Digital Audio Playback not working

Ok, but then the AIC23B supports 8Khz sampling rate, in that case what should be the input from the I2S side (BCLK, LRCIN etc) to run codec @ 8Khz sampling rate.

Because my simple calculation was, my BCLK is 384Khz (which I cannot change), so if I send 24 bit samples L/R and based on that the sampling rate would be (384/(24*2)) = 8Khz.

Remember that I am running the Codec in Slave.

If the AIC23B is slave, then BCLK=384 kHz is ok. LRCK would be 8 = 8kHz and must be exactly divided from your Master clock.

here is a good app note on the topic of clocking the audio serial interface (ASI):

http://www.ti.com/lit/an/slaa469/slaa469.pdf

Thank you, that solves quite a bit of doubt. 

I was referring to the app note and had another query on that.

I am referring to the page 4, figure 3 (ASI Master clock, independent Master clock), this is actually the scenario on our development board. The note says that with this configuration it may cause skipped or/and repeated samples.

Can this cause the audio playback to not work (skipped).

Note:

1. I am continuously sending data from the I2S Master (DIN).

2. I am not able to hear anything on the codec output (headphone/speaker)