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teensy-audio/output_pt8211.cpp

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  1. /* Audio Library for Teensy 3.X

  2.  * Copyright (c) 2016, Paul Stoffregen, paul@pjrc.com

  3.  *

  4.  * Development of this audio library was funded by PJRC.COM, LLC by sales of

  5.  * Teensy and Audio Adaptor boards.  Please support PJRC's efforts to develop

  6.  * open source software by purchasing Teensy or other PJRC products.

  7.  *

  8.  * Permission is hereby granted, free of charge, to any person obtaining a copy

  9.  * of this software and associated documentation files (the "Software"), to deal

  10.  * in the Software without restriction, including without limitation the rights

  11.  * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell

  12.  * copies of the Software, and to permit persons to whom the Software is

  13.  * furnished to do so, subject to the following conditions:

  14.  *

  15.  * The above copyright notice, development funding notice, and this permission

  16.  * notice shall be included in all copies or substantial portions of the Software.

  17.  *

  18.  * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR

  19.  * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,

  20.  * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE

  21.  * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER

  22.  * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,

  23.  * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN

  24.  * THE SOFTWARE.

  25.  */

  26. 

  27. //Adapted to PT8211, Frank Bösing, Ben-Rheinland

  28. 

  29. 

  30. #include <Arduino.h>

  31. #include "output_pt8211.h"

  32. #include "memcpy_audio.h"

  33. #include "utility/imxrt_hw.h"

  34. 

  35. audio_block_t * AudioOutputPT8211::block_left_1st = NULL;

  36. audio_block_t * AudioOutputPT8211::block_right_1st = NULL;

  37. audio_block_t * AudioOutputPT8211::block_left_2nd = NULL;

  38. audio_block_t * AudioOutputPT8211::block_right_2nd = NULL;

  39. uint16_t  AudioOutputPT8211::block_left_offset = 0;

  40. uint16_t  AudioOutputPT8211::block_right_offset = 0;

  41. bool AudioOutputPT8211::update_responsibility = false;

  42. #if defined(AUDIO_PT8211_OVERSAMPLING)

  43. 	static uint32_t i2s_tx_buffer[AUDIO_BLOCK_SAMPLES*4];

  44. #else

  45. 	static uint32_t i2s_tx_buffer[AUDIO_BLOCK_SAMPLES];

  46. #endif

  47. DMAChannel AudioOutputPT8211::dma(false);

  48. 

  49. void AudioOutputPT8211::begin(void)

  50. {

  51. 	dma.begin(true); // Allocate the DMA channel first

  52. 

  53. 	block_left_1st = NULL;

  54. 	block_right_1st = NULL;

  55. 

  56. 	// TODO: should we set & clear the I2S_TCSR_SR bit here?

  57. 	config_i2s();

  58. #if defined(KINETISK)

  59. 	CORE_PIN22_CONFIG = PORT_PCR_MUX(6); // pin 22, PTC1, I2S0_TXD0

  60. 

  61. 	dma.TCD->SADDR = i2s_tx_buffer;

  62. 	dma.TCD->SOFF = 2;

  63. 	dma.TCD->ATTR = DMA_TCD_ATTR_SSIZE(1) | DMA_TCD_ATTR_DSIZE(1);

  64. 	dma.TCD->NBYTES_MLNO = 2;

  65. 	dma.TCD->SLAST = -sizeof(i2s_tx_buffer);

  66. 	dma.TCD->DADDR = &I2S0_TDR0;

  67. 	dma.TCD->DOFF = 0;

  68. 	dma.TCD->CITER_ELINKNO = sizeof(i2s_tx_buffer) / 2;

  69. 	dma.TCD->DLASTSGA = 0;

  70. 	dma.TCD->BITER_ELINKNO = sizeof(i2s_tx_buffer) / 2;

  71. 	dma.TCD->CSR = DMA_TCD_CSR_INTHALF | DMA_TCD_CSR_INTMAJOR;

  72. 

  73. 	dma.triggerAtHardwareEvent(DMAMUX_SOURCE_I2S0_TX);

  74. 	I2S0_TCSR |= I2S_TCSR_TE | I2S_TCSR_BCE | I2S_TCSR_FRDE | I2S_TCSR_FR;

  75. 

  76. #elif defined(__IMXRT1052__) || defined(__IMXRT1062__)

  77. 

  78. #if defined(__IMXRT1052__)

  79. 	CORE_PIN6_CONFIG  = 3;  //1:TX_DATA0

  80. #elif defined(__IMXRT1062__)

  81. 	CORE_PIN7_CONFIG  = 3;  //1:TX_DATA0	

  82. #endif

  83. 

  84. 	dma.TCD->SADDR = i2s_tx_buffer;

  85. 	dma.TCD->SOFF = 2;

  86. 	dma.TCD->ATTR = DMA_TCD_ATTR_SSIZE(1) | DMA_TCD_ATTR_DSIZE(1);

  87. 	dma.TCD->NBYTES_MLNO = 2;

  88. 	dma.TCD->SLAST = -sizeof(i2s_tx_buffer);

  89. 	dma.TCD->DOFF = 0;

  90. 	dma.TCD->CITER_ELINKNO = sizeof(i2s_tx_buffer) / 2;

  91. 	dma.TCD->DLASTSGA = 0;

  92. 	dma.TCD->BITER_ELINKNO = sizeof(i2s_tx_buffer) / 2;

  93. 	dma.TCD->CSR = DMA_TCD_CSR_INTHALF | DMA_TCD_CSR_INTMAJOR;

  94. 	dma.TCD->DADDR = (void *)((uint32_t)&I2S1_TDR0);

  95. 	dma.triggerAtHardwareEvent(DMAMUX_SOURCE_SAI1_TX);

  96. 

  97. 	I2S1_RCSR |= I2S_RCSR_RE;

  98. 	I2S1_TCSR |= I2S_TCSR_TE | I2S_TCSR_BCE | I2S_TCSR_FRDE;

  99. #endif

  100. 	update_responsibility = update_setup();

  101. 	dma.attachInterrupt(isr);

  102. 	dma.enable();

  103. }

  104. 

  105. void AudioOutputPT8211::isr(void)

  106. {

  107. 	int16_t *dest;

  108. 	audio_block_t *blockL, *blockR;

  109. 	uint32_t saddr, offsetL, offsetR;

  110. 

  111. 	saddr = (uint32_t)(dma.TCD->SADDR);

  112. 	dma.clearInterrupt();

  113. 	if (saddr < (uint32_t)i2s_tx_buffer + sizeof(i2s_tx_buffer) / 2) {

  114. 		// DMA is transmitting the first half of the buffer

  115. 		// so we must fill the second half

  116. 		#if defined(AUDIO_PT8211_OVERSAMPLING)

  117. 			dest = (int16_t *)&i2s_tx_buffer[(AUDIO_BLOCK_SAMPLES/2)*4];

  118. 		#else

  119. 			dest = (int16_t *)&i2s_tx_buffer[AUDIO_BLOCK_SAMPLES/2];

  120. 		#endif

  121. 		if (AudioOutputPT8211::update_responsibility) AudioStream::update_all();

  122. 	} else {

  123. 		// DMA is transmitting the second half of the buffer

  124. 		// so we must fill the first half

  125. 		dest = (int16_t *)i2s_tx_buffer;

  126. 	}

  127. 

  128. 	blockL = AudioOutputPT8211::block_left_1st;

  129. 	blockR = AudioOutputPT8211::block_right_1st;

  130. 	offsetL = AudioOutputPT8211::block_left_offset;

  131. 	offsetR = AudioOutputPT8211::block_right_offset;

  132. 

  133. 	#if defined(AUDIO_PT8211_OVERSAMPLING)

  134. 		static int32_t oldL = 0;

  135. 		static int32_t oldR = 0;

  136. 	#endif

  137. 	if (blockL && blockR) {

  138. 		#if defined(AUDIO_PT8211_OVERSAMPLING)

  139. 			#if defined(AUDIO_PT8211_INTERPOLATION_LINEAR)

  140. 				for (int i=0; i< AUDIO_BLOCK_SAMPLES / 2; i++, offsetL++, offsetR++) {

  141. 					int32_t valL = blockL->data[offsetL];

  142. 					int32_t valR = blockR->data[offsetR];

  143. 					int32_t nL = (oldL+valL) >> 1;

  144. 					int32_t nR = (oldR+valR) >> 1;

  145. 					*(dest+0) = (oldL+nL) >> 1;

  146. 					*(dest+1) = (oldR+nR) >> 1;

  147. 					*(dest+2) = nL;

  148. 					*(dest+3) = nR;

  149. 					*(dest+4) = (nL+valL) >> 1;

  150. 					*(dest+5) = (nR+valR) >> 1;

  151. 					*(dest+6) = valL;

  152. 					*(dest+7) = valR;

  153. 					dest+=8;

  154. 					oldL = valL;

  155. 					oldR = valR;

  156. 				}

  157. 			#elif defined(AUDIO_PT8211_INTERPOLATION_CIC)

  158. 				for (int i=0; i< AUDIO_BLOCK_SAMPLES / 2; i++, offsetL++, offsetR++) {

  159. 					int32_t valL = blockL->data[offsetL];

  160. 					int32_t valR = blockR->data[offsetR];

  161. 

  162. 					int32_t combL[3] = {0};

  163. 					static int32_t combLOld[2] = {0};

  164. 					int32_t combR[3] = {0};

  165. 					static int32_t combROld[2] = {0};

  166. 

  167. 					combL[0] = valL - oldL;

  168. 					combR[0] = valR - oldR;

  169. 					combL[1] = combL[0] - combLOld[0];

  170. 					combR[1] = combR[0] - combROld[0];

  171. 					combL[2] = combL[1] - combLOld[1];

  172. 					combR[2] = combR[1] - combROld[1];

  173. 					// combL[2] now holds input val

  174. 					// combR[2] now holds input val

  175. 					oldL = valL;

  176. 					oldR = valR;

  177. 					combLOld[0] = combL[0];

  178. 					combROld[0] = combR[0];

  179. 					combLOld[1] = combL[1];

  180. 					combROld[1] = combR[1];

  181. 					for (int j = 0; j < 4; j++) {

  182. 						int32_t integrateL[3];

  183. 						int32_t integrateR[3];

  184. 						static int32_t integrateLOld[3] = {0};

  185. 						static int32_t integrateROld[3] = {0};

  186. 						integrateL[0] = ( (j==0) ? (combL[2]) : (0) ) + integrateLOld[0];

  187. 						integrateR[0] = ( (j==0) ? (combR[2]) : (0) ) + integrateROld[0];

  188. 						integrateL[1] = integrateL[0] + integrateLOld[1];

  189. 						integrateR[1] = integrateR[0] + integrateROld[1];

  190. 						integrateL[2] = integrateL[1] + integrateLOld[2];

  191. 						integrateR[2] = integrateR[1] + integrateROld[2];

  192. 						// integrateL[2] now holds j'th upsampled value

  193. 						// integrateR[2] now holds j'th upsampled value

  194. 						*(dest+j*2) = integrateL[2] >> 4;

  195. 						*(dest+j*2+1) = integrateR[2] >> 4;

  196. 						integrateLOld[0] = integrateL[0];

  197. 						integrateROld[0] = integrateR[0];

  198. 						integrateLOld[1] = integrateL[1];

  199. 						integrateROld[1] = integrateR[1];

  200. 						integrateLOld[2] = integrateL[2];

  201. 						integrateROld[2] = integrateR[2];

  202. 					}

  203. 					dest+=8;

  204. 				}

  205. 			#else

  206. 				#error no interpolation method defined for oversampling.

  207. 			#endif //defined(AUDIO_PT8211_INTERPOLATION_LINEAR)

  208. 		#else

  209. 			memcpy_tointerleaveLR(dest, blockL->data + offsetL, blockR->data + offsetR);

  210. 			offsetL += AUDIO_BLOCK_SAMPLES / 2;

  211. 			offsetR += AUDIO_BLOCK_SAMPLES / 2;

  212. 		#endif //defined(AUDIO_PT8211_OVERSAMPLING)

  213. 

  214. 	} else if (blockL) {

  215. 		#if defined(AUDIO_PT8211_OVERSAMPLING)

  216. 			#if defined(AUDIO_PT8211_INTERPOLATION_LINEAR)

  217. 				for (int i=0; i< AUDIO_BLOCK_SAMPLES / 2; i++, offsetL++) {

  218. 					int32_t val = blockL->data[offsetL];

  219. 					int32_t n = (oldL+val) >> 1;

  220. 					*(dest+0) = (oldL+n) >> 1;

  221. 					*(dest+1) = 0;

  222. 					*(dest+2) = n;

  223. 					*(dest+3) = 0;

  224. 					*(dest+4) = (n+val) >> 1;

  225. 					*(dest+5) = 0;

  226. 					*(dest+6) = val;

  227. 					*(dest+7) = 0;

  228. 					dest+=8;

  229. 					oldL = val;

  230. 				}

  231. 			#elif defined(AUDIO_PT8211_INTERPOLATION_CIC)

  232. 				for (int i=0; i< AUDIO_BLOCK_SAMPLES / 2; i++, offsetL++, offsetR++) {

  233. 					int32_t valL = blockL->data[offsetL];

  234. 

  235. 					int32_t combL[3] = {0};

  236. 					static int32_t combLOld[2] = {0};

  237. 

  238. 					combL[0] = valL - oldL;

  239. 					combL[1] = combL[0] - combLOld[0];

  240. 					combL[2] = combL[1] - combLOld[1];

  241. 					// combL[2] now holds input val

  242. 					combLOld[0] = combL[0];

  243. 					combLOld[1] = combL[1];

  244. 

  245. 					for (int j = 0; j < 4; j++) {

  246. 						int32_t integrateL[3];

  247. 						static int32_t integrateLOld[3] = {0};

  248. 						integrateL[0] = ( (j==0) ? (combL[2]) : (0) ) + integrateLOld[0];

  249. 						integrateL[1] = integrateL[0] + integrateLOld[1];

  250. 						integrateL[2] = integrateL[1] + integrateLOld[2];

  251. 						// integrateL[2] now holds j'th upsampled value

  252. 						*(dest+j*2) = integrateL[2] >> 4;

  253. 						integrateLOld[0] = integrateL[0];

  254. 						integrateLOld[1] = integrateL[1];

  255. 						integrateLOld[2] = integrateL[2];

  256. 					}

  257. 

  258. 					// fill right channel with zeros:

  259. 					*(dest+1) = 0;

  260. 					*(dest+3) = 0;

  261. 					*(dest+5) = 0;

  262. 					*(dest+7) = 0;

  263. 					dest+=8;

  264. 					oldL = valL;

  265. 				}

  266. 			#else

  267. 				#error no interpolation method defined for oversampling.

  268. 			#endif //defined(AUDIO_PT8211_INTERPOLATION_LINEAR)

  269. 		#else

  270. 			memcpy_tointerleaveL(dest, blockL->data + offsetL);

  271. 			offsetL += (AUDIO_BLOCK_SAMPLES / 2);

  272. 		#endif //defined(AUDIO_PT8211_OVERSAMPLING)

  273. 	} else if (blockR) {

  274. 		#if defined(AUDIO_PT8211_OVERSAMPLING)

  275. 			#if defined(AUDIO_PT8211_INTERPOLATION_LINEAR)

  276. 				for (int i=0; i< AUDIO_BLOCK_SAMPLES / 2; i++, offsetR++) {

  277. 					int32_t val = blockR->data[offsetR];

  278. 					int32_t n = (oldR+val) >> 1;

  279. 					*(dest+0) = 0;

  280. 					*(dest+1) = ((oldR+n) >> 1);

  281. 					*(dest+2) = 0;

  282. 					*(dest+3) = n;

  283. 					*(dest+4) = 0;

  284. 					*(dest+5) = ((n+val) >> 1);

  285. 					*(dest+6) = 0;

  286. 					*(dest+7) = val;

  287. 					dest+=8;

  288. 					oldR = val;

  289. 				}

  290. 			#elif defined(AUDIO_PT8211_INTERPOLATION_CIC)

  291. 				for (int i=0; i< AUDIO_BLOCK_SAMPLES / 2; i++, offsetL++, offsetR++) {

  292. 					int32_t valR = blockR->data[offsetR];

  293. 

  294. 					int32_t combR[3] = {0};

  295. 					static int32_t combROld[2] = {0};

  296. 

  297. 					combR[0] = valR - oldR;

  298. 					combR[1] = combR[0] - combROld[0];

  299. 					combR[2] = combR[1] - combROld[1];

  300. 					// combR[2] now holds input val

  301. 					combROld[0] = combR[0];

  302. 					combROld[1] = combR[1];

  303. 

  304. 					for (int j = 0; j < 4; j++) {

  305. 						int32_t integrateR[3];

  306. 						static int32_t integrateROld[3] = {0};

  307. 						integrateR[0] = ( (j==0) ? (combR[2]) : (0) ) + integrateROld[0];

  308. 						integrateR[1] = integrateR[0] + integrateROld[1];

  309. 						integrateR[2] = integrateR[1] + integrateROld[2];

  310. 						// integrateR[2] now holds j'th upsampled value

  311. 						*(dest+j*2+1) = integrateR[2] >> 4;

  312. 						integrateROld[0] = integrateR[0];

  313. 						integrateROld[1] = integrateR[1];

  314. 						integrateROld[2] = integrateR[2];

  315. 					}

  316. 

  317. 					// fill left channel with zeros:

  318. 					*(dest+0) = 0;

  319. 					*(dest+2) = 0;

  320. 					*(dest+4) = 0;

  321. 					*(dest+6) = 0;

  322. 					dest+=8;

  323. 					oldR = valR;

  324. 				}

  325. 			#else

  326. 				#error no interpolation method defined for oversampling.

  327. 			#endif //defined(AUDIO_PT8211_INTERPOLATION_LINEAR)

  328. 		#else

  329. 			memcpy_tointerleaveR(dest, blockR->data + offsetR);

  330. 			offsetR += AUDIO_BLOCK_SAMPLES / 2;

  331. 		#endif //defined(AUDIO_PT8211_OVERSAMPLING)

  332. 	} else {

  333. 		#if defined(AUDIO_PT8211_OVERSAMPLING)

  334. 			memset(dest,0,AUDIO_BLOCK_SAMPLES*8);

  335. 		#else

  336. 			memset(dest,0,AUDIO_BLOCK_SAMPLES*2);

  337. 		#endif

  338. 		return;

  339. 	}

  340. 

  341. 	if (offsetL < AUDIO_BLOCK_SAMPLES) {

  342. 		AudioOutputPT8211::block_left_offset = offsetL;

  343. 	} else {

  344. 		AudioOutputPT8211::block_left_offset = 0;

  345. 		AudioStream::release(blockL);

  346. 		AudioOutputPT8211::block_left_1st = AudioOutputPT8211::block_left_2nd;

  347. 		AudioOutputPT8211::block_left_2nd = NULL;

  348. 	}

  349. 	if (offsetR < AUDIO_BLOCK_SAMPLES) {

  350. 		AudioOutputPT8211::block_right_offset = offsetR;

  351. 	} else {

  352. 		AudioOutputPT8211::block_right_offset = 0;

  353. 		AudioStream::release(blockR);

  354. 		AudioOutputPT8211::block_right_1st = AudioOutputPT8211::block_right_2nd;

  355. 		AudioOutputPT8211::block_right_2nd = NULL;

  356. 	}

  357. }

  358. 

  359. 

  360. 

  361. void AudioOutputPT8211::update(void)

  362. {

  363. 

  364. 	audio_block_t *block;

  365. 	block = receiveReadOnly(0); // input 0 = left channel

  366. 	if (block) {

  367. 		__disable_irq();

  368. 		if (block_left_1st == NULL) {

  369. 			block_left_1st = block;

  370. 			block_left_offset = 0;

  371. 			__enable_irq();

  372. 		} else if (block_left_2nd == NULL) {

  373. 			block_left_2nd = block;

  374. 			__enable_irq();

  375. 		} else {

  376. 			audio_block_t *tmp = block_left_1st;

  377. 			block_left_1st = block_left_2nd;

  378. 			block_left_2nd = block;

  379. 			block_left_offset = 0;

  380. 			__enable_irq();

  381. 			release(tmp);

  382. 		}

  383. 	}

  384. 	block = receiveReadOnly(1); // input 1 = right channel

  385. 	if (block) {

  386. 		__disable_irq();

  387. 		if (block_right_1st == NULL) {

  388. 			block_right_1st = block;

  389. 			block_right_offset = 0;

  390. 			__enable_irq();

  391. 		} else if (block_right_2nd == NULL) {

  392. 			block_right_2nd = block;

  393. 			__enable_irq();

  394. 		} else {

  395. 			audio_block_t *tmp = block_right_1st;

  396. 			block_right_1st = block_right_2nd;

  397. 			block_right_2nd = block;

  398. 			block_right_offset = 0;

  399. 			__enable_irq();

  400. 			release(tmp);

  401. 		}

  402. 	}

  403. }

  404. 

  405. #if defined(KINETISK)

  406. // MCLK needs to be 48e6 / 1088 * 256 = 11.29411765 MHz -> 44.117647 kHz sample rate

  407. //

  408. #if F_CPU == 96000000 || F_CPU == 48000000 || F_CPU == 24000000

  409.   // PLL is at 96 MHz in these modes

  410.   #define MCLK_MULT 2

  411.   #define MCLK_DIV  17

  412. #elif F_CPU == 72000000

  413.   #define MCLK_MULT 8

  414.   #define MCLK_DIV  51

  415. #elif F_CPU == 120000000

  416.   #define MCLK_MULT 8

  417.   #define MCLK_DIV  85

  418. #elif F_CPU == 144000000

  419.   #define MCLK_MULT 4

  420.   #define MCLK_DIV  51

  421. #elif F_CPU == 168000000

  422.   #define MCLK_MULT 8

  423.   #define MCLK_DIV  119

  424. #elif F_CPU == 180000000

  425.   #define MCLK_MULT 16

  426.   #define MCLK_DIV  255

  427.   #define MCLK_SRC  0

  428. #elif F_CPU == 192000000

  429.   #define MCLK_MULT 1

  430.   #define MCLK_DIV  17

  431. #elif F_CPU == 216000000

  432.   #define MCLK_MULT 8

  433.   #define MCLK_DIV  153

  434.   #define MCLK_SRC  0

  435. #elif F_CPU == 240000000

  436.   #define MCLK_MULT 4

  437.   #define MCLK_DIV  85

  438. #elif F_CPU == 16000000

  439.   #define MCLK_MULT 12

  440.   #define MCLK_DIV  17

  441. #else

  442.   #error "This CPU Clock Speed is not supported by the Audio library";

  443. #endif

  444. 

  445. #ifndef MCLK_SRC

  446. #if F_CPU >= 20000000

  447.   #define MCLK_SRC  3  // the PLL

  448. #else

  449.   #define MCLK_SRC  0  // system clock

  450. #endif

  451. #endif

  452. #endif

  453. 

  454. void AudioOutputPT8211::config_i2s(void)

  455. {

  456. #if defined(KINETISK)

  457. 	SIM_SCGC6 |= SIM_SCGC6_I2S;

  458. 	SIM_SCGC7 |= SIM_SCGC7_DMA;

  459. 	SIM_SCGC6 |= SIM_SCGC6_DMAMUX;

  460. 

  461. 	// if transmitter is enabled, do nothing

  462. 	if (I2S0_TCSR & I2S_TCSR_TE) return;

  463. 

  464. 

  465. 	// enable MCLK output

  466. 	I2S0_MCR = I2S_MCR_MICS(MCLK_SRC) | I2S_MCR_MOE;

  467. 	while (I2S0_MCR & I2S_MCR_DUF) ;

  468. 	I2S0_MDR = I2S_MDR_FRACT((MCLK_MULT-1)) | I2S_MDR_DIVIDE((MCLK_DIV-1));

  469. 

  470. 	// configure transmitter

  471. 	I2S0_TMR = 0;

  472. 	I2S0_TCR1 = I2S_TCR1_TFW(1);  // watermark at half fifo size

  473. 	#if defined(AUDIO_PT8211_OVERSAMPLING)

  474. 		I2S0_TCR2 = I2S_TCR2_SYNC(0) | I2S_TCR2_BCP | I2S_TCR2_MSEL(1) | I2S_TCR2_BCD | I2S_TCR2_DIV(0);

  475. 	#else

  476. 		I2S0_TCR2 = I2S_TCR2_SYNC(0) | I2S_TCR2_BCP | I2S_TCR2_MSEL(1) | I2S_TCR2_BCD | I2S_TCR2_DIV(3);

  477. 	#endif

  478. 	I2S0_TCR3 = I2S_TCR3_TCE;

  479. //	I2S0_TCR4 = I2S_TCR4_FRSZ(1) | I2S_TCR4_SYWD(15) | I2S_TCR4_MF | I2S_TCR4_FSE | I2S_TCR4_FSP | I2S_TCR4_FSD; //TDA1543

  480. 	I2S0_TCR4 = I2S_TCR4_FRSZ(1) | I2S_TCR4_SYWD(15) | I2S_TCR4_MF /*| I2S_TCR4_FSE*/ | I2S_TCR4_FSP | I2S_TCR4_FSD; //PT8211

  481. 	I2S0_TCR5 = I2S_TCR5_WNW(15) | I2S_TCR5_W0W(15) | I2S_TCR5_FBT(15);

  482. 

  483. 	// configure pin mux for 3 clock signals

  484. 	CORE_PIN23_CONFIG = PORT_PCR_MUX(6); // pin 23, PTC2, I2S0_TX_FS (LRCLK)

  485. 	CORE_PIN9_CONFIG  = PORT_PCR_MUX(6); // pin  9, PTC3, I2S0_TX_BCLK

  486. 	//CORE_PIN11_CONFIG = PORT_PCR_MUX(6); // pin 11, PTC6, I2S0_MCLK

  487. 	

  488. #elif ( defined(__IMXRT1052__) || defined(__IMXRT1062__) )

  489. 

  490. 	CCM_CCGR5 |= CCM_CCGR5_SAI1(CCM_CCGR_ON);

  491. //PLL:

  492. 	int fs = AUDIO_SAMPLE_RATE_EXACT;

  493. 	// PLL between 27*24 = 648MHz und 54*24=1296MHz

  494. 	int n1 = 4; //SAI prescaler 4 => (n1*n2) = multiple of 4

  495. 	int n2 = 1 + (24000000 * 27) / (fs * 256 * n1);

  496. 

  497. 	double C = ((double)fs * 256 * n1 * n2) / 24000000;

  498. 	int c0 = C;

  499. 	int c2 = 10000;

  500. 	int c1 = C * c2 - (c0 * c2);

  501. 	set_audioClock(c0, c1, c2);

  502. 

  503. 	// clear SAI1_CLK register locations

  504. 	CCM_CSCMR1 = (CCM_CSCMR1 & ~(CCM_CSCMR1_SAI1_CLK_SEL_MASK))

  505. 		   | CCM_CSCMR1_SAI1_CLK_SEL(2); // &0x03 // (0,1,2): PLL3PFD0, PLL5, PLL4

  506. 	CCM_CS1CDR = (CCM_CS1CDR & ~(CCM_CS1CDR_SAI1_CLK_PRED_MASK | CCM_CS1CDR_SAI1_CLK_PODF_MASK))

  507. 		   | CCM_CS1CDR_SAI1_CLK_PRED(n1-1) // &0x07

  508. 		   | CCM_CS1CDR_SAI1_CLK_PODF(n2-1); // &0x3f

  509. 

  510. 	IOMUXC_GPR_GPR1 = (IOMUXC_GPR_GPR1 & ~(IOMUXC_GPR_GPR1_SAI1_MCLK1_SEL_MASK))

  511. 			| (IOMUXC_GPR_GPR1_SAI1_MCLK_DIR | IOMUXC_GPR_GPR1_SAI1_MCLK1_SEL(0));	//Select MCLK

  512. 

  513. 	if (I2S1_TCSR & I2S_TCSR_TE) return;

  514. 

  515. //	CORE_PIN23_CONFIG = 3;  //1:MCLK

  516. 	CORE_PIN21_CONFIG = 3;  //1:RX_BCLK

  517. 	CORE_PIN20_CONFIG = 3;  //1:RX_SYNC

  518. //	CORE_PIN6_CONFIG  = 3;  //1:TX_DATA0

  519. //	CORE_PIN7_CONFIG  = 3;  //1:RX_DATA0

  520. 

  521. 	int rsync = 0;

  522. 	int tsync = 1;

  523. 	#if defined(AUDIO_PT8211_OVERSAMPLING)

  524. 	int div = 0;

  525. 	#else

  526. 	int div = 3;

  527. 	#endif

  528. 	// configure transmitter

  529. 	I2S1_TMR = 0;

  530. 	I2S1_TCR1 = I2S_TCR1_RFW(0);

  531. 	I2S1_TCR2 = I2S_TCR2_SYNC(tsync) | I2S_TCR2_BCP | I2S_TCR2_MSEL(1) | I2S_TCR2_BCD | I2S_TCR2_DIV(div);

  532. 	I2S1_TCR3 = I2S_TCR3_TCE;

  533. //	I2S1_TCR4 = I2S_TCR4_FRSZ(1) | I2S_TCR4_SYWD(15) | I2S_TCR4_MF | I2S_TCR4_FSE | I2S_TCR4_FSP | I2S_TCR4_FSD; //TDA1543

  534. 	I2S1_TCR4 = I2S_TCR4_FRSZ(1) | I2S_TCR4_SYWD(15) | I2S_TCR4_MF /*| I2S_TCR4_FSE*/ | I2S_TCR4_FSP | I2S_TCR4_FSD; //PT8211

  535. 	I2S1_TCR5 = I2S_TCR5_WNW(15) | I2S_TCR5_W0W(15) | I2S_TCR5_FBT(15);

  536. 

  537. 	I2S1_RMR = 0;

  538. 	//I2S1_RCSR = (1<<25); //Reset

  539. 	I2S1_RCR1 = I2S_RCR1_RFW(0);

  540. 	I2S1_RCR2 = I2S_RCR2_SYNC(rsync) | I2S_RCR2_BCP | I2S_RCR2_MSEL(1) | I2S_TCR2_BCD | I2S_TCR2_DIV(div);

  541. 	I2S1_RCR3 = I2S_RCR3_RCE;

  542. //	I2S1_TCR4 = I2S_TCR4_FRSZ(1) | I2S_TCR4_SYWD(15) | I2S_TCR4_MF | I2S_TCR4_FSE | I2S_TCR4_FSP | I2S_TCR4_FSD; //TDA1543	

  543. 	I2S1_RCR4 = I2S_RCR4_FRSZ(1) | I2S_RCR4_SYWD(15) | I2S_RCR4_MF /*| I2S_RCR4_FSE*/ | I2S_RCR4_FSP | I2S_RCR4_FSD; //PT8211

  544. 	I2S1_RCR5 = I2S_RCR5_WNW(15) | I2S_RCR5_W0W(15) | I2S_RCR5_FBT(15);

  545. 

  546. #endif

  547. }