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Bughead Emperor


supra

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This seems interesting to me, and like you I would really like to see some sort of manual or explanation of the upsampling colors ect...

Forrest:

Win10 i9 9900KS/GTX1060 HQPlayer4>Win10 NAA

DSD>Pavel's DSC2.6>Bent Audio TAP>

Parasound JC1>"Naked" Quad ESL63/Tannoy PS350B subs<100Hz

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This was posted on another forum- it explains the complexity of the player, which is quite unique, and very good sounding. Actually its more than very good.....

 

 

A Russian thread has about 74 pages on this player. On this page http://forum.doctorh...c=24713&st=1650 someone is trying to explain this player with pictures. Run it through translator, see if it makes sense.

"

It is not simple to understand . The main difference is that the author, Hiroyuki , considering the source format 441000/16 not as a model , but as a lossy format . Accordingly, all these Black, Emerald , armor and other - these are very sophisticated algorithms restore lost when digitizing information. It uses spline interpolation , and as a model of multiple - model schemes recording and CD production . Accordingly, a huge load on the processor.

On a button - start when the first dialog box buttons and hold Shift + Z mode is activated Z-tune ( only on 64 bit systems and 8 gigs of memory ) , which optimizes the algorithm of memory , then the choice of ASIO or WASAPI output. First preferred.

After self window. Starting with version 2.92 of the dialogue removed buttons that are not harnessing the specific modes. Normal - mode output similar to Foobar and differs mainly in algorithms only work with memory. Modes with sound BufHead - a bit about them .

top row - selection X1 X2 X4 X8 - degree sampling transition , Basik, Emerald, ... Brownie- level and signal processing . LPF-free and under it - filtering the output signal different algorithms Free- unfiltered - important - if filtering is enabled it is necessary to ensure that the level would not exceed the maximum allowable .

 

Self sound optimaiser- create their own algorithms

image

1 - mode volume control output -AUTO player will reduce the volume if it is outside the permitted , pressing aotklyuchaet automatic control , then a box under a continuation of the arrow indicates the maximum output level for a particular algorithm - for example Browine 5x2 it will be 13 , he should be put on the volume

image

Next 2 - allows you to select the size of the cluster and the precision of the calculation for the spline interpolation , the higher accuracy and the more stages , the more accurate the result and the more demand resources. ( Workstation 6 on two nuclear Cheon loaded at 100% and believes the sound is 180-300 seconds at the maximum values ​​of the image )

3 - selection phase of the output signal

4 - Select the algorithm clear memory 9 different types of memory are different algorithms provide the highest quality sound , empirically )

5 - choice player in the harnessing of cores and threads

6 - Select the processed sound Bagh- HS - output sound from Bug LV5- maximum BUG LV1- minimal -

mode Self sound optimizer has the ability to export the processed stream player in crude RAW file for export WAV data from it and the possibility to listen to any player . There is no support CUE!!!

image

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Been playing with this today. It is a terrific sounding player and I don't really find the interface so ugly. It's a simple interface which belies the powerful tools it puts at your disposal for improving SQ.

 

When you launch it a panel appears inviting you to click on it for set up. You would do this the first time or if you want to change your audio interface/driver;etc.

 

Once you've chosen the driver/interface (mine was the ASIO driver for the Zodiac Platinum DAC) another small window offers 3 choices of what are basically upsampling features and their related options. Finally there's a check box at the bottom of that window which allows you to select a "Pink" interface. If you like Pink, well, why not?

 

I chose the top "Standard" option for upsampling and related features.

 

Following that, the main interface window appears. Across the top from left to right are buttons marked; 1X, 2X, 4X, 8X. Each of these, of course refers to how many times you wish to upsample the rate of the file(s) you're going to play. Clicking on each of these buttons in turn gets you a drop down menu of different algorithms which can be selected in conjunction with that level of upsampling. In each drop down menu as we move down the list each algorithm apparently gets more sophisticated and demanding of your CPU.

 

Each of these algorithms is, quite helpfully I think, given an easily remembered name, such as a Emerald, Saphire, Snow, Black; etc. Selecting any of these provides a further dropdown menu allowing you to select numbered levels of sophistication for that algorithm.

 

Example: If I have a 44.1 file(s) queued up to play in the playlist and I select the X2 button and I choose "Snow" as my algorithm, I then need to choose what level of "Snow" to use. If I choose Snow level 7, my selection is 7x2 (x2 being my upsampling level of 88.2 in this case).

 

Now in a pane on the right hand side a number of little icons will appear; 16 in this example, to be precise. In this pane it will also be indicated as 16 lines (the number of "lines" is equal to the number of these icons). This is important, because you need to find the limit that your CPU can handle. In my case it was 16 lines for 2x oversampling. The next algorithm in the dropdown list, "Black" is apparently more complex, and you will hit 16 lines at a lower level in the Black algorithm than the Snow algorithm. I determined I could use any algorithm/level at 2x oversampling that would yield 16 or less lines.

 

Depending on your CPU, you may get away with more lines or you may be limited to less. Different combinations of algorithms and their levels will yield this limit at 2x upsampling, which was 16 lines in my case. For 2x oversampling I knew I was safe as long as I didn't exceed this 16 line limit. What happens at 4x upsampling? Easy, the limit is halved each time we double our upsampling. So at 4x I can only select any algorithm combination within an 8 lines/icons limit, at 8x oversapling this becomes 4 lines or icons.

 

What happens if you exceed the limit? Simple, playback simply stops and a warning appears in the interface to tell you the CPU requirement is not satisfied.

 

This limit is for realtime playback. However, you can also choose to output a RAW file instead of playing back in real time. This process is not subject to the above limits. So you can produce a RAW PCM file processed by the most sophistocated algorithm without limit. This RAW file then needs to be converted to a playable format and can then be played back easily by any supported player for the file format in question. e.g. J River MC, or any other suitable player.

 

As this post is already very long, I'll try to write some more about this tomorrow and post some images.

 

An interesting little player indeed!

 

geoff

Owner of: Sound Galleries, High-End Audio Dealer, Monaco

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Geoff thanks for your useful observations. How do you find the sound? I started to think it was a bit " tone control" in operation . Some of the settings i used even made the sound a bit mono like. I guess there are so many options it's just as easy to get it wrong as right.

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Geoff thanks for your useful observations. How do you find the sound? I started to think it was a bit " tone control" in operation . Some of the settings i used even made the sound a bit mono like. I guess there are so many options it's just as easy to get it wrong as right.

 

It can make really dramatic differences to the sound! Truly night and day differences.

 

i agree with you though, that it's not always obvious whether the changes are improvements.

 

I've been using it as an offline processor instead of a real time player.

 

This allows me to push the settings to the limits, except I don't go further than 2x upsampling. I then use Brownie x9, which I'm assuming is the most advanced as it has the most processing (31 lines).

 

I click on the "self sound optimiser" button and double click on the further "Raw" button which appears.

 

Then when I click on play to play the one file I've loaded, it will count down the 31 lines through several iterations. At the end of the process a raw file will be produced in the same folder as the original file.

 

Don't do anything else with your machine while processing, as this could result in glitches in the raw file.

 

Go into the folder and you will see the raw file with the extension of the original file tacked on the end, e.g. ".flac" I need to remove this for conversion to a playable format.

 

I use Audacity to import the raw file. Audacity will present an options box during importation. Be sure to indicate in here the sample rate of the raw file. In my case this was 88200.

 

When it's imported into Audacity, export it out again as a flac file. flac preserves the 24 bit 88.2khz raw file characteristics. This is why I choose flac in Audacity rather than wav, which only seems to offer a 16bit option.

 

Now you should have a flac version which is playable in any other player such as J R MC.

 

By producing several versions of the same track this way and listening to them in another player you'll be able to determine which settings work best for you.

 

You'll also be able to push the settings way beyond the cpu limits you'd be able to achieve in Bug Head with real time playback.

 

One example I tried was Holly Cole's I want you from her best of album. The processed file sounded like a different woman singing. She was singing far more from the chest. I came to the conclusion that the original, which sounded lighter and hyper transparent by comparison, was in fact a little artificial and the Bug Head Bownie 9x at 88.2 version, more natural.

 

it will be great if you and other members would try this. We might reach a consensus on whether it can dramatically improve the sound and which settings are best.

 

 

I stuck to 88.2 to keep the file size and processing times reasonable.

 

I'll be interested to read about further experiments.

 

geoff

Owner of: Sound Galleries, High-End Audio Dealer, Monaco

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Thank you for your input Geoff. If I may ask, have you info(subject listening counts) indicating that the pull down ascended in quality the way you described- brownie being the best, not simply an algorithm type? I got a bit lost the first few times I tried it, and used it different than you described. I stayed at 4x/176k levels 1 & 2 attempting to learn the differences in the colors. It never wouldn't load, just took 45 seconds to start at times.

Forrest:

Win10 i9 9900KS/GTX1060 HQPlayer4>Win10 NAA

DSD>Pavel's DSC2.6>Bent Audio TAP>

Parasound JC1>"Naked" Quad ESL63/Tannoy PS350B subs<100Hz

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Thank you for your input Geoff. If I may ask, have you info(subject listening counts) indicating that the pull down ascended in quality the way you described- brownie being the best, not simply an algorithm type? I got a bit lost the first few times I tried it, and used it different than you described. I stayed at 4x/176k levels 1 & 2 attempting to learn the differences in the colors. It never wouldn't load, just took 45 seconds to start at times.

 

I'm making the simple assumption that the more demanding it is on your CPU, the more sophisticated the algorithm should be and the more sophisticated the algorithm, the better the sound should be, at least theoretically.

 

I've not yet compared the different algorithms. I just pushed it to the limits as an offline processor; but limited to 2x upsampling.

 

I'm going by the number of lines (icons) that appear in the panel on the right, to determine how sophisticated the algorithm, rather than the position in the menu.

 

if you use it as a player, it will count down those number of lines before playback begins. If I use it as a player, I can"t get beyond 16 lines at 2x or 8 lines at 4x. So I prefer to use it as an offline processor where no such limits apply. Brownie at 2x upsampling at level 9 yields 31 lines.

 

Of course I recognise that less sophisticated algorithms may produce preferable results for some listeners on some systems, subjectively speaking.

 

Now I'm using it purely offline as described in my last post and pushing 2x upsampling to the limits using 9x Brownie. I haven't made comparisons with any other algorithms.

 

I also had trouble converting raw files to flac at anything greater than 88200 using Audacity, so I stuck to 88200.

 

 

 

 

geoff

Owner of: Sound Galleries, High-End Audio Dealer, Monaco

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  • 8 months later...

Version 4.20

https://onedrive.live.com/?cid=95b15e620f5d211d&id=95B15E620F5D211D%21105

 

Facebook:

https://ja-jp.facebook.com/pages/Bug-head-technology/221354151338856

 

its in japan,but translated you can understand how the Player works.

 

shpongle

Gigabyte-GA-Z97 Sniper with separate usb-dac out,5v usb-power disabeld in Bios. i3 4130T(35 W)@800mhz,16gb [email protected] @1.25v. Cpu+Ram powered by Pico-160w+Voltcraft Lab.PSU.

Cpu-Fan+Music Hdd with 2 separate  LPsu. JLsounds Dac-Kit: USB-I2S XMOS-board+ AK4490 DAC+DVR603 FB+2 JLS LPSU. Player: wtfplay and Daphile

 

Tubeamp: Lua 4040 C . LS: Chario Academy II. Conections/Filters: Reson DNM / Fisch Audio.

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I've just been trying the latest version and it's sounding really great, up-sampling 44.1 to 352.8 using Brownie 9 * 8 (39 Lines).

 

When I load a DSD (.dff) file it plays; but converts to PCM. On the button/menu labeled PCM there is an option to choose DSD instead or to convert DSD to PCM. The DSD option is greyed out (not available). I'm using an exasound e22, which is DSD256 capable.

 

Can anyone tell me if this is for a future version or requires a paid version?

 

Also, here is a wish list of features I would like to see in the next version:

 

1/ Ability to up-sample/convert PCM to DSD up to the max supported by a DSD capable DAC.

2/ Load and play playlist files (m3u, m3u8; etc.)

3/ Support playback of internet lossless streams from Qobuz and Tidal from a list of URL's in a playlist file.

4/ Add option to start the play process automatically as soon as a file (including a playlist) is loaded.

 

I would gladly pay for a version which has these features.

 

geoff

Owner of: Sound Galleries, High-End Audio Dealer, Monaco

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hi geoff

for dsd nativ change the output to raw!

in DSD playback mode you have 2 filters, slash and shine and some Streaming SIMD Extensions. like mmx sse avx.

depends on the cpu and ram what you can use.

for high class professionel you need at least i3 haswell cpu +16gb of double sided ram,otherwise Sound degradition.

 

i use infinity blade hq in high class prof. mode normal.OS Server 2012 min.core mode +cad v2-6a.

 

my Player Setup for DSD playback:raw, slash10, avx+(slash/shine Level 1-10, you can Change with a click on normal)

Gigabyte-GA-Z97 Sniper with separate usb-dac out,5v usb-power disabeld in Bios. i3 4130T(35 W)@800mhz,16gb [email protected] @1.25v. Cpu+Ram powered by Pico-160w+Voltcraft Lab.PSU.

Cpu-Fan+Music Hdd with 2 separate  LPsu. JLsounds Dac-Kit: USB-I2S XMOS-board+ AK4490 DAC+DVR603 FB+2 JLS LPSU. Player: wtfplay and Daphile

 

Tubeamp: Lua 4040 C . LS: Chario Academy II. Conections/Filters: Reson DNM / Fisch Audio.

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hi geoff

for dsd nativ change the output to raw!

in DSD playback mode you have 2 filters, slash and shine and some Streaming SIMD Extensions. like mmx sse avx.

depends on the cpu and ram what you can use.

for high class professionel you need at least i3 haswell cpu +16gb of double sided ram,otherwise Sound degradition.

 

i use infinity blade hq in high class prof. mode normal.OS Server 2012 min.core mode +cad v2-6a.

 

my Player Setup for DSD playback:raw, slash10, avx+(slash/shine Level 1-10, you can Change with a click on normal)

 

OK, got it. Thanks very much shpongle. Sounds very rich, full warm and big, through KEF Blades using the exasound e22 and NCore Mono Blocks.

 

This with DSD64 files ripped from SACD (Using a Sony PS3). I could use a little more clarity/transparency, which I get when I up-sample these files to DSD256.

 

It's a very appealing sound in many ways though.

 

What does the slash/shine tuning do?

 

Thanks again,

 

Geoff

Owner of: Sound Galleries, High-End Audio Dealer, Monaco

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I can now hear there are a lot of possibilities to influence the sound of DSD in Normal mode. The degree of transparency versus richness seems to be influenced by those "Slash and Shine" values. It'll take me a while to investigate all these possibilities, just for DSD and in Normal mode.

 

Just out of interest which settings are you using for PCM? Are you up-sampling?

 

Also, are you in contact with the developer? I'm wondering how to send feature requests to him when I can't speak Japanese.

 

Thanks again,

 

geoff

Owner of: Sound Galleries, High-End Audio Dealer, Monaco

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hi geoff

always welcome,i like to help in the search of the best Sound!

.

 

i don't listen PCM with the Player.

i do upsampling/converting to dsd with korg audiogate,so i have only 4% cpu load.

 

here the Settings for PCM from hiroyuki,the developer:

 

Normal only:. / Fstar 10 + mmx / mmx + mmx + +]

Up-samping: [Galaxy] + mmx / mmx +] / [mmx + +]

[mmx]: Rewrite memory after decode

mmx +: Rewrite memory 2 after decode

[mmx + +]: [mmx +] asio send buffer memory after and Rewrite

Fstar 10: Ultra high quality send to asio buffer

Galaxy: High quality send to asio buffer

 

 

i'm not in contact with him.

i don't like Facebook.

i think he speaks/write english.

his Facebook blog:

 

https://ja-jp.facebook.com/pages/Bug-head-technology/221354151338856

 

 

 

enjoy the Sound!

 

shpongle

Gigabyte-GA-Z97 Sniper with separate usb-dac out,5v usb-power disabeld in Bios. i3 4130T(35 W)@800mhz,16gb [email protected] @1.25v. Cpu+Ram powered by Pico-160w+Voltcraft Lab.PSU.

Cpu-Fan+Music Hdd with 2 separate  LPsu. JLsounds Dac-Kit: USB-I2S XMOS-board+ AK4490 DAC+DVR603 FB+2 JLS LPSU. Player: wtfplay and Daphile

 

Tubeamp: Lua 4040 C . LS: Chario Academy II. Conections/Filters: Reson DNM / Fisch Audio.

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hi geoff

always welcome,i like to help in the search of the best Sound!

.

 

i don't listen PCM with the Player.

i do upsampling/converting to dsd with korg audiogate,so i have only 4% cpu load.

 

here the Settings for PCM from hiroyuki,the developer:

 

Normal only:. / Fstar 10 + mmx / mmx + mmx + +]

Up-samping: [Galaxy] + mmx / mmx +] / [mmx + +]

[mmx]: Rewrite memory after decode

mmx +: Rewrite memory 2 after decode

[mmx + +]: [mmx +] asio send buffer memory after and Rewrite

Fstar 10: Ultra high quality send to asio buffer

Galaxy: High quality send to asio buffer

 

 

i'm not in contact with him.

i don't like Facebook.

i think he speaks/write english.

his Facebook blog:

 

https://ja-jp.facebook.com/pages/Bug-head-technology/221354151338856

 

 

 

enjoy the Sound!

 

shpongle

 

Well thanks again, that's very useful. I am enjoying the sound and will try to contact him on FB,

 

geoff

Owner of: Sound Galleries, High-End Audio Dealer, Monaco

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hi geoff

before i forget,with a rightclick on the Player you will have more Settings

 

 

shpongle

Gigabyte-GA-Z97 Sniper with separate usb-dac out,5v usb-power disabeld in Bios. i3 4130T(35 W)@800mhz,16gb [email protected] @1.25v. Cpu+Ram powered by Pico-160w+Voltcraft Lab.PSU.

Cpu-Fan+Music Hdd with 2 separate  LPsu. JLsounds Dac-Kit: USB-I2S XMOS-board+ AK4490 DAC+DVR603 FB+2 JLS LPSU. Player: wtfplay and Daphile

 

Tubeamp: Lua 4040 C . LS: Chario Academy II. Conections/Filters: Reson DNM / Fisch Audio.

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Hi shpongle,

Great you share all the hints. Is there anywhere some kind of doc in english about the player? I would like to understand a bit more of the approach and the possibilities behind it.

Many thanks

Thomas

DIY coax tractrix horn system 2 corner subwoofer /// 6 full digital amplifier D802 floating PSU 12V battery & caps/filter /// Active crossover @ Acourate Convolver & room [email protected] /// General 2 PC setup: floating PSU picoless battery & caps/filter powered Bicker DC160W: PC1(Player) - individual stripped MS RamOS with JPlay /// PC2: Server 2016 RamOS - AO 2.0b5 - Acourate Convolver - online convolving & crossover /// Chain: PC1 - USB - F-1 - SPdif coax - Mutec MC3.1+ USB - SPdif coax - FireFace UCX floating PSU 12V battery & caps/filter - USB - PC2 - FireFace UCX - Adat LWL - Mutec MC-4 - 3 x SPdif coax - D802 Low/ D802 Middle / D802 High - 2 Stereo Lab KWH250 with BMS 4590 plus 2 modified corner subwoofer Abacus ABS210

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hi geoff

before i forget,with a rightclick on the Player you will have more Settings

 

 

shpongle

 

Thanks again shpongle

 

So you prefer all your PCM converted to DSD, right? I'm the same, except I've been doing the conversion real-time with HQPlayer.

 

HQPlayer is my reference player using its poly-sinc filter and DSD7 modulator to convert to DSD256 and send to the exasound DAC.

 

This uses about 14% max CPU on my Mac pro under Windows 8.1

 

I see you use 2012 server and prefer to keep CPU even lower.

 

The different possibilities with Bug Head intrigue me. The sound in general is different to HQPlayer, though enjoyable in its own way and as i said, I'm keen to explore it further.

 

The earlier version which I tried a year ago allowed offline up-sampling. It would produce a RAW file which then needed to be converted to flac or WAV for playback. Do you happen to know if this is still possible with the new version?

 

Also, as i recall, the previous version supported WASAPI and Kernel streaming, whereas this latest version only seems to work with ASIO drivers, unless I'm missing something.

Owner of: Sound Galleries, High-End Audio Dealer, Monaco

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hallo thomas

yes,there is a pdf.

install the Player, then you will find it under programs\bughead.not realy useful,only few sides in english!

when you use the Player,take infinity blade hq.with your hardeware you will get high class prof.mode.

Looks like this:

 

[ATTACH=CONFIG]15986[/ATTACH]

when all lines are red:high class prof.=highest possible Sound.

officially bughead suports win7/8.1.i use Server 2012 in mini Server mode+cad v2-6a,in core mode some dll are missing.

Server 2012r2 will also work.because its win 8.1.

 

 

the Player Historie has different parts:

first was pcm upsampling,Geoff+supra explained it very well,some posts above.

today normal mode is prefered from Hiroyuki.

he wrotes some new algorithms(filters) for normal mode plus Integration of Streaming SIMD Extensions.

on his Facebook blog, scroll down and you will see some pics, he is working with SIMD.

i don't understand what he is doing but results are great!

 

gutes testen!

 

daniel

Gigabyte-GA-Z97 Sniper with separate usb-dac out,5v usb-power disabeld in Bios. i3 4130T(35 W)@800mhz,16gb [email protected] @1.25v. Cpu+Ram powered by Pico-160w+Voltcraft Lab.PSU.

Cpu-Fan+Music Hdd with 2 separate  LPsu. JLsounds Dac-Kit: USB-I2S XMOS-board+ AK4490 DAC+DVR603 FB+2 JLS LPSU. Player: wtfplay and Daphile

 

Tubeamp: Lua 4040 C . LS: Chario Academy II. Conections/Filters: Reson DNM / Fisch Audio.

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