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A Couple of Questions about Amps

post #1 of 9
4/4/11 at 10:22am
Thread Starter 

I've been following a couple of threads on the Schiit Lyr. One thing that's come up over the last couple of days is how different the Lyr sounds from other amps, particularly SS ones.

 

I can understand why tube amplifiers sound different from SS amplifiers, the design of circuit topologies etc. all makes sense. What I'm not understanding is what people mean by one amp sounding faster than another. With phones, there's a reason for a faster response, but for amps, how is that supposed to happen? Could someone explain this?

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post #2 of 9
4/4/11 at 10:35am


 

Quote:
Originally Posted by rroseperry View Post

I've been following a couple of threads on the Schiit Lyr. One thing that's come up over the last couple of days is how different the Lyr sounds from other amps, particularly SS ones.

 

I can understand why tube amplifiers sound different from SS amplifiers, the design of circuit topologies etc. all makes sense. What I'm not understanding is what people mean by one amp sounding faster than another. With phones, there's a reason for a faster response, but for amps, how is that supposed to happen? Could someone explain this?

 

It is mostly a nonsense term relating to an unverified subjective evaluation, ask the reviewer how exactly they measured the speed.

 

Amps do have a measurable speed it is called the rise time, the time it takes to get from 10% signal amplitude to 90% , there is also the related measurement called the slew rate, which is the amount of voltage change over time normally measured in V/µs but that is not what the reviewer is talking about.

 


Edited by nick_charles - 4/4/11 at 10:47am
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post #3 of 9
4/4/11 at 11:46am
Thread Starter 

Thank you for the answer. Somehow I'm not surprised.

 

Oh, btw, if you ever want that analysis of your cable data, let me know.

 

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post #4 of 9
4/4/11 at 11:57am
Quote:
Originally Posted by rroseperry View Post

Thank you for the answer. Somehow I'm not surprised.

 

Oh, btw, if you ever want that analysis of your cable data, let me know.

 


Sadly I discovered that I deleted the raw data, I thought the thread had run its course, the charts are all that is left now and I do not have the time to recreate my tests...

 

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post #5 of 9
4/4/11 at 12:15pm
I have a question about DAC/Amps too, and I think this is a better place to post it rather than create my own thread.

I'm wondering about upsampling. if my source audio file is 16 bit 44.1khz and I upsample to 24/192khz, shouldnt that make absolutely no difference? For example, if a file is sampled at 1hz, ie once every second the computer figures out what the audio data is and represents its value using a single on off switch (1 bit). Then I tell the computer to pass that once per second data to a device that upsamples to 2 bits, 2 hz.

Suddenly, that device CAN provide 4 different output levels and it can check for what that output level is TWICE every second, however , every second the computer will tell that device the same piece of data at either 0 or 1. that value will then be converted to either 0 or 3 (0% or 100% output) by the upsampling device, and instead of giving this 0 or 100% value once every second, it will give that value twice every second. Technically, you have done nothing to the original signal except make it represented a bit redundantly.

So what, really, is upsampling and why is it considered a "good thing" if at all? I figured that somethign that upsamples will do nothing but cause added distortion
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post #6 of 9
4/8/11 at 10:30pm

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post #7 of 9
4/9/11 at 8:41am
Quote:
Originally Posted by shrimants View Post

I have a question about DAC/Amps too, and I think this is a better place to post it rather than create my own thread.

I'm wondering about upsampling. if my source audio file is 16 bit 44.1khz and I upsample to 24/192khz, shouldnt that make absolutely no difference? For example, if a file is sampled at 1hz, ie once every second the computer figures out what the audio data is and represents its value using a single on off switch (1 bit). Then I tell the computer to pass that once per second data to a device that upsamples to 2 bits, 2 hz.

Suddenly, that device CAN provide 4 different output levels and it can check for what that output level is TWICE every second, however , every second the computer will tell that device the same piece of data at either 0 or 1. that value will then be converted to either 0 or 3 (0% or 100% output) by the upsampling device, and instead of giving this 0 or 100% value once every second, it will give that value twice every second. Technically, you have done nothing to the original signal except make it represented a bit redundantly.

So what, really, is upsampling and why is it considered a "good thing" if at all? I figured that somethign that upsamples will do nothing but cause added distortion

From a signal processing point of view you're right. Practically you cannot add new samples in between the old ones precisely at their theoretically exact positions. Even if an added sample is spot-on all what happened is that the signal now contains redundant information.

From a hardware point of view however things looks differently. By letting the DAC oversample the signal it is possible to drive a 1-bit DAC with a pulse-density modulated signal. The technique is called sigma-delta modulation. This results in higher linearity and lower cost as opposed to a simple PWM-type DAC.
500

The output is filtered with a low-pass. From this picture you can probably imagine that it is advantageous for the DAC to use a very high rate. (actually, this is called oversampling and not upsampling)
Edited by xnor - 4/9/11 at 8:56am
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post #8 of 9
4/9/11 at 5:12pm

Dan Lavry wrote a good post about the need for oversampling in digital to analog conversion.  He's an engineer so describes the reasons in wonderful detail.

 

The basic idea is that digital signals are converted to continuous/smooth analog signals by a low-pass reconstruction filter.  As described by the Nyquist theorem, an oscillatory signal can be reconstituted from individual samples at only 1/2 the frequency of the samples.  So to not lop off the top 1/2 frequencies of the original waveform, 2x oversampling nicely solves the problem of preserving information that would otherwise be lost in the in the reconstruction of continuous sound waves from a digital signal. 


Edited by eucariote - 4/10/11 at 6:17am
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post #9 of 9
4/11/11 at 9:53am
this is rather good news for me since I am trying to build the Gamma 1 and Gamma 2 so that I can get some USB output from my computer to my A5's.
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