All ADCs/DACs have some sort internal data format that is used to get from digital to analogue. The converter circuit and the internal format are chosen together, depending on the designer's judgment or expectation of which combo is optimal.
There are only two cases where the internal format coincides with an existing audio format. On one extreme are R2R DACs that directly convert the PCM data you feed them by controlling 24 switchable current sources, each of which has half the current of the previous one. On the other extreme sit 1-bit converters that switch a single current or voltage source but at a high sampling rate. Both these extremes have limitations and the most common choice these days is a kind of compromise. They use a small number of bits (typically 5) that are used to control 32 (2^5) current sources with nominally equal currents, plus some trick to make the conversion minimally sensitive to any imbalances in those currents. The Tambaqui sits more or less in this camp: the PWM signal has a switching frequency of 3.125MHz, and can take 33 discrete lengths from 0*10ns to 32*10ns. So it's basically a 5-bit, 3.125MHz converter. The choice for PWM was given by the need to get rid of those idle tones I mentioned.
In any DAC design the internal format is essentially fixed, as it's a design choice. So any type of digital signal first gets converted to that format. That conversion consists of two stages. The first is upsampling and that takes the sample rate up to the sample rate of the internal format. Then the second is sigma-delta modulation (or noise shaping, however you want to call it) which takes the word length down to the word length of the internal format while preserving the SNR inside the audio band. Unfortunately, a succession of audio product and sales managers have conspired to conflate the second stage with the first, thereby confusing just about anyone with a passing interest in the technical background.
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The latency is indeed around 100ms, for all inputs. There's no real reason for it other than at some point we finally got the ASRC algorithm working glitch-free by making the input data buffer rather long. Much later we figured out how to get that down to less than a millisecond but never got down to applying that insight to the Tambaqui. Oh well.
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I hope I'm not offending anyone if I don't take part in the discussion further, but I have amplifier and speaker work on my plate. As I said the DAC project was done in 2013... Before Purifi, and before Kii even...