tim3320070
Headphoneus Supremus
Plus you most likely cannot hear the difference between 192 and 96 (or 44.1 for that matter). It's very subtle with my full-on system let alone the NFB-12.
Upsampling 16/44 to 24/96 or 24/192 is a good thing. It doesn't reduce the quality, and if it does then it's far from audible.
Transcoding 192kbps to 320kbps is a bad thing. This is because transcoding a lossy format always reduces quality. MP3 is lossy, stuff such as FLAC is lossless, in which case transcoding does not change the audio quality in any way whatsoever. Lossy codecs such as MP3 throw away some of the audio information that is present on lossless files such as FLAC to save more space. With every transcodation they throw away a bit more information, that's why transcoding lossy formats is always bad.
Upsampling is fundamentally different from transcoding. Transcoding involves changing from one way of encoding (i.e. compressing) the audio. Upsampling on the other hand just changes the detail by which the information is stored. No information is thrown away, but rather information is added. This added information is basically just describing audio signal, but by using more words.
For some reason DAC's such as the NFB-12 respond better to upsampled signals. Therefore there is no real reason not to upsample the signal, since it's only gonna improve the sound quality at the end of the chain.
With foobar2000 adding upsampling to the output is as simple as adding the standard resampler DSP.
File -> Preferences -> Playback -> DSP Manager
Then add 'Resampler (PPHS)'. Select it and click 'Configure selected' after adding.
Type in either 96000Hz or 192000Hz depending on whether you use USB or SPDIF. USB can only support up to 24/96, SPDIF can do 24/192 as well.
You then might have to enable 24/96 or 24/192 in Windows for the NFB-12.
Control Panel -> Sound. Then double-click the audio device corresponding to the NFB-12. For me the standard name was 'TE7022 Audio w/ SPDIF'.
Then hit 'Supported Formats' and check all the boxes for sample rate.
Then go to 'Advanced' and select '2 channel, 24 bit, 192000 Hz (Studio Quality)' or whatever sample rate you want to use.
Now it should work. You might not hear any differences, but theoretically they should be there. But hey, aren't most of us audiophiles acting on placebos and biased listening in at least some way?
Don't confuse sample rate conversion with compression level conversion.
When we talk about up sampling or oversampling, these are ways to take content with a 44.1KHz sample rate and "resample it" to make it look like the data was captured at a higher sample rate of 96, 192 etc. This allows the digital filter cutoff frequency to be implemented so far above the top audio frequencies that it ensures you will avoid aliasing (I'll cheat and simply call this a bad form of "distortion). You see in the RMAA up sampled to 96/96 chart above that aliasing is gone, whereas it is clearly visible in the 44 sampled at 96 (you can see it as the fading mirror image above the curves). This lets you use digital filters that have way less effect on the signal, and also eliminates the need for analog "brick wall" filters at 20KHz, which also improves performance.
Compression level is totally different than sample rate. Pretty much all 192, 256 or 320 content is SAMPLED at 44.1KHz. What the 192kbps is is essentially how much data per second is left after the file has been compacted using LOSSY compression. Since all these formats are lossy, if you take a low quality 128 or 192 and "recompress it" at 320, you're actually going to lose more of the audio signal and degrade the sound further, as every pass through a lossy compression algorithm will by definition throw out some data.
I use Pure Music and upsample all my 44.1KHz content, whether it's 256kbps or fully lossless, to 88.2KHz. This
Thanks guys, that helped a lot! I'm new to these FLAC stuff, don't even have an audiophile level headphones yet. But you know what they say, upgrade your source before upgrading headphones
Well I don't know about that. An LCD-2 driven from a front port will still sound better than iBuds driven from a Leben CS-300x. (The LCD-2 might be incredibly soft, though)
After the recording itself, headphones/speakers are always the most important component in the chain. Source and amplification are only there to compliment the headphones.
Just posted some suggestions for Mike to try. The wrong digital filter setting can give a bad impression, as can Low gain setting. Hopefully he can try a new setting and post impressions. I wonder if it's burned in too.
Was just looking at that before coming here ahaha. From the looks of it, Mike's not online though I was ready to look for a better alternative, but now shall wait for Mike to do a re-review
It doesn't look like he's inclined to do a re-review based on his response to my comments - I'm sure he gets lots of requests, so I understand. . .But it leaves a perhaps unfairly earned black eye on this unit.
If he was reviewing with the 8x linear half-band as the photo implies, that's not the default digital filter setting on either the earlier units which weren't switchable or the current units with the 9-way adjustable filter. So just bad luck that the unit was reviewed with the worst setting (to my ears), and it wasn't at all a subtle difference if you look at my earlier post/review. The 8x linear half-band filter only goes up to 48kHz, so it wouldn't ship out that way. The default/shipped setting on my units was 2x linear phase brickwall (supports up to 192kHz).
Anyway, for prospective buyers, there's definitely upside/better SQ settings than what was reviewed.
With Mike's settings, he says the only good thing about it is the bass. Do you think if he changed the settings to the one you mentioned, the other parts will also improve significantly?If you have the time, would you try and set yours to the settings Mike used? I'd like to see if you'd also get the same results Mike did. So I'll know that the settings is probably why Mike hated his and that it can be changed. Also, can you state what improved with your settings? Thanks!
Check post#1403 - I already tried the settings Mike used and compared that with my preferred setting. If you have more questions after reading that post, just let me know. But basically the detail, texture, etc was much better with the 8x min apodising vs the 8x half-band.
Different people, different cans, different ears. NFB-12 sounds great to me with HD600, DT880, and also MS1i. That's all that matters. To my ears, it also sounds much better than my old E7/E9 combo (now sold). I agree with CapTouch that the default setting from Audio-gd is not the best. But that's why you have filters - so you can change them to your own personal preferences.
Mike does some great reviews - but some I agree with and some I don't. This one I don't. I'll carry on enjoying my NFB-12 - that's allt hat really matters in the long run - what I personally think of it.
yeap, just read it. Now I don't wanna buy this anymore