Anyone familiar with Audio codecs,specifically the the one(WM8994) on stm32f7dis

[sjpatel1992]

Hi All! I am having a hard time making the communication between the audio codec and the microprocessor to work. I am using the new stm32F7 discovery board

I was able to configure the audio codec to output the audio picked up by two digital mics directly to the headphone jack through the digital sidetone path. Now, I would like to send the mic data to the micro controller to do some signal processing.

However, I don't really know how the audio codec outputs the raw digital mic data through AIF1 interface so that I can properly configure the SAI2 to receive it. I know the data coming from the digital mics is in PDM format. Does the codec convert it into PCM internally before outputting?

Here is how I have configured the audio codec

Codec is set to operate as a slave.

Audio Protocol ----> I2S Standard

Sampling Rate ----> 48kHz

Data Size -----> 16 bits

Timeslot 0 is disabled and Timeslot 1 is enabled to receive the mic data.

On the microcontroller side, I have configured SAI2 as follows

SAI2_BlockA -- Master TX

SAI2_BlockB -- Slave RX ( Synchronus with BlockA)

MCLK -- ~13 Mhz

Total Slots -- 2

Frame Length -- 32

Active Frame Length -- 16

Data Size(bits) -- 16

Please let me know if more details are required. Thanks

I have read the datasheets numerous times and haven't been able to find anything on this. I also posted this on the stm32 forum, but haven't been able to get any help.