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Removed old commented out code. No functional change

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pull/16/head
boblark 3 years ago
parent 2a7e5c62dc
commit 03c16e07d2
4 changed files with 34 additions and 718 deletions
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  1. 349
      input_i2s_f32.cpp
  2. 21
      input_i2s_f32.h
  3. 375
      output_i2s_f32.cpp
  4. 7
      output_i2s_f32.h

349
input_i2s_f32.cpp
Unescape View File

@ -1,6 +1,6 @@
/*
* input_i2s_f32.cpp
*
*
* Audio Library for Teensy 3.X
* Copyright (c) 2014, Paul Stoffregen, paul@pjrc.com
*
@ -26,20 +26,20 @@
* 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.
*/
// Updated OpenAudio F32 with this version from Chip Audette's Tympan Library Jan 2021 RSL
// Removed unused pieces. RSL 30 May 2022
#include <Arduino.h> //do we really need this? (Chip: 2020-10-31)
#include "input_i2s_f32.h"
#include "output_i2s_f32.h"
#include <arm_math.h>
//DMAMEM __attribute__((aligned(32)))
//DMAMEM __attribute__((aligned(32)))
static uint64_t i2s_rx_buffer[AUDIO_BLOCK_SAMPLES]; // Two 32-bit transfers per sample.
audio_block_f32_t * AudioInputI2S_F32::block_left_f32 = NULL;
audio_block_f32_t * AudioInputI2S_F32::block_right_f32 = NULL;
@ -69,7 +69,7 @@ void AudioInputI2S_F32::begin(bool transferUsing32bit) {
AudioOutputI2S_F32::sample_rate_Hz = sample_rate_Hz; //these were given in the AudioSettings in the contructor
AudioOutputI2S_F32::audio_block_samples = audio_block_samples;//these were given in the AudioSettings in the contructor
//block_left_1st = NULL;
//block_right_1st = NULL;
@ -100,7 +100,7 @@ void AudioInputI2S_F32::begin(bool transferUsing32bit) {
#elif defined(__IMXRT1062__)
CORE_PIN8_CONFIG = 3; //1:RX_DATA0
IOMUXC_SAI1_RX_DATA0_SELECT_INPUT = 2;
dma.TCD->SADDR = (void *)((uint32_t)&I2S1_RDR0 + 0);
dma.TCD->SOFF = 0;
dma.TCD->ATTR = DMA_TCD_ATTR_SSIZE(2) | DMA_TCD_ATTR_DSIZE(2);
@ -122,145 +122,9 @@ void AudioInputI2S_F32::begin(bool transferUsing32bit) {
update_responsibility = update_setup();
dma.enable();
dma.attachInterrupt(isr);
update_counter = 0;
}
/* void AudioInputI2S_F32::begin(bool transferUsing32bit) {
dma.begin(true); // Allocate the DMA channel first
AudioOutputI2S_F32::sample_rate_Hz = sample_rate_Hz; //these were given in the AudioSettings in the contructor
AudioOutputI2S_F32::audio_block_samples = audio_block_samples;//these were given in the AudioSettings in the contructor
//setup I2S parameters
AudioOutputI2S_F32::config_i2s(transferUsing32bit);
// TODO: should we set & clear the I2S_RCSR_SR bit here?
CORE_PIN13_CONFIG = PORT_PCR_MUX(4); // pin 13, PTC5, I2S0_RXD0
// setup DMA parameters
//if (transferUsing32bit) {
sub_begin_i32();
//} else {
// sub_begin_i16();
//}
// finish DMA setup
dma.triggerAtHardwareEvent(DMAMUX_SOURCE_I2S0_RX);
update_responsibility = update_setup();
dma.enable();
// finish I2S parameters
I2S0_RCSR |= I2S_RCSR_RE | I2S_RCSR_BCE | I2S_RCSR_FRDE | I2S_RCSR_FR;
I2S0_TCSR |= I2S_TCSR_TE | I2S_TCSR_BCE; // TX clock enable, because sync'd to TX
//if (transferUsing32bit) {
dma.attachInterrupt(isr_32);
//} else {
// dma.attachInterrupt(isr_16);
//}
update_counter = 0;
} */
/* void AudioInputI2S_F32::sub_begin_i16(void)
{
dma.TCD->SADDR = &I2S0_RDR0;
dma.TCD->SOFF = 0;
dma.TCD->ATTR = DMA_TCD_ATTR_SSIZE(1) | DMA_TCD_ATTR_DSIZE(1);
dma.TCD->NBYTES_MLNO = 2;
dma.TCD->SLAST = 0;
dma.TCD->DADDR = i2s_rx_buffer;
dma.TCD->DOFF = 2;
//dma.TCD->CITER_ELINKNO = sizeof(i2s_tx_buffer) / 2; //original
dma.TCD->CITER_ELINKNO = I2S_BUFFER_TO_USE_BYTES / 2;
//dma.TCD->DLASTSGA = -sizeof(i2s_rx_buffer); //original
dma.TCD->DLASTSGA = -I2S_BUFFER_TO_USE_BYTES;
//dma.TCD->BITER_ELINKNO = sizeof(i2s_rx_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 AudioInputI2S_F32::sub_begin_i32(void)
{
//let's assume that we'll transfer one sample (left or right) each call. So, it'll transfer 4 bytes (32-bits)
dma.TCD->SADDR = (void *)((uint32_t)&I2S0_RDR0);
dma.TCD->SOFF = 0; //do not increment the source memory pointer
dma.TCD->ATTR = DMA_TCD_ATTR_SSIZE(DMA_TCD_ATTR_SIZE_32BIT) | DMA_TCD_ATTR_DSIZE(DMA_TCD_ATTR_SIZE_32BIT);
dma.TCD->NBYTES_MLNO = 4; //one sample (32bits = 4bytes). should be 4 or 8? https://forum.pjrc.com/threads/42233-I2S-Input-Question
dma.TCD->SLAST = 0;
dma.TCD->DADDR = i2s_rx_buffer;
dma.TCD->DOFF = 4; //increment one sample (32bits = 4bytes) in the destination memory
//dma.TCD->CITER_ELINKNO = sizeof(i2s_tx_buffer) / 2; //original, 16-bit
//dma.TCD->CITER_ELINKNO = I2S_BUFFER_TO_USE_BYTES / 2; //revised WEA 16-bit
//dma.TCD->CITER_ELINKNO = sizeof(i2s_rx_buffer_32) / 4; //original, 32-bit
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? https://forum.pjrc.com/threads/42233-I2S-Input-Question
//dma.TCD->DLASTSGA = -sizeof(i2s_rx_buffer); //original, 16-bit
//dma.TCD->DLASTSGA = -I2S_BUFFER_TO_USE_BYTES;//revised WEA 16-bit
//dma.TCD->DLASTSGA = -sizeof(i2s_rx_buffer_32);//original, 32-bit
dma.TCD->DLASTSGA = -I2S_BUFFER_TO_USE_BYTES;//revised WEA 32-bit
//dma.TCD->BITER_ELINKNO = sizeof(i2s_rx_buffer) / 2; //original, 16-bit
//dma.TCD->BITER_ELINKNO = I2S_BUFFER_TO_USE_BYTES / 2;//revised WEA 16-bit
//dma.TCD->BITER_ELINKNO = sizeof(i2s_rx_buffer_32) / 4; //original, 32-bit
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? https://forum.pjrc.com/threads/42233-I2S-Input-Question
dma.TCD->CSR = DMA_TCD_CSR_INTHALF | DMA_TCD_CSR_INTMAJOR;
}; */
/* void AudioInputI2S_F32::isr_16(void)
{
uint32_t daddr, offset;
const int16_t *src, *end;
int16_t *dest_left, *dest_right;
audio_block_t *left, *right;
//digitalWriteFast(3, HIGH);
#if defined(KINETISK)
daddr = (uint32_t)(dma.TCD->DADDR);
#endif
dma.clearInterrupt();
//if (daddr < (uint32_t)i2s_rx_buffer + sizeof(i2s_rx_buffer) / 2) {
if (daddr < (uint32_t)i2s_rx_buffer + I2S_BUFFER_TO_USE_BYTES / 2) {
// DMA is receiving to the first half of the buffer
// need to remove data from the second half
//src = (int16_t *)&i2s_rx_buffer[AUDIO_BLOCK_SAMPLES/2]; //original
//end = (int16_t *)&i2s_rx_buffer[AUDIO_BLOCK_SAMPLES]; //original
src = (int16_t *)&i2s_rx_buffer[audio_block_samples/2];
end = (int16_t *)&i2s_rx_buffer[audio_block_samples];
if (AudioInputI2S_F32::update_responsibility) AudioStream_F32::update_all();
} else {
// DMA is receiving to the second half of the buffer
// need to remove data from the first half
src = (int16_t *)&i2s_rx_buffer[0];
//end = (int16_t *)&i2s_rx_buffer[AUDIO_BLOCK_SAMPLES/2]; //original
end = (int16_t *)&i2s_rx_buffer[audio_block_samples/2];
}
left = AudioInputI2S_F32::block_left;
right = AudioInputI2S_F32::block_right;
if (left != NULL && right != NULL) {
offset = AudioInputI2S_F32::block_offset;
//if (offset <= AUDIO_BLOCK_SAMPLES/2) { //original
if (offset <= ((uint32_t) audio_block_samples/2)) {
dest_left = &(left->data[offset]);
dest_right = &(right->data[offset]);
//AudioInputI2S_F32::block_offset = offset + AUDIO_BLOCK_SAMPLES/2; //original
AudioInputI2S_F32::block_offset = offset + audio_block_samples/2;
do {
//n = *src++;
// *dest_left++ = (int16_t)n;
// *dest_right++ = (int16_t)(n >> 16);
*dest_left++ = *src++;
*dest_right++ = *src++;
} while (src < end);
}
}
//digitalWriteFast(3, LOW);
} */
}
void AudioInputI2S_F32::isr(void)
{
@ -275,7 +139,6 @@ void AudioInputI2S_F32::isr(void)
daddr = (uint32_t)(dma.TCD->DADDR);
#endif
dma.clearInterrupt();
//Serial.println("isr");
//if (daddr < (uint32_t)i2s_rx_buffer + sizeof(i2s_rx_buffer) / 2) { //original Teensy Audio Library
if (daddr < (uint32_t)i2s_rx_buffer + I2S_BUFFER_TO_USE_BYTES / 2) {
@ -284,7 +147,7 @@ void AudioInputI2S_F32::isr(void)
//src = (int16_t *)&i2s_rx_buffer[AUDIO_BLOCK_SAMPLES/2]; //original Teensy Audio Library
//end = (int16_t *)&i2s_rx_buffer[AUDIO_BLOCK_SAMPLES]; //original Teensy Audio Library
src = (int32_t *)&i2s_rx_buffer[audio_block_samples/2];
end = (int32_t *)&i2s_rx_buffer[audio_block_samples];
end = (int32_t *)&i2s_rx_buffer[audio_block_samples];
update_counter++; //let's increment the counter here to ensure that we get every ISR resulting in audio
if (AudioInputI2S_F32::update_responsibility) AudioStream_F32::update_all();
} else {
@ -316,68 +179,7 @@ void AudioInputI2S_F32::isr(void)
}
}
/* void AudioInputI2S_F32::isr_32(void)
{
static bool flag_beenSuccessfullOnce = false;
uint32_t daddr, offset;
const int32_t *src_i32, *end_i32;
//int16_t *dest_left, *dest_right;
float32_t *dest_left_f32, *dest_right_f32;
audio_block_f32_t *left_f32, *right_f32;
daddr = (uint32_t)(dma.TCD->DADDR);
dma.clearInterrupt();
//if (daddr < (uint32_t)i2s_rx_buffer + sizeof(i2s_rx_buffer) / 2) {
if (daddr < (uint32_t)i2s_rx_buffer + I2S_BUFFER_TO_USE_BYTES / 2) {
// DMA is receiving to the first half of the buffer
// need to remove data from the second half
//src = (int32_t *)&i2s_rx_buffer_32[AUDIO_BLOCK_SAMPLES];
//end = (int32_t *)&i2s_rx_buffer_32[AUDIO_BLOCK_SAMPLES*2];
src_i32 = (int32_t *)&i2s_rx_buffer[audio_block_samples]; //WEA revised
end_i32 = (int32_t *)&i2s_rx_buffer[audio_block_samples*2]; //WEA revised
update_counter++; //let's increment the counter here to ensure that we get every ISR resulting in audio
if (AudioInputI2S_F32::update_responsibility) AudioStream_F32::update_all();
} else {
// DMA is receiving to the second half of the buffer
// need to remove data from the first half
//src = (int32_t *)&i2s_rx_buffer_32[0];
//end = (int32_t *)&i2s_rx_buffer_32[AUDIO_BLOCK_SAMPLES];
src_i32 = (int32_t *)&i2s_rx_buffer[0];
end_i32 = (int32_t *)&i2s_rx_buffer[audio_block_samples];
}
// OLD COMMENT: extract 16 but from 32 bit I2S buffer but shift to right first
// OLD COMMENT: there will be two buffers with each having "AUDIO_BLOCK_SAMPLES" samples
left_f32 = AudioInputI2S_F32::block_left_f32;
right_f32 = AudioInputI2S_F32::block_right_f32;
if ((left_f32 != NULL) && (right_f32 != NULL)) {
offset = AudioInputI2S_F32::block_offset;
//if (offset <= AUDIO_BLOCK_SAMPLES/2) { //original
if (offset <= ((uint32_t) audio_block_samples/2)) {
dest_left_f32 = &(left_f32->data[offset]);
dest_right_f32 = &(right_f32->data[offset]);
//AudioInputI2S_F32::block_offset = offset + AUDIO_BLOCK_SAMPLES/2; //original
AudioInputI2S_F32::block_offset = offset + audio_block_samples/2;
do {
//n = *src++;
// *dest_left++ = (int16_t)n;
// *dest_right++ = (int16_t)(n >> 16);
*dest_left_f32++ = (float32_t) *src_i32++;
*dest_right_f32++ = (float32_t) *src_i32++;
} while (src_i32 < end_i32);
}
flag_beenSuccessfullOnce = true;
} else {
if (flag_beenSuccessfullOnce) {
//but we were not successful this time
Serial.println("Input I2S: isr_32: WARNING!!! Null memory block.");
}
}
} */
#define I16_TO_F32_NORM_FACTOR (3.051850947599719e-05) //which is 1/32767
#define I16_TO_F32_NORM_FACTOR (3.051850947599719e-05) //which is 1/32767
void AudioInputI2S_F32::scale_i16_to_f32( float32_t *p_i16, float32_t *p_f32, int len) {
for (int i=0; i<len; i++) { *p_f32++ = ((*p_i16++) * I16_TO_F32_NORM_FACTOR); }
}
@ -390,114 +192,28 @@ void AudioInputI2S_F32::scale_i32_to_f32( float32_t *p_i32, float32_t *p_f32, in
for (int i=0; i<len; i++) { *p_f32++ = ((*p_i32++) * I32_TO_F32_NORM_FACTOR); }
}
/* void AudioInputI2S_F32::update_i16(void)
{
audio_block_t *new_left=NULL, *new_right=NULL, *out_left=NULL, *out_right=NULL;
// allocate 2 new blocks, but if one fails, allocate neither
new_left = AudioStream::allocate();
if (new_left != NULL) {
new_right = AudioStream::allocate();
if (new_right == NULL) {
flag_out_of_memory = 1;
AudioStream::release(new_left);
new_left = NULL;
}
} else {
flag_out_of_memory = 1;
}
__disable_irq();
//if (block_offset >= AUDIO_BLOCK_SAMPLES) { //original
if (block_offset >= audio_block_samples) {
// the DMA filled 2 blocks, so grab them and get the
// 2 new blocks to the DMA, as quickly as possible
out_left = block_left;
block_left = new_left;
out_right = block_right;
block_right = new_right;
block_offset = 0;
__enable_irq();
// then transmit the DMA's former blocks
// but, first, convert them to F32
audio_block_f32_t *out_left_f32=NULL, *out_right_f32=NULL;
out_left_f32 = AudioStream_F32::allocate_f32();
if (out_left_f32 != NULL) {
out_right_f32 = AudioStream_F32::allocate_f32();
if (out_right_f32 == NULL) {
flag_out_of_memory = 2;
AudioStream_F32::release(out_left_f32);
out_left_f32 = NULL;
}
} else {
flag_out_of_memory = 2;
}
if (out_left_f32 != NULL) {
//convert int16 to float 32
scale_i16_to_f32(out_left->data, out_left_f32->data, audio_block_samples);
scale_i16_to_f32(out_right->data, out_right_f32->data, audio_block_samples);
//prepare to transmit
update_counter++;
out_left_f32->id = update_counter;
out_right_f32->id = update_counter;
//transmit the f32 data!
AudioStream_F32::transmit(out_left_f32,0);
AudioStream_F32::transmit(out_right_f32,1);
AudioStream_F32::release(out_left_f32);
AudioStream_F32::release(out_right_f32);
}
AudioStream::release(out_left);
AudioStream::release(out_right);
//Serial.print(".");
} else if (new_left != NULL) {
// the DMA didn't fill blocks, but we allocated blocks
if (block_left == NULL) {
// the DMA doesn't have any blocks to fill, so
// give it the ones we just allocated
block_left = new_left;
block_right = new_right;
block_offset = 0;
__enable_irq();
} else {
// the DMA already has blocks, doesn't need these
__enable_irq();
AudioStream::release(new_left);
AudioStream::release(new_right);
}
} else {
// The DMA didn't fill blocks, and we could not allocate
// memory... the system is likely starving for memory!
// Sadly, there's nothing we can do.
__enable_irq();
}
}
*/
void AudioInputI2S_F32::update_1chan(int chan, audio_block_f32_t *&out_f32) {
if (!out_f32) return;
//scale the float values so that the maximum possible audio values span -1.0 to + 1.0
scale_i32_to_f32(out_f32->data, out_f32->data, audio_block_samples);
//scale_i16_to_f32(out_f32->data, out_f32->data, audio_block_samples);
//prepare to transmit by setting the update_counter (which helps tell if data is skipped or out-of-order)
out_f32->id = update_counter;
//transmit the f32 data!
AudioStream_F32::transmit(out_f32,chan);
//release the memory blocks
AudioStream_F32::release(out_f32);
}
void AudioInputI2S_F32::update(void)
{
static bool flag_beenSuccessfullOnce = false;
audio_block_f32_t *new_left=NULL, *new_right=NULL, *out_left=NULL, *out_right=NULL;
new_left = AudioStream_F32::allocate_f32();
new_right = AudioStream_F32::allocate_f32();
if ((!new_left) || (!new_right)) {
@ -510,7 +226,7 @@ void AudioInputI2S_F32::update(void)
} else {
flag_beenSuccessfullOnce = true;
}
__disable_irq();
if (block_offset >= audio_block_samples) {
// the DMA filled 2 blocks, so grab them and get the
@ -521,12 +237,12 @@ void AudioInputI2S_F32::update(void)
block_right_f32 = new_right;
block_offset = 0;
__enable_irq();
//update_counter++; //I chose to update it in the ISR instead.
update_1chan(0,out_left); //uses audio_block_samples and update_counter
update_1chan(1,out_right); //uses audio_block_samples and update_counter
} else if (new_left != NULL) {
// the DMA didn't fill blocks, but we allocated blocks
if (block_left_f32 == NULL) {
@ -583,35 +299,6 @@ void AudioInputI2Sslave_F32::begin(void)
I2S0_RCSR |= I2S_RCSR_RE | I2S_RCSR_BCE | I2S_RCSR_FRDE | I2S_RCSR_FR;
I2S0_TCSR |= I2S_TCSR_TE | I2S_TCSR_BCE; // TX clock enable, because sync'd to TX
dma.attachInterrupt(isr);
#endif
}
/*
void AudioInputI2Sslave::begin(void)
{
dma.begin(true); // Allocate the DMA channel first
//block_left_1st = NULL;
//block_right_1st = NULL;
AudioOutputI2Sslave::config_i2s();
CORE_PIN13_CONFIG = PORT_PCR_MUX(4); // pin 13, PTC5, I2S0_RXD0
#if defined(KINETISK)
dma.TCD->SADDR = &I2S0_RDR0;
dma.TCD->SOFF = 0;
dma.TCD->ATTR = DMA_TCD_ATTR_SSIZE(1) | DMA_TCD_ATTR_DSIZE(1);
dma.TCD->NBYTES_MLNO = 2;
dma.TCD->SLAST = 0;
dma.TCD->DADDR = i2s_rx_buffer;
dma.TCD->DOFF = 2;
dma.TCD->CITER_ELINKNO = sizeof(i2s_rx_buffer) / 2;
dma.TCD->DLASTSGA = -sizeof(i2s_rx_buffer);
dma.TCD->BITER_ELINKNO = sizeof(i2s_rx_buffer) / 2;
dma.TCD->CSR = DMA_TCD_CSR_INTHALF | DMA_TCD_CSR_INTMAJOR;
#endif
dma.triggerAtHardwareEvent(DMAMUX_SOURCE_I2S0_RX);
update_responsibility = update_setup();
dma.enable();
I2S0_RCSR |= I2S_RCSR_RE | I2S_RCSR_BCE | I2S_RCSR_FRDE | I2S_RCSR_FR;
I2S0_TCSR |= I2S_TCSR_TE | I2S_TCSR_BCE; // TX clock enable, because sync'd to TX
dma.attachInterrupt(isr);
}
*/

21
input_i2s_f32.h
Unescape View File

@ -1,6 +1,6 @@
/*
* ***** input_i2s_f32.h ******
*
*
* Audio Library for Teensy 3.X
* Copyright (c) 2014, Paul Stoffregen, paul@pjrc.com
*
@ -26,20 +26,21 @@
* 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.
*/
// Updated OpenAudio F32 with this version from Chip Audette's Tympan Library Jan 2021 RSL
// Removed unused pieces. RSL 30 May 2022
#ifndef _input_i2s_f32_h_
#define _input_i2s_f32_h_
#include <Arduino.h>
#include <arm_math.h>
#include <arm_math.h>
#include "AudioStream_F32.h"
#include "AudioStream.h" //Do we really need this?? (Chip, 2020-10-31)
// #include "AudioStream.h" // included AudioStream_F32.h > Audio.h > AudioStream.h
#include "DMAChannel.h"
class AudioInputI2S_F32 : public AudioStream_F32
@ -47,24 +48,24 @@ class AudioInputI2S_F32 : public AudioStream_F32
//GUI: inputs:0, outputs:2 //this line used for automatic generation of GUI nodes
public:
AudioInputI2S_F32(void) : AudioStream_F32(0, NULL) { begin(); } //uses default AUDIO_SAMPLE_RATE and BLOCK_SIZE_SAMPLES from AudioStream.h
AudioInputI2S_F32(const AudioSettings_F32 &settings) : AudioStream_F32(0, NULL) {
AudioInputI2S_F32(const AudioSettings_F32 &settings) : AudioStream_F32(0, NULL) {
sample_rate_Hz = settings.sample_rate_Hz;
audio_block_samples = settings.audio_block_samples;
begin();
begin();
}
virtual void update(void);
static void scale_i16_to_f32( float32_t *p_i16, float32_t *p_f32, int len) ;
static void scale_i24_to_f32( float32_t *p_i24, float32_t *p_f32, int len) ;
static void scale_i32_to_f32( float32_t *p_i32, float32_t *p_f32, int len);
void begin(void);
void begin(bool);
void sub_begin_i32(void);
//void sub_begin_i16(void);
// void sub_begin_i32(void); // These 2 are prototypes without functions RSL May 22
// void sub_begin_i16(void);
int get_isOutOfMemory(void) { return flag_out_of_memory; }
void clear_isOutOfMemory(void) { flag_out_of_memory = 0; }
//friend class AudioOutputI2S_F32;
protected:
protected:
AudioInputI2S_F32(int dummy): AudioStream_F32(0, NULL) {} // to be used only inside AudioInputI2Sslave !!
static bool update_responsibility;
static DMAChannel dma;

375
output_i2s_f32.cpp
Unescape View File

@ -32,88 +32,12 @@
* The F32 conversion is under the MIT License. Use at your own risk.
*/
// Updated OpenAudio F32 with this version from Chip Audette's Tympan Library Jan 2021 RSL
// Removed old commented out code. RSL 30 May 2022
#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);
@ -329,288 +253,6 @@ void AudioOutputI2S_F32::isr(void)
#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)); }
@ -667,16 +309,7 @@ void AudioOutputI2S_F32::update(void)
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
//now process the data blocks
__disable_irq();
if (block_left_1st == NULL) {
block_left_1st = block_f32_scaled;
@ -897,11 +530,11 @@ void AudioOutputI2S_F32::config_i2s(bool transferUsing32bit, float fs_Hz)
/******************************************************************/
// From Chip: The I2SSlave functionality has NOT been extended to allow for different block sizes or sample rates (2020-10-31)
// 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);

7
output_i2s_f32.h
Unescape View File

@ -32,6 +32,7 @@
* The F32 conversion is under the MIT License. Use at your own risk.
*/
// Updated OpenAudio F32 with this version from Chip Audette's Tympan Library Jan 2021 RSL
// Removed old commented out code. RSL 30 May 2022
#ifndef output_i2s_f32_h_
#define output_i2s_f32_h_
@ -39,7 +40,6 @@
#include <Arduino.h>
#include <arm_math.h>
#include "AudioStream_F32.h"
//include "AudioStream.h"
#include "DMAChannel.h"
class AudioOutputI2S_F32 : public AudioStream_F32
@ -64,10 +64,7 @@ public:
virtual void update(void);
void begin(void);
void begin(bool);
void sub_begin_i32(void);
void sub_begin_i16(void);
friend class AudioInputI2S_F32;
friend class AudioInputI2S_F32;
#if defined(__IMXRT1062__)
friend class AudioOutputI2SQuad_F32;
@ -89,8 +86,6 @@ protected:
static void config_i2s(bool);
static void config_i2s(float);
static void config_i2s(bool, float);
//static void config_i2s_i16(void,float);
//static void config_i2s_i32(void,float);
static audio_block_f32_t *block_left_1st;
static audio_block_f32_t *block_right_1st;
static bool update_responsibility;

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