I am not sure about the approach on sampling my digitized audio from an external source. My current setup consists of an MCLK clock of 24.576MHz which is also tied to the BCLK. With this clock speed, I can achieve super high quality audio but I only need 8bit 8kHz quality audio.
I've already posted another message on this and nobody answered although my post got 31 views. Can someone please help?
On my EVM board, I physically tied a piece of wire between my DOUT and my DIN pins to ensure that what was coming-in on my input IN1 (J1) was going out on my output (J9). That all works fine. I can also see my data with my scop on the DOUT (to DIN) pin.
I can read/write all registers on the AIC3204 through a command-line interface tied to a PIC over RS232.
So, I figured that if I have to read each bit of data on each BCLK tick, 24.576MHz is way too fast for my PIC running a 16.384MHz clock therefore I figured that I could perhaps lower the BCLK speed to 512kHz using a PWM on my PIC.
If I was to provide MCLK and BCLK a 50% duty cycle square wave from a PWM at 512kHz, how would I be able to achieve 8bit 8kHz audio? Do I need to use the PLL rather than simple MCLK for the CODEC_CLKIN?
I am just not sure what approach to take to be able to sample the data on the DOUT pin. I can use the WCLK to get the left-right audio, the BCLK to clock-in the bits, but I need someone to help please.
PLEASE HELP!!
Thanks.
Benoit