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Hello,
I already asked this question in different topic and solved my issue with cheap mixer. Still for the future I would like to know your opinion. Which of the solution A or B is better regarding:
1.) Latency
2.) Sound quality

I think sound quality is probably the same, but latency should be very slightly lower with solution A, as there is "one less traveling" on USB-USB cable between USB audio device and PC.
Thanx

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I get less latency with A; that has been my experience.


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If you are using virtual instruments (plug ins/vst's) latency depends a lot on:

1.- Quality of USB interface drivers (the little program that handles communicaiton between interface and PC).
2.- Audio settings.- Buffer size(the lower the better, if you CPu can handle it), Sample rate (44,100 is good enough, more is just not really that audible).

In both set ups you have MIDI has to travel to the PC first, so it really doesnt matter which way it gets there. The only significant difference would be if the USB MIDI drivers of the digital piano (if it uses any) are very bad which is quite rare.

If you are listening to the DP audio out then latency depends on the same things except that some audiointerfaces have the option for direct monitoring so the audio you listen doesn't go through the PC.

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Sound quality should be the same. Latency could be higher in B, but it is also possible that there isn't a distinguishable difference in latency. Most latency has to do with the audio processing and output in the computer, not so much the MIDI. Of course, that could depend on how well your audio interface is at passing MIDI through.

Last edited by gvfarns; 01/22/14 07:46 PM. Reason: Accidentally wrote A when I meant B
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My set-up is like your "A" schematic - except that I use a mixer in place of the USB audio device. It works well for my purposes.

But I'm curious to what advantages the USB audio device would have over a mixer in this case?


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I'd go for A with one modification: sending the digital signal from your PC to the "USB" audio device (DAC) via an optical link rather than via USB, if the device has such an input.

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Originally Posted by gvfarns
Sound quality should be the same. Latency *could* be higher in A, but it is also possible that there isn't a distinguishable difference in latency.

Why do you think so ? In B version there is one "cable travel" more from DP to PC.

Originally Posted by newbert

But I'm curious to what advantages the USB audio device would have over a mixer in this case?

If you have crappy built-in sound card in laptop usb interface gives you better sound quality usually.


I have one question regarding sample rate as it has a big impact on latency. When you are choosing sample rate in VST, what influences maximum value ? Is it the sample rate of audio interface ? For example with Steinberg UR22 I can go up to 192khz, while with Presonus USB up to 48khz only.


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I tried "A" before but got ground problems going over the USB MIDI connection between the DP and the computer. Now I go with "B" without any latency or ground problems.

I also didn't like having my USB ports blocked in version"A".

Last edited by Cmin; 01/22/14 03:41 PM.

Cheers,
Lenny

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I used "C" (not shown) back when I didn't own an outboard audio interface. The piano has a USB-to-host port, so I used that ... USB piano to USB PC.

When I bought an audio interface I switched to "B".

But regardless ... the MIDI data is traveling over USB to get into the PC.
In "A", the MIDI-to-USB converter carries the data over USB.
In "B", the audio interface does the MIDI-to-USB conversion.
In "C", the audio goes USB all the way.

In all three cases you have accept the PC's USB latency, which is the biggest contributor to interface latency. So choose any of these connections without concern. If latency is a problem, look at the PC for a fix.

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Originally Posted by jarosujo
Originally Posted by gvfarns
Sound quality should be the same. Latency *could* be higher in A, but it is also possible that there isn't a distinguishable difference in latency.

Why do you think so ? In B version there is one "cable travel" more from DP to PC.


That was a typo on my part. I mean to write B. Correcting and leaving a little note about why. Sorry.

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jarosujo, perhaps you can test both setups and post back with the latency readings?

James
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Originally Posted by Kawai James
jarosujo, perhaps you can test both setups and post back with the latency readings?

James
x

I don't have audio interface yet. At the moment I am using my main PC, which happens to sit right next to DP. It has PCI Sound Blaster X-FI sound card inside and I am very happy regarding latency (3ms in Pianoteq) and sound quality (much better when comparing to built-in sound card in cheap Atom laptop I have). However I might move my DP to living room so I will need to buy new PC (laptop probably) and then I will get Audio interface as well.
There are some which don't have MIDI ports, like
http://www.native-instruments.com/en/products/traktor/dj-audio-interfaces/traktor-audio-2/
That's why I was asking regarding connection, but as it seems, it is better to buy proper interface with MIDI as well and than you have more options in case something goes wrong with one type of connection.


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Originally Posted by jarosujo



I have one question regarding sample rate as it has a big impact on latency. When you are choosing sample rate in VST, what influences maximum value ? Is it the sample rate of audio interface ? For example with Steinberg UR22 I can go up to 192khz, while with Presonus USB up to 48khz only.


Yes the max value depends on the interface, some allow 44,1, 48, 96 and up to 128. But sample rate determines the audible spectrum , from wikipedia:

Quote
In digital audio the most common sampling rates are 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz and 192 kHz.[5] Lower sampling rates have the benefit of smaller data size and easier storage and transport. Because of the Nyquist-Shannon theorem, sampling rates higher than about 50 kHz to 60 kHz cannot supply more usable information for human listeners. Early professional audio equipment manufacturers chose sampling rates in the region of 50 kHz for this reason. 88.2 kHz and 96 kHz are often used in modern professional audio equipment, along with 44.1 kHz and 48 kHz. Higher rates such as 192 kHz are prone to ultrasonic artifacts causing audible intermodulation distortion, and inaccurate sampling caused by too much speed.[6] The Audio Engineering Society recommends 48 kHz sample rate for most applications but gives recognition to 44.1 kHz for Compact Disc and other consumer uses, 32 kHz for transmission-related application and 96 kHz for higher bandwidth or relaxed anti-aliasing filtering.[7]


So 44,100 is optimal.

Lower buffer size allows lower latency, but in order to be able to set it low (64 for example) you need to reduce the other cpu intensive settinga, like a high sample rate or 24 bit depth.


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I would not say that 44kHz is optimal. Optimal is as high as possible without any pops/dropouts. I use 96kHz in Pianoteq, but I would go even higher, but my X-FI sound card with native Creative ASIO doesn't allow more. With Pianoteq all you need is good CPU, I have Core i5 4GHz so it can handle low buffer and high sample rate (=low latency) without problem.
Regarding sound quality, I agree that without expensive high-end audiophile equipment you can't really hear much difference if going over 48kHz.


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Originally Posted by jarosujo
I would not say that 44kHz is optimal. Optimal is as high as possible without any pops/dropouts. I use 96kHz in Pianoteq, but I would go even higher, but my X-FI sound card with native Creative ASIO doesn't allow more. With Pianoteq all you need is good CPU, I have Core i5 4GHz so it can handle low buffer and high sample rate (=low latency) without problem.
Regarding sound quality, I agree that without expensive high-end audiophile equipment you can't really hear much difference if going over 48kHz.


You can't hear anything any difference above 48,000, it's out of human hearing range. http://en.wikipedia.org/wiki/Hearing_range

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When I talked to my audiophile friend he said something about frequencies over 48khz add some color to lower frequencies, kind of symphatetic resonance I guess. But these are maniacs who are willing to pay thousands $ just for power cable :-)


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Originally Posted by jarosujo
When I talked to my audiophile friend he said something about frequencies over 48khz add some color to lower frequencies, kind of symphatetic resonance I guess. But these are maniacs who are willing to pay thousands $ just for power cable :-)


The problem with "audiophiles" is that anyone can be one. All you have to do is be willing to buy audio equipment that other people will not. The result is that confirmation bias and post-purchase rationalization are exceptionally prevalent in this demographic.

Last edited by gvfarns; 01/23/14 12:22 PM.
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Well in the above quoted wikipedia article it states that the Audio engineering Society recommends 48,000 as a standard. Anything above is only usefull for virtuals instrument which use DSP (not samples) to generate sound in order to reduse aliasing, but still the sound generated out of the 20,000 hz wont be audible to humans.

Plug ins as Pianoteq can work at 96hz to produce it's sound (and reduce aliasing) but the system as a whole can still output at a lower sample rate with no degradation to sound.



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