Dear sir or madam,
I successfully tested the SSM3582. Recently I found out that the output seems to clip when the input signal is above 20 bits wide. Why is this so? Is this due to the clipping of the DAC interpolator? What is the maximum input signal then? Can I avoid this by some settings? I use PVDD = 16 V. The Limiter does not seem to be triggered. I also already monitored the I2S signal conditioning and this seems to be perfect.
In addition I observed when working with an prbs input signal with amplitude 2^20, where only the MSB toggles, the output is zero. Using a sine signal with amplitude 2^20 the output sounds fine, however a lot of harmonic distortions can be seen on the spectrum of the speaker output. DAC Frequency is set to 8 kHz. I read the fault registers but everything seems to be fine. Can the high frequency clipping be disabled?
OK, I saw the first post mentioned PVDD = 16V and wanted to match gain settings. For 12V this is fine. Using your project, I connect the Audio Precision I2S Transmitter to a EVAL-SSM3582Z board…
OK, I saw the first post mentioned PVDD = 16V and wanted to match gain settings. For 12V this is fine. Using your project, I connect the Audio Precision I2S Transmitter to a EVAL-SSM3582Z board and can see valid output all the way up to 0dBFS with the settings that are shown in the attached image.
A few notes: the MCLK is not being used in this case. The BCLK = 8kHzx256=2.048MHz. Also, I found if the formatting of the Audio Precision and the SSM3582 are not aligned properly, there may be a shift of the MSB, causing invalid output at digital input amplitudes above -6dBFS. I am wondering if this is the problem you are having. Are you able to see the formatting of the TDM source and compare to these settings? Likewise, note settings such as the "FSYNC Mode," and "Serial Data Format" in the SigmaStudio project can be used by the SSM3525 to align it's formatting with the source. So if the TDM source has different settings, your SigmaStudio project may need to adjust these settings.