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Analog Monitor and Control at 16-bit resolution

BrianBrianBrian
Expert 1085 points

Other Parts Discussed in Thread: ADS1262, ADS1256, ADS1271, ADS1255, ADS1274, ADS1278, OPA1632

My application requires multiple channels of synchronized precision ADC and DAC of at least 16-bits of resolution and at least 48kHz sampling rate.  The AMCxxxx chips satisfy all the requirements except for the resolution (maximum is 12-bits).  Why are there no 16-bit systems?  Is there a reliable way to get synchronized ADC/DAC from separate (i.e. non-integrated) ADC and DAC chips?

Let me emphasize that I am interested in precision ADC/DAC (i.e. DC accurate) which to my knowledge rules out audio codecs.

  • 0
    TI__Guru** 101005 points
    Brian,



    The AMC chips were design for a specific function that generally require only 12 bits of resolution.

    I'm sure that you could get 16 bits of performance from other devices, but it depends on what you need from your system. I generally advise on precision ADCs that come from the delta-sigma products (most are 24 bit, but some 16 bit devices). Most of the devices are a bit slower than the 48 kSPS that you require and the great majority of those have some sort of frequency response in the digital filter that might make it difficult for you to use.

    Of the products that I know, you might be able to use the ADS1271, ADS1274, ADS1278, ADS1255, ADS1256, and perhaps the new ADS1262. Again, you'd have to check out the datasheet. The digital filtering may significantly reduce the bandwidth. However, I suspect a SAR ADC might work better, I'll see if one of the SAR application engineers can help with that. I'll also see if the DAC engineers can help with something suitable as well.

    As for your requirements, what are you building? How many channels are you going to use?


    Joseph Wu
  • 0 in reply to Joseph Wu
    Expert 1085 points

    Hi Joseph

    The application domain requires using analog voltages (-10V to +10V) to control various parameters of a user's audio synthesis and processing units.  These analog control voltages need to be DC accurate (for accurate control of pitch and so on) but also be able to cover the audio frequency range (for audio-rate modulations and so on).  

    My product will have ideally have about 8-16 channels of ADC (16-bits over -10V to +10V) feeding into a uC/DSP and then back out to 4-8 channels of DAC (also 16-bits over -10V to 10V).  Also, ideally I would like to be able to do sample-based processing rather than frame-based sampling (for low latency).  This implies to me that it is important that the ADC and DAC are clocked together (one sample in on the ADC for every one sample out on the DAC).  Strict phase coherence across the ADC or DAC channels is not as important to me.  If the ADC and DAC are not clocked together than I am assuming there will be some sample rate drift between the two and I will need to buffer and resample the ADC inputs to match the DAC output sample rate.  I would prefer to avoid this complexity and the resulting latency.

    Looking at the audio codec and AMC offerings, strongly suggests that having the ADC and DAC in the same chip, sharing the same clock is the way this is achieved but unfortunately my resolution and bandwidth requirements seem to fall exactly between the two product groups.  I can find separate ADC and DAC chips that match my signal criterion but then how do I synchronize them?

    So I guess my question now becomes:  if I was using separate chips for ADC and DAC with potentially different interfaces is there a way to synchronize their sampling/latching to achieve the required "one sample in one sample out"?

    Kind regards,

    Brian

  • 0 in reply to BrianBrianBrian
    TI__Genius 11750 points

    Hi Brian,


    Thanks for you interest in the AMC7XXX.  The first thing to note is that the on-chip oscillator is primarily responsible for the sampling frequency of the ADC.  The sampling frequency of the ADC is dynamic and dependent on the acquisition and conversion time of each channel -- the sample rate is determined through register settings.  This information is included is most datasheets.


    The DACs are typically independent of the internal oscillator and update when their corresponding registers are changed and set. Although the DAC block is independent of the oscillator frequency, you can try to synchronize and update through SPI register writes/reads.

    Do you plan to constantly change the DAC output when making new ADC reads?


    Best Regards,

    Matt

  • 0 in reply to BrianBrianBrian
    TI__Guru** 101005 points
    Brian,


    As a follow-up most ADCs operate from 0 - 5V and you'd need to have some sort of buffering to get and input from +/-10V down to that range. I'd guess that you might want to use something like the OPA1632.

    As for initiating a conversion, you can use the master to initiate a conversion. Most devices have some method to give a conversion on demand. In most of the devices I deal with, we sometimes have a START pin to initiate or start a new conversion or there is SYNC command word that is issued through SPI to start a new conversion. I believe that many SAR devices use something like a START pin to initiate a conversion.


    Joseph Wu