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[Official] OCN Headphones and Earphones Club - Page 340

post #3391 of 37823
Quote:
Originally Posted by s1rrah View Post
We should start a 12 step program or something.

Even that might not help.
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post #3392 of 37823
Hey guys, what's the general consensus on the Senneiser RS / Other series of cordless 'phones?

Can't imagine them being awesome myself, but am drawn the the lack of wires for some reason.
post #3393 of 37823
Quote:
Originally Posted by wigseryc View Post
Hey guys, what's the general consensus on the Senneiser RS / Other series of cordless 'phones?

Can't imagine them being awesome myself, but am drawn the the lack of wires for some reason.
I believe they're good as far as wireless headphones go, but that's not very far.
post #3394 of 37823
Didn't think so. Weighing up my options see. I'm thinking about looking for a new set of cans, preferably full sized ones that cover the entire ear.

The search continues.
post #3395 of 37823
@ X-fi owners:

...

As you may know already ... I use my X-fi to send a digital signal to my desktop DAC.

When messing around with the X-fi control panel, I notice I can change the master sampling rate from 44.1khz up to 96khz.

...



...

Each time I do this, my DAC's synch light flickers and then locks. I'm guessing it has to re-synch/re-lock the signal each time I make a sampling rate change.

What I'm wondering is whether this is, essentially the same as introducing an upsampler into the chain of devices.

I swear there is a marked difference in the sound when I change, mid-song, between 44.1khz and 96khz. The 96khz setting sounds a bit more high frequency heavy. A bit sharper. Not necessarily better or worse, just more of an edge to things. Maybe I'm just imagining it, dunno.

So is this basically an upsampler type capability of the X-fi?

I was using Foobar with the Secret Rabbit code software upsampler which I found to be an interesting DSP plug in to play with.

Is this X-fi setting doing the same thing?

Just curious.
    
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post #3396 of 37823
Yes - you're basically introducing an upsampler into your audio chain. That could be particularly handy when using DSP or other processing, as the DSP can be applied more times within a given segment of the song (often mitigating the overly flat sound that can be a telltale of digital processing).
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post #3397 of 37823
Quote:
Originally Posted by Chipp View Post
Yes - you're basically introducing an upsampler into your audio chain. That could be particularly handy when using DSP or other processing, as the DSP can be applied more times within a given segment of the song (often mitigating the overly flat sound that can be a telltale of digital processing).
Thanks.

BTW: could you describe a brief scenario as an example of the bit you mention regarding the DSP being "applied more times"? Sounds interesting but I confess to not knowing a whole lot about the digital music thing.
    
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post #3398 of 37823
Khz is waves per second - the signal from the original source is sent more times per second (96,000 instead of 44,100). Since the source is only 44.1k, there will be no audible difference if all other factors are kept the same. This is because you are basically making duplicates of every few waves because in the original file, there is only so much information. I will try and make a diagram that explains that better, I know I did a terrible job.

So, when you are applying some DSP (let's just say a reverb to keep it simple) your reverb effect will be applied 96000 times per second instead of 44100 times. Though the music is not changed and will sound the same as the lower bandwidth file, the reverb will be more uniform on the high bandwidth file because it is being applied to smaller pieces of the original recording.
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post #3399 of 37823
Quote:
Originally Posted by Chipp View Post
Khz is waves per second - the signal from the original source is sent more times per second (96,000 instead of 44,100). Since the source is only 44.1k, there will be no audible difference if all other factors are kept the same. This is because you are basically making duplicates of every few waves because in the original file, there is only so much information. I will try and make a diagram that explains that better, I know I did a terrible job.

So, when you are applying some DSP (let's just say a reverb to keep it simple) your reverb effect will be applied 96000 times per second instead of 44100 times. Though the music is not changed and will sound the same as the lower bandwidth file, the reverb will be more uniform on the high bandwidth file because it is being applied to smaller pieces of the original recording.
Thanks for the breakdown, Chipp ... makes perfect sense.

I'd rep you if I could.

...

On another note:

Since getting my Paradisea DAC, I've wondered why on earth my particular version has 2.2uf output caps when MHDT Labs states quite clearly on their website that the early versions of the Paradisea shipped with 1.5uf caps and the newer versions are shipping with 2.0uf output caps.

In fact, one of the main marketing tools MHDT uses to pimp the newer Paradiseas is the upgrade from 1.5uf to 2.0uf output caps.

So whyfore my uber fat 2.2uf output caps?

...





...

So earlier I emailed the builder of my DAC with some questions.

Much to my surprise, he emailed back almost immediately:

...

Quote:

Hi Joel,

Output cap can be 0.47uf ~ 10uf

Few Paradisea install 2.2uf, mostly 2.0uf.

The larger value of output cap results in more apparent low frequency response (bass).

Some will think 2.2uf is adequate; some think it is to much bass.

Some even use 10uf.

Finally, I decide to use 2.0uf.

Thanks!!

Best Regards,
Jiun-Hsien
Mhdt Labs
...

Well ...

For what it's worth.

I think the 2.2uf output caps are *PERFECT* in so far as bass presence. Especially with my RS1's!

Very taught and natural bass and also very up front and accessible without any EQ'ing necessary.

...

Anyway ...

It's fun when a bit of gear offers up some mysteries!

    
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post #3400 of 37823
its always nice to get a response, let alone a quick and detailed response.
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