Why usb via schiit sounds thin

may be by mistake absolute sound has given R2R term..see what schiit says about their own yggdrasil DAC

Whats this bullschiit about a closed form digital filter, and why does it matter?

Most digital filters destroy the original samples in the process of upsampling. Theyre just like sample rate converters or delta-sigma DACs. Were all about the original samples, so we created a digital filter with a true closed-form solution, which means it retains all the original samples. This is a major difference between Schiit multibit DACs like Yggdrasil and every other DAC in the world.

From this link of schiit- Schiit Audio, Headphone amps and DACs made in USA.
 
Dheerajin, I think in the past they used to call these chips R2R chips. May not have been the right term but that's what they were referred to in the early 2000s.

Doesn't really matter. What's in a name?:)
 
Amit11, as far as I know oversampling was done to simplify design of anti aliasing filter
 
Most digital filters destroy the original samples in the process of upsampling. Theyre just like sample rate converters or delta-sigma DACs. Were all about the original samples, so we created a digital filter with a true closed-form solution, which means it retains all the original samples. This is a major difference between Schiit multibit DACs like Yggdrasil and every other DAC in the world.

From this link of schiit- Schiit Audio, Headphone amps and DACs made in USA.

Well.. I do appreciate what Schiit has written. But keeping the original sample data point while upsampling to integer multiples i.e.. upsampling to twice , thrice , four times i.e. 44.1 to 88.2, 132.3 , 176.4 etc, that by default makes sense and it should be done.. i.e. 1 original data point now has 3 extra points. The first is kept as it is and the remaining 3 are calculated. This is done for all the samples.
so yes its a good thing to atleast keep the first original sample point and it should be done. Not doing it would be something bad and that is what probably the older DAC might be doing and hence Schiit might have highlighted that their filter keeps the original values intact.

In case of non-integer sampling... e.g. 44.1 to 48... the original data point will tend to get overwritten with a new calculated one.
 
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Mathematically, it can be proven using the Nyquist-Shannon theorem that 44.1 kHz sampling is enough to reproduce the recorded sound completely.

My understanding is upsampling is done to move the aliases far away from the desired audio band, so that a gentle filter can be used to filter out unwanted aliases. Otherwise, for example, if we were to filter off anything above 22050 Hz as required for the basic PCM sampling rate, it would require a brick filter with a very steep roll off slope. A steep roll off increases the complexity of the filter while also introducing its own intermodulation products and messing up the phase (any resonance produces phase inversions). So upsampling isn't such a bad thing. Will upsampling produce better sound? Well, that's still open for debate.
 
AFAIK, Schitt multibit dacs oversample. They don't upsample.

Upsampling is done after the signal is sampled. Oversampling is done before the signal is sampled. They both sound very different
 
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Hi friends,

Recently i used async usb with my schiit dac. The source was ipad and also android phone ( with the app usb audio player pro, so that native usb driver of android which is 48khz is bypassed for bit perfect output of 44.1 ).

In general i noticed that usb playback sounded a bit thin compared to optical.

In some songs usb was sounding better and in some songs i preferred the optical.
Also i felt some difference between usb from ipad and usb from android.

Not sure if the "Thin sound" is due to the edge especially in the higher frequencies which can make the sound perceived as thin.


Sent from Note5

I have always found Optical to sound more "warmer" an with less edge than SPDIF or Coax connectors. maybe because it is less susceptible to RFI and as long as the converter from the electrical to optical and vice versa is of good quality you have hardly any corruption of the signal due to RFI.
 
I have always found Optical to sound more "warmer" an with less edge than SPDIF or Coax connectors. maybe because it is less susceptible to RFI and as long as the converter from the electrical to optical and vice versa is of good quality you have hardly any corruption of the signal due to RFI.

I have always found Optical to sound more "warmer" an with less edge than SPDIF or Coax connectors. maybe because it is less susceptible to RFI and as long as the converter from the electrical to optical and vice versa is of good quality you have hardly any corruption of the signal due to RFI.

Agree...My understanding is, though I may be grossly incorrect:

When optical is used, the DACs do not do any kind of massaging the data either via upsampling or either via filter. So the end sound then only depends upon two things: a) the hardware/chip of transport which is transmitting the optical data b) The DAC chip of the external DAC and its output stage.
So when we pass 44.1 to the DAC, it will process it at the SAME rate.


However when we use USB, the manufacturers are free to do almost all kind of massaging the data via upsampling or filters. Though the transmission is asynchronous and bitperfect, the mathematics fiddle the data and bring in some sort of artificialness or goodness. So even if we pass 44.1, the DAC may upsample to another rate e.g. 176.4 and output based on that rate.

Regarding the harshness, since USB adds it own data, the amplitude of high frequencies gets increased during the computation and brings in those kind of artifacts. This does not happen in optical since mathematics is not injected in the processing.
 
When optical is used, the DACs do not do any kind of massaging the data either via upsampling or either via filter

Ummm no. I had posted this clearly upthread (and that was a quote from the designer of the Schiit DACs)

So when we pass 44.1 to the DAC, it will process it at the SAME rate.

Wrong again. I had posted from the Schiit product page - the manufacturers pages says otherwise.


However when we use USB, the manufacturers are free to do almost all kind of massaging the data via upsampling or filters. <snip>This does not happen in optical since mathematics is not injected in the processing.

No. That is not what happens.

ciao
gr
 
The oversampling and filter will behave in the same way irrespective of whether you use optical, coax or USB. It's just that each digital input has its limitations in terms of the signal it can receive

The difference you are listening is because optical is way more electrically immune compared to USB
 
The difference you are listening is because optical is way more electrically immune compared to USB

In my case the difference would be related to transport.
For optical I am using ipad + airport express. So the hardware/firmware in airport express + apple can make the sound alter.

For USB I checked via apple camera connector kit from the ipad.(I also checked with laptop, andrioid aswell)

However I think the importance given to USB electrical isolation / etc compared to optical is so much hype. A USB within standards will work fine. I have tested for myself and there does not seem to be any loss of packets. My experiment may not be full fledged but I personally feel there is undue importance given to USB electrical isolation...
 
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The oversampling and filter will behave in the same way irrespective of whether you use optical, coax or USB.

Well quite possible.. but if the DAC has 44.1 osciallator / clock, then that clock is only used for receiving 44.1 via optical / coaxial and then later the higher clock (e.g. 176.4) does the actual output ?

If the DAC is only USB input, and DAC is supposed to output in 176.4, then I don't think 44.1 clock is required inside the DAC.

So for optical, the fact that 44.1 clock is used for getting the synchronous input, is also used for output, rather I should say it should be used for output aswell. That is what I can think of.
 
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Hi,

Some stuff you might like to look over. These were originally written by the designer of the Schiit DACs - Mike Moffat

What is this magic filter

An SOF (Schiit only feature) The Schiit Footlong Mega Burrito Supersauce Digital Filter:

It is a digital filter/sample rate converter designed to convert all audio to 352.8 or 396KHz sample rates so that it may drive our DACs. You get it from us; it is our filter. It keeps all original samples; those samples contain rudimentary frequency and phase information which can be optimized not only in the time domain but in the frequency domain.​

So what does it do ?

The below are the claims of the Digital Filter/Interpolator/Sample Rate Converter in the Yggy:

1. The filter is absolutely proprietary.
2. The development tools and coefficient calculator to derive the above filters are also proprietary.
3. The math involved in developing the filter and calculating has a closed form solution. It is not an approximation, as all other filters I have studied (most, if not all of them). Therefore, all of the original samples are output. This could be referred to fairly as bit perfect; what comes in goes out.
4. Oversimplified, however essentially correct: The filter is also time domain optimized which means the phase info in the original samples are averaged in the time domain with the filter generated interpolated samples to for corrected minimum phase shift as a function of frequency from DC to the percentage of nyquist - in our case .968. Time domain is well defined at DC - the playback device behaves as a window fan at DC - it either blows (in phase) or sucks (out). It is our time domain optimization that gives the uncanny sonic hologram that only Thetas and Yggys do. (It also allows the filter to disappear. Has to be heard to understand.) Since lower frequency wavelengths are measured in tens of feet, placement in image gets increasingly wrong as a function of decreasing frequency in non time domain optimized recordings - these keep the listener's ability to hear the venue - not to mention the sum of all of the phase errors in the microphones, mixing boards, eq, etc on the record side. An absolute phase switch is of little to no value in a non time domain optimized, stochastic time domain replay system. It makes a huge difference with an Yggy
5. This is combined with a frequency domain optimization which does not otherwise affect the phase optimization. The 0.968 of nyquist also gives us a small advantage that none of the off-the shelf FIR filters (0.907) provide: frequency response out to 21.344KHz, 42.688KHz, 85.3776KHz, and 170.5772KHz bandwidth for native 1,2,4, and 8x 44.1KHz SR multiple recordings - the 48KHz table is 23.232, 46.464, 92.868, and 185.856KHz respectively for 1,2,4, and 8x. This was the portion of the filter that had the divide by zero problem which John Lediaev worked out in 1983, to combine with #4 above AND retain the original samples.

This is what the competition offers:
5. Frequency domain optimization FIR filters with Parks-McClellan optimization. The development tools for these types of filters can be downloaded for a price range of free to $300 on the internet. Parks-McClellan is the goto filter optimization for audio design. These filters are derived with no closed form math; only successive approximation. The original samples are lost. The output is approximated. An educated guess. This optimization is ubiquitous in the front end of delta sigma dacs as well as standalone digital filters. While there is no inherent phase shift within Parks-McClellan filters, there is no optimization of phase either. The listener is left with what remains from the mixing boards, transducers, brick-wall filters, etc which can and usually do destroy proper phase/position information. Finally, it is processor efficient and economical to implement. Read cheap.

Any avoidance of the Parks-McClellan pablum requires a lot of original DSP work. Am I a prophet who received the tablets from God or some other high-end audio drivel. Hell, no. I was the producer and director of this project and worked with Dave Kerstetter (hardware-software), John Lediaev (Math), Tom Lippiat (DSP Code), Warren Goldman (Coefficient Generator and development tools) for a total of 15 or so man years. These folks either taught math at The University of Iowa, Computer Science at Carnegie-Mellon University, worked at think tanks like the Rand Corporation you get the idea. We did this for no money - What we all had in common was that we loved audio. All other audio pros were interested in Parks-McClellan and pointed and laughed at us. That's the way it happened.
It was worth it, every hour, day, and year. So go for it if you want. For what it is, it is not a lot of money.​


ciao
gr
 
I think the importance given to USB electrical isolation / etc compared to optical is so much hype. A USB within standards will work fine. I have tested for myself and there does not seem to be any loss of packets. My experiment may not be full fledged but I personally feel there is undue importance given to USB electrical isolation...

These might be worth a read.
http://www.hifivision.com/reviews/59218-uptone-audio-usb-regen.html
http://www.hifivision.com/dac/58313-chord-2qute-dac.html
 
Agree...My understanding is, though I may be grossly incorrect:

When optical is used, the DACs do not do any kind of massaging the data either via upsampling or either via filter. So the end sound then only depends upon two things: a) the hardware/chip of transport which is transmitting the optical data b) The DAC chip of the external DAC and its output stage.
So when we pass 44.1 to the DAC, it will process it at the SAME rate.


However when we use USB, the manufacturers are free to do almost all kind of massaging the data via upsampling or filters. Though the transmission is asynchronous and bitperfect, the mathematics fiddle the data and bring in some sort of artificialness or goodness. So even if we pass 44.1, the DAC may upsample to another rate e.g. 176.4 and output based on that rate.

Regarding the harshness, since USB adds it own data, the amplitude of high frequencies gets increased during the computation and brings in those kind of artifacts. This does not happen in optical since mathematics is not injected in the processing.

You may be right. but to really see the effect of upsampling/oversampling you will need to keep the output medium constant else will be very very difficult to isolate the effect of each
I will not be surprised if the difference you find is due to effect if lack of propose electrical isolation than the handling of data in itself as those differences are far more subtle than what you are mentioning...especially around sources like iPad/Android which in itself will have noise which is why people invest so much in the Music server and make it as isolated and focused on music as possible.

However I think the importance given to USB electrical isolation / etc compared to optical is so much hype. A USB within standards will work fine. I have tested for myself and there does not seem to be any loss of packets. My experiment may not be full fledged but I personally feel there is undue importance given to USB electrical isolation...
Personally i have experimented enough and also made my on DIY cable and found USB to be very dependent on the type of cable as well ..not the brand, but the topology. eg those where the power lines are isolated from the data is far superior in noise isolation
 
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I have been using the metrum octave Mk 2 for quite sometime now.
My experience;
As far as transports like CD/DVD (marantz cd 6005 /pioneer DVD player) vs the USB (dell inspiron mini) go, USB sounded the best.I suspect a top class CD transport will outperform the USB ,but they are expensive.More jitter in average CD transports as compared to M2 tech USB convertor inside the metrum
may be the case.
I also suspect power supplies have something to do in the USB conversion process since I have compared the usb input in the marantz cd player (yes, it has one and its possible to take a digital coaxial/optical out to an external DAC) to the metrum USB input. The metrum has better power supply system and hence sounds better.
cheers.
 
I had ordered the modi multibit too for myself and I was disappointed too. However the reason for my disappointment was that this and other schiit multibit dacs produce a lot of 'extra energy'. The upsampling etc gives the effect of smooth sound.

@Amit11 check whether you are disliking the smooth sound. I liked it. But due to the extra energy that i mentioned, I was stopping listening to songs after i had heard 4-5 songs.

I doubted if this was energy captured as band performs live (equipment energy etc).
On contacting schiit it was confirmed. But they were too proud of the fact that they were reproducing 'sound as it was recorded'. They did not care if it was causing too much energy reproduction or whether such thing is harmful for people in general. I even doubted if such energy capture is possible but no reply from their side even after complaining to cus service deptt.

Regarding that NOS DAC...check youtube...you can hear a different kind of sound there.
 
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