**Hifiman HE-400 Impressions and Discussion Thread**
Nov 9, 2013 at 4:33 PM Post #14,282 of 22,116
Did a lot of changes over the last few weeks in EQ (that drove me mad for a while) but I seem to have hit on a setup that is working really well. These are the things I have done differently recently...
 
1).
For a long time I used Electri-q that I was quite happy with but then slowly started not to like anymore. I Moved from using Electri-q to using the equalizer built into AIMP3. This also does parametric eq but it uses sliders to produce the curve. I am now using a curve that is similar to my previous curve I used in Electri-q but it has more flat tops rather than rounded tops that has a more natural sound.
 
2).
I got sloppy somewhere along the way and stopped using the preamp volume in the eq to match any boost in dB because I couldn't immediately tell any difference and I was changing the eq so often, none the less its safer to engage the preamp. It is possible this was causing some distortion that was hard to detect because of the volume level change. The consequence of now using the preamp is that I had to increase the physical volume on the amp to get the same volume as before (went from 11 o'clock to 12 on the dial). I have always felt that the HE400 sounded better at higher volumes but this was not possible because it made my ears hurt after a few hours. It could be that the amp is now in more of a sweet spot with the volume set higher.
 
3).
I set AIMP3 to use 24bit/192kHz and set the DAC driver (Tenor TE8802) to also use this. I tried this due to this video I had seen (http://www.youtube.com/watch?v=JYjHKv2_OqQ). In another video in those seminars, one of the experts says that 16bit/44kHz source that is upscaled to 24bit/192kHz does sound better. I moved to using WASAPI Exclusive mode instead of just normal WASAPI so that AIMP3 allowed me to use 24bit/192kHz.
 
The result of all the above is that I now have a more natural sound that sounds very balanced and 95% of the time I detect no tizz. The mids don't sound recessed at all and the vocals are very clear and crisp. The mids and the vocals sound just right now.
 
The biggest difference is made by 1) but I have a feeling 2) and 3) has something to do with it also but this is harder to detect. It took a few days of listening to songs that I was very familiar with to confirm my initial impressions.
 
I had decided to move to a more expensive setup (one of HE500/LCD3/HD800 and new amp/dac) but now I will stick with the HE400/NFB15. It is sounding the best I have ever heard it.
 
Nov 9, 2013 at 4:44 PM Post #14,283 of 22,116
I totally agree about the detail! Extremely detailed headphone! The bass is very well defined, mid range is superb, and treble is sparkling! It's the end of the road. I still have my Sennheiser hd650 with after market cable, but hifiman he-400 is on the top of the heap! :)
 
Nov 11, 2013 at 3:32 PM Post #14,286 of 22,116
  Did a lot of changes over the last few weeks in EQ (that drove me mad for a while) but I seem to have hit on a setup that is working really well. These are the things I have done differently recently...
 
1).
For a long time I used Electri-q that I was quite happy with but then slowly started not to like anymore. I Moved from using Electri-q to using the equalizer built into AIMP3. This also does parametric eq but it uses sliders to produce the curve. I am now using a curve that is similar to my previous curve I used in Electri-q but it has more flat tops rather than rounded tops that has a more natural sound.
 
Personally I found that Electri-Q gives the best results if you get it right because you can lower only the frequencies that you need to. I think that all graphic equalisers do what you are saying... but the one you are using is just showing you it on the interface? In which case you might be better with a 31 band EQ because it gives you more accuracy while still being easier to use than ElectriQ.. The advantage of Parametric is that it allows you much better accuracy by changing the gain, bandwidth and centre of each frequency, seems like you have downgraded by going to a 16 band graphic EQ, but if you are 100% happy with the sound then does not matter, if you want to get better accuracy (which works especially with the HE400 due to the crazy FR)... Maybe give electriQ another shot or try a 31 band graphic EQ. Just get a few presets you are happy with on electriQ then avoid changing it any more, 1 standard EQ, 1 for low quality sources and whatever else, if you keep trying to tweak it for every song it will indeed drive you mad.
 
2).
I got sloppy somewhere along the way and stopped using the preamp volume in the eq to match any boost in dB because I couldn't immediately tell any difference and I was changing the eq so often, none the less its safer to engage the preamp. It is possible this was causing some distortion that was hard to detect because of the volume level change. The consequence of now using the preamp is that I had to increase the physical volume on the amp to get the same volume as before (went from 11 o'clock to 12 on the dial). I have always felt that the HE400 sounded better at higher volumes but this was not possible because it made my ears hurt after a few hours. It could be that the amp is now in more of a sweet spot with the volume set higher.
 
3).
I set AIMP3 to use 24bit/192kHz and set the DAC driver (Tenor TE8802) to also use this. I tried this due to this video I had seen (http://www.youtube.com/watch?v=JYjHKv2_OqQ). In another video in those seminars, one of the experts says that 16bit/44kHz source that is upscaled to 24bit/192kHz does sound better. I moved to using WASAPI Exclusive mode instead of just normal WASAPI so that AIMP3 allowed me to use 24bit/192kHz.
 
I think you are probably better leaving your media player at the exact bitrate which the recording is, for example 16/44100 90% of the time, let the DAC do any upscaling, you do not get any advantage from setting high bitrates in software, at best a load of zero's added to the output, at worse it might cause problems with the sound.... Anyway just saying that if you get the urge to improve it more, give electriQ another chance, it should give better results than a 16 band graphic EQ, especially with the HE400.
 
The result of all the above is that I now have a more natural sound that sounds very balanced and 95% of the time I detect no tizz. The mids don't sound recessed at all and the vocals are very clear and crisp. The mids and the vocals sound just right now.
 
The biggest difference is made by 1) but I have a feeling 2) and 3) has something to do with it also but this is harder to detect. It took a few days of listening to songs that I was very familiar with to confirm my initial impressions.
 
I had decided to move to a more expensive setup (one of HE500/LCD3/HD800 and new amp/dac) but now I will stick with the HE400/NFB15. It is sounding the best I have ever heard it.

 
Nov 11, 2013 at 5:26 PM Post #14,287 of 22,116
I agree that electri-q is more advanced and probably better, that's why I used that before for 1.5 years. The problem is its hard to get the curve I need and the curve in the simple eq sounds more natural to me even when I tried to duplicate it exactly in electri-q. At the end of the day, it doesn't matter which eq as long as it sounds good to me.
 
With regards to the bit rate, I was of that opinion also but watching those videos, there is more to it. You would think that adding 0's to get from 16 bit to 24 bit wouldn't make any difference but it does. Watch that video I linked and it explains it. It could be my dac does that internally anyway (I am not sure) but it could also be that windows using 16 bits has already affected the signal before sending it to the dac. For example, electri-q when it attenuates/boosts a 16 bit source as part of its equalization process, it will reduce the bit rate. This happens because of how math in the binary world occurs, to divide by 2 (for example to reduce the volume), you shift 1 bit along, so you have lost a data bit, you go from 16 data bits to 15 data bits, the 16th bit now becomes a 0 and some of you data is lost. If it has 24 bits that were filled with 0's from a 16 bit source then the actual bit rate will not be reduced because the bits that are lost are the 0 bits that were added and not the actual data bits. So its gone from 24 bits to 23 bits effectively but your original 16 bits are still all there (watch that video about 8 mins in). I am definitely getting clearer crisper vocals now, like I said, I am not 100% sure why, but I am afraid to touch anything because its the best I have ever heard it as is. I think its getting a bit off topic so I will leave it at that.
 
Nov 11, 2013 at 5:35 PM Post #14,288 of 22,116
Yeh thats why I said that you can set the bitrate on windows audio control to 24bit but leave the media player the same. That is the bits though not the sampling rate in khz - eg. playing 44100khz file at 192000khz in your media player.... With the volume.. If you are bypassing it straight to your DAC with WASAPI exclusive it should not affect that anyway. This is off topic so lets not discuss it any further!
 
Nov 11, 2013 at 5:38 PM Post #14,289 of 22,116
That is very true about 16 vs. 24 bits. Does anyone know -- if I'm using WASAPI and set it at 24 bits in Foobar, does this mean that the extra 8 zeros are added before EasyQ sees the signal, or after? Meaning, is Easy Q taking me from 16 to 15 bits (or however many), and then the extra 8 are added, or are the extra 8 added first, so EasyQ takes me from 24 to 23? 

I'm getting pretty extreme with digital volume now that I have the Emotiva Mini-x, though interestingly, my EQ has gotten a lot simpler with this amp vs my old Little Dot. 

Also, listening impression on the Emotiva will be coming soon once I get the parts for my speaker taps adapter tomorrow. Then I can use my resistors. Right now, I'm just listening with a super ghetto adapter I made by mutilating a 3.5mm extension cable from Radioshack. Hopefully, the resistors will reduce my use of digital volume control...
 
Nov 11, 2013 at 6:25 PM Post #14,290 of 22,116
Out of interest what do you mean by "getting extreme with digital volume"?
smile.gif

 
Nov 11, 2013 at 6:39 PM Post #14,291 of 22,116
  Out of interest what do you mean by "getting extreme with digital volume"?
smile.gif


Setting it to -20 dB at times. Maybe extreme is an exaggeration for dramatic effect, but I never went that low with my old amp. However, I don't always have it this low. Depends on the loudness of the song. I have it at 0 for some songs. I try to keep the volume knob on the Emotiva above 9 o'clock, usually around 10 o'clock. There is a slight noise floor that increases with the volume knob, but I want stay above the channel imbalance region of the knob. 

My inline resistors should help with this once I get them hooked up, but I'm also trying to learn how to make a voltage divider network so I can see if that works any better. 
 
Nov 11, 2013 at 7:42 PM Post #14,292 of 22,116
  Yeh thats why I said that you can set the bitrate on windows audio control to 24bit but leave the media player the same. That is the bits though not the sampling rate in khz - eg. playing 44100khz file at 192000khz in your media player.... With the volume.. If you are bypassing it straight to your DAC with WASAPI exclusive it should not affect that anyway. This is off topic so lets not discuss it any further!

 
According to one of the experts in another video in that youtube channel (can't seem to find which video it is in now) said that re-sampling 44khz to 192khz sounds better also, this is due to being able to move the noise up to higher frequencies so you can't hear them. They mentioned how a lot of SACD's that were just up-converts from CD to DSD on a SACD sounded better for this reason. Maybe this is crap and maybe it doesn't do the same on a PC, but since its sounding good to me now, think I won't mess with it.
 
 
 
That is very true about 16 vs. 24 bits. Does anyone know -- if I'm using WASAPI and set it at 24 bits in Foobar, does this mean that the extra 8 zeros are added before EasyQ sees the signal, or after? Meaning, is Easy Q taking me from 16 to 15 bits (or however many), and then the extra 8 are added, or are the extra 8 added first, so EasyQ takes me from 24 to 23? 

 
I think if you leave everything at 16 bit on the computer side, then the software equalizing process will result in a drop in bit rate because it is always limited to 16 bit on its output side even if its doing its maths at 24 bit internally. I think its better to set it to 24 bit in foobar before the eq, or as a plug in, it will pass the data back to foobar as 16 bit if it is set to 16 in foobar producing data loss.
 
Nov 11, 2013 at 8:08 PM Post #14,293 of 22,116
  I think if you leave everything at 16 bit on the computer side, then the software equalizing process will result in a drop in bit rate because it is always limited to 16bit on its output side even if its doing its maths at 24bit internally. So I think its needs to be set to 24 in foobar before the eq.


Why is it limited to 16 bits on the output side? I have a 24 bit DAC and I set the WASAPI output to 24 bits.
 
Nov 11, 2013 at 8:24 PM Post #14,294 of 22,116
 
Why is it limited to 16 bits on the output side? I have a 24 bit DAC and I set the WASAPI output to 24 bits.

 
If you set foobar to 16 bit, EasyQ is a plug in to foobar so it will pass the data back in 16 bit to foobar. Foobar will then pass the data with data loss to WASAPI. Its been a long time since I used foobar but if by setting WASAPI to 24 bit that means that foobar is also set to 24 bit, then that is OK.
 

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