Is High-Resolution Audio Irrelevant? - Page 22 - AVS Forum
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post #631 of 640 Old 06-09-2014, 02:19 AM
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Quote:
Originally Posted by steveting99 View Post


Some may argue that half-waveform resolution is sufficient as you would simply inverse at the zero crossover (180 deg) point to re-construct the original signal.
Due to the addition of odd and even harmonics into the waveform I've assumed it to be non-symmetrical. I.e. the top half may not be the same as the bottom half in the worst case scenario. So to re-construct the entire waveform, the peak to peak full resolution is required. I thought this would be a more conservative way of estimating / determining the dynamic range.

There is another way to look at this situation.

If a wave is a perfect sine wave with no harmonics, then it is symmetrical over its positive and negative halves.

If a wave is not symmetrical, then it carries both the fundamental and some harmonics.

The situation with a 44.1 KHz sampled wave where the signal is represented by 2 but less than 4 points covers the octave covers the range from 11.025 KHz to 22.05 KHz. However, there is a brick wall filter at 22.05 KHz so none of the harmonics of those signals can be reproduced. Therefore it is the brick wall filter that answers the question as to whether or not signals in this range are reproduced with any harmonics. It can't happen! A 44.1 KHz sampled system can reproduce only pure sine waves above 11.025 KHz because the first harmonic of a wave at any higher frequency than 11.025 KHz is above the brick wall frequency of 22.05 KHz.
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post #632 of 640 Old 06-11-2014, 11:58 PM
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^arnyk,

I think you've got much more understanding of limitations brick wall filters employed in the recording process than me. From what you're saying the implementations of these filters would limit the harmonic frequencies from 11kHz upwards? How about IMD from the lower frequencies imposing on the waveforms from below 11kHz?

I note that most human hearing is most sensitive around 2kHz to 6kHz. So this is the range where everything is supposed to be reproduced as clean as possible.

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post #633 of 640 Old 06-12-2014, 05:25 AM
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Quote:
Originally Posted by eljr View Post
Quote:Originally Posted by WiWavelength 

Trust me,

AJ


LOL, thought this was an objective science forum?
Figure of speech, no?

The basic statement agrees with objective truth.
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post #634 of 640 Old 06-12-2014, 05:35 AM
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Quote:
Originally Posted by steveting99 View Post
^arnyk,

From what you're saying the implementations of these filters would limit the harmonic frequencies from 11kHz upwards?
Yes.

Quote:
How about IMD from the lower frequencies imposing on the waveforms from below 11kHz?
That, too. The thing about IM is that the original signals stimulating it can be at just about any frequency, audible or not.

Quote:
I note that most human hearing is most sensitive around 2kHz to 6kHz. So this is the range where everything is supposed to be reproduced as clean as possible.

Agreed. The thing about IM is that signals all over the frequency spectrum can create IM products in this range. All you need is two frequencies that are 2 to 6 KHz apart. So they could be 8 KHz and 10 KHz (for a 2 KHz difference) or it could be 50 KHz and 56 Khz (fpor 6 Khz difference).

The rules for creating IM have more opportunities for distortion creation than harmonic distortion. Harmonic distortion is easy, its always integer multiples. The IM products can be at sums, differences, sums of harmonics, differences of harmonics, all sorts of things.
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post #635 of 640 Old 06-14-2014, 03:52 AM
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Is HD audio irrelevant? - an experiment

Take a look at the result of this experiment (I used Adobe Audition):

What I did:

1. Pick an HD track that is sampled at, say, 96k/24bit.
2. Convert it to 44.1/16bit PCM (.wav)
3. Upsample the .wav file to the original 96k.
4. Invert one of the waves, say, the converted and upsampled one.
5. Mix the two tracks and listen to the result.
VERY IMPORTANT! You must align the tracks PERFECTLY for the experiment to work.

The results:

First, take a look at the spectrograms:

Original 96k/24bit track

Converted 44.1k/16bit track upsampled to 96k.

Mixed track (What you see is what you cut when you downsample the track. There is certainly a huge amount of data there!)

Now listen to the mixed track (that's exactly what you are losing when you convert this track from HD to CD).
After all we know about the physiology of our sense of hearing, could we have expected a different result?

My conclusion: neither bit depth further than 16bit nor high frequency sampling have anything to offer to improve the audible part of the signal. It's high time we made a clear distinction between data loss and degraded sound quality.

Last edited by gabebcn; 06-14-2014 at 03:57 AM.
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post #636 of 640 Old 06-14-2014, 05:19 AM
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By the way, read this excellent article on the subject, also referred to by Scott Wilkinson.
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post #637 of 640 Old 06-22-2014, 01:15 PM
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I think high-resolution audio is irrelevant for listening. For listening, I don't see any reason to go higher than 16/48. For mastering, that is where I could see using something like 24/176.4 or 24/192, etc.
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post #638 of 640 Old 07-12-2014, 07:26 AM
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Quote:
Originally Posted by gabebcn View Post
My conclusion: neither bit depth further than 16bit nor high frequency sampling have anything to offer to improve the audible part of the signal. It's high time we made a clear distinction between data loss and degraded sound quality.
That is certainly the theory. In practice my recent testing shows otherwise. Arny was kind enough to give us a file with good bit of high frequency information (keys jingling) that he then resampled to 44.1 Khz and 32 Khz. The files were then converted back to the original bit depth and sample rate of 24/96 Khz. Arny's position was that not only 44.1 Khz was transparent but so was the 16 Khz as there is "no useful information" above that. He said he had run both blind and sighted tests and no one could tell the difference.

I performed a double blind using Foobar2000 with its ABX plug-in. Here are the results as I posted here:

Quote:
Originally Posted by amirm View Post
I don't know that I can even hear 14 Khz! Yet these are my results as i post in the other thread: Debate Thread: Scott's Hi-res Audio Test

--------------


Good morning Arny. I was going to say "thank you" for posting these files but after having to listen to jingling keys so many times while our two dogs barked and barked away, not sure I am that thankful .

Here are my results:

32 Khz versus 96 Khz
=================================
foo_abx 1.3.4 report
foobar2000 v1.3.2
2014/07/09 06:10:07

File A: C:\Users\Amir\Music\Arnys Filter Test\keys jangling band resolution limited 3216 2496.wav
File B: C:\Users\Amir\Music\Arnys Filter Test\keys jangling full band 2496.wav

06:10:07 : Test started.
06:10:38 : 01/01 50.0%
06:10:50 : 02/02 25.0%
06:11:07 : 03/03 12.5%
06:11:23 : 04/04 6.3%
06:11:36 : 05/05 3.1%
06:12:00 : 06/06 1.6%
06:12:14 : 07/07 0.8%
06:12:26 : 08/08 0.4%
06:12:38 : 09/09 0.2%
06:12:49 : 10/10 0.1%
06:13:00 : 11/11 0.0%
06:13:23 : 12/12 0.0%
06:13:42 : 13/13 0.0%
06:13:48 : Test finished.

----------
Total: 13/13 (0.0%)


44.1 versus 96 Khz
---------------------------------

foo_abx 1.3.4 report
foobar2000 v1.3.2
2014/07/09 06:32:02

File A: C:\Users\Amir\Music\Arnys Filter Test\keys jangling band resolution limited 4416 2496.wav
File B: C:\Users\Amir\Music\Arnys Filter Test\keys jangling full band 2496.wav

06:32:02 : Test started.
06:33:07 : 01/01 50.0%
06:33:17 : 02/02 25.0%
06:33:24 : 03/03 12.5%
06:33:36 : 04/04 6.3%
06:33:47 : 05/05 3.1%
06:33:58 : 06/06 1.6%
06:34:12 : 07/07 0.8%
06:34:15 : Test finished.

----------
Total: 7/7 (0.8%)

===============================

I don't know why Foobar stopped all of a sudden at 7 trials on 44.1 vs 96. While I could clearly hear the difference between the files, I would want to run more trials later as I did not expect to be able to tell them apart this easily.

Anyway, how did you do Arny?
As you can see, I was able to differentiate the files completely. The 32 Khz sampling was very distorted. Another poster, Frank, got the same scores as I did in that file. 44.1 Khz was harder to distinguish but I managed to do so with 7 out of 7 trials correctly identified. No one else has post results for that so perhaps this is dependent on one's listening ability.

Scott was kind enough to create another set of files for the same comparison. See: AVS/AIX High-Resolution Audio Test: Take 2

I ran those files double blind with these results:

Quote:
Originally Posted by amirm View Post
Thank you Scott! Much appreciated the effort you have put on this project Scott. For the first time I feel that the forum is moving forward toward better understanding of this topic.

foo_abx 1.3.4 report
foobar2000 v1.3.2
2014/07/10 18:50:44

File A: C:\Users\Amir\Music\AIX AVS Test files\On_The_Street_Where_You_Live_A2.wav
File B: C:\Users\Amir\Music\AIX AVS Test files\On_The_Street_Where_You_Live_B2.wav

18:50:44 : Test started.
18:51:25 : 00/01 100.0%
18:51:38 : 01/02 75.0%
18:51:47 : 02/03 50.0%
18:51:55 : 03/04 31.3%
18:52:05 : 04/05 18.8%
18:52:21 : 05/06 10.9%
18:52:32 : 06/07 6.3%
18:52:43 : 07/08 3.5%
18:52:59 : 08/09 2.0%
18:53:10 : 09/10 1.1%
18:53:19 : 10/11 0.6%
18:53:23 : Test finished.

----------
Total: 10/11 (0.6%)

Quote:
Originally Posted by amirm View Post
The third track was pretty easy. First segment picked was quite revealing:

foo_abx 1.3.4 report
foobar2000 v1.3.2
2014/07/10 21:01:16

File A: C:\Users\Amir\Music\AIX AVS Test files\Just_My_Imagination_A2.wav
File B: C:\Users\Amir\Music\AIX AVS Test files\Just_My_Imagination_B2.wav

21:01:16 : Test started.
21:02:11 : 01/01 50.0%
21:02:20 : 02/02 25.0%
21:02:28 : 03/03 12.5%
21:02:38 : 04/04 6.3%
21:02:47 : 05/05 3.1%
21:02:56 : 06/06 1.6%
21:03:06 : 07/07 0.8%
21:03:16 : 08/08 0.4%
21:03:26 : 09/09 0.2%
21:03:45 : 10/10 0.1%
21:03:54 : 11/11 0.0%
21:04:11 : 12/12 0.0%
21:04:24 : Test finished.

----------
Total: 12/12 (0.0%)


Quote:
Originally Posted by amirm View Post
foo_abx 1.3.4 report
foobar2000 v1.3.2
2014/07/11 06:18:47

File A: C:\Users\Amir\Music\AIX AVS Test files\Mosaic_A2.wav
File B: C:\Users\Amir\Music\AIX AVS Test files\Mosaic_B2.wav

06:18:47 : Test started.
06:19:38 : 00/01 100.0%
06:20:15 : 00/02 100.0%
06:20:47 : 01/03 87.5%
06:21:01 : 01/04 93.8%
06:21:20 : 02/05 81.3%
06:21:32 : 03/06 65.6%
06:21:48 : 04/07 50.0%
06:22:01 : 04/08 63.7%
06:22:15 : 05/09 50.0%
06:22:24 : 05/10 62.3%
06:23:15 : 06/11 50.0% <---- difference found reliably. Note the 100% correct votes from here on.
06:23:27 : 07/12 38.7%
06:23:36 : 08/13 29.1%
06:23:49 : 09/14 21.2%
06:24:02 : 10/15 15.1%
06:24:10 : 11/16 10.5%
06:24:20 : 12/17 7.2%
06:24:27 : 13/18 4.8%
06:24:35 : 14/19 3.2%
06:24:40 : 15/20 2.1%
06:24:46 : 16/21 1.3%
06:24:56 : 17/22 0.8%
06:25:04 : 18/23 0.5%
06:25:13 : 19/24 0.3%
06:25:25 : 20/25 0.2%
06:25:32 : 21/26 0.1%
06:25:38 : 22/27 0.1%
06:25:45 : 23/28 0.0%
06:25:51 : 24/29 0.0%
06:25:58 : 25/30 0.0%

06:26:24 : Test finished.

----------
Total: 25/30 (0.0%)


So we now have 3 out of 3 positive detection of differences in Scott's clips.
Summarizing, I managed to consistently tell all three files apart from their downsampled 44.1 Khz/16 bit versions.

I have not seen anyone else post their results. I suspect some have PMed them to Scott. Would be interesting to see where other folks land.

It sure would have been nice to have more people run Arny's files as the case there is simpler than Scott's. Unfortunately no one else other than Frank reported them. Sadly Arny and other vocal members against high-resolution audio would not post any data as to how they did in these tests.

I suspect if we had that data we would find that not everyone has the same listening ability. This is counter to the common argument that "there are no golden ears." I am a believer that when it comes to non-linear distortions, there are people who have far better ability than average.

I will be summarizing the content of the other long thread where the posts came from but for now, I am hoping this information makes people think and gets them motivated to run the listening tests and post them. I suspect we are at the cusp of radically reshaping these conversations in the future given the data we now have.

Amir
Retired Technology Insider
Founder, Madrona Digital
"Insist on Quality Engineering"
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post #639 of 640 Old 07-12-2014, 01:36 PM
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post #640 of 640 Old 07-14-2014, 09:19 AM
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Quote:
Originally Posted by arnyk View Post


Actually a large proportion of all professional microphones start rolling off an octave lower - at 10 or 12 KHz.

This is the manufacturer's FR spec for what may be the most widely used professional microphone in the world:

http://cdn.shure.com/specification_s..._specsheet.pdf



it is pretty typical.

I am not chiming in to discuss or argue any of the points made in this conversation, but feel compelled to comment on this factoid that you bring up. For a number of years I was involved in audio engineering as the owner of a small recording studio. While it is true that the Shure SM58 is a widely used microphone - and some engineers are avid fans of the SM57/SM58, it is most common in live PA sound. There is a time and place for this microphone, but I can assure you that very little professional recording is done using the SM58, so I would sway away from using this to reinforce your point.


That being said, it is true that most mics have pretty steep roll-off curves and their own quirky frequency responses. I've even used the equivalent of a Dictaphone mic on a few occasions - with a frequency curve that would make that SM58 graph look like a Neumann in comparison. It is really about fitting the unique signature of a mic to the unique sound of what is being recorded to fit within the context of the track you are recording. I have been known to throw a dozen different mics on a vocalist or instrument to find that fit...and sometimes - yes - it might even be an SM58.
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