Some questions about settings (sample rate, latency, etc.) for an external USB DAC running off of ASIO
Jul 31, 2015 at 12:13 PM Post #16 of 138
   
My suggestion is to experiment and see if any of them sound different to you.

I've tried that.  The problem is it's hard to tell what I'm hearing >_<  I definitely don't notice a difference in sound-quality, because all the different choices are delivering bit-perfect sound output to my DAC, so they'll sound the same on my speakers or headphones.  But some I do seem to notice some latency on, like a delay in the audio when I'm watching a movie.  I guess you're right, I'll just have to experiment.  Thank you for all your advice!
 
Jul 31, 2015 at 12:16 PM Post #17 of 138
  I've tried that.  The problem is it's hard to tell what I'm hearing >_<  I definitely don't notice a difference in sound-quality, because all the different choices are delivering bit-perfect sound output to my DAC, so they'll sound the same on my speakers or headphones.  But some I do seem to notice some latency on, like a delay in the audio when I'm watching a movie.  I guess you're right, I'll just have to experiment.  Thank you for all your advice!

 
If it's video you're worried about, the best route (in my opinion) is to use VLC Media Player and tweak the settings in the software. There are many online guides about the best settings to use. Oh, and it automatically synchronizes the audio and video so there is no error, if that's what you meant by delay.
 
Jul 31, 2015 at 12:40 PM Post #18 of 138
   
If it's video you're worried about, the best route (in my opinion) is to use VLC Media Player and tweak the settings in the software. There are many online guides about the best settings to use. Oh, and it automatically synchronizes the audio and video so there is no error, if that's what you meant by delay.


Meh, I don't like VLC nearly as much as Media Player Classic, which has MORE customizable settings (although as a result is far more complicated), or PotPlayer which has the MOST customizable settings :)  And for live-action movies and shows (I watch a lot of anime, that's why I make the distinction) I prefer to use CyberLink PowerDVD because it has some EXCELLENT proprietary video-enhancement features that can, for example, make 480p look like 720p and such.  PowerDVD happened to come pre-installed on my HP Envy, and is some of the ONLY 3rd-party pre-installed software on this laptop that I didn't choose to get rid of. . .I like it a lot.

So wait, the auto-synchronization, all these players are doing it, right?  So if I seem to notice some slight difference in the sync between different latency settings, it seems so slight because I really am just imagining it, then?
 
Jul 31, 2015 at 1:00 PM Post #19 of 138
Yikes, there is an awful lot of incorrect science out there......
 
1) While the differences between the various sample rates and bit depths may not be obvious to all listeners, or with all equipment, or will all music, I think it's pretty widely agreed that they are audible a significant percentage of the time. Besides that, whenever you convert one sample rate to another you introduce slight alterations to your content. For both of those reasons, it seems to make obvious sense that you want to play your music bit-perfect. (Assuming that you want it to be as close to the original as possible, then you don't introduce extra changes, right?) I would suggest anyone wondering about this to listen for themselves.
 
2) Latency per-se is meaningless for listening to music. If you're in a studio, recording multiple tracks that have to be synchronized, then it is critical that latency (delay) be kept to an absolute minimum. However, when you're listening to music, you aren't going to hear the difference in whether the music plays 3 milliseconds after you hit the Play button or 30 milliseconds after you hit it. The real issue for playback is that you want to minimize variations in timing - and, arguably, systems with the most latency overall are also likely to have latency that varies the most.
 
Luckily for us, most USB DACs these days use an asynchronous USB input, which means that the DAC controls the timing, and the computer really doesn't make much difference.
 
3) The biggest problems for USB audio DACs are things like dropouts (where the computer gets distracted, and forgets to send packets for long enough that the music stops until things catch up). Changing those latency and buffer settings can help in this situation; a bigger buffer takes longer to fill up, and longer to empty; whether this is good or bad will depend on where the bottleneck lies in your particular system. Likewise, with some player software, and some DAC drivers, and some computers, you may find that ASIO works better than WASAPI - or vice versa. However, in the end, it really doesn't matter as long as the bits all get there intact. (Often a particular buffer setting will work better with a particular computer and drivers; however, there is no generally preferable setting; it's simply a matter of trying the different options and finding settings for buffer size and latency that work best on your particular system. DO not assume that the lowest latency - or the highest - will be the best choice for your setup.)
 
I mentioned that most USB DACs are asynch these days. if yours isn't, then, as they say, all bets are off - because, if the DAC doesn't control the timing, then the computer does.... in which case, in order to get the best sound quality, you DO need to optimize everything on the computer to give you the "smoothest" data feed (with the least variation in timing). In general, though, if you're doing it this way, then you're spending a lot of effort to try and match the performance that an asynch input gives you by default.
 
However, do try to avoid sweeping generalized assumptions. ASIO is NOT "better" or "worse" than WASAPI; it's more accurate to say that, on some computers, and with some DAC drivers, one or the other works better.
 
Jul 31, 2015 at 1:16 PM Post #20 of 138
  Yikes, there is an awful lot of incorrect science out there......
 
1) While the differences between the various sample rates and bit depths may not be obvious to all listeners, or with all equipment, or will all music, I think it's pretty widely agreed that they are audible a significant percentage of the time. Besides that, whenever you convert one sample rate to another you introduce slight alterations to your content. For both of those reasons, it seems to make obvious sense that you want to play your music bit-perfect. (Assuming that you want it to be as close to the original as possible, then you don't introduce extra changes, right?) I would suggest anyone wondering about this to listen for themselves.
 
2) Latency per-se is meaningless for listening to music. If you're in a studio, recording multiple tracks that have to be synchronized, then it is critical that latency (delay) be kept to an absolute minimum. However, when you're listening to music, you aren't going to hear the difference in whether the music plays 3 milliseconds after you hit the Play button or 30 milliseconds after you hit it. The real issue for playback is that you want to minimize variations in timing - and, arguably, systems with the most latency overall are also likely to have latency that varies the most.
 
Luckily for us, most USB DACs these days use an asynchronous USB input, which means that the DAC controls the timing, and the computer really doesn't make much difference.
 
3) The biggest problems for USB audio DACs are things like dropouts (where the computer gets distracted, and forgets to send packets for long enough that the music stops until things catch up). Changing those latency and buffer settings can help in this situation; a bigger buffer takes longer to fill up, and longer to empty; whether this is good or bad will depend on where the bottleneck lies in your particular system. Likewise, with some player software, and some DAC drivers, and some computers, you may find that ASIO works better than WASAPI - or vice versa. However, in the end, it really doesn't matter as long as the bits all get there intact. (Often a particular buffer setting will work better with a particular computer and drivers; however, there is no generally preferable setting; it's simply a matter of trying the different options and finding settings for buffer size and latency that work best on your particular system. DO not assume that the lowest latency - or the highest - will be the best choice for your setup.)
 
I mentioned that most USB DACs are asynch these days. if yours isn't, then, as they say, all bets are off - because, if the DAC doesn't control the timing, then the computer does.... in which case, in order to get the best sound quality, you DO need to optimize everything on the computer to give you the "smoothest" data feed (with the least variation in timing). In general, though, if you're doing it this way, then you're spending a lot of effort to try and match the performance that an asynch input gives you by default.
 
However, do try to avoid sweeping generalized assumptions. ASIO is NOT "better" or "worse" than WASAPI; it's more accurate to say that, on some computers, and with some DAC drivers, one or the other works better.


Thank you very much for this enlightening little mini-essay!  I actually do agree with you that there are audible differences when it comes to bit-depth, and it's not like I have Golden Ears or anything even close to that level of refined hearing-ability.  I've listened to Mozart's and Beethoven's symphonies in 16-bit/44.1 files before, and now that I have them in 24/96, I DEFINITELY notice a greater level of detail and subtlety in them, especially in certain sections like the woodwinds, and the overall instrumental separation.  Obviously with less complex musical genres I don't necessarily notice as much of a difference, but believe it or not, I DO even notice a difference between 16/44.1 and 24/96 in stuff like rap by the Beastie Boys!  The song "Intergalactic," especially, for some reason sounds noticably better in 24/96 :)  So, are you confirming my suspicion that for the best possible listening experience, I should always go into the Windows Advanced Sound and change the digital output to my DAC to precisely match the bit-depth and sample-rate to those of the file being played?  It seems you are, but I want to make sure.
 
The DAC I am using is my Fiio X3 2nd Gen that doubles as a DAC, and yes, it is asynchronous, as it should be :)  And I've never had any issues with it cutting-out, so I guess I'll just leave the settings as they are. . .middle-of-the-road (standard) latency setting, middle-of-the-road (2048 samples, max available is 8192) buffer size.

I'm still wondering about movies and shows, though.  Is it true that all media players (PotPlayer, Media Player Classic, PowerDVD) will automatically sync the video to the audio to make up for any potential latency issues, even when using an external DAC?
 
Jul 31, 2015 at 2:20 PM Post #21 of 138
  Meh, I don't like VLC nearly as much as Media Player Classic, which has MORE customizable settings (although as a result is far more complicated), or PotPlayer which has the MOST customizable settings :)  And for live-action movies and shows (I watch a lot of anime, that's why I make the distinction) I prefer to use CyberLink PowerDVD because it has some EXCELLENT proprietary video-enhancement features that can, for example, make 480p look like 720p and such.  PowerDVD happened to come pre-installed on my HP Envy, and is some of the ONLY 3rd-party pre-installed software on this laptop that I didn't choose to get rid of. . .I like it a lot.

So wait, the auto-synchronization, all these players are doing it, right?  So if I seem to notice some slight difference in the sync between different latency settings, it seems so slight because I really am just imagining it, then?

 
Oh, realllly?
ksc75smile.gif

 
http://myanimelist.net/animelist/MusicAlchemist
 
I've spent something like 1% of my entire life watching anime. Feel free to PM me about anime!
 
Looks like you're set for the software; just gotta figure out the specifics.
 
  Thank you very much for this enlightening little mini-essay!  I actually do agree with you that there are audible differences when it comes to bit-depth, and it's not like I have Golden Ears or anything even close to that level of refined hearing-ability.  I've listened to Mozart's and Beethoven's symphonies in 16-bit/44.1 files before, and now that I have them in 24/96, I DEFINITELY notice a greater level of detail and subtlety in them, especially in certain sections like the woodwinds, and the overall instrumental separation.  Obviously with less complex musical genres I don't necessarily notice as much of a difference, but believe it or not, I DO even notice a difference between 16/44.1 and 24/96 in stuff like rap by the Beastie Boys!  The song "Intergalactic," especially, for some reason sounds noticably better in 24/96 :)  So, are you confirming my suspicion that for the best possible listening experience, I should always go into the Windows Advanced Sound and change the digital output to my DAC to precisely match the bit-depth and sample-rate to those of the file being played?  It seems you are, but I want to make sure.
 
The DAC I am using is my Fiio X3 2nd Gen that doubles as a DAC, and yes, it is asynchronous, as it should be :)  And I've never had any issues with it cutting-out, so I guess I'll just leave the settings as they are. . .middle-of-the-road (standard) latency setting, middle-of-the-road (2048 samples, max available is 8192) buffer size.

I'm still wondering about movies and shows, though.  Is it true that all media players (PotPlayer, Media Player Classic, PowerDVD) will automatically sync the video to the audio to make up for any potential latency issues, even when using an external DAC?

 
If you like, I can show you how to convert your 24-bit files to 16-bit / 44.1 kHz, so you can hear for yourself that there is no audible difference. The only reason it sounds different is because you are listening to two masters of the recording. You're not comparing two resolutions; you're comparing two different masters. 16-bit already has more than enough dynamic range for all recordings. 24-bit just adds more dynamic range, but has no audible benefit in terms of audio playback. And humans can't hear frequencies above roughly 20 kHz, so a higher sample rate also has no benefit. I again suggest that you read the article I originally linked you to, to give you a much better understanding of the technical stuff.
 
Most DACs automatically select the right settings for you. Although some systems can make things sound different with different settings, it's not common.
 
Not all video players sync the audio and video all the time. If you are hearing a delay in the sense of audio playing too early or too late in relation to the video, then you have a problem to sort out.
 
Jul 31, 2015 at 2:30 PM Post #22 of 138
   
Oh, realllly?
ksc75smile.gif

 
http://myanimelist.net/animelist/MusicAlchemist
 
I've spent something like 1% of my entire life watching anime. Feel free to PM me about anime!
 
Looks like you're set for the software; just gotta figure out the specifics.
 
 
If you like, I can show you how to convert your 24-bit files to 16-bit / 44.1 kHz, so you can hear for yourself that there is no audible difference. The only reason it sounds different is because you are listening to two masters of the recording. You're not comparing two resolutions; you're comparing two different masters. 16-bit already has more than enough dynamic range for all recordings. 24-bit just adds more dynamic range, but has no audible benefit in terms of audio playbaack. And humans can't hear frequencies above roughly 20 kHz, so a higher sample rate also has no benefit. I again suggest that you read the article I originally linked you to, to give you a much better understanding of the technical stuff.
 
Most DACs automatically select the right settings for you. Although some systems can make things sound different with different settings, it's not common.
 
Not all video players sync the audio and video all the time. If you are hearing a delay in the sense of audio playing too early or too late in relation to the video, then you have a problem to sort out.

Okiedokies, thanks :)
 
Jul 31, 2015 at 2:35 PM Post #23 of 138
  Okiedokies, thanks :)

 
I'm going to assume this is confirming you want a tutorial.
 
Download and install the free trial of dBpoweramp. Right-click a 24-bit audio file and click Convert To. Next to Converting To, select either Wave or AIFF. In the settings, select 16-bit and 44.1 kHz. Configure the output location, then click Convert. You can rename the file(s) to make things easier.
 
Alternately, you could use foobar2000 to do the conversion.
 
Jul 31, 2015 at 2:48 PM Post #24 of 138
   
I'm going to assume this is confirming you want a tutorial.
 
Download and install the free trial of dBpoweramp. Right-click a 24-bit audio file and click Convert To. Next to Converting To, select either Wave or AIFF. In the settings, select 16-bit and 44.1 kHz. Configure the output location, then click Convert. You can rename the file(s) to make things easier.
 
Alternately, you could use foobar2000 to do the conversion.


Ah, there's conversion available in Foobar?  I'll just use that, then.  Thanks for teh tips.  I'll try down-sampling some 24/96 files to 16/48 or 16/44.1 this evening after going grocery shopping, and see if I can notice any difference :p
 
Jul 31, 2015 at 3:06 PM Post #25 of 138
  Oh, realllly?
ksc75smile.gif

 
http://myanimelist.net/animelist/MusicAlchemist
 
I've spent something like 1% of my entire life watching anime. Feel free to PM me about anime!
 
Looks like you're set for the software; just gotta figure out the specifics.

when you can't win with skill, beat them with experience and old age http://myanimelist.net/animelist/castleofargh&show=0&order=4
that makes more nolife than you! (ok maybe I shouldn't be too proud of that). ^_^ but going older I'm reverting back to mangas, also when you know the manga, the anime is spoiled. so the more manga I read, the less anime I want to watch. first world problem!
 
  I'm still wondering about movies and shows, though.  Is it true that all media players (PotPlayer, Media Player Classic, PowerDVD) will automatically sync the video to the audio to make up for any potential latency issues, even when using an external DAC?

they will automatically do nothing at all. but it's not like you would change your buffering settings or DSP chain 5times a day, so you set a little lag to compensate for it all once and you're done.
I used some crap running with convolver and DSPs to make action movies even more Michael Bay-ish at some point (you can use a few stuff already integrated to KMPpotplayer). with viper4android I had some noticeable delay, so I just used the audio sync option in potplayer untill I didn't notice it anymore.
I feel like you're making a mountain of something that doesn't matter. try to solve problems when you actually have them
wink_face.gif
. as for your DAC I told you, if the driver the manufacturer offered uses asio, then use that.
 
Jul 31, 2015 at 3:10 PM Post #26 of 138
  when you can't win with skill, beat them with experience and old age http://myanimelist.net/animelist/castleofargh&show=0&order=4
that makes more nolife than you! (ok maybe I shouldn't be too proud of that). ^_^ but going older I'm reverting back to mangas, also when you know the manga, the anime is spoiled. so the more manga I read, the less anime I want to watch. first world problem!

 
hehe, yeah, I saw your MAL awhile ago (when we talked about it) and was just like "Duuuude." I'll need to add some of the stuff on yours to my gargantuan plan to watch list.
 
But wait...didn't you give Elfen Lied a really low score before? It has a 10 now.
confused_face_2.gif

 
Jul 31, 2015 at 3:20 PM Post #27 of 138
.....
I'm still wondering about movies and shows, though.  Is it true that all media players (PotPlayer, Media Player Classic, PowerDVD) will automatically sync the video to the audio to make up for any potential latency issues, even when using an external DAC?

 
In general, when playing a video, it's the video processing and decoding that takes up most of the processing power. Since separating the audio and video from each other, as well as the video processing itself, is done by your player program, the hard part is making sure that it sends them both out to the computer in synch. If the player program gets that part right, any slight additional latency added by the DAC itself is going to be insignificant by comparison. (And, if the video takes too much processing power, and falls behind, the only really solution is to find a more capable video player or get a faster computer - the DAC won't be able to help.)
 
Jul 31, 2015 at 3:28 PM Post #28 of 138
 
Thank you very much for this enlightening little mini-essay!  I actually do agree with you that there are audible differences when it comes to bit-depth, and it's not like I have Golden Ears or anything even close to that level of refined hearing-ability.  I've listened to Mozart's and Beethoven's symphonies in 16-bit/44.1 files before, and now that I have them in 24/96, I DEFINITELY notice a greater level of detail and subtlety in them, especially in certain sections like the woodwinds, and the overall instrumental separation.  Obviously with less complex musical genres I don't necessarily notice as much of a difference, but believe it or not, I DO even notice a difference between 16/44.1 and 24/96 in stuff like rap by the Beastie Boys!  The song "Intergalactic," especially, for some reason sounds noticably better in 24/96 :)  So, are you confirming my suspicion that for the best possible listening experience, I should always go into the Windows Advanced Sound and change the digital output to my DAC to precisely match the bit-depth and sample-rate to those of the file being played?  It seems you are, but I want to make sure.
 
The DAC I am using is my Fiio X3 2nd Gen that doubles as a DAC, and yes, it is asynchronous, as it should be :)  And I've never had any issues with it cutting-out, so I guess I'll just leave the settings as they are. . .middle-of-the-road (standard) latency setting, middle-of-the-road (2048 samples, max available is 8192) buffer size.

I'm still wondering about movies and shows, though.  Is it true that all media players (PotPlayer, Media Player Classic, PowerDVD) will automatically sync the video to the audio to make up for any potential latency issues, even when using an external DAC?

 
By default, Windows expects you to set a particular sample rate, and plays everything at that sample rate - and setting that sample rate to match the files you're playing would allow you to play them back without conversion. However, there's an easier way. Windows has a special audio mode called WASAPI; and, when you use WASAPI mode, each file will automatically be played back at its native sample rate. The catch is that not all programs support WASAPI mode, including most of the ones that come with Windows, (although all "real audiophile music players" do).
 
If you really want the best sound quality when playing high-res music, the easiest thing is to switch to a program that does support WASAPI mode. (Incidentally, WASAPI mode will take care of matching the sample rate; the bit depth generally takes care of itself in most cases.)
 
If you prefer not to do that, the next easiest thing is to set your default output sample rate to either 24/96 or 24/192 and just leave it there.... Basically, this will play high sample rate files at their proper sample rate, and up-sample those recorded at a lower sample rate (which, while it can't add information, and so can't actually improve them, generally won't do much damage).
 
Jul 31, 2015 at 3:43 PM Post #29 of 138
   
By default, Windows expects you to set a particular sample rate, and plays everything at that sample rate - and setting that sample rate to match the files you're playing would allow you to play them back without conversion. However, there's an easier way. Windows has a special audio mode called WASAPI; and, when you use WASAPI mode, each file will automatically be played back at its native sample rate. The catch is that not all programs support WASAPI mode, including most of the ones that come with Windows, (although all "real audiophile music players" do).
 
If you really want the best sound quality when playing high-res music, the easiest thing is to switch to a program that does support WASAPI mode. (Incidentally, WASAPI mode will take care of matching the sample rate; the bit depth generally takes care of itself in most cases.)
 
If you prefer not to do that, the next easiest thing is to set your default output sample rate to either 24/96 or 24/192 and just leave it there.... Basically, this will play high sample rate files at their proper sample rate, and up-sample those recorded at a lower sample rate (which, while it can't add information, and so can't actually improve them, generally won't do much damage).


Will Kernel Streaming or ASIO also play files at their native sample rate, just like WASAPI, or what?

Why then, when I use WASAPI, does the display on my Fiio X3 (in DAC mode) still always say it's receiving input at the default Windows rate (16/44.1) rather than the rate of the file being played?
 

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