daze84 said:
Just setup my B&W's btw, not played em yet..
You still got your rotel amp(s)? Dunno which model to get
i use two [FONT=Arial, Helvetica, sans-serif]
RB 1091[/FONT] mono amps.
the RB 1030 might be perfect for compact speakers, but the bigger the better. in any way, even smaller models (rotel) will still knock any kind of "consumer" amp you've heard before.
73* said:
After setting them up and calibrating them to the K-System, I sat my girlfriend down and played a track off of Raising Sand and about halfway through she says "This is creeping me out, it sounds like Robert Plant is right there!" (while pointing between the speakers). So glad I was able to get these speakers...
yeah, REAL stereo! not just left and right (i think you know what i mean..
)
Emmapeel9 said:
I think there is a german ME who thinks it is more of a marketing gimmick with 96Khz and 192Khz so there are opposing views on this
this topic has 3 different "categories", you can talk about recording, processing or playback at different rates. the benefit of recording and playback at higher rates than needed ("oversampling") is really, really small (especially with modern ADACs) and has many drawbacks (more noise, more memory needed, jitter, etc).
but an important detail is why modern converters allow you to capture an excellent representation of the incoming signal at a sample-rate of 44.1kHz: ...they over-sample and down-sample the signal internally.
so, it's all about the technical depth of the discussion.
audio processing on the opposite is clearly "high rate friendly". the difference between cubases cheesy build in EQs and a beautifull sounding EQ from UAD is oversampling. just render your project at 192kHz and downsample it with a great sample-rate-converter (check the freebie from voxengo) back to 44.1kHz and compare it to a direct 44.1kHz rendering. the difference is bigger than you might think. especially synths sound much better when you give them more "room".
never forget that you're working in the digital domain. frequencies above the bandwidth limit (nyquist frequency) don't "smooth out" like in an analog system. they are fully mirrored back (!) into the spectrum, this is "aliasing". aliases of the original signal are pushed back into the spectrum, but in a highly unharmonic way. that's the main reason why synths, comps and saturation sound much cleaner and "creamier" at higher rates, their algorithms create much less aliases at higher rates.
just an example, creating a 0dB tone at 44 kHz in a 44.1kHz sample-rate system (22.05kHz nyquist freq.) will in fact result in a 0dB 100Hz tone. we have a hyper-thin peak beeing transformed into a big bass tone! now imagine what happens when adding series of even and odd harmonics (multiples of the incoming signal) to a full mix through slight saturation... ...there will be a lot unharmonic trash in the resulting signal. trash we can minimize through oversampled processing.
just have a sample-rate search for more
