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/* Audio Library for Teensy 3.X |
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* Copyright (c) 2016, Paul Stoffregen, paul@pjrc.com |
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* |
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* Development of this audio library was funded by PJRC.COM, LLC by sales of |
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* Teensy and Audio Adaptor boards. Please support PJRC's efforts to develop |
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* open source software by purchasing Teensy or other PJRC products. |
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* |
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* Permission is hereby granted, free of charge, to any person obtaining a copy |
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* of this software and associated documentation files (the "Software"), to deal |
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* in the Software without restriction, including without limitation the rights |
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* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell |
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* copies of the Software, and to permit persons to whom the Software is |
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* furnished to do so, subject to the following conditions: |
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* |
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* The above copyright notice, development funding notice, and this permission |
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* notice shall be included in all copies or substantial portions of the Software. |
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* |
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR |
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, |
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE |
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER |
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, |
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN |
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* THE SOFTWARE. |
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*/ |
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//Adapted to PT8211, Frank Bösing. |
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#include "output_pt8211.h" |
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#include "memcpy_audio.h" |
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//uncomment to enable oversampling: |
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#define OVERSAMPLING |
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//uncomment ONE of these to define interpolation type for oversampling: |
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// #define INTERPOLATION_LINEAR |
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#define INTERPOLATION_CIC |
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audio_block_t * AudioOutputPT8211::block_left_1st = NULL; |
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audio_block_t * AudioOutputPT8211::block_right_1st = NULL; |
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audio_block_t * AudioOutputPT8211::block_left_2nd = NULL; |
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audio_block_t * AudioOutputPT8211::block_right_2nd = NULL; |
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uint16_t AudioOutputPT8211::block_left_offset = 0; |
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uint16_t AudioOutputPT8211::block_right_offset = 0; |
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bool AudioOutputPT8211::update_responsibility = false; |
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#if defined(OVERSAMPLING) |
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DMAMEM static uint32_t i2s_tx_buffer[AUDIO_BLOCK_SAMPLES*4]; |
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#else |
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DMAMEM static uint32_t i2s_tx_buffer[AUDIO_BLOCK_SAMPLES]; |
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#endif |
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DMAChannel AudioOutputPT8211::dma(false); |
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void AudioOutputPT8211::begin(void) |
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{ |
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dma.begin(true); // Allocate the DMA channel first |
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block_left_1st = NULL; |
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block_right_1st = NULL; |
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// TODO: should we set & clear the I2S_TCSR_SR bit here? |
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config_i2s(); |
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CORE_PIN22_CONFIG = PORT_PCR_MUX(6); // pin 22, PTC1, I2S0_TXD0 |
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#if defined(KINETISK) |
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dma.TCD->SADDR = i2s_tx_buffer; |
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dma.TCD->SOFF = 2; |
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dma.TCD->ATTR = DMA_TCD_ATTR_SSIZE(1) | DMA_TCD_ATTR_DSIZE(1); |
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dma.TCD->NBYTES_MLNO = 2; |
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dma.TCD->SLAST = -sizeof(i2s_tx_buffer); |
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dma.TCD->DADDR = &I2S0_TDR0; |
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dma.TCD->DOFF = 0; |
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dma.TCD->CITER_ELINKNO = sizeof(i2s_tx_buffer) / 2; |
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dma.TCD->DLASTSGA = 0; |
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dma.TCD->BITER_ELINKNO = sizeof(i2s_tx_buffer) / 2; |
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dma.TCD->CSR = DMA_TCD_CSR_INTHALF | DMA_TCD_CSR_INTMAJOR; |
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#endif |
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dma.triggerAtHardwareEvent(DMAMUX_SOURCE_I2S0_TX); |
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update_responsibility = update_setup(); |
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dma.enable(); |
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I2S0_TCSR |= I2S_TCSR_TE | I2S_TCSR_BCE | I2S_TCSR_FRDE | I2S_TCSR_FR; |
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dma.attachInterrupt(isr); |
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} |
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void AudioOutputPT8211::isr(void) |
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{ |
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int16_t *dest; |
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audio_block_t *blockL, *blockR; |
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uint32_t saddr, offsetL, offsetR; |
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saddr = (uint32_t)(dma.TCD->SADDR); |
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dma.clearInterrupt(); |
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if (saddr < (uint32_t)i2s_tx_buffer + sizeof(i2s_tx_buffer) / 2) { |
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// DMA is transmitting the first half of the buffer |
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// so we must fill the second half |
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#if defined(OVERSAMPLING) |
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dest = (int16_t *)&i2s_tx_buffer[(AUDIO_BLOCK_SAMPLES/2)*4]; |
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#else |
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dest = (int16_t *)&i2s_tx_buffer[AUDIO_BLOCK_SAMPLES/2]; |
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#endif |
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if (AudioOutputPT8211::update_responsibility) AudioStream::update_all(); |
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} else { |
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// DMA is transmitting the second half of the buffer |
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// so we must fill the first half |
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dest = (int16_t *)i2s_tx_buffer; |
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} |
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blockL = AudioOutputPT8211::block_left_1st; |
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blockR = AudioOutputPT8211::block_right_1st; |
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offsetL = AudioOutputPT8211::block_left_offset; |
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offsetR = AudioOutputPT8211::block_right_offset; |
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#if defined(OVERSAMPLING) |
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static int32_t oldL = 0; |
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static int32_t oldR = 0; |
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#endif |
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if (blockL && blockR) { |
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#if defined(OVERSAMPLING) |
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#if defined(INTERPOLATION_LINEAR) |
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for (int i=0; i< AUDIO_BLOCK_SAMPLES / 2; i++, offsetL++, offsetR++) { |
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int32_t valL = blockL->data[offsetL]; |
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int32_t valR = blockR->data[offsetR]; |
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int32_t nL = (oldL+valL) >> 1; |
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int32_t nR = (oldR+valR) >> 1; |
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*(dest+0) = (oldL+nL) >> 1; |
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*(dest+1) = (oldR+nR) >> 1; |
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*(dest+2) = nL; |
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*(dest+3) = nR; |
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*(dest+4) = (nL+valL) >> 1; |
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*(dest+5) = (nR+valR) >> 1; |
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*(dest+6) = valL; |
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*(dest+7) = valR; |
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dest+=8; |
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oldL = valL; |
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oldR = valR; |
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} |
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#elif defined(INTERPOLATION_CIC) |
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for (int i=0; i< AUDIO_BLOCK_SAMPLES / 2; i++, offsetL++, offsetR++) { |
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int32_t valL = blockL->data[offsetL]; |
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int32_t valR = blockR->data[offsetR]; |
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int32_t combL[3] = {0}; |
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static int32_t combLOld[2] = {0}; |
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int32_t combR[3] = {0}; |
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static int32_t combROld[2] = {0}; |
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combL[0] = valL - oldL; |
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combL[1] = combL[0] - combLOld[0]; |
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combL[2] = combL[1] - combLOld[1]; |
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// combL[2] now holds input val |
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combLOld[0] = combL[0]; |
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combLOld[1] = combL[1]; |
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for (int j = 0; j < 4; j++) { |
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int32_t integrateL[3]; |
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static int32_t integrateLOld[3] = {0}; |
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integrateL[0] = ( (j==0) ? (combL[2]) : (0) ) + integrateLOld[0]; |
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integrateL[1] = integrateL[0] + integrateLOld[1]; |
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integrateL[2] = integrateL[1] + integrateLOld[2]; |
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// integrateL[2] now holds j'th upsampled value |
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*(dest+j*2) = integrateL[2] >> 4; |
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integrateLOld[0] = integrateL[0]; |
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integrateLOld[1] = integrateL[1]; |
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integrateLOld[2] = integrateL[2]; |
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} |
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combR[0] = valR - oldR; |
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combR[1] = combR[0] - combROld[0]; |
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combR[2] = combR[1] - combROld[1]; |
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// combR[2] now holds input val |
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combROld[0] = combR[0]; |
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combROld[1] = combR[1]; |
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for (int j = 0; j < 4; j++) { |
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int32_t integrateR[3]; |
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static int32_t integrateROld[3] = {0}; |
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integrateR[0] = ( (j==0) ? (combR[2]) : (0) ) + integrateROld[0]; |
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integrateR[1] = integrateR[0] + integrateROld[1]; |
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integrateR[2] = integrateR[1] + integrateROld[2]; |
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// integrateR[2] now holds j'th upsampled value |
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*(dest+j*2+1) = integrateR[2] >> 4; |
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integrateROld[0] = integrateR[0]; |
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integrateROld[1] = integrateR[1]; |
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integrateROld[2] = integrateR[2]; |
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} |
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dest+=8; |
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oldL = valL; |
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oldR = valR; |
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} |
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#else |
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#error no interpolation method defined for oversampling. |
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#endif //defined(INTERPOLATION_LINEAR) |
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#else |
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memcpy_tointerleaveLR(dest, blockL->data + offsetL, blockR->data + offsetR); |
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offsetL += AUDIO_BLOCK_SAMPLES / 2; |
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offsetR += AUDIO_BLOCK_SAMPLES / 2; |
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#endif //defined(OVERSAMPLING) |
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} else if (blockL) { |
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#if defined(OVERSAMPLING) |
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#if defined(INTERPOLATION_LINEAR) |
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for (int i=0; i< AUDIO_BLOCK_SAMPLES / 2; i++, offsetL++) { |
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int32_t val = blockL->data[offsetL]; |
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int32_t n = (oldL+val) >> 1; |
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*(dest+0) = (oldL+n) >> 1; |
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*(dest+1) = 0; |
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*(dest+2) = n; |
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*(dest+3) = 0; |
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*(dest+4) = (n+val) >> 1; |
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*(dest+5) = 0; |
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*(dest+6) = val; |
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*(dest+7) = 0; |
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dest+=8; |
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oldL = val; |
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} |
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#elif defined(INTERPOLATION_CIC) |
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for (int i=0; i< AUDIO_BLOCK_SAMPLES / 2; i++, offsetL++, offsetR++) { |
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int32_t valL = blockL->data[offsetL]; |
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int32_t combL[3] = {0}; |
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static int32_t combLOld[2] = {0}; |
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combL[0] = valL - oldL; |
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combL[1] = combL[0] - combLOld[0]; |
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combL[2] = combL[1] - combLOld[1]; |
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// combL[2] now holds input val |
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combLOld[0] = combL[0]; |
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combLOld[1] = combL[1]; |
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for (int j = 0; j < 4; j++) { |
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int32_t integrateL[3]; |
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static int32_t integrateLOld[3] = {0}; |
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integrateL[0] = ( (j==0) ? (combL[2]) : (0) ) + integrateLOld[0]; |
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integrateL[1] = integrateL[0] + integrateLOld[1]; |
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integrateL[2] = integrateL[1] + integrateLOld[2]; |
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// integrateL[2] now holds j'th upsampled value |
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*(dest+j*2) = integrateL[2] >> 4; |
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integrateLOld[0] = integrateL[0]; |
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integrateLOld[1] = integrateL[1]; |
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integrateLOld[2] = integrateL[2]; |
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} |
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// fill right channel with zeros: |
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*(dest+1) = 0; |
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*(dest+3) = 0; |
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*(dest+5) = 0; |
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*(dest+7) = 0; |
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dest+=8; |
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oldL = valL; |
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} |
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#else |
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#error no interpolation method defined for oversampling. |
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#endif //defined(INTERPOLATION_LINEAR) |
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#else |
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memcpy_tointerleaveL(dest, blockL->data + offsetL); |
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offsetL += (AUDIO_BLOCK_SAMPLES / 2); |
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#endif //defined(OVERSAMPLING) |
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} else if (blockR) { |
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#if defined(OVERSAMPLING) |
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#if defined(INTERPOLATION_LINEAR) |
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for (int i=0; i< AUDIO_BLOCK_SAMPLES / 2; i++, offsetR++) { |
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int32_t val = blockR->data[offsetR]; |
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int32_t n = (oldR+val) >> 1; |
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*(dest+0) = 0; |
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*(dest+1) = ((oldR+n) >> 1); |
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*(dest+2) = 0; |
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*(dest+3) = n; |
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*(dest+4) = 0; |
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*(dest+5) = ((n+val) >> 1); |
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*(dest+6) = 0; |
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*(dest+7) = val; |
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dest+=8; |
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oldR = val; |
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} |
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#elif defined(INTERPOLATION_CIC) |
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for (int i=0; i< AUDIO_BLOCK_SAMPLES / 2; i++, offsetL++, offsetR++) { |
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int32_t valR = blockR->data[offsetR]; |
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int32_t combR[3] = {0}; |
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static int32_t combROld[2] = {0}; |
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combR[0] = valR - oldR; |
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combR[1] = combR[0] - combROld[0]; |
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combR[2] = combR[1] - combROld[1]; |
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// combR[2] now holds input val |
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combROld[0] = combR[0]; |
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combROld[1] = combR[1]; |
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for (int j = 0; j < 4; j++) { |
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int32_t integrateR[3]; |
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static int32_t integrateROld[3] = {0}; |
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integrateR[0] = ( (j==0) ? (combR[2]) : (0) ) + integrateROld[0]; |
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integrateR[1] = integrateR[0] + integrateROld[1]; |
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integrateR[2] = integrateR[1] + integrateROld[2]; |
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// integrateR[2] now holds j'th upsampled value |
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*(dest+j*2+1) = integrateR[2] >> 4; |
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integrateROld[0] = integrateR[0]; |
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integrateROld[1] = integrateR[1]; |
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integrateROld[2] = integrateR[2]; |
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} |
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// fill left channel with zeros: |
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*(dest+0) = 0; |
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*(dest+2) = 0; |
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*(dest+4) = 0; |
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*(dest+6) = 0; |
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dest+=8; |
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oldR = valR; |
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} |
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#else |
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#error no interpolation method defined for oversampling. |
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#endif //defined(INTERPOLATION_LINEAR) |
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#else |
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memcpy_tointerleaveR(dest, blockR->data + offsetR); |
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offsetR += AUDIO_BLOCK_SAMPLES / 2; |
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#endif //defined(OVERSAMPLING) |
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} else { |
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memset(dest,0,AUDIO_BLOCK_SAMPLES * 2); |
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return; |
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} |
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if (offsetL < AUDIO_BLOCK_SAMPLES) { |
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AudioOutputPT8211::block_left_offset = offsetL; |
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} else { |
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AudioOutputPT8211::block_left_offset = 0; |
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AudioStream::release(blockL); |
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AudioOutputPT8211::block_left_1st = AudioOutputPT8211::block_left_2nd; |
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AudioOutputPT8211::block_left_2nd = NULL; |
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} |
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if (offsetR < AUDIO_BLOCK_SAMPLES) { |
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AudioOutputPT8211::block_right_offset = offsetR; |
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} else { |
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AudioOutputPT8211::block_right_offset = 0; |
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AudioStream::release(blockR); |
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AudioOutputPT8211::block_right_1st = AudioOutputPT8211::block_right_2nd; |
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AudioOutputPT8211::block_right_2nd = NULL; |
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} |
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} |
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void AudioOutputPT8211::update(void) |
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{ |
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audio_block_t *block; |
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block = receiveReadOnly(0); // input 0 = left channel |
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if (block) { |
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__disable_irq(); |
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if (block_left_1st == NULL) { |
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block_left_1st = block; |
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block_left_offset = 0; |
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__enable_irq(); |
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} else if (block_left_2nd == NULL) { |
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block_left_2nd = block; |
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__enable_irq(); |
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} else { |
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audio_block_t *tmp = block_left_1st; |
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block_left_1st = block_left_2nd; |
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block_left_2nd = block; |
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block_left_offset = 0; |
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__enable_irq(); |
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release(tmp); |
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} |
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} |
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block = receiveReadOnly(1); // input 1 = right channel |
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if (block) { |
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__disable_irq(); |
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if (block_right_1st == NULL) { |
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block_right_1st = block; |
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block_right_offset = 0; |
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__enable_irq(); |
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} else if (block_right_2nd == NULL) { |
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block_right_2nd = block; |
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__enable_irq(); |
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} else { |
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audio_block_t *tmp = block_right_1st; |
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block_right_1st = block_right_2nd; |
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block_right_2nd = block; |
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block_right_offset = 0; |
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__enable_irq(); |
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release(tmp); |
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} |
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} |
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} |
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// MCLK needs to be 48e6 / 1088 * 256 = 11.29411765 MHz -> 44.117647 kHz sample rate |
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// |
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#if F_CPU == 96000000 || F_CPU == 48000000 || F_CPU == 24000000 |
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// PLL is at 96 MHz in these modes |
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#define MCLK_MULT 2 |
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#define MCLK_DIV 17 |
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#elif F_CPU == 72000000 |
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#define MCLK_MULT 8 |
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#define MCLK_DIV 51 |
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#elif F_CPU == 120000000 |
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#define MCLK_MULT 8 |
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#define MCLK_DIV 85 |
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#elif F_CPU == 144000000 |
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#define MCLK_MULT 4 |
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#define MCLK_DIV 51 |
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#elif F_CPU == 168000000 |
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#define MCLK_MULT 8 |
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#define MCLK_DIV 119 |
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#elif F_CPU == 180000000 |
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#define MCLK_MULT 16 |
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#define MCLK_DIV 255 |
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#define MCLK_SRC 0 |
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#elif F_CPU == 192000000 |
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#define MCLK_MULT 1 |
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#define MCLK_DIV 17 |
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#elif F_CPU == 216000000 |
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#define MCLK_MULT 8 |
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#define MCLK_DIV 153 |
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#define MCLK_SRC 0 |
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#elif F_CPU == 240000000 |
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#define MCLK_MULT 4 |
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#define MCLK_DIV 85 |
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#elif F_CPU == 16000000 |
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#define MCLK_MULT 12 |
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#define MCLK_DIV 17 |
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#else |
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#error "This CPU Clock Speed is not supported by the Audio library"; |
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#endif |
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#ifndef MCLK_SRC |
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#if F_CPU >= 20000000 |
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#define MCLK_SRC 3 // the PLL |
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#else |
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#define MCLK_SRC 0 // system clock |
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#endif |
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#endif |
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void AudioOutputPT8211::config_i2s(void) |
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{ |
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SIM_SCGC6 |= SIM_SCGC6_I2S; |
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SIM_SCGC7 |= SIM_SCGC7_DMA; |
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SIM_SCGC6 |= SIM_SCGC6_DMAMUX; |
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// if transmitter is enabled, do nothing |
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if (I2S0_TCSR & I2S_TCSR_TE) return; |
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// enable MCLK output |
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I2S0_MCR = I2S_MCR_MICS(MCLK_SRC) | I2S_MCR_MOE; |
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while (I2S0_MCR & I2S_MCR_DUF) ; |
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I2S0_MDR = I2S_MDR_FRACT((MCLK_MULT-1)) | I2S_MDR_DIVIDE((MCLK_DIV-1)); |
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// configure transmitter |
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I2S0_TMR = 0; |
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I2S0_TCR1 = I2S_TCR1_TFW(1); // watermark at half fifo size |
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#if defined(OVERSAMPLING) |
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I2S0_TCR2 = I2S_TCR2_SYNC(0) | I2S_TCR2_BCP | I2S_TCR2_MSEL(1) | I2S_TCR2_BCD | I2S_TCR2_DIV(0); |
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#else |
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I2S0_TCR2 = I2S_TCR2_SYNC(0) | I2S_TCR2_BCP | I2S_TCR2_MSEL(1) | I2S_TCR2_BCD | I2S_TCR2_DIV(3); |
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#endif |
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I2S0_TCR3 = I2S_TCR3_TCE; |
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// I2S0_TCR4 = I2S_TCR4_FRSZ(1) | I2S_TCR4_SYWD(15) | I2S_TCR4_MF | I2S_TCR4_FSE | I2S_TCR4_FSP | I2S_TCR4_FSD; |
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|
I2S0_TCR4 = I2S_TCR4_FRSZ(1) | I2S_TCR4_SYWD(15) | I2S_TCR4_MF /*| I2S_TCR4_FSE*/ | I2S_TCR4_FSP | I2S_TCR4_FSD; //PT8211 |
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I2S0_TCR5 = I2S_TCR5_WNW(15) | I2S_TCR5_W0W(15) | I2S_TCR5_FBT(15); |
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// configure pin mux for 3 clock signals |
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|
CORE_PIN23_CONFIG = PORT_PCR_MUX(6); // pin 23, PTC2, I2S0_TX_FS (LRCLK) |
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CORE_PIN9_CONFIG = PORT_PCR_MUX(6); // pin 9, PTC3, I2S0_TX_BCLK |
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#if 0 |
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CORE_PIN11_CONFIG = PORT_PCR_MUX(6); // pin 11, PTC6, I2S0_MCLK |
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#endif |
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} |
Paul Stoffregen
8 years ago
