[mike1127]

I'm interested in the best way to play sound files so that I know the software is not corrupting the samples by putting them through any volume processing or other processing. I read in Stereophile that foobar2000 is the only way to do this properly. I installed foobar2000. I'm curious if anyone knows more about going into the foobar2000 preferences and choosing the type of audio interface for best sound (like ASIO, direct-x, etc.)
 
-Mike
 

 

[Audiophile1811]

Good question. However, I do think the default interface works well for most people. 

 

[hduong]

I always wonder if people can tell the difference between ASIO and Windows Default Sound.  They both sound the same to me.

 

[Portioli]

it depends....
 
what filies you are listening to and what speakers or headphones you have.
 
using only the WASAPI  plug-in can have improvement in every system
 

 

[Drake22]

According to this test here http://vst-asio.ru/news/2010-02-18-8

cplay and xmplay are supposedly "better"

 

[mike1127]

Quote:

it depends....
 
what filies you are listening to and what speakers or headphones you have.
 
using only the WASAPI  plug-in can have improvement in every system
 

 
I think this is the key to what I am looking for... a guarantee of bit-exact playback. Some of my files are lossy but let's assume I care most about lossless files. My soundcard can handle 24 bit/ 96 kHz and so can my DAC, although I have only downloaded or purchased 16/44.1 files.
 

 

[patsyleung]

Without getting into technical details, I use the following settings gathered from various threads:
 
Windows:
Set soundcard levels to 100%
Disable any sound enhancements
Set to sampling rate to match source material (16/44.1, 24/96, etc.)
Configure soundcard to 2 channel with full-range L and R
 
foobar2000:
Use WASAPI or ASIO so that your audio stream from the program has exclusive rights to the soundcard
Clear the DSP field
Set output device to WASAPI:[your_device] or whatever kernel streaming method you are using
Set buffer to minimum without dropping out (eg. 800ms)
Set sampling rate to match source material
Turn off dithering
Set playback priority to 7 (1-7, 7-highest) to prevent other processing threads causing dropouts
 

 

[Audiophile1811]

Quote:

 
My soundcard can handle 24 bit/ 96 kHz and so can my DAC, although I have only downloaded or purchased 16/44.1 files.
 

If I buy a DAC that accepts 24Bit/96KHz, will there be any sonic difference between my music files played at 16Bit/44.1KHz (my entire music library) or would it be best at the the highest sample rate and bits per sample the DAC accepts?
Thanks,
Andrew (Confused with DACs)

 

[Marcus_C]

http://www.head-fi.org/wiki/foobar2000-a-guide-to-set-up#user_anchor5
 
If it's any help.

 

[Portioli]

when the original file is in 16/44.1 (CD format) that most mp3s are, your player will keep this bitrate/samplerate
if you buy a    24Bit/96KHzDAC ,your dac will also have the ability to play the flac files with this bitrate.
with your standard soundcrad your player downsample these files to the 16/44.1
Only the newest Realtek on board soundcards can outplay this kind of format. 
if you follow patsyleung instuctions you will have the opportunity to listen to the   24Bit/96KHz, using your dac, at original format without downsampling with better results
 

 

[Roseval]

Quote:

If I buy a DAC that accepts 24Bit/96KHz, will there be any sonic difference between my music files played at 16Bit/44.1KHz (my entire music library) or would it be best at the the highest sample rate and bits per sample the DAC accepts?
Thanks,
Andrew (Confused with DACs)

 
As usual it depends on the implementation.
Does this DAC plays all at its native sample rate?
Does it upsample to 96?
Does apply ARC to e.g. 110?
Etc, etc.
 
In general you can only try
Play 16/44 - feed it to the DAC
Play 16/44 - upsample it on the PC to 24/96 and feed it to the DAC
Compare
 
You can find as much claims that upsampling in hardware beats upsampling in software as the reverse.