Follow along with the video below to see how to install our site as a web app on your home screen.
Note: This feature may not be available in some browsers.
Welcome to ASR. There are many reviews of audio hardware and expert members to help answer your questions.
Click here to have your audio equipment measured for free!
Hardware improve. I would say I now have the bass that class D amps usually provide. I know, it is incredible but it is true. And with a second-hand AVR that cost me less than € 120 put at home! Much more bass and fast. It looks like it has other loudspeakers, much bigger.
Hardware improve. I would say I now have the bass that class D amps usually provide. I know, it is incredible but it is true. And with a second-hand AVR that cost me less than € 120 put at home! Much more bass and fast. It looks like it has other loudspeakers, much bigger.
And how an amp can provide "better" bass? Doesn't that mean that it could be adding something such as distortion or tone controls?
In Spanish: ¿Cómo un amplificador puede añadir frecuencias al menos que no sea por distorsión o control de tono? Es raro que opines que un amplificador puede cambiar tanto las características del sonido.
It is true that we perceive sound intensity logarithmically. An increase in SPL of 10 dB subjectively sounds about twice as loud. Plotting the time-domain waveform with a logarithmic scale doesn't make sense, though. Simply put, the logarithmic part of hearing takes place after the sound has been separated into frequencies. That's why we usually use logarithmic scales for spectrum plots.
It is true that we perceive sound intensity logarithmically. An increase in SPL of 10 dB subjectively sounds about twice as loud. Plotting the time-domain waveform with a logarithmic scale doesn't make sense, though. Simply put, the logarithmic part of hearing takes place after the sound has been separated into frequencies. That's why we usually use logarithmic scales for spectrum plots.
Another thing. I've often heard the pro-min phase people say that speakers and headphones are minimum phase devices, so it doesn't make sense to use linear phase filters.
If we look at headphones (easier because it's a single driver) and linear phase filters, is the driver attempting to play these pre-echo signals coming out of the DAC?
Another thing. I've often heard the pro-min phase people say that speakers and headphones are minimum phase devices, so it doesn't make sense to use linear phase filters.
If we look at headphones (easier because it's a single driver) and linear phase filters, is the driver attempting to play these pre-echo signals coming out of the DAC?
For testing purpose you bypass the required bandwidth limiting ? that's what i understood about it . One can argue if similar signals appear in post process (ADC at inputs will bandwidth limit ) well written DAW plugins to , but finally the sample rate conversion to your CD or download file ?
For testing purpose you bypass the required bandwidth limiting ? that's what i understood about it . One can argue if similar signals appear in post process (ADC at inputs will bandwidth limits ) weel written DAW plugins to , but finnally the sample rate conversion to your CD or download file ?
It uses/assumes an artificial signal with infinite bandwidth, without any limiting. You will not get that kind of signals from post processing or DAW plugins.
It uses/assumes an artificial signal with infinite bandwidth, without any limiting. You will not get that kind of signals from post processing or DAW plugins.
Another thing. I've often heard the pro-min phase people say that speakers and headphones are minimum phase devices, so it doesn't make sense to use linear phase filters.
If we look at headphones (easier because it's a single driver) and linear phase filters, is the driver attempting to play these pre-echo signals coming out of the DAC?
If you’re doing EQ in the audible range then you can hear a difference between linear and minimum phase. It’s easy to demonstrate this, here’s an example.
From what I’ve read, when EQing speakers you use min-phase filters as a way of inverting the minimum phase errors in the speakers’ frequency response, thus cancelling out both FR and phase error.
None of this has any relevance to over-sampling filters used in DACs, though.
Abuot raedibaltiy: I awlyas touhgt tihs was fnuny:
It deosn't mttaer in waht oredr the ltteers in a wrod are, the olny iprmoetnt tihng is taht the frist and lsat ltteer be at the rghit pclae. The rset can be a toatl mses and you can sitll raed it wouthit a porbelm. Tihs is bcuseae the huamn mnid deos not raed ervey lteter by istlef, but the wrod as a wlohe.
RIAA equalisation should be minimum phase. But you need a pretty steep filter slope to produce enough ringing to be audible, which is probably why people can get away with linear phase. Also, well, we’re talking about vinyl here ...
[Edit] Oops, replied before I noticed your name. I’m sure you know a lot more about RIAA than I do.
in the pro world
