Audio Delay versus Sample Rate
When operating the ADAU 1701 with double to fourfold sample rate the audio delay (analoge input to analog output) is not 1/2 or 1/4 compared to 48 kSps as expected.
I measure 1/3 and 1/9 compared to 48 kSps.
What is the reason behind that? Are the filters in the sigma-delta ADCs and DACs changed with the sample rate?
Audio Signal Processors
hardware
Audio Delay versus Sample Rate
SigmaDSP Audio Processors
adau1701
Hello audiochiemgau,
Yes, the filters do change but as you noted not quite as much as expected. I need to start by saying I am not 100% certain of the ratio here. Generally the ADCs we design will run with an internal clock of 6.144MHz. This is with a 12.288Mhz MCLK and will be decimated by 128 to get to 48kHz. So it is a 128x oversampling filter on the ADC. When you switch to 96kHz it switches to 64x and for 192kHz it is 64xfs. Some of our DACs will operate at 64x, 32x and 16x. I am not sure why that does not halve the time through the filter. I would ask for your project to look it over but there really is not much control over the ADC and DAC rates.
Is this turning into a problem for your design? We have some other parts that have lower latency filters on the converters.
Dave T
Hi Dave
Thanks for your answer. I need very low latency from analoge input to analog output of the DSP because it is in a feedback loop. Can you recommend DSPs with low latency (<< 1ms) and four ADC and two DACs? I would like to stay with SigmaStudio.
Hello
The ADAU1787 fits this best.
Dave T
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