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Add PT8211 w/ experimental 4x oversampling 

PT8211 with oversampling and linear interpolation by Frank Bösing.
3 Stage CIC interpolation by Ben Rheinland
dds
Benjamin 8 anos atrás
pai
6381c771c9
commit
0db2e23ba1
4 arquivos alterados com 534 adições e 0 exclusões
Visão dividida Mostrar estatísticas do Diff
  1. +1
    -0
      Audio.h
  2. +1
    -0
      keywords.txt
  3. +470
    -0
      output_pt8211.cpp
  4. +62
    -0
      output_pt8211.h

+ 1
- 0
Audio.h Ver arquivo

@@ -90,6 +90,7 @@
#include "output_i2s_quad.h"
#include "output_pwm.h"
#include "output_spdif.h"
#include "output_pt8211.h"
#include "play_memory.h"
#include "play_queue.h"
#include "play_sd_raw.h"

+ 1
- 0
keywords.txt Ver arquivo

@@ -8,6 +8,7 @@ AudioOutputI2S KEYWORD2
AudioOutputI2SQuad KEYWORD2
AudioOutputI2Sslave KEYWORD2
AudioOutputSPDIF KEYWORD2
AudioOutputPT8211 KEYWORD2
AudioOutputPWM KEYWORD2
AudioOutputUSB KEYWORD2
AudioControlSGTL5000 KEYWORD2

+ 470
- 0
output_pt8211.cpp Ver arquivo

@@ -0,0 +1,470 @@
/* 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.

#include "output_pt8211.h"
#include "memcpy_audio.h"

//uncomment to enable oversampling:
#define OVERSAMPLING
//uncomment ONE of these to define interpolation type for oversampling:
// #define INTERPOLATION_LINEAR
#define INTERPOLATION_CIC

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(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)
{
digitalWriteFast(LED_BUILTIN, HIGH);
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(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(OVERSAMPLING)
static int32_t oldL = 0;
static int32_t oldR = 0;
#endif
if (blockL && blockR) {
#if defined(OVERSAMPLING)
#if defined(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(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;
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];
}
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];
}

dest+=8;
oldL = valL;
oldR = valR;
}
#else
#error no interpolation method defined for oversampling.
#endif //defined(INTERPOLATION_LINEAR)
#else
memcpy_tointerleaveLR(dest, blockL->data + offsetL, blockR->data + offsetR);
offsetL += AUDIO_BLOCK_SAMPLES / 2;
offsetR += AUDIO_BLOCK_SAMPLES / 2;
#endif //defined(OVERSAMPLING)
} else if (blockL) {
#if defined(OVERSAMPLING)
#if defined(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(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(INTERPOLATION_LINEAR)
#else
memcpy_tointerleaveL(dest, blockL->data + offsetL);
offsetL += (AUDIO_BLOCK_SAMPLES / 2);
#endif //defined(OVERSAMPLING)
} else if (blockR) {
#if defined(OVERSAMPLING)
#if defined(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(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(INTERPOLATION_LINEAR)
#else
memcpy_tointerleaveR(dest, blockR->data + offsetR);
offsetR += AUDIO_BLOCK_SAMPLES / 2;
#endif //defined(OVERSAMPLING)
} else {
memset(dest,0,AUDIO_BLOCK_SAMPLES * 2);
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;
}
digitalWriteFast(LED_BUILTIN, LOW);
}




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(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;
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
}

+ 62
- 0
output_pt8211.h Ver arquivo

@@ -0,0 +1,62 @@
/* 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.

#ifndef output_pt8211_h_
#define output_pt8211_h_

#include "Arduino.h"
#include "AudioStream.h"
#include "DMAChannel.h"

class AudioOutputPT8211 : public AudioStream
{
public:
AudioOutputPT8211(void) : AudioStream(2, inputQueueArray) { begin(); }
virtual void update(void);
void begin(void);
//friend class AudioInputI2S;
protected:
//AudioOutputI2S(int dummy): AudioStream(2, inputQueueArray) {} // to be used only inside AudioOutputI2Sslave !!
static void config_i2s(void);
static audio_block_t *block_left_1st;
static audio_block_t *block_right_1st;
static bool update_responsibility;
static DMAChannel dma;
static void isr(void);
private:
static audio_block_t *block_left_2nd;
static audio_block_t *block_right_2nd;
static uint16_t block_left_offset;
static uint16_t block_right_offset;
audio_block_t *inputQueueArray[2];
};




#endif

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