Posted on September 26, 2016 at 23:44
I'm already using two halves of the buffer when outputting the audio. Otherwise the audio will be affected by the updating process in the callback routine. This is how i did it for outputting a sine wave on the left channel and a triangle on the right channel.
#define BUFLEN 128
#define FREQ 11650000
uint32_t samplecounter = 0;
// the function ''readinterpolated'' outputs the waveform amplitude. The input phase is given as a uint32_t.
void
BSP_AUDIO_OUT_HalfTransfer_CallBack()
{
// update first half of buffer
for
(
int
i=0; i<BUFLEN/2; i++)
{
// left channel
buffer[i*2] = readinterpolated(samplecounter*FREQ, sindata)*32000.0f;
// right channel
buffer[i*2+1] = readinterpolated(samplecounter*FREQ, triangledata)*32000.0f;
samplecounter++;
}
}
void
BSP_AUDIO_OUT_TransferComplete_CallBack()
{
// update second part of buffer
for
(
int
i=BUFLEN/2; i<BUFLEN; i++)
{
// left channel
buffer[i*2] = readinterpolated(samplecounter*FREQ, sindata)*32000.0f;
// right channel
buffer[i*2+1] = readinterpolated(samplecounter*FREQ, triangledata)*32000.0f;
samplecounter++;
}
}
However when initializing the BSP_AUDIO_IN_Init and BSP_AUDIO_OUT_Init separately it seems like the data written to the buffer from the input jack is out of sync with the updating of the output buffer. One could of course have the input to a separate circular buffer and fetch that data from the output buffer but I see no reason why such a workaround should be needed.
Edit: I forgot to mention that I want to be able to superimpose the incoming audio from line-in to some generated waveform such as the example above.
