void AudioOutputI2S2::isr(void)
{
int16_t *dest, *dc;
audio_block_t *blockL, *blockR;
uint32_t saddr, offsetL, offsetR;
saddr = (uint32_t)(dma.TCD->SADDR);
dma.clearInterrupt();
if (saddr < (uint32_t)i2s2_tx_buffer + sizeof(i2s2_tx_buffer) / 2) {
// DMA is transmitting the first half of the buffer
// so we must fill the second half
dest = (int16_t *)&i2s2_tx_buffer[AUDIO_BLOCK_SAMPLES/2];
if (AudioOutputI2S2::update_responsibility) AudioStream::update_all();
} else {
// DMA is transmitting the second half of the buffer
// so we must fill the first half
dest = (int16_t *)i2s2_tx_buffer;
}
blockL = AudioOutputI2S2::block_left_1st;
blockR = AudioOutputI2S2::block_right_1st;
offsetL = AudioOutputI2S2::block_left_offset;
offsetR = AudioOutputI2S2::block_right_offset;
if (blockL && blockR) {
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 {
memset(dest,0,AUDIO_BLOCK_SAMPLES * 2);
}
#if IMXRT_CACHE_ENABLED >= 2
arm_dcache_flush_delete(dest, sizeof(i2s2_tx_buffer) / 2 );
#endif
if (offsetL < AUDIO_BLOCK_SAMPLES) {
AudioOutputI2S2::block_left_offset = offsetL;
} else {
AudioOutputI2S2::block_left_offset = 0;
AudioStream::release(blockL);
AudioOutputI2S2::block_left_1st = AudioOutputI2S2::block_left_2nd;
AudioOutputI2S2::block_left_2nd = NULL;
}
if (offsetR < AUDIO_BLOCK_SAMPLES) {
AudioOutputI2S2::block_right_offset = offsetR;
} else {
AudioOutputI2S2::block_right_offset = 0;
AudioStream::release(blockR);
AudioOutputI2S2::block_right_1st = AudioOutputI2S2::block_right_2nd;
AudioOutputI2S2::block_right_2nd = NULL;
}
}
void AudioOutputI2S::isr(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) {
// 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::update_responsibility) AudioStream::update_all();
} else {
// DMA is transmitting the second half of the buffer
// so we must fill the first half
dest = (int16_t *)i2s_tx_buffer;
}
blockL = AudioOutputI2S::block_left_1st;
blockR = AudioOutputI2S::block_right_1st;
offsetL = AudioOutputI2S::block_left_offset;
offsetR = AudioOutputI2S::block_right_offset;
if (blockL && blockR) {
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 {
memset(dest,0,AUDIO_BLOCK_SAMPLES * 2);
return;
}
if (offsetL < AUDIO_BLOCK_SAMPLES) {
AudioOutputI2S::block_left_offset = offsetL;
} else {
AudioOutputI2S::block_left_offset = 0;
AudioStream::release(blockL);
AudioOutputI2S::block_left_1st = AudioOutputI2S::block_left_2nd;
AudioOutputI2S::block_left_2nd = NULL;
}
if (offsetR < AUDIO_BLOCK_SAMPLES) {
AudioOutputI2S::block_right_offset = offsetR;
} else {
AudioOutputI2S::block_right_offset = 0;
AudioStream::release(blockR);
AudioOutputI2S::block_right_1st = AudioOutputI2S::block_right_2nd;
AudioOutputI2S::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::update_responsibility) AudioStream::update_all();
} else {
// DMA is transmitting the second half of the buffer
// so we must fill the first half
dest = (int16_t *)i2s_tx_buffer;
end = (int16_t *)&i2s_tx_buffer[AUDIO_BLOCK_SAMPLES/2];
}
block = AudioOutputI2S::block_left_1st;
if (block) {
offset = AudioOutputI2S::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::block_left_offset = offset;
} else {
AudioOutputI2S::block_left_offset = 0;
AudioStream::release(block);
AudioOutputI2S::block_left_1st = AudioOutputI2S::block_left_2nd;
AudioOutputI2S::block_left_2nd = NULL;
}
} else {
do {
*dest = 0;
dest += 2;
} while (dest < end);
}
dest -= AUDIO_BLOCK_SAMPLES - 1;
block = AudioOutputI2S::block_right_1st;
if (block) {
offset = AudioOutputI2S::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::block_right_offset = offset;
} else {
AudioOutputI2S::block_right_offset = 0;
AudioStream::release(block);
AudioOutputI2S::block_right_1st = AudioOutputI2S::block_right_2nd;
AudioOutputI2S::block_right_2nd = NULL;
}
} else {
do {
*dest = 0;
dest += 2;
} while (dest < end);
}
#endif
}
void AudioOutputPT8211::isr(void)
{
int16_t *dest;
audio_block_t *blockL, *blockR;
uint32_t saddr, offsetL, offsetR;
saddr = (uint32_t)(dma.TCD->SADDR);
dma.clearInterrupt();
if (saddr < (uint32_t)i2s_tx_buffer + sizeof(i2s_tx_buffer) / 2) {
// DMA is transmitting the first half of the buffer
// so we must fill the second half
#if defined(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];
#endif
if (AudioOutputPT8211::update_responsibility) AudioStream::update_all();
} else {
// DMA is transmitting the second half of the buffer
// so we must fill the first half
dest = (int16_t *)i2s_tx_buffer;
}
blockL = AudioOutputPT8211::block_left_1st;
blockR = AudioOutputPT8211::block_right_1st;
offsetL = AudioOutputPT8211::block_left_offset;
offsetR = AudioOutputPT8211::block_right_offset;
#if defined(AUDIO_PT8211_OVERSAMPLING)
static int32_t oldL = 0;
static int32_t oldR = 0;
#endif
if (blockL && blockR) {
#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];
int32_t nL = (oldL+valL) >> 1;
int32_t nR = (oldR+valR) >> 1;
*(dest+0) = (oldL+nL) >> 1;
*(dest+1) = (oldR+nR) >> 1;
*(dest+2) = nL;
*(dest+3) = nR;
*(dest+4) = (nL+valL) >> 1;
*(dest+5) = (nR+valR) >> 1;
*(dest+6) = valL;
*(dest+7) = valR;
dest+=8;
oldL = valL;
oldR = valR;
}
#elif defined(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];
int32_t combL[3] = {0};
static int32_t combLOld[2] = {0};
int32_t combR[3] = {0};
static int32_t combROld[2] = {0};
combL[0] = valL - oldL;
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;
}
#else
#error no interpolation method defined for oversampling.
#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(AUDIO_PT8211_OVERSAMPLING)
} else if (blockL) {