Tps65950 ADC timing

Hi,

Hi Volkan,

I think you are right that samples will be interleaved/delayed (although I admit it isn't 100% clear from the datasheet) and I as well would like somebody from TI to confirm :-)

This being said: In case you need 100% synchronized samples I think you should "easily" (on the connected OMAP?) be able to resample/interpolate the data to move either right or left sample point ½ sample back or forth in time and thereby get right/left samples 100% synchronized.

I hope this helps you forward?
  Søren

Hi,

Thanks for your help. If i undestant correctly you say data interleaved/delayed but.

referring to page 691 of swcu050e.pdf (Technical Reference Manual ver E):

Option 1: Four simultaneous multimedia analog channels (four multimedia DACs and no modem voice

path) and two simultaneous ADCs or four channels coming from digital microphones.

And also at page 682 of swcu050e.pdf :

NOTE: The main microphone captures the voice and surrounding noise, whereas the

submicrophone captures only the surrounding noise. The noise cancellation is performed by

the external host processor by subtracting the submicrophone data from the main

microphone data, thus keeping only the voice data.

I couldn't see any explanation about it. Method or how to do. If they simple subtracting like they say is it hardware or is there LMS algorithm as soft. implemantation.

I don't think noise cancelation can be achieve with time delay like 1/2 sample.

If there is any time diagram of samples or any explanation about, it will be very helpful.

Volkan

Hi Volkan,

On the PMIC side the sampling is always on the rising and falling edges of the clock. This is probably very standard in all types of interfaces. On rising edge one digital microphone provide the PDM data, and on the falling edge the other digital microphone provide the PDM data.

In case you are asking for the sampling done on the digital microphones then this is something you may have to find out from the digital microphone provider. I dont know how the ADC within the digital microphone would be functioning, they may be taking analog data at the same time or may have some delay.

I hope this helps. 

Regards,

Gandhar.

Hi Gandhar,

I think Volkan refers to the internal Audio ADCs in TPS65950, but just refers a wrong figure, but I admit that I'm not sure :-). Anyway, I at least would like to know - both for ADCs and DACs -  if the samples for right and left channel are aligned (sampled/reconstructed at the same time) or interleaved (sampled/reconstructed as they leave/arrive in the I2S data streamto/from OMAP). Not that it really matters for most use cases, but Volkans question made me curious and I think it should be pretty easy to answer although the TRM seems to be pretty weak on this point - At least I couldn't find any clear indication?

Best regards and thanks
  Søren

Hi,

Again i'm sorry for my weak explanations.

But  Søren is right. I'm not asking digital mic.s, i only used that figure to explain my problem(There is time diagram for dig. mic but not analog inputs). 

I'm asking Audio ADC. To be clear:

If i will use AUXL and AUXR inputs or Main mic and sub mic inputs.

Can i achieve concurrent sampling, is there any time diagram or how to do document(for exp. programming referans).

Is there any further explanation other then i quote upper comment.

Thanks...

Hi Gandhar,

Thanks for your quick and clear reply and for proving me wrong :-(

Do you know if the same apply for the DAC outputs? I'm just curious...

Best regards and thanks
  Søren