What are you guys settings for foobar 2000
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 Oct 9 2010, 12:04 AM
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Senior Member
1,649 posts Joined: Sep 2008 
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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  |
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Oct 9 2010, 12:49 AM
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Senior Member
1,242 posts Joined: Sep 2008 From: Cheras 
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QUOTE(andrew9292 @ Oct 8 2010, 10:31 AM) [attachmentid=1823178] Is this have advantage over normal priority because set it into realtime can cause unstability?[attachmentid=1823219] set foobar's priority to realtime  |
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Oct 9 2010, 01:03 AM
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Senior Member
902 posts Joined: Apr 2009 From: Feel like i'm in Mars
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Myself foobar/windows volume/soundcard at 90%-100%. Control volume from the stereo amp.
Sounds good to me. |
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Oct 9 2010, 01:22 AM
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Senior Member
5,211 posts Joined: Feb 2005 From: Konohana
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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 |
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Oct 9 2010, 01:53 PM
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Senior Member
1,649 posts Joined: Sep 2008
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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 |
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Oct 9 2010, 03:43 PM
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Senior Member
677 posts Joined: May 2005 From: 1s and 0s
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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.
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Oct 14 2010, 02:27 AM
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Senior Member
1,649 posts Joined: Sep 2008
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@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 |
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Oct 14 2010, 05:00 AM
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Senior Member
8,686 posts Joined: Mar 2009 
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but i didn't feel the different.
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Oct 14 2010, 05:12 AM
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Senior Member
1,649 posts Joined: Sep 2008
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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 |
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Oct 15 2010, 10:08 AM
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Senior Member
646 posts Joined: Feb 2005
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QUOTE(Categg @ Oct 9 2010, 04:43 PM) 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. ya upsampling really open the stage... but did u try JRMC?Added on October 15, 2010, 10:10 am QUOTE(Bonchi @ Oct 14 2010, 03:27 AM) @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 buffering the file into memory will have resulted in a less jitter.thanks for the guide This post has been edited by vir___killer: Oct 15 2010, 10:10 AM |
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Oct 15 2010, 04:09 PM
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Senior Member
3,037 posts Joined: Dec 2007 From: 6-feet under
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Hey guys,
I'm using Go-vibe external DAC. How do I ASIO it? I want sound isolation like I did last time |
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Oct 22 2010, 11:37 AM
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Newbie
4 posts Joined: Oct 2010
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QUOTE(andrew9292 @ Oct 8 2010, 10:31 AM) [attachmentid=1823178] hi bro andrew..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  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.. Added on October 22, 2010, 11:40 am QUOTE(Bonchi @ Oct 14 2010, 02:27 AM) @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 what's u'r audio system bonchi?thanks for the guide 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 |
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Oct 22 2010, 01:13 PM
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Senior Member
544 posts Joined: Mar 2010 From: Audiophile Land
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QUOTE(andrew9292 @ Oct 8 2010, 10:31 AM) [attachmentid=1823178] bro, can share how u make ASIO4ALL work? mind give me the dl link???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 |
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Oct 22 2010, 02:10 PM
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Senior Member
1,271 posts Joined: Sep 2008 From: Sometimes here, sometimes there.
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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 ....  |
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Oct 22 2010, 02:56 PM
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Senior Member
8,186 posts Joined: May 2005 From: Beaumont, Baile Ath Cliath, EIRE. 
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Exactly! Just click to play!
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Oct 22 2010, 05:43 PM
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Senior Member
1,649 posts Joined: Sep 2008
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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 offor 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 |
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Oct 22 2010, 05:47 PM
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Senior Member
1,242 posts Joined: Sep 2008 From: Cheras
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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.
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Oct 22 2010, 05:57 PM
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Senior Member
5,211 posts Joined: Feb 2005 From: Konohana
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QUOTE(Angel of Deth @ Oct 22 2010, 05:47 PM) 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 |
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Oct 22 2010, 09:41 PM
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Senior Member
1,242 posts Joined: Sep 2008 From: Cheras
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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.
Added on October 22, 2010, 9:42 pm QUOTE(Bonchi @ Oct 9 2010, 12:04 AM) 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 That is why AccurateRip exist. To ensure that your rip is perfect.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 This post has been edited by Angel of Deth: Oct 22 2010, 09:42 PM |
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Oct 22 2010, 11:44 PM
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Newbie
4 posts Joined: Oct 2010
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QUOTE(Najmods @ Oct 9 2010, 01:22 AM) 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. Hi Najmods, btw we're using same SC...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 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? |
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