Can tell me your settings/plug-in and what type of song u use the settings for? Thanks

This post has been edited by genjo: Oct 7 2010, 04:08 PM

No special settings. Only plugin I have is the APE decoder. Other than that-- nothing.

i only use the built-in EQ to add little bass.

has none dsp installed, deleted everything to save that few KB of space and only asio support
then everything else is for GUI, like wshpanel and columnsui

I still use DS with Resampler(PPHS) upsampling to 192000, then Monkey Audio plug in.

APE, Tak, TTA plugins... with output setting to DAC. No changes in the EQ or such.

I have loads of DSPs, but haven't used any at all. Not for months now. Bro Bonchi, I believe that DSPs are only those extras that affects sound production and processing in the player: all the other things we download are necessarily not considered DSPs, yes? I agree however that the GUI is MUCH better with WSHPanel and Columns UI. Can't live without them!

if the .dll is a dsp, usually it is mentioned in the filename itself basically i deleted all the dsp.dll to make room for png files which will later become custom buttons ... i love to keep my foobar under 3mb .. makes it alot easier to back up too

[attachmentid=1823178]
Foobar buffer @ 50ms
ASIO buffer @ 64 samples

[attachmentid=1823181]
Full file buffering max 2GB
cpu usage nearly zero and mem usage orignally is 6MB+size of song i'm playing(max 2GB)

[attachmentid=1823219]
set foobar's priority to realtime

btw, 64samples ASIO buffer on my lappy is only possible when i enable full file buffering, i think it's because it's faster to access from RAM than HD...
Windows volume control @ 100%,
Speaker volume @80%+. Volume is controlled from foobar... said to give better headroom/dynamic range, and prevents overload/clipping when using lossy formats...

[attachmentid=1823234]
Peak meter when playing an MP3 shows crossing the 0dB line... by reducing the volume in foobar it wont clip the windows volume when sent to hardware...

but try it yourself and see the results...

Remember to set the volume in foobar at at least -35dB if u set 100% at windows to avoid sudden burst. To see volume figure, right click on the bottom bar and tick "show volume"

no dsp used, dont wanna color the sound



This post has been edited by andrew9292: Oct 8 2010, 10:48 AM

Hmm...it does help to lower clipping.
I haven't realize it does give some difference

@bro andrew.. nice guide there ..thanks!
also did the same on the hardware buffering part tho i didnt set mine to 2gb on full file buffering but i will try that later (i think giving it 2gb of space will also imrpove smoother song playback)

btw i would like to suggest setting apply gain under playback to "none" as that will eliminate album gains.. those situation where some songs are loud while some are very soft.

but it won't be bit-perfect anymore doesn't it?

@andrew9262

wah nice settings down there bro. I'm thinking that with the full file buffering, probably help me a little when using foobar2000 on battery with less HDD activity.

Bit perfect?

I just slight reduce the volume from foobar....not related to data those...

i read somewhere it is best to change the volume on your audio interface/speaker volume knob and leave foobar and windows audio volume to 100%. Because when you lower the volume on foobar you'll lose some data. I'm not sure and i don't remember the original detailed term, but that person told me it won't be close to bit-perfect (as it is hard to get 100% bit-perfect, lots of factor can manipulate it). I for sure didn't hear any difference wheter it is bit-perfect or not, so it does not matters to me.

Need to search on the architecture 1st.

But for windows volume, i'm not sure too about the bit perfect stuff but i can confirm that some amplifiers if you low down the input power, it will loose out the details even though you crank up the amplifier volume because the input signal is too low.

hmm, lower the volume on foobar2000 can lose data?

Used to have many setting, but now none.

what is replay gain for? any setting for tat?

The RAM buffering settings refers to the maximum RAM Foobar can use. The memory usage will be Foobar's original memory usage+the file size of your song. Unless you have a 2GB song, it will never go that far... In fact for most users, setting to 512MB (524288kB) should be enough as most lossless files wont cross 200MB per track


True statement there. Lowing the software volume will affect the noise floor as it pushes the bits near the noise floor "down", to be out of range of most DACs... but unless you have music that has data/music at near -100dB, there wont really be a difference... I used to be a big fan of bit perfect (100% all digital volume control, 44100khz 16bit) until i realize it's disadvantages

-Setting 100% at application volume will be bit perfect, when outputing from the application to windows mixer.
-Setting the windows volume to 100% will allow Foobar to play in the full dynamic range.
-Setting both of above to be at 100% will allow True bit perfect in theory. Eg... what is at -3dB at foobar will also be -3dB at windows mixer, and that will be that.
-But setting both at 100% will cause clipping with lossy formats. Eg... +1dB overflow at foobar, will be "flattened" or clipped to 0dB at windows volume. Disadvantage 1.
-Setting both to 100% will cause very high output in the hardware, dac, opamps, whatever u name it. And such high output isn't ideal for most usage, you are driving the hardware at 100% or sometimes even overloading, oftenly this will cause distortion+heat. Disadvantage 2.
-Setting both to 100% will send a very high analogue output to your speaker, which has it's own preamp/amp inside, if the music is over the range of the amps, it will clip again+heat. And advance/high end amps with anti distortion control will further clip the sound by removing the peaks that are causing the amp to clip, either by using a hard limiter, intelligent limiter, or dynamic range compressor (also a kind of limiter) and this will affect the sound. Disadvantage 3 & 4.

-Setting the volume in foobar will prevent clipping at windows mixer.
-Setting the volume at your speaker/amp to 60-100% (varies amp to amp) will again open the dynamic range (what is soft can now be heard clearer in most cases)
-Setting the volume at foobar low, windows volume@100 = no clipping + high or [full volume at speaker(low gain amps)] = u get a loud sound without overloading anything
-Setting the volume at foobar too low, u push the lower range of the music out of range, but oftenly there is no information at such low level (around -100dB for most DACs)

I'm actually using live sound concept, in live sound setups, the final output stage or amps are normally at full or around 80%, so that there can be huge range to play with at the mixing console. If u set low output volume at final level, u will have to push the volume at processing stage to a very high level to get the same volume a.k.a loudness maximization... basically, the input must not be too loud, neither must it be too soft to the point where the really soft passages of instruments go missing...

but again, this is for most cases... different hardware will behave differently...
use yor ears to decide

This is also true, but it really depends on setup to setup

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My setting might work well for me, and for some, but not all Use your ears to judge what's best! This issue on bit perfect has been debated many many times and is very very controversial... most importantly... enjoy the music

This post has been edited by andrew9292: Oct 8 2010, 11:29 PM

getting full 100% bit perfect is not easy lol.. even our downloaded/EAC ripped flacs may not be 100% some times they are 99.9% accurate according to logs
i believe you cant get 100% on CDs as well if you compare it to the master copy

but the effort put in to achieve that in its own method is what makes and audiophile an audiophile:P

oh well.. as long as there's no audible fault .. lets just enjoy the music

Is this have advantage over normal priority because set it into realtime can cause unstability?

Myself foobar/windows volume/soundcard at 90%-100%. Control volume from the stereo amp.

Sounds good to me.

My setting is use ASIO plugin, with ASIO buffer latency 10ms and foobar buffer is at default 1000ms (used to put at lowest, but no audible difference to me). I use SoX resampler to resample to 44.1kHz if any of my file is below that. Other than that, everything is untouched.

Used to equalize my headphone by using VST wrapper plugin and Electri-Q equalizer but at the end I ditch it since I don't use headphone much nowadays

anyone uses secret rabbit code re-sampler here? what you guys think about this
currently i have my re-sampler remove but i have to use it sometimes for weird 48khz mastered albums and sometimes vinyl rips which are 96khz.. tho i hear audible difference if i have it on @ 44.1khz vs having it turned off

Currently using SSRC resampler at 88.2Khz as i feel it slightly opens up the soundstage and i can hear abit more details in the highs. PPHS softens the transients abit too much, useful for sharp recordings but i stick to SSRC most of the time. Before i got a DAC upgrade resampling or not didn't really make a difference.

@bro andrew.. finally got back to my rig and tested the file buffer from memory up to 512mb .. heck its sounds alot cleaner as compared to value set 0
thanks for the guide

but i didn't feel the different.

prolly cuz the HDD containing my songs is slow and old
file buffering removed stuttering .. less noise too giving me kinda like a clean experience

ya upsampling really open the stage... but did u try JRMC?

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Added on October 15, 2010, 10:10 ambuffering the file into memory will have resulted in a less jitter.

This post has been edited by vir___killer: Oct 15 2010, 10:10 AM

Hey guys,

I'm using Go-vibe external DAC. How do I ASIO it? I want sound isolation like I did last time

hi bro andrew..
Can u help me to find best setting for my system?
i'm using creative sb xtrememusic support asio native, f2k, and win 7 64bit.
i'm very glad see your setting for u'r system, may u can help on my system too..
thanks before..

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Added on October 22, 2010, 11:40 amwhat's u'r audio system bonchi?
may u can help me to find best setting for my audio system?
i'm using creative sb xtrememusic and f2k.
thanks before..

This post has been edited by dtonies75: Oct 22 2010, 11:40 AM

bro, can share how u make ASIO4ALL work? mind give me the dl link???

settings are what it came with, output via USB Audio DAC, select track and click play is all there is for me, then adjust volume to taste ....

Exactly! Just click to play!

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Added on October 22, 2010, 11:40 am
what's u'r audio system bonchi?
may u can help me to find best setting for my audio system?
i'm using creative sb xtrememusic and f2k.
thanks before..

[/quote]

mine is just some cheapo soundcard weird to a headphone directly nothing to be proud of
for best setting is hard cuz we all have diff taste.. will need to resort to using some DSP
what bro Andrew here suggest is for low latency only doing the file buffer thing gave me less crackling due to RAM is faster than HDD

there are no point obsessing of Asio output. There are no audible difference between Direct Sound and Asio unless your Direct Sound driver are broken. Asio purpose is to reduce latency when doing recording. And the full file buffering also not meant to produce greater sound quality.

No, the main point of using ASIO is to bypass kMixer during Windows XP days. In Vista and 7 its now possible to achieve bit perfect out without the needs of ASIO. DirectSound is more than enough unless the soundcard you use do some processing, if you concern WASAPI will do just that, easier as you don't need to install additional stuff in your PC if your soundcard don't have native ASIO

yes that is very true. Kmixer in Windows XP done a lousy job and if i'm correct it also perform resampling. And it is also noticeable. Now, i think most people use Windows Vista or 7 they doesn't need to bypass the mixer anymore since they're used new architecture.

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Added on October 22, 2010, 9:42 pmThat is why AccurateRip exist. To ensure that your rip is perfect.

This post has been edited by Angel of Deth: Oct 22 2010, 09:42 PM

Hi Najmods, btw we're using same SC...
Can you share to me how u setting your system to get best performance?
I'm really confused to chose wasapi or asio for xmusic, cause both give me good sound, how about you?

Najmod has stated before that he set his foobar2k output to Wasapi and set the playback devices sample rate to 16-bit 44.1 kHz to match his playback format. Honestly, there are no audible difference for me between DS, KS, Asio or Wasapi. I choose to use Asio because it is convenient for me because it mute other playback except for my music. Wasapi does the better in term of isolating the music and other playback sound but it have issue with my sound card driver, maybe because the plugin is still buggy (most of foobar plugin are buggy anyway). The only thing i dislike about Asio is i must close any other source of playback such as Youtube, Flash Player, sometime must kill internet browser process in the task bar in order to get Asio works, while Wasapi didn't have this issue.

Angel of Deth already say what I wanted to say. Since you use XtremeMusic just use ASIO. If ASIO have some problem when browsing (watching flash videos), just use either WASAPI so only your music is played, or simply use DirectSound, both sound won't be uninterrupted.

Oh and also open up Creative Control Panel, set to Audio Creation mode, and enable Bit-Matched playback and you set to go. In this state even DirectSound should provide bit perfect playback

Yeah thanks a lot brother..

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Added on October 23, 2010, 8:48 amOk Najmods i'll try to my rig..
BTW thanks for your suggestion.
How about you setting buffer length on f2k and buffer latency on Creative ASIO?
Now i'am using f2k buffer length 50ms, and full file buffering up to 512mb (like suggestion bro andrew), on Creative ASIO buffer latency set to 10ms (like your setting).
And on Audio Creation Mode EAX effects set to disable.
Any suggestion for me?

This post has been edited by dtonies75: Oct 23 2010, 08:56 AM

As long as it does not produce any clipping/distortion/cracking, it'll be fine. Low ASIO buffer latency could produce clipping/cracking on some slower system, but even my system at 800MHz low power setting it still work fine. It vary from one system to another.

EAX effects on Audio Creation Mode disable or enabled?
Can I set Bass and Treble on Audio Creation Mode?
Because my speaker too bad on bass.

I'll just use resampler and advance limiter for DSP and WASAPI for output with 16-bit output data format.

The purpose of using ASIO is to get bit perfect output, if you going to play with treble and bass you'd better off with the normal DirectSound because its already defeats the purpose of using ASIO in the first place. EAX is used for games, unless if its on Entertainment mode where you could do room emulation.

is there a component to replace the default playback button?

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Added on October 25, 2010, 11:50 pmP.s. except column ui

This post has been edited by Sphenix: Oct 25 2010, 11:50 PM

right click > customize buttons

you need to use icon file

ic...... it works....

how to add artwork on a song?

Find the album art online... Set a panel to display album art.

wat i meant is add the art to the song......
I right click > Properties the song i dont see add pic option......

yeah...... although is not perfect.....

Spectrum Analyzer
Biography
ArtworkView
NG Playlist
Chronoflow
Lyrics Show2
Item Properties
ESPlaylist
Waveform Seekbar

this is my current setting......

This post has been edited by Sphenix: Oct 30 2010, 12:18 AM


Attached thumbnail(s)

For audio purist one thing for sure turn off all dsp/eq. Remember to DISABLE the replay gain feature. It really sucks if turn on.

Install Asio and sound great already, I doubt you can improve further... The sound improves as u improve your hardware... that's my experience

The most important is your hardwares such as DAC/soundcards, amps, speakers/headphones. If not, no matter how you set it would sound good in sennsonic or SonicGear.

+1

Use boss c5 la!! guarantee for sound quality..

anyone know how to make the lyrics show work?

update: I'm using dark one UI. just found how to click the lyrics show panel out but anyway to make it like brother Sphenix that it's nicely inside the windows?

Update2: shit, i press add lyrics panel and it's covering my whole GUI! anyone how to close it back instead of reinstalling the software?

This post has been edited by Andy0625: Mar 18 2011, 02:29 AM

the default butten is always there.
*edit* do export your fcl theme often (include all the scripts as well) to prevent things like this to happen

This post has been edited by Bonchi: Mar 18 2011, 03:20 AM

where's the default button, bonchi?

hi... what its mean ?? mean if using usb DAC = nothing to set ?

no.

its not really a default button but more like a quick setup that allows you to change it to default layout (im assuming that youre using columnsUI)

but there's a workaround to remove a certain panel.. you can try this first before resorting to go back to default..

yes, just right click on the panel you wanna edit and click remove to remove it

Thanks man, I just choose the Theme to default and then change back to Columns UI and it works again - except for all the song list gone.

Borrow this thread for a question.
I have a Hippo Cri Cri+, well it does support up to 24-bit/96Khz, but I have a FLAC which is 24-bit/192KHz, and I can't use the WASAPI to send audio directly to the DAC, any other solution other than using the DS plugins?

use the resampler in foobar and set the output to 96KHz.

Oh great thanks, it works... lol my first time on Foobar

Pure vanilla Foobar2k. I rarely use EQ except when I listen to electronic. I usually turn my EQ off (flat) when listening to classical, orchestra and jazz.