The megaburrito filter runs at 32 bits -- the input 16 bit signals have zeros added at the input and are all output as original - that is the same 16 bits followed by zeros. The megaburrito filter also does 32 bit computations on the original samples and outputs 7 interpolated samples at 32 bit resolution, rounded to 20 bits (Yggy), and 18 bits (Gumby). For the Bimby, there are the same 32 bit calculations resulting in three interpolated samples at 32 bit original resolution, rounded to 16 bits.
So for the Bimby only, there are only 16 bits of data output. The advantage is the extra sample interpolation with the frequency and time domain optimization.
Would it be possible to explain what the megaburrito filter does in the most basic, easy to understand way possible? I have read tons about it on hear but don't fully understand what it does. If it is only on the interpolations why not just use the original samples for it? I don't know if I'm understanding that correctly.
I was able to read an insightful article somewhere to get the difference between delta sigma and multibit.
The process of digital-to-analog conversion creates spurious high frequency noise that must be filtered out.
If you convert a 16-bit 44.1khz CD to analog, then to keep the spurious high frequency noise out of the analog output, you need a really steep low-pass filter (if you do not understand "steep low-pass filter" then you need to read Wikipedia on EQ (equalization).)
A really steep analog low-pass filter is either really expensive or else it sounds like crap.
So, early in the 1980s, "oversampling" was developed. This means that the audio signal is converted to 88.2khz (" 2X "), 176.4khz (" 4x "), or 352.8khz ( " 8x " ). This is done, as Mike explained above, by "interpolating", i.e. creating data points in between the existing data points. For example, if there are two data points that are 17 and 21, then the oversampling filter calculates that halfway in between, it was probably 19 (oversimplified example).
Anyway, when the sample rate is 176.4khz, then the analog low-pass filter can have a much gentler curve, and can be done inexpensively in a way that is relatively transparent.
But, all the new data that was added by the oversampling filter - is also part of the analog output of the DAC. So, the fact that Mike's oversampling filter does a better job of predicting the missing data, means that what is produced by the DAC, will sound more like the original analog signal.
(
The reason that both the multibit DAC and the high quality oversampling are innovations, is that DACs are used in so many new devices - Smartphones, laptops, TVs, Chromecast sticks, etc - that all of the major development has been in the opposite direction - simple and cheap approximations rather than complex, more expensive, but more exact output. )