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OpenAudio_ArduinoLibrary/output_i2s_f32.cpp

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/*
* ***** output_i2s_f32.cpp *****
*
* 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>
#include <Audio.h> //to get access to Audio/utlity/imxrt_hw.h...do we really need this??? WEA 2020-10-31
/* Comment this out
//taken from Teensy Audio utility/imxrt_hw.h and imxrt_hw.cpp...
#if defined(__IMXRT1062__)
#ifndef imxr_hw_h_
#define imxr_hw_h_
#define IMXRT_CACHE_ENABLED 2 // 0=disabled, 1=WT, 2= WB
#include <Arduino.h>
#include <imxrt.h>
PROGMEM
void set_audioClock_tympan(int nfact, int32_t nmult, uint32_t ndiv, bool force = false) // sets PLL4
{
if (!force && (CCM_ANALOG_PLL_AUDIO & CCM_ANALOG_PLL_AUDIO_ENABLE)) return;
CCM_ANALOG_PLL_AUDIO = CCM_ANALOG_PLL_AUDIO_BYPASS | CCM_ANALOG_PLL_AUDIO_ENABLE
| CCM_ANALOG_PLL_AUDIO_POST_DIV_SELECT(2) // 2: 1/4; 1: 1/2; 0: 1/1
| CCM_ANALOG_PLL_AUDIO_DIV_SELECT(nfact);
CCM_ANALOG_PLL_AUDIO_NUM = nmult & CCM_ANALOG_PLL_AUDIO_NUM_MASK;
CCM_ANALOG_PLL_AUDIO_DENOM = ndiv & CCM_ANALOG_PLL_AUDIO_DENOM_MASK;
CCM_ANALOG_PLL_AUDIO &= ~CCM_ANALOG_PLL_AUDIO_POWERDOWN;//Switch on PLL
while (!(CCM_ANALOG_PLL_AUDIO & CCM_ANALOG_PLL_AUDIO_LOCK)) {}; //Wait for pll-lock
const int div_post_pll = 1; // other values: 2,4
CCM_ANALOG_MISC2 &= ~(CCM_ANALOG_MISC2_DIV_MSB | CCM_ANALOG_MISC2_DIV_LSB);
if(div_post_pll>1) CCM_ANALOG_MISC2 |= CCM_ANALOG_MISC2_DIV_LSB;
if(div_post_pll>3) CCM_ANALOG_MISC2 |= CCM_ANALOG_MISC2_DIV_MSB;
CCM_ANALOG_PLL_AUDIO &= ~CCM_ANALOG_PLL_AUDIO_BYPASS;//Disable Bypass
}
#endif
#else
//No IMXRT
#define IMXRT_CACHE_ENABLED 0
#endif
*/ //end of commented block
////////////
//
// Changing the sample rate based on changing the I2S bus freuqency (just for Teensy 3.x??)
//
//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...(again, just for Teensy 3.x??)
/*
%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_T3(const float freq_Hz) {
#if defined(KINETISK) //for Teensy 3.x only!
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);
}
}
#endif
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;
DMAChannel AudioOutputI2S_F32::dma(false);
DMAMEM __attribute__((aligned(32))) static uint32_t i2s_tx_buffer[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"
float AudioOutputI2S_F32::sample_rate_Hz = AUDIO_SAMPLE_RATE;
int AudioOutputI2S_F32::audio_block_samples = AUDIO_BLOCK_SAMPLES;
#if defined(__IMXRT1062__)
#include <utility/imxrt_hw.h> //from Teensy Audio library. For set_audioClock()
#endif
//#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 = false;
begin(transferUsing32bit);
}
void AudioOutputI2S_F32::begin(bool transferUsing32bit) {
dma.begin(true); // Allocate the DMA channel first
block_left_1st = NULL;
block_right_1st = NULL;
AudioOutputI2S_F32::config_i2s(transferUsing32bit, sample_rate_Hz);
#if defined(KINETISK)
CORE_PIN22_CONFIG = PORT_PCR_MUX(6); // pin 22, PTC1, I2S0_TXD0
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);//orig from Teensy Audio Library 2020-10-31
dma.TCD->SLAST = -I2S_BUFFER_TO_USE_BYTES;
dma.TCD->DADDR = (void *)((uint32_t)&I2S0_TDR0 + 2);
dma.TCD->DOFF = 0;
//dma.TCD->CITER_ELINKNO = sizeof(i2s_tx_buffer) / 2; //orig from Teensy Audio Library 2020-10-31
dma.TCD->CITER_ELINKNO = I2S_BUFFER_TO_USE_BYTES / 2;
dma.TCD->DLASTSGA = 0;
//dma.TCD->BITER_ELINKNO = sizeof(i2s_tx_buffer) / 2;//orig from Teensy Audio Library 2020-10-31
dma.TCD->BITER_ELINKNO = I2S_BUFFER_TO_USE_BYTES / 2;
dma.TCD->CSR = DMA_TCD_CSR_INTHALF | DMA_TCD_CSR_INTMAJOR;
dma.triggerAtHardwareEvent(DMAMUX_SOURCE_I2S0_TX);
dma.enable(); //newer location of this line in Teensy Audio library
I2S0_TCSR = I2S_TCSR_SR;
I2S0_TCSR = I2S_TCSR_TE | I2S_TCSR_BCE | I2S_TCSR_FRDE;
#elif defined(__IMXRT1062__)
CORE_PIN7_CONFIG = 3; //1:TX_DATA0
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);//orig from Teensy Audio Library 2020-10-31
dma.TCD->SLAST = -I2S_BUFFER_TO_USE_BYTES;
dma.TCD->DOFF = 0;
//dma.TCD->CITER_ELINKNO = sizeof(i2s_tx_buffer) / 2; //orig from Teensy Audio Library 2020-10-31
dma.TCD->CITER_ELINKNO = I2S_BUFFER_TO_USE_BYTES / 2;
dma.TCD->DLASTSGA = 0;
//dma.TCD->BITER_ELINKNO = sizeof(i2s_tx_buffer) / 2;//orig from Teensy Audio Library 2020-10-31
dma.TCD->BITER_ELINKNO = I2S_BUFFER_TO_USE_BYTES / 2;
dma.TCD->CSR = DMA_TCD_CSR_INTHALF | DMA_TCD_CSR_INTMAJOR;
dma.TCD->DADDR = (void *)((uint32_t)&I2S1_TDR0 + 2);
dma.triggerAtHardwareEvent(DMAMUX_SOURCE_SAI1_TX);
dma.enable(); //newer location of this line in Teensy Audio library
I2S1_RCSR |= I2S_RCSR_RE | I2S_RCSR_BCE;
I2S1_TCSR = I2S_TCSR_TE | I2S_TCSR_BCE | I2S_TCSR_FRDE;
#endif
update_responsibility = update_setup();
dma.attachInterrupt(AudioOutputI2S_F32::isr);
//dma.enable(); //original location of this line in older Tympan_Library
enabled = 1;
//AudioInputI2S_F32::begin_guts();
}
void AudioOutputI2S_F32::isr(void)
{
#if defined(KINETISK) || defined(__IMXRT1062__)
int16_t *dest;
audio_block_f32_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 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 Teensy Audio
dest = (int16_t *)&i2s_tx_buffer[audio_block_samples/2]; //this will be diff if we were to do 32-bit 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;
}
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);
float32_t *pL = blockL->data + offsetL;
float32_t *pR = blockR->data + offsetR;
for (int i=0; i < audio_block_samples/2; i++) {
*d++ = (int16_t) *pL++;
*d++ = (int16_t) *pR++; //interleave
//*d++ = 0;
//*d++ = 0;
}
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 * 2; i+=2) { *(d+i) = (int16_t) *pL++; } //interleave
offsetL += audio_block_samples / 2;
} else if (blockR) {
float32_t *pR = blockR->data + offsetR;
for (int i=0; i < audio_block_samples /2 * 2; i+=2) { *(d+i) = (int16_t) *pR++; } //interleave
offsetR += audio_block_samples / 2;
} else {
//memset(dest,0,AUDIO_BLOCK_SAMPLES * 2);
memset(dest,0,audio_block_samples * 2);
return;
}
arm_dcache_flush_delete(dest, sizeof(i2s_tx_buffer) / 2 );
//if (offsetL < AUDIO_BLOCK_SAMPLES) { //orig Teensy Audio
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) { //orig Teensy Audio
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;
}
#endif
}
/* 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;
}
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;
}
}
*/
#define F32_TO_I16_NORM_FACTOR (32767) //which is 2^15-1
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(-F32_TO_I16_NORM_FACTOR,min(F32_TO_I16_NORM_FACTOR,(*p_f32++) * F32_TO_I16_NORM_FACTOR)); }
}
#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.
//static int count = 0;
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);
scale_f32_to_i16(block_f32->data, block_f32_scaled->data, audio_block_samples);
//count++;
//if (count > 100) {
// Serial.print("AudioOutputI2S_F32::update() orig, scaled = ");
// Serial.print(block_f32->data[30]);
// Serial.print(", ");
// Serial.println(block_f32_scaled->data[30]);
// count=0;
//}
//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);
scale_f32_to_i16(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);
}
}
#if defined(KINETISK) || defined(KINETISL)
// 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 12
#define MCLK_DIV 17
#define MCLK_SRC 1
#elif F_CPU == 240000000
#define MCLK_MULT 2
#define MCLK_DIV 85
#define MCLK_SRC 0
#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
#endif
void AudioOutputI2S_F32::config_i2s(void) { config_i2s(false, AudioOutputI2S_F32::sample_rate_Hz); }
void AudioOutputI2S_F32::config_i2s(bool transferUsing32bit) { config_i2s(transferUsing32bit, AudioOutputI2S_F32::sample_rate_Hz); }
void AudioOutputI2S_F32::config_i2s(float fs_Hz) { config_i2s(false, fs_Hz); }
void AudioOutputI2S_F32::config_i2s(bool transferUsing32bit, float fs_Hz)
{
#if defined(KINETISK) || defined(KINETISL)
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;
// 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(1);
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);
I2S0_RCR2 = I2S_RCR2_SYNC(1) | I2S_TCR2_BCP | I2S_RCR2_MSEL(1)
| I2S_RCR2_BCD | I2S_RCR2_DIV(1);
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);
// 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
// change the I2S frequencies to make the requested sample rate
setI2SFreq_T3(fs_Hz); //for T3.x only!
#elif defined(__IMXRT1062__)
CCM_CCGR5 |= CCM_CCGR5_SAI1(CCM_CCGR_ON);
// if either transmitter or receiver is enabled, do nothing
if (I2S1_TCSR & I2S_TCSR_TE) return;
if (I2S1_RCSR & I2S_RCSR_RE) return;
//PLL:
//int fs = AUDIO_SAMPLE_RATE_EXACT; //original from Teensy Audio Library
int fs = fs_Hz;
// PLL between 27*24 = 648MHz und 54*24=1296MHz
int n1 = 4; //SAI prescaler 4 => (n1*n2) = multiple of 4
int n2 = 1 + (24000000 * 27) / (fs * 256 * n1);
double C = ((double)fs * 256 * n1 * n2) / 24000000;
int c0 = C;
int c2 = 10000;
int c1 = C * c2 - (c0 * c2);
set_audioClock(c0, c1, c2);
// clear SAI1_CLK register locations
CCM_CSCMR1 = (CCM_CSCMR1 & ~(CCM_CSCMR1_SAI1_CLK_SEL_MASK))
| CCM_CSCMR1_SAI1_CLK_SEL(2); // &0x03 // (0,1,2): PLL3PFD0, PLL5, PLL4
CCM_CS1CDR = (CCM_CS1CDR & ~(CCM_CS1CDR_SAI1_CLK_PRED_MASK | CCM_CS1CDR_SAI1_CLK_PODF_MASK))
| CCM_CS1CDR_SAI1_CLK_PRED(n1-1) // &0x07
| CCM_CS1CDR_SAI1_CLK_PODF(n2-1); // &0x3f
// Select MCLK
IOMUXC_GPR_GPR1 = (IOMUXC_GPR_GPR1
& ~(IOMUXC_GPR_GPR1_SAI1_MCLK1_SEL_MASK))
| (IOMUXC_GPR_GPR1_SAI1_MCLK_DIR | IOMUXC_GPR_GPR1_SAI1_MCLK1_SEL(0));
CORE_PIN23_CONFIG = 3; //1:MCLK
CORE_PIN21_CONFIG = 3; //1:RX_BCLK
CORE_PIN20_CONFIG = 3; //1:RX_SYNC
int rsync = 0;
int tsync = 1;
I2S1_TMR = 0;
//I2S1_TCSR = (1<<25); //Reset
I2S1_TCR1 = I2S_TCR1_RFW(1);
I2S1_TCR2 = I2S_TCR2_SYNC(tsync) | I2S_TCR2_BCP // sync=0; tx is async;
| (I2S_TCR2_BCD | I2S_TCR2_DIV((1)) | I2S_TCR2_MSEL(1));
I2S1_TCR3 = I2S_TCR3_TCE;
I2S1_TCR4 = I2S_TCR4_FRSZ((2-1)) | I2S_TCR4_SYWD((32-1)) | I2S_TCR4_MF
| I2S_TCR4_FSD | I2S_TCR4_FSE | I2S_TCR4_FSP;
I2S1_TCR5 = I2S_TCR5_WNW((32-1)) | I2S_TCR5_W0W((32-1)) | I2S_TCR5_FBT((32-1));
I2S1_RMR = 0;
//I2S1_RCSR = (1<<25); //Reset
I2S1_RCR1 = I2S_RCR1_RFW(1);
I2S1_RCR2 = I2S_RCR2_SYNC(rsync) | I2S_RCR2_BCP // sync=0; rx is async;
| (I2S_RCR2_BCD | I2S_RCR2_DIV((1)) | I2S_RCR2_MSEL(1));
I2S1_RCR3 = I2S_RCR3_RCE;
I2S1_RCR4 = I2S_RCR4_FRSZ((2-1)) | I2S_RCR4_SYWD((32-1)) | I2S_RCR4_MF
| I2S_RCR4_FSE | I2S_RCR4_FSP | I2S_RCR4_FSD;
I2S1_RCR5 = I2S_RCR5_WNW((32-1)) | I2S_RCR5_W0W((32-1)) | I2S_RCR5_FBT((32-1));
#endif
}
/******************************************************************/
// From Chip: The I2SSlave functionality has NOT been extended to allow for different block sizes or sample rates (2020-10-31)
void AudioOutputI2Sslave_F32::begin(void)
{
dma.begin(true); // Allocate the DMA channel first
//pinMode(2, OUTPUT);
block_left_1st = NULL;
block_right_1st = NULL;
AudioOutputI2Sslave_F32::config_i2s();
#if defined(KINETISK)
CORE_PIN22_CONFIG = PORT_PCR_MUX(6); // pin 22, PTC1, I2S0_TXD0
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 = (void *)((uint32_t)&I2S0_TDR0 + 2);
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;
dma.triggerAtHardwareEvent(DMAMUX_SOURCE_I2S0_TX);
dma.enable();
I2S0_TCSR = I2S_TCSR_SR;
I2S0_TCSR = I2S_TCSR_TE | I2S_TCSR_BCE | I2S_TCSR_FRDE;
#elif defined(__IMXRT1062__)
CORE_PIN7_CONFIG = 3; //1:TX_DATA0
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 = (void *)((uint32_t)&I2S1_TDR1 + 2);
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.triggerAtHardwareEvent(DMAMUX_SOURCE_SAI2_TX);
dma.TCD->DADDR = (void *)((uint32_t)&I2S1_TDR0 + 2);
dma.TCD->CSR = DMA_TCD_CSR_INTHALF | DMA_TCD_CSR_INTMAJOR;
dma.triggerAtHardwareEvent(DMAMUX_SOURCE_SAI1_TX);
dma.enable();
I2S1_RCSR |= I2S_RCSR_RE | I2S_RCSR_BCE;
I2S1_TCSR = I2S_TCSR_TE | I2S_TCSR_BCE | I2S_TCSR_FRDE;
#endif
update_responsibility = update_setup();
//dma.enable();
dma.attachInterrupt(AudioOutputI2S_F32::isr);
}
void AudioOutputI2Sslave_F32::config_i2s(void)
{
#if defined(KINETISK)
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(31) | I2S_TCR4_MF
| I2S_TCR4_FSE | I2S_TCR4_FSP;
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);
I2S0_RCR2 = I2S_RCR2_SYNC(1) | I2S_TCR2_BCP;
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);
// 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
#elif defined(__IMXRT1062__)
CCM_CCGR5 |= CCM_CCGR5_SAI1(CCM_CCGR_ON);
// if either transmitter or receiver is enabled, do nothing
if (I2S1_TCSR & I2S_TCSR_TE) return;
if (I2S1_RCSR & I2S_RCSR_RE) return;
// not using MCLK in slave mode - hope that's ok?
//CORE_PIN23_CONFIG = 3; // AD_B1_09 ALT3=SAI1_MCLK
CORE_PIN21_CONFIG = 3; // AD_B1_11 ALT3=SAI1_RX_BCLK
CORE_PIN20_CONFIG = 3; // AD_B1_10 ALT3=SAI1_RX_SYNC
IOMUXC_SAI1_RX_BCLK_SELECT_INPUT = 1; // 1=GPIO_AD_B1_11_ALT3, page 868
IOMUXC_SAI1_RX_SYNC_SELECT_INPUT = 1; // 1=GPIO_AD_B1_10_ALT3, page 872
// configure transmitter
I2S1_TMR = 0;
I2S1_TCR1 = I2S_TCR1_RFW(1); // watermark at half fifo size
I2S1_TCR2 = I2S_TCR2_SYNC(1) | I2S_TCR2_BCP;
I2S1_TCR3 = I2S_TCR3_TCE;
I2S1_TCR4 = I2S_TCR4_FRSZ(1) | I2S_TCR4_SYWD(31) | I2S_TCR4_MF
| I2S_TCR4_FSE | I2S_TCR4_FSP | I2S_RCR4_FSD;
I2S1_TCR5 = I2S_TCR5_WNW(31) | I2S_TCR5_W0W(31) | I2S_TCR5_FBT(31);
// configure receiver
I2S1_RMR = 0;
I2S1_RCR1 = I2S_RCR1_RFW(1);
I2S1_RCR2 = I2S_RCR2_SYNC(0) | I2S_TCR2_BCP;
I2S1_RCR3 = I2S_RCR3_RCE;
I2S1_RCR4 = I2S_RCR4_FRSZ(1) | I2S_RCR4_SYWD(31) | I2S_RCR4_MF
| I2S_RCR4_FSE | I2S_RCR4_FSP;
I2S1_RCR5 = I2S_RCR5_WNW(31) | I2S_RCR5_W0W(31) | I2S_RCR5_FBT(31);
#endif
}
////////////////////////////////////////////////////////////////////////
////////////////////////////////////////////////////////////////////////
#if 0
/*
* output_i2s_OA_F32.cpp (T3.6, T4.0 4Jan20 RSL)
* Revised for floating point input, but totally based
* on and forked from the Teensy Audio Library object output_i2s.cpp. Thus:
* 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.
*/
#include <Arduino.h>
#include "output_i2s_OA_F32.h"
#include "memcpy_audio.h"
#if defined(__IMXRT1062__)
#include "utility/imxrt_hw.h"
#endif
audio_block_t * AudioOutputI2S_OA_F32::block_left_1st = NULL;
audio_block_t * AudioOutputI2S_OA_F32::block_right_1st = NULL;
audio_block_t * AudioOutputI2S_OA_F32::block_left_2nd = NULL;
audio_block_t * AudioOutputI2S_OA_F32::block_right_2nd = NULL;
uint16_t AudioOutputI2S_OA_F32::block_left_offset = 0;
uint16_t AudioOutputI2S_OA_F32::block_right_offset = 0;
bool AudioOutputI2S_OA_F32::update_responsibility = false;
DMAChannel AudioOutputI2S_OA_F32::dma(false);
DMAMEM __attribute__((aligned(32))) static uint32_t i2s_tx_buffer[AUDIO_BLOCK_SAMPLES];
/*
static inline int32_t f32_to_i32(float32_t f) {
const float fullscale = 1LL << 31;
return max(-(fullscale - 1), min(fullscale - 1, f * fullscale));
} */
static void convert_F32toI16(float *in, int16_t *out, int len) {
//WEA Method. Should look at CMSIS arm_float_to_q15 instead:
// https://www.keil.com/pack/doc/CMSIS/DSP/html/group__float__to__x.html#ga215456e35a18db86882e1d3f0d24e1f2
const float MAX_INT = 32678.0;
for (int i = 0; i < len; i++) {
out[i] = (int16_t)(max(min( (in[i] * MAX_INT), MAX_INT), -MAX_INT));
}
}
// high-level explanation of how this I2S & DMA code works:
// https://forum.pjrc.com/threads/65229?p=263104&viewfull=1#post263104
void AudioOutputI2S_OA_F32::begin(void) {
dma.begin(true); // Allocate the DMA channel first
AudioOutputI2S_OA_F32::block_left_1st = NULL;
AudioOutputI2S_OA_F32::block_right_1st = NULL;
AudioOutputI2S_OA_F32::config_i2s(); // Get i2s clocks started
#if defined(KINETISK)
CORE_PIN22_CONFIG = PORT_PCR_MUX(6); // pin 22, PTC1, I2S0_TXD0
dma.TCD->SADDR = i2s_tx_buffer; // i2s_tx_buffer is buffer for I2S
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 = (void *)((uint32_t)&I2S0_TDR0 + 2);
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;
dma.triggerAtHardwareEvent(DMAMUX_SOURCE_I2S0_TX);
dma.enable();
I2S0_TCSR = I2S_TCSR_SR;
I2S0_TCSR = I2S_TCSR_TE | I2S_TCSR_BCE | I2S_TCSR_FRDE;
#elif defined(__IMXRT1062__)
CORE_PIN7_CONFIG = 3; //1:TX_DATA0
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->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;
dma.TCD->DADDR = (void *)((uint32_t)&I2S1_TDR0 + 2);
dma.triggerAtHardwareEvent(DMAMUX_SOURCE_SAI1_TX);
dma.enable();
I2S1_RCSR |= I2S_RCSR_RE | I2S_RCSR_BCE;
I2S1_TCSR = I2S_TCSR_TE | I2S_TCSR_BCE | I2S_TCSR_FRDE;
#endif
update_responsibility = AudioStream::update_setup();
dma.attachInterrupt(isr);
/* // change the I2S frequencies to make the requested sample rate
setI2SFreq(AudioOutputI2S_F32::sample_rate_Hz);
enabled = 1;
*/
}
// isr() every half update period, about 1450 uSec. Hardware i2s generated.
void AudioOutputI2S_OA_F32::isr(void) {
// T3.x or T4.x
#if defined(KINETISK) || defined(__IMXRT1062__)
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
dest = (int16_t *)&i2s_tx_buffer[AUDIO_BLOCK_SAMPLES/2];
if (AudioOutputI2S_OA_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_OA_F32::block_left_1st; // These are I16 audio blocks
blockR = AudioOutputI2S_OA_F32::block_right_1st;
offsetL = AudioOutputI2S_OA_F32::block_left_offset;
offsetR = AudioOutputI2S_OA_F32::block_right_offset;
if (blockL && blockR) { // This is the stereo case
memcpy_tointerleaveLR(dest, blockL->data + offsetL, blockR->data + offsetR);
offsetL += AUDIO_BLOCK_SAMPLES / 2;
offsetR += AUDIO_BLOCK_SAMPLES / 2;
} else if (blockL) {
memcpy_tointerleaveL(dest, blockL->data + offsetL);
offsetL += AUDIO_BLOCK_SAMPLES / 2;
} else if (blockR) {
memcpy_tointerleaveR(dest, blockR->data + offsetR);
offsetR += AUDIO_BLOCK_SAMPLES / 2;
} else {
// fill dest with zeros, AUDIO_BLOCK_SAMPLES *2 bytes (not enough?
// Maybe *4 bytes? This is not usually ever used.
memset(dest,0,AUDIO_BLOCK_SAMPLES * 2);
}
arm_dcache_flush_delete(dest, sizeof(i2s_tx_buffer) / 2 );
if (offsetL < AUDIO_BLOCK_SAMPLES) {
AudioOutputI2S_OA_F32::block_left_offset = offsetL;
} else {
AudioOutputI2S_OA_F32::block_left_offset = 0;
AudioStream::release(blockL); // I16 block
AudioOutputI2S_OA_F32::block_left_1st = AudioOutputI2S_OA_F32::block_left_2nd;
AudioOutputI2S_OA_F32::block_left_2nd = NULL;
}
if (offsetR < AUDIO_BLOCK_SAMPLES) {
AudioOutputI2S_OA_F32::block_right_offset = offsetR;
} else {
AudioOutputI2S_OA_F32::block_right_offset = 0;
AudioStream::release(blockR);
AudioOutputI2S_OA_F32::block_right_1st = AudioOutputI2S_OA_F32::block_right_2nd;
AudioOutputI2S_OA_F32::block_right_2nd = NULL;
}
#else
#if 0
//REMOVE <T3.x as not best for floating point library, switch to T3.x if using fP???
// Or just use T3,6 or T4.x and be happy
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_OA_F32::update_responsibility) AudioStream_F32::update_all(); // _F32
} 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_OA_F32::block_left_1st;
if (block) {
offset = AudioOutputI2S_OA_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_OA_F32::block_left_offset = offset;
} else {
AudioOutputI2S_OA_F32::block_left_offset = 0;
AudioStream::release(block);
AudioOutputI2S_OA_F32::block_left_1st = AudioOutputI2S_OA_F32::block_left_2nd;
AudioOutputI2S_OA_F32::block_left_2nd = NULL;
}
} else {
do {
*dest = 0;
dest += 2;
} while (dest < end);
}
dest -= AUDIO_BLOCK_SAMPLES - 1;
block = AudioOutputI2S_OA_F32::block_right_1st;
if (block) {
offset = AudioOutputI2S_OA_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_OA_F32::block_right_offset = offset;
} else {
AudioOutputI2S_OA_F32::block_right_offset = 0;
AudioStream::release(block);
AudioOutputI2S_OA_F32::block_right_1st = AudioOutputI2S_OA_F32::block_right_2nd;
AudioOutputI2S_OA_F32::block_right_2nd = NULL;
}
} else {
do {
*dest = 0;
dest += 2;
} while (dest < end);
}
#endif
// End of removed < T3.x support
#endif
}
void AudioOutputI2S_OA_F32::update(void) {
bool leftExist = false; // Kludge to get to R if L doesn't exist
// Get a float block to be used for getting L & R data from stream _F32
// if(millis() > 1200) { Serial.println(" ** update() ** "); } // <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
audio_block_f32_t *float_block;
float_block = AudioStream_F32::receiveReadOnly_f32(0); //float data block Left
// if(millis() > 1200) { Serial.print("float_block address="); Serial.println((uint32_t)float_block); } // <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
if (float_block==NULL)
return;
else
leftExist = true;
audio_block_t *block; // Int16 block
block = AudioStream::allocate(); // Integer 16 system
if (block==NULL) {
AudioStream_F32::release(float_block);
return;
}
if (leftExist) {
convert_F32toI16(float_block->data, block->data, float_block->length);
// if(millis() > 1200) { Serial.print(float_block->data[37], 6); Serial.print(" F32 L [37] I16 "); Serial.println(block->data[37]);} //<<<<<<<<<<<<<<<<
}
// Starting here left channels is regular 16-bit integer
if (leftExist && block != NULL) { // Redundant, block must exist or we wouldn't be here
__disable_irq();
if (AudioOutputI2S_OA_F32::block_left_1st == NULL) {
AudioOutputI2S_OA_F32::block_left_1st = block;
AudioOutputI2S_OA_F32::block_left_offset = 0;
__enable_irq();
} else if (block_left_2nd == NULL) {
AudioOutputI2S_OA_F32::block_left_2nd = block;
__enable_irq();
} else {
audio_block_t *tmp = AudioOutputI2S_OA_F32::block_left_1st; //Check for NULL??
AudioOutputI2S_OA_F32::block_left_1st = AudioOutputI2S_OA_F32::block_left_2nd;
AudioOutputI2S_OA_F32::block_left_2nd = block;
AudioOutputI2S_OA_F32::block_left_offset = 0;
__enable_irq();
AudioStream::release(tmp); // I16 block
}
}
AudioStream_F32::release(float_block);
// Same for the Right channel, if it exists
float_block = AudioStream_F32::receiveReadOnly_f32(1); //float data block Right
if (float_block != NULL) {
convert_F32toI16(float_block->data, block->data, float_block->length);
// if(millis() > 1200) { Serial.print(float_block->data[37], 6); Serial.print(" F32 R [37] I16 "); Serial.println(block->data[37]);} //<<<<<<<<<<<<
}
if (block && (float_block != NULL)) {
__disable_irq();
if (AudioOutputI2S_OA_F32::block_right_1st == NULL) {
AudioOutputI2S_OA_F32::block_right_1st = block;
AudioOutputI2S_OA_F32::block_right_offset = 0;
__enable_irq();
} else if (AudioOutputI2S_OA_F32::block_right_2nd == NULL) {
AudioOutputI2S_OA_F32::block_right_2nd = block;
__enable_irq();
} else {
audio_block_t *tmp = AudioOutputI2S_OA_F32::block_right_1st;
AudioOutputI2S_OA_F32::block_right_1st = AudioOutputI2S_OA_F32::block_right_2nd;
AudioOutputI2S_OA_F32::block_right_2nd = block;
AudioOutputI2S_OA_F32::block_right_offset = 0;
__enable_irq();
AudioStream::release(tmp);
}
}
/*
if(millis() > 1200) { Serial.print("Endupdate, fb="); Serial.println((uint32_t)float_block);
Serial.print(" b="); Serial.println((uint32_t)block); } // <<<<<<<<<<<<<<<<
*/
AudioStream_F32::release(float_block);
AudioStream::release(block);
} // End update()
#if defined(KINETISK) || defined(KINETISL)
// 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 12
#define MCLK_DIV 17
#define MCLK_SRC 1
#elif F_CPU == 240000000
#define MCLK_MULT 2
#define MCLK_DIV 85
#define MCLK_SRC 0
#elif F_CPU == 256000000
#define MCLK_MULT 12
#define MCLK_DIV 17
#define MCLK_SRC 1
#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
#endif
void AudioOutputI2S_OA_F32::config_i2s(void)
{
#if defined(KINETISK) || defined(KINETISL)
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;
// 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; // 32-bit Transmit Word Mask, 0=enabled
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(1);
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);
I2S0_RCR2 = I2S_RCR2_SYNC(1) | I2S_TCR2_BCP | I2S_RCR2_MSEL(1)
| I2S_RCR2_BCD | I2S_RCR2_DIV(1);
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);
// 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
#elif defined(__IMXRT1062__)
CCM_CCGR5 |= CCM_CCGR5_SAI1(CCM_CCGR_ON);
// if either transmitter or receiver is enabled, do nothing
if (I2S1_TCSR & I2S_TCSR_TE) return;
if (I2S1_RCSR & I2S_RCSR_RE) return;
//PLL:
int fs = AUDIO_SAMPLE_RATE_EXACT;
// PLL between 27*24 = 648MHz und 54*24=1296MHz
int n1 = 4; //SAI prescaler 4 => (n1*n2) = multiple of 4
int n2 = 1 + (24000000 * 27) / (fs * 256 * n1);
double C = ((double)fs * 256 * n1 * n2) / 24000000;
int c0 = C;
int c2 = 10000;
int c1 = C * c2 - (c0 * c2);
set_audioClock(c0, c1, c2);
// clear SAI1_CLK register locations
CCM_CSCMR1 = (CCM_CSCMR1 & ~(CCM_CSCMR1_SAI1_CLK_SEL_MASK))
| CCM_CSCMR1_SAI1_CLK_SEL(2); // &0x03 // (0,1,2): PLL3PFD0, PLL5, PLL4
CCM_CS1CDR = (CCM_CS1CDR & ~(CCM_CS1CDR_SAI1_CLK_PRED_MASK | CCM_CS1CDR_SAI1_CLK_PODF_MASK))
| CCM_CS1CDR_SAI1_CLK_PRED(n1-1) // &0x07
| CCM_CS1CDR_SAI1_CLK_PODF(n2-1); // &0x3f
// Select MCLK
IOMUXC_GPR_GPR1 = (IOMUXC_GPR_GPR1
& ~(IOMUXC_GPR_GPR1_SAI1_MCLK1_SEL_MASK))
| (IOMUXC_GPR_GPR1_SAI1_MCLK_DIR | IOMUXC_GPR_GPR1_SAI1_MCLK1_SEL(0));
CORE_PIN23_CONFIG = 3; //1:MCLK
CORE_PIN21_CONFIG = 3; //1:RX_BCLK
CORE_PIN20_CONFIG = 3; //1:RX_SYNC
int rsync = 0;
int tsync = 1;
I2S1_TMR = 0;
//I2S1_TCSR = (1<<25); //Reset
I2S1_TCR1 = I2S_TCR1_RFW(1);
I2S1_TCR2 = I2S_TCR2_SYNC(tsync) | I2S_TCR2_BCP // sync=0; tx is async;
| (I2S_TCR2_BCD | I2S_TCR2_DIV((1)) | I2S_TCR2_MSEL(1));
I2S1_TCR3 = I2S_TCR3_TCE;
I2S1_TCR4 = I2S_TCR4_FRSZ((2-1)) | I2S_TCR4_SYWD((32-1)) | I2S_TCR4_MF
| I2S_TCR4_FSD | I2S_TCR4_FSE | I2S_TCR4_FSP;
I2S1_TCR5 = I2S_TCR5_WNW((32-1)) | I2S_TCR5_W0W((32-1)) | I2S_TCR5_FBT((32-1));
I2S1_RMR = 0;
//I2S1_RCSR = (1<<25); //Reset
I2S1_RCR1 = I2S_RCR1_RFW(1);
I2S1_RCR2 = I2S_RCR2_SYNC(rsync) | I2S_RCR2_BCP // sync=0; rx is async;
| (I2S_RCR2_BCD | I2S_RCR2_DIV((1)) | I2S_RCR2_MSEL(1));
I2S1_RCR3 = I2S_RCR3_RCE;
I2S1_RCR4 = I2S_RCR4_FRSZ((2-1)) | I2S_RCR4_SYWD((32-1)) | I2S_RCR4_MF
| I2S_RCR4_FSE | I2S_RCR4_FSP | I2S_RCR4_FSD;
I2S1_RCR5 = I2S_RCR5_WNW((32-1)) | I2S_RCR5_W0W((32-1)) | I2S_RCR5_FBT((32-1));
#endif
}
/******************************************************************/
void AudioOutputI2Sslave_OA_F32::begin(void)
{
dma.begin(true); // Allocate the DMA channel first
block_left_1st = NULL;
block_right_1st = NULL;
AudioOutputI2Sslave_OA_F32::config_i2s();
#if defined(KINETISK)
CORE_PIN22_CONFIG = PORT_PCR_MUX(6); // pin 22, PTC1, I2S0_TXD0
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 = (void *)((uint32_t)&I2S0_TDR0 + 2);
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;
dma.triggerAtHardwareEvent(DMAMUX_SOURCE_I2S0_TX);
dma.enable();
I2S0_TCSR = I2S_TCSR_SR;
I2S0_TCSR = I2S_TCSR_TE | I2S_TCSR_BCE | I2S_TCSR_FRDE;
#elif defined(__IMXRT1062__)
CORE_PIN7_CONFIG = 3; //1:TX_DATA0
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->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->DADDR = (void *)((uint32_t)&I2S1_TDR0 + 2);
dma.TCD->CSR = DMA_TCD_CSR_INTHALF | DMA_TCD_CSR_INTMAJOR;
dma.triggerAtHardwareEvent(DMAMUX_SOURCE_SAI1_TX);
dma.enable();
I2S1_RCSR |= I2S_RCSR_RE | I2S_RCSR_BCE;
I2S1_TCSR = I2S_TCSR_TE | I2S_TCSR_BCE | I2S_TCSR_FRDE;
#endif
update_responsibility = AudioStream_F32::update_setup(); // <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
dma.attachInterrupt(isr);
}
void AudioOutputI2Sslave_OA_F32::config_i2s(void)
{
#if defined(KINETISK)
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(31) | I2S_TCR4_MF
| I2S_TCR4_FSE | I2S_TCR4_FSP;
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);
I2S0_RCR2 = I2S_RCR2_SYNC(1) | I2S_TCR2_BCP;
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);
// 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
#elif defined(__IMXRT1062__)
CCM_CCGR5 |= CCM_CCGR5_SAI1(CCM_CCGR_ON);
// if either transmitter or receiver is enabled, do nothing
if (I2S1_TCSR & I2S_TCSR_TE) return;
if (I2S1_RCSR & I2S_RCSR_RE) return;
// not using MCLK in slave mode - hope that's ok?
//CORE_PIN23_CONFIG = 3; // AD_B1_09 ALT3=SAI1_MCLK
CORE_PIN21_CONFIG = 3; // AD_B1_11 ALT3=SAI1_RX_BCLK
CORE_PIN20_CONFIG = 3; // AD_B1_10 ALT3=SAI1_RX_SYNC
IOMUXC_SAI1_RX_BCLK_SELECT_INPUT = 1; // 1=GPIO_AD_B1_11_ALT3, page 868
IOMUXC_SAI1_RX_SYNC_SELECT_INPUT = 1; // 1=GPIO_AD_B1_10_ALT3, page 872
// configure transmitter
I2S1_TMR = 0;
I2S1_TCR1 = I2S_TCR1_RFW(1); // watermark at half fifo size
I2S1_TCR2 = I2S_TCR2_SYNC(1) | I2S_TCR2_BCP;
I2S1_TCR3 = I2S_TCR3_TCE;
I2S1_TCR4 = I2S_TCR4_FRSZ(1) | I2S_TCR4_SYWD(31) | I2S_TCR4_MF
| I2S_TCR4_FSE | I2S_TCR4_FSP | I2S_RCR4_FSD;
I2S1_TCR5 = I2S_TCR5_WNW(31) | I2S_TCR5_W0W(31) | I2S_TCR5_FBT(31);
// configure receiver
I2S1_RMR = 0;
I2S1_RCR1 = I2S_RCR1_RFW(1);
I2S1_RCR2 = I2S_RCR2_SYNC(0) | I2S_TCR2_BCP;
I2S1_RCR3 = I2S_RCR3_RCE;
I2S1_RCR4 = I2S_RCR4_FRSZ(1) | I2S_RCR4_SYWD(31) | I2S_RCR4_MF
| I2S_RCR4_FSE | I2S_RCR4_FSP;
I2S1_RCR5 = I2S_RCR5_WNW(31) | I2S_RCR5_W0W(31) | I2S_RCR5_FBT(31);
#endif
}
#endif