I’m using Acon Digital’s Equalize 2 EQ plugin to apply loudness compensation for low volume listening which works great (in boosting the bass and treble according to relative loudness contours).
I’m streaming a 16-bit 44.1kHz FLAC file, which has the bit depth increased to 64-bit before entering the VST3 plugin (please see screenshot).
However when it leaves the processing chain (to go to a WiiM Ultra streamer) Audirvana applies a bit depth reduction, I imagine to match the original FLAC file (please see screenshot of the DAC input signal).
I have the UPnP settings to maximum bit depth and the streamer can handle 24 bit.
Is there anyway please that there could be a feature that avoids bit depth reduction (to match the input FLAC), but instead could match a predetermined user-configurable value in the settings (i.e. that supported by the DAC)?
(So not another link in the chain as such, like the up-sampling, but more a ‘preserving’ feature on the DAC input settings although actually the Volume Control stage of the screenshot is the stage that shows the bit depth reduction.)
Note also that when streaming an originally 24-bit FLAC the signal leaving the chain is also 24-bit so Audirvana’s doing the bit reduction from 64-bit to 24-bit, as expected. I mention this because if it can do both 64 → 16 bit and 62 → 24 bit then a customisable user setting to determine which (for FLACs with lesser original resolution) might be quite simple to implement?
(In order to preserve the higher dynamic range, and with respect to the higher resolution of the VST3 processing.)
Note: Your screen shot shows the DAC receiving a 16-bit signal… This does not reflect that the DAC is operating on a 24 or 32-bit signal…
Audirvana is processing the plug-in with 64-bit precision as is the case with every 64-bit application/plug-in, and subsequently decimating the result to a real-world bit-depth associated to the DAC capabilities… You are not losing anything in this process, but you are gaining in mathematical precision of the plug-in DSP… The sample-rate is not affected, … This is different than bit-depth/sample-rate conversion processing for output to your DAC.
Yes the plug-in is processing with 64-bit precision, and altering the signal with EQ, within the dynamic range that 64-bit offers. It produces a modified waveform within the 64-bit space.
Audirvana is then subsequently (at the end of the processing chain) doing bit depth reduction to 16-bit, presumably to match the original input FLAC’s resolution, which is 16-bit.
My request is: it would be better if Audirvana did bit depth reduction only to 24-bit since 24-bit has a dynamic range of 144 dB, whereas 16-bit only has a dynamic range of 95 dB. So in bit depth reducing all the way down to 16-bit there must be some loss of information, compared to bit depth reducing just to 24-bit.
So, when VST processing (64-bit) is active, rather than Audirvana doing bit depth reduction to match the INPUT file, it should do bit depth reduction to match the OUTPUT capabilities (the input capabilities of the DAC).
When processing a 24-bit input file, the VST plugin processes in 64-bit still, and Audirvana subsequently bit depth reduces to 24-bit (to match the input file).
So my point is: if Audirvana can bit depth reduce to both 16-bit and 24-bit (depending on the input resolution) then it might be straightforward to let the user specify which (based on the DAC capabilities)?
That way a greater dynamic range will be preserved, thus increasing the quality of the music.
Currently the current bit depth reduction is losing information and dynamic range, and therefore the implementation is flawed.
In the DSP calculations the registers/accumulators are capable of handling the buffering of the 64 bit processing operation calculations… It is not simply adding zero’s to the file data… the processing is altering the file data by adding or subtracting bits when the parametric adjustments are defined and applied… The headroom of the DSP calculations are related to the capabilities of the accumulators in a 64 bit calculation where data is buffered during the algorithmic operations of the plug-in.
Bit-depth is relative to the source ADC and not related to the math involved in the DSP calculations.
Again you are conflating ‘bit-depth’ related dynamic range of the file with the 64 bit mathematics employed in the DSP calculations and accumulator/buffers… What is happening with the file, is the bit-depth does not change, otherwise your plug-in won’t operate… the DSP is calculated related to the bit-depth and sample-rate of the file that is introduced to the plug-in…
Yes agreed. But the DSP still operates in a 64-bit space. The EQ is modifying what’s originally a 16-bit input, and in doing so is creating details and waveforms with more precision than the original 16-bit.
Imagine if 16-bit means 1 decimal place and 64-bit means 2 decimal places.
For simplicity:
Say an input sample has a value of 2.5 (16-bit).
Audirvana converts to 2.50 (64-bit).
Say a gain of 2.50 (64-bit) is applied.
Say the VST calculation is 2.50 x 2.50 = 6.25 (64-bit).
When I reduce the bit depth back to 16-bit (1 decimal place) I get 6.3 when rounded.
So you’re losing 0.05 of detail introduced by the VST plugin.
The resolution of the calculation is the issue… not the dynamic range… the calculations are done out to 64 zero’s and then truncated to a real-world operational resolution… this is intrinsically tied to the capabilities of the plug-in operation…
No… the 64 bit math is applied to the bit-depth and sample-rate of the file introduced to the plug-in architecture… Again you are conflating the ADC bit-depth with the mathematics used in the plug-in architecture… If you did not use the plug-in architecture, your DAC will only operate on the bit-depth and sample-rate of the source file…
It’s easy to understand… Using your logic, if the file is being upsampled to 64 bit-depth before being processed by the plug-in, it (the plug-in) will not work when this 64 bit-depth file is introduced to it.
Only the resolution of the mathematics used in the DSP calculations… Again, you are conflating bit-depth dynamic range of a file with the resolution of the DSP processing calculations.
Do you know of any digital-audio files that are recorded at or upsampled to 64/96kHz…? (your plug-in is good to 96kHz) can you upsample your files to 64/96kHz… No.
