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On High Sample Rate Audio Codec Pass Band

Category: Datasheet/Specs
Product Number: ADAU1860 ADAU1861


I had a few doubts/questions regarding the High Sample Rate Audio Codecs (192k to 768k) Audio Codecs available from Analog Devices.  I was trying to use the Audio Codecs for waveform generation and capture at frequencies above 20kHz and below 60kHz.  While looking into the many data sheets of high speed audio codecs, which have sample rates of 192kHz and 768kHz, I am finding a reference of a filter that provides a cutoff at 20kHz or nearby (A Weighted Filter and 20kHz Flat Cutoff Filters).  I was wondering about the following questions:

1.  Is it possible to bypass this filter so that the full sample rate capacity of the codec can be used to sample relatively high frequency voltage waveforms above the human auditory range ?  I am particularly interested in this because then it lets me use the I2S interfaces and software layers available with popular DSPs and embedded Operating Systems, without designing an interface for an ADC/DAC and it would help me reduce my circuit foot print, for building ultrasound applications.

2.  Are there any audio codecs that is available from Analog Devices that can do 192Ksps or above sample rate at 24 bits without the filters and providing cutoff at or near half Nyquist Rates ?

3.  If these filters are not removable from the processing chain, and if these codecs are limited in operation to 20kHz and below,  what additional benefit is provided by the high sample rates provided by 192k and 768k codecs ?  Does it improve fidelity in some sense ?  Isnt 48k or 96k sufficient enough ?  What kind of applications actually use these high sample rates in an audio codec with a filter that limits audio pass band at 20kHz ?

Thank you very much and Best Regards

Ajith Peter

  • Hello Ajith,

    Sorry I missed your post. I have been checking this forum for unanswered questions a lot and somehow yours was missed. 

    First I need to add that the ADAU1860 and 1861 were designed out of our China office and is supported by the group in China. They should have responded when you used the part number. 

    The part designed and supported in the USA is the ADAU1787 and is very similar except it has a SigmaDSP core rather than a HiFi core. 

    Your three questions are somewhat interrelated so I will respond and I should shed some light on several of your questions. 

    First, the purpose of these high sample rates is low latency. These parts are designed for ANC headphones so this is the signal path it is meant to use: ADC-->FDSP-->DAC. So if you run the ADC, the FDSP and the DACs at 768kHz fs you get a latency in the single digit micro-seconds! The FDSP is short for "Fast DSP" which is a limited instruction DSP core which is basically a Bi-Quad filter engine to calculate Bi-Quads really fast and efficiently. 

    The filter coefficients for the Bi-Quads need to adapt and change to changes in the environment and audio noise around the headphones. This does not need to happen super quickly. So the parts have synchronous sample rate converters to take in the audio at the high rate and slow it down and send it to the other DSP core or out of the part to another external processor. In the on-chip core there is usually filtering and tracking of levels that is done and possibly some calculations of filter coefficients but often this pre-processed data is then passed on out of the serial ports via an ASRC to an external processor which determines updates to the filter coefficients and then performs an I2C/SPI write to change the filters running in the FDSP. 

    The serial ports cannot operate above 192kHz in most real-world situations. So if you take the ADC output you must go through an ASRC to get it out of the serial port. The ASRC will scale the filters to the Nyquist frequency so you should get close to 1/2fs performance at the rate of the serial port. Keep in mind that this entire part is designed for low latency so the filters involved are optimized for latency not pass band flatness or for a steep cutoff frequency. The aliasing performance will not be stellar due to these design tradeoffs. 

    I would give the ASRC something to change to and from so I suggest you run the ADCs at 384kHz fs and then run the serial port at 192kHz and you should get close to 90kHz of bandwidth. It should be fairly flat to 70 or 80kHz. 

    There are other codecs and DS parts that would also do this. The ADAU1372 will do this and the ADAU1772 and ADAU1777. 

    Keep in mind that these are SigmaDelta converters so it will not work well for any arbitrary waveform generator. It will not do a square wave well or a ramp. but, any sine wave based waves it will be fine. 

    Dave T