# The foobar2000 help thread. Got problems or questions? Ask here!



## Tilpo

The foobar2000 help thread.

With reasonable frequency I get technical questions about foobar2000 from people all around the forum. While I love to help those people, usually it's simply off-topic.

Furthermore, I don't regularly check the forums for new threads, and I assume it's the same thing for many others. Therefore asking it in a thread like this will mean that your answer will be answered quicker too. 

*You can ask anything here related to foobar2000.*
Looking for a plugin? Ask here!
Got technical problems? Ask here!
Do you want a comparison of foobar2000 with other players? Ask here!
Do you want to discuss the sound science of bit-perfect playback? Heck, go ahead and ask here!


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## JoeinLA

I just started working with Foobar, based on all the threads on this forum. 
   
  My first noob question: recommended skins?  I'd like one that displays playlist and currently playing - the default skin, at least as I'm able to configure it, only show the "now playing" on the top and bottom of the windows (both positions are small and hard to see), and doesn't move the playlist to show the currently playing tune...
   
  Also, I've recorded most of my CDs using Windows Media Player at 320; but I'm reading here alot about FLAC and "lossless".  I may redo my CD's, but would love to hear that it's worth it before spending all that time   
   
  Also, FLAC recording seems quite complicated (whole CD with a track list vs. by track, etc.); is there a program for idiots that just does "record my tracks in lossless FLAC format please"...?
   
  Thanks!!!


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## Tilpo

joeinla said:


> I just started working with Foobar, based on all the threads on this forum.
> 
> My first noob question: recommended skins?  I'd like one that displays playlist and currently playing - the default skin, at least as I'm able to configure it, only show the "now playing" on the top and bottom of the windows (both positions are small and hard to see), and doesn't move the playlist to show the currently playing tune...
> 
> ...



If you want a premade skin, have a look around DeviantArt

If you want to do it yourself (which I prefer myself), then I suggest you read a guide on how to do it first. I can recommend this one.

Furthermore, it sounds like you're looking for a different playlist panel. I personally love ESPlaylist, but I have some others listed in my foobar2000 resource thread.


Regarding FLAC vs. MP3:
Flac is theoratically better, but in most cases the differences between Flac and 320 are inaudible. Unless you know _exactly_ what to listen for, you probably won't ever notice the difference. 
I personally like to get Flac instead of MP3 if I have the choice. If not only for the 'safe' feeling that I'm not missing out on anything. 

My advice is to not rerip everything from CD. That'll just be wasted effort. 
However, you can choose to rip new CD's on Flac instead of MP3 if you wish. Instead of Windows Media Player, most people here use EAC.
Please read this guide (and also the guide mentioned there) for using EAC with Flac for optimal results. 


If you have any more questions, I'd be happy to help.


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## imackler

This shows how little I know:
   
  If Foobar can play ALAC now that its become open source, can any DAC play those files as well since Foobar can? So I guess I'm asking does the data go from comptuer > foobar > dac or is foobar taken out of the flow when you're going from computer > DAC...
   
  My DAC has yet to arrive and I'm trying to get my computer ready to go...


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## Tilpo

imackler said:


> This shows how little I know:
> 
> If Foobar can play ALAC now that its become open source, can any DAC play those files as well since Foobar can? So I guess I'm asking does the data go from comptuer > foobar > dac or is foobar taken out of the flow when you're going from computer > DAC...
> 
> My DAC has yet to arrive and I'm trying to get my computer ready to go...




As long as foobar can play it, your DAC can play it.

Foobar2000, as well as any other thing that makes sound in the operating system (even system sounds), will send information to the sound drivers. 
The sound drivers will then mix everything together and send it to the DAC, or whatever audio output device you have selected.


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## imackler

Hello!
   
  What am I supposed to do with the data from accuraterip when burning tracks from cd into flac in foobar? My correction offset should be +6 if I'm reading the chart correctly. But that is data I don't know what to do with! Does accuraterip with foobar correct it automatically or do I need to reburn the tracks? 
   
  Thanks!


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## Tilpo

imackler said:


> Hello!
> 
> What am I supposed to do with the data from accuraterip when burning tracks from cd into flac in foobar? My correction offset should be +6 if I'm reading the chart correctly. But that is data I don't know what to do with! Does accuraterip with foobar correct it automatically or do I need to reburn the tracks?
> 
> Thanks!




I have never heard of AccurateRip, so I had to do a quick google. 

AccurateRip is a utility for the ripping software itself, and is separate from foobar. Once you have ripped the tracks with a program that uses the AccurateRip plugin, you can import them in foobar and you're done.
If you're asking how to use AccurateRip itself then I might not be able to help, since this is my first time hearing about it. 


Also note that 'ripping' is the term used when retrieving data from a CD and putting it one a PC, and 'burning' is used for putting data on a CD from a PC. You seemed to mix the two terms up.


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## AurorionWing

Hi Tilpo, I used iTunes to attach multiple images to my music files (mp3 and so on) but want to move to foobar. However, when I play my songs on foobar, the wrong album art shows up. It displays correctly in iTunes, WMP, and even Windows Explorer, so I'm pretty sure it's foobar that's the problem. I'd really love some help for this.
   
  Thanks!


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## Tilpo

aurorionwing said:


> Hi Tilpo, I used iTunes to attach multiple images to my music files (mp3 and so on) but want to move to foobar. However, when I play my songs on foobar, the wrong album art shows up. It displays correctly in iTunes, WMP, and even Windows Explorer, so I'm pretty sure it's foobar that's the problem. I'd really love some help for this.
> 
> Thanks!




If you navigate to the folder where the album is located, is there an image file for the album art? If so, what is it called (give a few examples if they are different).
Then go to options -> Display.
You'll see a box titled 'Album art'. Try playing around with the search patterns there. If you don't understand how they work, you can take a screenshot and I'll try to help you further. 

If there is no image file containing the album art in the folders then your foobar has a problem with embedded album art. It's very unlikely that this is the case, but if it is, I'd be happy to do a bit of googling to see if I can come up with a possible fix.


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## AurorionWing

Quote: 





tilpo said:


> If you navigate to the folder where the album is located, is there an image file for the album art? If so, what is it called (give a few examples if they are different).
> Then go to options -> Display.
> You'll see a box titled 'Album art'. Try playing around with the search patterns there. If you don't understand how they work, you can take a screenshot and I'll try to help you further.
> If there is no image file containing the album art in the folders then your foobar has a problem with embedded album art. It's very unlikely that this is the case, but if it is, I'd be happy to do a bit of googling to see if I can come up with a possible fix.


 
  Nope, I made sure that no images are in the folder with my album (even after un-hiding protected OS files). For safe measure, I copied song files individually to a different location (not the folder), but it still displays the wrong cover for foobar. It's so strange that other players show the right one :\


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## AurorionWing

This includes a sample file which shows the problem.
http://www.hydrogenaudio.org/forums/index.php?showtopic=97336


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## lg777

I think I have a similar problem with artwork in Foobar 2000.   I ripped about 300 CD's using EAC - FLAC and added album artwork.  The artwork is usually a jpg and named Artist - Album Title.jpg and is in each folder. 
   
  When I play them in Foobar, it doesn't recognize the album artwork but would have to manually add them which would take quite a long time.  I still have a few hundred more CD's to rip so I just want to make sure I can save myself some grief with managing album artwork.
   
  So the questions are:  When I rip to FLAC in EAC, is there a way to embed the artwork into the file versus a jpg file in a folder?  Or is there an easy way to get Foobar to look at the album folder for artwork?
   
  Thanks,


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## Destroysall

So what would be the proper rip settings to use with foobar2000?


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## Roller

Quote: 





imackler said:


> Hello!
> 
> What am I supposed to do with the data from accuraterip when burning tracks from cd into flac in foobar? My correction offset should be +6 if I'm reading the chart correctly. But that is data I don't know what to do with! Does accuraterip with foobar correct it automatically or do I need to reburn the tracks?
> 
> Thanks!


 
   
  Basically, AccurateRip offset settings are only required during the ripping process. Afterwards, it's just a data reference file for which drives with respective settings were used, as well as tracks that were ripped.


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## cb2222

Hi,

I was just wondering if you know of a way that Foobar can retrieve CD information automatically when I decide to listen to the physical CD instead of MP3/FLAC. I have hundreds of CDs that I don't have time to rip to FLAC format and would love to see the album information when I put it in my computer's dvd/cd drive. Is there a way that Foobar can do this automatically? The way it is right now, I have to do it manually and it is quite tedious. Any tips or ideas would be appreciated.


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## Tilpo

cb2222 said:


> Hi,
> I was just wondering if you know of a way that Foobar can retrieve CD information automatically when I decide to listen to the physical CD instead of MP3/FLAC. I have hundreds of CDs that I don't have time to rip to FLAC format and would love to see the album information when I put it in my computer's dvd/cd drive. Is there a way that Foobar can do this automatically? The way it is right now, I have to do it manually and it is quite tedious. Any tips or ideas would be appreciated.




You can automatically rip album information from freeDB and musicbrainz, though I don't know if their plugins work with CD's as well, but I think they should.
I'll take a look for you when I get home.


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## daigo

Are there any good lyrics retrieval and/or embedding plugins for foobar now?  I used to use one that doesn't seem to work for me anymore, and lost it when I had to rebuild foobar for my new computer.


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## Roller

Quote: 





daigo said:


> Are there any good lyrics retrieval and/or embedding plugins for foobar now?  I used to use one that doesn't seem to work for me anymore, and lost it when I had to rebuild foobar for my new computer.


 
   
  Doesn't this work for you?


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## Tilpo

daigo said:


> Are there any good lyrics retrieval and/or embedding plugins for foobar now?  I used to use one that doesn't seem to work for me anymore, and lost it when I had to rebuild foobar for my new computer.




I personally use Lyrics Grabber 2. It is excellent, but it is know to make people's foobar crash. Never had the problem myself, so I think it's worth a try at least.

Once I have put the lyrics in the tags of the music, I display them using foo_textdisplay, but I have tried other text display panels as well, and they work just as well.


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## Rhymenoceros

hard to describe, but i'm wondering if there's a way to lay out foobar so you can get all the interface options/playlists, etc laid over a large reproduction of the album art.  maybe have them slightly transparent so you can see the artwork but the interface is still clear.  has anyone seen someone with a configuration like this and if so can they point me in its direction?  not sure if i'm determined enough to learn how to configure foobar myself to the point where i can make this possible (it's probably simpler than i think, or at least it'd be simple for most people i'm assuming..).


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## Tilpo

rhymenoceros said:


> hard to describe, but i'm wondering if there's a way to lay out foobar so you can get all the interface options/playlists, etc laid over a large reproduction of the album art.  maybe have them slightly transparent so you can see the artwork but the interface is still clear.  has anyone seen someone with a configuration like this and if so can they point me in its direction?  not sure if i'm determined enough to learn how to configure foobar myself to the point where i can make this possible (it's probably simpler than i think, or at least it'd be simple for most people i'm assuming..).




As far as I know there is no plugin that does that. 
There is the possibility to make your entire playlist transparent and have a cool wallpaper behind it, but you can't make the wallpaper automatically change in that case.


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## cb2222

tilpo said:


> You can automatically rip album information from freeDB and musicbrainz, though I don't know if their plugins work with CD's as well, but I think they should.
> I'll take a look for you when I get home.




Yeah, I found those plugins and they are great when you want to rip CDs however, I was just hoping to find something that simply just grabs the information when I put a CD in to play. I know that Windows Media Player does it. 

Thanks for your time. I will post something here if I find a solution.


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## daigo

I'll give those lyric plug ins a try.


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## sonci

I recently moved my music files to NAS, and I can`t play .cue files anymore, I dont see them in the playlist,
  Nas hard drives are formated in EXT3?


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## Tilpo

sonci said:


> I recently moved my music files to NAS, and I can`t play .cue files anymore, I dont see them in the playlist,
> Nas hard drives are formated in EXT3?




Could you try navigating to the NAS manually in your file browser, and then drag and dropping the cue file in a foobar2000 with an empty playlist?
If nothing happens, see if the console says anything. (View -> Console)


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## Roller

Is the NAS running with a drive letter? Also, why are you running EXT3?


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## sonci

Its ok, It have been exluded .cue in Shell Extension, I havent touched it anyway,
  Thanks, 
  I have mapped a network drive, and put audio files in there, don`t know nothing of UPNP AV or itune server etc..
  Why not EXT3, I think my nas doesn`t support ext4..


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## Rhymenoceros

Quote: 





tilpo said:


> As far as I know there is no plugin that does that.
> There is the possibility to make your entire playlist transparent and have a cool wallpaper behind it, but you can't make the wallpaper automatically change in that case.


 
  thanks for the response!  i wonder if there's any music player that's like that, or at least puts a nice emphasis on artwork without having to do heavy tweaking.
   
  i like foobar from what i've used of it, but it seems to have a high barrier for entry and even when i DID get it configured the way i wanted it it seemed to just magically fail on its own after a few weeks of use (very strange, i changed ABSOLUTELY NOTHING and it suddenly doesn't like any of my configs/plugins.  also had the same problems occur whenever i try to tweak media player classic.  really frustrating.).  maybe have to dabble in programming...


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## Roller

Quote: 





rhymenoceros said:


> thanks for the response!  i wonder if there's any music player that's like that, or at least puts a nice emphasis on artwork without having to do heavy tweaking.
> 
> i like foobar from what i've used of it, but it seems to have a high barrier for entry and even when i DID get it configured the way i wanted it it seemed to just magically fail on its own after a few weeks of use (very strange, i changed ABSOLUTELY NOTHING and it suddenly doesn't like any of my configs/plugins.  also had the same problems occur whenever i try to tweak media player classic.  really frustrating.).  maybe have to dabble in programming...


 
   
  I recommend doing regular backups to configs, data corruption can occur, and backups are always good for safekeeping if that occurs. I also do the same with MPC-HC, with a single .ini requiring backup.


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## Tilpo

roller said:


> I recommend doing regular backups to configs, data corruption can occur, and backups are always good for safekeeping if that occurs. I also do the same with MPC-HC, with a single .ini requiring backup.




Which is also where foo_jesus comes in.


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## Roller

Quote: 





tilpo said:


> Which is also where foo_jesus comes in.


 
   
  I thought you were joking, I didn't remember the component being called that way  But yes.


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## KamijoIsMyHero

I am getting curious just as to what the WASAPI is for? What does it do? Microsoft description is quite little vague
   
  I will have the Schiit Modi/Magni soon and will use it with my laptop running foobar


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## xnor

Exclusive access to your audio interface.


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## KamijoIsMyHero

to improve sound/clarity?


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## pabloaugustus

just wait until PowerAmp releases a win7 version..they are so dumb porting it to other portables first. 
   
  I've been a winamp guy, a foobar guy, now I hate em all except Power AMP...Exact Audio Copy (EAC) can do what are describing....it gets the CD info not just the ID tags.


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## KamijoIsMyHero

what?


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## xnor

Quote: 





> WASAPI is a new audio output method introduced in Windows Vista; among other things, it provides an exclusive mode that allows applications to take full control over soundcard's resources (muting any sounds played by other applications) and play unaltered bitstream without passing it through the Windows mixer.


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## KamijoIsMyHero

I wouldn't post on a help thread if I didn't try to first understand what I am trying to understand. I wondered why it was an extension for foobar when it was already part of Windows OS which is why I got curious as to what it is and how it may affect my system. Quite a dick move to just give a definition.


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## xnor

You could have written that you read the wiki page. Many people that post here do in fact post question before doing research themselves.
   
  The default sound API is DirectSound (DS) which on >Vista is implemented as a layer above WASPI _shared _mode. All sounds form all applications will be mixed together. For this to work there has to be a common format, which is configurable in the control panel. If there's a mismatch the Windows audio engine (aka Mixer) will resampler/requantize.
   
  WASAPI _exclusive _mode on the other hand gives exclusive access and therefore the application can choose the format that matches the format of the track that is playing (e.g. 16/44.1 or 24/96 ..). To use WASAPI exclusive you need the component. I guess it will be built into foobar2000 in future versions.


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## KamijoIsMyHero

cool I didn't knew windows uses a mixer like that


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## Freekman

I have searched around on how to get my date added info from itunes into foobar I have found
  this
  http://www.hydrogenaudio.org/forums/index.php?showtopic=97728
  this
  http://www.hydrogenaudio.org/forums/index.php?showtopic=77767
  and this
  http://www.hydrogenaudio.org/forums/index.php?showtopic=71536&st=0&p=631310&#entry631310
   
  I don't know if it just because I am overtired, or just not as savvy as I thought I was, but I cannot figure out what they are telling me to do. I seems as though all the solutions are just a do this and you know the rest kind of deal. Problem is I don't know the rest. Can anybody give me a simple step by step guide on how to do this. Also anytime I try to open the itunes library.xml it crashes what ever editor I am in. I really like the layout, quality and lite-ness of foobar but this is keeping me from using it


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## Aurowave

I have an issue with facets/foobar I didn't have on my last install. My songs are being displayed individually, and aren't being grouped into their respective albums (I get the same album picture displayed 20 times instead of once). My albums are in folders so that's not an issue.


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## Tilpo

aurowave said:


> I have an issue with facets/foobar I didn't have on my last install. My songs are being displayed individually, and aren't being grouped into their respective albums (I get the same album picture displayed 20 times instead of once). My albums are in folders so that's not an issue.




Could you post a screenshot?

Also, if I recall correctly, facets doesn't have a grouping function in any case, so you may be confusing it with a different playlist display component.


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## Aurowave

Maybe it was Columns UI, i'll report back on that.





EDIT: I also have playback statistics. I cannot get Columns UI to work with Facets, so I doubt that's the problem.


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## Destroysall

I have a question which regards ripping CDs with Foobar2000. Is there any way to rip directly to ALAC as you can to FLAC?


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## hulja

Hi, i don't know where else to put my question. I am using foobar2k with WASAPI enabled, ALAC's and I am using Fiio e07k and Grado sr325is. I also use Akg k450. Yesterday I noticed kind of rattling,cracking sound coming when playing deep bass songs. First I have feared it was grattle, but plugging AKG's and IEM's delivered same result. It was also present without Fiio, and not present if I connected headphones to iPhone. It is the same file that I have on pc. Now, I tested it on iTunes and it did that again so sorry if question is missplaced. It is masked more when all frequencies are engaged thus it can be heard more at more quite, but bass heavy parts, like the beggings of songs. Music is dl-ed from bandcamp directly as ALAC. Also not every song with deep bass produce the noise. It's driving me nutts! I just can't get around it! Please help.


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## dakanao

Does anybody know where I can get a skin that looks kind of like Windows Media Player 12? I did my research, but couldn't find anything working.


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## ejong7

hey guys I'm new to foobar and i was just wondering does the iPod Manager automatically converts FLAC into ALAC when set to convert to playable format? Or does it convert to AAC?


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## HPiper

When ripping a cd to flac file, it doesn't display any album/track information. is there something I need to enable or change to get it to do that?


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## xnor

Right click album - tagging - get tags from freedb.
   
  There's also a plugin for musicbrainz which I prefer (link).


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## akinari-kun

Digital volume control: 

Due to my studio monitors having the volume knobs on the rear of each unit, I must use digital volume attenuation as adjusting my monitor volume would be extremely inconvenient requiring me to rotate each speaker around to adjust the volume.

Is it best to keep:

Windows volume 100% and adjust foobar volume? 

Or 

foobar volume 100% and adjust Windows volume? 

Which option would lead to the least bit degradation, assuming that this bit degradation is actually perceivable when playing 16 bit 320kbps mp3 files or FLAC or 16 bit CDs (which probably isn't perceivable anyway)?

I'm running an Auzentech X-Meridian 7.1 2G which is giving me insane distortion when Windows volume is anything more than 60% (tried pretty much anything and everything to diagnose, concluded with a bad cap slowly degrading) which will soon be swapped out for an ODAC or Modi on Windows 7. 

I just want to know if there's a right or wrong way, or a more preferable or correct way to adjust digital volume given that Windows and foobar both have volume adjustment.


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## xnor

Once you have an interface or DAC that doesn't distort near 100% I'd set the monitor knobs a bit above what is a comfortable level for you and control the system volume.
   
  foobar2000 vs. windows volume control: it doesn't matter if the DAC doesn't have a digitally controlled analoge volume control, which ODAC/Modi don't have afaik. Else I'd prefer the system volume control.


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## akinari-kun

Ah, I see then. I'm just wondering if lowering Windows volume or lowering foobar volume instead, one or the other would allow for a lower noise floor.


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## xnor

I see no reason why they should not be equal with the DACs above.


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## akinari-kun

I think I may have been misunderstood. What I meant was, since my DAC does not have volume control built in, is it better to lower volume with the Windows volume in the lower right corner of the screen in the taskbar, or lower the volume through my audio player volume slider in foobar to preserve the best quality, if it makes a difference at all?


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## xnor

I don't see any difference in that case.


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## Tilpo

Do whatever you find most convenient. I'd be surprised differences are anything close to audible.


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## OiCU812

Have an older computer running on xp, installed ASIO and ASIO4all v2, I 'think' it's bypassing the extra processing. It does show asio4all under device and the green box on the taskbar. How can I check to be certain? 

 Have a newer computer running windows8. Have installed WASAPI, it shows in the installed components window, even checked the components folder, it's there but under playback>outputs>device>it's not there. Computer volume and sounds play through the external dac which also indicates no wasapi? Where am I going wrong???

 This would be feeding a modi and/or by later next week an aune x1 through usb.


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## imeem

what's the point of up-sampling if it does not improve audio quality?


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## xnor

Upsampling may be used in audio-plugins that do signal processing, especially those with nonlinear effects like compression. Some do a bad job at it, so you can upsampling before, then let the plugin process the higher rate signal, then downsample again.
  
 It can also be used to kinda "bypass" the filter in your DAC or raise the sampling rate to a frequency where the DAC performs a bit better. Both gains are small but measurable.
 Resampling to a fixed sampling rate can be used to change the rate of a file to a rate your DAC likes (for example some tracks have 32 kHz but many DACs only support 44.1 kHz and up) or to bypass the Windows audio engine's resampler.


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## kcazbarach

oicu812 said:


> Have an older computer running on xp, installed ASIO and ASIO4all v2, I 'think' it's bypassing the extra processing. It does show asio4all under device and the green box on the taskbar. How can I check to be certain?
> 
> Have a newer computer running windows8. Have installed WASAPI, it shows in the installed components window, even checked the components folder, it's there but under playback>outputs>device>it's not there. Computer volume and sounds play through the external dac which also indicates no wasapi? Where am I going wrong???
> 
> This would be feeding a modi and/or by later next week an aune x1 through usb.


 
  
  
 for ur older computer and asio, the greenbox shows up on the task bar, and the only way it isn't functioning is if you click on that greenbox and it shows and X or something to indicate it's not working.
  
 I guess you can be certain by starting foobar with asio and trying to play something on youtube or any other audio. (it shouldn't work in conjunction).
  
 Your asio set up seems fine from how you explain it.
  
 I can't help you on your 2nd problem though, that's stumping me as well.


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## OiCU812

kcazbarach said:


> for ur older computer and asio, the greenbox shows up on the task bar, and the only way it isn't functioning is if you click on that greenbox and it shows and X or something to indicate it's not working.
> 
> I guess you can be certain by starting foobar with asio and trying to play something on youtube or any other audio. (it shouldn't work in conjunction).
> 
> ...


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## kcazbarach

if there is a red X, that indicates the asio drivers aren't in use.
  
 sometimes (I'm on windows 7, i realize you're using xp).
 I'd have to close firefox before the asio drivers would start because I might've been using pandora/youtube or anything that can have audio.
  
  
 and sometimes I'd have to restart foobar, but if you're looking to listen to music while browsing (via ASIO). STart up foobar first and play a song (this isn't as big of an issue with WASAPI in my experience) and then open the browser.


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## intheturese

If not only for the 'safe' feeling that I'm not missing out on anything.


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## KamijoIsMyHero

does anybody know why windows can affect the volume when I am playing music in foobar? I thought foobar automatically disables it and I can only control the volume using my amp. I never had this problem before and not sure when it started. 
  
 set up: Laptop->windows 7-> latest ver. foobar2000 -> ASIO -> USB-> AudioGD Compass 384 DAC/AMP-> headphones


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## PiotrTheGreat

Hey guys! Got a little issue with Foobar
  
 The story is such: I used to just drag and drop .cue files into Foobar and they would automatically split into separate tracks. However, when I then tried to drag these tracks into the folder of my MP3 player, they remained .cue files, instead of the .flac files the .cues should've been associated with.
  
 What am I doing wrong? I ended up caving and getting Medieval Cue Splitter, but I know there's a way to get the same result using only Foobar.
  
 If someone would guide this layman through how to do it via Foobar, I would be quite grateful


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## DonaldDraper

Hi, can you set Foobar to play just a portion of a song ? 



 
 e.g.  i want to start playing a track 12secs in.
I had been using this feature occasionally in iTunes and it is quite easy to set the start and stop points in a song, but i can not find the equivalent in Foobar.  
  
 can anybody please help?


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## adri123

Hi all,
  
 I'm using foobar2000 with Facets, How can I show album art when I have an album with multiple CDs?
  
 My folder organisation is like this:
  
 Artist
 |
 |_Album
    |
    |_CD1
    |   |
    |   |_Tracks...
    |
    |_CD2
       |
       |_Tracks...
  
 I have tried putting the cover.jpg picture in the Album folder and in the CD1 and CD2 folder but it never shows.
 The covers of albums with only one CD (one folder) show normally. The image is in that single folder and that's easy.
  
 The tracks are in FLAC format.
 I tag the files with Mp3tag.
 The discnumber tag contains 1 and 2 accordingly.
 I've tried putting 1/1 and 2/2 but no positive change.
 I would like to keep the discnumber tag to be 1 and 2.
  
 In Facets the album appears as one facet but with no image (no cover).
 If I put all the tracks (of both CDs) in one folder, then it works of course.
 But I want to keep one folder per CD.
  
 If any of you have managed to do this please tell me how.
  
 My foobar version is 1.3.1


----------



## adri123

adri123 said:


> Hi all,
> 
> I'm using foobar2000 with Facets, How can I show album art when I have an album with multiple CDs?
> 
> ...


 
  
 Here is the solution:
  
 On the Menu, click File > Preferences.
 Once there, you have a menu-tree on the left side of the window.
 Click on "Display".
 On the right side of the window there is an "Album art" section.
 Click the "Front cover" tab.
  
 There you can see a list of search patterns. Something like this:
  
 front.jpg
 cover.jpg
 %filename%.jpg
 %album%.jpg
 folder.jpg
  
 Now you have to tell the program to look in the "parent" folder.
 So if your album art is called *cover.jpg* you add this line at the end of the list: 
  
*..\cover.jpg*
  
 (exactly as you see it, with the two dots included)
  
*..\ **means parent folder*
  
 so it will look for a file named cover.jpg in the parent folder. That is, in my structure, the Album folder.
 Of course if your album art file is called something else, for example folder.jpg, then you should add the line:
  
*..\folder.jpg*
  
*Good luck everybody ! *


----------



## whirlwind

My artist pictures no longer work.....they use to download them from last fm
  
 That has stopped working....any suggestions


----------



## PiotrTheGreat

Still got a problem with Foobar that I'm hoping you folks could help me out with.

 So, this is what I've got in the folder




 From my understanding of how to have Foobar split up a .cue file, I'm supposed to drag the .cue file in this folder into a new playlist, select all the files, hit convert, and convert to FLAC, yes?

 Well, here's what happens when I do just that





 Please tell me this is just me not doing something, and not Foobar itself. I don't want to go back to Medieval CUE Splitter, and CUE Tools melted my mind (since I never could figure out if I could even separate .cue files that WEREN'T on a CD)


----------



## A_Man_Eating_Duck

Can you open up the CUE file in notepad and copy and paste everything in to your reply.

I have a feeling that the file referenced in the CUE doesn't have the correct file extension.


----------



## OICWUTUDIDTHAR

does anyone know if there is a way to make a 1-2 second delay between when a track starts playing and when its clicked?


----------



## A_Man_Eating_Duck

Give this a try.

http://www.foobar2000.org/components/view/foo_dsp_silence


----------



## OICWUTUDIDTHAR

works great thx!
  
 does this effect bitperfect play back though? (asio, wasapi)


----------



## A_Man_Eating_Duck

It shouldn't do since it's just adding silence between the tracks.


----------



## jaed

Hi all, 
 I edited the layout so i only have 3 columns displaying Artist, Album and Track, respectively. And the toolbar.
 The thing is: im using Facets and i´d like to increase the size of the album covers, is there a way to do this?
 My foobar is 1.3.2
 Thank you.


----------



## A_Man_Eating_Duck

You can increase the album art size by put the mouse cursor over the art work and hold the CTRL key and scroll the mouse wheel.


----------



## koshtramba

Been wondering about this for ages, so I thought I'd ask:
  
 How does increasing or decreasing the buffer size in Foobar affect the actual output? Do you guys alter the default value, or leave it as it is?


----------



## A_Man_Eating_Duck

The basic rule is to leave the buffer set to the default unless you're having audio drops while playing, If you are having dropouts, then increase it slowly until the dropouts stop. 

Audio quality wise there no difference between a small buffer or big buffer, all it's doing holding data in memory. larger buffer, more memory used. 

e.g. a 3000ms buffer makes my foobar install use 54MB. a 30000ms buffer makes my foobar install use 90MB.


----------



## koshtramba

Got it, thank you. But the buffer isn't set to the lowest setting by default. What would be the sort of scenario when you'd get an improvement by lowering the buffer value, instead of increasing it? When would that be necessary?


----------



## A_Man_Eating_Duck

You don't need to lower the buffer, all you are doing is running more risk of a buffer under runs when the computer can't keep up for a second or 2. 

Sometime setting a very very low butter is advised for WASAPI playback when the normal buffer length causes glitches. 

So we are once again back to If it's working fine don't mess with the buffer


----------



## heishiro

Is it possible that when i click my folders on _*All Music*_ the list of all songs inside the folder will go automatic on the playing window?


----------



## A_Man_Eating_Duck

Yup, in the foobar2000 preferences > Media library | Library viewer selection playlists | Tick enabled and tick Activate when changed.


----------



## heishiro

^ it works!!! many many thanks!


----------



## Boffy

I have a problem. Foobar2000 shows wrong album art. I deleted all the embedded album art and re-embedded them. Even after that, Foobar2000 shows wrong album art on some of the albums. I searched for a fix to this problem for days and I still can't find a solution.


----------



## A_Man_Eating_Duck

So when you delete the embedded art work foobar2000 no longer show any art work for the files? or is it still showing the incorrect art work? 

have you checked the folder where the files are for any JPG's that have been set to hidden (you'll need to set the file manager to view all files in the options).


----------



## Boffy

Foobar2000 is showing incorrect art works.
  
 As for the files, I did go to Windows Explorer and set the settings to view all files. Here is the thing, I basically throw in all my music file into one big folder so it is very disorganized. In the folder, I do not have any JPG or image files, just MP3 files. I embedded the album art by going to Foobar2000-Tagging-Attach Album Art. All JPG and Album Art are in a separate folder.


----------



## A_Man_Eating_Duck

Are you sure there are no jpg's that follow this naming scheme in that folder?

front.jpg
cover.jpg
%filename%.jpg
%album%.jpg
folder.jpg

Another place to look is if you go in to the Preferences > display and see if you have a path set for the stub image path.

in Foobar2000 right click on one of the files that shows the incorrect artwork and select tagging > manage attached pictures and see if there is more than 1 art work attached.


----------



## Boffy

Yes, there are no other files in the folder except for .mp3 files.
  
 When I go to Preferences-Display-and Stub Image Path, it tells me to select a file not a folder.
  
 Tagging-Manage Attached Pictures and there is only 1 Album Art, which is the correct album art. However, Foobar2000 does not display that correct Album Art.


----------



## A_Man_Eating_Duck

running out of ideas. could you post a picture of your current layout.

Maybe something is up with the tags.

can you make a backup of the file and remove all the tags and then embed the art work again.


----------



## Boffy

This is my setup.
  

  
 As for your suggestion, I have already tried that. I deleted/removed all tags and embedded all the art work again. That did not fix the problem.


----------



## A_Man_Eating_Duck

Any chance you can upload one of the songs to cloud storage and PM me the link so I can test?


----------



## Boffy

Unfortunately, I do not know how to use Cloud Storage. Is there any other way I can send you or upload a file?


----------



## A_Man_Eating_Duck

you can sign up for a free mediafire account and upload the files there. once uploaded you share the file and it will give you a link for me to access the file. PM that link to me.


----------



## zomgliekwtf

Lots of great info in this thread!


----------



## A_Man_Eating_Duck

BTW the workaround for Boffy's problem was to do the following.

Foobar2000 preferences > Advanced | Display > Album Art > Embedded vs external | set this to Prefer embedded.


----------



## castleofargh

most usefull duck I've ever seen. thanks for your relentless help.


----------



## adisib

1. How can I set my output to WASAPI Shared, as opposed to WASAPI exclusive or DirectSound?
  
 2. Is it possible to display the current song progress in digit form anywhere other than the bottom left corner where it is inconvenient to see? Near the seekbar is preferable.
  
 3. Is it possible to change the style of the seekbar without changing skins? Or at least a skin that will only change a minimal amount of things such as the toolbar?
  
 4. Is there an option or plugin anywhere that will automatically start a playlist when I open it?
  
 5. Is there a way to make a hotkey to restart a song other than a hotkey to seek back 10 minutes?
  
 6. Does the SoX resampler (http://www.hydrogenaud.io/forums/index.php?showtopic=67373) resample playback when necessary, or is it a tool for permanently resampling a file?
  
 7. Are there alternative Windows file icons available other than the default foobar ones?
  
 8. Is there a way to load files or folders into a playlist with Windows navigation instead of having to drag and drop it or using search?
  
 9. Is there any way to add a UI element that isn't on the popup list? (i.e. This equalizer plugin: http://www.foobar2000.org/components/view/foo_dsp_xgeq as opposed to the built-in equalizer which doesn't show numerically how far I've moved the slider but can be added easily as a UI element)
  
 10. Is there an option or plugin to fade volume muting and unmuting? (NOT fading between songs, but a short and smooth volume change fade or fade to pause/stop)
  
 11. What is the generally recommended crossfeed plugin?


----------



## A_Man_Eating_Duck

I'll try and answer a few of these 



adisib said:


> 1. How can I set my output to WASAPI Shared, as opposed to WASAPI exclusive or DirectSound?


 You can't, the plugin only does exclusive mode. Use DirectSound to allow other applications to make sound. 



> 2. Is it possible to display the current song progress in digit form anywhere other than the bottom left corner where it is inconvenient to see? Near the seekbar is preferable.


 I'm pretty sure this can be done using the Text Display 1.1 beta 1 plugin



> 3. Is it possible to change the style of the seekbar without changing skins? Or at least a skin that will only change a minimal amount of things such as the toolbar?


 I don't think this is possible. EDIT: Would the Waveform Seekbar 0.2.45 plugin do what your asking?



> 4. Is there an option or plugin anywhere that will automatically start a playlist when I open it?


 I don't think there is a plugin, but i'm pretty sure that there are swtiches for the foobar2000.exe that will allow you to do this.



> 5. Is there a way to make a hotkey to restart a song other than a hotkey to seek back 10 minutes?


 You could make a new keyboard shortcut for Main > Playback > Play that will restart the current playing song. 



> 6. Does the SoX resampler (http://www.hydrogenaud.io/forums/index.php?showtopic=67373) resample playback when necessary, or is it a tool for permanently resampling a file?


 There are options in that plugin (modded version) that allows you set certain sample rates not to resample or set certain sample rates to resample. EDIT: if you are using the built in PPHS resampler (which is very good) then it will only resample any sample rates that are not the same as the target sampling rate setting. 



> 7. Are there alternative Windows file icons available other than the default foobar ones?


 I'm sure there are if you put your google shoes on.



> 8. Is there a way to load files or folders into a playlist with Windows navigation instead of having to drag and drop it or using search?


 I think there is a limit to how many files can be selected and right click enqueued at once. Also look in the Foobar2000 Preferences > Shell integration to enable Folder context menus. You should really start using the media library for this type of stuff.



> 9. Is there any way to add a UI element that isn't on the popup list? (i.e. This equalizer plugin: http://www.foobar2000.org/components/view/foo_dsp_xgeq as opposed to the built-in equalizer which doesn't show numerically how far I've moved the slider but can be added easily as a UI element)


 sorry can't help here



> 10. Is there an option or plugin to fade volume muting and unmuting? (NOT fading between songs, but a short and smooth volume change fade or fade to pause/stop)


 using DirectSound as output allows you to do this. I think the other output methods are designed to be bit perfect so they don't have the option to fade out.



> 11. What is the generally recommended crossfeed plugin?


I can't help here either


----------



## castleofargh

I can do 11. \o/ I'm such a star. I can at long last pretend I knew something the Duck didn't !!!!!!!! 
	

	
	
		
		

		
			




 http://bs2b.sourceforge.net/download.html  that's what I use after trying a few crossfeeds. it's subtle with 3 presets and you can tweak a little if you want to.
  
 else if you're really curious about what you can do, I suggest xnor's crossfeed http://www.hydrogenaud.io/forums/index.php?showtopic=90761
  
  
 I don't think the rest would bring much to the table.


----------



## SunTanScanMan

I get the following message when playback output is set to "WASAPI (event): SPDIF Out (4-USB2.0 High Speed True HD Audio)".
  
 "Unrecoverable playback error: Invalid argument"
  
 I had been using WASAPI event with digital coaxial cable before, and switched back to USB. Windows sound settings has been adjusted to USB as default.
  
 Music plays fine when playback out is set to "DS: SPDIF Out (4-USB2.0 High-Speed True HD Audio)". Selecting audio output to WASAPI exclusive mode-USB on qobuz music streaming service, presents no problem either.
  
 I am using Windows 7.
  
 I would like to play music via WASAPI event - USB on foobar. What have I missed settings-wise?


----------



## castleofargh

suntanscanman said:


> I get the following message when playback output is set to "WASAPI (event): SPDIF Out (4-USB2.0 High Speed True HD Audio)".
> 
> "Unrecoverable playback error: Invalid argument"
> 
> ...


 

 I'm really not a pro with spdif, but what I think I know(feel the confidence of the guy ^_^) brings 2 points:
 -wasapi event is made so that the DAC "asks" for packets and kind of decides the timing. spdif can't use that as it's a continuous stream of stuff. does it work with push? if yes, wel use push ^_^.
 -maybe a speed problem? I know spdif can have limits with heavy dolby signals(I ask because I wonder what that "high speed true hd audio is?), so it's possible that going through DS the mixer takes it all down to whatever you have set in windows default ouptut, so the data stream is now small enough to pass through spdif. something that doesn't apply as soon as you tell foobar to use wasapi. 
 -maybe just that you go with some 24bit signal and have 16bit output selected (on the screen where you select wasapi so I guess you would have seen it).
  
  
 edit: the observant guy might notice I make 3 points. that's just me giving 150%


----------



## SunTanScanMan

castleofargh said:


>


 
 Hey castleofargh thanks very much for taking the time 
	

	
	
		
		

		
		
	


	



  
 - When I'm using the coaxial cable the audio output is set to "WASAPI (event): Digital Audio (S/PDIF) (High Definition Audio Device)" and it works flawlessly. So this would mean that it's not a SPDIF problem?
  
 - "high speed true hd audio" is the default name given to the playback device in my Windows sounds. I had to download a usb driver (Cmedia) for my DAC, the name could have come from there.
  
 - I disabled dolby along with DTS Audio and Microsoft WMA Pro Audio
  
 - All my music files are 16/44.1 CDs copied to ALAC format. Not interested in 24-bit.
  
 * I switched to my Microstreamer, in the same USB port. I am able to play music with foobar playback output as "WASAPI (event): HRT Microstreamer"
** Yes it works with WASAPI push. I'd wanted it as (event) as I'd read it was the optimum setting for foobar. Guess it does not matter much.*
*Would I be correct in concluding that it's my DAC that does not allow WASAPI event via USB?*


----------



## castleofargh

suntanscanman said:


> castleofargh said:
> 
> 
> >
> ...


 

  
 USB can use several kinds of protocols for the computer to "talk" to the DAC it's usually an automated transparent process, but for advanced stuff(often with DAC doing more than 24/96), you can need a dedicated driver and some special settings. at that point not knowing what the driver is, your guess is as good as mine. 
 if you got what you want from push, I'd use that and don't second guess it. even if event is an extra security for some DACs to decide on the streaming speed, there is nothing to say that the speed imposed by the drivers or by selecting "push" on the computer will do anything wrong. and even then, if there is a buffer and a clock in the DAC, chances are that both options wouldn't matter as then the DAC would most likely reclock the streaming itself and don't care how the signal came from the PC.
  
 as a side note, if it's ok with your DAC, I would recommend to set the output in wasapi to 24bit(and that works for everybody that doesn't have something like an old NOS DAC limited to 16bit). it costs really nothing but it gives a bigger place for the bits to move up and down if you use any digital volume control before the DAC. even by being overly dramatic and pessimistic, it should give you at least 20db to play with to set the volume on foobar without any danger of losing the least significant bits of your track.


----------



## Peti

Finally I just found this thread. I have two specific technical questions. First: in order to change buttons on foobar you need to use the column plugin? There's no way to change buttons without that?
  
 And second, I am having a really hard time make my foobar to play back sacd material. I use O2/ODAC if it does matter. Anyone could help me please?
  
 Thank you in advance and let's not die this superb thread!


----------



## The Other One

I’ve been trying to get Foobar to work with my daughters Ipod Touch.  I can connect my Nano with no problems but hers comes up as “No Ipod found” when it’s connected. 
Her computer is running windows 7 x64.  Foobar is version 1.36 set as a portable player.  Her Ipod Touch 2nd generation running IOS 4.2.1. The support page says it’s supported
I have downloaded and installed the following Apple Mobile Device Support, Apple Application Support versions 10.5 and foo_dop.dll. I also checked the enable mobile device support under preferences. I also tried adding Quicktime as well as doing a full I-Tunes install. I latter removed Quicktime and I-tunes.
Am I missing something?  Thanks


----------



## asilker

oh my goodness what a great thread idea.
  
 I'm so sorry if this is a redundant question, but I haven't been able to find it anywhere else so here goes:
  
 How can I change the appearance of the toolbar, seek bar panel, and button panel? The grey is functional but I would love something a bit more sleek 
	

	
	
		
		

		
			




  
 If I can get those parts customized I might be good for a bit


----------



## Sodafish

Hi all, brand new member here. So much great info on these forums 
  
 I have quite a specific inquiry about foobar regarding use with external USB DACs. I just recently got the iFi Micro iDSD (which is fabulous btw), and am having an issue with foobar where, upon a track transition involving a change in PCM sample rate or format type (PCM to DSD, or vice-versa) the first half second or so of the track is cut off. This appears to happen regardless of the buffer settings I use within the iFi driver or foobar's output settings. As I understand it, this is due to the hardware having to re-sync at the new sample rate, correct? JRiver has a setting specifically to address this: "play silence at startup for hardware synchronization", which solves the issue. Is there an equivalent functionality achievable within foobar (perhaps via plugins)? I have my whole library organised/tagged with foobar, and would prefer to stick with that if possible, but this problem is very annoying.
  
 Any help would be greatly appreciated. Thanks!


----------



## asilker

Bump! ( I believe we have four questions needs helpin')


----------



## Sodafish

peti said:


> Finally I just found this thread. I have two specific technical questions. First: in order to change buttons on foobar you need to use the column plugin? There's no way to change buttons without that?
> 
> And second, I am having a really hard time make my foobar to play back sacd material. I use O2/ODAC if it does matter. Anyone could help me please?
> 
> Thank you in advance and let's not die this superb thread!


 
  
 I can answer your second question. Your DAC won't decode DSD, so you need to have foobar transcode it on the fly to PCM.
  
 You will need the foo_input_sacd plugin from the zip file here: http://sourceforge.net/projects/sacddecoder/files/
  
 Extract the zip's contents, then in foobar go to preferences > components > install button > browse to the foo_input_sacd file and select it... foobar will restart.
  
 Now for the transcoding part. Your DAC supports up to 96 KHz speeds, so you should output at 88.2 KHz (multiples of 44.1 are best for DSD transcodes).
  
 Go to preferences > tools > SACD, and set:
  
 - ASIO driver mode to PCM
 - PCM volume +0dB
 - PCM sample rate 88200
 - DSD2PCM Mode to whatever you think sounds best (this is where you select the low pass filter).
  
 You should now be able to play your DSD files.


----------



## Peti

Thanks man,
  
 but in the mean time I figured that out!
  
 But, again, thanks for the effort replying!


----------



## Giogio

Hi guys, I wanted to ask if you have any idea of which criteria should I follow for the order of the plugins.
 I have noticed that the order completely change the result.
 In this moment I am mostly interested in the following plugins:
 Resampler (to resample my 44100 to 48000 and match with the settings of the audio card)
 Programmable Reverb
 Dynamic Compressor
 Advanced Limiter
 Graphic Equalizer
 Dolby Headphones Wrapper (but I still must find a recent dll for dolby headphones)
 Channel Mixer and Matrix Mixer
  
 I also wanted to know which Plugins are the best to EQ, to boost the bass, to increase  the soundstage, to improve the sound, and to normalize/anti-clip/limiter/compressor.
  
 Thanks


----------



## castleofargh

giogio said:


> Hi guys, I wanted to ask if you have any idea of which criteria should I follow for the order of the plugins.
> I have noticed that the order completely change the result.
> In this moment I am mostly interested in the following plugins:
> Resampler (to resample my 44100 to 48000 and match with the settings of the audio card)
> ...


 

 I would suspect some DSP to be made to run at a given sample rate. but I have no idea if there is a pre-conversion included in the DSP, or if it would just not work(thus giving a different sound depending on when you change the rate? or work but making a different sound because it would apply calculation onto the wrong sample rate??? I'm really too much of a noob in that domain, but maybe the sound differences come from that??????
 (you might guess from the number of question marks, that I'm really confident about this^_^)


----------



## Giogio

I did not think about that, so you already have at least some interesting ideas 
 I was thinking more about the fact that there is an order to place these things also in a real studio, physical effects.
 There is a logic, it is just different if you apply EQ to something compressed or compress something EQed, if you reverb before EQing or EQ before reverbering.
 I know that sound technicians know these things.
 I just am not one of them


----------



## castleofargh

well aren't we 2 overly confident people? ^_^
  in your case, I would probably go with compressor/limiter/EQ last, that way you have real control over the result, because I don't know if something compressed then going through dolby and the matrix stuff, couldn't again reach higher dynamics? but then can't he compressor ruin in part the 3D FX? I have no clue.
 and the EQ last simply because there is little point in EQing beforehand.
 for the resampler, it's back to my previous post and unwarranted suspicions ^_^.


----------



## virgopunk

Hi, hopefully someone may be able to give me some advice. I have a pretty decent laptop with Realtek ALC892 soundcard (optical and/or HDMI output) plus Foobar2000. I recently got a 24/96 5.1 ISO of a Beach Boys album and whilst I can hear it across my 5.1 setup ok I'm not 100% sure I've everything config'ed as good as it could be. I've heard that 5.1 HD files can only be played across HDMI as the optical SPDIF doesn't have the bandwith. Is this true?

 Additionally, should I be using a bit perfect plug-in such as WASAPI or ASIO?
  
 It seems there's so many areas to fiddle with the sound output I just don't know if I've got the cleanest route through to my 5.1 amp.
  
 I'd be extremely grateful for any help or advice.
  
 Thanks.


----------



## A_Man_Eating_Duck

Yes optical has a limit of 2 channel PCM. You can send 5.1 through optical if it's compressed as AC3 so the amp will convert it.

Just install the WASAPI plugin and use the hdmi output. if WASAPI throws any type of error when playing it just telling you that the soundcard\dac\amp doesn't support that bit depth, sample rate or amount of channels.


----------



## xrodx

sodafish said:


> Hi all, brand new member here. So much great info on these forums
> 
> I have quite a specific inquiry about foobar regarding use with external USB DACs. I just recently got the iFi Micro iDSD (which is fabulous btw), and am having an issue with foobar where, upon a track transition involving a change in PCM sample rate or format type (PCM to DSD, or vice-versa) the first half second or so of the track is cut off. This appears to happen regardless of the buffer settings I use within the iFi driver or foobar's output settings. As I understand it, this is due to the hardware having to re-sync at the new sample rate, correct? JRiver has a setting specifically to address this: "play silence at startup for hardware synchronization", which solves the issue. Is there an equivalent functionality achievable within foobar (perhaps via plugins)? I have my whole library organised/tagged with foobar, and would prefer to stick with that if possible, but this problem is very annoying.
> 
> Any help would be greatly appreciated. Thanks!


 

 I'm having the same issue, downloaded the trial of Jriver and that option makes the cutoff dissapear, i wish there is a way to do it in foobar. Any news on this issue?


----------



## PleasantSounds

xrodx said:


> sodafish said:
> 
> 
> > Hi all, brand new member here. So much great info on these forums
> ...


 
  
 There is a f2k component that plays silence. I have never used it, but looks like it may help with this issue.


----------



## xrodx

that component does the trick, thank you


----------



## hgpsemaj

Superseded - Problem Fixed.


----------



## bmcelvan

I have a question about streaming audio. I just got a new Sony SRS-x77 and when connected to my home network via either Wi-Fi or LAN cable, I can see it on my windows computer. I can "playto" just fine through windows and can use my galaxy s4 mini with Bubbleupnp to stream music to it just fine as well.
  
 What I can't do however is use foobar2000 in windows to find the device, it doesn't show up. Any idea why? Under View-->upnp controller in foobar, my yamaha receiver populates (which I use all the time and have been for a while) as does my Sony TV...but not the srs-x77.
  
 Any thoughts?
  
 Do I have to tell foobar the device is there?
  
 Thanks


----------



## Mozhoven

Hello all,

I've just recently invested in a Marantz HDDAC-1 and a set of Sennheiser HD700's. Like everyone else, I want to make sure I'm getting the best sound I can, without constantly flipping settings. I'm not new to foobar, but am when it comes to streaming PCM/DSD audio. I'll start off by saying that everything is working and I'm getting good sound, but I can't help by wondering if I've got everything setup right. Bit Rate settings, Bit Depth, ASIO, and SACD, etc..., I would like some advice on the best settings, and perhaps some help with playing DXD (if it's even possible with my unit, I can't seem to get any to work). If anything, I would love some explanations to what the various options mean (specifically Output Devcie and DSD2PCM options...)

So, I've downloaded and installed the WASAPI, ASIO, and SACD plugins and can play every format (DSD64/128/256, .iso, .dsf, flac, etc....) *except DXD. *

Here are a few of my hang ups: (related to my specific DAC)

*1)* Under Playback>Output: (5 options)
Buffer Length: 180ms

ASIO: Marantz ASIO Device
DS: Digital Audio Interface (2-Marantz USB Audio)
KS: Marantz USB Audio Wave
WASAPI (Event) Digital Audio Interface (2-Marantz USB Audio)
WASAPI (Push) Digital Audio Interface (2-Marantz USB Audio)

*Which one do I use?* All work, but only a few (like KS: Marantz Audio Wave) have a noticeable immediate difference. What do DS and KS stand for?


*2)* Under ASIO drivers, I've checked to use *64-bit drivers* and to run with *High Process Priority*


*3)* Under SACD:

Output Mode: PCM
PCM Volume +6db *(too much?)*
PCM Samplerate: 176400 *(highest that works, next up is 352800 which doesn't work)*
DSD2PCM: Multistage (double-precision) - *I have no idea about these....*
Preferable Area: Multi-channel

Other info:
*There are no active DSP's*


Thanks in advance folks!


----------



## PleasantSounds

Looks to me like your DAC still receives a PCM stream.
  
 Configuring foobar to play native DSD is a bit of a pain. You can use this link as a guide, but maybe Marantz gas it's own setup instruction somewhere. Most steps will be the same, just in the foo_dsd_asio dialog you need to point to the ASIO driver for your DAC and the DSD Playback Method may be different.


----------



## Mozhoven

pleasantsounds said:


> Looks to me like your DAC still receives a PCM stream.
> 
> Configuring foobar to play native DSD is a bit of a pain. You can use this link as a guide, but maybe Marantz gas it's own setup instruction somewhere. Most steps will be the same, just in the foo_dsd_asio dialog you need to point to the ASIO driver for your DAC and the DSD Playback Method may be different.


 

 Thanks for the link, but I'm still having trouble. I'm able to select DSD (in stead of PCM), but it grays out all other options except for [Preferable Area] (PCM Volume, PCM Samplerate, & DSD2PCM mode all unclickable). The only change is that, when selected, I am able to hear the DSD (and my player actually registers it as a DSD on the front panel), but the sound is awful. There is music there, but mostly static and digital artifacts. I've played around with different [playback methods] under the ASIO menu, but none help. Native and DoP are as I describe, while DSx and exD are only digital noise, no music.
  
 I will say there was one thing different in the install of the SACD Decoder plug-in and the instructions. After double-clicking the ASIOProxyInstall.exe file, there was *no *foo_input_sacd.dll file to be found in the unpacked directory or in Foobar's component directory. foo_dsd did not show up under ASIO and the DSD's I have would not play at all. On a lark I did a search for it on my computer and found it buried somewhere. I then copied it over to the component directory and then restarted with the results listed above.  
  
 My DAC (Marantz HDDAC-1) is supposed to decode DSD natively.....


----------



## PleasantSounds

The static in output could be a result of buffer underruns. Try opening your native ASIO config panel and increasing the buffer. I have no experience with the Marantz, but in my Hilo I found that to play native DSD I need a really large buffer.
  
 The SACD dialog is doing the right thing: when you play DSD natively, the options related to conversion from DSD to PCM are not applicable. You only need them if you want f2k to convert the stream to PCM.


----------



## Mozhoven

Well, I'm not exactly sure what I did, but it works perfectly now. I had a brainwave that since the foo_input_sacd.dll file wasn't present in the directory after install of the proxy file, I shouldn't have put it  in manually. So, I deleted and restarted. Still didn't help, but after playing around with the DSD Playback method for the umpteenth-time, it started working. So, I installed foo-jesus to save my settings and and leaving it alone. Thanks again for the help!


----------



## Tambourine Guy

I have been trying to get foobar to naively stream DSD256 for days now. : ( I think I am getting close.


----------



## Mozhoven

Ok, so I've got a new issue:
  
*Why does it take so long (sometimes never) for playback to start in Foobar?* Rather than using the media directories in FB, I prefer to navigate and select my collection via Window Explorer folders. About half the time my files start immediately (or close to it), but the other half the files could take as little as 15 seconds to start, a minute or more, or sometimes not at all. Sometimes it says that it "can't create ASIO buffer".
  
 This even happens if I select a group of files for queuing in foobar (.FLAC, .DSS, .MP3,  or ISO) and decide to jump to another track....it'll say "playback starting" in the lower left corner but takes ages to start. On a few occasions it never starts and I have to shut down FB. I don't believe the issue is with my DAC (Marantz HDDAC-01) as I can choose the same files to play in another player and they start immediately every time. *Is there a way to improve startup speed? *
  
*Here are my stats:*
  
 Windows 10
 Foobar2000 v. 1.3.8
 ASIO: foo_out_asio
 Buffer: 17,860ms
 Driver: ASIO Driver: Marantz ASIO Device
 DSD Playback Method: ASIO Native
 Output mode: DSD


----------



## x_lk

mozhoven said:


> ASIO: foo_out_asio
> Buffer: 17,860ms


 
 Try reducing the buffer size to hundreds of milliseconds instead of thousands.


----------



## Mozhoven

Thanks, that seems to help. I also realized that the longest delays were because of pop-ups/ads in my browser taking control of my DAC between albums. My DAC unlocks between tracks ( or even when jumping around in a track) giving just enough time for another source to take over. Super annoying.


----------



## Tambourine Guy

arh.. I still get the odd zee zee distortion from DSD256. : ( I will start reading from page 1 again.


----------



## Mozhoven

All of the sudden my SACD ISO's are skipping ahead every 4 seconds. It was working fine, but after I installed the command-line plug in for ffmpeg decoding and Monkey audio decoder it started acting up. I removed them, but the problem persists. Anybody know how to fix this?


----------



## sonci

OK,
 another autoplaylist question,
 basically I want to find all albums with MFSL or SHM, which are written in the filename,
 so I try with: %filename% HAS MFSL OR SHM, but I got syntax error, when I try with only one label the playlist is created ok,
 any help?
 thanks


----------



## Soundsgoodtome

Hello thread. Anyone know if the f2k volume control is a 32-bit float point?


----------



## sonci

sonci said:


> OK,
> another autoplaylist question,
> basically I want to find all albums with MFSL or SHM, which are written in the filename,
> so I try with: %filename% HAS MFSL OR SHM, but I got syntax error, when I try with only one label the playlist is created ok,
> ...


 
 BUMP


----------



## PleasantSounds

sonci said:


> sonci said:
> 
> 
> > OK,
> ...


 
 try:
  
 %filename% HAS MFSL OR %filename% HAS SHM


----------



## Pott

Just got Foobar2k: my lossy collection is controlled through Media Monkey, and got F2k for my building-up FLAC files.
  
 However I don't seem to be able to find a good layout for it. It all seems optimized for playlists. I never use playlists and never will. 0 interest in them. Rather, I have my entire music collection on shuffle at all times.
 If I want a specific album or artist I just stop the shuffling. That simple.
  
 So I named all my files in Artist - Album - Track# - Title (felt very dirty putting the track number in the filename, but without tags... ah well). Unfortunately I not only had to put my whole files list into a single playlist. Since I'm building my library this isn't very convenient...
  
 I'm a total noob with F2k and it's true I've not had THAT much time to look around the layout. However it took me minutes to figure it out with MediaMonkey, and I'm still stuck on F2k... any help is greatly appreciated. Thanks!


----------



## PleasantSounds

pott said:


> Just got Foobar2k: my lossy collection is controlled through Media Monkey, and got F2k for my building-up FLAC files.
> 
> However I don't seem to be able to find a good layout for it. It all seems optimized for playlists. I never use playlists and never will. 0 interest in them. Rather, I have my entire music collection on shuffle at all times.
> If I want a specific album or artist I just stop the shuffling. That simple.
> ...


 
  
 It's been ages since I've seen a vanilla F2k setup, but let's try.
  
 First of all, make sure your Preferences->Media Library contains the path to your music library. Once that's done, the Album List window will provide a tree view of your entire music collection. You can use it to navigate your collection and select tracks for listening.
 Among your playlists you should have one called "Library Sel" - it is kept in sync with Album List, so whatever you select in the Album List will be shown track by track in Library Sel. To shuffle you can right click on the playlist and select Sort->Randomize.


----------



## mortarman

I have EAC to rip my CD's to flac and pull album art for each one. I set up Foobar with an album display used from custom files provided by Jaguar Audio. But for some reason the album art does not display. All I get is a big black square. What am I doing wrong? Not sure if it's a Foobar problem or an EAC issue.


----------



## Peti

Houston, We have a problem here. Namely, I have finally managed to set up on my Foobar the HDCD playback and in the bottom status bar it properly indicates when I play back my HDCD albums, but, for some reaon, in the main information field it shows different data. This is the case when a picture tells more than a thousand words, so here is a snapshot:
  
 (I have marked the differing data)
  

  
 If anyone has a clue, please share it!
  
 Piece,
  
 Peter


----------



## Peti

To further complicate the situation, I just made my Foobar to display the same HDCD info in the main field and it only shows a question mark. And keep says that it's 16 bits.


----------



## Peti

And finally, same issue with SACD playback:


----------



## Modular

Really newbie question: where can I download LAME safely?


----------



## Aaromtaar

modular said:


> Really newbie question: where can I download LAME safely?




I think it is safe to assume the following links:
F2K encoder pack ( does not contain the lame encoder, but is useful) - https://www.foobar2000.org/encoderpack

Lame - sourceforge.net/projects/lame/files/lame/3.99/


----------



## Aaromtaar

lg777 said:


> I think I have a similar problem with artwork in Foobar 2000.   I ripped about 300 CD's using EAC - FLAC and added album artwork.  The artwork is usually a jpg and named Artist - Album Title.jpg and is in each folder.
> 
> When I play them in Foobar, it doesn't recognize the album artwork but would have to manually add them which would take quite a long time.  I still have a few hundred more CD's to rip so I just want to make sure I can save myself some grief with managing album artwork.
> 
> ...





Absolutely ! In my past experiences, I had the same problem when I was trying to edit the attributes of old mp3's (encoded somewhere in the early 00s); the issue is related with the possibly obsolete or unsupported id tags. As of now, using F2K v1.3.9, under tagging you'll find the MP3 tag types tab, and tdit the active tags. And yes, you can embed the artwork in the mp3, additionally F2K allows you to manually manage the attached artwork (attach, extract, replace), under the tagging tab. 
Hope this helps.


----------



## Aaromtaar

destroysall said:


> I have a question which regards ripping CDs with Foobar2000. Is there any way to rip directly to ALAC as you can to FLAC?




Yep, install this !
https://www.foobar2000.org/encoderpack
For the ALAC and Apple AAC you'll need to install iTunes aswell.


----------



## Modular

aaromtaar said:


> I think it is safe to assume the following links:
> F2K encoder pack ( does not contain the lame encoder, but is useful) - https://www.foobar2000.org/encoderpack
> 
> Lame - sourceforge.net/projects/lame/files/lame/3.99/


 
  
 I downloaded the LAME file from Sourceforge, but I'm not sure how to get Foobar to use it as the encoder for converting FLAC to 320kb MP3. It asks me for the lame encoder and I point it to the folder, but there is nothing there that Foobar can use. 
  
 What am I doing wrong? Do I need to compile the code somehow? Maybe I should just use dbpoweramp and call it a day?
  
 I followed this tutorial, but I cannot find the "lame.exe" file anywhere in what I downloaded:  http://www.cnet.com/how-to/how-to-convert-flac-audio-files-to-mp3-with-foobar2000/
  
  
 Thanks for your help so far!


----------



## PleasantSounds

modular said:


> I downloaded the LAME file from Sourceforge, but I'm not sure how to get Foobar to use it as the encoder for converting FLAC to 320kb MP3. It asks me for the lame encoder and I point it to the folder, but there is nothing there that Foobar can use.
> 
> What am I doing wrong? Do I need to compile the code somehow? Maybe I should just use dbpoweramp and call it a day?
> 
> ...


 
  
 Sourceforge distributes Lame only in the source code version.
 For binaries you can try this link.


----------



## VNandor

Hi guys,
  
 Does  the foobar 2000 has a built in parametric EQ or at least can I download one for it?


----------



## PleasantSounds

vnandor said:


> Hi guys,
> 
> Does  the foobar 2000 has a built in parametric EQ or at least can I download one for it?


 
  
 Only a basic graphic equalizer is built in, but there is no shortage of third party components.
 The most popular parametric EQ component seems to be the Electri-Q, but for that to work you will need also a VST wrapper.


----------



## VNandor

Thanks for the help, I managed to install the VST wrapper however the Electri-Q asks to select the VST plugin directory. Is it where I installed the VST wrapper? (Sorry for being dumb btw )


----------



## PleasantSounds

vnandor said:


> Thanks for the help, I managed to install the VST wrapper however the Electri-Q asks to select the VST plugin directory. Is it where I installed the VST wrapper? (Sorry for being dumb btw )


 
 foobar is incredibly flexible, but the price for that is a fair bit of complexity. I guess you just got a taste of it...
 VST is an interface that professional audio plugins commonly use. f2k does not support it out of the box, but the component you installed does just that. 
  
 Now if I remember correctly, the installer provides you with a default value - if that's the case then just accept it (but make sure you remember what it is). In my system it is C:\Program Files (x86)\VstPlugins. If you had some other VST plugins then it would matter to show where they are, but I guess this is your first one. The installer will put the core binary files (.dll) into that folder.
  
 But that won't be the end of your woes: you will have the Electri-Q installed on your system, but foobar will be still blissfully unaware of that... The reason is that the Electri-Q installer doesn't know which VST-enabled application you are going to use (and if it did, foobar by default is not one of them).
  
 What you need to do is open f2k Preferences->Components->VST plug-ins and [Add...] the Electri-Q plugin by navigating to the VST Plugins filder and selecting the right dll.
  
 Restart f2k and in Preferences->Playback->DSP Manager you should be able to find Electri-Q among the Available DSPs and move it to the Active DSPs window. You can configure it there and then, but it may be more practical to set your EQ through the View->DSP menu.
 Enjoy!


----------



## VNandor

So I installed the Electri-Q and found the Electri-Q (posihfopit edition).dll under Program files x86)\VstPlugins.
  
 I installed the VST wrapper and copied the foo_dsp_vstwrap.dll into foobar2000\components.
  
 But when i open foobar2000 and go to preferences I don't find anything named VST plug-ins. I see the George Yohng's VST Wrapper  (and a couple of other things of course) but i guess that is not what I'm looking for?
 I thought when I go to  preferences it makes a list of my installed stuff but if that's the case. why is there an install button?
 Also whenever I open foobar2000 i get this: 
 Am I supposed to do anything with that?
  
 And thanks for your patience I really appreciate it!


----------



## castleofargh

vnandor said:


> So I installed the Electri-Q and found the Electri-Q (posihfopit edition).dll under Program files x86)\VstPlugins.
> 
> I installed the VST wrapper and copied the foo_dsp_vstwrap.dll into foobar2000\components.
> 
> ...


 
in "preferences" / "components" you must get a sub path called VST(or whatever name you got/gave).
there you have a list of all the VST plug ins you have added, if electriQ isn't in it, you go get it at "C:\Program Files (x86)\vstplugin".
when it now appears in the list of VSTs, you go to "preferences" / "playback" / "DSP manager" and get electriQ moved to the left panel where the "active DSPs" are. 
  
I can't look at your picture so I don't really know what you're showing.  maybe I'm not helping ^_^.
 edit: idiot me talking, disregard. I was explaining the wrong vst wrapper.


----------



## VNandor

castleofargh said:


> in "preferences" / "components" you must get a sub path called VST(or whatever name you got/gave).
> there you have a list of all the VST plug ins you have added, if electriQ isn't in it, you go get it at "C:\Program Files (x86)\vstplugin".
> when it now appears in the list of VSTs, you go to "preferences" / "playback" / "DSP manager" and get electriQ moved to the left panel where the "active DSPs" are.
> 
> I can't look at your picture so I don't really know what you're showing.  maybe I'm not helping ^_^.


 

 Okay, I have sub path called George Yohng's VST Wrapper, under the module coloumn it writes foo_dsp_vstwrap. I hope thit is what you are talking about because it's the only thing that contains "VST". If I select it and click on Install a browser pops up however if I go where the Electri-Q.dll is located, the dll just simply doesn't show up. Actually I don't even know if I have to look for the dll located in the vstplugins, or for something entirely else.
 I don't know if that makes sense, I have no clue what I'm talking about. :/


----------



## castleofargh

lol, no I'm the one who doesn't know what he's talking about. I have VST 2.4 adapter right now, I was so very sure I had yohng's one(I know for a fact I started with it but I must have switched at some point and forgot about it).
 anyway sorry for misguiding you mate. maybe you can try VST 2.4 and see if you get better luck with it.


----------



## PleasantSounds

castleofargh said:


> lol, no I'm the one who doesn't know what he's talking about. I have VST 2.4 adapter right now, I was so very sure I had yohng's one(I know for a fact I started with it but I must have switched at some point and forgot about it).
> anyway sorry for misguiding you mate. maybe you can try VST 2.4 and see if you get better luck with it.


 
  
 You're not the only one: I have just checked and I'm also using the foo_vst 2.4  
	

	
	
		
		

		
			




 Looks like I have supplied a link to the VST wrapper that has been superseded. Here's the foo_vst 2.4 .
 Sorry for the confusion!


----------



## VNandor

Great, I got it working, thanks for the help guys!


----------



## Bob A (SD)

I'm a fairly recent convert to foobar2000.  I've opted to keep things rather basic with very few additional components.   I've been searching for an answer to two related issues without success so I'm posting here for help 
  
 I've finished ripping the bulk of my CD collection using EAC.  What remains are on hold until I can solve some foobar2000 issues with "collection" CDs.   I'm using the facets plugin for the library viewer with the hierarchy set for artist, albums, tracks.  So collection CDs end up spraying all the various artists all over with many, especially oldies but goodies, having just a track or three nested with the artist name.  I really don't want that.   I have found that if I make the artist column in exact audio copy (EAC) "various" and append the artist to the track title I can stop that.... only trouble is then the album goes under an artist named "various." 
  
 The quandry goes another way too.  When you have an album where the artists are a well known group, no sweat e.g. Simon & Garfunkle.  But when some get together for just one album you end up with an artist listing of their collaboration e.g. JJ Cale & Eric Clapton separate from their individual listings.   Haven't broken the code to be able to have the collaboration album nested under both of the artists individually.  I'd like to be able to look under Clapton and find all his indivual works plus the collaborations with Duane Allman and JJ Cale for example.
  
   Anyhow is there a foobar2000 solution for these issues?  Of not?
  
   Thanks!
  
 My foobar2000 layout:


----------



## canthearyou

Foobar worked perfect to rip 4 cds. During ripping my last cds it dropped drive speed to 0.00x and just sat there. Tried a different, brand new cd and same thing. It will rip about 75% and just craps out. Tried restarting computer and same thing.

Edit: It was the optical drive in my 1 month old laptop died.


----------



## Bob A (SD)

No one have any input on my post 161?


----------



## Bob A (SD)

Well I've continued to search for an answer and finally found one:  https://www.reddit.com/r/foobar2000/comments/36etnp/is_it_possible_to_have_an_album_displayed_under/
  
  
   "Well, there's a few things you need to do to use multivalue tags

 go to Properties -> Advanced -> Display -> Propoerties dialog -> Multivalue fields: and add "ARTIST" to the list if it's not already there (it probably is, but we have to make sure)
 in your config change any places that use %artist% to %<artist>% (the "<>" brackets tell foobar to use multiple values when they are available)
 tag the desired songs with multivalue tags (separated by a semicolon)
 at this point you need to select the multivalue songs -> right click -> properties -> right click Artist Name -> Split Values -> make sure ";" is in the list on it's own line (should be by default) -> OK -> Apply (this step is so foobar will realize the field is now multivalue instead of single value)
 some components may be hit or miss with their ability to use multivalue tags, but i know they work just fine with Facets"


----------



## Giogio

Hello guys, I was wondering which resampler do you consider the best.
 I still only have the stock one, PPHS, but I had read something about a PRHS? Or something like that. Dunno if it was just a typo of the poster.
 Cannot find it anymore.
 In Hydrogen I read that many people prefer the SSRC or SoX.
 Do you perceive any difference in sound?
 I do not perceive any difference, for example, if I use or not use the PPHS.
  
 And do you guys have experience with the Bauer Binaural plugin? I use headphones a lot and it is tiring. Does this plugin help?


----------



## Bob A (SD)

Can't help you with the resampler, but for crossfeed, there's a whole thread on page 3 on the subject:
 http://www.head-fi.org/t/202365/best-crossfeed-plugin-for-foobar
  
 ADDENDUM:
 You piqued my curiosity so I did some searching.  Ended up installing SoX resampler and following the guidance here: http://www.head-fi.org/t/587481/foobar2000-wasapi-sox  which nicely summarizes what I found in discussions elsewhere.  I did however opt for a target sample rate of 192000 vice 96000 because my USB DAC handles 24/192.
  
 I've also further tweaked my configuration.  Using a Nakamichi Dragon VU meter as I happen to own that cassette deck


----------



## Joeybgood

Is there a recommended output buffer setting or do folks generally just ideally set it as low as the DAC can handle without drop out etc and call it a day? Wondering if there is a setting that most folks found be ideal/the 'sweet spot'. Also, do most of you prefer ASIO over WASAPI or are either regarded as relatively equal? Tks.


----------



## A_Man_Eating_Duck

Just leave the buffer length as default unless you're getting dropouts. I tend to use WASAPI as output since the visualisations are more responsive.


----------



## PleasantSounds

I don't think the DAC is a limiting factor here - it's more about how reliably can your PC keep the buffer populated. Smaller buffer will result in lower latency, so the adjustments you make in foobar (e.g. volume or EQ) will be less delayed. Too small buffer however will result in playback gaps which are much worse.
 I'm not aware of any general buffer size sweet spot - the PCs I'm using are set to between 500 and 1200 ms, depending mostly on how much concurrent load they typically handle. If you chose WASAPI connection then remember to review also the hardware buffers under Preferences -> Advanced -> Playback -> WASAPI. I have them set to 250 ms for push, 180 ms for event and the high priority process check box checked.
 ASIO may give you slightly lower latency than WASAPI, but it makes sense only if you have a driver supplied by your DAC manufacturer. You will also need it if you want native DSD playback (provided that your DAC supports it). Don't bother with ASIO4All, as it is just and ASIO interface to your standard Windows driver.


----------



## Joeybgood

a_man_eating_duck said:


> Just leave the buffer length as default unless you're getting dropouts. I tend to use WASAPI as output since the visualisations are more responsive.


 
 Thank you. What is the difference between 'push' and 'event' in Wasapi? I can't say as I can truly hear a difference between them.


----------



## A_Man_Eating_Duck

No sure what the exact differences are but I know that some DACs might glitch when using 1 of the modes. My DAC works fine with push or event.


----------



## Joeybgood

pleasantsounds said:


> I don't think the DAC is a limiting factor here - it's more about how reliably can your PC keep the buffer populated. Smaller buffer will result in lower latency, so the adjustments you make in foobar (e.g. volume or EQ) will be less delayed. Too small buffer however will result in playback gaps which are much worse.
> I'm not aware of any general buffer size sweet spot - the PCs I'm using are set to between 500 and 1200 ms, depending mostly on how much concurrent load they typically handle. If you chose WASAPI connection then remember to review also the hardware buffers under Preferences -> Advanced -> Playback -> WASAPI. I have them set to 250 ms for push, 180 ms for event and the high priority process check box checked.
> ASIO may give you slightly lower latency than WASAPI, but it makes sense only if you have a driver supplied by your DAC manufacturer. You will also need it if you want native DSD playback (provided that your DAC supports it). Don't bother with ASIO4All, as it is just and ASIO interface to your standard Windows driver.


 
 great info. Tks!


----------



## PleasantSounds

joeybgood said:


> Thank you. What is the difference between 'push' and 'event' in Wasapi? I can't say as I can truly hear a difference between them.


 
  
 Event and Push are two different ways of feeding data to the DAC.
 Push is the older model where the source (PC) is responsible for controlling the timely delivery of data.
 Event is a more modern approach where the DAC controls the timing and requests next data packet when it has room for it in the buffer.
 If both methods work, I'd recommend using Event as it is more efficient.


----------



## Youth

Is it possible to measure DR on the played album with foobar2000?


----------



## VNandor

youth said:


> Is it possible to measure DR on the played album with foobar2000?


 
 http://www.pleasurizemusic.com/de/free-downloads
 You can download a plugin here however it expired in 2011 so if you want to use it you will have to set back the date on your computer. I don't know if there are later versions of dynamic range meter plugins though.


----------



## Bob A (SD)

Take a look at this 2104 article with a different link for download (don't know if it is more current or not):
 http://teribil-audio.com/2014/02/how-to-display-dynamic-range-rating-in-foobar2000/
 http://dr.loudness-war.info/downloads/foo_dynamic_range_1.1.1.zip


----------



## castleofargh

just maybe point out that DR is cool to use because it's so easy, but it isn't faultless. also it's but one way to estimate a dynamic based on their own set of rules. I use the website a lot to get a kind of overview and avoid buying albums that are overly compressed(too bad that justin bieber looked good ^_^).


----------



## dhmcclain1

Hi folks. I am confused about something. It seems that most (But not all?????) of the music i ripped from my cds does not show up in either Foobar or iTunes.
  
 Don't understand why they are not part of my library, nor do I know where they would be or how they got there. It may be obvious that I am a bit challenged by
  
 computers, etc. Can anyone tell me, speaking slowly and loudly LOL, where my songs have gone?  And how to make them part of my library?
  
 Thanks very much.


----------



## canthearyou

dhmcclain1 said:


> Hi folks. I am confused about something. It seems that most (But not all?????) of the music i ripped from my cds does not show up in either Foobar or iTunes.
> 
> Don't understand why they are not part of my library, nor do I know where they would be or how they got there. It may be obvious that I am a bit challenged by
> 
> ...




In Foobar click "file" "add folder". Now browse for the folder you ripped to(most likely "Music". Once you find it select it and click "ok"(or add).


----------



## Giogio

pleasantsounds said:


> Smaller buffer will result in lower latency, so the adjustments you make in foobar (e.g. volume or EQ) will be less delayed.
> ASIO may give you slightly lower latency than WASAPI, but it makes sense only if you have a driver supplied by your DAC manufacturer. You will also need it if you want native DSD playback (provided that your DAC supports it). Don't bother with ASIO4All, as it is just and ASIO interface to your standard Windows driver.


 
 So you mean that Wasabi give better sound than Asio4all? I was just wondering about that, as I do not have Asio4all...
  
 About the buffer, I try to keep it as low as possible and generally even with a BT headphone I can even keep it at 50 actually, but I leave it on 300 mostly.
 I put it on the max buffer possible when I have other CPU sucking programs running in background, like the MP3Gain (which correct me if I am wrong but it is still the best software to normalize MP3), which even in idle mode is terribly heavy. A huge buffer, and putting Foobar in high priority, solve the unwanted problems.


----------



## Giogio

bob a (sd) said:


> Can't help you with the resampler, but for crossfeed, there's a whole thread on page 3 on the subject:
> http://www.head-fi.org/t/202365/best-crossfeed-plugin-for-foobar
> 
> ADDENDUM:
> You piqued my curiosity so I did some searching.  Ended up installing SoX resampler


 
 And, which resampler do you prefer now? DO you really notice any difference in SOUND, between the stock Foobar one and any other resampler?
 Would the difference be noticeable in Mp3 too?
  
 Thanks for the other link. I was wandering, if I use crossfeed, I have less stereo effect, right? Will this not affect negatively the soundstage? Should I, and is it even possible and meaningful, use at same time a crossfeed and a dolby or whatever else surround plugin (not yet found a simple, good, effective one, and I hate when they just add a cheap metallic reverb).


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## castleofargh

giogio said:


> So you mean that Wasabi give better sound than Asio4all? I was just wondering about that, as I do not have Asio4all...
> About the buffer, I try to keep it as low as possible and generally even with a BT headphone I can even keep it at 50 actually, but I leave it on 300 mostly.
> I put it on the max buffer possible when I have other CPU sucking programs running in background, like the MP3Gain (which correct me if I am wrong but it is still the best software to normalize MP3), which even in idle mode is terribly heavy. A huge buffer, and putting Foobar in high priority, solve the unwanted problems.


 

 asio4all is like an adapter to make stuff that like to talk to asio, able to talk to other protocol(I think I read somewhere it was kernel streaming but I'm not sure). so the rational thing to do when you have a choice, would be to use the real asio drivers provided for your DAC if they happen to exist. or go for wasapi or even directly to KS if that works for you.
 but there is no actual situation where it really make sense to use asio4all for a basic listening. so more than saying "why not use it?", I would say "why use it? " ^_^.
  
  
 Quote:


giogio said:


> And, which resampler do you prefer now? DO you really notice any difference in SOUND, between the stock Foobar one and any other resampler?
> Would the difference be noticeable in Mp3 too?
> 
> Thanks for the other link. I was wandering, if I use crossfeed, I have less stereo effect, right? Will this not affect negatively the soundstage? Should I, and is it even possible and meaningful, use at same time a crossfeed and a dolby or whatever else surround plugin (not yet found a simple, good, effective one, and I hate when they just add a cheap metallic reverb).


 
 sox is known to be pretty good, can work fine with basic settings but you can add some magic spells if you're into making your own stuff. it's fine and versatile, known to measure pretty well, so there really is no reason not to use it IMO.
  
  
 crossfeed will put some frequencies that were on the left, and copy them on the right, usually attenuated by a value and delayed by a value. the aim is to simulate in part what would happen to a real sound coming from the left, you would get it in your left ear, then a little latter and a little quieter, into the right ear.  no sound would ever come only into the left ear, that doesn't exist in real life. but it exists in headphones. so unless the album was mastered for headphones(99.9% aren't), you end up with panning that is too strong and unnatural.
 it is my experience that you get used to whatever panning you get after half an hour or less(the brain is amazing), but most crossfeeds feel less fatiguing to me in the long run. at first it is less impressive, and often at least the bass suffers a little from it(voices too for some IMO). so some people just never get into it.
 crossfeed isn't speakers and is far from a perfect simulation, that must be clear for everyone. it's just slightly more of a realistic experience than music mastered on speakers and used on headphone.
  
  
 ps: I was desapoint by the naked picture. I expecting a cool juggling panda or something.


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## Giogio

castleofargh said:


> sox is known to be pretty good, can work fine with basic settings but you can add some magic spells if you're into making your own stuff. it's fine and versatile, known to measure pretty well, so there really is no reason not to use it IMO.
> 
> 
> crossfeed will put some frequencies that were on the left, and copy them on the right, usually attenuated by a value and delayed by a value.
> ...


 
 Well, you can still watch Kung Fu Panda, one of the best films ever, and get some of that stuff 
  
 The reason why not to use SoX (or the other one, SS something, I forgot how it is called, which soma people think it is better, for what I have read in the first page of the sox thread on hydrogenaudio), is that Foobar comes with a resampler already, which does not have to be configured. Now, if I have to use something more complicated where I have to set parameters, given how obsessive and perfectionist I am I may lose ages to configure it. SO I would only get into that hassle if that resampler really give a noticeable better audio quality than the stock one (also with MP3, which I listen a lot, because most of my music I got it from people met during travels).
 So... Does it, sound noticeably better? 
	

	
	
		
		

		
		
	


	



  
 About crossfeed, I had that in mind (Dmitry of Neutron explained that once to me, confused by the fact that I was using the Surround instead of the Crossfead effect with Headphones. But I like to have a spacious sound, and some BT headphones do not have it... Unfortunately you cannot use both effects on Neutron).
 But thanks! One never knows.
 What I am trying to find out is which Crossfeed plugin is better on Foobar. Apparently I am not the only one!
 The stock "crossfader" does nothing to my ears, I do not even know what is it supposed to do.
 I have downloaded the naive, which I still do not know how to configure (no help file, no faq, nothing, no idea what those three parameters really do).
 I have noticed the slight bad effect on bass on some tracks, which anyway does not affect me too badly because I have Real Bass Exciter doing a very good job.
 But on other tracks it was even better, because some tracks have a panned bass (never pan bass!), which with crossfeed becomes more central. Just in these days I was listening to the wonderful Electro Swing track "True Love Sweet Georgia Brown" of Ecklektic Mick (unfortunately not available online), where the punch/bass is almost all on the right. Very annoying.
 Crossfeed made it better.
 Btw, do you put it at the beginning or end of the chain? I have it just after the resampler, as second plugin, so that the Real Bass Exciter will excite the centred bass, not accentuating eventual panning of the recording.
  
 In the while I have other questions:
 what the Advanced Limiter and the Hard -6 Limiter do?
 I know, more or less, what a limiter does (stopping all sounds which passes a certain peak level, right?), so I though I could use it at the end of the chain, or bewteen (hmmm, I swear it was not a porn thought, just a typo) Real Bass Exciter and Graphic EQ, to minimize some eventual distortion (sometimes I do not reduce so much the gain in the EQ, to have more loudness. But apparently bass and gain do not understand each other perfectly well).
 I did not notice any effect with the Advanced, but I have noticed that the Hard even creates more distortion sometimes.
 Do you use them? Which? How?
 Does it makes sense to combine them?


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## Bob A (SD)

giogio said:


> And, which resampler do you prefer now? DO you really notice any difference in SOUND, between the stock Foobar one and any other resampler?
> Would the difference be noticeable in Mp3 too?
> 
> Thanks for the other link. I was wandering, if I use crossfeed, I have less stereo effect, right? Will this not affect negatively the soundstage? Should I, and is it even possible and meaningful, use at same time a crossfeed and a dolby or whatever else surround plugin (not yet found a simple, good, effective one, and I hate when they just add a cheap metallic reverb).


 

 I enjoy using crossfeed algorigthms and have for years (since Tyll introduced his in the Supreme headphone amp in 1993).  They simply take what is there and mix the channels ever so slightly to lessen the "in your head" sound of headphones/IEMs.   Subjectively it seems to broaden the soundstage.
  
 Your reverb and other special effects, colors the music in my view distorting what the artist has created.  I avoid them.
  
 I don't use dolby multichannel playback unless the tracks were originally recorded that way and I have the equipment to properly play them (e.g. dolby 5.1 music tracks on a home theater system).
  
 A resampler simplistically can reduce digital distortion.  And no, I have no heard any substantive differences with the SoX resampler engaged.  But as castleofargh stated there is no reason not to use one and SoX is touted as the best of what is available for foobar2000.
  
 So as far as DSP in my foobar2000 setup, all I use is foo_dsp_xfeed and foo_dsp_resampler.   The graphic equilizer foo_dsp_xgeq is active  but I have yet found a reason to use it.


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## castleofargh

> Originally Posted by *Giogio* /img/forum/go_quote.gif
> 
> Well, you can still watch Kung Fu Panda, one of the best films ever, and get some of that stuff
> 
> ...


 
 a lot of questions.
 does it sound noticeably better, I don't think so, but then again I'm one of those deaf objectivists that doesn't hear much of anything ^_^. I do use the default foobar converter sometimes and it didn't ruin my life. but I always use sox for real time downsampling.
 at least one of the reasons why I believe everybody doesn't agree on a preferred crossfeed, could be simply that we don't all have the same distance between our ears. meaning that we don't all need the same delay to get something similar to real life delays. xnor's crossfeed let you set the delay yourself, so you could try and measure your head, and estimate the value with the speed of sound. or you can just move the slider around until you feel that it's right for you. anyway the changes aren't dramatic(I actually struggle a lot to notice a change unless I change the values a lot) and even if we're a little off, again our brain usually takes care of what's wrong.
  
 crossfade isn't a crossfeed ^_^.
  
 limiters are used to set a maximum volume level. on the digital side it can been seen as an anti clipping tool. I'm pretty conservative with my gain values+ I use replay gain and my EQ gets mad when it clips so I know about it. so I don't feel like I need the limiter. I always wondered if the "prevent clipping" option when choosing what replay gain to use was not doing the exact same thing? anyway, in general the limiter will do nothing as long as the signal doesn't clip, so there is nothing wrong with using it as a safety measure. you can put it at the end of the DSP chain and forget about it. but it should not serve as an excuse not to correctly set the gain on your EQ to go with whatever boost you applied. a limiter is an improvement over hard clipping, but it isn't something you want to activate too often, it could sound really bad too if your signal is pushed up too much too often.
 and no using 2 limiters only means the lower one may do something, the higher threshold one will never get the opportunity to do a thing.


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## Giogio

bob a (sd) said:


> Your reverb and other special effects, colors the music in my view distorting what the artist has created.  I avoid them.
> I don't use dolby multichannel playback unless the tracks were originally recorded that way and I have the equipment to properly play them (e.g. dolby 5.1 music tracks on a home theater system).
> 
> A resampler simplistically can reduce digital distortion.  And no, I have no heard any substantive differences with the SoX resampler engaged.  But as castleofargh stated there is no reason not to use one and SoX is touted as the best of what is available for foobar2000.
> ...


 
 I meant if you or anybody ever heard a difference between the different resamplers. I am dangerous when I start with something new, so before I open a pandora box I want to be sure what could be the advantage respect to the stock foobar resampler.
  
 I do not use reverb and I understand what you mean but I do not agree and you may read why in the section "my sound ideology" of the opening post in the first link of my signature.
 Beside what written there, which is mostly objective, I also have other more personal reasons why I like to customize sound: I do not consider music like a mystical ipse dixit which must be venerated as it is.
 If I listen to Dubstep I doubt the "artist" would ever possibly care less if I push the bass or add this or that. And same I would say of most musicians, given that I seriously doubt that in pop and rock music each single little detail of the composition, and of the recording and mastering and mixing, is thoroughly thought (apart Pink Floyd, Alan Parsons Project, and guys like that). I think more artists do some jam, find something cool, work on it, and fix it when it is good enough. I have talked to musicians and producers of electronic music, and I have done some myself, and the creation of music is not that mystical in the 90% of cases. Artists know that they could stay ages working and reworking on something without ever feeling "oh, now it is ready".
 So, I do not see all this mystical need to respect the "sound as it was meant to be".
 Besides, when they record and mix, they have THOSE equipment, THOSE monitors, THOSE headphones. They listen to their own record through limited technology (nothing is uncoloured) and have no idea how that music will sound on your devices, nor you will ever be sure that how your devices make it sound is exactly how it sounded in the studio.
 For me it is like cooking. You can follow a recipe in each little detail, but, do you have the SAME ingredients? Same quality? Same origin? And do you have same mouth, do you perceive tastes in the same way? Are you sure that if you follow the recipe as it is, you will taste the same taste the creator of the recipe tasted?
 So, I change things. It is an active listening, where I participate in the creation process, in a dialogue with the artist.
 Unless I am facing an absolute masterpiece of music, a real ultimate work of art where each single little sound has a mystical reason the way it is, and where the recording was made state of the art with extremely sofisticated technology. In that case I would definitely do my best to organize a setup which would let me hear the recording as it was supposed to sound (just tweaking what I need to, to adapt the sound to the limitations of my devices and of my ears).
  
 But to be honest, mostly what I do is just adding salt and pepper. I can eq a bit, mostly one single preset to adjust the basic sound of my headphones overcoming their limitations/colour and tuning them to my ears and psychoacoustic traits.
 I can push a bit the bass because I like so and I could not care less if Sade would agree with me.
 I can apply crossfeed.
 And I can try to find something which can improve the soundstage, which in my opinion is NOT altering the creation of the artist but overcoming the limitations of the equipment I have (I do not have 2000$ open back planar magnetic with 2000$ top class amp). On a little AKG Y45BT, if you do not apply something, you have almost no soundstage, and I will bet my life each second that that is not what the artist wanted.
 If I was sure that the artist wanted a direct sound with no soundstage and I respect and admire the artist more than I like spatious sound, I would definitely avoid effects.
  
 But I avoid them now already, as I did not find ANY, at least, not any free, which can improve soundstage without sounding like I am in a case underground (as said, I do not like reverb. Although I would argue that artist do use reverb, but, whatever).
  
 I felt in writing mode tonight.
 I will try the xfeed as soon as hydrogenaudio is back online.


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## Giogio

> Originally Posted by *castleofargh* /img/forum/go_quote.gif
> 
> I always use sox for real time downsampling.
> 
> ...


 
 Thanks 
  
 I use a resampler because some of my Mp3 are 44, some are 48. Both my intel soundcard and my bt headphones can get to 48, so I put the resampler to 48. So, it is mostly upsampling.
 Would that make a difference?
 I do not even know if it would make a sense setting it at 96 or so, which with the output as "direct sound" it does not crash foobar (with wasabi I must set 48 if my device can only 48).
 Anyway, I will try sox 
  
 So crossfade is like, a track ends and another begin and the two mix? I hate that. Just used it for a party at home with Musicbee once.
  
 I have all my Mp3 set at 88,8 with mp3gain and option "do not clip" (so some are lower than 88,8).
 I also select the apply replay gain and do not clip in foobar.
 As EQ I use the graphic EQ, and it does not ma to me but I did not understand what you mean  What do you use? I like that because it goes deep till 20 and I can save presets.
 I do set the gain a bit lower if I am pushing the bass a lot (which I do in the presets I use for Techno, and even more for Dubstep).
 But I have no idea how much should I reduce the gain to avoid clipping, and despite replay gain there are tracks which clip and distort, and some which does not, although the perceived volume is the same.
 One which distorts very easily is Balloons (club mix) of Nils Hoffmann. I use it like a standard to set the gain!
 An autogain option like in Neutron would be cool. Is there any such plugin? It reduces the gain of x% when a peak is detected. Each new track it resets the gain at 0 (or the value you want).
 The graphic eq have a fake autogain, it does not analize the music real time like neutron does, it just set the gain as many db down as you have pushed some frequency up. This is way too much.
 So, basically I use the limiter "just in case". I do want power and try to reduce the gain as few as possible, but I definitely do not want distortion and clip.
  
 Using the two limiters would be in different moments of the chains. One after the Real bass Exciter, and one after the EQ. Because I push both till their limits.
 And I was just wondering how the two limiters differ.
  
 I like what you say of crossfeed, it sounds exactly like what I wrote in "my sound ideology" on my thread (feel free to read and feedback if you agree).
 I find it much more clever offering a customizable crossfeed than 20 different ones where people must try each to see if by chance it come close to what their ears need.
 I did not know this one you mention. Ah, ok, it must probably be the xfeed, right?
 I will try that soon.
  
 EDIT: as we are here, do you know if it is better 16 bit plus dither, or 24 bit without dither?


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## castleofargh

it was supposed to read "my EQ gets ma*d* when it clips" sorry about that, depending on the keyboard I use I tend to eat up some letters ^_^. I just meant there is a warning for clipping.
 for EQ there is an endless supply of VSTs that will work with foobar as long as you also add a vst adapter. I would always recommend parametric EQs as they tend to have cleaner impact and better controls. for years I've used electri-Q, great but crashes way too often for my taste. so I moved on to easyQ that offers a little less but also crashes less. now I've moved on to some paid EQ(DMG equilibrium), but both easyQ and electriQ were really cool ones and I never regretted using them and learning on them.
  
 what I meant about setting the gain correctly on the EQ, if you boost 10db, you should also move the gain to -10db at the same time. a digital signal is written with one value per sample, that value gives an amplitude and goes from 0db to -96 or -144db(16 or 24bit) the loudest sound being 0db. so anytime you boost something that was close to 0db with the EQ, you risk pushing the value above 0db which is impossible as there is no number for that, so the value that will be used instead is still 0. that's clipping. all the stuff that would go above ends up being 0db. at this point if you use nothing it clips, if you use the limiter, it will reduce the value of the signal by a given value for a given time and sound more or less natural depnding on the kind of limiter, but it's not a transparent process. so whatever boost you had decided upon will not be applied anyway if the music was already close to the max, one way or another, clip or limiter you will still be stuck to 0db(of course that's in the digital domain, on the analog side any amp can be pushed to be louder).
 so the proper method to avoid having to alter the music in an undesired way, is to simply make sure the sample values will always stay below 0db. it's easy, as long as you do nothing that will boost the signal you will not risk degrading said signal. that means trying to EQ by removing values instead of boosting. or if you boost, lowering the main gain setting in the EQ by the same value you used on the biggest boost. if your maximum boost is 20db, then you need to set the global gain of the EQ to -20db. the result will be that the loudest frequency will stay where it was and never risk clipping, while the rest will go lower.
 as far as digital go it is the right method. if your only concern is that you can't get a 20db boost in the bass, I was talking about it on another topic a few days ago, with easyQ you can make up to 48db boosts!!!!!!!!! ^_^

 here it is, let's say I want to apply this made up curve with my massive up to +48db bass boost. I can't leave it like this, instead I will lower the global gain by the value of the maximum boost. that way the loudest signal getting out of the EQ cannot try to be higher than 0db no matter what was recorded on the music. and the EQ I will use is in fact this one:

 this one cannot clip the music and won't need a limiter to try and save the day. I'm not using the EQ to boost the bass, I'm using it to lower everything else!!!!
  
  
  
  
 the only reason direct sound seems to work with 96khz when your devices can only do up to 48 is because windows must resample back down to 44 or 48 in the end. wasapi bypasses windows mixer so it doesn't resample down and your device says "no speako el 96khz". in both cases you do no end up with 96khz.
  
 now on the other hand setting the output to 24bit is very fine if your DAC can use 24bit(be it connected DAC or bluetooth DAC inside the headphone(does the streaming protocol accept 24bit????).
 if the DAC part is ok with 24bit then do use it, that gives some headroom for those EQ things we're doing and for some more volume setting by foobar or windows. if the DAC doesn't accept 24bit(or the bluetooth protocol), well then obviously select the 16bit output. ^_^
  
 and yes it's xfeed I was talking about for xnor's crossfeed, although I suppose several .dll are called the same.


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## Giogio

So, well, thanks again.
 I had read about subtractive EQ and I had used for a while but got sick of having to drag down all the other sliders (I mostly push the bass, so with subtractive I must drag down all other sliders on the Graphic EQ).
 I did not realize, well, I had partially realized or suspected that decreasing the main gain was the same as doing like the images you have posted.
  
 With MP3gain I have seen (analysing tracks) that most modern music has a value of 94, +/-.
 But older music is less. So MP3 gain suggests 89. As I do that, I have a lower 0, and I can push the EQ more without decreasing the gain so much, or this is what I thought till now.
 Also, decreasing the gain of 6 if you push one freq 6 up, it is only preventive, just in case, but we do not know if that freq was near to 0, so, maybe it war -3, and I just have to reduce the gain -3.
 The other problem would be that as I only push the bass, the overall sound will lose lot of loudness if I reduce a lot the main gain. With Dubstep I push the bass even more. More or less, 3db the 40-80hz with Techno, 6db with Dubstep. Plus the Real Bass Exciter, this is lot of bass (which the ATH-WS99BT do very well).
 I go by ear, I know which of my tracks have the most devastating (near 0) bass, and I use them to test my EQ and gain setting, reducing the gain just the min needed.
 Lately I am addicted to the highs of Dirac, which I use on the ATH although I only have the filters made to compensate FR and IR of the XTZ Headphone Divine.
 Dirac has got a clipping led so I can see what my ears do not hear.
  
 So, I wonder, does any of your EQ have both a clipping led and an automatic gain protection (like Neutron)? I think it is the best solution. I leave the EQ gain at 0, and the EQ automatically sees the peaks and decreases the gain as much as needed.
 I think MusicBee had a function, "dynamically normalize tracks". I wonder if it is the same.
 And would that be the same as Peak Normalization, but on the fly?
 Although Neutron as said does something different, it does not constantly increase and decrease the gain for every peak, it just reduces it a bit when the 0 is reached, and it leaves it like that till the end of the track also if there are no other peaks.


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## castleofargh

even with the basic foobar equalizer, once you're done with your EQ you just click "auto level" and boom it will move the loudest boost to zero. unless you're failing to go loud enough with your sound system, I don't really get why you wouldn't just do it as it should?
 + if you really use something that reduces the loudness for a lot of potential clipping that you would have created yourself, in the end you're ruining the job that was done with mp3gain to make all our songs feel as though they were at the same loudness.
  
  
  
 if you insist on using the EQ wrongly, you could always try and add something like this https://www.foobar2000.org/components/view/foo_r128norm  at the end just before the limiter, it would most likely do something close to what you're asking for (didn't try it so IDK if it's any good).  and it would make mp3gain redundant.


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## Giogio

My system are BT Headphones. Which means, I do not have speakers, nor wired headphone, nor anything I can push in loudness with an analogical Amp.
 And although my favourite BT Headphone is the probably loudest of all BT headphones or among the loudest ones, if I do an EQ preset for Dubstep with 6db push on the sub-bass plus other hardcore push on Real Bass Exciter (for that I also need to get headroom by lowering the gain on the EQ, as the Exciter does not have any levelling option), and as said I only push the bass, and I have to reduce the main gain something like 9 to 12 db, I do not have much power left to enjoy music loud enough.
 Another story is with the setting for Techno and Electro Swing, where I push bass 3db and the Exciter mush less than with Dubstep (the bass of Dubstep is so tricky, it needs lot of push to be really juicy, fat and dense).
 In this last case I normally see that -6db main gain are ok and I can live with that. That is anyway the "border value" of my loudest headphone, meaning that already with that I am just "loud enough". Which is indeed louder than many people would like to listen to music for many hours (me included) but which is not loud enough to me for occasional moments when a special track make me wish to jump around my room doing club moves.
  
 That's why I try to find a way to avoid reducing the main gain too much.
  
 So, you gave me one. I will look at what it does, but if you say it nullify Replay Gain then it must probably constantly do real time peak normalization? Does that not reduce dynamics? Besides, I could never understand why the creator of MP3Gain says that peak normalization is not the best way to make tracks sound at same volume.
  
 Another option would be maybe (I really have no idea if it would work, maybe you can confirm) using an external DAC or Soundcard, something I can connect to my notebook either via USB or with the 3.5 jack, and which would give a much higher loudness than the one I can get now.
 I see that some Soundforge have USB ports where you can stick an USB BT Dongle like the Creative BT-W2, otherwise I could use the 3.5 Avantree Priva 2 adapter.
 But I am not sure if once into the BT adapter the sound of the external card, even if originally louder than in my notebook, would be levelled digitally and have same exact loudness than those BT adapters would give me when connected to my notebook...


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## castleofargh

ok so you're a special case with a special problem. anybody else should avoid clipping and get a more powerful amp, but that can't be done with a bluetooth headphone as the amp is in the headphone. if the aim is really to feel louder, then the best known option is a dynamic compressor. like has been done for so many years on CD and on the radio (loudness war). it does alter your music but it's a proved method to feel loud.
  
 about peak limit vs mp3gain: a limiter or peak normalizer will make sure the maximum signal in the song doesn't get over a fixed value. while mp3 gain set a preferred target for how loud the song will feel! not how loud the peaks will go will change depending on the dynamic of the song. some pop song will have little dynamic range so the peaks are actually very close to the main music. if you do peak normalization, then that song will sound very loud. if you do the same for some classical music, the music will sound way quieter(probably hard to hear compared to the pop song).  if you use mp3gain or replaygain or some R128 stuff, your music is set to feel like all musics are as loud psycho acoustically. not electrically .so it will take the pop song down a lot so that it feels as loud as the high dynamic classical song. by doing that mp3gain tends to make most music quieter than what they were before as a mean to make them all sound about the same in perceived loudness.
 so each tool has it's own purpose.
  
  
  
  
  
 and no adding an external whatever wouldn't change a thing as you're most likely already pushing things to the max with your EQ+limiter. as I said, with digital music the maximum loudness can't be more than 0db. and the only part that can boost the analog domain is your bluetooth headphone in your particular situation.
  
  
 should I mention that listening to music loud for long periods is bad for your ears? ^_^


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## Joe Bloggs

giogio said:


> My system are BT Headphones. Which means, I do not have speakers, nor wired headphone, nor anything I can push in loudness with an analogical Amp.
> And although my favourite BT Headphone is the probably loudest of all BT headphones or among the loudest ones, if I do an EQ preset for Dubstep with 6db push on the sub-bass plus other hardcore push on Real Bass Exciter (for that I also need to get headroom by lowering the gain on the EQ, as the Exciter does not have any levelling option), and as said I only push the bass, and I have to reduce the main gain something like 9 to 12 db, I do not have much power left to enjoy music loud enough.
> Another story is with the setting for Techno and Electro Swing, where I push bass 3db and the Exciter mush less than with Dubstep (the bass of Dubstep is so tricky, it needs lot of push to be really juicy, fat and dense).
> In this last case I normally see that -6db main gain are ok and I can live with that. That is anyway the "border value" of my loudest headphone, meaning that already with that I am just "loud enough". Which is indeed louder than many people would like to listen to music for many hours (me included) but which is not loud enough to me for occasional moments when a special track make me wish to jump around my room doing club moves.
> ...




Why not just gauge the volume using the foobar peak meter? It indicates clipping whenever it happens. So just set the EQ gain however you like and turn it down if you see / hear clipping. I agree that you don't need to strictly follow the "everything below zero" rule, especially if Replaygain has been applied, but it's anybody's guess what a safe setting is for all the stuff you listen to. I don't listen loud, listen to wired earphones and have an amp that goes crazy loud, but if I were in your shoes I'd either watch the meter manually, or add the Advanced Limiter to the end of the DSP chain and call it a day.

Also, you do know there's volume buttons on the bluetooth headphones right?


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## castleofargh

joe bloggs said:


> Why not just gauge the volume using the foobar peak meter? It indicates clipping whenever it happens. So just set the EQ gain however you like and turn it down if you see / hear clipping. I agree that you don't need to strictly follow the "everything below zero" rule, especially if Replaygain has been applied, but it's anybody's guess what a safe setting is for all the stuff you listen to. I don't listen loud, listen to wired earphones and have an amp that goes crazy loud, but if I were in your shoes I'd either watch the meter manually, or add the Advanced Limiter to the end of the DSP chain and call it a day.
> 
> Also, you do know there's volume buttons on the bluetooth headphones right?


 
 it's also the first thing I thought about, but as you say it's a great song by song method, but you can't tell much as a general purpose. I have songs where I could add +30db at 40hz without clipping, but I also have songs where 40hz almost reaches the max level of the entire song. so I didn't mention it. but of course it's a perfect song by song control. 
  
 second point, when people know what they are doing and understand the signal chain, I'm with you that the everything below zero in the EQ is rarely needed. but for anybody who isn't yet confident with what he's doing, basic rules are how you learn how to do it right IMO. maybe we would have less EQ haters on headfi if they had cared for the overall gain.


----------



## Giogio

castleofargh said:


> if the aim is really to feel louder, then the best known option is a dynamic compressor. like has been done for so many years on CD and on the radio (loudness war). it does alter your music but it's a proved method to feel loud.
> 
> about peak limit vs mp3gain
> 
> ...


 
 I could not hear your last sentence, can you repeat?
 Anyway, you deserved THIS  (you know the original but maybe not this, so eventually watch till the end).
  
 Aim 1: quality
 Aim 2: bass
 Aim 3: loudness
 I have tried the Dynamic Compressor of Foobar, it makes some tracks louder, some other strangely quieter, and it mostly ruins the dynamics. All feels constipated. Compressed, exactly.
 No, that's not for me.
  
 I understood your explanation of peak vs gain. Thanks. So, a Techno track with a constant loud low kick pushed near 0 would be treated from mp3gain like the classical music or like the pop one in your example? I would say like the pop, as these peaks are constant, so I am not sure how could mp3gain understand that they are just the low kick and give priority to the main melody. But if mp3gain would do that, then it would push the low kick above the 0 (which is not possible but, you know what I mean), right? And to avoid this, there is the option "avoid clipping". Did I understand all right?
  
 Now, as I have already asked and nobody proposed any option, I must suppose there are none, but, I ask again just in case: are there any plugin (usable in Foobar2000) which would analyse the peaks in real time (probably before time, thanks to the buffer) and quickly reduce the gain as much as needed to avoid clipping?
 You may argue that this is what the option "apply replay gain and avoid clipping according to peak" is doing, but I would argue back that it is not doing it well. It is mostly wishing to do it than really doing it.
  
 About the 0 thing, the notebook must have some DAC to make the audio come out of the 3.5, right? And you said that with analog we can go above 0.
 In this case there may be an external DAC which is louder than my notebook. Right?
 The real question is, would the greater higher loudness of the new DAC result in a higher loudness on my headphones?
 I do not know if the ADC of the BT adapter, which can just go till 0, would make the sound equally loud no matter how loud was when it entered in the ADC.
 Do you?


joe bloggs said:


> Why not just gauge the volume using the foobar peak meter? It indicates clipping whenever it happens.





> or add the Advanced Limiter to the end of the DSP chain and call it a day.
> 
> Also, you do know there's volume buttons on the bluetooth headphones right?


 
 I have just tried the Peak Meter, it does NOT shows clipping. There is no red, no led for clipping. Or am I doing something wrong?
 Anyway I can at least see when I am getting close to 0. It can be useful at times.
 The advancer limiter is always there. And it definitely does not avoid clipping. I can notice it from the bad distortion when I am on red zone and from the clip led on Dirac flashing.
 I think it is not a very good limiter. But the Hard -6 is not better.
  
 Which buttons? Ah, yes, you  mean, that one with a + on it, which is used to turn headphone on and off? 
 (joke explained for dumb: all or nothing. loud or off.)
 Nah, do not build a bad impression of me, I only listen loud to energetic music. And only when I feel like energizing myself.


----------



## castleofargh

the EQ hater note wasn't about you(nor Joe obviously), you love your EQ as much as we do(even if we do for different uses). ^_^
  
 your bluetooth headphone receives digital signal, so whatever you use upstream, it will still come down to the loudest amplitude being sent to the headphone's DAC as being 0db and no higher. so no number of DAC to amp to adc to headphone's DAC will help going louder. the problem is your BT headphone not having a higher max voltage.
  
  
  
  
 by default a song that reaches 0db and has very compressed music(no dynamic), will almost entirely sound like it's around 0db and it is, so it will feel loud.
 now with a very dynamic classical music, with loud passages, quiet passages etc. if the loudest sounds are near 0db, the rest of the music can still be 10 or 15db lower for the quieter passages or instruments(+10db feels twice as loud). so overall that classical track will feel like it's a good deal quieter compared to the low dynamic song before, when both could reach 0db peaks.
 any replay gain solutions care about the overall dynamic of the song, instead of looking only at the highest value the peaks are reaching. if the target is 89db and the song has very compressed dynamic, then almost all the song will end up at around 89db. while the classical song might not be touched at all to prevent clipping, so now you feel like both songs sound as loud. but in reality the overly compressed one was made to play quieter to fit the dynamic one.
  
  
 and as I said several post above, if you look for sound quality, then you want to avoid clipping and avoid having the limiter to activate as much as possible. because they are both not totally transparent solutions as you noticed.
 about what you're asking for from the start, I don't know that it exist on foobar(but I'm not expert) there probably is a VST that does just that somewhere on the net. the thing is you're asking for peak normalization, and while it's perfectly possible, nobody listening to music would want to use that as a listening default setting. because as I explained for the low dynamic song vs highly dynamic song, you would end up with songs having very wild perceive loudness differences. so very annoying in the long run. the only reason why we use replay gain, mp3gain ,R128... in the first place is so that musics don't have too much perceived loudness disparities. so even if you find what you're asking for, I'm not sure how you would like it.


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## Joe Bloggs

giogio said:


> I have just tried the Peak Meter, it does NOT shows clipping. There is no red, no led for clipping. Or am I doing something wrong?
> Anyway I can at least see when I am getting close to 0. It can be useful at times.
> The advancer limiter is always there. And it definitely does not avoid clipping. I can notice it from the bad distortion when I am on red zone and from the clip led on Dirac flashing.




Oh, now I get it. You're using the Dirac D.A.P. DSP program, which is downstream of foobar.

Are you also doing the bass boost in Dirac?

foobar isn't clipping. (side note: the peak meter clipping indicator to the right will never light up when Advanced Limiter is engaged. At worst you'll hear volume pumping if the Advanced Limiter is working hard--if no post processing is being done after foobar)

There's nothing foobar can do to ensure that the signal doesn't clip downstream of it if it's post-processed.

But, Dirac should be upstream of the final master volume control of the PC (for the BT transmitter or whatever), and you should make sure that the latter is maxed out if you want the maximum digital volume without clipping (since, say, if the BT master volume is set at 50%, then digital signals from Dirac will clip at 50% rather than 100%).



> Which buttons? Ah, yes, you  mean, that one with a + on it, which is used to turn headphone on and off?
> (joke explained for dumb: all or nothing. loud or off.)




No...? On my Divine there are + and - buttons that adjust the volume?


----------



## Joe Bloggs

Also, if Dirac has taken over as the default sound output device, it may not be obvious where the actual volume adjustment for the physical output device (be it the 3.5mm jack or the BT driver or whatever) has gone. You'd have to look for it manually:



I have a similar situation on my computer. I set Line 1 (Virtual Audio Cable) as the default sound device, it captures all system sounds for me to process in a VST effects host before outputting to the actual sound card. Now, when I click the volume icon in the taskbar, the volume bar that shows up is the output to VSTHost (or in your case Dirac). You need to turn go to control panel and manually open the settings for your actual physical sound output device (be it "loudspeaker" or some BT USB device) and adjust its volume there under the "Levels" tab. Make sure the volume there is maxed out.


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## Mosstrekker192

How do you split a FLAC cue using foobar? I try opening the .cue, but I get error messages like this:
  
 "Unable to open item for playback (Error parsing cuesheet: invalid index list"
  
 I've tried Medieval Splitter, but that only worked once for me, where it almost seemed to split an album automatically. I'm trying it on another and I don't see any sort of options on how to do so. Wouldn't that just entail specifying start and end on a track and cutting songs from it? I haven't been able to find any good info on this.


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## dhmcclain1

canthearyou said:


> In Foobar click "file" "add folder". Now browse for the folder you ripped to(most likely "Music". Once you find it select it and click "ok"(or add).


 

 Thanks, canthearyou. I browsed thru any and all folders I could find. I did find one folder that had 4 of the missing songs, one of which was tagged incorrectly.
 Aside from that, I could find nothing. I'm fairly sure that I am not doing something that I should be doing. I have no recollection of what folder(s) they went to as
 alot of time has passed. Maybe if I rip another cd, I might be led to where the others are located. Don't understand why they don't remain in iTunes or Foobar or
 in whatever program they were ripped with. I know they were there before, as I put them on my iPod. Now they are gone.


----------



## oAmadeuso

mosstrekker192 said:


> How do you split a FLAC cue using foobar? I try opening the .cue, but I get error messages like this:
> 
> "Unable to open item for playback (Error parsing cuesheet: invalid index list"
> 
> I've tried Medieval Splitter, but that only worked once for me, where it almost seemed to split an album automatically. I'm trying it on another and I don't see any sort of options on how to do so. Wouldn't that just entail specifying start and end on a track and cutting songs from it? I haven't been able to find any good info on this.


 
 I use CUE Tools for jobs like that.
 http://www.cuetools.net/wiki/CUETools


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## Giogio

I was going to answer you in the last days guys, but Firefox deleted my answer before I could send it, and I got no wish to write again then.
 Here we go.
  
 Quote:


castleofargh said:


> for the low dynamic song vs highly dynamic song, you would end up with songs having very wild perceive loudness differences. so very annoying in the long run


 
 All clear, thanks, but you seem to have misunderstood something.
 The AGP (Automatic Gain Protection) of Neutron, the function I wish to find for Foobar, is not "instead of" Replay Gain. It works together with it.
 RG does its job and makes all tracks sound at same loudness. There should be no peaks, specially if you selected "avoid clipping" in mp3gain. So, AGP does nothing.
 But despite RG, you may still have peaks after you EQ.
 There and only there is where AGP intervenes.
 So, what you have described never happens. EVER.
 For it to happen you should have BOTH following situation together:
 1) you should push the EQ a LOT, specially in freq which are already near 0 (like bass in Techno)
 2) you should be listening to ALL your music with "shuffle all", mixing completely different things with different pace, style, dynamic etc
  
 I never do both things together:
 1) I only push EQ with Bassy music. Which you may say "why, if they are already bassy?". Because I am basshead but also audiophile. I do not want all music to be bassy. And bassy music still needs push, because it is music for clubs, not for home listening. I have first listened it in clubs. I am used to dance it, feeling the subs in my bones, in my stomach.
 And when I listen to that music at home, I need to FEEL the bass, its PHYSICAL POWER, not just hear its notes. Can you imagine what I mean?
 So, with all other music AGP would not intervene.
 2) I very very rarely do shuffle all and when I do it I do not have the EQ pushed. But generally I listen to music for genres, because each genre enhances a different emotion.
 I never mix classical with anything else.
 I never mix ethnic with anything else.
 I never mix jazz with anything else.
 I never mix chillout with anything else.
 I mix pop and rock, no big deal here.
 I never mix Bassy music with anything else. And I even divide Bassy music in different genera and I do not mix Dubstep with Techno. And peaks are generally only with Techno, and only with a few tracks which somehow must have been mastered that way and I can't solve it even by down RG manually at lower value.
  
 In other words, AGP would only intervene for few tracks, and only within one same genre, so among tracks with same dynamic.
 So, the scenario you have described, is only a scenario for me. It does not happen.
  


joe bloggs said:


> Oh, now I get it. You're using the Dirac D.A.P. DSP program, which is downstream of foobar.
> *yes*





> Are you also doing the bass boost in Dirac?
> *I have experimented with it. I use it for films or streamed music where I cannot EQ nor use other psychoacoustic bass enhancers. But I have seen that I can generally get a better result if I leave Dirac on the reference (neutral) filter, and I EQ+enhance the bass with plugins in foobar. This gives me a same or sometimes more powerful bass, without much sacrificing the bright highs of the reference filter (the boost filters lose some of that, specially the 3 and 4).** But I know what you wanted to say, they can cause Dirac to clip.*





> foobar isn't clipping. (side note: the peak meter clipping indicator to the right will never light up when Advanced Limiter is engaged. At worst you'll hear volume pumping if the Advanced Limiter is working hard--if no post processing is being done after foobar)
> *Foobar IS clipping. You are erroneously taking for granted that Dirac is clipping. I use Dirac since few weeks. I had the problem already only with Foobar.*
> *And I still have it when I do not use Dirac. Besides, the Advancer Limiter does help (I did some test yesterday observing with the Peak Meter) but it does not prevent all clipping nor distortion. And there is no clipping indicator in my Peak Meter. Where do you see such thing? Send me a screenshot please *





> But, Dirac should be upstream of the final master volume control of the PC (for the BT transmitter or whatever), and you should make sure that the latter is maxed out





> *It is, always. Although it makes no difference at all. Once Dirac is in control, the vol of the Notebook, or of the BT Device, can be either min or max and Dirac will sound the same.*





> On my Divine there are + and - buttons that adjust the volume?





> *It was a joke, meaning that I never reduce volume, so for me it only exists the +. Less than max vol is like "turned off". I am of course exaggerating to make it funny, but that is.*


 
  


joe bloggs said:


> Also, if Dirac has taken over as the default sound output device, it may not be obvious where the actual volume adjustment for the physical output device (be it the 3.5mm jack or the BT driver or whatever) has gone. You'd have to look for it manually


 
 I know that 
 But thanks for the good intention


----------



## castleofargh

at this point IDK. look for some VST that might do what you're asking for. I keep seeing the let's clip stuff and then ask for a dsp to lower the entire volume, to be malpractice. so I'm not totally sure anybody would think of making a DSP/VST for that specifically as a standalone. I suggested a solution many many post ago where you would convert your music for that very purpose (keep a back up!!!!!!!!!!!!!).  have a DSP EQ+ replay gain (in that order) in the "processing" option of the convert pop up.
 in the advanced option you can select the target for replaygain and maybe see how high you can get before it becomes clearly useless at doing some perceived loudness normalization. so that most musics would stay close to 0db instead of going for a target of average 89db or 77 or whatever you usually use.
  
 by doing this, you still get the risk of intersample clipping if you get too close to 0db, an the perceived loudness will at times feel like it's not working, so it's a slight fail in both instances, but the closest I can think of that would mostly do what you're asking for.
  
 in the end IMO the answer is to give up on those massive bass boosts, or to give up on BT headphone. without the EQ I suspect you can go loud enough with your headphone(or you simply are ruining your ears or something else is lowering the volume(like windows output not being at 100% on that output or foobar?).
 and without BT headphone you could have an amp making you o as loud as you could possibly wish without having to go clip the signal or even digitally push it to the max at all times(you could use the EQ the way it's supposed to be used 
	

	
	
		
		

		
		
	


	




).


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## Giogio

Now, I may have found something. It is not what I was looking for, but it allows to EQ with better bass results, and also pushing loudness, without clipping. I am still experimenting:
 It is called BootEQ and it is a VST normally used in music production, but which can be used with Foobar too.
 I have found it in *this list of free parametric EQ plugins*, while searching for an alternative to ElectriQ which goes DEEPLY on my nervs and I will never ever again use, as it crashes all the time (and I read that I am not the only one).
 I have downloaded many of those and I will keep testing, as this BootEQ is not really an alternative to ElectriQ, I use it mostly for the bass. It also has got one knob for low mids, one for upper mids, and one for highs, all of them with possibility to chose the frequency, so it can be used as general EQ, but I wanted something where I can SEE how the EQ curve is.
 What I love in BootEQ is that there is a pre-amp with drive, low-freq phase (I suppose it is why it makes bass so fat, a bit like Real Bass Exciter), and a Tube simulator.
 All with very high quality (as said, used to produce music).


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## Giogio

There are also some interesting free plugins on the KVR website.
 I have found one limiter called Maxwell Smart which is better than foobar's Automatic Limiter, specially in the smooth setting. Very simple, nothing to setup.
 And a peak meter (more than that) called kmeter, where it is easier to see the peaks. But as it installs as vst, you do not have it all the time open.


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## Joe Bloggs

I showed you how to install Electri-Q as a winamp plugin (which will not crash).

Looking at your replies, at this point I don't really feel like helping you.


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## Giogio

joe bloggs said:


> I showed you how to install Electri-Q as a winamp plugin (which will not crash).
> 
> Looking at your replies, at this point I don't really feel like helping you.


 

 You think you know, you think too much (Socrates) 
	

	
	
		
		

		
		
	


	



 This is the result I get by using your method, setting the winamp bridge at 16 or 24bit.
 Setting it at 32 bit does not crash but there is no change in the sound. Maybe you can tell me what I am doing wrong?
  
 Btw, with "I can't help you" do you mean because I did not use your winamp bridge and you are offended???


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## Giogio

@Joe Bloggs, my new friend, I just can't get your winamp thing to work in my foobar. It crashes and crashes and crashes. As said, 16 and 24 bit crash, 32 does not affect sound in any way.
 I have tried any possible combination of output (primary device, wasabi, dirac, no dirac).
 Same story.
  
 One more strange thing, your winamp plugin cannot find the electric-q in the components folder of my foobar, not in the vst folder where the electric-q exe (which I have downloaded from their site) install the dll.
 Your winamp plugin only recognize the electric-q dll in the components folder of your portable foobar.
 Maybe I should use your portable foobar to have the electric-q to work, but I am using my foobar, I want to use my foobar...


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## castleofargh

I went for easyQ instead some years back because the crashes annoyed me. it's IMO an inferior EQ(functionalities wise) but not crashing is a nice subjective bonus ^_^.
 anyway you can use any parametric EQ, that's not really an issue.


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## Giogio

Yeah, I have installed that too. It is less easy to use than the Electric-Q but it works.
 I am experimenting now with PhaseEQ, it has got a knobs interface, no eq curve, but has got HP, LP, LS, two peak freq (the normal kind of parametric eq but with knobs) and one HS.
 It has got a function to set the channels in l+r or m+s (mid + sides) which supposedly should give more focus on the low frequencies, which can also be achieved (they suggest) by HP to side (I have tried the HP at 320hz on "side", this let all what is above 320 go only on the sides, not in the middle, the effect is subtle). It also can be set to do 2x sampling which supposedly gives better sound.
 The BootEQ is also with knobs and half parametric. I have the impression that it gives a warmer sound even with no preamp. It is nice on same things. And the phase knob for the bass gives a fatter rumble.
 I find easyEQ a bit aseptic.
  
 A question: my BT headphones get only till 16 bit.
 But Dirac can be set at 16 or 24.
 Considering that I use wasapi (do you also think about eating sushi each time you talk of this plugin?), would I get better results if I set Dirac and Wasapi on 24, or at 16 like my headphones?
 This gets more complicated when I use the 3.5 BT Adapter and no Dirac. In that case I can set my soundcard will 24 bit at 192000hz. In this moment I still use the stock Foobar resampler so I would anyway have to "limit" my soundcard to 96000 otherwise it does not coincide with the resampler and wasapi does not like it. But, it makes any sense to set my soundcard (and resampler) so high, if this goes through the BT adapter and to my bt headphones?


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## castleofargh

24bit, they're the same samples as 16bit but with a bunch of zeros added at the end. so you can go back and forth 50times it won't create any problem adding or removing them is non destructive for the music. output 24bit anywhere you can. as long as the BT headphone is ok with the final signal, it's no problem.


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## Giogio

If you do not yet know TDR Nova you may want to give it a try.
 It looks MUCH better than EasyEQ and I think it sounds better and offers more possibilities.
  
  
 Question: what's the difference between "random" and "shuffle (tracks)" in the Playback/order menu?


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## castleofargh

shuffle can be customized(advanced option), and tries to follow a certain pattern within the randomness(same artist, or 1 full album then another random album in full. etc). random seems to be what it says .
  
 I no longer use easyQ, because it works fine for the most part and I try to suggest free stuff to people. I use it still for some simple examples that people can try to replicate if they like. last time I was "disappointed" by it, it was because using an EQ for left channel and one different for right created some too obvious phase changes. so compensating for some IEMs that had a few db too much around 5khz on the left(from my own measurements with a mic), didn't end up sounding balanced at all. the soundstage was shifted even thought they now measured the same in frequency response.
 it's no big deal and quite the specific problem, so I can't honestly blame a free EQ for trying to offer such a feature. ^_^
 Bob Katz explained to me some stuff about IIR vs FIR and linear filter or not, that could affect the imaging. TBH I thought it was all bollocks, but after testing, Mr Katz was absolutely right.
  
 anyway now I've bought DMG equilibrium, which is overkill for my noob amateur needs, but at least it does all I can ever ask of an EQ so I'm done wondering if it me or the EQ. when it sounds bad, it 100% on me now 
	

	
	
		
		

		
			





.


----------



## Giogio

What are FIR and IIR? I know they are filters, but, what do they do, how do they differ, and what does that acronym mean?
 Please in less mathematical and complicated terms as possible.


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## castleofargh

giogio said:


> What are FIR and IIR? I know they are filters, but, what do they do, how do they differ, and what does that acronym mean?
> Please in less mathematical and complicated terms as possible.


 
  
 it is a vast subject and it has really nothing to do with foobar problems and this topic. maybe go to sound science sub forum to ask about those stuff or look at what already might exist (if you really care)?
 even more so that in this case, I'm talking about simulating filter with a digital software so it's yet another topic compared to real FIR filters and stuff in the analog domain.


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## Giogio

castleofargh said:


> it is a vast subject and it has really nothing to do with foobar problems and this topic. maybe go to sound science sub forum to ask about those stuff or look at what already might exist (if you really care)?


 
 ???
 I thought you were talking of something for Foobar?
  
 I do care, I just have some problems when I have to read long complex technical stuff. It costs me to stay focused on text for more than 5 minutes.
 You know, ADS and that stuff. It takes me many hours to write/read something that other people could in half an hour.
  
 So I generally prefer to find people who can explain me things in a simple way, which does not require me to look in Wikipedia every two words.
  
 But if you were not talking of something related to Foobar, it's ok. I'll figure it out.


----------



## Maconi

castleofargh said:


> anyway now I've bought DMG equilibrium, which is overkill for my noob amateur needs, but at least it does all I can ever ask of an EQ so I'm done wondering if it me or the EQ. when it sounds bad, it 100% on me now
> 
> 
> 
> ...


 
 Wow, you went straight for the top eh? lol
  
 What made you go with DMG EQuilibrium over EQuality or EQuick? Possibly even FabFilter Pro-Q 2 (which compares more with EQuick I think)? About to dive into the world of VST EQ's myself.


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## Joe Bloggs

Hey Giogio,

My Electri-Q is the free 1.0 version whereas yours is... something else. I've also read another report of the two versions not playing well with each other and causing both to crash until "the something else" is uninstalled. So give that a try 

Regarding the peak meter, there's a small one that docks into the top panel. That's the one with black clipping "lights" to the right. The big Peak Meter that pops out instead shows clipped levels all the way out to +10dB the same way as it shows unclipped levels. However if your foobar master volume is not maxed out, you will have as much headroom above 0dB as the master volume is below zero. Hope that helps 

I also made sure that the winamp bridge for foobar that I got is the latest (last) available version; the earlier versions that are widely available on the net right now are explicitly stated by the author to crash with new versions of foobar. So, in short, uninstall other versions of Electri-Q, and copy dsp_eqfree and foo_dsp_winamp.dll from my foobar install to yours.


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## Giogio

joe bloggs said:


> @Giogio


 
 Thanks, but, my Electric-Q is free too. I have downloaded directly from their website.
 We may want to go more in detail in PM eventually.
  
 Anyway, I kind of like the TDR Nova more (it also has clipping led).
  
 Where do I find the dockable Peak Meter?
 I am using K-Meter which is very precise. It is a plugin, so I use it to check if my EQ setting is clipping or not, and then I close it.
 I am experimenting now because since yesterday I have a problem of Foobar ot ending its process in the task manager when I close it, and so not starting again when I want to re-open it. Instead, it opens a new process each time I click on it...
 I think it is a fault of Fidelizer which I have installed yesterday, but as sometimes I was able to solve this by taking the k-meter away from the chain, I am still not sure.
 The only sure thing is that all this sucks, time. Lot of time.
  
 Did any of you ever perceiv any improvement in sound using Foobar and Fidelizer together? It is difficult to compare because the free version of Fidelizer will not undo its changes to the system till you restart the machine. Ending the process wont work.


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## Joe Bloggs

giogio said:


> joe bloggs said:
> 
> 
> > [@=/u/410770/Giogio]@Giogio[/@]
> ...




I'm not saying yours isn't free  I know there were lots of free versions culminating in Christian Budde closing shop and offering the last version for free (?) on his website.

But I am saying that the original v1.0 "posihfopit" version (no longer appearing on Christian's website) is the one that works 

Whatever the case, we all ought to buy Christian a case of beer for the great work he did. Function-wise it's as though he outdid all his paid offering with the original free trial 



> Where do I find the dockable Peak Meter?



Right click on the dock...



> Did any of you ever perceiv any improvement in sound using Foobar and Fidelizer together? It is difficult to compare because the free version of Fidelizer will not undo its changes to the system till you restart the machine. Ending the process wont work.




I haven't tried Fidelizer. What windows version are you on?


----------



## castleofargh

maconi said:


> castleofargh said:
> 
> 
> > anyway now I've bought DMG equilibrium, which is overkill for my noob amateur needs, but at least it does all I can ever ask of an EQ so I'm done wondering if it me or the EQ. when it sounds bad, it 100% on me now
> ...


 

 yeah it was a little excessive, no doubt about it. I'm not sure I remember right, but I think EQuick didn't offer left and right EQ(that I really don't use much outside of making some dedicated µSD for a given IEM that has a big imbalance(so I do that maybe 3 times a year, but I'm glad I have it).
 and I didn't try equality but it seemed as if there was a limit in the number of points we can use???(would need to check I might be wrong about that). same thing, for day to day EQ I do not need 20 points, but sometimes when I go crazy and do some of the stuff Joe Bloggs shows in his EQ tutos, I'm glad to have big supply of points.
 and the DMG guy is clever, there are little settings and UI tricks you only have on the more expensive stuff. you can do the same with all, but it's a little faster/friendlier on equilibrium  once you have customized the UI(very customizable).
 also some have the linear filter option, other don't I think. it ends up with a lot of little things in settings or interface that most would never care about. but I guess actual pros might have a rational need for those stuff. but a pro I'm not ^_^.
  
 didn't try proQ, and I can't imagine why it wouldn't be an amazing EQ(as like DMG, so many pros use it). I sure value the proQ tutos on youtube, they're a treasure of information and very well made even for a noob like me. so I guess learning from them while using proQ could also be a nice bonus to get up to speed in the EQ world. but even with another EQ those videos are very clear.
  
 for day to day EQ without fancy left/right channel tweaks, I believe EQuick to be great. I paid extra for some stuff most people don't care about and there would be no point in paying for something we don't use. in fact I believe most free EQs to be great for day to day EQ. I just happened to test the trial version of equilibrium and cried when I had to go back to using easyQ. but it's really ease of use more than sound difference that decided me to pay. I'm not trained or young enough to tell if an EQ actually sounds better than another(unless it's a really crappy non parametric EQ).


----------



## Giogio

Does anybody know where Foobar stores the presets of the DSP chain? I have a lot of them which I need to delete. Doing it one by one is going to be slooow...


----------



## Soundsgoodtome

Hello guys, I'm resampling some hi-res flac (88, 96, 176, 192khz - 24bit) to 44.1 or 48khz - 16bit files on foobar. I'm using SOX resampler which I've heard has built-in dither (please confirm). 

Question is if SOX resampler plugin does have built-in dither, do I need to apply dither at the file output setting as well (pic below) or is that redundant and not necessary? If the latter, how does it affect sound quality?


----------



## Giogio

Does anybody know how to make Foobar the default program to play Audio CD (I find no option for that in Foobar and by opening AutoPlay in Win7 I am only listed VCL)?


----------



## PleasantSounds

giogio said:


> Does anybody know how to make Foobar the default program to play Audio CD (I find no option for that in Foobar and by opening AutoPlay in Win7 I am only listed VCL)?


 
  
 Under Preferences -> Shell Integration there is a checkbox "Set as the default CD player"


----------



## Giogio

pleasantsounds said:


> Under Preferences -> Shell Integration there is a checkbox "Set as the default CD player"


 

 I am deeply ashamed.
 I was looking for it under Display.
  
 Thanks


----------



## Giogio

And, ready for the next dumb question, is there a way to get an option to physically delete a file from the hard drive, not just remove it from the playlist?
 I find it unpractical that, while selecting music I want to keep or not, I have to open the folder and delete the file...
 Some players offer a way to do it within the player.
  
 And, a way to move a track (in this case not physically) to the next tab/playlist.


----------



## PleasantSounds

giogio said:


> And, ready for the next dumb question, is there a way to get an option to physically delete a file from the hard drive, not just remove it from the playlist?
> I find it unpractical that, while selecting music I want to keep or not, I have to open the folder and delete the file...
> Some players offer a way to do it within the player.


 
  
 Right click on the playlist item.
 Select File Operations -> Delete File


----------



## Giogio

pleasantsounds said:


> Right click on the playlist item.
> Select File Operations -> Delete File


 
 I told you it was the next dumb question.
 I must be tired.
 I had edited the question anyway.
 Do you by chance know the answer to the other matter?


----------



## PleasantSounds

You mean to move a track between playlists?
  
 If it is within the same playlist you can just drag the line to where you want it to be. Other than that the cut and paste works really well with the playlists.


----------



## Giogio

pleasantsounds said:


> Other than that the cut and paste works really well with the playlists.


 
 Yeah, it does, but somehow I felt that on a massive playlist it could be faster one single command "move to adjacent playlist".
 In the while I was using copy and paste.
  
 Do you know if it is possible to make the deleting faster? Like one shortcut, and no confirmation dialogue?


----------



## PleasantSounds

You can always select multiple files using Ctrl+click or Shift+click before you delete them.
  
 There is also a way to create custom shortcuts: check the Preferences ->: Keyboard Shortcuts.

Press the [Add New] button
in the tree view in the Action section scroll down to Context -> File Operations -> Delete File(s)
click the white field in the Key section and enter your preferred key combination, e.g. Ctrl+D 
press [OK] and you're done
  
 From then on pressing Ctrl+D should delete the selected items in the playlist and their corresponding files. But the conformation dialog will still show up and I don't know the way to bypass that. And personally I wouldn't want it, but let's not start discussing preferences....


----------



## Giogio

pleasantsounds said:


> From then on pressing Ctrl+D should delete the selected items in the playlist and their corresponding files. But the conformation dialog will still show up and I don't know the way to bypass that. And personally I wouldn't want it, but let's not start discussing preferences....


 
 works like a charm.
 Thanks again.
 I go to the oculist on Monday.


----------



## NorCa

Hey guys, anyone using Cd Art Display or something similar? I'm trying to make it work but well... if it did I wouldn't be here asking xD. I've seen some foobar components to integrate cad but no luck so far.


----------



## PleasantSounds

Have you seen this post?  It contains a paragraph on displaying album art, which should be easy enough to follow.


----------



## NorCa

Hey thanks for the link, however that's not what I meant. CdArt Display is a small program that shows a minified (skinnable) player on the desktop which shows currently playing song and may or may not have simple controls like pause next stop for the music player.


----------



## PleasantSounds

Well, as you can guess I had no idea that such a thing exists, so I'm afraid I won't be of much help.
  
 If you don't get any help here you can always try the CD Art Display forum. Not much traffic there, but hopefully authors of the software scan it sometimes.


----------



## Soundsgoodtome

How do I use a VST wrapper and something like Electri-Q? I don't quite understand how to get the two to worth together..
  
  
 Also, can anybody input with my question below?
 Quote:


soundsgoodtome said:


> Hello guys, I'm resampling some hi-res flac (88, 96, 176, 192khz - 24bit) to 44.1 or 48khz - 16bit files on foobar. I'm using SOX resampler which I've heard has built-in dither (please confirm).
> 
> Question is if SOX resampler plugin does have built-in dither, do I need to apply dither at the file output setting as well (pic below) or is that redundant and not necessary? If the latter, how does it affect sound quality?


----------



## Giogio

soundsgoodtome said:


>


 

 I am not an expert but I want to answer because nobody does.
 For the wrapper, you open the wrapper and use its search function to look for the dsp you want to load. It is that easy.
 So, you just have to install the wrapper in Foobar, not the other dsp. You load the wrapper in the dsp chain, open it, search for the dsp to load, end.
 I had problems with ElectriQ and I use now other EQ which I find much better, but if you want that one you should ask @Joe Bloggs because he has more experience in that.
  
 For SoX in my very limited knowledge I can tell you I understood that dither is only useful when downsampling. And in that case should always be used. I suppose it is absolutely harmless if you leave it active also when you are playing files which are already at 16bit, because they would not be resampled, I think.


----------



## Giogio

Now I have a question too: does anybody know how to set the advanced settings relative to playback?
 I have understood that it may be beneficial to change them, to have better performances. But I ignore what they really do and which settings would be better.
 Particularly I would like to understand:

full file buffering up to
wasapi
thread priority
  
 EDIT: other question: why when I scan a file (flac or mp3) for replay gain, the result for that file is always the same amount of db, no matter which target value I set in the advanced options? I would think that if I set 70 I should get less db than if I set 100...
 EDIT2: strange, the same track, ripped in mp3, if I scan for replay gain with foobar it suggests -5.25db (as said, always, no matter which target value I had set), but if I apply first replay gain with mp3gain at 88.8 and in foobar I right click the file for properties, foobar shows that the value is -0.43db, and if I scan it foobar suggests -0.73. If I delete all replay gain values in that track, foobar still suggests -0.73...
 What???
 Why does foobar suggest two different value for the same track when no replay gain info is present?


----------



## castleofargh

soundsgoodtome said:


> How do I use a VST wrapper and something like Electri-Q? I don't quite understand how to get the two to worth together..
> 
> 
> Also, can anybody input with my question below?
> ...


 
 you install VST 2.4 adapter in foobar's components tab( in file->preferences->components)
 you install any VST(like electriQ) in the VST plug in tab (should be right under the component one in preferences).
 you need to restart foobar so that it will load the added stuff, then go to the DSP manager tab (under playback) and move electriQ to the left to use it. at that point electriQ should show up in foobar under view->DSP along with any other DSP you have added to the left in preferences with the DSP manager.
  
  
 about SOX, I hoped that someone with actual knowledge would show up but I guess I'm all you've got, which isn't much. the SOX plugin for foobar is a resampler and doesn't care about bit depth and so doesn't care about dither. with SOX ( the real deal outside of foobar) and advanced commands you can specify a all lot of things, including bit depth and dither. but I don't think that applies to the default sox plugin for foobar which really is a basic resampler(or maybe I missed something?).
 now if you convert to a lower bit depth, IMO dither is always welcome. the circled option on your screenshot is applied at the end after all the possible DSPs you included to your conversion.


----------



## Giogio

In the sox plugin it's called aliasing but s far s i understood is same as dither.

About the vst wrapper he asked for electriq and that would not work with the wrapper you say. It would crash.


----------



## castleofargh

giogio said:


> In the sox plugin it's called aliasing but s far s i understood is same as dither.
> 
> About the vst wrapper he asked for electriq and that would not work with the wrapper you say. It would crash.


 

 I'd say no to both ^_^.   anti aliasing is a matter of where/how to low pass the signal, to get it band limited and remove any possible unwanted ultrasound energy. so it makes sense in a resampling process to apply a new frequency limit.
 dither is all about noise near the least significant bit, to mask the quantization noise. so it makes sense when you lower the bit depth of a signal, as it creates a new louder quantization noise that may benefit dithering.
 those are 2 very different things.
  
  
 and about the wrapper crashing with electriq, I just added electriQ and made this amazing EQ and screenshot. no crash.

 not that electriQ is stable, it does crash a lot when you manipulate settings, then it's fine if left alone to play music. but when I started with foobar I had the other wrapper and electriQ was still the most unstable VST I had. I don't believe the wrapper is the cause of the problem.


----------



## Giogio

castleofargh said:


> those are 2 very different things.
> I don't believe the wrapper is the cause of the problem.


 
 Thanks, I did not know.
  
 I also believe the problem is ElectricQ, but Joe seems to think that some wrappers solve the problem.
 Anyway you are right, when no manipulation, no crash.


----------



## Soundsgoodtome

Thanks for the input guys, the SOX issue is still up in the air.

 As for the VST wrapper, I actually need a parametric EQ so it doesn't necessarily have to be in a wrapper if there's a plugin for one. What do you guys use for Parametric EQ?


----------



## Giogio

soundsgoodtome said:


> What do you guys use for Parametric EQ?


 
 First of all, you should consider that there are parametric EQ which are not graphic, I mean, without the graphic curve, but just with knobs.
 As long as you can decide freq and Q, it is parametric.
 Phase EQ is excellent.
 NCL EQ is from the same guy, different approach.
 Then, with graphic curves:
 Filtrate LE
 ReaEQ and ReaFir
 TB Equalizer V3
 TB Nova
  
 None of them needs any wrapper. None of them ever crashes.


----------



## leggy

Hi everyone,

I searched this thread and some other and couldn't find an answer to my question below:
I have Windows 7 Pro, Foobar2000 and iBasso D14 dac/amp that supports up to 384/32 bit. However, I am unable to play 352/24 flacs from my laptop using Foobar but I can play it from my android phone connected to the D14 mentioned above without problem using UAPP player.
Is there a way to play those files from my computer wether through Foobar or another software?

Thank you


----------



## PleasantSounds

Not a lot of info to work with...
 Have you installed any drivers for the iBasso? If so, do they show up in foobar? Is there a control panel for them? Can you play any files at all? Up to what resolution?
  
 One thing that comes to mind if the native playback at this resolution is not available, you can always try using a resampler and get the bit rate down to 96k or lower if necessary.


----------



## castleofargh

leggy said:


> Hi everyone,
> 
> I searched this thread and some other and couldn't find an answer to my question below:
> I have Windows 7 Pro, Foobar2000 and iBasso D14 dac/amp that supports up to 384/32 bit. However, I am unable to play 352/24 flacs from my laptop using Foobar but I can play it from my android phone connected to the D14 mentioned above without problem using UAPP player.
> ...


 

 .
 are you sure your phone doesn't just downsample the file before sending it? is there a way to check the resolution the D14 is receiving? sorry I'm bringing more questions than answers, I have never even owned that kind of file, and have no device that could play them , so I can't even hope to test your situation.


----------



## leggy

pleasantsounds said:


> Not a lot of info to work with...
> Have you installed any drivers for the iBasso? If so, do they show up in foobar? Is there a control panel for them? Can you play any files at all? Up to what resolution?
> 
> One thing that comes to mind if the native playback at this resolution is not available, you can always try using a resampler and get the bit rate down to 96k or lower if necessary.



Latest Windows driver is already installed and the device does show up under output in Foobar control panel and that is the one selected. 
The next highest flac that I have beside the 352/24 is 192/24 which plays normal.
Whenever I try to play the 352 I get Foobar error message.
As for down sampling, I bought one album at this resolution to compare it with mp3 320kbps so I am not looking to down sample it.
Thank you


----------



## leggy

castleofargh said:


> .
> are you sure your phone doesn't just downsample the file before sending it? is there a way to check the resolution the D14 is receiving? sorry I'm bringing more questions than answers, I have never even owned that kind of file, and have no device that could play them , so I can't even hope to test your situation.



UAPP on android shows the file rate being played and at which rate the device is receiving/playing the song and in this case both are showing 352/24.


----------



## leggy

leggy said:


> Hi everyone,
> 
> I searched this thread and some other and couldn't find an answer to my question below:
> I have Windows 7 Pro, Foobar2000 and iBasso D14 dac/amp that supports up to 384/32 bit. However, I am unable to play 352/24 flacs from my laptop using Foobar but I can play it from my android phone connected to the D14 mentioned above without problem using UAPP player.
> ...


 
 Problem solved. Now I can play flac 352/24 on my pc through my D14 dac/amp. I follow these steps
 http://audinst.com/en/faqs/2441
  
 However, there are tiny skipping/jitter/ticking, not sure how to explain it, when playing these files. Have anyone had such effect when playing other formats or rates and how it was solved?


----------



## Sp12er3

It's a small question, but important for me nonetheless, How do you make the playlist work when you change your music directory? I move most of (if not all) of my saved track into external HD as some app need room to be installed, but now i'm at lost on how to use my extensively used Playlist... will be anxiously waiting for response from you guys... thanks


----------



## PleasantSounds

The trick is to move the files using the f2k itself, not the explorer.
 You can highlight the content of a playlist (or part of it) and go File Operations -> Move To... 
 This way all the open playlists referring to that file will be updated at the same time.
  
 I'm not sure though if there is a smart way to restore the orphan entries in a playlist - for that you may have to re-insert them into the playlist from the new location.


----------



## Sp12er3

pleasantsounds said:


> The trick is to move the files using the f2k itself, not the explorer.
> You can highlight the content of a playlist (or part of it) and go File Operations -> Move To...
> This way all the open playlists referring to that file will be updated at the same time.
> 
> I'm not sure though if there is a smart way to restore the orphan entries in a playlist - for that you may have to re-insert them into the playlist from the new location.


 Thanks ☺ will try it out.
Though thinking again, i means i have to create a duplicate as back up first isn't it? As it will break my current sort by Artist folder structure... And choosing them one by one when one track isn't playable is a bit of a chore.


----------



## konseki

Need serious help here.
  
 I ripped CDS using EAC to wav........sound pretty good but not wow wow fantastic to me.
 So thinking is the problem with my laptop sound card?
  
 Or will ripping CDs to wav with Foobar do the trick? i tried google and followed the instruction...gives me errors!! Even converting to flac gives me errors!!!!
 Playing with Foobar makes my CDs sound wayyyy better.
  
 Help help,


----------



## A_Man_Eating_Duck

Both rippers will sound the same as long you are not manipulating audio afterwards (DSP\EQ\Volume change). I have tested DBPoweramp, EAC, Foobar2000 and Cuetools ripper with a clean disc and all the outputs are exactly the same, checked with checksums, all match. 

Any chance of getting a screenshot of your foobar2000 error or could you provide a step by step of what you are doing?


----------



## konseki

a_man_eating_duck said:


> Both rippers will sound the same as long you are not manipulating audio afterwards (DSP\EQ\Volume change). I have tested DBPoweramp, EAC, Foobar2000 and Cuetools ripper with a clean disc and all the outputs are exactly the same, checked with checksums, all match.
> 
> Any chance of getting a screenshot of your foobar2000 error or could you provide a step by step of what you are doing?




I see. For EAC i use microsoft pcm 44100khz/16 bits n convert to wav....dont sound like CD quality still as played on sony CD walkman. For foobar i set to standard ripping mode n controlled speed...choose 16 bits and no tagging or whatsoever and output to Wav. Foobar seems to sound more consistent but not as 100% clear as EAC. Ripped n store all in western digital hard disk. So not sure if its CD quality either...or soundcard.

Not sure if i should use flac. Coz external compression to flac sure it sounds same as output wav.
But the non external compression to only wav sounds better to me.

If i add DSP will the songs sound better?


----------



## A_Man_Eating_Duck

Install this bit compare plugin for foobar2000. http://www.foobar2000.org/components/view/foo_bitcompare

the best thing you can do is rip the same song with EAC and Foobar2000. Then load both wav files in to Foobar2000, select both files and then right click utilities > bit compare tracks. 

If the results come back as no difference found then the file are exactly the same (it cannot be the source files that sound different).

please post your results.

Use FLAC or another lossless compression method of your choice (Wavpack, Monkeys audio, TAK), there are a lot of advantages to use lossless compression over WAV, HD space saving, Proper tagging support, checksums... FLAC is a popular choice since it is compatible with a lot of media players. 

Don't change audio when you are ripping music, later on you might think why the hell did i make those changes to the source files they sound terrible with my new gear and you'll need to re rip everything again. If you want to tweak the sound do it on playback.


----------



## konseki

a_man_eating_duck said:


> Install this bit compare plugin for foobar2000. http://www.foobar2000.org/components/view/foo_bitcompare
> 
> the best thing you can do is rip the same song with EAC and Foobar2000. Then load both wav files in to Foobar2000, select both files and then right click utilities > bit compare tracks.
> 
> ...


 
 Downloaded the plugin. Read the guide but still cant figure out how to load both outputs and compare??? 
	

	
	
		
		

		
		
	


	




 
  
 Can also compare wav to flac right? 
	

	
	
		
		

		
		
	


	




 
  
 I have certain collection where my discs have scratches....very not clean.


----------



## A_Man_Eating_Duck

Drag and drop both songs in to foobar2000s playlist. then select both, right click one of the selected files > utilities >bit compare tracks.

start with a clean disc first, error correction can differ depending on drive and software.


----------



## konseki

a_man_eating_duck said:


> Drag and drop both songs in to foobar2000s playlist. then select both, right click one of the selected files > utilities >bit compare tracks.
> 
> start with a clean disc first, error correction can differ depending on drive and software.


 

  
 No difference.


----------



## A_Man_Eating_Duck

Thats good, the ripping is clean and a match. it looks like you haven't configured the drive offset of -6 on either EAC or Foobar2000. 

I would assume EAC hasn't got it configured.

open EAC > click EAC menu > Drive options | Offset / Speed tab. Select *Use read sample offset correct* and enter *-6*.

rerip the same song in EAC and compare the results with the Foobar2000 rip. 

please post your results.


----------



## Joe Bloggs

A 6 sample offset is only going to affect where the song starts, not how the song sounds like... of course if your reference is playing the CD on a whole different system (the Discman) then all bets are off... could just be you preferring the amplifier / tone settings on the Discman to that on the computer and its sound card.


----------



## A_Man_Eating_Duck

I don't recall saying that there would be a change in sound.

All i was suggesting was to correctly setting up the offset for future ripping. the re ripping and comparing was to ensure that the configuration was changed correctly.


----------



## Joe Bloggs

That's nice, but a perceived sound difference is what sent the OP posting here in the first place.


----------



## A_Man_Eating_Duck

hopefully konseki has twigged that no differences found means that the source files are the same so any changes in sound has to be from the playback chain.


----------



## JacobAllison

Question that I can't seem to find anywhere: Does Foobar2000 mobile for android support DSD output when used with a DAC such as the OPPO HA-1?
  
 If it is I can't figure out how to do it, want to make sure I'm not wasting my time.
  
 Is there a particular point? Not really, no. But hey, nice toys, it's fun. DSD is like the vinyl of the computer audio world.  (The thing that really attracted me to vinyl was the expense and the inconvenience.)


----------



## 514077

Glad I found this thread, and that DSD popped up.
 I've gotten DSD to work somewhat through my Hugo and configured it with ASIO4ALL and the DOP thing.  I can play DSD with the PCM setting at 352,000.  When, however, I try to play in DSD mode, the FB status bar shows DSD645000 or so, followed by the 2.8...MHz indicating it's DSD.  But, I can't hear the full track as FB seems to run through the track at least 4X speed.  The track plays normally right up until the song ends on the timer.  Can anyone help?  From what I've heard from my portable, I like the sound of the format.  I will not get into a spitting contest about the worth of DSD or not.  I choose to use it and enjoy it.
 Thanks.


----------



## JacobAllison

uelong said:


> Glad I found this thread, and that DSD popped up.
> I've gotten DSD to work somewhat through my Hugo and configured it with ASIO4ALL and the DOP thing.  I can play DSD with the PCM setting at 352,000.  When, however, I try to play in DSD mode, the FB status bar shows DSD645000 or so, followed by the 2.8...MHz indicating it's DSD.  But, I can't hear the full track as FB seems to run through the track at least 4X speed.  The track plays normally right up until the song ends on the timer.  Can anyone help?  From what I've heard from my portable, I like the sound of the format.  I will not get into a spitting contest about the worth of DSD or not.  I choose to use it and enjoy it.
> Thanks.


 
  
 So in my experience with DSD and FooBar is that it completely depends on the specific driver implementation and the specific SACD plugin you use. My Oppo HA-2 works just fine with the very newest SACD plugin implementation, but my HiFiMeDIY 9018 async DAC only worked with one specific version of the SACD plugin. When using the latest driver I'd have problems where it would play too fast, or not play at all.
  
 Try and look on the DAC manufacturer's page to see if they have information on how to get it working!


----------



## 514077

jacoballison said:


> uelong said:
> 
> 
> > Glad I found this thread, and that DSD popped up.
> ...


 
 Out of curiosity, which version worked with your mediy 9018?  I'm using Foo-input-sacd.097.


----------



## JacobAllison

Uh, I beleive it was 0.8.4, the last one with the seperated out configurations.


----------



## 514077

jacoballison said:


> Uh, I beleive it was 0.8.4, the last one with the seperated out configurations.


 

 You were right!  I uninstalled 097 and installed 0.8.4, and it looks like it fixed the problem.
 Thanks!
 Kevin


----------



## JacobAllison

uelong said:


> You were right!  I uninstalled 097 and installed 0.8.4, and it looks like it fixed the problem.
> Thanks!
> Kevin




Glad I could help!


----------



## theaudiologist

guys i have a problem:
  
 i downloaded a 32bit 384kHz Wavpack file and i try to convert it to flac. the sample rate stays the same (thankfully) but whenever I convert the bit depth keeps changing to 24-bit.


----------



## canthearyou

theaudiologist said:


> guys i have a problem:
> 
> i downloaded a 32bit 384kHz Wavpack file and i try to convert it to flac. the sample rate stays the same (thankfully) but whenever I convert the bit depth keeps changing to 24-bit.




Does your DAC support 32bit? Do you have 32bit enabled in Windows control panel?

I may have read that wrong. Is it going to 24bit during the converting process, or during playback after converting?


----------



## theaudiologist

through the converting process. (i use foobar on my mac using wine but only for converting files).
  
 i want to convert a 32-bit 384kHz wavpack into a flac one, but instead it gives out a 24-bit 384kHz Flac file, not a 32-bit one. I put the output bit depth on flac conversion settings on both "auto" and "32-bit" but both give out a 24-bit file.


----------



## 514077

theaudiologist said:


> through the converting process. (i use foobar on my mac using wine but only for converting files).
> 
> i want to convert a 32-bit 384kHz wavpack into a flac one, but instead it gives out a 24-bit 384kHz Flac file, not a 32-bit one. I put the output bit depth on flac conversion settings on both "auto" and "32-bit" but both give out a 24-bit file.


 

 Just a thought, but have you tried AIFF or WAV?  Just a straight uncompressed file?  IDK if there would be a difference, but it could be an exercise in trouble shooting.
 Curious; where is that file available?  I didn't know 32-bit files were that common.


----------



## theaudiologist

i converted to flac both straight from wavpack and after converting to wav. but either way ended up 24 bit


----------



## timb5881

Nice thread with lots of info on foobar, I like it.


----------



## 514077

theaudiologist said:


> i converted to flac both straight from wavpack and after converting to wav. but either way ended up 24 bit


 
 I use DBPowerAmp which is now available for Mac.  You could get a 21-day trial and see if that works.  Problem is, the bit depth of a 32-bit file can't be heard on most systems, if I understand correctly.  A 24-bit file goes down to about -120db.  And most DACs can get, at most 20 bits of useful content from a 24-bit file.
 I'm not talking down the desire and your choice of files you use.  I happen-by choice-to like DSD, and got some good help on this forum without all that "YOU DON'T NEED DSD," stuff.
 I wish I could help more.  But, I'd give the DBPA a try.


----------



## PleasantSounds

theaudiologist said:


> i converted to flac both straight from wavpack and after converting to wav. but either way ended up 24 bit


 
  
 I have tried this and my results are the same: selection of 32-bit output results with a 24-bit file. 
 Also, the converted file loses a lot of quality. I expected I wouldn't be able to detect any difference, but the converted file is veiled and much less dynamic, as if it had max 12-bit resolution.
 Looks to me like it's a f2k bug.


----------



## A_Man_Eating_Duck

theaudiologist said:


> through the converting process. (i use foobar on my mac using wine but only for converting files).
> 
> i want to convert a 32-bit 384kHz wavpack into a flac one, but instead it gives out a 24-bit 384kHz Flac file, not a 32-bit one. I put the output bit depth on flac conversion settings on both "auto" and "32-bit" but both give out a 24-bit file.


FLAC doesn't support 32bit.


----------



## PleasantSounds

a_man_eating_duck said:


> FLAC doesn't support 32bit.


 
  
 Where did you get that from?
  
 According to xiph.org it does


----------



## A_Man_Eating_Duck

Sorry mixing up 32bit floats and 32bit int. Could this source file be a 32bit float?


----------



## theaudiologist

apparently yes, it's 32 bit floating point PCM


----------



## A_Man_Eating_Duck

If you want to use FLAC you'll need to convert it from float to integer, which foobar2000 is doing automatically for you. You can force it to convert from 32bit float to 32bit int by going in to the converter setup > output format | change the output bit depth from auto to 32-bit (sorry, just tested this and you are right, just converts to 24bit). This is not a lossless process but it should be very difficult \ impossible to hear the difference.

If you do find a very noticeable difference please post a small source sample (10-20seconds) so we can demonstrate the problem to the Foobar2000 devs just in case you have uncovered a bug.


----------



## castleofargh

are people really selling 32bit records? I can understand having 32bit ability in a DAW to get ever more headroom for some reason, but I don't know of an ADC capable of resolving 32bit(and for which purpose? listen to ants in the wall 2 rooms apart? and then of course there is that other small detail, no sound system can output even 24 bit resolution to this day, let alone 32bit.
  so why would music be exported to 32bit?


----------



## headdict

castleofargh said:


> are people really selling 32bit records? I can understand having 32bit ability in a DAW to get ever more headroom for some reason, but I don't know of an ADC capable of resolving 32bit(and for which purpose? listen to ants in the wall 2 rooms apart? and then of course there is that other small detail, no sound system can output even 24 bit resolution to this day, let alone 32bit.
> so why would music be exported to 32bit?


 

 Come on, you already know the reason. More bits sell for more bucks. Simple math.
  
 Just out of curiosity: Does anybody know how to calculate the dynamic range of 32 bit floating point PCM? I'm sure it makes a huge difference, maybe even a measurable one. Too bad we human beings can't hear it. We can still see and compare the numbers, though, and we tend to take numbers very seriously. Today many smartphones have a 64 bit architecture. When will we see 64 bit audio files?


----------



## RRod

headdict said:


> Come on, you already know the reason. More bits sell for more bucks. Simple math.
> 
> Just out of curiosity: Does anybody know how to calculate the dynamic range of 32 bit floating point PCM? I'm sure it makes a huge difference, maybe even a measurable one. Too bad we human beings can't hear it. We can still see and compare the numbers, though, and we tend to take numbers very seriously. Today many smartphones have a 64 bit architecture. When will we see 64 bit audio files?


 
  
 http://diwaves.com/tmp/sample_bits.htm


----------



## headdict

rrod said:


> http://diwaves.com/tmp/sample_bits.htm


 

 Very interesting! So floating point has no dynamic range advantage over integer. Why is it used then?


----------



## RRod

headdict said:


> Very interesting! So floating point has no dynamic range advantage over integer. Why is it used then?


 
  
 The article lists these advantages:


> Unlike with integers, and because the mantissa is normalized, the quantization error is not fixed. It is relative to each sample value. This has important advantages over integers:
> 1)The effective resolution for every sample is the full resolution that the mantissa allows, while the effective resolution of an equivalent size integer corresponds only to the significant bits used by the actual value.
> 2)The SNR is constant regardless of the signal level.
> 3)The quantization noise is relative to each sample value.  For any sample value smaller than 0 dB, the quantization noise is also smaller by the same ratio.
> 4)Even if a signal is normalized so its peaks are 0 dB, samples that are not peaks, but smaller values, exhibit a smaller quantization noise, always a fixed amount relative to the value itself. So while the maximum error in this case is the same (for the 0 dB values), the average is smaller (every other value has smaller error). Note also that our ear is more sensitive to distortion and noise in the parts of the waveform closer to zero, and less in the peaks. More exactly is more sensitive to the differential or slope, which is highest near zero and minimal at the peaks for a sine wave, and statistically similar for any signal.


 
  
 So it seems to sum up as "it's better for the softer parts", which is where we tend to notice quantization error. Practically I doubt this counts much versus straight up # of bits, in that I'd be surprised if the difference between a good and bad workflow was int vs float. There's also the "no clipping" thing, but we're talking about stuff happening in a DAW; for the ADC you still have clipping to worry about and you don't have 32-bits of dynamic range or SNR.


----------



## timb5881

A couple of questions about the trial foobar mobile. I have it on my iPad and iPhone. Will it stream 24/96 files with the plugin on my computer to my iPhone through foobar? So far it seems to be, when I do a stream on the foobar on the computer. It only seems to allow me to only acces my iTunes library without streaming music at the same time. So if I use a dac on my iPhone , like my Cyberdrive feather, can I stream dsd and other high resolution music? Can I acces them as files on my phone, high Rez, that are not in my iTunes folders?


----------



## Traveller

tilpo said:


> Do you want to discuss the sound science of bit-perfect playback?


  
 Actually, I do 
	

	
	
		
		

		
			





  
 A silly question perhaps, but why does Foobar2000 allow for volume control with DACS that have own (HW) volume controls?
 I use Foobar2000 on my Windows-based Notebooks and UAPP on my Android-based Smartphone. UAPP is able to determine which of my DACs have HW volume controls (ex. Mojo) and which do not (ex. AQ Dragonfly).
  
 When it comes to assuring "bit-perfect" data transmission, source-side volume control makes me nervous... 
	

	
	
		
		

		
		
	


	



  
 Thanks for any feedback you may have


----------



## harpo1

I have a weird issue and I'm not sure what is causing it so any help would be appreciated.  I have a 1 TB hard drive connected to my Asus router.  I copy a foobar folder over to it and it's fine for a while.  Then usually within a week I open foobar2000 and my library is gone.  Look on the hard drive and the foobar folder is gone.  This has been going on now for about 6 weeks.  Any ideas?


----------



## canthearyou

harpo1 said:


> I have a weird issue and I'm not sure what is causing it so any help would be appreciated.  I have a 1 TB hard drive connected to my Asus router.  I copy a foobar folder over to it and it's fine for a while.  Then usually within a week I open foobar2000 and my library is gone.  Look on the hard drive and the foobar folder is gone.  This has been going on now for about 6 weeks.  Any ideas?




Did you set a static IP address for the storage?


----------



## harpo1

canthearyou said:


> Did you set a static IP address for the storage?


 
 It's connected to the usb port on the router so it doesn't show up in the network map.


----------



## Traveller

traveller said:


> ...why does Foobar2000 allow for volume control with DACS that have own (HW) volume controls?  I use Foobar2000 on my Windows-based Notebooks and UAPP on my Android-based Smartphone. UAPP is able to determine which of my DACs have HW volume controls (ex. Mojo) and which do not (ex. AQ Dragonfly)...


 
 One more for the list; why is it that with WASAPI one has to select the Bit-depth? Again, I would expect Foobar to set the bit-depth based on the source... 
	

	
	
		
		

		
		
	


	



 ASIO does not have this option but I'm not quite happy with the sonic signature and prefer to use WASAPI.


----------



## PleasantSounds

traveller said:


> One more for the list; why is it that with WASAPI one has to select the Bit-depth? Again, I would expect Foobar to set the bit-depth based on the source...
> 
> 
> 
> ...


 
  
 Possible that switching between 16 and 24 on the fly has some side effects, or it's simply a design choice that could have been different, but evidently considered no big deal. If you're really curious why then check at the hydrogen audio where the f2k developers hang out.
  
 In practice if your DAC supports 24-bit then set it to 24, otherwise 16. No adverse effects with playing 16-bit files through 24-bit interface.


----------



## Traveller

pleasantsounds said:


> ...In practice if your DAC supports 24-bit then set it to 24, otherwise 16. *No adverse effects *with playing 16-bit files through 24-bit interface.


 
 Thanks for the feedback, sir! Will check out Hydrogen when I get a chance. [2/3 of] my USB DACs support 24-bit+.
  
 Is this then a case of "padding" the remaining bits with zeros when working with lower-bit-depth source music? And we're sure the DAC isn't _processing_ the addition bits _(waste of resources / battery, etc)?_


----------



## PleasantSounds

Correct on the "padding".
  
 Regarding what impact does it have on the DAC's working temperature, RF emissions or other side effects - I would expect that the 24-bit DACs do the "padding" internally on 16-bit streams just to have one uniform way of processing. So in the end exactly the same stream would be converted whether you send a 16-bit stream or padded to 24-bit.


----------



## castleofargh

Quote:
   


traveller said:


> One more for the list; why is it that with WASAPI one has to select the Bit-depth? Again, I would expect Foobar to set the bit-depth based on the source...
> 
> 
> 
> ...


 

 it's an asio/wasapi thing. foobar does what most players would do. asio by default switches to the highest bitdepth it can detect(based on asio drivers provided for the device). it's the default action and the recommended action for asio, so there is little meaning in adding a bith depth setting when using asio.
 now wasapi doesn't care and usually doesn't have device specific drivers. so it uses what you want it to use(if the DAC can handle it), so of course a bit depth setting makes sense.


----------



## Traveller

castleofargh said:


> ...so it uses what you want it to use(if the DAC can handle it), so of course a bit depth setting makes sense.


 
 It makes sense to someone with a little more knowledge on the subject than me... 
	

	
	
		
		

		
		
	


	




 So what happens when a 24b96KHz source is fed via WASAPI set to 16bit? Do the additional 8 bits of data get dropped


----------



## castleofargh

traveller said:


> castleofargh said:
> 
> 
> > ...so it uses what you want it to use(if the DAC can handle it), so of course a bit depth setting makes sense.
> ...


 
 yes, that's why you have "dither" option for the 16bit setting(well you may have a use for dither for other reasons, but the obvious one is when the original file is 24bit played at 16).


----------



## Soundsgoodtome

His issue is there is no dither option on asio mode...is there? 





castleofargh said:


> yes, that's why you have "dither" option for the 16bit setting(well you may have a use for dither for other reasons, but the obvious one is when the original file is 24bit played at 16).


----------



## Traveller

Sigh... but that's my point re. WASAPI - why on Earth would anyone want to pipe their $$ 24-bit hi-res to their $$24-bit+ DACs using a 16-bit pipeline 
	

	
	
		
		

		
		
	


	



 But as recommended, I have set it to 24 bits and any 16-bit source data will simply be padded.


----------



## castleofargh

wasapi is a simple more universal approach. you don't need to install specific wasapi drivers for every new DAC you get like for asio. if the you don't get custom asio drivers with your DAC, that's it. it's IMO the cool part of wasapi. but the consequence I guess is that wasapi doesn't have a custom made driver with all the information about the DAC. it's the same thing with direct sound in windows, you set the output yourself.
  
 in general, if your DAC came with asio drivers(not the asio4all mess), and it works fine, I would suggest to use asio.


----------



## 514077

castleofargh said:


> wasapi is a simple more universal approach. you don't need to install specific wasapi drivers for every new DAC you get like for asio. if the you don't get custom asio drivers with your DAC, that's it. it's IMO the cool part of wasapi. but the consequence I guess is that wasapi doesn't have a custom made driver with all the information about the DAC. it's the same thing with direct sound in windows, you set the output yourself.
> 
> in general, if your DAC came with asio drivers(not the asio4all mess), and it works fine, I would suggest to use asio.


 

 I didn't realize ASIO4All was junk.  I'm using a Hugo.  If I removed A4All, would it remove asio use, or is the Chord driver enough?  Bare in mind, I'm still stuck with XP; can't afford a new machine yet.
 Thanks for any advice you'd care to give.


----------



## Soundsgoodtome

traveller said:


> Sigh... but that's my point re. WASAPI - why on Earth would anyone want to pipe their $$ 24-bit hi-res to their $$24-bit+ DACs using a 16-bit pipeline
> 
> 
> 
> ...


 

 You're saying WASAPI is a 16bit pipeline?


----------



## castleofargh

uelong said:


> I didn't realize ASIO4All was junk.  I'm using a Hugo.  If I removed A4All, would it remove asio use, or is the Chord driver enough?  Bare in mind, I'm still stuck with XP; can't afford a new machine yet.
> Thanks for any advice you'd care to give.


 
 doesn't the hugo come with it's own asio driver? I thought it did.
 asio4all as it's name implies is a tool that pretends to be asio when no proper asio driver are is provided for the DAC. but the actual system used I believe is kernel streaming(I'm not sure of that but I've read it somewhere). so if you already have actual asio drivers for the hugo, it would be meaningless to have asio4all.


----------



## castleofargh

soundsgoodtome said:


> traveller said:
> 
> 
> > Sigh... but that's my point re. WASAPI - why on Earth would anyone want to pipe their $$ 24-bit hi-res to their $$24-bit+ DACs using a 16-bit pipeline
> ...


 

 if set to 16bit ^_^


----------



## 514077

castleofargh said:


> uelong said:
> 
> 
> > I didn't realize ASIO4All was junk.  I'm using a Hugo.  If I removed A4All, would it remove asio use, or is the Chord driver enough?  Bare in mind, I'm still stuck with XP; can't afford a new machine yet.
> ...


 

 It does.  Guess I'll dump A4A.  Thanks.


----------



## PleasantSounds

> Originally Posted by *castleofargh* /img/forum/go_quote.gif
> 
> asio4all as it's name implies is a tool that pretends to be asio when no proper asio driver are is provided for the DAC. but the actual system used I believe is kernel streaming(I'm not sure of that but I've read it somewhere). so if you already have actual asio drivers for the hugo, it would be meaningless to have asio4all.


 
  
 That's correct: asio4all makes sense in the best case if your player does not support any bit perfect connections, but otherwise is just an unnecessary complexity and latency to deal with. Chord drivers support native ASIO so there's no need to use asio4all in that case.


----------



## Traveller

Next up... DSD playback 
	

	
	
		
		

		
		
	


	



  
 1. DSDIFF - "reports" sample rate (to external DAC) based on what you set its one & only parameter: "Max Sample Rate". Oddly, it defaults to 88200 (and max is 192000.)
 In the end, I have no idea what Foobar is sending my DAC... 
	

	
	
		
		

		
		
	


	



  
 2. foo-input-SACD (ver 0.9.11) - obviously designed to do more than just DSD playback but it seems to play well with DSD. Aside from selecting either ASIO or WSAPI (event / push), the only parm (AFAIK) is its "output mode" _(PCM or DSD, the latter being the obvious choice)_. I'm assuming it is a DoP based deal. It correctly "reports" the defined sample rates for both DSD128 & 256 (5.6MHz & 11.2MHz).
 Oddly enough, the only issue I've come across is that it does not feed the Visualizations (Spectrum, etc.) when playing DSDs (PCMs are fine)... . Certainly not the worse issue and I prefer it over DSDIFF's poor "reporting"... .
  
 I would be curious to know what you think (or know) to be the best DSD solution for Foobar. Important to me is that said solution does not degrade PCM playback in any way as my media in 99.5% PCM. Not that Foobar's Output-Selection feature could be any easier, but I would hope that at the end of this rainbow there is one perfect solution...


----------



## castleofargh

others will have to help here, I'm a total DSD hater so I never even tried to play any. sorry.


----------



## 514077

traveller said:


> Next up... DSD playback
> 
> 
> 
> ...


 

 I tried version 0.9.7  (latest version) and couldn't get it to work.  At someone's advice, I went to version 0.8.4 and run DSD through DOP.  White light on Hugo and all.
 I've been wondering if there's a way to convert DFF (DSDIFF) to DSF, just for tagging purposes.


----------



## Traveller

castleofargh said:


> others will have to help here, I'm a total DSD hater so I never even tried to play any. sorry.


 
 No worries 
	

	
	
		
		

		
		
	


	



 I suppose there's a thread (or three) with the pros and cons of the two formats...


----------



## Soundsgoodtome

I've asked if Foobar2k's digital volume was 32-bit float. Now I wonder if it is a 64-bit float on the volume? I've been doing some reading and been doing some digital attenuation until a preamp comes in between my dac and active speaker, wondering how much I can drop it before losing data.


----------



## 514077

pleasantsounds said:


> > Originally Posted by *castleofargh* /img/forum/go_quote.gif
> >
> > asio4all as it's name implies is a tool that pretends to be asio when no proper asio driver are is provided for the DAC. but the actual system used I believe is kernel streaming(I'm not sure of that but I've read it somewhere). so if you already have actual asio drivers for the hugo, it would be meaningless to have asio4all.
> 
> ...


 
 Which leads me to a stupid, but I feel necessary question.  What is latency, what does it affect, and what should it be set to?


----------



## PleasantSounds

uelong said:


> Which leads me to a stupid, but I feel necessary question.  What is latency, what does it affect, and what should it be set to?


 
  
 In generic terms latency is a time lag between cause and effect. In f2k it may surface as a delay when adjusting volume, EQ settings or other DSP functions it performs.
  
 You cannot set it to a specific value - it is a side effect of things that have to happen between reading the data from a file and sending it to the DAC. Indirectly you can reduce latency by keeping buffer sizes to a reasonable minimum (without causing pops and clicks) and not going overboard with your DSP chain.
  
 While it is quite critical to keep the latency to a minimum in the music production, it is much less important (but still desirable) for casual listening.


----------



## theaudiologist

does 32-bit(float)/384khz and 24/384 khz make a difference?


----------



## PleasantSounds

theaudiologist said:


> does 32-bit(float)/384khz and 24/384 khz make a difference?


 
  
 Yes it does:
 - music in these formats tends to be more expensive and not that easy to find
 - it takes 10 x the space of a redbook CD
 - it forces your PC to work harder
 - pops and clicks are more likely to happen 
 That's as much as can be objectively ascertained.
  
 The impact on sound quality is debatable at best. Personally I know that with my aging ears on my fairly high end equipment I can't tell them apart from 24/48kHz copies made off the same master. There are those who claim differences from subtle to "night and day", but unless you empirically find out that belong in that camp, I'd recommend sticking to the redbook CD which is capable of delivering stellar quality. Focus on finding the best possible mastered edition instead, as this will make much bigger difference.


----------



## 514077

pleasantsounds said:


> uelong said:
> 
> 
> > Which leads me to a stupid, but I feel necessary question.  What is latency, what does it affect, and what should it be set to?
> ...


 

 Thanks.  I think I confused latency with the adjustable buffer size.  In FB2K, it seems to usually come at 1000 (I think) ms.  Can or should it be set lower, and how much is reasonable?


----------



## PleasantSounds

uelong said:


> Thanks.  I think I confused latency with the adjustable buffer size.  In FB2K, it seems to usually come at 1000 (I think) ms.  Can or should it be set lower, and how much is reasonable?


 
  
 This depends on your setup. I have 3 installations and each requires different settings. 1000 ms is a good starting point, but my main PC requires 1200 ms to completely avoid buffer underruns, especially when heavily multitasking.
 My tablet is happy with 400 ms, but that's a much newer architecture and I don't use it for things like video rendering while playing music.
  
 From my experience the right buffer size has two boundaries: one is when pops and clicks start getting in the audio stream (the lower limit) and the other when the lag between setting controls and hearing the result is more than you can tolerate (happens with too large buffer). The  problem is when the two overlap. Sometimes updating USB drivers helps, sometimes you have to stop some services which get too much in the way (latency monitor is a great tool for that). The last resort is moving your audio setup to a platform that can handle it better.


----------



## 514077

pleasantsounds said:


> uelong said:
> 
> 
> > Thanks.  I think I confused latency with the adjustable buffer size.  In FB2K, it seems to usually come at 1000 (I think) ms.  Can or should it be set lower, and how much is reasonable?
> ...


 

 Thanks for everything.  I'll try that monitor.


----------



## ExtremeGamerBR

traveller said:


> Next up... DSD playback
> 
> 
> 
> ...


 

 foo-input-SACD is the best. Just read the readme.txt file and be happy.


----------



## ScareDe2

edited


----------



## PinkyPowers

I'm trying to get Electri-Q to work, using the George Yohng's VST Wrapper.

After hitting Rescan All a few times, it finally finds \dsp_eqfree.dll and CLAIMS to be "Testing" said plugin. But all it's really doing is locking the F up.

Is there by any chance legitimate software that can give foobar a decent EQ?


----------



## castleofargh

is your problem with electriQ while other VSTs work fine? or is it a problem with the wrapper?


----------



## PinkyPowers

Good question. I'm not using any other VSTs, so I don't know.


----------



## Michgelsen

I noticed there's an exciting new feature in version 1.3.13 beta: "Enabled WavPack 5 DSD decoding."
  
 For DSD-playing folks, this means that foobar can now use the new DSD features of WavPack, so that WavPack can be used as one format for both PCM and DSD files, with tagging support. The latest version 5 of WavPack, which is still under development, has a new lossless DSD compressor. See here: https://hydrogenaud.io/index.php/topic,112529.0.html​ .
 Previously, you could either use Sony's .dsf format, which doesn't support DST compression but can be tagged, or Philips' .dff format, which can't be tagged but can be compressed with DST. Now you can have DSD files with both compression and tags, in the WavPack format.
  
 Did anybody try it yet? I haven't, but am excited nonetheless. I hope foobar will soon support WavPack's DSD encoding part too.


----------



## ExtremeGamerBR

pinkypowers said:


> I'm trying to get Electri-Q to work, using the George Yohng's VST Wrapper.
> 
> After hitting Rescan All a few times, it finally finds \dsp_eqfree.dll and CLAIMS to be "Testing" said plugin. But all it's really doing is locking the F up.
> 
> Is there by any chance legitimate software that can give foobar a decent EQ?


 
 Did you tried this VST host?
  
 http://wiki.hydrogenaud.io/index.php?title=Foobar2000:Components/VST_adapter
  
 It works fine on my Foobar.
  
 For EQ I use EasyQ: http://rs-met.com/freebies.html


----------



## PinkyPowers

extremegamerbr said:


> Did you tried this VST host?
> 
> http://wiki.hydrogenaud.io/index.php?title=Foobar2000:Components/VST_adapter
> 
> ...




Thanks! These are working perfect together.


----------



## 514077

extremegamerbr said:


> pinkypowers said:
> 
> 
> > I'm trying to get Electri-Q to work, using the George Yohng's VST Wrapper.
> ...


 

 I have a large .dff file with a q or is cue file.  Is there a way to divide the file into its smaller individual songs without converting them to .wav?
 I have no trouble with PCM files, but don't know what to use with DSD?
 Thanks.


----------



## PinkyPowers

uelong said:


> I have a large .dff file with a q or is cue file.  Is there a way to divide the file into its smaller individual songs without converting them to .wav?
> I have no trouble with PCM files, but don't know what to use with DSD?
> Thanks.




I've never seen an album turned into a .dff file.

But anytime I wish to break up one large file into the individual tracks, I use foobar to open the .cue file. This will import the individual tracks from the larger file. Then I right-click on the album (which should start playing right away) and go down to Convert. Simply setup your conversion however you desire, and it will break the album down exactly how the .cue file lists.

You must have the codec installed in foobar for whichever format you want. You would think foobar already has MP3 and FLAC installed, since it can play those file-types, but it does not. If you want to convert to them, you must manually install those codecs.


----------



## 514077

Thanks, PP.  I don't usually like DFF, but running XP still, I can't get any software to convert to DSF.
 I'll try with FB, but I hope it will still leave me in DSD.  Guess we'll see.
 Again thanks.


----------



## Michgelsen

uelong said:


> Thanks, PP.  I don't usually like DFF, but running XP still, I can't get any software to convert to DSF.
> I'll try with FB, but I hope it will still leave me in DSD.  Guess we'll see.
> Again thanks.


 

 ​Doesn't this program work for you: http://www.signalyst.com/professional.html ? I don't know whether it runs on XP, or whether you are familiar with this tool. It can convert DFF to DSF. It is used for instance by the very useful ISO2DSF utility: http://www.computeraudiophile.com/f11-software/how-do-you-store-dff-untagged-files-21780/index8.html#post368551 .


----------



## 514077 (Jul 6, 2017)

Oh yes, I have iso2dsd which works, and dff2dsf which unfortunately, doesn't work on my computer.  I assume it's because I'm stuck on XP, for now.
 I bet I'll get results on W7, when I upgrade.  PS:  I did.  DFF2DSF works now.
 Thanks.


----------



## isaccasi

Yes there is a program for newbies such as my self,if you are interested in ripping your cd collection to FLAC.
 The answer is dbpoweramp,it is very easy to use,but pricy.
 If you look at the Foobar2000 componets website you will see it there.


----------



## Michgelsen

But the question is: is there a program which allows editing (such as splitting) of DSD files for free? I don't know of any, and would be interested to hear about it.


----------



## PleasantSounds

michgelsen said:


> But the question is: is there a program which allows editing (such as splitting) of DSD files for free? I don't know of any, and would be interested to hear about it.


 
  
This program will split or merge dsf and dff files and it's free.


----------



## Joeybgood

I just purchased my first DSD capable DAC. I use Foobar2K. I am as NON tech savvy of a guy  as can possibly be. REALLY can use some help here. I dl'd this http://www.foobar2000.org/components/view/foo_input_dsdiff  but the couple of DSD files I picked up(a couple 128 and 256 files) will still not play. I get the 'white box/black lettering" popup stating this file is not supported etc. Can someone please tell me what else needs to be done so DSD playback will be allowed? Tks much Joe


----------



## Cotnijoe

Has anyone been experiencing some distortion when using Foobar? Windows 10 updated itself a few weeks ago and I've been noticing this issue ever since. It seems to happen when there are multiple programs running and simply restarting Foobar fixes the problem.
  
 I've tried reinstalling Foobar without any results.


----------



## ExtremeGamerBR

joeybgood said:


> I just purchased my first DSD capable DAC. I use Foobar2K. I am as NON tech savvy of a guy  as can possibly be. REALLY can use some help here. I dl'd this http://www.foobar2000.org/components/view/foo_input_dsdiff  but the couple of DSD files I picked up(a couple 128 and 256 files) will still not play. I get the 'white box/black lettering" popup stating this file is not supported etc. Can someone please tell me what else needs to be done so DSD playback will be allowed? Tks much Joe


 
 You could start from here: http://www.head-fi.org/t/624628/the-foobar2000-help-thread-got-problems-or-questions-ask-here/300#post_12811523
  
 And here: https://sourceforge.net/projects/sacddecoder/files/foo_input_sacd/
  
 It will have a readme file with instructions.


----------



## Joeybgood

extremegamerbr said:


> You could start from here: http://www.head-fi.org/t/624628/the-foobar2000-help-thread-got-problems-or-questions-ask-here/300#post_12811523
> 
> And here: https://sourceforge.net/projects/sacddecoder/files/foo_input_sacd/
> 
> It will have a readme file with instructions.


 
 awesome.. Tks so much!


----------



## Joeybgood

joeybgood said:


> awesome.. Tks so much!


 
 I dl'd those files. followed the Read Me instructions(at least I think I it correctly) but when I attempt to play a DSD file I get this pop up error msg "Unrecoverable playback error: Unsupported stream format: 176400 Hz / 24-bit / 6 channels" This makes no sense to me. I am attempting to play a DSD64 file. Anyone have a thought on why I'm getting this error msg?


----------



## 514077

Do they have a version for XP dynosaur users?  I have a couple of large DSD files with cue files.


----------



## 514077

joeybgood said:


> joeybgood said:
> 
> 
> > awesome.. Tks so much!
> ...


 

 I'm not great at this, but got a lot of help.  1)  Is your DAC native DSD or DOP, DSD Over PCM?
 2)  What version of the components are you using?  That might be a good place to start.


----------



## Joeybgood

uelong said:


> I'm not great at this, but got a lot of help.  1)  Is your DAC native DSD or DOP, DSD Over PCM?
> 2)  What version of the components are you using?  That might be a good place to start.


 
 Tks for your help. My DAC does Native DSD and the components are SACD Decoder 1.0.2 and DSD Processor 1.0.0


----------



## fiske

Also read this site explaining installing the SACD plugin.
 https://diyaudioheaven.wordpress.com/digital/pc-software/foobar-2000-for-dummies-part-3-new-experimental-sacd-plugin-v-0-9-x/


----------



## Joeybgood

fiske said:


> Also read this site explaining installing the SACD plugin.
> https://diyaudioheaven.wordpress.com/digital/pc-software/foobar-2000-for-dummies-part-3-new-experimental-sacd-plugin-v-0-9-x/


 
 Tks.. yes actually that is the link I was using. The oddest thing....I'm getting it to work currently. well.. sort of,,, As it changes tracks, I get a bit of distortionIcrackle/light popping) which lasts for just a second. It also does that when I click ahead on a track. When it changes tracks I get a popup each time "No installed FIR, continue with the default"  Anyone enlighten me to what a FIR is ? spoke too soon. This popped up and ceased playback again.  "Unrecoverable playback error: Unsupported stream format: 176400 Hz / 24-bit / 6 channels" frustrating...


----------



## Joeybgood

joeybgood said:


> Tks.. yes actually that is the link I was using. The oddest thing....I'm getting it to work currently. well.. sort of,,, As it changes tracks, I get a bit of distortionIcrackle/light popping) which lasts for just a second. It also does that when I click ahead on a track. When it changes tracks I get a popup each time "No installed FIR, continue with the default"  Anyone enlighten me to what a FIR is ? spoke too soon. This popped up and ceased playback again.  "Unrecoverable playback error: Unsupported stream format: 176400 Hz / 24-bit / 6 channels" frustrating...


 
 Since I haven't figured it out using Foobar2K for Dummies, does anyone know if there is a Foobar2K for total Idiots??!


----------



## Michgelsen

joeybgood said:


> Tks.. yes actually that is the link I was using. The oddest thing....I'm getting it to work currently. well.. sort of,,, As it changes tracks, I get a bit of distortionIcrackle/light popping) which lasts for just a second. It also does that when I click ahead on a track. When it changes tracks I get a popup each time "No installed FIR, continue with the default"  Anyone enlighten me to what a FIR is ? spoke too soon. This popped up and ceased playback again.  "Unrecoverable playback error: Unsupported stream format: 176400 Hz / 24-bit / 6 channels" frustrating...


 
  
 Under preferences > tools > SACD, did you:
 1. set the output mode to DSD, or is it still in PCM?
 2. select 'stereo' as the preferable area?
  
 If you have it set to PCM mode, the SACD plugin will convert DSD to PCM, for example with a 176.4kHz sample rate, and some SACDs have a multichannel area, which you probably don't want to play. Hence maybe the 6-channel error. If you have set it to DSD mode, there should be no mention of a PCM sample rate such as 176.4kHz, nor of a filter (FIR). These options are all greyed out once you select DSD mode.
  
 If you have a DSD-capable DAC, you should set the plugin tot DSD mode, or else it will convert DSD to PCM.
 The DSD signal will be sent to the DAC coded as a PCM stream, which is called DoP (DSD over PCM), and which is not a conversion to PCM, but simply a means of transportation. Your DAC will detect that the PCM stream actually doesn't contain a PCM signal, but a DSD signal instead, and will automatically decode that.


----------



## 514077

michgelsen said:


> joeybgood said:
> 
> 
> > Tks.. yes actually that is the link I was using. The oddest thing....I'm getting it to work currently. well.. sort of,,, As it changes tracks, I get a bit of distortionIcrackle/light popping) which lasts for just a second. It also does that when I click ahead on a track. When it changes tracks I get a popup each time "No installed FIR, continue with the default"  Anyone enlighten me to what a FIR is ? spoke too soon. This popped up and ceased playback again.  "Unrecoverable playback error: Unsupported stream format: 176400 Hz / 24-bit / 6 channels" frustrating...
> ...


 

 After the preferable area combo box, there are two more checkboxes: one is just 'edit' and the next one is 'master playback' in tools/sacd.  Are they important?  And if so, what are they used for?
 Thanks.


----------



## Michgelsen

I'm not sure. I'm still using version 0.9.11 and see that there have been a few new verions in the last two weeks, so it could be that we're not seeing the same things.
 In my config panel there's 'editable tags' (checked), 'store tags with iso' (checked), 'linked 2ch/mch tags' (checked), and 'editable master playback' (checked, and this is for a special kind of DSD files I believe, but I'm not sure).


----------



## VNandor

Hi guys,
  
 I had to reinstall my foobar2000 and its components however the dynamic range meter I used before is not working anymore.
 I used the one that can be downloaded here: http://dynamicrange.de/de/free-downloads
  
 The problem is that it has an expiration date set to 2011. I'm quite sure it worked before the reinstall without changing my computer's date settings.
 So, any ideas for a workaround or maybe a DL link for a newer version please?


----------



## GumbyDammit223

vnandor said:


> Hi guys,
> 
> 
> 
> ...



 


Try this: https://www.dropbox.com/s/wki2m3yotlvmoim/foo_dynamic_range_1.1.1.zip


----------



## canthearyou

Foobar refuses to RIP entire Beatles 1 album. Tried 2 different PCs and also a plug in CD drive. Fails every time at second from last track. CD is brand new. What else can be done?
Hell! It won't even play "Let It Be" directly from the CD without skipping and locking-up. Is it possible to get a bad cd?


----------



## PinkyPowers

canthearyou said:


> Foobar refuses to RIP entire Beatles 1 album. Tried 2 different PCs and also a plug in CD drive. Fails every time at second from last track. CD is brand new. What else can be done?
> Hell! It won't even play "Let It Be" directly from the CD without skipping and locking-up. Is it possible to get a bad cd?




Download Exact Audio Copy (EAC). If it's possible to rip, EAC will do it.


----------



## canthearyou

pinkypowers said:


> Download Exact Audio Copy (EAC). If it's possible to rip, EAC will do it.




Cool. Thanks! Is it a stand alone software or a Foobar add-on.

Edit: I see it now. Will DL and give it a try.


----------



## castleofargh

canthearyou said:


> Foobar refuses to RIP entire Beatles 1 album. Tried 2 different PCs and also a plug in CD drive. Fails every time at second from last track. CD is brand new. What else can be done?
> Hell! It won't even play "Let It Be" directly from the CD without skipping and locking-up. Is it possible to get a bad cd?


 

 it's probably the music industry thanking you for buying the album by making your life harder with copy protections. "we can't stop pirates, so let's annoy honest people! muhahaha".
 +1 for EAC. it's a powerful tool.


----------



## Michgelsen

There's also the possibility of using a CD player with digital output, hooked up to a digital input on your computer if you have one, and then record it digitally with Audacity. I ripped a few copy controlled CDs that way once. The CD player somehow interpolated the missing frames, while the ripping software either choked on them or ripped the tracks with missing frames in some places.


----------



## canthearyou

I don't think it's a copy protection system much as a faulty disc. It's been trying to rip for the last 3 hours using EAC. The disc drive sounds like a dot matrix printer.

Estimated time remaining 14:36:35

It finally gave up and skipped last track. It ripped 26 of 27 tracks successfully.


----------



## AtrafCreez

Hi,
 I had downloaded foobar200 yesterday and saw what was what.
 My source for music is only YouTube.  I have many music lists.
 Just now I went into google search to see if foobar and YouTube are compatible, I see there is a download.
 being non technical  could you tell me how to set it up?
 Thank You.


----------



## PleasantSounds

Download the YT component and double-click it. This should activate foobar and the component will be added to the components list. After that just press OK on the installed components window. Foobar will restart.

 The next part is trickier, because you have to place the installed components somewhere on the Foobar window. The first step is enabling the layout editing (View -> Layout -> Enable layout editing mode).
  
 I have no idea how your interface is organised, so I'm not able to be very specific. In my installation I have created a new tab in a window that displays album list, selection properties and a few other things (right-click on any of the tab headers and select "Add new tab"). Creating a new tab gives you a blank surface to work with. Next I added in it a horizontal splitter (right-click -> Add new UI element -> Splitter(top/bottom). Then in the upper part I have added the Youtube search component (it should be accessible in the same UI element list), and in the lower part the YT video. 
  
 If you want to add several components in one window, you have to divide it's surface before you do that. Once you select UI element, it will take the entire available surface and to add something else first you will have to remove it, split the window and add new elements to the parts of the window. That's why it helps to know upfront how you want the GUI arranged.
  
 When done don't forget to turn off the layout editing mode (the same way as turning it on) and restart foobar. After that you should be able to do searches in the newly added component and the results will display in the playlist window.
  
 I don't use youtube playlists so don't ask me how to get that into Foobar.


----------



## 514077

canthearyou said:


> I don't think it's a copy protection system much as a faulty disc. It's been trying to rip for the last 3 hours using EAC. The disc drive sounds like a dot matrix printer.
> 
> Estimated time remaining 14:36:35
> 
> It finally gave up and skipped last track. It ripped 26 of 27 tracks successfully.


 

 It probably is faulty.  I had a cd of Discovery by mike Oldfield, bought new, and the 1st two tracks were unplayable and unripable using DBPowerAmp.  It happens, I'm afraid.


----------



## bwanaaa

Hardware:
 Win10 PC-> USB Gustard X20->schitt headphone amp-> Senns
  
 Software:
 Just updated foobar to 1.3.12, SACD decoder to  1.0.0, and still have ASIO support 2.1.2
  
 Preferences in foobar:
 Playback->output:                        ASIO: XMOS USB Audio 2.0 ST 3033
 Playback->output-> ASIO:            use 64 bit ASIO drivers checked
 Tools->SACD:                             DSD+PCM
  
 DSD64 files play and the dac says it's a PCM stream.
  
  
 Change preferences to:
 Playback->output:                        ASIO: XMOS USB Audio 2.0 ST 3033
 Playback->output-> ASIO:            use 64 bit ASIO drivers checked
 Tools->SACD:                             DSD
  
 Trying to play some DSD files -> no sound
 playhead moves in foobar but no sound and no dancing bars in the meter
  
 Change preferences to:
 Playback->output:                        DSD: ASIO: XMOS USB Audio 2.0 ST 3033
 Playback->output-> ASIO:            use 64 bit ASIO drivers checked
 Tools->SACD:                             DSD
  
 Trying to play some DSD files -> no sound
 playhead moves in foobar but no sound and no dancing bars in the meter
  
 Change preferences to:
 Playback->output:                        DSD: ASIO: XMOS USB Audio 2.0 ST 3033
 Playback->output-> ASIO:            use 64 bit ASIO drivers checked
 Tools->SACD:                             DSD+PCM
  
 Trying to play some DSD files -> no sound
 playhead moves in foobar but no sound and DO GET dancing bars in the meter
  
 I know this dac will decode DSD but am having this issue getting the foobar and pc to output it as DSD.


----------



## canthearyou

Change your device to "foo_dsd_asio".


----------



## Michgelsen

Or try it with the WASAPI output plugin instead of ASIO: http://www.foobar2000.org/components/view/foo_out_wasapi


----------



## shortstuff

Where can I download "Real Bass Exciter" for foobar2k? I only see a way out of date "Bass Exciter" when googling...


----------



## 514077

I have a minor problem with volume to my Hugo.  Sometimes, I hear some distortion in tracks that are, admittedly, overcooked in treble.  Is there a master volume in Foobar that I can try to reduce?  I can't use wasapi as I'm still on XP.


----------



## Joe Bloggs

uelong said:


> I have a minor problem with volume to my Hugo.  Sometimes, I hear some distortion in tracks that are, admittedly, overcooked in treble.  Is there a master volume in Foobar that I can try to reduce?  I can't use wasapi as I'm still on XP.




Uh, yea, the big volume slider?


----------



## seoman

Is there any way to get a darker skin on foobar.
 Nothing fancy just something like a night-mode to ease the eyes.


----------



## 514077

joe bloggs said:


> uelong said:
> 
> 
> > I have a minor problem with volume to my Hugo.  Sometimes, I hear some distortion in tracks that are, admittedly, overcooked in treble.  Is there a master volume in Foobar that I can try to reduce?  I can't use wasapi as I'm still on XP.
> ...


 

 With all due respect, and not meaning to whine:  I'm a blind dude.  What slider.  If my JAWS (Job Access With Speech) program can't recognize the slider on the screen, I'm not going to see it either.  Thanks for the help, but if your answer was supposed to imply "Duh,"  it was misplaced, sir.
 Is there a way I can access the 'slider'  through the menu choices?


----------



## bwanaaa

thank you. turns out my usb card was acting up so foobar could not talk to it properly. reinstalled drivers and now ok. i think i should have stuck with motherboard usb and bought a singxer instead. Still, this usb card(paul pang) is dead silent.


----------



## PleasantSounds

uelong said:


> joe bloggs said:
> 
> 
> > uelong said:
> ...


 
  
 Whether you see the volume slider on screen or not will dpend on the actual GUI configuration you are using. For the standard GUI the way to bring the volume slider on screen is to right click on the main menu bar and put a tick next to Volume Control.
  
 If you want to lower the volume permanently, then probably a better way is to do it through Preferences->Playback. On that screen you have the Preamp section where volume can be adjusted. If you're using the ReplayGain, then use the upper slider, otherwise the lower one. Or just change both.


----------



## 514077

pleasantsounds said:


> uelong said:
> 
> 
> > joe bloggs said:
> ...


 

 Thanks @pleasant sounds.  I think that's what I was looking for.


----------



## Joe Bloggs

uelong said:


> joe bloggs said:
> 
> 
> > uelong said:
> ...




Sorry, I do remember your disability but did not think the volume slider would be inaccessible (or any more inaccessible than the other elements) because of this. Rather I thought you might be pegging the volume slider on purpose because of a desire for bit-perfect output (a typical audiophile obsession).

You can assign a keyboard shortcut to volume up and down if that helps. Go to File->Preferences->Keyboard Shortcuts and select Add New. The Action for volume is called Playback / Volume / Up and Playback / Volume / Down.


----------



## 514077

Excellent!  Thanks Joe.  I'll give that a go.  It's true that some controls i.e. the position slider at the top right is not useable by a totally blind guy.  I only know it's there because one time, I could see it.
 Just the way it is.
 Thanks again.


----------



## Joe Bloggs

uelong said:


> Excellent!  Thanks Joe.  I'll give that a go.  It's true that some controls i.e. the position slider at the top right is not useable by a totally blind guy.  I only know it's there because one time, I could see it.
> Just the way it is.
> Thanks again.




Hey--you can assign buttons to seek forward and backward within a track too from the same menu. It's called Playback / Seek / Back by ? seconds and Playback / Seek / Ahead by ? seconds


----------



## Book

I have just got a fresh install of windows 10 and had to reinstall foobar and equilibrium. I am getting the following error when installing the vst: 
 Could not load VST DLL. You may have tried to use x64 DLL or there is an internal error in the plug-in.
 D:\Programs\foobar2000\components\EQuilibrium.dll
 I have no clue how to fix this as the only thing i found so far is "Put all files from the convolver into your foobar folder and then try again installing the vst. Sometimes this helps." to help someone else with a similar problem, and i think i did that with equilibrium (Installed it straight to the foobar folder) but i still keep getting an error when i try to add the VST.


----------



## castleofargh

book said:


> I have just got a fresh install of windows 10 and had to reinstall foobar and equilibrium. I am getting the following error when installing the vst:
> Could not load VST DLL. You may have tried to use x64 DLL or there is an internal error in the plug-in.
> D:\Programs\foobar2000\components\EQuilibrium.dll
> I have no clue how to fix this as the only thing i found so far is "Put all files from the convolver into your foobar folder and then try again installing the vst. Sometimes this helps." to help someone else with a similar problem, and i think i did that with equilibrium (Installed it straight to the foobar folder) but i still keep getting an error when i try to add the VST.


 
 first, the obvious, you added a VST wrapper?
 past that, foobar won't work with the 64bit version of the vst. same thing for VST3 version, I don't know if there is a vst wrapper for foobar that is vst3 compatible.


----------



## Book

Yeah this is the wrapper that i got: https://hydrogenaud.io/index.php?PHPSESSID=3bfan5v3cicpjgchpkf56dovd6&topic=84947.0 , and it cant be incomparable because before i installed a new version of windows it was working fine. That's why i am complacently lost for what to do.


----------



## Book

Yeah i got Foobar2000 VST 2.4 adapter, and equilibrium should not be incomparable with foobar as i have run it before i got a fresh copy of windows with this problem cropping up but i managed to resolve it (though i dont know what i did). I have tried resolving it myself for a few hours and failed (guess i am not as lucky as before), and right now i have no clue what to do


----------



## castleofargh

sure equilibrium works in foobar, I use it.
 did you get another VST to work? to be sure it's equilibrium and not the wrapper installation that's a problem?


----------



## Book

It happened when i tried to install s(M)exoscope. But as far as i know all i need to do with the wrapper is placing it in the components folder.


----------



## Book

It happened when i tried to install another VST, so does that mean i installed the wrapper wrong, as far as i know all you need to do is place it in components folder right? I tried installing smexoscope. To install VST both times i followed this guide:
  
 "Download the file "foo_vst_0903.zip" and place the unpacked file in the Foobar2000 components directory. Restart Foobar and open up the Foobar Preferences window. At the top of the list on the left under Components you should now have something called "VST plug-ins". Done for now.
  
 Both EQ-ing plug-ins will install roughly the same, during installation they will place themselves in a directory on your computer. I believe the default is "C:\Program Files (x86)\VstPlugIns". Once that is done go back to Foobar2000 and look under [Preferences --> Components --> VST plug-ins] which we set up before. Click "Add" at the bottom and navigate to where the vst plug-ins were installed. Select the plug-in you want and restart Foobar.

 Once Foobar is restarted you should be able to select either Electri-Q or EQuilibrium in [Preferences --> Playback --> DSP Manager]. Move the plug-in across to the left and click apply to activate. Once you've activated either plug-in close the Preferences window and go to [view --> DSP]. Start the EQ program of your choice."
  
 But stuck on part 2


----------



## Joe Bloggs

The VST plugin should install with 32 and 64 bit editions. Run the equilibrium installer again and note where it asks you to install the 32 bit version dll, whether you agreed to install it and which directory you set it to install to. Then point foobar to that 32 bit version of the plugin dll.


----------



## GumbyDammit223

Greetings All:
  
 Not really knowing what's what, I've installed a bunch of plugins and now I have a plethora of output options that have the following abbreviations:
 DS: _audio driver_
 DSD : WASAPI (push) _audio driver_
 DSD : WASAPI (event) _audio driver_
 DSD : KS _audio driver_
 KS : _audio driver_
 WASAPI (push) _audio driver_
 WASAPI (event) _audio driver_
  
 where _audio driver_ can be "speakers (Sound Blaster Audigy 5/Rx)" or "Bifrost (Schiit USB Audio Gen 2)" for example.
  
 99.95% of the time my FLACs go from Foobar2k via USB to my Schiit Bifrost/Valhalla 2 combo to my HD700's.  The other 0.05% of the time it's split between my computer speakers or my PXC 550's via Bluetooth.  The Direct Sound selections seem to work, but I'm wondering if any of the other options will give better results/performance.
  
 Can anyone point this idiot in the right direction? 
	

	
	
		
		

		
			




 Thanks!


----------



## castleofargh

gumbydammit223 said:


> Greetings All:
> 
> Not really knowing what's what, I've installed a bunch of plugins and now I have a plethora of output options that have the following abbreviations:
> DS: _audio driver_
> ...


 

 wasapi push or event depends on the DAC but if both work, chances are they will give the same result. and if one creates problems or doesn't work at all, then you've got your answer ^_^ .
 KS is kernel streaming, another "bit perfect" solution like wasapi. again if it works without trouble, then it probably also gives the same kind of quality, usually people pick one at random or because the rest doesn't work on their specific system.
 direct sound while still usually audibly fine, will often measure a little below bit perfect solutions in exclusive mode. but depending on your usage, having a mixer can really simplify things(fidelity vs convenience).
 now to know for sure, I would very very strongly advise against believing posts like mine about what is the best ^_^. to check measurements, just get yourself a cable to go from your DAC to your soundcard's input (if it's not too crappy), and use the free RMAA to check the different solutions. it will take you an evening to test plenty of settings and know for good if something goes wrong. I've done that myself not too long ago when I was testing the recording side of my loop with various settings and usb configurations. but the idea is the same as long as the input isn't too bad to measure anything but itself.
  
 or if you go all subjective about this, just try and pick whatever feels right to you for whatever reason you have. your system, your sound.


----------



## GumbyDammit223

castleofargh said:


> wasapi push or event depends on the DAC but if both work, chances are they will give the same result. and if one creates problems or doesn't work at all, then you've got your answer ^_^ .
> KS is kernel streaming, another "bit perfect" solution like wasapi. again if it works without trouble, then it probably also gives the same kind of quality, usually people pick one at random or because the rest doesn't work on their specific system.
> direct sound while still usually audibly fine, will often measure a little below bit perfect solutions in exclusive mode. but depending on your usage, having a mixer can really simplify things(fidelity vs convenience).
> now to know for sure, I would very very strongly advise against believing posts like mine about what is the best ^_^. to check measurements, just get yourself a cable to go from your DAC to your soundcard's input (if it's not too crappy), and use the free RMAA to check the different solutions. it will take you an evening to test plenty of settings and know for good if something goes wrong. I've done that myself not too long ago when I was testing the recording side of my loop with various settings and usb configurations. but the idea is the same as long as the input isn't too bad to measure anything but itself.
> ...


 
  
 Thanks for the advice - good thing it's the start of the weekend! 
	

	
	
		
		

		
			




  
 I did see that quite a few of the options crashed when trying to use the Bluetooth path which helps narrow things down, but the USB path seems to be pretty robust.  Glad I'm into a hobby that's so cut and dry for options!


----------



## Michgelsen

If you're on Windows 10, I suggest using DSD: WASAPI (event). Push mode is also fine. Kernel streaming is something used on older Windows versions, while WASAPI is the latest, supported way of making a direct audio path from software to device, without passing through some form of mixing or sample rate conversion software layer. This makes sure that your DAC receives a bit perfect stream. With normal PCM audio, such as flac files, a bit perfect stream is something to aim for, but it is not essential for playback. On the other hand, when playing DSD, it IS essential, because the DSD stream is packed into a PCM stream. This is done so that it wasn't necessary to invent a new way for DSD delivery. Instead the normal PCM delivery path can be used. Only your DAC sees that the PCM stream in fact contains a DSD stream, and will decode DSD. However, when the PCM stream (or more precisely, DoP = DSDoverPCM stream) gets altered in any way by mixing or sample rate conversion, this no longer works, and you can't play back DSD.


----------



## GumbyDammit223

michgelsen said:


> If you're on Windows 10, I suggest using DSD: WASAPI (event). Push mode is also fine. Kernel streaming is something used on older Windows versions, while WASAPI is the latest, supported way of making a direct audio path from software to device, without passing through some form of mixing or sample rate conversion software layer. This makes sure that your DAC receives a bit perfect stream. With normal PCM audio, such as flac files, a bit perfect stream is something to aim for, but it is not essential for playback. On the other hand, when playing DSD, it IS essential, because the DSD stream is packed into a PCM stream. This is done so that it wasn't necessary to invent a new way for DSD delivery. Instead the normal PCM delivery path can be used. Only your DAC sees that the PCM stream in fact contains a DSD stream, and will decode DSD. However, when the PCM stream (or more precisely, DoP = DSDoverPCM stream) gets altered in any way by mixing or sample rate conversion, this no longer works, and you can't play back DSD.


 

 Thanks for the advice.  I'm running Win7 Pro and have settled on DSD:WASPI because it seemed like the cleanest path by description.  It was hard for me to tell much difference acoustically.  I tried running RMAA over the weekend and it worked with one setting (MME:x - I'm at work so can't look up exactly what it said) and it wouldn't work with others that I thought were more direct-stream-ish.  The MME (not finding what MME stood for) selections seemed to be more like the DirectSound settings - more generic and maybe reduced quality but more inclined to work.
  
 That's good to know about playing DSD material.  I've looked into getting a couple albums to try but hadn't taken the plunge yet because I had a feeling I would run into problems.  It's still worth trying though so maybe I'll play around some more in the upcoming weeks.  Thanks!


----------



## Joe Bloggs

michgelsen said:


> On the other hand, when playing DSD, it IS essential, because the DSD stream is packed into a PCM stream. This is done so that it wasn't necessary to invent a new way for DSD delivery. Instead the normal PCM delivery path can be used.




This is only true for DSD over PCM (DoP). Not for "native" DSD stream (but nomenclature aside, a correctly decoded DoP stream is not any less "native")


----------



## Michgelsen

joe bloggs said:


> This is only true for DSD over PCM (DoP). Not for "native" DSD stream (but nomenclature aside, a correctly decoded DoP stream is not any less "native")


 

 ​Native DSD only works over ASIO 2.2, and if your DAC supports it, right? Does foobar support this?


----------



## simon740

Hello,

 I'm pretty new to this field. Therefore, I am interested in having an app on the iPad which can control foobar2000? What needs to be set in foobar2000? Please help because I am in the dark

 greetings,
 Simon


----------



## GumbyDammit223

Just downloaded my first DSD files - Strauss' _Also Sprach Zarathustra_ from HDtracks.  Awesome detail!
  
 Silly question - I know the .dsf files have embedded album artwork because mp3tag shows it.  However foobar doesn't.  Foobar shows all the artwork in all my other FLACs and MP3s so is there a switch somewhere in the configuration I didn't throw?
  
 Thanks in advance,
 -Scott


----------



## RuFrost

Hello there guys,
 I'm very newbe to foobar.
 I own Fiio X7 as an external DAC.
 Can you please guide me (very preferably in algorithm - step by step from 0 to the last one) how to make it play DSD?
 I do not know which version I need to download, nothing about plugins, internal settings and so on.
 Hope for your help!


----------



## PsyMed

How do you change the foobar playlists to read them as D drive and E drive? Current playlists are on C drive and D drive ( new comp with a smaller OS drive / SSD ) How do you edit the playlist file?


----------



## Michgelsen

rufrost said:


> Hello there guys,
> I'm very newbe to foobar.
> I own Fiio X7 as an external DAC.
> Can you please guide me (very preferably in algorithm - step by step from 0 to the last one) how to make it play DSD?
> ...


 
  
  
 1. Download and install foobar: http://www.foobar2000.org/download
 2. Download and install the WASAPI output component: http://www.foobar2000.org/components/view/foo_out_wasapi
 3. Download and install the SACD input component: https://sourceforge.net/projects/sacddecoder/files/foo_input_sacd/ This component can play SACD rips in .iso format, or separate DSD files in DSF or DFF format. It can either output in PCM mode or DSD mode. See next step.
 4. Follow the instructions in the readme file of the SACD input component: https://heanet.dl.sourceforge.net/project/sacddecoder/foo_input_sacd/readme.txt As you can read in point 4 of the readme, you need to set the output mode of foobar to DSD : WASAPI (event mode or push mode, doesn't really matter as far as I know) : YOURDAC. This is where the WASAPI output plugin is used for which you downloaded earlier. Obviously, also make sure to select DSD mode instead of PCM mode in the config panel of the SACD output component. This is also in the readme.
 5. Make sure to enable the WASAPI exclusive mode in Windows. To do this, go to the Windows sound configuration window, go to Playback tab, select your DAC, click Properties, go to Advanced tab, and check the two boxes for exclusive mode. This will make sure that Windows will not mess with the audio stream going from foobar to the DAC, and perhaps making it impossible to decode. You need a bit perfect stream for DoP (= DSD over PCM) to work. Using this exclusive mode will prevent other programs from playing music while foobar is playing (or paused!), so don't expect YouTube to work at the same time anymore. You must stop the music in foobar in order to get any sound from other programs.
 6. You might need to enable a certain setting, for example for USB2 mode, on your DAC as well. See its manual to find out whether this is required. I can't help you with this.


----------



## PleasantSounds

psymed said:


> How do you change the foobar playlists to read them as D drive and E drive? Current playlists are on C drive and D drive ( new comp with a smaller OS drive / SSD ) How do you edit the playlist file?


 
  
 Go to Preferences -> Media Library and edit the Music Folders list.


----------



## PsyMed

pleasantsounds said:


> Go to Preferences -> Media Library and edit the Music Folders list.


 

 is there a way to just open those flv files and control+H D:\ to -> F:\ ?


----------



## PleasantSounds

psymed said:


> pleasantsounds said:
> 
> 
> > Go to Preferences -> Media Library and edit the Music Folders list.
> ...


 
  
 oh sorry - I misunderstood your question. 
  
 Unfortunately playlists are binary files, not text and editing them is much harder. You would need a binary file (or hex) editor. They are actually .fpl files, not .flv and are located in the %username%\AppData\Roaming\foobar2000\playlists-v1.3 folder (note that the AppData folder is by default hidden). Google hex editor and you'll get plenty of choices, many of them free.
  
 I haven't had a need to use a hex editor for years so can't tell you if you should expect any sort of search-and-replace functionality there, but it's not out of question. But always replace strings of the same length only.
  
 Just make sure you have backup copies of these files as it's very easy to corrupt them.
  
 EDIT: there's also a much safer way: you can export the playlist content as .m3u file (Utilities -> Save as Playlist) which you can edit with a notepad and then import to an empty playlist.


----------



## NemoReborn

found it !


----------



## 514077

Deleted


----------



## eschell27

Any of you tried playing around with any of the remote control components? I have a comfy chair that i like to lounge in and listen to my headphones when i'm not in front of my PC listening and had been getting annoyed by having to get up to change tracks etc so i thought HMMMmmmmm there must be some kind of android/ios remote control with an addin component for the pc version of foobar2000....sure enough there it is. Has more functionality that i actually wanted and seems to work fairly well. If any of you find yourself in a similar predicament give this a try. I'm using it on my android phone.
  
 Here are the links to the playstore android app and their site explaining the rather easy setup with a link to the pc side software download.
  
https://play.google.com/store/apps/details?id=com.cav.foobar2000controller&hl=en
  
http://foobar2000controller.blogspot.com.es/p/how-to-start.html
  
 If anyone gives it a shot, i'd be interested to see what you think! Only been using it for a little bit tonight but haven't really encountered any bugs....yet.


----------



## canthearyou

eschell27 said:


> Any of you tried playing around with any of the remote control components? I have a comfy chair that i like to lounge in and listen to my headphones when i'm not in front of my PC listening and had been getting annoyed by having to get up to change tracks etc so i thought HMMMmmmmm there must be some kind of android/ios remote control with an addin component for the pc version of foobar2000....sure enough there it is. Has more functionality that i actually wanted and seems to work fairly well. If any of you find yourself in a similar predicament give this a try. I'm using it on my android phone.
> 
> Here are the links to the playstore android app and their site explaining the rather easy setup with a link to the pc side software download.
> 
> ...




Foobarcon works for me.


----------



## 514077

I have a question about converting izo to dsf.  I've used izo2dsd up to now, but being freeware, it doesn't work with my screenreader.  A screenreader is a program which lets blind people use their computers with speech and without the need to use a mouse.  I2D isn't desined to use that way.  So my quetion is simply; is there a plug-in to use foobar2000 to convert iso to dsd?  Sorry for the typing mistakes, but whenever I backspace to correct, I'm thrown out of the edit field in this new version of Head-fi.
Thanks in advance.


----------



## Claude (Jun 5, 2017)

Concerning remote control:

I use foobar2000controller, with 2 servers alternatively (desktop PC in living room, laptop in bedroom), and it works fine for me, the way I use it (I don't use the library function a lot, but copy albums manually into a playlist on the PC and then use the Android app to control playback from my listening position).

@UELong: SACD ISO support in foobar2000 depends on the SACD plugin, which does not offer ISO to DSF conversion. You can only convert tracks from the ISO to PCM (to built-in formats).


----------



## 514077

Claude said:


> Concerning remote control:
> 
> I use foobar2000controller, with 2 servers alternatively (desktop PC in living room, laptop in bedroom), and it works fine for me, the way I use it (I don't use the library function a lot, but copy albums manually into a playlist on the PC and then use the Android app to control playback from my listening position).
> 
> @UELong: SACD ISO support in foobar2000 depends on the SACD plugin, which does not offer ISO to DSF conversion. You can only convert tracks from the ISO to PCM (to built-in formats).


Thanks.  I was afraid of that.


----------



## whirlwind

Are you guys saying that foobar does not do DSD


----------



## 514077

whirlwind said:


> Are you guys saying that foobar does not do DSD


No.  It does DSD with the propper add-on.  I was looking for a way to use FB to convert iso images to DSF for tagging and using in my player which doesn't play ISOs.  I'm also a blind dude, and the usual program, ISO2DSD, doesn't read with my blind-guy software.  Foobar is more usable in that way.  That was the reason for my inquiry.  HTH


----------



## canthearyou (Jun 5, 2017)

I can't for the life of me get Foobar to convert PCM to DSD and play. I've tried every single guide that I could find anywhere. I've uninstalled and reinstalled both an old and newer version. Hell, I can't even get it to play an actual DSD file!

Not sure what I did but it's working!


----------



## 514077

canthearyou said:


> I can't for the life of me get Foobar to convert PCM to DSD and play. I've tried every single guide that I could find anywhere. I've uninstalled and reinstalled both an old and newer version. Hell, I can't even get it to play an actual DSD file!
> 
> Not sure what I did but it's working!


Did you or are you now using Foo_input_sacd 1.0.5?  I found that worked for me.  But I've not tried to sample from PCM2DSD.


----------



## canthearyou

UELong said:


> Did you or are you now using Foo_input_sacd 1.0.5?  I found that worked for me.  But I've not tried to sample from PCM2DSD.


I'm using 0.9.7


----------



## 514077

canthearyou said:


> I'm using 0.9.7


Funny; 0.9.7 never worked for me.  I used 0.8.4 for the longest time.  Then, to keep up with the times, I went to Windows 7 in January.  I had to change up to 1.0.5, which worked like a charm.  But as long as it works now, that's cool.


----------



## Claude

With the DSD processor which is part of the recent versions of the SACD plugin I can set conversion of PCM signals up to DSD128. The higher settings don't work. The DAC confirms that it receives DSD128.


----------



## Makiah S

I don't know why, but using Kernal Streaming with my Audio GD NFb10ES2 some times when I go from Pause to play I get a LOT of static. I'm now in the habbit of JUST stopping instead of pausing, as that has never created the noise for me, but has any one else had this issue?


----------



## canthearyou

Mshenay said:


> I don't know why, but using Kernal Streaming with my Audio GD NFb10ES2 some times when I go from Pause to play I get a LOT of static. I'm now in the habbit of JUST stopping instead of pausing, as that has never created the noise for me, but has any one else had this issue?


Have you tried adjusting the buffer?


----------



## Makiah S

canthearyou said:


> Have you tried adjusting the buffer?


 yeap I have it set 1000, lowering it to 800 helps a little but it's still an issue. 

I guess I'll need to see what the lowest possible buffer value I can have before the visualizations stop working


----------



## 514077

Claude said:


> With the DSD processor which is part of the recent versions of the SACD plugin I can set conversion of PCM signals up to DSD128. The higher settings don't work. The DAC confirms that it receives DSD128.


Have you, or has anyone noticed a small delay when listening to DSD when the track starts?  I usually lose the first few notes on the first track when starting FB.  I'm using Hugo andSACD version 1.0.5.


----------



## Claude (Jun 8, 2017)

Yes. That happens too with my DAC (T+A DAC8 DSD), when playing DSD.

I've read somewhere else that this is a Foobar2000 issue. The player switches to DSD output only when playback starts, and then the DAC switches to DSD input. With files where the music starts without delay, a second of music gets lost at the start. It's annoying.

Even when pausing, skipping tracks or fast forwarding in Foobar2000, the output signal switches to PCM, but that can be avoided of course.

I would have changed to other players (HQplayer) for DSD playback, but all my SACD rips are in ISO format (I don't want to convert them to DSD files, as I have a large collection) and AFAIK only Foobar2000 supports SACD ISO and DSD output.


----------



## 514077

Claude said:


> Yes. That happens too with my DAC (T+A DAC8 DSD), when playing DSD.
> 
> I've read somewhere else that this is a Foobar2000 issue. The player switches to DSD output only when playback starts, and then the DAC switches to DSD input. With files where the music starts without delay, a second of music gets lost at the start. It's annoying.
> 
> ...


That's good to know where the problem lies.  I thought it might have been due to Rob Watts' preferences in designing the Hugo; I don't believe he's a big fan of DSD.  Frankly, I like the sound of my DSD files, and I get tired of reading a lot of neh-sayers who are just sniping DSD to promote their specific prejudices and flogging various products.  The Schiit Loki comes to mind.  Naming a product for a format they didn't care much for, after the worst norst god.  Sorry, I'm done now.  Just would be cool if FB comes out with a workaround for this.


----------



## Claude

Maybe DACs which can switch from PCM to DSD without any delay don't have this problem with Foobar2000 and DSD. 

My T+A DAC8 DSD has two seperate DACs inside for PCM and DSD, so it's inevitable that there is a certain delay when the input signal changes between PCM and DSD.


----------



## canthearyou (Jun 8, 2017)

Is it not possible to play a 24/96 audio file as DSD? All my 16/44.1 play fine at both DSD512 & DSD256. When I try to play a 24/96 file it says 9600 not supported. DAC is Matrix X-Saber Pro.


----------



## 514077

canthearyou said:


> Is it not possible to play a 24/96 audio file as DSD? All my 16/44.1 play fine at both DSD512 & DSD256. When I try to play a 24/96 file it says 9600 not supported. DAC is Matrix X-Saber Pro.


Would you have a 24/88.2 file to try?  And have you tried a 24/192 file?  I just wondered if it had something to do with files that are non-multiples of 44.1.  Just a thought.


----------



## canthearyou

UELong said:


> Would you have a 24/88.2 file to try?  And have you tried a 24/192 file?  I just wondered if it had something to do with files that are non-multiples of 44.1.  Just a thought.



It's only the X-Saber that will not play 24/96 as DSD. When I swap in the X-20 it plays everything perfectly in DSD. I believe I've tried every possible setting, but it will not play a 24/96 as DSD.


----------



## 514077

I have a new question.  I have a DSF album with an included cue file.  Is there a program to break up the file into its individual tracks without converting it to a PCM format?  It's easy to break up flac etc with FB2K's 'quick convert' function.  But I wish to avoid converting DSD to PCM if it can be helped.


----------



## PleasantSounds

Google the Tascam Hi Res Editor. It's free and handles DSF/DFF files.


----------



## 514077

PleasantSounds said:


> Google the Tascam Hi Res Editor. It's free and handles DSF/DFF files.


Thanks once again, @Pleasant Sounds.  I'll look it up.  You've been a great help to me in the past.


----------



## castleofargh

UELong said:


> Thanks once again, @Pleasant Sounds.  I'll look it up.  You've been a great help to me in the past.


I tell him, extortion, control of the information, propaganda... those are the good tools to benefit from people in need. altruism has very little economical or even social value, and yet he keeps on reading help topics and trying to answer questions when he can. he's been a constant disappointment for all aspiring dictators around the world.


----------



## PleasantSounds

castleofargh said:


> I tell him, extortion, control of the information, propaganda... those are the good tools to benefit from people in need. altruism has very little economical or even social value, and yet he keeps on reading help topics and trying to answer questions when he can. he's been a constant disappointment for all aspiring dictators around the world.



At least I'm not alone


----------



## alphanumerix1

What the best way to split dff files? Foobar is reading the albums a full blocks and not splitting.


----------



## Stanl3h

Hi guys I am really sorry but I have tried for days to get this working. I have a oppo ha-1 and I want to be able to play dsd 256 files on foobar, I followed the guides mentioned earlier but I must be missing something or I am just a idiot (entirely possible) it just says Unrecoverable playback error: Sample rate of 705600 Hz not supported by this device followed by No installed FIR, continue with the default
Would it be possible for one of you amazing people to rar your foobar that is working with dsd256 and upload it and I can try that ?

Thanks


----------



## 514077

Ryan Sayce said:


> Hi guys I am really sorry but I have tried for days to get this working. I have a oppo ha-1 and I want to be able to play dsd 256 files on foobar, I followed the guides mentioned earlier but I must be missing something or I am just a idiot (entirely possible) it just says Unrecoverable playback error: Sample rate of 705600 Hz not supported by this device followed by No installed FIR, continue with the default
> Would it be possible for one of you amazing people to rar your foobar that is working with dsd256 and upload it and I can try that ?
> 
> Thanks


I tried to .rar my version, but could only get it down to 8MB using best compression.  I'm afraid you'll have to wait for someone greater to come on and help.


----------



## Stanl3h

8mb isn't much to upload, do you have a very slow internet ?


----------



## 514077

Ryan Sayce said:


> 8mb isn't much to upload, do you have a very slow internet ?


No.  But the file size would have to be broken up to go through.


----------



## Stanl3h (Sep 3, 2017)

Ok thanks anyway, I will wait for someone else to hopefully be my saviour.


----------



## WoodyLuvr

This thread is a fountain of knowledge!

FYI: A new thread has been started over at Sound Science titled: *The DSP Rolling & How-To Thread*


----------



## 514077

WoodyLuvr said:


> This thread is a fountain of knowledge!
> 
> FYI: A new thread has been started over at Sound Science titled: *The DSP Rolling & How-To Thread*


Do they mention the VSP rolling and headake problems over there?  Sounds interesting, anyhow.  I'll give it a looksee.


----------



## theaudiologist

is foobar's convertion good? if i convert a flac to wav of the same bit depth and sample rate will it lose quality?
if i undersample the flac file into a lower sampling rate will foobar give it more noise and quantization errors?
if i change a 6-channel flac file to a 2-channel will it lose quality and the errors in the line above?


----------



## terry81

Using a 24/96 DAC on USB and WASAPI event in foobar, is it possible to play 24/192 files? I'm getting an error unless I select DS.


----------



## Slogra

You need to resample to 24/96 to make wasapi work on your dac. So go Options -> DSP and select a resampler and set it to 96khz. 

While you are at it, you might wanna try 44.1khz to find out if you can hear any difference 

Personally i use the ssrc resampler.


----------



## terry81

Slogra said:


> You need to resample to 24/96 to make wasapi work on your dac. So go Options -> DSP and select a resampler and set it to 96khz.
> 
> While you are at it, you might wanna try 44.1khz to find out if you can hear any difference
> 
> Personally i use the ssrc resampler.



Makes sense. Thank you and Merry Christmas!


----------



## PleasantSounds

theaudiologist said:


> is foobar's convertion good? if i convert a flac to wav of the same bit depth and sample rate will it lose quality?
> if i undersample the flac file into a lower sampling rate will foobar give it more noise and quantization errors?
> if i change a 6-channel flac file to a 2-channel will it lose quality and the errors in the line above?



Flac and wav are both lossless formats - this means the content fed to the DAC is identical. All you lose converting flac to wav is disc space and ability to embed metatags in the audio file.
Resampling results may vary depending on the resampler used and the parameters chosen. I use SoX which is an add-on component and find it very good for both up- and down-sampling.
Downmixing from 5.1 to stereo is rarely as good as getting the proper stereo mix, but if that's the only source you have it's still OK. The resulting stereo image may be not as clear as in the stereo mix.


----------



## 514077

PleasantSounds said:


> Flac and wav are both lossless formats - this means the content fed to the DAC is identical. All you lose converting flac to wav is disc space and ability to embed metatags in the audio file.
> Resampling results may vary depending on the resampler used and the parameters chosen. I use SoX which is an add-on component and find it very good for both up- and down-sampling.
> Downmixing from 5.1 to stereo is rarely as good as getting the proper stereo mix, but if that's the only source you have it's still OK. The resulting stereo image may be not as clear as in the stereo mix.


Thanks for being a great resource on this thread.


----------



## timb5881

Does anyone know how I can get my Korg ds-dac100m to play dsd files on foobar?   The only thing I can do now is let the output be pcm.


----------



## seoman

timb5881 said:


> Does anyone know how I can get my Korg ds-dac100m to play dsd files on foobar?   The only thing I can do now is let the output be pcm.



Have you installed the asio component in foobar?

Korg sais:


> ASIO (DSD playback is supported by the DSD option of ASIO 2.1 and later)


----------



## 514077 (Jan 27, 2018)

seoman said:


> Have you installed the asio component in foobar?
> 
> Korg sais:


I'm using the Hugo2 which can receive true DSD (no need for DoP'ing).  What I'm not sure of, is how to make sure I'm sending DSD without DoP?  I'm using the version 1.0.X components, now.  I'm an eegit.  I run DSD256 just fine.  So I think that answers my question.


----------



## timb5881

I did try that, and it wiped out all of my dsd drivers for other DAC’s, it still did not play dsd files, only if the output was set to pcm.


----------



## PleasantSounds

There's a bit more to playing DSD in f2k than just installing the ASIO driver, but the truth is you need it.
Then you can follow this tutorial and that should work.


----------



## timb5881

I have been playing native dsd for a while with Foobar2000, with my cyberdrive feather dac, and also using my Fiio X3 II as a dac.


----------



## 514077

Does someone know how to use 'VinylStudio' for splitting up a DSF file into its individual songs?  The lazy howto manual writers only know how to take snapshots, not write coherent explanations.  Thanks.


----------



## timb5881

Still no luck with DSD files playing on FooBar with the Korg DAC.


----------



## sandsha2

Could someone tell me, how to remove missing files from playlist? I created these by dragging songs into it.

Thanks,
Sandeep


----------



## Dawnrazor

Is there a way to get Foobar to send files or output to another player??


----------



## PleasantSounds

Dawnrazor said:


> Is there a way to get Foobar to send files or output to another player??



It would help knowing what player and what exactly you're trying to achieve and why.

Foobar can act as an UPnP/DLNA server  (and renderer) - if that would work for you then google foobar upnp and you should get on the right path quickly.


----------



## Dawnrazor

PleasantSounds said:


> It would help knowing what player and what exactly you're trying to achieve and why.
> 
> Foobar can act as an UPnP/DLNA server  (and renderer) - if that would work for you then google foobar upnp and you should get on the right path quickly.


Hi PleasantSounds.  

I have used Foobar as a renderer but would like it to send its sound to cplay.  Cplay sounds better but doesnt function as a renderer.  Renaissance player is a great renderer but doesnt support ASIO etc.

Thanks


----------



## jcn3

Dawnrazor said:


> Hi PleasantSounds.
> 
> I have used Foobar as a renderer but would like it to send its sound to cplay.  Cplay sounds better but doesnt function as a renderer.  Renaissance player is a great renderer but doesnt support ASIO etc.
> 
> Thanks



foobar can be set up as a dlna server, but will be looking for dlna renderers to communicate with.

i'm not clear on why asio is so important -- what's the thinking there? (e.g. what's wrong with wasapi?)


----------



## seoman

jcn3 said:


> i'm not clear on why asio is so important -- what's the thinking there? (e.g. what's wrong with wasapi?)



With asio you have control. you can assign output channels to actual outputs.
A favorite asio setting of mine was
FR >> FR and FR >> RR
FL >> FL and FL >> RL

wasapi seems to do other stuff like mixing FL FR together and send it to the center.
Which results in a clipping signal. Not very nice!


----------



## Dawnrazor

jcn3 said:


> foobar can be set up as a dlna server, but will be looking for dlna renderers to communicate with.
> 
> i'm not clear on why asio is so important -- what's the thinking there? (e.g. what's wrong with wasapi?)


Wasapi doesnt seem to work that well with my hardware or at all.  Both are pro cards and asio just works.  And Soeman details other benefits.  And add lower latencies.  

So no way to get foobar to output to another player it sounds like.


----------



## seoman

Dawnrazor said:


> Wasapi doesnt seem to work that well with my hardware or at all.  Both are pro cards and asio just works.  And Soeman details other benefits.  And add lower latencies.
> 
> So no way to get foobar to output to another player it sounds like.



Still why Cplay? is asio from foobar not good enough?
or have you migrated to w8/10? and get ticks and pops?


----------



## Dawnrazor

seoman said:


> Still why Cplay? is asio from foobar not good enough?
> or have you migrated to w8/10? and get ticks and pops?


I am not 100% i tried Foobar with ASIO in the situation i am doing currently but Cplay has always worked with any ASIO driver.  Not every player works.  Winamp I know doesnt with my m-audio revo sound card and at best is buggy.  Not sure if Foobar works either.  

Also when there were no compatibility issues to contend with, cPlay just sounded better than Foobar.  Also I am not sure how foobar implements its vc which is probably not as good as cPlay does.  And i think cPlay has a smaller footprint than Foobar since Foobar has many many many more but useless features.  

I may be forced to use Foobar since there are few renderers that support ASIO.  It seems to be the only player that does that.  Winamp which I like better than foobar for sound is useless as a renderer or I cant get it and asio to work at the same time!


----------



## seoman

I tried Cplay. 
What: last version 2010. Build for XP (!!!!) 
But it ran and sounded indeed very good. 
Horrible player though.
My wasapi is f*cked up now. Asio still works ok. But wasapi became extremely noisy. I don't understand so maybe i have to do a system restore.

Still since Microsoft declared the war on audio. It all went dawn to a horrible shitshow. (vista and later) 
Wasapi is the only way for feeding multi channel hi-res though HDMI. Still, MS doen't give a damn and it shows on all fronts. 
But i thank god that it works. 8PCM channels upto 96Khz or 192khz for 6channel PCM.

still use my soundcard and foobar2000 but my oppo203 and nas renders a windows based player to a redundant media platform.  
The only thing that works more convenient on windows is finding/adding a subtitle.

So i'm wondering if i will ever (re)build a media PC??
as for Cplay. it seems that it's still alive on linux


----------



## Dawnrazor

seoman said:


> I tried Cplay.
> What: last version 2010. Build for XP (!!!!)
> But it ran and sounded indeed very good.
> Horrible player though.
> ...


I could have saved you the trouble.  Cplay is one of the best sounding players but it only shines in the context of a cmp2 build using cmp as a windows explorer replacement and using cue sheets. It will do 1/10 what most players will do and yeah it was designed for XP since thats the smallest footprint and can be run with a much slower processor and under 512mb of ram

Here is a vid of cmp which acts as the album selector and hands the files to cplay to play.  Its all conrolled buy the mouse scroll wheel so no key board or dragging the mouse around.


Its this arrangement i want to replicate just with a dlna renderer acting as cmp.

Cplay only does ASIO so no reason why it would affect the wasapi afaik.  In a cmp2 build windows audio service is disabled but you have to manually do that- cplay doesnt mess with windows audio so its bizarre you are having issues.  Have you rebooted?


----------



## d34dj3d1

Can someone please tell me how to properly set up the crossfade so one song blends seamlessly into the next with no gaps between them.


----------



## 514077

I have a question about using Wavpack 5.1 and BatchEncoder to losslessly compress DSD files.  I've followed the instructions as I interpreted them, but can't get it to work.  I'm using Windows 7.  Is there someone familiar with this who could give me a bit of help?


----------



## Slogra

d34dj3d1, not sure if you still need help with crossfade, but here goes:

Go to File -> Preferences -> Playback -> DSP Manager, then: 
- move Crossfader to the left column
- Select it and click "Configure selected" if you want to change the default 2 second crossfade duration
- I'm not sure where it should be in the chain (move up/down), maybe somebody else knows? Anyway, I moved to the bottom; so after downmix, resample and convert stereo to mono. This way the frequency and channel will be the same when crossfading


----------



## d34dj3d1

I know how to get to it I just don't know what settings do what and how to set it to do what I want it to do.


----------



## 514077

UELong said:


> Oh yes, I have iso2dsd which works, and dff2dsf which unfortunately, doesn't work on my computer.  I assume it's because I'm stuck on XP, for now.
> I bet I'll get results on W7, when I upgrade.  PS:  I did.  DFF2DSF works now.
> Thanks.


Update:  DFF2DSF has stopped working for me, suddenly.  I tried to reinstall it, but the .exe file doesn't execute.  (Windows 7).  Any thoughts?
Thanks.


----------



## music_man

You will be glad you see this through. For free yet.


----------



## 514077

music_man said:


> You will be glad you see this through. For free yet.


Were you responding to some post?  Just curious.


----------



## jcn3

UELong said:


> Were you responding to some post?  Just curious.



i believe that he was responding to your post about dff2dsf


----------



## 514077

jcn3 said:


> i believe that he was responding to your post about dff2dsf


That's the thing about D2D.  It's free, and it works well when it works.  But, being free, there's no way I know how to find out how and why it stopped working.  Not quite sure where I went wrong, and no way to find out.  If I'd payed for it, I'd be more frustrated, especially when there's so much paid for software that isn't writen with accessability in mind.  For example:  I had to have my son help me with ISO2DSD converter, until I came across Scarlet SACD comverter, which is perfectly accessable.  I also wanted to ssplit some DSF files with a cue file without converting to pcm.  Corg Audiogate and Tascam high-res editor are supposed to accomplish just that.  But, my speech program just doesn't see the menus or controls, to make them usable.
As Bruce Hornsby said:  "That's just the way it is."


----------



## music_man

No, I was just saying to everyone here if you see it through because it is tricky it will reward you with great sound for free. That's all. No one in particular, everyone. I am not good at explaining things but figuring them out. Foobar imo is the best overall sound program. No need to pay money but nothing wrong with that either if something is worth it.  It just so happens that Foobar, when working is outstanding. I was just giving everyone encouragement even though I am not of much help.


----------



## d34dj3d1

We all realize that it's a great program. But this thread is for people who need help with learning how to set it up/use it. Hence why it's entitled "The foobar2000 help thread".


----------



## music_man

I was just trying to offer encouragement. I am very good using software but no good explaining it. I mean that, not being a jerk. I do not know why you think I am such a jerk.


----------



## 514077

music_man said:


> You will be glad you see this through. For free yet.


Well, you are right in that FB is a great piece of software; completely customizable.  I was harping on a program unrelated to FB except that it was free.  If it helps, I don't think you're a jerk.


----------



## music_man

Thank you. In case anyone did not know I have advanced Parkinson's disease. I am lucky to be alive. that's all. I was much more keen. I have to hire fab to do all my projects now. I can no longer even use solder. I am not looking for pity. Just explaining why I am having trouble remembering things. I mean if I make a setting etc. I will forget what I did in 5 minutes and remember 2 weeks later. So I am not the person you want explaining things lol.

I seriously hope people pitch in help. Since the program is outstanding. In many ways more customizable than Jriver but very difficult. Which is obviously why people need help. Trust me I am the last guy you want to ask. I just mean because my short term memory is like my long term memory now. If that even makes sense.

I will keep out of here though and just hope someone is willing to help everyone. Since it really is a good program. Even things I do not understand. Like why Jriver takes exclusive of Asio but Foobar does not. I mean there is something I am stumped on. Even when I was well I did not know everything.


----------



## music_man

Okay I actually have a question! this has stumped me. Is there a way to force exclusive in Foobar while using a 3RD party ASIO driver? AFAIK, no but I do not know everything. 
I honestly did not mean to be a jerk. I hope I explained above. I literally do not know what I am doing anymore. So I was just saying I am like the last person to ask.


----------



## MelonHead

music_man said:


> Okay I actually have a question! this has stumped me. Is there a way to force exclusive in Foobar while using a 3RD party ASIO driver? AFAIK, no but I do not know everything.
> I honestly did not mean to be a jerk. I hope I explained above. I literally do not know what I am doing anymore. So I was just saying I am like the last person to ask.


I don't know your equipemt and setup, and I'm not a power-user as well, but these links helped me a lot to understand the detailed configuration of Foobar 2000.
https://diyaudioheaven.wordpress.com/digital/pc-software/foobar-2000-for-dummies/
https://diyaudioheaven.wordpress.co...-part-3-new-experimental-sacd-plugin-v-0-9-x/
Sorry if you know them already.


----------



## music_man

I did not know those. Thank you I am pretty sure the issue is the DACS driver. I used to be able to modify the ini or whatever but not up to it now. Not a huge deal if I do not play any other sounds but not exactly bit perfect either I gather.


----------



## VNandor

How hard is it to get foobar to play back hi-res files? I recently tried it and out of curiosity I measured the output of my soundcard and to my surprise it didn't have any content over ~20kHz. 
After that I made some sine sweep that goes to like 80kHz to make 100% sure there's going to be high frequency content. In the output menu I tried using the "primary sound driver" which cut off everything over ~20kHz and "wasapi (event)" which caused aliasing. I have the spectrum of how the sweep should look like, as well as the spectrum of the actual analog output if that could help.

My soundcard's (asus xonar essence stx which according to foobar's FAQ apparently doesn't work very well) sample rate is set to 192kHz. I'm also using windows 10. 

Basically my question is, how to make sure that everything is set right on foobar so I can move to the soundcard or windows or whatever else that could cause this problem?


----------



## ComradeDylie (May 23, 2018)

VNandor said:


> How hard is it to get foobar to play back hi-res files? I recently tried it and out of curiosity I measured the output of my soundcard and to my surprise it didn't have any content over ~20kHz.
> After that I made some sine sweep that goes to like 80kHz to make 100% sure there's going to be high frequency content. In the output menu I tried using the "primary sound driver" which cut off everything over ~20kHz and "wasapi (event)" which caused aliasing. I have the spectrum of how the sweep should look like, as well as the spectrum of the actual analog output if that could help.
> 
> My soundcard's (asus xonar essence stx which according to foobar's FAQ apparently doesn't work very well) sample rate is set to 192kHz. I'm also using windows 10.
> ...




I would try to use KS or ASIO and see what happens.  I typically use KS for all of my stuff and have never had a problem.  I mean there isn't a good way to check most dacs but my dragonfly always lights up correctly.

How are you measuring the output?  I would like to try this.  I have the same card but I typically just use the optical out into my DAC for games/web browser audio and use the direct USB connection for my DAC when I play music from foobar.     

Let me know how to test the output and I can check to see if I can get mine working.


----------



## VNandor

ComradeDylie said:


> I would try to use KS or ASIO and see what happens.  I typically use KS for all of my stuff and have never had a problem.  I mean there isn't a good way to check most dacs but my dragonfly always lights up correctly.
> 
> How are you measuring the output?  I would like to try this.  I have the same card but I typically just use the optical out into my DAC for games/web browser audio and use the direct USB connection for my DAC when I play music from foobar.
> 
> Let me know how to test the output and I can check to see if I can get mine working.



I connected the headphone out where my mic normally goes, I think its called the line in. I used audacity for the actual recording (project sample rate set to 192kHz as well) and set the level in a way I would get a good SNR but no clipping. I then exported the file, and used SPEK to check its spectrum.
Now, the A/D converter  after the line in might have caused the aliasing but I also have an analog oscilloscope that I used to check what's on the output and it confirmed the sine sweep either cuts out, or start "wrapping" around 20kHz and instead of the frequency going upward, it goes down.
Point is, I know for a fact my output is already bad and it's not just the recording that messes up the signal.

I'll try and see what happens if I use KS or ASIO.


----------



## VNandor (May 23, 2018)

With Kernel Streaming I got aliasing, ASIO just cut off everything around ~20kHz.

I just checked both with the scope as well and it seems ASIO works well for the output, however there's probably a filter somewhere in the line in stage that cuts offf everything after 20kHz.


----------



## seoman (May 23, 2018)

VNandor said:


> How hard is it to get foobar to play back hi-res files? I recently tried it and out of curiosity I measured the output of my soundcard and to my surprise it didn't have any content over ~20kHz.
> After that I made some sine sweep that goes to like 80kHz to make 100% sure there's going to be high frequency content. In the output menu I tried using the "primary sound driver" which cut off everything over ~20kHz and "wasapi (event)" which caused aliasing. I have the spectrum of how the sweep should look like, as well as the spectrum of the actual analog output if that could help.
> 
> My soundcard's (asus xonar essence stx which according to foobar's FAQ apparently doesn't work very well) sample rate is set to 192kHz. I'm also using windows 10.
> ...



I have no issues with my stxII or had any with my ST
But since you select "primary sound driver" you are trying to go through the windows driver but that one needs to be configured first!
Basicly ALL possible outputs in W10 need to be configured (except for asio i gues)
So tell windows the xonar essence is capable of running 192K


----------



## 514077

VNandor said:


> With Kernel Streaming I got aliasing, ASIO just cut off everything around ~20kHz.
> 
> I just checked both with the scope as well and it seems ASIO works well for the output, however there's probably a filter somewhere in the line in stage that cuts offf everything after 20kHz.


Am I wrong, or was kernal done away with since Windows 7?  I thought WASAPI replaced kernal.  Meanwhile, I use asio, my old SB card can be configured for 96k.  But, my DAC goes as high as the content from USB.  Hope some of this helps.


----------



## VNandor (May 23, 2018)

seoman said:


> But since you select "primary sound driver" you are trying to go through the windows driver but that one needs to be configured first!
> Basicly ALL possible outputs in W10 need to be configured (except for asio i gues)
> So tell windows the xonar essence is capable of running 192K



Oh, I haven't done that yet. That probably caused pretty much all my problems.

EDIT: I just did it and for the inputs as well. I get aliasing with the primary sound driver this time with the line in and the scope as well.

With ASIO it seems the output is good, but if I connect it to the line in, it cuts out everything over 20kHz (no aliasing with it tough). I'm not sure could the sound card, at least in theory, capture frequencies over20kHz?

Should I move to the STX thread? It seems like this doesn't really have to do much with foobar.


----------



## betula

I have got a smaller problem in Foobar, I wonder if anyone can help.

When I start to skip tracks relatively quickly, or skip bigger sections more often in a track, I start to hear loud or less loud pops. Sounds like a small electric shock to the drivers in the headphones which is a bit worrying. If I don't skip tracks or skip only every now and then, I can't hear this 'pop' sound. 

What I tried so far: 
- using my laptop on battery only or without battery from main plug
- changing all the cables
- changing buffer size in Foobar (50ms, 1000ms, 2000ms, 18-20000ms etc)
- changing output data format (24/32 bit)
- changing from ASIO to WASAPI and backwards

Any more suggestions? I use a DELL laptop and Mojo.


----------



## MelonHead

betula said:


> I have got a smaller problem in Foobar, I wonder if anyone can help.
> 
> When I start to skip tracks relatively quickly, or skip bigger sections more often in a track, I start to hear loud or less loud pops. Sounds like a small electric shock to the drivers in the headphones which is a bit worrying. If I don't skip tracks or skip only every now and then, I can't hear this 'pop' sound.
> 
> ...


Do you hear the same pop sound with any other player, using the same asio or wasapi driver?


----------



## betula

MelonHead said:


> Do you hear the same pop sound with any other player, using the same asio or wasapi driver?


I don't have any other player that uses asio/wasapi but for the sake of troubleshooting I might install one. Any suggestion?


----------



## castleofargh

betula said:


> I have got a smaller problem in Foobar, I wonder if anyone can help.
> 
> When I start to skip tracks relatively quickly, or skip bigger sections more often in a track, I start to hear loud or less loud pops. Sounds like a small electric shock to the drivers in the headphones which is a bit worrying. If I don't skip tracks or skip only every now and then, I can't hear this 'pop' sound.
> 
> ...


I'm going to assume that you don't have a bunch of DSPs activated on your computer(like some of the crap "enhancements" in the Windows properties of your DAC)? if you do try disabling those when playing music that could perhaps help reduce the occurrence of noises when changing songs. 
else most likely what happens is something similar to when some DAC has to change the resolution to comply with a file of a different sample rate. there can be some little "tac" noise like an electrical switch going off. of course there is absolutely no danger for any gear, and it's absolutely not a sign that something is defective. remember that you're not discussing proper casual music playback, but you using rapid fire mode on your skip button or randomly moving the playback slider^_^. it's like actively trying to mess with the rather fragile connection between 2 devices which attempt to follow your successive demands almost in real time and start loading and playing each and every stuff you selected along the way. so a little noise here and there is really no big deal IMO(and I'm not sure it's foobar's fault).
if you had those in the music while doing nothing, then I'd worry and try to find the issue.


----------



## ajreynol

I could use a little bit of help, actually. I've been using a columnsUI-based skin called eola for a very long time and I've always been happy with it except for one thing: for albums in which I have both a FLAC and MP3 version of, it will put all of the songs together, resulting in playing each song twice.

I can sidestep this issue by sorting by folder (I have all FLAC album variants in a FLAC folder), but I feel like this is a work-around and that there is a solution out there. I've done some custom grouping which helps on the front end, but it doesn't stop Foobar from playing the MP3 version of the same song and bouncing back and forth between versions if I don't specify the FLAC folder. I know this sort of grouping is possible as I know other skins can do it, but I really like this skin save this single issue. I'm just not sure where to look anymore as I am new to the configuration and setup side of Foobar.

Any help or suggestions will be appreciated.


----------



## music_man

On my system Wasapi is exclusive but not bit perfect. Asio is bit perfect but not exclusive. Instead of asking about trying to force it exclusive.... If I am not playing any other sounds, does having the program take exclusive access improve the sound quality in any way? If it does I am not sure what I can do to get it to take this Asio driver as exclusive. It does in Jriver but not in Foobar. I have no idea why.


----------



## seoman (May 28, 2018)

music_man said:


> On my system Wasapi is exclusive but not bit perfect. Asio is bit perfect but not exclusive. Instead of asking about trying to force it exclusive.... If I am not playing any other sounds, does having the program take exclusive access improve the sound quality in any way? If it does I am not sure what I can do to get it to take this Asio driver as exclusive. It does in Jriver but not in Foobar. I have no idea why.



As far as i know asio has always been the same: it can only connect 1 output to 1 sound source.
"exclusive" it is by design.
Back in the days it was designed to bypass all crappy OS/Driver. And it still functions like that.
it is so exclusive  not even windows can get to it.

ASIO was developed by steinberg a company that build music software for profesionals.
To get it all working at a pro level they had to bypass windows.
And in my opinion there is still a valid reason to keep windows away from your audio.


----------



## music_man

Yes, I use Nuendo every day. The issue is for instance, while Foobar is playing with the ASIO driver I can simultaneously play a YouTube video. This makes no sense. Then if I try to use Wasapi it does take control but it is not Bit perfect. The ASIO driver works as intended in Jriver and HQplayer. I just do not understand this. there is nothing to "set" that could be wrong. So I have nothing to try to get it to work. For instance with Jriver if a YouTube video is attempted it says "This content was unable to load". Foobar just goes ahead and lets both play. I wonder if it is in fact even using ASIO to begin with. That is the driver selected but as you say, by nature it is exclusive. Very strange. I suppose it is fine so long as I do not attempt to play anything else, system sounds off etc. I do wish I could fix it or find out if in fact it is even using the ASIO driver to begin with. It very well may not be. Even though it is selected. I am using Foobar because it is the only software that can play this FLAC radio stream. Which is also very odd since it is a big radio station and I would think they would want it to work universally. I am doing at least a couple of things wrong but I cannot figure out either. I do prefer Foobar though. I paid for other software and like better the free one go figure. Perhaps reloading the drivers or something. I have no clue.


----------



## PleasantSounds

ajreynol said:


> I could use a little bit of help, actually. I've been using a columnsUI-based skin called eola for a very long time and I've always been happy with it except for one thing: for albums in which I have both a FLAC and MP3 version of, it will put all of the songs together, resulting in playing each song twice.
> 
> I can sidestep this issue by sorting by folder (I have all FLAC album variants in a FLAC folder), but I feel like this is a work-around and that there is a solution out there. I've done some custom grouping which helps on the front end, but it doesn't stop Foobar from playing the MP3 version of the same song and bouncing back and forth between versions if I don't specify the FLAC folder. I know this sort of grouping is possible as I know other skins can do it, but I really like this skin save this single issue. I'm just not sure where to look anymore as I am new to the configuration and setup side of Foobar.
> 
> Any help or suggestions will be appreciated.



I'm not using columns ui but had a similar problem with multiple versions of the same album. What worked for me was editing the album title tag, adding some descriptor at the end (or even a space). From then on F2k sees it as two different albums.


----------



## seoman

music_man said:


> Yes, I use Nuendo every day. The issue is for instance, while Foobar is playing with the ASIO driver I can simultaneously play a YouTube video. This makes no sense. Then if I try to use Wasapi it does take control but it is not Bit perfect. The ASIO driver works as intended in Jriver and HQplayer. I just do not understand this. there is nothing to "set" that could be wrong. So I have nothing to try to get it to work. For instance with Jriver if a YouTube video is attempted it says "This content was unable to load". Foobar just goes ahead and lets both play. I wonder if it is in fact even using ASIO to begin with. That is the driver selected but as you say, by nature it is exclusive. Very strange. I suppose it is fine so long as I do not attempt to play anything else, system sounds off etc. I do wish I could fix it or find out if in fact it is even using the ASIO driver to begin with. It very well may not be. Even though it is selected. I am using Foobar because it is the only software that can play this FLAC radio stream. Which is also very odd since it is a big radio station and I would think they would want it to work universally. I am doing at least a couple of things wrong but I cannot figure out either. I do prefer Foobar though. I paid for other software and like better the free one go figure. Perhaps reloading the drivers or something. I have no clue.



What asio component did you install in foobar? 
i use:
foo_out_asio.dll (2018-03-06 22:30:03 UTC)
    ASIO support 2.1.2


----------



## music_man

Well this is odd. With Jriver if trying to use YouTube for example while playing on ASIO it says "this content cannot be loaded now". Or something to that effect. With Foobar You tube constantly buffers the video and there is no audio from youtrube whatsoever. So I am guessing the answer is their engines just handle exclusive mode differently. There is also no audio playback from any other source while playing in Foobar. It seems as if it is playing, bar moving but no audio at all. So I assume it has taken exclusive access it just does so differently than Jriver. This is actually good however. Since reading the news etc. the system will not freeze like it does with Jriver. Since YouTube gives the message but many other video sources do not. By not trying to force it closed the system does not freeze. Once again Foobar wins for me.

I was just wondering what people think about using bit perfect playback vs using dsp,upsampling etc? My DAC already upsamples into the stratosphere so not sure that is any better. In fact, to me bit perfect sounds better. Depending on your dac ymmv I suppose.


----------



## music_man

I installed that but it does not show in the list. I am using my dac's native drivers.


----------



## seoman

Well there's your answer.
Make sure you make the component visible in foobar so you can acces the asio driver of the card.


----------



## music_man

So use Foo_Asio and not the actual drivers for my specific DAC? It says something like Chord Electronics DSD Asio Driver is what I have selected. Foo_asio is in the components list but does not show up in the output list. It says(from the Chord Driver) Bit perfect.


----------



## seoman

I have dedicated asio drivers for all outputs. Even the onboard realtek.
I don't have "Chord Electronics DSD Asio"
My Mojo USB dac (from Chord Electronics) asio driver is called 'Asio Chord 1.05'

What asio drivers did you install?
What device are we even talking about?


----------



## 514077

music_man said:


> So use Foo_Asio and not the actual drivers for my specific DAC? It says something like Chord Electronics DSD Asio Driver is what I have selected. Foo_asio is in the components list but does not show up in the output list. It says(from the Chord Driver) Bit perfect.


I use the Hugo2 and have installed the Chord 1.05 driver.  In my 'output' listbox, I have both Ds Chord Asio 1.05 (44.1-768) loosely quoted.
There's also 'Chord DSD 1.05.  Choose whatever works.  I use the DSD choice,as I have the DSD components installed.


----------



## music_man

You are right. Me stupid. There are two, neither seem to actually take exclusive. One says Chord_Asio 1.05 and the other says Chord DSD 1.05. It is a DAVE. I do not know why it does not work. I tried Directstream too and same thing. Well, I mean it works but for instance if you try to use another application you should get a message saying this application cannot access the sound device, it may be in use or busy. Right? Unless Foobar just works different. It will not play any other sounds but no error message from other application.


----------



## 514077

music_man said:


> You are right. Me stupid. There are two, neither seem to actually take exclusive. One says Chord_Asio 1.05 and the other says Chord DSD 1.05. It is a DAVE. I do not know why it does not work. I tried Directstream too and same thing. Well, I mean it works but for instance if you try to use another application you should get a message saying this application cannot access the sound device, it may be in use or busy. Right? Unless Foobar just works different. It will not play any other sounds but no error message from other application.


One small point:  If using DSD, and if you're on Fb version 1.4, which is in beta form right now, it's important to move your DSD drivers up to version 1.1.0.  You'll get weird highspeed playback if you don't.  Otherwise, I'm not really sure what's happening with your system.  Sorry.


----------



## castleofargh

music_man said:


> You are right. Me stupid. There are two, neither seem to actually take exclusive. One says Chord_Asio 1.05 and the other says Chord DSD 1.05. It is a DAVE. I do not know why it does not work. I tried Directstream too and same thing. Well, I mean it works but for instance if you try to use another application you should get a message saying this application cannot access the sound device, it may be in use or busy. Right? Unless Foobar just works different. It will not play any other sounds but no error message from other application.


I was intrigued by your issue so I tried my scarlett2i2 on the win10 computer. it's usually plugged to the old fanless PC in win7 that I use to measure stuff in "silence" with a mic). I get the same result as you on win10, but sorry I don't have a fix to get exclusivity back. 
the funny thing is that I can disable all playback devices in windowzz, and mute any of the 700 volume sliders you can find in win10 in various places, nothing impacts foobar's playback set to the asio driver of that DAC. so asio seems to work fine and to bypass everything. there is just an issue when it comes to the exclusive mode kicking in and telling windosss' mixer to ****. obviously I've checked that exclusive mode option was ticked in windose playback options, that is not the cause and I imagine it's not yours either. 

now one thing worth noting maybe is that there has been a lot of compatibility issues with this DAC's driver, it was pretty bad before and in win10 I had to wait for a "fixed" driver that's still not bug free. in 5 reboots to test stuff right now, twice the device was checked as muted and I have to pick another one as default output then come back. garbage compatibility.  
wasapi works fine. it's exclusive when I want to and not when I don't.


----------



## 514077

music_man said:


> You are right. Me stupid. There are two, neither seem to actually take exclusive. One says Chord_Asio 1.05 and the other says Chord DSD 1.05. It is a DAVE. I do not know why it does not work. I tried Directstream too and same thing. Well, I mean it works but for instance if you try to use another application you should get a message saying this application cannot access the sound device, it may be in use or busy. Right? Unless Foobar just works different. It will not play any other sounds but no error message from other application.


Just a quick, dumb question.  Do you have the correct driver for your Windows version?  I seem to recall that I had to download the Win7 driver.  And, of course, Win10 just had its 6-month major upgrade.  Glad I stayed with 7.


----------



## seoman

I tried my mojo on W10 with the asio driver and the driver is crappy and buggy.
Tracktor crashes on selekting the Mojo and foobar sometimes is able to to play but stops after a few secconds.

You might try the Asio4all driver to see if that's working better.
But my conclusion: the asio driver of chord electronics asio 1.05 sucks when running on Windows10.
On Win7 i had no problems with the driver 

look into asio4all. 
I tried it once and i was able to hookup my HDMI but the channel setup did not work.
I thought it would be nice to see if i could adres some individual speakers of my 5.1.4 system.
Maybe asio4all can adres the dac for you


----------



## seoman

UELong said:


> Just a quick, dumb question.  Do you have the correct driver for your Windows version?  I seem to recall that I had to download the Win7 driver.  And, of course, Win10 just had its 6-month major upgrade.  Glad I stayed with 7.



I just reinstalled the latest w10 driver, still garbage


----------



## seoman

i just installed the asio4all driver but the driver does not see the Mojo.


----------



## 514077

seoman said:


> I tried my mojo on W10 with the asio driver and the driver is crappy and buggy.
> Tracktor crashes on selekting the Mojo and foobar sometimes is able to to play but stops after a few secconds.
> 
> You might try the Asio4all driver to see if that's working better.
> ...


At this rate, I may never go to 10; security issues be damned.


----------



## seoman

UELong said:


> At this rate, I may never go to 10; security issues be damned.



It is really the worst. It surpasses the milenium edition in crappyness. I have never seen such an unstable platform as W10 (and i started with dos4.2)
Stay away and hope you never have to update your hardware.


----------



## 514077

seoman said:


> It is really the worst. It surpasses the milenium edition in crappyness. I have never seen such an unstable platform as W10 (and i started with dos4.2)
> Stay away and hope you never have to update your hardware.


Which might you say, was the allround best Windows platform?


----------



## seoman

UELong said:


> Which might you say, was the allround best Windows platform?


I think XP was the most universal platform.
7 was ok but when it comes to audio win7 all ready lost it.

But pushing updates wich renders working solutions into garbage is the ultimate failure 
And if someone would start a class act against MS i would donate a lot of money.

Sorry for the off-topic talk


----------



## 514077

seoman said:


> I think XP was the most universal platform.
> 7 was ok but when it comes to audio win7 all ready lost it.
> 
> But pushing updates wich renders working solutions into garbage is the ultimate failure
> ...


Funny, I found that 7 did some of the things XP wasn't advanced enough for.  i.e.  DSD, which I kinda like.  But, in general, XP was the best, I thought.


----------



## music_man

Got latest driver same bs. WASAPI works fine. No biggie since WASAPI is probably better anyways. In recording so I know it is really for multi chan recording buffer way too low. Playback may be better with WASAPI. ASIO on it does pass as bit perfect however. I tried 2 other good DACS plus a pro one and same bs. So it is not Chord. It is WIN10! This Os honestly blows. I think I am going to load server. Go figure Server has much tighter security. The exploits you should not be on the internet with this. Certainly not with edge. Problem is not any player or Chord or MSB or anyone except Microsoft! Seriously if you were doing anything mission critical IOS is not okay either have to use Linux. The problem is going to be a lot of people do not understand it. I mean I do just fine. Windows is more fun until you get butt jacked. I already have a resident Viri that I cannot eradicate. I am not worried because it is just a browser exploit I shut down each time. This is still crazy though. Do you know how many people are using this in the world? My bank uses it, Gee Whiz.


----------



## 514077

music_man said:


> Got latest driver same bs. WASAPI works fine. No biggie since WASAPI is probably better anyways. In recording so I know it is really for multi chan recording buffer way too low. Playback may be better with WASAPI. ASIO on it does pass as bit perfect however. I tried 2 other good DACS plus a pro one and same bs. So it is not Chord. It is WIN10! This Os honestly blows. I think I am going to load server. Go figure Server has much tighter security. The exploits you should not be on the internet with this. Certainly not with edge. Problem is not any player or Chord or MSB or anyone except Microsoft! Seriously if you were doing anything mission critical IOS is not okay either have to use Linux. The problem is going to be a lot of people do not understand it. I mean I do just fine. Windows is more fun until you get butt jacked. I already have a resident Viri that I cannot eradicate. I am not worried because it is just a browser exploit I shut down each time. This is still crazy though. Do you know how many people are using this in the world? My bank uses it, Gee Whiz.


Gee Whizz?  (with a scoff)  That's because Windows is built for people to use computers; not to design and upgrade.  A computer is a tool to be used by the average person.  Linux is for DIY'ers who have the knowledge for computers.  I for one, use my machine for music enjoyment, not for the sake of the machine.  Linux is an elitist platform for those who care about that stuff.  Windows is supposed to be for people who need a job done.  So are vacuum cleaners, dishwashers etc.  If there are some vandals who love to infect computers out there, well, there are people who love to break into homes, smash windows....  I just won't be going to w10 anytime soon.


----------



## RuFrost

I found Night Mode in google chrome, opera, Music Folder Player, Acrobat Reader on the Android. Does night mode exist for foobar? I have default white version and it is just way to bright at night when there is no external lightning except for display.


----------



## music_man

You will need some skin. There are hundreds but many are quite complex to install. Unfortunately it does not have night/dark afaik built in. You could just turn down your backlight even during the day. It is easier on the eyes and makes the display last much longer. Like a display torched to 50% adds about 80,000 hours to it's lifetime if it is LED. Just an idea because skins with this can be pretty difficult. Or get Deskband and minimize it. If you need to run an ir remote then use Ghost. See, this is the only way like Jriver is better. Everything is built in. If you spend hours customizing I say Foobar is the best program for this task. Although I might try Winamp and Musicbee for the heck of it. Doubt they are as good but who knows. I paid for Jriver and HQplayer but I feel Foobar is better.


----------



## castleofargh

RuFrost said:


> I found Night Mode in google chrome, opera, Music Folder Player, Acrobat Reader on the Android. Does night mode exist for foobar? I have default white version and it is just way to bright at night when there is no external lightning except for display.


the default foobar skin will follow whatever Windows theme you have(I'm captain obvious).
for the list of songs, library, and other stuff inside foobar's "boxes", the super easy solution is to go to 'file' -> 'preferences'  and under 'display' -> 'colors and fonts' you just change the color of the font and set a black/darker background. 

for more customization, you can see just a few examples of the actual possibilities here. https://www.head-fi.org/threads/whats-your-foobar2000-setup.556919/page-34 
I'm the kind of guy to set windows to show Win98 theme, so I can't really help much with fancy eye candy. but I'm guessing Deviantart and places like that, still have creative dudes sharing their massive work for others to use and enjoy.


----------



## music_man

Well, I have another problem nagging me since day 1 with this. I cannot get MCE remote to work. Should just record in core application but does not. Ghost does not work nore MCEremote. Honestly I would be using Jriver as it is much easier. However to no avail will it play a certain FLAC stream I require. I know Jriver, easy. Just does not work. Could get remote functioning in Foobar would be completely satisfied. I do think it is the best sounding one even though they admit they all sound the same... I just need Play, Stop of all things to pull out my (no) hair over this! Any ideas? Some app I do not know of? Thanx


----------



## PleasantSounds

music_man said:


> Well, I have another problem nagging me since day 1 with this. I cannot get MCE remote to work. Should just record in core application but does not. Ghost does not work nore MCEremote. Honestly I would be using Jriver as it is much easier. However to no avail will it play a certain FLAC stream I require. I know Jriver, easy. Just does not work. Could get remote functioning in Foobar would be completely satisfied. I do think it is the best sounding one even though they admit they all sound the same... I just need Play, Stop of all things to pull out my (no) hair over this! Any ideas? Some app I do not know of? Thanx



There is a pretty good Android remote app if you happen to use this platform: check the foobar2000 controller.
You'll get much more than start/stop.


----------



## waspa

What setings should make to foobar when i am using 2.1 sound system?Have a 2 rear and 1 sub.Thank you for advice.


----------



## d34dj3d1

Just got a pair of Sennheiser HD 380 Pros what DSPs or other things should I be using to complement them?


----------



## music_man

Jriver has sub management with LFE. I imagine there is a plugin for Foobar.

I want to use my URC remote. Just need play and stop. It should do it right from the main program with keyboard shortcut but will not work for me.

I prefer Foobar. Jriver is much easier. It will not play the FLAC stream I listen to. So it is out. The remote is less of a deal breaker but I am sure I am just setting it up wrong.


----------



## seoman

d34dj3d1 said:


> Just got a pair of Sennheiser HD 380 Pros what DSPs or other things should I be using to complement them?



The only thing the HD380 pro needs is a normal cable.
I use the sennheiser 531406


----------



## gimmeheadroom

Got a new Audiolab 8300CD player with ESS 9018 USB DAC. Foobar plays all bitrate flac and DSD 2.8 and 5.6. The manual says this unit can play DSD256 (11.2) but foobar pops up a console saying something like the device doesn't support bit rate of 705060. Probably not exactly the right number but don't have the windows box running now.  I'm still searching and found a few hits but so far all from like 2012 solved by upgrading to newer asio plugin and some not resolved. Not exactly getting helpful responses from Audiolab. Buy our player for 1200 Euros = we will direct you to somebody closer to you and further from the people who actually might know enough to be able to help.

Win 10 Pro, foobar2000 with everything updated. Just in case anybody knows what the problem could be. Thanks.


----------



## 514077

gimmeheadroom said:


> Got a new Audiolab 8300CD player with ESS 9018 USB DAC. Foobar plays all bitrate flac and DSD 2.8 and 5.6. The manual says this unit can play DSD256 (11.2) but foobar pops up a console saying something like the device doesn't support bit rate of 705060. Probably not exactly the right number but don't have the windows box running now.  I'm still searching and found a few hits but so far all from like 2012 solved by upgrading to newer asio plugin and some not resolved. Not exactly getting helpful responses from Audiolab. Buy our player for 1200 Euros = we will direct you to somebody closer to you and further from the people who actually might know enough to be able to help.
> 
> Win 10 Pro, foobar2000 with everything updated. Just in case anybody knows what the problem could be. Thanks.


I'm using Win7 with FB2K 1.3.19, and can get DSD256 into my Hugo2.  Are you using the Foo_input_sacd 1.0.X or something earlier?


----------



## gimmeheadroom

UELong said:


> I'm using Win7 with FB2K 1.3.19, and can get DSD256 into my Hugo2.  Are you using the Foo_input_sacd 1.0.X or something earlier?



Thanks UELong. Components says:

foo_out_asio  2.1.2
foobar2000 core 1.3.17
Super Audio CD Decoder 1.1.0 (foo_input_sacd)

Everything has been updated to the latest versions.

Actual error message from the console popup is:

Unrecoverable playback error: Sample rate of 705600 Hz not supported by this device

Thanks


----------



## music_man

Cannot get MCE to work still. It is straightforward. No plugin. Won't work. Obviously I am not doing it right. I have no clue how else to do it. Otherwise Foobar IMO is the top player.


----------



## 514077

gimmeheadroom said:


> Thanks UELong. Components says:
> 
> foo_out_asio  2.1.2
> foobar2000 core 1.3.17
> ...


I'm not positive about this.  But, why not try Foo_input_sacd v1.0.11?  I think v1.1.0  works better with the FB v1.4 beta series.  But it might help.


----------



## gimmeheadroom

UELong said:


> I'm not positive about this.  But, why not try Foo_input_sacd v1.0.11?  I think v1.1.0  works better with the FB v1.4 beta series.  But it might help.



Thanks, it's worth a try. I'll get back to you in a day or two. I would hope the later version didn't break anything, that would kinda suck. But you never know.

Weird thing is I get the error even with the device switched off. So it does kinda point to an error in foobar rather than the device. The error message was reported by a lot of people. In many cases it was not solved at least the guy saying he had the problem never confirmed it got fixed. About half the time the suggestions were to update the firmware on the device (not possible in my case AFAIK) or to update foobar or components. My foobar is completely updated. I also uninstalled all the drivers for every other device on this Windows box. I had three other audio device drivers installed but uninstalled them. There should be no driver conflicts.


----------



## gimmeheadroom

@UELong sorry for the delay. I didn't try this yet. I am waiting for my Oppo 205 to show up and if that works then the problem is narrowed down to my Audiolab. If not, it would suggest a foobar or plugin issue. I'll update when I have any info. Thanks.


----------



## 514077

gimmeheadroom said:


> Thanks UELong. Components says:
> 
> foo_out_asio  2.1.2
> foobar2000 core 1.3.17
> ...


I just reread your post, this time with my brain in circuit.  Just didn't recognize the number you quoted.  I usually see it as 705.6KHz.  That's a .PCM multiple of 44,100Hz.  That's not DSD.  I think your DAC may not be able to decode 705.6.  If you can arrow down in the preferences listbox to SACD, then check to see if its output is set to PCM or DSD, that might help.


----------



## gimmeheadroom

UELong said:


> I just reread your post, this time with my brain in circuit.  Just didn't recognize the number you quoted.  I usually see it as 705.6KHz.  That's a .PCM multiple of 44,100Hz.  That's not DSD.  I think your DAC may not be able to decode 705.6.  If you can arrow down in the preferences listbox to SACD, then check to see if its output is set to PCM or DSD, that might help.



Thanks, it bothered me too and I couldn't put my finger on it. The doc on the Audiolab DAC refers to a version of Windows I don't have and to (very) old foobar2000 and it seems to contradict itself about what the settings actually required are. None of the settings screens are the same. I did try DSD and DSD+PCM and it didn't seem to have any influence. It is beyond difficult to find anybody who cares at Audiolab and I am beginning to think they made an incorrect or inaccurate statement in the doc. Their answer was to avoid foobar and try iRiver. But I am not interested in doing that, especially with no statement in the doc that only iRiver is supported and proof that it works.

I'll know more when the Oppo shows up. That is if people can confirm Oppo actually supports what it claims. Oppo's doc is pretty minimal on what formats it will play.


----------



## 514077 (Jul 26, 2018)

Hi guys:
I now notice a new DSD problem.  I have the new stable version of FB v1.4 with the DSD versions 1.1.0 for the new version of FB.  DSD64 plays fine, but 128 and above does the old playing part of a song normally, while the counter on the status bar rips through the file in a few seconds.  I'm using the Hugo2 with its Windows 7 driver.  Has anyone noticed a similar problem, and/or can offer a solution?
Thanks in advance, of course.
Oops!  I made a mistake.  It reacts this way only through the Hugo2 at all DSD rates.  If I play the track with my internal soundcard that can't reproduce DSD sound, the counter reacts normally.


----------



## gimmeheadroom

Sorry, I have not tried a newer version. I believe my Oppo does play DSD256 fine btw, sorry I didn't update. I'll check again over the weekend, lately I have been playing mostly from dlna rather than foobar.

How are you liking your Hugo 2?


----------



## 514077

gimmeheadroom said:


> Sorry, I have not tried a newer version. I believe my Oppo does play DSD256 fine btw, sorry I didn't update. I'll check again over the weekend, lately I have been playing mostly from dlna rather than foobar.
> 
> How are you liking your Hugo 2?


I have to say I love it.  It's much better than the Hugo1; a big improvement.  I like anylitical sound.  Not a 'warm' person.


----------



## sakujou (Aug 12, 2018)

Hey everyone,

I am having a really hard time configuring foobar to play DSD files through my AVR. Read tons of stuff online, foobar for dummies, installed plagins, etc.

So my setup is as follows;
Windows 10 PC -> foobar-> - HDMI -> Denon X3300 AVR (in Pure Direct)

The AVR is suppose to support DSD through HDMI, even though info on that topic is scarce.

My foobar SACD settings look like this:
https://ibb.co/fiZ9ap






When I try this output setting:
https://ibb.co/bHY2vp





Or any of the other wasapi or dsd : wasapi settings with the Denon flacs  &other files play fine. My AVR info says the sample rate is 192kHz as it should.
But when I try to play dsd 64/128 I get these errors and there is no playback and sound:
https://ibb.co/cRDP89





When I setup the output setting like this (the highlighted option), dsd files are playing:
https://ibb.co/hKfhvp





But I get this error at the beginning of every track.
https://ibb.co/nEBBo9

And also on this setting even lower bitrate FLACs (192kHz) are still being played in 48kHz according to the info on my AVR menus), so something is definitely not right and I should be using some other setting, probably WASAPI).





Basically only with the last setting dsd files are playing, but even if I decide I can live with the error message pop up on every track - am I playing the dsd files properly? Is there no downsampling happening? As mentioned above in my receiver setup - general - information - audio - it says the stereo, PCM, 48 kHz, so that can not be fine.
Should I aim for direct DSD playback from my AVR? Is it OK for the output to be converted to PCM by foobar? If yes - shouldn't it at least have a higher sample rate, probably the same as the source DSD?

Any help would be greatly appreciated.


EDIT:
OK, I manged to get rid of the annoying FIR error message with the last setting by changing the DSD2PCM to Direct (64fp...). But other than that I still believe DSD file playback is not proper and there is some downsampling happening.


----------



## gimmeheadroom

I think foobar should be set not to convert anything if the device is supposed to support DSD. Can you get it to work first with USB? why are you using custom FIR? I think you can take defaults for everything.

And you have multiple drivers in the list for WSAPI for that device.. try them all. I don't know why so many are listed.. usually I see one for ASIO, one for WASAPI etc.


----------



## 514077

sakujou said:


> Hey everyone,
> 
> I am having a really hard time configuring foobar to play DSD files through my AVR. Read tons of stuff online, foobar for dummies, installed plagins, etc.
> 
> ...


Since you chose to post images instead of describing what your settings are, I'm OUT!


----------



## MelonHead (Aug 12, 2018)

sakujou said:


> Hey everyone,
> 
> I am having a really hard time configuring foobar to play DSD files through my AVR. Read tons of stuff online, foobar for dummies, installed plagins, etc.
> 
> ...


I'm not an expert by  any means, but my FB 2000 plays DSD flawlessly with my Audio GD R2R-11
The first thing which caught my eyes is the HDMI connection. I'm not sure whether your AV receiver can accept DSD or your PC can stream DSD over HDMI .
If you send me a PM, I can share my full setup with you. My installation is portable.

Edit: Yepp, the problem seems to be here. The PC can't  output DSD over HDMI. 
https://yabb.jriver.com/interact/index.php?topic=96197.0


----------



## sakujou (Aug 13, 2018)

Yeah it turns out the PC GPU and Nvidia are the culprit - currently no video cards can output DSD through HDMI.
Thanks a ton for the help! Saved me countless hours of digging online and messing around with settings.

I posted these questions on another forum, but I hope someone here can help also. So here it goes...

So are there any workarounds besides buying a dedicated DAC with a USB input and RCA output?

I understand that the Denon X3300 can accept DSD through USB as well, but there are some mentions a PC can't be connected directly, it has to be a USB stick or hardrive. Maybe with a LAN cable? I will dig online about some info, but wouldn't mind some help if someone is aware how it can be done. 


Also - can foobar or Jriver convert those DSDs to PCM and downsample them to 192kHz (which is my AVR limit)?
I tried setting up foobar to output PCM only and use the Sox (mod2) DSP resampler to resample everything to 192kHz, but it doesn't seem to do anything.
When I try these settings:

Output data format: 24-bit
SACD - Output Mode - PCM
PCM Samplerate - 44100
DCD2PCM Mode - any of Multistage or Direct options

Foobar plays the DSD files, I guess they are converted to PCM 44100. But when I try to setup a higher samplerate of 88200 or 176400 I get this error:
"Unrecoverable playback error: Unsupported stream format: 176400 Hz / 24-bit / 2 channels"

I guess the bit-rate and samplerate do not work together.
But basically how can I at least get the max out of the PCMs? Like 176400 or something similar?
Again I would really appreciate help on this. 


And another general question - even though my AVR says it reads DSD (I believe DSD128), it also has a limit of 192kHz I think. How would this be possible, when DSD 128 is way higher with 5,6mHz? Same goes for some USB DACs I guess, as I have seen a lot of them that say they support DSD 64, 128, even 256 or 512, but yet they have a max limit of either 192kHz or 384kHz. I guess I am missing something, but just wanna make sure if I invest in a separate DAC it would at least do the job.

Huge thanks to everyone again. I really appreciate it.

EDIT:
OK, managed to solve one of the problems - when trying to resample the PCM files, I hadn't specified a sample rate for Sox mod2 to resample to 192kHz.
So first in the SACD settings I set the output to 352800 and then I put the same value in the Sox mod2 "Resample ONLY frequencies" field. And it works - my AVR is now receiver is now playing 192kHz PCM! A small win. 
	

	
	
		
		

		
		
	


	



But the other questions stand, I guess. Even though I wonder is all the trouble worth it. And would I even be able to tell the difference between a bit-perfect DSD and the resample 192kHz PCM I am playing...


----------



## PleasantSounds

sakujou said:


> [..]
> And another general question - even though my AVR says it reads DSD (I believe DSD128), it also has a limit of 192kHz I think. How would this be possible, when DSD 128 is way higher with 5,6mHz? Same goes for some USB DACs I guess, as I have seen a lot of them that say they support DSD 64, 128, even 256 or 512, but yet they have a max limit of either 192kHz or 384kHz. I guess I am missing something, but just wanna make sure if I invest in a separate DAC it would at least do the job.
> [..]



The difference is that PCM typically uses 16 or 24-bit samples, while DSD is 1-bit. So if you multiply 24*192k you get pretty close to the DSD 64 bit rate (both about 5Mbps).

In my setup I have archived the original DSD files and most of the time use their versions converted to PCM 24/96 or even 24/48. If there is a quality difference, my ageing ears are not able to detect it. But in practice I prefer to be able to use replay gain, or if I feel like it turn on some DSP - something that if it is possible to do with DSD at all, so far I haven't fugured out how to.


----------



## sakujou

I see. Thank you for the information and help.

I actually had everything setup correctly, I was using WASAPI (event) with my AVR, converting to PCM than downsampling using sox Mod2 resampler to 192kHz. My receiver was recognizing the signal as 192kHz PCM and sound was awesome.
Then my Windows 10 updated, or I do not what happened and now everything is messed up again. :/
When using the previous settings which worked for 3 days flawlessly, now I get this error message when trying to playback a song.

"Unrecoverable playback error: Device invalidated"

In the windows setting I have allowed apps to take exclusive control for sound, I read a lot online, I was resetting some audio stuff through command prop, reinstalled foobar, tried every possible advice I found online with no luck. 
Basically I can no longer play files through foobar using WASAPI output.
When I switch to DS output it works, files play without issues, but I guess this is not the way to go if I want good quality and/or bit-perfect conversion to PCM.

Again - advice and help would be most welcome and greatly appreciated!


----------



## gimmeheadroom

I can't unravel everything you have said but it seems like you did some stuff that doesn't make sense to me. I am far, far from an expert in Windows so I could be totally wrong.

Anyway maybe it's a good idea to uninstall the driver and install it again. With Windows the technique seems to be to just keep uninstalling and reinstalling anything that doesn't work (including Windows itself.) I hate Windows but there's no other choice for home audio.

I hate Linux also, don't get me wrong. But I'm smart enough not to keep trying to hurt myself using Linux for audio


----------



## 514077

sakujou said:


> I see. Thank you for the information and help.
> 
> I actually had everything setup correctly, I was using WASAPI (event) with my AVR, converting to PCM than downsampling using sox Mod2 resampler to 192kHz. My receiver was recognizing the signal as 192kHz PCM and sound was awesome.
> Then my Windows 10 updated, or I do not what happened and now everything is messed up again. :/
> ...


Are you using V1.4  for FB?  I'm beginning to wonder if it's unstable for DSD period.  I've noticed that DSD will work for a short time, then disfunction.  I've tried both the 1.0.11 and 1.1.0  plugins alternatively and neither of them work for long with the new release of FB.
Thoughts?


----------



## Meelis

In 1.4 the DSF file will not display album art. The mp3tag shows the album art and it's id3v2.4
Foobar2000 can read other tag info, but not album art. When inserting album art in Foobar2000, it gives error
"Attached picture operations not supported for this file format"


----------



## Michgelsen

Meelis said:


> In 1.4 the DSF file will not display album art. The mp3tag shows the album art and it's id3v2.4
> Foobar2000 can read other tag info, but not album art. When inserting album art in Foobar2000, it gives error
> "Attached picture operations not supported for this file format"


Maybe you can convert the DSF file to Wavpack. Wavpack also supports DSD nowadays, and the conversion is lossless. Foobar will probably be able to read the album art from the Wavpack file.


----------



## 514077

Michgelsen said:


> Maybe you can convert the DSF file to Wavpack. Wavpack also supports DSD nowadays, and the conversion is lossless. Foobar will probably be able to read the album art from the Wavpack file.


Do you know how to compress DSF to Wavpack?  I tried it without success.  I use W7-64 bit.


----------



## Michgelsen (Sep 16, 2018)

Yes, I compressed my whole DSF library not too long ago. It works really well, and saves a lot of space. The easiest is to use the command line. Here is the wavpack manual: http://www.wavpack.com/wavpack_doc.html As you can see, the wavpack encoder supports dsf files as input.

If you have the latest wavpack software, the easiest is to copy wavpack.exe (the encoder) to the same folder as the files you want to compress. Then, in Windows Explorer, shift+rightclick in that folder in empty space, and choose 'open powershell window here' or 'open command prompt here'. Then type: wavpack.exe -h yoursong.dsf
The -h activates the high compression mode for DSD. If you have any spaces in the filename of your song, you should put its filename between " ". For example: wavpack.exe -h "your song.dsf"
Then it will compress it, and foobar will play it just like any other DSD file.

In case you use Powershell instead of the old skool command prompt, use ./wavpack.exe -h "your song.dsf"
Powershell needs the ./ in front of a .exe to work properly.


----------



## 514077

Michgelsen said:


> Yes, I compressed my whole DSF library not too long ago. It works really well, and saves a lot of space. The easiest is to use the command line. Here is the wavpack manual: http://www.wavpack.com/wavpack_doc.html As you can see, the wavpack encoder supports dsf files as input.
> 
> If you have the latest wavpack software, the easiest is to copy wavpack.exe (the encoder) to the same folder as the files you want to compress. Then, in Windows Explorer, shift+rightclick in that folder in empty space, and choose 'open powershell window here' or 'open command prompt here'. Then type: wavpack.exe -h yoursong.dsf
> The -h activates the high compression mode for DSD. If you have any spaces in the filename of your song, you should put its filename between " ". For example: wavpack.exe -h "your song.dsf"
> ...


Thanks.  I didn't have detailed instructions like that before.  I'll give it a go, and hope for the best.


----------



## Meelis

The wavpack looks interesting, thanks for suggestion.


----------



## 514077

Meelis said:


> The wavpack looks interesting, thanks for suggestion.


For me, I think it's going to be a bit of a learning curve.  I'm not great with command-line stuff, trying to remember what exactly to type.  But, I need something new to master.  Is there a commandline for dummies somewhere?  It seems that some people are more comfortable with it.


----------



## Michgelsen (Sep 17, 2018)

Well, one further tip I can give you, is that you can use the up arrow and the tab key to save a lot of typing time. With the up arrow, you can cycle through commands you typed in previously, and with tab you can autocomplete filenames and commands. Especially tab is very useful when you're compressing music files, which in my case always have long filenames with spaces and capitals in it. Tab will even automatically add the " " around the filename if needed.

Here you can find some basic commands: https://www.makeuseof.com/tag/a-beginners-guide-to-the-windows-command-line/
Navigating through folders is something you should learn first and is very easy. Use the cd (change directory) and dir (directory; show contents of a directory) commands for that. Note that you don't need cd for changing the drive letter. Just type D:\ or C:\ and you'll change drive.
Almost all command line programs have help functions built in. Usually you can see them by typing, for example: wavpack.exe --help , or wavpack.exe /? . Then you'll see how a command line program can be used, and which options there are.


----------



## 514077

Michgelsen said:


> Well, one further tip I can give you, is that you can use the up arrow and the tab key to save a lot of typing time. With the up arrow, you can cycle through commands you typed in previously, and with tab you can autocomplete filenames and commands. Especially tab is very useful when you're compressing music files, which in my case always have long filenames with spaces and capitals in it. Tab will even automatically add the " " around the filename if needed.
> 
> Here you can find some basic commands: https://www.makeuseof.com/tag/a-beginners-guide-to-the-windows-command-line/
> Navigating through folders is something you should learn first and is very easy. Use the cd (change directory) and dir (directory; show contents of a directory) commands for that. Note that you don't need cd for changing the drive letter. Just type D:\ or C:\ and you'll change drive.
> Almost all command line programs have help functions built in. Usually you can see them by typing, for example: wavpack.exe --help , or wavpack.exe /? . Then you'll see how a command line program can be used, and which options there are.


Thanks!  This is going to be a big help.  Thanks for taking the time.


----------



## 514077 (Sep 17, 2018)

Actually, I found a little program called Wavpack front end, which seems to help with a graphic interface with checkboxes.  I think I just compressed three DSF files.  I'll use it that way for now, although I'll persue some of this commandline operation.  It'll give me something new to work on.I guess I'm not sure what I'm doing.  I opened a folder with an album of DSF files, in which I copied the WavPack.exe file, using the rightclick open in command prompt.  I then typed hh for highest compression, which is (not a proper command).
I'm going to have to study some more.  I was foolish enough to think just clicking on the WavPack.exe would start something like a command window.  I'm not giving up, but I have to walk away for a while.


----------



## Michgelsen

The -hh switch doesn't work for DSD, it is the same as the -h switch. If you want to compress a whole folder, you should type: wavpack.exe -h *.dsf


----------



## Michgelsen

Today I found out that my wavpack DSD files were actually played back as 352kHz PCM files, while I remembered them working fine as DSD earlier (earlier being foobar 1.3.x). As it turns out, something has changed in foobar 1.4 causing this, but it can be easily fixed. You need to go to Preferences > Playback > Decoding, and then drag foo_input_sacd into a higher position than the WavPack encoder. Then WavPack can output DSD again.


----------



## 514077

Michgelsen said:


> Today I found out that my wavpack DSD files were actually played back as 352kHz PCM files, while I remembered them working fine as DSD earlier (earlier being foobar 1.3.x). As it turns out, something has changed in foobar 1.4 causing this, but it can be easily fixed. You need to go to Preferences > Playback > Decoding, and then drag foo_input_sacd into a higher position than the WavPack encoder. Then WavPack can output DSD again.


Thanks, I'll check that out.  The status bar says DSD(XXX) and the 2.8 or 5.6 bitrate.  But,, I'll have a look at what my DAC is seeing to make sure.  On the bright side, I've reclaimed about 100GB of drivespace so far, and I'm only into the 'B's of my library.


----------



## Michgelsen

Yes my status bar said that too, but I found out that my DAC was saying 352.


----------



## 514077 (Sep 21, 2018)

Michgelsen said:


> Yes my status bar said that too, but I found out that my DAC was saying 352.


Oh damn.  I'd better take a  look. Just checked.  Foo_input)SACD is at the top of the list.  Maybe due to my buggering around with swapping 1.1.0 with 1.0.11.  With FB1.4, I finally stuck with 1.1.0, even though I get that highspeed tracktiming on the statusbar, resulting in only a few seconds of play.  Seems to be resolved by a comp restart.  Compression/conversionis going quite well.


----------



## Meelis

Tested the WavPack. I was able to reduce 16GB of 352,8 kHz 32bit wav files into 10GB with slowest encoding settings.
Not bad.


----------



## Milck

Quick question but what themes are people using? Currently on Fusion Beta but thinking of changing it up a bit


----------



## 514077

Meelis said:


> Tested the WavPack. I was able to reduce 16GB of 352,8 kHz 32bit wav files into 10GB with slowest encoding settings.
> Not bad.


Was that originally a .wav or a .flac file?  I have a couple of 352.8  albums in .flac which might be worth doing as well.


----------



## Meelis

UELong said:


> Was that originally a .wav or a .flac file?  I have a couple of 352.8  albums in .flac which might be worth doing as well.


Original was DSD256 converted to WAV 32bit 352.8 kHz. But maybe DSD compresses better.

BTW this compression was insanely slow, ~ 1,5 hours for 2 albums.


----------



## 514077

Meelis said:


> Original was DSD256 converted to WAV 32bit 352.8 kHz. But maybe DSD compresses better.
> 
> BTW this compression was insanely slow, ~ 1,5 hours for 2 albums.


Wow!  I was able to compress about 300 files in about a couple of hours from various rates of DSD.  Maybe the 352.8 takes longer?  I'll have to try it.  I started a batch of 401 DSDs about an hour ago, and right now I'm about 18% complete.  Getting a  Lot of harddrive space back.


----------



## Meelis

UELong said:


> Wow!  I was able to compress about 300 files in about a couple of hours from various rates of DSD.  Maybe the 352.8 takes longer?  I'll have to try it.  I started a batch of 401 DSDs about an hour ago, and right now I'm about 18% complete.  Getting a  Lot of harddrive space back.


I used extra high compression mode with level 6, should have used default values.


----------



## 514077

Meelis said:


> I used extra high compression mode with level 6, should have used default values.


I used the highest for the DSD files.  So far I got 225GB space back.


----------



## Meelis

UELong said:


> I used the highest for the DSD files.  So far I got 225GB space back.


extra encode processing (optional n = 1-6, 1 = default)
The highest levels (n = 4-6) are extremely slow but can provide significant improvement in special situations (i.e. synthesized sounds).
This option is not applicable to DSD audio and is simply ignored.

I should not have used that option for the type of music i was encoding, no compression benefit, but very slow encode for PCM.
Should have read the documentation first.


----------



## Michgelsen

UELong said:


> I have a couple of 352.8  albums in .flac which might be worth doing as well.


There's no point. While wavpack may offer slightly more compression on the highest settings, the difference between flac and wavpack will be tiny, so it's not worth the hassle.




Milck said:


> Quick question but what themes are people using? Currently on Fusion Beta but thinking of changing it up a bit


I've been using the Columns UI plugin for years now. It's highly customizable but may take some time setting it up the way you like it.​


----------



## 514077

Michgelsen said:


> There's no point. While wavpack may offer slightly more compression on the highest settings, the difference between flac and wavpack will be tiny, so it's not worth the hassle.
> 
> 
> I've been using the Columns UI plugin for years now. It's highly customizable but may take some time setting it up the way you like it.​


On a similar note:  I used to use DFF2DSF in windows 7, developed, I think, by DSD master Ted B.  Have you ever tried it?  I ask because it stopped working, disappeared from my 'send-to' menu.  Just wondered if you had any thoughts?


----------



## Michgelsen (Sep 24, 2018)

I've always used ISO2DSF, a freeware utility which functions as a wrapper for sacd_extract.exe and DFF2DSF, so that it can transfer the tags from the sacd ISO to DSF. DFF doesn't support tagging as far as I know.
ISO2DSF used to be available for download over at computeraudiophile.com but I can't seem to find it anymore. I have attached it to this post. It also contains a manual, but it doesn't contain the context menu installer anymore because I didn't use that. I just drag and drop an ISO onto ISO2DSF.exe and then it works.

Edit: how do attachments even work on this forum nowadays? I can't see my attachment, but when editing this post it says there is. Anyway, shoot me a PM if you would like to have a copy of ISO2DSF.


----------



## tintinsnowydog

Just got the gustard x20 pro dac and singer su-1 usb bridge, so am totally new to proper computer audio. I want to try out the higher sample rates to see if I can hear a difference for myself. I have 2 questions:

1. is there a way to stream tidal through foobar
2. if so, is there any way to up-sample this to dsd256/512. currently using no music software at all gets tidal up to PCM 384kHz. 
3. if not, is there any other program out there that can do the above, information is very sparse about this, especially as I have no clue what i'm doing  If anything i typed above is nonsense please feel free to educate me as well!


----------



## gimmeheadroom

tintinsnowydog said:


> Just got the gustard x20 pro dac and singer su-1 usb bridge, so am totally new to proper computer audio. I want to try out the higher sample rates to see if I can hear a difference for myself. I have 2 questions:
> 
> 1. is there a way to stream tidal through foobar
> 2. if so, is there any way to up-sample this to dsd256/512. currently using no music software at all gets tidal up to PCM 384kHz.
> 3. if not, is there any other program out there that can do the above, information is very sparse about this, especially as I have no clue what i'm doing  If anything i typed above is nonsense please feel free to educate me as well!



Hi, Tidal has its own Windows app and that is the best way to get the best quality if you have Tidal hifi. But for best results you need an MQA DAC. I don't know about yours but if the gustard supports it, it will show up on the Tidal app as one of the output devices you can select as long as you installed any drivers necessary for the gustard. Select it in your preferences and select master quality in prefs and that is all you should need.

Upsampling does not improve quality.


----------



## GumbyDammit223

Dumb question from an idiot noob:
   Can I pipe my FLACs through Foobar, through my hardwired network, to my home theater receiver?  I have a Denon AV-4k receiver connected to my router which sees my computer and my computer sees it.  The problem is the receiver only sees a small fraction of my music library and I would like to sit at my computer and direct a playlist from that library to play on my big speakers instead of my headphones at my computer.  Is this possible?  The computer is running Win7 if that matters.


----------



## gimmeheadroom

GumbyDammit223 said:


> Dumb question from an idiot noob:
> Can I pipe my FLACs through Foobar, through my hardwired network, to my home theater receiver?  I have a Denon AV-4k receiver connected to my router which sees my computer and my computer sees it.  The problem is the receiver only sees a small fraction of my music library and I would like to sit at my computer and direct a playlist from that library to play on my big speakers instead of my headphones at my computer.  Is this possible?  The computer is running Win7 if that matters.



Sounds reasonable. You just have to connect your Denon to your Windows box via USB, optical, whatever you have. Then select that device in Windows sound manager or whatever it's called, and foobar will play through it. I can't remember if you can select output devices directly from foobar but maybe you can.


----------



## PleasantSounds

GumbyDammit223 said:


> Dumb question from an idiot noob:
> Can I pipe my FLACs through Foobar, through my hardwired network, to my home theater receiver?  I have a Denon AV-4k receiver connected to my router which sees my computer and my computer sees it.  The problem is the receiver only sees a small fraction of my music library and I would like to sit at my computer and direct a playlist from that library to play on my big speakers instead of my headphones at my computer.  Is this possible?  The computer is running Win7 if that matters.



I think this component may help you.


----------



## 514077

gimmeheadroom said:


> Sounds reasonable. You just have to connect your Denon to your Windows box via USB, optical, whatever you have. Then select that device in Windows sound manager or whatever it's called, and foobar will play through it. I can't remember if you can select output devices directly from foobar but maybe you can.


File menu>Preferences>Output>choices for output.


----------



## GumbyDammit223

gimmeheadroom said:


> Sounds reasonable. You just have to connect your Denon to your Windows box via USB, optical, whatever you have. Then select that device in Windows sound manager or whatever it's called, and foobar will play through it. I can't remember if you can select output devices directly from foobar but maybe you can.


I want to stream it through the network, NOT a direct connection.  Also, the computer does not see the Denon as an audio device, just a generic "device".



PleasantSounds said:


> I think this component may help you.


Ooooh!  This might do it!!!  Now I have to wait to go home to try it. :\



UELong said:


> File menu>Preferences>Output>choices for output.


Yeah, that's the obvious place to look, but I only see my computer speakers and my Schiit Bifrost.


----------



## timb5881

Ok, so I have had Foobar2000 installed for a long time.  I have it set to play DSD, ISO, dsf files.  Recently it has started going like 4x through the file, plays speed correctly, but scroll goes through a song, keeps playing that song but list it as the 2n, then to the 3rd.  After it goes through all the files, it has actually only 1 or 2 songs, then stops.  All the files play correctly through other players with no issues, only on Foobar.  Any ides or suggestions?


----------



## 514077

timb5881 said:


> Ok, so I have had Foobar2000 installed for a long time.  I have it set to play DSD, ISO, dsf files.  Recently it has started going like 4x through the file, plays speed correctly, but scroll goes through a song, keeps playing that song but list it as the 2n, then to the 3rd.  After it goes through all the files, it has actually only 1 or 2 songs, then stops.  All the files play correctly through other players with no issues, only on Foobar.  Any ides or suggestions?


Which versions of FB and plugins?


----------



## timb5881

UELong said:


> Which versions of FB and plugins?


Foobar  ver 1.3.17
ASIO support ver 2.1.2
DSD processor ver 1.1.0
DSDIFF ver 1.6
Super Audio cd decoder 1.10


----------



## 514077

timb5881 said:


> Foobar  ver 1.3.17
> ASIO support ver 2.1.2
> DSD processor ver 1.1.0
> DSDIFF ver 1.6
> Super Audio cd decoder 1.10


I believe the v1.1.0 DSD plugins are for, or work better with Foobar V1.4.  Update to that version first and see if that helps.
Or lower your DSD versions to 1.0.11.


----------



## timb5881

I ended up going to a previous version of Super Audio CD decoder.  Now it works fine again.


----------



## 514077

timb5881 said:


> I ended up going to a previous version of Super Audio CD decoder.  Now it works fine again.


I found the same thing.  I was juggling around with FB and SACD plugins for a while.  I'm sure that 1.1.0  is meant for the 1.4 version of FB2K.  There were some other things I liked to add, which only worked if I advanced the FB version.  As long as it works for you, that's good.


----------



## adlevision

Ok, so I've used Foobar since Jesus was a little boy and have used the in_SACD plugin for several years without a problem.  All of a sudden, my DACs all refuse to play in DSD mode.  

Tried two computers with Foobar 1.4 and 1.4.1, In_SACD 1.10, WASAPI 3.3.  I don't even think I'd upgraded anything since it worked last.  Three different DACs, so I think it's software rather than hardware.  

Perhaps I've changed a buffer setting someplace, in which case I need reminding of anything in the Output or Advanced sections that are finicky for DSD playback.  

I'm using the Windows native class-II USB audio driver, so maybe Windows did an update that broke it lately (I've run version 1803 since maybe May).
  I've also noticed that the SourceForge page for the SACD plugin now has an older file, not the usual in_SACD package.  What's up there now is older than In-SACD 1.10.  So, has something broken?  I can't find evidence of anyone else having this problem.

Thanks in advance.


----------



## 514077

adlevision said:


> Ok, so I've used Foobar since Jesus was a little boy and have used the in_SACD plugin for several years without a problem.  All of a sudden, my DACs all refuse to play in DSD mode.
> 
> Tried two computers with Foobar 1.4 and 1.4.1, In_SACD 1.10, WASAPI 3.3.  I don't even think I'd upgraded anything since it worked last.  Three different DACs, so I think it's software rather than hardware.
> 
> ...


It's still working for me.  I don't remember, but did you try asio over wasapi?


----------



## adlevision

It just occurred to me what I had changed: adding in the graphic equalizer DSP plugin to cope with my HD800s.  That was the culprit.  I tried lowering the overall volume in the EQ in case it was a clipping issue, but no joy.  Anybody have a workaround, or does ASIO not have the issue with the DSP plugin and in_SACD?
I can obviously switch to PCM mode when using the HD800s, but would hope for something else.  It's not that DSD is very important at all, but downsampling always sounds worse than whatever the original format was.
Man, you talk about scary: the TEAC UD-501 has relays or something that click mechanically whenever the sampling rate changes.  Maybe it's switching from 44.1 to 48kHz clocks.  Whatever it is, it was clicking rapidfire for as long as I cared to let it attempt to play DSD while that EQ plugin was in the chain.  No way it could have survived much more of that, judging from the sound it was making.
Also, while we're on yet another thread together Governor Long, have you found any parametric EQ plugins that are accessible with screen readers?  I've never come across a VST or AU plugin that presented standard UI elements to a screen reader on Mac or PC.  Or do you just put on a different pair of cans?

Thanks.


----------



## PleasantSounds

I don't think you will be able to use equalizer or any other DSP with the DSD sources. They only work with PCM.


----------



## HiFiRebel (Nov 16, 2018)

Is there a way to save the layout so that I don't have to set everything from scratch every time my computer crashes and Foobar resets to defaults?


----------



## adlevision

HiFiRebel said:


> Is there a way to save the layout so that I don't have to set everything from scratch every time my computer crashes and Foobar resets to defaults?





HiFiRebel said:


> Is there a way to save the layout so that I don't have to set everything from scratch every time my computer crashes and Foobar resets to defaults?


Sure is.  Assuming you installed Foobar with default options rather than "portable" install, then go to your user folder > appdata > roaming > foobar2000 and make a copy of the "configuration" folder.  That will back up just about all your settings.  To get more fine-grained, make backup copies of the relevant files in that folder.  Core.cfg contains most of it, such as DSP chain and current output device.  I seem to recall a separate UI cfg file that would contain layout.

For many years, users like me have relied on writing simple batch files to swap files in this directory in and out to act as different preference profiles.


----------



## HiFiRebel

adlevision said:


> Sure is.  Assuming you installed Foobar with default options rather than "portable" install, then go to your user folder > appdata > roaming > foobar2000 and make a copy of the "configuration" folder.  That will back up just about all your settings.  To get more fine-grained, make backup copies of the relevant files in that folder.  Core.cfg contains most of it, such as DSP chain and current output device.  I seem to recall a separate UI cfg file that would contain layout.
> 
> For many years, users like me have relied on writing simple batch files to swap files in this directory in and out to act as different preference profiles.


Thank you! I just backed up the whole folder.


----------



## adlevision

HiFiRebel said:


> Thank you! I just backed up the whole folder.


If that doesn't do it, I just noticed the "theme.fth" file in the C:\users\[yourname]\appdata\roaming\foobar2000 folder that might contain much of what you're looking for.  The UI config file is foo_ui_std.dll.cfg, looks like.  I suggest messing with your layout, restoring your backup files, and seeing if you've caught the right files.


----------



## 514077

adlevision said:


> Sure is.  Assuming you installed Foobar with default options rather than "portable" install, then go to your user folder > appdata > roaming > foobar2000 and make a copy of the "configuration" folder.  That will back up just about all your settings.  To get more fine-grained, make backup copies of the relevant files in that folder.  Core.cfg contains most of it, such as DSP chain and current output device.  I seem to recall a separate UI cfg file that would contain layout.
> 
> For many years, users like me have relied on writing simple batch files to swap files in this directory in and out to act as different preference profiles.


Thanks; I never even thought of doing that.


----------



## PleasantSounds

adlevision said:


> Sure is.  Assuming you installed Foobar with default options rather than "portable" install, then go to your user folder > appdata > roaming > foobar2000 and make a copy of the "configuration" folder.  That will back up just about all your settings.  To get more fine-grained, make backup copies of the relevant files in that folder.  Core.cfg contains most of it, such as DSP chain and current output device.  I seem to recall a separate UI cfg file that would contain layout.
> 
> For many years, users like me have relied on writing simple batch files to swap files in this directory in and out to act as different preference profiles.



Other users though have been using the foo_jesus component which backs up all the relevant files and maintains a configurable number of backup copies, all neatly zipped for minimum footprint....


----------



## adlevision

PleasantSounds said:


> Other users though have been using the foo_jesus component which backs up all the relevant files and maintains a configurable number of backup copies, all neatly zipped for minimum footprint....


Oh yeah!  I used this several years ago, but forgot the name and couldn't turn anything up on Google when I searched recently.  I stopped using this for some reason, but will look at it again.


----------



## adlevision (Nov 17, 2018)

Woe, so the simplest answer to HiFiRebel's question is actually already built into Foobar, turns out.  The "secret" file menu has a Save Configuration option.  If you're not familiar with the secret menus, shift+alt+f brings up the file menu (the same with "p" brings up the secret playback menu with output device options).  I think control-click works, too.  But does anybody know where Foobar actually puts the saved config?  I'm not finding it.  Just curious, since I also have Foo_Jesus going as of 10 minutes ago...


----------



## 514077

I have a question about building in hotkeys.  Does anyone know if Peter Poloski is open to suggestions?  
In VLC player, one can use the numpad keys for volume, but mainly quick movement through a file.  While the #4  and #6 keys move back and forth at about 10-second intervals, holding the 0 'Insert' key can move along at a larger rate-I believe a minute.
Since I prefer to use FB for my listening, it would be convenient if that key function could be added.
Thoughts?


----------



## adlevision

UELong said:


> I have a question about building in hotkeys.  Does anyone know if Peter Poloski is open to suggestions?
> In VLC player, one can use the numpad keys for volume, but mainly quick movement through a file.  While the #4  and #6 keys move back and forth at about 10-second intervals, holding the 0 'Insert' key can move along at a larger rate-I believe a minute.
> Since I prefer to use FB for my listening, it would be convenient if that key function could be added.
> Thoughts?


I've never known Peter to be open to anything, and there have always been issues with pop-up windows not receiving focus for screen readers, for instance, which I've emailed him about, plus several users have asked him for SMB access in Foobar Mobile without result.  Still my favorite program ever, though.  
Foobar hasn't really changed in over a decade.  But the good news is that keyboard hotkeys are already immensely configurable.  Just go to Preferences > Keyboard Shortcuts and hit "add."  A new entry will appear in the list of shortcuts and will automatically have focus.  Tab to the list of functions and find what you want.  In the case you mention, it's under "Seeking," and you have several options.  Simply cursor down to what you want and then hit Tab (not enter or space).  You'll be in the hotkey field.  After that, you can hit tab and then back-tab to hear if the key you entered was accepted.  Tab a couple more times to check the box whether it's global or only when the Foobar window is active, then tab over to "Apply."  You're now ready to add another one.  I'm giving screen reader-friendly directions here, obviously.
    My own numpad includes the top row for selecting audio output devices, 7 and 9 for seeking back 5 seconds or forward 10 seconds, 4 and 6 for prev/next, 8 and 2 for volume, and I'd have to look at the rest.  In fact, the volume keys actually are tied to EventGhost running in the background, which IR blasts volume commands directly to my amp through a USB-UIRT, another decade-old piece of equipment still working fine.


----------



## HiFiRebel

Thank you! I used the secret menu to save the configuration. Also installed the foo_jesus component, just to be on the safe side


----------



## PleasantSounds

One hint for all those who have just installed the foo_jesus component: if you want your playlists backed up (and I stringly recommend that), then go to 
Preferences->Advanced->Autosave & Autobackup and make sure that your playlists folder name is CORRECTLY listed in the Files and directories to back up section.
This component is several foobar updates behind and the playlists folder name has been changed since, so the default settings won't catch it. 
For the current f2k version the playlists folder is  playlists-v1.4


----------



## PleasantSounds

UELong said:


> I have a question about building in hotkeys.  Does anyone know if Peter Poloski is open to suggestions?
> In VLC player, one can use the numpad keys for volume, but mainly quick movement through a file.  While the #4  and #6 keys move back and forth at about 10-second intervals, holding the 0 'Insert' key can move along at a larger rate-I believe a minute.
> Since I prefer to use FB for my listening, it would be convenient if that key function could be added.
> Thoughts?



So the seekbar  doesn't cut it for you? You can click on any point on the track timeline and it takes the playback there.
There are also enhanced versions of the seekbar which display the track's waveform and control the play position through the mouse wheel - check the Waveform minibar (mod) if interested.


----------



## 514077

PleasantSounds said:


> So the seekbar  doesn't cut it for you? You can click on any point on the track timeline and it takes the playback there.
> There are also enhanced versions of the seekbar which display the track's waveform and control the play position through the mouse wheel - check the Waveform minibar (mod) if interested.


I actually installed the seekbox component.  One just has to convert minutes and hours into seconds, which can get tricky after a while.  But it works.
It would just  be quicker to do it with the NumPad.  Attlevision has given me a method of adding shortcut keys, which are extremely useful to blind guys.  I like the idea of a quick key for switching outputs.


----------



## 514077

adlevision said:


> I've never known Peter to be open to anything, and there have always been issues with pop-up windows not receiving focus for screen readers, for instance, which I've emailed him about, plus several users have asked him for SMB access in Foobar Mobile without result.  Still my favorite program ever, though.
> Foobar hasn't really changed in over a decade.  But the good news is that keyboard hotkeys are already immensely configurable.  Just go to Preferences > Keyboard Shortcuts and hit "add."  A new entry will appear in the list of shortcuts and will automatically have focus.  Tab to the list of functions and find what you want.  In the case you mention, it's under "Seeking," and you have several options.  Simply cursor down to what you want and then hit Tab (not enter or space).  You'll be in the hotkey field.  After that, you can hit tab and then back-tab to hear if the key you entered was accepted.  Tab a couple more times to check the box whether it's global or only when the Foobar window is active, then tab over to "Apply."  You're now ready to add another one.  I'm giving screen reader-friendly directions here, obviously.
> My own numpad includes the top row for selecting audio output devices, 7 and 9 for seeking back 5 seconds or forward 10 seconds, 4 and 6 for prev/next, 8 and 2 for volume, and I'd have to look at the rest.  In fact, the volume keys actually are tied to EventGhost running in the background, which IR blasts volume commands directly to my amp through a USB-UIRT, another decade-old piece of equipment still working fine.


Thanks.  It worked.  Fantastic!  I ended up using 7 and 9 for soundcard speakers and the Foo_dop to native.  2 and 8 are volume, 4 and 6 are 10-second jumpps and the same with insert are 1-minute.  This has been bugging me in the background of my mind for years.
One thing:  If I was to have checked the global box, would that overridden keystrokes outside the FB program?  I'm not sure what happens.
But thanks for your clear instructions.
Kevin


----------



## adlevision

UELong said:


> Thanks.  It worked.  Fantastic!  I ended up using 7 and 9 for soundcard speakers and the Foo_dop to native.  2 and 8 are volume, 4 and 6 are 10-second jumpps and the same with insert are 1-minute.  This has been bugging me in the background of my mind for years.
> One thing:  If I was to have checked the global box, would that overridden keystrokes outside the FB program?  I'm not sure what happens.
> But thanks for your clear instructions.
> Kevin


In GENERAL, YES.  BUT THE TEST IS WHETHER YOU CAN ENTER THE KEY OR IF ANOTHER BACKGROUND PROGRAM LIKE NVDA INTERCEPTS IT WHILE YOU'RE TRYING TO CREATE IT.  i DO GLOBAL ON THE NUMPAD BECAUSE NO OTHER PROGRAM USES IT AND IT ALLOWS playback controls without the window in the foreground.  Setting a global key to "album list" is a great way to quickly dive from writing that Head-Fi post to changing what album is playing, for example.
(my Caps lock key was out of control there for a minute...)
Quasi Off Topic: Another plugin I recommend is MonkeyMote, which lets an IPhone act as a remote control and even stream music directly to the phone from the PC's playlist.  Easier to navigate the library for Voiceover users than any native IPhone app I've found.
Yea! My 15th post!  Now I can start selling my drawer full of gear...


----------



## 514077

adlevision said:


> In GENERAL, YES.  BUT THE TEST IS WHETHER YOU CAN ENTER THE KEY OR IF ANOTHER BACKGROUND PROGRAM LIKE NVDA INTERCEPTS IT WHILE YOU'RE TRYING TO CREATE IT.  i DO GLOBAL ON THE NUMPAD BECAUSE NO OTHER PROGRAM USES IT AND IT ALLOWS playback controls without the window in the foreground.  Setting a global key to "album list" is a great way to quickly dive from writing that Head-Fi post to changing what album is playing, for example.
> (my Caps lock key was out of control there for a minute...)
> Quasi Off Topic: Another plugin I recommend is MonkeyMote, which lets an IPhone act as a remote control and even stream music directly to the phone from the PC's playlist.  Easier to navigate the library for Voiceover users than any native IPhone app I've found.
> Yea! My 15th post!  Now I can start selling my drawer full of gear...


Thanks again for all your help.
Kevin


----------



## timb5881

Ok so DSD dsf  playback is acting weird again.  Pop's or ticks on playback, and also periodically it is running fast.  It is almost like double speed.  Tried 2 different DAC's, not them.


----------



## timb5881

timb5881 said:


> Ok so DSD dsf  playback is acting weird again.  Pop's or ticks on playback, and also periodically it is running fast.  It is almost like double speed.  Tried 2 different DAC's, not them.


Ok so I reinstalled the Super Audio CD decoder and it works fine. Strange


----------



## gimmeheadroom

timb5881 said:


> Ok so I reinstalled the Super Audio CD decoder and it works fine. Strange



This is why technical people love Windows. Doesn't work? Reboot. Doesn't work after rebooting? Reboot again. Doesn't work after 2 or 3 more reboots? Reinstall.

How do you get Windows to understand when you install something you really meant it? And how do you get Microsloth to stop putting out crap repair tools for Outhouse, Orifice365, etc. Don't put out repair tools. Code things so they don't corrupt data and then you won't *need* repair tools. Why isn't this obvious?


----------



## GumbyDammit223

Several weeks ago I had posted in this and a couple other forums asking for help in playing the digital music collection I have on my computer through my home stereo system.  The computer is a Win7 box with foobar 2000 and the heart of my audio system is a Denon AVR-4000 receiver hard-wired to my network.  I had asked if there was a way to push the music to the Denon without having to spend any more money and preferably use foobar.  All the answers received were not helpful or were pushing me to spend more money on hardware that I don't need.  I did find a foobar addon* UPnP/DLNA Renderer, Server, Control Point* _By: bubbleguuum _that was quite buggy and locked up my receiver.  Then, just yesterday I found *UPnP MediaRenderer Output *_By: Peter._  This addon, beside the fact that it is currently being supported, does exactly what I want!  I can set up a playlist containing FLACs running the gamut from 16-bit / 44.1 kHz to 24-bit / 192k and they are played seamlessly through the Denon.  The tag information isn't displayed correctly on the receiver, but so what.  The music is coming through.

This post is closure for me for trying to find an answer to this problem that has been plaguing me for months.  Were it not for suggestions from this forum, I would not even know where to begin looking for an answer.


----------



## gimmeheadroom (Dec 15, 2018)

GumbyDammit223 said:


> Several weeks ago I had posted in this and a couple other forums asking for help in playing the digital music collection I have on my computer through my home stereo system.  The computer is a Win7 box with foobar 2000 and the heart of my audio system is a Denon AVR-4000 receiver hard-wired to my network.  I had asked if there was a way to push the music to the Denon without having to spend any more money and preferably use foobar.  All the answers received were not helpful or were pushing me to spend more money on hardware that I don't need.  I did find a foobar addon* UPnP/DLNA Renderer, Server, Control Point* _By: bubbleguuum _that was quite buggy and locked up my receiver.  Then, just yesterday I found *UPnP MediaRenderer Output *_By: Peter._  This addon, beside the fact that it is currently being supported, does exactly what I want!  I can set up a playlist containing FLACs running the gamut from 16-bit / 44.1 kHz to 24-bit / 192k and they are played seamlessly through the Denon.  The tag information isn't displayed correctly on the receiver, but so what.  The music is coming through.
> 
> This post is closure for me for trying to find an answer to this problem that has been plaguing me for months.  Were it not for suggestions from this forum, I would not even know where to begin looking for an answer.



I think you got some good answers with people trying to help based on what you asked. I said you could run foobar2000 into your Denon via USB or optical. That's about 5 bucks worth of cable, either way, and it is correct. If you're using a dlna server on your windows box it is not pushing anything to your clients (of which the Denon is apparently one). The clients pull it from the server. And there is latency there and various other problems. Cable is the way to go. Network is a compromise when you can't use a cable.


----------



## GumbyDammit223 (Dec 15, 2018)

gimmeheadroom said:


> I think you got some good answers with people trying to help based on what you asked. I said you could run foobar2000 into your Denon via USB or optical. That's about 5 bucks worth of cable, either way, and it is correct. If you're using a dlna server on your windows box it is not pushing anything to your clients (of which the Denon is apparently one). The clients pull it from the server. And there is latency there and various other problems. Cable is the way to go. Network is a compromise when you can't use a cable.


Respectfully, no.  Yes, I _could_ run more cable.  I _could_ run fiber.  I choose not to for multiple reasons which are my own.  The problem is what I stated.  And yes, the client (Denon) _can_ pull from the foobar server. It could pull directly from my hard drive even without foobar. * HOWEVER*, it can not access the full 1.3 TB / 38,500 files I currently have on the hard drive.  Per an email directly from Denon, it can only access one thousand files and apparently only the first thousand that were stored on the hard drive.  Sorry not good enough.  I won't get into semantics, but when I can sit at my computer, queue up a foobar playlist, select the foobar output to be *UPnP: Denon AVR-X4000*, click play, and see the receiver turn on and begin playing the playlist in the order displayed by foobar, even random, I would say the content is being pushed to the receiver.

Maybe we are saying the same thing in different words and I'm too dense to see that.  Oh well.  The end result is my system is finally doing what I wanted it to do after much trial and error and I thought I would pass on my experiences in case somebody might want to do the same thing and were as stupid as me.  Thanks for taking the time to read my post and replying!


----------



## dakanao (Jan 14, 2019)

What is the difference in foobar2000 between Device output buffer length, and Networking buffer size? I want to make foobar a little faster when scrolling forwards or backwards in a song with the ASIO plugin, and also want less lag when using foobar ASIO with Microsoft Edge


----------



## andresrgv

hi people! I want to know is there any way to create playback shortcuts for albums? like same functionality just instead of say next song it would be next album or previous or random, something like that same thing playback just for albums instead of tracks, I wonder because I normally play my all library and if hit random when I want another song but sometimes it would be nice to hit a bottom and go to another album


----------



## gimmeheadroom

dakanao said:


> What is the difference in foobar2000 between Device output buffer length, and Networking buffer size? I want to make foobar a little faster when scrolling forwards or backwards in a song with the ASIO plugin, and also want less lag when using foobar ASIO with Microsoft Edge



I don't know but I guess the device output buffer has to do with how much storage foobar allocates to send to the output device and the networking buffer has to do with how much storage foobar allocates when it is serving dlna. It might be the opposite, how much it allocates for use when it is a dlna or samba client. Just guesses, but you could ask on their forums.


----------



## whohasaquestion (Feb 1, 2019)

Can't adjust volume via (iphone) iOS 12 home screen? 

Btw, possible to have volume bar while in the album view?


----------



## 514077

As a rule, I don't use MP3 for music; rather lo-fi radio or movie or book material.  I've noticed that MP3 seems to be broken in v1.4.2.  Has anyone else noticed this problem?


----------



## 514077

For anyone interested, there's an update for SACD decoder to V1.1.1.
https://sourceforge.net/projects/sa...dater&utm_medium=email&utm_source=subscribers


----------



## gwertheim

I want to hook up foobar through CCA to my DAC. Does the foobar controller app still work well, it hasn't been updated since 2016


----------



## hifinoob005 (Dec 20, 2019)

Setup: Chord Mojo (optical and usb) and foobar.

According to the manual (http://www.chordelectronics.co.uk/wp-content/uploads/2016/09/Mojo-Manual-28072016.pdf) :

_Mojo has 3 digital inputs.
1 x TOSLink optical capable of playing 44.1KHz to 192KHz PCM and DSD64 in DoP format.
1 x 3.5mm COAX SPDIF capable of playing 44.1KHz to 384Khz PCM (768KHz special operation) and DSD64, DSD128 in DoP format.
1x micro USB capable of 44KHz to 768KHz PCM and DSD64, DSD128 and DSD256 in DoP format. 

DSD decodingDSD playback is supported using the DoP Standard 1.0 with 0xFA / 0x05 markers. _

Using
_Mode 1: Bitperfect in this guide_:
https://diyaudioheaven.wordpress.co...-part-3-new-experimental-sacd-plugin-v-0-9-x/

Output _DSD: ASIO : ASIO Chord 1.05._
DSD selected in _Tools>SACD._


With the settings above, playing DSD and DST, the light on the mojo is pink:




 





Why is the light pink?
How to configure correctly for DSD playback?
Is DST considered a DSD format?


----------



## Joe Bloggs

hifinoob005 said:


> Setup: Chord Mojo (optical and usb) and foobar.
> 
> According to the manual (http://www.chordelectronics.co.uk/wp-content/uploads/2016/09/Mojo-Manual-28072016.pdf) :
> 
> ...



Try ticking DoP in Tools>SACD, your Mojo requires it I think

DST is DSD, but your DST is 6 channels which the Mojo won't support directly


----------



## hifinoob005 (Dec 20, 2019)

Joe Bloggs said:


> Try ticking DoP in Tools>SACD, your Mojo requires it I think
> 
> DST is DSD, but your DST is 6 channels which the Mojo won't support directly



Screenshot here:
https://www.head-fi.org/threads/the...-surround-sound.593050/page-270#post-15361622

In the link I inquired on how to use Waves NX for virtualization of multi channel audio.
Not sure if it's possible.

Not sure how I got it to work previously but now whenever I try to play a DSD64/DSD or DST64 with output DSD: ASIO : ASIO Chord 1.05 results in an error message:
_Unrecoverable playback error: Could not start ASIO playback_

I have to select DSD:WASAPI (push/event): Digital Output (Chord Async)...







 

It only works on stereo. If a multi channel file is played:
_Unrecoverable playback error: Unsupported stream format: 176400 Hz / 16-bit / 6 channels
_
The sound comes through but very is a very loud white noise.
While it plays selecting DSD: ASIO : ASIO Chord 1.05 clears up the white noise.
Now multi channel files can be played as well.

If the audio is paused/stopped for about a minute, and the file resumed or started again it results in an error:
_Unrecoverable playback error: Could not start ASIO playback_
Same if foobar is restarted.

Rebooted the PC, same fault.
Non DSD/DST files do not have the loud white noise when DSD:WASAPI is used.

Enabling DoP for Converter in Tools>SACD does not change this behavior.

The light stays pink whenever a DSD/DST file is playing.
From two different angles:


----------



## Joe Bloggs

hifinoob005 said:


> Screenshot here:
> https://www.head-fi.org/threads/the...-surround-sound.593050/page-270#post-15361622
> 
> In the link I inquired on how to use Waves NX for virtualization of multi channel audio.
> ...


Can't really tell if that's pink or white, and there's no reason to think it should be outputting at 768kHz PCM.


----------



## hifinoob005

Joe Bloggs said:


> Can't really tell if that's pink or white, and there's no reason to think it should be outputting at 768kHz PCM.



From a certain angle it's pink, from another only the edges are pink.
Can't tell what kind of pink it is, looks like 384Hz.

What is certain is that it's not white.


----------



## gimmeheadroom

What music player are you using? Trying to play DSD from Linux is gonna be hard in most cases


----------



## hifinoob005 (Dec 23, 2019)

gimmeheadroom said:


> What music player are you using? Trying to play DSD from Linux is gonna be hard in most cases



fobbar2000>Mojo. I've noted the details in a few posts above.
On Win10x64. It's a desktop setup, not portable.


----------



## gimmeheadroom

hifinoob005 said:


> fobbar2000>Mojo. I've noted the details in a few posts above.
> On Win10x64. It's a desktop setup, not portable.



<FACEPALM>

Sorry man, I will have to blame it on the beer. I hope you get it resolved but I have a Mojo and can double check in a week or so. I'm away for the hols right now.


----------



## hifinoob005 (Dec 23, 2019)

gimmeheadroom said:


> <FACEPALM>
> 
> Sorry man, I will have to blame it on the beer. I hope you get it resolved but I have a Mojo and can double check in a week or so. I'm away for the hols right now.



Thanks, that would be great.

I contacted Chord, but I'm still talking to them.

LE: Chord told me the light is white, even though it's actually purple. The purple, they claim, is light leakage from the other LED's, even though the other LED's in those two pictures are orange.



LE2:
In the chart below after blue there are supposed to be three shades of purple, after which white follows.




_From the manual:

The + and – volume balls are illuminated to indicate the volume setting. The standard range is from brown to white for maximum volume.
At maximum volume the – ball will remain on white whilst the + ball will change through the colours from red to white at maximum volume. This gives a finer adjustment at very high volume._

This is what volume close to 100% looks like while playing DSD.



As it's described above, close to 100% volume the - ball stays "white" while the + ball changes color from brown to white.
The ON/OFF ball in the picture is taken while playing DSD and is supposed to be "white" .

The - ball (at close to 100% volume) matches color with the ON/OFF ball while playing DSD.
I guess this is what "white" looks like on the Mojo.


100% volume:


----------



## hifinoob005

Bad post.


----------



## alexdemaet

I am able to play some dsf music using Foobar.
 It's not a Vinylstudio problem. I have some dsf Files ripped from SACD that I cannot play in Windows 10 nor with Ubuntu using jriver mediacenter 25. Thanks to Synology DSM, I al able to play them with foobar from the Diskstation NAS. Thanks you for the support


----------



## alexdemaet

As I say, to the best of my knowledge, we generate both file types correctly, but DSF files have a variable block size (I think we use 4k, from memory)


----------



## gimmeheadroom (Apr 19, 2020)

I have lost track of what @hifinoob005 was asking. But I was able to play a couple of SACD rips over my Mojo and the on/off ball is kinda off-white tending towards pink, not white as I would expect. As much as the colorful balls are kinda cool and remind me of a Superman episode from the 1960s or 70s, the interface gets old fast.

And BTW I am using the line out volume setting where both of the volume balls are blue because I'm driving a mini stereo stack with the Mojo, not using headphones.. This helps show that the on/off ball is more whitish but maybe because of the blue light from the other two balls it appears pink.

Give me an LED display any day over this nonsense...


----------



## hifinoob005

gimmeheadroom said:


> I have lost track of what @hifinoob005 was asking. But I was able to play a couple of SACD rips over my Mojo and the on/off ball is kinda off-white tending towards pink, not white as I would expect. As much as the colorful balls are kinda cool and remind me of a Superman episode from the 1960s or 70s, the interface gets old fast.
> 
> And BTW I am using the line out volume setting where both of the volume balls are blue because I'm driving a mini stereo stack with the Mojo, not using headphones.. This helps show that the on/off ball is more whitish but maybe because of the blue light from the other two balls it appears pink.
> 
> Give me an LED display any day over this nonsense...



I've solved the problem , it's in the post above.


----------



## dakanao

Why does foobar2000 have an ASIOhost process in the taskbar of Windows, while JRiver Media Center doesn't?


----------



## gimmeheadroom

It does not necessarily have it. Are you using ASIO4ALL or some ASIO proxy or plugin?


----------



## dakanao

gimmeheadroom said:


> It does not necessarily have it. Are you using ASIO4ALL or some ASIO proxy or plugin?


I'm using Chord ASIO 1.5. 

Both programs support the codec and play through it fine.

But why does only foobar reveal the ASIOhost32 in the taskbar, and the JRiver doesn't?


----------



## gimmeheadroom

1.5 or 1.05? I have the 1.05 driver and I can play MQA from Tidal or anything from foobar (playing an SACD ISO rip right now) and there is nothing in the taskbar except the USB icon for removable devices. Win 10 Pro with latest updates.

I have another box far far away where I do remember seeing the Chord device driver does have some kind of icon in the taskbar now that I think about it. Also Win 10 Pro with latest updates. On that box it was always there and on this one it never is. I have no idea.


----------



## dakanao

gimmeheadroom said:


> 1.5 or 1.05? I have the 1.05 driver and I can play MQA from Tidal or anything from foobar (playing an SACD ISO rip right now) and there is nothing in the taskbar except the USB icon for removable devices. Win 10 Pro with latest updates.
> 
> I have another box far far away where I do remember seeing the Chord device driver does have some kind of icon in the taskbar now that I think about it. Also Win 10 Pro with latest updates. On that box it was always there and on this one it never is. I have no idea.


I have the Chord 1.05 ASIA

When you go to "details" on your taskbar-tab, can you see "ASIOhost32" in there?


----------



## gimmeheadroom

dakanao said:


> I have the Chord 1.05 ASIA
> 
> When you go to "details" on your taskbar-tab, can you see "ASIOhost32" in there?



I don't know what a taskbar tab is. Can you explain?


----------



## dakanao

gimmeheadroom said:


> I don't know what a taskbar tab is. Can you explain?


I'm sorry, I meant the taskmanager (ctrl + alt + del). There's a a tab there at the end that says ''details''

With foobar, there's always ASIOhost there when I use the native ASIO of my DAC.

However, on Jriver, when I use the same ASIO service, the ASIOHost in not in the taskmanager details tab.


----------



## gimmeheadroom

I have ASIOhost64.exe running when foobar2000 is playing.


----------



## dakanao

gimmeheadroom said:


> I have ASIOhost64.exe running when foobar2000 is playing.


Yes me too, and when you have Jriver Media Player running?


----------



## gimmeheadroom

I trialed Audirvana and I think Jriver but didn't like either one. Don't have them anymore.


----------



## gibsonsg87 (Nov 27, 2020)

Hey everybody. When I play my DSD files natively, I get a huge amount of hiss. I have always had issues using DSD with Foobar2000. I have all the components and other lossless/HD files work fine. Anybody know what is going on? Anyone have a fix? Anything that I am missing?

PS: I am using the Holo Audio Spring 2 Level 1


----------



## sajunky

gibsonsg87 said:


> Hey everybody. When I play my DSD files natively, I get a huge amount of hiss. I have always had issues using DSD with Foobar2000. I have all the components and other lossless/HD files work fine. Anybody know what is going on? Anyone have a fix? Anything that I am missing?


Noise instead of DSD is a *positive test result* for broken bit-perfect path. It also affect non-DSD files, but you didn't notice it yet.

Which output is selected in Foobar's Preferences -> Playback -> Output? ASIO or WASAPI? If WASAPI, did you enable option "Allow application to take exclusive control of this device" in Windows Sound Control Panel?

In Foobar it can be also broken. Check it up:
- move  Super Audio CD Decoder to the top of the list: Playback ->  Decoding
- empty an active DSP's list: Playback -> DSP Manager
- disable replay gain: Playback -> ReplayGain -> Processing = <none>

If doesn't work, uninstall Foobar, remove program folder and install again.


----------



## gibsonsg87

Ok. I will try all of those things. I will get back to you on what works. Thanks for your help.


----------



## gibsonsg87

I've reinstalled Foobar2000 and did all the other steps and I'm still getting the static. Do you think there are components interfering with each other?


----------



## gibsonsg87

I fixed the problem!! Now Foobar2000 is playing DSD natively!!


----------



## gimmeheadroom

gibsonsg87 said:


> I fixed the problem!! Now Foobar2000 is playing DSD natively!!


It would be even more amazing if you told us how you did it!

Because it did not happen to us


----------



## gibsonsg87

In my case, I updated Foobar2000 and reinstalled all the components. Then I reinstalled the drivers for my DAC. (Not on purpose!) It took me all night and a lot of reading.


----------



## dakanao

What does the “High definition audio output” option do on foobar iOS?


----------



## sajunky

dakanao said:


> What does the “High definition audio output” option do on foobar iOS?


Probably nothing yet, check release notes history. Geting Hires in iOS is tricky. You need lightning camera adapter ~$30  (not the cable supplied with a phone) and a compatible *external* DAC. Also try Vox app.


----------



## dakanao

sajunky said:


> Probably nothing yet, check release notes history. Geting Hires in iOS is tricky. You need lightning camera adapter ~$30  (not the cable supplied with a phone) and a compatible *external* DAC. Also try Vox app.


Does foobar iOS downsampls 24 bit files to 16 bit?


----------



## sajunky

dakanao said:


> Does foobar iOS downsampls 24 bit files to 16 bit?


What vox says?


----------



## dakanao

sajunky said:


> What vox says?


Original FLAC quality playback is only for the premium members of Vox.


----------



## dakanao

What advantages does Vox have over foobar iOS?


----------



## sajunky

dakanao said:


> Original FLAC quality playback is only for the premium members of Vox.


Sorry to hear, It brought my memory. It seems no features now unless you go premium. I switched to android two years ago.


----------



## dakanao

sajunky said:


> Sorry to hear, It brought my memory. It seems no features now unless you go premium. I switched to android two years ago.


Does Vox have better sq than foobar iOS?


----------



## dakanao (Dec 2, 2020)

Alright, so I did a quick comparison of foobar2000 and Vox, both on my iPad.

The Vox gives more details, better timing and dynamic range. Volume of different sounds are more varied by Vox, indicating better dynamic range. Foobar has everything more at the same volume in comparison.

Foobar has softer upper-treble, but it’s more of a de-emphasis than a refinement, since Vox is neutral here. Voices sound slightly thicker with foobar, but again this is due to the de-emphasis of the upper-treble, as vocals are less dynamic.

Basically, Vox sounds like everything is playing at a higher bitrate than foobar on the exact same files, at the exact same parts of the song.


----------



## Dawnrazor

dakanao said:


> Alright, so I did a quick comparison of foobar2000 and Vox, both on my iPad.
> 
> The Vox gives more details, better timing and dynamic range. Volume of different sounds are more varied by Vox, indicating better dynamic range. Foobar has everything more at the same volume in comparison.
> 
> ...


I always thought Foobar was average on the pc so its not a surprise it would be on IOS.


----------



## sajunky (Dec 3, 2020)

Dawnrazor said:


> I always thought Foobar was average on the pc so its not a surprise it would be on IOS.


Foobar is very good on PC. If you hear difference you probably didn't setup bit-perfect transfers properly, or a commercial player *play with you* using EQ, room echo DSP or other dirty tricks to fool your ears.

The same is perhaps on mobile, even more, as user has no information on the audio path, whether it goes through the system mixer.


----------



## Dawnrazor

sajunky said:


> Foobar is very good on PC. If you hear difference you probably didn't setup bit-perfect transfers properly, or a commercial player *play with you* using EQ, room echo DSP or other dirty tricks to fool your ears.
> 
> The same is perhaps on mobile, even more, as user has no information on the audio path, whether it goes through the system mixer.


Sorry my post didn't fit into your "all players sound the same unless you messed them  up somehow" paradigm.  None of your proposed excuses happened.  Hysolid for example doesn't have any eqs and while it was 10+ years ago when I tried and ditched foobar, I recall a bit perfect test of some sort and it passing.  I always use asio to cut down on the likelyhood of any dirty tricks.

Foobar sounded very very 2d compared to other players like cPlay or Hysolid.  Heck Winamp even sounded better than Foobar.


----------



## sajunky

Dawnrazor said:


> Sorry my post didn't fit into your "all players sound the same unless you messed them  up somehow" paradigm.  None of your proposed excuses happened.  Hysolid for example doesn't have any eqs and while it was 10+ years ago when I tried and ditched foobar, I recall a bit perfect test of some sort and it passing.  I always use asio to cut down on the likelyhood of any dirty tricks.
> 
> Foobar sounded very very 2d compared to other players like cPlay or Hysolid.  Heck Winamp even sounded better than Foobar.


How did you test bit perfect path?


----------



## Sterling2

Dawnrazor said:


> Sorry my post didn't fit into your "all players sound the same unless you messed them  up somehow" paradigm.  None of your proposed excuses happened.  Hysolid for example doesn't have any eqs and while it was 10+ years ago when I tried and ditched foobar, I recall a bit perfect test of some sort and it passing.  I always use asio to cut down on the likelyhood of any dirty tricks.
> 
> Foobar sounded very very 2d compared to other players like cPlay or Hysolid.  Heck Winamp even sounded better than Foobar.


I am satisfied with Apple Music; but, I have Foobar 2000 on my laptop too, to accomodate FLAC stereo and multi-channel downloads from Acoustic Sounds. I have not customized Foobar, it is running as downloaded and I have no complaint using it to deliver stereo music to my OPPO's DAC. It does not however deliver multi-channel to the OPPO via HDMI without dropouts every few minutes. I think perhaps I need to make some sort of adjustment to Foobar but being clueless about it, I now just enjoy my multi-channel music via a thumb drive connected to one of the OPPO's usb ports.


----------



## sajunky

Sterling2 said:


> I am satisfied with Apple Music; but, I have Foobar 2000 on my laptop too, to accomodate FLAC stereo and multi-channel downloads from Acoustic Sounds. I have not customized Foobar, it is running as downloaded and I have no complaint using it to deliver stereo music to my OPPO's DAC. It does not however deliver multi-channel to the OPPO via HDMI without dropouts every few minutes. I think perhaps I need to make some sort of adjustment to Foobar but being clueless about it, I now just enjoy my multi-channel music via a thumb drive connected to one of the OPPO's usb ports.


You need to configure Foobar to get good results. Basic configuration include at least installing plugins for bit perfect transfer: WASAPI or ASIO, I suggest both. Then setup task priority, bit lenght, output and some neccessary configuration is Windows Control Panel for WASAPI task priority and exclusive access. Playing multichannel can give Windows trouble, as it does unneccesary (and sound degrading) resampling. Playing multichannel should not break when Windows mixer is bypassed, it is why it is important to complete bit-perfect configuration. A link below will guide step by step with screenshots: https://diyaudioheaven.wordpress.com/digital/pc-software/foobar-2000-for-dummies/

GUI customisation in this guide is optional, skip installing extra plugins, HDCD, DTS, etc. if you don't need it.


----------



## manueljenkin (Dec 22, 2020)

Dawnrazor said:


> Sorry my post didn't fit into your "all players sound the same unless you messed them  up somehow" paradigm.  None of your proposed excuses happened.  Hysolid for example doesn't have any eqs and while it was 10+ years ago when I tried and ditched foobar, I recall a bit perfect test of some sort and it passing.  I always use asio to cut down on the likelyhood of any dirty tricks.
> 
> Foobar sounded very very 2d compared to other players like cPlay or Hysolid.  Heck Winamp even sounded better than Foobar.


+1 on this. I've set up both asio and wasapi on foobar (tried both peters and cases plugins) and it was nowhere close to even musicbee in terms of sq. Do try PlayPCMWin too! Another great free player. Recently I've been exploring installing the music playback software and audio driver in RAMDISK using some free ramdisk software (currently using AMD's one), and I'll recommend exploring that option too!.


----------



## Joe Bloggs

Dawnrazor said:


> Sorry my post didn't fit into your "all players sound the same unless you messed them  up somehow" paradigm.  None of your proposed excuses happened.  Hysolid for example doesn't have any eqs and while it was 10+ years ago when I tried and ditched foobar, I recall a bit perfect test of some sort and it passing.  I always use asio to cut down on the likelyhood of any dirty tricks.
> 
> Foobar sounded very very 2d compared to other players like cPlay or Hysolid.  Heck Winamp even sounded better than Foobar.


Foobar sounds just fine here in my premium home theatre and head-fi setups.  Bit perfect enabled or otherwise.  Are you sure it's not your brain playing tricks with you?  No offence--everyone's brain messes with everyone in some way or another.


----------



## Dawnrazor

Joe Bloggs said:


> Foobar sounds just fine here in my premium home theatre and head-fi setups.  Bit perfect enabled or otherwise.  Are you sure it's not your brain playing tricks with you?  No offence--everyone's brain messes with everyone in some way or another.


Trust me its not something I wanted to do....not like Foobar.  It sounded very 2d and flat.  I listened several times and doublechecked the settings.  But other players were just better.  Less lifeless and dull.  I had speakers at the time and they really imaged well and it was pretty easy to hear everything collapse when Foobar was playing.  Maybe it was just foobar's asio support but it was enough for me to ditch foobar and not look back....


----------



## Dawnrazor

manueljenkin said:


> +1 on this. I've set up both asio and wasapi on foobar (tried both peters and cases plugins) and it was nowhere close to even musicbee in terms of sq. Do try PlayPCMWin too! Another great free player. Recently I've been exploring installing the music playback software and audio driver in RAMDISK using some free ramdisk software (currently using AMD's one), and I'll recommend exploring that option too!.


Yeah playing from ramdisk did sound better.  I may check out the PlayPCMWIN, but no complaints with Hysolid at the moment.  For a long time I was doing cPlay in a cics memory player setup.  He had a shell that would replace windows and just do music.  That was one of the best sounding setups...but no anything...no art work, etc.  Just great sound.  It was interesting how he would make software enhancements to minimize resources used by the computer and how that would change the sound. He had a concept called the memory where everything was played back from memory.  Each version was a bit different and that was just the player.  There were so many os hacks and IIRC you got to only 2 or 3 services running, even ditching the windows audio service.  IIRC even memory settings in the bios were adjusted along with severe underclocking and undervolting.  Yea CMOS! Is your spark the sign of failure or future success?

 Hysolid says the the smartphone/ or tablet offloads the artwork/ cataloging so the computer doesn't have to do that and it also has a mode that allows you to log out of windows.  It sounds better than foobar to me for sure.   

I ditched an internal soundcard and went with an ethernet protocol thinking that the setting on the computer wouldn't make any difference.  man was I wrong.  Short version is that I tweaked the pc and it sounded great.  Pc got replaced and the same exact setup sounded horrible.  I had not yet done the tweaks...


----------



## Dawnrazor

sajunky said:


> How did you test bit perfect path?


I don't recall specifically.  There was some HD files or something and if the signal wasn't bit perfect it would yield static or something.  Or maybe it was dts or something.  I just recall trying it and getting the right results.  Also I only used ASIO which I think should always be bit perfect right?


----------



## manueljenkin (Dec 25, 2020)

Dawnrazor said:


> Yeah playing from ramdisk did sound better.  I may check out the PlayPCMWIN, but no complaints with Hysolid at the moment.  For a long time I was doing cPlay in a cics memory player setup.  He had a shell that would replace windows and just do music.  That was one of the best sounding setups...but no anything...no art work, etc.  Just great sound.  It was interesting how he would make software enhancements to minimize resources used by the computer and how that would change the sound. He had a concept called the memory where everything was played back from memory.  Each version was a bit different and that was just the player.  There were so many os hacks and IIRC you got to only 2 or 3 services running, even ditching the windows audio service.  IIRC even memory settings in the bios were adjusted along with severe underclocking and undervolting.  Yea CMOS! Is your spark the sign of failure or future success?
> 
> Hysolid says the the smartphone/ or tablet offloads the artwork/ cataloging so the computer doesn't have to do that and it also has a mode that allows you to log out of windows.  It sounds better than foobar to me for sure.
> 
> I ditched an internal soundcard and went with an ethernet protocol thinking that the setting on the computer wouldn't make any difference.  man was I wrong.  Short version is that I tweaked the pc and it sounded great.  Pc got replaced and the same exact setup sounded horrible.  I had not yet done the tweaks...



Hysolid is also among my favourites but in my system it has a "saturated sound" but clean and detailed. Wtfplay on the other hand is just as clean and detailed but without that saturated coloration. Wtfplay does similar to what the custom cics play code for you is doing. Runs from RAM with all kernels etc from RAM. Underclocking/undervolting is not in my radar since I use a laptop. If I go that serious, I might just skip pc and get something like sdtrans384. I've sent you the rest in PM.

Regarding foobar, asio will be bit perfect so is Hysolid, wtfplay, and so are many other players set up in asio or wasapi. There are atleast two asio plugins afaik, one from Peter and another from case. Case's one has a bug and it actually clips, while Peter's is functionally proper. Neither sound anywhere near as good as Hysolid though 😅.


----------



## sajunky (Dec 25, 2020)

@Dawnrazor.

Sorry , I have to depreciate your posts. You are not looking for help how to setup Foobar, you do use number of system tricks which are not needed. You just claim that Foobar sounds worse than other players, it cannot be accepted. I made number of installations helping others. None of these people make such claim, it is a nonsence.

You should use a fresh windows installation, follow a minimum Foobar bit-perfect setup according to the quide, no extra addons/features, enable WASAPI exclusive mode in Windows Sound Control Panel for this device and use the WASAPI event mode output in Foobar. If there is something is wrong with your PC, it will show as an occasional breaking up the stream, with some exceptions, see a note (*).

However it should be no sound difference between Foobar and other players using the same output and bit-perfect transfer. To test bit-perfect setup in Foobar try to play DSD files without use of DSD Processor, just default setup of SACD component. It will output DSD stream embedded in PCM frames, it is called DoP. If you get music, everything is fine. Any problem with a bit-perfect setup will result a noise.

You use ASIO, this is fine, but remember that quality of ASIO playback depends on the implementation of the driver. If there is no problem, it will sound almost the same as WASAPI event mode. It is easier to configure bit-perfect setup, but it doesn't give a reference for comparison.

* Why not always, why  'almost'? It is because some older USB implementations do not use USB asynchronous transfers, but this time is gone. If you have such device, you should rather replace it or use S/PDIF converter with full reclocking. In modern implementations there should be not a significant difference, no reason to worry about.

If your playback sound is better from a phone (as you wrote), then it indicate a big isssue. You must deal with it in a first place. Unfortunately very common problem, PC is a strong source of ground loops. It goes out of the scope of this thread and solutions are expensive. You can try Raspberry PC as a music source, probably the cheapest way to deal with this issue.


----------



## manueljenkin

dakanao said:


> Does Vox have better sq than foobar iOS?


Have you tried onkyo hfplayer? I don't have an iOS device but it sounds nice in my android device. The good features are paid though.


----------



## castleofargh

Oh boy, what a battle: 
On the red corner, a few guys who claim that foobar degrades the sound audibly, and you better take their word for it because that's all you'll get.
VS,
 on the blue corner, many measurements easily repeatable and suggesting that foobar can output audio just fine like pretty much any other player with similar settings. Those who can't try themselves can see stuff online, like for example the many RMAA, impulse responses, and jitter tests that Archimago shared online over the years. 

It's really hard to decide which side we should believe...



Last time I tried to take someone seriously on that subject of foobar sounding "bad", it was Manuel. His explanations made no sense to me, and honestly the stuff he posted(I think it was on SBAF?) would have anybody with slight technical knowledge go "wait what?".
What little I could decrypt about asio or whatever being bad in foobar, I tested myself with result contradicting his claims(just like Archimago's results when using asio in foobar TBH, and just like the very many people who on occasion have used foobar to play some test signals into some gear to measure them and didn't get any unexpected trouble). finally he showed some pics of an oscilloscope and some talk of...sample rate(I think). To this day I don't know what the pics are supposed to show or what were the testing conditions. I asked him to do his best to explain those stuff to me, he never did. But to come on various topics and crap on foobar with the same overconfident empty claims, you can be sure that he always finds the time for that.
anyway, I even asked if someone else reading the thread had understood what that was about and could explain it to me, no candidate so far.

I already said this several times, if there is really a problem with foobar, one that doesn't go away once you know what you're doing. Then we need to know about it and use another player or just have the guys maintaining foobar, to solve the issue. So instead of repeating the same empty claims over and over again as a sport, it would be real nice if one of the enlightened ones could some day pull his fingers out of his butt and actually back up his claims with a demonstration of the issue that we can replicate. that would make that person a real contributor to the community, instead of being a BS artist.


----------



## manueljenkin

castleofargh said:


> Oh boy, what a battle:
> On the red corner, a few guys who claim that foobar degrades the sound audibly, and you better take their word for it because that's all you'll get.
> VS,
> on the blue corner, many measurements easily repeatable and suggesting that foobar can output audio just fine like pretty much any other player with similar settings. Those who can't try themselves can see stuff online, like for example the many RMAA, impulse responses, and jitter tests that Archimago shared online over the years.
> ...



I did explain the test and there are people who tried the tweaks and heard similar improvements in sound. I am sorry for your inadequacy in critical listening or your know-it-all attitude. You're just craving for attention, hence trying to pull my name out of the blue everywhere. Most of the ones I've written in my thread are free stuff anyone can try, so there's not much need to go around seeking someone else's help to validate it.


----------



## sajunky (Dec 26, 2020)

manueljenkin said:


> I did explain the test and there are people who tried the tweaks and heard similar improvements in sound. I am sorry for your inadequacy in critical listening or your know-it-all attitude. You're just craving for attention, hence trying to pull my name out of the blue everywhere. Most of the ones I've written in my thread are free stuff anyone can try, so there's not much need to go around seeking someone else's help to validate it.


These tweeks to the OS are unnecesary and unwanted. As long as a pipe is not broken, it is not a problem.
A problems is usually caused by noise on the USB connection, it may cause some differences in sound, depends on a system setup. Eliminate main problem and it is done. You take attention to the wrong things.


----------



## bfreedma

manueljenkin said:


> I did explain the test and there are people who tried the tweaks and heard similar improvements in sound. I am sorry for your inadequacy in critical listening or your know-it-all attitude. You're just craving for attention, hence trying to pull my name out of the blue everywhere. Most of the ones I've written in my thread are free stuff anyone can try, so there's not much need to go around seeking someone else's help to validate it.



You didn’t explain anything.  As always, you posted a bunch of irrelevant word salad In response to a simple question which was directly answerable.

Your last sentence in the quoted post is telling.  Essentially, your position is that you get to make up whatever bs you like and then everyone else has to prove/disprove it.  Good luck with that...


----------



## manueljenkin (Dec 26, 2020)

sajunky said:


> These tweeks to the OS are unnecesary and unwanted. As long as a pipe is not broken, it is not a problem.
> A problems is usually caused by noise on the USB connection, it may cause some differences in sound, depends on a system setup. Eliminate main problem and it is done. You take attention to the wrong things.



My system is a laptop running on battery. And my dacs either run on separate battery or use usb power (I try on multiple dacs before coming to an assessment). So it's nothing relating to ground loops. Reducing usb ground plane noise (not the same as ground loop) is what these software tweaks do, the total number of access is lower when the song is fully loaded to ram and fetched from there instead of constantly having storage access in between these. And some of these softwares use sse or avx instructions to fetch larger chunk of data from ram at once so that reduces even the number of ram access. Any access will have a noise associated with it and since usb is a standard meant mainly for mass adoption (and not mainly over other fidelity related issues) almost all general implementations will not be immune to this. Ethernet is far less susceptible to this issue since it's standard is more rigorous than usb, and whenever possible it's wiser to use ethernet to stream it to some raspberry Pi or other endpoint (provided the endpoint is also fairly low noise, otherwise it's of no use). If you're rich maybe you can also buy something like a Jcat usb card 😅, or a signal regenerator like the ifi iusb 3.0. I'm not and at that price point I'll rather try to not use usb at all. It is also entirely possible to have crazy good isolation at the dac side (like the unison board on schiit) but it'll cost you, and also likely to affect latency, the latter is not a big deal for music. The software tweaks are free for the most part.


----------



## bfreedma

manueljenkin said:


> My system is a laptop running on battery. And my dacs either run on separate battery or use usb power (I try on multiple dacs before coming to an assessment). Nothing relating to ground loops. Reducing usb ground plane noise is what these software tweaks do, the total number of access is lower when the song is fully loaded to ram and fetched from there instead of constantly having storage access in between these. And some of these softwares use sse or avx instructions to fetch larger chunk of data from ram at once so that reduces even the number of ram access. Any access will have a noise associated with it and since usb is a standard meant mainly for mass adoption (and not mainly over other fidelity related issues) most general implementations will not be immune to this.



Sorry, but claims have no objective support.  And your position on RAM disk, disk topology, and USB operation shows a complete lack of understanding of theses technologies, their implementation, and their relationship with the data path.

Show some evidence of any of your claims being audible or move on.


----------



## sajunky (Dec 26, 2020)

bfreedma said:


> Sorry, but claims have no objective support.  And your position on RAM disk, disk topology, and USB operation shows a complete lack of understanding of theses technologies, their implementation, and their relationship with the data path.
> 
> Show some evidence of any of your claims being audible or move on.


In a defense, @manueljenkin has some level of understanding of jittery nature of the USB clock, but I guess he didn't explore WASAPI event driven mode in combination with a modern DAC. It is now more common that there is an explicit feedback endpoint in the interface chip which allow to take a full advantage of asynchronous transfers. Both popular XMOS and Amanero are capable of this type transfer. It is important to note that with a proper setup, USB transfers can be synchronised with a master clock on the DAC.

For example I have to take your attention to the Audio GD digital interface device DI-20. Developers found that with asynchronous USB transfers internal PLL gives more jitter than a direct clocking. DI-20(HE) is a first A-GD device that use a high precision fixed frequency clock for USB transfers, clock synchronisation is very simple and jitter free. A success of DI-20 prompted a company to make similar changes to the all family of DACs, now 2021 versions use the same clock synchronisation and more attention is taken to the galvanic isolation on the USB connector.


----------



## manueljenkin (Dec 26, 2020)

sajunky said:


> In a defense, @manueljenkin has some level of understanding of jittery nature of the USB clock, but I guess he didn't explore WASAPI event driven mode which in combination with a modern DAC. It is now more common that there is an explicit feedback endpoint in the interface chip which allow to take a full advantage of asynchronous transfers. Both XMOS and Amanero are capable of these transfers. It is important to note that with a proper setup, USB transfers can be synchronised with a master clock on the DAC.
> 
> For example I have to take your attention to the Audio GD digital interface device DI-20. Developers found that with asynchronous USB transfers internal PLL gives more jitter than a direct clocking. DI-20(HE) is a first A-GD device that use a high precision fixed frequency clock for USB transfers, clock synchronisation is very simple and jitter free. A success of DI-20 prompted a company to make similar changes to the all family of DACs, now 2021 versions use the same clock synchronisation and more attention is taken to the galvanic isolation on the USB connector.


Where can I read more about the DI 20. Curious to know further.

And btw, I was mostly talking about a general scenario, most of us use dacs that cost south of 400$. Good quality isolation itself would get pretty close to the same price, or even an acceptable one would cost north of 50$ if they implement something like the adum4160 + supporting circuits, or if you go the raspberry Pi way. A general inexpensive dac would most likely benefit from these noise reductions with these customisations to playback software.

I haven't followed the very recent xmos developments but my nx4dsd showed noticeable changes with these tweaks and my friends rme adi2 also did show similar benefits, and both of these are fairly modern. Is there any major improvement in the last few months? The only devices I have currently known to be almost completely immune to these issues are schiit yggdrasil with unison and holo may (and they cost serious cash).


----------



## bfreedma

sajunky said:


> In a defense, @manueljenkin has some level of understanding of jittery nature of the USB clock, but I guess he didn't explore WASAPI event driven mode which in combination with a modern DAC. It is now more common that there is an explicit feedback endpoint in the interface chip which allow to take a full advantage of asynchronous transfers. Both popular XMOS and Amanero are capable of this type transfer. It is important to note that with a proper setup, USB transfers can be synchronised with a master clock on the DAC.
> 
> For example I have to take your attention to the Audio GD digital interface device DI-20. Developers found that with asynchronous USB transfers internal PLL gives more jitter than a direct clocking. DI-20(HE) is a first A-GD device that use a high precision fixed frequency clock for USB transfers, clock synchronisation is very simple and jitter free. A success of DI-20 prompted a company to make similar changes to the all family of DACs, now 2021 versions use the same clock synchronisation and more attention is taken to the galvanic isolation on the USB connector.



I‘ve yet to see a modern DAC with jitter measurements near audible levels that wasn’t a victim of massively improper design.  Did Audio GD really engineer previous products so poorly that jitter was a real issue?  Good to see them changing to the same implementation model many other manufacturers have been using for years.

With many/most devices already clock syncing or reclocking, jitter isn’t an issue for the overwhelming majority of USB users.


----------



## sajunky

bfreedma said:


> I‘ve yet to see a modern DAC with jitter measurements near audible levels that wasn’t a victim of massively improper design.  Did Audio GD really engineer previous products so poorly that jitter was a real issue?  Good to see them changing to the same implementation model many other manufacturers have been using for years.


You are wrong regarding your biased opinion on other members, you are also wrong regarding Audio GD. Where did you get it? From botchered measurements on ASR? They did the same to a TotalDAC which has a good reputation in high-end market as a good sounding device. As for a new sans-reclocking design, I know for sure the only one, in a high-end territory. If you know more please come with details, I would be glad to see something in a price bracket below $1k.


----------



## sajunky (Dec 26, 2020)

manueljenkin said:


> Where can I read more about the DI 20. Curious to know further.
> 
> And btw, I was mostly talking about a general scenario, most of us use dacs that cost south of 400$. Good quality isolation itself would get pretty close to the same price, or even an acceptable one would cost north of 50$ if they implement something like the adum4160 + supporting circuits, or if you go the raspberry Pi way. A general inexpensive dac would most likely benefit from these noise reductions with these customisations to playback software.
> 
> I haven't followed the very recent xmos developments but my nx4dsd showed noticeable changes with these tweaks and my friends rme adi2 also did show similar benefits, and both of these are fairly modern. Is there any major improvement in the last few months? The only devices I have currently known to be almost completely immune to these issues are schiit yggdrasil with unison and holo may (and they cost serious cash).


There is discussion about a new feature on the DI-20 and R7/R8 thread:
https://www.head-fi.org/threads/audio-gd-di-20.918123/
https://www.head-fi.org/threads/new-audio-gd-r7-r2r-7-r8-flagship-resistor-ladder-dacs.853902/

I am sure that in the price range below $400 you can't find direct clocking device. With ADI2 you pay for features, not SQ. I would suggest Audio GD R-1 or Denafrips Ares II for the best SQ, but Ares use reclocking no matter what. R-1 2021 version is a right choice around $850.


----------



## manueljenkin

sajunky said:


> There is discussion about a new feature on the DI-20 and R7/R8 thread:
> https://www.head-fi.org/threads/audio-gd-di-20.918123/
> https://www.head-fi.org/threads/new-audio-gd-r7-r2r-7-r8-flagship-resistor-ladder-dacs.853902/
> 
> I am sure that in the price range below $400 you can't find direct clocking device. With ADI2 you pay for features, not SQ. I would suggest Audio GD R-1 or Denafrips Ares II for the best SQ, but Ares use reclocking no matter what. R-1 2021 version is a right choice around $850.


Thank you very much. I'll read into it.


----------



## AxelCloris

Let's please keep the discussion on-topic and away from personal comments. Thanks everyone, we appreciate it.


----------



## timb5881

Does any one else have a problem with Foobar when you get a windows 8 update?  Every time I get one I have to reconfigure foobar, the drivers seem to mall need to be reloaded.


----------



## gimmeheadroom

timb5881 said:


> Does any one else have a problem with Foobar when you get a windows 8 update?  Every time I get one I have to reconfigure foobar, the drivers seem to mall need to be reloaded.


I never had Windows 8. I'm a confirmed Windows hater but I have to admit, Windows 10 is pretty inoffensive. And that is saying a lot.

Of course it does weird things and sometimes drivers don't seem to work but 2 or three reboots seem to fix it. Luckily I have not had to reinstall any foobar components after an update.


----------



## dougms3

Trying to isolate if this is a problem with my SMSL SU-9, foobar settings or most likely both.  I'm pretty sure there is something wrong with the SU-9 because I can't change the sample rate higher than 384khz in windows sound control panel.

In foobar, under SACD settings. the pcm samplerate is capped at 352800 but on the dac display its only showing a 44khz samplerate.  

Also, another anomaly is that when swapping usb ports sometimes I get 705600hz max sample rate option but I can't get the dac to stream that, then that option usually disappears at some point.  I've never seen a 768khz option on that drop down list since I've had this device.

I've tried on Audirvana and its able to upsample to dsd512 but asio is always capped.

This is a link to the collection of screenshots.

https://imgur.com/a/rp5ah2w


----------



## sajunky

dougms3 said:


> Trying to isolate if this is a problem with my SMSL SU-9, foobar settings or most likely both.  I'm pretty sure there is something wrong with the SU-9 because I can't change the sample rate higher than 384khz in windows sound control panel.
> 
> In foobar, under SACD settings. the pcm samplerate is capped at 352800 but on the dac display its only showing a 44khz samplerate.
> 
> ...


Hi, I looked at photos and would like to add some comments, I hope it will help.

1. 384kHz is limit of a Windows mixer (unless something has changed). This is normal but I see on the last screenshot 705.6kHz (which you say it happens sometimes and dissappears), strange.

2. Windows panel settings only is valid when you select DS driver as an output in Foobar (or a default Windows sound device). For the WASAPI (push and event) output this settings will be ignored as long as exclusive mode is enabled in Windows (which is done according to a pic). For the ASIO output it doesn't matter at all. For both WASAPI exclusive and ASIO Foobar will decide based on a file sampling rate or oversampling rate you do in the DSP section. Foobar do not display these values. It cannot be higher than a value returned by a device, when you try to play a higher sampling rate, Foobar will return error. 

3. I don't know what you are trying to upsample PCM or DSD. It can be confusing:

For upsampling PCM you use some DSP add-on, settings is found in section: Components -> Playback -> DSP Manager.

For upsampling DSD, all DSP add-ons are bypassed, the only relevant  is a dedicated SACD DSP Processor if enabled in SACD section (which is according to a pic).  Upsampling settings for DSD Processor is found in a section: Components > Tools -> DSP Processor. I haven't used DSP Processor for a while, I think when enabled it takes over speed settings in SACD control panel. Please post screenshot of your DSP Processor settings.

4. I can't comment on the VST3 plug-in, but I have some suggestions:

- disable the option "DoP for Converter".  It is not needed when you only play music.

- I think you have enabled 32-bit output on the Foobar, it might limit your maximum sample rates by a half. Check: Components -> Payback -> Output [Output Format]. You save available bandwith by selecting 24-bit format. DSD DoP format is only 24-bit.

- Try to play DSD256/512 encoded files, instead upsampling in Foobar. Just for testing what is wrong.

 5. I am concern about unstable device parameters returned. Maybe Windows is testing maximum rate and overrides maximum values returned by a device when errors are detected. I know Windows can do it. Playing only 24-bit should can prevent this happening.


----------



## dougms3

sajunky said:


> Hi, I looked at photos and would like to add some comments, I hope it will help.
> 
> 1. 384kHz is limit of a Windows mixer (unless something has changed). This is normal but I see on the last screenshot 705.6kHz (which you say it happens sometimes and dissappears), strange.
> 
> ...



Thanks for chiming in, really appreciate it.

corgifall was helping me in the smsl su9 thread and took a screenshot of his setup and he's able to get 768khz in windows.

disabled dop for converter option, no change.  Unable to change to 24 bit because the option is blocked.

Downloaded a dsd file and plays in foobar but the lcd display on the SU-9 doesnt show that its playing a dsd file.  Its either at 44khz or 384khz.

I don't really use any DSPs except a skip silence dsp.


----------



## sajunky (Feb 6, 2021)

@dougms3

Thanks for following all suggestions. And..... I think I found a problem. A clue is giving Foobar changelog for version 1.6.1:


> Decoders are now made aware of output sample rate (from Windows Mixer settings) and can decode certain formats directly to the intended sample rate, skipping potential resampling steps.



Automatic output data format is only present when default output is selected. Why it is done with ASIO, I don't know (it is only needed to work with WASAPI shared access to avoid additional conversion in Windows - I even think ASIO output is broken in this release), but it is 32-bit, as you selected it in the Windows Sound Panel. It gave me a hint that your output device is wrong.

And it is wrong indeed. Your current output is ASIO:USB DAC ASIO

SACD component will only work properly when DSD output is selected.
Select this device: *DSD:*ASIO:USB DAC ASIO

Now you will be able to chose output format manually, use 24-bit. DSD source will upsampled to DSDxxx and also PCM will be converted and played correctly as DSDxxx.


----------



## dougms3

sajunky said:


> @dougms3
> 
> Thanks for following all suggestions. And..... I think I found a problem. A clue is giving Foobar changelog for version 1.6.1:
> 
> ...



Thanks for your help.

I've tried everything you suggested, its not showing that its upsampling to DSD or PCM on my SU-9. 

It just seems to be with foobar, I've tried with Audirvana and its upsampling to DSD512 and 768khz on the lcd.  

Also sometimes windows seems to recognize 768khz in the sound control panel properties.  Seems to appear here and there, haven't seen the 706khz option in a few days, don't know why it disappers.




Doesn't say studio quality and test tone doesn't work, don't know what that means. 

Maybe theres some emi or rf interference in the usb cable? I ordered a higher quality usb cable with shielding to make sure.

Gonna try AIMP or some other player and see if it works.


----------



## sajunky (Feb 8, 2021)

@dougms3

Stop trying to setup 32-bit output in Windows. I asked to not to try a top speed 32-bit option at all. *Not in the Windows sound panel, nor in Foobar*. Once you do it, go back to 24-bit, unplug and re-plug device. Windows can detect error on the top speed and remove a top speed option, It will also affect Foobar. It is also doing something wrong, so don't activate bugs in Windows. When using ASIO, do not look there at all. Select one of the 24-bit option, reset device and forget Windows sound panel. Also don't make this device a default Windows sound device.

Now about Foobar, and it was a critical question.
Is your output device this?
- *DSD:*ASIO:USB DAC ASIO
- Output format: 24-bit

I don't ask to try. I am asking to make it pernament. It is required to get a maximum speed working.


----------



## dougms3

sajunky said:


> @dougms3
> 
> Stop trying to setup 32-bit output in Windows. I asked to not to try a top speed 32-bit option at all. *Not in the Windows sound panel, nor in Foobar*. Once you do it, go back to 24-bit, unplug and re-plug device. Windows can detect error on the top speed and remove a top speed option, It will also affect Foobar. It is also doing something wrong, so don't activate bugs in Windows. When using ASIO, do not look there at all. Select one of the 24-bit option, reset device and forget Windows sound panel. Also don't make this device a default Windows sound device.
> 
> ...



Lol ok.

Both foobar and windows are set to 24bit, should I reduce the sample rate too?  Its at 384k.

I get this error when trying use - *DSD:*ASIO:USB DAC ASIO output, "Unrecoverable playback error: _Sample rate_ of _1411200 Hz_ not supported ".

If I use - ASIO:USB DAC ASIO it will play music but only at 44khz, it won't upsample.  Also now it won't play using the default device anymore.  Same sample rate error.


----------



## sajunky (Feb 9, 2021)

@dougms3

Thank you for testing as suggested. Now we've got a clue. Format_1411200 Hz_ not supported, it means that DSD Processor is trying to output DSD512 and Foobar returns error, which suggest that setup is done properly and SU-9 do not support this format. *It seems to be a true*, as a maximum for SU-9 is DSD512.

I think you did already pickup my mistake, right? 
No, there is no mistake. It is typical that when specsheet says a maximum supported is DSDxxx, but it is only valid for native DSD transfers.  DoP transfers are typically limited to a half of a maximum speed, but they didn't say so and you are a victim of such marketing. In this case DSD256 DoP transfers is a maximum.

Replace all values on the DSD Processor settings page from DSD512 to DSD256, and I am sure, it will be working. If it doesn't, for testing try to play files you downloaded with DSD Processor disabled. DSD256 file should play, of course you should use the oputput *DSD:*ASIO:USB DAC ASIO


A reason is that Foobar only support DSD PoP playback. To get a native DSD transfers you have to install on the Windows PC a proxy device driver. It is called DSD Transcoder. How to do that, it is covered on the Foobar for Dummies webpage. I will not try to help on this matter, as my experience with DSD Transcoder is negative. I had strange occasional behaviour. It is not worth efforts, trust me.

TL;DR, You are stuck with DSD256 or use a different player that can do a native DSD transfers.


----------



## dougms3

sajunky said:


> @dougms3
> 
> Thank you for testing as suggested. Now we've got a clue. Format_1411200 Hz_ not supported, it means that DSD Processor is trying to output DSD512 and Foobar returns error, which suggest that setup is done properly and SU-9 do not support this format. *It seems to be a true*, as a maximum for SU-9 is DSD512.
> 
> ...



You are a genius.

It is working with DSD256.  I downloaded DSD Transcoder and set all the configuration back to DSD512 and it is working with DSD : ASIO : DSD Transcoder (DoP/Native) output.

What a headache.

So this means SMSL SU-9 natively can only play up to DSD256?


----------



## sajunky

dougms3 said:


> So this means SMSL SU-9 natively can only play up to DSD256?


It means SMSL SU-9 natively can play up to DSD512, but DoP transfers are limited to DSD256.


----------



## dougms3

sajunky said:


> It means SMSL SU-9 natively can play up to DSD512, but DoP transfers are limited to DSD256.



Really enjoying the upsampling to dsd512, it is amazing.

Originally, I was thinking it might have been a hardware issue and when I contact Apos, they just ignored me bumped me around to 3 different reps then a week later said they don't know anything about software and just gave me smsl customer service email.  Never buying anything from them ever again.

Anyway, just wanted to say thanks again for all the help.


----------



## gemmoglock

Hi, does anyone use Foobar for mac here? How is it like on Catalina/Big Sur and is there any playlist sync functionality say to a microsd card? Couldn't find any info online. 

Thanks!


----------



## roskodan

Just dropping by to remind that *foobar2000* has a *convolver* plugin, *Impulse Response Convolver* plugin, once installed, to EQ your headphone to Harman, just download the respective *.wav file* for your headphone from  *AutoEQ* and load it into the plugin.


----------



## Nitreb

roskodan said:


> Just dropping by to remind that *foobar2000* has a *convolver* plugin, *Impulse Response Convolver* plugin, once installed, to EQ your headphone to Harman, just download the respective *.wav file* for your headphone from  *AutoEQ* and load it into the plugin.


I downloaded the plugin, installed it, then downloaded the wav file. but I don't see how to load it into the plugin since there's no option for that - I tried dragging the wav file on the plugin, but got an error message.


----------



## sajunky

Nitreb said:


> I downloaded the plugin, installed it, then downloaded the wav file. but I don't see how to load it into the plugin since there's no option for that - I tried dragging the wav file on the plugin, but got an error message.


Install/uninstall on the Preferences/Components page, then activate/remove and configure on the Preferences/Playback/DSP_Manager

When there are multiple entries on the active list panel, move up and down by dragging according to your requirements.

Some addons add configuration options in the Preferences/Advances section, check it out after installing.


----------



## Nitreb

sajunky said:


> Install/uninstall on the Preferences/Components page, then activate/remove and configure on the Preferences/Playback/DSP_Manager
> 
> When there are multiple entries on the active list panel, move up and down by dragging according to your requirements.
> 
> Some addons add configuration options in the Preferences/Advances section, check it out after installing.


Thanks


----------



## roskodan

file -> preferences -> DSP manager -> select the "Convoler" plugin from "Available DSPs"  list, so it shows up in "Active DSPs", then double click it, proprieties windows should pop up...


----------



## LittleAndy (Oct 10, 2021)

What is the best choice in DSP Manager? Convolver with headphone-matching presets or MathAudio Headphone EQ with preset? Can they work in parallel with each other? I apologize for the not very good, perhaps, the translation, because I can hardly form this specific question in my native language.


----------



## roskodan

The Convolver plugin seems not working right. MathAudio PEQ plugin is the next best free thing, in my experience, if you don't wanna go for Equalizer APO, PEACE.


----------



## ColdsnapBry (Feb 6, 2022)

Can someone explain to stupid me why with a Bifrost 2 in Foobar2000 WASAPI event causes distortion on 96k tracks? But it's fine on WASAPI push?

Am I alright using WASAPI in push? I've been using event for about a year now and just realised this is a problem because I just now tried to play a 96k track.

Here's my settings:


----------



## gimmeheadroom

Why not use ASIO?


----------



## Roseval

ColdsnapBry said:


> Am I alright using WASAPI in push?


Funny, because event mode (pull) in general is more stable. Might be a matter of implementation at the Unison.
Maybe simple increase buffer size.

Have a look at post #13 https://hydrogenaud.io/index.php/topic,109714.0.html#entry903677


----------



## sajunky

Roseval said:


> Funny, because event mode (pull) in general is more stable. Might be a matter of implementation at the Unison.
> Maybe simple increase buffer size.
> 
> Have a look at post #13 https://hydrogenaud.io/index.php/topic,109714.0.html#entry903677


Agreed, as WASAPI event takes less system resources and works more stable with asynchronous transfers. It can be a dormant driver loaded in memory with another instance driver active. Unplug USB cable and see whether there is still DAC driver listed in device manager. Remove this driver and try again or restart PC. Again, why not use ASIO?


----------



## gimmeheadroom

Isn't foobar's volume control supposed to be disabled when using ASIO? I just downloaded the latest version to install on a new PC and the output options have changed again. In addition to ASIO, WASAPI and all the traditional stuff they added Primary Sound exclusive mode. In all cases, foobar volume control is enabled. This is not what I remember..


----------



## sajunky (Mar 2, 2022)

Not. It wasn't disabled before. Volume control is only disabled for DSD playback. What is new? 

Foobar had major changes during last year and now is dropping Direct Sound for good. Default interface is WASAPI Shared. User interface follow requirements of the interface. By example, on a default (or primary) output Foobar doesn't allow to adjust bit depth anymore. It is because WASAPI Shared programming inteface requires from the application to match exactly sample rate and bit depth with a setting of the Windows Sound Control Panel, otherwise there is no sound. WASAPI support is now built in (both shared and exclusive), there is no requirement to install WASAPI plugins. The "Primary Sound exclusive" output is the same as the old WASAPI Exclusive, just a different name. 

You can still install WASAPI plug in, you will get back the old output modes.


----------



## gimmeheadroom

Thanks, just felt like some weird changes and I didn't remember that volume was enabled. Maybe because most of the time if I use foobar I'm playing SACD ISO. I'll check again and see if that changed. I never used WASAPI, so that would not be the reason.


----------



## gimmeheadroom

I think the UI got broken. The volume slider does move for DSD, but it does not have any effect. In previous version(s) the volume slider was greyed-out for DSD and could not be moved.


----------



## Thaddy

I've been having an issue with my Benchmark DAC3B while trying to use either the following device modes:

ASIO : Benchmark_DAC2 ASIO Driver
DSD : ASIO : Benchmark_DAC2 ASIO Driver
For now I'm using the device that's highlighted in the screenshot below.

The problem seems to be that when I'm using one of the two ASIO drivers, Foobar will eventually pause/timeout.  I have the USB 2.0 audio drivers installed from Benchmark, and the Streaming Mode set to 'Always On'.

These modes are the only way I can get the correct bitrate to display on the DAC3.  If I use the device selected below, it's locked at 24-bit/48,000 Hz.


----------



## gimmeheadroom

Write or call Benchmark, it should work without issues.


----------



## Roseval (Mar 7, 2022)

Thaddy said:


> If I use the device selected below, it's locked at 24-bit/48,000 Hz.


No surprise as this is Direct Sound. This is what is set in the Win audio panel so fixed by design.

If you are on Win10, you don't need the drivers by Benchmark as Win10 is fully UAC2 compliant.
https://benchmarkmedia.com/pages/dac-drivers

Try Foobar + WASAPI/Exclusive mode. This will provide automatic sample rate switching.
https://www.thewelltemperedcomputer.com/SW/Players/Foobar.htm


----------



## Thaddy (Mar 7, 2022)

Which output device should be selected?  Event vs. push?






edit:  It seems WASAPI (event) is the way to go.  I'll listen with this device mode for a bit to see if it's still pausing.


----------



## Roseval

Event style indeed (older hardware might need push)


----------



## gimmeheadroom

Found the following interesting info thanks to @Roseval link:

_On all operating systems, the DAC2 and DAC3 must be placed in USB AUDIO 2.0 mode to play DSD and high sample-rate PCM

The driver is not required for playback on recent versions of Windows 10 but adds features such as USB firmware updates and ASIO support.

Important: The USB mode on the DAC2 and DAC3 must then be set to USB AUDIO 2.0 MODE before completing the driver installation. Please note that the DAC ships with the USB mode set to the driverless USB AUDIO 1.1 MODE._

As long as you paid good money, you may as well take advantage of ASIO...


----------



## sajunky (Mar 7, 2022)

gimmeheadroom said:


> Important: The USB mode on the DAC2 and DAC3 must then be set to USB AUDIO 2.0 MODE before completing the driver installation. Please note that the DAC ships with the USB mode set to the driverless USB AUDIO 1.1 MODE.


^^^This. These are obsolete defaults.


----------



## Thaddy

gimmeheadroom said:


> Found the following interesting info thanks to @Roseval link:
> 
> _On all operating systems, the DAC2 and DAC3 must be placed in USB AUDIO 2.0 mode to play DSD and high sample-rate PCM
> 
> ...


I did this, and can play DSD and PCM files with no problem.  The issue is after a period of time Foobar pauses/craps out and stops playing.  I can get it to start up again by just clicking on the song I was listening to, however.


----------



## sajunky (Mar 7, 2022)

Thaddy said:


> I did this, and can play DSD and PCM files with no problem.  The issue is after a period of time Foobar pauses/craps out and stops playing.  I can get it to start up again by just clicking on the song I was listening to, however.


Follow this guide for setting up priority mode in ASIO and WASAPI amoung other things. I also see in your setup proxy driver. Uninstall if not using native DSD transfers. It requires a matching (older) version of Foobar and SACD plugin, a guide explains that.

I think you should start over with a clean install of the latest version of Foobar and plugins, it doesn't have support for DS (Direct Sound), WASAPI Shared is default and WASAPI plugin is not needed anymore. All WASAPI support is now built-in. You will find the 'event' option hidden in the Advanced section instead of listing as a separate output. The list of outputs will be less cluttered


----------



## gimmeheadroom

Thaddy said:


> I did this, and can play DSD and PCM files with no problem.  The issue is after a period of time Foobar pauses/craps out and stops playing.  I can get it to start up again by just clicking on the song I was listening to, however.


Frustrating, hope you get it fixed shortly.

Do you have a streaming service or any other app you can try to see if it's between foobar and your Benchmark or a general issue?


----------



## Thaddy

gimmeheadroom said:


> Frustrating, hope you get it fixed shortly.
> 
> Do you have a streaming service or any other app you can try to see if it's between foobar and your Benchmark or a general issue?


I'm going to try WASAPI for a bit to see if that resolves the issue.  I believe the issue is with Foobar or a component, because this doesn't happen with any other device modes.


----------



## Thaddy

I'm still unable to resolve this strange playback pausing issue.  I've installed the most recent version of Foobar, and have tried starting from scratch with the WASAPI and ASIO components.

Uninstalled the WASAPI component
Uninstalled the ASIO component and Windows driver
Uninstalled the Benchmark drivers
From there, I reinstalled what I needed to get ASIO (and only) to work.  Problem persists.  I then uninstalled everything again and started from scratch with the WASAPI component.  Again, the problem persists.  I'm unable to view any logging from Foobar when the problem happens.  The music simply stops after a random amount of time, but I can restart it again by just selecting the song again.

This is only an issue with ASIO and WASAPI.  If I switch to Direct Sound then problem disappears however the sample rate seems to be fixed at 44.1.


----------



## Roseval

Thaddy said:


> however the sample rate seems to be fixed at 44.1


Direct sound or WASAPI/Shared use the settings in the audio panel.
You have to change them manually.
https://www.thewelltemperedcomputer.com/SW/Windows/Win7/AudioPanel.htm

Wonder if you still need to install the WASAPI component (or if you create probelems by doing so)
https://diyaudioheaven.wordpress.com/digital/pc-software/foobar-2000-for-dummies/


----------



## sajunky

Thaddy said:


> This is only an issue with ASIO and WASAPI. If I switch to Direct Sound then problem disappears however the sample rate seems to be fixed at 44.1.


Wait... The latest Foobar do not support Direct Sound anymore. It is replaced with WASAPI Shared. Of course, it will play 44.1k, it is default settings of Windows Sound Control Panel. If you increase this value, sound might break at a higher sampling rate, the same way. Refer the previous post how to change settings and post results.

Check whether high priority is set for ASIO:
Playback -> Output -> ASIO

If you have installed WASAPI exclusive component, check the same for WASAPI
Advanced - > Playback - > (somewhere)

Or uninstall WASAPI component and use the output appended with word "(exclusive)".

If doesn't help, I suggest to uninstall Foobar, delete program folder and start over.


----------



## Thaddy

sajunky said:


> Wait... The latest Foobar do not support Direct Sound anymore. It is replaced with WASAPI Shared. Of course, it will play 44.1k, it is default settings of Windows Sound Control Panel. If you increase this value, sound might break at a higher sampling rate, the same way. Refer the previous post how to change settings and post results.
> 
> Check whether high priority is set for ASIO:
> Playback -> Output -> ASIO
> ...


Yes, you're correct and I failed to mention that.  No more Direct Sound in this version.  I do however see the default output with (exclusive) listed, which seems to be stable and also displays the correct sample rate on my DAC3.  I'm going to uninstall ASIO and WASAPI and test with this for a bit.


----------



## desertsilver

I've searched the thread but didn't find anything related, please let me know if it has been discussed before. I have a pretty big local HD holding my music and makes significant noise every time it spins up. I am trying to configurate Foobar2000 (the latest version): Read-ahead for local files(Kb) in Advanced settings, to at least 1G (with plenty of RAM, 1G is a good number for me even for some DSD content). My problem is that for setting "Read-ahead for local files(Kb)", no matter how big I set it, it changes to 16384. I think it might be an internal limit by design. The "Full file buffering up to (Kb)" setting is currently set to 1024000.

Is there a way I can set foobar to read up to 1G file content (as a full album, or all tracks in a single folder) at once and minimize local drive spin ups?

I know the "Prevent hard drive sleep while playing" setting, but I'd rather load the content at once and let the drive sleep as it's pretty noise any way.

There is a RAM-Disk component, but it's been 10 years old. Would it be the right way to solve my needs?


----------



## Roseval

I solved the noise problem this way.
As it is a USB HD, I attached it to my router, out of sight and out of hearing.
The router supports file sharing (SMB).


----------



## gimmeheadroom

Hey guys, can anybody recommend a good remote control app for foobar2000, either Windows or Android?


----------



## CAJames

I like MonkeyMote on iOS,  don’t know if it is available on your preferred platforms though.


----------



## gimmeheadroom

CAJames said:


> I like MonkeyMote on iOS,  don’t know if it is available on your preferred platforms though.


Thanks, there is an Android version of this. I'll check it out.


----------



## Hardwired

I just got 'foobar 200 controller pro', which was just made free recently. You need to install an httpd module on foobar but once you do you can control it from the phone nicely.


----------



## gimmeheadroom

Hardwired said:


> I just got 'foobar 200 controller pro', which was just made free recently. You need to install an httpd module on foobar but once you do you can control it from the phone nicely.


Thanks, I'll look at this also.


----------



## desertsilver

Roseval said:


> I solved the noise problem this way.
> As it is a USB HD, I attached it to my router, out of sight and out of hearing.
> The router supports file sharing (SMB).


Thanks for the suggestion, but my HD is a big internal SATA (22T) used as an archive/dump-all drive and not planning on put it as an external. Guess I might put it in a NAS sometime when I get around to build something.

On foobar, I found the "Full file buffering" setting actually works for single images, an ISO image or an album ripped in a single wav or flac. It just bothers me when an album is in multiply tracks, apparently the "Read ahead for local files" buffer is too small.


----------



## Subhasis

You can try MonkeyMote.


----------



## gimmeheadroom

I found this https://sites.google.com/site/foobarcon/ and I'm very happy with it.


----------



## Naguall

Hi, guys !
Is there any way to stream from Deezer, through Foobar ?
(Like you do from internet radio)


----------



## gimmeheadroom

Oi amigo!

I don't think so but maybe somebody will know. Or ask on the hydrogen audio forums.

Why do you want to play Deezer through foobar?


----------



## Naguall

gimmeheadroom said:


> Oi amigo!
> 
> I don't think so but maybe somebody will know. Or ask on the hydrogen audio forums.
> 
> Why do you want to play Deezer through foobar?


Salve, amigo !
Well, I think music has more "punch" in Foobar, compared to Deezer browser player.


----------



## sajunky

Naguall said:


> Salve, amigo !
> Well, I think music has more "punch" in Foobar, compared to Deezer browser player.


Check browser for ability to play using WASAPI Exclusive interface. I know that some can, but don't remember which one, probably Chrome. It will bypass system mixer and play with an original sample rate.

There are trick to capture output using a custom driver, but I don't know whether it will bypass system mixer.


----------



## gimmeheadroom

Naguall said:


> Salve, amigo !
> Well, I think music has more "punch" in Foobar, compared to Deezer browser player.



I understand. Deezer in the browser and even the desktop app is not bitperfect. As @sajunky suggested, check to see if it works better in Chrome.

I got bitperfect to work with Deezer desktop app but only if I know the sample rate. If you get the VB Audio ASIO bridge / Hifi cable app (it's free), and you have a DAC with ASIO drivers, you can get it to work. But since it requires knowing the sample rate all the time, I stopped using it after I got it to work in a test situation.

Here surprisingly Deezer hifi is more expensive than Tidal Hifi Plus. If the price is ok for you I can suggest adding Tidal to your setup because the desktop app is bitperfect.



sajunky said:


> Check browser for ability to play using WASAPI Exclusive interface. I know that some can, but don't remember which one, probably Chrome. It will bypass system mixer and play with an original sample rate.
> 
> There are trick to capture output using a custom driver, but I don't know whether it will bypass system mixer.



Yes, for example the one I mentioned does, I was able to get Deezer desktop app (which is not bitperfect) to talk to my Brooklyn and I was able to decode MQA from Deezer.


----------



## Naguall

gimmeheadroom said:


> I understand. Deezer in the browser and even the desktop app is not bitperfect. As @sajunky suggested, check to see if it works better in Chrome.
> 
> I got bitperfect to work with Deezer desktop app but only if I know the sample rate. If you get the VB Audio ASIO bridge / Hifi cable app (it's free), and you have a DAC with ASIO drivers, you can get it to work. But since it requires knowing the sample rate all the time, I stopped using it after I got it to work in a test situation.
> 
> ...


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## Naguall

Thanks, sajunky and gimmeheadroom, for your answers.  I´ll check Chrome for bitperfect.  I understood you heard difference enabling bitperfect , is that true ?  Anyway, I  thaught maybe there´s some kind of equalization in Dezeer, in browser mode, that makes it sound a little "holow", at least for me. Do you think is it possible ?


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## Naguall

Maybe I´ve had some trouble posting...


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## gimmeheadroom

Naguall said:


> Thanks, sajunky and gimmeheadroom, for your answers.  I´ll check Chrome for bitperfect.  I understood you heard difference enabling bitperfect , is that true ?  Anyway, I  thaught maybe there´s some kind of equalization in Dezeer, in browser mode, that makes it sound a little "holow", at least for me. Do you think is it possible ?



I don't think Chrome will do bitperfect, but it should at least use exclusive mode and sound better.

I don't necessarily hear an improvement using bitperfect, it is just that I never listen to anything that isn't bitperfect. I like to have things the way they should be technically.

There are tone controls in the Deezer desktop app and probably in the browser version also, you should check all your settings and see if you can turn them off. But, the limitation will be in Windows Audio Mixer, it will resample everything to the settings you pick (or defaults) in Sound Control Panel. This often does bad things, which is why @sajunky said it would be good to look into whether Chrome can run exclusive mode.


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## sajunky

There is a command-line flag for Chrome to enable exclusive mode. In some versions of a browser it doesn't do anything, must verify. I don't use Chrome anymore, can't confirm the current version.

Create a desktop shortcut for Chrome, then edit a shortcut command line field, adding to the end a space character and the following:

--enable-exclusive-audio

Veryfication: play music in Chrome (starting Chrome from the shortcut you created), then from the other application try to play something using the same sound device. If you get error, then it works, as Chrome took ownership of the device. A DAC should display the same sample rate as the Web source.


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## mrjayviper

is my DAC running in NOS mode? (using wasapi and connecting to a DAC via USB)

When using non-WASAPI output/playback device (example: default primary sound driver), the DAC shows PCM 384Khz on its display. When using WASAPI, it shows the sampling rate of the music file (usually Redbook specs as I have a lot of them)

Thanks


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## sajunky (Aug 16, 2022)

mrjayviper said:


> is my DAC running in NOS mode? (using wasapi and connecting to a DAC via USB)


NOS mode is subject to the internal settings in your DAC, it doesn't depend on interface or settings on the PC. It means no oversampling and no sound processing (like digital filtering). Depends on your DAC, this term can be missused, it can refer to the specific filtering settings, not every DAC marketed as NOS comply strictly to this definition. In all cases, a DAC display a current sampling rate of the incoming data stream irrespectively to the internal mode of operation.

As I read on, your question is related to PC and bit-perfect transfers:


mrjayviper said:


> When using non-WASAPI output/playback device (example: default primary sound driver), the DAC shows PCM 384Khz on its display. When using WASAPI, it shows the sampling rate of the music file (usually Redbook specs as I have a lot of them)


In the first case PC mixer is set up to upsample everything to 384kHz. This settings is found in the Sound Control Panel. It is not bit-perfect.

In the second case it may be bit-perfect, or not, depends on the output you select in the Foobar. Post screenshot of the output devices in Foobar, I will be able to tell you which output produce bit-perfect transfer. WASAPI can be shared and exclusive. Shared is default in Foobar.


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## mrjayviper

sajunky said:


> In the second case it may be bit-perfect, or not, depends on the output you select in the Foobar. Post screenshot of the output devices in Foobar, I will be able to tell you which output produce bit-perfect transfer. WASAPI can be shared and exclusive. Shared is default in Foobar.



I don't have the laptop with me but it's running on exclusive mode (push I believe)


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## gimmeheadroom

mrjayviper said:


> I don't have the laptop with me but it's running on exclusive mode (push I believe)


Use ASIO whenever possible.


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## gimmeheadroom

Guys, I got DSD 512 to work with foobar and my gear using DSD transcoder but I didn't use it that much. Lately I used it a bit more and it often causes foobar2000 to fail with some kind of message about the process ending. I thought it was because I was running on a small box without enough RAM so I installed it on a more powerful system with more memory and same thing. It is just not reliable for me.

Anybody have yeah or nay experience using DSD transcoder all the time?


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## sajunky

mrjayviper said:


> I don't have the laptop with me but it's running on exclusive mode (push I believe)


Try event mode, if your DAC support it. It helps OS task management. 

As you talk about push mode, it is indicative that you don't use the latest version of Foobar or the WASAPI add-on is installed. In the latest version there is no need to install WASAPI component, as WASAPI access is a default. Direct Sound access (DS) is dropped and a specific 'push' mode is depreciated. It can be enabled in the Options->Advanced section, but it doesn't show as a separate output device anymore. These changes makes output section much cleaner.


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## sajunky

@gimmeheadroom. If I remember correctly you have setup with asioproxy. It is sensitive to the installed version of SACD component. A compatible version is very old, there are many improvements. I suggest uninstall proxy in Control Panel and install a new version, now called DSDTranscoder, there is more chance to work with new software (SACD component and Foobar 1.6). It is covered in "Foobar for Dummies" Part 2, Mode 2.


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## gimmeheadroom

sajunky said:


> @gimmeheadroom. If I remember correctly you have setup with asioproxy. It is sensitive to the installed version of SACD component. A compatible version is very old, there are many improvements. I suggest uninstall proxy in Control Panel and install a new version, now called DSDTranscoder, there is more chance to work with new software (SACD component and Foobar 1.6). It is covered in "Foobar for Dummies" Part 2, Mode 2.



Thank you. I didn't try asioproxy, in fact it was you who alerted me to DSD transcoder and that's how I got the high rate DSD to work. But it doesn't work consistently.

I'll get back to this at some point but it's mostly academic now for several reasons in my setup.


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## Gilberto 62

Hello everyone. Does anyone know if it is possible, in Foobar, to make all the songs in a playlist sound the same, that is, at the same volume, being songs from different albums...? Thanks for your response.


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## gimmeheadroom

Gilberto 62 said:


> Hello everyone. Does anyone know if it is possible, in Foobar, to make all the songs in a playlist sound the same, that is, at the same volume, being songs from different albums...? Thanks for your response.


I don't use it but there is a volume normalization feature. I'm not sure if it needs metadata from each song being played though. Look at replay gain in the settings.


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## castleofargh

Gilberto 62 said:


> Hello everyone. Does anyone know if it is possible, in Foobar, to make all the songs in a playlist sound the same, that is, at the same volume, being songs from different albums...? Thanks for your response.


You can scan your tracks for replaygain data, with some app of sort or right into foobar if you want(if it's one playlist, click on one song then CTLR+A to select everything in that playlist, then right click->ReplayGain->Scan per-file track gain. 

For a big library it can take *a lot* of time and make your computer real hot depending on the resampling settings you have for the scanner(File->Preferences->Tools->ReplayGain Scanner). Basically, the more you oversample a file before scanning, the less you will have intersample clipping. So checking the True peak scan and setting a big oversampling value is technically very good except that it will seriously push your CPU(it was too much for my old PC) and take more time. I'd say that if your house is hot, it might be a really bad idea to use 8X oversampling. 

As a summer alternative you could instead uncheck True peak scan and just set the preamp to something like -3 or -4dB in the 'preamp' with 'RG info'(you can see that in the screenshot of @gimmeheadroom). The scan will go much faster and you'll have 3 or 4dB of headroom to avoid all but the most extreme intersample clips. 

Anyway once that's done, or if your tracks already have per track replaygain info, to tell foobar to use those metadata you again look at the screenshot above and at the top under ReplayGain in 'Source mode' you select 'track' . Next song should be good. 

Under is the 'Processing' option that may or may not be useful to you depending on how you scanned the files and if you have some preamp attenuation or not. The standard option is 'apply gain and prevent clipping according to peak'.  That last option has pros and cons:
1/ If the scan wasn't using oversampling, the peak measured will not be the true(highest) peak in the signal, so it might not totally avoid clipping. 
2/ If some track has positive gain after using the scanner, such a track might have a conflict between the clipping value and the "let's make it sound as loud as the rest" value. Meaning that preventing clipping will make such tracks a good deal quieter than the rest. It's a real issue with very dynamic tracks like some classical music, if you're like me and create degenerate playlists with Mozart, 2PAC, iron maiden and some anime opening and shuffle it all, it's near impossible to have everything sound the same and not clip at the same time with just automated processes.

But for the general concept, what I explained should do the job.


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## sajunky

I want to add to the above posts that you don't need to use oversampling option during scan if your DAC is not oversampling. For all others adjust oversampling rate to the value you normally play music. 

Foobar has hell of options, I didn't investigate all, as I don't use RG, but it is the only player that can use RG fully. Follow a link in the screenshot above, it brings to the Hydrogen audio web site, on the bottom there is a link dedicated to Foobar.


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## delapoer

When I switch Foobar output to my DAC's Exclusive mode I get an error saying: "Unrecoverable error unsupported stream format: 44100 Hz 16 bit 2 channels (0x3)"  why is this happening?


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## gimmeheadroom

delapoer said:


> When I switch Foobar output to my DAC's Exclusive mode I get an error saying: "Unrecoverable error unsupported stream format: 44100 Hz 16 bit 2 channels (0x3)"  why is this happening?


1. Has this ever worked?
2. What did you change?
3. Does it happen only with Foobar? It looks like Windows has broken some audio in the latest updates for Win 10.

Also, show some screenshots of your setup and the error, it might be easier to see what's going on.


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## delapoer

gimmeheadroom said:


> 1. Has this ever worked?
> 2. What did you change?
> 3. Does it happen only with Foobar? It looks like Windows has broken some audio in the latest updates for Win 10.
> 
> Also, show some screenshots of your setup and the error, it might be easier to see what's going on.


I use the non-exclusive version but noticed that there is an Exclusive version so I tried it to see what happened (nothing other than the error message).  The only thing I changed was the Output device in Library, Configurations, Output, nothing else was changed at all.  I only use Foobar and that is the whole extent of the error message.


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## delapoer

delapoer said:


> I use the non-exclusive version but noticed that there is an Exclusive version so I tried it to see what happened (nothing other than the error message).  The only thing I changed was the Output device in Library, Configurations, Output, nothing else was changed at all.  I only use Foobar and that is the whole extent of the error message.


Sorry, no this has never worked, I only tried it recently.


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## gimmeheadroom

Without seeing screenshots of your foobar2000 and soundcardd setup it's impossible to conclude anything.

Are you using onboard sound or an external DAC?

If onboard sound, you should go to sound settings and make sure you have allow applications exclusive use (or some similar verbiage) for your onboard sound card.

If you have an external DAC, does the maker have USB ASIO drivers available and have you installed them?


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## delapoer (Aug 31, 2022)

OK, am looking to see how to attach screenshot.  External DAC Audiolab 8200CDQ.  I believe that Audiolab don't have specific drivers and I did have ASIO drivers installed  but am not sure at the moment after an update, what's the easiest way to tell?  I'm new here and don't know how to attach images.


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## delapoer

Screenshot attached.


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## sajunky

delapoer said:


> Screenshot attached.


Change in Foobar output format from 16-bit to 24-bit. It seems your DAC do not support 16-bit format for 44.1kHz. It is also better to keep it all the time at 24-bits, as high resolution music will be not degraded.

If you go to the Windows Sound Control Panel, supported sample rate combination (as reported by a DAC) are listed in the Advanced tab. You will see confirmation to what I am saying.


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## delapoer

I have changed Foobar to 24 bits and it now works with the Exclusive mode.  This DAC works up to (at least) 24 bit 96 kHz.  Thanks for your help.  Can you point me to an explanation of what the Exclusive mode is/does?


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## Roseval

Maybe this link is of use: https://www.thewelltemperedcomputer.com/KB/WASAPI.htm


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## gimmeheadroom

Audiolab 8300 does have ASIO drivers and has quite a superb USB DAC, perhaps the 8000 does as well. Unfortunately they seem to come only on the mini CD that comes with the device in the box, a few years ago I checked but I didn't find drivers on their site. Perhaps you can now.


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## delapoer

gimmeheadroom said:


> Audiolab 8300 does have ASIO drivers and has quite a superb USB DAC, perhaps the 8000 does as well. Unfortunately they seem to come only on the mini CD that comes with the device in the box, a few years ago I checked but I didn't find drivers on their site. Perhaps you can now.


Sorry, my DAC is an Audiolab 8200 CDQ and works to 96 kHz and 24 bits.  It did not come with any drivers but use of ASIO4all was recommended.  Since the instructions only went to Win 7 I'm wondering which I should use out of ASIO, WASAPI or none of them.  The computer is only used for Foobar so there shouldn't be much in the way of other demands.  Thanks for your reply.


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## delapoer

Roseval said:


> Maybe this link is of use: https://www.thewelltemperedcomputer.com/KB/WASAPI.htm


Thanks I will take a look.


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## castleofargh

Asio4all isn’t actually asio. Asio is recommended because when someone makes a dedicated driver for his DAC, it will typically be an asio driver. The value in my opinion comes from it being a dedicated driver, so Asio4all being a generic solution has no benefit over wasapi.


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## delapoer

castleofargh said:


> Asio4all isn’t actually asio. Asio is recommended because when someone makes a dedicated driver for his DAC, it will typically be an asio driver. The value in my opinion comes from it being a dedicated driver, so Asio4all being a generic solution has no benefit over wasapi.


Thanks a lot for this, can you point me at any info on how best to use WASAPI?


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## redrol

castleofargh said:


> Asio4all isn’t actually asio. Asio is recommended because when someone makes a dedicated driver for his DAC, it will typically be an asio driver. The value in my opinion comes from it being a dedicated driver, so Asio4all being a generic solution has no benefit over wasapi.


Good catch.  ASIO4ALL is only a wrapper around the standard windows audio stack.  Has absolutely none of the advantages of real ASIO which bypasses the windows audio subsystem.  ASIO is made to talk directly to the audio hardware with no OS in the way.


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## castleofargh

delapoer said:


> Thanks a lot for this, can you point me at any info on how best to use WASAPI?


The exclusive mode you’ve tried is it. As you’ve experienced, things tend to work or not work with ”bit perfect” solutions. Once you get music, you’re usually done.


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## gimmeheadroom

delapoer said:


> Sorry, my DAC is an Audiolab 8200 CDQ and works to 96 kHz and 24 bits.  It did not come with any drivers but use of ASIO4all was recommended.  Since the instructions only went to Win 7 I'm wondering which I should use out of ASIO, WASAPI or none of them.  The computer is only used for Foobar so there shouldn't be much in the way of other demands.  Thanks for your reply.


ASIO was developed by Steinberg https://www.steinberg.net a pro audio company, for use in mastering recordings. It is the best possible protocol for bitperfect audio with Windows.

Since you show your address as UK, you might try contacting Audiolab https://www.audiolab.co.uk/contact-us/ to ask about drivers.


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## SpiegelEule

In case you install the Steinberg DAW, you'll get their ASIO-Driver for free. It ships even with the free version of their DAW. It is a good catch for those who use their PC for making music, I really like this driver and its easy setup menu. 

Michael


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## Peti

Hi, I have a question regarding DSD playback via Foobar2k. I have win10 +Woo WA8 and I can send DSD in DoP format to the Woo in Wasapi Event mode. Playback works just fine, however, in the bottom left corner it says "_*DSD128 I 5644800 Hz I 24 bits I Stereo I 11290 kbps*_"

My question is: do I get the dsd data converted to pcm or do I get the DSD via DoP? That "24bits" has got me confused. Thank you in advance!


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## newworld666 (Dec 12, 2022)

Any idea if there is a component like the "UPnP/DLNA Renderer, Server, Control Point" for foobar2000 v2 x64 ?
I would like to browse and play music from my personal Upnp home server when I am in my office as I can do till now with the 2015 component with foobar2000 (I have BubleUpnp on my personnal server at home for external access) ...
Till now; I haven't find any way to browse my upnp server from *foobar2000 v2* (from LAN and WAN).


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## sajunky (Dec 12, 2022)

Peti said:


> Hi, I have a question regarding DSD playback via Foobar2k. I have win10 +Woo WA8 and I can send DSD in DoP format to the Woo in Wasapi Event mode. Playback works just fine, however, in the bottom left corner it says "_*DSD128 I 5644800 Hz I 24 bits I Stereo I 11290 kbps*_"
> 
> My question is: do I get the dsd data converted to pcm or do I get the DSD via DoP? That "24bits" has got me confused. Thank you in advance!


Go to the Preferences -> Display -> Default User Interface
Press "Reset page" button

Or copy/paste a text from my setup in the status bar section:

```
%codec% | $if(%__bitspersample%,%__bitspersample%/)%samplerate% Hz | %channels% | %playback_time%[ / %length%]$if(%__hdcd%,' --------- HDCD: ['PE: %__hdcd_peak_extend% | LLE: %__hdcd_gain% | TF: %__hdcd_transient_filter%']',)
```

For PCM 96kHz 24-bit it display "24/96000 Hz", I wanted that way...
It also include HDCD reporting (if extension is installed and playing HDCD encoded material)


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