How to test "sound quality"of a Player software with objective method?

[Huang1997]

What i think is to test the Digital signal output after the player decodes. But how to determine the quality of the digital signal,what is the standard ?

 

[Kal Rubinson]

What i think is to test the Digital signal output after the player decodes.

Wouldn't the decoded signal be analog?

 

[kemmler3D]

The simplest way would be to compare the original file to the output of your software player using Deltawave or a similar program. It takes the two files and finds the difference between them. Ideally, and in general, there won't be any with the output of a software audio player. https://www.audiosciencereview.com/...test-deltawave-null-comparison-software.6633/

 

[GXAlan]

You don’t have to worry too much. For PCM, there really isn’t any decoding.

For DSD, different products may apply different amounts of dithering or high frequency filtration.

I will continue to advocate for Sony Music Center for PC. It allows you to enable DSEE HX and it’s free. At least DSEE is a measurable difference (though audibility is subject to debate).

 

[Huang1997]

最简单的方法是使用 Deltawave 或类似程序将原始文件与软件播放器的输出进行比较。它获取两个文件并找出它们之间的差异。理想情况下，一般来说，软件音频播放器的输出不会有任何输出。https://www.audiosciencereview.com/...test-deltawave-null-comparison-software.6633/

but how do you be certain that deltawave can decode the original file perfectly to be a standard?

 

[amirm]

You can get an audio interface with loopback and record the output and then compare it to input. There is also a driver in Windows that acts that way. I forget the name just now.

 

[Sokel]

There is also a driver in Windows that acts that way. I forget the name just now.

WASAPI does that,exposes the loopback feature (with the right devices,not to all of them)

 

[Huang1997]

Wouldn't the decoded signal be analog?

no ,The decoded signal is converted into an analog signal after undergoing DAC digital to analog conversion

 

[GXAlan]

but how do you be certain that deltawave can decode the original file perfectly to be a standard?

DeltaWave is written by @pkane and over the years, there has been all sorts of mathematical comparisons even against expensive paid-for products so it’s not a black box in terms of software.

 

[voodooless]

but how do you be certain that deltawave can decode the original file perfectly

If we’re talking about lossless compression codecs, because there is mathematical proof that they work, and the software themselves has plenty of unit and regression tests to validate its functionality.

Lossy compression is more difficult, but similar methods apply. You just never get the original back.

 

[Kal Rubinson]

no ,The decoded signal is converted into an analog signal after undergoing DAC digital to analog conversion

??? If the signal is a digital file, e.g., PCM, it is read by the player and output to the DAC for conversion to analog. Are you referring to the digital (PCM) stream between player and DAC with the suggestion that there are potential errors/distortion in reading the file?

 

[DVDdoug]

Usually, everything in the digital domain is fine.... Almost all "sound quality problems" are on the analog side (before or after digitizing) or they are in the recording/production.

And the biggest/worst "analog problems" are related to speakers & acoustics.

Assuming you're not using the player's EQ or other DSP the software simply sends the digital audio data to the DAC and it has no effect on "sound quality'.

Digital volume control means you aren't using all of the bits so it lowers resolution, but it's the same will all players and any digital volume adjustment.

Oh... That's also assuming that there are no glitches, gaps, interruptions of the audio, etc. But, that kind of thing is usually caused by the multitasking operating system, not the player software.

"Decoding" isn't much of an issue. If you have MP3 or other lossy compression, the encoding (compression) is the hard part where data is thrown-away to make a smaller file. Decoding (decompression) is pretty-well standardized and no data is being thrown-away during the process.

In some cases the drivers will down-sample... For example, if you are playing a high-resolution file on a "cheap" 16-bit soundcard it will be automatically down-sampled. Again, that has nothing to do with the player application.

 

[Chrispy]

no ,The decoded signal is converted into an analog signal after undergoing DAC digital to analog conversion

What is the particular format of digital signal (codec?) you're concerned with?

 

[Ken Tajalli]

??? If the signal is a digital file, e.g., PCM, it is read by the player and output to the DAC for conversion to analog. Are you referring to the digital (PCM) stream between player and DAC with the suggestion that there are potential errors/distortion in reading the file?

I think he is referring to digital convertion to PCM, from flac, wavepack etc. and whether it is actually outputting in bit-perfect or not.
As @voodooless has hinted, sometimes the original is a lossy format, being converted to PCM.

 

[popej]

You can get an audio interface with loopback and record the output and then compare it to input. There is also a driver in Windows that acts that way. I forget the name just now.

VB-Cable, Virtual Audio Cable (VAC).

 

[thulle]

VB-Cable, Virtual Audio Cable (VAC).

@Huang1997 Something like this to a recorder, then deltawave to compare to another decoding of the same file, or some bitperfect PCM. Whatever you want to use as reference.

In linux I can go straight from the music player to a recorder and dump it to an uncompressed file:

 

[GXAlan]

I think he is referring to digital convertion to PCM, from flac, wavepack etc. and whether it is actually outputting in bit-perfect or not.
As @voodooless has hinted, sometimes the original is a lossy format, being converted to PCM.

One nice, empiric way to verify this is to get a DTS-CD, and see if it is properly detected downstream. Any error from bit perfect will result in the DTS decoder failing.

That’s how you prove it to yourself.

On PC, you can just use Foobar2000 and be reassured. Ignoring DTS (haven’t tried), Sony Music Center is a really good option if you don’t want a community/homebrew style solution. Sony offers this for free and most importantly it also has DSEE HX which can add extra harmonics to season the sound. That is fully defeatable.

There are apps for Mac like Colibri.

Of course there is J. River.

For DSD, there is no bit perfect conversion to PCM. That is because there is variability in:
1) sample rate. Do you drop to 88 or 176 kHz. Pros and Cons for both.
2) Ultrasonic filtration. 40 kHz? 50 kHz? What slope?
3) Dithering? No, Yes — but if so, which pattern?

But even if there is no mathematical consistency, it’s very difficult to hear any differences between all of those 3 options I mentioned.

Sony is insistent in converting DSD2.8 to 24-176. They preserve all of the HF info. Oppo used to go to 24/88 only (same with Arcam). The argument for doing 24/88 is that it naturally filters out above 44 kHz. The argument for 24/176 is that it preserves more of the noise that’s higher up. You also have DACs that perform better when fed 24-176 instead of 24-88.

 

[Huang1997]

What is the particular format of digital signal (codec?) you're concerned with?

pcm and dsd

 

[Huang1997]

??? 如果信号是数字文件，例如PCM，则播放器读取该信号并将其输出到DAC以转换为模拟信号。您是否指的是播放器和 DAC 之间的数字 (PCM) 流，并建议读取文件时存在潜在错误/失真？

yes, thats what i mean

 

[Chrispy]

pcm and dsd

And how/where from are you getting the digital signal?