visus
/
teensy-audio
Watch 1
Star 0
Fork 0
Code Issues 0 Pull Requests 0 Releases 0 Wiki Activity
Browse Source

Merge pull request #194 from Ben-Rheinland/master 

Improve Oversampling for PT8211
dds
Paul Stoffregen 8 years ago
parent
6186a6a676 246a7f6977
commit
54974b6c7a
2 changed files with 47 additions and 49 deletions
Split View Show Diff Stats
  1. +36
    -48
      output_pt8211.cpp
  2. +11
    -1
      output_pt8211.h

+ 36
- 48
output_pt8211.cpp View File

@@ -29,12 +29,6 @@
#include "output_pt8211.h"
#include "memcpy_audio.h"

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

audio_block_t * AudioOutputPT8211::block_left_1st = NULL;
audio_block_t * AudioOutputPT8211::block_right_1st = NULL;
audio_block_t * AudioOutputPT8211::block_left_2nd = NULL;
@@ -42,7 +36,7 @@ audio_block_t * AudioOutputPT8211::block_right_2nd = NULL;
uint16_t AudioOutputPT8211::block_left_offset = 0;
uint16_t AudioOutputPT8211::block_right_offset = 0;
bool AudioOutputPT8211::update_responsibility = false;
#if defined(OVERSAMPLING)
#if defined(AUDIO_PT8211_OVERSAMPLING)
DMAMEM static uint32_t i2s_tx_buffer[AUDIO_BLOCK_SAMPLES*4];
#else
DMAMEM static uint32_t i2s_tx_buffer[AUDIO_BLOCK_SAMPLES];
@@ -81,7 +75,6 @@ void AudioOutputPT8211::begin(void)
dma.attachInterrupt(isr);
}


void AudioOutputPT8211::isr(void)
{
int16_t *dest;
@@ -93,7 +86,7 @@ void AudioOutputPT8211::isr(void)
if (saddr < (uint32_t)i2s_tx_buffer + sizeof(i2s_tx_buffer) / 2) {
// DMA is transmitting the first half of the buffer
// so we must fill the second half
#if defined(OVERSAMPLING)
#if defined(AUDIO_PT8211_OVERSAMPLING)
dest = (int16_t *)&i2s_tx_buffer[(AUDIO_BLOCK_SAMPLES/2)*4];
#else
dest = (int16_t *)&i2s_tx_buffer[AUDIO_BLOCK_SAMPLES/2];
@@ -110,13 +103,13 @@ void AudioOutputPT8211::isr(void)
offsetL = AudioOutputPT8211::block_left_offset;
offsetR = AudioOutputPT8211::block_right_offset;
#if defined(OVERSAMPLING)
#if defined(AUDIO_PT8211_OVERSAMPLING)
static int32_t oldL = 0;
static int32_t oldR = 0;
#endif
if (blockL && blockR) {
#if defined(OVERSAMPLING)
#if defined(INTERPOLATION_LINEAR)
#if defined(AUDIO_PT8211_OVERSAMPLING)
#if defined(AUDIO_PT8211_INTERPOLATION_LINEAR)
for (int i=0; i< AUDIO_BLOCK_SAMPLES / 2; i++, offsetL++, offsetR++) {
int32_t valL = blockL->data[offsetL];
int32_t valR = blockR->data[offsetR];
@@ -134,7 +127,7 @@ void AudioOutputPT8211::isr(void)
oldL = valL;
oldR = valR;
}
#elif defined(INTERPOLATION_CIC)
#elif defined(AUDIO_PT8211_INTERPOLATION_CIC)
for (int i=0; i< AUDIO_BLOCK_SAMPLES / 2; i++, offsetL++, offsetR++) {
int32_t valL = blockL->data[offsetL];
int32_t valR = blockR->data[offsetR];
@@ -145,59 +138,55 @@ void AudioOutputPT8211::isr(void)
static int32_t combROld[2] = {0};
combL[0] = valL - oldL;
combL[1] = combL[0] - combLOld[0];
combL[2] = combL[1] - combLOld[1];
// combL[2] now holds input val
combLOld[0] = combL[0];
combLOld[1] = combL[1];
for (int j = 0; j < 4; j++) {
int32_t integrateL[3];
static int32_t integrateLOld[3] = {0};
integrateL[0] = ( (j==0) ? (combL[2]) : (0) ) + integrateLOld[0];
integrateL[1] = integrateL[0] + integrateLOld[1];
integrateL[2] = integrateL[1] + integrateLOld[2];
// integrateL[2] now holds j'th upsampled value
*(dest+j*2) = integrateL[2] >> 4;
integrateLOld[0] = integrateL[0];
integrateLOld[1] = integrateL[1];
integrateLOld[2] = integrateL[2];
}
combR[0] = valR - oldR;
combL[1] = combL[0] - combLOld[0];
combR[1] = combR[0] - combROld[0];
combL[2] = combL[1] - combLOld[1];
combR[2] = combR[1] - combROld[1];
// combL[2] now holds input val
// combR[2] now holds input val
oldL = valL;
oldR = valR;
combLOld[0] = combL[0];
combROld[0] = combR[0];
combLOld[1] = combL[1];
combROld[1] = combR[1];
for (int j = 0; j < 4; j++) {
int32_t integrateL[3];
int32_t integrateR[3];
static int32_t integrateLOld[3] = {0};
static int32_t integrateROld[3] = {0};
integrateL[0] = ( (j==0) ? (combL[2]) : (0) ) + integrateLOld[0];
integrateR[0] = ( (j==0) ? (combR[2]) : (0) ) + integrateROld[0];
integrateL[1] = integrateL[0] + integrateLOld[1];
integrateR[1] = integrateR[0] + integrateROld[1];
integrateL[2] = integrateL[1] + integrateLOld[2];
integrateR[2] = integrateR[1] + integrateROld[2];
// integrateL[2] now holds j'th upsampled value
// integrateR[2] now holds j'th upsampled value
*(dest+j*2) = integrateL[2] >> 4;
*(dest+j*2+1) = integrateR[2] >> 4;
integrateLOld[0] = integrateL[0];
integrateROld[0] = integrateR[0];
integrateLOld[1] = integrateL[1];
integrateROld[1] = integrateR[1];
integrateLOld[2] = integrateL[2];
integrateROld[2] = integrateR[2];
}

dest+=8;
oldL = valL;
oldR = valR;
}
#else
#error no interpolation method defined for oversampling.
#endif //defined(INTERPOLATION_LINEAR)
#endif //defined(AUDIO_PT8211_INTERPOLATION_LINEAR)
#else
memcpy_tointerleaveLR(dest, blockL->data + offsetL, blockR->data + offsetR);
offsetL += AUDIO_BLOCK_SAMPLES / 2;
offsetR += AUDIO_BLOCK_SAMPLES / 2;
#endif //defined(OVERSAMPLING)
#endif //defined(AUDIO_PT8211_OVERSAMPLING)
} else if (blockL) {
#if defined(OVERSAMPLING)
#if defined(INTERPOLATION_LINEAR)
#if defined(AUDIO_PT8211_OVERSAMPLING)
#if defined(AUDIO_PT8211_INTERPOLATION_LINEAR)
for (int i=0; i< AUDIO_BLOCK_SAMPLES / 2; i++, offsetL++) {
int32_t val = blockL->data[offsetL];
int32_t n = (oldL+val) >> 1;
@@ -212,7 +201,7 @@ void AudioOutputPT8211::isr(void)
dest+=8;
oldL = val;
}
#elif defined(INTERPOLATION_CIC)
#elif defined(AUDIO_PT8211_INTERPOLATION_CIC)
for (int i=0; i< AUDIO_BLOCK_SAMPLES / 2; i++, offsetL++, offsetR++) {
int32_t valL = blockL->data[offsetL];

@@ -249,14 +238,14 @@ void AudioOutputPT8211::isr(void)
}
#else
#error no interpolation method defined for oversampling.
#endif //defined(INTERPOLATION_LINEAR)
#endif //defined(AUDIO_PT8211_INTERPOLATION_LINEAR)
#else
memcpy_tointerleaveL(dest, blockL->data + offsetL);
offsetL += (AUDIO_BLOCK_SAMPLES / 2);
#endif //defined(OVERSAMPLING)
#endif //defined(AUDIO_PT8211_OVERSAMPLING)
} else if (blockR) {
#if defined(OVERSAMPLING)
#if defined(INTERPOLATION_LINEAR)
#if defined(AUDIO_PT8211_OVERSAMPLING)
#if defined(AUDIO_PT8211_INTERPOLATION_LINEAR)
for (int i=0; i< AUDIO_BLOCK_SAMPLES / 2; i++, offsetR++) {
int32_t val = blockR->data[offsetR];
int32_t n = (oldR+val) >> 1;
@@ -271,7 +260,7 @@ void AudioOutputPT8211::isr(void)
dest+=8;
oldR = val;
}
#elif defined(INTERPOLATION_CIC)
#elif defined(AUDIO_PT8211_INTERPOLATION_CIC)
for (int i=0; i< AUDIO_BLOCK_SAMPLES / 2; i++, offsetL++, offsetR++) {
int32_t valR = blockR->data[offsetR];

@@ -308,11 +297,11 @@ void AudioOutputPT8211::isr(void)
}
#else
#error no interpolation method defined for oversampling.
#endif //defined(INTERPOLATION_LINEAR)
#endif //defined(AUDIO_PT8211_INTERPOLATION_LINEAR)
#else
memcpy_tointerleaveR(dest, blockR->data + offsetR);
offsetR += AUDIO_BLOCK_SAMPLES / 2;
#endif //defined(OVERSAMPLING)
#endif //defined(AUDIO_PT8211_OVERSAMPLING)
} else {
memset(dest,0,AUDIO_BLOCK_SAMPLES * 2);
return;
@@ -338,7 +327,6 @@ void AudioOutputPT8211::isr(void)




void AudioOutputPT8211::update(void)
{

@@ -449,7 +437,7 @@ void AudioOutputPT8211::config_i2s(void)
// configure transmitter
I2S0_TMR = 0;
I2S0_TCR1 = I2S_TCR1_TFW(1); // watermark at half fifo size
#if defined(OVERSAMPLING)
#if defined(AUDIO_PT8211_OVERSAMPLING)
I2S0_TCR2 = I2S_TCR2_SYNC(0) | I2S_TCR2_BCP | I2S_TCR2_MSEL(1) | I2S_TCR2_BCD | I2S_TCR2_DIV(0);
#else
I2S0_TCR2 = I2S_TCR2_SYNC(0) | I2S_TCR2_BCP | I2S_TCR2_MSEL(1) | I2S_TCR2_BCD | I2S_TCR2_DIV(3);

+ 11
- 1
output_pt8211.h View File

@@ -29,6 +29,12 @@
#ifndef output_pt8211_h_
#define output_pt8211_h_

//uncomment to enable oversampling:
#define AUDIO_PT8211_OVERSAMPLING
//uncomment ONE of these to define interpolation type for oversampling:
// #define AUDIO_PT8211_INTERPOLATION_LINEAR
#define AUDIO_PT8211_INTERPOLATION_CIC

#include "Arduino.h"
#include "AudioStream.h"
#include "DMAChannel.h"
@@ -47,7 +53,11 @@ protected:
static audio_block_t *block_right_1st;
static bool update_responsibility;
static DMAChannel dma;
static void isr(void);
static void isr(void)
#if defined(AUDIO_PT8211_OVERSAMPLING)
__attribute__((optimize("unroll-loops")))
#endif
;
private:
static audio_block_t *block_left_2nd;
static audio_block_t *block_right_2nd;

Write Preview
Loading…
Cancel
Save