Thoughts on software based filters/ modulators/ upsamplers - have you tried them with your hardware and what pros and cons ?
Who uses Software filters/modulators/upsampling?
- Thread starter Audiophile Neuroscience
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Just to clarify the question, do you mean using software based filters/modulators/upsamplers to override your DACs’ own filters/modulators/upsamplers?
Because all DACs have their internal hardware filters/modulators/upsamplers. Even for an NOS (non-oversampling) R2R ladder DAC, if the DAC natively runs at 352kHz or 394KHz (8fs) and you’re playing 16-bit 44.1KHz, you’re still running a sample and hold filter where each 16-bit sample plays 8 times until you move onto the next sample.
Or similarly, if you’re playing a 2.82MHz (1fs) DSD and your DSD DAC natively has 16 DSD modulators, your DAC is not playing the same DSD signal natively to each modulator at the same time. Native DSD playback is usually running a shift register so each modulator is playing the DSD signal shifted by 1 sample.
Since most DACs are hardware limited in computational power, the pros of running software filters/modulators/upsamplers is that you can get better noise shaping and more precise upsampling for the modulators of your DAC in theory so you should get more accurate transients and lower noise leading to better timbral accuracy and instrumental separation. The cons is that because you’re running more traditional PC hardware, it is very easy to introduce significantly more noise into the DAC which leads to harsher sound, more lower level noise, worse timbral accuracy and worse instrumental separation. Another con is that you have infinite number of filters/modulators/upsamplers to choose from so you can easily get lost in things or you might prefer one filter that is less optimal for one recording and then ended up choosing a less optimal filter for your DAC. And if you don’t understand how your DAC works, you could just be doing busy work as you can feed your DAC a signal that your DAC would just turn around and run it through its internal hardware, negating most of the benefits of the software filters/modulators/upsamplers.
Because all DACs have their internal hardware filters/modulators/upsamplers. Even for an NOS (non-oversampling) R2R ladder DAC, if the DAC natively runs at 352kHz or 394KHz (8fs) and you’re playing 16-bit 44.1KHz, you’re still running a sample and hold filter where each 16-bit sample plays 8 times until you move onto the next sample.
Or similarly, if you’re playing a 2.82MHz (1fs) DSD and your DSD DAC natively has 16 DSD modulators, your DAC is not playing the same DSD signal natively to each modulator at the same time. Native DSD playback is usually running a shift register so each modulator is playing the DSD signal shifted by 1 sample.
Since most DACs are hardware limited in computational power, the pros of running software filters/modulators/upsamplers is that you can get better noise shaping and more precise upsampling for the modulators of your DAC in theory so you should get more accurate transients and lower noise leading to better timbral accuracy and instrumental separation. The cons is that because you’re running more traditional PC hardware, it is very easy to introduce significantly more noise into the DAC which leads to harsher sound, more lower level noise, worse timbral accuracy and worse instrumental separation. Another con is that you have infinite number of filters/modulators/upsamplers to choose from so you can easily get lost in things or you might prefer one filter that is less optimal for one recording and then ended up choosing a less optimal filter for your DAC. And if you don’t understand how your DAC works, you could just be doing busy work as you can feed your DAC a signal that your DAC would just turn around and run it through its internal hardware, negating most of the benefits of the software filters/modulators/upsamplers.
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Many thanks for detailed reply
I have the Gryphon Kalliope DAC with "selectable32-bit/210 kHz asynchronous sample rate conversion and a dedicated ESS SABREES9018 32-bit D/A converter per channel, incorporating eight individual D/Aconverters in Dual Differential coupling"
You hit on an interesting point about "overriding" the DACS filters/modulators/upsamplers. I have only a rudimentary knowledge of such things but can understand that a digital signal can be upsampled prior to sending to DAC, and presumably delta sigma modulation from a multibit digital number to high speed 0 and 1s can also occur prior to the DAC and presumably filters like slow and fast, linear phase etc,apply to that SDM. Presumably the analog reconstruction filter can only occur in the actual DAC. If correct, yes I get that using a really powerful computer could be advantageous at taking these tasks out of the hands of the DAC, and you can add many more choices for better or worse.
I also understand that using PCs may introduce noise although they can be built for purpose with low noise PSUs and components
Still, what strikes me as odd is how does this new improved signal play with your DAC. Do you turn off oversampling for example etc
The next question would be, in the case of expensive DACs, one would expect that the computational power has been matched to the required functions
Then, as you say there may be a bewildering number of software settings and some may work better or worse depending on different circumstances or clash with your DAC
Not the least, there is also the spectre of PC and software glitches, learning curves, and many hours spent on forums to fix potential issues which can divert you from enjoying the music.
So, my original question is in an attempt to find out how people here view software based media players stacked with filters/modulators/upsamplers. I am also curious what the experience is in relation to higher end DACS which one would hope would perform well in the first place.
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Across multiple systems and DACs I’ve experimented with HQ Player’s upsampling (inline) and also PGGB (offline).
When the settings are “just right” there has been, in many of my setups, a noticeable improvement in “air”, “space” and lack of harshness. Settings have varied by DAC and even for a given DAC by the bit rate of the source material. The author of PGGP, which I apply offline, has offered great advice on settings to avoid the endless experimentation.
There’s a long PGGB thread over on Audiophile Style.
IMO it’ worth the small price to play with it, if only to better understand the sonics of alternative algorithms.
When the settings are “just right” there has been, in many of my setups, a noticeable improvement in “air”, “space” and lack of harshness. Settings have varied by DAC and even for a given DAC by the bit rate of the source material. The author of PGGP, which I apply offline, has offered great advice on settings to avoid the endless experimentation.
There’s a long PGGB thread over on Audiophile Style.
IMO it’ worth the small price to play with it, if only to better understand the sonics of alternative algorithms.
I have a Gryphon Diablo 300 with internal DAC, which I think is similar to the Kalliope DAC.
I also have a Holo May L2 R2R DAC.
I use HQ Player running on a Mac Mini on my network, the data connection to the streamer is fibre optic. The streamer is Innuos Pulsar, which has HQ Player Endpoint embedded onboard.
The Gryphon Diablo 300 DAC is improved by sending data upsampled to 32/384 PCM. There is a distinct improvement in resolution.
Using Holo May in non-oversampling mode with HQ Player upsampling to DSD256, there is more detail and air, but in comparison the Gryphon Diablo 300 DAC has a little more warmth and is pleasing in its own way.
Neither have any audible harshness or noise whatsoever.
The thing about the Holo May in a blind test with a Lumin was the increased air and resolution because the noise floor is about 20dB lower. Inaudible noise floors matter!
I hate the expression, by the Holo May is a touch more analogue, even though the 300 DAC is still very good. So I kept the Holo May.
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