/* Audio Library for Teensy 3.X * Copyright (c) 2016, 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. */ //Adapted to PT8211, Frank Bösing, Ben-Rheinland #include #include "output_pt8211.h" #include "memcpy_audio.h" audio_block_t * AudioOutputPT8211::block_left_1st = NULL; audio_block_t * AudioOutputPT8211::block_right_1st = NULL; audio_block_t * AudioOutputPT8211::block_left_2nd = NULL; audio_block_t * AudioOutputPT8211::block_right_2nd = NULL; uint16_t AudioOutputPT8211::block_left_offset = 0; uint16_t AudioOutputPT8211::block_right_offset = 0; bool AudioOutputPT8211::update_responsibility = false; #if defined(AUDIO_PT8211_OVERSAMPLING) DMAMEM static uint32_t i2s_tx_buffer[AUDIO_BLOCK_SAMPLES*4]; #else DMAMEM static uint32_t i2s_tx_buffer[AUDIO_BLOCK_SAMPLES]; #endif DMAChannel AudioOutputPT8211::dma(false); void AudioOutputPT8211::begin(void) { 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(); 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 AudioOutputPT8211::isr(void) { 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) { // DMA is transmitting the first half of the buffer // so we must fill the second half #if defined(AUDIO_PT8211_OVERSAMPLING) dest = (int16_t *)&i2s_tx_buffer[(AUDIO_BLOCK_SAMPLES/2)*4]; #else dest = (int16_t *)&i2s_tx_buffer[AUDIO_BLOCK_SAMPLES/2]; #endif if (AudioOutputPT8211::update_responsibility) AudioStream::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 = AudioOutputPT8211::block_left_1st; blockR = AudioOutputPT8211::block_right_1st; offsetL = AudioOutputPT8211::block_left_offset; offsetR = AudioOutputPT8211::block_right_offset; #if defined(AUDIO_PT8211_OVERSAMPLING) static int32_t oldL = 0; static int32_t oldR = 0; #endif if (blockL && blockR) { #if defined(AUDIO_PT8211_OVERSAMPLING) #if defined(AUDIO_PT8211_INTERPOLATION_LINEAR) for (int i=0; i< AUDIO_BLOCK_SAMPLES / 2; i++, offsetL++, offsetR++) { int32_t valL = blockL->data[offsetL]; int32_t valR = blockR->data[offsetR]; int32_t nL = (oldL+valL) >> 1; int32_t nR = (oldR+valR) >> 1; *(dest+0) = (oldL+nL) >> 1; *(dest+1) = (oldR+nR) >> 1; *(dest+2) = nL; *(dest+3) = nR; *(dest+4) = (nL+valL) >> 1; *(dest+5) = (nR+valR) >> 1; *(dest+6) = valL; *(dest+7) = valR; dest+=8; oldL = valL; oldR = valR; } #elif defined(AUDIO_PT8211_INTERPOLATION_CIC) for (int i=0; i< AUDIO_BLOCK_SAMPLES / 2; i++, offsetL++, offsetR++) { int32_t valL = blockL->data[offsetL]; int32_t valR = blockR->data[offsetR]; int32_t combL[3] = {0}; static int32_t combLOld[2] = {0}; int32_t combR[3] = {0}; static int32_t combROld[2] = {0}; combL[0] = valL - oldL; combR[0] = valR - oldR; combL[1] = combL[0] - combLOld[0]; combR[1] = combR[0] - combROld[0]; combL[2] = combL[1] - combLOld[1]; combR[2] = combR[1] - combROld[1]; // combL[2] now holds input val // combR[2] now holds input val oldL = valL; oldR = valR; combLOld[0] = combL[0]; combROld[0] = combR[0]; combLOld[1] = combL[1]; combROld[1] = combR[1]; for (int j = 0; j < 4; j++) { int32_t integrateL[3]; int32_t integrateR[3]; static int32_t integrateLOld[3] = {0}; static int32_t integrateROld[3] = {0}; integrateL[0] = ( (j==0) ? (combL[2]) : (0) ) + integrateLOld[0]; integrateR[0] = ( (j==0) ? (combR[2]) : (0) ) + integrateROld[0]; integrateL[1] = integrateL[0] + integrateLOld[1]; integrateR[1] = integrateR[0] + integrateROld[1]; integrateL[2] = integrateL[1] + integrateLOld[2]; integrateR[2] = integrateR[1] + integrateROld[2]; // integrateL[2] now holds j'th upsampled value // integrateR[2] now holds j'th upsampled value *(dest+j*2) = integrateL[2] >> 4; *(dest+j*2+1) = integrateR[2] >> 4; integrateLOld[0] = integrateL[0]; integrateROld[0] = integrateR[0]; integrateLOld[1] = integrateL[1]; integrateROld[1] = integrateR[1]; integrateLOld[2] = integrateL[2]; integrateROld[2] = integrateR[2]; } dest+=8; } #else #error no interpolation method defined for oversampling. #endif //defined(AUDIO_PT8211_INTERPOLATION_LINEAR) #else memcpy_tointerleaveLR(dest, blockL->data + offsetL, blockR->data + offsetR); offsetL += AUDIO_BLOCK_SAMPLES / 2; offsetR += AUDIO_BLOCK_SAMPLES / 2; #endif //defined(AUDIO_PT8211_OVERSAMPLING) } else if (blockL) { #if defined(AUDIO_PT8211_OVERSAMPLING) #if defined(AUDIO_PT8211_INTERPOLATION_LINEAR) for (int i=0; i< AUDIO_BLOCK_SAMPLES / 2; i++, offsetL++) { int32_t val = blockL->data[offsetL]; int32_t n = (oldL+val) >> 1; *(dest+0) = (oldL+n) >> 1; *(dest+1) = 0; *(dest+2) = n; *(dest+3) = 0; *(dest+4) = (n+val) >> 1; *(dest+5) = 0; *(dest+6) = val; *(dest+7) = 0; dest+=8; oldL = val; } #elif defined(AUDIO_PT8211_INTERPOLATION_CIC) for (int i=0; i< AUDIO_BLOCK_SAMPLES / 2; i++, offsetL++, offsetR++) { int32_t valL = blockL->data[offsetL]; int32_t combL[3] = {0}; static int32_t combLOld[2] = {0}; combL[0] = valL - oldL; combL[1] = combL[0] - combLOld[0]; combL[2] = combL[1] - combLOld[1]; // combL[2] now holds input val combLOld[0] = combL[0]; combLOld[1] = combL[1]; for (int j = 0; j < 4; j++) { int32_t integrateL[3]; static int32_t integrateLOld[3] = {0}; integrateL[0] = ( (j==0) ? (combL[2]) : (0) ) + integrateLOld[0]; integrateL[1] = integrateL[0] + integrateLOld[1]; integrateL[2] = integrateL[1] + integrateLOld[2]; // integrateL[2] now holds j'th upsampled value *(dest+j*2) = integrateL[2] >> 4; integrateLOld[0] = integrateL[0]; integrateLOld[1] = integrateL[1]; integrateLOld[2] = integrateL[2]; } // fill right channel with zeros: *(dest+1) = 0; *(dest+3) = 0; *(dest+5) = 0; *(dest+7) = 0; dest+=8; oldL = valL; } #else #error no interpolation method defined for oversampling. #endif //defined(AUDIO_PT8211_INTERPOLATION_LINEAR) #else memcpy_tointerleaveL(dest, blockL->data + offsetL); offsetL += (AUDIO_BLOCK_SAMPLES / 2); #endif //defined(AUDIO_PT8211_OVERSAMPLING) } else if (blockR) { #if defined(AUDIO_PT8211_OVERSAMPLING) #if defined(AUDIO_PT8211_INTERPOLATION_LINEAR) for (int i=0; i< AUDIO_BLOCK_SAMPLES / 2; i++, offsetR++) { int32_t val = blockR->data[offsetR]; int32_t n = (oldR+val) >> 1; *(dest+0) = 0; *(dest+1) = ((oldR+n) >> 1); *(dest+2) = 0; *(dest+3) = n; *(dest+4) = 0; *(dest+5) = ((n+val) >> 1); *(dest+6) = 0; *(dest+7) = val; dest+=8; oldR = val; } #elif defined(AUDIO_PT8211_INTERPOLATION_CIC) for (int i=0; i< AUDIO_BLOCK_SAMPLES / 2; i++, offsetL++, offsetR++) { int32_t valR = blockR->data[offsetR]; int32_t combR[3] = {0}; static int32_t combROld[2] = {0}; combR[0] = valR - oldR; combR[1] = combR[0] - combROld[0]; combR[2] = combR[1] - combROld[1]; // combR[2] now holds input val combROld[0] = combR[0]; combROld[1] = combR[1]; for (int j = 0; j < 4; j++) { int32_t integrateR[3]; static int32_t integrateROld[3] = {0}; integrateR[0] = ( (j==0) ? (combR[2]) : (0) ) + integrateROld[0]; integrateR[1] = integrateR[0] + integrateROld[1]; integrateR[2] = integrateR[1] + integrateROld[2]; // integrateR[2] now holds j'th upsampled value *(dest+j*2+1) = integrateR[2] >> 4; integrateROld[0] = integrateR[0]; integrateROld[1] = integrateR[1]; integrateROld[2] = integrateR[2]; } // fill left channel with zeros: *(dest+0) = 0; *(dest+2) = 0; *(dest+4) = 0; *(dest+6) = 0; dest+=8; oldR = valR; } #else #error no interpolation method defined for oversampling. #endif //defined(AUDIO_PT8211_INTERPOLATION_LINEAR) #else memcpy_tointerleaveR(dest, blockR->data + offsetR); offsetR += AUDIO_BLOCK_SAMPLES / 2; #endif //defined(AUDIO_PT8211_OVERSAMPLING) } else { #if defined(AUDIO_PT8211_OVERSAMPLING) memset(dest,0,AUDIO_BLOCK_SAMPLES*8); #else memset(dest,0,AUDIO_BLOCK_SAMPLES*2); #endif return; } if (offsetL < AUDIO_BLOCK_SAMPLES) { AudioOutputPT8211::block_left_offset = offsetL; } else { AudioOutputPT8211::block_left_offset = 0; AudioStream::release(blockL); AudioOutputPT8211::block_left_1st = AudioOutputPT8211::block_left_2nd; AudioOutputPT8211::block_left_2nd = NULL; } if (offsetR < AUDIO_BLOCK_SAMPLES) { AudioOutputPT8211::block_right_offset = offsetR; } else { AudioOutputPT8211::block_right_offset = 0; AudioStream::release(blockR); AudioOutputPT8211::block_right_1st = AudioOutputPT8211::block_right_2nd; AudioOutputPT8211::block_right_2nd = NULL; } } void AudioOutputPT8211::update(void) { audio_block_t *block; block = receiveReadOnly(0); // input 0 = left channel if (block) { __disable_irq(); if (block_left_1st == NULL) { block_left_1st = block; block_left_offset = 0; __enable_irq(); } else if (block_left_2nd == NULL) { block_left_2nd = block; __enable_irq(); } else { audio_block_t *tmp = block_left_1st; block_left_1st = block_left_2nd; block_left_2nd = block; block_left_offset = 0; __enable_irq(); release(tmp); } } block = receiveReadOnly(1); // input 1 = right channel if (block) { __disable_irq(); if (block_right_1st == NULL) { block_right_1st = block; block_right_offset = 0; __enable_irq(); } else if (block_right_2nd == NULL) { block_right_2nd = block; __enable_irq(); } else { audio_block_t *tmp = block_right_1st; block_right_1st = block_right_2nd; block_right_2nd = block; block_right_offset = 0; __enable_irq(); release(tmp); } } } // 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 AudioOutputPT8211::config_i2s(void) { SIM_SCGC6 |= SIM_SCGC6_I2S; SIM_SCGC7 |= SIM_SCGC7_DMA; SIM_SCGC6 |= SIM_SCGC6_DMAMUX; // if transmitter is enabled, do nothing if (I2S0_TCSR & I2S_TCSR_TE) return; // 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 #if defined(AUDIO_PT8211_OVERSAMPLING) I2S0_TCR2 = I2S_TCR2_SYNC(0) | I2S_TCR2_BCP | I2S_TCR2_MSEL(1) | I2S_TCR2_BCD | I2S_TCR2_DIV(0); #else I2S0_TCR2 = I2S_TCR2_SYNC(0) | I2S_TCR2_BCP | I2S_TCR2_MSEL(1) | I2S_TCR2_BCD | I2S_TCR2_DIV(3); #endif 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; //TDA1543 I2S0_TCR4 = I2S_TCR4_FRSZ(1) | I2S_TCR4_SYWD(15) | I2S_TCR4_MF /*| I2S_TCR4_FSE*/ | I2S_TCR4_FSP | I2S_TCR4_FSD; //PT8211 I2S0_TCR5 = I2S_TCR5_WNW(15) | I2S_TCR5_W0W(15) | I2S_TCR5_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 #if 0 CORE_PIN11_CONFIG = PORT_PCR_MUX(6); // pin 11, PTC6, I2S0_MCLK #endif }