Null test 88.2/24 and 44.1/16

[bachish]

Hi everyone!  First post here.

 

By way of introduction, I'm a professional cellist who has been dabbling in audio since I was a kid. I used to record my brother's band to cassette tape using radio shack microphones in the 1980s.  This eventually progressed to the point where I bought my first professional mics and mic pre and began doing my own recordings (a serious hobby since the mid 1990s).  Because I have a finite amount of funds and it gets split in three ways (cello, recording, monitoring and playback systems), I probably don't have as serious an audiophile system as many of you. My priorities are, in this order:  cello - recording equipment - playback system). But I do have some Paradigm Monitor 11s, v2 (tower speakers) that I bought back in the early 2000s, an older but transparent and detailed Harman Kardon Amp from which I bi-wire and bi-amp my Paradigms, a Grace M900 DAC, a Marantz CD player that set me back about $350 years ago, and an acoustically treated room to knock down the early reflections (makes a MASSIVE difference - highly recommended). 

 

Anyway, I posted a little null test I did over at gearslutz.com, the forum/hangout place for recording engineers.  It's not earth shattering but I thought as a first post I would share it here in a slightly different form.

 

For those who don't know, a null test is a way to see if two audio examples are identical and, if not, what the differences between them are.  All you have to do is flip the phase of one of the audio clips, make sure the levels are identical, mix them together, and see if they null.  If the audio is identical between them, the result will be a perfect phase cancellation.  No sound.  A perfect null. 

 

What I did was take an original recording I did of a jazz group at 88.2 Khz/24 bit and down sampled it to 44.1, flipped the phase, and dithered it to 16 bit. I then up-sampled back to 88.2Khz, which results in a 32 bit floating point audio (all processing results in more bits). I then truncated the 32 bit float back to 24 bits.

 

Here is the original file. Sorry the pics are so small. You may have to zoom in but you can see the extra high frequency content above 20KHz

 

 

Here is the down sampled then up-sampled version.  If you zoom in, you can see how the high frequencies are cut off in the down sampled and then re up-sampled version. The high frequencies are not regained despite the up-sampling. If you look carefully, you can see some of the added dither noise in the high frequencies as a band across the top. The other dither is hidden behind the audio.

 

 

 

I took the original 88.2/24 bit recording, loaded it onto a track in my DAW,  loaded the flipped phase version onto another track and mixed them together. The result was dead silence except at extremely high volumes during which I could hear some white noise. They nulled perfectly in the audible range (again, except for the extremely low level noise - mostly the dither noise) and so were identical in that respect. I heard zero music - not even slight residual sounds or artifacts.  Just hiss with the volume cranked full force!

 

You can see the two versions on tracks 1 and 2 and if you zoom in and look carefully, you will notice that the wave forms are mirror images of each other - they are inverted. You will also notice that the meters for the two tracks being compared are higher than the main meter to the right - so there isn't a perfect null.  But remember, the original has extra high frequency content. 

 

 

 

In a spectrogram, the nulled mix shows the dither noise (mainly shaped to the higher and lower frequencies and less in the mid-range) of the twice re-sampled version and the remaining high frequency content of the original 88.2 version - in other words, the only differences between the tracks.  All of the music from 20Hz to 20Khz nulled perfectly and so the original and re-sampled version are a perfect match in that frequency range.

 

 

 

Keep in mind that if any information was lost in the down sample to 44.1 and dithering to 16 bit, it cannot be regained by up sampling again to 88.2 and reverting to 24 bit. It's gone forever from that track.

 

This little test does not answer if the remaining high frequency content has any effect on the listener - that would necessitate another test. But it seems to me, assuming the re-sampling is done properly, that going from 88.2/24 bit to 44.1/16 bit does not change the audio between 20Hz and 20Khz - all stereo imaging, detail, transients, timbre, and anything else you can think of, remain perfectly in tact. If that were not the case, the two audio clips would not null as they did.

 

In a way it's comforting.

 

I hope this isn't seen as being too controversial as a first post!  Just a fun little experiment!

 

Here is the link at gearslutz for larger visuals and a more detailed description, if you are interested.

 

https://www.gearslutz.com/board/mastering-forum/1227563-my-null-test-88-2kh-24-bit-44-1kh-16-bit.html?posted=1#post13474865

 

Cheers

 

 

 

 

 

 

 

 

 

 

 

 

[wgscott]

Thanks. We need more of this kind of analysis   

[bachish]

14 minutes ago, wgscott said:

Thanks. We need more of this kind of analysis   

 

Glad it is useful. Thank you!

[firedog]

Thanks. Not surprising. On my present system I stopped worrying about recording format. I mostly buy what the original was recorded in, if it is available. But most important is a good recording. Redbook can sound perfectly satisfying, and I agree there are certainly lots of examples where the difference between it and Redbook are negligible, if they actually exist.
 

But there could still be reasons hi-res releases would sound better/different to some people, ranging from different mastering, to equipment doing a better job with hi-res. to DAC filtering differences changing the sound slightly between Redbook and hi-res.

There may also be other examples where transients and detail are easier to hear on the high res, even though they are also audible on the Redbook. I've also a couple of times thought I heard "new" detail in a hi-res version, only to then find out it was also audible on the Redbook version. But I didn't notice it on the Redbook till I heard it first on the hi-res version, where it was more apparent, or should I say easier to pick out?
 

Anyway, as a cellist, check out the just released Yo Yo Ma recordings of the six Bach Cello Suites, called "Six Evolutions".

[bachish]

11 minutes ago, firedog said:

Thanks. Not surprising. On my present system I stopped worrying about recording format. I mostly buy what the original was recorded in, if it is available. But most important is a good recording. Redbook can sound perfectly satisfying, and I agree there are certainly lots of examples where the difference between it and Redbook are negligible, if they actually exist.
 

But there could still be reasons hi-res would sound better/different to some people, ranging from equipment sweet spots to filtering differences changing the sound slightly between Redbook and hi-res.
There may also be other examples where transients and detail are easier to hear on the high res, even though they are also audible on the Redbook. I've also a couple of times thought I heard "new" detail in a hi-res version, only to then find out it was also audible on the Redbook version. But I didn't notice it on the Redbook till I heard it first on the hi-res version, where it was more apparent, or should I say easier to pick out?
 

Anyway, as a cellist, check out the just released Yo Yo Ma recordings of the six Bach Cello Suites, called "Six Evolutions".

Yes, there is the issue of how DACs handle Redbook audio regarding the filter. So this little test doesn't address that particular question. But I imagine, if your DAC is well made, and it probably is, that is does a pretty stellar job at 44.1/16 bit audio.  For my own recordings, I often listen to them in the high res original versions. It's partly psychological for me! haha

 

I haven't heard the new recording of the Bach suites. That's his third recording of them, I believe. The suites are something that never completely settle in a cellist's interpretation.  They are always changing. I suppose that is why the album is called 'Six Evolututions'. It's a great title. Thanks for letting me know! 

[mjb]

@bachish In your experience, does a well mic‘d solo cello benifit from 88.2 Khz/24 bit, or is there little to no activity above 22Khz? 

[pkane2001]

On 8/17/2018 at 10:45 PM, bachish said:

Hi everyone!  First post here.

 

By way of introduction, I'm a professional cellist who has been dabbling in audio since I was a kid. I used to record my brother's band to cassette tape using radio shack microphones in the 1980s.  This eventually progressed to the point where I bought my first professional mics and mic pre and began doing my own recordings (a serious hobby since the mid 1990s).  Because I have a finite amount of funds and it gets split in three ways (cello, recording, monitoring and playback systems), I probably don't have as serious an audiophile system as many of you. My priorities are, in this order:  cello - recording equipment - playback system). But I do have some Paradigm Monitor 11s, v2 (tower speakers) that I bought back in the early 2000s, an older but transparent and detailed Harman Kardon Amp from which I bi-wire and bi-amp my Paradigms, a Grace M900 DAC, a Marantz CD player that set me back about $350 years ago, and an acoustically treated room to knock down the early reflections (makes a MASSIVE difference - highly recommended). 

 

Anyway, I posted a little null test I did over at gearslutz.com, the forum/hangout place for recording engineers.  It's not earth shattering but I thought as a first post I would share it here in a slightly different form.

 

For those who don't know, a null test is a way to see if two audio examples are identical and, if not, what the differences between them are.  All you have to do is flip the phase of one of the audio clips, make sure the levels are identical, mix them together, and see if they null.  If the audio is identical between them, the result will be a perfect phase cancellation.  No sound.  A perfect null. 

 

What I did was take an original recording I did of a jazz group at 88.2 Khz/24 bit and down sampled it to 44.1, flipped the phase, and dithered it to 16 bit. I then up-sampled back to 88.2Khz, which results in a 32 bit floating point audio (all processing results in more bits). I then truncated the 32 bit float back to 24 bits.

 

Here is the original file. Sorry the pics are so small. You may have to zoom in but you can see the extra high frequency content above 20KHz

 

 

Here is the down sampled then up-sampled version.  If you zoom in, you can see how the high frequencies are cut off in the down sampled and then re up-sampled version. The high frequencies are not regained despite the up-sampling. If you look carefully, you can see some of the added dither noise in the high frequencies as a band across the top. The other dither is hidden behind the audio.

 

 

 

I took the original 88.2/24 bit recording, loaded it onto a track in my DAW,  loaded the flipped phase version onto another track and mixed them together. The result was dead silence except at extremely high volumes during which I could hear some white noise. They nulled perfectly in the audible range (again, except for the extremely low level noise - mostly the dither noise) and so were identical in that respect. I heard zero music - not even slight residual sounds or artifacts.  Just hiss with the volume cranked full force!

 

You can see the two versions on tracks 1 and 2 and if you zoom in and look carefully, you will notice that the wave forms are mirror images of each other - they are inverted. You will also notice that the meters for the two tracks being compared are higher than the main meter to the right - so there isn't a perfect null.  But remember, the original has extra high frequency content. 

 

 

 

In a spectrogram, the nulled mix shows the dither noise (mainly shaped to the higher and lower frequencies and less in the mid-range) of the twice re-sampled version and the remaining high frequency content of the original 88.2 version - in other words, the only differences between the tracks.  All of the music from 20Hz to 20Khz nulled perfectly and so the original and re-sampled version are a perfect match in that frequency range.

 

 

 

Keep in mind that if any information was lost in the down sample to 44.1 and dithering to 16 bit, it cannot be regained by up sampling again to 88.2 and reverting to 24 bit. It's gone forever from that track.

 

This little test does not answer if the remaining high frequency content has any effect on the listener - that would necessitate another test. But it seems to me, assuming the re-sampling is done properly, that going from 88.2/24 bit to 44.1/16 bit does not change the audio between 20Hz and 20Khz - all stereo imaging, detail, transients, timbre, and anything else you can think of, remain perfectly in tact. If that were not the case, the two audio clips would not null as they did.

 

In a way it's comforting.

 

I hope this isn't seen as being too controversial as a first post!  Just a fun little experiment!

 

Here is the link at gearslutz for larger visuals and a more detailed description, if you are interested.

 

https://www.gearslutz.com/board/mastering-forum/1227563-my-null-test-88-2kh-24-bit-44-1kh-16-bit.html?posted=1#post13474865

 

Cheers

 

 

 

 

 

 

 

 

 

 

 

 

 

Can you please share the audio files you used for this analysis? I’d like to run them through my own software to see if there’s anything interesting it can find.

[bachish]

There definitely are frequencies above 20Khz emanating from a cello.  We also can play pretty high.  I also tend to use ribbon mics for recording the cello for their smooth, silky, analog-like sound and they do roll off around 15Khz.  But looking at a spectrogram they still record info in the ultrasonic range, albeit kind of softly.

 

Here is my position. I know many disagree on this topic.

 

I'm not convinced the ultrasonics make a difference in playback. My own hearing at age 52 is limited to 15Khz.  I think a downsampled file, assuming it is done correctly, has identical musical information as the original file up to around 22Khz as the null test above shows.

 

However, the above null test doesn't show whether a particular converter sounds the same playing the 44.1/16 bit version compared to the high res version. This is harder to test. All the null test shows is that 44.1/16 file is identical to the 88.2/24 bit file in the audible range with added noise. That is what a good sample rate conversion is today, fortunately - a low pass filter with added very low level noise. But how a particular converter handles the two file versions is now the question.

 

My own opinion is, assuming a high quality DAC, that the differences between the original high res and 44/16 versions have to be quite nuanced, if at all audible.  

 

BUT, nonetheless, I do receive some sort of psychological satisfaction from listening to the high res versions of recordings.  It's a geeky thing for me.  I know that the high res file, technically, has all the original info (ultrasonics) and a lower noise floor. And I don't have to wonder if the converter is doing a good job with the lower res version, even though it probably is doing a stellar job. 

 

In other words, why do I like to listen to high res files whenever possible? Because I can and I like to. 

 

 

 

 

 

 

 

 

[bachish]

31 minutes ago, pkane2001 said:

 

Can you please share the audio files you used for this analysis? I’d like to run them through my own software to see if there’s anything interesting it can find.

Sure, they are uploading now to the cloud.

 

Can I send the link privately in a PM? 

 

 

[testikoff]

Shouldn't the spectrum of null-test delta look something like that (well, mine does)?.. 

 

[pkane2001]

34 minutes ago, bachish said:

Sure, they are uploading now to the cloud.

 

Can I send the link privately in a PM? 

 

 

 

Yes, of course.

[Jud]

57 minutes ago, bachish said:

I know that the high res file, technically, has all the original info (ultrasonics) 

 

Insofar as the studio mics were able to capture it, and it wasn't removed by the ADC(s) during sigma-delta modulation or any downsampling.

[semente]

The trumpet harmonics are some 30dB down at 20k and another 10dB or more down at 30k. 40dB is a massive dynamic amplitude but with the typical level of compression of a rock pop track 30k could be brought up to within 10dB. You'd still need mic that would work that far up in the frequency range.

 

[bachish]

47 minutes ago, Jud said:

 

Insofar as the studio mics were able to capture it, and it wasn't removed by the ADC(s) during sigma-delta modulation or any downsampling.

Yes, I'm not saying the wav file has a perfect representation of the original source recorded.

 

I am comparing listening back to either the high res, original version vs the down sampled version.  

 

 

[bachish]

43 minutes ago, pkane2001 said:

 

Yes, of course.

I found where to do it...Wasn't sure the policy here.  I'll let you know when they are done

[bachish]

1 hour ago, testikoff said:

Shouldn't the spectrum of null-test delta look something like that (well, mine does)?.. 

 

I think it depends on what is captured in the recording and the software. The program I used above is a picture of the whole jazz tune in time (at the bottom it shows the elapsed  time).

[testikoff]

47 minutes ago, bachish said:

I think it depends on what is captured in the recording and the software. The program I used above is a picture of the whole jazz tune in time (at the bottom it shows the elapsed  time).

Post the spectrum of your null-test delta signal for comparison.

[Jud]

1 hour ago, bachish said:

Yes, I'm not saying the wav file has a perfect representation of the original source recorded.

 

I am comparing listening back to either the high res, original version vs the down sampled version.  

 

Right, just wanting us to remember that A/D is the flip side of what goes on in a DAC - first converted from analog to digital with sigma-delta modulation, then (with most ADCs) converted to PCM at 352.8 or internally/externally downsampled to the more common resolutions, 192 or 176.4kHz or below.  So even the "hi res original" from the studio is almost always downsampled.

[bachish]

52 minutes ago, testikoff said:

Post the spectrum of your null-test delta signal for comparison.

And the magic word is? ?

 

I did post the spectrum analyzer from Izotope RX. That is how it looks in that program. It is the last pic in my original post.

[bachish]

20 minutes ago, Jud said:

 

Right, just wanting us to remember that A/D is the flip side of what goes on in a DAC - first converted from analog to digital with sigma-delta modulation, then (with most ADCs) converted to PCM at 352.8 or internally/externally downsampled to the more common resolutions, 192 or 176.4kHz or below.  So even the "hi res original" from the studio is almost always downsampled.

Got it! Thanks!

[testikoff]

6 minutes ago, bachish said:

And the magic word is? ?

 

I did post the spectrum analyzer from Izotope RX. That is how it looks in that program. It is the last pic in my original post.

What you posted is a spectrogram, not spectrum...

[bachish]

2 hours ago, testikoff said:

What you posted is a spectrogram, not spectrum...

It's all I got. Sorry

 

Opening it up and looking at it closely again, everything except the high frequency content and perhaps the dither across the top looks to me to be -80db and lower, which is pretty quiet. And it all looks like random noise to me except, again, for the high frequencies at the top. 

 

 

[bachish]

BTW, if you are interested to see the results of sample rate conversions of differing software and hardware, this is a very handy tool,

 

http://src.infinitewave.ca/

 

I used Izotope RX 

[testikoff]

7 hours ago, testikoff said:

Shouldn't the spectrum of null-test delta look something like that (well, mine does)?.. 

 

Oops, forgot to reset my Audacity setting of 16-bit sample precision to 32-bit float...

 

 

 

[pkane2001]

Here's what my own DeltaWave software as able to determine from about a 2.5 minutes extract from the middle of the recording.

 

The comparison is between the original 88.2Khz/24 bit recording and a downsampled version to 44.1Khz/16bit, then upsampled back to 88.2Khz/24 bit, as described by @bachish (thanks for the files!)

 

By the way, the recording sounds excellent! I also listened to the difference between the two files, which is primarily white noise. I was able to hear occasional notes come through buried in the noise, but only with software digital volume adjusted to about 70dB gain and my DAC set to 0dB (max volume).

 

The files were delivered with the upsampled version with inverted phase. Otherwise, the files match perfectly in level and phase, no phase offset or drift was found.

 

First, a comparison of the spectra of both waveforms. Original 88.2/24 is in blue:

 

 

The drop off do to downsampling/upsampling starts around 21800Hz at -92dB down. Here's the zoomed in portion where the cut-off starts. You can also see very tiny differences in level due to dither and computational error during resampling:

 

 

 

Now the actual waveforms overlaid on top each other:

 

 

Stats below show excellent correlation between the two: 76dB correlated null and -75dB difference (rms).

 

Spectrum of the difference of the two files. Well below -115dB in the audible range:

 

Spectrogram of the original 88.2/24 file:

 

And spectrogram of the downsampled/upsampled file:

 

Spectrogram of the differences of the two files:

 

Interesting results in the cepstrum analysis of the two files:

 

 

This shows that the downsampled/resampled file has some ringing/aliasing going on at a number of frequencies that are not present in the original 88.2/24 bit file. Probably the side-effect of the resampling process and filtering applied. I tried to label the main ones. Note that the vertical value (Y axis) is a correlation coefficient. It's an indication of how strong the ringing is in the measured file, while the X coordinate is the frequency at which this ringing was detected.

 

While the artifacts of the resampling process are visible in the Cepstrum plot, they are not at all noticeable in any of the measurements or in listening to the differences between the two files.