {205, 2179}, {8, 85}, {119,1166}, {64, 625}, {89, 473}, {16, 85}, {128, 625} };
  #endif
  
    for (int f = 0; f < numfreqs; f++) {
      if ( freq == samplefreqs[f] ) {
        while (I2S0_MCR & I2S_MCR_DUF) ;
  		I2S0_MDR = I2S_MDR_FRACT((clkArr[f].mult - 1)) | I2S_MDR_DIVIDE((clkArr[f].div - 1));
  		return (float)(F_PLL / 256 * clkArr[f].mult / clkArr[f].div);
      }
    }
    return 0.0f;
  }
  
  audio_block_f32_t * AudioOutputI2S_F32::block_left_1st = NULL;
  audio_block_f32_t * AudioOutputI2S_F32::block_right_1st = NULL;
  audio_block_f32_t * AudioOutputI2S_F32::block_left_2nd = NULL;
  audio_block_f32_t * AudioOutputI2S_F32::block_right_2nd = NULL;
  uint16_t  AudioOutputI2S_F32::block_left_offset = 0;
  uint16_t  AudioOutputI2S_F32::block_right_offset = 0;
  bool AudioOutputI2S_F32::update_responsibility = false;
  //DMAMEM static uint32_t i2s_tx_buffer[AUDIO_BLOCK_SAMPLES]; //local audio_block_samples should be no larger than global AUDIO_BLOCK_SAMPLES
  DMAMEM static int32_t i2s_tx_buffer[2*AUDIO_BLOCK_SAMPLES]; //2 channels at 32-bits per sample.  Local "audio_block_samples" should be no larger than global "AUDIO_BLOCK_SAMPLES"
  DMAChannel AudioOutputI2S_F32::dma(false);
  
  float AudioOutputI2S_F32::sample_rate_Hz = AUDIO_SAMPLE_RATE;
  int AudioOutputI2S_F32::audio_block_samples = AUDIO_BLOCK_SAMPLES;
  
  //#for 16-bit transfers
  //#define I2S_BUFFER_TO_USE_BYTES (AudioOutputI2S_F32::audio_block_samples*sizeof(i2s_tx_buffer[0]))
  
  //#for 32-bit transfers
  #define I2S_BUFFER_TO_USE_BYTES (AudioOutputI2S_F32::audio_block_samples*2*sizeof(i2s_tx_buffer[0]))
  
  
  void AudioOutputI2S_F32::begin(void)
  {
  	bool transferUsing32bit = true;
  	begin(transferUsing32bit);
  }
  void AudioOutputI2S_F32::begin(bool transferUsing32bit) {
  
  	dma.begin(true); // Allocate the DMA channel first
  
  	block_left_1st = NULL;
  	block_right_1st = NULL;
  
  	// TODO: should we set & clear the I2S_TCSR_SR bit here?
  	config_i2s(transferUsing32bit);
  
  	CORE_PIN22_CONFIG = PORT_PCR_MUX(6); // pin 22, PTC1, I2S0_TXD0
  
  	//setup DMA parameters
  	//if (transferUsing32bit) {
  		sub_begin_i32();
  	//} else {
  	//	sub_begin_i16();
  	//}
  	
  	dma.triggerAtHardwareEvent(DMAMUX_SOURCE_I2S0_TX);
  	update_responsibility = update_setup();
  	dma.enable();
  
  	I2S0_TCSR = I2S_TCSR_SR;
  	I2S0_TCSR = I2S_TCSR_TE | I2S_TCSR_BCE | I2S_TCSR_FRDE;
  	dma.attachInterrupt(isr_32);
  	
  	// change the I2S frequencies to make the requested sample rate
  	setI2SFreq(AudioOutputI2S_F32::sample_rate_Hz);
  	
  	enabled = 1;
  	
  	//AudioInputI2S_F32::begin_guts();
  }
  
  void AudioOutputI2S_F32::sub_begin_i16(void) {
  	dma.TCD->SADDR = i2s_tx_buffer;
  	dma.TCD->SOFF = 2;
  	dma.TCD->ATTR = DMA_TCD_ATTR_SSIZE(1) | DMA_TCD_ATTR_DSIZE(1);
  	dma.TCD->NBYTES_MLNO = 2;
  	//dma.TCD->SLAST = -sizeof(i2s_tx_buffer);	//original
  	dma.TCD->SLAST = -I2S_BUFFER_TO_USE_BYTES;
  	dma.TCD->DADDR = &I2S0_TDR0;
  	dma.TCD->DOFF = 0;
  	//dma.TCD->CITER_ELINKNO = sizeof(i2s_tx_buffer) / 2;	//original
  	dma.TCD->CITER_ELINKNO = I2S_BUFFER_TO_USE_BYTES / 2;
  	dma.TCD->DLASTSGA = 0;
  	//dma.TCD->BITER_ELINKNO = sizeof(i2s_tx_buffer) / 2;	//original
  	dma.TCD->BITER_ELINKNO = I2S_BUFFER_TO_USE_BYTES / 2;
  	dma.TCD->CSR = DMA_TCD_CSR_INTHALF | DMA_TCD_CSR_INTMAJOR;
  }
  
  void AudioOutputI2S_F32::sub_begin_i32(void) {
  	dma.TCD->SADDR = i2s_tx_buffer; //here's where to get the data from
  	
  	//let's assume that we'll transfer each sample (left or right) independently.  So 4-byte (32bit) transfers.
  	dma.TCD->SOFF = 4;	   //step forward pointer for source data by 4 bytes (ie, 32 bits) after each read
  	dma.TCD->ATTR = DMA_TCD_ATTR_SSIZE(DMA_TCD_ATTR_SIZE_32BIT) | DMA_TCD_ATTR_DSIZE(DMA_TCD_ATTR_SIZE_32BIT); //each read is 32 bits
  	dma.TCD->NBYTES_MLNO = 4;   //how many bytes to send per minor loop. Do each sample (left or right) independently.  So, 4 bytes?   Should be 4 or 8?
  	
  	//dma.TCD->SLAST = -sizeof(i2s_tx_buffer);	//original
  	dma.TCD->SLAST = -I2S_BUFFER_TO_USE_BYTES;  //jump back to beginning of source data when hit the end
  	dma.TCD->DADDR = &I2S0_TDR0;  //destination of DMA transfers
  	dma.TCD->DOFF = 0;  //do not increment the destination pointer
  	//dma.TCD->CITER_ELINKNO = sizeof(i2s_tx_buffer) / 2;	//original
  	dma.TCD->CITER_ELINKNO = I2S_BUFFER_TO_USE_BYTES / 4;   //number of minor loops in a major loop.  I2S_BUFFER_TO_USE_BYTES/NBYTES_MLNO?  Should be 4 or 8?
  	dma.TCD->DLASTSGA = 0;
  	//dma.TCD->BITER_ELINKNO = sizeof(i2s_tx_buffer) / 2;	//original
  	dma.TCD->BITER_ELINKNO = I2S_BUFFER_TO_USE_BYTES / 4;   //number of minor loops in a major loop.  I2S_BUFFER_TO_USE_BYTES/NBYTES_MLNO? should be 4 or 8?
  	dma.TCD->CSR = DMA_TCD_CSR_INTHALF | DMA_TCD_CSR_INTMAJOR;
  }
  
  
  /* void AudioOutputI2S_F32::isr_16(void)
  {
  #if defined(KINETISK)
  	int16_t *dest;
  	audio_block_t *blockL, *blockR;
  	uint32_t saddr, offsetL, offsetR;
  
  	saddr = (uint32_t)(dma.TCD->SADDR);
  	dma.clearInterrupt();
  	//if (saddr < (uint32_t)i2s_tx_buffer + sizeof(i2s_tx_buffer) / 2) {	//original
  	if (saddr < (uint32_t)i2s_tx_buffer + I2S_BUFFER_TO_USE_BYTES / 2) {	
  		// DMA is transmitting the first half of the buffer
  		// so we must fill the second half
  		//dest = (int16_t *)&i2s_tx_buffer[AUDIO_BLOCK_SAMPLES/2];	//original
  		dest = (int16_t *)&i2s_tx_buffer[audio_block_samples/2];
  		if (AudioOutputI2S_F32::update_responsibility) AudioStream_F32::update_all();
  	} else {
  		// DMA is transmitting the second half of the buffer
  		// so we must fill the first half
  		dest = (int16_t *)i2s_tx_buffer;
  	}
  
  	blockL = AudioOutputI2S_F32::block_left_1st;
  	blockR = AudioOutputI2S_F32::block_right_1st;
  	offsetL = AudioOutputI2S_F32::block_left_offset;
  	offsetR = AudioOutputI2S_F32::block_right_offset;
  
  	
  	int16_t *d = dest;
  	if (blockL && blockR) {
  		//memcpy_tointerleaveLR(dest, blockL->data + offsetL, blockR->data + offsetR);
  		//memcpy_tointerleaveLRwLen(dest, blockL->data + offsetL, blockR->data + offsetR, audio_block_samples/2);
  		int16_t *pL = blockL->data + offsetL;
  		int16_t *pR = blockR->data + offsetR;
  		for (int i=0; i < audio_block_samples/2; i++) {	*d++ = *pL++; *d++ = *pR++; } //interleave
  		offsetL += audio_block_samples / 2;
  		offsetR += audio_block_samples / 2;
  	} else if (blockL) {
  		//memcpy_tointerleaveLR(dest, blockL->data + offsetL, blockR->data + offsetR);
  		int16_t *pL = blockL->data + offsetL;
  		for (int i=0; i < audio_block_samples / 2 * 2; i+=2) { *(d+i) = *pL++; } //interleave
  		offsetL += audio_block_samples / 2;
  	} else if (blockR) {
  		int16_t *pR = blockR->data + offsetR;
  		for (int i=0; i < audio_block_samples /2 * 2; i+=2) { *(d+i) = *pR++; } //interleave
  		offsetR += audio_block_samples / 2;
  	} else {
  		//memset(dest,0,AUDIO_BLOCK_SAMPLES * 2);
  		memset(dest,0,audio_block_samples * 2);
  		return;
  	}
  
  	//if (offsetL < AUDIO_BLOCK_SAMPLES) { //original
  	if (offsetL < (uint16_t)audio_block_samples) {
  		AudioOutputI2S_F32::block_left_offset = offsetL;
  	} else {
  		AudioOutputI2S_F32::block_left_offset = 0;
  		AudioStream::release(blockL);
  		AudioOutputI2S_F32::block_left_1st = AudioOutputI2S_F32::block_left_2nd;
  		AudioOutputI2S_F32::block_left_2nd = NULL;
  	}
  	//if (offsetR < AUDIO_BLOCK_SAMPLES) {
  	if (offsetR < (uint16_t)audio_block_samples) {
  		AudioOutputI2S_F32::block_right_offset = offsetR;
  	} else {
  		AudioOutputI2S_F32::block_right_offset = 0;
  		AudioStream::release(blockR);
  		AudioOutputI2S_F32::block_right_1st = AudioOutputI2S_F32::block_right_2nd;
  		AudioOutputI2S_F32::block_right_2nd = NULL;
  	}
  #else
  	const int16_t *src, *end;
  	int16_t *dest;
  	audio_block_t *block;
  	uint32_t saddr, offset;
  
  	saddr = (uint32_t)(dma.CFG->SAR);
  	dma.clearInterrupt();
  	if (saddr < (uint32_t)i2s_tx_buffer + sizeof(i2s_tx_buffer) / 2) {
  		// DMA is transmitting the first half of the buffer
  		// so we must fill the second half
  		dest = (int16_t *)&i2s_tx_buffer[AUDIO_BLOCK_SAMPLES/2];
  		end = (int16_t *)&i2s_tx_buffer[AUDIO_BLOCK_SAMPLES];
  		if (AudioOutputI2S_F32::update_responsibility) AudioStream_F32::update_all();
  	} else {
  		// DMA is transmitting the second half of the buffer
  		// so we must fill the first half
  		dest = (int16_t *)i2s_tx_buffer;
  		end = (int16_t *)&i2s_tx_buffer[AUDIO_BLOCK_SAMPLES/2];
  	}
  
  	block = AudioOutputI2S_F32::block_left_1st;
  	if (block) {
  		offset = AudioOutputI2S_F32::block_left_offset;
  		src = &block->data[offset];
  		do {
  			*dest = *src++;
  			dest += 2;
  		} while (dest < end);
  		offset += AUDIO_BLOCK_SAMPLES/2;
  		if (offset < AUDIO_BLOCK_SAMPLES) {
  			AudioOutputI2S_F32::block_left_offset = offset;
  		} else {
  			AudioOutputI2S_F32::block_left_offset = 0;
  			AudioStream::release(block);
  			AudioOutputI2S_F32::block_left_1st = AudioOutputI2S_F32::block_left_2nd;
  			AudioOutputI2S_F32::block_left_2nd = NULL;
  		}
  	} else {
  		do {
  			*dest = 0;
  			dest += 2;
  		} while (dest < end);
  	}
  	dest -= AUDIO_BLOCK_SAMPLES - 1;
  	block = AudioOutputI2S_F32::block_right_1st;
  	if (block) {
  		offset = AudioOutputI2S_F32::block_right_offset;
  		src = &block->data[offset];
  		do {
  			*dest = *src++;
  			dest += 2;
  		} while (dest < end);
  		offset += AUDIO_BLOCK_SAMPLES/2;
  		if (offset < AUDIO_BLOCK_SAMPLES) {
  			AudioOutputI2S_F32::block_right_offset = offset;
  		} else {
  			AudioOutputI2S_F32::block_right_offset = 0;
  			AudioStream::release(block);
  			AudioOutputI2S_F32::block_right_1st = AudioOutputI2S_F32::block_right_2nd;
  			AudioOutputI2S_F32::block_right_2nd = NULL;
  		}
  	} else {
  		do {
  			*dest = 0;
  			dest += 2;
  		} while (dest < end);
  	}
  #endif
  } */
  
  void AudioOutputI2S_F32::isr_32(void)  //should be called every half of an audio block
  {
  	int32_t *dest;  //int32 is the data type being sent to the audio codec
  	audio_block_f32_t *blockL, *blockR;
  	uint32_t saddr; 
  	uint32_t offsetL, offsetR;
  	
  	saddr = (uint32_t)(dma.TCD->SADDR);
  	dma.clearInterrupt();
  	//if (saddr < (uint32_t)i2s_tx_buffer + sizeof(i2s_tx_buffer) / 2) {	//original 16-bit
  	if (saddr < (uint32_t)i2s_tx_buffer + I2S_BUFFER_TO_USE_BYTES / 2) {	//are we transmitting the first half or second half of the buffer?
  		// DMA is transmitting the first half of the buffer
  		// so we must fill the second half
  		//dest = (int16_t *)&i2s_tx_buffer[AUDIO_BLOCK_SAMPLES/2];	//original, half-way through buffer (buffer is 32-bit elements filled with 16-bit stereo samples)
  		dest = (int32_t *)&i2s_tx_buffer[2*(audio_block_samples/2)];  //half-way through the buffer..remember, buffer is 32-bit elements filled with 32-bit stereo samples)
  		if (AudioOutputI2S_F32::update_responsibility) AudioStream_F32::update_all();
  	} else {
  		// DMA is transmitting the second half of the buffer so we must fill the first half
  		dest = (int32_t *)i2s_tx_buffer;  //beginning of the buffer
  	}
  
  	blockL = AudioOutputI2S_F32::block_left_1st;
  	blockR = AudioOutputI2S_F32::block_right_1st;
  	offsetL = AudioOutputI2S_F32::block_left_offset;
  	offsetR = AudioOutputI2S_F32::block_right_offset;
  
  	
  	int32_t *d = dest;
  	if (blockL && blockR) {
  		//memcpy_tointerleaveLR(dest, blockL->data + offsetL, blockR->data + offsetR);
  		//memcpy_tointerleaveLRwLen(dest, blockL->data + offsetL, blockR->data + offsetR, audio_block_samples/2);
  		float32_t *pL = blockL->data + offsetL;
  		float32_t *pR = blockR->data + offsetR;
  		for (int i=0; i < audio_block_samples/2; i++) {	//loop over half of the audio block (this routine gets called every half an audio block)
  			*d++ = (int32_t) (*pL++); 	
  			*d++ = (int32_t) (*pR++); //cast and interleave
  		}
  		offsetL += (audio_block_samples / 2);
  		offsetR += (audio_block_samples / 2);
  	} else if (blockL) {
  		//memcpy_tointerleaveLR(dest, blockL->data + offsetL, blockR->data + offsetR);
  		float32_t *pL = blockL->data + offsetL;
  		for (int i=0; i < audio_block_samples /2; i++) { 
  			*d++ = (int32_t) *pL++;  //cast and interleave
  			*d++ = 0;
  		}
  		offsetL += (audio_block_samples / 2);
  	} else if (blockR) {
  		float32_t *pR = blockR->data + offsetR;
  		for (int i=0; i < audio_block_samples /2; i++) { 
  			*d++ = 0;
  			*d++ = (int32_t) *pR++;  //cast and interleave
  		} 
  		offsetR += (audio_block_samples / 2);
  	} else {
  		//memset(dest,0,AUDIO_BLOCK_SAMPLES * 2);  //half buffer (AUDIO_BLOCK_SAMPLES/2), 16-bits per sample (AUDIO_BLOCK_SAMPLES/2*2), stereo (AUDIO_BLOCK_SAMPLES/2*2*2)
  		//memset(dest,0,audio_block_samples * 2 * 4 / 2);//half buffer (AUDIO_BLOCK_SAMPLES/2), 32-bits per sample (AUDIO_BLOCK_SAMPLES/2*4), stereo (AUDIO_BLOCK_SAMPLES/2*4*2)
  		for (int i=0; i < audio_block_samples/2; i++) {	//loop over half of the audio block (this routine gets called every half an audio block)
  			*d++ = (int32_t) 0; 	
  			*d++ = (int32_t) 0; 
  			//*d++ = (int32_t) (-200000000L);
  		}
  		return;
  	}
  
  	//if (offsetL < AUDIO_BLOCK_SAMPLES) { //original
  	if (offsetL < (uint16_t)audio_block_samples) {
  		AudioOutputI2S_F32::block_left_offset = offsetL;
  	} else {
  		AudioOutputI2S_F32::block_left_offset = 0;
  		AudioStream_F32::release(blockL);
  		AudioOutputI2S_F32::block_left_1st = AudioOutputI2S_F32::block_left_2nd;
  		AudioOutputI2S_F32::block_left_2nd = NULL;
  	}
  	//if (offsetR < AUDIO_BLOCK_SAMPLES) {
  	if (offsetR < (uint16_t)audio_block_samples) {
  		AudioOutputI2S_F32::block_right_offset = offsetR;
  	} else {
  		AudioOutputI2S_F32::block_right_offset = 0;
  		AudioStream_F32::release(blockR);
  		AudioOutputI2S_F32::block_right_1st = AudioOutputI2S_F32::block_right_2nd;
  		AudioOutputI2S_F32::block_right_2nd = NULL;
  	}
  
  }
  
  void AudioOutputI2S_F32::scale_f32_to_i16(float32_t *p_f32, float32_t *p_i16, int len) {
  	for (int i=0; i<len; i++) { *p_i16++ = max(-32767,min(32767,(*p_f32++) * 32767.f)); }
  }
  #define F32_TO_I24_NORM_FACTOR (8388607)   //which is 2^23-1
  void AudioOutputI2S_F32::scale_f32_to_i24( float32_t *p_f32, float32_t *p_i24, int len) {
  	for (int i=0; i<len; i++) { *p_i24++ = max(-F32_TO_I24_NORM_FACTOR,min(F32_TO_I24_NORM_FACTOR,(*p_f32++) * F32_TO_I24_NORM_FACTOR)); }
  }
  #define F32_TO_I32_NORM_FACTOR (2147483647)   //which is 2^31-1
  //define F32_TO_I32_NORM_FACTOR (8388607)   //which is 2^23-1
  void AudioOutputI2S_F32::scale_f32_to_i32( float32_t *p_f32, float32_t *p_i32, int len) {
  	for (int i=0; i<len; i++) { *p_i32++ = max(-F32_TO_I32_NORM_FACTOR,min(F32_TO_I32_NORM_FACTOR,(*p_f32++) * F32_TO_I32_NORM_FACTOR)); }
  	//for (int i=0; i<len; i++) { *p_i32++ = (*p_f32++) * F32_TO_I32_NORM_FACTOR + 512.f*8388607.f; }
  }
  
  //update has to be carefully coded so that, if audio_blocks are not available, the code exits
  //gracefully and won't hang.  That'll cause the whole system to hang, which would be very bad.
  void AudioOutputI2S_F32::update(void)
  {
  	// null audio device: discard all incoming data
  	//if (!active) return;
  	//audio_block_t *block = receiveReadOnly();
  	//if (block) release(block);
  
  	audio_block_f32_t *block_f32;
  	audio_block_f32_t *block_f32_scaled = AudioStream_F32::allocate_f32();
  	audio_block_f32_t *block2_f32_scaled = AudioStream_F32::allocate_f32();
  	if ((!block_f32_scaled) || (!block2_f32_scaled)) {
  		//couldn't get some working memory.  Return.
  		if (block_f32_scaled) AudioStream_F32::release(block_f32_scaled);
  		if (block2_f32_scaled) AudioStream_F32::release(block2_f32_scaled);
  		return;
  	}
  	
  	//now that we have our working memory, proceed with getting the audio data and processing
  	block_f32 = receiveReadOnly_f32(0); // input 0 = left channel
  	if (block_f32) {
  		if (block_f32->length != audio_block_samples) {
  			Serial.print("AudioOutputI2S_F32: *** WARNING ***: audio_block says len = ");
  			Serial.print(block_f32->length);
  			Serial.print(", but I2S settings want it to be = ");
  			Serial.println(audio_block_samples);
  		}
  		//Serial.print("AudioOutputI2S_F32: audio_block_samples = ");
  		//Serial.println(audio_block_samples);
  	
  		//scale F32 to Int32
  		//block_f32_scaled = AudioStream_F32::allocate_f32();
  		scale_f32_to_i32(block_f32->data, block_f32_scaled->data, audio_block_samples);
  		
  		//now process the data blocks
  		__disable_irq();
  		if (block_left_1st == NULL) {
  			block_left_1st = block_f32_scaled;
  			block_left_offset = 0;
  			__enable_irq();
  		} else if (block_left_2nd == NULL) {
  			block_left_2nd = block_f32_scaled;
  			__enable_irq();
  		} else {
  			audio_block_f32_t *tmp = block_left_1st;
  			block_left_1st = block_left_2nd;
  			block_left_2nd = block_f32_scaled;
  			block_left_offset = 0;
  			__enable_irq();
  			AudioStream_F32::release(tmp);
  		}
  		AudioStream_F32::transmit(block_f32,0);	AudioStream_F32::release(block_f32); //echo the incoming audio out the outputs
  	} else {
  		//this branch should never get called, but if it does, let's release the buffer that was never used
  		AudioStream_F32::release(block_f32_scaled);
  	}
  	
  	block_f32_scaled = block2_f32_scaled;  //this is simply renaming the pre-allocated buffer
  	block_f32 = receiveReadOnly_f32(1); // input 1 = right channel
  	if (block_f32) {
  		//scale F32 to Int32
  		//block_f32_scaled = AudioStream_F32::allocate_f32();
  		scale_f32_to_i32(block_f32->data, block_f32_scaled->data, audio_block_samples);
  		
  		__disable_irq();
  		if (block_right_1st == NULL) {
  			block_right_1st = block_f32_scaled;
  			block_right_offset = 0;
  			__enable_irq();
  		} else if (block_right_2nd == NULL) {
  			block_right_2nd = block_f32_scaled;
  			__enable_irq();
  		} else {
  			audio_block_f32_t *tmp = block_right_1st;
  			block_right_1st = block_right_2nd;
  			block_right_2nd = block_f32_scaled;
  			block_right_offset = 0;
  			__enable_irq();
  			AudioStream_F32::release(tmp);
  		}
  		AudioStream_F32::transmit(block_f32,1);	AudioStream_F32::release(block_f32); //echo the incoming audio out the outputs
  	} else {
  		//this branch should never get called, but if it does, let's release the buffer that was never used
  		AudioStream_F32::release(block_f32_scaled);
  	}
  }
  
  
  // MCLK needs to be 48e6 / 1088 * 256 = 11.29411765 MHz -> 44.117647 kHz sample rate
  //
  #if F_CPU == 96000000 || F_CPU == 48000000 || F_CPU == 24000000
    // PLL is at 96 MHz in these modes
    #define MCLK_MULT 2
    #define MCLK_DIV  17
  #elif F_CPU == 72000000
    #define MCLK_MULT 8
    #define MCLK_DIV  51
  #elif F_CPU == 120000000
    #define MCLK_MULT 8
    #define MCLK_DIV  85
  #elif F_CPU == 144000000
    #define MCLK_MULT 4
    #define MCLK_DIV  51
  #elif F_CPU == 168000000
    #define MCLK_MULT 8
    #define MCLK_DIV  119
  #elif F_CPU == 180000000
    #define MCLK_MULT 16
    #define MCLK_DIV  255
    #define MCLK_SRC  0
  #elif F_CPU == 192000000
    #define MCLK_MULT 1
    #define MCLK_DIV  17
  #elif F_CPU == 216000000
    #define MCLK_MULT 8
    #define MCLK_DIV  153
    #define MCLK_SRC  0
  #elif F_CPU == 240000000
    #define MCLK_MULT 4
    #define MCLK_DIV  85
  #elif F_CPU == 16000000
    #define MCLK_MULT 12
    #define MCLK_DIV  17
  #else
    #error "This CPU Clock Speed is not supported by the Audio library";
  #endif
  
  #ifndef MCLK_SRC
  	#if (F_CPU >= 20000000)
  	  #define MCLK_SRC  3  // the PLL
  	#else
  	  #define MCLK_SRC  0  // system clock
  	#endif
  #endif
  
  void AudioOutputI2S_F32::config_i2s(void) {	config_i2s(false); }
  void AudioOutputI2S_F32::config_i2s(bool transferUsing32bit) {
  	SIM_SCGC6 |= SIM_SCGC6_I2S;
  	SIM_SCGC7 |= SIM_SCGC7_DMA;
  	SIM_SCGC6 |= SIM_SCGC6_DMAMUX;
  	
  	// if either transmitter or receiver is enabled, do nothing
  	if (I2S0_TCSR & I2S_TCSR_TE) return;
  	if (I2S0_RCSR & I2S_RCSR_RE) return;
  	
  	//if (transferUsing32bit) {
  		config_i2s_i32();
  	//} else {
  	//	config_i2s_i16();
  	//}
  
  	// configure pin mux for 3 clock signals
  	CORE_PIN23_CONFIG = PORT_PCR_MUX(6); // pin 23, PTC2, I2S0_TX_FS (LRCLK)
  	CORE_PIN9_CONFIG  = PORT_PCR_MUX(6); // pin  9, PTC3, I2S0_TX_BCLK
  	CORE_PIN11_CONFIG = PORT_PCR_MUX(6); // pin 11, PTC6, I2S0_MCLK	
  }
  
  void AudioOutputI2S_F32::config_i2s_i16(void)
  {
  
  	// enable MCLK output
  	I2S0_MCR = I2S_MCR_MICS(MCLK_SRC) | I2S_MCR_MOE;
  	while (I2S0_MCR & I2S_MCR_DUF) ;
  	I2S0_MDR = I2S_MDR_FRACT((MCLK_MULT-1)) | I2S_MDR_DIVIDE((MCLK_DIV-1));
  
  	// configure transmitter
  	I2S0_TMR = 0;
  	I2S0_TCR1 = I2S_TCR1_TFW(1);  // watermark at half fifo size
  	I2S0_TCR2 = I2S_TCR2_SYNC(0) | I2S_TCR2_BCP | I2S_TCR2_MSEL(1)
  		| I2S_TCR2_BCD | I2S_TCR2_DIV(3);  //for 32-bit, use I2S_TCR2_DIV(1)
  	I2S0_TCR3 = I2S_TCR3_TCE;
  	I2S0_TCR4 = I2S_TCR4_FRSZ(1) | I2S_TCR4_SYWD(15) | I2S_TCR4_MF
  		| I2S_TCR4_FSE | I2S_TCR4_FSP | I2S_TCR4_FSD;   //for 32-bit use  I2S_TCR4_SYWD(31)
  	I2S0_TCR5 = I2S_TCR5_WNW(15) | I2S_TCR5_W0W(15) | I2S_TCR5_FBT(15); //for 32-bit, change all 15 to 31
  
  	// configure receiver (sync'd to transmitter clocks)
  	I2S0_RMR = 0;
  	I2S0_RCR1 = I2S_RCR1_RFW(1);
  	I2S0_RCR2 = I2S_RCR2_SYNC(1) | I2S_TCR2_BCP | I2S_RCR2_MSEL(1)
  		| I2S_RCR2_BCD | I2S_RCR2_DIV(3); //for 32-bit, change I2S_RCR2_DIV(3) to I2S_RCR2_DIV(1)
  	I2S0_RCR3 = I2S_RCR3_RCE;
  	I2S0_RCR4 = I2S_RCR4_FRSZ(1) | I2S_RCR4_SYWD(15) | I2S_RCR4_MF
  		| I2S_RCR4_FSE | I2S_RCR4_FSP | I2S_RCR4_FSD; //for 32-bit, change I2S_RCR4_SYWD(15) to I2S_RCR4_SYWD(31)
  	I2S0_RCR5 = I2S_RCR5_WNW(15) | I2S_RCR5_W0W(15) | I2S_RCR5_FBT(15); //for 32-bit, change all 15 to 31
  
  
  }
  
  //32-bit transfers.  Taken from: https://github.com/WMXZ-EU/BasicAudioLogger/blob/master/I2S_32.h
  void AudioOutputI2S_F32::config_i2s_i32(void)
  {
  
    // enable MCLK output
    I2S0_MCR = I2S_MCR_MICS(MCLK_SRC) | I2S_MCR_MOE;
    while (I2S0_MCR & I2S_MCR_DUF) ;
    I2S0_MDR = I2S_MDR_FRACT((MCLK_MULT-1)) | I2S_MDR_DIVIDE((MCLK_DIV-1));
    //I2S0_MDR = I2S_MDR_FRACT((MCLK_MULT-1)) | I2S_MDR_DIVIDE((MCLK_DIV/2-1)); //For 32-bit?
  
    // configure transmitter
    I2S0_TMR = 0;
    I2S0_TCR1 = I2S_TCR1_TFW(1);  // watermark at half fifo size.  should be 1 or 2?
    I2S0_TCR2 = I2S_TCR2_SYNC(0) | I2S_TCR2_BCP | I2S_TCR2_MSEL(1)
      | I2S_TCR2_BCD | I2S_TCR2_DIV(1);  //transmitter must be set to asynchronous mode, 
    I2S0_TCR3 = I2S_TCR3_TCE;
    I2S0_TCR4 = I2S_TCR4_FRSZ(1) | I2S_TCR4_SYWD(31) | I2S_TCR4_MF
      | I2S_TCR4_FSE | I2S_TCR4_FSP | I2S_TCR4_FSD;
    I2S0_TCR5 = I2S_TCR5_WNW(31) | I2S_TCR5_W0W(31) | I2S_TCR5_FBT(31);
  
    // configure receiver (sync'd to transmitter clocks)
    I2S0_RMR = 0;
    I2S0_RCR1 = I2S_RCR1_RFW(1); // watermark at half fifo size.  should be 1 or 2?
    I2S0_RCR2 = I2S_RCR2_SYNC(1) | I2S_TCR2_BCP | I2S_RCR2_MSEL(1)
      | I2S_RCR2_BCD | I2S_RCR2_DIV(1);  //receiver set to syncrhonous
    I2S0_RCR3 = I2S_RCR3_RCE;
    I2S0_RCR4 = I2S_RCR4_FRSZ(1) | I2S_RCR4_SYWD(31) | I2S_RCR4_MF
      | I2S_RCR4_FSE | I2S_RCR4_FSP | I2S_RCR4_FSD;
    I2S0_RCR5 = I2S_RCR5_WNW(31) | I2S_RCR5_W0W(31) | I2S_RCR5_FBT(31);
  
  }
  /******************************************************************/
  
  /*
  void AudioOutputI2Sslave::begin(void)
  {
  	dma.begin(true); // Allocate the DMA channel first
  
  	//pinMode(2, OUTPUT);
  	block_left_1st = NULL;
  	block_right_1st = NULL;
  
  	AudioOutputI2Sslave::config_i2s();
  	CORE_PIN22_CONFIG = PORT_PCR_MUX(6); // pin 22, PTC1, I2S0_TXD0
  #if defined(KINETISK)
  	dma.TCD->SADDR = i2s_tx_buffer;
  	dma.TCD->SOFF = 2;
  	dma.TCD->ATTR = DMA_TCD_ATTR_SSIZE(1) | DMA_TCD_ATTR_DSIZE(1);
  	dma.TCD->NBYTES_MLNO = 2;
  	dma.TCD->SLAST = -sizeof(i2s_tx_buffer);
  	dma.TCD->DADDR = &I2S0_TDR0;
  	dma.TCD->DOFF = 0;
  	dma.TCD->CITER_ELINKNO = sizeof(i2s_tx_buffer) / 2;
  	dma.TCD->DLASTSGA = 0;
  	dma.TCD->BITER_ELINKNO = sizeof(i2s_tx_buffer) / 2;
  	dma.TCD->CSR = DMA_TCD_CSR_INTHALF | DMA_TCD_CSR_INTMAJOR;
  #endif
  	dma.triggerAtHardwareEvent(DMAMUX_SOURCE_I2S0_TX);
  	update_responsibility = update_setup();
  	dma.enable();
  
  	I2S0_TCSR |= I2S_TCSR_TE | I2S_TCSR_BCE | I2S_TCSR_FRDE | I2S_TCSR_FR;
  	dma.attachInterrupt(isr);
  }
  
  void AudioOutputI2Sslave::config_i2s(void)
  {
  	SIM_SCGC6 |= SIM_SCGC6_I2S;
  	SIM_SCGC7 |= SIM_SCGC7_DMA;
  	SIM_SCGC6 |= SIM_SCGC6_DMAMUX;
  
  	// if either transmitter or receiver is enabled, do nothing
  	if (I2S0_TCSR & I2S_TCSR_TE) return;
  	if (I2S0_RCSR & I2S_RCSR_RE) return;
  
  	// Select input clock 0
  	// Configure to input the bit-clock from pin, bypasses the MCLK divider
  	I2S0_MCR = I2S_MCR_MICS(0);
  	I2S0_MDR = 0;
  
  	// configure transmitter
  	I2S0_TMR = 0;
  	I2S0_TCR1 = I2S_TCR1_TFW(1);  // watermark at half fifo size
  	I2S0_TCR2 = I2S_TCR2_SYNC(0) | I2S_TCR2_BCP;
  
  	I2S0_TCR3 = I2S_TCR3_TCE;
  	I2S0_TCR4 = I2S_TCR4_FRSZ(1) | I2S_TCR4_SYWD(15) | I2S_TCR4_MF
  		| I2S_TCR4_FSE | I2S_TCR4_FSP;
  
  	I2S0_TCR5 = I2S_TCR5_WNW(15) | I2S_TCR5_W0W(15) | I2S_TCR5_FBT(15);
  
  	// configure receiver (sync'd to transmitter clocks)
  	I2S0_RMR = 0;
  	I2S0_RCR1 = I2S_RCR1_RFW(1);
  	I2S0_RCR2 = I2S_RCR2_SYNC(1) | I2S_TCR2_BCP;
  
  	I2S0_RCR3 = I2S_RCR3_RCE;
  	I2S0_RCR4 = I2S_RCR4_FRSZ(1) | I2S_RCR4_SYWD(15) | I2S_RCR4_MF
  		| I2S_RCR4_FSE | I2S_RCR4_FSP | I2S_RCR4_FSD;
  
  	I2S0_RCR5 = I2S_RCR5_WNW(15) | I2S_RCR5_W0W(15) | I2S_RCR5_FBT(15);
  
  	// configure pin mux for 3 clock signals
  	CORE_PIN23_CONFIG = PORT_PCR_MUX(6); // pin 23, PTC2, I2S0_TX_FS (LRCLK)
  	CORE_PIN9_CONFIG  = PORT_PCR_MUX(6); // pin  9, PTC3, I2S0_TX_BCLK
  	CORE_PIN11_CONFIG = PORT_PCR_MUX(6); // pin 11, PTC6, I2S0_MCLK
  }
  */
  
  